May 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ALSA: hda - Enforce CONFIG_SND_DYNAMIC_MINORS for HDMI/DP codec
by Takashi Iwai 02 May '18

02 May '18

The DP-MST support requires more PCM streams than usual, hence CONFIG_SND_DYNAMIC_MINORS is almost mandatory. Currently the driver just warns and continues even if streams are missing, but it doesn't seem to enough convince users to switch to the modern setup. This patch adds the enforced selection of CONFIG_SND_DYNAMIC_MINORS from CONFIG_SND_HDA_CODEC_HDMI for covering that. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/Kconfig | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index f7a492c382d9..4235907b7858 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -127,11 +127,15 @@ comment "Set to Y if you want auto-loading the codec driver" config SND_HDA_CODEC_HDMI tristate "Build HDMI/DisplayPort HD-audio codec support" + select SND_DYNAMIC_MINORS help Say Y or M here to include HDMI and DisplayPort HD-audio codec support in snd-hda-intel driver. This includes all AMD/ATI, Intel and Nvidia HDMI/DisplayPort codecs. + Note that this option mandatorily enables CONFIG_SND_DYNAMIC_MINORS + to assure the multiple streams for DP-MST support. + comment "Set to Y if you want auto-loading the codec driver" depends on SND_HDA=y && SND_HDA_CODEC_HDMI=m -- 2.16.3

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[alsa-devel] [PATCH 0/5] conf/ucm: chtrt5645: Various updates
by Hans de Goede 02 May '18

02 May '18

Hi All, Here is a series of chtrt5645 UCM updates, the first 2 patches bring in some fixes for the generic config from: https://github.com/plbossart/UCM The 3th patch adds some analog mic input fixes, together these 3 patches bring the generic chtrt5645 config in sync with the newer chtrt5645-mono-speaker-analog-mic config, but then for stereo speakers. The 4th and the 5th patch add long-name using board/model specific configs for the Asus T100HA and Lenovo Miix 320, these use a digital mic connected to the DMIC1 resp. DMIC2. input. The generic chtrt5645 config has been tested on a Lenovo Miix 310 and a Kazam Vision tablet, both of which feature stereo speakers and an analog microphone. The Asus T100HA and Lenovo Miix 320 configs have been tested on the resp. models they apply to. Regards, Hans

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[alsa-devel] WARNING in __snd_pcm_lib_xfer
by syzbot 02 May '18

02 May '18

Hello, syzbot found the following crash on: HEAD commit: fff75eb2a08c Merge tag 'errseq-v4.17' of git://git.kernel.o... git tree: upstream console output: https://syzkaller.appspot.com/x/log.txt?id=6530078334779392 kernel config: https://syzkaller.appspot.com/x/.config?id=6493557782959164711 dashboard link: https://syzkaller.appspot.com/bug?extid=1dac3a4f6bc9c1c675d4 compiler: gcc (GCC) 8.0.1 20180413 (experimental) userspace arch: i386 syzkaller repro:https://syzkaller.appspot.com/x/repro.syz?id=5691583680741376 IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+1dac3a4f6bc9c1c675d4(a)syzkaller.appspotmail.com random: sshd: uninitialized urandom read (32 bytes read) random: sshd: uninitialized urandom read (32 bytes read) random: sshd: uninitialized urandom read (32 bytes read) random: sshd: uninitialized urandom read (32 bytes read) IPVS: ftp: loaded support on port[0] = 21 WARNING: CPU: 1 PID: 4520 at sound/core/pcm_lib.c:2057 pcm_sanity_check sound/core/pcm_lib.c:2057 [inline] WARNING: CPU: 1 PID: 4520 at sound/core/pcm_lib.c:2057 __snd_pcm_lib_xfer+0x7c3/0x1d10 sound/core/pcm_lib.c:2122 Kernel panic - not syncing: panic_on_warn set ... CPU: 1 PID: 4520 Comm: syz-executor0 Not tainted 4.17.0-rc3+ #52 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+0x1b9/0x294 lib/dump_stack.c:113 panic+0x22f/0x4de kernel/panic.c:184 __warn.cold.8+0x163/0x1b3 kernel/panic.c:536 report_bug+0x252/0x2d0 lib/bug.c:186 fixup_bug arch/x86/kernel/traps.c:178 [inline] do_error_trap+0x1de/0x490 arch/x86/kernel/traps.c:296 do_invalid_op+0x1b/0x20 arch/x86/kernel/traps.c:315 invalid_op+0x14/0x20 arch/x86/entry/entry_64.S:992 RIP: 0010:pcm_sanity_check sound/core/pcm_lib.c:2057 [inline] RIP: 0010:__snd_pcm_lib_xfer+0x7c3/0x1d10 sound/core/pcm_lib.c:2122 RSP: 0018:ffff8801d97b7898 EFLAGS: 00010293 RAX: ffff8801ad00e0c0 RBX: ffff8801d965e100 RCX: 0000000000000000 RDX: 0000000000000000 RSI: ffffffff85c3d333 RDI: ffff8801d965e418 RBP: ffff8801d97b7b38 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000003 R11: 0000000000000000 R12: ffffffff884945c0 R13: 0000000000000000 R14: ffff8801d965e100 R15: ffff8801cdd2c940 snd_pcm_lib_readv include/sound/pcm.h:1113 [inline] snd_pcm_ioctl_xfern_compat+0x266/0x330 sound/core/pcm_compat.c:448 snd_pcm_ioctl_compat+0xc96/0x1c90 sound/core/pcm_compat.c:716 __do_compat_sys_ioctl fs/compat_ioctl.c:1461 [inline] __se_compat_sys_ioctl fs/compat_ioctl.c:1407 [inline] __ia32_compat_sys_ioctl+0x221/0x640 fs/compat_ioctl.c:1407 do_syscall_32_irqs_on arch/x86/entry/common.c:323 [inline] do_fast_syscall_32+0x345/0xf9b arch/x86/entry/common.c:394 entry_SYSENTER_compat+0x70/0x7f arch/x86/entry/entry_64_compat.S:139 RIP: 0023:0xf7f51cb9 RSP: 002b:00000000ffafd6ac EFLAGS: 00000282 ORIG_RAX: 0000000000000036 RAX: ffffffffffffffda RBX: 0000000000000003 RCX: 00000000800c4153 RDX: 0000000020000040 RSI: 0000000000000000 RDI: 0000000000000000 RBP: 0000000000000000 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000292 R12: 0000000000000000 R13: 0000000000000000 R14: 0000000000000000 R15: 0000000000000000 Dumping ftrace buffer: (ftrace buffer empty) Kernel Offset: disabled Rebooting in 86400 seconds.. --- 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. If you forgot to add the Reported-by tag, once the fix for this bug is merged into any tree, please reply to this email with: #syz fix: exact-commit-title If you want to test a patch for this bug, please reply with: #syz test: git://repo/address.git branch and provide the patch inline or as an attachment. To mark this as a duplicate of another syzbot report, please reply with: #syz dup: exact-subject-of-another-report If it's a one-off invalid bug report, please reply with: #syz invalid Note: if the crash happens again, it will cause creation of a new bug report. Note: all commands must start from beginning of the line in the email body.

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[alsa-devel] [alsa-lib][PATCH 1/2] ucm: adding the folder of card_long_name when finding verb conf file
by Hui Wang 02 May '18

02 May '18

The board configuration file and verb conf file are allowed to be in the folder named of card_long_name, so when finding the verb conf file, we need to check if it is in the folder of card_long_name or card_name. Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- src/ucm/parser.c | 10 ++++++++-- 1 file changed, 8 insertions(+), 2 deletions(-) diff --git a/src/ucm/parser.c b/src/ucm/parser.c index 2d76152..219edb9 100644 --- a/src/ucm/parser.c +++ b/src/ucm/parser.c @@ -1056,6 +1056,7 @@ static int parse_verb_file(snd_use_case_mgr_t *uc_mgr, char filename[MAX_FILE]; char *env = getenv(ALSA_CONFIG_UCM_VAR); int err; + char *folder_name; /* allocate verb */ verb = calloc(1, sizeof(struct use_case_verb)); @@ -1082,12 +1083,17 @@ static int parse_verb_file(snd_use_case_mgr_t *uc_mgr, } /* open Verb file for reading */ + if (!strncmp(uc_mgr->conf_file_name, uc_mgr->card_long_name, MAX_CARD_LONG_NAME)) + folder_name = uc_mgr->card_long_name; + else + folder_name = uc_mgr->card_name; + if (env) snprintf(filename, sizeof(filename), "%s/%s/%s", - env, uc_mgr->card_name, file); + env, folder_name, file); else snprintf(filename, sizeof(filename), "%s/ucm/%s/%s", - snd_config_topdir(), uc_mgr->card_name, file); + snd_config_topdir(), folder_name, file); filename[sizeof(filename)-1] = '\0'; err = uc_mgr_config_load(filename, &cfg); -- 2.7.4

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[alsa-devel] [PATCH] ASoC: mediatek: avoid using snd_soc_platform
by KaiChieh Chuang 01 May '18

01 May '18

avoid using snd_soc_platform, which is removed after 4.18 Signed-off-by: KaiChieh Chuang <kaichieh.chuang(a)mediatek.com> --- .../soc/mediatek/common/mtk-afe-platform-driver.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/common/mtk-afe-platform-driver.c b/sound/soc/mediatek/common/mtk-afe-platform-driver.c index d046ea8..8966ee1 100644 --- a/sound/soc/mediatek/common/mtk-afe-platform-driver.c +++ b/sound/soc/mediatek/common/mtk-afe-platform-driver.c @@ -62,7 +62,7 @@ static int mtk_afe_pcm_new(struct snd_soc_pcm_runtime *rtd) size = afe->mtk_afe_hardware->buffer_bytes_max; return snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, - rtd->platform->dev, + afe->dev, size, size); } -- 1.7.9.5

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[alsa-devel] Applied "ASoC: Intel: cht-bsw-rt5672: allow for topology-defined codec-dai setup" to the asoc tree
by Mark Brown 01 May '18

01 May '18

The patch ASoC: Intel: cht-bsw-rt5672: allow for topology-defined codec-dai setup has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From bf14adcc4ddd101088237179bee6daa19bac1669 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Fri, 27 Apr 2018 16:36:00 -0500 Subject: [PATCH] ASoC: Intel: cht-bsw-rt5672: allow for topology-defined codec-dai setup Hard-coded setups conflict with topology defined ones. Move this code to codec_fixup so that SOF can override codec dai settings, e.g. to only use 2 channels. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Acked-by: Vinod Koul <vkoul(a)kernel.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/cht_bsw_rt5672.c | 30 +++++++++++++++++-------- 1 file changed, 21 insertions(+), 9 deletions(-) diff --git a/sound/soc/intel/boards/cht_bsw_rt5672.c b/sound/soc/intel/boards/cht_bsw_rt5672.c index e68ec32720a4..e5aa13058dd7 100644 --- a/sound/soc/intel/boards/cht_bsw_rt5672.c +++ b/sound/soc/intel/boards/cht_bsw_rt5672.c @@ -189,13 +189,6 @@ static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) if (devm_acpi_dev_add_driver_gpios(component->dev, cht_rt5672_gpios)) dev_warn(runtime->dev, "Unable to add GPIO mapping table\n"); - /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ - ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xF, 0xF, 4, 24); - if (ret < 0) { - dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret); - return ret; - } - /* Select codec ASRC clock source to track I2S1 clock, because codec * is in slave mode and 100fs I2S format (BCLK = 100 * LRCLK) cannot * be supported by RT5672. Otherwise, ASRC will be disabled and cause @@ -252,6 +245,7 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, SNDRV_PCM_HW_PARAM_RATE); struct snd_interval *channels = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + int ret; /* The DSP will covert the FE rate to 48k, stereo, 24bits */ rate->min = rate->max = 48000; @@ -259,6 +253,26 @@ static int cht_codec_fixup(struct snd_soc_pcm_runtime *rtd, /* set SSP2 to 24-bit */ params_set_format(params, SNDRV_PCM_FORMAT_S24_LE); + + /* + * Default mode for SSP configuration is TDM 4 slot + */ + ret = snd_soc_dai_set_fmt(rtd->codec_dai, + SND_SOC_DAIFMT_DSP_B | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBS_CFS); + if (ret < 0) { + dev_err(rtd->dev, "can't set format to TDM %d\n", ret); + return ret; + } + + /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ + ret = snd_soc_dai_set_tdm_slot(rtd->codec_dai, 0xF, 0xF, 4, 24); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec TDM slot %d\n", ret); + return ret; + } + return 0; } @@ -315,8 +329,6 @@ static struct snd_soc_dai_link cht_dailink[] = { .nonatomic = true, .codec_dai_name = "rt5670-aif1", .codec_name = "i2c-10EC5670:00", - .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_IB_NF - | SND_SOC_DAIFMT_CBS_CFS, .init = cht_codec_init, .be_hw_params_fixup = cht_codec_fixup, .dpcm_playback = 1, -- 2.17.0

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[alsa-devel] [PATCH] ASoC: mediatek: add the .probe() callback in mt2701_afe_pcm_dai_component
by Ryder Lee 01 May '18

01 May '18

For the sake of uniformity, this patch adds a callback mt2701_afe_pcm_probe() in mt2701_afe_pcm_dai_component to retrieve the regmap - the canonical way to obtain the pointer.. Doing so, we could switch to use devm_snd_soc_register_component() to register the component driver. Signed-off-by: Ryder Lee <ryder.lee(a)mediatek.com> --- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 31 ++++++++++++++---------------- 1 file changed, 14 insertions(+), 17 deletions(-) diff --git a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c index d6eeb4c..828d11c 100644 --- a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c +++ b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c @@ -958,7 +958,17 @@ static int mt2701_irq_fs(struct snd_pcm_substream *substream, unsigned int rate) { "O31", "I35 Switch", "I35" }, }; +static int mt2701_afe_pcm_probe(struct snd_soc_component *component) +{ + struct mtk_base_afe *afe = snd_soc_component_get_drvdata(component); + + snd_soc_component_init_regmap(component, afe->regmap); + + return 0; +} + static const struct snd_soc_component_driver mt2701_afe_pcm_dai_component = { + .probe = mt2701_afe_pcm_probe, .name = "mt2701-afe-pcm-dai", .dapm_widgets = mt2701_afe_pcm_widgets, .num_dapm_widgets = ARRAY_SIZE(mt2701_afe_pcm_widgets), @@ -1316,22 +1326,6 @@ static int mt2701_afe_runtime_resume(struct device *dev) return mt2701_afe_enable_clock(afe); } -static int mt2701_afe_add_component(struct mtk_base_afe *afe) -{ - struct snd_soc_component *component; - - component = kzalloc(sizeof(*component), GFP_KERNEL); - if (!component) - return -ENOMEM; - - component->regmap = afe->regmap; - - return snd_soc_add_component(afe->dev, component, - &mt2701_afe_pcm_dai_component, - mt2701_afe_pcm_dais, - ARRAY_SIZE(mt2701_afe_pcm_dais)); -} - static int mt2701_afe_pcm_dev_probe(struct platform_device *pdev) { struct mtk_base_afe *afe; @@ -1442,7 +1436,10 @@ static int mt2701_afe_pcm_dev_probe(struct platform_device *pdev) goto err_platform; } - ret = mt2701_afe_add_component(afe); + ret = devm_snd_soc_register_component(&pdev->dev, + &mt2701_afe_pcm_dai_component, + mt2701_afe_pcm_dais, + ARRAY_SIZE(mt2701_afe_pcm_dais)); if (ret) { dev_warn(dev, "err_dai_component\n"); goto err_platform; -- 1.9.1

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[alsa-devel] [PATCH][next] ASoC: amd: fix spelling mistake: "failer" -> "failure"
by Colin King 01 May '18

01 May '18

From: Colin Ian King <colin.king(a)canonical.com> Trivial fix to spelling mistake in dev_err error message Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/amd/acp-pcm-dma.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 9c026c4d26d4..2b9a020e0b3e 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -1014,7 +1014,7 @@ static int acp_dma_new(struct snd_soc_pcm_runtime *rtd) } if (ret < 0) dev_err(component->dev, - "buffer preallocation failer error:%d\n", ret); + "buffer preallocation failure error:%d\n", ret); return ret; } -- 2.17.0

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[alsa-devel] [PATCH RFC] ASoC: dai playback_active and capture_active may be greater than 1
by Jerome Brunet 01 May '18

01 May '18

At the moment playback_active and capture_active are using only 1 bit so the maximum active count is 1. However, snd_soc_runtime_activate() may be called several time on the same dai. This happens when a dai is part of several dai_links. It is often the case for "snd-soc-dummy-dai". This is a problem if snd_soc_runtime_activate() is called an even number of times on a dai. In this case the active count overflow back to 0. As consequence, ASoC functions, such as soc_dpcm_runtime_update(), won't run correctly. Storing these usage counts on plain 'unsigned int' solves the problem. Fixes: f0fba2ad1b6b ("ASoC: multi-component - ASoC Multi-Component Support") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- Hi, I found this problem while working on ASoC support for a new platform. During a playback, using DPCM, if I changed the backend dai, nothing happened, old backend path was not pruned, new backend path was not added. Here is how it goes: 1 FE and 2 BE, all 3 using snd-soc-dummy-dai for the codec_dai (ATM). On playback start: - BE #1 activates and increments snd-soc-dummy-dai's playback_active - FE activates and increments snd-soc-dummy-dai's playback_active The playback works but snd-soc-dummy-dai's playback_active value is now 0. Through a mux kcontrol, change BE client of FE, from BE #1 to BE #2. -> trigger soc_dpcm_runtime_update(). -> checks fe's codec_dai playback_active which is zero so playback path processing is skipped, keeping the old invalid path. include/sound/soc-dai.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 8ad11669e4d8..35ebb0be5114 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -294,8 +294,8 @@ struct snd_soc_dai { struct snd_soc_dai_driver *driver; /* DAI runtime info */ - unsigned int capture_active:1; /* stream is in use */ - unsigned int playback_active:1; /* stream is in use */ + unsigned int capture_active; /* stream usage count */ + unsigned int playback_active; /* stream usage count */ unsigned int probed:1; unsigned int active; -- 2.14.3

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[alsa-devel] [PATCH v3 0/3] ASoc: TAS6424: Add support for mute, standby, and faster power on
by Jean-Jacques Hiblot 01 May '18

01 May '18

mute and standby pins are available on the codec. If they are connected, they should be managed by the driver, instead of relying on gpio hogs or on the initial state of the GPIOs. This series also includes a patch to improve the start-up time of the channels by disabling built-in DC diagnostics. Those diagnosdtics basically serve to detect : - wires shorted together - wire shorted to ground or vbat - wire disconnected This is not useful for all platforms and the addition to the startup time is quite noticeable (230ms). The diagnostics can be enabled/disabled using a dedicated ALSA control. Changes in v3: - use SOC_SINGLE_STROBE() to register the "auto diagniostics switch" control Changes in v2: - use a ALSA control to enable/disable the auto diagnostics instead of relying on a DTS property. - fixed typos - use gpios (plural) instead of gpio in the documentation of the bindings Jean-Jacques Hiblot (3): ASoC: tas6424: Add support for the standby pin ASoC: tas6424: Add support for the mute pin ASoC: tas6424: Allow disabling auto diagnostics for faster power-on .../devicetree/bindings/sound/ti,tas6424.txt | 2 + sound/soc/codecs/tas6424.c | 72 +++++++++++++++++++++- sound/soc/codecs/tas6424.h | 4 ++ 3 files changed, 75 insertions(+), 3 deletions(-) -- 2.7.4

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