July 2020 - Alsa-devel - mailman.alsa-project.org

[PATCH][next] ALSA: vmaster: Use flex_array_size() helper in memcpy()
by Gustavo A. R. Silva 30 Jul '20

30 Jul '20

Make use of the flex_array_size() helper to calculate the size of a flexible array member within an enclosing structure. This helper offers defense-in-depth against potential integer overflows and makes it explicitly clear that we are dealing with a flexible array member. Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- sound/core/vmaster.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c index ab36f9898711..21ce4082cb5f 100644 --- a/sound/core/vmaster.c +++ b/sound/core/vmaster.c @@ -262,7 +262,8 @@ int _snd_ctl_add_follower(struct snd_kcontrol *master, return -ENOMEM; srec->kctl = follower; srec->follower = *follower; - memcpy(srec->follower.vd, follower->vd, follower->count * sizeof(*follower->vd)); + memcpy(srec->follower.vd, follower->vd, flex_array_size(srec, follower.vd, + srec->follower.count)); srec->master = master_link; srec->flags = flags; -- 2.27.0

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HP NB right speaker no sound
by Kailang 30 Jul '20

30 Jul '20

Hi Takashi, HP NB platform right side speaker no sound. Attach patch was help EC to initial I2S Amp normally. Thanks. BR, Kailang

2 3

[PATCH] ASoC: fsl-asoc-card: Remove fsl_asoc_card_set_bias_level function
by Shengjiu Wang 30 Jul '20

30 Jul '20

With this case: aplay -Dhw:x 16khz.wav 24khz.wav There is sound distortion for 24khz.wav. The reason is that setting PLL of WM8962 with set_bias_level function, the bias level is not changed when 24khz.wav is played, then the PLL won't be reset, the clock is not correct, so distortion happens. The resolution of this issue is to remove fsl_asoc_card_set_bias_level. Move PLL configuration to hw_params and hw_free. After removing fsl_asoc_card_set_bias_level, also test WM8960 case, it can work. Fixes: 708b4351f08c ("ASoC: fsl: Add Freescale Generic ASoC Sound Card with ASRC support") Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl-asoc-card.c | 149 +++++++++++++++------------------- 1 file changed, 66 insertions(+), 83 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 4848ba61d083..0517dbb3e908 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -73,6 +73,7 @@ struct cpu_priv { * @codec_priv: CODEC private data * @cpu_priv: CPU private data * @card: ASoC card structure + * @is_stream_in_use: flags for release resource in hw_free * @sample_rate: Current sample rate * @sample_format: Current sample format * @asrc_rate: ASRC sample rate used by Back-Ends @@ -89,6 +90,7 @@ struct fsl_asoc_card_priv { struct codec_priv codec_priv; struct cpu_priv cpu_priv; struct snd_soc_card card; + bool is_stream_in_use[2]; u32 sample_rate; snd_pcm_format_t sample_format; u32 asrc_rate; @@ -151,21 +153,17 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct codec_priv *codec_priv = &priv->codec_priv; struct cpu_priv *cpu_priv = &priv->cpu_priv; struct device *dev = rtd->card->dev; + unsigned int pll_out; int ret; priv->sample_rate = params_rate(params); priv->sample_format = params_format(params); + priv->is_stream_in_use[tx] = true; - /* - * If codec-dai is DAI Master and all configurations are already in the - * set_bias_level(), bypass the remaining settings in hw_params(). - * Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS. - */ - if ((priv->card.set_bias_level && - priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) || - fsl_asoc_card_is_ac97(priv)) + if (fsl_asoc_card_is_ac97(priv)) return 0; /* Specific configurations of DAIs starts from here */ @@ -185,12 +183,72 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, return ret; } } + /* Specific configuration for PLL */ + if (codec_priv->pll_id && codec_priv->fll_id) { + if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) + pll_out = priv->sample_rate * 384; + else + pll_out = priv->sample_rate * 256; + + ret = snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0), + codec_priv->pll_id, + codec_priv->mclk_id, + codec_priv->mclk_freq, pll_out); + if (ret) { + dev_err(dev, "failed to start FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), + codec_priv->fll_id, + pll_out, SND_SOC_CLOCK_IN); + + if (ret && ret != -ENOTSUPP) { + dev_err(dev, "failed to set SYSCLK: %d\n", ret); + return ret; + } + } + + return 0; +} + +static int fsl_asoc_card_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(rtd->card); + bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + struct codec_priv *codec_priv = &priv->codec_priv; + struct device *dev = rtd->card->dev; + int ret; + + priv->is_stream_in_use[tx] = false; + + if (!priv->is_stream_in_use[!tx] && codec_priv->pll_id && + codec_priv->fll_id) { + /* Force freq to be 0 to avoid error message in codec */ + ret = snd_soc_dai_set_sysclk(asoc_rtd_to_codec(rtd, 0), + codec_priv->mclk_id, + 0, + SND_SOC_CLOCK_IN); + if (ret) { + dev_err(dev, "failed to switch away from FLL: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(asoc_rtd_to_codec(rtd, 0), + codec_priv->pll_id, 0, 0, 0); + if (ret && ret != -ENOTSUPP) { + dev_err(dev, "failed to stop FLL: %d\n", ret); + return ret; + } + } return 0; } static const struct snd_soc_ops fsl_asoc_card_ops = { .hw_params = fsl_asoc_card_hw_params, + .hw_free = fsl_asoc_card_hw_free, }; static int be_hw_params_fixup(struct snd_soc_pcm_runtime *rtd, @@ -254,75 +312,6 @@ static struct snd_soc_dai_link fsl_asoc_card_dai[] = { }, }; -static int fsl_asoc_card_set_bias_level(struct snd_soc_card *card, - struct snd_soc_dapm_context *dapm, - enum snd_soc_bias_level level) -{ - struct fsl_asoc_card_priv *priv = snd_soc_card_get_drvdata(card); - struct snd_soc_pcm_runtime *rtd; - struct snd_soc_dai *codec_dai; - struct codec_priv *codec_priv = &priv->codec_priv; - struct device *dev = card->dev; - unsigned int pll_out; - int ret; - - rtd = snd_soc_get_pcm_runtime(card, &card->dai_link[0]); - codec_dai = asoc_rtd_to_codec(rtd, 0); - if (dapm->dev != codec_dai->dev) - return 0; - - switch (level) { - case SND_SOC_BIAS_PREPARE: - if (dapm->bias_level != SND_SOC_BIAS_STANDBY) - break; - - if (priv->sample_format == SNDRV_PCM_FORMAT_S24_LE) - pll_out = priv->sample_rate * 384; - else - pll_out = priv->sample_rate * 256; - - ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, - codec_priv->mclk_id, - codec_priv->mclk_freq, pll_out); - if (ret) { - dev_err(dev, "failed to start FLL: %d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->fll_id, - pll_out, SND_SOC_CLOCK_IN); - if (ret && ret != -ENOTSUPP) { - dev_err(dev, "failed to set SYSCLK: %d\n", ret); - return ret; - } - break; - - case SND_SOC_BIAS_STANDBY: - if (dapm->bias_level != SND_SOC_BIAS_PREPARE) - break; - - ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, - codec_priv->mclk_freq, - SND_SOC_CLOCK_IN); - if (ret && ret != -ENOTSUPP) { - dev_err(dev, "failed to switch away from FLL: %d\n", ret); - return ret; - } - - ret = snd_soc_dai_set_pll(codec_dai, codec_priv->pll_id, 0, 0, 0); - if (ret) { - dev_err(dev, "failed to stop FLL: %d\n", ret); - return ret; - } - break; - - default: - break; - } - - return 0; -} - static int fsl_asoc_card_audmux_init(struct device_node *np, struct fsl_asoc_card_priv *priv) { @@ -611,7 +600,6 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Diversify the card configurations */ if (of_device_is_compatible(np, "fsl,imx-audio-cs42888")) { codec_dai_name = "cs42888"; - priv->card.set_bias_level = NULL; priv->cpu_priv.sysclk_freq[TX] = priv->codec_priv.mclk_freq; priv->cpu_priv.sysclk_freq[RX] = priv->codec_priv.mclk_freq; priv->cpu_priv.sysclk_dir[TX] = SND_SOC_CLOCK_OUT; @@ -628,26 +616,22 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8962")) { codec_dai_name = "wm8962"; - priv->card.set_bias_level = fsl_asoc_card_set_bias_level; priv->codec_priv.mclk_id = WM8962_SYSCLK_MCLK; priv->codec_priv.fll_id = WM8962_SYSCLK_FLL; priv->codec_priv.pll_id = WM8962_FLL; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) { codec_dai_name = "wm8960-hifi"; - priv->card.set_bias_level = fsl_asoc_card_set_bias_level; priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) { codec_dai_name = "ac97-hifi"; - priv->card.set_bias_level = NULL; priv->dai_fmt = SND_SOC_DAIFMT_AC97; priv->card.dapm_routes = audio_map_ac97; priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_ac97); } else if (of_device_is_compatible(np, "fsl,imx-audio-mqs")) { codec_dai_name = "fsl-mqs-dai"; - priv->card.set_bias_level = NULL; priv->dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_CBS_CFS | SND_SOC_DAIFMT_NB_NF; @@ -657,7 +641,6 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->card.num_dapm_routes = ARRAY_SIZE(audio_map_tx); } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8524")) { codec_dai_name = "wm8524-hifi"; - priv->card.set_bias_level = NULL; priv->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; priv->dai_link[1].dpcm_capture = 0; priv->dai_link[2].dpcm_capture = 0; -- 2.27.0

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Headset microphone not detected
by Ujjwal Sharma 30 Jul '20

30 Jul '20

Hi everyone! I am unable to get my headset microphone to work, and that's really puzzling to me since the headphone speakers show up as plugged and work perfectly (pulseaudio shows the headset microphone as unplugged). I reported this issue to the SOF issue tracker on GitHub and they pointed out that it's an ALSA issue. Here's the mentioned thread: https://github.com/thesofproject/sof/issues/3164. My alsa_info.sh: https://pastebin.com/raw/K99520LA Some snippets from there that might be interesting: [ 7612.883978] snd_hda_codec_realtek ehdaudio0D0: didn't find PCM for DAI Digital Codec DAI [ 7612.883986] snd_hda_codec_realtek ehdaudio0D0: ASoC: can't open DAI Digital Codec DAI: -22 [ 7612.883992] snd_hda_codec_realtek ehdaudio0D0: didn't find PCM for DAI Digital Codec DAI [ 7612.883998] Digital Playback and Capture: ASoC: BE open failed -22 [ 7612.884002] HDA Digital: ASoC: failed to start some BEs -22 state.sofhdadsp { .... name 'Headset Jack' value false ... name 'Mic Jack' value false I cannot possibly understand what's wrong. Is it a bug in the codecs? Thanks, Ujjwal

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KASAN: use-after-free Read in delete_and_unsubscribe_port (2)
by syzbot 30 Jul '20

30 Jul '20

Hello, syzbot found the following crash on: HEAD commit: 7cc2a8ea Merge tag 'block-5.8-2020-07-01' of git://git.ker.. git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?x=11561e7b100000 kernel config: https://syzkaller.appspot.com/x/.config?x=7be693511b29b338 dashboard link: https://syzkaller.appspot.com/bug?extid=1a54a94bd32716796edd compiler: gcc (GCC) 10.1.0-syz 20200507 userspace arch: i386 Unfortunately, I don't have any reproducer for this crash yet. IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+1a54a94bd32716796edd(a)syzkaller.appspotmail.com ================================================================== BUG: KASAN: use-after-free in list_empty include/linux/list.h:282 [inline] BUG: KASAN: use-after-free in delete_and_unsubscribe_port+0x420/0x4f0 sound/core/seq/seq_ports.c:530 Read of size 8 at addr ffff88809533de50 by task syz-executor.0/29034 CPU: 0 PID: 29034 Comm: syz-executor.0 Not tainted 5.8.0-rc3-syzkaller #0 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:77 [inline] dump_stack+0x18f/0x20d lib/dump_stack.c:118 print_address_description.constprop.0.cold+0xae/0x436 mm/kasan/report.c:383 __kasan_report mm/kasan/report.c:513 [inline] kasan_report.cold+0x1f/0x37 mm/kasan/report.c:530 list_empty include/linux/list.h:282 [inline] delete_and_unsubscribe_port+0x420/0x4f0 sound/core/seq/seq_ports.c:530 snd_seq_port_disconnect+0x433/0x5c0 sound/core/seq/seq_ports.c:610 snd_seq_ioctl_unsubscribe_port+0x1fc/0x400 sound/core/seq/seq_clientmgr.c:1525 snd_seq_ioctl+0x202/0x3e0 sound/core/seq/seq_clientmgr.c:2157 snd_seq_ioctl_compat+0xdc/0x2d0 sound/core/seq/seq_compat.c:109 __do_compat_sys_ioctl+0x1d3/0x230 fs/ioctl.c:847 do_syscall_32_irqs_on+0x3f/0x60 arch/x86/entry/common.c:403 __do_fast_syscall_32 arch/x86/entry/common.c:448 [inline] do_fast_syscall_32+0x7f/0x120 arch/x86/entry/common.c:474 entry_SYSENTER_compat+0x6d/0x7c arch/x86/entry/entry_64_compat.S:138 RIP: 0023:0xf7ff0569 Code: Bad RIP value. RSP: 002b:00000000f5deb0cc EFLAGS: 00000296 ORIG_RAX: 0000000000000036 RAX: ffffffffffffffda RBX: 0000000000000003 RCX: 0000000040505331 RDX: 0000000020000200 RSI: 0000000000000000 RDI: 0000000000000000 RBP: 0000000000000000 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000000 R12: 0000000000000000 R13: 0000000000000000 R14: 0000000000000000 R15: 0000000000000000 Allocated by task 29030: save_stack+0x1b/0x40 mm/kasan/common.c:48 set_track mm/kasan/common.c:56 [inline] __kasan_kmalloc.constprop.0+0xc2/0xd0 mm/kasan/common.c:494 kmem_cache_alloc_trace+0x14f/0x2d0 mm/slab.c:3551 kmalloc include/linux/slab.h:555 [inline] kzalloc include/linux/slab.h:669 [inline] snd_seq_port_connect+0x5d/0x520 sound/core/seq/seq_ports.c:553 snd_seq_ioctl_subscribe_port+0x1fc/0x400 sound/core/seq/seq_clientmgr.c:1484 snd_seq_ioctl+0x202/0x3e0 sound/core/seq/seq_clientmgr.c:2157 snd_seq_ioctl_compat+0xdc/0x2d0 sound/core/seq/seq_compat.c:109 __do_compat_sys_ioctl+0x1d3/0x230 fs/ioctl.c:847 do_syscall_32_irqs_on+0x3f/0x60 arch/x86/entry/common.c:403 __do_fast_syscall_32 arch/x86/entry/common.c:448 [inline] do_fast_syscall_32+0x7f/0x120 arch/x86/entry/common.c:474 entry_SYSENTER_compat+0x6d/0x7c arch/x86/entry/entry_64_compat.S:138 Freed by task 29035: save_stack+0x1b/0x40 mm/kasan/common.c:48 set_track mm/kasan/common.c:56 [inline] kasan_set_free_info mm/kasan/common.c:316 [inline] __kasan_slab_free+0xf5/0x140 mm/kasan/common.c:455 __cache_free mm/slab.c:3426 [inline] kfree+0x103/0x2c0 mm/slab.c:3757 snd_seq_port_disconnect+0x4c1/0x5c0 sound/core/seq/seq_ports.c:614 snd_seq_ioctl_unsubscribe_port+0x1fc/0x400 sound/core/seq/seq_clientmgr.c:1525 snd_seq_ioctl+0x202/0x3e0 sound/core/seq/seq_clientmgr.c:2157 snd_seq_ioctl_compat+0xdc/0x2d0 sound/core/seq/seq_compat.c:109 __do_compat_sys_ioctl+0x1d3/0x230 fs/ioctl.c:847 do_syscall_32_irqs_on+0x3f/0x60 arch/x86/entry/common.c:403 __do_fast_syscall_32 arch/x86/entry/common.c:448 [inline] do_fast_syscall_32+0x7f/0x120 arch/x86/entry/common.c:474 entry_SYSENTER_compat+0x6d/0x7c arch/x86/entry/entry_64_compat.S:138 The buggy address belongs to the object at ffff88809533de00 which belongs to the cache kmalloc-128 of size 128 The buggy address is located 80 bytes inside of 128-byte region [ffff88809533de00, ffff88809533de80) The buggy address belongs to the page: page:ffffea000254cf40 refcount:1 mapcount:0 mapping:0000000000000000 index:0xffff88809533d100 flags: 0xfffe0000000200(slab) raw: 00fffe0000000200 ffffea00029c2e48 ffffea0002257d48 ffff8880aa000700 raw: ffff88809533d100 ffff88809533d000 000000010000000a 0000000000000000 page dumped because: kasan: bad access detected Memory state around the buggy address: ffff88809533dd00: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ffff88809533dd80: fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc >ffff88809533de00: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ^ ffff88809533de80: fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc ffff88809533df00: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ================================================================== --- This bug is generated by a bot. It may contain errors. See https://goo.gl/tpsmEJ for more information about syzbot. syzbot engineers can be reached at syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. See: https://goo.gl/tpsmEJ#status for how to communicate with syzbot.

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[PATCH 0/7] ASoC: merge soc_pcm_open() rollback and soc_pcm_close()
by Kuninori Morimoto 30 Jul '20

30 Jul '20

Hi Mark soc_pcm_open() does rollback when failed (A), but, it is almost same as soc_pcm_close(). static int soc_pcm_open(xxx) { ... if (ret < 0) goto xxx_err; ... return 0; ^ config_err: | ... | rtd_startup_err: (A) ... | component_err: | ... v return ret; } This kind of duplicated code can be a hotbed of bugs, thus, this patch-set share soc_pcm_close() and rollback. Kuninori Morimoto (7): ASoC: soc-dai: add mark for snd_soc_dai_startup/shutdown() ASoC: soc-link: add mark for snd_soc_link_startup/shutdown() ASoC: soc-component: add mark for soc_pcm_components_open/close() ASoC: soc-component: add mark for snd_soc_pcm_component_pm_runtime_get/put() ASoC: soc-pcm: add soc_pcm_clean() and call it from soc_pcm_open/close() ASoC: soc-pcm: remove unneeded dev_err() for snd_soc_dai_startup() ASoC: soc-pcm: remove unneeded dev_err() for snd_soc_component_module/open() include/sound/soc-component.h | 28 +++++--- include/sound/soc-dai.h | 5 +- include/sound/soc-link.h | 3 +- include/sound/soc.h | 3 + sound/soc/soc-component.c | 73 ++++++++++++++++++++- sound/soc/soc-compress.c | 30 +++------ sound/soc/soc-dai.c | 21 +++++- sound/soc/soc-dapm.c | 4 +- sound/soc/soc-link.c | 21 +++++- sound/soc/soc-pcm.c | 120 ++++++++++++---------------------- 10 files changed, 190 insertions(+), 118 deletions(-) -- 2.25.1

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[PATCH RESEND] ASoC: wm8962: Do not access WM8962_GPIO_BASE
by Fabio Estevam 30 Jul '20

30 Jul '20

According to the WM8962 datasheet, there is no register at address 0x200. WM8962_GPIO_BASE is just a base address for the GPIO registers and not a real register, so remove it from wm8962_readable_register(). Also, Register 515 (WM8962_GPIO_BASE + 3) does not exist, so skip its access. This fixes the following errors: wm8962 0-001a: ASoC: error at soc_component_read_no_lock on wm8962.0-001a: -16 wm8962 0-001a: ASoC: error at soc_component_read_no_lock on wm8962.0-001a: -16 Signed-off-by: Fabio Estevam <festevam(a)gmail.com> Acked-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm8962.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/wm8962.c b/sound/soc/codecs/wm8962.c index df8cdc71357d..8159a3866cde 100644 --- a/sound/soc/codecs/wm8962.c +++ b/sound/soc/codecs/wm8962.c @@ -956,7 +956,6 @@ static bool wm8962_readable_register(struct device *dev, unsigned int reg) case WM8962_EQ39: case WM8962_EQ40: case WM8962_EQ41: - case WM8962_GPIO_BASE: case WM8962_GPIO_2: case WM8962_GPIO_3: case WM8962_GPIO_5: @@ -3437,7 +3436,13 @@ static int wm8962_probe(struct snd_soc_component *component) /* Save boards having to disable DMIC when not in use */ dmicclk = false; dmicdat = false; - for (i = 0; i < WM8962_MAX_GPIO; i++) { + for (i = 1; i < WM8962_MAX_GPIO; i++) { + /* + * Register 515 (WM8962_GPIO_BASE + 3) does not exist, + * so skip its access + */ + if (i == 3) + continue; switch (snd_soc_component_read(component, WM8962_GPIO_BASE + i) & WM8962_GP2_FN_MASK) { case WM8962_GPIO_FN_DMICCLK: -- 2.17.1

1 0

[PATCH v3] ASoC: dt-bindings: ak4613: switch to yaml base Documentation
by Kuninori Morimoto 30 Jul '20

30 Jul '20

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> This patch switches from .txt base to .yaml base Document. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Reviewed-by: Rob Herring <robh(a)kernel.org> Link: https://lore.kernel.org/r/87mu4cxlo2.wl-kuninori.morimoto.gx@renesas.com Link: https://lore.kernel.org/r/87o8pf3923.wl-kuninori.morimoto.gx@renesas.com --- v2 -> v3 - add Reviewd-by - add Link - use generic node name on examples .../devicetree/bindings/sound/ak4613.txt | 27 -------- .../devicetree/bindings/sound/ak4613.yaml | 65 +++++++++++++++++++ 2 files changed, 65 insertions(+), 27 deletions(-) delete mode 100644 Documentation/devicetree/bindings/sound/ak4613.txt create mode 100644 Documentation/devicetree/bindings/sound/ak4613.yaml diff --git a/Documentation/devicetree/bindings/sound/ak4613.txt b/Documentation/devicetree/bindings/sound/ak4613.txt deleted file mode 100644 index 49a2e74fd9cb..000000000000 --- a/Documentation/devicetree/bindings/sound/ak4613.txt +++ /dev/null @@ -1,27 +0,0 @@ -AK4613 I2C transmitter - -This device supports I2C mode only. - -Required properties: - -- compatible : "asahi-kasei,ak4613" -- reg : The chip select number on the I2C bus - -Optional properties: -- asahi-kasei,in1-single-end : Boolean. Indicate input / output pins are single-ended. -- asahi-kasei,in2-single-end rather than differential. -- asahi-kasei,out1-single-end -- asahi-kasei,out2-single-end -- asahi-kasei,out3-single-end -- asahi-kasei,out4-single-end -- asahi-kasei,out5-single-end -- asahi-kasei,out6-single-end - -Example: - -&i2c { - ak4613: ak4613@10 { - compatible = "asahi-kasei,ak4613"; - reg = <0x10>; - }; -}; diff --git a/Documentation/devicetree/bindings/sound/ak4613.yaml b/Documentation/devicetree/bindings/sound/ak4613.yaml new file mode 100644 index 000000000000..5aae6126c540 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4613.yaml @@ -0,0 +1,65 @@ +# SPDX-License-Identifier: GPL-2.0 +%YAML 1.2 +--- +$id: http://devicetree.org/schemas/sound/ak4613.yaml# +$schema: http://devicetree.org/meta-schemas/core.yaml# + +title: AK4613 I2C transmitter Device Tree Bindings + +maintainers: + - Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> + +properties: + compatible: + const: asahi-kasei,ak4613 + + reg: + maxItems: 1 + + clocks: + maxItems: 1 + + "#sound-dai-cells": + const: 0 + + # for OF-graph + port: + $ref: "audio-graph-card.yaml#definitions/port" + ports: + $ref: "audio-graph-card.yaml#definitions/ports" + +# use patternProperties to avoid naming "xxx,yyy" issue +patternProperties: + "^asahi-kasei,in1-single-end$": + $ref: /schemas/types.yaml#/definitions/flag + "^asahi-kasei,in2-single-end$": + $ref: /schemas/types.yaml#/definitions/flag + "^asahi-kasei,out1-single-end$": + $ref: /schemas/types.yaml#/definitions/flag + "^asahi-kasei,out2-single-end$": + $ref: /schemas/types.yaml#/definitions/flag + "^asahi-kasei,out3-single-end$": + $ref: /schemas/types.yaml#/definitions/flag + "^asahi-kasei,out4-single-end$": + $ref: /schemas/types.yaml#/definitions/flag + "^asahi-kasei,out5-single-end$": + $ref: /schemas/types.yaml#/definitions/flag + "^asahi-kasei,out6-single-end$": + $ref: /schemas/types.yaml#/definitions/flag + +required: + - compatible + - reg + +additionalProperties: false + +examples: + - | + i2c { + #address-cells = <1>; + #size-cells = <0>; + ak4613: codec@10 { + compatible = "asahi-kasei,ak4613"; + reg = <0x10>; + }; + }; -- 2.25.1

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[PATCH][next] ALSA: hda_codec: Use flex_array_size() helper in memcpy()
by Gustavo A. R. Silva 30 Jul '20

30 Jul '20

Make use of the flex_array_size() helper to calculate the size of a flexible array member within an enclosing structure. This helper offers defense-in-depth against potential integer overflows and makes it explicitly clear that we are dealing with a flexible array member. Signed-off-by: Gustavo A. R. Silva <gustavoars(a)kernel.org> --- sound/pci/hda/hda_codec.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index 58c698f4d131..453c4ec77c48 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -113,7 +113,7 @@ static int add_conn_list(struct hda_codec *codec, hda_nid_t nid, int len, return -ENOMEM; p->len = len; p->nid = nid; - memcpy(p->conns, list, len * sizeof(hda_nid_t)); + memcpy(p->conns, list, flex_array_size(p, conns, len)); list_add(&p->list, &codec->conn_list); return 0; } -- 2.27.0

1 0

Race condition when closing the Jack plugin
by GitHub issues - opened 29 Jul '20

29 Jul '20

alsa-project/alsa-plugins issue #11 was opened from jeberger: There is a potential race condition that can cause segmentation faults if the Jack process callback is called between the call to `snd_pcm_jack_stop` and `snd_pcm_jack_hw_free`. This was introduced by revision badd921. Issue URL : https://github.com/alsa-project/alsa-plugins/issues/11 Repository URL: https://github.com/alsa-project/alsa-plugins

1 0