From 8053f21675b073b379cbca258ee4a3f3850dfa94 Mon Sep 17 00:00:00 2001 From: Nicolin Chen Date: Tue, 26 Jul 2016 14:55:51 -0700 Subject: [PATCH 01/22] ASoC: dapm: Add a dummy snd_pcm_runtime to avoid NULL pointer access The SND_SOC_DAPM_PRE_PMU case would call startup()/hw_params() that might access substream->runtime through other functions. For example: Unable to handle kernel NULL pointer dereference at virtual address [....] PC is at snd_pcm_hw_rule_add+0x24/0x1b0 LR is at snd_pcm_hw_constraint_list+0x20/0x28 [....] Process arecord (pid: 424, stack limit = 0xffffffc1ecaf0020) Call trace: [] snd_pcm_hw_rule_add+0x24/0x1b0 [] snd_pcm_hw_constraint_list+0x20/0x28 [] cs53l30_pcm_startup+0x24/0x30 [] snd_soc_dai_link_event+0x290/0x354 [] dapm_seq_check_event.isra.31+0x134/0x2c8 [] dapm_seq_run_coalesced+0x94/0x1c8 [] dapm_seq_run+0xa4/0x404 [] dapm_power_widgets+0x524/0x984 [] snd_soc_dapm_stream_event+0x8c/0xa8 [] soc_pcm_prepare+0x10c/0x1ec [] snd_pcm_do_prepare+0x1c/0x38 [] snd_pcm_action_single+0x40/0x88 [] snd_pcm_action_nonatomic+0x70/0x90 [] snd_pcm_common_ioctl1+0xb6c/0xdd8 [] snd_pcm_capture_ioctl1+0x200/0x334 [] snd_pcm_ioctl_compat+0x648/0x95c [] compat_SyS_ioctl+0xac/0xfc4 [] el0_svc_naked+0x24/0x28 ---[ end trace 0dc4f99c2759c35c ]--- So this patch adds a dummy runtime for the original dummy substream to merely avoid the NULL pointer access. Signed-off-by: Nicolin Chen Signed-off-by: Mark Brown --- sound/soc/soc-dapm.c | 10 ++++++++++ 1 file changed, 10 insertions(+) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 8698c26773b3..d908ff8f9755 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3493,6 +3493,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, const struct snd_soc_pcm_stream *config = w->params + w->params_select; struct snd_pcm_substream substream; struct snd_pcm_hw_params *params = NULL; + struct snd_pcm_runtime *runtime = NULL; u64 fmt; int ret; @@ -3541,6 +3542,14 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, memset(&substream, 0, sizeof(substream)); + /* Allocate a dummy snd_pcm_runtime for startup() and other ops() */ + runtime = kzalloc(sizeof(*runtime), GFP_KERNEL); + if (!runtime) { + ret = -ENOMEM; + goto out; + } + substream.runtime = runtime; + switch (event) { case SND_SOC_DAPM_PRE_PMU: substream.stream = SNDRV_PCM_STREAM_CAPTURE; @@ -3606,6 +3615,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, } out: + kfree(runtime); kfree(params); return ret; } From 93ca33c99f22a0a096f83f19c9f887aeb67507a1 Mon Sep 17 00:00:00 2001 From: Hiroyuki Yokoyama Date: Mon, 25 Jul 2016 01:52:43 +0000 Subject: [PATCH 02/22] ASoC: rsnd: Fixup SRCm_IFSVR calculate method This patch fixes the calculation accuracy degradation of SRCm_IFSVR register value. Signed-off-by: Hiroyuki Yokoyama Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/sh/rcar/src.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/src.c b/sound/soc/sh/rcar/src.c index e39f916d0f2f..969a5169de25 100644 --- a/sound/soc/sh/rcar/src.c +++ b/sound/soc/sh/rcar/src.c @@ -226,8 +226,12 @@ static void rsnd_src_set_convert_rate(struct rsnd_dai_stream *io, ifscr = 0; fsrate = 0; if (fin != fout) { + u64 n; + ifscr = 1; - fsrate = 0x0400000 / fout * fin; + n = (u64)0x0400000 * fin; + do_div(n, fout); + fsrate = n; } /* From 2392f7fd695246df4fb5f0b5fb88ce37cdb01764 Mon Sep 17 00:00:00 2001 From: Vinod Koul Date: Tue, 26 Jul 2016 18:06:39 +0530 Subject: [PATCH 03/22] ASoC: Intel: Skylake: Check list empty while getting module info Module list can be NULL so check if the list is empty before accessing the list. Signed-off-by: Senthilnathan Veppur Signed-off-by: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl-sst-utils.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/soc/intel/skylake/skl-sst-utils.c b/sound/soc/intel/skylake/skl-sst-utils.c index 25fcb796bd86..ddcb52a51854 100644 --- a/sound/soc/intel/skylake/skl-sst-utils.c +++ b/sound/soc/intel/skylake/skl-sst-utils.c @@ -123,6 +123,11 @@ int snd_skl_get_module_info(struct skl_sst *ctx, u8 *uuid, uuid_mod = (uuid_le *)uuid; + if (list_empty(&ctx->uuid_list)) { + dev_err(ctx->dev, "Module list is empty\n"); + return -EINVAL; + } + list_for_each_entry(module, &ctx->uuid_list, list) { if (uuid_le_cmp(*uuid_mod, module->uuid) == 0) { dfw_config->module_id = module->id; From 1f85e118c81d15aa9e002604dfb69c823d4aac16 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Wed, 3 Aug 2016 01:24:05 +0000 Subject: [PATCH 04/22] ASoC: simple-card-utils: add missing MODULE_xxx() simple-card-utils might be used as module, but MODULE_xxx() information was missed. This patch adds it. Otherwise, we will have below error, and can't use it. Specil thanks to Kevin. > insmod simple-card-utils.ko simple_card_utils: module license 'unspecified' taints kernel. Disabling lock debugging due to kernel taint simple_card_utils: Unknown symbol snd_soc_of_parse_daifmt (err 0) simple_card_utils: Unknown symbol snd_soc_of_parse_card_name (err 0) insmod: can't insert 'simple-card-utils.ko': \ unknown symbol in module, or unknown parameter Reported-by: Kevin Hilman Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- sound/soc/generic/Makefile | 6 +++--- sound/soc/generic/simple-card-utils.c | 6 ++++++ 2 files changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/soc/generic/Makefile b/sound/soc/generic/Makefile index 45602ca8536e..2d53c8d70705 100644 --- a/sound/soc/generic/Makefile +++ b/sound/soc/generic/Makefile @@ -1,5 +1,5 @@ -obj-$(CONFIG_SND_SIMPLE_CARD_UTILS) := simple-card-utils.o - +snd-soc-simple-card-utils-objs := simple-card-utils.o snd-soc-simple-card-objs := simple-card.o -obj-$(CONFIG_SND_SIMPLE_CARD) += snd-soc-simple-card.o +obj-$(CONFIG_SND_SIMPLE_CARD_UTILS) += snd-soc-simple-card-utils.o +obj-$(CONFIG_SND_SIMPLE_CARD) += snd-soc-simple-card.o diff --git a/sound/soc/generic/simple-card-utils.c b/sound/soc/generic/simple-card-utils.c index d89a9a1b2471..9599de69a880 100644 --- a/sound/soc/generic/simple-card-utils.c +++ b/sound/soc/generic/simple-card-utils.c @@ -7,6 +7,7 @@ * it under the terms of the GNU General Public License version 2 as * published by the Free Software Foundation. */ +#include #include #include @@ -95,3 +96,8 @@ int asoc_simple_card_parse_card_name(struct snd_soc_card *card, return 0; } EXPORT_SYMBOL_GPL(asoc_simple_card_parse_card_name); + +/* Module information */ +MODULE_AUTHOR("Kuninori Morimoto "); +MODULE_DESCRIPTION("ALSA SoC Simple Card Utils"); +MODULE_LICENSE("GPL v2"); From 1bc610e7a17dcf5165f1ed4e0201ee080ba1a0df Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki Date: Thu, 4 Aug 2016 11:51:25 +0200 Subject: [PATCH 05/22] ASoC: samsung: Fix clock handling in S3C24XX_UDA134X card There is no "pclk" alias in the s3c2440 clk driver for "soc-audio" device so related clk_get() fails, which prevents any operation of the S3C24XX_UDA134X sound card. Instead we get the clock on behalf of the I2S device, i.e. we use the I2S block gate clock which has PCLK is its parent clock. Without this patch there is an error like: s3c24xx_uda134x_startup cannot get pclk ASoC: UDA134X startup failed: -2 Signed-off-by: Sylwester Nawrocki Signed-off-by: Mark Brown --- sound/soc/samsung/s3c24xx_uda134x.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/samsung/s3c24xx_uda134x.c b/sound/soc/samsung/s3c24xx_uda134x.c index 50849e137fc0..92e88bca386e 100644 --- a/sound/soc/samsung/s3c24xx_uda134x.c +++ b/sound/soc/samsung/s3c24xx_uda134x.c @@ -58,10 +58,12 @@ static struct platform_device *s3c24xx_uda134x_snd_device; static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream) { - int ret = 0; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->cpu_dai; #ifdef ENFORCE_RATES struct snd_pcm_runtime *runtime = substream->runtime; #endif + int ret = 0; mutex_lock(&clk_lock); pr_debug("%s %d\n", __func__, clk_users); @@ -71,8 +73,7 @@ static int s3c24xx_uda134x_startup(struct snd_pcm_substream *substream) printk(KERN_ERR "%s cannot get xtal\n", __func__); ret = PTR_ERR(xtal); } else { - pclk = clk_get(&s3c24xx_uda134x_snd_device->dev, - "pclk"); + pclk = clk_get(cpu_dai->dev, "iis"); if (IS_ERR(pclk)) { printk(KERN_ERR "%s cannot get pclk\n", __func__); From ca6ac305f017472a172e53345264abdb495eba46 Mon Sep 17 00:00:00 2001 From: John Hsu Date: Thu, 4 Aug 2016 16:52:06 +0800 Subject: [PATCH 06/22] ASoC: nau8825: fix bug in playback when suspend In chromium, the following steps will make codec function fail. \1. plug in headphones, Play music \2. run "powerd_dbus_suspend" \3. resume from S3 After resume, the jack detection will restart and make configuration for the headset. Meanwhile, the playback prepares and starts to work. The two sequences will conflict and make wrong register configuration. Originally, the driver adds protection for the case when it finds the playback is active. But the "powerd_dbus_suspend" command will close the pcm stream before suspend. Therefore, the driver can't detect the playback after resume, and the protection not works. For the issue, the driver raises protection every time after resume. The protection will release after jack detection and configuration completes, and then the playback just will goes on. Signed-off-by: John Hsu Signed-off-by: Mark Brown --- sound/soc/codecs/nau8825.c | 51 +++++++++----------------------------- 1 file changed, 12 insertions(+), 39 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 5c9707ac4bbf..20d2f1578b87 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -212,31 +212,6 @@ static const unsigned short logtable[256] = { 0xfa2f, 0xfaea, 0xfba5, 0xfc60, 0xfd1a, 0xfdd4, 0xfe8e, 0xff47 }; -static struct snd_soc_dai *nau8825_get_codec_dai(struct nau8825 *nau8825) -{ - struct snd_soc_codec *codec = snd_soc_dapm_to_codec(nau8825->dapm); - struct snd_soc_component *component = &codec->component; - struct snd_soc_dai *codec_dai, *_dai; - - list_for_each_entry_safe(codec_dai, _dai, &component->dai_list, list) { - if (!strncmp(codec_dai->name, NUVOTON_CODEC_DAI, - strlen(NUVOTON_CODEC_DAI))) - return codec_dai; - } - return NULL; -} - -static bool nau8825_dai_is_active(struct nau8825 *nau8825) -{ - struct snd_soc_dai *codec_dai = nau8825_get_codec_dai(nau8825); - - if (codec_dai) { - if (codec_dai->playback_active || codec_dai->capture_active) - return true; - } - return false; -} - /** * nau8825_sema_acquire - acquire the semaphore of nau88l25 * @nau8825: component to register the codec private data with @@ -1205,6 +1180,8 @@ static int nau8825_hw_params(struct snd_pcm_substream *substream, struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); unsigned int val_len = 0; + nau8825_sema_acquire(nau8825, 2 * HZ); + switch (params_width(params)) { case 16: val_len |= NAU8825_I2S_DL_16; @@ -1225,6 +1202,9 @@ static int nau8825_hw_params(struct snd_pcm_substream *substream, regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL1, NAU8825_I2S_DL_MASK, val_len); + /* Release the semaphone. */ + nau8825_sema_release(nau8825); + return 0; } @@ -1234,6 +1214,8 @@ static int nau8825_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); unsigned int ctrl1_val = 0, ctrl2_val = 0; + nau8825_sema_acquire(nau8825, 2 * HZ); + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { case SND_SOC_DAIFMT_CBM_CFM: ctrl2_val |= NAU8825_I2S_MS_MASTER; @@ -1282,6 +1264,9 @@ static int nau8825_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, NAU8825_I2S_MS_MASK, ctrl2_val); + /* Release the semaphone. */ + nau8825_sema_release(nau8825); + return 0; } @@ -2241,20 +2226,8 @@ static int __maybe_unused nau8825_resume(struct snd_soc_codec *codec) regcache_cache_only(nau8825->regmap, false); regcache_sync(nau8825->regmap); - if (nau8825_is_jack_inserted(nau8825->regmap)) { - /* If the jack is inserted, we need to check whether the play- - * back is active before suspend. If active, the driver has to - * raise the protection for cross talk function to avoid the - * playback recovers before cross talk process finish. Other- - * wise, the playback will be interfered by cross talk func- - * tion. It is better to apply hardware related parameters - * before starting playback or record. - */ - if (nau8825_dai_is_active(nau8825)) { - nau8825->xtalk_protect = true; - nau8825_sema_acquire(nau8825, 0); - } - } + nau8825->xtalk_protect = true; + nau8825_sema_acquire(nau8825, 0); enable_irq(nau8825->irq); return 0; From 06746c672cdd60715ee57ab1aced95a9e536fd4d Mon Sep 17 00:00:00 2001 From: John Hsu Date: Thu, 4 Aug 2016 16:52:07 +0800 Subject: [PATCH 07/22] ASoC: nau8825: fix static check error about semaphone control The patch is to fix the static check error as the following. The patch commit b50455fab459 ("ASoC: nau8825: cross talk suppression measurement function") from Jun 7, 2016, leads to the following static checker warning: sound/soc/codecs/nau8825.c:265 nau8825_sema_acquire() warn: 'sem:&nau8825->xtalk_sem' is sometimes locked here and sometimes unlocked. The semaphone acquire function has return value, and some callers can do error handling when lock fails. Signed-off-by: John Hsu Signed-off-by: Mark Brown --- sound/soc/codecs/nau8825.c | 29 +++++++++++++++++++++-------- 1 file changed, 21 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c index 20d2f1578b87..2e59a85e360b 100644 --- a/sound/soc/codecs/nau8825.c +++ b/sound/soc/codecs/nau8825.c @@ -225,19 +225,26 @@ static const unsigned short logtable[256] = { * Acquires the semaphore without jiffies. If no more tasks are allowed * to acquire the semaphore, calling this function will put the task to * sleep until the semaphore is released. - * It returns if the semaphore was acquired. + * If the semaphore is not released within the specified number of jiffies, + * this function returns -ETIME. + * If the sleep is interrupted by a signal, this function will return -EINTR. + * It returns 0 if the semaphore was acquired successfully. */ -static void nau8825_sema_acquire(struct nau8825 *nau8825, long timeout) +static int nau8825_sema_acquire(struct nau8825 *nau8825, long timeout) { int ret; - if (timeout) + if (timeout) { ret = down_timeout(&nau8825->xtalk_sem, timeout); - else + if (ret < 0) + dev_warn(nau8825->dev, "Acquire semaphone timeout\n"); + } else { ret = down_interruptible(&nau8825->xtalk_sem); + if (ret < 0) + dev_warn(nau8825->dev, "Acquire semaphone fail\n"); + } - if (ret < 0) - dev_warn(nau8825->dev, "Acquire semaphone fail\n"); + return ret; } /** @@ -1596,8 +1603,11 @@ static irqreturn_t nau8825_interrupt(int irq, void *data) * cess and restore changes if process * is ongoing when ejection. */ + int ret; nau8825->xtalk_protect = true; - nau8825_sema_acquire(nau8825, 0); + ret = nau8825_sema_acquire(nau8825, 0); + if (ret < 0) + nau8825->xtalk_protect = false; } /* Startup cross talk detection process */ nau8825->xtalk_state = NAU8825_XTALK_PREPARE; @@ -2223,11 +2233,14 @@ static int __maybe_unused nau8825_suspend(struct snd_soc_codec *codec) static int __maybe_unused nau8825_resume(struct snd_soc_codec *codec) { struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + int ret; regcache_cache_only(nau8825->regmap, false); regcache_sync(nau8825->regmap); nau8825->xtalk_protect = true; - nau8825_sema_acquire(nau8825, 0); + ret = nau8825_sema_acquire(nau8825, 0); + if (ret < 0) + nau8825->xtalk_protect = false; enable_irq(nau8825->irq); return 0; From 5d764912a0ee6db83e962c1501b5b9e58ba14e15 Mon Sep 17 00:00:00 2001 From: Adam Thomson Date: Thu, 4 Aug 2016 15:35:37 +0100 Subject: [PATCH 08/22] ASoC: da7213: Default to 64 BCLKs per WCLK to support all formats Previously code defaulted to 32 BCLKS per WCLK which meant 24 and 32 bit DAI formats would not work properly. This patch fixes the issue by defaulting to 64 BCLKs per WCLK. Signed-off-by: Adam Thomson Signed-off-by: Mark Brown --- sound/soc/codecs/da7213.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/da7213.c b/sound/soc/codecs/da7213.c index e5527bc570ae..bcf1834c5648 100644 --- a/sound/soc/codecs/da7213.c +++ b/sound/soc/codecs/da7213.c @@ -1247,8 +1247,8 @@ static int da7213_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) return -EINVAL; } - /* By default only 32 BCLK per WCLK is supported */ - dai_clk_mode |= DA7213_DAI_BCLKS_PER_WCLK_32; + /* By default only 64 BCLK per WCLK is supported */ + dai_clk_mode |= DA7213_DAI_BCLKS_PER_WCLK_64; snd_soc_write(codec, DA7213_DAI_CLK_MODE, dai_clk_mode); snd_soc_update_bits(codec, DA7213_DAI_CTRL, DA7213_DAI_FORMAT_MASK, From 70fcad495b9903d362bdbf9ffba23259294064c8 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 11 Aug 2016 14:39:00 +0100 Subject: [PATCH 09/22] ASoC: Fix leak of rtd in soc_bind_dai_link If we fail to find a platform we simply return EPROBE_DEFER, but we have allocated the rtd pointer. All error paths before soc_add_pcm_runtime need to call soc_free_pcm_runtime first to avoid leaking the rtd pointer. A suitable error path already exists and is used else where in the function so simply use that here as well. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 16369cad4803..b0e23db83695 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1056,7 +1056,7 @@ static int soc_bind_dai_link(struct snd_soc_card *card, if (!rtd->platform) { dev_err(card->dev, "ASoC: platform %s not registered\n", dai_link->platform_name); - return -EPROBE_DEFER; + goto _err_defer; } soc_add_pcm_runtime(card, rtd); From fa54aade5338937146bb6b8af93d2c27f6787ae1 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Thu, 11 Aug 2016 14:40:04 +0100 Subject: [PATCH 10/22] ASoC: wm2000: Fix return of uninitialised varible Anything that sets ret in wm2000_anc_transition will have immediately returned anyway as such we will always return an uninitialised ret at the bottom of the function. Simply replace the return with a return 0; Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/wm2000.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm2000.c b/sound/soc/codecs/wm2000.c index a67ea10f41a1..f2664396be6f 100644 --- a/sound/soc/codecs/wm2000.c +++ b/sound/soc/codecs/wm2000.c @@ -581,7 +581,7 @@ static int wm2000_anc_transition(struct wm2000_priv *wm2000, if (anc_transitions[i].dest == ANC_OFF) clk_disable_unprepare(wm2000->mclk); - return ret; + return 0; } static int wm2000_anc_set_mode(struct wm2000_priv *wm2000) From 979cf59acc9d634cc140aadd0d2915947ab303cc Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Fri, 12 Aug 2016 11:45:18 +0000 Subject: [PATCH 11/22] ASoC: Intel: Skylake: Fix error return code in skl_probe() Fix to return error code -ENODEV from the error handling case instead of 0, as done elsewhere in this function. Fixes: 87b2bdf02278 ("ASoC: Intel: Skylake: Initialize NHLT table") Signed-off-by: Wei Yongjun Acked-By: Vinod Koul Signed-off-by: Mark Brown --- sound/soc/intel/skylake/skl.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index cd59536a761d..e3e764167765 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -672,8 +672,10 @@ static int skl_probe(struct pci_dev *pci, skl->nhlt = skl_nhlt_init(bus->dev); - if (skl->nhlt == NULL) + if (skl->nhlt == NULL) { + err = -ENODEV; goto out_free; + } skl_nhlt_update_topology_bin(skl); From b0f12c61de013ecb0abe19d5e64bdab45989de5a Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Tue, 16 Aug 2016 11:24:46 +0100 Subject: [PATCH 12/22] ASoC: compress: Fix leak of a widget list in soc_compr_open_fe After we have called dpcm_path_get we should make sure to call dpcm_path_put on all error paths. This was not happening causing the allocated widget list to be leaked, this patch corrects this by adding a dpcm_path_put to the error path. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/soc-compress.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index d2df46c14c68..bf7b52fce597 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -121,7 +121,7 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) dpcm_be_disconnect(fe, stream); fe->dpcm[stream].runtime = NULL; - goto fe_err; + goto path_err; } dpcm_clear_pending_state(fe, stream); @@ -136,6 +136,8 @@ static int soc_compr_open_fe(struct snd_compr_stream *cstream) return 0; +path_err: + dpcm_path_put(&list); fe_err: if (fe->dai_link->compr_ops && fe->dai_link->compr_ops->shutdown) fe->dai_link->compr_ops->shutdown(cstream); From 3e103a65514c2947e53f3171b21255fbde8b60c6 Mon Sep 17 00:00:00 2001 From: Christoph Huber Date: Mon, 15 Aug 2016 18:59:25 +0200 Subject: [PATCH 13/22] ASoC: atmel_ssc_dai: Don't unconditionally reset SSC on stream startup commit cbaadf0f90d6 ("ASoC: atmel_ssc_dai: refactor the startup and shutdown") refactored code such that the SSC is reset on every startup; this breaks duplex audio (e.g. first start audio playback, then start record, causing the playback to stop/hang) Fixes: cbaadf0f90d6 (ASoC: atmel_ssc_dai: refactor the startup and shutdown) Signed-off-by: Christoph Huber Signed-off-by: Peter Meerwald-Stadler Signed-off-by: Mark Brown Cc: stable@vger.kernel.org --- sound/soc/atmel/atmel_ssc_dai.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 54c09acd3fed..16e459aedffe 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -299,8 +299,9 @@ static int atmel_ssc_startup(struct snd_pcm_substream *substream, clk_enable(ssc_p->ssc->clk); ssc_p->mck_rate = clk_get_rate(ssc_p->ssc->clk); - /* Reset the SSC to keep it at a clean status */ - ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); + /* Reset the SSC unless initialized to keep it in a clean state */ + if (!ssc_p->initialized) + ssc_writel(ssc_p->ssc->regs, CR, SSC_BIT(CR_SWRST)); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { dir = 0; From 479e2a86dc6aeaec6013165e1bd3525db6914f3a Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Thu, 18 Aug 2016 14:00:59 +0300 Subject: [PATCH 14/22] ASoC: omap-mcpdm: Drop pdmclk clock handling This reverts commit 65aca64d05b5eaa5ce15e18b458a8d338ddbd478. The patches for twl6040 MFD and clk missed the merge window and causing the McPDM driver to never probe since it is put back to the deferred list because the missing drivers. Signed-off-by: Peter Ujfalusi Signed-off-by: Mark Brown --- .../devicetree/bindings/sound/omap-mcpdm.txt | 10 ---------- sound/soc/omap/omap-mcpdm.c | 17 ----------------- 2 files changed, 27 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/omap-mcpdm.txt b/Documentation/devicetree/bindings/sound/omap-mcpdm.txt index 6f6c2f8e908d..0741dff048dd 100644 --- a/Documentation/devicetree/bindings/sound/omap-mcpdm.txt +++ b/Documentation/devicetree/bindings/sound/omap-mcpdm.txt @@ -8,8 +8,6 @@ Required properties: - interrupts: Interrupt number for McPDM - interrupt-parent: The parent interrupt controller - ti,hwmods: Name of the hwmod associated to the McPDM -- clocks: phandle for the pdmclk provider, likely <&twl6040> -- clock-names: Must be "pdmclk" Example: @@ -21,11 +19,3 @@ mcpdm: mcpdm@40132000 { interrupt-parent = <&gic>; ti,hwmods = "mcpdm"; }; - -In board DTS file the pdmclk needs to be added: - -&mcpdm { - clocks = <&twl6040>; - clock-names = "pdmclk"; - status = "okay"; -}; diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index e7cdc51fd806..74d6e6fdcfd0 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -31,7 +31,6 @@ #include #include #include -#include #include #include #include @@ -55,7 +54,6 @@ struct omap_mcpdm { unsigned long phys_base; void __iomem *io_base; int irq; - struct clk *pdmclk; struct mutex mutex; @@ -390,7 +388,6 @@ static int omap_mcpdm_probe(struct snd_soc_dai *dai) struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); int ret; - clk_prepare_enable(mcpdm->pdmclk); pm_runtime_enable(mcpdm->dev); /* Disable lines while request is ongoing */ @@ -425,7 +422,6 @@ static int omap_mcpdm_remove(struct snd_soc_dai *dai) pm_runtime_disable(mcpdm->dev); - clk_disable_unprepare(mcpdm->pdmclk); return 0; } @@ -445,8 +441,6 @@ static int omap_mcpdm_suspend(struct snd_soc_dai *dai) mcpdm->pm_active_count++; } - clk_disable_unprepare(mcpdm->pdmclk); - return 0; } @@ -454,8 +448,6 @@ static int omap_mcpdm_resume(struct snd_soc_dai *dai) { struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); - clk_prepare_enable(mcpdm->pdmclk); - if (mcpdm->pm_active_count) { while (mcpdm->pm_active_count--) pm_runtime_get_sync(mcpdm->dev); @@ -549,15 +541,6 @@ static int asoc_mcpdm_probe(struct platform_device *pdev) mcpdm->dev = &pdev->dev; - mcpdm->pdmclk = devm_clk_get(&pdev->dev, "pdmclk"); - if (IS_ERR(mcpdm->pdmclk)) { - if (PTR_ERR(mcpdm->pdmclk) == -EPROBE_DEFER) - return -EPROBE_DEFER; - dev_warn(&pdev->dev, "Error getting pdmclk (%ld)!\n", - PTR_ERR(mcpdm->pdmclk)); - mcpdm->pdmclk = NULL; - } - ret = devm_snd_soc_register_component(&pdev->dev, &omap_mcpdm_component, &omap_mcpdm_dai, 1); From d1e81428826221d7ff8e4d83db784d099cd232a7 Mon Sep 17 00:00:00 2001 From: Mark Brown Date: Thu, 18 Aug 2016 19:32:59 +0100 Subject: [PATCH 15/22] ASoC: core: Clean up DAPM before the card debugfs Both the card and DAPM cleanups recursively delete their debugfs directories. Since the DAPM debugfs subdirectory for the card is located within the card debugfs this means we end up trying to double free the DAPM subdirectory. Reorder the cleanup to free the card debugfs after we've cleaned up DAPM and it has deleted its own subdirectory. Reported-by: Russell King - ARM Linux Tested-by: Russell King Signed-off-by: Mark Brown --- sound/soc/soc-core.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 16369cad4803..66a33f1e4881 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2083,14 +2083,13 @@ static int soc_cleanup_card_resources(struct snd_soc_card *card) /* remove auxiliary devices */ soc_remove_aux_devices(card); + snd_soc_dapm_free(&card->dapm); soc_cleanup_card_debugfs(card); /* remove the card */ if (card->remove) card->remove(card); - snd_soc_dapm_free(&card->dapm); - snd_card_free(card->snd_card); return 0; From 21eb45db282317543ca46c821bbb8d5075e02cbe Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Fri, 19 Aug 2016 09:34:24 +0300 Subject: [PATCH 16/22] ASoC: omap-abe-twl6040: Correct dmic-codec device registration The dmic-codec was registered within the platform_driver's probe function, which can cause deferred probe to run in loops as reported and analyzed by Russell King. Use module_init/exit in the driver and handle the dmic-codec device registration and removal at that level instead of the platform_driver probe/remove. Signed-off-by: Peter Ujfalusi Reported-by: Russell King Tested-by: Russell King Signed-off-by: Mark Brown --- sound/soc/omap/omap-abe-twl6040.c | 61 ++++++++++++++++--------------- 1 file changed, 32 insertions(+), 29 deletions(-) diff --git a/sound/soc/omap/omap-abe-twl6040.c b/sound/soc/omap/omap-abe-twl6040.c index 0843a68f277c..f61b3b58083b 100644 --- a/sound/soc/omap/omap-abe-twl6040.c +++ b/sound/soc/omap/omap-abe-twl6040.c @@ -38,10 +38,10 @@ struct abe_twl6040 { int jack_detection; /* board can detect jack events */ int mclk_freq; /* MCLK frequency speed for twl6040 */ - - struct platform_device *dmic_codec_dev; }; +struct platform_device *dmic_codec_dev; + static int omap_abe_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -258,8 +258,6 @@ static int omap_abe_probe(struct platform_device *pdev) if (priv == NULL) return -ENOMEM; - priv->dmic_codec_dev = ERR_PTR(-EINVAL); - if (snd_soc_of_parse_card_name(card, "ti,model")) { dev_err(&pdev->dev, "Card name is not provided\n"); return -ENODEV; @@ -284,13 +282,6 @@ static int omap_abe_probe(struct platform_device *pdev) num_links = 2; abe_twl6040_dai_links[1].cpu_of_node = dai_node; abe_twl6040_dai_links[1].platform_of_node = dai_node; - - priv->dmic_codec_dev = platform_device_register_simple( - "dmic-codec", -1, NULL, 0); - if (IS_ERR(priv->dmic_codec_dev)) { - dev_err(&pdev->dev, "Can't instantiate dmic-codec\n"); - return PTR_ERR(priv->dmic_codec_dev); - } } else { num_links = 1; } @@ -299,16 +290,14 @@ static int omap_abe_probe(struct platform_device *pdev) of_property_read_u32(node, "ti,mclk-freq", &priv->mclk_freq); if (!priv->mclk_freq) { dev_err(&pdev->dev, "MCLK frequency not provided\n"); - ret = -EINVAL; - goto err_unregister; + return -EINVAL; } card->fully_routed = 1; if (!priv->mclk_freq) { dev_err(&pdev->dev, "MCLK frequency missing\n"); - ret = -ENODEV; - goto err_unregister; + return -ENODEV; } card->dai_link = abe_twl6040_dai_links; @@ -317,17 +306,9 @@ static int omap_abe_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(card, priv); ret = snd_soc_register_card(card); - if (ret) { + if (ret) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); - goto err_unregister; - } - - return 0; - -err_unregister: - if (!IS_ERR(priv->dmic_codec_dev)) - platform_device_unregister(priv->dmic_codec_dev); return ret; } @@ -335,13 +316,9 @@ err_unregister: static int omap_abe_remove(struct platform_device *pdev) { struct snd_soc_card *card = platform_get_drvdata(pdev); - struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card); snd_soc_unregister_card(card); - if (!IS_ERR(priv->dmic_codec_dev)) - platform_device_unregister(priv->dmic_codec_dev); - return 0; } @@ -361,7 +338,33 @@ static struct platform_driver omap_abe_driver = { .remove = omap_abe_remove, }; -module_platform_driver(omap_abe_driver); +static int __init omap_abe_init(void) +{ + int ret; + + dmic_codec_dev = platform_device_register_simple("dmic-codec", -1, NULL, + 0); + if (IS_ERR(dmic_codec_dev)) { + pr_err("%s: dmic-codec device registration failed\n", __func__); + return PTR_ERR(dmic_codec_dev); + } + + ret = platform_driver_register(&omap_abe_driver); + if (ret) { + pr_err("%s: platform driver registration failed\n", __func__); + platform_device_unregister(dmic_codec_dev); + } + + return ret; +} +module_init(omap_abe_init); + +static void __exit omap_abe_exit(void) +{ + platform_driver_unregister(&omap_abe_driver); + platform_device_unregister(dmic_codec_dev); +} +module_exit(omap_abe_exit); MODULE_AUTHOR("Misael Lopez Cruz "); MODULE_DESCRIPTION("ALSA SoC for OMAP boards with ABE and twl6040 codec"); From 7e4379eae0e31994ea645db1d13006ea8e5ce539 Mon Sep 17 00:00:00 2001 From: Andrej Krutak Date: Thu, 18 Aug 2016 23:52:10 +0200 Subject: [PATCH 17/22] ALSA: line6: Remove double line6_pcm_release() after failed acquire. If there's an error, pcm is released in line6_pcm_acquire already. Fixes: 247d95ee6dd2 ('ALSA: line6: Handle error from line6_pcm_acquire()') Cc: # v4.0+ Reviewed-by: Stefan Hajnoczi Signed-off-by: Andrej Krutak Signed-off-by: Takashi Iwai --- sound/usb/line6/pcm.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/usb/line6/pcm.c b/sound/usb/line6/pcm.c index 204cc074adb9..2bc8879757f9 100644 --- a/sound/usb/line6/pcm.c +++ b/sound/usb/line6/pcm.c @@ -55,7 +55,6 @@ static int snd_line6_impulse_volume_put(struct snd_kcontrol *kcontrol, err = line6_pcm_acquire(line6pcm, LINE6_STREAM_IMPULSE); if (err < 0) { line6pcm->impulse_volume = 0; - line6_pcm_release(line6pcm, LINE6_STREAM_IMPULSE); return err; } } else { From adc8a43a6d6688272ebffa81789fa857e603dec6 Mon Sep 17 00:00:00 2001 From: Andrej Krutak Date: Thu, 18 Aug 2016 23:52:11 +0200 Subject: [PATCH 18/22] ALSA: line6: Give up on the lock while URBs are released. Done, because line6_stream_stop() locks and calls line6_unlink_audio_urbs(), which in turn invokes audio_out_callback(), which tries to lock 2nd time. Fixes: ============================================= [ INFO: possible recursive locking detected ] 4.4.15+ #15 Not tainted --------------------------------------------- mplayer/3591 is trying to acquire lock: (&(&line6pcm->out.lock)->rlock){-.-...}, at: [] audio_out_callback+0x70/0x110 [snd_usb_line6] but task is already holding lock: (&(&line6pcm->out.lock)->rlock){-.-...}, at: [] line6_stream_stop+0x24/0x5c [snd_usb_line6] other info that might help us debug this: Possible unsafe locking scenario: CPU0 ---- lock(&(&line6pcm->out.lock)->rlock); lock(&(&line6pcm->out.lock)->rlock); *** DEADLOCK *** May be due to missing lock nesting notation 3 locks held by mplayer/3591: #0: (snd_pcm_link_rwlock){.-.-..}, at: [] snd_pcm_stream_lock+0x1e/0x40 [snd_pcm] #1: (&(&substream->self_group.lock)->rlock){-.-...}, at: [] snd_pcm_stream_lock+0x26/0x40 [snd_pcm] #2: (&(&line6pcm->out.lock)->rlock){-.-...}, at: [] line6_stream_stop+0x24/0x5c [snd_usb_line6] stack backtrace: CPU: 0 PID: 3591 Comm: mplayer Not tainted 4.4.15+ #15 Hardware name: Generic AM33XX (Flattened Device Tree) [] (unwind_backtrace) from [] (show_stack+0x11/0x14) [] (show_stack) from [] (dump_stack+0x8b/0xac) [] (dump_stack) from [] (__lock_acquire+0xc8b/0x1780) [] (__lock_acquire) from [] (lock_acquire+0x99/0x1c0) [] (lock_acquire) from [] (_raw_spin_lock_irqsave+0x3f/0x4c) [] (_raw_spin_lock_irqsave) from [] (audio_out_callback+0x70/0x110 [snd_usb_line6]) [] (audio_out_callback [snd_usb_line6]) from [] (__usb_hcd_giveback_urb+0x53/0xd0) [] (__usb_hcd_giveback_urb) from [] (musb_giveback+0x3d/0x98) [] (musb_giveback) from [] (musb_urb_dequeue+0x6d/0x114) [] (musb_urb_dequeue) from [] (usb_hcd_unlink_urb+0x39/0x98) [] (usb_hcd_unlink_urb) from [] (line6_unlink_audio_urbs+0x6a/0x6c [snd_usb_line6]) [] (line6_unlink_audio_urbs [snd_usb_line6]) from [] (line6_stream_stop+0x42/0x5c [snd_usb_line6]) [] (line6_stream_stop [snd_usb_line6]) from [] (snd_line6_trigger+0xb6/0xf4 [snd_usb_line6]) [] (snd_line6_trigger [snd_usb_line6]) from [] (snd_pcm_do_stop+0x36/0x38 [snd_pcm]) [] (snd_pcm_do_stop [snd_pcm]) from [] (snd_pcm_action_single+0x22/0x40 [snd_pcm]) [] (snd_pcm_action_single [snd_pcm]) from [] (snd_pcm_action+0xac/0xb0 [snd_pcm]) [] (snd_pcm_action [snd_pcm]) from [] (snd_pcm_drop+0x38/0x64 [snd_pcm]) [] (snd_pcm_drop [snd_pcm]) from [] (snd_pcm_common_ioctl1+0x7fe/0xbe8 [snd_pcm]) [] (snd_pcm_common_ioctl1 [snd_pcm]) from [] (snd_pcm_playback_ioctl1+0x15c/0x51c [snd_pcm]) [] (snd_pcm_playback_ioctl1 [snd_pcm]) from [] (snd_pcm_playback_ioctl+0x20/0x28 [snd_pcm]) [] (snd_pcm_playback_ioctl [snd_pcm]) from [] (do_vfs_ioctl+0x3af/0x5c8) Fixes: 63e20df1e5b2 ('ALSA: line6: Reorganize PCM stream handling') Cc: # v4.0+ Reviewed-by: Stefan Hajnoczi Signed-off-by: Andrej Krutak Signed-off-by: Takashi Iwai --- sound/usb/line6/pcm.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/usb/line6/pcm.c b/sound/usb/line6/pcm.c index 2bc8879757f9..41aa3355e920 100644 --- a/sound/usb/line6/pcm.c +++ b/sound/usb/line6/pcm.c @@ -210,7 +210,9 @@ static void line6_stream_stop(struct snd_line6_pcm *line6pcm, int direction, spin_lock_irqsave(&pstr->lock, flags); clear_bit(type, &pstr->running); if (!pstr->running) { + spin_unlock_irqrestore(&pstr->lock, flags); line6_unlink_audio_urbs(line6pcm, pstr); + spin_lock_irqsave(&pstr->lock, flags); if (direction == SNDRV_PCM_STREAM_CAPTURE) { line6pcm->prev_fbuf = NULL; line6pcm->prev_fsize = 0; From b027d11263836a0cd335520175257dcb99b43757 Mon Sep 17 00:00:00 2001 From: Andrej Krutak Date: Thu, 18 Aug 2016 23:52:12 +0200 Subject: [PATCH 19/22] ALSA: line6: Fix POD sysfs attributes segfault The commit 02fc76f6a changed base of the sysfs attributes from device to card. The "show" callbacks dereferenced wrong objects because of this. Fixes: 02fc76f6a7db ('ALSA: line6: Create sysfs via snd_card_add_dev_attr()') Cc: # v4.0+ Reviewed-by: Stefan Hajnoczi Signed-off-by: Andrej Krutak Signed-off-by: Takashi Iwai --- sound/usb/line6/pod.c | 12 ++++++------ 1 file changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/usb/line6/pod.c b/sound/usb/line6/pod.c index daf81d169a42..45dd34874f43 100644 --- a/sound/usb/line6/pod.c +++ b/sound/usb/line6/pod.c @@ -244,8 +244,8 @@ static int pod_set_system_param_int(struct usb_line6_pod *pod, int value, static ssize_t serial_number_show(struct device *dev, struct device_attribute *attr, char *buf) { - struct usb_interface *interface = to_usb_interface(dev); - struct usb_line6_pod *pod = usb_get_intfdata(interface); + struct snd_card *card = dev_to_snd_card(dev); + struct usb_line6_pod *pod = card->private_data; return sprintf(buf, "%u\n", pod->serial_number); } @@ -256,8 +256,8 @@ static ssize_t serial_number_show(struct device *dev, static ssize_t firmware_version_show(struct device *dev, struct device_attribute *attr, char *buf) { - struct usb_interface *interface = to_usb_interface(dev); - struct usb_line6_pod *pod = usb_get_intfdata(interface); + struct snd_card *card = dev_to_snd_card(dev); + struct usb_line6_pod *pod = card->private_data; return sprintf(buf, "%d.%02d\n", pod->firmware_version / 100, pod->firmware_version % 100); @@ -269,8 +269,8 @@ static ssize_t firmware_version_show(struct device *dev, static ssize_t device_id_show(struct device *dev, struct device_attribute *attr, char *buf) { - struct usb_interface *interface = to_usb_interface(dev); - struct usb_line6_pod *pod = usb_get_intfdata(interface); + struct snd_card *card = dev_to_snd_card(dev); + struct usb_line6_pod *pod = card->private_data; return sprintf(buf, "%d\n", pod->device_id); } From 209c721ce27022656a9d792b89572c4fdd7d31a7 Mon Sep 17 00:00:00 2001 From: Wei Yongjun Date: Sun, 21 Aug 2016 15:37:16 +0000 Subject: [PATCH 20/22] ASoC: max98371: Add terminate entry for i2c_device_id tables Make sure i2c_device_id tables are NULL terminated. Signed-off-by: Wei Yongjun Signed-off-by: Mark Brown --- sound/soc/codecs/max98371.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/max98371.c b/sound/soc/codecs/max98371.c index cf0a39bb631a..02352ed8961c 100644 --- a/sound/soc/codecs/max98371.c +++ b/sound/soc/codecs/max98371.c @@ -412,6 +412,7 @@ static int max98371_i2c_remove(struct i2c_client *client) static const struct i2c_device_id max98371_i2c_id[] = { { "max98371", 0 }, + { } }; MODULE_DEVICE_TABLE(i2c, max98371_i2c_id); From a8719670687c46ed2e904c0d05fa4cd7e4950cd1 Mon Sep 17 00:00:00 2001 From: Peter Ujfalusi Date: Tue, 23 Aug 2016 10:27:19 +0300 Subject: [PATCH 21/22] ASoC: omap-mcpdm: Fix irq resource handling Fixes: ddd17531ad908 ("ASoC: omap-mcpdm: Clean up with devm_* function") Managed irq request will not doing any good in ASoC probe level as it is not going to free up the irq when the driver is unbound from the sound card. Signed-off-by: Peter Ujfalusi Reported-by: Russell King Signed-off-by: Mark Brown --- sound/soc/omap/omap-mcpdm.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/omap-mcpdm.c b/sound/soc/omap/omap-mcpdm.c index 74d6e6fdcfd0..64609c77a79d 100644 --- a/sound/soc/omap/omap-mcpdm.c +++ b/sound/soc/omap/omap-mcpdm.c @@ -394,8 +394,8 @@ static int omap_mcpdm_probe(struct snd_soc_dai *dai) pm_runtime_get_sync(mcpdm->dev); omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, 0x00); - ret = devm_request_irq(mcpdm->dev, mcpdm->irq, omap_mcpdm_irq_handler, - 0, "McPDM", (void *)mcpdm); + ret = request_irq(mcpdm->irq, omap_mcpdm_irq_handler, 0, "McPDM", + (void *)mcpdm); pm_runtime_put_sync(mcpdm->dev); @@ -420,6 +420,7 @@ static int omap_mcpdm_remove(struct snd_soc_dai *dai) { struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai); + free_irq(mcpdm->irq, (void *)mcpdm); pm_runtime_disable(mcpdm->dev); return 0; From abaa2274811d607679e8687b4118c4922a3517ac Mon Sep 17 00:00:00 2001 From: Anisse Astier Date: Wed, 24 Aug 2016 09:14:13 +0200 Subject: [PATCH 22/22] ALSA: hda/realtek - fix headset mic detection for MSI MS-B120 MSI Cubi MS-B120 needs the same fixup as the Gigabyte BXBT-2807 for its mic to work. They both use a single 3-way jack for both mic and headset with an ALC283 codec, with the same pins used. Cc: Daniel Drake Signed-off-by: Anisse Astier Signed-off-by: Takashi Iwai --- sound/pci/hda/patch_realtek.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 574b1b48996f..7100f05e651a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4828,7 +4828,7 @@ enum { ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, ALC292_FIXUP_TPT440_DOCK, ALC292_FIXUP_TPT440, - ALC283_FIXUP_BXBT2807_MIC, + ALC283_FIXUP_HEADSET_MIC, ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED, ALC282_FIXUP_ASPIRE_V5_PINS, ALC280_FIXUP_HP_GPIO4, @@ -5321,7 +5321,7 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC292_FIXUP_TPT440_DOCK, }, - [ALC283_FIXUP_BXBT2807_MIC] = { + [ALC283_FIXUP_HEADSET_MIC] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { { 0x19, 0x04a110f0 }, @@ -5651,7 +5651,8 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x10cf, 0x1757, "Lifebook E752", ALC269_FIXUP_LIFEBOOK_HP_PIN), SND_PCI_QUIRK(0x10cf, 0x1845, "Lifebook U904", ALC269_FIXUP_LIFEBOOK_EXTMIC), SND_PCI_QUIRK(0x144d, 0xc109, "Samsung Ativ book 9 (NP900X3G)", ALC269_FIXUP_INV_DMIC), - SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_BXBT2807_MIC), + SND_PCI_QUIRK(0x1458, 0xfa53, "Gigabyte BXBT-2807", ALC283_FIXUP_HEADSET_MIC), + SND_PCI_QUIRK(0x1462, 0xb120, "MSI Cubi MS-B120", ALC283_FIXUP_HEADSET_MIC), SND_PCI_QUIRK(0x17aa, 0x20f2, "Thinkpad SL410/510", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x215e, "Thinkpad L512", ALC269_FIXUP_SKU_IGNORE), SND_PCI_QUIRK(0x17aa, 0x21b8, "Thinkpad Edge 14", ALC269_FIXUP_SKU_IGNORE),