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

[alsa-devel] [PATCH v2] ASoC: Intel: Baytrail: Fix implicit fallthrough warning
by Cezary Rojewski 30 Aug '19

30 Aug '19

Append fallthrough statement to fix warning reported during compilation. Fixes: b80d19c166c4 ("ASoC: Intel: Restore Baytrail ADSP streams only when ADSP was in reset") Signed-off-by: Cezary Rojewski <cezary.rojewski(a)intel.com> --- Changes from v1: - made use of fallthrough statement as suggested by Pierre - updated commit message sound/soc/intel/baytrail/sst-baytrail-pcm.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/baytrail/sst-baytrail-pcm.c b/sound/soc/intel/baytrail/sst-baytrail-pcm.c index 9cbc982d46a9..54f2ee3010ee 100644 --- a/sound/soc/intel/baytrail/sst-baytrail-pcm.c +++ b/sound/soc/intel/baytrail/sst-baytrail-pcm.c @@ -193,6 +193,7 @@ static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd) break; case SNDRV_PCM_TRIGGER_SUSPEND: pdata->restore_stream = false; + /* fallthrough */ case SNDRV_PCM_TRIGGER_PAUSE_PUSH: sst_byt_stream_pause(byt, pcm_data->stream); break; -- 2.17.1

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[alsa-devel] [PATCH] ASoC: fsl_ssi: Fix clock control issue in master mode
by Shengjiu Wang 30 Aug '19

30 Aug '19

The test case is arecord -Dhw:0 -d 10 -f S16_LE -r 48000 -c 2 temp.wav & aplay -Dhw:0 -d 30 -f S16_LE -r 48000 -c 2 test.wav There will be error after end of arecord: aplay: pcm_write:2051: write error: Input/output error Capture and Playback work in parallel in master mode, one substream stops, the other substream is impacted, the reason is that clock is disabled wrongly. The clock's reference count is not increased when second substream starts, the hw_param() function returns in the beginning because first substream is enabled, then in end of first substream, the hw_free() disables the clock. This patch is to move the clock enablement to the place before checking of the device enablement in hw_param(). Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- sound/soc/fsl/fsl_ssi.c | 18 +++++++++--------- 1 file changed, 9 insertions(+), 9 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index b0a6fead1a6a..537dc69256f0 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -799,15 +799,6 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, u32 wl = SSI_SxCCR_WL(sample_size); int ret; - /* - * SSI is properly configured if it is enabled and running in - * the synchronous mode; Note that AC97 mode is an exception - * that should set separate configurations for STCCR and SRCCR - * despite running in the synchronous mode. - */ - if (ssi->streams && ssi->synchronous) - return 0; - if (fsl_ssi_is_i2s_master(ssi)) { ret = fsl_ssi_set_bclk(substream, dai, hw_params); if (ret) @@ -823,6 +814,15 @@ static int fsl_ssi_hw_params(struct snd_pcm_substream *substream, } } + /* + * SSI is properly configured if it is enabled and running in + * the synchronous mode; Note that AC97 mode is an exception + * that should set separate configurations for STCCR and SRCCR + * despite running in the synchronous mode. + */ + if (ssi->streams && ssi->synchronous) + return 0; + if (!fsl_ssi_is_ac97(ssi)) { /* * Keep the ssi->i2s_net intact while having a local variable -- 2.21.0

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[alsa-devel] [PATCH] ASoC: SOF/Intel: fix selection of SND_INTEL_NHTL
by Pierre-Louis Bossart 30 Aug '19

30 Aug '19

We should only select SND_INTEL_NHLT when ACPI is defined. This was done for the legacy HDAudio driver but not for DSP-enabled cases, leading to compilation errors with randconfig. Fix by aligning on the same solution. For the Skylake driver this is overkill since there is a top-level dependency on ACPI, but it doesn't hurt and it's better to have consistency. Fixes: 68b953aeb50d9 ('ASoC: SOF: Intel: hda: fixup HDaudio topology name with DMIC number') Reported-by: Randy Dunlap <rdunlap(a)infradead.org> Signed-off-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> --- sound/soc/intel/Kconfig | 2 +- sound/soc/sof/intel/Kconfig | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index a3ec17fd63cd..01c99750212a 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -215,7 +215,7 @@ config SND_SOC_INTEL_SKYLAKE_COMMON select SND_SOC_INTEL_SST select SND_SOC_HDAC_HDA if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC select SND_SOC_ACPI_INTEL_MATCH - select SND_INTEL_NHLT + select SND_INTEL_NHLT if ACPI help If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/ GeminiLake or CannonLake platform with the DSP enabled in the BIOS diff --git a/sound/soc/sof/intel/Kconfig b/sound/soc/sof/intel/Kconfig index 889b6202d054..479ba249e219 100644 --- a/sound/soc/sof/intel/Kconfig +++ b/sound/soc/sof/intel/Kconfig @@ -286,7 +286,7 @@ config SND_SOC_SOF_HDA tristate select SND_HDA_EXT_CORE if SND_SOC_SOF_HDA_LINK select SND_SOC_HDAC_HDA if SND_SOC_SOF_HDA_AUDIO_CODEC - select SND_INTEL_NHLT + select SND_INTEL_NHLT if ACPI help This option is not user-selectable but automagically handled by 'select' statements at a higher level -- 2.20.1

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[alsa-devel] [PATCH 0/4] ASoC: Intel: Haswell: Adjust machine device private
by Cezary Rojewski 30 Aug '19

30 Aug '19

Apart from Haswell machines, all other devices have their private data set to snd_soc_acpi_mach instance. Changes for HSW/ BDW boards introduced with series: https://patchwork.kernel.org/cover/10782035/ added support for dai_link platform_name adjustments within card probe routines. These take for granted private_data points to snd_soc_acpi_mach whereas for Haswell, it's sst_pdata instead. Change private context of platform_device - representing machine board - to address this. Caught by recent cleanups where content of sst_pdata was moved. Currently, despite the incorrect cast, dereferenced field points happily to NULL (uninitialized field), so no panics were observed. Cezary Rojewski (4): ASoC: Intel: Haswell: Adjust machine device private context ASoC: Intel: haswell: Simplify device probe ASoC: Intel: bdw-rt5677: Simplify device probe ASoC: Intel: broadwell: Simplify device probe sound/soc/intel/boards/bdw-rt5677.c | 6 +----- sound/soc/intel/boards/broadwell.c | 6 +----- sound/soc/intel/boards/haswell.c | 6 +----- sound/soc/intel/common/sst-acpi.c | 3 ++- 4 files changed, 5 insertions(+), 16 deletions(-) -- 2.17.1

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[alsa-devel] [PATCH] ASoC: rsnd: fixup 6ch settings to 8ch
by Kuninori Morimoto 30 Aug '19

30 Aug '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> rsnd need to use 8ch clock settings for 6ch for TDM. Otherwise, it can't work correctly. This patch fixup it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 18 +++++++++++++----- sound/soc/sh/rcar/rsnd.h | 1 + sound/soc/sh/rcar/ssi.c | 2 ++ 3 files changed, 16 insertions(+), 5 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 4fe83e61..37cb615 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -300,6 +300,18 @@ int rsnd_runtime_channel_after_ctu_with_params(struct rsnd_dai_stream *io, return chan; } +int rsnd_channel_normalization(int chan) +{ + if ((chan > 8) || (chan < 0)) + return 0; + + /* TDM Extend Mode needs 8ch */ + if (chan == 6) + chan = 8; + + return chan; +} + int rsnd_runtime_channel_for_ssi_with_params(struct rsnd_dai_stream *io, struct snd_pcm_hw_params *params) { @@ -312,11 +324,7 @@ int rsnd_runtime_channel_for_ssi_with_params(struct rsnd_dai_stream *io, if (rsnd_runtime_is_multi_ssi(io)) chan /= rsnd_rdai_ssi_lane_get(rdai); - /* TDM Extend Mode needs 8ch */ - if (chan == 6) - chan = 8; - - return chan; + return rsnd_channel_normalization(chan); } int rsnd_runtime_is_multi_ssi(struct rsnd_dai_stream *io) diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 0e6ef4e1..7727add 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -446,6 +446,7 @@ void rsnd_parse_connect_common(struct rsnd_dai *rdai, struct device_node *playback, struct device_node *capture); +int rsnd_channel_normalization(int chan); #define rsnd_runtime_channel_original(io) \ rsnd_runtime_channel_original_with_params(io, NULL) int rsnd_runtime_channel_original_with_params(struct rsnd_dai_stream *io, diff --git a/sound/soc/sh/rcar/ssi.c b/sound/soc/sh/rcar/ssi.c index f5afab6..44bda21 100644 --- a/sound/soc/sh/rcar/ssi.c +++ b/sound/soc/sh/rcar/ssi.c @@ -303,6 +303,8 @@ static int rsnd_ssi_master_clk_start(struct rsnd_mod *mod, if (rsnd_runtime_is_tdm_split(io)) chan = rsnd_io_converted_chan(io); + chan = rsnd_channel_normalization(chan); + main_rate = rsnd_ssi_clk_query(rdai, rate, chan, &idx); if (!main_rate) { dev_err(dev, "unsupported clock rate\n"); -- 2.7.4

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[alsa-devel] [PATCH] ALSA: hda/realtek - Fix overridden device-specific initialization
by Takashi Iwai 30 Aug '19

30 Aug '19

The recent change to shuffle the codec initialization procedure for Realtek via commit 607ca3bd220f ("ALSA: hda/realtek - EAPD turn on later") caused the silent output on some machines. This change was supposed to be safe, but it isn't actually; some devices have quirk setups to override the EAPD via COEF or BTL in the additional verb table, which is applied at the beginning of snd_hda_gen_init(). And this EAPD setup is again overridden in alc_auto_init_amp(). For recovering from the regression, tell snd_hda_gen_init() not to apply the verbs there by a new flag, then apply the verbs in alc_init(). BugLink: https://bugzilla.kernel.org/show_bug.cgi?id=204727 Fixes: 607ca3bd220f ("ALSA: hda/realtek - EAPD turn on later") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_generic.c | 3 ++- sound/pci/hda/hda_generic.h | 1 + sound/pci/hda/patch_realtek.c | 2 ++ 3 files changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 5bf24fb819d2..10d502328b76 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -6009,7 +6009,8 @@ int snd_hda_gen_init(struct hda_codec *codec) if (spec->init_hook) spec->init_hook(codec); - snd_hda_apply_verbs(codec); + if (!spec->skip_verbs) + snd_hda_apply_verbs(codec); init_multi_out(codec); init_extra_out(codec); diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 5f199dcb0d18..fb9f1a90238b 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -243,6 +243,7 @@ struct hda_gen_spec { unsigned int indep_hp_enabled:1; /* independent HP enabled */ unsigned int have_aamix_ctl:1; unsigned int hp_mic_jack_modes:1; + unsigned int skip_verbs:1; /* don't apply verbs at snd_hda_gen_init() */ /* additional mute flags (only effective with auto_mute_via_amp=1) */ u64 mute_bits; diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index e333b3e30e31..03f903f9247d 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -837,9 +837,11 @@ static int alc_init(struct hda_codec *codec) if (spec->init_hook) spec->init_hook(codec); + spec->gen.skip_verbs = 1; /* applied in below */ snd_hda_gen_init(codec); alc_fix_pll(codec); alc_auto_init_amp(codec, spec->init_amp); + snd_hda_apply_verbs(codec); /* apply verbs here after own init */ snd_hda_apply_fixup(codec, HDA_FIXUP_ACT_INIT); -- 2.16.4

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[alsa-devel] [PATCH] [PATCH v2] ALSA: hda: Retry codec read on first time failure
by vani.ganji＠intel.com 30 Aug '19

30 Aug '19

From: Vani Ganji <vani.ganji(a)intel.com> Sometimes HDMI audio fails while fast switching between HDMI and Speaker due to codec Get Power state verb read failure. Verb is codec driver to access codec HW. Since issue scenario involves fast switching between HDMI and speaker, there might be chance to fail read from HW and same happened in the error scenario. Based on Display/Graphics architect suggestion, added a retry mechanism for failure on first attempt. Changes from v1: -Corrected missing braces. Signed-off-by: Vani Ganji <vani.ganji(a)intel.com> --- sound/hda/hdac_device.c | 15 +++++++++++---- 1 file changed, 11 insertions(+), 4 deletions(-) diff --git a/sound/hda/hdac_device.c b/sound/hda/hdac_device.c index b26cc93e7e10..3874d54aa686 100644 --- a/sound/hda/hdac_device.c +++ b/sound/hda/hdac_device.c @@ -984,10 +984,17 @@ static unsigned int codec_read(struct hdac_device *hdac, hda_nid_t nid, { unsigned int cmd = snd_hdac_make_cmd(hdac, nid, verb, parm); unsigned int res; - - if (snd_hdac_exec_verb(hdac, cmd, flags, &res)) - return -1; - + unsigned int count; + + /* Retry verb once more if it fails first time */ + /* recommended to retry since it is HW related */ + for (count = 0; count <= 1; count++) { + if (snd_hdac_exec_verb(hdac, cmd, flags, &res)) { + if (count == 1) + return -1; + } else + break; + } return res; } -- 2.17.1

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[alsa-devel] [PATCH] ALSA: firewire-motu: add support for MOTU 4pre
by Takashi Sakamoto 30 Aug '19

30 Aug '19

MOTU 4pre was launched in 2012 by MOTU, Inc. This commit allows userspace applications can transmit and receive PCM frames and MIDI messages for this model via ALSA PCM interface and RawMidi/Sequencer interfaces. The device supports MOTU protocol version 3. Unlike the other devices, the device is simply designed. The size of data block is fixed to 10 quadlets during available sampling rates (44.1 - 96.0 kHz). Each data block includes 1 source packet header, 2 data chunks for messages, 8 data chunks for PCM samples and 2 data chunks for padding to quadlet alignment. The device has no MIDI, optical, BNC and AES/EBU interfaces. Like support for the other MOTU devices, the quality of playback sound is not enough good with periodical noise yet. $ python2 crpp < ~/git/am-config-rom/motu/motu-4pre.img ROM header and bus information block ----------------------------------------------------------------- 400 041078cc bus_info_length 4, crc_length 16, crc 30924 404 31333934 bus_name "1394" 408 20ff7000 irmc 0, cmc 0, isc 1, bmc 0, cyc_clk_acc 255, max_rec 7 (256) 40c 0001f200 company_id 0001f2 | 410 000a41c5 device_id 00000a41c5 | EUI-64 0001f200000a41c5 root directory ----------------------------------------------------------------- 414 0004ef04 directory_length 4, crc 61188 418 030001f2 vendor 41c 0c0083c0 node capabilities per IEEE 1394 420 d1000002 --> unit directory at 428 424 8d000005 --> eui-64 leaf at 438 unit directory at 428 ----------------------------------------------------------------- 428 0003ceda directory_length 3, crc 52954 42c 120001f2 specifier id 430 13000045 version 434 17103800 model eui-64 leaf at 438 ----------------------------------------------------------------- 438 0002d248 leaf_length 2, crc 53832 43c 0001f200 company_id 0001f2 | 440 000a41c5 device_id 00000a41c5 | EUI-64 0001f200000a41c5 Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/motu/motu.c | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/sound/firewire/motu/motu.c b/sound/firewire/motu/motu.c index 03cda2166ea3..72908b4de77c 100644 --- a/sound/firewire/motu/motu.c +++ b/sound/firewire/motu/motu.c @@ -247,6 +247,17 @@ static const struct snd_motu_spec motu_audio_express = { .analog_out_ports = 4, }; +static const struct snd_motu_spec motu_4pre = { + .name = "4pre", + .protocol = &snd_motu_protocol_v3, + .flags = SND_MOTU_SPEC_SUPPORT_CLOCK_X2 | + SND_MOTU_SPEC_TX_MICINST_CHUNK | + SND_MOTU_SPEC_TX_RETURN_CHUNK | + SND_MOTU_SPEC_RX_SEPARETED_MAIN, + .analog_in_ports = 2, + .analog_out_ports = 2, +}; + #define SND_MOTU_DEV_ENTRY(model, data) \ { \ .match_flags = IEEE1394_MATCH_VENDOR_ID | \ @@ -265,6 +276,7 @@ static const struct ieee1394_device_id motu_id_table[] = { SND_MOTU_DEV_ENTRY(0x000015, &motu_828mk3), /* FireWire only. */ SND_MOTU_DEV_ENTRY(0x000035, &motu_828mk3), /* Hybrid. */ SND_MOTU_DEV_ENTRY(0x000033, &motu_audio_express), + SND_MOTU_DEV_ENTRY(0x000045, &motu_4pre), { } }; MODULE_DEVICE_TABLE(ieee1394, motu_id_table); -- 2.20.1

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[alsa-devel] [PATCH v5 0/4] ASoC: codecs: Add WSA881x Smart Speaker amplifier support
by Srinivas Kandagatla 29 Aug '19

29 Aug '19

Thanks for reviewing v4 patchset, here is v5 with addressing the comments in v4 This patchset adds support to WSA8810/WSA8815 Class-D Smart Speaker Amplifier which is SoundWire interfaced. This also adds support to some missing bits in SoundWire bus layer like Device Tree support. This patchset along with DB845c machine driver and WCD934x codec driver has been tested on SDM845 SoC based DragonBoard DB845c with two WSA8810 speakers. Most of the code in this driver is rework of Qualcomm downstream drivers used in Andriod. Credits to Banajit Goswami and Patrick Lai's Team. TODO: Add thermal sensor support in WSA881x. Thanks, srini Changes since v4: - updated slave bindings according to Rob's and Pierre Suggestion. - fixup warnings from make dt_binding_check Srinivas Kandagatla (4): dt-bindings: soundwire: add slave bindings soundwire: core: add device tree support for slave devices dt-bindings: ASoC: Add WSA881x bindings ASoC: codecs: add wsa881x amplifier support .../bindings/sound/qcom,wsa881x.yaml | 41 + .../soundwire/soundwire-controller.yaml | 72 ++ drivers/soundwire/bus.c | 2 + drivers/soundwire/bus.h | 1 + drivers/soundwire/slave.c | 52 + sound/soc/codecs/Kconfig | 10 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/wsa881x.c | 1134 +++++++++++++++++ 8 files changed, 1314 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/qcom,wsa881x.yaml create mode 100644 Documentation/devicetree/bindings/soundwire/soundwire-controller.yaml create mode 100644 sound/soc/codecs/wsa881x.c -- 2.21.0

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[alsa-devel] [ANNOUNCE] 2019 Linux Audio miniconference
by Mark Brown 29 Aug '19

29 Aug '19

Hi, As in previous years we're going to have an audio miniconference so we can get together and talk through issues, especially design decisions, face to face. This year's event will be held on October 31st in Lyon, France, the day after ELC-E. This will be held at the Lyon Convention Center (the ELC-E venue), generously sponsored by Intel. As with previous years let's pull together an agenda through a mailing list discussion - if people could reply to this mail with any topics they'd like to discuss we can take it from there. Of course if we can sort things out more quickly via the mailing list that's even better! If you're planning to attend please fill out the form here: https://docs.google.com/forms/d/e/1FAIpQLSd6FV-tWYZ1fHlCmyzQhG98OtYH9W9GX-1… This event will be covered by the same code of conduct as ELC-E. Thanks again to Intel for supporting this event. Thanks, Mark

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