From 1037d567fdfc7ab9d3e2328e27bdc1300d3fdb1e Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 21 Nov 2013 12:38:44 -0200 Subject: ASoC: wm8737: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/wm8737.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index 2f167a8ca01b..22de2420bec8 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -644,7 +644,7 @@ static const struct regmap_config wm8737_regmap = { .volatile_reg = wm8737_volatile, }; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm8737_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -758,7 +758,7 @@ static struct spi_driver wm8737_spi_driver = { static int __init wm8737_modinit(void) { int ret; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&wm8737_i2c_driver); if (ret != 0) { printk(KERN_ERR "Failed to register WM8737 I2C driver: %d\n", @@ -781,7 +781,7 @@ static void __exit wm8737_exit(void) #if defined(CONFIG_SPI_MASTER) spi_unregister_driver(&wm8737_spi_driver); #endif -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&wm8737_i2c_driver); #endif } -- cgit v1.2.3 From 060ec80a27bf7a00bafb0c7495bbeec92e7903d3 Mon Sep 17 00:00:00 2001 From: Fabio Estevam Date: Thu, 21 Nov 2013 12:38:51 -0200 Subject: ASoC: wm8983: Use IS_ENABLED() macro Using the IS_ENABLED() macro can make the code shorter and simpler. Signed-off-by: Fabio Estevam Signed-off-by: Mark Brown --- sound/soc/codecs/wm8983.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index aa41ba0dfff4..3bae71d990e6 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -1129,7 +1129,7 @@ static struct spi_driver wm8983_spi_driver = { }; #endif -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) static int wm8983_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -1182,7 +1182,7 @@ static int __init wm8983_modinit(void) { int ret = 0; -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) ret = i2c_add_driver(&wm8983_i2c_driver); if (ret) { printk(KERN_ERR "Failed to register wm8983 I2C driver: %d\n", @@ -1202,7 +1202,7 @@ module_init(wm8983_modinit); static void __exit wm8983_exit(void) { -#if defined(CONFIG_I2C) || defined(CONFIG_I2C_MODULE) +#if IS_ENABLED(CONFIG_I2C) i2c_del_driver(&wm8983_i2c_driver); #endif #if defined(CONFIG_SPI_MASTER) -- cgit v1.2.3 From 2a67e796da38dff41e5b2aa9352c81189407acb9 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 21 Jan 2014 15:58:06 +0000 Subject: ASoC: wm5102: Improve EQ coefficient controls The EQ coefficient binary controls overlapped with the volume controls for the B4 and B5 volumes, which were controllable from either the coefficient control or the volume control itself. This patch adds controls for the mode and moves the coefficient control to only cover the coefficients. Signed-off-by: Charles Keepax Signed-off-by: Richard Fitzgerald Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 17 ++++++++--------- 1 file changed, 8 insertions(+), 9 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index ce9c8e14d4bd..5e03a83aecaa 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -685,15 +685,8 @@ ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES_MASK("EQ1 Coefficients", ARIZONA_EQ1_1, 21, - ARIZONA_EQ1_ENA_MASK), -SND_SOC_BYTES_MASK("EQ2 Coefficients", ARIZONA_EQ2_1, 21, - ARIZONA_EQ2_ENA_MASK), -SND_SOC_BYTES_MASK("EQ3 Coefficients", ARIZONA_EQ3_1, 21, - ARIZONA_EQ3_ENA_MASK), -SND_SOC_BYTES_MASK("EQ4 Coefficients", ARIZONA_EQ4_1, 21, - ARIZONA_EQ4_ENA_MASK), - +SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 18), +SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, @@ -705,6 +698,8 @@ SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, 24, 0, eq_tlv), +SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 18), +SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, @@ -716,6 +711,8 @@ SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, 24, 0, eq_tlv), +SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 18), +SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT, @@ -727,6 +724,8 @@ SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, 24, 0, eq_tlv), +SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 18), +SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT, -- cgit v1.2.3 From 552694c65b1c563dbdbda4082527774a373ae720 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 21 Jan 2014 15:58:07 +0000 Subject: ASoC: wm5110: Improve EQ coefficient controls The EQ coefficient binary controls overlapped with the volume controls for the B4 and B5 volumes, which were controllable from either the coefficient control or the volume control itself. This patch adds controls for the mode and moves the coefficient control to only cover the coefficients. Signed-off-by: Charles Keepax Signed-off-by: Richard Fitzgerald Signed-off-by: Mark Brown --- sound/soc/codecs/wm5110.c | 17 ++++++++--------- 1 file changed, 8 insertions(+), 9 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 2c3c962d9a85..7c2b0d669d29 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -247,15 +247,8 @@ ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES_MASK("EQ1 Coefficients", ARIZONA_EQ1_1, 21, - ARIZONA_EQ1_ENA_MASK), -SND_SOC_BYTES_MASK("EQ2 Coefficients", ARIZONA_EQ2_1, 21, - ARIZONA_EQ2_ENA_MASK), -SND_SOC_BYTES_MASK("EQ3 Coefficients", ARIZONA_EQ3_1, 21, - ARIZONA_EQ3_ENA_MASK), -SND_SOC_BYTES_MASK("EQ4 Coefficients", ARIZONA_EQ4_1, 21, - ARIZONA_EQ4_ENA_MASK), - +SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 18), +SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, @@ -267,6 +260,8 @@ SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, 24, 0, eq_tlv), +SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 18), +SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, @@ -278,6 +273,8 @@ SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, 24, 0, eq_tlv), +SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 18), +SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT, @@ -289,6 +286,8 @@ SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, 24, 0, eq_tlv), +SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 18), +SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT, -- cgit v1.2.3 From fb0def0ceda7b2a79be978093704c1c71e26ba22 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 21 Jan 2014 15:58:08 +0000 Subject: ASoC: wm8997: Improve EQ coefficient controls The EQ coefficient binary controls overlapped with the volume controls for the B4 and B5 volumes, which were controllable from either the coefficient control or the volume control itself. This patch adds controls for the mode and moves the coefficient control to only cover the coefficients. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8997.c | 17 ++++++++--------- 1 file changed, 8 insertions(+), 9 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 555115ee2159..4a382495976c 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -170,15 +170,8 @@ ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES_MASK("EQ1 Coefficients", ARIZONA_EQ1_1, 21, - ARIZONA_EQ1_ENA_MASK), -SND_SOC_BYTES_MASK("EQ2 Coefficients", ARIZONA_EQ2_1, 21, - ARIZONA_EQ2_ENA_MASK), -SND_SOC_BYTES_MASK("EQ3 Coefficients", ARIZONA_EQ3_1, 21, - ARIZONA_EQ3_ENA_MASK), -SND_SOC_BYTES_MASK("EQ4 Coefficients", ARIZONA_EQ4_1, 21, - ARIZONA_EQ4_ENA_MASK), - +SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 18), +SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ1 B2 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B2_GAIN_SHIFT, @@ -190,6 +183,8 @@ SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, 24, 0, eq_tlv), +SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 18), +SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ2 B2 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B2_GAIN_SHIFT, @@ -201,6 +196,8 @@ SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, 24, 0, eq_tlv), +SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 18), +SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ3 B2 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B2_GAIN_SHIFT, @@ -212,6 +209,8 @@ SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, 24, 0, eq_tlv), +SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 18), +SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, 24, 0, eq_tlv), SOC_SINGLE_TLV("EQ4 B2 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B2_GAIN_SHIFT, -- cgit v1.2.3 From ddbc5efed0f9064287acead56bbf0dce3ca28ee2 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 22 Jan 2014 10:09:11 +0000 Subject: ASoC: wm_adsp: Add debug print to note that the DSP has shutdown It can be useful for debugging purposes to see at what point the DSP has powered down, so add a message to inform us of this. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 444626fcab40..f9fd56444a14 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -1679,6 +1679,8 @@ int wm_adsp2_event(struct snd_soc_dapm_widget *w, list_del(&alg_region->list); kfree(alg_region); } + + adsp_dbg(dsp, "Shutdown complete\n"); break; default: -- cgit v1.2.3 From 1371105731a7c72168d0e464a51203fec829390b Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jan 2014 13:45:34 +0000 Subject: ASoC: wm5102: Correct typo in EQ coefficient sizes Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 5e03a83aecaa..ebffe81daa1d 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -685,7 +685,7 @@ ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 18), +SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 19), SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, 24, 0, eq_tlv), @@ -698,7 +698,7 @@ SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 18), +SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 19), SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, 24, 0, eq_tlv), @@ -711,7 +711,7 @@ SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 18), +SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 19), SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, 24, 0, eq_tlv), @@ -724,7 +724,7 @@ SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 18), +SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 19), SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, 24, 0, eq_tlv), -- cgit v1.2.3 From 3c379ef97e7f6cd305a4873150319c2355b4316d Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jan 2014 13:45:35 +0000 Subject: ASoC: wm5110: Correct type in EQ coefficient sizes Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm5110.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 7c2b0d669d29..4de2bf16dc74 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -247,7 +247,7 @@ ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 18), +SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 19), SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, 24, 0, eq_tlv), @@ -260,7 +260,7 @@ SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 18), +SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 19), SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, 24, 0, eq_tlv), @@ -273,7 +273,7 @@ SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 18), +SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 19), SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, 24, 0, eq_tlv), @@ -286,7 +286,7 @@ SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 18), +SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 19), SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, 24, 0, eq_tlv), -- cgit v1.2.3 From 2b14cd3af9d98ce135e503e91e3973bdd5e4baeb Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 23 Jan 2014 13:45:36 +0000 Subject: ASoC: wm8997: Correct typo in EQ coefficient sizes Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8997.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 4a382495976c..6107108228b6 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -170,7 +170,7 @@ ARIZONA_MIXER_CONTROLS("EQ2", ARIZONA_EQ2MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ3", ARIZONA_EQ3MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("EQ4", ARIZONA_EQ4MIX_INPUT_1_SOURCE), -SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 18), +SND_SOC_BYTES("EQ1 Coefficients", ARIZONA_EQ1_3, 19), SOC_SINGLE("EQ1 Mode Switch", ARIZONA_EQ1_2, ARIZONA_EQ1_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ1 B1 Volume", ARIZONA_EQ1_1, ARIZONA_EQ1_B1_GAIN_SHIFT, 24, 0, eq_tlv), @@ -183,7 +183,7 @@ SOC_SINGLE_TLV("EQ1 B4 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ1 B5 Volume", ARIZONA_EQ1_2, ARIZONA_EQ1_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 18), +SND_SOC_BYTES("EQ2 Coefficients", ARIZONA_EQ2_3, 19), SOC_SINGLE("EQ2 Mode Switch", ARIZONA_EQ2_2, ARIZONA_EQ2_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ2 B1 Volume", ARIZONA_EQ2_1, ARIZONA_EQ2_B1_GAIN_SHIFT, 24, 0, eq_tlv), @@ -196,7 +196,7 @@ SOC_SINGLE_TLV("EQ2 B4 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ2 B5 Volume", ARIZONA_EQ2_2, ARIZONA_EQ2_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 18), +SND_SOC_BYTES("EQ3 Coefficients", ARIZONA_EQ3_3, 19), SOC_SINGLE("EQ3 Mode Switch", ARIZONA_EQ3_2, ARIZONA_EQ3_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ3 B1 Volume", ARIZONA_EQ3_1, ARIZONA_EQ3_B1_GAIN_SHIFT, 24, 0, eq_tlv), @@ -209,7 +209,7 @@ SOC_SINGLE_TLV("EQ3 B4 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B4_GAIN_SHIFT, SOC_SINGLE_TLV("EQ3 B5 Volume", ARIZONA_EQ3_2, ARIZONA_EQ3_B5_GAIN_SHIFT, 24, 0, eq_tlv), -SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 18), +SND_SOC_BYTES("EQ4 Coefficients", ARIZONA_EQ4_3, 19), SOC_SINGLE("EQ4 Mode Switch", ARIZONA_EQ4_2, ARIZONA_EQ4_B1_MODE, 1, 0), SOC_SINGLE_TLV("EQ4 B1 Volume", ARIZONA_EQ4_1, ARIZONA_EQ4_B1_GAIN_SHIFT, 24, 0, eq_tlv), -- cgit v1.2.3 From a74ab5121f8d91fb7f13ac1c86e72e9d35e0bc29 Mon Sep 17 00:00:00 2001 From: Markus Pargmann Date: Thu, 16 Jan 2014 16:30:25 +0100 Subject: ASoC: tlv320aic32x4: Use gpio_is_valid Use function gpio_is_valid to check for gpio ports. Signed-off-by: Markus Pargmann Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 688151ba309a..36a9cb90a585 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -588,7 +588,7 @@ static int aic32x4_probe(struct snd_soc_codec *codec) snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (aic32x4->rstn_gpio >= 0) { + if (gpio_is_valid(aic32x4->rstn_gpio)) { ndelay(10); gpio_set_value(aic32x4->rstn_gpio, 1); } @@ -693,7 +693,7 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c, aic32x4->rstn_gpio = -1; } - if (aic32x4->rstn_gpio >= 0) { + if (gpio_is_valid(aic32x4->rstn_gpio)) { ret = devm_gpio_request_one(&i2c->dev, aic32x4->rstn_gpio, GPIOF_OUT_INIT_LOW, "tlv320aic32x4 rstn"); if (ret != 0) -- cgit v1.2.3 From c671e79d6c2d5a525496fbf18103841c68fe3305 Mon Sep 17 00:00:00 2001 From: Markus Pargmann Date: Mon, 27 Jan 2014 13:03:07 +0100 Subject: ASoC: tlv320aic32x4: Use signed int mixer controls There are a number of mixer controls that support negative values. They use signed values for this with different number of bits for the values. Currently they only support the positive range. This patch replaces the unsigned mixers with signed mixers to support the full range. Signed-off-by: Markus Pargmann Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 26 ++++++++++++++++---------- 1 file changed, 16 insertions(+), 10 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 36a9cb90a585..c541213b1edf 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -68,18 +68,24 @@ struct aic32x4_priv { int rstn_gpio; }; -/* 0dB min, 1dB steps */ -static DECLARE_TLV_DB_SCALE(tlv_step_1, 0, 100, 0); /* 0dB min, 0.5dB steps */ static DECLARE_TLV_DB_SCALE(tlv_step_0_5, 0, 50, 0); +/* -63.5dB min, 0.5dB steps */ +static DECLARE_TLV_DB_SCALE(tlv_pcm, -6350, 50, 0); +/* -6dB min, 1dB steps */ +static DECLARE_TLV_DB_SCALE(tlv_driver_gain, -600, 100, 0); +/* -12dB min, 0.5dB steps */ +static DECLARE_TLV_DB_SCALE(tlv_adc_vol, -1200, 50, 0); static const struct snd_kcontrol_new aic32x4_snd_controls[] = { - SOC_DOUBLE_R_TLV("PCM Playback Volume", AIC32X4_LDACVOL, - AIC32X4_RDACVOL, 0, 0x30, 0, tlv_step_0_5), - SOC_DOUBLE_R_TLV("HP Driver Gain Volume", AIC32X4_HPLGAIN, - AIC32X4_HPRGAIN, 0, 0x1D, 0, tlv_step_1), - SOC_DOUBLE_R_TLV("LO Driver Gain Volume", AIC32X4_LOLGAIN, - AIC32X4_LORGAIN, 0, 0x1D, 0, tlv_step_1), + SOC_DOUBLE_R_S_TLV("PCM Playback Volume", AIC32X4_LDACVOL, + AIC32X4_RDACVOL, 0, -0x7f, 0x30, 7, 0, tlv_pcm), + SOC_DOUBLE_R_S_TLV("HP Driver Gain Volume", AIC32X4_HPLGAIN, + AIC32X4_HPRGAIN, 0, -0x6, 0x1d, 5, 0, + tlv_driver_gain), + SOC_DOUBLE_R_S_TLV("LO Driver Gain Volume", AIC32X4_LOLGAIN, + AIC32X4_LORGAIN, 0, -0x6, 0x1d, 5, 0, + tlv_driver_gain), SOC_DOUBLE_R("HP DAC Playback Switch", AIC32X4_HPLGAIN, AIC32X4_HPRGAIN, 6, 0x01, 1), SOC_DOUBLE_R("LO DAC Playback Switch", AIC32X4_LOLGAIN, @@ -90,8 +96,8 @@ static const struct snd_kcontrol_new aic32x4_snd_controls[] = { SOC_SINGLE("ADCFGA Left Mute Switch", AIC32X4_ADCFGA, 7, 1, 0), SOC_SINGLE("ADCFGA Right Mute Switch", AIC32X4_ADCFGA, 3, 1, 0), - SOC_DOUBLE_R_TLV("ADC Level Volume", AIC32X4_LADCVOL, - AIC32X4_RADCVOL, 0, 0x28, 0, tlv_step_0_5), + SOC_DOUBLE_R_S_TLV("ADC Level Volume", AIC32X4_LADCVOL, + AIC32X4_RADCVOL, 0, -0x18, 0x28, 6, 0, tlv_adc_vol), SOC_DOUBLE_R_TLV("PGA Level Volume", AIC32X4_LMICPGAVOL, AIC32X4_RMICPGAVOL, 0, 0x5f, 0, tlv_step_0_5), -- cgit v1.2.3 From 4d16700dd926d4c4a66a91a138c34eef4fd342b4 Mon Sep 17 00:00:00 2001 From: Markus Pargmann Date: Mon, 27 Jan 2014 13:03:08 +0100 Subject: ASoC: tlv320aic32x4: DT support Add DT support for this codec. The bindings differ a bit from the aic3x codec bindings, so I created a new binding documentation. Signed-off-by: Markus Pargmann Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/tlv320aic32x4.txt | 18 ++++++++++++++++ sound/soc/codecs/tlv320aic32x4.c | 25 ++++++++++++++++++++++ 2 files changed, 43 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic32x4.txt (limited to 'sound/soc/codecs') diff --git a/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt new file mode 100644 index 000000000000..db0551088cc4 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt @@ -0,0 +1,18 @@ +Texas Instruments - tlv320aic32x4 Codec module + +The tlv320aic32x4 serial control bus communicates through I2C protocols + +Required properties: + - compatible: Should be "ti,tlv320aic32x4" + - reg: I2C slave address + +Optional properties: + - reset-gpios: Reset-GPIO phandle with args as described in gpio/gpio.txt + + +Example: + +codec: tlv320aic32x4@18 { + compatible = "ti,tlv320aic32x4"; + reg = <0x18>; +}; diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index c541213b1edf..1dd50e48934c 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -29,6 +29,7 @@ #include #include #include +#include #include #include #include @@ -669,11 +670,22 @@ static struct snd_soc_codec_driver soc_codec_dev_aic32x4 = { .num_dapm_routes = ARRAY_SIZE(aic32x4_dapm_routes), }; +static int aic32x4_parse_dt(struct aic32x4_priv *aic32x4, + struct device_node *np) +{ + aic32x4->swapdacs = false; + aic32x4->micpga_routing = 0; + aic32x4->rstn_gpio = of_get_named_gpio(np, "reset-gpios", 0); + + return 0; +} + static int aic32x4_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct aic32x4_pdata *pdata = i2c->dev.platform_data; struct aic32x4_priv *aic32x4; + struct device_node *np = i2c->dev.of_node; int ret; aic32x4 = devm_kzalloc(&i2c->dev, sizeof(struct aic32x4_priv), @@ -692,6 +704,12 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c, aic32x4->swapdacs = pdata->swapdacs; aic32x4->micpga_routing = pdata->micpga_routing; aic32x4->rstn_gpio = pdata->rstn_gpio; + } else if (np) { + ret = aic32x4_parse_dt(aic32x4, np); + if (ret) { + dev_err(&i2c->dev, "Failed to parse DT node\n"); + return ret; + } } else { aic32x4->power_cfg = 0; aic32x4->swapdacs = false; @@ -723,10 +741,17 @@ static const struct i2c_device_id aic32x4_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, aic32x4_i2c_id); +static const struct of_device_id aic32x4_of_id[] = { + { .compatible = "ti,tlv320aic32x4", }, + { /* senitel */ } +}; +MODULE_DEVICE_TABLE(of, aic32x4_of_id); + static struct i2c_driver aic32x4_i2c_driver = { .driver = { .name = "tlv320aic32x4", .owner = THIS_MODULE, + .of_match_table = aic32x4_of_id, }, .probe = aic32x4_i2c_probe, .remove = aic32x4_i2c_remove, -- cgit v1.2.3 From 7e9614ebcf4d74edba864cc91e1e8a3ec6b32fc2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 30 Jan 2014 19:55:45 +0000 Subject: ASoC: wm8962: Hold a runtime PM reference while handling interrupts If the device is runtime suspended then we can't interact with it as it may have been powered off and the register map will be in cache only mode. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8962.c | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 97db3b45b411..1996567346c6 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3003,9 +3003,16 @@ static irqreturn_t wm8962_irq(int irq, void *data) unsigned int active; int reg, ret; + ret = pm_runtime_get_sync(dev); + if (ret < 0) { + dev_err(dev, "Failed to resume: %d\n", ret); + return IRQ_NONE; + } + ret = regmap_read(wm8962->regmap, WM8962_INTERRUPT_STATUS_2_MASK, &mask); if (ret != 0) { + pm_runtime_put(dev); dev_err(dev, "Failed to read interrupt mask: %d\n", ret); return IRQ_NONE; @@ -3013,14 +3020,17 @@ static irqreturn_t wm8962_irq(int irq, void *data) ret = regmap_read(wm8962->regmap, WM8962_INTERRUPT_STATUS_2, &active); if (ret != 0) { + pm_runtime_put(dev); dev_err(dev, "Failed to read interrupt: %d\n", ret); return IRQ_NONE; } active &= ~mask; - if (!active) + if (!active) { + pm_runtime_put(dev); return IRQ_NONE; + } /* Acknowledge the interrupts */ ret = regmap_write(wm8962->regmap, WM8962_INTERRUPT_STATUS_2, active); @@ -3070,6 +3080,8 @@ static irqreturn_t wm8962_irq(int irq, void *data) msecs_to_jiffies(250)); } + pm_runtime_put(dev); + return IRQ_HANDLED; } -- cgit v1.2.3 From df6ab65f2fef3d7b769f3ba87c7bb265ace80b4e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 30 Jan 2014 19:59:31 +0000 Subject: ASoC: wm8962: Check if we runtime resume the device when starting FLL Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8962.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 1996567346c6..d7d43c9371f4 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2886,7 +2886,11 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source, try_wait_for_completion(&wm8962->fll_lock); - pm_runtime_get_sync(codec->dev); + ret = pm_runtime_get_sync(codec->dev); + if (ret < 0) { + dev_err(codec->dev, "Failed to resume device: %d\n", ret); + return ret; + } snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1, WM8962_FLL_FRAC | WM8962_FLL_REFCLK_SRC_MASK | -- cgit v1.2.3 From d6f95e5407674d2f7d61feef81fef96b364d9188 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 30 Jan 2014 20:04:34 +0000 Subject: ASoC: wm8962: Clean up error handling for failed FLL start Don't record the FLL as having started and leave the hardware disabled ensuring we are in a better state if this does happen to be a transient error and making debugging easier. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8962.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index d7d43c9371f4..cd96d463a505 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2898,8 +2898,6 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source, dev_dbg(codec->dev, "FLL configured for %dHz->%dHz\n", Fref, Fout); - ret = 0; - /* This should be a massive overestimate but go even * higher if we'll error out */ @@ -2913,14 +2911,17 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source, if (timeout == 0 && wm8962->irq) { dev_err(codec->dev, "FLL lock timed out"); - ret = -ETIMEDOUT; + snd_soc_update_bits(codec, WM8962_FLL_CONTROL_1, + WM8962_FLL_ENA, 0); + pm_runtime_put(codec->dev); + return -ETIMEDOUT; } wm8962->fll_fref = Fref; wm8962->fll_fout = Fout; wm8962->fll_src = source; - return ret; + return 0; } static int wm8962_mute(struct snd_soc_dai *dai, int mute) -- cgit v1.2.3 From 9d7433b064a6349aae8a266e8243ef75637bec45 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 30 Jan 2014 20:32:06 +0000 Subject: ASoC: wm8962: Reinitialise the IRQ completion rather than just trying it This is better practice. Signed-off-by: Mark Brown Acked-by: Charles Keepax --- sound/soc/codecs/wm8962.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index cd96d463a505..c06bb5088e60 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -2884,7 +2884,7 @@ static int wm8962_set_fll(struct snd_soc_codec *codec, int fll_id, int source, snd_soc_write(codec, WM8962_FLL_CONTROL_7, fll_div.lambda); snd_soc_write(codec, WM8962_FLL_CONTROL_8, fll_div.n); - try_wait_for_completion(&wm8962->fll_lock); + reinit_completion(&wm8962->fll_lock); ret = pm_runtime_get_sync(codec->dev); if (ret < 0) { -- cgit v1.2.3 From 0d724f8a3bbc9b0ffd658732714d23694ff4abca Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 23 Jan 2014 12:50:51 +0000 Subject: ASoC: ak4554: Add to SND_SOC_ALL_CODECS For build coverage. Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 983d087aa92a..034130b336aa 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -25,6 +25,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C + select SND_SOC_AK4554 select SND_SOC_AK4641 if I2C select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C -- cgit v1.2.3 From 50a68fb4bc2516f593ceffa6617c93090d335f31 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 24 Jan 2014 16:07:05 +0000 Subject: ASoC: pcm1681: Convert to params_width() This will help support future enhancements in the way we negotiate parameters in the core. Signed-off-by: Mark Brown --- sound/soc/codecs/pcm1681.c | 15 ++++++++++----- 1 file changed, 10 insertions(+), 5 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/pcm1681.c b/sound/soc/codecs/pcm1681.c index 73f9c3630e2c..e427544183d7 100644 --- a/sound/soc/codecs/pcm1681.c +++ b/sound/soc/codecs/pcm1681.c @@ -172,16 +172,21 @@ static int pcm1681_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; struct pcm1681_private *priv = snd_soc_codec_get_drvdata(codec); int val = 0, ret; - int pcm_format = params_format(params); priv->rate = params_rate(params); switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_RIGHT_J: - if (pcm_format == SNDRV_PCM_FORMAT_S24_LE) - val = 0x00; - else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE) - val = 0x03; + switch (params_width(params)) { + case 24: + val = 0; + break; + case 16: + val = 3; + break; + default: + return -EINVAL; + } break; case SND_SOC_DAIFMT_I2S: val = 0x04; -- cgit v1.2.3 From e48c6d7f01032c42907dc2cd81b73f95490c81a1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 24 Jan 2014 16:07:16 +0000 Subject: ASoC: pcm1792a: Convert to params_width() This will help support future enhancements in the way we negotiate parameters in the core. Signed-off-by: Mark Brown Acked-by: Michael Trimarchi --- sound/soc/codecs/pcm1792a.c | 33 ++++++++++++++++++++++----------- 1 file changed, 22 insertions(+), 11 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/pcm1792a.c b/sound/soc/codecs/pcm1792a.c index 7146653a8e16..3a80ba4452df 100644 --- a/sound/soc/codecs/pcm1792a.c +++ b/sound/soc/codecs/pcm1792a.c @@ -107,24 +107,35 @@ static int pcm1792a_hw_params(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; struct pcm1792a_private *priv = snd_soc_codec_get_drvdata(codec); int val = 0, ret; - int pcm_format = params_format(params); priv->rate = params_rate(params); switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_RIGHT_J: - if (pcm_format == SNDRV_PCM_FORMAT_S24_LE || - pcm_format == SNDRV_PCM_FORMAT_S32_LE) - val = 0x02; - else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE) - val = 0x00; + switch (params_width(params)) { + case 24: + case 32: + val = 2; + break; + case 16: + val = 0; + break; + default: + return -EINVAL; + } break; case SND_SOC_DAIFMT_I2S: - if (pcm_format == SNDRV_PCM_FORMAT_S24_LE || - pcm_format == SNDRV_PCM_FORMAT_S32_LE) - val = 0x05; - else if (pcm_format == SNDRV_PCM_FORMAT_S16_LE) - val = 0x04; + switch (params_width(params)) { + case 24: + case 32: + val = 5; + break; + case 16: + val = 4; + break; + default: + return -EINVAL; + } break; default: dev_err(codec->dev, "Invalid DAI format\n"); -- cgit v1.2.3 From 1291e14175e6b83efe1464f32189acb21bc4be09 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 23 Jan 2014 12:58:16 +0000 Subject: ASoC: codecs: Make OF supported CODECs visible in Kconfig Now that we have a generic card driver we can't rely on the card driver selecting the CODECs for us so make the CODECs that can be enabled with OF directly selectable in Kconfig. For the platforms not using OF it's not clear that we don't still want to have some board specific selection since the kernel needs to contain code to register the devices; ACPI could provide this from firmware does not yet support any kind of generic card. It may also be desirable to hide these if OF is not enabled to reduce noise. Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 97 +++++++++++++++++++++++++++++++----------------- 1 file changed, 63 insertions(+), 34 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 034130b336aa..f0bdcc5abe83 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -190,8 +190,9 @@ config SND_SOC_AD73311 tristate config SND_SOC_ADAU1701 + tristate "Analog Devices ADAU1701 CODEC" + depends on I2C select SND_SOC_SIGMADSP - tristate config SND_SOC_ADAU1373 tristate @@ -203,28 +204,31 @@ config SND_SOC_ADS117X tristate config SND_SOC_AK4104 - tristate + tristate "AKM AK4104 CODEC" + depends on SPI_MASTER config SND_SOC_AK4535 tristate config SND_SOC_AK4554 - tristate + tristate "AKM AK4554 CODEC" config SND_SOC_AK4641 tristate config SND_SOC_AK4642 - tristate + tristate "AKM AK4642 CODEC" + depends on I2C config SND_SOC_AK4671 tristate config SND_SOC_AK5386 - tristate + tristate "AKM AK5638 CODEC" config SND_SOC_ALC5623 tristate + config SND_SOC_ALC5632 tristate @@ -235,14 +239,17 @@ config SND_SOC_CS42L51 tristate config SND_SOC_CS42L52 - tristate + tristate "Cirrus Logic CS42L52 CODEC" + depends on I2C config SND_SOC_CS42L73 - tristate + tristate "Cirrus Logic CS42L73 CODEC" + depends on I2C # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 - tristate + tristate "Cirrus Logic CS4270 CODEC" + depends on I2C # Cirrus Logic CS4270 Codec VD = 3.3V Errata # Select if you are affected by the errata where the part will not function @@ -253,7 +260,8 @@ config SND_SOC_CS4270_VD33_ERRATA depends on SND_SOC_CS4270 config SND_SOC_CS4271 - tristate + tristate "Cirrus Logic CS4271 CODEC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_CX20442 tristate @@ -284,6 +292,9 @@ config SND_SOC_BT_SCO config SND_SOC_DMIC tristate +config SND_SOC_HDMI_CODEC + tristate "HDMI stub CODEC" + config SND_SOC_ISABELLE tristate @@ -302,14 +313,13 @@ config SND_SOC_MAX98095 config SND_SOC_MAX9850 tristate -config SND_SOC_HDMI_CODEC - tristate - config SND_SOC_PCM1681 - tristate + tristate "Texas Instruments PCM1681 CODEC" + depends on I2C config SND_SOC_PCM1792A - tristate + tristate "Texas Instruments PCM1792A CODEC" + depends on SPI_MASTER config SND_SOC_PCM3008 tristate @@ -322,7 +332,8 @@ config SND_SOC_RT5640 #Freescale sgtl5000 codec config SND_SOC_SGTL5000 - tristate + tristate "Freescale SGTL5000 CODEC" + depends on I2C config SND_SOC_SI476X tristate @@ -335,7 +346,7 @@ config SND_SOC_SN95031 tristate config SND_SOC_SPDIF - tristate + tristate "S/PDIF CODEC" config SND_SOC_SSM2518 tristate @@ -353,7 +364,8 @@ config SND_SOC_STAC9766 tristate config SND_SOC_TAS5086 - tristate + tristate "Texas Instruments TAS5086 speaker amplifier" + depends on I2C config SND_SOC_TLV320AIC23 tristate @@ -366,7 +378,8 @@ config SND_SOC_TLV320AIC32X4 tristate config SND_SOC_TLV320AIC3X - tristate + tristate "Texas Instruments TLV320AIC3x CODECs" + depends on I2C config SND_SOC_TLV320DAC33 tristate @@ -415,55 +428,69 @@ config SND_SOC_WM8400 tristate config SND_SOC_WM8510 - tristate + tristate "Wolfson Microelectronics WM8510 CODEC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8523 - tristate + tristate "Wolfson Microelectronics WM8523 DAC" + depends on I2C config SND_SOC_WM8580 - tristate + tristate "Wolfson Microelectronics WM8523 CODEC" + depends on I2C config SND_SOC_WM8711 - tristate + tristate "Wolfson Microelectronics WM8711 CODEC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8727 tristate config SND_SOC_WM8728 - tristate + tristate "Wolfson Microelectronics WM8728 DAC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8731 - tristate + tristate "Wolfson Microelectronics WM8731 CODEC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8737 - tristate + tristate "Wolfson Microelectronics WM8737 ADC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8741 - tristate + tristate "Wolfson Microelectronics WM8737 DAC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8750 - tristate + tristate "Wolfson Microelectronics WM8750 CODEC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8753 - tristate + tristate "Wolfson Microelectronics WM8753 CODEC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8770 - tristate + tristate "Wolfson Microelectronics WM8770 CODEC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8776 - tristate + tristate "Wolfson Microelectronics WM8776 CODEC" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8782 tristate config SND_SOC_WM8804 - tristate + tristate "Wolfson Microelectronics WM8804 S/PDIF transceiver" + depends on SND_SOC_I2C_AND_SPI config SND_SOC_WM8900 tristate config SND_SOC_WM8903 - tristate + tristate "Wolfson Microelectronics WM8903 CODEC" + depends on I2C config SND_SOC_WM8904 tristate @@ -481,7 +508,8 @@ config SND_SOC_WM8961 tristate config SND_SOC_WM8962 - tristate + tristate "Wolfson Microelectronics WM8962 CODEC" + depends on I2C config SND_SOC_WM8971 tristate @@ -554,4 +582,5 @@ config SND_SOC_ML26124 tristate config SND_SOC_TPA6130A2 - tristate + tristate "Texas Instruments TPA6130A2 headphone amplifier" + depends on I2C -- cgit v1.2.3 From b224e9b857438afbd802f47008ab36863f71d8d1 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 4 Feb 2014 14:48:22 +0100 Subject: ASoC: cs4271: Remove outdated comment Commit 1b1861ead ("ASoC: cs4271: convert to direct regmap API usage") removed the bus_type field from the cs4271_private struct, but left the comment that described the field in there. This patch removes the comment. Signed-off-by: Lars-Peter Clausen Acked-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/codecs/cs4271.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index ce05fd93dc74..f7bbe6fdba67 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -159,7 +159,6 @@ static bool cs4271_volatile_reg(struct device *dev, unsigned int reg) } struct cs4271_private { - /* SND_SOC_I2C or SND_SOC_SPI */ unsigned int mclk; bool master; bool deemph; -- cgit v1.2.3 From 6e84b9768dfb299a9881895b331e3e532041fae4 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 4 Feb 2014 20:55:31 +0100 Subject: ASoC: cs42l73: Don't mix SNDRV_PCM_RATE_KNOT with specific rates SNDRV_PCM_RATE_KNOT means that the device can support rates that can not be expressed using the rate bits. The driver will provide a list of those rates specified through constraints. Any rate bits that are set in the rates mask will be ignored. So setting other rate bits besides SNDRV_PCM_RATE_KNOT wont have any effect, but might be confusing to the casual reader, so remove them. Signed-off-by: Lars-Peter Clausen Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l73.c | 15 ++++++--------- 1 file changed, 6 insertions(+), 9 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 549d5d6a3fef..7cae046c7dd0 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1255,9 +1255,6 @@ static int cs42l73_pcm_startup(struct snd_pcm_substream *substream, return 0; } -/* SNDRV_PCM_RATE_KNOT -> 12000, 24000 Hz, limit with constraint list */ -#define CS42L73_RATES (SNDRV_PCM_RATE_8000_48000 | SNDRV_PCM_RATE_KNOT) - #define CS42L73_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\ SNDRV_PCM_FMTBIT_S24_LE) @@ -1278,14 +1275,14 @@ static struct snd_soc_dai_driver cs42l73_dai[] = { .stream_name = "XSP Playback", .channels_min = 1, .channels_max = 2, - .rates = CS42L73_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = CS42L73_FORMATS, }, .capture = { .stream_name = "XSP Capture", .channels_min = 1, .channels_max = 2, - .rates = CS42L73_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = CS42L73_FORMATS, }, .ops = &cs42l73_ops, @@ -1298,14 +1295,14 @@ static struct snd_soc_dai_driver cs42l73_dai[] = { .stream_name = "ASP Playback", .channels_min = 2, .channels_max = 2, - .rates = CS42L73_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = CS42L73_FORMATS, }, .capture = { .stream_name = "ASP Capture", .channels_min = 2, .channels_max = 2, - .rates = CS42L73_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = CS42L73_FORMATS, }, .ops = &cs42l73_ops, @@ -1318,14 +1315,14 @@ static struct snd_soc_dai_driver cs42l73_dai[] = { .stream_name = "VSP Playback", .channels_min = 1, .channels_max = 2, - .rates = CS42L73_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = CS42L73_FORMATS, }, .capture = { .stream_name = "VSP Capture", .channels_min = 1, .channels_max = 2, - .rates = CS42L73_RATES, + .rates = SNDRV_PCM_RATE_KNOT, .formats = CS42L73_FORMATS, }, .ops = &cs42l73_ops, -- cgit v1.2.3 From 5a3af1293194d07610fd7fdf680b3e7f916dca84 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 6 Feb 2014 12:03:27 +0000 Subject: ASoC: pcm512x: Add PCM512x driver The PCM512x devices are a family of monolithic CMOS integrated circuits that include a stereo digital-to-analog converter and additional support circuitry. This is an initial driver which supports some core functionality for the device which covers common use cases but does not cover all features. Currently only slave clocking modes with automatic clock configuration are supported and most of the DSP configuration for the device is not enabled. Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/pcm512x.txt | 30 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/pcm512x.c | 672 +++++++++++++++++++++ sound/soc/codecs/pcm512x.h | 142 +++++ 5 files changed, 850 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/pcm512x.txt create mode 100644 sound/soc/codecs/pcm512x.c create mode 100644 sound/soc/codecs/pcm512x.h (limited to 'sound/soc/codecs') diff --git a/Documentation/devicetree/bindings/sound/pcm512x.txt b/Documentation/devicetree/bindings/sound/pcm512x.txt new file mode 100644 index 000000000000..faff75e64573 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/pcm512x.txt @@ -0,0 +1,30 @@ +PCM512x audio CODECs + +These devices support both I2C and SPI (configured with pin strapping +on the board). + +Required properties: + + - compatible : One of "ti,pcm5121" or "ti,pcm5122" + + - reg : the I2C address of the device for I2C, the chip select + number for SPI. + + - AVDD-supply, DVDD-supply, and CPVDD-supply : power supplies for the + device, as covered in bindings/regulator/regulator.txt + +Optional properties: + + - clocks : A clock specifier for the clock connected as SCLK. If this + is absent the device will be configured to clock from BCLK. + +Example: + + pcm5122: pcm5122@4c { + compatible = "ti,pcm5122"; + reg = <0x4c>; + + AVDD-supply = <®_3v3_analog>; + DVDD-supply = <®_1v8>; + CPVDD-supply = <®_3v3>; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 983d087aa92a..56d0c2845680 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -59,6 +59,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM1681 if I2C select SND_SOC_PCM1792A if SPI_MASTER select SND_SOC_PCM3008 + select SND_SOC_PCM512x if SND_SOC_I2C_AND_SPI select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C select SND_SOC_SGTL5000 if I2C @@ -313,6 +314,9 @@ config SND_SOC_PCM1792A config SND_SOC_PCM3008 tristate +config SND_SOC_PCM512x + tristate "Texas Instruments PCM512x CODECs" + config SND_SOC_RT5631 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index bc126764a44d..d3b536fc075d 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -46,6 +46,7 @@ snd-soc-hdmi-codec-objs := hdmi.o snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm1792a-codec-objs := pcm1792a.o snd-soc-pcm3008-objs := pcm3008.o +snd-soc-pcm512x-objs := pcm512x.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o snd-soc-sgtl5000-objs := sgtl5000.o @@ -179,6 +180,7 @@ obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o +obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c new file mode 100644 index 000000000000..5ad3e9aa3cb4 --- /dev/null +++ b/sound/soc/codecs/pcm512x.c @@ -0,0 +1,672 @@ +/* + * Driver for the PCM512x CODECs + * + * Author: Mark Brown + * Copyright 2014 Linaro Ltd + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "pcm512x.h" + +#define PCM512x_NUM_SUPPLIES 3 +static const char *pcm512x_supply_names[PCM512x_NUM_SUPPLIES] = { + "AVDD", + "DVDD", + "CPVDD", +}; + +struct pcm512x_priv { + struct regmap *regmap; + struct clk *sclk; + struct regulator_bulk_data supplies[PCM512x_NUM_SUPPLIES]; + struct notifier_block supply_nb[PCM512x_NUM_SUPPLIES]; +}; + +/* + * We can't use the same notifier block for more than one supply and + * there's no way I can see to get from a callback to the caller + * except container_of(). + */ +#define PCM512x_REGULATOR_EVENT(n) \ +static int pcm512x_regulator_event_##n(struct notifier_block *nb, \ + unsigned long event, void *data) \ +{ \ + struct pcm512x_priv *pcm512x = container_of(nb, struct pcm512x_priv, \ + supply_nb[n]); \ + if (event & REGULATOR_EVENT_DISABLE) { \ + regcache_mark_dirty(pcm512x->regmap); \ + regcache_cache_only(pcm512x->regmap, true); \ + } \ + return 0; \ +} + +PCM512x_REGULATOR_EVENT(0) +PCM512x_REGULATOR_EVENT(1) +PCM512x_REGULATOR_EVENT(2) + +static const struct reg_default pcm512x_reg_defaults[] = { + { PCM512x_RESET, 0x00 }, + { PCM512x_POWER, 0x00 }, + { PCM512x_MUTE, 0x00 }, + { PCM512x_DSP, 0x00 }, + { PCM512x_PLL_REF, 0x00 }, + { PCM512x_DAC_ROUTING, 0x11 }, + { PCM512x_DSP_PROGRAM, 0x01 }, + { PCM512x_CLKDET, 0x00 }, + { PCM512x_AUTO_MUTE, 0x00 }, + { PCM512x_ERROR_DETECT, 0x00 }, + { PCM512x_DIGITAL_VOLUME_1, 0x00 }, + { PCM512x_DIGITAL_VOLUME_2, 0x30 }, + { PCM512x_DIGITAL_VOLUME_3, 0x30 }, + { PCM512x_DIGITAL_MUTE_1, 0x22 }, + { PCM512x_DIGITAL_MUTE_2, 0x00 }, + { PCM512x_DIGITAL_MUTE_3, 0x07 }, +}; + +static bool pcm512x_readable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case PCM512x_RESET: + case PCM512x_POWER: + case PCM512x_MUTE: + case PCM512x_PLL_EN: + case PCM512x_SPI_MISO_FUNCTION: + case PCM512x_DSP: + case PCM512x_GPIO_EN: + case PCM512x_BCLK_LRCLK_CFG: + case PCM512x_DSP_GPIO_INPUT: + case PCM512x_MASTER_MODE: + case PCM512x_PLL_REF: + case PCM512x_PLL_COEFF_0: + case PCM512x_PLL_COEFF_1: + case PCM512x_PLL_COEFF_2: + case PCM512x_PLL_COEFF_3: + case PCM512x_PLL_COEFF_4: + case PCM512x_DSP_CLKDIV: + case PCM512x_DAC_CLKDIV: + case PCM512x_NCP_CLKDIV: + case PCM512x_OSR_CLKDIV: + case PCM512x_MASTER_CLKDIV_1: + case PCM512x_MASTER_CLKDIV_2: + case PCM512x_FS_SPEED_MODE: + case PCM512x_IDAC_1: + case PCM512x_IDAC_2: + case PCM512x_ERROR_DETECT: + case PCM512x_I2S_1: + case PCM512x_I2S_2: + case PCM512x_DAC_ROUTING: + case PCM512x_DSP_PROGRAM: + case PCM512x_CLKDET: + case PCM512x_AUTO_MUTE: + case PCM512x_DIGITAL_VOLUME_1: + case PCM512x_DIGITAL_VOLUME_2: + case PCM512x_DIGITAL_VOLUME_3: + case PCM512x_DIGITAL_MUTE_1: + case PCM512x_DIGITAL_MUTE_2: + case PCM512x_DIGITAL_MUTE_3: + case PCM512x_GPIO_OUTPUT_1: + case PCM512x_GPIO_OUTPUT_2: + case PCM512x_GPIO_OUTPUT_3: + case PCM512x_GPIO_OUTPUT_4: + case PCM512x_GPIO_OUTPUT_5: + case PCM512x_GPIO_OUTPUT_6: + case PCM512x_GPIO_CONTROL_1: + case PCM512x_GPIO_CONTROL_2: + case PCM512x_OVERFLOW: + case PCM512x_RATE_DET_1: + case PCM512x_RATE_DET_2: + case PCM512x_RATE_DET_3: + case PCM512x_RATE_DET_4: + case PCM512x_ANALOG_MUTE_DET: + case PCM512x_GPIN: + case PCM512x_DIGITAL_MUTE_DET: + return true; + default: + return false; + } +} + +static bool pcm512x_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case PCM512x_PLL_EN: + case PCM512x_OVERFLOW: + case PCM512x_RATE_DET_1: + case PCM512x_RATE_DET_2: + case PCM512x_RATE_DET_3: + case PCM512x_RATE_DET_4: + case PCM512x_ANALOG_MUTE_DET: + case PCM512x_GPIN: + case PCM512x_DIGITAL_MUTE_DET: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(digital_tlv, -10350, 50, 1); + +static const char *pcm512x_dsp_program_texts[] = { + "FIR interpolation with de-emphasis", + "Low latency IIR with de-emphasis", + "Fixed process flow", + "High attenuation with de-emphasis", + "Ringing-less low latency FIR", +}; + +static const unsigned int pcm512x_dsp_program_values[] = { + 1, + 2, + 3, + 5, + 7, +}; + +static const SOC_VALUE_ENUM_SINGLE_DECL(pcm512x_dsp_program, + PCM512x_DSP_PROGRAM, 0, 0x1f, + pcm512x_dsp_program_texts, + pcm512x_dsp_program_values); + +static const char *pcm512x_clk_missing_text[] = { + "1s", "2s", "3s", "4s", "5s", "6s", "7s", "8s" +}; + +static const struct soc_enum pcm512x_clk_missing = + SOC_ENUM_SINGLE(PCM512x_CLKDET, 0, 8, pcm512x_clk_missing_text); + +static const char *pcm512x_autom_text[] = { + "21ms", "106ms", "213ms", "533ms", "1.07s", "2.13s", "5.33s", "10.66s" +}; + +static const struct soc_enum pcm512x_autom_l = + SOC_ENUM_SINGLE(PCM512x_AUTO_MUTE, PCM512x_ATML_SHIFT, 8, + pcm512x_autom_text); + +static const struct soc_enum pcm512x_autom_r = + SOC_ENUM_SINGLE(PCM512x_AUTO_MUTE, PCM512x_ATMR_SHIFT, 8, + pcm512x_autom_text); + +static const char *pcm512x_ramp_rate_text[] = { + "1 sample/update", "2 samples/update", "4 samples/update", + "Immediate" +}; + +static const struct soc_enum pcm512x_vndf = + SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_1, PCM512x_VNDF_SHIFT, 4, + pcm512x_ramp_rate_text); + +static const struct soc_enum pcm512x_vnuf = + SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_1, PCM512x_VNUF_SHIFT, 4, + pcm512x_ramp_rate_text); + +static const struct soc_enum pcm512x_vedf = + SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_2, PCM512x_VEDF_SHIFT, 4, + pcm512x_ramp_rate_text); + +static const char *pcm512x_ramp_step_text[] = { + "4dB/step", "2dB/step", "1dB/step", "0.5dB/step" +}; + +static const struct soc_enum pcm512x_vnds = + SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_1, PCM512x_VNDS_SHIFT, 4, + pcm512x_ramp_step_text); + +static const struct soc_enum pcm512x_vnus = + SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_1, PCM512x_VNUS_SHIFT, 4, + pcm512x_ramp_step_text); + +static const struct soc_enum pcm512x_veds = + SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_2, PCM512x_VEDS_SHIFT, 4, + pcm512x_ramp_step_text); + +static const struct snd_kcontrol_new pcm512x_controls[] = { +SOC_DOUBLE_R_TLV("Playback Digital Volume", PCM512x_DIGITAL_VOLUME_2, + PCM512x_DIGITAL_VOLUME_3, 0, 255, 1, digital_tlv), +SOC_DOUBLE("Playback Digital Switch", PCM512x_MUTE, PCM512x_RQML_SHIFT, + PCM512x_RQMR_SHIFT, 1, 1), + +SOC_SINGLE("Deemphasis Switch", PCM512x_DSP, PCM512x_DEMP_SHIFT, 1, 1), +SOC_VALUE_ENUM("DSP Program", pcm512x_dsp_program), + +SOC_ENUM("Clock Missing Period", pcm512x_clk_missing), +SOC_ENUM("Auto Mute Time Left", pcm512x_autom_l), +SOC_ENUM("Auto Mute Time Right", pcm512x_autom_r), +SOC_SINGLE("Auto Mute Mono Switch", PCM512x_DIGITAL_MUTE_3, + PCM512x_ACTL_SHIFT, 1, 0), +SOC_DOUBLE("Auto Mute Switch", PCM512x_DIGITAL_MUTE_3, PCM512x_AMLE_SHIFT, + PCM512x_AMLR_SHIFT, 1, 0), + +SOC_ENUM("Volume Ramp Down Rate", pcm512x_vndf), +SOC_ENUM("Volume Ramp Down Step", pcm512x_vnds), +SOC_ENUM("Volume Ramp Up Rate", pcm512x_vnuf), +SOC_ENUM("Volume Ramp Up Step", pcm512x_vnus), +SOC_ENUM("Volume Ramp Down Emergency Rate", pcm512x_vedf), +SOC_ENUM("Volume Ramp Down Emergency Step", pcm512x_veds), +}; + +static const struct snd_soc_dapm_widget pcm512x_dapm_widgets[] = { +SND_SOC_DAPM_DAC("DACL", NULL, SND_SOC_NOPM, 0, 0), +SND_SOC_DAPM_DAC("DACR", NULL, SND_SOC_NOPM, 0, 0), + +SND_SOC_DAPM_OUTPUT("OUTL"), +SND_SOC_DAPM_OUTPUT("OUTR"), +}; + +static const struct snd_soc_dapm_route pcm512x_dapm_routes[] = { + { "DACL", NULL, "Playback" }, + { "DACR", NULL, "Playback" }, + + { "OUTL", NULL, "DACL" }, + { "OUTR", NULL, "DACR" }, +}; + +static int pcm512x_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct pcm512x_priv *pcm512x = dev_get_drvdata(codec->dev); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + ret = regmap_update_bits(pcm512x->regmap, PCM512x_POWER, + PCM512x_RQST, 0); + if (ret != 0) { + dev_err(codec->dev, "Failed to remove standby: %d\n", + ret); + return ret; + } + break; + + case SND_SOC_BIAS_OFF: + ret = regmap_update_bits(pcm512x->regmap, PCM512x_POWER, + PCM512x_RQST, PCM512x_RQST); + if (ret != 0) { + dev_err(codec->dev, "Failed to request standby: %d\n", + ret); + return ret; + } + break; + } + + codec->dapm.bias_level = level; + + return 0; +} + +static struct snd_soc_dai_driver pcm512x_dai = { + .name = "pcm512x-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE + }, +}; + +static struct snd_soc_codec_driver pcm512x_codec_driver = { + .set_bias_level = pcm512x_set_bias_level, + .idle_bias_off = true, + + .controls = pcm512x_controls, + .num_controls = ARRAY_SIZE(pcm512x_controls), + .dapm_widgets = pcm512x_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(pcm512x_dapm_widgets), + .dapm_routes = pcm512x_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(pcm512x_dapm_routes), +}; + +static const struct regmap_config pcm512x_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .readable_reg = pcm512x_readable, + .volatile_reg = pcm512x_volatile, + + .max_register = PCM512x_MAX_REGISTER, + .reg_defaults = pcm512x_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(pcm512x_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + +static const struct of_device_id pcm512x_of_match[] = { + { .compatible = "ti,pcm5121", }, + { .compatible = "ti,pcm5122", }, + { } +}; +MODULE_DEVICE_TABLE(of, pcm512x_of_match); + +static int pcm512x_probe(struct device *dev, struct regmap *regmap) +{ + struct pcm512x_priv *pcm512x; + int i, ret; + + pcm512x = devm_kzalloc(dev, sizeof(struct pcm512x_priv), GFP_KERNEL); + if (!pcm512x) + return -ENOMEM; + + dev_set_drvdata(dev, pcm512x); + pcm512x->regmap = regmap; + + for (i = 0; i < ARRAY_SIZE(pcm512x->supplies); i++) + pcm512x->supplies[i].supply = pcm512x_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, ARRAY_SIZE(pcm512x->supplies), + pcm512x->supplies); + if (ret != 0) { + dev_err(dev, "Failed to get supplies: %d\n", ret); + return ret; + } + + pcm512x->supply_nb[0].notifier_call = pcm512x_regulator_event_0; + pcm512x->supply_nb[1].notifier_call = pcm512x_regulator_event_1; + pcm512x->supply_nb[2].notifier_call = pcm512x_regulator_event_2; + + for (i = 0; i < ARRAY_SIZE(pcm512x->supplies); i++) { + ret = regulator_register_notifier(pcm512x->supplies[i].consumer, + &pcm512x->supply_nb[i]); + if (ret != 0) { + dev_err(dev, + "Failed to register regulator notifier: %d\n", + ret); + } + } + + ret = regulator_bulk_enable(ARRAY_SIZE(pcm512x->supplies), + pcm512x->supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + /* Reset the device, verifying I/O in the process for I2C */ + ret = regmap_write(regmap, PCM512x_RESET, + PCM512x_RSTM | PCM512x_RSTR); + if (ret != 0) { + dev_err(dev, "Failed to reset device: %d\n", ret); + goto err; + } + + ret = regmap_write(regmap, PCM512x_RESET, 0); + if (ret != 0) { + dev_err(dev, "Failed to reset device: %d\n", ret); + goto err; + } + + pcm512x->sclk = devm_clk_get(dev, NULL); + if (IS_ERR(pcm512x->sclk)) { + if (PTR_ERR(pcm512x->sclk) == -EPROBE_DEFER) + return -EPROBE_DEFER; + + dev_info(dev, "No SCLK, using BCLK: %ld\n", + PTR_ERR(pcm512x->sclk)); + + /* Disable reporting of missing SCLK as an error */ + regmap_update_bits(regmap, PCM512x_ERROR_DETECT, + PCM512x_IDCH, PCM512x_IDCH); + + /* Switch PLL input to BCLK */ + regmap_update_bits(regmap, PCM512x_PLL_REF, + PCM512x_SREF, PCM512x_SREF); + } else { + ret = clk_prepare_enable(pcm512x->sclk); + if (ret != 0) { + dev_err(dev, "Failed to enable SCLK: %d\n", ret); + return ret; + } + } + + /* Default to standby mode */ + ret = regmap_update_bits(pcm512x->regmap, PCM512x_POWER, + PCM512x_RQST, PCM512x_RQST); + if (ret != 0) { + dev_err(dev, "Failed to request standby: %d\n", + ret); + goto err_clk; + } + + pm_runtime_set_active(dev); + pm_runtime_enable(dev); + pm_runtime_idle(dev); + + ret = snd_soc_register_codec(dev, &pcm512x_codec_driver, + &pcm512x_dai, 1); + if (ret != 0) { + dev_err(dev, "Failed to register CODEC: %d\n", ret); + goto err_pm; + } + + return 0; + +err_pm: + pm_runtime_disable(dev); +err_clk: + if (!IS_ERR(pcm512x->sclk)) + clk_disable_unprepare(pcm512x->sclk); +err: + regulator_bulk_disable(ARRAY_SIZE(pcm512x->supplies), + pcm512x->supplies); + return ret; +} + +static void pcm512x_remove(struct device *dev) +{ + struct pcm512x_priv *pcm512x = dev_get_drvdata(dev); + + snd_soc_unregister_codec(dev); + pm_runtime_disable(dev); + if (!IS_ERR(pcm512x->sclk)) + clk_disable_unprepare(pcm512x->sclk); + regulator_bulk_disable(ARRAY_SIZE(pcm512x->supplies), + pcm512x->supplies); +} + +static int pcm512x_suspend(struct device *dev) +{ + struct pcm512x_priv *pcm512x = dev_get_drvdata(dev); + int ret; + + ret = regmap_update_bits(pcm512x->regmap, PCM512x_POWER, + PCM512x_RQPD, PCM512x_RQPD); + if (ret != 0) { + dev_err(dev, "Failed to request power down: %d\n", ret); + return ret; + } + + ret = regulator_bulk_disable(ARRAY_SIZE(pcm512x->supplies), + pcm512x->supplies); + if (ret != 0) { + dev_err(dev, "Failed to disable supplies: %d\n", ret); + return ret; + } + + if (!IS_ERR(pcm512x->sclk)) + clk_disable_unprepare(pcm512x->sclk); + + return 0; +} + +static int pcm512x_resume(struct device *dev) +{ + struct pcm512x_priv *pcm512x = dev_get_drvdata(dev); + int ret; + + if (!IS_ERR(pcm512x->sclk)) { + ret = clk_prepare_enable(pcm512x->sclk); + if (ret != 0) { + dev_err(dev, "Failed to enable SCLK: %d\n", ret); + return ret; + } + } + + ret = regulator_bulk_enable(ARRAY_SIZE(pcm512x->supplies), + pcm512x->supplies); + if (ret != 0) { + dev_err(dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + regcache_cache_only(pcm512x->regmap, false); + ret = regcache_sync(pcm512x->regmap); + if (ret != 0) { + dev_err(dev, "Failed to sync cache: %d\n", ret); + return ret; + } + + ret = regmap_update_bits(pcm512x->regmap, PCM512x_POWER, + PCM512x_RQPD, 0); + if (ret != 0) { + dev_err(dev, "Failed to remove power down: %d\n", ret); + return ret; + } + + return 0; +} + +static const struct dev_pm_ops pcm512x_pm_ops = { + SET_RUNTIME_PM_OPS(pcm512x_suspend, pcm512x_resume, NULL) +}; + +#if IS_ENABLED(CONFIG_I2C) +static int pcm512x_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct regmap *regmap; + + regmap = devm_regmap_init_i2c(i2c, &pcm512x_regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return pcm512x_probe(&i2c->dev, regmap); +} + +static int pcm512x_i2c_remove(struct i2c_client *i2c) +{ + pcm512x_remove(&i2c->dev); + return 0; +} + +static const struct i2c_device_id pcm512x_i2c_id[] = { + { "pcm5121", }, + { "pcm5122", }, + { } +}; +MODULE_DEVICE_TABLE(i2c, pcm512x_i2c_id); + +static struct i2c_driver pcm512x_i2c_driver = { + .probe = pcm512x_i2c_probe, + .remove = pcm512x_i2c_remove, + .id_table = pcm512x_i2c_id, + .driver = { + .name = "pcm512x", + .owner = THIS_MODULE, + .of_match_table = pcm512x_of_match, + .pm = &pcm512x_pm_ops, + }, +}; +#endif + +#if defined(CONFIG_SPI_MASTER) +static int pcm512x_spi_probe(struct spi_device *spi) +{ + struct regmap *regmap; + int ret; + + regmap = devm_regmap_init_spi(spi, &pcm512x_regmap); + if (IS_ERR(regmap)) { + ret = PTR_ERR(regmap); + return ret; + } + + return pcm512x_probe(&spi->dev, regmap); +} + +static int pcm512x_spi_remove(struct spi_device *spi) +{ + pcm512x_remove(&spi->dev); + return 0; +} + +static const struct spi_device_id pcm512x_spi_id[] = { + { "pcm5121", }, + { "pcm5122", }, + { }, +}; +MODULE_DEVICE_TABLE(spi, pcm512x_spi_id); + +static struct spi_driver pcm512x_spi_driver = { + .probe = pcm512x_spi_probe, + .remove = pcm512x_spi_remove, + .id_table = pcm512x_spi_id, + .driver = { + .name = "pcm512x", + .owner = THIS_MODULE, + .of_match_table = pcm512x_of_match, + .pm = &pcm512x_pm_ops, + }, +}; +#endif + +static int __init pcm512x_modinit(void) +{ + int ret = 0; + +#if IS_ENABLED(CONFIG_I2C) + ret = i2c_add_driver(&pcm512x_i2c_driver); + if (ret) { + printk(KERN_ERR "Failed to register pcm512x I2C driver: %d\n", + ret); + } +#endif +#if defined(CONFIG_SPI_MASTER) + ret = spi_register_driver(&pcm512x_spi_driver); + if (ret != 0) { + printk(KERN_ERR "Failed to register pcm512x SPI driver: %d\n", + ret); + } +#endif + return ret; +} +module_init(pcm512x_modinit); + +static void __exit pcm512x_exit(void) +{ +#if IS_ENABLED(CONFIG_I2C) + i2c_del_driver(&pcm512x_i2c_driver); +#endif +#if defined(CONFIG_SPI_MASTER) + spi_unregister_driver(&pcm512x_spi_driver); +#endif +} +module_exit(pcm512x_exit); + +MODULE_DESCRIPTION("ASoC PCM512x codec driver"); +MODULE_AUTHOR("Mark Brown "); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/pcm512x.h b/sound/soc/codecs/pcm512x.h new file mode 100644 index 000000000000..b2f518ecb35c --- /dev/null +++ b/sound/soc/codecs/pcm512x.h @@ -0,0 +1,142 @@ +/* + * Driver for the PCM512x CODECs + * + * Author: Mark Brown + * Copyright 2014 Linaro Ltd + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#ifndef _SND_SOC_PCM512X +#define _SND_SOC_PCM512X + +#define PCM512x_PAGE_0_BASE 0 + +#define PCM512x_PAGE 0 + +#define PCM512x_RESET (PCM512x_PAGE_0_BASE + 1) +#define PCM512x_POWER (PCM512x_PAGE_0_BASE + 2) +#define PCM512x_MUTE (PCM512x_PAGE_0_BASE + 3) +#define PCM512x_PLL_EN (PCM512x_PAGE_0_BASE + 4) +#define PCM512x_SPI_MISO_FUNCTION (PCM512x_PAGE_0_BASE + 6) +#define PCM512x_DSP (PCM512x_PAGE_0_BASE + 7) +#define PCM512x_GPIO_EN (PCM512x_PAGE_0_BASE + 8) +#define PCM512x_BCLK_LRCLK_CFG (PCM512x_PAGE_0_BASE + 9) +#define PCM512x_DSP_GPIO_INPUT (PCM512x_PAGE_0_BASE + 10) +#define PCM512x_MASTER_MODE (PCM512x_PAGE_0_BASE + 12) +#define PCM512x_PLL_REF (PCM512x_PAGE_0_BASE + 13) +#define PCM512x_PLL_COEFF_0 (PCM512x_PAGE_0_BASE + 20) +#define PCM512x_PLL_COEFF_1 (PCM512x_PAGE_0_BASE + 21) +#define PCM512x_PLL_COEFF_2 (PCM512x_PAGE_0_BASE + 22) +#define PCM512x_PLL_COEFF_3 (PCM512x_PAGE_0_BASE + 23) +#define PCM512x_PLL_COEFF_4 (PCM512x_PAGE_0_BASE + 24) +#define PCM512x_DSP_CLKDIV (PCM512x_PAGE_0_BASE + 27) +#define PCM512x_DAC_CLKDIV (PCM512x_PAGE_0_BASE + 28) +#define PCM512x_NCP_CLKDIV (PCM512x_PAGE_0_BASE + 29) +#define PCM512x_OSR_CLKDIV (PCM512x_PAGE_0_BASE + 30) +#define PCM512x_MASTER_CLKDIV_1 (PCM512x_PAGE_0_BASE + 32) +#define PCM512x_MASTER_CLKDIV_2 (PCM512x_PAGE_0_BASE + 33) +#define PCM512x_FS_SPEED_MODE (PCM512x_PAGE_0_BASE + 34) +#define PCM512x_IDAC_1 (PCM512x_PAGE_0_BASE + 35) +#define PCM512x_IDAC_2 (PCM512x_PAGE_0_BASE + 36) +#define PCM512x_ERROR_DETECT (PCM512x_PAGE_0_BASE + 37) +#define PCM512x_I2S_1 (PCM512x_PAGE_0_BASE + 40) +#define PCM512x_I2S_2 (PCM512x_PAGE_0_BASE + 41) +#define PCM512x_DAC_ROUTING (PCM512x_PAGE_0_BASE + 42) +#define PCM512x_DSP_PROGRAM (PCM512x_PAGE_0_BASE + 43) +#define PCM512x_CLKDET (PCM512x_PAGE_0_BASE + 44) +#define PCM512x_AUTO_MUTE (PCM512x_PAGE_0_BASE + 59) +#define PCM512x_DIGITAL_VOLUME_1 (PCM512x_PAGE_0_BASE + 60) +#define PCM512x_DIGITAL_VOLUME_2 (PCM512x_PAGE_0_BASE + 61) +#define PCM512x_DIGITAL_VOLUME_3 (PCM512x_PAGE_0_BASE + 62) +#define PCM512x_DIGITAL_MUTE_1 (PCM512x_PAGE_0_BASE + 63) +#define PCM512x_DIGITAL_MUTE_2 (PCM512x_PAGE_0_BASE + 64) +#define PCM512x_DIGITAL_MUTE_3 (PCM512x_PAGE_0_BASE + 65) +#define PCM512x_GPIO_OUTPUT_1 (PCM512x_PAGE_0_BASE + 80) +#define PCM512x_GPIO_OUTPUT_2 (PCM512x_PAGE_0_BASE + 81) +#define PCM512x_GPIO_OUTPUT_3 (PCM512x_PAGE_0_BASE + 82) +#define PCM512x_GPIO_OUTPUT_4 (PCM512x_PAGE_0_BASE + 83) +#define PCM512x_GPIO_OUTPUT_5 (PCM512x_PAGE_0_BASE + 84) +#define PCM512x_GPIO_OUTPUT_6 (PCM512x_PAGE_0_BASE + 85) +#define PCM512x_GPIO_CONTROL_1 (PCM512x_PAGE_0_BASE + 86) +#define PCM512x_GPIO_CONTROL_2 (PCM512x_PAGE_0_BASE + 87) +#define PCM512x_OVERFLOW (PCM512x_PAGE_0_BASE + 90) +#define PCM512x_RATE_DET_1 (PCM512x_PAGE_0_BASE + 91) +#define PCM512x_RATE_DET_2 (PCM512x_PAGE_0_BASE + 92) +#define PCM512x_RATE_DET_3 (PCM512x_PAGE_0_BASE + 93) +#define PCM512x_RATE_DET_4 (PCM512x_PAGE_0_BASE + 94) +#define PCM512x_ANALOG_MUTE_DET (PCM512x_PAGE_0_BASE + 108) +#define PCM512x_GPIN (PCM512x_PAGE_0_BASE + 119) +#define PCM512x_DIGITAL_MUTE_DET (PCM512x_PAGE_0_BASE + 120) + +#define PCM512x_MAX_REGISTER (PCM512x_PAGE_0_BASE + 120) + +/* Page 0, Register 1 - reset */ +#define PCM512x_RSTR (1 << 0) +#define PCM512x_RSTM (1 << 4) + +/* Page 0, Register 2 - power */ +#define PCM512x_RQPD (1 << 0) +#define PCM512x_RQPD_SHIFT 0 +#define PCM512x_RQST (1 << 4) +#define PCM512x_RQST_SHIFT 4 + +/* Page 0, Register 3 - mute */ +#define PCM512x_RQMR_SHIFT 0 +#define PCM512x_RQML_SHIFT 4 + +/* Page 0, Register 4 - PLL */ +#define PCM512x_PLCE (1 << 0) +#define PCM512x_RLCE_SHIFT 0 +#define PCM512x_PLCK (1 << 4) +#define PCM512x_PLCK_SHIFT 4 + +/* Page 0, Register 7 - DSP */ +#define PCM512x_SDSL (1 << 0) +#define PCM512x_SDSL_SHIFT 0 +#define PCM512x_DEMP (1 << 4) +#define PCM512x_DEMP_SHIFT 4 + +/* Page 0, Register 13 - PLL reference */ +#define PCM512x_SREF (1 << 4) + +/* Page 0, Register 37 - Error detection */ +#define PCM512x_IPLK (1 << 0) +#define PCM512x_DCAS (1 << 1) +#define PCM512x_IDCM (1 << 2) +#define PCM512x_IDCH (1 << 3) +#define PCM512x_IDSK (1 << 4) +#define PCM512x_IDBK (1 << 5) +#define PCM512x_IDFS (1 << 6) + +/* Page 0, Register 42 - DAC routing */ +#define PCM512x_AUPR_SHIFT 0 +#define PCM512x_AUPL_SHIFT 4 + +/* Page 0, Register 59 - auto mute */ +#define PCM512x_ATMR_SHIFT 0 +#define PCM512x_ATML_SHIFT 4 + +/* Page 0, Register 63 - ramp rates */ +#define PCM512x_VNDF_SHIFT 6 +#define PCM512x_VNDS_SHIFT 4 +#define PCM512x_VNUF_SHIFT 2 +#define PCM512x_VNUS_SHIFT 0 + +/* Page 0, Register 64 - emergency ramp rates */ +#define PCM512x_VEDF_SHIFT 6 +#define PCM512x_VEDS_SHIFT 4 + +/* Page 0, Register 65 - Digital mute enables */ +#define PCM512x_ACTL_SHIFT 2 +#define PCM512x_AMLE_SHIFT 1 +#define PCM512x_AMLR_SHIFT 0 + +#endif -- cgit v1.2.3 From 06d0ffcc5c12ad49786141fa9768da38485a8a61 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 6 Feb 2014 14:33:52 +0000 Subject: ASoC: pcm512x: More constification Since the core now takes const strings for enums we should be constifying them (and the regulator supplies while we're at it). Reported-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/pcm512x.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 5ad3e9aa3cb4..1150381fc373 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -30,7 +30,7 @@ #include "pcm512x.h" #define PCM512x_NUM_SUPPLIES 3 -static const char *pcm512x_supply_names[PCM512x_NUM_SUPPLIES] = { +static const char * const pcm512x_supply_names[PCM512x_NUM_SUPPLIES] = { "AVDD", "DVDD", "CPVDD", @@ -167,7 +167,7 @@ static bool pcm512x_volatile(struct device *dev, unsigned int reg) static const DECLARE_TLV_DB_SCALE(digital_tlv, -10350, 50, 1); -static const char *pcm512x_dsp_program_texts[] = { +static const char * const pcm512x_dsp_program_texts[] = { "FIR interpolation with de-emphasis", "Low latency IIR with de-emphasis", "Fixed process flow", @@ -188,14 +188,14 @@ static const SOC_VALUE_ENUM_SINGLE_DECL(pcm512x_dsp_program, pcm512x_dsp_program_texts, pcm512x_dsp_program_values); -static const char *pcm512x_clk_missing_text[] = { +static const char * const pcm512x_clk_missing_text[] = { "1s", "2s", "3s", "4s", "5s", "6s", "7s", "8s" }; static const struct soc_enum pcm512x_clk_missing = SOC_ENUM_SINGLE(PCM512x_CLKDET, 0, 8, pcm512x_clk_missing_text); -static const char *pcm512x_autom_text[] = { +static const char * const pcm512x_autom_text[] = { "21ms", "106ms", "213ms", "533ms", "1.07s", "2.13s", "5.33s", "10.66s" }; @@ -207,7 +207,7 @@ static const struct soc_enum pcm512x_autom_r = SOC_ENUM_SINGLE(PCM512x_AUTO_MUTE, PCM512x_ATMR_SHIFT, 8, pcm512x_autom_text); -static const char *pcm512x_ramp_rate_text[] = { +static const char * const pcm512x_ramp_rate_text[] = { "1 sample/update", "2 samples/update", "4 samples/update", "Immediate" }; @@ -224,7 +224,7 @@ static const struct soc_enum pcm512x_vedf = SOC_ENUM_SINGLE(PCM512x_DIGITAL_MUTE_2, PCM512x_VEDF_SHIFT, 4, pcm512x_ramp_rate_text); -static const char *pcm512x_ramp_step_text[] = { +static const char * const pcm512x_ramp_step_text[] = { "4dB/step", "2dB/step", "1dB/step", "0.5dB/step" }; -- cgit v1.2.3 From 096ae5444b8600bbee0501b01987094657a1458e Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 5 Feb 2014 21:54:32 +0100 Subject: ASoC: cs42l73: Constify rate constraints The rate constraints in this driver are shared between all device instances. It should not be (and is not) modified at runtime, so make them const. While we are at it also change the type for the rates array from u32 to unsigned int. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l73.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 7cae046c7dd0..69c8e2de7d0e 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1108,7 +1108,7 @@ static int cs42l73_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) return 0; } -static u32 cs42l73_asrc_rates[] = { +static const unsigned int cs42l73_asrc_rates[] = { 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 }; @@ -1241,7 +1241,7 @@ static int cs42l73_set_tristate(struct snd_soc_dai *dai, int tristate) 0x7F, tristate << 7); } -static struct snd_pcm_hw_constraint_list constraints_12_24 = { +static const struct snd_pcm_hw_constraint_list constraints_12_24 = { .count = ARRAY_SIZE(cs42l73_asrc_rates), .list = cs42l73_asrc_rates, }; -- cgit v1.2.3 From f75ac2d9bd8211c1890ad591046aef0783ad6416 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 5 Feb 2014 21:54:33 +0100 Subject: ASoC: ssm2602: Constify rate constraints The rate constraints in this driver are shared between all device instances. It should not be (and is not) modified at runtime, so make them const. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2602.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index af76bbd1b24f..f444d585b916 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -52,7 +52,7 @@ enum ssm2602_type { /* codec private data */ struct ssm2602_priv { unsigned int sysclk; - struct snd_pcm_hw_constraint_list *sysclk_constraints; + const struct snd_pcm_hw_constraint_list *sysclk_constraints; struct regmap *regmap; @@ -197,7 +197,7 @@ static const unsigned int ssm2602_rates_12288000[] = { 8000, 16000, 32000, 48000, 96000, }; -static struct snd_pcm_hw_constraint_list ssm2602_constraints_12288000 = { +static const struct snd_pcm_hw_constraint_list ssm2602_constraints_12288000 = { .list = ssm2602_rates_12288000, .count = ARRAY_SIZE(ssm2602_rates_12288000), }; @@ -206,7 +206,7 @@ static const unsigned int ssm2602_rates_11289600[] = { 8000, 44100, 88200, }; -static struct snd_pcm_hw_constraint_list ssm2602_constraints_11289600 = { +static const struct snd_pcm_hw_constraint_list ssm2602_constraints_11289600 = { .list = ssm2602_rates_11289600, .count = ARRAY_SIZE(ssm2602_rates_11289600), }; -- cgit v1.2.3 From 0d76fc6a47b183d519c47e40971e74cfd96cca85 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 5 Feb 2014 21:54:34 +0100 Subject: ASoC: twl6040: Constify rate constraints The rate constraints in this driver are shared between all device instances. It should not be (and is not) modified at runtime, so make them const. Signed-off-by: Lars-Peter Clausen Acked-by: Peter Ujfalusi Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 0afe8bef6765..cb642c927dc8 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -81,7 +81,7 @@ struct twl6040_data { }; /* set of rates for each pll: low-power and high-performance */ -static unsigned int lp_rates[] = { +static const unsigned int lp_rates[] = { 8000, 11250, 16000, @@ -93,7 +93,7 @@ static unsigned int lp_rates[] = { 96000, }; -static unsigned int hp_rates[] = { +static const unsigned int hp_rates[] = { 8000, 16000, 32000, @@ -101,7 +101,7 @@ static unsigned int hp_rates[] = { 96000, }; -static struct snd_pcm_hw_constraint_list sysclk_constraints[] = { +static const struct snd_pcm_hw_constraint_list sysclk_constraints[] = { { .count = ARRAY_SIZE(lp_rates), .list = lp_rates, }, { .count = ARRAY_SIZE(hp_rates), .list = hp_rates, }, }; -- cgit v1.2.3 From 70bad2c780abc6744ab2ab5832b0c9f3de608dad Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 5 Feb 2014 21:54:35 +0100 Subject: ASoC: wm8741: Constify rate constraints The rate constraints in this driver are shared between all device instances. It should not be (and is not) modified at runtime, so make them const. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8741.c | 34 +++++++++++++++++----------------- 1 file changed, 17 insertions(+), 17 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 2895c8d3b5e4..dd02ebf88015 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -44,7 +44,7 @@ struct wm8741_priv { struct regmap *regmap; struct regulator_bulk_data supplies[WM8741_NUM_SUPPLIES]; unsigned int sysclk; - struct snd_pcm_hw_constraint_list *sysclk_constraints; + const struct snd_pcm_hw_constraint_list *sysclk_constraints; }; static const struct reg_default wm8741_reg_defaults[] = { @@ -122,74 +122,74 @@ static struct { { 6, 768 }, }; -static unsigned int rates_11289[] = { +static const unsigned int rates_11289[] = { 44100, 88235, }; -static struct snd_pcm_hw_constraint_list constraints_11289 = { +static const struct snd_pcm_hw_constraint_list constraints_11289 = { .count = ARRAY_SIZE(rates_11289), .list = rates_11289, }; -static unsigned int rates_12288[] = { +static const unsigned int rates_12288[] = { 32000, 48000, 96000, }; -static struct snd_pcm_hw_constraint_list constraints_12288 = { +static const struct snd_pcm_hw_constraint_list constraints_12288 = { .count = ARRAY_SIZE(rates_12288), .list = rates_12288, }; -static unsigned int rates_16384[] = { +static const unsigned int rates_16384[] = { 32000, }; -static struct snd_pcm_hw_constraint_list constraints_16384 = { +static const struct snd_pcm_hw_constraint_list constraints_16384 = { .count = ARRAY_SIZE(rates_16384), .list = rates_16384, }; -static unsigned int rates_16934[] = { +static const unsigned int rates_16934[] = { 44100, 88235, }; -static struct snd_pcm_hw_constraint_list constraints_16934 = { +static const struct snd_pcm_hw_constraint_list constraints_16934 = { .count = ARRAY_SIZE(rates_16934), .list = rates_16934, }; -static unsigned int rates_18432[] = { +static const unsigned int rates_18432[] = { 48000, 96000, }; -static struct snd_pcm_hw_constraint_list constraints_18432 = { +static const struct snd_pcm_hw_constraint_list constraints_18432 = { .count = ARRAY_SIZE(rates_18432), .list = rates_18432, }; -static unsigned int rates_22579[] = { +static const unsigned int rates_22579[] = { 44100, 88235, 1764000 }; -static struct snd_pcm_hw_constraint_list constraints_22579 = { +static const struct snd_pcm_hw_constraint_list constraints_22579 = { .count = ARRAY_SIZE(rates_22579), .list = rates_22579, }; -static unsigned int rates_24576[] = { +static const unsigned int rates_24576[] = { 32000, 48000, 96000, 192000 }; -static struct snd_pcm_hw_constraint_list constraints_24576 = { +static const struct snd_pcm_hw_constraint_list constraints_24576 = { .count = ARRAY_SIZE(rates_24576), .list = rates_24576, }; -static unsigned int rates_36864[] = { +static const unsigned int rates_36864[] = { 48000, 96000, 19200 }; -static struct snd_pcm_hw_constraint_list constraints_36864 = { +static const struct snd_pcm_hw_constraint_list constraints_36864 = { .count = ARRAY_SIZE(rates_36864), .list = rates_36864, }; -- cgit v1.2.3 From b57efda1f0372d6107ced5a255384be9fd449260 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 5 Feb 2014 21:54:36 +0100 Subject: ASoC: wm8988: Constify rate constraints The rate constraints in this driver are shared between all device instances. It should not be (and is not) modified at runtime, so make them const. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8988.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index a55e1c2c382e..c6e4aba25b77 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -116,7 +116,7 @@ static bool wm8988_writeable(struct device *dev, unsigned int reg) struct wm8988_priv { struct regmap *regmap; unsigned int sysclk; - struct snd_pcm_hw_constraint_list *sysclk_constraints; + const struct snd_pcm_hw_constraint_list *sysclk_constraints; }; #define wm8988_reset(c) snd_soc_write(c, WM8988_RESET, 0) @@ -521,30 +521,30 @@ static inline int get_coeff(int mclk, int rate) /* The set of rates we can generate from the above for each SYSCLK */ -static unsigned int rates_12288[] = { +static const unsigned int rates_12288[] = { 8000, 12000, 16000, 24000, 24000, 32000, 48000, 96000, }; -static struct snd_pcm_hw_constraint_list constraints_12288 = { +static const struct snd_pcm_hw_constraint_list constraints_12288 = { .count = ARRAY_SIZE(rates_12288), .list = rates_12288, }; -static unsigned int rates_112896[] = { +static const unsigned int rates_112896[] = { 8000, 11025, 22050, 44100, }; -static struct snd_pcm_hw_constraint_list constraints_112896 = { +static const struct snd_pcm_hw_constraint_list constraints_112896 = { .count = ARRAY_SIZE(rates_112896), .list = rates_112896, }; -static unsigned int rates_12[] = { +static const unsigned int rates_12[] = { 8000, 11025, 12000, 16000, 22050, 2400, 32000, 41100, 48000, 48000, 88235, 96000, }; -static struct snd_pcm_hw_constraint_list constraints_12 = { +static const struct snd_pcm_hw_constraint_list constraints_12 = { .count = ARRAY_SIZE(rates_12), .list = rates_12, }; -- cgit v1.2.3 From 4d1a77224bac47c994ecdc776f0c09c9a0ed9d49 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 7 Feb 2014 15:40:35 +0000 Subject: ASoC: codecs: Put the CODEC drivers in a menu Now they're visible they get a bit noisy. Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index f0bdcc5abe83..7da528a2c1c5 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -8,6 +8,8 @@ config SND_SOC_I2C_AND_SPI default y if I2C=y default y if SPI_MASTER=y +menu "CODEC drivers" + config SND_SOC_ALL_CODECS tristate "Build all ASoC CODEC drivers" depends on COMPILE_TEST @@ -584,3 +586,5 @@ config SND_SOC_ML26124 config SND_SOC_TPA6130A2 tristate "Texas Instruments TPA6130A2 headphone amplifier" depends on I2C + +endmenu -- cgit v1.2.3 From e479d85ced01ea7f85c7dd21dc858af25f1493b4 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 7 Feb 2014 15:57:50 +0000 Subject: ASoC: wm8770: Depend on SPI only The device has no I2C support so it shouldn't be buildable if only I2C is enabled. Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7da528a2c1c5..a6afdf5004cb 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -474,7 +474,7 @@ config SND_SOC_WM8753 config SND_SOC_WM8770 tristate "Wolfson Microelectronics WM8770 CODEC" - depends on SND_SOC_I2C_AND_SPI + depends on SPI_MASTER config SND_SOC_WM8776 tristate "Wolfson Microelectronics WM8776 CODEC" -- cgit v1.2.3 From 806d6466076a0aebbe0a9c17294d1a13e93fabcf Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 7 Feb 2014 19:08:11 +0000 Subject: ASoC: pcm512x: Implement paging support The PCM512x devices use a paged register map covering the entire register range. Implement support for this, mapping pages in at addresses starting at 0x100 for ease of use (though since the pages are numbered from 0 there is going to be an off by one when looking at the first byte as a page number). Also mark the new registers as accessible with the exception of the coefficient RAM which is a bit fiddly and may benefit from some extra handling to linearise the blocks. Signed-off-by: Mark Brown --- sound/soc/codecs/pcm512x.c | 65 ++++++++++++++++------- sound/soc/codecs/pcm512x.h | 126 +++++++++++++++++++++++++-------------------- 2 files changed, 116 insertions(+), 75 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 1150381fc373..cdcb51e4c86f 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -66,22 +66,29 @@ PCM512x_REGULATOR_EVENT(1) PCM512x_REGULATOR_EVENT(2) static const struct reg_default pcm512x_reg_defaults[] = { - { PCM512x_RESET, 0x00 }, - { PCM512x_POWER, 0x00 }, - { PCM512x_MUTE, 0x00 }, - { PCM512x_DSP, 0x00 }, - { PCM512x_PLL_REF, 0x00 }, - { PCM512x_DAC_ROUTING, 0x11 }, - { PCM512x_DSP_PROGRAM, 0x01 }, - { PCM512x_CLKDET, 0x00 }, - { PCM512x_AUTO_MUTE, 0x00 }, - { PCM512x_ERROR_DETECT, 0x00 }, - { PCM512x_DIGITAL_VOLUME_1, 0x00 }, - { PCM512x_DIGITAL_VOLUME_2, 0x30 }, - { PCM512x_DIGITAL_VOLUME_3, 0x30 }, - { PCM512x_DIGITAL_MUTE_1, 0x22 }, - { PCM512x_DIGITAL_MUTE_2, 0x00 }, - { PCM512x_DIGITAL_MUTE_3, 0x07 }, + { PCM512x_RESET, 0x00 }, + { PCM512x_POWER, 0x00 }, + { PCM512x_MUTE, 0x00 }, + { PCM512x_DSP, 0x00 }, + { PCM512x_PLL_REF, 0x00 }, + { PCM512x_DAC_ROUTING, 0x11 }, + { PCM512x_DSP_PROGRAM, 0x01 }, + { PCM512x_CLKDET, 0x00 }, + { PCM512x_AUTO_MUTE, 0x00 }, + { PCM512x_ERROR_DETECT, 0x00 }, + { PCM512x_DIGITAL_VOLUME_1, 0x00 }, + { PCM512x_DIGITAL_VOLUME_2, 0x30 }, + { PCM512x_DIGITAL_VOLUME_3, 0x30 }, + { PCM512x_DIGITAL_MUTE_1, 0x22 }, + { PCM512x_DIGITAL_MUTE_2, 0x00 }, + { PCM512x_DIGITAL_MUTE_3, 0x07 }, + { PCM512x_OUTPUT_AMPLITUDE, 0x00 }, + { PCM512x_ANALOG_GAIN_CTRL, 0x00 }, + { PCM512x_UNDERVOLTAGE_PROT, 0x00 }, + { PCM512x_ANALOG_MUTE_CTRL, 0x00 }, + { PCM512x_ANALOG_GAIN_BOOST, 0x00 }, + { PCM512x_VCOM_CTRL_1, 0x00 }, + { PCM512x_VCOM_CTRL_2, 0x01 }, }; static bool pcm512x_readable(struct device *dev, unsigned int reg) @@ -141,9 +148,18 @@ static bool pcm512x_readable(struct device *dev, unsigned int reg) case PCM512x_ANALOG_MUTE_DET: case PCM512x_GPIN: case PCM512x_DIGITAL_MUTE_DET: + case PCM512x_OUTPUT_AMPLITUDE: + case PCM512x_ANALOG_GAIN_CTRL: + case PCM512x_UNDERVOLTAGE_PROT: + case PCM512x_ANALOG_MUTE_CTRL: + case PCM512x_ANALOG_GAIN_BOOST: + case PCM512x_VCOM_CTRL_1: + case PCM512x_VCOM_CTRL_2: + case PCM512x_CRAM_CTRL: return true; default: - return false; + /* There are 256 raw register addresses */ + return reg < 0xff; } } @@ -159,9 +175,11 @@ static bool pcm512x_volatile(struct device *dev, unsigned int reg) case PCM512x_ANALOG_MUTE_DET: case PCM512x_GPIN: case PCM512x_DIGITAL_MUTE_DET: + case PCM512x_CRAM_CTRL: return true; default: - return false; + /* There are 256 raw register addresses */ + return reg < 0xff; } } @@ -343,6 +361,14 @@ static struct snd_soc_codec_driver pcm512x_codec_driver = { .num_dapm_routes = ARRAY_SIZE(pcm512x_dapm_routes), }; +static const struct regmap_range_cfg pcm512x_range = { + .name = "Pages", .range_min = PCM512x_VIRT_BASE, + .range_max = PCM512x_MAX_REGISTER, + .selector_reg = PCM512x_PAGE, + .selector_mask = 0xff, + .window_start = 0, .window_len = 0x100, +}; + static const struct regmap_config pcm512x_regmap = { .reg_bits = 8, .val_bits = 8, @@ -350,6 +376,9 @@ static const struct regmap_config pcm512x_regmap = { .readable_reg = pcm512x_readable, .volatile_reg = pcm512x_volatile, + .ranges = &pcm512x_range, + .num_ranges = 1, + .max_register = PCM512x_MAX_REGISTER, .reg_defaults = pcm512x_reg_defaults, .num_reg_defaults = ARRAY_SIZE(pcm512x_reg_defaults), diff --git a/sound/soc/codecs/pcm512x.h b/sound/soc/codecs/pcm512x.h index b2f518ecb35c..e58743c965d6 100644 --- a/sound/soc/codecs/pcm512x.h +++ b/sound/soc/codecs/pcm512x.h @@ -17,66 +17,78 @@ #ifndef _SND_SOC_PCM512X #define _SND_SOC_PCM512X -#define PCM512x_PAGE_0_BASE 0 +#define PCM512x_VIRT_BASE 0x100 +#define PCM512x_PAGE_LEN 0x100 +#define PCM512x_PAGE_BASE(n) (PCM512x_VIRT_BASE + (PCM512x_PAGE_LEN * n)) #define PCM512x_PAGE 0 -#define PCM512x_RESET (PCM512x_PAGE_0_BASE + 1) -#define PCM512x_POWER (PCM512x_PAGE_0_BASE + 2) -#define PCM512x_MUTE (PCM512x_PAGE_0_BASE + 3) -#define PCM512x_PLL_EN (PCM512x_PAGE_0_BASE + 4) -#define PCM512x_SPI_MISO_FUNCTION (PCM512x_PAGE_0_BASE + 6) -#define PCM512x_DSP (PCM512x_PAGE_0_BASE + 7) -#define PCM512x_GPIO_EN (PCM512x_PAGE_0_BASE + 8) -#define PCM512x_BCLK_LRCLK_CFG (PCM512x_PAGE_0_BASE + 9) -#define PCM512x_DSP_GPIO_INPUT (PCM512x_PAGE_0_BASE + 10) -#define PCM512x_MASTER_MODE (PCM512x_PAGE_0_BASE + 12) -#define PCM512x_PLL_REF (PCM512x_PAGE_0_BASE + 13) -#define PCM512x_PLL_COEFF_0 (PCM512x_PAGE_0_BASE + 20) -#define PCM512x_PLL_COEFF_1 (PCM512x_PAGE_0_BASE + 21) -#define PCM512x_PLL_COEFF_2 (PCM512x_PAGE_0_BASE + 22) -#define PCM512x_PLL_COEFF_3 (PCM512x_PAGE_0_BASE + 23) -#define PCM512x_PLL_COEFF_4 (PCM512x_PAGE_0_BASE + 24) -#define PCM512x_DSP_CLKDIV (PCM512x_PAGE_0_BASE + 27) -#define PCM512x_DAC_CLKDIV (PCM512x_PAGE_0_BASE + 28) -#define PCM512x_NCP_CLKDIV (PCM512x_PAGE_0_BASE + 29) -#define PCM512x_OSR_CLKDIV (PCM512x_PAGE_0_BASE + 30) -#define PCM512x_MASTER_CLKDIV_1 (PCM512x_PAGE_0_BASE + 32) -#define PCM512x_MASTER_CLKDIV_2 (PCM512x_PAGE_0_BASE + 33) -#define PCM512x_FS_SPEED_MODE (PCM512x_PAGE_0_BASE + 34) -#define PCM512x_IDAC_1 (PCM512x_PAGE_0_BASE + 35) -#define PCM512x_IDAC_2 (PCM512x_PAGE_0_BASE + 36) -#define PCM512x_ERROR_DETECT (PCM512x_PAGE_0_BASE + 37) -#define PCM512x_I2S_1 (PCM512x_PAGE_0_BASE + 40) -#define PCM512x_I2S_2 (PCM512x_PAGE_0_BASE + 41) -#define PCM512x_DAC_ROUTING (PCM512x_PAGE_0_BASE + 42) -#define PCM512x_DSP_PROGRAM (PCM512x_PAGE_0_BASE + 43) -#define PCM512x_CLKDET (PCM512x_PAGE_0_BASE + 44) -#define PCM512x_AUTO_MUTE (PCM512x_PAGE_0_BASE + 59) -#define PCM512x_DIGITAL_VOLUME_1 (PCM512x_PAGE_0_BASE + 60) -#define PCM512x_DIGITAL_VOLUME_2 (PCM512x_PAGE_0_BASE + 61) -#define PCM512x_DIGITAL_VOLUME_3 (PCM512x_PAGE_0_BASE + 62) -#define PCM512x_DIGITAL_MUTE_1 (PCM512x_PAGE_0_BASE + 63) -#define PCM512x_DIGITAL_MUTE_2 (PCM512x_PAGE_0_BASE + 64) -#define PCM512x_DIGITAL_MUTE_3 (PCM512x_PAGE_0_BASE + 65) -#define PCM512x_GPIO_OUTPUT_1 (PCM512x_PAGE_0_BASE + 80) -#define PCM512x_GPIO_OUTPUT_2 (PCM512x_PAGE_0_BASE + 81) -#define PCM512x_GPIO_OUTPUT_3 (PCM512x_PAGE_0_BASE + 82) -#define PCM512x_GPIO_OUTPUT_4 (PCM512x_PAGE_0_BASE + 83) -#define PCM512x_GPIO_OUTPUT_5 (PCM512x_PAGE_0_BASE + 84) -#define PCM512x_GPIO_OUTPUT_6 (PCM512x_PAGE_0_BASE + 85) -#define PCM512x_GPIO_CONTROL_1 (PCM512x_PAGE_0_BASE + 86) -#define PCM512x_GPIO_CONTROL_2 (PCM512x_PAGE_0_BASE + 87) -#define PCM512x_OVERFLOW (PCM512x_PAGE_0_BASE + 90) -#define PCM512x_RATE_DET_1 (PCM512x_PAGE_0_BASE + 91) -#define PCM512x_RATE_DET_2 (PCM512x_PAGE_0_BASE + 92) -#define PCM512x_RATE_DET_3 (PCM512x_PAGE_0_BASE + 93) -#define PCM512x_RATE_DET_4 (PCM512x_PAGE_0_BASE + 94) -#define PCM512x_ANALOG_MUTE_DET (PCM512x_PAGE_0_BASE + 108) -#define PCM512x_GPIN (PCM512x_PAGE_0_BASE + 119) -#define PCM512x_DIGITAL_MUTE_DET (PCM512x_PAGE_0_BASE + 120) - -#define PCM512x_MAX_REGISTER (PCM512x_PAGE_0_BASE + 120) +#define PCM512x_RESET (PCM512x_PAGE_BASE(0) + 1) +#define PCM512x_POWER (PCM512x_PAGE_BASE(0) + 2) +#define PCM512x_MUTE (PCM512x_PAGE_BASE(0) + 3) +#define PCM512x_PLL_EN (PCM512x_PAGE_BASE(0) + 4) +#define PCM512x_SPI_MISO_FUNCTION (PCM512x_PAGE_BASE(0) + 6) +#define PCM512x_DSP (PCM512x_PAGE_BASE(0) + 7) +#define PCM512x_GPIO_EN (PCM512x_PAGE_BASE(0) + 8) +#define PCM512x_BCLK_LRCLK_CFG (PCM512x_PAGE_BASE(0) + 9) +#define PCM512x_DSP_GPIO_INPUT (PCM512x_PAGE_BASE(0) + 10) +#define PCM512x_MASTER_MODE (PCM512x_PAGE_BASE(0) + 12) +#define PCM512x_PLL_REF (PCM512x_PAGE_BASE(0) + 13) +#define PCM512x_PLL_COEFF_0 (PCM512x_PAGE_BASE(0) + 20) +#define PCM512x_PLL_COEFF_1 (PCM512x_PAGE_BASE(0) + 21) +#define PCM512x_PLL_COEFF_2 (PCM512x_PAGE_BASE(0) + 22) +#define PCM512x_PLL_COEFF_3 (PCM512x_PAGE_BASE(0) + 23) +#define PCM512x_PLL_COEFF_4 (PCM512x_PAGE_BASE(0) + 24) +#define PCM512x_DSP_CLKDIV (PCM512x_PAGE_BASE(0) + 27) +#define PCM512x_DAC_CLKDIV (PCM512x_PAGE_BASE(0) + 28) +#define PCM512x_NCP_CLKDIV (PCM512x_PAGE_BASE(0) + 29) +#define PCM512x_OSR_CLKDIV (PCM512x_PAGE_BASE(0) + 30) +#define PCM512x_MASTER_CLKDIV_1 (PCM512x_PAGE_BASE(0) + 32) +#define PCM512x_MASTER_CLKDIV_2 (PCM512x_PAGE_BASE(0) + 33) +#define PCM512x_FS_SPEED_MODE (PCM512x_PAGE_BASE(0) + 34) +#define PCM512x_IDAC_1 (PCM512x_PAGE_BASE(0) + 35) +#define PCM512x_IDAC_2 (PCM512x_PAGE_BASE(0) + 36) +#define PCM512x_ERROR_DETECT (PCM512x_PAGE_BASE(0) + 37) +#define PCM512x_I2S_1 (PCM512x_PAGE_BASE(0) + 40) +#define PCM512x_I2S_2 (PCM512x_PAGE_BASE(0) + 41) +#define PCM512x_DAC_ROUTING (PCM512x_PAGE_BASE(0) + 42) +#define PCM512x_DSP_PROGRAM (PCM512x_PAGE_BASE(0) + 43) +#define PCM512x_CLKDET (PCM512x_PAGE_BASE(0) + 44) +#define PCM512x_AUTO_MUTE (PCM512x_PAGE_BASE(0) + 59) +#define PCM512x_DIGITAL_VOLUME_1 (PCM512x_PAGE_BASE(0) + 60) +#define PCM512x_DIGITAL_VOLUME_2 (PCM512x_PAGE_BASE(0) + 61) +#define PCM512x_DIGITAL_VOLUME_3 (PCM512x_PAGE_BASE(0) + 62) +#define PCM512x_DIGITAL_MUTE_1 (PCM512x_PAGE_BASE(0) + 63) +#define PCM512x_DIGITAL_MUTE_2 (PCM512x_PAGE_BASE(0) + 64) +#define PCM512x_DIGITAL_MUTE_3 (PCM512x_PAGE_BASE(0) + 65) +#define PCM512x_GPIO_OUTPUT_1 (PCM512x_PAGE_BASE(0) + 80) +#define PCM512x_GPIO_OUTPUT_2 (PCM512x_PAGE_BASE(0) + 81) +#define PCM512x_GPIO_OUTPUT_3 (PCM512x_PAGE_BASE(0) + 82) +#define PCM512x_GPIO_OUTPUT_4 (PCM512x_PAGE_BASE(0) + 83) +#define PCM512x_GPIO_OUTPUT_5 (PCM512x_PAGE_BASE(0) + 84) +#define PCM512x_GPIO_OUTPUT_6 (PCM512x_PAGE_BASE(0) + 85) +#define PCM512x_GPIO_CONTROL_1 (PCM512x_PAGE_BASE(0) + 86) +#define PCM512x_GPIO_CONTROL_2 (PCM512x_PAGE_BASE(0) + 87) +#define PCM512x_OVERFLOW (PCM512x_PAGE_BASE(0) + 90) +#define PCM512x_RATE_DET_1 (PCM512x_PAGE_BASE(0) + 91) +#define PCM512x_RATE_DET_2 (PCM512x_PAGE_BASE(0) + 92) +#define PCM512x_RATE_DET_3 (PCM512x_PAGE_BASE(0) + 93) +#define PCM512x_RATE_DET_4 (PCM512x_PAGE_BASE(0) + 94) +#define PCM512x_ANALOG_MUTE_DET (PCM512x_PAGE_BASE(0) + 108) +#define PCM512x_GPIN (PCM512x_PAGE_BASE(0) + 119) +#define PCM512x_DIGITAL_MUTE_DET (PCM512x_PAGE_BASE(0) + 120) + +#define PCM512x_OUTPUT_AMPLITUDE (PCM512x_PAGE_BASE(1) + 1) +#define PCM512x_ANALOG_GAIN_CTRL (PCM512x_PAGE_BASE(1) + 2) +#define PCM512x_UNDERVOLTAGE_PROT (PCM512x_PAGE_BASE(1) + 5) +#define PCM512x_ANALOG_MUTE_CTRL (PCM512x_PAGE_BASE(1) + 6) +#define PCM512x_ANALOG_GAIN_BOOST (PCM512x_PAGE_BASE(1) + 7) +#define PCM512x_VCOM_CTRL_1 (PCM512x_PAGE_BASE(1) + 8) +#define PCM512x_VCOM_CTRL_2 (PCM512x_PAGE_BASE(1) + 9) + +#define PCM512x_CRAM_CTRL (PCM512x_PAGE_BASE(44) + 1) + +#define PCM512x_MAX_REGISTER (PCM512x_PAGE_BASE(44) + 1) /* Page 0, Register 1 - reset */ #define PCM512x_RSTR (1 << 0) -- cgit v1.2.3 From 5be2fc20b101b5138c4f54a584dc11790ef16598 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 7 Feb 2014 19:16:56 +0000 Subject: ASoC: pcm512x: Implement analogue volume control There are some analogue volume controls in page 1 of the register map so implement support for them now that we can access the registers. Signed-off-by: Mark Brown --- sound/soc/codecs/pcm512x.c | 6 ++++++ sound/soc/codecs/pcm512x.h | 8 ++++++++ 2 files changed, 14 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index cdcb51e4c86f..3a0bbb6ab242 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -184,6 +184,8 @@ static bool pcm512x_volatile(struct device *dev, unsigned int reg) } static const DECLARE_TLV_DB_SCALE(digital_tlv, -10350, 50, 1); +static const DECLARE_TLV_DB_SCALE(analog_tlv, -600, 600, 0); +static const DECLARE_TLV_DB_SCALE(boost_tlv, 0, 80, 0); static const char * const pcm512x_dsp_program_texts[] = { "FIR interpolation with de-emphasis", @@ -261,6 +263,10 @@ static const struct soc_enum pcm512x_veds = static const struct snd_kcontrol_new pcm512x_controls[] = { SOC_DOUBLE_R_TLV("Playback Digital Volume", PCM512x_DIGITAL_VOLUME_2, PCM512x_DIGITAL_VOLUME_3, 0, 255, 1, digital_tlv), +SOC_DOUBLE_TLV("Playback Volume", PCM512x_ANALOG_GAIN_CTRL, + PCM512x_LAGN_SHIFT, PCM512x_RAGN_SHIFT, 1, 1, analog_tlv), +SOC_DOUBLE_TLV("Playback Boost Volume", PCM512x_ANALOG_GAIN_BOOST, + PCM512x_AGBL_SHIFT, PCM512x_AGBR_SHIFT, 1, 0, boost_tlv), SOC_DOUBLE("Playback Digital Switch", PCM512x_MUTE, PCM512x_RQML_SHIFT, PCM512x_RQMR_SHIFT, 1, 1), diff --git a/sound/soc/codecs/pcm512x.h b/sound/soc/codecs/pcm512x.h index e58743c965d6..ac4a52c9ccf4 100644 --- a/sound/soc/codecs/pcm512x.h +++ b/sound/soc/codecs/pcm512x.h @@ -151,4 +151,12 @@ #define PCM512x_AMLE_SHIFT 1 #define PCM512x_AMLR_SHIFT 0 +/* Page 1, Register 2 - analog volume control */ +#define PCM512x_RAGN_SHIFT 0 +#define PCM512x_LAGN_SHIFT 4 + +/* Page 1, Register 7 - analog boost control */ +#define PCM512x_AGBR_SHIFT 0 +#define PCM512x_AGBL_SHIFT 4 + #endif -- cgit v1.2.3 From e156291c7bab1771df34acd583c3c1b1ad71231c Mon Sep 17 00:00:00 2001 From: Christian Engelmayer Date: Fri, 7 Feb 2014 19:43:19 +0100 Subject: ASoC: wm8991: Remove unused pointer in wm8991_probe() Remove unused pointer 'wm8991' in function wm8991_probe(). The last user vanished with a86652e5 (ASoC: wm8991: Convert to direct regmap API usage) Detected by Coverity: CID 1162831 Signed-off-by: Christian Engelmayer Signed-off-by: Mark Brown --- sound/soc/codecs/wm8991.c | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index dba0306c42a5..244eb09ffa43 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -1251,11 +1251,8 @@ static int wm8991_remove(struct snd_soc_codec *codec) static int wm8991_probe(struct snd_soc_codec *codec) { - struct wm8991_priv *wm8991; int ret; - wm8991 = snd_soc_codec_get_drvdata(codec); - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); -- cgit v1.2.3 From 9f10b36ffde2b732def037c1e764a0c71745a372 Mon Sep 17 00:00:00 2001 From: Adam Thomson Date: Thu, 6 Feb 2014 18:03:09 +0000 Subject: ASoC: da9055: Add DT support for CODEC Signed-off-by: Adam Thomson Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/da9055.txt | 22 ++++++++++++++++++++++ sound/soc/codecs/da9055.c | 8 ++++++++ 2 files changed, 30 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/da9055.txt (limited to 'sound/soc/codecs') diff --git a/Documentation/devicetree/bindings/sound/da9055.txt b/Documentation/devicetree/bindings/sound/da9055.txt new file mode 100644 index 000000000000..ed1b7cc6f249 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/da9055.txt @@ -0,0 +1,22 @@ +* Dialog DA9055 Audio CODEC + +DA9055 provides Audio CODEC support (I2C only). + +The Audio CODEC device in DA9055 has it's own I2C address which is configurable, +so the device is instantiated separately from the PMIC (MFD) device. + +For details on accompanying PMIC I2C device, see the following: +Documentation/devicetree/bindings/mfd/da9055.txt + +Required properties: + + - compatible: "dlg,da9055-codec" + - reg: Specifies the I2C slave address + + +Example: + + codec: da9055-codec@1a { + compatible = "dlg,da9055-codec"; + reg = <0x1a>; + }; diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index 422812613a28..be31f3cfd46e 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -18,6 +18,8 @@ #include #include #include +#include +#include #include #include #include @@ -1536,11 +1538,17 @@ static const struct i2c_device_id da9055_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, da9055_i2c_id); +static const struct of_device_id da9055_of_match[] = { + { .compatible = "dlg,da9055-codec", }, + { } +}; + /* I2C codec control layer */ static struct i2c_driver da9055_i2c_driver = { .driver = { .name = "da9055-codec", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(da9055_of_match), }, .probe = da9055_i2c_probe, .remove = da9055_remove, -- cgit v1.2.3 From 8955f28dba9edb04305504a06d7f395ca3b32904 Mon Sep 17 00:00:00 2001 From: Christian Engelmayer Date: Fri, 7 Feb 2014 23:37:16 +0100 Subject: ASoC: wm8995: Remove unused pointer in hp_supply_event() Remove unused driver data pointer 'wm8995' in function hp_supply_event(). Detected by Coverity: CID 141181. Signed-off-by: Christian Engelmayer Signed-off-by: Mark Brown --- sound/soc/codecs/wm8995.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 4300caff1783..403e1db6870f 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -561,10 +561,8 @@ static int hp_supply_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { struct snd_soc_codec *codec; - struct wm8995_priv *wm8995; codec = w->codec; - wm8995 = snd_soc_codec_get_drvdata(codec); switch (event) { case SND_SOC_DAPM_PRE_PMU: -- cgit v1.2.3 From dfd72a68aa0f6cf87575f3181319bde8a2d4c01b Mon Sep 17 00:00:00 2001 From: Thomas Petazzoni Date: Thu, 30 Jan 2014 18:14:05 +0100 Subject: ASoC: cs42l51: add Device Tree binding to cs42l51 This commit adds a trivial Device Tree binding to the I2C-based cs42l51 sound codec, so that it can be used from Device Tree based platforms. Signed-off-by: Thomas Petazzoni Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/i2c/trivial-devices.txt | 1 + sound/soc/codecs/cs42l51.c | 7 +++++++ 2 files changed, 8 insertions(+) (limited to 'sound/soc/codecs') diff --git a/Documentation/devicetree/bindings/i2c/trivial-devices.txt b/Documentation/devicetree/bindings/i2c/trivial-devices.txt index 1a1ac2e560e9..f47e56bcf78d 100644 --- a/Documentation/devicetree/bindings/i2c/trivial-devices.txt +++ b/Documentation/devicetree/bindings/i2c/trivial-devices.txt @@ -18,6 +18,7 @@ atmel,24c02 i2c serial eeprom (24cxx) atmel,at97sc3204t i2c trusted platform module (TPM) capella,cm32181 CM32181: Ambient Light Sensor catalyst,24c32 i2c serial eeprom +cirrus,cs42l51 Cirrus Logic CS42L51 audio codec dallas,ds1307 64 x 8, Serial, I2C Real-Time Clock dallas,ds1338 I2C RTC with 56-Byte NV RAM dallas,ds1339 I2C Serial Real-Time Clock diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 6e9ea8379a91..824cdf4d4974 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -600,10 +600,17 @@ static const struct i2c_device_id cs42l51_id[] = { }; MODULE_DEVICE_TABLE(i2c, cs42l51_id); +static const struct of_device_id cs42l51_of_match[] = { + { .compatible = "cirrus,cs42l51", }, + { } +}; +MODULE_DEVICE_TABLE(of, cs42l51_of_match); + static struct i2c_driver cs42l51_i2c_driver = { .driver = { .name = "cs42l51-codec", .owner = THIS_MODULE, + .of_match_table = cs42l51_of_match, }, .id_table = cs42l51_id, .probe = cs42l51_i2c_probe, -- cgit v1.2.3 From 5be736442ed94217c6521ae0c948abab995f281f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 11 Feb 2014 19:21:16 +0000 Subject: ASoC: cs42l51: Don't log if we fail to allocate memory The VM subsystem already logs quite loudly if we run out of memory so don't bother here. Signed-off-by: Mark Brown Acked-by: Brian Austin --- sound/soc/codecs/cs42l51.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 824cdf4d4974..b11079b38f15 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -574,10 +574,8 @@ static int cs42l51_i2c_probe(struct i2c_client *i2c_client, cs42l51 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l51_private), GFP_KERNEL); - if (!cs42l51) { - dev_err(&i2c_client->dev, "could not allocate codec\n"); + if (!cs42l51) return -ENOMEM; - } i2c_set_clientdata(i2c_client, cs42l51); cs42l51->control_type = SND_SOC_I2C; -- cgit v1.2.3 From da071489762499a3635cb3563d32792cea20c087 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 11 Feb 2014 19:24:31 +0000 Subject: ASoC: cs42l51: Convert to direct regmap API usage As part of phasing out the ASoC level register I/O code (which is now just a thin wrapper around regmap anyway) convert the cs42l51 driver to use the regmap API directly. We now no longer initialise the cache from hardware at startup, the regmap caches are smart enough to understand which registers are actually cached and read on demand. This should have no visible effect on the system. Signed-off-by: Mark Brown Acked-by: Brian Austin --- sound/soc/codecs/cs42l51.c | 59 ++++++++++++++++++++-------------------------- 1 file changed, 25 insertions(+), 34 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index b11079b38f15..e53c8714591f 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -30,6 +30,7 @@ #include #include #include +#include #include "cs42l51.h" @@ -40,7 +41,6 @@ enum master_slave_mode { }; struct cs42l51_private { - enum snd_soc_control_type control_type; unsigned int mclk; unsigned int audio_mode; /* The mode (I2S or left-justified) */ enum master_slave_mode func; @@ -52,24 +52,6 @@ struct cs42l51_private { SNDRV_PCM_FMTBIT_S20_3LE | SNDRV_PCM_FMTBIT_S20_3BE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE) -static int cs42l51_fill_cache(struct snd_soc_codec *codec) -{ - u8 *cache = codec->reg_cache + 1; - struct i2c_client *i2c_client = to_i2c_client(codec->dev); - s32 length; - - length = i2c_smbus_read_i2c_block_data(i2c_client, - CS42L51_FIRSTREG | 0x80, CS42L51_NUMREGS, cache); - if (length != CS42L51_NUMREGS) { - dev_err(&i2c_client->dev, - "I2C read failure, addr=0x%x (ret=%d vs %d)\n", - i2c_client->addr, length, CS42L51_NUMREGS); - return -EIO; - } - - return 0; -} - static int cs42l51_get_chan_mix(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { @@ -508,13 +490,7 @@ static int cs42l51_probe(struct snd_soc_codec *codec) struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec); int ret, reg; - ret = cs42l51_fill_cache(codec); - if (ret < 0) { - dev_err(codec->dev, "failed to fill register cache\n"); - return ret; - } - - ret = snd_soc_codec_set_cache_io(codec, 8, 8, cs42l51->control_type); + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -538,8 +514,6 @@ static int cs42l51_probe(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_device_cs42l51 = { .probe = cs42l51_probe, - .reg_cache_size = CS42L51_NUMREGS + 1, - .reg_word_size = sizeof(u8), .controls = cs42l51_snd_controls, .num_controls = ARRAY_SIZE(cs42l51_snd_controls), @@ -549,28 +523,46 @@ static struct snd_soc_codec_driver soc_codec_device_cs42l51 = { .num_dapm_routes = ARRAY_SIZE(cs42l51_routes), }; +static const struct regmap_config cs42l51_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS42L51_CHARGE_FREQ, + .cache_type = REGCACHE_RBTREE, +}; + static int cs42l51_i2c_probe(struct i2c_client *i2c_client, const struct i2c_device_id *id) { struct cs42l51_private *cs42l51; + struct regmap *regmap; + unsigned int val; int ret; + regmap = devm_regmap_init_i2c(i2c_client, &cs42l51_regmap); + if (IS_ERR(regmap)) { + ret = PTR_ERR(regmap); + dev_err(&i2c_client->dev, "Failed to create regmap: %d\n", + ret); + return ret; + } + /* Verify that we have a CS42L51 */ - ret = i2c_smbus_read_byte_data(i2c_client, CS42L51_CHIP_REV_ID); + ret = regmap_read(regmap, CS42L51_CHIP_REV_ID, &val); if (ret < 0) { dev_err(&i2c_client->dev, "failed to read I2C\n"); goto error; } - if ((ret != CS42L51_MK_CHIP_REV(CS42L51_CHIP_ID, CS42L51_CHIP_REV_A)) && - (ret != CS42L51_MK_CHIP_REV(CS42L51_CHIP_ID, CS42L51_CHIP_REV_B))) { - dev_err(&i2c_client->dev, "Invalid chip id\n"); + if ((val != CS42L51_MK_CHIP_REV(CS42L51_CHIP_ID, CS42L51_CHIP_REV_A)) && + (val != CS42L51_MK_CHIP_REV(CS42L51_CHIP_ID, CS42L51_CHIP_REV_B))) { + dev_err(&i2c_client->dev, "Invalid chip id: %x\n", val); ret = -ENODEV; goto error; } dev_info(&i2c_client->dev, "found device cs42l51 rev %d\n", - ret & 7); + val & 7); cs42l51 = devm_kzalloc(&i2c_client->dev, sizeof(struct cs42l51_private), GFP_KERNEL); @@ -578,7 +570,6 @@ static int cs42l51_i2c_probe(struct i2c_client *i2c_client, return -ENOMEM; i2c_set_clientdata(i2c_client, cs42l51); - cs42l51->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&i2c_client->dev, &soc_codec_device_cs42l51, &cs42l51_dai, 1); -- cgit v1.2.3 From 78c51bc6475982a843947a556853affd2c360b19 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 11 Feb 2014 19:51:58 +0000 Subject: ASoC: ak4671: Convert to table based control init Saves code and adds error handling. Signed-off-by: Mark Brown --- sound/soc/codecs/ak4671.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 25bdf6ad4a54..456bd0a065b1 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -628,9 +628,6 @@ static int ak4671_probe(struct snd_soc_codec *codec) return ret; } - snd_soc_add_codec_controls(codec, ak4671_snd_controls, - ARRAY_SIZE(ak4671_snd_controls)); - ak4671_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return ret; @@ -646,6 +643,8 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4671 = { .probe = ak4671_probe, .remove = ak4671_remove, .set_bias_level = ak4671_set_bias_level, + .controls = ak4671_snd_controls, + .num_controls = ARRAY_SIZE(ak4671_snd_controls), .reg_cache_size = AK4671_CACHEREGNUM, .reg_word_size = sizeof(u8), .reg_cache_default = ak4671_reg, -- cgit v1.2.3 From 7ac5a47886ae4b211f08657d0d06027285eda2d0 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 11 Feb 2014 19:53:25 +0000 Subject: ASoC: ak4671: Convert to direct regmap API usage This helps us remove the ASoC level I/O functionality which is now just a thin wrapper around regmap. Signed-off-by: Mark Brown --- sound/soc/codecs/ak4671.c | 221 +++++++++++++++++++++++----------------------- sound/soc/codecs/ak4671.h | 2 - 2 files changed, 111 insertions(+), 112 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 456bd0a065b1..743bbe31bc08 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -15,6 +15,7 @@ #include #include #include +#include #include #include #include @@ -23,104 +24,99 @@ #include "ak4671.h" -/* codec private data */ -struct ak4671_priv { - enum snd_soc_control_type control_type; -}; - /* ak4671 register cache & default register settings */ -static const u8 ak4671_reg[AK4671_CACHEREGNUM] = { - 0x00, /* AK4671_AD_DA_POWER_MANAGEMENT (0x00) */ - 0xf6, /* AK4671_PLL_MODE_SELECT0 (0x01) */ - 0x00, /* AK4671_PLL_MODE_SELECT1 (0x02) */ - 0x02, /* AK4671_FORMAT_SELECT (0x03) */ - 0x00, /* AK4671_MIC_SIGNAL_SELECT (0x04) */ - 0x55, /* AK4671_MIC_AMP_GAIN (0x05) */ - 0x00, /* AK4671_MIXING_POWER_MANAGEMENT0 (0x06) */ - 0x00, /* AK4671_MIXING_POWER_MANAGEMENT1 (0x07) */ - 0xb5, /* AK4671_OUTPUT_VOLUME_CONTROL (0x08) */ - 0x00, /* AK4671_LOUT1_SIGNAL_SELECT (0x09) */ - 0x00, /* AK4671_ROUT1_SIGNAL_SELECT (0x0a) */ - 0x00, /* AK4671_LOUT2_SIGNAL_SELECT (0x0b) */ - 0x00, /* AK4671_ROUT2_SIGNAL_SELECT (0x0c) */ - 0x00, /* AK4671_LOUT3_SIGNAL_SELECT (0x0d) */ - 0x00, /* AK4671_ROUT3_SIGNAL_SELECT (0x0e) */ - 0x00, /* AK4671_LOUT1_POWER_MANAGERMENT (0x0f) */ - 0x00, /* AK4671_LOUT2_POWER_MANAGERMENT (0x10) */ - 0x80, /* AK4671_LOUT3_POWER_MANAGERMENT (0x11) */ - 0x91, /* AK4671_LCH_INPUT_VOLUME_CONTROL (0x12) */ - 0x91, /* AK4671_RCH_INPUT_VOLUME_CONTROL (0x13) */ - 0xe1, /* AK4671_ALC_REFERENCE_SELECT (0x14) */ - 0x00, /* AK4671_DIGITAL_MIXING_CONTROL (0x15) */ - 0x00, /* AK4671_ALC_TIMER_SELECT (0x16) */ - 0x00, /* AK4671_ALC_MODE_CONTROL (0x17) */ - 0x02, /* AK4671_MODE_CONTROL1 (0x18) */ - 0x01, /* AK4671_MODE_CONTROL2 (0x19) */ - 0x18, /* AK4671_LCH_OUTPUT_VOLUME_CONTROL (0x1a) */ - 0x18, /* AK4671_RCH_OUTPUT_VOLUME_CONTROL (0x1b) */ - 0x00, /* AK4671_SIDETONE_A_CONTROL (0x1c) */ - 0x02, /* AK4671_DIGITAL_FILTER_SELECT (0x1d) */ - 0x00, /* AK4671_FIL3_COEFFICIENT0 (0x1e) */ - 0x00, /* AK4671_FIL3_COEFFICIENT1 (0x1f) */ - 0x00, /* AK4671_FIL3_COEFFICIENT2 (0x20) */ - 0x00, /* AK4671_FIL3_COEFFICIENT3 (0x21) */ - 0x00, /* AK4671_EQ_COEFFICIENT0 (0x22) */ - 0x00, /* AK4671_EQ_COEFFICIENT1 (0x23) */ - 0x00, /* AK4671_EQ_COEFFICIENT2 (0x24) */ - 0x00, /* AK4671_EQ_COEFFICIENT3 (0x25) */ - 0x00, /* AK4671_EQ_COEFFICIENT4 (0x26) */ - 0x00, /* AK4671_EQ_COEFFICIENT5 (0x27) */ - 0xa9, /* AK4671_FIL1_COEFFICIENT0 (0x28) */ - 0x1f, /* AK4671_FIL1_COEFFICIENT1 (0x29) */ - 0xad, /* AK4671_FIL1_COEFFICIENT2 (0x2a) */ - 0x20, /* AK4671_FIL1_COEFFICIENT3 (0x2b) */ - 0x00, /* AK4671_FIL2_COEFFICIENT0 (0x2c) */ - 0x00, /* AK4671_FIL2_COEFFICIENT1 (0x2d) */ - 0x00, /* AK4671_FIL2_COEFFICIENT2 (0x2e) */ - 0x00, /* AK4671_FIL2_COEFFICIENT3 (0x2f) */ - 0x00, /* AK4671_DIGITAL_FILTER_SELECT2 (0x30) */ - 0x00, /* this register not used */ - 0x00, /* AK4671_E1_COEFFICIENT0 (0x32) */ - 0x00, /* AK4671_E1_COEFFICIENT1 (0x33) */ - 0x00, /* AK4671_E1_COEFFICIENT2 (0x34) */ - 0x00, /* AK4671_E1_COEFFICIENT3 (0x35) */ - 0x00, /* AK4671_E1_COEFFICIENT4 (0x36) */ - 0x00, /* AK4671_E1_COEFFICIENT5 (0x37) */ - 0x00, /* AK4671_E2_COEFFICIENT0 (0x38) */ - 0x00, /* AK4671_E2_COEFFICIENT1 (0x39) */ - 0x00, /* AK4671_E2_COEFFICIENT2 (0x3a) */ - 0x00, /* AK4671_E2_COEFFICIENT3 (0x3b) */ - 0x00, /* AK4671_E2_COEFFICIENT4 (0x3c) */ - 0x00, /* AK4671_E2_COEFFICIENT5 (0x3d) */ - 0x00, /* AK4671_E3_COEFFICIENT0 (0x3e) */ - 0x00, /* AK4671_E3_COEFFICIENT1 (0x3f) */ - 0x00, /* AK4671_E3_COEFFICIENT2 (0x40) */ - 0x00, /* AK4671_E3_COEFFICIENT3 (0x41) */ - 0x00, /* AK4671_E3_COEFFICIENT4 (0x42) */ - 0x00, /* AK4671_E3_COEFFICIENT5 (0x43) */ - 0x00, /* AK4671_E4_COEFFICIENT0 (0x44) */ - 0x00, /* AK4671_E4_COEFFICIENT1 (0x45) */ - 0x00, /* AK4671_E4_COEFFICIENT2 (0x46) */ - 0x00, /* AK4671_E4_COEFFICIENT3 (0x47) */ - 0x00, /* AK4671_E4_COEFFICIENT4 (0x48) */ - 0x00, /* AK4671_E4_COEFFICIENT5 (0x49) */ - 0x00, /* AK4671_E5_COEFFICIENT0 (0x4a) */ - 0x00, /* AK4671_E5_COEFFICIENT1 (0x4b) */ - 0x00, /* AK4671_E5_COEFFICIENT2 (0x4c) */ - 0x00, /* AK4671_E5_COEFFICIENT3 (0x4d) */ - 0x00, /* AK4671_E5_COEFFICIENT4 (0x4e) */ - 0x00, /* AK4671_E5_COEFFICIENT5 (0x4f) */ - 0x88, /* AK4671_EQ_CONTROL_250HZ_100HZ (0x50) */ - 0x88, /* AK4671_EQ_CONTROL_3500HZ_1KHZ (0x51) */ - 0x08, /* AK4671_EQ_CONTRO_10KHZ (0x52) */ - 0x00, /* AK4671_PCM_IF_CONTROL0 (0x53) */ - 0x00, /* AK4671_PCM_IF_CONTROL1 (0x54) */ - 0x00, /* AK4671_PCM_IF_CONTROL2 (0x55) */ - 0x18, /* AK4671_DIGITAL_VOLUME_B_CONTROL (0x56) */ - 0x18, /* AK4671_DIGITAL_VOLUME_C_CONTROL (0x57) */ - 0x00, /* AK4671_SIDETONE_VOLUME_CONTROL (0x58) */ - 0x00, /* AK4671_DIGITAL_MIXING_CONTROL2 (0x59) */ - 0x00, /* AK4671_SAR_ADC_CONTROL (0x5a) */ +static const struct reg_default ak4671_reg_defaults[] = { + { 0x00, 0x00 }, /* AK4671_AD_DA_POWER_MANAGEMENT (0x00) */ + { 0x01, 0xf6 }, /* AK4671_PLL_MODE_SELECT0 (0x01) */ + { 0x02, 0x00 }, /* AK4671_PLL_MODE_SELECT1 (0x02) */ + { 0x03, 0x02 }, /* AK4671_FORMAT_SELECT (0x03) */ + { 0x04, 0x00 }, /* AK4671_MIC_SIGNAL_SELECT (0x04) */ + { 0x05, 0x55 }, /* AK4671_MIC_AMP_GAIN (0x05) */ + { 0x06, 0x00 }, /* AK4671_MIXING_POWER_MANAGEMENT0 (0x06) */ + { 0x07, 0x00 }, /* AK4671_MIXING_POWER_MANAGEMENT1 (0x07) */ + { 0x08, 0xb5 }, /* AK4671_OUTPUT_VOLUME_CONTROL (0x08) */ + { 0x09, 0x00 }, /* AK4671_LOUT1_SIGNAL_SELECT (0x09) */ + { 0x0a, 0x00 }, /* AK4671_ROUT1_SIGNAL_SELECT (0x0a) */ + { 0x0b, 0x00 }, /* AK4671_LOUT2_SIGNAL_SELECT (0x0b) */ + { 0x0c, 0x00 }, /* AK4671_ROUT2_SIGNAL_SELECT (0x0c) */ + { 0x0d, 0x00 }, /* AK4671_LOUT3_SIGNAL_SELECT (0x0d) */ + { 0x0e, 0x00 }, /* AK4671_ROUT3_SIGNAL_SELECT (0x0e) */ + { 0x0f, 0x00 }, /* AK4671_LOUT1_POWER_MANAGERMENT (0x0f) */ + { 0x10, 0x00 }, /* AK4671_LOUT2_POWER_MANAGERMENT (0x10) */ + { 0x11, 0x80 }, /* AK4671_LOUT3_POWER_MANAGERMENT (0x11) */ + { 0x12, 0x91 }, /* AK4671_LCH_INPUT_VOLUME_CONTROL (0x12) */ + { 0x13, 0x91 }, /* AK4671_RCH_INPUT_VOLUME_CONTROL (0x13) */ + { 0x14, 0xe1 }, /* AK4671_ALC_REFERENCE_SELECT (0x14) */ + { 0x15, 0x00 }, /* AK4671_DIGITAL_MIXING_CONTROL (0x15) */ + { 0x16, 0x00 }, /* AK4671_ALC_TIMER_SELECT (0x16) */ + { 0x17, 0x00 }, /* AK4671_ALC_MODE_CONTROL (0x17) */ + { 0x18, 0x02 }, /* AK4671_MODE_CONTROL1 (0x18) */ + { 0x19, 0x01 }, /* AK4671_MODE_CONTROL2 (0x19) */ + { 0x1a, 0x18 }, /* AK4671_LCH_OUTPUT_VOLUME_CONTROL (0x1a) */ + { 0x1b, 0x18 }, /* AK4671_RCH_OUTPUT_VOLUME_CONTROL (0x1b) */ + { 0x1c, 0x00 }, /* AK4671_SIDETONE_A_CONTROL (0x1c) */ + { 0x1d, 0x02 }, /* AK4671_DIGITAL_FILTER_SELECT (0x1d) */ + { 0x1e, 0x00 }, /* AK4671_FIL3_COEFFICIENT0 (0x1e) */ + { 0x1f, 0x00 }, /* AK4671_FIL3_COEFFICIENT1 (0x1f) */ + { 0x20, 0x00 }, /* AK4671_FIL3_COEFFICIENT2 (0x20) */ + { 0x21, 0x00 }, /* AK4671_FIL3_COEFFICIENT3 (0x21) */ + { 0x22, 0x00 }, /* AK4671_EQ_COEFFICIENT0 (0x22) */ + { 0x23, 0x00 }, /* AK4671_EQ_COEFFICIENT1 (0x23) */ + { 0x24, 0x00 }, /* AK4671_EQ_COEFFICIENT2 (0x24) */ + { 0x25, 0x00 }, /* AK4671_EQ_COEFFICIENT3 (0x25) */ + { 0x26, 0x00 }, /* AK4671_EQ_COEFFICIENT4 (0x26) */ + { 0x27, 0x00 }, /* AK4671_EQ_COEFFICIENT5 (0x27) */ + { 0x28, 0xa9 }, /* AK4671_FIL1_COEFFICIENT0 (0x28) */ + { 0x29, 0x1f }, /* AK4671_FIL1_COEFFICIENT1 (0x29) */ + { 0x2a, 0xad }, /* AK4671_FIL1_COEFFICIENT2 (0x2a) */ + { 0x2b, 0x20 }, /* AK4671_FIL1_COEFFICIENT3 (0x2b) */ + { 0x2c, 0x00 }, /* AK4671_FIL2_COEFFICIENT0 (0x2c) */ + { 0x2d, 0x00 }, /* AK4671_FIL2_COEFFICIENT1 (0x2d) */ + { 0x2e, 0x00 }, /* AK4671_FIL2_COEFFICIENT2 (0x2e) */ + { 0x2f, 0x00 }, /* AK4671_FIL2_COEFFICIENT3 (0x2f) */ + { 0x30, 0x00 }, /* AK4671_DIGITAL_FILTER_SELECT2 (0x30) */ + + { 0x32, 0x00 }, /* AK4671_E1_COEFFICIENT0 (0x32) */ + { 0x33, 0x00 }, /* AK4671_E1_COEFFICIENT1 (0x33) */ + { 0x34, 0x00 }, /* AK4671_E1_COEFFICIENT2 (0x34) */ + { 0x35, 0x00 }, /* AK4671_E1_COEFFICIENT3 (0x35) */ + { 0x36, 0x00 }, /* AK4671_E1_COEFFICIENT4 (0x36) */ + { 0x37, 0x00 }, /* AK4671_E1_COEFFICIENT5 (0x37) */ + { 0x38, 0x00 }, /* AK4671_E2_COEFFICIENT0 (0x38) */ + { 0x39, 0x00 }, /* AK4671_E2_COEFFICIENT1 (0x39) */ + { 0x3a, 0x00 }, /* AK4671_E2_COEFFICIENT2 (0x3a) */ + { 0x3b, 0x00 }, /* AK4671_E2_COEFFICIENT3 (0x3b) */ + { 0x3c, 0x00 }, /* AK4671_E2_COEFFICIENT4 (0x3c) */ + { 0x3d, 0x00 }, /* AK4671_E2_COEFFICIENT5 (0x3d) */ + { 0x3e, 0x00 }, /* AK4671_E3_COEFFICIENT0 (0x3e) */ + { 0x3f, 0x00 }, /* AK4671_E3_COEFFICIENT1 (0x3f) */ + { 0x40, 0x00 }, /* AK4671_E3_COEFFICIENT2 (0x40) */ + { 0x41, 0x00 }, /* AK4671_E3_COEFFICIENT3 (0x41) */ + { 0x42, 0x00 }, /* AK4671_E3_COEFFICIENT4 (0x42) */ + { 0x43, 0x00 }, /* AK4671_E3_COEFFICIENT5 (0x43) */ + { 0x44, 0x00 }, /* AK4671_E4_COEFFICIENT0 (0x44) */ + { 0x45, 0x00 }, /* AK4671_E4_COEFFICIENT1 (0x45) */ + { 0x46, 0x00 }, /* AK4671_E4_COEFFICIENT2 (0x46) */ + { 0x47, 0x00 }, /* AK4671_E4_COEFFICIENT3 (0x47) */ + { 0x48, 0x00 }, /* AK4671_E4_COEFFICIENT4 (0x48) */ + { 0x49, 0x00 }, /* AK4671_E4_COEFFICIENT5 (0x49) */ + { 0x4a, 0x00 }, /* AK4671_E5_COEFFICIENT0 (0x4a) */ + { 0x4b, 0x00 }, /* AK4671_E5_COEFFICIENT1 (0x4b) */ + { 0x4c, 0x00 }, /* AK4671_E5_COEFFICIENT2 (0x4c) */ + { 0x4d, 0x00 }, /* AK4671_E5_COEFFICIENT3 (0x4d) */ + { 0x4e, 0x00 }, /* AK4671_E5_COEFFICIENT4 (0x4e) */ + { 0x4f, 0x00 }, /* AK4671_E5_COEFFICIENT5 (0x4f) */ + { 0x50, 0x88 }, /* AK4671_EQ_CONTROL_250HZ_100HZ (0x50) */ + { 0x51, 0x88 }, /* AK4671_EQ_CONTROL_3500HZ_1KHZ (0x51) */ + { 0x52, 0x08 }, /* AK4671_EQ_CONTRO_10KHZ (0x52) */ + { 0x53, 0x00 }, /* AK4671_PCM_IF_CONTROL0 (0x53) */ + { 0x54, 0x00 }, /* AK4671_PCM_IF_CONTROL1 (0x54) */ + { 0x55, 0x00 }, /* AK4671_PCM_IF_CONTROL2 (0x55) */ + { 0x56, 0x18 }, /* AK4671_DIGITAL_VOLUME_B_CONTROL (0x56) */ + { 0x57, 0x18 }, /* AK4671_DIGITAL_VOLUME_C_CONTROL (0x57) */ + { 0x58, 0x00 }, /* AK4671_SIDETONE_VOLUME_CONTROL (0x58) */ + { 0x59, 0x00 }, /* AK4671_DIGITAL_MIXING_CONTROL2 (0x59) */ + { 0x5a, 0x00 }, /* AK4671_SAR_ADC_CONTROL (0x5a) */ }; /* @@ -619,10 +615,9 @@ static struct snd_soc_dai_driver ak4671_dai = { static int ak4671_probe(struct snd_soc_codec *codec) { - struct ak4671_priv *ak4671 = snd_soc_codec_get_drvdata(codec); int ret; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, ak4671->control_type); + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; @@ -645,28 +640,34 @@ static struct snd_soc_codec_driver soc_codec_dev_ak4671 = { .set_bias_level = ak4671_set_bias_level, .controls = ak4671_snd_controls, .num_controls = ARRAY_SIZE(ak4671_snd_controls), - .reg_cache_size = AK4671_CACHEREGNUM, - .reg_word_size = sizeof(u8), - .reg_cache_default = ak4671_reg, .dapm_widgets = ak4671_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(ak4671_dapm_widgets), .dapm_routes = ak4671_intercon, .num_dapm_routes = ARRAY_SIZE(ak4671_intercon), }; +static const struct regmap_config ak4671_regmap = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = AK4671_SAR_ADC_CONTROL, + .reg_defaults = ak4671_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(ak4671_reg_defaults), + .cache_type = REGCACHE_RBTREE, +}; + static int ak4671_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { - struct ak4671_priv *ak4671; + struct regmap *regmap; int ret; - ak4671 = devm_kzalloc(&client->dev, sizeof(struct ak4671_priv), - GFP_KERNEL); - if (ak4671 == NULL) - return -ENOMEM; - - i2c_set_clientdata(client, ak4671); - ak4671->control_type = SND_SOC_I2C; + regmap = devm_regmap_init_i2c(client, &ak4671_regmap); + if (IS_ERR(regmap)) { + ret = PTR_ERR(regmap); + dev_err(&client->dev, "Failed to create regmap: %d\n", ret); + return ret; + } ret = snd_soc_register_codec(&client->dev, &soc_codec_dev_ak4671, &ak4671_dai, 1); diff --git a/sound/soc/codecs/ak4671.h b/sound/soc/codecs/ak4671.h index 61cb7ab7552c..394a34d3f50a 100644 --- a/sound/soc/codecs/ak4671.h +++ b/sound/soc/codecs/ak4671.h @@ -105,8 +105,6 @@ #define AK4671_DIGITAL_MIXING_CONTROL2 0x59 #define AK4671_SAR_ADC_CONTROL 0x5a -#define AK4671_CACHEREGNUM (AK4671_SAR_ADC_CONTROL + 1) - /* Bitfield Definitions */ /* AK4671_AD_DA_POWER_MANAGEMENT (0x00) Fields */ -- cgit v1.2.3 From f951f835a9719996e08e5c2932afc65ebfdbf47a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 16 Feb 2014 08:29:21 +0800 Subject: ASoC: pcm512x: Add regmap select We need at least the core regmap code to build. Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 1 + 1 file changed, 1 insertion(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 56d0c2845680..fa47a8336be3 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -316,6 +316,7 @@ config SND_SOC_PCM3008 config SND_SOC_PCM512x tristate "Texas Instruments PCM512x CODECs" + select REGMAP config SND_SOC_RT5631 tristate -- cgit v1.2.3 From 6c3d713e6d32706999689e379a9098afb4cd8a2c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 17 Feb 2014 13:16:54 +0100 Subject: ASoC: ad193x: Split SPI and I2C code into different modules There are a few known (minor) problems with having the support code for both I2C and SPI in the same module: * We need to be extra careful to make sure to not build the driver into the kernel if one of the subsystems is build as a module (Currently only I2C can be build as a module). * The module init path error handling is rather ugly. E.g. what should be done if either the SPI or the I2C driver fails to register? Most drivers that implement SPI and I2C in the same module currently fallback to undefined behavior in that case. Splitting the the driver into two modules, one for each bus, allows the registration of the other bus driver to continue without problems if one of them fails. This patch splits the AD193X driver into 3 modules. One core module that implements the device logic, but is independent of the bus method used. And one module for SPI and I2C each that registers the drivers and sets up the regmap struct for the bus. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/blackfin/Kconfig | 3 +- sound/soc/codecs/Kconfig | 11 +++- sound/soc/codecs/Makefile | 4 ++ sound/soc/codecs/ad193x-i2c.c | 54 ++++++++++++++++ sound/soc/codecs/ad193x-spi.c | 48 +++++++++++++++ sound/soc/codecs/ad193x.c | 140 +++++------------------------------------- sound/soc/codecs/ad193x.h | 7 +++ 7 files changed, 140 insertions(+), 127 deletions(-) create mode 100644 sound/soc/codecs/ad193x-i2c.c create mode 100644 sound/soc/codecs/ad193x-spi.c (limited to 'sound/soc/codecs') diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 54f74f8cbb75..359136777c4d 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -68,7 +68,8 @@ config SND_BF5XX_SOC_AD193X tristate "SoC AD193X Audio support for Blackfin" depends on SND_BF5XX_I2S select SND_BF5XX_SOC_I2S - select SND_SOC_AD193X + select SND_SOC_AD193X_I2C if I2C + select SND_SOC_AD193X_SPI if SPI_MASTER help Say Y if you want to add support for AD193X codec on Blackfin. This driver supports AD1936, AD1937, AD1938 and AD1939. diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 983d087aa92a..d7da6191fc55 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -16,7 +16,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_AB8500_CODEC if ABX500_CORE select SND_SOC_AC97_CODEC if SND_SOC_AC97_BUS select SND_SOC_AD1836 if SPI_MASTER - select SND_SOC_AD193X if SND_SOC_I2C_AND_SPI + select SND_SOC_AD193X_SPI if SPI_MASTER + select SND_SOC_AD193X_I2C if I2C select SND_SOC_AD1980 if SND_SOC_AC97_BUS select SND_SOC_AD73311 select SND_SOC_ADAU1373 if I2C @@ -182,6 +183,14 @@ config SND_SOC_AD1836 config SND_SOC_AD193X tristate +config SND_SOC_AD193X_SPI + tristate + select SND_SOC_AD193X + +config SND_SOC_AD193X_I2C + tristate + select SND_SOC_AD193X + config SND_SOC_AD1980 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index bc126764a44d..7795e37d313f 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -3,6 +3,8 @@ snd-soc-ab8500-codec-objs := ab8500-codec.o snd-soc-ac97-objs := ac97.o snd-soc-ad1836-objs := ad1836.o snd-soc-ad193x-objs := ad193x.o +snd-soc-ad193x-spi-objs := ad193x-spi.o +snd-soc-ad193x-i2c-objs := ad193x-i2c.o snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o snd-soc-adau1701-objs := adau1701.o @@ -134,6 +136,8 @@ obj-$(CONFIG_SND_SOC_AB8500_CODEC) += snd-soc-ab8500-codec.o obj-$(CONFIG_SND_SOC_AC97_CODEC) += snd-soc-ac97.o obj-$(CONFIG_SND_SOC_AD1836) += snd-soc-ad1836.o obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o +obj-$(CONFIG_SND_SOC_AD193X_SPI) += snd-soc-ad193x-spi.o +obj-$(CONFIG_SND_SOC_AD193X_I2C) += snd-soc-ad193x-i2c.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_ADAU1373) += snd-soc-adau1373.o diff --git a/sound/soc/codecs/ad193x-i2c.c b/sound/soc/codecs/ad193x-i2c.c new file mode 100644 index 000000000000..df3a1a415825 --- /dev/null +++ b/sound/soc/codecs/ad193x-i2c.c @@ -0,0 +1,54 @@ +/* + * AD1936/AD1937 audio driver + * + * Copyright 2014 Analog Devices Inc. + * + * Licensed under the GPL-2. + */ + +#include +#include +#include + +#include + +#include "ad193x.h" + +static const struct i2c_device_id ad193x_id[] = { + { "ad1936", 0 }, + { "ad1937", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ad193x_id); + +static int ad193x_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct regmap_config config; + + config = ad193x_regmap_config; + config.val_bits = 8; + config.reg_bits = 8; + + return ad193x_probe(&client->dev, devm_regmap_init_i2c(client, &config)); +} + +static int ad193x_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static struct i2c_driver ad193x_i2c_driver = { + .driver = { + .name = "ad193x", + }, + .probe = ad193x_i2c_probe, + .remove = ad193x_i2c_remove, + .id_table = ad193x_id, +}; +module_i2c_driver(ad193x_i2c_driver); + +MODULE_DESCRIPTION("ASoC AD1936/AD1937 audio CODEC driver"); +MODULE_AUTHOR("Barry Song <21cnbao@gmail.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ad193x-spi.c b/sound/soc/codecs/ad193x-spi.c new file mode 100644 index 000000000000..390cef9b9dc2 --- /dev/null +++ b/sound/soc/codecs/ad193x-spi.c @@ -0,0 +1,48 @@ +/* + * AD1938/AD1939 audio driver + * + * Copyright 2014 Analog Devices Inc. + * + * Licensed under the GPL-2. + */ + +#include +#include +#include + +#include + +#include "ad193x.h" + +static int ad193x_spi_probe(struct spi_device *spi) +{ + struct regmap_config config; + + config = ad193x_regmap_config; + config.val_bits = 8; + config.reg_bits = 16; + config.read_flag_mask = 0x09; + config.write_flag_mask = 0x08; + + return ad193x_probe(&spi->dev, devm_regmap_init_spi(spi, &config)); +} + +static int ad193x_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + return 0; +} + +static struct spi_driver ad193x_spi_driver = { + .driver = { + .name = "ad193x", + .owner = THIS_MODULE, + }, + .probe = ad193x_spi_probe, + .remove = ad193x_spi_remove, +}; +module_spi_driver(ad193x_spi_driver); + +MODULE_DESCRIPTION("ASoC AD1938/AD1939 audio CODEC driver"); +MODULE_AUTHOR("Barry Song <21cnbao@gmail.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 5a42dca535b7..f644a34a28de 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -6,12 +6,10 @@ * Licensed under the GPL-2 or later. */ -#include #include #include #include -#include -#include +#include #include #include #include @@ -19,6 +17,7 @@ #include #include #include + #include "ad193x.h" /* codec private data */ @@ -320,7 +319,7 @@ static struct snd_soc_dai_driver ad193x_dai = { .ops = &ad193x_dai_ops, }; -static int ad193x_probe(struct snd_soc_codec *codec) +static int ad193x_codec_probe(struct snd_soc_codec *codec) { struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec); int ret; @@ -352,7 +351,7 @@ static int ad193x_probe(struct snd_soc_codec *codec) } static struct snd_soc_codec_driver soc_codec_dev_ad193x = { - .probe = ad193x_probe, + .probe = ad193x_codec_probe, .controls = ad193x_snd_controls, .num_controls = ARRAY_SIZE(ad193x_snd_controls), .dapm_widgets = ad193x_dapm_widgets, @@ -366,140 +365,31 @@ static bool adau193x_reg_volatile(struct device *dev, unsigned int reg) return false; } -#if defined(CONFIG_SPI_MASTER) - -static const struct regmap_config ad193x_spi_regmap_config = { - .val_bits = 8, - .reg_bits = 16, - .read_flag_mask = 0x09, - .write_flag_mask = 0x08, - +const struct regmap_config ad193x_regmap_config = { .max_register = AD193X_NUM_REGS - 1, .volatile_reg = adau193x_reg_volatile, }; +EXPORT_SYMBOL_GPL(ad193x_regmap_config); -static int ad193x_spi_probe(struct spi_device *spi) +int ad193x_probe(struct device *dev, struct regmap *regmap) { struct ad193x_priv *ad193x; - ad193x = devm_kzalloc(&spi->dev, sizeof(struct ad193x_priv), - GFP_KERNEL); - if (ad193x == NULL) - return -ENOMEM; - - ad193x->regmap = devm_regmap_init_spi(spi, &ad193x_spi_regmap_config); - if (IS_ERR(ad193x->regmap)) - return PTR_ERR(ad193x->regmap); - - spi_set_drvdata(spi, ad193x); - - return snd_soc_register_codec(&spi->dev, &soc_codec_dev_ad193x, - &ad193x_dai, 1); -} - -static int ad193x_spi_remove(struct spi_device *spi) -{ - snd_soc_unregister_codec(&spi->dev); - return 0; -} - -static struct spi_driver ad193x_spi_driver = { - .driver = { - .name = "ad193x", - .owner = THIS_MODULE, - }, - .probe = ad193x_spi_probe, - .remove = ad193x_spi_remove, -}; -#endif - -#if IS_ENABLED(CONFIG_I2C) - -static const struct regmap_config ad193x_i2c_regmap_config = { - .val_bits = 8, - .reg_bits = 8, - - .max_register = AD193X_NUM_REGS - 1, - .volatile_reg = adau193x_reg_volatile, -}; - -static const struct i2c_device_id ad193x_id[] = { - { "ad1936", 0 }, - { "ad1937", 0 }, - { } -}; -MODULE_DEVICE_TABLE(i2c, ad193x_id); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); -static int ad193x_i2c_probe(struct i2c_client *client, - const struct i2c_device_id *id) -{ - struct ad193x_priv *ad193x; - - ad193x = devm_kzalloc(&client->dev, sizeof(struct ad193x_priv), - GFP_KERNEL); + ad193x = devm_kzalloc(dev, sizeof(*ad193x), GFP_KERNEL); if (ad193x == NULL) return -ENOMEM; - ad193x->regmap = devm_regmap_init_i2c(client, &ad193x_i2c_regmap_config); - if (IS_ERR(ad193x->regmap)) - return PTR_ERR(ad193x->regmap); - - i2c_set_clientdata(client, ad193x); - - return snd_soc_register_codec(&client->dev, &soc_codec_dev_ad193x, - &ad193x_dai, 1); -} - -static int ad193x_i2c_remove(struct i2c_client *client) -{ - snd_soc_unregister_codec(&client->dev); - return 0; -} + ad193x->regmap = regmap; -static struct i2c_driver ad193x_i2c_driver = { - .driver = { - .name = "ad193x", - }, - .probe = ad193x_i2c_probe, - .remove = ad193x_i2c_remove, - .id_table = ad193x_id, -}; -#endif - -static int __init ad193x_modinit(void) -{ - int ret; - -#if IS_ENABLED(CONFIG_I2C) - ret = i2c_add_driver(&ad193x_i2c_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register AD193X I2C driver: %d\n", - ret); - } -#endif - -#if defined(CONFIG_SPI_MASTER) - ret = spi_register_driver(&ad193x_spi_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register AD193X SPI driver: %d\n", - ret); - } -#endif - return ret; -} -module_init(ad193x_modinit); - -static void __exit ad193x_modexit(void) -{ -#if defined(CONFIG_SPI_MASTER) - spi_unregister_driver(&ad193x_spi_driver); -#endif + dev_set_drvdata(dev, ad193x); -#if IS_ENABLED(CONFIG_I2C) - i2c_del_driver(&ad193x_i2c_driver); -#endif + return snd_soc_register_codec(dev, &soc_codec_dev_ad193x, + &ad193x_dai, 1); } -module_exit(ad193x_modexit); +EXPORT_SYMBOL_GPL(ad193x_probe); MODULE_DESCRIPTION("ASoC ad193x driver"); MODULE_AUTHOR("Barry Song <21cnbao@gmail.com>"); diff --git a/sound/soc/codecs/ad193x.h b/sound/soc/codecs/ad193x.h index 473388049992..ab9a998f15be 100644 --- a/sound/soc/codecs/ad193x.h +++ b/sound/soc/codecs/ad193x.h @@ -9,6 +9,13 @@ #ifndef __AD193X_H__ #define __AD193X_H__ +#include + +struct device; + +extern const struct regmap_config ad193x_regmap_config; +int ad193x_probe(struct device *dev, struct regmap *regmap); + #define AD193X_PLL_CLK_CTRL0 0x00 #define AD193X_PLL_POWERDOWN 0x01 #define AD193X_PLL_INPUT_MASK 0x6 -- cgit v1.2.3 From c924dc68f7371582cb420c003faadb700cd4f76c Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 17 Feb 2014 13:16:53 +0100 Subject: ASoC: ssm2602: Split SPI and I2C code into different modules There are a few known (minor) problems with having the support code for both I2C and SPI in the same module: * We need to be extra careful to make sure to not build the driver into the kernel if one of the subsystems is build as a module (Currently only I2C can be build as a module). * The module init path error handling is rather ugly. E.g. what should be done if either the SPI or the I2C driver fails to register? Most drivers that implement SPI and I2C in the same module currently fallback to undefined behavior in that case. Splitting the the driver into two modules, one for each bus allows the registration of the other bus driver to continue without problems if one of them fails. This patch splits the ssm2602 driver into 3 modules. One core module that implements the device logic, but is independent of the bus method used. And one module for SPI and I2C each that registers the drivers and sets up the regmap struct for the bus. While we are at it also cleanup the include section of the ssm2602 driver and remove unneeded includes. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/blackfin/Kconfig | 3 +- sound/soc/codecs/Kconfig | 11 ++- sound/soc/codecs/Makefile | 4 ++ sound/soc/codecs/ssm2602-i2c.c | 57 +++++++++++++++ sound/soc/codecs/ssm2602-spi.c | 41 +++++++++++ sound/soc/codecs/ssm2602.c | 158 ++++++----------------------------------- sound/soc/codecs/ssm2602.h | 14 ++++ 7 files changed, 148 insertions(+), 140 deletions(-) create mode 100644 sound/soc/codecs/ssm2602-i2c.c create mode 100644 sound/soc/codecs/ssm2602-spi.c (limited to 'sound/soc/codecs') diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 54f74f8cbb75..f9118dc98853 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -14,7 +14,8 @@ config SND_BF5XX_SOC_SSM2602 depends on SND_BF5XX_I2S && (SPI_MASTER || I2C) select SND_BF5XX_SOC_I2S if !BF60x select SND_BF6XX_SOC_I2S if BF60x - select SND_SOC_SSM2602 + select SND_SOC_SSM2602_SPI if SPI_MASTER + select SND_SOC_SSM2602_I2C if I2C help Say Y if you want to add support for the Analog Devices SSM2602 Audio Codec Add-On Card. diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 983d087aa92a..f17e6da53ce7 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -66,7 +66,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_SN95031 if INTEL_SCU_IPC select SND_SOC_SPDIF select SND_SOC_SSM2518 if I2C - select SND_SOC_SSM2602 if SND_SOC_I2C_AND_SPI + select SND_SOC_SSM2602_SPI if SPI_MASTER + select SND_SOC_SSM2602_I2C if I2C select SND_SOC_STA32X if I2C select SND_SOC_STA529 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS @@ -342,6 +343,14 @@ config SND_SOC_SSM2518 config SND_SOC_SSM2602 tristate +config SND_SOC_SSM2602_SPI + select SND_SOC_SSM2602 + tristate + +config SND_SOC_SSM2602_I2C + select SND_SOC_SSM2602 + tristate + config SND_SOC_STA32X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index bc126764a44d..e7a2fb91148f 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -58,6 +58,8 @@ snd-soc-spdif-tx-objs := spdif_transmitter.o snd-soc-spdif-rx-objs := spdif_receiver.o snd-soc-ssm2518-objs := ssm2518.o snd-soc-ssm2602-objs := ssm2602.o +snd-soc-ssm2602-spi-objs := ssm2602-spi.o +snd-soc-ssm2602-i2c-objs := ssm2602-i2c.o snd-soc-sta32x-objs := sta32x.o snd-soc-sta529-objs := sta529.o snd-soc-stac9766-objs := stac9766.o @@ -188,6 +190,8 @@ obj-$(CONFIG_SND_SOC_SN95031) +=snd-soc-sn95031.o obj-$(CONFIG_SND_SOC_SPDIF) += snd-soc-spdif-rx.o snd-soc-spdif-tx.o obj-$(CONFIG_SND_SOC_SSM2518) += snd-soc-ssm2518.o obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o +obj-$(CONFIG_SND_SOC_SSM2602_SPI) += snd-soc-ssm2602-spi.o +obj-$(CONFIG_SND_SOC_SSM2602_I2C) += snd-soc-ssm2602-i2c.o obj-$(CONFIG_SND_SOC_STA32X) += snd-soc-sta32x.o obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o diff --git a/sound/soc/codecs/ssm2602-i2c.c b/sound/soc/codecs/ssm2602-i2c.c new file mode 100644 index 000000000000..abd63d537173 --- /dev/null +++ b/sound/soc/codecs/ssm2602-i2c.c @@ -0,0 +1,57 @@ +/* + * SSM2602/SSM2603/SSM2604 I2C audio driver + * + * Copyright 2014 Analog Devices Inc. + * + * Licensed under the GPL-2. + */ + +#include +#include +#include + +#include + +#include "ssm2602.h" + +/* + * ssm2602 2 wire address is determined by GPIO5 + * state during powerup. + * low = 0x1a + * high = 0x1b + */ +static int ssm2602_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + return ssm2602_probe(&client->dev, id->driver_data, + devm_regmap_init_i2c(client, &ssm2602_regmap_config)); +} + +static int ssm2602_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id ssm2602_i2c_id[] = { + { "ssm2602", SSM2602 }, + { "ssm2603", SSM2602 }, + { "ssm2604", SSM2604 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ssm2602_i2c_id); + +static struct i2c_driver ssm2602_i2c_driver = { + .driver = { + .name = "ssm2602", + .owner = THIS_MODULE, + }, + .probe = ssm2602_i2c_probe, + .remove = ssm2602_i2c_remove, + .id_table = ssm2602_i2c_id, +}; +module_i2c_driver(ssm2602_i2c_driver); + +MODULE_DESCRIPTION("ASoC SSM2602/SSM2603/SSM2604 I2C driver"); +MODULE_AUTHOR("Cliff Cai"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ssm2602-spi.c b/sound/soc/codecs/ssm2602-spi.c new file mode 100644 index 000000000000..2bf55e24a7bb --- /dev/null +++ b/sound/soc/codecs/ssm2602-spi.c @@ -0,0 +1,41 @@ +/* + * SSM2602 SPI audio driver + * + * Copyright 2014 Analog Devices Inc. + * + * Licensed under the GPL-2. + */ + +#include +#include +#include + +#include + +#include "ssm2602.h" + +static int ssm2602_spi_probe(struct spi_device *spi) +{ + return ssm2602_probe(&spi->dev, SSM2602, + devm_regmap_init_spi(spi, &ssm2602_regmap_config)); +} + +static int ssm2602_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + return 0; +} + +static struct spi_driver ssm2602_spi_driver = { + .driver = { + .name = "ssm2602", + .owner = THIS_MODULE, + }, + .probe = ssm2602_spi_probe, + .remove = ssm2602_spi_remove, +}; +module_spi_driver(ssm2602_spi_driver); + +MODULE_DESCRIPTION("ASoC SSM2602 SPI driver"); +MODULE_AUTHOR("Cliff Cai"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index f444d585b916..49d28eaa6d73 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -27,28 +27,16 @@ */ #include -#include -#include -#include -#include -#include -#include #include #include -#include + #include #include #include -#include #include #include "ssm2602.h" -enum ssm2602_type { - SSM2602, - SSM2604, -}; - /* codec private data */ struct ssm2602_priv { unsigned int sysclk; @@ -529,7 +517,7 @@ static int ssm2602_resume(struct snd_soc_codec *codec) return 0; } -static int ssm2602_probe(struct snd_soc_codec *codec) +static int ssm2602_codec_probe(struct snd_soc_codec *codec) { struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; @@ -554,7 +542,7 @@ static int ssm2602_probe(struct snd_soc_codec *codec) ARRAY_SIZE(ssm2602_routes)); } -static int ssm2604_probe(struct snd_soc_codec *codec) +static int ssm2604_codec_probe(struct snd_soc_codec *codec) { struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; @@ -568,7 +556,7 @@ static int ssm2604_probe(struct snd_soc_codec *codec) ARRAY_SIZE(ssm2604_routes)); } -static int ssm260x_probe(struct snd_soc_codec *codec) +static int ssm260x_codec_probe(struct snd_soc_codec *codec) { struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); int ret; @@ -597,10 +585,10 @@ static int ssm260x_probe(struct snd_soc_codec *codec) switch (ssm2602->type) { case SSM2602: - ret = ssm2602_probe(codec); + ret = ssm2602_codec_probe(codec); break; case SSM2604: - ret = ssm2604_probe(codec); + ret = ssm2604_codec_probe(codec); break; } @@ -620,7 +608,7 @@ static int ssm2602_remove(struct snd_soc_codec *codec) } static struct snd_soc_codec_driver soc_codec_dev_ssm2602 = { - .probe = ssm260x_probe, + .probe = ssm260x_codec_probe, .remove = ssm2602_remove, .suspend = ssm2602_suspend, .resume = ssm2602_resume, @@ -639,7 +627,7 @@ static bool ssm2602_register_volatile(struct device *dev, unsigned int reg) return reg == SSM2602_RESET; } -static const struct regmap_config ssm2602_regmap_config = { +const struct regmap_config ssm2602_regmap_config = { .val_bits = 9, .reg_bits = 7, @@ -650,134 +638,28 @@ static const struct regmap_config ssm2602_regmap_config = { .reg_defaults_raw = ssm2602_reg, .num_reg_defaults_raw = ARRAY_SIZE(ssm2602_reg), }; +EXPORT_SYMBOL_GPL(ssm2602_regmap_config); -#if defined(CONFIG_SPI_MASTER) -static int ssm2602_spi_probe(struct spi_device *spi) +int ssm2602_probe(struct device *dev, enum ssm2602_type type, + struct regmap *regmap) { struct ssm2602_priv *ssm2602; - int ret; - - ssm2602 = devm_kzalloc(&spi->dev, sizeof(struct ssm2602_priv), - GFP_KERNEL); - if (ssm2602 == NULL) - return -ENOMEM; - - spi_set_drvdata(spi, ssm2602); - ssm2602->type = SSM2602; - - ssm2602->regmap = devm_regmap_init_spi(spi, &ssm2602_regmap_config); - if (IS_ERR(ssm2602->regmap)) - return PTR_ERR(ssm2602->regmap); - - ret = snd_soc_register_codec(&spi->dev, - &soc_codec_dev_ssm2602, &ssm2602_dai, 1); - return ret; -} -static int ssm2602_spi_remove(struct spi_device *spi) -{ - snd_soc_unregister_codec(&spi->dev); - return 0; -} - -static struct spi_driver ssm2602_spi_driver = { - .driver = { - .name = "ssm2602", - .owner = THIS_MODULE, - }, - .probe = ssm2602_spi_probe, - .remove = ssm2602_spi_remove, -}; -#endif - -#if IS_ENABLED(CONFIG_I2C) -/* - * ssm2602 2 wire address is determined by GPIO5 - * state during powerup. - * low = 0x1a - * high = 0x1b - */ -static int ssm2602_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) -{ - struct ssm2602_priv *ssm2602; - int ret; + if (IS_ERR(regmap)) + return PTR_ERR(regmap); - ssm2602 = devm_kzalloc(&i2c->dev, sizeof(struct ssm2602_priv), - GFP_KERNEL); + ssm2602 = devm_kzalloc(dev, sizeof(*ssm2602), GFP_KERNEL); if (ssm2602 == NULL) return -ENOMEM; - i2c_set_clientdata(i2c, ssm2602); - ssm2602->type = id->driver_data; - - ssm2602->regmap = devm_regmap_init_i2c(i2c, &ssm2602_regmap_config); - if (IS_ERR(ssm2602->regmap)) - return PTR_ERR(ssm2602->regmap); - - ret = snd_soc_register_codec(&i2c->dev, - &soc_codec_dev_ssm2602, &ssm2602_dai, 1); - return ret; -} - -static int ssm2602_i2c_remove(struct i2c_client *client) -{ - snd_soc_unregister_codec(&client->dev); - return 0; -} - -static const struct i2c_device_id ssm2602_i2c_id[] = { - { "ssm2602", SSM2602 }, - { "ssm2603", SSM2602 }, - { "ssm2604", SSM2604 }, - { } -}; -MODULE_DEVICE_TABLE(i2c, ssm2602_i2c_id); - -/* corgi i2c codec control layer */ -static struct i2c_driver ssm2602_i2c_driver = { - .driver = { - .name = "ssm2602", - .owner = THIS_MODULE, - }, - .probe = ssm2602_i2c_probe, - .remove = ssm2602_i2c_remove, - .id_table = ssm2602_i2c_id, -}; -#endif - - -static int __init ssm2602_modinit(void) -{ - int ret = 0; - -#if defined(CONFIG_SPI_MASTER) - ret = spi_register_driver(&ssm2602_spi_driver); - if (ret) - return ret; -#endif - -#if IS_ENABLED(CONFIG_I2C) - ret = i2c_add_driver(&ssm2602_i2c_driver); - if (ret) - return ret; -#endif - - return ret; -} -module_init(ssm2602_modinit); - -static void __exit ssm2602_exit(void) -{ -#if defined(CONFIG_SPI_MASTER) - spi_unregister_driver(&ssm2602_spi_driver); -#endif + dev_set_drvdata(dev, ssm2602); + ssm2602->type = SSM2602; + ssm2602->regmap = regmap; -#if IS_ENABLED(CONFIG_I2C) - i2c_del_driver(&ssm2602_i2c_driver); -#endif + return snd_soc_register_codec(dev, &soc_codec_dev_ssm2602, + &ssm2602_dai, 1); } -module_exit(ssm2602_exit); +EXPORT_SYMBOL_GPL(ssm2602_probe); MODULE_DESCRIPTION("ASoC SSM2602/SSM2603/SSM2604 driver"); MODULE_AUTHOR("Cliff Cai"); diff --git a/sound/soc/codecs/ssm2602.h b/sound/soc/codecs/ssm2602.h index fbd07d7b73ca..747538847689 100644 --- a/sound/soc/codecs/ssm2602.h +++ b/sound/soc/codecs/ssm2602.h @@ -28,6 +28,20 @@ #ifndef _SSM2602_H #define _SSM2602_H +#include + +struct device; + +enum ssm2602_type { + SSM2602, + SSM2604, +}; + +extern const struct regmap_config ssm2602_regmap_config; + +int ssm2602_probe(struct device *dev, enum ssm2602_type type, + struct regmap *regmap); + /* SSM2602 Codec Register definitions */ #define SSM2602_LINVOL 0x00 -- cgit v1.2.3 From f96a5d3f1c09ce85ac1a90d733ca3585b9f2f70a Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 17 Feb 2014 13:16:55 +0100 Subject: ASoC: adav80x: Use devm_kzalloc() Use devm_kzalloc() to allocate the device state struct. Saves use from having to free it manually on the error path and in the remove callback. Now that the adav80x_bus_probe() function is only a call to snd_soc_unregister_codec() also inline that. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adav80x.c | 22 ++++++---------------- 1 file changed, 6 insertions(+), 16 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index f78b27a7c461..a4bd051c5430 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -872,27 +872,15 @@ static int adav80x_bus_probe(struct device *dev, struct regmap *regmap) if (IS_ERR(regmap)) return PTR_ERR(regmap); - adav80x = kzalloc(sizeof(*adav80x), GFP_KERNEL); + adav80x = devm_kzalloc(dev, sizeof(*adav80x), GFP_KERNEL); if (!adav80x) return -ENOMEM; - dev_set_drvdata(dev, adav80x); adav80x->regmap = regmap; - ret = snd_soc_register_codec(dev, &adav80x_codec_driver, + return snd_soc_register_codec(dev, &adav80x_codec_driver, adav80x_dais, ARRAY_SIZE(adav80x_dais)); - if (ret) - kfree(adav80x); - - return ret; -} - -static int adav80x_bus_remove(struct device *dev) -{ - snd_soc_unregister_codec(dev); - kfree(dev_get_drvdata(dev)); - return 0; } #if defined(CONFIG_SPI_MASTER) @@ -923,7 +911,8 @@ static int adav80x_spi_probe(struct spi_device *spi) static int adav80x_spi_remove(struct spi_device *spi) { - return adav80x_bus_remove(&spi->dev); + snd_soc_unregister_codec(dev); + return 0; } static struct spi_driver adav80x_spi_driver = { @@ -965,7 +954,8 @@ static int adav80x_i2c_probe(struct i2c_client *client, static int adav80x_i2c_remove(struct i2c_client *client) { - return adav80x_bus_remove(&client->dev); + snd_soc_unregister_codec(dev); + return 0; } static struct i2c_driver adav80x_i2c_driver = { -- cgit v1.2.3 From 0c2d6964562835501280409cac5d4ee28e07e8c2 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Mon, 17 Feb 2014 13:16:56 +0100 Subject: ASoC: adav80x: Split SPI and I2C code into different modules There are a few known (minor) problems with having the support code for both I2C and SPI in the same module: * We need to be extra careful to make sure to not build the driver into the kernel if one of the subsystems is build as a module (Currently only I2C can be build as a module). * The module init path error handling is rather ugly. E.g. what should be done if either the SPI or the I2C driver fails to register. Most drivers that implement SPI and I2C in the same module currently fallback to undefined behavior in that case. Splitting the the driver into two modules, one for each bus, allows the registration of the other bus drive to continue without problems if one of them fails. This patch splits the ADAV80X driver into 3 modules. One core module that implements the device logic, but is independent of the bus method used. And one module for SPI and I2C each that registers the drivers and sets up the regmap struct for the bus. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/blackfin/Kconfig | 3 +- sound/soc/codecs/Kconfig | 11 ++++- sound/soc/codecs/Makefile | 4 ++ sound/soc/codecs/adav801.c | 53 ++++++++++++++++++++ sound/soc/codecs/adav803.c | 50 +++++++++++++++++++ sound/soc/codecs/adav80x.c | 117 +++------------------------------------------ sound/soc/codecs/adav80x.h | 7 +++ 7 files changed, 133 insertions(+), 112 deletions(-) create mode 100644 sound/soc/codecs/adav801.c create mode 100644 sound/soc/codecs/adav803.c (limited to 'sound/soc/codecs') diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig index 54f74f8cbb75..18abd884d84f 100644 --- a/sound/soc/blackfin/Kconfig +++ b/sound/soc/blackfin/Kconfig @@ -47,7 +47,8 @@ config SND_SOC_BFIN_EVAL_ADAV80X tristate "Support for the EVAL-ADAV80X boards on Blackfin eval boards" depends on SND_BF5XX_I2S && (SPI_MASTER || I2C) select SND_BF5XX_SOC_I2S - select SND_SOC_ADAV80X + select SND_SOC_ADAV801 if SPI_MASTER + select SND_SOC_ADAV803 if I2C help Say Y if you want to add support for the Analog Devices EVAL-ADAV801 or EVAL-ADAV803 board connected to one of the Blackfin evaluation boards diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 983d087aa92a..9556321c7b8b 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -20,7 +20,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_AD1980 if SND_SOC_AC97_BUS select SND_SOC_AD73311 select SND_SOC_ADAU1373 if I2C - select SND_SOC_ADAV80X if SND_SOC_I2C_AND_SPI + select SND_SOC_ADAV801 if SPI_MASTER + select SND_SOC_ADAV803 if I2C select SND_SOC_ADAU1701 if I2C select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER @@ -198,6 +199,14 @@ config SND_SOC_ADAU1373 config SND_SOC_ADAV80X tristate +config SND_SOC_ADAV801 + tristate + select SND_SOC_ADAV80X + +config SND_SOC_ADAV803 + tristate + select SND_SOC_ADAV80X + config SND_SOC_ADS117X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index bc126764a44d..1b2c6eca863f 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -8,6 +8,8 @@ snd-soc-ad73311-objs := ad73311.o snd-soc-adau1701-objs := adau1701.o snd-soc-adau1373-objs := adau1373.o snd-soc-adav80x-objs := adav80x.o +snd-soc-adav801-objs := adav801.o +snd-soc-adav803-objs := adav803.o snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o @@ -139,6 +141,8 @@ obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_ADAU1373) += snd-soc-adau1373.o obj-$(CONFIG_SND_SOC_ADAU1701) += snd-soc-adau1701.o obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o +obj-$(CONFIG_SND_SOC_ADAV801) += snd-soc-adav801.o +obj-$(CONFIG_SND_SOC_ADAV803) += snd-soc-adav803.o obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o diff --git a/sound/soc/codecs/adav801.c b/sound/soc/codecs/adav801.c new file mode 100644 index 000000000000..790fce33ab10 --- /dev/null +++ b/sound/soc/codecs/adav801.c @@ -0,0 +1,53 @@ +/* + * ADAV801 audio driver + * + * Copyright 2014 Analog Devices Inc. + * + * Licensed under the GPL-2. + */ + +#include +#include +#include + +#include + +#include "adav80x.h" + +static const struct spi_device_id adav80x_spi_id[] = { + { "adav801", 0 }, + { } +}; +MODULE_DEVICE_TABLE(spi, adav80x_spi_id); + +static int adav80x_spi_probe(struct spi_device *spi) +{ + struct regmap_config config; + + config = adav80x_regmap_config; + config.read_flag_mask = 0x01; + + return adav80x_bus_probe(&spi->dev, devm_regmap_init_spi(spi, &config)); +} + +static int adav80x_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + return 0; +} + +static struct spi_driver adav80x_spi_driver = { + .driver = { + .name = "adav801", + .owner = THIS_MODULE, + }, + .probe = adav80x_spi_probe, + .remove = adav80x_spi_remove, + .id_table = adav80x_spi_id, +}; +module_spi_driver(adav80x_spi_driver); + +MODULE_DESCRIPTION("ASoC ADAV801 driver"); +MODULE_AUTHOR("Lars-Peter Clausen "); +MODULE_AUTHOR("Yi Li >"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adav803.c b/sound/soc/codecs/adav803.c new file mode 100644 index 000000000000..66d9fce34e62 --- /dev/null +++ b/sound/soc/codecs/adav803.c @@ -0,0 +1,50 @@ +/* + * ADAV803 audio driver + * + * Copyright 2014 Analog Devices Inc. + * + * Licensed under the GPL-2. + */ + +#include +#include +#include + +#include + +#include "adav80x.h" + +static const struct i2c_device_id adav803_id[] = { + { "adav803", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, adav803_id); + +static int adav803_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + return adav80x_bus_probe(&client->dev, + devm_regmap_init_i2c(client, &adav80x_regmap_config)); +} + +static int adav803_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static struct i2c_driver adav803_driver = { + .driver = { + .name = "adav803", + .owner = THIS_MODULE, + }, + .probe = adav803_probe, + .remove = adav803_remove, + .id_table = adav803_id, +}; +module_i2c_driver(adav803_driver); + +MODULE_DESCRIPTION("ASoC ADAV803 driver"); +MODULE_AUTHOR("Lars-Peter Clausen "); +MODULE_AUTHOR("Yi Li >"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index a4bd051c5430..09d560962e8d 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -8,17 +8,15 @@ * Licensed under the GPL-2 or later. */ -#include #include #include -#include -#include +#include #include -#include + #include #include -#include #include +#include #include "adav80x.h" @@ -864,10 +862,9 @@ static struct snd_soc_codec_driver adav80x_codec_driver = { .num_dapm_routes = ARRAY_SIZE(adav80x_dapm_routes), }; -static int adav80x_bus_probe(struct device *dev, struct regmap *regmap) +int adav80x_bus_probe(struct device *dev, struct regmap *regmap) { struct adav80x *adav80x; - int ret; if (IS_ERR(regmap)) return PTR_ERR(regmap); @@ -882,9 +879,9 @@ static int adav80x_bus_probe(struct device *dev, struct regmap *regmap) return snd_soc_register_codec(dev, &adav80x_codec_driver, adav80x_dais, ARRAY_SIZE(adav80x_dais)); } +EXPORT_SYMBOL_GPL(adav80x_bus_probe); -#if defined(CONFIG_SPI_MASTER) -static const struct regmap_config adav80x_spi_regmap_config = { +const struct regmap_config adav80x_regmap_config = { .val_bits = 8, .pad_bits = 1, .reg_bits = 7, @@ -896,107 +893,7 @@ static const struct regmap_config adav80x_spi_regmap_config = { .reg_defaults = adav80x_reg_defaults, .num_reg_defaults = ARRAY_SIZE(adav80x_reg_defaults), }; - -static const struct spi_device_id adav80x_spi_id[] = { - { "adav801", 0 }, - { } -}; -MODULE_DEVICE_TABLE(spi, adav80x_spi_id); - -static int adav80x_spi_probe(struct spi_device *spi) -{ - return adav80x_bus_probe(&spi->dev, - devm_regmap_init_spi(spi, &adav80x_spi_regmap_config)); -} - -static int adav80x_spi_remove(struct spi_device *spi) -{ - snd_soc_unregister_codec(dev); - return 0; -} - -static struct spi_driver adav80x_spi_driver = { - .driver = { - .name = "adav801", - .owner = THIS_MODULE, - }, - .probe = adav80x_spi_probe, - .remove = adav80x_spi_remove, - .id_table = adav80x_spi_id, -}; -#endif - -#if IS_ENABLED(CONFIG_I2C) -static const struct regmap_config adav80x_i2c_regmap_config = { - .val_bits = 8, - .pad_bits = 1, - .reg_bits = 7, - - .max_register = ADAV80X_PLL_OUTE, - - .cache_type = REGCACHE_RBTREE, - .reg_defaults = adav80x_reg_defaults, - .num_reg_defaults = ARRAY_SIZE(adav80x_reg_defaults), -}; - -static const struct i2c_device_id adav80x_i2c_id[] = { - { "adav803", 0 }, - { } -}; -MODULE_DEVICE_TABLE(i2c, adav80x_i2c_id); - -static int adav80x_i2c_probe(struct i2c_client *client, - const struct i2c_device_id *id) -{ - return adav80x_bus_probe(&client->dev, - devm_regmap_init_i2c(client, &adav80x_i2c_regmap_config)); -} - -static int adav80x_i2c_remove(struct i2c_client *client) -{ - snd_soc_unregister_codec(dev); - return 0; -} - -static struct i2c_driver adav80x_i2c_driver = { - .driver = { - .name = "adav803", - .owner = THIS_MODULE, - }, - .probe = adav80x_i2c_probe, - .remove = adav80x_i2c_remove, - .id_table = adav80x_i2c_id, -}; -#endif - -static int __init adav80x_init(void) -{ - int ret = 0; - -#if IS_ENABLED(CONFIG_I2C) - ret = i2c_add_driver(&adav80x_i2c_driver); - if (ret) - return ret; -#endif - -#if defined(CONFIG_SPI_MASTER) - ret = spi_register_driver(&adav80x_spi_driver); -#endif - - return ret; -} -module_init(adav80x_init); - -static void __exit adav80x_exit(void) -{ -#if IS_ENABLED(CONFIG_I2C) - i2c_del_driver(&adav80x_i2c_driver); -#endif -#if defined(CONFIG_SPI_MASTER) - spi_unregister_driver(&adav80x_spi_driver); -#endif -} -module_exit(adav80x_exit); +EXPORT_SYMBOL_GPL(adav80x_regmap_config); MODULE_DESCRIPTION("ASoC ADAV80x driver"); MODULE_AUTHOR("Lars-Peter Clausen "); diff --git a/sound/soc/codecs/adav80x.h b/sound/soc/codecs/adav80x.h index adb0fc76d4e3..8a1d7c09dca5 100644 --- a/sound/soc/codecs/adav80x.h +++ b/sound/soc/codecs/adav80x.h @@ -9,6 +9,13 @@ #ifndef _ADAV80X_H #define _ADAV80X_H +#include + +struct device; + +extern const struct regmap_config adav80x_regmap_config; +int adav80x_bus_probe(struct device *dev, struct regmap *regmap); + enum adav80x_pll_src { ADAV80X_PLL_SRC_XIN, ADAV80X_PLL_SRC_XTAL, -- cgit v1.2.3 From 423f0c4a3d32cc83dff204324f59aecb4516f3cf Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Tue, 18 Feb 2014 14:32:48 +0530 Subject: ASoC: cs42l51: Remove unused variable MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit ‘cs42l51’ is not used. Remove it. Signed-off-by: Sachin Kamat Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l51.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index e53c8714591f..3eab46020a30 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -487,7 +487,6 @@ static struct snd_soc_dai_driver cs42l51_dai = { static int cs42l51_probe(struct snd_soc_codec *codec) { - struct cs42l51_private *cs42l51 = snd_soc_codec_get_drvdata(codec); int ret, reg; ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); -- cgit v1.2.3 From 603597c9375b8162edae3231dd4cc7f1f500de79 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 18 Feb 2014 15:57:35 +0100 Subject: ASoC: Add ADAU1977 CODEC driver This patch adds support for the ADAU1977, ADAU1978 and ADAU1979 audio CODEC devices. They are a family of 4-channel differential input audio ADC devices. They can be connected to either a SPI or I2C bus. The driver is implemented in three modules, one main module (adau1977.ko) which implements the device logic and one module each for SPI (adau1977-spi.ko) and I2C (adau1977-i2c.ko) bus access. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/linux/platform_data/adau1977.h | 45 ++ sound/soc/codecs/Kconfig | 15 + sound/soc/codecs/Makefile | 6 + sound/soc/codecs/adau1977-i2c.c | 59 ++ sound/soc/codecs/adau1977-spi.c | 76 +++ sound/soc/codecs/adau1977.c | 1018 ++++++++++++++++++++++++++++++++ sound/soc/codecs/adau1977.h | 37 ++ 7 files changed, 1256 insertions(+) create mode 100644 include/linux/platform_data/adau1977.h create mode 100644 sound/soc/codecs/adau1977-i2c.c create mode 100644 sound/soc/codecs/adau1977-spi.c create mode 100644 sound/soc/codecs/adau1977.c create mode 100644 sound/soc/codecs/adau1977.h (limited to 'sound/soc/codecs') diff --git a/include/linux/platform_data/adau1977.h b/include/linux/platform_data/adau1977.h new file mode 100644 index 000000000000..bed11d908f92 --- /dev/null +++ b/include/linux/platform_data/adau1977.h @@ -0,0 +1,45 @@ +/* + * ADAU1977/ADAU1978/ADAU1979 driver + * + * Copyright 2014 Analog Devices Inc. + * Author: Lars-Peter Clausen + * + * Licensed under the GPL-2. + */ + +#ifndef __LINUX_PLATFORM_DATA_ADAU1977_H__ +#define __LINUX_PLATFORM_DATA_ADAU1977_H__ + +/** + * enum adau1977_micbias - ADAU1977 MICBIAS pin voltage setting + * @ADAU1977_MICBIAS_5V0: MICBIAS is set to 5.0 V + * @ADAU1977_MICBIAS_5V5: MICBIAS is set to 5.5 V + * @ADAU1977_MICBIAS_6V0: MICBIAS is set to 6.0 V + * @ADAU1977_MICBIAS_6V5: MICBIAS is set to 6.5 V + * @ADAU1977_MICBIAS_7V0: MICBIAS is set to 7.0 V + * @ADAU1977_MICBIAS_7V5: MICBIAS is set to 7.5 V + * @ADAU1977_MICBIAS_8V0: MICBIAS is set to 8.0 V + * @ADAU1977_MICBIAS_8V5: MICBIAS is set to 8.5 V + * @ADAU1977_MICBIAS_9V0: MICBIAS is set to 9.0 V + */ +enum adau1977_micbias { + ADAU1977_MICBIAS_5V0 = 0x0, + ADAU1977_MICBIAS_5V5 = 0x1, + ADAU1977_MICBIAS_6V0 = 0x2, + ADAU1977_MICBIAS_6V5 = 0x3, + ADAU1977_MICBIAS_7V0 = 0x4, + ADAU1977_MICBIAS_7V5 = 0x5, + ADAU1977_MICBIAS_8V0 = 0x6, + ADAU1977_MICBIAS_8V5 = 0x7, + ADAU1977_MICBIAS_9V0 = 0x8, +}; + +/** + * struct adau1977_platform_data - Platform configuration data for the ADAU1977 + * @micbias: Specifies the voltage for the MICBIAS pin + */ +struct adau1977_platform_data { + enum adau1977_micbias micbias; +}; + +#endif diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 9ea428bdd872..4535674f6eba 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -24,6 +24,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_ADAU1373 if I2C select SND_SOC_ADAV801 if SPI_MASTER select SND_SOC_ADAV803 if I2C + select SND_SOC_ADAU1977_SPI if SPI_MASTER + select SND_SOC_ADAU1977_I2C if I2C select SND_SOC_ADAU1701 if I2C select SND_SOC_ADS117X select SND_SOC_AK4104 if SPI_MASTER @@ -200,6 +202,19 @@ config SND_SOC_ADAU1701 config SND_SOC_ADAU1373 tristate +config SND_SOC_ADAU1977 + tristate + +config SND_SOC_ADAU1977_SPI + tristate + select SND_SOC_ADAU1977 + select REGMAP_SPI + +config SND_SOC_ADAU1977_I2C + tristate + select SND_SOC_ADAU1977 + select REGMAP_I2C + config SND_SOC_ADAV80X tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 1b2c6eca863f..bfd85ec2dfa8 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -7,6 +7,9 @@ snd-soc-ad1980-objs := ad1980.o snd-soc-ad73311-objs := ad73311.o snd-soc-adau1701-objs := adau1701.o snd-soc-adau1373-objs := adau1373.o +snd-soc-adau1977-objs := adau1977.o +snd-soc-adau1977-spi-objs := adau1977-spi.o +snd-soc-adau1977-i2c-objs := adau1977-i2c.o snd-soc-adav80x-objs := adav80x.o snd-soc-adav801-objs := adav801.o snd-soc-adav803-objs := adav803.o @@ -139,6 +142,9 @@ obj-$(CONFIG_SND_SOC_AD193X) += snd-soc-ad193x.o obj-$(CONFIG_SND_SOC_AD1980) += snd-soc-ad1980.o obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_ADAU1373) += snd-soc-adau1373.o +obj-$(CONFIG_SND_SOC_ADAU1977) += snd-soc-adau1977.o +obj-$(CONFIG_SND_SOC_ADAU1977_SPI) += snd-soc-adau1977-spi.o +obj-$(CONFIG_SND_SOC_ADAU1977_I2C) += snd-soc-adau1977-i2c.o obj-$(CONFIG_SND_SOC_ADAU1701) += snd-soc-adau1701.o obj-$(CONFIG_SND_SOC_ADAV80X) += snd-soc-adav80x.o obj-$(CONFIG_SND_SOC_ADAV801) += snd-soc-adav801.o diff --git a/sound/soc/codecs/adau1977-i2c.c b/sound/soc/codecs/adau1977-i2c.c new file mode 100644 index 000000000000..9700e8c838c9 --- /dev/null +++ b/sound/soc/codecs/adau1977-i2c.c @@ -0,0 +1,59 @@ +/* + * ADAU1977/ADAU1978/ADAU1979 driver + * + * Copyright 2014 Analog Devices Inc. + * Author: Lars-Peter Clausen + * + * Licensed under the GPL-2. + */ + +#include +#include +#include +#include +#include + +#include "adau1977.h" + +static int adau1977_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct regmap_config config; + + config = adau1977_regmap_config; + config.val_bits = 8; + config.reg_bits = 8; + + return adau1977_probe(&client->dev, + devm_regmap_init_i2c(client, &config), + id->driver_data, NULL); +} + +static int adau1977_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id adau1977_i2c_ids[] = { + { "adau1977", ADAU1977 }, + { "adau1978", ADAU1978 }, + { "adau1979", ADAU1978 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, adau1977_i2c_ids); + +static struct i2c_driver adau1977_i2c_driver = { + .driver = { + .name = "adau1977", + .owner = THIS_MODULE, + }, + .probe = adau1977_i2c_probe, + .remove = adau1977_i2c_remove, + .id_table = adau1977_i2c_ids, +}; +module_i2c_driver(adau1977_i2c_driver); + +MODULE_DESCRIPTION("ASoC ADAU1977/ADAU1978/ADAU1979 driver"); +MODULE_AUTHOR("Lars-Peter Clausen "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau1977-spi.c b/sound/soc/codecs/adau1977-spi.c new file mode 100644 index 000000000000..b05cf5da3a94 --- /dev/null +++ b/sound/soc/codecs/adau1977-spi.c @@ -0,0 +1,76 @@ +/* + * ADAU1977/ADAU1978/ADAU1979 driver + * + * Copyright 2014 Analog Devices Inc. + * Author: Lars-Peter Clausen + * + * Licensed under the GPL-2. + */ + +#include +#include +#include +#include +#include + +#include "adau1977.h" + +static void adau1977_spi_switch_mode(struct device *dev) +{ + struct spi_device *spi = to_spi_device(dev); + + /* + * To get the device into SPI mode CLATCH has to be pulled low three + * times. Do this by issuing three dummy reads. + */ + spi_w8r8(spi, 0x00); + spi_w8r8(spi, 0x00); + spi_w8r8(spi, 0x00); +} + +static int adau1977_spi_probe(struct spi_device *spi) +{ + const struct spi_device_id *id = spi_get_device_id(spi); + struct regmap_config config; + + if (!id) + return -EINVAL; + + config = adau1977_regmap_config; + config.val_bits = 8; + config.reg_bits = 16; + config.read_flag_mask = 0x1; + + return adau1977_probe(&spi->dev, + devm_regmap_init_spi(spi, &config), + id->driver_data, adau1977_spi_switch_mode); +} + +static int adau1977_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + return 0; +} + +static const struct spi_device_id adau1977_spi_ids[] = { + { "adau1977", ADAU1977 }, + { "adau1978", ADAU1978 }, + { "adau1979", ADAU1978 }, + { } +}; +MODULE_DEVICE_TABLE(spi, adau1977_spi_ids); + +static struct spi_driver adau1977_spi_driver = { + .driver = { + .name = "adau1977", + .owner = THIS_MODULE, + }, + .probe = adau1977_spi_probe, + .remove = adau1977_spi_remove, + .id_table = adau1977_spi_ids, +}; +module_spi_driver(adau1977_spi_driver); + +MODULE_DESCRIPTION("ASoC ADAU1977/ADAU1978/ADAU1979 driver"); +MODULE_AUTHOR("Lars-Peter Clausen "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau1977.c b/sound/soc/codecs/adau1977.c new file mode 100644 index 000000000000..fd55da7cb9d4 --- /dev/null +++ b/sound/soc/codecs/adau1977.c @@ -0,0 +1,1018 @@ +/* + * ADAU1977/ADAU1978/ADAU1979 driver + * + * Copyright 2014 Analog Devices Inc. + * Author: Lars-Peter Clausen + * + * Licensed under the GPL-2. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include +#include + +#include "adau1977.h" + +#define ADAU1977_REG_POWER 0x00 +#define ADAU1977_REG_PLL 0x01 +#define ADAU1977_REG_BOOST 0x02 +#define ADAU1977_REG_MICBIAS 0x03 +#define ADAU1977_REG_BLOCK_POWER_SAI 0x04 +#define ADAU1977_REG_SAI_CTRL0 0x05 +#define ADAU1977_REG_SAI_CTRL1 0x06 +#define ADAU1977_REG_CMAP12 0x07 +#define ADAU1977_REG_CMAP34 0x08 +#define ADAU1977_REG_SAI_OVERTEMP 0x09 +#define ADAU1977_REG_POST_ADC_GAIN(x) (0x0a + (x)) +#define ADAU1977_REG_MISC_CONTROL 0x0e +#define ADAU1977_REG_DIAG_CONTROL 0x10 +#define ADAU1977_REG_STATUS(x) (0x11 + (x)) +#define ADAU1977_REG_DIAG_IRQ1 0x15 +#define ADAU1977_REG_DIAG_IRQ2 0x16 +#define ADAU1977_REG_ADJUST1 0x17 +#define ADAU1977_REG_ADJUST2 0x18 +#define ADAU1977_REG_ADC_CLIP 0x19 +#define ADAU1977_REG_DC_HPF_CAL 0x1a + +#define ADAU1977_POWER_RESET BIT(7) +#define ADAU1977_POWER_PWUP BIT(0) + +#define ADAU1977_PLL_CLK_S BIT(4) +#define ADAU1977_PLL_MCS_MASK 0x7 + +#define ADAU1977_MICBIAS_MB_VOLTS_MASK 0xf0 +#define ADAU1977_MICBIAS_MB_VOLTS_OFFSET 4 + +#define ADAU1977_BLOCK_POWER_SAI_LR_POL BIT(7) +#define ADAU1977_BLOCK_POWER_SAI_BCLK_EDGE BIT(6) +#define ADAU1977_BLOCK_POWER_SAI_LDO_EN BIT(5) + +#define ADAU1977_SAI_CTRL0_FMT_MASK (0x3 << 6) +#define ADAU1977_SAI_CTRL0_FMT_I2S (0x0 << 6) +#define ADAU1977_SAI_CTRL0_FMT_LJ (0x1 << 6) +#define ADAU1977_SAI_CTRL0_FMT_RJ_24BIT (0x2 << 6) +#define ADAU1977_SAI_CTRL0_FMT_RJ_16BIT (0x3 << 6) + +#define ADAU1977_SAI_CTRL0_SAI_MASK (0x7 << 3) +#define ADAU1977_SAI_CTRL0_SAI_I2S (0x0 << 3) +#define ADAU1977_SAI_CTRL0_SAI_TDM_2 (0x1 << 3) +#define ADAU1977_SAI_CTRL0_SAI_TDM_4 (0x2 << 3) +#define ADAU1977_SAI_CTRL0_SAI_TDM_8 (0x3 << 3) +#define ADAU1977_SAI_CTRL0_SAI_TDM_16 (0x4 << 3) + +#define ADAU1977_SAI_CTRL0_FS_MASK (0x7) +#define ADAU1977_SAI_CTRL0_FS_8000_12000 (0x0) +#define ADAU1977_SAI_CTRL0_FS_16000_24000 (0x1) +#define ADAU1977_SAI_CTRL0_FS_32000_48000 (0x2) +#define ADAU1977_SAI_CTRL0_FS_64000_96000 (0x3) +#define ADAU1977_SAI_CTRL0_FS_128000_192000 (0x4) + +#define ADAU1977_SAI_CTRL1_SLOT_WIDTH_MASK (0x3 << 5) +#define ADAU1977_SAI_CTRL1_SLOT_WIDTH_32 (0x0 << 5) +#define ADAU1977_SAI_CTRL1_SLOT_WIDTH_24 (0x1 << 5) +#define ADAU1977_SAI_CTRL1_SLOT_WIDTH_16 (0x2 << 5) +#define ADAU1977_SAI_CTRL1_DATA_WIDTH_MASK (0x1 << 4) +#define ADAU1977_SAI_CTRL1_DATA_WIDTH_16BIT (0x1 << 4) +#define ADAU1977_SAI_CTRL1_DATA_WIDTH_24BIT (0x0 << 4) +#define ADAU1977_SAI_CTRL1_LRCLK_PULSE BIT(3) +#define ADAU1977_SAI_CTRL1_MSB BIT(2) +#define ADAU1977_SAI_CTRL1_BCLKRATE_16 (0x1 << 1) +#define ADAU1977_SAI_CTRL1_BCLKRATE_32 (0x0 << 1) +#define ADAU1977_SAI_CTRL1_BCLKRATE_MASK (0x1 << 1) +#define ADAU1977_SAI_CTRL1_MASTER BIT(0) + +#define ADAU1977_SAI_OVERTEMP_DRV_C(x) BIT(4 + (x)) +#define ADAU1977_SAI_OVERTEMP_DRV_HIZ BIT(3) + +#define ADAU1977_MISC_CONTROL_SUM_MODE_MASK (0x3 << 6) +#define ADAU1977_MISC_CONTROL_SUM_MODE_1CH (0x2 << 6) +#define ADAU1977_MISC_CONTROL_SUM_MODE_2CH (0x1 << 6) +#define ADAU1977_MISC_CONTROL_SUM_MODE_4CH (0x0 << 6) +#define ADAU1977_MISC_CONTROL_MMUTE BIT(4) +#define ADAU1977_MISC_CONTROL_DC_CAL BIT(0) + +#define ADAU1977_CHAN_MAP_SECOND_SLOT_OFFSET 4 +#define ADAU1977_CHAN_MAP_FIRST_SLOT_OFFSET 0 + +struct adau1977 { + struct regmap *regmap; + bool right_j; + unsigned int sysclk; + enum adau1977_sysclk_src sysclk_src; + struct gpio_desc *reset_gpio; + enum adau1977_type type; + + struct regulator *avdd_reg; + struct regulator *dvdd_reg; + + struct snd_pcm_hw_constraint_list constraints; + + struct device *dev; + void (*switch_mode)(struct device *dev); + + unsigned int max_master_fs; + unsigned int slot_width; + bool enabled; + bool master; +}; + +static const struct reg_default adau1977_reg_defaults[] = { + { 0x00, 0x00 }, + { 0x01, 0x41 }, + { 0x02, 0x4a }, + { 0x03, 0x7d }, + { 0x04, 0x3d }, + { 0x05, 0x02 }, + { 0x06, 0x00 }, + { 0x07, 0x10 }, + { 0x08, 0x32 }, + { 0x09, 0xf0 }, + { 0x0a, 0xa0 }, + { 0x0b, 0xa0 }, + { 0x0c, 0xa0 }, + { 0x0d, 0xa0 }, + { 0x0e, 0x02 }, + { 0x10, 0x0f }, + { 0x15, 0x20 }, + { 0x16, 0x00 }, + { 0x17, 0x00 }, + { 0x18, 0x00 }, + { 0x1a, 0x00 }, +}; + +static const DECLARE_TLV_DB_MINMAX_MUTE(adau1977_adc_gain, -3562, 6000); + +static const struct snd_soc_dapm_widget adau1977_micbias_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("MICBIAS", ADAU1977_REG_MICBIAS, + 3, 0, NULL, 0) +}; + +static const struct snd_soc_dapm_widget adau1977_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("Vref", ADAU1977_REG_BLOCK_POWER_SAI, + 4, 0, NULL, 0), + + SND_SOC_DAPM_ADC("ADC1", "Capture", ADAU1977_REG_BLOCK_POWER_SAI, 0, 0), + SND_SOC_DAPM_ADC("ADC2", "Capture", ADAU1977_REG_BLOCK_POWER_SAI, 1, 0), + SND_SOC_DAPM_ADC("ADC3", "Capture", ADAU1977_REG_BLOCK_POWER_SAI, 2, 0), + SND_SOC_DAPM_ADC("ADC4", "Capture", ADAU1977_REG_BLOCK_POWER_SAI, 3, 0), + + SND_SOC_DAPM_INPUT("AIN1"), + SND_SOC_DAPM_INPUT("AIN2"), + SND_SOC_DAPM_INPUT("AIN3"), + SND_SOC_DAPM_INPUT("AIN4"), + + SND_SOC_DAPM_OUTPUT("VREF"), +}; + +static const struct snd_soc_dapm_route adau1977_dapm_routes[] = { + { "ADC1", NULL, "AIN1" }, + { "ADC2", NULL, "AIN2" }, + { "ADC3", NULL, "AIN3" }, + { "ADC4", NULL, "AIN4" }, + + { "ADC1", NULL, "Vref" }, + { "ADC2", NULL, "Vref" }, + { "ADC3", NULL, "Vref" }, + { "ADC4", NULL, "Vref" }, + + { "VREF", NULL, "Vref" }, +}; + +#define ADAU1977_VOLUME(x) \ + SOC_SINGLE_TLV("ADC" #x " Capture Volume", \ + ADAU1977_REG_POST_ADC_GAIN((x) - 1), \ + 0, 255, 1, adau1977_adc_gain) + +#define ADAU1977_HPF_SWITCH(x) \ + SOC_SINGLE("ADC" #x " Highpass-Filter Capture Switch", \ + ADAU1977_REG_DC_HPF_CAL, (x) - 1, 1, 0) + +#define ADAU1977_DC_SUB_SWITCH(x) \ + SOC_SINGLE("ADC" #x " DC Substraction Capture Switch", \ + ADAU1977_REG_DC_HPF_CAL, (x) + 3, 1, 0) + +static const struct snd_kcontrol_new adau1977_snd_controls[] = { + ADAU1977_VOLUME(1), + ADAU1977_VOLUME(2), + ADAU1977_VOLUME(3), + ADAU1977_VOLUME(4), + + ADAU1977_HPF_SWITCH(1), + ADAU1977_HPF_SWITCH(2), + ADAU1977_HPF_SWITCH(3), + ADAU1977_HPF_SWITCH(4), + + ADAU1977_DC_SUB_SWITCH(1), + ADAU1977_DC_SUB_SWITCH(2), + ADAU1977_DC_SUB_SWITCH(3), + ADAU1977_DC_SUB_SWITCH(4), +}; + +static int adau1977_reset(struct adau1977 *adau1977) +{ + int ret; + + /* + * The reset bit is obviously volatile, but we need to be able to cache + * the other bits in the register, so we can't just mark the whole + * register as volatile. Since this is the only place where we'll ever + * touch the reset bit just bypass the cache for this operation. + */ + regcache_cache_bypass(adau1977->regmap, true); + ret = regmap_write(adau1977->regmap, ADAU1977_REG_POWER, + ADAU1977_POWER_RESET); + regcache_cache_bypass(adau1977->regmap, false); + if (ret) + return ret; + + return ret; +} + +/* + * Returns the appropriate setting for ths FS field in the CTRL0 register + * depending on the rate. + */ +static int adau1977_lookup_fs(unsigned int rate) +{ + if (rate >= 8000 && rate <= 12000) + return ADAU1977_SAI_CTRL0_FS_8000_12000; + else if (rate >= 16000 && rate <= 24000) + return ADAU1977_SAI_CTRL0_FS_16000_24000; + else if (rate >= 32000 && rate <= 48000) + return ADAU1977_SAI_CTRL0_FS_32000_48000; + else if (rate >= 64000 && rate <= 96000) + return ADAU1977_SAI_CTRL0_FS_64000_96000; + else if (rate >= 128000 && rate <= 192000) + return ADAU1977_SAI_CTRL0_FS_128000_192000; + else + return -EINVAL; +} + +static int adau1977_lookup_mcs(struct adau1977 *adau1977, unsigned int rate, + unsigned int fs) +{ + unsigned int mcs; + + /* + * rate = sysclk / (512 * mcs_lut[mcs]) * 2**fs + * => mcs_lut[mcs] = sysclk / (512 * rate) * 2**fs + * => mcs_lut[mcs] = sysclk / ((512 / 2**fs) * rate) + */ + + rate *= 512 >> fs; + + if (adau1977->sysclk % rate != 0) + return -EINVAL; + + mcs = adau1977->sysclk / rate; + + /* The factors configured by MCS are 1, 2, 3, 4, 6 */ + if (mcs < 1 || mcs > 6 || mcs == 5) + return -EINVAL; + + mcs = mcs - 1; + if (mcs == 5) + mcs = 4; + + return mcs; +} + +static int adau1977_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(codec); + unsigned int rate = params_rate(params); + unsigned int slot_width; + unsigned int ctrl0, ctrl0_mask; + unsigned int ctrl1; + int mcs, fs; + int ret; + + fs = adau1977_lookup_fs(rate); + if (fs < 0) + return fs; + + if (adau1977->sysclk_src == ADAU1977_SYSCLK_SRC_MCLK) { + mcs = adau1977_lookup_mcs(adau1977, rate, fs); + if (mcs < 0) + return mcs; + } else { + mcs = 0; + } + + ctrl0_mask = ADAU1977_SAI_CTRL0_FS_MASK; + ctrl0 = fs; + + if (adau1977->right_j) { + switch (params_width(params)) { + case 16: + ctrl0 |= ADAU1977_SAI_CTRL0_FMT_RJ_16BIT; + break; + case 24: + ctrl0 |= ADAU1977_SAI_CTRL0_FMT_RJ_24BIT; + break; + default: + return -EINVAL; + } + ctrl0_mask |= ADAU1977_SAI_CTRL0_FMT_MASK; + } + + if (adau1977->master) { + switch (params_width(params)) { + case 16: + ctrl1 = ADAU1977_SAI_CTRL1_DATA_WIDTH_16BIT; + slot_width = 16; + break; + case 24: + case 32: + ctrl1 = ADAU1977_SAI_CTRL1_DATA_WIDTH_24BIT; + slot_width = 32; + break; + default: + return -EINVAL; + } + + /* In TDM mode there is a fixed slot width */ + if (adau1977->slot_width) + slot_width = adau1977->slot_width; + + if (slot_width == 16) + ctrl1 |= ADAU1977_SAI_CTRL1_BCLKRATE_16; + else + ctrl1 |= ADAU1977_SAI_CTRL1_BCLKRATE_32; + + ret = regmap_update_bits(adau1977->regmap, + ADAU1977_REG_SAI_CTRL1, + ADAU1977_SAI_CTRL1_DATA_WIDTH_MASK | + ADAU1977_SAI_CTRL1_BCLKRATE_MASK, + ctrl1); + if (ret < 0) + return ret; + } + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_CTRL0, + ctrl0_mask, ctrl0); + if (ret < 0) + return ret; + + return regmap_update_bits(adau1977->regmap, ADAU1977_REG_PLL, + ADAU1977_PLL_MCS_MASK, mcs); +} + +static int adau1977_power_disable(struct adau1977 *adau1977) +{ + int ret = 0; + + if (!adau1977->enabled) + return 0; + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_POWER, + ADAU1977_POWER_PWUP, 0); + if (ret) + return ret; + + regcache_mark_dirty(adau1977->regmap); + + if (adau1977->reset_gpio) + gpiod_set_value_cansleep(adau1977->reset_gpio, 0); + + regcache_cache_only(adau1977->regmap, true); + + regulator_disable(adau1977->avdd_reg); + if (adau1977->dvdd_reg) + regulator_disable(adau1977->dvdd_reg); + + adau1977->enabled = false; + + return 0; +} + +static int adau1977_power_enable(struct adau1977 *adau1977) +{ + unsigned int val; + int ret = 0; + + if (adau1977->enabled) + return 0; + + ret = regulator_enable(adau1977->avdd_reg); + if (ret) + return ret; + + if (adau1977->dvdd_reg) { + ret = regulator_enable(adau1977->dvdd_reg); + if (ret) + goto err_disable_avdd; + } + + if (adau1977->reset_gpio) + gpiod_set_value_cansleep(adau1977->reset_gpio, 1); + + regcache_cache_only(adau1977->regmap, false); + + if (adau1977->switch_mode) + adau1977->switch_mode(adau1977->dev); + + ret = adau1977_reset(adau1977); + if (ret) + goto err_disable_dvdd; + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_POWER, + ADAU1977_POWER_PWUP, ADAU1977_POWER_PWUP); + if (ret) + goto err_disable_dvdd; + + ret = regcache_sync(adau1977->regmap); + if (ret) + goto err_disable_dvdd; + + /* + * The PLL register is not affected by the software reset. It is + * possible that the value of the register was changed to the + * default value while we were in cache only mode. In this case + * regcache_sync will skip over it and we have to manually sync + * it. + */ + ret = regmap_read(adau1977->regmap, ADAU1977_REG_PLL, &val); + if (ret) + goto err_disable_dvdd; + + if (val == 0x41) { + regcache_cache_bypass(adau1977->regmap, true); + ret = regmap_write(adau1977->regmap, ADAU1977_REG_PLL, + 0x41); + if (ret) + goto err_disable_dvdd; + regcache_cache_bypass(adau1977->regmap, false); + } + + adau1977->enabled = true; + + return ret; + +err_disable_dvdd: + if (adau1977->dvdd_reg) + regulator_disable(adau1977->dvdd_reg); +err_disable_avdd: + regulator_disable(adau1977->avdd_reg); + return ret; +} + +static int adau1977_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + break; + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) + ret = adau1977_power_enable(adau1977); + break; + case SND_SOC_BIAS_OFF: + ret = adau1977_power_disable(adau1977); + break; + } + + if (ret) + return ret; + + codec->dapm.bias_level = level; + + return 0; +} + +static int adau1977_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int width) +{ + struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int ctrl0, ctrl1, drv; + unsigned int slot[4]; + unsigned int i; + int ret; + + if (slots == 0) { + /* 0 = No fixed slot width */ + adau1977->slot_width = 0; + adau1977->max_master_fs = 192000; + return regmap_update_bits(adau1977->regmap, + ADAU1977_REG_SAI_CTRL0, ADAU1977_SAI_CTRL0_SAI_MASK, + ADAU1977_SAI_CTRL0_SAI_I2S); + } + + if (rx_mask == 0 || tx_mask != 0) + return -EINVAL; + + drv = 0; + for (i = 0; i < 4; i++) { + slot[i] = __ffs(rx_mask); + drv |= ADAU1977_SAI_OVERTEMP_DRV_C(i); + rx_mask &= ~(1 << slot[i]); + if (slot[i] >= slots) + return -EINVAL; + if (rx_mask == 0) + break; + } + + if (rx_mask != 0) + return -EINVAL; + + switch (width) { + case 16: + ctrl1 = ADAU1977_SAI_CTRL1_SLOT_WIDTH_16; + break; + case 24: + /* We can only generate 16 bit or 32 bit wide slots */ + if (adau1977->master) + return -EINVAL; + ctrl1 = ADAU1977_SAI_CTRL1_SLOT_WIDTH_24; + break; + case 32: + ctrl1 = ADAU1977_SAI_CTRL1_SLOT_WIDTH_32; + break; + default: + return -EINVAL; + } + + switch (slots) { + case 2: + ctrl0 = ADAU1977_SAI_CTRL0_SAI_TDM_2; + break; + case 4: + ctrl0 = ADAU1977_SAI_CTRL0_SAI_TDM_4; + break; + case 8: + ctrl0 = ADAU1977_SAI_CTRL0_SAI_TDM_8; + break; + case 16: + ctrl0 = ADAU1977_SAI_CTRL0_SAI_TDM_16; + break; + default: + return -EINVAL; + } + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_OVERTEMP, + ADAU1977_SAI_OVERTEMP_DRV_C(0) | + ADAU1977_SAI_OVERTEMP_DRV_C(1) | + ADAU1977_SAI_OVERTEMP_DRV_C(2) | + ADAU1977_SAI_OVERTEMP_DRV_C(3), drv); + if (ret) + return ret; + + ret = regmap_write(adau1977->regmap, ADAU1977_REG_CMAP12, + (slot[1] << ADAU1977_CHAN_MAP_SECOND_SLOT_OFFSET) | + (slot[0] << ADAU1977_CHAN_MAP_FIRST_SLOT_OFFSET)); + if (ret) + return ret; + + ret = regmap_write(adau1977->regmap, ADAU1977_REG_CMAP34, + (slot[3] << ADAU1977_CHAN_MAP_SECOND_SLOT_OFFSET) | + (slot[2] << ADAU1977_CHAN_MAP_FIRST_SLOT_OFFSET)); + if (ret) + return ret; + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_CTRL0, + ADAU1977_SAI_CTRL0_SAI_MASK, ctrl0); + if (ret) + return ret; + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_CTRL1, + ADAU1977_SAI_CTRL1_SLOT_WIDTH_MASK, ctrl1); + if (ret) + return ret; + + adau1977->slot_width = width; + + /* In master mode the maximum bitclock is 24.576 MHz */ + adau1977->max_master_fs = min(192000, 24576000 / width / slots); + + return 0; +} + +static int adau1977_mute(struct snd_soc_dai *dai, int mute, int stream) +{ + struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int val; + + if (mute) + val = ADAU1977_MISC_CONTROL_MMUTE; + else + val = 0; + + return regmap_update_bits(adau1977->regmap, ADAU1977_REG_MISC_CONTROL, + ADAU1977_MISC_CONTROL_MMUTE, val); +} + +static int adau1977_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int ctrl0 = 0, ctrl1 = 0, block_power = 0; + bool invert_lrclk; + int ret; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + adau1977->master = false; + break; + case SND_SOC_DAIFMT_CBM_CFM: + ctrl1 |= ADAU1977_SAI_CTRL1_MASTER; + adau1977->master = true; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + invert_lrclk = false; + break; + case SND_SOC_DAIFMT_IB_NF: + block_power |= ADAU1977_BLOCK_POWER_SAI_BCLK_EDGE; + invert_lrclk = false; + break; + case SND_SOC_DAIFMT_NB_IF: + invert_lrclk = true; + break; + case SND_SOC_DAIFMT_IB_IF: + block_power |= ADAU1977_BLOCK_POWER_SAI_BCLK_EDGE; + invert_lrclk = true; + break; + default: + return -EINVAL; + } + + adau1977->right_j = false; + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl0 |= ADAU1977_SAI_CTRL0_FMT_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl0 |= ADAU1977_SAI_CTRL0_FMT_LJ; + invert_lrclk = !invert_lrclk; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl0 |= ADAU1977_SAI_CTRL0_FMT_RJ_24BIT; + adau1977->right_j = true; + invert_lrclk = !invert_lrclk; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl1 |= ADAU1977_SAI_CTRL1_LRCLK_PULSE; + ctrl0 |= ADAU1977_SAI_CTRL0_FMT_I2S; + invert_lrclk = false; + break; + case SND_SOC_DAIFMT_DSP_B: + ctrl1 |= ADAU1977_SAI_CTRL1_LRCLK_PULSE; + ctrl0 |= ADAU1977_SAI_CTRL0_FMT_LJ; + invert_lrclk = false; + break; + default: + return -EINVAL; + } + + if (invert_lrclk) + block_power |= ADAU1977_BLOCK_POWER_SAI_LR_POL; + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_BLOCK_POWER_SAI, + ADAU1977_BLOCK_POWER_SAI_LR_POL | + ADAU1977_BLOCK_POWER_SAI_BCLK_EDGE, block_power); + if (ret) + return ret; + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_CTRL0, + ADAU1977_SAI_CTRL0_FMT_MASK, + ctrl0); + if (ret) + return ret; + + return regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_CTRL1, + ADAU1977_SAI_CTRL1_MASTER | ADAU1977_SAI_CTRL1_LRCLK_PULSE, + ctrl1); +} + +static int adau1977_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(dai->codec); + u64 formats = 0; + + if (adau1977->slot_width == 16) + formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE; + else if (adau1977->right_j || adau1977->slot_width == 24) + formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S24_BE; + + snd_pcm_hw_constraint_list(substream->runtime, 0, + SNDRV_PCM_HW_PARAM_RATE, &adau1977->constraints); + + if (adau1977->master) + snd_pcm_hw_constraint_minmax(substream->runtime, + SNDRV_PCM_HW_PARAM_RATE, 8000, adau1977->max_master_fs); + + if (formats != 0) + snd_pcm_hw_constraint_mask64(substream->runtime, + SNDRV_PCM_HW_PARAM_FORMAT, formats); + + return 0; +} + +static int adau1977_set_tristate(struct snd_soc_dai *dai, int tristate) +{ + struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(dai->codec); + unsigned int val; + + if (tristate) + val = ADAU1977_SAI_OVERTEMP_DRV_HIZ; + else + val = 0; + + return regmap_update_bits(adau1977->regmap, ADAU1977_REG_SAI_OVERTEMP, + ADAU1977_SAI_OVERTEMP_DRV_HIZ, val); +} + +static const struct snd_soc_dai_ops adau1977_dai_ops = { + .startup = adau1977_startup, + .hw_params = adau1977_hw_params, + .mute_stream = adau1977_mute, + .set_fmt = adau1977_set_dai_fmt, + .set_tdm_slot = adau1977_set_tdm_slot, + .set_tristate = adau1977_set_tristate, +}; + +static struct snd_soc_dai_driver adau1977_dai = { + .name = "adau1977-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 4, + .rates = SNDRV_PCM_RATE_KNOT, + .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .sig_bits = 24, + }, + .ops = &adau1977_dai_ops, +}; + +static const unsigned int adau1977_rates[] = { + 8000, 16000, 32000, 64000, 128000, + 11025, 22050, 44100, 88200, 172400, + 12000, 24000, 48000, 96000, 192000, +}; + +#define ADAU1977_RATE_CONSTRAINT_MASK_32000 0x001f +#define ADAU1977_RATE_CONSTRAINT_MASK_44100 0x03e0 +#define ADAU1977_RATE_CONSTRAINT_MASK_48000 0x7c00 +/* All rates >= 32000 */ +#define ADAU1977_RATE_CONSTRAINT_MASK_LRCLK 0x739c + +static bool adau1977_check_sysclk(unsigned int mclk, unsigned int base_freq) +{ + unsigned int mcs; + + if (mclk % (base_freq * 128) != 0) + return false; + + mcs = mclk / (128 * base_freq); + if (mcs < 1 || mcs > 6 || mcs == 5) + return false; + + return true; +} + +static int adau1977_set_sysclk(struct snd_soc_codec *codec, + int clk_id, int source, unsigned int freq, int dir) +{ + struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(codec); + unsigned int mask = 0; + unsigned int clk_src; + unsigned int ret; + + if (dir != SND_SOC_CLOCK_IN) + return -EINVAL; + + if (clk_id != ADAU1977_SYSCLK) + return -EINVAL; + + switch (source) { + case ADAU1977_SYSCLK_SRC_MCLK: + clk_src = 0; + break; + case ADAU1977_SYSCLK_SRC_LRCLK: + clk_src = ADAU1977_PLL_CLK_S; + break; + default: + return -EINVAL; + } + + if (freq != 0 && source == ADAU1977_SYSCLK_SRC_MCLK) { + if (freq < 4000000 || freq > 36864000) + return -EINVAL; + + if (adau1977_check_sysclk(freq, 32000)) + mask |= ADAU1977_RATE_CONSTRAINT_MASK_32000; + if (adau1977_check_sysclk(freq, 44100)) + mask |= ADAU1977_RATE_CONSTRAINT_MASK_44100; + if (adau1977_check_sysclk(freq, 48000)) + mask |= ADAU1977_RATE_CONSTRAINT_MASK_48000; + + if (mask == 0) + return -EINVAL; + } else if (source == ADAU1977_SYSCLK_SRC_LRCLK) { + mask = ADAU1977_RATE_CONSTRAINT_MASK_LRCLK; + } + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_PLL, + ADAU1977_PLL_CLK_S, clk_src); + if (ret) + return ret; + + adau1977->constraints.mask = mask; + adau1977->sysclk_src = source; + adau1977->sysclk = freq; + + return 0; +} + +static int adau1977_codec_probe(struct snd_soc_codec *codec) +{ + struct adau1977 *adau1977 = snd_soc_codec_get_drvdata(codec); + int ret; + + switch (adau1977->type) { + case ADAU1977: + ret = snd_soc_dapm_new_controls(&codec->dapm, + adau1977_micbias_dapm_widgets, + ARRAY_SIZE(adau1977_micbias_dapm_widgets)); + if (ret < 0) + return ret; + break; + default: + break; + } + + return 0; +} + +static struct snd_soc_codec_driver adau1977_codec_driver = { + .probe = adau1977_codec_probe, + .set_bias_level = adau1977_set_bias_level, + .set_sysclk = adau1977_set_sysclk, + .idle_bias_off = true, + + .controls = adau1977_snd_controls, + .num_controls = ARRAY_SIZE(adau1977_snd_controls), + .dapm_widgets = adau1977_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(adau1977_dapm_widgets), + .dapm_routes = adau1977_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(adau1977_dapm_routes), +}; + +static int adau1977_setup_micbias(struct adau1977 *adau1977) +{ + struct adau1977_platform_data *pdata = adau1977->dev->platform_data; + unsigned int micbias; + + if (pdata) { + micbias = pdata->micbias; + if (micbias > ADAU1977_MICBIAS_9V0) + return -EINVAL; + + } else { + micbias = ADAU1977_MICBIAS_8V5; + } + + return regmap_update_bits(adau1977->regmap, ADAU1977_REG_MICBIAS, + ADAU1977_MICBIAS_MB_VOLTS_MASK, + micbias << ADAU1977_MICBIAS_MB_VOLTS_OFFSET); +} + +int adau1977_probe(struct device *dev, struct regmap *regmap, + enum adau1977_type type, void (*switch_mode)(struct device *dev)) +{ + unsigned int power_off_mask; + struct adau1977 *adau1977; + int ret; + + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + adau1977 = devm_kzalloc(dev, sizeof(*adau1977), GFP_KERNEL); + if (adau1977 == NULL) + return -ENOMEM; + + adau1977->dev = dev; + adau1977->type = type; + adau1977->regmap = regmap; + adau1977->switch_mode = switch_mode; + adau1977->max_master_fs = 192000; + + adau1977->constraints.list = adau1977_rates; + adau1977->constraints.count = ARRAY_SIZE(adau1977_rates); + + adau1977->avdd_reg = devm_regulator_get(dev, "AVDD"); + if (IS_ERR(adau1977->avdd_reg)) + return PTR_ERR(adau1977->avdd_reg); + + adau1977->dvdd_reg = devm_regulator_get_optional(dev, "DVDD"); + if (IS_ERR(adau1977->dvdd_reg)) { + if (PTR_ERR(adau1977->dvdd_reg) != -ENODEV) + return PTR_ERR(adau1977->dvdd_reg); + adau1977->dvdd_reg = NULL; + } + + adau1977->reset_gpio = devm_gpiod_get(dev, "reset"); + if (IS_ERR(adau1977->reset_gpio)) { + ret = PTR_ERR(adau1977->reset_gpio); + if (ret != -ENOENT && ret != -ENOSYS) + return PTR_ERR(adau1977->reset_gpio); + adau1977->reset_gpio = NULL; + } + + dev_set_drvdata(dev, adau1977); + + if (adau1977->reset_gpio) { + ret = gpiod_direction_output(adau1977->reset_gpio, 0); + if (ret) + return ret; + ndelay(100); + } + + ret = adau1977_power_enable(adau1977); + if (ret) + return ret; + + if (type == ADAU1977) { + ret = adau1977_setup_micbias(adau1977); + if (ret) + goto err_poweroff; + } + + if (adau1977->dvdd_reg) + power_off_mask = ~0; + else + power_off_mask = ~ADAU1977_BLOCK_POWER_SAI_LDO_EN; + + ret = regmap_update_bits(adau1977->regmap, ADAU1977_REG_BLOCK_POWER_SAI, + power_off_mask, 0x00); + if (ret) + goto err_poweroff; + + ret = adau1977_power_disable(adau1977); + if (ret) + return ret; + + return snd_soc_register_codec(dev, &adau1977_codec_driver, + &adau1977_dai, 1); + +err_poweroff: + adau1977_power_disable(adau1977); + return ret; + +} +EXPORT_SYMBOL_GPL(adau1977_probe); + +static bool adau1977_register_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case ADAU1977_REG_STATUS(0): + case ADAU1977_REG_STATUS(1): + case ADAU1977_REG_STATUS(2): + case ADAU1977_REG_STATUS(3): + case ADAU1977_REG_ADC_CLIP: + return true; + } + + return false; +} + +const struct regmap_config adau1977_regmap_config = { + .max_register = ADAU1977_REG_DC_HPF_CAL, + .volatile_reg = adau1977_register_volatile, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = adau1977_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(adau1977_reg_defaults), +}; +EXPORT_SYMBOL_GPL(adau1977_regmap_config); + +MODULE_DESCRIPTION("ASoC ADAU1977/ADAU1978/ADAU1979 driver"); +MODULE_AUTHOR("Lars-Peter Clausen "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/adau1977.h b/sound/soc/codecs/adau1977.h new file mode 100644 index 000000000000..95e714345a86 --- /dev/null +++ b/sound/soc/codecs/adau1977.h @@ -0,0 +1,37 @@ +/* + * ADAU1977/ADAU1978/ADAU1979 driver + * + * Copyright 2014 Analog Devices Inc. + * Author: Lars-Peter Clausen + * + * Licensed under the GPL-2. + */ + +#ifndef __SOUND_SOC_CODECS_ADAU1977_H__ +#define __SOUND_SOC_CODECS_ADAU1977_H__ + +#include + +struct device; + +enum adau1977_type { + ADAU1977, + ADAU1978, + ADAU1979, +}; + +int adau1977_probe(struct device *dev, struct regmap *regmap, + enum adau1977_type type, void (*switch_mode)(struct device *dev)); + +extern const struct regmap_config adau1977_regmap_config; + +enum adau1977_clk_id { + ADAU1977_SYSCLK, +}; + +enum adau1977_sysclk_src { + ADAU1977_SYSCLK_SRC_MCLK, + ADAU1977_SYSCLK_SRC_LRCLK, +}; + +#endif -- cgit v1.2.3 From d6cf89ee07cbfd980f189cc12ae924c811b00ee4 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 19 Feb 2014 14:05:54 +0100 Subject: ASoC: cs4271: claim reset GPIO in bus probe function Move the GPIO acquisition from the codec to the bus probe functions. Signed-off-by: Daniel Mack Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs4271.c | 60 +++++++++++++++++++++++++++++++---------------- 1 file changed, 40 insertions(+), 20 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index f7bbe6fdba67..96c309777208 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -539,14 +539,10 @@ static int cs4271_probe(struct snd_soc_codec *codec) struct cs4271_private *cs4271 = snd_soc_codec_get_drvdata(codec); struct cs4271_platform_data *cs4271plat = codec->dev->platform_data; int ret; - int gpio_nreset = -EINVAL; bool amutec_eq_bmutec = false; #ifdef CONFIG_OF if (of_match_device(cs4271_dt_ids, codec->dev)) { - gpio_nreset = of_get_named_gpio(codec->dev->of_node, - "reset-gpio", 0); - if (of_get_property(codec->dev->of_node, "cirrus,amutec-eq-bmutec", NULL)) amutec_eq_bmutec = true; @@ -558,27 +554,19 @@ static int cs4271_probe(struct snd_soc_codec *codec) #endif if (cs4271plat) { - if (gpio_is_valid(cs4271plat->gpio_nreset)) - gpio_nreset = cs4271plat->gpio_nreset; - amutec_eq_bmutec = cs4271plat->amutec_eq_bmutec; cs4271->enable_soft_reset = cs4271plat->enable_soft_reset; } - if (gpio_nreset >= 0) - if (devm_gpio_request(codec->dev, gpio_nreset, "CS4271 Reset")) - gpio_nreset = -EINVAL; - if (gpio_nreset >= 0) { + if (gpio_is_valid(cs4271->gpio_nreset)) { /* Reset codec */ - gpio_direction_output(gpio_nreset, 0); + gpio_direction_output(cs4271->gpio_nreset, 0); udelay(1); - gpio_set_value(gpio_nreset, 1); + gpio_set_value(cs4271->gpio_nreset, 1); /* Give the codec time to wake up */ udelay(1); } - cs4271->gpio_nreset = gpio_nreset; - ret = regmap_update_bits(cs4271->regmap, CS4271_MODE2, CS4271_MODE2_PDN | CS4271_MODE2_CPEN, CS4271_MODE2_PDN | CS4271_MODE2_CPEN); @@ -640,14 +628,45 @@ static const struct regmap_config cs4271_spi_regmap = { .volatile_reg = cs4271_volatile_reg, }; -static int cs4271_spi_probe(struct spi_device *spi) +static int cs4271_common_probe(struct device *dev, + struct cs4271_private **c) { + struct cs4271_platform_data *cs4271plat = dev->platform_data; struct cs4271_private *cs4271; - cs4271 = devm_kzalloc(&spi->dev, sizeof(*cs4271), GFP_KERNEL); + cs4271 = devm_kzalloc(dev, sizeof(*cs4271), GFP_KERNEL); if (!cs4271) return -ENOMEM; + if (of_match_device(cs4271_dt_ids, dev)) + cs4271->gpio_nreset = + of_get_named_gpio(dev->of_node, "reset-gpio", 0); + + if (cs4271plat) + cs4271->gpio_nreset = cs4271plat->gpio_nreset; + + if (gpio_is_valid(cs4271->gpio_nreset)) { + int ret; + + ret = devm_gpio_request(dev, cs4271->gpio_nreset, + "CS4271 Reset"); + if (ret < 0) + return ret; + } + + *c = cs4271; + return 0; +} + +static int cs4271_spi_probe(struct spi_device *spi) +{ + struct cs4271_private *cs4271; + int ret; + + ret = cs4271_common_probe(&spi->dev, &cs4271); + if (ret < 0) + return ret; + spi_set_drvdata(spi, cs4271); cs4271->regmap = devm_regmap_init_spi(spi, &cs4271_spi_regmap); if (IS_ERR(cs4271->regmap)) @@ -697,10 +716,11 @@ static int cs4271_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { struct cs4271_private *cs4271; + int ret; - cs4271 = devm_kzalloc(&client->dev, sizeof(*cs4271), GFP_KERNEL); - if (!cs4271) - return -ENOMEM; + ret = cs4271_common_probe(&client->dev, &cs4271); + if (ret < 0) + return ret; i2c_set_clientdata(client, cs4271); cs4271->regmap = devm_regmap_init_i2c(client, &cs4271_i2c_regmap); -- cgit v1.2.3 From 7ec02609739d82f7786e5a169e5a900dbaf0d1a1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 08:59:39 +0100 Subject: ASoC: da732x: Remove superfluous DA732X_SOC_ENUM_DOUBLE_R() It's nowhere used. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/da732x.h | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/da732x.h b/sound/soc/codecs/da732x.h index c8ce5475de22..1dceafeec415 100644 --- a/sound/soc/codecs/da732x.h +++ b/sound/soc/codecs/da732x.h @@ -113,9 +113,6 @@ #define DA732X_EQ_OVERALL_VOL_DB_MIN -1800 #define DA732X_EQ_OVERALL_VOL_DB_INC 600 -#define DA732X_SOC_ENUM_DOUBLE_R(xreg, xrreg, xmax, xtext) \ - {.reg = xreg, .reg2 = xrreg, .max = xmax, .texts = xtext} - enum da732x_sysctl { DA732X_SR_8KHZ = 0x1, DA732X_SR_11_025KHZ = 0x2, -- cgit v1.2.3 From b38fbe30739afa83695c4969c8e966d919d828b0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 09:17:44 +0100 Subject: ASoC: ssm2602: Omit superfluous elements in input select array The array contains too many elements although it should have only two. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2602.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 49d28eaa6d73..12947096897c 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -63,15 +63,16 @@ static const u16 ssm2602_reg[SSM2602_CACHEREGNUM] = { /*Appending several "None"s just for OSS mixer use*/ static const char *ssm2602_input_select[] = { - "Line", "Mic", "None", "None", "None", - "None", "None", "None", + "Line", "Mic", }; static const char *ssm2602_deemph[] = {"None", "32Khz", "44.1Khz", "48Khz"}; static const struct soc_enum ssm2602_enum[] = { - SOC_ENUM_SINGLE(SSM2602_APANA, 2, 2, ssm2602_input_select), - SOC_ENUM_SINGLE(SSM2602_APDIGI, 1, 4, ssm2602_deemph), + SOC_ENUM_SINGLE(SSM2602_APANA, 2, ARRAY_SIZE(ssm2602_input_select), + ssm2602_input_select), + SOC_ENUM_SINGLE(SSM2602_APDIGI, 1, ARRAY_SIZE(ssm2602_deemph), + ssm2602_deemph), }; static const unsigned int ssm260x_outmix_tlv[] = { -- cgit v1.2.3 From 830b501138ab50dd413143dce47d1ae6dd1e39a5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 09:43:49 +0100 Subject: ASoC: wm8990: Fix the wrong number of enum items wm8990 codec driver has a few places wrongly defining the number of enum items. Use SOC_ENUM_SINGLE_DECL() macro and they are automatically fixed. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Charles Keepax Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8990.c | 41 +++++++++++++++++++---------------------- 1 file changed, 19 insertions(+), 22 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 0ccd4d8d043b..33f53ab1e7b0 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -157,26 +157,23 @@ static int wm899x_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol, static const char *wm8990_digital_sidetone[] = {"None", "Left ADC", "Right ADC", "Reserved"}; -static const struct soc_enum wm8990_left_digital_sidetone_enum = -SOC_ENUM_SINGLE(WM8990_DIGITAL_SIDE_TONE, - WM8990_ADC_TO_DACL_SHIFT, - WM8990_ADC_TO_DACL_MASK, - wm8990_digital_sidetone); - -static const struct soc_enum wm8990_right_digital_sidetone_enum = -SOC_ENUM_SINGLE(WM8990_DIGITAL_SIDE_TONE, - WM8990_ADC_TO_DACR_SHIFT, - WM8990_ADC_TO_DACR_MASK, - wm8990_digital_sidetone); +static SOC_ENUM_SINGLE_DECL(wm8990_left_digital_sidetone_enum, + WM8990_DIGITAL_SIDE_TONE, + WM8990_ADC_TO_DACL_SHIFT, + wm8990_digital_sidetone); + +static SOC_ENUM_SINGLE_DECL(wm8990_right_digital_sidetone_enum, + WM8990_DIGITAL_SIDE_TONE, + WM8990_ADC_TO_DACR_SHIFT, + wm8990_digital_sidetone); static const char *wm8990_adcmode[] = {"Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3"}; -static const struct soc_enum wm8990_right_adcmode_enum = -SOC_ENUM_SINGLE(WM8990_ADC_CTRL, - WM8990_ADC_HPF_CUT_SHIFT, - WM8990_ADC_HPF_CUT_MASK, - wm8990_adcmode); +static SOC_ENUM_SINGLE_DECL(wm8990_right_adcmode_enum, + WM8990_ADC_CTRL, + WM8990_ADC_HPF_CUT_SHIFT, + wm8990_adcmode); static const struct snd_kcontrol_new wm8990_snd_controls[] = { /* INMIXL */ @@ -475,9 +472,9 @@ SOC_DAPM_SINGLE("RINPGA34 Switch", WM8990_INPUT_MIXER3, WM8990_L34MNB_BIT, static const char *wm8990_ainlmux[] = {"INMIXL Mix", "RXVOICE Mix", "DIFFINL Mix"}; -static const struct soc_enum wm8990_ainlmux_enum = -SOC_ENUM_SINGLE(WM8990_INPUT_MIXER1, WM8990_AINLMODE_SHIFT, - ARRAY_SIZE(wm8990_ainlmux), wm8990_ainlmux); +static SOC_ENUM_SINGLE_DECL(wm8990_ainlmux_enum, + WM8990_INPUT_MIXER1, WM8990_AINLMODE_SHIFT, + wm8990_ainlmux); static const struct snd_kcontrol_new wm8990_dapm_ainlmux_controls = SOC_DAPM_ENUM("Route", wm8990_ainlmux_enum); @@ -488,9 +485,9 @@ SOC_DAPM_ENUM("Route", wm8990_ainlmux_enum); static const char *wm8990_ainrmux[] = {"INMIXR Mix", "RXVOICE Mix", "DIFFINR Mix"}; -static const struct soc_enum wm8990_ainrmux_enum = -SOC_ENUM_SINGLE(WM8990_INPUT_MIXER1, WM8990_AINRMODE_SHIFT, - ARRAY_SIZE(wm8990_ainrmux), wm8990_ainrmux); +static SOC_ENUM_SINGLE_DECL(wm8990_ainrmux_enum, + WM8990_INPUT_MIXER1, WM8990_AINRMODE_SHIFT, + wm8990_ainrmux); static const struct snd_kcontrol_new wm8990_dapm_ainrmux_controls = SOC_DAPM_ENUM("Route", wm8990_ainrmux_enum); -- cgit v1.2.3 From d07338b0f2e1697c99fcd52bf1895227ce5e1a14 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 09:44:23 +0100 Subject: ASoC: wm8991: Fix the wrong number of enum items wm8991 codec driver has a few places wrongly defining the number of enum items. Use SOC_ENUM_SINGLE_DECL() macro and they are automatically fixed. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Charles Keepax Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8991.c | 41 +++++++++++++++++++---------------------- 1 file changed, 19 insertions(+), 22 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 244eb09ffa43..32d219570cca 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -171,26 +171,23 @@ static int wm899x_outpga_put_volsw_vu(struct snd_kcontrol *kcontrol, static const char *wm8991_digital_sidetone[] = {"None", "Left ADC", "Right ADC", "Reserved"}; -static const struct soc_enum wm8991_left_digital_sidetone_enum = - SOC_ENUM_SINGLE(WM8991_DIGITAL_SIDE_TONE, - WM8991_ADC_TO_DACL_SHIFT, - WM8991_ADC_TO_DACL_MASK, - wm8991_digital_sidetone); - -static const struct soc_enum wm8991_right_digital_sidetone_enum = - SOC_ENUM_SINGLE(WM8991_DIGITAL_SIDE_TONE, - WM8991_ADC_TO_DACR_SHIFT, - WM8991_ADC_TO_DACR_MASK, - wm8991_digital_sidetone); +static SOC_ENUM_SINGLE_DECL(wm8991_left_digital_sidetone_enum, + WM8991_DIGITAL_SIDE_TONE, + WM8991_ADC_TO_DACL_SHIFT, + wm8991_digital_sidetone); + +static SOC_ENUM_SINGLE_DECL(wm8991_right_digital_sidetone_enum, + WM8991_DIGITAL_SIDE_TONE, + WM8991_ADC_TO_DACR_SHIFT, + wm8991_digital_sidetone); static const char *wm8991_adcmode[] = {"Hi-fi mode", "Voice mode 1", "Voice mode 2", "Voice mode 3"}; -static const struct soc_enum wm8991_right_adcmode_enum = - SOC_ENUM_SINGLE(WM8991_ADC_CTRL, - WM8991_ADC_HPF_CUT_SHIFT, - WM8991_ADC_HPF_CUT_MASK, - wm8991_adcmode); +static SOC_ENUM_SINGLE_DECL(wm8991_right_adcmode_enum, + WM8991_ADC_CTRL, + WM8991_ADC_HPF_CUT_SHIFT, + wm8991_adcmode); static const struct snd_kcontrol_new wm8991_snd_controls[] = { /* INMIXL */ @@ -486,9 +483,9 @@ static const struct snd_kcontrol_new wm8991_dapm_inmixr_controls[] = { static const char *wm8991_ainlmux[] = {"INMIXL Mix", "RXVOICE Mix", "DIFFINL Mix"}; -static const struct soc_enum wm8991_ainlmux_enum = - SOC_ENUM_SINGLE(WM8991_INPUT_MIXER1, WM8991_AINLMODE_SHIFT, - ARRAY_SIZE(wm8991_ainlmux), wm8991_ainlmux); +static SOC_ENUM_SINGLE_DECL(wm8991_ainlmux_enum, + WM8991_INPUT_MIXER1, WM8991_AINLMODE_SHIFT, + wm8991_ainlmux); static const struct snd_kcontrol_new wm8991_dapm_ainlmux_controls = SOC_DAPM_ENUM("Route", wm8991_ainlmux_enum); @@ -499,9 +496,9 @@ static const struct snd_kcontrol_new wm8991_dapm_ainlmux_controls = static const char *wm8991_ainrmux[] = {"INMIXR Mix", "RXVOICE Mix", "DIFFINR Mix"}; -static const struct soc_enum wm8991_ainrmux_enum = - SOC_ENUM_SINGLE(WM8991_INPUT_MIXER1, WM8991_AINRMODE_SHIFT, - ARRAY_SIZE(wm8991_ainrmux), wm8991_ainrmux); +static SOC_ENUM_SINGLE_DECL(wm8991_ainrmux_enum, + WM8991_INPUT_MIXER1, WM8991_AINRMODE_SHIFT, + wm8991_ainrmux); static const struct snd_kcontrol_new wm8991_dapm_ainrmux_controls = SOC_DAPM_ENUM("Route", wm8991_ainrmux_enum); -- cgit v1.2.3 From b6592d88ec37440c88cc3bc2c9c08a61d0de3eec Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 09:55:27 +0100 Subject: ASoC: ad193x: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ad193x.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index f644a34a28de..9381a767e75f 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -31,8 +31,8 @@ struct ad193x_priv { */ static const char * const ad193x_deemp[] = {"None", "48kHz", "44.1kHz", "32kHz"}; -static const struct soc_enum ad193x_deemp_enum = - SOC_ENUM_SINGLE(AD193X_DAC_CTRL2, 1, 4, ad193x_deemp); +static SOC_ENUM_SINGLE_DECL(ad193x_deemp_enum, AD193X_DAC_CTRL2, 1, + ad193x_deemp); static const DECLARE_TLV_DB_MINMAX(adau193x_tlv, -9563, 0); -- cgit v1.2.3 From 9a8d38db030f016bee45b927af02d9b46398ed46 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 08:11:42 +0100 Subject: ASoC: Rename soc_enum.max field with items The name "max" in struct soc_enum is rather confusing since it actually takes the number of items. With "max", one might try to assign (nitems - 1) value. Rename the field to a more appropriate one, "items", which is also used in struct snd_ctl_elem_info, too. This patch also rewrites some code like "if (x > e->nitems - 1)" with "if (x >= e->nitems)". Not only the latter improves the readability, it also fixes a potential bug when e->items is zero. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 24 ++++++++++++------------ sound/soc/codecs/max98088.c | 2 +- sound/soc/codecs/max98095.c | 4 ++-- sound/soc/codecs/twl4030.c | 4 ++-- sound/soc/codecs/wm8904.c | 4 ++-- sound/soc/codecs/wm8958-dsp2.c | 8 ++++---- sound/soc/codecs/wm8994.c | 4 ++-- sound/soc/codecs/wm8996.c | 2 +- sound/soc/omap/ams-delta.c | 2 +- sound/soc/soc-core.c | 18 +++++++++--------- sound/soc/soc-dapm.c | 22 +++++++++++----------- 11 files changed, 47 insertions(+), 47 deletions(-) (limited to 'sound/soc/codecs') diff --git a/include/sound/soc.h b/include/sound/soc.h index 9a001472b96a..66de6a70581e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -162,19 +162,19 @@ .private_value = (unsigned long)&(struct soc_mixer_control) \ {.reg = xreg, .min = xmin, .max = xmax, \ .platform_max = xmax} } -#define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmax, xtexts) \ +#define SOC_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xitems, xtexts) \ { .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ - .max = xmax, .texts = xtexts, \ - .mask = xmax ? roundup_pow_of_two(xmax) - 1 : 0} -#define SOC_ENUM_SINGLE(xreg, xshift, xmax, xtexts) \ - SOC_ENUM_DOUBLE(xreg, xshift, xshift, xmax, xtexts) -#define SOC_ENUM_SINGLE_EXT(xmax, xtexts) \ -{ .max = xmax, .texts = xtexts } -#define SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xmax, xtexts, xvalues) \ + .items = xitems, .texts = xtexts, \ + .mask = xitems ? roundup_pow_of_two(xitems) - 1 : 0} +#define SOC_ENUM_SINGLE(xreg, xshift, xitems, xtexts) \ + SOC_ENUM_DOUBLE(xreg, xshift, xshift, xitems, xtexts) +#define SOC_ENUM_SINGLE_EXT(xitems, xtexts) \ +{ .items = xitems, .texts = xtexts } +#define SOC_VALUE_ENUM_DOUBLE(xreg, xshift_l, xshift_r, xmask, xitems, xtexts, xvalues) \ { .reg = xreg, .shift_l = xshift_l, .shift_r = xshift_r, \ - .mask = xmask, .max = xmax, .texts = xtexts, .values = xvalues} -#define SOC_VALUE_ENUM_SINGLE(xreg, xshift, xmask, xmax, xtexts, xvalues) \ - SOC_VALUE_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xmax, xtexts, xvalues) + .mask = xmask, .items = xitems, .texts = xtexts, .values = xvalues} +#define SOC_VALUE_ENUM_SINGLE(xreg, xshift, xmask, xnitmes, xtexts, xvalues) \ + SOC_VALUE_ENUM_DOUBLE(xreg, xshift, xshift, xmask, xnitmes, xtexts, xvalues) #define SOC_ENUM(xname, xenum) \ { .iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname,\ .info = snd_soc_info_enum_double, \ @@ -1089,7 +1089,7 @@ struct soc_enum { unsigned short reg2; unsigned char shift_l; unsigned char shift_r; - unsigned int max; + unsigned int items; unsigned int mask; const char * const *texts; const unsigned int *values; diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index ee660e2d3df3..bb1ecfc4459b 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1849,7 +1849,7 @@ static void max98088_handle_eq_pdata(struct snd_soc_codec *codec) /* Now point the soc_enum to .texts array items */ max98088->eq_enum.texts = max98088->eq_texts; - max98088->eq_enum.max = max98088->eq_textcnt; + max98088->eq_enum.items = max98088->eq_textcnt; ret = snd_soc_add_codec_controls(codec, controls, ARRAY_SIZE(controls)); if (ret != 0) diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 3ba1170ebb53..5bce9cde4a6d 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -1861,7 +1861,7 @@ static void max98095_handle_eq_pdata(struct snd_soc_codec *codec) /* Now point the soc_enum to .texts array items */ max98095->eq_enum.texts = max98095->eq_texts; - max98095->eq_enum.max = max98095->eq_textcnt; + max98095->eq_enum.items = max98095->eq_textcnt; ret = snd_soc_add_codec_controls(codec, controls, ARRAY_SIZE(controls)); if (ret != 0) @@ -2016,7 +2016,7 @@ static void max98095_handle_bq_pdata(struct snd_soc_codec *codec) /* Now point the soc_enum to .texts array items */ max98095->bq_enum.texts = max98095->bq_texts; - max98095->bq_enum.max = max98095->bq_textcnt; + max98095->bq_enum.items = max98095->bq_textcnt; ret = snd_soc_add_codec_controls(codec, controls, ARRAY_SIZE(controls)); if (ret != 0) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 00665ada23e2..1eb13d586309 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -975,13 +975,13 @@ static int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol, return -EBUSY; } - if (ucontrol->value.enumerated.item[0] > e->max - 1) + if (ucontrol->value.enumerated.item[0] >= e->items) return -EINVAL; val = ucontrol->value.enumerated.item[0] << e->shift_l; mask = e->mask << e->shift_l; if (e->shift_l != e->shift_r) { - if (ucontrol->value.enumerated.item[1] > e->max - 1) + if (ucontrol->value.enumerated.item[1] >= e->items) return -EINVAL; val |= ucontrol->value.enumerated.item[1] << e->shift_r; mask |= e->mask << e->shift_r; diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 53bbfac6a83a..b2664ec901b9 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -1981,7 +1981,7 @@ static void wm8904_handle_retune_mobile_pdata(struct snd_soc_codec *codec) dev_dbg(codec->dev, "Allocated %d unique ReTune Mobile names\n", wm8904->num_retune_mobile_texts); - wm8904->retune_mobile_enum.max = wm8904->num_retune_mobile_texts; + wm8904->retune_mobile_enum.items = wm8904->num_retune_mobile_texts; wm8904->retune_mobile_enum.texts = wm8904->retune_mobile_texts; ret = snd_soc_add_codec_controls(codec, &control, 1); @@ -2022,7 +2022,7 @@ static void wm8904_handle_pdata(struct snd_soc_codec *codec) for (i = 0; i < pdata->num_drc_cfgs; i++) wm8904->drc_texts[i] = pdata->drc_cfgs[i].name; - wm8904->drc_enum.max = pdata->num_drc_cfgs; + wm8904->drc_enum.items = pdata->num_drc_cfgs; wm8904->drc_enum.texts = wm8904->drc_texts; ret = snd_soc_add_codec_controls(codec, &control, 1); diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index b7488f190d2b..19743779bf4d 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -944,7 +944,7 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) for (i = 0; i < pdata->num_mbc_cfgs; i++) wm8994->mbc_texts[i] = pdata->mbc_cfgs[i].name; - wm8994->mbc_enum.max = pdata->num_mbc_cfgs; + wm8994->mbc_enum.items = pdata->num_mbc_cfgs; wm8994->mbc_enum.texts = wm8994->mbc_texts; ret = snd_soc_add_codec_controls(wm8994->hubs.codec, @@ -973,7 +973,7 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) for (i = 0; i < pdata->num_vss_cfgs; i++) wm8994->vss_texts[i] = pdata->vss_cfgs[i].name; - wm8994->vss_enum.max = pdata->num_vss_cfgs; + wm8994->vss_enum.items = pdata->num_vss_cfgs; wm8994->vss_enum.texts = wm8994->vss_texts; ret = snd_soc_add_codec_controls(wm8994->hubs.codec, @@ -1003,7 +1003,7 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) for (i = 0; i < pdata->num_vss_hpf_cfgs; i++) wm8994->vss_hpf_texts[i] = pdata->vss_hpf_cfgs[i].name; - wm8994->vss_hpf_enum.max = pdata->num_vss_hpf_cfgs; + wm8994->vss_hpf_enum.items = pdata->num_vss_hpf_cfgs; wm8994->vss_hpf_enum.texts = wm8994->vss_hpf_texts; ret = snd_soc_add_codec_controls(wm8994->hubs.codec, @@ -1034,7 +1034,7 @@ void wm8958_dsp2_init(struct snd_soc_codec *codec) for (i = 0; i < pdata->num_enh_eq_cfgs; i++) wm8994->enh_eq_texts[i] = pdata->enh_eq_cfgs[i].name; - wm8994->enh_eq_enum.max = pdata->num_enh_eq_cfgs; + wm8994->enh_eq_enum.items = pdata->num_enh_eq_cfgs; wm8994->enh_eq_enum.texts = wm8994->enh_eq_texts; ret = snd_soc_add_codec_controls(wm8994->hubs.codec, diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index b9be9cbc4603..8253c3c6db0e 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3237,7 +3237,7 @@ static void wm8994_handle_retune_mobile_pdata(struct wm8994_priv *wm8994) dev_dbg(codec->dev, "Allocated %d unique ReTune Mobile names\n", wm8994->num_retune_mobile_texts); - wm8994->retune_mobile_enum.max = wm8994->num_retune_mobile_texts; + wm8994->retune_mobile_enum.items = wm8994->num_retune_mobile_texts; wm8994->retune_mobile_enum.texts = wm8994->retune_mobile_texts; ret = snd_soc_add_codec_controls(wm8994->hubs.codec, controls, @@ -3293,7 +3293,7 @@ static void wm8994_handle_pdata(struct wm8994_priv *wm8994) for (i = 0; i < pdata->num_drc_cfgs; i++) wm8994->drc_texts[i] = pdata->drc_cfgs[i].name; - wm8994->drc_enum.max = pdata->num_drc_cfgs; + wm8994->drc_enum.items = pdata->num_drc_cfgs; wm8994->drc_enum.texts = wm8994->drc_texts; ret = snd_soc_add_codec_controls(wm8994->hubs.codec, controls, diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 1a7655b0aa22..ea6b587f65c1 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2595,7 +2595,7 @@ static void wm8996_retune_mobile_pdata(struct snd_soc_codec *codec) dev_dbg(codec->dev, "Allocated %d unique ReTune Mobile names\n", wm8996->num_retune_mobile_texts); - wm8996->retune_mobile_enum.max = wm8996->num_retune_mobile_texts; + wm8996->retune_mobile_enum.items = wm8996->num_retune_mobile_texts; wm8996->retune_mobile_enum.texts = wm8996->retune_mobile_texts; ret = snd_soc_add_codec_controls(codec, controls, ARRAY_SIZE(controls)); diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c index 629446482a91..2eca91a51f51 100644 --- a/sound/soc/omap/ams-delta.c +++ b/sound/soc/omap/ams-delta.c @@ -103,7 +103,7 @@ static int ams_delta_set_audio_mode(struct snd_kcontrol *kcontrol, if (!codec->hw_write) return -EUNATCH; - if (ucontrol->value.enumerated.item[0] >= control->max) + if (ucontrol->value.enumerated.item[0] >= control->items) return -EINVAL; mutex_lock(&codec->mutex); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index fe1df50805a3..4372efb4e033 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2571,10 +2571,10 @@ int snd_soc_info_enum_double(struct snd_kcontrol *kcontrol, uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; uinfo->count = e->shift_l == e->shift_r ? 1 : 2; - uinfo->value.enumerated.items = e->max; + uinfo->value.enumerated.items = e->items; - if (uinfo->value.enumerated.item > e->max - 1) - uinfo->value.enumerated.item = e->max - 1; + if (uinfo->value.enumerated.item >= e->items) + uinfo->value.enumerated.item = e->items - 1; strlcpy(uinfo->value.enumerated.name, e->texts[uinfo->value.enumerated.item], sizeof(uinfo->value.enumerated.name)); @@ -2626,12 +2626,12 @@ int snd_soc_put_enum_double(struct snd_kcontrol *kcontrol, unsigned int val; unsigned int mask; - if (ucontrol->value.enumerated.item[0] > e->max - 1) + if (ucontrol->value.enumerated.item[0] >= e->items) return -EINVAL; val = ucontrol->value.enumerated.item[0] << e->shift_l; mask = e->mask << e->shift_l; if (e->shift_l != e->shift_r) { - if (ucontrol->value.enumerated.item[1] > e->max - 1) + if (ucontrol->value.enumerated.item[1] >= e->items) return -EINVAL; val |= ucontrol->value.enumerated.item[1] << e->shift_r; mask |= e->mask << e->shift_r; @@ -2662,14 +2662,14 @@ int snd_soc_get_value_enum_double(struct snd_kcontrol *kcontrol, reg_val = snd_soc_read(codec, e->reg); val = (reg_val >> e->shift_l) & e->mask; - for (mux = 0; mux < e->max; mux++) { + for (mux = 0; mux < e->items; mux++) { if (val == e->values[mux]) break; } ucontrol->value.enumerated.item[0] = mux; if (e->shift_l != e->shift_r) { val = (reg_val >> e->shift_r) & e->mask; - for (mux = 0; mux < e->max; mux++) { + for (mux = 0; mux < e->items; mux++) { if (val == e->values[mux]) break; } @@ -2700,12 +2700,12 @@ int snd_soc_put_value_enum_double(struct snd_kcontrol *kcontrol, unsigned int val; unsigned int mask; - if (ucontrol->value.enumerated.item[0] > e->max - 1) + if (ucontrol->value.enumerated.item[0] >= e->items) return -EINVAL; val = e->values[ucontrol->value.enumerated.item[0]] << e->shift_l; mask = e->mask << e->shift_l; if (e->shift_l != e->shift_r) { - if (ucontrol->value.enumerated.item[1] > e->max - 1) + if (ucontrol->value.enumerated.item[1] >= e->items) return -EINVAL; val |= e->values[ucontrol->value.enumerated.item[1]] << e->shift_r; mask |= e->mask << e->shift_r; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dc8ff13187f7..2026a64a0afb 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -535,7 +535,7 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, soc_widget_read(w, e->reg, &val); item = (val >> e->shift_l) & e->mask; - if (item < e->max && !strcmp(p->name, e->texts[item])) + if (item < e->items && !strcmp(p->name, e->texts[item])) p->connect = 1; else p->connect = 0; @@ -563,12 +563,12 @@ static void dapm_set_path_status(struct snd_soc_dapm_widget *w, soc_widget_read(w, e->reg, &val); val = (val >> e->shift_l) & e->mask; - for (item = 0; item < e->max; item++) { + for (item = 0; item < e->items; item++) { if (val == e->values[item]) break; } - if (item < e->max && !strcmp(p->name, e->texts[item])) + if (item < e->items && !strcmp(p->name, e->texts[item])) p->connect = 1; else p->connect = 0; @@ -616,7 +616,7 @@ static int dapm_connect_mux(struct snd_soc_dapm_context *dapm, struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; int i; - for (i = 0; i < e->max; i++) { + for (i = 0; i < e->items; i++) { if (!(strcmp(control_name, e->texts[i]))) { list_add(&path->list, &dapm->card->paths); list_add(&path->list_sink, &dest->sources); @@ -2967,13 +2967,13 @@ int snd_soc_dapm_put_enum_double(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_update update; int ret = 0; - if (ucontrol->value.enumerated.item[0] > e->max - 1) + if (ucontrol->value.enumerated.item[0] >= e->items) return -EINVAL; mux = ucontrol->value.enumerated.item[0]; val = mux << e->shift_l; mask = e->mask << e->shift_l; if (e->shift_l != e->shift_r) { - if (ucontrol->value.enumerated.item[1] > e->max - 1) + if (ucontrol->value.enumerated.item[1] >= e->items) return -EINVAL; val |= ucontrol->value.enumerated.item[1] << e->shift_r; mask |= e->mask << e->shift_r; @@ -3036,7 +3036,7 @@ int snd_soc_dapm_put_enum_virt(struct snd_kcontrol *kcontrol, int change; int ret = 0; - if (ucontrol->value.enumerated.item[0] >= e->max) + if (ucontrol->value.enumerated.item[0] >= e->items) return -EINVAL; mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); @@ -3077,14 +3077,14 @@ int snd_soc_dapm_get_value_enum_double(struct snd_kcontrol *kcontrol, reg_val = snd_soc_read(codec, e->reg); val = (reg_val >> e->shift_l) & e->mask; - for (mux = 0; mux < e->max; mux++) { + for (mux = 0; mux < e->items; mux++) { if (val == e->values[mux]) break; } ucontrol->value.enumerated.item[0] = mux; if (e->shift_l != e->shift_r) { val = (reg_val >> e->shift_r) & e->mask; - for (mux = 0; mux < e->max; mux++) { + for (mux = 0; mux < e->items; mux++) { if (val == e->values[mux]) break; } @@ -3119,13 +3119,13 @@ int snd_soc_dapm_put_value_enum_double(struct snd_kcontrol *kcontrol, struct snd_soc_dapm_update update; int ret = 0; - if (ucontrol->value.enumerated.item[0] > e->max - 1) + if (ucontrol->value.enumerated.item[0] >= e->items) return -EINVAL; mux = ucontrol->value.enumerated.item[0]; val = e->values[ucontrol->value.enumerated.item[0]] << e->shift_l; mask = e->mask << e->shift_l; if (e->shift_l != e->shift_r) { - if (ucontrol->value.enumerated.item[1] > e->max - 1) + if (ucontrol->value.enumerated.item[1] >= e->items) return -EINVAL; val |= e->values[ucontrol->value.enumerated.item[1]] << e->shift_r; mask |= e->mask << e->shift_r; -- cgit v1.2.3 From 6b207c0f166e7f19c9d9dc48feb25e276e36c43f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 08:56:39 +0100 Subject: ASoC: twl4030: Clean up duplicated code Remove the open code in snd_soc_put_twl4030_opmode_enum_double() but just call snd_soc_put_enum_double() instead, which does the very same thing (even correctly with a lock). Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 17 +---------------- 1 file changed, 1 insertion(+), 16 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 1eb13d586309..682e4ac88939 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -965,9 +965,6 @@ static int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol, { struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); struct twl4030_priv *twl4030 = snd_soc_codec_get_drvdata(codec); - struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; - unsigned short val; - unsigned short mask; if (twl4030->configured) { dev_err(codec->dev, @@ -975,19 +972,7 @@ static int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol, return -EBUSY; } - if (ucontrol->value.enumerated.item[0] >= e->items) - return -EINVAL; - - val = ucontrol->value.enumerated.item[0] << e->shift_l; - mask = e->mask << e->shift_l; - if (e->shift_l != e->shift_r) { - if (ucontrol->value.enumerated.item[1] >= e->items) - return -EINVAL; - val |= ucontrol->value.enumerated.item[1] << e->shift_r; - mask |= e->mask << e->shift_r; - } - - return snd_soc_update_bits(codec, e->reg, mask, val); + return snd_soc_put_enum_double(kcontrol, ucontrol); } /* -- cgit v1.2.3 From c28b14f49978af2fc864dd2faeb89f517746f9f9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 08:59:39 +0100 Subject: ALSA: da732x: Remove superfluous DA732X_SOC_ENUM_DOUBLE_R() It's nowhere used. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/da732x.h | 3 --- 1 file changed, 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/da732x.h b/sound/soc/codecs/da732x.h index c8ce5475de22..1dceafeec415 100644 --- a/sound/soc/codecs/da732x.h +++ b/sound/soc/codecs/da732x.h @@ -113,9 +113,6 @@ #define DA732X_EQ_OVERALL_VOL_DB_MIN -1800 #define DA732X_EQ_OVERALL_VOL_DB_INC 600 -#define DA732X_SOC_ENUM_DOUBLE_R(xreg, xrreg, xmax, xtext) \ - {.reg = xreg, .reg2 = xrreg, .max = xmax, .texts = xtext} - enum da732x_sysctl { DA732X_SR_8KHZ = 0x1, DA732X_SR_11_025KHZ = 0x2, -- cgit v1.2.3 From 4988aff78bfd7c97cc67f0390555a9c2b6825f40 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 09:40:51 +0100 Subject: ASoC: adau1373: Remove superfluous const As SOC_ENUM_SINGLE_DECL() itself contains const modifier now, we can reduce const from its users. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adau1373.c | 28 ++++++++++++++-------------- 1 file changed, 14 insertions(+), 14 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index eb836ed5271f..46b1595d7c50 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -345,15 +345,15 @@ static const char *adau1373_fdsp_sel_text[] = { "Channel 5", }; -static const SOC_ENUM_SINGLE_DECL(adau1373_drc1_channel_enum, +static SOC_ENUM_SINGLE_DECL(adau1373_drc1_channel_enum, ADAU1373_FDSP_SEL1, 4, adau1373_fdsp_sel_text); -static const SOC_ENUM_SINGLE_DECL(adau1373_drc2_channel_enum, +static SOC_ENUM_SINGLE_DECL(adau1373_drc2_channel_enum, ADAU1373_FDSP_SEL1, 0, adau1373_fdsp_sel_text); -static const SOC_ENUM_SINGLE_DECL(adau1373_drc3_channel_enum, +static SOC_ENUM_SINGLE_DECL(adau1373_drc3_channel_enum, ADAU1373_FDSP_SEL2, 0, adau1373_fdsp_sel_text); -static const SOC_ENUM_SINGLE_DECL(adau1373_hpf_channel_enum, +static SOC_ENUM_SINGLE_DECL(adau1373_hpf_channel_enum, ADAU1373_FDSP_SEL3, 0, adau1373_fdsp_sel_text); -static const SOC_ENUM_SINGLE_DECL(adau1373_bass_channel_enum, +static SOC_ENUM_SINGLE_DECL(adau1373_bass_channel_enum, ADAU1373_FDSP_SEL4, 4, adau1373_fdsp_sel_text); static const char *adau1373_hpf_cutoff_text[] = { @@ -362,7 +362,7 @@ static const char *adau1373_hpf_cutoff_text[] = { "800Hz", }; -static const SOC_ENUM_SINGLE_DECL(adau1373_hpf_cutoff_enum, +static SOC_ENUM_SINGLE_DECL(adau1373_hpf_cutoff_enum, ADAU1373_HPF_CTRL, 3, adau1373_hpf_cutoff_text); static const char *adau1373_bass_lpf_cutoff_text[] = { @@ -388,14 +388,14 @@ static const unsigned int adau1373_bass_tlv[] = { 5, 7, TLV_DB_SCALE_ITEM(1400, 150, 0), }; -static const SOC_ENUM_SINGLE_DECL(adau1373_bass_lpf_cutoff_enum, +static SOC_ENUM_SINGLE_DECL(adau1373_bass_lpf_cutoff_enum, ADAU1373_BASS1, 5, adau1373_bass_lpf_cutoff_text); -static const SOC_VALUE_ENUM_SINGLE_DECL(adau1373_bass_clip_level_enum, +static SOC_VALUE_ENUM_SINGLE_DECL(adau1373_bass_clip_level_enum, ADAU1373_BASS1, 2, 7, adau1373_bass_clip_level_text, adau1373_bass_clip_level_values); -static const SOC_ENUM_SINGLE_DECL(adau1373_bass_hpf_cutoff_enum, +static SOC_ENUM_SINGLE_DECL(adau1373_bass_hpf_cutoff_enum, ADAU1373_BASS1, 0, adau1373_bass_hpf_cutoff_text); static const char *adau1373_3d_level_text[] = { @@ -409,9 +409,9 @@ static const char *adau1373_3d_cutoff_text[] = { "0.16875 fs", "0.27083 fs" }; -static const SOC_ENUM_SINGLE_DECL(adau1373_3d_level_enum, +static SOC_ENUM_SINGLE_DECL(adau1373_3d_level_enum, ADAU1373_3D_CTRL1, 4, adau1373_3d_level_text); -static const SOC_ENUM_SINGLE_DECL(adau1373_3d_cutoff_enum, +static SOC_ENUM_SINGLE_DECL(adau1373_3d_cutoff_enum, ADAU1373_3D_CTRL1, 0, adau1373_3d_cutoff_text); static const unsigned int adau1373_3d_tlv[] = { @@ -427,11 +427,11 @@ static const char *adau1373_lr_mux_text[] = { "Stereo", }; -static const SOC_ENUM_SINGLE_DECL(adau1373_lineout1_lr_mux_enum, +static SOC_ENUM_SINGLE_DECL(adau1373_lineout1_lr_mux_enum, ADAU1373_OUTPUT_CTRL, 4, adau1373_lr_mux_text); -static const SOC_ENUM_SINGLE_DECL(adau1373_lineout2_lr_mux_enum, +static SOC_ENUM_SINGLE_DECL(adau1373_lineout2_lr_mux_enum, ADAU1373_OUTPUT_CTRL, 6, adau1373_lr_mux_text); -static const SOC_ENUM_SINGLE_DECL(adau1373_speaker_lr_mux_enum, +static SOC_ENUM_SINGLE_DECL(adau1373_speaker_lr_mux_enum, ADAU1373_LS_CTRL, 4, adau1373_lr_mux_text); static const struct snd_kcontrol_new adau1373_controls[] = { -- cgit v1.2.3 From 51e5b59c3559e63e8a943350714a8c070fba8cbf Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 09:42:16 +0100 Subject: ASoC: lm49453: Remove superfluous const As SOC_ENUM_SINGLE_DECL() itself contains const modifier now, we can reduce const from its users. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/lm49453.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index e19490cfb3a8..6b7fe5e54881 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -195,18 +195,18 @@ struct lm49453_priv { static const char *lm49453_mic2mode_text[] = {"Single Ended", "Differential"}; -static const SOC_ENUM_SINGLE_DECL(lm49453_mic2mode_enum, LM49453_P0_MICR_REG, 5, - lm49453_mic2mode_text); +static SOC_ENUM_SINGLE_DECL(lm49453_mic2mode_enum, LM49453_P0_MICR_REG, 5, + lm49453_mic2mode_text); static const char *lm49453_dmic_cfg_text[] = {"DMICDAT1", "DMICDAT2"}; -static const SOC_ENUM_SINGLE_DECL(lm49453_dmic12_cfg_enum, - LM49453_P0_DIGITAL_MIC1_CONFIG_REG, - 7, lm49453_dmic_cfg_text); +static SOC_ENUM_SINGLE_DECL(lm49453_dmic12_cfg_enum, + LM49453_P0_DIGITAL_MIC1_CONFIG_REG, 7, + lm49453_dmic_cfg_text); -static const SOC_ENUM_SINGLE_DECL(lm49453_dmic34_cfg_enum, - LM49453_P0_DIGITAL_MIC2_CONFIG_REG, - 7, lm49453_dmic_cfg_text); +static SOC_ENUM_SINGLE_DECL(lm49453_dmic34_cfg_enum, + LM49453_P0_DIGITAL_MIC2_CONFIG_REG, 7, + lm49453_dmic_cfg_text); /* MUX Controls */ static const char *lm49453_adcl_mux_text[] = { "MIC1", "Aux_L" }; -- cgit v1.2.3 From a750987443cd4239c2b4dd742a59474aea10b179 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 09:42:50 +0100 Subject: ASoC: mc13783: Remove superfluous const As SOC_ENUM_SINGLE_DECL() itself contains const modifier now, we can reduce const from its users. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/mc13783.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 582c2bbd42cb..147c2e53797b 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -430,8 +430,8 @@ static const struct snd_kcontrol_new samp_ctl = static const char * const speaker_amp_source_text[] = { "CODEC", "Right" }; -static const SOC_ENUM_SINGLE_DECL(speaker_amp_source, MC13783_AUDIO_RX0, 4, - speaker_amp_source_text); +static SOC_ENUM_SINGLE_DECL(speaker_amp_source, MC13783_AUDIO_RX0, 4, + speaker_amp_source_text); static const struct snd_kcontrol_new speaker_amp_source_mux = SOC_DAPM_ENUM("Speaker Amp Source MUX", speaker_amp_source); @@ -439,8 +439,8 @@ static const char * const headset_amp_source_text[] = { "CODEC", "Mixer" }; -static const SOC_ENUM_SINGLE_DECL(headset_amp_source, MC13783_AUDIO_RX0, 11, - headset_amp_source_text); +static SOC_ENUM_SINGLE_DECL(headset_amp_source, MC13783_AUDIO_RX0, 11, + headset_amp_source_text); static const struct snd_kcontrol_new headset_amp_source_mux = SOC_DAPM_ENUM("Headset Amp Source MUX", headset_amp_source); -- cgit v1.2.3 From f843cdf2f767a88e7cfe9300f37e1c02b97cc389 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 09:43:08 +0100 Subject: ASoC: rt5631: Remove superfluous const As SOC_ENUM_SINGLE_DECL() itself contains const modifier now, we can reduce const from its users. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/rt5631.c | 75 +++++++++++++++++++---------------------------- 1 file changed, 30 insertions(+), 45 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 912c9cbc2724..ce199d375209 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -210,26 +210,22 @@ static int rt5631_dmic_put(struct snd_kcontrol *kcontrol, static const char *rt5631_input_mode[] = { "Single ended", "Differential"}; -static const SOC_ENUM_SINGLE_DECL( - rt5631_mic1_mode_enum, RT5631_MIC_CTRL_1, - RT5631_MIC1_DIFF_INPUT_SHIFT, rt5631_input_mode); +static SOC_ENUM_SINGLE_DECL(rt5631_mic1_mode_enum, RT5631_MIC_CTRL_1, + RT5631_MIC1_DIFF_INPUT_SHIFT, rt5631_input_mode); -static const SOC_ENUM_SINGLE_DECL( - rt5631_mic2_mode_enum, RT5631_MIC_CTRL_1, - RT5631_MIC2_DIFF_INPUT_SHIFT, rt5631_input_mode); +static SOC_ENUM_SINGLE_DECL(rt5631_mic2_mode_enum, RT5631_MIC_CTRL_1, + RT5631_MIC2_DIFF_INPUT_SHIFT, rt5631_input_mode); /* MONO Input Type */ -static const SOC_ENUM_SINGLE_DECL( - rt5631_monoin_mode_enum, RT5631_MONO_INPUT_VOL, - RT5631_MONO_DIFF_INPUT_SHIFT, rt5631_input_mode); +static SOC_ENUM_SINGLE_DECL(rt5631_monoin_mode_enum, RT5631_MONO_INPUT_VOL, + RT5631_MONO_DIFF_INPUT_SHIFT, rt5631_input_mode); /* SPK Ratio Gain Control */ static const char *rt5631_spk_ratio[] = {"1.00x", "1.09x", "1.27x", "1.44x", "1.56x", "1.68x", "1.99x", "2.34x"}; -static const SOC_ENUM_SINGLE_DECL( - rt5631_spk_ratio_enum, RT5631_GEN_PUR_CTRL_REG, - RT5631_SPK_AMP_RATIO_CTRL_SHIFT, rt5631_spk_ratio); +static SOC_ENUM_SINGLE_DECL(rt5631_spk_ratio_enum, RT5631_GEN_PUR_CTRL_REG, + RT5631_SPK_AMP_RATIO_CTRL_SHIFT, rt5631_spk_ratio); static const struct snd_kcontrol_new rt5631_snd_controls[] = { /* MIC */ @@ -759,9 +755,8 @@ static const struct snd_kcontrol_new rt5631_monomix_mixer_controls[] = { /* Left SPK Volume Input */ static const char *rt5631_spkvoll_sel[] = {"Vmid", "SPKMIXL"}; -static const SOC_ENUM_SINGLE_DECL( - rt5631_spkvoll_enum, RT5631_SPK_OUT_VOL, - RT5631_L_EN_SHIFT, rt5631_spkvoll_sel); +static SOC_ENUM_SINGLE_DECL(rt5631_spkvoll_enum, RT5631_SPK_OUT_VOL, + RT5631_L_EN_SHIFT, rt5631_spkvoll_sel); static const struct snd_kcontrol_new rt5631_spkvoll_mux_control = SOC_DAPM_ENUM("Left SPKVOL SRC", rt5631_spkvoll_enum); @@ -769,9 +764,8 @@ static const struct snd_kcontrol_new rt5631_spkvoll_mux_control = /* Left HP Volume Input */ static const char *rt5631_hpvoll_sel[] = {"Vmid", "OUTMIXL"}; -static const SOC_ENUM_SINGLE_DECL( - rt5631_hpvoll_enum, RT5631_HP_OUT_VOL, - RT5631_L_EN_SHIFT, rt5631_hpvoll_sel); +static SOC_ENUM_SINGLE_DECL(rt5631_hpvoll_enum, RT5631_HP_OUT_VOL, + RT5631_L_EN_SHIFT, rt5631_hpvoll_sel); static const struct snd_kcontrol_new rt5631_hpvoll_mux_control = SOC_DAPM_ENUM("Left HPVOL SRC", rt5631_hpvoll_enum); @@ -779,9 +773,8 @@ static const struct snd_kcontrol_new rt5631_hpvoll_mux_control = /* Left Out Volume Input */ static const char *rt5631_outvoll_sel[] = {"Vmid", "OUTMIXL"}; -static const SOC_ENUM_SINGLE_DECL( - rt5631_outvoll_enum, RT5631_MONO_AXO_1_2_VOL, - RT5631_L_EN_SHIFT, rt5631_outvoll_sel); +static SOC_ENUM_SINGLE_DECL(rt5631_outvoll_enum, RT5631_MONO_AXO_1_2_VOL, + RT5631_L_EN_SHIFT, rt5631_outvoll_sel); static const struct snd_kcontrol_new rt5631_outvoll_mux_control = SOC_DAPM_ENUM("Left OUTVOL SRC", rt5631_outvoll_enum); @@ -789,9 +782,8 @@ static const struct snd_kcontrol_new rt5631_outvoll_mux_control = /* Right Out Volume Input */ static const char *rt5631_outvolr_sel[] = {"Vmid", "OUTMIXR"}; -static const SOC_ENUM_SINGLE_DECL( - rt5631_outvolr_enum, RT5631_MONO_AXO_1_2_VOL, - RT5631_R_EN_SHIFT, rt5631_outvolr_sel); +static SOC_ENUM_SINGLE_DECL(rt5631_outvolr_enum, RT5631_MONO_AXO_1_2_VOL, + RT5631_R_EN_SHIFT, rt5631_outvolr_sel); static const struct snd_kcontrol_new rt5631_outvolr_mux_control = SOC_DAPM_ENUM("Right OUTVOL SRC", rt5631_outvolr_enum); @@ -799,9 +791,8 @@ static const struct snd_kcontrol_new rt5631_outvolr_mux_control = /* Right HP Volume Input */ static const char *rt5631_hpvolr_sel[] = {"Vmid", "OUTMIXR"}; -static const SOC_ENUM_SINGLE_DECL( - rt5631_hpvolr_enum, RT5631_HP_OUT_VOL, - RT5631_R_EN_SHIFT, rt5631_hpvolr_sel); +static SOC_ENUM_SINGLE_DECL(rt5631_hpvolr_enum, RT5631_HP_OUT_VOL, + RT5631_R_EN_SHIFT, rt5631_hpvolr_sel); static const struct snd_kcontrol_new rt5631_hpvolr_mux_control = SOC_DAPM_ENUM("Right HPVOL SRC", rt5631_hpvolr_enum); @@ -809,9 +800,8 @@ static const struct snd_kcontrol_new rt5631_hpvolr_mux_control = /* Right SPK Volume Input */ static const char *rt5631_spkvolr_sel[] = {"Vmid", "SPKMIXR"}; -static const SOC_ENUM_SINGLE_DECL( - rt5631_spkvolr_enum, RT5631_SPK_OUT_VOL, - RT5631_R_EN_SHIFT, rt5631_spkvolr_sel); +static SOC_ENUM_SINGLE_DECL(rt5631_spkvolr_enum, RT5631_SPK_OUT_VOL, + RT5631_R_EN_SHIFT, rt5631_spkvolr_sel); static const struct snd_kcontrol_new rt5631_spkvolr_mux_control = SOC_DAPM_ENUM("Right SPKVOL SRC", rt5631_spkvolr_enum); @@ -820,9 +810,8 @@ static const struct snd_kcontrol_new rt5631_spkvolr_mux_control = static const char *rt5631_spol_src_sel[] = { "SPOLMIX", "MONOIN_RX", "VDAC", "DACL"}; -static const SOC_ENUM_SINGLE_DECL( - rt5631_spol_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL, - RT5631_SPK_L_MUX_SEL_SHIFT, rt5631_spol_src_sel); +static SOC_ENUM_SINGLE_DECL(rt5631_spol_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL, + RT5631_SPK_L_MUX_SEL_SHIFT, rt5631_spol_src_sel); static const struct snd_kcontrol_new rt5631_spol_mux_control = SOC_DAPM_ENUM("SPOL SRC", rt5631_spol_src_enum); @@ -831,9 +820,8 @@ static const struct snd_kcontrol_new rt5631_spol_mux_control = static const char *rt5631_spor_src_sel[] = { "SPORMIX", "MONOIN_RX", "VDAC", "DACR"}; -static const SOC_ENUM_SINGLE_DECL( - rt5631_spor_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL, - RT5631_SPK_R_MUX_SEL_SHIFT, rt5631_spor_src_sel); +static SOC_ENUM_SINGLE_DECL(rt5631_spor_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL, + RT5631_SPK_R_MUX_SEL_SHIFT, rt5631_spor_src_sel); static const struct snd_kcontrol_new rt5631_spor_mux_control = SOC_DAPM_ENUM("SPOR SRC", rt5631_spor_src_enum); @@ -841,9 +829,8 @@ static const struct snd_kcontrol_new rt5631_spor_mux_control = /* MONO Input */ static const char *rt5631_mono_src_sel[] = {"MONOMIX", "MONOIN_RX", "VDAC"}; -static const SOC_ENUM_SINGLE_DECL( - rt5631_mono_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL, - RT5631_MONO_MUX_SEL_SHIFT, rt5631_mono_src_sel); +static SOC_ENUM_SINGLE_DECL(rt5631_mono_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL, + RT5631_MONO_MUX_SEL_SHIFT, rt5631_mono_src_sel); static const struct snd_kcontrol_new rt5631_mono_mux_control = SOC_DAPM_ENUM("MONO SRC", rt5631_mono_src_enum); @@ -851,9 +838,8 @@ static const struct snd_kcontrol_new rt5631_mono_mux_control = /* Left HPO Input */ static const char *rt5631_hpl_src_sel[] = {"Left HPVOL", "Left DAC"}; -static const SOC_ENUM_SINGLE_DECL( - rt5631_hpl_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL, - RT5631_HP_L_MUX_SEL_SHIFT, rt5631_hpl_src_sel); +static SOC_ENUM_SINGLE_DECL(rt5631_hpl_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL, + RT5631_HP_L_MUX_SEL_SHIFT, rt5631_hpl_src_sel); static const struct snd_kcontrol_new rt5631_hpl_mux_control = SOC_DAPM_ENUM("HPL SRC", rt5631_hpl_src_enum); @@ -861,9 +847,8 @@ static const struct snd_kcontrol_new rt5631_hpl_mux_control = /* Right HPO Input */ static const char *rt5631_hpr_src_sel[] = {"Right HPVOL", "Right DAC"}; -static const SOC_ENUM_SINGLE_DECL( - rt5631_hpr_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL, - RT5631_HP_R_MUX_SEL_SHIFT, rt5631_hpr_src_sel); +static SOC_ENUM_SINGLE_DECL(rt5631_hpr_src_enum, RT5631_SPK_MONO_HP_OUT_CTRL, + RT5631_HP_R_MUX_SEL_SHIFT, rt5631_hpr_src_sel); static const struct snd_kcontrol_new rt5631_hpr_mux_control = SOC_DAPM_ENUM("HPR SRC", rt5631_hpr_src_enum); -- cgit v1.2.3 From 4c03cb6f86c1e171c3b277c207a6b9003402fbb6 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 09:43:21 +0100 Subject: ASoC: rt5640: Remove superfluous const As SOC_ENUM_SINGLE_DECL() itself contains const modifier now, we can reduce const from its users. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/rt5640.c | 75 +++++++++++++++++++++-------------------------- 1 file changed, 34 insertions(+), 41 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index a3fb41179636..fe4a5c2d4845 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -361,25 +361,24 @@ static unsigned int bst_tlv[] = { static const char * const rt5640_data_select[] = { "Normal", "left copy to right", "right copy to left", "Swap"}; -static const SOC_ENUM_SINGLE_DECL(rt5640_if1_dac_enum, RT5640_DIG_INF_DATA, - RT5640_IF1_DAC_SEL_SFT, rt5640_data_select); +static SOC_ENUM_SINGLE_DECL(rt5640_if1_dac_enum, RT5640_DIG_INF_DATA, + RT5640_IF1_DAC_SEL_SFT, rt5640_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5640_if1_adc_enum, RT5640_DIG_INF_DATA, - RT5640_IF1_ADC_SEL_SFT, rt5640_data_select); +static SOC_ENUM_SINGLE_DECL(rt5640_if1_adc_enum, RT5640_DIG_INF_DATA, + RT5640_IF1_ADC_SEL_SFT, rt5640_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5640_if2_dac_enum, RT5640_DIG_INF_DATA, - RT5640_IF2_DAC_SEL_SFT, rt5640_data_select); +static SOC_ENUM_SINGLE_DECL(rt5640_if2_dac_enum, RT5640_DIG_INF_DATA, + RT5640_IF2_DAC_SEL_SFT, rt5640_data_select); -static const SOC_ENUM_SINGLE_DECL(rt5640_if2_adc_enum, RT5640_DIG_INF_DATA, - RT5640_IF2_ADC_SEL_SFT, rt5640_data_select); +static SOC_ENUM_SINGLE_DECL(rt5640_if2_adc_enum, RT5640_DIG_INF_DATA, + RT5640_IF2_ADC_SEL_SFT, rt5640_data_select); /* Class D speaker gain ratio */ static const char * const rt5640_clsd_spk_ratio[] = {"1.66x", "1.83x", "1.94x", "2x", "2.11x", "2.22x", "2.33x", "2.44x", "2.55x", "2.66x", "2.77x"}; -static const SOC_ENUM_SINGLE_DECL( - rt5640_clsd_spk_ratio_enum, RT5640_CLS_D_OUT, - RT5640_CLSD_RATIO_SFT, rt5640_clsd_spk_ratio); +static SOC_ENUM_SINGLE_DECL(rt5640_clsd_spk_ratio_enum, RT5640_CLS_D_OUT, + RT5640_CLSD_RATIO_SFT, rt5640_clsd_spk_ratio); static const struct snd_kcontrol_new rt5640_snd_controls[] = { /* Speaker Output Volume */ @@ -753,9 +752,8 @@ static const char * const rt5640_stereo_adc1_src[] = { "DIG MIX", "ADC" }; -static const SOC_ENUM_SINGLE_DECL( - rt5640_stereo_adc1_enum, RT5640_STO_ADC_MIXER, - RT5640_ADC_1_SRC_SFT, rt5640_stereo_adc1_src); +static SOC_ENUM_SINGLE_DECL(rt5640_stereo_adc1_enum, RT5640_STO_ADC_MIXER, + RT5640_ADC_1_SRC_SFT, rt5640_stereo_adc1_src); static const struct snd_kcontrol_new rt5640_sto_adc_1_mux = SOC_DAPM_ENUM("Stereo ADC1 Mux", rt5640_stereo_adc1_enum); @@ -764,9 +762,8 @@ static const char * const rt5640_stereo_adc2_src[] = { "DMIC1", "DMIC2", "DIG MIX" }; -static const SOC_ENUM_SINGLE_DECL( - rt5640_stereo_adc2_enum, RT5640_STO_ADC_MIXER, - RT5640_ADC_2_SRC_SFT, rt5640_stereo_adc2_src); +static SOC_ENUM_SINGLE_DECL(rt5640_stereo_adc2_enum, RT5640_STO_ADC_MIXER, + RT5640_ADC_2_SRC_SFT, rt5640_stereo_adc2_src); static const struct snd_kcontrol_new rt5640_sto_adc_2_mux = SOC_DAPM_ENUM("Stereo ADC2 Mux", rt5640_stereo_adc2_enum); @@ -776,9 +773,8 @@ static const char * const rt5640_mono_adc_l1_src[] = { "Mono DAC MIXL", "ADCL" }; -static const SOC_ENUM_SINGLE_DECL( - rt5640_mono_adc_l1_enum, RT5640_MONO_ADC_MIXER, - RT5640_MONO_ADC_L1_SRC_SFT, rt5640_mono_adc_l1_src); +static SOC_ENUM_SINGLE_DECL(rt5640_mono_adc_l1_enum, RT5640_MONO_ADC_MIXER, + RT5640_MONO_ADC_L1_SRC_SFT, rt5640_mono_adc_l1_src); static const struct snd_kcontrol_new rt5640_mono_adc_l1_mux = SOC_DAPM_ENUM("Mono ADC1 left source", rt5640_mono_adc_l1_enum); @@ -787,9 +783,8 @@ static const char * const rt5640_mono_adc_l2_src[] = { "DMIC L1", "DMIC L2", "Mono DAC MIXL" }; -static const SOC_ENUM_SINGLE_DECL( - rt5640_mono_adc_l2_enum, RT5640_MONO_ADC_MIXER, - RT5640_MONO_ADC_L2_SRC_SFT, rt5640_mono_adc_l2_src); +static SOC_ENUM_SINGLE_DECL(rt5640_mono_adc_l2_enum, RT5640_MONO_ADC_MIXER, + RT5640_MONO_ADC_L2_SRC_SFT, rt5640_mono_adc_l2_src); static const struct snd_kcontrol_new rt5640_mono_adc_l2_mux = SOC_DAPM_ENUM("Mono ADC2 left source", rt5640_mono_adc_l2_enum); @@ -798,9 +793,8 @@ static const char * const rt5640_mono_adc_r1_src[] = { "Mono DAC MIXR", "ADCR" }; -static const SOC_ENUM_SINGLE_DECL( - rt5640_mono_adc_r1_enum, RT5640_MONO_ADC_MIXER, - RT5640_MONO_ADC_R1_SRC_SFT, rt5640_mono_adc_r1_src); +static SOC_ENUM_SINGLE_DECL(rt5640_mono_adc_r1_enum, RT5640_MONO_ADC_MIXER, + RT5640_MONO_ADC_R1_SRC_SFT, rt5640_mono_adc_r1_src); static const struct snd_kcontrol_new rt5640_mono_adc_r1_mux = SOC_DAPM_ENUM("Mono ADC1 right source", rt5640_mono_adc_r1_enum); @@ -809,9 +803,8 @@ static const char * const rt5640_mono_adc_r2_src[] = { "DMIC R1", "DMIC R2", "Mono DAC MIXR" }; -static const SOC_ENUM_SINGLE_DECL( - rt5640_mono_adc_r2_enum, RT5640_MONO_ADC_MIXER, - RT5640_MONO_ADC_R2_SRC_SFT, rt5640_mono_adc_r2_src); +static SOC_ENUM_SINGLE_DECL(rt5640_mono_adc_r2_enum, RT5640_MONO_ADC_MIXER, + RT5640_MONO_ADC_R2_SRC_SFT, rt5640_mono_adc_r2_src); static const struct snd_kcontrol_new rt5640_mono_adc_r2_mux = SOC_DAPM_ENUM("Mono ADC2 right source", rt5640_mono_adc_r2_enum); @@ -826,9 +819,9 @@ static int rt5640_dac_l2_values[] = { 3, }; -static const SOC_VALUE_ENUM_SINGLE_DECL( - rt5640_dac_l2_enum, RT5640_DSP_PATH2, RT5640_DAC_L2_SEL_SFT, - 0x3, rt5640_dac_l2_src, rt5640_dac_l2_values); +static SOC_VALUE_ENUM_SINGLE_DECL(rt5640_dac_l2_enum, + RT5640_DSP_PATH2, RT5640_DAC_L2_SEL_SFT, + 0x3, rt5640_dac_l2_src, rt5640_dac_l2_values); static const struct snd_kcontrol_new rt5640_dac_l2_mux = SOC_DAPM_VALUE_ENUM("DAC2 left channel source", rt5640_dac_l2_enum); @@ -841,9 +834,9 @@ static int rt5640_dac_r2_values[] = { 0, }; -static const SOC_VALUE_ENUM_SINGLE_DECL( - rt5640_dac_r2_enum, RT5640_DSP_PATH2, RT5640_DAC_R2_SEL_SFT, - 0x3, rt5640_dac_r2_src, rt5640_dac_r2_values); +static SOC_VALUE_ENUM_SINGLE_DECL(rt5640_dac_r2_enum, + RT5640_DSP_PATH2, RT5640_DAC_R2_SEL_SFT, + 0x3, rt5640_dac_r2_src, rt5640_dac_r2_values); static const struct snd_kcontrol_new rt5640_dac_r2_mux = SOC_DAPM_ENUM("DAC2 right channel source", rt5640_dac_r2_enum); @@ -860,9 +853,10 @@ static int rt5640_dai_iis_map_values[] = { 7, }; -static const SOC_VALUE_ENUM_SINGLE_DECL( - rt5640_dai_iis_map_enum, RT5640_I2S1_SDP, RT5640_I2S_IF_SFT, - 0x7, rt5640_dai_iis_map, rt5640_dai_iis_map_values); +static SOC_VALUE_ENUM_SINGLE_DECL(rt5640_dai_iis_map_enum, + RT5640_I2S1_SDP, RT5640_I2S_IF_SFT, + 0x7, rt5640_dai_iis_map, + rt5640_dai_iis_map_values); static const struct snd_kcontrol_new rt5640_dai_mux = SOC_DAPM_VALUE_ENUM("DAI select", rt5640_dai_iis_map_enum); @@ -872,9 +866,8 @@ static const char * const rt5640_sdi_sel[] = { "IF1", "IF2" }; -static const SOC_ENUM_SINGLE_DECL( - rt5640_sdi_sel_enum, RT5640_I2S2_SDP, - RT5640_I2S2_SDI_SFT, rt5640_sdi_sel); +static SOC_ENUM_SINGLE_DECL(rt5640_sdi_sel_enum, RT5640_I2S2_SDP, + RT5640_I2S2_SDI_SFT, rt5640_sdi_sel); static const struct snd_kcontrol_new rt5640_sdi_mux = SOC_DAPM_ENUM("SDI select", rt5640_sdi_sel_enum); -- cgit v1.2.3 From 655e3652db7c5fd73d8b7949f24e9d7432cdb806 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 09:43:35 +0100 Subject: ASoC: ssm2518: Remove superfluous const As SOC_ENUM_SINGLE_DECL() itself contains const modifier now, we can reduce const from its users. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ssm2518.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index cc8debce752f..806f3d826ffb 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -169,19 +169,19 @@ static const char * const ssm2518_drc_hold_time_text[] = { "682.24 ms", "1364 ms", }; -static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_peak_detector_attack_time_enum, +static SOC_ENUM_SINGLE_DECL(ssm2518_drc_peak_detector_attack_time_enum, SSM2518_REG_DRC_2, 4, ssm2518_drc_peak_detector_attack_time_text); -static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_peak_detector_release_time_enum, +static SOC_ENUM_SINGLE_DECL(ssm2518_drc_peak_detector_release_time_enum, SSM2518_REG_DRC_2, 0, ssm2518_drc_peak_detector_release_time_text); -static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_attack_time_enum, +static SOC_ENUM_SINGLE_DECL(ssm2518_drc_attack_time_enum, SSM2518_REG_DRC_6, 4, ssm2518_drc_peak_detector_attack_time_text); -static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_decay_time_enum, +static SOC_ENUM_SINGLE_DECL(ssm2518_drc_decay_time_enum, SSM2518_REG_DRC_6, 0, ssm2518_drc_peak_detector_release_time_text); -static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_hold_time_enum, +static SOC_ENUM_SINGLE_DECL(ssm2518_drc_hold_time_enum, SSM2518_REG_DRC_7, 4, ssm2518_drc_hold_time_text); -static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_noise_gate_hold_time_enum, +static SOC_ENUM_SINGLE_DECL(ssm2518_drc_noise_gate_hold_time_enum, SSM2518_REG_DRC_7, 0, ssm2518_drc_hold_time_text); -static const SOC_ENUM_SINGLE_DECL(ssm2518_drc_rms_averaging_time_enum, +static SOC_ENUM_SINGLE_DECL(ssm2518_drc_rms_averaging_time_enum, SSM2518_REG_DRC_9, 0, ssm2518_drc_peak_detector_release_time_text); static const struct snd_kcontrol_new ssm2518_snd_controls[] = { -- cgit v1.2.3 From f281205422d8924bee025af79f222d0c79646ae7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 09:43:49 +0100 Subject: ASoC: sta529: Remove superfluous const As SOC_ENUM_SINGLE_DECL() itself contains const modifier now, we can reduce const from its users. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/sta529.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index 40c07be9b581..f15b0e37274c 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -141,7 +141,7 @@ static const char *pwm_mode_text[] = { "Binary", "Headphone", "Ternary", static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -9150, 50, 0); static const DECLARE_TLV_DB_SCALE(master_vol_tlv, -12750, 50, 0); -static const SOC_ENUM_SINGLE_DECL(pwm_src, STA529_FFXCFG1, 4, pwm_mode_text); +static SOC_ENUM_SINGLE_DECL(pwm_src, STA529_FFXCFG1, 4, pwm_mode_text); static const struct snd_kcontrol_new sta529_snd_controls[] = { SOC_DOUBLE_R_TLV("Digital Playback Volume", STA529_LVOL, STA529_RVOL, 0, -- cgit v1.2.3 From da9f39f512eb2ac322d0ee340479d2739759919e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 09:44:00 +0100 Subject: ASoC: wm8804: Remove superfluous const As SOC_ENUM_SINGLE_DECL() itself contains const modifier now, we can reduce const from its users. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Charles Keepax Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8804.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 9bc8206a6807..72d12bbe1a56 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -92,7 +92,7 @@ WM8804_REGULATOR_EVENT(0) WM8804_REGULATOR_EVENT(1) static const char *txsrc_text[] = { "S/PDIF RX", "AIF" }; -static const SOC_ENUM_SINGLE_EXT_DECL(txsrc, txsrc_text); +static SOC_ENUM_SINGLE_EXT_DECL(txsrc, txsrc_text); static const struct snd_kcontrol_new wm8804_snd_controls[] = { SOC_ENUM_EXT("Input Source", txsrc, txsrc_get, txsrc_put), -- cgit v1.2.3 From 11a544bb2fe5bcacd20d2cdb9b792e035bd82eb2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 09:44:10 +0100 Subject: ASoC: wm8978: Remove superfluous const As SOC_ENUM_SINGLE_DECL() itself contains const modifier now, we can reduce const from its users. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Charles Keepax Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8978.c | 30 +++++++++++++++--------------- 1 file changed, 15 insertions(+), 15 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index d8fc531c0e59..a9e2f465c331 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -117,21 +117,21 @@ static const char *wm8978_eq5[] = {"5.3kHz", "6.9kHz", "9kHz", "11.7kHz"}; static const char *wm8978_alc3[] = {"ALC", "Limiter"}; static const char *wm8978_alc1[] = {"Off", "Right", "Left", "Both"}; -static const SOC_ENUM_SINGLE_DECL(adc_compand, WM8978_COMPANDING_CONTROL, 1, - wm8978_companding); -static const SOC_ENUM_SINGLE_DECL(dac_compand, WM8978_COMPANDING_CONTROL, 3, - wm8978_companding); -static const SOC_ENUM_SINGLE_DECL(eqmode, WM8978_EQ1, 8, wm8978_eqmode); -static const SOC_ENUM_SINGLE_DECL(eq1, WM8978_EQ1, 5, wm8978_eq1); -static const SOC_ENUM_SINGLE_DECL(eq2bw, WM8978_EQ2, 8, wm8978_bw); -static const SOC_ENUM_SINGLE_DECL(eq2, WM8978_EQ2, 5, wm8978_eq2); -static const SOC_ENUM_SINGLE_DECL(eq3bw, WM8978_EQ3, 8, wm8978_bw); -static const SOC_ENUM_SINGLE_DECL(eq3, WM8978_EQ3, 5, wm8978_eq3); -static const SOC_ENUM_SINGLE_DECL(eq4bw, WM8978_EQ4, 8, wm8978_bw); -static const SOC_ENUM_SINGLE_DECL(eq4, WM8978_EQ4, 5, wm8978_eq4); -static const SOC_ENUM_SINGLE_DECL(eq5, WM8978_EQ5, 5, wm8978_eq5); -static const SOC_ENUM_SINGLE_DECL(alc3, WM8978_ALC_CONTROL_3, 8, wm8978_alc3); -static const SOC_ENUM_SINGLE_DECL(alc1, WM8978_ALC_CONTROL_1, 7, wm8978_alc1); +static SOC_ENUM_SINGLE_DECL(adc_compand, WM8978_COMPANDING_CONTROL, 1, + wm8978_companding); +static SOC_ENUM_SINGLE_DECL(dac_compand, WM8978_COMPANDING_CONTROL, 3, + wm8978_companding); +static SOC_ENUM_SINGLE_DECL(eqmode, WM8978_EQ1, 8, wm8978_eqmode); +static SOC_ENUM_SINGLE_DECL(eq1, WM8978_EQ1, 5, wm8978_eq1); +static SOC_ENUM_SINGLE_DECL(eq2bw, WM8978_EQ2, 8, wm8978_bw); +static SOC_ENUM_SINGLE_DECL(eq2, WM8978_EQ2, 5, wm8978_eq2); +static SOC_ENUM_SINGLE_DECL(eq3bw, WM8978_EQ3, 8, wm8978_bw); +static SOC_ENUM_SINGLE_DECL(eq3, WM8978_EQ3, 5, wm8978_eq3); +static SOC_ENUM_SINGLE_DECL(eq4bw, WM8978_EQ4, 8, wm8978_bw); +static SOC_ENUM_SINGLE_DECL(eq4, WM8978_EQ4, 5, wm8978_eq4); +static SOC_ENUM_SINGLE_DECL(eq5, WM8978_EQ5, 5, wm8978_eq5); +static SOC_ENUM_SINGLE_DECL(alc3, WM8978_ALC_CONTROL_3, 8, wm8978_alc3); +static SOC_ENUM_SINGLE_DECL(alc1, WM8978_ALC_CONTROL_1, 7, wm8978_alc1); static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1); static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); -- cgit v1.2.3 From ae170688da955e9fb0536a117be63ee031cd1f6d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 09:44:56 +0100 Subject: ASoC: wm8983: Remove superfluous const As SOC_ENUM_SINGLE_DECL() itself contains const modifier now, we can reduce const from its users. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Charles Keepax Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8983.c | 39 +++++++++++++++++---------------------- 1 file changed, 17 insertions(+), 22 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index aa41ba0dfff4..770e5a705851 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -205,49 +205,44 @@ static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0); static const DECLARE_TLV_DB_SCALE(pga_boost_tlv, 0, 2000, 0); static const char *alc_sel_text[] = { "Off", "Right", "Left", "Stereo" }; -static const SOC_ENUM_SINGLE_DECL(alc_sel, WM8983_ALC_CONTROL_1, 7, - alc_sel_text); +static SOC_ENUM_SINGLE_DECL(alc_sel, WM8983_ALC_CONTROL_1, 7, alc_sel_text); static const char *alc_mode_text[] = { "ALC", "Limiter" }; -static const SOC_ENUM_SINGLE_DECL(alc_mode, WM8983_ALC_CONTROL_3, 8, - alc_mode_text); +static SOC_ENUM_SINGLE_DECL(alc_mode, WM8983_ALC_CONTROL_3, 8, alc_mode_text); static const char *filter_mode_text[] = { "Audio", "Application" }; -static const SOC_ENUM_SINGLE_DECL(filter_mode, WM8983_ADC_CONTROL, 7, - filter_mode_text); +static SOC_ENUM_SINGLE_DECL(filter_mode, WM8983_ADC_CONTROL, 7, + filter_mode_text); static const char *eq_bw_text[] = { "Narrow", "Wide" }; static const char *eqmode_text[] = { "Capture", "Playback" }; -static const SOC_ENUM_SINGLE_EXT_DECL(eqmode, eqmode_text); +static SOC_ENUM_SINGLE_EXT_DECL(eqmode, eqmode_text); static const char *eq1_cutoff_text[] = { "80Hz", "105Hz", "135Hz", "175Hz" }; -static const SOC_ENUM_SINGLE_DECL(eq1_cutoff, WM8983_EQ1_LOW_SHELF, 5, - eq1_cutoff_text); +static SOC_ENUM_SINGLE_DECL(eq1_cutoff, WM8983_EQ1_LOW_SHELF, 5, + eq1_cutoff_text); static const char *eq2_cutoff_text[] = { "230Hz", "300Hz", "385Hz", "500Hz" }; -static const SOC_ENUM_SINGLE_DECL(eq2_bw, WM8983_EQ2_PEAK_1, 8, eq_bw_text); -static const SOC_ENUM_SINGLE_DECL(eq2_cutoff, WM8983_EQ2_PEAK_1, 5, - eq2_cutoff_text); +static SOC_ENUM_SINGLE_DECL(eq2_bw, WM8983_EQ2_PEAK_1, 8, eq_bw_text); +static SOC_ENUM_SINGLE_DECL(eq2_cutoff, WM8983_EQ2_PEAK_1, 5, eq2_cutoff_text); static const char *eq3_cutoff_text[] = { "650Hz", "850Hz", "1.1kHz", "1.4kHz" }; -static const SOC_ENUM_SINGLE_DECL(eq3_bw, WM8983_EQ3_PEAK_2, 8, eq_bw_text); -static const SOC_ENUM_SINGLE_DECL(eq3_cutoff, WM8983_EQ3_PEAK_2, 5, - eq3_cutoff_text); +static SOC_ENUM_SINGLE_DECL(eq3_bw, WM8983_EQ3_PEAK_2, 8, eq_bw_text); +static SOC_ENUM_SINGLE_DECL(eq3_cutoff, WM8983_EQ3_PEAK_2, 5, eq3_cutoff_text); static const char *eq4_cutoff_text[] = { "1.8kHz", "2.4kHz", "3.2kHz", "4.1kHz" }; -static const SOC_ENUM_SINGLE_DECL(eq4_bw, WM8983_EQ4_PEAK_3, 8, eq_bw_text); -static const SOC_ENUM_SINGLE_DECL(eq4_cutoff, WM8983_EQ4_PEAK_3, 5, - eq4_cutoff_text); +static SOC_ENUM_SINGLE_DECL(eq4_bw, WM8983_EQ4_PEAK_3, 8, eq_bw_text); +static SOC_ENUM_SINGLE_DECL(eq4_cutoff, WM8983_EQ4_PEAK_3, 5, eq4_cutoff_text); static const char *eq5_cutoff_text[] = { "5.3kHz", "6.9kHz", "9kHz", "11.7kHz" }; -static const SOC_ENUM_SINGLE_DECL(eq5_cutoff, WM8983_EQ5_HIGH_SHELF, 5, - eq5_cutoff_text); +static SOC_ENUM_SINGLE_DECL(eq5_cutoff, WM8983_EQ5_HIGH_SHELF, 5, + eq5_cutoff_text); static const char *depth_3d_text[] = { "Off", @@ -267,8 +262,8 @@ static const char *depth_3d_text[] = { "93.3%", "100%" }; -static const SOC_ENUM_SINGLE_DECL(depth_3d, WM8983_3D_CONTROL, 0, - depth_3d_text); +static SOC_ENUM_SINGLE_DECL(depth_3d, WM8983_3D_CONTROL, 0, + depth_3d_text); static const struct snd_kcontrol_new wm8983_snd_controls[] = { SOC_SINGLE("Digital Loopback Switch", WM8983_COMPANDING_CONTROL, -- cgit v1.2.3 From d0a4eec16adf3a528dbd1b52c3fc653a2bfee8c4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 09:45:08 +0100 Subject: ASoC: wm8985: Remove superfluous const As SOC_ENUM_SINGLE_DECL() itself contains const modifier now, we can reduce const from its users. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Charles Keepax Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8985.c | 39 +++++++++++++++++---------------------- 1 file changed, 17 insertions(+), 22 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index 271b517911a4..d786f2b39764 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -226,52 +226,48 @@ static const DECLARE_TLV_DB_SCALE(bypass_tlv, -1500, 300, 0); static const DECLARE_TLV_DB_SCALE(pga_boost_tlv, 0, 2000, 0); static const char *alc_sel_text[] = { "Off", "Right", "Left", "Stereo" }; -static const SOC_ENUM_SINGLE_DECL(alc_sel, WM8985_ALC_CONTROL_1, 7, - alc_sel_text); +static SOC_ENUM_SINGLE_DECL(alc_sel, WM8985_ALC_CONTROL_1, 7, alc_sel_text); static const char *alc_mode_text[] = { "ALC", "Limiter" }; -static const SOC_ENUM_SINGLE_DECL(alc_mode, WM8985_ALC_CONTROL_3, 8, - alc_mode_text); +static SOC_ENUM_SINGLE_DECL(alc_mode, WM8985_ALC_CONTROL_3, 8, alc_mode_text); static const char *filter_mode_text[] = { "Audio", "Application" }; -static const SOC_ENUM_SINGLE_DECL(filter_mode, WM8985_ADC_CONTROL, 7, - filter_mode_text); +static SOC_ENUM_SINGLE_DECL(filter_mode, WM8985_ADC_CONTROL, 7, + filter_mode_text); static const char *eq_bw_text[] = { "Narrow", "Wide" }; static const char *eqmode_text[] = { "Capture", "Playback" }; -static const SOC_ENUM_SINGLE_EXT_DECL(eqmode, eqmode_text); +static SOC_ENUM_SINGLE_EXT_DECL(eqmode, eqmode_text); static const char *eq1_cutoff_text[] = { "80Hz", "105Hz", "135Hz", "175Hz" }; -static const SOC_ENUM_SINGLE_DECL(eq1_cutoff, WM8985_EQ1_LOW_SHELF, 5, - eq1_cutoff_text); +static SOC_ENUM_SINGLE_DECL(eq1_cutoff, WM8985_EQ1_LOW_SHELF, 5, + eq1_cutoff_text); static const char *eq2_cutoff_text[] = { "230Hz", "300Hz", "385Hz", "500Hz" }; -static const SOC_ENUM_SINGLE_DECL(eq2_bw, WM8985_EQ2_PEAK_1, 8, eq_bw_text); -static const SOC_ENUM_SINGLE_DECL(eq2_cutoff, WM8985_EQ2_PEAK_1, 5, - eq2_cutoff_text); +static SOC_ENUM_SINGLE_DECL(eq2_bw, WM8985_EQ2_PEAK_1, 8, eq_bw_text); +static SOC_ENUM_SINGLE_DECL(eq2_cutoff, WM8985_EQ2_PEAK_1, 5, eq2_cutoff_text); static const char *eq3_cutoff_text[] = { "650Hz", "850Hz", "1.1kHz", "1.4kHz" }; -static const SOC_ENUM_SINGLE_DECL(eq3_bw, WM8985_EQ3_PEAK_2, 8, eq_bw_text); -static const SOC_ENUM_SINGLE_DECL(eq3_cutoff, WM8985_EQ3_PEAK_2, 5, - eq3_cutoff_text); +static SOC_ENUM_SINGLE_DECL(eq3_bw, WM8985_EQ3_PEAK_2, 8, eq_bw_text); +static SOC_ENUM_SINGLE_DECL(eq3_cutoff, WM8985_EQ3_PEAK_2, 5, + eq3_cutoff_text); static const char *eq4_cutoff_text[] = { "1.8kHz", "2.4kHz", "3.2kHz", "4.1kHz" }; -static const SOC_ENUM_SINGLE_DECL(eq4_bw, WM8985_EQ4_PEAK_3, 8, eq_bw_text); -static const SOC_ENUM_SINGLE_DECL(eq4_cutoff, WM8985_EQ4_PEAK_3, 5, - eq4_cutoff_text); +static SOC_ENUM_SINGLE_DECL(eq4_bw, WM8985_EQ4_PEAK_3, 8, eq_bw_text); +static SOC_ENUM_SINGLE_DECL(eq4_cutoff, WM8985_EQ4_PEAK_3, 5, eq4_cutoff_text); static const char *eq5_cutoff_text[] = { "5.3kHz", "6.9kHz", "9kHz", "11.7kHz" }; -static const SOC_ENUM_SINGLE_DECL(eq5_cutoff, WM8985_EQ5_HIGH_SHELF, 5, +static SOC_ENUM_SINGLE_DECL(eq5_cutoff, WM8985_EQ5_HIGH_SHELF, 5, eq5_cutoff_text); static const char *speaker_mode_text[] = { "Class A/B", "Class D" }; -static const SOC_ENUM_SINGLE_DECL(speaker_mode, 0x17, 8, speaker_mode_text); +static SOC_ENUM_SINGLE_DECL(speaker_mode, 0x17, 8, speaker_mode_text); static const char *depth_3d_text[] = { "Off", @@ -291,8 +287,7 @@ static const char *depth_3d_text[] = { "93.3%", "100%" }; -static const SOC_ENUM_SINGLE_DECL(depth_3d, WM8985_3D_CONTROL, 0, - depth_3d_text); +static SOC_ENUM_SINGLE_DECL(depth_3d, WM8985_3D_CONTROL, 0, depth_3d_text); static const struct snd_kcontrol_new wm8985_snd_controls[] = { SOC_SINGLE("Digital Loopback Switch", WM8985_COMPANDING_CONTROL, -- cgit v1.2.3 From 48e50ce37fcf4c376845d1c42177eeb1601f99ad Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 09:45:18 +0100 Subject: ASoC: wm8995: Remove superfluous const As SOC_ENUM_SINGLE_DECL() itself contains const modifier now, we can reduce const from its users. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Charles Keepax Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8995.c | 32 ++++++++++++++++---------------- 1 file changed, 16 insertions(+), 16 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 4300caff1783..dcd1e72c19cf 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -423,24 +423,24 @@ static const char *in1l_text[] = { "Differential", "Single-ended IN1LN", "Single-ended IN1LP" }; -static const SOC_ENUM_SINGLE_DECL(in1l_enum, WM8995_LEFT_LINE_INPUT_CONTROL, - 2, in1l_text); +static SOC_ENUM_SINGLE_DECL(in1l_enum, WM8995_LEFT_LINE_INPUT_CONTROL, + 2, in1l_text); static const char *in1r_text[] = { "Differential", "Single-ended IN1RN", "Single-ended IN1RP" }; -static const SOC_ENUM_SINGLE_DECL(in1r_enum, WM8995_LEFT_LINE_INPUT_CONTROL, - 0, in1r_text); +static SOC_ENUM_SINGLE_DECL(in1r_enum, WM8995_LEFT_LINE_INPUT_CONTROL, + 0, in1r_text); static const char *dmic_src_text[] = { "DMICDAT1", "DMICDAT2", "DMICDAT3" }; -static const SOC_ENUM_SINGLE_DECL(dmic_src1_enum, WM8995_POWER_MANAGEMENT_5, - 8, dmic_src_text); -static const SOC_ENUM_SINGLE_DECL(dmic_src2_enum, WM8995_POWER_MANAGEMENT_5, - 6, dmic_src_text); +static SOC_ENUM_SINGLE_DECL(dmic_src1_enum, WM8995_POWER_MANAGEMENT_5, + 8, dmic_src_text); +static SOC_ENUM_SINGLE_DECL(dmic_src2_enum, WM8995_POWER_MANAGEMENT_5, + 6, dmic_src_text); static const struct snd_kcontrol_new wm8995_snd_controls[] = { SOC_DOUBLE_R_TLV("DAC1 Volume", WM8995_DAC1_LEFT_VOLUME, @@ -899,14 +899,14 @@ static const char *spk_src_text[] = { "DAC1L", "DAC1R", "DAC2L", "DAC2R" }; -static const SOC_ENUM_SINGLE_DECL(spk1l_src_enum, WM8995_LEFT_PDM_SPEAKER_1, - 0, spk_src_text); -static const SOC_ENUM_SINGLE_DECL(spk1r_src_enum, WM8995_RIGHT_PDM_SPEAKER_1, - 0, spk_src_text); -static const SOC_ENUM_SINGLE_DECL(spk2l_src_enum, WM8995_LEFT_PDM_SPEAKER_2, - 0, spk_src_text); -static const SOC_ENUM_SINGLE_DECL(spk2r_src_enum, WM8995_RIGHT_PDM_SPEAKER_2, - 0, spk_src_text); +static SOC_ENUM_SINGLE_DECL(spk1l_src_enum, WM8995_LEFT_PDM_SPEAKER_1, + 0, spk_src_text); +static SOC_ENUM_SINGLE_DECL(spk1r_src_enum, WM8995_RIGHT_PDM_SPEAKER_1, + 0, spk_src_text); +static SOC_ENUM_SINGLE_DECL(spk2l_src_enum, WM8995_LEFT_PDM_SPEAKER_2, + 0, spk_src_text); +static SOC_ENUM_SINGLE_DECL(spk2r_src_enum, WM8995_RIGHT_PDM_SPEAKER_2, + 0, spk_src_text); static const struct snd_kcontrol_new spk1l_mux = SOC_DAPM_ENUM("SPK1L SRC", spk1l_src_enum); -- cgit v1.2.3 From 27ca2c30f4fec73d53f727fe2fe729e16d9a43b2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:01:43 +0100 Subject: ASoC: arizona: Fix wrong number of items in enum ctls arizona codec driver has a few places wrongly defining the number of enum items. Use SOC_ENUM_SINGLE_DECL() macro and they are automatically fixed. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 71 ++++++++++++++++++++++++++-------------------- 1 file changed, 40 insertions(+), 31 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index e4295fee8f13..a32b84ac03f6 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -542,67 +542,76 @@ static const char *arizona_vol_ramp_text[] = { "15ms/6dB", "30ms/6dB", }; -const struct soc_enum arizona_in_vd_ramp = - SOC_ENUM_SINGLE(ARIZONA_INPUT_VOLUME_RAMP, - ARIZONA_IN_VD_RAMP_SHIFT, 7, arizona_vol_ramp_text); +SOC_ENUM_SINGLE_DECL(arizona_in_vd_ramp, + ARIZONA_INPUT_VOLUME_RAMP, + ARIZONA_IN_VD_RAMP_SHIFT, + arizona_vol_ramp_text); EXPORT_SYMBOL_GPL(arizona_in_vd_ramp); -const struct soc_enum arizona_in_vi_ramp = - SOC_ENUM_SINGLE(ARIZONA_INPUT_VOLUME_RAMP, - ARIZONA_IN_VI_RAMP_SHIFT, 7, arizona_vol_ramp_text); +SOC_ENUM_SINGLE_DECL(arizona_in_vi_ramp, + ARIZONA_INPUT_VOLUME_RAMP, + ARIZONA_IN_VI_RAMP_SHIFT, + arizona_vol_ramp_text); EXPORT_SYMBOL_GPL(arizona_in_vi_ramp); -const struct soc_enum arizona_out_vd_ramp = - SOC_ENUM_SINGLE(ARIZONA_OUTPUT_VOLUME_RAMP, - ARIZONA_OUT_VD_RAMP_SHIFT, 7, arizona_vol_ramp_text); +SOC_ENUM_SINGLE_DECL(arizona_out_vd_ramp, + ARIZONA_OUTPUT_VOLUME_RAMP, + ARIZONA_OUT_VD_RAMP_SHIFT, + arizona_vol_ramp_text); EXPORT_SYMBOL_GPL(arizona_out_vd_ramp); -const struct soc_enum arizona_out_vi_ramp = - SOC_ENUM_SINGLE(ARIZONA_OUTPUT_VOLUME_RAMP, - ARIZONA_OUT_VI_RAMP_SHIFT, 7, arizona_vol_ramp_text); +SOC_ENUM_SINGLE_DECL(arizona_out_vi_ramp, + ARIZONA_OUTPUT_VOLUME_RAMP, + ARIZONA_OUT_VI_RAMP_SHIFT, + arizona_vol_ramp_text); EXPORT_SYMBOL_GPL(arizona_out_vi_ramp); static const char *arizona_lhpf_mode_text[] = { "Low-pass", "High-pass" }; -const struct soc_enum arizona_lhpf1_mode = - SOC_ENUM_SINGLE(ARIZONA_HPLPF1_1, ARIZONA_LHPF1_MODE_SHIFT, 2, - arizona_lhpf_mode_text); +SOC_ENUM_SINGLE_DECL(arizona_lhpf1_mode, + ARIZONA_HPLPF1_1, + ARIZONA_LHPF1_MODE_SHIFT, + arizona_lhpf_mode_text); EXPORT_SYMBOL_GPL(arizona_lhpf1_mode); -const struct soc_enum arizona_lhpf2_mode = - SOC_ENUM_SINGLE(ARIZONA_HPLPF2_1, ARIZONA_LHPF2_MODE_SHIFT, 2, - arizona_lhpf_mode_text); +SOC_ENUM_SINGLE_DECL(arizona_lhpf2_mode, + ARIZONA_HPLPF2_1, + ARIZONA_LHPF2_MODE_SHIFT, + arizona_lhpf_mode_text); EXPORT_SYMBOL_GPL(arizona_lhpf2_mode); -const struct soc_enum arizona_lhpf3_mode = - SOC_ENUM_SINGLE(ARIZONA_HPLPF3_1, ARIZONA_LHPF3_MODE_SHIFT, 2, - arizona_lhpf_mode_text); +SOC_ENUM_SINGLE_DECL(arizona_lhpf3_mode, + ARIZONA_HPLPF3_1, + ARIZONA_LHPF3_MODE_SHIFT, + arizona_lhpf_mode_text); EXPORT_SYMBOL_GPL(arizona_lhpf3_mode); -const struct soc_enum arizona_lhpf4_mode = - SOC_ENUM_SINGLE(ARIZONA_HPLPF4_1, ARIZONA_LHPF4_MODE_SHIFT, 2, - arizona_lhpf_mode_text); +SOC_ENUM_SINGLE_DECL(arizona_lhpf4_mode, + ARIZONA_HPLPF4_1, + ARIZONA_LHPF4_MODE_SHIFT, + arizona_lhpf_mode_text); EXPORT_SYMBOL_GPL(arizona_lhpf4_mode); static const char *arizona_ng_hold_text[] = { "30ms", "120ms", "250ms", "500ms", }; -const struct soc_enum arizona_ng_hold = - SOC_ENUM_SINGLE(ARIZONA_NOISE_GATE_CONTROL, ARIZONA_NGATE_HOLD_SHIFT, - 4, arizona_ng_hold_text); +SOC_ENUM_SINGLE_DECL(arizona_ng_hold, + ARIZONA_NOISE_GATE_CONTROL, + ARIZONA_NGATE_HOLD_SHIFT, + arizona_ng_hold_text); EXPORT_SYMBOL_GPL(arizona_ng_hold); static const char * const arizona_in_hpf_cut_text[] = { "2.5Hz", "5Hz", "10Hz", "20Hz", "40Hz" }; -const struct soc_enum arizona_in_hpf_cut_enum = - SOC_ENUM_SINGLE(ARIZONA_HPF_CONTROL, ARIZONA_IN_HPF_CUT_SHIFT, - ARRAY_SIZE(arizona_in_hpf_cut_text), - arizona_in_hpf_cut_text); +SOC_ENUM_SINGLE_DECL(arizona_in_hpf_cut_enum, + ARIZONA_HPF_CONTROL, + ARIZONA_IN_HPF_CUT_SHIFT, + arizona_in_hpf_cut_text); EXPORT_SYMBOL_GPL(arizona_in_hpf_cut_enum); static const char * const arizona_in_dmic_osr_text[] = { -- cgit v1.2.3 From 1c38450b9fe52a86da7f4fd2a2c97935b71f2b62 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 09:58:45 +0100 Subject: ASoC: adau1373: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adau1373.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index eb836ed5271f..83c0acd5f67d 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -576,8 +576,8 @@ static const char *adau1373_decimator_text[] = { "DMIC1", }; -static const struct soc_enum adau1373_decimator_enum = - SOC_ENUM_SINGLE(0, 0, 2, adau1373_decimator_text); +static SOC_ENUM_SINGLE_DECL(adau1373_decimator_enum, + 0, 0, adau1373_decimator_text); static const struct snd_kcontrol_new adau1373_decimator_mux = SOC_DAPM_ENUM_VIRT("Decimator Mux", adau1373_decimator_enum); -- cgit v1.2.3 From 98bf1b5e788e63536705dcc418cc97b36481067e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:00:29 +0100 Subject: ASoC: alc5623: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/alc5623.c | 30 ++++++++++++++++++------------ 1 file changed, 18 insertions(+), 12 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index d3036283482a..ba61c07ebbb2 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -228,32 +228,37 @@ static const char *alc5623_aux_out_input_sel[] = { "Vmid", "HPOut Mix", "Speaker Mix", "Mono Mix"}; /* auxout output mux */ -static const struct soc_enum alc5623_aux_out_input_enum = -SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 6, 4, alc5623_aux_out_input_sel); +static SOC_ENUM_SINGLE_DECL(alc5623_aux_out_input_enum, + ALC5623_OUTPUT_MIXER_CTRL, 6, + alc5623_aux_out_input_sel); static const struct snd_kcontrol_new alc5623_auxout_mux_controls = SOC_DAPM_ENUM("Route", alc5623_aux_out_input_enum); /* speaker output mux */ -static const struct soc_enum alc5623_spkout_input_enum = -SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 10, 4, alc5623_spkout_input_sel); +static SOC_ENUM_SINGLE_DECL(alc5623_spkout_input_enum, + ALC5623_OUTPUT_MIXER_CTRL, 10, + alc5623_spkout_input_sel); static const struct snd_kcontrol_new alc5623_spkout_mux_controls = SOC_DAPM_ENUM("Route", alc5623_spkout_input_enum); /* headphone left output mux */ -static const struct soc_enum alc5623_hpl_out_input_enum = -SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 9, 2, alc5623_hpl_out_input_sel); +static SOC_ENUM_SINGLE_DECL(alc5623_hpl_out_input_enum, + ALC5623_OUTPUT_MIXER_CTRL, 9, + alc5623_hpl_out_input_sel); static const struct snd_kcontrol_new alc5623_hpl_out_mux_controls = SOC_DAPM_ENUM("Route", alc5623_hpl_out_input_enum); /* headphone right output mux */ -static const struct soc_enum alc5623_hpr_out_input_enum = -SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 8, 2, alc5623_hpr_out_input_sel); +static SOC_ENUM_SINGLE_DECL(alc5623_hpr_out_input_enum, + ALC5623_OUTPUT_MIXER_CTRL, 8, + alc5623_hpr_out_input_sel); static const struct snd_kcontrol_new alc5623_hpr_out_mux_controls = SOC_DAPM_ENUM("Route", alc5623_hpr_out_input_enum); /* speaker output N select */ -static const struct soc_enum alc5623_spk_n_sour_enum = -SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 14, 4, alc5623_spk_n_sour_sel); +static SOC_ENUM_SINGLE_DECL(alc5623_spk_n_sour_enum, + ALC5623_OUTPUT_MIXER_CTRL, 14, + alc5623_spk_n_sour_sel); static const struct snd_kcontrol_new alc5623_spkoutn_mux_controls = SOC_DAPM_ENUM("Route", alc5623_spk_n_sour_enum); @@ -338,8 +343,9 @@ SND_SOC_DAPM_VMID("Vmid"), }; static const char *alc5623_amp_names[] = {"AB Amp", "D Amp"}; -static const struct soc_enum alc5623_amp_enum = - SOC_ENUM_SINGLE(ALC5623_OUTPUT_MIXER_CTRL, 13, 2, alc5623_amp_names); +static SOC_ENUM_SINGLE_DECL(alc5623_amp_enum, + ALC5623_OUTPUT_MIXER_CTRL, 13, + alc5623_amp_names); static const struct snd_kcontrol_new alc5623_amp_mux_controls = SOC_DAPM_ENUM("Route", alc5623_amp_enum); -- cgit v1.2.3 From 4682a0a2b8e5d9633727276455f445e0b402767c Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:05:01 +0100 Subject: ASoC: cs42l52: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l52.c | 92 ++++++++++++++++++++++------------------------ 1 file changed, 43 insertions(+), 49 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 0bac6d5a4ac8..be455ea5f2fe 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -210,13 +210,11 @@ static const char * const cs42l52_adca_text[] = { static const char * const cs42l52_adcb_text[] = { "Input1B", "Input2B", "Input3B", "Input4B", "PGA Input Right"}; -static const struct soc_enum adca_enum = - SOC_ENUM_SINGLE(CS42L52_ADC_PGA_A, 5, - ARRAY_SIZE(cs42l52_adca_text), cs42l52_adca_text); +static SOC_ENUM_SINGLE_DECL(adca_enum, + CS42L52_ADC_PGA_A, 5, cs42l52_adca_text); -static const struct soc_enum adcb_enum = - SOC_ENUM_SINGLE(CS42L52_ADC_PGA_B, 5, - ARRAY_SIZE(cs42l52_adcb_text), cs42l52_adcb_text); +static SOC_ENUM_SINGLE_DECL(adcb_enum, + CS42L52_ADC_PGA_B, 5, cs42l52_adcb_text); static const struct snd_kcontrol_new adca_mux = SOC_DAPM_ENUM("Left ADC Input Capture Mux", adca_enum); @@ -229,26 +227,22 @@ static const char * const mic_bias_level_text[] = { "0.8 +VA", "0.83 +VA", "0.91 +VA" }; -static const struct soc_enum mic_bias_level_enum = - SOC_ENUM_SINGLE(CS42L52_IFACE_CTL2, 0, - ARRAY_SIZE(mic_bias_level_text), mic_bias_level_text); +static SOC_ENUM_SINGLE_DECL(mic_bias_level_enum, + CS42L52_IFACE_CTL2, 0, mic_bias_level_text); static const char * const cs42l52_mic_text[] = { "MIC1", "MIC2" }; -static const struct soc_enum mica_enum = - SOC_ENUM_SINGLE(CS42L52_MICA_CTL, 5, - ARRAY_SIZE(cs42l52_mic_text), cs42l52_mic_text); +static SOC_ENUM_SINGLE_DECL(mica_enum, + CS42L52_MICA_CTL, 5, cs42l52_mic_text); -static const struct soc_enum micb_enum = - SOC_ENUM_SINGLE(CS42L52_MICB_CTL, 5, - ARRAY_SIZE(cs42l52_mic_text), cs42l52_mic_text); +static SOC_ENUM_SINGLE_DECL(micb_enum, + CS42L52_MICB_CTL, 5, cs42l52_mic_text); static const char * const digital_output_mux_text[] = {"ADC", "DSP"}; -static const struct soc_enum digital_output_mux_enum = - SOC_ENUM_SINGLE(CS42L52_ADC_MISC_CTL, 6, - ARRAY_SIZE(digital_output_mux_text), - digital_output_mux_text); +static SOC_ENUM_SINGLE_DECL(digital_output_mux_enum, + CS42L52_ADC_MISC_CTL, 6, + digital_output_mux_text); static const struct snd_kcontrol_new digital_output_mux = SOC_DAPM_ENUM("Digital Output Mux", digital_output_mux_enum); @@ -258,18 +252,18 @@ static const char * const hp_gain_num_text[] = { "0.7099", "0.8399", "1.000", "1.1430" }; -static const struct soc_enum hp_gain_enum = - SOC_ENUM_SINGLE(CS42L52_PB_CTL1, 5, - ARRAY_SIZE(hp_gain_num_text), hp_gain_num_text); +static SOC_ENUM_SINGLE_DECL(hp_gain_enum, + CS42L52_PB_CTL1, 5, + hp_gain_num_text); static const char * const beep_pitch_text[] = { "C4", "C5", "D5", "E5", "F5", "G5", "A5", "B5", "C6", "D6", "E6", "F6", "G6", "A6", "B6", "C7" }; -static const struct soc_enum beep_pitch_enum = - SOC_ENUM_SINGLE(CS42L52_BEEP_FREQ, 4, - ARRAY_SIZE(beep_pitch_text), beep_pitch_text); +static SOC_ENUM_SINGLE_DECL(beep_pitch_enum, + CS42L52_BEEP_FREQ, 4, + beep_pitch_text); static const char * const beep_ontime_text[] = { "86 ms", "430 ms", "780 ms", "1.20 s", "1.50 s", @@ -277,66 +271,66 @@ static const char * const beep_ontime_text[] = { "3.50 s", "3.80 s", "4.20 s", "4.50 s", "4.80 s", "5.20 s" }; -static const struct soc_enum beep_ontime_enum = - SOC_ENUM_SINGLE(CS42L52_BEEP_FREQ, 0, - ARRAY_SIZE(beep_ontime_text), beep_ontime_text); +static SOC_ENUM_SINGLE_DECL(beep_ontime_enum, + CS42L52_BEEP_FREQ, 0, + beep_ontime_text); static const char * const beep_offtime_text[] = { "1.23 s", "2.58 s", "3.90 s", "5.20 s", "6.60 s", "8.05 s", "9.35 s", "10.80 s" }; -static const struct soc_enum beep_offtime_enum = - SOC_ENUM_SINGLE(CS42L52_BEEP_VOL, 5, - ARRAY_SIZE(beep_offtime_text), beep_offtime_text); +static SOC_ENUM_SINGLE_DECL(beep_offtime_enum, + CS42L52_BEEP_VOL, 5, + beep_offtime_text); static const char * const beep_config_text[] = { "Off", "Single", "Multiple", "Continuous" }; -static const struct soc_enum beep_config_enum = - SOC_ENUM_SINGLE(CS42L52_BEEP_TONE_CTL, 6, - ARRAY_SIZE(beep_config_text), beep_config_text); +static SOC_ENUM_SINGLE_DECL(beep_config_enum, + CS42L52_BEEP_TONE_CTL, 6, + beep_config_text); static const char * const beep_bass_text[] = { "50 Hz", "100 Hz", "200 Hz", "250 Hz" }; -static const struct soc_enum beep_bass_enum = - SOC_ENUM_SINGLE(CS42L52_BEEP_TONE_CTL, 1, - ARRAY_SIZE(beep_bass_text), beep_bass_text); +static SOC_ENUM_SINGLE_DECL(beep_bass_enum, + CS42L52_BEEP_TONE_CTL, 1, + beep_bass_text); static const char * const beep_treble_text[] = { "5 kHz", "7 kHz", "10 kHz", " 15 kHz" }; -static const struct soc_enum beep_treble_enum = - SOC_ENUM_SINGLE(CS42L52_BEEP_TONE_CTL, 3, - ARRAY_SIZE(beep_treble_text), beep_treble_text); +static SOC_ENUM_SINGLE_DECL(beep_treble_enum, + CS42L52_BEEP_TONE_CTL, 3, + beep_treble_text); static const char * const ng_threshold_text[] = { "-34dB", "-37dB", "-40dB", "-43dB", "-46dB", "-52dB", "-58dB", "-64dB" }; -static const struct soc_enum ng_threshold_enum = - SOC_ENUM_SINGLE(CS42L52_NOISE_GATE_CTL, 2, - ARRAY_SIZE(ng_threshold_text), ng_threshold_text); +static SOC_ENUM_SINGLE_DECL(ng_threshold_enum, + CS42L52_NOISE_GATE_CTL, 2, + ng_threshold_text); static const char * const cs42l52_ng_delay_text[] = { "50ms", "100ms", "150ms", "200ms"}; -static const struct soc_enum ng_delay_enum = - SOC_ENUM_SINGLE(CS42L52_NOISE_GATE_CTL, 0, - ARRAY_SIZE(cs42l52_ng_delay_text), cs42l52_ng_delay_text); +static SOC_ENUM_SINGLE_DECL(ng_delay_enum, + CS42L52_NOISE_GATE_CTL, 0, + cs42l52_ng_delay_text); static const char * const cs42l52_ng_type_text[] = { "Apply Specific", "Apply All" }; -static const struct soc_enum ng_type_enum = - SOC_ENUM_SINGLE(CS42L52_NOISE_GATE_CTL, 6, - ARRAY_SIZE(cs42l52_ng_type_text), cs42l52_ng_type_text); +static SOC_ENUM_SINGLE_DECL(ng_type_enum, + CS42L52_NOISE_GATE_CTL, 6, + cs42l52_ng_type_text); static const char * const left_swap_text[] = { "Left", "LR 2", "Right"}; -- cgit v1.2.3 From e34042d850a7117b9acafeabd50287d5d8e61849 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:06:00 +0100 Subject: ASoC: da7210: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/da7210.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index e62e294a8033..01e55fc72307 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -307,29 +307,29 @@ static const char * const da7210_hpf_cutoff_txt[] = { "Fs/8192*pi", "Fs/4096*pi", "Fs/2048*pi", "Fs/1024*pi" }; -static const struct soc_enum da7210_dac_hpf_cutoff = - SOC_ENUM_SINGLE(DA7210_DAC_HPF, 0, 4, da7210_hpf_cutoff_txt); +static SOC_ENUM_SINGLE_DECL(da7210_dac_hpf_cutoff, + DA7210_DAC_HPF, 0, da7210_hpf_cutoff_txt); -static const struct soc_enum da7210_adc_hpf_cutoff = - SOC_ENUM_SINGLE(DA7210_ADC_HPF, 0, 4, da7210_hpf_cutoff_txt); +static SOC_ENUM_SINGLE_DECL(da7210_adc_hpf_cutoff, + DA7210_ADC_HPF, 0, da7210_hpf_cutoff_txt); /* ADC and DAC voice (8kHz) high pass cutoff value */ static const char * const da7210_vf_cutoff_txt[] = { "2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz" }; -static const struct soc_enum da7210_dac_vf_cutoff = - SOC_ENUM_SINGLE(DA7210_DAC_HPF, 4, 8, da7210_vf_cutoff_txt); +static SOC_ENUM_SINGLE_DECL(da7210_dac_vf_cutoff, + DA7210_DAC_HPF, 4, da7210_vf_cutoff_txt); -static const struct soc_enum da7210_adc_vf_cutoff = - SOC_ENUM_SINGLE(DA7210_ADC_HPF, 4, 8, da7210_vf_cutoff_txt); +static SOC_ENUM_SINGLE_DECL(da7210_adc_vf_cutoff, + DA7210_ADC_HPF, 4, da7210_vf_cutoff_txt); static const char *da7210_hp_mode_txt[] = { "Class H", "Class G" }; -static const struct soc_enum da7210_hp_mode_sel = - SOC_ENUM_SINGLE(DA7210_HP_CFG, 0, 2, da7210_hp_mode_txt); +static SOC_ENUM_SINGLE_DECL(da7210_hp_mode_sel, + DA7210_HP_CFG, 0, da7210_hp_mode_txt); /* ALC can be enabled only if noise suppression is disabled */ static int da7210_put_alc_sw(struct snd_kcontrol *kcontrol, -- cgit v1.2.3 From c99f8b216c39b1fbeb8b6830b95e461db551afa9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:06:47 +0100 Subject: ASoC: da7213: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/da7213.c | 151 ++++++++++++++++++++++++---------------------- 1 file changed, 80 insertions(+), 71 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 0c77e7ad7423..439d10387f10 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -63,30 +63,30 @@ static const char * const da7213_voice_hpf_corner_txt[] = { "2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz" }; -static const struct soc_enum da7213_dac_voice_hpf_corner = - SOC_ENUM_SINGLE(DA7213_DAC_FILTERS1, DA7213_VOICE_HPF_CORNER_SHIFT, - DA7213_VOICE_HPF_CORNER_MAX, - da7213_voice_hpf_corner_txt); +static SOC_ENUM_SINGLE_DECL(da7213_dac_voice_hpf_corner, + DA7213_DAC_FILTERS1, + DA7213_VOICE_HPF_CORNER_SHIFT, + da7213_voice_hpf_corner_txt); -static const struct soc_enum da7213_adc_voice_hpf_corner = - SOC_ENUM_SINGLE(DA7213_ADC_FILTERS1, DA7213_VOICE_HPF_CORNER_SHIFT, - DA7213_VOICE_HPF_CORNER_MAX, - da7213_voice_hpf_corner_txt); +static SOC_ENUM_SINGLE_DECL(da7213_adc_voice_hpf_corner, + DA7213_ADC_FILTERS1, + DA7213_VOICE_HPF_CORNER_SHIFT, + da7213_voice_hpf_corner_txt); /* ADC and DAC high pass filter cutoff value */ static const char * const da7213_audio_hpf_corner_txt[] = { "Fs/24000", "Fs/12000", "Fs/6000", "Fs/3000" }; -static const struct soc_enum da7213_dac_audio_hpf_corner = - SOC_ENUM_SINGLE(DA7213_DAC_FILTERS1, DA7213_AUDIO_HPF_CORNER_SHIFT, - DA7213_AUDIO_HPF_CORNER_MAX, - da7213_audio_hpf_corner_txt); +static SOC_ENUM_SINGLE_DECL(da7213_dac_audio_hpf_corner, + DA7213_DAC_FILTERS1 + , DA7213_AUDIO_HPF_CORNER_SHIFT, + da7213_audio_hpf_corner_txt); -static const struct soc_enum da7213_adc_audio_hpf_corner = - SOC_ENUM_SINGLE(DA7213_ADC_FILTERS1, DA7213_AUDIO_HPF_CORNER_SHIFT, - DA7213_AUDIO_HPF_CORNER_MAX, - da7213_audio_hpf_corner_txt); +static SOC_ENUM_SINGLE_DECL(da7213_adc_audio_hpf_corner, + DA7213_ADC_FILTERS1, + DA7213_AUDIO_HPF_CORNER_SHIFT, + da7213_audio_hpf_corner_txt); /* Gain ramping rate value */ static const char * const da7213_gain_ramp_rate_txt[] = { @@ -94,52 +94,50 @@ static const char * const da7213_gain_ramp_rate_txt[] = { "nominal rate / 32" }; -static const struct soc_enum da7213_gain_ramp_rate = - SOC_ENUM_SINGLE(DA7213_GAIN_RAMP_CTRL, DA7213_GAIN_RAMP_RATE_SHIFT, - DA7213_GAIN_RAMP_RATE_MAX, da7213_gain_ramp_rate_txt); +static SOC_ENUM_SINGLE_DECL(da7213_gain_ramp_rate, + DA7213_GAIN_RAMP_CTRL, + DA7213_GAIN_RAMP_RATE_SHIFT, + da7213_gain_ramp_rate_txt); /* DAC noise gate setup time value */ static const char * const da7213_dac_ng_setup_time_txt[] = { "256 samples", "512 samples", "1024 samples", "2048 samples" }; -static const struct soc_enum da7213_dac_ng_setup_time = - SOC_ENUM_SINGLE(DA7213_DAC_NG_SETUP_TIME, - DA7213_DAC_NG_SETUP_TIME_SHIFT, - DA7213_DAC_NG_SETUP_TIME_MAX, - da7213_dac_ng_setup_time_txt); +static SOC_ENUM_SINGLE_DECL(da7213_dac_ng_setup_time, + DA7213_DAC_NG_SETUP_TIME, + DA7213_DAC_NG_SETUP_TIME_SHIFT, + da7213_dac_ng_setup_time_txt); /* DAC noise gate rampup rate value */ static const char * const da7213_dac_ng_rampup_txt[] = { "0.02 ms/dB", "0.16 ms/dB" }; -static const struct soc_enum da7213_dac_ng_rampup_rate = - SOC_ENUM_SINGLE(DA7213_DAC_NG_SETUP_TIME, - DA7213_DAC_NG_RAMPUP_RATE_SHIFT, - DA7213_DAC_NG_RAMP_RATE_MAX, - da7213_dac_ng_rampup_txt); +static SOC_ENUM_SINGLE_DECL(da7213_dac_ng_rampup_rate, + DA7213_DAC_NG_SETUP_TIME, + DA7213_DAC_NG_RAMPUP_RATE_SHIFT, + da7213_dac_ng_rampup_txt); /* DAC noise gate rampdown rate value */ static const char * const da7213_dac_ng_rampdown_txt[] = { "0.64 ms/dB", "20.48 ms/dB" }; -static const struct soc_enum da7213_dac_ng_rampdown_rate = - SOC_ENUM_SINGLE(DA7213_DAC_NG_SETUP_TIME, - DA7213_DAC_NG_RAMPDN_RATE_SHIFT, - DA7213_DAC_NG_RAMP_RATE_MAX, - da7213_dac_ng_rampdown_txt); +static SOC_ENUM_SINGLE_DECL(da7213_dac_ng_rampdown_rate, + DA7213_DAC_NG_SETUP_TIME, + DA7213_DAC_NG_RAMPDN_RATE_SHIFT, + da7213_dac_ng_rampdown_txt); /* DAC soft mute rate value */ static const char * const da7213_dac_soft_mute_rate_txt[] = { "1", "2", "4", "8", "16", "32", "64" }; -static const struct soc_enum da7213_dac_soft_mute_rate = - SOC_ENUM_SINGLE(DA7213_DAC_FILTERS5, DA7213_DAC_SOFTMUTE_RATE_SHIFT, - DA7213_DAC_SOFTMUTE_RATE_MAX, - da7213_dac_soft_mute_rate_txt); +static SOC_ENUM_SINGLE_DECL(da7213_dac_soft_mute_rate, + DA7213_DAC_FILTERS5, + DA7213_DAC_SOFTMUTE_RATE_SHIFT, + da7213_dac_soft_mute_rate_txt); /* ALC Attack Rate select */ static const char * const da7213_alc_attack_rate_txt[] = { @@ -147,9 +145,10 @@ static const char * const da7213_alc_attack_rate_txt[] = { "5632/fs", "11264/fs", "22528/fs", "45056/fs", "90112/fs", "180224/fs" }; -static const struct soc_enum da7213_alc_attack_rate = - SOC_ENUM_SINGLE(DA7213_ALC_CTRL2, DA7213_ALC_ATTACK_SHIFT, - DA7213_ALC_ATTACK_MAX, da7213_alc_attack_rate_txt); +static SOC_ENUM_SINGLE_DECL(da7213_alc_attack_rate, + DA7213_ALC_CTRL2, + DA7213_ALC_ATTACK_SHIFT, + da7213_alc_attack_rate_txt); /* ALC Release Rate select */ static const char * const da7213_alc_release_rate_txt[] = { @@ -157,9 +156,10 @@ static const char * const da7213_alc_release_rate_txt[] = { "11264/fs", "22528/fs", "45056/fs", "90112/fs", "180224/fs" }; -static const struct soc_enum da7213_alc_release_rate = - SOC_ENUM_SINGLE(DA7213_ALC_CTRL2, DA7213_ALC_RELEASE_SHIFT, - DA7213_ALC_RELEASE_MAX, da7213_alc_release_rate_txt); +static SOC_ENUM_SINGLE_DECL(da7213_alc_release_rate, + DA7213_ALC_CTRL2, + DA7213_ALC_RELEASE_SHIFT, + da7213_alc_release_rate_txt); /* ALC Hold Time select */ static const char * const da7213_alc_hold_time_txt[] = { @@ -168,22 +168,25 @@ static const char * const da7213_alc_hold_time_txt[] = { "253952/fs", "507904/fs", "1015808/fs", "2031616/fs" }; -static const struct soc_enum da7213_alc_hold_time = - SOC_ENUM_SINGLE(DA7213_ALC_CTRL3, DA7213_ALC_HOLD_SHIFT, - DA7213_ALC_HOLD_MAX, da7213_alc_hold_time_txt); +static SOC_ENUM_SINGLE_DECL(da7213_alc_hold_time, + DA7213_ALC_CTRL3, + DA7213_ALC_HOLD_SHIFT, + da7213_alc_hold_time_txt); /* ALC Input Signal Tracking rate select */ static const char * const da7213_alc_integ_rate_txt[] = { "1/4", "1/16", "1/256", "1/65536" }; -static const struct soc_enum da7213_alc_integ_attack_rate = - SOC_ENUM_SINGLE(DA7213_ALC_CTRL3, DA7213_ALC_INTEG_ATTACK_SHIFT, - DA7213_ALC_INTEG_MAX, da7213_alc_integ_rate_txt); +static SOC_ENUM_SINGLE_DECL(da7213_alc_integ_attack_rate, + DA7213_ALC_CTRL3, + DA7213_ALC_INTEG_ATTACK_SHIFT, + da7213_alc_integ_rate_txt); -static const struct soc_enum da7213_alc_integ_release_rate = - SOC_ENUM_SINGLE(DA7213_ALC_CTRL3, DA7213_ALC_INTEG_RELEASE_SHIFT, - DA7213_ALC_INTEG_MAX, da7213_alc_integ_rate_txt); +static SOC_ENUM_SINGLE_DECL(da7213_alc_integ_release_rate, + DA7213_ALC_CTRL3, + DA7213_ALC_INTEG_RELEASE_SHIFT, + da7213_alc_integ_rate_txt); /* @@ -584,15 +587,17 @@ static const char * const da7213_mic_amp_in_sel_txt[] = { "Differential", "MIC_P", "MIC_N" }; -static const struct soc_enum da7213_mic_1_amp_in_sel = - SOC_ENUM_SINGLE(DA7213_MIC_1_CTRL, DA7213_MIC_AMP_IN_SEL_SHIFT, - DA7213_MIC_AMP_IN_SEL_MAX, da7213_mic_amp_in_sel_txt); +static SOC_ENUM_SINGLE_DECL(da7213_mic_1_amp_in_sel, + DA7213_MIC_1_CTRL, + DA7213_MIC_AMP_IN_SEL_SHIFT, + da7213_mic_amp_in_sel_txt); static const struct snd_kcontrol_new da7213_mic_1_amp_in_sel_mux = SOC_DAPM_ENUM("Mic 1 Amp Source MUX", da7213_mic_1_amp_in_sel); -static const struct soc_enum da7213_mic_2_amp_in_sel = - SOC_ENUM_SINGLE(DA7213_MIC_2_CTRL, DA7213_MIC_AMP_IN_SEL_SHIFT, - DA7213_MIC_AMP_IN_SEL_MAX, da7213_mic_amp_in_sel_txt); +static SOC_ENUM_SINGLE_DECL(da7213_mic_2_amp_in_sel, + DA7213_MIC_2_CTRL, + DA7213_MIC_AMP_IN_SEL_SHIFT, + da7213_mic_amp_in_sel_txt); static const struct snd_kcontrol_new da7213_mic_2_amp_in_sel_mux = SOC_DAPM_ENUM("Mic 2 Amp Source MUX", da7213_mic_2_amp_in_sel); @@ -601,15 +606,17 @@ static const char * const da7213_dai_src_txt[] = { "ADC Left", "ADC Right", "DAI Input Left", "DAI Input Right" }; -static const struct soc_enum da7213_dai_l_src = - SOC_ENUM_SINGLE(DA7213_DIG_ROUTING_DAI, DA7213_DAI_L_SRC_SHIFT, - DA7213_DAI_SRC_MAX, da7213_dai_src_txt); +static SOC_ENUM_SINGLE_DECL(da7213_dai_l_src, + DA7213_DIG_ROUTING_DAI, + DA7213_DAI_L_SRC_SHIFT, + da7213_dai_src_txt); static const struct snd_kcontrol_new da7213_dai_l_src_mux = SOC_DAPM_ENUM("DAI Left Source MUX", da7213_dai_l_src); -static const struct soc_enum da7213_dai_r_src = - SOC_ENUM_SINGLE(DA7213_DIG_ROUTING_DAI, DA7213_DAI_R_SRC_SHIFT, - DA7213_DAI_SRC_MAX, da7213_dai_src_txt); +static SOC_ENUM_SINGLE_DECL(da7213_dai_r_src, + DA7213_DIG_ROUTING_DAI, + DA7213_DAI_R_SRC_SHIFT, + da7213_dai_src_txt); static const struct snd_kcontrol_new da7213_dai_r_src_mux = SOC_DAPM_ENUM("DAI Right Source MUX", da7213_dai_r_src); @@ -619,15 +626,17 @@ static const char * const da7213_dac_src_txt[] = { "DAI Input Right" }; -static const struct soc_enum da7213_dac_l_src = - SOC_ENUM_SINGLE(DA7213_DIG_ROUTING_DAC, DA7213_DAC_L_SRC_SHIFT, - DA7213_DAC_SRC_MAX, da7213_dac_src_txt); +static SOC_ENUM_SINGLE_DECL(da7213_dac_l_src, + DA7213_DIG_ROUTING_DAC, + DA7213_DAC_L_SRC_SHIFT, + da7213_dac_src_txt); static const struct snd_kcontrol_new da7213_dac_l_src_mux = SOC_DAPM_ENUM("DAC Left Source MUX", da7213_dac_l_src); -static const struct soc_enum da7213_dac_r_src = - SOC_ENUM_SINGLE(DA7213_DIG_ROUTING_DAC, DA7213_DAC_R_SRC_SHIFT, - DA7213_DAC_SRC_MAX, da7213_dac_src_txt); +static SOC_ENUM_SINGLE_DECL(da7213_dac_r_src, + DA7213_DIG_ROUTING_DAC, + DA7213_DAC_R_SRC_SHIFT, + da7213_dac_src_txt); static const struct snd_kcontrol_new da7213_dac_r_src_mux = SOC_DAPM_ENUM("DAC Right Source MUX", da7213_dac_r_src); -- cgit v1.2.3 From 6415e307b1d270e4e5f7ee7bd9aac4ac4f1daf65 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:09:52 +0100 Subject: ASoC: lm4857: Use SOC_ENUM_SINGLE_EXT_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/lm4857.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/lm4857.c b/sound/soc/codecs/lm4857.c index 0e5743ea79df..4f048db9f55f 100644 --- a/sound/soc/codecs/lm4857.c +++ b/sound/soc/codecs/lm4857.c @@ -101,8 +101,7 @@ static const char *lm4857_mode[] = { "Headphone", }; -static const struct soc_enum lm4857_mode_enum = - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(lm4857_mode), lm4857_mode); +static SOC_ENUM_SINGLE_EXT_DECL(lm4857_mode_enum, lm4857_mode); static const struct snd_soc_dapm_widget lm4857_dapm_widgets[] = { SND_SOC_DAPM_INPUT("IN"), -- cgit v1.2.3 From 9839ce9360c4410bdf30d2670a2329c444160f02 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:07:25 +0100 Subject: ASoC: da9055: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/da9055.c | 84 +++++++++++++++++++++++------------------------ 1 file changed, 42 insertions(+), 42 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index be31f3cfd46e..f118daa91234 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -323,22 +323,22 @@ static const char * const da9055_hpf_cutoff_txt[] = { "Fs/24000", "Fs/12000", "Fs/6000", "Fs/3000" }; -static const struct soc_enum da9055_dac_hpf_cutoff = - SOC_ENUM_SINGLE(DA9055_DAC_FILTERS1, 4, 4, da9055_hpf_cutoff_txt); +static SOC_ENUM_SINGLE_DECL(da9055_dac_hpf_cutoff, + DA9055_DAC_FILTERS1, 4, da9055_hpf_cutoff_txt); -static const struct soc_enum da9055_adc_hpf_cutoff = - SOC_ENUM_SINGLE(DA9055_ADC_FILTERS1, 4, 4, da9055_hpf_cutoff_txt); +static SOC_ENUM_SINGLE_DECL(da9055_adc_hpf_cutoff, + DA9055_ADC_FILTERS1, 4, da9055_hpf_cutoff_txt); /* ADC and DAC voice mode (8kHz) high pass cutoff value */ static const char * const da9055_vf_cutoff_txt[] = { "2.5Hz", "25Hz", "50Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz" }; -static const struct soc_enum da9055_dac_vf_cutoff = - SOC_ENUM_SINGLE(DA9055_DAC_FILTERS1, 0, 8, da9055_vf_cutoff_txt); +static SOC_ENUM_SINGLE_DECL(da9055_dac_vf_cutoff, + DA9055_DAC_FILTERS1, 0, da9055_vf_cutoff_txt); -static const struct soc_enum da9055_adc_vf_cutoff = - SOC_ENUM_SINGLE(DA9055_ADC_FILTERS1, 0, 8, da9055_vf_cutoff_txt); +static SOC_ENUM_SINGLE_DECL(da9055_adc_vf_cutoff, + DA9055_ADC_FILTERS1, 0, da9055_vf_cutoff_txt); /* Gain ramping rate value */ static const char * const da9055_gain_ramping_txt[] = { @@ -346,44 +346,44 @@ static const char * const da9055_gain_ramping_txt[] = { "nominal rate / 8" }; -static const struct soc_enum da9055_gain_ramping_rate = - SOC_ENUM_SINGLE(DA9055_GAIN_RAMP_CTRL, 0, 4, da9055_gain_ramping_txt); +static SOC_ENUM_SINGLE_DECL(da9055_gain_ramping_rate, + DA9055_GAIN_RAMP_CTRL, 0, da9055_gain_ramping_txt); /* DAC noise gate setup time value */ static const char * const da9055_dac_ng_setup_time_txt[] = { "256 samples", "512 samples", "1024 samples", "2048 samples" }; -static const struct soc_enum da9055_dac_ng_setup_time = - SOC_ENUM_SINGLE(DA9055_DAC_NG_SETUP_TIME, 0, 4, - da9055_dac_ng_setup_time_txt); +static SOC_ENUM_SINGLE_DECL(da9055_dac_ng_setup_time, + DA9055_DAC_NG_SETUP_TIME, 0, + da9055_dac_ng_setup_time_txt); /* DAC noise gate rampup rate value */ static const char * const da9055_dac_ng_rampup_txt[] = { "0.02 ms/dB", "0.16 ms/dB" }; -static const struct soc_enum da9055_dac_ng_rampup_rate = - SOC_ENUM_SINGLE(DA9055_DAC_NG_SETUP_TIME, 2, 2, - da9055_dac_ng_rampup_txt); +static SOC_ENUM_SINGLE_DECL(da9055_dac_ng_rampup_rate, + DA9055_DAC_NG_SETUP_TIME, 2, + da9055_dac_ng_rampup_txt); /* DAC noise gate rampdown rate value */ static const char * const da9055_dac_ng_rampdown_txt[] = { "0.64 ms/dB", "20.48 ms/dB" }; -static const struct soc_enum da9055_dac_ng_rampdown_rate = - SOC_ENUM_SINGLE(DA9055_DAC_NG_SETUP_TIME, 3, 2, - da9055_dac_ng_rampdown_txt); +static SOC_ENUM_SINGLE_DECL(da9055_dac_ng_rampdown_rate, + DA9055_DAC_NG_SETUP_TIME, 3, + da9055_dac_ng_rampdown_txt); /* DAC soft mute rate value */ static const char * const da9055_dac_soft_mute_rate_txt[] = { "1", "2", "4", "8", "16", "32", "64" }; -static const struct soc_enum da9055_dac_soft_mute_rate = - SOC_ENUM_SINGLE(DA9055_DAC_FILTERS5, 4, 7, - da9055_dac_soft_mute_rate_txt); +static SOC_ENUM_SINGLE_DECL(da9055_dac_soft_mute_rate, + DA9055_DAC_FILTERS5, 4, + da9055_dac_soft_mute_rate_txt); /* DAC routing select */ static const char * const da9055_dac_src_txt[] = { @@ -391,40 +391,40 @@ static const char * const da9055_dac_src_txt[] = { "AIF input right" }; -static const struct soc_enum da9055_dac_l_src = - SOC_ENUM_SINGLE(DA9055_DIG_ROUTING_DAC, 0, 4, da9055_dac_src_txt); +static SOC_ENUM_SINGLE_DECL(da9055_dac_l_src, + DA9055_DIG_ROUTING_DAC, 0, da9055_dac_src_txt); -static const struct soc_enum da9055_dac_r_src = - SOC_ENUM_SINGLE(DA9055_DIG_ROUTING_DAC, 4, 4, da9055_dac_src_txt); +static SOC_ENUM_SINGLE_DECL(da9055_dac_r_src, + DA9055_DIG_ROUTING_DAC, 4, da9055_dac_src_txt); /* MIC PGA Left source select */ static const char * const da9055_mic_l_src_txt[] = { "MIC1_P_N", "MIC1_P", "MIC1_N", "MIC2_L" }; -static const struct soc_enum da9055_mic_l_src = - SOC_ENUM_SINGLE(DA9055_MIXIN_L_SELECT, 4, 4, da9055_mic_l_src_txt); +static SOC_ENUM_SINGLE_DECL(da9055_mic_l_src, + DA9055_MIXIN_L_SELECT, 4, da9055_mic_l_src_txt); /* MIC PGA Right source select */ static const char * const da9055_mic_r_src_txt[] = { "MIC2_R_L", "MIC2_R", "MIC2_L" }; -static const struct soc_enum da9055_mic_r_src = - SOC_ENUM_SINGLE(DA9055_MIXIN_R_SELECT, 4, 3, da9055_mic_r_src_txt); +static SOC_ENUM_SINGLE_DECL(da9055_mic_r_src, + DA9055_MIXIN_R_SELECT, 4, da9055_mic_r_src_txt); /* ALC Input Signal Tracking rate select */ static const char * const da9055_signal_tracking_rate_txt[] = { "1/4", "1/16", "1/256", "1/65536" }; -static const struct soc_enum da9055_integ_attack_rate = - SOC_ENUM_SINGLE(DA9055_ALC_CTRL3, 4, 4, - da9055_signal_tracking_rate_txt); +static SOC_ENUM_SINGLE_DECL(da9055_integ_attack_rate, + DA9055_ALC_CTRL3, 4, + da9055_signal_tracking_rate_txt); -static const struct soc_enum da9055_integ_release_rate = - SOC_ENUM_SINGLE(DA9055_ALC_CTRL3, 6, 4, - da9055_signal_tracking_rate_txt); +static SOC_ENUM_SINGLE_DECL(da9055_integ_release_rate, + DA9055_ALC_CTRL3, 6, + da9055_signal_tracking_rate_txt); /* ALC Attack Rate select */ static const char * const da9055_attack_rate_txt[] = { @@ -432,8 +432,8 @@ static const char * const da9055_attack_rate_txt[] = { "5632/fs", "11264/fs", "22528/fs", "45056/fs", "90112/fs", "180224/fs" }; -static const struct soc_enum da9055_attack_rate = - SOC_ENUM_SINGLE(DA9055_ALC_CTRL2, 0, 13, da9055_attack_rate_txt); +static SOC_ENUM_SINGLE_DECL(da9055_attack_rate, + DA9055_ALC_CTRL2, 0, da9055_attack_rate_txt); /* ALC Release Rate select */ static const char * const da9055_release_rate_txt[] = { @@ -441,8 +441,8 @@ static const char * const da9055_release_rate_txt[] = { "11264/fs", "22528/fs", "45056/fs", "90112/fs", "180224/fs" }; -static const struct soc_enum da9055_release_rate = - SOC_ENUM_SINGLE(DA9055_ALC_CTRL2, 4, 11, da9055_release_rate_txt); +static SOC_ENUM_SINGLE_DECL(da9055_release_rate, + DA9055_ALC_CTRL2, 4, da9055_release_rate_txt); /* ALC Hold Time select */ static const char * const da9055_hold_time_txt[] = { @@ -451,8 +451,8 @@ static const char * const da9055_hold_time_txt[] = { "253952/fs", "507904/fs", "1015808/fs", "2031616/fs" }; -static const struct soc_enum da9055_hold_time = - SOC_ENUM_SINGLE(DA9055_ALC_CTRL3, 0, 16, da9055_hold_time_txt); +static SOC_ENUM_SINGLE_DECL(da9055_hold_time, + DA9055_ALC_CTRL3, 0, da9055_hold_time_txt); static int da9055_get_alc_data(struct snd_soc_codec *codec, u8 reg_val) { -- cgit v1.2.3 From a0628934d6d3fcca5588fd9617270f63c5387f1b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:12:32 +0100 Subject: ASoC: max98088: Use SOC_*_ENUM_SINGLE_DECL() Just replace with the helper macros. No functional change at all. Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/max98088.c | 39 +++++++++++++++++++-------------------- 1 file changed, 19 insertions(+), 20 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index ee660e2d3df3..25ce1355b8fb 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -597,28 +597,27 @@ static const unsigned int max98088_exmode_values[] = { 0x00, 0x43, 0x10, 0x20, 0x30, 0x40, 0x11, 0x22, 0x32 }; -static const struct soc_enum max98088_exmode_enum = - SOC_VALUE_ENUM_SINGLE(M98088_REG_41_SPKDHP, 0, 127, - ARRAY_SIZE(max98088_exmode_texts), - max98088_exmode_texts, - max98088_exmode_values); +static SOC_VALUE_ENUM_SINGLE_DECL(max98088_exmode_enum, + M98088_REG_41_SPKDHP, 0, 127, + max98088_exmode_texts, + max98088_exmode_values); static const char *max98088_ex_thresh[] = { /* volts PP */ "0.6", "1.2", "1.8", "2.4", "3.0", "3.6", "4.2", "4.8"}; -static const struct soc_enum max98088_ex_thresh_enum[] = { - SOC_ENUM_SINGLE(M98088_REG_42_SPKDHP_THRESH, 0, 8, - max98088_ex_thresh), -}; +static SOC_ENUM_SINGLE_DECL(max98088_ex_thresh_enum, + M98088_REG_42_SPKDHP_THRESH, 0, + max98088_ex_thresh); static const char *max98088_fltr_mode[] = {"Voice", "Music" }; -static const struct soc_enum max98088_filter_mode_enum[] = { - SOC_ENUM_SINGLE(M98088_REG_18_DAI1_FILTERS, 7, 2, max98088_fltr_mode), -}; +static SOC_ENUM_SINGLE_DECL(max98088_filter_mode_enum, + M98088_REG_18_DAI1_FILTERS, 7, + max98088_fltr_mode); static const char *max98088_extmic_text[] = { "None", "MIC1", "MIC2" }; -static const struct soc_enum max98088_extmic_enum = - SOC_ENUM_SINGLE(M98088_REG_48_CFG_MIC, 0, 3, max98088_extmic_text); +static SOC_ENUM_SINGLE_DECL(max98088_extmic_enum, + M98088_REG_48_CFG_MIC, 0, + max98088_extmic_text); static const struct snd_kcontrol_new max98088_extmic_mux = SOC_DAPM_ENUM("External MIC Mux", max98088_extmic_enum); @@ -626,12 +625,12 @@ static const struct snd_kcontrol_new max98088_extmic_mux = static const char *max98088_dai1_fltr[] = { "Off", "fc=258/fs=16k", "fc=500/fs=16k", "fc=258/fs=8k", "fc=500/fs=8k", "fc=200"}; -static const struct soc_enum max98088_dai1_dac_filter_enum[] = { - SOC_ENUM_SINGLE(M98088_REG_18_DAI1_FILTERS, 0, 6, max98088_dai1_fltr), -}; -static const struct soc_enum max98088_dai1_adc_filter_enum[] = { - SOC_ENUM_SINGLE(M98088_REG_18_DAI1_FILTERS, 4, 6, max98088_dai1_fltr), -}; +static SOC_ENUM_SINGLE_DECL(max98088_dai1_dac_filter_enum, + M98088_REG_18_DAI1_FILTERS, 0, + max98088_dai1_fltr); +static SOC_ENUM_SINGLE_DECL(max98088_dai1_adc_filter_enum, + M98088_REG_18_DAI1_FILTERS, 4, + max98088_dai1_fltr); static int max98088_mic1pre_set(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -- cgit v1.2.3 From 2907cbcc120e0c388df499fcb1be7d093ade3993 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:14:13 +0100 Subject: ASoC: max98090: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 172 +++++++++++++++++++++++++------------------- 1 file changed, 97 insertions(+), 75 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 51f9b3d16b41..c7b9e901bdac 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -512,65 +512,75 @@ static const char *max98090_perf_pwr_text[] = static const char *max98090_pwr_perf_text[] = { "Low Power", "High Performance" }; -static const struct soc_enum max98090_vcmbandgap_enum = - SOC_ENUM_SINGLE(M98090_REG_BIAS_CONTROL, M98090_VCM_MODE_SHIFT, - ARRAY_SIZE(max98090_pwr_perf_text), max98090_pwr_perf_text); +static SOC_ENUM_SINGLE_DECL(max98090_vcmbandgap_enum, + M98090_REG_BIAS_CONTROL, + M98090_VCM_MODE_SHIFT, + max98090_pwr_perf_text); static const char *max98090_osr128_text[] = { "64*fs", "128*fs" }; -static const struct soc_enum max98090_osr128_enum = - SOC_ENUM_SINGLE(M98090_REG_ADC_CONTROL, M98090_OSR128_SHIFT, - ARRAY_SIZE(max98090_osr128_text), max98090_osr128_text); +static SOC_ENUM_SINGLE_DECL(max98090_osr128_enum, + M98090_REG_ADC_CONTROL, + M98090_OSR128_SHIFT, + max98090_osr128_text); static const char *max98090_mode_text[] = { "Voice", "Music" }; -static const struct soc_enum max98090_mode_enum = - SOC_ENUM_SINGLE(M98090_REG_FILTER_CONFIG, M98090_MODE_SHIFT, - ARRAY_SIZE(max98090_mode_text), max98090_mode_text); +static SOC_ENUM_SINGLE_DECL(max98090_mode_enum, + M98090_REG_FILTER_CONFIG, + M98090_MODE_SHIFT, + max98090_mode_text); -static const struct soc_enum max98090_filter_dmic34mode_enum = - SOC_ENUM_SINGLE(M98090_REG_FILTER_CONFIG, - M98090_FLT_DMIC34MODE_SHIFT, - ARRAY_SIZE(max98090_mode_text), max98090_mode_text); +static SOC_ENUM_SINGLE_DECL(max98090_filter_dmic34mode_enum, + M98090_REG_FILTER_CONFIG, + M98090_FLT_DMIC34MODE_SHIFT, + max98090_mode_text); static const char *max98090_drcatk_text[] = { "0.5ms", "1ms", "5ms", "10ms", "25ms", "50ms", "100ms", "200ms" }; -static const struct soc_enum max98090_drcatk_enum = - SOC_ENUM_SINGLE(M98090_REG_DRC_TIMING, M98090_DRCATK_SHIFT, - ARRAY_SIZE(max98090_drcatk_text), max98090_drcatk_text); +static SOC_ENUM_SINGLE_DECL(max98090_drcatk_enum, + M98090_REG_DRC_TIMING, + M98090_DRCATK_SHIFT, + max98090_drcatk_text); static const char *max98090_drcrls_text[] = { "8s", "4s", "2s", "1s", "0.5s", "0.25s", "0.125s", "0.0625s" }; -static const struct soc_enum max98090_drcrls_enum = - SOC_ENUM_SINGLE(M98090_REG_DRC_TIMING, M98090_DRCRLS_SHIFT, - ARRAY_SIZE(max98090_drcrls_text), max98090_drcrls_text); +static SOC_ENUM_SINGLE_DECL(max98090_drcrls_enum, + M98090_REG_DRC_TIMING, + M98090_DRCRLS_SHIFT, + max98090_drcrls_text); static const char *max98090_alccmp_text[] = { "1:1", "1:1.5", "1:2", "1:4", "1:INF" }; -static const struct soc_enum max98090_alccmp_enum = - SOC_ENUM_SINGLE(M98090_REG_DRC_COMPRESSOR, M98090_DRCCMP_SHIFT, - ARRAY_SIZE(max98090_alccmp_text), max98090_alccmp_text); +static SOC_ENUM_SINGLE_DECL(max98090_alccmp_enum, + M98090_REG_DRC_COMPRESSOR, + M98090_DRCCMP_SHIFT, + max98090_alccmp_text); static const char *max98090_drcexp_text[] = { "1:1", "2:1", "3:1" }; -static const struct soc_enum max98090_drcexp_enum = - SOC_ENUM_SINGLE(M98090_REG_DRC_EXPANDER, M98090_DRCEXP_SHIFT, - ARRAY_SIZE(max98090_drcexp_text), max98090_drcexp_text); +static SOC_ENUM_SINGLE_DECL(max98090_drcexp_enum, + M98090_REG_DRC_EXPANDER, + M98090_DRCEXP_SHIFT, + max98090_drcexp_text); -static const struct soc_enum max98090_dac_perfmode_enum = - SOC_ENUM_SINGLE(M98090_REG_DAC_CONTROL, M98090_PERFMODE_SHIFT, - ARRAY_SIZE(max98090_perf_pwr_text), max98090_perf_pwr_text); +static SOC_ENUM_SINGLE_DECL(max98090_dac_perfmode_enum, + M98090_REG_DAC_CONTROL, + M98090_PERFMODE_SHIFT, + max98090_perf_pwr_text); -static const struct soc_enum max98090_dachp_enum = - SOC_ENUM_SINGLE(M98090_REG_DAC_CONTROL, M98090_DACHP_SHIFT, - ARRAY_SIZE(max98090_pwr_perf_text), max98090_pwr_perf_text); +static SOC_ENUM_SINGLE_DECL(max98090_dachp_enum, + M98090_REG_DAC_CONTROL, + M98090_DACHP_SHIFT, + max98090_pwr_perf_text); -static const struct soc_enum max98090_adchp_enum = - SOC_ENUM_SINGLE(M98090_REG_ADC_CONTROL, M98090_ADCHP_SHIFT, - ARRAY_SIZE(max98090_pwr_perf_text), max98090_pwr_perf_text); +static SOC_ENUM_SINGLE_DECL(max98090_adchp_enum, + M98090_REG_ADC_CONTROL, + M98090_ADCHP_SHIFT, + max98090_pwr_perf_text); static const struct snd_kcontrol_new max98090_snd_controls[] = { SOC_ENUM("MIC Bias VCM Bandgap", max98090_vcmbandgap_enum), @@ -841,39 +851,42 @@ static int max98090_micinput_event(struct snd_soc_dapm_widget *w, static const char *mic1_mux_text[] = { "IN12", "IN56" }; -static const struct soc_enum mic1_mux_enum = - SOC_ENUM_SINGLE(M98090_REG_INPUT_MODE, M98090_EXTMIC1_SHIFT, - ARRAY_SIZE(mic1_mux_text), mic1_mux_text); +static SOC_ENUM_SINGLE_DECL(mic1_mux_enum, + M98090_REG_INPUT_MODE, + M98090_EXTMIC1_SHIFT, + mic1_mux_text); static const struct snd_kcontrol_new max98090_mic1_mux = SOC_DAPM_ENUM("MIC1 Mux", mic1_mux_enum); static const char *mic2_mux_text[] = { "IN34", "IN56" }; -static const struct soc_enum mic2_mux_enum = - SOC_ENUM_SINGLE(M98090_REG_INPUT_MODE, M98090_EXTMIC2_SHIFT, - ARRAY_SIZE(mic2_mux_text), mic2_mux_text); +static SOC_ENUM_SINGLE_DECL(mic2_mux_enum, + M98090_REG_INPUT_MODE, + M98090_EXTMIC2_SHIFT, + mic2_mux_text); static const struct snd_kcontrol_new max98090_mic2_mux = SOC_DAPM_ENUM("MIC2 Mux", mic2_mux_enum); static const char *dmic_mux_text[] = { "ADC", "DMIC" }; -static const struct soc_enum dmic_mux_enum = - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(dmic_mux_text), dmic_mux_text); +static SOC_ENUM_SINGLE_EXT_DECL(dmic_mux_enum, dmic_mux_text); static const struct snd_kcontrol_new max98090_dmic_mux = SOC_DAPM_ENUM_VIRT("DMIC Mux", dmic_mux_enum); static const char *max98090_micpre_text[] = { "Off", "On" }; -static const struct soc_enum max98090_pa1en_enum = - SOC_ENUM_SINGLE(M98090_REG_MIC1_INPUT_LEVEL, M98090_MIC_PA1EN_SHIFT, - ARRAY_SIZE(max98090_micpre_text), max98090_micpre_text); +static SOC_ENUM_SINGLE_DECL(max98090_pa1en_enum, + M98090_REG_MIC1_INPUT_LEVEL, + M98090_MIC_PA1EN_SHIFT, + max98090_micpre_text); -static const struct soc_enum max98090_pa2en_enum = - SOC_ENUM_SINGLE(M98090_REG_MIC2_INPUT_LEVEL, M98090_MIC_PA2EN_SHIFT, - ARRAY_SIZE(max98090_micpre_text), max98090_micpre_text); +static SOC_ENUM_SINGLE_DECL(max98090_pa2en_enum, + M98090_REG_MIC2_INPUT_LEVEL, + M98090_MIC_PA2EN_SHIFT, + max98090_micpre_text); /* LINEA mixer switch */ static const struct snd_kcontrol_new max98090_linea_mixer_controls[] = { @@ -937,13 +950,15 @@ static const struct snd_kcontrol_new max98090_right_adc_mixer_controls[] = { static const char *lten_mux_text[] = { "Normal", "Loopthrough" }; -static const struct soc_enum ltenl_mux_enum = - SOC_ENUM_SINGLE(M98090_REG_IO_CONFIGURATION, M98090_LTEN_SHIFT, - ARRAY_SIZE(lten_mux_text), lten_mux_text); +static SOC_ENUM_SINGLE_DECL(ltenl_mux_enum, + M98090_REG_IO_CONFIGURATION, + M98090_LTEN_SHIFT, + lten_mux_text); -static const struct soc_enum ltenr_mux_enum = - SOC_ENUM_SINGLE(M98090_REG_IO_CONFIGURATION, M98090_LTEN_SHIFT, - ARRAY_SIZE(lten_mux_text), lten_mux_text); +static SOC_ENUM_SINGLE_DECL(ltenr_mux_enum, + M98090_REG_IO_CONFIGURATION, + M98090_LTEN_SHIFT, + lten_mux_text); static const struct snd_kcontrol_new max98090_ltenl_mux = SOC_DAPM_ENUM("LTENL Mux", ltenl_mux_enum); @@ -953,13 +968,15 @@ static const struct snd_kcontrol_new max98090_ltenr_mux = static const char *lben_mux_text[] = { "Normal", "Loopback" }; -static const struct soc_enum lbenl_mux_enum = - SOC_ENUM_SINGLE(M98090_REG_IO_CONFIGURATION, M98090_LBEN_SHIFT, - ARRAY_SIZE(lben_mux_text), lben_mux_text); +static SOC_ENUM_SINGLE_DECL(lbenl_mux_enum, + M98090_REG_IO_CONFIGURATION, + M98090_LBEN_SHIFT, + lben_mux_text); -static const struct soc_enum lbenr_mux_enum = - SOC_ENUM_SINGLE(M98090_REG_IO_CONFIGURATION, M98090_LBEN_SHIFT, - ARRAY_SIZE(lben_mux_text), lben_mux_text); +static SOC_ENUM_SINGLE_DECL(lbenr_mux_enum, + M98090_REG_IO_CONFIGURATION, + M98090_LBEN_SHIFT, + lben_mux_text); static const struct snd_kcontrol_new max98090_lbenl_mux = SOC_DAPM_ENUM("LBENL Mux", lbenl_mux_enum); @@ -971,13 +988,15 @@ static const char *stenl_mux_text[] = { "Normal", "Sidetone Left" }; static const char *stenr_mux_text[] = { "Normal", "Sidetone Right" }; -static const struct soc_enum stenl_mux_enum = - SOC_ENUM_SINGLE(M98090_REG_ADC_SIDETONE, M98090_DSTSL_SHIFT, - ARRAY_SIZE(stenl_mux_text), stenl_mux_text); +static SOC_ENUM_SINGLE_DECL(stenl_mux_enum, + M98090_REG_ADC_SIDETONE, + M98090_DSTSL_SHIFT, + stenl_mux_text); -static const struct soc_enum stenr_mux_enum = - SOC_ENUM_SINGLE(M98090_REG_ADC_SIDETONE, M98090_DSTSR_SHIFT, - ARRAY_SIZE(stenr_mux_text), stenr_mux_text); +static SOC_ENUM_SINGLE_DECL(stenr_mux_enum, + M98090_REG_ADC_SIDETONE, + M98090_DSTSR_SHIFT, + stenr_mux_text); static const struct snd_kcontrol_new max98090_stenl_mux = SOC_DAPM_ENUM("STENL Mux", stenl_mux_enum); @@ -1085,9 +1104,10 @@ static const struct snd_kcontrol_new max98090_right_rcv_mixer_controls[] = { static const char *linmod_mux_text[] = { "Left Only", "Left and Right" }; -static const struct soc_enum linmod_mux_enum = - SOC_ENUM_SINGLE(M98090_REG_LOUTR_MIXER, M98090_LINMOD_SHIFT, - ARRAY_SIZE(linmod_mux_text), linmod_mux_text); +static SOC_ENUM_SINGLE_DECL(linmod_mux_enum, + M98090_REG_LOUTR_MIXER, + M98090_LINMOD_SHIFT, + linmod_mux_text); static const struct snd_kcontrol_new max98090_linmod_mux = SOC_DAPM_ENUM("LINMOD Mux", linmod_mux_enum); @@ -1097,16 +1117,18 @@ static const char *mixhpsel_mux_text[] = { "DAC Only", "HP Mixer" }; /* * This is a mux as it selects the HP output, but to DAPM it is a Mixer enable */ -static const struct soc_enum mixhplsel_mux_enum = - SOC_ENUM_SINGLE(M98090_REG_HP_CONTROL, M98090_MIXHPLSEL_SHIFT, - ARRAY_SIZE(mixhpsel_mux_text), mixhpsel_mux_text); +static SOC_ENUM_SINGLE_DECL(mixhplsel_mux_enum, + M98090_REG_HP_CONTROL, + M98090_MIXHPLSEL_SHIFT, + mixhpsel_mux_text); static const struct snd_kcontrol_new max98090_mixhplsel_mux = SOC_DAPM_ENUM("MIXHPLSEL Mux", mixhplsel_mux_enum); -static const struct soc_enum mixhprsel_mux_enum = - SOC_ENUM_SINGLE(M98090_REG_HP_CONTROL, M98090_MIXHPRSEL_SHIFT, - ARRAY_SIZE(mixhpsel_mux_text), mixhpsel_mux_text); +static SOC_ENUM_SINGLE_DECL(mixhprsel_mux_enum, + M98090_REG_HP_CONTROL, + M98090_MIXHPRSEL_SHIFT, + mixhpsel_mux_text); static const struct snd_kcontrol_new max98090_mixhprsel_mux = SOC_DAPM_ENUM("MIXHPRSEL Mux", mixhprsel_mux_enum); -- cgit v1.2.3 From af1f0a50823a3eb8bb7a11731c02b77d145fff70 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:15:26 +0100 Subject: ASoC: max98095: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/max98095.c | 50 ++++++++++++++++++++++++--------------------- 1 file changed, 27 insertions(+), 23 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 3ba1170ebb53..ddbb4164e8d2 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -560,25 +560,27 @@ static void m98095_biquad_band(struct snd_soc_codec *codec, unsigned int dai, } static const char * const max98095_fltr_mode[] = { "Voice", "Music" }; -static const struct soc_enum max98095_dai1_filter_mode_enum[] = { - SOC_ENUM_SINGLE(M98095_02E_DAI1_FILTERS, 7, 2, max98095_fltr_mode), -}; -static const struct soc_enum max98095_dai2_filter_mode_enum[] = { - SOC_ENUM_SINGLE(M98095_038_DAI2_FILTERS, 7, 2, max98095_fltr_mode), -}; +static SOC_ENUM_SINGLE_DECL(max98095_dai1_filter_mode_enum, + M98095_02E_DAI1_FILTERS, 7, + max98095_fltr_mode); +static SOC_ENUM_SINGLE_DECL(max98095_dai2_filter_mode_enum, + M98095_038_DAI2_FILTERS, 7, + max98095_fltr_mode); static const char * const max98095_extmic_text[] = { "None", "MIC1", "MIC2" }; -static const struct soc_enum max98095_extmic_enum = - SOC_ENUM_SINGLE(M98095_087_CFG_MIC, 0, 3, max98095_extmic_text); +static SOC_ENUM_SINGLE_DECL(max98095_extmic_enum, + M98095_087_CFG_MIC, 0, + max98095_extmic_text); static const struct snd_kcontrol_new max98095_extmic_mux = SOC_DAPM_ENUM("External MIC Mux", max98095_extmic_enum); static const char * const max98095_linein_text[] = { "INA", "INB" }; -static const struct soc_enum max98095_linein_enum = - SOC_ENUM_SINGLE(M98095_086_CFG_LINE, 6, 2, max98095_linein_text); +static SOC_ENUM_SINGLE_DECL(max98095_linein_enum, + M98095_086_CFG_LINE, 6, + max98095_linein_text); static const struct snd_kcontrol_new max98095_linein_mux = SOC_DAPM_ENUM("Linein Input Mux", max98095_linein_enum); @@ -586,24 +588,26 @@ static const struct snd_kcontrol_new max98095_linein_mux = static const char * const max98095_line_mode_text[] = { "Stereo", "Differential"}; -static const struct soc_enum max98095_linein_mode_enum = - SOC_ENUM_SINGLE(M98095_086_CFG_LINE, 7, 2, max98095_line_mode_text); +static SOC_ENUM_SINGLE_DECL(max98095_linein_mode_enum, + M98095_086_CFG_LINE, 7, + max98095_line_mode_text); -static const struct soc_enum max98095_lineout_mode_enum = - SOC_ENUM_SINGLE(M98095_086_CFG_LINE, 4, 2, max98095_line_mode_text); +static SOC_ENUM_SINGLE_DECL(max98095_lineout_mode_enum, + M98095_086_CFG_LINE, 4, + max98095_line_mode_text); static const char * const max98095_dai_fltr[] = { "Off", "Elliptical-HPF-16k", "Butterworth-HPF-16k", "Elliptical-HPF-8k", "Butterworth-HPF-8k", "Butterworth-HPF-Fs/240"}; -static const struct soc_enum max98095_dai1_dac_filter_enum[] = { - SOC_ENUM_SINGLE(M98095_02E_DAI1_FILTERS, 0, 6, max98095_dai_fltr), -}; -static const struct soc_enum max98095_dai2_dac_filter_enum[] = { - SOC_ENUM_SINGLE(M98095_038_DAI2_FILTERS, 0, 6, max98095_dai_fltr), -}; -static const struct soc_enum max98095_dai3_dac_filter_enum[] = { - SOC_ENUM_SINGLE(M98095_042_DAI3_FILTERS, 0, 6, max98095_dai_fltr), -}; +static SOC_ENUM_SINGLE_DECL(max98095_dai1_dac_filter_enum, + M98095_02E_DAI1_FILTERS, 0, + max98095_dai_fltr); +static SOC_ENUM_SINGLE_DECL(max98095_dai2_dac_filter_enum, + M98095_038_DAI2_FILTERS, 0, + max98095_dai_fltr); +static SOC_ENUM_SINGLE_DECL(max98095_dai3_dac_filter_enum, + M98095_042_DAI3_FILTERS, 0, + max98095_dai_fltr); static int max98095_mic1pre_set(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -- cgit v1.2.3 From 1bf1b8cf4faf279c3643c5c8045bec53b047ca9a Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 18 Feb 2014 15:22:16 +0000 Subject: ASoC: adav80x: Update locking around use of DAPM pin API The pin updates in this driver look like they are intended to be done atomically, update to do so. Signed-off-by: Charles Keepax Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adav80x.c | 17 +++++++++++------ 1 file changed, 11 insertions(+), 6 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index f78b27a7c461..ab790d5fe53d 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -541,6 +541,7 @@ static int adav80x_set_sysclk(struct snd_soc_codec *codec, unsigned int freq, int dir) { struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; if (dir == SND_SOC_CLOCK_IN) { switch (clk_id) { @@ -573,7 +574,7 @@ static int adav80x_set_sysclk(struct snd_soc_codec *codec, regmap_write(adav80x->regmap, ADAV80X_ICLK_CTRL2, iclk_ctrl2); - snd_soc_dapm_sync(&codec->dapm); + snd_soc_dapm_sync(dapm); } } else { unsigned int mask; @@ -600,17 +601,21 @@ static int adav80x_set_sysclk(struct snd_soc_codec *codec, adav80x->sysclk_pd[clk_id] = false; } + snd_soc_dapm_mutex_lock(dapm); + if (adav80x->sysclk_pd[0]) - snd_soc_dapm_disable_pin(&codec->dapm, "PLL1"); + snd_soc_dapm_disable_pin_unlocked(dapm, "PLL1"); else - snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL1"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL1"); if (adav80x->sysclk_pd[1] || adav80x->sysclk_pd[2]) - snd_soc_dapm_disable_pin(&codec->dapm, "PLL2"); + snd_soc_dapm_disable_pin_unlocked(dapm, "PLL2"); else - snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL2"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "PLL2"); - snd_soc_dapm_sync(&codec->dapm); + snd_soc_dapm_sync_unlocked(dapm); + + snd_soc_dapm_mutex_unlock(dapm); } return 0; -- cgit v1.2.3 From e951f267fd042ce6ca66449dd6d537b6126a10d7 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 18 Feb 2014 15:22:17 +0000 Subject: ASoC: wm5100: Update locking around use of DAPM pin API The pin updates in this driver look like they are intended to be done atomically, update to do so. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 4e3e31aaf509..492fe846ae68 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2100,6 +2100,7 @@ static void wm5100_micd_irq(struct wm5100_priv *wm5100) int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) { struct wm5100_priv *wm5100 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; if (jack) { wm5100->jack = jack; @@ -2117,9 +2118,14 @@ int wm5100_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) WM5100_ACCDET_RATE_MASK); /* We need the charge pump to power MICBIAS */ - snd_soc_dapm_force_enable_pin(&codec->dapm, "CP2"); - snd_soc_dapm_force_enable_pin(&codec->dapm, "SYSCLK"); - snd_soc_dapm_sync(&codec->dapm); + snd_soc_dapm_mutex_lock(dapm); + + snd_soc_dapm_force_enable_pin_unlocked(dapm, "CP2"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "SYSCLK"); + + snd_soc_dapm_sync_unlocked(dapm); + + snd_soc_dapm_mutex_unlock(dapm); /* We start off just enabling microphone detection - even a * plain headphone will trigger detection. -- cgit v1.2.3 From f1a3b8d9d4818b88cd7de369da3bb1804c2ad7da Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 18 Feb 2014 15:22:18 +0000 Subject: ASoC: wm8962: Update locking around use of DAPM pin API The pin updates in this driver look like they are intended to be done atomically, update to do so. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 13 +++++++++---- 1 file changed, 9 insertions(+), 4 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 97db3b45b411..9e6233633c44 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3089,6 +3089,7 @@ static irqreturn_t wm8962_irq(int irq, void *data) int wm8962_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) { struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; int irq_mask, enable; wm8962->jack = jack; @@ -3109,14 +3110,18 @@ int wm8962_mic_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack) snd_soc_jack_report(wm8962->jack, 0, SND_JACK_MICROPHONE | SND_JACK_BTN_0); + snd_soc_dapm_mutex_lock(dapm); + if (jack) { - snd_soc_dapm_force_enable_pin(&codec->dapm, "SYSCLK"); - snd_soc_dapm_force_enable_pin(&codec->dapm, "MICBIAS"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "SYSCLK"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "MICBIAS"); } else { - snd_soc_dapm_disable_pin(&codec->dapm, "SYSCLK"); - snd_soc_dapm_disable_pin(&codec->dapm, "MICBIAS"); + snd_soc_dapm_disable_pin_unlocked(dapm, "SYSCLK"); + snd_soc_dapm_disable_pin_unlocked(dapm, "MICBIAS"); } + snd_soc_dapm_mutex_unlock(dapm); + return 0; } EXPORT_SYMBOL_GPL(wm8962_mic_detect); -- cgit v1.2.3 From babce8211b194acdc41bc47975b50969a48b84ab Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 18 Feb 2014 15:22:19 +0000 Subject: ASoC: wm8994: Update locking around use of DAPM pin API The pin updates in this driver look like they are intended to be done atomically, update to do so. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 45 +++++++++++++++++++++++++++------------------ 1 file changed, 27 insertions(+), 18 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index b9be9cbc4603..e8daf55d37e3 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2549,43 +2549,52 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, int wm8994_vmid_mode(struct snd_soc_codec *codec, enum wm8994_vmid_mode mode) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; switch (mode) { case WM8994_VMID_NORMAL: + snd_soc_dapm_mutex_lock(dapm); + if (wm8994->hubs.lineout1_se) { - snd_soc_dapm_disable_pin(&codec->dapm, - "LINEOUT1N Driver"); - snd_soc_dapm_disable_pin(&codec->dapm, - "LINEOUT1P Driver"); + snd_soc_dapm_disable_pin_unlocked(dapm, + "LINEOUT1N Driver"); + snd_soc_dapm_disable_pin_unlocked(dapm, + "LINEOUT1P Driver"); } if (wm8994->hubs.lineout2_se) { - snd_soc_dapm_disable_pin(&codec->dapm, - "LINEOUT2N Driver"); - snd_soc_dapm_disable_pin(&codec->dapm, - "LINEOUT2P Driver"); + snd_soc_dapm_disable_pin_unlocked(dapm, + "LINEOUT2N Driver"); + snd_soc_dapm_disable_pin_unlocked(dapm, + "LINEOUT2P Driver"); } /* Do the sync with the old mode to allow it to clean up */ - snd_soc_dapm_sync(&codec->dapm); + snd_soc_dapm_sync_unlocked(dapm); wm8994->vmid_mode = mode; + + snd_soc_dapm_mutex_unlock(dapm); break; case WM8994_VMID_FORCE: + snd_soc_dapm_mutex_lock(dapm); + if (wm8994->hubs.lineout1_se) { - snd_soc_dapm_force_enable_pin(&codec->dapm, - "LINEOUT1N Driver"); - snd_soc_dapm_force_enable_pin(&codec->dapm, - "LINEOUT1P Driver"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, + "LINEOUT1N Driver"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, + "LINEOUT1P Driver"); } if (wm8994->hubs.lineout2_se) { - snd_soc_dapm_force_enable_pin(&codec->dapm, - "LINEOUT2N Driver"); - snd_soc_dapm_force_enable_pin(&codec->dapm, - "LINEOUT2P Driver"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, + "LINEOUT2N Driver"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, + "LINEOUT2P Driver"); } wm8994->vmid_mode = mode; - snd_soc_dapm_sync(&codec->dapm); + snd_soc_dapm_sync_unlocked(dapm); + + snd_soc_dapm_mutex_unlock(dapm); break; default: -- cgit v1.2.3 From 02afc6a23863367f59a3c792567cf1c9bb9d626c Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 18 Feb 2014 15:22:20 +0000 Subject: ASoC: wm8996: Update locking around use of DAPM pin API The pin updates in this driver look like they are intended to be done atomically, update to do so. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8996.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 1a7655b0aa22..d565d0ac7a11 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2251,6 +2251,7 @@ int wm8996_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, wm8996_polarity_fn polarity_cb) { struct wm8996_priv *wm8996 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; wm8996->jack = jack; wm8996->detecting = true; @@ -2267,8 +2268,12 @@ int wm8996_detect(struct snd_soc_codec *codec, struct snd_soc_jack *jack, WM8996_MICB2_DISCH, 0); /* LDO2 powers the microphones, SYSCLK clocks detection */ - snd_soc_dapm_force_enable_pin(&codec->dapm, "LDO2"); - snd_soc_dapm_force_enable_pin(&codec->dapm, "SYSCLK"); + snd_soc_dapm_mutex_lock(dapm); + + snd_soc_dapm_force_enable_pin_unlocked(dapm, "LDO2"); + snd_soc_dapm_force_enable_pin_unlocked(dapm, "SYSCLK"); + + snd_soc_dapm_mutex_unlock(dapm); /* We start off just enabling microphone detection - even a * plain headphone will trigger detection. -- cgit v1.2.3 From d1755bb75c68abaa2024dc9507d1c78521adadeb Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:16:08 +0100 Subject: ASoC: mc13783: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/mc13783.c | 14 +++++++------- 1 file changed, 7 insertions(+), 7 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 582c2bbd42cb..21c8baa2dd22 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -408,8 +408,8 @@ static const char * const adcl_enum_text[] = { "MC1L", "RXINL", }; -static const struct soc_enum adcl_enum = - SOC_ENUM_SINGLE(0, 0, ARRAY_SIZE(adcl_enum_text), adcl_enum_text); +static SOC_ENUM_SINGLE_DECL(adcl_enum, + 0, 0, adcl_enum_text); static const struct snd_kcontrol_new left_input_mux = SOC_DAPM_ENUM_VIRT("Route", adcl_enum); @@ -418,8 +418,8 @@ static const char * const adcr_enum_text[] = { "MC1R", "MC2", "RXINR", "TXIN", }; -static const struct soc_enum adcr_enum = - SOC_ENUM_SINGLE(0, 0, ARRAY_SIZE(adcr_enum_text), adcr_enum_text); +static SOC_ENUM_SINGLE_DECL(adcr_enum, + 0, 0, adcr_enum_text); static const struct snd_kcontrol_new right_input_mux = SOC_DAPM_ENUM_VIRT("Route", adcr_enum); @@ -580,9 +580,9 @@ static struct snd_soc_dapm_route mc13783_routes[] = { static const char * const mc13783_3d_mixer[] = {"Stereo", "Phase Mix", "Mono", "Mono Mix"}; -static const struct soc_enum mc13783_enum_3d_mixer = - SOC_ENUM_SINGLE(MC13783_AUDIO_RX1, 16, ARRAY_SIZE(mc13783_3d_mixer), - mc13783_3d_mixer); +static SOC_ENUM_SINGLE_DECL(mc13783_enum_3d_mixer, + MC13783_AUDIO_RX1, 16, + mc13783_3d_mixer); static struct snd_kcontrol_new mc13783_control_list[] = { SOC_SINGLE("Loudspeaker enable", MC13783_AUDIO_RX0, 5, 1, 0), -- cgit v1.2.3 From c8ed6504218c5a1951159033d1b1e7927665f109 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:16:31 +0100 Subject: ASoC: sgtl5000: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 0fcbe90f3ef2..ab4754a7a88c 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -187,8 +187,9 @@ static const char *adc_mux_text[] = { "MIC_IN", "LINE_IN" }; -static const struct soc_enum adc_enum = -SOC_ENUM_SINGLE(SGTL5000_CHIP_ANA_CTRL, 2, 2, adc_mux_text); +static SOC_ENUM_SINGLE_DECL(adc_enum, + SGTL5000_CHIP_ANA_CTRL, 2, + adc_mux_text); static const struct snd_kcontrol_new adc_mux = SOC_DAPM_ENUM("Capture Mux", adc_enum); @@ -198,8 +199,9 @@ static const char *dac_mux_text[] = { "DAC", "LINE_IN" }; -static const struct soc_enum dac_enum = -SOC_ENUM_SINGLE(SGTL5000_CHIP_ANA_CTRL, 6, 2, dac_mux_text); +static SOC_ENUM_SINGLE_DECL(dac_enum, + SGTL5000_CHIP_ANA_CTRL, 6, + dac_mux_text); static const struct snd_kcontrol_new dac_mux = SOC_DAPM_ENUM("Headphone Mux", dac_enum); -- cgit v1.2.3 From f3c1196e16a84e81978bdf60690cb82fe33e880f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:17:24 +0100 Subject: ASoC: sn95031: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/sn95031.c | 44 ++++++++++++++++++++++---------------------- 1 file changed, 22 insertions(+), 22 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 13045f2af4d3..bca7d02b362a 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -312,14 +312,14 @@ static int sn95031_dmic56_event(struct snd_soc_dapm_widget *w, /* mux controls */ static const char *sn95031_mic_texts[] = { "AMIC", "LineIn" }; -static const struct soc_enum sn95031_micl_enum = - SOC_ENUM_SINGLE(SN95031_ADCCONFIG, 1, 2, sn95031_mic_texts); +static SOC_ENUM_SINGLE_DECL(sn95031_micl_enum, + SN95031_ADCCONFIG, 1, sn95031_mic_texts); static const struct snd_kcontrol_new sn95031_micl_mux_control = SOC_DAPM_ENUM("Route", sn95031_micl_enum); -static const struct soc_enum sn95031_micr_enum = - SOC_ENUM_SINGLE(SN95031_ADCCONFIG, 3, 2, sn95031_mic_texts); +static SOC_ENUM_SINGLE_DECL(sn95031_micr_enum, + SN95031_ADCCONFIG, 3, sn95031_mic_texts); static const struct snd_kcontrol_new sn95031_micr_mux_control = SOC_DAPM_ENUM("Route", sn95031_micr_enum); @@ -328,26 +328,26 @@ static const char *sn95031_input_texts[] = { "DMIC1", "DMIC2", "DMIC3", "DMIC4", "DMIC5", "DMIC6", "ADC Left", "ADC Right" }; -static const struct soc_enum sn95031_input1_enum = - SOC_ENUM_SINGLE(SN95031_AUDIOMUX12, 0, 8, sn95031_input_texts); +static SOC_ENUM_SINGLE_DECL(sn95031_input1_enum, + SN95031_AUDIOMUX12, 0, sn95031_input_texts); static const struct snd_kcontrol_new sn95031_input1_mux_control = SOC_DAPM_ENUM("Route", sn95031_input1_enum); -static const struct soc_enum sn95031_input2_enum = - SOC_ENUM_SINGLE(SN95031_AUDIOMUX12, 4, 8, sn95031_input_texts); +static SOC_ENUM_SINGLE_DECL(sn95031_input2_enum, + SN95031_AUDIOMUX12, 4, sn95031_input_texts); static const struct snd_kcontrol_new sn95031_input2_mux_control = SOC_DAPM_ENUM("Route", sn95031_input2_enum); -static const struct soc_enum sn95031_input3_enum = - SOC_ENUM_SINGLE(SN95031_AUDIOMUX34, 0, 8, sn95031_input_texts); +static SOC_ENUM_SINGLE_DECL(sn95031_input3_enum, + SN95031_AUDIOMUX34, 0, sn95031_input_texts); static const struct snd_kcontrol_new sn95031_input3_mux_control = SOC_DAPM_ENUM("Route", sn95031_input3_enum); -static const struct soc_enum sn95031_input4_enum = - SOC_ENUM_SINGLE(SN95031_AUDIOMUX34, 4, 8, sn95031_input_texts); +static SOC_ENUM_SINGLE_DECL(sn95031_input4_enum, + SN95031_AUDIOMUX34, 4, sn95031_input_texts); static const struct snd_kcontrol_new sn95031_input4_mux_control = SOC_DAPM_ENUM("Route", sn95031_input4_enum); @@ -359,19 +359,19 @@ static const char *sn95031_micmode_text[] = {"Single Ended", "Differential"}; /* 0dB to 30dB in 10dB steps */ static const DECLARE_TLV_DB_SCALE(mic_tlv, 0, 10, 0); -static const struct soc_enum sn95031_micmode1_enum = - SOC_ENUM_SINGLE(SN95031_MICAMP1, 1, 2, sn95031_micmode_text); -static const struct soc_enum sn95031_micmode2_enum = - SOC_ENUM_SINGLE(SN95031_MICAMP2, 1, 2, sn95031_micmode_text); +static SOC_ENUM_SINGLE_DECL(sn95031_micmode1_enum, + SN95031_MICAMP1, 1, sn95031_micmode_text); +static SOC_ENUM_SINGLE_DECL(sn95031_micmode2_enum, + SN95031_MICAMP2, 1, sn95031_micmode_text); static const char *sn95031_dmic_cfg_text[] = {"GPO", "DMIC"}; -static const struct soc_enum sn95031_dmic12_cfg_enum = - SOC_ENUM_SINGLE(SN95031_DMICMUX, 0, 2, sn95031_dmic_cfg_text); -static const struct soc_enum sn95031_dmic34_cfg_enum = - SOC_ENUM_SINGLE(SN95031_DMICMUX, 1, 2, sn95031_dmic_cfg_text); -static const struct soc_enum sn95031_dmic56_cfg_enum = - SOC_ENUM_SINGLE(SN95031_DMICMUX, 2, 2, sn95031_dmic_cfg_text); +static SOC_ENUM_SINGLE_DECL(sn95031_dmic12_cfg_enum, + SN95031_DMICMUX, 0, sn95031_dmic_cfg_text); +static SOC_ENUM_SINGLE_DECL(sn95031_dmic34_cfg_enum, + SN95031_DMICMUX, 1, sn95031_dmic_cfg_text); +static SOC_ENUM_SINGLE_DECL(sn95031_dmic56_cfg_enum, + SN95031_DMICMUX, 2, sn95031_dmic_cfg_text); static const struct snd_kcontrol_new sn95031_snd_controls[] = { SOC_ENUM("Mic1Mode Capture Route", sn95031_micmode1_enum), -- cgit v1.2.3 From 3cd7ca58827234c40bb683de2d1d8512631be8e4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:26:27 +0100 Subject: ASoC: stac9766: Use SOC_ENUM_{SINGLE|DOUBLE}_DECL() Just replace with the helper macros. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/stac9766.c | 38 +++++++++++++++++++------------------- 1 file changed, 19 insertions(+), 19 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c index a5455c1aea42..53b810d23fea 100644 --- a/sound/soc/codecs/stac9766.c +++ b/sound/soc/codecs/stac9766.c @@ -62,25 +62,25 @@ static const char *stac9766_boost1[] = {"0dB", "10dB"}; static const char *stac9766_boost2[] = {"0dB", "20dB"}; static const char *stac9766_stereo_mic[] = {"Off", "On"}; -static const struct soc_enum stac9766_record_enum = - SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 8, stac9766_record_mux); -static const struct soc_enum stac9766_mono_enum = - SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 9, 2, stac9766_mono_mux); -static const struct soc_enum stac9766_mic_enum = - SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, stac9766_mic_mux); -static const struct soc_enum stac9766_SPDIF_enum = - SOC_ENUM_SINGLE(AC97_STAC_DA_CONTROL, 1, 2, stac9766_SPDIF_mux); -static const struct soc_enum stac9766_popbypass_enum = - SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, stac9766_popbypass_mux); -static const struct soc_enum stac9766_record_all_enum = - SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2, - stac9766_record_all_mux); -static const struct soc_enum stac9766_boost1_enum = - SOC_ENUM_SINGLE(AC97_MIC, 6, 2, stac9766_boost1); /* 0/10dB */ -static const struct soc_enum stac9766_boost2_enum = - SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 2, 2, stac9766_boost2); /* 0/20dB */ -static const struct soc_enum stac9766_stereo_mic_enum = - SOC_ENUM_SINGLE(AC97_STAC_STEREO_MIC, 2, 1, stac9766_stereo_mic); +static SOC_ENUM_DOUBLE_DECL(stac9766_record_enum, + AC97_REC_SEL, 8, 0, stac9766_record_mux); +static SOC_ENUM_SINGLE_DECL(stac9766_mono_enum, + AC97_GENERAL_PURPOSE, 9, stac9766_mono_mux); +static SOC_ENUM_SINGLE_DECL(stac9766_mic_enum, + AC97_GENERAL_PURPOSE, 8, stac9766_mic_mux); +static SOC_ENUM_SINGLE_DECL(stac9766_SPDIF_enum, + AC97_STAC_DA_CONTROL, 1, stac9766_SPDIF_mux); +static SOC_ENUM_SINGLE_DECL(stac9766_popbypass_enum, + AC97_GENERAL_PURPOSE, 15, stac9766_popbypass_mux); +static SOC_ENUM_SINGLE_DECL(stac9766_record_all_enum, + AC97_STAC_ANALOG_SPECIAL, 12, + stac9766_record_all_mux); +static SOC_ENUM_SINGLE_DECL(stac9766_boost1_enum, + AC97_MIC, 6, stac9766_boost1); /* 0/10dB */ +static SOC_ENUM_SINGLE_DECL(stac9766_boost2_enum, + AC97_STAC_ANALOG_SPECIAL, 2, stac9766_boost2); /* 0/20dB */ +static SOC_ENUM_SINGLE_DECL(stac9766_stereo_mic_enum, + AC97_STAC_STEREO_MIC, 2, stac9766_stereo_mic); static const DECLARE_TLV_DB_LINEAR(master_tlv, -4600, 0); static const DECLARE_TLV_DB_LINEAR(record_tlv, 0, 2250); -- cgit v1.2.3 From 6900ab55dda48825b1cddc87e5e1908d51e96b95 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:27:40 +0100 Subject: ASoC: tlv320aic26: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic26.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 94a658fa6d97..ff5f23d482b7 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -238,8 +238,9 @@ static struct snd_soc_dai_driver aic26_dai = { * ALSA controls */ static const char *aic26_capture_src_text[] = {"Mic", "Aux"}; -static const struct soc_enum aic26_capture_src_enum = - SOC_ENUM_SINGLE(AIC26_REG_AUDIO_CTRL1, 12, 2, aic26_capture_src_text); +static SOC_ENUM_SINGLE_DECL(aic26_capture_src_enum, + AIC26_REG_AUDIO_CTRL1, 12, + aic26_capture_src_text); static const struct snd_kcontrol_new aic26_snd_controls[] = { /* Output */ -- cgit v1.2.3 From 9f04fba79781fb3ba39eac631f4bd6762f9717db Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:28:25 +0100 Subject: ASoC: twl4030: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/twl4030.c | 84 ++++++++++++++++++++-------------------------- 1 file changed, 36 insertions(+), 48 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 00665ada23e2..e084df7e27a5 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -415,10 +415,9 @@ static const struct snd_kcontrol_new twl4030_dapm_carkitr_controls[] = { static const char *twl4030_handsfreel_texts[] = {"Voice", "AudioL1", "AudioL2", "AudioR2"}; -static const struct soc_enum twl4030_handsfreel_enum = - SOC_ENUM_SINGLE(TWL4030_REG_HFL_CTL, 0, - ARRAY_SIZE(twl4030_handsfreel_texts), - twl4030_handsfreel_texts); +static SOC_ENUM_SINGLE_DECL(twl4030_handsfreel_enum, + TWL4030_REG_HFL_CTL, 0, + twl4030_handsfreel_texts); static const struct snd_kcontrol_new twl4030_dapm_handsfreel_control = SOC_DAPM_ENUM("Route", twl4030_handsfreel_enum); @@ -431,10 +430,9 @@ static const struct snd_kcontrol_new twl4030_dapm_handsfreelmute_control = static const char *twl4030_handsfreer_texts[] = {"Voice", "AudioR1", "AudioR2", "AudioL2"}; -static const struct soc_enum twl4030_handsfreer_enum = - SOC_ENUM_SINGLE(TWL4030_REG_HFR_CTL, 0, - ARRAY_SIZE(twl4030_handsfreer_texts), - twl4030_handsfreer_texts); +static SOC_ENUM_SINGLE_DECL(twl4030_handsfreer_enum, + TWL4030_REG_HFR_CTL, 0, + twl4030_handsfreer_texts); static const struct snd_kcontrol_new twl4030_dapm_handsfreer_control = SOC_DAPM_ENUM("Route", twl4030_handsfreer_enum); @@ -448,10 +446,9 @@ static const struct snd_kcontrol_new twl4030_dapm_handsfreermute_control = static const char *twl4030_vibra_texts[] = {"AudioL1", "AudioR1", "AudioL2", "AudioR2"}; -static const struct soc_enum twl4030_vibra_enum = - SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 2, - ARRAY_SIZE(twl4030_vibra_texts), - twl4030_vibra_texts); +static SOC_ENUM_SINGLE_DECL(twl4030_vibra_enum, + TWL4030_REG_VIBRA_CTL, 2, + twl4030_vibra_texts); static const struct snd_kcontrol_new twl4030_dapm_vibra_control = SOC_DAPM_ENUM("Route", twl4030_vibra_enum); @@ -460,10 +457,9 @@ SOC_DAPM_ENUM("Route", twl4030_vibra_enum); static const char *twl4030_vibrapath_texts[] = {"Local vibrator", "Audio"}; -static const struct soc_enum twl4030_vibrapath_enum = - SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 4, - ARRAY_SIZE(twl4030_vibrapath_texts), - twl4030_vibrapath_texts); +static SOC_ENUM_SINGLE_DECL(twl4030_vibrapath_enum, + TWL4030_REG_VIBRA_CTL, 4, + twl4030_vibrapath_texts); static const struct snd_kcontrol_new twl4030_dapm_vibrapath_control = SOC_DAPM_ENUM("Route", twl4030_vibrapath_enum); @@ -490,10 +486,9 @@ static const struct snd_kcontrol_new twl4030_dapm_analogrmic_controls[] = { static const char *twl4030_micpathtx1_texts[] = {"Analog", "Digimic0"}; -static const struct soc_enum twl4030_micpathtx1_enum = - SOC_ENUM_SINGLE(TWL4030_REG_ADCMICSEL, 0, - ARRAY_SIZE(twl4030_micpathtx1_texts), - twl4030_micpathtx1_texts); +static SOC_ENUM_SINGLE_DECL(twl4030_micpathtx1_enum, + TWL4030_REG_ADCMICSEL, 0, + twl4030_micpathtx1_texts); static const struct snd_kcontrol_new twl4030_dapm_micpathtx1_control = SOC_DAPM_ENUM("Route", twl4030_micpathtx1_enum); @@ -502,10 +497,9 @@ SOC_DAPM_ENUM("Route", twl4030_micpathtx1_enum); static const char *twl4030_micpathtx2_texts[] = {"Analog", "Digimic1"}; -static const struct soc_enum twl4030_micpathtx2_enum = - SOC_ENUM_SINGLE(TWL4030_REG_ADCMICSEL, 2, - ARRAY_SIZE(twl4030_micpathtx2_texts), - twl4030_micpathtx2_texts); +static SOC_ENUM_SINGLE_DECL(twl4030_micpathtx2_enum, + TWL4030_REG_ADCMICSEL, 2, + twl4030_micpathtx2_texts); static const struct snd_kcontrol_new twl4030_dapm_micpathtx2_control = SOC_DAPM_ENUM("Route", twl4030_micpathtx2_enum); @@ -955,10 +949,9 @@ static const char *twl4030_op_modes_texts[] = { "Option 2 (voice/audio)", "Option 1 (audio)" }; -static const struct soc_enum twl4030_op_modes_enum = - SOC_ENUM_SINGLE(TWL4030_REG_CODEC_MODE, 0, - ARRAY_SIZE(twl4030_op_modes_texts), - twl4030_op_modes_texts); +static SOC_ENUM_SINGLE_DECL(twl4030_op_modes_enum, + TWL4030_REG_CODEC_MODE, 0, + twl4030_op_modes_texts); static int snd_soc_put_twl4030_opmode_enum_double(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -1044,10 +1037,9 @@ static const char *twl4030_avadc_clk_priority_texts[] = { "Voice high priority", "HiFi high priority" }; -static const struct soc_enum twl4030_avadc_clk_priority_enum = - SOC_ENUM_SINGLE(TWL4030_REG_AVADC_CTL, 2, - ARRAY_SIZE(twl4030_avadc_clk_priority_texts), - twl4030_avadc_clk_priority_texts); +static SOC_ENUM_SINGLE_DECL(twl4030_avadc_clk_priority_enum, + TWL4030_REG_AVADC_CTL, 2, + twl4030_avadc_clk_priority_texts); static const char *twl4030_rampdelay_texts[] = { "27/20/14 ms", "55/40/27 ms", "109/81/55 ms", "218/161/109 ms", @@ -1055,40 +1047,36 @@ static const char *twl4030_rampdelay_texts[] = { "3495/2581/1748 ms" }; -static const struct soc_enum twl4030_rampdelay_enum = - SOC_ENUM_SINGLE(TWL4030_REG_HS_POPN_SET, 2, - ARRAY_SIZE(twl4030_rampdelay_texts), - twl4030_rampdelay_texts); +static SOC_ENUM_SINGLE_DECL(twl4030_rampdelay_enum, + TWL4030_REG_HS_POPN_SET, 2, + twl4030_rampdelay_texts); /* Vibra H-bridge direction mode */ static const char *twl4030_vibradirmode_texts[] = { "Vibra H-bridge direction", "Audio data MSB", }; -static const struct soc_enum twl4030_vibradirmode_enum = - SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 5, - ARRAY_SIZE(twl4030_vibradirmode_texts), - twl4030_vibradirmode_texts); +static SOC_ENUM_SINGLE_DECL(twl4030_vibradirmode_enum, + TWL4030_REG_VIBRA_CTL, 5, + twl4030_vibradirmode_texts); /* Vibra H-bridge direction */ static const char *twl4030_vibradir_texts[] = { "Positive polarity", "Negative polarity", }; -static const struct soc_enum twl4030_vibradir_enum = - SOC_ENUM_SINGLE(TWL4030_REG_VIBRA_CTL, 1, - ARRAY_SIZE(twl4030_vibradir_texts), - twl4030_vibradir_texts); +static SOC_ENUM_SINGLE_DECL(twl4030_vibradir_enum, + TWL4030_REG_VIBRA_CTL, 1, + twl4030_vibradir_texts); /* Digimic Left and right swapping */ static const char *twl4030_digimicswap_texts[] = { "Not swapped", "Swapped", }; -static const struct soc_enum twl4030_digimicswap_enum = - SOC_ENUM_SINGLE(TWL4030_REG_MISC_SET_1, 0, - ARRAY_SIZE(twl4030_digimicswap_texts), - twl4030_digimicswap_texts); +static SOC_ENUM_SINGLE_DECL(twl4030_digimicswap_enum, + TWL4030_REG_MISC_SET_1, 0, + twl4030_digimicswap_texts); static const struct snd_kcontrol_new twl4030_snd_controls[] = { /* Codec operation mode control */ -- cgit v1.2.3 From a1d0d786af587c50ea948439e610c90525af36d7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:30:20 +0100 Subject: ASoC: twl6040: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. Also, use ARRAY_SIZE() in some ASOC_ENUM_SINGLE() lines. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/twl6040.c | 11 ++++++----- 1 file changed, 6 insertions(+), 5 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index cb642c927dc8..bd3a20647fdf 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -392,8 +392,10 @@ static const char *twl6040_amicr_texts[] = {"Headset Mic", "Sub Mic", "Aux/FM Right", "Off"}; static const struct soc_enum twl6040_enum[] = { - SOC_ENUM_SINGLE(TWL6040_REG_MICLCTL, 3, 4, twl6040_amicl_texts), - SOC_ENUM_SINGLE(TWL6040_REG_MICRCTL, 3, 4, twl6040_amicr_texts), + SOC_ENUM_SINGLE(TWL6040_REG_MICLCTL, 3, + ARRAY_SIZE(twl6040_amicl_texts), twl6040_amicl_texts), + SOC_ENUM_SINGLE(TWL6040_REG_MICRCTL, 3, + ARRAY_SIZE(twl6040_amicr_texts), twl6040_amicr_texts), }; static const char *twl6040_hs_texts[] = { @@ -476,9 +478,8 @@ static const char *twl6040_power_mode_texts[] = { "Low-Power", "High-Performance", }; -static const struct soc_enum twl6040_power_mode_enum = - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(twl6040_power_mode_texts), - twl6040_power_mode_texts); +static SOC_ENUM_SINGLE_EXT_DECL(twl6040_power_mode_enum, + twl6040_power_mode_texts); static int twl6040_headset_power_get_enum(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -- cgit v1.2.3 From 1dbb348d904baed36789078e1202344f5a6ecc84 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:32:16 +0100 Subject: ASoC: uda1380: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. Also, use ARRAY_SIZE() in some ASOC_ENUM_SINGLE() lines. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/uda1380.c | 38 ++++++++++++++++++++------------------ 1 file changed, 20 insertions(+), 18 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 726df6d43c2b..72ee8e1838e5 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -237,25 +237,27 @@ static const char *uda1380_os_setting[] = { }; static const struct soc_enum uda1380_deemp_enum[] = { - SOC_ENUM_SINGLE(UDA1380_DEEMP, 8, 5, uda1380_deemp), - SOC_ENUM_SINGLE(UDA1380_DEEMP, 0, 5, uda1380_deemp), + SOC_ENUM_SINGLE(UDA1380_DEEMP, 8, ARRAY_SIZE(uda1380_deemp), + uda1380_deemp), + SOC_ENUM_SINGLE(UDA1380_DEEMP, 0, ARRAY_SIZE(uda1380_deemp), + uda1380_deemp), }; -static const struct soc_enum uda1380_input_sel_enum = - SOC_ENUM_SINGLE(UDA1380_ADC, 2, 4, uda1380_input_sel); /* SEL_MIC, SEL_LNA */ -static const struct soc_enum uda1380_output_sel_enum = - SOC_ENUM_SINGLE(UDA1380_PM, 7, 2, uda1380_output_sel); /* R02_EN_AVC */ -static const struct soc_enum uda1380_spf_enum = - SOC_ENUM_SINGLE(UDA1380_MODE, 14, 4, uda1380_spf_mode); /* M */ -static const struct soc_enum uda1380_capture_sel_enum = - SOC_ENUM_SINGLE(UDA1380_IFACE, 6, 2, uda1380_capture_sel); /* SEL_SOURCE */ -static const struct soc_enum uda1380_sel_ns_enum = - SOC_ENUM_SINGLE(UDA1380_MIXER, 14, 2, uda1380_sel_ns); /* SEL_NS */ -static const struct soc_enum uda1380_mix_enum = - SOC_ENUM_SINGLE(UDA1380_MIXER, 12, 4, uda1380_mix_control); /* MIX, MIX_POS */ -static const struct soc_enum uda1380_sdet_enum = - SOC_ENUM_SINGLE(UDA1380_MIXER, 4, 4, uda1380_sdet_setting); /* SD_VALUE */ -static const struct soc_enum uda1380_os_enum = - SOC_ENUM_SINGLE(UDA1380_MIXER, 0, 3, uda1380_os_setting); /* OS */ +static SOC_ENUM_SINGLE_DECL(uda1380_input_sel_enum, + UDA1380_ADC, 2, uda1380_input_sel); /* SEL_MIC, SEL_LNA */ +static SOC_ENUM_SINGLE_DECL(uda1380_output_sel_enum, + UDA1380_PM, 7, uda1380_output_sel); /* R02_EN_AVC */ +static SOC_ENUM_SINGLE_DECL(uda1380_spf_enum, + UDA1380_MODE, 14, uda1380_spf_mode); /* M */ +static SOC_ENUM_SINGLE_DECL(uda1380_capture_sel_enum, + UDA1380_IFACE, 6, uda1380_capture_sel); /* SEL_SOURCE */ +static SOC_ENUM_SINGLE_DECL(uda1380_sel_ns_enum, + UDA1380_MIXER, 14, uda1380_sel_ns); /* SEL_NS */ +static SOC_ENUM_SINGLE_DECL(uda1380_mix_enum, + UDA1380_MIXER, 12, uda1380_mix_control); /* MIX, MIX_POS */ +static SOC_ENUM_SINGLE_DECL(uda1380_sdet_enum, + UDA1380_MIXER, 4, uda1380_sdet_setting); /* SD_VALUE */ +static SOC_ENUM_SINGLE_DECL(uda1380_os_enum, + UDA1380_MIXER, 0, uda1380_os_setting); /* OS */ /* * from -48 dB in 1.5 dB steps (mute instead of -49.5 dB) -- cgit v1.2.3 From 4ec20a9700f6f4fa140d8586f4f8ff5f76ea4ba7 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:33:06 +0100 Subject: ASoC: wl1273: Use SOC_ENUM_SINGLE_EXT_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wl1273.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index b7ab2ef567c8..7485285d08d2 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -209,8 +209,7 @@ static int snd_wl1273_set_audio_route(struct snd_kcontrol *kcontrol, return 1; } -static const struct soc_enum wl1273_enum = - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(wl1273_audio_route), wl1273_audio_route); +static SOC_ENUM_SINGLE_EXT_DECL(wl1273_enum, wl1273_audio_route); static int snd_wl1273_fm_audio_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -247,9 +246,7 @@ static int snd_wl1273_fm_audio_put(struct snd_kcontrol *kcontrol, static const char * const wl1273_audio_strings[] = { "Digital", "Analog" }; -static const struct soc_enum wl1273_audio_enum = - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(wl1273_audio_strings), - wl1273_audio_strings); +static SOC_ENUM_SINGLE_EXT_DECL(wl1273_audio_enum, wl1273_audio_strings); static int snd_wl1273_fm_volume_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -- cgit v1.2.3 From 9614be73d810a02f23f3cc19d8eb97b81213545e Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:33:59 +0100 Subject: ASoC: wm2200: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Charles Keepax Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm2200.c | 18 ++++++++---------- 1 file changed, 8 insertions(+), 10 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 57ba315d0c84..1e0a083d8345 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1113,11 +1113,10 @@ static const char *wm2200_rxanc_input_sel_texts[] = { "None", "IN1", "IN2", "IN3", }; -static const struct soc_enum wm2200_rxanc_input_sel = - SOC_ENUM_SINGLE(WM2200_RXANC_SRC, - WM2200_IN_RXANC_SEL_SHIFT, - ARRAY_SIZE(wm2200_rxanc_input_sel_texts), - wm2200_rxanc_input_sel_texts); +static SOC_ENUM_SINGLE_DECL(wm2200_rxanc_input_sel, + WM2200_RXANC_SRC, + WM2200_IN_RXANC_SEL_SHIFT, + wm2200_rxanc_input_sel_texts); static const struct snd_kcontrol_new wm2200_snd_controls[] = { SOC_SINGLE("IN1 High Performance Switch", WM2200_IN1L_CONTROL, @@ -1288,11 +1287,10 @@ static const char *wm2200_aec_loopback_texts[] = { "OUT1L", "OUT1R", "OUT2L", "OUT2R", }; -static const struct soc_enum wm2200_aec_loopback = - SOC_ENUM_SINGLE(WM2200_DAC_AEC_CONTROL_1, - WM2200_AEC_LOOPBACK_SRC_SHIFT, - ARRAY_SIZE(wm2200_aec_loopback_texts), - wm2200_aec_loopback_texts); +static SOC_ENUM_SINGLE_DECL(wm2200_aec_loopback, + WM2200_DAC_AEC_CONTROL_1, + WM2200_AEC_LOOPBACK_SRC_SHIFT, + wm2200_aec_loopback_texts); static const struct snd_kcontrol_new wm2200_aec_loopback_mux = SOC_DAPM_ENUM("AEC Loopback", wm2200_aec_loopback); -- cgit v1.2.3 From fed08d94bf3f930ebe9ad2f3ad7744b8e2eab6bc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:35:49 +0100 Subject: ASoC: wm8523: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Charles Keepax Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8523.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 74d106dc7667..5dfd571b1a03 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -75,8 +75,8 @@ static const char *wm8523_zd_count_text[] = { "2048", }; -static const struct soc_enum wm8523_zc_count = - SOC_ENUM_SINGLE(WM8523_ZERO_DETECT, 0, 2, wm8523_zd_count_text); +static SOC_ENUM_SINGLE_DECL(wm8523_zc_count, WM8523_ZERO_DETECT, 0, + wm8523_zd_count_text); static const struct snd_kcontrol_new wm8523_controls[] = { SOC_DOUBLE_R_TLV("Playback Volume", WM8523_DAC_GAINL, WM8523_DAC_GAINR, -- cgit v1.2.3 From 9e74b14ad5140837cfa0dc639dc13acaa2b05b0f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:36:16 +0100 Subject: ASoC: wm8731: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Charles Keepax Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8731.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 029720366ff8..d9655f981df1 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -83,8 +83,8 @@ static bool wm8731_writeable(struct device *dev, unsigned int reg) static const char *wm8731_input_select[] = {"Line In", "Mic"}; -static const struct soc_enum wm8731_insel_enum = - SOC_ENUM_SINGLE(WM8731_APANA, 2, 2, wm8731_input_select); +static SOC_ENUM_SINGLE_DECL(wm8731_insel_enum, + WM8731_APANA, 2, wm8731_input_select); static int wm8731_deemph[] = { 0, 32000, 44100, 48000 }; -- cgit v1.2.3 From ca275811fd2eef7e0121fefbc46cc5b47d680b10 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:37:14 +0100 Subject: ASoC: wm8737: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Charles Keepax Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8737.c | 44 ++++++++++++++++++++++---------------------- 1 file changed, 22 insertions(+), 22 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index 22de2420bec8..ecc4e8725d5b 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -99,29 +99,29 @@ static const char *micbias_enum_text[] = { "100%", }; -static const struct soc_enum micbias_enum = - SOC_ENUM_SINGLE(WM8737_MIC_PREAMP_CONTROL, 0, 4, micbias_enum_text); +static SOC_ENUM_SINGLE_DECL(micbias_enum, + WM8737_MIC_PREAMP_CONTROL, 0, micbias_enum_text); static const char *low_cutoff_text[] = { "Low", "High" }; -static const struct soc_enum low_3d = - SOC_ENUM_SINGLE(WM8737_3D_ENHANCE, 6, 2, low_cutoff_text); +static SOC_ENUM_SINGLE_DECL(low_3d, + WM8737_3D_ENHANCE, 6, low_cutoff_text); static const char *high_cutoff_text[] = { "High", "Low" }; -static const struct soc_enum high_3d = - SOC_ENUM_SINGLE(WM8737_3D_ENHANCE, 5, 2, high_cutoff_text); +static SOC_ENUM_SINGLE_DECL(high_3d, + WM8737_3D_ENHANCE, 5, high_cutoff_text); static const char *alc_fn_text[] = { "Disabled", "Right", "Left", "Stereo" }; -static const struct soc_enum alc_fn = - SOC_ENUM_SINGLE(WM8737_ALC1, 7, 4, alc_fn_text); +static SOC_ENUM_SINGLE_DECL(alc_fn, + WM8737_ALC1, 7, alc_fn_text); static const char *alc_hold_text[] = { "0", "2.67ms", "5.33ms", "10.66ms", "21.32ms", "42.64ms", "85.28ms", @@ -129,24 +129,24 @@ static const char *alc_hold_text[] = { "10.916s", "21.832s", "43.691s" }; -static const struct soc_enum alc_hold = - SOC_ENUM_SINGLE(WM8737_ALC2, 0, 16, alc_hold_text); +static SOC_ENUM_SINGLE_DECL(alc_hold, + WM8737_ALC2, 0, alc_hold_text); static const char *alc_atk_text[] = { "8.4ms", "16.8ms", "33.6ms", "67.2ms", "134.4ms", "268.8ms", "537.6ms", "1.075s", "2.15s", "4.3s", "8.6s" }; -static const struct soc_enum alc_atk = - SOC_ENUM_SINGLE(WM8737_ALC3, 0, 11, alc_atk_text); +static SOC_ENUM_SINGLE_DECL(alc_atk, + WM8737_ALC3, 0, alc_atk_text); static const char *alc_dcy_text[] = { "33.6ms", "67.2ms", "134.4ms", "268.8ms", "537.6ms", "1.075s", "2.15s", "4.3s", "8.6s", "17.2s", "34.41s" }; -static const struct soc_enum alc_dcy = - SOC_ENUM_SINGLE(WM8737_ALC3, 4, 11, alc_dcy_text); +static SOC_ENUM_SINGLE_DECL(alc_dcy, + WM8737_ALC3, 4, alc_dcy_text); static const struct snd_kcontrol_new wm8737_snd_controls[] = { SOC_DOUBLE_R_TLV("Mic Boost Volume", WM8737_AUDIO_PATH_L, WM8737_AUDIO_PATH_R, @@ -191,8 +191,8 @@ static const char *linsel_text[] = { "LINPUT1", "LINPUT2", "LINPUT3", "LINPUT1 DC", }; -static const struct soc_enum linsel_enum = - SOC_ENUM_SINGLE(WM8737_AUDIO_PATH_L, 7, 4, linsel_text); +static SOC_ENUM_SINGLE_DECL(linsel_enum, + WM8737_AUDIO_PATH_L, 7, linsel_text); static const struct snd_kcontrol_new linsel_mux = SOC_DAPM_ENUM("LINSEL", linsel_enum); @@ -202,8 +202,8 @@ static const char *rinsel_text[] = { "RINPUT1", "RINPUT2", "RINPUT3", "RINPUT1 DC", }; -static const struct soc_enum rinsel_enum = - SOC_ENUM_SINGLE(WM8737_AUDIO_PATH_R, 7, 4, rinsel_text); +static SOC_ENUM_SINGLE_DECL(rinsel_enum, + WM8737_AUDIO_PATH_R, 7, rinsel_text); static const struct snd_kcontrol_new rinsel_mux = SOC_DAPM_ENUM("RINSEL", rinsel_enum); @@ -212,15 +212,15 @@ static const char *bypass_text[] = { "Direct", "Preamp" }; -static const struct soc_enum lbypass_enum = - SOC_ENUM_SINGLE(WM8737_MIC_PREAMP_CONTROL, 2, 2, bypass_text); +static SOC_ENUM_SINGLE_DECL(lbypass_enum, + WM8737_MIC_PREAMP_CONTROL, 2, bypass_text); static const struct snd_kcontrol_new lbypass_mux = SOC_DAPM_ENUM("Left Bypass", lbypass_enum); -static const struct soc_enum rbypass_enum = - SOC_ENUM_SINGLE(WM8737_MIC_PREAMP_CONTROL, 3, 2, bypass_text); +static SOC_ENUM_SINGLE_DECL(rbypass_enum, + WM8737_MIC_PREAMP_CONTROL, 3, bypass_text); static const struct snd_kcontrol_new rbypass_mux = SOC_DAPM_ENUM("Left Bypass", rbypass_enum); -- cgit v1.2.3 From a21bc5c5bdeb19314976b79db6dc3993c9b227c8 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:40:16 +0100 Subject: ASoC: wm8903: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Charles Keepax Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 108 +++++++++++++++++++++++----------------------- 1 file changed, 54 insertions(+), 54 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index eebcb1da3b7b..b82b70a3b3d3 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -489,28 +489,28 @@ static const char *hpf_mode_text[] = { "Hi-fi", "Voice 1", "Voice 2", "Voice 3" }; -static const struct soc_enum hpf_mode = - SOC_ENUM_SINGLE(WM8903_ADC_DIGITAL_0, 5, 4, hpf_mode_text); +static SOC_ENUM_SINGLE_DECL(hpf_mode, + WM8903_ADC_DIGITAL_0, 5, hpf_mode_text); static const char *osr_text[] = { "Low power", "High performance" }; -static const struct soc_enum adc_osr = - SOC_ENUM_SINGLE(WM8903_ANALOGUE_ADC_0, 0, 2, osr_text); +static SOC_ENUM_SINGLE_DECL(adc_osr, + WM8903_ANALOGUE_ADC_0, 0, osr_text); -static const struct soc_enum dac_osr = - SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_1, 0, 2, osr_text); +static SOC_ENUM_SINGLE_DECL(dac_osr, + WM8903_DAC_DIGITAL_1, 0, osr_text); static const char *drc_slope_text[] = { "1", "1/2", "1/4", "1/8", "1/16", "0" }; -static const struct soc_enum drc_slope_r0 = - SOC_ENUM_SINGLE(WM8903_DRC_2, 3, 6, drc_slope_text); +static SOC_ENUM_SINGLE_DECL(drc_slope_r0, + WM8903_DRC_2, 3, drc_slope_text); -static const struct soc_enum drc_slope_r1 = - SOC_ENUM_SINGLE(WM8903_DRC_2, 0, 6, drc_slope_text); +static SOC_ENUM_SINGLE_DECL(drc_slope_r1, + WM8903_DRC_2, 0, drc_slope_text); static const char *drc_attack_text[] = { "instantaneous", @@ -518,125 +518,125 @@ static const char *drc_attack_text[] = { "46.4ms", "92.8ms", "185.6ms" }; -static const struct soc_enum drc_attack = - SOC_ENUM_SINGLE(WM8903_DRC_1, 12, 11, drc_attack_text); +static SOC_ENUM_SINGLE_DECL(drc_attack, + WM8903_DRC_1, 12, drc_attack_text); static const char *drc_decay_text[] = { "186ms", "372ms", "743ms", "1.49s", "2.97s", "5.94s", "11.89s", "23.87s", "47.56s" }; -static const struct soc_enum drc_decay = - SOC_ENUM_SINGLE(WM8903_DRC_1, 8, 9, drc_decay_text); +static SOC_ENUM_SINGLE_DECL(drc_decay, + WM8903_DRC_1, 8, drc_decay_text); static const char *drc_ff_delay_text[] = { "5 samples", "9 samples" }; -static const struct soc_enum drc_ff_delay = - SOC_ENUM_SINGLE(WM8903_DRC_0, 5, 2, drc_ff_delay_text); +static SOC_ENUM_SINGLE_DECL(drc_ff_delay, + WM8903_DRC_0, 5, drc_ff_delay_text); static const char *drc_qr_decay_text[] = { "0.725ms", "1.45ms", "5.8ms" }; -static const struct soc_enum drc_qr_decay = - SOC_ENUM_SINGLE(WM8903_DRC_1, 4, 3, drc_qr_decay_text); +static SOC_ENUM_SINGLE_DECL(drc_qr_decay, + WM8903_DRC_1, 4, drc_qr_decay_text); static const char *drc_smoothing_text[] = { "Low", "Medium", "High" }; -static const struct soc_enum drc_smoothing = - SOC_ENUM_SINGLE(WM8903_DRC_0, 11, 3, drc_smoothing_text); +static SOC_ENUM_SINGLE_DECL(drc_smoothing, + WM8903_DRC_0, 11, drc_smoothing_text); static const char *soft_mute_text[] = { "Fast (fs/2)", "Slow (fs/32)" }; -static const struct soc_enum soft_mute = - SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_1, 10, 2, soft_mute_text); +static SOC_ENUM_SINGLE_DECL(soft_mute, + WM8903_DAC_DIGITAL_1, 10, soft_mute_text); static const char *mute_mode_text[] = { "Hard", "Soft" }; -static const struct soc_enum mute_mode = - SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_1, 9, 2, mute_mode_text); +static SOC_ENUM_SINGLE_DECL(mute_mode, + WM8903_DAC_DIGITAL_1, 9, mute_mode_text); static const char *companding_text[] = { "ulaw", "alaw" }; -static const struct soc_enum dac_companding = - SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 0, 2, companding_text); +static SOC_ENUM_SINGLE_DECL(dac_companding, + WM8903_AUDIO_INTERFACE_0, 0, companding_text); -static const struct soc_enum adc_companding = - SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 2, 2, companding_text); +static SOC_ENUM_SINGLE_DECL(adc_companding, + WM8903_AUDIO_INTERFACE_0, 2, companding_text); static const char *input_mode_text[] = { "Single-Ended", "Differential Line", "Differential Mic" }; -static const struct soc_enum linput_mode_enum = - SOC_ENUM_SINGLE(WM8903_ANALOGUE_LEFT_INPUT_1, 0, 3, input_mode_text); +static SOC_ENUM_SINGLE_DECL(linput_mode_enum, + WM8903_ANALOGUE_LEFT_INPUT_1, 0, input_mode_text); -static const struct soc_enum rinput_mode_enum = - SOC_ENUM_SINGLE(WM8903_ANALOGUE_RIGHT_INPUT_1, 0, 3, input_mode_text); +static SOC_ENUM_SINGLE_DECL(rinput_mode_enum, + WM8903_ANALOGUE_RIGHT_INPUT_1, 0, input_mode_text); static const char *linput_mux_text[] = { "IN1L", "IN2L", "IN3L" }; -static const struct soc_enum linput_enum = - SOC_ENUM_SINGLE(WM8903_ANALOGUE_LEFT_INPUT_1, 2, 3, linput_mux_text); +static SOC_ENUM_SINGLE_DECL(linput_enum, + WM8903_ANALOGUE_LEFT_INPUT_1, 2, linput_mux_text); -static const struct soc_enum linput_inv_enum = - SOC_ENUM_SINGLE(WM8903_ANALOGUE_LEFT_INPUT_1, 4, 3, linput_mux_text); +static SOC_ENUM_SINGLE_DECL(linput_inv_enum, + WM8903_ANALOGUE_LEFT_INPUT_1, 4, linput_mux_text); static const char *rinput_mux_text[] = { "IN1R", "IN2R", "IN3R" }; -static const struct soc_enum rinput_enum = - SOC_ENUM_SINGLE(WM8903_ANALOGUE_RIGHT_INPUT_1, 2, 3, rinput_mux_text); +static SOC_ENUM_SINGLE_DECL(rinput_enum, + WM8903_ANALOGUE_RIGHT_INPUT_1, 2, rinput_mux_text); -static const struct soc_enum rinput_inv_enum = - SOC_ENUM_SINGLE(WM8903_ANALOGUE_RIGHT_INPUT_1, 4, 3, rinput_mux_text); +static SOC_ENUM_SINGLE_DECL(rinput_inv_enum, + WM8903_ANALOGUE_RIGHT_INPUT_1, 4, rinput_mux_text); static const char *sidetone_text[] = { "None", "Left", "Right" }; -static const struct soc_enum lsidetone_enum = - SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_0, 2, 3, sidetone_text); +static SOC_ENUM_SINGLE_DECL(lsidetone_enum, + WM8903_DAC_DIGITAL_0, 2, sidetone_text); -static const struct soc_enum rsidetone_enum = - SOC_ENUM_SINGLE(WM8903_DAC_DIGITAL_0, 0, 3, sidetone_text); +static SOC_ENUM_SINGLE_DECL(rsidetone_enum, + WM8903_DAC_DIGITAL_0, 0, sidetone_text); static const char *adcinput_text[] = { "ADC", "DMIC" }; -static const struct soc_enum adcinput_enum = - SOC_ENUM_SINGLE(WM8903_CLOCK_RATE_TEST_4, 9, 2, adcinput_text); +static SOC_ENUM_SINGLE_DECL(adcinput_enum, + WM8903_CLOCK_RATE_TEST_4, 9, adcinput_text); static const char *aif_text[] = { "Left", "Right" }; -static const struct soc_enum lcapture_enum = - SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 7, 2, aif_text); +static SOC_ENUM_SINGLE_DECL(lcapture_enum, + WM8903_AUDIO_INTERFACE_0, 7, aif_text); -static const struct soc_enum rcapture_enum = - SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 6, 2, aif_text); +static SOC_ENUM_SINGLE_DECL(rcapture_enum, + WM8903_AUDIO_INTERFACE_0, 6, aif_text); -static const struct soc_enum lplay_enum = - SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 5, 2, aif_text); +static SOC_ENUM_SINGLE_DECL(lplay_enum, + WM8903_AUDIO_INTERFACE_0, 5, aif_text); -static const struct soc_enum rplay_enum = - SOC_ENUM_SINGLE(WM8903_AUDIO_INTERFACE_0, 4, 2, aif_text); +static SOC_ENUM_SINGLE_DECL(rplay_enum, + WM8903_AUDIO_INTERFACE_0, 4, aif_text); static const struct snd_kcontrol_new wm8903_snd_controls[] = { -- cgit v1.2.3 From d12bfd62fa936f7549a02811b4168493c67c98be Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:40:50 +0100 Subject: ASoC: wm8904: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Charles Keepax Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8904.c | 73 ++++++++++++++++++++++++----------------------- 1 file changed, 37 insertions(+), 36 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 53bbfac6a83a..cf12a9023bae 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -552,18 +552,20 @@ static const char *input_mode_text[] = { "Single-Ended", "Differential Line", "Differential Mic" }; -static const struct soc_enum lin_mode = - SOC_ENUM_SINGLE(WM8904_ANALOGUE_LEFT_INPUT_1, 0, 3, input_mode_text); +static SOC_ENUM_SINGLE_DECL(lin_mode, + WM8904_ANALOGUE_LEFT_INPUT_1, 0, + input_mode_text); -static const struct soc_enum rin_mode = - SOC_ENUM_SINGLE(WM8904_ANALOGUE_RIGHT_INPUT_1, 0, 3, input_mode_text); +static SOC_ENUM_SINGLE_DECL(rin_mode, + WM8904_ANALOGUE_RIGHT_INPUT_1, 0, + input_mode_text); static const char *hpf_mode_text[] = { "Hi-fi", "Voice 1", "Voice 2", "Voice 3" }; -static const struct soc_enum hpf_mode = - SOC_ENUM_SINGLE(WM8904_ADC_DIGITAL_0, 5, 4, hpf_mode_text); +static SOC_ENUM_SINGLE_DECL(hpf_mode, WM8904_ADC_DIGITAL_0, 5, + hpf_mode_text); static int wm8904_adc_osr_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -611,8 +613,7 @@ static const char *drc_path_text[] = { "ADC", "DAC" }; -static const struct soc_enum drc_path = - SOC_ENUM_SINGLE(WM8904_DRC_0, 14, 2, drc_path_text); +static SOC_ENUM_SINGLE_DECL(drc_path, WM8904_DRC_0, 14, drc_path_text); static const struct snd_kcontrol_new wm8904_dac_snd_controls[] = { SOC_SINGLE_TLV("Digital Playback Boost Volume", @@ -858,14 +859,14 @@ static const char *lin_text[] = { "IN1L", "IN2L", "IN3L" }; -static const struct soc_enum lin_enum = - SOC_ENUM_SINGLE(WM8904_ANALOGUE_LEFT_INPUT_1, 2, 3, lin_text); +static SOC_ENUM_SINGLE_DECL(lin_enum, WM8904_ANALOGUE_LEFT_INPUT_1, 2, + lin_text); static const struct snd_kcontrol_new lin_mux = SOC_DAPM_ENUM("Left Capture Mux", lin_enum); -static const struct soc_enum lin_inv_enum = - SOC_ENUM_SINGLE(WM8904_ANALOGUE_LEFT_INPUT_1, 4, 3, lin_text); +static SOC_ENUM_SINGLE_DECL(lin_inv_enum, WM8904_ANALOGUE_LEFT_INPUT_1, 4, + lin_text); static const struct snd_kcontrol_new lin_inv_mux = SOC_DAPM_ENUM("Left Capture Inveting Mux", lin_inv_enum); @@ -874,14 +875,14 @@ static const char *rin_text[] = { "IN1R", "IN2R", "IN3R" }; -static const struct soc_enum rin_enum = - SOC_ENUM_SINGLE(WM8904_ANALOGUE_RIGHT_INPUT_1, 2, 3, rin_text); +static SOC_ENUM_SINGLE_DECL(rin_enum, WM8904_ANALOGUE_RIGHT_INPUT_1, 2, + rin_text); static const struct snd_kcontrol_new rin_mux = SOC_DAPM_ENUM("Right Capture Mux", rin_enum); -static const struct soc_enum rin_inv_enum = - SOC_ENUM_SINGLE(WM8904_ANALOGUE_RIGHT_INPUT_1, 4, 3, rin_text); +static SOC_ENUM_SINGLE_DECL(rin_inv_enum, WM8904_ANALOGUE_RIGHT_INPUT_1, 4, + rin_text); static const struct snd_kcontrol_new rin_inv_mux = SOC_DAPM_ENUM("Right Capture Inveting Mux", rin_inv_enum); @@ -890,26 +891,26 @@ static const char *aif_text[] = { "Left", "Right" }; -static const struct soc_enum aifoutl_enum = - SOC_ENUM_SINGLE(WM8904_AUDIO_INTERFACE_0, 7, 2, aif_text); +static SOC_ENUM_SINGLE_DECL(aifoutl_enum, WM8904_AUDIO_INTERFACE_0, 7, + aif_text); static const struct snd_kcontrol_new aifoutl_mux = SOC_DAPM_ENUM("AIFOUTL Mux", aifoutl_enum); -static const struct soc_enum aifoutr_enum = - SOC_ENUM_SINGLE(WM8904_AUDIO_INTERFACE_0, 6, 2, aif_text); +static SOC_ENUM_SINGLE_DECL(aifoutr_enum, WM8904_AUDIO_INTERFACE_0, 6, + aif_text); static const struct snd_kcontrol_new aifoutr_mux = SOC_DAPM_ENUM("AIFOUTR Mux", aifoutr_enum); -static const struct soc_enum aifinl_enum = - SOC_ENUM_SINGLE(WM8904_AUDIO_INTERFACE_0, 5, 2, aif_text); +static SOC_ENUM_SINGLE_DECL(aifinl_enum, WM8904_AUDIO_INTERFACE_0, 5, + aif_text); static const struct snd_kcontrol_new aifinl_mux = SOC_DAPM_ENUM("AIFINL Mux", aifinl_enum); -static const struct soc_enum aifinr_enum = - SOC_ENUM_SINGLE(WM8904_AUDIO_INTERFACE_0, 4, 2, aif_text); +static SOC_ENUM_SINGLE_DECL(aifinr_enum, WM8904_AUDIO_INTERFACE_0, 4, + aif_text); static const struct snd_kcontrol_new aifinr_mux = SOC_DAPM_ENUM("AIFINR Mux", aifinr_enum); @@ -991,26 +992,26 @@ static const char *out_mux_text[] = { "DAC", "Bypass" }; -static const struct soc_enum hpl_enum = - SOC_ENUM_SINGLE(WM8904_ANALOGUE_OUT12_ZC, 3, 2, out_mux_text); +static SOC_ENUM_SINGLE_DECL(hpl_enum, WM8904_ANALOGUE_OUT12_ZC, 3, + out_mux_text); static const struct snd_kcontrol_new hpl_mux = SOC_DAPM_ENUM("HPL Mux", hpl_enum); -static const struct soc_enum hpr_enum = - SOC_ENUM_SINGLE(WM8904_ANALOGUE_OUT12_ZC, 2, 2, out_mux_text); +static SOC_ENUM_SINGLE_DECL(hpr_enum, WM8904_ANALOGUE_OUT12_ZC, 2, + out_mux_text); static const struct snd_kcontrol_new hpr_mux = SOC_DAPM_ENUM("HPR Mux", hpr_enum); -static const struct soc_enum linel_enum = - SOC_ENUM_SINGLE(WM8904_ANALOGUE_OUT12_ZC, 1, 2, out_mux_text); +static SOC_ENUM_SINGLE_DECL(linel_enum, WM8904_ANALOGUE_OUT12_ZC, 1, + out_mux_text); static const struct snd_kcontrol_new linel_mux = SOC_DAPM_ENUM("LINEL Mux", linel_enum); -static const struct soc_enum liner_enum = - SOC_ENUM_SINGLE(WM8904_ANALOGUE_OUT12_ZC, 0, 2, out_mux_text); +static SOC_ENUM_SINGLE_DECL(liner_enum, WM8904_ANALOGUE_OUT12_ZC, 0, + out_mux_text); static const struct snd_kcontrol_new liner_mux = SOC_DAPM_ENUM("LINER Mux", liner_enum); @@ -1019,14 +1020,14 @@ static const char *sidetone_text[] = { "None", "Left", "Right" }; -static const struct soc_enum dacl_sidetone_enum = - SOC_ENUM_SINGLE(WM8904_DAC_DIGITAL_0, 2, 3, sidetone_text); +static SOC_ENUM_SINGLE_DECL(dacl_sidetone_enum, WM8904_DAC_DIGITAL_0, 2, + sidetone_text); static const struct snd_kcontrol_new dacl_sidetone_mux = SOC_DAPM_ENUM("Left Sidetone Mux", dacl_sidetone_enum); -static const struct soc_enum dacr_sidetone_enum = - SOC_ENUM_SINGLE(WM8904_DAC_DIGITAL_0, 0, 3, sidetone_text); +static SOC_ENUM_SINGLE_DECL(dacr_sidetone_enum, WM8904_DAC_DIGITAL_0, 0, + sidetone_text); static const struct snd_kcontrol_new dacr_sidetone_mux = SOC_DAPM_ENUM("Right Sidetone Mux", dacr_sidetone_enum); -- cgit v1.2.3 From 47ef34271b8714b01ec1647d9013abf8f11fda3d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:42:06 +0100 Subject: ASoC: wm8940: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Charles Keepax Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8940.c | 20 ++++++++++---------- 1 file changed, 10 insertions(+), 10 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index b404c26c1753..87f032d0d19f 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -154,22 +154,22 @@ static const struct reg_default wm8940_reg_defaults[] = { }; static const char *wm8940_companding[] = { "Off", "NC", "u-law", "A-law" }; -static const struct soc_enum wm8940_adc_companding_enum -= SOC_ENUM_SINGLE(WM8940_COMPANDINGCTL, 1, 4, wm8940_companding); -static const struct soc_enum wm8940_dac_companding_enum -= SOC_ENUM_SINGLE(WM8940_COMPANDINGCTL, 3, 4, wm8940_companding); +static SOC_ENUM_SINGLE_DECL(wm8940_adc_companding_enum, + WM8940_COMPANDINGCTL, 1, wm8940_companding); +static SOC_ENUM_SINGLE_DECL(wm8940_dac_companding_enum, + WM8940_COMPANDINGCTL, 3, wm8940_companding); static const char *wm8940_alc_mode_text[] = {"ALC", "Limiter"}; -static const struct soc_enum wm8940_alc_mode_enum -= SOC_ENUM_SINGLE(WM8940_ALC3, 8, 2, wm8940_alc_mode_text); +static SOC_ENUM_SINGLE_DECL(wm8940_alc_mode_enum, + WM8940_ALC3, 8, wm8940_alc_mode_text); static const char *wm8940_mic_bias_level_text[] = {"0.9", "0.65"}; -static const struct soc_enum wm8940_mic_bias_level_enum -= SOC_ENUM_SINGLE(WM8940_INPUTCTL, 8, 2, wm8940_mic_bias_level_text); +static SOC_ENUM_SINGLE_DECL(wm8940_mic_bias_level_enum, + WM8940_INPUTCTL, 8, wm8940_mic_bias_level_text); static const char *wm8940_filter_mode_text[] = {"Audio", "Application"}; -static const struct soc_enum wm8940_filter_mode_enum -= SOC_ENUM_SINGLE(WM8940_ADC, 7, 2, wm8940_filter_mode_text); +static SOC_ENUM_SINGLE_DECL(wm8940_filter_mode_enum, + WM8940_ADC, 7, wm8940_filter_mode_text); static DECLARE_TLV_DB_SCALE(wm8940_spk_vol_tlv, -5700, 100, 1); static DECLARE_TLV_DB_SCALE(wm8940_att_tlv, -1000, 1000, 0); -- cgit v1.2.3 From a6616cda405c9b42732a481a71d69f0b60dafc38 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:42:45 +0100 Subject: ASoC: wm8961: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Charles Keepax Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8961.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 900328e28a15..ce8fa6e01cb4 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -317,15 +317,15 @@ static const char *adc_hpf_text[] = { "Hi-fi", "Voice 1", "Voice 2", "Voice 3", }; -static const struct soc_enum adc_hpf = - SOC_ENUM_SINGLE(WM8961_ADC_DAC_CONTROL_2, 7, 4, adc_hpf_text); +static SOC_ENUM_SINGLE_DECL(adc_hpf, + WM8961_ADC_DAC_CONTROL_2, 7, adc_hpf_text); static const char *dac_deemph_text[] = { "None", "32kHz", "44.1kHz", "48kHz", }; -static const struct soc_enum dac_deemph = - SOC_ENUM_SINGLE(WM8961_ADC_DAC_CONTROL_1, 1, 4, dac_deemph_text); +static SOC_ENUM_SINGLE_DECL(dac_deemph, + WM8961_ADC_DAC_CONTROL_1, 1, dac_deemph_text); static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1); static const DECLARE_TLV_DB_SCALE(hp_sec_tlv, -700, 100, 0); @@ -385,11 +385,11 @@ static const char *sidetone_text[] = { "None", "Left", "Right" }; -static const struct soc_enum dacl_sidetone = - SOC_ENUM_SINGLE(WM8961_DSP_SIDETONE_0, 2, 3, sidetone_text); +static SOC_ENUM_SINGLE_DECL(dacl_sidetone, + WM8961_DSP_SIDETONE_0, 2, sidetone_text); -static const struct soc_enum dacr_sidetone = - SOC_ENUM_SINGLE(WM8961_DSP_SIDETONE_1, 2, 3, sidetone_text); +static SOC_ENUM_SINGLE_DECL(dacr_sidetone, + WM8961_DSP_SIDETONE_1, 2, sidetone_text); static const struct snd_kcontrol_new dacl_mux = SOC_DAPM_ENUM("DACL Sidetone", dacl_sidetone); -- cgit v1.2.3 From da6ebf83bb8f3ad5e12b2543f15962e27939ff7f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:43:07 +0100 Subject: ASoC: wm8962: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Charles Keepax Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 32 ++++++++++++++++---------------- 1 file changed, 16 insertions(+), 16 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index c06bb5088e60..3924ee243745 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1658,16 +1658,16 @@ static const char *cap_hpf_mode_text[] = { "Hi-fi", "Application" }; -static const struct soc_enum cap_hpf_mode = - SOC_ENUM_SINGLE(WM8962_ADC_DAC_CONTROL_2, 10, 2, cap_hpf_mode_text); +static SOC_ENUM_SINGLE_DECL(cap_hpf_mode, + WM8962_ADC_DAC_CONTROL_2, 10, cap_hpf_mode_text); static const char *cap_lhpf_mode_text[] = { "LPF", "HPF" }; -static const struct soc_enum cap_lhpf_mode = - SOC_ENUM_SINGLE(WM8962_LHPF1, 1, 2, cap_lhpf_mode_text); +static SOC_ENUM_SINGLE_DECL(cap_lhpf_mode, + WM8962_LHPF1, 1, cap_lhpf_mode_text); static const struct snd_kcontrol_new wm8962_snd_controls[] = { SOC_DOUBLE("Input Mixer Switch", WM8962_INPUT_MIXER_CONTROL_1, 3, 2, 1, 1), @@ -2014,40 +2014,40 @@ static int dsp2_event(struct snd_soc_dapm_widget *w, static const char *st_text[] = { "None", "Left", "Right" }; -static const struct soc_enum str_enum = - SOC_ENUM_SINGLE(WM8962_DAC_DSP_MIXING_1, 2, 3, st_text); +static SOC_ENUM_SINGLE_DECL(str_enum, + WM8962_DAC_DSP_MIXING_1, 2, st_text); static const struct snd_kcontrol_new str_mux = SOC_DAPM_ENUM("Right Sidetone", str_enum); -static const struct soc_enum stl_enum = - SOC_ENUM_SINGLE(WM8962_DAC_DSP_MIXING_2, 2, 3, st_text); +static SOC_ENUM_SINGLE_DECL(stl_enum, + WM8962_DAC_DSP_MIXING_2, 2, st_text); static const struct snd_kcontrol_new stl_mux = SOC_DAPM_ENUM("Left Sidetone", stl_enum); static const char *outmux_text[] = { "DAC", "Mixer" }; -static const struct soc_enum spkoutr_enum = - SOC_ENUM_SINGLE(WM8962_SPEAKER_MIXER_2, 7, 2, outmux_text); +static SOC_ENUM_SINGLE_DECL(spkoutr_enum, + WM8962_SPEAKER_MIXER_2, 7, outmux_text); static const struct snd_kcontrol_new spkoutr_mux = SOC_DAPM_ENUM("SPKOUTR Mux", spkoutr_enum); -static const struct soc_enum spkoutl_enum = - SOC_ENUM_SINGLE(WM8962_SPEAKER_MIXER_1, 7, 2, outmux_text); +static SOC_ENUM_SINGLE_DECL(spkoutl_enum, + WM8962_SPEAKER_MIXER_1, 7, outmux_text); static const struct snd_kcontrol_new spkoutl_mux = SOC_DAPM_ENUM("SPKOUTL Mux", spkoutl_enum); -static const struct soc_enum hpoutr_enum = - SOC_ENUM_SINGLE(WM8962_HEADPHONE_MIXER_2, 7, 2, outmux_text); +static SOC_ENUM_SINGLE_DECL(hpoutr_enum, + WM8962_HEADPHONE_MIXER_2, 7, outmux_text); static const struct snd_kcontrol_new hpoutr_mux = SOC_DAPM_ENUM("HPOUTR Mux", hpoutr_enum); -static const struct soc_enum hpoutl_enum = - SOC_ENUM_SINGLE(WM8962_HEADPHONE_MIXER_1, 7, 2, outmux_text); +static SOC_ENUM_SINGLE_DECL(hpoutl_enum, + WM8962_HEADPHONE_MIXER_1, 7, outmux_text); static const struct snd_kcontrol_new hpoutl_mux = SOC_DAPM_ENUM("HPOUTL Mux", hpoutl_enum); -- cgit v1.2.3 From de461bd2908c44ac51ff3be6a7bf8d13ad23ce1f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:43:31 +0100 Subject: ASoC: wm8974: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Charles Keepax Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8974.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 15f45c7bd833..6e16c4306461 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -84,8 +84,8 @@ static const struct soc_enum wm8974_enum[] = { static const char *wm8974_auxmode_text[] = { "Buffer", "Mixer" }; -static const struct soc_enum wm8974_auxmode = - SOC_ENUM_SINGLE(WM8974_INPUT, 3, 2, wm8974_auxmode_text); +static SOC_ENUM_SINGLE_DECL(wm8974_auxmode, + WM8974_INPUT, 3, wm8974_auxmode_text); static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1); static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); -- cgit v1.2.3 From 2d075611c67c7a351fead73ac6bf0864049305b3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:45:12 +0100 Subject: ASoC: wm8993: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Charles Keepax Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8993.c | 64 +++++++++++++++++++++++------------------------ 1 file changed, 32 insertions(+), 32 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 2ee23a39622c..1c12f2c9418a 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -646,8 +646,8 @@ static const char *dac_deemph_text[] = { "48kHz", }; -static const struct soc_enum dac_deemph = - SOC_ENUM_SINGLE(WM8993_DAC_CTRL, 4, 4, dac_deemph_text); +static SOC_ENUM_SINGLE_DECL(dac_deemph, + WM8993_DAC_CTRL, 4, dac_deemph_text); static const char *adc_hpf_text[] = { "Hi-Fi", @@ -656,16 +656,16 @@ static const char *adc_hpf_text[] = { "Voice 3", }; -static const struct soc_enum adc_hpf = - SOC_ENUM_SINGLE(WM8993_ADC_CTRL, 5, 4, adc_hpf_text); +static SOC_ENUM_SINGLE_DECL(adc_hpf, + WM8993_ADC_CTRL, 5, adc_hpf_text); static const char *drc_path_text[] = { "ADC", "DAC" }; -static const struct soc_enum drc_path = - SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_1, 14, 2, drc_path_text); +static SOC_ENUM_SINGLE_DECL(drc_path, + WM8993_DRC_CONTROL_1, 14, drc_path_text); static const char *drc_r0_text[] = { "1", @@ -676,8 +676,8 @@ static const char *drc_r0_text[] = { "0", }; -static const struct soc_enum drc_r0 = - SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_3, 8, 6, drc_r0_text); +static SOC_ENUM_SINGLE_DECL(drc_r0, + WM8993_DRC_CONTROL_3, 8, drc_r0_text); static const char *drc_r1_text[] = { "1", @@ -687,8 +687,8 @@ static const char *drc_r1_text[] = { "0", }; -static const struct soc_enum drc_r1 = - SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_4, 13, 5, drc_r1_text); +static SOC_ENUM_SINGLE_DECL(drc_r1, + WM8993_DRC_CONTROL_4, 13, drc_r1_text); static const char *drc_attack_text[] = { "Reserved", @@ -705,8 +705,8 @@ static const char *drc_attack_text[] = { "185.6ms", }; -static const struct soc_enum drc_attack = - SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_2, 12, 12, drc_attack_text); +static SOC_ENUM_SINGLE_DECL(drc_attack, + WM8993_DRC_CONTROL_2, 12, drc_attack_text); static const char *drc_decay_text[] = { "186ms", @@ -720,16 +720,16 @@ static const char *drc_decay_text[] = { "47.56ms", }; -static const struct soc_enum drc_decay = - SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_2, 8, 9, drc_decay_text); +static SOC_ENUM_SINGLE_DECL(drc_decay, + WM8993_DRC_CONTROL_2, 8, drc_decay_text); static const char *drc_ff_text[] = { "5 samples", "9 samples", }; -static const struct soc_enum drc_ff = - SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_3, 7, 2, drc_ff_text); +static SOC_ENUM_SINGLE_DECL(drc_ff, + WM8993_DRC_CONTROL_3, 7, drc_ff_text); static const char *drc_qr_rate_text[] = { "0.725ms", @@ -737,8 +737,8 @@ static const char *drc_qr_rate_text[] = { "5.8ms", }; -static const struct soc_enum drc_qr_rate = - SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_3, 0, 3, drc_qr_rate_text); +static SOC_ENUM_SINGLE_DECL(drc_qr_rate, + WM8993_DRC_CONTROL_3, 0, drc_qr_rate_text); static const char *drc_smooth_text[] = { "Low", @@ -746,8 +746,8 @@ static const char *drc_smooth_text[] = { "High", }; -static const struct soc_enum drc_smooth = - SOC_ENUM_SINGLE(WM8993_DRC_CONTROL_1, 4, 3, drc_smooth_text); +static SOC_ENUM_SINGLE_DECL(drc_smooth, + WM8993_DRC_CONTROL_1, 4, drc_smooth_text); static const struct snd_kcontrol_new wm8993_snd_controls[] = { SOC_DOUBLE_TLV("Digital Sidetone Volume", WM8993_DIGITAL_SIDE_TONE, @@ -841,26 +841,26 @@ static const char *aif_text[] = { "Left", "Right" }; -static const struct soc_enum aifoutl_enum = - SOC_ENUM_SINGLE(WM8993_AUDIO_INTERFACE_1, 15, 2, aif_text); +static SOC_ENUM_SINGLE_DECL(aifoutl_enum, + WM8993_AUDIO_INTERFACE_1, 15, aif_text); static const struct snd_kcontrol_new aifoutl_mux = SOC_DAPM_ENUM("AIFOUTL Mux", aifoutl_enum); -static const struct soc_enum aifoutr_enum = - SOC_ENUM_SINGLE(WM8993_AUDIO_INTERFACE_1, 14, 2, aif_text); +static SOC_ENUM_SINGLE_DECL(aifoutr_enum, + WM8993_AUDIO_INTERFACE_1, 14, aif_text); static const struct snd_kcontrol_new aifoutr_mux = SOC_DAPM_ENUM("AIFOUTR Mux", aifoutr_enum); -static const struct soc_enum aifinl_enum = - SOC_ENUM_SINGLE(WM8993_AUDIO_INTERFACE_2, 15, 2, aif_text); +static SOC_ENUM_SINGLE_DECL(aifinl_enum, + WM8993_AUDIO_INTERFACE_2, 15, aif_text); static const struct snd_kcontrol_new aifinl_mux = SOC_DAPM_ENUM("AIFINL Mux", aifinl_enum); -static const struct soc_enum aifinr_enum = - SOC_ENUM_SINGLE(WM8993_AUDIO_INTERFACE_2, 14, 2, aif_text); +static SOC_ENUM_SINGLE_DECL(aifinr_enum, + WM8993_AUDIO_INTERFACE_2, 14, aif_text); static const struct snd_kcontrol_new aifinr_mux = SOC_DAPM_ENUM("AIFINR Mux", aifinr_enum); @@ -869,14 +869,14 @@ static const char *sidetone_text[] = { "None", "Left", "Right" }; -static const struct soc_enum sidetonel_enum = - SOC_ENUM_SINGLE(WM8993_DIGITAL_SIDE_TONE, 2, 3, sidetone_text); +static SOC_ENUM_SINGLE_DECL(sidetonel_enum, + WM8993_DIGITAL_SIDE_TONE, 2, sidetone_text); static const struct snd_kcontrol_new sidetonel_mux = SOC_DAPM_ENUM("Left Sidetone", sidetonel_enum); -static const struct soc_enum sidetoner_enum = - SOC_ENUM_SINGLE(WM8993_DIGITAL_SIDE_TONE, 0, 3, sidetone_text); +static SOC_ENUM_SINGLE_DECL(sidetoner_enum, + WM8993_DIGITAL_SIDE_TONE, 0, sidetone_text); static const struct snd_kcontrol_new sidetoner_mux = SOC_DAPM_ENUM("Right Sidetone", sidetoner_enum); -- cgit v1.2.3 From 15b49e73d375a952e34d31f7aac92944df003ca3 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:46:47 +0100 Subject: ASoC: wm8995: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Charles Keepax Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8995.c | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 403e1db6870f..0c539bf1105c 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -781,14 +781,12 @@ static const char *sidetone_text[] = { "ADC/DMIC1", "DMIC2", }; -static const struct soc_enum sidetone1_enum = - SOC_ENUM_SINGLE(WM8995_SIDETONE, 0, 2, sidetone_text); +static SOC_ENUM_SINGLE_DECL(sidetone1_enum, WM8995_SIDETONE, 0, sidetone_text); static const struct snd_kcontrol_new sidetone1_mux = SOC_DAPM_ENUM("Left Sidetone Mux", sidetone1_enum); -static const struct soc_enum sidetone2_enum = - SOC_ENUM_SINGLE(WM8995_SIDETONE, 1, 2, sidetone_text); +static SOC_ENUM_SINGLE_DECL(sidetone2_enum, WM8995_SIDETONE, 1, sidetone_text); static const struct snd_kcontrol_new sidetone2_mux = SOC_DAPM_ENUM("Right Sidetone Mux", sidetone2_enum); @@ -884,8 +882,7 @@ static const char *adc_mux_text[] = { "DMIC", }; -static const struct soc_enum adc_enum = - SOC_ENUM_SINGLE(0, 0, 2, adc_mux_text); +static const SOC_ENUM_SINGLE_DECL(adc_enum, 0, 0, adc_mux_text); static const struct snd_kcontrol_new adcl_mux = SOC_DAPM_ENUM_VIRT("ADCL Mux", adc_enum); -- cgit v1.2.3 From 5cca5a916f7ade910e2996e3b44ddffee8e08caf Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:47:05 +0100 Subject: ASoC: wm8996: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Charles Keepax Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8996.c | 56 +++++++++++++++++++++++------------------------ 1 file changed, 28 insertions(+), 28 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 1a7655b0aa22..92bb02185c46 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -311,28 +311,28 @@ static const char *sidetone_hpf_text[] = { "2.9kHz", "1.5kHz", "735Hz", "403Hz", "196Hz", "98Hz", "49Hz" }; -static const struct soc_enum sidetone_hpf = - SOC_ENUM_SINGLE(WM8996_SIDETONE, 7, 7, sidetone_hpf_text); +static SOC_ENUM_SINGLE_DECL(sidetone_hpf, + WM8996_SIDETONE, 7, sidetone_hpf_text); static const char *hpf_mode_text[] = { "HiFi", "Custom", "Voice" }; -static const struct soc_enum dsp1tx_hpf_mode = - SOC_ENUM_SINGLE(WM8996_DSP1_TX_FILTERS, 3, 3, hpf_mode_text); +static SOC_ENUM_SINGLE_DECL(dsp1tx_hpf_mode, + WM8996_DSP1_TX_FILTERS, 3, hpf_mode_text); -static const struct soc_enum dsp2tx_hpf_mode = - SOC_ENUM_SINGLE(WM8996_DSP2_TX_FILTERS, 3, 3, hpf_mode_text); +static SOC_ENUM_SINGLE_DECL(dsp2tx_hpf_mode, + WM8996_DSP2_TX_FILTERS, 3, hpf_mode_text); static const char *hpf_cutoff_text[] = { "50Hz", "75Hz", "100Hz", "150Hz", "200Hz", "300Hz", "400Hz" }; -static const struct soc_enum dsp1tx_hpf_cutoff = - SOC_ENUM_SINGLE(WM8996_DSP1_TX_FILTERS, 0, 7, hpf_cutoff_text); +static SOC_ENUM_SINGLE_DECL(dsp1tx_hpf_cutoff, + WM8996_DSP1_TX_FILTERS, 0, hpf_cutoff_text); -static const struct soc_enum dsp2tx_hpf_cutoff = - SOC_ENUM_SINGLE(WM8996_DSP2_TX_FILTERS, 0, 7, hpf_cutoff_text); +static SOC_ENUM_SINGLE_DECL(dsp2tx_hpf_cutoff, + WM8996_DSP2_TX_FILTERS, 0, hpf_cutoff_text); static void wm8996_set_retune_mobile(struct snd_soc_codec *codec, int block) { @@ -780,14 +780,14 @@ static const char *sidetone_text[] = { "IN1", "IN2", }; -static const struct soc_enum left_sidetone_enum = - SOC_ENUM_SINGLE(WM8996_SIDETONE, 0, 2, sidetone_text); +static SOC_ENUM_SINGLE_DECL(left_sidetone_enum, + WM8996_SIDETONE, 0, sidetone_text); static const struct snd_kcontrol_new left_sidetone = SOC_DAPM_ENUM("Left Sidetone", left_sidetone_enum); -static const struct soc_enum right_sidetone_enum = - SOC_ENUM_SINGLE(WM8996_SIDETONE, 1, 2, sidetone_text); +static SOC_ENUM_SINGLE_DECL(right_sidetone_enum, + WM8996_SIDETONE, 1, sidetone_text); static const struct snd_kcontrol_new right_sidetone = SOC_DAPM_ENUM("Right Sidetone", right_sidetone_enum); @@ -796,14 +796,14 @@ static const char *spk_text[] = { "DAC1L", "DAC1R", "DAC2L", "DAC2R" }; -static const struct soc_enum spkl_enum = - SOC_ENUM_SINGLE(WM8996_LEFT_PDM_SPEAKER, 0, 4, spk_text); +static SOC_ENUM_SINGLE_DECL(spkl_enum, + WM8996_LEFT_PDM_SPEAKER, 0, spk_text); static const struct snd_kcontrol_new spkl_mux = SOC_DAPM_ENUM("SPKL", spkl_enum); -static const struct soc_enum spkr_enum = - SOC_ENUM_SINGLE(WM8996_RIGHT_PDM_SPEAKER, 0, 4, spk_text); +static SOC_ENUM_SINGLE_DECL(spkr_enum, + WM8996_RIGHT_PDM_SPEAKER, 0, spk_text); static const struct snd_kcontrol_new spkr_mux = SOC_DAPM_ENUM("SPKR", spkr_enum); @@ -812,8 +812,8 @@ static const char *dsp1rx_text[] = { "AIF1", "AIF2" }; -static const struct soc_enum dsp1rx_enum = - SOC_ENUM_SINGLE(WM8996_POWER_MANAGEMENT_8, 0, 2, dsp1rx_text); +static SOC_ENUM_SINGLE_DECL(dsp1rx_enum, + WM8996_POWER_MANAGEMENT_8, 0, dsp1rx_text); static const struct snd_kcontrol_new dsp1rx = SOC_DAPM_ENUM("DSP1RX", dsp1rx_enum); @@ -822,8 +822,8 @@ static const char *dsp2rx_text[] = { "AIF2", "AIF1" }; -static const struct soc_enum dsp2rx_enum = - SOC_ENUM_SINGLE(WM8996_POWER_MANAGEMENT_8, 4, 2, dsp2rx_text); +static SOC_ENUM_SINGLE_DECL(dsp2rx_enum, + WM8996_POWER_MANAGEMENT_8, 4, dsp2rx_text); static const struct snd_kcontrol_new dsp2rx = SOC_DAPM_ENUM("DSP2RX", dsp2rx_enum); @@ -832,8 +832,8 @@ static const char *aif2tx_text[] = { "DSP2", "DSP1", "AIF1" }; -static const struct soc_enum aif2tx_enum = - SOC_ENUM_SINGLE(WM8996_POWER_MANAGEMENT_8, 6, 3, aif2tx_text); +static SOC_ENUM_SINGLE_DECL(aif2tx_enum, + WM8996_POWER_MANAGEMENT_8, 6, aif2tx_text); static const struct snd_kcontrol_new aif2tx = SOC_DAPM_ENUM("AIF2TX", aif2tx_enum); @@ -842,14 +842,14 @@ static const char *inmux_text[] = { "ADC", "DMIC1", "DMIC2" }; -static const struct soc_enum in1_enum = - SOC_ENUM_SINGLE(WM8996_POWER_MANAGEMENT_7, 0, 3, inmux_text); +static SOC_ENUM_SINGLE_DECL(in1_enum, + WM8996_POWER_MANAGEMENT_7, 0, inmux_text); static const struct snd_kcontrol_new in1_mux = SOC_DAPM_ENUM("IN1 Mux", in1_enum); -static const struct soc_enum in2_enum = - SOC_ENUM_SINGLE(WM8996_POWER_MANAGEMENT_7, 4, 3, inmux_text); +static SOC_ENUM_SINGLE_DECL(in2_enum, + WM8996_POWER_MANAGEMENT_7, 4, inmux_text); static const struct snd_kcontrol_new in2_mux = SOC_DAPM_ENUM("IN2 Mux", in2_enum); -- cgit v1.2.3 From abc4b4fb94a274055f5da900976d0970f73a00cc Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:47:17 +0100 Subject: ASoC: wm_hubs: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Charles Keepax Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm_hubs.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index b371066dd5bc..b6209662ab13 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -50,16 +50,16 @@ static const char *speaker_ref_text[] = { "VMID", }; -static const struct soc_enum speaker_ref = - SOC_ENUM_SINGLE(WM8993_SPEAKER_MIXER, 8, 2, speaker_ref_text); +static SOC_ENUM_SINGLE_DECL(speaker_ref, + WM8993_SPEAKER_MIXER, 8, speaker_ref_text); static const char *speaker_mode_text[] = { "Class D", "Class AB", }; -static const struct soc_enum speaker_mode = - SOC_ENUM_SINGLE(WM8993_SPKMIXR_ATTENUATION, 8, 2, speaker_mode_text); +static SOC_ENUM_SINGLE_DECL(speaker_mode, + WM8993_SPKMIXR_ATTENUATION, 8, speaker_mode_text); static void wait_for_dc_servo(struct snd_soc_codec *codec, unsigned int op) { @@ -735,15 +735,15 @@ static const char *hp_mux_text[] = { "DAC", }; -static const struct soc_enum hpl_enum = - SOC_ENUM_SINGLE(WM8993_OUTPUT_MIXER1, 8, 2, hp_mux_text); +static SOC_ENUM_SINGLE_DECL(hpl_enum, + WM8993_OUTPUT_MIXER1, 8, hp_mux_text); const struct snd_kcontrol_new wm_hubs_hpl_mux = WM_HUBS_ENUM_W("Left Headphone Mux", hpl_enum); EXPORT_SYMBOL_GPL(wm_hubs_hpl_mux); -static const struct soc_enum hpr_enum = - SOC_ENUM_SINGLE(WM8993_OUTPUT_MIXER2, 8, 2, hp_mux_text); +static SOC_ENUM_SINGLE_DECL(hpr_enum, + WM8993_OUTPUT_MIXER2, 8, hp_mux_text); const struct snd_kcontrol_new wm_hubs_hpr_mux = WM_HUBS_ENUM_W("Right Headphone Mux", hpr_enum); -- cgit v1.2.3 From 6eb0e8f90f5156bb8441c8427b1c11ed9401178f Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:43:13 +0100 Subject: ASoC: 88pm860x: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/88pm860x-codec.c | 116 +++++++++++++++++++------------------- 1 file changed, 58 insertions(+), 58 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 75d0ad5d2dcb..697d24be7897 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -448,38 +448,38 @@ static const char *pm860x_opamp_texts[] = {"-50%", "-25%", "0%", "75%"}; static const char *pm860x_pa_texts[] = {"-33%", "0%", "33%", "66%"}; -static const struct soc_enum pm860x_hs1_opamp_enum = - SOC_ENUM_SINGLE(PM860X_HS1_CTRL, 5, 4, pm860x_opamp_texts); +static SOC_ENUM_SINGLE_DECL(pm860x_hs1_opamp_enum, + PM860X_HS1_CTRL, 5, pm860x_opamp_texts); -static const struct soc_enum pm860x_hs2_opamp_enum = - SOC_ENUM_SINGLE(PM860X_HS2_CTRL, 5, 4, pm860x_opamp_texts); +static SOC_ENUM_SINGLE_DECL(pm860x_hs2_opamp_enum, + PM860X_HS2_CTRL, 5, pm860x_opamp_texts); -static const struct soc_enum pm860x_hs1_pa_enum = - SOC_ENUM_SINGLE(PM860X_HS1_CTRL, 3, 4, pm860x_pa_texts); +static SOC_ENUM_SINGLE_DECL(pm860x_hs1_pa_enum, + PM860X_HS1_CTRL, 3, pm860x_pa_texts); -static const struct soc_enum pm860x_hs2_pa_enum = - SOC_ENUM_SINGLE(PM860X_HS2_CTRL, 3, 4, pm860x_pa_texts); +static SOC_ENUM_SINGLE_DECL(pm860x_hs2_pa_enum, + PM860X_HS2_CTRL, 3, pm860x_pa_texts); -static const struct soc_enum pm860x_lo1_opamp_enum = - SOC_ENUM_SINGLE(PM860X_LO1_CTRL, 5, 4, pm860x_opamp_texts); +static SOC_ENUM_SINGLE_DECL(pm860x_lo1_opamp_enum, + PM860X_LO1_CTRL, 5, pm860x_opamp_texts); -static const struct soc_enum pm860x_lo2_opamp_enum = - SOC_ENUM_SINGLE(PM860X_LO2_CTRL, 5, 4, pm860x_opamp_texts); +static SOC_ENUM_SINGLE_DECL(pm860x_lo2_opamp_enum, + PM860X_LO2_CTRL, 5, pm860x_opamp_texts); -static const struct soc_enum pm860x_lo1_pa_enum = - SOC_ENUM_SINGLE(PM860X_LO1_CTRL, 3, 4, pm860x_pa_texts); +static SOC_ENUM_SINGLE_DECL(pm860x_lo1_pa_enum, + PM860X_LO1_CTRL, 3, pm860x_pa_texts); -static const struct soc_enum pm860x_lo2_pa_enum = - SOC_ENUM_SINGLE(PM860X_LO2_CTRL, 3, 4, pm860x_pa_texts); +static SOC_ENUM_SINGLE_DECL(pm860x_lo2_pa_enum, + PM860X_LO2_CTRL, 3, pm860x_pa_texts); -static const struct soc_enum pm860x_spk_pa_enum = - SOC_ENUM_SINGLE(PM860X_EAR_CTRL_1, 5, 4, pm860x_pa_texts); +static SOC_ENUM_SINGLE_DECL(pm860x_spk_pa_enum, + PM860X_EAR_CTRL_1, 5, pm860x_pa_texts); -static const struct soc_enum pm860x_ear_pa_enum = - SOC_ENUM_SINGLE(PM860X_EAR_CTRL_2, 0, 4, pm860x_pa_texts); +static SOC_ENUM_SINGLE_DECL(pm860x_ear_pa_enum, + PM860X_EAR_CTRL_2, 0, pm860x_pa_texts); -static const struct soc_enum pm860x_spk_ear_opamp_enum = - SOC_ENUM_SINGLE(PM860X_EAR_CTRL_1, 3, 4, pm860x_opamp_texts); +static SOC_ENUM_SINGLE_DECL(pm860x_spk_ear_opamp_enum, + PM860X_EAR_CTRL_1, 3, pm860x_opamp_texts); static const struct snd_kcontrol_new pm860x_snd_controls[] = { SOC_DOUBLE_R_TLV("ADC Capture Volume", PM860X_ADC_ANA_2, @@ -561,8 +561,8 @@ static const char *aif1_text[] = { "PCM L", "PCM R", }; -static const struct soc_enum aif1_enum = - SOC_ENUM_SINGLE(PM860X_PCM_IFACE_3, 6, 2, aif1_text); +static SOC_ENUM_SINGLE_DECL(aif1_enum, + PM860X_PCM_IFACE_3, 6, aif1_text); static const struct snd_kcontrol_new aif1_mux = SOC_DAPM_ENUM("PCM Mux", aif1_enum); @@ -572,8 +572,8 @@ static const char *i2s_din_text[] = { "DIN", "DIN1", }; -static const struct soc_enum i2s_din_enum = - SOC_ENUM_SINGLE(PM860X_I2S_IFACE_3, 1, 2, i2s_din_text); +static SOC_ENUM_SINGLE_DECL(i2s_din_enum, + PM860X_I2S_IFACE_3, 1, i2s_din_text); static const struct snd_kcontrol_new i2s_din_mux = SOC_DAPM_ENUM("I2S DIN Mux", i2s_din_enum); @@ -583,8 +583,8 @@ static const char *i2s_mic_text[] = { "Ex PA", "ADC", }; -static const struct soc_enum i2s_mic_enum = - SOC_ENUM_SINGLE(PM860X_I2S_IFACE_3, 4, 2, i2s_mic_text); +static SOC_ENUM_SINGLE_DECL(i2s_mic_enum, + PM860X_I2S_IFACE_3, 4, i2s_mic_text); static const struct snd_kcontrol_new i2s_mic_mux = SOC_DAPM_ENUM("I2S Mic Mux", i2s_mic_enum); @@ -594,8 +594,8 @@ static const char *adcl_text[] = { "ADCR", "ADCL", }; -static const struct soc_enum adcl_enum = - SOC_ENUM_SINGLE(PM860X_PCM_IFACE_3, 4, 2, adcl_text); +static SOC_ENUM_SINGLE_DECL(adcl_enum, + PM860X_PCM_IFACE_3, 4, adcl_text); static const struct snd_kcontrol_new adcl_mux = SOC_DAPM_ENUM("ADC Left Mux", adcl_enum); @@ -605,8 +605,8 @@ static const char *adcr_text[] = { "ADCL", "ADCR", }; -static const struct soc_enum adcr_enum = - SOC_ENUM_SINGLE(PM860X_PCM_IFACE_3, 2, 2, adcr_text); +static SOC_ENUM_SINGLE_DECL(adcr_enum, + PM860X_PCM_IFACE_3, 2, adcr_text); static const struct snd_kcontrol_new adcr_mux = SOC_DAPM_ENUM("ADC Right Mux", adcr_enum); @@ -616,8 +616,8 @@ static const char *adcr_ec_text[] = { "ADCR", "EC", }; -static const struct soc_enum adcr_ec_enum = - SOC_ENUM_SINGLE(PM860X_ADC_EN_2, 3, 2, adcr_ec_text); +static SOC_ENUM_SINGLE_DECL(adcr_ec_enum, + PM860X_ADC_EN_2, 3, adcr_ec_text); static const struct snd_kcontrol_new adcr_ec_mux = SOC_DAPM_ENUM("ADCR EC Mux", adcr_ec_enum); @@ -627,8 +627,8 @@ static const char *ec_text[] = { "Left", "Right", "Left + Right", }; -static const struct soc_enum ec_enum = - SOC_ENUM_SINGLE(PM860X_EC_PATH, 1, 3, ec_text); +static SOC_ENUM_SINGLE_DECL(ec_enum, + PM860X_EC_PATH, 1, ec_text); static const struct snd_kcontrol_new ec_mux = SOC_DAPM_ENUM("EC Mux", ec_enum); @@ -638,36 +638,36 @@ static const char *dac_text[] = { }; /* DAC Headset 1 Mux / Mux10 */ -static const struct soc_enum dac_hs1_enum = - SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 0, 4, dac_text); +static SOC_ENUM_SINGLE_DECL(dac_hs1_enum, + PM860X_ANA_INPUT_SEL_1, 0, dac_text); static const struct snd_kcontrol_new dac_hs1_mux = SOC_DAPM_ENUM("DAC HS1 Mux", dac_hs1_enum); /* DAC Headset 2 Mux / Mux11 */ -static const struct soc_enum dac_hs2_enum = - SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 2, 4, dac_text); +static SOC_ENUM_SINGLE_DECL(dac_hs2_enum, + PM860X_ANA_INPUT_SEL_1, 2, dac_text); static const struct snd_kcontrol_new dac_hs2_mux = SOC_DAPM_ENUM("DAC HS2 Mux", dac_hs2_enum); /* DAC Lineout 1 Mux / Mux12 */ -static const struct soc_enum dac_lo1_enum = - SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 4, 4, dac_text); +static SOC_ENUM_SINGLE_DECL(dac_lo1_enum, + PM860X_ANA_INPUT_SEL_1, 4, dac_text); static const struct snd_kcontrol_new dac_lo1_mux = SOC_DAPM_ENUM("DAC LO1 Mux", dac_lo1_enum); /* DAC Lineout 2 Mux / Mux13 */ -static const struct soc_enum dac_lo2_enum = - SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_1, 6, 4, dac_text); +static SOC_ENUM_SINGLE_DECL(dac_lo2_enum, + PM860X_ANA_INPUT_SEL_1, 6, dac_text); static const struct snd_kcontrol_new dac_lo2_mux = SOC_DAPM_ENUM("DAC LO2 Mux", dac_lo2_enum); /* DAC Spearker Earphone Mux / Mux14 */ -static const struct soc_enum dac_spk_ear_enum = - SOC_ENUM_SINGLE(PM860X_ANA_INPUT_SEL_2, 0, 4, dac_text); +static SOC_ENUM_SINGLE_DECL(dac_spk_ear_enum, + PM860X_ANA_INPUT_SEL_2, 0, dac_text); static const struct snd_kcontrol_new dac_spk_ear_mux = SOC_DAPM_ENUM("DAC SP Mux", dac_spk_ear_enum); @@ -677,29 +677,29 @@ static const char *in_text[] = { "Digital", "Analog", }; -static const struct soc_enum hs1_enum = - SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 0, 2, in_text); +static SOC_ENUM_SINGLE_DECL(hs1_enum, + PM860X_ANA_TO_ANA, 0, in_text); static const struct snd_kcontrol_new hs1_mux = SOC_DAPM_ENUM("Headset1 Mux", hs1_enum); /* Headset 2 Mux / Mux16 */ -static const struct soc_enum hs2_enum = - SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 1, 2, in_text); +static SOC_ENUM_SINGLE_DECL(hs2_enum, + PM860X_ANA_TO_ANA, 1, in_text); static const struct snd_kcontrol_new hs2_mux = SOC_DAPM_ENUM("Headset2 Mux", hs2_enum); /* Lineout 1 Mux / Mux17 */ -static const struct soc_enum lo1_enum = - SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 2, 2, in_text); +static SOC_ENUM_SINGLE_DECL(lo1_enum, + PM860X_ANA_TO_ANA, 2, in_text); static const struct snd_kcontrol_new lo1_mux = SOC_DAPM_ENUM("Lineout1 Mux", lo1_enum); /* Lineout 2 Mux / Mux18 */ -static const struct soc_enum lo2_enum = - SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 3, 2, in_text); +static SOC_ENUM_SINGLE_DECL(lo2_enum, + PM860X_ANA_TO_ANA, 3, in_text); static const struct snd_kcontrol_new lo2_mux = SOC_DAPM_ENUM("Lineout2 Mux", lo2_enum); @@ -709,8 +709,8 @@ static const char *spk_text[] = { "Earpiece", "Speaker", }; -static const struct soc_enum spk_enum = - SOC_ENUM_SINGLE(PM860X_ANA_TO_ANA, 6, 2, spk_text); +static SOC_ENUM_SINGLE_DECL(spk_enum, + PM860X_ANA_TO_ANA, 6, spk_text); static const struct snd_kcontrol_new spk_demux = SOC_DAPM_ENUM("Speaker Earpiece Demux", spk_enum); @@ -720,8 +720,8 @@ static const char *mic_text[] = { "Mic 1", "Mic 2", }; -static const struct soc_enum mic_enum = - SOC_ENUM_SINGLE(PM860X_ADC_ANA_4, 4, 2, mic_text); +static SOC_ENUM_SINGLE_DECL(mic_enum, + PM860X_ADC_ANA_4, 4, mic_text); static const struct snd_kcontrol_new mic_mux = SOC_DAPM_ENUM("MIC Mux", mic_enum); -- cgit v1.2.3 From cbf6242281069c3405b57fc10563d094aae6c9c0 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:43:45 +0100 Subject: ASoC: ak4641: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ak4641.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 94cbe508dd37..684b56f2856a 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -113,14 +113,14 @@ static const DECLARE_TLV_DB_SCALE(alc_tlv, -800, 50, 0); static const DECLARE_TLV_DB_SCALE(aux_in_tlv, -2100, 300, 0); -static const struct soc_enum ak4641_mono_out_enum = - SOC_ENUM_SINGLE(AK4641_SIG1, 6, 2, ak4641_mono_out); -static const struct soc_enum ak4641_hp_out_enum = - SOC_ENUM_SINGLE(AK4641_MODE2, 2, 2, ak4641_hp_out); -static const struct soc_enum ak4641_mic_select_enum = - SOC_ENUM_SINGLE(AK4641_MIC, 1, 2, ak4641_mic_select); -static const struct soc_enum ak4641_mic_or_dac_enum = - SOC_ENUM_SINGLE(AK4641_BTIF, 4, 2, ak4641_mic_or_dac); +static SOC_ENUM_SINGLE_DECL(ak4641_mono_out_enum, + AK4641_SIG1, 6, ak4641_mono_out); +static SOC_ENUM_SINGLE_DECL(ak4641_hp_out_enum, + AK4641_MODE2, 2, ak4641_hp_out); +static SOC_ENUM_SINGLE_DECL(ak4641_mic_select_enum, + AK4641_MIC, 1, ak4641_mic_select); +static SOC_ENUM_SINGLE_DECL(ak4641_mic_or_dac_enum, + AK4641_BTIF, 4, ak4641_mic_or_dac); static const struct snd_kcontrol_new ak4641_snd_controls[] = { SOC_ENUM("Mono 1 Output", ak4641_mono_out_enum), -- cgit v1.2.3 From 64e6b58db9c3ee550092c73e18e36ad9dc168334 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:43:58 +0100 Subject: ASoC: ak4671: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ak4671.c | 14 ++++++-------- 1 file changed, 6 insertions(+), 8 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index 743bbe31bc08..deb2b44669de 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -237,19 +237,17 @@ static const struct snd_kcontrol_new ak4671_rout3_mixer_controls[] = { /* Input MUXs */ static const char *ak4671_lin_mux_texts[] = {"LIN1", "LIN2", "LIN3", "LIN4"}; -static const struct soc_enum ak4671_lin_mux_enum = - SOC_ENUM_SINGLE(AK4671_MIC_SIGNAL_SELECT, 0, - ARRAY_SIZE(ak4671_lin_mux_texts), - ak4671_lin_mux_texts); +static SOC_ENUM_SINGLE_DECL(ak4671_lin_mux_enum, + AK4671_MIC_SIGNAL_SELECT, 0, + ak4671_lin_mux_texts); static const struct snd_kcontrol_new ak4671_lin_mux_control = SOC_DAPM_ENUM("Route", ak4671_lin_mux_enum); static const char *ak4671_rin_mux_texts[] = {"RIN1", "RIN2", "RIN3", "RIN4"}; -static const struct soc_enum ak4671_rin_mux_enum = - SOC_ENUM_SINGLE(AK4671_MIC_SIGNAL_SELECT, 2, - ARRAY_SIZE(ak4671_rin_mux_texts), - ak4671_rin_mux_texts); +static SOC_ENUM_SINGLE_DECL(ak4671_rin_mux_enum, + AK4671_MIC_SIGNAL_SELECT, 2, + ak4671_rin_mux_texts); static const struct snd_kcontrol_new ak4671_rin_mux_control = SOC_DAPM_ENUM("Route", ak4671_rin_mux_enum); -- cgit v1.2.3 From e84f2463765ff34a322511be815ad235dcc1f218 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:44:32 +0100 Subject: ASoC: alc5632: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/alc5632.c | 40 ++++++++++++++++++++++++---------------- 1 file changed, 24 insertions(+), 16 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index fb001c56cf8d..d885056ad8f2 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -293,51 +293,59 @@ static const char * const alc5632_i2s_out_sel[] = { "ADC LR", "Voice Stereo Digital"}; /* auxout output mux */ -static const struct soc_enum alc5632_aux_out_input_enum = -SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 6, 4, alc5632_aux_out_input_sel); +static SOC_ENUM_SINGLE_DECL(alc5632_aux_out_input_enum, + ALC5632_OUTPUT_MIXER_CTRL, 6, + alc5632_aux_out_input_sel); static const struct snd_kcontrol_new alc5632_auxout_mux_controls = SOC_DAPM_ENUM("AuxOut Mux", alc5632_aux_out_input_enum); /* speaker output mux */ -static const struct soc_enum alc5632_spkout_input_enum = -SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 10, 4, alc5632_spkout_input_sel); +static SOC_ENUM_SINGLE_DECL(alc5632_spkout_input_enum, + ALC5632_OUTPUT_MIXER_CTRL, 10, + alc5632_spkout_input_sel); static const struct snd_kcontrol_new alc5632_spkout_mux_controls = SOC_DAPM_ENUM("SpeakerOut Mux", alc5632_spkout_input_enum); /* headphone left output mux */ -static const struct soc_enum alc5632_hpl_out_input_enum = -SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 9, 2, alc5632_hpl_out_input_sel); +static SOC_ENUM_SINGLE_DECL(alc5632_hpl_out_input_enum, + ALC5632_OUTPUT_MIXER_CTRL, 9, + alc5632_hpl_out_input_sel); static const struct snd_kcontrol_new alc5632_hpl_out_mux_controls = SOC_DAPM_ENUM("Left Headphone Mux", alc5632_hpl_out_input_enum); /* headphone right output mux */ -static const struct soc_enum alc5632_hpr_out_input_enum = -SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 8, 2, alc5632_hpr_out_input_sel); +static SOC_ENUM_SINGLE_DECL(alc5632_hpr_out_input_enum, + ALC5632_OUTPUT_MIXER_CTRL, 8, + alc5632_hpr_out_input_sel); static const struct snd_kcontrol_new alc5632_hpr_out_mux_controls = SOC_DAPM_ENUM("Right Headphone Mux", alc5632_hpr_out_input_enum); /* speaker output N select */ -static const struct soc_enum alc5632_spk_n_sour_enum = -SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 14, 4, alc5632_spk_n_sour_sel); +static SOC_ENUM_SINGLE_DECL(alc5632_spk_n_sour_enum, + ALC5632_OUTPUT_MIXER_CTRL, 14, + alc5632_spk_n_sour_sel); static const struct snd_kcontrol_new alc5632_spkoutn_mux_controls = SOC_DAPM_ENUM("SpeakerOut N Mux", alc5632_spk_n_sour_enum); /* speaker amplifier */ static const char *alc5632_amp_names[] = {"AB Amp", "D Amp"}; -static const struct soc_enum alc5632_amp_enum = - SOC_ENUM_SINGLE(ALC5632_OUTPUT_MIXER_CTRL, 13, 2, alc5632_amp_names); +static SOC_ENUM_SINGLE_DECL(alc5632_amp_enum, + ALC5632_OUTPUT_MIXER_CTRL, 13, + alc5632_amp_names); static const struct snd_kcontrol_new alc5632_amp_mux_controls = SOC_DAPM_ENUM("AB-D Amp Mux", alc5632_amp_enum); /* ADC output select */ -static const struct soc_enum alc5632_adcr_func_enum = - SOC_ENUM_SINGLE(ALC5632_DAC_FUNC_SELECT, 5, 2, alc5632_adcr_func_sel); +static SOC_ENUM_SINGLE_DECL(alc5632_adcr_func_enum, + ALC5632_DAC_FUNC_SELECT, 5, + alc5632_adcr_func_sel); static const struct snd_kcontrol_new alc5632_adcr_func_controls = SOC_DAPM_ENUM("ADCR Mux", alc5632_adcr_func_enum); /* I2S out select */ -static const struct soc_enum alc5632_i2s_out_enum = - SOC_ENUM_SINGLE(ALC5632_I2S_OUT_CTL, 5, 2, alc5632_i2s_out_sel); +static SOC_ENUM_SINGLE_DECL(alc5632_i2s_out_enum, + ALC5632_I2S_OUT_CTL, 5, + alc5632_i2s_out_sel); static const struct snd_kcontrol_new alc5632_i2s_out_controls = SOC_DAPM_ENUM("I2SOut Mux", alc5632_i2s_out_enum); -- cgit v1.2.3 From 0cd257bf9b9b0cbb4fa1a5c988a232506997867c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 20 Feb 2014 09:04:06 +0900 Subject: ASoC: alc5623: Convert to direct regmap API usage Convert to directly use the regmap API, allowing us to eliminate the last user of the ASoC level I/O implementations (there are still open coded I/O implementations in drivers), avoiding duplicating code in regmap. We no longer cache the entire CODEC register map on probe since the more advanced cache infrastructure in regmap is able to fill the cache on demand. Signed-off-by: Mark Brown --- sound/soc/codecs/alc5623.c | 87 ++++++++++++++++++++++++++-------------------- 1 file changed, 49 insertions(+), 38 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index ba61c07ebbb2..ed506253a914 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -21,6 +21,7 @@ #include #include #include +#include #include #include #include @@ -38,26 +39,13 @@ MODULE_PARM_DESC(caps_charge, "ALC5623 cap charge time (msecs)"); /* codec private data */ struct alc5623_priv { - enum snd_soc_control_type control_type; + struct regmap *regmap; u8 id; unsigned int sysclk; - u16 reg_cache[ALC5623_VENDOR_ID2+2]; unsigned int add_ctrl; unsigned int jack_det_ctrl; }; -static void alc5623_fill_cache(struct snd_soc_codec *codec) -{ - int i, step = codec->driver->reg_cache_step; - u16 *cache = codec->reg_cache; - - /* not really efficient ... */ - codec->cache_bypass = 1; - for (i = 0 ; i < codec->driver->reg_cache_size ; i += step) - cache[i] = snd_soc_read(codec, i); - codec->cache_bypass = 0; -} - static inline int alc5623_reset(struct snd_soc_codec *codec) { return snd_soc_write(codec, ALC5623_RESET, 0); @@ -875,18 +863,28 @@ static struct snd_soc_dai_driver alc5623_dai = { static int alc5623_suspend(struct snd_soc_codec *codec) { + struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); + alc5623_set_bias_level(codec, SND_SOC_BIAS_OFF); + regcache_cache_only(alc5623->regmap, true); + return 0; } static int alc5623_resume(struct snd_soc_codec *codec) { - int i, step = codec->driver->reg_cache_step; - u16 *cache = codec->reg_cache; + struct alc5623_priv *alc5623 = snd_soc_codec_get_drvdata(codec); + int ret; /* Sync reg_cache with the hardware */ - for (i = 2 ; i < codec->driver->reg_cache_size ; i += step) - snd_soc_write(codec, i, cache[i]); + regcache_cache_only(alc5623->regmap, false); + ret = regcache_sync(alc5623->regmap); + if (ret != 0) { + dev_err(codec->dev, "Failed to sync register cache: %d\n", + ret); + regcache_cache_only(alc5623->regmap, true); + return ret; + } alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -906,14 +904,14 @@ static int alc5623_probe(struct snd_soc_codec *codec) struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, alc5623->control_type); + codec->control_data = alc5623->regmap; + ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); if (ret < 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; } alc5623_reset(codec); - alc5623_fill_cache(codec); /* power on device */ alc5623_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -986,9 +984,15 @@ static struct snd_soc_codec_driver soc_codec_device_alc5623 = { .suspend = alc5623_suspend, .resume = alc5623_resume, .set_bias_level = alc5623_set_bias_level, - .reg_cache_size = ALC5623_VENDOR_ID2+2, - .reg_word_size = sizeof(u16), - .reg_cache_step = 2, +}; + +static const struct regmap_config alc5623_regmap = { + .reg_bits = 8, + .val_bits = 16, + .reg_stride = 2, + + .max_register = ALC5623_VENDOR_ID2, + .cache_type = REGCACHE_RBTREE, }; /* @@ -1002,19 +1006,32 @@ static int alc5623_i2c_probe(struct i2c_client *client, { struct alc5623_platform_data *pdata; struct alc5623_priv *alc5623; - int ret, vid1, vid2; + unsigned int vid1, vid2; + int ret; - vid1 = i2c_smbus_read_word_data(client, ALC5623_VENDOR_ID1); - if (vid1 < 0) { - dev_err(&client->dev, "failed to read I2C\n"); - return -EIO; + alc5623 = devm_kzalloc(&client->dev, sizeof(struct alc5623_priv), + GFP_KERNEL); + if (alc5623 == NULL) + return -ENOMEM; + + alc5623->regmap = devm_regmap_init_i2c(client, &alc5623_regmap); + if (IS_ERR(alc5623->regmap)) { + ret = PTR_ERR(alc5623->regmap); + dev_err(&client->dev, "Failed to initialise I/O: %d\n", ret); + return ret; + } + + ret = regmap_read(alc5623->regmap, ALC5623_VENDOR_ID1, &vid1); + if (ret < 0) { + dev_err(&client->dev, "failed to read vendor ID1: %d\n", ret); + return ret; } vid1 = ((vid1 & 0xff) << 8) | (vid1 >> 8); - vid2 = i2c_smbus_read_byte_data(client, ALC5623_VENDOR_ID2); - if (vid2 < 0) { - dev_err(&client->dev, "failed to read I2C\n"); - return -EIO; + ret = regmap_read(alc5623->regmap, ALC5623_VENDOR_ID2, &vid2); + if (ret < 0) { + dev_err(&client->dev, "failed to read vendor ID2: %d\n", ret); + return ret; } if ((vid1 != 0x10ec) || (vid2 != id->driver_data)) { @@ -1027,11 +1044,6 @@ static int alc5623_i2c_probe(struct i2c_client *client, dev_dbg(&client->dev, "Found codec id : alc56%02x\n", vid2); - alc5623 = devm_kzalloc(&client->dev, sizeof(struct alc5623_priv), - GFP_KERNEL); - if (alc5623 == NULL) - return -ENOMEM; - pdata = client->dev.platform_data; if (pdata) { alc5623->add_ctrl = pdata->add_ctrl; @@ -1054,7 +1066,6 @@ static int alc5623_i2c_probe(struct i2c_client *client, } i2c_set_clientdata(client, alc5623); - alc5623->control_type = SND_SOC_I2C; ret = snd_soc_register_codec(&client->dev, &soc_codec_device_alc5623, &alc5623_dai, 1); -- cgit v1.2.3 From 6109ab2bfc22c903e4e8592bdcce268758f1dd5b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:44:57 +0100 Subject: ASoC: wm2200: Use SOC_ENUM_SINGLE_*_DECL() Just replace with the helper macros. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Charles Keepax Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l51.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 6e9ea8379a91..1703b9c988db 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -135,8 +135,7 @@ static const char *chan_mix[] = { "R L", }; -static const struct soc_enum cs42l51_chan_mix = - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(chan_mix), chan_mix); +static SOC_ENUM_SINGLE_EXT_DECL(cs42l51_chan_mix, chan_mix); static const struct snd_kcontrol_new cs42l51_snd_controls[] = { SOC_DOUBLE_R_SX_TLV("PCM Playback Volume", @@ -192,22 +191,22 @@ static int cs42l51_pdn_event(struct snd_soc_dapm_widget *w, static const char *cs42l51_dac_names[] = {"Direct PCM", "DSP PCM", "ADC"}; -static const struct soc_enum cs42l51_dac_mux_enum = - SOC_ENUM_SINGLE(CS42L51_DAC_CTL, 6, 3, cs42l51_dac_names); +static SOC_ENUM_SINGLE_DECL(cs42l51_dac_mux_enum, + CS42L51_DAC_CTL, 6, cs42l51_dac_names); static const struct snd_kcontrol_new cs42l51_dac_mux_controls = SOC_DAPM_ENUM("Route", cs42l51_dac_mux_enum); static const char *cs42l51_adcl_names[] = {"AIN1 Left", "AIN2 Left", "MIC Left", "MIC+preamp Left"}; -static const struct soc_enum cs42l51_adcl_mux_enum = - SOC_ENUM_SINGLE(CS42L51_ADC_INPUT, 4, 4, cs42l51_adcl_names); +static SOC_ENUM_SINGLE_DECL(cs42l51_adcl_mux_enum, + CS42L51_ADC_INPUT, 4, cs42l51_adcl_names); static const struct snd_kcontrol_new cs42l51_adcl_mux_controls = SOC_DAPM_ENUM("Route", cs42l51_adcl_mux_enum); static const char *cs42l51_adcr_names[] = {"AIN1 Right", "AIN2 Right", "MIC Right", "MIC+preamp Right"}; -static const struct soc_enum cs42l51_adcr_mux_enum = - SOC_ENUM_SINGLE(CS42L51_ADC_INPUT, 6, 4, cs42l51_adcr_names); +static SOC_ENUM_SINGLE_DECL(cs42l51_adcr_mux_enum, + CS42L51_ADC_INPUT, 6, cs42l51_adcr_names); static const struct snd_kcontrol_new cs42l51_adcr_mux_controls = SOC_DAPM_ENUM("Route", cs42l51_adcr_mux_enum); -- cgit v1.2.3 From 7e5091087246b620dfbfed2ce4f17c53898b66d4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:45:17 +0100 Subject: ASoC: da732x: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/da732x.c | 160 +++++++++++++++++++++------------------------- 1 file changed, 72 insertions(+), 88 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index f295b6569910..3219fa1f3cf5 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -269,81 +269,65 @@ static const char *da732x_hpf_voice[] = { "150Hz", "200Hz", "300Hz", "400Hz" }; -static const struct soc_enum da732x_dac1_hpf_mode_enum[] = { - SOC_ENUM_SINGLE(DA732X_REG_DAC1_HPF, DA732X_HPF_MODE_SHIFT, - DA732X_HPF_MODE_MAX, da732x_hpf_mode) -}; - -static const struct soc_enum da732x_dac2_hpf_mode_enum[] = { - SOC_ENUM_SINGLE(DA732X_REG_DAC2_HPF, DA732X_HPF_MODE_SHIFT, - DA732X_HPF_MODE_MAX, da732x_hpf_mode) -}; +static SOC_ENUM_SINGLE_DECL(da732x_dac1_hpf_mode_enum, + DA732X_REG_DAC1_HPF, DA732X_HPF_MODE_SHIFT, + da732x_hpf_mode); -static const struct soc_enum da732x_dac3_hpf_mode_enum[] = { - SOC_ENUM_SINGLE(DA732X_REG_DAC3_HPF, DA732X_HPF_MODE_SHIFT, - DA732X_HPF_MODE_MAX, da732x_hpf_mode) -}; +static SOC_ENUM_SINGLE_DECL(da732x_dac2_hpf_mode_enum, + DA732X_REG_DAC2_HPF, DA732X_HPF_MODE_SHIFT, + da732x_hpf_mode); -static const struct soc_enum da732x_adc1_hpf_mode_enum[] = { - SOC_ENUM_SINGLE(DA732X_REG_ADC1_HPF, DA732X_HPF_MODE_SHIFT, - DA732X_HPF_MODE_MAX, da732x_hpf_mode) -}; +static SOC_ENUM_SINGLE_DECL(da732x_dac3_hpf_mode_enum, + DA732X_REG_DAC3_HPF, DA732X_HPF_MODE_SHIFT, + da732x_hpf_mode); -static const struct soc_enum da732x_adc2_hpf_mode_enum[] = { - SOC_ENUM_SINGLE(DA732X_REG_ADC2_HPF, DA732X_HPF_MODE_SHIFT, - DA732X_HPF_MODE_MAX, da732x_hpf_mode) -}; +static SOC_ENUM_SINGLE_DECL(da732x_adc1_hpf_mode_enum, + DA732X_REG_ADC1_HPF, DA732X_HPF_MODE_SHIFT, + da732x_hpf_mode); -static const struct soc_enum da732x_dac1_hp_filter_enum[] = { - SOC_ENUM_SINGLE(DA732X_REG_DAC1_HPF, DA732X_HPF_MUSIC_SHIFT, - DA732X_HPF_MUSIC_MAX, da732x_hpf_music) -}; +static SOC_ENUM_SINGLE_DECL(da732x_adc2_hpf_mode_enum, + DA732X_REG_ADC2_HPF, DA732X_HPF_MODE_SHIFT, + da732x_hpf_mode); -static const struct soc_enum da732x_dac2_hp_filter_enum[] = { - SOC_ENUM_SINGLE(DA732X_REG_DAC2_HPF, DA732X_HPF_MUSIC_SHIFT, - DA732X_HPF_MUSIC_MAX, da732x_hpf_music) -}; +static SOC_ENUM_SINGLE_DECL(da732x_dac1_hp_filter_enum, + DA732X_REG_DAC1_HPF, DA732X_HPF_MUSIC_SHIFT, + da732x_hpf_music); -static const struct soc_enum da732x_dac3_hp_filter_enum[] = { - SOC_ENUM_SINGLE(DA732X_REG_DAC3_HPF, DA732X_HPF_MUSIC_SHIFT, - DA732X_HPF_MUSIC_MAX, da732x_hpf_music) -}; +static SOC_ENUM_SINGLE_DECL(da732x_dac2_hp_filter_enum, + DA732X_REG_DAC2_HPF, DA732X_HPF_MUSIC_SHIFT, + da732x_hpf_music); -static const struct soc_enum da732x_adc1_hp_filter_enum[] = { - SOC_ENUM_SINGLE(DA732X_REG_ADC1_HPF, DA732X_HPF_MUSIC_SHIFT, - DA732X_HPF_MUSIC_MAX, da732x_hpf_music) -}; +static SOC_ENUM_SINGLE_DECL(da732x_dac3_hp_filter_enum, + DA732X_REG_DAC3_HPF, DA732X_HPF_MUSIC_SHIFT, + da732x_hpf_music); -static const struct soc_enum da732x_adc2_hp_filter_enum[] = { - SOC_ENUM_SINGLE(DA732X_REG_ADC2_HPF, DA732X_HPF_MUSIC_SHIFT, - DA732X_HPF_MUSIC_MAX, da732x_hpf_music) -}; +static SOC_ENUM_SINGLE_DECL(da732x_adc1_hp_filter_enum, + DA732X_REG_ADC1_HPF, DA732X_HPF_MUSIC_SHIFT, + da732x_hpf_music); -static const struct soc_enum da732x_dac1_voice_filter_enum[] = { - SOC_ENUM_SINGLE(DA732X_REG_DAC1_HPF, DA732X_HPF_VOICE_SHIFT, - DA732X_HPF_VOICE_MAX, da732x_hpf_voice) -}; +static SOC_ENUM_SINGLE_DECL(da732x_adc2_hp_filter_enum, + DA732X_REG_ADC2_HPF, DA732X_HPF_MUSIC_SHIFT, + da732x_hpf_music); -static const struct soc_enum da732x_dac2_voice_filter_enum[] = { - SOC_ENUM_SINGLE(DA732X_REG_DAC2_HPF, DA732X_HPF_VOICE_SHIFT, - DA732X_HPF_VOICE_MAX, da732x_hpf_voice) -}; +static SOC_ENUM_SINGLE_DECL(da732x_dac1_voice_filter_enum, + DA732X_REG_DAC1_HPF, DA732X_HPF_VOICE_SHIFT, + da732x_hpf_voice); -static const struct soc_enum da732x_dac3_voice_filter_enum[] = { - SOC_ENUM_SINGLE(DA732X_REG_DAC3_HPF, DA732X_HPF_VOICE_SHIFT, - DA732X_HPF_VOICE_MAX, da732x_hpf_voice) -}; +static SOC_ENUM_SINGLE_DECL(da732x_dac2_voice_filter_enum, + DA732X_REG_DAC2_HPF, DA732X_HPF_VOICE_SHIFT, + da732x_hpf_voice); -static const struct soc_enum da732x_adc1_voice_filter_enum[] = { - SOC_ENUM_SINGLE(DA732X_REG_ADC1_HPF, DA732X_HPF_VOICE_SHIFT, - DA732X_HPF_VOICE_MAX, da732x_hpf_voice) -}; +static SOC_ENUM_SINGLE_DECL(da732x_dac3_voice_filter_enum, + DA732X_REG_DAC3_HPF, DA732X_HPF_VOICE_SHIFT, + da732x_hpf_voice); -static const struct soc_enum da732x_adc2_voice_filter_enum[] = { - SOC_ENUM_SINGLE(DA732X_REG_ADC2_HPF, DA732X_HPF_VOICE_SHIFT, - DA732X_HPF_VOICE_MAX, da732x_hpf_voice) -}; +static SOC_ENUM_SINGLE_DECL(da732x_adc1_voice_filter_enum, + DA732X_REG_ADC1_HPF, DA732X_HPF_VOICE_SHIFT, + da732x_hpf_voice); +static SOC_ENUM_SINGLE_DECL(da732x_adc2_voice_filter_enum, + DA732X_REG_ADC2_HPF, DA732X_HPF_VOICE_SHIFT, + da732x_hpf_voice); static int da732x_hpf_set(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) @@ -714,65 +698,65 @@ static const char *enable_text[] = { }; /* ADC1LMUX */ -static const struct soc_enum adc1l_enum = - SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC1L_MUX_SEL_SHIFT, - DA732X_ADCL_MUX_MAX, adcl_text); +static SOC_ENUM_SINGLE_DECL(adc1l_enum, + DA732X_REG_INP_MUX, DA732X_ADC1L_MUX_SEL_SHIFT, + adcl_text); static const struct snd_kcontrol_new adc1l_mux = SOC_DAPM_ENUM("ADC Route", adc1l_enum); /* ADC1RMUX */ -static const struct soc_enum adc1r_enum = - SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC1R_MUX_SEL_SHIFT, - DA732X_ADCR_MUX_MAX, adcr_text); +static SOC_ENUM_SINGLE_DECL(adc1r_enum, + DA732X_REG_INP_MUX, DA732X_ADC1R_MUX_SEL_SHIFT, + adcr_text); static const struct snd_kcontrol_new adc1r_mux = SOC_DAPM_ENUM("ADC Route", adc1r_enum); /* ADC2LMUX */ -static const struct soc_enum adc2l_enum = - SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC2L_MUX_SEL_SHIFT, - DA732X_ADCL_MUX_MAX, adcl_text); +static SOC_ENUM_SINGLE_DECL(adc2l_enum, + DA732X_REG_INP_MUX, DA732X_ADC2L_MUX_SEL_SHIFT, + adcl_text); static const struct snd_kcontrol_new adc2l_mux = SOC_DAPM_ENUM("ADC Route", adc2l_enum); /* ADC2RMUX */ -static const struct soc_enum adc2r_enum = - SOC_ENUM_SINGLE(DA732X_REG_INP_MUX, DA732X_ADC2R_MUX_SEL_SHIFT, - DA732X_ADCR_MUX_MAX, adcr_text); +static SOC_ENUM_SINGLE_DECL(adc2r_enum, + DA732X_REG_INP_MUX, DA732X_ADC2R_MUX_SEL_SHIFT, + adcr_text); static const struct snd_kcontrol_new adc2r_mux = SOC_DAPM_ENUM("ADC Route", adc2r_enum); -static const struct soc_enum da732x_hp_left_output = - SOC_ENUM_SINGLE(DA732X_REG_HPL, DA732X_HP_OUT_DAC_EN_SHIFT, - DA732X_DAC_EN_MAX, enable_text); +static SOC_ENUM_SINGLE_DECL(da732x_hp_left_output, + DA732X_REG_HPL, DA732X_HP_OUT_DAC_EN_SHIFT, + enable_text); static const struct snd_kcontrol_new hpl_mux = SOC_DAPM_ENUM("HPL Switch", da732x_hp_left_output); -static const struct soc_enum da732x_hp_right_output = - SOC_ENUM_SINGLE(DA732X_REG_HPR, DA732X_HP_OUT_DAC_EN_SHIFT, - DA732X_DAC_EN_MAX, enable_text); +static SOC_ENUM_SINGLE_DECL(da732x_hp_right_output, + DA732X_REG_HPR, DA732X_HP_OUT_DAC_EN_SHIFT, + enable_text); static const struct snd_kcontrol_new hpr_mux = SOC_DAPM_ENUM("HPR Switch", da732x_hp_right_output); -static const struct soc_enum da732x_speaker_output = - SOC_ENUM_SINGLE(DA732X_REG_LIN3, DA732X_LOUT_DAC_EN_SHIFT, - DA732X_DAC_EN_MAX, enable_text); +static SOC_ENUM_SINGLE_DECL(da732x_speaker_output, + DA732X_REG_LIN3, DA732X_LOUT_DAC_EN_SHIFT, + enable_text); static const struct snd_kcontrol_new spk_mux = SOC_DAPM_ENUM("SPK Switch", da732x_speaker_output); -static const struct soc_enum da732x_lout4_output = - SOC_ENUM_SINGLE(DA732X_REG_LIN4, DA732X_LOUT_DAC_EN_SHIFT, - DA732X_DAC_EN_MAX, enable_text); +static SOC_ENUM_SINGLE_DECL(da732x_lout4_output, + DA732X_REG_LIN4, DA732X_LOUT_DAC_EN_SHIFT, + enable_text); static const struct snd_kcontrol_new lout4_mux = SOC_DAPM_ENUM("LOUT4 Switch", da732x_lout4_output); -static const struct soc_enum da732x_lout2_output = - SOC_ENUM_SINGLE(DA732X_REG_LIN2, DA732X_LOUT_DAC_EN_SHIFT, - DA732X_DAC_EN_MAX, enable_text); +static SOC_ENUM_SINGLE_DECL(da732x_lout2_output, + DA732X_REG_LIN2, DA732X_LOUT_DAC_EN_SHIFT, + enable_text); static const struct snd_kcontrol_new lout2_mux = SOC_DAPM_ENUM("LOUT2 Switch", da732x_lout2_output); -- cgit v1.2.3 From c3518d1bd45ebfdfd88480e5781054fed378fb22 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:45:36 +0100 Subject: ASoC: ml26124: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ml26124.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c index 185fa3bc3052..577fb8776ce7 100644 --- a/sound/soc/codecs/ml26124.c +++ b/sound/soc/codecs/ml26124.c @@ -73,11 +73,11 @@ static const DECLARE_TLV_DB_SCALE(ngth, -7650, 150, 0); static const char * const ml26124_companding[] = {"16bit PCM", "u-law", "A-law"}; -static const struct soc_enum ml26124_adc_companding_enum - = SOC_ENUM_SINGLE(ML26124_SAI_TRANS_CTL, 6, 3, ml26124_companding); +static SOC_ENUM_SINGLE_DECL(ml26124_adc_companding_enum, + ML26124_SAI_TRANS_CTL, 6, ml26124_companding); -static const struct soc_enum ml26124_dac_companding_enum - = SOC_ENUM_SINGLE(ML26124_SAI_RCV_CTL, 6, 3, ml26124_companding); +static SOC_ENUM_SINGLE_DECL(ml26124_dac_companding_enum, + ML26124_SAI_RCV_CTL, 6, ml26124_companding); static const struct snd_kcontrol_new ml26124_snd_controls[] = { SOC_SINGLE_TLV("Capture Digital Volume", ML26124_RECORD_DIG_VOL, 0, @@ -136,8 +136,8 @@ static const struct snd_kcontrol_new ml26124_output_mixer_controls[] = { static const char * const ml26124_input_select[] = {"Analog MIC SingleEnded in", "Digital MIC in", "Analog MIC Differential in"}; -static const struct soc_enum ml26124_insel_enum = - SOC_ENUM_SINGLE(ML26124_MIC_IF_CTL, 0, 3, ml26124_input_select); +static SOC_ENUM_SINGLE_DECL(ml26124_insel_enum, + ML26124_MIC_IF_CTL, 0, ml26124_input_select); static const struct snd_kcontrol_new ml26124_input_mux_controls = SOC_DAPM_ENUM("Input Select", ml26124_insel_enum); -- cgit v1.2.3 From 806057cc75ef641cd9b012b0278c1f179090bab2 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:45:53 +0100 Subject: ASoC: tlv320aic23: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic23.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 5d430cc56f51..139f11f4dd8b 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -64,16 +64,16 @@ static const struct regmap_config tlv320aic23_regmap = { static const char *rec_src_text[] = { "Line", "Mic" }; static const char *deemph_text[] = {"None", "32Khz", "44.1Khz", "48Khz"}; -static const struct soc_enum rec_src_enum = - SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text); +static SOC_ENUM_SINGLE_DECL(rec_src_enum, + TLV320AIC23_ANLG, 2, rec_src_text); static const struct snd_kcontrol_new tlv320aic23_rec_src_mux_controls = SOC_DAPM_ENUM("Input Select", rec_src_enum); -static const struct soc_enum tlv320aic23_rec_src = - SOC_ENUM_SINGLE(TLV320AIC23_ANLG, 2, 2, rec_src_text); -static const struct soc_enum tlv320aic23_deemph = - SOC_ENUM_SINGLE(TLV320AIC23_DIGT, 1, 4, deemph_text); +static SOC_ENUM_SINGLE_DECL(tlv320aic23_rec_src, + TLV320AIC23_ANLG, 2, rec_src_text); +static SOC_ENUM_SINGLE_DECL(tlv320aic23_deemph, + TLV320AIC23_DIGT, 1, deemph_text); static const DECLARE_TLV_DB_SCALE(out_gain_tlv, -12100, 100, 0); static const DECLARE_TLV_DB_SCALE(input_gain_tlv, -1725, 75, 0); -- cgit v1.2.3 From 0224ba6a01983a52ed809bdc8647510d02656143 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:46:09 +0100 Subject: ASoC: wm5100: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Charles Keepax Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm5100.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 4e3e31aaf509..14beefd7e58c 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -506,21 +506,21 @@ static const char *wm5100_lhpf_mode_text[] = { "Low-pass", "High-pass" }; -static const struct soc_enum wm5100_lhpf1_mode = - SOC_ENUM_SINGLE(WM5100_HPLPF1_1, WM5100_LHPF1_MODE_SHIFT, 2, - wm5100_lhpf_mode_text); +static SOC_ENUM_SINGLE_DECL(wm5100_lhpf1_mode, + WM5100_HPLPF1_1, WM5100_LHPF1_MODE_SHIFT, + wm5100_lhpf_mode_text); -static const struct soc_enum wm5100_lhpf2_mode = - SOC_ENUM_SINGLE(WM5100_HPLPF2_1, WM5100_LHPF2_MODE_SHIFT, 2, - wm5100_lhpf_mode_text); +static SOC_ENUM_SINGLE_DECL(wm5100_lhpf2_mode, + WM5100_HPLPF2_1, WM5100_LHPF2_MODE_SHIFT, + wm5100_lhpf_mode_text); -static const struct soc_enum wm5100_lhpf3_mode = - SOC_ENUM_SINGLE(WM5100_HPLPF3_1, WM5100_LHPF3_MODE_SHIFT, 2, - wm5100_lhpf_mode_text); +static SOC_ENUM_SINGLE_DECL(wm5100_lhpf3_mode, + WM5100_HPLPF3_1, WM5100_LHPF3_MODE_SHIFT, + wm5100_lhpf_mode_text); -static const struct soc_enum wm5100_lhpf4_mode = - SOC_ENUM_SINGLE(WM5100_HPLPF4_1, WM5100_LHPF4_MODE_SHIFT, 2, - wm5100_lhpf_mode_text); +static SOC_ENUM_SINGLE_DECL(wm5100_lhpf4_mode, + WM5100_HPLPF4_1, WM5100_LHPF4_MODE_SHIFT, + wm5100_lhpf_mode_text); static const struct snd_kcontrol_new wm5100_snd_controls[] = { SOC_SINGLE("IN1 High Performance Switch", WM5100_IN1L_CONTROL, -- cgit v1.2.3 From 54db41c116c31b8829ba636895556a9fa45df987 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:46:22 +0100 Subject: ASoC: wm8955: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Charles Keepax Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8955.c | 11 +++++------ 1 file changed, 5 insertions(+), 6 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index 82c8ba975720..d4dcaecc8a5f 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -416,22 +416,21 @@ static const char *bass_mode_text[] = { "Linear", "Adaptive", }; -static const struct soc_enum bass_mode = - SOC_ENUM_SINGLE(WM8955_BASS_CONTROL, 7, 2, bass_mode_text); +static SOC_ENUM_SINGLE_DECL(bass_mode, WM8955_BASS_CONTROL, 7, bass_mode_text); static const char *bass_cutoff_text[] = { "Low", "High" }; -static const struct soc_enum bass_cutoff = - SOC_ENUM_SINGLE(WM8955_BASS_CONTROL, 6, 2, bass_cutoff_text); +static SOC_ENUM_SINGLE_DECL(bass_cutoff, WM8955_BASS_CONTROL, 6, + bass_cutoff_text); static const char *treble_cutoff_text[] = { "High", "Low" }; -static const struct soc_enum treble_cutoff = - SOC_ENUM_SINGLE(WM8955_TREBLE_CONTROL, 6, 2, treble_cutoff_text); +static SOC_ENUM_SINGLE_DECL(treble_cutoff, WM8955_TREBLE_CONTROL, 2, + treble_cutoff_text); static const DECLARE_TLV_DB_SCALE(digital_tlv, -12750, 50, 1); static const DECLARE_TLV_DB_SCALE(atten_tlv, -600, 600, 0); -- cgit v1.2.3 From b13a054aed6f6f3103df14f8bb0f9da4c4edb514 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:46:35 +0100 Subject: ASoC: wm8988: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Charles Keepax Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8988.c | 48 +++++++++++++++++++++++------------------------ 1 file changed, 24 insertions(+), 24 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index c6e4aba25b77..0277a76c6bef 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -126,46 +126,46 @@ struct wm8988_priv { */ static const char *bass_boost_txt[] = {"Linear Control", "Adaptive Boost"}; -static const struct soc_enum bass_boost = - SOC_ENUM_SINGLE(WM8988_BASS, 7, 2, bass_boost_txt); +static SOC_ENUM_SINGLE_DECL(bass_boost, + WM8988_BASS, 7, bass_boost_txt); static const char *bass_filter_txt[] = { "130Hz @ 48kHz", "200Hz @ 48kHz" }; -static const struct soc_enum bass_filter = - SOC_ENUM_SINGLE(WM8988_BASS, 6, 2, bass_filter_txt); +static SOC_ENUM_SINGLE_DECL(bass_filter, + WM8988_BASS, 6, bass_filter_txt); static const char *treble_txt[] = {"8kHz", "4kHz"}; -static const struct soc_enum treble = - SOC_ENUM_SINGLE(WM8988_TREBLE, 6, 2, treble_txt); +static SOC_ENUM_SINGLE_DECL(treble, + WM8988_TREBLE, 6, treble_txt); static const char *stereo_3d_lc_txt[] = {"200Hz", "500Hz"}; -static const struct soc_enum stereo_3d_lc = - SOC_ENUM_SINGLE(WM8988_3D, 5, 2, stereo_3d_lc_txt); +static SOC_ENUM_SINGLE_DECL(stereo_3d_lc, + WM8988_3D, 5, stereo_3d_lc_txt); static const char *stereo_3d_uc_txt[] = {"2.2kHz", "1.5kHz"}; -static const struct soc_enum stereo_3d_uc = - SOC_ENUM_SINGLE(WM8988_3D, 6, 2, stereo_3d_uc_txt); +static SOC_ENUM_SINGLE_DECL(stereo_3d_uc, + WM8988_3D, 6, stereo_3d_uc_txt); static const char *stereo_3d_func_txt[] = {"Capture", "Playback"}; -static const struct soc_enum stereo_3d_func = - SOC_ENUM_SINGLE(WM8988_3D, 7, 2, stereo_3d_func_txt); +static SOC_ENUM_SINGLE_DECL(stereo_3d_func, + WM8988_3D, 7, stereo_3d_func_txt); static const char *alc_func_txt[] = {"Off", "Right", "Left", "Stereo"}; -static const struct soc_enum alc_func = - SOC_ENUM_SINGLE(WM8988_ALC1, 7, 4, alc_func_txt); +static SOC_ENUM_SINGLE_DECL(alc_func, + WM8988_ALC1, 7, alc_func_txt); static const char *ng_type_txt[] = {"Constant PGA Gain", "Mute ADC Output"}; -static const struct soc_enum ng_type = - SOC_ENUM_SINGLE(WM8988_NGATE, 1, 2, ng_type_txt); +static SOC_ENUM_SINGLE_DECL(ng_type, + WM8988_NGATE, 1, ng_type_txt); static const char *deemph_txt[] = {"None", "32Khz", "44.1Khz", "48Khz"}; -static const struct soc_enum deemph = - SOC_ENUM_SINGLE(WM8988_ADCDAC, 1, 4, deemph_txt); +static SOC_ENUM_SINGLE_DECL(deemph, + WM8988_ADCDAC, 1, deemph_txt); static const char *adcpol_txt[] = {"Normal", "L Invert", "R Invert", "L + R Invert"}; -static const struct soc_enum adcpol = - SOC_ENUM_SINGLE(WM8988_ADCDAC, 5, 4, adcpol_txt); +static SOC_ENUM_SINGLE_DECL(adcpol, + WM8988_ADCDAC, 5, adcpol_txt); static const DECLARE_TLV_DB_SCALE(pga_tlv, -1725, 75, 0); static const DECLARE_TLV_DB_SCALE(adc_tlv, -9750, 50, 1); @@ -317,16 +317,16 @@ static const struct snd_kcontrol_new wm8988_right_pga_controls = /* Differential Mux */ static const char *wm8988_diff_sel[] = {"Line 1", "Line 2"}; -static const struct soc_enum diffmux = - SOC_ENUM_SINGLE(WM8988_ADCIN, 8, 2, wm8988_diff_sel); +static SOC_ENUM_SINGLE_DECL(diffmux, + WM8988_ADCIN, 8, wm8988_diff_sel); static const struct snd_kcontrol_new wm8988_diffmux_controls = SOC_DAPM_ENUM("Route", diffmux); /* Mono ADC Mux */ static const char *wm8988_mono_mux[] = {"Stereo", "Mono (Left)", "Mono (Right)", "Digital Mono"}; -static const struct soc_enum monomux = - SOC_ENUM_SINGLE(WM8988_ADCIN, 6, 4, wm8988_mono_mux); +static SOC_ENUM_SINGLE_DECL(monomux, + WM8988_ADCIN, 6, wm8988_mono_mux); static const struct snd_kcontrol_new wm8988_monomux_controls = SOC_DAPM_ENUM("Route", monomux); -- cgit v1.2.3 From aedbfd9649c5a9ee8e601f8f35e74ea617371ae5 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:46:48 +0100 Subject: ASoC: wm9081: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Charles Keepax Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm9081.c | 23 +++++++++-------------- 1 file changed, 9 insertions(+), 14 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 0982c1d38ec4..721cee71d5fc 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -268,8 +268,7 @@ static const char *drc_high_text[] = { "0", }; -static const struct soc_enum drc_high = - SOC_ENUM_SINGLE(WM9081_DRC_3, 3, 6, drc_high_text); +static SOC_ENUM_SINGLE_DECL(drc_high, WM9081_DRC_3, 3, drc_high_text); static const char *drc_low_text[] = { "1", @@ -279,8 +278,7 @@ static const char *drc_low_text[] = { "0", }; -static const struct soc_enum drc_low = - SOC_ENUM_SINGLE(WM9081_DRC_3, 0, 5, drc_low_text); +static SOC_ENUM_SINGLE_DECL(drc_low, WM9081_DRC_3, 0, drc_low_text); static const char *drc_atk_text[] = { "181us", @@ -297,8 +295,7 @@ static const char *drc_atk_text[] = { "185.6ms", }; -static const struct soc_enum drc_atk = - SOC_ENUM_SINGLE(WM9081_DRC_2, 12, 12, drc_atk_text); +static SOC_ENUM_SINGLE_DECL(drc_atk, WM9081_DRC_2, 12, drc_atk_text); static const char *drc_dcy_text[] = { "186ms", @@ -312,8 +309,7 @@ static const char *drc_dcy_text[] = { "47.56s", }; -static const struct soc_enum drc_dcy = - SOC_ENUM_SINGLE(WM9081_DRC_2, 8, 9, drc_dcy_text); +static SOC_ENUM_SINGLE_DECL(drc_dcy, WM9081_DRC_2, 8, drc_dcy_text); static const char *drc_qr_dcy_text[] = { "0.725ms", @@ -321,8 +317,7 @@ static const char *drc_qr_dcy_text[] = { "5.8ms", }; -static const struct soc_enum drc_qr_dcy = - SOC_ENUM_SINGLE(WM9081_DRC_2, 4, 3, drc_qr_dcy_text); +static SOC_ENUM_SINGLE_DECL(drc_qr_dcy, WM9081_DRC_2, 4, drc_qr_dcy_text); static const char *dac_deemph_text[] = { "None", @@ -331,16 +326,16 @@ static const char *dac_deemph_text[] = { "48kHz", }; -static const struct soc_enum dac_deemph = - SOC_ENUM_SINGLE(WM9081_DAC_DIGITAL_2, 1, 4, dac_deemph_text); +static SOC_ENUM_SINGLE_DECL(dac_deemph, WM9081_DAC_DIGITAL_2, 1, + dac_deemph_text); static const char *speaker_mode_text[] = { "Class D", "Class AB", }; -static const struct soc_enum speaker_mode = - SOC_ENUM_SINGLE(WM9081_ANALOGUE_SPEAKER_2, 6, 2, speaker_mode_text); +static SOC_ENUM_SINGLE_DECL(speaker_mode, WM9081_ANALOGUE_SPEAKER_2, 6, + speaker_mode_text); static int speaker_mode_get(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) -- cgit v1.2.3 From 09981dcb77eba05974eed4def32931fb12421fae Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:47:00 +0100 Subject: ASoC: wm9705: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Charles Keepax Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm9705.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c index 70ce6793c5bd..c0b7f45dfa37 100644 --- a/sound/soc/codecs/wm9705.c +++ b/sound/soc/codecs/wm9705.c @@ -67,12 +67,12 @@ static const char *wm9705_mic[] = {"Mic 1", "Mic 2"}; static const char *wm9705_rec_sel[] = {"Mic", "CD", "NC", "NC", "Line", "Stereo Mix", "Mono Mix", "Phone"}; -static const struct soc_enum wm9705_enum_mic = - SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, wm9705_mic); -static const struct soc_enum wm9705_enum_rec_l = - SOC_ENUM_SINGLE(AC97_REC_SEL, 8, 8, wm9705_rec_sel); -static const struct soc_enum wm9705_enum_rec_r = - SOC_ENUM_SINGLE(AC97_REC_SEL, 0, 8, wm9705_rec_sel); +static SOC_ENUM_SINGLE_DECL(wm9705_enum_mic, + AC97_GENERAL_PURPOSE, 8, wm9705_mic); +static SOC_ENUM_SINGLE_DECL(wm9705_enum_rec_l, + AC97_REC_SEL, 8, wm9705_rec_sel); +static SOC_ENUM_SINGLE_DECL(wm9705_enum_rec_r, + AC97_REC_SEL, 0, wm9705_rec_sel); /* Headphone Mixer */ static const struct snd_kcontrol_new wm9705_hp_mixer_controls[] = { -- cgit v1.2.3 From 2e86434f9eeb4a6d9379895b889ac85fa51b0ac9 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:54:31 +0100 Subject: ASoC: ad1836: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/ad1836.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 77f459868579..685998dd086e 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -40,8 +40,8 @@ struct ad1836_priv { */ static const char *ad1836_deemp[] = {"None", "44.1kHz", "32kHz", "48kHz"}; -static const struct soc_enum ad1836_deemp_enum = - SOC_ENUM_SINGLE(AD1836_DAC_CTRL1, 8, 4, ad1836_deemp); +static SOC_ENUM_SINGLE_DECL(ad1836_deemp_enum, + AD1836_DAC_CTRL1, 8, ad1836_deemp); #define AD1836_DAC_VOLUME(x) \ SOC_DOUBLE_R("DAC" #x " Playback Volume", AD1836_DAC_L_VOL(x), \ -- cgit v1.2.3 From 52a5b545bc09ebc7b1e4a55d765ccb76286ca48d Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 10:57:55 +0100 Subject: ASoC: cs42l73: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l73.c | 36 ++++++++++++++++++------------------ 1 file changed, 18 insertions(+), 18 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 69c8e2de7d0e..06f429184821 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -278,13 +278,13 @@ static const DECLARE_TLV_DB_SCALE(attn_tlv, -6300, 100, 1); static const char * const cs42l73_pgaa_text[] = { "Line A", "Mic 1" }; static const char * const cs42l73_pgab_text[] = { "Line B", "Mic 2" }; -static const struct soc_enum pgaa_enum = - SOC_ENUM_SINGLE(CS42L73_ADCIPC, 3, - ARRAY_SIZE(cs42l73_pgaa_text), cs42l73_pgaa_text); +static SOC_ENUM_SINGLE_DECL(pgaa_enum, + CS42L73_ADCIPC, 3, + cs42l73_pgaa_text); -static const struct soc_enum pgab_enum = - SOC_ENUM_SINGLE(CS42L73_ADCIPC, 7, - ARRAY_SIZE(cs42l73_pgab_text), cs42l73_pgab_text); +static SOC_ENUM_SINGLE_DECL(pgab_enum, + CS42L73_ADCIPC, 7, + cs42l73_pgab_text); static const struct snd_kcontrol_new pgaa_mux = SOC_DAPM_ENUM("Left Analog Input Capture Mux", pgaa_enum); @@ -309,9 +309,9 @@ static const struct snd_kcontrol_new input_right_mixer[] = { static const char * const cs42l73_ng_delay_text[] = { "50ms", "100ms", "150ms", "200ms" }; -static const struct soc_enum ng_delay_enum = - SOC_ENUM_SINGLE(CS42L73_NGCAB, 0, - ARRAY_SIZE(cs42l73_ng_delay_text), cs42l73_ng_delay_text); +static SOC_ENUM_SINGLE_DECL(ng_delay_enum, + CS42L73_NGCAB, 0, + cs42l73_ng_delay_text); static const char * const cs42l73_mono_mix_texts[] = { "Left", "Right", "Mono Mix"}; @@ -357,19 +357,19 @@ static const struct snd_kcontrol_new esl_xsp_mixer = static const char * const cs42l73_ip_swap_text[] = { "Stereo", "Mono A", "Mono B", "Swap A-B"}; -static const struct soc_enum ip_swap_enum = - SOC_ENUM_SINGLE(CS42L73_MIOPC, 6, - ARRAY_SIZE(cs42l73_ip_swap_text), cs42l73_ip_swap_text); +static SOC_ENUM_SINGLE_DECL(ip_swap_enum, + CS42L73_MIOPC, 6, + cs42l73_ip_swap_text); static const char * const cs42l73_spo_mixer_text[] = {"Mono", "Stereo"}; -static const struct soc_enum vsp_output_mux_enum = - SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 5, - ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text); +static SOC_ENUM_SINGLE_DECL(vsp_output_mux_enum, + CS42L73_MIXERCTL, 5, + cs42l73_spo_mixer_text); -static const struct soc_enum xsp_output_mux_enum = - SOC_ENUM_SINGLE(CS42L73_MIXERCTL, 4, - ARRAY_SIZE(cs42l73_spo_mixer_text), cs42l73_spo_mixer_text); +static SOC_ENUM_SINGLE_DECL(xsp_output_mux_enum, + CS42L73_MIXERCTL, 4, + cs42l73_spo_mixer_text); static const struct snd_kcontrol_new vsp_output_mux = SOC_DAPM_ENUM("Route", vsp_output_mux_enum); -- cgit v1.2.3 From 26d04ca8c46e3af510f0bbbe4d0b1aca8e18b393 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 11:00:13 +0100 Subject: ASoC: lm49453: Use SOC_ENUM_SINGLE_DECL() Just replace with the helper macro. No functional change at all. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/lm49453.c | 16 +++++++--------- 1 file changed, 7 insertions(+), 9 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index e19490cfb3a8..d6f391a0f37e 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -213,15 +213,13 @@ static const char *lm49453_adcl_mux_text[] = { "MIC1", "Aux_L" }; static const char *lm49453_adcr_mux_text[] = { "MIC2", "Aux_R" }; -static const struct soc_enum lm49453_adcl_enum = - SOC_ENUM_SINGLE(LM49453_P0_ANALOG_MIXER_ADC_REG, 0, - ARRAY_SIZE(lm49453_adcl_mux_text), - lm49453_adcl_mux_text); - -static const struct soc_enum lm49453_adcr_enum = - SOC_ENUM_SINGLE(LM49453_P0_ANALOG_MIXER_ADC_REG, 1, - ARRAY_SIZE(lm49453_adcr_mux_text), - lm49453_adcr_mux_text); +static SOC_ENUM_SINGLE_DECL(lm49453_adcl_enum, + LM49453_P0_ANALOG_MIXER_ADC_REG, 0, + lm49453_adcl_mux_text); + +static SOC_ENUM_SINGLE_DECL(lm49453_adcr_enum, + LM49453_P0_ANALOG_MIXER_ADC_REG, 1, + lm49453_adcr_mux_text); static const struct snd_kcontrol_new lm49453_adcl_mux_control = SOC_DAPM_ENUM("ADC Left Mux", lm49453_adcl_enum); -- cgit v1.2.3 From d77c290af76637e87dc07df28536231ee5042c98 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 11:07:32 +0100 Subject: ASoC: tlv320dac33: Use SOC_ENUM_SINGLE_*_DECL() Just replace with the helper macros. No functional change at all. Acked-by: Liam Girdwood Acked-by: Lars-Peter Clausen Signed-off-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320dac33.c | 32 +++++++++++++------------------- 1 file changed, 13 insertions(+), 19 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 4f358393d6d6..9ce849623a0a 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -478,9 +478,7 @@ static const char *dac33_fifo_mode_texts[] = { "Bypass", "Mode 1", "Mode 7" }; -static const struct soc_enum dac33_fifo_mode_enum = - SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(dac33_fifo_mode_texts), - dac33_fifo_mode_texts); +static SOC_ENUM_SINGLE_EXT_DECL(dac33_fifo_mode_enum, dac33_fifo_mode_texts); /* L/R Line Output Gain */ static const char *lr_lineout_gain_texts[] = { @@ -488,15 +486,13 @@ static const char *lr_lineout_gain_texts[] = { "Line 0dB DAC 12dB", "Line 6dB DAC 18dB", }; -static const struct soc_enum l_lineout_gain_enum = - SOC_ENUM_SINGLE(DAC33_LDAC_PWR_CTRL, 0, - ARRAY_SIZE(lr_lineout_gain_texts), - lr_lineout_gain_texts); +static SOC_ENUM_SINGLE_DECL(l_lineout_gain_enum, + DAC33_LDAC_PWR_CTRL, 0, + lr_lineout_gain_texts); -static const struct soc_enum r_lineout_gain_enum = - SOC_ENUM_SINGLE(DAC33_RDAC_PWR_CTRL, 0, - ARRAY_SIZE(lr_lineout_gain_texts), - lr_lineout_gain_texts); +static SOC_ENUM_SINGLE_DECL(r_lineout_gain_enum, + DAC33_RDAC_PWR_CTRL, 0, + lr_lineout_gain_texts); /* * DACL/R digital volume control: @@ -534,18 +530,16 @@ static const struct snd_kcontrol_new dac33_dapm_abypassr_control = /* LOP L/R invert selection */ static const char *dac33_lr_lom_texts[] = {"DAC", "LOP"}; -static const struct soc_enum dac33_left_lom_enum = - SOC_ENUM_SINGLE(DAC33_OUT_AMP_CTRL, 3, - ARRAY_SIZE(dac33_lr_lom_texts), - dac33_lr_lom_texts); +static SOC_ENUM_SINGLE_DECL(dac33_left_lom_enum, + DAC33_OUT_AMP_CTRL, 3, + dac33_lr_lom_texts); static const struct snd_kcontrol_new dac33_dapm_left_lom_control = SOC_DAPM_ENUM("Route", dac33_left_lom_enum); -static const struct soc_enum dac33_right_lom_enum = - SOC_ENUM_SINGLE(DAC33_OUT_AMP_CTRL, 2, - ARRAY_SIZE(dac33_lr_lom_texts), - dac33_lr_lom_texts); +static SOC_ENUM_SINGLE_DECL(dac33_right_lom_enum, + DAC33_OUT_AMP_CTRL, 2, + dac33_lr_lom_texts); static const struct snd_kcontrol_new dac33_dapm_right_lom_control = SOC_DAPM_ENUM("Route", dac33_right_lom_enum); -- cgit v1.2.3 From daf152a21d361967bd9d2d7a976c125de842d589 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 11:49:02 +0100 Subject: ASoC: wm5102: Use ARRAY_SIZE() for SOC_VALUE_ENUM_SINGLE() ... to make clear the meaning of the argument. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Charles Keepax Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index ebffe81daa1d..293dffcb1d2f 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -622,13 +622,16 @@ static const unsigned int wm5102_osr_val[] = { static const struct soc_enum wm5102_hpout_osr[] = { SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_1L, - ARIZONA_OUT1_OSR_SHIFT, 0x7, 3, + ARIZONA_OUT1_OSR_SHIFT, 0x7, + ARRAY_SIZE(wm5102_osr_text), wm5102_osr_text, wm5102_osr_val), SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_2L, - ARIZONA_OUT2_OSR_SHIFT, 0x7, 3, + ARIZONA_OUT2_OSR_SHIFT, 0x7, + ARRAY_SIZE(wm5102_osr_text), wm5102_osr_text, wm5102_osr_val), SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_3L, - ARIZONA_OUT3_OSR_SHIFT, 0x7, 3, + ARIZONA_OUT3_OSR_SHIFT, 0x7, + ARRAY_SIZE(wm5102_osr_text), wm5102_osr_text, wm5102_osr_val), }; -- cgit v1.2.3 From 347e5512642da44d15bf8b48ee0fe196b37a78f1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 18 Feb 2014 11:49:56 +0100 Subject: ASoC: wm8997: Use ARRAY_SIZE() for SOC_VALUE_ENUM_SINGLE() ... to make clear the meaning of the argument. Signed-off-by: Takashi Iwai Acked-by: Liam Girdwood Acked-by: Charles Keepax Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8997.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 6107108228b6..4e6442ce9a2a 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -123,10 +123,12 @@ static const unsigned int wm8997_osr_val[] = { static const struct soc_enum wm8997_hpout_osr[] = { SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_1L, - ARIZONA_OUT1_OSR_SHIFT, 0x7, 3, + ARIZONA_OUT1_OSR_SHIFT, 0x7, + ARRAY_SIZE(wm8997_osr_text), wm8997_osr_text, wm8997_osr_val), SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_3L, - ARIZONA_OUT3_OSR_SHIFT, 0x7, 3, + ARIZONA_OUT3_OSR_SHIFT, 0x7, + ARRAY_SIZE(wm8997_osr_text), wm8997_osr_text, wm8997_osr_val), }; -- cgit v1.2.3 From 98b664e2ceddd40120e8cd2aa56f7eb9a51870cf Mon Sep 17 00:00:00 2001 From: Markus Pargmann Date: Thu, 20 Feb 2014 18:22:58 +0100 Subject: ASoC: tlv320aic32x4: Support for master clock Add support for a master clock passed through DT. The master clock of the codec is only active when the codec is in use. Signed-off-by: Markus Pargmann Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/tlv320aic32x4.txt | 4 ++++ sound/soc/codecs/tlv320aic32x4.c | 21 +++++++++++++++++++++ 2 files changed, 25 insertions(+) (limited to 'sound/soc/codecs') diff --git a/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt index db0551088cc4..352be7b1f7e2 100644 --- a/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt +++ b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt @@ -8,6 +8,8 @@ Required properties: Optional properties: - reset-gpios: Reset-GPIO phandle with args as described in gpio/gpio.txt + - clocks/clock-names: Clock named 'mclk' for the master clock of the codec. + See clock/clock-bindings.txt for information about the detailed format. Example: @@ -15,4 +17,6 @@ Example: codec: tlv320aic32x4@18 { compatible = "ti,tlv320aic32x4"; reg = <0x18>; + clocks = <&clks 201>; + clock-names = "mclk"; }; diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 1dd50e48934c..643fa53beaab 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -33,6 +33,7 @@ #include #include #include +#include #include #include @@ -67,6 +68,7 @@ struct aic32x4_priv { u32 micpga_routing; bool swapdacs; int rstn_gpio; + struct clk *mclk; }; /* 0dB min, 0.5dB steps */ @@ -487,8 +489,18 @@ static int aic32x4_mute(struct snd_soc_dai *dai, int mute) static int aic32x4_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { + struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); + int ret; + switch (level) { case SND_SOC_BIAS_ON: + /* Switch on master clock */ + ret = clk_prepare_enable(aic32x4->mclk); + if (ret) { + dev_err(codec->dev, "Failed to enable master clock\n"); + return ret; + } + /* Switch on PLL */ snd_soc_update_bits(codec, AIC32X4_PLLPR, AIC32X4_PLLEN, AIC32X4_PLLEN); @@ -539,6 +551,9 @@ static int aic32x4_set_bias_level(struct snd_soc_codec *codec, /* Switch off BCLK_N Divider */ snd_soc_update_bits(codec, AIC32X4_BCLKN, AIC32X4_BCLKEN, 0); + + /* Switch off master clock */ + clk_disable_unprepare(aic32x4->mclk); break; case SND_SOC_BIAS_OFF: break; @@ -717,6 +732,12 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c, aic32x4->rstn_gpio = -1; } + aic32x4->mclk = devm_clk_get(&i2c->dev, "mclk"); + if (IS_ERR(aic32x4->mclk)) { + dev_err(&i2c->dev, "Failed getting the mclk. The current implementation does not support the usage of this codec without mclk\n"); + return PTR_ERR(aic32x4->mclk); + } + if (gpio_is_valid(aic32x4->rstn_gpio)) { ret = devm_gpio_request_one(&i2c->dev, aic32x4->rstn_gpio, GPIOF_OUT_INIT_LOW, "tlv320aic32x4 rstn"); -- cgit v1.2.3 From 239b669b2dedc46d5e6b07d87c3d1dedf8d9477c Mon Sep 17 00:00:00 2001 From: Markus Pargmann Date: Thu, 20 Feb 2014 18:22:59 +0100 Subject: ASoC: tlv320aic32x4: Support for regulators Support regulators to power up the codec. This patch also enables the AVDD LDO if no AV regulator was found. Signed-off-by: Markus Pargmann Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/tlv320aic32x4.txt | 8 ++ sound/soc/codecs/tlv320aic32x4.c | 126 ++++++++++++++++++++- 2 files changed, 133 insertions(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt index 352be7b1f7e2..5e2741af27be 100644 --- a/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt +++ b/Documentation/devicetree/bindings/sound/tlv320aic32x4.txt @@ -5,6 +5,14 @@ The tlv320aic32x4 serial control bus communicates through I2C protocols Required properties: - compatible: Should be "ti,tlv320aic32x4" - reg: I2C slave address + - supply-*: Required supply regulators are: + "iov" - digital IO power supply + "ldoin" - LDO power supply + "dv" - Digital core power supply + "av" - Analog core power supply + If you supply ldoin, dv and av are optional. Otherwise they are required + See regulator/regulator.txt for more information about the detailed binding + format. Optional properties: - reset-gpios: Reset-GPIO phandle with args as described in gpio/gpio.txt diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index 643fa53beaab..d69c61ffcda8 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -34,6 +34,7 @@ #include #include #include +#include #include #include @@ -69,6 +70,11 @@ struct aic32x4_priv { bool swapdacs; int rstn_gpio; struct clk *mclk; + + struct regulator *supply_ldo; + struct regulator *supply_iov; + struct regulator *supply_dv; + struct regulator *supply_av; }; /* 0dB min, 0.5dB steps */ @@ -695,6 +701,106 @@ static int aic32x4_parse_dt(struct aic32x4_priv *aic32x4, return 0; } +static void aic32x4_disable_regulators(struct aic32x4_priv *aic32x4) +{ + regulator_disable(aic32x4->supply_iov); + + if (!IS_ERR(aic32x4->supply_ldo)) + regulator_disable(aic32x4->supply_ldo); + + if (!IS_ERR(aic32x4->supply_dv)) + regulator_disable(aic32x4->supply_dv); + + if (!IS_ERR(aic32x4->supply_av)) + regulator_disable(aic32x4->supply_av); +} + +static int aic32x4_setup_regulators(struct device *dev, + struct aic32x4_priv *aic32x4) +{ + int ret = 0; + + aic32x4->supply_ldo = devm_regulator_get_optional(dev, "ldoin"); + aic32x4->supply_iov = devm_regulator_get(dev, "iov"); + aic32x4->supply_dv = devm_regulator_get_optional(dev, "dv"); + aic32x4->supply_av = devm_regulator_get_optional(dev, "av"); + + /* Check if the regulator requirements are fulfilled */ + + if (IS_ERR(aic32x4->supply_iov)) { + dev_err(dev, "Missing supply 'iov'\n"); + return PTR_ERR(aic32x4->supply_iov); + } + + if (IS_ERR(aic32x4->supply_ldo)) { + if (PTR_ERR(aic32x4->supply_ldo) == -EPROBE_DEFER) + return -EPROBE_DEFER; + + if (IS_ERR(aic32x4->supply_dv)) { + dev_err(dev, "Missing supply 'dv' or 'ldoin'\n"); + return PTR_ERR(aic32x4->supply_dv); + } + if (IS_ERR(aic32x4->supply_av)) { + dev_err(dev, "Missing supply 'av' or 'ldoin'\n"); + return PTR_ERR(aic32x4->supply_av); + } + } else { + if (IS_ERR(aic32x4->supply_dv) && + PTR_ERR(aic32x4->supply_dv) == -EPROBE_DEFER) + return -EPROBE_DEFER; + if (IS_ERR(aic32x4->supply_av) && + PTR_ERR(aic32x4->supply_av) == -EPROBE_DEFER) + return -EPROBE_DEFER; + } + + ret = regulator_enable(aic32x4->supply_iov); + if (ret) { + dev_err(dev, "Failed to enable regulator iov\n"); + return ret; + } + + if (!IS_ERR(aic32x4->supply_ldo)) { + ret = regulator_enable(aic32x4->supply_ldo); + if (ret) { + dev_err(dev, "Failed to enable regulator ldo\n"); + goto error_ldo; + } + } + + if (!IS_ERR(aic32x4->supply_dv)) { + ret = regulator_enable(aic32x4->supply_dv); + if (ret) { + dev_err(dev, "Failed to enable regulator dv\n"); + goto error_dv; + } + } + + if (!IS_ERR(aic32x4->supply_av)) { + ret = regulator_enable(aic32x4->supply_av); + if (ret) { + dev_err(dev, "Failed to enable regulator av\n"); + goto error_av; + } + } + + if (!IS_ERR(aic32x4->supply_ldo) && IS_ERR(aic32x4->supply_av)) + aic32x4->power_cfg |= AIC32X4_PWR_AIC32X4_LDO_ENABLE; + + return 0; + +error_av: + if (!IS_ERR(aic32x4->supply_dv)) + regulator_disable(aic32x4->supply_dv); + +error_dv: + if (!IS_ERR(aic32x4->supply_ldo)) + regulator_disable(aic32x4->supply_ldo); + +error_ldo: + regulator_disable(aic32x4->supply_iov); + return ret; +} + static int aic32x4_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { @@ -745,13 +851,31 @@ static int aic32x4_i2c_probe(struct i2c_client *i2c, return ret; } + ret = aic32x4_setup_regulators(&i2c->dev, aic32x4); + if (ret) { + dev_err(&i2c->dev, "Failed to setup regulators\n"); + return ret; + } + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_aic32x4, &aic32x4_dai, 1); - return ret; + if (ret) { + dev_err(&i2c->dev, "Failed to register codec\n"); + aic32x4_disable_regulators(aic32x4); + return ret; + } + + i2c_set_clientdata(i2c, aic32x4); + + return 0; } static int aic32x4_i2c_remove(struct i2c_client *client) { + struct aic32x4_priv *aic32x4 = i2c_get_clientdata(client); + + aic32x4_disable_regulators(aic32x4); + snd_soc_unregister_codec(&client->dev); return 0; } -- cgit v1.2.3 From 3154cc7404506700ff270b6f123ec9c734f002fd Mon Sep 17 00:00:00 2001 From: Markus Pargmann Date: Thu, 20 Feb 2014 18:23:00 +0100 Subject: ASoC: tlv320aic32x4: Rearrange clock tree shutdown Rearrange clock tree shutdown to disable them in the reversed order of startup. First disable all dividers, then PLL followed by master clock. Signed-off-by: Markus Pargmann Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic32x4.c | 32 ++++++++++++++++---------------- 1 file changed, 16 insertions(+), 16 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index d69c61ffcda8..c6bd7e75352d 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -534,29 +534,29 @@ static int aic32x4_set_bias_level(struct snd_soc_codec *codec, case SND_SOC_BIAS_PREPARE: break; case SND_SOC_BIAS_STANDBY: - /* Switch off PLL */ - snd_soc_update_bits(codec, AIC32X4_PLLPR, - AIC32X4_PLLEN, 0); - - /* Switch off NDAC Divider */ - snd_soc_update_bits(codec, AIC32X4_NDAC, - AIC32X4_NDACEN, 0); + /* Switch off BCLK_N Divider */ + snd_soc_update_bits(codec, AIC32X4_BCLKN, + AIC32X4_BCLKEN, 0); - /* Switch off MDAC Divider */ - snd_soc_update_bits(codec, AIC32X4_MDAC, - AIC32X4_MDACEN, 0); + /* Switch off MADC Divider */ + snd_soc_update_bits(codec, AIC32X4_MADC, + AIC32X4_MADCEN, 0); /* Switch off NADC Divider */ snd_soc_update_bits(codec, AIC32X4_NADC, AIC32X4_NADCEN, 0); - /* Switch off MADC Divider */ - snd_soc_update_bits(codec, AIC32X4_MADC, - AIC32X4_MADCEN, 0); + /* Switch off MDAC Divider */ + snd_soc_update_bits(codec, AIC32X4_MDAC, + AIC32X4_MDACEN, 0); - /* Switch off BCLK_N Divider */ - snd_soc_update_bits(codec, AIC32X4_BCLKN, - AIC32X4_BCLKEN, 0); + /* Switch off NDAC Divider */ + snd_soc_update_bits(codec, AIC32X4_NDAC, + AIC32X4_NDACEN, 0); + + /* Switch off PLL */ + snd_soc_update_bits(codec, AIC32X4_PLLPR, + AIC32X4_PLLEN, 0); /* Switch off master clock */ clk_disable_unprepare(aic32x4->mclk); -- cgit v1.2.3 From eeecf1a3f9f4791371bf35035ab9b95ab6aab5e7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 24 Feb 2014 11:38:11 +0900 Subject: ASoC: da732x: Remove leftover cache size setting The da732x driver no longer uses the ASoC level register cache but the cache size setting had been left in the driver by mistake. Remove it. Signed-off-by: Mark Brown --- sound/soc/codecs/da732x.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index 3219fa1f3cf5..05f07fb49860 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1538,7 +1538,6 @@ static struct snd_soc_codec_driver soc_codec_dev_da732x = { .dapm_routes = da732x_dapm_routes, .num_dapm_routes = ARRAY_SIZE(da732x_dapm_routes), .set_pll = da732x_set_dai_pll, - .reg_cache_size = ARRAY_SIZE(da732x_reg_cache), }; static int da732x_i2c_probe(struct i2c_client *i2c, -- cgit v1.2.3 From 30812cca63600a3e4b44b93ed2ec7e00ce572ee0 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 22 Feb 2014 18:34:45 +0100 Subject: ASoC: da732x: Use da732x->regmap instead of codec->control_data With the ongoing component-ization of the ASoC framework and the continuing migration to using regmap for IO the control_data field of the snd_soc_codec struct will eventually be removed. Prepare the da732x driver for this by using da732x->regmap instead of accessing the CODEC's control_data field. Signed-off-by: Lars-Peter Clausen Acked-by: Adam Thomson Signed-off-by: Mark Brown --- sound/soc/codecs/da732x.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index 05f07fb49860..cf9b15472c85 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1471,8 +1471,8 @@ static int da732x_set_bias_level(struct snd_soc_codec *codec, da732x_hp_dc_offset_cancellation(codec); - regcache_cache_only(codec->control_data, false); - regcache_sync(codec->control_data); + regcache_cache_only(da732x->regmap, false); + regcache_sync(da732x->regmap); } else { snd_soc_update_bits(codec, DA732X_REG_BIAS_EN, DA732X_BIAS_BOOST_MASK, @@ -1483,7 +1483,7 @@ static int da732x_set_bias_level(struct snd_soc_codec *codec, } break; case SND_SOC_BIAS_OFF: - regcache_cache_only(codec->control_data, true); + regcache_cache_only(da732x->regmap, true); da732x_set_charge_pump(codec, DA732X_DISABLE_CP); snd_soc_update_bits(codec, DA732X_REG_BIAS_EN, DA732X_BIAS_EN, DA732X_BIAS_DIS); -- cgit v1.2.3 From 180c275eb8a65c919d09af49179b99e4c01a3d1e Mon Sep 17 00:00:00 2001 From: Christian Engelmayer Date: Sat, 22 Feb 2014 16:00:16 +0100 Subject: ASoC: wm8993: Remove unused pointer in wm8993_remove() Commit 88b5bdfd (ASoC: wm8993: drop regulator_bulk_free of devm_ allocated data) eliminated the last user of driver data pointer 'wm8993' in function wm8993_remove() - Thus remove it. Detected by Coverity: CID 1186208. Signed-off-by: Christian Engelmayer Signed-off-by: Mark Brown --- sound/soc/codecs/wm8993.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 1c12f2c9418a..7b0630a076fa 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1559,8 +1559,6 @@ static int wm8993_probe(struct snd_soc_codec *codec) static int wm8993_remove(struct snd_soc_codec *codec) { - struct wm8993_priv *wm8993 = snd_soc_codec_get_drvdata(codec); - wm8993_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } -- cgit v1.2.3 From b7c1b73097b48a7922d1320d5a8c8315ff4fae71 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 22 Feb 2014 18:32:04 +0100 Subject: ASoC: wm8996: Replace codec->control_data with wm8996->regmap With the ongoing component-ization of the ASoC framework and the continuing migration to using regmap for IO the control_data field of the snd_soc_codec struct will eventually be removed. Prepare the wm8996 driver for this by using wm8996->regmap instead of accessing the CODEC's control_data field. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8996.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 92bb02185c46..36a52b9e97e7 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -1608,8 +1608,8 @@ static int wm8996_set_bias_level(struct snd_soc_codec *codec, msleep(5); } - regcache_cache_only(codec->control_data, false); - regcache_sync(codec->control_data); + regcache_cache_only(wm8996->regmap, false); + regcache_sync(wm8996->regmap); } /* Bypass the MICBIASes for lowest power */ @@ -1620,10 +1620,10 @@ static int wm8996_set_bias_level(struct snd_soc_codec *codec, break; case SND_SOC_BIAS_OFF: - regcache_cache_only(codec->control_data, true); + regcache_cache_only(wm8996->regmap, true); if (wm8996->pdata.ldo_ena >= 0) { gpio_set_value_cansleep(wm8996->pdata.ldo_ena, 0); - regcache_cache_only(codec->control_data, true); + regcache_cache_only(wm8996->regmap, true); } regulator_bulk_disable(ARRAY_SIZE(wm8996->supplies), wm8996->supplies); -- cgit v1.2.3 From d7f31d3c898e3e621a34d5d64966f7b830df66f1 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 22 Feb 2014 18:32:05 +0100 Subject: ASoC: wm8962: Replace codec->control_data with wm8962->regmap With the ongoing component-ization of the ASoC framework and the continuing migration to using regmap for IO the control_data field of the snd_soc_codec struct will eventually be removed. Prepare the wm8962 driver for this by using wm8962->regmap instead of accessing the CODEC's control_data field. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/wm8962.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 3924ee243745..6ff1ff83b4dd 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -1479,7 +1479,9 @@ static const DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static int wm8962_dsp2_write_config(struct snd_soc_codec *codec) { - return regcache_sync_region(codec->control_data, + struct wm8962_priv *wm8962 = snd_soc_codec_get_drvdata(codec); + + return regcache_sync_region(wm8962->regmap, WM8962_HDBASS_AI_1, WM8962_MAX_REGISTER); } -- cgit v1.2.3 From 30519cb8d2ecb7f0f0cdc42d709da0d9f7a04bcb Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Thu, 27 Feb 2014 17:49:53 +0800 Subject: ASoC: sgtl5000: Simplify ASoC probe code Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/codecs/sgtl5000.c | 8 -------- 1 file changed, 8 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c index 0fcbe90f3ef2..c8c37e431e7c 100644 --- a/sound/soc/codecs/sgtl5000.c +++ b/sound/soc/codecs/sgtl5000.c @@ -1350,14 +1350,6 @@ static int sgtl5000_probe(struct snd_soc_codec *codec) int ret; struct sgtl5000_priv *sgtl5000 = snd_soc_codec_get_drvdata(codec); - /* setup i2c data ops */ - codec->control_data = sgtl5000->regmap; - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = sgtl5000_enable_regulators(codec); if (ret) return ret; -- cgit v1.2.3 From 28963178427386e83788d314d2a05f0c093f836a Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 28 Feb 2014 08:31:06 +0100 Subject: ASoC: adau1373: Use SOC_ENUM_SINGLE_VIRT_DECL() For the upcoming consolidation for MUXs and virtual MUXs we need to mark virtual enums as such. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adau1373.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 5765c224cd9a..5223800775ad 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -576,8 +576,8 @@ static const char *adau1373_decimator_text[] = { "DMIC1", }; -static SOC_ENUM_SINGLE_DECL(adau1373_decimator_enum, - 0, 0, adau1373_decimator_text); +static SOC_ENUM_SINGLE_VIRT_DECL(adau1373_decimator_enum, + adau1373_decimator_text); static const struct snd_kcontrol_new adau1373_decimator_mux = SOC_DAPM_ENUM_VIRT("Decimator Mux", adau1373_decimator_enum); -- cgit v1.2.3 From ba513116403bc93072dd54a1f89dacdb4d89fcab Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 28 Feb 2014 08:31:07 +0100 Subject: ASoC: max98090: Use SOC_ENUM_SINGLE_VIRT_DECL() For the upcoming consolidation for MUXs and virtual MUXs we need to mark virtual enums as such. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index c7b9e901bdac..1686ade2a429 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -871,7 +871,7 @@ static const struct snd_kcontrol_new max98090_mic2_mux = static const char *dmic_mux_text[] = { "ADC", "DMIC" }; -static SOC_ENUM_SINGLE_EXT_DECL(dmic_mux_enum, dmic_mux_text); +static SOC_ENUM_SINGLE_VIRT_DECL(dmic_mux_enum, dmic_mux_text); static const struct snd_kcontrol_new max98090_dmic_mux = SOC_DAPM_ENUM_VIRT("DMIC Mux", dmic_mux_enum); -- cgit v1.2.3 From 15ab40a9a83035d92b4f4c83cb88c7529bf73c71 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 28 Feb 2014 08:31:08 +0100 Subject: ASoC: mc13783: Use SOC_ENUM_SINGLE_VIRT_DECL() MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit For the upcoming consolidation for MUXs and virtual MUXs we need to mark virtual enums as such. Signed-off-by: Lars-Peter Clausen Tested-by: Philippe Rétornaz Signed-off-by: Mark Brown --- sound/soc/codecs/mc13783.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index c605036cc0b0..ec89b8f90a64 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -408,8 +408,7 @@ static const char * const adcl_enum_text[] = { "MC1L", "RXINL", }; -static SOC_ENUM_SINGLE_DECL(adcl_enum, - 0, 0, adcl_enum_text); +static SOC_ENUM_SINGLE_VIRT_DECL(adcl_enum, adcl_enum_text); static const struct snd_kcontrol_new left_input_mux = SOC_DAPM_ENUM_VIRT("Route", adcl_enum); @@ -418,8 +417,7 @@ static const char * const adcr_enum_text[] = { "MC1R", "MC2", "RXINR", "TXIN", }; -static SOC_ENUM_SINGLE_DECL(adcr_enum, - 0, 0, adcr_enum_text); +static SOC_ENUM_SINGLE_VIRT_DECL(adcr_enum, adcr_enum_text); static const struct snd_kcontrol_new right_input_mux = SOC_DAPM_ENUM_VIRT("Route", adcr_enum); -- cgit v1.2.3 From 86d4c9ab28b73f9eeb8cbd3b11cb2f2aee079a00 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 28 Feb 2014 08:31:09 +0100 Subject: ASoC: wm8994: Use SOC_ENUM_SINGLE_VIRT_DECL() For the upcoming consolidation for MUXs and virtual MUXs we need to mark virtual enums as such. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 699b527e2a77..79854cb7feb6 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -1344,8 +1344,7 @@ static const char *adc_mux_text[] = { "DMIC", }; -static SOC_ENUM_SINGLE_DECL(adc_enum, - 0, 0, adc_mux_text); +static SOC_ENUM_SINGLE_VIRT_DECL(adc_enum, adc_mux_text); static const struct snd_kcontrol_new adcl_mux = SOC_DAPM_ENUM_VIRT("ADCL Mux", adc_enum); -- cgit v1.2.3 From f6b45c28f451e8e415db49f693b2fec90c2cb557 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 28 Feb 2014 08:31:10 +0100 Subject: ASoC: wm8995: Use SOC_ENUM_SINGLE_VIRT_DECL() For the upcoming consolidation for MUXs and virtual MUXs we need to mark virtual enums as such. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8995.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 33ff361250a8..ddb197dc1d53 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -882,7 +882,7 @@ static const char *adc_mux_text[] = { "DMIC", }; -static const SOC_ENUM_SINGLE_DECL(adc_enum, 0, 0, adc_mux_text); +static SOC_ENUM_SINGLE_VIRT_DECL(adc_enum, adc_mux_text); static const struct snd_kcontrol_new adcl_mux = SOC_DAPM_ENUM_VIRT("ADCL Mux", adc_enum); -- cgit v1.2.3 From 48b5e1fb883c8c72b50ee1ccd3ee51e4f53f632f Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 1 Mar 2014 13:46:44 +0100 Subject: ASoC: wm8753: Remove superfluous 'codec->cache_sync = 1' The wm8763 driver uses regmap for IO which means that codec->cache_sync is not used. The line was added in commit d3398ff ('ASoC: Convert WM8753 to direct regmap API usage'). Presumably this was meant to be regcache_mark_dirty(), but since we already call regcache_mark_dirty() in the core when suspending the CODEC it is safe to just remove the line. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm8753.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index be85da93a268..a02e76c248f6 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1440,7 +1440,6 @@ static void wm8753_work(struct work_struct *work) static int wm8753_suspend(struct snd_soc_codec *codec) { wm8753_set_bias_level(codec, SND_SOC_BIAS_OFF); - codec->cache_sync = 1; return 0; } -- cgit v1.2.3 From 055bbe2df957343fece60fe1f60553a9c1005217 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Sat, 1 Mar 2014 13:45:32 +0100 Subject: ASoC: wm{5102, 5110, 8997}: Replace codec->control_data with arizona->regmap With the ongoing component-ization of the ASoC framework and the continuing migration to using regmap for IO the control_data field of the snd_soc_codec struct will eventually be removed. Prepare the wm5192, wm5110 and wm8997 drivers for this by using arizona->regmap instead of accessing the CODEC's control_data field. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 2 +- sound/soc/codecs/wm5110.c | 2 +- sound/soc/codecs/wm8997.c | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 293dffcb1d2f..34109050ceed 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -582,7 +582,7 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = w->codec; struct arizona *arizona = dev_get_drvdata(codec->dev->parent); - struct regmap *regmap = codec->control_data; + struct regmap *regmap = arizona->regmap; const struct reg_default *patch = NULL; int i, patch_size; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 4de2bf16dc74..d7bf8848174a 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -136,7 +136,7 @@ static int wm5110_sysclk_ev(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = w->codec; struct arizona *arizona = dev_get_drvdata(codec->dev->parent); - struct regmap *regmap = codec->control_data; + struct regmap *regmap = arizona->regmap; const struct reg_default *patch = NULL; int i, patch_size; diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 4e6442ce9a2a..e10f44d7fdb7 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -86,7 +86,7 @@ static int wm8997_sysclk_ev(struct snd_soc_dapm_widget *w, { struct snd_soc_codec *codec = w->codec; struct arizona *arizona = dev_get_drvdata(codec->dev->parent); - struct regmap *regmap = codec->control_data; + struct regmap *regmap = arizona->regmap; const struct reg_default *patch = NULL; int i, patch_size; -- cgit v1.2.3 From d4179c1deafd216b9358f76f5f399220cb8451ab Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 4 Mar 2014 16:54:58 +0800 Subject: ASoC: da732x: Replace hw_read usage with snd_soc_read() Pre-merge code was using direct hw_read() calls as an out of framework way of doing volatile register I/O when not using regmap. This has never functioned correctly in mainline due to the regmap conversion, the hw_read() implementation still does caching. In order to facilitate removal of the subsystem level I/O code convert to use snd_soc_read(), there should be no functional impact. Signed-off-by: Mark Brown Acked-by: Adam Thomson --- sound/soc/codecs/da732x.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index f295b6569910..8053e0e7f4a7 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1301,9 +1301,9 @@ static void da732x_dac_offset_adjust(struct snd_soc_codec *codec) msleep(DA732X_WAIT_FOR_STABILIZATION); /* Check DAC offset sign */ - sign[DA732X_HPL_DAC] = (codec->hw_read(codec, DA732X_REG_HPL_DAC_OFF_CNTL) & + sign[DA732X_HPL_DAC] = (snd_soc_read(codec, DA732X_REG_HPL_DAC_OFF_CNTL) & DA732X_HP_DAC_OFF_CNTL_COMPO); - sign[DA732X_HPR_DAC] = (codec->hw_read(codec, DA732X_REG_HPR_DAC_OFF_CNTL) & + sign[DA732X_HPR_DAC] = (snd_soc_read(codec, DA732X_REG_HPR_DAC_OFF_CNTL) & DA732X_HP_DAC_OFF_CNTL_COMPO); /* Binary search DAC offset values (both channels at once) */ @@ -1320,10 +1320,10 @@ static void da732x_dac_offset_adjust(struct snd_soc_codec *codec) msleep(DA732X_WAIT_FOR_STABILIZATION); - if ((codec->hw_read(codec, DA732X_REG_HPL_DAC_OFF_CNTL) & + if ((snd_soc_read(codec, DA732X_REG_HPL_DAC_OFF_CNTL) & DA732X_HP_DAC_OFF_CNTL_COMPO) ^ sign[DA732X_HPL_DAC]) offset[DA732X_HPL_DAC] &= ~step; - if ((codec->hw_read(codec, DA732X_REG_HPR_DAC_OFF_CNTL) & + if ((snd_soc_read(codec, DA732X_REG_HPR_DAC_OFF_CNTL) & DA732X_HP_DAC_OFF_CNTL_COMPO) ^ sign[DA732X_HPR_DAC]) offset[DA732X_HPR_DAC] &= ~step; @@ -1364,9 +1364,9 @@ static void da732x_output_offset_adjust(struct snd_soc_codec *codec) msleep(DA732X_WAIT_FOR_STABILIZATION); /* Check output offset sign */ - sign[DA732X_HPL_AMP] = codec->hw_read(codec, DA732X_REG_HPL) & + sign[DA732X_HPL_AMP] = snd_soc_read(codec, DA732X_REG_HPL) & DA732X_HP_OUT_COMPO; - sign[DA732X_HPR_AMP] = codec->hw_read(codec, DA732X_REG_HPR) & + sign[DA732X_HPR_AMP] = snd_soc_read(codec, DA732X_REG_HPR) & DA732X_HP_OUT_COMPO; snd_soc_write(codec, DA732X_REG_HPL, DA732X_HP_OUT_COMP | @@ -1387,10 +1387,10 @@ static void da732x_output_offset_adjust(struct snd_soc_codec *codec) msleep(DA732X_WAIT_FOR_STABILIZATION); - if ((codec->hw_read(codec, DA732X_REG_HPL) & + if ((snd_soc_read(codec, DA732X_REG_HPL) & DA732X_HP_OUT_COMPO) ^ sign[DA732X_HPL_AMP]) offset[DA732X_HPL_AMP] &= ~step; - if ((codec->hw_read(codec, DA732X_REG_HPR) & + if ((snd_soc_read(codec, DA732X_REG_HPR) & DA732X_HP_OUT_COMPO) ^ sign[DA732X_HPR_AMP]) offset[DA732X_HPR_AMP] &= ~step; -- cgit v1.2.3 From 57d4325a4fa67436b05f30c02a36e204c6cd1282 Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Tue, 4 Mar 2014 09:19:37 +0100 Subject: ASoC: ak4104: Remove superfluous codec->control_data initialization The driver uses automatic IO setup, which will also initialize the control_data field of the CODEC, no need to do it manually. Signed-off-by: Lars-Peter Clausen Acked-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/codecs/ak4104.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index b4819dcd4f4d..10adf25d4c14 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -174,8 +174,6 @@ static int ak4104_probe(struct snd_soc_codec *codec) struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec); int ret; - codec->control_data = ak4104->regmap; - /* set power-up and non-reset bits */ ret = regmap_update_bits(ak4104->regmap, AK4104_REG_CONTROL1, AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN, -- cgit v1.2.3 From 9c369c6e885599818d98ff7130d6ef62ce6ae8d4 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 5 Mar 2014 17:53:31 +0100 Subject: ASoC: cs4271: Fix build error without CONFIG_SPI_MASTER MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit cs4271_common_probe() is called from cs4271_i2c_probe() but defined in CONFIG_SPI_MASTER block, thus it results in a build error when CONFIG_SPI_MASTER=n: sound/soc/codecs/cs4271.c:721:2: error: implicit declaration of function ‘cs4271_common_probe’ [-Werror=implicit-function-declaration] Move the function out of #if block. Fixes: d6cf89ee07cb ('ASoC: cs4271: claim reset GPIO in bus probe function') Signed-off-by: Takashi Iwai Acked-by: Daniel Mack Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs4271.c | 32 ++++++++++++++++---------------- 1 file changed, 16 insertions(+), 16 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/cs4271.c b/sound/soc/codecs/cs4271.c index 96c309777208..aef4965750c7 100644 --- a/sound/soc/codecs/cs4271.c +++ b/sound/soc/codecs/cs4271.c @@ -612,22 +612,6 @@ static struct snd_soc_codec_driver soc_codec_dev_cs4271 = { .num_dapm_routes = ARRAY_SIZE(cs4271_dapm_routes), }; -#if defined(CONFIG_SPI_MASTER) - -static const struct regmap_config cs4271_spi_regmap = { - .reg_bits = 16, - .val_bits = 8, - .max_register = CS4271_LASTREG, - .read_flag_mask = 0x21, - .write_flag_mask = 0x20, - - .reg_defaults = cs4271_reg_defaults, - .num_reg_defaults = ARRAY_SIZE(cs4271_reg_defaults), - .cache_type = REGCACHE_RBTREE, - - .volatile_reg = cs4271_volatile_reg, -}; - static int cs4271_common_probe(struct device *dev, struct cs4271_private **c) { @@ -658,6 +642,22 @@ static int cs4271_common_probe(struct device *dev, return 0; } +#if defined(CONFIG_SPI_MASTER) + +static const struct regmap_config cs4271_spi_regmap = { + .reg_bits = 16, + .val_bits = 8, + .max_register = CS4271_LASTREG, + .read_flag_mask = 0x21, + .write_flag_mask = 0x20, + + .reg_defaults = cs4271_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(cs4271_reg_defaults), + .cache_type = REGCACHE_RBTREE, + + .volatile_reg = cs4271_volatile_reg, +}; + static int cs4271_spi_probe(struct spi_device *spi) { struct cs4271_private *cs4271; -- cgit v1.2.3 From c1a7898d655fd265feefcf6fe82ab0096e6d078e Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 5 Mar 2014 14:28:16 +0000 Subject: ASoC: wm_adsp: Split firmware load into smaller chunks The firmware files can be quite large and allocating the whole firmware a single DMA safe buffer can be problematic if the system is under a high memory load. Ease the requirements slightly by writing the firmware out in page sized chunks. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 48 ++++++++++++++++++++++++++++++---------------- 1 file changed, 31 insertions(+), 17 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f9fd56444a14..937af6f31ffa 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -684,24 +684,38 @@ static int wm_adsp_load(struct wm_adsp *dsp) } if (reg) { - buf = wm_adsp_buf_alloc(region->data, - le32_to_cpu(region->len), - &buf_list); - if (!buf) { - adsp_err(dsp, "Out of memory\n"); - ret = -ENOMEM; - goto out_fw; - } + size_t to_write = PAGE_SIZE; + size_t remain = le32_to_cpu(region->len); + const u8 *data = region->data; + + while (remain > 0) { + if (remain < PAGE_SIZE) + to_write = remain; + + buf = wm_adsp_buf_alloc(data, + to_write, + &buf_list); + if (!buf) { + adsp_err(dsp, "Out of memory\n"); + ret = -ENOMEM; + goto out_fw; + } - ret = regmap_raw_write_async(regmap, reg, buf->buf, - le32_to_cpu(region->len)); - if (ret != 0) { - adsp_err(dsp, - "%s.%d: Failed to write %d bytes at %d in %s: %d\n", - file, regions, - le32_to_cpu(region->len), offset, - region_name, ret); - goto out_fw; + ret = regmap_raw_write_async(regmap, reg, + buf->buf, + to_write); + if (ret != 0) { + adsp_err(dsp, + "%s.%d: Failed to write %d bytes at %d in %s: %d\n", + file, regions, + to_write, offset, + region_name, ret); + goto out_fw; + } + + data += to_write; + reg += to_write / 2; + remain -= to_write; } } -- cgit v1.2.3 From f516e368dcb5eb5fbe23246c09bf69573d67cd18 Mon Sep 17 00:00:00 2001 From: Rongjun Ying Date: Wed, 5 Mar 2014 16:34:34 +0800 Subject: ASoC: sirf: Add SiRF internal audio codec driver SiRF internal audio codec is integrated in SiRF atlas6 and prima2 SoC. Features include: 1. Stereo DAC and ADC with 16-bit resolution amd 48KHz sample rate 2. Support headphone and/or speaker output 3. Integrate headphone and speaker output amp 4. Support LINE and MIC input 5. Support single ended and differential input mode Signed-off-by: Rongjun Ying --v5: 1. Drop all inlines. 2. Reordering the Kconfig and Makefile 3. Remove the sirf_audio_codec_reg_bits struct, use the new controls instead it. 4. Add some SND_SOC_DAPM_OUT_DRV instead of HP and SPK enable driver 5. Add audio codec clock supply instead of adc event callback 6. Fixed playback and capture can't concurrent work bug. -- .../devicetree/bindings/sound/sirf-audio-codec.txt | 17 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 1 + sound/soc/codecs/sirf-audio-codec.c | 533 ++++++++++++++++++++ sound/soc/codecs/sirf-audio-codec.h | 75 +++ 5 files changed, 631 insertions(+), 0 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-codec.txt create mode 100644 sound/soc/codecs/sirf-audio-codec.c create mode 100644 sound/soc/codecs/sirf-audio-codec.h Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/sirf-audio-codec.txt | 17 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 1 + sound/soc/codecs/sirf-audio-codec.c | 533 +++++++++++++++++++++ sound/soc/codecs/sirf-audio-codec.h | 75 +++ 5 files changed, 631 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/sirf-audio-codec.txt create mode 100644 sound/soc/codecs/sirf-audio-codec.c create mode 100644 sound/soc/codecs/sirf-audio-codec.h (limited to 'sound/soc/codecs') diff --git a/Documentation/devicetree/bindings/sound/sirf-audio-codec.txt b/Documentation/devicetree/bindings/sound/sirf-audio-codec.txt new file mode 100644 index 000000000000..062f5ec36f9b --- /dev/null +++ b/Documentation/devicetree/bindings/sound/sirf-audio-codec.txt @@ -0,0 +1,17 @@ +SiRF internal audio CODEC + +Required properties: + + - compatible : "sirf,atlas6-audio-codec" or "sirf,prima2-audio-codec" + + - reg : the register address of the device. + + - clocks: the clock of SiRF internal audio codec + +Example: + +audiocodec: audiocodec@b0040000 { + compatible = "sirf,atlas6-audio-codec"; + reg = <0xb0040000 0x10000>; + clocks = <&clks 27>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 983d087aa92a..bf9b12c1a9c0 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -63,6 +63,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_RT5640 if I2C select SND_SOC_SGTL5000 if I2C select SND_SOC_SI476X if MFD_SI476X_CORE + select SND_SOC_SIRF_AUDIO_CODEC select SND_SOC_SN95031 if INTEL_SCU_IPC select SND_SOC_SPDIF select SND_SOC_SSM2518 if I2C @@ -330,6 +331,10 @@ config SND_SOC_SIGMADSP tristate select CRC32 +config SND_SOC_SIRF_AUDIO_CODEC + tristate "SiRF SoC internal audio codec" + select REGMAP_MMIO + config SND_SOC_SN95031 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index bc126764a44d..de6d7f81b5f6 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -53,6 +53,7 @@ snd-soc-alc5623-objs := alc5623.o snd-soc-alc5632-objs := alc5632.o snd-soc-sigmadsp-objs := sigmadsp.o snd-soc-si476x-objs := si476x.o +snd-soc-sirf-audio-codec-objs := sirf-audio-codec.o snd-soc-sn95031-objs := sn95031.o snd-soc-spdif-tx-objs := spdif_transmitter.o snd-soc-spdif-rx-objs := spdif_receiver.o diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c new file mode 100644 index 000000000000..90e3a228bae4 --- /dev/null +++ b/sound/soc/codecs/sirf-audio-codec.c @@ -0,0 +1,533 @@ +/* + * SiRF audio codec driver + * + * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company. + * + * Licensed under GPLv2 or later. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "sirf-audio-codec.h" + +struct sirf_audio_codec { + struct clk *clk; + struct regmap *regmap; + u32 reg_ctrl0, reg_ctrl1; +}; + +static const char * const input_mode_mux[] = {"Single-ended", + "Differential"}; + +static const struct soc_enum input_mode_mux_enum = + SOC_ENUM_SINGLE(AUDIO_IC_CODEC_CTRL1, 4, 2, input_mode_mux); + +static const struct snd_kcontrol_new sirf_audio_codec_input_mode_control = + SOC_DAPM_ENUM("Route", input_mode_mux_enum); + +static const DECLARE_TLV_DB_SCALE(playback_vol_tlv, -12400, 100, 0); +static const DECLARE_TLV_DB_SCALE(capture_vol_tlv_prima2, 500, 100, 0); +static const DECLARE_TLV_DB_RANGE(capture_vol_tlv_atlas6, + 0, 7, TLV_DB_SCALE_ITEM(-100, 100, 0), + 0x22, 0x3F, TLV_DB_SCALE_ITEM(700, 100, 0), +); + +static struct snd_kcontrol_new volume_controls_atlas6[] = { + SOC_DOUBLE_TLV("Playback Volume", AUDIO_IC_CODEC_CTRL0, 21, 14, + 0x7F, 0, playback_vol_tlv), + SOC_DOUBLE_TLV("Capture Volume", AUDIO_IC_CODEC_CTRL1, 16, 10, + 0x3F, 0, capture_vol_tlv_atlas6), +}; + +static struct snd_kcontrol_new volume_controls_prima2[] = { + SOC_DOUBLE_TLV("Speaker Volume", AUDIO_IC_CODEC_CTRL0, 21, 14, + 0x7F, 0, playback_vol_tlv), + SOC_DOUBLE_TLV("Capture Volume", AUDIO_IC_CODEC_CTRL1, 15, 10, + 0x1F, 0, capture_vol_tlv_prima2), +}; + +static struct snd_kcontrol_new left_input_path_controls[] = { + SOC_DAPM_SINGLE("Line Left Switch", AUDIO_IC_CODEC_CTRL1, 6, 1, 0), + SOC_DAPM_SINGLE("Mic Left Switch", AUDIO_IC_CODEC_CTRL1, 3, 1, 0), +}; + +static struct snd_kcontrol_new right_input_path_controls[] = { + SOC_DAPM_SINGLE("Line Right Switch", AUDIO_IC_CODEC_CTRL1, 5, 1, 0), + SOC_DAPM_SINGLE("Mic Right Switch", AUDIO_IC_CODEC_CTRL1, 2, 1, 0), +}; + +static struct snd_kcontrol_new left_dac_to_hp_left_amp_switch_control = + SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 9, 1, 0); + +static struct snd_kcontrol_new left_dac_to_hp_right_amp_switch_control = + SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 8, 1, 0); + +static struct snd_kcontrol_new right_dac_to_hp_left_amp_switch_control = + SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 7, 1, 0); + +static struct snd_kcontrol_new right_dac_to_hp_right_amp_switch_control = + SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 6, 1, 0); + +static struct snd_kcontrol_new left_dac_to_speaker_lineout_switch_control = + SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 11, 1, 0); + +static struct snd_kcontrol_new right_dac_to_speaker_lineout_switch_control = + SOC_DAPM_SINGLE("Switch", AUDIO_IC_CODEC_CTRL0, 10, 1, 0); + +/* After enable adc, Delay 200ms to avoid pop noise */ +static int adc_enable_delay_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + msleep(200); + break; + default: + break; + } + + return 0; +} + +static void enable_and_reset_codec(struct regmap *regmap, + u32 codec_enable_bits, u32 codec_reset_bits) +{ + regmap_update_bits(regmap, AUDIO_IC_CODEC_CTRL1, + codec_enable_bits | codec_reset_bits, + codec_enable_bits | ~codec_reset_bits); + msleep(20); + regmap_update_bits(regmap, AUDIO_IC_CODEC_CTRL1, + codec_reset_bits, codec_reset_bits); +} + +static int atlas6_codec_enable_and_reset_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ +#define ATLAS6_CODEC_ENABLE_BITS (1 << 29) +#define ATLAS6_CODEC_RESET_BITS (1 << 28) + struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(w->codec->dev); + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + enable_and_reset_codec(sirf_audio_codec->regmap, + ATLAS6_CODEC_ENABLE_BITS, ATLAS6_CODEC_RESET_BITS); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(sirf_audio_codec->regmap, + AUDIO_IC_CODEC_CTRL1, ATLAS6_CODEC_ENABLE_BITS, + ~ATLAS6_CODEC_ENABLE_BITS); + break; + default: + break; + } + + return 0; +} + +static int prima2_codec_enable_and_reset_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ +#define PRIMA2_CODEC_ENABLE_BITS (1 << 27) +#define PRIMA2_CODEC_RESET_BITS (1 << 26) + struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(w->codec->dev); + switch (event) { + case SND_SOC_DAPM_POST_PMU: + enable_and_reset_codec(sirf_audio_codec->regmap, + PRIMA2_CODEC_ENABLE_BITS, PRIMA2_CODEC_RESET_BITS); + break; + case SND_SOC_DAPM_POST_PMD: + regmap_update_bits(sirf_audio_codec->regmap, + AUDIO_IC_CODEC_CTRL1, PRIMA2_CODEC_ENABLE_BITS, + ~PRIMA2_CODEC_ENABLE_BITS); + break; + default: + break; + } + + return 0; +} + +static const struct snd_soc_dapm_widget atlas6_output_driver_dapm_widgets[] = { + SND_SOC_DAPM_OUT_DRV("HP Left Driver", AUDIO_IC_CODEC_CTRL1, + 25, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("HP Right Driver", AUDIO_IC_CODEC_CTRL1, + 26, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Speaker Driver", AUDIO_IC_CODEC_CTRL1, + 27, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_widget prima2_output_driver_dapm_widgets[] = { + SND_SOC_DAPM_OUT_DRV("HP Left Driver", AUDIO_IC_CODEC_CTRL1, + 23, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("HP Right Driver", AUDIO_IC_CODEC_CTRL1, + 24, 0, NULL, 0), + SND_SOC_DAPM_OUT_DRV("Speaker Driver", AUDIO_IC_CODEC_CTRL1, + 25, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_widget atlas6_codec_clock_dapm_widget = + SND_SOC_DAPM_SUPPLY("codecclk", SND_SOC_NOPM, 0, 0, + atlas6_codec_enable_and_reset_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD); + +static const struct snd_soc_dapm_widget prima2_codec_clock_dapm_widget = + SND_SOC_DAPM_SUPPLY("codecclk", SND_SOC_NOPM, 0, 0, + prima2_codec_enable_and_reset_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD); + +static const struct snd_soc_dapm_widget sirf_audio_codec_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC left", NULL, AUDIO_IC_CODEC_CTRL0, 1, 0), + SND_SOC_DAPM_DAC("DAC right", NULL, AUDIO_IC_CODEC_CTRL0, 0, 0), + SND_SOC_DAPM_SWITCH("Left dac to hp left amp", SND_SOC_NOPM, 0, 0, + &left_dac_to_hp_left_amp_switch_control), + SND_SOC_DAPM_SWITCH("Left dac to hp right amp", SND_SOC_NOPM, 0, 0, + &left_dac_to_hp_right_amp_switch_control), + SND_SOC_DAPM_SWITCH("Right dac to hp left amp", SND_SOC_NOPM, 0, 0, + &right_dac_to_hp_left_amp_switch_control), + SND_SOC_DAPM_SWITCH("Right dac to hp right amp", SND_SOC_NOPM, 0, 0, + &right_dac_to_hp_right_amp_switch_control), + SND_SOC_DAPM_OUT_DRV("HP amp left driver", AUDIO_IC_CODEC_CTRL0, 3, 0, + NULL, 0), + SND_SOC_DAPM_OUT_DRV("HP amp right driver", AUDIO_IC_CODEC_CTRL0, 3, 0, + NULL, 0), + + SND_SOC_DAPM_SWITCH("Left dac to speaker lineout", SND_SOC_NOPM, 0, 0, + &left_dac_to_speaker_lineout_switch_control), + SND_SOC_DAPM_SWITCH("Right dac to speaker lineout", SND_SOC_NOPM, 0, 0, + &right_dac_to_speaker_lineout_switch_control), + SND_SOC_DAPM_OUT_DRV("Speaker amp driver", AUDIO_IC_CODEC_CTRL0, 4, 0, + NULL, 0), + + SND_SOC_DAPM_OUTPUT("HPOUTL"), + SND_SOC_DAPM_OUTPUT("HPOUTR"), + SND_SOC_DAPM_OUTPUT("SPKOUT"), + + SND_SOC_DAPM_ADC_E("ADC left", NULL, AUDIO_IC_CODEC_CTRL1, 8, 0, + adc_enable_delay_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_ADC_E("ADC right", NULL, AUDIO_IC_CODEC_CTRL1, 7, 0, + adc_enable_delay_event, SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_MIXER("Left PGA mixer", AUDIO_IC_CODEC_CTRL1, 1, 0, + &left_input_path_controls[0], + ARRAY_SIZE(left_input_path_controls)), + SND_SOC_DAPM_MIXER("Right PGA mixer", AUDIO_IC_CODEC_CTRL1, 0, 0, + &right_input_path_controls[0], + ARRAY_SIZE(right_input_path_controls)), + + SND_SOC_DAPM_MUX("Mic input mode mux", SND_SOC_NOPM, 0, 0, + &sirf_audio_codec_input_mode_control), + SND_SOC_DAPM_MICBIAS("Mic Bias", AUDIO_IC_CODEC_PWR, 3, 0), + SND_SOC_DAPM_INPUT("MICIN1"), + SND_SOC_DAPM_INPUT("MICIN2"), + SND_SOC_DAPM_INPUT("LINEIN1"), + SND_SOC_DAPM_INPUT("LINEIN2"), + + SND_SOC_DAPM_SUPPLY("HSL Phase Opposite", AUDIO_IC_CODEC_CTRL0, + 30, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_route sirf_audio_codec_map[] = { + {"SPKOUT", NULL, "Speaker Driver"}, + {"Speaker Driver", NULL, "Speaker amp driver"}, + {"Speaker amp driver", NULL, "Left dac to speaker lineout"}, + {"Speaker amp driver", NULL, "Right dac to speaker lineout"}, + {"Left dac to speaker lineout", "Switch", "DAC left"}, + {"Right dac to speaker lineout", "Switch", "DAC right"}, + {"HPOUTL", NULL, "HP Left Driver"}, + {"HPOUTR", NULL, "HP Right Driver"}, + {"HP Left Driver", NULL, "HP amp left driver"}, + {"HP Right Driver", NULL, "HP amp right driver"}, + {"HP amp left driver", NULL, "Right dac to hp left amp"}, + {"HP amp right driver", NULL , "Right dac to hp right amp"}, + {"HP amp left driver", NULL, "Left dac to hp left amp"}, + {"HP amp right driver", NULL , "Right dac to hp right amp"}, + {"Right dac to hp left amp", "Switch", "DAC left"}, + {"Right dac to hp right amp", "Switch", "DAC right"}, + {"Left dac to hp left amp", "Switch", "DAC left"}, + {"Left dac to hp right amp", "Switch", "DAC right"}, + {"DAC left", NULL, "codecclk"}, + {"DAC right", NULL, "codecclk"}, + {"DAC left", NULL, "Playback"}, + {"DAC right", NULL, "Playback"}, + {"DAC left", NULL, "HSL Phase Opposite"}, + {"DAC right", NULL, "HSL Phase Opposite"}, + + {"Capture", NULL, "ADC left"}, + {"Capture", NULL, "ADC right"}, + {"ADC left", NULL, "codecclk"}, + {"ADC right", NULL, "codecclk"}, + {"ADC left", NULL, "Left PGA mixer"}, + {"ADC right", NULL, "Right PGA mixer"}, + {"Left PGA mixer", "Line Left Switch", "LINEIN2"}, + {"Right PGA mixer", "Line Right Switch", "LINEIN1"}, + {"Left PGA mixer", "Mic Left Switch", "MICIN2"}, + {"Right PGA mixer", "Mic Right Switch", "Mic input mode mux"}, + {"Mic input mode mux", "Single-ended", "MICIN1"}, + {"Mic input mode mux", "Differential", "MICIN1"}, +}; + +static int sirf_audio_codec_trigger(struct snd_pcm_substream *substream, + int cmd, + struct snd_soc_dai *dai) +{ + int playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct snd_soc_codec *codec = dai->codec; + u32 val = 0; + + /* + * This is a workaround, When stop playback, + * need disable HP amp, avoid the current noise. + */ + switch (cmd) { + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + break; + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (playback) + val = IC_HSLEN | IC_HSREN; + break; + default: + return -EINVAL; + } + + if (playback) + snd_soc_update_bits(codec, AUDIO_IC_CODEC_CTRL0, + IC_HSLEN | IC_HSREN, val); + return 0; +} + +struct snd_soc_dai_ops sirf_audio_codec_dai_ops = { + .trigger = sirf_audio_codec_trigger, +}; + +struct snd_soc_dai_driver sirf_audio_codec_dai = { + .name = "sirf-audio-codec", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &sirf_audio_codec_dai_ops, +}; + +static int sirf_audio_codec_probe(struct snd_soc_codec *codec) +{ + int ret; + struct snd_soc_dapm_context *dapm = &codec->dapm; + struct sirf_audio_codec *sirf_audio_codec = snd_soc_codec_get_drvdata(codec); + + pm_runtime_enable(codec->dev); + codec->control_data = sirf_audio_codec->regmap; + + ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + if (ret != 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + return ret; + } + + if (of_device_is_compatible(codec->dev->of_node, "sirf,prima2-audio-codec")) { + snd_soc_dapm_new_controls(dapm, + prima2_output_driver_dapm_widgets, + ARRAY_SIZE(prima2_output_driver_dapm_widgets)); + snd_soc_dapm_new_controls(dapm, + &prima2_codec_clock_dapm_widget, 1); + return snd_soc_add_codec_controls(codec, + volume_controls_prima2, + ARRAY_SIZE(volume_controls_prima2)); + } + if (of_device_is_compatible(codec->dev->of_node, "sirf,atlas6-audio-codec")) { + snd_soc_dapm_new_controls(dapm, + atlas6_output_driver_dapm_widgets, + ARRAY_SIZE(atlas6_output_driver_dapm_widgets)); + snd_soc_dapm_new_controls(dapm, + &atlas6_codec_clock_dapm_widget, 1); + return snd_soc_add_codec_controls(codec, + volume_controls_atlas6, + ARRAY_SIZE(volume_controls_atlas6)); + } + + return -EINVAL; +} + +static int sirf_audio_codec_remove(struct snd_soc_codec *codec) +{ + pm_runtime_disable(codec->dev); + return 0; +} + +static struct snd_soc_codec_driver soc_codec_device_sirf_audio_codec = { + .probe = sirf_audio_codec_probe, + .remove = sirf_audio_codec_remove, + .dapm_widgets = sirf_audio_codec_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sirf_audio_codec_dapm_widgets), + .dapm_routes = sirf_audio_codec_map, + .num_dapm_routes = ARRAY_SIZE(sirf_audio_codec_map), + .idle_bias_off = true, +}; + +static const struct of_device_id sirf_audio_codec_of_match[] = { + { .compatible = "sirf,prima2-audio-codec" }, + { .compatible = "sirf,atlas6-audio-codec" }, + {} +}; +MODULE_DEVICE_TABLE(of, sirf_audio_codec_of_match); + +static const struct regmap_config sirf_audio_codec_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = AUDIO_IC_CODEC_CTRL3, + .cache_type = REGCACHE_NONE, +}; + +static int sirf_audio_codec_driver_probe(struct platform_device *pdev) +{ + int ret; + struct sirf_audio_codec *sirf_audio_codec; + void __iomem *base; + struct resource *mem_res; + const struct of_device_id *match; + + match = of_match_node(sirf_audio_codec_of_match, pdev->dev.of_node); + + sirf_audio_codec = devm_kzalloc(&pdev->dev, + sizeof(struct sirf_audio_codec), GFP_KERNEL); + if (!sirf_audio_codec) + return -ENOMEM; + + platform_set_drvdata(pdev, sirf_audio_codec); + + mem_res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, mem_res); + if (base == NULL) + return -ENOMEM; + + sirf_audio_codec->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &sirf_audio_codec_regmap_config); + if (IS_ERR(sirf_audio_codec->regmap)) + return PTR_ERR(sirf_audio_codec->regmap); + + sirf_audio_codec->clk = devm_clk_get(&pdev->dev, NULL); + if (IS_ERR(sirf_audio_codec->clk)) { + dev_err(&pdev->dev, "Get clock failed.\n"); + return PTR_ERR(sirf_audio_codec->clk); + } + + ret = clk_prepare_enable(sirf_audio_codec->clk); + if (ret) { + dev_err(&pdev->dev, "Enable clock failed.\n"); + return ret; + } + + ret = snd_soc_register_codec(&(pdev->dev), + &soc_codec_device_sirf_audio_codec, + &sirf_audio_codec_dai, 1); + if (ret) { + dev_err(&pdev->dev, "Register Audio Codec dai failed.\n"); + goto err_clk_put; + } + + /* + * Always open charge pump, if not, when the charge pump closed the + * adc will not stable + */ + regmap_update_bits(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL0, + IC_CPFREQ, IC_CPFREQ); + + if (of_device_is_compatible(pdev->dev.of_node, "sirf,atlas6-audio-codec")) + regmap_update_bits(sirf_audio_codec->regmap, + AUDIO_IC_CODEC_CTRL0, IC_CPEN, IC_CPEN); + return 0; + +err_clk_put: + clk_disable_unprepare(sirf_audio_codec->clk); + return ret; +} + +static int sirf_audio_codec_driver_remove(struct platform_device *pdev) +{ + struct sirf_audio_codec *sirf_audio_codec = platform_get_drvdata(pdev); + + clk_disable_unprepare(sirf_audio_codec->clk); + snd_soc_unregister_codec(&(pdev->dev)); + + return 0; +} + +#ifdef CONFIG_PM_SLEEP +static int sirf_audio_codec_suspend(struct device *dev) +{ + struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(dev); + + regmap_read(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL0, + &sirf_audio_codec->reg_ctrl0); + regmap_read(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL1, + &sirf_audio_codec->reg_ctrl1); + clk_disable_unprepare(sirf_audio_codec->clk); + + return 0; +} + +static int sirf_audio_codec_resume(struct device *dev) +{ + struct sirf_audio_codec *sirf_audio_codec = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(sirf_audio_codec->clk); + if (ret) + return ret; + + regmap_write(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL0, + sirf_audio_codec->reg_ctrl0); + regmap_write(sirf_audio_codec->regmap, AUDIO_IC_CODEC_CTRL1, + sirf_audio_codec->reg_ctrl1); + + return 0; +} +#endif + +static const struct dev_pm_ops sirf_audio_codec_pm_ops = { + SET_SYSTEM_SLEEP_PM_OPS(sirf_audio_codec_suspend, sirf_audio_codec_resume) +}; + +static struct platform_driver sirf_audio_codec_driver = { + .driver = { + .name = "sirf-audio-codec", + .owner = THIS_MODULE, + .of_match_table = sirf_audio_codec_of_match, + .pm = &sirf_audio_codec_pm_ops, + }, + .probe = sirf_audio_codec_driver_probe, + .remove = sirf_audio_codec_driver_remove, +}; + +module_platform_driver(sirf_audio_codec_driver); + +MODULE_DESCRIPTION("SiRF audio codec driver"); +MODULE_AUTHOR("RongJun Ying "); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/sirf-audio-codec.h b/sound/soc/codecs/sirf-audio-codec.h new file mode 100644 index 000000000000..d4c187b8e54a --- /dev/null +++ b/sound/soc/codecs/sirf-audio-codec.h @@ -0,0 +1,75 @@ +/* + * SiRF inner codec controllers define + * + * Copyright (c) 2011 Cambridge Silicon Radio Limited, a CSR plc group company. + * + * Licensed under GPLv2 or later. + */ + +#ifndef _SIRF_AUDIO_CODEC_H +#define _SIRF_AUDIO_CODEC_H + + +#define AUDIO_IC_CODEC_PWR (0x00E0) +#define AUDIO_IC_CODEC_CTRL0 (0x00E4) +#define AUDIO_IC_CODEC_CTRL1 (0x00E8) +#define AUDIO_IC_CODEC_CTRL2 (0x00EC) +#define AUDIO_IC_CODEC_CTRL3 (0x00F0) + +#define MICBIASEN (1 << 3) + +#define IC_RDACEN (1 << 0) +#define IC_LDACEN (1 << 1) +#define IC_HSREN (1 << 2) +#define IC_HSLEN (1 << 3) +#define IC_SPEN (1 << 4) +#define IC_CPEN (1 << 5) + +#define IC_HPRSELR (1 << 6) +#define IC_HPLSELR (1 << 7) +#define IC_HPRSELL (1 << 8) +#define IC_HPLSELL (1 << 9) +#define IC_SPSELR (1 << 10) +#define IC_SPSELL (1 << 11) + +#define IC_MONOR (1 << 12) +#define IC_MONOL (1 << 13) + +#define IC_RXOSRSEL (1 << 28) +#define IC_CPFREQ (1 << 29) +#define IC_HSINVEN (1 << 30) + +#define IC_MICINREN (1 << 0) +#define IC_MICINLEN (1 << 1) +#define IC_MICIN1SEL (1 << 2) +#define IC_MICIN2SEL (1 << 3) +#define IC_MICDIFSEL (1 << 4) +#define IC_LINEIN1SEL (1 << 5) +#define IC_LINEIN2SEL (1 << 6) +#define IC_RADCEN (1 << 7) +#define IC_LADCEN (1 << 8) +#define IC_ALM (1 << 9) + +#define IC_DIGMICEN (1 << 22) +#define IC_DIGMICFREQ (1 << 23) +#define IC_ADC14B_12 (1 << 24) +#define IC_FIRDAC_HSL_EN (1 << 25) +#define IC_FIRDAC_HSR_EN (1 << 26) +#define IC_FIRDAC_LOUT_EN (1 << 27) +#define IC_POR (1 << 28) +#define IC_CODEC_CLK_EN (1 << 29) +#define IC_HP_3DB_BOOST (1 << 30) + +#define IC_ADC_LEFT_GAIN_SHIFT 16 +#define IC_ADC_RIGHT_GAIN_SHIFT 10 +#define IC_ADC_GAIN_MASK 0x3F +#define IC_MIC_MAX_GAIN 0x39 + +#define IC_RXPGAR_MASK 0x3F +#define IC_RXPGAR_SHIFT 14 +#define IC_RXPGAL_MASK 0x3F +#define IC_RXPGAL_SHIFT 21 +#define IC_RXPGAR 0x7B +#define IC_RXPGAL 0x7B + +#endif /*__SIRF_AUDIO_CODEC_H*/ -- cgit v1.2.3 From 5c898e74d135a23ce12e0263c1a3c78eeae1b52b Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Wed, 5 Mar 2014 13:17:45 +0100 Subject: ASoC: Add helper function to check whether a CODEC is active Instead of directly checking the 'active' field of the CODEC struct add a new helper function that will return either true or false depending on whether the CODEC is active. This will make the migration to the component level easier. The patch also updates all CODEC drivers that check the active attribute to use the new helper function. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- include/sound/soc.h | 5 +++++ sound/soc/codecs/adav80x.c | 4 ++-- sound/soc/codecs/tlv320aic23.c | 2 +- sound/soc/codecs/tlv320dac33.c | 2 +- sound/soc/codecs/uda1380.c | 2 +- sound/soc/codecs/wl1273.c | 2 +- sound/soc/codecs/wm8711.c | 2 +- sound/soc/codecs/wm8753.c | 4 ++-- 8 files changed, 14 insertions(+), 9 deletions(-) (limited to 'sound/soc/codecs') diff --git a/include/sound/soc.h b/include/sound/soc.h index 53d15e0e6e89..5c2b4f4b5cfa 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1172,6 +1172,11 @@ static inline bool snd_soc_volsw_is_stereo(struct soc_mixer_control *mc) return 1; } +static inline bool snd_soc_codec_is_active(struct snd_soc_codec *codec) +{ + return codec->active != 0; +} + int snd_soc_util_init(void); void snd_soc_util_exit(void); diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index f78b27a7c461..d50cf5b29a27 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -722,7 +722,7 @@ static int adav80x_dai_startup(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); - if (!codec->active || !adav80x->rate) + if (!snd_soc_codec_is_active(codec) || !adav80x->rate) return 0; return snd_pcm_hw_constraint_minmax(substream->runtime, @@ -735,7 +735,7 @@ static void adav80x_dai_shutdown(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); - if (!codec->active) + if (!snd_soc_codec_is_active(codec)) adav80x->rate = 0; } diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 5d430cc56f51..458a6aed203e 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -400,7 +400,7 @@ static void tlv320aic23_shutdown(struct snd_pcm_substream *substream, struct aic23 *aic23 = snd_soc_codec_get_drvdata(codec); /* deactivate */ - if (!codec->active) { + if (!snd_soc_codec_is_active(codec)) { udelay(50); snd_soc_write(codec, TLV320AIC23_ACTIVE, 0x0); } diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 4f358393d6d6..35b2d244e42e 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -461,7 +461,7 @@ static int dac33_set_fifo_mode(struct snd_kcontrol *kcontrol, if (dac33->fifo_mode == ucontrol->value.integer.value[0]) return 0; /* Do not allow changes while stream is running*/ - if (codec->active) + if (snd_soc_codec_is_active(codec)) return -EPERM; if (ucontrol->value.integer.value[0] < 0 || diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 726df6d43c2b..8e3940dcff20 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -108,7 +108,7 @@ static int uda1380_write(struct snd_soc_codec *codec, unsigned int reg, /* the interpolator & decimator regs must only be written when the * codec DAI is active. */ - if (!codec->active && (reg >= UDA1380_MVOL)) + if (!snd_soc_codec_is_active(codec) && (reg >= UDA1380_MVOL)) return 0; pr_debug("uda1380: hw write %x val %x\n", reg, value); if (codec->hw_write(codec->control_data, data, 3) == 3) { diff --git a/sound/soc/codecs/wl1273.c b/sound/soc/codecs/wl1273.c index b7ab2ef567c8..47e96ff30064 100644 --- a/sound/soc/codecs/wl1273.c +++ b/sound/soc/codecs/wl1273.c @@ -197,7 +197,7 @@ static int snd_wl1273_set_audio_route(struct snd_kcontrol *kcontrol, return 0; /* Do not allow changes while stream is running */ - if (codec->active) + if (snd_soc_codec_is_active(codec)) return -EPERM; if (ucontrol->value.integer.value[0] < 0 || diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index d99f948c513c..6efcc40a7cb3 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -201,7 +201,7 @@ static void wm8711_shutdown(struct snd_pcm_substream *substream, struct snd_soc_codec *codec = dai->codec; /* deactivate */ - if (!codec->active) { + if (!snd_soc_codec_is_active(codec)) { udelay(50); snd_soc_write(codec, WM8711_ACTIVE, 0x0); } diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index be85da93a268..5cf4bebc5d89 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -251,7 +251,7 @@ static int wm8753_set_dai(struct snd_kcontrol *kcontrol, if (wm8753->dai_func == ucontrol->value.integer.value[0]) return 0; - if (codec->active) + if (snd_soc_codec_is_active(codec)) return -EBUSY; ioctl = snd_soc_read(codec, WM8753_IOCTL); @@ -1314,7 +1314,7 @@ static int wm8753_mute(struct snd_soc_dai *dai, int mute) /* the digital mute covers the HiFi and Voice DAC's on the WM8753. * make sure we check if they are not both active when we mute */ if (mute && wm8753->dai_func == 1) { - if (!codec->active) + if (!snd_soc_codec_is_active(codec)) snd_soc_write(codec, WM8753_DAC, mute_reg | 0x8); } else { if (mute) -- cgit v1.2.3 From fab800cc33e98378336faf75688ea0961eac21b6 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 6 Mar 2014 10:00:18 +0000 Subject: ASoC: wm_adsp: Correct type specifier in printf Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 937af6f31ffa..bb5f7b4e3ebb 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -706,7 +706,7 @@ static int wm_adsp_load(struct wm_adsp *dsp) to_write); if (ret != 0) { adsp_err(dsp, - "%s.%d: Failed to write %d bytes at %d in %s: %d\n", + "%s.%d: Failed to write %zd bytes at %d in %s: %d\n", file, regions, to_write, offset, region_name, ret); -- cgit v1.2.3 From b3fc5725967cea8b661383742ccce21fdeb3ef72 Mon Sep 17 00:00:00 2001 From: Max Filippov Date: Thu, 6 Mar 2014 14:04:41 +0400 Subject: ASoC: tlv320aic23: add support for SPI control mode tlv320aic23 chip control interface may work in either I2C or SPI mode depending on the MODE pin state. Functionality and register layout are independent of the control mode. Implement bus-specific parts as separate modules. Signed-off-by: Max Filippov Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 11 ++++++- sound/soc/codecs/Makefile | 4 +++ sound/soc/codecs/tlv320aic23-i2c.c | 59 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic23-spi.c | 57 ++++++++++++++++++++++++++++++++++++ sound/soc/codecs/tlv320aic23.c | 55 +++++++---------------------------- sound/soc/codecs/tlv320aic23.h | 6 ++++ 6 files changed, 147 insertions(+), 45 deletions(-) create mode 100644 sound/soc/codecs/tlv320aic23-i2c.c create mode 100644 sound/soc/codecs/tlv320aic23-spi.c (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 983d087aa92a..5e4fc048d42a 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -71,7 +71,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_STA529 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS select SND_SOC_TAS5086 if I2C - select SND_SOC_TLV320AIC23 if I2C + select SND_SOC_TLV320AIC23_I2C if I2C + select SND_SOC_TLV320AIC23_SPI if SPI_MASTER select SND_SOC_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC32X4 if I2C select SND_SOC_TLV320AIC3X if I2C @@ -357,6 +358,14 @@ config SND_SOC_TAS5086 config SND_SOC_TLV320AIC23 tristate +config SND_SOC_TLV320AIC23_I2C + tristate + select SND_SOC_TLV320AIC23 + +config SND_SOC_TLV320AIC23_SPI + tristate + select SND_SOC_TLV320AIC23 + config SND_SOC_TLV320AIC26 tristate depends on SPI diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index bc126764a44d..ed1fd8925e43 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -63,6 +63,8 @@ snd-soc-sta529-objs := sta529.o snd-soc-stac9766-objs := stac9766.o snd-soc-tas5086-objs := tas5086.o snd-soc-tlv320aic23-objs := tlv320aic23.o +snd-soc-tlv320aic23-i2c-objs := tlv320aic23-i2c.o +snd-soc-tlv320aic23-spi-objs := tlv320aic23-spi.o snd-soc-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o @@ -193,6 +195,8 @@ obj-$(CONFIG_SND_SOC_STA529) += snd-soc-sta529.o obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o +obj-$(CONFIG_SND_SOC_TLV320AIC23_I2C) += snd-soc-tlv320aic23-i2c.o +obj-$(CONFIG_SND_SOC_TLV320AIC23_SPI) += snd-soc-tlv320aic23-spi.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TLV320AIC32X4) += snd-soc-tlv320aic32x4.o diff --git a/sound/soc/codecs/tlv320aic23-i2c.c b/sound/soc/codecs/tlv320aic23-i2c.c new file mode 100644 index 000000000000..20fc46092c2c --- /dev/null +++ b/sound/soc/codecs/tlv320aic23-i2c.c @@ -0,0 +1,59 @@ +/* + * ALSA SoC TLV320AIC23 codec driver I2C interface + * + * Author: Arun KS, + * Copyright: (C) 2008 Mistral Solutions Pvt Ltd., + * + * Based on sound/soc/codecs/wm8731.c by Richard Purdie + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include + +#include "tlv320aic23.h" + +static int tlv320aic23_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *i2c_id) +{ + struct regmap *regmap; + + if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA)) + return -EINVAL; + + regmap = devm_regmap_init_i2c(i2c, &tlv320aic23_regmap); + return tlv320aic23_probe(&i2c->dev, regmap); +} + +static int __exit tlv320aic23_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + return 0; +} + +static const struct i2c_device_id tlv320aic23_id[] = { + {"tlv320aic23", 0}, + {} +}; + +MODULE_DEVICE_TABLE(i2c, tlv320aic23_id); + +static struct i2c_driver tlv320aic23_i2c_driver = { + .driver = { + .name = "tlv320aic23-codec", + }, + .probe = tlv320aic23_i2c_probe, + .remove = __exit_p(tlv320aic23_i2c_remove), + .id_table = tlv320aic23_id, +}; + +module_i2c_driver(tlv320aic23_i2c_driver); + +MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver I2C"); +MODULE_AUTHOR("Arun KS "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic23-spi.c b/sound/soc/codecs/tlv320aic23-spi.c new file mode 100644 index 000000000000..585aea436c6a --- /dev/null +++ b/sound/soc/codecs/tlv320aic23-spi.c @@ -0,0 +1,57 @@ +/* + * ALSA SoC TLV320AIC23 codec driver SPI interface + * + * Author: Arun KS, + * Copyright: (C) 2008 Mistral Solutions Pvt Ltd., + * + * Based on sound/soc/codecs/wm8731.c by Richard Purdie + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include + +#include "tlv320aic23.h" + +static int aic23_spi_probe(struct spi_device *spi) +{ + int ret; + struct regmap *regmap; + + dev_dbg(&spi->dev, "probing tlv320aic23 spi device\n"); + + spi->bits_per_word = 16; + spi->mode = SPI_MODE_0; + ret = spi_setup(spi); + if (ret < 0) + return ret; + + regmap = devm_regmap_init_spi(spi, &tlv320aic23_regmap); + return tlv320aic23_probe(&spi->dev, regmap); +} + +static int aic23_spi_remove(struct spi_device *spi) +{ + snd_soc_unregister_codec(&spi->dev); + return 0; +} + +static struct spi_driver aic23_spi = { + .driver = { + .name = "tlv320aic23", + .owner = THIS_MODULE, + }, + .probe = aic23_spi_probe, + .remove = aic23_spi_remove, +}; + +module_spi_driver(aic23_spi); + +MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver SPI"); +MODULE_AUTHOR("Arun KS "); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 139f11f4dd8b..ab369ae76b8c 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -23,7 +23,6 @@ #include #include #include -#include #include #include #include @@ -51,7 +50,7 @@ static const struct reg_default tlv320aic23_reg[] = { { 9, 0x0000 }, }; -static const struct regmap_config tlv320aic23_regmap = { +const struct regmap_config tlv320aic23_regmap = { .reg_bits = 7, .val_bits = 9, @@ -557,7 +556,7 @@ static int tlv320aic23_resume(struct snd_soc_codec *codec) return 0; } -static int tlv320aic23_probe(struct snd_soc_codec *codec) +static int tlv320aic23_codec_probe(struct snd_soc_codec *codec) { int ret; @@ -604,7 +603,7 @@ static int tlv320aic23_remove(struct snd_soc_codec *codec) } static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = { - .probe = tlv320aic23_probe, + .probe = tlv320aic23_codec_probe, .remove = tlv320aic23_remove, .suspend = tlv320aic23_suspend, .resume = tlv320aic23_resume, @@ -617,57 +616,25 @@ static struct snd_soc_codec_driver soc_codec_dev_tlv320aic23 = { .num_dapm_routes = ARRAY_SIZE(tlv320aic23_intercon), }; -/* - * If the i2c layer weren't so broken, we could pass this kind of data - * around - */ -static int tlv320aic23_codec_probe(struct i2c_client *i2c, - const struct i2c_device_id *i2c_id) +int tlv320aic23_probe(struct device *dev, struct regmap *regmap) { struct aic23 *aic23; - int ret; - if (!i2c_check_functionality(i2c->adapter, I2C_FUNC_SMBUS_BYTE_DATA)) - return -EINVAL; + if (IS_ERR(regmap)) + return PTR_ERR(regmap); - aic23 = devm_kzalloc(&i2c->dev, sizeof(struct aic23), GFP_KERNEL); + aic23 = devm_kzalloc(dev, sizeof(struct aic23), GFP_KERNEL); if (aic23 == NULL) return -ENOMEM; - aic23->regmap = devm_regmap_init_i2c(i2c, &tlv320aic23_regmap); - if (IS_ERR(aic23->regmap)) - return PTR_ERR(aic23->regmap); + aic23->regmap = regmap; - i2c_set_clientdata(i2c, aic23); + dev_set_drvdata(dev, aic23); - ret = snd_soc_register_codec(&i2c->dev, - &soc_codec_dev_tlv320aic23, &tlv320aic23_dai, 1); - return ret; -} -static int __exit tlv320aic23_i2c_remove(struct i2c_client *i2c) -{ - snd_soc_unregister_codec(&i2c->dev); - return 0; + return snd_soc_register_codec(dev, &soc_codec_dev_tlv320aic23, + &tlv320aic23_dai, 1); } -static const struct i2c_device_id tlv320aic23_id[] = { - {"tlv320aic23", 0}, - {} -}; - -MODULE_DEVICE_TABLE(i2c, tlv320aic23_id); - -static struct i2c_driver tlv320aic23_i2c_driver = { - .driver = { - .name = "tlv320aic23-codec", - }, - .probe = tlv320aic23_codec_probe, - .remove = __exit_p(tlv320aic23_i2c_remove), - .id_table = tlv320aic23_id, -}; - -module_i2c_driver(tlv320aic23_i2c_driver); - MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver"); MODULE_AUTHOR("Arun KS "); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic23.h b/sound/soc/codecs/tlv320aic23.h index e804120bd3da..3a7235a04a89 100644 --- a/sound/soc/codecs/tlv320aic23.h +++ b/sound/soc/codecs/tlv320aic23.h @@ -12,6 +12,12 @@ #ifndef _TLV320AIC23_H #define _TLV320AIC23_H +struct device; +struct regmap_config; + +extern const struct regmap_config tlv320aic23_regmap; +int tlv320aic23_probe(struct device *dev, struct regmap *regmap); + /* Codec TLV320AIC23 */ #define TLV320AIC23_LINVOL 0x00 #define TLV320AIC23_RINVOL 0x01 -- cgit v1.2.3 From 22066226b50e40591d67aef1d5525abce7515df2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 7 Mar 2014 11:44:08 +0800 Subject: ASoC: pcm512x: Split out bus drivers Move to the new style of defining the bus interfaces in separate modules in order to simplify dependencies. Signed-off-by: Mark Brown --- sound/soc/codecs/Kconfig | 18 +++++- sound/soc/codecs/Makefile | 4 ++ sound/soc/codecs/pcm512x-i2c.c | 71 ++++++++++++++++++++++ sound/soc/codecs/pcm512x-spi.c | 69 +++++++++++++++++++++ sound/soc/codecs/pcm512x.c | 134 +++-------------------------------------- sound/soc/codecs/pcm512x.h | 9 +++ 6 files changed, 176 insertions(+), 129 deletions(-) create mode 100644 sound/soc/codecs/pcm512x-i2c.c create mode 100644 sound/soc/codecs/pcm512x-spi.c (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index fa47a8336be3..cebe3ceef4bd 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -59,7 +59,8 @@ config SND_SOC_ALL_CODECS select SND_SOC_PCM1681 if I2C select SND_SOC_PCM1792A if SPI_MASTER select SND_SOC_PCM3008 - select SND_SOC_PCM512x if SND_SOC_I2C_AND_SPI + select SND_SOC_PCM512x_I2C if I2C + select SND_SOC_PCM512x_SPI if SPI_MASTER select SND_SOC_RT5631 if I2C select SND_SOC_RT5640 if I2C select SND_SOC_SGTL5000 if I2C @@ -315,8 +316,19 @@ config SND_SOC_PCM3008 tristate config SND_SOC_PCM512x - tristate "Texas Instruments PCM512x CODECs" - select REGMAP + tristate + +config SND_SOC_PCM512x_I2C + tristate "Texas Instruments PCM512x CODECs - I2C" + depends on I2C + select SND_SOC_PCM512x + select REGMAP_I2C + +config SND_SOC_PCM512x_SPI + tristate "Texas Instruments PCM512x CODECs - SPI" + depends on SPI_MASTER + select SND_SOC_PCM512x + select REGMAP_SPI config SND_SOC_RT5631 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index d3b536fc075d..c1191c05c88e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -47,6 +47,8 @@ snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm1792a-codec-objs := pcm1792a.o snd-soc-pcm3008-objs := pcm3008.o snd-soc-pcm512x-objs := pcm512x.o +snd-soc-pcm512x-i2c-objs := pcm512x-i2c.o +snd-soc-pcm512x-spi-objs := pcm512x-spi.o snd-soc-rt5631-objs := rt5631.o snd-soc-rt5640-objs := rt5640.o snd-soc-sgtl5000-objs := sgtl5000.o @@ -181,6 +183,8 @@ obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o obj-$(CONFIG_SND_SOC_PCM3008) += snd-soc-pcm3008.o obj-$(CONFIG_SND_SOC_PCM512x) += snd-soc-pcm512x.o +obj-$(CONFIG_SND_SOC_PCM512x_I2C) += snd-soc-pcm512x-i2c.o +obj-$(CONFIG_SND_SOC_PCM512x_SPI) += snd-soc-pcm512x-spi.o obj-$(CONFIG_SND_SOC_RT5631) += snd-soc-rt5631.o obj-$(CONFIG_SND_SOC_RT5640) += snd-soc-rt5640.o obj-$(CONFIG_SND_SOC_SGTL5000) += snd-soc-sgtl5000.o diff --git a/sound/soc/codecs/pcm512x-i2c.c b/sound/soc/codecs/pcm512x-i2c.c new file mode 100644 index 000000000000..4d62230bd378 --- /dev/null +++ b/sound/soc/codecs/pcm512x-i2c.c @@ -0,0 +1,71 @@ +/* + * Driver for the PCM512x CODECs + * + * Author: Mark Brown + * Copyright 2014 Linaro Ltd + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include +#include +#include + +#include "pcm512x.h" + +static int pcm512x_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct regmap *regmap; + + regmap = devm_regmap_init_i2c(i2c, &pcm512x_regmap); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return pcm512x_probe(&i2c->dev, regmap); +} + +static int pcm512x_i2c_remove(struct i2c_client *i2c) +{ + pcm512x_remove(&i2c->dev); + return 0; +} + +static const struct i2c_device_id pcm512x_i2c_id[] = { + { "pcm5121", }, + { "pcm5122", }, + { } +}; +MODULE_DEVICE_TABLE(i2c, pcm512x_i2c_id); + +static const struct of_device_id pcm512x_of_match[] = { + { .compatible = "ti,pcm5121", }, + { .compatible = "ti,pcm5122", }, + { } +}; +MODULE_DEVICE_TABLE(of, pcm512x_of_match); + +static struct i2c_driver pcm512x_i2c_driver = { + .probe = pcm512x_i2c_probe, + .remove = pcm512x_i2c_remove, + .id_table = pcm512x_i2c_id, + .driver = { + .name = "pcm512x", + .owner = THIS_MODULE, + .of_match_table = pcm512x_of_match, + .pm = &pcm512x_pm_ops, + }, +}; + +module_i2c_driver(pcm512x_i2c_driver); + +MODULE_DESCRIPTION("ASoC PCM512x codec driver - I2C"); +MODULE_AUTHOR("Mark Brown "); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/pcm512x-spi.c b/sound/soc/codecs/pcm512x-spi.c new file mode 100644 index 000000000000..f297058c0038 --- /dev/null +++ b/sound/soc/codecs/pcm512x-spi.c @@ -0,0 +1,69 @@ +/* + * Driver for the PCM512x CODECs + * + * Author: Mark Brown + * Copyright 2014 Linaro Ltd + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + */ + +#include +#include +#include + +#include "pcm512x.h" + +static int pcm512x_spi_probe(struct spi_device *spi) +{ + struct regmap *regmap; + int ret; + + regmap = devm_regmap_init_spi(spi, &pcm512x_regmap); + if (IS_ERR(regmap)) { + ret = PTR_ERR(regmap); + return ret; + } + + return pcm512x_probe(&spi->dev, regmap); +} + +static int pcm512x_spi_remove(struct spi_device *spi) +{ + pcm512x_remove(&spi->dev); + return 0; +} + +static const struct spi_device_id pcm512x_spi_id[] = { + { "pcm5121", }, + { "pcm5122", }, + { }, +}; +MODULE_DEVICE_TABLE(spi, pcm512x_spi_id); + +static const struct of_device_id pcm512x_of_match[] = { + { .compatible = "ti,pcm5121", }, + { .compatible = "ti,pcm5122", }, + { } +}; +MODULE_DEVICE_TABLE(of, pcm512x_of_match); + +static struct spi_driver pcm512x_spi_driver = { + .probe = pcm512x_spi_probe, + .remove = pcm512x_spi_remove, + .id_table = pcm512x_spi_id, + .driver = { + .name = "pcm512x", + .owner = THIS_MODULE, + .of_match_table = pcm512x_of_match, + .pm = &pcm512x_pm_ops, + }, +}; + +module_spi_driver(pcm512x_spi_driver); diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 3a0bbb6ab242..0c907051cc7b 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -18,11 +18,9 @@ #include #include #include -#include #include #include #include -#include #include #include #include @@ -375,7 +373,7 @@ static const struct regmap_range_cfg pcm512x_range = { .window_start = 0, .window_len = 0x100, }; -static const struct regmap_config pcm512x_regmap = { +const struct regmap_config pcm512x_regmap = { .reg_bits = 8, .val_bits = 8, @@ -390,15 +388,9 @@ static const struct regmap_config pcm512x_regmap = { .num_reg_defaults = ARRAY_SIZE(pcm512x_reg_defaults), .cache_type = REGCACHE_RBTREE, }; +EXPORT_SYMBOL_GPL(pcm512x_regmap); -static const struct of_device_id pcm512x_of_match[] = { - { .compatible = "ti,pcm5121", }, - { .compatible = "ti,pcm5122", }, - { } -}; -MODULE_DEVICE_TABLE(of, pcm512x_of_match); - -static int pcm512x_probe(struct device *dev, struct regmap *regmap) +int pcm512x_probe(struct device *dev, struct regmap *regmap) { struct pcm512x_priv *pcm512x; int i, ret; @@ -510,8 +502,9 @@ err: pcm512x->supplies); return ret; } +EXPORT_SYMBOL_GPL(pcm512x_probe); -static void pcm512x_remove(struct device *dev) +void pcm512x_remove(struct device *dev) { struct pcm512x_priv *pcm512x = dev_get_drvdata(dev); @@ -522,6 +515,7 @@ static void pcm512x_remove(struct device *dev) regulator_bulk_disable(ARRAY_SIZE(pcm512x->supplies), pcm512x->supplies); } +EXPORT_SYMBOL_GPL(pcm512x_remove); static int pcm512x_suspend(struct device *dev) { @@ -585,122 +579,10 @@ static int pcm512x_resume(struct device *dev) return 0; } -static const struct dev_pm_ops pcm512x_pm_ops = { +const struct dev_pm_ops pcm512x_pm_ops = { SET_RUNTIME_PM_OPS(pcm512x_suspend, pcm512x_resume, NULL) }; - -#if IS_ENABLED(CONFIG_I2C) -static int pcm512x_i2c_probe(struct i2c_client *i2c, - const struct i2c_device_id *id) -{ - struct regmap *regmap; - - regmap = devm_regmap_init_i2c(i2c, &pcm512x_regmap); - if (IS_ERR(regmap)) - return PTR_ERR(regmap); - - return pcm512x_probe(&i2c->dev, regmap); -} - -static int pcm512x_i2c_remove(struct i2c_client *i2c) -{ - pcm512x_remove(&i2c->dev); - return 0; -} - -static const struct i2c_device_id pcm512x_i2c_id[] = { - { "pcm5121", }, - { "pcm5122", }, - { } -}; -MODULE_DEVICE_TABLE(i2c, pcm512x_i2c_id); - -static struct i2c_driver pcm512x_i2c_driver = { - .probe = pcm512x_i2c_probe, - .remove = pcm512x_i2c_remove, - .id_table = pcm512x_i2c_id, - .driver = { - .name = "pcm512x", - .owner = THIS_MODULE, - .of_match_table = pcm512x_of_match, - .pm = &pcm512x_pm_ops, - }, -}; -#endif - -#if defined(CONFIG_SPI_MASTER) -static int pcm512x_spi_probe(struct spi_device *spi) -{ - struct regmap *regmap; - int ret; - - regmap = devm_regmap_init_spi(spi, &pcm512x_regmap); - if (IS_ERR(regmap)) { - ret = PTR_ERR(regmap); - return ret; - } - - return pcm512x_probe(&spi->dev, regmap); -} - -static int pcm512x_spi_remove(struct spi_device *spi) -{ - pcm512x_remove(&spi->dev); - return 0; -} - -static const struct spi_device_id pcm512x_spi_id[] = { - { "pcm5121", }, - { "pcm5122", }, - { }, -}; -MODULE_DEVICE_TABLE(spi, pcm512x_spi_id); - -static struct spi_driver pcm512x_spi_driver = { - .probe = pcm512x_spi_probe, - .remove = pcm512x_spi_remove, - .id_table = pcm512x_spi_id, - .driver = { - .name = "pcm512x", - .owner = THIS_MODULE, - .of_match_table = pcm512x_of_match, - .pm = &pcm512x_pm_ops, - }, -}; -#endif - -static int __init pcm512x_modinit(void) -{ - int ret = 0; - -#if IS_ENABLED(CONFIG_I2C) - ret = i2c_add_driver(&pcm512x_i2c_driver); - if (ret) { - printk(KERN_ERR "Failed to register pcm512x I2C driver: %d\n", - ret); - } -#endif -#if defined(CONFIG_SPI_MASTER) - ret = spi_register_driver(&pcm512x_spi_driver); - if (ret != 0) { - printk(KERN_ERR "Failed to register pcm512x SPI driver: %d\n", - ret); - } -#endif - return ret; -} -module_init(pcm512x_modinit); - -static void __exit pcm512x_exit(void) -{ -#if IS_ENABLED(CONFIG_I2C) - i2c_del_driver(&pcm512x_i2c_driver); -#endif -#if defined(CONFIG_SPI_MASTER) - spi_unregister_driver(&pcm512x_spi_driver); -#endif -} -module_exit(pcm512x_exit); +EXPORT_SYMBOL_GPL(pcm512x_pm_ops); MODULE_DESCRIPTION("ASoC PCM512x codec driver"); MODULE_AUTHOR("Mark Brown "); diff --git a/sound/soc/codecs/pcm512x.h b/sound/soc/codecs/pcm512x.h index ac4a52c9ccf4..6ee76aaca09a 100644 --- a/sound/soc/codecs/pcm512x.h +++ b/sound/soc/codecs/pcm512x.h @@ -17,6 +17,9 @@ #ifndef _SND_SOC_PCM512X #define _SND_SOC_PCM512X +#include +#include + #define PCM512x_VIRT_BASE 0x100 #define PCM512x_PAGE_LEN 0x100 #define PCM512x_PAGE_BASE(n) (PCM512x_VIRT_BASE + (PCM512x_PAGE_LEN * n)) @@ -159,4 +162,10 @@ #define PCM512x_AGBR_SHIFT 0 #define PCM512x_AGBL_SHIFT 4 +extern const struct dev_pm_ops pcm512x_pm_ops; +extern const struct regmap_config pcm512x_regmap; + +int pcm512x_probe(struct device *dev, struct regmap *regmap); +void pcm512x_remove(struct device *dev); + #endif -- cgit v1.2.3 From e97db9abf99280e6ff7d9b339dd2ca4846ce2eea Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 7 Mar 2014 11:43:04 +0800 Subject: ASoC: pcm512x: Fix duplicate const warning Signed-off-by: Mark Brown --- sound/soc/codecs/pcm512x.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/pcm512x.c b/sound/soc/codecs/pcm512x.c index 0c907051cc7b..4b4c0c7bb918 100644 --- a/sound/soc/codecs/pcm512x.c +++ b/sound/soc/codecs/pcm512x.c @@ -201,10 +201,10 @@ static const unsigned int pcm512x_dsp_program_values[] = { 7, }; -static const SOC_VALUE_ENUM_SINGLE_DECL(pcm512x_dsp_program, - PCM512x_DSP_PROGRAM, 0, 0x1f, - pcm512x_dsp_program_texts, - pcm512x_dsp_program_values); +static SOC_VALUE_ENUM_SINGLE_DECL(pcm512x_dsp_program, + PCM512x_DSP_PROGRAM, 0, 0x1f, + pcm512x_dsp_program_texts, + pcm512x_dsp_program_values); static const char * const pcm512x_clk_missing_text[] = { "1s", "2s", "3s", "4s", "5s", "6s", "7s", "8s" -- cgit v1.2.3 From 40423285a10e317b8e89e430779633eaef0b4add Mon Sep 17 00:00:00 2001 From: Max Filippov Date: Sat, 8 Mar 2014 13:31:06 +0400 Subject: ASoC: tlv320aic23: add missing EXPORT_SYMBOLs This fixes the following build errors when aic23 is configured as module: >> ERROR: "tlv320aic23_probe" >> [sound/soc/codecs/snd-soc-tlv320aic23-i2c.ko] undefined! >> ERROR: "tlv320aic23_regmap" >> [sound/soc/codecs/snd-soc-tlv320aic23-i2c.ko] undefined! Signed-off-by: Max Filippov Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic23.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index ab369ae76b8c..7b4cfef232ea 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -59,6 +59,7 @@ const struct regmap_config tlv320aic23_regmap = { .num_reg_defaults = ARRAY_SIZE(tlv320aic23_reg), .cache_type = REGCACHE_RBTREE, }; +EXPORT_SYMBOL(tlv320aic23_regmap); static const char *rec_src_text[] = { "Line", "Mic" }; static const char *deemph_text[] = {"None", "32Khz", "44.1Khz", "48Khz"}; @@ -634,6 +635,7 @@ int tlv320aic23_probe(struct device *dev, struct regmap *regmap) return snd_soc_register_codec(dev, &soc_codec_dev_tlv320aic23, &tlv320aic23_dai, 1); } +EXPORT_SYMBOL(tlv320aic23_probe); MODULE_DESCRIPTION("ASoC TLV320AIC23 codec driver"); MODULE_AUTHOR("Arun KS "); -- cgit v1.2.3 From da28ed585b26dc6eb0c8d897a9b842a86dd6a659 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 7 Mar 2014 16:34:17 +0000 Subject: ASoC: arizona: An OUTDIV of 1 is not valid, avoid this One is not a valid value for the OUTDIV start searching at 2 instead. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index e4295fee8f13..d90804686e4e 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1406,7 +1406,7 @@ static int arizona_calc_fll(struct arizona_fll *fll, Fref /= div; /* Fvco should be over the targt; don't check the upper bound */ - div = 1; + div = 2; while (Fout * div < 90000000 * fll->vco_mult) { div++; if (div > 7) { -- cgit v1.2.3 From 87383ac5a73ff34c60d3ea483bf24cabb27fb522 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 7 Mar 2014 16:34:18 +0000 Subject: ASoC: arizona: Add defines for FLL configuration constants Improve readability by adding defines for some of the constants associated with FLL configuration. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 16 +++++++++++----- 1 file changed, 11 insertions(+), 5 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index d90804686e4e..3d4408db075f 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -53,6 +53,12 @@ #define ARIZONA_AIF_RX_ENABLES 0x1A #define ARIZONA_AIF_FORCE_WRITE 0x1B +#define ARIZONA_FLL_MAX_FREF 13500000 +#define ARIZONA_FLL_MIN_FVCO 90000000 +#define ARIZONA_FLL_MAX_REFDIV 8 +#define ARIZONA_FLL_MIN_OUTDIV 2 +#define ARIZONA_FLL_MAX_OUTDIV 7 + #define arizona_fll_err(_fll, fmt, ...) \ dev_err(_fll->arizona->dev, "FLL%d: " fmt, _fll->id, ##__VA_ARGS__) #define arizona_fll_warn(_fll, fmt, ...) \ @@ -1390,11 +1396,11 @@ static int arizona_calc_fll(struct arizona_fll *fll, /* Fref must be <=13.5MHz */ div = 1; cfg->refdiv = 0; - while ((Fref / div) > 13500000) { + while ((Fref / div) > ARIZONA_FLL_MAX_FREF) { div *= 2; cfg->refdiv++; - if (div > 8) { + if (div > ARIZONA_FLL_MAX_REFDIV) { arizona_fll_err(fll, "Can't scale %dMHz in to <=13.5MHz\n", Fref); @@ -1406,10 +1412,10 @@ static int arizona_calc_fll(struct arizona_fll *fll, Fref /= div; /* Fvco should be over the targt; don't check the upper bound */ - div = 2; - while (Fout * div < 90000000 * fll->vco_mult) { + div = ARIZONA_FLL_MIN_OUTDIV; + while (Fout * div < ARIZONA_FLL_MIN_FVCO * fll->vco_mult) { div++; - if (div > 7) { + if (div > ARIZONA_FLL_MAX_OUTDIV) { arizona_fll_err(fll, "No FLL_OUTDIV for Fout=%uHz\n", Fout); return -EINVAL; -- cgit v1.2.3 From 61719db8141acde1a6293bbbddc733655defcc3c Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 7 Mar 2014 16:34:19 +0000 Subject: ASoC: arizona: Move set of OUTDIV in to arizona_apply_fll Since we know in arizona_apply_fll if we are setting the sync or ref path there is no need to set the outdiv seperately anymore. This patch moves this from arizona_enable_fll to arizona_apply_fll. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 28 ++++++++++++---------------- 1 file changed, 12 insertions(+), 16 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 3d4408db075f..9afd8c41d143 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1502,14 +1502,18 @@ static void arizona_apply_fll(struct arizona *arizona, unsigned int base, cfg->refdiv << ARIZONA_FLL1_CLK_REF_DIV_SHIFT | source << ARIZONA_FLL1_CLK_REF_SRC_SHIFT); - if (sync) - regmap_update_bits_async(arizona->regmap, base + 0x7, - ARIZONA_FLL1_GAIN_MASK, - cfg->gain << ARIZONA_FLL1_GAIN_SHIFT); - else - regmap_update_bits_async(arizona->regmap, base + 0x9, - ARIZONA_FLL1_GAIN_MASK, - cfg->gain << ARIZONA_FLL1_GAIN_SHIFT); + if (sync) { + regmap_update_bits(arizona->regmap, base + 0x7, + ARIZONA_FLL1_GAIN_MASK, + cfg->gain << ARIZONA_FLL1_GAIN_SHIFT); + } else { + regmap_update_bits(arizona->regmap, base + 0x5, + ARIZONA_FLL1_OUTDIV_MASK, + cfg->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); + regmap_update_bits(arizona->regmap, base + 0x9, + ARIZONA_FLL1_GAIN_MASK, + cfg->gain << ARIZONA_FLL1_GAIN_SHIFT); + } regmap_update_bits_async(arizona->regmap, base + 2, ARIZONA_FLL1_CTRL_UPD | ARIZONA_FLL1_N_MASK, @@ -1546,10 +1550,6 @@ static void arizona_enable_fll(struct arizona_fll *fll, */ if (fll->ref_src >= 0 && fll->ref_freq && fll->ref_src != fll->sync_src) { - regmap_update_bits_async(arizona->regmap, fll->base + 5, - ARIZONA_FLL1_OUTDIV_MASK, - ref->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); - arizona_apply_fll(arizona, fll->base, ref, fll->ref_src, false); if (fll->sync_src >= 0) { @@ -1558,10 +1558,6 @@ static void arizona_enable_fll(struct arizona_fll *fll, use_sync = true; } } else if (fll->sync_src >= 0) { - regmap_update_bits_async(arizona->regmap, fll->base + 5, - ARIZONA_FLL1_OUTDIV_MASK, - sync->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); - arizona_apply_fll(arizona, fll->base, sync, fll->sync_src, false); -- cgit v1.2.3 From 23f785a8bc33a98c4c384a653b9bff9c0cc3591d Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 7 Mar 2014 16:34:20 +0000 Subject: ASoC: arizona: Move calculation of FLL configuration Currently the FLL configuration is calculated before it is known which FLL path the configuration will be applied to. Newer versions of the IP have differences in the configuration required for each FLL path, which makes it complicated to calculate the FLL configuration in advance. This patch simply checks the validity of a requested input and output frequency before we know which FLL path they will be applied to and saves the actual calculation of the configuration until we know where the settings will be applied. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 79 ++++++++++++++++++++++++++-------------------- 1 file changed, 44 insertions(+), 35 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 9afd8c41d143..7398c69192cb 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1383,6 +1383,29 @@ struct arizona_fll_cfg { int gain; }; +static int arizona_validate_fll(struct arizona_fll *fll, + unsigned int Fref, + unsigned int Fout) +{ + unsigned int Fvco_min; + + if (Fref / ARIZONA_FLL_MAX_REFDIV > ARIZONA_FLL_MAX_FREF) { + arizona_fll_err(fll, + "Can't scale %dMHz in to <=13.5MHz\n", + Fref); + return -EINVAL; + } + + Fvco_min = ARIZONA_FLL_MIN_FVCO * fll->vco_mult; + if (Fout * ARIZONA_FLL_MAX_OUTDIV < Fvco_min) { + arizona_fll_err(fll, "No FLL_OUTDIV for Fout=%uHz\n", + Fout); + return -EINVAL; + } + + return 0; +} + static int arizona_calc_fll(struct arizona_fll *fll, struct arizona_fll_cfg *cfg, unsigned int Fref, @@ -1400,12 +1423,8 @@ static int arizona_calc_fll(struct arizona_fll *fll, div *= 2; cfg->refdiv++; - if (div > ARIZONA_FLL_MAX_REFDIV) { - arizona_fll_err(fll, - "Can't scale %dMHz in to <=13.5MHz\n", - Fref); + if (div > ARIZONA_FLL_MAX_REFDIV) return -EINVAL; - } } /* Apply the division for our remaining calculations */ @@ -1415,11 +1434,8 @@ static int arizona_calc_fll(struct arizona_fll *fll, div = ARIZONA_FLL_MIN_OUTDIV; while (Fout * div < ARIZONA_FLL_MIN_FVCO * fll->vco_mult) { div++; - if (div > ARIZONA_FLL_MAX_OUTDIV) { - arizona_fll_err(fll, "No FLL_OUTDIV for Fout=%uHz\n", - Fout); + if (div > ARIZONA_FLL_MAX_OUTDIV) return -EINVAL; - } } target = Fout * div / fll->vco_mult; cfg->outdiv = div; @@ -1536,13 +1552,12 @@ static bool arizona_is_enabled_fll(struct arizona_fll *fll) return reg & ARIZONA_FLL1_ENA; } -static void arizona_enable_fll(struct arizona_fll *fll, - struct arizona_fll_cfg *ref, - struct arizona_fll_cfg *sync) +static void arizona_enable_fll(struct arizona_fll *fll) { struct arizona *arizona = fll->arizona; int ret; bool use_sync = false; + struct arizona_fll_cfg cfg; /* * If we have both REFCLK and SYNCCLK then enable both, @@ -1550,15 +1565,21 @@ static void arizona_enable_fll(struct arizona_fll *fll, */ if (fll->ref_src >= 0 && fll->ref_freq && fll->ref_src != fll->sync_src) { - arizona_apply_fll(arizona, fll->base, ref, fll->ref_src, + arizona_calc_fll(fll, &cfg, fll->ref_freq, fll->fout); + + arizona_apply_fll(arizona, fll->base, &cfg, fll->ref_src, false); if (fll->sync_src >= 0) { - arizona_apply_fll(arizona, fll->base + 0x10, sync, + arizona_calc_fll(fll, &cfg, fll->sync_freq, fll->fout); + + arizona_apply_fll(arizona, fll->base + 0x10, &cfg, fll->sync_src, true); use_sync = true; } } else if (fll->sync_src >= 0) { - arizona_apply_fll(arizona, fll->base, sync, + arizona_calc_fll(fll, &cfg, fll->sync_freq, fll->fout); + + arizona_apply_fll(arizona, fll->base, &cfg, fll->sync_src, false); regmap_update_bits_async(arizona->regmap, fll->base + 0x11, @@ -1620,32 +1641,22 @@ static void arizona_disable_fll(struct arizona_fll *fll) int arizona_set_fll_refclk(struct arizona_fll *fll, int source, unsigned int Fref, unsigned int Fout) { - struct arizona_fll_cfg ref, sync; int ret; if (fll->ref_src == source && fll->ref_freq == Fref) return 0; - if (fll->fout) { - if (Fref > 0) { - ret = arizona_calc_fll(fll, &ref, Fref, fll->fout); - if (ret != 0) - return ret; - } - - if (fll->sync_src >= 0) { - ret = arizona_calc_fll(fll, &sync, fll->sync_freq, - fll->fout); - if (ret != 0) - return ret; - } + if (fll->fout && Fref > 0) { + ret = arizona_validate_fll(fll, Fref, fll->fout); + if (ret != 0) + return ret; } fll->ref_src = source; fll->ref_freq = Fref; if (fll->fout && Fref > 0) { - arizona_enable_fll(fll, &ref, &sync); + arizona_enable_fll(fll); } return 0; @@ -1655,7 +1666,6 @@ EXPORT_SYMBOL_GPL(arizona_set_fll_refclk); int arizona_set_fll(struct arizona_fll *fll, int source, unsigned int Fref, unsigned int Fout) { - struct arizona_fll_cfg ref, sync; int ret; if (fll->sync_src == source && @@ -1664,13 +1674,12 @@ int arizona_set_fll(struct arizona_fll *fll, int source, if (Fout) { if (fll->ref_src >= 0) { - ret = arizona_calc_fll(fll, &ref, fll->ref_freq, - Fout); + ret = arizona_validate_fll(fll, fll->ref_freq, Fout); if (ret != 0) return ret; } - ret = arizona_calc_fll(fll, &sync, Fref, Fout); + ret = arizona_validate_fll(fll, Fref, Fout); if (ret != 0) return ret; } @@ -1680,7 +1689,7 @@ int arizona_set_fll(struct arizona_fll *fll, int source, fll->fout = Fout; if (Fout) { - arizona_enable_fll(fll, &ref, &sync); + arizona_enable_fll(fll); } else { arizona_disable_fll(fll); } -- cgit v1.2.3 From 8ccefcd265b486186c94ea70c77511e7c570347d Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 7 Mar 2014 16:34:21 +0000 Subject: ASoC: arizona: Don't pass Fout into arizona_calc_fll As we now calculate the FLL configuration at a later stage in the process the fout member of the FLL structure will contain the desired Fout frequency so no need to pass this in seperately. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 15 +++++++-------- 1 file changed, 7 insertions(+), 8 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 7398c69192cb..7b1354ae337b 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1408,13 +1408,12 @@ static int arizona_validate_fll(struct arizona_fll *fll, static int arizona_calc_fll(struct arizona_fll *fll, struct arizona_fll_cfg *cfg, - unsigned int Fref, - unsigned int Fout) + unsigned int Fref) { unsigned int target, div, gcd_fll; int i, ratio; - arizona_fll_dbg(fll, "Fref=%u Fout=%u\n", Fref, Fout); + arizona_fll_dbg(fll, "Fref=%u Fout=%u\n", Fref, fll->fout); /* Fref must be <=13.5MHz */ div = 1; @@ -1432,12 +1431,12 @@ static int arizona_calc_fll(struct arizona_fll *fll, /* Fvco should be over the targt; don't check the upper bound */ div = ARIZONA_FLL_MIN_OUTDIV; - while (Fout * div < ARIZONA_FLL_MIN_FVCO * fll->vco_mult) { + while (fll->fout * div < ARIZONA_FLL_MIN_FVCO * fll->vco_mult) { div++; if (div > ARIZONA_FLL_MAX_OUTDIV) return -EINVAL; } - target = Fout * div / fll->vco_mult; + target = fll->fout * div / fll->vco_mult; cfg->outdiv = div; arizona_fll_dbg(fll, "Fvco=%dHz\n", target); @@ -1565,19 +1564,19 @@ static void arizona_enable_fll(struct arizona_fll *fll) */ if (fll->ref_src >= 0 && fll->ref_freq && fll->ref_src != fll->sync_src) { - arizona_calc_fll(fll, &cfg, fll->ref_freq, fll->fout); + arizona_calc_fll(fll, &cfg, fll->ref_freq); arizona_apply_fll(arizona, fll->base, &cfg, fll->ref_src, false); if (fll->sync_src >= 0) { - arizona_calc_fll(fll, &cfg, fll->sync_freq, fll->fout); + arizona_calc_fll(fll, &cfg, fll->sync_freq); arizona_apply_fll(arizona, fll->base + 0x10, &cfg, fll->sync_src, true); use_sync = true; } } else if (fll->sync_src >= 0) { - arizona_calc_fll(fll, &cfg, fll->sync_freq, fll->fout); + arizona_calc_fll(fll, &cfg, fll->sync_freq); arizona_apply_fll(arizona, fll->base, &cfg, fll->sync_src, false); -- cgit v1.2.3 From f641aec62c948c7754429136ad176824fbb97238 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 7 Mar 2014 16:34:22 +0000 Subject: ASoC: arizona: Calculate OUTDIV first OUTDIV will remain unchanged whilst the rest of the FLL configuration is calculated so do this first. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 7b1354ae337b..1f106abf1bb0 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1415,6 +1415,18 @@ static int arizona_calc_fll(struct arizona_fll *fll, arizona_fll_dbg(fll, "Fref=%u Fout=%u\n", Fref, fll->fout); + /* Fvco should be over the targt; don't check the upper bound */ + div = ARIZONA_FLL_MIN_OUTDIV; + while (fll->fout * div < ARIZONA_FLL_MIN_FVCO * fll->vco_mult) { + div++; + if (div > ARIZONA_FLL_MAX_OUTDIV) + return -EINVAL; + } + target = fll->fout * div / fll->vco_mult; + cfg->outdiv = div; + + arizona_fll_dbg(fll, "Fvco=%dHz\n", target); + /* Fref must be <=13.5MHz */ div = 1; cfg->refdiv = 0; @@ -1429,18 +1441,6 @@ static int arizona_calc_fll(struct arizona_fll *fll, /* Apply the division for our remaining calculations */ Fref /= div; - /* Fvco should be over the targt; don't check the upper bound */ - div = ARIZONA_FLL_MIN_OUTDIV; - while (fll->fout * div < ARIZONA_FLL_MIN_FVCO * fll->vco_mult) { - div++; - if (div > ARIZONA_FLL_MAX_OUTDIV) - return -EINVAL; - } - target = fll->fout * div / fll->vco_mult; - cfg->outdiv = div; - - arizona_fll_dbg(fll, "Fvco=%dHz\n", target); - /* Find an appropraite FLL_FRATIO and factor it out of the target */ for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) { if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) { -- cgit v1.2.3 From 5a3935c7643966e4172e7a704a48a35f9b4dc668 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 7 Mar 2014 16:34:23 +0000 Subject: ASoC: arizona: Calculate FLL gain last No part of the FLL calculation depends on the value determined for the gain but the gain does depend on other values. In preparation for future updates this patch moves the gain to be the last thing calculated. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 24 ++++++++++++------------ 1 file changed, 12 insertions(+), 12 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 1f106abf1bb0..219d1d54f536 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1455,18 +1455,6 @@ static int arizona_calc_fll(struct arizona_fll *fll, return -EINVAL; } - for (i = 0; i < ARRAY_SIZE(fll_gains); i++) { - if (fll_gains[i].min <= Fref && Fref <= fll_gains[i].max) { - cfg->gain = fll_gains[i].gain; - break; - } - } - if (i == ARRAY_SIZE(fll_gains)) { - arizona_fll_err(fll, "Unable to find gain for Fref=%uHz\n", - Fref); - return -EINVAL; - } - cfg->n = target / (ratio * Fref); if (target % (ratio * Fref)) { @@ -1490,6 +1478,18 @@ static int arizona_calc_fll(struct arizona_fll *fll, cfg->lambda >>= 1; } + for (i = 0; i < ARRAY_SIZE(fll_gains); i++) { + if (fll_gains[i].min <= Fref && Fref <= fll_gains[i].max) { + cfg->gain = fll_gains[i].gain; + break; + } + } + if (i == ARRAY_SIZE(fll_gains)) { + arizona_fll_err(fll, "Unable to find gain for Fref=%uHz\n", + Fref); + return -EINVAL; + } + arizona_fll_dbg(fll, "N=%x THETA=%x LAMBDA=%x\n", cfg->n, cfg->theta, cfg->lambda); arizona_fll_dbg(fll, "FRATIO=%x(%d) OUTDIV=%x REFCLK_DIV=%x\n", -- cgit v1.2.3 From d0800342bba71e7f11b2a67a29cf00c41ad1a3e4 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 7 Mar 2014 16:34:25 +0000 Subject: ASoC: arizona: Support new fratio encoding on the wm5110 rev D The reference clock path on newer IP FLLs requires a different configuration, and should avoid integer mode operation. This patch adds support for both the new encoding and updates the calculation. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 130 +++++++++++++++++++++++++++++++++++---------- 1 file changed, 101 insertions(+), 29 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 219d1d54f536..c3884861e8cb 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -53,8 +53,10 @@ #define ARIZONA_AIF_RX_ENABLES 0x1A #define ARIZONA_AIF_FORCE_WRITE 0x1B +#define ARIZONA_FLL_VCO_CORNER 141900000 #define ARIZONA_FLL_MAX_FREF 13500000 #define ARIZONA_FLL_MIN_FVCO 90000000 +#define ARIZONA_FLL_MAX_FRATIO 16 #define ARIZONA_FLL_MAX_REFDIV 8 #define ARIZONA_FLL_MIN_OUTDIV 2 #define ARIZONA_FLL_MAX_OUTDIV 7 @@ -1406,9 +1408,99 @@ static int arizona_validate_fll(struct arizona_fll *fll, return 0; } +static int arizona_find_fratio(unsigned int Fref, int *fratio) +{ + int i; + + /* Find an appropriate FLL_FRATIO */ + for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) { + if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) { + if (fratio) + *fratio = fll_fratios[i].fratio; + return fll_fratios[i].ratio; + } + } + + return -EINVAL; +} + +static int arizona_calc_fratio(struct arizona_fll *fll, + struct arizona_fll_cfg *cfg, + unsigned int target, + unsigned int Fref, bool sync) +{ + int init_ratio, ratio; + int refdiv, div; + + /* Fref must be <=13.5MHz, find initial refdiv */ + div = 1; + cfg->refdiv = 0; + while (Fref > ARIZONA_FLL_MAX_FREF) { + div *= 2; + Fref /= 2; + cfg->refdiv++; + + if (div > ARIZONA_FLL_MAX_REFDIV) + return -EINVAL; + } + + /* Find an appropriate FLL_FRATIO */ + init_ratio = arizona_find_fratio(Fref, &cfg->fratio); + if (init_ratio < 0) { + arizona_fll_err(fll, "Unable to find FRATIO for Fref=%uHz\n", + Fref); + return init_ratio; + } + + switch (fll->arizona->type) { + case WM5110: + if (fll->arizona->rev < 3 || sync) + return init_ratio; + break; + default: + return init_ratio; + } + + cfg->fratio = init_ratio - 1; + + /* Adjust FRATIO/refdiv to avoid integer mode if possible */ + refdiv = cfg->refdiv; + + while (div <= ARIZONA_FLL_MAX_REFDIV) { + for (ratio = init_ratio; ratio <= ARIZONA_FLL_MAX_FRATIO; + ratio++) { + if (target % (ratio * Fref)) { + cfg->refdiv = refdiv; + cfg->fratio = ratio - 1; + return ratio; + } + } + + for (ratio = init_ratio - 1; ratio >= 0; ratio--) { + if (ARIZONA_FLL_VCO_CORNER / (fll->vco_mult * ratio) < + Fref) + break; + + if (target % (ratio * Fref)) { + cfg->refdiv = refdiv; + cfg->fratio = ratio - 1; + return ratio; + } + } + + div *= 2; + Fref /= 2; + refdiv++; + init_ratio = arizona_find_fratio(Fref, NULL); + } + + arizona_fll_warn(fll, "Falling back to integer mode operation\n"); + return cfg->fratio + 1; +} + static int arizona_calc_fll(struct arizona_fll *fll, struct arizona_fll_cfg *cfg, - unsigned int Fref) + unsigned int Fref, bool sync) { unsigned int target, div, gcd_fll; int i, ratio; @@ -1427,33 +1519,13 @@ static int arizona_calc_fll(struct arizona_fll *fll, arizona_fll_dbg(fll, "Fvco=%dHz\n", target); - /* Fref must be <=13.5MHz */ - div = 1; - cfg->refdiv = 0; - while ((Fref / div) > ARIZONA_FLL_MAX_FREF) { - div *= 2; - cfg->refdiv++; - - if (div > ARIZONA_FLL_MAX_REFDIV) - return -EINVAL; - } + /* Find an appropriate FLL_FRATIO and refdiv */ + ratio = arizona_calc_fratio(fll, cfg, target, Fref, sync); + if (ratio < 0) + return ratio; /* Apply the division for our remaining calculations */ - Fref /= div; - - /* Find an appropraite FLL_FRATIO and factor it out of the target */ - for (i = 0; i < ARRAY_SIZE(fll_fratios); i++) { - if (fll_fratios[i].min <= Fref && Fref <= fll_fratios[i].max) { - cfg->fratio = fll_fratios[i].fratio; - ratio = fll_fratios[i].ratio; - break; - } - } - if (i == ARRAY_SIZE(fll_fratios)) { - arizona_fll_err(fll, "Unable to find FRATIO for Fref=%uHz\n", - Fref); - return -EINVAL; - } + Fref = Fref / (1 << cfg->refdiv); cfg->n = target / (ratio * Fref); @@ -1564,19 +1636,19 @@ static void arizona_enable_fll(struct arizona_fll *fll) */ if (fll->ref_src >= 0 && fll->ref_freq && fll->ref_src != fll->sync_src) { - arizona_calc_fll(fll, &cfg, fll->ref_freq); + arizona_calc_fll(fll, &cfg, fll->ref_freq, false); arizona_apply_fll(arizona, fll->base, &cfg, fll->ref_src, false); if (fll->sync_src >= 0) { - arizona_calc_fll(fll, &cfg, fll->sync_freq); + arizona_calc_fll(fll, &cfg, fll->sync_freq, true); arizona_apply_fll(arizona, fll->base + 0x10, &cfg, fll->sync_src, true); use_sync = true; } } else if (fll->sync_src >= 0) { - arizona_calc_fll(fll, &cfg, fll->sync_freq); + arizona_calc_fll(fll, &cfg, fll->sync_freq, false); arizona_apply_fll(arizona, fll->base, &cfg, fll->sync_src, false); -- cgit v1.2.3 From b46f2c5c0054153a6e5f76b6a943df5d837879f6 Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Fri, 7 Mar 2014 14:58:32 +0200 Subject: ASoC: tlv320aic32x4: Sort Makefile in alphabetic order The tlv320aic32x4 related lines were wrongly placed after tlv320aic3x lines. Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- sound/soc/codecs/Makefile | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index bc126764a44d..1deeb20fd411 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -64,8 +64,8 @@ snd-soc-stac9766-objs := stac9766.o snd-soc-tas5086-objs := tas5086.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o -snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o +snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-tlv320dac33-objs := tlv320dac33.o snd-soc-twl4030-objs := twl4030.o snd-soc-twl6040-objs := twl6040.o @@ -194,8 +194,8 @@ obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o -obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TLV320AIC32X4) += snd-soc-tlv320aic32x4.o +obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o obj-$(CONFIG_SND_SOC_TWL4030) += snd-soc-twl4030.o obj-$(CONFIG_SND_SOC_TWL6040) += snd-soc-twl6040.o -- cgit v1.2.3 From 5d6be5aa6becc750c5c2aa0ef8f7209ce19aa328 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Tue, 11 Mar 2014 12:43:20 +0800 Subject: ASoC: codec: Simplify ASoC probe code. For some CODEC drivers like who act as the MFDs children are ignored by this patch. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- sound/soc/codecs/ad193x.c | 10 +--------- sound/soc/codecs/adau1373.c | 7 ------- sound/soc/codecs/adav80x.c | 7 ------- sound/soc/codecs/ak4535.c | 9 --------- sound/soc/codecs/ak4641.c | 8 -------- sound/soc/codecs/ak4642.c | 8 -------- sound/soc/codecs/ak4671.c | 12 +----------- sound/soc/codecs/alc5623.c | 7 ------- sound/soc/codecs/alc5632.c | 8 -------- sound/soc/codecs/cs4270.c | 9 --------- sound/soc/codecs/cs42l51.c | 6 ------ sound/soc/codecs/cs42l52.c | 9 +-------- sound/soc/codecs/cs42l73.c | 11 +---------- sound/soc/codecs/da7210.c | 8 -------- sound/soc/codecs/da7213.c | 8 -------- sound/soc/codecs/da732x.c | 13 ++----------- sound/soc/codecs/da9055.c | 8 -------- sound/soc/codecs/isabelle.c | 16 ---------------- sound/soc/codecs/lm49453.c | 17 ----------------- sound/soc/codecs/max9768.c | 5 ----- sound/soc/codecs/max98088.c | 6 ------ sound/soc/codecs/max98090.c | 8 -------- sound/soc/codecs/max98095.c | 6 ------ sound/soc/codecs/max9850.c | 8 -------- sound/soc/codecs/ml26124.c | 10 ---------- sound/soc/codecs/rt5631.c | 9 --------- sound/soc/codecs/rt5640.c | 8 -------- sound/soc/codecs/sn95031.c | 2 -- sound/soc/codecs/ssm2518.c | 10 ---------- sound/soc/codecs/ssm2602.c | 7 ------- sound/soc/codecs/sta32x.c | 14 -------------- sound/soc/codecs/sta529.c | 10 ---------- sound/soc/codecs/tlv320aic23.c | 8 -------- sound/soc/codecs/tlv320aic26.c | 2 -- sound/soc/codecs/tlv320aic32x4.c | 2 -- sound/soc/codecs/tlv320aic3x.c | 6 ------ sound/soc/codecs/wm2000.c | 2 -- sound/soc/codecs/wm2200.c | 7 ------- sound/soc/codecs/wm5100.c | 7 ------- sound/soc/codecs/wm8510.c | 10 +--------- sound/soc/codecs/wm8523.c | 7 ------- sound/soc/codecs/wm8580.c | 6 ------ sound/soc/codecs/wm8711.c | 6 ------ sound/soc/codecs/wm8728.c | 11 +---------- sound/soc/codecs/wm8731.c | 7 ------- sound/soc/codecs/wm8737.c | 6 ------ sound/soc/codecs/wm8741.c | 6 ------ sound/soc/codecs/wm8750.c | 6 ------ sound/soc/codecs/wm8753.c | 7 ------- sound/soc/codecs/wm8770.c | 6 ------ sound/soc/codecs/wm8776.c | 6 ------ sound/soc/codecs/wm8804.c | 8 -------- sound/soc/codecs/wm8900.c | 8 +------- sound/soc/codecs/wm8903.c | 10 +--------- sound/soc/codecs/wm8904.c | 9 --------- sound/soc/codecs/wm8940.c | 6 ------ sound/soc/codecs/wm8955.c | 8 -------- sound/soc/codecs/wm8960.c | 6 ------ sound/soc/codecs/wm8961.c | 7 ------- sound/soc/codecs/wm8962.c | 7 ------- sound/soc/codecs/wm8971.c | 6 ------ sound/soc/codecs/wm8974.c | 6 ------ sound/soc/codecs/wm8978.c | 8 +------- sound/soc/codecs/wm8983.c | 6 ------ sound/soc/codecs/wm8985.c | 7 ------- sound/soc/codecs/wm8988.c | 8 -------- sound/soc/codecs/wm8990.c | 8 -------- sound/soc/codecs/wm8991.c | 8 -------- sound/soc/codecs/wm8993.c | 7 ------- sound/soc/codecs/wm8995.c | 7 ------- sound/soc/codecs/wm8996.c | 12 +----------- sound/soc/codecs/wm9081.c | 11 +---------- sound/soc/codecs/wm9090.c | 10 ---------- 73 files changed, 13 insertions(+), 562 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ad193x.c b/sound/soc/codecs/ad193x.c index 9381a767e75f..6844d0b2af68 100644 --- a/sound/soc/codecs/ad193x.c +++ b/sound/soc/codecs/ad193x.c @@ -322,14 +322,6 @@ static struct snd_soc_dai_driver ad193x_dai = { static int ad193x_codec_probe(struct snd_soc_codec *codec) { struct ad193x_priv *ad193x = snd_soc_codec_get_drvdata(codec); - int ret; - - codec->control_data = ad193x->regmap; - ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); - return ret; - } /* default setting for ad193x */ @@ -347,7 +339,7 @@ static int ad193x_codec_probe(struct snd_soc_codec *codec) regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL0, 0x99); /* mclk=24.576Mhz: 0x9D; mclk=12.288Mhz: 0x99 */ regmap_write(ad193x->regmap, AD193X_PLL_CLK_CTRL1, 0x04); - return ret; + return 0; } static struct snd_soc_codec_driver soc_codec_dev_ad193x = { diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index eb836ed5271f..db5c303dcdbc 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -1376,15 +1376,8 @@ static int adau1373_probe(struct snd_soc_codec *codec) struct adau1373_platform_data *pdata = codec->dev->platform_data; bool lineout_differential = false; unsigned int val; - int ret; int i; - ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); - if (ret) { - dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); - return ret; - } - if (pdata) { if (pdata->num_drc > ARRAY_SIZE(pdata->drc_setting)) return -EINVAL; diff --git a/sound/soc/codecs/adav80x.c b/sound/soc/codecs/adav80x.c index f78b27a7c461..8d79c3fe02dc 100644 --- a/sound/soc/codecs/adav80x.c +++ b/sound/soc/codecs/adav80x.c @@ -798,15 +798,8 @@ static struct snd_soc_dai_driver adav80x_dais[] = { static int adav80x_probe(struct snd_soc_codec *codec) { - int ret; struct adav80x *adav80x = snd_soc_codec_get_drvdata(codec); - ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); - if (ret) { - dev_err(codec->dev, "failed to set cache I/O: %d\n", ret); - return ret; - } - /* Force PLLs on for SYSCLK output */ snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL1"); snd_soc_dapm_force_enable_pin(&codec->dapm, "PLL2"); diff --git a/sound/soc/codecs/ak4535.c b/sound/soc/codecs/ak4535.c index 684fe910669f..30e297890fec 100644 --- a/sound/soc/codecs/ak4535.c +++ b/sound/soc/codecs/ak4535.c @@ -388,15 +388,6 @@ static int ak4535_resume(struct snd_soc_codec *codec) static int ak4535_probe(struct snd_soc_codec *codec) { - struct ak4535_priv *ak4535 = snd_soc_codec_get_drvdata(codec); - int ret; - - codec->control_data = ak4535->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } /* power on device */ ak4535_set_bias_level(codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/codecs/ak4641.c b/sound/soc/codecs/ak4641.c index 94cbe508dd37..a7b7d9858f8a 100644 --- a/sound/soc/codecs/ak4641.c +++ b/sound/soc/codecs/ak4641.c @@ -519,14 +519,6 @@ static int ak4641_resume(struct snd_soc_codec *codec) static int ak4641_probe(struct snd_soc_codec *codec) { - int ret; - - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* power on device */ ak4641_set_bias_level(codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index 1f646c6e90c6..92655cc189ae 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -465,14 +465,6 @@ static int ak4642_resume(struct snd_soc_codec *codec) static int ak4642_probe(struct snd_soc_codec *codec) { - int ret; - - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ak4642_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c index deb2b44669de..998fa0c5a0b9 100644 --- a/sound/soc/codecs/ak4671.c +++ b/sound/soc/codecs/ak4671.c @@ -613,17 +613,7 @@ static struct snd_soc_dai_driver ak4671_dai = { static int ak4671_probe(struct snd_soc_codec *codec) { - int ret; - - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - - ak4671_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - - return ret; + return ak4671_set_bias_level(codec, SND_SOC_BIAS_STANDBY); } static int ak4671_remove(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/alc5623.c b/sound/soc/codecs/alc5623.c index ed506253a914..09f7e773bafb 100644 --- a/sound/soc/codecs/alc5623.c +++ b/sound/soc/codecs/alc5623.c @@ -904,13 +904,6 @@ static int alc5623_probe(struct snd_soc_codec *codec) struct snd_soc_dapm_context *dapm = &codec->dapm; int ret; - codec->control_data = alc5623->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - alc5623_reset(codec); /* power on device */ diff --git a/sound/soc/codecs/alc5632.c b/sound/soc/codecs/alc5632.c index d885056ad8f2..ec071a6306ef 100644 --- a/sound/soc/codecs/alc5632.c +++ b/sound/soc/codecs/alc5632.c @@ -1063,14 +1063,6 @@ static int alc5632_probe(struct snd_soc_codec *codec) struct alc5632_priv *alc5632 = snd_soc_codec_get_drvdata(codec); int ret; - codec->control_data = alc5632->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* power on device */ alc5632_set_bias_level(codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index 83c835d9fd88..3920e6264948 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -506,15 +506,6 @@ static int cs4270_probe(struct snd_soc_codec *codec) struct cs4270_private *cs4270 = snd_soc_codec_get_drvdata(codec); int ret; - /* Tell ASoC what kind of I/O to use to read the registers. ASoC will - * then do the I2C transactions itself. - */ - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "failed to set cache I/O (ret=%i)\n", ret); - return ret; - } - /* Disable auto-mute. This feature appears to be buggy. In some * situations, auto-mute will not deactivate when it should, so we want * this feature disabled by default. An application (e.g. alsactl) can diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 3eab46020a30..a0c6060df4b4 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -489,12 +489,6 @@ static int cs42l51_probe(struct snd_soc_codec *codec) { int ret, reg; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* * DAC configuration * - Use signal processor diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 0bac6d5a4ac8..4bd59cea4922 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -1115,14 +1115,7 @@ static void cs42l52_free_beep(struct snd_soc_codec *codec) static int cs42l52_probe(struct snd_soc_codec *codec) { struct cs42l52_private *cs42l52 = snd_soc_codec_get_drvdata(codec); - int ret; - codec->control_data = cs42l52->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } regcache_cache_only(cs42l52->regmap, true); cs42l52_add_mic_controls(codec); @@ -1134,7 +1127,7 @@ static int cs42l52_probe(struct snd_soc_codec *codec) cs42l52->sysclk = CS42L52_DEFAULT_CLK; cs42l52->config.format = CS42L52_DEFAULT_FORMAT; - return ret; + return 0; } static int cs42l52_remove(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 549d5d6a3fef..b9aa009b5b01 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1348,17 +1348,8 @@ static int cs42l73_resume(struct snd_soc_codec *codec) static int cs42l73_probe(struct snd_soc_codec *codec) { - int ret; struct cs42l73_private *cs42l73 = snd_soc_codec_get_drvdata(codec); - codec->control_data = cs42l73->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - cs42l73_set_bias_level(codec, SND_SOC_BIAS_STANDBY); /* Set Charge Pump Frequency */ @@ -1371,7 +1362,7 @@ static int cs42l73_probe(struct snd_soc_codec *codec) cs42l73->mclksel = CS42L73_CLKID_MCLK1; cs42l73->mclk = 0; - return ret; + return 0; } static int cs42l73_remove(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index e62e294a8033..a5838ba69e4e 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -1071,17 +1071,9 @@ static struct snd_soc_dai_driver da7210_dai = { static int da7210_probe(struct snd_soc_codec *codec) { struct da7210_priv *da7210 = snd_soc_codec_get_drvdata(codec); - int ret; dev_info(codec->dev, "DA7210 Audio Codec %s\n", DA7210_VERSION); - codec->control_data = da7210->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - da7210->mclk_rate = 0; /* This will be set from set_sysclk() */ da7210->master = 0; /* This will be set from set_fmt() */ diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index 0c77e7ad7423..110f4dd1a89e 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -1384,17 +1384,9 @@ static int da7213_set_bias_level(struct snd_soc_codec *codec, static int da7213_probe(struct snd_soc_codec *codec) { - int ret; struct da7213_priv *da7213 = snd_soc_codec_get_drvdata(codec); struct da7213_platform_data *pdata = da7213->pdata; - codec->control_data = da7213->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* Default to using ALC auto offset calibration mode. */ snd_soc_update_bits(codec, DA7213_ALC_CTRL1, DA7213_ALC_CALIB_MODE_MAN, 0); diff --git a/sound/soc/codecs/da732x.c b/sound/soc/codecs/da732x.c index 8053e0e7f4a7..fdefc4bb8bc4 100644 --- a/sound/soc/codecs/da732x.c +++ b/sound/soc/codecs/da732x.c @@ -1516,23 +1516,14 @@ static int da732x_probe(struct snd_soc_codec *codec) { struct da732x_priv *da732x = snd_soc_codec_get_drvdata(codec); struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret = 0; da732x->codec = codec; dapm->idle_bias_off = false; - codec->control_data = da732x->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to register codec.\n"); - goto err; - } - da732x_set_bias_level(codec, SND_SOC_BIAS_STANDBY); -err: - return ret; + + return 0; } static int da732x_remove(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index 52b79a487ac7..f0a371dbaee3 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -1381,16 +1381,8 @@ static int da9055_set_bias_level(struct snd_soc_codec *codec, static int da9055_probe(struct snd_soc_codec *codec) { - int ret; struct da9055_priv *da9055 = snd_soc_codec_get_drvdata(codec); - codec->control_data = da9055->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* Enable all Gain Ramps */ snd_soc_update_bits(codec, DA9055_AUX_L_CTRL, DA9055_GAIN_RAMPING_EN, DA9055_GAIN_RAMPING_EN); diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index 5839048ec467..087b3cb2dcdf 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -1082,23 +1082,7 @@ static struct snd_soc_dai_driver isabelle_dai[] = { }, }; -static int isabelle_probe(struct snd_soc_codec *codec) -{ - int ret = 0; - - codec->control_data = dev_get_regmap(codec->dev, NULL); - - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - - return 0; -} - static struct snd_soc_codec_driver soc_codec_dev_isabelle = { - .probe = isabelle_probe, .set_bias_level = isabelle_set_bias_level, .controls = isabelle_snd_controls, .num_controls = ARRAY_SIZE(isabelle_snd_controls), diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index e19490cfb3a8..069cb0359932 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1409,22 +1409,6 @@ static int lm49453_resume(struct snd_soc_codec *codec) return 0; } -static int lm49453_probe(struct snd_soc_codec *codec) -{ - struct lm49453_priv *lm49453 = snd_soc_codec_get_drvdata(codec); - int ret = 0; - - codec->control_data = lm49453->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - - return 0; -} - /* power down chip */ static int lm49453_remove(struct snd_soc_codec *codec) { @@ -1433,7 +1417,6 @@ static int lm49453_remove(struct snd_soc_codec *codec) } static struct snd_soc_codec_driver soc_codec_dev_lm49453 = { - .probe = lm49453_probe, .remove = lm49453_remove, .suspend = lm49453_suspend, .resume = lm49453_resume, diff --git a/sound/soc/codecs/max9768.c b/sound/soc/codecs/max9768.c index 31f91560e9f6..ec481fc428c7 100644 --- a/sound/soc/codecs/max9768.c +++ b/sound/soc/codecs/max9768.c @@ -135,11 +135,6 @@ static int max9768_probe(struct snd_soc_codec *codec) struct max9768 *max9768 = snd_soc_codec_get_drvdata(codec); int ret; - codec->control_data = max9768->regmap; - ret = snd_soc_codec_set_cache_io(codec, 2, 6, SND_SOC_REGMAP); - if (ret) - return ret; - if (max9768->flags & MAX9768_FLAG_CLASSIC_PWM) { ret = snd_soc_write(codec, MAX9768_CTRL, MAX9768_CTRL_PWM); if (ret) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index ee660e2d3df3..64965005a41e 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1915,12 +1915,6 @@ static int max98088_probe(struct snd_soc_codec *codec) regcache_mark_dirty(max98088->regmap); - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* initialize private data */ max98088->sysclk = (unsigned)-1; diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 51f9b3d16b41..4ac3b67b2006 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2195,14 +2195,6 @@ static int max98090_probe(struct snd_soc_codec *codec) max98090->codec = codec; - codec->control_data = max98090->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* Reset the codec, the DSP core, and disable all interrupts */ max98090_reset(max98090); diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 3ba1170ebb53..2a9bfefb7d26 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -2234,12 +2234,6 @@ static int max98095_probe(struct snd_soc_codec *codec) struct i2c_client *client; int ret = 0; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* reset the codec, the DSP core, and disable all interrupts */ max98095_reset(codec); diff --git a/sound/soc/codecs/max9850.c b/sound/soc/codecs/max9850.c index 82757ebf0301..4fdf5aaa236f 100644 --- a/sound/soc/codecs/max9850.c +++ b/sound/soc/codecs/max9850.c @@ -312,14 +312,6 @@ static int max9850_resume(struct snd_soc_codec *codec) static int max9850_probe(struct snd_soc_codec *codec) { - int ret; - - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* enable zero-detect */ snd_soc_update_bits(codec, MAX9850_GENERAL_PURPOSE, 1, 1); /* enable slew-rate control */ diff --git a/sound/soc/codecs/ml26124.c b/sound/soc/codecs/ml26124.c index 185fa3bc3052..b9f21fe5a7dc 100644 --- a/sound/soc/codecs/ml26124.c +++ b/sound/soc/codecs/ml26124.c @@ -586,16 +586,6 @@ static int ml26124_resume(struct snd_soc_codec *codec) static int ml26124_probe(struct snd_soc_codec *codec) { - int ret; - struct ml26124_priv *priv = snd_soc_codec_get_drvdata(codec); - codec->control_data = priv->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* Software Reset */ snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 1); snd_soc_update_bits(codec, ML26124_SW_RST, 0x01, 0); diff --git a/sound/soc/codecs/rt5631.c b/sound/soc/codecs/rt5631.c index 912c9cbc2724..73fcd3cff327 100644 --- a/sound/soc/codecs/rt5631.c +++ b/sound/soc/codecs/rt5631.c @@ -1585,15 +1585,6 @@ static int rt5631_probe(struct snd_soc_codec *codec) { struct rt5631_priv *rt5631 = snd_soc_codec_get_drvdata(codec); unsigned int val; - int ret; - - codec->control_data = rt5631->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } val = rt5631_read_index(codec, RT5631_ADDA_MIXER_INTL_REG3); if (val & 0x0002) diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index a3fb41179636..074cff100f54 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1943,16 +1943,8 @@ static int rt5640_set_bias_level(struct snd_soc_codec *codec, static int rt5640_probe(struct snd_soc_codec *codec) { struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); - int ret; rt5640->codec = codec; - codec->control_data = rt5640->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } codec->dapm.idle_bias_off = 1; rt5640_set_bias_level(codec, SND_SOC_BIAS_OFF); diff --git a/sound/soc/codecs/sn95031.c b/sound/soc/codecs/sn95031.c index 13045f2af4d3..193760ec98ef 100644 --- a/sound/soc/codecs/sn95031.c +++ b/sound/soc/codecs/sn95031.c @@ -825,8 +825,6 @@ static int sn95031_codec_probe(struct snd_soc_codec *codec) { pr_debug("codec_probe called\n"); - snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); - /* PCM interface config * This sets the pcm rx slot conguration to max 6 slots * for max 4 dais (2 stereo and 2 mono) diff --git a/sound/soc/codecs/ssm2518.c b/sound/soc/codecs/ssm2518.c index cc8debce752f..cf4a6572f35b 100644 --- a/sound/soc/codecs/ssm2518.c +++ b/sound/soc/codecs/ssm2518.c @@ -648,16 +648,6 @@ static struct snd_soc_dai_driver ssm2518_dai = { static int ssm2518_probe(struct snd_soc_codec *codec) { - struct ssm2518 *ssm2518 = snd_soc_codec_get_drvdata(codec); - int ret; - - codec->control_data = ssm2518->regmap; - ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - return ssm2518_set_bias_level(codec, SND_SOC_BIAS_OFF); } diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index af76bbd1b24f..9fe4d485e699 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -573,13 +573,6 @@ static int ssm260x_probe(struct snd_soc_codec *codec) struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); int ret; - codec->control_data = ssm2602->regmap; - ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = regmap_write(ssm2602->regmap, SSM2602_RESET, 0); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset: %d\n", ret); diff --git a/sound/soc/codecs/sta32x.c b/sound/soc/codecs/sta32x.c index 06edb396e733..1cab6f62be7c 100644 --- a/sound/soc/codecs/sta32x.c +++ b/sound/soc/codecs/sta32x.c @@ -872,16 +872,6 @@ static int sta32x_probe(struct snd_soc_codec *codec) return ret; } - /* Tell ASoC what kind of I/O to use to read the registers. ASoC will - * then do the I2C transactions itself. - */ - codec->control_data = sta32x->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "failed to set cache I/O (ret=%i)\n", ret); - goto err; - } - /* Chip documentation explicitly requires that the reset values * of reserved register bits are left untouched. * Write the register default value to cache for reserved registers, @@ -946,10 +936,6 @@ static int sta32x_probe(struct snd_soc_codec *codec) regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); return 0; - -err: - regulator_bulk_disable(ARRAY_SIZE(sta32x->supplies), sta32x->supplies); - return ret; } static int sta32x_remove(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index 40c07be9b581..30aae61bfdb3 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -322,16 +322,6 @@ static struct snd_soc_dai_driver sta529_dai = { static int sta529_probe(struct snd_soc_codec *codec) { - struct sta529 *sta529 = snd_soc_codec_get_drvdata(codec); - int ret; - - codec->control_data = sta529->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } sta529_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 7b4cfef232ea..46b8a5073857 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -559,14 +559,6 @@ static int tlv320aic23_resume(struct snd_soc_codec *codec) static int tlv320aic23_codec_probe(struct snd_soc_codec *codec) { - int ret; - - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* Reset codec */ snd_soc_write(codec, TLV320AIC23_RESET, 0); diff --git a/sound/soc/codecs/tlv320aic26.c b/sound/soc/codecs/tlv320aic26.c index 94a658fa6d97..8037beabd030 100644 --- a/sound/soc/codecs/tlv320aic26.c +++ b/sound/soc/codecs/tlv320aic26.c @@ -295,8 +295,6 @@ static int aic26_probe(struct snd_soc_codec *codec) struct aic26 *aic26 = dev_get_drvdata(codec->dev); int ret, reg; - snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); - aic26->codec = codec; /* Reset the codec to power on defaults */ diff --git a/sound/soc/codecs/tlv320aic32x4.c b/sound/soc/codecs/tlv320aic32x4.c index c6bd7e75352d..1d9b117345a3 100644 --- a/sound/soc/codecs/tlv320aic32x4.c +++ b/sound/soc/codecs/tlv320aic32x4.c @@ -614,8 +614,6 @@ static int aic32x4_probe(struct snd_soc_codec *codec) struct aic32x4_priv *aic32x4 = snd_soc_codec_get_drvdata(codec); u32 tmp_reg; - snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (gpio_is_valid(aic32x4->rstn_gpio)) { ndelay(10); gpio_set_value(aic32x4->rstn_gpio, 1); diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 470fbfb4b386..b1835103e9b4 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1344,12 +1344,6 @@ static int aic3x_probe(struct snd_soc_codec *codec) INIT_LIST_HEAD(&aic3x->list); aic3x->codec = codec; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - for (i = 0; i < ARRAY_SIZE(aic3x->supplies); i++) { aic3x->disable_nb[i].nb.notifier_call = aic3x_regulator_event; aic3x->disable_nb[i].aic3x = aic3x; diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index 8ae50274ea8f..83a2c872925c 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -786,8 +786,6 @@ static int wm2000_probe(struct snd_soc_codec *codec) { struct wm2000_priv *wm2000 = dev_get_drvdata(codec->dev); - snd_soc_codec_set_cache_io(codec, 16, 8, SND_SOC_REGMAP); - /* This will trigger a transition to standby mode by default */ wm2000_anc_set_mode(wm2000); diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index 57ba315d0c84..5129d91a6da4 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1556,15 +1556,8 @@ static int wm2200_probe(struct snd_soc_codec *codec) int ret; wm2200->codec = codec; - codec->control_data = wm2200->regmap; codec->dapm.bias_level = SND_SOC_BIAS_OFF; - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = snd_soc_add_codec_controls(codec, wm_adsp1_fw_controls, 2); if (ret != 0) return ret; diff --git a/sound/soc/codecs/wm5100.c b/sound/soc/codecs/wm5100.c index 4e3e31aaf509..bac848f009e7 100644 --- a/sound/soc/codecs/wm5100.c +++ b/sound/soc/codecs/wm5100.c @@ -2337,13 +2337,6 @@ static int wm5100_probe(struct snd_soc_codec *codec) int ret, i; wm5100->codec = codec; - codec->control_data = wm5100->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } for (i = 0; i < ARRAY_SIZE(wm5100_dig_vu); i++) snd_soc_update_bits(codec, wm5100_dig_vu[i], WM5100_OUT_VU, diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index 7df7d4572755..1c1e328feeb8 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -589,20 +589,12 @@ static int wm8510_resume(struct snd_soc_codec *codec) static int wm8510_probe(struct snd_soc_codec *codec) { - int ret; - - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - printk(KERN_ERR "wm8510: failed to set cache I/O: %d\n", ret); - return ret; - } - wm8510_reset(codec); /* power on device */ wm8510_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return ret; + return 0; } /* power down chip */ diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 74d106dc7667..e6116aff0943 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -392,18 +392,11 @@ static int wm8523_resume(struct snd_soc_codec *codec) static int wm8523_probe(struct snd_soc_codec *codec) { struct wm8523_priv *wm8523 = snd_soc_codec_get_drvdata(codec); - int ret; wm8523->rate_constraint.list = &wm8523->rate_constraint_list[0]; wm8523->rate_constraint.count = ARRAY_SIZE(wm8523->rate_constraint_list); - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* Change some default settings - latch VU and enable ZC */ snd_soc_update_bits(codec, WM8523_DAC_GAINR, WM8523_DACR_VU, WM8523_DACR_VU); diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 318989acbbe5..7558c838193d 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -869,12 +869,6 @@ static int wm8580_probe(struct snd_soc_codec *codec) struct wm8580_priv *wm8580 = snd_soc_codec_get_drvdata(codec); int ret = 0; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = regulator_bulk_enable(ARRAY_SIZE(wm8580->supplies), wm8580->supplies); if (ret != 0) { diff --git a/sound/soc/codecs/wm8711.c b/sound/soc/codecs/wm8711.c index d99f948c513c..ef6cbc7ba489 100644 --- a/sound/soc/codecs/wm8711.c +++ b/sound/soc/codecs/wm8711.c @@ -367,12 +367,6 @@ static int wm8711_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = wm8711_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset\n"); diff --git a/sound/soc/codecs/wm8728.c b/sound/soc/codecs/wm8728.c index cd89033e84c0..bac7fc28fe71 100644 --- a/sound/soc/codecs/wm8728.c +++ b/sound/soc/codecs/wm8728.c @@ -228,19 +228,10 @@ static int wm8728_resume(struct snd_soc_codec *codec) static int wm8728_probe(struct snd_soc_codec *codec) { - int ret; - - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - printk(KERN_ERR "wm8728: failed to configure cache I/O: %d\n", - ret); - return ret; - } - /* power on device */ wm8728_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return ret; + return 0; } static int wm8728_remove(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 029720366ff8..2c95b633e7df 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -583,13 +583,6 @@ static int wm8731_probe(struct snd_soc_codec *codec) struct wm8731_priv *wm8731 = snd_soc_codec_get_drvdata(codec); int ret = 0, i; - codec->control_data = wm8731->regmap; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - for (i = 0; i < ARRAY_SIZE(wm8731->supplies); i++) wm8731->supplies[i].supply = wm8731_supply_names[i]; diff --git a/sound/soc/codecs/wm8737.c b/sound/soc/codecs/wm8737.c index 2f167a8ca01b..3693479b43d5 100644 --- a/sound/soc/codecs/wm8737.c +++ b/sound/soc/codecs/wm8737.c @@ -570,12 +570,6 @@ static int wm8737_probe(struct snd_soc_codec *codec) struct wm8737_priv *wm8737 = snd_soc_codec_get_drvdata(codec); int ret; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = regulator_bulk_enable(ARRAY_SIZE(wm8737->supplies), wm8737->supplies); if (ret != 0) { diff --git a/sound/soc/codecs/wm8741.c b/sound/soc/codecs/wm8741.c index 2895c8d3b5e4..ecf4fcfa99fd 100644 --- a/sound/soc/codecs/wm8741.c +++ b/sound/soc/codecs/wm8741.c @@ -429,12 +429,6 @@ static int wm8741_probe(struct snd_soc_codec *codec) goto err_get; } - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - goto err_enable; - } - ret = wm8741_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset\n"); diff --git a/sound/soc/codecs/wm8750.c b/sound/soc/codecs/wm8750.c index 78616a638a55..33990b63d214 100644 --- a/sound/soc/codecs/wm8750.c +++ b/sound/soc/codecs/wm8750.c @@ -702,12 +702,6 @@ static int wm8750_probe(struct snd_soc_codec *codec) { int ret; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - printk(KERN_ERR "wm8750: failed to set cache I/O: %d\n", ret); - return ret; - } - ret = wm8750_reset(codec); if (ret < 0) { printk(KERN_ERR "wm8750: failed to reset: %d\n", ret); diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index be85da93a268..0d1670b70702 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1471,13 +1471,6 @@ static int wm8753_probe(struct snd_soc_codec *codec) INIT_DELAYED_WORK(&codec->dapm.delayed_work, wm8753_work); - codec->control_data = wm8753->regmap; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = wm8753_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset: %d\n", ret); diff --git a/sound/soc/codecs/wm8770.c b/sound/soc/codecs/wm8770.c index 89a18d82f303..32e736320526 100644 --- a/sound/soc/codecs/wm8770.c +++ b/sound/soc/codecs/wm8770.c @@ -580,12 +580,6 @@ static int wm8770_probe(struct snd_soc_codec *codec) wm8770 = snd_soc_codec_get_drvdata(codec); wm8770->codec = codec; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = regulator_bulk_enable(ARRAY_SIZE(wm8770->supplies), wm8770->supplies); if (ret) { diff --git a/sound/soc/codecs/wm8776.c b/sound/soc/codecs/wm8776.c index ef8246725232..70952ceb278b 100644 --- a/sound/soc/codecs/wm8776.c +++ b/sound/soc/codecs/wm8776.c @@ -430,12 +430,6 @@ static int wm8776_probe(struct snd_soc_codec *codec) { int ret = 0; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = wm8776_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset: %d\n", ret); diff --git a/sound/soc/codecs/wm8804.c b/sound/soc/codecs/wm8804.c index 9bc8206a6807..448a943e5a1b 100644 --- a/sound/soc/codecs/wm8804.c +++ b/sound/soc/codecs/wm8804.c @@ -546,14 +546,6 @@ static int wm8804_probe(struct snd_soc_codec *codec) wm8804 = snd_soc_codec_get_drvdata(codec); - codec->control_data = wm8804->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); - return ret; - } - for (i = 0; i < ARRAY_SIZE(wm8804->supplies); i++) wm8804->supplies[i].supply = wm8804_supply_names[i]; diff --git a/sound/soc/codecs/wm8900.c b/sound/soc/codecs/wm8900.c index e98bc7038a08..637be63cd474 100644 --- a/sound/soc/codecs/wm8900.c +++ b/sound/soc/codecs/wm8900.c @@ -1178,13 +1178,7 @@ static int wm8900_resume(struct snd_soc_codec *codec) static int wm8900_probe(struct snd_soc_codec *codec) { - int ret = 0, reg; - - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } + int reg; reg = snd_soc_read(codec, WM8900_REG_ID); if (reg != 0x8900) { diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index eebcb1da3b7b..cf2f49fdb3bd 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -1897,21 +1897,13 @@ static void wm8903_free_gpio(struct wm8903_priv *wm8903) static int wm8903_probe(struct snd_soc_codec *codec) { struct wm8903_priv *wm8903 = snd_soc_codec_get_drvdata(codec); - int ret; wm8903->codec = codec; - codec->control_data = wm8903->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } /* power on device */ wm8903_set_bias_level(codec, SND_SOC_BIAS_STANDBY); - return ret; + return 0; } /* power down chip */ diff --git a/sound/soc/codecs/wm8904.c b/sound/soc/codecs/wm8904.c index 53bbfac6a83a..817dddc7f8a1 100644 --- a/sound/soc/codecs/wm8904.c +++ b/sound/soc/codecs/wm8904.c @@ -2047,9 +2047,6 @@ static void wm8904_handle_pdata(struct snd_soc_codec *codec) static int wm8904_probe(struct snd_soc_codec *codec) { struct wm8904_priv *wm8904 = snd_soc_codec_get_drvdata(codec); - int ret; - - codec->control_data = wm8904->regmap; switch (wm8904->devtype) { case WM8904: @@ -2063,12 +2060,6 @@ static int wm8904_probe(struct snd_soc_codec *codec) return -EINVAL; } - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - wm8904_handle_pdata(codec); wm8904_add_widgets(codec); diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c index b404c26c1753..1cdabaf07639 100644 --- a/sound/soc/codecs/wm8940.c +++ b/sound/soc/codecs/wm8940.c @@ -712,12 +712,6 @@ static int wm8940_probe(struct snd_soc_codec *codec) int ret; u16 reg; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = wm8940_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset\n"); diff --git a/sound/soc/codecs/wm8955.c b/sound/soc/codecs/wm8955.c index 82c8ba975720..a94d1858930a 100644 --- a/sound/soc/codecs/wm8955.c +++ b/sound/soc/codecs/wm8955.c @@ -896,14 +896,6 @@ static int wm8955_probe(struct snd_soc_codec *codec) struct wm8955_pdata *pdata = dev_get_platdata(codec->dev); int ret, i; - codec->control_data = wm8955->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - for (i = 0; i < ARRAY_SIZE(wm8955->supplies); i++) wm8955->supplies[i].supply = wm8955_supply_names[i]; diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index f156010e52bc..d04e9cad445c 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -976,12 +976,6 @@ static int wm8960_probe(struct snd_soc_codec *codec) wm8960->set_bias_level = wm8960_set_bias_level_capless; } - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = wm8960_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset\n"); diff --git a/sound/soc/codecs/wm8961.c b/sound/soc/codecs/wm8961.c index 900328e28a15..db84507c1ec5 100644 --- a/sound/soc/codecs/wm8961.c +++ b/sound/soc/codecs/wm8961.c @@ -836,15 +836,8 @@ static struct snd_soc_dai_driver wm8961_dai = { static int wm8961_probe(struct snd_soc_codec *codec) { struct snd_soc_dapm_context *dapm = &codec->dapm; - int ret = 0; u16 reg; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* Enable class W */ reg = snd_soc_read(codec, WM8961_CHARGE_PUMP_B); reg |= WM8961_CP_DYN_PWR_MASK; diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index 97db3b45b411..1d556c945bb2 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -3400,13 +3400,6 @@ static int wm8962_probe(struct snd_soc_codec *codec) bool dmicclk, dmicdat; wm8962->codec = codec; - codec->control_data = wm8962->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } wm8962->disable_nb[0].notifier_call = wm8962_regulator_event_0; wm8962->disable_nb[1].notifier_call = wm8962_regulator_event_1; diff --git a/sound/soc/codecs/wm8971.c b/sound/soc/codecs/wm8971.c index 67aba78a7ca5..09b7b4200221 100644 --- a/sound/soc/codecs/wm8971.c +++ b/sound/soc/codecs/wm8971.c @@ -648,12 +648,6 @@ static int wm8971_probe(struct snd_soc_codec *codec) int ret = 0; u16 reg; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - printk(KERN_ERR "wm8971: failed to set cache I/O: %d\n", ret); - return ret; - } - INIT_DELAYED_WORK(&codec->dapm.delayed_work, wm8971_work); wm8971_workq = create_workqueue("wm8971"); if (wm8971_workq == NULL) diff --git a/sound/soc/codecs/wm8974.c b/sound/soc/codecs/wm8974.c index 15f45c7bd833..ea0de269a472 100644 --- a/sound/soc/codecs/wm8974.c +++ b/sound/soc/codecs/wm8974.c @@ -593,12 +593,6 @@ static int wm8974_probe(struct snd_soc_codec *codec) { int ret = 0; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = wm8974_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset\n"); diff --git a/sound/soc/codecs/wm8978.c b/sound/soc/codecs/wm8978.c index d8fc531c0e59..13de3688e86f 100644 --- a/sound/soc/codecs/wm8978.c +++ b/sound/soc/codecs/wm8978.c @@ -975,19 +975,13 @@ static const int update_reg[] = { static int wm8978_probe(struct snd_soc_codec *codec) { struct wm8978_priv *wm8978 = snd_soc_codec_get_drvdata(codec); - int ret = 0, i; + int i; /* * Set default system clock to PLL, it is more precise, this is also the * default hardware setting */ wm8978->sysclk = WM8978_PLL; - codec->control_data = wm8978->regmap; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } /* * Set the update bit in all registers, that have one. This way all diff --git a/sound/soc/codecs/wm8983.c b/sound/soc/codecs/wm8983.c index aa41ba0dfff4..84aa09319ba1 100644 --- a/sound/soc/codecs/wm8983.c +++ b/sound/soc/codecs/wm8983.c @@ -1000,12 +1000,6 @@ static int wm8983_probe(struct snd_soc_codec *codec) int ret; int i; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); - return ret; - } - ret = snd_soc_write(codec, WM8983_SOFTWARE_RESET, 0); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset: %d\n", ret); diff --git a/sound/soc/codecs/wm8985.c b/sound/soc/codecs/wm8985.c index 271b517911a4..64e211cb4bcf 100644 --- a/sound/soc/codecs/wm8985.c +++ b/sound/soc/codecs/wm8985.c @@ -1000,13 +1000,6 @@ static int wm8985_probe(struct snd_soc_codec *codec) int ret; wm8985 = snd_soc_codec_get_drvdata(codec); - codec->control_data = wm8985->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); - return ret; - } for (i = 0; i < ARRAY_SIZE(wm8985->supplies); i++) wm8985->supplies[i].supply = wm8985_supply_names[i]; diff --git a/sound/soc/codecs/wm8988.c b/sound/soc/codecs/wm8988.c index a55e1c2c382e..424bbf752f3f 100644 --- a/sound/soc/codecs/wm8988.c +++ b/sound/soc/codecs/wm8988.c @@ -810,16 +810,8 @@ static int wm8988_resume(struct snd_soc_codec *codec) static int wm8988_probe(struct snd_soc_codec *codec) { - struct wm8988_priv *wm8988 = snd_soc_codec_get_drvdata(codec); int ret = 0; - codec->control_data = wm8988->regmap; - ret = snd_soc_codec_set_cache_io(codec, 7, 9, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - ret = wm8988_reset(codec); if (ret < 0) { dev_err(codec->dev, "Failed to issue reset\n"); diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index 0ccd4d8d043b..1487625551e7 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -1292,14 +1292,6 @@ static int wm8990_resume(struct snd_soc_codec *codec) */ static int wm8990_probe(struct snd_soc_codec *codec) { - int ret; - - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret < 0) { - printk(KERN_ERR "wm8990: failed to set cache I/O: %d\n", ret); - return ret; - } - wm8990_reset(codec); /* charge output caps */ diff --git a/sound/soc/codecs/wm8991.c b/sound/soc/codecs/wm8991.c index 32d219570cca..844cc4a60d66 100644 --- a/sound/soc/codecs/wm8991.c +++ b/sound/soc/codecs/wm8991.c @@ -1248,14 +1248,6 @@ static int wm8991_remove(struct snd_soc_codec *codec) static int wm8991_probe(struct snd_soc_codec *codec) { - int ret; - - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); - return ret; - } - wm8991_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; diff --git a/sound/soc/codecs/wm8993.c b/sound/soc/codecs/wm8993.c index 433d59a0f3ef..1674a1f33ba0 100644 --- a/sound/soc/codecs/wm8993.c +++ b/sound/soc/codecs/wm8993.c @@ -1493,13 +1493,6 @@ static int wm8993_probe(struct snd_soc_codec *codec) wm8993->hubs_data.dcs_codes_r = -2; wm8993->hubs_data.series_startup = 1; - codec->control_data = wm8993->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* Latch volume update bits and default ZC on */ snd_soc_update_bits(codec, WM8993_RIGHT_DAC_DIGITAL_VOLUME, WM8993_DAC_VU, WM8993_DAC_VU); diff --git a/sound/soc/codecs/wm8995.c b/sound/soc/codecs/wm8995.c index 4300caff1783..9fd76c9c4e36 100644 --- a/sound/soc/codecs/wm8995.c +++ b/sound/soc/codecs/wm8995.c @@ -2047,13 +2047,6 @@ static int wm8995_probe(struct snd_soc_codec *codec) wm8995 = snd_soc_codec_get_drvdata(codec); wm8995->codec = codec; - codec->control_data = wm8995->regmap; - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); - if (ret < 0) { - dev_err(codec->dev, "Failed to set cache i/o: %d\n", ret); - return ret; - } - for (i = 0; i < ARRAY_SIZE(wm8995->supplies); i++) wm8995->supplies[i].supply = wm8995_supply_names[i]; diff --git a/sound/soc/codecs/wm8996.c b/sound/soc/codecs/wm8996.c index 1a7655b0aa22..54066436d72c 100644 --- a/sound/soc/codecs/wm8996.c +++ b/sound/soc/codecs/wm8996.c @@ -2628,14 +2628,6 @@ static int wm8996_probe(struct snd_soc_codec *codec) init_completion(&wm8996->dcs_done); init_completion(&wm8996->fll_lock); - codec->control_data = wm8996->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - goto err; - } - if (wm8996->pdata.num_retune_mobile_cfgs) wm8996_retune_mobile_pdata(codec); else @@ -2674,13 +2666,11 @@ static int wm8996_probe(struct snd_soc_codec *codec) } else { dev_err(codec->dev, "Failed to request IRQ: %d\n", ret); + return ret; } } return 0; - -err: - return ret; } static int wm8996_remove(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/wm9081.c b/sound/soc/codecs/wm9081.c index 0982c1d38ec4..cda185d436f7 100644 --- a/sound/soc/codecs/wm9081.c +++ b/sound/soc/codecs/wm9081.c @@ -1265,15 +1265,6 @@ static struct snd_soc_dai_driver wm9081_dai = { static int wm9081_probe(struct snd_soc_codec *codec) { struct wm9081_priv *wm9081 = snd_soc_codec_get_drvdata(codec); - int ret; - - codec->control_data = wm9081->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } /* Enable zero cross by default */ snd_soc_update_bits(codec, WM9081_ANALOGUE_LINEOUT, @@ -1288,7 +1279,7 @@ static int wm9081_probe(struct snd_soc_codec *codec) ARRAY_SIZE(wm9081_eq_controls)); } - return ret; + return 0; } static int wm9081_remove(struct snd_soc_codec *codec) diff --git a/sound/soc/codecs/wm9090.c b/sound/soc/codecs/wm9090.c index a07fe1618eec..87934171f063 100644 --- a/sound/soc/codecs/wm9090.c +++ b/sound/soc/codecs/wm9090.c @@ -522,16 +522,6 @@ static int wm9090_set_bias_level(struct snd_soc_codec *codec, static int wm9090_probe(struct snd_soc_codec *codec) { - struct wm9090_priv *wm9090 = dev_get_drvdata(codec->dev); - int ret; - - codec->control_data = wm9090->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } - /* Configure some defaults; they will be written out when we * bring the bias up. */ -- cgit v1.2.3 From 092eba937d948a76ff55825922eff4df010f6a17 Mon Sep 17 00:00:00 2001 From: Xiubo Li Date: Tue, 11 Mar 2014 12:43:21 +0800 Subject: ASoC: io: New signature for snd_soc_codec_set_cache_io() Now that all users have been converted to regmap and the config.reg_bits and config.val_bits can be setted by each user through regmap core API. So these two params are redundant here. Since the only control type that left is SND_SOC_REGMAP, so remove it. Drop the control params and add struct regmap *regmap to simplify the code. Signed-off-by: Xiubo Li Signed-off-by: Mark Brown --- include/sound/soc.h | 7 +----- sound/soc/codecs/88pm860x-codec.c | 3 +-- sound/soc/codecs/cq93vc.c | 3 +-- sound/soc/codecs/mc13783.c | 4 ++-- sound/soc/codecs/si476x.c | 6 +++-- sound/soc/codecs/tlv320dac33.c | 1 - sound/soc/codecs/wm5102.c | 4 +--- sound/soc/codecs/wm5110.c | 3 +-- sound/soc/codecs/wm8350.c | 4 +--- sound/soc/codecs/wm8400.c | 3 +-- sound/soc/codecs/wm8994.c | 3 +-- sound/soc/codecs/wm8997.c | 4 +--- sound/soc/soc-core.c | 2 +- sound/soc/soc-io.c | 47 +++++++++++++++++---------------------- 14 files changed, 36 insertions(+), 58 deletions(-) (limited to 'sound/soc/codecs') diff --git a/include/sound/soc.h b/include/sound/soc.h index 2d8982db0344..85a5b7bbe39a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -354,10 +354,6 @@ typedef int (*hw_write_t)(void *,const char* ,int); extern struct snd_ac97_bus_ops *soc_ac97_ops; -enum snd_soc_control_type { - SND_SOC_REGMAP, -}; - enum snd_soc_pcm_subclass { SND_SOC_PCM_CLASS_PCM = 0, SND_SOC_PCM_CLASS_BE = 1, @@ -404,8 +400,7 @@ int snd_soc_codec_readable_register(struct snd_soc_codec *codec, int snd_soc_codec_writable_register(struct snd_soc_codec *codec, unsigned int reg); int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, - int addr_bits, int data_bits, - enum snd_soc_control_type control); + struct regmap *regmap); int snd_soc_cache_sync(struct snd_soc_codec *codec); int snd_soc_cache_init(struct snd_soc_codec *codec); int snd_soc_cache_exit(struct snd_soc_codec *codec); diff --git a/sound/soc/codecs/88pm860x-codec.c b/sound/soc/codecs/88pm860x-codec.c index 647a72cda005..773b53366ada 100644 --- a/sound/soc/codecs/88pm860x-codec.c +++ b/sound/soc/codecs/88pm860x-codec.c @@ -1327,8 +1327,7 @@ static int pm860x_probe(struct snd_soc_codec *codec) pm860x->codec = codec; - codec->control_data = pm860x->regmap; - ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, pm860x->regmap); if (ret) return ret; diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 43737a27d79c..1e25c7af853b 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -138,9 +138,8 @@ static int cq93vc_probe(struct snd_soc_codec *codec) struct davinci_vc *davinci_vc = codec->dev->platform_data; davinci_vc->cq93vc.codec = codec; - codec->control_data = davinci_vc->regmap; - snd_soc_codec_set_cache_io(codec, 32, 32, SND_SOC_REGMAP); + snd_soc_codec_set_cache_io(codec, davinci_vc->regmap); /* Off, with power on */ cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 582c2bbd42cb..fc28b20f6c69 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -614,8 +614,8 @@ static int mc13783_probe(struct snd_soc_codec *codec) struct mc13783_priv *priv = snd_soc_codec_get_drvdata(codec); int ret; - codec->control_data = dev_get_regmap(codec->dev->parent, NULL); - ret = snd_soc_codec_set_cache_io(codec, 8, 24, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, + dev_get_regmap(codec->dev->parent, NULL)); if (ret != 0) { dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); return ret; diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index fa2b8e07f420..244c097cd905 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -21,6 +21,7 @@ #include #include #include +#include #include #include @@ -209,8 +210,9 @@ out: static int si476x_codec_probe(struct snd_soc_codec *codec) { - codec->control_data = dev_get_regmap(codec->dev->parent, NULL); - return snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + struct regmap *regmap = dev_get_regmap(codec->dev->parent, NULL); + + return snd_soc_codec_set_cache_io(codec, regmap); } static struct snd_soc_dai_ops si476x_dai_ops = { diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 4f358393d6d6..64afda740c80 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -122,7 +122,6 @@ struct tlv320dac33_priv { unsigned int uthr; enum dac33_state state; - enum snd_soc_control_type control_type; void *control_data; }; diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index ce9c8e14d4bd..5613d0efe19b 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1758,9 +1758,7 @@ static int wm5102_codec_probe(struct snd_soc_codec *codec) struct wm5102_priv *priv = snd_soc_codec_get_drvdata(codec); int ret; - codec->control_data = priv->core.arizona->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, priv->core.arizona->regmap); if (ret != 0) return ret; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 2c3c962d9a85..66d3ad4176c3 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -1588,10 +1588,9 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) struct wm5110_priv *priv = snd_soc_codec_get_drvdata(codec); int ret; - codec->control_data = priv->core.arizona->regmap; priv->core.arizona->dapm = &codec->dapm; - ret = snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, priv->core.arizona->regmap); if (ret != 0) return ret; diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c index a183dcf3d5c1..757256bf7672 100644 --- a/sound/soc/codecs/wm8350.c +++ b/sound/soc/codecs/wm8350.c @@ -1505,9 +1505,7 @@ static int wm8350_codec_probe(struct snd_soc_codec *codec) if (ret != 0) return ret; - codec->control_data = wm8350->regmap; - - snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + snd_soc_codec_set_cache_io(codec, wm8350->regmap); /* Put the codec into reset if it wasn't already */ wm8350_clear_bits(wm8350, WM8350_POWER_MGMT_5, WM8350_CODEC_ENA); diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index 48dc7d2fee36..939baf83bb59 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1310,10 +1310,9 @@ static int wm8400_codec_probe(struct snd_soc_codec *codec) snd_soc_codec_set_drvdata(codec, priv); priv->wm8400 = wm8400; - codec->control_data = wm8400->regmap; priv->codec = codec; - snd_soc_codec_set_cache_io(codec, 8, 16, SND_SOC_REGMAP); + snd_soc_codec_set_cache_io(codec, wm8400->regmap); ret = devm_regulator_bulk_get(wm8400->dev, ARRAY_SIZE(power), &power[0]); diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index b9be9cbc4603..32cc83e3f1ff 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -3985,9 +3985,8 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) int ret, i; wm8994->hubs.codec = codec; - codec->control_data = control->regmap; - snd_soc_codec_set_cache_io(codec, 16, 16, SND_SOC_REGMAP); + snd_soc_codec_set_cache_io(codec, control->regmap); mutex_init(&wm8994->accdet_lock); INIT_DELAYED_WORK(&wm8994->jackdet_bootstrap, diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 555115ee2159..e3d1522daf64 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -1052,9 +1052,7 @@ static int wm8997_codec_probe(struct snd_soc_codec *codec) struct wm8997_priv *priv = snd_soc_codec_get_drvdata(codec); int ret; - codec->control_data = priv->core.arizona->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 32, 16, SND_SOC_REGMAP); + ret = snd_soc_codec_set_cache_io(codec, priv->core.arizona->regmap); if (ret != 0) return ret; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ad2dd14f0e3e..6510a8e4a5af 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1139,7 +1139,7 @@ static int soc_probe_codec(struct snd_soc_card *card, /* Set the default I/O up try regmap */ if (dev_get_regmap(codec->dev, NULL)) - snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); + snd_soc_codec_set_cache_io(codec, NULL); if (driver->probe) { ret = driver->probe(codec); diff --git a/sound/soc/soc-io.c b/sound/soc/soc-io.c index 18353f111b6a..8aa086996866 100644 --- a/sound/soc/soc-io.c +++ b/sound/soc/soc-io.c @@ -69,9 +69,7 @@ static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) * snd_soc_codec_set_cache_io: Set up standard I/O functions. * * @codec: CODEC to configure. - * @addr_bits: Number of bits of register address data. - * @data_bits: Number of bits of data per register. - * @control: Control bus used. + * @map: Register map to write to * * Register formats are frequently shared between many I2C and SPI * devices. In order to promote code reuse the ASoC core provides @@ -85,41 +83,36 @@ static unsigned int hw_read(struct snd_soc_codec *codec, unsigned int reg) * volatile registers. */ int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, - int addr_bits, int data_bits, - enum snd_soc_control_type control) + struct regmap *regmap) { int ret; + /* Device has made its own regmap arrangements */ + if (!regmap) + codec->control_data = dev_get_regmap(codec->dev, NULL); + else + codec->control_data = regmap; + + if (IS_ERR(codec->control_data)) + return PTR_ERR(codec->control_data); + codec->write = hw_write; codec->read = hw_read; - switch (control) { - case SND_SOC_REGMAP: - /* Device has made its own regmap arrangements */ - codec->using_regmap = true; - if (!codec->control_data) - codec->control_data = dev_get_regmap(codec->dev, NULL); - - if (codec->control_data) { - ret = regmap_get_val_bytes(codec->control_data); - /* Errors are legitimate for non-integer byte - * multiples */ - if (ret > 0) - codec->val_bytes = ret; - } - break; - - default: - return -EINVAL; - } + ret = regmap_get_val_bytes(codec->control_data); + /* Errors are legitimate for non-integer byte + * multiples */ + if (ret > 0) + codec->val_bytes = ret; + + codec->using_regmap = true; - return PTR_ERR_OR_ZERO(codec->control_data); + return 0; } EXPORT_SYMBOL_GPL(snd_soc_codec_set_cache_io); #else int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec, - int addr_bits, int data_bits, - enum snd_soc_control_type control) + struct regmap *regmap) { return -ENOTSUPP; } -- cgit v1.2.3 From 051389e250d018f7c38c8043c54aa8979d4b2cab Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 12 Mar 2014 13:32:34 +0000 Subject: ASoC: tlv320aic23: Remove spurious bits per word setting regmap should handle any byte ordering issues required, it is looking for a byte stream from the bus, so don't set 16 bits per word. This is likely to have tested out OK due to use of an unmerged SPI controller driver. Signed-off-by: Mark Brown Tested-by: Max Filippov --- sound/soc/codecs/tlv320aic23-spi.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic23-spi.c b/sound/soc/codecs/tlv320aic23-spi.c index 585aea436c6a..3b387e41d75d 100644 --- a/sound/soc/codecs/tlv320aic23-spi.c +++ b/sound/soc/codecs/tlv320aic23-spi.c @@ -25,7 +25,6 @@ static int aic23_spi_probe(struct spi_device *spi) dev_dbg(&spi->dev, "probing tlv320aic23 spi device\n"); - spi->bits_per_word = 16; spi->mode = SPI_MODE_0; ret = spi_setup(spi); if (ret < 0) -- cgit v1.2.3 From e00447fafbf7daf2cd49205b97e63d9734068a4f Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Tue, 11 Mar 2014 12:57:32 +0200 Subject: ASoC: tlv320aic31xx: Add basic codec driver implementation This commit adds a bare bones driver support for TLV320AIC31XX family audio codecs. The driver adds basic stereo playback trough headphone and speaker outputs and mono capture trough microphone inputs. The driver is currently missing support at least for mini DSP features and jack detection. I have tested the driver only on TLV320AIC3111, but based on the data sheets TLV320AIC3100, TLV320AIC3110, and TLV320AIC3120 should work Ok too. The base for the implementation was taken from: git@gitorious.org:ti-codecs/ti-codecs.git ajitk/topics/k3.10.1-aic31xx -branch at commit 77504eba0294764e9e63b4a0c696b44db187cd13. Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/tlv320aic31xx.txt | 61 + include/dt-bindings/sound/tlv320aic31xx-micbias.h | 8 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tlv320aic31xx.c | 1295 ++++++++++++++++++++ sound/soc/codecs/tlv320aic31xx.h | 258 ++++ 6 files changed, 1628 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic31xx.txt create mode 100644 include/dt-bindings/sound/tlv320aic31xx-micbias.h create mode 100644 sound/soc/codecs/tlv320aic31xx.c create mode 100644 sound/soc/codecs/tlv320aic31xx.h (limited to 'sound/soc/codecs') diff --git a/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt new file mode 100644 index 000000000000..74c66dee3e14 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tlv320aic31xx.txt @@ -0,0 +1,61 @@ +Texas Instruments - tlv320aic31xx Codec module + +The tlv320aic31xx serial control bus communicates through I2C protocols + +Required properties: + +- compatible - "string" - One of: + "ti,tlv320aic310x" - Generic TLV320AIC31xx with mono speaker amp + "ti,tlv320aic311x" - Generic TLV320AIC31xx with stereo speaker amp + "ti,tlv320aic3100" - TLV320AIC3100 (mono speaker amp, no MiniDSP) + "ti,tlv320aic3110" - TLV320AIC3110 (stereo speaker amp, no MiniDSP) + "ti,tlv320aic3120" - TLV320AIC3120 (mono speaker amp, MiniDSP) + "ti,tlv320aic3111" - TLV320AIC3111 (stereo speaker amp, MiniDSP) + +- reg - - I2C slave address + + +Optional properties: + +- gpio-reset - gpio pin number used for codec reset +- ai31xx-micbias-vg - MicBias Voltage setting + 1 or MICBIAS_2_0V - MICBIAS output is powered to 2.0V + 2 or MICBIAS_2_5V - MICBIAS output is powered to 2.5V + 3 or MICBIAS_AVDD - MICBIAS output is connected to AVDD + If this node is not mentioned or if the value is unknown, then + micbias is set to 2.0V. +- HPVDD-supply, SPRVDD-supply, SPLVDD-supply, AVDD-supply, IOVDD-supply, + DVDD-supply : power supplies for the device as covered in + Documentation/devicetree/bindings/regulator/regulator.txt + +CODEC output pins: + * HPL + * HPR + * SPL, devices with stereo speaker amp + * SPR, devices with stereo speaker amp + * SPK, devices with mono speaker amp + * MICBIAS + +CODEC input pins: + * MIC1LP + * MIC1RP + * MIC1LM + +The pins can be used in referring sound node's audio-routing property. + +Example: +#include + +tlv320aic31xx: tlv320aic31xx@18 { + compatible = "ti,tlv320aic311x"; + reg = <0x18>; + + ai31xx-micbias-vg = ; + + HPVDD-supply = <®ulator>; + SPRVDD-supply = <®ulator>; + SPLVDD-supply = <®ulator>; + AVDD-supply = <®ulator>; + IOVDD-supply = <®ulator>; + DVDD-supply = <®ulator>; +}; diff --git a/include/dt-bindings/sound/tlv320aic31xx-micbias.h b/include/dt-bindings/sound/tlv320aic31xx-micbias.h new file mode 100644 index 000000000000..f5cb772ab9c8 --- /dev/null +++ b/include/dt-bindings/sound/tlv320aic31xx-micbias.h @@ -0,0 +1,8 @@ +#ifndef __DT_TLV320AIC31XX_MICBIAS_H +#define __DT_TLV320AIC31XX_MICBIAS_H + +#define MICBIAS_2_0V 1 +#define MICBIAS_2_5V 2 +#define MICBIAS_AVDDV 3 + +#endif /* __DT_TLV320AIC31XX_MICBIAS_H */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 983d087aa92a..66f6c53ea328 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -73,6 +73,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_TAS5086 if I2C select SND_SOC_TLV320AIC23 if I2C select SND_SOC_TLV320AIC26 if SPI_MASTER + select SND_SOC_TLV320AIC31XX if I2C select SND_SOC_TLV320AIC32X4 if I2C select SND_SOC_TLV320AIC3X if I2C select SND_SOC_TPA6130A2 if I2C @@ -361,6 +362,9 @@ config SND_SOC_TLV320AIC26 tristate depends on SPI +config SND_SOC_TLV320AIC31XX + tristate + config SND_SOC_TLV320AIC32X4 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 1deeb20fd411..ff1775c562fe 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -64,6 +64,7 @@ snd-soc-stac9766-objs := stac9766.o snd-soc-tas5086-objs := tas5086.o snd-soc-tlv320aic23-objs := tlv320aic23.o snd-soc-tlv320aic26-objs := tlv320aic26.o +snd-soc-tlv320aic31xx-objs := tlv320aic31xx.o snd-soc-tlv320aic32x4-objs := tlv320aic32x4.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o snd-soc-tlv320dac33-objs := tlv320dac33.o @@ -194,6 +195,7 @@ obj-$(CONFIG_SND_SOC_STAC9766) += snd-soc-stac9766.o obj-$(CONFIG_SND_SOC_TAS5086) += snd-soc-tas5086.o obj-$(CONFIG_SND_SOC_TLV320AIC23) += snd-soc-tlv320aic23.o obj-$(CONFIG_SND_SOC_TLV320AIC26) += snd-soc-tlv320aic26.o +obj-$(CONFIG_SND_SOC_TLV320AIC31XX) += snd-soc-tlv320aic31xx.o obj-$(CONFIG_SND_SOC_TLV320AIC32X4) += snd-soc-tlv320aic32x4.o obj-$(CONFIG_SND_SOC_TLV320AIC3X) += snd-soc-tlv320aic3x.o obj-$(CONFIG_SND_SOC_TLV320DAC33) += snd-soc-tlv320dac33.o diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c new file mode 100644 index 000000000000..e60e37b43a1b --- /dev/null +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -0,0 +1,1295 @@ +/* + * ALSA SoC TLV320AIC31XX codec driver + * + * Copyright (C) 2014 Texas Instruments, Inc. + * + * Author: Jyri Sarha + * + * Based on ground work by: Ajit Kulkarni + * + * This package is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * THIS PACKAGE IS PROVIDED AS IS AND WITHOUT ANY EXPRESS OR + * IMPLIED WARRANTIES, INCLUDING, WITHOUT LIMITATION, THE IMPLIED + * WARRANTIES OF MERCHANTIBILITY AND FITNESS FOR A PARTICULAR PURPOSE. + * + * The TLV320AIC31xx series of audio codec is a low-power, highly integrated + * high performance codec which provides a stereo DAC, a mono ADC, + * and mono/stereo Class-D speaker driver. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "tlv320aic31xx.h" + +static const struct reg_default aic31xx_reg_defaults[] = { + { AIC31XX_CLKMUX, 0x00 }, + { AIC31XX_PLLPR, 0x11 }, + { AIC31XX_PLLJ, 0x04 }, + { AIC31XX_PLLDMSB, 0x00 }, + { AIC31XX_PLLDLSB, 0x00 }, + { AIC31XX_NDAC, 0x01 }, + { AIC31XX_MDAC, 0x01 }, + { AIC31XX_DOSRMSB, 0x00 }, + { AIC31XX_DOSRLSB, 0x80 }, + { AIC31XX_NADC, 0x01 }, + { AIC31XX_MADC, 0x01 }, + { AIC31XX_AOSR, 0x80 }, + { AIC31XX_IFACE1, 0x00 }, + { AIC31XX_DATA_OFFSET, 0x00 }, + { AIC31XX_IFACE2, 0x00 }, + { AIC31XX_BCLKN, 0x01 }, + { AIC31XX_DACSETUP, 0x14 }, + { AIC31XX_DACMUTE, 0x0c }, + { AIC31XX_LDACVOL, 0x00 }, + { AIC31XX_RDACVOL, 0x00 }, + { AIC31XX_ADCSETUP, 0x00 }, + { AIC31XX_ADCFGA, 0x80 }, + { AIC31XX_ADCVOL, 0x00 }, + { AIC31XX_HPDRIVER, 0x04 }, + { AIC31XX_SPKAMP, 0x06 }, + { AIC31XX_DACMIXERROUTE, 0x00 }, + { AIC31XX_LANALOGHPL, 0x7f }, + { AIC31XX_RANALOGHPR, 0x7f }, + { AIC31XX_LANALOGSPL, 0x7f }, + { AIC31XX_RANALOGSPR, 0x7f }, + { AIC31XX_HPLGAIN, 0x02 }, + { AIC31XX_HPRGAIN, 0x02 }, + { AIC31XX_SPLGAIN, 0x00 }, + { AIC31XX_SPRGAIN, 0x00 }, + { AIC31XX_MICBIAS, 0x00 }, + { AIC31XX_MICPGA, 0x80 }, + { AIC31XX_MICPGAPI, 0x00 }, + { AIC31XX_MICPGAMI, 0x00 }, +}; + +static bool aic31xx_volatile(struct device *dev, unsigned int reg) +{ + switch (reg) { + case AIC31XX_PAGECTL: /* regmap implementation requires this */ + case AIC31XX_RESET: /* always clears after write */ + case AIC31XX_OT_FLAG: + case AIC31XX_ADCFLAG: + case AIC31XX_DACFLAG1: + case AIC31XX_DACFLAG2: + case AIC31XX_OFFLAG: /* Sticky interrupt flags */ + case AIC31XX_INTRDACFLAG: /* Sticky interrupt flags */ + case AIC31XX_INTRADCFLAG: /* Sticky interrupt flags */ + case AIC31XX_INTRDACFLAG2: + case AIC31XX_INTRADCFLAG2: + return true; + } + return false; +} + +static bool aic31xx_writeable(struct device *dev, unsigned int reg) +{ + switch (reg) { + case AIC31XX_OT_FLAG: + case AIC31XX_ADCFLAG: + case AIC31XX_DACFLAG1: + case AIC31XX_DACFLAG2: + case AIC31XX_OFFLAG: /* Sticky interrupt flags */ + case AIC31XX_INTRDACFLAG: /* Sticky interrupt flags */ + case AIC31XX_INTRADCFLAG: /* Sticky interrupt flags */ + case AIC31XX_INTRDACFLAG2: + case AIC31XX_INTRADCFLAG2: + return false; + } + return true; +} + +static const struct regmap_range_cfg aic31xx_ranges[] = { + { + .range_min = 0, + .range_max = 12 * 128, + .selector_reg = AIC31XX_PAGECTL, + .selector_mask = 0xff, + .selector_shift = 0, + .window_start = 0, + .window_len = 128, + }, +}; + +struct regmap_config aic31xx_i2c_regmap = { + .reg_bits = 8, + .val_bits = 8, + .writeable_reg = aic31xx_writeable, + .volatile_reg = aic31xx_volatile, + .reg_defaults = aic31xx_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(aic31xx_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .ranges = aic31xx_ranges, + .num_ranges = ARRAY_SIZE(aic31xx_ranges), + .max_register = 12 * 128, +}; + +#define AIC31XX_NUM_SUPPLIES 6 +static const char * const aic31xx_supply_names[AIC31XX_NUM_SUPPLIES] = { + "HPVDD", + "SPRVDD", + "SPLVDD", + "AVDD", + "IOVDD", + "DVDD", +}; + +struct aic31xx_disable_nb { + struct notifier_block nb; + struct aic31xx_priv *aic31xx; +}; + +struct aic31xx_priv { + struct snd_soc_codec *codec; + u8 i2c_regs_status; + struct device *dev; + struct regmap *regmap; + struct aic31xx_pdata pdata; + struct regulator_bulk_data supplies[AIC31XX_NUM_SUPPLIES]; + struct aic31xx_disable_nb disable_nb[AIC31XX_NUM_SUPPLIES]; + unsigned int sysclk; + int rate_div_line; +}; + +struct aic31xx_rate_divs { + u32 mclk; + u32 rate; + u8 p_val; + u8 pll_j; + u16 pll_d; + u16 dosr; + u8 ndac; + u8 mdac; + u8 aosr; + u8 nadc; + u8 madc; +}; + +/* ADC dividers can be disabled by cofiguring them to 0 */ +static const struct aic31xx_rate_divs aic31xx_divs[] = { + /* mclk rate pll: p j d dosr ndac mdac aors nadc madc */ + /* 8k rate */ + {12000000, 8000, 1, 8, 1920, 128, 48, 2, 128, 48, 2}, + {24000000, 8000, 2, 8, 1920, 128, 48, 2, 128, 48, 2}, + {25000000, 8000, 2, 7, 8643, 128, 48, 2, 128, 48, 2}, + /* 11.025k rate */ + {12000000, 11025, 1, 7, 5264, 128, 32, 2, 128, 32, 2}, + {24000000, 11025, 2, 7, 5264, 128, 32, 2, 128, 32, 2}, + {25000000, 11025, 2, 7, 2253, 128, 32, 2, 128, 32, 2}, + /* 16k rate */ + {12000000, 16000, 1, 8, 1920, 128, 24, 2, 128, 24, 2}, + {24000000, 16000, 2, 8, 1920, 128, 24, 2, 128, 24, 2}, + {25000000, 16000, 2, 7, 8643, 128, 24, 2, 128, 24, 2}, + /* 22.05k rate */ + {12000000, 22050, 1, 7, 5264, 128, 16, 2, 128, 16, 2}, + {24000000, 22050, 2, 7, 5264, 128, 16, 2, 128, 16, 2}, + {25000000, 22050, 2, 7, 2253, 128, 16, 2, 128, 16, 2}, + /* 32k rate */ + {12000000, 32000, 1, 8, 1920, 128, 12, 2, 128, 12, 2}, + {24000000, 32000, 2, 8, 1920, 128, 12, 2, 128, 12, 2}, + {25000000, 32000, 2, 7, 8643, 128, 12, 2, 128, 12, 2}, + /* 44.1k rate */ + {12000000, 44100, 1, 7, 5264, 128, 8, 2, 128, 8, 2}, + {24000000, 44100, 2, 7, 5264, 128, 8, 2, 128, 8, 2}, + {25000000, 44100, 2, 7, 2253, 128, 8, 2, 128, 8, 2}, + /* 48k rate */ + {12000000, 48000, 1, 8, 1920, 128, 8, 2, 128, 8, 2}, + {24000000, 48000, 2, 8, 1920, 128, 8, 2, 128, 8, 2}, + {25000000, 48000, 2, 7, 8643, 128, 8, 2, 128, 8, 2}, + /* 88.2k rate */ + {12000000, 88200, 1, 7, 5264, 64, 8, 2, 64, 8, 2}, + {24000000, 88200, 2, 7, 5264, 64, 8, 2, 64, 8, 2}, + {25000000, 88200, 2, 7, 2253, 64, 8, 2, 64, 8, 2}, + /* 96k rate */ + {12000000, 96000, 1, 8, 1920, 64, 8, 2, 64, 8, 2}, + {24000000, 96000, 2, 8, 1920, 64, 8, 2, 64, 8, 2}, + {25000000, 96000, 2, 7, 8643, 64, 8, 2, 64, 8, 2}, + /* 176.4k rate */ + {12000000, 176400, 1, 7, 5264, 32, 8, 2, 32, 8, 2}, + {24000000, 176400, 2, 7, 5264, 32, 8, 2, 32, 8, 2}, + {25000000, 176400, 2, 7, 2253, 32, 8, 2, 32, 8, 2}, + /* 192k rate */ + {12000000, 192000, 1, 8, 1920, 32, 8, 2, 32, 8, 2}, + {24000000, 192000, 2, 8, 1920, 32, 8, 2, 32, 8, 2}, + {25000000, 192000, 2, 7, 8643, 32, 8, 2, 32, 8, 2}, +}; + +static const char * const ldac_in_text[] = { + "Off", "Left Data", "Right Data", "Mono" +}; + +static const char * const rdac_in_text[] = { + "Off", "Right Data", "Left Data", "Mono" +}; + +static SOC_ENUM_SINGLE_DECL(ldac_in_enum, AIC31XX_DACSETUP, 4, ldac_in_text); + +static SOC_ENUM_SINGLE_DECL(rdac_in_enum, AIC31XX_DACSETUP, 2, rdac_in_text); + +static const char * const mic_select_text[] = { + "Off", "FFR 10 Ohm", "FFR 20 Ohm", "FFR 40 Ohm" +}; + +static const +SOC_ENUM_SINGLE_DECL(mic1lp_p_enum, AIC31XX_MICPGAPI, 6, mic_select_text); +static const +SOC_ENUM_SINGLE_DECL(mic1rp_p_enum, AIC31XX_MICPGAPI, 4, mic_select_text); +static const +SOC_ENUM_SINGLE_DECL(mic1lm_p_enum, AIC31XX_MICPGAPI, 2, mic_select_text); + +static const +SOC_ENUM_SINGLE_DECL(cm_m_enum, AIC31XX_MICPGAMI, 6, mic_select_text); +static const +SOC_ENUM_SINGLE_DECL(mic1lm_m_enum, AIC31XX_MICPGAMI, 4, mic_select_text); + +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -6350, 50, 0); +static const DECLARE_TLV_DB_SCALE(adc_fgain_tlv, 0, 10, 0); +static const DECLARE_TLV_DB_SCALE(adc_cgain_tlv, -2000, 50, 0); +static const DECLARE_TLV_DB_SCALE(mic_pga_tlv, 0, 50, 0); +static const DECLARE_TLV_DB_SCALE(hp_drv_tlv, 0, 100, 0); +static const DECLARE_TLV_DB_SCALE(class_D_drv_tlv, 600, 600, 0); +static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -6350, 50, 0); +static const DECLARE_TLV_DB_SCALE(sp_vol_tlv, -6350, 50, 0); + +/* + * controls to be exported to the user space + */ +static const struct snd_kcontrol_new aic31xx_snd_controls[] = { + SOC_DOUBLE_R_S_TLV("DAC Playback Volume", AIC31XX_LDACVOL, + AIC31XX_RDACVOL, 0, -127, 48, 7, 0, dac_vol_tlv), + + SOC_SINGLE_TLV("ADC Fine Capture Volume", AIC31XX_ADCFGA, 4, 4, 1, + adc_fgain_tlv), + + SOC_SINGLE("ADC Capture Switch", AIC31XX_ADCFGA, 7, 1, 1), + SOC_DOUBLE_R_S_TLV("ADC Capture Volume", AIC31XX_ADCVOL, AIC31XX_ADCVOL, + 0, -24, 40, 6, 0, adc_cgain_tlv), + + SOC_SINGLE_TLV("Mic PGA Capture Volume", AIC31XX_MICPGA, 0, + 119, 0, mic_pga_tlv), + + SOC_DOUBLE_R("HP Driver Playback Switch", AIC31XX_HPLGAIN, + AIC31XX_HPRGAIN, 2, 1, 0), + SOC_DOUBLE_R_TLV("HP Driver Playback Volume", AIC31XX_HPLGAIN, + AIC31XX_HPRGAIN, 3, 0x09, 0, hp_drv_tlv), + + SOC_DOUBLE_R_TLV("HP Analog Playback Volume", AIC31XX_LANALOGHPL, + AIC31XX_RANALOGHPR, 0, 0x7F, 1, hp_vol_tlv), +}; + +static const struct snd_kcontrol_new aic311x_snd_controls[] = { + SOC_DOUBLE_R("Speaker Driver Playback Switch", AIC31XX_SPLGAIN, + AIC31XX_SPRGAIN, 2, 1, 0), + SOC_DOUBLE_R_TLV("Speaker Driver Playback Volume", AIC31XX_SPLGAIN, + AIC31XX_SPRGAIN, 3, 3, 0, class_D_drv_tlv), + + SOC_DOUBLE_R_TLV("Speaker Analog Playback Volume", AIC31XX_LANALOGSPL, + AIC31XX_RANALOGSPR, 0, 0x7F, 1, sp_vol_tlv), +}; + +static const struct snd_kcontrol_new aic310x_snd_controls[] = { + SOC_SINGLE("Speaker Driver Playback Switch", AIC31XX_SPLGAIN, + 2, 1, 0), + SOC_SINGLE_TLV("Speaker Driver Playback Volume", AIC31XX_SPLGAIN, + 3, 3, 0, class_D_drv_tlv), + + SOC_SINGLE_TLV("Speaker Analog Playback Volume", AIC31XX_LANALOGSPL, + 0, 0x7F, 1, sp_vol_tlv), +}; + +static const struct snd_kcontrol_new ldac_in_control = + SOC_DAPM_ENUM("DAC Left Input", ldac_in_enum); + +static const struct snd_kcontrol_new rdac_in_control = + SOC_DAPM_ENUM("DAC Right Input", rdac_in_enum); + +int aic31xx_wait_bits(struct aic31xx_priv *aic31xx, unsigned int reg, + unsigned int mask, unsigned int wbits, int sleep, + int count) +{ + unsigned int bits; + int counter = count; + int ret = regmap_read(aic31xx->regmap, reg, &bits); + while ((bits & mask) != wbits && counter && !ret) { + usleep_range(sleep, sleep * 2); + ret = regmap_read(aic31xx->regmap, reg, &bits); + counter--; + } + if ((bits & mask) != wbits) { + dev_err(aic31xx->dev, + "%s: Failed! 0x%x was 0x%x expected 0x%x (%d, 0x%x, %d us)\n", + __func__, reg, bits, wbits, ret, mask, + (count - counter) * sleep); + ret = -1; + } + return ret; +} + +#define WIDGET_BIT(reg, shift) (((shift) << 8) | (reg)) + +static int aic31xx_dapm_power_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(w->codec); + unsigned int reg = AIC31XX_DACFLAG1; + unsigned int mask; + + switch (WIDGET_BIT(w->reg, w->shift)) { + case WIDGET_BIT(AIC31XX_DACSETUP, 7): + mask = AIC31XX_LDACPWRSTATUS_MASK; + break; + case WIDGET_BIT(AIC31XX_DACSETUP, 6): + mask = AIC31XX_RDACPWRSTATUS_MASK; + break; + case WIDGET_BIT(AIC31XX_HPDRIVER, 7): + mask = AIC31XX_HPLDRVPWRSTATUS_MASK; + break; + case WIDGET_BIT(AIC31XX_HPDRIVER, 6): + mask = AIC31XX_HPRDRVPWRSTATUS_MASK; + break; + case WIDGET_BIT(AIC31XX_SPKAMP, 7): + mask = AIC31XX_SPLDRVPWRSTATUS_MASK; + break; + case WIDGET_BIT(AIC31XX_SPKAMP, 6): + mask = AIC31XX_SPRDRVPWRSTATUS_MASK; + break; + case WIDGET_BIT(AIC31XX_ADCSETUP, 7): + mask = AIC31XX_ADCPWRSTATUS_MASK; + reg = AIC31XX_ADCFLAG; + break; + default: + dev_err(w->codec->dev, "Unknown widget '%s' calling %s/n", + w->name, __func__); + return -EINVAL; + } + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + return aic31xx_wait_bits(aic31xx, reg, mask, mask, 5000, 100); + case SND_SOC_DAPM_POST_PMD: + return aic31xx_wait_bits(aic31xx, reg, mask, 0, 5000, 100); + default: + dev_dbg(w->codec->dev, + "Unhandled dapm widget event %d from %s\n", + event, w->name); + } + return 0; +} + +static const struct snd_kcontrol_new left_output_switches[] = { + SOC_DAPM_SINGLE("From Left DAC", AIC31XX_DACMIXERROUTE, 6, 1, 0), + SOC_DAPM_SINGLE("From MIC1LP", AIC31XX_DACMIXERROUTE, 5, 1, 0), + SOC_DAPM_SINGLE("From MIC1RP", AIC31XX_DACMIXERROUTE, 4, 1, 0), +}; + +static const struct snd_kcontrol_new right_output_switches[] = { + SOC_DAPM_SINGLE("From Right DAC", AIC31XX_DACMIXERROUTE, 2, 1, 0), + SOC_DAPM_SINGLE("From MIC1RP", AIC31XX_DACMIXERROUTE, 1, 1, 0), +}; + +static const struct snd_kcontrol_new p_term_mic1lp = + SOC_DAPM_ENUM("MIC1LP P-Terminal", mic1lp_p_enum); + +static const struct snd_kcontrol_new p_term_mic1rp = + SOC_DAPM_ENUM("MIC1RP P-Terminal", mic1rp_p_enum); + +static const struct snd_kcontrol_new p_term_mic1lm = + SOC_DAPM_ENUM("MIC1LM P-Terminal", mic1lm_p_enum); + +static const struct snd_kcontrol_new m_term_mic1lm = + SOC_DAPM_ENUM("MIC1LM M-Terminal", mic1lm_m_enum); + +static const struct snd_kcontrol_new aic31xx_dapm_hpl_switch = + SOC_DAPM_SINGLE("Switch", AIC31XX_LANALOGHPL, 7, 1, 0); + +static const struct snd_kcontrol_new aic31xx_dapm_hpr_switch = + SOC_DAPM_SINGLE("Switch", AIC31XX_RANALOGHPR, 7, 1, 0); + +static const struct snd_kcontrol_new aic31xx_dapm_spl_switch = + SOC_DAPM_SINGLE("Switch", AIC31XX_LANALOGSPL, 7, 1, 0); + +static const struct snd_kcontrol_new aic31xx_dapm_spr_switch = + SOC_DAPM_SINGLE("Switch", AIC31XX_RANALOGSPR, 7, 1, 0); + +static int mic_bias_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* change mic bias voltage to user defined */ + snd_soc_update_bits(codec, AIC31XX_MICBIAS, + AIC31XX_MICBIAS_MASK, + aic31xx->pdata.micbias_vg << + AIC31XX_MICBIAS_SHIFT); + dev_dbg(codec->dev, "%s: turned on\n", __func__); + break; + case SND_SOC_DAPM_PRE_PMD: + /* turn mic bias off */ + snd_soc_update_bits(codec, AIC31XX_MICBIAS, + AIC31XX_MICBIAS_MASK, 0); + dev_dbg(codec->dev, "%s: turned off\n", __func__); + break; + } + return 0; +} + +static const struct snd_soc_dapm_widget aic31xx_dapm_widgets[] = { + SND_SOC_DAPM_AIF_IN("DAC IN", "DAC Playback", 0, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_MUX("DAC Left Input", + SND_SOC_NOPM, 0, 0, &ldac_in_control), + SND_SOC_DAPM_MUX("DAC Right Input", + SND_SOC_NOPM, 0, 0, &rdac_in_control), + /* DACs */ + SND_SOC_DAPM_DAC_E("DAC Left", "Left Playback", + AIC31XX_DACSETUP, 7, 0, aic31xx_dapm_power_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_DAC_E("DAC Right", "Right Playback", + AIC31XX_DACSETUP, 6, 0, aic31xx_dapm_power_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("Output Left", SND_SOC_NOPM, 0, 0, + left_output_switches, + ARRAY_SIZE(left_output_switches)), + SND_SOC_DAPM_MIXER("Output Right", SND_SOC_NOPM, 0, 0, + right_output_switches, + ARRAY_SIZE(right_output_switches)), + + SND_SOC_DAPM_SWITCH("HP Left", SND_SOC_NOPM, 0, 0, + &aic31xx_dapm_hpl_switch), + SND_SOC_DAPM_SWITCH("HP Right", SND_SOC_NOPM, 0, 0, + &aic31xx_dapm_hpr_switch), + + /* Output drivers */ + SND_SOC_DAPM_OUT_DRV_E("HPL Driver", AIC31XX_HPDRIVER, 7, 0, + NULL, 0, aic31xx_dapm_power_event, + SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_OUT_DRV_E("HPR Driver", AIC31XX_HPDRIVER, 6, 0, + NULL, 0, aic31xx_dapm_power_event, + SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_POST_PMU), + + /* ADC */ + SND_SOC_DAPM_ADC_E("ADC", "Capture", AIC31XX_ADCSETUP, 7, 0, + aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + + /* Input Selection to MIC_PGA */ + SND_SOC_DAPM_MUX("MIC1LP P-Terminal", SND_SOC_NOPM, 0, 0, + &p_term_mic1lp), + SND_SOC_DAPM_MUX("MIC1RP P-Terminal", SND_SOC_NOPM, 0, 0, + &p_term_mic1rp), + SND_SOC_DAPM_MUX("MIC1LM P-Terminal", SND_SOC_NOPM, 0, 0, + &p_term_mic1lm), + + SND_SOC_DAPM_MUX("MIC1LM M-Terminal", SND_SOC_NOPM, 0, 0, + &m_term_mic1lm), + /* Enabling & Disabling MIC Gain Ctl */ + SND_SOC_DAPM_PGA("MIC_GAIN_CTL", AIC31XX_MICPGA, + 7, 1, NULL, 0), + + /* Mic Bias */ + SND_SOC_DAPM_SUPPLY("MICBIAS", SND_SOC_NOPM, 0, 0, mic_bias_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("HPL"), + SND_SOC_DAPM_OUTPUT("HPR"), + + /* Inputs */ + SND_SOC_DAPM_INPUT("MIC1LP"), + SND_SOC_DAPM_INPUT("MIC1RP"), + SND_SOC_DAPM_INPUT("MIC1LM"), +}; + +static const struct snd_soc_dapm_widget aic311x_dapm_widgets[] = { + /* AIC3111 and AIC3110 have stereo class-D amplifier */ + SND_SOC_DAPM_OUT_DRV_E("SPL ClassD", AIC31XX_SPKAMP, 7, 0, NULL, 0, + aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_OUT_DRV_E("SPR ClassD", AIC31XX_SPKAMP, 6, 0, NULL, 0, + aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SWITCH("Speaker Left", SND_SOC_NOPM, 0, 0, + &aic31xx_dapm_spl_switch), + SND_SOC_DAPM_SWITCH("Speaker Right", SND_SOC_NOPM, 0, 0, + &aic31xx_dapm_spr_switch), + SND_SOC_DAPM_OUTPUT("SPL"), + SND_SOC_DAPM_OUTPUT("SPR"), +}; + +/* AIC3100 and AIC3120 have only mono class-D amplifier */ +static const struct snd_soc_dapm_widget aic310x_dapm_widgets[] = { + SND_SOC_DAPM_OUT_DRV_E("SPK ClassD", AIC31XX_SPKAMP, 7, 0, NULL, 0, + aic31xx_dapm_power_event, SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_SWITCH("Speaker", SND_SOC_NOPM, 0, 0, + &aic31xx_dapm_spl_switch), + SND_SOC_DAPM_OUTPUT("SPK"), +}; + +static const struct snd_soc_dapm_route +aic31xx_audio_map[] = { + /* DAC Input Routing */ + {"DAC Left Input", "Left Data", "DAC IN"}, + {"DAC Left Input", "Right Data", "DAC IN"}, + {"DAC Left Input", "Mono", "DAC IN"}, + {"DAC Right Input", "Left Data", "DAC IN"}, + {"DAC Right Input", "Right Data", "DAC IN"}, + {"DAC Right Input", "Mono", "DAC IN"}, + {"DAC Left", NULL, "DAC Left Input"}, + {"DAC Right", NULL, "DAC Right Input"}, + + /* Mic input */ + {"MIC1LP P-Terminal", "FFR 10 Ohm", "MIC1LP"}, + {"MIC1LP P-Terminal", "FFR 20 Ohm", "MIC1LP"}, + {"MIC1LP P-Terminal", "FFR 40 Ohm", "MIC1LP"}, + {"MIC1RP P-Terminal", "FFR 10 Ohm", "MIC1RP"}, + {"MIC1RP P-Terminal", "FFR 20 Ohm", "MIC1RP"}, + {"MIC1RP P-Terminal", "FFR 40 Ohm", "MIC1RP"}, + {"MIC1LM P-Terminal", "FFR 10 Ohm", "MIC1LM"}, + {"MIC1LM P-Terminal", "FFR 20 Ohm", "MIC1LM"}, + {"MIC1LM P-Terminal", "FFR 40 Ohm", "MIC1LM"}, + + {"MIC1LM M-Terminal", "FFR 10 Ohm", "MIC1LM"}, + {"MIC1LM M-Terminal", "FFR 20 Ohm", "MIC1LM"}, + {"MIC1LM M-Terminal", "FFR 40 Ohm", "MIC1LM"}, + + {"MIC_GAIN_CTL", NULL, "MIC1LP P-Terminal"}, + {"MIC_GAIN_CTL", NULL, "MIC1RP P-Terminal"}, + {"MIC_GAIN_CTL", NULL, "MIC1LM P-Terminal"}, + {"MIC_GAIN_CTL", NULL, "MIC1LM M-Terminal"}, + + {"ADC", NULL, "MIC_GAIN_CTL"}, + + /* Left Output */ + {"Output Left", "From Left DAC", "DAC Left"}, + {"Output Left", "From MIC1LP", "MIC1LP"}, + {"Output Left", "From MIC1RP", "MIC1RP"}, + + /* Right Output */ + {"Output Right", "From Right DAC", "DAC Right"}, + {"Output Right", "From MIC1RP", "MIC1RP"}, + + /* HPL path */ + {"HP Left", "Switch", "Output Left"}, + {"HPL Driver", NULL, "HP Left"}, + {"HPL", NULL, "HPL Driver"}, + + /* HPR path */ + {"HP Right", "Switch", "Output Right"}, + {"HPR Driver", NULL, "HP Right"}, + {"HPR", NULL, "HPR Driver"}, +}; + +static const struct snd_soc_dapm_route +aic311x_audio_map[] = { + /* SP L path */ + {"Speaker Left", "Switch", "Output Left"}, + {"SPL ClassD", NULL, "Speaker Left"}, + {"SPL", NULL, "SPL ClassD"}, + + /* SP R path */ + {"Speaker Right", "Switch", "Output Right"}, + {"SPR ClassD", NULL, "Speaker Right"}, + {"SPR", NULL, "SPR ClassD"}, +}; + +static const struct snd_soc_dapm_route +aic310x_audio_map[] = { + /* SP L path */ + {"Speaker", "Switch", "Output Left"}, + {"SPK ClassD", NULL, "Speaker"}, + {"SPK", NULL, "SPK ClassD"}, +}; + +static int aic31xx_add_controls(struct snd_soc_codec *codec) +{ + int ret = 0; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + + if (aic31xx->pdata.codec_type & AIC31XX_STEREO_CLASS_D_BIT) + ret = snd_soc_add_codec_controls( + codec, aic311x_snd_controls, + ARRAY_SIZE(aic311x_snd_controls)); + else + ret = snd_soc_add_codec_controls( + codec, aic310x_snd_controls, + ARRAY_SIZE(aic310x_snd_controls)); + + return ret; +} + +static int aic31xx_add_widgets(struct snd_soc_codec *codec) +{ + struct snd_soc_dapm_context *dapm = &codec->dapm; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + if (aic31xx->pdata.codec_type & AIC31XX_STEREO_CLASS_D_BIT) { + ret = snd_soc_dapm_new_controls( + dapm, aic311x_dapm_widgets, + ARRAY_SIZE(aic311x_dapm_widgets)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, aic311x_audio_map, + ARRAY_SIZE(aic311x_audio_map)); + if (ret) + return ret; + } else { + ret = snd_soc_dapm_new_controls( + dapm, aic310x_dapm_widgets, + ARRAY_SIZE(aic310x_dapm_widgets)); + if (ret) + return ret; + + ret = snd_soc_dapm_add_routes(dapm, aic310x_audio_map, + ARRAY_SIZE(aic310x_audio_map)); + if (ret) + return ret; + } + + return 0; +} + +static int aic31xx_setup_pll(struct snd_soc_codec *codec, + struct snd_pcm_hw_params *params) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int bclk_n = 0; + int i; + + /* Use PLL as CODEC_CLKIN and DAC_CLK as BDIV_CLKIN */ + snd_soc_update_bits(codec, AIC31XX_CLKMUX, + AIC31XX_CODEC_CLKIN_MASK, AIC31XX_CODEC_CLKIN_PLL); + snd_soc_update_bits(codec, AIC31XX_IFACE2, + AIC31XX_BDIVCLK_MASK, AIC31XX_DAC2BCLK); + + for (i = 0; i < ARRAY_SIZE(aic31xx_divs); i++) { + if (aic31xx_divs[i].rate == params_rate(params) && + aic31xx_divs[i].mclk == aic31xx->sysclk) + break; + } + + if (i == ARRAY_SIZE(aic31xx_divs)) { + dev_err(codec->dev, "%s: Sampling rate %u not supported\n", + __func__, params_rate(params)); + return -EINVAL; + } + + /* PLL configuration */ + snd_soc_update_bits(codec, AIC31XX_PLLPR, AIC31XX_PLL_MASK, + (aic31xx_divs[i].p_val << 4) | 0x01); + snd_soc_write(codec, AIC31XX_PLLJ, aic31xx_divs[i].pll_j); + + snd_soc_write(codec, AIC31XX_PLLDMSB, + aic31xx_divs[i].pll_d >> 8); + snd_soc_write(codec, AIC31XX_PLLDLSB, + aic31xx_divs[i].pll_d & 0xff); + + /* DAC dividers configuration */ + snd_soc_update_bits(codec, AIC31XX_NDAC, AIC31XX_PLL_MASK, + aic31xx_divs[i].ndac); + snd_soc_update_bits(codec, AIC31XX_MDAC, AIC31XX_PLL_MASK, + aic31xx_divs[i].mdac); + + snd_soc_write(codec, AIC31XX_DOSRMSB, aic31xx_divs[i].dosr >> 8); + snd_soc_write(codec, AIC31XX_DOSRLSB, aic31xx_divs[i].dosr & 0xff); + + /* ADC dividers configuration. Write reset value 1 if not used. */ + snd_soc_update_bits(codec, AIC31XX_NADC, AIC31XX_PLL_MASK, + aic31xx_divs[i].nadc ? aic31xx_divs[i].nadc : 1); + snd_soc_update_bits(codec, AIC31XX_MADC, AIC31XX_PLL_MASK, + aic31xx_divs[i].madc ? aic31xx_divs[i].madc : 1); + + snd_soc_write(codec, AIC31XX_AOSR, aic31xx_divs[i].aosr); + + /* Bit clock divider configuration. */ + bclk_n = (aic31xx_divs[i].dosr * aic31xx_divs[i].mdac) + / snd_soc_params_to_frame_size(params); + if (bclk_n == 0) { + dev_err(codec->dev, "%s: Not enough BLCK bandwidth\n", + __func__); + return -EINVAL; + } + + snd_soc_update_bits(codec, AIC31XX_BCLKN, + AIC31XX_PLL_MASK, bclk_n); + + aic31xx->rate_div_line = i; + + dev_dbg(codec->dev, + "pll %d.%04d/%d dosr %d n %d m %d aosr %d n %d m %d bclk_n %d\n", + aic31xx_divs[i].pll_j, aic31xx_divs[i].pll_d, + aic31xx_divs[i].p_val, aic31xx_divs[i].dosr, + aic31xx_divs[i].ndac, aic31xx_divs[i].mdac, + aic31xx_divs[i].aosr, aic31xx_divs[i].nadc, + aic31xx_divs[i].madc, bclk_n); + + return 0; +} + +static int aic31xx_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *tmp) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + u8 data = 0; + + dev_dbg(codec->dev, "## %s: format %d width %d rate %d\n", + __func__, params_format(params), params_width(params), + params_rate(params)); + + switch (params_width(params)) { + case 16: + break; + case 20: + data = (AIC31XX_WORD_LEN_20BITS << + AIC31XX_IFACE1_DATALEN_SHIFT); + break; + case 24: + data = (AIC31XX_WORD_LEN_24BITS << + AIC31XX_IFACE1_DATALEN_SHIFT); + break; + case 32: + data = (AIC31XX_WORD_LEN_32BITS << + AIC31XX_IFACE1_DATALEN_SHIFT); + break; + default: + dev_err(codec->dev, "%s: Unsupported format %d\n", + __func__, params_format(params)); + return -EINVAL; + } + + snd_soc_update_bits(codec, AIC31XX_IFACE1, + AIC31XX_IFACE1_DATALEN_MASK, + data); + + return aic31xx_setup_pll(codec, params); +} + +static int aic31xx_dac_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + if (mute) { + snd_soc_update_bits(codec, AIC31XX_DACMUTE, + AIC31XX_DACMUTE_MASK, + AIC31XX_DACMUTE_MASK); + } else { + snd_soc_update_bits(codec, AIC31XX_DACMUTE, + AIC31XX_DACMUTE_MASK, 0x0); + } + + return 0; +} + +static int aic31xx_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + u8 iface_reg1 = 0; + u8 iface_reg3 = 0; + u8 dsp_a_val = 0; + + dev_dbg(codec->dev, "## %s: fmt = 0x%x\n", __func__, fmt); + + /* set master/slave audio interface */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + iface_reg1 |= AIC31XX_BCLK_MASTER | AIC31XX_WCLK_MASTER; + break; + default: + dev_alert(codec->dev, "Invalid DAI master/slave interface\n"); + return -EINVAL; + } + + /* interface format */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_DSP_A: + dsp_a_val = 0x1; + case SND_SOC_DAIFMT_DSP_B: + /* NOTE: BCLKINV bit value 1 equas NB and 0 equals IB */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + iface_reg3 |= AIC31XX_BCLKINV_MASK; + break; + case SND_SOC_DAIFMT_IB_NF: + break; + default: + return -EINVAL; + } + iface_reg1 |= (AIC31XX_DSP_MODE << + AIC31XX_IFACE1_DATATYPE_SHIFT); + break; + case SND_SOC_DAIFMT_RIGHT_J: + iface_reg1 |= (AIC31XX_RIGHT_JUSTIFIED_MODE << + AIC31XX_IFACE1_DATATYPE_SHIFT); + break; + case SND_SOC_DAIFMT_LEFT_J: + iface_reg1 |= (AIC31XX_LEFT_JUSTIFIED_MODE << + AIC31XX_IFACE1_DATATYPE_SHIFT); + break; + default: + dev_err(codec->dev, "Invalid DAI interface format\n"); + return -EINVAL; + } + + snd_soc_update_bits(codec, AIC31XX_IFACE1, + AIC31XX_IFACE1_DATATYPE_MASK | + AIC31XX_IFACE1_MASTER_MASK, + iface_reg1); + snd_soc_update_bits(codec, AIC31XX_DATA_OFFSET, + AIC31XX_DATA_OFFSET_MASK, + dsp_a_val); + snd_soc_update_bits(codec, AIC31XX_IFACE2, + AIC31XX_BCLKINV_MASK, + iface_reg3); + + return 0; +} + +static int aic31xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int i; + + dev_dbg(codec->dev, "## %s: clk_id = %d, freq = %d, dir = %d\n", + __func__, clk_id, freq, dir); + + for (i = 0; aic31xx_divs[i].mclk != freq; i++) { + if (i == ARRAY_SIZE(aic31xx_divs)) { + dev_err(aic31xx->dev, "%s: Unsupported frequency %d\n", + __func__, freq); + return -EINVAL; + } + } + + /* set clock on MCLK, BCLK, or GPIO1 as PLL input */ + snd_soc_update_bits(codec, AIC31XX_CLKMUX, AIC31XX_PLL_CLKIN_MASK, + clk_id << AIC31XX_PLL_CLKIN_SHIFT); + + aic31xx->sysclk = freq; + return 0; +} + +static int aic31xx_regulator_event(struct notifier_block *nb, + unsigned long event, void *data) +{ + struct aic31xx_disable_nb *disable_nb = + container_of(nb, struct aic31xx_disable_nb, nb); + struct aic31xx_priv *aic31xx = disable_nb->aic31xx; + + if (event & REGULATOR_EVENT_DISABLE) { + /* + * Put codec to reset and as at least one of the + * supplies was disabled. + */ + if (gpio_is_valid(aic31xx->pdata.gpio_reset)) + gpio_set_value(aic31xx->pdata.gpio_reset, 0); + + regcache_mark_dirty(aic31xx->regmap); + dev_dbg(aic31xx->dev, "## %s: DISABLE received\n", __func__); + } + + return 0; +} + +static void aic31xx_clk_on(struct snd_soc_codec *codec) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + u8 mask = AIC31XX_PM_MASK; + u8 on = AIC31XX_PM_MASK; + + dev_dbg(codec->dev, "codec clock -> on (rate %d)\n", + aic31xx_divs[aic31xx->rate_div_line].rate); + snd_soc_update_bits(codec, AIC31XX_PLLPR, mask, on); + mdelay(10); + snd_soc_update_bits(codec, AIC31XX_NDAC, mask, on); + snd_soc_update_bits(codec, AIC31XX_MDAC, mask, on); + if (aic31xx_divs[aic31xx->rate_div_line].nadc) + snd_soc_update_bits(codec, AIC31XX_NADC, mask, on); + if (aic31xx_divs[aic31xx->rate_div_line].madc) + snd_soc_update_bits(codec, AIC31XX_MADC, mask, on); + snd_soc_update_bits(codec, AIC31XX_BCLKN, mask, on); +} + +static void aic31xx_clk_off(struct snd_soc_codec *codec) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + u8 mask = AIC31XX_PM_MASK; + u8 off = 0; + + dev_dbg(codec->dev, "codec clock -> off\n"); + snd_soc_update_bits(codec, AIC31XX_BCLKN, mask, off); + snd_soc_update_bits(codec, AIC31XX_MADC, mask, off); + snd_soc_update_bits(codec, AIC31XX_NADC, mask, off); + snd_soc_update_bits(codec, AIC31XX_MDAC, mask, off); + snd_soc_update_bits(codec, AIC31XX_NDAC, mask, off); + snd_soc_update_bits(codec, AIC31XX_PLLPR, mask, off); +} + +static int aic31xx_power_on(struct snd_soc_codec *codec) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + ret = regulator_bulk_enable(ARRAY_SIZE(aic31xx->supplies), + aic31xx->supplies); + if (ret) + return ret; + + if (gpio_is_valid(aic31xx->pdata.gpio_reset)) { + gpio_set_value(aic31xx->pdata.gpio_reset, 1); + udelay(100); + } + regcache_cache_only(aic31xx->regmap, false); + ret = regcache_sync(aic31xx->regmap); + if (ret != 0) { + dev_err(codec->dev, + "Failed to restore cache: %d\n", ret); + regcache_cache_only(aic31xx->regmap, true); + regulator_bulk_disable(ARRAY_SIZE(aic31xx->supplies), + aic31xx->supplies); + return ret; + } + return 0; +} + +static int aic31xx_power_off(struct snd_soc_codec *codec) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int ret = 0; + + regcache_cache_only(aic31xx->regmap, true); + ret = regulator_bulk_disable(ARRAY_SIZE(aic31xx->supplies), + aic31xx->supplies); + + return ret; +} + +static int aic31xx_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + dev_dbg(codec->dev, "## %s: %d -> %d\n", __func__, + codec->dapm.bias_level, level); + + switch (level) { + case SND_SOC_BIAS_ON: + break; + case SND_SOC_BIAS_PREPARE: + if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) + aic31xx_clk_on(codec); + break; + case SND_SOC_BIAS_STANDBY: + switch (codec->dapm.bias_level) { + case SND_SOC_BIAS_OFF: + aic31xx_power_on(codec); + break; + case SND_SOC_BIAS_PREPARE: + aic31xx_clk_off(codec); + break; + default: + BUG(); + } + break; + case SND_SOC_BIAS_OFF: + aic31xx_power_off(codec); + break; + } + codec->dapm.bias_level = level; + + return 0; +} + +static int aic31xx_suspend(struct snd_soc_codec *codec) +{ + aic31xx_set_bias_level(codec, SND_SOC_BIAS_OFF); + return 0; +} + +static int aic31xx_resume(struct snd_soc_codec *codec) +{ + aic31xx_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + return 0; +} + +static int aic31xx_codec_probe(struct snd_soc_codec *codec) +{ + int ret = 0; + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int i; + + dev_dbg(aic31xx->dev, "## %s\n", __func__); + + aic31xx = snd_soc_codec_get_drvdata(codec); + codec->control_data = aic31xx->regmap; + + aic31xx->codec = codec; + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); + + if (ret != 0) { + dev_err(codec->dev, "snd_soc_codec_set_cache_io failed %d\n", + ret); + return ret; + } + + for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++) { + aic31xx->disable_nb[i].nb.notifier_call = + aic31xx_regulator_event; + aic31xx->disable_nb[i].aic31xx = aic31xx; + ret = regulator_register_notifier(aic31xx->supplies[i].consumer, + &aic31xx->disable_nb[i].nb); + if (ret) { + dev_err(codec->dev, + "Failed to request regulator notifier: %d\n", + ret); + return ret; + } + } + + regcache_cache_only(aic31xx->regmap, true); + regcache_mark_dirty(aic31xx->regmap); + + ret = aic31xx_add_controls(codec); + if (ret) + return ret; + + ret = aic31xx_add_widgets(codec); + + return ret; +} + +static int aic31xx_codec_remove(struct snd_soc_codec *codec) +{ + struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); + int i; + /* power down chip */ + aic31xx_set_bias_level(codec, SND_SOC_BIAS_OFF); + + for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++) + regulator_unregister_notifier(aic31xx->supplies[i].consumer, + &aic31xx->disable_nb[i].nb); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_driver_aic31xx = { + .probe = aic31xx_codec_probe, + .remove = aic31xx_codec_remove, + .suspend = aic31xx_suspend, + .resume = aic31xx_resume, + .set_bias_level = aic31xx_set_bias_level, + .controls = aic31xx_snd_controls, + .num_controls = ARRAY_SIZE(aic31xx_snd_controls), + .dapm_widgets = aic31xx_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(aic31xx_dapm_widgets), + .dapm_routes = aic31xx_audio_map, + .num_dapm_routes = ARRAY_SIZE(aic31xx_audio_map), +}; + +static struct snd_soc_dai_ops aic31xx_dai_ops = { + .hw_params = aic31xx_hw_params, + .set_sysclk = aic31xx_set_dai_sysclk, + .set_fmt = aic31xx_set_dai_fmt, + .digital_mute = aic31xx_dac_mute, +}; + +static struct snd_soc_dai_driver aic31xx_dai_driver[] = { + { + .name = "tlv320aic31xx-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = AIC31XX_RATES, + .formats = AIC31XX_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = AIC31XX_RATES, + .formats = AIC31XX_FORMATS, + }, + .ops = &aic31xx_dai_ops, + .symmetric_rates = 1, + } +}; + +#if defined(CONFIG_OF) +static const struct of_device_id tlv320aic31xx_of_match[] = { + { .compatible = "ti,tlv320aic310x" }, + { .compatible = "ti,tlv320aic311x" }, + { .compatible = "ti,tlv320aic3100" }, + { .compatible = "ti,tlv320aic3110" }, + { .compatible = "ti,tlv320aic3120" }, + { .compatible = "ti,tlv320aic3111" }, + {}, +}; +MODULE_DEVICE_TABLE(of, tlv320aic31xx_of_match); + +static void aic31xx_pdata_from_of(struct aic31xx_priv *aic31xx) +{ + struct device_node *np = aic31xx->dev->of_node; + unsigned int value = MICBIAS_2_0V; + int ret; + + of_property_read_u32(np, "ai31xx-micbias-vg", &value); + switch (value) { + case MICBIAS_2_0V: + case MICBIAS_2_5V: + case MICBIAS_AVDDV: + aic31xx->pdata.micbias_vg = value; + break; + default: + dev_err(aic31xx->dev, + "Bad ai31xx-micbias-vg value %d DT\n", + value); + aic31xx->pdata.micbias_vg = MICBIAS_2_0V; + } + + ret = of_get_named_gpio(np, "gpio-reset", 0); + if (ret > 0) + aic31xx->pdata.gpio_reset = ret; +} +#else /* CONFIG_OF */ +static void aic31xx_pdata_from_of(struct aic31xx_priv *aic31xx) +{ +} +#endif /* CONFIG_OF */ + +void aic31xx_device_init(struct aic31xx_priv *aic31xx) +{ + int ret, i; + + dev_set_drvdata(aic31xx->dev, aic31xx); + + if (dev_get_platdata(aic31xx->dev)) + memcpy(&aic31xx->pdata, dev_get_platdata(aic31xx->dev), + sizeof(aic31xx->pdata)); + else if (aic31xx->dev->of_node) + aic31xx_pdata_from_of(aic31xx); + + if (aic31xx->pdata.gpio_reset) { + ret = devm_gpio_request_one(aic31xx->dev, + aic31xx->pdata.gpio_reset, + GPIOF_OUT_INIT_HIGH, + "aic31xx-reset-pin"); + if (ret < 0) { + dev_err(aic31xx->dev, "not able to acquire gpio\n"); + return; + } + } + + for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++) + aic31xx->supplies[i].supply = aic31xx_supply_names[i]; + + ret = devm_regulator_bulk_get(aic31xx->dev, + ARRAY_SIZE(aic31xx->supplies), + aic31xx->supplies); + if (ret != 0) + dev_err(aic31xx->dev, "Failed to request supplies: %d\n", ret); + +} + +static int aic31xx_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct aic31xx_priv *aic31xx; + int ret; + const struct regmap_config *regmap_config; + + dev_dbg(&i2c->dev, "## %s: %s codec_type = %d\n", __func__, + id->name, (int) id->driver_data); + + regmap_config = &aic31xx_i2c_regmap; + + aic31xx = devm_kzalloc(&i2c->dev, sizeof(*aic31xx), GFP_KERNEL); + if (aic31xx == NULL) + return -ENOMEM; + + aic31xx->regmap = devm_regmap_init_i2c(i2c, regmap_config); + + if (IS_ERR(aic31xx->regmap)) { + ret = PTR_ERR(aic31xx->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + aic31xx->dev = &i2c->dev; + + aic31xx->pdata.codec_type = id->driver_data; + + aic31xx_device_init(aic31xx); + + ret = snd_soc_register_codec(&i2c->dev, &soc_codec_driver_aic31xx, + aic31xx_dai_driver, + ARRAY_SIZE(aic31xx_dai_driver)); + + return ret; +} + +static int aic31xx_i2c_remove(struct i2c_client *i2c) +{ + struct aic31xx_priv *aic31xx = dev_get_drvdata(&i2c->dev); + + kfree(aic31xx); + return 0; +} + +static const struct i2c_device_id aic31xx_i2c_id[] = { + { "tlv320aic310x", AIC3100 }, + { "tlv320aic311x", AIC3110 }, + { "tlv320aic3100", AIC3100 }, + { "tlv320aic3110", AIC3110 }, + { "tlv320aic3120", AIC3120 }, + { "tlv320aic3111", AIC3111 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, aic31xx_i2c_id); + +static struct i2c_driver aic31xx_i2c_driver = { + .driver = { + .name = "tlv320aic31xx-codec", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(tlv320aic31xx_of_match), + }, + .probe = aic31xx_i2c_probe, + .remove = (aic31xx_i2c_remove), + .id_table = aic31xx_i2c_id, +}; + +module_i2c_driver(aic31xx_i2c_driver); + +MODULE_DESCRIPTION("ASoC TLV320AIC3111 codec driver"); +MODULE_AUTHOR("Jyri Sarha"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h new file mode 100644 index 000000000000..52ed57c69dfa --- /dev/null +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -0,0 +1,258 @@ +/* + * ALSA SoC TLV320AIC31XX codec driver + * + * Copyright (C) 2013 Texas Instruments, Inc. + * + * This package is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + * + * THIS PACKAGE IS PROVIDED ``AS IS'' AND WITHOUT ANY EXPRESS OR + * IMPLIED WARRANTIES, INCLUDING, WITHOUT LIMITATION, THE IMPLIED + * WARRANTIES OF MERCHANTIBILITY AND FITNESS FOR A PARTICULAR PURPOSE. + * + */ +#ifndef _TLV320AIC31XX_H +#define _TLV320AIC31XX_H + +#define AIC31XX_RATES SNDRV_PCM_RATE_8000_192000 + +#define AIC31XX_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + + +#define AIC31XX_STEREO_CLASS_D_BIT 0x1 +#define AIC31XX_MINIDSP_BIT 0x2 + +enum aic31xx_type { + AIC3100 = 0, + AIC3110 = AIC31XX_STEREO_CLASS_D_BIT, + AIC3120 = AIC31XX_MINIDSP_BIT, + AIC3111 = (AIC31XX_STEREO_CLASS_D_BIT | AIC31XX_MINIDSP_BIT), +}; + +struct aic31xx_pdata { + enum aic31xx_type codec_type; + unsigned int gpio_reset; + int micbias_vg; +}; + +/* Page Control Register */ +#define AIC31XX_PAGECTL 0x00 + +/* Page 0 Registers */ +/* Software reset register */ +#define AIC31XX_RESET 0x01 +/* OT FLAG register */ +#define AIC31XX_OT_FLAG 0x03 +/* Clock clock Gen muxing, Multiplexers*/ +#define AIC31XX_CLKMUX 0x04 +/* PLL P and R-VAL register */ +#define AIC31XX_PLLPR 0x05 +/* PLL J-VAL register */ +#define AIC31XX_PLLJ 0x06 +/* PLL D-VAL MSB register */ +#define AIC31XX_PLLDMSB 0x07 +/* PLL D-VAL LSB register */ +#define AIC31XX_PLLDLSB 0x08 +/* DAC NDAC_VAL register*/ +#define AIC31XX_NDAC 0x0B +/* DAC MDAC_VAL register */ +#define AIC31XX_MDAC 0x0C +/* DAC OSR setting register 1, MSB value */ +#define AIC31XX_DOSRMSB 0x0D +/* DAC OSR setting register 2, LSB value */ +#define AIC31XX_DOSRLSB 0x0E +#define AIC31XX_MINI_DSP_INPOL 0x10 +/* Clock setting register 8, PLL */ +#define AIC31XX_NADC 0x12 +/* Clock setting register 9, PLL */ +#define AIC31XX_MADC 0x13 +/* ADC Oversampling (AOSR) Register */ +#define AIC31XX_AOSR 0x14 +/* Clock setting register 9, Multiplexers */ +#define AIC31XX_CLKOUTMUX 0x19 +/* Clock setting register 10, CLOCKOUT M divider value */ +#define AIC31XX_CLKOUTMVAL 0x1A +/* Audio Interface Setting Register 1 */ +#define AIC31XX_IFACE1 0x1B +/* Audio Data Slot Offset Programming */ +#define AIC31XX_DATA_OFFSET 0x1C +/* Audio Interface Setting Register 2 */ +#define AIC31XX_IFACE2 0x1D +/* Clock setting register 11, BCLK N Divider */ +#define AIC31XX_BCLKN 0x1E +/* Audio Interface Setting Register 3, Secondary Audio Interface */ +#define AIC31XX_IFACESEC1 0x1F +/* Audio Interface Setting Register 4 */ +#define AIC31XX_IFACESEC2 0x20 +/* Audio Interface Setting Register 5 */ +#define AIC31XX_IFACESEC3 0x21 +/* I2C Bus Condition */ +#define AIC31XX_I2C 0x22 +/* ADC FLAG */ +#define AIC31XX_ADCFLAG 0x24 +/* DAC Flag Registers */ +#define AIC31XX_DACFLAG1 0x25 +#define AIC31XX_DACFLAG2 0x26 +/* Sticky Interrupt flag (overflow) */ +#define AIC31XX_OFFLAG 0x27 +/* Sticy DAC Interrupt flags */ +#define AIC31XX_INTRDACFLAG 0x2C +/* Sticy ADC Interrupt flags */ +#define AIC31XX_INTRADCFLAG 0x2D +/* DAC Interrupt flags 2 */ +#define AIC31XX_INTRDACFLAG2 0x2E +/* ADC Interrupt flags 2 */ +#define AIC31XX_INTRADCFLAG2 0x2F +/* INT1 interrupt control */ +#define AIC31XX_INT1CTRL 0x30 +/* INT2 interrupt control */ +#define AIC31XX_INT2CTRL 0x31 +/* GPIO1 control */ +#define AIC31XX_GPIO1 0x33 + +#define AIC31XX_DACPRB 0x3C +/* ADC Instruction Set Register */ +#define AIC31XX_ADCPRB 0x3D +/* DAC channel setup register */ +#define AIC31XX_DACSETUP 0x3F +/* DAC Mute and volume control register */ +#define AIC31XX_DACMUTE 0x40 +/* Left DAC channel digital volume control */ +#define AIC31XX_LDACVOL 0x41 +/* Right DAC channel digital volume control */ +#define AIC31XX_RDACVOL 0x42 +/* Headset detection */ +#define AIC31XX_HSDETECT 0x43 +/* ADC Digital Mic */ +#define AIC31XX_ADCSETUP 0x51 +/* ADC Digital Volume Control Fine Adjust */ +#define AIC31XX_ADCFGA 0x52 +/* ADC Digital Volume Control Coarse Adjust */ +#define AIC31XX_ADCVOL 0x53 + + +/* Page 1 Registers */ +/* Headphone drivers */ +#define AIC31XX_HPDRIVER 0x9F +/* Class-D Speakear Amplifier */ +#define AIC31XX_SPKAMP 0xA0 +/* HP Output Drivers POP Removal Settings */ +#define AIC31XX_HPPOP 0xA1 +/* Output Driver PGA Ramp-Down Period Control */ +#define AIC31XX_SPPGARAMP 0xA2 +/* DAC_L and DAC_R Output Mixer Routing */ +#define AIC31XX_DACMIXERROUTE 0xA3 +/* Left Analog Vol to HPL */ +#define AIC31XX_LANALOGHPL 0xA4 +/* Right Analog Vol to HPR */ +#define AIC31XX_RANALOGHPR 0xA5 +/* Left Analog Vol to SPL */ +#define AIC31XX_LANALOGSPL 0xA6 +/* Right Analog Vol to SPR */ +#define AIC31XX_RANALOGSPR 0xA7 +/* HPL Driver */ +#define AIC31XX_HPLGAIN 0xA8 +/* HPR Driver */ +#define AIC31XX_HPRGAIN 0xA9 +/* SPL Driver */ +#define AIC31XX_SPLGAIN 0xAA +/* SPR Driver */ +#define AIC31XX_SPRGAIN 0xAB +/* HP Driver Control */ +#define AIC31XX_HPCONTROL 0xAC +/* MIC Bias Control */ +#define AIC31XX_MICBIAS 0xAE +/* MIC PGA*/ +#define AIC31XX_MICPGA 0xAF +/* Delta-Sigma Mono ADC Channel Fine-Gain Input Selection for P-Terminal */ +#define AIC31XX_MICPGAPI 0xB0 +/* ADC Input Selection for M-Terminal */ +#define AIC31XX_MICPGAMI 0xB1 +/* Input CM Settings */ +#define AIC31XX_MICPGACM 0xB2 + +/* Bits, masks and shifts */ + +/* AIC31XX_CLKMUX */ +#define AIC31XX_PLL_CLKIN_MASK 0x0c +#define AIC31XX_PLL_CLKIN_SHIFT 2 +#define AIC31XX_PLL_CLKIN_MCLK 0 +#define AIC31XX_CODEC_CLKIN_MASK 0x03 +#define AIC31XX_CODEC_CLKIN_SHIFT 0 +#define AIC31XX_CODEC_CLKIN_PLL 3 +#define AIC31XX_CODEC_CLKIN_BCLK 1 + +/* AIC31XX_PLLPR, AIC31XX_NDAC, AIC31XX_MDAC, AIC31XX_NADC, AIC31XX_MADC, + AIC31XX_BCLKN */ +#define AIC31XX_PLL_MASK 0x7f +#define AIC31XX_PM_MASK 0x80 + +/* AIC31XX_IFACE1 */ +#define AIC31XX_WORD_LEN_16BITS 0x00 +#define AIC31XX_WORD_LEN_20BITS 0x01 +#define AIC31XX_WORD_LEN_24BITS 0x02 +#define AIC31XX_WORD_LEN_32BITS 0x03 +#define AIC31XX_IFACE1_DATALEN_MASK 0x30 +#define AIC31XX_IFACE1_DATALEN_SHIFT (4) +#define AIC31XX_IFACE1_DATATYPE_MASK 0xC0 +#define AIC31XX_IFACE1_DATATYPE_SHIFT (6) +#define AIC31XX_I2S_MODE 0x00 +#define AIC31XX_DSP_MODE 0x01 +#define AIC31XX_RIGHT_JUSTIFIED_MODE 0x02 +#define AIC31XX_LEFT_JUSTIFIED_MODE 0x03 +#define AIC31XX_IFACE1_MASTER_MASK 0x0C +#define AIC31XX_BCLK_MASTER 0x08 +#define AIC31XX_WCLK_MASTER 0x04 + +/* AIC31XX_DATA_OFFSET */ +#define AIC31XX_DATA_OFFSET_MASK 0xFF + +/* AIC31XX_IFACE2 */ +#define AIC31XX_BCLKINV_MASK 0x08 +#define AIC31XX_BDIVCLK_MASK 0x03 +#define AIC31XX_DAC2BCLK 0x00 +#define AIC31XX_DACMOD2BCLK 0x01 +#define AIC31XX_ADC2BCLK 0x02 +#define AIC31XX_ADCMOD2BCLK 0x03 + +/* AIC31XX_ADCFLAG */ +#define AIC31XX_ADCPWRSTATUS_MASK 0x40 + +/* AIC31XX_DACFLAG1 */ +#define AIC31XX_LDACPWRSTATUS_MASK 0x80 +#define AIC31XX_RDACPWRSTATUS_MASK 0x08 +#define AIC31XX_HPLDRVPWRSTATUS_MASK 0x20 +#define AIC31XX_HPRDRVPWRSTATUS_MASK 0x02 +#define AIC31XX_SPLDRVPWRSTATUS_MASK 0x10 +#define AIC31XX_SPRDRVPWRSTATUS_MASK 0x01 + +/* AIC31XX_INTRDACFLAG */ +#define AIC31XX_HPSCDETECT_MASK 0x80 +#define AIC31XX_BUTTONPRESS_MASK 0x20 +#define AIC31XX_HSPLUG_MASK 0x10 +#define AIC31XX_LDRCTHRES_MASK 0x08 +#define AIC31XX_RDRCTHRES_MASK 0x04 +#define AIC31XX_DACSINT_MASK 0x02 +#define AIC31XX_DACAINT_MASK 0x01 + +/* AIC31XX_INT1CTRL */ +#define AIC31XX_HSPLUGDET_MASK 0x80 +#define AIC31XX_BUTTONPRESSDET_MASK 0x40 +#define AIC31XX_DRCTHRES_MASK 0x20 +#define AIC31XX_AGCNOISE_MASK 0x10 +#define AIC31XX_OC_MASK 0x08 +#define AIC31XX_ENGINE_MASK 0x04 + +/* AIC31XX_DACSETUP */ +#define AIC31XX_SOFTSTEP_MASK 0x03 + +/* AIC31XX_DACMUTE */ +#define AIC31XX_DACMUTE_MASK 0x0C + +/* AIC31XX_MICBIAS */ +#define AIC31XX_MICBIAS_MASK 0x03 +#define AIC31XX_MICBIAS_SHIFT 0 + +#endif /* _TLV320AIC31XX_H */ -- cgit v1.2.3 From a2d57678ce98534a87de42e55e599cae730d17ca Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Thu, 13 Mar 2014 17:37:52 +0200 Subject: ASoC: tlv320aic31xx: Fix unused variable warning from aic31xx_clk_off Fix "warning: unused variable 'aic31xx'" from function 'aic31xx_clk_off'. Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic31xx.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index e60e37b43a1b..bdc0d8bd47b4 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -943,7 +943,6 @@ static void aic31xx_clk_on(struct snd_soc_codec *codec) static void aic31xx_clk_off(struct snd_soc_codec *codec) { - struct aic31xx_priv *aic31xx = snd_soc_codec_get_drvdata(codec); u8 mask = AIC31XX_PM_MASK; u8 off = 0; -- cgit v1.2.3 From bc236fa7301c6ca0ccf5470a964842d1a60e786f Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Thu, 13 Mar 2014 18:22:35 +0200 Subject: ASoC: tlv320aic31xx: Remove snd_soc_codec_set_cache_io() call Remove snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP) call and codec->control_data = aic31xx->regmap assignment since that already done by core. Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic31xx.c | 9 --------- 1 file changed, 9 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index bdc0d8bd47b4..dcdc5751048f 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1049,18 +1049,9 @@ static int aic31xx_codec_probe(struct snd_soc_codec *codec) dev_dbg(aic31xx->dev, "## %s\n", __func__); aic31xx = snd_soc_codec_get_drvdata(codec); - codec->control_data = aic31xx->regmap; aic31xx->codec = codec; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - - if (ret != 0) { - dev_err(codec->dev, "snd_soc_codec_set_cache_io failed %d\n", - ret); - return ret; - } - for (i = 0; i < ARRAY_SIZE(aic31xx->supplies); i++) { aic31xx->disable_nb[i].nb.notifier_call = aic31xx_regulator_event; -- cgit v1.2.3 From 9296f4da3bafa23d8b9abc5cd271a66ea8f90cd2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 13 Mar 2014 17:44:22 +0000 Subject: ASoC: tlv320aic31xx: Staticise non-exported symbols Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic31xx.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index dcdc5751048f..d3517a919776 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -129,7 +129,7 @@ static const struct regmap_range_cfg aic31xx_ranges[] = { }, }; -struct regmap_config aic31xx_i2c_regmap = { +static const struct regmap_config aic31xx_i2c_regmap = { .reg_bits = 8, .val_bits = 8, .writeable_reg = aic31xx_writeable, @@ -321,9 +321,9 @@ static const struct snd_kcontrol_new ldac_in_control = static const struct snd_kcontrol_new rdac_in_control = SOC_DAPM_ENUM("DAC Right Input", rdac_in_enum); -int aic31xx_wait_bits(struct aic31xx_priv *aic31xx, unsigned int reg, - unsigned int mask, unsigned int wbits, int sleep, - int count) +static int aic31xx_wait_bits(struct aic31xx_priv *aic31xx, unsigned int reg, + unsigned int mask, unsigned int wbits, int sleep, + int count) { unsigned int bits; int counter = count; @@ -1177,7 +1177,7 @@ static void aic31xx_pdata_from_of(struct aic31xx_priv *aic31xx) } #endif /* CONFIG_OF */ -void aic31xx_device_init(struct aic31xx_priv *aic31xx) +static void aic31xx_device_init(struct aic31xx_priv *aic31xx) { int ret, i; -- cgit v1.2.3 From e585ca342dbbfe7102985d9ed4eae3f9e1d77ced Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 14 Mar 2014 09:33:36 +0100 Subject: ASoC: max98090: Remove unused control_data field The driver assigns a value to the control_data field of the driver's state struct, but never reads it again. Which means it is unused and can be removed. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 1 - sound/soc/codecs/max98090.h | 1 - 2 files changed, 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index c7b9e901bdac..361862d4fa65 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2350,7 +2350,6 @@ static int max98090_i2c_probe(struct i2c_client *i2c, max98090->devtype = id->driver_data; i2c_set_clientdata(i2c, max98090); - max98090->control_data = i2c; max98090->pdata = i2c->dev.platform_data; max98090->irq = i2c->irq; diff --git a/sound/soc/codecs/max98090.h b/sound/soc/codecs/max98090.h index 7e103f249053..1a4e2334a7b2 100644 --- a/sound/soc/codecs/max98090.h +++ b/sound/soc/codecs/max98090.h @@ -1523,7 +1523,6 @@ struct max98090_priv { struct regmap *regmap; struct snd_soc_codec *codec; enum max98090_type devtype; - void *control_data; struct max98090_pdata *pdata; unsigned int sysclk; unsigned int bclk; -- cgit v1.2.3 From fd218aa3e5d4ee522cbfe88ad4dd83eb891096fb Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Mon, 17 Mar 2014 09:31:31 +0200 Subject: ASoC: tlv320aic31xx: Turn power off only once. Regulator code keep count of enables and disables. Double disable causes an ugly warning. Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic31xx.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index d3517a919776..1f243c2c98fd 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1020,7 +1020,8 @@ static int aic31xx_set_bias_level(struct snd_soc_codec *codec, } break; case SND_SOC_BIAS_OFF: - aic31xx_power_off(codec); + if (codec->dapm.bias_level == SND_SOC_BIAS_STANDBY) + aic31xx_power_off(codec); break; } codec->dapm.bias_level = level; -- cgit v1.2.3 From bfe723f6eae285e399615d99f297d1646a6253fe Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Fri, 14 Mar 2014 09:26:11 +0100 Subject: ASoC: sirf-audio-codec: Remove snd_soc_codec_set_cache_io() call There was a overlap between the snd_soc_codec_set_cache_io() cleanup and the addition of the sirf-audio-codec resulting in the sirf-audio-codec driver still using the old signature of snd_soc_codec_set_cache_io(), which will cause a compile error. Since the core is able to automatically setup IO for this driver we can just remove both the snd_soc_set_cache_io() call and the control_data assignment. Signed-off-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/sirf-audio-codec.c | 9 --------- 1 file changed, 9 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/sirf-audio-codec.c b/sound/soc/codecs/sirf-audio-codec.c index 90e3a228bae4..58e7c1f23771 100644 --- a/sound/soc/codecs/sirf-audio-codec.c +++ b/sound/soc/codecs/sirf-audio-codec.c @@ -337,18 +337,9 @@ struct snd_soc_dai_driver sirf_audio_codec_dai = { static int sirf_audio_codec_probe(struct snd_soc_codec *codec) { - int ret; struct snd_soc_dapm_context *dapm = &codec->dapm; - struct sirf_audio_codec *sirf_audio_codec = snd_soc_codec_get_drvdata(codec); pm_runtime_enable(codec->dev); - codec->control_data = sirf_audio_codec->regmap; - - ret = snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); - if (ret != 0) { - dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); - return ret; - } if (of_device_is_compatible(codec->dev->of_node, "sirf,prima2-audio-codec")) { snd_soc_dapm_new_controls(dapm, -- cgit v1.2.3 From dac7e40404a6b1e7442c01ef4c2e7e149b9627e5 Mon Sep 17 00:00:00 2001 From: Axel Lin Date: Sun, 16 Mar 2014 23:06:25 +0800 Subject: ASoC: tlv320aic31xx: Don't call kfree for memory allocated by devm_kzalloc The kfree call is not necessary, but we need to call snd_soc_unregister_codec() in remove(). Signed-off-by: Axel Lin Acked-by: Jyri Sarha Signed-off-by: Mark Brown --- sound/soc/codecs/tlv320aic31xx.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index 1f243c2c98fd..e463ae7fe1f4 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1229,7 +1229,6 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, return -ENOMEM; aic31xx->regmap = devm_regmap_init_i2c(i2c, regmap_config); - if (IS_ERR(aic31xx->regmap)) { ret = PTR_ERR(aic31xx->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", @@ -1242,18 +1241,14 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, aic31xx_device_init(aic31xx); - ret = snd_soc_register_codec(&i2c->dev, &soc_codec_driver_aic31xx, + return snd_soc_register_codec(&i2c->dev, &soc_codec_driver_aic31xx, aic31xx_dai_driver, ARRAY_SIZE(aic31xx_dai_driver)); - - return ret; } static int aic31xx_i2c_remove(struct i2c_client *i2c) { - struct aic31xx_priv *aic31xx = dev_get_drvdata(&i2c->dev); - - kfree(aic31xx); + snd_soc_unregister_codec(&i2c->dev); return 0; } @@ -1275,7 +1270,7 @@ static struct i2c_driver aic31xx_i2c_driver = { .of_match_table = of_match_ptr(tlv320aic31xx_of_match), }, .probe = aic31xx_i2c_probe, - .remove = (aic31xx_i2c_remove), + .remove = aic31xx_i2c_remove, .id_table = aic31xx_i2c_id, }; -- cgit v1.2.3 From ab64246cf8c31f70a390dcabd134097c3aec45ab Mon Sep 17 00:00:00 2001 From: Lars-Peter Clausen Date: Thu, 13 Mar 2014 21:24:54 +0100 Subject: ASoC: codecs: Replace instances of rtd->codec with dai->codec With CODEC to CODEC links rtd->codec does not necessarily point to the driver's CODEC. CODEC drivers should always use dai->codec and never even look at the PCM runtime. Signed-off-by: Lars-Peter Clausen Acked-by: Charles Keepax Acked-by: Jyri Sarha Signed-off-by: Mark Brown --- sound/soc/codecs/isabelle.c | 3 +-- sound/soc/codecs/mc13783.c | 6 ++---- sound/soc/codecs/rt5640.c | 3 +-- sound/soc/codecs/sta529.c | 3 +-- sound/soc/codecs/tlv320aic31xx.c | 5 ++--- sound/soc/codecs/uda134x.c | 3 +-- sound/soc/codecs/uda1380.c | 3 +-- sound/soc/codecs/wm8580.c | 3 +-- 8 files changed, 10 insertions(+), 19 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/isabelle.c b/sound/soc/codecs/isabelle.c index 5839048ec467..7c8ba02a6d1b 100644 --- a/sound/soc/codecs/isabelle.c +++ b/sound/soc/codecs/isabelle.c @@ -910,8 +910,7 @@ static int isabelle_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 aif = 0; unsigned int fs_val = 0; diff --git a/sound/soc/codecs/mc13783.c b/sound/soc/codecs/mc13783.c index 582c2bbd42cb..bebba7fb8819 100644 --- a/sound/soc/codecs/mc13783.c +++ b/sound/soc/codecs/mc13783.c @@ -106,8 +106,7 @@ static int mc13783_pcm_hw_params_dac(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; unsigned int rate = params_rate(params); int i; @@ -126,8 +125,7 @@ static int mc13783_pcm_hw_params_codec(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; unsigned int rate = params_rate(params); unsigned int val; diff --git a/sound/soc/codecs/rt5640.c b/sound/soc/codecs/rt5640.c index a3fb41179636..7877a5e14c54 100644 --- a/sound/soc/codecs/rt5640.c +++ b/sound/soc/codecs/rt5640.c @@ -1601,8 +1601,7 @@ static int get_clk_info(int sclk, int rate) static int rt5640_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct rt5640_priv *rt5640 = snd_soc_codec_get_drvdata(codec); unsigned int val_len = 0, val_clk, mask_clk; int dai_sel, pre_div, bclk_ms, frame_size; diff --git a/sound/soc/codecs/sta529.c b/sound/soc/codecs/sta529.c index 40c07be9b581..fa4b050fe6e6 100644 --- a/sound/soc/codecs/sta529.c +++ b/sound/soc/codecs/sta529.c @@ -193,8 +193,7 @@ static int sta529_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; int pdata, play_freq_val, record_freq_val; int bclk_to_fs_ratio; diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index e463ae7fe1f4..fa158cfe9b32 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -753,10 +753,9 @@ static int aic31xx_setup_pll(struct snd_soc_codec *codec, static int aic31xx_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, - struct snd_soc_dai *tmp) + struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u8 data = 0; dev_dbg(codec->dev, "## %s: format %d width %d rate %d\n", diff --git a/sound/soc/codecs/uda134x.c b/sound/soc/codecs/uda134x.c index c94d4c1e3dac..edf27acc1d77 100644 --- a/sound/soc/codecs/uda134x.c +++ b/sound/soc/codecs/uda134x.c @@ -203,8 +203,7 @@ static int uda134x_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct uda134x_priv *uda134x = snd_soc_codec_get_drvdata(codec); u8 hw_params; diff --git a/sound/soc/codecs/uda1380.c b/sound/soc/codecs/uda1380.c index 726df6d43c2b..dc47f6f5cd83 100644 --- a/sound/soc/codecs/uda1380.c +++ b/sound/soc/codecs/uda1380.c @@ -564,8 +564,7 @@ static int uda1380_pcm_hw_params(struct snd_pcm_substream *substream, static void uda1380_pcm_shutdown(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; u16 clk = uda1380_read_reg_cache(codec, UDA1380_CLK); /* shut down WSPLL power if running from this clock */ diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 318989acbbe5..49c2869b681c 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -504,8 +504,7 @@ static int wm8580_paif_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct wm8580_priv *wm8580 = snd_soc_codec_get_drvdata(codec); u16 paifa = 0; u16 paifb = 0; -- cgit v1.2.3 From 1555b652970e541fa1cb80c61ffc696bbfb92bb7 Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Tue, 18 Mar 2014 13:56:21 -0500 Subject: ASoC: cs42l73: Fix mask bits for SOC_VALUE_ENUM_SINGLE The mask bits values were wrong for the SOC_VALUE_ENUM_SINGLE for the mono mix controls. Reported-by: Takashi Iwai Signed-off-by: Brian Austin Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/cs42l73.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 549d5d6a3fef..7b95f7cbc515 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -319,7 +319,7 @@ static const char * const cs42l73_mono_mix_texts[] = { static const unsigned int cs42l73_mono_mix_values[] = { 0, 1, 2 }; static const struct soc_enum spk_asp_enum = - SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 6, 1, + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 6, 3, ARRAY_SIZE(cs42l73_mono_mix_texts), cs42l73_mono_mix_texts, cs42l73_mono_mix_values); @@ -337,7 +337,7 @@ static const struct snd_kcontrol_new spk_xsp_mixer = SOC_DAPM_ENUM("Route", spk_xsp_enum); static const struct soc_enum esl_asp_enum = - SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 2, 5, + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 2, 3, ARRAY_SIZE(cs42l73_mono_mix_texts), cs42l73_mono_mix_texts, cs42l73_mono_mix_values); @@ -346,7 +346,7 @@ static const struct snd_kcontrol_new esl_asp_mixer = SOC_DAPM_ENUM("Route", esl_asp_enum); static const struct soc_enum esl_xsp_enum = - SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 0, 7, + SOC_VALUE_ENUM_SINGLE(CS42L73_MMIXCTL, 0, 3, ARRAY_SIZE(cs42l73_mono_mix_texts), cs42l73_mono_mix_texts, cs42l73_mono_mix_values); -- cgit v1.2.3 From d31a33dd7792c7d6c11fda226a3b9e4fb7f86f95 Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Tue, 18 Mar 2014 14:01:47 -0500 Subject: ASoC: cs42l52: Fix mask bits for SOC_VALUE_ENUM_SINGLE The mask bits values were wrong for the SOC_VALUE_ENUM_SINGLE for the PCM/ADC Swap controls Reported-by: Takashi Iwai Signed-off-by: Brian Austin Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/cs42l52.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/cs42l52.c b/sound/soc/codecs/cs42l52.c index 0bac6d5a4ac8..1102ced9b20e 100644 --- a/sound/soc/codecs/cs42l52.c +++ b/sound/soc/codecs/cs42l52.c @@ -347,7 +347,7 @@ static const char * const right_swap_text[] = { static const unsigned int swap_values[] = { 0, 1, 3 }; static const struct soc_enum adca_swap_enum = - SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 2, 1, + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 2, 3, ARRAY_SIZE(left_swap_text), left_swap_text, swap_values); @@ -356,7 +356,7 @@ static const struct snd_kcontrol_new adca_mixer = SOC_DAPM_ENUM("Route", adca_swap_enum); static const struct soc_enum pcma_swap_enum = - SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 6, 1, + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 6, 3, ARRAY_SIZE(left_swap_text), left_swap_text, swap_values); @@ -365,7 +365,7 @@ static const struct snd_kcontrol_new pcma_mixer = SOC_DAPM_ENUM("Route", pcma_swap_enum); static const struct soc_enum adcb_swap_enum = - SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 0, 1, + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 0, 3, ARRAY_SIZE(right_swap_text), right_swap_text, swap_values); @@ -374,7 +374,7 @@ static const struct snd_kcontrol_new adcb_mixer = SOC_DAPM_ENUM("Route", adcb_swap_enum); static const struct soc_enum pcmb_swap_enum = - SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 4, 1, + SOC_VALUE_ENUM_SINGLE(CS42L52_ADC_PCM_MIXER, 4, 3, ARRAY_SIZE(right_swap_text), right_swap_text, swap_values); -- cgit v1.2.3 From 7272e051157ccd5871b5d939548d0ba5a94a2965 Mon Sep 17 00:00:00 2001 From: Brian Austin Date: Wed, 19 Mar 2014 10:40:02 -0500 Subject: ASoC: cs42l51: Fix SOC_DOUBLE_R_SX_TLV shift values for ADC, PCM, and Analog kcontrols The shift values for the ADC,PCM, and Analog kcontrols were wrong causing wrong values for the SOC_DOUBLE_R_SX_TLV macros Fixed the TLV for aout_tlv to show -102dB correctly Fixes: 1d99f2436d (ASoC: core: Rework SOC_DOUBLE_R_SX_TLV add SOC_SINGLE_SX_TLV) Reported-by: Thomas Petazzoni Signed-off-by: Brian Austin Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/codecs/cs42l51.c | 9 ++++----- 1 file changed, 4 insertions(+), 5 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/cs42l51.c b/sound/soc/codecs/cs42l51.c index 6e9ea8379a91..7a272fa90b39 100644 --- a/sound/soc/codecs/cs42l51.c +++ b/sound/soc/codecs/cs42l51.c @@ -124,9 +124,8 @@ static int cs42l51_set_chan_mix(struct snd_kcontrol *kcontrol, static const DECLARE_TLV_DB_SCALE(adc_pcm_tlv, -5150, 50, 0); static const DECLARE_TLV_DB_SCALE(tone_tlv, -1050, 150, 0); -/* This is a lie. after -102 db, it stays at -102 */ -/* maybe a range would be better */ -static const DECLARE_TLV_DB_SCALE(aout_tlv, -11550, 50, 0); + +static const DECLARE_TLV_DB_SCALE(aout_tlv, -10200, 50, 0); static const DECLARE_TLV_DB_SCALE(boost_tlv, 1600, 1600, 0); static const char *chan_mix[] = { @@ -141,7 +140,7 @@ static const struct soc_enum cs42l51_chan_mix = static const struct snd_kcontrol_new cs42l51_snd_controls[] = { SOC_DOUBLE_R_SX_TLV("PCM Playback Volume", CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, - 6, 0x19, 0x7F, adc_pcm_tlv), + 0, 0x19, 0x7F, adc_pcm_tlv), SOC_DOUBLE_R("PCM Playback Switch", CS42L51_PCMA_VOL, CS42L51_PCMB_VOL, 7, 1, 1), SOC_DOUBLE_R_SX_TLV("Analog Playback Volume", @@ -149,7 +148,7 @@ static const struct snd_kcontrol_new cs42l51_snd_controls[] = { 0, 0x34, 0xE4, aout_tlv), SOC_DOUBLE_R_SX_TLV("ADC Mixer Volume", CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, - 6, 0x19, 0x7F, adc_pcm_tlv), + 0, 0x19, 0x7F, adc_pcm_tlv), SOC_DOUBLE_R("ADC Mixer Switch", CS42L51_ADCA_VOL, CS42L51_ADCB_VOL, 7, 1, 1), SOC_SINGLE("Playback Deemphasis Switch", CS42L51_DAC_CTL, 3, 1, 0), -- cgit v1.2.3 From 0c516b4ff85c0be4cee5b30ae59c9565c7f91a00 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Thu, 20 Mar 2014 18:18:37 +0800 Subject: ASoC: cs42xx8: Add codec driver support for CS42448/CS42888 This patch adds support for the Cirrus Logic CS42448/CS42888 Audio CODEC that has six/four 24-bit AD and eight 24-bit DA converters. [ CS42448/CS42888 supports both I2C and SPI control ports. As initial patch, this patch only adds the support for I2C. ] Signed-off-by: Nicolin Chen Acked-by: Brian Austin Acked-by: Paul Handrigan Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/cs42xx8.txt | 28 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/cs42xx8-i2c.c | 64 +++ sound/soc/codecs/cs42xx8.c | 602 +++++++++++++++++++++ sound/soc/codecs/cs42xx8.h | 238 ++++++++ 6 files changed, 946 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/cs42xx8.txt create mode 100644 sound/soc/codecs/cs42xx8-i2c.c create mode 100644 sound/soc/codecs/cs42xx8.c create mode 100644 sound/soc/codecs/cs42xx8.h (limited to 'sound/soc/codecs') diff --git a/Documentation/devicetree/bindings/sound/cs42xx8.txt b/Documentation/devicetree/bindings/sound/cs42xx8.txt new file mode 100644 index 000000000000..f631fbca6284 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/cs42xx8.txt @@ -0,0 +1,28 @@ +CS42448/CS42888 audio CODEC + +Required properties: + + - compatible : must contain one of "cirrus,cs42448" and "cirrus,cs42888" + + - reg : the I2C address of the device for I2C + + - clocks : a list of phandles + clock-specifiers, one for each entry in + clock-names + + - clock-names : must contain "mclk" + + - VA-supply, VD-supply, VLS-supply, VLC-supply: power supplies for the device, + as covered in Documentation/devicetree/bindings/regulator/regulator.txt + +Example: + +codec: cs42888@48 { + compatible = "cirrus,cs42888"; + reg = <0x48>; + clocks = <&codec_mclk 0>; + clock-names = "mclk"; + VA-supply = <®_audio>; + VD-supply = <®_audio>; + VLS-supply = <®_audio>; + VLC-supply = <®_audio>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 983d087aa92a..a79c0d141f90 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -37,6 +37,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_CS42L73 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_CS4271 if SND_SOC_I2C_AND_SPI + select SND_SOC_CS42XX8_I2C if I2C select SND_SOC_CX20442 if TTY select SND_SOC_DA7210 if I2C select SND_SOC_DA7213 if I2C @@ -254,6 +255,15 @@ config SND_SOC_CS4270_VD33_ERRATA config SND_SOC_CS4271 tristate +config SND_SOC_CS42XX8 + tristate + +config SND_SOC_CS42XX8_I2C + tristate "Cirrus Logic CS42448/CS42888 CODEC (I2C)" + depends on I2C + select SND_SOC_CS42XX8 + select REGMAP_I2C + config SND_SOC_CX20442 tristate depends on TTY diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index bc126764a44d..cfe5d634c812 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -23,6 +23,8 @@ snd-soc-cs42l52-objs := cs42l52.o snd-soc-cs42l73-objs := cs42l73.o snd-soc-cs4270-objs := cs4270.o snd-soc-cs4271-objs := cs4271.o +snd-soc-cs42xx8-objs := cs42xx8.o +snd-soc-cs42xx8-i2c-objs := cs42xx8-i2c.o snd-soc-cx20442-objs := cx20442.o snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o @@ -156,6 +158,8 @@ obj-$(CONFIG_SND_SOC_CS42L52) += snd-soc-cs42l52.o obj-$(CONFIG_SND_SOC_CS42L73) += snd-soc-cs42l73.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CS4271) += snd-soc-cs4271.o +obj-$(CONFIG_SND_SOC_CS42XX8) += snd-soc-cs42xx8.o +obj-$(CONFIG_SND_SOC_CS42XX8_I2C) += snd-soc-cs42xx8-i2c.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o diff --git a/sound/soc/codecs/cs42xx8-i2c.c b/sound/soc/codecs/cs42xx8-i2c.c new file mode 100644 index 000000000000..657dce27eade --- /dev/null +++ b/sound/soc/codecs/cs42xx8-i2c.c @@ -0,0 +1,64 @@ +/* + * Cirrus Logic CS42448/CS42888 Audio CODEC DAI I2C driver + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include +#include +#include +#include + +#include "cs42xx8.h" + +static int cs42xx8_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + u32 ret = cs42xx8_probe(&i2c->dev, + devm_regmap_init_i2c(i2c, &cs42xx8_regmap_config)); + if (ret) + return ret; + + pm_runtime_enable(&i2c->dev); + pm_request_idle(&i2c->dev); + + return 0; +} + +static int cs42xx8_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + pm_runtime_disable(&i2c->dev); + + return 0; +} + +static struct i2c_device_id cs42xx8_i2c_id[] = { + {"cs42448", (kernel_ulong_t)&cs42448_data}, + {"cs42888", (kernel_ulong_t)&cs42888_data}, + {} +}; +MODULE_DEVICE_TABLE(i2c, cs42xx8_i2c_id); + +static struct i2c_driver cs42xx8_i2c_driver = { + .driver = { + .name = "cs42xx8", + .owner = THIS_MODULE, + .pm = &cs42xx8_pm, + }, + .probe = cs42xx8_i2c_probe, + .remove = cs42xx8_i2c_remove, + .id_table = cs42xx8_i2c_id, +}; + +module_i2c_driver(cs42xx8_i2c_driver); + +MODULE_DESCRIPTION("Cirrus Logic CS42448/CS42888 ALSA SoC Codec I2C Driver"); +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c new file mode 100644 index 000000000000..082299a4e2fa --- /dev/null +++ b/sound/soc/codecs/cs42xx8.c @@ -0,0 +1,602 @@ +/* + * Cirrus Logic CS42448/CS42888 Audio CODEC Digital Audio Interface (DAI) driver + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "cs42xx8.h" + +#define CS42XX8_NUM_SUPPLIES 4 +static const char *const cs42xx8_supply_names[CS42XX8_NUM_SUPPLIES] = { + "VA", + "VD", + "VLS", + "VLC", +}; + +#define CS42XX8_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +/* codec private data */ +struct cs42xx8_priv { + struct regulator_bulk_data supplies[CS42XX8_NUM_SUPPLIES]; + const struct cs42xx8_driver_data *drvdata; + struct regmap *regmap; + struct clk *clk; + + bool slave_mode; + unsigned long sysclk; +}; + +/* -127.5dB to 0dB with step of 0.5dB */ +static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1); +/* -64dB to 24dB with step of 0.5dB */ +static const DECLARE_TLV_DB_SCALE(adc_tlv, -6400, 50, 0); + +static const char *const cs42xx8_adc_single[] = { "Differential", "Single-Ended" }; +static const char *const cs42xx8_szc[] = { "Immediate Change", "Zero Cross", + "Soft Ramp", "Soft Ramp on Zero Cross" }; + +static const struct soc_enum adc1_single_enum = + SOC_ENUM_SINGLE(CS42XX8_ADCCTL, 4, 2, cs42xx8_adc_single); +static const struct soc_enum adc2_single_enum = + SOC_ENUM_SINGLE(CS42XX8_ADCCTL, 3, 2, cs42xx8_adc_single); +static const struct soc_enum adc3_single_enum = + SOC_ENUM_SINGLE(CS42XX8_ADCCTL, 2, 2, cs42xx8_adc_single); +static const struct soc_enum dac_szc_enum = + SOC_ENUM_SINGLE(CS42XX8_TXCTL, 5, 4, cs42xx8_szc); +static const struct soc_enum adc_szc_enum = + SOC_ENUM_SINGLE(CS42XX8_TXCTL, 0, 4, cs42xx8_szc); + +static const struct snd_kcontrol_new cs42xx8_snd_controls[] = { + SOC_DOUBLE_R_TLV("DAC1 Playback Volume", CS42XX8_VOLAOUT1, + CS42XX8_VOLAOUT2, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC2 Playback Volume", CS42XX8_VOLAOUT3, + CS42XX8_VOLAOUT4, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC3 Playback Volume", CS42XX8_VOLAOUT5, + CS42XX8_VOLAOUT6, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_TLV("DAC4 Playback Volume", CS42XX8_VOLAOUT7, + CS42XX8_VOLAOUT8, 0, 0xff, 1, dac_tlv), + SOC_DOUBLE_R_S_TLV("ADC1 Capture Volume", CS42XX8_VOLAIN1, + CS42XX8_VOLAIN2, 0, -0x80, 0x30, 7, 0, adc_tlv), + SOC_DOUBLE_R_S_TLV("ADC2 Capture Volume", CS42XX8_VOLAIN3, + CS42XX8_VOLAIN4, 0, -0x80, 0x30, 7, 0, adc_tlv), + SOC_DOUBLE("DAC1 Invert Switch", CS42XX8_DACINV, 0, 1, 1, 0), + SOC_DOUBLE("DAC2 Invert Switch", CS42XX8_DACINV, 2, 3, 1, 0), + SOC_DOUBLE("DAC3 Invert Switch", CS42XX8_DACINV, 4, 5, 1, 0), + SOC_DOUBLE("DAC4 Invert Switch", CS42XX8_DACINV, 6, 7, 1, 0), + SOC_DOUBLE("ADC1 Invert Switch", CS42XX8_ADCINV, 0, 1, 1, 0), + SOC_DOUBLE("ADC2 Invert Switch", CS42XX8_ADCINV, 2, 3, 1, 0), + SOC_SINGLE("ADC High-Pass Filter Switch", CS42XX8_ADCCTL, 7, 1, 1), + SOC_SINGLE("DAC De-emphasis Switch", CS42XX8_ADCCTL, 5, 1, 0), + SOC_ENUM("ADC1 Single Ended Mode Switch", adc1_single_enum), + SOC_ENUM("ADC2 Single Ended Mode Switch", adc2_single_enum), + SOC_SINGLE("DAC Single Volume Control Switch", CS42XX8_TXCTL, 7, 1, 0), + SOC_ENUM("DAC Soft Ramp & Zero Cross Control Switch", dac_szc_enum), + SOC_SINGLE("DAC Auto Mute Switch", CS42XX8_TXCTL, 4, 1, 0), + SOC_SINGLE("Mute ADC Serial Port Switch", CS42XX8_TXCTL, 3, 1, 0), + SOC_SINGLE("ADC Single Volume Control Switch", CS42XX8_TXCTL, 2, 1, 0), + SOC_ENUM("ADC Soft Ramp & Zero Cross Control Switch", adc_szc_enum), +}; + +static const struct snd_kcontrol_new cs42xx8_adc3_snd_controls[] = { + SOC_DOUBLE_R_S_TLV("ADC3 Capture Volume", CS42XX8_VOLAIN5, + CS42XX8_VOLAIN6, 0, -0x80, 0x30, 7, 0, adc_tlv), + SOC_DOUBLE("ADC3 Invert Switch", CS42XX8_ADCINV, 4, 5, 1, 0), + SOC_ENUM("ADC3 Single Ended Mode Switch", adc3_single_enum), +}; + +static const struct snd_soc_dapm_widget cs42xx8_dapm_widgets[] = { + SND_SOC_DAPM_DAC("DAC1", "Playback", CS42XX8_PWRCTL, 1, 1), + SND_SOC_DAPM_DAC("DAC2", "Playback", CS42XX8_PWRCTL, 2, 1), + SND_SOC_DAPM_DAC("DAC3", "Playback", CS42XX8_PWRCTL, 3, 1), + SND_SOC_DAPM_DAC("DAC4", "Playback", CS42XX8_PWRCTL, 4, 1), + + SND_SOC_DAPM_OUTPUT("AOUT1L"), + SND_SOC_DAPM_OUTPUT("AOUT1R"), + SND_SOC_DAPM_OUTPUT("AOUT2L"), + SND_SOC_DAPM_OUTPUT("AOUT2R"), + SND_SOC_DAPM_OUTPUT("AOUT3L"), + SND_SOC_DAPM_OUTPUT("AOUT3R"), + SND_SOC_DAPM_OUTPUT("AOUT4L"), + SND_SOC_DAPM_OUTPUT("AOUT4R"), + + SND_SOC_DAPM_ADC("ADC1", "Capture", CS42XX8_PWRCTL, 5, 1), + SND_SOC_DAPM_ADC("ADC2", "Capture", CS42XX8_PWRCTL, 6, 1), + + SND_SOC_DAPM_INPUT("AIN1L"), + SND_SOC_DAPM_INPUT("AIN1R"), + SND_SOC_DAPM_INPUT("AIN2L"), + SND_SOC_DAPM_INPUT("AIN2R"), + + SND_SOC_DAPM_SUPPLY("PWR", CS42XX8_PWRCTL, 0, 1, NULL, 0), +}; + +static const struct snd_soc_dapm_widget cs42xx8_adc3_dapm_widgets[] = { + SND_SOC_DAPM_ADC("ADC3", "Capture", CS42XX8_PWRCTL, 7, 1), + + SND_SOC_DAPM_INPUT("AIN3L"), + SND_SOC_DAPM_INPUT("AIN3R"), +}; + +static const struct snd_soc_dapm_route cs42xx8_dapm_routes[] = { + /* Playback */ + { "AOUT1L", NULL, "DAC1" }, + { "AOUT1R", NULL, "DAC1" }, + { "DAC1", NULL, "PWR" }, + + { "AOUT2L", NULL, "DAC2" }, + { "AOUT2R", NULL, "DAC2" }, + { "DAC2", NULL, "PWR" }, + + { "AOUT3L", NULL, "DAC3" }, + { "AOUT3R", NULL, "DAC3" }, + { "DAC3", NULL, "PWR" }, + + { "AOUT4L", NULL, "DAC4" }, + { "AOUT4R", NULL, "DAC4" }, + { "DAC4", NULL, "PWR" }, + + /* Capture */ + { "ADC1", NULL, "AIN1L" }, + { "ADC1", NULL, "AIN1R" }, + { "ADC1", NULL, "PWR" }, + + { "ADC2", NULL, "AIN2L" }, + { "ADC2", NULL, "AIN2R" }, + { "ADC2", NULL, "PWR" }, +}; + +static const struct snd_soc_dapm_route cs42xx8_adc3_dapm_routes[] = { + /* Capture */ + { "ADC3", NULL, "AIN3L" }, + { "ADC3", NULL, "AIN3R" }, + { "ADC3", NULL, "PWR" }, +}; + +struct cs42xx8_ratios { + unsigned int ratio; + unsigned char speed; + unsigned char mclk; +}; + +static const struct cs42xx8_ratios cs42xx8_ratios[] = { + { 64, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_256(4) }, + { 96, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_384(4) }, + { 128, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_512(4) }, + { 192, CS42XX8_FM_QUAD, CS42XX8_FUNCMOD_MFREQ_768(4) }, + { 256, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_256(1) }, + { 384, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_384(1) }, + { 512, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_512(1) }, + { 768, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_768(1) }, + { 1024, CS42XX8_FM_SINGLE, CS42XX8_FUNCMOD_MFREQ_1024(1) } +}; + +static int cs42xx8_set_dai_sysclk(struct snd_soc_dai *codec_dai, + int clk_id, unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec); + + cs42xx8->sysclk = freq; + + return 0; +} + +static int cs42xx8_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec); + u32 val; + + /* Set DAI format */ + switch (format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_LEFT_J: + val = CS42XX8_INTF_DAC_DIF_LEFTJ | CS42XX8_INTF_ADC_DIF_LEFTJ; + break; + case SND_SOC_DAIFMT_I2S: + val = CS42XX8_INTF_DAC_DIF_I2S | CS42XX8_INTF_ADC_DIF_I2S; + break; + case SND_SOC_DAIFMT_RIGHT_J: + val = CS42XX8_INTF_DAC_DIF_RIGHTJ | CS42XX8_INTF_ADC_DIF_RIGHTJ; + break; + default: + dev_err(codec->dev, "unsupported dai format\n"); + return -EINVAL; + } + + regmap_update_bits(cs42xx8->regmap, CS42XX8_INTF, + CS42XX8_INTF_DAC_DIF_MASK | + CS42XX8_INTF_ADC_DIF_MASK, val); + + /* Set master/slave audio interface */ + switch (format & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + cs42xx8->slave_mode = true; + break; + case SND_SOC_DAIFMT_CBM_CFM: + cs42xx8->slave_mode = false; + break; + default: + dev_err(codec->dev, "unsupported master/slave mode\n"); + return -EINVAL; + } + + return 0; +} + +static int cs42xx8_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_codec *codec = rtd->codec; + struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + u32 ratio = cs42xx8->sysclk / params_rate(params); + u32 i, fm, val, mask; + + for (i = 0; i < ARRAY_SIZE(cs42xx8_ratios); i++) { + if (cs42xx8_ratios[i].ratio == ratio) + break; + } + + if (i == ARRAY_SIZE(cs42xx8_ratios)) { + dev_err(codec->dev, "unsupported sysclk ratio\n"); + return -EINVAL; + } + + mask = CS42XX8_FUNCMOD_MFREQ_MASK; + val = cs42xx8_ratios[i].mclk; + + fm = cs42xx8->slave_mode ? CS42XX8_FM_AUTO : cs42xx8_ratios[i].speed; + + regmap_update_bits(cs42xx8->regmap, CS42XX8_FUNCMOD, + CS42XX8_FUNCMOD_xC_FM_MASK(tx) | mask, + CS42XX8_FUNCMOD_xC_FM(tx, fm) | val); + + return 0; +} + +static int cs42xx8_digital_mute(struct snd_soc_dai *dai, int mute) +{ + struct snd_soc_codec *codec = dai->codec; + struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(cs42xx8->regmap, CS42XX8_DACMUTE, + CS42XX8_DACMUTE_ALL, mute ? CS42XX8_DACMUTE_ALL : 0); + + return 0; +} + +static const struct snd_soc_dai_ops cs42xx8_dai_ops = { + .set_fmt = cs42xx8_set_dai_fmt, + .set_sysclk = cs42xx8_set_dai_sysclk, + .hw_params = cs42xx8_hw_params, + .digital_mute = cs42xx8_digital_mute, +}; + +static struct snd_soc_dai_driver cs42xx8_dai = { + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = CS42XX8_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = CS42XX8_FORMATS, + }, + .ops = &cs42xx8_dai_ops, +}; + +static const struct reg_default cs42xx8_reg[] = { + { 0x01, 0x01 }, /* Chip I.D. and Revision Register */ + { 0x02, 0x00 }, /* Power Control */ + { 0x03, 0xF0 }, /* Functional Mode */ + { 0x04, 0x46 }, /* Interface Formats */ + { 0x05, 0x00 }, /* ADC Control & DAC De-Emphasis */ + { 0x06, 0x10 }, /* Transition Control */ + { 0x07, 0x00 }, /* DAC Channel Mute */ + { 0x08, 0x00 }, /* Volume Control AOUT1 */ + { 0x09, 0x00 }, /* Volume Control AOUT2 */ + { 0x0a, 0x00 }, /* Volume Control AOUT3 */ + { 0x0b, 0x00 }, /* Volume Control AOUT4 */ + { 0x0c, 0x00 }, /* Volume Control AOUT5 */ + { 0x0d, 0x00 }, /* Volume Control AOUT6 */ + { 0x0e, 0x00 }, /* Volume Control AOUT7 */ + { 0x0f, 0x00 }, /* Volume Control AOUT8 */ + { 0x10, 0x00 }, /* DAC Channel Invert */ + { 0x11, 0x00 }, /* Volume Control AIN1 */ + { 0x12, 0x00 }, /* Volume Control AIN2 */ + { 0x13, 0x00 }, /* Volume Control AIN3 */ + { 0x14, 0x00 }, /* Volume Control AIN4 */ + { 0x15, 0x00 }, /* Volume Control AIN5 */ + { 0x16, 0x00 }, /* Volume Control AIN6 */ + { 0x17, 0x00 }, /* ADC Channel Invert */ + { 0x18, 0x00 }, /* Status Control */ + { 0x1a, 0x00 }, /* Status Mask */ + { 0x1b, 0x00 }, /* MUTEC Pin Control */ +}; + +static bool cs42xx8_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42XX8_STATUS: + return true; + default: + return false; + } +} + +static bool cs42xx8_writeable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case CS42XX8_CHIPID: + case CS42XX8_STATUS: + return false; + default: + return true; + } +} + +const struct regmap_config cs42xx8_regmap_config = { + .reg_bits = 8, + .val_bits = 8, + + .max_register = CS42XX8_LASTREG, + .reg_defaults = cs42xx8_reg, + .num_reg_defaults = ARRAY_SIZE(cs42xx8_reg), + .volatile_reg = cs42xx8_volatile_register, + .writeable_reg = cs42xx8_writeable_register, + .cache_type = REGCACHE_RBTREE, +}; +EXPORT_SYMBOL_GPL(cs42xx8_regmap_config); + +static int cs42xx8_codec_probe(struct snd_soc_codec *codec) +{ + struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec); + struct snd_soc_dapm_context *dapm = &codec->dapm; + + switch (cs42xx8->drvdata->num_adcs) { + case 3: + snd_soc_add_codec_controls(codec, cs42xx8_adc3_snd_controls, + ARRAY_SIZE(cs42xx8_adc3_snd_controls)); + snd_soc_dapm_new_controls(dapm, cs42xx8_adc3_dapm_widgets, + ARRAY_SIZE(cs42xx8_adc3_dapm_widgets)); + snd_soc_dapm_add_routes(dapm, cs42xx8_adc3_dapm_routes, + ARRAY_SIZE(cs42xx8_adc3_dapm_routes)); + break; + default: + break; + } + + /* Mute all DAC channels */ + regmap_write(cs42xx8->regmap, CS42XX8_DACMUTE, CS42XX8_DACMUTE_ALL); + + return 0; +} + +static const struct snd_soc_codec_driver cs42xx8_driver = { + .probe = cs42xx8_codec_probe, + .idle_bias_off = true, + + .controls = cs42xx8_snd_controls, + .num_controls = ARRAY_SIZE(cs42xx8_snd_controls), + .dapm_widgets = cs42xx8_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cs42xx8_dapm_widgets), + .dapm_routes = cs42xx8_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(cs42xx8_dapm_routes), +}; + +const struct cs42xx8_driver_data cs42448_data = { + .name = "cs42448", + .num_adcs = 3, +}; +EXPORT_SYMBOL_GPL(cs42448_data); + +const struct cs42xx8_driver_data cs42888_data = { + .name = "cs42888", + .num_adcs = 2, +}; +EXPORT_SYMBOL_GPL(cs42888_data); + +const struct of_device_id cs42xx8_of_match[] = { + { .compatible = "cirrus,cs42448", .data = &cs42448_data, }, + { .compatible = "cirrus,cs42888", .data = &cs42888_data, }, + { /* sentinel */ } +}; +MODULE_DEVICE_TABLE(of, cs42xx8_of_match); +EXPORT_SYMBOL_GPL(cs42xx8_of_match); + +int cs42xx8_probe(struct device *dev, struct regmap *regmap) +{ + const struct of_device_id *of_id = of_match_device(cs42xx8_of_match, dev); + struct cs42xx8_priv *cs42xx8; + int ret, val, i; + + cs42xx8 = devm_kzalloc(dev, sizeof(*cs42xx8), GFP_KERNEL); + if (cs42xx8 == NULL) + return -ENOMEM; + + dev_set_drvdata(dev, cs42xx8); + + if (of_id) + cs42xx8->drvdata = of_id->data; + + if (!cs42xx8->drvdata) { + dev_err(dev, "failed to find driver data\n"); + return -EINVAL; + } + + cs42xx8->clk = devm_clk_get(dev, "mclk"); + if (IS_ERR(cs42xx8->clk)) { + dev_err(dev, "failed to get the clock: %ld\n", + PTR_ERR(cs42xx8->clk)); + return -EINVAL; + } + + cs42xx8->sysclk = clk_get_rate(cs42xx8->clk); + + for (i = 0; i < ARRAY_SIZE(cs42xx8->supplies); i++) + cs42xx8->supplies[i].supply = cs42xx8_supply_names[i]; + + ret = devm_regulator_bulk_get(dev, + ARRAY_SIZE(cs42xx8->supplies), cs42xx8->supplies); + if (ret) { + dev_err(dev, "failed to request supplies: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(cs42xx8->supplies), + cs42xx8->supplies); + if (ret) { + dev_err(dev, "failed to enable supplies: %d\n", ret); + return ret; + } + + /* Make sure hardware reset done */ + msleep(5); + + cs42xx8->regmap = regmap; + if (IS_ERR(cs42xx8->regmap)) { + ret = PTR_ERR(cs42xx8->regmap); + dev_err(dev, "failed to allocate regmap: %d\n", ret); + goto err_enable; + } + + /* + * We haven't marked the chip revision as volatile due to + * sharing a register with the right input volume; explicitly + * bypass the cache to read it. + */ + regcache_cache_bypass(cs42xx8->regmap, true); + + /* Validate the chip ID */ + regmap_read(cs42xx8->regmap, CS42XX8_CHIPID, &val); + if (val < 0) { + dev_err(dev, "failed to get device ID: %x", val); + ret = -EINVAL; + goto err_enable; + } + + /* The top four bits of the chip ID should be 0000 */ + if ((val & CS42XX8_CHIPID_CHIP_ID_MASK) != 0x00) { + dev_err(dev, "unmatched chip ID: %d\n", + val & CS42XX8_CHIPID_CHIP_ID_MASK); + ret = -EINVAL; + goto err_enable; + } + + dev_info(dev, "found device, revision %X\n", + val & CS42XX8_CHIPID_REV_ID_MASK); + + regcache_cache_bypass(cs42xx8->regmap, false); + + cs42xx8_dai.name = cs42xx8->drvdata->name; + + /* Each adc supports stereo input */ + cs42xx8_dai.capture.channels_max = cs42xx8->drvdata->num_adcs * 2; + + ret = snd_soc_register_codec(dev, &cs42xx8_driver, &cs42xx8_dai, 1); + if (ret) { + dev_err(dev, "failed to register codec:%d\n", ret); + goto err_enable; + } + + regcache_cache_only(cs42xx8->regmap, true); + +err_enable: + regulator_bulk_disable(ARRAY_SIZE(cs42xx8->supplies), + cs42xx8->supplies); + + return ret; +} +EXPORT_SYMBOL_GPL(cs42xx8_probe); + +#ifdef CONFIG_PM_RUNTIME +static int cs42xx8_runtime_resume(struct device *dev) +{ + struct cs42xx8_priv *cs42xx8 = dev_get_drvdata(dev); + int ret; + + ret = clk_prepare_enable(cs42xx8->clk); + if (ret) { + dev_err(dev, "failed to enable mclk: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(cs42xx8->supplies), + cs42xx8->supplies); + if (ret) { + dev_err(dev, "failed to enable supplies: %d\n", ret); + goto err_clk; + } + + /* Make sure hardware reset done */ + msleep(5); + + regcache_cache_only(cs42xx8->regmap, false); + + ret = regcache_sync(cs42xx8->regmap); + if (ret) { + dev_err(dev, "failed to sync regmap: %d\n", ret); + goto err_bulk; + } + + return 0; + +err_bulk: + regulator_bulk_disable(ARRAY_SIZE(cs42xx8->supplies), + cs42xx8->supplies); +err_clk: + clk_disable_unprepare(cs42xx8->clk); + + return ret; +} + +static int cs42xx8_runtime_suspend(struct device *dev) +{ + struct cs42xx8_priv *cs42xx8 = dev_get_drvdata(dev); + + regcache_cache_only(cs42xx8->regmap, true); + + regulator_bulk_disable(ARRAY_SIZE(cs42xx8->supplies), + cs42xx8->supplies); + + clk_disable_unprepare(cs42xx8->clk); + + return 0; +} +#endif + +const struct dev_pm_ops cs42xx8_pm = { + SET_RUNTIME_PM_OPS(cs42xx8_runtime_suspend, cs42xx8_runtime_resume, NULL) +}; +EXPORT_SYMBOL_GPL(cs42xx8_pm); + +MODULE_DESCRIPTION("Cirrus Logic CS42448/CS42888 ALSA SoC Codec Driver"); +MODULE_AUTHOR("Freescale Semiconductor, Inc."); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/cs42xx8.h b/sound/soc/codecs/cs42xx8.h new file mode 100644 index 000000000000..da0b94aee419 --- /dev/null +++ b/sound/soc/codecs/cs42xx8.h @@ -0,0 +1,238 @@ +/* + * cs42xx8.h - Cirrus Logic CS42448/CS42888 Audio CODEC driver header file + * + * Copyright (C) 2014 Freescale Semiconductor, Inc. + * + * Author: Nicolin Chen + * + * This file is licensed under the terms of the GNU General Public License + * version 2. This program is licensed "as is" without any warranty of any + * kind, whether express or implied. + */ + +#ifndef _CS42XX8_H +#define _CS42XX8_H + +struct cs42xx8_driver_data { + char name[32]; + int num_adcs; +}; + +extern const struct dev_pm_ops cs42xx8_pm; +extern const struct cs42xx8_driver_data cs42448_data; +extern const struct cs42xx8_driver_data cs42888_data; +extern const struct regmap_config cs42xx8_regmap_config; +int cs42xx8_probe(struct device *dev, struct regmap *regmap); + +/* CS42888 register map */ +#define CS42XX8_CHIPID 0x01 /* Chip ID */ +#define CS42XX8_PWRCTL 0x02 /* Power Control */ +#define CS42XX8_FUNCMOD 0x03 /* Functional Mode */ +#define CS42XX8_INTF 0x04 /* Interface Formats */ +#define CS42XX8_ADCCTL 0x05 /* ADC Control */ +#define CS42XX8_TXCTL 0x06 /* Transition Control */ +#define CS42XX8_DACMUTE 0x07 /* DAC Mute Control */ +#define CS42XX8_VOLAOUT1 0x08 /* Volume Control AOUT1 */ +#define CS42XX8_VOLAOUT2 0x09 /* Volume Control AOUT2 */ +#define CS42XX8_VOLAOUT3 0x0A /* Volume Control AOUT3 */ +#define CS42XX8_VOLAOUT4 0x0B /* Volume Control AOUT4 */ +#define CS42XX8_VOLAOUT5 0x0C /* Volume Control AOUT5 */ +#define CS42XX8_VOLAOUT6 0x0D /* Volume Control AOUT6 */ +#define CS42XX8_VOLAOUT7 0x0E /* Volume Control AOUT7 */ +#define CS42XX8_VOLAOUT8 0x0F /* Volume Control AOUT8 */ +#define CS42XX8_DACINV 0x10 /* DAC Channel Invert */ +#define CS42XX8_VOLAIN1 0x11 /* Volume Control AIN1 */ +#define CS42XX8_VOLAIN2 0x12 /* Volume Control AIN2 */ +#define CS42XX8_VOLAIN3 0x13 /* Volume Control AIN3 */ +#define CS42XX8_VOLAIN4 0x14 /* Volume Control AIN4 */ +#define CS42XX8_VOLAIN5 0x15 /* Volume Control AIN5 */ +#define CS42XX8_VOLAIN6 0x16 /* Volume Control AIN6 */ +#define CS42XX8_ADCINV 0x17 /* ADC Channel Invert */ +#define CS42XX8_STATUSCTL 0x18 /* Status Control */ +#define CS42XX8_STATUS 0x19 /* Status */ +#define CS42XX8_STATUSM 0x1A /* Status Mask */ +#define CS42XX8_MUTEC 0x1B /* MUTEC Pin Control */ + +#define CS42XX8_FIRSTREG CS42XX8_CHIPID +#define CS42XX8_LASTREG CS42XX8_MUTEC +#define CS42XX8_NUMREGS (CS42XX8_LASTREG - CS42XX8_FIRSTREG + 1) +#define CS42XX8_I2C_INCR 0x80 + +/* Chip I.D. and Revision Register (Address 01h) */ +#define CS42XX8_CHIPID_CHIP_ID_MASK 0xF0 +#define CS42XX8_CHIPID_REV_ID_MASK 0x0F + +/* Power Control (Address 02h) */ +#define CS42XX8_PWRCTL_PDN_ADC3_SHIFT 7 +#define CS42XX8_PWRCTL_PDN_ADC3_MASK (1 << CS42XX8_PWRCTL_PDN_ADC3_SHIFT) +#define CS42XX8_PWRCTL_PDN_ADC3 (1 << CS42XX8_PWRCTL_PDN_ADC3_SHIFT) +#define CS42XX8_PWRCTL_PDN_ADC2_SHIFT 6 +#define CS42XX8_PWRCTL_PDN_ADC2_MASK (1 << CS42XX8_PWRCTL_PDN_ADC2_SHIFT) +#define CS42XX8_PWRCTL_PDN_ADC2 (1 << CS42XX8_PWRCTL_PDN_ADC2_SHIFT) +#define CS42XX8_PWRCTL_PDN_ADC1_SHIFT 5 +#define CS42XX8_PWRCTL_PDN_ADC1_MASK (1 << CS42XX8_PWRCTL_PDN_ADC1_SHIFT) +#define CS42XX8_PWRCTL_PDN_ADC1 (1 << CS42XX8_PWRCTL_PDN_ADC1_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC4_SHIFT 4 +#define CS42XX8_PWRCTL_PDN_DAC4_MASK (1 << CS42XX8_PWRCTL_PDN_DAC4_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC4 (1 << CS42XX8_PWRCTL_PDN_DAC4_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC3_SHIFT 3 +#define CS42XX8_PWRCTL_PDN_DAC3_MASK (1 << CS42XX8_PWRCTL_PDN_DAC3_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC3 (1 << CS42XX8_PWRCTL_PDN_DAC3_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC2_SHIFT 2 +#define CS42XX8_PWRCTL_PDN_DAC2_MASK (1 << CS42XX8_PWRCTL_PDN_DAC2_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC2 (1 << CS42XX8_PWRCTL_PDN_DAC2_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC1_SHIFT 1 +#define CS42XX8_PWRCTL_PDN_DAC1_MASK (1 << CS42XX8_PWRCTL_PDN_DAC1_SHIFT) +#define CS42XX8_PWRCTL_PDN_DAC1 (1 << CS42XX8_PWRCTL_PDN_DAC1_SHIFT) +#define CS42XX8_PWRCTL_PDN_SHIFT 0 +#define CS42XX8_PWRCTL_PDN_MASK (1 << CS42XX8_PWRCTL_PDN_SHIFT) +#define CS42XX8_PWRCTL_PDN (1 << CS42XX8_PWRCTL_PDN_SHIFT) + +/* Functional Mode (Address 03h) */ +#define CS42XX8_FUNCMOD_DAC_FM_SHIFT 6 +#define CS42XX8_FUNCMOD_DAC_FM_WIDTH 2 +#define CS42XX8_FUNCMOD_DAC_FM_MASK (((1 << CS42XX8_FUNCMOD_DAC_FM_WIDTH) - 1) << CS42XX8_FUNCMOD_DAC_FM_SHIFT) +#define CS42XX8_FUNCMOD_DAC_FM(v) ((v) << CS42XX8_FUNCMOD_DAC_FM_SHIFT) +#define CS42XX8_FUNCMOD_ADC_FM_SHIFT 4 +#define CS42XX8_FUNCMOD_ADC_FM_WIDTH 2 +#define CS42XX8_FUNCMOD_ADC_FM_MASK (((1 << CS42XX8_FUNCMOD_ADC_FM_WIDTH) - 1) << CS42XX8_FUNCMOD_ADC_FM_SHIFT) +#define CS42XX8_FUNCMOD_ADC_FM(v) ((v) << CS42XX8_FUNCMOD_ADC_FM_SHIFT) +#define CS42XX8_FUNCMOD_xC_FM_MASK(x) ((x) ? CS42XX8_FUNCMOD_DAC_FM_MASK : CS42XX8_FUNCMOD_ADC_FM_MASK) +#define CS42XX8_FUNCMOD_xC_FM(x, v) ((x) ? CS42XX8_FUNCMOD_DAC_FM(v) : CS42XX8_FUNCMOD_ADC_FM(v)) +#define CS42XX8_FUNCMOD_MFREQ_SHIFT 1 +#define CS42XX8_FUNCMOD_MFREQ_WIDTH 3 +#define CS42XX8_FUNCMOD_MFREQ_MASK (((1 << CS42XX8_FUNCMOD_MFREQ_WIDTH) - 1) << CS42XX8_FUNCMOD_MFREQ_SHIFT) +#define CS42XX8_FUNCMOD_MFREQ_256(s) ((0 << CS42XX8_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42XX8_FUNCMOD_MFREQ_384(s) ((1 << CS42XX8_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42XX8_FUNCMOD_MFREQ_512(s) ((2 << CS42XX8_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42XX8_FUNCMOD_MFREQ_768(s) ((3 << CS42XX8_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) +#define CS42XX8_FUNCMOD_MFREQ_1024(s) ((4 << CS42XX8_FUNCMOD_MFREQ_SHIFT) >> (s >> 1)) + +#define CS42XX8_FM_SINGLE 0 +#define CS42XX8_FM_DOUBLE 1 +#define CS42XX8_FM_QUAD 2 +#define CS42XX8_FM_AUTO 3 + +/* Interface Formats (Address 04h) */ +#define CS42XX8_INTF_FREEZE_SHIFT 7 +#define CS42XX8_INTF_FREEZE_MASK (1 << CS42XX8_INTF_FREEZE_SHIFT) +#define CS42XX8_INTF_FREEZE (1 << CS42XX8_INTF_FREEZE_SHIFT) +#define CS42XX8_INTF_AUX_DIF_SHIFT 6 +#define CS42XX8_INTF_AUX_DIF_MASK (1 << CS42XX8_INTF_AUX_DIF_SHIFT) +#define CS42XX8_INTF_AUX_DIF (1 << CS42XX8_INTF_AUX_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_SHIFT 3 +#define CS42XX8_INTF_DAC_DIF_WIDTH 3 +#define CS42XX8_INTF_DAC_DIF_MASK (((1 << CS42XX8_INTF_DAC_DIF_WIDTH) - 1) << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_LEFTJ (0 << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_I2S (1 << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_RIGHTJ (2 << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_RIGHTJ_16 (3 << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_ONELINE_20 (4 << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_ONELINE_24 (6 << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_DAC_DIF_TDM (7 << CS42XX8_INTF_DAC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_SHIFT 0 +#define CS42XX8_INTF_ADC_DIF_WIDTH 3 +#define CS42XX8_INTF_ADC_DIF_MASK (((1 << CS42XX8_INTF_ADC_DIF_WIDTH) - 1) << CS42XX8_INTF_ADC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_LEFTJ (0 << CS42XX8_INTF_ADC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_I2S (1 << CS42XX8_INTF_ADC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_RIGHTJ (2 << CS42XX8_INTF_ADC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_RIGHTJ_16 (3 << CS42XX8_INTF_ADC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_ONELINE_20 (4 << CS42XX8_INTF_ADC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_ONELINE_24 (6 << CS42XX8_INTF_ADC_DIF_SHIFT) +#define CS42XX8_INTF_ADC_DIF_TDM (7 << CS42XX8_INTF_ADC_DIF_SHIFT) + +/* ADC Control & DAC De-Emphasis (Address 05h) */ +#define CS42XX8_ADCCTL_ADC_HPF_FREEZE_SHIFT 7 +#define CS42XX8_ADCCTL_ADC_HPF_FREEZE_MASK (1 << CS42XX8_ADCCTL_ADC_HPF_FREEZE_SHIFT) +#define CS42XX8_ADCCTL_ADC_HPF_FREEZE (1 << CS42XX8_ADCCTL_ADC_HPF_FREEZE_SHIFT) +#define CS42XX8_ADCCTL_DAC_DEM_SHIFT 5 +#define CS42XX8_ADCCTL_DAC_DEM_MASK (1 << CS42XX8_ADCCTL_DAC_DEM_SHIFT) +#define CS42XX8_ADCCTL_DAC_DEM (1 << CS42XX8_ADCCTL_DAC_DEM_SHIFT) +#define CS42XX8_ADCCTL_ADC1_SINGLE_SHIFT 4 +#define CS42XX8_ADCCTL_ADC1_SINGLE_MASK (1 << CS42XX8_ADCCTL_ADC1_SINGLE_SHIFT) +#define CS42XX8_ADCCTL_ADC1_SINGLE (1 << CS42XX8_ADCCTL_ADC1_SINGLE_SHIFT) +#define CS42XX8_ADCCTL_ADC2_SINGLE_SHIFT 3 +#define CS42XX8_ADCCTL_ADC2_SINGLE_MASK (1 << CS42XX8_ADCCTL_ADC2_SINGLE_SHIFT) +#define CS42XX8_ADCCTL_ADC2_SINGLE (1 << CS42XX8_ADCCTL_ADC2_SINGLE_SHIFT) +#define CS42XX8_ADCCTL_ADC3_SINGLE_SHIFT 2 +#define CS42XX8_ADCCTL_ADC3_SINGLE_MASK (1 << CS42XX8_ADCCTL_ADC3_SINGLE_SHIFT) +#define CS42XX8_ADCCTL_ADC3_SINGLE (1 << CS42XX8_ADCCTL_ADC3_SINGLE_SHIFT) +#define CS42XX8_ADCCTL_AIN5_MUX_SHIFT 1 +#define CS42XX8_ADCCTL_AIN5_MUX_MASK (1 << CS42XX8_ADCCTL_AIN5_MUX_SHIFT) +#define CS42XX8_ADCCTL_AIN5_MUX (1 << CS42XX8_ADCCTL_AIN5_MUX_SHIFT) +#define CS42XX8_ADCCTL_AIN6_MUX_SHIFT 0 +#define CS42XX8_ADCCTL_AIN6_MUX_MASK (1 << CS42XX8_ADCCTL_AIN6_MUX_SHIFT) +#define CS42XX8_ADCCTL_AIN6_MUX (1 << CS42XX8_ADCCTL_AIN6_MUX_SHIFT) + +/* Transition Control (Address 06h) */ +#define CS42XX8_TXCTL_DAC_SNGVOL_SHIFT 7 +#define CS42XX8_TXCTL_DAC_SNGVOL_MASK (1 << CS42XX8_TXCTL_DAC_SNGVOL_SHIFT) +#define CS42XX8_TXCTL_DAC_SNGVOL (1 << CS42XX8_TXCTL_DAC_SNGVOL_SHIFT) +#define CS42XX8_TXCTL_DAC_SZC_SHIFT 5 +#define CS42XX8_TXCTL_DAC_SZC_WIDTH 2 +#define CS42XX8_TXCTL_DAC_SZC_MASK (((1 << CS42XX8_TXCTL_DAC_SZC_WIDTH) - 1) << CS42XX8_TXCTL_DAC_SZC_SHIFT) +#define CS42XX8_TXCTL_DAC_SZC_IC (0 << CS42XX8_TXCTL_DAC_SZC_SHIFT) +#define CS42XX8_TXCTL_DAC_SZC_ZC (1 << CS42XX8_TXCTL_DAC_SZC_SHIFT) +#define CS42XX8_TXCTL_DAC_SZC_SR (2 << CS42XX8_TXCTL_DAC_SZC_SHIFT) +#define CS42XX8_TXCTL_DAC_SZC_SRZC (3 << CS42XX8_TXCTL_DAC_SZC_SHIFT) +#define CS42XX8_TXCTL_AMUTE_SHIFT 4 +#define CS42XX8_TXCTL_AMUTE_MASK (1 << CS42XX8_TXCTL_AMUTE_SHIFT) +#define CS42XX8_TXCTL_AMUTE (1 << CS42XX8_TXCTL_AMUTE_SHIFT) +#define CS42XX8_TXCTL_MUTE_ADC_SP_SHIFT 3 +#define CS42XX8_TXCTL_MUTE_ADC_SP_MASK (1 << CS42XX8_TXCTL_MUTE_ADC_SP_SHIFT) +#define CS42XX8_TXCTL_MUTE_ADC_SP (1 << CS42XX8_TXCTL_MUTE_ADC_SP_SHIFT) +#define CS42XX8_TXCTL_ADC_SNGVOL_SHIFT 2 +#define CS42XX8_TXCTL_ADC_SNGVOL_MASK (1 << CS42XX8_TXCTL_ADC_SNGVOL_SHIFT) +#define CS42XX8_TXCTL_ADC_SNGVOL (1 << CS42XX8_TXCTL_ADC_SNGVOL_SHIFT) +#define CS42XX8_TXCTL_ADC_SZC_SHIFT 0 +#define CS42XX8_TXCTL_ADC_SZC_MASK (((1 << CS42XX8_TXCTL_ADC_SZC_WIDTH) - 1) << CS42XX8_TXCTL_ADC_SZC_SHIFT) +#define CS42XX8_TXCTL_ADC_SZC_IC (0 << CS42XX8_TXCTL_ADC_SZC_SHIFT) +#define CS42XX8_TXCTL_ADC_SZC_ZC (1 << CS42XX8_TXCTL_ADC_SZC_SHIFT) +#define CS42XX8_TXCTL_ADC_SZC_SR (2 << CS42XX8_TXCTL_ADC_SZC_SHIFT) +#define CS42XX8_TXCTL_ADC_SZC_SRZC (3 << CS42XX8_TXCTL_ADC_SZC_SHIFT) + +/* DAC Channel Mute (Address 07h) */ +#define CS42XX8_DACMUTE_AOUT(n) (0x1 << n) +#define CS42XX8_DACMUTE_ALL 0xff + +/* Status Control (Address 18h)*/ +#define CS42XX8_STATUSCTL_INI_SHIFT 2 +#define CS42XX8_STATUSCTL_INI_WIDTH 2 +#define CS42XX8_STATUSCTL_INI_MASK (((1 << CS42XX8_STATUSCTL_INI_WIDTH) - 1) << CS42XX8_STATUSCTL_INI_SHIFT) +#define CS42XX8_STATUSCTL_INT_ACTIVE_HIGH (0 << CS42XX8_STATUSCTL_INI_SHIFT) +#define CS42XX8_STATUSCTL_INT_ACTIVE_LOW (1 << CS42XX8_STATUSCTL_INI_SHIFT) +#define CS42XX8_STATUSCTL_INT_OPEN_DRAIN (2 << CS42XX8_STATUSCTL_INI_SHIFT) + +/* Status (Address 19h)*/ +#define CS42XX8_STATUS_DAC_CLK_ERR_SHIFT 4 +#define CS42XX8_STATUS_DAC_CLK_ERR_MASK (1 << CS42XX8_STATUS_DAC_CLK_ERR_SHIFT) +#define CS42XX8_STATUS_ADC_CLK_ERR_SHIFT 3 +#define CS42XX8_STATUS_ADC_CLK_ERR_MASK (1 << CS42XX8_STATUS_ADC_CLK_ERR_SHIFT) +#define CS42XX8_STATUS_ADC3_OVFL_SHIFT 2 +#define CS42XX8_STATUS_ADC3_OVFL_MASK (1 << CS42XX8_STATUS_ADC3_OVFL_SHIFT) +#define CS42XX8_STATUS_ADC2_OVFL_SHIFT 1 +#define CS42XX8_STATUS_ADC2_OVFL_MASK (1 << CS42XX8_STATUS_ADC2_OVFL_SHIFT) +#define CS42XX8_STATUS_ADC1_OVFL_SHIFT 0 +#define CS42XX8_STATUS_ADC1_OVFL_MASK (1 << CS42XX8_STATUS_ADC1_OVFL_SHIFT) + +/* Status Mask (Address 1Ah) */ +#define CS42XX8_STATUS_DAC_CLK_ERR_M_SHIFT 4 +#define CS42XX8_STATUS_DAC_CLK_ERR_M_MASK (1 << CS42XX8_STATUS_DAC_CLK_ERR_M_SHIFT) +#define CS42XX8_STATUS_ADC_CLK_ERR_M_SHIFT 3 +#define CS42XX8_STATUS_ADC_CLK_ERR_M_MASK (1 << CS42XX8_STATUS_ADC_CLK_ERR_M_SHIFT) +#define CS42XX8_STATUS_ADC3_OVFL_M_SHIFT 2 +#define CS42XX8_STATUS_ADC3_OVFL_M_MASK (1 << CS42XX8_STATUS_ADC3_OVFL_M_SHIFT) +#define CS42XX8_STATUS_ADC2_OVFL_M_SHIFT 1 +#define CS42XX8_STATUS_ADC2_OVFL_M_MASK (1 << CS42XX8_STATUS_ADC2_OVFL_M_SHIFT) +#define CS42XX8_STATUS_ADC1_OVFL_M_SHIFT 0 +#define CS42XX8_STATUS_ADC1_OVFL_M_MASK (1 << CS42XX8_STATUS_ADC1_OVFL_M_SHIFT) + +/* MUTEC Pin Control (Address 1Bh) */ +#define CS42XX8_MUTEC_MCPOLARITY_SHIFT 1 +#define CS42XX8_MUTEC_MCPOLARITY_MASK (1 << CS42XX8_MUTEC_MCPOLARITY_SHIFT) +#define CS42XX8_MUTEC_MCPOLARITY_ACTIVE_LOW (0 << CS42XX8_MUTEC_MCPOLARITY_SHIFT) +#define CS42XX8_MUTEC_MCPOLARITY_ACTIVE_HIGH (1 << CS42XX8_MUTEC_MCPOLARITY_SHIFT) +#define CS42XX8_MUTEC_MUTEC_ACTIVE_SHIFT 0 +#define CS42XX8_MUTEC_MUTEC_ACTIVE_MASK (1 << CS42XX8_MUTEC_MUTEC_ACTIVE_SHIFT) +#define CS42XX8_MUTEC_MUTEC_ACTIVE (1 << CS42XX8_MUTEC_MUTEC_ACTIVE_SHIFT) +#endif /* _CS42XX8_H */ -- cgit v1.2.3