September 2015 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] SoC/Codecs/SGTL5000 - Non working MIC_BIAS Voltage settings
by gianluca 30 Sep '15

30 Sep '15

Hello, in attachment there is a small patch to ensure the correct register value for MIC BIAS Voltage. Best regards -- Eurek s.r.l. | Electronic Engineering | http://www.eurek.it via Celletta 8/B, 40026 Imola, Italy | Phone: +39-(0)542-609120 p.iva 00690621206 - c.f. 04020030377 | Fax: +39-(0)542-609212

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[alsa-devel] Found bugs in tlv320aic3x.c driver, how do I report them?
by Rick Mann 30 Sep '15

30 Sep '15

Hi. I came across some bugs in sound/soc/codecs/tlv320aic3x.c, wherein it writes non-zero values to reserved registers on the tlv320aic3104 (I found it in 4.1.4, but it's still in 4.3-rc3). I have an untested proposed patch. But I don't know anything about how to report it or offer it. What should I do? Thanks. For reference: $ git diff -- sound/soc/codecs/tlv320aic3x.c diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 51c4713..79c0ca0 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -1509,14 +1509,24 @@ static int aic3x_init(struct snd_soc_codec *codec) snd_soc_write(codec, PGAL_2_LLOPM_VOL, DEFAULT_VOL); snd_soc_write(codec, PGAR_2_RLOPM_VOL, DEFAULT_VOL); - /* Line2 to HP Bypass default volume, disconnect from Output Mixer */ - snd_soc_write(codec, LINE2L_2_HPLOUT_VOL, DEFAULT_VOL); - snd_soc_write(codec, LINE2R_2_HPROUT_VOL, DEFAULT_VOL); - snd_soc_write(codec, LINE2L_2_HPLCOM_VOL, DEFAULT_VOL); - snd_soc_write(codec, LINE2R_2_HPRCOM_VOL, DEFAULT_VOL); - /* Line2 Line Out default volume, disconnect from Output Mixer */ - snd_soc_write(codec, LINE2L_2_LLOPM_VOL, DEFAULT_VOL); - snd_soc_write(codec, LINE2R_2_RLOPM_VOL, DEFAULT_VOL); + if (aic3x->model == AIC3X_MODEL_3104) { + /* On tlv320aic3014, these registers are reserved and must be written 0 */ + snd_soc_write(codec, LINE2L_2_HPLOUT_VOL, 0); + snd_soc_write(codec, LINE2R_2_HPROUT_VOL, 0); + snd_soc_write(codec, LINE2L_2_HPLCOM_VOL, 0); + snd_soc_write(codec, LINE2R_2_HPRCOM_VOL, 0); + snd_soc_write(codec, LINE2L_2_LLOPM_VOL, 0); + snd_soc_write(codec, LINE2R_2_RLOPM_VOL, 0); + } else { + /* Line2 to HP Bypass default volume, disconnect from Output Mixer */ + snd_soc_write(codec, LINE2L_2_HPLOUT_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2R_2_HPROUT_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2L_2_HPLCOM_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2R_2_HPRCOM_VOL, DEFAULT_VOL); + /* Line2 Line Out default volume, disconnect from Output Mixer */ + snd_soc_write(codec, LINE2L_2_LLOPM_VOL, DEFAULT_VOL); + snd_soc_write(codec, LINE2R_2_RLOPM_VOL, DEFAULT_VOL); + } switch (aic3x->model) { case AIC3X_MODEL_3X: -- Rick Mann rmann(a)latencyzero.com

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[alsa-devel] [PATCH 0/2] ASoC: Add support for the Allwinner A10 codec
by Maxime Ripard 30 Sep '15

30 Sep '15

Hi everyone, This patch set adds the support for what Allwinner calls the codec on their SoCs. This codec is actually a combination of a codec and DAI, tied together in a single memory-mapped IP. It is completely standalone, and outputs directly the analog signal. While it supports both playback and capture, the capture is not implemented in this patch, and will be posted eventually as a separate one. This set, in order to be functional, has a dependency on the audio clocks patch set posted separately. However, it doesn't needs this to compile properly, so I guess it can be merged without really caring for the merging status of the clock patches. Let me know what you think, Maxime Emilio López (1): ASoC: sunxi: add support for the on-chip codec on early Allwinner SoCs Maxime Ripard (1): Documentation: bindings: Add the Allwinner A10 codec bindings .../devicetree/bindings/sound/sun4i-codec.txt | 33 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/sunxi/Kconfig | 11 + sound/soc/sunxi/Makefile | 2 + sound/soc/sunxi/sun4i-codec.c | 720 +++++++++++++++++++++ 6 files changed, 767 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/sun4i-codec.txt create mode 100644 sound/soc/sunxi/Kconfig create mode 100644 sound/soc/sunxi/Makefile create mode 100644 sound/soc/sunxi/sun4i-codec.c -- 2.5.1

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[alsa-devel] [PATCH 0/7 v3] firewire-digi00x: new driver for Digidesign 002/003 family
by Takashi Sakamoto 30 Sep '15

30 Sep '15

This patchset updates a part of my previous post: [alsa-devel] [PATCH 00/25 v2] ALSA: support AMDTP variants http://mailman.alsa-project.org/pipermail/alsa-devel/2015-August/096739.html This patchset adds support for Digidesign Digi 002/003 family without MIDI functionality. Although most part of AMDTP stream functionality is clear, a mechanism of clock synchronization with internal clock source is not clear (see the last patch). Takashi Sakamoto (7): ALSA: firewire-digi00x: add skeleton for Digi 002/003 family ALSA: firewire-digi00x: add data block processing layer ALSA: firewire-digi00x: add stream functionality ALSA: firewire-digi00x: add proc node to show clock status ALSA: firewire-digi00x: add PCM functionality ALSA: firewire-digi00x: add hwdep interface ALSA: firewire-digi00x: add support for asynchronous messaging include/uapi/sound/asound.h | 3 +- include/uapi/sound/firewire.h | 8 + sound/firewire/Kconfig | 15 + sound/firewire/Makefile | 1 + sound/firewire/digi00x/Makefile | 4 + sound/firewire/digi00x/amdtp-dot.c | 330 +++++++++++++++++++++ sound/firewire/digi00x/digi00x-hwdep.c | 200 +++++++++++++ sound/firewire/digi00x/digi00x-pcm.c | 373 ++++++++++++++++++++++++ sound/firewire/digi00x/digi00x-proc.c | 99 +++++++ sound/firewire/digi00x/digi00x-stream.c | 420 +++++++++++++++++++++++++++ sound/firewire/digi00x/digi00x-transaction.c | 81 ++++++ sound/firewire/digi00x/digi00x.c | 169 +++++++++++ sound/firewire/digi00x/digi00x.h | 146 ++++++++++ 13 files changed, 1848 insertions(+), 1 deletion(-) create mode 100644 sound/firewire/digi00x/Makefile create mode 100644 sound/firewire/digi00x/amdtp-dot.c create mode 100644 sound/firewire/digi00x/digi00x-hwdep.c create mode 100644 sound/firewire/digi00x/digi00x-pcm.c create mode 100644 sound/firewire/digi00x/digi00x-proc.c create mode 100644 sound/firewire/digi00x/digi00x-stream.c create mode 100644 sound/firewire/digi00x/digi00x-transaction.c create mode 100644 sound/firewire/digi00x/digi00x.c create mode 100644 sound/firewire/digi00x/digi00x.h -- 2.1.4

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[alsa-devel] [PATCH] drm/i915: Add missing const to audio_rate_need_prog()
by Takashi Iwai 30 Sep '15

30 Sep '15

The lack of const leads to a compile warning after merging i915 upstream tree: drivers/gpu/drm/i915/intel_audio.c:147:13: note: expected 'struct drm_display_mode *' but argument is of type 'const struct drm_display_mode *' Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- drivers/gpu/drm/i915/intel_audio.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/gpu/drm/i915/intel_audio.c b/drivers/gpu/drm/i915/intel_audio.c index 30f6859dcb36..dffd0b4c5f17 100644 --- a/drivers/gpu/drm/i915/intel_audio.c +++ b/drivers/gpu/drm/i915/intel_audio.c @@ -144,7 +144,7 @@ static uint32_t audio_config_setup_n_reg(int n, uint32_t val) /* check whether N/CTS/M need be set manually */ static bool audio_rate_need_prog(struct intel_crtc *crtc, - struct drm_display_mode *mode) + const struct drm_display_mode *mode) { if (((mode->clock == TMDS_297M) || (mode->clock == TMDS_296M)) && -- 2.5.3

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[alsa-devel] [PATCH RFC 0/7] ASoC: tlv320aic26: Add device tree support and bug fixes
by Cormier, Jonathan 30 Sep '15

30 Sep '15

Hi, This is my first patch so I'm including the RFC to help identify if I've missed something. The following patches were created to get the tlv320aic26 working on our AM335x SoM. This patch series relies on commit 93d0ad8f374c "ASoC: tlv320aic26: Convert to params_width()" for proper operation. Thanks -Jonathan Cormier, Jonathan (7): ASoC: tlv320aic26: Add device tree binding ASoC: tlv320aic26: Fix module autoload ASoC: tlv320aic26: Make SND_SOC_TLV320AIC26 user-visible ASoC: tlv320aic26: Fix regmap by setting reg_defaults and reg_stride ASoC: tlv320aic26: hw_params was unintentionally clearing AIC26 master mode ASoC: tlv320aic26: Add support for DSP_B data format ASoC: tlv320aic26: Change Capture Mute to Capture Switch to match alsa documentation. .../devicetree/bindings/sound/tlv320aic26.txt | 65 ++++++++++++++++++++++ sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/tlv320aic26.c | 60 ++++++++++++++++++-- 3 files changed, 120 insertions(+), 7 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic26.txt -- 1.9.1

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[alsa-devel] speaker-test error using sgtl5000 and imx27
by Alexandre Schnegg 30 Sep '15

30 Sep '15

Hello, I would like to use the Freescale SGTl5000 audio codec with the Freescale imx27 processor. Unfortunately, I get this error when I try to run < speaker-test >: # speaker-test speaker-test 1.0.28 Playback device is default Stream parameters are 48000Hz, S16_LE, 1 channels Using 16 octaves of pink noise Rate set to 48000Hz (requested 48000Hz) Buffer size range from 128 to 32768 Period size range from 64 to 16384 Using max buffer size 32768 Periods = 4 was set period_size = 8192 was set buffer_size = 32768 0 - Front Left Write error: -5,Input/output error xrun_recovery failed: -5,Input/output error Transfer failed: Input/output error The codec is connected to SSI2 pins and I would like to use the SSI0 internal port in asynchronous mode (4 wires). The command interface is I2C1. The codec seems to work because it generates both bit and frame clocks (the codec is the master) but no data are coming from the processor. I'm using kernel 4.1.7 but I have got the same error with versions 3.19.8, 4.0.5 and 4.0.9. I have also tried the < imx-sgtl5000 > driver and I get the same error. I can use < alsamixer > with no issue. Here are my device tree bindings: sound { compatible = "simple-audio-card"; simple-audio-card,name = "sgtl5000-audio"; simple-audio-card,format = "i2s"; simple-audio-card,bitclock-master = <&dailink0_master>; simple-audio-card,frame-master = <&dailink0_master>; simple-audio-card,widgets = "Microphone", "Mic Jack", "Headphone", "Headphone Jack"; simple-audio-card,routing = "MIC_IN", "Mic Jack", "Mic Jack", "Mic Bias", "Headphone Jack", "HP_OUT"; simple-audio-card,cpu { sound-dai = <&ssi1 0>; }; dailink0_master: simple-audio-card,codec { sound-dai = <&sgtl5000>; clocks = <&mclk>; }; }; mclk: asvmpc { #clock-cells = <0>; compatible = "fixed-clock"; clock-frequency = <12288000>; }; &ssi1 { #sound-dai-cells = <1>; status = "okay"; fsl,audmux-port = <MX27_AUDMUX_PPCR2_SSI_PINS_2>; fsl,port-config = < (IMX_AUDMUX_V1_PCR_SYN | IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_HPCR1_SSI0)) >; }; &audmux { pinctrl-names = "default"; pinctrl-0 = <&pinctrl_ssi2>; status = "okay"; fsl,audmux-port = <MX27_AUDMUX_HPCR1_SSI0>; fsl,port-config = < (IMX_AUDMUX_V1_PCR_SYN | IMX_AUDMUX_V1_PCR_TFCSEL(MX27_AUDMUX_PPCR2_SSI_PINS_2) | IMX_AUDMUX_V1_PCR_TFSDIR | IMX_AUDMUX_V1_PCR_TCLKDIR | IMX_AUDMUX_V1_PCR_RXDSEL(MX27_AUDMUX_PPCR2_SSI_PINS_2)) >; }; &i2c1 { clock-frequency = <100000>; pinctrl-names = "default"; pinctrl-0 = <&pinctrl_i2c1>; status = "okay"; sgtl5000: sgtl5000@0a { #sound-dai-cells = <0>; compatible = "fsl,sgtl5000"; reg = <0x0a>; clocks = <&mclk>; micbias-resistor-k-ohms = <2>; micbias-voltage-m-volts = <2250>; VDDA-supply = <&reg_2p8v>; VDDIO-supply = <&reg_2p8v>; }; }; pinctrl_ssi2: ssi2grp { fsl,pins = < MX27_PAD_SSI2_FS__SSI2_FS 0x0 MX27_PAD_SSI2_RXDAT__SSI2_RXDAT 0x0 MX27_PAD_SSI2_TXDAT__SSI2_TXDAT 0x0 MX27_PAD_SSI2_CLK__SSI2_CLK 0x0 >; }; And the modules I have loaded before running < speaker test > : # modprobe snd-soc-sgtl5000 [ 23.274351] sgtl5000 0-000a: sgtl5000 revision 0x11 # modprobe snd-soc-simple-card # modprobe snd-soc-fsl-ssi [ 79.752288] sgtl5000 0-000a: Using internal LDO instead of VDDD # [ 79.853711] asoc-simple-card sound: sgtl5000 <-> 10010000.ssi mapping ok # modprobe snd-soc-imx-audmux Does anybody know what is the mistake? Thank you for your help. Best regards. Alexandre Schnegg

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[alsa-devel] [PATCH 1/2] ASoC: sgtl5000: fix wrong register MIC_BIAS_VOLTAGE setup on probe
by Gianluca Renzi 29 Sep '15

29 Sep '15

Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Fabio Estevam <fabio.estevam(a)freescale.com> Cc: alsa-devel(a)alsa-project.org Cc: linux-kernel(a)vger.kernel.org Signed-off-by: Gianluca Renzi <gianlucarenzi(a)eurekelettronica.it> --- sound/soc/codecs/sgtl5000.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) --- a/sound/soc/codecs/sgtl5000.c 2015-09-25 15:24:10.597051191 +0200 +++ b/sound/soc/codecs/sgtl5000.c 2015-09-25 21:26:07.102020194 +0200 @@ -1376,8 +1376,8 @@ static int sgtl5000_probe(struct snd_soc sgtl5000->micbias_resistor << SGTL5000_BIAS_R_SHIFT); snd_soc_update_bits(codec, SGTL5000_CHIP_MIC_CTRL, - SGTL5000_BIAS_R_MASK, - sgtl5000->micbias_voltage << SGTL5000_BIAS_R_SHIFT); + SGTL5000_BIAS_VOLT_MASK, + sgtl5000->micbias_voltage << SGTL5000_BIAS_VOLT_SHIFT); /* * disable DAP * TODO:

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[alsa-devel] [PATCH alsa-utils] alsactl: Manage both save and restore in a single unit
by Takashi Iwai 29 Sep '15

29 Sep '15

With RemainAfterExit=true, we can manage both save and restore of the card state in a single unit file. This will fix also the case where systemd reloads the service; with two individual units, it will restore the previous state before saving, and may lead to inconsistent state suddenly. Also fix alsa-state.service as well to make both start and stop working in a simpler way. Bugzilla: https://bugzilla.novell.com/show_bug.cgi?id=929619 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- alsactl/Makefile.am | 14 ++------------ alsactl/alsa-restore.service.in | 7 +++---- alsactl/alsa-state.service.in | 3 --- alsactl/alsa-store.service.in | 16 ---------------- 4 files changed, 5 insertions(+), 35 deletions(-) delete mode 100644 alsactl/alsa-store.service.in diff --git a/alsactl/Makefile.am b/alsactl/Makefile.am index 47f06e906ced..cac8094687de 100644 --- a/alsactl/Makefile.am +++ b/alsactl/Makefile.am @@ -25,20 +25,15 @@ if HAVE_SYSTEMD systemdsystemunit_DATA = \ alsa-state.service \ - alsa-restore.service \ - alsa-store.service + alsa-restore.service install-data-hook: $(MKDIR_P) -m 0755 \ - $(DESTDIR)$(systemdsystemunitdir)/basic.target.wants \ - $(DESTDIR)$(systemdsystemunitdir)/shutdown.target.wants + $(DESTDIR)$(systemdsystemunitdir)/basic.target.wants ( cd $(DESTDIR)$(systemdsystemunitdir)/basic.target.wants && \ rm -f alsa-state.service alsa-restore.service && \ $(LN_S) ../alsa-state.service alsa-state.service && \ $(LN_S) ../alsa-restore.service alsa-restore.service) - ( cd $(DESTDIR)$(systemdsystemunitdir)/shutdown.target.wants && \ - rm -f alsa-store.service && \ - $(LN_S) ../alsa-store.service alsa-store.service ) endif @@ -54,22 +49,17 @@ alsa-state.service: alsa-state.service.in alsa-restore.service: alsa-restore.service.in $(edit) -alsa-store.service: alsa-store.service.in - $(edit) - 90-alsa-restore.rules: 90-alsa-restore.rules.in $(edit) EXTRA_DIST += \ alsa-state.service.in \ alsa-restore.service.in \ - alsa-store.service.in \ 90-alsa-restore.rules.in CLEANFILES = \ alsa-state.service \ alsa-restore.service \ - alsa-store.service \ 90-alsa-restore.rules %.7: %.xml diff --git a/alsactl/alsa-restore.service.in b/alsactl/alsa-restore.service.in index d1a74d637457..2884098c26f0 100644 --- a/alsactl/alsa-restore.service.in +++ b/alsactl/alsa-restore.service.in @@ -4,15 +4,14 @@ # [Unit] -Description=Restore Sound Card State +Description=Save/Restore Sound Card State ConditionPathExists=!@daemonswitch@ ConditionPathExistsGlob=/dev/snd/control* -DefaultDependencies=no After=alsa-state.service -Before=shutdown.target -Conflicts=shutdown.target [Service] Type=oneshot +RemainAfterExit=true ExecStart=-@sbindir@/alsactl restore +ExecStop=-@sbindir@/alsactl store StandardOutput=syslog diff --git a/alsactl/alsa-state.service.in b/alsactl/alsa-state.service.in index 56c12c104565..f1321d63a0ca 100644 --- a/alsactl/alsa-state.service.in +++ b/alsactl/alsa-state.service.in @@ -6,10 +6,7 @@ [Unit] Description=Manage Sound Card State (restore and store) ConditionPathExists=@daemonswitch@ -DefaultDependencies=no After=sysinit.target -Before=shutdown.target -Conflicts=shutdown.target [Service] Type=simple diff --git a/alsactl/alsa-store.service.in b/alsactl/alsa-store.service.in deleted file mode 100644 index c89cfff04f35..000000000000 --- a/alsactl/alsa-store.service.in +++ /dev/null @@ -1,16 +0,0 @@ -# -# Note that two different ALSA card state management schemes exist and they -# can be switched using a file exist check - /etc/alsa/state-daemon.conf . -# - -[Unit] -Description=Store Sound Card State -ConditionPathExists=!@daemonswitch@ -ConditionPathExistsGlob=/dev/snd/control* -DefaultDependencies=no -Before=shutdown.target - -[Service] -Type=oneshot -ExecStart=@sbindir@/alsactl store -StandardOutput=syslog -- 2.5.3

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[alsa-devel] [PATCH v3 0/4] ASoC: Add support for DA7219 audio codec
by Adam Thomson 29 Sep '15

29 Sep '15

This patch set adds support for the DA7219 audio codec with built-in advanced accessory detection functionality. Patch set includes codec driver, associated DT bindings documentation and MAINTAINERS file updates to cover new bindings. This patch set is baselined against the v4.3-rc3 kernel version. Changes in v3: - Actually set clk_src in dai_sysclk() function of codec, so future calls can correctly identify if work is to be done or not. Changes in v2: - Add SOC_DOUBLE_R_EXT support to ASoC core. - Use of upper-case naming for controls, where missing. - Use of Playback/Capture Digital naming for certain ADC/DAC controls. - CP mchange and vol_thresh DT bindings removed, now Kcontrols - Kcontrols and DAPM now locked as part of hptest work, and specific controls affected have had locking introduced. - Timeout for btn_det work when checking micbias status. - Added comments to describe reasoning for work use with hptest, btn_det. - Added comments to tonegen_freq_put/get to clarify endian conversion, and code now uses regmap_raw_read/write calls instead of bulk. - DT parsing code now prints dev_warn() for invalid settings. - SOC_VALUE_ENUM_SINGLE now used for hpf_mode handling, instead of customer get/put handlers. - DTMF key control now a SOC_ENUM to correctly describe supported keys, including non-numeric. - MIXIN_L_MIX_EN bit set as part of DAPM now (SUPPLY), rather than in probe. - Fix dai_sysclk() function to correctly check for no change, using clk_id as well as freq. - Error checking added for clk_set_rate() call. - Improved error checking for devm_clk_get() call. - Removed explicit use of device_may_wakeup() and enable/disable_irq_wake(). Now taken care of by I2C core so DT bindings updated accordingly to document this. - hptest work now uses cache to keep state, instead of reading registers individually. Cache is bypassed, then when test executed and complete, regcache_sync_region() used to restore settings to pre-test values. - Regulator consumer framework usage added, in particular to determine VDDIO level in use so device can be configured accordingly. DT documents updated to reflect this. Adam Thomson (4): ASoC: Add SOC_DOUBLE_R_EXT ASoC: codecs: Add da7219 codec driver ASoC: da7219: Add bindings documentation for DA7219 audio codec MAINTAINERS: da7219: Add entry to cover DA7219 bindings document Documentation/devicetree/bindings/sound/da7219.txt | 106 ++ MAINTAINERS | 1 + include/sound/da7219-aad.h | 99 + include/sound/da7219.h | 55 + include/sound/soc.h | 7 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/da7219-aad.c | 823 +++++++++ sound/soc/codecs/da7219-aad.h | 212 +++ sound/soc/codecs/da7219.c | 1940 ++++++++++++++++++++ sound/soc/codecs/da7219.h | 820 +++++++++ 11 files changed, 4069 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/da7219.txt create mode 100644 include/sound/da7219-aad.h create mode 100644 include/sound/da7219.h create mode 100644 sound/soc/codecs/da7219-aad.c create mode 100644 sound/soc/codecs/da7219-aad.h create mode 100644 sound/soc/codecs/da7219.c create mode 100644 sound/soc/codecs/da7219.h -- 1.9.3

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