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

[alsa-devel] [PATCH] drm: bridge/dw_hdmi: add audio sample channel status setting
by Cheng-Yi Chiang 04 Sep '19

04 Sep '19

From: Yakir Yang <ykk(a)rock-chips.com> When transmitting IEC60985 linear PCM audio, we configure the Audio Sample Channel Status information of all the channel status bits in the IEC60958 frame. Refer to 60958-3 page 10 for frequency, original frequency, and wordlength setting. This fix the issue that audio does not come out on some monitors (e.g. LG 22CV241) Signed-off-by: Yakir Yang <ykk(a)rock-chips.com> Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- drivers/gpu/drm/bridge/synopsys/dw-hdmi.c | 59 +++++++++++++++++++++++ drivers/gpu/drm/bridge/synopsys/dw-hdmi.h | 20 ++++++++ 2 files changed, 79 insertions(+) diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c index bd65d0479683..34d46e25d610 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.c @@ -582,6 +582,63 @@ static unsigned int hdmi_compute_n(unsigned int freq, unsigned long pixel_clk) return n; } +static void hdmi_set_schnl(struct dw_hdmi *hdmi) +{ + u8 aud_schnl_samplerate; + u8 aud_schnl_8; + + /* These registers are on RK3288 using version 2.0a. */ + if (hdmi->version != 0x200a) + return; + + switch (hdmi->sample_rate) { + case 32000: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_32K; + break; + case 44100: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_44K1; + break; + case 48000: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_48K; + break; + case 88200: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_88K2; + break; + case 96000: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_96K; + break; + case 176400: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_176K4; + break; + case 192000: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_192K; + break; + case 768000: + aud_schnl_samplerate = HDMI_FC_AUDSCHNLS7_SMPRATE_768K; + break; + default: + dev_warn(hdmi->dev, "Unsupported audio sample rate (%u)\n", + hdmi->sample_rate); + return; + } + + /* set channel status register */ + hdmi_modb(hdmi, aud_schnl_samplerate, HDMI_FC_AUDSCHNLS7_SMPRATE_MASK, + HDMI_FC_AUDSCHNLS7); + + /* + * Set original frequency to be the same as frequency. + * Use one-complement value as stated in IEC60958-3 page 13. + */ + aud_schnl_8 = (~aud_schnl_samplerate) << + HDMI_FC_AUDSCHNLS8_ORIGSAMPFREQ_OFFSET; + + /* This means word length is 16 bit. Refer to IEC60958-3 page 12. */ + aud_schnl_8 |= 2 << HDMI_FC_AUDSCHNLS8_WORDLEGNTH_OFFSET; + + hdmi_writeb(hdmi, aud_schnl_8, HDMI_FC_AUDSCHNLS8); +} + static void hdmi_set_clk_regenerator(struct dw_hdmi *hdmi, unsigned long pixel_clk, unsigned int sample_rate) { @@ -620,6 +677,8 @@ static void hdmi_set_clk_regenerator(struct dw_hdmi *hdmi, hdmi->audio_cts = cts; hdmi_set_cts_n(hdmi, cts, hdmi->audio_enable ? n : 0); spin_unlock_irq(&hdmi->audio_lock); + + hdmi_set_schnl(hdmi); } static void hdmi_init_clk_regenerator(struct dw_hdmi *hdmi) diff --git a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h index 6988f12d89d9..619ebc1c8354 100644 --- a/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h +++ b/drivers/gpu/drm/bridge/synopsys/dw-hdmi.h @@ -158,6 +158,17 @@ #define HDMI_FC_SPDDEVICEINF 0x1062 #define HDMI_FC_AUDSCONF 0x1063 #define HDMI_FC_AUDSSTAT 0x1064 +#define HDMI_FC_AUDSV 0x1065 +#define HDMI_FC_AUDSU 0x1066 +#define HDMI_FC_AUDSCHNLS0 0x1067 +#define HDMI_FC_AUDSCHNLS1 0x1068 +#define HDMI_FC_AUDSCHNLS2 0x1069 +#define HDMI_FC_AUDSCHNLS3 0x106a +#define HDMI_FC_AUDSCHNLS4 0x106b +#define HDMI_FC_AUDSCHNLS5 0x106c +#define HDMI_FC_AUDSCHNLS6 0x106d +#define HDMI_FC_AUDSCHNLS7 0x106e +#define HDMI_FC_AUDSCHNLS8 0x106f #define HDMI_FC_DATACH0FILL 0x1070 #define HDMI_FC_DATACH1FILL 0x1071 #define HDMI_FC_DATACH2FILL 0x1072 @@ -706,6 +717,15 @@ enum { /* HDMI_FC_AUDSCHNLS7 field values */ HDMI_FC_AUDSCHNLS7_ACCURACY_OFFSET = 4, HDMI_FC_AUDSCHNLS7_ACCURACY_MASK = 0x30, + HDMI_FC_AUDSCHNLS7_SMPRATE_MASK = 0x0f, + HDMI_FC_AUDSCHNLS7_SMPRATE_192K = 0xe, + HDMI_FC_AUDSCHNLS7_SMPRATE_176K4 = 0xc, + HDMI_FC_AUDSCHNLS7_SMPRATE_96K = 0xa, + HDMI_FC_AUDSCHNLS7_SMPRATE_768K = 0x9, + HDMI_FC_AUDSCHNLS7_SMPRATE_88K2 = 0x8, + HDMI_FC_AUDSCHNLS7_SMPRATE_32K = 0x3, + HDMI_FC_AUDSCHNLS7_SMPRATE_48K = 0x2, + HDMI_FC_AUDSCHNLS7_SMPRATE_44K1 = 0x0, /* HDMI_FC_AUDSCHNLS8 field values */ HDMI_FC_AUDSCHNLS8_ORIGSAMPFREQ_MASK = 0xf0, -- 2.23.0.187.g17f5b7556c-goog

5 13

[alsa-devel] [PATCH] soundwire: bus: set initial value to port_status
by Bard liao 04 Sep '19

04 Sep '19

From: Bard Liao <yung-chuan.liao(a)linux.intel.com> port_status[port_num] are assigned for each port_num in some if conditions. So some of the port_status may not be initialized. Signed-off-by: Bard Liao <yung-chuan.liao(a)linux.intel.com> --- drivers/soundwire/bus.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/soundwire/bus.c b/drivers/soundwire/bus.c index f6a1e4b4813d..33f41b3e642e 100644 --- a/drivers/soundwire/bus.c +++ b/drivers/soundwire/bus.c @@ -835,7 +835,7 @@ static int sdw_handle_port_interrupt(struct sdw_slave *slave, static int sdw_handle_slave_alerts(struct sdw_slave *slave) { struct sdw_slave_intr_status slave_intr; - u8 clear = 0, bit, port_status[15]; + u8 clear = 0, bit, port_status[15] = {0}; int port_num, stat, ret, count = 0; unsigned long port; bool slave_notify = false; -- 2.17.1

3 2

[alsa-devel] some lock needed for flush_delayed_work ?
by Kuninori Morimoto 04 Sep '19

04 Sep '19

Hi ALSA ML I got below error when I unbind/bind CPU DAI. It seems we need to have some kind of lock ? I think => message came from linux/kernel/locking/lockdep.c :: assign_lock_key ... [ 6.392640] asoc-audio-graph-card sound: ASoC: CPU DAI not registered => [ 6.399430] INFO: trying to register non-static key. => [ 6.404393] the code is fine but needs lockdep annotation. => [ 6.409875] turning off the locking correctness validator. [ 6.415362] CPU: 2 PID: 193 Comm: test-bind-unbin Not tainted 5.3.0-rc1+ #625 [ 6.422495] Hardware name: Renesas H3ULCB Kingfisher board based on r8a7795 ES2.0+ (DT) [ 6.430497] Call trace: [ 6.432949] dump_backtrace+0x0/0x140 [ 6.436609] show_stack+0x24/0x30 [ 6.439927] dump_stack+0xc4/0x10c [ 6.443328] register_lock_class+0x4e0/0x618 [ 6.447594] __lock_acquire+0x7c/0x1de8 [ 6.451426] lock_acquire+0xdc/0x1e0 [ 6.455000] del_timer_sync+0x60/0xe8 [ 6.458660] flush_delayed_work+0x2c/0x70 [ 6.462671] soc_cleanup_card_resources+0x50/0x348 [ 6.467459] snd_soc_bind_card+0x1e4/0x990 [ 6.471551] snd_soc_add_component+0x248/0x310 [ 6.475992] snd_soc_register_component+0x54/0x70 [ 6.480696] devm_snd_soc_register_component+0x60/0xb8 [ 6.485833] rsnd_probe+0x164/0x448 [ 6.489321] platform_drv_probe+0x58/0xa8 [ 6.493330] really_probe+0xec/0x2a0 [ 6.496902] driver_probe_device+0x5c/0xf0 [ 6.500996] device_driver_attach+0x74/0x80 [ 6.505177] bind_store+0xdc/0x120 [ 6.508575] drv_attr_store+0x40/0x58 [ 6.512237] sysfs_kf_write+0x50/0x78 [ 6.515896] kernfs_fop_write+0xf0/0x1e8 [ 6.519819] __vfs_write+0x48/0x90 [ 6.523218] vfs_write+0xac/0x1b8 [ 6.526529] ksys_write+0x74/0xf8 [ 6.529840] __arm64_sys_write+0x24/0x30 [ 6.533762] el0_svc_common.constprop.0+0x98/0x170 [ 6.538550] el0_svc_compat_handler+0x2c/0x38 [ 6.542904] el0_svc_compat+0x8/0x10 Thank you for your help !! Best regards --- Kuninori Morimoto

2 2

[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.

1 0

[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

2 1

[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

2 3

[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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