From 353bb6a5f2ac495f289b7c7a528c7d134c9a8ec4 Mon Sep 17 00:00:00 2001 From: Simon Trimmer Date: Wed, 9 Mar 2022 13:00:17 +0000 Subject: ASoC: wm_adsp: Compressed stream DSP memory structs should be __packed The compressed stream code has a bunch of structs that are used to represent DSP memory but have not been marked __packed. This isn't safe, they could get padded on a 64-bit build. Signed-off-by: Simon Trimmer Link: https://lore.kernel.org/r/20220309130017.2816-1-simont@opensource.cirrus.com Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm_adsp.c | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 0582585236a2..5a57bb04a0ae 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -97,13 +97,13 @@ struct wm_adsp_system_config_xm_hdr { __be32 wdma[8]; __be32 build_job_name[3]; __be32 build_job_number; -}; +} __packed; struct wm_halo_system_config_xm_hdr { __be32 halo_heartbeat; __be32 build_job_name[3]; __be32 build_job_number; -}; +} __packed; struct wm_adsp_alg_xm_struct { __be32 magic; @@ -114,13 +114,13 @@ struct wm_adsp_alg_xm_struct { __be32 high_water_mark; __be32 low_water_mark; __be64 smoothed_power; -}; +} __packed; struct wm_adsp_host_buf_coeff_v1 { __be32 host_buf_ptr; /* Host buffer pointer */ __be32 versions; /* Version numbers */ __be32 name[4]; /* The buffer name */ -}; +} __packed; struct wm_adsp_buffer { __be32 buf1_base; /* Base addr of first buffer area */ @@ -141,7 +141,7 @@ struct wm_adsp_buffer { __be32 min_free; /* min free space since stream start */ __be32 blocks_written[2]; /* total blocks written (64 bit) */ __be32 words_written[2]; /* total words written (64 bit) */ -}; +} __packed; struct wm_adsp_compr; -- cgit 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 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 From b4f5c6b2e52b27462c0599e64e96e53b58438de1 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 16 Apr 2022 13:54:08 +0100 Subject: ASoC: wm8958: Fix change notifications for DSP controls The WM8958 DSP controls all return 0 on successful write, not a boolean value indicating if the write changed the value of the control. Fix this by returning 1 after a change, there is already a check at the start of each put() that skips the function in the case that there is no change. Signed-off-by: Mark Brown Acked-by: Charles Keepax Link: https://lore.kernel.org/r/20220416125408.197440-1-broonie@kernel.org Cc: stable@vger.kernel.org --- sound/soc/codecs/wm8958-dsp2.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/wm8958-dsp2.c b/sound/soc/codecs/wm8958-dsp2.c index e4018ba3b19a..7878c7a58ff1 100644 --- a/sound/soc/codecs/wm8958-dsp2.c +++ b/sound/soc/codecs/wm8958-dsp2.c @@ -530,7 +530,7 @@ static int wm8958_mbc_put(struct snd_kcontrol *kcontrol, wm8958_dsp_apply(component, mbc, wm8994->mbc_ena[mbc]); - return 0; + return 1; } #define WM8958_MBC_SWITCH(xname, xval) {\ @@ -656,7 +656,7 @@ static int wm8958_vss_put(struct snd_kcontrol *kcontrol, wm8958_dsp_apply(component, vss, wm8994->vss_ena[vss]); - return 0; + return 1; } @@ -730,7 +730,7 @@ static int wm8958_hpf_put(struct snd_kcontrol *kcontrol, wm8958_dsp_apply(component, hpf % 3, ucontrol->value.integer.value[0]); - return 0; + return 1; } #define WM8958_HPF_SWITCH(xname, xval) {\ @@ -824,7 +824,7 @@ static int wm8958_enh_eq_put(struct snd_kcontrol *kcontrol, wm8958_dsp_apply(component, eq, ucontrol->value.integer.value[0]); - return 0; + return 1; } #define WM8958_ENH_EQ_SWITCH(xname, xval) {\ -- cgit 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 From 3756aa16fadaef2873cfbd2659dfa1978a7e1859 Mon Sep 17 00:00:00 2001 From: Olivier Moysan Date: Tue, 12 Apr 2022 13:16:58 +0200 Subject: ASoC: simple-card-utils: fix sysclk shutdown In asoc_simple_shutdown() the snd_soc_dai_set_sysclk() function is called twice with input direction SND_SOC_CLOCK_IN. Restore one call with output direction SND_SOC_CLOCK_OUT. Fixes: 5ca2ab459817 ("ASoC: simple-card-utils: Add new system-clock-fixed flag") Signed-off-by: Olivier Moysan Link: https://lore.kernel.org/r/20220412111658.11015-1-olivier.moysan@foss.st.com Signed-off-by: Mark Brown --- sound/soc/generic/simple-card-utils.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index f2157944247f..da0c27828ce6 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -322,7 +322,7 @@ void asoc_simple_shutdown(struct snd_pcm_substream *substream) if (props->mclk_fs && !dai->clk_fixed && !snd_soc_dai_active(cpu_dai)) snd_soc_dai_set_sysclk(cpu_dai, - 0, 0, SND_SOC_CLOCK_IN); + 0, 0, SND_SOC_CLOCK_OUT); asoc_simple_clk_disable(dai); } -- cgit From 08ef48404965cfef99343d6bbbcf75b88c74aa0e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 20 Apr 2022 14:34:37 +0100 Subject: ASoC: da7219: Fix change notifications for tone generator frequency The tone generator frequency control just returns 0 on successful write, not a boolean value indicating if there was a change or not. Compare what was written with the value that was there previously so that notifications are generated appropriately when the value changes. Signed-off-by: Mark Brown Reviewed-by: Adam Thomson Cc: stable@vger.kernel.org Link: https://lore.kernel.org/r/20220420133437.569229-1-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/da7219.c | 14 ++++++++++---- 1 file changed, 10 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 13009d08b09a..c7493549a9a5 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -446,7 +446,7 @@ static int da7219_tonegen_freq_put(struct snd_kcontrol *kcontrol, struct soc_mixer_control *mixer_ctrl = (struct soc_mixer_control *) kcontrol->private_value; unsigned int reg = mixer_ctrl->reg; - __le16 val; + __le16 val_new, val_old; int ret; /* @@ -454,13 +454,19 @@ static int da7219_tonegen_freq_put(struct snd_kcontrol *kcontrol, * Therefore we need to convert to little endian here to align with * HW registers. */ - val = cpu_to_le16(ucontrol->value.integer.value[0]); + val_new = cpu_to_le16(ucontrol->value.integer.value[0]); mutex_lock(&da7219->ctrl_lock); - ret = regmap_raw_write(da7219->regmap, reg, &val, sizeof(val)); + ret = regmap_raw_read(da7219->regmap, reg, &val_old, sizeof(val_old)); + if (ret == 0 && (val_old != val_new)) + ret = regmap_raw_write(da7219->regmap, reg, + &val_new, sizeof(val_new)); mutex_unlock(&da7219->ctrl_lock); - return ret; + if (ret < 0) + return ret; + + return val_old != val_new; } -- cgit 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 From 2e3a0d1bfa95b54333f7add3e50e288769373873 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 21 Apr 2022 13:38:01 +0100 Subject: ASoC: meson: Fix event generation for AUI ACODEC mux The AIU ACODEC has a custom put() operation which returns 0 when the value of the mux changes, meaning that events are not generated for userspace. Change to return 1 in this case, the function returns early in the case where there is no change. Signed-off-by: Mark Brown Reviewed-by: Jerome Brunet Link: https://lore.kernel.org/r/20220421123803.292063-2-broonie@kernel.org Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/meson/aiu-acodec-ctrl.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/meson/aiu-acodec-ctrl.c b/sound/soc/meson/aiu-acodec-ctrl.c index 22e181646bc3..3776b073a3db 100644 --- a/sound/soc/meson/aiu-acodec-ctrl.c +++ b/sound/soc/meson/aiu-acodec-ctrl.c @@ -58,7 +58,7 @@ static int aiu_acodec_ctrl_mux_put_enum(struct snd_kcontrol *kcontrol, snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL); - return 0; + return 1; } static SOC_ENUM_SINGLE_DECL(aiu_acodec_ctrl_mux_enum, AIU_ACODEC_CTRL, -- cgit From fce49921a22262736cdc3cc74fa67915b75e9363 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 21 Apr 2022 13:38:02 +0100 Subject: ASoC: meson: Fix event generation for AUI CODEC mux The AIU CODEC has a custom put() operation which returns 0 when the value of the mux changes, meaning that events are not generated for userspace. Change to return 1 in this case, the function returns early in the case where there is no change. Signed-off-by: Mark Brown Reviewed-by: Jerome Brunet Link: https://lore.kernel.org/r/20220421123803.292063-3-broonie@kernel.org Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/meson/aiu-codec-ctrl.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/meson/aiu-codec-ctrl.c b/sound/soc/meson/aiu-codec-ctrl.c index 59ee66fc2bcd..286ac4983d40 100644 --- a/sound/soc/meson/aiu-codec-ctrl.c +++ b/sound/soc/meson/aiu-codec-ctrl.c @@ -57,7 +57,7 @@ static int aiu_codec_ctrl_mux_put_enum(struct snd_kcontrol *kcontrol, snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL); - return 0; + return 1; } static SOC_ENUM_SINGLE_DECL(aiu_hdmi_ctrl_mux_enum, AIU_HDMI_CLK_DATA_CTRL, -- cgit From 12131008fc13ff7f7690d170b7a8f72d24fd7d1e Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 21 Apr 2022 13:38:03 +0100 Subject: ASoC: meson: Fix event generation for G12A tohdmi mux The G12A tohdmi has a custom put() operation which returns 0 when the value of the mux changes, meaning that events are not generated for userspace. Change to return 1 in this case, the function returns early in the case where there is no change. Signed-off-by: Mark Brown Reviewed-by: Jerome Brunet Link: https://lore.kernel.org/r/20220421123803.292063-4-broonie@kernel.org Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/meson/g12a-tohdmitx.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/meson/g12a-tohdmitx.c b/sound/soc/meson/g12a-tohdmitx.c index 9b2b59536ced..6c99052feafd 100644 --- a/sound/soc/meson/g12a-tohdmitx.c +++ b/sound/soc/meson/g12a-tohdmitx.c @@ -67,7 +67,7 @@ static int g12a_tohdmitx_i2s_mux_put_enum(struct snd_kcontrol *kcontrol, snd_soc_dapm_mux_update_power(dapm, kcontrol, mux, e, NULL); - return 0; + return 1; } static SOC_ENUM_SINGLE_DECL(g12a_tohdmitx_i2s_mux_enum, TOHDMITX_CTRL0, -- cgit From eb5773201b1c5d603424bd21f161c8c2d1075b42 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart Date: Thu, 21 Apr 2022 11:23:28 -0500 Subject: ASoC: soc-ops: fix error handling MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit cppcheck throws the following warning: sound/soc/soc-ops.c:461:8: style: Variable 'ret' is assigned a value that is never used. [unreadVariable] ret = err; ^ This seems to be a missing change in the return value. Fixes: 7f3d90a351968 ("ASoC: ops: Fix stereo change notifications in snd_soc_put_volsw_sx()") Signed-off-by: Pierre-Louis Bossart Reviewed-by: Bard Liao Reviewed-by: Rander Wang Reviewed-by: Péter Ujfalusi Link: https://lore.kernel.org/r/20220421162328.302017-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/soc-ops.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index a0ca58ba1627..58347eadd219 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -461,7 +461,7 @@ int snd_soc_put_volsw_sx(struct snd_kcontrol *kcontrol, ret = err; } } - return err; + return ret; } EXPORT_SYMBOL_GPL(snd_soc_put_volsw_sx); -- cgit From c26830b6c5c534d273ce007eb33d5a2d2ad4e969 Mon Sep 17 00:00:00 2001 From: Neil Armstrong Date: Thu, 21 Apr 2022 17:57:24 +0200 Subject: ASoC: meson: axg-tdm-interface: Fix formatters in trigger" This reverts commit bf5e4887eeddb48480568466536aa08ec7f179a5 because the following and required commit e138233e56e9829e65b6293887063a1a3ccb2d68 causes the following system crash when using audio: BUG: sleeping function called from invalid context at kernel/locking/mutex.c:282 Fixes: bf5e4887eeddb4848056846 ("ASoC: meson: axg-tdm-interface: manage formatters in trigger") Reported-by: Dmitry Shmidt Signed-off-by: Neil Armstrong Acked-by: Jerome Brunet Link: https://lore.kernel.org/r/20220421155725.2589089-1-narmstrong@baylibre.com Signed-off-by: Mark Brown --- sound/soc/meson/axg-tdm-interface.c | 26 +++++--------------------- 1 file changed, 5 insertions(+), 21 deletions(-) diff --git a/sound/soc/meson/axg-tdm-interface.c b/sound/soc/meson/axg-tdm-interface.c index 0c31934a9630..e076ced30025 100644 --- a/sound/soc/meson/axg-tdm-interface.c +++ b/sound/soc/meson/axg-tdm-interface.c @@ -351,29 +351,13 @@ static int axg_tdm_iface_hw_free(struct snd_pcm_substream *substream, return 0; } -static int axg_tdm_iface_trigger(struct snd_pcm_substream *substream, - int cmd, +static int axg_tdm_iface_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { - struct axg_tdm_stream *ts = - snd_soc_dai_get_dma_data(dai, substream); - - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - case SNDRV_PCM_TRIGGER_RESUME: - case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - axg_tdm_stream_start(ts); - break; - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - case SNDRV_PCM_TRIGGER_STOP: - axg_tdm_stream_stop(ts); - break; - default: - return -EINVAL; - } + struct axg_tdm_stream *ts = snd_soc_dai_get_dma_data(dai, substream); - return 0; + /* Force all attached formatters to update */ + return axg_tdm_stream_reset(ts); } static int axg_tdm_iface_remove_dai(struct snd_soc_dai *dai) @@ -413,8 +397,8 @@ static const struct snd_soc_dai_ops axg_tdm_iface_ops = { .set_fmt = axg_tdm_iface_set_fmt, .startup = axg_tdm_iface_startup, .hw_params = axg_tdm_iface_hw_params, + .prepare = axg_tdm_iface_prepare, .hw_free = axg_tdm_iface_hw_free, - .trigger = axg_tdm_iface_trigger, }; /* TDM Backend DAIs */ -- cgit From 0c9b152c72e53016e96593bdbb8cffe2176694b9 Mon Sep 17 00:00:00 2001 From: Neil Armstrong Date: Thu, 21 Apr 2022 17:57:25 +0200 Subject: ASoC: meson: axg-card: Fix nonatomic links This commit e138233e56e9829e65b6293887063a1a3ccb2d68 causes the following system crash when using audio on G12A/G12B & SM1 systems: BUG: sleeping function called from invalid context at kernel/locking/mutex.c:282 in_atomic(): 1, irqs_disabled(): 128, non_block: 0, pid: 0, name: swapper/0 preempt_count: 10001, expected: 0 RCU nest depth: 0, expected: 0 Preemption disabled at: schedule_preempt_disabled+0x20/0x2c mutex_lock+0x24/0x60 _snd_pcm_stream_lock_irqsave+0x20/0x3c snd_pcm_period_elapsed+0x24/0xa4 axg_fifo_pcm_irq_block+0x64/0xdc __handle_irq_event_percpu+0x104/0x264 handle_irq_event+0x48/0xb4 ... start_kernel+0x3f0/0x484 __primary_switched+0xc0/0xc8 Revert this commit until the crash is fixed. Fixes: e138233e56e9829e65b6 ("ASoC: meson: axg-card: make links nonatomic") Reported-by: Dmitry Shmidt Signed-off-by: Neil Armstrong Acked-by: Jerome Brunet Link: https://lore.kernel.org/r/20220421155725.2589089-2-narmstrong@baylibre.com Signed-off-by: Mark Brown --- sound/soc/meson/axg-card.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/meson/axg-card.c b/sound/soc/meson/axg-card.c index cbbaa55d92a6..2b77010c2c5c 100644 --- a/sound/soc/meson/axg-card.c +++ b/sound/soc/meson/axg-card.c @@ -320,7 +320,6 @@ static int axg_card_add_link(struct snd_soc_card *card, struct device_node *np, dai_link->cpus = cpu; dai_link->num_cpus = 1; - dai_link->nonatomic = true; ret = meson_card_parse_dai(card, np, &dai_link->cpus->of_node, &dai_link->cpus->dai_name); -- cgit 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 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 From a6ac60b36dade525c13c5bb0838589619533efb7 Mon Sep 17 00:00:00 2001 From: Hui Wang Date: Fri, 22 Apr 2022 15:39:37 +0800 Subject: ALSA: hda/realtek: Fix mute led issue on thinkpad with cs35l41 s-codec The quirk ALC287_FIXUP_CS35L41_I2C_2 needs to chain the quirk ALC269_FIXUP_THINKPAD_ACPI, otherwise the mute led will not work if a thinkpad machine applies that quirk. And it will be safe if non-thinkpad machines apply that quirk since hda_fixup_thinkpad_acpi() will check and return in this case. Fixes: ae7abe36e352e ("ALSA: hda/realtek: Add CS35L41 support for Thinkpad laptops") Signed-off-by: Hui Wang Link: https://lore.kernel.org/r/20220422073937.10073-1-hui.wang@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4c0c593f3c0a..f9c3b2c9ca12 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -8769,6 +8769,8 @@ static const struct hda_fixup alc269_fixups[] = { [ALC287_FIXUP_CS35L41_I2C_2] = { .type = HDA_FIXUP_FUNC, .v.func = cs35l41_fixup_i2c_two, + .chained = true, + .chain_id = ALC269_FIXUP_THINKPAD_ACPI, }, [ALC287_FIXUP_CS35L41_I2C_2_HP_GPIO_LED] = { .type = HDA_FIXUP_FUNC, -- cgit From 5f5d8890789c90470d9571a283f0b789acd594af Mon Sep 17 00:00:00 2001 From: Andy Chi Date: Fri, 22 Apr 2022 17:08:43 +0800 Subject: ALSA: hda/realtek: Enable mute/micmute LEDs support for HP Laptops On HP Laptops, requires the same ALC285_FIXUP_HP_GPIO_LED quirk to make its audio LEDs work. So apply the quirk, and make it the last one since it's an LED quirk. Signed-off-by: Andy Chi Fixes: 07bcab93946c ("ALSA: hda/realtek: Add support for HP Laptops") Link: https://lore.kernel.org/r/20220422090845.230071-1-andy.chi@canonical.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f9c3b2c9ca12..c65d3dbc6cc9 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9025,12 +9025,12 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x103c, 0x8896, "HP EliteBook 855 G8 Notebook PC", ALC285_FIXUP_HP_MUTE_LED), SND_PCI_QUIRK(0x103c, 0x8898, "HP EliteBook 845 G8 Notebook PC", ALC285_FIXUP_HP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x103c, 0x88d0, "HP Pavilion 15-eh1xxx (mainboard 88D0)", ALC287_FIXUP_HP_GPIO_LED), - SND_PCI_QUIRK(0x103c, 0x896e, "HP EliteBook x360 830 G9", ALC245_FIXUP_CS35L41_SPI_2), - SND_PCI_QUIRK(0x103c, 0x8971, "HP EliteBook 830 G9", ALC245_FIXUP_CS35L41_SPI_2), - SND_PCI_QUIRK(0x103c, 0x8972, "HP EliteBook 840 G9", ALC245_FIXUP_CS35L41_SPI_2), - SND_PCI_QUIRK(0x103c, 0x8973, "HP EliteBook 860 G9", ALC245_FIXUP_CS35L41_SPI_2), - SND_PCI_QUIRK(0x103c, 0x8974, "HP EliteBook 840 Aero G9", ALC245_FIXUP_CS35L41_SPI_2), - SND_PCI_QUIRK(0x103c, 0x8975, "HP EliteBook x360 840 Aero G9", ALC245_FIXUP_CS35L41_SPI_2), + SND_PCI_QUIRK(0x103c, 0x896e, "HP EliteBook x360 830 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8971, "HP EliteBook 830 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8972, "HP EliteBook 840 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8973, "HP EliteBook 860 G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8974, "HP EliteBook 840 Aero G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), + SND_PCI_QUIRK(0x103c, 0x8975, "HP EliteBook x360 840 Aero G9", ALC245_FIXUP_CS35L41_SPI_2_HP_GPIO_LED), 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), -- cgit From 87c18514bb8477563a61f50b4285da156296edc4 Mon Sep 17 00:00:00 2001 From: ChiYuan Huang Date: Fri, 22 Apr 2022 14:26:50 +0800 Subject: ASoC: rt9120: Correct the reg 0x09 size to one byte Correct the reg 0x09 size to one byte. Signed-off-by: ChiYuan Huang Cc: stable@vger.kernel.org Link: https://lore.kernel.org/r/1650608810-3829-1-git-send-email-u0084500@gmail.com Signed-off-by: Mark Brown --- sound/soc/codecs/rt9120.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/rt9120.c b/sound/soc/codecs/rt9120.c index 7aa1772a915f..6e0d7cf0c8c9 100644 --- a/sound/soc/codecs/rt9120.c +++ b/sound/soc/codecs/rt9120.c @@ -341,7 +341,6 @@ static int rt9120_get_reg_size(unsigned int reg) { switch (reg) { case 0x00: - case 0x09: case 0x20 ... 0x27: return 2; case 0x30 ... 0x3D: -- cgit From b7c81f80246fac44077166f3e07103affe6db8ff Mon Sep 17 00:00:00 2001 From: Chengfeng Ye Date: Sat, 9 Apr 2022 13:12:41 +0900 Subject: firewire: fix potential uaf in outbound_phy_packet_callback() &e->event and e point to the same address, and &e->event could be freed in queue_event. So there is a potential uaf issue if we dereference e after calling queue_event(). Fix this by adding a temporary variable to maintain e->client in advance, this can avoid the potential uaf issue. Cc: Signed-off-by: Chengfeng Ye Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20220409041243.603210-2-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- drivers/firewire/core-cdev.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/drivers/firewire/core-cdev.c b/drivers/firewire/core-cdev.c index 9f89c17730b1..708e417200f4 100644 --- a/drivers/firewire/core-cdev.c +++ b/drivers/firewire/core-cdev.c @@ -1500,6 +1500,7 @@ static void outbound_phy_packet_callback(struct fw_packet *packet, { struct outbound_phy_packet_event *e = container_of(packet, struct outbound_phy_packet_event, p); + struct client *e_client; switch (status) { /* expected: */ @@ -1516,9 +1517,10 @@ static void outbound_phy_packet_callback(struct fw_packet *packet, } e->phy_packet.data[0] = packet->timestamp; + e_client = e->client; queue_event(e->client, &e->event, &e->phy_packet, sizeof(e->phy_packet) + e->phy_packet.length, NULL, 0); - client_put(e->client); + client_put(e_client); } static int ioctl_send_phy_packet(struct client *client, union ioctl_arg *arg) -- cgit From 9423973869bd4632ffe669f950510c49296656e0 Mon Sep 17 00:00:00 2001 From: Jakob Koschel Date: Sat, 9 Apr 2022 13:12:42 +0900 Subject: firewire: remove check of list iterator against head past the loop body When list_for_each_entry() completes the iteration over the whole list without breaking the loop, the iterator value will be a bogus pointer computed based on the head element. While it is safe to use the pointer to determine if it was computed based on the head element, either with list_entry_is_head() or &pos->member == head, using the iterator variable after the loop should be avoided. In preparation to limit the scope of a list iterator to the list traversal loop, use a dedicated pointer to point to the found element [1]. Link: https://lore.kernel.org/all/CAHk-=wgRr_D8CB-D9Kg-c=EHreAsk5SqXPwr9Y7k9sA6cWXJ6w@mail.gmail.com/ [1] Cc: Signed-off-by: Jakob Koschel Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20220409041243.603210-3-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- drivers/firewire/core-transaction.c | 30 ++++++++++++++++-------------- drivers/firewire/sbp2.c | 13 +++++++------ 2 files changed, 23 insertions(+), 20 deletions(-) diff --git a/drivers/firewire/core-transaction.c b/drivers/firewire/core-transaction.c index ac487c96bb71..6c20815cc8d1 100644 --- a/drivers/firewire/core-transaction.c +++ b/drivers/firewire/core-transaction.c @@ -73,24 +73,25 @@ static int try_cancel_split_timeout(struct fw_transaction *t) static int close_transaction(struct fw_transaction *transaction, struct fw_card *card, int rcode) { - struct fw_transaction *t; + struct fw_transaction *t = NULL, *iter; unsigned long flags; spin_lock_irqsave(&card->lock, flags); - list_for_each_entry(t, &card->transaction_list, link) { - if (t == transaction) { - if (!try_cancel_split_timeout(t)) { + list_for_each_entry(iter, &card->transaction_list, link) { + if (iter == transaction) { + if (!try_cancel_split_timeout(iter)) { spin_unlock_irqrestore(&card->lock, flags); goto timed_out; } - list_del_init(&t->link); - card->tlabel_mask &= ~(1ULL << t->tlabel); + list_del_init(&iter->link); + card->tlabel_mask &= ~(1ULL << iter->tlabel); + t = iter; break; } } spin_unlock_irqrestore(&card->lock, flags); - if (&t->link != &card->transaction_list) { + if (t) { t->callback(card, rcode, NULL, 0, t->callback_data); return 0; } @@ -935,7 +936,7 @@ EXPORT_SYMBOL(fw_core_handle_request); void fw_core_handle_response(struct fw_card *card, struct fw_packet *p) { - struct fw_transaction *t; + struct fw_transaction *t = NULL, *iter; unsigned long flags; u32 *data; size_t data_length; @@ -947,20 +948,21 @@ void fw_core_handle_response(struct fw_card *card, struct fw_packet *p) rcode = HEADER_GET_RCODE(p->header[1]); spin_lock_irqsave(&card->lock, flags); - list_for_each_entry(t, &card->transaction_list, link) { - if (t->node_id == source && t->tlabel == tlabel) { - if (!try_cancel_split_timeout(t)) { + list_for_each_entry(iter, &card->transaction_list, link) { + if (iter->node_id == source && iter->tlabel == tlabel) { + if (!try_cancel_split_timeout(iter)) { spin_unlock_irqrestore(&card->lock, flags); goto timed_out; } - list_del_init(&t->link); - card->tlabel_mask &= ~(1ULL << t->tlabel); + list_del_init(&iter->link); + card->tlabel_mask &= ~(1ULL << iter->tlabel); + t = iter; break; } } spin_unlock_irqrestore(&card->lock, flags); - if (&t->link == &card->transaction_list) { + if (!t) { timed_out: fw_notice(card, "unsolicited response (source %x, tlabel %x)\n", source, tlabel); diff --git a/drivers/firewire/sbp2.c b/drivers/firewire/sbp2.c index 85cd379fd383..60051c0cabea 100644 --- a/drivers/firewire/sbp2.c +++ b/drivers/firewire/sbp2.c @@ -408,7 +408,7 @@ static void sbp2_status_write(struct fw_card *card, struct fw_request *request, void *payload, size_t length, void *callback_data) { struct sbp2_logical_unit *lu = callback_data; - struct sbp2_orb *orb; + struct sbp2_orb *orb = NULL, *iter; struct sbp2_status status; unsigned long flags; @@ -433,17 +433,18 @@ static void sbp2_status_write(struct fw_card *card, struct fw_request *request, /* Lookup the orb corresponding to this status write. */ spin_lock_irqsave(&lu->tgt->lock, flags); - list_for_each_entry(orb, &lu->orb_list, link) { + list_for_each_entry(iter, &lu->orb_list, link) { if (STATUS_GET_ORB_HIGH(status) == 0 && - STATUS_GET_ORB_LOW(status) == orb->request_bus) { - orb->rcode = RCODE_COMPLETE; - list_del(&orb->link); + STATUS_GET_ORB_LOW(status) == iter->request_bus) { + iter->rcode = RCODE_COMPLETE; + list_del(&iter->link); + orb = iter; break; } } spin_unlock_irqrestore(&lu->tgt->lock, flags); - if (&orb->link != &lu->orb_list) { + if (orb) { orb->callback(orb, &status); kref_put(&orb->kref, free_orb); /* orb callback reference */ } else { -- cgit From a7ecbe92b9243edbe94772f6f2c854e4142a3345 Mon Sep 17 00:00:00 2001 From: Niels Dossche Date: Sat, 9 Apr 2022 13:12:43 +0900 Subject: firewire: core: extend card->lock in fw_core_handle_bus_reset card->local_node and card->bm_retries are both always accessed under card->lock. fw_core_handle_bus_reset has a check whose condition depends on card->local_node and whose body writes to card->bm_retries. Both of these accesses are not under card->lock. Move the lock acquiring of card->lock to before this check such that these accesses do happen when card->lock is held. fw_destroy_nodes is called inside the check. Since fw_destroy_nodes already acquires card->lock inside its function body, move this out to the callsites of fw_destroy_nodes. Also add a comment to indicate which locking is necessary when calling fw_destroy_nodes. Cc: Signed-off-by: Niels Dossche Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20220409041243.603210-4-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- drivers/firewire/core-card.c | 3 +++ drivers/firewire/core-topology.c | 9 +++------ 2 files changed, 6 insertions(+), 6 deletions(-) diff --git a/drivers/firewire/core-card.c b/drivers/firewire/core-card.c index 54be88167c60..f3b3953cac83 100644 --- a/drivers/firewire/core-card.c +++ b/drivers/firewire/core-card.c @@ -668,6 +668,7 @@ EXPORT_SYMBOL_GPL(fw_card_release); void fw_core_remove_card(struct fw_card *card) { struct fw_card_driver dummy_driver = dummy_driver_template; + unsigned long flags; card->driver->update_phy_reg(card, 4, PHY_LINK_ACTIVE | PHY_CONTENDER, 0); @@ -682,7 +683,9 @@ void fw_core_remove_card(struct fw_card *card) dummy_driver.stop_iso = card->driver->stop_iso; card->driver = &dummy_driver; + spin_lock_irqsave(&card->lock, flags); fw_destroy_nodes(card); + spin_unlock_irqrestore(&card->lock, flags); /* Wait for all users, especially device workqueue jobs, to finish. */ fw_card_put(card); diff --git a/drivers/firewire/core-topology.c b/drivers/firewire/core-topology.c index b63d55f5ebd3..f40c81534381 100644 --- a/drivers/firewire/core-topology.c +++ b/drivers/firewire/core-topology.c @@ -375,16 +375,13 @@ static void report_found_node(struct fw_card *card, card->bm_retries = 0; } +/* Must be called with card->lock held */ void fw_destroy_nodes(struct fw_card *card) { - unsigned long flags; - - spin_lock_irqsave(&card->lock, flags); card->color++; if (card->local_node != NULL) for_each_fw_node(card, card->local_node, report_lost_node); card->local_node = NULL; - spin_unlock_irqrestore(&card->lock, flags); } static void move_tree(struct fw_node *node0, struct fw_node *node1, int port) @@ -510,6 +507,8 @@ void fw_core_handle_bus_reset(struct fw_card *card, int node_id, int generation, struct fw_node *local_node; unsigned long flags; + spin_lock_irqsave(&card->lock, flags); + /* * If the selfID buffer is not the immediate successor of the * previously processed one, we cannot reliably compare the @@ -521,8 +520,6 @@ void fw_core_handle_bus_reset(struct fw_card *card, int node_id, int generation, card->bm_retries = 0; } - spin_lock_irqsave(&card->lock, flags); - card->broadcast_channel_allocated = card->broadcast_channel_auto_allocated; card->node_id = node_id; /* -- cgit From 3b79954fd00d540677c97a560622b73f3a1f4e28 Mon Sep 17 00:00:00 2001 From: Zihao Wang Date: Sun, 24 Apr 2022 16:41:20 +0800 Subject: ALSA: hda/realtek: Add quirk for Yoga Duet 7 13ITL6 speakers Lenovo Yoga Duet 7 13ITL6 has Realtek ALC287 and built-in speakers do not work out of the box. The fix developed for Yoga 7i 14ITL5 also enables speaker output for this model. Signed-off-by: Zihao Wang Cc: Link: https://lore.kernel.org/r/20220424084120.74125-1-wzhd@ustc.edu 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 c65d3dbc6cc9..cf531c1efa13 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9247,6 +9247,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x3813, "Legion 7i 15IMHG05", ALC287_FIXUP_LEGION_15IMHG05_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3818, "Lenovo C940", ALC298_FIXUP_LENOVO_SPK_VOLUME), SND_PCI_QUIRK(0x17aa, 0x3819, "Lenovo 13s Gen2 ITL", ALC287_FIXUP_13S_GEN2_SPEAKERS), + SND_PCI_QUIRK(0x17aa, 0x3820, "Yoga Duet 7 13ITL6", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3824, "Legion Y9000X 2020", ALC285_FIXUP_LEGION_Y9000X_SPEAKERS), SND_PCI_QUIRK(0x17aa, 0x3827, "Ideapad S740", ALC285_FIXUP_IDEAPAD_S740_COEF), SND_PCI_QUIRK(0x17aa, 0x3834, "Lenovo IdeaPad Slim 9i 14ITL5", ALC287_FIXUP_YOGA7_14ITL_SPEAKERS), -- cgit From eb9d84b0ffe39893cb23b0b6712bbe3637fa25fa Mon Sep 17 00:00:00 2001 From: Takashi Sakamoto Date: Sun, 24 Apr 2022 19:24:28 +0900 Subject: ALSA: fireworks: fix wrong return count shorter than expected by 4 bytes ALSA fireworks driver has a bug in its initial state to return count shorter than expected by 4 bytes to userspace applications when handling response frame for Echo Audio Fireworks transaction. It's due to missing addition of the size for the type of event in ALSA firewire stack. Fixes: 555e8a8f7f14 ("ALSA: fireworks: Add command/response functionality into hwdep interface") Cc: Signed-off-by: Takashi Sakamoto Link: https://lore.kernel.org/r/20220424102428.21109-1-o-takashi@sakamocchi.jp Signed-off-by: Takashi Iwai --- sound/firewire/fireworks/fireworks_hwdep.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/firewire/fireworks/fireworks_hwdep.c b/sound/firewire/fireworks/fireworks_hwdep.c index 626c0c34b0b6..3a53914277d3 100644 --- a/sound/firewire/fireworks/fireworks_hwdep.c +++ b/sound/firewire/fireworks/fireworks_hwdep.c @@ -34,6 +34,7 @@ hwdep_read_resp_buf(struct snd_efw *efw, char __user *buf, long remained, type = SNDRV_FIREWIRE_EVENT_EFW_RESPONSE; if (copy_to_user(buf, &type, sizeof(type))) return -EFAULT; + count += sizeof(type); remained -= sizeof(type); buf += sizeof(type); -- cgit From 2fbe467bcbfc760a08f08475eea6bbd4c2874319 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 20 Apr 2022 20:34:53 +0100 Subject: ASoC: max98090: Reject invalid values in custom control put() The max98090 driver has a custom put function for some controls which can only be updated in certain circumstances which makes no effort to validate that input is suitable for the control, allowing out of spec values to be written to the hardware and presented to userspace. Fix this by returning an error when invalid values are written. Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20220420193454.2647908-1-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index b45ec35cd63c..6d9261346842 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -413,6 +413,9 @@ static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol, val = (val >> mc->shift) & mask; + if (sel < 0 || sel > mc->max) + return -EINVAL; + *select = sel; /* Setting a volume is only valid if it is already On */ -- cgit From 13fcf676d9e102594effc686d98521ff5c90b925 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Wed, 20 Apr 2022 20:34:54 +0100 Subject: ASoC: max98090: Generate notifications on changes for custom control The max98090 driver has some custom controls which share a put() function which returns 0 unconditionally, meaning that events are not generated when the value changes. Fix that. Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20220420193454.2647908-2-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/codecs/max98090.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/max98090.c b/sound/soc/codecs/max98090.c index 6d9261346842..62b41ca050a2 100644 --- a/sound/soc/codecs/max98090.c +++ b/sound/soc/codecs/max98090.c @@ -430,7 +430,7 @@ static int max98090_put_enab_tlv(struct snd_kcontrol *kcontrol, mask << mc->shift, sel << mc->shift); - return 0; + return *select != val; } static const char *max98090_perf_pwr_text[] = -- cgit From 2bde1985e39173d8cb64005dad6f34e9bee4c750 Mon Sep 17 00:00:00 2001 From: Codrin Ciubotariu Date: Thu, 21 Apr 2022 15:54:03 +0300 Subject: ASoC: atmel: mchp-pdmc: set prepare_slave_config Since a pointer to struct snd_dmaengine_pcm_config is passed, snd_dmaengine_pcm_prepare_slave_config() is no longer called unless it's explicitly set in prepare_slave_config. Fixes: 50291652af52 ("ASoC: atmel: mchp-pdmc: add PDMC driver") Suggested-by: Sascha Hauer Signed-off-by: Codrin Ciubotariu Link: https://lore.kernel.org/r/20220421125403.2180824-2-codrin.ciubotariu@microchip.com Signed-off-by: Mark Brown --- sound/soc/atmel/mchp-pdmc.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/atmel/mchp-pdmc.c b/sound/soc/atmel/mchp-pdmc.c index 1a7802fbf23c..a3856c73e221 100644 --- a/sound/soc/atmel/mchp-pdmc.c +++ b/sound/soc/atmel/mchp-pdmc.c @@ -966,6 +966,7 @@ static int mchp_pdmc_process(struct snd_pcm_substream *substream, static struct snd_dmaengine_pcm_config mchp_pdmc_config = { .process = mchp_pdmc_process, + .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, }; static int mchp_pdmc_probe(struct platform_device *pdev) -- cgit From 660564fc9a92a893a14f255be434f7ea0b967901 Mon Sep 17 00:00:00 2001 From: Codrin Ciubotariu Date: Thu, 21 Apr 2022 15:54:02 +0300 Subject: ASoC: dmaengine: Restore NULL prepare_slave_config() callback As pointed out by Sascha Hauer, this patch changes: if (pmc->config && !pcm->config->prepare_slave_config) to: if (pmc->config && !pcm->config->prepare_slave_config) snd_dmaengine_pcm_prepare_slave_config() This breaks the drivers that do not need a call to dmaengine_slave_config(). Drivers that still need to call snd_dmaengine_pcm_prepare_slave_config(), but have a NULL pcm->config->prepare_slave_config should use snd_dmaengine_pcm_prepare_slave_config() as their prepare_slave_config callback. Fixes: 9a1e13440a4f ("ASoC: dmaengine: do not use a NULL prepare_slave_config() callback") Reported-by: Sascha Hauer Signed-off-by: Codrin Ciubotariu Link: https://lore.kernel.org/r/20220421125403.2180824-1-codrin.ciubotariu@microchip.com Signed-off-by: Mark Brown --- sound/soc/soc-generic-dmaengine-pcm.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 2ab2ddc1294d..285441d6aeed 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -86,10 +86,10 @@ static int dmaengine_pcm_hw_params(struct snd_soc_component *component, memset(&slave_config, 0, sizeof(slave_config)); - if (pcm->config && pcm->config->prepare_slave_config) - prepare_slave_config = pcm->config->prepare_slave_config; - else + if (!pcm->config) prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config; + else + prepare_slave_config = pcm->config->prepare_slave_config; if (prepare_slave_config) { int ret = prepare_slave_config(substream, params, &slave_config); -- cgit From aa22125c57f9e577f0a667e4fa07fc3fa8ca1e60 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Sat, 23 Apr 2022 14:12:39 +0100 Subject: ASoC: ops: Validate input values in snd_soc_put_volsw_range() Check that values written via snd_soc_put_volsw_range() are within the range advertised by the control, ensuring that we don't write out of spec values to the hardware. Signed-off-by: Mark Brown Link: https://lore.kernel.org/r/20220423131239.3375261-1-broonie@kernel.org Signed-off-by: Mark Brown --- sound/soc/soc-ops.c | 18 +++++++++++++++++- 1 file changed, 17 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-ops.c b/sound/soc/soc-ops.c index 58347eadd219..e693070f51fe 100644 --- a/sound/soc/soc-ops.c +++ b/sound/soc/soc-ops.c @@ -519,7 +519,15 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, unsigned int mask = (1 << fls(max)) - 1; unsigned int invert = mc->invert; unsigned int val, val_mask; - int err, ret; + int err, ret, tmp; + + tmp = ucontrol->value.integer.value[0]; + if (tmp < 0) + return -EINVAL; + if (mc->platform_max && tmp > mc->platform_max) + return -EINVAL; + if (tmp > mc->max - mc->min + 1) + return -EINVAL; if (invert) val = (max - ucontrol->value.integer.value[0]) & mask; @@ -534,6 +542,14 @@ int snd_soc_put_volsw_range(struct snd_kcontrol *kcontrol, ret = err; if (snd_soc_volsw_is_stereo(mc)) { + tmp = ucontrol->value.integer.value[1]; + if (tmp < 0) + return -EINVAL; + if (mc->platform_max && tmp > mc->platform_max) + return -EINVAL; + if (tmp > mc->max - mc->min + 1) + return -EINVAL; + if (invert) val = (max - ucontrol->value.integer.value[1]) & mask; else -- cgit From c61711c1c95791850be48dd65a1d72eb34ba719f Mon Sep 17 00:00:00 2001 From: Ajit Kumar Pandey Date: Tue, 26 Apr 2022 13:33:57 -0500 Subject: ASoC: SOF: Fix NULL pointer exception in sof_pci_probe callback We are accessing "desc->ops" in sof_pci_probe without checking "desc" pointer. This results in NULL pointer exception if pci_id->driver_data i.e desc pointer isn't defined in sof device probe: BUG: kernel NULL pointer dereference, address: 0000000000000060 PGD 0 P4D 0 Oops: 0000 [#1] PREEMPT SMP NOPTI RIP: 0010:sof_pci_probe+0x1e/0x17f [snd_sof_pci] Code: Unable to access opcode bytes at RIP 0xffffffffc043dff4. RSP: 0018:ffffac4b03b9b8d8 EFLAGS: 00010246 Add NULL pointer check for sof_dev_desc pointer to avoid such exception. Reviewed-by: Ranjani Sridharan Signed-off-by: Ajit Kumar Pandey Signed-off-by: Pierre-Louis Bossart Link: https://lore.kernel.org/r/20220426183357.102155-1-pierre-louis.bossart@linux.intel.com Signed-off-by: Mark Brown --- sound/soc/sof/sof-pci-dev.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/sof/sof-pci-dev.c b/sound/soc/sof/sof-pci-dev.c index 12f5cff22448..7fa2649e56e5 100644 --- a/sound/soc/sof/sof-pci-dev.c +++ b/sound/soc/sof/sof-pci-dev.c @@ -153,6 +153,11 @@ int sof_pci_probe(struct pci_dev *pci, const struct pci_device_id *pci_id) dev_dbg(&pci->dev, "PCI DSP detected"); + if (!desc) { + dev_err(dev, "error: no matching PCI descriptor\n"); + return -ENODEV; + } + if (!desc->ops) { dev_err(dev, "error: no matching PCI descriptor ops\n"); return -ENODEV; -- cgit From d7be213849232a2accb219d537edf056d29186b4 Mon Sep 17 00:00:00 2001 From: Forest Crossman Date: Tue, 3 May 2022 19:24:44 -0500 Subject: ALSA: usb-audio: Don't get sample rate for MCT Trigger 5 USB-to-HDMI This device doesn't support reading the sample rate, so we need to apply this quirk to avoid a 15-second delay waiting for three timeouts. Signed-off-by: Forest Crossman Link: https://lore.kernel.org/r/20220504002444.114011-2-cyrozap@gmail.com Signed-off-by: Takashi Iwai --- sound/usb/quirks.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index ab9f3da49941..fbbe59054c3f 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1822,6 +1822,8 @@ static const struct usb_audio_quirk_flags_table quirk_flags_table[] = { QUIRK_FLAG_IGNORE_CTL_ERROR), DEVICE_FLG(0x06f8, 0xd002, /* Hercules DJ Console (Macintosh Edition) */ QUIRK_FLAG_IGNORE_CTL_ERROR), + DEVICE_FLG(0x0711, 0x5800, /* MCT Trigger 5 USB-to-HDMI */ + QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x074d, 0x3553, /* Outlaw RR2150 (Micronas UAC3553B) */ QUIRK_FLAG_GET_SAMPLE_RATE), DEVICE_FLG(0x08bb, 0x2702, /* LineX FM Transmitter */ -- cgit From c3d9ca93f1e3bd3d1adfc4479a12c82fed424c87 Mon Sep 17 00:00:00 2001 From: Randy Dunlap Date: Sat, 30 Apr 2022 12:33:18 -0700 Subject: ALSA: hda - fix unused Realtek function when PM is not enabled When CONFIG_PM is not enabled, alc_shutup() is not needed, so move it inside the #ifdef CONFIG_PM guard. Also drop some contiguous #endif / #ifdef CONFIG_PM for simplicity. Fixes this build warning: sound/pci/hda/patch_realtek.c:886:20: warning: unused function 'alc_shutup' Fixes: 08c189f2c552 ("ALSA: hda - Use generic parser codes for Realtek driver") Signed-off-by: Randy Dunlap Reported-by: kernel test robot Link: https://lore.kernel.org/r/20220430193318.29024-1-rdunlap@infradead.org Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 8 +++----- 1 file changed, 3 insertions(+), 5 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index cf531c1efa13..89b705d38f5b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -937,6 +937,9 @@ static int alc_init(struct hda_codec *codec) return 0; } +#define alc_free snd_hda_gen_free + +#ifdef CONFIG_PM static inline void alc_shutup(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -950,9 +953,6 @@ static inline void alc_shutup(struct hda_codec *codec) alc_shutup_pins(codec); } -#define alc_free snd_hda_gen_free - -#ifdef CONFIG_PM static void alc_power_eapd(struct hda_codec *codec) { alc_auto_setup_eapd(codec, false); @@ -966,9 +966,7 @@ static int alc_suspend(struct hda_codec *codec) spec->power_hook(codec); return 0; } -#endif -#ifdef CONFIG_PM static int alc_resume(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; -- cgit From 1efcdd9c1f34f5a6590bc9ac5471e562fb011386 Mon Sep 17 00:00:00 2001 From: Gabriele Mazzotta Date: Sun, 1 May 2022 14:42:37 +0200 Subject: ALSA: hda/realtek: Add quirk for Dell Latitude 7520 The driver is currently using ALC269_FIXUP_DELL4_MIC_NO_PRESENCE for the Latitude 7520, but this fixup chain has some issues: - The internal mic is really loud and the recorded audio is distorted at "standard" audio levels. - There are pop noises at system startup and when plugging/unplugging headphone jacks. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215885 Signed-off-by: Gabriele Mazzotta Link: https://lore.kernel.org/r/20220501124237.4667-1-gabriele.mzt@gmail.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 43 +++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 43 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 89b705d38f5b..37ad7b6780a2 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6778,6 +6778,41 @@ static void alc256_fixup_mic_no_presence_and_resume(struct hda_codec *codec, } } +static void alc_fixup_dell4_mic_no_presence_quiet(struct hda_codec *codec, + const struct hda_fixup *fix, + int action) +{ + struct alc_spec *spec = codec->spec; + struct hda_input_mux *imux = &spec->gen.input_mux; + int i; + + alc269_fixup_limit_int_mic_boost(codec, fix, action); + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + /** + * Set the vref of pin 0x19 (Headset Mic) and pin 0x1b (Headphone Mic) + * to Hi-Z to avoid pop noises at startup and when plugging and + * unplugging headphones. + */ + snd_hda_codec_set_pin_target(codec, 0x19, PIN_VREFHIZ); + snd_hda_codec_set_pin_target(codec, 0x1b, PIN_VREFHIZ); + break; + case HDA_FIXUP_ACT_PROBE: + /** + * Make the internal mic (0x12) the default input source to + * prevent pop noises on cold boot. + */ + for (i = 0; i < imux->num_items; i++) { + if (spec->gen.imux_pins[i] == 0x12) { + spec->gen.cur_mux[0] = i; + break; + } + } + break; + } +} + enum { ALC269_FIXUP_GPIO2, ALC269_FIXUP_SONY_VAIO, @@ -6819,6 +6854,7 @@ enum { ALC269_FIXUP_DELL2_MIC_NO_PRESENCE, ALC269_FIXUP_DELL3_MIC_NO_PRESENCE, ALC269_FIXUP_DELL4_MIC_NO_PRESENCE, + ALC269_FIXUP_DELL4_MIC_NO_PRESENCE_QUIET, ALC269_FIXUP_HEADSET_MODE, ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC, ALC269_FIXUP_ASPIRE_HEADSET_MIC, @@ -8806,6 +8842,12 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC285_FIXUP_HP_MUTE_LED, }, + [ALC269_FIXUP_DELL4_MIC_NO_PRESENCE_QUIET] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc_fixup_dell4_mic_no_presence_quiet, + .chained = true, + .chain_id = ALC269_FIXUP_DELL4_MIC_NO_PRESENCE, + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -8896,6 +8938,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x09bf, "Dell Precision", ALC233_FIXUP_ASUS_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x0a2e, "Dell", ALC236_FIXUP_DELL_AIO_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0a30, "Dell", ALC236_FIXUP_DELL_AIO_HEADSET_MIC), + SND_PCI_QUIRK(0x1028, 0x0a38, "Dell Latitude 7520", ALC269_FIXUP_DELL4_MIC_NO_PRESENCE_QUIET), SND_PCI_QUIRK(0x1028, 0x0a58, "Dell", ALC255_FIXUP_DELL_HEADSET_MIC), SND_PCI_QUIRK(0x1028, 0x0a61, "Dell XPS 15 9510", ALC289_FIXUP_DUAL_SPK), SND_PCI_QUIRK(0x1028, 0x0a62, "Dell Precision 5560", ALC289_FIXUP_DUAL_SPK), -- cgit From a34ae6c0660d3b96b0055f68ef74dc9478852245 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Tue, 10 May 2022 12:36:26 +0200 Subject: ALSA: wavefront: Proper check of get_user() error The antient ISA wavefront driver reads its sample patch data (uploaded over an ioctl) via __get_user() with no good reason; likely just for some performance optimizations in the past. Let's change this to the standard get_user() and the error check for handling the fault case properly. Reported-by: Linus Torvalds Cc: Link: https://lore.kernel.org/r/20220510103626.16635-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/isa/wavefront/wavefront_synth.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/isa/wavefront/wavefront_synth.c b/sound/isa/wavefront/wavefront_synth.c index 69cbc79fbb71..2aaaa6807174 100644 --- a/sound/isa/wavefront/wavefront_synth.c +++ b/sound/isa/wavefront/wavefront_synth.c @@ -1094,7 +1094,8 @@ wavefront_send_sample (snd_wavefront_t *dev, if (dataptr < data_end) { - __get_user (sample_short, dataptr); + if (get_user(sample_short, dataptr)) + return -EFAULT; dataptr += skip; if (data_is_unsigned) { /* GUS ? */ -- cgit From 309d7363ca3d9fcdb92ff2d958be14d7e8707f68 Mon Sep 17 00:00:00 2001 From: "Dustin L. Howett" Date: Tue, 10 May 2022 20:07:59 -0500 Subject: ALSA: hda/realtek: Add quirk for the Framework Laptop Some board revisions of the Framework Laptop have an ALC295 with a disconnected or faulty headset mic presence detect. The "dell-headset-multi" fixup addresses this issue, but also enables an inoperative "Headphone Mic" input device whenever a headset is connected. Adding a new quirk chain specific to the Framework Laptop resolves this issue. The one introduced here is based on the System76 "no headphone mic" quirk chain. The VID:PID f111:0001 have been allocated to Framework Computer for this board revision. Revision history: - v2: Moved to a custom quirk chain to suppress the "Headphone Mic" pincfg. Signed-off-by: Dustin L. Howett Link: https://lore.kernel.org/r/20220511010759.3554-1-dustin@howett.net Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 37ad7b6780a2..33c439578a61 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -7046,6 +7046,7 @@ enum { ALC245_FIXUP_CS35L41_SPI_4, ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED, ALC285_FIXUP_HP_SPEAKERS_MICMUTE_LED, + ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE, }; static const struct hda_fixup alc269_fixups[] = { @@ -8848,6 +8849,15 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC269_FIXUP_DELL4_MIC_NO_PRESENCE, }, + [ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE] = { + .type = HDA_FIXUP_PINS, + .v.pins = (const struct hda_pintbl[]) { + { 0x19, 0x02a1112c }, /* use as headset mic, without its own jack detect */ + { } + }, + .chained = true, + .chain_id = ALC269_FIXUP_HEADSET_MODE_NO_HP_MIC + }, }; static const struct snd_pci_quirk alc269_fixup_tbl[] = { @@ -9338,6 +9348,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x8086, 0x2074, "Intel NUC 8", ALC233_FIXUP_INTEL_NUC8_DMIC), SND_PCI_QUIRK(0x8086, 0x2080, "Intel NUC 8 Rugged", ALC256_FIXUP_INTEL_NUC8_RUGGED), SND_PCI_QUIRK(0x8086, 0x2081, "Intel NUC 10", ALC256_FIXUP_INTEL_NUC10), + SND_PCI_QUIRK(0xf111, 0x0001, "Framework Laptop", ALC295_FIXUP_FRAMEWORK_LAPTOP_MIC_NO_PRESENCE), #if 0 /* Below is a quirk table taken from the old code. -- cgit From 8b3b2392ed68bcd17c7eb84ca615ce1e5f115b99 Mon Sep 17 00:00:00 2001 From: Werner Sembach Date: Thu, 12 May 2022 20:09:56 +0200 Subject: ALSA: hda/realtek: Add quirk for TongFang devices with pop noise When audio stops playing there is an audible "pop"-noise when using headphones on the TongFang GMxMRxx, GKxNRxx, GMxZGxx, GMxTGxx and GMxAGxx. This quirk fixes this mostly. Signed-off-by: Werner Sembach Cc: stable@vger.kernel.org Link: https://lore.kernel.org/r/20220512180956.281804-1-wse@tuxedocomputers.com Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 33c439578a61..ce2cb1986677 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9341,6 +9341,14 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1c06, 0x2013, "Lemote A1802", ALC269_FIXUP_LEMOTE_A1802), SND_PCI_QUIRK(0x1c06, 0x2015, "Lemote A190X", ALC269_FIXUP_LEMOTE_A190X), SND_PCI_QUIRK(0x1d05, 0x1132, "TongFang PHxTxX1", ALC256_FIXUP_SET_COEF_DEFAULTS), + SND_PCI_QUIRK(0x1d05, 0x1096, "TongFang GMxMRxx", ALC269_FIXUP_NO_SHUTUP), + SND_PCI_QUIRK(0x1d05, 0x1100, "TongFang GKxNRxx", ALC269_FIXUP_NO_SHUTUP), + SND_PCI_QUIRK(0x1d05, 0x1111, "TongFang GMxZGxx", ALC269_FIXUP_NO_SHUTUP), + SND_PCI_QUIRK(0x1d05, 0x1119, "TongFang GMxZGxx", ALC269_FIXUP_NO_SHUTUP), + SND_PCI_QUIRK(0x1d05, 0x1129, "TongFang GMxZGxx", ALC269_FIXUP_NO_SHUTUP), + SND_PCI_QUIRK(0x1d05, 0x1147, "TongFang GMxTGxx", ALC269_FIXUP_NO_SHUTUP), + SND_PCI_QUIRK(0x1d05, 0x115c, "TongFang GMxTGxx", ALC269_FIXUP_NO_SHUTUP), + SND_PCI_QUIRK(0x1d05, 0x121b, "TongFang GMxAGxx", ALC269_FIXUP_NO_SHUTUP), SND_PCI_QUIRK(0x1d72, 0x1602, "RedmiBook", ALC255_FIXUP_XIAOMI_HEADSET_MIC), SND_PCI_QUIRK(0x1d72, 0x1701, "XiaomiNotebook Pro", ALC298_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1d72, 0x1901, "RedmiBook 14", ALC256_FIXUP_ASUS_HEADSET_MIC), -- cgit From 024a7ad9eb4df626ca8c77fef4f67fd0ebd559d2 Mon Sep 17 00:00:00 2001 From: Andy Chi Date: Fri, 13 May 2022 20:16:45 +0800 Subject: ALSA: hda/realtek: fix right sounds and mute/micmute LEDs for HP machine The HP EliteBook 630 is using ALC236 codec which used 0x02 to control mute LED and 0x01 to control micmute LED. Therefore, add a quirk to make it works. Signed-off-by: Andy Chi Cc: Link: https://lore.kernel.org/r/20220513121648.28584-1-andy.chi@canonical.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 ce2cb1986677..ad292df7d805 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -9091,6 +9091,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { 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), + SND_PCI_QUIRK(0x103c, 0x89aa, "HP EliteBook 630 G9", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89ac, "HP EliteBook 640 G9", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89ae, "HP EliteBook 650 G9", ALC236_FIXUP_HP_GPIO_LED), SND_PCI_QUIRK(0x103c, 0x89c3, "Zbook Studio G9", ALC245_FIXUP_CS35L41_SPI_4_HP_GPIO_LED), -- cgit From 5c62383c06837b5719cd5447a5758b791279e653 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Mon, 16 May 2022 12:31:12 +0200 Subject: ALSA: usb-audio: Restore Rane SL-1 quirk At cleaning up and moving the device rename from the quirk table to its own table, we removed the entry for Rane SL-1 as we thought it's only for renaming. It turned out, however, that the quirk is required for matching with the device that declares itself as no standard audio but only as vendor-specific. Restore the quirk entry for Rane SL-1 to fix the regression. BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=215887 Fixes: 5436f59bc5bc ("ALSA: usb-audio: Move device rename and profile quirks to an internal table") Cc: Link: https://lore.kernel.org/r/20220516103112.12950-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 0ea39565e623..40a5e3eb4ef2 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -3235,6 +3235,15 @@ YAMAHA_DEVICE(0x7010, "UB99"), } }, +/* Rane SL-1 */ +{ + USB_DEVICE(0x13e5, 0x0001), + .driver_info = (unsigned long) & (const struct snd_usb_audio_quirk) { + .ifnum = QUIRK_ANY_INTERFACE, + .type = QUIRK_AUDIO_STANDARD_INTERFACE + } +}, + /* disabled due to regression for other devices; * see https://bugzilla.kernel.org/show_bug.cgi?id=199905 */ -- cgit From 12626530de84d2b00ff6480142f7f4374662c0c3 Mon Sep 17 00:00:00 2001 From: Kees Cook Date: Fri, 20 May 2022 09:55:37 -0700 Subject: ALSA: lola: Bounds check loop iterator against streams array size GCC 12 sees that it's technically possible for num_streams to be larger than ARRAY_SIZE(pcm->streams). Bounds-check the iterator. ../sound/pci/lola/lola_pcm.c: In function 'lola_pcm_update': ../sound/pci/lola/lola_pcm.c:567:64: warning: array subscript [0, 31] is outside array bounds of 'struct lola_stream[16]' [-Warray-bounds] 567 | struct lola_stream *str = &pcm->streams[i]; | ~~~~~~~~~~~~^~~ In file included from ../sound/pci/lola/lola_pcm.c:15: ../sound/pci/lola/lola.h:307:28: note: while referencing 'streams' 307 | struct lola_stream streams[MAX_STREAM_COUNT]; | ^~~~~~~ Signed-off-by: Kees Cook Link: https://lore.kernel.org/r/20220520165537.2139826-1-keescook@chromium.org Signed-off-by: Takashi Iwai --- sound/pci/lola/lola_pcm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/pci/lola/lola_pcm.c b/sound/pci/lola/lola_pcm.c index 738ec987000a..32193fae978d 100644 --- a/sound/pci/lola/lola_pcm.c +++ b/sound/pci/lola/lola_pcm.c @@ -561,8 +561,9 @@ static snd_pcm_uframes_t lola_pcm_pointer(struct snd_pcm_substream *substream) void lola_pcm_update(struct lola *chip, struct lola_pcm *pcm, unsigned int bits) { int i; + u8 num_streams = min_t(u8, pcm->num_streams, ARRAY_SIZE(pcm->streams)); - for (i = 0; bits && i < pcm->num_streams; i++) { + for (i = 0; bits && i < num_streams; i++) { if (bits & (1 << i)) { struct lola_stream *str = &pcm->streams[i]; if (str->substream && str->running) -- cgit From 5ce0b06ae5e69e23142e73c5c3c0260e9f2ccb4b Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 21 May 2022 08:46:27 +0200 Subject: ALSA: usb-audio: Workaround for clock setup on TEAC devices Maris reported that TEAC UD-501 (0644:8043) doesn't work with the typical "clock source 41 is not valid, cannot use" errors on the recent kernels. The currently known workaround so far is to restore (partially) what we've done unconditionally at the clock setup; namely, re-setup the USB interface immediately after the clock is changed. This patch re-introduces the behavior conditionally for TEAC devices. Further notes: - The USB interface shall be set later in snd_usb_endpoint_configure(), but this seems to be too late. - Even calling usb_set_interface() right after sne_usb_init_sample_rate() doesn't help; so this must be related with the clock validation, too. - The device may still spew the "clock source 41 is not valid" error at the first clock setup. This seems happening at the very first try of clock setup, but it disappears at later attempts. The error is likely harmless because the driver retries the clock setup (such an error is more or less expected on some devices). Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management") Reported-and-tested-by: Maris Abele Cc: Link: https://lore.kernel.org/r/20220521064627.29292-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/clock.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/usb/clock.c b/sound/usb/clock.c index 4dfe76416794..3c435d379306 100644 --- a/sound/usb/clock.c +++ b/sound/usb/clock.c @@ -572,6 +572,13 @@ static int set_sample_rate_v2v3(struct snd_usb_audio *chip, /* continue processing */ } + /* FIXME - TEAC devices require the immediate interface setup */ + if (rate != prev_rate && USB_ID_VENDOR(chip->usb_id) == 0x0644) { + usb_set_interface(chip->dev, fmt->iface, fmt->altsetting); + if (chip->quirk_flags & QUIRK_FLAG_IFACE_DELAY) + msleep(50); + } + validation: /* validate clock after rate change */ if (!uac_clock_source_is_valid(chip, fmt, clock)) -- cgit From 7b0efea4baf02f5e2f89e5f9b75ef891571b45f1 Mon Sep 17 00:00:00 2001 From: Takashi Iwai Date: Sat, 21 May 2022 08:53:25 +0200 Subject: ALSA: usb-audio: Add missing ep_idx in fixed EP quirks MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The quirk entry for Focusrite Saffire 6 had no proper ep_idx for the capture endpoint, and this confused the driver, resulting in the broken sound. This patch adds the missing ep_idx in the entry. While we are at it, a couple of other entries (for Digidesign MBox and MOTU MicroBook II) seem to have the same problem, and those are covered as well. Fixes: bf6313a0ff76 ("ALSA: usb-audio: Refactor endpoint management") Reported-by: André Kapelrud Cc: Link: https://lore.kernel.org/r/20220521065325.426-1-tiwai@suse.de Signed-off-by: Takashi Iwai --- sound/usb/quirks-table.h | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/usb/quirks-table.h b/sound/usb/quirks-table.h index 40a5e3eb4ef2..78eb41b621d6 100644 --- a/sound/usb/quirks-table.h +++ b/sound/usb/quirks-table.h @@ -2672,6 +2672,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .altset_idx = 1, .attributes = 0, .endpoint = 0x82, + .ep_idx = 1, .ep_attr = USB_ENDPOINT_XFER_ISOC, .datainterval = 1, .maxpacksize = 0x0126, @@ -2875,6 +2876,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .altset_idx = 1, .attributes = 0x4, .endpoint = 0x81, + .ep_idx = 1, .ep_attr = USB_ENDPOINT_XFER_ISOC | USB_ENDPOINT_SYNC_ASYNC, .maxpacksize = 0x130, @@ -3391,6 +3393,7 @@ YAMAHA_DEVICE(0x7010, "UB99"), .altset_idx = 1, .attributes = 0, .endpoint = 0x03, + .ep_idx = 1, .rates = SNDRV_PCM_RATE_96000, .ep_attr = USB_ENDPOINT_XFER_ISOC | USB_ENDPOINT_SYNC_ASYNC, -- cgit