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

[alsa-devel] [PATCH v2 resend 0/3] ASoC: SOF: fix suspend ordering with runtime idle
by Kai Vehmanen 06 Jul '19

06 Jul '19

Hi all, this series addresses issues with ordering of HDA codec and controller suspends in the runtime PM flows. The implemented logic for SOF is similar to what has been used by the Intel AZX HDA driver. To implement this, first a fix is needed to hdac_hdmic ASoC codec driver. SOF framework also needs to be extended to allow SOF devices to implement a runtime_idle callback. Third, concrete implementation is in a separate patch for APL/CNL Intel hardware, for which strict ordering of codec-controller power down sequence needs to be maintained. As this extends the SOF device interface, Pierre asked me to send to the list for wider review. This series has been prereviewd at SOF github as: https://github.com/thesofproject/linux/pull/1003 v2 resend: - no changes since v2 -- resending as 2 weeks has passed and multiple reviewed-by acks already received v2: - Rebased on broonie/for-next, there was a conflict with patch "ASoC: SOF: Intel: hda: release link DMA for paused streams during suspend" - Added Reviewed-by tags from v1 round Kai Vehmanen (3): ASoC: hdac_hdmi: report codec link up/down status to bus ASoC: SOF: add runtime idle callback ASoC: SOF: Intel: implement runtime idle for CNL/APL sound/soc/codecs/hdac_hdmi.c | 2 ++ sound/soc/sof/intel/apl.c | 1 + sound/soc/sof/intel/cnl.c | 1 + sound/soc/sof/intel/hda-dsp.c | 13 +++++++++++++ sound/soc/sof/intel/hda.h | 1 + sound/soc/sof/ops.h | 8 ++++++++ sound/soc/sof/pm.c | 8 ++++++++ sound/soc/sof/sof-acpi-dev.c | 2 +- sound/soc/sof/sof-pci-dev.c | 2 +- sound/soc/sof/sof-priv.h | 2 ++ 10 files changed, 38 insertions(+), 2 deletions(-) -- 2.17.1

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[alsa-devel] [PATCH -next] ASoC: SOF: debug: fix possible memory leak in sof_dfsentry_write()
by Wei Yongjun 05 Jul '19

05 Jul '19

'string' is malloced in sof_dfsentry_write() and should be freed before leaving from the error handling cases, otherwise it will cause memory leak. Fixes: 091c12e1f50c ("ASoC: SOF: debug: add new debugfs entries for IPC flood test") Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/sof/debug.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/sof/debug.c b/sound/soc/sof/debug.c index 54bb53bfc81b..2388477a965e 100644 --- a/sound/soc/sof/debug.c +++ b/sound/soc/sof/debug.c @@ -162,7 +162,7 @@ static ssize_t sof_dfsentry_write(struct file *file, const char __user *buffer, else ret = kstrtoul(string, 0, &ipc_count); if (ret < 0) - return ret; + goto out; /* limit max duration/ipc count for flood test */ if (flood_duration_test) { @@ -191,7 +191,7 @@ static ssize_t sof_dfsentry_write(struct file *file, const char __user *buffer, "error: debugfs write failed to resume %d\n", ret); pm_runtime_put_noidle(sdev->dev); - return ret; + goto out; } /* flood test */

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[alsa-devel] [PATCH] ASoC: rt5665: remove redundant assignment to variable idx
by Colin King 05 Jul '19

05 Jul '19

From: Colin Ian King <colin.king(a)canonical.com> The variable idx is being initialized with a value that is never read and it is being updated later with a new value. The initialization is redundant and can be removed. Addresses-Coverity: ("Unused value") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/rt5665.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5665.c b/sound/soc/codecs/rt5665.c index 87263317085a..c050d84a6916 100644 --- a/sound/soc/codecs/rt5665.c +++ b/sound/soc/codecs/rt5665.c @@ -1478,7 +1478,7 @@ static int set_dmic_clk(struct snd_soc_dapm_widget *w, { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct rt5665_priv *rt5665 = snd_soc_component_get_drvdata(component); - int pd, idx = -EINVAL; + int pd, idx; pd = rl6231_get_pre_div(rt5665->regmap, RT5665_ADDA_CLK_1, RT5665_I2S_PD1_SFT); -- 2.20.1

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[alsa-devel] [PATCH v2] ASoC: sunxi: sun50i-codec-analog: Add earpiece
by Luca Weiss 05 Jul '19

05 Jul '19

This adds the necessary registers and audio routes to play audio using the Earpiece, that's supported on the A64. Signed-off-by: Luca Weiss <luca(a)z3ntu.xyz> --- Changes v1 -> v2: * Make the earpiece enable register a DAPM widget * Adjust the audio routes to include the new Earpiece Amp widget * Left/Right Analog Mixer => Left/Right Mixer sound/soc/sunxi/sun50i-codec-analog.c | 50 +++++++++++++++++++++++++++ 1 file changed, 50 insertions(+) diff --git a/sound/soc/sunxi/sun50i-codec-analog.c b/sound/soc/sunxi/sun50i-codec-analog.c index d105c90c3706..6d1de565350e 100644 --- a/sound/soc/sunxi/sun50i-codec-analog.c +++ b/sound/soc/sunxi/sun50i-codec-analog.c @@ -49,6 +49,15 @@ #define SUN50I_ADDA_OR_MIX_CTRL_DACR 1 #define SUN50I_ADDA_OR_MIX_CTRL_DACL 0 +#define SUN50I_ADDA_EARPIECE_CTRL0 0x03 +#define SUN50I_ADDA_EARPIECE_CTRL0_EAR_RAMP_TIME 4 +#define SUN50I_ADDA_EARPIECE_CTRL0_ESPSR 0 + +#define SUN50I_ADDA_EARPIECE_CTRL1 0x04 +#define SUN50I_ADDA_EARPIECE_CTRL1_ESPPA_EN 7 +#define SUN50I_ADDA_EARPIECE_CTRL1_ESPPA_MUTE 6 +#define SUN50I_ADDA_EARPIECE_CTRL1_ESP_VOL 0 + #define SUN50I_ADDA_LINEOUT_CTRL0 0x05 #define SUN50I_ADDA_LINEOUT_CTRL0_LEN 7 #define SUN50I_ADDA_LINEOUT_CTRL0_REN 6 @@ -172,6 +181,10 @@ static const DECLARE_TLV_DB_RANGE(sun50i_codec_lineout_vol_scale, 2, 31, TLV_DB_SCALE_ITEM(-4350, 150, 0), ); +static const DECLARE_TLV_DB_RANGE(sun50i_codec_earpiece_vol_scale, + 0, 1, TLV_DB_SCALE_ITEM(TLV_DB_GAIN_MUTE, 0, 1), + 2, 31, TLV_DB_SCALE_ITEM(-4350, 150, 0), +); /* volume / mute controls */ static const struct snd_kcontrol_new sun50i_a64_codec_controls[] = { @@ -225,6 +238,15 @@ static const struct snd_kcontrol_new sun50i_a64_codec_controls[] = { SUN50I_ADDA_LINEOUT_CTRL0_LEN, SUN50I_ADDA_LINEOUT_CTRL0_REN, 1, 0), + SOC_SINGLE_TLV("Earpiece Playback Volume", + SUN50I_ADDA_EARPIECE_CTRL1, + SUN50I_ADDA_EARPIECE_CTRL1_ESP_VOL, 0x1f, 0, + sun50i_codec_earpiece_vol_scale), + + SOC_SINGLE("Earpiece Playback Switch", + SUN50I_ADDA_EARPIECE_CTRL1, + SUN50I_ADDA_EARPIECE_CTRL1_ESPPA_MUTE, 1, 0), + }; static const char * const sun50i_codec_hp_src_enum_text[] = { @@ -257,6 +279,20 @@ static const struct snd_kcontrol_new sun50i_codec_lineout_src[] = { sun50i_codec_lineout_src_enum), }; +static const char * const sun50i_codec_earpiece_src_enum_text[] = { + "DACR", "DACL", "Right Mixer", "Left Mixer", +}; + +static SOC_ENUM_SINGLE_DECL(sun50i_codec_earpiece_src_enum, + SUN50I_ADDA_EARPIECE_CTRL0, + SUN50I_ADDA_EARPIECE_CTRL0_ESPSR, + sun50i_codec_earpiece_src_enum_text); + +static const struct snd_kcontrol_new sun50i_codec_earpiece_src[] = { + SOC_DAPM_ENUM("Earpiece Source Playback Route", + sun50i_codec_earpiece_src_enum), +}; + static const struct snd_soc_dapm_widget sun50i_a64_codec_widgets[] = { /* DAC */ SND_SOC_DAPM_DAC("Left DAC", NULL, SUN50I_ADDA_MIX_DAC_CTRL, @@ -285,6 +321,12 @@ static const struct snd_soc_dapm_widget sun50i_a64_codec_widgets[] = { SND_SOC_NOPM, 0, 0, sun50i_codec_lineout_src), SND_SOC_DAPM_OUTPUT("LINEOUT"), + SND_SOC_DAPM_MUX("Earpiece Source Playback Route", + SND_SOC_NOPM, 0, 0, sun50i_codec_earpiece_src), + SND_SOC_DAPM_OUT_DRV("Earpiece Amp", SUN50I_ADDA_EARPIECE_CTRL1, + SUN50I_ADDA_EARPIECE_CTRL1_ESPPA_EN, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("EARPIECE"), + /* Microphone inputs */ SND_SOC_DAPM_INPUT("MIC1"), @@ -388,6 +430,14 @@ static const struct snd_soc_dapm_route sun50i_a64_codec_routes[] = { { "Line Out Source Playback Route", "Mono Differential", "Right Mixer" }, { "LINEOUT", NULL, "Line Out Source Playback Route" }, + + /* Earpiece Routes */ + { "Earpiece Source Playback Route", "DACL", "Left DAC" }, + { "Earpiece Source Playback Route", "DACR", "Right DAC" }, + { "Earpiece Source Playback Route", "Left Mixer", "Left Mixer" }, + { "Earpiece Source Playback Route", "Right Mixer", "Right Mixer" }, + { "Earpiece Amp", NULL, "Earpiece Source Playback Route" }, + { "EARPIECE", NULL, "Earpiece Amp" }, }; static const struct snd_soc_component_driver sun50i_codec_analog_cmpnt_drv = { -- 2.22.0

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[alsa-devel] [PATCH 2/3 v2] pcm_file: improve error checking in write_wav_header function
by Adam Miartus 05 Jul '19

05 Jul '19

previously errno would be returned even for cases where it may have not been populated, for example one of the write functions failing, or writing only partial buffer, now progress through write operations separately and report errno when appropriate Signed-off-by: Adam Miartus <amiartus(a)de.adit-jv.com> Reviewed-by: Timo Wischer <twischer(a)de.adit-jv.com> --- src/pcm/pcm_file.c | 38 ++++++++++++++++++++++++++++++-------- 1 file changed, 30 insertions(+), 8 deletions(-) diff --git a/src/pcm/pcm_file.c b/src/pcm/pcm_file.c index 03c1640..da7f087 100644 --- a/src/pcm/pcm_file.c +++ b/src/pcm/pcm_file.c @@ -327,6 +327,8 @@ static void setup_wav_header(snd_pcm_t *pcm, struct wav_fmt *fmt) static int write_wav_header(snd_pcm_t *pcm) { snd_pcm_file_t *file = pcm->private_data; + ssize_t res; + static const char header[] = { 'R', 'I', 'F', 'F', 0x24, 0, 0, 0, @@ -341,15 +343,35 @@ static int write_wav_header(snd_pcm_t *pcm) setup_wav_header(pcm, &file->wav_header); - if (write(file->fd, header, sizeof(header)) != sizeof(header) || - write(file->fd, &file->wav_header, sizeof(file->wav_header)) != - sizeof(file->wav_header) || - write(file->fd, header2, sizeof(header2)) != sizeof(header2)) { - int err = errno; - SYSERR("%s write header failed, file data may be corrupt", file->fname); - return -err; - } + res = write(file->fd, header, sizeof(header)); + if (res != sizeof(header)) + goto write_error; + + res = write(file->fd, &file->wav_header, sizeof(file->wav_header)); + if (res != sizeof(file->wav_header)) + goto write_error; + + res = write(file->fd, header2, sizeof(header2)); + if (res != sizeof(header2)) + goto write_error; + return 0; + +write_error: + /* + * print real errno if available and return EIO, reason for this is + * to block possible EPIPE in case file->fd is a pipe. EPIPE from + * file->fd conflicts with EPIPE from playback stream which should + * be used to signal XRUN on playback device + */ + if (res < 0) + SYSERR("%s write header failed, file data may be corrupt", file->fname); + else + SNDERR("%s write header incomplete, file data may be corrupt", file->fname); + + memset(&file->wav_header, 0, sizeof(struct wav_fmt)); + + return -EIO; } /* fix up the length fields in WAV header */ -- 2.7.4

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[alsa-devel] [PATCH] ALSA: cs4281: remove redundant assignment to variable val and remove a goto
by Colin King 05 Jul '19

05 Jul '19

From: Colin Ian King <colin.king(a)canonical.com> The variable val is being assigned with a value that is never read and it is being updated later with a new value. The assignment is redundant and can be removed. Also remove a goto statement and a label and replace with a break statement. Addresses-Coverity: ("Unused value") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/pci/cs4281.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/pci/cs4281.c b/sound/pci/cs4281.c index a2cce3ecda6f..04c712647853 100644 --- a/sound/pci/cs4281.c +++ b/sound/pci/cs4281.c @@ -694,7 +694,7 @@ static int snd_cs4281_trigger(struct snd_pcm_substream *substream, int cmd) static unsigned int snd_cs4281_rate(unsigned int rate, unsigned int *real_rate) { - unsigned int val = ~0; + unsigned int val; if (real_rate) *real_rate = rate; @@ -707,9 +707,8 @@ static unsigned int snd_cs4281_rate(unsigned int rate, unsigned int *real_rate) case 44100: return 1; case 48000: return 0; default: - goto __variable; + break; } - __variable: val = 1536000 / rate; if (real_rate) *real_rate = 1536000 / val; -- 2.20.1

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[alsa-devel] bug in playing 6-channel sound
by Julian Bradfield 05 Jul '19

05 Jul '19

Some five years ago, the following post https://mailman.alsa-project.org/pipermail/alsa-devel/2014-February/072265.… reported a failure to play six-channel sound, that could be worked around by setting the prealloc value for the device to various values *other than* the default 64. I have just run into this playing a movie from my laptop (Thinkpad Yoga X1, Intel HDA PCH sound card outputting through HDMI) running Xubuntu 18.04, up to date as at the time of posting. I see exactly the same problem as in that post, and the same workaround works for me - I tried prealloc values 63,65,32,128, and any of them is ok. I'm a bit baffled that such a problem has survived for several years. If any developer would like me to provide debugging information, please get in touch - I have some time available at present, but not enough to learn ALSA and debug it myself! I will remain subscribed to the list for a few weeks to read any followups.

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[alsa-devel] [GIT PULL]: soundwire updates for v5.3-rc1
by Vinod Koul 05 Jul '19

05 Jul '19

Hello Greg, Please pull to receive the updates for soundwire for v5.3. The following changes since commit a188339ca5a396acc588e5851ed7e19f66b0ebd9: Linux 5.2-rc1 (2019-05-19 15:47:09 -0700) are available in the Git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/vkoul/soundwire.git tags/soundwire-5.3-rc1 for you to fetch changes up to 2aeac95d1a4cc85aae57ab842d5c3340df0f817f: soundwire: add module_sdw_driver helper macro (2019-07-01 11:52:43 +0530) ---------------------------------------------------------------- soundwire updates for v5.3-rc1 Updates for 5.3 include: - module_sdw_driver macro for drivers - Documentation updates for code-blocks - Improvement from Pierre on intel and cadence driver - Clarification of DisCo properties and updates ---------------------------------------------------------------- Jan Kotas (1): soundwire: cdns: Fix compilation error on arm64 Mauro Carvalho Chehab (1): docs: soundwire: locking: fix tags for a code-block Pierre-Louis Bossart (15): soundwire: intel: filter SoundWire controller device search soundwire: mipi_disco: fix master/link error soundwire: add port-related definitions soundwire: remove master data port properties soundwire: mipi-disco: remove master_count property for masters soundwire: rename 'freq' fields soundwire: mipi-disco: fix clock stop modes soundwire: clarify comment soundwire: rename/clarify MIPI DisCo properties soundwire: cadence_master: use rate_limited dynamic debug soundwire: cadence_master: log Slave status mask on errors soundwire: cadence_master: check the number of bidir PDIs soundwire: Intel: add log for number of PCM and PDM PDIs soundwire: fix typo in comments soundwire: intel_init: add checks on link numbers Srinivas Kandagatla (1): soundwire: add module_sdw_driver helper macro Documentation/driver-api/soundwire/locking.rst | 4 +- drivers/soundwire/bus.c | 6 +- drivers/soundwire/cadence_master.c | 30 +++++---- drivers/soundwire/intel.c | 17 +++-- drivers/soundwire/intel.h | 2 +- drivers/soundwire/intel_init.c | 25 +++++++- drivers/soundwire/mipi_disco.c | 35 +++++----- drivers/soundwire/stream.c | 8 +-- include/linux/soundwire/sdw.h | 88 ++++++++++++++++++++------ include/linux/soundwire/sdw_type.h | 11 ++++ 10 files changed, 162 insertions(+), 64 deletions(-) -- ~Vinod

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[alsa-devel] [PATCH] sound: pci: asihpi: Remove unneeded variable change
by Hariprasad Kelam 05 Jul '19

05 Jul '19

this patch fixes below issue reported by coccicheck sound/pci/asihpi/asihpi.c:1558:5-11: Unneeded variable: "change". Return "1" on line 1564 Signed-off-by: Hariprasad Kelam <hariprasad.kelam(a)gmail.com> --- sound/pci/asihpi/asihpi.c | 7 ++----- 1 file changed, 2 insertions(+), 5 deletions(-) diff --git a/sound/pci/asihpi/asihpi.c b/sound/pci/asihpi/asihpi.c index e7234f3..2a21a3d 100644 --- a/sound/pci/asihpi/asihpi.c +++ b/sound/pci/asihpi/asihpi.c @@ -1519,7 +1519,6 @@ static int snd_asihpi_volume_get(struct snd_kcontrol *kcontrol, static int snd_asihpi_volume_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - int change; u32 h_control = kcontrol->private_value; short an_gain_mB[HPI_MAX_CHANNELS]; @@ -1530,9 +1529,8 @@ static int snd_asihpi_volume_put(struct snd_kcontrol *kcontrol, /* change = asihpi->mixer_volume[addr][0] != left || asihpi->mixer_volume[addr][1] != right; */ - change = 1; hpi_handle_error(hpi_volume_set_gain(h_control, an_gain_mB)); - return change; + return 1; } static const DECLARE_TLV_DB_SCALE(db_scale_100, -10000, VOL_STEP_mB, 0); @@ -1555,13 +1553,12 @@ static int snd_asihpi_volume_mute_put(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { u32 h_control = kcontrol->private_value; - int change = 1; /* HPI currently only supports all or none muting of multichannel volume ALSA Switch element has opposite sense to HPI mute: on==unmuted, off=muted */ int mute = ucontrol->value.integer.value[0] ? 0 : HPI_BITMASK_ALL_CHANNELS; hpi_handle_error(hpi_volume_set_mute(h_control, mute)); - return change; + return 1; } static int snd_asihpi_volume_add(struct snd_card_asihpi *asihpi, -- 2.7.4

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[alsa-devel] audio lost from speaker after reboot from windows on the device ALC295
by He, Bo 05 Jul '19

05 Jul '19

Hi, patch_realtek.c maintainer: I see one issue that reboot from windows and boot to ubuntu, the audio lost from speaker, I suspect there are some bugs in patch_realtek.c drivers, the device is ALC295 and the device id is 0x10ec0295. I have done the below experiments: 1. reboot from windows to windows, the audio is persist . 2. reboot from windows to ubuntu, the audio lost from speaker, but can hear if I hotplug one earphone. 3. if the issue reproduce after reboot from windows, reboot the ubuntu can't restore the audio, I suspect it's warm reset. 4. if I write the port 0xcf9 with 0xe to do cold reset, the audio can restore. 5. if I do suspend/resume, the audio can restore, I suspect do cold boot and suspend will trigger the platform reset to reset the ALC295. 6. if I do double function reset (write the verb 0x7ff in alc_init), the audio is still can't restore. snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_CODEC_RESET, 0); /* Function reset */ snd_hda_codec_write(codec, 0x01, 0, AC_VERB_SET_CODEC_RESET, 0); /* double Function reset */ 7. the issue is first found on kernel 4.19.50, I still see the issue with the latest kernel 5.2-rc2, is it possible windows change some default registers, but ALC295 don't initialize the register?

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