From 19b34bdc6d267723f3fc526ae775efba0ca4c39b Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 20 Feb 2013 17:28:34 +0000 Subject: ASoC: arizona: Move selection of FLL REFCLK into init In preparation for additional features on the FLL this patch moves the code selecting the REFCLK source based on the 32kHz clock into the FLL initialisation function. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 28 ++++++++++++++++------------ sound/soc/codecs/arizona.h | 3 +++ 2 files changed, 19 insertions(+), 12 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index ac948a671ea6..c14e7551a332 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1079,7 +1079,7 @@ int arizona_set_fll(struct arizona_fll *fll, int source, { struct arizona *arizona = fll->arizona; struct arizona_fll_cfg cfg, sync; - unsigned int reg, val; + unsigned int reg; int syncsrc; bool ena; int ret; @@ -1096,16 +1096,7 @@ int arizona_set_fll(struct arizona_fll *fll, int source, ena = reg & ARIZONA_FLL1_ENA; if (Fout) { - /* Do we have a 32kHz reference? */ - regmap_read(arizona->regmap, ARIZONA_CLOCK_32K_1, &val); - switch (val & ARIZONA_CLK_32K_SRC_MASK) { - case ARIZONA_CLK_SRC_MCLK1: - case ARIZONA_CLK_SRC_MCLK2: - syncsrc = val & ARIZONA_CLK_32K_SRC_MASK; - break; - default: - syncsrc = -1; - } + syncsrc = fll->ref_src; if (source == syncsrc) syncsrc = -1; @@ -1115,7 +1106,7 @@ int arizona_set_fll(struct arizona_fll *fll, int source, if (ret != 0) return ret; - ret = arizona_calc_fll(fll, &cfg, 32768, Fout); + ret = arizona_calc_fll(fll, &cfg, fll->ref_freq, Fout); if (ret != 0) return ret; } else { @@ -1178,6 +1169,7 @@ int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq, int ok_irq, struct arizona_fll *fll) { int ret; + unsigned int val; init_completion(&fll->ok); @@ -1185,6 +1177,18 @@ int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq, fll->base = base; fll->arizona = arizona; + /* Configure default refclk to 32kHz if we have one */ + regmap_read(arizona->regmap, ARIZONA_CLOCK_32K_1, &val); + switch (val & ARIZONA_CLK_32K_SRC_MASK) { + case ARIZONA_CLK_SRC_MCLK1: + case ARIZONA_CLK_SRC_MCLK2: + fll->ref_src = val & ARIZONA_CLK_32K_SRC_MASK; + break; + default: + fll->ref_src = -1; + } + fll->ref_freq = 32768; + snprintf(fll->lock_name, sizeof(fll->lock_name), "FLL%d lock", id); snprintf(fll->clock_ok_name, sizeof(fll->clock_ok_name), "FLL%d clock OK", id); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 116372c91f5d..124f9f0ef1ac 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -201,6 +201,9 @@ struct arizona_fll { unsigned int fref; unsigned int fout; + int ref_src; + unsigned int ref_freq; + char lock_name[ARIZONA_FLL_NAME_LEN]; char clock_ok_name[ARIZONA_FLL_NAME_LEN]; }; -- cgit v1.2.3 From 9e359c645fa86daf0e3e5cc2dcbe7388f6e4d16a Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 20 Feb 2013 17:28:35 +0000 Subject: ASoC: arizona: Tidy up SYNCCLK selection and cache values This patch caches the current SYNCCLK settings in the arizona_fll struct and uses these to simplify the code which determines which source should be used for the REFCLK and SYNCCLK inputs. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 101 ++++++++++++++++++++++----------------------- sound/soc/codecs/arizona.h | 2 + 2 files changed, 52 insertions(+), 51 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index c14e7551a332..03076efa4d9e 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1078,15 +1078,39 @@ int arizona_set_fll(struct arizona_fll *fll, int source, unsigned int Fref, unsigned int Fout) { struct arizona *arizona = fll->arizona; - struct arizona_fll_cfg cfg, sync; + struct arizona_fll_cfg ref, sync; unsigned int reg; - int syncsrc; bool ena; int ret; if (fll->fref == Fref && fll->fout == Fout) return 0; + if (fll->ref_src < 0 || fll->ref_src == source) { + if (Fout) { + ret = arizona_calc_fll(fll, &ref, Fref, Fout); + if (ret != 0) + return ret; + } + + fll->sync_src = -1; + fll->ref_src = source; + fll->ref_freq = Fref; + } else { + if (Fout) { + ret = arizona_calc_fll(fll, &ref, fll->ref_freq, Fout); + if (ret != 0) + return ret; + + ret = arizona_calc_fll(fll, &sync, Fref, Fout); + if (ret != 0) + return ret; + } + + fll->sync_src = source; + fll->sync_freq = Fref; + } + ret = regmap_read(arizona->regmap, fll->base + 1, ®); if (ret != 0) { arizona_fll_err(fll, "Failed to read current state: %d\n", @@ -1096,24 +1120,32 @@ int arizona_set_fll(struct arizona_fll *fll, int source, ena = reg & ARIZONA_FLL1_ENA; if (Fout) { - syncsrc = fll->ref_src; + regmap_update_bits(arizona->regmap, fll->base + 5, + ARIZONA_FLL1_OUTDIV_MASK, + ref.outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); - if (source == syncsrc) - syncsrc = -1; + arizona_apply_fll(arizona, fll->base, &ref, fll->ref_src); + if (fll->sync_src >= 0) + arizona_apply_fll(arizona, fll->base + 0x10, &sync, + fll->sync_src); - if (syncsrc >= 0) { - ret = arizona_calc_fll(fll, &sync, Fref, Fout); - if (ret != 0) - return ret; + if (!ena) + pm_runtime_get(arizona->dev); - ret = arizona_calc_fll(fll, &cfg, fll->ref_freq, Fout); - if (ret != 0) - return ret; - } else { - ret = arizona_calc_fll(fll, &cfg, Fref, Fout); - if (ret != 0) - return ret; - } + /* Clear any pending completions */ + try_wait_for_completion(&fll->ok); + + regmap_update_bits(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); + if (fll->sync_src >= 0) + regmap_update_bits(arizona->regmap, fll->base + 0x11, + ARIZONA_FLL1_SYNC_ENA, + ARIZONA_FLL1_SYNC_ENA); + + ret = wait_for_completion_timeout(&fll->ok, + msecs_to_jiffies(250)); + if (ret == 0) + arizona_fll_warn(fll, "Timed out waiting for lock\n"); } else { regmap_update_bits(arizona->regmap, fll->base + 1, ARIZONA_FLL1_ENA, 0); @@ -1122,42 +1154,8 @@ int arizona_set_fll(struct arizona_fll *fll, int source, if (ena) pm_runtime_put_autosuspend(arizona->dev); - - fll->fref = Fref; - fll->fout = Fout; - - return 0; } - regmap_update_bits(arizona->regmap, fll->base + 5, - ARIZONA_FLL1_OUTDIV_MASK, - cfg.outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); - - if (syncsrc >= 0) { - arizona_apply_fll(arizona, fll->base, &cfg, syncsrc); - arizona_apply_fll(arizona, fll->base + 0x10, &sync, source); - } else { - arizona_apply_fll(arizona, fll->base, &cfg, source); - } - - if (!ena) - pm_runtime_get(arizona->dev); - - /* Clear any pending completions */ - try_wait_for_completion(&fll->ok); - - regmap_update_bits(arizona->regmap, fll->base + 1, - ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); - if (syncsrc >= 0) - regmap_update_bits(arizona->regmap, fll->base + 0x11, - ARIZONA_FLL1_SYNC_ENA, - ARIZONA_FLL1_SYNC_ENA); - - ret = wait_for_completion_timeout(&fll->ok, - msecs_to_jiffies(250)); - if (ret == 0) - arizona_fll_warn(fll, "Timed out waiting for lock\n"); - fll->fref = Fref; fll->fout = Fout; @@ -1176,6 +1174,7 @@ int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq, fll->id = id; fll->base = base; fll->arizona = arizona; + fll->sync_src = -1; /* Configure default refclk to 32kHz if we have one */ regmap_read(arizona->regmap, ARIZONA_CLOCK_32K_1, &val); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 124f9f0ef1ac..37766b547b9d 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -201,6 +201,8 @@ struct arizona_fll { unsigned int fref; unsigned int fout; + int sync_src; + unsigned int sync_freq; int ref_src; unsigned int ref_freq; -- cgit v1.2.3 From d122d6c974e35c940a638c26aa70bea363141d27 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 20 Feb 2013 17:28:36 +0000 Subject: ASoC: arizona: Factor out check for enabled FLL In preparation for additional features on the FLL this patch factors out the code which checks if an FLL is currently enabled into a seperate function. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 25 +++++++++++++++++-------- 1 file changed, 17 insertions(+), 8 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 03076efa4d9e..4640bccbfba2 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1074,12 +1074,27 @@ static void arizona_apply_fll(struct arizona *arizona, unsigned int base, ARIZONA_FLL1_CTRL_UPD | cfg->n); } +static bool arizona_is_enabled_fll(struct arizona_fll *fll) +{ + struct arizona *arizona = fll->arizona; + unsigned int reg; + int ret; + + ret = regmap_read(arizona->regmap, fll->base + 1, ®); + if (ret != 0) { + arizona_fll_err(fll, "Failed to read current state: %d\n", + ret); + return ret; + } + + return reg & ARIZONA_FLL1_ENA; +} + int arizona_set_fll(struct arizona_fll *fll, int source, unsigned int Fref, unsigned int Fout) { struct arizona *arizona = fll->arizona; struct arizona_fll_cfg ref, sync; - unsigned int reg; bool ena; int ret; @@ -1111,13 +1126,7 @@ int arizona_set_fll(struct arizona_fll *fll, int source, fll->sync_freq = Fref; } - ret = regmap_read(arizona->regmap, fll->base + 1, ®); - if (ret != 0) { - arizona_fll_err(fll, "Failed to read current state: %d\n", - ret); - return ret; - } - ena = reg & ARIZONA_FLL1_ENA; + ena = arizona_is_enabled_fll(fll); if (Fout) { regmap_update_bits(arizona->regmap, fll->base + 5, -- cgit v1.2.3 From 7604054e13897c2da3570e33a67ecb76462212d8 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 20 Feb 2013 17:28:37 +0000 Subject: ASoC: arizona: Factor out FLL disable In preparation for additional features on the FLL this patch factors out the code for disabling an FLL into a seperate function. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 22 +++++++++++++++------- 1 file changed, 15 insertions(+), 7 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 4640bccbfba2..a8821a819adc 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1090,6 +1090,20 @@ static bool arizona_is_enabled_fll(struct arizona_fll *fll) return reg & ARIZONA_FLL1_ENA; } +static void arizona_disable_fll(struct arizona_fll *fll) +{ + struct arizona *arizona = fll->arizona; + bool change; + + regmap_update_bits_check(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_ENA, 0, &change); + regmap_update_bits(arizona->regmap, fll->base + 0x11, + ARIZONA_FLL1_SYNC_ENA, 0); + + if (change) + pm_runtime_put_autosuspend(arizona->dev); +} + int arizona_set_fll(struct arizona_fll *fll, int source, unsigned int Fref, unsigned int Fout) { @@ -1156,13 +1170,7 @@ int arizona_set_fll(struct arizona_fll *fll, int source, if (ret == 0) arizona_fll_warn(fll, "Timed out waiting for lock\n"); } else { - regmap_update_bits(arizona->regmap, fll->base + 1, - ARIZONA_FLL1_ENA, 0); - regmap_update_bits(arizona->regmap, fll->base + 0x11, - ARIZONA_FLL1_SYNC_ENA, 0); - - if (ena) - pm_runtime_put_autosuspend(arizona->dev); + arizona_disable_fll(fll); } fll->fref = Fref; -- cgit v1.2.3 From 357228153b4a158bdeb05f1c46ee13ef60a675a6 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 20 Feb 2013 17:28:38 +0000 Subject: ASoC: arizona: Factor out FLL enable In preparation for additional features on the FLL this patch factors out the code for enabling an FLL into a seperate function. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 66 +++++++++++++++++++++++++--------------------- 1 file changed, 36 insertions(+), 30 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index a8821a819adc..e770945fa019 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1090,6 +1090,41 @@ 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) +{ + struct arizona *arizona = fll->arizona; + int ret; + + regmap_update_bits(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); + if (fll->sync_src >= 0) + arizona_apply_fll(arizona, fll->base + 0x10, sync, + fll->sync_src); + + if (!arizona_is_enabled_fll(fll)) + pm_runtime_get(arizona->dev); + + /* Clear any pending completions */ + try_wait_for_completion(&fll->ok); + + regmap_update_bits(arizona->regmap, fll->base + 1, + ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); + if (fll->sync_src >= 0) + regmap_update_bits(arizona->regmap, fll->base + 0x11, + ARIZONA_FLL1_SYNC_ENA, + ARIZONA_FLL1_SYNC_ENA); + + ret = wait_for_completion_timeout(&fll->ok, + msecs_to_jiffies(250)); + if (ret == 0) + arizona_fll_warn(fll, "Timed out waiting for lock\n"); +} + static void arizona_disable_fll(struct arizona_fll *fll) { struct arizona *arizona = fll->arizona; @@ -1107,9 +1142,7 @@ static void arizona_disable_fll(struct arizona_fll *fll) int arizona_set_fll(struct arizona_fll *fll, int source, unsigned int Fref, unsigned int Fout) { - struct arizona *arizona = fll->arizona; struct arizona_fll_cfg ref, sync; - bool ena; int ret; if (fll->fref == Fref && fll->fout == Fout) @@ -1140,35 +1173,8 @@ int arizona_set_fll(struct arizona_fll *fll, int source, fll->sync_freq = Fref; } - ena = arizona_is_enabled_fll(fll); - if (Fout) { - regmap_update_bits(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); - if (fll->sync_src >= 0) - arizona_apply_fll(arizona, fll->base + 0x10, &sync, - fll->sync_src); - - if (!ena) - pm_runtime_get(arizona->dev); - - /* Clear any pending completions */ - try_wait_for_completion(&fll->ok); - - regmap_update_bits(arizona->regmap, fll->base + 1, - ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); - if (fll->sync_src >= 0) - regmap_update_bits(arizona->regmap, fll->base + 0x11, - ARIZONA_FLL1_SYNC_ENA, - ARIZONA_FLL1_SYNC_ENA); - - ret = wait_for_completion_timeout(&fll->ok, - msecs_to_jiffies(250)); - if (ret == 0) - arizona_fll_warn(fll, "Timed out waiting for lock\n"); + arizona_enable_fll(fll, &ref, &sync); } else { arizona_disable_fll(fll); } -- cgit v1.2.3 From de1e6eedddeab2fa417c38c231d896198f903129 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 20 Feb 2013 17:28:39 +0000 Subject: ASoC: arizona: Improve suppression of noop FLL updates Previously updates that only changes FLL source would be missed, this patch corrects this. We also ensures that both REFCLK and SYNCCLK frequency changes are considered, in preparation for future updates. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 16 ++++++++++------ sound/soc/codecs/arizona.h | 3 +-- 2 files changed, 11 insertions(+), 8 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index e770945fa019..149e44f42f84 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1145,10 +1145,12 @@ int arizona_set_fll(struct arizona_fll *fll, int source, struct arizona_fll_cfg ref, sync; int ret; - if (fll->fref == Fref && fll->fout == Fout) - return 0; - if (fll->ref_src < 0 || fll->ref_src == source) { + if (fll->sync_src == -1 && + fll->ref_src == source && fll->ref_freq == Fref && + fll->fout == Fout) + return 0; + if (Fout) { ret = arizona_calc_fll(fll, &ref, Fref, Fout); if (ret != 0) @@ -1159,6 +1161,10 @@ int arizona_set_fll(struct arizona_fll *fll, int source, fll->ref_src = source; fll->ref_freq = Fref; } else { + if (fll->sync_src == source && + fll->sync_freq == Fref && fll->fout == Fout) + return 0; + if (Fout) { ret = arizona_calc_fll(fll, &ref, fll->ref_freq, Fout); if (ret != 0) @@ -1172,6 +1178,7 @@ int arizona_set_fll(struct arizona_fll *fll, int source, fll->sync_src = source; fll->sync_freq = Fref; } + fll->fout = Fout; if (Fout) { arizona_enable_fll(fll, &ref, &sync); @@ -1179,9 +1186,6 @@ int arizona_set_fll(struct arizona_fll *fll, int source, arizona_disable_fll(fll); } - fll->fref = Fref; - fll->fout = Fout; - return 0; } EXPORT_SYMBOL_GPL(arizona_set_fll); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 37766b547b9d..bedf12a527e5 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -198,9 +198,8 @@ struct arizona_fll { unsigned int base; unsigned int vco_mult; struct completion ok; - unsigned int fref; - unsigned int fout; + unsigned int fout; int sync_src; unsigned int sync_freq; int ref_src; -- cgit v1.2.3 From ee929a9780605f21ad67a1ccb626baa41e038c1a Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 20 Feb 2013 17:28:40 +0000 Subject: ASoC: arizona: Add support for directly setting the FLL REFCLK This patch allows the REFCLK to be set directly allowing much greater flexibility in how the FLLs are configured. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 39 +++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/arizona.h | 2 ++ sound/soc/codecs/wm5102.c | 6 ++++++ sound/soc/codecs/wm5102.h | 6 ++++-- sound/soc/codecs/wm5110.c | 6 ++++++ sound/soc/codecs/wm5110.h | 6 ++++-- 6 files changed, 61 insertions(+), 4 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 149e44f42f84..2bebfae3485f 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1139,6 +1139,45 @@ static void arizona_disable_fll(struct arizona_fll *fll) pm_runtime_put_autosuspend(arizona->dev); } +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 (source < 0) + return -EINVAL; + + if (fll->ref_src == source && fll->ref_freq == Fref && + fll->fout == Fout) + return 0; + + if (Fout) { + ret = arizona_calc_fll(fll, &ref, Fref, Fout); + if (ret != 0) + return ret; + + if (fll->sync_src >= 0) { + ret = arizona_calc_fll(fll, &sync, fll->sync_freq, Fout); + if (ret != 0) + return ret; + } + } + + fll->ref_src = source; + fll->ref_freq = Fref; + fll->fout = Fout; + + if (Fout) { + arizona_enable_fll(fll, &ref, &sync); + } else { + arizona_disable_fll(fll); + } + + return 0; +} +EXPORT_SYMBOL_GPL(arizona_set_fll_refclk); + int arizona_set_fll(struct arizona_fll *fll, int source, unsigned int Fref, unsigned int Fout) { diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index bedf12a527e5..f2ca41f8dc83 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -211,6 +211,8 @@ struct arizona_fll { extern int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq, int ok_irq, struct arizona_fll *fll); +extern int arizona_set_fll_refclk(struct arizona_fll *fll, int source, + unsigned int Fref, unsigned int Fout); extern int arizona_set_fll(struct arizona_fll *fll, int source, unsigned int Fref, unsigned int Fout); diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index b8d461db369f..5515d85fd82f 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1483,6 +1483,12 @@ static int wm5102_set_fll(struct snd_soc_codec *codec, int fll_id, int source, return arizona_set_fll(&wm5102->fll[0], source, Fref, Fout); case WM5102_FLL2: return arizona_set_fll(&wm5102->fll[1], source, Fref, Fout); + case WM5102_FLL1_REFCLK: + return arizona_set_fll_refclk(&wm5102->fll[0], source, Fref, + Fout); + case WM5102_FLL2_REFCLK: + return arizona_set_fll_refclk(&wm5102->fll[1], source, Fref, + Fout); default: return -EINVAL; } diff --git a/sound/soc/codecs/wm5102.h b/sound/soc/codecs/wm5102.h index d30477f3070c..adb38040f661 100644 --- a/sound/soc/codecs/wm5102.h +++ b/sound/soc/codecs/wm5102.h @@ -15,7 +15,9 @@ #include "arizona.h" -#define WM5102_FLL1 1 -#define WM5102_FLL2 2 +#define WM5102_FLL1 1 +#define WM5102_FLL2 2 +#define WM5102_FLL1_REFCLK 3 +#define WM5102_FLL2_REFCLK 4 #endif diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index cd17b477781d..2d9b55f57eed 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -880,6 +880,12 @@ static int wm5110_set_fll(struct snd_soc_codec *codec, int fll_id, int source, return arizona_set_fll(&wm5110->fll[0], source, Fref, Fout); case WM5110_FLL2: return arizona_set_fll(&wm5110->fll[1], source, Fref, Fout); + case WM5110_FLL1_REFCLK: + return arizona_set_fll_refclk(&wm5110->fll[0], source, Fref, + Fout); + case WM5110_FLL2_REFCLK: + return arizona_set_fll_refclk(&wm5110->fll[1], source, Fref, + Fout); default: return -EINVAL; } diff --git a/sound/soc/codecs/wm5110.h b/sound/soc/codecs/wm5110.h index 75e9351ccab0..e6c0cd4235c5 100644 --- a/sound/soc/codecs/wm5110.h +++ b/sound/soc/codecs/wm5110.h @@ -15,7 +15,9 @@ #include "arizona.h" -#define WM5110_FLL1 1 -#define WM5110_FLL2 2 +#define WM5110_FLL1 1 +#define WM5110_FLL2 2 +#define WM5110_FLL1_REFCLK 3 +#define WM5110_FLL2_REFCLK 4 #endif -- cgit v1.2.3 From f3f1163d19ebd5aa374e5df5372a8f932f2bd5f9 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 20 Feb 2013 17:28:41 +0000 Subject: ASoC: arizona: Add convience define for clearing SYNCCLK Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 8 ++++---- sound/soc/codecs/arizona.h | 1 + 2 files changed, 5 insertions(+), 4 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 2bebfae3485f..6837863b582d 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1185,7 +1185,7 @@ int arizona_set_fll(struct arizona_fll *fll, int source, int ret; if (fll->ref_src < 0 || fll->ref_src == source) { - if (fll->sync_src == -1 && + if (fll->sync_src == ARIZONA_FLL_SRC_NONE && fll->ref_src == source && fll->ref_freq == Fref && fll->fout == Fout) return 0; @@ -1196,7 +1196,7 @@ int arizona_set_fll(struct arizona_fll *fll, int source, return ret; } - fll->sync_src = -1; + fll->sync_src = ARIZONA_FLL_SRC_NONE; fll->ref_src = source; fll->ref_freq = Fref; } else { @@ -1240,7 +1240,7 @@ int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq, fll->id = id; fll->base = base; fll->arizona = arizona; - fll->sync_src = -1; + fll->sync_src = ARIZONA_FLL_SRC_NONE; /* Configure default refclk to 32kHz if we have one */ regmap_read(arizona->regmap, ARIZONA_CLOCK_32K_1, &val); @@ -1250,7 +1250,7 @@ int arizona_init_fll(struct arizona *arizona, int id, int base, int lock_irq, fll->ref_src = val & ARIZONA_CLK_32K_SRC_MASK; break; default: - fll->ref_src = -1; + fll->ref_src = ARIZONA_FLL_SRC_NONE; } fll->ref_freq = 32768; diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index f2ca41f8dc83..3f84943b23bf 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -32,6 +32,7 @@ #define ARIZONA_CLK_SRC_AIF2BCLK 0x9 #define ARIZONA_CLK_SRC_AIF3BCLK 0xa +#define ARIZONA_FLL_SRC_NONE -1 #define ARIZONA_FLL_SRC_MCLK1 0 #define ARIZONA_FLL_SRC_MCLK2 1 #define ARIZONA_FLL_SRC_SLIMCLK 3 -- cgit v1.2.3 From ddbce97cd1798ba4661e33662c659b168e9f51ed Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 15 Feb 2013 17:27:22 +0000 Subject: ASoC: arizona: Only allow input volume updates when inputs are enabled Since we are automatically managing the mutes we may as well also manage the volume update bits, disabling volume updates while none of the inputs are active. Since we are doing this we may as well allow the volumes to ramp together so only enable volume updates once at the end of power up. Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 39 +++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/arizona.h | 3 +++ sound/soc/codecs/wm5102.c | 26 +++++++++++++------------- sound/soc/codecs/wm5110.c | 34 +++++++++++++++++----------------- 4 files changed, 70 insertions(+), 32 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 6837863b582d..debd184cc706 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -10,6 +10,7 @@ * published by the Free Software Foundation. */ +#include #include #include #include @@ -332,9 +333,27 @@ const struct soc_enum arizona_ng_hold = 4, arizona_ng_hold_text); EXPORT_SYMBOL_GPL(arizona_ng_hold); +static void arizona_in_set_vu(struct snd_soc_codec *codec, int ena) +{ + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + unsigned int val; + int i; + + if (ena) + val = ARIZONA_IN_VU; + else + val = 0; + + for (i = 0; i < priv->num_inputs; i++) + snd_soc_update_bits(codec, + ARIZONA_ADC_DIGITAL_VOLUME_1L + (i * 4), + ARIZONA_IN_VU, val); +} + int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + struct arizona_priv *priv = snd_soc_codec_get_drvdata(w->codec); unsigned int reg; if (w->shift % 2) @@ -343,13 +362,29 @@ int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, reg = ARIZONA_ADC_DIGITAL_VOLUME_1R + ((w->shift / 2) * 8); switch (event) { + case SND_SOC_DAPM_PRE_PMU: + priv->in_pending++; + break; case SND_SOC_DAPM_POST_PMU: snd_soc_update_bits(w->codec, reg, ARIZONA_IN1L_MUTE, 0); + + /* If this is the last input pending then allow VU */ + priv->in_pending--; + if (priv->in_pending == 0) { + msleep(1); + arizona_in_set_vu(w->codec, 1); + } break; case SND_SOC_DAPM_PRE_PMD: - snd_soc_update_bits(w->codec, reg, ARIZONA_IN1L_MUTE, - ARIZONA_IN1L_MUTE); + snd_soc_update_bits(w->codec, reg, + ARIZONA_IN1L_MUTE | ARIZONA_IN_VU, + ARIZONA_IN1L_MUTE | ARIZONA_IN_VU); break; + case SND_SOC_DAPM_POST_PMD: + /* Disable volume updates if no inputs are enabled */ + reg = snd_soc_read(w->codec, ARIZONA_INPUT_ENABLES); + if (reg == 0) + arizona_in_set_vu(w->codec, 0); } return 0; diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 3f84943b23bf..d592adcc969c 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -65,6 +65,9 @@ struct arizona_priv { int sysclk; int asyncclk; struct arizona_dai_priv dai[ARIZONA_MAX_DAI]; + + int num_inputs; + unsigned int in_pending; }; #define ARIZONA_NUM_MIXER_INPUTS 99 diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 5515d85fd82f..44d4c69d25e5 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -973,22 +973,28 @@ SND_SOC_DAPM_INPUT("IN3R"), SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT, 0, NULL, 0, arizona_in_ev, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT, 0, NULL, 0, arizona_in_ev, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("IN2L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT, 0, NULL, 0, arizona_in_ev, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("IN2R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2R_ENA_SHIFT, 0, NULL, 0, arizona_in_ev, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("IN3L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3L_ENA_SHIFT, 0, NULL, 0, arizona_in_ev, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("IN3R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3R_ENA_SHIFT, 0, NULL, 0, arizona_in_ev, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("MICBIAS1", ARIZONA_MIC_BIAS_CTRL_1, ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0), @@ -1599,13 +1605,6 @@ static int wm5102_codec_remove(struct snd_soc_codec *codec) #define WM5102_DIG_VU 0x0200 static unsigned int wm5102_digital_vu[] = { - ARIZONA_ADC_DIGITAL_VOLUME_1L, - ARIZONA_ADC_DIGITAL_VOLUME_1R, - ARIZONA_ADC_DIGITAL_VOLUME_2L, - ARIZONA_ADC_DIGITAL_VOLUME_2R, - ARIZONA_ADC_DIGITAL_VOLUME_3L, - ARIZONA_ADC_DIGITAL_VOLUME_3R, - ARIZONA_DAC_DIGITAL_VOLUME_1L, ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_DAC_DIGITAL_VOLUME_2L, @@ -1648,6 +1647,7 @@ static int wm5102_probe(struct platform_device *pdev) platform_set_drvdata(pdev, wm5102); wm5102->core.arizona = arizona; + wm5102->core.num_inputs = 6; wm5102->core.adsp[0].part = "wm5102"; wm5102->core.adsp[0].num = 1; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 2d9b55f57eed..a64d3b8bc3b4 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -416,28 +416,36 @@ SND_SOC_DAPM_INPUT("IN4R"), SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT, 0, NULL, 0, arizona_in_ev, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT, 0, NULL, 0, arizona_in_ev, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("IN2L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT, 0, NULL, 0, arizona_in_ev, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("IN2R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2R_ENA_SHIFT, 0, NULL, 0, arizona_in_ev, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("IN3L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3L_ENA_SHIFT, 0, NULL, 0, arizona_in_ev, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("IN3R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3R_ENA_SHIFT, 0, NULL, 0, arizona_in_ev, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("IN4L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN4L_ENA_SHIFT, 0, NULL, 0, arizona_in_ev, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("IN4R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN4R_ENA_SHIFT, 0, NULL, 0, arizona_in_ev, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("MICBIAS1", ARIZONA_MIC_BIAS_CTRL_1, ARIZONA_MICB1_ENA_SHIFT, 0, NULL, 0), @@ -993,15 +1001,6 @@ static int wm5110_codec_remove(struct snd_soc_codec *codec) #define WM5110_DIG_VU 0x0200 static unsigned int wm5110_digital_vu[] = { - ARIZONA_ADC_DIGITAL_VOLUME_1L, - ARIZONA_ADC_DIGITAL_VOLUME_1R, - ARIZONA_ADC_DIGITAL_VOLUME_2L, - ARIZONA_ADC_DIGITAL_VOLUME_2R, - ARIZONA_ADC_DIGITAL_VOLUME_3L, - ARIZONA_ADC_DIGITAL_VOLUME_3R, - ARIZONA_ADC_DIGITAL_VOLUME_4L, - ARIZONA_ADC_DIGITAL_VOLUME_4R, - ARIZONA_DAC_DIGITAL_VOLUME_1L, ARIZONA_DAC_DIGITAL_VOLUME_1R, ARIZONA_DAC_DIGITAL_VOLUME_2L, @@ -1046,6 +1045,7 @@ static int wm5110_probe(struct platform_device *pdev) platform_set_drvdata(pdev, wm5110); wm5110->core.arizona = arizona; + wm5110->core.num_inputs = 8; for (i = 0; i < ARRAY_SIZE(wm5110->fll); i++) wm5110->fll[i].vco_mult = 3; -- cgit v1.2.3 From 1c5617fc230b399c1d84711b8a2e316199387eb9 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 22 Feb 2013 17:10:37 +0000 Subject: ASoC: arizona: Don't enable FLL on REFCLK configuration Enabling the FLL when REFCLK is being configured is not what the user would expect and can cause issues if SYNCCLK has no specified frequency. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 15 ++++++--------- 1 file changed, 6 insertions(+), 9 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index debd184cc706..e456cb4b196e 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1183,17 +1183,17 @@ int arizona_set_fll_refclk(struct arizona_fll *fll, int source, if (source < 0) return -EINVAL; - if (fll->ref_src == source && fll->ref_freq == Fref && - fll->fout == Fout) + if (fll->ref_src == source && fll->ref_freq == Fref) return 0; - if (Fout) { - ret = arizona_calc_fll(fll, &ref, Fref, Fout); + if (fll->fout) { + 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, Fout); + ret = arizona_calc_fll(fll, &sync, fll->sync_freq, + fll->fout); if (ret != 0) return ret; } @@ -1201,12 +1201,9 @@ int arizona_set_fll_refclk(struct arizona_fll *fll, int source, fll->ref_src = source; fll->ref_freq = Fref; - fll->fout = Fout; - if (Fout) { + if (fll->fout) { arizona_enable_fll(fll, &ref, &sync); - } else { - arizona_disable_fll(fll); } return 0; -- cgit v1.2.3 From c751a1f49b3fbdce0fbbb2c9b56544a7e6833fff Mon Sep 17 00:00:00 2001 From: Dylan Reid Date: Fri, 15 Feb 2013 08:55:10 -0800 Subject: ASoC: max98088: Add TLV data for volume controls. Specify volumes as defined in the MAX98088/9 data sheet. Allows ALSA lib snd_mixer_selem_get_playback_dB_range and related functions to work. Signed-off-by: Dylan Reid Signed-off-by: Mark Brown --- sound/soc/codecs/max98088.c | 30 ++++++++++++++++++++++++------ 1 file changed, 24 insertions(+), 6 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index a4c16fd70f77..3a7b7fd14e3e 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -739,14 +739,32 @@ static const unsigned int max98088_micboost_tlv[] = { 2, 2, TLV_DB_SCALE_ITEM(3000, 0, 0), }; +static const unsigned int max98088_hp_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 6, TLV_DB_SCALE_ITEM(-6700, 400, 0), + 7, 14, TLV_DB_SCALE_ITEM(-4000, 300, 0), + 15, 21, TLV_DB_SCALE_ITEM(-1700, 200, 0), + 22, 27, TLV_DB_SCALE_ITEM(-400, 100, 0), + 28, 31, TLV_DB_SCALE_ITEM(150, 50, 0), +}; + +static const unsigned int max98088_spk_tlv[] = { + TLV_DB_RANGE_HEAD(5), + 0, 6, TLV_DB_SCALE_ITEM(-6200, 400, 0), + 7, 14, TLV_DB_SCALE_ITEM(-3500, 300, 0), + 15, 21, TLV_DB_SCALE_ITEM(-1200, 200, 0), + 22, 27, TLV_DB_SCALE_ITEM(100, 100, 0), + 28, 31, TLV_DB_SCALE_ITEM(650, 50, 0), +}; + static const struct snd_kcontrol_new max98088_snd_controls[] = { - SOC_DOUBLE_R("Headphone Volume", M98088_REG_39_LVL_HP_L, - M98088_REG_3A_LVL_HP_R, 0, 31, 0), - SOC_DOUBLE_R("Speaker Volume", M98088_REG_3D_LVL_SPK_L, - M98088_REG_3E_LVL_SPK_R, 0, 31, 0), - SOC_DOUBLE_R("Receiver Volume", M98088_REG_3B_LVL_REC_L, - M98088_REG_3C_LVL_REC_R, 0, 31, 0), + SOC_DOUBLE_R_TLV("Headphone Volume", M98088_REG_39_LVL_HP_L, + M98088_REG_3A_LVL_HP_R, 0, 31, 0, max98088_hp_tlv), + SOC_DOUBLE_R_TLV("Speaker Volume", M98088_REG_3D_LVL_SPK_L, + M98088_REG_3E_LVL_SPK_R, 0, 31, 0, max98088_spk_tlv), + SOC_DOUBLE_R_TLV("Receiver Volume", M98088_REG_3B_LVL_REC_L, + M98088_REG_3C_LVL_REC_R, 0, 31, 0, max98088_spk_tlv), SOC_DOUBLE_R("Headphone Switch", M98088_REG_39_LVL_HP_L, M98088_REG_3A_LVL_HP_R, 7, 1, 1), -- cgit v1.2.3 From f314cbe84fd81082a286af685c59c2dc4048bc77 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 18 Feb 2013 17:02:10 +0530 Subject: ASoC: max98090: Remove unneeded version.h header include version.h header file inclusion is not required as detected by versioncheck. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 2 -- 1 file changed, 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index fc176044994d..9ea73aa71198 100755 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -23,8 +23,6 @@ #include #include "max98090.h" -#include - #define DEBUG #define EXTMIC_METHOD #define EXTMIC_METHOD_TEST -- cgit v1.2.3 From a3a6cc84652d82ff795c519c6187d37baa1d9697 Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 18 Feb 2013 17:02:11 +0530 Subject: ASoC: max98090: Convert to devm_regmap_init_i2c() devm_regmap_init_i2c() is device managed and makes error handling and code cleanup simpler. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 7 +------ 1 file changed, 1 insertion(+), 6 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 9ea73aa71198..fef370e03305 100755 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2322,7 +2322,7 @@ static int max98090_i2c_probe(struct i2c_client *i2c, max98090->pdata = i2c->dev.platform_data; max98090->irq = i2c->irq; - max98090->regmap = regmap_init_i2c(i2c, &max98090_regmap); + max98090->regmap = devm_regmap_init_i2c(i2c, &max98090_regmap); if (IS_ERR(max98090->regmap)) { ret = PTR_ERR(max98090->regmap); dev_err(&i2c->dev, "Failed to allocate regmap: %d\n", ret); @@ -2332,18 +2332,13 @@ static int max98090_i2c_probe(struct i2c_client *i2c, ret = snd_soc_register_codec(&i2c->dev, &soc_codec_dev_max98090, max98090_dai, ARRAY_SIZE(max98090_dai)); - if (ret < 0) - regmap_exit(max98090->regmap); - err_enable: return ret; } static int max98090_i2c_remove(struct i2c_client *client) { - struct max98090_priv *max98090 = dev_get_drvdata(&client->dev); snd_soc_unregister_codec(&client->dev); - regmap_exit(max98090->regmap); return 0; } -- cgit v1.2.3 From 3e12af7e139275e5822383e210c86e0ff1ea185c Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Mon, 18 Feb 2013 17:02:12 +0530 Subject: ASoC: max98090: Make struct dev_pm_ops const Silences the following checkpatch warning: WARNING: struct dev_pm_ops should normally be const. Signed-off-by: Sachin Kamat 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 fef370e03305..1cf017f2f587 100755 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -2362,7 +2362,7 @@ static int max98090_runtime_suspend(struct device *dev) return 0; } -static struct dev_pm_ops max98090_pm = { +static const struct dev_pm_ops max98090_pm = { SET_RUNTIME_PM_OPS(max98090_runtime_suspend, max98090_runtime_resume, NULL) }; -- cgit v1.2.3 From 4ca74feb6ceb3031e8cf9ef88dedb3ebb984a59a Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Thu, 21 Feb 2013 12:24:59 +0530 Subject: ASoC: max98090: Fix checkpatch errors related to spacing Fixes the following type of checkpatch errors: ERROR: "foo * bar" should be "foo *bar" Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 1cf017f2f587..89f83f8e56b2 100755 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -507,16 +507,16 @@ static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol, return 0; } -static const char * max98090_perf_pwr_text[] = +static const char *max98090_perf_pwr_text[] = { "High Performance", "Low Power" }; -static const char * max98090_pwr_perf_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 const char * max98090_osr128_text[] = { "64*fs", "128*fs" }; +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, @@ -533,28 +533,28 @@ static const struct soc_enum max98090_filter_dmic34mode_enum = M98090_FLT_DMIC34MODE_SHIFT, ARRAY_SIZE(max98090_mode_text), max98090_mode_text); -static const char * max98090_drcatk_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 const char * max98090_drcrls_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 const char * max98090_alccmp_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 const char * max98090_drcexp_text[] = { "1:1", "2:1", "3:1" }; +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, @@ -857,7 +857,7 @@ static const struct soc_enum mic2_mux_enum = static const struct snd_kcontrol_new max98090_mic2_mux = SOC_DAPM_ENUM("MIC2 Mux", mic2_mux_enum); -static const char * max98090_micpre_text[] = { "Off", "On" }; +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, -- cgit v1.2.3 From 959b6250dbf8398e3c63544f771ff1682a09987e Mon Sep 17 00:00:00 2001 From: Sachin Kamat Date: Thu, 21 Feb 2013 12:25:00 +0530 Subject: ASoC: max98090: Remove unnecessary braces Braces are not required for single line statements. Silences the following checkpatch warnings: WARNING: braces {} are not necessary for single statement blocks. Signed-off-by: Sachin Kamat Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 18 ++++++------------ 1 file changed, 6 insertions(+), 12 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 89f83f8e56b2..ce0d36412c97 100755 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -1701,9 +1701,8 @@ static int max98090_dai_set_fmt(struct snd_soc_dai *codec_dai, * seen for the case of TDM mode. The remaining cases have * normal logic. */ - if (max98090->tdm_slots > 1) { + if (max98090->tdm_slots > 1) regval ^= M98090_BCI_MASK; - } snd_soc_write(codec, M98090_REG_INTERFACE_FORMAT, regval); @@ -2057,17 +2056,14 @@ static irqreturn_t max98090_interrupt(int irq, void *data) if (!active) return IRQ_NONE; - if (active & M98090_CLD_MASK) { + if (active & M98090_CLD_MASK) dev_err(codec->dev, "M98090_CLD_MASK\n"); - } - if (active & M98090_SLD_MASK) { + if (active & M98090_SLD_MASK) dev_dbg(codec->dev, "M98090_SLD_MASK\n"); - } - if (active & M98090_ULK_MASK) { + if (active & M98090_ULK_MASK) dev_err(codec->dev, "M98090_ULK_MASK\n"); - } if (active & M98090_JDET_MASK) { dev_dbg(codec->dev, "M98090_JDET_MASK\n"); @@ -2078,13 +2074,11 @@ static irqreturn_t max98090_interrupt(int irq, void *data) msecs_to_jiffies(100)); } - if (active & M98090_DRCACT_MASK) { + if (active & M98090_DRCACT_MASK) dev_dbg(codec->dev, "M98090_DRCACT_MASK\n"); - } - if (active & M98090_DRCCLP_MASK) { + if (active & M98090_DRCCLP_MASK) dev_err(codec->dev, "M98090_DRCCLP_MASK\n"); - } return IRQ_HANDLED; } -- cgit v1.2.3 From d686500ae87275ed58a074f9e5e8b35b9afe30d8 Mon Sep 17 00:00:00 2001 From: Andrey Smirnov Date: Mon, 18 Feb 2013 19:59:34 -0800 Subject: ASoC: si476x: Convert SI476X codec to use regmap The latest radio and MFD drivers for SI476X radio chips use regmap API to provide access to the registers and allow for caching of their values when the actual chip is powered off. Convert the codec driver to do the same, so it would not loose the settings when the radio driver powers the chip down. Signed-off-by: Andrey Smirnov Signed-off-by: Mark Brown --- sound/soc/codecs/si476x.c | 22 +++++++++++++++++++--- 1 file changed, 19 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index f2d61a187830..30aebbe6815e 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -45,13 +45,23 @@ static unsigned int si476x_codec_read(struct snd_soc_codec *codec, unsigned int reg) { int err; + unsigned int val; struct si476x_core *core = codec->control_data; si476x_core_lock(core); - err = si476x_core_cmd_get_property(core, reg); + if (!si476x_core_is_powered_up(core)) + regcache_cache_only(core->regmap, true); + + err = regmap_read(core->regmap, reg, &val); + + if (!si476x_core_is_powered_up(core)) + regcache_cache_only(core->regmap, false); si476x_core_unlock(core); - return err; + if (err < 0) + return err; + + return val; } static int si476x_codec_write(struct snd_soc_codec *codec, @@ -61,7 +71,13 @@ static int si476x_codec_write(struct snd_soc_codec *codec, struct si476x_core *core = codec->control_data; si476x_core_lock(core); - err = si476x_core_cmd_set_property(core, reg, val); + if (!si476x_core_is_powered_up(core)) + regcache_cache_only(core->regmap, true); + + err = regmap_write(core->regmap, reg, val); + + if (!si476x_core_is_powered_up(core)) + regcache_cache_only(core->regmap, false); si476x_core_unlock(core); return err; -- cgit v1.2.3 From 06d7c13325228a2272e21caa4aa60805bc4d0fe4 Mon Sep 17 00:00:00 2001 From: Andrey Smirnov Date: Mon, 18 Feb 2013 19:59:35 -0800 Subject: ASoC: si476x: Cosmetic changes to SI476X codec driver - Add appropriate license header - Change email address in MODULE_AUTHOR - Remove trailing whitespaces Signed-off-by: Andrey Smirnov Signed-off-by: Mark Brown --- sound/soc/codecs/si476x.c | 25 ++++++++++++++++++++++--- 1 file changed, 22 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/si476x.c b/sound/soc/codecs/si476x.c index 30aebbe6815e..68b648aabdb9 100644 --- a/sound/soc/codecs/si476x.c +++ b/sound/soc/codecs/si476x.c @@ -1,3 +1,22 @@ +/* + * sound/soc/codecs/si476x.c -- Codec driver for SI476X chips + * + * Copyright (C) 2012 Innovative Converged Devices(ICD) + * Copyright (C) 2013 Andrey Smirnov + * + * Author: Andrey Smirnov + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * 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 @@ -156,7 +175,7 @@ static int si476x_codec_set_dai_fmt(struct snd_soc_dai *codec_dai, dev_err(codec_dai->codec->dev, "Failed to set output format\n"); return err; } - + return 0; } @@ -197,7 +216,7 @@ static int si476x_codec_hw_params(struct snd_pcm_substream *substream, err = snd_soc_update_bits(dai->codec, SI476X_DIGITAL_IO_OUTPUT_FORMAT, SI476X_DIGITAL_IO_OUTPUT_WIDTH_MASK, - (width << SI476X_DIGITAL_IO_SLOT_SIZE_SHIFT) | + (width << SI476X_DIGITAL_IO_SLOT_SIZE_SHIFT) | (width << SI476X_DIGITAL_IO_SAMPLE_SIZE_SHIFT)); if (err < 0) { dev_err(dai->codec->dev, "Failed to set output width\n"); @@ -266,6 +285,6 @@ static struct platform_driver si476x_platform_driver = { }; module_platform_driver(si476x_platform_driver); -MODULE_AUTHOR("Andrey Smirnov "); +MODULE_AUTHOR("Andrey Smirnov "); MODULE_DESCRIPTION("ASoC Si4761/64 codec driver"); MODULE_LICENSE("GPL"); -- cgit v1.2.3 From 21eb2693dd3bb701f831588977f92c4b63eeb132 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 26 Feb 2013 23:36:37 +0000 Subject: ASoC: wm8960: Add input boost volume control Signed-off-by: Mark Brown --- sound/soc/codecs/wm8960.c | 10 ++++++++++ 1 file changed, 10 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index 9bb927325993..3fea242cbf1a 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -204,6 +204,7 @@ static const DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 50, 0); static const DECLARE_TLV_DB_SCALE(dac_tlv, -12700, 50, 1); static const DECLARE_TLV_DB_SCALE(bypass_tlv, -2100, 300, 0); static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1); +static const DECLARE_TLV_DB_SCALE(boost_tlv, -1200, 300, 1); static const struct snd_kcontrol_new wm8960_snd_controls[] = { SOC_DOUBLE_R_TLV("Capture Volume", WM8960_LINVOL, WM8960_RINVOL, @@ -213,6 +214,15 @@ SOC_DOUBLE_R("Capture Volume ZC Switch", WM8960_LINVOL, WM8960_RINVOL, SOC_DOUBLE_R("Capture Switch", WM8960_LINVOL, WM8960_RINVOL, 7, 1, 0), +SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT3 Volume", + WM8960_INBMIX1, 4, 7, 0, boost_tlv), +SOC_SINGLE_TLV("Right Input Boost Mixer RINPUT2 Volume", + WM8960_INBMIX1, 1, 7, 0, boost_tlv), +SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT3 Volume", + WM8960_INBMIX2, 4, 7, 0, boost_tlv), +SOC_SINGLE_TLV("Left Input Boost Mixer LINPUT2 Volume", + WM8960_INBMIX2, 1, 7, 0, boost_tlv), + SOC_DOUBLE_R_TLV("Playback Volume", WM8960_LDAC, WM8960_RDAC, 0, 255, 0, dac_tlv), -- cgit v1.2.3 From ff680a173506e0f5f15c1d9c70251e7e3208c761 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 4 Mar 2013 16:00:19 +0800 Subject: ASoC: arizona: If we only have a clock to synchronise with make it REFCLK If there is only one clock active the FLL should use REFCLK rather than SYNCCLK as the clock to synchronise with since REFCLK is always required. Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 76 +++++++++++++++++++++++----------------------- 1 file changed, 38 insertions(+), 38 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index e456cb4b196e..0599ff8ea935 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1132,14 +1132,30 @@ static void arizona_enable_fll(struct arizona_fll *fll, struct arizona *arizona = fll->arizona; int ret; - regmap_update_bits(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); - if (fll->sync_src >= 0) - arizona_apply_fll(arizona, fll->base + 0x10, sync, + /* + * If we have both REFCLK and SYNCCLK then enable both, + * otherwise apply the SYNCCLK settings to REFCLK. + */ + if (fll->ref_src >= 0 && fll->ref_src != fll->sync_src) { + regmap_update_bits(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); + if (fll->sync_src >= 0) + arizona_apply_fll(arizona, fll->base + 0x10, sync, + fll->sync_src); + } else if (fll->sync_src >= 0) { + regmap_update_bits(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); + } else { + arizona_fll_err(fll, "No clocks provided\n"); + return; + } if (!arizona_is_enabled_fll(fll)) pm_runtime_get(arizona->dev); @@ -1149,7 +1165,8 @@ static void arizona_enable_fll(struct arizona_fll *fll, regmap_update_bits(arizona->regmap, fll->base + 1, ARIZONA_FLL1_ENA, ARIZONA_FLL1_ENA); - if (fll->sync_src >= 0) + if (fll->ref_src >= 0 && fll->sync_src >= 0 && + fll->ref_src != fll->sync_src) regmap_update_bits(arizona->regmap, fll->base + 0x11, ARIZONA_FLL1_SYNC_ENA, ARIZONA_FLL1_SYNC_ENA); @@ -1180,9 +1197,6 @@ int arizona_set_fll_refclk(struct arizona_fll *fll, int source, struct arizona_fll_cfg ref, sync; int ret; - if (source < 0) - return -EINVAL; - if (fll->ref_src == source && fll->ref_freq == Fref) return 0; @@ -1216,39 +1230,25 @@ int arizona_set_fll(struct arizona_fll *fll, int source, struct arizona_fll_cfg ref, sync; int ret; - if (fll->ref_src < 0 || fll->ref_src == source) { - if (fll->sync_src == ARIZONA_FLL_SRC_NONE && - fll->ref_src == source && fll->ref_freq == Fref && - fll->fout == Fout) - return 0; - - if (Fout) { - ret = arizona_calc_fll(fll, &ref, Fref, Fout); - if (ret != 0) - return ret; - } - - fll->sync_src = ARIZONA_FLL_SRC_NONE; - fll->ref_src = source; - fll->ref_freq = Fref; - } else { - if (fll->sync_src == source && - fll->sync_freq == Fref && fll->fout == Fout) - return 0; - - if (Fout) { - ret = arizona_calc_fll(fll, &ref, fll->ref_freq, Fout); - if (ret != 0) - return ret; + if (fll->sync_src == source && + fll->sync_freq == Fref && fll->fout == Fout) + return 0; - ret = arizona_calc_fll(fll, &sync, Fref, Fout); + if (Fout) { + if (fll->ref_src >= 0) { + ret = arizona_calc_fll(fll, &ref, fll->ref_freq, + Fout); if (ret != 0) return ret; } - fll->sync_src = source; - fll->sync_freq = Fref; + ret = arizona_calc_fll(fll, &sync, Fref, Fout); + if (ret != 0) + return ret; } + + fll->sync_src = source; + fll->sync_freq = Fref; fll->fout = Fout; if (Fout) { -- cgit v1.2.3 From cadf2120ff756789a3adaac07c5b85a09649c66e Mon Sep 17 00:00:00 2001 From: Paul Handrigan Date: Tue, 5 Mar 2013 13:12:56 -0600 Subject: ASoC: cs42l73: If Internal MCLK is >= 6.4MHz, then set SCLK to 64*Fs. Signed-off-by: Paul Handrigan Acked-by: Brian Austin Signed-off-by: Mark Brown --- sound/soc/codecs/cs42l73.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/cs42l73.c b/sound/soc/codecs/cs42l73.c index 6361dab48bd1..3b20c86cdb01 100644 --- a/sound/soc/codecs/cs42l73.c +++ b/sound/soc/codecs/cs42l73.c @@ -1180,7 +1180,11 @@ static int cs42l73_pcm_hw_params(struct snd_pcm_substream *substream, priv->config[id].mmcc &= 0xC0; priv->config[id].mmcc |= cs42l73_mclk_coeffs[mclk_coeff].mmcc; priv->config[id].spc &= 0xFC; - priv->config[id].spc |= MCK_SCLK_MCLK; + /* Use SCLK=64*Fs if internal MCLK >= 6.4MHz */ + if (priv->mclk >= 6400000) + priv->config[id].spc |= MCK_SCLK_64FS; + else + priv->config[id].spc |= MCK_SCLK_MCLK; } else { /* CS42L73 Slave */ priv->config[id].spc &= 0xFC; -- cgit v1.2.3 From 576411be200ee0e0801f1fe57d5e7ee787bb1a90 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 5 Mar 2013 12:07:16 +0800 Subject: ASoC: arizona: Increase FLL synchroniser bandwidth for high frequencies If we are using a high freqency SYNCCLK then increasing the bandwidth of the synchroniser improves performance. Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 11 +++++++++++ 1 file changed, 11 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 0599ff8ea935..e3aee143487e 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1157,6 +1157,17 @@ static void arizona_enable_fll(struct arizona_fll *fll, return; } + /* + * Increase the bandwidth if we're not using a low frequency + * sync source. + */ + if (fll->sync_src >= 0 && fll->sync_freq > 100000) + regmap_update_bits(arizona->regmap, fll->base + 0x17, + ARIZONA_FLL1_SYNC_BW, 0); + else + regmap_update_bits(arizona->regmap, fll->base + 0x17, + ARIZONA_FLL1_SYNC_BW, ARIZONA_FLL1_SYNC_BW); + if (!arizona_is_enabled_fll(fll)) pm_runtime_get(arizona->dev); -- cgit v1.2.3 From 8f113d7d2606003e485c4e8452977750d916dbc6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 5 Mar 2013 12:08:57 +0800 Subject: ASoC: arizona: Optimise FLL loop gains For optimal performance the FLL loop gain should be adjusted depending on the frequency of the input clock for the loop. Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 43 +++++++++++++++++++++++++++++++++++++++---- 1 file changed, 39 insertions(+), 4 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index e3aee143487e..8b7855df99de 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -990,6 +990,16 @@ static struct { { 1000000, 13500000, 0, 1 }, }; +static struct { + unsigned int min; + unsigned int max; + u16 gain; +} fll_gains[] = { + { 0, 256000, 0 }, + { 256000, 1000000, 2 }, + { 1000000, 13500000, 4 }, +}; + struct arizona_fll_cfg { int n; int theta; @@ -997,6 +1007,7 @@ struct arizona_fll_cfg { int refdiv; int outdiv; int fratio; + int gain; }; static int arizona_calc_fll(struct arizona_fll *fll, @@ -1056,6 +1067,18 @@ 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)) { @@ -1083,13 +1106,15 @@ static int arizona_calc_fll(struct arizona_fll *fll, cfg->n, cfg->theta, cfg->lambda); arizona_fll_dbg(fll, "FRATIO=%x(%d) OUTDIV=%x REFCLK_DIV=%x\n", cfg->fratio, cfg->fratio, cfg->outdiv, cfg->refdiv); + arizona_fll_dbg(fll, "GAIN=%d\n", cfg->gain); return 0; } static void arizona_apply_fll(struct arizona *arizona, unsigned int base, - struct arizona_fll_cfg *cfg, int source) + struct arizona_fll_cfg *cfg, int source, + bool sync) { regmap_update_bits(arizona->regmap, base + 3, ARIZONA_FLL1_THETA_MASK, cfg->theta); @@ -1104,6 +1129,15 @@ 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(arizona->regmap, base + 0x7, + ARIZONA_FLL1_GAIN_MASK, + cfg->gain << ARIZONA_FLL1_GAIN_SHIFT); + else + regmap_update_bits(arizona->regmap, base + 0x9, + ARIZONA_FLL1_GAIN_MASK, + cfg->gain << ARIZONA_FLL1_GAIN_SHIFT); + regmap_update_bits(arizona->regmap, base + 2, ARIZONA_FLL1_CTRL_UPD | ARIZONA_FLL1_N_MASK, ARIZONA_FLL1_CTRL_UPD | cfg->n); @@ -1141,17 +1175,18 @@ static void arizona_enable_fll(struct arizona_fll *fll, ARIZONA_FLL1_OUTDIV_MASK, ref->outdiv << ARIZONA_FLL1_OUTDIV_SHIFT); - arizona_apply_fll(arizona, fll->base, ref, fll->ref_src); + arizona_apply_fll(arizona, fll->base, ref, fll->ref_src, + false); if (fll->sync_src >= 0) arizona_apply_fll(arizona, fll->base + 0x10, sync, - fll->sync_src); + fll->sync_src, true); } else if (fll->sync_src >= 0) { regmap_update_bits(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); + fll->sync_src, false); } else { arizona_fll_err(fll, "No clocks provided\n"); return; -- cgit v1.2.3 From b0ec761b99291f3c0f28ac370f94c145ec806095 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 6 Mar 2013 22:22:15 +0100 Subject: ASoC: ak4104: convert to direct regmap API usage Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/codecs/ak4104.c | 31 ++++++++++++++++--------------- 1 file changed, 16 insertions(+), 15 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index 6f6c335a5baa..58f390d3ea7a 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -55,6 +55,7 @@ static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int format) { struct snd_soc_codec *codec = codec_dai->codec; + struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec); int val = 0; int ret; @@ -77,9 +78,9 @@ static int ak4104_set_dai_fmt(struct snd_soc_dai *codec_dai, if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) return -EINVAL; - ret = snd_soc_update_bits(codec, AK4104_REG_CONTROL1, - AK4104_CONTROL1_DIF0 | AK4104_CONTROL1_DIF1, - val); + ret = regmap_update_bits(ak4104->regmap, AK4104_REG_CONTROL1, + AK4104_CONTROL1_DIF0 | AK4104_CONTROL1_DIF1, + val); if (ret < 0) return ret; @@ -91,11 +92,12 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; + struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec); int val = 0; /* set the IEC958 bits: consumer mode, no copyright bit */ val |= IEC958_AES0_CON_NOT_COPYRIGHT; - snd_soc_write(codec, AK4104_REG_CHN_STATUS(0), val); + regmap_write(ak4104->regmap, AK4104_REG_CHN_STATUS(0), val); val = 0; @@ -132,7 +134,7 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - return snd_soc_write(codec, AK4104_REG_CHN_STATUS(3), val); + return regmap_write(ak4104->regmap, AK4104_REG_CHN_STATUS(3), val); } static const struct snd_soc_dai_ops ak4101_dai_ops = { @@ -160,20 +162,17 @@ static int ak4104_probe(struct snd_soc_codec *codec) int ret; codec->control_data = ak4104->regmap; - ret = snd_soc_codec_set_cache_io(codec, 8, 8, SND_SOC_REGMAP); - if (ret != 0) - return ret; /* set power-up and non-reset bits */ - ret = snd_soc_update_bits(codec, AK4104_REG_CONTROL1, - AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN, - AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN); + ret = regmap_update_bits(ak4104->regmap, AK4104_REG_CONTROL1, + AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN, + AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN); if (ret < 0) return ret; /* enable transmitter */ - ret = snd_soc_update_bits(codec, AK4104_REG_TX, - AK4104_TX_TXE, AK4104_TX_TXE); + ret = regmap_update_bits(ak4104->regmap, AK4104_REG_TX, + AK4104_TX_TXE, AK4104_TX_TXE); if (ret < 0) return ret; @@ -182,8 +181,10 @@ static int ak4104_probe(struct snd_soc_codec *codec) static int ak4104_remove(struct snd_soc_codec *codec) { - snd_soc_update_bits(codec, AK4104_REG_CONTROL1, - AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN, 0); + struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec); + + regmap_update_bits(ak4104->regmap, AK4104_REG_CONTROL1, + AK4104_CONTROL1_PW | AK4104_CONTROL1_RSTN, 0); return 0; } -- cgit v1.2.3 From b692a436e1dc7227f2b7cf447797c3dc6ece5c29 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Wed, 6 Mar 2013 22:22:16 +0100 Subject: ASoC: ak4104: correct tranceiver enable handling Move the enabling of the TX diode to hw_params() and disable it again in hw_free(). This way, the diode is only switched on as long as it needs to be. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/codecs/ak4104.c | 26 ++++++++++++++++++++++++-- 1 file changed, 24 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index 58f390d3ea7a..c7cfdf957e4d 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -93,7 +93,7 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec); - int val = 0; + int ret, val = 0; /* set the IEC958 bits: consumer mode, no copyright bit */ val |= IEC958_AES0_CON_NOT_COPYRIGHT; @@ -134,11 +134,33 @@ static int ak4104_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - return regmap_write(ak4104->regmap, AK4104_REG_CHN_STATUS(3), val); + ret = regmap_write(ak4104->regmap, AK4104_REG_CHN_STATUS(3), val); + if (ret < 0) + return ret; + + /* enable transmitter */ + ret = regmap_update_bits(ak4104->regmap, AK4104_REG_TX, + AK4104_TX_TXE, AK4104_TX_TXE); + if (ret < 0) + return ret; + + return 0; +} + +static int ak4104_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak4104_private *ak4104 = snd_soc_codec_get_drvdata(codec); + + /* disable transmitter */ + return regmap_update_bits(ak4104->regmap, AK4104_REG_TX, + AK4104_TX_TXE, 0); } static const struct snd_soc_dai_ops ak4101_dai_ops = { .hw_params = ak4104_hw_params, + .hw_free = ak4104_hw_free, .set_fmt = ak4104_set_dai_fmt, }; -- cgit v1.2.3 From eca2e8e24a0c712c2613ce5704e9e73b693d2e98 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 6 Mar 2013 00:09:59 +0800 Subject: ASoC: arizona: Ensure synchroniser is disabled when not needed When live configuring a FLL configuration with no synchroniser disable the synchroniser in case the previous configuration used one. Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 3 +++ 1 file changed, 3 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 8b7855df99de..53ddd529769c 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1187,6 +1187,9 @@ static void arizona_enable_fll(struct arizona_fll *fll, arizona_apply_fll(arizona, fll->base, sync, fll->sync_src, false); + + regmap_update_bits(arizona->regmap, fll->base + 0x11, + ARIZONA_FLL1_SYNC_ENA, 0); } else { arizona_fll_err(fll, "No clocks provided\n"); return; -- cgit v1.2.3 From 86cd684fcb3220f4aa20cf9e32fd1059373a608a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 7 Mar 2013 16:14:04 +0800 Subject: ASoC: arizona: Suppress reference calculations when setting REFCLK to 0 Allow users to keep on specifying their output frequency when disabling the reference clock. Reported-by: Kyung Kwee Ryu Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 53ddd529769c..ad21d8255341 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -1249,7 +1249,7 @@ int arizona_set_fll_refclk(struct arizona_fll *fll, int source, if (fll->ref_src == source && fll->ref_freq == Fref) return 0; - if (fll->fout) { + if (fll->fout && Fref > 0) { ret = arizona_calc_fll(fll, &ref, Fref, fll->fout); if (ret != 0) return ret; @@ -1265,7 +1265,7 @@ int arizona_set_fll_refclk(struct arizona_fll *fll, int source, fll->ref_src = source; fll->ref_freq = Fref; - if (fll->fout) { + if (fll->fout && Fref > 0) { arizona_enable_fll(fll, &ref, &sync); } -- cgit v1.2.3 From cc289be8c913006a43275dfd8ed4ac56b43140a8 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Fri, 8 Mar 2013 12:07:28 +0100 Subject: ASoC: Add codec driver for AK5386 Adds a driver for Asahi Kasei's AK5386 Single-ended 24-Bit 192kHz delta-sigma ADC. The device has no control port interface but an optional RESET/PDN GPIO pin. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- Documentation/devicetree/bindings/sound/ak5386.txt | 19 +++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak5386.c | 152 +++++++++++++++++++++ 4 files changed, 177 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ak5386.txt create mode 100644 sound/soc/codecs/ak5386.c (limited to 'sound/soc/codecs') diff --git a/Documentation/devicetree/bindings/sound/ak5386.txt b/Documentation/devicetree/bindings/sound/ak5386.txt new file mode 100644 index 000000000000..dc3914fe6ce8 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak5386.txt @@ -0,0 +1,19 @@ +AK5386 Single-ended 24-Bit 192kHz delta-sigma ADC + +This device has no control interface. + +Required properties: + + - compatible : "asahi-kasei,ak5386" + +Optional properties: + + - reset-gpio : a GPIO spec for the reset/power down pin. + If specified, it will be deasserted at probe time. + +Example: + +spdif: ak5386@0 { + compatible = "asahi-kasei,ak5386"; + reset-gpio = <&gpio0 23>; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 45b72561c615..500f666c3875 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -26,6 +26,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4641 if I2C select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C + select SND_SOC_AK5386 select SND_SOC_ALC5623 if I2C select SND_SOC_ALC5632 if I2C select SND_SOC_CQ0093VC if MFD_DAVINCI_VOICECODEC @@ -203,6 +204,9 @@ config SND_SOC_AK4642 config SND_SOC_AK4671 tristate +config SND_SOC_AK5386 + tristate + config SND_SOC_ALC5623 tristate config SND_SOC_ALC5632 diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 6a3b3c3b8b41..3a7ec1c39741 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -14,6 +14,7 @@ snd-soc-ak4535-objs := ak4535.o snd-soc-ak4641-objs := ak4641.o snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o +snd-soc-ak5386-objs := ak5386.o snd-soc-arizona-objs := arizona.o snd-soc-cq93vc-objs := cq93vc.o snd-soc-cs42l51-objs := cs42l51.o @@ -137,6 +138,7 @@ obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o +obj-$(CONFIG_SND_SOC_AK5386) += snd-soc-ak5386.o obj-$(CONFIG_SND_SOC_ALC5623) += snd-soc-alc5623.o obj-$(CONFIG_SND_SOC_ALC5632) += snd-soc-alc5632.o obj-$(CONFIG_SND_SOC_ARIZONA) += snd-soc-arizona.o diff --git a/sound/soc/codecs/ak5386.c b/sound/soc/codecs/ak5386.c new file mode 100644 index 000000000000..1f303983ae02 --- /dev/null +++ b/sound/soc/codecs/ak5386.c @@ -0,0 +1,152 @@ +/* + * ALSA SoC driver for + * Asahi Kasei AK5386 Single-ended 24-Bit 192kHz delta-sigma ADC + * + * (c) 2013 Daniel Mack + * + * 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 +#include +#include +#include + +struct ak5386_priv { + int reset_gpio; +}; + +static struct snd_soc_codec_driver soc_codec_ak5386; + +static int ak5386_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + + format &= SND_SOC_DAIFMT_FORMAT_MASK; + if (format != SND_SOC_DAIFMT_LEFT_J && + format != SND_SOC_DAIFMT_I2S) { + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + return 0; +} + +static int ak5386_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 ak5386_priv *priv = snd_soc_codec_get_drvdata(codec); + + /* + * From the datasheet: + * + * All external clocks (MCLK, SCLK and LRCK) must be present unless + * PDN pin = “L”. If these clocks are not provided, the AK5386 may + * draw excess current due to its use of internal dynamically + * refreshed logic. If the external clocks are not present, place + * the AK5386 in power-down mode (PDN pin = “L”). + */ + + if (gpio_is_valid(priv->reset_gpio)) + gpio_set_value(priv->reset_gpio, 1); + + return 0; +} + +static int ak5386_hw_free(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak5386_priv *priv = snd_soc_codec_get_drvdata(codec); + + if (gpio_is_valid(priv->reset_gpio)) + gpio_set_value(priv->reset_gpio, 0); + + return 0; +} + +static const struct snd_soc_dai_ops ak5386_dai_ops = { + .set_fmt = ak5386_set_dai_fmt, + .hw_params = ak5386_hw_params, + .hw_free = ak5386_hw_free, +}; + +static struct snd_soc_dai_driver ak5386_dai = { + .name = "ak5386-hifi", + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_192000, + .formats = SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | + SNDRV_PCM_FMTBIT_S24_3LE, + }, + .ops = &ak5386_dai_ops, +}; + +#ifdef CONFIG_OF +static const struct of_device_id ak5386_dt_ids[] = { + { .compatible = "asahi-kasei,ak5386", }, + { } +}; +MODULE_DEVICE_TABLE(of, ak5386_dt_ids); +#endif + +static int ak5386_probe(struct platform_device *pdev) +{ + struct device *dev = &pdev->dev; + struct ak5386_priv *priv; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->reset_gpio = -EINVAL; + dev_set_drvdata(dev, priv); + + if (of_match_device(of_match_ptr(ak5386_dt_ids), dev)) + priv->reset_gpio = of_get_named_gpio(dev->of_node, + "reset-gpio", 0); + + if (gpio_is_valid(priv->reset_gpio)) + if (devm_gpio_request_one(dev, priv->reset_gpio, + GPIOF_OUT_INIT_LOW, + "AK5386 Reset")) + priv->reset_gpio = -EINVAL; + + return snd_soc_register_codec(dev, &soc_codec_ak5386, + &ak5386_dai, 1); +} + +static int ak5386_remove(struct platform_device *pdev) +{ + snd_soc_unregister_codec(&pdev->dev); + return 0; +} + +static struct platform_driver ak5386_driver = { + .probe = ak5386_probe, + .remove = ak5386_remove, + .driver = { + .name = "ak5386", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(ak5386_dt_ids), + }, +}; + +module_platform_driver(ak5386_driver); + +MODULE_DESCRIPTION("ASoC driver for AK5386 ADC"); +MODULE_AUTHOR("Daniel Mack "); +MODULE_LICENSE("GPL"); -- cgit v1.2.3 From 4fa89346fbc34750f96ec0c1b2b59b15596ab333 Mon Sep 17 00:00:00 2001 From: Daniel Mack Date: Fri, 8 Mar 2013 13:52:09 +0100 Subject: ALSA: ASoC: add codec driver for TI TAS5086 This patch adds a driver for TI's TA5086 6-channel PWM processor. This chip has a very unusual register layout, specifically because the registers are of unequal size, and multi-byte registers require bulk writes to take effect. Regmap does not support these kind of mappings. Currently, the driver does not touch any of the registers >= 0x20, so it doesn't matter, because the register map is mapped to an 8-bit array. In case more features will be added in the future that require access to higher registers, the entire regmap H/W I/O routines have to be open-coded. Signed-off-by: Daniel Mack Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/ti,tas5086.txt | 32 ++ include/sound/tas5086.h | 7 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tas5086.c | 601 +++++++++++++++++++++ 5 files changed, 646 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ti,tas5086.txt create mode 100644 include/sound/tas5086.h create mode 100644 sound/soc/codecs/tas5086.c (limited to 'sound/soc/codecs') diff --git a/Documentation/devicetree/bindings/sound/ti,tas5086.txt b/Documentation/devicetree/bindings/sound/ti,tas5086.txt new file mode 100644 index 000000000000..8ea4f5b4818d --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ti,tas5086.txt @@ -0,0 +1,32 @@ +Texas Instruments TAS5086 6-channel PWM Processor + +Required properties: + + - compatible: Should contain "ti,tas5086". + - reg: The i2c address. Should contain <0x1b>. + +Optional properties: + + - reset-gpio: A GPIO spec to define which pin is connected to the + chip's !RESET pin. If specified, the driver will + assert a hardware reset at probe time. + + - ti,charge-period: This property should contain the time in microseconds + that closely matches the external single-ended + split-capacitor charge period. The hardware chip + waits for this period of time before starting the + PWM signals. This helps reduce pops and clicks. + + When not specified, the hardware default of 1300ms + is retained. + +Examples: + + i2c_bus { + tas5086@1b { + compatible = "ti,tas5086"; + reg = <0x1b>; + reset-gpio = <&gpio 23 0>; + ti,charge-period = <156000>; + }; + }; diff --git a/include/sound/tas5086.h b/include/sound/tas5086.h new file mode 100644 index 000000000000..aac481b7db8f --- /dev/null +++ b/include/sound/tas5086.h @@ -0,0 +1,7 @@ +#ifndef _SND_SOC_CODEC_TAS5086_H_ +#define _SND_SOC_CODEC_TAS5086_H_ + +#define TAS5086_CLK_IDX_MCLK 0 +#define TAS5086_CLK_IDX_SCLK 1 + +#endif /* _SND_SOC_CODEC_TAS5086_H_ */ diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 45b72561c615..86b35245e559 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -63,6 +63,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_STA32X if I2C 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_TLV320AIC26 if SPI_MASTER select SND_SOC_TLV320AIC32X4 if I2C @@ -320,6 +321,9 @@ config SND_SOC_STA529 config SND_SOC_STAC9766 tristate +config SND_SOC_TAS5086 + tristate + config SND_SOC_TLV320AIC23 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 6a3b3c3b8b41..8077bc2b4f43 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -55,6 +55,7 @@ snd-soc-ssm2602-objs := ssm2602.o snd-soc-sta32x-objs := sta32x.o 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-tlv320aic26-objs := tlv320aic26.o snd-soc-tlv320aic3x-objs := tlv320aic3x.o @@ -177,6 +178,7 @@ obj-$(CONFIG_SND_SOC_SSM2602) += snd-soc-ssm2602.o obj-$(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 +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 diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c new file mode 100644 index 000000000000..008bea4d6208 --- /dev/null +++ b/sound/soc/codecs/tas5086.c @@ -0,0 +1,601 @@ +/* + * TAS5086 ASoC codec driver + * + * Copyright (c) 2013 Daniel Mack + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * + * 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. + * + * TODO: + * - implement DAPM and input muxing + * - implement modulation limit + * - implement non-default PWM start + * + * Note that this chip has a very unusual register layout, specifically + * because the registers are of unequal size, and multi-byte registers + * require bulk writes to take effect. Regmap does not support that kind + * of devices. + * + * Currently, the driver does not touch any of the registers >= 0x20, so + * it doesn't matter because the entire map can be accessed as 8-bit + * array. In case more features will be added in the future + * that require access to higher registers, the entire regmap H/W I/O + * routines have to be open-coded. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#define TAS5086_PCM_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_3LE) + +#define TAS5086_PCM_RATES (SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | \ + SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 | \ + SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_176400 | \ + SNDRV_PCM_RATE_192000) + +/* + * TAS5086 registers + */ +#define TAS5086_CLOCK_CONTROL 0x00 /* Clock control register */ +#define TAS5086_CLOCK_RATE(val) (val << 5) +#define TAS5086_CLOCK_RATE_MASK (0x7 << 5) +#define TAS5086_CLOCK_RATIO(val) (val << 2) +#define TAS5086_CLOCK_RATIO_MASK (0x7 << 2) +#define TAS5086_CLOCK_SCLK_RATIO_48 (1 << 1) +#define TAS5086_CLOCK_VALID (1 << 0) + +#define TAS5086_DEEMPH_MASK 0x03 +#define TAS5086_SOFT_MUTE_ALL 0x3f + +#define TAS5086_DEV_ID 0x01 /* Device ID register */ +#define TAS5086_ERROR_STATUS 0x02 /* Error status register */ +#define TAS5086_SYS_CONTROL_1 0x03 /* System control register 1 */ +#define TAS5086_SERIAL_DATA_IF 0x04 /* Serial data interface register */ +#define TAS5086_SYS_CONTROL_2 0x05 /* System control register 2 */ +#define TAS5086_SOFT_MUTE 0x06 /* Soft mute register */ +#define TAS5086_MASTER_VOL 0x07 /* Master volume */ +#define TAS5086_CHANNEL_VOL(X) (0x08 + (X)) /* Channel 1-6 volume */ +#define TAS5086_VOLUME_CONTROL 0x09 /* Volume control register */ +#define TAS5086_MOD_LIMIT 0x10 /* Modulation limit register */ +#define TAS5086_PWM_START 0x18 /* PWM start register */ +#define TAS5086_SURROUND 0x19 /* Surround register */ +#define TAS5086_SPLIT_CAP_CHARGE 0x1a /* Split cap charge period register */ +#define TAS5086_OSC_TRIM 0x1b /* Oscillator trim register */ +#define TAS5086_BKNDERR 0x1c + +/* + * Default TAS5086 power-up configuration + */ +static const struct reg_default tas5086_reg_defaults[] = { + { 0x00, 0x6c }, + { 0x01, 0x03 }, + { 0x02, 0x00 }, + { 0x03, 0xa0 }, + { 0x04, 0x05 }, + { 0x05, 0x60 }, + { 0x06, 0x00 }, + { 0x07, 0xff }, + { 0x08, 0x30 }, + { 0x09, 0x30 }, + { 0x0a, 0x30 }, + { 0x0b, 0x30 }, + { 0x0c, 0x30 }, + { 0x0d, 0x30 }, + { 0x0e, 0xb1 }, + { 0x0f, 0x00 }, + { 0x10, 0x02 }, + { 0x11, 0x00 }, + { 0x12, 0x00 }, + { 0x13, 0x00 }, + { 0x14, 0x00 }, + { 0x15, 0x00 }, + { 0x16, 0x00 }, + { 0x17, 0x00 }, + { 0x18, 0x3f }, + { 0x19, 0x00 }, + { 0x1a, 0x18 }, + { 0x1b, 0x82 }, + { 0x1c, 0x05 }, +}; + +static bool tas5086_accessible_reg(struct device *dev, unsigned int reg) +{ + return !((reg == 0x0f) || (reg >= 0x11 && reg <= 0x17)); +} + +static bool tas5086_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case TAS5086_DEV_ID: + case TAS5086_ERROR_STATUS: + return true; + } + + return false; +} + +static bool tas5086_writeable_reg(struct device *dev, unsigned int reg) +{ + return tas5086_accessible_reg(dev, reg) && (reg != TAS5086_DEV_ID); +} + +struct tas5086_private { + struct regmap *regmap; + unsigned int mclk, sclk; + unsigned int format; + bool deemph; + /* Current sample rate for de-emphasis control */ + int rate; + /* GPIO driving Reset pin, if any */ + int gpio_nreset; +}; + +static int tas5086_deemph[] = { 0, 32000, 44100, 48000 }; + +static int tas5086_set_deemph(struct snd_soc_codec *codec) +{ + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + int i, val = 0; + + if (priv->deemph) + for (i = 0; i < ARRAY_SIZE(tas5086_deemph); i++) + if (tas5086_deemph[i] == priv->rate) + val = i; + + return regmap_update_bits(priv->regmap, TAS5086_SYS_CONTROL_1, + TAS5086_DEEMPH_MASK, val); +} + +static int tas5086_get_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + ucontrol->value.enumerated.item[0] = priv->deemph; + + return 0; +} + +static int tas5086_put_deemph(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol); + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + priv->deemph = ucontrol->value.enumerated.item[0]; + + return tas5086_set_deemph(codec); +} + + +static int tas5086_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 tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + switch (clk_id) { + case TAS5086_CLK_IDX_MCLK: + priv->mclk = freq; + break; + case TAS5086_CLK_IDX_SCLK: + priv->sclk = freq; + break; + } + + return 0; +} + +static int tas5086_set_dai_fmt(struct snd_soc_dai *codec_dai, + unsigned int format) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + /* The TAS5086 can only be slave to all clocks */ + if ((format & SND_SOC_DAIFMT_MASTER_MASK) != SND_SOC_DAIFMT_CBS_CFS) { + dev_err(codec->dev, "Invalid clocking mode\n"); + return -EINVAL; + } + + /* we need to refer to the data format from hw_params() */ + priv->format = format; + + return 0; +} + +static const int tas5086_sample_rates[] = { + 32000, 38000, 44100, 48000, 88200, 96000, 176400, 192000 +}; + +static const int tas5086_ratios[] = { + 64, 128, 192, 256, 384, 512 +}; + +static int index_in_array(const int *array, int len, int needle) +{ + int i; + + for (i = 0; i < len; i++) + if (array[i] == needle) + return i; + + return -ENOENT; +} + +static int tas5086_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 tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int val; + int ret; + + priv->rate = params_rate(params); + + /* Look up the sample rate and refer to the offset in the list */ + val = index_in_array(tas5086_sample_rates, + ARRAY_SIZE(tas5086_sample_rates), priv->rate); + + if (val < 0) { + dev_err(codec->dev, "Invalid sample rate\n"); + return -EINVAL; + } + + ret = regmap_update_bits(priv->regmap, TAS5086_CLOCK_CONTROL, + TAS5086_CLOCK_RATE_MASK, + TAS5086_CLOCK_RATE(val)); + if (ret < 0) + return ret; + + /* MCLK / Fs ratio */ + val = index_in_array(tas5086_ratios, ARRAY_SIZE(tas5086_ratios), + priv->mclk / priv->rate); + if (val < 0) { + dev_err(codec->dev, "Inavlid MCLK / Fs ratio\n"); + return -EINVAL; + } + + ret = regmap_update_bits(priv->regmap, TAS5086_CLOCK_CONTROL, + TAS5086_CLOCK_RATIO_MASK, + TAS5086_CLOCK_RATIO(val)); + if (ret < 0) + return ret; + + + ret = regmap_update_bits(priv->regmap, TAS5086_CLOCK_CONTROL, + TAS5086_CLOCK_SCLK_RATIO_48, + (priv->sclk == 48 * priv->rate) ? + TAS5086_CLOCK_SCLK_RATIO_48 : 0); + if (ret < 0) + return ret; + + /* + * The chip has a very unituitive register mapping and muxes information + * about data format and sample depth into the same register, but not on + * a logical bit-boundary. Hence, we have to refer to the format passed + * in the set_dai_fmt() callback and set up everything from here. + * + * First, determine the 'base' value, using the format ... + */ + switch (priv->format & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + val = 0x00; + break; + case SND_SOC_DAIFMT_I2S: + val = 0x03; + break; + case SND_SOC_DAIFMT_LEFT_J: + val = 0x06; + break; + default: + dev_err(codec->dev, "Invalid DAI format\n"); + return -EINVAL; + } + + /* ... then add the offset for the sample bit depth. */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + val += 0; + break; + case SNDRV_PCM_FORMAT_S20_3LE: + val += 1; + break; + case SNDRV_PCM_FORMAT_S24_3LE: + val += 2; + break; + default: + dev_err(codec->dev, "Invalid bit width\n"); + return -EINVAL; + }; + + ret = regmap_write(priv->regmap, TAS5086_SERIAL_DATA_IF, val); + if (ret < 0) + return ret; + + /* clock is considered valid now */ + ret = regmap_update_bits(priv->regmap, TAS5086_CLOCK_CONTROL, + TAS5086_CLOCK_VALID, TAS5086_CLOCK_VALID); + if (ret < 0) + return ret; + + return tas5086_set_deemph(codec); +} + +static int tas5086_mute_stream(struct snd_soc_dai *dai, int mute, int stream) +{ + struct snd_soc_codec *codec = dai->codec; + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + unsigned int val = 0; + + if (mute) + val = TAS5086_SOFT_MUTE_ALL; + + return regmap_write(priv->regmap, TAS5086_SOFT_MUTE, val); +} + +/* TAS5086 controls */ +static const DECLARE_TLV_DB_SCALE(tas5086_dac_tlv, -10350, 50, 1); + +static const struct snd_kcontrol_new tas5086_controls[] = { + SOC_SINGLE_TLV("Master Playback Volume", TAS5086_MASTER_VOL, + 0, 0xff, 1, tas5086_dac_tlv), + SOC_DOUBLE_R_TLV("Channel 1/2 Playback Volume", + TAS5086_CHANNEL_VOL(0), TAS5086_CHANNEL_VOL(1), + 0, 0xff, 1, tas5086_dac_tlv), + SOC_DOUBLE_R_TLV("Channel 3/4 Playback Volume", + TAS5086_CHANNEL_VOL(2), TAS5086_CHANNEL_VOL(3), + 0, 0xff, 1, tas5086_dac_tlv), + SOC_DOUBLE_R_TLV("Channel 5/6 Playback Volume", + TAS5086_CHANNEL_VOL(4), TAS5086_CHANNEL_VOL(5), + 0, 0xff, 1, tas5086_dac_tlv), + SOC_SINGLE_BOOL_EXT("De-emphasis Switch", 0, + tas5086_get_deemph, tas5086_put_deemph), +}; + +static const struct snd_soc_dai_ops tas5086_dai_ops = { + .hw_params = tas5086_hw_params, + .set_sysclk = tas5086_set_dai_sysclk, + .set_fmt = tas5086_set_dai_fmt, + .mute_stream = tas5086_mute_stream, +}; + +static struct snd_soc_dai_driver tas5086_dai = { + .name = "tas5086-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 6, + .rates = TAS5086_PCM_RATES, + .formats = TAS5086_PCM_FORMATS, + }, + .ops = &tas5086_dai_ops, +}; + +#ifdef CONFIG_PM +static int tas5086_soc_resume(struct snd_soc_codec *codec) +{ + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + /* Restore codec state */ + return regcache_sync(priv->regmap); +} +#else +#define tas5086_soc_resume NULL +#endif /* CONFIG_PM */ + +#ifdef CONFIG_OF +static const struct of_device_id tas5086_dt_ids[] = { + { .compatible = "ti,tas5086", }, + { } +}; +MODULE_DEVICE_TABLE(of, tas5086_dt_ids); +#endif + +/* charge period values in microseconds */ +static const int tas5086_charge_period[] = { + 13000, 16900, 23400, 31200, 41600, 54600, 72800, 96200, + 130000, 156000, 234000, 312000, 416000, 546000, 728000, 962000, + 1300000, 169000, 2340000, 3120000, 4160000, 5460000, 7280000, 9620000, +}; + +static int tas5086_probe(struct snd_soc_codec *codec) +{ + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + int charge_period = 1300000; /* hardware default is 1300 ms */ + int i, ret; + + if (of_match_device(of_match_ptr(tas5086_dt_ids), codec->dev)) { + struct device_node *of_node = codec->dev->of_node; + of_property_read_u32(of_node, "ti,charge-period", &charge_period); + } + + /* lookup and set split-capacitor charge period */ + if (charge_period == 0) { + regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, 0); + } else { + i = index_in_array(tas5086_charge_period, + ARRAY_SIZE(tas5086_charge_period), + charge_period); + if (i >= 0) + regmap_write(priv->regmap, TAS5086_SPLIT_CAP_CHARGE, + i + 0x08); + else + dev_warn(codec->dev, + "Invalid split-cap charge period of %d ns.\n", + charge_period); + } + + /* enable factory trim */ + ret = regmap_write(priv->regmap, TAS5086_OSC_TRIM, 0x00); + if (ret < 0) + return ret; + + /* start all channels */ + ret = regmap_write(priv->regmap, TAS5086_SYS_CONTROL_2, 0x20); + if (ret < 0) + return ret; + + /* set master volume to 0 dB */ + ret = regmap_write(priv->regmap, TAS5086_MASTER_VOL, 0x30); + if (ret < 0) + return ret; + + /* mute all channels for now */ + ret = regmap_write(priv->regmap, TAS5086_SOFT_MUTE, + TAS5086_SOFT_MUTE_ALL); + if (ret < 0) + return ret; + + return 0; +} + +static int tas5086_remove(struct snd_soc_codec *codec) +{ + struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); + + if (gpio_is_valid(priv->gpio_nreset)) + /* Set codec to the reset state */ + gpio_set_value(priv->gpio_nreset, 0); + + return 0; +}; + +static struct snd_soc_codec_driver soc_codec_dev_tas5086 = { + .probe = tas5086_probe, + .remove = tas5086_remove, + .resume = tas5086_soc_resume, + .controls = tas5086_controls, + .num_controls = ARRAY_SIZE(tas5086_controls), +}; + +static const struct i2c_device_id tas5086_i2c_id[] = { + { "tas5086", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, tas5086_i2c_id); + +static const struct regmap_config tas5086_regmap = { + .reg_bits = 8, + .val_bits = 8, + .max_register = ARRAY_SIZE(tas5086_reg_defaults), + .reg_defaults = tas5086_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(tas5086_reg_defaults), + .cache_type = REGCACHE_RBTREE, + .volatile_reg = tas5086_volatile_reg, + .writeable_reg = tas5086_writeable_reg, + .readable_reg = tas5086_accessible_reg, +}; + +static int tas5086_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct tas5086_private *priv; + struct device *dev = &i2c->dev; + int gpio_nreset = -EINVAL; + int i, ret; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->regmap = devm_regmap_init_i2c(i2c, &tas5086_regmap); + if (IS_ERR(priv->regmap)) { + ret = PTR_ERR(priv->regmap); + dev_err(&i2c->dev, "Failed to create regmap: %d\n", ret); + return ret; + } + + i2c_set_clientdata(i2c, priv); + + if (of_match_device(of_match_ptr(tas5086_dt_ids), dev)) { + struct device_node *of_node = dev->of_node; + gpio_nreset = of_get_named_gpio(of_node, "reset-gpio", 0); + } + + if (gpio_is_valid(gpio_nreset)) + if (devm_gpio_request(dev, gpio_nreset, "TAS5086 Reset")) + gpio_nreset = -EINVAL; + + if (gpio_is_valid(gpio_nreset)) { + /* Reset codec - minimum assertion time is 400ns */ + gpio_direction_output(gpio_nreset, 0); + udelay(1); + gpio_set_value(gpio_nreset, 1); + + /* Codec needs ~15ms to wake up */ + msleep(15); + } + + priv->gpio_nreset = gpio_nreset; + + /* The TAS5086 always returns 0x03 in its TAS5086_DEV_ID register */ + ret = regmap_read(priv->regmap, TAS5086_DEV_ID, &i); + if (ret < 0) + return ret; + + if (i != 0x3) { + dev_err(dev, + "Failed to identify TAS5086 codec (got %02x)\n", i); + return -ENODEV; + } + + return snd_soc_register_codec(&i2c->dev, &soc_codec_dev_tas5086, + &tas5086_dai, 1); +} + +static int tas5086_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_codec(&i2c->dev); + return 0; +} + +static struct i2c_driver tas5086_i2c_driver = { + .driver = { + .name = "tas5086", + .owner = THIS_MODULE, + .of_match_table = of_match_ptr(tas5086_dt_ids), + }, + .id_table = tas5086_i2c_id, + .probe = tas5086_i2c_probe, + .remove = tas5086_i2c_remove, +}; + +static int __init tas5086_modinit(void) +{ + return i2c_add_driver(&tas5086_i2c_driver); +} +module_init(tas5086_modinit); + +static void __exit tas5086_modexit(void) +{ + i2c_del_driver(&tas5086_i2c_driver); +} +module_exit(tas5086_modexit); + +MODULE_AUTHOR("Daniel Mack "); +MODULE_DESCRIPTION("Texas Instruments TAS5086 ALSA SoC Codec Driver"); +MODULE_LICENSE("GPL"); -- cgit v1.2.3 From c300d6de53ae029576b2805f08d8596d2e511b08 Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Tue, 12 Mar 2013 21:36:24 +0800 Subject: ASoC: tas5086: use module_i2c_driver to simplify the code Use the module_i2c_driver() macro to make the code smaller and a bit simpler. Signed-off-by: Wei Yongjun Acked-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/codecs/tas5086.c | 12 +----------- 1 file changed, 1 insertion(+), 11 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 008bea4d6208..40cee844f0cf 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -584,17 +584,7 @@ static struct i2c_driver tas5086_i2c_driver = { .remove = tas5086_i2c_remove, }; -static int __init tas5086_modinit(void) -{ - return i2c_add_driver(&tas5086_i2c_driver); -} -module_init(tas5086_modinit); - -static void __exit tas5086_modexit(void) -{ - i2c_del_driver(&tas5086_i2c_driver); -} -module_exit(tas5086_modexit); +module_i2c_driver(tas5086_i2c_driver); MODULE_AUTHOR("Daniel Mack "); MODULE_DESCRIPTION("Texas Instruments TAS5086 ALSA SoC Codec Driver"); -- cgit v1.2.3 From 14a1b8ca172f4cfbc544051a729d85a380447a82 Mon Sep 17 00:00:00 2001 From: Tim Gardner Date: Mon, 11 Mar 2013 13:18:23 -0600 Subject: ASoC: adau1373: adau1373_hw_params: Silence overflow warning ADAU1373_BCLKDIV_SOURCE is defined as BIT(5) which uses UL constants. On amd64 the result of the ones complement operator is then truncated to unsigned int according to the prototype of snd_soc_update_bits(). I think gcc is correctly warning that the upper 32 bits are lost. sound/soc/codecs/adau1373.c: In function 'adau1373_hw_params': sound/soc/codecs/adau1373.c:940:3: warning: large integer implicitly truncated to unsigned type [-Woverflow] gcc version 4.6.3 Add 2 more BCLKDIV mask macros as explained by Lars: The BCLKDIV has three fields. The bitclock divider (bit 0-1), the samplerate (bit 2-4) and the source select (bit 5). Here we want to update the bitclock divider field and the samplerate field. When I wrote the code I was lazy and used ~ADAU1373_BCLKDIV_SOURCE as the mask, which for this register is functionally equivalent to ADAU1373_BCLKDIV_SR_MASK | ADAU1373_BCLKDIV_BCLK_MASK. Signed-off-by: Tim Gardner Acked-by: Lars-Peter Clausen Signed-off-by: Mark Brown --- sound/soc/codecs/adau1373.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/adau1373.c b/sound/soc/codecs/adau1373.c index 068b3ae56a17..1aa10ddf3a61 100644 --- a/sound/soc/codecs/adau1373.c +++ b/sound/soc/codecs/adau1373.c @@ -133,6 +133,8 @@ struct adau1373 { #define ADAU1373_DAI_FORMAT_DSP 0x3 #define ADAU1373_BCLKDIV_SOURCE BIT(5) +#define ADAU1373_BCLKDIV_SR_MASK (0x07 << 2) +#define ADAU1373_BCLKDIV_BCLK_MASK 0x03 #define ADAU1373_BCLKDIV_32 0x03 #define ADAU1373_BCLKDIV_64 0x02 #define ADAU1373_BCLKDIV_128 0x01 @@ -937,7 +939,8 @@ static int adau1373_hw_params(struct snd_pcm_substream *substream, adau1373_dai->enable_src = (div != 0); snd_soc_update_bits(codec, ADAU1373_BCLKDIV(dai->id), - ~ADAU1373_BCLKDIV_SOURCE, (div << 2) | ADAU1373_BCLKDIV_64); + ADAU1373_BCLKDIV_SR_MASK | ADAU1373_BCLKDIV_BCLK_MASK, + (div << 2) | ADAU1373_BCLKDIV_64); switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: -- cgit v1.2.3 From 1f5353e765fe2a1168477bfe55e4dd7cdd96b477 Mon Sep 17 00:00:00 2001 From: Tim Gardner Date: Sun, 10 Mar 2013 10:58:21 -0600 Subject: ASoC: wm_hubs: Silence reg_r and reg_l 'may be used uninitialized' warnings Return an error from wm_hubs_read_dc_servo() if hubs->dcs_readback_mode is not correctly initialized. You might as well bail out since nothing is likely to work correctly afterwards. sound/soc/codecs/wm_hubs.c:321:11: warning: 'reg_r' may be used uninitialized in this function [-Wuninitialized] sound/soc/codecs/wm_hubs.c:251:13: note: 'reg_r' was declared here sound/soc/codecs/wm_hubs.c:322:11: warning: 'reg_l' may be used uninitialized in this function [-Wuninitialized] sound/soc/codecs/wm_hubs.c:251:6: note: 'reg_l' was declared here gcc version 4.6.3 Signed-off-by: Tim Gardner Signed-off-by: Mark Brown --- sound/soc/codecs/wm_hubs.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm_hubs.c b/sound/soc/codecs/wm_hubs.c index 867ae97ddcec..f5d81b948759 100644 --- a/sound/soc/codecs/wm_hubs.c +++ b/sound/soc/codecs/wm_hubs.c @@ -199,11 +199,12 @@ static void wm_hubs_dcs_cache_set(struct snd_soc_codec *codec, u16 dcs_cfg) list_add_tail(&cache->list, &hubs->dcs_cache); } -static void wm_hubs_read_dc_servo(struct snd_soc_codec *codec, +static int wm_hubs_read_dc_servo(struct snd_soc_codec *codec, u16 *reg_l, u16 *reg_r) { struct wm_hubs_data *hubs = snd_soc_codec_get_drvdata(codec); u16 dcs_reg, reg; + int ret = 0; switch (hubs->dcs_readback_mode) { case 2: @@ -236,8 +237,9 @@ static void wm_hubs_read_dc_servo(struct snd_soc_codec *codec, break; default: WARN(1, "Unknown DCS readback method\n"); - return; + ret = -1; } + return ret; } /* @@ -286,7 +288,8 @@ static void enable_dc_servo(struct snd_soc_codec *codec) WM8993_DCS_TRIG_STARTUP_1); } - wm_hubs_read_dc_servo(codec, ®_l, ®_r); + if (wm_hubs_read_dc_servo(codec, ®_l, ®_r) < 0) + return; dev_dbg(codec->dev, "DCS input: %x %x\n", reg_l, reg_r); -- cgit v1.2.3 From 28dbd1611f5701c9b5b8c07924c1bd2ad6f64435 Mon Sep 17 00:00:00 2001 From: Dan Carpenter Date: Wed, 13 Mar 2013 08:32:53 +0300 Subject: ASoC: tas5086: signedness bug in tas5086_hw_params() "val" has to be signed for the error handling to work. Signed-off-by: Dan Carpenter Acked-by: Daniel Mack Signed-off-by: Mark Brown --- sound/soc/codecs/tas5086.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/tas5086.c b/sound/soc/codecs/tas5086.c index 40cee844f0cf..d447c4aa1d5e 100644 --- a/sound/soc/codecs/tas5086.c +++ b/sound/soc/codecs/tas5086.c @@ -251,7 +251,7 @@ static int tas5086_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct tas5086_private *priv = snd_soc_codec_get_drvdata(codec); - unsigned int val; + int val; int ret; priv->rate = params_rate(params); -- cgit v1.2.3 From 3f341f741de956980775761370e3abc4122be53a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 8 Mar 2013 15:22:29 +0800 Subject: ASoC: arizona: Provide defines for the clock rates Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 12 ++++++------ sound/soc/codecs/arizona.h | 8 ++++++++ 2 files changed, 14 insertions(+), 6 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index ad21d8255341..0c70d503fd32 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -504,27 +504,27 @@ int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id, break; case 11289600: case 12288000: - val |= 1 << ARIZONA_SYSCLK_FREQ_SHIFT; + val |= ARIZONA_CLK_12MHZ << ARIZONA_SYSCLK_FREQ_SHIFT; break; case 22579200: case 24576000: - val |= 2 << ARIZONA_SYSCLK_FREQ_SHIFT; + val |= ARIZONA_CLK_24MHZ << ARIZONA_SYSCLK_FREQ_SHIFT; break; case 45158400: case 49152000: - val |= 3 << ARIZONA_SYSCLK_FREQ_SHIFT; + val |= ARIZONA_CLK_49MHZ << ARIZONA_SYSCLK_FREQ_SHIFT; break; case 67737600: case 73728000: - val |= 4 << ARIZONA_SYSCLK_FREQ_SHIFT; + val |= ARIZONA_CLK_73MHZ << ARIZONA_SYSCLK_FREQ_SHIFT; break; case 90316800: case 98304000: - val |= 5 << ARIZONA_SYSCLK_FREQ_SHIFT; + val |= ARIZONA_CLK_98MHZ << ARIZONA_SYSCLK_FREQ_SHIFT; break; case 135475200: case 147456000: - val |= 6 << ARIZONA_SYSCLK_FREQ_SHIFT; + val |= ARIZONA_CLK_147MHZ << ARIZONA_SYSCLK_FREQ_SHIFT; break; case 0: dev_dbg(arizona->dev, "%s cleared\n", name); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index d592adcc969c..572f11bc90b4 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -49,6 +49,14 @@ #define ARIZONA_MIXER_VOL_SHIFT 1 #define ARIZONA_MIXER_VOL_WIDTH 7 +#define ARIZONA_CLK_6MHZ 0 +#define ARIZONA_CLK_12MHZ 1 +#define ARIZONA_CLK_24MHZ 2 +#define ARIZONA_CLK_49MHZ 3 +#define ARIZONA_CLK_73MHZ 4 +#define ARIZONA_CLK_98MHZ 5 +#define ARIZONA_CLK_147MHZ 6 + #define ARIZONA_MAX_DAI 4 #define ARIZONA_MAX_ADSP 4 -- cgit v1.2.3 From f395a21853935ab7a2d0d760cda206ae55300194 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 5 Mar 2013 22:39:54 +0800 Subject: ASoC: wm_adsp: Handle old .bin files Older .bin files report the global coefficients as absolute address writes to zero; maintain compatibility with them. Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 26 ++++++++++++++++++++++---- sound/soc/codecs/wm_adsp.h | 2 ++ 2 files changed, 24 insertions(+), 4 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f3f7e75f8628..febb4c76535e 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -549,8 +549,9 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) buf_size = sizeof(adsp1_id); algs = be32_to_cpu(adsp1_id.algs); + dsp->fw_id = be32_to_cpu(adsp1_id.fw.id); adsp_info(dsp, "Firmware: %x v%d.%d.%d, %zu algorithms\n", - be32_to_cpu(adsp1_id.fw.id), + dsp->fw_id, (be32_to_cpu(adsp1_id.fw.ver) & 0xff0000) >> 16, (be32_to_cpu(adsp1_id.fw.ver) & 0xff00) >> 8, be32_to_cpu(adsp1_id.fw.ver) & 0xff, @@ -573,8 +574,9 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) buf_size = sizeof(adsp2_id); algs = be32_to_cpu(adsp2_id.algs); + dsp->fw_id = be32_to_cpu(adsp2_id.fw.id); adsp_info(dsp, "Firmware: %x v%d.%d.%d, %zu algorithms\n", - be32_to_cpu(adsp2_id.fw.id), + dsp->fw_id, (be32_to_cpu(adsp2_id.fw.ver) & 0xff0000) >> 16, (be32_to_cpu(adsp2_id.fw.ver) & 0xff00) >> 8, be32_to_cpu(adsp2_id.fw.ver) & 0xff, @@ -781,8 +783,24 @@ static int wm_adsp_load_coeff(struct wm_adsp *dsp) case (WMFW_INFO_TEXT << 8): break; case (WMFW_ABSOLUTE << 8): - region_name = "register"; - reg = offset; + /* + * Old files may use this for global + * coefficients. + */ + if (le32_to_cpu(blk->id) == dsp->fw_id && + offset == 0) { + region_name = "global coefficients"; + mem = wm_adsp_find_region(dsp, type); + if (!mem) { + adsp_err(dsp, "No ZM\n"); + break; + } + reg = wm_adsp_region_to_reg(mem, 0); + + } else { + region_name = "register"; + reg = offset; + } break; case WMFW_ADSP1_DM: diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index cb8871a3ec00..d6fd8af53b5d 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -46,6 +46,8 @@ struct wm_adsp { struct list_head alg_regions; + int fw_id; + const struct wm_adsp_region *mem; int num_mems; -- cgit v1.2.3 From 76bf969e6f86e5de788dd943ff2d4340bac71822 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 5 Mar 2013 14:17:47 +0800 Subject: ASoC: arizona: Ensure we clock two channels for I2S mode I2S requires stereo clocking even for mono data. Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 0c70d503fd32..2b0803ec8234 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -818,7 +818,7 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, struct arizona *arizona = priv->arizona; int base = dai->driver->base; const int *rates; - int i, ret; + int i, ret, val; int chan_limit = arizona->pdata.max_channels_clocked[dai->id - 1]; int bclk, lrclk, wl, frame, bclk_target; @@ -834,6 +834,13 @@ static int arizona_hw_params(struct snd_pcm_substream *substream, bclk_target *= chan_limit; } + /* Force stereo for I2S mode */ + val = snd_soc_read(codec, base + ARIZONA_AIF_FORMAT); + if (params_channels(params) == 1 && (val & ARIZONA_AIF1_FMT_MASK)) { + arizona_aif_dbg(dai, "Forcing stereo mode\n"); + bclk_target *= 2; + } + for (i = 0; i < ARRAY_SIZE(arizona_44k1_bclk_rates); i++) { if (rates[i] >= bclk_target && rates[i] % params_rate(params) == 0) { -- cgit v1.2.3 From 00aa0fac76c2f3b5a3543a63798af12c6d48b9b1 Mon Sep 17 00:00:00 2001 From: Alban Bedel Date: Wed, 20 Mar 2013 17:37:32 +0100 Subject: ASoC: wm8903: Add the DAC boost control Signed-off-by: Alban Bedel Signed-off-by: Mark Brown --- sound/soc/codecs/wm8903.c | 4 ++++ 1 file changed, 4 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8903.c b/sound/soc/codecs/wm8903.c index 134e41c870b9..70e5eb200485 100644 --- a/sound/soc/codecs/wm8903.c +++ b/sound/soc/codecs/wm8903.c @@ -478,6 +478,8 @@ static int wm8903_put_deemph(struct snd_kcontrol *kcontrol, /* ALSA can only do steps of .01dB */ static const DECLARE_TLV_DB_SCALE(digital_tlv, -7200, 75, 1); +static const DECLARE_TLV_DB_SCALE(dac_boost_tlv, 0, 600, 0); + static const DECLARE_TLV_DB_SCALE(digital_sidetone_tlv, -3600, 300, 0); static const DECLARE_TLV_DB_SCALE(out_tlv, -5700, 100, 0); @@ -698,6 +700,8 @@ SOC_ENUM("DAC Mute Mode", mute_mode), SOC_SINGLE("DAC Mono Switch", WM8903_DAC_DIGITAL_1, 12, 1, 0), SOC_ENUM("DAC Companding Mode", dac_companding), SOC_SINGLE("DAC Companding Switch", WM8903_AUDIO_INTERFACE_0, 1, 1, 0), +SOC_SINGLE_TLV("DAC Boost Volume", WM8903_AUDIO_INTERFACE_0, 9, 3, 0, + dac_boost_tlv), SOC_SINGLE_BOOL_EXT("Playback Deemphasis Switch", 0, wm8903_get_deemph, wm8903_put_deemph), -- cgit v1.2.3 From 3cf956eebe54cdb7cf1701642085507f0354e56a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 20 Mar 2013 10:12:10 +0100 Subject: ASoC: wm8994: Support constraining the maximum number of channels clocked Some systems use the audio CODEC to clock a DAI with multiple data lines in parallel, meaning that bit clocks are only required for a smaller number of channels than data is sent for. In some cases providing the extra bit clocks can take the other devices on the audio bus out of spec. Support such systems by allowing a maximum number of channels to be specified. Signed-off-by: Mark Brown --- include/linux/mfd/wm8994/pdata.h | 8 ++++++++ sound/soc/codecs/wm8994.c | 13 +++++++++++-- 2 files changed, 19 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/include/linux/mfd/wm8994/pdata.h b/include/linux/mfd/wm8994/pdata.h index 8e21a094836d..68e776594889 100644 --- a/include/linux/mfd/wm8994/pdata.h +++ b/include/linux/mfd/wm8994/pdata.h @@ -17,6 +17,7 @@ #define WM8994_NUM_LDO 2 #define WM8994_NUM_GPIO 11 +#define WM8994_NUM_AIF 3 struct wm8994_ldo_pdata { /** GPIOs to enable regulator, 0 or less if not available */ @@ -215,6 +216,13 @@ struct wm8994_pdata { * system. */ bool spkmode_pu; + + /** + * Maximum number of channels clocks will be generated for, + * useful for systems where and I2S bus with multiple data + * lines is mastered. + */ + int max_channels_clocked[WM8994_NUM_AIF]; }; #endif diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index c9bd445c4976..318ea64b9800 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2656,6 +2656,8 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_codec *codec = dai->codec; struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + struct wm8994 *control = wm8994->wm8994; + struct wm8994_pdata *pdata = &control->pdata; int aif1_reg; int aif2_reg; int bclk_reg; @@ -2723,7 +2725,14 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, } wm8994->channels[id] = params_channels(params); - switch (params_channels(params)) { + if (pdata->max_channels_clocked[id] && + wm8994->channels[id] > pdata->max_channels_clocked[id]) { + dev_dbg(dai->dev, "Constraining channels to %d from %d\n", + pdata->max_channels_clocked[id], wm8994->channels[id]); + wm8994->channels[id] = pdata->max_channels_clocked[id]; + } + + switch (wm8994->channels[id]) { case 1: case 2: bclk_rate *= 2; @@ -2745,7 +2754,7 @@ static int wm8994_hw_params(struct snd_pcm_substream *substream, dev_dbg(dai->dev, "AIF%dCLK is %dHz, target BCLK %dHz\n", dai->id, wm8994->aifclk[id], bclk_rate); - if (params_channels(params) == 1 && + if (wm8994->channels[id] == 1 && (snd_soc_read(codec, aif1_reg) & 0x18) == 0x18) aif2 |= WM8994_AIF1_MONO; -- cgit v1.2.3 From 56447e1324009d7e3cec40e3cc2987843b59a00f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 10 Jan 2013 14:45:58 +0000 Subject: ASoC: arizona: Factor out speaker widgets from CODEC drivers Some system designs have been identified which repurpose portions of the speaker driver circuits for other functions which will require that they not be managed using DAPM. Prepare for this by factoring out the creation of the speaker widgets into the core driver, the widgets will be replaced by dummy ones when the additional functions are enabled. Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 82 ++++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/arizona.h | 5 +++ sound/soc/codecs/wm5102.c | 52 ++--------------------------- sound/soc/codecs/wm5110.c | 6 ---- 4 files changed, 89 insertions(+), 56 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 2b0803ec8234..009810b8c667 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -14,6 +14,7 @@ #include #include #include +#include #include #include #include @@ -66,6 +67,87 @@ #define arizona_aif_dbg(_dai, fmt, ...) \ dev_dbg(_dai->dev, "AIF%d: " fmt, _dai->id, ##__VA_ARGS__) +static int arizona_spk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct snd_soc_codec *codec = w->codec; + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + bool manual_ena = false; + + switch (arizona->type) { + case WM5102: + switch (arizona->rev) { + case 0: + break; + default: + manual_ena = true; + break; + } + default: + break; + } + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + if (!priv->spk_ena && manual_ena) { + snd_soc_write(codec, 0x4f5, 0x25a); + priv->spk_ena_pending = true; + } + break; + case SND_SOC_DAPM_POST_PMU: + if (priv->spk_ena_pending) { + msleep(75); + snd_soc_write(codec, 0x4f5, 0xda); + priv->spk_ena_pending = false; + priv->spk_ena++; + } + break; + case SND_SOC_DAPM_PRE_PMD: + if (manual_ena) { + priv->spk_ena--; + if (!priv->spk_ena) + snd_soc_write(codec, 0x4f5, 0x25a); + } + break; + case SND_SOC_DAPM_POST_PMD: + if (manual_ena) { + if (!priv->spk_ena) + snd_soc_write(codec, 0x4f5, 0x0da); + } + break; + } + + return 0; +} + +static const struct snd_soc_dapm_widget arizona_spkl = + SND_SOC_DAPM_PGA_E("OUT4L", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, arizona_spk_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU); + +static const struct snd_soc_dapm_widget arizona_spkr = + SND_SOC_DAPM_PGA_E("OUT4R", ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, arizona_spk_ev, + SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU); + +int arizona_init_spk(struct snd_soc_codec *codec) +{ + int ret; + + ret = snd_soc_dapm_new_controls(&codec->dapm, &arizona_spkl, 1); + if (ret != 0) + return ret; + + ret = snd_soc_dapm_new_controls(&codec->dapm, &arizona_spkr, 1); + if (ret != 0) + return ret; + + return 0; +} +EXPORT_SYMBOL_GPL(arizona_init_spk); + const char *arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = { "None", "Tone Generator 1", diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 572f11bc90b4..9399940f700d 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -76,6 +76,9 @@ struct arizona_priv { int num_inputs; unsigned int in_pending; + + unsigned int spk_ena:2; + unsigned int spk_ena_pending:1; }; #define ARIZONA_NUM_MIXER_INPUTS 99 @@ -228,6 +231,8 @@ extern int arizona_set_fll_refclk(struct arizona_fll *fll, int source, extern int arizona_set_fll(struct arizona_fll *fll, int source, unsigned int Fref, unsigned int Fout); +extern int arizona_init_spk(struct snd_soc_codec *codec); + extern int arizona_init_dai(struct arizona_priv *priv, int dai); int arizona_set_output_mode(struct snd_soc_codec *codec, int output, diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 44d4c69d25e5..97757bc5fd0e 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -36,9 +36,6 @@ struct wm5102_priv { struct arizona_priv core; struct arizona_fll fll[2]; - - unsigned int spk_ena:2; - unsigned int spk_ena_pending:1; }; static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); @@ -817,47 +814,6 @@ ARIZONA_MIXER_CONTROLS("AIF3TX1", ARIZONA_AIF3TX1MIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("AIF3TX2", ARIZONA_AIF3TX2MIX_INPUT_1_SOURCE), }; -static int wm5102_spk_ev(struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, - int event) -{ - struct snd_soc_codec *codec = w->codec; - struct arizona *arizona = dev_get_drvdata(codec->dev->parent); - struct wm5102_priv *wm5102 = snd_soc_codec_get_drvdata(codec); - - if (arizona->rev < 1) - return 0; - - switch (event) { - case SND_SOC_DAPM_PRE_PMU: - if (!wm5102->spk_ena) { - snd_soc_write(codec, 0x4f5, 0x25a); - wm5102->spk_ena_pending = true; - } - break; - case SND_SOC_DAPM_POST_PMU: - if (wm5102->spk_ena_pending) { - msleep(75); - snd_soc_write(codec, 0x4f5, 0xda); - wm5102->spk_ena_pending = false; - wm5102->spk_ena++; - } - break; - case SND_SOC_DAPM_PRE_PMD: - wm5102->spk_ena--; - if (!wm5102->spk_ena) - snd_soc_write(codec, 0x4f5, 0x25a); - break; - case SND_SOC_DAPM_POST_PMD: - if (!wm5102->spk_ena) - snd_soc_write(codec, 0x4f5, 0x0da); - break; - } - - return 0; -} - - ARIZONA_MIXER_ENUMS(EQ1, ARIZONA_EQ1MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(EQ2, ARIZONA_EQ2MIX_INPUT_1_SOURCE); ARIZONA_MIXER_ENUMS(EQ3, ARIZONA_EQ3MIX_INPUT_1_SOURCE); @@ -1141,12 +1097,6 @@ SND_SOC_DAPM_PGA_E("OUT2R", ARIZONA_OUTPUT_ENABLES_1, SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1, ARIZONA_OUT3L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), -SND_SOC_DAPM_PGA_E("OUT4L", ARIZONA_OUTPUT_ENABLES_1, - ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, wm5102_spk_ev, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), -SND_SOC_DAPM_PGA_E("OUT4R", ARIZONA_OUTPUT_ENABLES_1, - ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, wm5102_spk_ev, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1, ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), @@ -1586,6 +1536,8 @@ static int wm5102_codec_probe(struct snd_soc_codec *codec) if (ret != 0) return ret; + arizona_init_spk(codec); + snd_soc_dapm_disable_pin(&codec->dapm, "HAPTICS"); priv->core.arizona->dapm = &codec->dapm; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index a64d3b8bc3b4..b6329c8c19df 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -577,12 +577,6 @@ SND_SOC_DAPM_PGA_E("OUT3L", ARIZONA_OUTPUT_ENABLES_1, SND_SOC_DAPM_PGA_E("OUT3R", ARIZONA_OUTPUT_ENABLES_1, ARIZONA_OUT3R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), -SND_SOC_DAPM_PGA_E("OUT4L", ARIZONA_OUTPUT_ENABLES_1, - ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), -SND_SOC_DAPM_PGA_E("OUT4R", ARIZONA_OUTPUT_ENABLES_1, - ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, - SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("OUT5L", ARIZONA_OUTPUT_ENABLES_1, ARIZONA_OUT5L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), -- cgit v1.2.3 From 899817e27a58038546b53bc42eeaa4aae5a886cb Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 13 Mar 2013 12:32:10 +0000 Subject: ASoC: arizona: Log thermal events Help with debuggability. Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 54 ++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 54 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 009810b8c667..895ddf007de2 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -122,6 +122,42 @@ static int arizona_spk_ev(struct snd_soc_dapm_widget *w, return 0; } +static irqreturn_t arizona_thermal_warn(int irq, void *data) +{ + struct arizona *arizona = data; + unsigned int val; + int ret; + + ret = regmap_read(arizona->regmap, ARIZONA_INTERRUPT_RAW_STATUS_3, + &val); + if (ret != 0) { + dev_err(arizona->dev, "Failed to read thermal status: %d\n", + ret); + } else if (val & ARIZONA_SPK_SHUTDOWN_WARN_STS) { + dev_crit(arizona->dev, "Thermal warning\n"); + } + + return IRQ_HANDLED; +} + +static irqreturn_t arizona_thermal_shutdown(int irq, void *data) +{ + struct arizona *arizona = data; + unsigned int val; + int ret; + + ret = regmap_read(arizona->regmap, ARIZONA_INTERRUPT_RAW_STATUS_3, + &val); + if (ret != 0) { + dev_err(arizona->dev, "Failed to read thermal status: %d\n", + ret); + } else if (val & ARIZONA_SPK_SHUTDOWN_STS) { + dev_crit(arizona->dev, "Thermal shutdown\n"); + } + + return IRQ_HANDLED; +} + static const struct snd_soc_dapm_widget arizona_spkl = SND_SOC_DAPM_PGA_E("OUT4L", ARIZONA_OUTPUT_ENABLES_1, ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, arizona_spk_ev, @@ -134,6 +170,8 @@ static const struct snd_soc_dapm_widget arizona_spkr = int arizona_init_spk(struct snd_soc_codec *codec) { + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; int ret; ret = snd_soc_dapm_new_controls(&codec->dapm, &arizona_spkl, 1); @@ -144,6 +182,22 @@ int arizona_init_spk(struct snd_soc_codec *codec) if (ret != 0) return ret; + ret = arizona_request_irq(arizona, ARIZONA_IRQ_SPK_SHUTDOWN_WARN, + "Thermal warning", arizona_thermal_warn, + arizona); + if (ret != 0) + dev_err(arizona->dev, + "Failed to get thermal warning IRQ: %d\n", + ret); + + ret = arizona_request_irq(arizona, ARIZONA_IRQ_SPK_SHUTDOWN, + "Thermal shutdown", arizona_thermal_shutdown, + arizona); + if (ret != 0) + dev_err(arizona->dev, + "Failed to get thermal shutdown IRQ: %d\n", + ret); + return 0; } EXPORT_SYMBOL_GPL(arizona_init_spk); -- cgit v1.2.3 From f4a76e7cc6d1c402e990e2111fb94afb305fb974 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 13 Mar 2013 12:22:39 +0000 Subject: ASoC: arizona: Suppress speaker enable if thermal shutdown is flagged Ensure that the device state does not diverge from the state we have set in the register map in order to make the behaviour clearer. Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 26 ++++++++++++++++++++++++-- 1 file changed, 24 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 895ddf007de2..6c773804ffe0 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -75,6 +75,7 @@ static int arizona_spk_ev(struct snd_soc_dapm_widget *w, struct arizona *arizona = dev_get_drvdata(codec->dev->parent); struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); bool manual_ena = false; + int val; switch (arizona->type) { case WM5102: @@ -97,6 +98,16 @@ static int arizona_spk_ev(struct snd_soc_dapm_widget *w, } break; case SND_SOC_DAPM_POST_PMU: + val = snd_soc_read(codec, ARIZONA_INTERRUPT_RAW_STATUS_3); + if (val & ARIZONA_SPK_SHUTDOWN_STS) { + dev_crit(arizona->dev, + "Speaker not enabled due to temperature\n"); + return -EBUSY; + } + + snd_soc_update_bits(codec, ARIZONA_OUTPUT_ENABLES_1, + 1 << w->shift, 1 << w->shift); + if (priv->spk_ena_pending) { msleep(75); snd_soc_write(codec, 0x4f5, 0xda); @@ -110,6 +121,9 @@ static int arizona_spk_ev(struct snd_soc_dapm_widget *w, if (!priv->spk_ena) snd_soc_write(codec, 0x4f5, 0x25a); } + + snd_soc_update_bits(codec, ARIZONA_OUTPUT_ENABLES_1, + 1 << w->shift, 0); break; case SND_SOC_DAPM_POST_PMD: if (manual_ena) { @@ -153,18 +167,26 @@ static irqreturn_t arizona_thermal_shutdown(int irq, void *data) ret); } else if (val & ARIZONA_SPK_SHUTDOWN_STS) { dev_crit(arizona->dev, "Thermal shutdown\n"); + ret = regmap_update_bits(arizona->regmap, + ARIZONA_OUTPUT_ENABLES_1, + ARIZONA_OUT4L_ENA | + ARIZONA_OUT4R_ENA, 0); + if (ret != 0) + dev_crit(arizona->dev, + "Failed to disable speaker outputs: %d\n", + ret); } return IRQ_HANDLED; } static const struct snd_soc_dapm_widget arizona_spkl = - SND_SOC_DAPM_PGA_E("OUT4L", ARIZONA_OUTPUT_ENABLES_1, + SND_SOC_DAPM_PGA_E("OUT4L", SND_SOC_NOPM, ARIZONA_OUT4L_ENA_SHIFT, 0, NULL, 0, arizona_spk_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU); static const struct snd_soc_dapm_widget arizona_spkr = - SND_SOC_DAPM_PGA_E("OUT4R", ARIZONA_OUTPUT_ENABLES_1, + SND_SOC_DAPM_PGA_E("OUT4R", SND_SOC_NOPM, ARIZONA_OUT4R_ENA_SHIFT, 0, NULL, 0, arizona_spk_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU); -- cgit v1.2.3 From dc91428a6152b2c8428a39a27ab9b5e429848f55 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 18 Feb 2013 19:09:23 +0000 Subject: ASoC: arizona: Basic support for ISRC rate selection Since ASoC does not yet really have the framework features needed to support propagating sample rates through the device well yet implement basic support for the ISRCs equivalent to that we currently have for the ASRCs. The user can opt for 8kHz or 16kHz as the rate for the DSP blocks in addition to the main audio rate, these being the primary use cases. Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 27 +++++++++++++++++++++++++++ sound/soc/codecs/arizona.h | 6 ++++++ sound/soc/codecs/wm5102.c | 11 ++++++++++- sound/soc/codecs/wm_adsp.c | 24 ++++++++++++++++++++++++ 4 files changed, 67 insertions(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 6c773804ffe0..26e1579c36cb 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -433,6 +433,33 @@ EXPORT_SYMBOL_GPL(arizona_mixer_values); const DECLARE_TLV_DB_SCALE(arizona_mixer_tlv, -3200, 100, 0); EXPORT_SYMBOL_GPL(arizona_mixer_tlv); +const char *arizona_rate_text[ARIZONA_RATE_ENUM_SIZE] = { + "SYNCCLK rate", "8kHz", "16kHz", "ASYNCCLK rate", +}; +EXPORT_SYMBOL_GPL(arizona_rate_text); + +const int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE] = { + 0, 1, 2, 8, +}; +EXPORT_SYMBOL_GPL(arizona_rate_val); + + +const struct soc_enum arizona_isrc_fsl[] = { + SOC_VALUE_ENUM_SINGLE(ARIZONA_ISRC_1_CTRL_2, + ARIZONA_ISRC1_FSL_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_ISRC_2_CTRL_2, + ARIZONA_ISRC2_FSL_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_ISRC_3_CTRL_2, + ARIZONA_ISRC3_FSL_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), +}; +EXPORT_SYMBOL_GPL(arizona_isrc_fsl); + static const char *arizona_vol_ramp_text[] = { "0ms/6dB", "0.5ms/6dB", "1ms/6dB", "2ms/6dB", "4ms/6dB", "8ms/6dB", "15ms/6dB", "30ms/6dB", diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 9399940f700d..a754a1c0217f 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -180,6 +180,12 @@ extern int arizona_mixer_values[ARIZONA_NUM_MIXER_INPUTS]; ARIZONA_MIXER_ROUTES(name, name "L"), \ ARIZONA_MIXER_ROUTES(name, name "R") +#define ARIZONA_RATE_ENUM_SIZE 4 +extern const char *arizona_rate_text[ARIZONA_RATE_ENUM_SIZE]; +extern const int arizona_rate_val[ARIZONA_RATE_ENUM_SIZE]; + +extern const struct soc_enum arizona_isrc_fsl[]; + extern const struct soc_enum arizona_in_vi_ramp; extern const struct soc_enum arizona_in_vd_ramp; diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 97757bc5fd0e..a0084b1febdb 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -731,6 +731,9 @@ SOC_ENUM("LHPF2 Mode", arizona_lhpf2_mode), SOC_ENUM("LHPF3 Mode", arizona_lhpf3_mode), SOC_ENUM("LHPF4 Mode", arizona_lhpf4_mode), +SOC_ENUM("ISRC1 FSL", arizona_isrc_fsl[0]), +SOC_ENUM("ISRC2 FSL", arizona_isrc_fsl[1]), + ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE), @@ -1532,7 +1535,7 @@ static int wm5102_codec_probe(struct snd_soc_codec *codec) if (ret != 0) return ret; - ret = snd_soc_add_codec_controls(codec, wm_adsp_fw_controls, 1); + ret = snd_soc_add_codec_controls(codec, wm_adsp_fw_controls, 2); if (ret != 0) return ret; @@ -1624,6 +1627,12 @@ static int wm5102_probe(struct platform_device *pdev) ARIZONA_IRQ_FLL2_LOCK, ARIZONA_IRQ_FLL2_CLOCK_OK, &wm5102->fll[1]); + /* SR2 fixed at 8kHz, SR3 fixed at 16kHz */ + regmap_update_bits(arizona->regmap, ARIZONA_SAMPLE_RATE_2, + ARIZONA_SAMPLE_RATE_2_MASK, 0x11); + regmap_update_bits(arizona->regmap, ARIZONA_SAMPLE_RATE_3, + ARIZONA_SAMPLE_RATE_3_MASK, 0x12); + for (i = 0; i < ARRAY_SIZE(wm5102_dai); i++) arizona_init_dai(&wm5102->core, i); diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index f3f7e75f8628..3a481fd1bf9a 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -31,6 +31,7 @@ #include +#include "arizona.h" #include "wm_adsp.h" #define adsp_crit(_dsp, fmt, ...) \ @@ -246,15 +247,38 @@ static const struct soc_enum wm_adsp_fw_enum[] = { SOC_ENUM_SINGLE(0, 3, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text), }; +static const struct soc_enum wm_adsp_rate_enum[] = { + SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP1_CONTROL_1, + ARIZONA_DSP1_RATE_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP2_CONTROL_1, + ARIZONA_DSP1_RATE_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP3_CONTROL_1, + ARIZONA_DSP1_RATE_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP3_CONTROL_1, + ARIZONA_DSP1_RATE_SHIFT, 0xf, + ARIZONA_RATE_ENUM_SIZE, + arizona_rate_text, arizona_rate_val), +}; + const struct snd_kcontrol_new wm_adsp_fw_controls[] = { SOC_ENUM_EXT("DSP1 Firmware", wm_adsp_fw_enum[0], wm_adsp_fw_get, wm_adsp_fw_put), + SOC_ENUM("DSP1 Rate", wm_adsp_rate_enum[0]), SOC_ENUM_EXT("DSP2 Firmware", wm_adsp_fw_enum[1], wm_adsp_fw_get, wm_adsp_fw_put), + SOC_ENUM("DSP2 Rate", wm_adsp_rate_enum[1]), SOC_ENUM_EXT("DSP3 Firmware", wm_adsp_fw_enum[2], wm_adsp_fw_get, wm_adsp_fw_put), + SOC_ENUM("DSP3 Rate", wm_adsp_rate_enum[2]), SOC_ENUM_EXT("DSP4 Firmware", wm_adsp_fw_enum[3], wm_adsp_fw_get, wm_adsp_fw_put), + SOC_ENUM("DSP4 Rate", wm_adsp_rate_enum[3]), }; EXPORT_SYMBOL_GPL(wm_adsp_fw_controls); -- cgit v1.2.3 From d3725761ee3d4813c6071ea1d952de1094d8b68f Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 29 Jan 2013 23:17:12 +0800 Subject: ASoC: wm8994: Restore AIFnCLK after reducing it for low clock rates This helps to ensure a smooth startup when we restore. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 32 ++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8994.h | 1 + 2 files changed, 33 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 318ea64b9800..1c02a47910e4 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2268,10 +2268,26 @@ out: */ if (max(wm8994->aifclk[0], wm8994->aifclk[1]) < 50000) { dev_dbg(codec->dev, "Configuring AIFs for 128fs\n"); + + wm8994->aifdiv[0] = snd_soc_read(codec, WM8994_AIF1_RATE) + & WM8994_AIF1CLK_RATE_MASK; + wm8994->aifdiv[1] = snd_soc_read(codec, WM8994_AIF2_RATE) + & WM8994_AIF1CLK_RATE_MASK; + snd_soc_update_bits(codec, WM8994_AIF1_RATE, WM8994_AIF1CLK_RATE_MASK, 0x1); snd_soc_update_bits(codec, WM8994_AIF2_RATE, WM8994_AIF2CLK_RATE_MASK, 0x1); + } else if (wm8994->aifdiv[0]) { + snd_soc_update_bits(codec, WM8994_AIF1_RATE, + WM8994_AIF1CLK_RATE_MASK, + wm8994->aifdiv[0]); + snd_soc_update_bits(codec, WM8994_AIF2_RATE, + WM8994_AIF2CLK_RATE_MASK, + wm8994->aifdiv[1]); + + wm8994->aifdiv[0] = 0; + wm8994->aifdiv[1] = 0; } return 0; @@ -2368,10 +2384,26 @@ static int wm8994_set_dai_sysclk(struct snd_soc_dai *dai, */ if (max(wm8994->aifclk[0], wm8994->aifclk[1]) < 50000) { dev_dbg(codec->dev, "Configuring AIFs for 128fs\n"); + + wm8994->aifdiv[0] = snd_soc_read(codec, WM8994_AIF1_RATE) + & WM8994_AIF1CLK_RATE_MASK; + wm8994->aifdiv[1] = snd_soc_read(codec, WM8994_AIF2_RATE) + & WM8994_AIF1CLK_RATE_MASK; + snd_soc_update_bits(codec, WM8994_AIF1_RATE, WM8994_AIF1CLK_RATE_MASK, 0x1); snd_soc_update_bits(codec, WM8994_AIF2_RATE, WM8994_AIF2CLK_RATE_MASK, 0x1); + } else if (wm8994->aifdiv[0]) { + snd_soc_update_bits(codec, WM8994_AIF1_RATE, + WM8994_AIF1CLK_RATE_MASK, + wm8994->aifdiv[0]); + snd_soc_update_bits(codec, WM8994_AIF2_RATE, + WM8994_AIF2CLK_RATE_MASK, + wm8994->aifdiv[1]); + + wm8994->aifdiv[0] = 0; + wm8994->aifdiv[1] = 0; } return 0; diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index 45f192702024..928e2c256450 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -79,6 +79,7 @@ struct wm8994_priv { int sysclk_rate[2]; int mclk[2]; int aifclk[2]; + int aifdiv[2]; int channels[2]; struct wm8994_fll_config fll[2], fll_suspend[2]; struct completion fll_locked[2]; -- cgit v1.2.3 From aed9913e6fad5a7eccce2b7a3ee6daa96b575157 Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Tue, 26 Mar 2013 14:47:08 +0800 Subject: ASoC: arizona: remove duplicated include from arizona.c Remove duplicated include. Signed-off-by: Wei Yongjun Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 1 - 1 file changed, 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 26e1579c36cb..c979ff2b4191 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -14,7 +14,6 @@ #include #include #include -#include #include #include #include -- cgit v1.2.3 From f607e31ce3963327f749b56c65dfec2642aa623c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 22 Feb 2013 18:36:53 +0000 Subject: ASoC: arizona: Fix interaction between headphone outputs and identification Running HPDET while the headphone outputs are enabled can disrupt the operation of HPDET. In order to avoid this HPDET needs to disable the headphone outputs and ASoC needs to not enable them while HPDET is running. Do the ASoC side of this by storing the enable state in the core driver structure and only writing to the device if a flag indicating that the accessory detection side is in a state where it can have the headphone output stage enabled. Signed-off-by: Mark Brown --- include/linux/mfd/arizona/core.h | 3 +++ sound/soc/codecs/arizona.c | 33 +++++++++++++++++++++++++++++++++ sound/soc/codecs/arizona.h | 3 +++ sound/soc/codecs/wm5102.c | 8 ++++---- sound/soc/codecs/wm5110.c | 8 ++++---- 5 files changed, 47 insertions(+), 8 deletions(-) (limited to 'sound/soc/codecs') diff --git a/include/linux/mfd/arizona/core.h b/include/linux/mfd/arizona/core.h index a710255528d7..cc281368dc55 100644 --- a/include/linux/mfd/arizona/core.h +++ b/include/linux/mfd/arizona/core.h @@ -100,6 +100,9 @@ struct arizona { struct regmap_irq_chip_data *aod_irq_chip; struct regmap_irq_chip_data *irq_chip; + bool hpdet_magic; + unsigned int hp_ena; + struct mutex clk_lock; int clk32k_ref; diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index ac948a671ea6..e7d34711412c 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -364,6 +364,39 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w, } EXPORT_SYMBOL_GPL(arizona_out_ev); +int arizona_hp_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event) +{ + struct arizona_priv *priv = snd_soc_codec_get_drvdata(w->codec); + unsigned int mask = 1 << w->shift; + unsigned int val; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + val = mask; + break; + case SND_SOC_DAPM_PRE_PMD: + val = 0; + break; + default: + return -EINVAL; + } + + /* Store the desired state for the HP outputs */ + priv->arizona->hp_ena &= ~mask; + priv->arizona->hp_ena |= val; + + /* Force off if HPDET magic is active */ + if (priv->arizona->hpdet_magic) + val = 0; + + snd_soc_update_bits(w->codec, ARIZONA_OUTPUT_ENABLES_1, mask, val); + + return arizona_out_ev(w, kcontrol, event); +} +EXPORT_SYMBOL_GPL(arizona_hp_ev); + static unsigned int arizona_sysclk_48k_rates[] = { 6144000, 12288000, diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 116372c91f5d..13dd2916b721 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -184,6 +184,9 @@ extern int arizona_in_ev(struct snd_soc_dapm_widget *w, extern int arizona_out_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); +extern int arizona_hp_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, + int event); extern int arizona_set_sysclk(struct snd_soc_codec *codec, int clk_id, int source, unsigned int freq, int dir); diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index b82bbf584146..2657aad3f8b1 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1131,11 +1131,11 @@ ARIZONA_DSP_WIDGETS(DSP1, "DSP1"), SND_SOC_DAPM_VALUE_MUX("AEC Loopback", ARIZONA_DAC_AEC_CONTROL_1, ARIZONA_AEC_LOOPBACK_ENA, 0, &wm5102_aec_loopback_mux), -SND_SOC_DAPM_PGA_E("OUT1L", ARIZONA_OUTPUT_ENABLES_1, - ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, +SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM, + ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), -SND_SOC_DAPM_PGA_E("OUT1R", ARIZONA_OUTPUT_ENABLES_1, - ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, +SND_SOC_DAPM_PGA_E("OUT1R", SND_SOC_NOPM, + ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("OUT2L", ARIZONA_OUTPUT_ENABLES_1, ARIZONA_OUT2L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index cdeb301da1f6..7841b42a819c 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -551,11 +551,11 @@ SND_SOC_DAPM_AIF_IN("AIF3RX1", NULL, 0, SND_SOC_DAPM_AIF_IN("AIF3RX2", NULL, 0, ARIZONA_AIF3_RX_ENABLES, ARIZONA_AIF3RX2_ENA_SHIFT, 0), -SND_SOC_DAPM_PGA_E("OUT1L", ARIZONA_OUTPUT_ENABLES_1, - ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, +SND_SOC_DAPM_PGA_E("OUT1L", SND_SOC_NOPM, + ARIZONA_OUT1L_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), -SND_SOC_DAPM_PGA_E("OUT1R", ARIZONA_OUTPUT_ENABLES_1, - ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, +SND_SOC_DAPM_PGA_E("OUT1R", SND_SOC_NOPM, + ARIZONA_OUT1R_ENA_SHIFT, 0, NULL, 0, arizona_hp_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_PGA_E("OUT2L", ARIZONA_OUTPUT_ENABLES_1, ARIZONA_OUT2L_ENA_SHIFT, 0, NULL, 0, arizona_out_ev, -- cgit v1.2.3 From 1a2c7d568f624307c5821f31e54727a4b374855c Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sun, 24 Mar 2013 22:50:23 +0000 Subject: ASoC: arizona: Add delay after powering up line level outputs Ensure that the outputs are fully enabled before we begin passing audio through them. Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 18 ++++++++++++++++++ 1 file changed, 18 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index abdd019c5b6e..389f23253831 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -579,6 +579,24 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { + switch (event) { + case SND_SOC_DAPM_POST_PMU: + switch (w->shift) { + case ARIZONA_OUT1L_ENA_SHIFT: + case ARIZONA_OUT1R_ENA_SHIFT: + case ARIZONA_OUT2L_ENA_SHIFT: + case ARIZONA_OUT2R_ENA_SHIFT: + case ARIZONA_OUT3L_ENA_SHIFT: + case ARIZONA_OUT3R_ENA_SHIFT: + msleep(17); + break; + + default: + break; + } + break; + } + return 0; } EXPORT_SYMBOL_GPL(arizona_out_ev); -- cgit v1.2.3 From 658e6101d045ae0bc97d31f5d6a5ea117a86c92a Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Mon, 25 Mar 2013 15:50:22 +0000 Subject: ASoC: wm5102: Implement OSR support Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 26 ++++++++++++++++++++++++++ 1 file changed, 26 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index b7a3fdceec7f..a1ff43c8b479 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -612,6 +612,26 @@ static int wm5102_sysclk_ev(struct snd_soc_dapm_widget *w, return 0; } +static const char *wm5102_osr_text[] = { + "Low power", "Normal", "High performance", +}; + +static const unsigned int wm5102_osr_val[] = { + 0x0, 0x3, 0x5, +}; + +static const struct soc_enum wm5102_hpout_osr[] = { + SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_1L, + ARIZONA_OUT1_OSR_SHIFT, 0x7, 3, + wm5102_osr_text, wm5102_osr_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_2L, + ARIZONA_OUT2_OSR_SHIFT, 0x7, 3, + wm5102_osr_text, wm5102_osr_val), + SOC_VALUE_ENUM_SINGLE(ARIZONA_OUTPUT_PATH_CONFIG_3L, + ARIZONA_OUT3_OSR_SHIFT, 0x7, 3, + wm5102_osr_text, wm5102_osr_val), +}; + #define WM5102_NG_SRC(name, base) \ SOC_SINGLE(name " NG HPOUT1L Switch", base, 0, 1, 0), \ SOC_SINGLE(name " NG HPOUT1R Switch", base, 1, 1, 0), \ @@ -761,6 +781,8 @@ ARIZONA_MIXER_CONTROLS("SPKOUTR", ARIZONA_OUT4RMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SPKDAT1L", ARIZONA_OUT5LMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("SPKDAT1R", ARIZONA_OUT5RMIX_INPUT_1_SOURCE), +SOC_SINGLE("Speaker High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_4L, + ARIZONA_OUT4_OSR_SHIFT, 1, 0), SOC_SINGLE("SPKDAT1 High Performance Switch", ARIZONA_OUTPUT_PATH_CONFIG_5L, ARIZONA_OUT5_OSR_SHIFT, 1, 0), @@ -790,6 +812,10 @@ SOC_DOUBLE_R_TLV("SPKDAT1 Digital Volume", ARIZONA_DAC_DIGITAL_VOLUME_5L, ARIZONA_DAC_DIGITAL_VOLUME_5R, ARIZONA_OUT5L_VOL_SHIFT, 0xbf, 0, digital_tlv), +SOC_VALUE_ENUM("HPOUT1 OSR", wm5102_hpout_osr[0]), +SOC_VALUE_ENUM("HPOUT2 OSR", wm5102_hpout_osr[1]), +SOC_VALUE_ENUM("HPOUT3 OSR", wm5102_hpout_osr[2]), + SOC_ENUM("Output Ramp Up", arizona_out_vi_ramp), SOC_ENUM("Output Ramp Down", arizona_out_vd_ramp), -- cgit v1.2.3 From a96f5e9394d298689eb3b876e6619166f1a37cc4 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Wed, 27 Mar 2013 10:50:53 +0000 Subject: ASoC: wm5102: Correctly use SOC_VALUE_ENUM for ISRC FSL controls Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm5102.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index a1ff43c8b479..d1b43ebec087 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -762,8 +762,8 @@ SOC_ENUM("LHPF2 Mode", arizona_lhpf2_mode), SOC_ENUM("LHPF3 Mode", arizona_lhpf3_mode), SOC_ENUM("LHPF4 Mode", arizona_lhpf4_mode), -SOC_ENUM("ISRC1 FSL", arizona_isrc_fsl[0]), -SOC_ENUM("ISRC2 FSL", arizona_isrc_fsl[1]), +SOC_VALUE_ENUM("ISRC1 FSL", arizona_isrc_fsl[0]), +SOC_VALUE_ENUM("ISRC2 FSL", arizona_isrc_fsl[1]), ARIZONA_MIXER_CONTROLS("Mic", ARIZONA_MICMIX_INPUT_1_SOURCE), ARIZONA_MIXER_CONTROLS("Noise", ARIZONA_NOISEMIX_INPUT_1_SOURCE), -- cgit v1.2.3 From 961b0fc840bf70511ef87d2f799eab014b4d2d37 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 29 Mar 2013 09:45:34 +0000 Subject: ASoC: wm0010: Constify usage of firmware filenames Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm0010.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index ad2fee4bb4cd..55fdf0f4406c 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -342,7 +342,7 @@ static void byte_swap_64(u64 *data_in, u64 *data_out, u32 len) data_out[i] = cpu_to_be64(le64_to_cpu(data_in[i])); } -static int wm0010_firmware_load(char *name, struct snd_soc_codec *codec) +static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) { struct spi_device *spi = to_spi_device(codec->dev); struct wm0010_priv *wm0010 = snd_soc_codec_get_drvdata(codec); -- cgit v1.2.3 From 3e112af51eedda46fe87d2cba427d48c4b7695fd Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Fri, 29 Mar 2013 09:45:35 +0000 Subject: ASoC: wm0010: Report filename when we fail to load firmware Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm0010.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 55fdf0f4406c..8df2b6e1a1a6 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -361,8 +361,8 @@ static int wm0010_firmware_load(const char *name, struct snd_soc_codec *codec) ret = request_firmware(&fw, name, codec->dev); if (ret != 0) { - dev_err(codec->dev, "Failed to request application: %d\n", - ret); + dev_err(codec->dev, "Failed to request application(%s): %d\n", + name, ret); return ret; } -- cgit v1.2.3 From 939dc51bddc245df51c1e8ee44bf136621475149 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 29 Mar 2013 13:03:39 +0800 Subject: ASoC: wm2000: Expose some more registers for diagnostics Signed-off-by: Mark Brown --- sound/soc/codecs/wm2000.c | 4 +++- sound/soc/codecs/wm2000.h | 2 ++ 2 files changed, 5 insertions(+), 1 deletion(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index f2ac38b61a1b..7fefd766b582 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -761,6 +761,8 @@ static bool wm2000_readable_reg(struct device *dev, unsigned int reg) case WM2000_REG_SYS_CTL2: case WM2000_REG_ANC_STAT: case WM2000_REG_IF_CTL: + case WM2000_REG_ANA_MIC_CTL: + case WM2000_REG_SPK_CTL: return true; default: return false; @@ -771,7 +773,7 @@ static const struct regmap_config wm2000_regmap = { .reg_bits = 16, .val_bits = 8, - .max_register = WM2000_REG_IF_CTL, + .max_register = WM2000_REG_SPK_CTL, .readable_reg = wm2000_readable_reg, }; diff --git a/sound/soc/codecs/wm2000.h b/sound/soc/codecs/wm2000.h index fb812cd9e77d..3870c0e1d246 100644 --- a/sound/soc/codecs/wm2000.h +++ b/sound/soc/codecs/wm2000.h @@ -30,6 +30,8 @@ #define WM2000_REG_SYS_CTL2 0xf004 #define WM2000_REG_ANC_STAT 0xf005 #define WM2000_REG_IF_CTL 0xf006 +#define WM2000_REG_ANA_MIC_CTL 0xf028 +#define WM2000_REG_SPK_CTL 0xf034 /* SPEECH_CLARITY */ #define WM2000_SPEECH_CLARITY 0x01 -- cgit v1.2.3 From dd84f9259bfe8454ee7c9e6faf6ac13f45bb1ed2 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 8 Mar 2013 15:25:58 +0800 Subject: ASoC: wm_adsp: Provide defines for firmwares For future work to have specific handling for some firmwares. Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 18 +++++++++++++----- 1 file changed, 13 insertions(+), 5 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index febb4c76535e..68eda929fbde 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -193,17 +193,25 @@ static void wm_adsp_buf_free(struct list_head *list) #define WM_ADSP_NUM_FW 4 +#define WM_ADSP_FW_MBC_VSS 0 +#define WM_ADSP_FW_TX 1 +#define WM_ADSP_FW_TX_SPK 2 +#define WM_ADSP_FW_RX_ANC 3 + static const char *wm_adsp_fw_text[WM_ADSP_NUM_FW] = { - "MBC/VSS", "Tx", "Tx Speaker", "Rx ANC" + [WM_ADSP_FW_MBC_VSS] = "MBC/VSS", + [WM_ADSP_FW_TX] = "Tx", + [WM_ADSP_FW_TX_SPK] = "Tx Speaker", + [WM_ADSP_FW_RX_ANC] = "Rx ANC", }; static struct { const char *file; } wm_adsp_fw[WM_ADSP_NUM_FW] = { - { .file = "mbc-vss" }, - { .file = "tx" }, - { .file = "tx-spk" }, - { .file = "rx-anc" }, + [WM_ADSP_FW_MBC_VSS] = { .file = "mbc-vss" }, + [WM_ADSP_FW_TX] = { .file = "tx" }, + [WM_ADSP_FW_TX_SPK] = { .file = "tx-spk" }, + [WM_ADSP_FW_RX_ANC] = { .file = "rx-anc" }, }; static int wm_adsp_fw_get(struct snd_kcontrol *kcontrol, -- cgit v1.2.3 From b6ed61cfa24786e36164869b593d44d411a700ad Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 29 Mar 2013 18:00:24 +0000 Subject: ASoC: wm_adsp: Split ADSP1 and ADSP2 firmware controls Now that we have regular register mapped controls we should be splitting the control sets for ADSP1 and ADSP2 as the register maps are not identical. Do that. Signed-off-by: Mark Brown --- sound/soc/codecs/wm2200.c | 2 +- sound/soc/codecs/wm5102.c | 2 +- sound/soc/codecs/wm_adsp.c | 26 +++++++++++++++++++------- sound/soc/codecs/wm_adsp.h | 3 ++- 4 files changed, 23 insertions(+), 10 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm2200.c b/sound/soc/codecs/wm2200.c index ddc98f02ecbd..57ba315d0c84 100644 --- a/sound/soc/codecs/wm2200.c +++ b/sound/soc/codecs/wm2200.c @@ -1565,7 +1565,7 @@ static int wm2200_probe(struct snd_soc_codec *codec) return ret; } - ret = snd_soc_add_codec_controls(codec, wm_adsp_fw_controls, 2); + ret = snd_soc_add_codec_controls(codec, wm_adsp1_fw_controls, 2); if (ret != 0) return ret; diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index d1b43ebec087..cb03cc448da6 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1572,7 +1572,7 @@ static int wm5102_codec_probe(struct snd_soc_codec *codec) if (ret != 0) return ret; - ret = snd_soc_add_codec_controls(codec, wm_adsp_fw_controls, 2); + ret = snd_soc_add_codec_controls(codec, wm_adsp2_fw_controls, 2); if (ret != 0) return ret; diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 3a481fd1bf9a..bc03baef39fa 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -247,7 +247,18 @@ static const struct soc_enum wm_adsp_fw_enum[] = { SOC_ENUM_SINGLE(0, 3, ARRAY_SIZE(wm_adsp_fw_text), wm_adsp_fw_text), }; -static const struct soc_enum wm_adsp_rate_enum[] = { +const struct snd_kcontrol_new wm_adsp1_fw_controls[] = { + SOC_ENUM_EXT("DSP1 Firmware", wm_adsp_fw_enum[0], + wm_adsp_fw_get, wm_adsp_fw_put), + SOC_ENUM_EXT("DSP2 Firmware", wm_adsp_fw_enum[1], + wm_adsp_fw_get, wm_adsp_fw_put), + SOC_ENUM_EXT("DSP3 Firmware", wm_adsp_fw_enum[2], + wm_adsp_fw_get, wm_adsp_fw_put), +}; +EXPORT_SYMBOL_GPL(wm_adsp1_fw_controls); + +#if IS_ENABLED(CONFIG_SND_SOC_ARIZONA) +static const struct soc_enum wm_adsp2_rate_enum[] = { SOC_VALUE_ENUM_SINGLE(ARIZONA_DSP1_CONTROL_1, ARIZONA_DSP1_RATE_SHIFT, 0xf, ARIZONA_RATE_ENUM_SIZE, @@ -266,21 +277,22 @@ static const struct soc_enum wm_adsp_rate_enum[] = { arizona_rate_text, arizona_rate_val), }; -const struct snd_kcontrol_new wm_adsp_fw_controls[] = { +const struct snd_kcontrol_new wm_adsp2_fw_controls[] = { SOC_ENUM_EXT("DSP1 Firmware", wm_adsp_fw_enum[0], wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM("DSP1 Rate", wm_adsp_rate_enum[0]), + SOC_ENUM("DSP1 Rate", wm_adsp2_rate_enum[0]), SOC_ENUM_EXT("DSP2 Firmware", wm_adsp_fw_enum[1], wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM("DSP2 Rate", wm_adsp_rate_enum[1]), + SOC_ENUM("DSP2 Rate", wm_adsp2_rate_enum[1]), SOC_ENUM_EXT("DSP3 Firmware", wm_adsp_fw_enum[2], wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM("DSP3 Rate", wm_adsp_rate_enum[2]), + SOC_ENUM("DSP3 Rate", wm_adsp2_rate_enum[2]), SOC_ENUM_EXT("DSP4 Firmware", wm_adsp_fw_enum[3], wm_adsp_fw_get, wm_adsp_fw_put), - SOC_ENUM("DSP4 Rate", wm_adsp_rate_enum[3]), + SOC_ENUM("DSP4 Rate", wm_adsp2_rate_enum[3]), }; -EXPORT_SYMBOL_GPL(wm_adsp_fw_controls); +EXPORT_SYMBOL_GPL(wm_adsp2_fw_controls); +#endif static struct wm_adsp_region const *wm_adsp_find_region(struct wm_adsp *dsp, int type) diff --git a/sound/soc/codecs/wm_adsp.h b/sound/soc/codecs/wm_adsp.h index cb8871a3ec00..9f90c9fea842 100644 --- a/sound/soc/codecs/wm_adsp.h +++ b/sound/soc/codecs/wm_adsp.h @@ -65,7 +65,8 @@ struct wm_adsp { .shift = num, .event = wm_adsp2_event, \ .event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD } -extern const struct snd_kcontrol_new wm_adsp_fw_controls[]; +extern const struct snd_kcontrol_new wm_adsp1_fw_controls[]; +extern const struct snd_kcontrol_new wm_adsp2_fw_controls[]; int wm_adsp1_init(struct wm_adsp *adsp); int wm_adsp2_init(struct wm_adsp *adsp, bool dvfs); -- cgit v1.2.3 From da445afe357ae656f6baddd8fd58b01e923f1fc6 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 12 Mar 2013 17:46:09 +0000 Subject: ASoC: wm8994: Remove duplicate revision cache There's already a device revision stored in the core data structure, don't duplicate it in the CODEC driver. Signed-off-by: Mark Brown --- sound/soc/codecs/wm8994.c | 23 +++++++++++------------ sound/soc/codecs/wm8994.h | 2 -- 2 files changed, 11 insertions(+), 14 deletions(-) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index 1c02a47910e4..14094f558e03 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2209,7 +2209,7 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, vmid_reference(codec); break; case WM8958: - if (wm8994->revision < 1) + if (control->revision < 1) vmid_reference(codec); break; default: @@ -2244,7 +2244,7 @@ static int _wm8994_set_fll(struct snd_soc_codec *codec, int id, int src, vmid_dereference(codec); break; case WM8958: - if (wm8994->revision < 1) + if (control->revision < 1) vmid_dereference(codec); break; default: @@ -2443,7 +2443,7 @@ static int wm8994_set_bias_level(struct snd_soc_codec *codec, if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { switch (control->type) { case WM8958: - if (wm8994->revision == 0) { + if (control->revision == 0) { /* Optimise performance for rev A */ snd_soc_update_bits(codec, WM8958_CHARGE_PUMP_2, @@ -3094,7 +3094,7 @@ static int wm8994_codec_resume(struct snd_soc_codec *codec) int i, ret; unsigned int val, mask; - if (wm8994->revision < 4) { + if (control->revision < 4) { /* force a HW read */ ret = regmap_read(control->regmap, WM8994_POWER_MANAGEMENT_5, &val); @@ -3911,7 +3911,6 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) codec->dapm.idle_bias_off = 1; /* Set revision-specific configuration */ - wm8994->revision = snd_soc_read(codec, WM8994_CHIP_REVISION); switch (control->type) { case WM8994: /* Single ended line outputs should have VMID on. */ @@ -3919,7 +3918,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) !control->pdata.lineout2_diff) codec->dapm.idle_bias_off = 0; - switch (wm8994->revision) { + switch (control->revision) { case 2: case 3: wm8994->hubs.dcs_codes_l = -5; @@ -3938,7 +3937,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) wm8994->hubs.dcs_readback_mode = 1; wm8994->hubs.hp_startup_mode = 1; - switch (wm8994->revision) { + switch (control->revision) { case 0: break; default: @@ -4041,7 +4040,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) switch (control->type) { case WM1811: - if (control->cust_id > 1 || wm8994->revision > 1) { + if (control->cust_id > 1 || control->revision > 1) { ret = wm8994_request_irq(wm8994->wm8994, WM8994_IRQ_GPIO(6), wm1811_jackdet_irq, "JACKDET", @@ -4155,7 +4154,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) case WM8994: snd_soc_dapm_new_controls(dapm, wm8994_specific_dapm_widgets, ARRAY_SIZE(wm8994_specific_dapm_widgets)); - if (wm8994->revision < 4) { + if (control->revision < 4) { snd_soc_dapm_new_controls(dapm, wm8994_lateclk_revd_widgets, ARRAY_SIZE(wm8994_lateclk_revd_widgets)); snd_soc_dapm_new_controls(dapm, wm8994_adc_revd_widgets, @@ -4176,7 +4175,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) ARRAY_SIZE(wm8958_snd_controls)); snd_soc_dapm_new_controls(dapm, wm8958_dapm_widgets, ARRAY_SIZE(wm8958_dapm_widgets)); - if (wm8994->revision < 1) { + if (control->revision < 1) { snd_soc_dapm_new_controls(dapm, wm8994_lateclk_revd_widgets, ARRAY_SIZE(wm8994_lateclk_revd_widgets)); snd_soc_dapm_new_controls(dapm, wm8994_adc_revd_widgets, @@ -4215,7 +4214,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) snd_soc_dapm_add_routes(dapm, wm8994_intercon, ARRAY_SIZE(wm8994_intercon)); - if (wm8994->revision < 4) { + if (control->revision < 4) { snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon, ARRAY_SIZE(wm8994_revd_intercon)); snd_soc_dapm_add_routes(dapm, wm8994_lateclk_revd_intercon, @@ -4226,7 +4225,7 @@ static int wm8994_codec_probe(struct snd_soc_codec *codec) } break; case WM8958: - if (wm8994->revision < 1) { + if (control->revision < 1) { snd_soc_dapm_add_routes(dapm, wm8994_intercon, ARRAY_SIZE(wm8994_intercon)); snd_soc_dapm_add_routes(dapm, wm8994_revd_intercon, diff --git a/sound/soc/codecs/wm8994.h b/sound/soc/codecs/wm8994.h index 928e2c256450..55ddf4d57d9b 100644 --- a/sound/soc/codecs/wm8994.h +++ b/sound/soc/codecs/wm8994.h @@ -147,8 +147,6 @@ struct wm8994_priv { wm1811_mic_id_cb mic_id_cb; void *mic_id_cb_data; - int revision; - unsigned int aif1clk_enable:1; unsigned int aif2clk_enable:1; -- cgit v1.2.3 From ac50009f64c65f0ec8406a33846c1e41d3b33ff7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Tue, 9 Apr 2013 17:08:24 +0100 Subject: ASoC: wm_adsp: Add support for firmware wide coefficient blocks Firmwares may provide some firmware wide configuration regions which can be configured by the coefficient files using the firmware ID as the algorithm ID, include these in the algorithm list. Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 40 ++++++++++++++++++++++++++++++++++++++++ 1 file changed, 40 insertions(+) (limited to 'sound/soc/codecs') diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 68eda929fbde..a793c7d951ce 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -565,6 +565,22 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) be32_to_cpu(adsp1_id.fw.ver) & 0xff, algs); + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP1_ZM; + region->alg = be32_to_cpu(adsp1_id.fw.id); + region->base = be32_to_cpu(adsp1_id.zm); + list_add_tail(®ion->list, &dsp->alg_regions); + + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP1_DM; + region->alg = be32_to_cpu(adsp1_id.fw.id); + region->base = be32_to_cpu(adsp1_id.dm); + list_add_tail(®ion->list, &dsp->alg_regions); + pos = sizeof(adsp1_id) / 2; term = pos + ((sizeof(*adsp1_alg) * algs) / 2); break; @@ -590,6 +606,30 @@ static int wm_adsp_setup_algs(struct wm_adsp *dsp) be32_to_cpu(adsp2_id.fw.ver) & 0xff, algs); + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP2_XM; + region->alg = be32_to_cpu(adsp2_id.fw.id); + region->base = be32_to_cpu(adsp2_id.xm); + list_add_tail(®ion->list, &dsp->alg_regions); + + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP2_YM; + region->alg = be32_to_cpu(adsp2_id.fw.id); + region->base = be32_to_cpu(adsp2_id.ym); + list_add_tail(®ion->list, &dsp->alg_regions); + + region = kzalloc(sizeof(*region), GFP_KERNEL); + if (!region) + return -ENOMEM; + region->type = WMFW_ADSP2_ZM; + region->alg = be32_to_cpu(adsp2_id.fw.id); + region->base = be32_to_cpu(adsp2_id.zm); + list_add_tail(®ion->list, &dsp->alg_regions); + pos = sizeof(adsp2_id) / 2; term = pos + ((sizeof(*adsp2_alg) * algs) / 2); break; -- cgit v1.2.3