December 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v4 08/11] ASoC: Intel: atom: Make PCI dependency explicit
by Sinan Kaya 02 Jan '19

02 Jan '19

Code does unconditional select for IOSF_MBI. IOSF_MBI driver depends on CONFIG_PCI set but this is not specified anywhere. Fixes: 5d32a66541c46 ("PCI/ACPI: Allow ACPI to be built without CONFIG_PCI set") Signed-off-by: Sinan Kaya <okaya(a)kernel.org> --- sound/soc/intel/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 2fd1b61e8331..b0764b2fe001 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -91,7 +91,7 @@ config SND_SST_ATOM_HIFI2_PLATFORM_PCI config SND_SST_ATOM_HIFI2_PLATFORM_ACPI tristate "ACPI HiFi2 (Baytrail, Cherrytrail) Platforms" default ACPI - depends on X86 && ACPI + depends on X86 && ACPI && PCI select SND_SST_IPC_ACPI select SND_SST_ATOM_HIFI2_PLATFORM select SND_SOC_ACPI_INTEL_MATCH -- 2.19.0

4 11

[alsa-devel] [PATCH 1/4] conf/ucm: bytcr-rt5651: Enable Stereo? ADC MIXL ADC? switches when enabling inputs
by Hans de Goede 01 Jan '19

01 Jan '19

Explicitly enable the right 'Stereo? ADC MIXL ADC?' when selecting an input, rather then just enabling them all in ucm/codecs/rt5651/EnableSeq.conf . This is a preparation patch for adding digital mic support. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- src/conf/ucm/codecs/rt5651/EnableSeq.conf | 16 ++++++++-------- src/conf/ucm/codecs/rt5651/IN1-InternalMic.conf | 4 ++++ src/conf/ucm/codecs/rt5651/IN12-InternalMic.conf | 4 ++++ src/conf/ucm/codecs/rt5651/IN2-InternalMic.conf | 4 ++++ src/conf/ucm/codecs/rt5651/IN3-HeadsetMic.conf | 4 ++++ 5 files changed, 24 insertions(+), 8 deletions(-) diff --git a/src/conf/ucm/codecs/rt5651/EnableSeq.conf b/src/conf/ucm/codecs/rt5651/EnableSeq.conf index 6fb3c1f2..db2fd328 100644 --- a/src/conf/ucm/codecs/rt5651/EnableSeq.conf +++ b/src/conf/ucm/codecs/rt5651/EnableSeq.conf @@ -14,12 +14,14 @@ cset "name='Stereo DAC MIXL DAC L1 Switch' on" cset "name='Stereo DAC MIXL DAC R1 Switch' off" cset "name='Stereo DAC MIXR DAC L1 Switch' off" cset "name='Stereo DAC MIXR DAC R1 Switch' on" -cset "name='Stereo1 ADC MIXR ADC1 Switch' on" -cset "name='Stereo1 ADC MIXR ADC2 Switch' on" -cset "name='Stereo2 ADC MIXL ADC1 Switch' on" -cset "name='Stereo2 ADC MIXL ADC2 Switch' on" -cset "name='Stereo2 ADC MIXR ADC1 Switch' on" -cset "name='Stereo2 ADC MIXR ADC2 Switch' on" +cset "name='Stereo1 ADC MIXL ADC1 Switch' off" +cset "name='Stereo1 ADC MIXR ADC1 Switch' off" +cset "name='Stereo1 ADC MIXL ADC2 Switch' off" +cset "name='Stereo1 ADC MIXR ADC2 Switch' off" +cset "name='Stereo2 ADC MIXL ADC1 Switch' off" +cset "name='Stereo2 ADC MIXR ADC1 Switch' off" +cset "name='Stereo2 ADC MIXL ADC2 Switch' off" +cset "name='Stereo2 ADC MIXR ADC2 Switch' off" cset "name='Headphone Switch' off" cset "name='Speaker Switch' off" @@ -41,8 +43,6 @@ cset "name='IN3 Boost' 1" # input cset "name='Stereo1 ADC L1 Mux' ADC" cset "name='Stereo1 ADC R1 Mux' ADC" -cset "name='Stereo1 ADC MIXL ADC1 Switch' on" -cset "name='Stereo1 ADC MIXR ADC1 Switch' on" cset "name='RECMIXL BST1 Switch' off" cset "name='RECMIXR BST1 Switch' off" cset "name='RECMIXL BST3 Switch' off" diff --git a/src/conf/ucm/codecs/rt5651/IN1-InternalMic.conf b/src/conf/ucm/codecs/rt5651/IN1-InternalMic.conf index 85e25a5c..b2ce17b6 100644 --- a/src/conf/ucm/codecs/rt5651/IN1-InternalMic.conf +++ b/src/conf/ucm/codecs/rt5651/IN1-InternalMic.conf @@ -12,10 +12,14 @@ SectionDevice."InternalMic-IN1" { cset "name='Internal Mic Switch' on" cset "name='RECMIXL BST1 Switch' on" cset "name='RECMIXR BST1 Switch' on" + cset "name='Stereo1 ADC MIXL ADC1 Switch' on" + cset "name='Stereo1 ADC MIXR ADC1 Switch' on" ] DisableSequence [ cdev "hw:bytcrrt5651" + cset "name='Stereo1 ADC MIXL ADC1 Switch' off" + cset "name='Stereo1 ADC MIXR ADC1 Switch' off" cset "name='Internal Mic Switch' off" cset "name='RECMIXL BST1 Switch' off" cset "name='RECMIXR BST1 Switch' off" diff --git a/src/conf/ucm/codecs/rt5651/IN12-InternalMic.conf b/src/conf/ucm/codecs/rt5651/IN12-InternalMic.conf index f1cc17fe..e8041912 100644 --- a/src/conf/ucm/codecs/rt5651/IN12-InternalMic.conf +++ b/src/conf/ucm/codecs/rt5651/IN12-InternalMic.conf @@ -14,10 +14,14 @@ SectionDevice."InternalMic-IN12" { cset "name='RECMIXR BST1 Switch' on" cset "name='RECMIXL BST2 Switch' on" cset "name='RECMIXR BST2 Switch' on" + cset "name='Stereo1 ADC MIXL ADC1 Switch' on" + cset "name='Stereo1 ADC MIXR ADC1 Switch' on" ] DisableSequence [ cdev "hw:bytcrrt5651" + cset "name='Stereo1 ADC MIXL ADC1 Switch' off" + cset "name='Stereo1 ADC MIXR ADC1 Switch' off" cset "name='Internal Mic Switch' off" cset "name='RECMIXL BST1 Switch' off" cset "name='RECMIXR BST1 Switch' off" diff --git a/src/conf/ucm/codecs/rt5651/IN2-InternalMic.conf b/src/conf/ucm/codecs/rt5651/IN2-InternalMic.conf index 18f74e1a..048fcc95 100644 --- a/src/conf/ucm/codecs/rt5651/IN2-InternalMic.conf +++ b/src/conf/ucm/codecs/rt5651/IN2-InternalMic.conf @@ -12,10 +12,14 @@ SectionDevice."InternalMic-IN2" { cset "name='Internal Mic Switch' on" cset "name='RECMIXL BST2 Switch' on" cset "name='RECMIXR BST2 Switch' on" + cset "name='Stereo1 ADC MIXL ADC1 Switch' on" + cset "name='Stereo1 ADC MIXR ADC1 Switch' on" ] DisableSequence [ cdev "hw:bytcrrt5651" + cset "name='Stereo1 ADC MIXL ADC1 Switch' off" + cset "name='Stereo1 ADC MIXR ADC1 Switch' off" cset "name='Internal Mic Switch' off" cset "name='RECMIXL BST2 Switch' off" cset "name='RECMIXR BST2 Switch' off" diff --git a/src/conf/ucm/codecs/rt5651/IN3-HeadsetMic.conf b/src/conf/ucm/codecs/rt5651/IN3-HeadsetMic.conf index bfd0b009..712d14bb 100644 --- a/src/conf/ucm/codecs/rt5651/IN3-HeadsetMic.conf +++ b/src/conf/ucm/codecs/rt5651/IN3-HeadsetMic.conf @@ -12,10 +12,14 @@ SectionDevice."HeadsetMic-IN3" { cset "name='Headset Mic Switch' on" cset "name='RECMIXL BST3 Switch' on" cset "name='RECMIXR BST3 Switch' on" + cset "name='Stereo1 ADC MIXL ADC1 Switch' on" + cset "name='Stereo1 ADC MIXR ADC1 Switch' on" ] DisableSequence [ cdev "hw:bytcrrt5651" + cset "name='Stereo1 ADC MIXL ADC1 Switch' off" + cset "name='Stereo1 ADC MIXR ADC1 Switch' off" cset "name='Headset Mic Switch' off" cset "name='RECMIXL BST3 Switch' off" cset "name='RECMIXR BST3 Switch' off" -- 2.20.1

2 6

[alsa-devel] [PATCH v2] staging: bcm2835-audio: interpolate audio delay
by Mike Brady 01 Jan '19

01 Jan '19

When the BCM2835 audio output is used, userspace sees a jitter up to 10ms in the audio position, aka "delay" -- the number of frames that must be output before a new frame would be played. Make this a bit nicer for userspace by interpolating the position using the CPU clock. The overhead is small -- an extra ktime_get() every time a GPU message is sent -- and another call and a few calculations whenever the delay is sought from userland. At 48,000 frames per second, i.e. approximately 20 microseconds per frame, it would take a clock inaccuracy of 20 microseconds in 10 milliseconds -- 2,000 parts per million -- to result in an inaccurate estimate, whereas crystal- or resonator-based clocks typically have an inaccuracy of 10s to 100s of parts per million. Signed-off-by: Mike Brady <mikebrady(a)eircom.net> --- Changes in v2 -- remove inappropriate addition of SNDRV_PCM_INFO_BATCH flag .../vc04_services/bcm2835-audio/bcm2835-pcm.c | 20 +++++++++++++++++++ .../vc04_services/bcm2835-audio/bcm2835.h | 1 + 2 files changed, 21 insertions(+) diff --git a/drivers/staging/vc04_services/bcm2835-audio/bcm2835-pcm.c b/drivers/staging/vc04_services/bcm2835-audio/bcm2835-pcm.c index e66da11af5cf..9053b996cada 100644 --- a/drivers/staging/vc04_services/bcm2835-audio/bcm2835-pcm.c +++ b/drivers/staging/vc04_services/bcm2835-audio/bcm2835-pcm.c @@ -74,6 +74,7 @@ void bcm2835_playback_fifo(struct bcm2835_alsa_stream *alsa_stream, atomic_set(&alsa_stream->pos, pos); alsa_stream->period_offset += bytes; + alsa_stream->interpolate_start = ktime_get(); if (alsa_stream->period_offset >= alsa_stream->period_size) { alsa_stream->period_offset %= alsa_stream->period_size; snd_pcm_period_elapsed(substream); @@ -243,6 +244,7 @@ static int snd_bcm2835_pcm_prepare(struct snd_pcm_substream *substream) atomic_set(&alsa_stream->pos, 0); alsa_stream->period_offset = 0; alsa_stream->draining = false; + alsa_stream->interpolate_start = ktime_get(); return 0; } @@ -292,6 +294,24 @@ snd_bcm2835_pcm_pointer(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct bcm2835_alsa_stream *alsa_stream = runtime->private_data; + ktime_t now = ktime_get(); + + /* Give userspace better delay reporting by interpolating between GPU + * notifications, assuming audio speed is close enough to the clock + * used for ktime + */ + + if ((ktime_to_ns(alsa_stream->interpolate_start)) && + (ktime_compare(alsa_stream->interpolate_start, now) < 0)) { + u64 interval = + (ktime_to_ns(ktime_sub(now, + alsa_stream->interpolate_start))); + u64 frames_output_in_interval = + div_u64((interval * runtime->rate), 1000000000); + snd_pcm_sframes_t frames_output_in_interval_sized = + -frames_output_in_interval; + runtime->delay = frames_output_in_interval_sized; + } return snd_pcm_indirect_playback_pointer(substream, &alsa_stream->pcm_indirect, diff --git a/drivers/staging/vc04_services/bcm2835-audio/bcm2835.h b/drivers/staging/vc04_services/bcm2835-audio/bcm2835.h index e13435d1c205..595ad584243f 100644 --- a/drivers/staging/vc04_services/bcm2835-audio/bcm2835.h +++ b/drivers/staging/vc04_services/bcm2835-audio/bcm2835.h @@ -78,6 +78,7 @@ struct bcm2835_alsa_stream { unsigned int period_offset; unsigned int buffer_size; unsigned int period_size; + ktime_t interpolate_start; struct bcm2835_audio_instance *instance; int idx; -- 2.17.1

4 17

[alsa-devel] [PATCH] aplay: support no period wakeup option in argument
by S.j. Wang 01 Jan '19

01 Jan '19

In the case that alsa driver can't support period wakeup, we need to set the no period wakeup flag Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- aplay/aplay.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/aplay/aplay.c b/aplay/aplay.c index efc1eb4cae3a..4f562bfe2884 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -137,6 +137,7 @@ static int use_strftime = 0; volatile static int recycle_capture_file = 0; static long term_c_lflag = -1; static int dump_hw_params = 0; +static int no_period_wakeup = 0; static int fd = -1; static off64_t pbrec_count = LLONG_MAX, fdcount; @@ -243,6 +244,7 @@ _("Usage: %s [OPTION]... [FILE]...\n" " --use-strftime apply the strftime facility to the output file name\n" " --dump-hw-params dump hw_params of the device\n" " --fatal-errors treat all errors as fatal\n" +" --no-period-wakeup set no period wakeup flag if necessary\n" ) , command); printf(_("Recognized sample formats are:")); @@ -429,6 +431,7 @@ enum { OPT_USE_STRFTIME, OPT_DUMP_HWPARAMS, OPT_FATAL_ERRORS, + OPT_NO_PERIOD_WAKEUP, }; /* @@ -516,6 +519,7 @@ int main(int argc, char *argv[]) {"interactive", 0, 0, 'i'}, {"dump-hw-params", 0, 0, OPT_DUMP_HWPARAMS}, {"fatal-errors", 0, 0, OPT_FATAL_ERRORS}, + {"no-period-wakeup", 0, 0, OPT_NO_PERIOD_WAKEUP}, #ifdef CONFIG_SUPPORT_CHMAP {"chmap", 1, 0, 'm'}, #endif @@ -799,6 +803,9 @@ int main(int argc, char *argv[]) case OPT_FATAL_ERRORS: fatal_errors = 1; break; + case OPT_NO_PERIOD_WAKEUP: + no_period_wakeup = 1; + break; #ifdef CONFIG_SUPPORT_CHMAP case 'm': channel_map = snd_pcm_chmap_parse_string(optarg); @@ -1396,6 +1403,12 @@ static void set_params(void) &buffer_frames); } assert(err >= 0); + + if (no_period_wakeup) { + err = snd_pcm_hw_params_set_period_wakeup(handle, params, 0); + assert(err >= 0); + } + monotonic = snd_pcm_hw_params_is_monotonic(params); can_pause = snd_pcm_hw_params_can_pause(params); err = snd_pcm_hw_params(handle, params); -- 1.9.1

3 6

[alsa-devel] [alsa-lib][PATCH] conf/ucm: kblrt5660: Add ucm setting for Dell Edge IoT platform
by Hui Wang 01 Jan '19

01 Jan '19

The new generation of Dell Edge IoT platform is based on Intel Kabylake platform, and the audio codec is ALC3277 which is 100% compatible with RT5660 in I2S mode. The audio design on this IoT platform is as below: - Intel kabylake platform - connect the codec ALC3277 via SSP0 - line-out and line-in with Micbias jacks - line-out mute control and jack detection of line-out and line-in - two HDMI ports with audio capability Signed-off-by: Hui Wang <hui.wang(a)canonical.com> --- configure.ac | 1 + src/conf/ucm/Makefile.am | 1 + src/conf/ucm/kblrt5660/Hdmi1 | 20 +++++++ src/conf/ucm/kblrt5660/Hdmi2 | 20 +++++++ src/conf/ucm/kblrt5660/HiFi | 85 +++++++++++++++++++++++++++ src/conf/ucm/kblrt5660/Makefile.am | 4 ++ src/conf/ucm/kblrt5660/kblrt5660.conf | 18 ++++++ 7 files changed, 149 insertions(+) create mode 100755 src/conf/ucm/kblrt5660/Hdmi1 create mode 100755 src/conf/ucm/kblrt5660/Hdmi2 create mode 100755 src/conf/ucm/kblrt5660/HiFi create mode 100755 src/conf/ucm/kblrt5660/Makefile.am create mode 100755 src/conf/ucm/kblrt5660/kblrt5660.conf diff --git a/configure.ac b/configure.ac index d0cab2d6..6185a7eb 100644 --- a/configure.ac +++ b/configure.ac @@ -745,6 +745,7 @@ AC_OUTPUT(Makefile doc/Makefile doc/pictures/Makefile doc/doxygen.cfg \ src/conf/ucm/gpd-win-pocket-rt5645/Makefile \ src/conf/ucm/HDAudio-Gigabyte-ALC1220DualCodecs/Makefile \ src/conf/ucm/HDAudio-Lenovo-DualCodecs/Makefile \ + src/conf/ucm/kblrt5660/Makefile \ src/conf/ucm/LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216/Makefile \ src/conf/ucm/PandaBoard/Makefile \ src/conf/ucm/PandaBoardES/Makefile \ diff --git a/src/conf/ucm/Makefile.am b/src/conf/ucm/Makefile.am index 85c15090..993d02bb 100644 --- a/src/conf/ucm/Makefile.am +++ b/src/conf/ucm/Makefile.am @@ -29,6 +29,7 @@ GoogleNyan \ gpd-win-pocket-rt5645 \ HDAudio-Gigabyte-ALC1220DualCodecs \ HDAudio-Lenovo-DualCodecs \ +kblrt5660 \ LENOVO-80XF-LenovoMIIX320_10ICR-LNVNB161216 \ PandaBoard \ PandaBoardES \ diff --git a/src/conf/ucm/kblrt5660/Hdmi1 b/src/conf/ucm/kblrt5660/Hdmi1 new file mode 100755 index 00000000..95e6187a --- /dev/null +++ b/src/conf/ucm/kblrt5660/Hdmi1 @@ -0,0 +1,20 @@ +# Usecase for device HDMI1/Display Port stereo playback on Intel KABYLAKE platforms +# For Audio in I2S mode + +SectionDevice."Hdmi1" { + Comment "HDMI/Display Port 1 Stereo" + + EnableSequence [ + ] + + DisableSequence [ + ] + + Value { + PlaybackPCM "hw:kblrt5660,2" + PlaybackChannels "2" + PlaybackPriority "2" + JackControl "HDMI/DP, pcm=4 Jack" + } +} + diff --git a/src/conf/ucm/kblrt5660/Hdmi2 b/src/conf/ucm/kblrt5660/Hdmi2 new file mode 100755 index 00000000..f6d2feee --- /dev/null +++ b/src/conf/ucm/kblrt5660/Hdmi2 @@ -0,0 +1,20 @@ +# Usecase for device HDMI2/Display Port stereo playback on Intel KABYLAKE platforms +# For Audio in I2S mode + +SectionDevice."Hdmi2" { + Comment "HDMI/Display Port 2 Stereo" + + EnableSequence [ + ] + + DisableSequence [ + ] + + Value { + PlaybackPCM "hw:kblrt5660,3" + PlaybackChannels "2" + PlaybackPriority "3" + JackControl "HDMI/DP, pcm=5 Jack" + } +} + diff --git a/src/conf/ucm/kblrt5660/HiFi b/src/conf/ucm/kblrt5660/HiFi new file mode 100755 index 00000000..dbdb9006 --- /dev/null +++ b/src/conf/ucm/kblrt5660/HiFi @@ -0,0 +1,85 @@ +# UCM for Intel Kabylake platforms with RT5660 +# For Audio in I2S mode + +SectionVerb { + + EnableSequence [ + ] + + DisableSequence [ + ] + + # ALSA PCM + Value { + # ALSA PCM device for HiFi + PlaybackPCM "hw:kblrt5660,0" + CapturePCM "hw:kblrt5660,1" + } +} + +SectionDevice."LineOut" { + Comment "LineOut playback" + + EnableSequence [ + cdev "hw:kblrt5660" + cset "name='DAC1 Playback Volume' 30" + cset "name='OUT Playback Volume' 19" + cset "name='OUT Playback Switch' on" + cset "name='codec0_out mo media0_in mi Switch' on" + 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 MIXR DAC L1 Switch' off" + cset "name='Stereo DAC MIXR DAC R1 Switch' on" + cset "name='Stereo DAC MIXL DAC R1 Switch' off" + cset "name='LOUT MIX DAC Switch' on" + cset "name='Line Out Switch' on" + ] + + DisableSequence [ + cdev "hw:kblrt5660" + cset "name='Line Out Switch' off" + cset "name='DAC1 MIXL DAC1' off" + cset "name='DAC1 MIXR DAC1' off" + ] + + Value { + PlaybackChannels "2" + JackDev "rt5660-jack" + JackControl "Line Out Jack" + } +} + +SectionDevice."LineIn" { + Comment "LineIn capture" + + EnableSequence [ + cdev "hw:kblrt5660" + + cset "name='ADC Capture Volume' 30" + cset "name='IN1 Boost Volume' 30" + cset "name='IN2 Boost Volume' 30" + cset "name='STO1 ADC Boost Gain Volume' 1" + cset "name='RECMIXL BST1 Switch' off" + cset "name='RECMIXR BST1 Switch' on" + cset "name='RECMIXL BST2 Switch' on" + cset "name='RECMIXR BST2 Switch' off" + cset "name='Line In Switch' on" + cset "name='Sto1 ADC MIXR ADC1 Switch' on" + cset "name='Sto1 ADC MIXL ADC1 Switch' on" + cset "name='media0_out mo codec0_in mi Switch' on" + ] + + DisableSequence [ + cdev "hw:kblrt5660" + cset "name='Line In Switch' off" + cset "name='Sto1 ADC MIXR ADC1 Switch' off" + cset "name='Sto1 ADC MIXL ADC1 Switch' off" + ] + + Value { + CaptureChannels "2" + JackDev "rt5660-jack" + JackControl "Line In Jack" + } +} diff --git a/src/conf/ucm/kblrt5660/Makefile.am b/src/conf/ucm/kblrt5660/Makefile.am new file mode 100755 index 00000000..83c3743a --- /dev/null +++ b/src/conf/ucm/kblrt5660/Makefile.am @@ -0,0 +1,4 @@ +alsaconfigdir = @ALSA_CONFIG_DIR@ +ucmdir = $(alsaconfigdir)/ucm/kblrt5660 +ucm_DATA = kblrt5660.conf HiFi Hdmi1 Hdmi2 +EXTRA_DIST = $(ucm_DATA) diff --git a/src/conf/ucm/kblrt5660/kblrt5660.conf b/src/conf/ucm/kblrt5660/kblrt5660.conf new file mode 100755 index 00000000..c5e9420e --- /dev/null +++ b/src/conf/ucm/kblrt5660/kblrt5660.conf @@ -0,0 +1,18 @@ +SectionUseCase."HiFi" { + File "HiFi" + Comment "Play HiFi quality Music" +} + +SectionUseCase."Hdmi1" { + File "Hdmi1" + Comment "Play on Hdmi/DP 1" +} + +SectionUseCase."Hdmi2" { + File "Hdmi2" + Comment "Play on Hdmi/DP 2" +} + +SectionDefaults [ + cdev "hw:kblrt5660" +] -- 2.17.1

2 1

Re: [alsa-devel] [PATCH] sound: atom: fix a missing check of snd_pcm_lib_malloc_pages
by Pierre-Louis Bossart 01 Jan '19

01 Jan '19

On 12/25/18 8:29 PM, Kangjie Lu wrote: > snd_pcm_lib_malloc_pages() may fail, so let's check its status and > return its error code upstream. Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> > > Signed-off-by: Kangjie Lu <kjlu(a)umn.edu> > --- > sound/soc/intel/atom/sst-mfld-platform-pcm.c | 8 +++++++- > 1 file changed, 7 insertions(+), 1 deletion(-) > > diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c > index afc559866095..91a2436ce952 100644 > --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c > +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c > @@ -399,7 +399,13 @@ static int sst_media_hw_params(struct snd_pcm_substream *substream, > struct snd_pcm_hw_params *params, > struct snd_soc_dai *dai) > { > - snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params)); > + int ret; > + > + ret = > + snd_pcm_lib_malloc_pages(substream, > + params_buffer_bytes(params)); > + if (ret) > + return ret; > memset(substream->runtime->dma_area, 0, params_buffer_bytes(params)); > return 0; > }

2 1

[alsa-devel] [PATCH] ALSA: hda/tegra: clear pending irq handlers
by Sameer Pujar 01 Jan '19

01 Jan '19

Even after disabling interrupts on the module, it could be possible that irq handlers are still running. System hang is seen during suspend path. It was found that, there were pending writes on the HDA bus and clock was disabled by that time. Above mentioned issue is fixed by clearing any pending irq handlers before disabling clocks and returning from hda suspend. Suggested-by: Mohan Kumar <mkumard(a)nvidia.com> Suggested-by: Dara Ramesh <dramesh(a)nvidia.com> Signed-off-by: Sameer Pujar <spujar(a)nvidia.com> --- sound/pci/hda/hda_tegra.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/hda_tegra.c b/sound/pci/hda/hda_tegra.c index 83befd8..97a176d 100644 --- a/sound/pci/hda/hda_tegra.c +++ b/sound/pci/hda/hda_tegra.c @@ -234,10 +234,12 @@ static int hda_tegra_suspend(struct device *dev) struct snd_card *card = dev_get_drvdata(dev); struct azx *chip = card->private_data; struct hda_tegra *hda = container_of(chip, struct hda_tegra, chip); + struct hdac_bus *bus = azx_bus(chip); snd_power_change_state(card, SNDRV_CTL_POWER_D3hot); azx_stop_chip(chip); + synchronize_irq(bus->irq); azx_enter_link_reset(chip); hda_tegra_disable_clocks(hda); -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: hda/realtek: Enable the headset mic auto detection for ASUS laptops
by Jian-Hong Pan 01 Jan '19

01 Jan '19

The headset mic of ASUS laptops like UX533FD, UX433FN and UX333FA, whose CODEC is Realtek ALC294 has jack auto detection feature. This patch enables the feature. Signed-off-by: Daniel Drake <drake(a)endlessm.com> Signed-off-by: Jian-Hong Pan <jian-hong(a)endlessm.com> --- sound/pci/hda/patch_realtek.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a4f4a9dd488d..aee4cbd29d53 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6501,7 +6501,7 @@ static const struct hda_fixup alc269_fixups[] = { [ALC294_FIXUP_ASUS_HEADSET_MIC] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { - { 0x19, 0x01a1113c }, /* use as headset mic, without its own jack detect */ + { 0x19, 0x01a1103c }, /* use as headset mic */ { } }, .chained = true, -- 2.11.0

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[alsa-devel] [PATCH] ALSA: hda - Revert DSP detection on legacy HD-audio driver
by Takashi Iwai 01 Jan '19

01 Jan '19

This essentially reverts the commits c337104b1a16 ("ALSA: HD-Audio: SKL+: abort probe if DSP is present and Skylake driver selected") and d82b51c855a2 ("ALSA: HD-Audio: SKL+: force HDaudio legacy or SKL+ driver selection") for the path of legacy HD-audio controller (snd-hda-intel). The automatic DSP detection and skip of binding with the legacy driver caused regressions on several machines like Dell XPS13. They give the PCI class 0x40380 indicating the availability of DSP while they don't work with ASoC SKL driver (yet). As the support of ASoC driver for such devices isn't available, it's better to revert the whole DSP-detection-and-skip behavior of the legacy driver, so that we can get the old good driver working on such devices. The pci_binding option for ASoC SKL driver is still kept so that it can work without blacklisting. Fixes: c337104b1a16 ("ALSA: HD-Audio: SKL+: abort probe if DSP is present and Skylake driver selected") Reported-by: Linus Torvalds <torvalds(a)linux-foundation.org> Reported-by: Hans de Goede <hdegoede(a)redhat.com> Reported-by: Azat Khuzhin <dohardgopro(a)gmail.com> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hdaudio.h | 6 ---- sound/pci/hda/Kconfig | 62 ---------------------------------- sound/pci/hda/hda_controller.h | 2 +- sound/pci/hda/hda_intel.c | 54 ++++------------------------- sound/soc/intel/Kconfig | 6 ---- 5 files changed, 8 insertions(+), 122 deletions(-) diff --git a/include/sound/hdaudio.h b/include/sound/hdaudio.h index b4fa1c775251..f9b2b6330d27 100644 --- a/include/sound/hdaudio.h +++ b/include/sound/hdaudio.h @@ -98,12 +98,6 @@ enum { HDA_DEV_ASOC, }; -enum { - SND_SKL_PCI_BIND_AUTO, /* automatic selection based on pci class */ - SND_SKL_PCI_BIND_LEGACY,/* bind only with legacy driver */ - SND_SKL_PCI_BIND_ASOC /* bind only with ASoC driver */ -}; - /* direction */ enum { HDA_INPUT, HDA_OUTPUT diff --git a/sound/pci/hda/Kconfig b/sound/pci/hda/Kconfig index 0d38c006e182..4235907b7858 100644 --- a/sound/pci/hda/Kconfig +++ b/sound/pci/hda/Kconfig @@ -226,68 +226,6 @@ config SND_HDA_POWER_SAVE_DEFAULT The default time-out value in seconds for HD-audio automatic power-save mode. 0 means to disable the power-save mode. -if SND_HDA_INTEL - -# The options below should not be enabled by distributions or -# users. They are selected by Intel/Skylake or SOF drivers when they -# register for a PCI ID which is also handled by the HDAudio legacy -# driver. When this option is selected and the DSP is detected based on -# the PCI class/subclass/prog-if, the probe of the HDAudio legacy -# aborts. This mechanism removes the need for distributions to use -# blacklists. It can be bypassed with module parameters should the -# Intel/Skylake or SOF drivers fail to handle a specific platform. - -config SND_HDA_INTEL_DSP_DETECTION_SKL - bool - help - This option is selected by SOF or SST drivers, not users or distros. - It enables DSP detection based on PCI class information for - Skylake machines. - -config SND_HDA_INTEL_DSP_DETECTION_APL - bool - help - This option is selected by SOF or SST drivers, not users or distros. - It enables DSP detection based on PCI class information for - Broxton/ApolloLake machines - -config SND_HDA_INTEL_DSP_DETECTION_KBL - bool - help - This option is selected by SOF or SST drivers, not users or distros. - It enables DSP detection based on PCI class information for - KabyLake machines - -config SND_HDA_INTEL_DSP_DETECTION_GLK - bool - help - This option is selected by SOF or SST drivers, not users or distros. - It enables DSP detection based on PCI class information for - GeminiLake machines - -config SND_HDA_INTEL_DSP_DETECTION_CNL - bool - help - This option is selected by SOF or SST drivers, not users or distros. - It enables DSP detection based on PCI class information for - CannonLake machines - -config SND_HDA_INTEL_DSP_DETECTION_CFL - bool - help - This option is selected by SOF or SST drivers, not users or distros. - It enables DSP detection based on PCI class information for - CoffeeLake machines - -config SND_HDA_INTEL_DSP_DETECTION_ICL - bool - help - This option is selected by SOF or SST drivers, not users or distros. - It enables DSP detection based on PCI class information for - IceLake machines - -endif ## SND_HDA_INTEL - endif endmenu diff --git a/sound/pci/hda/hda_controller.h b/sound/pci/hda/hda_controller.h index e0c3fcbaa028..7185ed574b41 100644 --- a/sound/pci/hda/hda_controller.h +++ b/sound/pci/hda/hda_controller.h @@ -37,7 +37,7 @@ #else #define AZX_DCAPS_I915_COMPONENT 0 /* NOP */ #endif -#define AZX_DCAPS_INTEL_SHARED (1 << 14) /* shared with ASoC */ +/* 14 unused */ #define AZX_DCAPS_CTX_WORKAROUND (1 << 15) /* X-Fi workaround */ #define AZX_DCAPS_POSFIX_LPIB (1 << 16) /* Use LPIB as default */ /* 17 unused */ diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index e42cc2230977..e784130ea4e0 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -172,9 +172,6 @@ module_param_array(beep_mode, bool, NULL, 0444); MODULE_PARM_DESC(beep_mode, "Select HDA Beep registration mode " "(0=off, 1=on) (default=1)."); #endif -static int skl_pci_binding; -module_param_named(pci_binding, skl_pci_binding, int, 0444); -MODULE_PARM_DESC(pci_binding, "PCI binding (0=auto, 1=only legacy, 2=only asoc"); #ifdef CONFIG_PM static int param_set_xint(const char *val, const struct kernel_param *kp); @@ -360,7 +357,6 @@ enum { AZX_DCAPS_NO_64BIT |\ AZX_DCAPS_4K_BDLE_BOUNDARY | AZX_DCAPS_SNOOP_OFF) -#define AZX_DCAPS_INTEL_DSP_DETECTION(conf) (IS_ENABLED(CONFIG_SND_HDA_INTEL_DSP_DETECTION_##conf) ? AZX_DCAPS_INTEL_SHARED : 0) /* * vga_switcheroo support */ @@ -2052,28 +2048,6 @@ static int azx_probe(struct pci_dev *pci, bool schedule_probe; int err; - /* check if this driver can be used on SKL+ Intel platforms */ - if (pci_id->driver_data & AZX_DCAPS_INTEL_SHARED) { - switch (skl_pci_binding) { - case SND_SKL_PCI_BIND_AUTO: - if (pci->class != 0x040300) { - dev_info(&pci->dev, "The DSP is enabled on this platform, aborting probe\n"); - return -ENODEV; - } - dev_info(&pci->dev, "No DSP detected, continuing HDaudio legacy probe\n"); - break; - case SND_SKL_PCI_BIND_LEGACY: - dev_info(&pci->dev, "Module parameter forced binding with HDaudio legacy, bypassed detection logic\n"); - break; - case SND_SKL_PCI_BIND_ASOC: - dev_info(&pci->dev, "Module parameter forced binding with SKL+ ASoC driver, aborting probe\n"); - return -ENODEV; - default: - dev_err(&pci->dev, "invalid value for skl_pci_binding module parameter, ignored\n"); - break; - } - } - if (dev >= SNDRV_CARDS) return -ENODEV; if (!enable[dev]) { @@ -2380,48 +2354,34 @@ static const struct pci_device_id azx_ids[] = { .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE }, /* Sunrise Point-LP */ { PCI_DEVICE(0x8086, 0x9d70), - .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE | - AZX_DCAPS_INTEL_DSP_DETECTION(SKL) - }, + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE }, /* Kabylake */ { PCI_DEVICE(0x8086, 0xa171), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE }, /* Kabylake-LP */ { PCI_DEVICE(0x8086, 0x9d71), - .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE | - AZX_DCAPS_INTEL_DSP_DETECTION(KBL) - }, + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE }, /* Kabylake-H */ { PCI_DEVICE(0x8086, 0xa2f0), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE }, /* Coffelake */ { PCI_DEVICE(0x8086, 0xa348), - .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE | - AZX_DCAPS_INTEL_DSP_DETECTION(CFL) - }, + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Cannonlake */ { PCI_DEVICE(0x8086, 0x9dc8), - .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE | - AZX_DCAPS_INTEL_DSP_DETECTION(CNL) - }, + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Icelake */ { PCI_DEVICE(0x8086, 0x34c8), - .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE | - AZX_DCAPS_INTEL_DSP_DETECTION(ICL) - }, + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Broxton-P(Apollolake) */ { PCI_DEVICE(0x8086, 0x5a98), - .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_BROXTON | - AZX_DCAPS_INTEL_DSP_DETECTION(APL) - }, + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_BROXTON }, /* Broxton-T */ { PCI_DEVICE(0x8086, 0x1a98), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_BROXTON }, /* Gemini-Lake */ { PCI_DEVICE(0x8086, 0x3198), - .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_BROXTON | - AZX_DCAPS_INTEL_DSP_DETECTION(GLK) - }, + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_BROXTON }, /* Haswell */ { PCI_DEVICE(0x8086, 0x0a0c), .driver_data = AZX_DRIVER_HDMI | AZX_DCAPS_INTEL_HASWELL }, diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 2fd1b61e8331..99a62ba409df 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -188,12 +188,6 @@ config SND_SOC_INTEL_SKYLAKE_COMMON select SND_SOC_TOPOLOGY select SND_SOC_INTEL_SST select SND_SOC_HDAC_HDA if SND_SOC_INTEL_SKYLAKE_HDAUDIO_CODEC - select SND_HDA_INTEL_DSP_DETECTION_SKL if SND_SOC_INTEL_SKL - select SND_HDA_INTEL_DSP_DETECTION_APL if SND_SOC_INTEL_APL - select SND_HDA_INTEL_DSP_DETECTION_KBL if SND_SOC_INTEL_KBL - select SND_HDA_INTEL_DSP_DETECTION_GLK if SND_SOC_INTEL_GLK - select SND_HDA_INTEL_DSP_DETECTION_CNL if SND_SOC_INTEL_CNL - select SND_HDA_INTEL_DSP_DETECTION_CFL if SND_SOC_INTEL_CFL select SND_SOC_ACPI_INTEL_MATCH help If you have a Intel Skylake/Broxton/ApolloLake/KabyLake/ -- 2.19.2

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[alsa-devel] "ALSA: HD-Audio: SKL+: abort probe if DSP is present and Skylake driver selected" breaks sound, suspend on Apollo Lake
by Hans de Goede 31 Dec '18

31 Dec '18

Hi Pierre-Louis, Since I recently have been doing a bunch of i915 work I'm currently running drm-tip. This brought in this commit from next: https://cgit.freedesktop.org/drm-tip/commit/?id=c337104b1a16becc486fdc95d54… This causes the hda driver to not bind to the audio on my Apollo Lake based laptop, and the asoc driver errors out with: [ 19.176417] skl_hda_dsp_generic skl_hda_dsp_generic: Unsupported HDAudio/iDisp configuration found [ 19.176428] skl_hda_dsp_generic: probe of skl_hda_dsp_generic failed with error -22 After which I get no sound and worse, the machine hangs on suspend. It seems to never reach its fully suspended state (the power-led never starts to fade on/off as it normally does when suspended). I've been looking through the Kconfig option and it seems that currently selecting SND_SOC_INTEL_APL makes little sense since there are 0 machine drivers for this setup ? I guess I can fix this by unselecting SND_SOC_INTEL_SKYLAKE and then during the manual platform selection not selecting SND_SOC_INTEL_APL, but IMHO we need to put something in place where the hda driver actually checks if there is a machine driver and if there isn't one then we should keep using the hda driver on the machine in question. Regards, Hans p.s. Merry Christmas!

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