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

[PATCH 1/1] ASoC: dpcm: allow start or stop during pause for backend
by 이경택 30 Mar '20

30 Mar '20

soc_compr_trigger_fe() allows start or stop after pause_push. In dpcm_be_dai_trigger(), however, only pause_release is allowed command after pause_push. To fix this problem, SND_SOC_DPCM_STATE_PAUSED should be allowed also for start or stop command. Signed-off-by: Gyeongtaek Lee <gt82.lee(a)samsung.com> --- sound/soc/soc-pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 2c59b3688ca0..8f6f0ad50288 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2236,7 +2236,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_PREPARE) && - (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP)) + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_STOP) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; ret = dpcm_do_trigger(dpcm, be_substream, cmd); @@ -2266,7 +2267,8 @@ int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, be->dpcm[stream].state = SND_SOC_DPCM_STATE_START; break; case SNDRV_PCM_TRIGGER_STOP: - if (be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) + if ((be->dpcm[stream].state != SND_SOC_DPCM_STATE_START) && + (be->dpcm[stream].state != SND_SOC_DPCM_STATE_PAUSED)) continue; if (!snd_soc_dpcm_can_be_free_stop(fe, be, stream)) -- 2.21.0

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[PATCH 0/1] ASoC: dpcm: allow started or stop during pause for backend
by 이경택 30 Mar '20

30 Mar '20

Hi, When compress offload is used with DPCM, error can be occurred on start or stop after pause. soc_compr_trigger_fe() allows start or stop after pause_push. In dpcm_be_dai_trigger(), however, only pause_release is allowed command after pause_push. Any feedback will be greatly appreciated. Gyeongtaek Lee (1): ASoC: dpcm: allow start or stop during pause for backend sound/soc/soc-pcm.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) base-commit: 76ccd234269bd05debdbc12c96eafe62dd9a6180 -- 2.21.0

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[PATCH 0/4] ASoC: remove rtd->cpu/codec_dai{s} v2
by Kuninori Morimoto 30 Mar '20

30 Mar '20

Hi Mark These are rebased "remove rtd->cpu/codec_dai" patches. Kuninori Morimoto (4): ASoC: soc: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: soc-core: set rtd->num_cpu/codec at soc_new_pcm_runtime() ASoC: soc-core: tidyup soc_new_pcm_runtime() rtd setups ASoC: soc-core: remove cpu_dai/codec_dai/cpu_dais/codec_dais include/sound/soc.h | 26 +++++++++-------- sound/soc/soc-compress.c | 36 +++++++++++------------ sound/soc/soc-core.c | 42 ++++++++++----------------- sound/soc/soc-dapm.c | 4 +-- sound/soc/soc-generic-dmaengine-pcm.c | 6 ++-- sound/soc/soc-pcm.c | 30 +++++++++---------- 6 files changed, 68 insertions(+), 76 deletions(-) -- 2.17.1

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[PATCH v2] fix some output pcms being namehint-ed as input pcms
by sylvain.bertrand＠gmail.com 29 Mar '20

29 Mar '20

This is additional cleanup following: commit d434638a683c01bd6decf8f41863044055237a50 namehint: remember the direction from the upper levels signed-off-by: Sylvain BERTRAND <sylvain.bertrand(a)legeek.net> --- based on arecord -L output --- a/share/alsa/pcm/front.conf +++ b/share/alsa/pcm/front.conf @@ -53,6 +53,7 @@ name defaults.namehint.basic } description "Front speakers" - device $DEV + device_output $DEV + omit_noargs true } } --- a/share/alsa/pcm/iec958.conf +++ b/share/alsa/pcm/iec958.conf @@ -78,6 +78,6 @@ name defaults.namehint.basic } description "IEC958 (S/PDIF) Digital Audio Output" - device $DEV + device_output $DEV } }

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[PATCH] ASoC: bcm2835-i2s: substream alignment now independent in hwparams
by Matt Flax 29 Mar '20

29 Mar '20

Substream sample alignment was being set in hwparams for both substreams at the same time. This became a problem when the Audio Injector isolated sound card needed to offset sample alignment for high sample rates. The latency difference between playback and capture occurs because of the digital isolation chip propagation time, particularly when the codec is master and the DAC return is twice delayed. This patch sets sample alignment registers based on the substream direction in hwparams. This gives the machine driver more control over sample alignment in the bcm2835 i2s driver. Signed-off-by: Matt Flax <flatmax(a)flatmax.org> --- sound/soc/bcm/bcm2835-i2s.c | 36 +++++++++++++++++++----------------- 1 file changed, 19 insertions(+), 17 deletions(-) diff --git a/sound/soc/bcm/bcm2835-i2s.c b/sound/soc/bcm/bcm2835-i2s.c index e6a12e271b07..9db542699a13 100644 --- a/sound/soc/bcm/bcm2835-i2s.c +++ b/sound/soc/bcm/bcm2835-i2s.c @@ -493,11 +493,6 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - bcm2835_i2s_calc_channel_pos(&rx_ch1_pos, &rx_ch2_pos, - rx_mask, slot_width, data_delay, odd_slot_offset); - bcm2835_i2s_calc_channel_pos(&tx_ch1_pos, &tx_ch2_pos, - tx_mask, slot_width, data_delay, odd_slot_offset); - /* * Transmitting data immediately after frame start, eg * in left-justified or DSP mode A, only works stable @@ -508,19 +503,26 @@ static int bcm2835_i2s_hw_params(struct snd_pcm_substream *substream, "Unstable slave config detected, L/R may be swapped"); /* - * Set format for both streams. - * We cannot set another frame length - * (and therefore word length) anyway, - * so the format will be the same. + * Set format on a per stream basis. + * The alignment format can be different depending on direction. */ - regmap_write(dev->i2s_regmap, BCM2835_I2S_RXC_A_REG, - format - | BCM2835_I2S_CH1_POS(rx_ch1_pos) - | BCM2835_I2S_CH2_POS(rx_ch2_pos)); - regmap_write(dev->i2s_regmap, BCM2835_I2S_TXC_A_REG, - format - | BCM2835_I2S_CH1_POS(tx_ch1_pos) - | BCM2835_I2S_CH2_POS(tx_ch2_pos)); + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + bcm2835_i2s_calc_channel_pos(&rx_ch1_pos, &rx_ch2_pos, + rx_mask, slot_width, data_delay, odd_slot_offset); + regmap_write(dev->i2s_regmap, BCM2835_I2S_RXC_A_REG, + format + | BCM2835_I2S_CH1_POS(rx_ch1_pos) + | BCM2835_I2S_CH2_POS(rx_ch2_pos)); + } + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + bcm2835_i2s_calc_channel_pos(&tx_ch1_pos, &tx_ch2_pos, + tx_mask, slot_width, data_delay, odd_slot_offset); + regmap_write(dev->i2s_regmap, BCM2835_I2S_TXC_A_REG, + format + | BCM2835_I2S_CH1_POS(tx_ch1_pos) + | BCM2835_I2S_CH2_POS(tx_ch2_pos)); + } /* Setup the I2S mode */ -- 2.20.1

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[PATCH v2] ALSA: hda/realtek - a fake key event is triggered by running shutup
by Hui Wang 29 Mar '20

29 Mar '20

On the Lenovo X1C7 machines, after we plug the headset, the rt_resume() and rt_suspend() of the codec driver will be called periodically, the driver can't stay in the rt_suspend state even users doen't use the sound card. Through debugging, I found when running rt_suspend(), it will call alc225_shutup(), in this function, it will change 3k pull down control by alc_update_coef_idx(codec, 0x4a, 0, 3 << 10), this will trigger a fake key event and that event will resume the codec, when codec suspend agin, it will trigger the fake key event one more time, this process will repeat. If disable the key event before changing the pull down control, it will not trigger fake key event. It also needs to restore the pull down control and re-enable the key event, otherwise the system can't get key event when codec is in rt_suspend state. Also move some functions ahead of alc225_shutup(), this can save the function declaration. Fixes: 76f7dec08fd6 (ALSA: hda/realtek - Add Headset Button supported for ThinkPad X1) Cc: Kailang Yang <kailang(a)realtek.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 170 +++++++++++++++++++++------------- 1 file changed, 107 insertions(+), 63 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 1ad8c2e2d1af..0af33f00617a 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -107,6 +107,7 @@ struct alc_spec { unsigned int done_hp_init:1; unsigned int no_shutup_pins:1; unsigned int ultra_low_power:1; + unsigned int has_hs_key:1; /* for PLL fix */ hda_nid_t pll_nid; @@ -2982,6 +2983,107 @@ static int alc269_parse_auto_config(struct hda_codec *codec) return alc_parse_auto_config(codec, alc269_ignore, ssids); } +static const struct hda_jack_keymap alc_headset_btn_keymap[] = { + { SND_JACK_BTN_0, KEY_PLAYPAUSE }, + { SND_JACK_BTN_1, KEY_VOICECOMMAND }, + { SND_JACK_BTN_2, KEY_VOLUMEUP }, + { SND_JACK_BTN_3, KEY_VOLUMEDOWN }, + {} +}; + +static void alc_headset_btn_callback(struct hda_codec *codec, + struct hda_jack_callback *jack) +{ + int report = 0; + + if (jack->unsol_res & (7 << 13)) + report |= SND_JACK_BTN_0; + + if (jack->unsol_res & (1 << 16 | 3 << 8)) + report |= SND_JACK_BTN_1; + + /* Volume up key */ + if (jack->unsol_res & (7 << 23)) + report |= SND_JACK_BTN_2; + + /* Volume down key */ + if (jack->unsol_res & (7 << 10)) + report |= SND_JACK_BTN_3; + + jack->jack->button_state = report; +} + +static void alc_disable_headset_jack_key(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (!spec->has_hs_key) + return; + + switch (codec->core.vendor_id) { + case 0x10ec0215: + case 0x10ec0225: + case 0x10ec0285: + case 0x10ec0295: + case 0x10ec0289: + case 0x10ec0299: + alc_write_coef_idx(codec, 0x48, 0x0); + alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); + alc_update_coef_idx(codec, 0x44, 0x0045 << 8, 0x0); + break; + case 0x10ec0236: + case 0x10ec0256: + alc_write_coef_idx(codec, 0x48, 0x0); + alc_update_coef_idx(codec, 0x49, 0x0045, 0x0); + break; + } +} + +static void alc_enable_headset_jack_key(struct hda_codec *codec) +{ + struct alc_spec *spec = codec->spec; + + if (!spec->has_hs_key) + return; + + switch (codec->core.vendor_id) { + case 0x10ec0215: + case 0x10ec0225: + case 0x10ec0285: + case 0x10ec0295: + case 0x10ec0289: + case 0x10ec0299: + alc_write_coef_idx(codec, 0x48, 0xd011); + alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); + alc_update_coef_idx(codec, 0x44, 0x007f << 8, 0x0045 << 8); + break; + case 0x10ec0236: + case 0x10ec0256: + alc_write_coef_idx(codec, 0x48, 0xd011); + alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); + break; + } +} + +static void alc_fixup_headset_jack(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + spec->has_hs_key = 1; + snd_hda_jack_detect_enable_callback(codec, 0x55, + alc_headset_btn_callback); + snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", false, + SND_JACK_HEADSET, alc_headset_btn_keymap); + break; + case HDA_FIXUP_ACT_INIT: + alc_enable_headset_jack_key(codec); + break; + } +} + static void alc269vb_toggle_power_output(struct hda_codec *codec, int power_up) { alc_update_coef_idx(codec, 0x04, 1 << 11, power_up ? (1 << 11) : 0); @@ -3372,6 +3474,8 @@ static void alc225_shutup(struct hda_codec *codec) if (!hp_pin) hp_pin = 0x21; + + alc_disable_headset_jack_key(codec); /* 3k pull low control for Headset jack. */ alc_update_coef_idx(codec, 0x4a, 0, 3 << 10); @@ -3411,6 +3515,9 @@ static void alc225_shutup(struct hda_codec *codec) alc_update_coef_idx(codec, 0x4a, 3<<4, 2<<4); msleep(30); } + + alc_update_coef_idx(codec, 0x4a, 3 << 10, 0); + alc_enable_headset_jack_key(codec); } static void alc_default_init(struct hda_codec *codec) @@ -5668,69 +5775,6 @@ static void alc285_fixup_invalidate_dacs(struct hda_codec *codec, snd_hda_override_wcaps(codec, 0x03, 0); } -static const struct hda_jack_keymap alc_headset_btn_keymap[] = { - { SND_JACK_BTN_0, KEY_PLAYPAUSE }, - { SND_JACK_BTN_1, KEY_VOICECOMMAND }, - { SND_JACK_BTN_2, KEY_VOLUMEUP }, - { SND_JACK_BTN_3, KEY_VOLUMEDOWN }, - {} -}; - -static void alc_headset_btn_callback(struct hda_codec *codec, - struct hda_jack_callback *jack) -{ - int report = 0; - - if (jack->unsol_res & (7 << 13)) - report |= SND_JACK_BTN_0; - - if (jack->unsol_res & (1 << 16 | 3 << 8)) - report |= SND_JACK_BTN_1; - - /* Volume up key */ - if (jack->unsol_res & (7 << 23)) - report |= SND_JACK_BTN_2; - - /* Volume down key */ - if (jack->unsol_res & (7 << 10)) - report |= SND_JACK_BTN_3; - - jack->jack->button_state = report; -} - -static void alc_fixup_headset_jack(struct hda_codec *codec, - const struct hda_fixup *fix, int action) -{ - - switch (action) { - case HDA_FIXUP_ACT_PRE_PROBE: - snd_hda_jack_detect_enable_callback(codec, 0x55, - alc_headset_btn_callback); - snd_hda_jack_add_kctl(codec, 0x55, "Headset Jack", false, - SND_JACK_HEADSET, alc_headset_btn_keymap); - break; - case HDA_FIXUP_ACT_INIT: - switch (codec->core.vendor_id) { - case 0x10ec0215: - case 0x10ec0225: - case 0x10ec0285: - case 0x10ec0295: - case 0x10ec0289: - case 0x10ec0299: - alc_write_coef_idx(codec, 0x48, 0xd011); - alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); - alc_update_coef_idx(codec, 0x44, 0x007f << 8, 0x0045 << 8); - break; - case 0x10ec0236: - case 0x10ec0256: - alc_write_coef_idx(codec, 0x48, 0xd011); - alc_update_coef_idx(codec, 0x49, 0x007f, 0x0045); - break; - } - break; - } -} - static void alc295_fixup_chromebook(struct hda_codec *codec, const struct hda_fixup *fix, int action) { -- 2.17.1

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about alc880 no sound and eapd coef
by 尤晓杰 29 Mar '20

29 Mar '20

dear friend, I am using ecs g420 laptop, use options snd-hda-intel model=generic the speaker and mic correctly, 2 pop sounds,when the kernel load(left channel) and the x11 load(right channel), but with standard way only one pop sound when kernel load and silent always. the mic is ok. I have got datasheet of alc880 and see the eapd function, but in your code I see this need coef parameter. so what should I write using verb to open eapd? the attachment is the schematic of sound part.in summary: the 0x14 hp out, 0x18 mic in,0x1c cd in, 0x1e spif out, no jack sense. eapd shared with spdif in. all are correct with default of datasheet's describe. I use verb to dectt each pin widget's capability , take 0x14 as exaple, no eapd capability.To set 0x14's eapd with hda-verb /dev/snd/hwC0D0 0x70c 0x2, failed to read it again alwas 0. and bios also reported 0x16, I think it is bios's bug. I what to use "Early Patching" method to deal with it without changing source code, is it feasible? could you give me some idea? thanks!!!! dmesg: [ 8.009944] snd_hda_codec_realtek hdaudioC0D0: autoconfig for ALC880: line_outs=2 (0x14/0x16/0x0/0x0/0x0) type:hp [ 8.014023] snd_hda_codec_realtek hdaudioC0D0: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 8.017999] snd_hda_codec_realtek hdaudioC0D0: hp_outs=0 (0x0/0x0/0x0/0x0/0x0) [ 8.021625] snd_hda_codec_realtek hdaudioC0D0: mono: mono_out=0x0 [ 8.024255] snd_hda_codec_realtek hdaudioC0D0: dig-out=0x1e/0x0 [ 8.026741] snd_hda_codec_realtek hdaudioC0D0: inputs: [ 8.038468] mc: Linux media interface: v0.10 [ 8.067515] snd_hda_codec_realtek hdaudioC0D0: Mic=0x18 [ 8.078984] snd_hda_codec_realtek hdaudioC0D0: CD=0x1c [ 8.290306] input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:1b.0/sound/card0/input13

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Port HDA Analyzer to python 3
by andythe_great 29 Mar '20

29 Mar '20

Hello, Is it possible to port HDA Analyzer to python 3? https://www.alsa-project.org/main/index.php/HDA_Analyzer Because openSUSE is retiring python 2. I tried using python-modernize https://python-modernize.readthedocs.io/en/latest/ but many could not get it to run. Thanks. Andy

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[PATCH] ALSA: hda: default enable CA0132 DSP support
by Rouven Czerwinski 29 Mar '20

29 Mar '20

From: Rouven Czerwinski <rouven(a)czerwinskis.de> If SND_HDA_CODEC_CA0132 is enabled, the DSP support should be enabled as well. Disabled DSP support leads to a hanging alsa system and no sound output on the card otherwise. Tested on: 06:00.0 Audio device: Creative Labs Sound Core3D [Sound Blaster Recon3D / Z-Series] (rev 01) Signed-off-by: Rouven Czerwinski <rouven(a)czerwinskis.de> --- sound/pci/hda/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index bd48335d09d7..e1d3082a4fe9 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -184,6 +184,7 @@ comment "Set to Y if you want auto-loading the codec driver" config SND_HDA_CODEC_CA0132_DSP bool "Support new DSP code for CA0132 codec" depends on SND_HDA_CODEC_CA0132 + default y select SND_HDA_DSP_LOADER select FW_LOADER help base-commit: e595dd94515ed6bc5ba38fce0f9598db8c0ee9a9 -- 2.25.1

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[alsa-devel] [PATCH 0/3] ALSA: bebob: misc improvements for starting/stopping packet streaming
by Takashi Sakamoto 27 Mar '20

27 Mar '20

Hi, This patchset improves model-specific issues in a point of protocol version supported by the model. The changes are relevant to packet streaming, thus it can brings some regressions. For testers, I prepare for a remote branch to backport the latest codes to Linux v4.17 or later. I'm happy to receive any report. https://github.com/takaswie/snd-firewire-improve/tree/topic/bebob-connectio… Takashi Sakamoto (3): ALSA: bebob: expand sleep just after breaking connections for protocol version 1 ALSA: bebob: expand delay of start for IR context just for version 3 firmware ALSA: bebob: link the order of establishing connections and Syt-match clock mode sound/firewire/bebob/bebob_stream.c | 76 ++++++++++++++--------------- 1 file changed, 38 insertions(+), 38 deletions(-) -- 2.20.1

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[PATCH] ASoC: topology: Add missing memory checks
by Amadeusz Sławiński 27 Mar '20

27 Mar '20

kstrdup is an allocation function and it can fail, so its return value should be checked and handled appropriately. In order to check all cases, we need to modify set_stream_info to return a value, so check that everything went correctly when doing kstrdup(). Later add proper checks and error handlers. Signed-off-by: Amadeusz Sławiński <amadeuszx.slawinski(a)linux.intel.com> --- sound/soc/soc-topology.c | 65 +++++++++++++++++++++++++++++++--------- 1 file changed, 51 insertions(+), 14 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 575da6aba807..0bec3ff782c1 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1766,10 +1766,13 @@ static int soc_tplg_dapm_complete(struct soc_tplg *tplg) return 0; } -static void set_stream_info(struct snd_soc_pcm_stream *stream, +static int set_stream_info(struct snd_soc_pcm_stream *stream, struct snd_soc_tplg_stream_caps *caps) { stream->stream_name = kstrdup(caps->name, GFP_KERNEL); + if (!stream->stream_name) + return -ENOMEM; + stream->channels_min = le32_to_cpu(caps->channels_min); stream->channels_max = le32_to_cpu(caps->channels_max); stream->rates = le32_to_cpu(caps->rates); @@ -1777,6 +1780,8 @@ static void set_stream_info(struct snd_soc_pcm_stream *stream, stream->rate_max = le32_to_cpu(caps->rate_max); stream->formats = le64_to_cpu(caps->formats); stream->sig_bits = le32_to_cpu(caps->sig_bits); + + return 0; } static void set_dai_flags(struct snd_soc_dai_driver *dai_drv, @@ -1812,20 +1817,29 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, if (dai_drv == NULL) return -ENOMEM; - if (strlen(pcm->dai_name)) + if (strlen(pcm->dai_name)) { dai_drv->name = kstrdup(pcm->dai_name, GFP_KERNEL); + if (!dai_drv->name) { + ret = -ENOMEM; + goto err; + } + } dai_drv->id = le32_to_cpu(pcm->dai_id); if (pcm->playback) { stream = &dai_drv->playback; caps = &pcm->caps[SND_SOC_TPLG_STREAM_PLAYBACK]; - set_stream_info(stream, caps); + ret = set_stream_info(stream, caps); + if (ret < 0) + goto err; } if (pcm->capture) { stream = &dai_drv->capture; caps = &pcm->caps[SND_SOC_TPLG_STREAM_CAPTURE]; - set_stream_info(stream, caps); + ret = set_stream_info(stream, caps); + if (ret < 0) + goto err; } if (pcm->compress) @@ -1835,11 +1849,7 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, ret = soc_tplg_dai_load(tplg, dai_drv, pcm, NULL); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n"); - kfree(dai_drv->playback.stream_name); - kfree(dai_drv->capture.stream_name); - kfree(dai_drv->name); - kfree(dai_drv); - return ret; + goto err; } dai_drv->dobj.index = tplg->index; @@ -1857,9 +1867,17 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, if (ret != 0) { dev_err(dai->dev, "Failed to create DAI widgets %d\n", ret); snd_soc_unregister_dai(dai); - return ret; + goto err; } + return 0; + +err: + kfree(dai_drv->playback.stream_name); + kfree(dai_drv->capture.stream_name); + kfree(dai_drv->name); + kfree(dai_drv); + return ret; } @@ -1916,11 +1934,20 @@ static int soc_tplg_fe_link_create(struct soc_tplg *tplg, if (strlen(pcm->pcm_name)) { link->name = kstrdup(pcm->pcm_name, GFP_KERNEL); link->stream_name = kstrdup(pcm->pcm_name, GFP_KERNEL); + if (!link->name || !link->stream_name) { + ret = -ENOMEM; + goto err; + } } link->id = le32_to_cpu(pcm->pcm_id); - if (strlen(pcm->dai_name)) + if (strlen(pcm->dai_name)) { link->cpus->dai_name = kstrdup(pcm->dai_name, GFP_KERNEL); + if (!link->cpus->dai_name) { + ret = -ENOMEM; + goto err; + } + } link->codecs->name = "snd-soc-dummy"; link->codecs->dai_name = "snd-soc-dummy-dai"; @@ -2436,13 +2463,17 @@ static int soc_tplg_dai_config(struct soc_tplg *tplg, if (d->playback) { stream = &dai_drv->playback; caps = &d->caps[SND_SOC_TPLG_STREAM_PLAYBACK]; - set_stream_info(stream, caps); + ret = set_stream_info(stream, caps); + if (ret < 0) + goto err; } if (d->capture) { stream = &dai_drv->capture; caps = &d->caps[SND_SOC_TPLG_STREAM_CAPTURE]; - set_stream_info(stream, caps); + ret = set_stream_info(stream, caps); + if (ret < 0) + goto err; } if (d->flag_mask) @@ -2454,10 +2485,16 @@ static int soc_tplg_dai_config(struct soc_tplg *tplg, ret = soc_tplg_dai_load(tplg, dai_drv, NULL, dai); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n"); - return ret; + goto err; } return 0; + +err: + kfree(dai_drv->playback.stream_name); + kfree(dai_drv->capture.stream_name); + + return ret; } /* load physical DAI elements */ -- 2.17.1

2 2

[PATCH 0/4] ASoC: Intel: add SoundWire machine driver
by Pierre-Louis Bossart 27 Mar '20

27 Mar '20

To handle multiple hardware combinations, this patchset suggests a single machine driver which will create and initialize dailinks dynamically. This allows us to support new configurations easily, as shown with the TigerLake rt5682 example. Each configuration updates the card component string, and UCM can test for the presence of components to configure them as needed. Since we use a single the machine driver name, all previous ACPI tables need to be updated. That should have no impact since the machine drivers listed at the time were not upstreamed and are no longer maintained. Naveen Manohar (2): ASoC: Intel: common: add match table for TGL RT5682 SoundWire driver ASoC: Intel: sof_sdw: Add Volteer support with RT5682 SNDW helper function Pierre-Louis Bossart (1): ASoC: Intel: boards: add sof_sdw machine driver Rander Wang (1): ASoC: Intel: soc-acpi: update topology and driver name for SoundWire platforms sound/soc/intel/boards/Kconfig | 24 + sound/soc/intel/boards/Makefile | 8 +- sound/soc/intel/boards/sof_sdw.c | 962 ++++++++++++++++++ sound/soc/intel/boards/sof_sdw_common.h | 114 +++ sound/soc/intel/boards/sof_sdw_dmic.c | 42 + sound/soc/intel/boards/sof_sdw_hdmi.c | 97 ++ sound/soc/intel/boards/sof_sdw_rt1308.c | 151 +++ sound/soc/intel/boards/sof_sdw_rt5682.c | 126 +++ sound/soc/intel/boards/sof_sdw_rt700.c | 125 +++ sound/soc/intel/boards/sof_sdw_rt711.c | 156 +++ sound/soc/intel/boards/sof_sdw_rt715.c | 42 + .../intel/common/soc-acpi-intel-cml-match.c | 24 +- .../intel/common/soc-acpi-intel-icl-match.c | 6 +- .../intel/common/soc-acpi-intel-tgl-match.c | 30 +- 14 files changed, 1896 insertions(+), 11 deletions(-) create mode 100644 sound/soc/intel/boards/sof_sdw.c create mode 100644 sound/soc/intel/boards/sof_sdw_common.h create mode 100644 sound/soc/intel/boards/sof_sdw_dmic.c create mode 100644 sound/soc/intel/boards/sof_sdw_hdmi.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt1308.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt5682.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt700.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt711.c create mode 100644 sound/soc/intel/boards/sof_sdw_rt715.c -- 2.20.1

2 11

[PATCH] dt-bindings: sound: tlv320adcx140: Remove undocumented property
by Dan Murphy 27 Mar '20

27 Mar '20

Remove undocumented and unneeded ti,use-internal-reg from the example as it was an artifact from initial development. The code does not query for this property and as the document indicates if areg-supply is undefined then the internal regulator is used. Fixes: 302c0b7490cd ("dt-bindings: sound: Add TLV320ADCx140 dt bindings") CC: Rob Herring <robh(a)kernel.org> Signed-off-by: Dan Murphy <dmurphy(a)ti.com> --- Documentation/devicetree/bindings/sound/tlv320adcx140.yaml | 1 - 1 file changed, 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml index 1433ff62b14f..ab2268c0ee67 100644 --- a/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml +++ b/Documentation/devicetree/bindings/sound/tlv320adcx140.yaml @@ -76,7 +76,6 @@ examples: codec: codec@4c { compatible = "ti,tlv320adc5140"; reg = <0x4c>; - ti,use-internal-areg; ti,mic-bias-source = <6>; reset-gpios = <&gpio0 14 GPIO_ACTIVE_HIGH>; }; -- 2.25.1

2 1

[PATCH] ASoC: amd: acp3x-pcm-dma: clean up two indentation issues
by Colin King 27 Mar '20

27 Mar '20

From: Colin Ian King <colin.king(a)canonical.com> There are a couple of statements that are not indented correctly, add in the missing tab and break the lines to address a checkpatch warning. Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/amd/raven/acp3x-pcm-dma.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/raven/acp3x-pcm-dma.c b/sound/soc/amd/raven/acp3x-pcm-dma.c index d62c0d90c41e..e362f0bc9e46 100644 --- a/sound/soc/amd/raven/acp3x-pcm-dma.c +++ b/sound/soc/amd/raven/acp3x-pcm-dma.c @@ -458,7 +458,8 @@ static int acp3x_resume(struct device *dev) reg_val = mmACP_I2STDM_ITER; frmt_val = mmACP_I2STDM_TXFRMT; } - rv_writel((rtd->xfer_resolution << 3), rtd->acp3x_base + reg_val); + rv_writel((rtd->xfer_resolution << 3), + rtd->acp3x_base + reg_val); } if (adata->capture_stream && adata->capture_stream->runtime) { struct i2s_stream_instance *rtd = @@ -474,7 +475,8 @@ static int acp3x_resume(struct device *dev) reg_val = mmACP_I2STDM_IRER; frmt_val = mmACP_I2STDM_RXFRMT; } - rv_writel((rtd->xfer_resolution << 3), rtd->acp3x_base + reg_val); + rv_writel((rtd->xfer_resolution << 3), + rtd->acp3x_base + reg_val); } if (adata->tdm_mode == TDM_ENABLE) { rv_writel(adata->tdm_fmt, adata->acp3x_base + frmt_val); -- 2.25.1

2 1

[PATCH] additional cleanup for "arecord -L show entries which are not capture pcms"
by sylvain.bertrand＠gmail.com 27 Mar '20

27 Mar '20

I did post to github, my bad. On Fri, Mar 27, 2020 at 04:01:57AM -0700, Jaroslav Kysela wrote: > Closed #39 via d434638a683c01bd6decf8f41863044055237a50. Tested with the mic from my usb webcam, and it actually needs a little bit more tidying. -- Sylvain --- a/share/alsa/pcm/front.conf +++ b/share/alsa/pcm/front.conf @@ -53,6 +53,7 @@ name defaults.namehint.basic } description "Front speakers" - device $DEV + device_output $DEV + omit_noargs true } } --- a/share/alsa/pcm/iec958.conf +++ b/share/alsa/pcm/iec958.conf @@ -78,6 +78,6 @@ name defaults.namehint.basic } description "IEC958 (S/PDIF) Digital Audio Output" - device $DEV + device_output $DEV } }

1 0

route and chmap
by Jonas Holmberg 27 Mar '20

27 Mar '20

Hi, I use the route plugin to copy mono to both FL and FR (front left and front right stereo) of the slave during playback. When get_chmap() is called in the route plugin it will call get_chmap() of the slave, which returns 2 channels with FL and FR positions, and then route itself returns 1 channel with FL position. But the application that called snd_pcm_get_chmap() on the playback device expected to get position MONO instead of FL since the number of channels is 1. Is the application wrong in not accepting the chmap or should the route plugin have changed position from FL to MONO? Is it possible to add something to the config (asound.conf) to make FL become MONO in the reply of get_chmap() or solve the problem in any other way? BR /Jonas

2 2

[PATCH 00/36] ASoC: remove rtd->cpu/codec_dai{s}
by Kuninori Morimoto 27 Mar '20

27 Mar '20

Hi Mark Now, CPU/Codec DAI(s) were replaced by rtd->dais. Thus, We don't need rtd->cpu/codec_dai{s} anymore. This pathset replaces it by new macro. Kuninori Morimoto (36): ASoC: soc-core: add asoc_rtd_to_cpu/codec() macro ASoC: amd: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: atmel: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: au1x: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: bcm: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: cirrus: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: dwc: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: fsl: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: generic: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: img: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: intel: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: kirkwood: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: mediatek: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: meson: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: mxs: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: pxa: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: qcom: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: rockchip: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: samsung: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: sh: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: sof: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: sprd: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: stm: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: sunxi: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: tegra: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: ti: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: txx9: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: uniphier: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: ux500: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: xtensa: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: arm: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: codecs: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: soc: use asoc_rtd_to_cpu() / asoc_rtd_to_codec() macro for DAI pointer ASoC: soc-core: set rtd->num_cpu/codec at soc_new_pcm_runtime() ASoC: soc-core: tidyup soc_new_pcm_runtime() rtd setups ASoC: soc-core: remove cpu_dai/codec_dai/cpu_dais/codec_dais include/sound/soc.h | 30 +++++++------ sound/arm/pxa2xx-pcm-lib.c | 8 ++-- sound/soc/amd/acp-da7219-max98357a.c | 2 +- sound/soc/amd/acp-rt5645.c | 4 +- sound/soc/amd/acp3x-rt5682-max9836.c | 6 +-- sound/soc/atmel/atmel-pcm-dma.c | 4 +- sound/soc/atmel/atmel-pcm-pdc.c | 2 +- sound/soc/atmel/atmel_wm8904.c | 2 +- sound/soc/atmel/mikroe-proto.c | 2 +- sound/soc/atmel/sam9g20_wm8731.c | 2 +- sound/soc/atmel/sam9x5_wm8731.c | 2 +- sound/soc/au1x/db1200.c | 2 +- sound/soc/au1x/dbdma2.c | 2 +- sound/soc/au1x/dma.c | 2 +- sound/soc/au1x/psc-ac97.c | 2 +- sound/soc/bcm/bcm63xx-pcm-whistler.c | 16 +++---- sound/soc/bcm/cygnus-pcm.c | 22 +++++----- sound/soc/cirrus/edb93xx.c | 4 +- sound/soc/cirrus/snappercl15.c | 4 +- sound/soc/codecs/cs47l15.c | 4 +- sound/soc/codecs/cs47l24.c | 6 +-- sound/soc/codecs/cs47l35.c | 6 +-- sound/soc/codecs/cs47l85.c | 6 +-- sound/soc/codecs/cs47l90.c | 6 +-- sound/soc/codecs/cs47l92.c | 4 +- sound/soc/codecs/wm5110.c | 6 +-- sound/soc/codecs/wm_adsp.c | 10 ++--- sound/soc/dwc/dwc-pcm.c | 2 +- sound/soc/fsl/eukrea-tlv320.c | 4 +- sound/soc/fsl/fsl-asoc-card.c | 10 ++--- sound/soc/fsl/fsl_asrc_dma.c | 6 +-- sound/soc/fsl/fsl_spdif.c | 10 ++--- sound/soc/fsl/fsl_ssi.c | 8 ++-- sound/soc/fsl/imx-audmix.c | 8 ++-- sound/soc/fsl/imx-mc13783.c | 4 +- sound/soc/fsl/imx-sgtl5000.c | 2 +- sound/soc/fsl/mpc5200_dma.c | 10 ++--- sound/soc/fsl/mpc5200_psc_i2s.c | 2 +- sound/soc/fsl/mpc8610_hpcd.c | 4 +- sound/soc/fsl/mx27vis-aic32x4.c | 4 +- sound/soc/fsl/p1022_ds.c | 4 +- sound/soc/fsl/p1022_rdk.c | 4 +- sound/soc/fsl/wm1133-ev1.c | 6 +-- sound/soc/generic/simple-card-utils.c | 12 +++--- sound/soc/img/img-i2s-in.c | 2 +- sound/soc/img/img-i2s-out.c | 2 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 6 +-- sound/soc/intel/boards/bdw-rt5650.c | 6 +-- sound/soc/intel/boards/bdw-rt5677.c | 6 +-- sound/soc/intel/boards/broadwell.c | 4 +- sound/soc/intel/boards/bxt_da7219_max98357a.c | 8 ++-- sound/soc/intel/boards/bxt_rt298.c | 8 ++-- sound/soc/intel/boards/byt-max98090.c | 2 +- sound/soc/intel/boards/byt-rt5640.c | 4 +- sound/soc/intel/boards/bytcht_cx2072x.c | 10 ++--- sound/soc/intel/boards/bytcht_da7213.c | 8 ++-- sound/soc/intel/boards/bytcht_es8316.c | 8 ++-- sound/soc/intel/boards/bytcht_nocodec.c | 4 +- sound/soc/intel/boards/bytcr_rt5640.c | 8 ++-- sound/soc/intel/boards/bytcr_rt5651.c | 8 ++-- sound/soc/intel/boards/cht_bsw_max98090_ti.c | 6 +-- sound/soc/intel/boards/cht_bsw_nau8824.c | 4 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 14 +++---- sound/soc/intel/boards/cht_bsw_rt5672.c | 8 ++-- sound/soc/intel/boards/cml_rt1011_rt5682.c | 6 +-- sound/soc/intel/boards/glk_rt5682_max98357a.c | 10 ++--- sound/soc/intel/boards/haswell.c | 2 +- sound/soc/intel/boards/kbl_da7219_max98357a.c | 8 ++-- sound/soc/intel/boards/kbl_da7219_max98927.c | 6 +-- sound/soc/intel/boards/kbl_rt5660.c | 6 +-- sound/soc/intel/boards/kbl_rt5663_max98927.c | 8 ++-- .../intel/boards/kbl_rt5663_rt5514_max98927.c | 8 ++-- .../soc/intel/boards/skl_nau88l25_max98357a.c | 12 +++--- sound/soc/intel/boards/skl_nau88l25_ssm4567.c | 16 +++---- sound/soc/intel/boards/skl_rt286.c | 8 ++-- sound/soc/intel/boards/sof_da7219_max98373.c | 8 ++-- sound/soc/intel/boards/sof_pcm512x.c | 8 ++-- sound/soc/intel/boards/sof_rt5682.c | 6 +-- sound/soc/intel/haswell/sst-haswell-pcm.c | 26 ++++++------ sound/soc/intel/skylake/skl-pcm.c | 10 ++--- sound/soc/kirkwood/armada-370-db.c | 2 +- sound/soc/kirkwood/kirkwood-dma.c | 2 +- sound/soc/mediatek/common/mtk-afe-fe-dai.c | 10 ++--- .../mediatek/common/mtk-afe-platform-driver.c | 2 +- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 2 +- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 4 +- sound/soc/mediatek/mt2701/mt2701-wm8960.c | 4 +- sound/soc/mediatek/mt6797/mt6797-afe-pcm.c | 2 +- sound/soc/mediatek/mt8173/mt8173-afe-pcm.c | 2 +- sound/soc/mediatek/mt8173/mt8173-max98090.c | 4 +- .../mediatek/mt8173/mt8173-rt5650-rt5514.c | 2 +- .../mediatek/mt8173/mt8173-rt5650-rt5676.c | 4 +- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 6 +-- sound/soc/mediatek/mt8183/mt8183-afe-pcm.c | 2 +- .../mediatek/mt8183/mt8183-da7219-max98357.c | 4 +- .../mt8183/mt8183-mt6358-ts3a227-max98357.c | 2 +- sound/soc/meson/aiu-fifo.c | 2 +- sound/soc/meson/axg-card.c | 8 ++-- sound/soc/meson/axg-fifo.c | 2 +- sound/soc/meson/meson-card-utils.c | 2 +- sound/soc/mxs/mxs-sgtl5000.c | 4 +- sound/soc/pxa/brownstone.c | 4 +- sound/soc/pxa/corgi.c | 4 +- sound/soc/pxa/hx4700.c | 4 +- sound/soc/pxa/imote2.c | 4 +- sound/soc/pxa/magician.c | 8 ++-- sound/soc/pxa/mioa701_wm9713.c | 4 +- sound/soc/pxa/mmp-pcm.c | 2 +- sound/soc/pxa/mmp-sspa.c | 2 +- sound/soc/pxa/poodle.c | 4 +- sound/soc/pxa/pxa2xx-i2s.c | 2 +- sound/soc/pxa/spitz.c | 4 +- sound/soc/pxa/ttc-dkb.c | 2 +- sound/soc/pxa/z2.c | 4 +- sound/soc/pxa/zylonite.c | 6 +-- sound/soc/qcom/apq8016_sbc.c | 2 +- sound/soc/qcom/apq8096.c | 6 +-- sound/soc/qcom/lpass-platform.c | 2 +- sound/soc/qcom/qdsp6/q6asm-dai.c | 4 +- sound/soc/qcom/qdsp6/q6routing.c | 2 +- sound/soc/qcom/sdm845.c | 22 +++++----- sound/soc/qcom/storm.c | 2 +- sound/soc/rockchip/rk3288_hdmi_analog.c | 4 +- sound/soc/rockchip/rk3399_gru_sound.c | 16 +++---- sound/soc/rockchip/rockchip_max98090.c | 6 +-- sound/soc/rockchip/rockchip_rt5645.c | 6 +-- sound/soc/samsung/arndale.c | 6 +-- sound/soc/samsung/bells.c | 16 +++---- sound/soc/samsung/h1940_uda1380.c | 2 +- sound/soc/samsung/i2s.c | 2 +- sound/soc/samsung/jive_wm8750.c | 4 +- sound/soc/samsung/littlemill.c | 14 +++---- sound/soc/samsung/lowland.c | 4 +- sound/soc/samsung/neo1973_wm8753.c | 10 ++--- sound/soc/samsung/odroid.c | 2 +- sound/soc/samsung/pcm.c | 4 +- sound/soc/samsung/rx1950_uda1380.c | 2 +- sound/soc/samsung/s3c-i2s-v2.c | 2 +- sound/soc/samsung/s3c24xx_simtec.c | 4 +- sound/soc/samsung/s3c24xx_uda134x.c | 6 +-- sound/soc/samsung/smartq_wm8987.c | 4 +- sound/soc/samsung/smdk_spdif.c | 2 +- sound/soc/samsung/smdk_wm8580.c | 2 +- sound/soc/samsung/smdk_wm8994.c | 2 +- sound/soc/samsung/smdk_wm8994pcm.c | 4 +- sound/soc/samsung/snow.c | 4 +- sound/soc/samsung/spdif.c | 8 ++-- sound/soc/samsung/speyside.c | 8 ++-- sound/soc/samsung/tm2_wm5110.c | 16 +++---- sound/soc/samsung/tobermory.c | 8 ++-- sound/soc/sh/dma-sh7760.c | 16 +++---- sound/soc/sh/fsi.c | 2 +- sound/soc/sh/migor.c | 6 +-- sound/soc/sh/rcar/core.c | 2 +- sound/soc/soc-compress.c | 36 ++++++++-------- sound/soc/soc-core.c | 42 +++++++------------ sound/soc/soc-dapm.c | 4 +- sound/soc/soc-generic-dmaengine-pcm.c | 6 +-- sound/soc/soc-pcm.c | 30 ++++++------- sound/soc/sof/intel/hda-dai.c | 6 +-- sound/soc/sof/intel/hda-dsp.c | 2 +- sound/soc/sprd/sprd-pcm-compress.c | 4 +- sound/soc/sprd/sprd-pcm-dma.c | 2 +- sound/soc/stm/stm32_adfsdm.c | 12 +++--- sound/soc/stm/stm32_sai_sub.c | 2 +- sound/soc/sunxi/sun4i-spdif.c | 2 +- sound/soc/tegra/tegra_alc5632.c | 2 +- sound/soc/tegra/tegra_max98090.c | 2 +- sound/soc/tegra/tegra_rt5640.c | 2 +- sound/soc/tegra/tegra_rt5677.c | 2 +- sound/soc/tegra/tegra_sgtl5000.c | 2 +- sound/soc/tegra/tegra_wm8753.c | 2 +- sound/soc/tegra/tegra_wm8903.c | 6 +-- sound/soc/tegra/trimslice.c | 2 +- sound/soc/ti/ams-delta.c | 4 +- sound/soc/ti/davinci-evm.c | 4 +- sound/soc/ti/davinci-vcif.c | 4 +- sound/soc/ti/n810.c | 2 +- sound/soc/ti/omap-abe-twl6040.c | 6 +-- sound/soc/ti/omap-mcbsp-st.c | 2 +- sound/soc/ti/omap-mcbsp.c | 4 +- sound/soc/ti/omap-mcpdm.c | 2 +- sound/soc/ti/omap3pandora.c | 4 +- sound/soc/ti/osk5912.c | 2 +- sound/soc/ti/rx51.c | 2 +- sound/soc/txx9/txx9aclc.c | 2 +- sound/soc/uniphier/aio-compress.c | 22 +++++----- sound/soc/uniphier/aio-dma.c | 6 +-- sound/soc/ux500/mop500_ab8500.c | 6 +-- sound/soc/ux500/ux500_pcm.c | 8 ++-- sound/soc/xtensa/xtfpga-i2s.c | 2 +- 191 files changed, 573 insertions(+), 577 deletions(-) -- 2.17.1

3 72

[PATCH v2 00/11] ASoC: SOF: Intel: add SoundWire support
by Pierre-Louis Bossart 27 Mar '20

27 Mar '20

This patchset provides the support for SoundWire support on Intel CometLake, IcelLake and TigerLake RVP platforms and form-factor devices to be released 'soon'. The bulk of the code is about detecting a valid SoundWire configuration from ACPI, and implementing the interfaces suggested in '[PATCH 0/8] soundwire: remove platform devices, add SOF interfaces' for interrupts, PCI wakes and clock-stop configurations. Since that SoundWire series will not be in 5.7, the build support for SOF w/ SoundWire is not provided for now, and fall-back functions will be used. This code is tested on a daily basis in the SOF tree and is not expected to change in significant ways. Changes since v2: Corrected error in ACPI table (thanks Amadeusz) Added patch 11 to add reset cycle required on some SoundWire platforms Bard Liao (1): ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire interrupt handlers Pierre-Louis Bossart (8): ASoC: soc-acpi: expand description of _ADR-based devices ASoC: SOF: Intel: add SoundWire configuration interface ASoC: SOF: IPC: dai-intel: move ALH declarations in header file ASoC: SOF: Intel: hda: add SoundWire stream config/free callbacks ASoC: SOF: Intel: hda: initial SoundWire machine driver autodetect ASoC: SOF: Intel: hda: disable SoundWire interrupts on suspend ASoC: SOF: Intel: hda: add parameter to control SoundWire clock stop quirks ASoC: SOF: Intel: hda-ctrl: add reset cycle before parsing capabilities Rander Wang (2): ASoC: SOF: Intel: hda: add WAKEEN interrupt support for SoundWire Asoc: SOF: Intel: hda: check SoundWire wakeen interrupt in irq thread include/sound/soc-acpi.h | 39 +- include/sound/sof/dai-intel.h | 18 +- .../intel/common/soc-acpi-intel-cml-match.c | 87 +++- .../intel/common/soc-acpi-intel-icl-match.c | 97 ++++- .../intel/common/soc-acpi-intel-tgl-match.c | 49 ++- sound/soc/sof/intel/hda-ctrl.c | 25 +- sound/soc/sof/intel/hda-dsp.c | 2 + sound/soc/sof/intel/hda-loader.c | 31 ++ sound/soc/sof/intel/hda.c | 400 ++++++++++++++++++ sound/soc/sof/intel/hda.h | 66 +++ 10 files changed, 750 insertions(+), 64 deletions(-) -- 2.20.1

2 14

Applied "ASoC: SOF: Intel: add SoundWire configuration interface" to the asoc tree
by Mark Brown 27 Mar '20

27 Mar '20

The patch ASoC: SOF: Intel: add SoundWire configuration interface 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 51dfed1e178a38202960b98f6e29df009a06050f Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Wed, 25 Mar 2020 16:50:18 -0500 Subject: [PATCH] ASoC: SOF: Intel: add SoundWire configuration interface Now that the SoundWire core supports the multi-step initialization, call the relevant APIs. The actual hardware enablement can be done in two places, ideally we'd want to startup the SoundWire IP as soon as possible (while still taking power rail dependencies into account) However when suspend/resume is implemented, the DSP device will be resumed first, and only when the DSP firmware is downloaded/booted would the SoundWire child devices be resumed, so there are only marginal benefits in starting the IP earlier for the first probe. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200325215027.28716-3-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/intel/hda-loader.c | 13 ++++ sound/soc/sof/intel/hda.c | 120 +++++++++++++++++++++++++++++++ sound/soc/sof/intel/hda.h | 44 ++++++++++++ 3 files changed, 177 insertions(+) diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 0633b76dab49..2ae94ea53122 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -399,6 +399,19 @@ int hda_dsp_pre_fw_run(struct snd_sof_dev *sdev) /* post fw run operations */ int hda_dsp_post_fw_run(struct snd_sof_dev *sdev) { + int ret; + + if (sdev->first_boot) { + ret = hda_sdw_startup(sdev); + if (ret < 0) { + dev_err(sdev->dev, + "error: could not startup SoundWire links\n"); + return ret; + } + } + + hda_sdw_int_enable(sdev, true); + /* re-enable clock gating and power gating */ return hda_dsp_ctrl_clock_power_gating(sdev, true); } diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index ee1edb53840c..c1fe94800da1 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -18,7 +18,9 @@ #include <sound/hdaudio_ext.h> #include <sound/hda_register.h> +#include <linux/acpi.h> #include <linux/module.h> +#include <linux/soundwire/sdw_intel.h> #include <sound/intel-nhlt.h> #include <sound/sof.h> #include <sound/sof/xtensa.h> @@ -34,6 +36,98 @@ #define EXCEPT_MAX_HDR_SIZE 0x400 +#if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) + +void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable) +{ + sdw_intel_enable_irq(sdev->bar[HDA_DSP_BAR], enable); +} + +static int hda_sdw_acpi_scan(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + acpi_handle handle; + int ret; + + handle = ACPI_HANDLE(sdev->dev); + + /* save ACPI info for the probe step */ + hdev = sdev->pdata->hw_pdata; + + ret = sdw_intel_acpi_scan(handle, &hdev->info); + if (ret < 0) { + dev_err(sdev->dev, "%s failed\n", __func__); + return -EINVAL; + } + + return 0; +} + +static int hda_sdw_probe(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + struct sdw_intel_res res; + acpi_handle handle; + void *sdw; + + handle = ACPI_HANDLE(sdev->dev); + + hdev = sdev->pdata->hw_pdata; + + memset(&res, 0, sizeof(res)); + + res.mmio_base = sdev->bar[HDA_DSP_BAR]; + res.irq = sdev->ipc_irq; + res.handle = hdev->info.handle; + res.parent = sdev->dev; + + /* + * ops and arg fields are not populated for now, + * they will be needed when the DAI callbacks are + * provided + */ + + /* we could filter links here if needed, e.g for quirks */ + res.count = hdev->info.count; + res.link_mask = hdev->info.link_mask; + + sdw = sdw_intel_probe(&res); + if (!sdw) { + dev_err(sdev->dev, "error: SoundWire probe failed\n"); + return -EINVAL; + } + + /* save context */ + hdev->sdw = sdw; + + return 0; +} + +int hda_sdw_startup(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + + hdev = sdev->pdata->hw_pdata; + + return sdw_intel_startup(hdev->sdw); +} + +static int hda_sdw_exit(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + + hdev = sdev->pdata->hw_pdata; + + hda_sdw_int_enable(sdev, false); + + if (hdev->sdw) + sdw_intel_exit(hdev->sdw); + hdev->sdw = NULL; + + return 0; +} +#endif + /* * Debug */ @@ -346,9 +440,12 @@ static const char *fixup_tplg_name(struct snd_sof_dev *sdev, static int hda_init_caps(struct snd_sof_dev *sdev) { struct hdac_bus *bus = sof_to_bus(sdev); + struct snd_sof_pdata *pdata = sdev->pdata; #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) struct hdac_ext_link *hlink; #endif + struct sof_intel_hda_dev *hdev = pdata->hw_pdata; + u32 link_mask; int ret = 0; device_disable_async_suspend(bus->dev); @@ -365,6 +462,27 @@ static int hda_init_caps(struct snd_sof_dev *sdev) return ret; } + /* scan SoundWire capabilities exposed by DSDT */ + ret = hda_sdw_acpi_scan(sdev); + if (ret < 0) { + dev_err(sdev->dev, "error: SoundWire ACPI scan error\n"); + return ret; + } + + link_mask = hdev->info.link_mask; + if (!link_mask) { + /* + * probe/allocated SoundWire resources. + * The hardware configuration takes place in hda_sdw_startup + * after power rails are enabled. + */ + ret = hda_sdw_probe(sdev); + if (ret < 0) { + dev_err(sdev->dev, "error: SoundWire probe error\n"); + return ret; + } + } + #if IS_ENABLED(CONFIG_SND_SOC_SOF_HDA) if (bus->mlcap) snd_hdac_ext_bus_get_ml_capabilities(bus); @@ -622,6 +740,8 @@ int hda_dsp_remove(struct snd_sof_dev *sdev) snd_hdac_ext_bus_device_remove(bus); #endif + hda_sdw_exit(sdev); + if (!IS_ERR_OR_NULL(hda->dmic_dev)) platform_device_unregister(hda->dmic_dev); diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 2a57fc9f3e58..928a3432e9e6 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -11,6 +11,8 @@ #ifndef __SOF_INTEL_HDA_H #define __SOF_INTEL_HDA_H +#include <linux/soundwire/sdw.h> +#include <linux/soundwire/sdw_intel.h> #include <sound/compress_driver.h> #include <sound/hda_codec.h> #include <sound/hdaudio_ext.h> @@ -436,6 +438,12 @@ struct sof_intel_hda_dev { /* delayed work to enter D0I3 opportunistically */ struct delayed_work d0i3_work; + + /* ACPI information stored between scan and probe steps */ + struct sdw_intel_acpi_info info; + + /* sdw context allocated by SoundWire driver */ + struct sdw_intel_ctx *sdw; }; static inline struct hdac_bus *sof_to_bus(struct snd_sof_dev *s) @@ -654,6 +662,42 @@ int hda_dsp_trace_init(struct snd_sof_dev *sdev, u32 *stream_tag); int hda_dsp_trace_release(struct snd_sof_dev *sdev); int hda_dsp_trace_trigger(struct snd_sof_dev *sdev, int cmd); +/* + * SoundWire support + */ +#if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) + +int hda_sdw_startup(struct snd_sof_dev *sdev); +void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable); + +#else + +static inline int hda_sdw_acpi_scan(struct snd_sof_dev *sdev) +{ + return 0; +} + +static inline int hda_sdw_probe(struct snd_sof_dev *sdev) +{ + return 0; +} + +static inline int hda_sdw_startup(struct snd_sof_dev *sdev) +{ + return 0; +} + +static inline int hda_sdw_exit(struct snd_sof_dev *sdev) +{ + return 0; +} + +static inline void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable) +{ +} + +#endif + /* common dai driver */ extern struct snd_soc_dai_driver skl_dai[]; -- 2.20.1

1 0

Applied "ASoC: SOF: IPC: dai-intel: move ALH declarations in header file" to the asoc tree
by Mark Brown 27 Mar '20

27 Mar '20

The patch ASoC: SOF: IPC: dai-intel: move ALH declarations in header file 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 f8e25018801548314b3d908315cbf271e8eceba8 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Wed, 25 Mar 2020 16:50:19 -0500 Subject: [PATCH] ASoC: SOF: IPC: dai-intel: move ALH declarations in header file ALH was inserted in the wrong place during integration, add after DMIC to mirror the file used by SOF firmware. No functional change, just text move in the same file to better track changes, if any. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200325215027.28716-4-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/sof/dai-intel.h | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/include/sound/sof/dai-intel.h b/include/sound/sof/dai-intel.h index 5f1ef5565be6..04e48227f542 100644 --- a/include/sound/sof/dai-intel.h +++ b/include/sound/sof/dai-intel.h @@ -87,6 +87,15 @@ struct sof_ipc_dai_hda_params { uint32_t link_dma_ch; } __packed; +/* ALH Configuration Request - SOF_IPC_DAI_ALH_CONFIG */ +struct sof_ipc_dai_alh_params { + struct sof_ipc_hdr hdr; + uint32_t stream_id; + + /* reserved for future use */ + uint32_t reserved[15]; +} __packed; + /* DMIC Configuration Request - SOF_IPC_DAI_DMIC_CONFIG */ /* This struct is defined per 2ch PDM controller available in the platform. @@ -179,13 +188,4 @@ struct sof_ipc_dai_dmic_params { struct sof_ipc_dai_dmic_pdm_ctrl pdm[0]; } __packed; -/* ALH Configuration Request - SOF_IPC_DAI_ALH_CONFIG */ -struct sof_ipc_dai_alh_params { - struct sof_ipc_hdr hdr; - uint32_t stream_id; - - /* reserved for future use */ - uint32_t reserved[15]; -} __packed; - #endif -- 2.20.1

1 0

Applied "ASoC: SOF: Intel: hda: add SoundWire stream config/free callbacks" to the asoc tree
by Mark Brown 27 Mar '20

27 Mar '20

The patch ASoC: SOF: Intel: hda: add SoundWire stream config/free callbacks 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 d2c383aa49dce9c5b59930c999099ecc580857fe Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Wed, 25 Mar 2020 16:50:20 -0500 Subject: [PATCH] ASoC: SOF: Intel: hda: add SoundWire stream config/free callbacks These callbacks are invoked when a matching hw_params/hw_free() DAI operation takes place, and will result in IPC operations with the SOF firmware. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200325215027.28716-5-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/intel/hda.c | 71 +++++++++++++++++++++++++++++++++++++++ 1 file changed, 71 insertions(+) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index c1fe94800da1..1f93124a63cd 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -24,6 +24,7 @@ #include <sound/intel-nhlt.h> #include <sound/sof.h> #include <sound/sof/xtensa.h> +#include "../sof-audio.h" #include "../ops.h" #include "hda.h" @@ -38,6 +39,74 @@ #if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) +static int sdw_params_stream(struct device *dev, + struct sdw_intel_stream_params_data *params_data) +{ + struct snd_sof_dev *sdev = dev_get_drvdata(dev); + struct snd_soc_dai *d = params_data->dai; + struct sof_ipc_dai_config config; + struct sof_ipc_reply reply; + int link_id = params_data->link_id; + int alh_stream_id = params_data->alh_stream_id; + int ret; + u32 size = sizeof(config); + + memset(&config, 0, size); + config.hdr.size = size; + config.hdr.cmd = SOF_IPC_GLB_DAI_MSG | SOF_IPC_DAI_CONFIG; + config.type = SOF_DAI_INTEL_ALH; + config.dai_index = (link_id << 8) | (d->id); + config.alh.stream_id = alh_stream_id; + + /* send message to DSP */ + ret = sof_ipc_tx_message(sdev->ipc, + config.hdr.cmd, &config, size, &reply, + sizeof(reply)); + if (ret < 0) { + dev_err(sdev->dev, + "error: failed to set DAI hw_params for link %d dai->id %d ALH %d\n", + link_id, d->id, alh_stream_id); + } + + return ret; +} + +static int sdw_free_stream(struct device *dev, + struct sdw_intel_stream_free_data *free_data) +{ + struct snd_sof_dev *sdev = dev_get_drvdata(dev); + struct snd_soc_dai *d = free_data->dai; + struct sof_ipc_dai_config config; + struct sof_ipc_reply reply; + int link_id = free_data->link_id; + int ret; + u32 size = sizeof(config); + + memset(&config, 0, size); + config.hdr.size = size; + config.hdr.cmd = SOF_IPC_GLB_DAI_MSG | SOF_IPC_DAI_CONFIG; + config.type = SOF_DAI_INTEL_ALH; + config.dai_index = (link_id << 8) | d->id; + config.alh.stream_id = 0xFFFF; /* invalid value on purpose */ + + /* send message to DSP */ + ret = sof_ipc_tx_message(sdev->ipc, + config.hdr.cmd, &config, size, &reply, + sizeof(reply)); + if (ret < 0) { + dev_err(sdev->dev, + "error: failed to free stream for link %d dai->id %d\n", + link_id, d->id); + } + + return ret; +} + +static const struct sdw_intel_ops sdw_callback = { + .params_stream = sdw_params_stream, + .free_stream = sdw_free_stream, +}; + void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable) { sdw_intel_enable_irq(sdev->bar[HDA_DSP_BAR], enable); @@ -80,6 +149,8 @@ static int hda_sdw_probe(struct snd_sof_dev *sdev) res.irq = sdev->ipc_irq; res.handle = hdev->info.handle; res.parent = sdev->dev; + res.ops = &sdw_callback; + res.dev = sdev->dev; /* * ops and arg fields are not populated for now, -- 2.20.1

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Applied "ASoC: SOF: Intel: hda: disable SoundWire interrupts on suspend" to the asoc tree
by Mark Brown 27 Mar '20

27 Mar '20

The patch ASoC: SOF: Intel: hda: disable SoundWire interrupts on suspend 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 3eadff5639b01c17f5f2ffeb209d05cc19706687 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Wed, 25 Mar 2020 16:50:22 -0500 Subject: [PATCH] ASoC: SOF: Intel: hda: disable SoundWire interrupts on suspend Doing this avoid conflicts and errors reported on the bus. The interrupts are only re-enabled on resume after the firmware is downloaded, so the behavior is not fully symmetric Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200325215027.28716-7-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/intel/hda-dsp.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/sof/intel/hda-dsp.c b/sound/soc/sof/intel/hda-dsp.c index c396b7ef0328..1aff90042694 100644 --- a/sound/soc/sof/intel/hda-dsp.c +++ b/sound/soc/sof/intel/hda-dsp.c @@ -594,6 +594,8 @@ static int hda_suspend(struct snd_sof_dev *sdev, bool runtime_suspend) #endif int ret; + hda_sdw_int_enable(sdev, false); + /* disable IPC interrupts */ hda_dsp_ipc_int_disable(sdev); -- 2.20.1

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Applied "ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire interrupt handlers" to the asoc tree
by Mark Brown 27 Mar '20

27 Mar '20

The patch ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire interrupt handlers 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 722ba5f1f530a919d28a0dd9e8e0ec63af18270d Mon Sep 17 00:00:00 2001 From: Bard Liao <yung-chuan.liao(a)linux.intel.com> Date: Wed, 25 Mar 2020 16:50:23 -0500 Subject: [PATCH] ASoC: SOF: Intel: hda: merge IPC, stream and SoundWire interrupt handlers We have a single irq handler for SOF interrupts. We can further merge SoundWire ones to completely remove MSI interrupts handling issues leading to timeouts. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> Link: https://lore.kernel.org/r/20200325215027.28716-8-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/intel/hda.c | 36 ++++++++++++++++++++++++++++++++++++ sound/soc/sof/intel/hda.h | 11 +++++++++++ 2 files changed, 47 insertions(+) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index dea3c385b664..ee4f1ceca883 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -198,6 +198,38 @@ static int hda_sdw_exit(struct snd_sof_dev *sdev) return 0; } + +static bool hda_dsp_check_sdw_irq(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + bool ret = false; + u32 irq_status; + + hdev = sdev->pdata->hw_pdata; + + if (!hdev->sdw) + return ret; + + /* store status */ + irq_status = snd_sof_dsp_read(sdev, HDA_DSP_BAR, HDA_DSP_REG_ADSPIS2); + + /* invalid message ? */ + if (irq_status == 0xffffffff) + goto out; + + /* SDW message ? */ + if (irq_status & HDA_DSP_REG_ADSPIS2_SNDW) + ret = true; + +out: + return ret; +} + +static irqreturn_t hda_dsp_sdw_thread(int irq, void *context) +{ + return sdw_intel_thread(irq, context); +} + #endif /* @@ -618,6 +650,7 @@ static irqreturn_t hda_dsp_interrupt_handler(int irq, void *context) static irqreturn_t hda_dsp_interrupt_thread(int irq, void *context) { struct snd_sof_dev *sdev = context; + struct sof_intel_hda_dev *hdev = sdev->pdata->hw_pdata; /* deal with streams and controller first */ if (hda_dsp_check_stream_irq(sdev)) @@ -626,6 +659,9 @@ static irqreturn_t hda_dsp_interrupt_thread(int irq, void *context) if (hda_dsp_check_ipc_irq(sdev)) sof_ops(sdev)->irq_thread(irq, sdev); + if (hda_dsp_check_sdw_irq(sdev)) + hda_dsp_sdw_thread(irq, hdev->sdw); + /* enable GIE interrupt */ snd_sof_dsp_update_bits(sdev, HDA_DSP_HDA_BAR, SOF_HDA_INTCTL, diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index 928a3432e9e6..fc104c5ba006 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -232,6 +232,8 @@ #define HDA_DSP_REG_ADSPIC2 (HDA_DSP_GEN_BASE + 0x10) #define HDA_DSP_REG_ADSPIS2 (HDA_DSP_GEN_BASE + 0x14) +#define HDA_DSP_REG_ADSPIS2_SNDW BIT(5) + /* Intel HD Audio Inter-Processor Communication Registers */ #define HDA_DSP_IPC_BASE 0x40 #define HDA_DSP_REG_HIPCT (HDA_DSP_IPC_BASE + 0x00) @@ -696,6 +698,15 @@ static inline void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable) { } +static inline bool hda_dsp_check_sdw_irq(struct snd_sof_dev *sdev) +{ + return false; +} + +static inline irqreturn_t hda_dsp_sdw_thread(int irq, void *context) +{ + return IRQ_HANDLED; +} #endif /* common dai driver */ -- 2.20.1

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Applied "ASoC: SOF: Intel: hda: add parameter to control SoundWire clock stop quirks" to the asoc tree
by Mark Brown 27 Mar '20

27 Mar '20

The patch ASoC: SOF: Intel: hda: add parameter to control SoundWire clock stop quirks 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 02df8f4364b070428af0e5b6c8739c884c8ad4e7 Mon Sep 17 00:00:00 2001 From: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Date: Wed, 25 Mar 2020 16:50:24 -0500 Subject: [PATCH] ASoC: SOF: Intel: hda: add parameter to control SoundWire clock stop quirks Add module parameter so that the different modes can be quickly tested. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Link: https://lore.kernel.org/r/20200325215027.28716-9-pierre-louis.bossart@linux… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/intel/hda.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index ee4f1ceca883..1e69cfcee8e0 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -40,6 +40,16 @@ #if IS_ENABLED(CONFIG_SND_SOC_SOF_INTEL_SOUNDWIRE) +/* + * The default for SoundWire clock stop quirks is to power gate the IP + * and do a Bus Reset, this will need to be modified when the DSP + * needs to remain in D0i3 so that the Master does not lose context + * and enumeration is not required on clock restart + */ +static int sdw_clock_stop_quirks = SDW_INTEL_CLK_STOP_BUS_RESET; +module_param(sdw_clock_stop_quirks, int, 0444); +MODULE_PARM_DESC(sdw_clock_stop_quirks, "SOF SoundWire clock stop quirks"); + static int sdw_params_stream(struct device *dev, struct sdw_intel_stream_params_data *params_data) { @@ -149,6 +159,7 @@ static int hda_sdw_probe(struct snd_sof_dev *sdev) res.parent = sdev->dev; res.ops = &sdw_callback; res.dev = sdev->dev; + res.clock_stop_quirks = sdw_clock_stop_quirks; /* * ops and arg fields are not populated for now, -- 2.20.1

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Applied "ASoC: SOF: Intel: hda: add WAKEEN interrupt support for SoundWire" to the asoc tree
by Mark Brown 27 Mar '20

27 Mar '20

The patch ASoC: SOF: Intel: hda: add WAKEEN interrupt support for SoundWire 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 bbd19cdca8279cf244a301c6a13ae5ec9e4ef976 Mon Sep 17 00:00:00 2001 From: Rander Wang <rander.wang(a)intel.com> Date: Wed, 25 Mar 2020 16:50:25 -0500 Subject: [PATCH] ASoC: SOF: Intel: hda: add WAKEEN interrupt support for SoundWire When a SoundWire link is in clock stop state, a Slave device may wake up the Master for some events such as jack detection. The WAKEEN interrupt will be triggered and processed by the audio pci device. If audio device is in D3, the interrupt will be routed to PME, or aggregated at cAVS level as interrupt when audio device is in D0. This patch only supports D3 case, where the audio pci device will be resumed by a PME event and the WAKEEN interrupt will be processed after audio pci device is powered up and ROM is initialized successfully. The WAKEEN handling is only enabled after the first boot due to dependencies on a shim_lock mutex being initialized. Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Rander Wang <rander.wang(a)intel.com> Link: https://lore.kernel.org/r/20200325215027.28716-10-pierre-louis.bossart@linu… Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/sof/intel/hda-loader.c | 18 ++++++++++++++++++ sound/soc/sof/intel/hda.c | 11 +++++++++++ sound/soc/sof/intel/hda.h | 5 +++++ 3 files changed, 34 insertions(+) diff --git a/sound/soc/sof/intel/hda-loader.c b/sound/soc/sof/intel/hda-loader.c index 2ae94ea53122..e1550ccd0a49 100644 --- a/sound/soc/sof/intel/hda-loader.c +++ b/sound/soc/sof/intel/hda-loader.c @@ -346,6 +346,24 @@ int hda_dsp_cl_boot_firmware(struct snd_sof_dev *sdev) goto cleanup; } + /* + * When a SoundWire link is in clock stop state, a Slave + * device may trigger in-band wakes for events such as jack + * insertion or acoustic event detection. This event will lead + * to a WAKEEN interrupt, handled by the PCI device and routed + * to PME if the PCI device is in D3. The resume function in + * audio PCI driver will be invoked by ACPI for PME event and + * initialize the device and process WAKEEN interrupt. + * + * The WAKEEN interrupt should be processed ASAP to prevent an + * interrupt flood, otherwise other interrupts, such IPC, + * cannot work normally. The WAKEEN is handled after the ROM + * is initialized successfully, which ensures power rails are + * enabled before accessing the SoundWire SHIM registers + */ + if (!sdev->first_boot) + hda_sdw_process_wakeen(sdev); + /* * at this point DSP ROM has been initialized and * should be ready for code loading and firmware boot diff --git a/sound/soc/sof/intel/hda.c b/sound/soc/sof/intel/hda.c index 1e69cfcee8e0..7d1aa4c7d82c 100644 --- a/sound/soc/sof/intel/hda.c +++ b/sound/soc/sof/intel/hda.c @@ -241,6 +241,17 @@ static irqreturn_t hda_dsp_sdw_thread(int irq, void *context) return sdw_intel_thread(irq, context); } +void hda_sdw_process_wakeen(struct snd_sof_dev *sdev) +{ + struct sof_intel_hda_dev *hdev; + + hdev = sdev->pdata->hw_pdata; + if (!hdev->sdw) + return; + + sdw_intel_process_wakeen_event(hdev->sdw); +} + #endif /* diff --git a/sound/soc/sof/intel/hda.h b/sound/soc/sof/intel/hda.h index fc104c5ba006..6f1765b1ed1d 100644 --- a/sound/soc/sof/intel/hda.h +++ b/sound/soc/sof/intel/hda.h @@ -671,6 +671,7 @@ int hda_dsp_trace_trigger(struct snd_sof_dev *sdev, int cmd); int hda_sdw_startup(struct snd_sof_dev *sdev); void hda_sdw_int_enable(struct snd_sof_dev *sdev, bool enable); +void hda_sdw_process_wakeen(struct snd_sof_dev *sdev); #else @@ -707,6 +708,10 @@ static inline irqreturn_t hda_dsp_sdw_thread(int irq, void *context) { return IRQ_HANDLED; } + +static inline void hda_sdw_process_wakeen(struct snd_sof_dev *sdev) +{ +} #endif /* common dai driver */ -- 2.20.1

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