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

[alsa-devel] snd_usb_audio - ALSA woke us up to write...was actually nothing to write
by Nathan Royce 03 Sep '19

03 Sep '19

***** ... Sep 03 14:32:58 username kernel: perf: interrupt took too long (4174 > 4156), lowering kernel.perf_event_max_sample_rate to 47700 Sep 03 14:33:03 username plasmashell[1057]: qml: temp unit: 0 Sep 03 14:33:14 username org_kde_powerdevil[1089]: powerdevil: Releasing inhibition with cookie 14 Sep 03 14:33:14 username org_kde_powerdevil[1089]: powerdevil: Releasing inhibition with cookie 15 Sep 03 14:33:14 username org_kde_powerdevil[1089]: powerdevil: Restoring DPMS features after inhibition release Sep 03 14:33:14 username org_kde_powerdevil[1089]: powerdevil: Can't contact ck Sep 03 14:33:14 username org_kde_powerdevil[1089]: powerdevil: Releasing inhibition with cookie 16 Sep 03 14:33:14 username org_kde_powerdevil[1089]: powerdevil: Can't contact ck Sep 03 14:33:15 username plasmashell[1057]: qml: temp unit: 0 ... Sep 03 14:33:23 username pulseaudio[1081]: E: [alsa-sink-USB Audio] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write. Sep 03 14:33:23 username pulseaudio[1081]: E: [alsa-sink-USB Audio] alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_usb_audio'. Please report this issue to the ALSA developers. Sep 03 14:33:23 username pulseaudio[1081]: E: [alsa-sink-USB Audio] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. Sep 03 14:33:24 username plasmashell[1057]: qml: temp unit: 0 ... ***** During this time period, I do believe I was starting an android emulator and also watching a twitter video that had audio.

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[alsa-devel] [PATCH] ASoC: soc-core: add comment to jack at soc_remove_component()
by Kuninori Morimoto 03 Sep '19

03 Sep '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Basically, driver which setups snd_soc_component_set_jack() need to release it by themselves. But, as framework level robustness, soc_remove_component() also releases it. To avoid code reader confuse, this patch makes it clarify. This patch makes it clarify. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 659b90a..60f7e8b 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1171,7 +1171,9 @@ static void soc_remove_component(struct snd_soc_component *component) if (component->probed) snd_soc_component_remove(component); + /* For framework level robustness */ snd_soc_component_set_jack(component, NULL, NULL); + list_del(&component->card_list); snd_soc_dapm_free(snd_soc_component_get_dapm(component)); soc_cleanup_component_debugfs(component); -- 2.7.4

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[alsa-devel] sound/usb kernel modules as a DKMS package.
by Hin-Tak Leung 03 Sep '19

03 Sep '19

Hi all, I (We... this is kind of my 9-5 job) have been having some reliability issue with a Focusrite Scarlett Gen 2 18i20, and I was almost tempted to start looking at the windows driver, when I came upon Geoffrey's work. So I made the bleeding-edge sound/usb modules (for the 5.4 time frame) from Takashi's tree built as dkms modules against main line 5.2 . I hope this is useful for other people using USB sound devices. https://github.com/HinTak/sound-usb-dkms This make it easier / faster to try out new code, without building the whole kernel. I would appreciate tips and tricks about the Scarlett 18i20, as well as general recommendation for new professional gear of the same sort to buy. Incidentally, Tobias Hoffmann (who wrote the first gen driver about 5 years ago) was my GSoC student working on printer-related matters more than a decade ago; I did some wifi/USB related work in the kernel (am listed in linux/MAINTAINERS) so you can assume that I am relatively happy with driver-related matters. Cheers, Hin-Tak

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[alsa-devel] [PATCH 1/2] conf/ucm: Add UCM profile for cht-bsw-rt5672 boards
by Hans de Goede 03 Sep '19

03 Sep '19

Add an UCM profile for Intel boards with a RT5672 codec. Re-use the existing platform enable and disable sequences for BYT/CHT SST support and add a codecs/rt5672 dir with codec specific enable / disable sequences for the various inputs and outputs. This is partly based on earlier work done here: https://github.com/plbossart/UCM/tree/master/cht-bsw-rt5672 Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- configure.ac | 2 + src/conf/ucm/Makefile.am | 1 + src/conf/ucm/cht-bsw-rt5672/HiFi.conf | 27 +++++++++++ src/conf/ucm/cht-bsw-rt5672/Makefile.am | 4 ++ .../ucm/cht-bsw-rt5672/cht-bsw-rt5672.conf | 6 +++ src/conf/ucm/codecs/Makefile.am | 1 + src/conf/ucm/codecs/rt5672/DMIC1.conf | 28 +++++++++++ src/conf/ucm/codecs/rt5672/DMIC2.conf | 28 +++++++++++ src/conf/ucm/codecs/rt5672/EnableSeq.conf | 48 +++++++++++++++++++ src/conf/ucm/codecs/rt5672/HeadPhones.conf | 23 +++++++++ src/conf/ucm/codecs/rt5672/HeadsetMic.conf | 38 +++++++++++++++ src/conf/ucm/codecs/rt5672/Makefile.am | 5 ++ src/conf/ucm/codecs/rt5672/MonoSpeaker.conf | 26 ++++++++++ src/conf/ucm/codecs/rt5672/Speaker.conf | 22 +++++++++ 14 files changed, 259 insertions(+) create mode 100644 src/conf/ucm/cht-bsw-rt5672/HiFi.conf create mode 100644 src/conf/ucm/cht-bsw-rt5672/Makefile.am create mode 100644 src/conf/ucm/cht-bsw-rt5672/cht-bsw-rt5672.conf create mode 100644 src/conf/ucm/codecs/rt5672/DMIC1.conf create mode 100644 src/conf/ucm/codecs/rt5672/DMIC2.conf create mode 100644 src/conf/ucm/codecs/rt5672/EnableSeq.conf create mode 100644 src/conf/ucm/codecs/rt5672/HeadPhones.conf create mode 100644 src/conf/ucm/codecs/rt5672/HeadsetMic.conf create mode 100644 src/conf/ucm/codecs/rt5672/Makefile.am create mode 100644 src/conf/ucm/codecs/rt5672/MonoSpeaker.conf create mode 100644 src/conf/ucm/codecs/rt5672/Speaker.conf diff --git a/configure.ac b/configure.ac index 2e955760..9ccca7aa 100644 --- a/configure.ac +++ b/configure.ac @@ -741,6 +741,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/bytcr-rt5651-stereo-spk-in1-mic/Makefile \ src/conf/ucm/bytcr-rt5651-stereo-spk-in2-mic/Makefile \ src/conf/ucm/bytcr-rt5651-stereo-spk-in12-mic/Makefile \ + src/conf/ucm/cht-bsw-rt5672/Makefile \ src/conf/ucm/chtnau8824/Makefile \ src/conf/ucm/chtrt5645/Makefile \ src/conf/ucm/chtrt5645-mono-speaker-analog-mic/Makefile \ @@ -772,6 +773,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/codecs/rt5640/Makefile \ src/conf/ucm/codecs/rt5645/Makefile \ src/conf/ucm/codecs/rt5651/Makefile \ + src/conf/ucm/codecs/rt5672/Makefile \ src/conf/ucm/platforms/Makefile \ src/conf/ucm/platforms/bytcr/Makefile \ src/conf/topology/Makefile \ diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am index 02257048..2ed4e1a3 100644 --- a/src/conf/ucm/Makefile.am +++ b/src/conf/ucm/Makefile.am @@ -25,6 +25,7 @@ bytcr-rt5651-stereo-spk-dmic-mic \ bytcr-rt5651-stereo-spk-in1-mic \ bytcr-rt5651-stereo-spk-in2-mic \ bytcr-rt5651-stereo-spk-in12-mic \ +cht-bsw-rt5672 \ chtnau8824 \ chtrt5645 \ chtrt5645-mono-speaker-analog-mic \ diff --git a/src/conf/ucm/cht-bsw-rt5672/HiFi.conf b/src/conf/ucm/cht-bsw-rt5672/HiFi.conf new file mode 100644 index 00000000..b83099d8 --- /dev/null +++ b/src/conf/ucm/cht-bsw-rt5672/HiFi.conf @@ -0,0 +1,27 @@ +# Adapted from https://github.com/plbossart/UCM/tree/master/cht-bsw-rt5672 + +SectionVerb { + EnableSequence [ + cdev "hw:chtbswrt5672" + <platforms/bytcr/PlatformEnableSeq.conf> + <codecs/rt5672/EnableSeq.conf> + ] + + DisableSequence [ + cdev "hw:chtbswrt5672" + <platforms/bytcr/PlatformDisableSeq.conf> + ] + + Value { + PlaybackPCM "hw:chtbswrt5672" + CapturePCM "hw:chtbswrt5672" + } +} + +<codecs/rt5672/Speaker.conf> +<codecs/rt5672/MonoSpeaker.conf> +<codecs/rt5672/HeadPhones.conf> + +<codecs/rt5672/DMIC1.conf> +<codecs/rt5672/DMIC2.conf> +<codecs/rt5672/HeadsetMic.conf> diff --git a/src/conf/ucm/cht-bsw-rt5672/Makefile.am b/src/conf/ucm/cht-bsw-rt5672/Makefile.am new file mode 100644 index 00000000..c3d557fa --- /dev/null +++ b/src/conf/ucm/cht-bsw-rt5672/Makefile.am @@ -0,0 +1,4 @@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +ucmdir = $(alsaconfigdir)/ucm/cht-bsw-rt5672 +ucm_DATA = cht-bsw-rt5672.conf HiFi.conf +EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/cht-bsw-rt5672/cht-bsw-rt5672.conf b/src/conf/ucm/cht-bsw-rt5672/cht-bsw-rt5672.conf new file mode 100644 index 00000000..c79024d9 --- /dev/null +++ b/src/conf/ucm/cht-bsw-rt5672/cht-bsw-rt5672.conf @@ -0,0 +1,6 @@ +# Adapted from https://github.com/plbossart/UCM/tree/master/cht-bsw-rt5672 + +SectionUseCase."HiFi" { + File "HiFi.conf" + Comment "Play HiFi quality Music" +} diff --git a/src/conf/ucm/codecs/Makefile.am b/src/conf/ucm/codecs/Makefile.am index 5987b9cf..17c4ebf9 100644 --- a/src/conf/ucm/codecs/Makefile.am +++ b/src/conf/ucm/codecs/Makefile.am @@ -4,4 +4,5 @@ es8316 \ rt5640 \ rt5645 \ rt5651 \ +rt5672 \ nau8824 diff --git a/src/conf/ucm/codecs/rt5672/DMIC1.conf b/src/conf/ucm/codecs/rt5672/DMIC1.conf new file mode 100644 index 00000000..9a8db1a5 --- /dev/null +++ b/src/conf/ucm/codecs/rt5672/DMIC1.conf @@ -0,0 +1,28 @@ +SectionDevice."DigitalMic-DMIC1" { + Comment "Internal Digital Microphone on DMIC1" + + ConflictingDevice [ + "DigitalMic-DMIC2" + "HeadsetMic" + ] + + EnableSequence [ + cdev "hw:chtbswrt5672" + cset "name='Int Mic Switch' on" + cset "name='Stereo1 DMIC Mux' DMIC1" + cset "name='Stereo1 ADC 2 Mux' DMIC" + cset "name='Sto1 ADC MIXL ADC2 Switch' on" + cset "name='Sto1 ADC MIXR ADC2 Switch' on" + ] + + DisableSequence [ + cdev "hw:chtbswrt5672" + cset "name='Int Mic Switch' off" + cset "name='Sto1 ADC MIXL ADC2 Switch' off" + cset "name='Sto1 ADC MIXR ADC2 Switch' off" + ] + + Value { + CaptureChannels 2 + } +} diff --git a/src/conf/ucm/codecs/rt5672/DMIC2.conf b/src/conf/ucm/codecs/rt5672/DMIC2.conf new file mode 100644 index 00000000..b960436e --- /dev/null +++ b/src/conf/ucm/codecs/rt5672/DMIC2.conf @@ -0,0 +1,28 @@ +SectionDevice."DigitalMic-DMIC2" { + Comment "Internal Digital Microphone on DMIC2" + + ConflictingDevice [ + "DigitalMic-DMIC1" + "HeadsetMic" + ] + + EnableSequence [ + cdev "hw:chtbswrt5672" + cset "name='Int Mic Switch' on" + cset "name='Stereo1 DMIC Mux' DMIC2" + cset "name='Stereo1 ADC 2 Mux' DMIC" + cset "name='Sto1 ADC MIXL ADC2 Switch' on" + cset "name='Sto1 ADC MIXR ADC2 Switch' on" + ] + + DisableSequence [ + cdev "hw:chtbswrt5672" + cset "name='Int Mic Switch' off" + cset "name='Sto1 ADC MIXL ADC2 Switch' off" + cset "name='Sto1 ADC MIXR ADC2 Switch' off" + ] + + Value { + CaptureChannels 2 + } +} diff --git a/src/conf/ucm/codecs/rt5672/EnableSeq.conf b/src/conf/ucm/codecs/rt5672/EnableSeq.conf new file mode 100644 index 00000000..bd4f357c --- /dev/null +++ b/src/conf/ucm/codecs/rt5672/EnableSeq.conf @@ -0,0 +1,48 @@ +# Playback (I2S1) + +# common +cset "name='DAC1 L Mux' IF1 DAC" +cset "name='DAC1 R Mux' IF1 DAC" +cset "name='DAC1 MIXL DAC1 Switch' on" +cset "name='DAC1 MIXR DAC1 Switch' on" +cset "name='Stereo DAC MIXL DAC L1 Switch' on" +cset "name='Stereo DAC MIXL DAC R1 Switch' off" +cset "name='Stereo DAC MIXR DAC R1 Switch' on" +cset "name='Stereo DAC MIXR DAC L1 Switch' off" + +# speaker +cset "name='PDM1 L Mux' Stereo DAC" +cset "name='PDM1 R Mux' Stereo DAC" + +# headphone +cset "name='HPOVOL MIXL DAC1 Switch' on" +cset "name='HPOVOL MIXR DAC1 Switch' on" +cset "name='HPO MIX HPVOL Switch' on" + +# line out +# cset "name='OUT MIXL DAC L1 Switch' on" +# cset "name='OUT MIXR DAC R1 Switch' on" +# cset "name='LOUT MIX OUTMIX L Switch' on" +# cset "name='LOUT MIX OUTMIX R Switch' on" + +# 47=0dB, 0.375 dB/step +cset "name='ADC Capture Volume' 47" +cset "name='ADC Capture Switch' on" + +# Headset mic is quite soft, boost it a bit, 1 = 20dB which is the first +# available boost step +cset "name='IN1 Boost Volume' 1" + +# Start with all outputs / inputs disabled +cset "name='Ext Spk Switch' off" +cset "name='Headphone Switch' off" +cset "name='Int Mic Switch' off" +cset "name='Headset Mic Switch' off" +cset "name='Sto1 ADC MIXL ADC1 Switch' off" +cset "name='Sto1 ADC MIXR ADC1 Switch' off" +cset "name='Sto1 ADC MIXL ADC2 Switch' off" +cset "name='Sto1 ADC MIXR ADC2 Switch' off" +cset "name='RECMIXL BST1 Switch' off" +cset "name='RECMIXR BST1 Switch' off" +cset "name='RECMIXL BST2 Switch' off" +cset "name='RECMIXR BST2 Switch' off" diff --git a/src/conf/ucm/codecs/rt5672/HeadPhones.conf b/src/conf/ucm/codecs/rt5672/HeadPhones.conf new file mode 100644 index 00000000..cb8aad2d --- /dev/null +++ b/src/conf/ucm/codecs/rt5672/HeadPhones.conf @@ -0,0 +1,23 @@ +SectionDevice."Headphones" { + Comment "Headphones" + + ConflictingDevice [ + "Speaker" + "MonoSpeaker" + ] + + EnableSequence [ + cdev "hw:chtbswrt5672" + cset "name='Headphone Switch' on" + ] + + DisableSequence [ + cdev "hw:chtbswrt5672" + cset "name='Headphone Switch' off" + ] + + Value { + PlaybackChannels "2" + JackControl "Headphone Jack" + } +} diff --git a/src/conf/ucm/codecs/rt5672/HeadsetMic.conf b/src/conf/ucm/codecs/rt5672/HeadsetMic.conf new file mode 100644 index 00000000..cc6d8d56 --- /dev/null +++ b/src/conf/ucm/codecs/rt5672/HeadsetMic.conf @@ -0,0 +1,38 @@ +SectionDevice."HeadsetMic" { + Comment "Headset Microphone" + + ConflictingDevice [ + "DigitalMic-DMIC1" + "DigitalMic-DMIC2" + ] + + EnableSequence [ + cdev "hw:chtbswrt5672" + cset "name='Headset Mic Switch' on" + cset "name='RECMIXL BST1 Switch' on" + cset "name='RECMIXR BST1 Switch' on" + cset "name='Sto1 ADC MIXL ADC1 Switch' on" + cset "name='Sto1 ADC MIXR ADC1 Switch' on" + # For unknown reasons the headset mic sound is only recorded + # on the left channel, we tweak the data-stream reception to + # use the left samples for both slots, so that we get a stereo + # stream with the mic sound on both channels + cset "name='codec_in rx deinterleaver codec_in0_1' 1" + ] + + DisableSequence [ + cdev "hw:chtbswrt5672" + cset "name='Headset Mic Switch' off" + cset "name='RECMIXL BST1 Switch' off" + cset "name='RECMIXR BST1 Switch' off" + cset "name='Sto1 ADC MIXL ADC1 Switch' off" + cset "name='Sto1 ADC MIXR ADC1 Switch' off" + # Undo data-stream reception mono-mix workaround + cset "name='codec_in rx deinterleaver codec_in0_1' 2" + ] + + Value { + CaptureChannels "2" + JackControl "Headset Mic Jack" + } +} diff --git a/src/conf/ucm/codecs/rt5672/Makefile.am b/src/conf/ucm/codecs/rt5672/Makefile.am new file mode 100644 index 00000000..feba0b76 --- /dev/null +++ b/src/conf/ucm/codecs/rt5672/Makefile.am @@ -0,0 +1,5 @@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +ucmdir = $(alsaconfigdir)/ucm/codecs/rt5672 +ucm_DATA = DMIC1.conf DMIC2.conf EnableSeq.conf HeadPhones.conf \ + HeadsetMic.conf MonoSpeaker.conf Speaker.conf +EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/codecs/rt5672/MonoSpeaker.conf b/src/conf/ucm/codecs/rt5672/MonoSpeaker.conf new file mode 100644 index 00000000..828a1154 --- /dev/null +++ b/src/conf/ucm/codecs/rt5672/MonoSpeaker.conf @@ -0,0 +1,26 @@ +SectionDevice."MonoSpeaker" { + Comment "Mono Speaker" + + ConflictingDevice [ + "Speaker" + "Headphones" + ] + + EnableSequence [ + cdev "hw:chtbswrt5672" + cset "name='Stereo DAC MIXR DAC R1 Switch' off" + cset "name='Stereo DAC MIXL DAC R1 Switch' on" + cset "name='Ext Spk Switch' on" + ] + + DisableSequence [ + cdev "hw:chtbswrt5672" + cset "name='Ext Spk Switch' off" + cset "name='Stereo DAC MIXL DAC R1 Switch' off" + cset "name='Stereo DAC MIXR DAC R1 Switch' on" + ] + + Value { + PlaybackChannels "2" + } +} diff --git a/src/conf/ucm/codecs/rt5672/Speaker.conf b/src/conf/ucm/codecs/rt5672/Speaker.conf new file mode 100644 index 00000000..f9291e53 --- /dev/null +++ b/src/conf/ucm/codecs/rt5672/Speaker.conf @@ -0,0 +1,22 @@ +SectionDevice."Speaker" { + Comment "Speakers" + + ConflictingDevice [ + "MonoSpeaker" + "Headphones" + ] + + EnableSequence [ + cdev "hw:chtbswrt5672" + cset "name='Ext Spk Switch' on" + ] + + DisableSequence [ + cdev "hw:chtbswrt5672" + cset "name='Ext Spk Switch' off" + ] + + Value { + PlaybackChannels "2" + } +} -- 2.23.0

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[alsa-devel] [RFC PATCH 0/7] adapt SOF to use snd-hda-codec-hdmi
by Kai Vehmanen 03 Sep '19

03 Sep '19

Hi all, here's a RFC patch series that adapts SOF (and one example machine driver) to use snd-hda-codec-hdmi (patch_hdmi.c) codec driver instead of hdac_hdmi (soc/codecs/hdac_hdmi.c). The primary goal is unify the HDMI codec implementation between DSP and non-DSP HDA configurations, offer same interface to user-space and reduce maintenance load for all. Main points I'd like your input on: 1) Is the high-level approach ok? - SOF already uses pci/hda/ for all others codecs, so HDMI has been the sole exception where code has been duplicated. - I've tried to keep changes to hda/hdmi minimal. - This series implements logic to parse the PCM topology in a dynamic fashion, so we do not have to change all existing (and future) DSP topologis to use fixed PCM device numbers for HDMI, and we avoid need to hardcode PCM device numbers in machine driver code. 2) Can we drop hdac_hdmi and its support from machine drivers, or do we need to make it optional and keep it around? - Current series does not add any Kconfig options, but simply switches SOF to use HDA codecs for all, including HDMI/DP. This means hdac_hdmi is never used with SOF and could be dropped (if SST is ok as well). - This may break some usage with SST (input is welcome!) - The change is visible to applications. The ALSA mixer interface is different (OTOH with the new driver, playback works out-of-the-box while with hdac_hdmi you needed to set the multiple controls first, to have any audio out). - Alternatively I could add a KConfig option and we could have a deprecation period for hdac_hdmi, allowing people to compile with the old driver during transition time. This will require #ifdef'ing in all the machine drivers. .. once these are addressed, I can proceed to extend the patchset with all affected machine drivers. Feature and testing info: - Tested on multiple Intel platforms supported by SOF. - Tested with ALSA console tools as well as with Pulseaudio. - requires Pulseaudio 12.x or newer, see https://lists.freedesktop.org/archives/pulseaudio-discuss/2019-August/03135… - HDMI, DP, DP-MST with multi-monitor use-scenarios work ok. - New feature for SOF: ELD /proc fs works just like in DSP-less mode. - New feature for SOF: jack detection works out-of-the-box with Pulseaudio (no need for card specific UCM for HDMI) - Pre-reviewed at: https://github.com/thesofproject/linux/pull/1155 Kai Vehmanen (7): ALSA: hda - add mst_no_extra_pcms flag to hda_codec ASoC: Intel: skl-hda-dsp-generic: use snd-hda-codec-hdmi ASoC: hdac_hda: add support for HDMI/DP as a HDA codec ALSA: hda/hdmi - allow control creation without a linked pcm ALSA: hda/hdmi - implement mst_no_extra_pcms flag ALSA: hda/hdmi - complete pcm_setup_pin without snd_pcm link ASoC: SOF: Intel: load hda codec module also for HDMI/DP include/sound/hda_codec.h | 1 + sound/pci/hda/patch_hdmi.c | 31 ++++--- sound/soc/codecs/hdac_hda.c | 95 +++++++++++++++++--- sound/soc/codecs/hdac_hda.h | 10 ++- sound/soc/intel/boards/skl_hda_dsp_common.c | 27 ++---- sound/soc/intel/boards/skl_hda_dsp_common.h | 61 +++++++++++++ sound/soc/intel/boards/skl_hda_dsp_generic.c | 7 -- sound/soc/sof/intel/hda-codec.c | 11 +-- sound/soc/sof/intel/hda.h | 5 +- 9 files changed, 189 insertions(+), 59 deletions(-) -- 2.17.1

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[alsa-devel] [PATCH v2 1/3] ASoC: es8316: judge PCM rate at later timing
by Katsuhiro Suzuki 03 Sep '19

03 Sep '19

This patch change the judge timing about playing/capturing PCM rate. Original code set constraints list of PCM rate limits at set_sysclk. This strategy works well if system is using fixed rate clock. But some boards and SoC (such as RockPro64 and RockChip I2S) has connected SoC MCLK out to ES8316 MCLK in. In this case, SoC side I2S will choose suitable frequency of MCLK such as fs * mclk-fs when user starts playing or capturing. Bad scenario as follows (mclk-fs = 256): - Initialize sysclk by correct value (Ex. 12.288MHz) - ES8316 set constraints of PCM rate by sysclk 48kHz (1/256), 32kHz (1/384), 30.720kHz (1/400), 24kHz (1/512), 16kHz (1/768), 12kHz (1/1024) - Play 48kHz sound, it's acceptable - Sysclk is not changed - Play 32kHz sound, it's acceptable - Set sysclk by 8.192MHz (= fs * mclk-fs = 32k * 256) - ES8316 set constraints of PCM rate by sysclk 32kHz (1/256), 21.33kHz (1/384), 20.48kHz (1/400), 16kHz (1/512), 10.66kHz (1/768), 8kHz (1/1024) - Play 48kHz again, but it's NOT acceptable because constraints list does not allow 48kHz Root cause of this strange behavior is changing constraints list at set_sysclk timing. It seems that is too early to determine. So this patch does not use constraints list and check PCM rate limit more later timing at hw_params. Signed-off-by: Katsuhiro Suzuki <katsuhiro(a)katsuster.net> --- sound/soc/codecs/es8316.c | 37 +++++++++++++++---------------------- 1 file changed, 15 insertions(+), 22 deletions(-) diff --git a/sound/soc/codecs/es8316.c b/sound/soc/codecs/es8316.c index ed2959dbe1fb..229808fa627c 100644 --- a/sound/soc/codecs/es8316.c +++ b/sound/soc/codecs/es8316.c @@ -363,27 +363,12 @@ static int es8316_set_dai_sysclk(struct snd_soc_dai *codec_dai, { struct snd_soc_component *component = codec_dai->component; struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); - int i; - int count = 0; es8316->sysclk = freq; if (freq == 0) return 0; - /* Limit supported sample rates to ones that can be autodetected - * by the codec running in slave mode. - */ - for (i = 0; i < NR_SUPPORTED_MCLK_LRCK_RATIOS; i++) { - const unsigned int ratio = supported_mclk_lrck_ratios[i]; - - if (freq % ratio == 0) - es8316->allowed_rates[count++] = freq / ratio; - } - - es8316->sysclk_constraints.list = es8316->allowed_rates; - es8316->sysclk_constraints.count = count; - return 0; } @@ -449,13 +434,6 @@ static int es8316_pcm_startup(struct snd_pcm_substream *substream, return -EINVAL; } - /* The set of sample rates that can be supported depends on the - * MCLK supplied to the CODEC. - */ - snd_pcm_hw_constraint_list(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_RATE, - &es8316->sysclk_constraints); - return 0; } @@ -466,12 +444,27 @@ static int es8316_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_component *component = dai->component; struct es8316_priv *es8316 = snd_soc_component_get_drvdata(component); u8 wordlen = 0; + int i; if (!es8316->sysclk) { dev_err(component->dev, "No MCLK configured\n"); return -EINVAL; } + /* Limit supported sample rates to ones that can be autodetected + * by the codec running in slave mode. + */ + for (i = 0; i < NR_SUPPORTED_MCLK_LRCK_RATIOS; i++) { + const unsigned int ratio = supported_mclk_lrck_ratios[i]; + + if (es8316->sysclk % ratio != 0) + continue; + if (es8316->sysclk / ratio == params_rate(params)) + break; + } + if (i == NR_SUPPORTED_MCLK_LRCK_RATIOS) + return -EINVAL; + switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: wordlen = ES8316_SERDATA2_LEN_16; -- 2.23.0.rc1

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[alsa-devel] [PATCH] MAINTAINERS: update entry for firewire audio drivers with UAPI header
by Takashi Sakamoto 03 Sep '19

03 Sep '19

The most of drivers in ALSA firewire stack supports common ioctl commands to enable/disable packet streaming as well as some ioctl commands for model-specific features. An UAPI header is exported to userspace. This commit adds supplement for entry of ALSA firewire stack with a path of the UAPI header. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- MAINTAINERS | 1 + 1 file changed, 1 insertion(+) diff --git a/MAINTAINERS b/MAINTAINERS index 4200194e69ea..6bd54ce6dc66 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -6271,6 +6271,7 @@ L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) T: git git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git S: Maintained F: sound/firewire/ +F: include/uapi/sound/firewire.h FIREWIRE MEDIA DRIVERS (firedtv) M: Stefan Richter <stefanr(a)s5r6.in-berlin.de> -- 2.20.1

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[alsa-devel] [PATCH] ASoC: simple-card: indicate rebind issue
by Kuninori Morimoto 03 Sep '19

03 Sep '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> ALSA SoC try to rebind Sound Card if Card/CPU/Codec/Platform were unbinded and re-binded again. But, Simple Card might can't rebind again if user do for example unbind CPU or Codec driver bind CPU or Codec driver Because Simple Card is still pointing old/unbinded CPU or Codec driver's DAI name at dlc->dai_name. To avoid this issue, it needs to alloc memory and keep DAI name even though if CPU or Codec driver was unbinded. Or, always do unbind/bind at Sound Card. For now, this patch indicates this issue as FIXME. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/simple-card.c | 20 +++++++++++++++++++- 1 file changed, 19 insertions(+), 1 deletion(-) diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 0eac3bc..fc9c753 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -46,7 +46,25 @@ static int asoc_simple_parse_dai(struct device_node *node, if (ret) return ret; - /* Get dai->name */ + /* + * FIXME + * + * Here, dlc->dai_name is pointer to CPU/Codec DAI name. + * If user unbinded CPU or Codec driver, but not for Sound Card, + * dlc->dai_name is keeping unbinded CPU or Codec + * driver's pointer. + * + * If user re-bind CPU or Codec driver again, ALSA SoC will try + * to rebind Card via snd_soc_try_rebind_card(), but because of + * above reason, it might can't bind Sound Card. + * Because Sound Card is pointing to released dai_name pointer. + * + * To avoid this rebind Card issue, + * 1) It needs to alloc memory to keep dai_name eventhough + * CPU or Codec driver was unbinded, or + * 2) user need to rebind Sound Card everytime + * if he unbinded CPU or Codec. + */ ret = snd_soc_of_get_dai_name(node, &dlc->dai_name); if (ret < 0) return ret; -- 2.7.4

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[alsa-devel] [PATCH] ASoC: audio-graph: indicate rebind issue
by Kuninori Morimoto 03 Sep '19

03 Sep '19

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> ALSA SoC try to rebind Sound Card if Card/CPU/Codec/Platform were unbinded and re-binded again. But, Audio Graph Card might can't rebind again if user do for example unbind CPU or Codec driver bind CPU or Codec driver Because Audio Graph Card is still pointing old/unbinded CPU or Codec driver's DAI name at dlc->dai_name. To avoid this issue, it needs to alloc memory and keep DAI name even though if CPU or Codec driver was unbinded. Or, always do unbind/bind at Sound Card. For now, this patch indicates this issue as FIXME. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/generic/audio-graph-card.c | 19 +++++++++++++++++++ 1 file changed, 19 insertions(+) diff --git a/sound/soc/generic/audio-graph-card.c b/sound/soc/generic/audio-graph-card.c index 288df24..6007e63 100644 --- a/sound/soc/generic/audio-graph-card.c +++ b/sound/soc/generic/audio-graph-card.c @@ -129,6 +129,25 @@ static int asoc_simple_parse_dai(struct device_node *ep, args.args[0] = graph_get_dai_id(ep); args.args_count = (of_graph_get_endpoint_count(node) > 1); + /* + * FIXME + * + * Here, dlc->dai_name is pointer to CPU/Codec DAI name. + * If user unbinded CPU or Codec driver, but not for Sound Card, + * dlc->dai_name is keeping unbinded CPU or Codec + * driver's pointer. + * + * If user re-bind CPU or Codec driver again, ALSA SoC will try + * to rebind Card via snd_soc_try_rebind_card(), but because of + * above reason, it might can't bind Sound Card. + * Because Sound Card is pointing to released dai_name pointer. + * + * To avoid this rebind Card issue, + * 1) It needs to alloc memory to keep dai_name eventhough + * CPU or Codec driver was unbinded, or + * 2) user need to rebind Sound Card everytime + * if he unbinded CPU or Codec. + */ ret = snd_soc_get_dai_name(&args, &dlc->dai_name); if (ret < 0) return ret; -- 2.7.4

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[alsa-devel] [PATCH] alsamixer: Remove exp10 usage
by Rosen Penev 03 Sep '19

03 Sep '19

exp10 is a GNU extension and not available everywhere (eg. uClibc-ng). Signed-off-by: Rosen Penev <rosenp(a)gmail.com> --- alsamixer/volume_mapping.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/alsamixer/volume_mapping.c b/alsamixer/volume_mapping.c index 48cfbe2..29ab061 100644 --- a/alsamixer/volume_mapping.c +++ b/alsamixer/volume_mapping.c @@ -108,9 +108,9 @@ static double get_normalized_volume(snd_mixer_elem_t *elem, if (use_linear_dB_scale(min, max)) return (value - min) / (double)(max - min); - normalized = exp10((value - max) / 6000.0); + normalized = pow(10, (value - max) / 6000.0); if (min != SND_CTL_TLV_DB_GAIN_MUTE) { - min_norm = exp10((min - max) / 6000.0); + min_norm = pow(10, (min - max) / 6000.0); normalized = (normalized - min_norm) / (1 - min_norm); } @@ -143,7 +143,7 @@ static int set_normalized_volume(snd_mixer_elem_t *elem, } if (min != SND_CTL_TLV_DB_GAIN_MUTE) { - min_norm = exp10((min - max) / 6000.0); + min_norm = pow(10, (min - max) / 6000.0); volume = volume * (1 - min_norm) + min_norm; } value = lrint_dir(6000.0 * log10(volume), dir) + max; -- 2.17.1

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