From c775cbf62ed4911e4f0f23880f01815753123690 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Fri, 25 Mar 2022 15:42:39 +0000 Subject: ASoC: atmel: Remove system clock tree configuration for at91sam9g20ek The MCLK of the WM8731 on the AT91SAM9G20-EK board is connected to the PCK0 output of the SoC, intended in the reference software to be supplied using PLLB and programmed to 12MHz. As originally written for use with a board file the audio driver was responsible for configuring the entire tree but in the conversion to the common clock framework the registration of the named pck0 and pllb clocks was removed so the driver has failed to instantiate ever since. Since the WM8731 driver has had support for managing a MCLK provided via the common clock framework for some time we can simply drop all the clock management code from the machine driver other than configuration of the sysclk rate, the CODEC driver still respects that configuration from the machine driver. Fixes: ff78a189b0ae55f ("ARM: at91: remove old at91-specific clock driver") Signed-off-by: Mark Brown Reviewed-by: Codrin Ciubotariu Link: https://lore.kernel.org/r/20220325154241.1600757-2-broonie@kernel.org --- sound/soc/atmel/sam9g20_wm8731.c | 61 ---------------------------------------- 1 file changed, 61 deletions(-) diff --git a/sound/soc/atmel/sam9g20_wm8731.c b/sound/soc/atmel/sam9g20_wm8731.c index 33e43013ff77..0d639a33ad96 100644 --- a/sound/soc/atmel/sam9g20_wm8731.c +++ b/sound/soc/atmel/sam9g20_wm8731.c @@ -46,35 +46,6 @@ */ #undef ENABLE_MIC_INPUT -static struct clk *mclk; - -static int at91sam9g20ek_set_bias_level(struct snd_soc_card *card, - struct snd_soc_dapm_context *dapm, - enum snd_soc_bias_level level) -{ - static int mclk_on; - int ret = 0; - - switch (level) { - case SND_SOC_BIAS_ON: - case SND_SOC_BIAS_PREPARE: - if (!mclk_on) - ret = clk_enable(mclk); - if (ret == 0) - mclk_on = 1; - break; - - case SND_SOC_BIAS_OFF: - case SND_SOC_BIAS_STANDBY: - if (mclk_on) - clk_disable(mclk); - mclk_on = 0; - break; - } - - return ret; -} - static const struct snd_soc_dapm_widget at91sam9g20ek_dapm_widgets[] = { SND_SOC_DAPM_MIC("Int Mic", NULL), SND_SOC_DAPM_SPK("Ext Spk", NULL), @@ -135,7 +106,6 @@ static struct snd_soc_card snd_soc_at91sam9g20ek = { .owner = THIS_MODULE, .dai_link = &at91sam9g20ek_dai, .num_links = 1, - .set_bias_level = at91sam9g20ek_set_bias_level, .dapm_widgets = at91sam9g20ek_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(at91sam9g20ek_dapm_widgets), @@ -148,7 +118,6 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev) { struct device_node *np = pdev->dev.of_node; struct device_node *codec_np, *cpu_np; - struct clk *pllb; struct snd_soc_card *card = &snd_soc_at91sam9g20ek; int ret; @@ -162,31 +131,6 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev) return -EINVAL; } - /* - * Codec MCLK is supplied by PCK0 - set it up. - */ - mclk = clk_get(NULL, "pck0"); - if (IS_ERR(mclk)) { - dev_err(&pdev->dev, "Failed to get MCLK\n"); - ret = PTR_ERR(mclk); - goto err; - } - - pllb = clk_get(NULL, "pllb"); - if (IS_ERR(pllb)) { - dev_err(&pdev->dev, "Failed to get PLLB\n"); - ret = PTR_ERR(pllb); - goto err_mclk; - } - ret = clk_set_parent(mclk, pllb); - clk_put(pllb); - if (ret != 0) { - dev_err(&pdev->dev, "Failed to set MCLK parent\n"); - goto err_mclk; - } - - clk_set_rate(mclk, MCLK_RATE); - card->dev = &pdev->dev; /* Parse device node info */ @@ -230,9 +174,6 @@ static int at91sam9g20ek_audio_probe(struct platform_device *pdev) return ret; -err_mclk: - clk_put(mclk); - mclk = NULL; err: atmel_ssc_put_audio(0); return ret; @@ -242,8 +183,6 @@ static int at91sam9g20ek_audio_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); - clk_disable(mclk); - mclk = NULL; snd_soc_unregister_card(card); atmel_ssc_put_audio(0); -- cgit v1.2.3 From 9c363532413cda3e2c6dfa10e5cca7cd221877a0 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 31 Mar 2022 14:49:57 +0300 Subject: ASoC: topology: Correct error handling in soc_tplg_dapm_widget_create() Academic correction of error handling: In case the allocation of kc or kcontrol_type fails the correct label to jump is hdr_err since the template.sname has been also allocated at this point. Fixes: d29d41e28eea6 ("ASoC: topology: Add support for multiple kcontrol types to a widget") Signed-off-by: Peter Ujfalusi Reviewed-by: Ranjani Sridharan Reviewed-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220331114957.519-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/soc-topology.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 72e50df7052c..3bb90a819650 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1436,12 +1436,12 @@ static int soc_tplg_dapm_widget_create(struct soc_tplg *tplg, template.num_kcontrols = le32_to_cpu(w->num_kcontrols); kc = devm_kcalloc(tplg->dev, le32_to_cpu(w->num_kcontrols), sizeof(*kc), GFP_KERNEL); if (!kc) - goto err; + goto hdr_err; kcontrol_type = devm_kcalloc(tplg->dev, le32_to_cpu(w->num_kcontrols), sizeof(unsigned int), GFP_KERNEL); if (!kcontrol_type) - goto err; + goto hdr_err; for (i = 0; i < le32_to_cpu(w->num_kcontrols); i++) { control_hdr = (struct snd_soc_tplg_ctl_hdr *)tplg->pos; -- cgit v1.2.3 From acc72863e0f11cd0bedc888b663700229f9ba5ff Mon Sep 17 00:00:00 2001 From: Xiaomeng Tong Date: Sun, 27 Mar 2022 16:13:00 +0800 Subject: codecs: rt5682s: fix an incorrect NULL check on list iterator The bug is here: if (!dai) { The list iterator value 'dai' will *always* be set and non-NULL by for_each_component_dais(), so it is incorrect to assume that the iterator value will be NULL if the list is empty or no element is found (In fact, it will be a bogus pointer to an invalid struct object containing the HEAD). Otherwise it will bypass the check 'if (!dai) {' (never call dev_err() and never return -ENODEV;) and lead to invalid memory access lately when calling 'rt5682s_set_bclk1_ratio(dai, factor);'. To fix the bug, just return rt5682s_set_bclk1_ratio(dai, factor); when found the 'dai', otherwise dev_err() and return -ENODEV; Cc: stable@vger.kernel.org Fixes: bdd229ab26be9 ("ASoC: rt5682s: Add driver for ALC5682I-VS codec") Signed-off-by: Xiaomeng Tong Link: https://lore.kernel.org/r/20220327081300.12962-1-xiam0nd.tong@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682s.c | 11 ++++------- 1 file changed, 4 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt5682s.c b/sound/soc/codecs/rt5682s.c index 1cba8ec7cedb..b55f3ac3a267 100644 --- a/sound/soc/codecs/rt5682s.c +++ b/sound/soc/codecs/rt5682s.c @@ -2687,14 +2687,11 @@ static int rt5682s_bclk_set_rate(struct clk_hw *hw, unsigned long rate, for_each_component_dais(component, dai) if (dai->id == RT5682S_AIF1) - break; - if (!dai) { - dev_err(component->dev, "dai %d not found in component\n", - RT5682S_AIF1); - return -ENODEV; - } + return rt5682s_set_bclk1_ratio(dai, factor); - return rt5682s_set_bclk1_ratio(dai, factor); + dev_err(component->dev, "dai %d not found in component\n", + RT5682S_AIF1); + return -ENODEV; } static const struct clk_ops rt5682s_dai_clk_ops[RT5682S_DAI_NUM_CLKS] = { -- cgit v1.2.3 From 5708cc2f4b50c7bf27234eee77e1d9487533bbd3 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 31 Mar 2022 14:48:45 +0300 Subject: ASoC: SOF: topology: Fix memory leak of scontrol->name The scontrol->name is allocated with kstrdup, it must be freed before the scontrol is freed to avoid leaking memory. The constant leaking happens via sof_widget_unload() path on every module removal. Fixes: b5cee8feb1d48 ("ASoC: SOF: topology: Make control parsing IPC agnostic") Signed-off-by: Peter Ujfalusi Reviewed-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Link: https://lore.kernel.org/r/20220331114845.32747-1-peter.ujfalusi@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/topology.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 9b11e9795a7a..75d78f9178a3 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -941,11 +941,13 @@ static int sof_control_load(struct snd_soc_component *scomp, int index, default: dev_warn(scomp->dev, "control type not supported %d:%d:%d\n", hdr->ops.get, hdr->ops.put, hdr->ops.info); + kfree(scontrol->name); kfree(scontrol); return 0; } if (ret < 0) { + kfree(scontrol->name); kfree(scontrol); return ret; } @@ -1380,6 +1382,7 @@ static int sof_widget_unload(struct snd_soc_component *scomp, } kfree(scontrol->ipc_control_data); list_del(&scontrol->list); + kfree(scontrol->name); kfree(scontrol); } -- cgit v1.2.3 From fb6d679fee95d272c0a94912c4e534146823ee89 Mon Sep 17 00:00:00 2001 From: Christophe JAILLET Date: Thu, 31 Mar 2022 22:19:44 +0200 Subject: ASoC: soc-pcm: use GFP_KERNEL when the code is sleepable At the kzalloc() call in dpcm_be_connect(), there is no spin lock involved. It's merely protected by card->pcm_mutex, instead. The spinlock is applied at the later call with snd_soc_pcm_stream_lock_irq() only for the list manipulations. (See it's *_irq(), not *_irqsave(); that means the context being sleepable at that point.) So, we can use GFP_KERNEL safely there. This patch revert commit d8a9c6e1f676 ("ASoC: soc-pcm: use GFP_ATOMIC for dpcm structure") which is no longer needed since commit b7898396f4bb ("ASoC: soc-pcm: Fix and cleanup DPCM locking"). Signed-off-by: Christophe JAILLET Link: https://lore.kernel.org/r/e740f1930843060e025e3c0f17ec1393cfdafb26.1648757961.git.christophe.jaillet@wanadoo.fr Signed-off-by: Mark Brown --- sound/soc/soc-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 9a954680d492..11c9853e9e80 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1214,7 +1214,7 @@ static int dpcm_be_connect(struct snd_soc_pcm_runtime *fe, be_substream->pcm->nonatomic = 1; } - dpcm = kzalloc(sizeof(struct snd_soc_dpcm), GFP_ATOMIC); + dpcm = kzalloc(sizeof(struct snd_soc_dpcm), GFP_KERNEL); if (!dpcm) return -ENOMEM; -- cgit v1.2.3 From f730a46b931d894816af34a0ff8e4ad51565b39f Mon Sep 17 00:00:00 2001 From: Xiaomeng Tong Date: Tue, 29 Mar 2022 09:21:34 +0800 Subject: ASoC: soc-dapm: fix two incorrect uses of list iterator These two bug are here: list_for_each_entry_safe_continue(w, n, list, power_list); list_for_each_entry_safe_continue(w, n, list, power_list); After the list_for_each_entry_safe_continue() exits, the list iterator will always be a bogus pointer which point to an invalid struct objdect containing HEAD member. The funciton poniter 'w->event' will be a invalid value which can lead to a control-flow hijack if the 'w' can be controlled. The original intention was to continue the outer list_for_each_entry_safe() loop with the same entry if w->event is NULL, but misunderstanding the meaning of list_for_each_entry_safe_continue(). So just add a 'continue;' to fix the bug. Cc: stable@vger.kernel.org Fixes: 163cac061c973 ("ASoC: Factor out DAPM sequence execution") Signed-off-by: Xiaomeng Tong Link: https://lore.kernel.org/r/20220329012134.9375-1-xiam0nd.tong@gmail.com Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index b435b5c4cfb7..ca917a849c42 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1687,8 +1687,7 @@ static void dapm_seq_run(struct snd_soc_card *card, switch (w->id) { case snd_soc_dapm_pre: if (!w->event) - list_for_each_entry_safe_continue(w, n, list, - power_list); + continue; if (event == SND_SOC_DAPM_STREAM_START) ret = w->event(w, @@ -1700,8 +1699,7 @@ static void dapm_seq_run(struct snd_soc_card *card, case snd_soc_dapm_post: if (!w->event) - list_for_each_entry_safe_continue(w, n, list, - power_list); + continue; if (event == SND_SOC_DAPM_STREAM_START) ret = w->event(w, -- cgit v1.2.3 From c8618d65007ba68d7891130642d73e89372101e8 Mon Sep 17 00:00:00 2001 From: Xiaomeng Tong Date: Sun, 27 Mar 2022 16:10:02 +0800 Subject: ASoC: rt5682: fix an incorrect NULL check on list iterator The bug is here: if (!dai) { The list iterator value 'dai' will *always* be set and non-NULL by for_each_component_dais(), so it is incorrect to assume that the iterator value will be NULL if the list is empty or no element is found (In fact, it will be a bogus pointer to an invalid struct object containing the HEAD). Otherwise it will bypass the check 'if (!dai) {' (never call dev_err() and never return -ENODEV;) and lead to invalid memory access lately when calling 'rt5682_set_bclk1_ratio(dai, factor);'. To fix the bug, just return rt5682_set_bclk1_ratio(dai, factor); when found the 'dai', otherwise dev_err() and return -ENODEV; Cc: stable@vger.kernel.org Fixes: ebbfabc16d23d ("ASoC: rt5682: Add CCF usage for providing I2S clks") Signed-off-by: Xiaomeng Tong Link: https://lore.kernel.org/r/20220327081002.12684-1-xiam0nd.tong@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682.c | 11 ++++------- 1 file changed, 4 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index be68d573a490..c9ff9c89adf7 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -2822,14 +2822,11 @@ static int rt5682_bclk_set_rate(struct clk_hw *hw, unsigned long rate, for_each_component_dais(component, dai) if (dai->id == RT5682_AIF1) - break; - if (!dai) { - dev_err(rt5682->i2c_dev, "dai %d not found in component\n", - RT5682_AIF1); - return -ENODEV; - } + return rt5682_set_bclk1_ratio(dai, factor); - return rt5682_set_bclk1_ratio(dai, factor); + dev_err(rt5682->i2c_dev, "dai %d not found in component\n", + RT5682_AIF1); + return -ENODEV; } static const struct clk_ops rt5682_dai_clk_ops[RT5682_DAI_NUM_CLKS] = { -- cgit v1.2.3 From c598ccfbeb26cb9452f99e7beb92ef779dcb16b1 Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Thu, 24 Mar 2022 16:18:38 +0800 Subject: ASoC: cs35l41: Add one more variable in the debug log otp_map[].size is a key variable to compute the value of otp_val and to update the bit_offset, it is helpful to debug if could put it in the debug log. Signed-off-by: Hui Wang Reviewed-by: Lucas Tanure Link: https://lore.kernel.org/r/20220324081839.62009-1-hui.wang@canonical.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l41-lib.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs35l41-lib.c b/sound/soc/codecs/cs35l41-lib.c index e5a56bcbb223..d0a480c40231 100644 --- a/sound/soc/codecs/cs35l41-lib.c +++ b/sound/soc/codecs/cs35l41-lib.c @@ -822,8 +822,8 @@ int cs35l41_otp_unpack(struct device *dev, struct regmap *regmap) word_offset = otp_map_match->word_offset; for (i = 0; i < otp_map_match->num_elements; i++) { - dev_dbg(dev, "bitoffset= %d, word_offset=%d, bit_sum mod 32=%d\n", - bit_offset, word_offset, bit_sum % 32); + dev_dbg(dev, "bitoffset= %d, word_offset=%d, bit_sum mod 32=%d otp_map[i].size = %d\n", + bit_offset, word_offset, bit_sum % 32, otp_map[i].size); if (bit_offset + otp_map[i].size - 1 >= 32) { otp_val = (otp_mem[word_offset] & GENMASK(31, bit_offset)) >> bit_offset; -- cgit v1.2.3 From 0b3d5d2e358ca6772fc3662fca27acb12a682fbf Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Thu, 24 Mar 2022 16:18:39 +0800 Subject: ASoC: cs35l41: Fix a shift-out-of-bounds warning found by UBSAN We enabled UBSAN in the ubuntu kernel, and the cs35l41 driver triggers a warning calltrace like below: cs35l41-hda i2c-CSC3551:00-cs35l41-hda.0: bitoffset= 8, word_offset=23, bit_sum mod 32=0, otp_map[i].size = 24 cs35l41-hda i2c-CSC3551:00-cs35l41-hda.0: bitoffset= 0, word_offset=24, bit_sum mod 32=24, otp_map[i].size = 0 ================================================================================ UBSAN: shift-out-of-bounds in linux-kernel-src/sound/soc/codecs/cs35l41-lib.c:836:8 shift exponent 64 is too large for 64-bit type 'long unsigned int' CPU: 10 PID: 595 Comm: systemd-udevd Not tainted 5.15.0-23-generic #23 Hardware name: LENOVO \x02MFG_IN_GO/\x02MFG_IN_GO, BIOS N3GET19W (1.00 ) 03/11/2022 Call Trace: show_stack+0x52/0x58 dump_stack_lvl+0x4a/0x5f dump_stack+0x10/0x12 ubsan_epilogue+0x9/0x45 __ubsan_handle_shift_out_of_bounds.cold+0x61/0xef ? regmap_unlock_mutex+0xe/0x10 cs35l41_otp_unpack.cold+0x1c6/0x2b2 [snd_soc_cs35l41_lib] cs35l41_hda_probe+0x24f/0x33a [snd_hda_scodec_cs35l41] cs35l41_hda_i2c_probe+0x65/0x90 [snd_hda_scodec_cs35l41_i2c] When both bitoffset and otp_map[i].size are 0, the line 836 will result in GENMASK(-1, 0), this triggers the shift-out-of-bounds calltrace. Here add a checking, if both bitoffset and otp_map[i].size are 0, do not run GENMASK() and directly set otp_val to 0, this will not bring any function change on the driver but could avoid the calltrace. Signed-off-by: Hui Wang Link: https://lore.kernel.org/r/20220324081839.62009-2-hui.wang@canonical.com Signed-off-by: Mark Brown --- sound/soc/codecs/cs35l41-lib.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs35l41-lib.c b/sound/soc/codecs/cs35l41-lib.c index d0a480c40231..aa6823fbd1a4 100644 --- a/sound/soc/codecs/cs35l41-lib.c +++ b/sound/soc/codecs/cs35l41-lib.c @@ -831,12 +831,14 @@ int cs35l41_otp_unpack(struct device *dev, struct regmap *regmap) GENMASK(bit_offset + otp_map[i].size - 33, 0)) << (32 - bit_offset); bit_offset += otp_map[i].size - 32; - } else { + } else if (bit_offset + otp_map[i].size - 1 >= 0) { otp_val = (otp_mem[word_offset] & GENMASK(bit_offset + otp_map[i].size - 1, bit_offset) ) >> bit_offset; bit_offset += otp_map[i].size; - } + } else /* both bit_offset and otp_map[i].size are 0 */ + otp_val = 0; + bit_sum += otp_map[i].size; if (bit_offset == 32) { -- cgit v1.2.3 From 8ba08d3a367a70f707b7c5d53ad92b98b960ee88 Mon Sep 17 00:00:00 2001 From: Miaoqian Lin Date: Mon, 4 Apr 2022 09:07:46 +0000 Subject: ASoC: rk817: Use devm_clk_get() in rk817_platform_probe We need to call clk_put() to undo clk_get() in the error path. Use devm_clk_get() to obtain a reference to the clock, It has the benefit that clk_put() is no longer required. Fixes: 0d6a04da9b25 ("ASoC: Add Rockchip rk817 audio CODEC support") Signed-off-by: Miaoqian Lin Link: https://lore.kernel.org/r/20220404090753.17940-1-linmq006@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/rk817_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rk817_codec.c b/sound/soc/codecs/rk817_codec.c index 8fffe378618d..cce6f4e7992f 100644 --- a/sound/soc/codecs/rk817_codec.c +++ b/sound/soc/codecs/rk817_codec.c @@ -489,7 +489,7 @@ static int rk817_platform_probe(struct platform_device *pdev) rk817_codec_parse_dt_property(&pdev->dev, rk817_codec_data); - rk817_codec_data->mclk = clk_get(pdev->dev.parent, "mclk"); + rk817_codec_data->mclk = devm_clk_get(pdev->dev.parent, "mclk"); if (IS_ERR(rk817_codec_data->mclk)) { dev_dbg(&pdev->dev, "Unable to get mclk\n"); ret = -ENXIO; -- cgit v1.2.3 From e927b05f3cc20de87f6b7d912a5bbe556931caca Mon Sep 17 00:00:00 2001 From: Miaoqian Lin Date: Sun, 3 Apr 2022 11:52:39 +0000 Subject: ASoC: msm8916-wcd-digital: Check failure for devm_snd_soc_register_component devm_snd_soc_register_component() may fails, we should check the error and do the corresponding error handling. Fixes: 150db8c5afa1 ("ASoC: codecs: Add msm8916-wcd digital codec") Signed-off-by: Miaoqian Lin Link: https://lore.kernel.org/r/20220403115239.30140-1-linmq006@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/msm8916-wcd-digital.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/msm8916-wcd-digital.c b/sound/soc/codecs/msm8916-wcd-digital.c index 9ad7fc0baf07..20a07c92b2fc 100644 --- a/sound/soc/codecs/msm8916-wcd-digital.c +++ b/sound/soc/codecs/msm8916-wcd-digital.c @@ -1206,9 +1206,16 @@ static int msm8916_wcd_digital_probe(struct platform_device *pdev) dev_set_drvdata(dev, priv); - return devm_snd_soc_register_component(dev, &msm8916_wcd_digital, + ret = devm_snd_soc_register_component(dev, &msm8916_wcd_digital, msm8916_wcd_digital_dai, ARRAY_SIZE(msm8916_wcd_digital_dai)); + if (ret) + goto err_mclk; + + return 0; + +err_mclk: + clk_disable_unprepare(priv->mclk); err_clk: clk_disable_unprepare(priv->ahbclk); return ret; -- cgit v1.2.3 From d462f6ed2aeac30c0b440a91fb05d964956935f9 Mon Sep 17 00:00:00 2001 From: Heiner Kallweit Date: Wed, 9 Mar 2022 21:21:55 +0100 Subject: ASoC: soc-core: add debugfs_prefix member to snd_soc_component_driver Allow the component debugfs_prefix to be set from snd_soc_component_driver. First use case is avoiding a duplicate debugfs entry error in case a device has multiple components which have the same name therefore. Note that we don't set component->debugfs_prefix if it's set already. That's needed because partially component->debugfs_prefix is set before calling snd_soc_component_initialize(). Signed-off-by: Heiner Kallweit Link: https://lore.kernel.org/r/d18bff6a-1df1-5f95-0cf8-10dbaa62d7be@gmail.com Signed-off-by: Mark Brown --- include/sound/soc-component.h | 4 ++++ sound/soc/soc-core.c | 5 +++++ 2 files changed, 9 insertions(+) diff --git a/include/sound/soc-component.h b/include/sound/soc-component.h index a52080407b98..766dc6f009c0 100644 --- a/include/sound/soc-component.h +++ b/include/sound/soc-component.h @@ -179,6 +179,10 @@ struct snd_soc_component_driver { struct snd_pcm_hw_params *params); bool use_dai_pcm_id; /* use DAI link PCM ID as PCM device number */ int be_pcm_base; /* base device ID for all BE PCMs */ + +#ifdef CONFIG_DEBUG_FS + const char *debugfs_prefix; +#endif }; struct snd_soc_component { diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ce153ac2c3ab..8c7da82a62ca 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2587,6 +2587,11 @@ int snd_soc_component_initialize(struct snd_soc_component *component, component->dev = dev; component->driver = driver; +#ifdef CONFIG_DEBUG_FS + if (!component->debugfs_prefix) + component->debugfs_prefix = driver->debugfs_prefix; +#endif + return 0; } EXPORT_SYMBOL_GPL(snd_soc_component_initialize); -- cgit v1.2.3 From fc35880d198d9f2023bf231c120e1a69ad4db841 Mon Sep 17 00:00:00 2001 From: Heiner Kallweit Date: Wed, 9 Mar 2022 21:23:06 +0100 Subject: ASoC: meson: aiu: fix duplicate debugfs directory error On a S905W-based system I get the following error: debugfs: Directory 'c1105400.audio-controller' with parent 'P230-Q200' already present! Turned out that multiple components having the same name triggers this error in soc_init_component_debugfs(). With the patch the error is gone and that's the debugfs entries. /sys/kernel/debug/asoc/P230-Q200/acodec:c1105400.audio-controller /sys/kernel/debug/asoc/P230-Q200/hdmi:c1105400.audio-controller /sys/kernel/debug/asoc/P230-Q200/cpu:c1105400.audio-controller Signed-off-by: Heiner Kallweit Link: https://lore.kernel.org/r/38053baf-c33b-7fdf-7593-99b22153a9c0@gmail.com Signed-off-by: Mark Brown --- sound/soc/meson/aiu-acodec-ctrl.c | 3 +++ sound/soc/meson/aiu-codec-ctrl.c | 3 +++ sound/soc/meson/aiu.c | 3 +++ 3 files changed, 9 insertions(+) diff --git a/sound/soc/meson/aiu-acodec-ctrl.c b/sound/soc/meson/aiu-acodec-ctrl.c index 27a6d3259c50..22e181646bc3 100644 --- a/sound/soc/meson/aiu-acodec-ctrl.c +++ b/sound/soc/meson/aiu-acodec-ctrl.c @@ -193,6 +193,9 @@ static const struct snd_soc_component_driver aiu_acodec_ctrl_component = { .of_xlate_dai_name = aiu_acodec_of_xlate_dai_name, .endianness = 1, .non_legacy_dai_naming = 1, +#ifdef CONFIG_DEBUG_FS + .debugfs_prefix = "acodec", +#endif }; int aiu_acodec_ctrl_register_component(struct device *dev) diff --git a/sound/soc/meson/aiu-codec-ctrl.c b/sound/soc/meson/aiu-codec-ctrl.c index c3ea733fce91..59ee66fc2bcd 100644 --- a/sound/soc/meson/aiu-codec-ctrl.c +++ b/sound/soc/meson/aiu-codec-ctrl.c @@ -140,6 +140,9 @@ static const struct snd_soc_component_driver aiu_hdmi_ctrl_component = { .of_xlate_dai_name = aiu_hdmi_of_xlate_dai_name, .endianness = 1, .non_legacy_dai_naming = 1, +#ifdef CONFIG_DEBUG_FS + .debugfs_prefix = "hdmi", +#endif }; int aiu_hdmi_ctrl_register_component(struct device *dev) diff --git a/sound/soc/meson/aiu.c b/sound/soc/meson/aiu.c index d299a70db7e5..88e611e64d14 100644 --- a/sound/soc/meson/aiu.c +++ b/sound/soc/meson/aiu.c @@ -103,6 +103,9 @@ static const struct snd_soc_component_driver aiu_cpu_component = { .pointer = aiu_fifo_pointer, .probe = aiu_cpu_component_probe, .remove = aiu_cpu_component_remove, +#ifdef CONFIG_DEBUG_FS + .debugfs_prefix = "cpu", +#endif }; static struct snd_soc_dai_driver aiu_cpu_dai_drv[] = { -- cgit v1.2.3 From 51a630a7051f7f4f1cfdd64c20c7110f9907c230 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Mon, 4 Apr 2022 12:32:52 +0100 Subject: ASoC: simple-card-utils: Avoid NULL deref in asoc_simple_set_tdm() Don't dereference simple_dai before it has been checked for NULL. Signed-off-by: Richard Fitzgerald Fixes: 1e974e5b82b3 ("ASoC: audio_graph_card2: Add support for variable slot widths") Reported-by: kernel test robot Reported-by: Dan Carpenter Link: https://lore.kernel.org/r/20220404113252.1152659-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- sound/soc/generic/simple-card-utils.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index 8e037835bc58..f2157944247f 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -364,13 +364,15 @@ static int asoc_simple_set_tdm(struct snd_soc_dai *dai, struct snd_pcm_hw_params *params) { int sample_bits = params_width(params); - int slot_width = simple_dai->slot_width; - int slot_count = simple_dai->slots; + int slot_width, slot_count; int i, ret; if (!simple_dai || !simple_dai->tdm_width_map) return 0; + slot_width = simple_dai->slot_width; + slot_count = simple_dai->slots; + if (slot_width == 0) slot_width = sample_bits; -- cgit v1.2.3 From d00887c106dac47b9af6ed70e8d5c45b69c4bd52 Mon Sep 17 00:00:00 2001 From: Ahmad Fatoum Date: Tue, 5 Apr 2022 17:57:31 +0200 Subject: ASoC: fsl_sai: fix 1:1 bclk:mclk ratio support Refactoring in commit a50b7926d015 ("ASoC: fsl_sai: implement 1:1 bclk:mclk ratio support") led to the bypass never happening as (ratio = 1) was caught in the existing if (ratio & 1) continue; check. The correct check sequence instead is: - skip all ratios lower than one and higher than 512 - skip all odd ratios except for 1:1 - skip 1:1 ratio if and only if !support_1_1_ratio And for all others, calculate the appropriate divider. Adjust the code to facilitate this. Fixes: a50b7926d015 ("ASoC: fsl_sai: implement 1:1 bclk:mclk ratio support") Signed-off-by: Ahmad Fatoum Acked-by: Shengjiu Wang Reviewed-by: Sascha Hauer Link: https://lore.kernel.org/r/20220405155731.745413-1-a.fatoum@pengutronix.de Signed-off-by: Mark Brown --- sound/soc/fsl/fsl_sai.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 4650a6931a94..ffc24afb5a7a 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -372,7 +372,7 @@ static int fsl_sai_set_bclk(struct snd_soc_dai *dai, bool tx, u32 freq) continue; if (ratio == 1 && !support_1_1_ratio) continue; - else if (ratio & 1) + if ((ratio & 1) && ratio > 1) continue; diff = abs((long)clk_rate - ratio * freq); -- cgit v1.2.3 From fcd1e39cca6e3a262f2badfcd5edd76c910ad3bc Mon Sep 17 00:00:00 2001 From: Ajye Huang Date: Thu, 24 Mar 2022 16:47:08 +0800 Subject: ASoC: Intel: sof_rt5682: Add support for max98360a speaker amp on SSP2 Follow Intel's design to replace max98360a amp SSP2 reather than SSP1 by judging DMI_OEM_STRING in sof_rt5682_quirk_table struct. And reusing max98357's topology since DAI setting could be leveraged. Signed-off-by: Ajye Huang Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220324084708.2009375-1-ajye_huang@compal.corp-partner.google.com Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_rt5682.c | 13 +++++++++++++ sound/soc/sof/sof-pci-dev.c | 9 ++++++++- 2 files changed, 21 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/boards/sof_rt5682.c b/sound/soc/intel/boards/sof_rt5682.c index ebec4d15edaa..7126fcb63d90 100644 --- a/sound/soc/intel/boards/sof_rt5682.c +++ b/sound/soc/intel/boards/sof_rt5682.c @@ -212,6 +212,19 @@ static const struct dmi_system_id sof_rt5682_quirk_table[] = { SOF_SSP_BT_OFFLOAD_PRESENT), }, + { + .callback = sof_rt5682_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Brya"), + DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98360_ALC5682I_I2S_AMP_SSP2"), + }, + .driver_data = (void *)(SOF_RT5682_MCLK_EN | + SOF_RT5682_SSP_CODEC(0) | + SOF_SPEAKER_AMP_PRESENT | + SOF_MAX98360A_SPEAKER_AMP_PRESENT | + SOF_RT5682_SSP_AMP(2) | + SOF_RT5682_NUM_HDMIDEV(4)), + }, {} }; diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index 4c9596742844..12f5cff22448 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -83,7 +83,14 @@ static const struct dmi_system_id sof_tplg_table[] = { }, .driver_data = "sof-adl-max98357a-rt5682-2way.tplg", }, - + { + .callback = sof_tplg_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_FAMILY, "Google_Brya"), + DMI_MATCH(DMI_OEM_STRING, "AUDIO-MAX98360_ALC5682I_I2S_AMP_SSP2"), + }, + .driver_data = "sof-adl-max98357a-rt5682.tplg", + }, {} }; -- cgit v1.2.3 From 770f3d992a3f7330f801dfeee98429b2885c9fdb Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Wed, 6 Apr 2022 14:20:05 -0500 Subject: ASoC: rt711/5682: check if bus is active before deferred jack detection This patch takes a defensive programming and paranoid approach in case the parent device (SoundWire) is pm_runtime resumed but the rt711 device is not. In that case, during the attachment and initialization, a jack detection workqueue can be scheduled. Since the pm_runtime suspend routines will not be invoked, the sequence to cancel all deferred work is not executed, and the jack detection could happen after the bus stops operating, leading to a timeout. This patch applies the same solution to rt5682, based on the similarities between codec drivers. The race condition with rt5682 was not detected experimentally though. BugLink: https://github.com/thesofproject/linux/issues/3459 Signed-off-by: Pierre-Louis Bossart Reviewed-by: Rander Wang Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220406192005.262996-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt5682.c | 9 +++++++++ sound/soc/codecs/rt711.c | 7 +++++++ 2 files changed, 16 insertions(+) diff --git a/sound/soc/codecs/rt5682.c b/sound/soc/codecs/rt5682.c index c9ff9c89adf7..2b6c6d6b9771 100644 --- a/sound/soc/codecs/rt5682.c +++ b/sound/soc/codecs/rt5682.c @@ -1100,6 +1100,15 @@ void rt5682_jack_detect_handler(struct work_struct *work) return; } + if (rt5682->is_sdw) { + if (pm_runtime_status_suspended(rt5682->slave->dev.parent)) { + dev_dbg(&rt5682->slave->dev, + "%s: parent device is pm_runtime_status_suspended, skipping jack detection\n", + __func__); + return; + } + } + dapm = snd_soc_component_get_dapm(rt5682->component); snd_soc_dapm_mutex_lock(dapm); diff --git a/sound/soc/codecs/rt711.c b/sound/soc/codecs/rt711.c index 6770825d037a..ea25fd58d43a 100644 --- a/sound/soc/codecs/rt711.c +++ b/sound/soc/codecs/rt711.c @@ -245,6 +245,13 @@ static void rt711_jack_detect_handler(struct work_struct *work) if (!rt711->component->card->instantiated) return; + if (pm_runtime_status_suspended(rt711->slave->dev.parent)) { + dev_dbg(&rt711->slave->dev, + "%s: parent device is pm_runtime_status_suspended, skipping jack detection\n", + __func__); + return; + } + reg = RT711_VERB_GET_PIN_SENSE | RT711_HP_OUT; ret = regmap_read(rt711->regmap, reg, &jack_status); if (ret < 0) -- cgit v1.2.3 From 20744617bdbafe2e7fb7bf5401f616e24bde4471 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Wed, 6 Apr 2022 14:16:06 -0500 Subject: ASoC: SOF: topology: cleanup dailinks on widget unload MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit We set the cpu_dai capture_ or playback_widget on widget_ready but never clear them, which leads to failures when unloading/reloading a topology in modprobe/rmmod tests BugLink: https://github.com/thesofproject/linux/issues/3535 Fixes: 311ce4fe7637 ("ASoC: SOF: Add support for loading topologies") Signed-off-by: Pierre-Louis Bossart Reviewed-by: Ranjani Sridharan Reviewed-by: Péter Ujfalusi Reviewed-by: Bard Liao Link: https://lore.kernel.org/r/20220406191606.254576-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/topology.c | 43 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 43 insertions(+) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 75d78f9178a3..5953d1050cc9 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -1070,6 +1070,46 @@ static int sof_connect_dai_widget(struct snd_soc_component *scomp, return 0; } +static void sof_disconnect_dai_widget(struct snd_soc_component *scomp, + struct snd_soc_dapm_widget *w) +{ + struct snd_soc_card *card = scomp->card; + struct snd_soc_pcm_runtime *rtd; + struct snd_soc_dai *cpu_dai; + int i; + + if (!w->sname) + return; + + list_for_each_entry(rtd, &card->rtd_list, list) { + /* does stream match DAI link ? */ + if (!rtd->dai_link->stream_name || + strcmp(w->sname, rtd->dai_link->stream_name)) + continue; + + switch (w->id) { + case snd_soc_dapm_dai_out: + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { + if (cpu_dai->capture_widget == w) { + cpu_dai->capture_widget = NULL; + break; + } + } + break; + case snd_soc_dapm_dai_in: + for_each_rtd_cpu_dais(rtd, i, cpu_dai) { + if (cpu_dai->playback_widget == w) { + cpu_dai->playback_widget = NULL; + break; + } + } + break; + default: + break; + } + } +} + /* bind PCM ID to host component ID */ static int spcm_bind(struct snd_soc_component *scomp, struct snd_sof_pcm *spcm, int dir) @@ -1355,6 +1395,9 @@ static int sof_widget_unload(struct snd_soc_component *scomp, if (dai) list_del(&dai->list); + + sof_disconnect_dai_widget(scomp, widget); + break; default: break; -- cgit v1.2.3 From 9b91d0ece22b9ab37fc185511c7f992e51c93d6e Mon Sep 17 00:00:00 2001 From: Yu Liao Date: Fri, 18 Mar 2022 10:16:16 +0800 Subject: ASoC: SOF: topology: Fix memory leak in sof_control_load() scontrol doesn't get freed when kstrdup returns NULL. Fix by free iscontrol in that case. scontrol = kzalloc(sizeof(*scontrol), GFP_KERNEL); if (!scontrol) return -ENOMEM; scontrol->name = kstrdup(hdr->name, GFP_KERNEL); if (!scontrol->name) return -ENOMEM; Signed-off-by: Yu Liao Link: https://lore.kernel.org/r/20220318021616.2599630-1-liaoyu15@huawei.com Signed-off-by: Mark Brown --- sound/soc/sof/topology.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/sof/topology.c b/sound/soc/sof/topology.c index 5953d1050cc9..3e5b319b44c7 100644 --- a/sound/soc/sof/topology.c +++ b/sound/soc/sof/topology.c @@ -904,8 +904,10 @@ static int sof_control_load(struct snd_soc_component *scomp, int index, return -ENOMEM; scontrol->name = kstrdup(hdr->name, GFP_KERNEL); - if (!scontrol->name) + if (!scontrol->name) { + kfree(scontrol); return -ENOMEM; + } scontrol->scomp = scomp; scontrol->access = kc->access; -- cgit v1.2.3 From db6dd1bee63d1d88fbddfe07af800af5948ac28e Mon Sep 17 00:00:00 2001 From: Srinivas Kandagatla Date: Thu, 7 Apr 2022 10:43:13 +0100 Subject: ASoC: codecs: wcd934x: do not switch off SIDO Buck when codec is in use SIDO(Single-Inductor Dual-Ouput) Buck powers up both analog and digital circuits along with internal memory, powering off this is the last thing that codec should do when going to very low power. Current code was powering off this Buck if there are no users of sysclk, which is not correct. Powering off this buck will result in no register access. This code path was never tested until recently after adding pm support in SoundWire controller. Fix this by removing the buck poweroff when the codec is active and also the code that is not used. Without this patch all the read/write transactions will never complete and results in SLIMBus Errors like: qcom,slim-ngd qcom,slim-ngd.1: Tx:MT:0x0, MC:0x60, LA:0xcf failed:-110 wcd934x-codec wcd934x-codec.1.auto: ASoC: error at soc_component_read_no_lock on wcd934x-codec.1.auto for register: [0x00000d05] -110 qcom,slim-ngd-ctrl 171c0000.slim: Error Interrupt received 0x82000000 Reported-by: Amit Pundir Fixes: a61f3b4f476e ("ASoC: wcd934x: add support to wcd9340/wcd9341 codec") Signed-off-by: Srinivas Kandagatla Tested-by: Amit Pundir Link: https://lore.kernel.org/r/20220407094313.2880-1-srinivas.kandagatla@linaro.org Signed-off-by: Mark Brown --- sound/soc/codecs/wcd934x.c | 26 +------------------------- 1 file changed, 1 insertion(+), 25 deletions(-) diff --git a/sound/soc/codecs/wcd934x.c b/sound/soc/codecs/wcd934x.c index 1e75e93cf28f..6298ebe96e94 100644 --- a/sound/soc/codecs/wcd934x.c +++ b/sound/soc/codecs/wcd934x.c @@ -1274,29 +1274,7 @@ static int wcd934x_set_sido_input_src(struct wcd934x_codec *wcd, int sido_src) if (sido_src == wcd->sido_input_src) return 0; - if (sido_src == SIDO_SOURCE_INTERNAL) { - regmap_update_bits(wcd->regmap, WCD934X_ANA_BUCK_CTL, - WCD934X_ANA_BUCK_HI_ACCU_EN_MASK, 0); - usleep_range(100, 110); - regmap_update_bits(wcd->regmap, WCD934X_ANA_BUCK_CTL, - WCD934X_ANA_BUCK_HI_ACCU_PRE_ENX_MASK, 0x0); - usleep_range(100, 110); - regmap_update_bits(wcd->regmap, WCD934X_ANA_RCO, - WCD934X_ANA_RCO_BG_EN_MASK, 0); - usleep_range(100, 110); - regmap_update_bits(wcd->regmap, WCD934X_ANA_BUCK_CTL, - WCD934X_ANA_BUCK_PRE_EN1_MASK, - WCD934X_ANA_BUCK_PRE_EN1_ENABLE); - usleep_range(100, 110); - regmap_update_bits(wcd->regmap, WCD934X_ANA_BUCK_CTL, - WCD934X_ANA_BUCK_PRE_EN2_MASK, - WCD934X_ANA_BUCK_PRE_EN2_ENABLE); - usleep_range(100, 110); - regmap_update_bits(wcd->regmap, WCD934X_ANA_BUCK_CTL, - WCD934X_ANA_BUCK_HI_ACCU_EN_MASK, - WCD934X_ANA_BUCK_HI_ACCU_ENABLE); - usleep_range(100, 110); - } else if (sido_src == SIDO_SOURCE_RCO_BG) { + if (sido_src == SIDO_SOURCE_RCO_BG) { regmap_update_bits(wcd->regmap, WCD934X_ANA_RCO, WCD934X_ANA_RCO_BG_EN_MASK, WCD934X_ANA_RCO_BG_ENABLE); @@ -1382,8 +1360,6 @@ static int wcd934x_disable_ana_bias_and_syclk(struct wcd934x_codec *wcd) regmap_update_bits(wcd->regmap, WCD934X_CLK_SYS_MCLK_PRG, WCD934X_EXT_CLK_BUF_EN_MASK | WCD934X_MCLK_EN_MASK, 0x0); - wcd934x_set_sido_input_src(wcd, SIDO_SOURCE_INTERNAL); - regmap_update_bits(wcd->regmap, WCD934X_ANA_BIAS, WCD934X_ANA_BIAS_EN_MASK, 0); regmap_update_bits(wcd->regmap, WCD934X_ANA_BIAS, -- cgit v1.2.3 From 97326be14df7bacc6ba5c62c0556298c27ea0432 Mon Sep 17 00:00:00 2001 From: Chao Song Date: Wed, 6 Apr 2022 14:23:41 -0500 Subject: ASoC: Intel: soc-acpi: correct device endpoints for max98373 The left speaker of max98373 uses spk_r_endpoint, and right speaker uses spk_l_endpoint, this is obviously wrong. This patch corrects the endpoints for max98373 codec. Signed-off-by: Chao Song Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220406192341.271465-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/intel/common/soc-acpi-intel-tgl-match.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c index 6edc9b7108cd..ef19150e7b2e 100644 --- a/sound/soc/intel/common/soc-acpi-intel-tgl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-tgl-match.c @@ -132,13 +132,13 @@ static const struct snd_soc_acpi_adr_device mx8373_1_adr[] = { { .adr = 0x000123019F837300ull, .num_endpoints = 1, - .endpoints = &spk_l_endpoint, + .endpoints = &spk_r_endpoint, .name_prefix = "Right" }, { .adr = 0x000127019F837300ull, .num_endpoints = 1, - .endpoints = &spk_r_endpoint, + .endpoints = &spk_l_endpoint, .name_prefix = "Left" } }; -- cgit v1.2.3 From 92ccbf17eeacf510cf1eed9c252d9332ca24f02d Mon Sep 17 00:00:00 2001 From: Zheyu Ma Date: Tue, 5 Apr 2022 20:10:38 +0800 Subject: ASoC: wm8731: Disable the regulator when probing fails When the driver fails during probing, the driver should disable the regulator, not just handle it in wm8731_hw_init(). The following log reveals it: [ 17.812483] WARNING: CPU: 1 PID: 364 at drivers/regulator/core.c:2257 _regulator_put+0x3ec/0x4e0 [ 17.815958] RIP: 0010:_regulator_put+0x3ec/0x4e0 [ 17.824467] Call Trace: [ 17.824774] [ 17.825040] regulator_bulk_free+0x82/0xe0 [ 17.825514] devres_release_group+0x319/0x3d0 [ 17.825882] i2c_device_probe+0x766/0x940 [ 17.829198] i2c_register_driver+0xb5/0x130 Signed-off-by: Zheyu Ma Link: https://lore.kernel.org/r/20220405121038.4094051-1-zheyuma97@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/wm8731.c | 19 +++++++++++-------- 1 file changed, 11 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/wm8731.c b/sound/soc/codecs/wm8731.c index 5d4949c2ec9b..b14c6d104e6d 100644 --- a/sound/soc/codecs/wm8731.c +++ b/sound/soc/codecs/wm8731.c @@ -602,7 +602,7 @@ static int wm8731_hw_init(struct device *dev, struct wm8731_priv *wm8731) ret = wm8731_reset(wm8731->regmap); if (ret < 0) { dev_err(dev, "Failed to issue reset: %d\n", ret); - goto err_regulator_enable; + goto err; } /* Clear POWEROFF, keep everything else disabled */ @@ -619,10 +619,7 @@ static int wm8731_hw_init(struct device *dev, struct wm8731_priv *wm8731) regcache_mark_dirty(wm8731->regmap); -err_regulator_enable: - /* Regulators will be enabled by bias management */ - regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); - +err: return ret; } @@ -760,21 +757,27 @@ static int wm8731_i2c_probe(struct i2c_client *i2c, ret = PTR_ERR(wm8731->regmap); dev_err(&i2c->dev, "Failed to allocate register map: %d\n", ret); - return ret; + goto err_regulator_enable; } ret = wm8731_hw_init(&i2c->dev, wm8731); if (ret != 0) - return ret; + goto err_regulator_enable; ret = devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_wm8731, &wm8731_dai, 1); if (ret != 0) { dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); - return ret; + goto err_regulator_enable; } return 0; + +err_regulator_enable: + /* Regulators will be enabled by bias management */ + regulator_bulk_disable(ARRAY_SIZE(wm8731->supplies), wm8731->supplies); + + return ret; } static const struct i2c_device_id wm8731_i2c_id[] = { -- cgit v1.2.3 From 890a4087a6c2045911b5002566d1528f710cd723 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 7 Apr 2022 20:49:56 +0200 Subject: ASoC: Intel: sof_es8336: simplify speaker gpio naming In preparation for the support of an additional gpio for headphone control, rename GPIOs to make explicit references to speakers and gpio0 or gpio1. No functionality change. Signed-off-by: Pierre-Louis Bossart Signed-off-by: Mauro Carvalho Chehab Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/3008c576ca45d5cc99ad4a18d1d30de45a0aff80.1649357263.git.mchehab@kernel.org Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_es8336.c | 42 ++++++++++++++++++------------------- 1 file changed, 21 insertions(+), 21 deletions(-) diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c index 5e0529aa4f1d..e4829a376b79 100644 --- a/sound/soc/intel/boards/sof_es8336.c +++ b/sound/soc/intel/boards/sof_es8336.c @@ -27,7 +27,7 @@ #define SOF_ES8336_SSP_CODEC(quirk) ((quirk) & GENMASK(3, 0)) #define SOF_ES8336_SSP_CODEC_MASK (GENMASK(3, 0)) -#define SOF_ES8336_TGL_GPIO_QUIRK BIT(4) +#define SOF_ES8336_SPEAKERS_EN_GPIO1_QUIRK BIT(4) #define SOF_ES8336_ENABLE_DMIC BIT(5) #define SOF_ES8336_JD_INVERTED BIT(6) @@ -39,7 +39,7 @@ MODULE_PARM_DESC(quirk, "Board-specific quirk override"); struct sof_es8336_private { struct device *codec_dev; - struct gpio_desc *gpio_pa; + struct gpio_desc *gpio_speakers; struct snd_soc_jack jack; struct list_head hdmi_pcm_list; bool speaker_en; @@ -51,19 +51,19 @@ struct sof_hdmi_pcm { int device; }; -static const struct acpi_gpio_params pa_enable_gpio = { 0, 0, true }; -static const struct acpi_gpio_mapping acpi_es8336_gpios[] = { - { "pa-enable-gpios", &pa_enable_gpio, 1 }, +static const struct acpi_gpio_params speakers_enable_gpio0 = { 0, 0, true }; +static const struct acpi_gpio_mapping acpi_speakers_enable_gpio0[] = { + { "speakers-enable-gpios", &speakers_enable_gpio0, 1 }, { } }; -static const struct acpi_gpio_params quirk_pa_enable_gpio = { 1, 0, true }; -static const struct acpi_gpio_mapping quirk_acpi_es8336_gpios[] = { - { "pa-enable-gpios", &quirk_pa_enable_gpio, 1 }, +static const struct acpi_gpio_params speakers_enable_gpio1 = { 1, 0, true }; +static const struct acpi_gpio_mapping acpi_speakers_enable_gpio1[] = { + { "speakers-enable-gpios", &speakers_enable_gpio1, 1 }, { } }; -static const struct acpi_gpio_mapping *gpio_mapping = acpi_es8336_gpios; +static const struct acpi_gpio_mapping *gpio_mapping = acpi_speakers_enable_gpio0; static void log_quirks(struct device *dev) { @@ -71,8 +71,8 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk SSP%ld\n", SOF_ES8336_SSP_CODEC(quirk)); if (quirk & SOF_ES8336_ENABLE_DMIC) dev_info(dev, "quirk DMIC enabled\n"); - if (quirk & SOF_ES8336_TGL_GPIO_QUIRK) - dev_info(dev, "quirk TGL GPIO enabled\n"); + if (quirk & SOF_ES8336_SPEAKERS_EN_GPIO1_QUIRK) + dev_info(dev, "Speakers GPIO1 quirk enabled\n"); if (quirk & SOF_ES8336_JD_INVERTED) dev_info(dev, "quirk JD inverted enabled\n"); } @@ -88,7 +88,7 @@ static int sof_es8316_speaker_power_event(struct snd_soc_dapm_widget *w, else priv->speaker_en = true; - gpiod_set_value_cansleep(priv->gpio_pa, priv->speaker_en); + gpiod_set_value_cansleep(priv->gpio_speakers, priv->speaker_en); return 0; } @@ -233,8 +233,8 @@ static int sof_es8336_quirk_cb(const struct dmi_system_id *id) { quirk = (unsigned long)id->driver_data; - if (quirk & SOF_ES8336_TGL_GPIO_QUIRK) - gpio_mapping = quirk_acpi_es8336_gpios; + if (quirk & SOF_ES8336_SPEAKERS_EN_GPIO1_QUIRK) + gpio_mapping = acpi_speakers_enable_gpio1; return 1; } @@ -257,7 +257,7 @@ static const struct dmi_system_id sof_es8336_quirk_table[] = { DMI_MATCH(DMI_SYS_VENDOR, "IP3 tech"), DMI_MATCH(DMI_BOARD_NAME, "WN1"), }, - .driver_data = (void *)(SOF_ES8336_TGL_GPIO_QUIRK) + .driver_data = (void *)(SOF_ES8336_SPEAKERS_EN_GPIO1_QUIRK) }, {} }; @@ -585,10 +585,10 @@ static int sof_es8336_probe(struct platform_device *pdev) if (ret) dev_warn(codec_dev, "unable to add GPIO mapping table\n"); - priv->gpio_pa = gpiod_get_optional(codec_dev, "pa-enable", GPIOD_OUT_LOW); - if (IS_ERR(priv->gpio_pa)) { - ret = dev_err_probe(dev, PTR_ERR(priv->gpio_pa), - "could not get pa-enable GPIO\n"); + priv->gpio_speakers = gpiod_get_optional(codec_dev, "speakers-enable", GPIOD_OUT_LOW); + if (IS_ERR(priv->gpio_speakers)) { + ret = dev_err_probe(dev, PTR_ERR(priv->gpio_speakers), + "could not get speakers-enable GPIO\n"); goto err_put_codec; } @@ -604,7 +604,7 @@ static int sof_es8336_probe(struct platform_device *pdev) ret = devm_snd_soc_register_card(dev, card); if (ret) { - gpiod_put(priv->gpio_pa); + gpiod_put(priv->gpio_speakers); dev_err(dev, "snd_soc_register_card failed: %d\n", ret); goto err_put_codec; } @@ -622,7 +622,7 @@ static int sof_es8336_remove(struct platform_device *pdev) struct snd_soc_card *card = platform_get_drvdata(pdev); struct sof_es8336_private *priv = snd_soc_card_get_drvdata(card); - gpiod_put(priv->gpio_pa); + gpiod_put(priv->gpio_speakers); device_remove_software_node(priv->codec_dev); put_device(priv->codec_dev); -- cgit v1.2.3 From 6e1ff1459e0086312e61c2d1ff8b74395a082fcb Mon Sep 17 00:00:00 2001 From: Mauro Carvalho Chehab Date: Thu, 7 Apr 2022 20:49:57 +0200 Subject: ASoC: Intel: sof_es8336: support a separate gpio to control headphone Some devices may use both gpio0 and gpio1 to independently switch the speaker and the headphone. Add support for that. Acked-by: Hans de Goede Signed-off-by: Mauro Carvalho Chehab Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/535454c0c598a8454487fe29b164527370e2db81.1649357263.git.mchehab@kernel.org Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_es8336.c | 59 ++++++++++++++++++++++++++++++------- 1 file changed, 49 insertions(+), 10 deletions(-) diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c index e4829a376b79..d15a58666cc6 100644 --- a/sound/soc/intel/boards/sof_es8336.c +++ b/sound/soc/intel/boards/sof_es8336.c @@ -30,6 +30,7 @@ #define SOF_ES8336_SPEAKERS_EN_GPIO1_QUIRK BIT(4) #define SOF_ES8336_ENABLE_DMIC BIT(5) #define SOF_ES8336_JD_INVERTED BIT(6) +#define SOF_ES8336_HEADPHONE_GPIO BIT(7) static unsigned long quirk; @@ -39,7 +40,7 @@ MODULE_PARM_DESC(quirk, "Board-specific quirk override"); struct sof_es8336_private { struct device *codec_dev; - struct gpio_desc *gpio_speakers; + struct gpio_desc *gpio_speakers, *gpio_headphone; struct snd_soc_jack jack; struct list_head hdmi_pcm_list; bool speaker_en; @@ -51,15 +52,27 @@ struct sof_hdmi_pcm { int device; }; -static const struct acpi_gpio_params speakers_enable_gpio0 = { 0, 0, true }; +static const struct acpi_gpio_params enable_gpio0 = { 0, 0, true }; +static const struct acpi_gpio_params enable_gpio1 = { 1, 0, true }; + static const struct acpi_gpio_mapping acpi_speakers_enable_gpio0[] = { - { "speakers-enable-gpios", &speakers_enable_gpio0, 1 }, + { "speakers-enable-gpios", &enable_gpio0, 1 }, { } }; -static const struct acpi_gpio_params speakers_enable_gpio1 = { 1, 0, true }; static const struct acpi_gpio_mapping acpi_speakers_enable_gpio1[] = { - { "speakers-enable-gpios", &speakers_enable_gpio1, 1 }, + { "speakers-enable-gpios", &enable_gpio1, 1 }, +}; + +static const struct acpi_gpio_mapping acpi_enable_both_gpios[] = { + { "speakers-enable-gpios", &enable_gpio0, 1 }, + { "headphone-enable-gpios", &enable_gpio1, 1 }, + { } +}; + +static const struct acpi_gpio_mapping acpi_enable_both_gpios_rev_order[] = { + { "speakers-enable-gpios", &enable_gpio1, 1 }, + { "headphone-enable-gpios", &enable_gpio0, 1 }, { } }; @@ -73,6 +86,8 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk DMIC enabled\n"); if (quirk & SOF_ES8336_SPEAKERS_EN_GPIO1_QUIRK) dev_info(dev, "Speakers GPIO1 quirk enabled\n"); + if (quirk & SOF_ES8336_HEADPHONE_GPIO) + dev_info(dev, "quirk headphone GPIO enabled\n"); if (quirk & SOF_ES8336_JD_INVERTED) dev_info(dev, "quirk JD inverted enabled\n"); } @@ -83,13 +98,24 @@ static int sof_es8316_speaker_power_event(struct snd_soc_dapm_widget *w, struct snd_soc_card *card = w->dapm->card; struct sof_es8336_private *priv = snd_soc_card_get_drvdata(card); + if (priv->speaker_en == !SND_SOC_DAPM_EVENT_ON(event)) + return 0; + + priv->speaker_en = !SND_SOC_DAPM_EVENT_ON(event); + if (SND_SOC_DAPM_EVENT_ON(event)) - priv->speaker_en = false; - else - priv->speaker_en = true; + msleep(70); gpiod_set_value_cansleep(priv->gpio_speakers, priv->speaker_en); + if (!(quirk & SOF_ES8336_HEADPHONE_GPIO)) + return 0; + + if (SND_SOC_DAPM_EVENT_ON(event)) + msleep(70); + + gpiod_set_value_cansleep(priv->gpio_headphone, priv->speaker_en); + return 0; } @@ -114,7 +140,7 @@ static const struct snd_soc_dapm_route sof_es8316_audio_map[] = { /* * There is no separate speaker output instead the speakers are muxed to - * the HP outputs. The mux is controlled by the "Speaker Power" supply. + * the HP outputs. The mux is controlled Speaker and/or headphone switch. */ {"Speaker", NULL, "HPOL"}, {"Speaker", NULL, "HPOR"}, @@ -233,8 +259,14 @@ static int sof_es8336_quirk_cb(const struct dmi_system_id *id) { quirk = (unsigned long)id->driver_data; - if (quirk & SOF_ES8336_SPEAKERS_EN_GPIO1_QUIRK) + if (quirk & SOF_ES8336_HEADPHONE_GPIO) { + if (quirk & SOF_ES8336_SPEAKERS_EN_GPIO1_QUIRK) + gpio_mapping = acpi_enable_both_gpios; + else + gpio_mapping = acpi_enable_both_gpios_rev_order; + } else if (quirk & SOF_ES8336_SPEAKERS_EN_GPIO1_QUIRK) { gpio_mapping = acpi_speakers_enable_gpio1; + } return 1; } @@ -592,6 +624,13 @@ static int sof_es8336_probe(struct platform_device *pdev) goto err_put_codec; } + priv->gpio_headphone = gpiod_get_optional(codec_dev, "headphone-enable", GPIOD_OUT_LOW); + if (IS_ERR(priv->gpio_headphone)) { + ret = dev_err_probe(dev, PTR_ERR(priv->gpio_headphone), + "could not get headphone-enable GPIO\n"); + goto err_put_codec; + } + INIT_LIST_HEAD(&priv->hdmi_pcm_list); snd_soc_card_set_drvdata(card, priv); -- cgit v1.2.3 From 7c7bb2a059b226ebadb14ce07460f6357023d56c Mon Sep 17 00:00:00 2001 From: Mauro Carvalho Chehab Date: Thu, 7 Apr 2022 20:49:58 +0200 Subject: ASoC: Intel: sof_es8336: add a quirk for headset at mic1 port The headset/internal mic can either be routed as mic1/mic2 or vice-versa. By default, the driver assumes that the headset is mapped as mic2, but not all devices map this way. So, add a quirk to support changing it to mic1, using mic2 for the internal analog mic (if any). Signed-off-by: Mauro Carvalho Chehab Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/5d88fc29b79be7ab77dae391c8e5ee929fd36c27.1649357263.git.mchehab@kernel.org Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_es8336.c | 19 ++++++++++++++++--- 1 file changed, 16 insertions(+), 3 deletions(-) diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c index d15a58666cc6..c71842be9d59 100644 --- a/sound/soc/intel/boards/sof_es8336.c +++ b/sound/soc/intel/boards/sof_es8336.c @@ -31,6 +31,7 @@ #define SOF_ES8336_ENABLE_DMIC BIT(5) #define SOF_ES8336_JD_INVERTED BIT(6) #define SOF_ES8336_HEADPHONE_GPIO BIT(7) +#define SOC_ES8336_HEADSET_MIC1 BIT(8) static unsigned long quirk; @@ -90,6 +91,8 @@ static void log_quirks(struct device *dev) dev_info(dev, "quirk headphone GPIO enabled\n"); if (quirk & SOF_ES8336_JD_INVERTED) dev_info(dev, "quirk JD inverted enabled\n"); + if (quirk & SOC_ES8336_HEADSET_MIC1) + dev_info(dev, "quirk headset at mic1 port enabled\n"); } static int sof_es8316_speaker_power_event(struct snd_soc_dapm_widget *w, @@ -147,11 +150,16 @@ static const struct snd_soc_dapm_route sof_es8316_audio_map[] = { {"Speaker", NULL, "Speaker Power"}, }; -static const struct snd_soc_dapm_route sof_es8316_intmic_in1_map[] = { +static const struct snd_soc_dapm_route sof_es8316_headset_mic2_map[] = { {"MIC1", NULL, "Internal Mic"}, {"MIC2", NULL, "Headset Mic"}, }; +static const struct snd_soc_dapm_route sof_es8316_headset_mic1_map[] = { + {"MIC2", NULL, "Internal Mic"}, + {"MIC1", NULL, "Headset Mic"}, +}; + static const struct snd_soc_dapm_route dmic_map[] = { /* digital mics */ {"DMic", NULL, "SoC DMIC"}, @@ -225,8 +233,13 @@ static int sof_es8316_init(struct snd_soc_pcm_runtime *runtime) card->dapm.idle_bias_off = true; - custom_map = sof_es8316_intmic_in1_map; - num_routes = ARRAY_SIZE(sof_es8316_intmic_in1_map); + if (quirk & SOC_ES8336_HEADSET_MIC1) { + custom_map = sof_es8316_headset_mic1_map; + num_routes = ARRAY_SIZE(sof_es8316_headset_mic1_map); + } else { + custom_map = sof_es8316_headset_mic2_map; + num_routes = ARRAY_SIZE(sof_es8316_headset_mic2_map); + } ret = snd_soc_dapm_add_routes(&card->dapm, custom_map, num_routes); if (ret) -- cgit v1.2.3 From c7cb4717f641db68e8117635bfcf62a9c27dc8d3 Mon Sep 17 00:00:00 2001 From: Mauro Carvalho Chehab Date: Thu, 7 Apr 2022 20:49:59 +0200 Subject: ASoC: Intel: sof_es8336: Add a quirk for Huawei Matebook D15 Based on experimental tests, Huawei Matebook D15 actually uses both gpio0 and gpio1: the first one controls the speaker, while the other one controls the headphone. Also, the headset is mapped as MIC1, instead of MIC2. So, add a quirk for it. Signed-off-by: Mauro Carvalho Chehab Acked-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/d678aef9fc9a07aced611aa7cb8c9b800c649e5a.1649357263.git.mchehab@kernel.org Signed-off-by: Mark Brown --- sound/soc/intel/boards/sof_es8336.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/intel/boards/sof_es8336.c b/sound/soc/intel/boards/sof_es8336.c index c71842be9d59..9d617831dd20 100644 --- a/sound/soc/intel/boards/sof_es8336.c +++ b/sound/soc/intel/boards/sof_es8336.c @@ -304,6 +304,15 @@ static const struct dmi_system_id sof_es8336_quirk_table[] = { }, .driver_data = (void *)(SOF_ES8336_SPEAKERS_EN_GPIO1_QUIRK) }, + { + .callback = sof_es8336_quirk_cb, + .matches = { + DMI_MATCH(DMI_SYS_VENDOR, "HUAWEI"), + DMI_MATCH(DMI_BOARD_NAME, "BOHB-WAX9-PCB-B2"), + }, + .driver_data = (void *)(SOF_ES8336_HEADPHONE_GPIO | + SOC_ES8336_HEADSET_MIC1) + }, {} }; -- cgit v1.2.3 From ddfd534528146660de75ee84d6db10f10e778f95 Mon Sep 17 00:00:00 2001 From: Christophe JAILLET Date: Sun, 3 Apr 2022 08:58:27 +0200 Subject: ASoC: codecs: Fix an error handling path in (rx|tx|va)_macro_probe() After a successful lpass_macro_pds_init() call, lpass_macro_pds_exit() must be called. Add the missing call in the error handling path of the probe function and use it. Fixes: 9e3d83c52844 ("ASoC: codecs: Add power domains support in digital macro codecs") Signed-off-by: Christophe JAILLET Link: https://lore.kernel.org/r/5b5a015a9b1dc8011c6a4053fa49da1f2531e47c.1648969065.git.christophe.jaillet@wanadoo.fr Signed-off-by: Mark Brown --- sound/soc/codecs/lpass-rx-macro.c | 14 ++++++++++---- sound/soc/codecs/lpass-tx-macro.c | 14 ++++++++++---- sound/soc/codecs/lpass-va-macro.c | 8 ++++++-- 3 files changed, 26 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/lpass-rx-macro.c b/sound/soc/codecs/lpass-rx-macro.c index 6884ae505e33..3143f9cd7277 100644 --- a/sound/soc/codecs/lpass-rx-macro.c +++ b/sound/soc/codecs/lpass-rx-macro.c @@ -3566,12 +3566,16 @@ static int rx_macro_probe(struct platform_device *pdev) return PTR_ERR(rx->pds); base = devm_platform_ioremap_resource(pdev, 0); - if (IS_ERR(base)) - return PTR_ERR(base); + if (IS_ERR(base)) { + ret = PTR_ERR(base); + goto err; + } rx->regmap = devm_regmap_init_mmio(dev, base, &rx_regmap_config); - if (IS_ERR(rx->regmap)) - return PTR_ERR(rx->regmap); + if (IS_ERR(rx->regmap)) { + ret = PTR_ERR(rx->regmap); + goto err; + } dev_set_drvdata(dev, rx); @@ -3632,6 +3636,8 @@ err_mclk: err_dcodec: clk_disable_unprepare(rx->macro); err: + lpass_macro_pds_exit(rx->pds); + return ret; } diff --git a/sound/soc/codecs/lpass-tx-macro.c b/sound/soc/codecs/lpass-tx-macro.c index 714a411d5337..55503ba480bb 100644 --- a/sound/soc/codecs/lpass-tx-macro.c +++ b/sound/soc/codecs/lpass-tx-macro.c @@ -1828,8 +1828,10 @@ static int tx_macro_probe(struct platform_device *pdev) return PTR_ERR(tx->pds); base = devm_platform_ioremap_resource(pdev, 0); - if (IS_ERR(base)) - return PTR_ERR(base); + if (IS_ERR(base)) { + ret = PTR_ERR(base); + goto err; + } /* Update defaults for lpass sc7280 */ if (of_device_is_compatible(np, "qcom,sc7280-lpass-tx-macro")) { @@ -1846,8 +1848,10 @@ static int tx_macro_probe(struct platform_device *pdev) } tx->regmap = devm_regmap_init_mmio(dev, base, &tx_regmap_config); - if (IS_ERR(tx->regmap)) - return PTR_ERR(tx->regmap); + if (IS_ERR(tx->regmap)) { + ret = PTR_ERR(tx->regmap); + goto err; + } dev_set_drvdata(dev, tx); @@ -1907,6 +1911,8 @@ err_mclk: err_dcodec: clk_disable_unprepare(tx->macro); err: + lpass_macro_pds_exit(tx->pds); + return ret; } diff --git a/sound/soc/codecs/lpass-va-macro.c b/sound/soc/codecs/lpass-va-macro.c index f3cb596058e0..d18b56e60433 100644 --- a/sound/soc/codecs/lpass-va-macro.c +++ b/sound/soc/codecs/lpass-va-macro.c @@ -1434,8 +1434,10 @@ static int va_macro_probe(struct platform_device *pdev) va->dmic_clk_div = VA_MACRO_CLK_DIV_2; } else { ret = va_macro_validate_dmic_sample_rate(sample_rate, va); - if (!ret) - return -EINVAL; + if (!ret) { + ret = -EINVAL; + goto err; + } } base = devm_platform_ioremap_resource(pdev, 0); @@ -1492,6 +1494,8 @@ err_mclk: err_dcodec: clk_disable_unprepare(va->macro); err: + lpass_macro_pds_exit(va->pds); + return ret; } -- cgit v1.2.3 From 5b933c7262c5b0ea11ea3c3b3ea81add04895954 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald Date: Tue, 12 Apr 2022 17:39:27 +0100 Subject: firmware: cs_dsp: Fix overrun of unterminated control name string For wmfw format v2 and later the coefficient name strings have a length field and are NOT null-terminated. Use kasprintf() to convert the unterminated string into a null-terminated string in an allocated buffer. The previous code handled this duplication incorrectly using kmemdup() and getting the length from a strlen() of the (unterminated) source string. This resulted in creating a string that continued up to the next byte in the firmware file that just happened to be 0x00. Signed-off-by: Richard Fitzgerald Fixes: f6bc909e7673 ("firmware: cs_dsp: add driver to support firmware loading on Cirrus Logic DSPs") Link: https://lore.kernel.org/r/20220412163927.1303470-1-rf@opensource.cirrus.com Signed-off-by: Mark Brown --- drivers/firmware/cirrus/cs_dsp.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/drivers/firmware/cirrus/cs_dsp.c b/drivers/firmware/cirrus/cs_dsp.c index e48108e694f8..7dad6f57d970 100644 --- a/drivers/firmware/cirrus/cs_dsp.c +++ b/drivers/firmware/cirrus/cs_dsp.c @@ -955,8 +955,7 @@ static int cs_dsp_create_control(struct cs_dsp *dsp, ctl->alg_region = *alg_region; if (subname && dsp->fw_ver >= 2) { ctl->subname_len = subname_len; - ctl->subname = kmemdup(subname, - strlen(subname) + 1, GFP_KERNEL); + ctl->subname = kasprintf(GFP_KERNEL, "%.*s", subname_len, subname); if (!ctl->subname) { ret = -ENOMEM; goto err_ctl; -- cgit v1.2.3 From 357ad4d898286b94aaae0cb7e3f573459e5b98b9 Mon Sep 17 00:00:00 2001 From: Miles Chen Date: Thu, 14 Apr 2022 17:19:38 +0800 Subject: sound/oss/dmasound: fix 'dmasound_setup' defined but not used We observed: 'dmasound_setup' defined but not used error with COMPILER=gcc ARCH=m68k DEFCONFIG=allmodconfig build. Fix it by adding __maybe_unused to dmasound_setup. Error(s): sound/oss/dmasound/dmasound_core.c:1431:12: error: 'dmasound_setup' defined but not used [-Werror=unused-function] Fixes: 9dd7c46346ca ("sound/oss/dmasound: fix build when drivers are mixed =y/=m") Signed-off-by: Miles Chen Acked-by: Randy Dunlap Link: https://lore.kernel.org/r/20220414091940.2216-1-miles.chen@mediatek.com Signed-off-by: Takashi Iwai --- sound/oss/dmasound/dmasound_core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/oss/dmasound/dmasound_core.c b/sound/oss/dmasound/dmasound_core.c index 9c48f3a9e3d1..164335d3c200 100644 --- a/sound/oss/dmasound/dmasound_core.c +++ b/sound/oss/dmasound/dmasound_core.c @@ -1428,7 +1428,7 @@ void dmasound_deinit(void) unregister_sound_dsp(sq_unit); } -static int dmasound_setup(char *str) +static int __maybe_unused dmasound_setup(char *str) { int ints[6], size; -- cgit v1.2.3 From c74193787b2f683751a67603fb5f15c7584f355f Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Thu, 14 Apr 2022 18:05:16 +0300 Subject: ALSA: hda/hdmi: fix warning about PCM count when used with SOF With commit 13046370c4d1 ("ALSA: hda/hdmi: let new platforms assign the pcm slot dynamically"), old behaviour to consider the HDA pin number, when choosing PCM to assign, was dropped. Build on this change and limit the number of PCMs created to number of converters (= maximum number of concurrent display/receivers) when "mst_no_extra_pcms" and "dyn_pcm_no_legacy" quirks are both set. Fix the check in hdmi_find_pcm_slot() to ensure only spec->pcm_used entries are considered in the search. Elsewhere in the driver spec->pcm_used is already checked properly. Doing this avoids following warning at SOF driver probe for multiple machine drivers: [ 112.425297] sof_sdw sof_sdw: hda_dsp_hdmi_build_controls: no PCM in topology for HDMI converter 4 [ 112.425298] sof_sdw sof_sdw: hda_dsp_hdmi_build_controls: no PCM in topology for HDMI converter 5 [ 112.425299] sof_sdw sof_sdw: hda_dsp_hdmi_build_controls: no PCM in topology for HDMI converter 6 Fixes: 13046370c4d1 ("ALSA: hda/hdmi: let new platforms assign the pcm slot dynamically") BugLink: https://github.com/thesofproject/linux/issues/2573 Signed-off-by: Kai Vehmanen Link: https://lore.kernel.org/r/20220414150516.3638283-1-kai.vehmanen@linux.intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 3e086eebf88d..f9d67058d69d 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -1395,7 +1395,7 @@ static int hdmi_find_pcm_slot(struct hdmi_spec *spec, last_try: /* the last try; check the empty slots in pins */ - for (i = 0; i < spec->num_nids; i++) { + for (i = 0; i < spec->pcm_used; i++) { if (!test_bit(i, &spec->pcm_bitmap)) return i; } @@ -2325,7 +2325,9 @@ static int generic_hdmi_build_pcms(struct hda_codec *codec) * dev_num is the device entry number in a pin */ - if (codec->mst_no_extra_pcms) + if (spec->dyn_pcm_no_legacy && codec->mst_no_extra_pcms) + pcm_num = spec->num_cvts; + else if (codec->mst_no_extra_pcms) pcm_num = spec->num_nids; else pcm_num = spec->num_nids + spec->dev_num - 1; -- cgit v1.2.3 From 6624fb41f5126c7205e866e58d4aaae0453f0914 Mon Sep 17 00:00:00 2001 From: Kai Vehmanen Date: Thu, 14 Apr 2022 19:01:29 +0300 Subject: ALSA: hda/hdmi: add HDMI codec VID for Raptorlake-P Add HDMI codec VID for Intel Raptorlake-P platform. Signed-off-by: Kai Vehmanen Link: https://lore.kernel.org/r/20220414160129.3641411-1-kai.vehmanen@linux.intel.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_hdmi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index f9d67058d69d..31fe41795571 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -4553,6 +4553,7 @@ HDA_CODEC_ENTRY(0x80862819, "DG2 HDMI", patch_i915_adlp_hdmi), HDA_CODEC_ENTRY(0x8086281a, "Jasperlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281b, "Elkhartlake HDMI", patch_i915_icl_hdmi), HDA_CODEC_ENTRY(0x8086281c, "Alderlake-P HDMI", patch_i915_adlp_hdmi), +HDA_CODEC_ENTRY(0x8086281f, "Raptorlake-P HDMI", patch_i915_adlp_hdmi), HDA_CODEC_ENTRY(0x80862880, "CedarTrail HDMI", patch_generic_hdmi), HDA_CODEC_ENTRY(0x80862882, "Valleyview2 HDMI", patch_i915_byt_hdmi), HDA_CODEC_ENTRY(0x80862883, "Braswell HDMI", patch_i915_byt_hdmi), -- cgit v1.2.3 From 00fd7cfad0548b6b7234c93370076f9b9c2e39f8 Mon Sep 17 00:00:00 2001 From: Lucas De Marchi Date: Fri, 15 Apr 2022 23:44:18 -0700 Subject: ALSA: hda/i915: Fix one too many pci_dev_put() pci_get_class() will already unref the pci device passed as argument. So if it's unconditionally unref'ed, even if the loop is not stopped, there will be one too many unref for each device not matched. Closes: https://gitlab.freedesktop.org/drm/intel/-/issues/5701 Fixes: c9db8a30d9f0 ("ALSA: hda/i915 - skip acomp init if no matching display") Signed-off-by: Lucas De Marchi Reviewed-by: Kai Vehmanen Link: https://lore.kernel.org/r/20220416064418.2364582-1-lucas.demarchi@intel.com Signed-off-by: Takashi Iwai --- sound/hda/hdac_i915.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/hda/hdac_i915.c b/sound/hda/hdac_i915.c index 48b8ed752b69..3f35972e1cf7 100644 --- a/sound/hda/hdac_i915.c +++ b/sound/hda/hdac_i915.c @@ -127,11 +127,10 @@ static int i915_gfx_present(struct pci_dev *hdac_pci) display_dev = pci_get_class(class, display_dev); if (display_dev && display_dev->vendor == PCI_VENDOR_ID_INTEL && - connectivity_check(display_dev, hdac_pci)) + connectivity_check(display_dev, hdac_pci)) { + pci_dev_put(display_dev); match = true; - - pci_dev_put(display_dev); - + } } while (!match && display_dev); return match; -- cgit v1.2.3 From 4ddef9c4d70aae0c9029bdec7c3f7f1c1c51ff8c Mon Sep 17 00:00:00 2001 From: Maurizio Avogadro Date: Mon, 18 Apr 2022 15:16:12 +0200 Subject: ALSA: usb-audio: add mapping for MSI MAG X570S Torpedo MAX. The USB audio device 0db0:a073 based on the Realtek ALC4080 chipset exposes all playback volume controls as "PCM". This makes distinguishing the individual functions hard. The mapping already adopted for device 0db0:419c based on the same chipset fixes the issue, apply it for this device too. Signed-off-by: Maurizio Avogadro Cc: Link: https://lore.kernel.org/r/Yl1ykPaGgsFf3SnW@ryzen Signed-off-by: Takashi Iwai --- sound/usb/mixer_maps.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index 64f5544d0a0a..7ef7a8abcc2b 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -599,6 +599,10 @@ static const struct usbmix_ctl_map usbmix_ctl_maps[] = { .id = USB_ID(0x0db0, 0x419c), .map = msi_mpg_x570s_carbon_max_wifi_alc4080_map, }, + { /* MSI MAG X570S Torpedo Max */ + .id = USB_ID(0x0db0, 0xa073), + .map = msi_mpg_x570s_carbon_max_wifi_alc4080_map, + }, { /* MSI TRX40 */ .id = USB_ID(0x0db0, 0x543d), .map = trx40_mobo_map, -- cgit v1.2.3 From 0665886ad1392e6b5bae85d7a6ccbed48dca1522 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Wed, 20 Apr 2022 15:02:47 +0200 Subject: ALSA: usb-audio: Clear MIDI port active flag after draining When a rawmidi output stream is closed, it calls the drain at first, then does trigger-off only when the drain returns -ERESTARTSYS as a fallback. It implies that each driver should turn off the stream properly after the drain. Meanwhile, USB-audio MIDI interface didn't change the port->active flag after the drain. This may leave the output work picking up the port that is closed right now, which eventually leads to a use-after-free for the already released rawmidi object. This patch fixes the bug by properly clearing the port->active flag after the output drain. Reported-by: syzbot+70e777a39907d6d5fd0a@syzkaller.appspotmail.com Cc: Link: https://lore.kernel.org/r/00000000000011555605dceaff03@google.com Link: https://lore.kernel.org/r/20220420130247.22062-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/midi.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/midi.c b/sound/usb/midi.c index 2c01649c70f6..7c6ca2b433a5 100644 --- a/sound/usb/midi.c +++ b/sound/usb/midi.c @@ -1194,6 +1194,7 @@ static void snd_usbmidi_output_drain(struct snd_rawmidi_substream *substream) } while (drain_urbs && timeout); finish_wait(&ep->drain_wait, &wait); } + port->active = 0; spin_unlock_irq(&ep->buffer_lock); } -- cgit v1.2.3 From b3fbe53610b5ed8f0370ec4c7e6c8a1f261ddf70 Mon Sep 17 00:00:00 2001 From: Andy Chi Date: Thu, 21 Apr 2022 14:36:04 +0800 Subject: ALSA: hda/realtek: Enable mute/micmute LEDs and limit mic boost on EliteBook 845/865 G9 On HP EliteBook 845 G9 and EliteBook 865 G9, the audio LEDs can be enabled by ALC285_FIXUP_HP_MUTE_LED. So use it accordingly. Signed-off-by: Andy Chi Fixes: 07bcab93946c ("ALSA: hda/realtek: Add support for HP Laptops") Link: https://lore.kernel.org/r/20220421063606.39772-1-andy.chi@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 11 +++++++++-- 1 file changed, 9 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 62fbf3772b41..0cba2f19a772 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7006,6 +7006,7 @@ enum { ALC285_FIXUP_LEGION_Y9000X_AUTOMUTE, ALC287_FIXUP_LEGION_16ACHG6, ALC287_FIXUP_CS35L41_I2C_2, + ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED, ALC245_FIXUP_CS35L41_SPI_2, ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED, ALC245_FIXUP_CS35L41_SPI_4, @@ -8769,6 +8770,12 @@ static const struct hda_fixup alc269_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = cs35l41_fixup_i2c_two, }, + [ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED] = { + .type = HDA_FIXUP_FUNC, + .v.func = cs35l41_fixup_i2c_two, + .chained = true, + .chain_id = ALC285_FIXUP_HP_MUTE_LED, + }, [ALC245_FIXUP_CS35L41_SPI_2] = { .type = HDA_FIXUP_FUNC, .v.func = cs35l41_fixup_spi_two, @@ -9025,9 +9032,9 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8981, "HP Elite Dragonfly G3", ALC245_FIXUP_CS35L41_SPI_4), SND_PCI_QUIRK(0x103c, 0x898e, "HP EliteBook 835 G9", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x898f, "HP EliteBook 835 G9", ALC287_FIXUP_CS35L41_I2C_2), - SND_PCI_QUIRK(0x103c, 0x8991, "HP EliteBook 845 G9", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8991, "HP EliteBook 845 G9", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8992, "HP EliteBook 845 G9", ALC287_FIXUP_CS35L41_I2C_2), - SND_PCI_QUIRK(0x103c, 0x8994, "HP EliteBook 855 G9", ALC287_FIXUP_CS35L41_I2C_2), + SND_PCI_QUIRK(0x103c, 0x8994, "HP EliteBook 855 G9", ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x8995, "HP EliteBook 855 G9", ALC287_FIXUP_CS35L41_I2C_2), SND_PCI_QUIRK(0x103c, 0x89a4, "HP ProBook 440 G9", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89a6, "HP ProBook 450 G9", ALC236_FIXUP_HP_GPIO_LED), -- cgit v1.2.3 From b07908ab26ceab51165c13714277c19252e62594 Mon Sep 17 00:00:00 2001 From: Gongjun Song Date: Thu, 21 Apr 2022 11:35:46 -0500 Subject: ALSA: hda: intel-dsp-config: Add RaptorLake PCI IDs Add RaptorLake-P PCI IDs Reviewed-by: Kai Vehmanen Signed-off-by: Gongjun Song Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220421163546.319604-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Takashi Iwai --- sound/hda/intel-dsp-config.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/hda/intel-dsp-config.c b/sound/hda/intel-dsp-config.c index 8b0a16ba27d3..a8fe01764b25 100644 --- a/sound/hda/intel-dsp-config.c +++ b/sound/hda/intel-dsp-config.c @@ -424,6 +424,15 @@ static const struct config_entry config_table[] = { .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, .device = 0x54c8, }, + /* RaptorLake-P */ + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, + .device = 0x51ca, + }, + { + .flags = FLAG_SOF | FLAG_SOF_ONLY_IF_DMIC_OR_SOUNDWIRE, + .device = 0x51cb, + }, #endif }; -- cgit v1.2.3 From 86222af07abf1f5f07a5873cc399c29ab8a9b8b8 Mon Sep 17 00:00:00 2001 From: Tim Crawford Date: Thu, 21 Apr 2022 11:04:12 -0600 Subject: ALSA: hda/realtek: Add quirk for Clevo NP70PNP Fixes headset detection on Clevo NP70PNP. Signed-off-by: Tim Crawford Cc: Link: https://lore.kernel.org/r/20220421170412.3697-1-tcrawford@system76.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 0cba2f19a772..4c0c593f3c0a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9170,6 +9170,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1558, 0x8562, "Clevo NH[57][0-9]RZ[Q]", ALC269_FIXUP_DMIC), SND_PCI_QUIRK(0x1558, 0x8668, "Clevo NP50B[BE]", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x866d, "Clevo NP5[05]PN[HJK]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), + SND_PCI_QUIRK(0x1558, 0x867c, "Clevo NP7[01]PNP", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x867d, "Clevo NP7[01]PN[HJK]", ALC256_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8680, "Clevo NJ50LU", ALC293_FIXUP_SYSTEM76_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1558, 0x8686, "Clevo NH50[CZ]U", ALC256_FIXUP_MIC_NO_PRESENCE_AND_RESUME), -- cgit v1.2.3