April 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] alsa kernel panic
by Victor L 02 Apr '19

02 Apr '19

Hi , I have Linux panic when started play music with one of my USB audio device. first please see this logs: [ 71.684730] usbcore: registered new interface driver snd-usb-audio [ 73.835355] BUG: unable to handle kernel NULL pointer dereference at 0000000000000008 [ 73.835412] IP: [<ffffffffc0b45d3c>] snd_usb_endpoint_start+0x23c/0x2f0 [snd_usb_audio] [ 73.835414] PGD 0 [ 73.835415] [ 73.835418] Oops: 0000 [#1] PREEMPT SMP [ 73.835482] Modules linked in: snd_usb_audio snd_usbmidi_lib snd_hwdep snd_pcm_oss snd_mixer_oss snd_ pcm snd_seq_midi snd_seq_midi_event snd_rawmidi snd_seq snd_seq_device snd_timer snd soundcore coretemp iwl3945 iwlegacy i915 ehci_pci iTCO_wdt ehci_hcd pcmcia iTCO_vendor_support kvm usbcore mac80211 e1000e yenta_socket sr_mod pcmcia_rsrc cfg80211 pcmcia_core lpc_ich mfd_core drm_kms_helper cdrom rfkill ata_ge neric irqbypass ptp pps_core joydev sg pcspkr i2c_i801 drm battery i2c_algo_bit ata_piix i2c_smbus usb_c ommon shpchp rng_core ac acpi_cpufreq video button nls_utf8 cifs sha256_ssse3 cmac md4 des_generic arc4 dns_resolver fscache sunrpc loop ip_tables x_tables autofs4 ext4 crc16 jbd2 crc32c_generic fscrypto ecb glue_helper lrw gf128mul ablk_helper cryptd aes_x86_64 mbcache sd_mod ahci libahci [ 73.835486] evdev libata psmouse serio_raw scsi_mod thermal [ 73.835491] CPU: 1 PID: 1099 Comm: output:oops Not tainted 4.9.144-rt93 #1 [ 73.835492] Hardware name: LENOVO 636962M/636962M, BIOS 7IET38WW (1.19 ) 03/22/2011 [ 73.835494] task: ffff8beb97502c40 task.stack: ffffac1901068000 [ 73.835505] RIP: 0010:[<ffffffffc0b45d3c>] [<ffffffffc0b45d3c>] snd_usb_endpoint_start+0x23c/0x2f0 [ snd_usb_audio] [ 73.835507] RSP: 0000:ffffac190106bcc0 EFLAGS: 00010246 [ 73.835508] RAX: 00000000ffffffea RBX: ffff8beb96c00000 RCX: 00000000ffffffea [ 73.835509] RDX: 0000000000000000 RSI: ffffffffc0b5bf98 RDI: ffff8beb99a18000 [ 73.835511] RBP: 0000000000000002 R08: ffffffffc0b5a8d3 R09: 0000000000000000 [ 73.835512] R10: 00000000fffffe00 R11: 0000000000000200 R12: ffff8beb96c80070 [ 73.835513] R13: ffff8beb99a18000 R14: 0000000000000002 R15: 0000000000000000 [ 73.835516] FS: 00007fdfedc66700(0000) GS:ffff8bebbf300000(0000) knlGS:0000000000000000 [ 73.835518] CS: 0010 DS: 0000 ES: 0000 CR0: 0000000080050033 [ 73.835519] CR2: 0000000000000008 CR3: 00000000b4ab6000 CR4: 0000000000000670 [ 73.835520] Stack: [ 73.835525] ffff8bebb8888c18 ffff8beb96c00000 0000000000000000 0000000000008002 [ 73.835528] 00007fdfe40197d0 ffffffffc0b4e7cd ffff8bebb8888c18 ffff8beb974b2400 [ 73.835531] ffffffffc0b5088a ffff8beb0000ac44 ffff8bebb206d300 0000000000000000 [ 73.835531] Call Trace: [ 73.835545] [<ffffffffc0b4e7cd>] ? start_endpoints+0x3d/0x150 [snd_usb_audio] [ 73.835555] [<ffffffffc0b5088a>] ? snd_usb_pcm_prepare+0x2ea/0x520 [snd_usb_audio] [ 73.835577] [<ffffffffc0b0c176>] ? snd_pcm_sw_params_user+0x56/0x90 [snd_pcm] [ 73.835584] [<ffffffffc0b0b2d9>] ? snd_pcm_do_prepare+0x19/0x30 [snd_pcm] [ 73.835591] [<ffffffffc0b09ea6>] ? snd_pcm_action_single+0x36/0x80 [snd_pcm] [ 73.835598] [<ffffffffc0b0a311>] ? snd_pcm_action_nonatomic+0x61/0x70 [snd_pcm] [ 73.835606] [<ffffffffc0b0e219>] ? snd_pcm_common_ioctl1+0x3a9/0xc60 [snd_pcm] [ 73.835674] [<ffffffffc03a73ce>] ? ext4_file_write_iter+0x22e/0x370 [ext4] [ 73.835682] [<ffffffffc0b0ecaa>] ? snd_pcm_playback_ioctl1+0x1da/0x240 [snd_pcm] [ 73.835690] [<ffffffffb9736a38>] ? __seccomp_filter+0x78/0x2a0 [ 73.835698] [<ffffffffc0b0ed33>] ? snd_pcm_playback_ioctl+0x23/0x30 [snd_pcm] [ 73.835702] [<ffffffffb982c9c4>] ? do_vfs_ioctl+0xa4/0x620 [ 73.835706] [<ffffffffb96033b4>] ? syscall_trace_enter+0x144/0x300 [ 73.835709] [<ffffffffb982cfb4>] ? SyS_ioctl+0x74/0x80 [ 73.835711] [<ffffffffb9603c56>] ? do_syscall_64+0x76/0x100 [ 73.835717] [<ffffffffb9c38b4e>] ? entry_SYSCALL_64_after_swapgs+0x58/0xc6 [ 73.835747] Code: c0 d3 a8 b5 c0 83 f8 a6 74 15 83 f8 e0 49 c7 c0 8c a8 b5 c0 48 c7 c2 74 a8 b5 c0 4c 0f 45 c2 48 8b 13 89 c1 48 c7 c6 98 bf b5 c0 <48> 8b 7a 08 89 ea 48 81 c7 98 00 00 00 e8 e2 7b f4 f8 f0 80 63 [ 73.835757] RIP [<ffffffffc0b45d3c>] snd_usb_endpoint_start+0x23c/0x2f0 [snd_usb_audio] [ 73.835758] RSP <ffffac190106bcc0> [ 73.835759] CR2: 0000000000000008 I give you more detail, I have Debian Linux 4.9.144-rt93. I tested under Debian 4.19 and Ubuntu same panic present. Please help me to fix this problem, this device working fine under Windows. I believe this problem in Linux audio driver. I ready to provide any additional details if needs. -- Regards, Viktor L

1 2

[alsa-devel] [PATCH] ASoC: dpcm: skip missing substream while applying symmetry
by Jerome Brunet 02 Apr '19

02 Apr '19

If for any reason, the backend does not have the requested substream (like capture on a playback only backend), the BE will be skipped in dpcm_be_dai_startup(). However, dpcm_apply_symmetry() does not skip those BE and will dereference the be_substream (NULL) pointer anyway. Like in dpcm_be_dai_startup(), just skip those BE. Fixes: 906c7d690c3b ("ASoC: dpcm: Apply symmetry for DPCM") Signed-off-by: Jerome Brunet <jbrunet(a)baylibre.com> --- sound/soc/soc-pcm.c | 7 ++++++- 1 file changed, 6 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 0d5ec68a1e50..99bbd724d2a6 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1899,10 +1899,15 @@ static int dpcm_apply_symmetry(struct snd_pcm_substream *fe_substream, struct snd_soc_pcm_runtime *be = dpcm->be; struct snd_pcm_substream *be_substream = snd_soc_dpcm_get_substream(be, stream); - struct snd_soc_pcm_runtime *rtd = be_substream->private_data; + struct snd_soc_pcm_runtime *rtd; struct snd_soc_dai *codec_dai; int i; + /* A backend may not have the requested substream */ + if (!be_substream) + continue; + + rtd = be_substream->private_data; if (rtd->dai_link->be_hw_params_fixup) continue; -- 2.20.1

2 1

[alsa-devel] [PATCH v4 1/2] ASoC: tlv320aic31xx: Add headphone/headset detection
by Andrew F. Davis 02 Apr '19

02 Apr '19

This device can detect the insertion/removal of headphones and headsets. Enable reporting this status by enabling this interrupt and forwarding this to upper-layers if a jack has been defined. This jack definition and the resulting operation from a jack detection event must currently be defined by sound card platform code until CODEC outputs to jack mappings can be defined generically. Signed-off-by: Andrew F. Davis <afd(a)ti.com> --- Changes from v3: - Rebase on broonie/for-next sound/soc/codecs/tlv320aic31xx.c | 48 +++++++++++++++++++++++++++++++- sound/soc/codecs/tlv320aic31xx.h | 11 ++++++++ 2 files changed, 58 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index c544a1e35f5e..a60cd8c08a14 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -25,6 +25,7 @@ #include <linux/of_gpio.h> #include <linux/slab.h> #include <sound/core.h> +#include <sound/jack.h> #include <sound/pcm.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -89,6 +90,7 @@ static bool aic31xx_volatile(struct device *dev, unsigned int reg) case AIC31XX_INTRADCFLAG: /* Sticky interrupt flags */ case AIC31XX_INTRDACFLAG2: case AIC31XX_INTRADCFLAG2: + case AIC31XX_HSDETECT: return true; } return false; @@ -163,6 +165,7 @@ struct aic31xx_priv { struct aic31xx_pdata pdata; struct regulator_bulk_data supplies[AIC31XX_NUM_SUPPLIES]; struct aic31xx_disable_nb disable_nb[AIC31XX_NUM_SUPPLIES]; + struct snd_soc_jack *jack; unsigned int sysclk; u8 p_div; int rate_div_line; @@ -1261,6 +1264,20 @@ static int aic31xx_set_bias_level(struct snd_soc_component *component, return 0; } +int aic31xx_set_jack(struct snd_soc_component *component, + struct snd_soc_jack *jack, void *data) +{ + struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component); + + aic31xx->jack = jack; + + /* Enable/Disable jack detection */ + regmap_write(aic31xx->regmap, AIC31XX_HSDETECT, + jack ? AIC31XX_HSD_ENABLE : 0); + + return 0; +} + static int aic31xx_codec_probe(struct snd_soc_component *component) { struct aic31xx_priv *aic31xx = snd_soc_component_get_drvdata(component); @@ -1301,6 +1318,7 @@ static int aic31xx_codec_probe(struct snd_soc_component *component) static const struct snd_soc_component_driver soc_codec_driver_aic31xx = { .probe = aic31xx_codec_probe, + .set_jack = aic31xx_set_jack, .set_bias_level = aic31xx_set_bias_level, .controls = common31xx_snd_controls, .num_controls = ARRAY_SIZE(common31xx_snd_controls), @@ -1405,8 +1423,35 @@ static irqreturn_t aic31xx_irq(int irq, void *data) dev_err(dev, "Short circuit on Left output is detected\n"); if (value & AIC31XX_HPRSCDETECT) dev_err(dev, "Short circuit on Right output is detected\n"); + if (value & AIC31XX_HSPLUG) { + unsigned int val; + int status = 0; + + ret = regmap_read(aic31xx->regmap, AIC31XX_HSDETECT, &val); + if (ret) { + dev_err(dev, "Failed to read headset type: %d\n", ret); + goto exit; + } + + switch ((val & AIC31XX_HSD_TYPE_MASK) >> + AIC31XX_HSD_TYPE_SHIFT) { + case AIC31XX_HSD_HP: + status |= SND_JACK_HEADPHONE; + break; + case AIC31XX_HSD_HS: + status |= SND_JACK_HEADSET; + break; + default: + break; + } + + if (aic31xx->jack) + snd_soc_jack_report(aic31xx->jack, status, + AIC31XX_JACK_MASK); + } if (value & ~(AIC31XX_HPLSCDETECT | - AIC31XX_HPRSCDETECT)) + AIC31XX_HPRSCDETECT | + AIC31XX_HSPLUG)) dev_err(dev, "Unknown DAC interrupt flags: 0x%08x\n", value); read_overflow: @@ -1518,6 +1563,7 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, AIC31XX_GPIO1_FUNC_SHIFT); regmap_write(aic31xx->regmap, AIC31XX_INT1CTRL, + AIC31XX_HSPLUGDET | AIC31XX_SC | AIC31XX_ENGINE); diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index 2636f2c6bc79..9985ceed97ae 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -20,6 +20,9 @@ #define AIC31XX_MINIDSP_BIT BIT(2) #define DAC31XX_BIT BIT(3) +#define AIC31XX_JACK_MASK (SND_JACK_HEADPHONE | \ + SND_JACK_HEADSET) + enum aic31xx_type { AIC3100 = 0, AIC3110 = AIC31XX_STEREO_CLASS_D_BIT, @@ -220,6 +223,14 @@ struct aic31xx_pdata { /* AIC31XX_DACMUTE */ #define AIC31XX_DACMUTE_MASK GENMASK(3, 2) +/* AIC31XX_HSDETECT */ +#define AIC31XX_HSD_ENABLE BIT(7) +#define AIC31XX_HSD_TYPE_MASK GENMASK(6, 5) +#define AIC31XX_HSD_TYPE_SHIFT 5 +#define AIC31XX_HSD_NONE 0x00 +#define AIC31XX_HSD_HP 0x01 +#define AIC31XX_HSD_HS 0x03 + /* AIC31XX_MICBIAS */ #define AIC31XX_MICBIAS_MASK GENMASK(1, 0) #define AIC31XX_MICBIAS_SHIFT 0 -- 2.21.0

2 2

[alsa-devel] Applied "ASoC: tlv320aic31xx: Add button press detection" to the asoc tree
by Mark Brown 02 Apr '19

02 Apr '19

The patch ASoC: tlv320aic31xx: Add button press detection 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 90f0202bf2e15e8933a550b7dfd439b500a76e1c Mon Sep 17 00:00:00 2001 From: "Andrew F. Davis" <afd(a)ti.com> Date: Mon, 1 Apr 2019 10:13:29 -0500 Subject: [PATCH] ASoC: tlv320aic31xx: Add button press detection This device can optionally detect headset or microphone button presses. Add support for this by passing this event to the jack layer. Signed-off-by: Andrew F. Davis <afd(a)ti.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/tlv320aic31xx.c | 17 +++++++++++++++-- sound/soc/codecs/tlv320aic31xx.h | 3 ++- 2 files changed, 17 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tlv320aic31xx.c b/sound/soc/codecs/tlv320aic31xx.c index a60cd8c08a14..fa47bfcb93e9 100644 --- a/sound/soc/codecs/tlv320aic31xx.c +++ b/sound/soc/codecs/tlv320aic31xx.c @@ -1423,10 +1423,21 @@ static irqreturn_t aic31xx_irq(int irq, void *data) dev_err(dev, "Short circuit on Left output is detected\n"); if (value & AIC31XX_HPRSCDETECT) dev_err(dev, "Short circuit on Right output is detected\n"); - if (value & AIC31XX_HSPLUG) { + if (value & (AIC31XX_HSPLUG | AIC31XX_BUTTONPRESS)) { unsigned int val; int status = 0; + ret = regmap_read(aic31xx->regmap, AIC31XX_INTRDACFLAG2, + &val); + if (ret) { + dev_err(dev, "Failed to read interrupt mask: %d\n", + ret); + goto exit; + } + + if (val & AIC31XX_BUTTONPRESS) + status |= SND_JACK_BTN_0; + ret = regmap_read(aic31xx->regmap, AIC31XX_HSDETECT, &val); if (ret) { dev_err(dev, "Failed to read headset type: %d\n", ret); @@ -1451,7 +1462,8 @@ static irqreturn_t aic31xx_irq(int irq, void *data) } if (value & ~(AIC31XX_HPLSCDETECT | AIC31XX_HPRSCDETECT | - AIC31XX_HSPLUG)) + AIC31XX_HSPLUG | + AIC31XX_BUTTONPRESS)) dev_err(dev, "Unknown DAC interrupt flags: 0x%08x\n", value); read_overflow: @@ -1564,6 +1576,7 @@ static int aic31xx_i2c_probe(struct i2c_client *i2c, regmap_write(aic31xx->regmap, AIC31XX_INT1CTRL, AIC31XX_HSPLUGDET | + AIC31XX_BUTTONPRESSDET | AIC31XX_SC | AIC31XX_ENGINE); diff --git a/sound/soc/codecs/tlv320aic31xx.h b/sound/soc/codecs/tlv320aic31xx.h index 9985ceed97ae..cb024955c978 100644 --- a/sound/soc/codecs/tlv320aic31xx.h +++ b/sound/soc/codecs/tlv320aic31xx.h @@ -21,7 +21,8 @@ #define DAC31XX_BIT BIT(3) #define AIC31XX_JACK_MASK (SND_JACK_HEADPHONE | \ - SND_JACK_HEADSET) + SND_JACK_HEADSET | \ + SND_JACK_BTN_0) enum aic31xx_type { AIC3100 = 0, -- 2.20.1

1 0

[alsa-devel] [PATCH V2] ASoC: hdac_hdmi: add device PM dependency
by libin.yang＠intel.com 02 Apr '19

02 Apr '19

From: Libin Yang <libin.yang(a)intel.com> HDMI audio codec register operation depends on the display audio power domain. The hdmi audio codec dapm event callback must be called after display audio power is turned on. Add hdac_hdmi_device_link_add() in HDMI audio codec driver. The customer audio driver, such as cAVS/SST and SOF audio driver, can call the function to set the audio power dependency. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 12 ++++++++++++ sound/soc/codecs/hdac_hdmi.h | 6 ++++++ 2 files changed, 18 insertions(+) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index 5eeb0fe..c991407 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -1792,6 +1792,18 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int device, } EXPORT_SYMBOL_GPL(hdac_hdmi_jack_init); +struct device_link * +hdac_hdmi_device_link_add(struct device *consumer, + struct snd_soc_component *component, + u32 flags) +{ + struct hdac_hdmi_priv *hdmi = snd_soc_component_get_drvdata(component); + struct hdac_device *hdev = hdmi->hdev; + + return device_link_add(consumer, &hdev->dev, flags); +} +EXPORT_SYMBOL_GPL(hdac_hdmi_device_link_add); + static void hdac_hdmi_present_sense_all_pins(struct hdac_device *hdev, struct hdac_hdmi_priv *hdmi, bool detect_pin_caps) { diff --git a/sound/soc/codecs/hdac_hdmi.h b/sound/soc/codecs/hdac_hdmi.h index 4fa2fc9..9239c6b 100644 --- a/sound/soc/codecs/hdac_hdmi.h +++ b/sound/soc/codecs/hdac_hdmi.h @@ -7,4 +7,10 @@ int hdac_hdmi_jack_init(struct snd_soc_dai *dai, int pcm, int hdac_hdmi_jack_port_init(struct snd_soc_component *component, struct snd_soc_dapm_context *dapm); + +struct device_link * +hdac_hdmi_device_link_add(struct device *consumer, + struct snd_soc_component *component, + u32 flags); + #endif /* __HDAC_HDMI_H__ */ -- 2.7.4

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[alsa-devel] regression: bad sound card type detection in audio graph
by Arnaud Pouliquen 02 Apr '19

02 Apr '19

Hello Kuninori, We are seeing a regression while rebasing on kernel V5.1 (from v4.19). The regression is linked to your update of the audio graph to merge the support of the normal and DPCM sound cards. In our sound card (described here: https://github.com/STMicroelectronics/linux/blob/v4.19-stm32mp/arch/arm/boo…) We declare a normal sound card using the WM8994 codec: Playback: CPU-DAI (SAI2a)-> CODEC-DAI/port0 Record: CPU-DAI (SAI2B)<- CODEC-DAI/port1 ... The WM8994 codec is declared with 2 ports as it supports 2 interfaces. When we probe the sound card we are detecting a wrong dai link type because the condition if (of_get_child_count(codec_port) > 1) return true. The dai link is detected as DPCM instead of the expected normal link. I checked your last commits sent on mailing list, seems that this criteria is still valid. Please could you elaborate a little bit on this criteria for the normal/DPCM selection? What is you feeling about this limitation. Do you see a way to refine the condition to allow normal sound card with multi-port codecs. Or do we have to migrate our sound card to DPCM. Thanks in advance for your answer Regards Arnaud

2 7

[alsa-devel] [PATCH] Documentation: soundwire: Ensure that code is inside the code blocks
by Jonathan Neuschäfer 02 Apr '19

02 Apr '19

The way the document is written now, Sphinx renders empty code blocks followed by the lines that should be inside them. Unindent the code-block directives to fix this. Fixes: 502c00d9c315 ("Documentation: soundwire: fix stream.rst markup warnings") Signed-off-by: Jonathan Neuschäfer <j.neuschaefer(a)gmx.net> --- Documentation/driver-api/soundwire/stream.rst | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/Documentation/driver-api/soundwire/stream.rst b/Documentation/driver-api/soundwire/stream.rst index 26a6064503fd..5351bd2f34a8 100644 --- a/Documentation/driver-api/soundwire/stream.rst +++ b/Documentation/driver-api/soundwire/stream.rst @@ -201,7 +201,7 @@ Bus implements below API for allocate a stream which needs to be called once per stream. From ASoC DPCM framework, this stream state maybe linked to .startup() operation. - .. code-block:: c +.. code-block:: c int sdw_alloc_stream(char * stream_name); @@ -228,7 +228,7 @@ the respective Master(s) and Slave(s) associated with stream. These APIs can only be invoked once by respective Master(s) and Slave(s). From ASoC DPCM framework, this stream state is linked to .hw_params() operation. - .. code-block:: c +.. code-block:: c int sdw_stream_add_master(struct sdw_bus * bus, struct sdw_stream_config * stream_config, @@ -274,7 +274,7 @@ Bus implements below API for PREPARE state which needs to be called once per stream. From ASoC DPCM framework, this stream state is linked to .prepare() operation. - .. code-block:: c +.. code-block:: c int sdw_prepare_stream(struct sdw_stream_runtime * stream); @@ -304,7 +304,7 @@ Bus implements below API for ENABLE state which needs to be called once per stream. From ASoC DPCM framework, this stream state is linked to .trigger() start operation. - .. code-block:: c +.. code-block:: c int sdw_enable_stream(struct sdw_stream_runtime * stream); @@ -332,7 +332,7 @@ Bus implements below API for DISABLED state which needs to be called once per stream. From ASoC DPCM framework, this stream state is linked to .trigger() stop operation. - .. code-block:: c +.. code-block:: c int sdw_disable_stream(struct sdw_stream_runtime * stream); @@ -357,7 +357,7 @@ Bus implements below API for DEPREPARED state which needs to be called once per stream. From ASoC DPCM framework, this stream state is linked to .trigger() stop operation. - .. code-block:: c +.. code-block:: c int sdw_deprepare_stream(struct sdw_stream_runtime * stream); @@ -382,7 +382,7 @@ Bus implements below APIs for RELEASE state which needs to be called by all the Master(s) and Slave(s) associated with stream. From ASoC DPCM framework, this stream state is linked to .hw_free() operation. - .. code-block:: c +.. code-block:: c int sdw_stream_remove_master(struct sdw_bus * bus, struct sdw_stream_runtime * stream); @@ -395,7 +395,7 @@ stream assigned as part of ALLOCATED state. In .shutdown() the data structure maintaining stream state are freed up. - .. code-block:: c +.. code-block:: c void sdw_release_stream(struct sdw_stream_runtime * stream); -- 2.20.1

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[alsa-devel] [asoc:for-5.2 92/97] sound/soc//mediatek/mt8183/mt8183-da7219-max98357.c:413:13: error: 'struct snd_soc_dai_link' has no member named 'platform'; did you mean 'platforms'?
by kbuild test robot 01 Apr '19

01 Apr '19

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-5.2 head: 35146467bd43704c52d1d5760af9839a10ad5b12 commit: ebbddc75bbe8d8eab64d3285b0480f5ae9232b2e [92/97] ASoC: Mediatek: MT8183: Add machine driver with DA7219 config: arm-allmodconfig (attached as .config) compiler: arm-linux-gnueabi-gcc (Debian 7.2.0-11) 7.2.0 reproduce: wget https://raw.githubusercontent.com/intel/lkp-tests/master/sbin/make.cross -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout ebbddc75bbe8d8eab64d3285b0480f5ae9232b2e # save the attached .config to linux build tree GCC_VERSION=7.2.0 make.cross ARCH=arm All errors (new ones prefixed by >>): sound/soc//mediatek/mt8183/mt8183-da7219-max98357.c: In function 'mt8183_da7219_max98357_dev_probe': >> sound/soc//mediatek/mt8183/mt8183-da7219-max98357.c:413:13: error: 'struct snd_soc_dai_link' has no member named 'platform'; did you mean 'platforms'? dai_link->platform = NULL; ^~~~~~~~ platforms vim +413 sound/soc//mediatek/mt8183/mt8183-da7219-max98357.c 387 388 static int mt8183_da7219_max98357_dev_probe(struct platform_device *pdev) 389 { 390 struct snd_soc_card *card = &mt8183_da7219_max98357_card; 391 struct device_node *platform_node; 392 struct snd_soc_dai_link *dai_link; 393 struct pinctrl *default_pins; 394 int ret, i; 395 396 card->dev = &pdev->dev; 397 398 platform_node = of_parse_phandle(pdev->dev.of_node, 399 "mediatek,platform", 0); 400 if (!platform_node) { 401 dev_err(&pdev->dev, "Property 'platform' missing or invalid\n"); 402 return -EINVAL; 403 } 404 405 for_each_card_prelinks(card, i, dai_link) { 406 /* In the alsa soc-core, the "platform" will be 407 * allocated by devm_kzalloc if null. 408 * There is a special case that registerring 409 * sound card is failed at the first time, but 410 * the "platform" will not null when probe is trying 411 * again. It's not expected normally. 412 */ > 413 dai_link->platform = NULL; 414 415 if (dai_link->platform_name) 416 continue; 417 dai_link->platform_of_node = platform_node; 418 } 419 420 mt8183_da7219_max98357_headset_dev.codec_of_node = 421 of_parse_phandle(pdev->dev.of_node, 422 "mediatek,headset-codec", 0); 423 if (!mt8183_da7219_max98357_headset_dev.codec_of_node) { 424 dev_err(&pdev->dev, 425 "Property 'mediatek,headset-codec' missing/invalid\n"); 426 return -EINVAL; 427 } 428 429 ret = devm_snd_soc_register_card(&pdev->dev, card); 430 if (ret) { 431 dev_err(&pdev->dev, "%s snd_soc_register_card fail %d\n", 432 __func__, ret); 433 return ret; 434 } 435 436 default_pins = 437 devm_pinctrl_get_select(&pdev->dev, PINCTRL_STATE_DEFAULT); 438 if (IS_ERR(default_pins)) { 439 dev_err(&pdev->dev, "%s set pins failed\n", 440 __func__); 441 return PTR_ERR(default_pins); 442 } 443 444 return ret; 445 } 446 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH v14 0/6] Media Device Allocator API
by Shuah Khan 01 Apr '19

01 Apr '19

Media Device Allocator API to allows multiple drivers share a media device. This API solves a very common use-case for media devices where one physical device (an USB stick) provides both audio and video. When such media device exposes a standard USB Audio class, a proprietary Video class, two or more independent drivers will share a single physical USB bridge. In such cases, it is necessary to coordinate access to the shared resource. Using this API, drivers can allocate a media device with the shared struct device as the key. Once the media device is allocated by a driver, other drivers can get a reference to it. The media device is released when all the references are released. The primary focus for testing The patch series is making sure media device is released when both drivers release the media device with a series of unbind/binds on both drivers. - both au0828 and snd-usb-aduio as built-in - both au0828 and snd-usb-aduio as modules - au0828 as module and snd-usb-aduio as built-in - au0828 as built-in and snd-usb-aduio as module Test results can be found at: https://docs.google.com/document/d/1RMF8Rwj7xHJEoOx6_K2f-REgZJ63BMeAVEf-CV-… Changes since v13: - Minor changes to variable names and other minor changes to copyright and typos suggested by Hans Verkuil. Changes since v12: - Patch 1: Fixed prototype warns from media_dev_allocator.c. Removed dependency on find_module() by adding struct module to input args. Still need module name to pass into media_device API. - Patch 2 & 4: Update media dev allocator api calls to pass in struct module pointer. - No changes to Patches 3 & 5. - Added patch 6 with a test. It can go in separately. Changes since v11: - Patch 1: Add CONFIG_MODULES dependency in media_dev_allocator files. to fix compile errors when CONFIG_MODULES is disabled. - Patch 2, 3: No changes. - Patch 4: Fix sparse error reported by Dan Carpenter. - Patch 5: Fix warns found by Hans Verkuil. - v11 was tested on 5.0-rc7 and addresses comments on v10 series from Hans Verkuil. Fixed problems found in resource sharing logic in au0828 adding a patch 5 to this series. The test plan used for testing resource sharing could serve as a regression test plan and the test results can be found at: - v10 was tested on 5.0-rc3 and addresses comments on v9 series from Hans Verkuil. Changes since v10: - Patch 1: Fixed SPDX tag and removed redundant IS_ENABLED(CONFIG_USB) around media_device_usb_allocate() - Sakari Ailus's review comment. - Patch 2 and 3: No changes - Patch 4: Fixed SPDX tag - Sakari Ailus's review comment. - Carried Reviewed-by tag from Takashi Iwai for the sound from v9. - Patch 5: This is a new patch added to fix resource sharing inconsistencies and problem found during testing using Han's tests. Changes since v9: - Patch 1: Fix mutex assert warning from find_module() calls. This code was written before the change to find_module() that requires callers to hold module_mutex. I missed this during my testing on 4.20-rc6. Hans Verkuil reported the problem. - Patch 4: sound/usb: Initializes all the entities it can before registering the device based on comments from Hans Verkuil - Carried Reviewed-by tag from Takashi Iwai for the sound from v9. - No changes to Patches 2 and 3. References: https://lkml.org/lkml/2018/11/2/169 https://www.mail-archive.com/linux-media@vger.kernel.org/msg105854.html Shuah Khan (6): media: Media Device Allocator API media: change au0828 to use Media Device Allocator API media: media.h: Enable ALSA MEDIA_INTF_T* interface types sound/usb: Use Media Controller API to share media resources au0828: fix enable and disable source audio and video inconsistencies selftests: media_dev_allocator api test Documentation/media/kapi/mc-core.rst | 41 +++ drivers/media/Makefile | 6 + drivers/media/media-dev-allocator.c | 135 ++++++++ drivers/media/usb/au0828/Kconfig | 2 + drivers/media/usb/au0828/au0828-core.c | 195 ++++++++--- drivers/media/usb/au0828/au0828.h | 6 +- include/media/media-dev-allocator.h | 63 ++++ include/uapi/linux/media.h | 25 +- sound/usb/Kconfig | 4 + sound/usb/Makefile | 2 + sound/usb/card.c | 14 + sound/usb/card.h | 3 + sound/usb/media.c | 327 ++++++++++++++++++ sound/usb/media.h | 74 ++++ sound/usb/mixer.h | 3 + sound/usb/pcm.c | 29 +- sound/usb/quirks-table.h | 1 + sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 6 + .../media_tests/media_dev_allocator.sh | 85 +++++ 20 files changed, 962 insertions(+), 61 deletions(-) create mode 100644 drivers/media/media-dev-allocator.c create mode 100644 include/media/media-dev-allocator.h create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h create mode 100755 tools/testing/selftests/media_tests/media_dev_allocator.sh -- 2.17.1

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[alsa-devel] [PATCH v4] ASoC: Intel: boards: kbl_da7219_max98927: add dai_trigger function
by Jenny TC 01 Apr '19

01 Apr '19

From: Mac Chiang <mac.chiang(a)intel.com> amplifier feedback is not modeled as being dependent on any active output. Even when there is no playback happening, parts of the graph, specifically the IV sense->speaker protection->output remains active and this prevents the DSP from entering low-power states. This patch suggest a machine driver level approach where the speaker pins are enabled/disabled dynamically depending on stream start/stop events. DPAM graph representations show the feedback loop is indeed disabled and low-power states can be reached. Signed-off-by: Jenny TC <jenny.tc(a)intel.com> Signed-off-by: Harshapriya.n <harshapriya.n(a)intel.com> Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> --- Changelog: v4: - Changed device name in in kabylake_ssp0_trigger to address a conflict with recently merged patch MAXIM_DEV0_NAME -> MAX98927_DEV0_NAME ... v3: - fixed the return logic at the end of kabylake_card_late_probe() v2: - described clearly in commit message - added snd_soc_dapm_disable_pin in kabylake_card_late_probe() --- --- sound/soc/intel/boards/kbl_da7219_max98927.c | 73 +++++++++++++++++++++++++++- 1 file changed, 71 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 2768a57..a04a6bb 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -219,8 +219,60 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, return 0; } +static int kabylake_ssp0_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + int j, ret; + + for (j = 0; j < rtd->num_codecs; j++) { + struct snd_soc_dai *codec_dai = rtd->codec_dais[j]; + const char *name = codec_dai->component->name; + struct snd_soc_component *component = codec_dai->component; + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + char pin_name[20]; + + if (strcmp(name, MAX98927_DEV0_NAME) && + strcmp(name, MAX98927_DEV1_NAME) && + strcmp(name, MAX98373_DEV0_NAME) && + strcmp(name, MAX98373_DEV1_NAME)) + continue; + + snprintf(pin_name, ARRAY_SIZE(pin_name), "%s Spk", + codec_dai->component->name_prefix); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + ret = snd_soc_dapm_enable_pin(dapm, pin_name); + if (ret) { + dev_err(rtd->dev, "failed to enable %s: %d\n", + pin_name, ret); + return ret; + } + snd_soc_dapm_sync(dapm); + break; + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + ret = snd_soc_dapm_disable_pin(dapm, pin_name); + if (ret) { + dev_err(rtd->dev, "failed to disable %s: %d\n", + pin_name, ret); + return ret; + } + snd_soc_dapm_sync(dapm); + break; + } + } + + return 0; +} + static struct snd_soc_ops kabylake_ssp0_ops = { .hw_params = kabylake_ssp0_hw_params, + .trigger = kabylake_ssp0_trigger, }; static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, @@ -956,6 +1008,7 @@ static int kabylake_card_late_probe(struct snd_soc_card *card) { struct kbl_codec_private *ctx = snd_soc_card_get_drvdata(card); struct kbl_hdmi_pcm *pcm; + struct snd_soc_dapm_context *dapm = &card->dapm; struct snd_soc_component *component = NULL; int err, i = 0; char jack_name[NAME_SIZE]; @@ -982,9 +1035,25 @@ static int kabylake_card_late_probe(struct snd_soc_card *card) if (!component) return -EINVAL; - return hdac_hdmi_jack_port_init(component, &card->dapm); - return 0; + err = hdac_hdmi_jack_port_init(component, &card->dapm); + + if (err < 0) + return err; + + err = snd_soc_dapm_disable_pin(dapm, "Left Spk"); + if (err) { + dev_err(card->dev, "failed to disable Left Spk: %d\n", err); + return err; + } + + err = snd_soc_dapm_disable_pin(dapm, "Right Spk"); + if (err) { + dev_err(card->dev, "failed to disable Right Spk: %d\n", err); + return err; + } + + return snd_soc_dapm_sync(dapm); } /* kabylake audio machine driver for SPT + DA7219 */ -- 1.9.1

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