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[alsa-devel] [PATCH 0/3] Add support for Wolfson Micro WM8998 codec
by Richard Fitzgerald 28 Sep '15

28 Sep '15

This set of patches is based on broonie/regulator.git/for-next and broonie/sound.git/for-next branches. Richard Fitzgerald (3): regulator: arizona: add support for WM8998 and WM1814 ASoC: arizona: add defines for single-input gain control ASoC: wm8998: Initial WM8998 codec driver drivers/regulator/arizona-ldo1.c | 2 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/arizona.c | 33 + sound/soc/codecs/arizona.h | 12 +- sound/soc/codecs/wm8998.c | 1429 ++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/wm8998.h | 23 + 7 files changed, 1506 insertions(+), 1 deletion(-) create mode 100644 sound/soc/codecs/wm8998.c create mode 100644 sound/soc/codecs/wm8998.h -- 1.9.1

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[alsa-devel] [PATCH v2 1/1] ASoC: dwc: fix dma stop transferring issue
by yitian 28 Sep '15

28 Sep '15

Designware I2S uses tx empty and rx available signals as the DMA handshaking signals. during music playing, if XRUN occurs, i2s_stop() function will be executed and both tx and rx irq are masked, when music contintes to be played, i2s_start() is executed but both tx and rx irq are not unmasked which cause I2S stop sending DMA handshaking signal to DMA controller, and it finally cause music playing will be stopped once XRUN occurs for the first time. Signed-off-by: Yitian Bu <yitian.bu(a)tangramtek.com> --- changes in V2: - add definition for i and irq --- sound/soc/dwc/designware_i2s.c | 15 ++++++++++++--- 1 file changed, 12 insertions(+), 3 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index a3e97b4..76b2e19 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -141,13 +141,22 @@ static inline void i2s_clear_irqs(struct dw_i2s_dev *dev, u32 stream) static void i2s_start(struct dw_i2s_dev *dev, struct snd_pcm_substream *substream) { - + u32 i, irq; i2s_write_reg(dev->i2s_base, IER, 1); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + for (i = 0; i < 4; i++) { + irq = i2s_read_reg(dev->i2s_base, IMR(i)); + i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x30); + } i2s_write_reg(dev->i2s_base, ITER, 1); - else + } else { + for (i = 0; i < 4; i++) { + irq = i2s_read_reg(dev->i2s_base, IMR(i)); + i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x03); + } i2s_write_reg(dev->i2s_base, IRER, 1); + } i2s_write_reg(dev->i2s_base, CER, 1); } -- 1.7.12.4

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[alsa-devel] [PATCH 1/1] ASoC: dwc: fix dma stop transferring issue
by yitian 28 Sep '15

28 Sep '15

Designware I2S uses tx empty and rx available signals as the DMA handshaking signals. during music playing, if XRUN occurs, i2s_stop() function will be executed and both tx and rx irq are masked, when music contintes to be played, i2s_start() is executed but both tx and rx irq are not unmasked which cause I2S stop sending DMA handshaking signal to DMA controller, and it finally cause music playing will be stopped once XRUN occurs for the first time. Signed-off-by: Yitian Bu <yitian.bu(a)tangramtek.com> --- sound/soc/dwc/designware_i2s.c | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/sound/soc/dwc/designware_i2s.c b/sound/soc/dwc/designware_i2s.c index a3e97b4..4d3e1d3 100644 --- a/sound/soc/dwc/designware_i2s.c +++ b/sound/soc/dwc/designware_i2s.c @@ -144,10 +144,19 @@ static void i2s_start(struct dw_i2s_dev *dev, i2s_write_reg(dev->i2s_base, IER, 1); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + for (i = 0; i < 4; i++) { + irq = i2s_read_reg(dev->i2s_base, IMR(i)); + i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x30); + } i2s_write_reg(dev->i2s_base, ITER, 1); - else + } else { + for (i = 0; i < 4; i++) { + irq = i2s_read_reg(dev->i2s_base, IMR(i)); + i2s_write_reg(dev->i2s_base, IMR(i), irq & ~0x03); + } i2s_write_reg(dev->i2s_base, IRER, 1); + } i2s_write_reg(dev->i2s_base, CER, 1); } -- 1.7.12.4

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[alsa-devel] [PATCH v2 0/2] ASoC: atmel-classd: add driver for Atmel CLASSD
by Songjun Wu 28 Sep '15

28 Sep '15

The Audio Class D Amplifier driver includes two parts. 1) Driver code to implement the Audio Class D Amplifier function. 2) Device tree binding document, it describes how to add the Audio Class D Amplifier in device tree. Changes in v2: - Change the "Mono", "Swap" and "Deemphasis" controls to "Mono Switch", "Swap Switch" and "Deemphasis Switch". - Merge "EQ Bass", "EQ Medium" and "EQ Treble" controls into one "EQ" control. - Change the "single-ended" and "differential" to "Single ended", "Differential". - Merge separate left and right controls into one single stereo control. - Add a warning if the user trys to specify an invalid value in function "atmel_classd_codec_probe". - Use the "dev_get_regmap()" in function "atmel_classd_codec_get_remap". - Check for errors when invoking the "clk_prepare_enable". - Remove the "dev_info" in function "atmel_classd_probe". - Add some code to create a sound card in function "atmel_classd_probe". - Remove the DT node "Sound" and the related code. - Delete the device node "Sound". Songjun Wu (2): ASoC: atmel-classd: add the Audio Class D Amplifier ASoC: atmel-classd: DT binding for Class D audio amplifier driver .../devicetree/bindings/sound/atmel-classd.txt | 52 ++ sound/soc/atmel/Kconfig | 9 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel-classd.c | 680 ++++++++++++++++++++ sound/soc/atmel/atmel-classd.h | 120 ++++ 5 files changed, 863 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/atmel-classd.txt create mode 100644 sound/soc/atmel/atmel-classd.c create mode 100644 sound/soc/atmel/atmel-classd.h -- 1.7.9.5

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[alsa-devel] [PATCH] mfd: arizona: Call the runtime PM function if the state is runtime resumed
by Inha Song 27 Sep '15

27 Sep '15

This patch call runtiem PM function manually if the state is runtime resumed before enter suspend. This case has occurred when wake-up by the resume after entering to suspend during the playback. $ aplay -> runtime_resume() -> suspend() -> resume() In this case, Can occurred an error when enter suspend. -- [ 82.559234] PM: suspend of devices complete after 57.252 msecs [ 82.567978] arizona spi1.0: Failed to read IRQ status: -108 [ 82.567989] arizona spi1.0: Failed to read main IRQ status: -108 [ 82.568027] arizona spi1.0: Failed to read IRQ status: -108 [ 82.568036] arizona spi1.0: Failed to read main IRQ status: -108 .... So, we should call runtime_suspend() manually when enter suspend if the state is runtime resume. Also should call runtime_resume() when wake-up by resume if the state was runtime resume before entering suspend. Signed-off-by: Inha Song <ideal.song(a)samsung.com> --- drivers/mfd/arizona-core.c | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/drivers/mfd/arizona-core.c b/drivers/mfd/arizona-core.c index 908c69b..de55706 100644 --- a/drivers/mfd/arizona-core.c +++ b/drivers/mfd/arizona-core.c @@ -683,6 +683,10 @@ static int arizona_suspend(struct device *dev) struct arizona *arizona = dev_get_drvdata(dev); dev_dbg(arizona->dev, "Suspend, disabling IRQ\n"); + + if (!pm_runtime_status_suspended(dev)) + arizona_runtime_suspend(dev); + disable_irq(arizona->irq); return 0; @@ -713,6 +717,10 @@ static int arizona_resume(struct device *dev) struct arizona *arizona = dev_get_drvdata(dev); dev_dbg(arizona->dev, "Late resume, reenabling IRQ\n"); + + if (!pm_runtime_status_suspended(dev)) + arizona_runtime_resume(dev); + enable_irq(arizona->irq); return 0; -- 2.0.0.390.gcb682f8

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[alsa-devel] Baytrail based HDMI Stick Orbsmart AW-01: No sound
by Michael Leun 26 Sep '15

26 Sep '15

Hi, I've an baytrail / Z3735F based HDMI Stick "Orbsmart AW-01", but sound does not work. I'm a little lost finding out what is missing / required to make it work. I compiled an 4.3-rc2 vanilla kernel with hopefully each and everything turned on relating soc/sound. In dmesg I get: [...] [ 4.252522] sst-acpi 80860F28:00: No matching ASoC machine driver found [ 5.863177] intel_sst_acpi 80860F28:00: No matching machine driver found [...] It looks like there is rt5640 based sound recognized: [ml@orb ~]$ find /sys -name '*5640*' -print /sys/bus/i2c/drivers/rt5640 /sys/bus/acpi/devices/10EC5640:00 /sys/bus/acpi/devices/10EC5640:01 /sys/devices/LNXSYSTM:00/LNXSYBUS:00/80860F41:01/10EC5640:00 /sys/devices/LNXSYSTM:00/LNXSYBUS:00/80860F41:01/10EC5640:01 find: "/sys/kernel/debug": Keine Berechtigung /sys/module/i2c_core/holders/snd_soc_rt5640 /sys/module/snd_soc_rl6231/holders/snd_soc_rt5640 /sys/module/snd_soc_rt5640 /sys/module/snd_soc_rt5640/drivers/i2c:rt5640 /sys/module/snd_soc_core/holders/snd_soc_rt5640 /sys/module/regmap_i2c/holders/snd_soc_rt5640 /sys/module/snd_pcm/holders/snd_soc_rt5640 [ml@orb ~]$ This are the sound related modules loaded: [ml@orb ~]$ lsmod|grep snd snd_intel_sst_acpi 16384 0 snd_intel_sst_core 65536 1 snd_intel_sst_acpi snd_soc_rt5640 106496 0 snd_soc_sst_mfld_platform 73728 1 snd_intel_sst_core snd_soc_rl6231 16384 1 snd_soc_rt5640 snd_soc_core 155648 2 snd_soc_rt5640,snd_soc_sst_mfld_platform snd_compress 20480 1 snd_soc_core regmap_i2c 16384 1 snd_soc_rt5640 snd_pcm_dmaengine 16384 1 snd_soc_core ac97_bus 16384 1 snd_soc_core snd_pcm 86016 4 snd_soc_rt5640,snd_soc_core,snd_soc_sst_mfld_platform,snd_pcm_dmaengine snd_timer 28672 1 snd_pcm snd 65536 5 snd_soc_core,snd_timer,snd_pcm,snd_soc_sst_mfld_platform,snd_compress soundcore 16384 1 snd i2c_core 49152 7 drm,i915,snd_soc_rt5640,i2c_designware_platform,regmap_i2c,drm_kms_helper,i2c_algo_bit snd_soc_sst_acpi 16384 0 [ml@orb ~]$ -- Best Regards, Michael Leun

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[alsa-devel] [PATCH] ASoC: sgtl5000: fix bad register MIC_BIAS_VOLTAGE setup on probe
by Gianluca Renzi 25 Sep '15

25 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(-) diff -up a/sound/soc/codecs/sgtl5000.c b/sound/soc/codecs/sgtl5000.c --- a/sound/soc/codecs/sgtl5000.c 2015-09-25 15:24:10.597051191 +0200 +++ b/sound/soc/codecs/sgtl5000.c 2015-09-25 15:24:21.000000000 +0200 @@ -1377,7 +1377,7 @@ static int sgtl5000_probe(struct snd_soc snd_soc_update_bits(codec, SGTL5000_CHIP_MIC_CTRL, SGTL5000_BIAS_R_MASK, - sgtl5000->micbias_voltage << SGTL5000_BIAS_R_SHIFT); + sgtl5000->micbias_voltage << SGTL5000_BIAS_VOLT_SHIFT); /* * disable DAP * TODO: @@ -1549,7 +1549,7 @@ static int sgtl5000_i2c_probe(struct i2c else { sgtl5000->micbias_voltage = 0; dev_err(&client->dev, - "Unsuitable MicBias resistor\n"); + "Unsuitable MicBias voltage\n"); } } else { sgtl5000->micbias_voltage = 0;

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[alsa-devel] ASoC: please revert "ASoC: db1200: Use static DAI format setup"
by Manuel Lauss 25 Sep '15

25 Sep '15

Hi Lars, commit b4508d0f95fa4aaed889549e31391641d675d4bb ("ASoC: db1200: Use static DAI format setup") breaks I2S in that the set_dai_fmt() callbacks of neither the codec (wm8731) nor the I2S block (psc-i2s) are called. Please revert this change or fix it up so that the set_dai_fmt() callbacks are at least called once. (This was broken since 4.0, but I just now got around to test linux on the hardware again). Thanks! Manuel -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in

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[alsa-devel] [GIT PULL] ASoC updates for v4.4
by Mark Brown 25 Sep '15

25 Sep '15

The following changes since commit 1f93e4a96c9109378204c147b3eec0d0e8100fde: Linux 4.3-rc2 (2015-09-20 14:32:34 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v4.3-rc2 for you to fetch changes up to 4922e7a1b767fcf6f29ce121ad74e424f4045eff: Merge remote-tracking branch 'asoc/topic/ux500' into asoc-next (2015-09-23 11:01:30 -0700) ---------------------------------------------------------------- ASoC: Updates for v4.4 A first batch of updates targetted at v4.4. There are no substantial core fixes here, the biggest block of changes is updates to the rcar drivers and the addition of a CODEC driver for the AK4613. ---------------------------------------------------------------- Adam Thomson (1): ASoC: fsl_ssi: Fix checking of dai format for AC97 mode Anatol Pomozov (1): ASoC: Document snd-soc-dummy-dai purpose Arnaud Pouliquen (1): ASoC: sti: check return of of_property_read Axel Lin (11): ASoC: mediatek: Convert to devm_snd_soc_register_card ASoC: au1x: Convert to devm_snd_soc_register_card ASoC: blackfin: Convert to devm_snd_soc_register_card ASoC: mxs-sgtl5000: Convert to devm_snd_soc_register_card ASoC: pxa: Convert to devm_snd_soc_register_card ASoC: SPEAr: Make SND_SPEAR_SOC select SND_SOC_GENERIC_DMAENGINE_PCM ASoC: au1x: psc-i2s: Fix unused variable 'ret' warning ASoC: sh: siu_dai: Convert to use resource managed APIs ASoC: intel: broadwell: Convert to devm_snd_soc_register_card ASoC: SPEAr: Make SND_SPEAR_SOC select SND_SOC_GENERIC_DMAENGINE_PCM ASoC: rockchip: Fix missing CLKDEV_LOOKUP dependency Charles Keepax (4): ASoC: arizona: Add default cases for event switches ASoC: arizona: Add utility function to check if an input is analog ASoC: Add SOC_SINGLE_RANGE_EXT_TLV macro ASoC: wm5110: Add additional analogue input enable for early revs Emilio López (1): ASoC: sunxi: add support for the on-chip codec on early Allwinner SoCs Jarkko Nikula (1): spi: atmel: Remove needless bits_per_word and speed_hz tests Jeeja KP (1): ASoC: Intel: Skylake: Remove unused CPU dais John Lin (3): ASoC: rt5645: Add struct dmi_system_id "Google Ultima" for chrome platform ASoC: rt5645: Add dmi for Broadwell ASoC: rt5645: Add jd_invert for Broadwell Julia Lawall (1): ASoC: qcom: change PTR_ERR argument Jyri Sarha (7): ASoC: davinci-mcasp: Set .symmetric_rates = 1 in snd_soc_dai_driver ASoC: tlv320aic3x: Improve tdm support ASoC: davinci-mcasp: Add set_tdm_slots() support ASoC: simple-card: Add tdm slot mask support to simple-card ASoC: soc-core: Fix sparse warning in be32_to_cpup() call ASoC: davinci-mcasp: Get rid of bclk_lrclk_ratio in private data ASoC: hdmi: Remove obsolete dummy HDMI codec Koro Chen (1): ASoC: mediatek: Increase periods_min in capture Kuninori Morimoto (20): ASoC: ak4642: use *dev on ak4642_i2c_probe() ASoC: ak4642: enable to use MCKO as fixed rate output pin on DT ASoC: rsnd: tidyup rsnd_dma_to_xxx() position ASoC: rsnd: rename rsnd_src_pcm_new() to rsnd_src_pcm_new_gen2() ASoC: rsnd: remove unneeded sh_clk header ASoC: rsnd: add missing #sound-dai-cells explain on Document ASoC: rsnd: Add Gen3 initial support ASoC: rsnd: add rsnd_mod_get() macro and use it ASoC: rsnd: ADG uses mod base common method ASoC: rsnd: add common mod confirm method ASoC: rsnd: ADG initial setup on rsnd_adg_ssi_clk_init() ASoC: rsnd: tidyup ADG clock calculate method ASoC: rsnd: tidyup ADG debug message for clock selection ASoC: rsnd: adg: ignore undefined clock error ASoC: rsnd: add AUDIO_CLKOUT support ASoC: rsnd: SND_SOC_RCAR depends on COMMON_CLK ASoC: ak4642: exchange macro name to avoid redefinition ASoC: rsnd: fixup clkout_name[] index error for single AUDIO_CLKOUT SoC ASoC: add ak4613 support ASoC: ak4613: add Digital Playback Volume support Liam Girdwood (1): ASoC: intel: Fix SSP port configuration after RTD3 resume. Luis de Bethencourt (8): ASoC: atmel_wm8904: Fix module autoload for OF platform driver ASoC: fsl-asoc-card: Fix module autoload for OF platform driver ASoC: fsl_sai: Fix module autoload for OF platform driver ASoC: jz4740: Fix module autoload for OF platform driver ASoC: kirkwood: Fix module autoload for OF platform driver ASoC: pxa: Fix module autoload for OF platform drivers ASoC: mop500: Fix module autoload for OF platform driver ASoC: ux500: Fix module autoload for OF platform driver Maciej S. Szmigiero (1): ASoC: fsl-asoc-card: put ASRC OF node in case of unknown device Mark Brown (14): Merge branch 'fix/rt5645' of git://git.kernel.org/.../broonie/sound into asoc-fix-rt5645 Merge branch 'fix/rt5645' of git://git.kernel.org/.../broonie/sound into asoc-rt5645 Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/rt5645' into asoc-linus Merge remote-tracking branches 'asoc/fix/davinci', 'asoc/fix/doc', 'asoc/fix/fsl-card', 'asoc/fix/fsl-ssi', 'asoc/fix/intel' and 'asoc/fix/maintainers' into asoc-linus Merge remote-tracking branches 'asoc/fix/mtk', 'asoc/fix/psc', 'asoc/fix/pxa', 'asoc/fix/spear', 'asoc/fix/sti' and 'asoc/fix/wm0010' into asoc-linus Merge remote-tracking branches 'asoc/fix/wm8960' and 'asoc/fix/wm8962' into asoc-linus Merge remote-tracking branch 'asoc/topic/rt5645' into asoc-next Merge remote-tracking branches 'asoc/topic/ak4613', 'asoc/topic/ak4642', 'asoc/topic/arizona', 'asoc/topic/atmel' and 'asoc/topic/au1x' into asoc-next Merge remote-tracking branches 'asoc/topic/blackfin', 'asoc/topic/davinci', 'asoc/topic/fsl', 'asoc/topic/hdmi' and 'asoc/topic/intel' into asoc-next Merge remote-tracking branches 'asoc/topic/jz4740', 'asoc/topic/kirkwood', 'asoc/topic/mtk', 'asoc/topic/multi' and 'asoc/topic/mxs' into asoc-next Merge remote-tracking branches 'asoc/topic/pxa', 'asoc/topic/qcom', 'asoc/topic/rcar' and 'asoc/topic/rockchip' into asoc-next Merge remote-tracking branches 'asoc/topic/sh', 'asoc/topic/simple', 'asoc/topic/spear', 'asoc/topic/sunxi' and 'asoc/topic/tlv320aic3x' into asoc-next Merge remote-tracking branch 'asoc/topic/ux500' into asoc-next Maxime Ripard (1): ASoC: sunxi: Add the Allwinner A10 codec bindings Oder Chiou (5): ASoC: rt5645: Remove incorrect settings ASoC: rt5645: Remove the repeated definitions ASoC: rt5645: Use the type SOC_DAPM_SINGLE_AUTODISABLE to prevent the weird sound in runtime of power up ASoC: rt5645: Increase the delay time to remove the pop sound ASoC: rt5645: Prevent the pop sound in case of playback and the jack is plugging Peter Ujfalusi (2): ASoC: davinci-mcasp: Revise the FIFO threshold calculation ASoC: davinci-mcasp: Fix devm_kasprintf format string Ricard Wanderlof (1): ASoC: Handle multiple codecs with split playback / capture Richard Fitzgerald (1): MAINTAINERS: Update website and git repo for Wolfson Microelectronics Robert Jarzmik (2): ASoC: fix broken pxa SoC support ASoC: pxa: pxa2xx-ac97: fix dma requestor lines Sudip Mukherjee (5): ASoC: wm0010: fix memory leak ASoC: dapm: fix memory leak ASoC: Intel: remove unused function ASoC: wm0010: fix memory leak ASoC: wm0010: fix error path Zidan Wang (4): ASoC: wm8960: correct the min gain value of some PGA ASoC: wm8960: correct gain value for input PGA and add microphone PGA ASoC: wm8962: remove 64k sample rate support ASoC: wm8960: correct the max register value of mic boost pga kbuild test robot (1): ASoC: sunxi: fix platform_no_drv_owner.cocci warnings Documentation/devicetree/bindings/sound/ak4613.txt | 17 + Documentation/devicetree/bindings/sound/ak4642.txt | 22 +- .../devicetree/bindings/sound/renesas,rsnd.txt | 7 + .../devicetree/bindings/sound/sun4i-codec.txt | 33 + .../devicetree/bindings/sound/tdm-slot.txt | 11 +- MAINTAINERS | 9 +- drivers/spi/spi-atmel.c | 21 +- include/sound/rcar_snd.h | 1 - include/sound/rt5645.h | 2 + include/sound/simple_card.h | 2 + include/sound/soc.h | 14 + sound/arm/Kconfig | 15 +- sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/atmel/atmel_wm8904.c | 1 + sound/soc/au1x/db1000.c | 10 +- sound/soc/au1x/db1200.c | 10 +- sound/soc/au1x/psc-i2s.c | 1 - sound/soc/blackfin/bf5xx-ad1836.c | 11 +- sound/soc/blackfin/bfin-eval-adau1373.c | 12 +- sound/soc/blackfin/bfin-eval-adau1701.c | 12 +- sound/soc/blackfin/bfin-eval-adav80x.c | 12 +- sound/soc/codecs/Kconfig | 9 +- sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/ak4613.c | 497 ++++++++++++++ sound/soc/codecs/ak4642.c | 153 +++-- sound/soc/codecs/arizona.c | 16 + sound/soc/codecs/arizona.h | 2 + sound/soc/codecs/hdmi.c | 109 ---- sound/soc/codecs/rt5645.c | 57 +- sound/soc/codecs/rt5645.h | 6 +- sound/soc/codecs/tlv320aic3x.c | 30 +- sound/soc/codecs/wm0010.c | 23 +- sound/soc/codecs/wm5110.c | 187 +++++- sound/soc/codecs/wm8960.c | 26 +- sound/soc/codecs/wm8962.c | 3 +- sound/soc/davinci/davinci-mcasp.c | 319 +++++---- sound/soc/fsl/fsl-asoc-card.c | 4 +- sound/soc/fsl/fsl_sai.c | 1 + sound/soc/fsl/fsl_ssi.c | 5 +- sound/soc/generic/simple-card.c | 8 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 17 - sound/soc/intel/boards/broadwell.c | 9 +- sound/soc/intel/haswell/sst-haswell-ipc.c | 20 +- sound/soc/intel/skylake/skl-pcm.c | 33 - sound/soc/jz4740/jz4740-i2s.c | 1 + sound/soc/kirkwood/armada-370-db.c | 1 + sound/soc/mediatek/mt8173-max98090.c | 11 +- sound/soc/mediatek/mt8173-rt5650-rt5676.c | 11 +- sound/soc/mediatek/mtk-afe-pcm.c | 17 + sound/soc/mxs/mxs-sgtl5000.c | 6 +- sound/soc/pxa/Kconfig | 2 - sound/soc/pxa/brownstone.c | 9 +- sound/soc/pxa/corgi.c | 11 +- sound/soc/pxa/e740_wm9705.c | 5 +- sound/soc/pxa/e750_wm9705.c | 5 +- sound/soc/pxa/e800_wm9712.c | 5 +- sound/soc/pxa/hx4700.c | 4 +- sound/soc/pxa/imote2.c | 11 +- sound/soc/pxa/mioa701_wm9713.c | 11 +- sound/soc/pxa/palm27x.c | 9 +- sound/soc/pxa/poodle.c | 11 +- sound/soc/pxa/pxa-ssp.c | 1 + sound/soc/pxa/pxa2xx-ac97.c | 4 +- sound/soc/pxa/pxa2xx-pcm.c | 1 + sound/soc/pxa/spitz.c | 5 +- sound/soc/pxa/tosa.c | 5 +- sound/soc/pxa/ttc-dkb.c | 12 +- sound/soc/qcom/lpass-cpu.c | 3 +- sound/soc/rockchip/Kconfig | 4 +- sound/soc/sh/Kconfig | 1 + sound/soc/sh/rcar/adg.c | 303 +++++++-- sound/soc/sh/rcar/core.c | 12 + sound/soc/sh/rcar/ctu.c | 6 +- sound/soc/sh/rcar/dvc.c | 6 +- sound/soc/sh/rcar/mix.c | 6 +- sound/soc/sh/rcar/rsnd.h | 15 +- sound/soc/sh/rcar/src.c | 17 +- sound/soc/sh/rcar/ssi.c | 98 +-- sound/soc/sh/siu_dai.c | 85 +-- sound/soc/soc-core.c | 25 + sound/soc/soc-dapm.c | 2 +- sound/soc/soc-pcm.c | 49 ++ sound/soc/soc-utils.c | 9 + sound/soc/spear/Kconfig | 2 +- sound/soc/sti/uniperif_player.c | 14 +- sound/soc/sti/uniperif_reader.c | 6 +- sound/soc/sunxi/Kconfig | 11 + sound/soc/sunxi/Makefile | 2 + sound/soc/sunxi/sun4i-codec.c | 719 +++++++++++++++++++++ sound/soc/ux500/mop500.c | 1 + sound/soc/ux500/ux500_msp_dai.c | 1 + 92 files changed, 2484 insertions(+), 834 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/ak4613.txt create mode 100644 Documentation/devicetree/bindings/sound/sun4i-codec.txt create mode 100644 sound/soc/codecs/ak4613.c delete mode 100644 sound/soc/codecs/hdmi.c create mode 100644 sound/soc/sunxi/Kconfig create mode 100644 sound/soc/sunxi/Makefile create mode 100644 sound/soc/sunxi/sun4i-codec.c

2 2

[alsa-devel] [PATCH v2] sound: oss: ad1848: Fix returned errno code in ad1848_init()
by Luis de Bethencourt 25 Sep '15

25 Sep '15

The driver is using -1 instead of the -ENOMEM defined macro to specify that a buffer allocation failed. Since the error number is propagated, the caller will get a -EPERM which is the wrong error condition. Changed the checks of the return of ad1848_init() to be for >= 0, instead of != -1. Smatch tool warning: ad1848_init() warn: returning -1 instead of -ENOMEM is sloppy Signed-off-by: Luis de Bethencourt <luisbg(a)osg.samsung.com> --- sound/oss/ad1848.c | 4 ++-- sound/oss/dev_table.c | 2 +- sound/oss/pss.c | 2 +- 3 files changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/oss/ad1848.c b/sound/oss/ad1848.c index 10c8de1..58c6d31 100644 --- a/sound/oss/ad1848.c +++ b/sound/oss/ad1848.c @@ -1992,7 +1992,7 @@ int ad1848_init (char *name, struct resource *ports, int irq, int dma_playback, portc = kmalloc(sizeof(ad1848_port_info), GFP_KERNEL); if(portc==NULL) { release_region(devc->base, 4); - return -1; + return -ENOMEM; } if ((my_dev = sound_install_audiodrv(AUDIO_DRIVER_VERSION, @@ -2007,7 +2007,7 @@ int ad1848_init (char *name, struct resource *ports, int irq, int dma_playback, { release_region(devc->base, 4); kfree(portc); - return -1; + return my_dev; } audio_devs[my_dev]->portc = portc; diff --git a/sound/oss/dev_table.c b/sound/oss/dev_table.c index 6dad515..abb3db4 100644 --- a/sound/oss/dev_table.c +++ b/sound/oss/dev_table.c @@ -148,7 +148,7 @@ EXPORT_SYMBOL(sound_install_mixer); void sound_unload_audiodev(int dev) { - if (dev != -1) { + if (dev >= 0) { DMAbuf_deinit(dev); audio_devs[dev] = NULL; unregister_sound_dsp((dev<<4)+3); diff --git a/sound/oss/pss.c b/sound/oss/pss.c index 81314f9..e84aafe 100644 --- a/sound/oss/pss.c +++ b/sound/oss/pss.c @@ -1091,7 +1091,7 @@ static int __init probe_pss_mss(struct address_info *hw_config) pss_mixer_reset(devc); attach_ms_sound(hw_config, ports, THIS_MODULE); /* Slot 0 */ - if (hw_config->slots[0] != -1) + if (hw_config->slots[0] >= 0) { /* The MSS driver installed itself */ audio_devs[hw_config->slots[0]]->coproc = &pss_coproc_operations; -- 2.5.1

2 4

[alsa-devel] [PATCH] ASoC: db1200: Fix DAI link format for db1300 and db1550
by Lars-Peter Clausen 25 Sep '15

25 Sep '15

Commit b4508d0f95fa ("ASoC: db1200: Use static DAI format setup") switched the db1200 driver over to using static DAI format setup instead of a callback function. But the commit only added the dai_fmt field to one of the three DAI links in the driver. This breaks audio on db1300 and db1550. Add the two missing dai_fmt settings to fix the issue. Fixes: b4508d0f95fa ("ASoC: db1200: Use static DAI format setup") Reported-by: Manuel Lauss <manuel.lauss(a)gmail.com> Tested-by: Manuel Lauss <manuel.lauss(a)gmail.com> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/au1x/db1200.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c index 638ca0b..5c73061 100644 --- a/sound/soc/au1x/db1200.c +++ b/sound/soc/au1x/db1200.c @@ -129,6 +129,8 @@ static struct snd_soc_dai_link db1300_i2s_dai = { .cpu_dai_name = "au1xpsc_i2s.2", .platform_name = "au1xpsc-pcm.2", .codec_name = "wm8731.0-001b", + .dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .ops = &db1200_i2s_wm8731_ops, }; @@ -146,6 +148,8 @@ static struct snd_soc_dai_link db1550_i2s_dai = { .cpu_dai_name = "au1xpsc_i2s.3", .platform_name = "au1xpsc-pcm.3", .codec_name = "wm8731.0-001b", + .dai_fmt = SND_SOC_DAIFMT_LEFT_J | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBM_CFM, .ops = &db1200_i2s_wm8731_ops, }; -- 2.1.4

1 0

[alsa-devel] Missing snd_pcm_link() for the rate plugin
by Christian Gruber 25 Sep '15

25 Sep '15

Dear ALSA community, does anybody know, why the rate plugin does not support the snd_pcm_link() function? As I can see from the ALSA-Lib source code, snd_pcm_link() calls the fast_op "link", which is a function pointer, whose destination depends on the type of pcm device/plugin to be linked. If the device is a HW-device for instance, the function snd_pcm_hw_link() is called. Several other plugins (e.g. mmap_emul, linear, lfloat, mulaw, alaw, route) use the generic function snd_pcm_generic_link(), which forwards the function call to their slave devices/plugins. The rate plugin instead does not support snd_pcm_link(), since the function pointer "link" is not assigned in the definition of snd_pcm_rate_fast_ops. Is their any reason to not use the generic function snd_pcm_generic_link() within the rate plugin? Thanks in advance for answering this question, Christian

2 1

[alsa-devel] sunxi, Allwinner TDM Support?
by Caleb Crome 25 Sep '15

25 Sep '15

Hi, I was wondering if the sunxi parts support TDM mode on their DAI ports? Those parts sure have attractive pricing, and as you may have gathered, I'm all about many microphone channels. :-) Thanks, -Caleb

1 0

[alsa-devel] RT5640 jack detection
by Carlo Caione 24 Sep '15

24 Sep '15

Hi, I'm working with the ALC5640-VB audio codec based on the Realtek RT5640. On my setup the codec is only used for audio output through an headphone jack, otherwise the audio should be routed via HDMI. The problem is that the jack sensing only works when the codec is actively playing something, that is when the bias set for the codec is not SND_SOC_BIAS_OFF. When the codec is suspended and it is powered off I'm not able to detect anymore the presence of the jack. Because of this issue actively switching the audio from HDMI to HP when the jack is inserted is not working. The driver I'm using for the RT5640 in the .set_bias_level hook function among other things is gating the I2S input clock and this seems to prevent the generation of the IRQ signal I'm using to detect the presence/absence of the jack (the culprit here is MX-FA[0] bit for those interested). I can easily modify the codec driver to avoid gating the I2S clock when the bias is off but this seems in contrast with the whole DAPM principle. How this is usually supposed to be working? -- Carlo Caione | +39.340.80.30.096 | Endless

3 4

[alsa-devel] [PATCH 0/5] Add ASoC support for AMD APUs [v3]
by Alex Deucher 24 Sep '15

24 Sep '15

This patch set implements support for i2s audio and new AMD GPUs. The i2s codec is fed by a DMA engine on the GPU. To handle this we create mfd cells which we hang the i2s codec and DMA engine on. Because of this, this patch set covers two subsystems: drm and alsa. The drm patches add support for the ACP hw block which provides the DMA engine for the i2s codec. The alsa patches add the ASoC driver for the i2s codec. Since the alsa changes depend on the drm changes in this patch set as well as some other drm changes queued for 4.3, I'd like to take the alsa patches in via the drm tree. V2 changes: - Use the MFD subsystem rather than adding our own bus - Squash all sub-feature patches together - fix comments mentioned in previous review V3 changes: - Update the designware driver to handle slave mode, amd specific features - Use the designware driver directly for i2s - Move the DMA handling from the GPU driver into the AMD ASoC driver - Change the license on the ASoC driver to GPL Patch 4 adds the register headers for the ACP block which is a pretty big patch so I've excluded it from email. The entire patch set can be viewed here: http://cgit.freedesktop.org/~agd5f/linux/log/?h=acp-upstream3 Thanks, Alex Maruthi Bayyavarapu (1): drm/amd: add ACP driver support Maruthi Srinivas Bayyavarapu (4): ASoC : dwc : support dw i2s in slave mode ASoC : dwc : support dw i2s in AMD platform ASoC : AMD : add ACP 2.2 register headers ASoC: AMD: add AMD ASoC ACP-I2S driver drivers/gpu/drm/Kconfig | 2 + drivers/gpu/drm/amd/acp/Kconfig | 10 + drivers/gpu/drm/amd/acp/Makefile | 9 + drivers/gpu/drm/amd/acp/acp_hw.c | 127 ++ drivers/gpu/drm/amd/acp/include/acp_gfx_if.h | 49 + drivers/gpu/drm/amd/amdgpu/Makefile | 13 +- drivers/gpu/drm/amd/amdgpu/amdgpu.h | 12 + drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c | 269 +++ drivers/gpu/drm/amd/amdgpu/amdgpu_acp.h | 41 + drivers/gpu/drm/amd/amdgpu/vi.c | 12 + drivers/gpu/drm/amd/include/amd_shared.h | 1 + include/linux/mfd/amd_acp.h | 43 + include/sound/designware_i2s.h | 4 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/amd/Kconfig | 4 + sound/soc/amd/Makefile | 3 + sound/soc/amd/acp-pcm-dma.c | 518 ++++++ sound/soc/amd/acp.c | 736 +++++++++ sound/soc/amd/acp.h | 147 ++ sound/soc/amd/include/acp_2_2_d.h | 609 +++++++ sound/soc/amd/include/acp_2_2_enum.h | 1068 ++++++++++++ sound/soc/amd/include/acp_2_2_sh_mask.h | 2292 ++++++++++++++++++++++++++ sound/soc/dwc/designware_i2s.c | 270 +-- 24 files changed, 6141 insertions(+), 100 deletions(-) create mode 100644 drivers/gpu/drm/amd/acp/Kconfig create mode 100644 drivers/gpu/drm/amd/acp/Makefile create mode 100644 drivers/gpu/drm/amd/acp/acp_hw.c create mode 100644 drivers/gpu/drm/amd/acp/include/acp_gfx_if.h create mode 100644 drivers/gpu/drm/amd/amdgpu/amdgpu_acp.c create mode 100644 drivers/gpu/drm/amd/amdgpu/amdgpu_acp.h create mode 100644 include/linux/mfd/amd_acp.h create mode 100644 sound/soc/amd/Kconfig create mode 100644 sound/soc/amd/Makefile create mode 100644 sound/soc/amd/acp-pcm-dma.c create mode 100644 sound/soc/amd/acp.c create mode 100644 sound/soc/amd/acp.h create mode 100644 sound/soc/amd/include/acp_2_2_d.h create mode 100644 sound/soc/amd/include/acp_2_2_enum.h create mode 100644 sound/soc/amd/include/acp_2_2_sh_mask.h -- 1.8.3.1

2 7

[alsa-devel] [PATCH] ASoC: rl6347a: Clean up unneeded inclusion of header files
by Axel Lin 24 Sep '15

24 Sep '15

Also move the include of sound/hda_verbs.h to rl6347a.h because it is used in rl6347a.h. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/rl6347a.c | 19 +------------------ sound/soc/codecs/rl6347a.h | 2 ++ sound/soc/codecs/rt286.c | 1 - sound/soc/codecs/rt298.c | 1 - 4 files changed, 3 insertions(+), 20 deletions(-) diff --git a/sound/soc/codecs/rl6347a.c b/sound/soc/codecs/rl6347a.c index 91d5166..a4b910e 100644 --- a/sound/soc/codecs/rl6347a.c +++ b/sound/soc/codecs/rl6347a.c @@ -11,25 +11,8 @@ */ #include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/init.h> -#include <linux/delay.h> -#include <linux/pm.h> #include <linux/i2c.h> -#include <linux/platform_device.h> -#include <linux/spi/spi.h> -#include <linux/dmi.h> -#include <linux/acpi.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> -#include <sound/soc-dapm.h> -#include <sound/initval.h> -#include <sound/tlv.h> -#include <sound/jack.h> -#include <linux/workqueue.h> -#include <sound/hda_verbs.h> +#include <linux/regmap.h> #include "rl6347a.h" diff --git a/sound/soc/codecs/rl6347a.h b/sound/soc/codecs/rl6347a.h index 1cb56e5..e127919 100644 --- a/sound/soc/codecs/rl6347a.h +++ b/sound/soc/codecs/rl6347a.h @@ -12,6 +12,8 @@ #ifndef __RL6347A_H__ #define __RL6347A_H__ +#include <sound/hda_verbs.h> + #define VERB_CMD(V, N, D) ((N << 20) | (V << 8) | D) #define RL6347A_VENDOR_REGISTERS 0x20 diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index bd93658..1fbdb4f 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -29,7 +29,6 @@ #include <sound/jack.h> #include <linux/workqueue.h> #include <sound/rt286.h> -#include <sound/hda_verbs.h> #include "rl6347a.h" #include "rt286.h" diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index 3c2f0f8..2a6c2ba 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -28,7 +28,6 @@ #include <sound/jack.h> #include <linux/workqueue.h> #include <sound/rt298.h> -#include <sound/hda_verbs.h> #include "rl6347a.h" #include "rt298.h" -- 2.1.4

2 1

Re: [alsa-devel] ALSA: pcm_direct.c SND_PCM_DIRECT_MAGIC
by Takashi Iwai 24 Sep '15

24 Sep '15

On Wed, 23 Sep 2015 18:07:46 +0200, Mike Bjorge wrote: > > I see, interesting. > > I am running into an issue where pcm_dmix.c is occasionally failing with an > "unable to create IPC shm instance" (pcm_dmix.c@1004) because in > pcm_direct.c@134, (dmix->shmptr->magic != SND_PCM_DIRECT_MAGIC) when > buf.shm_nattch == 2. > > Do you know if there are any likely causes that would result in > dmix->shmptr->magic not being set correctly? What value is read there? If the value is completely different, it means that some other programs use the same shmem for other purpose. It'd be just a conflict of shmem key. Takashi > > Thanks for your help, > Mike > > > > On Wed, Sep 23, 2015 at 6:11 AM, Takashi Iwai <tiwai(a)suse.de> wrote: > > > On Sat, 19 Sep 2015 01:13:42 +0200, > > Mike Bjorge wrote: > > > > > > Hi Takashi, > > > > > > I know this is a bit out of the blue, but I was wondering if you could > > > explain the point of SND_PCM_DIRECT_MAGIC and dmix->shmptr->magic? > > > > > > Specifically, what it's doing in pcm_direct.c from this commit? > > > > > http://git.alsa-project.org/?p=alsa-lib.git;a=blobdiff;f=src/pcm/pcm_direct… > > > > It's to check the consistency of shared data via shmem. > > If we change the data structure in an incompatible way, we'd change > > the magic key so that each instance can detect it. > > > > > > Takashi > > > [2 <text/html; UTF-8 (quoted-printable)>] >

2 1

[alsa-devel] [GIT PULL] ASoC fixes for v4.3
by Mark Brown 24 Sep '15

24 Sep '15

The following changes since commit 1f93e4a96c9109378204c147b3eec0d0e8100fde: Linux 4.3-rc2 (2015-09-20 14:32:34 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.3-rc2 for you to fetch changes up to ed14ee0eea8b6808025356cecc87a8007885263f: Merge remote-tracking branches 'asoc/fix/wm8960' and 'asoc/fix/wm8962' into asoc-linus (2015-09-23 11:01:12 -0700) ---------------------------------------------------------------- ASoC: Fixes for v4.3 A disappointingly large set of fixes, though none of them very big and very widely spread over many different drivers. Nothing especially stands out, it's mostly all device specific and relatively minor. ---------------------------------------------------------------- Adam Thomson (1): ASoC: fsl_ssi: Fix checking of dai format for AC97 mode Anatol Pomozov (1): ASoC: Document snd-soc-dummy-dai purpose Arnaud Pouliquen (1): ASoC: sti: check return of of_property_read Axel Lin (2): ASoC: SPEAr: Make SND_SPEAR_SOC select SND_SOC_GENERIC_DMAENGINE_PCM ASoC: au1x: psc-i2s: Fix unused variable 'ret' warning John Lin (1): ASoC: rt5645: Add struct dmi_system_id "Google Ultima" for chrome platform Jyri Sarha (1): ASoC: davinci-mcasp: Set .symmetric_rates = 1 in snd_soc_dai_driver Koro Chen (1): ASoC: mediatek: Increase periods_min in capture Liam Girdwood (1): ASoC: intel: Fix SSP port configuration after RTD3 resume. Maciej S. Szmigiero (1): ASoC: fsl-asoc-card: put ASRC OF node in case of unknown device Mark Brown (6): Merge branch 'fix/rt5645' of git://git.kernel.org/.../broonie/sound into asoc-fix-rt5645 Merge remote-tracking branch 'asoc/fix/dapm' into asoc-linus Merge remote-tracking branch 'asoc/fix/rt5645' into asoc-linus Merge remote-tracking branches 'asoc/fix/davinci', 'asoc/fix/doc', 'asoc/fix/fsl-card', 'asoc/fix/fsl-ssi', 'asoc/fix/intel' and 'asoc/fix/maintainers' into asoc-linus Merge remote-tracking branches 'asoc/fix/mtk', 'asoc/fix/psc', 'asoc/fix/pxa', 'asoc/fix/spear', 'asoc/fix/sti' and 'asoc/fix/wm0010' into asoc-linus Merge remote-tracking branches 'asoc/fix/wm8960' and 'asoc/fix/wm8962' into asoc-linus Oder Chiou (4): ASoC: rt5645: Remove incorrect settings ASoC: rt5645: Use the type SOC_DAPM_SINGLE_AUTODISABLE to prevent the weird sound in runtime of power up ASoC: rt5645: Increase the delay time to remove the pop sound ASoC: rt5645: Prevent the pop sound in case of playback and the jack is plugging Peter Ujfalusi (2): ASoC: davinci-mcasp: Revise the FIFO threshold calculation ASoC: davinci-mcasp: Fix devm_kasprintf format string Richard Fitzgerald (1): MAINTAINERS: Update website and git repo for Wolfson Microelectronics Robert Jarzmik (2): ASoC: fix broken pxa SoC support ASoC: pxa: pxa2xx-ac97: fix dma requestor lines Sudip Mukherjee (4): ASoC: wm0010: fix memory leak ASoC: dapm: fix memory leak ASoC: wm0010: fix memory leak ASoC: wm0010: fix error path Zidan Wang (4): ASoC: wm8960: correct the min gain value of some PGA ASoC: wm8960: correct gain value for input PGA and add microphone PGA ASoC: wm8962: remove 64k sample rate support ASoC: wm8960: correct the max register value of mic boost pga MAINTAINERS | 9 +++------ sound/arm/Kconfig | 15 ++++++++------- sound/soc/au1x/psc-i2s.c | 1 - sound/soc/codecs/rt5645.c | 22 +++++++++++++--------- sound/soc/codecs/wm0010.c | 23 +++++++++-------------- sound/soc/codecs/wm8960.c | 26 ++++++++++++++++++-------- sound/soc/codecs/wm8962.c | 3 ++- sound/soc/davinci/davinci-mcasp.c | 14 +++++++------- sound/soc/fsl/fsl-asoc-card.c | 3 ++- sound/soc/fsl/fsl_ssi.c | 5 +++-- sound/soc/intel/haswell/sst-haswell-ipc.c | 20 ++++++++++++-------- sound/soc/mediatek/mtk-afe-pcm.c | 17 +++++++++++++++++ sound/soc/pxa/Kconfig | 2 -- sound/soc/pxa/pxa2xx-ac97.c | 4 ++-- sound/soc/soc-dapm.c | 2 +- sound/soc/soc-utils.c | 9 +++++++++ sound/soc/spear/Kconfig | 2 +- sound/soc/sti/uniperif_player.c | 14 ++++++++++---- sound/soc/sti/uniperif_reader.c | 6 +++++- 19 files changed, 122 insertions(+), 75 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: mediatek: Increase periods_min in capture
by Koro Chen 24 Sep '15

24 Sep '15

In capture, there is chance that hw_ptr reported at IRQ is a little smaller than period_size due to internal AFE buffer. In the case of ping-pong buffer: |xxxxxxxxxxxxxxxxxxxxxxxxxxxx--|-----------------------------| hw_ptr < period_size This available buffer will not be read since its size is smaller than avail_min (which is period_size by default), and read thread continues to sleep. If the next hw_ptr is just a little larger than buffer_size, overrun occurs. One more period can hold the possible unread buffer. Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- sound/soc/mediatek/mtk-afe-pcm.c | 17 +++++++++++++++++ 1 file changed, 17 insertions(+) diff --git a/sound/soc/mediatek/mtk-afe-pcm.c b/sound/soc/mediatek/mtk-afe-pcm.c index d190fe0..f5baf3c 100644 --- a/sound/soc/mediatek/mtk-afe-pcm.c +++ b/sound/soc/mediatek/mtk-afe-pcm.c @@ -549,6 +549,23 @@ static int mtk_afe_dais_startup(struct snd_pcm_substream *substream, memif->substream = substream; snd_soc_set_runtime_hwparams(substream, &mtk_afe_hardware); + + /* + * Capture cannot use ping-pong buffer since hw_ptr at IRQ may be + * smaller than period_size due to AFE's internal buffer. + * This easily leads to overrun when avail_min is period_size. + * One more period can hold the possible unread buffer. + */ + if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) { + ret = snd_pcm_hw_constraint_minmax(runtime, + SNDRV_PCM_HW_PARAM_PERIODS, + 3, + mtk_afe_hardware.periods_max); + if (ret < 0) { + dev_err(afe->dev, "hw_constraint_minmax failed\n"); + return ret; + } + } ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); if (ret < 0) -- 1.7.9.5

3 6

[alsa-devel] Applied "spi: atmel: Remove needless bits_per_word and speed_hz tests" to the asoc tree
by Mark Brown 24 Sep '15

24 Sep '15

The patch spi: atmel: Remove needless bits_per_word and speed_hz tests has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 3f8958cf9aaaac8a2bfc7417cb87db757eeb4465 Mon Sep 17 00:00:00 2001 From: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> Date: Tue, 15 Sep 2015 16:26:22 +0300 Subject: [PATCH] spi: atmel: Remove needless bits_per_word and speed_hz tests SPI core validates both bits_per_word and speed_hz transfer parameters and defaults to spi->bits_per_word and spi->max_speed_hz in case these per transfer parameters are not set. This makes possible to remove two if statements and remove one code block that is never executed. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> Acked-by: Nicolas Ferre <nicolas.ferre(a)atmel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- drivers/spi/spi-atmel.c | 21 ++++++--------------- 1 file changed, 6 insertions(+), 15 deletions(-) diff --git a/drivers/spi/spi-atmel.c b/drivers/spi/spi-atmel.c index bf9ed38..1cc1f1e 100644 --- a/drivers/spi/spi-atmel.c +++ b/drivers/spi/spi-atmel.c @@ -871,14 +871,7 @@ static int atmel_spi_set_xfer_speed(struct atmel_spi *as, * Calculate the lowest divider that satisfies the * constraint, assuming div32/fdiv/mbz == 0. */ - if (xfer->speed_hz) - scbr = DIV_ROUND_UP(bus_hz, xfer->speed_hz); - else - /* - * This can happend if max_speed is null. - * In this case, we set the lowest possible speed - */ - scbr = 0xff; + scbr = DIV_ROUND_UP(bus_hz, xfer->speed_hz); /* * If the resulting divider doesn't fit into the @@ -1300,14 +1293,12 @@ static int atmel_spi_one_transfer(struct spi_master *master, return -EINVAL; } - if (xfer->bits_per_word) { - asd = spi->controller_state; - bits = (asd->csr >> 4) & 0xf; - if (bits != xfer->bits_per_word - 8) { - dev_dbg(&spi->dev, + asd = spi->controller_state; + bits = (asd->csr >> 4) & 0xf; + if (bits != xfer->bits_per_word - 8) { + dev_dbg(&spi->dev, "you can't yet change bits_per_word in transfers\n"); - return -ENOPROTOOPT; - } + return -ENOPROTOOPT; } /* -- 2.5.0

2 2

[alsa-devel] [PATCH RFC v4 0/8] Implement generic ASoC HDMI codec and use it in tda998x
by Jyri Sarha 24 Sep '15

24 Sep '15

Changes since RFC v3, ASoC side: - Add "ALSA: pcm: add IEC958 channel status helper for hw_params" - Add "tda998x: Improve tda998x_configure_audio() audio related pdata" - use snd_pcm_create_iec958_consumer_hw_params() to construct the stream header - Remove set_clk() callback from hdmi-codec. It is not needed for now. - Refer to stream header in AIF as specified in HDMI standard - Set current_stream to NULL only after video side audio_shutdown() has been called. Avoid potential race if video side attempts to abort audio at the same time. - No need to have video side device pointer in the hdmi codec's pdata as it is found from dev->parent. - Fix hdmi-codec enum: DAI_ID_I2C > DAI_ID_I2S - Improve audio_startup API comment - Make improved checkpatch happy - BUG_ON > WARN_ON - put */ ending the block comment to a separate line DRM side: - Fix tda998x get_eld() locking - Change tda998x audio parameters in pdata to more generic, that can be readily used in tda998x_audio_config() - Rename and restructure audio port related private data members to be more descriptive - Require audio configuration trough ASoC hdmi-codec if HDMI audio is configured trough DT binding. DTS: - Increase McASP fifo usage form 1 to 32 The binding for tda998x is taken from Jean Francois' patch series[1] on the same subject. The implementation of the of-node parsing has some minor changes from my self. Here is what I think at least could or should still be done, but non of that stuff does not sounds critical right now. Missing from tda998x driver side - hdmi_codec_ops audio_startup() implementation for audio abort support - multi channel audio support (I would need specs and preferably some HW to test for this). Missing from ASoC side generic implementation: - channel_allocation handling is completely left for the video side driver, see if ASoC side could help in any way - snd_soc_jack functionality to handle hdmi cable plug/unplug events [1] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-July/095596.html Best regards, Jyri Jean-Francois Moine (1): drm/i2c: tda998x: Add support of a DT graph of ports Jyri Sarha (7): ASoC: hdmi: Remove obsolete dummy HDMI codec ALSA: pcm: add IEC958 channel status helper for hw_params ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders drm/i2c: tda998x: Remove include/sound/tda998x.h and fix graph parsing drm/i2c: tda998x: Improve tda998x_configure_audio() audio related pdata drm/i2c: tda998x: Register ASoC HDMI codec for audio functionality ARM: dts: am335x-boneblack: Add HDMI audio support .../devicetree/bindings/drm/i2c/tda998x.txt | 51 +++ arch/arm/boot/dts/am335x-boneblack.dts | 90 ++++- drivers/gpu/drm/armada/armada_drv.c | 25 +- drivers/gpu/drm/i2c/Kconfig | 1 + drivers/gpu/drm/i2c/tda998x_drv.c | 319 +++++++++++++--- include/drm/i2c/tda998x.h | 24 +- include/sound/hdmi-codec.h | 104 ++++++ include/sound/pcm_iec958.h | 2 + sound/core/pcm_iec958.c | 52 ++- sound/soc/codecs/Kconfig | 4 +- sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/hdmi-codec.c | 404 +++++++++++++++++++++ sound/soc/codecs/hdmi.c | 109 ------ 13 files changed, 985 insertions(+), 204 deletions(-) create mode 100644 include/sound/hdmi-codec.h create mode 100644 sound/soc/codecs/hdmi-codec.c delete mode 100644 sound/soc/codecs/hdmi.c -- 1.9.1

6 20

[alsa-devel] [PATCH] ALSA: hda - Disable power_save_node for Thinkpads
by Takashi Iwai 24 Sep '15

24 Sep '15

Lenovo Thinkpads with recent Realtek codecs seem suffering from click noises at power transition since the introduction of widget power saving in 4.1 kernel. Although this might be solved by some delays in appropriate points, as a quick workaround, just disable the power_save_node feature for now. The gain it gives is relatively small, and this makes the situation back to pre 4.1 time. This patch ended up with a bit more code changes than usual because the existing fixup for Thinkpads is highly chained. Instead of adding yet another chain, combine a few of them into a single fixup entry, as a gratis cleanup. Bugzilla: https://bugzilla.suse.com/show_bug.cgi?id=943982 Cc: <stable(a)vger.kernel.org> # v4.1+ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 31 +++++++++++++++++++------------ 1 file changed, 19 insertions(+), 12 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a75b5611d1e4..afec6dc9f91f 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4188,6 +4188,24 @@ static void alc_fixup_disable_aamix(struct hda_codec *codec, } } +/* fixup for Thinkpad docks: add dock pins, avoid HP parser fixup */ +static void alc_fixup_tpt440_dock(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + static const struct hda_pintbl pincfgs[] = { + { 0x16, 0x21211010 }, /* dock headphone */ + { 0x19, 0x21a11010 }, /* dock mic */ + { } + }; + struct alc_spec *spec = codec->spec; + + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + spec->parse_flags = HDA_PINCFG_NO_HP_FIXUP; + codec->power_save_node = 0; /* avoid click noises */ + snd_hda_apply_pincfgs(codec, pincfgs); + } +} + static void alc_shutup_dell_xps13(struct hda_codec *codec) { struct alc_spec *spec = codec->spec; @@ -4562,7 +4580,6 @@ enum { ALC255_FIXUP_HEADSET_MODE_NO_HP_MIC, ALC293_FIXUP_DELL1_MIC_NO_PRESENCE, ALC292_FIXUP_TPT440_DOCK, - ALC292_FIXUP_TPT440_DOCK2, ALC283_FIXUP_BXBT2807_MIC, ALC255_FIXUP_DELL_WMI_MIC_MUTE_LED, ALC282_FIXUP_ASPIRE_V5_PINS, @@ -5029,17 +5046,7 @@ static const struct hda_fixup alc269_fixups[] = { }, [ALC292_FIXUP_TPT440_DOCK] = { .type = HDA_FIXUP_FUNC, - .v.func = alc269_fixup_pincfg_no_hp_to_lineout, - .chained = true, - .chain_id = ALC292_FIXUP_TPT440_DOCK2 - }, - [ALC292_FIXUP_TPT440_DOCK2] = { - .type = HDA_FIXUP_PINS, - .v.pins = (const struct hda_pintbl[]) { - { 0x16, 0x21211010 }, /* dock headphone */ - { 0x19, 0x21a11010 }, /* dock mic */ - { } - }, + .v.func = alc_fixup_tpt440_dock, .chained = true, .chain_id = ALC269_FIXUP_LIMIT_INT_MIC_BOOST }, -- 2.5.3

1 0

[alsa-devel] [linux-sunxi][PATCH 0/3]Add SPDIF support for Allwinner SoCs
by codekipper＠gmail.com 24 Sep '15

24 Sep '15

From: Marcus Cooper <codekipper(a)gmail.com> This patch set adds support for the Allwinner SPDIF transceiver as present on the A10, A20 and A31 SoC boards. For now just the SPDIF transmitter has been tested on a Mele A2000. In order for this patch set to be functional we require audio clock patches which will be delivered separately. For those that are interested I've pushed the patches here with all the required changes to get SPDIF audio out of the device. https://github.com/codekipper/linux-sunxi/commits/spdif_delivery Thanks in advance, CK Marcus Cooper (3): dt-bindings: add sunxi SPDIF transceiver bindings dt-binding: Add sunxi SPDIF machine driver ASOC: sunxi: Add support for the spdif block .../devicetree/bindings/sound/sunxi,spdif.txt | 49 ++ .../bindings/sound/sunxi-audio-spdif.txt | 36 + sound/soc/sunxi/Kconfig | 10 + sound/soc/sunxi/Makefile | 4 + sound/soc/sunxi/sunxi-machine-spdif.c | 110 +++ sound/soc/sunxi/sunxi-spdif.c | 801 +++++++++++++++++++++ 6 files changed, 1010 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/sunxi,spdif.txt create mode 100644 Documentation/devicetree/bindings/sound/sunxi-audio-spdif.txt create mode 100644 sound/soc/sunxi/sunxi-machine-spdif.c create mode 100644 sound/soc/sunxi/sunxi-spdif.c -- 2.5.3

2 4

[alsa-devel] [PATCH] drm/i915: set proper N/CTS in modeset
by libin.yang＠intel.com 24 Sep '15

24 Sep '15

From: Libin Yang <libin.yang(a)intel.com> When modeset occurs and the TMDS frequency is set to some speical values, the N/CTS need to be set manually if audio is playing. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- drivers/gpu/drm/i915/intel_audio.c | 56 ++++++++++++++++++++++++++++++++------ include/drm/i915_component.h | 10 +++++++ 2 files changed, 57 insertions(+), 9 deletions(-) diff --git a/drivers/gpu/drm/i915/intel_audio.c b/drivers/gpu/drm/i915/intel_audio.c index e3c32ce..a12bf57 100644 --- a/drivers/gpu/drm/i915/intel_audio.c +++ b/drivers/gpu/drm/i915/intel_audio.c @@ -128,6 +128,20 @@ static int audio_config_get_n(const struct drm_display_mode *mode, int rate) return 0; } +static uint32_t audio_config_setup_n_reg(int n, uint32_t val) +{ + int n_low, n_up; + uint32_t tmp = val; + + n_low = n & 0xfff; + n_up = (n >> 12) & 0xff; + tmp &= ~(AUD_CONFIG_UPPER_N_MASK | AUD_CONFIG_LOWER_N_MASK); + tmp |= ((n_up << AUD_CONFIG_UPPER_N_SHIFT) | + (n_low << AUD_CONFIG_LOWER_N_SHIFT) | + AUD_CONFIG_N_PROG_ENABLE); + return tmp; +} + /* check whether N/CTS/M need be set manually */ static bool audio_rate_need_prog(struct intel_crtc *crtc, struct drm_display_mode *mode) @@ -262,9 +276,14 @@ static void hsw_audio_codec_enable(struct drm_connector *connector, struct drm_i915_private *dev_priv = connector->dev->dev_private; struct intel_crtc *intel_crtc = to_intel_crtc(encoder->base.crtc); enum pipe pipe = intel_crtc->pipe; + struct i915_audio_component *acomp = dev_priv->audio_component; const uint8_t *eld = connector->eld; + struct intel_digital_port *intel_dig_port = + enc_to_dig_port(&encoder->base); + enum port port = intel_dig_port->port; uint32_t tmp; int len, i; + int n, rate; DRM_DEBUG_KMS("Enable audio codec on pipe %c, %u bytes ELD\n", pipe_name(pipe), drm_eld_size(eld)); @@ -302,12 +321,29 @@ static void hsw_audio_codec_enable(struct drm_connector *connector, /* Enable timestamps */ tmp = I915_READ(HSW_AUD_CFG(pipe)); tmp &= ~AUD_CONFIG_N_VALUE_INDEX; - tmp &= ~AUD_CONFIG_N_PROG_ENABLE; tmp &= ~AUD_CONFIG_PIXEL_CLOCK_HDMI_MASK; if (intel_pipe_has_type(intel_crtc, INTEL_OUTPUT_DISPLAYPORT)) tmp |= AUD_CONFIG_N_VALUE_INDEX; else tmp |= audio_config_hdmi_pixel_clock(mode); + + tmp &= ~AUD_CONFIG_N_PROG_ENABLE; + if (audio_rate_need_prog(intel_crtc, mode)) { + if (!acomp) + rate = 0; + else if (port >= PORT_A && port <= PORT_E) + rate = acomp->aud_sample_rate[port]; + else { + DRM_ERROR("invalid port: %d\n", port); + rate = 0; + } + n = audio_config_get_n(mode, rate); + if (n != 0) + tmp = audio_config_setup_n_reg(n, tmp); + else + DRM_DEBUG_KMS("no suitable N value is found\n"); + } + I915_WRITE(HSW_AUD_CFG(pipe), tmp); mutex_unlock(&dev_priv->av_mutex); @@ -594,9 +630,10 @@ static int i915_audio_component_sync_audio_rate(struct device *dev, struct intel_digital_port *intel_dig_port; struct intel_crtc *crtc; struct drm_display_mode *mode; + struct i915_audio_component *acomp = dev_priv->audio_component; enum pipe pipe = -1; u32 tmp; - int n_low, n_up, n; + int n; /* HSW, BDW SKL need this fix */ if (!IS_SKYLAKE(dev_priv) && @@ -630,6 +667,9 @@ static int i915_audio_component_sync_audio_rate(struct device *dev, pipe_name(pipe), port_name(port)); mode = &crtc->config->base.adjusted_mode; + /* port must be valid now, otherwise the pipe will be invalid */ + acomp->aud_sample_rate[port] = rate; + /* 2. check whether to set the N/CTS/M manually or not */ if (!audio_rate_need_prog(crtc, mode)) { tmp = I915_READ(HSW_AUD_CFG(pipe)); @@ -649,15 +689,10 @@ static int i915_audio_component_sync_audio_rate(struct device *dev, mutex_unlock(&dev_priv->av_mutex); return 0; } - n_low = n & 0xfff; - n_up = (n >> 12) & 0xff; - /* 4. set the N/CTS/M */ + /* 3. set the N/CTS/M */ tmp = I915_READ(HSW_AUD_CFG(pipe)); - tmp &= ~(AUD_CONFIG_UPPER_N_MASK | AUD_CONFIG_LOWER_N_MASK); - tmp |= ((n_up << AUD_CONFIG_UPPER_N_SHIFT) | - (n_low << AUD_CONFIG_LOWER_N_SHIFT) | - AUD_CONFIG_N_PROG_ENABLE); + tmp = audio_config_setup_n_reg(n, tmp); I915_WRITE(HSW_AUD_CFG(pipe), tmp); mutex_unlock(&dev_priv->av_mutex); @@ -678,6 +713,7 @@ static int i915_audio_component_bind(struct device *i915_dev, { struct i915_audio_component *acomp = data; struct drm_i915_private *dev_priv = dev_to_i915(i915_dev); + int i; if (WARN_ON(acomp->ops || acomp->dev)) return -EEXIST; @@ -685,6 +721,8 @@ static int i915_audio_component_bind(struct device *i915_dev, drm_modeset_lock_all(dev_priv->dev); acomp->ops = &i915_audio_component_ops; acomp->dev = i915_dev; + for (i = 0; i < I915_MAX_PORTS; i++) + acomp->aud_sample_rate[i] = 0; dev_priv->audio_component = acomp; drm_modeset_unlock_all(dev_priv->dev); diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index e6d35d7..89dc7d6 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -24,8 +24,18 @@ #ifndef _I915_COMPONENT_H_ #define _I915_COMPONENT_H_ +/* MAX_PORT is the number of port + * It must be sync with I915_MAX_PORTS defined i915_drv.h + * 5 should be enough as only HSW, BDW, SKL need such fix. + */ +#define MAX_PORTS 5 + struct i915_audio_component { struct device *dev; + /** + * @aud_sample_rate: the array of audio sample rate per port + */ + int aud_sample_rate[MAX_PORTS]; const struct i915_audio_component_ops { struct module *owner; -- 1.9.1

4 4

[alsa-devel] Bug: LADSPA implementation - no up/downmix handling
by Andreas Hartmann 24 Sep '15

24 Sep '15

Hi, Using the surround_encoder of the swh-plugins I noticed, that the alsa implementation only assumes an equal number of in- and output channels for ladspa plugins. This occurs from Line 752 onwards in pcm_ladspa.c and leads to a Segfault shortly after. There needs to be a check and correct handling for up/downmix scenarios. Andreas

2 4

[alsa-devel] No stream timing info with alsa pulse plugin?
by sqweek 24 Sep '15

24 Sep '15

Hi guys, I discovered on my machine that alsa_snd_pcm_status_get_tstamp (and also get_htstamp) stop working when my asoundrc invokes the pulse plugin. That is, normally they work fine but once the sound is going via pulse the returned time stamp is always zero. I'm trying to understand why that is. I couldn't really find any discussion of the pulse plugin on the archives so I'm not entirely sure I've found the right list; I was directed here via the pulseaudio bugzilla[1]. [1] https://bugs.freedesktop.org/show_bug.cgi?id=88503 Thanks for any assistance! -sqweek

4 3

[alsa-devel] [PATCH 0/3] ALSA: hda - Avoid potential deadlock
by Thierry Reding 24 Sep '15

24 Sep '15

From: Thierry Reding <treding(a)nvidia.com> The Tegra HDA controller driver committed in v3.16 causes deadlocks when loaded as a module. The reason is that the driver core will lock the HDA controller device upon calling its probe callback and the probe callback then goes on to create child devices for detected codecs and loads their modules via a request_module() call. This is problematic because the new driver will immediately be bound to the device, which will in turn cause the parent of the codec device (the HDA controller device) to be locked again, causing a deadlock. This problem seems to have been present since the modularization of the HD-audio driver in commit 1289e9e8b42f ("ALSA: hda - Modularize HD-audio driver"). On Intel platforms this has been worked around by splitting up the probe sequence into a synchronous and an asynchronous part where the request_module() calls are asynchronous and hence avoid the deadlock. An alternative proposal is provided in this series of patches. Rather than relying on explicit request_module() calls to load kernel modules for HDA codec drivers, this implements a uevent callback for the HDA bus to advertises the MODALIAS information to the userspace helper. Effectively this results in the same modules being loaded, but it uses the more canonical infrastructure to perform this. Deferring the module loading to userspace removes the need for the explicit request_module() calls and works around the recursive locking issue because both drivers will be bound from separate contexts. Thierry Thierry Reding (3): ALSA: hda/hdmi - Add missing MODALIAS information ALSA: hda - Advertise MODALIAS in uevent ALSA: hda: Do not rely on explicit module loading sound/hda/hda_bus_type.c | 12 +++++++ sound/pci/hda/hda_bind.c | 80 ---------------------------------------------- sound/pci/hda/patch_hdmi.c | 3 ++ 3 files changed, 15 insertions(+), 80 deletions(-) -- 2.5.0

3 9

[alsa-devel] ALSA: core: check for underflow in snd_pcm_sw_params()
by Dan Carpenter 24 Sep '15

24 Sep '15

As far as I can see, having an invalid ->tstamp_mode is harmless, but adding a check silences a static checker warning. Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 75888dd..4863af5 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -650,7 +650,8 @@ static int snd_pcm_sw_params(struct snd_pcm_substream *substream, } snd_pcm_stream_unlock_irq(substream); - if (params->tstamp_mode > SNDRV_PCM_TSTAMP_LAST) + if (params->tstamp_mode < 0 || + params->tstamp_mode > SNDRV_PCM_TSTAMP_LAST) return -EINVAL; if (params->proto >= SNDRV_PROTOCOL_VERSION(2, 0, 12) && params->tstamp_type > SNDRV_PCM_TSTAMP_TYPE_LAST)

2 1

[alsa-devel] [PATCH] ALSA: oxfw: add Mackie Onyx Satellite quirk to inform wrong value in 'dbs' field in tx CIP header
by Takashi Sakamoto 24 Sep '15

24 Sep '15

Mackie Onyx Satellite has two usage; standalone and with base station. These two modes has different stream formats. In standalone mode, rx data block includes 2 Multi Bit Linear Audio (MBLA) data channels and tx data block includes 2. With base station, they're 6 and 2. Although, with base station, the actual tx packet include wrong value in 'dbs' field in its CIP header. This quirk causes packet streaming layer to detect packet discontinuity and to stop PCM substream. This is a response of 'single' subfunction 'extended stream format information' command in AV/C Stream Format Information Specification 1.1. It means that a data block in tx packets includes 2 MBLA data channels. $ ./firewire-request /dev/fw1 fcp 0x01ffbfc001000000ffffffff response: 000: 0c ff bf c0 01 00 00 00 ff 00 90 40 03 02 01 02 response: 010: 06 Current OXFW driver parses the response and detects stream formats correctly. $ cat /proc/asound/card1/firewire/formation ... Output Stream from device: Rate PCM MIDI * 48000 2 0 44100 2 0 88200 2 0 96000 2 0 On the other hand, in actual tx CIP, the 'dbs' field has 6. But the number of quadlets in CIP payload is not multiple of 6. Seeing the subtraction of two successive payload quadlets, it should be multiple of 2. payload CIP CIP quadlets header0 header1 24 00060052 9002ffff 24 0006005e 9002ffff 26 0006006a 9002ffff 24 00060077 9002ffff 24 00060083 9002ffff 26 0006008f 9002ffff 24 0006009c 9002ffff 24 000600a8 9002ffff 26 000600b4 9002ffff 24 000600c1 9002ffff This commit adds support for this quirk to OXFW driver, by using CIP_WRONG_DBS flag. When this flag is set, packet streaming layer uses the value of its 'data_block_quadlets' member instead of the value in CIP header. This value is already set by OXFW driver and no discontinuity is detected. Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/oxfw/oxfw-stream.c | 5 ++++- sound/firewire/oxfw/oxfw.c | 29 +++++++++++++++++++++++++++++ sound/firewire/oxfw/oxfw.h | 1 + 3 files changed, 34 insertions(+), 1 deletion(-) diff --git a/sound/firewire/oxfw/oxfw-stream.c b/sound/firewire/oxfw/oxfw-stream.c index 77ad5b9..863eb4a 100644 --- a/sound/firewire/oxfw/oxfw-stream.c +++ b/sound/firewire/oxfw/oxfw-stream.c @@ -238,9 +238,12 @@ int snd_oxfw_stream_init_simplex(struct snd_oxfw *oxfw, * packets. As a result, next isochronous packet includes more data * blocks than IEC 61883-6 defines. */ - if (stream == &oxfw->tx_stream) + if (stream == &oxfw->tx_stream) { oxfw->tx_stream.flags |= CIP_SKIP_INIT_DBC_CHECK | CIP_JUMBO_PAYLOAD; + if (oxfw->wrong_dbs) + oxfw->tx_stream.flags |= CIP_WRONG_DBS; + } end: return err; } diff --git a/sound/firewire/oxfw/oxfw.c b/sound/firewire/oxfw/oxfw.c index 8c6ce01..d606e3a 100644 --- a/sound/firewire/oxfw/oxfw.c +++ b/sound/firewire/oxfw/oxfw.c @@ -19,6 +19,8 @@ #define VENDOR_BEHRINGER 0x001564 #define VENDOR_LACIE 0x00d04b +#define MODEL_SATELLITE 0x00200f + #define SPECIFIER_1394TA 0x00a02d #define VERSION_AVC 0x010001 @@ -129,6 +131,31 @@ static void oxfw_card_free(struct snd_card *card) mutex_destroy(&oxfw->mutex); } +static void detect_quirks(struct snd_oxfw *oxfw) +{ + struct fw_device *fw_dev = fw_parent_device(oxfw->unit); + struct fw_csr_iterator it; + int key, val; + int vendor, model; + + /* Seek from Root Directory of Config ROM. */ + vendor = model = 0; + fw_csr_iterator_init(&it, fw_dev->config_rom + 5); + while (fw_csr_iterator_next(&it, &key, &val)) { + if (key == CSR_VENDOR) + vendor = val; + else if (key == CSR_MODEL) + model = val; + } + + /* + * Mackie Onyx Satellite with base station has a quirk to report a wrong + * value in 'dbs' field of CIP header against its format information. + */ + if (vendor == VENDOR_LOUD && model == MODEL_SATELLITE) + oxfw->wrong_dbs = true; +} + static int oxfw_probe(struct fw_unit *unit, const struct ieee1394_device_id *id) { @@ -157,6 +184,8 @@ static int oxfw_probe(struct fw_unit *unit, if (err < 0) goto error; + detect_quirks(oxfw); + err = name_card(oxfw); if (err < 0) goto error; diff --git a/sound/firewire/oxfw/oxfw.h b/sound/firewire/oxfw/oxfw.h index cace5ad..7b5dad8 100644 --- a/sound/firewire/oxfw/oxfw.h +++ b/sound/firewire/oxfw/oxfw.h @@ -49,6 +49,7 @@ struct snd_oxfw { struct mutex mutex; spinlock_t lock; + bool wrong_dbs; bool has_output; u8 *tx_stream_formats[SND_OXFW_STREAM_FORMAT_ENTRIES]; u8 *rx_stream_formats[SND_OXFW_STREAM_FORMAT_ENTRIES]; -- 2.1.4

2 1

Re: [alsa-devel] [Alsa-user] snd-usb-caiaq (NI Maschine Mikro) device not recognised by alsa
by sypwex 24 Sep '15

24 Sep '15

ow, sorry! I not mentioned that there is no PID of Maschine Mikro (0x1110) in caiaq device list *https://github.com/torvalds/linux/blob/v4.2/sound/usb/caiaq/device.h <https://github.com/torvalds/linux/blob/v4.2/sound/usb/caiaq/device.h>* yes there is some endless dump from usbhid but I don't know how to deal with it $sudo usbhid-dump --entity=all --address=1:15 http://pastebin.com/raw.php?i=KC96ERSV (300KiB) ok, I've tried to extend the driver and end up with following doodad https://github.com/krasnobaev/linux/commits/feature/caiaq-mmikro-1 for every commit I've generated dmesg log *http://pastebin.com/raw.php?i=pJzdAyf4 <http://pastebin.com/raw.php?i=pJzdAyf4>* $amidi -l have no devices $cat /proc/asound/cards 4 [MaschineMikro ]: snd-usb-caiaq - Maschine Mikro Native Instruments Maschine Mikro (usb-0000:00:02.1-8) $cat /proc/asound/devices 24: [ 4] : control $cat /proc/asound/oss/sndstat Sound Driver:3.8.1a-980706 (ALSA emulation code) Kernel: Linux ***** 3.19.0+ #1 SMP Sun Sep 13 10:44:23 MSK 2015 x86_64 Config options: 0 Installed drivers: Type 10: ALSA emulation Card config: X-Fi Extreme Audio [SB0790] at 0xdc00 irq 18 HDA NVidia at 0xfcf7c000 irq 19 Aureal Vortex au8810 at 0xfbfc0000 irq 17 Webcam C170 at usb-0000:00:02.1-4, high speed Native Instruments Maschine Mikro (usb-0000:00:02.1-8) Audio devices: 0: CA0106 (DUPLEX) 2: au8810 adb (DUPLEX) 3: USB Audio Synth devices: NOT ENABLED IN CONFIG Midi devices: 0: CA0106 MPU-401 (UART) 2: au8810 MIDI 2 Timers: 31: system timer Mixers: 0: CA0106 1: Nvidia GPU 1c HDMI/DP 2: Wolfson WM9701,WM9701A 3: USB Mixer 4: Maschine Mikro $cat /proc/asound/oss/devices 64: [4- 0]: mixer So, currently customized driver fails (all info in dmesg log) on every usb_submit_urb and usb_set_interface functions, so I've decided to watch wireshark usb logs of windows driver filtered windows log (usb.transfer_type != 0x01) && (usb.device_address == 3) http://filebin.ca/2G6kerv8x9TD/usbpcap-maschine-mikro-win7-x86-3maschinesof… RAW http://filebin.ca/2G6g6civjKfI/usbpcap-maschine-mikro-win7-x86-3maschinesof… filtered log for custom driver http://filebin.ca/2G6kFZxgTTKR/usbpcap-maschine-mikro-ubuntu15.04-linux3.19… RAW http://filebin.ca/2G6oNuU6c6b5/usbpcap-maschine-mikro-ubuntu15.04-linux3.19… the difference is... - windows driver do not send SET INTERFACE request while native caiaq driver is. - windows driver receives Response and Status on each request while caiaq recieves only Response - windows driver send two GET_REPORT with wValues: 0x03f8 and 0x0101 while caiaq driver sends three GET_REPORT requests with wValues: 0x0101, 0x0120, 0x03f8 - windows send 3 SET_REPORT requests with same wValue 0x03f8 while caiaq driver do not send any SET_REPORT request I do not know what means these wValues (0x0101, 0x0120, 0x03f8) and how to find out. Is there probability if linux driver send SET_REPORT requests in the same way as do windows driver - the device will set up correctly? How to send SET REPORT request? I believe that there is no big difference between Maschine Controller and Maschine Mikro. BUT there is no Maschine Mikro on caiaq portfolio http://caiaq.de/references_en.html On Sun, Sep 6, 2015 at 8:02 PM, Clemens Ladisch <cladisch(a)googlemail.com> wrote: > sypwex wrote: > > Native Instruments Mashine Mikro which is snd-usb-caiaq. > > No it's not; the Maschine Mikro is not the same as the Maschine Controller. > > As far as I can see from your lsusb output, the Mikro either works as > a HID input device, or needs a separate driver. To check the first > possibility, try a tool like <https://github.com/DIGImend/usbhid-dump>. > > > Regards, > Clemens > Cheers.

2 6

[alsa-devel] MIDI port using UART interface
by Francesco Lavra 24 Sep '15

24 Sep '15

Hi, I'm looking to use an UART interface as a MIDI in/out port. Since MIDI is an asynchronous serial interface just like UART, and provided a given UART controller is able to operate at 31.25 kbps (in my case, it is) a UART port can be transformed into a MIDI port by adding some hardware components (e.g. an optocopuler) to it, I would have expected to find a generic MIDI<->UART adapter layer in the kernel that bridges the ALSA MIDI interface with UART drivers and allows to operate UARTs as MIDI ports. But apparently there is no such generic adapter, the closest driver I found is for the UART 16550 but this is specific to a type of UART controller different from the one in the platform I'm using (which is i.MX6). So how would you approach the task of writing a MIDI driver that uses UART behind the scenes? Am I correct in assuming that I have to use the RawMIDI interface (as the UART 16550 driver does) to make my MIDI port available to ALSA applications? Or is there a better alternative? Thanks, Francesco

2 2

[alsa-devel] [PATCH] ASOC: fsl_spdif: fix bit order for spdif RX channel status
by Kevin Groeneveld 24 Sep '15

24 Sep '15

The bits within each byte of the hardware registers for the channel status bits are reversed. Therefore when the channel status is read from the hardware we need to reverse the bits. The code which writes the channel status bits for the TX already correctly reverses the bits. This patch implements similar functionality for the RX. I have tested this patch running on imx6 using several different SPDIF source devices. Signed-off-by: Kevin Groeneveld <kgroeneveld(a)lenbrook.com> --- sound/soc/fsl/fsl_spdif.c | 12 ++++++------ 1 files changed, 6 insertions(+), 6 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index ab729f2..e84e839 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -676,14 +676,14 @@ static int fsl_spdif_capture_get(struct snd_kcontrol *kcontrol, return -EAGAIN; regmap_read(regmap, REG_SPDIF_SRCSH, &cstatus); - ucontrol->value.iec958.status[0] = (cstatus >> 16) & 0xFF; - ucontrol->value.iec958.status[1] = (cstatus >> 8) & 0xFF; - ucontrol->value.iec958.status[2] = cstatus & 0xFF; + ucontrol->value.iec958.status[0] = bitrev8((cstatus >> 16) & 0xFF); + ucontrol->value.iec958.status[1] = bitrev8((cstatus >> 8) & 0xFF); + ucontrol->value.iec958.status[2] = bitrev8(cstatus & 0xFF); regmap_read(regmap, REG_SPDIF_SRCSL, &cstatus); - ucontrol->value.iec958.status[3] = (cstatus >> 16) & 0xFF; - ucontrol->value.iec958.status[4] = (cstatus >> 8) & 0xFF; - ucontrol->value.iec958.status[5] = cstatus & 0xFF; + ucontrol->value.iec958.status[3] = bitrev8((cstatus >> 16) & 0xFF); + ucontrol->value.iec958.status[4] = bitrev8((cstatus >> 8) & 0xFF); + ucontrol->value.iec958.status[5] = bitrev8(cstatus & 0xFF); /* Clear intr */ regmap_write(regmap, REG_SPDIF_SIC, INT_CNEW); -- 1.7.4.1

2 1

[alsa-devel] [PATCH 1/4] ASoC: rt5645: Remove the repeated definitions
by Oder Chiou 23 Sep '15

23 Sep '15

Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5645.h | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/rt5645.h b/sound/soc/codecs/rt5645.h index 90325d9..f45861c 100644 --- a/sound/soc/codecs/rt5645.h +++ b/sound/soc/codecs/rt5645.h @@ -777,8 +777,6 @@ #define RT5645_PWR_CLS_D_R_BIT 9 #define RT5645_PWR_CLS_D_L (0x1 << 8) #define RT5645_PWR_CLS_D_L_BIT 8 -#define RT5645_PWR_ADC_R (0x1 << 1) -#define RT5645_PWR_ADC_R_BIT 1 #define RT5645_PWR_DAC_L2 (0x1 << 7) #define RT5645_PWR_DAC_L2_BIT 7 #define RT5645_PWR_DAC_R2 (0x1 << 6) -- 1.8.1.1.439.g50a6b54

2 7

[alsa-devel] [PATCH] topology: Add API to set a vendor specific version number
by mengdong.lin＠intel.com 23 Sep '15

23 Sep '15

From: Mengdong Lin <mengdong.lin(a)intel.com> This vendor-specific version number is optional. It will be written to the 'version' field of each block header of the binary toplogy data file. The vendor driver can check this number for further processing in kernel. The topology ABI version number is still stored in the 'abi' field of block headers. Signed-off-by: Mengdong Lin <mengdong.lin(a)intel.com> Reviewed-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/topology.h | 8 ++++++++ src/topology/builder.c | 6 +++--- src/topology/parser.c | 7 +++++++ 3 files changed, 18 insertions(+), 3 deletions(-) diff --git a/include/topology.h b/include/topology.h index 6ff8c5f..9b84bd9 100644 --- a/include/topology.h +++ b/include/topology.h @@ -690,6 +690,14 @@ int snd_tplg_build(snd_tplg_t *tplg, const char *outfile); */ int snd_tplg_set_manifest_data(snd_tplg_t *tplg, const void *data, int len); +/** + * \brief Set an optional vendor specific version number. + * \param tplg Topology instance. + * \param version Vendor specific version number. + * \return Zero on success, otherwise a negative error code + */ +int snd_tplg_set_version(snd_tplg_t *tplg, unsigned int version); + /* \} */ #ifdef __cplusplus diff --git a/src/topology/builder.c b/src/topology/builder.c index 3bccd44..2f01137 100644 --- a/src/topology/builder.c +++ b/src/topology/builder.c @@ -89,7 +89,7 @@ static int write_data_block(snd_tplg_t *tplg, int size, int tplg_type, /* write the header for this block */ ret = write_block_header(tplg, tplg_type, 0, - SND_SOC_TPLG_ABI_VERSION, 0, size, 1); + tplg->version, 0, size, 1); if (ret < 0) { SNDERR("error: failed to write %s block %d\n", obj_name, ret); return ret; @@ -125,7 +125,7 @@ static int write_elem_block(snd_tplg_t *tplg, } ret = write_block_header(tplg, tplg_type, vendor_type, - SND_SOC_TPLG_ABI_VERSION, 0, size, count); + tplg->version, 0, size, count); if (ret < 0) { SNDERR("error: failed to write %s block %d\n", obj_name, ret); @@ -243,7 +243,7 @@ static int write_manifest_data(snd_tplg_t *tplg) /* write the header for this block */ ret = write_block_header(tplg, SND_SOC_TPLG_TYPE_MANIFEST, 0, - SND_SOC_TPLG_ABI_VERSION, 0, + tplg->version, 0, sizeof(tplg->manifest) + tplg->manifest.priv.size, 1); if (ret < 0) { SNDERR("error: failed to write manifest block %d\n", ret); diff --git a/src/topology/parser.c b/src/topology/parser.c index 7ca244c..c0ab5e4 100644 --- a/src/topology/parser.c +++ b/src/topology/parser.c @@ -362,6 +362,13 @@ int snd_tplg_set_manifest_data(snd_tplg_t *tplg, const void *data, int len) return 0; } +int snd_tplg_set_version(snd_tplg_t *tplg, unsigned int version) +{ + tplg->version = version; + + return 0; +} + void snd_tplg_verbose(snd_tplg_t *tplg, int verbose) { tplg->verbose = verbose; -- 1.9.1

5 4

[alsa-devel] [PATCH - 1/2] HDA - Add SND_JACK_HEADSET and Headset jack kctl
by Raymond Yau 23 Sep '15

23 Sep '15

>From b3ed187912eccf922000aee09c04f9546b068925 Mon Sep 17 00:00:00 2001 From: Raymond Yau <superquad.vortex2(a)gmail.com> Date: Sun, 20 Sep 2015 22:10:18 +0800 Subject: [PATCH - 1/2] HDA - Add SND_JACK_HEADSET and Headset jack kctl add headset jack kctl for alsa jack type SND_JACK_HEADSET skip creation of Headset Mic Phantom Jack use Headset Playback Volume and Headset Playback Switch allow auto mic selection by using HP pin for jack sense Signed-off-by: Raymond Yau <superquad.vortex2(a)gmail.com> diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 24f9111..ce6da6f 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -2140,7 +2140,8 @@ static int create_hp_out_ctls(struct hda_codec *codec) struct hda_gen_spec *spec = codec->spec; return create_extra_outs(codec, spec->autocfg.hp_outs, spec->hp_paths, - "Headphone"); + spec->hs_mic_use_hp_sense ? + "Headset" : "Headphone"); } static int create_speaker_out_ctls(struct hda_codec *codec) @@ -4362,6 +4363,16 @@ void snd_hda_gen_line_automute(struct hda_codec *codec, } EXPORT_SYMBOL_GPL(snd_hda_gen_line_automute); +bool is_headset_mic(struct auto_pin_cfg *cfg, hda_nid_t pin) +{ + int i; + + for (i = 0; i < cfg->num_inputs; i++) + if (cfg->inputs[i].pin == pin) + return cfg->inputs[i].is_headset_mic; + return 0; +} + /** * snd_hda_gen_mic_autoswitch - standard mic auto-switch helper * @codec: the HDA codec @@ -4381,6 +4392,9 @@ void snd_hda_gen_mic_autoswitch(struct hda_codec *codec, /* don't detect pins retasked as outputs */ if (snd_hda_codec_get_pin_target(codec, pin) & AC_PINCTL_OUT_EN) continue; + if (spec->hs_mic_use_hp_sense && + is_headset_mic(&spec->autocfg, pin)) + pin = auto_cfg_hp_pins(&spec->autocfg)[0]; if (snd_hda_jack_detect_state(codec, pin) == HDA_JACK_PRESENT) { mux_select(codec, 0, spec->am_entry[i].idx); return; @@ -4662,7 +4676,9 @@ static int check_auto_mic_availability(struct hda_codec *codec) if (!spec->line_in_auto_switch && cfg->inputs[i].type != AUTO_PIN_MIC) return 0; /* only mic is allowed */ - if (!is_jack_detectable(codec, nid)) + if (!is_jack_detectable(codec, nid) && + !(cfg->inputs[i].is_headset_mic && + spec->hs_mic_use_hp_sense)) return 0; /* no unsol support */ break; } diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 56e4139..96f8214 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -236,6 +236,7 @@ struct hda_gen_spec { unsigned int indep_hp_enabled:1; /* independent HP enabled */ unsigned int have_aamix_ctl:1; unsigned int hp_mic_jack_modes:1; + unsigned int hs_mic_use_hp_sense:1; /* additional mute flags (only effective with auto_mute_via_amp=1) */ u64 mute_bits; diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index 366efbf..5fa8ace 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -19,6 +19,7 @@ #include "hda_local.h" #include "hda_auto_parser.h" #include "hda_jack.h" +#include "hda_generic.h" /** * is_jack_detectable - Check whether the given pin is jack-detectable @@ -350,11 +351,14 @@ EXPORT_SYMBOL_GPL(snd_hda_jack_report_sync); static int get_input_jack_type(struct hda_codec *codec, hda_nid_t nid) { unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid); + struct hda_gen_spec *spec = codec->spec; switch (get_defcfg_device(def_conf)) { case AC_JACK_LINE_OUT: case AC_JACK_SPEAKER: return SND_JACK_LINEOUT; case AC_JACK_HP_OUT: + if (spec->hs_mic_use_hp_sense) + return SND_JACK_HEADSET; return SND_JACK_HEADPHONE; case AC_JACK_SPDIF_OUT: case AC_JACK_DIG_OTHER_OUT: @@ -434,6 +438,7 @@ static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, char name[SNDRV_CTL_ELEM_ID_NAME_MAXLEN]; int err; bool phantom_jack; + struct hda_gen_spec *spec = codec->spec; if (!nid) return 0; @@ -446,8 +451,13 @@ static int add_jack_kctl(struct hda_codec *codec, hda_nid_t nid, if (base_name) strlcpy(name, base_name, sizeof(name)); - else + else { snd_hda_get_pin_label(codec, nid, cfg, name, sizeof(name), NULL); + if ((strcmp(name, "Headphone") == 0) && + (nid == auto_cfg_hp_pins(cfg)[0]) && + spec->hs_mic_use_hp_sense) + strlcpy(name, "Headset", sizeof(name)); + } if (phantom_jack) /* Example final name: "Internal Mic Phantom Jack" */ strncat(name, " Phantom", sizeof(name) - strlen(name) - 1); @@ -469,6 +479,7 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { const hda_nid_t *p; + struct hda_gen_spec *spec = codec->spec; int i, err; for (i = 0; i < cfg->num_inputs; i++) { @@ -482,6 +493,10 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg, "Headphone Mic"); } else + if (cfg->inputs[i].is_headset_mic && + spec->hs_mic_use_hp_sense) + err = 0; + else err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg, NULL); if (err < 0) -- 2.5.0

2 2

[alsa-devel] [PATCH utils] aplay: fix VU meter for S24_LE etc formats
by Ricard Wanderlof 23 Sep '15

23 Sep '15

When recording or playing back audio in a format where the number of significant bits is less than the physical width (e.g. S24_LE), the VU meter code needs to consider the number of significant bits in the samples rather than the physical sample width (e.g. 24 vs 32 bits). Otherwise the resulting VU meter display will be far too low and it will just indicate 0% all the time. Tested with a device supporting the S24_LE format. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- alsa-utils/aplay/aplay.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index e58e1bc..19b6a92 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -124,7 +124,7 @@ static int fatal_errors = 0; static int verbose = 0; static int vumeter = VUMETER_NONE; static int buffer_pos = 0; -static size_t bits_per_sample, bits_per_frame; +static size_t significant_bits_per_sample, bits_per_sample, bits_per_frame; static size_t chunk_bytes; static int test_position = 0; static int test_coef = 8; @@ -1344,6 +1344,7 @@ static void set_params(void) snd_pcm_dump(handle, log); bits_per_sample = snd_pcm_format_physical_width(hwparams.format); + significant_bits_per_sample = snd_pcm_format_width(hwparams.format); bits_per_frame = bits_per_sample * hwparams.channels; chunk_bytes = chunk_size * bits_per_frame / 8; audiobuf = realloc(audiobuf, chunk_bytes); @@ -1745,7 +1746,7 @@ static void compute_max_peak(u_char *data, size_t count) } return; } - max = 1 << (bits_per_sample-1); + max = 1 << (significant_bits_per_sample-1); if (max <= 0) max = 0x7fffffff; -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 1

[alsa-devel] [PATCH RESEND 1/4] sound/onyx: remove redundant case entry
by Yaowei Bai 23 Sep '15

23 Sep '15

Case ti->tag == 0 can be included in the last return, remove it. No functional change. Signed-off-by: Yaowei Bai <bywxiaobai(a)163.com> --- sound/aoa/codecs/onyx.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c index a04edff..a407e1f 100644 --- a/sound/aoa/codecs/onyx.c +++ b/sound/aoa/codecs/onyx.c @@ -683,7 +683,6 @@ static int onyx_usable(struct codec_info_item *cii, mutex_unlock(&onyx->mutex); switch (ti->tag) { - case 0: return 1; case 1: return analog_enabled; case 2: return spdif_enabled; } -- 1.9.1

1 3

[alsa-devel] [PATCH 1/4] sound/onyx: remove redundant case entry
by Yaowei Bai 23 Sep '15

23 Sep '15

Case ti->tag == 0 can be included in the last return, remove it. No functional change. Signed-off-by: Yaowei Bai <bywxiaobai(a)163.com> --- sound/aoa/codecs/onyx.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c index a04edff..a407e1f 100644 --- a/sound/aoa/codecs/onyx.c +++ b/sound/aoa/codecs/onyx.c @@ -683,7 +683,6 @@ static int onyx_usable(struct codec_info_item *cii, mutex_unlock(&onyx->mutex); switch (ti->tag) { - case 0: return 1; case 1: return analog_enabled; case 2: return spdif_enabled; } -- 1.9.1

2 7

[alsa-devel] [PATCH - alsa-utils 1/1] arecord: Remove only regular files
by Alexander Volkov 23 Sep '15

23 Sep '15

arecord removes a file before writing into it. It's not appropriate in some cases. For example, if you a pass a symlink to a file, then the symlink will be removed while the user expects to record into the symlink's target. Another case is recording into the device file. Some modems provide a tty device file as a voice device. And it's not possible to write into it under root with arecord, because it removes the device file. So check the type of a file before writing into it and remove only regular files. Signed-off-by: Alexander Volkov <a.volkov(a)rusbitech.ru> diff --git a/aplay/aplay.c b/aplay/aplay.c index 459f7dd..1b2cdfc 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -2929,6 +2929,7 @@ static void capture(char *orig_name) char *name = orig_name; /* current filename */ char namebuf[PATH_MAX+1]; off64_t count, rest; /* number of bytes to capture */ + struct stat statbuf; /* get number of bytes to capture */ count = calc_count(); @@ -2973,7 +2974,10 @@ static void capture(char *orig_name) } /* open a new file */ - remove(name); + if (!lstat(name, &statbuf)) { + if (S_ISREG(statbuf.st_mode)) + remove(name); + } fd = safe_open(name); if (fd < 0) { perror(name); -- 2.4.0

4 3

[alsa-devel] [PATCH alsa-lib] build: Do not try to detect cross-compiler
by Tanu Kaskinen 23 Sep '15

23 Sep '15

From: Khem Raj <raj.khem(a)gmail.com> cross compilers are passed via path may not be a gcc based cross compiler in such cases this check fails and try's to force gcc based cross compiler detection, This code is a convenience that limits the build system Signed-off-by: Khem Raj <raj.khem(a)gmail.com> --- configure.ac | 14 -------------- 1 file changed, 14 deletions(-) diff --git a/configure.ac b/configure.ac index a482b3e..a14e52d 100644 --- a/configure.ac +++ b/configure.ac @@ -27,20 +27,6 @@ AC_PREFIX_DEFAULT(/usr) dnl Checks for programs. -dnl try to gues cross-compiler if not set -if test "x$host" != "x$build" -a -z "`echo $CC | grep -e '-gcc'`"; -then - AC_MSG_CHECKING(for cross-compiler) - - which ${program_prefix}gcc >/dev/null 2>&1 && CC=${program_prefix}gcc - which ${host_cpu}-${host_os}-gcc >/dev/null 2>&1 \ - && CC=${host_cpu}-${host_os}-gcc - which ${host_cpu}-${host_vendor}-${host_os}-gcc >/dev/null 2>&1 \ - && CC=${host_cpu}-${host_vendor}-${host_os}-gcc - - AC_MSG_RESULT($CC) -fi - CFLAGS="$CFLAGS -D_GNU_SOURCE" -- 2.5.1

2 1

[alsa-devel] [PATCH] sound: oss: ad1848: Fix returned errno code in ad1848_init()
by Luis de Bethencourt 23 Sep '15

23 Sep '15

The driver is using -1 instead of the -ENOMEM defined macro to specify that a buffer allocation failed. Since the error number is propagated, the caller will get a -EPERM which is the wrong error condition. Smatch tool warning: ad1848_init() warn: returning -1 instead of -ENOMEM is sloppy Signed-off-by: Luis de Bethencourt <luisbg(a)osg.samsung.com> --- sound/oss/ad1848.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/oss/ad1848.c b/sound/oss/ad1848.c index 10c8de1..6b35656 100644 --- a/sound/oss/ad1848.c +++ b/sound/oss/ad1848.c @@ -1992,7 +1992,7 @@ int ad1848_init (char *name, struct resource *ports, int irq, int dma_playback, portc = kmalloc(sizeof(ad1848_port_info), GFP_KERNEL); if(portc==NULL) { release_region(devc->base, 4); - return -1; + return -ENOMEM; } if ((my_dev = sound_install_audiodrv(AUDIO_DRIVER_VERSION, -- 2.5.3

2 3

[alsa-devel] [PATCH v2- ALSA 3/3] HDA - add headset-sonicmaster and headset-multi-sonicmaster models
by Raymond Yau 23 Sep '15

23 Sep '15

add headset only model for ALC668 with Sonic Master Subwoofer add 625ms delay after headset is plugged for correct detection of headset type for ALC668 add headset multi model with Sonice Master Subwoofer update HD-Audio-Models.txt Signed-off-by: Raymond Yau <superquad.vortex2(a)gmail.com> diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index ec099d4..8baddac 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -55,7 +55,12 @@ ALC66x/67x/892 asus-mode7 ASUS asus-mode8 ASUS inv-dmic Inverted internal mic workaround + +ALC668 +====== dell-headset-multi Headset jack, which can also be used as mic-in + headset-sonicmaster Headset jack and SonicMaster Subwoofer + headset-multi-sonicmaster Headset, HP or Mic and SonicMaster Subwoofer ALC680 ====== diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a75b561..91de00c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4029,7 +4029,13 @@ static void alc_update_headset_mode(struct hda_codec *codec) PIN_VREFHIZ); } spec->current_headset_mode = new_headset_mode; - + if (codec->core.vendor_id == 0x10ec0668) { + if (spec->gen.hs_mic_use_hp_sense && spec->gen.auto_mic) { + if (new_headset_mode == ALC_HEADSET_MODE_HEADPHONE) + msleep(625); + snd_hda_gen_mic_autoswitch(codec, NULL); + }; + } snd_hda_gen_update_outputs(codec); } @@ -6071,6 +6077,42 @@ static void alc662_fixup_led_gpio1(struct hda_codec *codec, } } +static void alc668_fixup_asus_headset_only(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + const struct hda_pintbl hs_mic_pin[] = { + { 0x1b, 0x03a1113c }, /* headset mic */ + { } + }; + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_apply_pincfgs(codec, hs_mic_pin); + spec->gen.hs_mic_use_hp_sense = 1; + alc_fixup_disable_aamix(codec, fix, action); + break; + } +} + +static void alc668_fixup_sonicmaster(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + const struct hda_pintbl bass_pin[] = { + { 0x1a, 0x04110011 }, /* external subwoofer */ + { } + }; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_apply_pincfgs(codec, bass_pin); + break; + case HDA_FIXUP_ACT_BUILD: + alc_fixup_bass_chmap(codec, fix, action); + break; + } +} + static struct coef_fw alc668_coefs[] = { WRITE_COEF(0x01, 0xbebe), WRITE_COEF(0x02, 0xaaaa), WRITE_COEF(0x03, 0x0), WRITE_COEF(0x04, 0x0180), WRITE_COEF(0x06, 0x0), WRITE_COEF(0x07, 0x0f80), @@ -6130,6 +6172,9 @@ enum { ALC668_FIXUP_AUTO_MUTE, ALC668_FIXUP_DELL_DISABLE_AAMIX, ALC668_FIXUP_DELL_XPS13, + ALC668_FIXUP_HEADSET_ONLY, + ALC668_FIXUP_HEADSET_SONICMASTER, + ALC668_FIXUP_HEADSET_MULTI_SONICMASTER, }; static const struct hda_fixup alc662_fixups[] = { @@ -6370,6 +6415,24 @@ static const struct hda_fixup alc662_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_bass_chmap, }, + [ALC668_FIXUP_HEADSET_ONLY] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc668_fixup_asus_headset_only, + .chained = true, + .chain_id = ALC668_FIXUP_HEADSET_MODE, + }, + [ALC668_FIXUP_HEADSET_SONICMASTER] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc668_fixup_sonicmaster, + .chained = true, + .chain_id = ALC668_FIXUP_HEADSET_ONLY, + }, + [ALC668_FIXUP_HEADSET_MULTI_SONICMASTER] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc668_fixup_sonicmaster, + .chained = true, + .chain_id = ALC668_FIXUP_DELL_DISABLE_AAMIX, + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -6476,6 +6539,9 @@ static const struct hda_model_fixup alc662_fixup_models[] = { {.id = ALC662_FIXUP_ASUS_MODE8, .name = "asus-mode8"}, {.id = ALC662_FIXUP_INV_DMIC, .name = "inv-dmic"}, {.id = ALC668_FIXUP_DELL_MIC_NO_PRESENCE, .name = "dell-headset-multi"}, + {.id = ALC668_FIXUP_HEADSET_SONICMASTER, .name = "headset-sonicmaster"}, + {.id = ALC668_FIXUP_HEADSET_MULTI_SONICMASTER, + .name = "headset-multi-sonicmaster"}, {} }; -- 2.5.0

1 0

[alsa-devel] [PATCH v2- ALSA 2/3] HDA - create headset mic jack which use headphone pin sense
by Raymond Yau 23 Sep '15

23 Sep '15

- use __snd_hda_jack_add_kctl to create headset mic jack which use headphone pin sense - set headset mic jack as slave of headphone jack - allow non detectable headset mic pin and hs_mic_use_hp_sense pass check auto mic availability Signed-off-by: Raymond Yau <superquad.vortex2(a)gmail.com> diff --git a/sound/pci/hda/hda_generic.c b/sound/pci/hda/hda_generic.c index 24f9111..e8e4e34 100644 --- a/sound/pci/hda/hda_generic.c +++ b/sound/pci/hda/hda_generic.c @@ -4662,7 +4662,9 @@ static int check_auto_mic_availability(struct hda_codec *codec) if (!spec->line_in_auto_switch && cfg->inputs[i].type != AUTO_PIN_MIC) return 0; /* only mic is allowed */ - if (!is_jack_detectable(codec, nid)) + if (!is_jack_detectable(codec, nid) && + !(spec->hs_mic_use_hp_sense && + cfg->inputs[i].is_headset_mic)) return 0; /* no unsol support */ break; } diff --git a/sound/pci/hda/hda_generic.h b/sound/pci/hda/hda_generic.h index 56e4139..96f8214 100644 --- a/sound/pci/hda/hda_generic.h +++ b/sound/pci/hda/hda_generic.h @@ -236,6 +236,7 @@ struct hda_gen_spec { unsigned int indep_hp_enabled:1; /* independent HP enabled */ unsigned int have_aamix_ctl:1; unsigned int hp_mic_jack_modes:1; + unsigned int hs_mic_use_hp_sense:1; /* additional mute flags (only effective with auto_mute_via_amp=1) */ u64 mute_bits; diff --git a/sound/pci/hda/hda_jack.c b/sound/pci/hda/hda_jack.c index d81d65f..c29f05e 100644 --- a/sound/pci/hda/hda_jack.c +++ b/sound/pci/hda/hda_jack.c @@ -480,6 +480,10 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, const struct auto_pin_cfg *cfg) { const hda_nid_t *p; + struct hda_jack_tbl *jack; + struct hda_gen_spec *spec = codec->spec; + hda_nid_t hp_pin = auto_cfg_hp_pins(cfg)[0]; + hda_nid_t hs_mic_pin = 0; int i, err; for (i = 0; i < cfg->num_inputs; i++) { @@ -493,6 +497,13 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg, "Headphone Mic"); } else + if (cfg->inputs[i].is_headset_mic && + !is_jack_detectable(codec, cfg->inputs[i].pin) && + spec->hs_mic_use_hp_sense) { + hs_mic_pin = cfg->inputs[i].pin; + err = __snd_hda_jack_add_kctl(codec, cfg->inputs[i].pin, + "Headset Mic", false, hp_pin); + } else err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg, NULL); if (err < 0) @@ -510,6 +521,11 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, err = add_jack_kctl(codec, *p, cfg, NULL); if (err < 0) return err; + if (*p == hp_pin && spec->hs_mic_use_hp_sense && hs_mic_pin) { + jack = snd_hda_jack_tbl_get(codec, *p); + if (jack) + jack->slave_nid = hs_mic_pin; + } } for (i = 0, p = cfg->speaker_pins; i < cfg->speaker_outs; i++, p++) { if (*p == *cfg->line_out_pins) /* might be duplicated */ -- 2.5.0

1 0

[alsa-devel] [PATCH v2] ASoC: pxa: pxa2xx-ac97: fix dma requestor lines
by Robert Jarzmik 22 Sep '15

22 Sep '15

PCM receive and transmit DMA requestor lines were reverted, breaking the PCM playback interface for PXA platforms using the sound/soc/ variant instead of the sound/arm variant. The commit below shows the inversion in the requestor lines. Fixes: d65a14587a9b ("ASoC: pxa: use snd_dmaengine_dai_dma_data") Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- Since v1: rebase on Linus's master, not my dmaengine tip branch. --- sound/soc/pxa/pxa2xx-ac97.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index 1f6054650991..9e4b04e0fbd1 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -49,7 +49,7 @@ static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { .reset = pxa2xx_ac97_cold_reset, }; -static unsigned long pxa2xx_ac97_pcm_stereo_in_req = 12; +static unsigned long pxa2xx_ac97_pcm_stereo_in_req = 11; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, @@ -57,7 +57,7 @@ static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_in = { .filter_data = &pxa2xx_ac97_pcm_stereo_in_req, }; -static unsigned long pxa2xx_ac97_pcm_stereo_out_req = 11; +static unsigned long pxa2xx_ac97_pcm_stereo_out_req = 12; static struct snd_dmaengine_dai_dma_data pxa2xx_ac97_pcm_stereo_out = { .addr = __PREG(PCDR), .addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES, -- 2.1.4

2 1

[alsa-devel] [PATCH v3 00/21] Update ALSA, and au0828 drivers to use Managed Media Controller API
by Shuah Khan 22 Sep '15

22 Sep '15

This patch series updates ALSA driver, and au0828 core driver to use Managed Media controller API to share tuner. Please note that Managed Media Controller API and DVB and V4L2 drivers updates to use Media Controller API have been added in a prior patch series. Media Controller API is enhanced with two new interfaces to register and unregister entity_notify hooks to allow drivers to take appropriate actions when as new entities get added to the shared media device. Tested exclusion between digital, analog, and audio to ensure when tuner has an active link to DVB FE, analog, and audio will detect and honor the tuner busy conditions and vice versa. Changes since v2: http://www.spinics.net/lists/linux-media/msg91926.html 1. An important change in this patch series is made to ALSA to check for resources and hold in snd_usb_hw_params(), and release from snd_usb_hw_free(). This change fixed the lockdep warnings seen when resources were held in TRIGGER_START and released from TRIGGER_STOP which could run in IRQ context. I acknowledge Clemens Ladisch for suggesting the correct places to hold/free resources to avoid IRQ path complications. 2. With the above change, the patch series is simpler without the need to change the graph_mutex into a spinlock. 3. I split the patches up differently for easy reviews - no code bloat from v2. 4. A second important change is now the Bridge driver (au0828) owns and drives the graph creation as well as enabling and disabling tuner. It also keeps state information to avoid graph walks in enable_source and disable_source handler. I acknowledge Hans Verkuil for his suggestions and ideas for this change. History: This patch series has been updated to address comments from 3 previous versions of this series. Links to v3 version for reference are: https://www.mail-archive.com/linux-media%40vger.kernel.org/msg89491.html https://www.mail-archive.com/linux-media@vger.kernel.org/msg89492.html https://www.mail-archive.com/linux-media%40vger.kernel.org/msg89493.html Shuah Khan (21): Revert "[media] media: media controller entity framework enhancements for ALSA" media: Media Controller register/unregister entity_notify API media: Add ALSA Media Controller devnodes media: Media Controller enable/disable source handler API media: Media Controller fix to not let stream_count go negative media: Media Controller export non locking __media_entity_setup_link() media: Media Controller non-locking __media_entity_pipeline_start/stop() media: v4l-core add v4l_enable/disable_media_tuner() helper functions media: Move au8522_media_pads enum to au8522.h from au8522_priv.h media: au8522 change to create MC pad for ALSA Audio Out media: au0828 Use au8522_media_pads enum for pad defines media: au0828 fix au0828_create_media_graph() entity checks media: Change v4l-core to check for tuner availability media: dvb-frontend invoke enable/disable_source handlers media: au0828 video remove au0828_enable_analog_tuner() media: au0828 video change to use v4l_enable_media_tuner() media: au0828 change to use Managed Media Controller API media: au0828 change to register/unregister entity_notify hook media: au0828 implement enable_source and disable_source handlers media: media: dvb-frontend fix enable_source error legs sound/usb: Update ALSA driver to use Managed Media Controller API drivers/media/dvb-core/dvb_frontend.c | 142 ++----------- drivers/media/dvb-core/dvb_frontend.h | 3 + drivers/media/dvb-frontends/au8522.h | 8 + drivers/media/dvb-frontends/au8522_decoder.c | 1 + drivers/media/dvb-frontends/au8522_priv.h | 8 - drivers/media/media-device.c | 46 +++- drivers/media/media-entity.c | 64 ++++-- drivers/media/usb/au0828/au0828-core.c | 305 ++++++++++++++++++++++----- drivers/media/usb/au0828/au0828-video.c | 76 ++----- drivers/media/usb/au0828/au0828.h | 8 + drivers/media/v4l2-core/v4l2-dev.c | 27 +++ drivers/media/v4l2-core/v4l2-fh.c | 1 + drivers/media/v4l2-core/v4l2-ioctl.c | 29 +++ drivers/media/v4l2-core/videobuf2-core.c | 3 + include/media/media-device.h | 41 ++++ include/media/media-entity.h | 7 +- include/media/v4l2-dev.h | 4 + include/uapi/linux/media.h | 5 + sound/usb/Makefile | 15 +- sound/usb/card.c | 5 + sound/usb/card.h | 1 + sound/usb/media.c | 178 ++++++++++++++++ sound/usb/media.h | 51 +++++ sound/usb/pcm.c | 29 ++- sound/usb/quirks-table.h | 1 + sound/usb/quirks.c | 9 +- sound/usb/stream.c | 2 + sound/usb/usbaudio.h | 1 + 28 files changed, 790 insertions(+), 280 deletions(-) create mode 100644 sound/usb/media.c create mode 100644 sound/usb/media.h -- 2.1.4

1 21

[alsa-devel] [PATCH v3 1/2] ASoC: rt5645: Add dmi for Broadwell
by John Lin 22 Sep '15

22 Sep '15

Add DMI data for Buddy project. Signed-off-by: Bard Liao <bardliao(a)realtek.com> Signed-off-by: John Lin <john.lin(a)realtek.com> --- sound/soc/codecs/rt5645.c | 28 +++++++++++++++++++++++++++- 1 file changed, 27 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index dbc1d76..e830688 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3209,6 +3209,31 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { { } }; +static struct rt5645_platform_data buddy_platform_data = { + .dmic1_data_pin = RT5645_DMIC_DATA_GPIO5, + .dmic2_data_pin = RT5645_DMIC_DATA_IN2P, + .jd_mode = 3, +}; + +static int buddy_quirk_cb(const struct dmi_system_id *id) +{ + rt5645_pdata = &buddy_platform_data; + + return 1; +} + +static struct dmi_system_id dmi_platform_intel_broadwell[] __initdata = { + { + .ident = "Chrome Buddy", + .callback = buddy_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Buddy"), + }, + }, + { } +}; + + static int rt5645_parse_dt(struct rt5645_priv *rt5645, struct device *dev) { rt5645->pdata.in2_diff = device_property_read_bool(dev, @@ -3241,7 +3266,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, if (pdata) rt5645->pdata = *pdata; - else if (dmi_check_system(dmi_platform_intel_braswell)) + else if (dmi_check_system(dmi_platform_intel_braswell) || + dmi_check_system(dmi_platform_intel_broadwell)) rt5645->pdata = *rt5645_pdata; else rt5645_parse_dt(rt5645, &i2c->dev); -- 1.8.1.1.439.g50a6b54

2 2

[alsa-devel] [PATCH 1/2] ASoC: rt5645: Add dmi for Broadwell
by Bard Liao 22 Sep '15

22 Sep '15

Add DMI data for Buddy project. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 28 +++++++++++++++++++++++++++- 1 file changed, 27 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 66e9ce6..2fc437f 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3210,6 +3210,31 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { { } }; +static struct rt5645_platform_data buddy_platform_data = { + .dmic1_data_pin = RT5645_DMIC_DATA_GPIO5, + .dmic2_data_pin = RT5645_DMIC_DATA_IN2P, + .jd_mode = 3, +}; + +static int buddy_quirk_cb(const struct dmi_system_id *id) +{ + rt5645_pdata = &buddy_platform_data; + + return 1; +} + +static struct dmi_system_id dmi_platform_intel_broadwell[] __initdata = { + { + .ident = "Chrome Buddy", + .callback = buddy_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Buddy"), + }, + }, + { } +}; + + static int rt5645_parse_dt(struct rt5645_priv *rt5645, struct device *dev) { rt5645->pdata.in2_diff = device_property_read_bool(dev, @@ -3242,7 +3267,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, if (pdata) rt5645->pdata = *pdata; - else if (dmi_check_system(dmi_platform_intel_braswell)) + else if (dmi_check_system(dmi_platform_intel_braswell) || + dmi_check_system(dmi_platform_intel_broadwel) rt5645->pdata = *rt5645_pdata; else rt5645_parse_dt(rt5645, &i2c->dev); -- 1.8.1.1.439.g50a6b54

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[alsa-devel] Applied "MAINTAINERS: Update website and git repo for Wolfson Microelectronics" to the asoc tree
by Mark Brown 22 Sep '15

22 Sep '15

The patch MAINTAINERS: Update website and git repo for Wolfson Microelectronics has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From f0e03dbd2d61d991bdd2d76b4e84681fe3077176 Mon Sep 17 00:00:00 2001 From: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> Date: Tue, 22 Sep 2015 15:30:33 +0100 Subject: [PATCH] MAINTAINERS: Update website and git repo for Wolfson Microelectronics Support for Wolfson Microelectronics devices is now part of Cirrus Logic and the relevant parts of the old opensource.wolfsonmicro.com site have moved to the Cirrus Logic GitHub area. This patch updates the website and git repo links, and also removes an obsolete website link for the voltage and current drivers. Signed-off-by: Richard Fitzgerald <rf(a)opensource.wolfsonmicro.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- MAINTAINERS | 9 +++------ 1 file changed, 3 insertions(+), 6 deletions(-) diff --git a/MAINTAINERS b/MAINTAINERS index 7ba7ab7..dffea21 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -11239,7 +11239,6 @@ VOLTAGE AND CURRENT REGULATOR FRAMEWORK M: Liam Girdwood <lgirdwood(a)gmail.com> M: Mark Brown <broonie(a)kernel.org> L: linux-kernel(a)vger.kernel.org -W: http://opensource.wolfsonmicro.com/node/15 W: http://www.slimlogic.co.uk/?p=48 T: git git://git.kernel.org/pub/scm/linux/kernel/git/broonie/regulator.git S: Supported @@ -11368,17 +11367,15 @@ WM97XX TOUCHSCREEN DRIVERS M: Mark Brown <broonie(a)kernel.org> M: Liam Girdwood <lrg(a)slimlogic.co.uk> L: linux-input(a)vger.kernel.org -T: git git://opensource.wolfsonmicro.com/linux-2.6-touch -W: http://opensource.wolfsonmicro.com/node/7 +W: https://github.com/CirrusLogic/linux-drivers/wiki S: Supported F: drivers/input/touchscreen/*wm97* F: include/linux/wm97xx.h WOLFSON MICROELECTRONICS DRIVERS L: patches(a)opensource.wolfsonmicro.com -T: git git://opensource.wolfsonmicro.com/linux-2.6-asoc -T: git git://opensource.wolfsonmicro.com/linux-2.6-audioplus -W: http://opensource.wolfsonmicro.com/content/linux-drivers-wolfson-devices +T: git https://github.com/CirrusLogic/linux-drivers.git +W: https://github.com/CirrusLogic/linux-drivers/wiki S: Supported F: Documentation/hwmon/wm83?? F: arch/arm/mach-s3c64xx/mach-crag6410* -- 2.5.0

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Re: [alsa-devel] [PATCH 1/3] ASoC: codecs: Add da7219 codec driver
by Mark Brown 22 Sep '15

22 Sep '15

On Thu, Sep 17, 2015 at 05:01:14PM +0100, Adam Thomson wrote: > + do { > + statusa = snd_soc_read(codec, DA7219_ACCDET_STATUS_A); > + if (statusa & DA7219_MICBIAS_UP_STS_MASK) > + micbias_up = true; > + } while (!micbias_up); This could go into an inifinite loop. > +static void da7219_aad_hptest_work(struct work_struct *work) > +{ > + struct da7219_aad_priv *da7219_aad = > + container_of(work, struct da7219_aad_priv, hptest_work); > + struct snd_soc_codec *codec = da7219_aad->codec; > + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); > + > + u8 tonegen_cfg1, tonegen_cfg2, tonegen_onper; > + u16 tonegen_freq1, tonegen_freq_hptest; > + u8 hpl_gain, hpr_gain, dacl_gain, dacr_gain, dac_filters1, dac_filters4; > + u8 dac_filters5, cp_ctrl, routing_dac, dacl_ctrl, dacr_ctrl; > + u8 mixoutl_sel, mixoutr_sel, st_outfilt_1l, st_outfilt_1r; > + u8 mixoutl_ctrl, mixoutr_ctrl, hpl_ctrl, hpr_ctrl, accdet_cfg8; > + int report = 0; > + > + /* Save current settings */ This is obviously a massive reconfiguration of the device. I'm not seeing anything here which prevents userspace coming in and change the configuration while we're in this function, that would obviously have serious issues. I'm also wondering if this might be more elegantly implemented by going into cache bypass mode, doing the test and then using a cache resync to restore the initial configuration. That will at least avoid issues with updates adding a new register but not modifying it. > + if (statusa & DA7219_JACK_TYPE_STS_MASK) { > + report |= SND_JACK_HEADSET; > + mask |= SND_JACK_HEADSET | SND_JACK_LINEOUT; > + schedule_work(&da7219_aad->btn_det_work); > + } else { > + schedule_work(&da7219_aad->hptest_work); > + } Why are we scheduling work - we're already in thread context? > + /* Un-drive headphones/lineout */ > + snd_soc_update_bits(codec, DA7219_HP_R_CTRL, > + DA7219_HP_R_AMP_OE_MASK, 0); > + snd_soc_update_bits(codec, DA7219_HP_L_CTRL, > + DA7219_HP_L_AMP_OE_MASK, 0); This looks like DAPM? > +static enum da7219_aad_jack_ins_deb da7219_aad_of_jack_ins_deb(u32 val) > +{ > + switch (val) { > + case 5: > + return DA7219_AAD_JACK_INS_DEB_5MS; > + case 10: > + return DA7219_AAD_JACK_INS_DEB_10MS; > + case 20: > + return DA7219_AAD_JACK_INS_DEB_20MS; > + case 50: > + return DA7219_AAD_JACK_INS_DEB_50MS; > + case 100: > + return DA7219_AAD_JACK_INS_DEB_100MS; > + case 200: > + return DA7219_AAD_JACK_INS_DEB_200MS; > + case 500: > + return DA7219_AAD_JACK_INS_DEB_500MS; > + case 1000: > + return DA7219_AAD_JACK_INS_DEB_1S; > + default: > + return DA7219_AAD_JACK_INS_DEB_20MS; This isn't an error? > +/* Input/Output Enums */ > +static const char * const da7219_gain_ramp_rate_txt[] = { > + "nominal rate * 8", "nominal rate", "nominal rate / 8", > + "nominal rate / 16" > +}; The ALSA ABI generally capitalises words. > +/* ToneGen */ > +static int da7219_tonegen_freq_get(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) > +{ > + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); > + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); > + struct soc_mixer_control *mixer_ctrl = > + (struct soc_mixer_control *) kcontrol->private_value; > + unsigned int reg = mixer_ctrl->reg; > + u16 val; > + int ret; > + > + ret = regmap_bulk_read(da7219->regmap, reg, &val, sizeof(val)); > + if (ret) > + return ret; > + > + ucontrol->value.integer.value[0] = le16_to_cpu(val); This is *weird*. We do a bulk read for a single register using an API that returns CPU endian data then make it CPU endian (without any annotations on variables...). Why not use regmap_read()? Why swap? Why not use raw I/O? > +static int da7219_hpf_put(struct snd_kcontrol *kcontrol, > + struct snd_ctl_elem_value *ucontrol) > +{ > + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); > + struct soc_enum *enum_ctrl = (struct soc_enum *)kcontrol->private_value; > + unsigned int reg = enum_ctrl->reg; > + unsigned int sel = ucontrol->value.integer.value[0]; > + unsigned int bits; > + > + switch (sel) { > + case DA7219_HPF_MODE_DISABLED: > + bits = DA7219_HPF_DISABLED; > + break; > + case DA7219_HPF_MODE_AUDIO: > + bits = DA7219_HPF_AUDIO_EN; > + break; > + case DA7219_HPF_MODE_VOICE: > + bits = DA7219_HPF_VOICE_EN; > + break; > + default: > + return -EINVAL; > + } > + > + snd_soc_update_bits(codec, reg, DA7219_HPF_MODE_MASK, bits); > + > + return 0; > +} This looks like a standard enumeration with a simple mapping to the register map? > + > + /* ADCs */ > + SOC_SINGLE_TLV("ADC Volume", DA7219_ADC_L_GAIN, > + DA7219_ADC_L_DIGITAL_GAIN_SHIFT, > + DA7219_ADC_L_DIGITAL_GAIN_MAX, DA7219_NO_INVERT, > + da7219_adc_dig_gain_tlv), > + SOC_SINGLE("ADC Switch", DA7219_ADC_L_CTRL, > + DA7219_ADC_L_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX, > + DA7219_INVERT), > + SOC_SINGLE("ADC Gain Ramp Switch", DA7219_ADC_L_CTRL, > + DA7219_ADC_L_RAMP_EN_SHIFT, DA7219_SWITCH_EN_MAX, > + DA7219_NO_INVERT), Capture Digital rather than ADC. > + SOC_SINGLE_TLV("ALC Max Attenuation", DA7219_ALC_GAIN_LIMITS, > + DA7219_ALC_ATTEN_MAX_SHIFT, DA7219_ALC_ATTEN_GAIN_MAX, > + DA7219_NO_INVERT, da7219_alc_gain_tlv), > + SOC_SINGLE_TLV("ALC Max Gain", DA7219_ALC_GAIN_LIMITS, > + DA7219_ALC_GAIN_MAX_SHIFT, DA7219_ALC_ATTEN_GAIN_MAX, > + DA7219_NO_INVERT, da7219_alc_gain_tlv), > + SOC_SINGLE_RANGE_TLV("ALC Min Analog Gain", DA7219_ALC_ANA_GAIN_LIMITS, > + DA7219_ALC_ANA_GAIN_MIN_SHIFT, > + DA7219_ALC_ANA_GAIN_MIN, DA7219_ALC_ANA_GAIN_MAX, > + DA7219_NO_INVERT, da7219_alc_ana_gain_tlv), > + SOC_SINGLE_RANGE_TLV("ALC Max Analog Gain", DA7219_ALC_ANA_GAIN_LIMITS, > + DA7219_ALC_ANA_GAIN_MAX_SHIFT, > + DA7219_ALC_ANA_GAIN_MIN, DA7219_ALC_ANA_GAIN_MAX, > + DA7219_NO_INVERT, da7219_alc_ana_gain_tlv), Volume not Gain. > + SOC_SINGLE("ToneGen DTMF Key", DA7219_TONE_GEN_CFG1, > + DA7219_DTMF_REG_SHIFT, DA7219_DTMF_REG_MAX, > + DA7219_NO_INVERT), Should this be an enumeration with the DTMF digits in it (# and * aren't numbers)? > + /* DACs */ > + SOC_DOUBLE_R_TLV("DAC Volume", DA7219_DAC_L_GAIN, DA7219_DAC_R_GAIN, > + DA7219_DAC_L_DIGITAL_GAIN_SHIFT, > + DA7219_DAC_DIGITAL_GAIN_MAX, DA7219_NO_INVERT, > + da7219_dac_dig_gain_tlv), > + SOC_DOUBLE_R("DAC Switch", DA7219_DAC_L_CTRL, DA7219_DAC_R_CTRL, > + DA7219_DAC_L_MUTE_EN_SHIFT, DA7219_SWITCH_EN_MAX, > + DA7219_INVERT), > + SOC_DOUBLE_R("DAC Gain Ramp Switch", DA7219_DAC_L_CTRL, > + DA7219_DAC_R_CTRL, DA7219_DAC_L_RAMP_EN_SHIFT, > + DA7219_SWITCH_EN_MAX, DA7219_NO_INVERT), All the DAC controls should probably be Playback Digital instead. > + > +static int da7219_set_dai_sysclk(struct snd_soc_dai *codec_dai, > + int clk_id, unsigned int freq, int dir) > +{ > + struct snd_soc_codec *codec = codec_dai->codec; > + struct da7219_priv *da7219 = snd_soc_codec_get_drvdata(codec); > + > + if (da7219->mclk_rate == freq) > + return 0; Given that we also have source selection is this safe - should we also be checking the source? > + if (da7219->mclk) { > + freq = clk_round_rate(da7219->mclk, freq); > + clk_set_rate(da7219->mclk, freq); > + } Missing error checking. > + /* Internal LDO */ > + if (da7219->use_int_ldo) > + snd_soc_update_bits(codec, DA7219_LDO_CTRL, > + DA7219_LDO_EN_MASK, > + DA7219_LDO_EN_MASK); If there is an option to use an external supply I would expect to see the regulator API used to discover the external LDO (and ideally also to configure the integrated LDO). If the driver works outside the frameworks then it is likely this will lead to integration issues later on. > + /* Internal LDO */ > + da7219->use_int_ldo = pdata->use_internal_ldo; This is likely to lead to surprises for example... > + /* Check if MCLK provided, if not the clock is NULL */ > + da7219->mclk = devm_clk_get(codec->dev, "mclk"); > + if (IS_ERR(da7219->mclk)) { > + if (PTR_ERR(da7219->mclk) == -EPROBE_DEFER) > + return -EPROBE_DEFER; > + da7219->mclk = NULL; > + } This should specifically look for the error code that corresponds to a clock not being mapped and only continue silently in that case rather than silently accepting all failures to obtain a clock - if we silently accept all failures that will mean that actual errors will be ignored. > + /* > + * There are multiple control bits for the input mixer. > + * The following can be enabled now as it's not power related. > + */ > + snd_soc_update_bits(codec, DA7219_MIXIN_L_CTRL, > + DA7219_MIXIN_L_MIX_EN_MASK, > + DA7219_MIXIN_L_MIX_EN_MASK); So the chip designers just put these in for randomness? Fun. It'd be more idiomatic to do something like making these supply widgets so they're controlled via DAPM even if they don't matter much. > + else if (device_may_wakeup(codec->dev)) > + disable_irq_wake(da7219->aad->irq); You can use dev_pm_set_wake_irq() and skip having to manage this stuff explicitly in the driver.

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[alsa-devel] unable to create IPC shm instance; SND_PCM_DIRECT_MAGIC
by Mike Bjorge 22 Sep '15

22 Sep '15

I was wondering if anyone could explain the point of SND_PCM_DIRECT_MAGIC and dmix->shmptr->magic? Specifically, what it's doing in pcm_direct.c from this commit? http://git.alsa-project.org/?p=alsa-lib.git;a=blobdiff;f=src/pcm/pcm_direct… I am trying to use pcm_dmix.c, but this is occasionally failing first_instance = ret = snd_pcm_direct_shm_create_or_connect(dmix); if (ret < 0) { SNDERR("unable to create IPC shm instance"); goto _err; } because dmix->shmptr->magic is not correctly set to SND_PCM_DIRECT_MAGIC. Thanks for any information! -Mike

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[alsa-devel] [PATCH v2 0/9] Add DSP topology management for SKL
by Vinod Koul 21 Sep '15

21 Sep '15

The SKL driver does not code DSP topology in driver. With this series the ASoC topology framework is used to parse the topology information to create widgets, controls and route map for the FW topology. It adds routines for SKL DSP module configuration, compute resources for modules, initialize and bind the pipe modules. It uses the SKL IPC library added earlier to send IPC for initialize the module, bind/unbind modules. Last patch in this series removes the unused dais. changes in v2: - add more explanation in changelog and code - add explanation of MCPS - fix whitespace issues - remove skl widget check on event handler as it is redundant - add switch for depth - add else for direction based code - remove cast for NHLT blob query - make debug prints to error - remove dump info - clean some debug prints Jeeja KP (9): ASoC: Intel: Skylake: Add pipe and modules handlers ASoC: Intel: Skylake: Add module configuration helpers ASoC: Intel: Skylake: add DSP platform widget event handlers ASoC: Intel: Skylake: Add FE and BE hw_params handling ASoC: Intel: Skylake: Add topology core init and handlers ASoC: Intel: Skylake: Initialize and load DSP controls ASoC: Intel: Skylake: Add DSP support and enable it ASoC: Intel: Skylake: Initialize NHLT table ASoC: Intel: Skylake: Remove unused CPU dai's sound/soc/intel/skylake/Makefile | 3 +- sound/soc/intel/skylake/skl-pcm.c | 171 ++-- sound/soc/intel/skylake/skl-topology.c | 1262 ++++++++++++++++++++++++++ sound/soc/intel/skylake/skl-topology.h | 21 + sound/soc/intel/skylake/skl-tplg-interface.h | 78 ++ sound/soc/intel/skylake/skl.c | 29 +- sound/soc/intel/skylake/skl.h | 11 + 7 files changed, 1507 insertions(+), 68 deletions(-) create mode 100644 sound/soc/intel/skylake/skl-topology.c -- 1.9.1

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[alsa-devel] Intel SST on a Bay Trail tablet
by Michele Curti 21 Sep '15

21 Sep '15

> On Thu, 25 Jun 2015 22:17:17 +0530 Vinod Koul wrote: > > > On Thu, Jun 25, 2015 at 12:21:24PM +0200, Antonio Ospite wrote: > > > On Thu, 25 Jun 2015 11:20:46 +0530 > > > Vinod Koul <vinod.koul at intel.com> wrote: > > > > > > > On Wed, Jun 24, 2015 at 03:46:13PM +0530, Vinod Koul wrote: > > > > > On Tue, Apr 14, 2015 at 05:06:27PM +0300, Jarkko Nikula wrote: > > > [...] > > > > > > > > > > > > Which makes me thinking how does those strings describe the SSP port > > > > > > setup? E.g. do they reflect what port is actually used or could it > > > > > > be possible that those are just driver strings but firmware could > > > > > > have been tuned for SSP0? If I looked at earlier right, Teclast has > > > > > > the low pin-count Baytrail without SSP2 but I'm not sure about that. > > > > > The FW doesnt provide way to changes ports from driver in this, so this > > > > > doesnt mean much. If we are sure it is not ssp2, I can provide ssp1 fw for > > > > > test > > > > I ahve pushed latest BYT version we have as well as added binary for second > > > > SSP port. Please give it a try: > > > > > > > > https://git.kernel.org/cgit/linux/kernel/git/vkoul/firmware.git/commit/?h=b… > > > > > > > > > > Thanks Vinod, but I see these are in the ELF format, and the mainline > > > driver does not support this format, at least the last time I checked it > > > didn't. > > My bad, updated the patch now with right format > > > > https://git.kernel.org/cgit/linux/kernel/git/vkoul/firmware.git/commit/?h=b… > > > > (Adding Michele to CC) > > OK, so Jarkko was right: after a quick test I confirm that with > fw_sst_0f28_ssp0.bin I get sound; this is quite an improvement, > thanks! Damn, completely missed.. Hi Vinod, if it's possible, could you please generate a fw_sst_0f28_ssp1.bin also, so I can test my driver (mine was for the Asus X205TA laptop, where the codec is on SSP1 port)? Thanks, Michele

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[alsa-devel] [PATCH BAT V1 0/7] BAT: Add Basic Audio Tester command line tool
by han.lu＠intel.com 21 Sep '15

21 Sep '15

From: "Lu, Han" <han.lu(a)intel.com> BAT (Basic Audio Tester) is a simple command line utility intended to automate audio driver and sound server QA testing with no human interaction. BAT works by playing an audio stream and capturing the same stream in either a digital or analog loopback. It then compares the captured stream to the original to determine if the testcase passes or fails. The main idea of frequency detecting is: The analysis function reads data from wav file, run fft against the data to get magnitude of frequency vectors, and then calculates the average value and standard deviation of frequency vectors. After that, we define a threshold: threshold = 3 * standard_deviation + average_value Frequencies with amplitude larger than threshold will be recognized as a peak, and the frequency with largest peak value will be recognized as a detected frequency. BAT then compares the detected frequency to target frequency, to decide if the detecting passes or fails. BAT supports 4 working modes: 1. single line playback; 2. single line capture and analysis; 3. playback and capture in loop, and analyze captured data; 4. local analyze without actual playback or capture. BAT will check devices input by user to decide which mode to use. BAT will create threads for playback and record, and will run spectrum analysis for mode 2, 3 or 4. Lu, Han (7): BAT: Add initial functions BAT: Add common definitions and functions BAT: Add playback and record functions BAT: Add signal generator BAT: Add converting functions BAT: Add spectrum analysis functions BAT: Add Makefile and configures Makefile.am | 3 + bat/Makefile.am | 14 ++ bat/alsa.c | 618 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++ bat/alsa.h | 20 ++ bat/analyze.c | 314 ++++++++++++++++++++++++++++ bat/analyze.h | 16 ++ bat/bat.c | 608 +++++++++++++++++++++++++++++++++++++++++++++++++++++++ bat/common.c | 198 ++++++++++++++++++ bat/common.h | 169 ++++++++++++++++ bat/convert.c | 113 +++++++++++ bat/convert.h | 23 +++ bat/signal.c | 88 ++++++++ configure.ac | 19 +- 13 files changed, 2202 insertions(+), 1 deletion(-) create mode 100644 bat/Makefile.am create mode 100644 bat/alsa.c create mode 100644 bat/alsa.h create mode 100644 bat/analyze.c create mode 100644 bat/analyze.h create mode 100644 bat/bat.c create mode 100644 bat/common.c create mode 100644 bat/common.h create mode 100644 bat/convert.c create mode 100644 bat/convert.h create mode 100644 bat/signal.c -- 1.9.1

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[alsa-devel] [PATCH - 2/2] HDA - add headset-sonicmaster and headset-multi-sonicmaster models
by Raymond Yau 21 Sep '15

21 Sep '15

>From e97383136359c5d3ef93282888540bbb20939a9e Mon Sep 17 00:00:00 2001 From: Raymond Yau <superquad.vortex2(a)gmail.com> Date: Sun, 20 Sep 2015 22:18:56 +0800 Subject: [PATCH - 2/2] HDA - add headset-sonicmaster and headset-multi-sonicmaster models add headset only model for ALC668 with Sonic Master Subwoofer add 625ms delay after headset is plugged for correct detection of headset type for ALC668 add headset multi model with Sonice Master Subwoofer update HD-Audio-Models.txt Signed-off-by: Raymond Yau <superquad.vortex2(a)gmail.com> diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index ec099d4..8baddac 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -55,7 +55,12 @@ ALC66x/67x/892 asus-mode7 ASUS asus-mode8 ASUS inv-dmic Inverted internal mic workaround + +ALC668 +====== dell-headset-multi Headset jack, which can also be used as mic-in + headset-sonicmaster Headset jack and SonicMaster Subwoofer + headset-multi-sonicmaster Headset, HP or Mic and SonicMaster Subwoofer ALC680 ====== diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index a75b561..91de00c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -4029,7 +4029,13 @@ static void alc_update_headset_mode(struct hda_codec *codec) PIN_VREFHIZ); } spec->current_headset_mode = new_headset_mode; - + if (codec->core.vendor_id == 0x10ec0668) { + if (spec->gen.hs_mic_use_hp_sense && spec->gen.auto_mic) { + if (new_headset_mode == ALC_HEADSET_MODE_HEADPHONE) + msleep(625); + snd_hda_gen_mic_autoswitch(codec, NULL); + }; + } snd_hda_gen_update_outputs(codec); } @@ -6071,6 +6077,42 @@ static void alc662_fixup_led_gpio1(struct hda_codec *codec, } } +static void alc668_fixup_asus_headset_only(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + const struct hda_pintbl hs_mic_pin[] = { + { 0x1b, 0x03a1113c }, /* headset mic */ + { } + }; + struct alc_spec *spec = codec->spec; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_apply_pincfgs(codec, hs_mic_pin); + spec->gen.hs_mic_use_hp_sense = 1; + alc_fixup_disable_aamix(codec, fix, action); + break; + } +} + +static void alc668_fixup_sonicmaster(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + const struct hda_pintbl bass_pin[] = { + { 0x1a, 0x04110011 }, /* external subwoofer */ + { } + }; + + switch (action) { + case HDA_FIXUP_ACT_PRE_PROBE: + snd_hda_apply_pincfgs(codec, bass_pin); + break; + case HDA_FIXUP_ACT_BUILD: + alc_fixup_bass_chmap(codec, fix, action); + break; + } +} + static struct coef_fw alc668_coefs[] = { WRITE_COEF(0x01, 0xbebe), WRITE_COEF(0x02, 0xaaaa), WRITE_COEF(0x03, 0x0), WRITE_COEF(0x04, 0x0180), WRITE_COEF(0x06, 0x0), WRITE_COEF(0x07, 0x0f80), @@ -6130,6 +6172,9 @@ enum { ALC668_FIXUP_AUTO_MUTE, ALC668_FIXUP_DELL_DISABLE_AAMIX, ALC668_FIXUP_DELL_XPS13, + ALC668_FIXUP_HEADSET_ONLY, + ALC668_FIXUP_HEADSET_SONICMASTER, + ALC668_FIXUP_HEADSET_MULTI_SONICMASTER, }; static const struct hda_fixup alc662_fixups[] = { @@ -6370,6 +6415,24 @@ static const struct hda_fixup alc662_fixups[] = { .type = HDA_FIXUP_FUNC, .v.func = alc_fixup_bass_chmap, }, + [ALC668_FIXUP_HEADSET_ONLY] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc668_fixup_asus_headset_only, + .chained = true, + .chain_id = ALC668_FIXUP_HEADSET_MODE, + }, + [ALC668_FIXUP_HEADSET_SONICMASTER] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc668_fixup_sonicmaster, + .chained = true, + .chain_id = ALC668_FIXUP_HEADSET_ONLY, + }, + [ALC668_FIXUP_HEADSET_MULTI_SONICMASTER] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc668_fixup_sonicmaster, + .chained = true, + .chain_id = ALC668_FIXUP_DELL_DISABLE_AAMIX, + }, }; static const struct snd_pci_quirk alc662_fixup_tbl[] = { @@ -6476,6 +6539,9 @@ static const struct hda_model_fixup alc662_fixup_models[] = { {.id = ALC662_FIXUP_ASUS_MODE8, .name = "asus-mode8"}, {.id = ALC662_FIXUP_INV_DMIC, .name = "inv-dmic"}, {.id = ALC668_FIXUP_DELL_MIC_NO_PRESENCE, .name = "dell-headset-multi"}, + {.id = ALC668_FIXUP_HEADSET_SONICMASTER, .name = "headset-sonicmaster"}, + {.id = ALC668_FIXUP_HEADSET_MULTI_SONICMASTER, + .name = "headset-multi-sonicmaster"}, {} }; -- 2.5.0

1 0

[alsa-devel] [PATCH v2 1/2] ASoC: rt5645: Add dmi for Broadwell
by Bard Liao 20 Sep '15

20 Sep '15

Add DMI data for Buddy project. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 28 +++++++++++++++++++++++++++- 1 file changed, 27 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 66e9ce6..2fc437f 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3210,6 +3210,31 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { { } }; +static struct rt5645_platform_data buddy_platform_data = { + .dmic1_data_pin = RT5645_DMIC_DATA_GPIO5, + .dmic2_data_pin = RT5645_DMIC_DATA_IN2P, + .jd_mode = 3, +}; + +static int buddy_quirk_cb(const struct dmi_system_id *id) +{ + rt5645_pdata = &buddy_platform_data; + + return 1; +} + +static struct dmi_system_id dmi_platform_intel_broadwell[] __initdata = { + { + .ident = "Chrome Buddy", + .callback = buddy_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Buddy"), + }, + }, + { } +}; + + static int rt5645_parse_dt(struct rt5645_priv *rt5645, struct device *dev) { rt5645->pdata.in2_diff = device_property_read_bool(dev, @@ -3242,7 +3267,8 @@ static int rt5645_i2c_probe(struct i2c_client *i2c, if (pdata) rt5645->pdata = *pdata; - else if (dmi_check_system(dmi_platform_intel_braswell)) + else if (dmi_check_system(dmi_platform_intel_braswell) || + dmi_check_system(dmi_platform_intel_broadwel) rt5645->pdata = *rt5645_pdata; else rt5645_parse_dt(rt5645, &i2c->dev); -- 1.8.1.1.439.g50a6b54

2 2

[alsa-devel] [PATCH v3 1/2] mfd: arizona: Add register bits to support the ANC block
by Charles Keepax 20 Sep '15

20 Sep '15

Some Arizona devices have a hardware ANC block present. This patch adds the registers necessary to configure this hardware block. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Changes since v2: - Added missing default for 0xF0A Thanks, Charles drivers/mfd/wm5110-tables.c | 186 +++++++++++++++++++++++++++++++++ include/linux/mfd/arizona/registers.h | 70 ++++++++++++ 2 files changed, 256 insertions(+), 0 deletions(-) diff --git a/drivers/mfd/wm5110-tables.c b/drivers/mfd/wm5110-tables.c index dae04dd..f154991 100644 --- a/drivers/mfd/wm5110-tables.c +++ b/drivers/mfd/wm5110-tables.c @@ -1637,6 +1637,185 @@ static const struct reg_default wm5110_reg_default[] = { { 0x00000EF8, 0x0000 }, /* R3832 - ISRC 3 CTRL 3 */ { 0x00000F00, 0x0000 }, /* R3840 - Clock Control */ { 0x00000F01, 0x0000 }, /* R3841 - ANC_SRC */ + { 0x00000F08, 0x001c }, /* R3848 - ANC Coefficient */ + { 0x00000F09, 0x0000 }, /* R3849 - ANC Coefficient */ + { 0x00000F0A, 0x0000 }, /* R3850 - ANC Coefficient */ + { 0x00000F0B, 0x0000 }, /* R3851 - ANC Coefficient */ + { 0x00000F0C, 0x0000 }, /* R3852 - ANC Coefficient */ + { 0x00000F0D, 0x0000 }, /* R3853 - ANC Coefficient */ + { 0x00000F0E, 0x0000 }, /* R3854 - ANC Coefficient */ + { 0x00000F0F, 0x0000 }, /* R3855 - ANC Coefficient */ + { 0x00000F10, 0x0000 }, /* R3856 - ANC Coefficient */ + { 0x00000F11, 0x0000 }, /* R3857 - ANC Coefficient */ + { 0x00000F12, 0x0000 }, /* R3858 - ANC Coefficient */ + { 0x00000F15, 0x0000 }, /* R3861 - FCL Filter Control */ + { 0x00000F17, 0x0004 }, /* R3863 - FCL ADC Reformatter Control */ + { 0x00000F18, 0x0004 }, /* R3864 - ANC Coefficient */ + { 0x00000F19, 0x0002 }, /* R3865 - ANC Coefficient */ + { 0x00000F1A, 0x0000 }, /* R3866 - ANC Coefficient */ + { 0x00000F1B, 0x0010 }, /* R3867 - ANC Coefficient */ + { 0x00000F1C, 0x0000 }, /* R3868 - ANC Coefficient */ + { 0x00000F1D, 0x0000 }, /* R3869 - ANC Coefficient */ + { 0x00000F1E, 0x0000 }, /* R3870 - ANC Coefficient */ + { 0x00000F1F, 0x0000 }, /* R3871 - ANC Coefficient */ + { 0x00000F20, 0x0000 }, /* R3872 - ANC Coefficient */ + { 0x00000F21, 0x0000 }, /* R3873 - ANC Coefficient */ + { 0x00000F22, 0x0000 }, /* R3874 - ANC Coefficient */ + { 0x00000F23, 0x0000 }, /* R3875 - ANC Coefficient */ + { 0x00000F24, 0x0000 }, /* R3876 - ANC Coefficient */ + { 0x00000F25, 0x0000 }, /* R3877 - ANC Coefficient */ + { 0x00000F26, 0x0000 }, /* R3878 - ANC Coefficient */ + { 0x00000F27, 0x0000 }, /* R3879 - ANC Coefficient */ + { 0x00000F28, 0x0000 }, /* R3880 - ANC Coefficient */ + { 0x00000F29, 0x0000 }, /* R3881 - ANC Coefficient */ + { 0x00000F2A, 0x0000 }, /* R3882 - ANC Coefficient */ + { 0x00000F2B, 0x0000 }, /* R3883 - ANC Coefficient */ + { 0x00000F2C, 0x0000 }, /* R3884 - ANC Coefficient */ + { 0x00000F2D, 0x0000 }, /* R3885 - ANC Coefficient */ + { 0x00000F2E, 0x0000 }, /* R3886 - ANC Coefficient */ + { 0x00000F2F, 0x0000 }, /* R3887 - ANC Coefficient */ + { 0x00000F30, 0x0000 }, /* R3888 - ANC Coefficient */ + { 0x00000F31, 0x0000 }, /* R3889 - ANC Coefficient */ + { 0x00000F32, 0x0000 }, /* R3890 - ANC Coefficient */ + { 0x00000F33, 0x0000 }, /* R3891 - ANC Coefficient */ + { 0x00000F34, 0x0000 }, /* R3892 - ANC Coefficient */ + { 0x00000F35, 0x0000 }, /* R3893 - ANC Coefficient */ + { 0x00000F36, 0x0000 }, /* R3894 - ANC Coefficient */ + { 0x00000F37, 0x0000 }, /* R3895 - ANC Coefficient */ + { 0x00000F38, 0x0000 }, /* R3896 - ANC Coefficient */ + { 0x00000F39, 0x0000 }, /* R3897 - ANC Coefficient */ + { 0x00000F3A, 0x0000 }, /* R3898 - ANC Coefficient */ + { 0x00000F3B, 0x0000 }, /* R3899 - ANC Coefficient */ + { 0x00000F3C, 0x0000 }, /* R3900 - ANC Coefficient */ + { 0x00000F3D, 0x0000 }, /* R3901 - ANC Coefficient */ + { 0x00000F3E, 0x0000 }, /* R3902 - ANC Coefficient */ + { 0x00000F3F, 0x0000 }, /* R3903 - ANC Coefficient */ + { 0x00000F40, 0x0000 }, /* R3904 - ANC Coefficient */ + { 0x00000F41, 0x0000 }, /* R3905 - ANC Coefficient */ + { 0x00000F42, 0x0000 }, /* R3906 - ANC Coefficient */ + { 0x00000F43, 0x0000 }, /* R3907 - ANC Coefficient */ + { 0x00000F44, 0x0000 }, /* R3908 - ANC Coefficient */ + { 0x00000F45, 0x0000 }, /* R3909 - ANC Coefficient */ + { 0x00000F46, 0x0000 }, /* R3910 - ANC Coefficient */ + { 0x00000F47, 0x0000 }, /* R3911 - ANC Coefficient */ + { 0x00000F48, 0x0000 }, /* R3912 - ANC Coefficient */ + { 0x00000F49, 0x0000 }, /* R3913 - ANC Coefficient */ + { 0x00000F4A, 0x0000 }, /* R3914 - ANC Coefficient */ + { 0x00000F4B, 0x0000 }, /* R3915 - ANC Coefficient */ + { 0x00000F4C, 0x0000 }, /* R3916 - ANC Coefficient */ + { 0x00000F4D, 0x0000 }, /* R3917 - ANC Coefficient */ + { 0x00000F4E, 0x0000 }, /* R3918 - ANC Coefficient */ + { 0x00000F4F, 0x0000 }, /* R3919 - ANC Coefficient */ + { 0x00000F50, 0x0000 }, /* R3920 - ANC Coefficient */ + { 0x00000F51, 0x0000 }, /* R3921 - ANC Coefficient */ + { 0x00000F52, 0x0000 }, /* R3922 - ANC Coefficient */ + { 0x00000F53, 0x0000 }, /* R3923 - ANC Coefficient */ + { 0x00000F54, 0x0000 }, /* R3924 - ANC Coefficient */ + { 0x00000F55, 0x0000 }, /* R3925 - ANC Coefficient */ + { 0x00000F56, 0x0000 }, /* R3926 - ANC Coefficient */ + { 0x00000F57, 0x0000 }, /* R3927 - ANC Coefficient */ + { 0x00000F58, 0x0000 }, /* R3928 - ANC Coefficient */ + { 0x00000F59, 0x0000 }, /* R3929 - ANC Coefficient */ + { 0x00000F5A, 0x0000 }, /* R3930 - ANC Coefficient */ + { 0x00000F5B, 0x0000 }, /* R3931 - ANC Coefficient */ + { 0x00000F5C, 0x0000 }, /* R3932 - ANC Coefficient */ + { 0x00000F5D, 0x0000 }, /* R3933 - ANC Coefficient */ + { 0x00000F5E, 0x0000 }, /* R3934 - ANC Coefficient */ + { 0x00000F5F, 0x0000 }, /* R3935 - ANC Coefficient */ + { 0x00000F60, 0x0000 }, /* R3936 - ANC Coefficient */ + { 0x00000F61, 0x0000 }, /* R3937 - ANC Coefficient */ + { 0x00000F62, 0x0000 }, /* R3938 - ANC Coefficient */ + { 0x00000F63, 0x0000 }, /* R3939 - ANC Coefficient */ + { 0x00000F64, 0x0000 }, /* R3940 - ANC Coefficient */ + { 0x00000F65, 0x0000 }, /* R3941 - ANC Coefficient */ + { 0x00000F66, 0x0000 }, /* R3942 - ANC Coefficient */ + { 0x00000F67, 0x0000 }, /* R3943 - ANC Coefficient */ + { 0x00000F68, 0x0000 }, /* R3944 - ANC Coefficient */ + { 0x00000F69, 0x0000 }, /* R3945 - ANC Coefficient */ + { 0x00000F70, 0x0000 }, /* R3952 - FCR Filter Control */ + { 0x00000F72, 0x0004 }, /* R3954 - FCR ADC Reformatter Control */ + { 0x00000F73, 0x0004 }, /* R3955 - ANC Coefficient */ + { 0x00000F74, 0x0002 }, /* R3956 - ANC Coefficient */ + { 0x00000F75, 0x0000 }, /* R3957 - ANC Coefficient */ + { 0x00000F76, 0x0010 }, /* R3958 - ANC Coefficient */ + { 0x00000F77, 0x0000 }, /* R3959 - ANC Coefficient */ + { 0x00000F78, 0x0000 }, /* R3960 - ANC Coefficient */ + { 0x00000F79, 0x0000 }, /* R3961 - ANC Coefficient */ + { 0x00000F7A, 0x0000 }, /* R3962 - ANC Coefficient */ + { 0x00000F7B, 0x0000 }, /* R3963 - ANC Coefficient */ + { 0x00000F7C, 0x0000 }, /* R3964 - ANC Coefficient */ + { 0x00000F7D, 0x0000 }, /* R3965 - ANC Coefficient */ + { 0x00000F7E, 0x0000 }, /* R3966 - ANC Coefficient */ + { 0x00000F7F, 0x0000 }, /* R3967 - ANC Coefficient */ + { 0x00000F80, 0x0000 }, /* R3968 - ANC Coefficient */ + { 0x00000F81, 0x0000 }, /* R3969 - ANC Coefficient */ + { 0x00000F82, 0x0000 }, /* R3970 - ANC Coefficient */ + { 0x00000F83, 0x0000 }, /* R3971 - ANC Coefficient */ + { 0x00000F84, 0x0000 }, /* R3972 - ANC Coefficient */ + { 0x00000F85, 0x0000 }, /* R3973 - ANC Coefficient */ + { 0x00000F86, 0x0000 }, /* R3974 - ANC Coefficient */ + { 0x00000F87, 0x0000 }, /* R3975 - ANC Coefficient */ + { 0x00000F88, 0x0000 }, /* R3976 - ANC Coefficient */ + { 0x00000F89, 0x0000 }, /* R3977 - ANC Coefficient */ + { 0x00000F8A, 0x0000 }, /* R3978 - ANC Coefficient */ + { 0x00000F8B, 0x0000 }, /* R3979 - ANC Coefficient */ + { 0x00000F8C, 0x0000 }, /* R3980 - ANC Coefficient */ + { 0x00000F8D, 0x0000 }, /* R3981 - ANC Coefficient */ + { 0x00000F8E, 0x0000 }, /* R3982 - ANC Coefficient */ + { 0x00000F8F, 0x0000 }, /* R3983 - ANC Coefficient */ + { 0x00000F90, 0x0000 }, /* R3984 - ANC Coefficient */ + { 0x00000F91, 0x0000 }, /* R3985 - ANC Coefficient */ + { 0x00000F92, 0x0000 }, /* R3986 - ANC Coefficient */ + { 0x00000F93, 0x0000 }, /* R3987 - ANC Coefficient */ + { 0x00000F94, 0x0000 }, /* R3988 - ANC Coefficient */ + { 0x00000F95, 0x0000 }, /* R3989 - ANC Coefficient */ + { 0x00000F96, 0x0000 }, /* R3990 - ANC Coefficient */ + { 0x00000F97, 0x0000 }, /* R3991 - ANC Coefficient */ + { 0x00000F98, 0x0000 }, /* R3992 - ANC Coefficient */ + { 0x00000F99, 0x0000 }, /* R3993 - ANC Coefficient */ + { 0x00000F9A, 0x0000 }, /* R3994 - ANC Coefficient */ + { 0x00000F9B, 0x0000 }, /* R3995 - ANC Coefficient */ + { 0x00000F9C, 0x0000 }, /* R3996 - ANC Coefficient */ + { 0x00000F9D, 0x0000 }, /* R3997 - ANC Coefficient */ + { 0x00000F9E, 0x0000 }, /* R3998 - ANC Coefficient */ + { 0x00000F9F, 0x0000 }, /* R3999 - ANC Coefficient */ + { 0x00000FA0, 0x0000 }, /* R4000 - ANC Coefficient */ + { 0x00000FA1, 0x0000 }, /* R4001 - ANC Coefficient */ + { 0x00000FA2, 0x0000 }, /* R4002 - ANC Coefficient */ + { 0x00000FA3, 0x0000 }, /* R4003 - ANC Coefficient */ + { 0x00000FA4, 0x0000 }, /* R4004 - ANC Coefficient */ + { 0x00000FA5, 0x0000 }, /* R4005 - ANC Coefficient */ + { 0x00000FA6, 0x0000 }, /* R4006 - ANC Coefficient */ + { 0x00000FA7, 0x0000 }, /* R4007 - ANC Coefficient */ + { 0x00000FA8, 0x0000 }, /* R4008 - ANC Coefficient */ + { 0x00000FA9, 0x0000 }, /* R4009 - ANC Coefficient */ + { 0x00000FAA, 0x0000 }, /* R4010 - ANC Coefficient */ + { 0x00000FAB, 0x0000 }, /* R4011 - ANC Coefficient */ + { 0x00000FAC, 0x0000 }, /* R4012 - ANC Coefficient */ + { 0x00000FAD, 0x0000 }, /* R4013 - ANC Coefficient */ + { 0x00000FAE, 0x0000 }, /* R4014 - ANC Coefficient */ + { 0x00000FAF, 0x0000 }, /* R4015 - ANC Coefficient */ + { 0x00000FB0, 0x0000 }, /* R4016 - ANC Coefficient */ + { 0x00000FB1, 0x0000 }, /* R4017 - ANC Coefficient */ + { 0x00000FB2, 0x0000 }, /* R4018 - ANC Coefficient */ + { 0x00000FB3, 0x0000 }, /* R4019 - ANC Coefficient */ + { 0x00000FB4, 0x0000 }, /* R4020 - ANC Coefficient */ + { 0x00000FB5, 0x0000 }, /* R4021 - ANC Coefficient */ + { 0x00000FB6, 0x0000 }, /* R4022 - ANC Coefficient */ + { 0x00000FB7, 0x0000 }, /* R4023 - ANC Coefficient */ + { 0x00000FB8, 0x0000 }, /* R4024 - ANC Coefficient */ + { 0x00000FB9, 0x0000 }, /* R4025 - ANC Coefficient */ + { 0x00000FBA, 0x0000 }, /* R4026 - ANC Coefficient */ + { 0x00000FBB, 0x0000 }, /* R4027 - ANC Coefficient */ + { 0x00000FBC, 0x0000 }, /* R4028 - ANC Coefficient */ + { 0x00000FBD, 0x0000 }, /* R4029 - ANC Coefficient */ + { 0x00000FBE, 0x0000 }, /* R4030 - ANC Coefficient */ + { 0x00000FBF, 0x0000 }, /* R4031 - ANC Coefficient */ + { 0x00000FC0, 0x0000 }, /* R4032 - ANC Coefficient */ + { 0x00000FC1, 0x0000 }, /* R4033 - ANC Coefficient */ + { 0x00000FC2, 0x0000 }, /* R4034 - ANC Coefficient */ + { 0x00000FC3, 0x0000 }, /* R4035 - ANC Coefficient */ + { 0x00000FC4, 0x0000 }, /* R4036 - ANC Coefficient */ { 0x00001100, 0x0010 }, /* R4352 - DSP1 Control 1 */ { 0x00001101, 0x0000 }, /* R4353 - DSP1 Clocking 1 */ { 0x00001200, 0x0010 }, /* R4608 - DSP2 Control 1 */ @@ -2716,6 +2895,13 @@ static bool wm5110_readable_register(struct device *dev, unsigned int reg) case ARIZONA_CLOCK_CONTROL: case ARIZONA_ANC_SRC: case ARIZONA_DSP_STATUS: + case ARIZONA_ANC_COEFF_START ... ARIZONA_ANC_COEFF_END: + case ARIZONA_FCL_FILTER_CONTROL: + case ARIZONA_FCL_ADC_REFORMATTER_CONTROL: + case ARIZONA_FCL_COEFF_START ... ARIZONA_FCL_COEFF_END: + case ARIZONA_FCR_FILTER_CONTROL: + case ARIZONA_FCR_ADC_REFORMATTER_CONTROL: + case ARIZONA_FCR_COEFF_START ... ARIZONA_FCR_COEFF_END: case ARIZONA_DSP1_CONTROL_1: case ARIZONA_DSP1_CLOCKING_1: case ARIZONA_DSP1_STATUS_1: diff --git a/include/linux/mfd/arizona/registers.h b/include/linux/mfd/arizona/registers.h index 11affb3..fdb12bf 100644 --- a/include/linux/mfd/arizona/registers.h +++ b/include/linux/mfd/arizona/registers.h @@ -1050,6 +1050,16 @@ #define ARIZONA_CLOCK_CONTROL 0xF00 #define ARIZONA_ANC_SRC 0xF01 #define ARIZONA_DSP_STATUS 0xF02 +#define ARIZONA_ANC_COEFF_START 0xF08 +#define ARIZONA_ANC_COEFF_END 0xF12 +#define ARIZONA_FCL_FILTER_CONTROL 0xF15 +#define ARIZONA_FCL_ADC_REFORMATTER_CONTROL 0xF17 +#define ARIZONA_FCL_COEFF_START 0xF18 +#define ARIZONA_FCL_COEFF_END 0xF69 +#define ARIZONA_FCR_FILTER_CONTROL 0xF70 +#define ARIZONA_FCR_ADC_REFORMATTER_CONTROL 0xF72 +#define ARIZONA_FCR_COEFF_START 0xF73 +#define ARIZONA_FCR_COEFF_END 0xFC4 #define ARIZONA_DSP1_CONTROL_1 0x1100 #define ARIZONA_DSP1_CLOCKING_1 0x1101 #define ARIZONA_DSP1_STATUS_1 0x1104 @@ -7823,6 +7833,66 @@ #define ARIZONA_ISRC3_NOTCH_ENA_WIDTH 1 /* ISRC3_NOTCH_ENA */ /* + * R3840 (0xF00) - Clock Control + */ +#define ARIZONA_EXT_NG_SEL_CLR 0x0080 /* EXT_NG_SEL_CLR */ +#define ARIZONA_EXT_NG_SEL_CLR_MASK 0x0080 /* EXT_NG_SEL_CLR */ +#define ARIZONA_EXT_NG_SEL_CLR_SHIFT 7 /* EXT_NG_SEL_CLR */ +#define ARIZONA_EXT_NG_SEL_CLR_WIDTH 1 /* EXT_NG_SEL_CLR */ +#define ARIZONA_EXT_NG_SEL_SET 0x0040 /* EXT_NG_SEL_SET */ +#define ARIZONA_EXT_NG_SEL_SET_MASK 0x0040 /* EXT_NG_SEL_SET */ +#define ARIZONA_EXT_NG_SEL_SET_SHIFT 6 /* EXT_NG_SEL_SET */ +#define ARIZONA_EXT_NG_SEL_SET_WIDTH 1 /* EXT_NG_SEL_SET */ +#define ARIZONA_CLK_R_ENA_CLR 0x0020 /* CLK_R_ENA_CLR */ +#define ARIZONA_CLK_R_ENA_CLR_MASK 0x0020 /* CLK_R_ENA_CLR */ +#define ARIZONA_CLK_R_ENA_CLR_SHIFT 5 /* CLK_R_ENA_CLR */ +#define ARIZONA_CLK_R_ENA_CLR_WIDTH 1 /* CLK_R_ENA_CLR */ +#define ARIZONA_CLK_R_ENA_SET 0x0010 /* CLK_R_ENA_SET */ +#define ARIZONA_CLK_R_ENA_SET_MASK 0x0010 /* CLK_R_ENA_SET */ +#define ARIZONA_CLK_R_ENA_SET_SHIFT 4 /* CLK_R_ENA_SET */ +#define ARIZONA_CLK_R_ENA_SET_WIDTH 1 /* CLK_R_ENA_SET */ +#define ARIZONA_CLK_NG_ENA_CLR 0x0008 /* CLK_NG_ENA_CLR */ +#define ARIZONA_CLK_NG_ENA_CLR_MASK 0x0008 /* CLK_NG_ENA_CLR */ +#define ARIZONA_CLK_NG_ENA_CLR_SHIFT 3 /* CLK_NG_ENA_CLR */ +#define ARIZONA_CLK_NG_ENA_CLR_WIDTH 1 /* CLK_NG_ENA_CLR */ +#define ARIZONA_CLK_NG_ENA_SET 0x0004 /* CLK_NG_ENA_SET */ +#define ARIZONA_CLK_NG_ENA_SET_MASK 0x0004 /* CLK_NG_ENA_SET */ +#define ARIZONA_CLK_NG_ENA_SET_SHIFT 2 /* CLK_NG_ENA_SET */ +#define ARIZONA_CLK_NG_ENA_SET_WIDTH 1 /* CLK_NG_ENA_SET */ +#define ARIZONA_CLK_L_ENA_CLR 0x0002 /* CLK_L_ENA_CLR */ +#define ARIZONA_CLK_L_ENA_CLR_MASK 0x0002 /* CLK_L_ENA_CLR */ +#define ARIZONA_CLK_L_ENA_CLR_SHIFT 1 /* CLK_L_ENA_CLR */ +#define ARIZONA_CLK_L_ENA_CLR_WIDTH 1 /* CLK_L_ENA_CLR */ +#define ARIZONA_CLK_L_ENA_SET 0x0001 /* CLK_L_ENA_SET */ +#define ARIZONA_CLK_L_ENA_SET_MASK 0x0001 /* CLK_L_ENA_SET */ +#define ARIZONA_CLK_L_ENA_SET_SHIFT 0 /* CLK_L_ENA_SET */ +#define ARIZONA_CLK_L_ENA_SET_WIDTH 1 /* CLK_L_ENA_SET */ + +/* + * R3841 (0xF01) - ANC SRC + */ +#define ARIZONA_IN_RXANCR_SEL_MASK 0x0070 /* IN_RXANCR_SEL - [4:6] */ +#define ARIZONA_IN_RXANCR_SEL_SHIFT 4 /* IN_RXANCR_SEL - [4:6] */ +#define ARIZONA_IN_RXANCR_SEL_WIDTH 3 /* IN_RXANCR_SEL - [4:6] */ +#define ARIZONA_IN_RXANCL_SEL_MASK 0x0007 /* IN_RXANCL_SEL - [0:2] */ +#define ARIZONA_IN_RXANCL_SEL_SHIFT 0 /* IN_RXANCL_SEL - [0:2] */ +#define ARIZONA_IN_RXANCL_SEL_WIDTH 3 /* IN_RXANCL_SEL - [0:2] */ + +/* + * R3863 (0xF17) - FCL ADC Reformatter Control + */ +#define ARIZONA_FCL_MIC_MODE_SEL 0x000C /* FCL_MIC_MODE_SEL - [2:3] */ +#define ARIZONA_FCL_MIC_MODE_SEL_SHIFT 2 /* FCL_MIC_MODE_SEL - [2:3] */ +#define ARIZONA_FCL_MIC_MODE_SEL_WIDTH 2 /* FCL_MIC_MODE_SEL - [2:3] */ + +/* + * R3954 (0xF72) - FCR ADC Reformatter Control + */ +#define ARIZONA_FCR_MIC_MODE_SEL 0x000C /* FCR_MIC_MODE_SEL - [2:3] */ +#define ARIZONA_FCR_MIC_MODE_SEL_SHIFT 2 /* FCR_MIC_MODE_SEL - [2:3] */ +#define ARIZONA_FCR_MIC_MODE_SEL_WIDTH 2 /* FCR_MIC_MODE_SEL - [2:3] */ + +/* * R4352 (0x1100) - DSP1 Control 1 */ #define ARIZONA_DSP1_RATE_MASK 0x7800 /* DSP1_RATE - [14:11] */ -- 1.7.2.5

2 2

[alsa-devel] [PATCH] ASoC: ak4613: add Digital Playback Volume support
by Kuninori Morimoto 20 Sep '15

20 Sep '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> For L/ROUT1 to L/ROUT6 Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/ak4613.c | 28 ++++++++++++++++++++++++++++ 1 file changed, 28 insertions(+) diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c index fd96a8f..07a2664 100644 --- a/sound/soc/codecs/ak4613.c +++ b/sound/soc/codecs/ak4613.c @@ -22,6 +22,7 @@ #include <linux/regmap.h> #include <sound/soc.h> #include <sound/pcm_params.h> +#include <sound/tlv.h> #define PW_MGMT1 0x00 /* Power Management 1 */ #define PW_MGMT2 0x01 /* Power Management 2 */ @@ -91,6 +92,31 @@ struct ak4613_interface { struct ak4613_formats playback; }; +/* + * Playback Volume + * + * max : 0x00 : 0 dB + * ( 0.5 dB step ) + * min : 0xFE : -127.0 dB + * mute: 0xFF + */ +static const DECLARE_TLV_DB_SCALE(out_tlv, -12750, 50, 1); + +static const struct snd_kcontrol_new ak4613_snd_controls[] = { + SOC_DOUBLE_R_TLV("Digital Playback Volume1", LOUT1, ROUT1, + 0, 0xFF, 1, out_tlv), + SOC_DOUBLE_R_TLV("Digital Playback Volume2", LOUT2, ROUT2, + 0, 0xFF, 1, out_tlv), + SOC_DOUBLE_R_TLV("Digital Playback Volume3", LOUT3, ROUT3, + 0, 0xFF, 1, out_tlv), + SOC_DOUBLE_R_TLV("Digital Playback Volume4", LOUT4, ROUT4, + 0, 0xFF, 1, out_tlv), + SOC_DOUBLE_R_TLV("Digital Playback Volume5", LOUT5, ROUT5, + 0, 0xFF, 1, out_tlv), + SOC_DOUBLE_R_TLV("Digital Playback Volume6", LOUT6, ROUT6, + 0, 0xFF, 1, out_tlv), +}; + static const struct reg_default ak4613_reg[] = { { 0x0, 0x0f }, { 0x1, 0x07 }, { 0x2, 0x3f }, { 0x3, 0x20 }, { 0x4, 0x20 }, { 0x5, 0x55 }, { 0x6, 0x05 }, { 0x7, 0x07 }, @@ -397,6 +423,8 @@ static int ak4613_resume(struct snd_soc_codec *codec) static struct snd_soc_codec_driver soc_codec_dev_ak4613 = { .resume = ak4613_resume, .set_bias_level = ak4613_set_bias_level, + .controls = ak4613_snd_controls, + .num_controls = ARRAY_SIZE(ak4613_snd_controls), .dapm_widgets = ak4613_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(ak4613_dapm_widgets), .dapm_routes = ak4613_intercon, -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: davinci-mcasp: Fix devm_kasprintf format string
by Peter Ujfalusi 20 Sep '15

20 Sep '15

The '\n' at the end of the format string is not needed. It adds an extra line break when doing cat /proc/interrupts Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 1260b315a96c..e631f96f9cf7 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1684,7 +1684,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) irq = platform_get_irq_byname(pdev, "common"); if (irq >= 0) { - irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_common\n", + irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_common", dev_name(&pdev->dev)); ret = devm_request_threaded_irq(&pdev->dev, irq, NULL, davinci_mcasp_common_irq_handler, @@ -1701,7 +1701,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) irq = platform_get_irq_byname(pdev, "rx"); if (irq >= 0) { - irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_rx\n", + irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_rx", dev_name(&pdev->dev)); ret = devm_request_threaded_irq(&pdev->dev, irq, NULL, davinci_mcasp_rx_irq_handler, @@ -1716,7 +1716,7 @@ static int davinci_mcasp_probe(struct platform_device *pdev) irq = platform_get_irq_byname(pdev, "tx"); if (irq >= 0) { - irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_tx\n", + irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_tx", dev_name(&pdev->dev)); ret = devm_request_threaded_irq(&pdev->dev, irq, NULL, davinci_mcasp_tx_irq_handler, -- 2.5.2

2 1

[alsa-devel] [PATCH] ASoC: rockchip: Fix missing CLKDEV_LOOKUP dependency
by Axel Lin 20 Sep '15

20 Sep '15

Make SND_SOC_ROCKCHIP_MAX98090 and SND_SOC_ROCKCHIP_RT5645 depend on CLKDEV_LOOKUP to fix below build warning: warning: (SND_SOC_ROCKCHIP_MAX98090 && SND_SOC_ROCKCHIP_RT5645) selects SND_SOC_ROCKCHIP_I2S which has unmet direct dependencies (SOUND && !M68K && !UML && SND && SND_SOC && CLKDEV_LOOKUP && SND_SOC_ROCKCHIP) Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/rockchip/Kconfig | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/rockchip/Kconfig b/sound/soc/rockchip/Kconfig index 58bae8e..5709057 100644 --- a/sound/soc/rockchip/Kconfig +++ b/sound/soc/rockchip/Kconfig @@ -17,7 +17,7 @@ config SND_SOC_ROCKCHIP_I2S config SND_SOC_ROCKCHIP_MAX98090 tristate "ASoC support for Rockchip boards using a MAX98090 codec" - depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB + depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB && CLKDEV_LOOKUP select SND_SOC_ROCKCHIP_I2S select SND_SOC_MAX98090 select SND_SOC_TS3A227E @@ -27,7 +27,7 @@ config SND_SOC_ROCKCHIP_MAX98090 config SND_SOC_ROCKCHIP_RT5645 tristate "ASoC support for Rockchip boards using a RT5645/RT5650 codec" - depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB + depends on SND_SOC_ROCKCHIP && I2C && GPIOLIB && CLKDEV_LOOKUP select SND_SOC_ROCKCHIP_I2S select SND_SOC_RT5645 help -- 2.1.4

2 1

[alsa-devel] [PATCH 1/3] ASoC: wm0010: disable regulator on error
by Sudip Mukherjee 20 Sep '15

20 Sep '15

We have done regulator_bulk_enable() while booting the DSP but on the error exit path we have not disbled it. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/soc/codecs/wm0010.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index 8434d45..79a7cd3 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -739,8 +739,6 @@ static int wm0010_boot(struct snd_soc_codec *codec) abort: /* Put the chip back into reset */ wm0010_halt(codec); - mutex_unlock(&wm0010->lock); - return ret; err_core: mutex_unlock(&wm0010->lock); -- 1.9.1

3 9

[alsa-devel] [PATCH RFC v2 0/7] Implement generic ASoC HDMI codec
by Jyri Sarha 20 Sep '15

20 Sep '15

Only the first three patches are meant for serious review. The ASoC part should be ready for review in other respects but the EDID SADs handling is waiting for Russell King's DRM ELD helper patches. There is a copy-pasted not-to-be-merged patch with the same functionality in the patch series. The tda998x part is not complete and I do not think Jean-Francois' binding are ready as such either. At least the usage of reg property in a port node for something that is not an address or even guaranteed to be unique is not a good idea. Some documentation of how to configure AP0-AP4 pins (REG_ENA_AP?) would help in making a proper binding for the tda998x audio configuration. These patches, my tilcdc refactoring[1], and my latest BCLK fixes for davinci-mcasp driver [2], can be found in a branch that produces working HDMI audio on Beaglebone-Black here: https://github.com/jsarha/linux.git linux-master-bbb-hdmi-20150512 [1] http://lists.freedesktop.org/archives/dri-devel/2015-May/082537.html [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2015-April/090974.html Jean-Francois Moine (1): drm/i2c: tda998x: Add support of a DT graph of ports DO NOT MERGE Jyri Sarha (6): ASoC: core: If component doesn't have of_node use parent's node instead ASoC: hdmi: Remove obsolete dummy HDMI codec ASoC: hdmi-codec: Add hdmi-codec for external HDMI-encoders ASoC: hdmi-codec: Add ELD based audio pcm rules DO NOT MERGE drm/i2c: tda998x: Register ASoC HDMI codec for audio functionality DO NOT MERGE ARM: dts: am335x-boneblack: Add HDMI audio support DO NOT MERGE .../devicetree/bindings/drm/i2c/tda998x.txt | 51 ++ arch/arm/boot/dts/am335x-boneblack.dts | 78 ++- drivers/gpu/drm/i2c/Kconfig | 1 + drivers/gpu/drm/i2c/tda998x_drv.c | 328 +++++++++++- include/sound/hdmi-codec.h | 99 ++++ include/sound/tda998x.h | 8 + sound/soc/codecs/Kconfig | 2 +- sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/hdmi-codec.c | 560 +++++++++++++++++++++ sound/soc/codecs/hdmi.c | 109 ---- sound/soc/soc-core.c | 14 +- 11 files changed, 1129 insertions(+), 125 deletions(-) create mode 100644 include/sound/hdmi-codec.h create mode 100644 include/sound/tda998x.h create mode 100644 sound/soc/codecs/hdmi-codec.c delete mode 100644 sound/soc/codecs/hdmi.c -- 1.9.1

2 12

[alsa-devel] [PATCH v2] ASoC: Handle multiple codecs with split playback / capture
by Ricard Wanderlof 20 Sep '15

20 Sep '15

Add the capability to use multiple codecs on the same DAI linke where one codec is used for playback and another one is used for capture. Tested on a setup using an SSM2518 for playback and an ICS43432 I2S MEMS microphone for capture. Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- The patch was created after a discussion on the alsa-devel mailing list as to how best to implement this functionality. (http://mailman.alsa-project.org/pipermail/alsa-devel/2015-June/093214.html). V2: Minor code change, otherwise the differences compared to V1 are solely related to comments, in particular considerations given to the potential consequences of the patch. After consideration, it seems to me that the patch as it stands augments the current framework with the functionality indicated in the commit message above, without restricting other current uses. Further functionality could be added, but IMHO that should be done as the need arises, when it can be properly tested in a real-world setup. sound/soc/soc-pcm.c | 49 +++++++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 49 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 35fe58f4..08407ba 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -34,6 +34,24 @@ #define DPCM_MAX_BE_USERS 8 +/* + * snd_soc_dai_stream_valid() - check if a DAI supports the given stream + * + * Returns true if the DAI supports the indicated stream type. + */ +static bool snd_soc_dai_stream_valid(struct snd_soc_dai *dai, int stream) +{ + struct snd_soc_pcm_stream *codec_stream; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + codec_stream = &dai->driver->playback; + else + codec_stream = &dai->driver->capture; + + /* If the codec specifies any rate at all, it supports the stream. */ + return codec_stream->rates; +} + /** * snd_soc_runtime_activate() - Increment active count for PCM runtime components * @rtd: ASoC PCM runtime that is activated @@ -371,6 +389,20 @@ static void soc_pcm_init_runtime_hw(struct snd_pcm_substream *substream) /* first calculate min/max only for CODECs in the DAI link */ for (i = 0; i < rtd->num_codecs; i++) { + + /* + * Skip CODECs which don't support the current stream type. + * Otherwise, since the rate, channel, and format values will + * zero in that case, we would have no usable settings left, + * causing the resulting setup to fail. + * At least one CODEC should match, otherwise we should have + * bailed out on a higher level, since there would be no + * CODEC to support the transfer direction in that case. + */ + if (!snd_soc_dai_stream_valid(rtd->codec_dais[i], + substream->stream)) + continue; + codec_dai_drv = rtd->codec_dais[i]->driver; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) codec_stream = &codec_dai_drv->playback; @@ -827,6 +859,23 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; struct snd_pcm_hw_params codec_params; + /* + * Skip CODECs which don't support the current stream type, + * the idea being that if a CODEC is not used for the currently + * set up transfer direction, it should not need to be + * configured, especially since the configuration used might + * not even be supported by that CODEC. There may be cases + * however where a CODEC needs to be set up although it is + * actually not being used for the transfer, e.g. if a + * capture-only CODEC is acting as an LRCLK and/or BCLK master + * for the DAI link including a playback-only CODEC. + * If this becomes necessary, we will have to augment the + * machine driver setup with information on how to act, so + * we can do the right thing here. + */ + if (!snd_soc_dai_stream_valid(codec_dai, substream->stream)) + continue; + /* copy params for each codec */ codec_params = *params; -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

3 8

[alsa-devel] [PATCH] ASoC: fsl_ssi: Fixing invalid fmt SND_SOC_DAIFMT_AC97 test
by Arnaud Mouiche 20 Sep '15

20 Sep '15

SND_SOC_DAIFMT_AC97 (6) has similar bits than SND_SOC_DAIFMT_DSP_A (4) and can't be tested with a simple AND operation Signed-off-by: Arnaud Mouiche <arnaud.mouiche(a)invoxia.com> --- sound/soc/fsl/fsl_ssi.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 6b0c8f7..c25a1e8 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -249,7 +249,8 @@ MODULE_DEVICE_TABLE(of, fsl_ssi_ids); static bool fsl_ssi_is_ac97(struct fsl_ssi_private *ssi_private) { - return !!(ssi_private->dai_fmt & SND_SOC_DAIFMT_AC97); + return (ssi_private->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) == + SND_SOC_DAIFMT_AC97; } static bool fsl_ssi_is_i2s_master(struct fsl_ssi_private *ssi_private) @@ -945,7 +946,7 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev, CCSR_SSI_SCR_TCH_EN); } - if (fmt & SND_SOC_DAIFMT_AC97) + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_AC97) fsl_ssi_setup_ac97(ssi_private); return 0; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: fsl-asoc-card: add wm8960 support
by Zidan Wang 19 Sep '15

19 Sep '15

add wm8960 support for fsl-asoc-card Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- .../devicetree/bindings/sound/fsl-asoc-card.txt | 2 ++ sound/soc/fsl/fsl-asoc-card.c | 37 ++++++++++++++++++++-- 2 files changed, 36 insertions(+), 3 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt index a96774c..fa0a10d 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt @@ -59,6 +59,8 @@ Optional properties: - audio-asrc : The phandle of ASRC. It can be absent if there's no need to add ASRC support via DPCM. + - gpr : The gpr handle + Example: sound-cs42888 { compatible = "fsl,imx-audio-cs42888"; diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index de43887..c058e68 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -16,6 +16,7 @@ #include <linux/of_platform.h> #include <sound/pcm_params.h> #include <sound/soc.h> +#include <linux/mfd/syscon.h> #include "fsl_esai.h" #include "fsl_sai.h" @@ -23,6 +24,7 @@ #include "../codecs/sgtl5000.h" #include "../codecs/wm8962.h" +#include "../codecs/wm8960.h" #define RX 0 #define TX 1 @@ -90,6 +92,7 @@ struct fsl_asoc_card_priv { u32 asrc_format; u32 dai_fmt; char name[32]; + struct regmap *gpr; }; /** @@ -400,7 +403,7 @@ static int fsl_asoc_card_late_probe(struct snd_soc_card *card) static int fsl_asoc_card_probe(struct platform_device *pdev) { - struct device_node *cpu_np, *codec_np, *asrc_np; + struct device_node *cpu_np, *codec_np, *asrc_np, *gpr_np; struct device_node *np = pdev->dev.of_node; struct platform_device *asrc_pdev = NULL; struct platform_device *cpu_pdev; @@ -443,6 +446,22 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) if (asrc_np) asrc_pdev = of_find_device_by_node(asrc_np); + gpr_np = of_parse_phandle(np, "gpr", 0); + if (gpr_np) { + priv->gpr = syscon_node_to_regmap(gpr_np); + if (IS_ERR(priv->gpr)) { + ret = PTR_ERR(priv->gpr); + dev_err(&pdev->dev, "failed to get gpr regmap\n"); + goto fail; + } + /* + * for imx6ul we should set SAI2_MCLK_DIR to enable + * codec MCLK + */ + if (of_device_is_compatible(np, "fsl,imx6ul-evk-wm8960")) + regmap_update_bits(priv->gpr, 4, 1 << 20, 1 << 20); + } + /* Get the MCLK rate only, and leave it controlled by CODEC drivers */ codec_clk = clk_get(&codec_dev->dev, NULL); if (!IS_ERR(codec_clk)) { @@ -475,6 +494,11 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->codec_priv.fll_id = WM8962_SYSCLK_FLL; priv->codec_priv.pll_id = WM8962_FLL; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } else if (of_device_is_compatible(np, "fsl,imx-audio-wm8960")) { + priv->card.set_bias_level = fsl_asoc_card_set_bias_level; + priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; + priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; + priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); return -EINVAL; @@ -521,7 +545,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Normal DAI Link */ priv->dai_link[0].cpu_of_node = cpu_np; priv->dai_link[0].codec_of_node = codec_np; - priv->dai_link[0].codec_dai_name = codec_dev->name; + if (strstr(codec_dev->name, "wm8960")) + priv->dai_link[0].codec_dai_name = "wm8960-hifi"; + else + priv->dai_link[0].codec_dai_name = codec_dev->name; priv->dai_link[0].platform_of_node = cpu_np; priv->dai_link[0].dai_fmt = priv->dai_fmt; priv->card.num_links = 1; @@ -530,7 +557,10 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* DPCM DAI Links only if ASRC exsits */ priv->dai_link[1].cpu_of_node = asrc_np; priv->dai_link[1].platform_of_node = asrc_np; - priv->dai_link[2].codec_dai_name = codec_dev->name; + if (strstr(codec_dev->name, "wm8960")) + priv->dai_link[2].codec_dai_name = "wm8960-hifi"; + else + priv->dai_link[2].codec_dai_name = codec_dev->name; priv->dai_link[2].codec_of_node = codec_np; priv->dai_link[2].cpu_of_node = cpu_np; priv->dai_link[2].dai_fmt = priv->dai_fmt; @@ -578,6 +608,7 @@ static const struct of_device_id fsl_asoc_card_dt_ids[] = { { .compatible = "fsl,imx-audio-cs42888", }, { .compatible = "fsl,imx-audio-sgtl5000", }, { .compatible = "fsl,imx-audio-wm8962", }, + { .compatible = "fsl,imx-audio-wm8960", }, {} }; -- 1.9.1

4 15

[alsa-devel] [RESEND PATCH 1/1] ASoC: fsl_ssi: Fix checking of dai format for AC97 mode
by Adam Thomson 19 Sep '15

19 Sep '15

Current code incorrectly treats dai format for AC97 as bit mask whereas it's actually an integer value. This causes DAI formats other than AC97 (e.g. DSP_B) to trigger AC97 related code, which is incorrect and breaks functionality. This patch fixes the code to correctly compare values to determine AC97 or not. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- Patch based on v4.3-rc1. sound/soc/fsl/fsl_ssi.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 8ec6fb2..37c5cd4 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -249,7 +249,8 @@ MODULE_DEVICE_TABLE(of, fsl_ssi_ids); static bool fsl_ssi_is_ac97(struct fsl_ssi_private *ssi_private) { - return !!(ssi_private->dai_fmt & SND_SOC_DAIFMT_AC97); + return (ssi_private->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) == + SND_SOC_DAIFMT_AC97; } static bool fsl_ssi_is_i2s_master(struct fsl_ssi_private *ssi_private) @@ -947,7 +948,7 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev, CCSR_SSI_SCR_TCH_EN); } - if (fmt & SND_SOC_DAIFMT_AC97) + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_AC97) fsl_ssi_setup_ac97(ssi_private); return 0; -- 1.9.3

3 2

[alsa-devel] [PATCH v2] ASoC: fsl-asoc-card: add AC'97 support
by Maciej S. Szmigiero 19 Sep '15

19 Sep '15

Add AC'97 support to fsl-asoc-card using generic ASoC AC'97 CODEC. The SSI controller will silently enable any TX AC'97 slots that have their bits set in SLOTREQ received from CODEC and then will redirect some of playback samples there. That's why it is important to make sure that any of CODEC playback slots that can pull samples are set to slots 3/4 (standard PCM playback slots). Currently, this applies to S/PDIF slots as they were seen to pull samples sometimes even with S/PDIF output being disabled. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- Changes from v1: don't use of_find_device_by_node() to get pointer to I2S/PCM CODEC device as this only matches platform bus devices while I2S/PCM CODECs supported by this driver sit on I2C bus. .../devicetree/bindings/sound/fsl-asoc-card.txt | 10 +- sound/soc/fsl/fsl-asoc-card.c | 140 ++++++++++++++++----- 2 files changed, 116 insertions(+), 34 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt index a96774c..ce55c0a 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt @@ -13,13 +13,15 @@ So having this generic sound card allows all Freescale SoC users to benefit from the simplification of a new card support and the capability of the wide sample rates support through ASRC. -Note: The card is initially designed for those sound cards who use I2S and - PCM DAI formats. However, it'll be also possible to support those non - I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as long - as the driver has been properly upgraded. +Note: The card is initially designed for those sound cards who use AC'97, I2S + and PCM DAI formats. However, it'll be also possible to support those non + AC'97/I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as + long as the driver has been properly upgraded. The compatible list for this generic sound card currently: + "fsl,imx-audio-ac97" + "fsl,imx-audio-cs42888" "fsl,imx-audio-wm8962" diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 0901d5e..1b05d1c 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -14,6 +14,9 @@ #include <linux/i2c.h> #include <linux/module.h> #include <linux/of_platform.h> +#if IS_ENABLED(CONFIG_SND_AC97_CODEC) +#include <sound/ac97_codec.h> +#endif #include <sound/pcm_params.h> #include <sound/soc.h> @@ -115,6 +118,11 @@ static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = { SND_SOC_DAPM_MIC("DMIC", NULL), }; +static bool fsl_asoc_card_is_ac97(struct fsl_asoc_card_priv *priv) +{ + return priv->dai_fmt == SND_SOC_DAIFMT_AC97; +} + static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -133,7 +141,9 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, * set_bias_level(), bypass the remaining settings in hw_params(). * Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS. */ - if (priv->card.set_bias_level && priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) + if ((priv->card.set_bias_level && + priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) || + fsl_asoc_card_is_ac97(priv)) return 0; /* Specific configurations of DAIs starts from here */ @@ -300,7 +310,7 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, ext_port--; /* - * Use asynchronous mode (6 wires) for all cases. + * Use asynchronous mode (6 wires) for all cases except AC97. * If only 4 wires are needed, just set SSI into * synchronous mode and enable 4 PADs in IOMUX. */ @@ -346,15 +356,30 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, IMX_AUDMUX_V2_PTCR_TCLKDIR; break; default: - return -EINVAL; + if (!fsl_asoc_card_is_ac97(priv)) + return -EINVAL; + } + + if (fsl_asoc_card_is_ac97(priv)) { + int_ptcr = IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCLKDIR; + ext_ptcr = IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSEL(int_port) | + IMX_AUDMUX_V2_PTCR_TFSDIR; } /* Asynchronous mode can not be set along with RCLKDIR */ - ret = imx_audmux_v2_configure_port(int_port, 0, - IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); - if (ret) { - dev_err(dev, "audmux internal port setup failed\n"); - return ret; + if (!fsl_asoc_card_is_ac97(priv)) { + unsigned int pdcr = + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port); + + ret = imx_audmux_v2_configure_port(int_port, 0, + pdcr); + if (ret) { + dev_err(dev, "audmux internal port setup failed\n"); + return ret; + } } ret = imx_audmux_v2_configure_port(int_port, int_ptcr, @@ -364,11 +389,16 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, return ret; } - ret = imx_audmux_v2_configure_port(ext_port, 0, - IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); - if (ret) { - dev_err(dev, "audmux external port setup failed\n"); - return ret; + if (!fsl_asoc_card_is_ac97(priv)) { + unsigned int pdcr = + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port); + + ret = imx_audmux_v2_configure_port(ext_port, 0, + pdcr); + if (ret) { + dev_err(dev, "audmux external port setup failed\n"); + return ret; + } } ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr, @@ -389,6 +419,23 @@ static int fsl_asoc_card_late_probe(struct snd_soc_card *card) struct device *dev = card->dev; int ret; + if (fsl_asoc_card_is_ac97(priv)) { +#if IS_ENABLED(CONFIG_SND_AC97_CODEC) + struct snd_soc_codec *codec = card->rtd[0].codec; + struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); + + /* + * Use slots 3/4 for S/PDIF so SSI won't try to enable + * other slots and send some samples there + * due to SLOTREQ bits for S/PDIF received from codec + */ + snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS, + AC97_EA_SPSA_SLOT_MASK, AC97_EA_SPSA_3_4); +#endif + + return 0; + } + ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, codec_priv->mclk_freq, SND_SOC_CLOCK_IN); if (ret) { @@ -407,7 +454,6 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) struct platform_device *cpu_pdev; struct fsl_asoc_card_priv *priv; struct i2c_client *codec_dev; - struct clk *codec_clk; const char *codec_dai_name; u32 width; int ret; @@ -420,9 +466,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Give a chance to old DT binding */ if (!cpu_np) cpu_np = of_parse_phandle(np, "ssi-controller", 0); - codec_np = of_parse_phandle(np, "audio-codec", 0); - if (!cpu_np || !codec_np) { - dev_err(&pdev->dev, "phandle missing or invalid\n"); + if (!cpu_np) { + dev_err(&pdev->dev, "CPU phandle missing or invalid\n"); ret = -EINVAL; goto fail; } @@ -434,22 +479,24 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) goto fail; } - codec_dev = of_find_i2c_device_by_node(codec_np); - if (!codec_dev) { - dev_err(&pdev->dev, "failed to find codec platform device\n"); - ret = -EINVAL; - goto fail; - } + codec_np = of_parse_phandle(np, "audio-codec", 0); + if (codec_np) + codec_dev = of_find_i2c_device_by_node(codec_np); + else + codec_dev = NULL; asrc_np = of_parse_phandle(np, "audio-asrc", 0); if (asrc_np) asrc_pdev = of_find_device_by_node(asrc_np); /* Get the MCLK rate only, and leave it controlled by CODEC drivers */ - codec_clk = clk_get(&codec_dev->dev, NULL); - if (!IS_ERR(codec_clk)) { - priv->codec_priv.mclk_freq = clk_get_rate(codec_clk); - clk_put(codec_clk); + if (codec_dev) { + struct clk *codec_clk = clk_get(&codec_dev->dev, NULL); + + if (!IS_ERR(codec_clk)) { + priv->codec_priv.mclk_freq = clk_get_rate(codec_clk); + clk_put(codec_clk); + } } /* Default sample rate and format, will be updated in hw_params() */ @@ -486,12 +533,22 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) { + codec_dai_name = "ac97-hifi"; + priv->card.set_bias_level = NULL; + priv->dai_fmt = SND_SOC_DAIFMT_AC97; } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); ret = -EINVAL; goto asrc_fail; } + if (!fsl_asoc_card_is_ac97(priv) && !codec_dev) { + dev_err(&pdev->dev, "failed to find codec device\n"); + ret = -EINVAL; + goto asrc_fail; + } + /* Common settings for corresponding Freescale CPU DAI driver */ if (strstr(cpu_np->name, "ssi")) { /* Only SSI needs to configure AUDMUX */ @@ -508,7 +565,9 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1; } - sprintf(priv->name, "%s-audio", codec_dev->name); + snprintf(priv->name, sizeof(priv->name), "%s-audio", + fsl_asoc_card_is_ac97(priv) ? "ac97" : + codec_dev->name); /* Initialize sound card */ priv->pdev = pdev; @@ -532,8 +591,26 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Normal DAI Link */ priv->dai_link[0].cpu_of_node = cpu_np; - priv->dai_link[0].codec_of_node = codec_np; priv->dai_link[0].codec_dai_name = codec_dai_name; + + if (!fsl_asoc_card_is_ac97(priv)) + priv->dai_link[0].codec_of_node = codec_np; + else { + u32 idx; + + ret = of_property_read_u32(cpu_np, "cell-index", &idx); + if (ret) { + dev_err(&pdev->dev, + "cannot get CPU index property\n"); + goto asrc_fail; + } + + priv->dai_link[0].codec_name = + devm_kasprintf(&pdev->dev, GFP_KERNEL, + "ac97-codec.%u", + (unsigned int)idx); + } + priv->dai_link[0].platform_of_node = cpu_np; priv->dai_link[0].dai_fmt = priv->dai_fmt; priv->card.num_links = 1; @@ -544,6 +621,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->dai_link[1].platform_of_node = asrc_np; priv->dai_link[2].codec_dai_name = codec_dai_name; priv->dai_link[2].codec_of_node = codec_np; + priv->dai_link[2].codec_name = + priv->dai_link[0].codec_name; priv->dai_link[2].cpu_of_node = cpu_np; priv->dai_link[2].dai_fmt = priv->dai_fmt; priv->card.num_links = 3; @@ -579,14 +658,15 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) asrc_fail: of_node_put(asrc_np); -fail: of_node_put(codec_np); +fail: of_node_put(cpu_np); return ret; } static const struct of_device_id fsl_asoc_card_dt_ids[] = { + { .compatible = "fsl,imx-audio-ac97", }, { .compatible = "fsl,imx-audio-cs42888", }, { .compatible = "fsl,imx-audio-sgtl5000", }, { .compatible = "fsl,imx-audio-wm8962", },

1 0

Re: [alsa-devel] Asus N751JK - push/merge patches on mailinglist
by Raymond Yau 18 Sep '15

18 Sep '15

2015-09-13 16:13 GMT+08:00 Takashi Iwai <tiwai(a)suse.de>: > On Sat, 12 Sep 2015 23:09:43 +0200, > Arthur Borsboom wrote: > > > > Hi Takashi, > > > > Previous month Raymond Yau has done great work to get the audio working > for > > the Asus N751JK laptop. > > He developed 7 patches, which I have tested and confirmed working on this > > laptop. > > > > Raymond is not responding for about two weeks (maybe on holiday?). > > Would you mind to post these patches on the mailinglist (the correct way) > > and/or merge them in a way the get into the mainline kernel a.s.a.p? > > Sorry, it can't be done like that. The patches aren't yet in > acceptable forms, and it needs more changes to be merged to mainline. > It must be done by someone who did it originally or who has the > hardware and test. > I don't have the hardware, the alternate approach is to create headset jack (SND_JACK_HEADSET) and skip the creation of headset mic phanton jack instead of headphone jack and headset mic jack since implement sense_nid and slave_nid is not easy to debug What you need is adding "Jack Headset" to pulseaudio analog-input-headset-mic.conf since "Headset Jack" kctl already in analog-out-headphones.conf static int get_input_jack_type(struct hda_codec *codec, hda_nid_t nid) { + struct hda_gen_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; unsigned int def_conf = snd_hda_codec_get_pincfg(codec, nid); switch (get_defcfg_device(def_conf)) { case AC_JACK_LINE_OUT: case AC_JACK_SPEAKER: return SND_JACK_LINEOUT; case AC_JACK_HP_OUT: + if (spec->hs_mic_use_hp_sense && + (nid == auto_cfg_hp_pins(cfg)[0])) + return SND_JACK_HEADSET; return SND_JACK_HEADPHONE; case AC_JACK_SPDIF_OUT: case AC_JACK_DIG_OTHER_OUT: const struct auto_pin_cfg *cfg) { const hda_nid_t *p; + struct hda_gen_spec *spec = codec->spec; int i, err; for (i = 0; i < cfg->num_inputs; i++) { @@ -482,6 +499,12 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg, "Headphone Mic"); } else + if (cfg->inputs[i].is_headset_mic && + spec->hs_mic_use_hp_sense && + !is_jack_detectable(codec, cfg->inputs[i].pin)) { + err = 0; + + } else err = add_jack_kctl(codec, cfg->inputs[i].pin, cfg, NULL); if (err < 0) @@ -489,14 +512,21 @@ int snd_hda_jack_add_kctls(struct hda_codec *codec, } for (i = 0, p = cfg->line_out_pins; i < cfg->line_outs; i++, p++) { - err = add_jack_kctl(codec, *p, cfg, NULL); + if (is_subwoofer(codec, *p, i, cfg->line_outs) && + is_external_pin(codec, *p)) + err = add_jack_kctl(codec, *p, cfg, "Ext Subwoofer"); + else + err = add_jack_kctl(codec, *p, cfg, NULL); if (err < 0) return err; } for (i = 0, p = cfg->hp_pins; i < cfg->hp_outs; i++, p++) { if (*p == *cfg->line_out_pins) /* might be duplicated */ break; - err = add_jack_kctl(codec, *p, cfg, NULL); + if (i == 0 && spec->hs_mic_use_hp_sense) + err = add_jack_kctl(codec, *p, cfg, "Headset"); + else + err = add_jack_kctl(codec, *p, cfg, NULL); if (err < 0) return err; } change the pin to hp pin when mic pin is headset mic +bool is_headset_mic(struct hda_codec *codec, hda_nid_t nid) +{ + struct hda_gen_spec *spec = codec->spec; + struct auto_pin_cfg *cfg = &spec->autocfg; + int i; + + for (i = 0; i < cfg->num_inputs; i++) + if (nid == cfg->inputs[i].pin) + return cfg->inputs[i].is_headset_mic; + return false; +} + @@ -4384,6 +4399,8 @@ void snd_hda_gen_mic_autoswitch(struct hda_codec *codec, /* don't detect pins retasked as outputs */ if (snd_hda_codec_get_pin_target(codec, pin) & AC_PINCTL_OUT_EN) continue; + if (is_headset_mic(codec, pin) && spec->hs_mic_use_hp_sense) + pin = auto_cfg_hp_pins(cfg)[0]; if (snd_hda_jack_detect_state(codec, pin) == HDA_JACK_PRESENT) { mux_select(codec, 0, spec->am_entry[i].idx); return;

2 5

[alsa-devel] [PATCH 1/4] ASoC: fsl_sai: Add driver suspend and resume to support MEGA Fast
by Zidan Wang 18 Sep '15

18 Sep '15

For i.MX6 SoloX, there is a mode of the SoC to shutdown all power source of modules during system suspend and resume procedure. Thus, SAI needs to save all the values of registers before the system suspend and restore them after the system resume. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- sound/soc/fsl/fsl_sai.c | 33 +++++++++++++++++++++++++++++++++ 1 file changed, 33 insertions(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index a18fd92..5c737f1 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -637,6 +637,8 @@ static bool fsl_sai_readable_reg(struct device *dev, unsigned int reg) static bool fsl_sai_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { + case FSL_SAI_TCSR: + case FSL_SAI_RCSR: case FSL_SAI_TFR: case FSL_SAI_RFR: case FSL_SAI_TDR: @@ -681,6 +683,7 @@ static const struct regmap_config fsl_sai_regmap_config = { .readable_reg = fsl_sai_readable_reg, .volatile_reg = fsl_sai_volatile_reg, .writeable_reg = fsl_sai_writeable_reg, + .cache_type = REGCACHE_FLAT, }; static int fsl_sai_probe(struct platform_device *pdev) @@ -802,10 +805,40 @@ static const struct of_device_id fsl_sai_ids[] = { { /* sentinel */ } }; +#if CONFIG_PM_SLEEP +static int fsl_sai_suspend(struct device *dev) +{ + struct fsl_sai *sai = dev_get_drvdata(dev); + + regcache_cache_only(sai->regmap, true); + regcache_mark_dirty(sai->regmap); + + return 0; +} + +static int fsl_sai_resume(struct device *dev) +{ + struct fsl_sai *sai = dev_get_drvdata(dev); + + regcache_cache_only(sai->regmap, false); + regmap_write(sai->regmap, FSL_SAI_TCSR, FSL_SAI_CSR_SR); + regmap_write(sai->regmap, FSL_SAI_RCSR, FSL_SAI_CSR_SR); + msleep(1); + regmap_write(sai->regmap, FSL_SAI_TCSR, 0); + regmap_write(sai->regmap, FSL_SAI_RCSR, 0); + return regcache_sync(sai->regmap); +} +#endif /* CONFIG_PM_SLEEP */ + +static const struct dev_pm_ops fsl_sai_pm_ops = { + SET_SYSTEM_SLEEP_PM_OPS(fsl_sai_suspend, fsl_sai_resume) +}; + static struct platform_driver fsl_sai_driver = { .probe = fsl_sai_probe, .driver = { .name = "fsl-sai", + .pm = &fsl_sai_pm_ops, .of_match_table = fsl_sai_ids, }, }; -- 1.9.1

1 3

[alsa-devel] Intel SST on a Bay Trail tablet
by Antonio Ospite 18 Sep '15

18 Sep '15

Hi, I am trying to get the Intel SST driver working on a Teclast X98 Air 3G [1], it's a Bay Trail tablet. The tests below have been made with 4.0.0-rc1 and the firmware files from https://git.kernel.org/cgit/linux/kernel/git/firmware/linux-firmware.git/tr… The DSDT on this device is quite broken, and I am patching it to get stuff working. The audio device works just fine under Android, I don't know if that's because the Android kernel is ignoring the DSDT and is hard-coding some platform data. Anyhow, with the untouched DSDT[2] I was getting this message in dmesg[3]: [ 11.344095] platform byt-rt5640: Driver byt-rt5640 requests probe deferral [ 11.404831] resource sanity check: requesting [mem 0x91000000-0x911fffff], which spans more than ADMA0F28:00 [mem 0x91000000-0x91000fff] [ 11.404833] ------------[ cut here ]------------ [ 11.404849] WARNING: CPU: 0 PID: 213 at arch/x86/mm/ioremap.c:197 __ioremap_caller+0x22c/0x350() ... I guessed this was because the DSDT defined an ADMA0F28 device with a memory resource overlapping with the LPMA device. So I removed the overlapping resources (ADMA0F28 and SSPX0000) in the DSDT and then the driver attempted to boot the audio device but failed with this message in dmesg[4]: [ 11.602291] baytrail-pcm-audio baytrail-pcm-audio: ipc: error DSP boot timeout ... Then I noticed that the Interrupt resources of the LPMA device were out of order in the DSDT, so I put the Interrupt 0x18 as the first one[5] and now the driver can boot the device and upload the firmware to it[6]: [ 11.433460] baytrail-pcm-audio baytrail-pcm-audio: FW version: 04.05.13.a0 [ 11.433468] baytrail-pcm-audio baytrail-pcm-audio: Build type: a0 [ 11.433473] baytrail-pcm-audio baytrail-pcm-audio: Build date: Apr 2 2014 14:14:39 [ 11.448257] byt-rt5640 byt-rt5640: rt5640-aif1 <-> baytrail-pcm-audio mapping ok ... But still no sound. I can see the mixer controls in alsamixer and I tried to initialize them copying from the alsactl output from the Android kernel[7], but no, still no sound. I do not know if that matters, but the driver still emits this message: [ 11.398304] baytrail-pcm-audio baytrail-pcm-audio: error: invalid DMA engine 0 [ 11.398314] baytrail-pcm-audio baytrail-pcm-audio: sst_dma_new failed -22 ... I noticed that on the Android kernel[8] the DMA channels for the ADMA0F28 device are actually configured, the intel_mid_dma driver sets up 4 channels for it along with 2 x 6 channels for the INTL9C60 device. On the mainline kernel only the INTL9C60 device gets configured, by the dw_dmac driver and with 2 x 8 channels. I do not know much about these details tho. Also, the codec driver on the Android kernel (byt_rt5642) seems to use one GPIO as an interrupt[9]: 390: 0 1 0 0 VLV-GPIO-gpio byt-codec-int Which is pin 4 on GPO2, but the codec driver in the mainline kernel does not seem to be using this; however AFAICS in sound/soc/codecs/rt5640.c a gpio pin in only used for suspend/resume, so it is not strictly needed if it's already in the correct state, right? Last question, what is the difference between having the device detected by sound/soc/intel/sst-acpi.c (like in my case) opposed to sound/soc/intel/sst/sst_acpi.c? I can see the 80860F28 id in both the files but they seem to load different firmwares. Any idea of what is missing? It's the DSDT which still needs adjusting, or are some changes needed (also) in the kernel driver? Thanks, Antonio [1] http://ao2.it/110 [2] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/dsdt_sst_original.dsl [3] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/dmesg_mainline_sst_ori… [4] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/dmesg_mainline_sst_wit… [5] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/dsdt_sst_without_ADMA0… [6] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/dmesg_mainline_sst_wit… [7] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/android_asound.state [8] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/dmesg_Android_Teclast_… [9] http://ao2.it/tmp/sst-baytrail-on-Teclast-X98-Air-3G/android_proc_interrupt… -- Antonio Ospite http://ao2.it A: Because it messes up the order in which people normally read text. See http://en.wikipedia.org/wiki/Posting_style Q: Why is top-posting such a bad thing?

6 23

[alsa-devel] Applied "ASoC: fsl-asoc-card: add AC'97 support" to the asoc tree
by Mark Brown 17 Sep '15

17 Sep '15

The patch ASoC: fsl-asoc-card: add AC'97 support has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From e06b508481e1916b25038289b945104009d774c9 Mon Sep 17 00:00:00 2001 From: "Maciej S. Szmigiero" <mail(a)maciej.szmigiero.name> Date: Mon, 31 Aug 2015 17:11:35 +0200 Subject: [PATCH] ASoC: fsl-asoc-card: add AC'97 support Add AC'97 support to fsl-asoc-card using generic ASoC AC'97 CODEC. The SSI controller will silently enable any TX AC'97 slots that have their bits set in SLOTREQ received from CODEC and then will redirect some of playback samples there. That's why it is important to make sure that any of CODEC playback slots that can pull samples are set to slots 3/4 (standard PCM playback slots). Currently, this applies to S/PDIF slots as they were seen to pull samples sometimes even with S/PDIF output being disabled. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- .../devicetree/bindings/sound/fsl-asoc-card.txt | 10 +- sound/soc/fsl/fsl-asoc-card.c | 145 ++++++++++++++++----- 2 files changed, 120 insertions(+), 35 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt index a96774c..ce55c0a 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt @@ -13,13 +13,15 @@ So having this generic sound card allows all Freescale SoC users to benefit from the simplification of a new card support and the capability of the wide sample rates support through ASRC. -Note: The card is initially designed for those sound cards who use I2S and - PCM DAI formats. However, it'll be also possible to support those non - I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as long - as the driver has been properly upgraded. +Note: The card is initially designed for those sound cards who use AC'97, I2S + and PCM DAI formats. However, it'll be also possible to support those non + AC'97/I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as + long as the driver has been properly upgraded. The compatible list for this generic sound card currently: + "fsl,imx-audio-ac97" + "fsl,imx-audio-cs42888" "fsl,imx-audio-wm8962" diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 5aeb6ed..86aa498 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -14,6 +14,9 @@ #include <linux/i2c.h> #include <linux/module.h> #include <linux/of_platform.h> +#if IS_ENABLED(CONFIG_SND_AC97_CODEC) +#include <sound/ac97_codec.h> +#endif #include <sound/pcm_params.h> #include <sound/soc.h> @@ -115,6 +118,11 @@ static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = { SND_SOC_DAPM_MIC("DMIC", NULL), }; +static bool fsl_asoc_card_is_ac97(struct fsl_asoc_card_priv *priv) +{ + return priv->dai_fmt == SND_SOC_DAIFMT_AC97; +} + static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -133,7 +141,9 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, * set_bias_level(), bypass the remaining settings in hw_params(). * Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS. */ - if (priv->card.set_bias_level && priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) + if ((priv->card.set_bias_level && + priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) || + fsl_asoc_card_is_ac97(priv)) return 0; /* Specific configurations of DAIs starts from here */ @@ -300,7 +310,7 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, ext_port--; /* - * Use asynchronous mode (6 wires) for all cases. + * Use asynchronous mode (6 wires) for all cases except AC97. * If only 4 wires are needed, just set SSI into * synchronous mode and enable 4 PADs in IOMUX. */ @@ -346,15 +356,30 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, IMX_AUDMUX_V2_PTCR_TCLKDIR; break; default: - return -EINVAL; + if (!fsl_asoc_card_is_ac97(priv)) + return -EINVAL; + } + + if (fsl_asoc_card_is_ac97(priv)) { + int_ptcr = IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCLKDIR; + ext_ptcr = IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSEL(int_port) | + IMX_AUDMUX_V2_PTCR_TFSDIR; } /* Asynchronous mode can not be set along with RCLKDIR */ - ret = imx_audmux_v2_configure_port(int_port, 0, - IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); - if (ret) { - dev_err(dev, "audmux internal port setup failed\n"); - return ret; + if (!fsl_asoc_card_is_ac97(priv)) { + unsigned int pdcr = + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port); + + ret = imx_audmux_v2_configure_port(int_port, 0, + pdcr); + if (ret) { + dev_err(dev, "audmux internal port setup failed\n"); + return ret; + } } ret = imx_audmux_v2_configure_port(int_port, int_ptcr, @@ -364,11 +389,16 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, return ret; } - ret = imx_audmux_v2_configure_port(ext_port, 0, - IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); - if (ret) { - dev_err(dev, "audmux external port setup failed\n"); - return ret; + if (!fsl_asoc_card_is_ac97(priv)) { + unsigned int pdcr = + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port); + + ret = imx_audmux_v2_configure_port(ext_port, 0, + pdcr); + if (ret) { + dev_err(dev, "audmux external port setup failed\n"); + return ret; + } } ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr, @@ -389,6 +419,23 @@ static int fsl_asoc_card_late_probe(struct snd_soc_card *card) struct device *dev = card->dev; int ret; + if (fsl_asoc_card_is_ac97(priv)) { +#if IS_ENABLED(CONFIG_SND_AC97_CODEC) + struct snd_soc_codec *codec = card->rtd[0].codec; + struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); + + /* + * Use slots 3/4 for S/PDIF so SSI won't try to enable + * other slots and send some samples there + * due to SLOTREQ bits for S/PDIF received from codec + */ + snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS, + AC97_EA_SPSA_SLOT_MASK, AC97_EA_SPSA_3_4); +#endif + + return 0; + } + ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, codec_priv->mclk_freq, SND_SOC_CLOCK_IN); if (ret) { @@ -404,10 +451,9 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) struct device_node *cpu_np, *codec_np, *asrc_np; struct device_node *np = pdev->dev.of_node; struct platform_device *asrc_pdev = NULL; - struct platform_device *cpu_pdev; + struct platform_device *cpu_pdev, *codec_pdev; struct fsl_asoc_card_priv *priv; - struct i2c_client *codec_dev; - struct clk *codec_clk; + struct i2c_client *codec_i2c_dev; const char *codec_dai_name; u32 width; int ret; @@ -420,9 +466,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Give a chance to old DT binding */ if (!cpu_np) cpu_np = of_parse_phandle(np, "ssi-controller", 0); - codec_np = of_parse_phandle(np, "audio-codec", 0); - if (!cpu_np || !codec_np) { - dev_err(&pdev->dev, "phandle missing or invalid\n"); + if (!cpu_np) { + dev_err(&pdev->dev, "CPU phandle missing or invalid\n"); ret = -EINVAL; goto fail; } @@ -434,11 +479,13 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) goto fail; } - codec_dev = of_find_i2c_device_by_node(codec_np); - if (!codec_dev) { - dev_err(&pdev->dev, "failed to find codec platform device\n"); - ret = -EINVAL; - goto fail; + codec_np = of_parse_phandle(np, "audio-codec", 0); + if (codec_np) { + codec_pdev = of_find_device_by_node(codec_np); + codec_i2c_dev = of_find_i2c_device_by_node(codec_np); + } else { + codec_pdev = NULL; + codec_i2c_dev = NULL; } asrc_np = of_parse_phandle(np, "audio-asrc", 0); @@ -446,10 +493,13 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) asrc_pdev = of_find_device_by_node(asrc_np); /* Get the MCLK rate only, and leave it controlled by CODEC drivers */ - codec_clk = clk_get(&codec_dev->dev, NULL); - if (!IS_ERR(codec_clk)) { - priv->codec_priv.mclk_freq = clk_get_rate(codec_clk); - clk_put(codec_clk); + if (codec_pdev) { + struct clk *codec_clk = clk_get(&codec_pdev->dev, NULL); + + if (!IS_ERR(codec_clk)) { + priv->codec_priv.mclk_freq = clk_get_rate(codec_clk); + clk_put(codec_clk); + } } /* Default sample rate and format, will be updated in hw_params() */ @@ -486,11 +536,21 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) { + codec_dai_name = "ac97-hifi"; + priv->card.set_bias_level = NULL; + priv->dai_fmt = SND_SOC_DAIFMT_AC97; } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); return -EINVAL; } + if (!fsl_asoc_card_is_ac97(priv) && !codec_pdev) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + ret = -EINVAL; + goto asrc_fail; + } + /* Common settings for corresponding Freescale CPU DAI driver */ if (strstr(cpu_np->name, "ssi")) { /* Only SSI needs to configure AUDMUX */ @@ -507,7 +567,9 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1; } - sprintf(priv->name, "%s-audio", codec_dev->name); + snprintf(priv->name, sizeof(priv->name), "%s-audio", + fsl_asoc_card_is_ac97(priv) ? "ac97" : + codec_i2c_dev ? codec_i2c_dev->name : codec_pdev->name); /* Initialize sound card */ priv->pdev = pdev; @@ -531,8 +593,26 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Normal DAI Link */ priv->dai_link[0].cpu_of_node = cpu_np; - priv->dai_link[0].codec_of_node = codec_np; priv->dai_link[0].codec_dai_name = codec_dai_name; + + if (!fsl_asoc_card_is_ac97(priv)) + priv->dai_link[0].codec_of_node = codec_np; + else { + u32 idx; + + ret = of_property_read_u32(cpu_np, "cell-index", &idx); + if (ret) { + dev_err(&pdev->dev, + "cannot get CPU index property\n"); + goto asrc_fail; + } + + priv->dai_link[0].codec_name = + devm_kasprintf(&pdev->dev, GFP_KERNEL, + "ac97-codec.%u", + (unsigned int)idx); + } + priv->dai_link[0].platform_of_node = cpu_np; priv->dai_link[0].dai_fmt = priv->dai_fmt; priv->card.num_links = 1; @@ -543,6 +623,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->dai_link[1].platform_of_node = asrc_np; priv->dai_link[2].codec_dai_name = codec_dai_name; priv->dai_link[2].codec_of_node = codec_np; + priv->dai_link[2].codec_name = + priv->dai_link[0].codec_name; priv->dai_link[2].cpu_of_node = cpu_np; priv->dai_link[2].dai_fmt = priv->dai_fmt; priv->card.num_links = 3; @@ -578,14 +660,15 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) asrc_fail: of_node_put(asrc_np); -fail: of_node_put(codec_np); +fail: of_node_put(cpu_np); return ret; } static const struct of_device_id fsl_asoc_card_dt_ids[] = { + { .compatible = "fsl,imx-audio-ac97", }, { .compatible = "fsl,imx-audio-cs42888", }, { .compatible = "fsl,imx-audio-sgtl5000", }, { .compatible = "fsl,imx-audio-wm8962", }, -- 2.5.0

3 4

[alsa-devel] [PATCH v2] audio: sai: Add Power Management support
by Alison Wang 17 Sep '15

17 Sep '15

This patch adds Power Management support for SAI. Activate regmap cache with REGCACHE_FLAT, and use regmap cache code to save and restore registers in suspend and resume. The Transmit Control Register (TCSR) and Receive Control Register(RCSR) should be volatile registers. Signed-off-by: Alison Wang <alison.wang(a)freescale.com> --- Changes since v1: - Use REGCACHE_FLAT instead of REGCACHE_RBTREE. - Use SIMPLE_DEV_PM_OPS to simplify the code. sound/soc/fsl/fsl_sai.c | 31 ++++++++++++++++++++++++++++++- 1 file changed, 30 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index a18fd92..030f71e 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -637,9 +637,11 @@ static bool fsl_sai_readable_reg(struct device *dev, unsigned int reg) static bool fsl_sai_volatile_reg(struct device *dev, unsigned int reg) { switch (reg) { + case FSL_SAI_TCSR: case FSL_SAI_TFR: - case FSL_SAI_RFR: case FSL_SAI_TDR: + case FSL_SAI_RCSR: + case FSL_SAI_RFR: case FSL_SAI_RDR: return true; default: @@ -681,6 +683,7 @@ static const struct regmap_config fsl_sai_regmap_config = { .readable_reg = fsl_sai_readable_reg, .volatile_reg = fsl_sai_volatile_reg, .writeable_reg = fsl_sai_writeable_reg, + .cache_type = REGCACHE_FLAT, }; static int fsl_sai_probe(struct platform_device *pdev) @@ -796,6 +799,31 @@ static int fsl_sai_probe(struct platform_device *pdev) return devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); } +#ifdef CONFIG_PM_SLEEP +static int fsl_sai_suspend(struct device *dev) +{ + struct fsl_sai *sai = dev_get_drvdata(dev); + + regcache_cache_only(sai->regmap, true); + regcache_mark_dirty(sai->regmap); + + return 0; +} + +static int fsl_sai_resume(struct device *dev) +{ + struct fsl_sai *sai = dev_get_drvdata(dev); + + /* Restore all registers */ + regcache_cache_only(sai->regmap, false); + regcache_sync(sai->regmap); + + return 0; +} +#endif /* CONFIG_PM_SLEEP */ + +static SIMPLE_DEV_PM_OPS(fsl_sai_pm, fsl_sai_suspend, fsl_sai_resume); + static const struct of_device_id fsl_sai_ids[] = { { .compatible = "fsl,vf610-sai", }, { .compatible = "fsl,imx6sx-sai", }, @@ -807,6 +835,7 @@ static struct platform_driver fsl_sai_driver = { .driver = { .name = "fsl-sai", .of_match_table = fsl_sai_ids, + .pm = &fsl_sai_pm, }, }; module_platform_driver(fsl_sai_driver); -- 2.1.0.27.g96db324

4 4

[alsa-devel] Multiple codecs on one sound card for multi-channel sound card
by Caleb Crome 17 Sep '15

17 Sep '15

Hi all, What's the current best practices to specify multiple codecs on one sound card, with all codecs sharing a single TDM bus? I have a card with up to 32 TLV320AIC33 codecs on it. For the moment, we can limit the discussion to only 16 codecs, so we don't have to get extra serializers involved... I currently have something like this in my am335x-boneblack.dts file: (i.e. attempting to set tdm slot width and num in the i2c codec) but it doesn't seem to be pickin up the fact that I want all the codecs linked together as one. (FYI, the first codec, i.e. tlv32aic3x_a should be the master, and all others including the CPU will be slaves). Thanks for any help! BTW, I'm happy to use whatever kernel will support me with the minimal amount of effort! Sincerely, -Caleb &i2c1 { clock-frequency = <100000>; status = "okay"; pinctrl-names = "default"; pinctrl-0 = <&i2c1_pins_default>; status="okay"; tlv320aic3x_a: tlv320aic3x@18 { #sound-dai-cells = <0>; compatible = "ti,tlv320aic3x"; reg = <0x18>; status = "okay"; dai-tdm-slot-width = <16>; dai-tdm-slot-num = <0>; }; tlv320aic3x_b: tlv320aic3x@19 { #sound-dai-cells = <0>; compatible = "ti,tlv320aic3x"; reg = <0x19>; status = "okay"; dai-tdm-slot-width = <16>; dai-tdm-slot-num = <2>; }; tlv320aic3x_c: tlv320aic3x@1a { compatible = "ti,tlv320aic3x"; reg = <0x1a>; status = "okay"; dai-tdm-slot-width = <16>; dai-tdm-slot-num = <4>; }; tlv320aic3x_d: tlv320aic3x@1b { compatible = "ti,tlv320aic3x"; reg = <0x1b>; status = "okay"; dai-tdm-slot-width = <16>; dai-tdm-slot-num = <6>; }; }; sound { compatible = "simple-audio-card"; simple-audio-card,name = "puck audio"; // simple-audio-card,widgets= // simple-audio-card,routing= simple-audio-card,mclk-fs = <256>; status="okay"; simple-audio-card,bitclock-master = <&tlv320aic3x_a>; simple-audio-card,frame-master = <&tlv320aic3x_a>; simple-audio-card,dai-link@0 { format = "left_j"; cpu { sound-dai = <&mcasp0 0>; }; // *** Here's where I'm really confused -- this part doesn't seem to be supported to // have all the codecs on a single DAI. // The examples given seem to be for different codec DAIs // sharing a single CPU DAI, but not used at the same time. codec { #sound-dai-cells = <0>; sound-dai = < &tlv320aic3x_a &tlv320aic3x_b &tlv320aic3x_c &tlv320aic3x_d >; }; }; };

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Re: [alsa-devel] [linux-sunxi] Re: [PATCH 2/2] ASoC: sunxi: add support for the on-chip codec on early Allwinner SoCs
by Mark Brown 17 Sep '15

17 Sep '15

On Thu, Sep 17, 2015 at 11:10:41PM +0800, Chen-Yu Tsai wrote: > On Thu, Sep 17, 2015 at 9:31 PM, Maxime Ripard > >> > + .formats = SNDRV_PCM_FMTBIT_S16_LE | > >> > + SNDRV_PCM_FMTBIT_S32_LE, > >> > + .sig_bits = 24, > >> So presumably also S24_LE (ie, 24 bits packed into a 32 bit word)? > > Hmm, probably yes, I'll test that. > IIRC when Emilio first wrote the driver, we tried 24 bit and no sound > came out. Turns out it's an alignment issue. The codec's FIFO register > is 32 bits wide, and takes the higher 24 bits as input when set to 24 > bit mode. The internal FIFO is only 24 bits wide. A20 user manual P174 > describes how the bits are copied. > So for 24 bit audio, you would actually send it 32 bit audio samples, > and let it truncate or drop the least significant 8 bits. This is why > we have SNDRV_PCM_FMTBIT_S32_LE with .sig_bits = 24. > I don't know if this is just a workaround, but a few other drivers do > this as well, for example twl3040 and omap-mcpdm. That's very common - it's essentially what S24_LE mode is.

1 0

[alsa-devel] [PATCH v2] ASoC: wm5110: Add additional analogue input enable for early revs
by Charles Keepax 17 Sep '15

17 Sep '15

Earlier revisions of the wm5110/8280 silicon require a slightly more complex procedure to enable analogue inputs. This patch adds this into the driver. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- Changes since v2: - Added a comment to explain why we need to hold the DAPM lock when setting the PGA volume for the inputs Thanks, Charles sound/soc/codecs/wm5110.c | 187 +++++++++++++++++++++++++++++++++++++++------ 1 files changed, 163 insertions(+), 24 deletions(-) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 9756578..c04c0bc 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -38,6 +38,12 @@ struct wm5110_priv { struct arizona_priv core; struct arizona_fll fll[2]; + + unsigned int in_value; + int in_pre_pending; + int in_post_pending; + + unsigned int in_pga_cache[6]; }; static const struct wm_adsp_region wm5110_dsp1_regions[] = { @@ -428,6 +434,127 @@ err: return ret; } +static int wm5110_in_pga_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct snd_soc_card *card = dapm->card; + int ret; + + /* + * PGA Volume is also used as part of the enable sequence, so + * usage of it should be avoided whilst that is running. + */ + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + + ret = snd_soc_get_volsw_range(kcontrol, ucontrol); + + mutex_unlock(&card->dapm_mutex); + + return ret; +} + +static int wm5110_in_pga_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); + struct snd_soc_dapm_context *dapm = snd_soc_codec_get_dapm(codec); + struct snd_soc_card *card = dapm->card; + int ret; + + /* + * PGA Volume is also used as part of the enable sequence, so + * usage of it should be avoided whilst that is running. + */ + mutex_lock_nested(&card->dapm_mutex, SND_SOC_DAPM_CLASS_RUNTIME); + + ret = snd_soc_put_volsw_range(kcontrol, ucontrol); + + mutex_unlock(&card->dapm_mutex); + + return ret; +} + +static int wm5110_in_analog_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct wm5110_priv *wm5110 = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + unsigned int reg, mask; + struct reg_sequence analog_seq[] = { + { 0x80, 0x3 }, + { 0x35d, 0 }, + { 0x80, 0x0 }, + }; + + reg = ARIZONA_IN1L_CONTROL + ((w->shift ^ 0x1) * 4); + mask = ARIZONA_IN1L_PGA_VOL_MASK; + + switch (event) { + case SND_SOC_DAPM_WILL_PMU: + wm5110->in_value |= 0x3 << ((w->shift ^ 0x1) * 2); + wm5110->in_pre_pending++; + wm5110->in_post_pending++; + return 0; + case SND_SOC_DAPM_PRE_PMU: + wm5110->in_pga_cache[w->shift] = snd_soc_read(codec, reg); + + snd_soc_update_bits(codec, reg, mask, + 0x40 << ARIZONA_IN1L_PGA_VOL_SHIFT); + + wm5110->in_pre_pending--; + if (wm5110->in_pre_pending == 0) { + analog_seq[1].def = wm5110->in_value; + regmap_multi_reg_write_bypassed(arizona->regmap, + analog_seq, + ARRAY_SIZE(analog_seq)); + + msleep(55); + + wm5110->in_value = 0; + } + + break; + case SND_SOC_DAPM_POST_PMU: + snd_soc_update_bits(codec, reg, mask, + wm5110->in_pga_cache[w->shift]); + + wm5110->in_post_pending--; + if (wm5110->in_post_pending == 0) + regmap_multi_reg_write_bypassed(arizona->regmap, + analog_seq, + ARRAY_SIZE(analog_seq)); + break; + default: + break; + } + + return 0; +} + +static int wm5110_in_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); + struct arizona *arizona = priv->arizona; + + switch (arizona->rev) { + case 0 ... 4: + if (arizona_input_analog(codec, w->shift)) + wm5110_in_analog_ev(w, kcontrol, event); + + break; + default: + break; + } + + return arizona_in_ev(w, kcontrol, event); +} + static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); @@ -454,18 +581,24 @@ SOC_ENUM("IN2 OSR", arizona_in_dmic_osr[1]), SOC_ENUM("IN3 OSR", arizona_in_dmic_osr[2]), SOC_ENUM("IN4 OSR", arizona_in_dmic_osr[3]), -SOC_SINGLE_RANGE_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL, - ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_SINGLE_RANGE_TLV("IN1R Volume", ARIZONA_IN1R_CONTROL, - ARIZONA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_SINGLE_RANGE_TLV("IN2L Volume", ARIZONA_IN2L_CONTROL, - ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_SINGLE_RANGE_TLV("IN2R Volume", ARIZONA_IN2R_CONTROL, - ARIZONA_IN2R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_SINGLE_RANGE_TLV("IN3L Volume", ARIZONA_IN3L_CONTROL, - ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), -SOC_SINGLE_RANGE_TLV("IN3R Volume", ARIZONA_IN3R_CONTROL, - ARIZONA_IN3R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, ana_tlv), +SOC_SINGLE_RANGE_EXT_TLV("IN1L Volume", ARIZONA_IN1L_CONTROL, + ARIZONA_IN1L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, + wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv), +SOC_SINGLE_RANGE_EXT_TLV("IN1R Volume", ARIZONA_IN1R_CONTROL, + ARIZONA_IN1R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, + wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv), +SOC_SINGLE_RANGE_EXT_TLV("IN2L Volume", ARIZONA_IN2L_CONTROL, + ARIZONA_IN2L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, + wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv), +SOC_SINGLE_RANGE_EXT_TLV("IN2R Volume", ARIZONA_IN2R_CONTROL, + ARIZONA_IN2R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, + wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv), +SOC_SINGLE_RANGE_EXT_TLV("IN3L Volume", ARIZONA_IN3L_CONTROL, + ARIZONA_IN3L_PGA_VOL_SHIFT, 0x40, 0x5f, 0, + wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv), +SOC_SINGLE_RANGE_EXT_TLV("IN3R Volume", ARIZONA_IN3R_CONTROL, + ARIZONA_IN3R_PGA_VOL_SHIFT, 0x40, 0x5f, 0, + wm5110_in_pga_get, wm5110_in_pga_put, ana_tlv), SOC_ENUM("IN HPF Cutoff Frequency", arizona_in_hpf_cut_enum), @@ -896,29 +1029,35 @@ SND_SOC_DAPM_OUTPUT("DRC1 Signal Activity"), SND_SOC_DAPM_OUTPUT("DRC2 Signal Activity"), SND_SOC_DAPM_PGA_E("IN1L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1L_ENA_SHIFT, - 0, NULL, 0, arizona_in_ev, + 0, NULL, 0, wm5110_in_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_WILL_PMU), SND_SOC_DAPM_PGA_E("IN1R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN1R_ENA_SHIFT, - 0, NULL, 0, arizona_in_ev, + 0, NULL, 0, wm5110_in_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_WILL_PMU), SND_SOC_DAPM_PGA_E("IN2L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2L_ENA_SHIFT, - 0, NULL, 0, arizona_in_ev, + 0, NULL, 0, wm5110_in_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_WILL_PMU), SND_SOC_DAPM_PGA_E("IN2R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN2R_ENA_SHIFT, - 0, NULL, 0, arizona_in_ev, + 0, NULL, 0, wm5110_in_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_WILL_PMU), SND_SOC_DAPM_PGA_E("IN3L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3L_ENA_SHIFT, - 0, NULL, 0, arizona_in_ev, + 0, NULL, 0, wm5110_in_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_WILL_PMU), SND_SOC_DAPM_PGA_E("IN3R PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN3R_ENA_SHIFT, - 0, NULL, 0, arizona_in_ev, + 0, NULL, 0, wm5110_in_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | - SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_WILL_PMU), SND_SOC_DAPM_PGA_E("IN4L PGA", ARIZONA_INPUT_ENABLES, ARIZONA_IN4L_ENA_SHIFT, 0, NULL, 0, arizona_in_ev, SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD | -- 1.7.2.5

2 1

[alsa-devel] [PATCH v2] ASoC: soc-core: Fix sparse warning in be32_to_cpup() call
by Jyri Sarha 17 Sep '15

17 Sep '15

Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1929f0e..af4999c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3294,7 +3294,7 @@ static int snd_soc_of_get_slot_mask(struct device_node *np, unsigned int *mask) { u32 val; - const u32 *of_slot_mask = of_get_property(np, prop_name, &val); + const __be32 *of_slot_mask = of_get_property(np, prop_name, &val); int i; if (!of_slot_mask) -- 1.9.1

2 1

[alsa-devel] [PATCH 0/2] change PTR_ERR argument
by Julia Lawall 17 Sep '15

17 Sep '15

Apply PTR_ERR to the value that was recently assigned. --- var/julia/linuxcopy/drivers/spi/spi-ep93xx.c | 3 ++- var/julia/linuxcopy/sound/soc/qcom/lpass-cpu.c | 3 ++- 2 files changed, 4 insertions(+), 2 deletions(-)

4 11

[alsa-devel] [PATCH v2] ASoC: davinci-mcasp: Get rid of bclk_lrclk_ratio in private data
by Jyri Sarha 17 Sep '15

17 Sep '15

The slot_width is for essentially same thing. Instead of storing bclk_lrclk_ratio, just store the slot_width. Comments has been updated accordingly and some variable names changed to more descriptive. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- Added the missing __func__ to dev_warn() warn call in __davinci_mcasp_set_clkdiv(). sound/soc/davinci/davinci-mcasp.c | 56 ++++++++++++++++++++++----------------- 1 file changed, 31 insertions(+), 25 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index de1e3a8..9a9d633 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -87,7 +87,6 @@ struct davinci_mcasp { u8 *serial_dir; u8 version; u8 bclk_div; - u16 bclk_lrclk_ratio; int streams; u32 irq_request[2]; int dma_request[2]; @@ -558,8 +557,21 @@ static int __davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id, mcasp->bclk_div = div; break; - case 2: /* BCLK/LRCLK ratio */ - mcasp->bclk_lrclk_ratio = div; + case 2: /* + * BCLK/LRCLK ratio descries how many bit-clock cycles + * fit into one frame. The clock ratio is given for a + * full period of data (for I2S format both left and + * right channels), so it has to be divided by number + * of tdm-slots (for I2S - divided by 2). + * Instead of storing this ratio, we calculate a new + * tdm_slot width by dividing the the ratio by the + * number of configured tdm slots. + */ + mcasp->slot_width = div / mcasp->tdm_slots; + if (div % mcasp->tdm_slots) + dev_warn(mcasp->dev, + "%s(): BCLK/LRCLK %d is not divisible by %d tdm slots", + __func__, div, mcasp->tdm_slots); break; default: @@ -682,11 +694,13 @@ static int davinci_mcasp_set_tdm_slot(struct snd_soc_dai *dai, } static int davinci_config_channel_size(struct davinci_mcasp *mcasp, - int word_length) + int sample_width) { u32 fmt; - u32 tx_rotate = (word_length / 4) & 0x7; - u32 mask = (1ULL << word_length) - 1; + u32 tx_rotate = (sample_width / 4) & 0x7; + u32 mask = (1ULL << sample_width) - 1; + u32 slot_width = sample_width; + /* * For captured data we should not rotate, inversion and masking is * enoguh to get the data to the right position: @@ -699,31 +713,23 @@ static int davinci_config_channel_size(struct davinci_mcasp *mcasp, u32 rx_rotate = 0; /* - * if s BCLK-to-LRCLK ratio has been configured via the set_clkdiv() - * callback, take it into account here. That allows us to for example - * send 32 bits per channel to the codec, while only 16 of them carry - * audio payload. - * The clock ratio is given for a full period of data (for I2S format - * both left and right channels), so it has to be divided by number of - * tdm-slots (for I2S - divided by 2). + * Setting the tdm slot width either with set_clkdiv() or + * set_tdm_slot() allows us to for example send 32 bits per + * channel to the codec, while only 16 of them carry audio + * payload. */ - if (mcasp->bclk_lrclk_ratio) { - u32 slot_length = mcasp->bclk_lrclk_ratio / mcasp->tdm_slots; - + if (mcasp->slot_width) { /* - * When we have more bclk then it is needed for the data, we - * need to use the rotation to move the received samples to have - * correct alignment. + * When we have more bclk then it is needed for the + * data, we need to use the rotation to move the + * received samples to have correct alignment. */ - rx_rotate = (slot_length - word_length) / 4; - word_length = slot_length; - } else if (mcasp->slot_width) { - rx_rotate = (mcasp->slot_width - word_length) / 4; - word_length = mcasp->slot_width; + slot_width = mcasp->slot_width; + rx_rotate = (slot_width - sample_width) / 4; } /* mapping of the XSSZ bit-field as described in the datasheet */ - fmt = (word_length >> 1) - 1; + fmt = (slot_width >> 1) - 1; if (mcasp->op_mode != DAVINCI_MCASP_DIT_MODE) { mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, RXSSZ(fmt), -- 1.9.1

2 1

[alsa-devel] [asoc:topic/sunxi 2/2] sound/soc/sunxi/sun4i-codec.c:708:3-8: No need to set .owner here. The core will do it.
by kbuild test robot 17 Sep '15

17 Sep '15

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/sunxi head: 45fb6b6f2aa3f6b22b81078db0dba4b26c9d0bdb commit: 45fb6b6f2aa3f6b22b81078db0dba4b26c9d0bdb [2/2] ASoC: sunxi: add support for the on-chip codec on early Allwinner SoCs coccinelle warnings: (new ones prefixed by >>) >> sound/soc/sunxi/sun4i-codec.c:708:3-8: No need to set .owner here. The core will do it. Please review and possibly fold the followup patch. --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

3 3

[alsa-devel] [PATCH v2] ASoC: add ak4613 support
by Kuninori Morimoto 17 Sep '15

17 Sep '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- v1 -> v2 - fix for checkpatch - adds ak4613 in SND_SOC_ALL_CODECS - ak4613_write() -> snd_soc_write() - ak4613_read() was removed - spin lock -> mutex - ak4613_resume() returns regcache_sync()'s result Documentation/devicetree/bindings/sound/ak4613.txt | 17 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4613.c | 469 +++++++++++++++++++++ 4 files changed, 493 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ak4613.txt create mode 100644 sound/soc/codecs/ak4613.c diff --git a/Documentation/devicetree/bindings/sound/ak4613.txt b/Documentation/devicetree/bindings/sound/ak4613.txt new file mode 100644 index 0000000..15a9195 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4613.txt @@ -0,0 +1,17 @@ +AK4613 I2C transmitter + +This device supports I2C mode only. + +Required properties: + +- compatible : "asahi-kasei,ak4613" +- reg : The chip select number on the I2C bus + +Example: + +&i2c { + ak4613: ak4613@0x10 { + compatible = "asahi-kasei,ak4613"; + reg = <0x10>; + }; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0c9733e..a92e4d4 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -36,6 +36,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C select SND_SOC_AK4554 + select SND_SOC_AK4613 if I2C select SND_SOC_AK4641 if I2C select SND_SOC_AK4642 if I2C select SND_SOC_AK4671 if I2C @@ -319,6 +320,10 @@ config SND_SOC_AK4535 config SND_SOC_AK4554 tristate "AKM AK4554 CODEC" +config SND_SOC_AK4613 + tristate "AKM AK4613 CODEC" + depends on I2C + config SND_SOC_AK4641 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 4a32077..5b6c8af 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -26,6 +26,7 @@ snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o snd-soc-ak4554-objs := ak4554.o +snd-soc-ak4613-objs := ak4613.o snd-soc-ak4641-objs := ak4641.o snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o @@ -216,6 +217,7 @@ obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4554) += snd-soc-ak4554.o +obj-$(CONFIG_SND_SOC_AK4613) += snd-soc-ak4613.o obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c new file mode 100644 index 0000000..fd96a8f --- /dev/null +++ b/sound/soc/codecs/ak4613.c @@ -0,0 +1,469 @@ +/* + * ak4613.c -- Asahi Kasei ALSA Soc Audio driver + * + * Copyright (C) 2015 Renesas Electronics Corporation + * Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> + * + * Based on ak4642.c by Kuninori Morimoto + * Based on wm8731.c by Richard Purdie + * Based on ak4535.c by Richard Purdie + * Based on wm8753.c by Liam Girdwood + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/clk.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/of_device.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> + +#define PW_MGMT1 0x00 /* Power Management 1 */ +#define PW_MGMT2 0x01 /* Power Management 2 */ +#define PW_MGMT3 0x02 /* Power Management 3 */ +#define CTRL1 0x03 /* Control 1 */ +#define CTRL2 0x04 /* Control 2 */ +#define DEMP1 0x05 /* De-emphasis1 */ +#define DEMP2 0x06 /* De-emphasis2 */ +#define OFD 0x07 /* Overflow Detect */ +#define ZRD 0x08 /* Zero Detect */ +#define ICTRL 0x09 /* Input Control */ +#define OCTRL 0x0a /* Output Control */ +#define LOUT1 0x0b /* LOUT1 Volume Control */ +#define ROUT1 0x0c /* ROUT1 Volume Control */ +#define LOUT2 0x0d /* LOUT2 Volume Control */ +#define ROUT2 0x0e /* ROUT2 Volume Control */ +#define LOUT3 0x0f /* LOUT3 Volume Control */ +#define ROUT3 0x10 /* ROUT3 Volume Control */ +#define LOUT4 0x11 /* LOUT4 Volume Control */ +#define ROUT4 0x12 /* ROUT4 Volume Control */ +#define LOUT5 0x13 /* LOUT5 Volume Control */ +#define ROUT5 0x14 /* ROUT5 Volume Control */ +#define LOUT6 0x15 /* LOUT6 Volume Control */ +#define ROUT6 0x16 /* ROUT6 Volume Control */ + +/* PW_MGMT1 */ +#define RSTN BIT(0) +#define PMDAC BIT(1) +#define PMADC BIT(2) +#define PMVR BIT(3) + +/* PW_MGMT2 */ +#define PMAD_ALL 0x7 + +/* PW_MGMT3 */ +#define PMDA_ALL 0x3f + +/* CTRL1 */ +#define DIF0 BIT(3) +#define DIF1 BIT(4) +#define DIF2 BIT(5) +#define TDM0 BIT(6) +#define TDM1 BIT(7) +#define NO_FMT (0xff) +#define FMT_MASK (0xf8) + +/* CTRL2 */ +#define DFS_NORMAL_SPEED (0 << 2) +#define DFS_DOUBLE_SPEED (1 << 2) +#define DFS_QUAD_SPEED (2 << 2) + +struct ak4613_priv { + struct mutex lock; + + unsigned int fmt; + u8 fmt_ctrl; + int cnt; +}; + +struct ak4613_formats { + unsigned int width; + unsigned int fmt; +}; + +struct ak4613_interface { + struct ak4613_formats capture; + struct ak4613_formats playback; +}; + +static const struct reg_default ak4613_reg[] = { + { 0x0, 0x0f }, { 0x1, 0x07 }, { 0x2, 0x3f }, { 0x3, 0x20 }, + { 0x4, 0x20 }, { 0x5, 0x55 }, { 0x6, 0x05 }, { 0x7, 0x07 }, + { 0x8, 0x0f }, { 0x9, 0x07 }, { 0xa, 0x3f }, { 0xb, 0x00 }, + { 0xc, 0x00 }, { 0xd, 0x00 }, { 0xe, 0x00 }, { 0xf, 0x00 }, + { 0x10, 0x00 }, { 0x11, 0x00 }, { 0x12, 0x00 }, { 0x13, 0x00 }, + { 0x14, 0x00 }, { 0x15, 0x00 }, { 0x16, 0x00 }, +}; + +#define AUDIO_IFACE_IDX_TO_VAL(i) (i << 3) +#define AUDIO_IFACE(b, fmt) { b, SND_SOC_DAIFMT_##fmt } +static const struct ak4613_interface ak4613_iface[] = { + /* capture */ /* playback */ + [0] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(16, RIGHT_J) }, + [1] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(20, RIGHT_J) }, + [2] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(24, RIGHT_J) }, + [3] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(24, LEFT_J) }, + [4] = { AUDIO_IFACE(24, I2S), AUDIO_IFACE(24, I2S) }, +}; + +static const struct regmap_config ak4613_regmap_cfg = { + .reg_bits = 8, + .val_bits = 8, + .max_register = 0x16, + .reg_defaults = ak4613_reg, + .num_reg_defaults = ARRAY_SIZE(ak4613_reg), +}; + +static const struct of_device_id ak4613_of_match[] = { + { .compatible = "asahi-kasei,ak4613", .data = &ak4613_regmap_cfg }, + {}, +}; +MODULE_DEVICE_TABLE(of, ak4613_of_match); + +static const struct i2c_device_id ak4613_i2c_id[] = { + { "ak4613", (kernel_ulong_t)&ak4613_regmap_cfg }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ak4613_i2c_id); + +static const struct snd_soc_dapm_widget ak4613_dapm_widgets[] = { + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("LOUT1"), + SND_SOC_DAPM_OUTPUT("LOUT2"), + SND_SOC_DAPM_OUTPUT("LOUT3"), + SND_SOC_DAPM_OUTPUT("LOUT4"), + SND_SOC_DAPM_OUTPUT("LOUT5"), + SND_SOC_DAPM_OUTPUT("LOUT6"), + + SND_SOC_DAPM_OUTPUT("ROUT1"), + SND_SOC_DAPM_OUTPUT("ROUT2"), + SND_SOC_DAPM_OUTPUT("ROUT3"), + SND_SOC_DAPM_OUTPUT("ROUT4"), + SND_SOC_DAPM_OUTPUT("ROUT5"), + SND_SOC_DAPM_OUTPUT("ROUT6"), + + /* Inputs */ + SND_SOC_DAPM_INPUT("LIN1"), + SND_SOC_DAPM_INPUT("LIN2"), + + SND_SOC_DAPM_INPUT("RIN1"), + SND_SOC_DAPM_INPUT("RIN2"), + + /* DAC */ + SND_SOC_DAPM_DAC("DAC1", NULL, PW_MGMT3, 0, 0), + SND_SOC_DAPM_DAC("DAC2", NULL, PW_MGMT3, 1, 0), + SND_SOC_DAPM_DAC("DAC3", NULL, PW_MGMT3, 2, 0), + SND_SOC_DAPM_DAC("DAC4", NULL, PW_MGMT3, 3, 0), + SND_SOC_DAPM_DAC("DAC5", NULL, PW_MGMT3, 4, 0), + SND_SOC_DAPM_DAC("DAC6", NULL, PW_MGMT3, 5, 0), + + /* ADC */ + SND_SOC_DAPM_ADC("ADC1", NULL, PW_MGMT2, 0, 0), + SND_SOC_DAPM_ADC("ADC2", NULL, PW_MGMT2, 1, 0), +}; + +static const struct snd_soc_dapm_route ak4613_intercon[] = { + {"LOUT1", NULL, "DAC1"}, + {"LOUT2", NULL, "DAC2"}, + {"LOUT3", NULL, "DAC3"}, + {"LOUT4", NULL, "DAC4"}, + {"LOUT5", NULL, "DAC5"}, + {"LOUT6", NULL, "DAC6"}, + + {"ROUT1", NULL, "DAC1"}, + {"ROUT2", NULL, "DAC2"}, + {"ROUT3", NULL, "DAC3"}, + {"ROUT4", NULL, "DAC4"}, + {"ROUT5", NULL, "DAC5"}, + {"ROUT6", NULL, "DAC6"}, + + {"DAC1", NULL, "Playback"}, + {"DAC2", NULL, "Playback"}, + {"DAC3", NULL, "Playback"}, + {"DAC4", NULL, "Playback"}, + {"DAC5", NULL, "Playback"}, + {"DAC6", NULL, "Playback"}, + + {"Capture", NULL, "ADC1"}, + {"Capture", NULL, "ADC2"}, + + {"ADC1", NULL, "LIN1"}, + {"ADC2", NULL, "LIN2"}, + + {"ADC1", NULL, "RIN1"}, + {"ADC2", NULL, "RIN2"}, +}; + +static void ak4613_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); + struct device *dev = codec->dev; + + mutex_lock(&priv->lock); + priv->cnt--; + if (priv->cnt < 0) { + dev_err(dev, "unexpected counter error\n"); + priv->cnt = 0; + } + if (!priv->cnt) + priv->fmt_ctrl = NO_FMT; + mutex_unlock(&priv->lock); +} + +static int ak4613_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); + + fmt &= SND_SOC_DAIFMT_FORMAT_MASK; + + switch (fmt) { + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_I2S: + priv->fmt = fmt; + + break; + default: + return -EINVAL; + } + + return 0; +} + +static int ak4613_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); + const struct ak4613_formats *fmts; + struct device *dev = codec->dev; + unsigned int width = params_width(params); + unsigned int fmt = priv->fmt; + unsigned int rate; + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int i, ret; + u8 fmt_ctrl, ctrl2; + + rate = params_rate(params); + switch (rate) { + case 32000: + case 44100: + case 48000: + ctrl2 = DFS_NORMAL_SPEED; + break; + case 88200: + case 96000: + ctrl2 = DFS_DOUBLE_SPEED; + break; + case 176400: + case 192000: + ctrl2 = DFS_QUAD_SPEED; + break; + default: + return -EINVAL; + } + + /* + * FIXME + * + * It doesn't support TDM at this point + */ + fmt_ctrl = NO_FMT; + for (i = 0; i < ARRAY_SIZE(ak4613_iface); i++) { + fmts = (is_play) ? &ak4613_iface[i].playback : + &ak4613_iface[i].capture; + + if (fmts->fmt != fmt) + continue; + + if (fmt == SND_SOC_DAIFMT_RIGHT_J) { + if (fmts->width != width) + continue; + } else { + if (fmts->width < width) + continue; + } + + fmt_ctrl = AUDIO_IFACE_IDX_TO_VAL(i); + break; + } + + ret = -EINVAL; + if (fmt_ctrl == NO_FMT) + goto hw_params_end; + + mutex_lock(&priv->lock); + if ((priv->fmt_ctrl == NO_FMT) || + (priv->fmt_ctrl == fmt_ctrl)) { + priv->fmt_ctrl = fmt_ctrl; + priv->cnt++; + ret = 0; + } + mutex_unlock(&priv->lock); + + if (ret < 0) + goto hw_params_end; + + snd_soc_update_bits(codec, CTRL1, FMT_MASK, fmt_ctrl); + snd_soc_write(codec, CTRL2, ctrl2); + +hw_params_end: + if (ret < 0) + dev_warn(dev, "unsupported data width/format combination\n"); + + return ret; +} + +static int ak4613_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u8 mgmt1 = 0; + + switch (level) { + case SND_SOC_BIAS_ON: + mgmt1 |= RSTN; + /* fall through */ + case SND_SOC_BIAS_PREPARE: + mgmt1 |= PMADC | PMDAC; + /* fall through */ + case SND_SOC_BIAS_STANDBY: + mgmt1 |= PMVR; + /* fall through */ + case SND_SOC_BIAS_OFF: + default: + break; + } + + snd_soc_write(codec, PW_MGMT1, mgmt1); + + return 0; +} + +static const struct snd_soc_dai_ops ak4613_dai_ops = { + .shutdown = ak4613_dai_shutdown, + .set_fmt = ak4613_dai_set_fmt, + .hw_params = ak4613_dai_hw_params, +}; + +#define AK4613_PCM_RATE (SNDRV_PCM_RATE_32000 |\ + SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_64000 |\ + SNDRV_PCM_RATE_88200 |\ + SNDRV_PCM_RATE_96000 |\ + SNDRV_PCM_RATE_176400 |\ + SNDRV_PCM_RATE_192000) +#define AK4613_PCM_FMTBIT (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver ak4613_dai = { + .name = "ak4613-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = AK4613_PCM_RATE, + .formats = AK4613_PCM_FMTBIT, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = AK4613_PCM_RATE, + .formats = AK4613_PCM_FMTBIT, + }, + .ops = &ak4613_dai_ops, + .symmetric_rates = 1, +}; + +static int ak4613_resume(struct snd_soc_codec *codec) +{ + struct regmap *regmap = dev_get_regmap(codec->dev, NULL); + + regcache_mark_dirty(regmap); + return regcache_sync(regmap); +} + +static struct snd_soc_codec_driver soc_codec_dev_ak4613 = { + .resume = ak4613_resume, + .set_bias_level = ak4613_set_bias_level, + .dapm_widgets = ak4613_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak4613_dapm_widgets), + .dapm_routes = ak4613_intercon, + .num_dapm_routes = ARRAY_SIZE(ak4613_intercon), +}; + +static int ak4613_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct device_node *np = dev->of_node; + const struct regmap_config *regmap_cfg; + struct regmap *regmap; + struct ak4613_priv *priv; + + regmap_cfg = NULL; + if (np) { + const struct of_device_id *of_id; + + of_id = of_match_device(ak4613_of_match, dev); + if (of_id) + regmap_cfg = of_id->data; + } else { + regmap_cfg = (const struct regmap_config *)id->driver_data; + } + + if (!regmap_cfg) + return -EINVAL; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->fmt_ctrl = NO_FMT; + priv->cnt = 0; + + mutex_init(&priv->lock); + + i2c_set_clientdata(i2c, priv); + + regmap = devm_regmap_init_i2c(i2c, regmap_cfg); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return snd_soc_register_codec(dev, &soc_codec_dev_ak4613, + &ak4613_dai, 1); +} + +static int ak4613_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static struct i2c_driver ak4613_i2c_driver = { + .driver = { + .name = "ak4613-codec", + .owner = THIS_MODULE, + .of_match_table = ak4613_of_match, + }, + .probe = ak4613_i2c_probe, + .remove = ak4613_i2c_remove, + .id_table = ak4613_i2c_id, +}; + +module_i2c_driver(ak4613_i2c_driver); + +MODULE_DESCRIPTION("Soc AK4613 driver"); +MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>"); +MODULE_LICENSE("GPL v2"); -- 1.9.1

2 1

[alsa-devel] [PATCH v3] ASoC: soc-core: Fix sparse warning in be32_to_cpup() call
by Jyri Sarha 17 Sep '15

17 Sep '15

Store return value of of_get_property() to a pointer of __be32 type as device tree has big endian type. This fixes a sparse warning couple of lines later when be32_to_cpup() is used to convert from big endian to cpu endian. The whole conversion is not really necessary, as we are only checking if the value is zero or not, but I wanted to add it to remind in the future that the data has to be converted before use. Compiler should optimize the unnecessary operations away. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- Damn, I wrote description ready but forgot to produce the patch again. So here is a v3 of the same one liner patch. sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1929f0e..af4999c 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3294,7 +3294,7 @@ static int snd_soc_of_get_slot_mask(struct device_node *np, unsigned int *mask) { u32 val; - const u32 *of_slot_mask = of_get_property(np, prop_name, &val); + const __be32 *of_slot_mask = of_get_property(np, prop_name, &val); int i; if (!of_slot_mask) -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: soc-core: Fix sparse warning in be32_to_cpup() call
by Jyri Sarha 17 Sep '15

17 Sep '15

Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 1929f0e..614b831 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3301,7 +3301,7 @@ static int snd_soc_of_get_slot_mask(struct device_node *np, return 0; val /= sizeof(u32); for (i = 0; i < val; i++) - if (be32_to_cpup(&of_slot_mask[i])) + if (be32_to_cpup((__be32 *)&of_slot_mask[i])) *mask |= (1 << i); return val; -- 1.9.1

2 2

[alsa-devel] [PATCH 1/4] ASoC: arizona: Add default cases for event switches
by Charles Keepax 17 Sep '15

17 Sep '15

Since the addition of the WILL_PMU / WILL_PMD several of the switches in arizona.c no longer cover all cases or have a default case. Whilst this isn't causing any problems in the interests of robustness add default cases. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/arizona.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index 8a2221a..1bc19f1 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -147,6 +147,8 @@ static int arizona_spk_ev(struct snd_soc_dapm_widget *w, 0x4f5, 0x0da); } break; + default: + break; } return 0; @@ -725,6 +727,9 @@ int arizona_in_ev(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, reg = snd_soc_read(codec, ARIZONA_INPUT_ENABLES); if (reg == 0) arizona_in_set_vu(codec, 0); + break; + default: + break; } return 0; @@ -806,6 +811,8 @@ int arizona_out_ev(struct snd_soc_dapm_widget *w, break; } break; + default: + break; } return 0; -- 1.7.2.5

2 8

[alsa-devel] Test device for automated testing
by Uli Franke 17 Sep '15

17 Sep '15

Hi everybody, we're currently writing a lot of ALSA user space applications. In order to test this code with our unit testing and other test environments we'd like to know if there's a dummy PCM device which provides the fundamental operations like reading/writing and setting the common hardware parameters. It doesn't even have to supply some useful functionality like loop back or similar. Neither the docs nor the goo led us to something useful. Thanks and regards Uli

2 4

[alsa-devel] [PATCH 0/2] ASoC: atmel-classd: add the Audio Class D Amplifier
by Songjun Wu 17 Sep '15

17 Sep '15

The Audio Class D Amplifier driver includes two parts. 1) Driver code to implement the Audio Class D Amplifier function. 2) Device tree binding document, it describes how to add the Audio Class D Amplifier in device tree. Songjun Wu (2): ASoC: atmel-classd: add the Audio Class D Amplifier code ASoC: atmel-classd: DT binding for Class D audio amplifier driver .../devicetree/bindings/sound/atmel-classd.txt | 73 ++ sound/soc/atmel/Kconfig | 9 + sound/soc/atmel/Makefile | 2 + sound/soc/atmel/atmel-classd.c | 801 ++++++++++++++++++++ sound/soc/atmel/atmel-classd.h | 120 +++ 5 files changed, 1005 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/atmel-classd.txt create mode 100644 sound/soc/atmel/atmel-classd.c create mode 100644 sound/soc/atmel/atmel-classd.h -- 1.7.9.5

3 21

[alsa-devel] [asoc:topic/simple 1/1] sound/soc/soc-core.c:3306:47: sparse: incorrect type in argument 1 (different base types)
by kbuild test robot 16 Sep '15

16 Sep '15

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/simple head: 6131084a0bc966107021d8c89489f9cd1663b902 commit: 6131084a0bc966107021d8c89489f9cd1663b902 [1/1] ASoC: simple-card: Add tdm slot mask support to simple-card reproduce: # apt-get install sparse git checkout 6131084a0bc966107021d8c89489f9cd1663b902 make ARCH=x86_64 allmodconfig make C=1 CF=-D__CHECK_ENDIAN__ sparse warnings: (new ones prefixed by >>) >> sound/soc/soc-core.c:3306:47: sparse: incorrect type in argument 1 (different base types) sound/soc/soc-core.c:3306:47: expected restricted __be32 const [usertype] *p sound/soc/soc-core.c:3306:47: got unsigned int const [usertype] * vim +3306 sound/soc/soc-core.c 3290 return 0; 3291 } 3292 EXPORT_SYMBOL_GPL(snd_soc_of_parse_audio_simple_widgets); 3293 3294 static int snd_soc_of_get_slot_mask(struct device_node *np, 3295 const char *prop_name, 3296 unsigned int *mask) 3297 { 3298 u32 val; 3299 const u32 *of_slot_mask = of_get_property(np, prop_name, &val); 3300 int i; 3301 3302 if (!of_slot_mask) 3303 return 0; 3304 val /= sizeof(u32); 3305 for (i = 0; i < val; i++) > 3306 if (be32_to_cpup(&of_slot_mask[i])) 3307 *mask |= (1 << i); 3308 3309 return val; 3310 } 3311 3312 int snd_soc_of_parse_tdm_slot(struct device_node *np, 3313 unsigned int *tx_mask, 3314 unsigned int *rx_mask, --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH] ASoC: SPEAr: Make SND_SPEAR_SOC select SND_SOC_GENERIC_DMAENGINE_PCM
by Axel Lin 16 Sep '15

16 Sep '15

devm_snd_dmaengine_pcm_register() is guarded by CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/spear/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/spear/Kconfig b/sound/soc/spear/Kconfig index 0a53053..4fb9141 100644 --- a/sound/soc/spear/Kconfig +++ b/sound/soc/spear/Kconfig @@ -1,6 +1,6 @@ config SND_SPEAR_SOC tristate - select SND_DMAENGINE_PCM + select SND_SOC_GENERIC_DMAENGINE_PCM config SND_SPEAR_SPDIF_OUT tristate -- 2.1.0

2 1

[alsa-devel] [PATCH RFC 0/5] ASoC: TDM stuff for simple-card, mcasp, and tlv320aic3x
by Jyri Sarha 16 Sep '15

16 Sep '15

The first patch is an independent fix. The symmetric_samplebits constraint could probably be removed too (with some changes to mcasp code too), but as tlv320aic3x family codecs won't support it, I could not test it. Jyri Sarha (5): ASoC: davinci-mcasp: Set .symmetric_rates = 1 in snd_soc_dai_driver ASoC: simple-card: Add tdm slot mask support to simple-card ASoC: davinci-mcasp: Add set_tdm_slots() support ASoC: davinci-mcasp: Get rid of bclk_lrclk_ratio in private data ASoC: tlv320aic3x: Improve tdm support .../devicetree/bindings/sound/tdm-slot.txt | 11 +- include/sound/simple_card.h | 2 + include/sound/soc.h | 2 + sound/soc/codecs/tlv320aic3x.c | 30 +- sound/soc/davinci/davinci-mcasp.c | 306 ++++++++++++++------- sound/soc/generic/simple-card.c | 8 +- sound/soc/soc-core.c | 25 ++ 7 files changed, 274 insertions(+), 110 deletions(-) -- 1.9.1

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[alsa-devel] [PATCH 1/2] ASoC: dapm: fix memory leak
by Sudip Mukherjee 16 Sep '15

16 Sep '15

Incase of an unknown event we were directly returning but we missed freeing params. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/soc/soc-dapm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index f4bf21a..ff8bda4 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3501,7 +3501,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, default: WARN(1, "Unknown event %d\n", event); - return -EINVAL; + ret = -EINVAL; } out: -- 1.9.1

2 2

[alsa-devel] Applied "ASoC: sunxi: add support for the on-chip codec on early Allwinner SoCs" to the asoc tree
by Mark Brown 16 Sep '15

16 Sep '15

The patch ASoC: sunxi: add support for the on-chip codec on early Allwinner SoCs has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 45fb6b6f2aa3f6b22b81078db0dba4b26c9d0bdb Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Emilio=20L=C3=B3pez?= <emilio(a)elopez.com.ar> Date: Sat, 12 Sep 2015 15:26:24 +0200 Subject: [PATCH] ASoC: sunxi: add support for the on-chip codec on early Allwinner SoCs MIME-Version: 1.0 Content-Type: text/plain; charset=UTF-8 Content-Transfer-Encoding: 8bit The sun4i, sun5i and sun7i SoC families have a built-in codec, capable of both audio capture and playback. While this is called a codec by Allwinner, it really is an in-SoC combination of a codec and a DAI, with its own DAC/ADC and amplifiers in a single memory-mapped controller. The capture part has been left out for now, and will be added eventually. Signed-off-by: Emilio López <emilio(a)elopez.com.ar> Signed-off-by: Maxime Ripard <maxime.ripard(a)free-electrons.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- 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 ++++++++++++++++++++++++++++++++++++++++++ 5 files changed, 735 insertions(+) create mode 100644 sound/soc/sunxi/Kconfig create mode 100644 sound/soc/sunxi/Makefile create mode 100644 sound/soc/sunxi/sun4i-codec.c diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 225bfda..7de792b 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -58,6 +58,7 @@ source "sound/soc/sh/Kconfig" source "sound/soc/sirf/Kconfig" source "sound/soc/spear/Kconfig" source "sound/soc/sti/Kconfig" +source "sound/soc/sunxi/Kconfig" source "sound/soc/tegra/Kconfig" source "sound/soc/txx9/Kconfig" source "sound/soc/ux500/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 134aca1..af0a571 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -40,6 +40,7 @@ obj-$(CONFIG_SND_SOC) += sh/ obj-$(CONFIG_SND_SOC) += sirf/ obj-$(CONFIG_SND_SOC) += spear/ obj-$(CONFIG_SND_SOC) += sti/ +obj-$(CONFIG_SND_SOC) += sunxi/ obj-$(CONFIG_SND_SOC) += tegra/ obj-$(CONFIG_SND_SOC) += txx9/ obj-$(CONFIG_SND_SOC) += ux500/ diff --git a/sound/soc/sunxi/Kconfig b/sound/soc/sunxi/Kconfig new file mode 100644 index 0000000..84c72ec --- /dev/null +++ b/sound/soc/sunxi/Kconfig @@ -0,0 +1,11 @@ +menu "Allwinner SoC Audio support" + +config SND_SUN4I_CODEC + tristate "Allwinner A10 Codec Support" + select SND_SOC_GENERIC_DMAENGINE_PCM + select REGMAP_MMIO + help + Select Y or M to add support for the Codec embedded in the Allwinner + A10 and affiliated SoCs. + +endmenu diff --git a/sound/soc/sunxi/Makefile b/sound/soc/sunxi/Makefile new file mode 100644 index 0000000..ea8a08c --- /dev/null +++ b/sound/soc/sunxi/Makefile @@ -0,0 +1,2 @@ +obj-$(CONFIG_SND_SUN4I_CODEC) += sun4i-codec.o + diff --git a/sound/soc/sunxi/sun4i-codec.c b/sound/soc/sunxi/sun4i-codec.c new file mode 100644 index 0000000..6e83e62 --- /dev/null +++ b/sound/soc/sunxi/sun4i-codec.c @@ -0,0 +1,720 @@ +/* + * Copyright 2014 Emilio López <emilio(a)elopez.com.ar> + * Copyright 2014 Jon Smirl <jonsmirl(a)gmail.com> + * Copyright 2015 Maxime Ripard <maxime.ripard(a)free-electrons.com> + * + * Based on the Allwinner SDK driver, released under the GPL. + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; either version 2 of the License, or + * (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + */ + +#include <linux/init.h> +#include <linux/kernel.h> +#include <linux/module.h> +#include <linux/platform_device.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <linux/of.h> +#include <linux/of_platform.h> +#include <linux/of_address.h> +#include <linux/clk.h> +#include <linux/regmap.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/tlv.h> +#include <sound/initval.h> +#include <sound/dmaengine_pcm.h> + +/* Codec DAC register offsets and bit fields */ +#define SUN4I_CODEC_DAC_DPC (0x00) +#define SUN4I_CODEC_DAC_DPC_EN_DA (31) +#define SUN4I_CODEC_DAC_DPC_DVOL (12) +#define SUN4I_CODEC_DAC_FIFOC (0x04) +#define SUN4I_CODEC_DAC_FIFOC_DAC_FS (29) +#define SUN4I_CODEC_DAC_FIFOC_FIR_VERSION (28) +#define SUN4I_CODEC_DAC_FIFOC_SEND_LASAT (26) +#define SUN4I_CODEC_DAC_FIFOC_TX_FIFO_MODE (24) +#define SUN4I_CODEC_DAC_FIFOC_DRQ_CLR_CNT (21) +#define SUN4I_CODEC_DAC_FIFOC_TX_TRIG_LEVEL (8) +#define SUN4I_CODEC_DAC_FIFOC_MONO_EN (6) +#define SUN4I_CODEC_DAC_FIFOC_TX_SAMPLE_BITS (5) +#define SUN4I_CODEC_DAC_FIFOC_DAC_DRQ_EN (4) +#define SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH (0) +#define SUN4I_CODEC_DAC_FIFOS (0x08) +#define SUN4I_CODEC_DAC_TXDATA (0x0c) +#define SUN4I_CODEC_DAC_ACTL (0x10) +#define SUN4I_CODEC_DAC_ACTL_DACAENR (31) +#define SUN4I_CODEC_DAC_ACTL_DACAENL (30) +#define SUN4I_CODEC_DAC_ACTL_MIXEN (29) +#define SUN4I_CODEC_DAC_ACTL_LDACLMIXS (15) +#define SUN4I_CODEC_DAC_ACTL_RDACRMIXS (14) +#define SUN4I_CODEC_DAC_ACTL_LDACRMIXS (13) +#define SUN4I_CODEC_DAC_ACTL_DACPAS (8) +#define SUN4I_CODEC_DAC_ACTL_MIXPAS (7) +#define SUN4I_CODEC_DAC_ACTL_PA_MUTE (6) +#define SUN4I_CODEC_DAC_ACTL_PA_VOL (0) +#define SUN4I_CODEC_DAC_TUNE (0x14) +#define SUN4I_CODEC_DAC_DEBUG (0x18) + +/* Codec ADC register offsets and bit fields */ +#define SUN4I_CODEC_ADC_FIFOC (0x1c) +#define SUN4I_CODEC_ADC_FIFOC_EN_AD (28) +#define SUN4I_CODEC_ADC_FIFOC_RX_FIFO_MODE (24) +#define SUN4I_CODEC_ADC_FIFOC_RX_TRIG_LEVEL (8) +#define SUN4I_CODEC_ADC_FIFOC_MONO_EN (7) +#define SUN4I_CODEC_ADC_FIFOC_RX_SAMPLE_BITS (6) +#define SUN4I_CODEC_ADC_FIFOC_ADC_DRQ_EN (4) +#define SUN4I_CODEC_ADC_FIFOC_FIFO_FLUSH (0) +#define SUN4I_CODEC_ADC_FIFOS (0x20) +#define SUN4I_CODEC_ADC_RXDATA (0x24) +#define SUN4I_CODEC_ADC_ACTL (0x28) +#define SUN4I_CODEC_ADC_ACTL_ADC_R_EN (31) +#define SUN4I_CODEC_ADC_ACTL_ADC_L_EN (30) +#define SUN4I_CODEC_ADC_ACTL_PREG1EN (29) +#define SUN4I_CODEC_ADC_ACTL_PREG2EN (28) +#define SUN4I_CODEC_ADC_ACTL_VMICEN (27) +#define SUN4I_CODEC_ADC_ACTL_VADCG (20) +#define SUN4I_CODEC_ADC_ACTL_ADCIS (17) +#define SUN4I_CODEC_ADC_ACTL_PA_EN (4) +#define SUN4I_CODEC_ADC_ACTL_DDE (3) +#define SUN4I_CODEC_ADC_DEBUG (0x2c) + +/* Other various ADC registers */ +#define SUN4I_CODEC_DAC_TXCNT (0x30) +#define SUN4I_CODEC_ADC_RXCNT (0x34) +#define SUN4I_CODEC_AC_SYS_VERI (0x38) +#define SUN4I_CODEC_AC_MIC_PHONE_CAL (0x3c) + +struct sun4i_codec { + struct device *dev; + struct regmap *regmap; + struct clk *clk_apb; + struct clk *clk_module; + + struct snd_dmaengine_dai_dma_data playback_dma_data; +}; + +static void sun4i_codec_start_playback(struct sun4i_codec *scodec) +{ + /* + * FIXME: according to the BSP, we might need to drive a PA + * GPIO high here on some boards + */ + + /* Flush TX FIFO */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH), + BIT(SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH)); + + /* Enable DAC DRQ */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_DAC_DRQ_EN), + BIT(SUN4I_CODEC_DAC_FIFOC_DAC_DRQ_EN)); +} + +static void sun4i_codec_stop_playback(struct sun4i_codec *scodec) +{ + /* + * FIXME: according to the BSP, we might need to drive a PA + * GPIO low here on some boards + */ + + /* Disable DAC DRQ */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_DAC_DRQ_EN), + 0); +} + +static int sun4i_codec_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card); + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return -ENOTSUPP; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + sun4i_codec_start_playback(scodec); + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + sun4i_codec_stop_playback(scodec); + break; + + default: + return -EINVAL; + } + + return 0; +} + +static int sun4i_codec_prepare(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card); + u32 val; + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return -ENOTSUPP; + + /* Flush the TX FIFO */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH), + BIT(SUN4I_CODEC_DAC_FIFOC_FIFO_FLUSH)); + + /* Set TX FIFO Empty Trigger Level */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + 0x3f << SUN4I_CODEC_DAC_FIFOC_TX_TRIG_LEVEL, + 0xf << SUN4I_CODEC_DAC_FIFOC_TX_TRIG_LEVEL); + + if (substream->runtime->rate > 32000) + /* Use 64 bits FIR filter */ + val = 0; + else + /* Use 32 bits FIR filter */ + val = BIT(SUN4I_CODEC_DAC_FIFOC_FIR_VERSION); + + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_FIR_VERSION), + val); + + /* Send zeros when we have an underrun */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_SEND_LASAT), + 0); + + return 0; +} + +static unsigned long sun4i_codec_get_mod_freq(struct snd_pcm_hw_params *params) +{ + unsigned int rate = params_rate(params); + + switch (rate) { + case 176400: + case 88200: + case 44100: + case 33075: + case 22050: + case 14700: + case 11025: + case 7350: + return 22579200; + + case 192000: + case 96000: + case 48000: + case 32000: + case 24000: + case 16000: + case 12000: + case 8000: + return 24576000; + + default: + return 0; + } +} + +static int sun4i_codec_get_hw_rate(struct snd_pcm_hw_params *params) +{ + unsigned int rate = params_rate(params); + + switch (rate) { + case 192000: + case 176400: + return 6; + + case 96000: + case 88200: + return 7; + + case 48000: + case 44100: + return 0; + + case 32000: + case 33075: + return 1; + + case 24000: + case 22050: + return 2; + + case 16000: + case 14700: + return 3; + + case 12000: + case 11025: + return 4; + + case 8000: + case 7350: + return 5; + + default: + return -EINVAL; + } +} + +static int sun4i_codec_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card); + unsigned long clk_freq; + int hwrate; + u32 val; + + if (substream->stream != SNDRV_PCM_STREAM_PLAYBACK) + return -ENOTSUPP; + + clk_freq = sun4i_codec_get_mod_freq(params); + if (!clk_freq) + return -EINVAL; + + if (clk_set_rate(scodec->clk_module, clk_freq)) + return -EINVAL; + + hwrate = sun4i_codec_get_hw_rate(params); + if (hwrate < 0) + return hwrate; + + /* Set DAC sample rate */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + 7 << SUN4I_CODEC_DAC_FIFOC_DAC_FS, + hwrate << SUN4I_CODEC_DAC_FIFOC_DAC_FS); + + /* Set the number of channels we want to use */ + if (params_channels(params) == 1) + val = BIT(SUN4I_CODEC_DAC_FIFOC_MONO_EN); + else + val = 0; + + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_MONO_EN), + val); + + /* Set the number of sample bits to either 16 or 24 bits */ + if (hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS)->min == 32) { + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_TX_SAMPLE_BITS), + BIT(SUN4I_CODEC_DAC_FIFOC_TX_SAMPLE_BITS)); + + /* Set TX FIFO mode to padding the LSBs with 0 */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_TX_FIFO_MODE), + 0); + + scodec->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES; + } else { + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_TX_SAMPLE_BITS), + 0); + + /* Set TX FIFO mode to repeat the MSB */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + BIT(SUN4I_CODEC_DAC_FIFOC_TX_FIFO_MODE), + BIT(SUN4I_CODEC_DAC_FIFOC_TX_FIFO_MODE)); + + scodec->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + } + + return 0; +} + +static int sun4i_codec_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card); + + /* + * Stop issuing DRQ when we have room for less than 16 samples + * in our TX FIFO + */ + regmap_update_bits(scodec->regmap, SUN4I_CODEC_DAC_FIFOC, + 3 << SUN4I_CODEC_DAC_FIFOC_DRQ_CLR_CNT, + 3 << SUN4I_CODEC_DAC_FIFOC_DRQ_CLR_CNT); + + return clk_prepare_enable(scodec->clk_module); +} + +static void sun4i_codec_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct sun4i_codec *scodec = snd_soc_card_get_drvdata(rtd->card); + + clk_disable_unprepare(scodec->clk_module); +} + +static const struct snd_soc_dai_ops sun4i_codec_dai_ops = { + .startup = sun4i_codec_startup, + .shutdown = sun4i_codec_shutdown, + .trigger = sun4i_codec_trigger, + .hw_params = sun4i_codec_hw_params, + .prepare = sun4i_codec_prepare, +}; + +static struct snd_soc_dai_driver sun4i_codec_dai = { + .name = "Codec", + .ops = &sun4i_codec_dai_ops, + .playback = { + .stream_name = "Codec Playback", + .channels_min = 1, + .channels_max = 2, + .rate_min = 8000, + .rate_max = 192000, + .rates = SNDRV_PCM_RATE_8000_48000 | + SNDRV_PCM_RATE_96000 | + SNDRV_PCM_RATE_192000 | + SNDRV_PCM_RATE_KNOT, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S32_LE, + .sig_bits = 24, + }, +}; + +/*** Codec ***/ +static const struct snd_kcontrol_new sun4i_codec_pa_mute = + SOC_DAPM_SINGLE("Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_PA_MUTE, 1, 0); + +static DECLARE_TLV_DB_SCALE(sun4i_codec_pa_volume_scale, -6300, 100, 1); + +static const struct snd_kcontrol_new sun4i_codec_widgets[] = { + SOC_SINGLE_TLV("PA Volume", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_PA_VOL, 0x3F, 0, + sun4i_codec_pa_volume_scale), +}; + +static const struct snd_kcontrol_new sun4i_codec_left_mixer_controls[] = { + SOC_DAPM_SINGLE("Left DAC Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_LDACLMIXS, 1, 0), +}; + +static const struct snd_kcontrol_new sun4i_codec_right_mixer_controls[] = { + SOC_DAPM_SINGLE("Right DAC Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_RDACRMIXS, 1, 0), + SOC_DAPM_SINGLE("Left DAC Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_LDACRMIXS, 1, 0), +}; + +static const struct snd_kcontrol_new sun4i_codec_pa_mixer_controls[] = { + SOC_DAPM_SINGLE("DAC Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_DACPAS, 1, 0), + SOC_DAPM_SINGLE("Mixer Playback Switch", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_MIXPAS, 1, 0), +}; + +static const struct snd_soc_dapm_widget sun4i_codec_dapm_widgets[] = { + /* Digital parts of the DACs */ + SND_SOC_DAPM_SUPPLY("DAC", SUN4I_CODEC_DAC_DPC, + SUN4I_CODEC_DAC_DPC_EN_DA, 0, + NULL, 0), + + /* Analog parts of the DACs */ + SND_SOC_DAPM_DAC("Left DAC", "Codec Playback", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_DACAENL, 0), + SND_SOC_DAPM_DAC("Right DAC", "Codec Playback", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_DACAENR, 0), + + /* Mixers */ + SND_SOC_DAPM_MIXER("Left Mixer", SND_SOC_NOPM, 0, 0, + sun4i_codec_left_mixer_controls, + ARRAY_SIZE(sun4i_codec_left_mixer_controls)), + SND_SOC_DAPM_MIXER("Right Mixer", SND_SOC_NOPM, 0, 0, + sun4i_codec_right_mixer_controls, + ARRAY_SIZE(sun4i_codec_right_mixer_controls)), + + /* Global Mixer Enable */ + SND_SOC_DAPM_SUPPLY("Mixer Enable", SUN4I_CODEC_DAC_ACTL, + SUN4I_CODEC_DAC_ACTL_MIXEN, 0, NULL, 0), + + /* Pre-Amplifier */ + SND_SOC_DAPM_MIXER("Pre-Amplifier", SUN4I_CODEC_ADC_ACTL, + SUN4I_CODEC_ADC_ACTL_PA_EN, 0, + sun4i_codec_pa_mixer_controls, + ARRAY_SIZE(sun4i_codec_pa_mixer_controls)), + SND_SOC_DAPM_SWITCH("Pre-Amplifier Mute", SND_SOC_NOPM, 0, 0, + &sun4i_codec_pa_mute), + + SND_SOC_DAPM_OUTPUT("HP Right"), + SND_SOC_DAPM_OUTPUT("HP Left"), +}; + +static const struct snd_soc_dapm_route sun4i_codec_dapm_routes[] = { + /* Left DAC Routes */ + { "Left DAC", NULL, "DAC" }, + + /* Right DAC Routes */ + { "Right DAC", NULL, "DAC" }, + + /* Right Mixer Routes */ + { "Right Mixer", NULL, "Mixer Enable" }, + { "Right Mixer", "Left DAC Playback Switch", "Left DAC" }, + { "Right Mixer", "Right DAC Playback Switch", "Right DAC" }, + + /* Left Mixer Routes */ + { "Left Mixer", NULL, "Mixer Enable" }, + { "Left Mixer", "Left DAC Playback Switch", "Left DAC" }, + + /* Pre-Amplifier Mixer Routes */ + { "Pre-Amplifier", "Mixer Playback Switch", "Left Mixer" }, + { "Pre-Amplifier", "Mixer Playback Switch", "Right Mixer" }, + { "Pre-Amplifier", "DAC Playback Switch", "Left DAC" }, + { "Pre-Amplifier", "DAC Playback Switch", "Right DAC" }, + + /* PA -> HP path */ + { "Pre-Amplifier Mute", "Switch", "Pre-Amplifier" }, + { "HP Right", NULL, "Pre-Amplifier Mute" }, + { "HP Left", NULL, "Pre-Amplifier Mute" }, +}; + +static struct snd_soc_codec_driver sun4i_codec_codec = { + .controls = sun4i_codec_widgets, + .num_controls = ARRAY_SIZE(sun4i_codec_widgets), + .dapm_widgets = sun4i_codec_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(sun4i_codec_dapm_widgets), + .dapm_routes = sun4i_codec_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(sun4i_codec_dapm_routes), +}; + +static const struct snd_soc_component_driver sun4i_codec_component = { + .name = "sun4i-codec", +}; + +#define SUN4I_CODEC_RATES SNDRV_PCM_RATE_8000_192000 +#define SUN4I_CODEC_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ + SNDRV_PCM_FMTBIT_S32_LE) + +static int sun4i_codec_dai_probe(struct snd_soc_dai *dai) +{ + struct snd_soc_card *card = snd_soc_dai_get_drvdata(dai); + struct sun4i_codec *scodec = snd_soc_card_get_drvdata(card); + + snd_soc_dai_init_dma_data(dai, &scodec->playback_dma_data, + NULL); + + return 0; +} + +static struct snd_soc_dai_driver dummy_cpu_dai = { + .name = "sun4i-codec-cpu-dai", + .probe = sun4i_codec_dai_probe, + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = SUN4I_CODEC_RATES, + .formats = SUN4I_CODEC_FORMATS, + .sig_bits = 24, + }, +}; + +static const struct regmap_config sun4i_codec_regmap_config = { + .reg_bits = 32, + .reg_stride = 4, + .val_bits = 32, + .max_register = SUN4I_CODEC_AC_MIC_PHONE_CAL, +}; + +static const struct of_device_id sun4i_codec_of_match[] = { + { .compatible = "allwinner,sun4i-a10-codec" }, + { .compatible = "allwinner,sun7i-a20-codec" }, + {} +}; +MODULE_DEVICE_TABLE(of, sun4i_codec_of_match); + +static struct snd_soc_dai_link *sun4i_codec_create_link(struct device *dev, + int *num_links) +{ + struct snd_soc_dai_link *link = devm_kzalloc(dev, sizeof(*link), + GFP_KERNEL); + if (!link) + return NULL; + + link->name = "cdc"; + link->stream_name = "CDC PCM"; + link->codec_dai_name = "Codec"; + link->cpu_dai_name = dev_name(dev); + link->codec_name = dev_name(dev); + link->platform_name = dev_name(dev); + link->dai_fmt = SND_SOC_DAIFMT_I2S; + + *num_links = 1; + + return link; +}; + +static struct snd_soc_card *sun4i_codec_create_card(struct device *dev) +{ + struct snd_soc_card *card; + int ret; + + card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL); + if (!card) + return NULL; + + card->dai_link = sun4i_codec_create_link(dev, &card->num_links); + if (!card->dai_link) + return NULL; + + card->dev = dev; + card->name = "sun4i-codec"; + + ret = snd_soc_of_parse_audio_routing(card, "routing"); + if (ret) { + dev_err(dev, "Failed to create our audio routing\n"); + return NULL; + } + + return card; +}; + +static int sun4i_codec_probe(struct platform_device *pdev) +{ + struct snd_soc_card *card; + struct sun4i_codec *scodec; + struct resource *res; + void __iomem *base; + int ret; + + scodec = devm_kzalloc(&pdev->dev, sizeof(*scodec), GFP_KERNEL); + if (!scodec) + return -ENOMEM; + + scodec->dev = &pdev->dev; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + base = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(base)) { + dev_err(&pdev->dev, "Failed to map the registers\n"); + return PTR_ERR(base); + } + + scodec->regmap = devm_regmap_init_mmio(&pdev->dev, base, + &sun4i_codec_regmap_config); + if (IS_ERR(scodec->regmap)) { + dev_err(&pdev->dev, "Failed to create our regmap\n"); + return PTR_ERR(scodec->regmap); + } + + /* Get the clocks from the DT */ + scodec->clk_apb = devm_clk_get(&pdev->dev, "apb"); + if (IS_ERR(scodec->clk_apb)) { + dev_err(&pdev->dev, "Failed to get the APB clock\n"); + return PTR_ERR(scodec->clk_apb); + } + + scodec->clk_module = devm_clk_get(&pdev->dev, "codec"); + if (IS_ERR(scodec->clk_module)) { + dev_err(&pdev->dev, "Failed to get the module clock\n"); + return PTR_ERR(scodec->clk_module); + } + + /* Enable the bus clock */ + if (clk_prepare_enable(scodec->clk_apb)) { + dev_err(&pdev->dev, "Failed to enable the APB clock\n"); + return -EINVAL; + } + + /* DMA configuration for TX FIFO */ + scodec->playback_dma_data.addr = res->start + SUN4I_CODEC_DAC_TXDATA; + scodec->playback_dma_data.maxburst = 4; + scodec->playback_dma_data.addr_width = DMA_SLAVE_BUSWIDTH_2_BYTES; + + ret = snd_soc_register_codec(&pdev->dev, &sun4i_codec_codec, + &sun4i_codec_dai, 1); + if (ret) { + dev_err(&pdev->dev, "Failed to register our codec\n"); + goto err_clk_disable; + } + + ret = devm_snd_soc_register_component(&pdev->dev, + &sun4i_codec_component, + &dummy_cpu_dai, 1); + if (ret) { + dev_err(&pdev->dev, "Failed to register our DAI\n"); + goto err_unregister_codec; + } + + ret = devm_snd_dmaengine_pcm_register(&pdev->dev, NULL, 0); + if (ret) { + dev_err(&pdev->dev, "Failed to register against DMAEngine\n"); + goto err_unregister_codec; + } + + card = sun4i_codec_create_card(&pdev->dev); + if (!card) { + dev_err(&pdev->dev, "Failed to create our card\n"); + goto err_unregister_codec; + } + + platform_set_drvdata(pdev, card); + snd_soc_card_set_drvdata(card, scodec); + + ret = snd_soc_register_card(card); + if (ret) { + dev_err(&pdev->dev, "Failed to register our card\n"); + goto err_unregister_codec; + } + + return 0; + +err_unregister_codec: + snd_soc_unregister_codec(&pdev->dev); +err_clk_disable: + clk_disable_unprepare(scodec->clk_apb); + return ret; +} + +static int sun4i_codec_remove(struct platform_device *pdev) +{ + struct snd_soc_card *card = platform_get_drvdata(pdev); + struct sun4i_codec *scodec = snd_soc_card_get_drvdata(card); + + snd_soc_unregister_card(card); + snd_soc_unregister_codec(&pdev->dev); + clk_disable_unprepare(scodec->clk_apb); + + return 0; +} + +static struct platform_driver sun4i_codec_driver = { + .driver = { + .name = "sun4i-codec", + .owner = THIS_MODULE, + .of_match_table = sun4i_codec_of_match, + }, + .probe = sun4i_codec_probe, + .remove = sun4i_codec_remove, +}; +module_platform_driver(sun4i_codec_driver); + +MODULE_DESCRIPTION("Allwinner A10 codec driver"); +MODULE_AUTHOR("Emilio López <emilio(a)elopez.com.ar>"); +MODULE_AUTHOR("Jon Smirl <jonsmirl(a)gmail.com>"); +MODULE_AUTHOR("Maxime Ripard <maxime.ripard(a)free-electrons.com>"); +MODULE_LICENSE("GPL"); -- 2.5.0

1 0

[alsa-devel] [PATCH] ASoC: rsnd: SND_SOC_RCAR depends on COMMON_CLK
by Kuninori Morimoto 16 Sep '15

16 Sep '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> 2a46db4a3("ASoC: rsnd: add AUDIO_CLKOUT support") uses of_clk_add_provider() which is requesting struct clk_onecell_data. But it is COMMON_CLK feature. SND_SOC_RCAR depends on COMMON_CLK This patch also solved compile error of 7486d80f7("ASoC: rsnd: remove unneeded sh_clk header") Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/Kconfig | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/sh/Kconfig b/sound/soc/sh/Kconfig index 07114b0..6ca90aa 100644 --- a/sound/soc/sh/Kconfig +++ b/sound/soc/sh/Kconfig @@ -37,6 +37,7 @@ config SND_SOC_SH4_SIU config SND_SOC_RCAR tristate "R-Car series SRU/SCU/SSIU/SSI support" depends on DMA_OF + depends on COMMON_CLK select SND_SIMPLE_CARD select REGMAP_MMIO help -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: ak4642: exchange macro name to avoid redefinition
by Kuninori Morimoto 16 Sep '15

16 Sep '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> 71a0138ab("ASoC: ak4642: enable to use MCKO as fixed rate output pin on DT") added new FS() macro, but x86 already has it in arch/x86/include/uapi/asm/ptrace-abi.h This patch exchange FS() to FSs() to avoid redefinition warning Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/ak4642.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/ak4642.c b/sound/soc/codecs/ak4642.c index b5c4981..cda27c2 100644 --- a/sound/soc/codecs/ak4642.c +++ b/sound/soc/codecs/ak4642.c @@ -130,8 +130,8 @@ #define I2S (3 << 0) /* MD_CTL2 */ -#define FS(val) (((val & 0x7) << 0) | ((val & 0x8) << 2)) -#define PS(val) ((val & 0x3) << 6) +#define FSs(val) (((val & 0x7) << 0) | ((val & 0x8) << 2)) +#define PSs(val) ((val & 0x3) << 6) /* MD_CTL3 */ #define BST1 (1 << 3) @@ -458,7 +458,8 @@ static int ak4642_set_mcko(struct snd_soc_codec *codec, for (ps = 0; ps < ARRAY_SIZE(ps_list); ps++) { for (fs = 0; fs < ARRAY_SIZE(fs_list); fs++) { if (frequency == ps_list[ps] * fs_list[fs]) { - snd_soc_write(codec, MD_CTL2, PS(ps) | FS(fs)); + snd_soc_write(codec, MD_CTL2, + PSs(ps) | FSs(fs)); return 0; } } -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: rsnd: fixup clkout_name[] index error for single AUDIO_CLKOUT SoC
by Kuninori Morimoto 16 Sep '15

16 Sep '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> 2a46db4a3("ASoC: rsnd: add AUDIO_CLKOUT support") added AUDIO_CLKOUT support for ADG. But single AUDIO_CLKOUT needs clkout_name[CLKOUT], not clkout_name[i]. Kernel will have NULL pointer access without this patch. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/adg.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sh/rcar/adg.c b/sound/soc/sh/rcar/adg.c index 3fecb87..c4ebbb7 100644 --- a/sound/soc/sh/rcar/adg.c +++ b/sound/soc/sh/rcar/adg.c @@ -532,7 +532,7 @@ static void rsnd_adg_get_clkout(struct rsnd_priv *priv, * for clkout */ if (!count) { - clk = clk_register_fixed_rate(dev, clkout_name[i], + clk = clk_register_fixed_rate(dev, clkout_name[CLKOUT], parent_clk_name, (parent_clk_name) ? 0 : CLK_IS_ROOT, req_rate); -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: fix broken pxa SoC support
by Robert Jarzmik 16 Sep '15

16 Sep '15

The previous fix of pxa library support, which was introduced to fix the library dependency, broke the previous SoC behavior, where a machine code binding pxa2xx-ac97 with a coded relied on : - sound/soc/pxa/pxa2xx-ac97.c - sound/soc/codecs/XXX.c For example, the mioa701_wm9713.c machine code is currently broken. The "select ARM" statement wrongly selects the soc/arm/pxa2xx-ac97 for compilation, as per an unfortunate fate SND_PXA2XX_AC97 is both declared in sound/arm/Kconfig and sound/soc/pxa/Kconfig. Fix this by ensuring that SND_PXA2XX_SOC correctly triggers the correct pxa2xx-ac97 compilation. Fixes: 846172dfe33c ("ASoC: fix SND_PXA2XX_LIB Kconfig warning") Signed-off-by: Robert Jarzmik <robert.jarzmik(a)free.fr> --- sound/arm/Kconfig | 15 ++++++++------- sound/soc/pxa/Kconfig | 2 -- 2 files changed, 8 insertions(+), 9 deletions(-) diff --git a/sound/arm/Kconfig b/sound/arm/Kconfig index 885683a3b0bd..e0406211716b 100644 --- a/sound/arm/Kconfig +++ b/sound/arm/Kconfig @@ -9,6 +9,14 @@ menuconfig SND_ARM Drivers that are implemented on ASoC can be found in "ALSA for SoC audio support" section. +config SND_PXA2XX_LIB + tristate + select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97 + select SND_DMAENGINE_PCM + +config SND_PXA2XX_LIB_AC97 + bool + if SND_ARM config SND_ARMAACI @@ -21,13 +29,6 @@ config SND_PXA2XX_PCM tristate select SND_PCM -config SND_PXA2XX_LIB - tristate - select SND_AC97_CODEC if SND_PXA2XX_LIB_AC97 - -config SND_PXA2XX_LIB_AC97 - bool - config SND_PXA2XX_AC97 tristate "AC97 driver for the Intel PXA2xx chip" depends on ARCH_PXA diff --git a/sound/soc/pxa/Kconfig b/sound/soc/pxa/Kconfig index 39cea80846c3..f2bf8661dd21 100644 --- a/sound/soc/pxa/Kconfig +++ b/sound/soc/pxa/Kconfig @@ -1,7 +1,6 @@ config SND_PXA2XX_SOC tristate "SoC Audio for the Intel PXA2xx chip" depends on ARCH_PXA - select SND_ARM select SND_PXA2XX_LIB help Say Y or M if you want to add support for codecs attached to @@ -25,7 +24,6 @@ config SND_PXA2XX_AC97 config SND_PXA2XX_SOC_AC97 tristate select AC97_BUS - select SND_ARM select SND_PXA2XX_LIB_AC97 select SND_SOC_AC97_BUS -- 2.1.4

2 1

[alsa-devel] CM6206 S/PDIF trouble
by Linus Walleij 15 Sep '15

15 Sep '15

Hi ALSA folks, feel like an amateur around here. Anyways, I got this USB sound card named "TRIXES External Sound Card USB 6 Channel 5.1 Audio" Looks like so: http://www.amazon.co.uk/External-Sound-Card-Channel-Audio/dp/B003TO3KHY/ref… Here is the alsa-info.sh output (yeah on a machine with hardwired HDA in it too, sorry): http://www.alsa-project.org/db/?f=f52ce9edc11918d70bc0f230f3cbd5f984bea5b5 The lsusb -v output is in the end of the mail, too. This is a 0d8c:0102 "C-Media Electronics, Inc. CM106 Like Sound Device" Actually, and according to the drivers on the CD that comes with it, this is a CM6206-based card. It seems like a nice and cheap 5.1 card actually. But I wanted to use the digital S/PDIF (IEC958) input/output and that does not work at all. It appears very nicely in the pulseaudio UI thing in GNOME, one output/input for analog line and S/PDIF respectively. First problem: running alsamixer -c <card> the "PCM Capture Source" appear in the Playback controls. This seems like a bug, selecting "IEC958 In" with the arrow keys work fine. "Line" and "Mic" work fine too. In the Capture controls, S/PDIF appears as mutable but without any input level setting, I guess this is normal for digital in. Anyways: no samples go out on the digital out and no samples get taken from the digital in. The endpoints are just dead. With an S/PDIF device listinging on the out line, it says "no signal" and I have tested the same equipment with the S/PDIF out on my HDA audio card and that works just fine. Since the CM6206 has that quirky quirk in usb/quirks.c that set up it's internal registers to work as expected there is a CM6206 datasheet on the web these days and it says interesting things like it needs bit 15 set in the first control register to use S/PDIF as master and also has to set the output frequency in the next three bits (etc etc) so unless someone is onto this already I'll just start hacking at it. Since Eric was patching this relating to S/PDIF I was under the impression it was working for him though? Anyways, Yours, Linus Walleij Here is lsusb -v: Bus 003 Device 004: ID 0d8c:0102 C-Media Electronics, Inc. CM106 Like Sound Device Device Descriptor: bLength 18 bDescriptorType 1 bcdUSB 1.10 bDeviceClass 0 (Defined at Interface level) bDeviceSubClass 0 bDeviceProtocol 0 bMaxPacketSize0 8 idVendor 0x0d8c C-Media Electronics, Inc. idProduct 0x0102 CM106 Like Sound Device bcdDevice 0.10 iManufacturer 0 iProduct 2 USB Sound Device iSerial 0 bNumConfigurations 1 Configuration Descriptor: bLength 9 bDescriptorType 2 wTotalLength 534 bNumInterfaces 4 bConfigurationValue 1 iConfiguration 0 bmAttributes 0x80 (Bus Powered) MaxPower 500mA Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 0 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 1 Control Device bInterfaceProtocol 0 iInterface 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 1 (HEADER) bcdADC 1.00 wTotalLength 200 bInCollection 2 baInterfaceNr( 0) 1 baInterfaceNr( 1) 2 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 1 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bNrChannels 8 wChannelConfig 0x063f Left Front (L) Right Front (R) Center Front (C) Low Freqency Enhancement (LFE) Left Surround (LS) Right Surround (RS) Side Left (SL) Side Right (SR) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 4 wTerminalType 0x0201 Microphone bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 8 bSourceID 4 bControlSize 1 bmaControls( 0) 0x01 Mute Control bmaControls( 1) 0x02 Volume Control bmaControls( 2) 0x02 Volume Control iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 9 bSourceID 4 bControlSize 1 bmaControls( 0) 0x01 Mute Control bmaControls( 1) 0x02 Volume Control bmaControls( 2) 0x02 Volume Control iFeature 0 AudioControl Interface Descriptor: bLength 25 bDescriptorType 36 bDescriptorSubtype 4 (MIXER_UNIT) bUnitID 12 bNrInPins 3 baSourceID( 0) 1 baSourceID( 1) 9 baSourceID( 2) 11 bNrChannels 8 wChannelConfig 0x063f Left Front (L) Right Front (R) Center Front (C) Low Freqency Enhancement (LFE) Left Surround (LS) Right Surround (RS) Side Left (SL) Side Right (SR) iChannelNames 0 bmControls 0x00 bmControls 0x00 bmControls 0x00 iMixer 0 junk at descriptor end: 00 00 00 00 00 00 00 00 00 AudioControl Interface Descriptor: bLength 16 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 13 bSourceID 12 bControlSize 1 bmaControls( 0) 0x01 Mute Control bmaControls( 1) 0x02 Volume Control bmaControls( 2) 0x02 Volume Control bmaControls( 3) 0x02 Volume Control bmaControls( 4) 0x02 Volume Control bmaControls( 5) 0x02 Volume Control bmaControls( 6) 0x02 Volume Control bmaControls( 7) 0x02 Volume Control bmaControls( 8) 0x02 Volume Control iFeature 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 3 wTerminalType 0x0301 Speaker bAssocTerminal 0 bSourceID 13 iTerminal 0 AudioControl Interface Descriptor: bLength 13 bDescriptorType 36 bDescriptorSubtype 4 (MIXER_UNIT) bUnitID 17 bNrInPins 2 baSourceID( 0) 9 baSourceID( 1) 11 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 bmControls 0x00 iMixer 0 AudioControl Interface Descriptor: bLength 9 bDescriptorType 36 bDescriptorSubtype 3 (OUTPUT_TERMINAL) bTerminalID 10 wTerminalType 0x0101 USB Streaming bAssocTerminal 0 bSourceID 7 iTerminal 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 5 (SELECTOR_UNIT) bUnitID 7 bNrInPins 4 baSource( 0) 8 baSource( 1) 15 baSource( 2) 16 baSource( 3) 2 iSelector 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 6 wTerminalType 0x0603 Line Connector bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 15 bSourceID 6 bControlSize 1 bmaControls( 0) 0x01 Mute Control bmaControls( 1) 0x02 Volume Control bmaControls( 2) 0x02 Volume Control iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 11 bSourceID 6 bControlSize 1 bmaControls( 0) 0x01 Mute Control bmaControls( 1) 0x02 Volume Control bmaControls( 2) 0x02 Volume Control iFeature 0 AudioControl Interface Descriptor: bLength 12 bDescriptorType 36 bDescriptorSubtype 2 (INPUT_TERMINAL) bTerminalID 5 wTerminalType 0x0605 SPDIF interface bAssocTerminal 0 bNrChannels 2 wChannelConfig 0x0003 Left Front (L) Right Front (R) iChannelNames 0 iTerminal 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 2 bSourceID 17 bControlSize 1 bmaControls( 0) 0x01 Mute Control bmaControls( 1) 0x02 Volume Control bmaControls( 2) 0x02 Volume Control iFeature 0 AudioControl Interface Descriptor: bLength 10 bDescriptorType 36 bDescriptorSubtype 6 (FEATURE_UNIT) bUnitID 16 bSourceID 5 bControlSize 1 bmaControls( 0) 0x01 Mute Control bmaControls( 1) 0x00 bmaControls( 2) 0x00 iFeature 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 8 bSubframeSize 2 bBitResolution 16 bSamFreqType 2 Discrete tSamFreq[ 0] 44100 tSamFreq[ 1] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x06 EP 6 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0310 1x 784 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 2 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 2 Discrete tSamFreq[ 0] 44100 tSamFreq[ 1] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x06 EP 6 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x00c4 1x 196 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 3 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 4 bSubframeSize 2 bBitResolution 16 bSamFreqType 2 Discrete tSamFreq[ 0] 44100 tSamFreq[ 1] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x06 EP 6 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0188 1x 392 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 4 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 6 bSubframeSize 2 bBitResolution 16 bSamFreqType 2 Discrete tSamFreq[ 0] 44100 tSamFreq[ 1] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x06 EP 6 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0248 1x 584 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 1 bAlternateSetting 5 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 1 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 11 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 1 Discrete tSamFreq[ 0] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x06 EP 6 OUT bmAttributes 9 Transfer Type Isochronous Synch Type Adaptive Usage Type Data wMaxPacketSize 0x0188 1x 392 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 0 bNumEndpoints 0 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 2 bAlternateSetting 1 bNumEndpoints 1 bInterfaceClass 1 Audio bInterfaceSubClass 2 Streaming bInterfaceProtocol 0 iInterface 0 AudioStreaming Interface Descriptor: bLength 7 bDescriptorType 36 bDescriptorSubtype 1 (AS_GENERAL) bTerminalLink 10 bDelay 1 frames wFormatTag 1 PCM AudioStreaming Interface Descriptor: bLength 14 bDescriptorType 36 bDescriptorSubtype 2 (FORMAT_TYPE) bFormatType 1 (FORMAT_TYPE_I) bNrChannels 2 bSubframeSize 2 bBitResolution 16 bSamFreqType 2 Discrete tSamFreq[ 0] 44100 tSamFreq[ 1] 48000 Endpoint Descriptor: bLength 9 bDescriptorType 5 bEndpointAddress 0x85 EP 5 IN bmAttributes 5 Transfer Type Isochronous Synch Type Asynchronous Usage Type Data wMaxPacketSize 0x00c8 1x 200 bytes bInterval 1 bRefresh 0 bSynchAddress 0 AudioControl Endpoint Descriptor: bLength 7 bDescriptorType 37 bDescriptorSubtype 1 (EP_GENERAL) bmAttributes 0x01 Sampling Frequency bLockDelayUnits 0 Undefined wLockDelay 0 Undefined Interface Descriptor: bLength 9 bDescriptorType 4 bInterfaceNumber 3 bAlternateSetting 0 bNumEndpoints 1 bInterfaceClass 3 Human Interface Device bInterfaceSubClass 0 No Subclass bInterfaceProtocol 0 None iInterface 0 HID Device Descriptor: bLength 9 bDescriptorType 33 bcdHID 1.00 bCountryCode 0 Not supported bNumDescriptors 1 bDescriptorType 34 Report wDescriptorLength 50 Report Descriptors: ** UNAVAILABLE ** Endpoint Descriptor: bLength 7 bDescriptorType 5 bEndpointAddress 0x81 EP 1 IN bmAttributes 3 Transfer Type Interrupt Synch Type None Usage Type Data wMaxPacketSize 0x0003 1x 3 bytes bInterval 1 Device Status: 0x0000 (Bus Powered)

2 1

[alsa-devel] [PATCH v2 1/2] ASoC: soc-compress: add a config item for soc-compress
by Jie Yang 15 Sep '15

15 Sep '15

We don't always need soc-compress in soc, here add a config item SND_SOC_COMPRESS, please add 'select SND_SOC_COMPRESS' to driver Kconfig when it is needed. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- include/sound/soc.h | 7 +++++++ sound/soc/Kconfig | 5 ++++- sound/soc/Makefile | 3 ++- sound/soc/intel/Kconfig | 1 + 4 files changed, 14 insertions(+), 2 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 93df8bf..cc1cd4f 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -440,7 +440,14 @@ int snd_soc_platform_read(struct snd_soc_platform *platform, int snd_soc_platform_write(struct snd_soc_platform *platform, unsigned int reg, unsigned int val); int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num); +#if IS_ENABLED(CONFIG_SND_SOC_COMPRESS) int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num); +#else +static inline int soc_new_compress(struct snd_soc_pcm_runtime *rtd, int num) +{ + return -EPERM; +} +#endif struct snd_pcm_substream *snd_soc_get_dai_substream(struct snd_soc_card *card, const char *dai_link, int stream); diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index e2828e1..a124759 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -9,7 +9,6 @@ menuconfig SND_SOC select SND_JACK if INPUT=y || INPUT=SND select REGMAP_I2C if I2C select REGMAP_SPI if SPI_MASTER - select SND_COMPRESS_OFFLOAD ---help--- If you want ASoC support, you should say Y here and also to the @@ -30,6 +29,10 @@ config SND_SOC_GENERIC_DMAENGINE_PCM bool select SND_DMAENGINE_PCM +config SND_SOC_COMPRESS + tristate + select SND_COMPRESS_OFFLOAD + # All the supported SoCs source "sound/soc/adi/Kconfig" source "sound/soc/atmel/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index a0e1ee6..184c1e6 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -1,6 +1,7 @@ snd-soc-core-objs := soc-core.o soc-dapm.o soc-jack.o soc-cache.o soc-utils.o -snd-soc-core-objs += soc-pcm.o soc-compress.o soc-io.o soc-devres.o soc-ops.o +snd-soc-core-objs += soc-pcm.o soc-io.o soc-devres.o soc-ops.o snd-soc-core-objs += soc-topology.o +snd-soc-core-$(CONFIG_SND_SOC_COMPRESS) += soc-compress.o ifneq ($(CONFIG_SND_SOC_GENERIC_DMAENGINE_PCM),) snd-soc-core-objs += soc-generic-dmaengine-pcm.o diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 791953f..e559174 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -12,6 +12,7 @@ config SND_MFLD_MACHINE config SND_SST_MFLD_PLATFORM tristate + select SND_SOC_COMPRESS config SND_SST_IPC tristate -- 1.9.1

6 23

[alsa-devel] Occasional channel swapping, rt5631 vs mmp2 sspa
by James Cameron 15 Sep '15

15 Sep '15

G'day, The rt5631 has a single serial data port shared by playback and capture at the same time, using I2S. Sometimes on playback or capture the channels are swapped left for right. The swap starts as a stream starts. The swap continues until the stream stops. The SoC is an Armada 610 with an SSPA peripheral. A workaround is to remove a clock domain reset when a stream is started. This is a reset bit in a register of the SSPA. My theory is that the clock domain reset in the SSPA while I2S is clocking interferes with LRCK (aka fsync) edge detection. I'm wondering how this kind of problem should be solved properly; avoiding the clock domain reset while I2S is clocking seems right, but which codec and SoC driver ops should start I2S and/or reset the host port? My post is probably incomplete and inadequate; please ask any questions. ;-) -- James Cameron http://quozl.linux.org.au/

1 0

[alsa-devel] [PATCH] ASoC: add ak4613 support
by Kuninori Morimoto 15 Sep '15

15 Sep '15

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- Documentation/devicetree/bindings/sound/ak4613.txt | 17 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4613.c | 493 +++++++++++++++++++++ 4 files changed, 516 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/ak4613.txt create mode 100644 sound/soc/codecs/ak4613.c diff --git a/Documentation/devicetree/bindings/sound/ak4613.txt b/Documentation/devicetree/bindings/sound/ak4613.txt new file mode 100644 index 0000000..15a9195 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/ak4613.txt @@ -0,0 +1,17 @@ +AK4613 I2C transmitter + +This device supports I2C mode only. + +Required properties: + +- compatible : "asahi-kasei,ak4613" +- reg : The chip select number on the I2C bus + +Example: + +&i2c { + ak4613: ak4613@0x10 { + compatible = "asahi-kasei,ak4613"; + reg = <0x10>; + }; +}; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0c9733e..06bbe75 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -319,6 +319,10 @@ config SND_SOC_AK4535 config SND_SOC_AK4554 tristate "AKM AK4554 CODEC" +config SND_SOC_AK4613 + tristate "AKM AK4613 CODEC" + depends on I2C + config SND_SOC_AK4641 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 4a32077..5b6c8af 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -26,6 +26,7 @@ snd-soc-ads117x-objs := ads117x.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o snd-soc-ak4554-objs := ak4554.o +snd-soc-ak4613-objs := ak4613.o snd-soc-ak4641-objs := ak4641.o snd-soc-ak4642-objs := ak4642.o snd-soc-ak4671-objs := ak4671.o @@ -216,6 +217,7 @@ obj-$(CONFIG_SND_SOC_ADS117X) += snd-soc-ads117x.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4554) += snd-soc-ak4554.o +obj-$(CONFIG_SND_SOC_AK4613) += snd-soc-ak4613.o obj-$(CONFIG_SND_SOC_AK4641) += snd-soc-ak4641.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o diff --git a/sound/soc/codecs/ak4613.c b/sound/soc/codecs/ak4613.c new file mode 100644 index 0000000..19e1741 --- /dev/null +++ b/sound/soc/codecs/ak4613.c @@ -0,0 +1,493 @@ +/* + * ak4613.c -- Asahi Kasei ALSA Soc Audio driver + * + * Copyright (C) 2015 Renesas Electronics Corporation + * Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> + * + * Based on ak4642.c by Kuninori Morimoto + * Based on wm8731.c by Richard Purdie + * Based on ak4535.c by Richard Purdie + * Based on wm8753.c by Liam Girdwood + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/clk.h> +#include <linux/i2c.h> +#include <linux/slab.h> +#include <linux/of_device.h> +#include <linux/module.h> +#include <linux/regmap.h> +#include <sound/soc.h> +#include <sound/pcm_params.h> + +#define PW_MGMT1 0x00 /* Power Management 1 */ +#define PW_MGMT2 0x01 /* Power Management 2 */ +#define PW_MGMT3 0x02 /* Power Management 3 */ +#define CTRL1 0x03 /* Control 1 */ +#define CTRL2 0x04 /* Control 2 */ +#define DEMP1 0x05 /* De-emphasis1 */ +#define DEMP2 0x06 /* De-emphasis2 */ +#define OFD 0x07 /* Overflow Detect */ +#define ZRD 0x08 /* Zero Detect */ +#define ICTRL 0x09 /* Input Control */ +#define OCTRL 0x0a /* Output Control */ +#define LOUT1 0x0b /* LOUT1 Volume Control */ +#define ROUT1 0x0c /* ROUT1 Volume Control */ +#define LOUT2 0x0d /* LOUT2 Volume Control */ +#define ROUT2 0x0e /* ROUT2 Volume Control */ +#define LOUT3 0x0f /* LOUT3 Volume Control */ +#define ROUT3 0x10 /* ROUT3 Volume Control */ +#define LOUT4 0x11 /* LOUT4 Volume Control */ +#define ROUT4 0x12 /* ROUT4 Volume Control */ +#define LOUT5 0x13 /* LOUT5 Volume Control */ +#define ROUT5 0x14 /* ROUT5 Volume Control */ +#define LOUT6 0x15 /* LOUT6 Volume Control */ +#define ROUT6 0x16 /* ROUT6 Volume Control */ + +/* PW_MGMT1 */ +#define RSTN BIT(0) +#define PMDAC BIT(1) +#define PMADC BIT(2) +#define PMVR BIT(3) + +/* PW_MGMT2 */ +#define PMAD_ALL 0x7 + +/* PW_MGMT3 */ +#define PMDA_ALL 0x3f + +/* CTRL1 */ +#define DIF0 BIT(3) +#define DIF1 BIT(4) +#define DIF2 BIT(5) +#define TDM0 BIT(6) +#define TDM1 BIT(7) +#define NO_FMT (0xff) +#define FMT_MASK (0xf8) + +/* CTRL2 */ +#define DFS_NORMAL_SPEED (0 << 2) +#define DFS_DOUBLE_SPEED (1 << 2) +#define DFS_QUAD_SPEED (2 << 2) + +struct ak4613_priv { + spinlock_t lock; + + unsigned int fmt; + u8 fmt_ctrl; + int cnt; +}; + +struct ak4613_formats { + unsigned int width; + unsigned int fmt; +}; + +struct ak4613_interface { + struct ak4613_formats capture; + struct ak4613_formats playback; +}; + +static const struct reg_default ak4613_reg[] = { + { 0x0, 0x0f }, { 0x1, 0x07 }, { 0x2, 0x3f }, { 0x3, 0x20 }, + { 0x4, 0x20 }, { 0x5, 0x55 }, { 0x6, 0x05 }, { 0x7, 0x07 }, + { 0x8, 0x0f }, { 0x9, 0x07 }, { 0xa, 0x3f }, { 0xb, 0x00 }, + { 0xc, 0x00 }, { 0xd, 0x00 }, { 0xe, 0x00 }, { 0xf, 0x00 }, + { 0x10, 0x00 }, { 0x11, 0x00 }, { 0x12, 0x00 }, { 0x13, 0x00 }, + { 0x14, 0x00 }, { 0x15, 0x00 }, { 0x16, 0x00 }, +}; + +#define AUDIO_IFACE_IDX_TO_VAL(i) (i << 3) +#define AUDIO_IFACE(b, fmt) { b, SND_SOC_DAIFMT_##fmt } +static const struct ak4613_interface ak4613_iface[] = { + /* capture */ /* playback */ + [0] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(16, RIGHT_J) }, + [1] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(20, RIGHT_J) }, + [2] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(24, RIGHT_J) }, + [3] = { AUDIO_IFACE(24, LEFT_J), AUDIO_IFACE(24, LEFT_J) }, + [4] = { AUDIO_IFACE(24, I2S), AUDIO_IFACE(24, I2S) }, +}; + +static const struct regmap_config ak4613_regmap_cfg = { + .reg_bits = 8, + .val_bits = 8, + .max_register = 0x16, + .reg_defaults = ak4613_reg, + .num_reg_defaults = ARRAY_SIZE(ak4613_reg), +}; + +static const struct of_device_id ak4613_of_match[] = { + { .compatible = "asahi-kasei,ak4613", .data = &ak4613_regmap_cfg }, + {}, +}; +MODULE_DEVICE_TABLE(of, ak4613_of_match); + +static const struct i2c_device_id ak4613_i2c_id[] = { + { "ak4613", (kernel_ulong_t)&ak4613_regmap_cfg }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ak4613_i2c_id); + +static const struct snd_soc_dapm_widget ak4613_dapm_widgets[] = { + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("LOUT1"), + SND_SOC_DAPM_OUTPUT("LOUT2"), + SND_SOC_DAPM_OUTPUT("LOUT3"), + SND_SOC_DAPM_OUTPUT("LOUT4"), + SND_SOC_DAPM_OUTPUT("LOUT5"), + SND_SOC_DAPM_OUTPUT("LOUT6"), + + SND_SOC_DAPM_OUTPUT("ROUT1"), + SND_SOC_DAPM_OUTPUT("ROUT2"), + SND_SOC_DAPM_OUTPUT("ROUT3"), + SND_SOC_DAPM_OUTPUT("ROUT4"), + SND_SOC_DAPM_OUTPUT("ROUT5"), + SND_SOC_DAPM_OUTPUT("ROUT6"), + + /* Inputs */ + SND_SOC_DAPM_INPUT("LIN1"), + SND_SOC_DAPM_INPUT("LIN2"), + + SND_SOC_DAPM_INPUT("RIN1"), + SND_SOC_DAPM_INPUT("RIN2"), + + /* DAC */ + SND_SOC_DAPM_DAC("DAC1", NULL, PW_MGMT3, 0, 0), + SND_SOC_DAPM_DAC("DAC2", NULL, PW_MGMT3, 1, 0), + SND_SOC_DAPM_DAC("DAC3", NULL, PW_MGMT3, 2, 0), + SND_SOC_DAPM_DAC("DAC4", NULL, PW_MGMT3, 3, 0), + SND_SOC_DAPM_DAC("DAC5", NULL, PW_MGMT3, 4, 0), + SND_SOC_DAPM_DAC("DAC6", NULL, PW_MGMT3, 5, 0), + + /* ADC */ + SND_SOC_DAPM_ADC("ADC1", NULL, PW_MGMT2, 0, 0), + SND_SOC_DAPM_ADC("ADC2", NULL, PW_MGMT2, 1, 0), +}; + +static const struct snd_soc_dapm_route ak4613_intercon[] = { + {"LOUT1", NULL, "DAC1"}, + {"LOUT2", NULL, "DAC2"}, + {"LOUT3", NULL, "DAC3"}, + {"LOUT4", NULL, "DAC4"}, + {"LOUT5", NULL, "DAC5"}, + {"LOUT6", NULL, "DAC6"}, + + {"ROUT1", NULL, "DAC1"}, + {"ROUT2", NULL, "DAC2"}, + {"ROUT3", NULL, "DAC3"}, + {"ROUT4", NULL, "DAC4"}, + {"ROUT5", NULL, "DAC5"}, + {"ROUT6", NULL, "DAC6"}, + + {"DAC1", NULL, "Playback"}, + {"DAC2", NULL, "Playback"}, + {"DAC3", NULL, "Playback"}, + {"DAC4", NULL, "Playback"}, + {"DAC5", NULL, "Playback"}, + {"DAC6", NULL, "Playback"}, + + {"Capture", NULL, "ADC1"}, + {"Capture", NULL, "ADC2"}, + + {"ADC1", NULL, "LIN1"}, + {"ADC2", NULL, "LIN2"}, + + {"ADC1", NULL, "RIN1"}, + {"ADC2", NULL, "RIN2"}, +}; + +static inline void ak4613_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int val) +{ + struct device *dev = codec->dev; + + dev_dbg(dev, "reg %02x w %02x\n", reg, val); + snd_soc_write(codec, reg, val); +} + +static inline unsigned int ak4613_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + struct device *dev = codec->dev; + unsigned int val = snd_soc_read(codec, reg); + + dev_dbg(dev, "reg %02x r %02x\n", reg, val); + + return val; +} + +static void ak4613_dai_shutdown(struct snd_pcm_substream *substream, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); + struct device *dev = codec->dev; + unsigned long flags; + + spin_lock_irqsave(&priv->lock, flags); + priv->cnt--; + if (priv->cnt < 0) { + dev_err(dev, "unexpected counter error\n"); + priv->cnt = 0; + } + if (0 == priv->cnt) + priv->fmt_ctrl = NO_FMT; + spin_unlock_irqrestore(&priv->lock, flags); +} + +static int ak4613_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); + + fmt &= SND_SOC_DAIFMT_FORMAT_MASK; + + switch (fmt) { + case SND_SOC_DAIFMT_RIGHT_J: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_I2S: + priv->fmt = fmt; + + break; + default: + return -EINVAL; + } + + return 0; +} + +static int ak4613_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct ak4613_priv *priv = snd_soc_codec_get_drvdata(codec); + const struct ak4613_formats *fmts; + struct device *dev = codec->dev; + unsigned int width = params_width(params); + unsigned int fmt = priv->fmt; + unsigned int rate; + unsigned long flags; + int is_play = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + int i, ret; + u8 fmt_ctrl, ctrl2; + + rate = params_rate(params); + switch (rate) { + case 32000: + case 44100: + case 48000: + ctrl2 = DFS_NORMAL_SPEED; + break; + case 88200: + case 96000: + ctrl2 = DFS_DOUBLE_SPEED; + break; + case 176400: + case 192000: + ctrl2 = DFS_QUAD_SPEED; + break; + default: + return -EINVAL; + } + + /* + * FIXME + * + * It doesn't support TDM at this point + */ + fmt_ctrl = NO_FMT; + for (i = 0; i < ARRAY_SIZE(ak4613_iface); i++) { + fmts = (is_play) ? &ak4613_iface[i].playback : + &ak4613_iface[i].capture; + + if (fmts->fmt != fmt) + continue; + + if (SND_SOC_DAIFMT_RIGHT_J == fmt) { + if (fmts->width != width) + continue; + } else { + if (fmts->width < width) + continue; + } + + fmt_ctrl = AUDIO_IFACE_IDX_TO_VAL(i); + break; + } + + ret = -EINVAL; + if (NO_FMT == fmt_ctrl) + goto hw_params_end; + + spin_lock_irqsave(&priv->lock, flags); + if ((NO_FMT == priv->fmt_ctrl) || + (fmt_ctrl == priv->fmt_ctrl)) { + priv->fmt_ctrl = fmt_ctrl; + priv->cnt++; + ret = 0; + } + spin_unlock_irqrestore(&priv->lock, flags); + + if (ret < 0) + goto hw_params_end; + + snd_soc_update_bits(codec, CTRL1, FMT_MASK, fmt_ctrl); + ak4613_write(codec, CTRL2, ctrl2); + +hw_params_end: + if (ret < 0) + dev_warn(dev, "unsupported data width/format combination\n"); + + return ret; +} + +static int ak4613_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u8 mgmt1 = 0; + + switch (level) { + case SND_SOC_BIAS_ON: + mgmt1 |= RSTN; + /* fall through */ + case SND_SOC_BIAS_PREPARE: + mgmt1 |= PMADC | PMDAC; + /* fall through */ + case SND_SOC_BIAS_STANDBY: + mgmt1 |= PMVR; + /* fall through */ + case SND_SOC_BIAS_OFF: + default: + break; + } + + ak4613_write(codec, PW_MGMT1, mgmt1); + + return 0; +} + +static const struct snd_soc_dai_ops ak4613_dai_ops = { + .shutdown = ak4613_dai_shutdown, + .set_fmt = ak4613_dai_set_fmt, + .hw_params = ak4613_dai_hw_params, +}; + +#define AK4613_PCM_RATE (SNDRV_PCM_RATE_32000 |\ + SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000 |\ + SNDRV_PCM_RATE_64000 |\ + SNDRV_PCM_RATE_88200 |\ + SNDRV_PCM_RATE_96000 |\ + SNDRV_PCM_RATE_176400 |\ + SNDRV_PCM_RATE_192000) +#define AK4613_PCM_FMTBIT (SNDRV_PCM_FMTBIT_S16_LE |\ + SNDRV_PCM_FMTBIT_S24_LE) + +static struct snd_soc_dai_driver ak4613_dai = { + .name = "ak4613-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 2, + .channels_max = 2, + .rates = AK4613_PCM_RATE, + .formats = AK4613_PCM_FMTBIT, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 2, + .channels_max = 2, + .rates = AK4613_PCM_RATE, + .formats = AK4613_PCM_FMTBIT, + }, + .ops = &ak4613_dai_ops, + .symmetric_rates = 1, +}; + +static int ak4613_resume(struct snd_soc_codec *codec) +{ + struct regmap *regmap = dev_get_regmap(codec->dev, NULL); + + regcache_mark_dirty(regmap); + regcache_sync(regmap); + + return 0; +} + +static struct snd_soc_codec_driver soc_codec_dev_ak4613 = { + .resume = ak4613_resume, + .set_bias_level = ak4613_set_bias_level, + .dapm_widgets = ak4613_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(ak4613_dapm_widgets), + .dapm_routes = ak4613_intercon, + .num_dapm_routes = ARRAY_SIZE(ak4613_intercon), +}; + +static int ak4613_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct device_node *np = dev->of_node; + const struct regmap_config *regmap_cfg; + struct regmap *regmap; + struct ak4613_priv *priv; + + regmap_cfg = NULL; + if (np) { + const struct of_device_id *of_id; + + of_id = of_match_device(ak4613_of_match, dev); + if (of_id) + regmap_cfg = of_id->data; + } else { + regmap_cfg = (const struct regmap_config *)id->driver_data; + } + + if (!regmap_cfg) + return -EINVAL; + + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + priv->fmt_ctrl = NO_FMT; + priv->cnt = 0; + + spin_lock_init(&priv->lock); + + i2c_set_clientdata(i2c, priv); + + regmap = devm_regmap_init_i2c(i2c, regmap_cfg); + if (IS_ERR(regmap)) + return PTR_ERR(regmap); + + return snd_soc_register_codec(dev, &soc_codec_dev_ak4613, + &ak4613_dai, 1); +} + +static int ak4613_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static struct i2c_driver ak4613_i2c_driver = { + .driver = { + .name = "ak4613-codec", + .owner = THIS_MODULE, + .of_match_table = ak4613_of_match, + }, + .probe = ak4613_i2c_probe, + .remove = ak4613_i2c_remove, + .id_table = ak4613_i2c_id, +}; + +module_i2c_driver(ak4613_i2c_driver); + +MODULE_DESCRIPTION("Soc AK4613 driver"); +MODULE_AUTHOR("Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com>"); +MODULE_LICENSE("GPL v2"); -- 1.9.1

2 2

[alsa-devel] [PATCH] ASoC: rt5645: Remove incorrect settings
by Oder Chiou 14 Sep '15

14 Sep '15

The patch removes the incorrect settings to avoid the pop sound in the first playback with headphone after boot. Signed-off-by: Oder Chiou <oder_chiou(a)realtek.com> --- sound/soc/codecs/rt5645.c | 6 ------ 1 file changed, 6 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 4972bf3..595ace9 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -2829,10 +2829,6 @@ static int rt5645_jack_detect(struct snd_soc_codec *codec, int jack_insert) snd_soc_dapm_sync(dapm); rt5645->jack_type = SND_JACK_HEADPHONE; } - - snd_soc_update_bits(codec, RT5645_CHARGE_PUMP, 0x0300, 0x0200); - snd_soc_write(codec, RT5645_DEPOP_M1, 0x001d); - snd_soc_write(codec, RT5645_DEPOP_M1, 0x0001); } else { /* jack out */ rt5645->jack_type = 0; @@ -2880,8 +2876,6 @@ int rt5645_set_jack_detect(struct snd_soc_codec *codec, rt5645->en_button_func = true; regmap_update_bits(rt5645->regmap, RT5645_GPIO_CTRL1, RT5645_GP1_PIN_IRQ, RT5645_GP1_PIN_IRQ); - regmap_update_bits(rt5645->regmap, RT5645_DEPOP_M1, - RT5645_HP_CB_MASK, RT5645_HP_CB_PU); regmap_update_bits(rt5645->regmap, RT5645_GEN_CTRL1, RT5645_DIG_GATE_CTRL, RT5645_DIG_GATE_CTRL); } -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH 00/10] Add support for img AXD audio hardware decoder
by Qais Yousef 14 Sep '15

14 Sep '15

This patch series adds AXD Alsa Compress Offload SoC driver. AXD is an audio hardware based on MIPS architecture that supports decoding, encoding, GEQ, resampling, mixing and synchronisation. At the moment only decoding support is added in hope to add the rest of the functionality on top of that once this is accepted. I divided the files into separate patches by functionality in hope it'll make the reviewing process easier. Worth noting that a lot of the cmd interface helper funtions in patch 7 are not used yet but will be as support for more functionality is added later. At the moment this code has been tested on Pistachio SoC using gstreamer patched with the code in this link https://bugzilla.gnome.org/show_bug.cgi?id=743192 Qais Yousef (10): irqchip: irq-mips-gic: export gic_send_ipi dt: add img,axd.txt device tree binding document ALSA: add AXD Audio Processing IP alsa driver ALSA: axd: add fw binary header manipulation files ALSA: axd: add buffers manipulation files ALSA: axd: add basic files for sending/receiving axd cmds ALSA: axd: add cmd interface helper functions ALSA: axd: add low level AXD platform setup files ALSA: axd: add alsa compress offload operations ALSA: axd: add Makefile .../devicetree/bindings/sound/img,axd.txt | 34 + drivers/irqchip/irq-mips-gic.c | 1 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/img/Kconfig | 11 + sound/soc/img/Makefile | 1 + sound/soc/img/axd/Makefile | 13 + sound/soc/img/axd/axd_alsa_ops.c | 211 ++ sound/soc/img/axd/axd_api.h | 649 ++++ sound/soc/img/axd/axd_buffers.c | 243 ++ sound/soc/img/axd/axd_buffers.h | 74 + sound/soc/img/axd/axd_cmds.c | 102 + sound/soc/img/axd/axd_cmds.h | 532 ++++ sound/soc/img/axd/axd_cmds_config.c | 1235 ++++++++ sound/soc/img/axd/axd_cmds_decoder_config.c | 422 +++ sound/soc/img/axd/axd_cmds_info.c | 1249 ++++++++ sound/soc/img/axd/axd_cmds_internal.c | 3264 ++++++++++++++++++++ sound/soc/img/axd/axd_cmds_internal.h | 317 ++ sound/soc/img/axd/axd_cmds_pipes.c | 1387 +++++++++ sound/soc/img/axd/axd_hdr.c | 64 + sound/soc/img/axd/axd_hdr.h | 24 + sound/soc/img/axd/axd_module.c | 742 +++++ sound/soc/img/axd/axd_module.h | 83 + sound/soc/img/axd/axd_platform.h | 35 + sound/soc/img/axd/axd_platform_mips.c | 416 +++ 25 files changed, 11111 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/img,axd.txt create mode 100644 sound/soc/img/Kconfig create mode 100644 sound/soc/img/Makefile create mode 100644 sound/soc/img/axd/Makefile create mode 100644 sound/soc/img/axd/axd_alsa_ops.c create mode 100644 sound/soc/img/axd/axd_api.h create mode 100644 sound/soc/img/axd/axd_buffers.c create mode 100644 sound/soc/img/axd/axd_buffers.h create mode 100644 sound/soc/img/axd/axd_cmds.c create mode 100644 sound/soc/img/axd/axd_cmds.h create mode 100644 sound/soc/img/axd/axd_cmds_config.c create mode 100644 sound/soc/img/axd/axd_cmds_decoder_config.c create mode 100644 sound/soc/img/axd/axd_cmds_info.c create mode 100644 sound/soc/img/axd/axd_cmds_internal.c create mode 100644 sound/soc/img/axd/axd_cmds_internal.h create mode 100644 sound/soc/img/axd/axd_cmds_pipes.c create mode 100644 sound/soc/img/axd/axd_hdr.c create mode 100644 sound/soc/img/axd/axd_hdr.h create mode 100644 sound/soc/img/axd/axd_module.c create mode 100644 sound/soc/img/axd/axd_module.h create mode 100644 sound/soc/img/axd/axd_platform.h create mode 100644 sound/soc/img/axd/axd_platform_mips.c Cc: Thomas Gleixner <tglx(a)linutronix.de> Cc: Jason Cooper <jason(a)lakedaemon.net> Cc: Marc Zyngier <marc.zyngier(a)arm.com> Cc: linux-kernel(a)vger.kernel.org Cc: linux-mips(a)linux-mips.org Cc: Rob Herring <robh+dt(a)kernel.org> Cc: Pawel Moll <pawel.moll(a)arm.com> Cc: Mark Rutland <mark.rutland(a)arm.com> Cc: Ian Campbell <ijc+devicetree(a)hellion.org.uk> Cc: Kumar Gala <galak(a)codeaurora.org> Cc: devicetree(a)vger.kernel.org Cc: Liam Girdwood <lgirdwood(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> -- 2.1.0

7 57

[alsa-devel] [PATCH] ASoC: nau8825: Add driver for headset chip Nuvoton 8825
by Anatol Pomozov 14 Sep '15

14 Sep '15

Sponsored-by: Google Chromium project Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- .../devicetree/bindings/sound/nau8825.txt | 88 ++ sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/nau8825.c | 1041 ++++++++++++++++++++ sound/soc/codecs/nau8825.h | 317 ++++++ 5 files changed, 1452 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nau8825.txt create mode 100644 sound/soc/codecs/nau8825.c create mode 100644 sound/soc/codecs/nau8825.h diff --git a/Documentation/devicetree/bindings/sound/nau8825.txt b/Documentation/devicetree/bindings/sound/nau8825.txt new file mode 100644 index 0000000..380e820 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8825.txt @@ -0,0 +1,88 @@ +Nuvoton NAU8825 audio codec + +This device supports I2C only. + +Required properties: + - compatible : Must be "nuvoton,nau8825" + + - reg : the I2C address of the device. This is either 0x1a (CSB=0) or 0x1b (CSB=1). + +Optional properties: + - nuvoton,jkdet-enable: Enable jack detection via JKDET pin. + - nuvoton,jkdet-pull-enable: Enable JKDET pin pull. If set - pin pull enabled, + otherwise pin in high impedance state. + - nuvoton,jkdet-pull-up: Pull-up JKDET pin. If set then JKDET pin is pull up, otherwise pull down. + - nuvoton,jkdet-polarity: JKDET pin polarity. 0 - active high, 1 - active low. + + - nuvoton,vref-impedance: VREF Impedance selection + 0 - Open + 1 - 25 kOhm + 2 - 125 kOhm + 3 - 2.5 kOhm + + - nuvoton,micbias-voltage: Micbias voltage level. + 0 - VDDA + 1 - VDDA + 2 - VDDA * 1.1 + 3 - VDDA * 1.2 + 4 - VDDA * 1.3 + 5 - VDDA * 1.4 + 6 - VDDA * 1.53 + 7 - VDDA * 1.53 + + - nuvoton,sar-threshold-num: Number of buttons supported + - nuvoton,sar-threshold: Impedance threshold for each button. Array that contains up to 8 buttons configuration. SAR value is calculated as + SAR = 255 * MICBIAS / SAR_VOLTAGE * R / (2000 + R) + where MICBIAS is configured by 'nuvoton,micbias-voltage', SAR_VOLTAGE is configured by 'nuvoton,sar-voltage', R - button impedance. + Refer datasheet section 10.2 for more information about threshold calculation. + + - nuvoton,sar-hysteresis: Button impedance measurement hysteresis. + + - nuvoton,sar-voltage: Reference voltage for button impedance measurement. + 0 - VDDA + 1 - VDDA + 2 - VDDA * 1.1 + 3 - VDDA * 1.2 + 4 - VDDA * 1.3 + 5 - VDDA * 1.4 + 6 - VDDA * 1.53 + 7 - VDDA * 1.53 + + - nuvoton,short-key-debounce: Button short key press debounce time. + 0 - 30ms + 1 - 50ms + 2 - 100ms + 3 - 30ms + + - nuvoton,jack-insert-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms + - nuvoton,jack-eject-debounce: number from 0 to 7 that sets debounce time to 2^(n+2) ms + + - clocks: list of phandle and clock specifier pairs according to common clock bindings for the + clocks described in clock-names + - clock-names: should include "mclk" for the MCLK master clock + +Example: + + headset: nau8825@1a { + compatible = "nuvoton,nau8825"; + reg = <0x1a>; + interrupt-parent = <&gpio>; + interrupts = <TEGRA_GPIO(E, 6) IRQ_TYPE_LEVEL_LOW>; + nuvoton,jkdet-enable; + nuvoton,jkdet-pull-enable; + nuvoton,jkdet-pull-up; + nuvoton,jkdet-polarity = <GPIO_ACTIVE_LOW>; + nuvoton,vref-impedance = <2>; + nuvoton,micbias-voltage = <6>; + // Setup 4 buttons impedance according to Android specification + nuvoton,sar-threshold-num = <4>; + nuvoton,sar-threshold = <0xc 0x1e 0x38 0x60>; + nuvoton,sar-hysteresis = <1>; + nuvoton,sar-voltage = <0>; + nuvoton,short-key-debounce = <2>; + nuvoton,jack-insert-debounce = <7>; + nuvoton,jack-eject-debounce = <7>; + + clock-names = "mclk"; + clocks = <&tegra_car TEGRA210_CLK_CLK_OUT_2>; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0c9733e..9cabc05 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -79,6 +79,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9877 if I2C select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C + select SND_SOC_NAU8825 if I2C select SND_SOC_HDMI_CODEC select SND_SOC_PCM1681 if I2C select SND_SOC_PCM1792A if SPI_MASTER @@ -896,6 +897,9 @@ config SND_SOC_MC13783 config SND_SOC_ML26124 tristate +config SND_SOC_NAU8825 + tristate + config SND_SOC_TPA6130A2 tristate "Texas Instruments TPA6130A2 headphone amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 4a32077..bfbbb2a 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -72,6 +72,7 @@ snd-soc-max98925-objs := max98925.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o +snd-soc-nau8825-objs := nau8825.o snd-soc-hdmi-codec-objs := hdmi.o snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm1792a-codec-objs := pcm1792a.o @@ -264,6 +265,7 @@ obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o +obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c new file mode 100644 index 0000000..ae36d8b --- /dev/null +++ b/sound/soc/codecs/nau8825.c @@ -0,0 +1,1041 @@ +/* + * Nuvoton NAU8825 audio codec driver + * + * Copyright 2015 Google Chromium project. + * Author: Anatol Pomozov <anatol(a)chromium.org> + * Copyright 2015 Nuvoton Technology Corp. + * Co-author: Meng-Huang Kuo <mhkuo(a)nuvoton.com> + * + * Licensed under the GPL-2. + */ + +#include <linux/module.h> +#include <linux/delay.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <linux/clk.h> + +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> + + +#include "nau8825.h" + +static const struct reg_default nau8825_reg_defaults[] = { + { NAU8825_REG_ENA_CTRL, 0x00ff }, + { NAU8825_REG_CLK_DIVIDER, 0x0050 }, + { NAU8825_REG_FLL1, 0x0 }, + { NAU8825_REG_FLL2, 0x3126 }, + { NAU8825_REG_FLL3, 0x0008 }, + { NAU8825_REG_FLL4, 0x0010 }, + { NAU8825_REG_FLL5, 0x0 }, + { NAU8825_REG_FLL6, 0x6000 }, + { NAU8825_REG_FLL_VCO_RSV, 0xf13c }, + { NAU8825_REG_HSD_CTRL, 0x000c }, + { NAU8825_REG_JACK_DET_CTRL, 0x0 }, + { NAU8825_REG_INTERRUPT_MASK, 0x0 }, + { NAU8825_REG_INTERRUPT_DIS_CTRL, 0xffff }, + { NAU8825_REG_SAR_CTRL, 0x0015 }, + { NAU8825_REG_KEYDET_CTRL, 0x0110 }, + { NAU8825_REG_VDET_THRESHOLD_1, 0x0 }, + { NAU8825_REG_VDET_THRESHOLD_2, 0x0 }, + { NAU8825_REG_VDET_THRESHOLD_3, 0x0 }, + { NAU8825_REG_VDET_THRESHOLD_4, 0x0 }, + { NAU8825_REG_GPIO34_CTRL, 0x0 }, + { NAU8825_REG_GPIO12_CTRL, 0x0 }, + { NAU8825_REG_TDM_CTRL, 0x0 }, + { NAU8825_REG_I2S_PCM_CTRL1, 0x000b }, + { NAU8825_REG_I2S_PCM_CTRL2, 0x8010 }, + { NAU8825_REG_LEFT_TIME_SLOT, 0x0 }, + { NAU8825_REG_RIGHT_TIME_SLOT, 0x0 }, + { NAU8825_REG_BIQ_CTRL, 0x0 }, + { NAU8825_REG_BIQ_COF1, 0x0 }, + { NAU8825_REG_BIQ_COF2, 0x0 }, + { NAU8825_REG_BIQ_COF3, 0x0 }, + { NAU8825_REG_BIQ_COF4, 0x0 }, + { NAU8825_REG_BIQ_COF5, 0x0 }, + { NAU8825_REG_BIQ_COF6, 0x0 }, + { NAU8825_REG_BIQ_COF7, 0x0 }, + { NAU8825_REG_BIQ_COF8, 0x0 }, + { NAU8825_REG_BIQ_COF9, 0x0 }, + { NAU8825_REG_BIQ_COF10, 0x0 }, + { NAU8825_REG_ADC_RATE, 0x0010 }, + { NAU8825_REG_DAC_CTRL1, 0x0001 }, + { NAU8825_REG_DAC_CTRL2, 0x0 }, + { NAU8825_REG_DAC_DGAIN_CTRL, 0x0 }, + { NAU8825_REG_ADC_DGAIN_CTRL, 0x00cf }, + { NAU8825_REG_MUTE_CTRL, 0x0 }, + { NAU8825_REG_HSVOL_CTRL, 0x0 }, + { NAU8825_REG_DACL_CTRL, 0x02cf }, + { NAU8825_REG_DACR_CTRL, 0x00cf }, + { NAU8825_REG_ADC_DRC_KNEE_IP12, 0x1486 }, + { NAU8825_REG_ADC_DRC_KNEE_IP34, 0x0f12 }, + { NAU8825_REG_ADC_DRC_SLOPES, 0x25ff }, + { NAU8825_REG_ADC_DRC_ATKDCY, 0x3457 }, + { NAU8825_REG_DAC_DRC_KNEE_IP12, 0x1486 }, + { NAU8825_REG_DAC_DRC_KNEE_IP34, 0x0f12 }, + { NAU8825_REG_DAC_DRC_SLOPES, 0x25f9 }, + { NAU8825_REG_DAC_DRC_ATKDCY, 0x3457 }, + { NAU8825_REG_IMM_MODE_CTRL, 0x0 }, + { NAU8825_REG_CLASSG_CTRL, 0x0 }, + { NAU8825_REG_OPT_EFUSE_CTRL, 0x0 }, + { NAU8825_REG_MISC_CTRL, 0x0 }, + { NAU8825_REG_BIAS_ADJ, 0x0 }, + { NAU8825_REG_TRIM_SETTINGS, 0x0 }, + { NAU8825_REG_ANALOG_CONTROL_1, 0x0 }, + { NAU8825_REG_ANALOG_CONTROL_2, 0x0 }, + { NAU8825_REG_ANALOG_ADC_1, 0x0011 }, + { NAU8825_REG_ANALOG_ADC_2, 0x0020 }, + { NAU8825_REG_RDAC, 0x0008 }, + { NAU8825_REG_MIC_BIAS, 0x0006 }, + { NAU8825_REG_BOOST, 0x0 }, + { NAU8825_REG_FEPGA, 0x0 }, + { NAU8825_REG_POWER_UP_CONTROL, 0x0 }, + { NAU8825_REG_CHARGE_PUMP, 0x0 }, +}; + +static bool nau8825_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_REG_ENA_CTRL: + case NAU8825_REG_CLK_DIVIDER ... NAU8825_REG_FLL_VCO_RSV: + case NAU8825_REG_HSD_CTRL ... NAU8825_REG_JACK_DET_CTRL: + case NAU8825_REG_INTERRUPT_MASK ... NAU8825_REG_KEYDET_CTRL: + case NAU8825_REG_VDET_THRESHOLD_1 ... NAU8825_REG_DACR_CTRL: + case NAU8825_REG_ADC_DRC_KNEE_IP12 ... NAU8825_REG_ADC_DRC_ATKDCY: + case NAU8825_REG_DAC_DRC_KNEE_IP12 ... NAU8825_REG_DAC_DRC_ATKDCY: + case NAU8825_REG_IMM_MODE_CTRL ... NAU8825_REG_IMM_RMS_R: + case NAU8825_REG_CLASSG_CTRL ... NAU8825_REG_OPT_EFUSE_CTRL: + case NAU8825_REG_MISC_CTRL: + case NAU8825_REG_I2C_DEVICE_ID ... NAU8825_REG_SARDOUT_RAM_STATUS: + case NAU8825_REG_BIAS_ADJ: + case NAU8825_REG_TRIM_SETTINGS ... NAU8825_REG_ANALOG_CONTROL_2: + case NAU8825_REG_ANALOG_ADC_1 ... NAU8825_REG_MIC_BIAS: + case NAU8825_REG_BOOST ... NAU8825_REG_FEPGA: + case NAU8825_REG_POWER_UP_CONTROL ... NAU8825_REG_GENERAL_STATUS: + return true; + default: + return false; + } + +} + +static bool nau8825_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_REG_RESET ... NAU8825_REG_ENA_CTRL: + case NAU8825_REG_CLK_DIVIDER ... NAU8825_REG_FLL_VCO_RSV: + case NAU8825_REG_HSD_CTRL ... NAU8825_REG_JACK_DET_CTRL: + case NAU8825_REG_INTERRUPT_MASK: + case NAU8825_REG_INT_CLR_KEY_STATUS ... NAU8825_REG_KEYDET_CTRL: + case NAU8825_REG_VDET_THRESHOLD_1 ... NAU8825_REG_DACR_CTRL: + case NAU8825_REG_ADC_DRC_KNEE_IP12 ... NAU8825_REG_ADC_DRC_ATKDCY: + case NAU8825_REG_DAC_DRC_KNEE_IP12 ... NAU8825_REG_DAC_DRC_ATKDCY: + case NAU8825_REG_IMM_MODE_CTRL: + case NAU8825_REG_CLASSG_CTRL ... NAU8825_REG_OPT_EFUSE_CTRL: + case NAU8825_REG_MISC_CTRL: + case NAU8825_REG_BIAS_ADJ: + case NAU8825_REG_TRIM_SETTINGS ... NAU8825_REG_ANALOG_CONTROL_2: + case NAU8825_REG_ANALOG_ADC_1 ... NAU8825_REG_MIC_BIAS: + case NAU8825_REG_BOOST ... NAU8825_REG_FEPGA: + case NAU8825_REG_POWER_UP_CONTROL ... NAU8825_REG_CHARGE_PUMP: + return true; + default: + return false; + } +} + +static bool nau8825_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_REG_RESET: + case NAU8825_REG_IRQ_STATUS: + case NAU8825_REG_INT_CLR_KEY_STATUS: + case NAU8825_REG_IMM_RMS_L: + case NAU8825_REG_IMM_RMS_R: + case NAU8825_REG_I2C_DEVICE_ID: + case NAU8825_REG_SARDOUT_RAM_STATUS: + case NAU8825_REG_CHARGE_PUMP_INPUT_READ: + case NAU8825_REG_GENERAL_STATUS: + return true; + default: + return false; + } +} + +static int nau8825_pump_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /* Prevent startup click by letting charge pump to ramp up */ + msleep(10); + break; + default: + return -EINVAL; + } + + return 0; +} + +static const char * const nau8825_adc_decimation[] = { + "32", "64", "128", "256" +}; + +static const struct soc_enum nau8825_adc_decimation_enum = + SOC_ENUM_SINGLE(NAU8825_REG_ADC_RATE, NAU8825_ADC_SYNC_DOWN_SFT, + ARRAY_SIZE(nau8825_adc_decimation), nau8825_adc_decimation); + +static const char * const nau8825_dac_oversampl[] = { + "64", "256", "128", "", "32" +}; + +static const struct soc_enum nau8825_dac_oversampl_enum = + SOC_ENUM_SINGLE(NAU8825_REG_DAC_CTRL1, NAU8825_DAC_OVERSAMPLE_SFT, + ARRAY_SIZE(nau8825_dac_oversampl), nau8825_dac_oversampl); + +static const DECLARE_TLV_DB_MINMAX_MUTE(adc_vol_tlv, -10300, 2400); +static const DECLARE_TLV_DB_MINMAX_MUTE(sidetone_vol_tlv, -4200, 0); +static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -5400, 0); +static const DECLARE_TLV_DB_MINMAX(fepga_gain_tlv, -100, 3600); + +static const struct snd_kcontrol_new nau8825_controls[] = { + SOC_SINGLE_TLV("MIC Volume", NAU8825_REG_ADC_DGAIN_CTRL, + 0, 0xff, 0, adc_vol_tlv), + SOC_DOUBLE_TLV("HP Sidetone Volume", NAU8825_REG_ADC_DGAIN_CTRL, + 12, 8, 0x0f, 0, sidetone_vol_tlv), + SOC_DOUBLE_TLV("HP Volume", NAU8825_REG_HSVOL_CTRL, + 6, 0, 0x3f, 1, dac_vol_tlv), + SOC_SINGLE_TLV("Frontend PGA Gain", NAU8825_REG_POWER_UP_CONTROL, + 8, 37, 0, fepga_gain_tlv), + + SOC_ENUM("ADC Decimation Rate", nau8825_adc_decimation_enum), + SOC_ENUM("DAC Oversampling Rate", nau8825_dac_oversampl_enum), +}; + +/* DAC Mux 0x33[9] and 0x34[9] */ +static const char * const nau8825_dac_src[] = { + "DACL", "DACR", +}; + +static SOC_ENUM_SINGLE_DECL( + nau8825_dacl_enum, NAU8825_REG_DACL_CTRL, + NAU8825_DACL_CH_SEL_SFT, nau8825_dac_src); + +static SOC_ENUM_SINGLE_DECL( + nau8825_dacr_enum, NAU8825_REG_DACR_CTRL, + NAU8825_DACR_CH_SEL_SFT, nau8825_dac_src); + +static const struct snd_kcontrol_new nau8825_dacl_mux = + SOC_DAPM_ENUM("DACL Source", nau8825_dacl_enum); + +static const struct snd_kcontrol_new nau8825_dacr_mux = + SOC_DAPM_ENUM("DACR Source", nau8825_dacr_enum); + + +static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { + SND_SOC_DAPM_AIF_OUT("AIFTX", "Capture", 0, NAU8825_REG_I2S_PCM_CTRL2, + 15, 1), + + SND_SOC_DAPM_INPUT("MIC"), + SND_SOC_DAPM_MICBIAS("MICBIAS", NAU8825_REG_MIC_BIAS, 8, 0), + + SND_SOC_DAPM_PGA("Frontend PGA", NAU8825_REG_POWER_UP_CONTROL, 14, 0, + NULL, 0), + + SND_SOC_DAPM_ADC("ADC", NULL, NAU8825_REG_ENA_CTRL, 8, 0), + SND_SOC_DAPM_SUPPLY("ADC Clock", NAU8825_REG_ENA_CTRL, 7, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC Power", NAU8825_REG_ANALOG_ADC_2, 6, 0, NULL, + 0), + + /* ADC for button press detection */ + SND_SOC_DAPM_ADC("SAR", NULL, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_ADC_EN_SFT, 0), + + SND_SOC_DAPM_DAC("ADACL", NULL, NAU8825_REG_RDAC, 12, 0), + SND_SOC_DAPM_DAC("ADACR", NULL, NAU8825_REG_RDAC, 13, 0), + SND_SOC_DAPM_SUPPLY("ADACL Clock", NAU8825_REG_RDAC, 8, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADACR Clock", NAU8825_REG_RDAC, 9, 0, NULL, 0), + + SND_SOC_DAPM_DAC("DDACR", NULL, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_DACR_SFT, 0), + SND_SOC_DAPM_DAC("DDACL", NULL, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_DACL_SFT, 0), + SND_SOC_DAPM_SUPPLY("DDAC Clock", NAU8825_REG_ENA_CTRL, 6, 0, NULL, 0), + + SND_SOC_DAPM_MUX("DACL Mux", SND_SOC_NOPM, 0, 0, &nau8825_dacl_mux), + SND_SOC_DAPM_MUX("DACR Mux", SND_SOC_NOPM, 0, 0, &nau8825_dacr_mux), + + SND_SOC_DAPM_PGA("HP amp L", NAU8825_REG_CLASSG_CTRL, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("HP amp R", NAU8825_REG_CLASSG_CTRL, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("HP amp power", NAU8825_REG_CLASSG_CTRL, 0, 0, NULL, + 0), + + SND_SOC_DAPM_SUPPLY("Charge Pump", NAU8825_REG_CHARGE_PUMP, 5, 0, + nau8825_pump_event, SND_SOC_DAPM_POST_PMU), + + SND_SOC_DAPM_PGA("Output Driver R Stage 1", + NAU8825_REG_POWER_UP_CONTROL, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("Output Driver L Stage 1", + NAU8825_REG_POWER_UP_CONTROL, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("Output Driver R Stage 2", + NAU8825_REG_POWER_UP_CONTROL, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("Output Driver L Stage 2", + NAU8825_REG_POWER_UP_CONTROL, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("Output Driver R Stage 3", 1, + NAU8825_REG_POWER_UP_CONTROL, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA_S("Output Driver L Stage 3", 1, + NAU8825_REG_POWER_UP_CONTROL, 0, 0, NULL, 0), + + SND_SOC_DAPM_PGA_S("Output DACL", 2, NAU8825_REG_CHARGE_PUMP, 8, 1, NULL, 0), + SND_SOC_DAPM_PGA_S("Output DACR", 2, NAU8825_REG_CHARGE_PUMP, 9, 1, NULL, 0), + + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route nau8825_dapm_routes[] = { + {"Frontend PGA", NULL, "MIC"}, + {"ADC", NULL, "Frontend PGA"}, + {"ADC", NULL, "ADC Clock"}, + {"ADC", NULL, "ADC Power"}, + {"AIFTX", NULL, "ADC"}, + + {"DDACL", NULL, "Playback"}, + {"DDACR", NULL, "Playback"}, + {"DDACL", NULL, "DDAC Clock"}, + {"DDACR", NULL, "DDAC Clock"}, + {"DACL Mux", "DACL", "DDACL"}, + {"DACL Mux", "DACR", "DDACR"}, + {"DACR Mux", "DACL", "DDACL"}, + {"DACR Mux", "DACR", "DDACR"}, + {"HP amp L", NULL, "DACL Mux"}, + {"HP amp R", NULL, "DACR Mux"}, + {"HP amp L", NULL, "HP amp power"}, + {"HP amp R", NULL, "HP amp power"}, + {"ADACL", NULL, "HP amp L"}, + {"ADACR", NULL, "HP amp R"}, + {"ADACL", NULL, "ADACL Clock"}, + {"ADACR", NULL, "ADACR Clock"}, + {"Output Driver L Stage 1", NULL, "ADACL"}, + {"Output Driver R Stage 1", NULL, "ADACR"}, + {"Output Driver L Stage 2", NULL, "Output Driver L Stage 1"}, + {"Output Driver R Stage 2", NULL, "Output Driver R Stage 1"}, + {"Output Driver L Stage 3", NULL, "Output Driver L Stage 2"}, + {"Output Driver R Stage 3", NULL, "Output Driver R Stage 2"}, + {"Output DACL", NULL, "Output Driver L Stage 3"}, + {"Output DACR", NULL, "Output Driver R Stage 3"}, + {"HPOL", NULL, "Output DACL"}, + {"HPOR", NULL, "Output DACR"}, + {"HPOL", NULL, "Charge Pump"}, + {"HPOR", NULL, "Charge Pump"}, +}; + +static int nau8825_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0; + + switch (params_width(params)) { + case 16: + val_len |= NAU8825_I2S_DL_16; + break; + case 20: + val_len |= NAU8825_I2S_DL_20; + break; + case 24: + val_len |= NAU8825_I2S_DL_24; + break; + case 32: + val_len |= NAU8825_I2S_DL_32; + break; + default: + return -EINVAL; + } + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL1, + NAU8825_I2S_DL_MASK, val_len); + + return 0; +} + +static int nau8825_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + unsigned int ctrl1_val = 0, ctrl2_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl2_val |= NAU8825_I2S_MS_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl1_val |= NAU8825_I2S_BP_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl1_val |= NAU8825_I2S_DF_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1_val |= NAU8825_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl1_val |= NAU8825_I2S_DF_RIGTH; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl1_val |= NAU8825_I2S_DF_PCM_AB; + break; + case SND_SOC_DAIFMT_DSP_B: + ctrl1_val |= NAU8825_I2S_DF_PCM_AB; + ctrl1_val |= NAU8825_I2S_PCMB_EN; + break; + default: + return -EINVAL; + } + + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL1, + NAU8825_I2S_DL_MASK | NAU8825_I2S_DF_MASK | + NAU8825_I2S_BP_MASK | NAU8825_I2S_PCMB_MASK, + ctrl1_val); + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, ctrl2_val); + + return 0; +} + +static const struct snd_soc_dai_ops nau8825_dai_ops = { + .hw_params = nau8825_hw_params, + .set_fmt = nau8825_set_dai_fmt, +}; + +#define NAU8825_RATES SNDRV_PCM_RATE_8000_192000 +#define NAU8825_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver nau8825_dai = { + .name = "nau8825-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8825_RATES, + .formats = NAU8825_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 1, + .rates = NAU8825_RATES, + .formats = NAU8825_FORMATS, + }, + .ops = &nau8825_dai_ops, +}; + +/** + * nau8825_enable_jack_detect - Specify a jack for event reporting + * + * @component: component to register the jack with + * @jack: jack to use to report headset and button events on + * + * After this function has been called the headset insert/remove and button + * events will be routed to the given jack. Jack can be null to stop + * reporting. + */ +int nau8825_enable_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + nau8825->jack = jack; + + return 0; +} +EXPORT_SYMBOL_GPL(nau8825_enable_jack_detect); + + +static bool nau8825_is_jack_inserted(struct regmap *regmap) +{ + int status; + + regmap_read(regmap, NAU8825_REG_I2C_DEVICE_ID, &status); + return !(status & NAU8825_GPIO2JD1); +} + +static void nau8825_restart_jack_detection(struct regmap *regmap) +{ + /* this will restart the entire jack detection process including MIC/GND + * switching and create interrupts. We have to go from 0 to 1 and back + * to 0 to restart. + */ + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_RESTART, NAU8825_JACK_DET_RESTART); + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_RESTART, 0); +} + +static void nau8825_eject_jack(struct nau8825 *nau8825) +{ + struct snd_soc_dapm_context *dapm = nau8825->dapm; + struct regmap *regmap = nau8825->regmap; + + snd_soc_dapm_disable_pin(dapm, "SAR"); + snd_soc_dapm_disable_pin(dapm, "MICBIAS"); + /* Detach 2kOhm Resistors from MICBIAS to MICGND1/2 */ + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, 0); + /* ground HPL/HPR, MICGRND1/2 */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0xf, 0xf); + + snd_soc_dapm_sync(dapm); +} + +static int nau8825_button_decode(int value) +{ + int buttons; + + /* The chip supports up to 8 buttons, but ALSA defines only 6 buttons */ + if (value & BIT(0)) + buttons |= SND_JACK_BTN_0; + if (value & BIT(1)) + buttons |= SND_JACK_BTN_1; + if (value & BIT(2)) + buttons |= SND_JACK_BTN_2; + if (value & BIT(3)) + buttons |= SND_JACK_BTN_3; + if (value & BIT(4)) + buttons |= SND_JACK_BTN_4; + if (value & BIT(5)) + buttons |= SND_JACK_BTN_5; + + return buttons; +} + +static int nau8825_jack_insert(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + struct snd_soc_dapm_context *dapm = nau8825->dapm; + int jack_status_reg, mic_detected; + int type = 0; + + regmap_read(regmap, NAU8825_REG_GENERAL_STATUS, &jack_status_reg); + mic_detected = (jack_status_reg >> 10) & 3; + + switch (mic_detected) { + case 0: + /* no mic */ + type = SND_JACK_HEADPHONE; + break; + case 1: + dev_dbg(nau8825->dev, "OMTP (micgnd1) mic connected\n"); + type = SND_JACK_HEADSET; + + /* Unground MICGND1 */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 3 << 2, + 1 << 2); + /* Attach 2kOhm Resistor from MICBIAS to MICGND1 */ + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, + NAU8825_MICBIAS_JKR2); + /* Attach SARADC to MICGND1 */ + regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_INPUT_MASK, + NAU8825_SAR_INPUT_JKR2); + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); + snd_soc_dapm_force_enable_pin(dapm, "SAR"); + snd_soc_dapm_sync(dapm); + break; + case 2: + case 3: + dev_dbg(nau8825->dev, "CTIA (micgnd2) mic connected\n"); + type = SND_JACK_HEADSET; + + /* Unground MICGND2 */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 3 << 2, + 2 << 2); + /* Attach 2kOhm Resistor from MICBIAS to MICGND2 */ + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, + NAU8825_MICBIAS_JKSLV); + /* Attach SARADC to MICGND2 */ + regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_INPUT_MASK, + NAU8825_SAR_INPUT_JKSLV); + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); + snd_soc_dapm_force_enable_pin(dapm, "SAR"); + snd_soc_dapm_sync(dapm); + break; + } + + if (type & SND_JACK_HEADPHONE) { + /* Unground HPL/R */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0x3, 0); + } + + return type; +} + +#define NAU8825_BUTTONS (SND_JACK_BTN_0 | SND_JACK_BTN_1 | \ + SND_JACK_BTN_2 | SND_JACK_BTN_3) + +static irqreturn_t nau8825_interrupt(int irq, void *data) +{ + struct nau8825 *nau8825 = (struct nau8825 *)data; + struct regmap *regmap = nau8825->regmap; + int active_irq, clear_irq = 0, event = 0, event_mask = 0; + + regmap_read(regmap, NAU8825_REG_IRQ_STATUS, &active_irq); + + if ((active_irq & NAU8825_JACK_EJECTION_IRQ_MASK) == + NAU8825_JACK_EJECTION_DETECTED) { + + nau8825_eject_jack(nau8825); + event_mask |= SND_JACK_HEADSET; + clear_irq = NAU8825_JACK_EJECTION_IRQ_MASK; + } else if (active_irq & NAU8825_KEY_SHORT_PRESS_IRQ) { + int key_status; + + regmap_read(regmap, NAU8825_REG_INT_CLR_KEY_STATUS, + &key_status); + + /* upper 8 bits of the register are for short pressed keys, + lower 8 bits - for long pressed buttons */ + nau8825->button_pressed = nau8825_button_decode( + key_status >> 8); + + event |= nau8825->button_pressed; + event_mask |= NAU8825_BUTTONS; + clear_irq = NAU8825_KEY_SHORT_PRESS_IRQ; + } else if (active_irq & NAU8825_KEY_RELEASE_IRQ) { + event_mask = NAU8825_BUTTONS; + clear_irq = NAU8825_KEY_RELEASE_IRQ; + } else if (active_irq & NAU8825_HEADSET_COMPLETION_IRQ) { + if (nau8825_is_jack_inserted(regmap)) { + event |= nau8825_jack_insert(nau8825); + } else { + dev_warn(nau8825->dev, "Headset completion IRQ fired but no headset connected\n"); + nau8825_eject_jack(nau8825); + } + + event_mask |= SND_JACK_HEADSET; + clear_irq = NAU8825_HEADSET_COMPLETION_IRQ; + } + + if (!clear_irq) + clear_irq = active_irq; + /* clears the rightmost interruption */ + regmap_write(regmap, NAU8825_REG_INT_CLR_KEY_STATUS, clear_irq); + + if (event_mask) + snd_soc_jack_report(nau8825->jack, event, event_mask); + + return IRQ_HANDLED; +} + +static void nau8825_setup_buttons(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + + regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_TRACKING_GAIN_MASK, + nau8825->sar_voltage << NAU8825_SAR_TRACKING_GAIN_SFT); + + regmap_update_bits(regmap, NAU8825_REG_KEYDET_CTRL, + NAU8825_KEYDET_LEVELS_NR_MASK, + (nau8825->sar_threshold_num - 1) << NAU8825_KEYDET_LEVELS_NR_SFT); + regmap_update_bits(regmap, NAU8825_REG_KEYDET_CTRL, + NAU8825_KEYDET_HYSTERESIS_MASK, + nau8825->sar_hysteresis << NAU8825_KEYDET_HYSTERESIS_SFT); + regmap_update_bits(regmap, NAU8825_REG_KEYDET_CTRL, + NAU8825_KEYDET_SHORTKEY_DEBOUNCE_MASK, + nau8825->key_debounce << NAU8825_KEYDET_SHORTKEY_DEBOUNCE_SFT); + + regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_1, + (nau8825->sar_threshold[0] << 8) | nau8825->sar_threshold[1]); + regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_2, + (nau8825->sar_threshold[2] << 8) | nau8825->sar_threshold[3]); + regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_3, + (nau8825->sar_threshold[4] << 8) | nau8825->sar_threshold[5]); + regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_4, + (nau8825->sar_threshold[6] << 8) | nau8825->sar_threshold[7]); + + /* Enable short press and release interruptions */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_KEY_SHORT_PRESS_EN | NAU8825_IRQ_KEY_RELEASE_EN, + 0); +} + +static void nau8825_init_regs(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + + /* Bias and clock are needed for button detection */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_BIAS_ADJ, + NAU8825_BIAS_VMID, NAU8825_BIAS_VMID); + regmap_update_bits(nau8825->regmap, NAU8825_REG_BOOST, + NAU8825_GLOBAL_BIAS_EN, NAU8825_GLOBAL_BIAS_EN); + + /* VMID Tieoff */ + regmap_update_bits(regmap, NAU8825_REG_BIAS_ADJ, + NAU8825_BIAS_VMID_SEL_MASK, + nau8825->vref_impedance << NAU8825_BIAS_VMID_SEL_SFT); + /* Disable Boost Driver, Automatic Short circuit protection enable */ + regmap_update_bits(regmap, NAU8825_REG_BOOST, + NAU8825_PRECHARGE_DIS | NAU8825_HP_BOOST_G_DIS | + NAU8825_SHORT_SHUTDOWN_EN, + NAU8825_PRECHARGE_DIS | NAU8825_HP_BOOST_G_DIS | + NAU8825_SHORT_SHUTDOWN_EN); + + /* Ground HP Outputs[1:0], needed for headset auto detection + * Enable Automatic Mic/Gnd switching reading on insert interrupt[6] */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, + NAU8825_HSD_AUTO_MODE | NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L, + NAU8825_HSD_AUTO_MODE | NAU8825_SPKR_DWN1R | NAU8825_SPKR_DWN1L); + + regmap_update_bits(regmap, NAU8825_REG_GPIO12_CTRL, + NAU8825_JKDET_OUTPUT_EN, + nau8825->jkdet_enable ? 0 : NAU8825_JKDET_OUTPUT_EN); + regmap_update_bits(regmap, NAU8825_REG_GPIO12_CTRL, + NAU8825_JKDET_PULL_EN, + nau8825->jkdet_pull_enable ? 0 : NAU8825_JKDET_PULL_EN); + regmap_update_bits(regmap, NAU8825_REG_GPIO12_CTRL, + NAU8825_JKDET_PULL_UP, + nau8825->jkdet_pull_up ? NAU8825_JKDET_PULL_UP : 0); + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_POLARITY, + /* jkdet_polarity - 1 is for active-low */ + nau8825->jkdet_polarity ? 0 : NAU8825_JACK_POLARITY); + + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_INSERT_DEBOUNCE_MASK, + nau8825->jack_insert_debounce << NAU8825_JACK_INSERT_DEBOUNCE_SFT); + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_EJECT_DEBOUNCE_MASK, + nau8825->jack_eject_debounce << NAU8825_JACK_EJECT_DEBOUNCE_SFT); + + /* IRQ Output Enable */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_OUTPUT_EN, NAU8825_IRQ_OUTPUT_EN); + /* Mask unneeded IRQs: 1 - disable, 0 - enable */ + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, 0x7ff, 0x7ff); + regmap_update_bits(regmap, NAU8825_REG_INTERRUPT_MASK, + NAU8825_IRQ_HEADSET_COMPLETE_EN | NAU8825_IRQ_EJECT_EN, 0); + + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_VOLTAGE_MASK, nau8825->micbias_voltage); + + if (nau8825->sar_threshold_num) + nau8825_setup_buttons(nau8825); + + /* Default oversampling/decimations settings are unusable + * (audible hiss). Set it to something better. */ + regmap_update_bits(regmap, NAU8825_REG_ADC_RATE, + NAU8825_ADC_SYNC_DOWN_MASK, NAU8825_ADC_SYNC_DOWN_128); + regmap_update_bits(regmap, NAU8825_REG_DAC_CTRL1, + NAU8825_DAC_OVERSAMPLE_MASK, NAU8825_DAC_OVERSAMPLE_128); +} + +static const struct regmap_config nau8825_regmap_config = { + .val_bits = 16, + .reg_bits = 16, + + .max_register = NAU8825_REG_MAX, + .readable_reg = nau8825_readable_reg, + .writeable_reg = nau8825_writeable_reg, + .volatile_reg = nau8825_volatile_reg, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = nau8825_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(nau8825_reg_defaults), +}; + +static int nau8825_codec_probe(struct snd_soc_codec *codec) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + nau8825->dapm = &codec->dapm; + + /* The interrupt clock is gated by x1[10:8], + * one of them needs to be enabled all the time for + * interrupts to happen. */ + snd_soc_dapm_force_enable_pin(&codec->dapm, "DDACR"); + snd_soc_dapm_sync(&codec->dapm); + + /* Unmask interruptions. Handler uses dapm object so we can enable + * interruptions only after dapm is fully initialized. */ + regmap_write(nau8825->regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, 0); + nau8825_restart_jack_detection(nau8825->regmap); + + return 0; +} + +static int nau8825_configure_sysclk(struct nau8825 *nau8825, int clk_id, + unsigned int freq) +{ + struct regmap *regmap = nau8825->regmap; + + switch (clk_id) { + case NAU8825_CLK_MCLK: + regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_MCLK); + regmap_update_bits(regmap, NAU8825_REG_FLL6, NAU8825_DCO_EN, 0); + + /* We selected MCLK source but the clock itself managed externally */ + if (!nau8825->mclk) + break; + + if (!nau8825->mclk_freq) + clk_prepare_enable(nau8825->mclk); + + if (nau8825->mclk_freq != freq) { + nau8825->mclk_freq = freq; + + freq = clk_round_rate(nau8825->mclk, freq); + clk_set_rate(nau8825->mclk, freq); + } + + break; + case NAU8825_CLK_INTERNAL: + regmap_update_bits(regmap, NAU8825_REG_FLL6, NAU8825_DCO_EN, + NAU8825_DCO_EN); + regmap_update_bits(regmap, NAU8825_REG_CLK_DIVIDER, + NAU8825_CLK_SRC_MASK, NAU8825_CLK_SRC_VCO); + + if (nau8825->mclk_freq) { + clk_disable_unprepare(nau8825->mclk); + nau8825->mclk_freq = 0; + } + + break; + default: + dev_err(nau8825->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + + dev_dbg(nau8825->dev, "Sysclk is %dHz and clock id is %d\n", freq, + clk_id); + return 0; +} + +static int nau8825_set_sysclk(struct snd_soc_codec *codec, int clk_id, + int source, unsigned int freq, int dir) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + return nau8825_configure_sysclk(nau8825, clk_id, freq); +} + +static int nau8825_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + int ret; + + switch (level) { + case SND_SOC_BIAS_ON: + break; + + case SND_SOC_BIAS_PREPARE: + break; + + case SND_SOC_BIAS_STANDBY: + if (codec->dapm.bias_level == SND_SOC_BIAS_OFF) { + if (nau8825->mclk_freq) + clk_prepare_enable(nau8825->mclk); + + ret = regcache_sync(nau8825->regmap); + if (ret != 0) { + dev_err(codec->dev, + "Failed to sync cache: %d\n", ret); + return ret; + } + } + + break; + + case SND_SOC_BIAS_OFF: + if (nau8825->mclk_freq) + clk_disable_unprepare(nau8825->mclk); + + regcache_mark_dirty(nau8825->regmap); + break; + } + return 0; +} + +static struct snd_soc_codec_driver nau8825_codec_driver = { + .probe = nau8825_codec_probe, + .set_sysclk = nau8825_set_sysclk, + .set_bias_level = nau8825_set_bias_level, + .suspend_bias_off = true, + + .controls = nau8825_controls, + .num_controls = ARRAY_SIZE(nau8825_controls), + .dapm_widgets = nau8825_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8825_dapm_widgets), + .dapm_routes = nau8825_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8825_dapm_routes), +}; + +static void nau8825_reset_chip(struct regmap *regmap) +{ + regmap_write(regmap, NAU8825_REG_RESET, 0x00); + regmap_write(regmap, NAU8825_REG_RESET, 0x00); +} + +static int nau8825_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct nau8825 *nau8825; + int ret, value; + + nau8825 = devm_kzalloc(dev, sizeof(*nau8825), GFP_KERNEL); + if (!nau8825) + return -ENOMEM; + + i2c_set_clientdata(i2c, nau8825); + + nau8825->regmap = devm_regmap_init_i2c(i2c, &nau8825_regmap_config); + if (IS_ERR(nau8825->regmap)) + return PTR_ERR(nau8825->regmap); + nau8825->irq = i2c->irq; + nau8825->dev = dev; + + nau8825->jkdet_enable = device_property_read_bool(dev, + "nuvoton,jkdet-enable"); + nau8825->jkdet_pull_enable = device_property_read_bool(dev, + "nuvoton,jkdet-pull-enable"); + nau8825->jkdet_pull_up = device_property_read_bool(dev, + "nuvoton,jkdet-pull-up"); + device_property_read_u32(dev, "nuvoton,jkdet-polarity", + &nau8825->jkdet_polarity); + device_property_read_u32(dev, "nuvoton,micbias-voltage", + &nau8825->micbias_voltage); + device_property_read_u32(dev, "nuvoton,vref-impedance", + &nau8825->vref_impedance); + device_property_read_u32(dev, "nuvoton,sar-threshold-num", + &nau8825->sar_threshold_num); + device_property_read_u32_array(dev, "nuvoton,sar-threshold", + nau8825->sar_threshold, nau8825->sar_threshold_num); + device_property_read_u32(dev, "nuvoton,sar-hysteresis", + &nau8825->sar_hysteresis); + device_property_read_u32(dev, "nuvoton,sar-voltage", + &nau8825->sar_voltage); + device_property_read_u32(dev, "nuvoton,short-key-debounce", + &nau8825->key_debounce); + device_property_read_u32(dev, "nuvoton,jack-insert-debounce", + &nau8825->jack_insert_debounce); + device_property_read_u32(dev, "nuvoton,jack-eject-debounce", + &nau8825->jack_eject_debounce); + + + nau8825->mclk = devm_clk_get(dev, "mclk"); + if (PTR_ERR(nau8825->mclk) == -EPROBE_DEFER) { + return -EPROBE_DEFER; + } else if (PTR_ERR(nau8825->mclk) == -ENOENT) { + /* The MCLK is managed externally or not used at all */ + nau8825->mclk = NULL; + dev_info(dev, "No 'mclk' clock found, assume MCLK is managed externally"); + } else if (IS_ERR(nau8825->mclk)) { + return -EINVAL; + } + + nau8825_reset_chip(nau8825->regmap); + ret = regmap_read(nau8825->regmap, NAU8825_REG_I2C_DEVICE_ID, &value); + if (ret < 0) { + dev_err(dev, "Failed to read device id from the NAU8825: %d\n", + ret); + return ret; + } + if ((value & NAU8825_SOFTWARE_ID_MASK) != + NAU8825_SOFTWARE_ID_NAU8825) { + dev_err(dev, "Not a NAU8825 chip\n"); + return -ENODEV; + } + + nau8825_init_regs(nau8825); + + /* Enable internal VCO needed for interruptions */ + nau8825_configure_sysclk(nau8825, NAU8825_CLK_INTERNAL, 0); + + /* Enable DDACR needed for interrupts + * It is the same as force_enable_pin("DDACR") we do later */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_DACR, NAU8825_ENABLE_DACR); + + /* Chip needs one FSCLK cycle in order to generate interrupts, + * as we cannot guarantee one will be provided by the system. Turning + * master mode on then off enables us to generate that FSCLK cycle + * with a minimum of contention on the clock bus. + */ + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_MASTER); + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, NAU8825_I2S_MS_SLAVE); + + if (nau8825->irq) { + int ret = devm_request_threaded_irq(dev, nau8825->irq, NULL, + nau8825_interrupt, IRQF_TRIGGER_LOW | IRQF_ONESHOT, + "nau8825", nau8825); + + if (ret) { + dev_err(dev, "Cannot request irq %d (%d)\n", + nau8825->irq, ret); + return ret; + } + } + + return snd_soc_register_codec(&i2c->dev, &nau8825_codec_driver, + &nau8825_dai, 1); +} + +static int nau8825_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id nau8825_i2c_ids[] = { + { "nau8825", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, nau8825_i2c_ids); + +static struct i2c_driver nau8825_driver = { + .driver = { + .name = "nau8825", + .owner = THIS_MODULE, + }, + .probe = nau8825_i2c_probe, + .remove = nau8825_i2c_remove, + .id_table = nau8825_i2c_ids, +}; +module_i2c_driver(nau8825_driver); + +MODULE_DESCRIPTION("ASoC nau8825 driver"); +MODULE_AUTHOR("Anatol Pomozov <anatol(a)chromium.org>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h new file mode 100644 index 0000000..b84148b --- /dev/null +++ b/sound/soc/codecs/nau8825.h @@ -0,0 +1,317 @@ +/* + * NAU8825 ALSA SoC audio driver + * + * Copyright 2015 Google Inc. + * Author: Anatol Pomozov <anatol.pomozov(a)chrominium.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __NAU8825_H__ +#define __NAU8825_H__ + +#define NAU8825_REG_RESET 0x00 +#define NAU8825_REG_ENA_CTRL 0x01 +#define NAU8825_REG_CLK_DIVIDER 0x03 +#define NAU8825_REG_FLL1 0x04 +#define NAU8825_REG_FLL2 0x05 +#define NAU8825_REG_FLL3 0x06 +#define NAU8825_REG_FLL4 0x07 +#define NAU8825_REG_FLL5 0x08 +#define NAU8825_REG_FLL6 0x09 +#define NAU8825_REG_FLL_VCO_RSV 0x0a +#define NAU8825_REG_HSD_CTRL 0x0c +#define NAU8825_REG_JACK_DET_CTRL 0x0d +#define NAU8825_REG_INTERRUPT_MASK 0x0f +#define NAU8825_REG_IRQ_STATUS 0x10 +#define NAU8825_REG_INT_CLR_KEY_STATUS 0x11 +#define NAU8825_REG_INTERRUPT_DIS_CTRL 0x12 +#define NAU8825_REG_SAR_CTRL 0x13 +#define NAU8825_REG_KEYDET_CTRL 0x14 +#define NAU8825_REG_VDET_THRESHOLD_1 0x15 +#define NAU8825_REG_VDET_THRESHOLD_2 0x16 +#define NAU8825_REG_VDET_THRESHOLD_3 0x17 +#define NAU8825_REG_VDET_THRESHOLD_4 0x18 +#define NAU8825_REG_GPIO34_CTRL 0x19 +#define NAU8825_REG_GPIO12_CTRL 0x1a +#define NAU8825_REG_TDM_CTRL 0x1b +#define NAU8825_REG_I2S_PCM_CTRL1 0x1c +#define NAU8825_REG_I2S_PCM_CTRL2 0x1d +#define NAU8825_REG_LEFT_TIME_SLOT 0x1e +#define NAU8825_REG_RIGHT_TIME_SLOT 0x1f +#define NAU8825_REG_BIQ_CTRL 0x20 +#define NAU8825_REG_BIQ_COF1 0x21 +#define NAU8825_REG_BIQ_COF2 0x22 +#define NAU8825_REG_BIQ_COF3 0x23 +#define NAU8825_REG_BIQ_COF4 0x24 +#define NAU8825_REG_BIQ_COF5 0x25 +#define NAU8825_REG_BIQ_COF6 0x26 +#define NAU8825_REG_BIQ_COF7 0x27 +#define NAU8825_REG_BIQ_COF8 0x28 +#define NAU8825_REG_BIQ_COF9 0x29 +#define NAU8825_REG_BIQ_COF10 0x2a +#define NAU8825_REG_ADC_RATE 0x2b +#define NAU8825_REG_DAC_CTRL1 0x2c +#define NAU8825_REG_DAC_CTRL2 0x2d +#define NAU8825_REG_DAC_DGAIN_CTRL 0x2f +#define NAU8825_REG_ADC_DGAIN_CTRL 0x30 +#define NAU8825_REG_MUTE_CTRL 0x31 +#define NAU8825_REG_HSVOL_CTRL 0x32 +#define NAU8825_REG_DACL_CTRL 0x33 +#define NAU8825_REG_DACR_CTRL 0x34 +#define NAU8825_REG_ADC_DRC_KNEE_IP12 0x38 +#define NAU8825_REG_ADC_DRC_KNEE_IP34 0x39 +#define NAU8825_REG_ADC_DRC_SLOPES 0x3a +#define NAU8825_REG_ADC_DRC_ATKDCY 0x3b +#define NAU8825_REG_DAC_DRC_KNEE_IP12 0x45 +#define NAU8825_REG_DAC_DRC_KNEE_IP34 0x46 +#define NAU8825_REG_DAC_DRC_SLOPES 0x47 +#define NAU8825_REG_DAC_DRC_ATKDCY 0x48 +#define NAU8825_REG_IMM_MODE_CTRL 0x4c +#define NAU8825_REG_IMM_RMS_L 0x4d +#define NAU8825_REG_IMM_RMS_R 0x4e +#define NAU8825_REG_CLASSG_CTRL 0x50 +#define NAU8825_REG_OPT_EFUSE_CTRL 0x51 +#define NAU8825_REG_MISC_CTRL 0x55 +#define NAU8825_REG_I2C_DEVICE_ID 0x58 +#define NAU8825_REG_SARDOUT_RAM_STATUS 0x59 +#define NAU8825_REG_BIAS_ADJ 0x66 +#define NAU8825_REG_TRIM_SETTINGS 0x68 +#define NAU8825_REG_ANALOG_CONTROL_1 0x69 +#define NAU8825_REG_ANALOG_CONTROL_2 0x6a +#define NAU8825_REG_ANALOG_ADC_1 0x71 +#define NAU8825_REG_ANALOG_ADC_2 0x72 +#define NAU8825_REG_RDAC 0x73 +#define NAU8825_REG_MIC_BIAS 0x74 +#define NAU8825_REG_BOOST 0x76 +#define NAU8825_REG_FEPGA 0x77 +#define NAU8825_REG_POWER_UP_CONTROL 0x7f +#define NAU8825_REG_CHARGE_PUMP 0x80 +#define NAU8825_REG_CHARGE_PUMP_INPUT_READ 0x81 +#define NAU8825_REG_GENERAL_STATUS 0x82 +#define NAU8825_REG_MAX NAU8825_REG_GENERAL_STATUS + +/* ENA_CTRL (0x1) */ +#define NAU8825_ENABLE_DACR_SFT 10 +#define NAU8825_ENABLE_DACR (1 << NAU8825_ENABLE_DACR_SFT) +#define NAU8825_ENABLE_DACL_SFT 9 +#define NAU8825_ENABLE_ADC_SFT 8 +#define NAU8825_ENABLE_SAR_SFT 1 + +/* CLK_DIVIDER (0x3) */ +#define NAU8825_CLK_SRC_SFT 15 +#define NAU8825_CLK_SRC_MASK (1 << NAU8825_CLK_SRC_SFT) +#define NAU8825_CLK_SRC_VCO (1 << NAU8825_CLK_SRC_SFT) +#define NAU8825_CLK_SRC_MCLK (0 << NAU8825_CLK_SRC_SFT) + +/* FLL6 (0x9) */ +#define NAU8825_DCO_EN (1 << 15) + +/* HSD_CTRL (0xc) */ +#define NAU8825_HSD_AUTO_MODE (1 << 6) +/* 0 - short to GND, 1 - open */ +#define NAU8825_SPKR_DWN1R (1 << 1) +#define NAU8825_SPKR_DWN1L (1 << 0) + +/* JACK_DET_CTRL (0xd) */ +#define NAU8825_JACK_DET_RESTART (1 << 9) +#define NAU8825_JACK_INSERT_DEBOUNCE_SFT 5 +#define NAU8825_JACK_INSERT_DEBOUNCE_MASK (0x7 << NAU8825_JACK_INSERT_DEBOUNCE_SFT) +#define NAU8825_JACK_EJECT_DEBOUNCE_SFT 2 +#define NAU8825_JACK_EJECT_DEBOUNCE_MASK (0x7 << NAU8825_JACK_EJECT_DEBOUNCE_SFT) +#define NAU8825_JACK_POLARITY (1 << 1) /* 0 - active low, 1 - active high */ + +/* INTERRUPT_MASK (0xf) */ +#define NAU8825_IRQ_OUTPUT_EN (1 << 11) +#define NAU8825_IRQ_HEADSET_COMPLETE_EN (1 << 10) +#define NAU8825_IRQ_KEY_RELEASE_EN (1 << 7) +#define NAU8825_IRQ_KEY_SHORT_PRESS_EN (1 << 5) +#define NAU8825_IRQ_EJECT_EN (1 << 2) + +/* IRQ_STATUS (0x10) */ +#define NAU8825_HEADSET_COMPLETION_IRQ (1 << 10) +#define NAU8825_SHORT_CIRCUIT_IRQ (1 << 9) +#define NAU8825_IMPEDANCE_MEAS_IRQ (1 << 8) +#define NAU8825_KEY_IRQ_MASK (0x7 << 5) +#define NAU8825_KEY_RELEASE_IRQ (1 << 7) +#define NAU8825_KEY_LONG_PRESS_IRQ (1 << 6) +#define NAU8825_KEY_SHORT_PRESS_IRQ (1 << 5) +#define NAU8825_MIC_DETECTION_IRQ (1 << 4) +#define NAU8825_JACK_EJECTION_IRQ_MASK (3 << 2) +#define NAU8825_JACK_EJECTION_DETECTED (1 << 2) +#define NAU8825_JACK_INSERTION_IRQ_MASK (3 << 0) +#define NAU8825_JACK_INSERTION_DETECTED (1 << 0) + +/* INTERRUPT_DIS_CTRL (0x12) */ +#define NAU8825_IRQ_HEADSET_COMPLETE_DIS (1 << 10) +#define NAU8825_IRQ_KEY_RELEASE_DIS (1 << 7) +#define NAU8825_IRQ_KEY_SHORT_PRESS_DIS (1 << 5) +#define NAU8825_IRQ_EJECT_DIS (1 << 2) + +/* SAR_CTRL (0x13) */ +#define NAU8825_SAR_ADC_EN_SFT 12 +#define NAU8825_SAR_ADC_EN (1 << NAU8825_SAR_ADC_EN_SFT) +#define NAU8825_SAR_INPUT_MASK (1 << 11) +#define NAU8825_SAR_INPUT_JKSLV (1 << 11) +#define NAU8825_SAR_INPUT_JKR2 (0 << 11) +#define NAU8825_SAR_TRACKING_GAIN_SFT 8 +#define NAU8825_SAR_TRACKING_GAIN_MASK (0x7 << NAU8825_SAR_TRACKING_GAIN_SFT) + +/* KEYDET_CTRL (0x14) */ +#define NAU8825_KEYDET_SHORTKEY_DEBOUNCE_SFT 12 +#define NAU8825_KEYDET_SHORTKEY_DEBOUNCE_MASK (0x3 << NAU8825_KEYDET_SHORTKEY_DEBOUNCE_SFT) +#define NAU8825_KEYDET_LEVELS_NR_SFT 8 +#define NAU8825_KEYDET_LEVELS_NR_MASK (0x7 << 8) +#define NAU8825_KEYDET_HYSTERESIS_SFT 0 +#define NAU8825_KEYDET_HYSTERESIS_MASK 0xf + +/* GPIO12_CTRL (0x1a) */ +#define NAU8825_JKDET_PULL_UP (1 << 11) /* 0 - pull down, 1 - pull up */ +#define NAU8825_JKDET_PULL_EN (1 << 9) /* 0 - enable pull, 1 - disable */ +#define NAU8825_JKDET_OUTPUT_EN (1 << 8) /* 0 - enable niput, 1 - enable output */ + +/* I2S_PCM_CTRL1 (0x1c) */ +#define NAU8825_I2S_BP_SFT 7 +#define NAU8825_I2S_BP_MASK (1 << NAU8825_I2S_BP_SFT) +#define NAU8825_I2S_BP_INV (1 << NAU8825_I2S_BP_SFT) +#define NAU8825_I2S_PCMB_SFT 6 +#define NAU8825_I2S_PCMB_MASK (1 << NAU8825_I2S_PCMB_SFT) +#define NAU8825_I2S_PCMB_EN (1 << NAU8825_I2S_PCMB_SFT) +#define NAU8825_I2S_DL_SFT 2 +#define NAU8825_I2S_DL_MASK (0x3 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_16 (0 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_20 (1 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_24 (2 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_32 (3 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DF_SFT 0 +#define NAU8825_I2S_DF_MASK (0x3 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_RIGTH (0 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_LEFT (1 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_I2S (2 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_PCM_AB (3 << NAU8825_I2S_DF_SFT) + +/* I2S_PCM_CTRL2 (0x1d) */ +#define NAU8825_I2S_TRISTATE (1 << 15) /* 0 - Normal mode, 1 - Output Hi-Z */ +#define NAU8825_I2S_MS_SFT 3 +#define NAU8825_I2S_MS_MASK (1 << NAU8825_I2S_MS_SFT) +#define NAU8825_I2S_MS_MASTER (1 << NAU8825_I2S_MS_SFT) +#define NAU8825_I2S_MS_SLAVE (0 << NAU8825_I2S_MS_SFT) + +/* ADC_RATE (0x2b) */ +#define NAU8825_ADC_SYNC_DOWN_SFT 0 +#define NAU8825_ADC_SYNC_DOWN_MASK 0x3 +#define NAU8825_ADC_SYNC_DOWN_32 0 +#define NAU8825_ADC_SYNC_DOWN_64 1 +#define NAU8825_ADC_SYNC_DOWN_128 2 +#define NAU8825_ADC_SYNC_DOWN_256 3 + +/* DAC_CTRL1 (0x2c) */ +#define NAU8825_DAC_CLIP_OFF (1 << 7) +#define NAU8825_DAC_OVERSAMPLE_SFT 0 +#define NAU8825_DAC_OVERSAMPLE_MASK 0x7 +#define NAU8825_DAC_OVERSAMPLE_64 0 +#define NAU8825_DAC_OVERSAMPLE_256 1 +#define NAU8825_DAC_OVERSAMPLE_128 2 +#define NAU8825_DAC_OVERSAMPLE_32 4 + +/* MUTE_CTRL (0x31) */ +#define NAU8825_DAC_ZERO_CROSSING_EN (1 << 9) +#define NAU8825_DAC_SOFT_MUTE (1 << 9) + +/* HSVOL_CTRL (0x32) */ +#define NAU8825_HP_MUTE (1 << 15) + +/* DACL_CTRL (0x33) */ +#define NAU8825_DACL_CH_SEL_SFT 9 + +/* DACR_CTRL (0x34) */ +#define NAU8825_DACR_CH_SEL_SFT 9 + +/* I2C_DEVICE_ID (0x58) */ +#define NAU8825_GPIO2JD1 (1 << 7) +#define NAU8825_SOFTWARE_ID_MASK 0x3 +#define NAU8825_SOFTWARE_ID_NAU8825 0x0 + +/* BIAS_ADJ (0x66) */ +#define NAU8825_BIAS_VMID (1 << 6) +#define NAU8825_BIAS_VMID_SEL_SFT 4 +#define NAU8825_BIAS_VMID_SEL_MASK (3 << NAU8825_BIAS_VMID_SEL_SFT) + +/* ANALOG_CONTROL_2 (0x6a) */ +#define NAU8825_HP_NON_CLASSG_CURRENT_2xADJ (1 << 12) +#define NAU8825_DAC_CAPACITOR_MSB (1 << 1) +#define NAU8825_DAC_CAPACITOR_LSB (1 << 0) + +/* ANALOG_ADC_2 (0x72) */ +#define NAU8825_ADC_VREFSEL_MASK (0x3 << 8) +#define NAU8825_ADC_VREFSEL_ANALOG (0 << 8) +#define NAU8825_ADC_VREFSEL_VMID (1 << 8) +#define NAU8825_ADC_VREFSEL_VMID_PLUS_0_5DB (2 << 8) +#define NAU8825_ADC_VREFSEL_VMID_PLUS_1DB (3 << 8) +#define NAU8825_POWERUP_ADCL (1 << 6) + +/* MIC_BIAS (0x74) */ +#define NAU8825_MICBIAS_JKSLV (1 << 14) +#define NAU8825_MICBIAS_JKR2 (1 << 12) +#define NAU8825_MICBIAS_POWERUP_SFT 8 +#define NAU8825_MICBIAS_VOLTAGE_SFT 0 +#define NAU8825_MICBIAS_VOLTAGE_MASK 0x7 + +/* BOOST (0x76) */ +#define NAU8825_PRECHARGE_DIS (1 << 13) +#define NAU8825_GLOBAL_BIAS_EN (1 << 12) +#define NAU8825_HP_BOOST_G_DIS (1 << 8) +#define NAU8825_SHORT_SHUTDOWN_EN (1 << 6) + +/* POWER_UP_CONTROL (0x7f) */ +#define NAU8825_POWERUP_INTEGR_R (1 << 5) +#define NAU8825_POWERUP_INTEGR_L (1 << 4) +#define NAU8825_POWERUP_DRV_IN_R (1 << 3) +#define NAU8825_POWERUP_DRV_IN_L (1 << 2) +#define NAU8825_POWERUP_HP_DRV_R (1 << 1) +#define NAU8825_POWERUP_HP_DRV_L (1 << 0) + +/* CHARGE_PUMP (0x80) */ +#define NAU8825_JAMNODCLOW (1 << 10) +#define NAU8825_POWER_DOWN_DACR (1 << 9) +#define NAU8825_POWER_DOWN_DACL (1 << 8) +#define NAU8825_CHANRGE_PUMP_EN (1 << 5) + + +/* System Clock Source */ +enum { + NAU8825_CLK_MCLK = 0, + NAU8825_CLK_INTERNAL, +}; + +struct nau8825 { + struct device *dev; + struct regmap *regmap; + struct snd_soc_dapm_context *dapm; + int irq; + struct snd_soc_jack *jack; + struct clk *mclk; + int mclk_freq; /* 0 - mclk is disabled */ + int button_pressed; + int micbias_voltage; + int vref_impedance; + bool jkdet_enable; + bool jkdet_pull_enable; + bool jkdet_pull_up; + int jkdet_polarity; + int sar_threshold_num; + int sar_threshold[8]; + int sar_hysteresis; + int sar_voltage; + int key_debounce; + int jack_insert_debounce; + int jack_eject_debounce; +}; + +int nau8825_enable_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack); + + +#endif /* __NAU8825_H__ */ -- 2.5.0.457.gab17608

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[alsa-devel] [PATCH] ASoC: davinci-mcasp: Revise the FIFO threshold calculation
by Peter Ujfalusi 14 Sep '15

14 Sep '15

The FIFO threshold for McASP should be <=[tx/rx]numevt so the initial value for the refining should meet this requirement as well. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/davinci/davinci-mcasp.c | 7 +++---- 1 file changed, 3 insertions(+), 4 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index add6bb99661d..1260b315a96c 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -663,7 +663,7 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, u8 rx_ser = 0; u8 slots = mcasp->tdm_slots; u8 max_active_serializers = (channels + slots - 1) / slots; - int active_serializers, numevt, n; + int active_serializers, numevt; u32 reg; /* Default configuration */ if (mcasp->version < MCASP_VERSION_3) @@ -745,9 +745,8 @@ static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream, * The number of words for numevt need to be in steps of active * serializers. */ - n = numevt % active_serializers; - if (n) - numevt += (active_serializers - n); + numevt = (numevt / active_serializers) * active_serializers; + while (period_words % numevt && numevt > 0) numevt -= active_serializers; if (numevt <= 0) -- 2.5.2

1 0

[alsa-devel] [PATCH] ALSA: pcm: remove structure member of 'struct snd_pcm_hwptr_log *' type because this structure had been removed
by Takashi Sakamoto 13 Sep '15

13 Sep '15

This structure was added by 4d96eb255c53 ('ALSA: pcm_lib - add possibility to log last 10 DMA ring buffer positions') to store PCM pointers information of latest 10 pointer movements (=XRUN_LOG_CNT). When CONFIG_SND_PCM_XRUN_DEBUG is configured, 'struct snd_pcm_runtime' has 'hwptr_log' member with a pointer to the structure. When calling xrun_log() in pcm_lib.c, the structure was allocated to the pointer. When calling snd_pcm_detach_substream() in pcm.c, the allocated pointer is released. In f5914908a5b7 ('ALSA: pcm: Replace PCM hwptr tracking with tracepoints'), the pointer logging is replaced with using Linux Kernel Tracepoints. The structure was also removed, while it's just declared. The member and kfree still remains. This commit removes the member and related codes. I think this was overlooked because it brings no errors/warnings to C compilers. Fixes: f5914908a5b7 ('ALSA: pcm: Replace PCM hwptr tracking with tracepoints') Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- include/sound/pcm.h | 6 ------ sound/core/pcm.c | 3 --- 2 files changed, 9 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index 691e7ee..a4fcc94 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -285,8 +285,6 @@ struct snd_pcm_hw_constraint_ranges { unsigned int mask; }; -struct snd_pcm_hwptr_log; - /* * userspace-provided audio timestamp config to kernel, * structure is for internal use only and filled with dedicated unpack routine @@ -428,10 +426,6 @@ struct snd_pcm_runtime { /* -- OSS things -- */ struct snd_pcm_oss_runtime oss; #endif - -#ifdef CONFIG_SND_PCM_XRUN_DEBUG - struct snd_pcm_hwptr_log *hwptr_log; -#endif }; struct snd_pcm_group { /* keep linked substreams */ diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 02bd969..308c9ec 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -1014,9 +1014,6 @@ void snd_pcm_detach_substream(struct snd_pcm_substream *substream) snd_free_pages((void*)runtime->control, PAGE_ALIGN(sizeof(struct snd_pcm_mmap_control))); kfree(runtime->hw_constraints.rules); -#ifdef CONFIG_SND_PCM_XRUN_DEBUG - kfree(runtime->hwptr_log); -#endif kfree(runtime); substream->runtime = NULL; put_pid(substream->pid); -- 2.1.4

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Re: [alsa-devel] [Alsa-user] 'control ... is already present' message
by Ran Shalit 12 Sep '15

12 Sep '15

On Fri, Sep 11, 2015 at 9:42 AM, Clemens Ladisch <cladisch(a)googlemail.com> wrote: > Ran Shalit wrote: >> asoc: failed to add dapm kcontrol Right HPCOM Mixer Line2L Bypass Switch: -16 >> asoc: failed to add kcontrol Right Line2R Mux >> ... >> What is the reason for the "asoc: failed to add dapm kcontrol" and the >> " 'control ... already present" messeges ? > > That there are two controls with the same name and index. > >> If I move each to be registered to its own card will it resolve this issue ? > > Probably, but a better idea might be to set index=1 for all controls of the > second codec. I don't know how ASoC handles this; ask on the alsa-devel > list. > > > Regards, > Clemens Hi Clemens, Can you please add how to set index=1 to second codec ? Thank you, Ran

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[alsa-devel] What is exact definition of "normal/inverted" FSYNC signal polarity?
by Anatol Pomozov 12 Sep '15

12 Sep '15

Hi I am trying to configure TegraX1 SoC and Realtek5677 codec. These two drivers have different meaning of FSYNC polarity in TDM mode A format. * TegraX1 driver thinks that "normal" polarity is when a frame starts with FSYNC falling edge. TegraX1 applies it to all formats (I2S, PCM/TDM mode A). And inverted FSYNC means that frame starts with rising edge of FSYNC. * RT5677 thinks that "normal" for I2S is "frame starts with falling edge". And for TMD mode A "normal" means "frame starts with rising FSYNC edge". Thus normal/inverted FSYNC in TDM mode A are opposite at these 2 chips. Which one is correct? What is exact definition of SND_SOC_DAIFMT_xB_yF constants in soc-dai.h? I tried to google if I2S spec [1] defines FSYNC polarity but no luck. It worth to clarify definition of BCLK polarity as well. I believe "normal" BCLK is "line sensing happens at rising edge", "inverted" BCLK - "line sensing happens at falling edge". Is it correct? [1] https://web.archive.org/web/20060821114007/http://www.semiconductors.philip…

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[alsa-devel] [PATCH] ALSA: hdsp: fix memory leak
by Sudip Mukherjee 11 Sep '15

11 Sep '15

If the size of the firmware is less than expected size then we are exiting with the error code but we missed releasing the firmware. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/pci/rme9652/hdsp.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/rme9652/hdsp.c b/sound/pci/rme9652/hdsp.c index 9bba275..2875b4f 100644 --- a/sound/pci/rme9652/hdsp.c +++ b/sound/pci/rme9652/hdsp.c @@ -5112,6 +5112,7 @@ static int hdsp_request_fw_loader(struct hdsp *hdsp) dev_err(hdsp->card->dev, "too short firmware size %d (expected %d)\n", (int)fw->size, HDSP_FIRMWARE_SIZE); + release_firmware(fw); return -EINVAL; } -- 1.9.1

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[alsa-devel] [PATCH v4 0/10] Fix broken DMAFLUSHP on Rockchips platform
by Shawn Lin 11 Sep '15

11 Sep '15

The purpose of the DMAFLUSHP instruction: - Tell the peripheral to clear its status and control registers. - Send a message to the peripheral to resend its level status. There are 3 timings described in PL330 Technical Reference Manual: - Timing 1: Burst request, can work well without DMAFLUSHP. - Timing 2: Single and burst request, DMAC will ignore the single transfer request. This timing happens if there are single and burst request. - Timing 3: Single transfers for a burst request, DMAC should signals datype to request the peripheral to flush the contents of any control registers. This timing happens if there is not enough MFIFO to places the burst data. A peripheral may signal a DMA request during the execution of DMAFLUSHP instruction, that cause DMA request being ignored by DMAC. But DMAC and all peripherals on RK3X SoCs DO NOT support DMAFLUSHP. It can't send a message to the peripheral to resend DMA request, and the peripheral can't acknowledge a flush request from DMAC. So all DMA requests should NOT be ignored by DMAC, and DMAC will not notify the peripheral to flush. To fix this problem, we need: - Do NOT execute DMAFLUSHP instruction. - Timing 2 and timing 3 should not happen. Because on RK3X SoCs, there are 6 or below channels and 32 MFIFO depth for DMAC_BUS, and 8 channels and 64 MFIFO depth for DMAC_PERI, it is impossible to hit the timing 3 if burst length is equal or less than 4. Since the request type signal by the peripheral can only be set by software. We can set Rockchip Soc's GRF_PERIDMAC_CON0[2:1] to select single or burst request, if it is set b01, all of the peripharals will signal a brust request. So the timing 2 will not happen, too. So DMAC on RK3X can support single or burst transfer, but can't support mixed transfer. Because burst transfer is more efficient than single transfer, this is confirmed by our ASIC team, who strongly suggest to use burst transfer. And this is confirmed by Addy's test on RK3288-Pink2 board, the speed of spi flash burst transfer will increase about two times than single transfer. Also, I have tested dw_mmc with pl330 on RK3188 platform to double confirm the result. That means burst transfer is reansonable. So we need a quirk not to execute DMAFLUSHP instruction and to use burst transfer. Note: - The Rockchip Soc default value of GRF_PERIDMAC_CON0[2:1] is b01. To support brust transfer, these bits should not be changed in bootloader. Changes in v4: - remove spi & i2s dts changes and query quirk from dmaengine API suggeseted by Mark Brown. - fix typo - Add dmaengine_get_quirk hook and implement it for pl330 Changes in v3: - add more rockchip drivers' changes in this patchset - add Reviewed-by: Sonny Rao <sonnyrao(a)chromium.org> Changes in v2: - amend the author - reorder the patches suggested by Doug - add Reviewed-by: Doug Anderson <dianders(a)chromium.org> for rk3288.dtsi patch and arm-pl330.txt patch - amend Olof's mail address Changes in v1: - rename broken-no-flushp to "arm,pl330-broken-no-flushp" suggested by Krzysztof. - add From original author. - remove Sunny's tag Addy Ke (3): DMA: pl330: add quirk for broken no flushp ARM: dts: Add arm,pl330-broken-no-flushp quirk for rk3288 platform spi: rockchip: modify DMA max burst to 1 Boojin Kim (1): DMA: pl330: support burst mode for dev-to-mem and mem-to-dev transmit Shawn Lin (5): Documentation: arm-pl330: add description of arm,pl330-broken-no-flushp ARM: dts: Add arm,pl330-broken-no-flushp quirk for rk3xxx platform dmaengine: add API for getting dma controller's quirk DMA: pl330: implement dmaengine_get_quirks hook snd: dmaengine-pcm: add snd_dmaengine_pcm_get_quirks interface Yiwei Cai (1): ASoC: rockchip_i2s: modify DMA max burst to 1 .../devicetree/bindings/dma/arm-pl330.txt | 1 + arch/arm/boot/dts/rk3288.dtsi | 3 + arch/arm/boot/dts/rk3xxx.dtsi | 3 + drivers/dma/pl330.c | 110 +++++++++++++++------ drivers/spi/spi-rockchip.c | 13 ++- include/linux/dmaengine.h | 9 ++ sound/soc/rockchip/rockchip_i2s.c | 16 ++- sound/soc/soc-generic-dmaengine-pcm.c | 24 +++++ 8 files changed, 147 insertions(+), 32 deletions(-) -- 2.3.7

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[alsa-devel] [PATCH] ASoC: Intel: Bsw: Read sst DSP DMA pointer after period elapse occurs
by yang.a.fang＠intel.com 11 Sep '15

11 Sep '15

From: "Fang, Yang A" <yang.a.fang(a)intel.com> We should not read sst hw_ptr and pcm_delay in sst_platform_pcm_pointer Since it will be ricky if sst_platform_pcm_pointer is called while dsp is updating the timestamp.Now read sst hw_ptr after period elapse interrupt occurs. Signed-off-by: Fang, Yang A <yang.a.fang(a)intel.com> Acked-by: Vinod Koul <vinod.koul(a)intel.com> --- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 18 +++++++++++------- 1 file changed, 11 insertions(+), 7 deletions(-) diff --git a/sound/soc/intel/atom/sst-mfld-platform-pcm.c b/sound/soc/intel/atom/sst-mfld-platform-pcm.c index 683e501..d58230b 100644 --- a/sound/soc/intel/atom/sst-mfld-platform-pcm.c +++ b/sound/soc/intel/atom/sst-mfld-platform-pcm.c @@ -263,17 +263,27 @@ static int sst_platform_alloc_stream(struct snd_pcm_substream *substream, static void sst_period_elapsed(void *arg) { struct snd_pcm_substream *substream = arg; + struct snd_soc_pcm_runtime *rtd; struct sst_runtime_stream *stream; + struct pcm_stream_info *str_info; int status; + int ret_val; if (!substream || !substream->runtime) return; + rtd = substream->private_data; stream = substream->runtime->private_data; if (!stream) return; status = sst_get_stream_status(stream); if (status != SST_PLATFORM_RUNNING) return; + str_info = &stream->stream_info; + ret_val = stream->ops->stream_read_tstamp(sst->dev, str_info); + if (ret_val) { + dev_err(rtd->dev, "sst: err code = %d\n", ret_val); + return; + } snd_pcm_period_elapsed(substream); } @@ -660,20 +670,14 @@ static snd_pcm_uframes_t sst_platform_pcm_pointer (struct snd_pcm_substream *substream) { struct sst_runtime_stream *stream; - int ret_val, status; + int status; struct pcm_stream_info *str_info; - struct snd_soc_pcm_runtime *rtd = substream->private_data; stream = substream->runtime->private_data; status = sst_get_stream_status(stream); if (status == SST_PLATFORM_INIT) return 0; str_info = &stream->stream_info; - ret_val = stream->ops->stream_read_tstamp(sst->dev, str_info); - if (ret_val) { - dev_err(rtd->dev, "sst: error code = %d\n", ret_val); - return ret_val; - } substream->runtime->delay = str_info->pcm_delay; return str_info->buffer_ptr; } -- 1.7.9.5

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Re: [alsa-devel] [PATCH 1/1] sound/usb: Add Asus Xonar U7 Mixer - input support
by Johan Aires Rastén 11 Sep '15

11 Sep '15

Hi Piotr, great to see someone else also working on getting the U7 working! Unfortunately I don't have the answer you're looking for as I don't know that much about ALSA actually. Most of what I've done was just comparing the UAC2 specification with what the driver does, and some light USB protocol sniffing. I'm not actively working on the driver since I bought a new sound card for my PC and am using the U7 with my Windows laptop, but send me a message on Google Hangouts if you think I can help with something. When I connect the U7 it starts with "mic" as input, but I have to switch it to line and back to mic before it works. Does this happen to you too? //Johan On 10 September 2015 at 16:47, Piotr SQ9FK <sq9fk(a)wp.pl> wrote: > Hello Takashi,Johan > My patches are unessesary since following last patch was applied to the git: > ALSA: usb-audio: Change internal PCM order > > Capture working but there are no volume sliders for Line-In Capture or > Mic-In Capture (I can only toggle on off). Probably the RANGES are > broken?. From ASUS Windows driver I can see following ranges: > > Master Volume: (Phisycal Knob on Asus) > from 100 to 1 correspond 0dB to -70dB, 0 is -127dB > Range from 1 to 100 -> 0dB to -70dB (logarythmic) > > Mic-In: > from 100 to 1 correspond 30dB to -10dB, 0 is -192dB > Range from 1 to 10 -> -10dB to 10dB (linear?) > Range from 11 to 100 -> 10dB to 30dB (logarythmic) > > Line-In: > from 100 to 1 correspond 0dB to -40dB, 0 is -192dB > Range from 11 to 100 -> 10dB to 30dB (logarythmic) > > Actual Situation: > Input Gain Pad Control - 3 steps [0 - 50 - 100] > Double Slider. In Capture and Playback Categories, both of them are > controlled in the same time. > Controls Playback Volume only, Capture not. > > "Mic" and "Line" Switches [on/off] > PCM Cature Source [Line/Mic] > Electret Mic doesn't work. With all possible configurations - Silent. > When I pluged Line level signal, I can record the signal without problems. > PCM Capture Source [Line] - Line [on] Mic [off] - Sound > PCM Capture Source [Line] - Line [on] Mic [on] - Sound > PCM Capture Source [Line] - Line [off] Mic [off] - No Sound > PCM Capture Source [Line] - Line [off] Mic [on] - No Sound > The same with selected [Mic]. There is no Capture Volume control. > > PCM 1 [mute/on] > I think this is SPDIF switch. It doesn't work. When I switch it on -> > Blue Led on Xonar doesn't light up. > > I attached screen of alsamixer, usbmix from proc and my mixer_map from > my old patch. > I think there are is still bugs in parsing. Please point me how I can > add manually Volume control for Capture? > > Piotr Gaska > > W dniu 2015-07-27 o 14:10, Takashi Iwai pisze: > >> On Sun, 26 Jul 2015 10:00:12 +0200, >> Piotr G. wrote: >>> >>> Hello, >>> This is my first patch to alsa-devel, and I'm not very skilled >>> programmer. From lsusb -vvv, I've draw Xonar U7 map with all IT OT FU CS >>> in mixer_map.c I don't know why the driver by default doesn't parse all >>> info correctly. With this patch you can see Line Switch and Mic Switch >>> in alsamixer. I can record my voice in Audacity with all supported bit >>> and sampling rates in Mono or Stereo >>> This is temporary solution. Please correct me if the patch is not >>> properly made. >>> >>> What doesn't work: >>> - Cannot change Capture Volume for Line/Mic. No mixer control in >>> alsamixer >>> - Switching between Line/Mic doesn't work (Mute doesn't work) >>> - SPDIF not tested >>> As what I wrote at beginning I'm not skilled programmer so any help >>> is >>> aprreciated >>> >>> >>> proc-usbmixer -> cat /proc/asound/*/usbmix >>> usb.debug -> lsusb -vv (only Asus Xonar U7) >> >> The changes in mixer_maps.c look good, but what's the reason for >> uac2_audio_feature_info[]? It seems adding "Control" suffix to each >> entry, which doesn't look correct. >> >> >> thanks, >> >> Takashi >> >> >>> --- >>> sound/usb/mixer.c | 23 +++++++++++++++++++++-- >>> sound/usb/mixer_maps.c | 48 >>> ++++++++++++++++++++++++++++++++++++++++++++++++ >>> 2 files changed, 69 insertions(+), 2 deletions(-) >>> >>> diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c >>> index 6b3acba..027d160 100644 >>> --- a/sound/usb/mixer.c >>> +++ b/sound/usb/mixer.c >>> @@ -818,6 +818,25 @@ static struct usb_feature_control_info >>> audio_feature_info[] = { >>> { "Phase Inverter Control", USB_MIXER_BOOLEAN }, >>> }; >>> >>> +static struct usb_feature_control_info uac2_audio_feature_info[] = { >>> + /* UAC2 specific */ >>> + { "Mute Control", USB_MIXER_INV_BOOLEAN }, >>> + { "Volume Control", USB_MIXER_S16 }, >>> + { "Bass Control", USB_MIXER_S8 }, >>> + { "Mid Control", USB_MIXER_S8 }, >>> + { "Treble Control", USB_MIXER_S8 }, >>> + { "Graphic Equalizer Control", USB_MIXER_S8 }, >>> + { "Automatic Gain Control", USB_MIXER_BOOLEAN }, >>> + { "Delay Control", USB_MIXER_S16 }, /* FIXME: U32 in >>> UAC2 */ >>> + { "Bass Boost Control", USB_MIXER_BOOLEAN }, >>> + { "Loudness Control", USB_MIXER_BOOLEAN }, >>> + { "Input Gain Control", USB_MIXER_S16 }, >>> + { "Input Gain Pad Control", USB_MIXER_S16 }, >>> + { "Phase Inverter Control", USB_MIXER_BOOLEAN }, >>> + { "Underflow Control", USB_MIXER_BOOLEAN }, >>> + { "Overflow Control", USB_MIXER_BOOLEAN }, >>> +}; >>> + >>> /* private_free callback */ >>> void snd_usb_mixer_elem_free(struct snd_kcontrol *kctl) >>> { >>> @@ -1240,7 +1259,7 @@ static void build_feature_ctl(struct mixer_build >>> *state, void *raw_desc, >>> snd_usb_mixer_elem_init_std(&cval->head, state->mixer, unitid); >>> cval->control = control; >>> cval->cmask = ctl_mask; >>> - cval->val_type = audio_feature_info[control-1].type; >>> + cval->val_type = uac2_audio_feature_info[control-1].type; >>> if (ctl_mask == 0) { >>> cval->channels = 1; /* master channel */ >>> cval->master_readonly = readonly_mask; >>> @@ -1318,7 +1337,7 @@ static void build_feature_ctl(struct mixer_build >>> *state, void *raw_desc, >>> break; >>> default: >>> if (!len) >>> - strlcpy(kctl->id.name, >>> audio_feature_info[control-1].name, >>> + strlcpy(kctl->id.name, >>> uac2_audio_feature_info[control-1].name, >>> sizeof(kctl->id.name)); >>> break; >>> } >>> diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c >>> index e5000da..07f510e 100644 >>> --- a/sound/usb/mixer_maps.c >>> +++ b/sound/usb/mixer_maps.c >>> @@ -341,6 +341,48 @@ static const struct usbmix_name_map >>> scms_usb3318_map[] = { >>> { 0 } >>> }; >>> >>> +/* ASUS Xonar U7 - USB CM6632A (ADC CS5361 // DAC Dolby CS4362A // DAC >>> Stereo CS4398 */ >>> +/* Topology: >>> + >>> +USB_IN[1]------->FU[13]-----------CS[18]--->Spk_OUT[7] >>> + >>> +Dig_IN[2]------->FU[14]-----------CS[19]--->Dig_OUT[8] >>> + >>> +Mic_IN[4]------->FU[16]---+ >>> + SU[20]--CS[22]--->USB_OUT[11] >>> +Line_IN[5]------>FU[17]---+ >>> + >>> +*/ >>> + >>> +static const struct usbmix_name_map xonar_u7_map[] = { >>> + /* 1: IT PCM Analog (USB Streaming) */ >>> + /* 2: IT PCM2 Digital (USB Streaming) */ >>> + /* 4: IT Microphone */ >>> + /* 5: IT Line IN */ >>> + /* 7: OT Speaker */ >>> + /* 8: OT SPDIF */ >>> + /* 11 OT Capture Playback (USB Streaming) */ >>> + { 13, "Speaker Playback" }, /* 13 FU Speaker Out Mute/Volume >>> */ >>> + { 14, "IEC958 Playback" }, /* 14 FU SPDIF Out Mute/Volume */ >>> + { 16, "Mic Capture" }, /* 16 FU Capture Mic-in >>> Mute/Volume */ >>> + { 17, "Line Capture" }, /* 17 FU Capture Line-in >>> Mute/Volume */ >>> + /* 18 CS for 1 PCM (USB Streaming) Clock Source */ >>> + /* 19 CS for 2 SPDIF (USB Streaming) Clock Source */ >>> + { 20, "Capture Source" }, /* 20 SU Capture Source Selector >>> */ >>> + /* 22 CS Line-in and Mic-in Clock Source */ >>> + { 0 } >>> +}; >>> + >>> +static struct usbmix_selector_map xonar_u7_selectors[] = { >>> +{ >>> + .id = 20, >>> + .count = 2, >>> + .names = (const char*[]) {"Mic", "Line"} >>> + }, >>> + { 0 } /* terminator */ >>> + >>> +}; >>> + >>> /* >>> * Control map entries >>> */ >>> @@ -451,6 +493,12 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { >>> .id = USB_ID(0x25c4, 0x0003), >>> .map = scms_usb3318_map, >>> }, >>> + { >>> + /* Asus Xonar U7 */ >>> + .id = USB_ID(0x1043, 0x857c), >>> + .map = xonar_u7_map, >>> + .selector_map = xonar_u7_selectors, >>> + }, >>> { 0 } /* terminator */ >>> }; >>> >>> -- >>> 2.3.6 >>> >

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[alsa-devel] Mixer control appears twice in alsamixer
by James Cameron 11 Sep '15

11 Sep '15

G'day, I've a mixer control for "Analog Mic Boost" which appears twice in alsamixer --view all, and once in amixer. How do I cause it to appear only once in alsamixer? amixer shows both Playback and Capture channels for the control: -- Simple mixer control 'Analog Mic Boost',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 8 Mono: 8 [100%] -- The control is declared with: /* {0, +20, +24, +30, +35, +40, +44, +50, +52}dB */ static unsigned int mic_bst_tlv[] = { TLV_DB_RANGE_HEAD(7), 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0), 3, 5, TLV_DB_SCALE_ITEM(3000, 500, 0), 6, 6, TLV_DB_SCALE_ITEM(4400, 0, 0), 7, 7, TLV_DB_SCALE_ITEM(5000, 0, 0), 8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0), }; static const struct snd_kcontrol_new rt5631_snd_controls[] = { ... SOC_SINGLE_TLV("Analog Mic Boost", RT5631_MIC_CTRL_2, RT5631_MIC1_BOOST_SHIFT, 8, 0, mic_bst_tlv), ... } I'm working on alc5631/rt5631 in a local branch of either 3.0 or 3.5 for the OLPC XO-1.75 and XO-4. But the declaration seems no different in 4.2, apart from the name of the control. https://git.kernel.org/cgit/linux/kernel/git/torvalds/linux.git/tree/sound/… https://git.kernel.org/cgit/linux/kernel/git/torvalds/linux.git/tree/sound/… -- James Cameron http://quozl.linux.org.au/

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[alsa-devel] 'control ... is already present' message
by Ran Shalit 11 Sep '15

11 Sep '15

Hello, I'm using 2 codecs (tlv320aic3x-codec) each which both are registered into same sound card, each as a single device. After boot I have the following devices: root@dm814x:~# cat /proc/asound/devices 0: [ 0] : control 16: [ 0- 0]: digital audio playback 17: [ 0- 1]: digital audio playback 24: [ 0- 0]: digital audio capture 25: [ 0- 1]: digital audio capture But I get the following error messeges during boot: usbcore: registered new interface driver snd-usb-audio Registered tvp5158 audio codec PM: Adding info for platform:soc-audio.0 PM: Adding info for No Bus:TLV320AIC3X_2 asoc: tlv320aic3x-hifi <-> davinci-mcasp.2 mapping ok snd_soc_instantiate_card 7 tlv320aic3x-codec 2-0019: tlv320aic3x-codec.2-0019: Failed to add PCM Playback Volume: -16 asoc: failed to add dapm kcontrol Right HPCOM Mixer Line2L Bypass Switch: -16 asoc: failed to add dapm kcontrol Left HPCOM Mixer Line2L Bypass Switch: -16 asoc: failed to add dapm kcontrol Right HP Mixer Line2L Bypass Switch: -16 asoc: failed to add dapm kcontrol Left HP Mixer Line2L Bypass Switch: -16 asoc: failed to add dapm kcontrol Mono Mixer Line2L Bypass Switch: -16 asoc: failed to add dapm kcontrol Right Line Mixer Line2L Bypass Switch: -16 asoc: failed to add dapm kcontrol Left Line Mixer Line2L Bypass Switch: -16 asoc: failed to add kcontrol Right Line2R Mux asoc: failed to add kcontrol Right Line1R Mux asoc: failed to add kcontrol Right Line1L Mux asoc: failed to add dapm kcontrol Right PGA Mixer Line1R Switch: -16 asoc: failed to add kcontrol Left Line2L Mux asoc: failed to add kcontrol Left Line1R Mux asoc: failed to add kcontrol Left Line1L Mux asoc: failed to add dapm kcontrol Left PGA Mixer Line1L Switch: -16 asoc: failed to add kcontrol Right HPCOM Mux asoc: failed to add kcontrol Right DAC Mux asoc: failed to add kcontrol Left HPCOM Mux asoc: failed to add kcontrol Left DAC Mux PM: Adding info for No Bus:TLV320AIC3X_3 asoc: tlv320aic3x-hifi <-> davinci-mcasp.3 mapping ok PM: Adding info for No Bus:card0 PM: Adding info for No Bus:pcmC0D1p PM: Adding info for No Bus:pcmC0D1c PM: Adding info for No Bus:pcmC0D0p PM: Adding info for No Bus:pcmC0D0c PM: Adding info for No Bus:controlC0 ALSA device list: #0: TI81XX SOUND0 What is the reason for the "asoc: failed to add dapm kcontrol" and the " 'control ... already present" messeges ? Is it becuase both codec are registered on same card (single card) ? 1. If I move each to be registered to its own card, will it resolve this issue ? 2. Should the following patch resolve this issue: http://mailman.alsa-project.org/pipermail/alsa-devel/2010-November/033593.h… Regards, Ran

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[alsa-devel] [PATCH 1/1] sound/usb: Add Asus Xonar U7 Mixer - input support
by Piotr G. 10 Sep '15

10 Sep '15

Hello, This is my first patch to alsa-devel, and I'm not very skilled programmer. From lsusb -vvv, I've draw Xonar U7 map with all IT OT FU CS in mixer_map.c I don't know why the driver by default doesn't parse all info correctly. With this patch you can see Line Switch and Mic Switch in alsamixer. I can record my voice in Audacity with all supported bit and sampling rates in Mono or Stereo This is temporary solution. Please correct me if the patch is not properly made. What doesn't work: - Cannot change Capture Volume for Line/Mic. No mixer control in alsamixer - Switching between Line/Mic doesn't work (Mute doesn't work) - SPDIF not tested As what I wrote at beginning I'm not skilled programmer so any help is aprreciated proc-usbmixer -> cat /proc/asound/*/usbmix usb.debug -> lsusb -vv (only Asus Xonar U7) --- sound/usb/mixer.c | 23 +++++++++++++++++++++-- sound/usb/mixer_maps.c | 48 ++++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 69 insertions(+), 2 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 6b3acba..027d160 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -818,6 +818,25 @@ static struct usb_feature_control_info audio_feature_info[] = { { "Phase Inverter Control", USB_MIXER_BOOLEAN }, }; +static struct usb_feature_control_info uac2_audio_feature_info[] = { + /* UAC2 specific */ + { "Mute Control", USB_MIXER_INV_BOOLEAN }, + { "Volume Control", USB_MIXER_S16 }, + { "Bass Control", USB_MIXER_S8 }, + { "Mid Control", USB_MIXER_S8 }, + { "Treble Control", USB_MIXER_S8 }, + { "Graphic Equalizer Control", USB_MIXER_S8 }, + { "Automatic Gain Control", USB_MIXER_BOOLEAN }, + { "Delay Control", USB_MIXER_S16 }, /* FIXME: U32 in UAC2 */ + { "Bass Boost Control", USB_MIXER_BOOLEAN }, + { "Loudness Control", USB_MIXER_BOOLEAN }, + { "Input Gain Control", USB_MIXER_S16 }, + { "Input Gain Pad Control", USB_MIXER_S16 }, + { "Phase Inverter Control", USB_MIXER_BOOLEAN }, + { "Underflow Control", USB_MIXER_BOOLEAN }, + { "Overflow Control", USB_MIXER_BOOLEAN }, +}; + /* private_free callback */ void snd_usb_mixer_elem_free(struct snd_kcontrol *kctl) { @@ -1240,7 +1259,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, snd_usb_mixer_elem_init_std(&cval->head, state->mixer, unitid); cval->control = control; cval->cmask = ctl_mask; - cval->val_type = audio_feature_info[control-1].type; + cval->val_type = uac2_audio_feature_info[control-1].type; if (ctl_mask == 0) { cval->channels = 1; /* master channel */ cval->master_readonly = readonly_mask; @@ -1318,7 +1337,7 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, break; default: if (!len) - strlcpy(kctl->id.name, audio_feature_info[control-1].name, + strlcpy(kctl->id.name, uac2_audio_feature_info[control-1].name, sizeof(kctl->id.name)); break; } diff --git a/sound/usb/mixer_maps.c b/sound/usb/mixer_maps.c index e5000da..07f510e 100644 --- a/sound/usb/mixer_maps.c +++ b/sound/usb/mixer_maps.c @@ -341,6 +341,48 @@ static const struct usbmix_name_map scms_usb3318_map[] = { { 0 } }; +/* ASUS Xonar U7 - USB CM6632A (ADC CS5361 // DAC Dolby CS4362A // DAC Stereo CS4398 */ +/* Topology: + +USB_IN[1]------->FU[13]-----------CS[18]--->Spk_OUT[7] + +Dig_IN[2]------->FU[14]-----------CS[19]--->Dig_OUT[8] + +Mic_IN[4]------->FU[16]---+ + SU[20]--CS[22]--->USB_OUT[11] +Line_IN[5]------>FU[17]---+ + +*/ + +static const struct usbmix_name_map xonar_u7_map[] = { + /* 1: IT PCM Analog (USB Streaming) */ + /* 2: IT PCM2 Digital (USB Streaming) */ + /* 4: IT Microphone */ + /* 5: IT Line IN */ + /* 7: OT Speaker */ + /* 8: OT SPDIF */ + /* 11 OT Capture Playback (USB Streaming) */ + { 13, "Speaker Playback" }, /* 13 FU Speaker Out Mute/Volume */ + { 14, "IEC958 Playback" }, /* 14 FU SPDIF Out Mute/Volume */ + { 16, "Mic Capture" }, /* 16 FU Capture Mic-in Mute/Volume */ + { 17, "Line Capture" }, /* 17 FU Capture Line-in Mute/Volume */ + /* 18 CS for 1 PCM (USB Streaming) Clock Source */ + /* 19 CS for 2 SPDIF (USB Streaming) Clock Source */ + { 20, "Capture Source" }, /* 20 SU Capture Source Selector */ + /* 22 CS Line-in and Mic-in Clock Source */ + { 0 } +}; + +static struct usbmix_selector_map xonar_u7_selectors[] = { +{ + .id = 20, + .count = 2, + .names = (const char*[]) {"Mic", "Line"} + }, + { 0 } /* terminator */ + +}; + /* * Control map entries */ @@ -451,6 +493,12 @@ static struct usbmix_ctl_map usbmix_ctl_maps[] = { .id = USB_ID(0x25c4, 0x0003), .map = scms_usb3318_map, }, + { + /* Asus Xonar U7 */ + .id = USB_ID(0x1043, 0x857c), + .map = xonar_u7_map, + .selector_map = xonar_u7_selectors, + }, { 0 } /* terminator */ }; -- 2.3.6

2 2

[alsa-devel] [PATCH 1/2] ASoC: wm8960: correct the min gain value of some PGA
by Zidan Wang 10 Sep '15

10 Sep '15

The min gain is the corresponding gain value when the register value is 0 instead of 1, just correct it. Signed-off-by: Zidan Wang <zidan.wang(a)freescale.com> --- sound/soc/codecs/wm8960.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/wm8960.c b/sound/soc/codecs/wm8960.c index e3b7d0c..6163474 100644 --- a/sound/soc/codecs/wm8960.c +++ b/sound/soc/codecs/wm8960.c @@ -211,11 +211,11 @@ static int wm8960_put_deemph(struct snd_kcontrol *kcontrol, return wm8960_set_deemph(codec); } -static const DECLARE_TLV_DB_SCALE(adc_tlv, -9700, 50, 0); -static const DECLARE_TLV_DB_SCALE(dac_tlv, -12700, 50, 1); +static const DECLARE_TLV_DB_SCALE(adc_tlv, -9750, 50, 1); +static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1); static const DECLARE_TLV_DB_SCALE(bypass_tlv, -2100, 300, 0); static const DECLARE_TLV_DB_SCALE(out_tlv, -12100, 100, 1); -static const DECLARE_TLV_DB_SCALE(boost_tlv, -1200, 300, 1); +static const DECLARE_TLV_DB_SCALE(boost_tlv, -1500, 300, 1); static const struct snd_kcontrol_new wm8960_snd_controls[] = { SOC_DOUBLE_R_TLV("Capture Volume", WM8960_LINVOL, WM8960_RINVOL, -- 1.9.1

3 10

[alsa-devel] [PATCH] ASoC: rt5645: Add struct dmi_system_id "Google Ultima" for chrome platform
by John Lin 10 Sep '15

10 Sep '15

Signed-off-by: John Lin <john.lin(a)realtek.com> --- sound/soc/codecs/rt5645.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 4972bf3..8e51242 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -3205,6 +3205,13 @@ static const struct dmi_system_id dmi_platform_intel_braswell[] = { DMI_MATCH(DMI_PRODUCT_NAME, "Celes"), }, }, + { + .ident = "Google Ultima", + .callback = strago_quirk_cb, + .matches = { + DMI_MATCH(DMI_PRODUCT_NAME, "Ultima"), + }, + }, { } }; -- 1.8.1.1.439.g50a6b54

2 1

[alsa-devel] [PATCH] ASoC: intel: Fix SSP port configuration after RTD3 resume.
by Liam Girdwood 10 Sep '15

10 Sep '15

Currently the SSP port settings are being clobbered as part of the DSP RTD3 restore logic. make sure we save the correct params and restore them at resume. The FW sadly does not save SSP settings as part of the PM context. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/intel/haswell/sst-haswell-ipc.c | 20 ++++++++++++-------- 1 file changed, 12 insertions(+), 8 deletions(-) diff --git a/sound/soc/intel/haswell/sst-haswell-ipc.c b/sound/soc/intel/haswell/sst-haswell-ipc.c index f6efa9d..b27f25f 100644 --- a/sound/soc/intel/haswell/sst-haswell-ipc.c +++ b/sound/soc/intel/haswell/sst-haswell-ipc.c @@ -302,6 +302,10 @@ struct sst_hsw { struct sst_hsw_ipc_dx_reply dx; void *dx_context; dma_addr_t dx_context_paddr; + enum sst_hsw_device_id dx_dev; + enum sst_hsw_device_mclk dx_mclk; + enum sst_hsw_device_mode dx_mode; + u32 dx_clock_divider; /* boot */ wait_queue_head_t boot_wait; @@ -1400,10 +1404,10 @@ int sst_hsw_device_set_config(struct sst_hsw *hsw, trace_ipc_request("set device config", dev); - config.ssp_interface = dev; - config.clock_frequency = mclk; - config.mode = mode; - config.clock_divider = clock_divider; + hsw->dx_dev = config.ssp_interface = dev; + hsw->dx_mclk = config.clock_frequency = mclk; + hsw->dx_mode = config.mode = mode; + hsw->dx_clock_divider = config.clock_divider = clock_divider; if (mode == SST_HSW_DEVICE_TDM_CLOCK_MASTER) config.channels = 4; else @@ -1704,10 +1708,10 @@ int sst_hsw_dsp_runtime_resume(struct sst_hsw *hsw) return -EIO; } - /* Set ADSP SSP port settings */ - ret = sst_hsw_device_set_config(hsw, SST_HSW_DEVICE_SSP_0, - SST_HSW_DEVICE_MCLK_FREQ_24_MHZ, - SST_HSW_DEVICE_CLOCK_MASTER, 9); + /* Set ADSP SSP port settings - sadly the FW does not store SSP port + settings as part of the PM context. */ + ret = sst_hsw_device_set_config(hsw, hsw->dx_dev, hsw->dx_mclk, + hsw->dx_mode, hsw->dx_clock_divider); if (ret < 0) dev_err(dev, "error: SSP re-initialization failed\n"); -- 2.1.4

2 1

[alsa-devel] [PATCH V2 1/1] ASoC: sti: check return of of_property_read
by Arnaud Pouliquen 10 Sep '15

10 Sep '15

Add check on of_property_read to return error when DT required property is not defined. Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> --- sound/soc/sti/uniperif_player.c | 14 ++++++++++---- sound/soc/sti/uniperif_reader.c | 6 +++++- 2 files changed, 15 insertions(+), 5 deletions(-) diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index f6eefe1..843f037 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -989,8 +989,8 @@ static int uni_player_parse_dt(struct platform_device *pdev, if (!info) return -ENOMEM; - of_property_read_u32(pnode, "version", &player->ver); - if (player->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN) { + if (of_property_read_u32(pnode, "version", &player->ver) || + player->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN) { dev_err(dev, "Unknown uniperipheral version "); return -EINVAL; } @@ -998,10 +998,16 @@ static int uni_player_parse_dt(struct platform_device *pdev, if (player->ver >= SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) info->underflow_enabled = 1; - of_property_read_u32(pnode, "uniperiph-id", &info->id); + if (of_property_read_u32(pnode, "uniperiph-id", &info->id)) { + dev_err(dev, "uniperipheral id not defined"); + return -EINVAL; + } /* Read the device mode property */ - of_property_read_string(pnode, "mode", &mode); + if (of_property_read_string(pnode, "mode", &mode)) { + dev_err(dev, "uniperipheral mode not defined"); + return -EINVAL; + } if (strcasecmp(mode, "hdmi") == 0) info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_HDMI; diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c index c502626..f791239 100644 --- a/sound/soc/sti/uniperif_reader.c +++ b/sound/soc/sti/uniperif_reader.c @@ -316,7 +316,11 @@ static int uni_reader_parse_dt(struct platform_device *pdev, if (!info) return -ENOMEM; - of_property_read_u32(node, "version", &reader->ver); + if (of_property_read_u32(node, "version", &reader->ver) || + reader->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN) { + dev_err(&pdev->dev, "Unknown uniperipheral version "); + return -EINVAL; + } /* Save the info structure */ reader->info = info; -- 1.9.1

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Re: [alsa-devel] [PATCH] ASoC: make dapm cache search depth configurable
by Mark Brown 10 Sep '15

10 Sep '15

On Thu, Sep 10, 2015 at 10:42:50AM +0100, Nikesh wrote: > Hi Mark, As I'm fairly sure I've had to tell you before don't top post. This means that when someone sees something like this: > The reason I though this will be usefull is as below: while reading they know what the "this" you are talking about is. > the widgets in wm5110_dapm_widgets[]) as defined in this macro. But if you > notice the way DSP widgets are created they use a WM_ADSP2 macro which > creates a preloader widget for every DSP widget and we want to skip past > this preloader widget when doing a cache search so increasing the cache > search depth helps here. Why is the solution to this problem to add device specific tuning? Is this extremely expensive so no other device would want it?

1 0

[alsa-devel] [PATCH] drm/i915: set proper N/CTS in modeset
by libin.yang＠intel.com 10 Sep '15

10 Sep '15

From: Libin Yang <libin.yang(a)intel.com> When modeset occurs and the TMDS frequency is set to some speical values, the N/CTS need to be set manually if audio is playing. Signed-off-by: Libin Yang <libin.yang(a)intel.com> --- drivers/gpu/drm/i915/intel_audio.c | 56 ++++++++++++++++++++++++++++++++------ include/drm/i915_component.h | 11 ++++++++ 2 files changed, 58 insertions(+), 9 deletions(-) diff --git a/drivers/gpu/drm/i915/intel_audio.c b/drivers/gpu/drm/i915/intel_audio.c index e3c32ce..a12bf57 100644 --- a/drivers/gpu/drm/i915/intel_audio.c +++ b/drivers/gpu/drm/i915/intel_audio.c @@ -128,6 +128,20 @@ static int audio_config_get_n(const struct drm_display_mode *mode, int rate) return 0; } +static uint32_t audio_config_setup_n_reg(int n, uint32_t val) +{ + int n_low, n_up; + uint32_t tmp = val; + + n_low = n & 0xfff; + n_up = (n >> 12) & 0xff; + tmp &= ~(AUD_CONFIG_UPPER_N_MASK | AUD_CONFIG_LOWER_N_MASK); + tmp |= ((n_up << AUD_CONFIG_UPPER_N_SHIFT) | + (n_low << AUD_CONFIG_LOWER_N_SHIFT) | + AUD_CONFIG_N_PROG_ENABLE); + return tmp; +} + /* check whether N/CTS/M need be set manually */ static bool audio_rate_need_prog(struct intel_crtc *crtc, struct drm_display_mode *mode) @@ -262,9 +276,14 @@ static void hsw_audio_codec_enable(struct drm_connector *connector, struct drm_i915_private *dev_priv = connector->dev->dev_private; struct intel_crtc *intel_crtc = to_intel_crtc(encoder->base.crtc); enum pipe pipe = intel_crtc->pipe; + struct i915_audio_component *acomp = dev_priv->audio_component; const uint8_t *eld = connector->eld; + struct intel_digital_port *intel_dig_port = + enc_to_dig_port(&encoder->base); + enum port port = intel_dig_port->port; uint32_t tmp; int len, i; + int n, rate; DRM_DEBUG_KMS("Enable audio codec on pipe %c, %u bytes ELD\n", pipe_name(pipe), drm_eld_size(eld)); @@ -302,12 +321,29 @@ static void hsw_audio_codec_enable(struct drm_connector *connector, /* Enable timestamps */ tmp = I915_READ(HSW_AUD_CFG(pipe)); tmp &= ~AUD_CONFIG_N_VALUE_INDEX; - tmp &= ~AUD_CONFIG_N_PROG_ENABLE; tmp &= ~AUD_CONFIG_PIXEL_CLOCK_HDMI_MASK; if (intel_pipe_has_type(intel_crtc, INTEL_OUTPUT_DISPLAYPORT)) tmp |= AUD_CONFIG_N_VALUE_INDEX; else tmp |= audio_config_hdmi_pixel_clock(mode); + + tmp &= ~AUD_CONFIG_N_PROG_ENABLE; + if (audio_rate_need_prog(intel_crtc, mode)) { + if (!acomp) + rate = 0; + else if (port >= PORT_A && port <= PORT_E) + rate = acomp->aud_sample_rate[port]; + else { + DRM_ERROR("invalid port: %d\n", port); + rate = 0; + } + n = audio_config_get_n(mode, rate); + if (n != 0) + tmp = audio_config_setup_n_reg(n, tmp); + else + DRM_DEBUG_KMS("no suitable N value is found\n"); + } + I915_WRITE(HSW_AUD_CFG(pipe), tmp); mutex_unlock(&dev_priv->av_mutex); @@ -594,9 +630,10 @@ static int i915_audio_component_sync_audio_rate(struct device *dev, struct intel_digital_port *intel_dig_port; struct intel_crtc *crtc; struct drm_display_mode *mode; + struct i915_audio_component *acomp = dev_priv->audio_component; enum pipe pipe = -1; u32 tmp; - int n_low, n_up, n; + int n; /* HSW, BDW SKL need this fix */ if (!IS_SKYLAKE(dev_priv) && @@ -630,6 +667,9 @@ static int i915_audio_component_sync_audio_rate(struct device *dev, pipe_name(pipe), port_name(port)); mode = &crtc->config->base.adjusted_mode; + /* port must be valid now, otherwise the pipe will be invalid */ + acomp->aud_sample_rate[port] = rate; + /* 2. check whether to set the N/CTS/M manually or not */ if (!audio_rate_need_prog(crtc, mode)) { tmp = I915_READ(HSW_AUD_CFG(pipe)); @@ -649,15 +689,10 @@ static int i915_audio_component_sync_audio_rate(struct device *dev, mutex_unlock(&dev_priv->av_mutex); return 0; } - n_low = n & 0xfff; - n_up = (n >> 12) & 0xff; - /* 4. set the N/CTS/M */ + /* 3. set the N/CTS/M */ tmp = I915_READ(HSW_AUD_CFG(pipe)); - tmp &= ~(AUD_CONFIG_UPPER_N_MASK | AUD_CONFIG_LOWER_N_MASK); - tmp |= ((n_up << AUD_CONFIG_UPPER_N_SHIFT) | - (n_low << AUD_CONFIG_LOWER_N_SHIFT) | - AUD_CONFIG_N_PROG_ENABLE); + tmp = audio_config_setup_n_reg(n, tmp); I915_WRITE(HSW_AUD_CFG(pipe), tmp); mutex_unlock(&dev_priv->av_mutex); @@ -678,6 +713,7 @@ static int i915_audio_component_bind(struct device *i915_dev, { struct i915_audio_component *acomp = data; struct drm_i915_private *dev_priv = dev_to_i915(i915_dev); + int i; if (WARN_ON(acomp->ops || acomp->dev)) return -EEXIST; @@ -685,6 +721,8 @@ static int i915_audio_component_bind(struct device *i915_dev, drm_modeset_lock_all(dev_priv->dev); acomp->ops = &i915_audio_component_ops; acomp->dev = i915_dev; + for (i = 0; i < I915_MAX_PORTS; i++) + acomp->aud_sample_rate[i] = 0; dev_priv->audio_component = acomp; drm_modeset_unlock_all(dev_priv->dev); diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index e6d35d7..11999eb 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -24,8 +24,19 @@ #ifndef _I915_COMPONENT_H_ #define _I915_COMPONENT_H_ +/* MAX_PORT is the number of port + * It must be sync with I915_MAX_PORTS defined i915_drv.h + * 5 should be enough as only HSW, BDW, SKL need such fix. + * The 4K resolution audio issue should be fixed on CNL + */ +#define MAX_PORTS 5 + struct i915_audio_component { struct device *dev; + /** + * @aud_sample_rate: the array of audio sample rate per port + */ + int aud_sample_rate[MAX_PORTS]; const struct i915_audio_component_ops { struct module *owner; -- 1.9.1

1 0

[alsa-devel] [PATCH 0/9] ASoC: rsnd: Gen3 support
by Kuninori Morimoto 10 Sep '15

10 Sep '15

Hi Mark These are Gen3 support patches for Renesas sound IP. These are based on my previous cleanup patches [PATCH 0/4] ASoC: rsnd: cleanup for Gen3 support Date: Thu, 10 Sep 2015 15:44:13 +0900 Kuninori Morimoto (9): ASoC: rsnd: Add Gen3 initial support ASoC: rsnd: add rsnd_mod_get() macro and use it ASoC: rsnd: ADG uses mod base common method ASoC: rsnd: add common mod confirm method ASoC: rsnd: ADG initial setup on rsnd_adg_ssi_clk_init() ASoC: rsnd: tidyup ADG clock calculate method ASoC: rsnd: tidyup ADG debug message for clock selection ASoC: rsnd: adg: ignore undefined clock error ASoC: rsnd: add AUDIO_CLKOUT support Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 5 ++ sound/soc/sh/rcar/adg.c | 302 ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++----------------------- sound/soc/sh/rcar/core.c | 12 +++++ sound/soc/sh/rcar/ctu.c | 6 +-- sound/soc/sh/rcar/dvc.c | 6 +-- sound/soc/sh/rcar/mix.c | 6 +-- sound/soc/sh/rcar/rsnd.h | 12 +++++ sound/soc/sh/rcar/src.c | 6 +-- sound/soc/sh/rcar/ssi.c | 98 ++++++++++++++++++---------------- 9 files changed, 326 insertions(+), 127 deletions(-)

1 9

[alsa-devel] [PATCH 0/4] ASoC: rsnd: cleanup for Gen3 support
by Kuninori Morimoto 10 Sep '15

10 Sep '15

Hi Mark These are not so important, but cleanup for preparing of Gen3 support Kuninori Morimoto (4): ASoC: rsnd: tidyup rsnd_dma_to_xxx() position ASoC: rsnd: rename rsnd_src_pcm_new() to rsnd_src_pcm_new_gen2() ASoC: rsnd: remove unneeded sh_clk header ASoC: rsnd: add missing #sound-dai-cells explain on Document Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 2 ++ include/sound/rcar_snd.h | 1 - sound/soc/sh/rcar/adg.c | 1 - sound/soc/sh/rcar/rsnd.h | 3 +-- sound/soc/sh/rcar/src.c | 11 ++--------- 5 files changed, 5 insertions(+), 13 deletions(-)

1 4

[alsa-devel] [PATCH 0/2] ASoC: ak4642: MCKO support
by Kuninori Morimoto 10 Sep '15

10 Sep '15

Hi Mark These patches adds MCKO support for ak4642. Kuninori Morimoto (2): ASoC: ak4642: use *dev on ak4642_i2c_probe() ASoC: ak4642: enable to use MCKO as fixed rate output pin on DT Documentation/devicetree/bindings/sound/ak4642.txt | 22 +++++++++++++++- sound/soc/codecs/ak4642.c | 152 +++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++-------------------------------------- 2 files changed, 120 insertions(+), 54 deletions(-) Best regards --- Kuninori Morimoto

1 2

[alsa-devel] [PATCH 1/1] ASoC: sti: fix compilation warnings
by Arnaud Pouliquen 09 Sep '15

09 Sep '15

Add check on of_property_read_string return to avoid compilation warning. Signed-off-by: Arnaud Pouliquen <arnaud.pouliquen(a)st.com> --- sound/soc/sti/uniperif_player.c | 32 +++++++++++++++++++------------- sound/soc/sti/uniperif_reader.c | 6 +++++- 2 files changed, 24 insertions(+), 14 deletions(-) diff --git a/sound/soc/sti/uniperif_player.c b/sound/soc/sti/uniperif_player.c index f6eefe1..d90a3fb 100644 --- a/sound/soc/sti/uniperif_player.c +++ b/sound/soc/sti/uniperif_player.c @@ -989,8 +989,8 @@ static int uni_player_parse_dt(struct platform_device *pdev, if (!info) return -ENOMEM; - of_property_read_u32(pnode, "version", &player->ver); - if (player->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN) { + if ((of_property_read_u32(pnode, "version", &player->ver)) || + (player->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN)) { dev_err(dev, "Unknown uniperipheral version "); return -EINVAL; } @@ -998,19 +998,25 @@ static int uni_player_parse_dt(struct platform_device *pdev, if (player->ver >= SND_ST_UNIPERIF_VERSION_UNI_PLR_TOP_1_0) info->underflow_enabled = 1; - of_property_read_u32(pnode, "uniperiph-id", &info->id); + if (of_property_read_u32(pnode, "uniperiph-id", &info->id)) { + dev_err(dev, "uniperipheral id not defined"); + return -EINVAL; + } /* Read the device mode property */ - of_property_read_string(pnode, "mode", &mode); - - if (strcasecmp(mode, "hdmi") == 0) - info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_HDMI; - else if (strcasecmp(mode, "pcm") == 0) - info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_PCM; - else if (strcasecmp(mode, "spdif") == 0) - info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_SPDIF; - else - info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_NONE; + if (!of_property_read_string(pnode, "mode", &mode)) { + if (strcasecmp(mode, "hdmi") == 0) + info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_HDMI; + else if (strcasecmp(mode, "pcm") == 0) + info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_PCM; + else if (strcasecmp(mode, "spdif") == 0) + info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_SPDIF; + else + info->player_type = SND_ST_UNIPERIF_PLAYER_TYPE_NONE; + } else { + dev_err(dev, "uniperipheral mode not defined"); + return -EINVAL; + } /* Save the info structure */ player->info = info; diff --git a/sound/soc/sti/uniperif_reader.c b/sound/soc/sti/uniperif_reader.c index c502626..7ddbd14 100644 --- a/sound/soc/sti/uniperif_reader.c +++ b/sound/soc/sti/uniperif_reader.c @@ -316,7 +316,11 @@ static int uni_reader_parse_dt(struct platform_device *pdev, if (!info) return -ENOMEM; - of_property_read_u32(node, "version", &reader->ver); + if ((of_property_read_u32(node, "version", &reader->ver)) || + (reader->ver == SND_ST_UNIPERIF_VERSION_UNKNOWN)) { + dev_err(&pdev->dev, "Unknown uniperipheral version "); + return -EINVAL; + } /* Save the info structure */ reader->info = info; -- 1.9.1

2 1

[alsa-devel] [PATCH lib 0/5] alsa-lib topology trivial fixes
by Takashi Iwai 09 Sep '15

09 Sep '15

Hi, here is a series of trivial fixes that have been spotted by gcc warnings. Takashi Takashi Iwai (5): topology: builder: Fix possibly uninitialized variable in write_elem_block() topology: ctl: Fix access type checks topology: data: Fix wrong size check in tplg_parse_data_hex() topology: parser: Add missing return value to snd_tplg_set_manifest_data() topology: pcm: Remove unused variables src/topology/builder.c | 9 ++++----- src/topology/ctl.c | 2 +- src/topology/data.c | 4 ++-- src/topology/parser.c | 1 + src/topology/pcm.c | 3 +-- 5 files changed, 9 insertions(+), 10 deletions(-) -- 2.5.1

2 9

[alsa-devel] [PATCH v3 0/9] asoc: Add audio for sti platforms
by Arnaud Pouliquen 09 Sep '15

09 Sep '15

version 3 : - patch dependency: ASoC: simple card: set cpu-dai sysclk with mclk-fs (Arnaud Pouliquen) - Remark not applied: Use Russell King's patch "sound/core: add IEC958 channel status helper". => This allows to create settings based on runtime, but not compatible with iec958 controls exposed to userland (settings overwritten by snd_pcm_create_iec958_consumer) - main corrections: - rework DAI CPU declaration to use generic dma - Supress abstraction layer for uniperiph player/reader - use dapm for DAC powerup/down version 2: corrections after first review - bindings: . suppress useless dts declaration - uniperif_player.c: . Suppress standby mode in uniplayer ( will be part of a dedicated patch) . Deliver IEC and clock adjustement controls in separate patch - uniperif_reader.c: . kept as separate file as behavior and feature not same than uniplayer . minor corrections - codec/sti-sas.c . implement DAPM for DAC . rework registers declaration to separate device and driver part version 1: - First patches submission Arnaud Pouliquen (9): ASoC: sti: add binding for ASoC driver ASoC: sti: Add uniperipheral header file ASoC: sti: Add CPU DAI driver for playback ASoC: sti: Add CPU DAI driver for capture ASoC: sti: Add uniperipheral dai driver ASoC: Add ability to build sti drivers ASoC: Codec: Add sti platform codec ASoC: sti: Add clock adjustement control ASoC: sti: Add IEC control .../devicetree/bindings/sound/st,sti-asoc-card.txt | 155 +++ sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/sti-sas.c | 627 ++++++++++ sound/soc/sti/Kconfig | 11 + sound/soc/sti/Makefile | 4 + sound/soc/sti/sti_uniperif.c | 254 ++++ sound/soc/sti/uniperif.h | 1229 ++++++++++++++++++++ sound/soc/sti/uniperif_player.c | 1109 ++++++++++++++++++ sound/soc/sti/uniperif_reader.c | 364 ++++++ 12 files changed, 3762 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt create mode 100644 sound/soc/codecs/sti-sas.c create mode 100644 sound/soc/sti/Kconfig create mode 100644 sound/soc/sti/Makefile create mode 100644 sound/soc/sti/sti_uniperif.c create mode 100644 sound/soc/sti/uniperif.h create mode 100644 sound/soc/sti/uniperif_player.c create mode 100644 sound/soc/sti/uniperif_reader.c -- 1.9.1

4 27

[alsa-devel] [PATCH] [alsa-utils.git] Fix a strange regression in aplay's '-d' option
by Grond 09 Sep '15

09 Sep '15

Hello. (J. Random Hacker here) While using aplay/arecord (version 1.0.28) to generate some noise, I ran across a regression in the use of the '-d' option. Specifically: According to the man page: -d, --duration=# Interrupt after # seconds. A value of zero means infinity. The default is zero, so if this option is omitted then the arecord process will run until it is killed. [snip] arecord -d 10 -f cd -t wav -D copy foobar.wav will record foobar.wav as a 10-second, CD-quality wave file, using the PCM "copy" (which might be defined in the user's .asoundrc file as: pcm.copy { type plug slave { pcm hw } route_policy copy } [snip] So the behavior I expect to see (and have experienced in the past) is: $ arecord -d 3 -f cd dump.wav Recording WAVE 'dump.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo $ ls -l -rw------- 1 grond66 grond66 529244 Sep 5 00:34 dump.wav What actually happens is: $ arecord -d 3 -f cd dump.wav Recording WAVE 'dump.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo [ program hangs here until SIGINT ] ^CAborted by signal Interrupt... arecord: pcm_read:2031: read error: Interrupted system call $ ls -l -rw------- 1 grond66 grond66 526444 Sep 5 00:39 dump-01.wav -rw------- 1 grond66 grond66 44 Sep 5 00:39 dump-02.wav -rw------- 1 grond66 grond66 44 Sep 5 00:39 dump-03.wav -rw------- 1 grond66 grond66 44 Sep 5 00:39 dump-04.wav -rw------- 1 grond66 grond66 44 Sep 5 00:39 dump-05.wav -rw------- 1 grond66 grond66 44 Sep 5 00:39 dump-06.wav -rw------- 1 grond66 grond66 44 Sep 5 00:39 dump-07.wav -rw------- 1 grond66 grond66 44 Sep 5 00:39 dump-08.wav -rw------- 1 grond66 grond66 44 Sep 5 00:39 dump-09.wav -rw------- 1 grond66 grond66 44 Sep 5 00:39 dump-100.wav -rw------- 1 grond66 grond66 44 Sep 5 00:39 dump-101.wav -rw------- 1 grond66 grond66 44 Sep 5 00:39 dump-102.wav -rw------- 1 grond66 grond66 44 Sep 5 00:39 dump-103.wav -rw------- 1 grond66 grond66 44 Sep 5 00:39 dump-104.wav -rw------- 1 grond66 grond66 44 Sep 5 00:39 dump-105.wav -rw------- 1 grond66 grond66 44 Sep 5 00:39 dump-106.wav -rw------- 1 grond66 grond66 44 Sep 5 00:39 dump-107.wav [ etcetera... ] $ aplay dump-01.wav [ three seconds of hiss, as expected ] $ less dump-02.wav RIFF$^@^@^@WAVEfmt ^P^@^@^@^A^@^B^@D<AC>^@^@^P<B1>^B^@^D^@^P^@data^@^@^@^@ $ So arecord is recording the requested audio and putting it into the first file, then spamming WAV headers into all the subsequent files. This is not terribly useful. The problem appears to be six lines of code in aplay/aplay.c, in the functions pcm_read() and pcm_readv(): 2006 static ssize_t pcm_read(u_char *data, size_t rcount) 2007 { 2008 ssize_t r; 2009 size_t result = 0; 2010 size_t count = rcount; 2011 )> 2012 if (count != chunk_size) { )> 2013 count = chunk_size; )> 2014 } 2015 2016 while (count > 0 && !in_aborting) { 2045 static ssize_t pcm_readv(u_char **data, unsigned int channels, size_t rcount) 2046 { 2047 ssize_t r; 2048 size_t result = 0; 2049 size_t count = rcount; 2050 )> 2051 if (count != chunk_size) { )> 2052 count = chunk_size; )> 2053 } 2054 2055 while (count > 0 && !in_aborting) { In this case, the 'rcount' arguments are the number of PCM integers to be read from the capture device. In theory. Because what lines 2012-2014 and 2051-2053 do is effectively: count = chunk_size; In other words, the argument is completely disregarded, and 'chunk_size' is used no matter what the functions are passed. But capture() and family use the following logic: "If pcm_read{,v}() returns with a value other than the number integers we told it to read off the wire, assume something went wrong": 2925 static void capture(char *orig_name) 2926 { 2927 int tostdout=0; /* boolean which describes output stream */ 2928 int filecount=0; /* number of files written */ 2929 char *name = orig_name; /* current filename */ 2930 char namebuf[PATH_MAX+1]; 2931 off64_t count, rest; /* number of bytes to capture */ [...] 2964 do { 2965 /* open a file to write */ 2966 if(!tostdout) { 2967 /* upon the second file we start the numbering scheme */ 2968 if (filecount || use_strftime) { 2969 filecount = new_capture_file(orig_name, namebuf, 2970 sizeof(namebuf), 2971 filecount); 2972 name = namebuf; 2973 } 2974 2975 /* open a new file */ 2976 remove(name); 2977 fd = safe_open(name); 2978 f (fd < 0) { 2979 perror(name); 2980 prg_exit(EXIT_FAILURE); 2981 } 2982 filecount++; 2983 } 2984 2985 rest = count; 2986 if (rest > fmt_rec_table[file_type].max_filesize) 2987 rest = fmt_rec_table[file_type].max_filesize; 2988 if (max_file_size && (rest > max_file_size)) 2989 rest = max_file_size; 2990 2991 /* setup sample header */ 2992 if (fmt_rec_table[file_type].start) 2993 fmt_rec_table[file_type].start(fd, rest); 2994 2995 /* capture */ 2996 fdcount = 0; 2997 while (rest > 0 && recycle_capture_file == 0 && !in_aborting) { )> 2998 size_t c = (rest <= (off64_t)chunk_bytes) ? )> 2999 (size_t)rest : chunk_bytes; )> 3000 size_t f = c * 8 / bits_per_frame; )> 3001 if (pcm_read(audiobuf, f) != f) )> 3002 break; 3003 if (write(fd, audiobuf, c) != c) { 3004 perror(name); 3005 prg_exit(EXIT_FAILURE); 3006 } 3007 count -= c; 3008 rest -= c; 3009 fdcount += c; 3010 } 3011 [...] 3027 /* repeat the loop when format is raw without timelimit or 3028 * requested counts of data are recorded 3029 */ 3030 } while ((file_type == FORMAT_RAW && !timelimit) || count > 0); 3031 } These two modes of operation are incompatible. However, if we nix the strange requirement that all calls to pcm_read{,v}() must be for exactly 'chunk_size' samples, the problem goes away. This is the purpose of the patch I've included below.

2 2

[alsa-devel] Getting notification of sound card load
by Mina Almasry 09 Sep '15

09 Sep '15

Hi, I'd like to get a notification when a new sound card becomes available in my system. What's the best way to do that? For example, when I 'cat /proc/asound/cards' I usually see one sound card available. But if I plug in USB speakers to my system and then run 'cat /proc/asound/cards', I see that the USB sound card is ready for the device I just plugged in. What's the best way via C API to get a notification that a new card has been loaded? Is there a better way that looking for changes on /proc/asound/cards? Thanks, Mina

2 2

[alsa-devel] [PATCH] ASoC: dapm: add kcontrol to switch regulator to regulated/bypass state
by Nikesh Oswal 08 Sep '15

08 Sep '15

When regulator is defined with SND_SOC_DAPM_REGULATOR_CONTROL_BYPASS flag, then a kcontrol will be created which can be used to switch regulator to regulated/bypass state. This will help to control the behaviour of the regulator based on a usecase. For example voice call may need a regulated voltage to acheive higher quality whereas voice trigger may need bypass voltage so as to save on power. Signed-off-by: Nikesh Oswal <nikesh(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 7 +- sound/soc/soc-dapm.c | 177 ++++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 183 insertions(+), 1 deletion(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 15717b4..778b847 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -343,7 +343,12 @@ struct device; (e & (SND_SOC_DAPM_PRE_PMD | SND_SOC_DAPM_POST_PMD)) /* regulator widget flags */ -#define SND_SOC_DAPM_REGULATOR_BYPASS 0x1 /* bypass when disabled */ +/* bypass when disabled and regulated when enabled */ +#define SND_SOC_DAPM_REGULATOR_BYPASS 0x1 +/* bypass when disabled and regulated when enable by default and a + kcontrol is created to explicitly switch between bypass/regulated */ +#define SND_SOC_DAPM_REGULATOR_CONTROL_BYPASS \ + (SND_SOC_DAPM_REGULATOR_BYPASS | 0x2) struct snd_soc_dapm_widget; enum snd_soc_dapm_type; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 36ab9cb..2d77eb9 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -965,6 +965,180 @@ static int dapm_new_mux(struct snd_soc_dapm_widget *w) return 0; } +static int snd_soc_dapm_regulator_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_dapm_widget_list *wlist = + dapm_kcontrol_get_wlist(kcontrol); + struct snd_soc_dapm_widget *widget = wlist->widgets[0]; + struct snd_soc_card *card = widget->dapm->card; + struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; + unsigned int *item = ucontrol->value.enumerated.item; + unsigned int new_val, val; + int ret; + bool bypass; + + if (item[0] >= e->items) + return -EINVAL; + + val = dapm_kcontrol_get_value(kcontrol); + new_val = item[0] == 0 ? SND_SOC_DAPM_REGULATOR_BYPASS : 0; + bypass = new_val == SND_SOC_DAPM_REGULATOR_BYPASS ? false : true; + + if (new_val != val) { + mutex_lock_nested(&card->dapm_mutex, + SND_SOC_DAPM_CLASS_RUNTIME); + if (regulator_is_enabled(widget->regulator)) { + ret = regulator_allow_bypass(widget->regulator, bypass); + if (ret != 0) + dev_warn(widget->dapm->dev, + "ASoC: Failed to change bypass %s: %d\n", + widget->name, ret); + } + dapm_kcontrol_set_value(kcontrol, new_val); + widget->on_val = new_val; + mutex_unlock(&card->dapm_mutex); + } + + return 0; +} + +static int snd_soc_dapm_regulator_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + unsigned int val; + + val = dapm_kcontrol_get_value(kcontrol); + + if (val == SND_SOC_DAPM_REGULATOR_BYPASS) + ucontrol->value.enumerated.item[0] = 0; + else + ucontrol->value.enumerated.item[0] = 1; + + return 0; +} + +static const char * const dapm_regulator_texts[] = { + "Regulated", + "Bypass", +}; + +/* create new dapm regulator control */ +static int dapm_new_regulator(struct snd_soc_dapm_widget *w) +{ + int ret = 0; + struct snd_soc_card *card = w->dapm->card; + unsigned long private_value; + struct snd_kcontrol *kcontrol; + struct snd_soc_dapm_path *path; + struct soc_enum regulator_enum[] = { + SOC_ENUM_SINGLE_VIRT(ARRAY_SIZE(dapm_regulator_texts), + dapm_regulator_texts), + }; + struct snd_kcontrol_new kcontrol_regulator[] = { + SOC_ENUM_EXT(NULL, regulator_enum[0], + snd_soc_dapm_regulator_get, + snd_soc_dapm_regulator_put), + }; + + + /* kcontrol creation is done only if client requests it */ + if (w->on_val != SND_SOC_DAPM_REGULATOR_CONTROL_BYPASS) + return 0; + + + /* create a kcontrol only if somebody is sourcing + from this regulator widget */ + if (list_empty(&w->edges[SND_SOC_DAPM_DIR_IN])) { + dev_err(w->dapm->dev, "ASoC: %s has no sinks\n", w->name); + return -EINVAL; + } + + w->num_kcontrols = 1; + + private_value = (unsigned long) devm_kmemdup(card->dev, + (void *)(kcontrol_regulator[0].private_value), + sizeof(struct soc_enum), GFP_KERNEL); + if (!private_value) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + w->name); + ret = -ENOMEM; + goto err_out; + } + + kcontrol_regulator[0].private_value = private_value; + + w->kcontrol_news = devm_kmemdup(card->dev, &kcontrol_regulator[0], + sizeof(struct snd_kcontrol_new), GFP_KERNEL); + if (!(w->kcontrol_news)) { + dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", + w->name); + ret = -ENOMEM; + goto err_private; + } + + + kcontrol = snd_soc_cnew(&w->kcontrol_news[0], NULL, + w->name, NULL); + + if (!kcontrol) { + ret = -ENOMEM; + goto err_kcontrol_news; + } + + kcontrol->private_free = dapm_kcontrol_free; + + ret = dapm_kcontrol_data_alloc(w, kcontrol); + if (ret) + goto err_kcontrol; + + + ret = snd_ctl_add(card->snd_card, kcontrol); + if (ret < 0) { + dev_err(w->dapm->dev, + "ASoC: failed to add widget %s dapm kcontrol %s: %d\n", + w->name, w->kcontrol_news[0].name, ret); + goto err_kcontrol; + } + + ret = dapm_kcontrol_add_widget(kcontrol, w); + if (ret) + goto err_kcontrol; + + + /* change on_val to remove the kcontrol creation bit + as kcontrol is already created */ + w->on_val = SND_SOC_DAPM_REGULATOR_BYPASS; + /* update the kcontrol value to reflect the initial value */ + dapm_kcontrol_set_value(kcontrol, w->on_val); + + w->kcontrols = kzalloc(w->num_kcontrols * + sizeof(struct snd_kcontrol *), + GFP_KERNEL); + if (!w->kcontrols) { + ret = -ENOMEM; + goto err_kcontrol; + } + + w->kcontrols[0] = kcontrol; + + snd_soc_dapm_widget_for_each_path(w, SND_SOC_DAPM_DIR_IN, path) { + if (path->name) + dapm_kcontrol_add_path(w->kcontrols[0], path); + } + + return 0; + +err_kcontrol: + snd_ctl_free_one(kcontrol); +err_kcontrol_news: + devm_kfree(card->dev, (void *)w->kcontrol_news); +err_private: + devm_kfree(card->dev, (void *)private_value); +err_out: + return ret; +} + /* create new dapm volume control */ static int dapm_new_pga(struct snd_soc_dapm_widget *w) { @@ -2922,6 +3096,9 @@ int snd_soc_dapm_new_widgets(struct snd_soc_card *card) case snd_soc_dapm_dai_link: dapm_new_dai_link(w); break; + case snd_soc_dapm_regulator_supply: + dapm_new_regulator(w); + break; default: break; } -- 1.7.9.5

2 1

[alsa-devel] [PATCH] ASoC: make dapm cache search depth configurable
by Nikesh Oswal 08 Sep '15

08 Sep '15

cache search depth will have a default value of 2 and can be modified by the respective component probe function Signed-off-by: Nikesh Oswal <nikesh(a)opensource.wolfsonmicro.com> --- include/sound/soc-dapm.h | 1 + sound/soc/soc-core.c | 2 ++ sound/soc/soc-dapm.c | 5 +++-- 3 files changed, 6 insertions(+), 2 deletions(-) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 5abba03..15717b4 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -629,6 +629,7 @@ struct snd_soc_dapm_context { struct snd_soc_dapm_wcache path_sink_cache; struct snd_soc_dapm_wcache path_source_cache; + unsigned int cache_search_depth; #ifdef CONFIG_DEBUG_FS struct dentry *debugfs_dapm; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 6173d15..1802883 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1595,6 +1595,7 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) card->dapm.bias_level = SND_SOC_BIAS_OFF; card->dapm.dev = card->dev; card->dapm.card = card; + card->dapm.cache_search_depth = 2; list_add(&card->dapm.list, &card->dapm_list); #ifdef CONFIG_DEBUG_FS @@ -2665,6 +2666,7 @@ static int snd_soc_component_initialize(struct snd_soc_component *component, component->remove = component->driver->remove; dapm = &component->dapm; + dapm->cache_search_depth = 2; dapm->dev = dev; dapm->component = component; dapm->bias_level = SND_SOC_BIAS_OFF; diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index f4bf21a..36ab9cb 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -580,11 +580,12 @@ dapm_wcache_lookup(struct snd_soc_dapm_wcache *wcache, const char *name) { struct snd_soc_dapm_widget *w = wcache->widget; struct list_head *wlist; - const int depth = 2; + int depth; int i = 0; - if (w) { + if (w && w->dapm->cache_search_depth) { wlist = &w->dapm->card->widgets; + depth = w->dapm->cache_search_depth; list_for_each_entry_from(w, wlist, list) { if (!strcmp(name, w->name)) -- 1.7.9.5

2 1

[alsa-devel] [PATCH 1/2] hda-emu: Fix hda_i915 build failure
by David Henningsson 07 Sep '15

07 Sep '15

This allows building hda-emu even after recent i915 -> hda patches. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- include/drm/i915_component.h | 1 + include/sound/hda_i915.h | 1 + 2 files changed, 2 insertions(+) create mode 100644 include/sound/hda_i915.h diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index e69de29..545c915 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -0,0 +1 @@ +#include "../../dist/include/drm/i915_component.h" diff --git a/include/sound/hda_i915.h b/include/sound/hda_i915.h new file mode 100644 index 0000000..1579430 --- /dev/null +++ b/include/sound/hda_i915.h @@ -0,0 +1 @@ +#include "../../dist/include/sound/hda_i915.h" -- 1.9.1

2 2

[alsa-devel] [PATCH 1/1] ALSA: usb-audio: Change internal PCM order
by johan＠oljud.se 07 Sep '15

07 Sep '15

From: Johan Rastén <johan(a)oljud.se> New PCMs will now be added to the end of the chip's PCM list instead of to the front. This changes the way streams are combined so that the first capture stream will now be merged with the first playback stream instead of the last. This fixes a problem with ASUS U7. Cards with one playback stream and cards without capture streams should be unaffected by this change. Exception added for M-Audio Audiophile USB (tm) since it seems to have a fix to swap capture stream numbering in alsa-lib conf/cards/USB-audio.conf Signed-off-by: Johan Rastén <johan(a)oljud.se> diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 310a382..01f5a86 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -377,8 +377,15 @@ int snd_usb_add_audio_stream(struct snd_usb_audio *chip, snd_usb_init_substream(as, stream, fp); - list_add(&as->list, &chip->pcm_list); + /* + * Keep using head insertion for M-Audio Audiophile USB (tm) which has a + * fix to swap capture stream order in conf/cards/USB-audio.conf + */ + if (chip->usb_id == USB_ID(0x0763, 0x2003)) + list_add(&as->list, &chip->pcm_list); + else + list_add_tail(&as->list, &chip->pcm_list); + chip->pcm_devs++; snd_usb_proc_pcm_format_add(as); -- 2.1.4

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Recurse before saving terminal properties
by Julian Scheel 06 Sep '15

06 Sep '15

When the descriptor parser recurses into the clock source descriptor, the terminals properties, like name and type, are set to the values of the clock selector as check_input_term overwrites term->name and others in the recursive call. As we want to only fill missing values using the descriptors higher in the recursion tree the properties taken from the actual descriptor must be applied after recursing. Signed-off-by: Julian Scheel <julian(a)jusst.de> --- sound/usb/mixer.c | 9 +++++---- 1 file changed, 5 insertions(+), 4 deletions(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 6b3acba..4a49944 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -715,15 +715,16 @@ static int check_input_term(struct mixer_build *state, int id, term->name = d->iTerminal; } else { /* UAC_VERSION_2 */ struct uac2_input_terminal_descriptor *d = p1; - term->type = le16_to_cpu(d->wTerminalType); - term->channels = d->bNrChannels; - term->chconfig = le32_to_cpu(d->bmChannelConfig); - term->name = d->iTerminal; /* call recursively to get the clock selectors */ err = check_input_term(state, d->bCSourceID, term); if (err < 0) return err; + + term->type = le16_to_cpu(d->wTerminalType); + term->channels = d->bNrChannels; + term->chconfig = le32_to_cpu(d->bmChannelConfig); + term->name = d->iTerminal; } return 0; case UAC_FEATURE_UNIT: { -- 2.5.0

4 7

[alsa-devel] [PATCH] ASoC: au1x: psc-i2s: Fix unused variable 'ret' warning
by Axel Lin 06 Sep '15

06 Sep '15

Fix below build warning: sound/soc/au1x/psc-i2s.c: In function 'au1xpsc_i2s_drvprobe': sound/soc/au1x/psc-i2s.c:299:6: warning: unused variable 'ret' [-Wunused-variable] Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/au1x/psc-i2s.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index 38e853a..0bf9d62 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -296,7 +296,6 @@ static int au1xpsc_i2s_drvprobe(struct platform_device *pdev) { struct resource *iores, *dmares; unsigned long sel; - int ret; struct au1xpsc_audio_data *wd; wd = devm_kzalloc(&pdev->dev, sizeof(struct au1xpsc_audio_data), -- 2.1.0

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda - Enable headphone jack detect on old Fujitsu laptops
by Takashi Iwai 05 Sep '15

05 Sep '15

According to the bug report, FSC Amilo laptops with ALC880 can detect the headphone jack but currently the driver disables it. It's partly intentionally, as non-working jack detect was reported in the past. Let's enable now. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=102501 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 764dd5bba9ee..9b43cb95c047 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -1135,7 +1135,7 @@ static const struct hda_fixup alc880_fixups[] = { /* override all pins as BIOS on old Amilo is broken */ .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { - { 0x14, 0x0121411f }, /* HP */ + { 0x14, 0x0121401f }, /* HP */ { 0x15, 0x99030120 }, /* speaker */ { 0x16, 0x99030130 }, /* bass speaker */ { 0x17, 0x411111f0 }, /* N/A */ @@ -1155,7 +1155,7 @@ static const struct hda_fixup alc880_fixups[] = { /* almost compatible with FUJITSU, but no bass and SPDIF */ .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { - { 0x14, 0x0121411f }, /* HP */ + { 0x14, 0x0121401f }, /* HP */ { 0x15, 0x99030120 }, /* speaker */ { 0x16, 0x411111f0 }, /* N/A */ { 0x17, 0x411111f0 }, /* N/A */ -- 2.5.1

1 1

[alsa-devel] No sound with Odys Winpad v10 - byt-rt5640
by Moritz 05 Sep '15

05 Sep '15

Hi, I'm having trouble getting sound to work and it looks like it's a bug in the driver. Full description: When loading the kernel-module for Bay Trail SoC TCO device (Version=3, TCOBASE=0x0460) the kernel throws a kernel trace depending on the firmware-image. First I provided the firmware-image from kernel.org: fw_sst_0f28.bin-48kHz_i2s_master With this image the kernel loads the firmware but no sound-devices are found. /proc/asound/cards reports no cards. dmesg just tells [ 11.196018] byt-rt5640 byt-rt5640: ASoC: CPU DAI baytrail-pcm-audio not registered If I'm loading a firmware-image from Intel directly I get a sound-device in /proc but the module failes later with a kernel trace. The download-link to the intel-firmware: https://downloadcenter.intel.com/download/23903 The relevant parts of my dmesg for this problem: [ 11.120805] byt-rt5640 byt-rt5640: ASoC: CPU DAI baytrail-pcm-audio not registered [ 11.214831] iTCO_vendor_support: vendor-support=0 [ 11.215294] iTCO_wdt: Intel TCO WatchDog Timer Driver v1.11 [ 11.215391] iTCO_wdt: Found a Bay Trail SoC TCO device (Version=3, TCOBASE=0x0460) [ 11.215655] iTCO_wdt: initialized. heartbeat=30 sec (nowayout=0) [ 11.215997] byt-rt5640 byt-rt5640: ASoC: CPU DAI baytrail-pcm-audio not registered [ 11.232805] baytrail-pcm-audio baytrail-pcm-audio: FW version: 04.05.12.02 [ 11.232808] baytrail-pcm-audio baytrail-pcm-audio: Build type: 2 [ 11.232811] baytrail-pcm-audio baytrail-pcm-audio: Build date: Jan 16 2014 18:17:00 [ 11.259456] byt-rt5640 byt-rt5640: rt5640-aif1 <-> baytrail-pcm-audio mapping ok [ 11.970293] audit: type=1305 audit(1441466901.770:2): audit_pid=608 old=0 auid=4294967295 ses=4294967295 res=1 [ 12.291031] irq 6: nobody cared (try booting with the "irqpoll" option) [ 12.294817] CPU: 0 PID: 0 Comm: swapper/0 Tainted: G O 4.2.0-4-desktop #1 [ 12.294822] Hardware name: AXDIA International GmbH WINPAD V10/Type2 - Board Product Name, BIOS AD1012.1.03.017 11/21/2014 [ 12.294826] 00000000 00000000 f582df50 c080ecdd f5b3f500 f582df70 c02abab9 c09c9bfc [ 12.294839] 00000006 00000000 f582df6c f5b3f500 00000006 f582df94 c02abf49 f582df94 [ 12.294850] c05f400c fffb910a 00000000 f5b3f500 f5b3f570 00000000 f582dfd0 c02a9a53 [ 12.294861] Call Trace: [ 12.294874] [<c080ecdd>] dump_stack+0x48/0x69 [ 12.294882] [<c02abab9>] __report_bad_irq+0x29/0xd0 [ 12.294888] [<c02abf49>] note_interrupt+0x1c9/0x210 [ 12.294896] [<c05f400c>] ? add_interrupt_randomness+0x15c/0x190 [ 12.294902] [<c02a9a53>] handle_irq_event_percpu+0xa3/0x1c0 [ 12.294908] [<c02a9ba3>] handle_irq_event+0x33/0x50 [ 12.294914] [<c02ac3b5>] handle_fasteoi_irq+0x75/0x130 [ 12.294919] [<c02ac340>] ? handle_simple_irq+0x70/0x70 [ 12.294925] [<c0204b11>] handle_irq+0x71/0x90 [ 12.294929] <IRQ> [<c081603c>] do_IRQ+0x3c/0xd0 [ 12.294941] [<c08155b3>] common_interrupt+0x33/0x38 [ 12.294948] [<c06e753f>] ? cpuidle_enter_state+0xaf/0x2e0 [ 12.294954] [<c06e77a4>] cpuidle_enter+0x14/0x20 [ 12.294961] [<c02923f3>] call_cpuidle+0x33/0x70 [ 12.294967] [<c029268e>] cpu_startup_entry+0x25e/0x320 [ 12.294973] [<c080743b>] rest_init+0x7b/0x80 [ 12.294980] [<c0b34cd8>] start_kernel+0x456/0x45d [ 12.294986] [<c0b3464e>] ? set_init_arg+0x52/0x52 [ 12.294992] [<c0b3438c>] i386_start_kernel+0x13a/0x13d [ 12.294995] handlers: [ 12.298499] [<f8725c80>] sst_byt_irq [snd_soc_sst_baytrail_pcm] threaded [<f8724360>] sst_byt_irq_thread [snd_soc_sst_baytrail_pcm] [ 12.302073] Disabling IRQ #6 with my limited understanding it looks like the irq isn't handled correctly. System information: Linux klappstulle.XXX 4.2.0-4-desktop #1 SMP PREEMPT Mon Aug 31 12:34:37 UTC 2015 (efc468a) i686 i686 i386 GNU/Linux Modules Loaded fuse dm_crypt drbg ansi_cprng algif_skcipher af_alg loop nf_log_ipv6 af_packet xt_pkttype nf_log_ipv4 nf_log_common xt_LOG xt_limit ip6t_REJECT xt_tcpudp nf_conntrack_ipv6 nf_defrag_ipv6 ip6table_raw ipt_REJECT iptable_raw xt_CT iptable_filter ip6table_mangle nf_conntrack_netbios_ns nf_conntrack_broadcast nf_conntrack_ipv4 nf_defrag_ipv4 ip_tables xt_conntrack nf_conntrack ip6table_filter ip6_tables x_tables xfs libcrc32c nls_iso8859_1 nls_cp437 vfat fat joydev hid_multitouch iTCO_wdt iTCO_vendor_support r8723bs snd_soc_sst_baytrail_pcm snd_soc_sst_ipc snd_soc_sst_dsp dw_dmac_core snd_soc_sst_byt_rt5640_mach intel_powerclamp coretemp kvm_intel kvm crc32_pclmul aesni_intel aes_i586 xts lrw gf128mul ablk_helper cryptd pcspkr efivars cfg80211 snd_intel_sst_acpi snd_intel_sst_core snd_soc_rt5640 snd_soc_sst_mfld_platform snd_soc_rl6231 snd_soc_core snd_compress lpc_ich mfd_core snd_pcm thermal mei_txe xhci_pci mei rfkill_gpio xhci_hcd 8250_fintek jsa1212 industrialio i2c_hid regmap_i2c battery rfkill snd_timer snd acpi_pad int3402_thermal int3403_thermal processor_thermal_device int340x_thermal_zone intel_soc_dts_iosf ac int3400_thermal acpi_thermal_rel soundcore ac97_bus snd_soc_sst_acpi i2c_designware_platform i2c_designware_core 8250_dw pwm_lpss_platform pwm_lpss processor dm_mod efivarfs btrfs xor raid6_pq mmc_block crc32c_intel i915 i2c_algo_bit drm_kms_helper drm button video sdhci_acpi sdhci mmc_core sg Package versions gcc-4.8-7.1.2.i586 make-4.0-2.2.3.i586 binutils-2.24-6.1.7.i586 util-linux-2.25.1-17.1.i586 mount from util-linux 2.25.1 (libmount 2.25.0: selinux, assert, debug) glibc-2.19-16.15.1.i686 linux-glibc-devel-3.16-2.1.7.noarch /proc/cmdline BOOT_IMAGE=/boot/vmlinuz-4.2.0-4-desktop root=UUID=fc064bad-46f0-4947-9364-3a7864bb0b6b ro nomodeset \ sdhci.debug_quirks=0x8000 quiet showopts /proc/cpuinfo processor : *** 0 - 3 *** vendor_id : GenuineIntel cpu family : 6 model : 55 model name : Intel(R) Atom(TM) CPU Z3735F @ 1.33GHz stepping : 8 microcode : 0x82a cpu MHz : 1751.642 cache size : 1024 KB physical id : 0 siblings : 4 core id : 0 cpu cores : 4 apicid : 0 initial apicid : 0 fdiv_bug : no f00f_bug : no coma_bug : no fpu : yes fpu_exception : yes cpuid level : 11 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe nx rdtscp lm constant_tsc arch_perfmon pebs bts xtopology nonstop_tsc aperfmperf pni pclmulqdq dtes64 monitor ds_cpl vmx est tm2 ssse3 cx16 xtpr pdcm sse4_1 sse4_2 movbe popcnt tsc_deadline_timer aes rdrand lahf_lm 3dnowprefetch ida arat epb dtherm tpr_shadow vnmi flexpriority ept vpid tsc_adjust smep erms bugs : bogomips : 2662.40 clflush size : 64 cache_alignment : 64 address sizes : 36 bits physical, 48 bits virtual power management: /proc/asound/cards 0 [bytrt5640 ]: byt-rt5640 - byt-rt5640 byt-rt5640 /proc/asound/devices 2: [ 0] : control 3: [ 0- 0]: digital audio playback 4: [ 0- 0]: digital audio capture 33: : timer Please see attached for full dmesg. Kind regards Moritz

1 0

[alsa-devel] [RFC PATCH (alsa-lib)] pcm: Modify check condition in snd_pcm_sw_params_set_avail_min
by Koro Chen 04 Sep '15

04 Sep '15

When we use a ping-ping buffer in capture, and if hw_ptr reported at IRQ is a little smaller than period_size: |xxxxxxxxxxxxxxxxxxxxxxxxxxxx--|-----------------------------| hw_ptr < period_size This available buffer will not be read since its size is smaller than avail_min (which is set to be period_size), and read thread continues to sleep. If the next hw_ptr is just a little larger than buffer_size, overrun occurs. This could be resolved by setting a small avail_min to kernel, for example, 1, so read thread wakes up and reads every data at IRQ. But current alsa-lib only allows avail_min to be at least period_size. Remove the constraint and only check for zero case. Signed-off-by: Koro Chen <koro.chen(a)mediatek.com> --- src/pcm/pcm.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index f5fc728..8492689 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -5958,12 +5958,8 @@ int snd_pcm_sw_params_set_avail_min(snd_pcm_t *pcm, snd_pcm_sw_params_t *params, #endif { assert(pcm && params); - /* Fix avail_min if it's below period size. The period_size - * defines the minimal wake-up timing accuracy, so it doesn't - * make sense to set below that. - */ - if (val < pcm->period_size) - val = pcm->period_size; + if (!val) + val = 1; params->avail_min = val; return 0; } -- 1.7.9.5

4 7

[alsa-devel] Tascam US-366 and Alsa Firmware
by Burton Samograd 04 Sep '15

04 Sep '15

Hi, I recently bought the Tascam US-366 and forgot to check for Linux support, which it doesn't have (yet! :). Looking through the Alsa firmware sources, there seems to be a firmware loader called usx2yloader which supports a number of Tascam cards, but the US-366 is not listed. It looks like really usx2yloader just needs the firmware and a `prepad` file. I`ve found the firmware for the card (I think) buried in the OSX firmware download package: http://tascam.com/content/downloads/products/789/us-322_us-366_fw_v103_mave… Now I think I just need the prepad file, but there's little documentation regarding exactly what that is (I'm assuming a number of 0s just to pad the firmware image to alignment, but I'm not sure of what alignment). Any ideas if usx2yloader will work at all with the US-366? Does anyone know what the pre-pad file should be/contain? Has anyone tried to get the US-366 working yet and failed? Reference: http://tascam.com/product/us-366/ Thanks. Burton Samograd

2 2

[alsa-devel] reconfigure cpu dai after resume from suspend
by maruthi srinivas 04 Sep '15

04 Sep '15

Hi, I came across a situation, where I need to reconfigure cpu dai ( similar to cpu dai's hw_params()) after system resumed from suspend. Should this be taken care by soc core or do I need to call the same logic(hw_params()) inside snd_soc_dai_driver's resume() ? Regards, Maruthi

1 0

[alsa-devel] [PATCH] drm/i915: Improve kernel-doc for i915_audio_component struct
by David Henningsson 04 Sep '15

04 Sep '15

To make kernel-doc happy, the i915_audio_component_audio_ops struct cannot be nested. Signed-off-by: David Henningsson <david.henningsson(a)canonical.com> --- Note that I didn't do the same un-nesting for i915_audio_component_ops. This is to make it easier to merge the pending sync_audio_rate patch set. It applies on top of my just sent patches so should probably be taken through Takashi's tree. Documentation/DocBook/drm.tmpl | 1 + include/drm/i915_component.h | 33 +++++++++++++++++++++++---------- 2 files changed, 24 insertions(+), 10 deletions(-) diff --git a/Documentation/DocBook/drm.tmpl b/Documentation/DocBook/drm.tmpl index 2fb9a54..7554679 100644 --- a/Documentation/DocBook/drm.tmpl +++ b/Documentation/DocBook/drm.tmpl @@ -4048,6 +4048,7 @@ int num_ioctls;</synopsis> <title>High Definition Audio</title> !Pdrivers/gpu/drm/i915/intel_audio.c High Definition Audio over HDMI and Display Port !Idrivers/gpu/drm/i915/intel_audio.c +!Iinclude/drm/i915_component.h </sect2> <sect2> <title>Panel Self Refresh PSR (PSR/SRD)</title> diff --git a/include/drm/i915_component.h b/include/drm/i915_component.h index b2d56dd..89d6362 100644 --- a/include/drm/i915_component.h +++ b/include/drm/i915_component.h @@ -24,6 +24,28 @@ #ifndef _I915_COMPONENT_H_ #define _I915_COMPONENT_H_ +/** + * struct i915_audio_component_audio_ops - i915 directly calling hda driver + * + * @audio_ptr: Pointer to pass when calling pin_eld_notify. + * @pin_eld_notify: Called from i915 driver, notifying the HDA driver that + * pin sense and/or ELD information has changed. + * + * These functions are implemented by hda driver and called by the i915 + * driver. + */ +struct i915_audio_component_audio_ops { + void *audio_ptr; + void (*pin_eld_notify)(void *audio_ptr, int port); +}; + +/** + * struct i915_audio_component - Used for direct communication between i915 and hda drivers + * + * @dev: i915 device, used as parameter for ops + * @ops: Ops implemented by i915 driver, called by hda driver + * @audio_ops: Ops implemented by hda driver, called by i915 driver + */ struct i915_audio_component { struct device *dev; @@ -35,16 +57,7 @@ struct i915_audio_component { int (*get_cdclk_freq)(struct device *); } *ops; - const struct i915_audio_component_audio_ops { - void *audio_ptr; - /** - * Call from i915 driver, notifying the HDA driver that - * pin sense and/or ELD information has changed. - * @audio_ptr: HDA driver object - * @port: Which port has changed (PORTA / PORTB / PORTC etc) - */ - void (*pin_eld_notify)(void *audio_ptr, int port); - } *audio_ops; + const struct i915_audio_component_audio_ops *audio_ops; }; #endif /* _I915_COMPONENT_H_ */ -- 1.9.1

3 3

[alsa-devel] [PATCH] ALSA: sparc: amd7930: Fix module autoload for OF platform driver
by Luis de Bethencourt 04 Sep '15

04 Sep '15

This platform driver has a OF device ID table but the OF module alias information is not created so module autoloading won't work. Signed-off-by: Luis de Bethencourt <luis(a)debethencourt.com> --- Hello, This patch adds the missing MODULE_DEVICE_TABLE() for OF to eport that information so modules have the correct aliases built-in and autoloading works correctly. A longer explanation by Javier Canillas can be found here: https://lkml.org/lkml/2015/7/30/519 Thanks, Luis sound/sparc/amd7930.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/sparc/amd7930.c b/sound/sparc/amd7930.c index 784ceb8..35c1f6a 100644 --- a/sound/sparc/amd7930.c +++ b/sound/sparc/amd7930.c @@ -1064,6 +1064,7 @@ static const struct of_device_id amd7930_match[] = { }, {}, }; +MODULE_DEVICE_TABLE(of, amd7930_match); static struct platform_driver amd7930_sbus_driver = { .driver = { -- 2.4.6

2 1

[alsa-devel] [PATCH] ALSA: hda - Add some FIXUP quirks for white noise on Dell laptop.
by woodrow.shen＠canonical.com 04 Sep '15

04 Sep '15

From: Woodrow Shen <woodrow.shen(a)canonical.com> Dell laptop has a series model to use the same codec but different subsystem ID. At the same time they happens the white noise by login screen and headphone; for fixing them together, I only can add these IDs to FIXUP function ALC292_FIXUP_DISABLE_AAMIX, then try to solve such the similar issues. Codec: Realtek ALC3235 Vendor Id: 0x10ec0293 Subsystem Id: 0x102806dd Subsystem Id: 0x102806df Subsystem Id: 0x102806e0 Cc: <stable(a)vger.kernel.org> BugLink: https://bugs.launchpad.net/bugs/1492132 Signed-off-by: Woodrow Shen <woodrow.shen(a)canonical.com> --- sound/pci/hda/patch_realtek.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 4e6b090..764dd5b 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5189,8 +5189,11 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x06c7, "Dell", ALC255_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x06d9, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x06da, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), - SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC292_FIXUP_DISABLE_AAMIX), SND_PCI_QUIRK(0x1028, 0x06db, "Dell", ALC292_FIXUP_DISABLE_AAMIX), + SND_PCI_QUIRK(0x1028, 0x06dd, "Dell", ALC292_FIXUP_DISABLE_AAMIX), + SND_PCI_QUIRK(0x1028, 0x06de, "Dell", ALC292_FIXUP_DISABLE_AAMIX), + SND_PCI_QUIRK(0x1028, 0x06df, "Dell", ALC292_FIXUP_DISABLE_AAMIX), + SND_PCI_QUIRK(0x1028, 0x06e0, "Dell", ALC292_FIXUP_DISABLE_AAMIX), SND_PCI_QUIRK(0x1028, 0x164a, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x164b, "Dell", ALC293_FIXUP_DELL1_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x103c, 0x1586, "HP", ALC269_FIXUP_HP_MUTE_LED_MIC2), -- 2.1.4

2 1

[alsa-devel] [PATCH v3 0/9] Fix broken DMAFLUSHP on Rockchips platform
by Shawn Lin 04 Sep '15

04 Sep '15

The purpose of the DMAFLUSHP instruction: - Tell the peripheral to clear its status and control registers. - Send a message to the peripheral to resend its level status. There are 3 timings described in PL330 Technical Reference Manual: - Timing 1: Burst request, can work well without DMAFLUSHP. - Timing 2: Single and burst request, DMAC will ignore the single transfer request. This timing happens if there are single and burst request. - Timing 3: Single transfers for a burst request, DMAC should signals datype to request the peripheral to flush the contents of any control registers. This timing happens if there is not enough MFIFO to places the burst data. A peripheral may signal a DMA request during the execution of DMAFLUSHP instruction, that cause DMA request being ignored by DMAC. But DMAC and all peripherals on RK3X SoCs DO NOT support DMAFLUSHP. It can't send a message to the peripheral to resend DMA request, and the peripheral can't acknowledge a flush request from DMAC. So all DMA requests should NOT be ignored by DMAC, and DMAC will not notify the peripheral to flush. To fix this problem, we need: - Do NOT execute DMAFLUSHP instruction. - Timing 2 and timing 3 should not happen. Because on RK3X SoCs, there are 6 or below channels and 32 MFIFO depth for DMAC_BUS, and 8 channels and 64 MFIFO depth for DMAC_PERI, it is impossible to hit the timing 3 if burst length is equal or less than 4. Since the request type signal by the peripheral can only be set by software. We can set Rockchip Soc's GRF_PERIDMAC_CON0[2:1] to select single or burst request, if it is set b01, all of the peripharals will signal a brust request. So the timing 2 will not happen, too. So DMAC on RK3X can support single or burst transfer, but can't support mixed transfer. Because burst transfer is more efficient than single transfer, this is confirmed by our ASIC team, who strongly suggest to use burst transfer. And this is confirmed by Addy's test on RK3288-Pink2 board, the speed of spi flash burst transfer will increase about two times than single transfer. Also, I have tested dw_mmc with pl330 on RK3188 platform to double confirm the result. That means burst transfer is reansonable. So we need a quirk not to execute DMAFLUSHP instruction and to use burst transfer. Note: - The Rockchip Soc default value of GRF_PERIDMAC_CON0[2:1] is b01. To support brust transfer, these bits should not be changed in bootloader. Changes in v3: - add more rockchip drivers' changes in this patchset - add Reviewed-by: Sonny Rao <sonnyrao(a)chromium.org> Changes in v2: - amend the author - reorder the patches suggested by Doug - add Reviewed-by: Doug Anderson <dianders(a)chromium.org> for rk3288.dtsi patch and arm-pl330.txt patch - amend Olof's mail address Changes in v1: - rename broken-no-flushp to "arm,pl330-broken-no-flushp" suggested by Krzysztof. - add From original author. - remove Sunny's tag Addy Ke (3): DMA: pl330: add quirk for broken no flushp ARM: dts: Add arm,pl330-broken-no-flushp quirk for rk3288 platform spi: rockchip: modify DMA max burst to 1 Boojin Kim (1): DMA: pl330: support burst mode for dev-to-mem and mem-to-dev transmit Shawn Lin (4): Documentation: arm-pl330: add description of arm,pl330-broken-no-flushp Documentation: spi-rockchip: add description of rockchip,spi-broken-burst-len Documentation: rockchip-i2s: add description of rockchip,i2s-broken-burst-len ARM: dts: Add arm,pl330-broken-no-flushp quirk for rk3xxx platform Yiwei Cai (1): ASoC: rockchip_i2s: modify DMA max burst to 1 .../devicetree/bindings/dma/arm-pl330.txt | 1 + .../devicetree/bindings/sound/rockchip-i2s.txt | 4 + .../devicetree/bindings/spi/spi-rockchip.txt | 3 +- arch/arm/boot/dts/rk3288.dtsi | 7 ++ arch/arm/boot/dts/rk3xxx.dtsi | 5 + drivers/dma/pl330.c | 101 +++++++++++++++------ drivers/spi/spi-rockchip.c | 16 +++- sound/soc/rockchip/rockchip_i2s.c | 10 +- 8 files changed, 113 insertions(+), 34 deletions(-) -- 2.3.7

2 13

[alsa-devel] [PATCH 00/12] ASoC: Fix module autoload for OF platform drivers
by Luis de Bethencourt 03 Sep '15

03 Sep '15

Hello, This series adds the missing MODULE_DEVICE_TABLE() for OF to eport that information so modules have the correct aliases built-in and autoloading works correctly. A longer explanation by Javier Canillas can be found here: https://lkml.org/lkml/2015/7/30/519 Thanks, Luis Luis de Bethencourt (12): ASoC: atmel_wm8904: Fix module autoload for OF platform driver ASoC: da9055: Fix module autoload for OF platform driver ASoC: wm8510: Fix module autoload for OF platform driver ASoC: wm8580: Fix module autoload for OF platform driver ASoC: wm8523: Fix module autoload for OF platform driver ASoC: fsl-asoc-card: Fix module autoload for OF platform driver ASoC: fsl_sai: Fix module autoload for OF platform driver ASoC: jz4740: Fix module autoload for OF platform driver ASoC: kirkwood: Fix module autoload for OF platform driver ASoC: pxa: Fix module autoload for OF platform drivers ASoC: mop500: Fix module autoload for OF platform driver ASoC: ux500: Fix module autoload for OF platform driver sound/soc/atmel/atmel_wm8904.c | 1 + sound/soc/codecs/da9055.c | 1 + sound/soc/codecs/wm8510.c | 1 + sound/soc/codecs/wm8523.c | 1 + sound/soc/codecs/wm8580.c | 1 + sound/soc/fsl/fsl-asoc-card.c | 1 + sound/soc/fsl/fsl_sai.c | 1 + sound/soc/jz4740/jz4740-i2s.c | 1 + sound/soc/kirkwood/armada-370-db.c | 1 + sound/soc/pxa/pxa-ssp.c | 1 + sound/soc/pxa/pxa2xx-pcm.c | 1 + sound/soc/ux500/mop500.c | 1 + sound/soc/ux500/ux500_msp_dai.c | 1 + 13 files changed, 13 insertions(+) -- 2.4.6

2 2

[alsa-devel] [PATCH 02/12] ASoC: da9055: Fix module autoload for OF platform driver
by Luis de Bethencourt 03 Sep '15

03 Sep '15

This platform driver has a OF device ID table but the OF module alias information is not created so module autoloading won't work. Signed-off-by: Luis de Bethencourt <luis(a)debethencourt.com> --- sound/soc/codecs/da9055.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/da9055.c b/sound/soc/codecs/da9055.c index 66bb446..ab39de6 100644 --- a/sound/soc/codecs/da9055.c +++ b/sound/soc/codecs/da9055.c @@ -1533,6 +1533,7 @@ static const struct of_device_id da9055_of_match[] = { { .compatible = "dlg,da9055-codec", }, { } }; +MODULE_DEVICE_TABLE(of, da9055_of_match); /* I2C codec control layer */ static struct i2c_driver da9055_i2c_driver = { -- 2.4.6

2 1

[alsa-devel] [PATCH 12/12] ASoC: ux500: Fix module autoload for OF platform driver
by Luis de Bethencourt 03 Sep '15

03 Sep '15

This platform driver has a OF device ID table but the OF module alias information is not created so module autoloading won't work. Signed-off-by: Luis de Bethencourt <luis(a)debethencourt.com> --- sound/soc/ux500/ux500_msp_dai.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/ux500/ux500_msp_dai.c b/sound/soc/ux500/ux500_msp_dai.c index 978f2d7..a57a9b8 100644 --- a/sound/soc/ux500/ux500_msp_dai.c +++ b/sound/soc/ux500/ux500_msp_dai.c @@ -852,6 +852,7 @@ static const struct of_device_id ux500_msp_i2s_match[] = { { .compatible = "stericsson,ux500-msp-i2s", }, {}, }; +MODULE_DEVICE_TABLE(of, ux500_msp_i2s_match); static struct platform_driver msp_i2s_driver = { .driver = { -- 2.4.6

1 0

[alsa-devel] [PATCH 11/12] ASoC: mop500: Fix module autoload for OF platform driver
by Luis de Bethencourt 03 Sep '15

03 Sep '15

This platform driver has a OF device ID table but the OF module alias information is not created so module autoloading won't work. Signed-off-by: Luis de Bethencourt <luis(a)debethencourt.com> --- sound/soc/ux500/mop500.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index 4e0c0e5..ba9fc09 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -152,6 +152,7 @@ static const struct of_device_id snd_soc_mop500_match[] = { { .compatible = "stericsson,snd-soc-mop500", }, {}, }; +MODULE_DEVICE_TABLE(of, snd_soc_mop500_match); static struct platform_driver snd_soc_mop500_driver = { .driver = { -- 2.4.6

1 0

[alsa-devel] [PATCH 10/12] ASoC: pxa: Fix module autoload for OF platform drivers
by Luis de Bethencourt 03 Sep '15

03 Sep '15

These platform drivers have a OF device ID table but the OF module alias information is not created so module autoloading won't work. Signed-off-by: Luis de Bethencourt <luis(a)debethencourt.com> --- sound/soc/pxa/pxa-ssp.c | 1 + sound/soc/pxa/pxa2xx-pcm.c | 1 + 2 files changed, 2 insertions(+) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index fbe2e93..c254a75 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -809,6 +809,7 @@ static const struct of_device_id pxa_ssp_of_ids[] = { { .compatible = "mrvl,pxa-ssp-dai" }, {} }; +MODULE_DEVICE_TABLE(of, pxa_ssp_of_ids); #endif static int asoc_ssp_probe(struct platform_device *pdev) diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index a51c9da..f1aea95 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -138,6 +138,7 @@ static const struct of_device_id snd_soc_pxa_audio_match[] = { { .compatible = "mrvl,pxa-pcm-audio" }, { } }; +MODULE_DEVICE_TABLE(of, snd_soc_pxa_audio_match); #endif static struct platform_driver pxa_pcm_driver = { -- 2.4.6

1 0

[alsa-devel] [PATCH 09/12] ASoC: kirkwood: Fix module autoload for OF platform driver
by Luis de Bethencourt 03 Sep '15

03 Sep '15

This platform driver has a OF device ID table but the OF module alias information is not created so module autoloading won't work. Signed-off-by: Luis de Bethencourt <luis(a)debethencourt.com> --- sound/soc/kirkwood/armada-370-db.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/kirkwood/armada-370-db.c b/sound/soc/kirkwood/armada-370-db.c index de7563b..e0304d5 100644 --- a/sound/soc/kirkwood/armada-370-db.c +++ b/sound/soc/kirkwood/armada-370-db.c @@ -130,6 +130,7 @@ static const struct of_device_id a370db_dt_ids[] = { { .compatible = "marvell,a370db-audio" }, { }, }; +MODULE_DEVICE_TABLE(of, a370db_dt_ids); static struct platform_driver a370db_driver = { .driver = { -- 2.4.6

1 0

[alsa-devel] [PATCH 08/12] ASoC: jz4740: Fix module autoload for OF platform driver
by Luis de Bethencourt 03 Sep '15

03 Sep '15

This platform driver has a OF device ID table but the OF module alias information is not created so module autoloading won't work. Signed-off-by: Luis de Bethencourt <luis(a)debethencourt.com> --- sound/soc/jz4740/jz4740-i2s.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/jz4740/jz4740-i2s.c b/sound/soc/jz4740/jz4740-i2s.c index b05fb1c..794a349 100644 --- a/sound/soc/jz4740/jz4740-i2s.c +++ b/sound/soc/jz4740/jz4740-i2s.c @@ -485,6 +485,7 @@ static const struct of_device_id jz4740_of_matches[] = { { .compatible = "ingenic,jz4780-i2s", .data = (void *)JZ_I2S_JZ4780 }, { /* sentinel */ } }; +MODULE_DEVICE_TABLE(of, jz4740_of_matches); #endif static int jz4740_i2s_dev_probe(struct platform_device *pdev) -- 2.4.6

1 0

[alsa-devel] [PATCH 07/12] ASoC: fsl_sai: Fix module autoload for OF platform driver
by Luis de Bethencourt 03 Sep '15

03 Sep '15

This platform driver has a OF device ID table but the OF module alias information is not created so module autoloading won't work. Signed-off-by: Luis de Bethencourt <luis(a)debethencourt.com> --- sound/soc/fsl/fsl_sai.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 5c73bea..ec14427 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -801,6 +801,7 @@ static const struct of_device_id fsl_sai_ids[] = { { .compatible = "fsl,imx6sx-sai", }, { /* sentinel */ } }; +MODULE_DEVICE_TABLE(of, fsl_sai_ids); static struct platform_driver fsl_sai_driver = { .probe = fsl_sai_probe, -- 2.4.6

1 0

[alsa-devel] [PATCH 06/12] ASoC: fsl-asoc-card: Fix module autoload for OF platform driver
by Luis de Bethencourt 03 Sep '15

03 Sep '15

This platform driver has a OF device ID table but the OF module alias information is not created so module autoloading won't work. Signed-off-by: Luis de Bethencourt <luis(a)debethencourt.com> --- sound/soc/fsl/fsl-asoc-card.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index de43887..ca8b469 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -580,6 +580,7 @@ static const struct of_device_id fsl_asoc_card_dt_ids[] = { { .compatible = "fsl,imx-audio-wm8962", }, {} }; +MODULE_DEVICE_TABLE(of, fsl_asoc_card_dt_ids); static struct platform_driver fsl_asoc_card_driver = { .probe = fsl_asoc_card_probe, -- 2.4.6

1 0

[alsa-devel] [PATCH 05/12] ASoC: wm8523: Fix module autoload for OF platform driver
by Luis de Bethencourt 03 Sep '15

03 Sep '15

This platform driver has a OF device ID table but the OF module alias information is not created so module autoloading won't work. Signed-off-by: Luis de Bethencourt <luis(a)debethencourt.com> --- sound/soc/codecs/wm8523.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8523.c b/sound/soc/codecs/wm8523.c index 43ea8ae..483ec72 100644 --- a/sound/soc/codecs/wm8523.c +++ b/sound/soc/codecs/wm8523.c @@ -430,6 +430,7 @@ static const struct of_device_id wm8523_of_match[] = { { .compatible = "wlf,wm8523" }, { }, }; +MODULE_DEVICE_TABLE(of, wm8523_of_match); static const struct regmap_config wm8523_regmap = { .reg_bits = 8, -- 2.4.6

1 0

[alsa-devel] [PATCH 04/12] ASoC: wm8580: Fix module autoload for OF platform driver
by Luis de Bethencourt 03 Sep '15

03 Sep '15

This platform driver has a OF device ID table but the OF module alias information is not created so module autoloading won't work. Signed-off-by: Luis de Bethencourt <luis(a)debethencourt.com> --- sound/soc/codecs/wm8580.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 759a792..a9f3900 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -916,6 +916,7 @@ static const struct of_device_id wm8580_of_match[] = { { .compatible = "wlf,wm8580" }, { }, }; +MODULE_DEVICE_TABLE(of, wm8580_of_match); static const struct regmap_config wm8580_regmap = { .reg_bits = 7, -- 2.4.6

1 0

[alsa-devel] [PATCH 03/12] ASoC: wm8510: Fix module autoload for OF platform driver
by Luis de Bethencourt 03 Sep '15

03 Sep '15

This platform driver has a OF device ID table but the OF module alias information is not created so module autoloading won't work. Signed-off-by: Luis de Bethencourt <luis(a)debethencourt.com> --- sound/soc/codecs/wm8510.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/codecs/wm8510.c b/sound/soc/codecs/wm8510.c index dac5beb..0617415 100644 --- a/sound/soc/codecs/wm8510.c +++ b/sound/soc/codecs/wm8510.c @@ -598,6 +598,7 @@ static const struct of_device_id wm8510_of_match[] = { { .compatible = "wlf,wm8510" }, { }, }; +MODULE_DEVICE_TABLE(of, wm8510_of_match); static const struct regmap_config wm8510_regmap = { .reg_bits = 7, -- 2.4.6

1 0

[alsa-devel] [PATCH 01/12] ASoC: atmel_wm8904: Fix module autoload for OF platform driver
by Luis de Bethencourt 03 Sep '15

03 Sep '15

This platform driver has a OF device ID table but the OF module alias information is not created so module autoloading won't work. Signed-off-by: Luis de Bethencourt <luis(a)debethencourt.com> --- sound/soc/atmel/atmel_wm8904.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/atmel/atmel_wm8904.c b/sound/soc/atmel/atmel_wm8904.c index aa354e1..1933bcd 100644 --- a/sound/soc/atmel/atmel_wm8904.c +++ b/sound/soc/atmel/atmel_wm8904.c @@ -176,6 +176,7 @@ static const struct of_device_id atmel_asoc_wm8904_dt_ids[] = { { .compatible = "atmel,asoc-wm8904", }, { } }; +MODULE_DEVICE_TABLE(of, atmel_asoc_wm8904_dt_ids); #endif static struct platform_driver atmel_asoc_wm8904_driver = { -- 2.4.6

1 0

[alsa-devel] [PATCH 0/2] Update v4 -> v5 for i915 -> HDA patch set
by David Henningsson 03 Sep '15

03 Sep '15

Takashi mistakenly merged v4 instead of v5 of the patch set. This updates the code changes. (Not the docbook, I'm still fighting to get that to look right) David Henningsson (2): drm/i915: Drop port_mst_index parameter from pin/eld callback drm/i915: Add locks around audio component bind/unbind drivers/gpu/drm/i915/intel_audio.c | 8 ++++++-- include/drm/i915_component.h | 3 +-- 2 files changed, 7 insertions(+), 4 deletions(-) -- 1.9.1

2 5

[alsa-devel] [PATCH v2 1/1] ASoC: fsl_sai: Add missing MODULE_DEVICE_TABLE
by Alexander Stein 03 Sep '15

03 Sep '15

This enables module auto-loading when using device tree. Signed-off-by: Alexander Stein <alexander.stein(a)systec-electronic.com> Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com> --- Change in v2: * Added "Acked-by: Nicolin Chen <nicoleotsuka(a)gmail.com>" sound/soc/fsl/fsl_sai.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/fsl/fsl_sai.c b/sound/soc/fsl/fsl_sai.c index 5c73bea..ec14427 100644 --- a/sound/soc/fsl/fsl_sai.c +++ b/sound/soc/fsl/fsl_sai.c @@ -801,6 +801,7 @@ static const struct of_device_id fsl_sai_ids[] = { { .compatible = "fsl,imx6sx-sai", }, { /* sentinel */ } }; +MODULE_DEVICE_TABLE(of, fsl_sai_ids); static struct platform_driver fsl_sai_driver = { .probe = fsl_sai_probe, -- 2.4.6

1 0

[alsa-devel] Linux 4.2 ALSA snd-usb-audio inconsistent lock state warn in PCM nonatomic mode
by Shuah Khan 02 Sep '15

02 Sep '15

Hi Takashi, I am seeing the following inconsistent lock state warning when PCM is run in nonatomic mode. This is on 4.2.0 and with the following change to force PCM on nonatomic mode: diff --git a/sound/usb/stream.c b/sound/usb/stream.c index 310a382..16bbb71 100644 --- a/sound/usb/stream.c +++ b/sound/usb/stream.c @@ -370,6 +370,7 @@ int snd_usb_add_audio_stream(struct snd_usb_audio *chip, pcm->private_data = as; pcm->private_free = snd_usb_audio_pcm_free; pcm->info_flags = 0; + pcm->nonatomic = true; if (chip->pcm_devs > 0) sprintf(pcm->name, "USB Audio #%d", chip->pcm_devs); else The device Bus 003 Device 002: ID 2040:7200 Hauppauge Please let me know if you need more information and any ideas on how to fix this problem. thanks, -- Shuah [ 120.283960] ================================= [ 120.283964] [ INFO: inconsistent lock state ] [ 120.283968] 4.2.0+ #29 Not tainted [ 120.283972] --------------------------------- [ 120.283975] inconsistent {HARDIRQ-ON-W} -> {IN-HARDIRQ-W} usage. [ 120.283980] swapper/1/0 [HC1[1]:SC0[0]:HE0:SE1] takes: [ 120.283983] (&(&subs->lock)->rlock){?.+...}, at: [<ffffffffa05ed210>] retire_capture_urb+0x140/0x2b0 [snd_usb_audio] [ 120.284005] {HARDIRQ-ON-W} state was registered at: [ 120.284008] [<ffffffff810af230>] __lock_acquire+0xc50/0x2380 [ 120.284016] [<ffffffff810b1631>] lock_acquire+0xb1/0x130 [ 120.284022] [<ffffffff817fd4a1>] _raw_spin_lock+0x31/0x40 [ 120.284028] [<ffffffffa05ee81d>] snd_usb_pcm_pointer+0x5d/0xc0 [snd_usb_audio] [ 120.284040] [<ffffffffa01c6498>] snd_pcm_update_hw_ptr0+0x38/0x3a0 [snd_pcm] [ 120.284052] [<ffffffffa01c73f0>] snd_pcm_update_hw_ptr+0x10/0x20 [snd_pcm] [ 120.284063] [<ffffffffa01beae5>] snd_pcm_hwsync+0x45/0xa0 [snd_pcm] [ 120.284071] [<ffffffffa01c1347>] snd_pcm_common_ioctl1+0x277/0xce0 [snd_pcm] [ 120.284081] [<ffffffffa01c227e>] snd_pcm_capture_ioctl1+0x1be/0x2d0 [snd_pcm] [ 120.284090] [<ffffffffa01c2444>] snd_pcm_capture_ioctl+0x34/0x40 [snd_pcm] [ 120.284100] [<ffffffff811ed7e1>] do_vfs_ioctl+0x301/0x560 [ 120.284107] [<ffffffff811edab9>] SyS_ioctl+0x79/0x90 [ 120.284112] [<ffffffff817fdf17>] entry_SYSCALL_64_fastpath+0x12/0x6f [ 120.284119] irq event stamp: 823304 [ 120.284122] hardirqs last enabled at (823301): [<ffffffff816707ad>] cpuidle_enter_state+0xed/0x230 [ 120.284129] hardirqs last disabled at (823302): [<ffffffff817fea28>] common_interrupt+0x68/0x6d [ 120.284135] softirqs last enabled at (823304): [<ffffffff81060351>] _local_bh_enable+0x21/0x50 [ 120.284139] softirqs last disabled at (823303): [<ffffffff810611ac>] irq_enter+0x4c/0x70 [ 120.284143] other info that might help us debug this: [ 120.284146] Possible unsafe locking scenario: [ 120.284149] CPU0 [ 120.284150] ---- [ 120.284152] lock(&(&subs->lock)->rlock); [ 120.284155] <Interrupt> [ 120.284157] lock(&(&subs->lock)->rlock); [ 120.284160] *** DEADLOCK *** [ 120.284163] no locks held by swapper/1/0. [ 120.284165] stack backtrace: [ 120.284170] CPU: 1 PID: 0 Comm: swapper/1 Not tainted 4.2.0+ #29 [ 120.284173] Hardware name: Hewlett-Packard HP ProBook 6475b/180F, BIOS 68TTU Ver. F.04 08/03/2012 [ 120.284176] ffffffff828d5630 ffff88023ec83a68 ffffffff817f4ea0 0000000000000007 [ 120.284181] ffff880235a6a500 ffff88023ec83ac8 ffffffff810ad97f 0000000000000000 [ 120.284186] 0000000000000000 ffff880200000001 ffffffff810134df ffffffff827c5390 [ 120.284192] Call Trace: [ 120.284194] <IRQ> [<ffffffff817f4ea0>] dump_stack+0x45/0x57 [ 120.284203] [<ffffffff810ad97f>] print_usage_bug+0x1ff/0x210 [ 120.284209] [<ffffffff810134df>] ? save_stack_trace+0x2f/0x50 [ 120.284214] [<ffffffff810adffe>] mark_lock+0x66e/0x6f0 [ 120.284218] [<ffffffff810aceb0>] ? print_shortest_lock_dependencies+0x1d0/0x1d0 [ 120.284222] [<ffffffff810af3ab>] __lock_acquire+0xdcb/0x2380 [ 120.284226] [<ffffffff81091a2c>] ? __enqueue_entity+0x6c/0x70 [ 120.284230] [<ffffffff810ab8dd>] ? __lock_is_held+0x4d/0x70 [ 120.284234] [<ffffffff810ab8dd>] ? __lock_is_held+0x4d/0x70 [ 120.284238] [<ffffffff810ab8dd>] ? __lock_is_held+0x4d/0x70 [ 120.284242] [<ffffffff810ab8dd>] ? __lock_is_held+0x4d/0x70 [ 120.284246] [<ffffffff810b1631>] lock_acquire+0xb1/0x130 [ 120.284256] [<ffffffffa05ed210>] ? retire_capture_urb+0x140/0x2b0 [snd_usb_audio] [ 120.284261] [<ffffffff817fd61c>] _raw_spin_lock_irqsave+0x3c/0x50 [ 120.284270] [<ffffffffa05ed210>] ? retire_capture_urb+0x140/0x2b0 [snd_usb_audio] [ 120.284276] [<ffffffff81595325>] ? usb_hcd_get_frame_number+0x25/0x30 [ 120.284285] [<ffffffffa05ed210>] retire_capture_urb+0x140/0x2b0 [snd_usb_audio] [ 120.284294] [<ffffffffa05e513c>] snd_complete_urb+0x13c/0x250 [snd_usb_audio] [ 120.284298] [<ffffffff81592cb2>] __usb_hcd_giveback_urb+0x72/0x110 [ 120.284303] [<ffffffff81592e53>] usb_hcd_giveback_urb+0x43/0x140 [ 120.284307] [<ffffffff815d1b02>] xhci_irq+0xd42/0x1fc0 [ 120.284312] [<ffffffff815d2d91>] xhci_msi_irq+0x11/0x20 [ 120.284317] [<ffffffff810c4570>] handle_irq_event_percpu+0x80/0x1a0 [ 120.284322] [<ffffffff810c46da>] handle_irq_event+0x4a/0x70 [ 120.284325] [<ffffffff810c7794>] ? handle_edge_irq+0x24/0x150 [ 120.284329] [<ffffffff810c77f1>] handle_edge_irq+0x81/0x150 [ 120.284333] [<ffffffff81005c85>] handle_irq+0x25/0x40 [ 120.284337] [<ffffffff818008ef>] do_IRQ+0x4f/0xe0 [ 120.284341] [<ffffffff817fea2d>] common_interrupt+0x6d/0x6d [ 120.284343] <EOI> [<ffffffff816707b2>] ? cpuidle_enter_state+0xf2/0x230 [ 120.284351] [<ffffffff816707ad>] ? cpuidle_enter_state+0xed/0x230 [ 120.284355] [<ffffffff81670927>] cpuidle_enter+0x17/0x20 [ 120.284360] [<ffffffff810a3862>] call_cpuidle+0x32/0x60 [ 120.284364] [<ffffffff81670903>] ? cpuidle_select+0x13/0x20 [ 120.284369] [<ffffffff810a3aa6>] cpu_startup_entry+0x216/0x2d0 [ 120.284374] [<ffffffff8103679d>] start_secondary+0x12d/0x150 [ 125.426447] device: 'ep_84': device_unregister [ 125.426511] PM: Removing info for No Bus:ep_84 [ 125.426542] device: 'ep_84': device_add [ 125.426602] PM: Adding info for No Bus:ep_84 -- Shuah Khan Sr. Linux Kernel Developer Open Source Innovation Group Samsung Research America (Silicon Valley) shuahkh(a)osg.samsung.com | (970) 217-8978

3 5

[alsa-devel] [PATCH] ASoC: wm0010: fix memory leak
by Sudip Mukherjee 02 Sep '15

02 Sep '15

We were aborting if the kzalloc of img_swap fails but without freeing the already allocated out. Similarly we were aborting if spi_sync fails without releasing out and img_swap. Signed-off-by: Sudip Mukherjee <sudip(a)vectorindia.org> --- sound/soc/codecs/wm0010.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm0010.c b/sound/soc/codecs/wm0010.c index f2c6ad4..8434d45 100644 --- a/sound/soc/codecs/wm0010.c +++ b/sound/soc/codecs/wm0010.c @@ -672,8 +672,10 @@ static int wm0010_boot(struct snd_soc_codec *codec) } img_swap = kzalloc(len, GFP_KERNEL | GFP_DMA); - if (!img_swap) + if (!img_swap) { + kfree(out); goto abort; + } /* We need to re-order for 0010 */ byte_swap_64((u64 *)&pll_rec, img_swap, len); @@ -690,6 +692,8 @@ static int wm0010_boot(struct snd_soc_codec *codec) ret = spi_sync(spi, &m); if (ret != 0) { dev_err(codec->dev, "First PLL write failed: %d\n", ret); + kfree(img_swap); + kfree(out); goto abort; } @@ -697,6 +701,8 @@ static int wm0010_boot(struct snd_soc_codec *codec) ret = spi_sync(spi, &m); if (ret != 0) { dev_err(codec->dev, "Second PLL write failed: %d\n", ret); + kfree(img_swap); + kfree(out); goto abort; } -- 1.9.1

3 2

[alsa-devel] [PATCH] ALSA: hda - Fix missing inline for dummy snd_hdac_set_codec_wakeup()
by Takashi Iwai 02 Sep '15

02 Sep '15

This seems overlooked. Fixes: 98d8fc6c5d36 ('ALSA: hda - Move hda_i915.c from sound/pci/hda to sound/hda') Cc: <stable(a)vger.kernel.org> # v4.2+ Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- include/sound/hda_i915.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/include/sound/hda_i915.h b/include/sound/hda_i915.h index a5b5cae03e87..930b41e5acf4 100644 --- a/include/sound/hda_i915.h +++ b/include/sound/hda_i915.h @@ -14,7 +14,7 @@ int snd_hdac_i915_init(struct hdac_bus *bus); int snd_hdac_i915_exit(struct hdac_bus *bus); int snd_hdac_i915_register_notifier(const struct i915_audio_component_audio_ops *); #else -static int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable) +static inline int snd_hdac_set_codec_wakeup(struct hdac_bus *bus, bool enable) { return 0; } -- 2.5.1

1 0

[alsa-devel] [PATCH v4 0/4] i915 to call hda driver on HDMI plug/unplug
by David Henningsson 02 Sep '15

02 Sep '15

It's been a while since the last patch set iteration, due to me being on vacation. But here's a new set, and I still hope that it can make it into the next merge window. Changes since v3 (with the person suggesting that change within parantheses): * Valleyview now has three pins like all the others (Libin Yang) * Renamed a few references from "hotplug" to "pin_eld" to reduce confusion with hotplug code on the i915 side (Jani Nikula) * Rewrote the dispatching from hda core to codec on the hda side (Takashi Iwai) This iteration has been tested and working on one Skylake machine. David Henningsson (4): drm/i915: Add audio pin sense / ELD callback drm/i915: Call audio pin/ELD notify function ALSA: hda - allow codecs to access the i915 pin/ELD callback ALSA: hda - Wake the codec up on pin/ELD notify events drivers/gpu/drm/i915/i915_drv.h | 1 + drivers/gpu/drm/i915/intel_audio.c | 23 ++++++++++++++++++++--- include/drm/i915_component.h | 12 ++++++++++++ include/sound/hda_i915.h | 7 +++++++ sound/hda/hdac_i915.c | 10 ++++++++++ sound/pci/hda/patch_hdmi.c | 22 +++++++++++++++++++++- 6 files changed, 71 insertions(+), 4 deletions(-) -- 1.9.1

4 18

[alsa-devel] [PATCH] ASoC: intel: broadwell: Convert to devm_snd_soc_register_card
by Axel Lin 02 Sep '15

02 Sep '15

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/intel/boards/broadwell.c | 9 +-------- 1 file changed, 1 insertion(+), 8 deletions(-) diff --git a/sound/soc/intel/boards/broadwell.c b/sound/soc/intel/boards/broadwell.c index 8bafaf6..3f8a1e1 100644 --- a/sound/soc/intel/boards/broadwell.c +++ b/sound/soc/intel/boards/broadwell.c @@ -266,18 +266,11 @@ static int broadwell_audio_probe(struct platform_device *pdev) { broadwell_rt286.dev = &pdev->dev; - return snd_soc_register_card(&broadwell_rt286); -} - -static int broadwell_audio_remove(struct platform_device *pdev) -{ - snd_soc_unregister_card(&broadwell_rt286); - return 0; + return devm_snd_soc_register_card(&pdev->dev, &broadwell_rt286); } static struct platform_driver broadwell_audio = { .probe = broadwell_audio_probe, - .remove = broadwell_audio_remove, .driver = { .name = "broadwell-audio", }, -- 2.1.0

1 0

[alsa-devel] [ANNOUNCE] Media Workshop at the Kernel Summit 2015 in Korea
by Mauro Carvalho Chehab 01 Sep '15

01 Sep '15

Hi, As we keep being very active on the development of the Linux Kernel Media subsystem, we'll be doing another workshop this year. This will be the first time we'll be doing it in Asia, so it is a great opportunity for the ones that are there to join us and participate, in order to improve the Linux Kernel support for A/V stream capture, webcam and analog/digital TV. As usual, we'll be using the media-workshop(a)linuxtv.org Mailing List for specific discussions about that, so the ones interested on participate are requested to subscribe it. This time, we'll be doing something different than what we've done on the last years, and use something that other subsystems like ALSA are doing for a while: we'll be using two google docs to track the proposals and the interested parties. If you're interested in participate and wants to be invited for the event, please put your data at this document: https://docs.google.com/spreadsheets/d/1YecCAUo0L1R_i3jPg3S-A5iqX7BYAd2c_Vf… As this is the first time we'll be using Google Docs, I kindly request you to also send an email for media-workshop(a)linuxtv.org if you have any troubles adding things there and/or a backup measure. If you have any proposal for themes, please add it here: https://docs.google.com/spreadsheets/d/1F67VcdhkDlVfRpx-GPhe61adZh81Q26q--_… It is useful if you could also send an email to the workshop ML in order to better explain the theme proposal. Hoping to see you there soon! Mauro

3 3

[alsa-devel] [PATCH] ASoC: sh: siu_dai: Convert to use resource managed APIs
by Axel Lin 01 Sep '15

01 Sep '15

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/sh/siu_dai.c | 85 ++++++++++++++------------------------------------ 1 file changed, 24 insertions(+), 61 deletions(-) diff --git a/sound/soc/sh/siu_dai.c b/sound/soc/sh/siu_dai.c index abb0d95..76b2ab8 100644 --- a/sound/soc/sh/siu_dai.c +++ b/sound/soc/sh/siu_dai.c @@ -738,7 +738,7 @@ static int siu_probe(struct platform_device *pdev) struct siu_info *info; int ret; - info = kmalloc(sizeof(*info), GFP_KERNEL); + info = devm_kmalloc(&pdev->dev, sizeof(*info), GFP_KERNEL); if (!info) return -ENOMEM; siu_i2s_data = info; @@ -746,7 +746,7 @@ static int siu_probe(struct platform_device *pdev) ret = request_firmware(&fw_entry, "siu_spb.bin", &pdev->dev); if (ret) - goto ereqfw; + return ret; /* * Loaded firmware is "const" - read only, but we have to modify it in @@ -757,89 +757,52 @@ static int siu_probe(struct platform_device *pdev) release_firmware(fw_entry); res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - ret = -ENODEV; - goto egetres; - } + if (!res) + return -ENODEV; - region = request_mem_region(res->start, resource_size(res), - pdev->name); + region = devm_request_mem_region(&pdev->dev, res->start, + resource_size(res), pdev->name); if (!region) { dev_err(&pdev->dev, "SIU region already claimed\n"); - ret = -EBUSY; - goto ereqmemreg; + return -EBUSY; } - ret = -ENOMEM; - info->pram = ioremap(res->start, PRAM_SIZE); + info->pram = devm_ioremap(&pdev->dev, res->start, PRAM_SIZE); if (!info->pram) - goto emappram; - info->xram = ioremap(res->start + XRAM_OFFSET, XRAM_SIZE); + return -ENOMEM; + info->xram = devm_ioremap(&pdev->dev, res->start + XRAM_OFFSET, + XRAM_SIZE); if (!info->xram) - goto emapxram; - info->yram = ioremap(res->start + YRAM_OFFSET, YRAM_SIZE); + return -ENOMEM; + info->yram = devm_ioremap(&pdev->dev, res->start + YRAM_OFFSET, + YRAM_SIZE); if (!info->yram) - goto emapyram; - info->reg = ioremap(res->start + REG_OFFSET, resource_size(res) - - REG_OFFSET); + return -ENOMEM; + info->reg = devm_ioremap(&pdev->dev, res->start + REG_OFFSET, + resource_size(res) - REG_OFFSET); if (!info->reg) - goto emapreg; + return -ENOMEM; dev_set_drvdata(&pdev->dev, info); /* register using ARRAY version so we can keep dai name */ - ret = snd_soc_register_component(&pdev->dev, &siu_i2s_component, - &siu_i2s_dai, 1); + ret = devm_snd_soc_register_component(&pdev->dev, &siu_i2s_component, + &siu_i2s_dai, 1); if (ret < 0) - goto edaiinit; + return ret; - ret = snd_soc_register_platform(&pdev->dev, &siu_platform); + ret = devm_snd_soc_register_platform(&pdev->dev, &siu_platform); if (ret < 0) - goto esocregp; + return ret; pm_runtime_enable(&pdev->dev); - return ret; - -esocregp: - snd_soc_unregister_component(&pdev->dev); -edaiinit: - iounmap(info->reg); -emapreg: - iounmap(info->yram); -emapyram: - iounmap(info->xram); -emapxram: - iounmap(info->pram); -emappram: - release_mem_region(res->start, resource_size(res)); -ereqmemreg: -egetres: -ereqfw: - kfree(info); - - return ret; + return 0; } static int siu_remove(struct platform_device *pdev) { - struct siu_info *info = dev_get_drvdata(&pdev->dev); - struct resource *res; - pm_runtime_disable(&pdev->dev); - - snd_soc_unregister_platform(&pdev->dev); - snd_soc_unregister_component(&pdev->dev); - - iounmap(info->reg); - iounmap(info->yram); - iounmap(info->xram); - iounmap(info->pram); - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (res) - release_mem_region(res->start, resource_size(res)); - kfree(info); - return 0; } -- 2.1.0

1 0

[alsa-devel] SSM2518 Device Tree support ?
by Ricard Wanderlof 01 Sep '15

01 Sep '15

We needed DT support in the SSM2518 codec driver, and I noticed that there is already a set of bindings specified in Documentation/devicetree, however, there doesn't seem to be any DT support in the driver itself. I don't know if this is the result of a missing commit ? Documentation/devicetree/bindings/sound/ssm2518.c seems to have been added in May 2013. I've added rudimentary support for at least the compatible property so that the driver can be used in a system with DT configuration (sent as a separate email). /Ricard -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 4

[alsa-devel] [PATCH v4] OMAPDSS: hdmi: Reconfigure and restart audio when display is enabled
by Jyri Sarha 01 Sep '15

01 Sep '15

Reconfigure and restart audio when display is enabled, if audio playback was active before. This is needed in a situation when an audio+video stream application opens the audio stream before the video. When video stream is opened the display mode may change and that aborts audio playback, because the display is momentarily turned off. The audio configuration is stored when it is successfully applied and a boolean is set when the audio playback is started and unset when stopped. This data is used to reconfigure the audio when display is re-enabled. The audio playback is aborted if the reconfiguration fails. A new spin lock is introduced in order to protect state variables related to audio playback status. This is needed for the transition from display enabled state (when audio start/stop commands can be written to HW) to display disabled state (when audio start/stop commands update only the hdmi.audio_playing variable) to always serialize correctly with the start/stop audio commands. The already existing mutex can not be used, because the audio start and stop commands are executed in atomic context. For example: when display is turned back on we take the spinlock and we can be sure that the audio start/stop status will not change while we update the HW according to hdmi.audio_playing state and set hdmi.display_enabled to true. After releasing the lock hdmi.display_enabled is true and all audio_start and audio_stop commands write their stuff directly to HW. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- Fixed one bad sentece in the description. drivers/video/fbdev/omap2/dss/hdmi.h | 9 ++++- drivers/video/fbdev/omap2/dss/hdmi4.c | 66 +++++++++++++++++++++++++----- drivers/video/fbdev/omap2/dss/hdmi5.c | 76 ++++++++++++++++++++++++++++------- 3 files changed, 124 insertions(+), 27 deletions(-) diff --git a/drivers/video/fbdev/omap2/dss/hdmi.h b/drivers/video/fbdev/omap2/dss/hdmi.h index e4a32fe..53616b0 100644 --- a/drivers/video/fbdev/omap2/dss/hdmi.h +++ b/drivers/video/fbdev/omap2/dss/hdmi.h @@ -351,13 +351,20 @@ struct omap_hdmi { struct regulator *vdda_reg; bool core_enabled; - bool display_enabled; struct omap_dss_device output; struct platform_device *audio_pdev; void (*audio_abort_cb)(struct device *dev); int wp_idlemode; + + bool audio_configured; + struct omap_dss_audio audio_config; + + /* This lock should be taken when booleans bellow are touched. */ + spinlock_t audio_playing_lock; + bool audio_playing; + bool display_enabled; }; #endif diff --git a/drivers/video/fbdev/omap2/dss/hdmi4.c b/drivers/video/fbdev/omap2/dss/hdmi4.c index 6d3aa3f..94c8d55 100644 --- a/drivers/video/fbdev/omap2/dss/hdmi4.c +++ b/drivers/video/fbdev/omap2/dss/hdmi4.c @@ -321,9 +321,22 @@ static int read_edid(u8 *buf, int len) return r; } +static void hdmi_start_audio_stream(struct omap_hdmi *hd) +{ + hdmi_wp_audio_enable(&hd->wp, true); + hdmi4_audio_start(&hd->core, &hd->wp); +} + +static void hdmi_stop_audio_stream(struct omap_hdmi *hd) +{ + hdmi4_audio_stop(&hd->core, &hd->wp); + hdmi_wp_audio_enable(&hd->wp, false); +} + static int hdmi_display_enable(struct omap_dss_device *dssdev) { struct omap_dss_device *out = &hdmi.output; + unsigned long flags; int r = 0; DSSDBG("ENTER hdmi_display_enable\n"); @@ -342,7 +355,21 @@ static int hdmi_display_enable(struct omap_dss_device *dssdev) goto err0; } + if (hdmi.audio_configured) { + r = hdmi4_audio_config(&hdmi.core, &hdmi.wp, &hdmi.audio_config, + hdmi.cfg.timings.pixelclock); + if (r) { + DSSERR("Error restoring audio configuration: %d", r); + hdmi.audio_abort_cb(&hdmi.pdev->dev); + hdmi.audio_configured = false; + } + } + + spin_lock_irqsave(&hdmi.audio_playing_lock, flags); + if (hdmi.audio_configured && hdmi.audio_playing) + hdmi_start_audio_stream(&hdmi); hdmi.display_enabled = true; + spin_unlock_irqrestore(&hdmi.audio_playing_lock, flags); mutex_unlock(&hdmi.lock); return 0; @@ -354,17 +381,19 @@ err0: static void hdmi_display_disable(struct omap_dss_device *dssdev) { + unsigned long flags; + DSSDBG("Enter hdmi_display_disable\n"); mutex_lock(&hdmi.lock); - if (hdmi.audio_pdev && hdmi.audio_abort_cb) - hdmi.audio_abort_cb(&hdmi.audio_pdev->dev); + spin_lock_irqsave(&hdmi.audio_playing_lock, flags); + hdmi_stop_audio_stream(&hdmi); + hdmi.display_enabled = false; + spin_unlock_irqrestore(&hdmi.audio_playing_lock, flags); hdmi_power_off_full(dssdev); - hdmi.display_enabled = false; - mutex_unlock(&hdmi.lock); } @@ -568,6 +597,8 @@ static int hdmi_audio_shutdown(struct device *dev) mutex_lock(&hd->lock); hd->audio_abort_cb = NULL; + hd->audio_configured = false; + hd->audio_playing = false; mutex_unlock(&hd->lock); return 0; @@ -576,25 +607,34 @@ static int hdmi_audio_shutdown(struct device *dev) static int hdmi_audio_start(struct device *dev) { struct omap_hdmi *hd = dev_get_drvdata(dev); + unsigned long flags; WARN_ON(!hdmi_mode_has_audio(&hd->cfg)); - WARN_ON(!hd->display_enabled); - hdmi_wp_audio_enable(&hd->wp, true); - hdmi4_audio_start(&hd->core, &hd->wp); + spin_lock_irqsave(&hd->audio_playing_lock, flags); + + if (hd->display_enabled) + hdmi_start_audio_stream(hd); + hd->audio_playing = true; + spin_unlock_irqrestore(&hd->audio_playing_lock, flags); return 0; } static void hdmi_audio_stop(struct device *dev) { struct omap_hdmi *hd = dev_get_drvdata(dev); + unsigned long flags; WARN_ON(!hdmi_mode_has_audio(&hd->cfg)); - WARN_ON(!hd->display_enabled); - hdmi4_audio_stop(&hd->core, &hd->wp); - hdmi_wp_audio_enable(&hd->wp, false); + spin_lock_irqsave(&hd->audio_playing_lock, flags); + + if (hd->display_enabled) + hdmi_stop_audio_stream(hd); + hd->audio_playing = false; + + spin_unlock_irqrestore(&hd->audio_playing_lock, flags); } static int hdmi_audio_config(struct device *dev, @@ -612,7 +652,10 @@ static int hdmi_audio_config(struct device *dev, ret = hdmi4_audio_config(&hd->core, &hd->wp, dss_audio, hd->cfg.timings.pixelclock); - + if (!ret) { + hd->audio_configured = true; + hd->audio_config = *dss_audio; + } out: mutex_unlock(&hd->lock); @@ -657,6 +700,7 @@ static int hdmi4_bind(struct device *dev, struct device *master, void *data) dev_set_drvdata(&pdev->dev, &hdmi); mutex_init(&hdmi.lock); + spin_lock_init(&hdmi.audio_playing_lock); if (pdev->dev.of_node) { r = hdmi_probe_of(pdev); diff --git a/drivers/video/fbdev/omap2/dss/hdmi5.c b/drivers/video/fbdev/omap2/dss/hdmi5.c index 7f87578..b6c767b 100644 --- a/drivers/video/fbdev/omap2/dss/hdmi5.c +++ b/drivers/video/fbdev/omap2/dss/hdmi5.c @@ -349,9 +349,24 @@ static int read_edid(u8 *buf, int len) return r; } +static void hdmi_start_audio_stream(struct omap_hdmi *hd) +{ + REG_FLD_MOD(hdmi.wp.base, HDMI_WP_SYSCONFIG, 1, 3, 2); + hdmi_wp_audio_enable(&hd->wp, true); + hdmi_wp_audio_core_req_enable(&hd->wp, true); +} + +static void hdmi_stop_audio_stream(struct omap_hdmi *hd) +{ + hdmi_wp_audio_core_req_enable(&hd->wp, false); + hdmi_wp_audio_enable(&hd->wp, false); + REG_FLD_MOD(hd->wp.base, HDMI_WP_SYSCONFIG, hd->wp_idlemode, 3, 2); +} + static int hdmi_display_enable(struct omap_dss_device *dssdev) { struct omap_dss_device *out = &hdmi.output; + unsigned long flags; int r = 0; DSSDBG("ENTER hdmi_display_enable\n"); @@ -370,7 +385,21 @@ static int hdmi_display_enable(struct omap_dss_device *dssdev) goto err0; } + if (hdmi.audio_configured) { + r = hdmi5_audio_config(&hdmi.core, &hdmi.wp, &hdmi.audio_config, + hdmi.cfg.timings.pixelclock); + if (r) { + DSSERR("Error restoring audio configuration: %d", r); + hdmi.audio_abort_cb(&hdmi.pdev->dev); + hdmi.audio_configured = false; + } + } + + spin_lock_irqsave(&hdmi.audio_playing_lock, flags); + if (hdmi.audio_configured && hdmi.audio_playing) + hdmi_start_audio_stream(&hdmi); hdmi.display_enabled = true; + spin_unlock_irqrestore(&hdmi.audio_playing_lock, flags); mutex_unlock(&hdmi.lock); return 0; @@ -382,17 +411,19 @@ err0: static void hdmi_display_disable(struct omap_dss_device *dssdev) { + unsigned long flags; + DSSDBG("Enter hdmi_display_disable\n"); mutex_lock(&hdmi.lock); - if (hdmi.audio_pdev && hdmi.audio_abort_cb) - hdmi.audio_abort_cb(&hdmi.audio_pdev->dev); + spin_lock_irqsave(&hdmi.audio_playing_lock, flags); + hdmi_stop_audio_stream(&hdmi); + hdmi.display_enabled = false; + spin_unlock_irqrestore(&hdmi.audio_playing_lock, flags); hdmi_power_off_full(dssdev); - hdmi.display_enabled = false; - mutex_unlock(&hdmi.lock); } @@ -596,6 +627,8 @@ static int hdmi_audio_shutdown(struct device *dev) mutex_lock(&hd->lock); hd->audio_abort_cb = NULL; + hd->audio_configured = false; + hd->audio_playing = false; mutex_unlock(&hd->lock); return 0; @@ -604,32 +637,35 @@ static int hdmi_audio_shutdown(struct device *dev) static int hdmi_audio_start(struct device *dev) { struct omap_hdmi *hd = dev_get_drvdata(dev); + unsigned long flags; WARN_ON(!hdmi_mode_has_audio(&hd->cfg)); - WARN_ON(!hd->display_enabled); - /* No-idle while playing audio, store the old value */ - hd->wp_idlemode = REG_GET(hdmi.wp.base, HDMI_WP_SYSCONFIG, 3, 2); - REG_FLD_MOD(hdmi.wp.base, HDMI_WP_SYSCONFIG, 1, 3, 2); + spin_lock_irqsave(&hd->audio_playing_lock, flags); - hdmi_wp_audio_enable(&hd->wp, true); - hdmi_wp_audio_core_req_enable(&hd->wp, true); + if (hd->display_enabled) { + hdmi_start_audio_stream(hd); + } + hd->audio_playing = true; + spin_unlock_irqrestore(&hd->audio_playing_lock, flags); return 0; } static void hdmi_audio_stop(struct device *dev) { struct omap_hdmi *hd = dev_get_drvdata(dev); + unsigned long flags; WARN_ON(!hdmi_mode_has_audio(&hd->cfg)); - WARN_ON(!hd->display_enabled); - hdmi_wp_audio_core_req_enable(&hd->wp, false); - hdmi_wp_audio_enable(&hd->wp, false); + spin_lock_irqsave(&hd->audio_playing_lock, flags); + + if (hd->display_enabled) + hdmi_stop_audio_stream(hd); + hd->audio_playing = false; - /* Playback stopped, restore original idlemode */ - REG_FLD_MOD(hdmi.wp.base, HDMI_WP_SYSCONFIG, hd->wp_idlemode, 3, 2); + spin_unlock_irqrestore(&hd->audio_playing_lock, flags); } static int hdmi_audio_config(struct device *dev, @@ -648,6 +684,10 @@ static int hdmi_audio_config(struct device *dev, ret = hdmi5_audio_config(&hd->core, &hd->wp, dss_audio, hd->cfg.timings.pixelclock); + if (!ret) { + hd->audio_configured = true; + hd->audio_config = *dss_audio; + } out: mutex_unlock(&hd->lock); @@ -678,6 +718,11 @@ static int hdmi_audio_register(struct device *dev) if (IS_ERR(hdmi.audio_pdev)) return PTR_ERR(hdmi.audio_pdev); + hdmi_runtime_get(); + hdmi.wp_idlemode = + REG_GET(hdmi.wp.base, HDMI_WP_SYSCONFIG, 3, 2); + hdmi_runtime_put(); + return 0; } @@ -692,6 +737,7 @@ static int hdmi5_bind(struct device *dev, struct device *master, void *data) dev_set_drvdata(&pdev->dev, &hdmi); mutex_init(&hdmi.lock); + spin_lock_init(&hdmi.audio_playing_lock); if (pdev->dev.of_node) { r = hdmi_probe_of(pdev); -- 1.9.1

2 1

[alsa-devel] Beaglebone Black: Transmit buffer underflows when trying to play audio using an externally attached codec board (simple-card, davinci-mcasp, cs4270)
by Shadi Abdu-Rahman 01 Sep '15

01 Sep '15

Hi all, I'm having trouble using an audio codec board with a custom device tree overlay (see below) with the Beaglebone Black (kernel 4.1.3-bone15). When exporting the overlay, the codecs seem to register fine: dmesg: [ 1375.144463] bone_capemgr bone_capemgr: part_number 'BB-BONE-XA-SK-AU', version 'N/A' [ 1375.152305] bone_capemgr bone_capemgr: slot #4: override [ 1375.157799] bone_capemgr bone_capemgr: Using override eeprom data at slot 4 [ 1375.164864] bone_capemgr bone_capemgr: slot #4: 'Override Board Name,00A0,Override Manuf,BB-BONE-XA-SK-AU' [ 1375.206351] bone_capemgr bone_capemgr: slot #4: dtbo 'BB-BONE-XA-SK-AU-00A0.dtbo' loaded; overlay id #0 [ 1375.464260] cs4270 2-0048: found device at i2c address 48 [ 1375.469699] cs4270 2-0048: hardware revision 3 [ 1375.485782] cs4270 2-0049: found device at i2c address 49 [ 1375.491222] cs4270 2-0049: hardware revision 3 [ 1375.504744] (NULL device *): Setting gpio_int_masterclk_enable = 0 [ 1375.513334] asoc-simple-card ocp:sound: cs4270-hifi <-> 48038000.mcasp mapping ok [ 1375.521095] (NULL device *): Setting gpio_int_masterclk_enable = 0 [ 1375.529893] asoc-simple-card ocp:sound: cs4270-hifi <-> 48038000.mcasp mapping ok aplay -l: **** List of PLAYBACK Hardware Devices **** card 0: XMOSAudioSlice [XMOS-Audio-Slice], device 0: davinci-mcasp.0-cs4270-hifi-a cs4270-hifi-0 [] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: XMOSAudioSlice [XMOS-Audio-Slice], device 1: davinci-mcasp.0-cs4270-hifi-b cs4270-hifi-1 [] Subdevices: 1/1 Subdevice #0: subdevice #0 lsmod: snd_soc_davinci_mcasp 14391 2 snd_soc_simple_card 8156 0 snd_soc_edma 1150 1 snd_soc_davinci_mcasp snd_soc_cs4270 6875 2 snd_soc_omap 2573 1 snd_soc_davinci_mcasp snd_soc_core 156881 5 snd_soc_davinci_mcasp,snd_soc_edma,snd_soc_omap,snd_soc_simple_card,snd_soc_cs4270 snd_compress 11668 1 snd_soc_core snd_pcm_dmaengine 5061 2 snd_soc_core,snd_soc_omap snd_pcm 77081 4 snd_soc_davinci_mcasp,snd_soc_core,snd_soc_omap,snd_pcm_dmaengine snd_timer 16860 1 snd_pcm snd 56902 4 snd_soc_core,snd_timer,snd_pcm,snd_compress soundcore 6869 1 snd pvrsrvkm 147014 0 omap_aes 13089 0 omap_sham 19190 0 tda998x 11683 1 tilcdc 27919 0 omap_rng 4354 0 rng_core 7270 1 omap_rng drm_kms_helper 106610 3 tda998x,tilcdc uio_pdrv_genirq 3317 0 leds_gpio 3102 0 uio 8330 1 uio_pdrv_genirq|| || | |but when I try to play an audio file I get a stream of transmit buffer underflows and no audio: aplay test128.wav: Playing WAVE 'test128.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Stereo underrun!!! (at least 1.630 ms long) underrun!!! (at least 5.413 ms long) underrun!!! (at least 0.636 ms long) underrun!!! (at least 5.369 ms long) underrun!!! (at least 0.217 ms long) underrun!!! (at least 0.641 ms long) underrun!!! (at least 5.254 ms long) dmesg: [ 1493.190031] davinci-mcasp 48038000.mcasp: Transmit buffer underflow [ 1493.201647] davinci-mcasp 48038000.mcasp: Transmit buffer underflow [ 1493.218719] davinci-mcasp 48038000.mcasp: Transmit buffer underflow [ 1493.229946] davinci-mcasp 48038000.mcasp: Transmit buffer underflow [ 1493.245328] davinci-mcasp 48038000.mcasp: Transmit buffer underflow [ 1493.259411] davinci-mcasp 48038000.mcasp: Transmit buffer underflow [ 1493.270549] davinci-mcasp 48038000.mcasp: Transmit buffer underflow Using a software oscilloscope, I can see that the frame clock starts on mcasp0_fsx frame clock line but stops after a few periods. There’s no activity on the on the mcasp0_axr2 audio out line. I'm not sure if I'm missing something trivial or if this is a general with audio out on the Beaglebone Black and 4.1.x kernels. Any pointers or help would be much appreciated. Additional information: I’m using the v4.1.3-bone15 linux kernel with debian-8.1-minimal-armhf-2015-06-09 compiled as per Robert C Nelsons instructions (https://eewiki.net/display/linuxonarm/BeagleBone+Black). The board I’m using is an XMOS XA-SK-AUDIO slice and consists of two Cirrus Logic cs4270 codecs and an onboard oscillator generating the master clock. I’ve interfaced the board to a XMOS micro controller to make sure the board is in working order (and it is). When connected to the BBB, reset de-assertion and some initial i2c codec configuration (setting the codecs in software mode) are handled by the same XMOS micro controller. The overlay I’m using looks like this: /dts-v1/; /plugin/; / { compatible = "ti,beaglebone", "ti,beaglebone-black"; part-number = "BB-BONE-XA-SK-AU"; version = "00A0"; exclusive-use = /* pin header usage */ "P9.25", /* mcasp0_ahclkx <- MCLK */ "P9.31", /* mcasp0_aclkx <-> BCLK*/ "P9.29", /* mcasp0_fsx <-> LRCLK */ "P9.30", /* mcasp0_axr0 <- ADC_DATA0 */ "P9.41", /* mcasp0_axr1 <- ADC_DATA1 */ "P9.28", /* mcasp0_axr2 -> DAC_DATA0 */ "P9.27", /* mcasp0_axr3 -> DAC_DATA1 */ /* hardware ip usage */ "gpio0_15", /* ext-mclk-sel -> MCLK_FSEL */ "gpio0_20", /* CLKOUT2 disabled */ "gpio1_27", /* internal 24.576 MHz oscillator disabled */ "mcasp0"; fragment@0 { target = <&am33xx_pinmux>; __overlay__ { i2c2_pins: pinmux_i2c2_pins { pinctrl-single,pins = < /* Pullup resistors available on codec board */ 0x17c 0x2b /* i2c2_scl, (PIN_INPUT | MUX_MODE3) */ 0x178 0x2b /* i2c2_sda, (PIN_INPUT | MUX_MODE3) */ >; }; mcasp0_pins: pinmux_mcasp0_pins { pinctrl-single,pins = < 0x1ac 0x20 /* mcasp0_ahclkx, (PIN_INPUT_PULLDOWN | MUX_MODE0) */ 0x190 0x20 /* mcasp0_aclkx, (PIN_INPUT_PULLDOWN | MUX_MODE0) */ 0x194 0x20 /* mcasp0_fsx, (PIN_INPUT_PULLDOWN | MUX_MODE0) */ 0x198 0x20 /* mcasp0_axr0, (PIN_INPUT_PULLDOWN | MUX_MODE0) */ 0x1a8 0x20 /* mcasp0_axr1, (PIN_INPUT_PULLDOWN | MUX_MODE0) */ 0x19c 0x02 /* mcasp0_axr2, (PIN_OUTPUT_PULLDOWN | MUX_MODE2) */ 0x1a4 0x02 /* mcasp0_axr3, (PIN_OUTPUT_PULLDOWN | MUX_MODE2) */ /* disable internal 24.576 MHz oscillator */ 0x06c 0x07 /* gpio1_27, (PIN_OUTPUT_PULLDOWN | MUX_MODE7) */ >; }; gpio_pins: pinmux_gpio_pins { pinctrl-single,pins = < /* ext-mclk-sel */ 0x184 0x17 /* gpio0_15, (PIN_OUTPUT_PULLUP | MUX_MODE7) */ /* disable CLKOUT2 to allow usage of mcasp0_axr1 on P9_41 */ 0x1b4 0x2b /* gpio0_20, (PIN_INPUT | MUX_MODE3) */ >; }; }; }; fragment@1 { target = <&i2c2>; __overlay__ { pinctrl-names = "default"; pinctrl-0 = <&i2c2_pins>; status = "okay"; #address-cells = <1>; #size-cells = <0>; clock-frequency = <100000>; cs4270m: cs4270@48 { compatible = "cirrus,cs4270"; status = "okay"; #sound-dai-cells = <0>; reg = <0x48>; name-prefix = "a"; va-supply = <&ldo4_reg>; vd-supply = <&ldo4_reg>; vlc-supply = <&ldo4_reg>; }; cs4270s: cs4270@49 { compatible = "cirrus,cs4270"; status = "okay"; #sound-dai-cells = <0>; reg = <0x49>; name-prefix = "b"; va-supply = <&ldo4_reg>; vd-supply = <&ldo4_reg>; vlc-supply = <&ldo4_reg>; }; }; }; fragment@2 { target = <&mcasp0>; __overlay__ { pinctrl-names = "default"; pinctrl-0 = <&mcasp0_pins>; status = "okay"; #sound-dai-cells = <0>; op-mode = <0>; // MCASP_IIS_MODE tdm-slots = <2>; num-serializer = <4>; serial-dir = < // 0: INACTIVE, 1: TX, 2: RX 2 2 1 1 >; tx-num-evt = <1>; rx-num-evt = <1>; }; }; fragment@3 { target = <&ocp>; __overlay__ { sound { compatible = "simple-audio-card"; simple-audio-card,name = "XMOS-Audio-Slice"; simple-audio-card,widgets = "Line", "Line In 1", "Line", "Line Out 1", "Line", "Line In 2", "Line", "Line Out 2" ; simple-audio-card,routing = "a AINA", "Line In 1", "a AINB", "Line In 1", "Line Out 1", "a AOUTA", "Line Out 1", "a AOUTB", "b AINA", "Line In 2", "b AINB", "Line In 2", "Line Out 2", "b AOUTA", "Line Out 2", "b AOUTB" ; simple-audio-card,int-masterclk-enable = <&gpio1 27 0>; simple-audio-card,dai-link@1 { format = "i2s"; frame-master = <&cpu_node1>; bitclock-master = <&cpu_node1>; cpu_node1: cpu { sound-dai = <&mcasp0>; system-clock-frequency = <24576000>; }; codec_node1: codec { sound-dai = <&cs4270s>; system-clock-frequency = <24576000>; }; }; simple-audio-card,dai-link@0 { format = "i2s"; frame-master = <&cpu_node0>; bitclock-master = <&cpu_node0>; cpu_node0: cpu { sound-dai = <&mcasp0>; system-clock-frequency = <24576000>; }; codec_node0: codec { sound-dai = <&cs4270m>; system-clock-frequency = <24576000>; }; }; }; }; }; }; I've also made modifications to cs4270.c, simple-card.c, and soc-core.c to add a mechanism to disable the internal BBB master clock (in simple-card.c), as well as to fix issues with missing source/sink widgets (this seems to be a bug in cs4270.c): [ 137.408549] cs4270 2-0048: ASoC: no source widget found for AINA [ 137.414737] cs4270 2-0048: ASoC: Failed to add route AINA -> direct -> Capture [ 137.422016] cs4270 2-0048: ASoC: no source widget found for AINB [ 137.428137] cs4270 2-0048: ASoC: Failed to add route AINB -> direct -> Capture [ 137.435477] cs4270 2-0048: ASoC: no sink widget found for AOUTA [ 137.441445] cs4270 2-0048: ASoC: Failed to add route Playback -> direct -> AOUTA [ 137.448928] cs4270 2-0048: ASoC: no sink widget found for AOUTB [ 137.454922] cs4270 2-0048: ASoC: Failed to add route Playback -> direct -> AOUTB and name collisions, arising from using two identical codecs (by hacking together name-prefix support in simple-card): [ 51.364492] asoc-simple-card ocp:sound: control 2:0:0:Master Playback Volume:0 is already present [ 51.373469] cs4270 2-0049: ASoC: Failed to add Master Playback Volume: -16 ... [ 51.393905] ------------[ cut here ]------------ [ 51.398561] WARNING: CPU: 0 PID: 44 at fs/sysfs/dir.c:31 sysfs_warn_dup+0x51/0x5c() [ 51.406281] sysfs: cannot create duplicate filename '/devices/platform/ocp/ocp:sound/davinci-mcasp.0-cs4270-hifi' The modifications are as follows: diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c index e6d4ff9..4805747 100644 --- a/sound/soc/codecs/cs4270.c +++ b/sound/soc/codecs/cs4270.c @@ -140,11 +140,11 @@ struct cs4270_private { }; static const struct snd_soc_dapm_widget cs4270_dapm_widgets[] = { -SND_SOC_DAPM_INPUT("AINL"), -SND_SOC_DAPM_INPUT("AINR"), +SND_SOC_DAPM_INPUT("AINA"), +SND_SOC_DAPM_INPUT("AINB"), -SND_SOC_DAPM_OUTPUT("AOUTL"), -SND_SOC_DAPM_OUTPUT("AOUTR"), +SND_SOC_DAPM_OUTPUT("AOUTA"), +SND_SOC_DAPM_OUTPUT("AOUTB"), }; static const struct snd_soc_dapm_route cs4270_dapm_routes[] = { diff --git a/sound/soc/generic/simple-card.c b/sound/soc/generic/simple-card.c index 33feee9..1383b0f 100644 --- a/sound/soc/generic/simple-card.c +++ b/sound/soc/generic/simple-card.c @@ -28,6 +28,7 @@ struct simple_card_data { struct asoc_simple_dai codec_dai; } *dai_props; unsigned int mclk_fs; + int gpio_int_masterclk_enable; int gpio_hp_det; int gpio_hp_det_invert; int gpio_mic_det; @@ -50,7 +51,7 @@ static int asoc_simple_card_startup(struct snd_pcm_substream *substream) ret = clk_prepare_enable(dai_props->cpu_dai.clk); if (ret) return ret; - + ret = clk_prepare_enable(dai_props->codec_dai.clk); if (ret) clk_disable_unprepare(dai_props->cpu_dai.clk); @@ -167,6 +168,12 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) if (ret < 0) return ret; + /* TODO: Logic for enabling internal clock if specified in DT*/ + if (gpio_is_valid(priv->gpio_int_masterclk_enable)) { + dev_info(NULL, "Setting gpio_int_masterclk_enable = 0"); + gpio_set_value(priv->gpio_int_masterclk_enable, 0); + } + if (gpio_is_valid(priv->gpio_hp_det)) { snd_soc_card_jack_new(rtd->card, "Headphones", SND_JACK_HEADPHONE, @@ -195,16 +202,18 @@ static int asoc_simple_card_dai_init(struct snd_soc_pcm_runtime *rtd) } static int -asoc_simple_card_sub_parse_of(struct device_node *np, - struct asoc_simple_dai *dai, - struct device_node **p_node, - const char **name, - int *args_count) +asoc_simple_card_sub_parse_of(struct simple_card_data *priv, + struct device_node *np, + struct asoc_simple_dai *dai, + struct device_node **p_node, + const char **name, + int *args_count, bool codec_device, int idx) { struct of_phandle_args args; struct clk *clk; u32 val; int ret; + const char *str = NULL; /* * Get node via "sound-dai = <&phandle port>" @@ -225,6 +234,18 @@ asoc_simple_card_sub_parse_of(struct device_node *np, if (ret < 0) return ret; + /* get name-prefix */ + if (codec_device) { + /* NULL bypasses the device name check in soc-core */ + priv->snd_card.codec_conf[idx].dev_name = NULL; + priv->snd_card.codec_conf[idx].of_node = *p_node; + of_property_read_string(*p_node, "name-prefix", &str); + if (str) + priv->snd_card.codec_conf[idx].name_prefix = str; + else + priv->snd_card.codec_conf[idx].name_prefix = NULL; + } + /* Parse TDM slot */ ret = snd_soc_of_parse_tdm_slot(np, &dai->slots, &dai->slot_width); if (ret) @@ -334,16 +355,16 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, if (ret < 0) goto dai_link_of_err; - ret = asoc_simple_card_sub_parse_of(cpu, &dai_props->cpu_dai, + ret = asoc_simple_card_sub_parse_of(priv, cpu, &dai_props->cpu_dai, &dai_link->cpu_of_node, &dai_link->cpu_dai_name, - &cpu_args); + &cpu_args, false, idx); if (ret < 0) goto dai_link_of_err; - ret = asoc_simple_card_sub_parse_of(codec, &dai_props->codec_dai, + ret = asoc_simple_card_sub_parse_of(priv, codec, &dai_props->codec_dai, &dai_link->codec_of_node, - &dai_link->codec_dai_name, NULL); + &dai_link->codec_dai_name, NULL, true, idx); if (ret < 0) goto dai_link_of_err; @@ -357,8 +378,10 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, /* DAI link name is created from CPU/CODEC dai name */ name = devm_kzalloc(dev, - strlen(dai_link->cpu_dai_name) + - strlen(dai_link->codec_dai_name) + 2, + strlen(dai_link->cpu_dai_name) + + strlen(dai_link->codec_dai_name) + + (priv->snd_card.codec_conf[idx].name_prefix ? + strlen(priv->snd_card.codec_conf[idx].name_prefix) + 3 : 2), GFP_KERNEL); if (!name) { ret = -ENOMEM; @@ -367,6 +390,9 @@ static int asoc_simple_card_dai_link_of(struct device_node *node, sprintf(name, "%s-%s", dai_link->cpu_dai_name, dai_link->codec_dai_name); + if (priv->snd_card.codec_conf[idx].name_prefix) + sprintf(name, "%s-%s", name, + priv->snd_card.codec_conf[idx].name_prefix); dai_link->name = dai_link->stream_name = name; dai_link->ops = &asoc_simple_card_ops; dai_link->init = asoc_simple_card_dai_init; @@ -459,6 +485,15 @@ static int asoc_simple_card_parse_of(struct device_node *node, return ret; } + /* request GPIO to control internal 24.576MHz oscillator */ + priv->gpio_int_masterclk_enable = of_get_named_gpio_flags(node, + "simple-audio-card,int-masterclk-enable", 0, &flags); + + if (priv->gpio_int_masterclk_enable == -EPROBE_DEFER) + return -EPROBE_DEFER; + gpio_request_one(priv->gpio_int_masterclk_enable, GPIOF_OUT_INIT_LOW, + "Internal Clock Enable Pin"); + priv->gpio_hp_det = of_get_named_gpio_flags(node, "simple-audio-card,hp-det-gpio", 0, &flags); priv->gpio_hp_det_invert = !!(flags & OF_GPIO_ACTIVE_LOW); @@ -498,13 +533,18 @@ static int asoc_simple_card_probe(struct platform_device *pdev) struct snd_soc_dai_link *dai_link; struct device_node *np = pdev->dev.of_node; struct device *dev = &pdev->dev; - int num_links, ret; + int num_links = (np && of_get_child_by_name(np, + "simple-audio-card,dai-link")) ? + of_get_child_count(np) : 1; + struct snd_soc_codec_conf codec_conf[num_links]; + int ret; - /* Get the number of DAI links */ + /* Get the number of DAI links if (np && of_get_child_by_name(np, "simple-audio-card,dai-link")) num_links = of_get_child_count(np); else num_links = 1; + */ /* Allocate the private data and the DAI link array */ priv = devm_kzalloc(dev, @@ -519,6 +559,8 @@ static int asoc_simple_card_probe(struct platform_device *pdev) dai_link = priv->dai_link; priv->snd_card.dai_link = dai_link; priv->snd_card.num_links = num_links; + priv->snd_card.codec_conf = codec_conf; + priv->snd_card.num_configs = num_links; priv->gpio_hp_det = -ENOENT; priv->gpio_mic_det = -ENOENT; @@ -589,6 +631,11 @@ static int asoc_simple_card_remove(struct platform_device *pdev) struct snd_soc_card *card = platform_get_drvdata(pdev); struct simple_card_data *priv = snd_soc_card_get_drvdata(card); + if (gpio_is_valid(priv->gpio_int_masterclk_enable)) { + gpio_set_value(priv->gpio_int_masterclk_enable, 0); + gpio_free(priv->gpio_int_masterclk_enable); + } + if (gpio_is_valid(priv->gpio_hp_det)) snd_soc_jack_free_gpios(&simple_card_hp_jack, 1, &simple_card_hp_jack_gpio); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 2373252..4808089 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1102,6 +1102,9 @@ static void soc_set_name_prefix(struct snd_soc_card *card, for (i = 0; i < card->num_configs; i++) { struct snd_soc_codec_conf *map = &card->codec_conf[i]; + + if (!map->name_prefix) + continue; if (map->of_node && component->dev->of_node != map->of_node) continue; if (map->dev_name && strcmp(component->name, map->dev_name)) Another possible bug: Adding name-prefix support to simple-card uncovers a possible bug. You'll notice in the device tree overlay that I'v listed the simple-audio-card dai-links in reverse order, with dai-link@1 before dai-link@0. I've done so because using the normal order the wrong prefix gets assigned to the wrong dai-link. I'm not really sure where this happens, and it's not a huge issue for me, but I thought I'd report it. Kind Regards Shadi Abdu-Rahman

1 0

[alsa-devel] [PATCH RFT] ASoC: pxa: Convert to devm_snd_soc_register_card
by Axel Lin 01 Sep '15

01 Sep '15

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Only compile tested. sound/soc/pxa/brownstone.c | 9 +-------- sound/soc/pxa/corgi.c | 11 +---------- sound/soc/pxa/e740_wm9705.c | 5 +---- sound/soc/pxa/e750_wm9705.c | 5 +---- sound/soc/pxa/e800_wm9712.c | 5 +---- sound/soc/pxa/hx4700.c | 4 +--- sound/soc/pxa/imote2.c | 11 +---------- sound/soc/pxa/mioa701_wm9713.c | 11 +---------- sound/soc/pxa/palm27x.c | 9 +-------- sound/soc/pxa/poodle.c | 11 +---------- sound/soc/pxa/spitz.c | 5 +---- sound/soc/pxa/tosa.c | 5 +---- sound/soc/pxa/ttc-dkb.c | 12 +----------- 13 files changed, 13 insertions(+), 90 deletions(-) diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c index 2b26318..6147e86 100644 --- a/sound/soc/pxa/brownstone.c +++ b/sound/soc/pxa/brownstone.c @@ -116,26 +116,19 @@ static int brownstone_probe(struct platform_device *pdev) int ret; brownstone.dev = &pdev->dev; - ret = snd_soc_register_card(&brownstone); + ret = devm_snd_soc_register_card(&pdev->dev, &brownstone); if (ret) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); return ret; } -static int brownstone_remove(struct platform_device *pdev) -{ - snd_soc_unregister_card(&brownstone); - return 0; -} - static struct platform_driver mmp_driver = { .driver = { .name = "brownstone-audio", .pm = &snd_soc_pm_ops, }, .probe = brownstone_probe, - .remove = brownstone_remove, }; module_platform_driver(mmp_driver); diff --git a/sound/soc/pxa/corgi.c b/sound/soc/pxa/corgi.c index 3580d10..c97dc13 100644 --- a/sound/soc/pxa/corgi.c +++ b/sound/soc/pxa/corgi.c @@ -295,28 +295,19 @@ static int corgi_probe(struct platform_device *pdev) card->dev = &pdev->dev; - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); return ret; } -static int corgi_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - return 0; -} - static struct platform_driver corgi_driver = { .driver = { .name = "corgi-audio", .pm = &snd_soc_pm_ops, }, .probe = corgi_probe, - .remove = corgi_remove, }; module_platform_driver(corgi_driver); diff --git a/sound/soc/pxa/e740_wm9705.c b/sound/soc/pxa/e740_wm9705.c index d72e124..1de8765 100644 --- a/sound/soc/pxa/e740_wm9705.c +++ b/sound/soc/pxa/e740_wm9705.c @@ -138,7 +138,7 @@ static int e740_probe(struct platform_device *pdev) card->dev = &pdev->dev; - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); @@ -149,10 +149,7 @@ static int e740_probe(struct platform_device *pdev) static int e740_remove(struct platform_device *pdev) { - struct snd_soc_card *card = platform_get_drvdata(pdev); - gpio_free_array(e740_audio_gpios, ARRAY_SIZE(e740_audio_gpios)); - snd_soc_unregister_card(card); return 0; } diff --git a/sound/soc/pxa/e750_wm9705.c b/sound/soc/pxa/e750_wm9705.c index 48f2d7c..b7eb7cd 100644 --- a/sound/soc/pxa/e750_wm9705.c +++ b/sound/soc/pxa/e750_wm9705.c @@ -120,7 +120,7 @@ static int e750_probe(struct platform_device *pdev) card->dev = &pdev->dev; - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); @@ -131,10 +131,7 @@ static int e750_probe(struct platform_device *pdev) static int e750_remove(struct platform_device *pdev) { - struct snd_soc_card *card = platform_get_drvdata(pdev); - gpio_free_array(e750_audio_gpios, ARRAY_SIZE(e750_audio_gpios)); - snd_soc_unregister_card(card); return 0; } diff --git a/sound/soc/pxa/e800_wm9712.c b/sound/soc/pxa/e800_wm9712.c index 45d4bd4..41bf714 100644 --- a/sound/soc/pxa/e800_wm9712.c +++ b/sound/soc/pxa/e800_wm9712.c @@ -119,7 +119,7 @@ static int e800_probe(struct platform_device *pdev) card->dev = &pdev->dev; - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); @@ -130,10 +130,7 @@ static int e800_probe(struct platform_device *pdev) static int e800_remove(struct platform_device *pdev) { - struct snd_soc_card *card = platform_get_drvdata(pdev); - gpio_free_array(e800_audio_gpios, ARRAY_SIZE(e800_audio_gpios)); - snd_soc_unregister_card(card); return 0; } diff --git a/sound/soc/pxa/hx4700.c b/sound/soc/pxa/hx4700.c index 9f8be7c..ecbf287 100644 --- a/sound/soc/pxa/hx4700.c +++ b/sound/soc/pxa/hx4700.c @@ -193,7 +193,7 @@ static int hx4700_audio_probe(struct platform_device *pdev) return ret; snd_soc_card_hx4700.dev = &pdev->dev; - ret = snd_soc_register_card(&snd_soc_card_hx4700); + ret = devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_hx4700); if (ret) gpio_free_array(hx4700_audio_gpios, ARRAY_SIZE(hx4700_audio_gpios)); @@ -203,8 +203,6 @@ static int hx4700_audio_probe(struct platform_device *pdev) static int hx4700_audio_remove(struct platform_device *pdev) { - snd_soc_unregister_card(&snd_soc_card_hx4700); - gpio_set_value(GPIO92_HX4700_HP_DRIVER, 0); gpio_set_value(GPIO107_HX4700_SPK_nSD, 0); diff --git a/sound/soc/pxa/imote2.c b/sound/soc/pxa/imote2.c index 29fabbf..9d0e407 100644 --- a/sound/soc/pxa/imote2.c +++ b/sound/soc/pxa/imote2.c @@ -72,28 +72,19 @@ static int imote2_probe(struct platform_device *pdev) card->dev = &pdev->dev; - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); return ret; } -static int imote2_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - return 0; -} - static struct platform_driver imote2_driver = { .driver = { .name = "imote2-audio", .pm = &snd_soc_pm_ops, }, .probe = imote2_probe, - .remove = imote2_remove, }; module_platform_driver(imote2_driver); diff --git a/sound/soc/pxa/mioa701_wm9713.c b/sound/soc/pxa/mioa701_wm9713.c index a9615a57..29bc60e8 100644 --- a/sound/soc/pxa/mioa701_wm9713.c +++ b/sound/soc/pxa/mioa701_wm9713.c @@ -181,7 +181,7 @@ static int mioa701_wm9713_probe(struct platform_device *pdev) return -ENODEV; mioa701.dev = &pdev->dev; - rc = snd_soc_register_card(&mioa701); + rc = devm_snd_soc_register_card(&pdev->dev, &mioa701); if (!rc) dev_warn(&pdev->dev, "Be warned that incorrect mixers/muxes setup will" "lead to overheating and possible destruction of your device." @@ -189,17 +189,8 @@ static int mioa701_wm9713_probe(struct platform_device *pdev) return rc; } -static int mioa701_wm9713_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - return 0; -} - static struct platform_driver mioa701_wm9713_driver = { .probe = mioa701_wm9713_probe, - .remove = mioa701_wm9713_remove, .driver = { .name = "mioa701-wm9713", .pm = &snd_soc_pm_ops, diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c index c20bbc0..4e74d95 100644 --- a/sound/soc/pxa/palm27x.c +++ b/sound/soc/pxa/palm27x.c @@ -140,22 +140,15 @@ static int palm27x_asoc_probe(struct platform_device *pdev) palm27x_asoc.dev = &pdev->dev; - ret = snd_soc_register_card(&palm27x_asoc); + ret = devm_snd_soc_register_card(&pdev->dev, &palm27x_asoc); if (ret) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); return ret; } -static int palm27x_asoc_remove(struct platform_device *pdev) -{ - snd_soc_unregister_card(&palm27x_asoc); - return 0; -} - static struct platform_driver palm27x_wm9712_driver = { .probe = palm27x_asoc_probe, - .remove = palm27x_asoc_remove, .driver = { .name = "palm27x-asoc", .pm = &snd_soc_pm_ops, diff --git a/sound/soc/pxa/poodle.c b/sound/soc/pxa/poodle.c index 80b457a..84d0e2e 100644 --- a/sound/soc/pxa/poodle.c +++ b/sound/soc/pxa/poodle.c @@ -267,28 +267,19 @@ static int poodle_probe(struct platform_device *pdev) card->dev = &pdev->dev; - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); return ret; } -static int poodle_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - return 0; -} - static struct platform_driver poodle_driver = { .driver = { .name = "poodle-audio", .pm = &snd_soc_pm_ops, }, .probe = poodle_probe, - .remove = poodle_remove, }; module_platform_driver(poodle_driver); diff --git a/sound/soc/pxa/spitz.c b/sound/soc/pxa/spitz.c index 461123a..b002226 100644 --- a/sound/soc/pxa/spitz.c +++ b/sound/soc/pxa/spitz.c @@ -305,7 +305,7 @@ static int spitz_probe(struct platform_device *pdev) card->dev = &pdev->dev; - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); @@ -322,9 +322,6 @@ err1: static int spitz_remove(struct platform_device *pdev) { - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); gpio_free(spitz_mic_gpio); return 0; } diff --git a/sound/soc/pxa/tosa.c b/sound/soc/pxa/tosa.c index f59f566..49518dd 100644 --- a/sound/soc/pxa/tosa.c +++ b/sound/soc/pxa/tosa.c @@ -233,7 +233,7 @@ static int tosa_probe(struct platform_device *pdev) card->dev = &pdev->dev; - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); @@ -244,10 +244,7 @@ static int tosa_probe(struct platform_device *pdev) static int tosa_remove(struct platform_device *pdev) { - struct snd_soc_card *card = platform_get_drvdata(pdev); - gpio_free(TOSA_GPIO_L_MUTE); - snd_soc_unregister_card(card); return 0; } diff --git a/sound/soc/pxa/ttc-dkb.c b/sound/soc/pxa/ttc-dkb.c index 1753c7d..65c20f7 100644 --- a/sound/soc/pxa/ttc-dkb.c +++ b/sound/soc/pxa/ttc-dkb.c @@ -128,7 +128,7 @@ static int ttc_dkb_probe(struct platform_device *pdev) card->dev = &pdev->dev; - ret = snd_soc_register_card(card); + ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n", ret); @@ -136,22 +136,12 @@ static int ttc_dkb_probe(struct platform_device *pdev) return ret; } -static int ttc_dkb_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - - snd_soc_unregister_card(card); - - return 0; -} - static struct platform_driver ttc_dkb_driver = { .driver = { .name = "ttc-dkb-audio", .pm = &snd_soc_pm_ops, }, .probe = ttc_dkb_probe, - .remove = ttc_dkb_remove, }; module_platform_driver(ttc_dkb_driver); -- 2.1.0

1 0

[alsa-devel] feature request: alsamixer should show alternatives to function key bindings, at least F1/Help
by frederik＠ofb.net 01 Sep '15

01 Sep '15

In alsamixer, many options are bound to function keys. These are used by my window manager, so I cannot use them in my terminal. The alsamixer man page mentions that 'gnome-terminal' cannot use function keys either. I noticed from the "Help" screen that many function keys have non-function-key alternatives, for instance instead of F6 I can press "s". This is great, but there is one remaining problem: The legend (at the top of the alsamixer screen) labels commands with their function-key bindings. This includes "Help" which is labelled with "F1". Seeing this, users are likely to assume that "F1" is the only binding for "Help". But it turns out that one can also access "Help" with "?" and "H". This can be learned by reading the man page or pressing keys experimentally, but it would be nice if "Help" was just labeled with either "?" or "H", as that allows people who are unable to use function keys to easily access the help screen and then see that alternate (non-function-key) bindings exist for other commands as well. As an aside, it seems odd to map "q" to "increase left volume"... (from https://bugs.freedesktop.org/show_bug.cgi?id=91782, edited)

1 0

[alsa-devel] feature request: "mute" indication in alsamixer is not self-evident
by frederik＠ofb.net 01 Sep '15

01 Sep '15

When a user presses 'm' for 'mute' in alsamixer, then a string "MM" appears at the bottom of a volume control. This change makes it obvious to the user that "MM" stands for "Mute". However, when a new user opens alsamixer after some software on his computer has (perhaps by mistake) automatically muted a volume control, the "MM" won't indicate anything obvious. Among other things, this makes it difficult for new users to debug issues with the auto-mute feature. The fact that the volume control can be at maximum, even when the control is muted, contributes to the potential for confusion. The problem would be solved if, for instance, the "Item: " string at the top of the alsamixer screen were changed to contain the word "(muted)" for muted controls. This way, users who don't have sound will see what the problem is when they open alsamixer. (from <https://bugs.freedesktop.org/show_bug.cgi?id=91781>, edited)

1 0

[alsa-devel] bug tracker down?
by frederik＠ofb.net 01 Sep '15

01 Sep '15

Hi guys, For some reason I can't connect to https://bugtrack.alsa-project.org/alsa-bug I want to file two bugs which I mistakenly sent to the Pulseaudio bug tracker, having to do with alsamixer. They are here and here: https://bugs.freedesktop.org/show_bug.cgi?id=91781 https://bugs.freedesktop.org/show_bug.cgi?id=91782 Thanks, Frederick

2 2

[alsa-devel] [PATCH] ASoC: fsl_ssi: Fix checking of dai format for AC97 mode
by Adam Thomson 31 Aug '15

31 Aug '15

Current code incorrectly treats dai format for AC97 as bit mask whereas it's actually an integer value. This causes DAI formats other than AC97 (e.g. DSP_B) to trigger AC97 related code, which is incorrect and breaks functionality. This patch fixes the code to correctly compare values to determine AC97 or not. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- sound/soc/fsl/fsl_ssi.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index c0b940e..ee47041 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -249,7 +249,8 @@ MODULE_DEVICE_TABLE(of, fsl_ssi_ids); static bool fsl_ssi_is_ac97(struct fsl_ssi_private *ssi_private) { - return !!(ssi_private->dai_fmt & SND_SOC_DAIFMT_AC97); + return (ssi_private->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) == + SND_SOC_DAIFMT_AC97; } static bool fsl_ssi_is_i2s_master(struct fsl_ssi_private *ssi_private) @@ -945,7 +946,7 @@ static int _fsl_ssi_set_dai_fmt(struct device *dev, CCSR_SSI_SCR_TCH_EN); } - if (fmt & SND_SOC_DAIFMT_AC97) + if ((fmt & SND_SOC_DAIFMT_FORMAT_MASK) == SND_SOC_DAIFMT_AC97) fsl_ssi_setup_ac97(ssi_private); return 0; -- 1.9.3

1 0

[alsa-devel] [PATCH] reset: Add (devm_)reset_control_get stub functions
by Axel Lin 31 Aug '15

31 Aug '15

So the drivers can be compiled with CONFIG_RESET_CONTROLLER disabled. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- include/linux/reset.h | 12 ++++++++++++ 1 file changed, 12 insertions(+) diff --git a/include/linux/reset.h b/include/linux/reset.h index da5602b..cd42c1b 100644 --- a/include/linux/reset.h +++ b/include/linux/reset.h @@ -74,6 +74,18 @@ static inline int device_reset_optional(struct device *dev) return -ENOSYS; } +static inline struct reset_control *reset_control_get( + struct device *dev, const char *id) +{ + return ERR_PTR(-ENOSYS); +} + +static inline struct reset_control *devm_reset_control_get( + struct device *dev, const char *id) +{ + return ERR_PTR(-ENOSYS); +} + static inline struct reset_control *reset_control_get_optional( struct device *dev, const char *id) { -- 2.1.0

3 2

[alsa-devel] [PATCH] ASoC: Document snd-soc-dummy-dai purpose
by Anatol Pomozov 31 Aug '15

31 Aug '15

Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- sound/soc/soc-utils.c | 9 +++++++++ 1 file changed, 9 insertions(+) diff --git a/sound/soc/soc-utils.c b/sound/soc/soc-utils.c index 362c69a..53dd085 100644 --- a/sound/soc/soc-utils.c +++ b/sound/soc/soc-utils.c @@ -101,6 +101,15 @@ static struct snd_soc_codec_driver dummy_codec; SNDRV_PCM_FMTBIT_S32_LE | \ SNDRV_PCM_FMTBIT_U32_LE | \ SNDRV_PCM_FMTBIT_IEC958_SUBFRAME_LE) +/* + * The dummy CODEC is only meant to be used in situations where there is no + * actual hardware. + * + * If there is actual hardware even if it does not have a control bus + * the hardware will still have constraints like supported samplerates, etc. + * which should be modelled. And the data flow graph also should be modelled + * using DAPM. + */ static struct snd_soc_dai_driver dummy_dai = { .name = "snd-soc-dummy-dai", .playback = { -- 2.5.0.457.gab17608

2 1

[alsa-devel] [PATCH 1/4] ASoC: au1x: Convert to devm_snd_soc_register_card
by Axel Lin 31 Aug '15

31 Aug '15

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/au1x/db1000.c | 10 +--------- sound/soc/au1x/db1200.c | 10 +--------- 2 files changed, 2 insertions(+), 18 deletions(-) diff --git a/sound/soc/au1x/db1000.c b/sound/soc/au1x/db1000.c index 452f404..e97c327 100644 --- a/sound/soc/au1x/db1000.c +++ b/sound/soc/au1x/db1000.c @@ -38,14 +38,7 @@ static int db1000_audio_probe(struct platform_device *pdev) { struct snd_soc_card *card = &db1000_ac97; card->dev = &pdev->dev; - return snd_soc_register_card(card); -} - -static int db1000_audio_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - snd_soc_unregister_card(card); - return 0; + return devm_snd_soc_register_card(&pdev->dev, card); } static struct platform_driver db1000_audio_driver = { @@ -54,7 +47,6 @@ static struct platform_driver db1000_audio_driver = { .pm = &snd_soc_pm_ops, }, .probe = db1000_audio_probe, - .remove = db1000_audio_remove, }; module_platform_driver(db1000_audio_driver); diff --git a/sound/soc/au1x/db1200.c b/sound/soc/au1x/db1200.c index 58c3164..638ca0b 100644 --- a/sound/soc/au1x/db1200.c +++ b/sound/soc/au1x/db1200.c @@ -174,14 +174,7 @@ static int db1200_audio_probe(struct platform_device *pdev) card = db1200_cards[pid->driver_data]; card->dev = &pdev->dev; - return snd_soc_register_card(card); -} - -static int db1200_audio_remove(struct platform_device *pdev) -{ - struct snd_soc_card *card = platform_get_drvdata(pdev); - snd_soc_unregister_card(card); - return 0; + return devm_snd_soc_register_card(&pdev->dev, card); } static struct platform_driver db1200_audio_driver = { @@ -191,7 +184,6 @@ static struct platform_driver db1200_audio_driver = { }, .id_table = db1200_pids, .probe = db1200_audio_probe, - .remove = db1200_audio_remove, }; module_platform_driver(db1200_audio_driver); -- 2.1.0

3 5

[alsa-devel] [PATCH 2/2] ASoC: fsl-asoc-card: add AC'97 support
by Maciej S. Szmigiero 31 Aug '15

31 Aug '15

Add AC'97 support to fsl-asoc-card using generic ASoC AC'97 CODEC. The SSI controller will silently enable any TX AC'97 slots that have their bits set in SLOTREQ received from CODEC and then will redirect some of playback samples there. That's why it is important to make sure that any of CODEC playback slots that can pull samples are set to slots 3/4 (standard PCM playback slots). Currently, this applies to S/PDIF slots as they were seen to pull samples sometimes even with S/PDIF output being disabled. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- .../devicetree/bindings/sound/fsl-asoc-card.txt | 10 +- sound/soc/fsl/fsl-asoc-card.c | 145 +++++++++++++++---- 2 files changed, 120 insertions(+), 35 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt index a96774c..ce55c0a 100644 --- a/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt +++ b/Documentation/devicetree/bindings/sound/fsl-asoc-card.txt @@ -13,13 +13,15 @@ So having this generic sound card allows all Freescale SoC users to benefit from the simplification of a new card support and the capability of the wide sample rates support through ASRC. -Note: The card is initially designed for those sound cards who use I2S and - PCM DAI formats. However, it'll be also possible to support those non - I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as long - as the driver has been properly upgraded. +Note: The card is initially designed for those sound cards who use AC'97, I2S + and PCM DAI formats. However, it'll be also possible to support those non + AC'97/I2S/PCM type sound cards, such as S/PDIF audio and HDMI audio, as + long as the driver has been properly upgraded. The compatible list for this generic sound card currently: + "fsl,imx-audio-ac97" + "fsl,imx-audio-cs42888" "fsl,imx-audio-wm8962" diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 96f55ae..385a9f8 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -14,6 +14,9 @@ #include <linux/i2c.h> #include <linux/module.h> #include <linux/of_platform.h> +#if IS_ENABLED(CONFIG_SND_AC97_CODEC) +#include <sound/ac97_codec.h> +#endif #include <sound/pcm_params.h> #include <sound/soc.h> @@ -115,6 +118,11 @@ static const struct snd_soc_dapm_widget fsl_asoc_card_dapm_widgets[] = { SND_SOC_DAPM_MIC("DMIC", NULL), }; +static bool fsl_asoc_card_is_ac97(struct fsl_asoc_card_priv *priv) +{ + return priv->dai_fmt == SND_SOC_DAIFMT_AC97; +} + static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params) { @@ -133,7 +141,9 @@ static int fsl_asoc_card_hw_params(struct snd_pcm_substream *substream, * set_bias_level(), bypass the remaining settings in hw_params(). * Note: (dai_fmt & CBM_CFM) includes CBM_CFM and CBM_CFS. */ - if (priv->card.set_bias_level && priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) + if ((priv->card.set_bias_level && + priv->dai_fmt & SND_SOC_DAIFMT_CBM_CFM) || + fsl_asoc_card_is_ac97(priv)) return 0; /* Specific configurations of DAIs starts from here */ @@ -300,7 +310,7 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, ext_port--; /* - * Use asynchronous mode (6 wires) for all cases. + * Use asynchronous mode (6 wires) for all cases except AC97. * If only 4 wires are needed, just set SSI into * synchronous mode and enable 4 PADs in IOMUX. */ @@ -346,15 +356,30 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, IMX_AUDMUX_V2_PTCR_TCLKDIR; break; default: - return -EINVAL; + if (!fsl_asoc_card_is_ac97(priv)) + return -EINVAL; + } + + if (fsl_asoc_card_is_ac97(priv)) { + int_ptcr = IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TCSEL(ext_port) | + IMX_AUDMUX_V2_PTCR_TCLKDIR; + ext_ptcr = IMX_AUDMUX_V2_PTCR_SYN | + IMX_AUDMUX_V2_PTCR_TFSEL(int_port) | + IMX_AUDMUX_V2_PTCR_TFSDIR; } /* Asynchronous mode can not be set along with RCLKDIR */ - ret = imx_audmux_v2_configure_port(int_port, 0, - IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port)); - if (ret) { - dev_err(dev, "audmux internal port setup failed\n"); - return ret; + if (!fsl_asoc_card_is_ac97(priv)) { + unsigned int pdcr = + IMX_AUDMUX_V2_PDCR_RXDSEL(ext_port); + + ret = imx_audmux_v2_configure_port(int_port, 0, + pdcr); + if (ret) { + dev_err(dev, "audmux internal port setup failed\n"); + return ret; + } } ret = imx_audmux_v2_configure_port(int_port, int_ptcr, @@ -364,11 +389,16 @@ static int fsl_asoc_card_audmux_init(struct device_node *np, return ret; } - ret = imx_audmux_v2_configure_port(ext_port, 0, - IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); - if (ret) { - dev_err(dev, "audmux external port setup failed\n"); - return ret; + if (!fsl_asoc_card_is_ac97(priv)) { + unsigned int pdcr = + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port); + + ret = imx_audmux_v2_configure_port(ext_port, 0, + pdcr); + if (ret) { + dev_err(dev, "audmux external port setup failed\n"); + return ret; + } } ret = imx_audmux_v2_configure_port(ext_port, ext_ptcr, @@ -389,6 +419,23 @@ static int fsl_asoc_card_late_probe(struct snd_soc_card *card) struct device *dev = card->dev; int ret; + if (fsl_asoc_card_is_ac97(priv)) { +#if IS_ENABLED(CONFIG_SND_AC97_CODEC) + struct snd_soc_codec *codec = card->rtd[0].codec; + struct snd_ac97 *ac97 = snd_soc_codec_get_drvdata(codec); + + /* + * Use slots 3/4 for S/PDIF so SSI won't try to enable + * other slots and send some samples there + * due to SLOTREQ bits for S/PDIF received from codec + */ + snd_ac97_update_bits(ac97, AC97_EXTENDED_STATUS, + AC97_EA_SPSA_SLOT_MASK, AC97_EA_SPSA_3_4); +#endif + + return 0; + } + ret = snd_soc_dai_set_sysclk(codec_dai, codec_priv->mclk_id, codec_priv->mclk_freq, SND_SOC_CLOCK_IN); if (ret) { @@ -404,10 +451,9 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) struct device_node *cpu_np, *codec_np, *asrc_np; struct device_node *np = pdev->dev.of_node; struct platform_device *asrc_pdev = NULL; - struct platform_device *cpu_pdev; + struct platform_device *cpu_pdev, *codec_pdev; struct fsl_asoc_card_priv *priv; - struct i2c_client *codec_dev; - struct clk *codec_clk; + struct i2c_client *codec_i2c_dev; const char *codec_dai_name; u32 width; int ret; @@ -420,9 +466,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Give a chance to old DT binding */ if (!cpu_np) cpu_np = of_parse_phandle(np, "ssi-controller", 0); - codec_np = of_parse_phandle(np, "audio-codec", 0); - if (!cpu_np || !codec_np) { - dev_err(&pdev->dev, "phandle missing or invalid\n"); + if (!cpu_np) { + dev_err(&pdev->dev, "CPU phandle missing or invalid\n"); ret = -EINVAL; goto fail; } @@ -434,11 +479,13 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) goto fail; } - codec_dev = of_find_i2c_device_by_node(codec_np); - if (!codec_dev) { - dev_err(&pdev->dev, "failed to find codec platform device\n"); - ret = -EINVAL; - goto fail; + codec_np = of_parse_phandle(np, "audio-codec", 0); + if (codec_np) { + codec_pdev = of_find_device_by_node(codec_np); + codec_i2c_dev = of_find_i2c_device_by_node(codec_np); + } else { + codec_pdev = NULL; + codec_i2c_dev = NULL; } asrc_np = of_parse_phandle(np, "audio-asrc", 0); @@ -446,10 +493,13 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) asrc_pdev = of_find_device_by_node(asrc_np); /* Get the MCLK rate only, and leave it controlled by CODEC drivers */ - codec_clk = clk_get(&codec_dev->dev, NULL); - if (!IS_ERR(codec_clk)) { - priv->codec_priv.mclk_freq = clk_get_rate(codec_clk); - clk_put(codec_clk); + if (codec_pdev) { + struct clk *codec_clk = clk_get(&codec_pdev->dev, NULL); + + if (!IS_ERR(codec_clk)) { + priv->codec_priv.mclk_freq = clk_get_rate(codec_clk); + clk_put(codec_clk); + } } /* Default sample rate and format, will be updated in hw_params() */ @@ -486,12 +536,22 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->codec_priv.fll_id = WM8960_SYSCLK_AUTO; priv->codec_priv.pll_id = WM8960_SYSCLK_AUTO; priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; + } else if (of_device_is_compatible(np, "fsl,imx-audio-ac97")) { + codec_dai_name = "ac97-hifi"; + priv->card.set_bias_level = NULL; + priv->dai_fmt = SND_SOC_DAIFMT_AC97; } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); ret = -EINVAL; goto asrc_fail; } + if (!fsl_asoc_card_is_ac97(priv) && !codec_pdev) { + dev_err(&pdev->dev, "failed to find codec platform device\n"); + ret = -EINVAL; + goto asrc_fail; + } + /* Common settings for corresponding Freescale CPU DAI driver */ if (strstr(cpu_np->name, "ssi")) { /* Only SSI needs to configure AUDMUX */ @@ -508,7 +568,9 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->cpu_priv.sysclk_id[0] = FSL_SAI_CLK_MAST1; } - sprintf(priv->name, "%s-audio", codec_dev->name); + snprintf(priv->name, sizeof(priv->name), "%s-audio", + fsl_asoc_card_is_ac97(priv) ? "ac97" : + codec_i2c_dev ? codec_i2c_dev->name : codec_pdev->name); /* Initialize sound card */ priv->pdev = pdev; @@ -532,8 +594,26 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) /* Normal DAI Link */ priv->dai_link[0].cpu_of_node = cpu_np; - priv->dai_link[0].codec_of_node = codec_np; priv->dai_link[0].codec_dai_name = codec_dai_name; + + if (!fsl_asoc_card_is_ac97(priv)) + priv->dai_link[0].codec_of_node = codec_np; + else { + u32 idx; + + ret = of_property_read_u32(cpu_np, "cell-index", &idx); + if (ret) { + dev_err(&pdev->dev, + "cannot get CPU index property\n"); + goto asrc_fail; + } + + priv->dai_link[0].codec_name = + devm_kasprintf(&pdev->dev, GFP_KERNEL, + "ac97-codec.%u", + (unsigned int)idx); + } + priv->dai_link[0].platform_of_node = cpu_np; priv->dai_link[0].dai_fmt = priv->dai_fmt; priv->card.num_links = 1; @@ -544,6 +624,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->dai_link[1].platform_of_node = asrc_np; priv->dai_link[2].codec_dai_name = codec_dai_name; priv->dai_link[2].codec_of_node = codec_np; + priv->dai_link[2].codec_name = + priv->dai_link[0].codec_name; priv->dai_link[2].cpu_of_node = cpu_np; priv->dai_link[2].dai_fmt = priv->dai_fmt; priv->card.num_links = 3; @@ -579,14 +661,15 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) asrc_fail: of_node_put(asrc_np); -fail: of_node_put(codec_np); +fail: of_node_put(cpu_np); return ret; } static const struct of_device_id fsl_asoc_card_dt_ids[] = { + { .compatible = "fsl,imx-audio-ac97", }, { .compatible = "fsl,imx-audio-cs42888", }, { .compatible = "fsl,imx-audio-sgtl5000", }, { .compatible = "fsl,imx-audio-wm8962", },

1 0

[alsa-devel] [PATCH 1/2] ASoC: fsl-asoc-card: put ASRC OF node in case of unknown device
by Maciej S. Szmigiero 31 Aug '15

31 Aug '15

In case of unknown DT compatible device the ASRC OF node possibly acquired earlier by of_parse_phandle() has to be put before returning from probe method. Signed-off-by: Maciej Szmigiero <mail(a)maciej.szmigiero.name> --- sound/soc/fsl/fsl-asoc-card.c | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/fsl-asoc-card.c b/sound/soc/fsl/fsl-asoc-card.c index 5aeb6ed..96f55ae 100644 --- a/sound/soc/fsl/fsl-asoc-card.c +++ b/sound/soc/fsl/fsl-asoc-card.c @@ -488,7 +488,8 @@ static int fsl_asoc_card_probe(struct platform_device *pdev) priv->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; } else { dev_err(&pdev->dev, "unknown Device Tree compatible\n"); - return -EINVAL; + ret = -EINVAL; + goto asrc_fail; } /* Common settings for corresponding Freescale CPU DAI driver */

2 1

[alsa-devel] [GIT PULL] ASoC updates for v4.3
by Mark Brown 31 Aug '15

31 Aug '15

The following changes since commit c13dcf9f2d6f5f06ef1bf79ec456df614c5e058b: Linux 4.2-rc8 (2015-08-23 20:52:59 -0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v4.2-rc8 for you to fetch changes up to 91931320cfbbcc1dd874c2d0aef62e7d90f07f70: Merge remote-tracking branches 'asoc/topic/wm8996', 'asoc/topic/xtensa' and 'asoc/topic/zx296702' into asoc-next (2015-08-30 15:59:18 +0100) ---------------------------------------------------------------- ASoC: Updates for v4.3 Not many updates to the core here, but an awful lot of driver updates this time round: - Factoring out of AC'97 reset code into the core - New drivers for Cirrus CS4349, GTM601, InvenSense ICS43432, Realtek RT298 and ST STI controllers. - Machine drivers for Rockchip systems with MAX98090 and RT5645 and RT5650. - Initial driver support for Intel Skylake devices. - A large number of cleanups for Lars-Peter Clausen and Axel Lin. ---------------------------------------------------------------- Anatol Pomozov (4): ASoC: max98357a: Make 'sdmode-gpios' dts property optional ASoC: max98357a: Use DAI input as a dapm widget ASoC: max98357a: Do not print error message on asoc/gpio errors ASoC: rl6231: Simplify DMIC divider calculation expression Andrew Morton (1): ASoC: cs4349: include gpio/consumer.h Andrzej Hajda (2): ASoC: intel: use kmemdup rather than duplicating its implementation ASoC: topology: use kmemdup rather than duplicating its implementation Antonio Borneo (1): ASoC: wm0010: Remove redundant spi driver bus initialization Arnaud Pouliquen (12): ASoC: simple card: set cpu-dai sysclk with mclk-fs ASoC: sti-sas: Add sti platform codec ASoC: sti: add binding for ASoC driver ASoC: sti: Add uniperipheral header file ASoC: sti: Add CPU DAI driver for playback ASoC: sti-sas: fix x86 compilation issue. ASoC: sti: minor corrections for uniplayer ASoC: sti: Add CPU DAI driver for capture ASoC: sti: Add uniperipheral dai driver ASoC: Add ability to build sti drivers ASoC: sti: Add clock adjustement control ASoC: sti: Add IEC control Axel Lin (73): ASoC: gtm601: Fix modalias ASoC: rt5645: Constify dmi_system_id table ASoC: cs42xx8: Move the code checking *regmap argument earlier ASoC: cs42xx8: Setup of_match_table ASoC: 88pm860x: Don't change pm860x->dir setting if pm860x_set_dai_sysclk fails ASoC: ak4642: Fix up max_register setting ASoC: lm49453: Remove fs_rate from struct lm49453_priv ASoC: gtm601: Constify soc_codec_dev_gtm601 ASoC: Constify reg_default tables ASoC: rt286: Constify dmi_system_id table ASoC: wm0010: Remove spurious missing IRQF_ONESHOT ASoC: wm5100: Remove spurious IRQF_ONESHOT flag ASoC: wm8996: Remove spurious IRQF_ONESHOT flag ASoC: sti-sas: Fix checking return value for ERR_PTR ASoC: sti-sas: Staticise local symbols ASoC: Constify snd_soc_dai_ops variables ASoC: da732x: Merge da732x_dai1_ops and da732x_dai2_ops ASoC: sirf-audio-codec: Staticise local symbols ASoC: Constify dev_pm_ops variables ASoC: cs4349: Fix max_register setting for cs4349_regmap ASoC: cs4349: Constify cs4349_regmap ASoC: cs4349: Fix setting digital interface format ASoC: cs4349: Set .writeable_reg for cs4349_regmap ASoC: cs4349: Drop platform data support ASoC: cs4349: Remove unneeded NULL test for cs4349->reset_gpio ASoC: cs4349: Fix up setting PWR_DWN bit ASoC: cs35l32: Fix define for CS35L32_SDOUT_3ST ASoC: cs35l32: Remove unneeded NULL test for cs35l32->reset_gpio ASoC: cs42l73: Fix mask for setting CS42L73_SP_3ST bit ASoC: rt5677: Remove NULL test for desc before gpiod_set_value_cansleep call ASoC: tas2552: Fix off-by-one for max_register setting ASoC: tas5086: Improve the logic for de-emphasis sampling rate selection ASoC: pcm1681: Improve the logic for de-emphasis sampling rate selection ASoC: sta529: Remove redundant I2C_FUNC_SMBUS_BYTE_DATA functionality check ASoC: cs4265: CS4265_INT_STATUS is readable register ASoC: Use params_width() at appropriate places ASoC: rt5677: Return error if devm_gpiod_get_optional return ERR_PTR ASoC: tas2552: Remove unneeded NULL test for tas2552->enable_gpio ASoC: max98088: Get rid of max98088_access table ASoC: max98095: Get rid of max98095_access table ASoC: wm8983: Get rid of wm8983_access_masks table ASoC: max98090: Simplify max98090_readable_register implementation ASoC: wm8741: Simplify wm8741_set_dai_sysclk implementation ASoC: wm8731: Drop wm8731_writeable function ASoC: wm8753: Drop wm8753_writeable function ASoC: topology: Use kcalloc instead of kzalloc for array allocation ASoC: adav80x: Remove .read_flag_mask setting from adav80x_regmap_config ASoC: cs35l32: Use case range at appropriate places ASoC: cs42l52: Use case range at appropriate place ASoC: cs42l56: Use case range at appropriate place ASoC: cs42l73: Use case range at appropriate place ASoC: wm8741: Drop misleading readable_reg callback implementation ASoC: Set missing card owner field ASoC: davinci: Convert to use devm_ioremap_resource ASoC: omap-mcbsp: Convert to use devm_ioremap_resource ASoC: SPEAr: Convert to use devm_ioremap_resource ASoC: tegra: Fix unused variable 'spdif' warning ASoC: tegra: Use devm_ioremap_resource instead of open code ASoC: ics43432: Fix module license ASoC: pxa-ssp: Convert to devm_snd_soc_register_component ASoC: pxa2xx-i2s: Convert to devm_snd_soc_register_component ASoC: sh: ssi: Convert to devm_snd_soc_register_component ASoC: qcom: Constify asoc_qcom_lpass_cpu_dai_ops ASoC: au1x: Convert to devm_snd_soc_register_platform ASoC: blackfin: Convert to devm_snd_soc_register_platform ASoC: nuc900: Convert to devm_snd_soc_register_platform ASoC: pxa: Convert to devm_snd_soc_register_platform ASoC: txx9: Convert to devm_snd_soc_register_platform ASoC: spear_pcm: Use devm_snd_dmaengine_pcm_register to fix resource leak ASoC: sh: dma-sh7760: Convert to devm_snd_soc_register_platform ASoC: au1x: psc-i2s: Convert to use devm_ioremap_resource ASoC: sti_uniperif: Ensure component is unregistered when unload module ASoC: zx296702-i2s: Fix resource leak when unload module Bard Liao (4): ASoC: rt5645: move RT5645 muxes to rt5645_specific_dapm_widgets ASoC: add rt298 codec driver ASoC: rt298: remove meanless pr_info ASoC: rl6231: add pll preset table Ben Zhang (3): ASoC: rt5677: Switch to use descriptor-based gpiod API ASoC: rt5677: Switch to use unified device property API ASoC: rt5677: Allow arbitrary block read/write via SPI Charles Keepax (8): ASoC: arizona: Implement stability check for EQ coefficients ASoC: arizona: Implement stability check for LHPF coefficients mfd: wm5110: Add registers for custom write sequence triggers mfd: wm5110: Add register patch for rev E and above ASoC: wm5110: Add special DRE on/off handling for the headphone path ASoC: arizona: Use the more idiomatic params_width ASoC: arizona: Fix error path in codec probe ASoC: arizona: Poll for FLL clock OK rather than use interrupts Dan Carpenter (3): ASoC: fsl: fsl_spdif: signedness bug in fsl_spdif_startup() ASoC: max98925: reading beyond the end of the array ASoC: sti: error handling bug in sti_uniperiph_cpu_dai_of() Fabio Estevam (6): ASoC: fsl: fsl_asrc: Check for clk_prepare_enable() error ASoC: fsl: fsl_spdif: Check for clk_prepare_enable() error ASoC: max98090: Check for clk_prepare_enable() error ASoC: wm8904: Check for clk_prepare_enable() error ASoC: wm8731: Check for clk_prepare_enable() error ASoC: max98095: Check for clk_prepare_enable() error Fang, Yang A (2): ASoC: ts3a227e: do not report jack status when there is i2c read err ASoC: max98090: Fix sequencing when starting additional routes Geert Uytterhoeven (2): ASoC: fsi: Remove obsolete sh_fsi2 platform_device_id entry ASoC: rsnd: Silence DMA slave ID compile warning on 64-bit Hardik T Shah (1): ASoC: Intel: Skylake: Add helpers for SRC and converter modules Harsha Priya (1): ASoC: ssm4567: Added ACPI entry for SSM4567 codec Jarkko Nikula (6): ASoC: rt286: Prefix hexadecimal ID register value with 0x in error print ASoC: rt5640: Prefix hexadecimal ID register value with 0x in error print ASoC: rt5645: Prefix hexadecimal ID register value with 0x in error print ASoC: rt5651: Prefix hexadecimal ID register value with 0x in error print ASoC: rt5670: Prefix hexadecimal ID register value with 0x in error print ASoC: rt5677: Prefix hexadecimal ID register value with 0x in error print Javier Martinez Canillas (1): ASoC: Export OF module alias information in missing codec drivers Jean-Francois Moine (1): ASoC: kirkwood: prevent double streaming Jeeja KP (11): ASoC: Intel: Add Skylake HDA platform driver ASoC: Intel: Add Skylake HDA audio driver ASoC: Intel: Add makefile support for SKL driver ASoC: Intel: Add support for decoupled mode in skl driver ASoC: Intel: Skylake: Add NHLT support to get BE config ASoC: Intel: Skylake: Add dsp and ipc init helpers ASoC: Intel: Skylake: Fix the NHLT rate size ASoC: Intel: Skylake: Use acpi header for NHLT header ASoC: Intel: Skylake: Add helpers for DSP module configuration ASoC: Intel: Skylake: Add DSP module init and binding routines ASoC: Intel: Skylake: Add pipe management helpers Joe Perches (1): ASoC: atmel_ssc_dai: Correct misuse of 0x%<decimal> John Lin (1): ASoC: rt5640: fix line out no sound issue Jonathan Corbet (1): ASoC: Clean up docbook warnings Julia Lawall (1): ASoC: ux500: adjust devm usage Junjie Mao (1): ASoC: Intel: fix runtime pm imbalance on error Jyri Sarha (1): ASoC: omap-hdmi-audio: Set buffer bytes step constraint to 128 Kenneth Westfield (2): ASoC: qcom: move board Kconfig deps to parent config ASoC: qcom: remove Kconfig deps from variant configs Koro Chen (5): ASoC: dpcm: Add checks of playback/capture before dpcm_get_be ASoC: rt5645: Fix missing free_irq ASoC: rt5645: Add regulator support ASoC: mediatek: Add suspend/resume callbacks ASoC: mediatek: Remove AIF widgets for backend DAIs Krzysztof Kozlowski (2): ASoC: mediatek: Drop owner assignment from platform_driver ASoC: drivers: Drop owner assignment from i2c_driver Kuninori Morimoto (36): regmap: add force_write option on _regmap_update_bits() regmap: add regmap_write_bits() regmap: add regmap_fields_force_write() ASoC: rsnd: gen: add rsnd_force_write() ASoC: core: add snd_soc_of_parse_audio_prefix() ASoC: rsrc-card: use snd_soc_of_parse_audio_route/prefix for routing ASoC: rsnd: remove unnecessary 'out of memory' message from SSI ASoC: rsnd: remove unnecessary 'out of memory' message from SRC ASoC: rsnd: remove unnecessary 'out of memory' message from DVC ASoC: rsnd: rename BUSIF_DALIGN to SSI_BUSIF_DALIGN ASoC: rsnd: rename INT_ENABLE to SSI_INT_ENABLE ASoC: rsnd: fixup each module counter on __rsnd_mod_call() ASoC: rsnd: add workaround for SRC sync convert + DVC ASoC: rsnd: rsnd_mod_id() return -1 if mod was NULL ASoC: rsnd: move DVC specific macro into dvc.c ASoC: rsnd: move SRC specific macro into src.c ASoC: rsnd: dvc: make sure DVC soft reset ASoC: rsnd: src: make sure SRC soft reset ASoC: rsnd: enable module multi connection ASoC: rsnd: rename rsnd_path_parse/break() into add/remove ASoC: rsnd: add rsnd_path_parse() for CTU/MIX/DVC route setting ASoC: rsnd: add rsnd_dvc_initialize_lock/unlock() ASoC: rsnd: add rsnd_src_initialize_lock/unlock() ASoC: rsnd: tidyup ADINR function name ASoC: rsnd: add rsnd_get_adinr_chan() ASoC: rsnd: tidyup data align position ASoC: rsnd: show debug message for SSI/SRC/DVC connection ASoC: rsnd: tidyup rsnd_dma_ops definition place ASoC: rsnd: check the Gen1 at the beginning of DVC probe ASoC: rsnd: dma: add DMA name on .ops ASoC: rsnd: add rsnd_io_to_mod() ASoC: rsnd: tidyup SRC position on each code ASoC: rsnd: update Audio DMA path search method ASoC: rsnd: add CTU (Channel Transfer Unit) prototype support ASoC: rsnd: add MIX (Mixer) support ASoC: rsnd: tidyup parameter assignment position Lars-Peter Clausen (83): ASoC: rt5645: Replace direct snd_soc_codec dapm field access ASoC: tas571x: Replace direct snd_soc_codec dapm field access ASoC: omap3pandora: Use card DAPM context to access widgets ASoC: cht_bsw_max98090_ti: Use card DAPM context to access widgets ASoC: dapm_widget_show_codec: Use component instead of CODEC ASoC: Remove snd_soc_codec dapm field ASoC: Prevent components from being bound to multiple cards ASoC: Use card field to indicate whether a component is bound ASoC: uda134x: Move state struct allocation to bus probe ASoC: uda134x: Explicitly handle in-data addresses ASoC: uda134x: Convert to regmap ASoC: uda134x: Use regmap_update_bits() were appropriate ASoC: max98925: Remove control_data initialization ASoC: max9877: Make driver global regmap struct local ASoC: max9877: Convert to component ASoC: max9768: Use managed gpio request ASoC: max9768: Convert to component ASoC: cs4349: Fix suspend/resume ASoC: cs4349: Don't use rtd->codec ASoC: dapm: Add helper function to free a widget ASoC: dapm: Avoid duplicating immutable strings ALSA: ac97: Add helper function to reset the AC97 device ASoC: ac97: Add support for resetting device before registration ASoC: ad1980: Use core AC'97 reset helper ASoC: stac9766: Use core reset helper ASoC: wm9705: Use core AC'97 reset helper ASoC: wm9712: Use core AC'97 reset helper ASoC: wm9713: Use core AC'97 reset helper ASoC: dapm: Drop always true checks ASoC: dapm: Simplify list creation in dapm_dai_get_connected_widgets() ASoC: dapm: Add widget path iterators ASoC: wm8804: Drop duplicate const ASoC: lpass-ipq806x: Staticise local symbols ASoC: rcar ctu: Staticise local symbols ASoC: uniperf: Add missing __iomem annotation ASoC: uniperf: Staticise local symbols ASoC: xtfpga-i2s: Add missing __rcu annotation ASoC: 88pm860x: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: ab8500: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: adau1373: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: alc5623: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: alc5632: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: cs42l52: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: cs42l56: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: cs42l73: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: da7210: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: da7213: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: da9055: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: jz4740: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: max9768: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: max98088: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: max98090: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: max98095: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: max9850: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: max9877: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: rt5631: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: rt5640: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: rt5645: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: rt5651: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: rt5670: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: rt5677: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: sgtl5000: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: ssm2602: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: tpa6130a2: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: twl4030: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: uda1380: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm8350: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm8400: Don't use range container for TLV with one entry ASoC: wm8737: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm8753: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm8961: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm8962: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm8990: Don't use range container for TLV with one entry ASoC: wm8991: Don't use range container for TLVs with one entry ASoC: wm8993: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm9081: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm9090: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm9713: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: wm_hubs: Replace TLV_DB_RANGE_HEAD with DECLARE_TLV_DB_RANGE ASoC: dapm: dapm_dai_get_connected_widgets: Fix missing mutex unlock ASoC: dapm: Consolidate input and output path handling ASoC: dapm: Consolidate path trace events ASoC: ssm4567: Add sense support Luis de Bethencourt (1): ASoC: bcm2835-i2s: Fix module autoload for OF platform drivers Maciej S. Szmigiero (6): ASoC: fsl_ssi: enable IPG clock during AC'97 reg access ASoC: fsl_ssi: AC'97 DAI driver needs probe method too ASoC: fsl_ssi: enable AC'97 asymmetric rates ASoC: fsl_ssi: add AC'97 ops setting check and cleanup ASoC: fsl_ssi: instantiate AC'97 CODEC ASoC: fsl_ssi: adjust set DAI format in AC'97 mode Marek Belisko (2): ASoC: Add gtm601 codec driver Documentation: vendor-prefixes: Add option prefix Mark Brown (43): Merge branch 'topic/codec-dapm' of git://git.kernel.org/.../broonie/sound into asoc-rt5645 Merge tag 'regmap-force' into asoc-rcar ASoC: sti-sas: Remove spurious dependency on SND_SOC_STI Merge branches 'topic/rt298', 'topic/ts3a227e' and 'topic/cs42xx8' of git://git.kernel.org/.../broonie/sound into asoc-owner Merge branches 'topic/owner' and 'topic/reg-default' of git://git.kernel.org/.../broonie/sound into asoc-max9877 Merge branch 'topic/of-name' of git://git.kernel.org/.../broonie/sound into asoc-rcar Merge tag 'regmap-seq-delay-api' of git://git.kernel.org/.../broonie/regmap into asoc-wm5110 Merge tag 'regmap-seq-delay-api' of git://git.kernel.org/.../broonie/regmap into asoc-reg-default Merge branch 'fix/rt5645' of git://git.kernel.org/.../broonie/sound into asoc-rt5645 ASoC: sti-uniperf: Fix implicit inclusion of pinctrl Merge branch 'fix/dapm' of git://git.kernel.org/.../broonie/sound into asoc-dapm Merge branch 'topic/owner' of git://git.kernel.org/.../broonie/sound into asoc-ssm4567 Merge branch 'fix/ssm4567' of git://git.kernel.org/.../broonie/sound into asoc-ssm4567 Merge tag 'v4.2-rc6' into asoc-topology Merge branch 'topic/gtm601' of git://git.kernel.org/.../broonie/sound into asoc-ics43432 Merge branch 'topic/ics43432' of git://git.kernel.org/.../broonie/sound into asoc-gtm601 Merge remote-tracking branch 'asoc/fix/rt5645' into asoc-linus Merge remote-tracking branches 'asoc/fix/adav80x', 'asoc/fix/arizona', 'asoc/fix/max98090', 'asoc/fix/rt5640', 'asoc/fix/samsung' and 'asoc/fix/wm8994' into asoc-linus Merge remote-tracking branch 'asoc/topic/core' into asoc-next Merge remote-tracking branch 'asoc/topic/dapm' into asoc-next Merge remote-tracking branch 'asoc/topic/pcm1681' into asoc-next Merge remote-tracking branch 'asoc/topic/rt5645' into asoc-next Merge remote-tracking branch 'asoc/topic/ssm4567' into asoc-next Merge remote-tracking branches 'asoc/topic/88pm860x', 'asoc/topic/ac97', 'asoc/topic/ak4542', 'asoc/topic/arizona' and 'asoc/topic/atmel' into asoc-next Merge remote-tracking branches 'asoc/topic/au1x', 'asoc/topic/bcm2835', 'asoc/topic/blackfin' and 'asoc/topic/card' into asoc-next Merge remote-tracking branches 'asoc/topic/const', 'asoc/topic/cs35l32', 'asoc/topic/cs4265' and 'asoc/topic/cs42l52' into asoc-next Merge remote-tracking branches 'asoc/topic/cs42l56', 'asoc/topic/cs42l73', 'asoc/topic/cs4349' and 'asoc/topic/da732x' into asoc-next Merge remote-tracking branches 'asoc/topic/davinci', 'asoc/topic/davinci-vcif', 'asoc/topic/doc' and 'asoc/topic/dpcm' into asoc-next Merge remote-tracking branches 'asoc/topic/fsi', 'asoc/topic/fsl', 'asoc/topic/fsl-asrc', 'asoc/topic/fsl-card' and 'asoc/topic/fsl-sai' into asoc-next Merge remote-tracking branches 'asoc/topic/fsl-spdif', 'asoc/topic/fsl-ssi', 'asoc/topic/gtm601', 'asoc/topic/ics43432' and 'asoc/topic/ids' into asoc-next Merge remote-tracking branches 'asoc/topic/intel', 'asoc/topic/kirkwood', 'asoc/topic/lm49453', 'asoc/topic/max9768' and 'asoc/topic/max98088' into asoc-next Merge remote-tracking branches 'asoc/topic/max98090', 'asoc/topic/max98095', 'asoc/topic/max98357a', 'asoc/topic/max9877' and 'asoc/topic/max98925' into asoc-next Merge remote-tracking branches 'asoc/topic/mediatek', 'asoc/topic/mtk', 'asoc/topic/nuc900', 'asoc/topic/of-name' and 'asoc/topic/omap' into asoc-next Merge remote-tracking branches 'asoc/topic/pm-ops', 'asoc/topic/pxa' and 'asoc/topic/qcom' into asoc-next Merge remote-tracking branches 'asoc/topic/rcar', 'asoc/topic/reg-default', 'asoc/topic/rl6231', 'asoc/topic/rockchip' and 'asoc/topic/rt286' into asoc-next Merge remote-tracking branches 'asoc/topic/rt298', 'asoc/topic/rt5640', 'asoc/topic/rt5651' and 'asoc/topic/rt5670' into asoc-next Merge remote-tracking branches 'asoc/topic/rt5677', 'asoc/topic/sh', 'asoc/topic/simple', 'asoc/topic/sirf-codec' and 'asoc/topic/spear' into asoc-next Merge remote-tracking branches 'asoc/topic/ssm2518', 'asoc/topic/sta529', 'asoc/topic/sti' and 'asoc/topic/sti-sas' into asoc-next Merge remote-tracking branches 'asoc/topic/tas2552', 'asoc/topic/tas5086', 'asoc/topic/tegra', 'asoc/topic/tlv' and 'asoc/topic/topology' into asoc-next Merge remote-tracking branches 'asoc/topic/txx9', 'asoc/topic/uda134x', 'asoc/topic/ux500' and 'asoc/topic/width' into asoc-next Merge remote-tracking branches 'asoc/topic/wm0010', 'asoc/topic/wm5100', 'asoc/topic/wm5110', 'asoc/topic/wm8004' and 'asoc/topic/wm8731' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8741', 'asoc/topic/wm8753', 'asoc/topic/wm8904', 'asoc/topic/wm8960' and 'asoc/topic/wm8983' into asoc-next Merge remote-tracking branches 'asoc/topic/wm8996', 'asoc/topic/xtensa' and 'asoc/topic/zx296702' into asoc-next Masanari Iida (2): ASoC: Fix warning while make xmldocs caused by soc-core.c ASoC: Fix warning caused by a typo in comments of snd_soc_add_platform() Mengdong Lin (4): ASoC: topology: Change pass number of DAI smaller than graph ASoC: topology: Bind vendor specific kcontrol handlers before standard ones ASoC: topology: Reduce arguments of soc_tplg_kcontrol_bind_io() ASoC: topology: Add support for TLV bytes controls Nariman Poushin (2): regmap: Use reg_sequence for multi_reg_write / register_patch ASoC: wm5110: Use reg_sequence for multi_reg_write/register_patch Nicolas Boichat (5): ASoC: rt5645: Remove irq_jack_detection function ASoC: rt5645: Update dapm pins when the card is not instantiated yet ASoC: rt5645: Simplify rt5645_enable_push_button_irq ASoC: rt5645: Remove unused rt5645 variable ASoC: rt5645: Remove return value in jack detect work Nicolin Chen (1): ASoC: fsl-asoc-card: Specify codec_dai_name for DAI links Nik Nyby (1): ASoC: codecs: max98088: fix typo in constant Nikesh Oswal (1): ASoC: arizona: Fix gain settings of FLL in free-run mode Oder Chiou (7): ASoC: Add function "rl6231_get_pre_div" to correct the dmic clock calculation ASoC: rt5645: Modify the headphone depop and calibration function to prevent the pop sound in the booting time ASoC: rt5645: Modify the jack detection function to prevent the pop sound while the jack plug in ASoC: rt5645: Remove the incorrect setting of the JD mode ASoC: rt5645: Add the register RT5645_CHARGE_PUMP to readable check function ASoC: rt5645: Add struct dmi_system_id "Google Celes" for chrome platform ASoC: rt5645: Prevent the pop sound of the headphone while rebooting or shutdowning Omair M Abdullah (1): ASoC: Intel: Skylake: Print error code in IPC for SKL Ricard Wanderlof (3): ASoC: ssm2518: Add explicit device tree support ASoC: ics43432: Add codec driver for InvenSense ICS-43432 ASoC: ics43432: Fix build errors Sergey Kiselev (1): ASoC: wm8731: initialize the hardware when loading the codec driver Shengjiu Wang (2): ASoC: fsl: Add dedicated DMA buffer size for each cpu dai ASoC: core: Don't probe the component which is dummy Subhransu S. Prusty (9): ASoC: Intel: Define SKL ADSP IPC and general purpose registers ASoC: Intel: Reorganize the common dsp structure ASoC: Intel: Add helper to poll register for DSP status ASoC: Intel: Add helper to update register bits with attr RWC ASoC: Intel: Add Skylake IPC library ASoC: Intel: Add DSP init and boot up functionality for SKL ASoC: Intel: Skylake: Add dsp loader ops ASoC: Intel: Skylake: Process code loader DMA interrupt ASoC: Intel: Skylake: Add SKL DSP initialization Tim Howe (1): ASoC: cs4349: Add support for Cirrus Logic CS4349 Vaishali Thakkar (7): ASoC: xtensa: Remove unnecessary snd_pcm_lib_preallocate_free_for_all() ASoC: tegra: Use devm_clk_get ASoC: tegra: Convert to managed resources ASoC: davinci-vcif: Use devm_snd_soc_register_component ASoC: tegra: Revert part of "ASoC: tegra: Convert to managed resources" ASoC: rockchip: i2s: Adjust devm usage ASoC: samsung: Remove redundant arndale_audio_remove Vinod Koul (7): ASoC: intel: kconfig - Move DW_DMAC_CORE dependency to machines ASoC: intel - kconfig: remove SND_SOC_INTEL_SST prompt ASoC: intel - atom: statify sst_slot_enum_info ASoC: intel - atom: fix sst_platform_compr_ops declaration ASoC: Intel: Skylake: Add code loader DMA registers ASoC: Intel: Skylake: add code loader DMA operations ASoC: Intel: Skylake: Add code loader DMA APIs Xing Zheng (1): ASoC: rockchip: fix a misjudgement by return Xiubo Li (1): ASoC: fsl: fix typos for sound/soc/fsl/* Zidan Wang (4): ASoC: wm8960: update pll and clock setting function ASoC: fsl-sai: add 32 bit word length support ASoC: fsl-asoc-card: add wm8960 support ASoC: wm8994: fix add dynamic path error kbuild test robot (8): ASoC: fix platform_no_drv_owner.cocci warnings ASoC: gtm601_dai can be static ASoC: sti-sas: fix platform_no_drv_owner.cocci warnings ASoC: sti: sti_uniperiph_dai_create_ctrl() can be static ASoC: cs4349: fix platform_no_drv_owner.cocci warnings ASoC: rockchip: fix platform_no_drv_owner.cocci warnings ASoC: rockchip: fix platform_no_drv_owner.cocci warnings ASoC: ics43432: fix platform_no_drv_owner.cocci warnings zhengxing (2): ASoC: rockchip: Add machine driver for max98090 codec ASoC: rockchip: Add machine driver for rt5645/rt5650 codec Documentation/DocBook/alsa-driver-api.tmpl | 2 +- Documentation/devicetree/bindings/sound/cs4349.txt | 19 + .../devicetree/bindings/sound/ics43432.txt | 17 + .../devicetree/bindings/sound/max98357a.txt | 6 +- .../devicetree/bindings/sound/renesas,rsnd.txt | 22 + .../bindings/sound/renesas,rsrc-card.txt | 7 + .../bindings/sound/rockchip-max98090.txt | 19 + .../devicetree/bindings/sound/rockchip-rt5645.txt | 17 + .../devicetree/bindings/sound/st,sti-asoc-card.txt | 155 +++ .../devicetree/bindings/vendor-prefixes.txt | 2 + drivers/base/regmap/internal.h | 2 +- drivers/base/regmap/regmap.c | 73 +- drivers/gpu/drm/i2c/adv7511.c | 2 +- drivers/input/misc/drv260x.c | 6 +- drivers/input/misc/drv2665.c | 2 +- drivers/input/misc/drv2667.c | 4 +- drivers/mfd/arizona-core.c | 2 +- drivers/mfd/twl6040.c | 2 +- drivers/mfd/wm5102-tables.c | 6 +- drivers/mfd/wm5110-tables.c | 22 +- drivers/mfd/wm8994-core.c | 8 +- drivers/mfd/wm8997-tables.c | 2 +- include/linux/mfd/arizona/registers.h | 37 + include/linux/regmap.h | 28 +- include/sound/ac97_codec.h | 2 + include/sound/rcar_snd.h | 14 + include/sound/rt298.h | 20 + include/sound/soc-dapm.h | 84 +- include/sound/soc-topology.h | 13 +- include/sound/soc.h | 29 +- include/trace/events/asoc.h | 53 +- sound/ac97_bus.c | 62 + sound/soc/Kconfig | 1 + sound/soc/Makefile | 1 + sound/soc/atmel/atmel_ssc_dai.c | 2 +- sound/soc/au1x/dbdma2.c | 11 +- sound/soc/au1x/dma.c | 11 +- sound/soc/au1x/psc-i2s.c | 16 +- sound/soc/bcm/bcm2835-i2s.c | 2 + sound/soc/blackfin/bf5xx-ac97-pcm.c | 10 +- sound/soc/blackfin/bf5xx-i2s-pcm.c | 10 +- sound/soc/blackfin/bfin-eval-adau1x61.c | 1 + sound/soc/codecs/88pm860x-codec.c | 32 +- sound/soc/codecs/Kconfig | 25 + sound/soc/codecs/Makefile | 10 + sound/soc/codecs/ab8500-codec.c | 7 +- sound/soc/codecs/ad1980.c | 36 +- sound/soc/codecs/adau1373.c | 22 +- sound/soc/codecs/adau1701.c | 1 - sound/soc/codecs/adau1761-i2c.c | 1 - sound/soc/codecs/adau1781-i2c.c | 1 - sound/soc/codecs/adau1977-i2c.c | 1 - sound/soc/codecs/adav803.c | 1 - sound/soc/codecs/adav80x.c | 3 +- sound/soc/codecs/ak4535.c | 1 - sound/soc/codecs/ak4641.c | 1 - sound/soc/codecs/ak4642.c | 34 +- sound/soc/codecs/ak4671.c | 1 - sound/soc/codecs/alc5623.c | 8 +- sound/soc/codecs/alc5632.c | 10 +- sound/soc/codecs/arizona.c | 129 +- sound/soc/codecs/arizona.h | 20 +- sound/soc/codecs/cs35l32.c | 60 +- sound/soc/codecs/cs35l32.h | 2 +- sound/soc/codecs/cs4265.c | 19 +- sound/soc/codecs/cs4270.c | 1 - sound/soc/codecs/cs4271-i2c.c | 1 - sound/soc/codecs/cs42l51-i2c.c | 1 - sound/soc/codecs/cs42l52.c | 65 +- sound/soc/codecs/cs42l56.c | 71 +- sound/soc/codecs/cs42l73.c | 115 +- sound/soc/codecs/cs42xx8-i2c.c | 4 +- sound/soc/codecs/cs42xx8.c | 19 +- sound/soc/codecs/cs42xx8.h | 1 + sound/soc/codecs/cs4349.c | 392 ++++++ sound/soc/codecs/cs4349.h | 136 +++ sound/soc/codecs/da7210.c | 29 +- sound/soc/codecs/da7213.c | 18 +- sound/soc/codecs/da732x.c | 15 +- sound/soc/codecs/da9055.c | 19 +- sound/soc/codecs/gtm601.c | 95 ++ sound/soc/codecs/ics43432.c | 76 ++ sound/soc/codecs/isabelle.c | 11 +- sound/soc/codecs/jz4740.c | 7 +- sound/soc/codecs/lm4857.c | 1 - sound/soc/codecs/lm49453.c | 19 +- sound/soc/codecs/max9768.c | 71 +- sound/soc/codecs/max98088.c | 329 +---- sound/soc/codecs/max98088.h | 2 +- sound/soc/codecs/max98090.c | 171 ++- sound/soc/codecs/max98090.h | 1 + sound/soc/codecs/max98095.c | 354 +----- sound/soc/codecs/max98357a.c | 25 +- sound/soc/codecs/max9850.c | 8 +- sound/soc/codecs/max9877.c | 33 +- sound/soc/codecs/max98925.c | 6 +- sound/soc/codecs/mc13783.c | 6 +- sound/soc/codecs/ml26124.c | 3 +- sound/soc/codecs/pcm1681.c | 14 +- sound/soc/codecs/pcm512x-i2c.c | 1 - sound/soc/codecs/pcm512x.c | 4 +- sound/soc/codecs/rl6231.c | 104 +- sound/soc/codecs/rl6231.h | 1 + sound/soc/codecs/rt286.c | 11 +- sound/soc/codecs/rt298.c | 1271 ++++++++++++++++++++ sound/soc/codecs/rt298.h | 206 ++++ sound/soc/codecs/rt5631.c | 8 +- sound/soc/codecs/rt5640.c | 59 +- sound/soc/codecs/rt5645.c | 443 +++---- sound/soc/codecs/rt5645.h | 27 +- sound/soc/codecs/rt5651.c | 19 +- sound/soc/codecs/rt5670.c | 21 +- sound/soc/codecs/rt5677-spi.c | 233 +++- sound/soc/codecs/rt5677-spi.h | 8 +- sound/soc/codecs/rt5677.c | 159 +-- sound/soc/codecs/rt5677.h | 5 +- sound/soc/codecs/sgtl5000.c | 8 +- sound/soc/codecs/si476x.c | 2 +- sound/soc/codecs/sirf-audio-codec.c | 4 +- sound/soc/codecs/ssm2518.c | 10 +- sound/soc/codecs/ssm2602-i2c.c | 1 - sound/soc/codecs/ssm2602.c | 7 +- sound/soc/codecs/ssm4567.c | 34 +- sound/soc/codecs/sta32x.c | 1 - sound/soc/codecs/sta350.c | 1 - sound/soc/codecs/sta529.c | 4 - sound/soc/codecs/stac9766.c | 57 +- sound/soc/codecs/sti-sas.c | 628 ++++++++++ sound/soc/codecs/tas2552.c | 20 +- sound/soc/codecs/tas2552.h | 2 +- sound/soc/codecs/tas5086.c | 11 +- sound/soc/codecs/tas571x.c | 2 +- sound/soc/codecs/tfa9879.c | 3 +- sound/soc/codecs/tlv320aic31xx.c | 3 +- sound/soc/codecs/tlv320aic32x4.c | 1 - sound/soc/codecs/tlv320aic3x.c | 3 +- sound/soc/codecs/tlv320dac33.c | 1 - sound/soc/codecs/tpa6130a2.c | 15 +- sound/soc/codecs/ts3a227e.c | 48 +- sound/soc/codecs/twl4030.c | 7 +- sound/soc/codecs/uda134x.c | 180 ++- sound/soc/codecs/uda134x.h | 2 - sound/soc/codecs/uda1380.c | 15 +- sound/soc/codecs/wm0010.c | 3 +- sound/soc/codecs/wm1250-ev1.c | 1 - sound/soc/codecs/wm2000.c | 1 - sound/soc/codecs/wm2200.c | 9 +- sound/soc/codecs/wm5100.c | 13 +- sound/soc/codecs/wm5102.c | 27 +- sound/soc/codecs/wm5110.c | 315 ++++- sound/soc/codecs/wm8350.c | 7 +- sound/soc/codecs/wm8400.c | 5 +- sound/soc/codecs/wm8510.c | 2 +- sound/soc/codecs/wm8523.c | 2 +- sound/soc/codecs/wm8580.c | 2 +- sound/soc/codecs/wm8711.c | 1 - sound/soc/codecs/wm8728.c | 1 - sound/soc/codecs/wm8731.c | 88 +- sound/soc/codecs/wm8737.c | 8 +- sound/soc/codecs/wm8741.c | 63 +- sound/soc/codecs/wm8750.c | 1 - sound/soc/codecs/wm8753.c | 14 +- sound/soc/codecs/wm8776.c | 5 +- sound/soc/codecs/wm8804-i2c.c | 1 - sound/soc/codecs/wm8804.c | 2 +- sound/soc/codecs/wm8900.c | 1 - sound/soc/codecs/wm8903.c | 1 - sound/soc/codecs/wm8904.c | 5 +- sound/soc/codecs/wm8940.c | 1 - sound/soc/codecs/wm8955.c | 1 - sound/soc/codecs/wm8960.c | 221 +++- sound/soc/codecs/wm8960.h | 1 + sound/soc/codecs/wm8961.c | 8 +- sound/soc/codecs/wm8962.c | 21 +- sound/soc/codecs/wm8971.c | 1 - sound/soc/codecs/wm8974.c | 1 - sound/soc/codecs/wm8978.c | 1 - sound/soc/codecs/wm8983.c | 81 +- sound/soc/codecs/wm8985.c | 1 - sound/soc/codecs/wm8988.c | 1 - sound/soc/codecs/wm8990.c | 6 +- sound/soc/codecs/wm8991.c | 53 +- sound/soc/codecs/wm8993.c | 12 +- sound/soc/codecs/wm8994.c | 18 +- sound/soc/codecs/wm8995.c | 1 - sound/soc/codecs/wm8996.c | 11 +- sound/soc/codecs/wm8997.c | 20 +- sound/soc/codecs/wm9081.c | 10 +- sound/soc/codecs/wm9090.c | 22 +- sound/soc/codecs/wm9705.c | 40 +- sound/soc/codecs/wm9712.c | 45 +- sound/soc/codecs/wm9713.c | 55 +- sound/soc/codecs/wm9713.h | 2 - sound/soc/codecs/wm_hubs.c | 7 +- sound/soc/davinci/davinci-i2s.c | 25 +- sound/soc/davinci/davinci-mcasp.c | 18 +- sound/soc/davinci/davinci-vcif.c | 14 +- sound/soc/fsl/eukrea-tlv320.c | 2 +- sound/soc/fsl/fsl-asoc-card.c | 16 +- sound/soc/fsl/fsl_asrc.c | 25 +- sound/soc/fsl/fsl_esai.c | 2 +- sound/soc/fsl/fsl_sai.c | 2 +- sound/soc/fsl/fsl_sai.h | 15 +- sound/soc/fsl/fsl_spdif.c | 25 +- sound/soc/fsl/fsl_ssi.c | 68 +- sound/soc/fsl/imx-pcm-dma.c | 25 +- sound/soc/fsl/imx-pcm.h | 9 +- sound/soc/fsl/imx-ssi.c | 2 +- sound/soc/generic/simple-card.c | 9 + sound/soc/intel/Kconfig | 29 +- sound/soc/intel/Makefile | 1 + sound/soc/intel/atom/sst-atom-controls.c | 6 +- sound/soc/intel/atom/sst-mfld-platform-pcm.c | 1 - sound/soc/intel/atom/sst-mfld-platform.h | 1 + sound/soc/intel/atom/sst/sst_drv_interface.c | 9 +- sound/soc/intel/atom/sst/sst_ipc.c | 3 +- sound/soc/intel/boards/byt-max98090.c | 1 + sound/soc/intel/boards/byt-rt5640.c | 1 + sound/soc/intel/boards/bytcr_rt5640.c | 1 + sound/soc/intel/boards/cht_bsw_max98090_ti.c | 19 +- sound/soc/intel/boards/cht_bsw_rt5645.c | 2 + sound/soc/intel/boards/cht_bsw_rt5672.c | 1 + sound/soc/intel/common/sst-dsp-priv.h | 23 +- sound/soc/intel/common/sst-dsp.c | 71 ++ sound/soc/intel/common/sst-dsp.h | 6 + sound/soc/intel/skylake/Makefile | 9 + sound/soc/intel/skylake/skl-messages.c | 884 ++++++++++++++ sound/soc/intel/skylake/skl-nhlt.c | 140 +++ sound/soc/intel/skylake/skl-nhlt.h | 106 ++ sound/soc/intel/skylake/skl-pcm.c | 916 ++++++++++++++ sound/soc/intel/skylake/skl-sst-cldma.c | 327 +++++ sound/soc/intel/skylake/skl-sst-cldma.h | 251 ++++ sound/soc/intel/skylake/skl-sst-dsp.c | 342 ++++++ sound/soc/intel/skylake/skl-sst-dsp.h | 145 +++ sound/soc/intel/skylake/skl-sst-ipc.c | 771 ++++++++++++ sound/soc/intel/skylake/skl-sst-ipc.h | 125 ++ sound/soc/intel/skylake/skl-sst.c | 280 +++++ sound/soc/intel/skylake/skl-topology.h | 286 +++++ sound/soc/intel/skylake/skl-tplg-interface.h | 88 ++ sound/soc/intel/skylake/skl.c | 536 +++++++++ sound/soc/intel/skylake/skl.h | 84 ++ sound/soc/kirkwood/kirkwood-dma.c | 4 + sound/soc/mediatek/mt8173-max98090.c | 2 +- sound/soc/mediatek/mt8173-rt5650-rt5676.c | 2 +- sound/soc/mediatek/mtk-afe-common.h | 8 - sound/soc/mediatek/mtk-afe-pcm.c | 89 +- sound/soc/nuc900/nuc900-pcm.c | 9 +- sound/soc/omap/mcbsp.c | 20 +- sound/soc/omap/omap-hdmi-audio.c | 10 +- sound/soc/omap/omap3pandora.c | 6 +- sound/soc/pxa/mmp-pcm.c | 9 +- sound/soc/pxa/pxa-ssp.c | 11 +- sound/soc/pxa/pxa2xx-i2s.c | 11 +- sound/soc/pxa/pxa2xx-pcm.c | 9 +- sound/soc/qcom/Kconfig | 7 +- sound/soc/qcom/lpass-cpu.c | 2 +- sound/soc/qcom/lpass-ipq806x.c | 2 +- sound/soc/qcom/lpass.h | 2 +- sound/soc/rockchip/Kconfig | 19 + sound/soc/rockchip/Makefile | 6 + sound/soc/rockchip/rockchip_i2s.c | 8 +- sound/soc/rockchip/rockchip_max98090.c | 236 ++++ sound/soc/rockchip/rockchip_rt5645.c | 225 ++++ sound/soc/samsung/arndale_rt5631.c | 11 +- sound/soc/samsung/snow.c | 1 + sound/soc/sh/dma-sh7760.c | 9 +- sound/soc/sh/fsi.c | 1 - sound/soc/sh/rcar/Makefile | 2 +- sound/soc/sh/rcar/core.c | 195 ++- sound/soc/sh/rcar/ctu.c | 171 +++ sound/soc/sh/rcar/dma.c | 128 +- sound/soc/sh/rcar/dvc.c | 73 +- sound/soc/sh/rcar/gen.c | 33 +- sound/soc/sh/rcar/mix.c | 200 +++ sound/soc/sh/rcar/rsnd.h | 111 +- sound/soc/sh/rcar/rsrc-card.c | 22 +- sound/soc/sh/rcar/src.c | 117 +- sound/soc/sh/rcar/ssi.c | 4 +- sound/soc/sh/ssi.c | 12 +- sound/soc/soc-ac97.c | 30 +- sound/soc/soc-core.c | 86 +- sound/soc/soc-dapm.c | 503 ++++---- sound/soc/soc-pcm.c | 16 +- sound/soc/soc-topology.c | 145 +-- sound/soc/spear/spdif_in.c | 20 +- sound/soc/spear/spear_pcm.c | 2 +- sound/soc/sti/Kconfig | 11 + sound/soc/sti/Makefile | 4 + sound/soc/sti/sti_uniperif.c | 254 ++++ sound/soc/sti/uniperif.h | 1229 +++++++++++++++++++ sound/soc/sti/uniperif_player.c | 1110 +++++++++++++++++ sound/soc/sti/uniperif_reader.c | 362 ++++++ sound/soc/tegra/tegra20_das.c | 23 +- sound/soc/tegra/tegra20_i2s.c | 23 +- sound/soc/tegra/tegra20_spdif.c | 47 +- sound/soc/tegra/tegra30_ahub.c | 80 +- sound/soc/tegra/tegra30_i2s.c | 23 +- sound/soc/txx9/txx9aclc.c | 10 +- sound/soc/ux500/ux500_msp_dai.c | 29 +- sound/soc/xtensa/xtfpga-i2s.c | 8 +- sound/soc/zte/zx296702-i2s.c | 6 +- 301 files changed, 16003 insertions(+), 3744 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/cs4349.txt create mode 100644 Documentation/devicetree/bindings/sound/ics43432.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip-max98090.txt create mode 100644 Documentation/devicetree/bindings/sound/rockchip-rt5645.txt create mode 100644 Documentation/devicetree/bindings/sound/st,sti-asoc-card.txt create mode 100644 include/sound/rt298.h create mode 100644 sound/soc/codecs/cs4349.c create mode 100644 sound/soc/codecs/cs4349.h create mode 100644 sound/soc/codecs/gtm601.c create mode 100644 sound/soc/codecs/ics43432.c create mode 100644 sound/soc/codecs/rt298.c create mode 100644 sound/soc/codecs/rt298.h create mode 100644 sound/soc/codecs/sti-sas.c create mode 100644 sound/soc/intel/skylake/Makefile create mode 100644 sound/soc/intel/skylake/skl-messages.c create mode 100644 sound/soc/intel/skylake/skl-nhlt.c create mode 100644 sound/soc/intel/skylake/skl-nhlt.h create mode 100644 sound/soc/intel/skylake/skl-pcm.c create mode 100644 sound/soc/intel/skylake/skl-sst-cldma.c create mode 100644 sound/soc/intel/skylake/skl-sst-cldma.h create mode 100644 sound/soc/intel/skylake/skl-sst-dsp.c create mode 100644 sound/soc/intel/skylake/skl-sst-dsp.h create mode 100644 sound/soc/intel/skylake/skl-sst-ipc.c create mode 100644 sound/soc/intel/skylake/skl-sst-ipc.h create mode 100644 sound/soc/intel/skylake/skl-sst.c create mode 100644 sound/soc/intel/skylake/skl-topology.h create mode 100644 sound/soc/intel/skylake/skl-tplg-interface.h create mode 100644 sound/soc/intel/skylake/skl.c create mode 100644 sound/soc/intel/skylake/skl.h create mode 100644 sound/soc/rockchip/rockchip_max98090.c create mode 100644 sound/soc/rockchip/rockchip_rt5645.c create mode 100644 sound/soc/sh/rcar/ctu.c create mode 100644 sound/soc/sh/rcar/mix.c create mode 100644 sound/soc/sti/Kconfig create mode 100644 sound/soc/sti/Makefile create mode 100644 sound/soc/sti/sti_uniperif.c create mode 100644 sound/soc/sti/uniperif.h create mode 100644 sound/soc/sti/uniperif_player.c create mode 100644 sound/soc/sti/uniperif_reader.c

2 6

[alsa-devel] [PATCH] ASoC: Make SND_SOC_STI_SAS depend on RESET_CONTROLLER
by Axel Lin 31 Aug '15

31 Aug '15

(devm_)reset_control_get function is available only when RESET_CONTROLLER is set. Make SND_SOC_STI_SAS depend on RESET_CONTROLLER to fix build error if !RESET_CONTROLLER. Reported-by: Takashi Iwai <tiwai(a)suse.com> Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/Kconfig | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0c9733e..f8e0022 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -106,7 +106,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_STA350 if I2C select SND_SOC_STA529 if I2C select SND_SOC_STAC9766 if SND_SOC_AC97_BUS - select SND_SOC_STI_SAS + select SND_SOC_STI_SAS if RESET_CONTROLLER select SND_SOC_TAS2552 if I2C select SND_SOC_TAS5086 if I2C select SND_SOC_TAS571X if I2C @@ -634,6 +634,7 @@ config SND_SOC_STAC9766 config SND_SOC_STI_SAS tristate "codec Audio support for STI SAS codec" + depends on RESET_CONTROLLER config SND_SOC_TAS2552 tristate "Texas Instruments TAS2552 Mono Audio amplifier" -- 1.9.1

3 3

[alsa-devel] [PATCH] ASoC: ics43432: Fix build errors
by Ricard Wanderlof 31 Aug '15

31 Aug '15

Fix build errors Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> --- Not sure how the incorrect name or the misspelled license string got through my local build, perhaps because I've been developing this for an ARM system? Have now set up a tree to build for x86 as well, where the errors manifest themselves, to avoid this type of stuff happening in the future. sound/soc/codecs/ics43432.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ics43432.c b/sound/soc/codecs/ics43432.c index d217fed..465d591 100644 --- a/sound/soc/codecs/ics43432.c +++ b/sound/soc/codecs/ics43432.c @@ -57,7 +57,7 @@ static const struct of_device_id ics43432_ids[] = { { .compatible = "invensense,ics43432", }, { } }; -MODULE_DEVICE_TABLE(of, ics43432_dt_ids); +MODULE_DEVICE_TABLE(of, ics43432_ids); #endif static struct platform_driver ics43432_driver = { @@ -73,4 +73,4 @@ module_platform_driver(ics43432_driver); MODULE_DESCRIPTION("ASoC ICS43432 driver"); MODULE_AUTHOR("Ricard Wanderlof <ricardw(a)axis.com>"); -MODULE_LICENSE("GPLv2"); +MODULE_LICENSE("GPL v2"); -- 1.7.10.4 -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

2 3

[alsa-devel] [PATCH] Revert part of "ASoC: tegra: Convert to managed resources"
by Vaishali Thakkar 31 Aug '15

31 Aug '15

Revert the problematic part of commit 470805eb9f31 ("ASoC: tegra: Convert to managed resources"). Before this commit, PM cleanup was performed after the component was unregistered. But returning directly will skip PM cleanup. So, to be on safe side it is better to use snd_soc_register_component instead of devm_snd_soc_register_component. Signed-off-by: Vaishali Thakkar <vthakkar1994(a)gmail.com> --- sound/soc/tegra/tegra20_spdif.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index a814da0..66237bc 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -329,9 +329,8 @@ static int tegra20_spdif_platform_probe(struct platform_device *pdev) goto err_pm_disable; } - ret = devm_snd_soc_register_component(&pdev->dev, - &tegra20_spdif_component, - &tegra20_spdif_dai, 1); + ret = snd_soc_register_component(&pdev->dev, &tegra20_spdif_component, + &tegra20_spdif_dai, 1); if (ret) { dev_err(&pdev->dev, "Could not register DAI: %d\n", ret); ret = -ENOMEM; @@ -341,11 +340,13 @@ static int tegra20_spdif_platform_probe(struct platform_device *pdev) ret = tegra_pcm_platform_register(&pdev->dev); if (ret) { dev_err(&pdev->dev, "Could not register PCM: %d\n", ret); - return ret; + goto err_unregister_component; } return 0; +err_unregister_component: + snd_soc_unregister_component(&pdev->dev); err_suspend: if (!pm_runtime_status_suspended(&pdev->dev)) tegra20_spdif_runtime_suspend(&pdev->dev); @@ -364,6 +365,7 @@ static int tegra20_spdif_platform_remove(struct platform_device *pdev) tegra20_spdif_runtime_suspend(&pdev->dev); tegra_pcm_platform_unregister(&pdev->dev); + snd_soc_unregister_component(&pdev->dev); return 0; } -- 1.9.1

3 4

[alsa-devel] [PATCH] ASoC: tegra: Use devm_ioremap_resource instead of open code
by Axel Lin 31 Aug '15

31 Aug '15

Use devm_ioremap_resource() to simplify the code. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/tegra/tegra20_das.c | 23 ++++------------------ sound/soc/tegra/tegra20_i2s.c | 23 ++++------------------ sound/soc/tegra/tegra20_spdif.c | 22 ++++----------------- sound/soc/tegra/tegra30_ahub.c | 43 +++++++---------------------------------- sound/soc/tegra/tegra30_i2s.c | 23 ++++------------------ 5 files changed, 23 insertions(+), 111 deletions(-) diff --git a/sound/soc/tegra/tegra20_das.c b/sound/soc/tegra/tegra20_das.c index f52600b..89add13 100644 --- a/sound/soc/tegra/tegra20_das.c +++ b/sound/soc/tegra/tegra20_das.c @@ -133,7 +133,7 @@ static const struct regmap_config tegra20_das_regmap_config = { static int tegra20_das_probe(struct platform_device *pdev) { - struct resource *res, *region; + struct resource *res; void __iomem *regs; int ret = 0; @@ -149,24 +149,9 @@ static int tegra20_das_probe(struct platform_device *pdev) das->dev = &pdev->dev; res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - dev_err(&pdev->dev, "No memory resource\n"); - ret = -ENODEV; - goto err; - } - - region = devm_request_mem_region(&pdev->dev, res->start, - resource_size(res), pdev->name); - if (!region) { - dev_err(&pdev->dev, "Memory region already claimed\n"); - ret = -EBUSY; - goto err; - } - - regs = devm_ioremap(&pdev->dev, res->start, resource_size(res)); - if (!regs) { - dev_err(&pdev->dev, "ioremap failed\n"); - ret = -ENOMEM; + regs = devm_ioremap_resource(&pdev->dev, res); + if (IS_ERR(regs)) { + ret = PTR_ERR(regs); goto err; } diff --git a/sound/soc/tegra/tegra20_i2s.c b/sound/soc/tegra/tegra20_i2s.c index 05f1c6e..14106fa 100644 --- a/sound/soc/tegra/tegra20_i2s.c +++ b/sound/soc/tegra/tegra20_i2s.c @@ -339,7 +339,7 @@ static const struct regmap_config tegra20_i2s_regmap_config = { static int tegra20_i2s_platform_probe(struct platform_device *pdev) { struct tegra20_i2s *i2s; - struct resource *mem, *memregion; + struct resource *mem; void __iomem *regs; int ret; @@ -362,24 +362,9 @@ static int tegra20_i2s_platform_probe(struct platform_device *pdev) } mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!mem) { - dev_err(&pdev->dev, "No memory resource\n"); - ret = -ENODEV; - goto err_clk_put; - } - - memregion = devm_request_mem_region(&pdev->dev, mem->start, - resource_size(mem), DRV_NAME); - if (!memregion) { - dev_err(&pdev->dev, "Memory region already claimed\n"); - ret = -EBUSY; - goto err_clk_put; - } - - regs = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); - if (!regs) { - dev_err(&pdev->dev, "ioremap failed\n"); - ret = -ENOMEM; + regs = devm_ioremap_resource(&pdev->dev, mem); + if (IS_ERR(regs)) { + ret = PTR_ERR(regs); goto err_clk_put; } diff --git a/sound/soc/tegra/tegra20_spdif.c b/sound/soc/tegra/tegra20_spdif.c index 0809b1e..a0c3640 100644 --- a/sound/soc/tegra/tegra20_spdif.c +++ b/sound/soc/tegra/tegra20_spdif.c @@ -265,7 +265,7 @@ static const struct regmap_config tegra20_spdif_regmap_config = { static int tegra20_spdif_platform_probe(struct platform_device *pdev) { struct tegra20_spdif *spdif; - struct resource *mem, *memregion, *dmareq; + struct resource *mem, *dmareq; void __iomem *regs; int ret; @@ -285,10 +285,9 @@ static int tegra20_spdif_platform_probe(struct platform_device *pdev) } mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!mem) { - dev_err(&pdev->dev, "No memory resource\n"); - return -ENODEV; - } + regs = devm_ioremap_resource(&pdev->dev, mem); + if (IS_ERR(regs)) + return PTR_ERR(regs); dmareq = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!dmareq) { @@ -296,19 +295,6 @@ static int tegra20_spdif_platform_probe(struct platform_device *pdev) return -ENODEV; } - memregion = devm_request_mem_region(&pdev->dev, mem->start, - resource_size(mem), DRV_NAME); - if (!memregion) { - dev_err(&pdev->dev, "Memory region already claimed\n"); - return -EBUSY; - } - - regs = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); - if (!regs) { - dev_err(&pdev->dev, "ioremap failed\n"); - return -ENOMEM; - } - spdif->regmap = devm_regmap_init_mmio(&pdev->dev, regs, &tegra20_spdif_regmap_config); if (IS_ERR(spdif->regmap)) { diff --git a/sound/soc/tegra/tegra30_ahub.c b/sound/soc/tegra/tegra30_ahub.c index 989b1e8..fef3b9a 100644 --- a/sound/soc/tegra/tegra30_ahub.c +++ b/sound/soc/tegra/tegra30_ahub.c @@ -521,7 +521,7 @@ static int tegra30_ahub_probe(struct platform_device *pdev) const struct tegra30_ahub_soc_data *soc_data; struct reset_control *rst; int i; - struct resource *res0, *res1, *region; + struct resource *res0, *res1; void __iomem *regs_apbif, *regs_ahub; int ret = 0; @@ -584,26 +584,12 @@ static int tegra30_ahub_probe(struct platform_device *pdev) } res0 = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res0) { - dev_err(&pdev->dev, "No apbif memory resource\n"); - return -ENODEV; - } + regs_apbif = devm_ioremap_resource(&pdev->dev, res0); + if (IS_ERR(regs_apbif)) + return PTR_ERR(regs_apbif); - region = devm_request_mem_region(&pdev->dev, res0->start, - resource_size(res0), DRV_NAME); - if (!region) { - dev_err(&pdev->dev, "request region apbif failed\n"); - return -EBUSY; - } ahub->apbif_addr = res0->start; - regs_apbif = devm_ioremap(&pdev->dev, res0->start, - resource_size(res0)); - if (!regs_apbif) { - dev_err(&pdev->dev, "ioremap apbif failed\n"); - return -ENOMEM; - } - ahub->regmap_apbif = devm_regmap_init_mmio(&pdev->dev, regs_apbif, &tegra30_ahub_apbif_regmap_config); if (IS_ERR(ahub->regmap_apbif)) { @@ -614,24 +600,9 @@ static int tegra30_ahub_probe(struct platform_device *pdev) regcache_cache_only(ahub->regmap_apbif, true); res1 = platform_get_resource(pdev, IORESOURCE_MEM, 1); - if (!res1) { - dev_err(&pdev->dev, "No ahub memory resource\n"); - return -ENODEV; - } - - region = devm_request_mem_region(&pdev->dev, res1->start, - resource_size(res1), DRV_NAME); - if (!region) { - dev_err(&pdev->dev, "request region ahub failed\n"); - return -EBUSY; - } - - regs_ahub = devm_ioremap(&pdev->dev, res1->start, - resource_size(res1)); - if (!regs_ahub) { - dev_err(&pdev->dev, "ioremap ahub failed\n"); - return -ENOMEM; - } + regs_ahub = devm_ioremap_resource(&pdev->dev, res1); + if (IS_ERR(regs_ahub)) + return PTR_ERR(regs_ahub); ahub->regmap_ahub = devm_regmap_init_mmio(&pdev->dev, regs_ahub, &tegra30_ahub_ahub_regmap_config); diff --git a/sound/soc/tegra/tegra30_i2s.c b/sound/soc/tegra/tegra30_i2s.c index fe36375..8e55583 100644 --- a/sound/soc/tegra/tegra30_i2s.c +++ b/sound/soc/tegra/tegra30_i2s.c @@ -379,7 +379,7 @@ static int tegra30_i2s_platform_probe(struct platform_device *pdev) struct tegra30_i2s *i2s; const struct of_device_id *match; u32 cif_ids[2]; - struct resource *mem, *memregion; + struct resource *mem; void __iomem *regs; int ret; @@ -419,24 +419,9 @@ static int tegra30_i2s_platform_probe(struct platform_device *pdev) } mem = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!mem) { - dev_err(&pdev->dev, "No memory resource\n"); - ret = -ENODEV; - goto err_clk_put; - } - - memregion = devm_request_mem_region(&pdev->dev, mem->start, - resource_size(mem), DRV_NAME); - if (!memregion) { - dev_err(&pdev->dev, "Memory region already claimed\n"); - ret = -EBUSY; - goto err_clk_put; - } - - regs = devm_ioremap(&pdev->dev, mem->start, resource_size(mem)); - if (!regs) { - dev_err(&pdev->dev, "ioremap failed\n"); - ret = -ENOMEM; + regs = devm_ioremap_resource(&pdev->dev, mem); + if (IS_ERR(regs)) { + ret = PTR_ERR(regs); goto err_clk_put; } -- 2.1.0

3 2

[alsa-devel] [asoc:topic/au1x 2/2] sound/soc/au1x/psc-i2s.c:299:6: warning: unused variable 'ret'
by kbuild test robot 30 Aug '15

30 Aug '15

tree: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/au1x head: fd63542fc5492b86158dbd53ce5de764f171b1b6 commit: fd63542fc5492b86158dbd53ce5de764f171b1b6 [2/2] ASoC: au1x: psc-i2s: Convert to use devm_ioremap_resource config: mips-db1xxx_defconfig (attached as .config) reproduce: wget https://git.kernel.org/cgit/linux/kernel/git/wfg/lkp-tests.git/plain/sbin/m… -O ~/bin/make.cross chmod +x ~/bin/make.cross git checkout fd63542fc5492b86158dbd53ce5de764f171b1b6 # save the attached .config to linux build tree make.cross ARCH=mips All warnings (new ones prefixed by >>): sound/soc/au1x/psc-i2s.c: In function 'au1xpsc_i2s_drvprobe': >> sound/soc/au1x/psc-i2s.c:299:6: warning: unused variable 'ret' [-Wunused-variable] int ret; ^ vim +/ret +299 sound/soc/au1x/psc-i2s.c 0f83d639 Manuel Lauss 2009-10-31 283 .rates = AU1XPSC_I2S_RATES, 0f83d639 Manuel Lauss 2009-10-31 284 .formats = AU1XPSC_I2S_FMTS, 0f83d639 Manuel Lauss 2009-10-31 285 .channels_min = 2, 0f83d639 Manuel Lauss 2009-10-31 286 .channels_max = 8, /* 2 without external help */ 0f83d639 Manuel Lauss 2009-10-31 287 }, 0f83d639 Manuel Lauss 2009-10-31 288 .ops = &au1xpsc_i2s_dai_ops, 0f83d639 Manuel Lauss 2009-10-31 289 }; 0f83d639 Manuel Lauss 2009-10-31 290 4edf87f5 Kuninori Morimoto 2013-03-21 291 static const struct snd_soc_component_driver au1xpsc_i2s_component = { 4edf87f5 Kuninori Morimoto 2013-03-21 292 .name = "au1xpsc-i2s", 4edf87f5 Kuninori Morimoto 2013-03-21 293 }; 4edf87f5 Kuninori Morimoto 2013-03-21 294 5c658be0 Bill Pemberton 2012-12-07 295 static int au1xpsc_i2s_drvprobe(struct platform_device *pdev) 0f83d639 Manuel Lauss 2009-10-31 296 { 226d0f22 Julia Lawall 2011-10-18 297 struct resource *iores, *dmares; 4a161d23 Manuel Lauss 2008-07-09 298 unsigned long sel; 4a161d23 Manuel Lauss 2008-07-09 @299 int ret; 0f83d639 Manuel Lauss 2009-10-31 300 struct au1xpsc_audio_data *wd; 4a161d23 Manuel Lauss 2008-07-09 301 aa4079c1 Julia Lawall 2011-12-29 302 wd = devm_kzalloc(&pdev->dev, sizeof(struct au1xpsc_audio_data), aa4079c1 Julia Lawall 2011-12-29 303 GFP_KERNEL); 0f83d639 Manuel Lauss 2009-10-31 304 if (!wd) 4a161d23 Manuel Lauss 2008-07-09 305 return -ENOMEM; 4a161d23 Manuel Lauss 2008-07-09 306 226d0f22 Julia Lawall 2011-10-18 307 iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); :::::: The code at line 299 was first introduced by commit :::::: 4a161d235b68eb7234f40106560c488a1bdb3851 ALSA: ASoC: Au12x0/Au1550 PSC Audio support :::::: TO: Manuel Lauss <mano(a)roarinelk.homelinux.net> :::::: CC: Jaroslav Kysela <perex(a)perex.cz> --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH] ASoC: spear_pcm: Use devm_snd_dmaengine_pcm_register to fix resource leak
by Axel Lin 30 Aug '15

30 Aug '15

All the callers assume devm_spear_pcm_platform_register is a devm_ API, so use devm_snd_dmaengine_pcm_register in devm_spear_pcm_platform_register. Fixes: e1771bcf99b0 ("ASoC: SPEAr: remove custom DMA alloc compat function") Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/spear/spear_pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/spear/spear_pcm.c b/sound/soc/spear/spear_pcm.c index a7dc3c5..e8476da 100644 --- a/sound/soc/spear/spear_pcm.c +++ b/sound/soc/spear/spear_pcm.c @@ -44,7 +44,7 @@ int devm_spear_pcm_platform_register(struct device *dev, *config = spear_dmaengine_pcm_config; config->compat_filter_fn = filter; - return snd_dmaengine_pcm_register(dev, config, + return devm_snd_dmaengine_pcm_register(dev, config, SND_DMAENGINE_PCM_FLAG_NO_DT | SND_DMAENGINE_PCM_FLAG_COMPAT); } -- 2.1.0

2 1

[alsa-devel] [PATCH] ASoC: sh: dma-sh7760: Convert to devm_snd_soc_register_platform
by Axel Lin 30 Aug '15

30 Aug '15

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/sh/dma-sh7760.c | 9 +-------- 1 file changed, 1 insertion(+), 8 deletions(-) diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index fd11404..8fad444 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -327,13 +327,7 @@ static struct snd_soc_platform_driver sh7760_soc_platform = { static int sh7760_soc_platform_probe(struct platform_device *pdev) { - return snd_soc_register_platform(&pdev->dev, &sh7760_soc_platform); -} - -static int sh7760_soc_platform_remove(struct platform_device *pdev) -{ - snd_soc_unregister_platform(&pdev->dev); - return 0; + return devm_snd_soc_register_platform(&pdev->dev, &sh7760_soc_platform); } static struct platform_driver sh7760_pcm_driver = { @@ -342,7 +336,6 @@ static struct platform_driver sh7760_pcm_driver = { }, .probe = sh7760_soc_platform_probe, - .remove = sh7760_soc_platform_remove, }; module_platform_driver(sh7760_pcm_driver); -- 2.1.0

3 2

[alsa-devel] [PATCH] ASoC: au1x: psc-i2s: Convert to use devm_ioremap_resource
by Axel Lin 30 Aug '15

30 Aug '15

Use devm_ioremap_resource() instead of open code. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/au1x/psc-i2s.c | 16 +++------------- 1 file changed, 3 insertions(+), 13 deletions(-) diff --git a/sound/soc/au1x/psc-i2s.c b/sound/soc/au1x/psc-i2s.c index e742ef6..38e853a 100644 --- a/sound/soc/au1x/psc-i2s.c +++ b/sound/soc/au1x/psc-i2s.c @@ -305,19 +305,9 @@ static int au1xpsc_i2s_drvprobe(struct platform_device *pdev) return -ENOMEM; iores = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!iores) - return -ENODEV; - - ret = -EBUSY; - if (!devm_request_mem_region(&pdev->dev, iores->start, - resource_size(iores), - pdev->name)) - return -EBUSY; - - wd->mmio = devm_ioremap(&pdev->dev, iores->start, - resource_size(iores)); - if (!wd->mmio) - return -EBUSY; + wd->mmio = devm_ioremap_resource(&pdev->dev, iores); + if (IS_ERR(wd->mmio)) + return PTR_ERR(wd->mmio); dmares = platform_get_resource(pdev, IORESOURCE_DMA, 0); if (!dmares) -- 2.1.0

3 2

[alsa-devel] [PATCH] ASoC: sti_uniperif: Ensure component is unregistered when unload module
by Axel Lin 30 Aug '15

30 Aug '15

Use devm_snd_soc_register_component to ensure component is unregistered when unload the module. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/sti/sti_uniperif.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/sti/sti_uniperif.c b/sound/soc/sti/sti_uniperif.c index dffabf3..39bcefe 100644 --- a/sound/soc/sti/sti_uniperif.c +++ b/sound/soc/sti/sti_uniperif.c @@ -224,9 +224,9 @@ static int sti_uniperiph_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, priv); - ret = snd_soc_register_component(&pdev->dev, - &sti_uniperiph_dai_component, - priv->dai, 1); + ret = devm_snd_soc_register_component(&pdev->dev, + &sti_uniperiph_dai_component, + priv->dai, 1); if (ret < 0) return ret; -- 2.1.0

2 1

[alsa-devel] [PATCH] ASoC: zx296702-i2s: Fix resource leak when unload module
by Axel Lin 30 Aug '15

30 Aug '15

Use devm_* API to fix leaks in current code. 1. Use devm_kzalloc to fix memory leak for zx_i2s when unload the module. 2. Use devm_snd_soc_register_component to ensure component is unregistered when unload the module. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/zte/zx296702-i2s.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/zte/zx296702-i2s.c b/sound/soc/zte/zx296702-i2s.c index 1930c42..1cad93d 100644 --- a/sound/soc/zte/zx296702-i2s.c +++ b/sound/soc/zte/zx296702-i2s.c @@ -380,7 +380,7 @@ static int zx_i2s_probe(struct platform_device *pdev) struct zx_i2s_info *zx_i2s; int ret; - zx_i2s = kzalloc(sizeof(*zx_i2s), GFP_KERNEL); + zx_i2s = devm_kzalloc(&pdev->dev, sizeof(*zx_i2s), GFP_KERNEL); if (!zx_i2s) return -ENOMEM; @@ -401,8 +401,8 @@ static int zx_i2s_probe(struct platform_device *pdev) writel_relaxed(0, zx_i2s->reg_base + ZX_I2S_FIFO_CTRL); platform_set_drvdata(pdev, zx_i2s); - ret = snd_soc_register_component(&pdev->dev, &zx_i2s_component, - &zx_i2s_dai, 1); + ret = devm_snd_soc_register_component(&pdev->dev, &zx_i2s_component, + &zx_i2s_dai, 1); if (ret) { dev_err(&pdev->dev, "Register DAI failed: %d\n", ret); return ret; -- 2.1.0

3 2

[alsa-devel] [PATCH 1/3] ASoC: pxa-ssp: Convert to devm_snd_soc_register_component
by Axel Lin 30 Aug '15

30 Aug '15

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/pxa/pxa-ssp.c | 11 ++--------- 1 file changed, 2 insertions(+), 9 deletions(-) diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index fbe2e93..3da485e 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -813,14 +813,8 @@ static const struct of_device_id pxa_ssp_of_ids[] = { static int asoc_ssp_probe(struct platform_device *pdev) { - return snd_soc_register_component(&pdev->dev, &pxa_ssp_component, - &pxa_ssp_dai, 1); -} - -static int asoc_ssp_remove(struct platform_device *pdev) -{ - snd_soc_unregister_component(&pdev->dev); - return 0; + return devm_snd_soc_register_component(&pdev->dev, &pxa_ssp_component, + &pxa_ssp_dai, 1); } static struct platform_driver asoc_ssp_driver = { @@ -830,7 +824,6 @@ static struct platform_driver asoc_ssp_driver = { }, .probe = asoc_ssp_probe, - .remove = asoc_ssp_remove, }; module_platform_driver(asoc_ssp_driver); -- 2.1.0

4 7

[alsa-devel] [PATCH 1/5] ASoC: au1x: Convert to devm_snd_soc_register_platform
by Axel Lin 30 Aug '15

30 Aug '15

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/au1x/dbdma2.c | 11 ++--------- sound/soc/au1x/dma.c | 11 ++--------- 2 files changed, 4 insertions(+), 18 deletions(-) diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index dd94fea..5741c0a 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -344,14 +344,8 @@ static int au1xpsc_pcm_drvprobe(struct platform_device *pdev) platform_set_drvdata(pdev, dmadata); - return snd_soc_register_platform(&pdev->dev, &au1xpsc_soc_platform); -} - -static int au1xpsc_pcm_drvremove(struct platform_device *pdev) -{ - snd_soc_unregister_platform(&pdev->dev); - - return 0; + return devm_snd_soc_register_platform(&pdev->dev, + &au1xpsc_soc_platform); } static struct platform_driver au1xpsc_pcm_driver = { @@ -359,7 +353,6 @@ static struct platform_driver au1xpsc_pcm_driver = { .name = "au1xpsc-pcm", }, .probe = au1xpsc_pcm_drvprobe, - .remove = au1xpsc_pcm_drvremove, }; module_platform_driver(au1xpsc_pcm_driver); diff --git a/sound/soc/au1x/dma.c b/sound/soc/au1x/dma.c index 24cc7f4..fcf5a9ad 100644 --- a/sound/soc/au1x/dma.c +++ b/sound/soc/au1x/dma.c @@ -312,14 +312,8 @@ static int alchemy_pcm_drvprobe(struct platform_device *pdev) platform_set_drvdata(pdev, ctx); - return snd_soc_register_platform(&pdev->dev, &alchemy_pcm_soc_platform); -} - -static int alchemy_pcm_drvremove(struct platform_device *pdev) -{ - snd_soc_unregister_platform(&pdev->dev); - - return 0; + return devm_snd_soc_register_platform(&pdev->dev, + &alchemy_pcm_soc_platform); } static struct platform_driver alchemy_pcmdma_driver = { @@ -327,7 +321,6 @@ static struct platform_driver alchemy_pcmdma_driver = { .name = "alchemy-pcm-dma", }, .probe = alchemy_pcm_drvprobe, - .remove = alchemy_pcm_drvremove, }; module_platform_driver(alchemy_pcmdma_driver); -- 2.1.0

3 10

Re: [alsa-devel] [pulseaudio-discuss] sound is distorted when pulseaudio system timer scheduling is used (tsched=1)
by Raymond Yau 29 Aug '15

29 Aug '15

> > I am having issue with pulseaudiosink, > I used following command to play a wav file > gst-launch filesrc location=<wav file path> ! wavparse ! pulseaudiosink > > if tsched is enabled, then play of wav file via pulseaudiosink gives me lots of Underrun, and the sound is distorted, following is the pulseaudio log when --log-level=3 > > I: [pulseaudio] protocol-native.c: Got credentials: uid=0 gid=0 success=1 > I: [pulseaudio] sink-input.c: Trying to change sample rate > I: [pulseaudio] alsa-sink.c: Updating rate for device hw:0, new rate is 48000 > I: [pulseaudio] source.c: Changed sampling rate successfully > I: [pulseaudio] sink.c: Changed sampling rate successfully > I: [pulseaudio] sink-input.c: Rate changed to 48000 Hz > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Trying resume... > I: [alsa-sink-HiFi wm8962-0] alsa-util.c: cannot disable ALSA period wakeups > I: [alsa-sink-HiFi wm8962-0] alsa-util.c: ALSA period wakeups were not disabled > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Time scheduling watermark is 18.38ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Resumed successfully... > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Starting playback. > I: [pulseaudio] sink-input.c: Created input 0 "pulsesink probe" on alsa_output.platform-sound.analog-s > tereo with sample spec s16le 2ch 48000Hz and channel map front-left,front-right > I: [pulseaudio] sink-input.c: media.name = "pulsesink probe" > I: [pulseaudio] sink-input.c: application.name = "gst-launch-0.10" > I: [pulseaudio] sink-input.c: native-protocol.peer = "UNIX socket client" > I: [pulseaudio] sink-input.c: native-protocol.version = "29" > I: [pulseaudio] sink-input.c: application.process.id = "985" > I: [pulseaudio] sink-input.c: application.process.user = "root" > I: [pulseaudio] sink-input.c: application.process.host = "mx6q" > I: [pulseaudio] sink-input.c: application.process.binary = "gst-launch-0.10" > I: [pulseaudio] sink-input.c: application.language = "C" > I: [pulseaudio] sink-input.c: application.process.machine_id = "0aa7a9566c734f608cbd36f187de8645" > I: [pulseaudio] sink-input.c: module-stream-restore.id = "sink-input-by-application-name:gst-launc > h-0.10" > I: [pulseaudio] protocol-native.c: Requested tlength=250.00 ms, minreq=20.00 ms > I: [pulseaudio] protocol-native.c: Final latency 250.00 ms = 105.00 ms + 2*20.00 ms + 105.00 ms > I: [pulseaudio] sink-input.c: Freeing input 0 "pulsesink probe" > I: [pulseaudio] sink-input.c: Created input 1 "Playback Stream" on alsa_output.platform-sound.analog-s > tereo with sample spec s16le 2ch 48000Hz and channel map front-left,front-right > I: [pulseaudio] sink-input.c: media.name = "Playback Stream" > I: [pulseaudio] sink-input.c: application.name = "gst-launch-0.10" > I: [pulseaudio] sink-input.c: native-protocol.peer = "UNIX socket client" > I: [pulseaudio] sink-input.c: native-protocol.version = "29" > I: [pulseaudio] sink-input.c: application.process.id = "985" > I: [pulseaudio] sink-input.c: application.process.user = "root" > I: [pulseaudio] sink-input.c: application.process.host = "mx6q" > I: [pulseaudio] sink-input.c: application.process.binary = "gst-launch-0.10" > I: [pulseaudio] sink-input.c: application.language = "C" > I: [pulseaudio] sink-input.c: application.process.machine_id = "0aa7a9566c734f608cbd36f187de8645" > I: [pulseaudio] sink-input.c: module-stream-restore.id = "sink-input-by-application-name:gst-launc > h-0.10" > I: [pulseaudio] protocol-native.c: Requested tlength=200.00 ms, minreq=10.00 ms > I: [pulseaudio] protocol-native.c: Final latency 200.00 ms = 90.00 ms + 2*10.00 ms + 90.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing wakeup watermark to 28.38 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing wakeup watermark to 38.38 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing wakeup watermark to 48.38 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing wakeup watermark to 58.38 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing wakeup watermark to 68.38 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing wakeup watermark to 78.38 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing wakeup watermark to 80.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 1.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 2.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 4.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 8.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 16.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 26.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 36.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 46.00 ms > E: [alsa-sink-HiFi wm8962-0] alsa-sink.c: ALSA woke us up to write new data to the device, but there w > as actually nothing to write! > E: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Most likely this is a bug in the ALSA driver '(null)'. Pleas > e report this issue to the ALSA developers. > E: [alsa-sink-HiFi wm8962-0] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent sn > d_pcm_avail() returned 0 or another value < min_avail. > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 56.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 66.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 76.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 86.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 96.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing wakeup watermark to 86.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 106.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing wakeup watermark to 96.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 116.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing wakeup watermark to 106.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 126.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing wakeup watermark to 116.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 136.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing wakeup watermark to 126.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Underrun! > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 146.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Underrun! > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing wakeup watermark to 136.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Underrun! > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 156.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing wakeup watermark to 146.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 166.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing wakeup watermark to 156.00 ms > I: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Increasing minimal latency to 176.00 ms > I: [pulseaudio] sink-input.c: Freeing input 1 "Playback Stream" > I: [pulseaudio] client.c: Freed 4 "gst-launch-0.10" > I: [pulseaudio] protocol-native.c: Connection died. > > from the log it seems like, when system timer based scheduling is used, then pulseaudio will try to keep minimum watermark/latency, only when Underrun occurs, it will increase watermark/latency. > > Is this the expected behaviour of pulseaudio 'glict free' feature? > And from the log there is some error message complain about ALSA driver > E: [alsa-sink-HiFi wm8962-0] alsa-sink.c: ALSA woke us up to write new data to the device, but there w > as actually nothing to write! > E: [alsa-sink-HiFi wm8962-0] alsa-sink.c: Most likely this is a bug in the ALSA driver '(null)'. Pleas > e report this issue to the ALSA developers. > E: [alsa-sink-HiFi wm8962-0] alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent sn > d_pcm_avail() returned 0 or another value < min_avail. > > does this mean, there is some bug in ALSA driver, or it just indicates the ALSA driver doesn't support glitch free feature very well The timer base scheduling require sound card driver provide accurate hw_ptr position at any time. So far only two sound card drivers snd-hda-intel and snd-oxygen can provide accurate hw_ptr position from hardware register static snd_pcm_uframes_t dmaengine_pcm_pointer( struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); if (pcm->flags & SND_DMAENGINE_PCM_FLAG_NO_RESIDUE) return snd_dmaengine_pcm_pointer_no_residue(substream); else return snd_dmaengine_pcm_pointer(substream); } http://lxr.free-electrons.com/ident?i=dmaengine_tx_status Seem none of soc sound drivers can report position accurate than period size Is there any reason to enable timer based scheduling by default ? http://mailman.alsa-project.org/pipermail/alsa-devel/2014-September/081501.… You can try alexandra 's program to find out how accuate hw_ptr can your sound card driver can report ?

1 0

[alsa-devel] [PATCH] ALSA: fireworks/bebob/dice/oxfw: fix substreams counting at vmalloc failure
by Takashi Sakamoto 29 Aug '15

29 Aug '15

In PCM core, when hw_params() in each driver returns error, the state of PCM substream is kept as 'open'. In this case, current drivers for sound units on IEEE 1394 bus doesn't decrement substream counter in hw_free() correctly. This causes these drivers to keep streams even if not required. This commit fixes this bug. When snd_pcm_lib_alloc_vmalloc_buffer() fails, hw_params function in each driver returns without incrementing the counter. Reported-by: Takashi Iwai <tiwai(a)suse.de> Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob_pcm.c | 20 ++++++++++++++++---- sound/firewire/dice/dice-pcm.c | 18 ++++++++++++++---- sound/firewire/fireworks/fireworks_pcm.c | 18 ++++++++++++++---- sound/firewire/oxfw/oxfw-pcm.c | 17 +++++++++++++---- 4 files changed, 57 insertions(+), 16 deletions(-) diff --git a/sound/firewire/bebob/bebob_pcm.c b/sound/firewire/bebob/bebob_pcm.c index 7a2c1f5..c0f018a 100644 --- a/sound/firewire/bebob/bebob_pcm.c +++ b/sound/firewire/bebob/bebob_pcm.c @@ -211,26 +211,38 @@ pcm_capture_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { struct snd_bebob *bebob = substream->private_data; + int err; + + err = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); + if (err < 0) + return err; if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) atomic_inc(&bebob->substreams_counter); amdtp_stream_set_pcm_format(&bebob->tx_stream, params_format(hw_params)); - return snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + + return 0; } static int pcm_playback_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { struct snd_bebob *bebob = substream->private_data; + int err; + + err = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); + if (err < 0) + return err; if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) atomic_inc(&bebob->substreams_counter); amdtp_stream_set_pcm_format(&bebob->rx_stream, params_format(hw_params)); - return snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + + return 0; } static int diff --git a/sound/firewire/dice/dice-pcm.c b/sound/firewire/dice/dice-pcm.c index f7771451..4e67b1d 100644 --- a/sound/firewire/dice/dice-pcm.c +++ b/sound/firewire/dice/dice-pcm.c @@ -230,6 +230,12 @@ static int capture_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { struct snd_dice *dice = substream->private_data; + int err; + + err = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); + if (err < 0) + return err; if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { mutex_lock(&dice->mutex); @@ -240,13 +246,18 @@ static int capture_hw_params(struct snd_pcm_substream *substream, amdtp_stream_set_pcm_format(&dice->tx_stream, params_format(hw_params)); - return snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + return 0; } static int playback_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { struct snd_dice *dice = substream->private_data; + int err; + + err = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); + if (err < 0) + return err; if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { mutex_lock(&dice->mutex); @@ -257,8 +268,7 @@ static int playback_hw_params(struct snd_pcm_substream *substream, amdtp_stream_set_pcm_format(&dice->rx_stream, params_format(hw_params)); - return snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + return 0; } static int capture_hw_free(struct snd_pcm_substream *substream) diff --git a/sound/firewire/fireworks/fireworks_pcm.c b/sound/firewire/fireworks/fireworks_pcm.c index 8a34753..c30b2ff 100644 --- a/sound/firewire/fireworks/fireworks_pcm.c +++ b/sound/firewire/fireworks/fireworks_pcm.c @@ -244,25 +244,35 @@ static int pcm_capture_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { struct snd_efw *efw = substream->private_data; + int err; + + err = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); + if (err < 0) + return err; if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) atomic_inc(&efw->capture_substreams); amdtp_stream_set_pcm_format(&efw->tx_stream, params_format(hw_params)); - return snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + return 0; } static int pcm_playback_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { struct snd_efw *efw = substream->private_data; + int err; + + err = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); + if (err < 0) + return err; if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) atomic_inc(&efw->playback_substreams); amdtp_stream_set_pcm_format(&efw->rx_stream, params_format(hw_params)); - return snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + return 0; } static int pcm_capture_hw_free(struct snd_pcm_substream *substream) diff --git a/sound/firewire/oxfw/oxfw-pcm.c b/sound/firewire/oxfw/oxfw-pcm.c index 67ade07..9c73930 100644 --- a/sound/firewire/oxfw/oxfw-pcm.c +++ b/sound/firewire/oxfw/oxfw-pcm.c @@ -231,7 +231,12 @@ static int pcm_capture_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { struct snd_oxfw *oxfw = substream->private_data; + int err; + err = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); + if (err < 0) + return err; if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { mutex_lock(&oxfw->mutex); @@ -241,13 +246,18 @@ static int pcm_capture_hw_params(struct snd_pcm_substream *substream, amdtp_stream_set_pcm_format(&oxfw->tx_stream, params_format(hw_params)); - return snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + return 0; } static int pcm_playback_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *hw_params) { struct snd_oxfw *oxfw = substream->private_data; + int err; + + err = snd_pcm_lib_alloc_vmalloc_buffer(substream, + params_buffer_bytes(hw_params)); + if (err < 0) + return err; if (substream->runtime->status->state == SNDRV_PCM_STATE_OPEN) { mutex_lock(&oxfw->mutex); @@ -257,8 +267,7 @@ static int pcm_playback_hw_params(struct snd_pcm_substream *substream, amdtp_stream_set_pcm_format(&oxfw->rx_stream, params_format(hw_params)); - return snd_pcm_lib_alloc_vmalloc_buffer(substream, - params_buffer_bytes(hw_params)); + return 0; } static int pcm_capture_hw_free(struct snd_pcm_substream *substream) -- 2.1.4

3 2

[alsa-devel] [PATCH 00/25 v2] ALSA: support AMDTP variants
by Takashi Sakamoto 29 Aug '15

29 Aug '15

This patchset for Linux 4.3 updates my previous one: [alsa-devel] [PATCH 00/25] ALSA: support AMDTP variants http://mailman.alsa-project.org/pipermail/alsa-devel/2015-August/096338.html As the same as the previous, the second patch is too large to be blasted by mailing list server. I request server administrators to grant it. Changes: * Enable to select as kernel module or the others. * Add a comment for memory barriers in firewire-lib. * Add a comment about using jiffies in firewire-lib. * Use jiffies_64 helpers instead of jiffies helpers in firewire-lib. Rest of work: * Using proper clocksource for MIDI throttle in asynchronous transaction helper. * Handling control/status messages for TASCAM driver. * Supporting MIDI exclusive message in outgoing packets for TASCAM driver. * Researching a mechanism to de-synchronize sometimes for Digi00x family. You can see this patchset as a part in this branch. https://github.com/takaswie/sound/tree/amdtp-variants Takashi Sakamoto (25): ALSA: firewire-lib: rename 'amdtp' to 'amdtp-stream' for functional separation ALSA: firewire-lib: functional separation for packet transmission layer and data processing layer ALSA: firewire-lib: add helper functions for asynchronous MIDI port ALSA: firewire-lib: add a restriction for a transaction at once ALSA: firewire-lib: schedule tasklet again when MIDI substream has rest of MIDI messages ALSA: firewire-lib: add throttle for MIDI data rate ALSA: firewire-lib: avoid endless loop to transfer MIDI messages at fatal error ALSA: firewire-digi00x: add skeleton for Digi 002/003 family ALSA: firewire-digi00x: add data 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 MIDI functionality ALSA: firewire-digi00x: add hwdep interface ALSA: firewire-digi00x: add support for asynchronous messaging ALSA: firewire-digi00x: add support for MIDI ports for machine control ALSA: firewire-tascam: add skeleton for TASCAM FireWire series ALSA: firewire-tascam: add a structure for model-dependent parameters. ALSA: firewire-tascam: add proc node to show firmware information ALSA: firewire-tascam: add data processing layer ALSA: firewire-tascam: add streaming functionality ALSA: firewire-tascam: add PCM functionality ALSA: firewire-tascam: add transaction functionality ALSA: firewire-tascam: add MIDI functionality ALSA: firewire-tascam: add hwdep interface include/uapi/sound/asound.h | 4 +- include/uapi/sound/firewire.h | 9 + sound/firewire/Kconfig | 27 ++ sound/firewire/Makefile | 4 +- sound/firewire/amdtp-am824.c | 451 ++++++++++++++++++++++++ sound/firewire/amdtp-am824.h | 47 +++ sound/firewire/{amdtp.c => amdtp-stream.c} | 375 ++++---------------- sound/firewire/{amdtp.h => amdtp-stream.h} | 113 ++---- sound/firewire/bebob/bebob.h | 2 +- sound/firewire/bebob/bebob_midi.c | 16 +- sound/firewire/bebob/bebob_pcm.c | 12 +- sound/firewire/bebob/bebob_stream.c | 34 +- sound/firewire/dice/dice-midi.c | 16 +- sound/firewire/dice/dice-pcm.c | 12 +- sound/firewire/dice/dice-stream.c | 41 ++- sound/firewire/dice/dice.h | 2 +- sound/firewire/digi00x/Makefile | 4 + sound/firewire/digi00x/amdtp-dot.c | 436 +++++++++++++++++++++++ sound/firewire/digi00x/digi00x-hwdep.c | 200 +++++++++++ sound/firewire/digi00x/digi00x-midi.c | 198 +++++++++++ sound/firewire/digi00x/digi00x-pcm.c | 353 +++++++++++++++++++ sound/firewire/digi00x/digi00x-proc.c | 99 ++++++ sound/firewire/digi00x/digi00x-stream.c | 422 +++++++++++++++++++++++ sound/firewire/digi00x/digi00x-transaction.c | 137 ++++++++ sound/firewire/digi00x/digi00x.c | 174 ++++++++++ sound/firewire/digi00x/digi00x.h | 157 +++++++++ sound/firewire/fcp.c | 2 +- sound/firewire/fireworks/fireworks.c | 12 +- sound/firewire/fireworks/fireworks.h | 2 +- sound/firewire/fireworks/fireworks_midi.c | 16 +- sound/firewire/fireworks/fireworks_pcm.c | 10 +- sound/firewire/fireworks/fireworks_stream.c | 8 +- sound/firewire/lib.c | 147 ++++++++ sound/firewire/lib.h | 52 +++ sound/firewire/oxfw/oxfw-midi.c | 16 +- sound/firewire/oxfw/oxfw-pcm.c | 10 +- sound/firewire/oxfw/oxfw-stream.c | 11 +- sound/firewire/oxfw/oxfw.h | 2 +- sound/firewire/tascam/Makefile | 4 + sound/firewire/tascam/amdtp-tascam.c | 248 +++++++++++++ sound/firewire/tascam/tascam-hwdep.c | 202 +++++++++++ sound/firewire/tascam/tascam-midi.c | 162 +++++++++ sound/firewire/tascam/tascam-pcm.c | 298 ++++++++++++++++ sound/firewire/tascam/tascam-proc.c | 88 +++++ sound/firewire/tascam/tascam-stream.c | 497 +++++++++++++++++++++++++++ sound/firewire/tascam/tascam-transaction.c | 291 ++++++++++++++++ sound/firewire/tascam/tascam.c | 230 +++++++++++++ sound/firewire/tascam/tascam.h | 140 ++++++++ 48 files changed, 5295 insertions(+), 498 deletions(-) create mode 100644 sound/firewire/amdtp-am824.c create mode 100644 sound/firewire/amdtp-am824.h rename sound/firewire/{amdtp.c => amdtp-stream.c} (70%) rename sound/firewire/{amdtp.h => amdtp-stream.h} (77%) 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-midi.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 create mode 100644 sound/firewire/tascam/Makefile create mode 100644 sound/firewire/tascam/amdtp-tascam.c create mode 100644 sound/firewire/tascam/tascam-hwdep.c create mode 100644 sound/firewire/tascam/tascam-midi.c create mode 100644 sound/firewire/tascam/tascam-pcm.c create mode 100644 sound/firewire/tascam/tascam-proc.c create mode 100644 sound/firewire/tascam/tascam-stream.c create mode 100644 sound/firewire/tascam/tascam-transaction.c create mode 100644 sound/firewire/tascam/tascam.c create mode 100644 sound/firewire/tascam/tascam.h -- 2.1.4

2 35

[alsa-devel] [PATCH] ASoC: nau8825: Add driver for headset chip Nuvoton 8825
by Anatol Pomozov 29 Aug '15

29 Aug '15

Signed-off-by: Anatol Pomozov <anatol.pomozov(a)gmail.com> --- .../devicetree/bindings/sound/nau8825.txt | 14 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/nau8825.c | 860 +++++++++++++++++++++ sound/soc/codecs/nau8825.h | 258 +++++++ 5 files changed, 1138 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/nau8825.txt create mode 100644 sound/soc/codecs/nau8825.c create mode 100644 sound/soc/codecs/nau8825.h diff --git a/Documentation/devicetree/bindings/sound/nau8825.txt b/Documentation/devicetree/bindings/sound/nau8825.txt new file mode 100644 index 0000000..0334be3 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/nau8825.txt @@ -0,0 +1,14 @@ +Nuvoton NAU8825 audio codec + +This device supports I2C only. + +Required properties: + - compatible : Must be "nuvoton,nau8825" + - reg : the I2C address of the device. This is either 0x1a (CSB=0) or 0x1b (CSB=1). + +Example: + + codec: nau8825@1a { + compatible = "nuvoton,nau8825"; + reg = <0x1a>; + }; diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 7fcbc22..1a3639c 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -78,6 +78,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_MAX9877 if I2C select SND_SOC_MC13783 if MFD_MC13XXX select SND_SOC_ML26124 if I2C + select SND_SOC_NAU8825 if I2C select SND_SOC_HDMI_CODEC select SND_SOC_PCM1681 if I2C select SND_SOC_PCM1792A if SPI_MASTER @@ -892,6 +893,9 @@ config SND_SOC_MC13783 config SND_SOC_ML26124 tristate +config SND_SOC_NAU8825 + tristate + config SND_SOC_TPA6130A2 tristate "Texas Instruments TPA6130A2 headphone amplifier" depends on I2C diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 5b46c8f..4874881 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -71,6 +71,7 @@ snd-soc-max98925-objs := max98925.o snd-soc-max9850-objs := max9850.o snd-soc-mc13783-objs := mc13783.o snd-soc-ml26124-objs := ml26124.o +snd-soc-nau8825-objs := nau8825.o snd-soc-hdmi-codec-objs := hdmi.o snd-soc-pcm1681-objs := pcm1681.o snd-soc-pcm1792a-codec-objs := pcm1792a.o @@ -262,6 +263,7 @@ obj-$(CONFIG_SND_SOC_MAX98925) += snd-soc-max98925.o obj-$(CONFIG_SND_SOC_MAX9850) += snd-soc-max9850.o obj-$(CONFIG_SND_SOC_MC13783) += snd-soc-mc13783.o obj-$(CONFIG_SND_SOC_ML26124) += snd-soc-ml26124.o +obj-$(CONFIG_SND_SOC_NAU8825) += snd-soc-nau8825.o obj-$(CONFIG_SND_SOC_HDMI_CODEC) += snd-soc-hdmi-codec.o obj-$(CONFIG_SND_SOC_PCM1681) += snd-soc-pcm1681.o obj-$(CONFIG_SND_SOC_PCM1792A) += snd-soc-pcm1792a-codec.o diff --git a/sound/soc/codecs/nau8825.c b/sound/soc/codecs/nau8825.c new file mode 100644 index 0000000..1c99308b --- /dev/null +++ b/sound/soc/codecs/nau8825.c @@ -0,0 +1,860 @@ +/* + * NAU8825 audio codec driver + * + * Copyright 2015 Nuvoton Technology Corp. + * Copyright 2015 Google Chromium project. + * Author: Anatol Pomozov <anatol(a)chromium.org> + * + * Licensed under the GPL-2. + */ + +#include <linux/module.h> +#include <linux/delay.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/regmap.h> +#include <linux/slab.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <linux/input.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include <sound/jack.h> + +#include "nau8825.h" + +struct nau8825 { + struct device *dev; + struct regmap *regmap; + struct snd_soc_dapm_context *dapm; + int irq; + struct snd_soc_jack *jack; + int button_pressed; +}; + +static const struct reg_default nau8825_reg_defaults[] = { + { NAU8825_REG_ENA_CTRL, 0x00ff }, + { NAU8825_REG_CLK_DIVIDER, 0x0050 }, + { NAU8825_REG_FLL1, 0x0 }, + { NAU8825_REG_FLL2, 0x3126 }, + { NAU8825_REG_FLL3, 0x0008 }, + { NAU8825_REG_FLL4, 0x0010 }, + { NAU8825_REG_FLL5, 0x0 }, + { NAU8825_REG_FLL6, 0x6000 }, + { NAU8825_REG_FLL_VCO_RSV, 0xf13c }, + { NAU8825_REG_HSD_CTRL, 0x000c }, + { NAU8825_REG_JACK_DET_CTRL, 0x0 }, + { NAU8825_REG_INTERRUPT_MASK, 0x0 }, + { NAU8825_REG_INTERRUPT_DIS_CTRL, 0xffff }, + { NAU8825_REG_SAR_CTRL, 0x0015 }, + { NAU8825_REG_KEYDET_CTRL, 0x0110 }, + { NAU8825_REG_VDET_THRESHOLD_1, 0x0 }, + { NAU8825_REG_VDET_THRESHOLD_2, 0x0 }, + { NAU8825_REG_VDET_THRESHOLD_3, 0x0 }, + { NAU8825_REG_VDET_THRESHOLD_4, 0x0 }, + { NAU8825_REG_GPIO34_CTRL, 0x0 }, + { NAU8825_REG_GPIO12_CTRL, 0x0 }, + { NAU8825_REG_TDM_CTRL, 0x0 }, + { NAU8825_REG_I2S_PCM_CTRL1, 0x000b }, + { NAU8825_REG_I2S_PCM_CTRL2, 0x8010 }, + { NAU8825_REG_LEFT_TIME_SLOT, 0x0 }, + { NAU8825_REG_RIGHT_TIME_SLOT, 0x0 }, + { NAU8825_REG_BIQ_CTRL, 0x0 }, + { NAU8825_REG_BIQ_COF1, 0x0 }, + { NAU8825_REG_BIQ_COF2, 0x0 }, + { NAU8825_REG_BIQ_COF3, 0x0 }, + { NAU8825_REG_BIQ_COF4, 0x0 }, + { NAU8825_REG_BIQ_COF5, 0x0 }, + { NAU8825_REG_BIQ_COF6, 0x0 }, + { NAU8825_REG_BIQ_COF7, 0x0 }, + { NAU8825_REG_BIQ_COF8, 0x0 }, + { NAU8825_REG_BIQ_COF9, 0x0 }, + { NAU8825_REG_BIQ_COF10, 0x0 }, + { NAU8825_REG_ADC_RATE, 0x0010 }, + { NAU8825_REG_DAC_CTRL1, 0x0001 }, + { NAU8825_REG_DAC_CTRL2, 0x0 }, + { NAU8825_REG_DAC_DGAIN_CTRL, 0x0 }, + { NAU8825_REG_ADC_DGAIN_CTRL, 0x00cf }, + { NAU8825_REG_MUTE_CTRL, 0x0 }, + { NAU8825_REG_HSVOL_CTRL, 0x0 }, + { NAU8825_REG_DACL_CTRL, 0x02cf }, + { NAU8825_REG_DACR_CTRL, 0x00cf }, + { NAU8825_REG_ADC_DRC_KNEE_IP12, 0x1486 }, + { NAU8825_REG_ADC_DRC_KNEE_IP34, 0x0f12 }, + { NAU8825_REG_ADC_DRC_SLOPES, 0x25ff }, + { NAU8825_REG_ADC_DRC_ATKDCY, 0x3457 }, + { NAU8825_REG_DAC_DRC_KNEE_IP12, 0x1486 }, + { NAU8825_REG_DAC_DRC_KNEE_IP34, 0x0f12 }, + { NAU8825_REG_DAC_DRC_SLOPES, 0x25f9 }, + { NAU8825_REG_DAC_DRC_ATKDCY, 0x3457 }, + { NAU8825_REG_IMM_MODE_CTRL, 0x0 }, + { NAU8825_REG_CLASSG_CTRL, 0x0 }, + { NAU8825_REG_OPT_EFUSE_CTRL, 0x0 }, + { NAU8825_REG_MISC_CTRL, 0x0 }, + { NAU8825_REG_BIAS_ADJ, 0x0 }, + { NAU8825_REG_TRIM_SETTINGS, 0x0 }, + { NAU8825_REG_ANALOG_CONTROL_1, 0x0 }, + { NAU8825_REG_ANALOG_CONTROL_2, 0x0 }, + { NAU8825_REG_ANALOG_ADC_1, 0x0011 }, + { NAU8825_REG_ANALOG_ADC_2, 0x0020 }, + { NAU8825_REG_RDAC, 0x0008 }, + { NAU8825_REG_MIC_BIAS, 0x0006 }, + { NAU8825_REG_BOOST, 0x0 }, + { NAU8825_REG_FEPGA, 0x0 }, + { NAU8825_REG_POWER_UP_CONTROL, 0x0 }, + { NAU8825_REG_CHARGE_PUMP, 0x0 }, +}; + +static bool nau8825_readable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_REG_ENA_CTRL: + case NAU8825_REG_CLK_DIVIDER ... NAU8825_REG_FLL_VCO_RSV: + case NAU8825_REG_HSD_CTRL ... NAU8825_REG_JACK_DET_CTRL: + case NAU8825_REG_INTERRUPT_MASK ... NAU8825_REG_KEYDET_CTRL: + case NAU8825_REG_VDET_THRESHOLD_1 ... NAU8825_REG_DACR_CTRL: + case NAU8825_REG_ADC_DRC_KNEE_IP12 ... NAU8825_REG_ADC_DRC_ATKDCY: + case NAU8825_REG_DAC_DRC_KNEE_IP12 ... NAU8825_REG_DAC_DRC_ATKDCY: + case NAU8825_REG_IMM_MODE_CTRL ... NAU8825_REG_IMM_RMS_R: + case NAU8825_REG_CLASSG_CTRL ... NAU8825_REG_OPT_EFUSE_CTRL: + case NAU8825_REG_MISC_CTRL: + case NAU8825_REG_I2C_DEVICE_ID ... NAU8825_REG_SARDOUT_RAM_STATUS: + case NAU8825_REG_BIAS_ADJ: + case NAU8825_REG_TRIM_SETTINGS ... NAU8825_REG_ANALOG_CONTROL_2: + case NAU8825_REG_ANALOG_ADC_1 ... NAU8825_REG_MIC_BIAS: + case NAU8825_REG_BOOST ... NAU8825_REG_FEPGA: + case NAU8825_REG_POWER_UP_CONTROL ... NAU8825_REG_GENERAL_STATUS: + return true; + default: + return false; + } + +} + +static bool nau8825_writeable_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_REG_RESET ... NAU8825_REG_ENA_CTRL: + case NAU8825_REG_CLK_DIVIDER ... NAU8825_REG_FLL_VCO_RSV: + case NAU8825_REG_HSD_CTRL ... NAU8825_REG_JACK_DET_CTRL: + case NAU8825_REG_INTERRUPT_MASK: + case NAU8825_REG_INT_CLR_KEY_STATUS ... NAU8825_REG_KEYDET_CTRL: + case NAU8825_REG_VDET_THRESHOLD_1 ... NAU8825_REG_DACR_CTRL: + case NAU8825_REG_ADC_DRC_KNEE_IP12 ... NAU8825_REG_ADC_DRC_ATKDCY: + case NAU8825_REG_DAC_DRC_KNEE_IP12 ... NAU8825_REG_DAC_DRC_ATKDCY: + case NAU8825_REG_IMM_MODE_CTRL: + case NAU8825_REG_CLASSG_CTRL ... NAU8825_REG_OPT_EFUSE_CTRL: + case NAU8825_REG_MISC_CTRL: + case NAU8825_REG_BIAS_ADJ: + case NAU8825_REG_TRIM_SETTINGS ... NAU8825_REG_ANALOG_CONTROL_2: + case NAU8825_REG_ANALOG_ADC_1 ... NAU8825_REG_MIC_BIAS: + case NAU8825_REG_BOOST ... NAU8825_REG_FEPGA: + case NAU8825_REG_POWER_UP_CONTROL ... NAU8825_REG_CHARGE_PUMP: + return true; + default: + return false; + } +} + +static bool nau8825_volatile_reg(struct device *dev, unsigned int reg) +{ + switch (reg) { + case NAU8825_REG_RESET: + case NAU8825_REG_IRQ_STATUS: + case NAU8825_REG_INT_CLR_KEY_STATUS: + case NAU8825_REG_IMM_RMS_L: + case NAU8825_REG_IMM_RMS_R: + case NAU8825_REG_I2C_DEVICE_ID: + case NAU8825_REG_SARDOUT_RAM_STATUS: + case NAU8825_REG_CHARGE_PUMP_INPUT_READ: + case NAU8825_REG_GENERAL_STATUS: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_MINMAX_MUTE(adc_vol_tlv, -10300, 2400); +static const DECLARE_TLV_DB_MINMAX_MUTE(sidetone_vol_tlv, -4200, 0); +static const DECLARE_TLV_DB_MINMAX(dac_vol_tlv, -5400, 0); +static const DECLARE_TLV_DB_MINMAX(fepga_gain_tlv, -100, 3600); + +static const struct snd_kcontrol_new nau8825_snd_controls[] = { + SOC_SINGLE_TLV("MIC Volume", NAU8825_REG_ADC_DGAIN_CTRL, + 0, 0xff, 0, adc_vol_tlv), + SOC_DOUBLE_TLV("HP Sidetone Volume", NAU8825_REG_ADC_DGAIN_CTRL, + 12, 8, 0x0f, 0, sidetone_vol_tlv), + SOC_DOUBLE_TLV("HP Volume", NAU8825_REG_HSVOL_CTRL, + 6, 0, 0x3f, 1, dac_vol_tlv), + SOC_SINGLE_TLV("Frontend PGA Gain", NAU8825_REG_POWER_UP_CONTROL, + 8, 37, 0, fepga_gain_tlv), +}; + +static int nau8825_charge_pump_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + struct regmap *regmap = nau8825->regmap; + + if (SND_SOC_DAPM_EVENT_ON(event)) { + regmap_update_bits(regmap, NAU8825_REG_CHARGE_PUMP, + NAU8825_JAMNODCLOW, NAU8825_JAMNODCLOW); + } else { + regmap_update_bits(regmap, NAU8825_REG_CHARGE_PUMP, + NAU8825_JAMNODCLOW, 0); + } + + return 0; +} + +static int nau8825_output_driver_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + struct regmap *regmap = nau8825->regmap; + + if (SND_SOC_DAPM_EVENT_ON(event)) { + /* Power Up L and R Output Drivers */ + regmap_update_bits(regmap, NAU8825_REG_POWER_UP_CONTROL, 0x3c, + 0x3c); + /* Power up Main Output Drivers (The main driver must be turned + on after the predriver to avoid pops) */ + regmap_update_bits(regmap, NAU8825_REG_POWER_UP_CONTROL, 0x3, + 0x3); + } else { + /* Power Down L and R Output Drivers */ + regmap_update_bits(regmap, NAU8825_REG_POWER_UP_CONTROL, 0x3f, + 0x0); + } + + return 0; +} + +/* DAC Mux 0x33[9] and 0x34[9] */ +static const char * const nau8825_dac_src[] = { + "DACL", "DACR", +}; + +static SOC_ENUM_SINGLE_DECL( + nau8825_dacl_enum, NAU8825_REG_DACL_CTRL, + NAU8825_DACL_CH_SEL_SFT, nau8825_dac_src); + +static SOC_ENUM_SINGLE_DECL( + nau8825_dacr_enum, NAU8825_REG_DACR_CTRL, + NAU8825_DACR_CH_SEL_SFT, nau8825_dac_src); + +static const struct snd_kcontrol_new nau8825_dacl_mux = + SOC_DAPM_ENUM("DACL Source", nau8825_dacl_enum); + +static const struct snd_kcontrol_new nau8825_dacr_mux = + SOC_DAPM_ENUM("DACR Source", nau8825_dacr_enum); + +static const struct snd_soc_dapm_widget nau8825_dapm_widgets[] = { + SND_SOC_DAPM_INPUT("MIC"), + SND_SOC_DAPM_MICBIAS("MICBIAS", NAU8825_REG_MIC_BIAS, 8, 0), + + SND_SOC_DAPM_PGA("Frontend PGA", NAU8825_REG_POWER_UP_CONTROL, 14, 0, + NULL, 0), + + SND_SOC_DAPM_ADC("ADC", NULL, NAU8825_REG_ENA_CTRL, 8, 0), + SND_SOC_DAPM_SUPPLY("ADC Clock", NAU8825_REG_ENA_CTRL, 7, 0, NULL, 0), + + /* ADC for button press detection */ + SND_SOC_DAPM_ADC("SAR", NULL, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_ADC_EN_SFT, 0), + + SND_SOC_DAPM_DAC("ADACL", NULL, NAU8825_REG_RDAC, 12, 0), + SND_SOC_DAPM_DAC("ADACR", NULL, NAU8825_REG_RDAC, 13, 0), + SND_SOC_DAPM_SUPPLY("ADACL Clock", NAU8825_REG_RDAC, 8, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADACR Clock", NAU8825_REG_RDAC, 9, 0, NULL, 0), + + SND_SOC_DAPM_DAC("DDACR", NULL, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_DACR_SFT, 0), + SND_SOC_DAPM_DAC("DDACL", NULL, NAU8825_REG_ENA_CTRL, + NAU8825_ENABLE_DACL_SFT, 0), + SND_SOC_DAPM_SUPPLY("DDAC Clock", NAU8825_REG_ENA_CTRL, 6, 0, NULL, 0), + + SND_SOC_DAPM_MUX("DACL Mux", SND_SOC_NOPM, 0, 0, + &nau8825_dacl_mux), + SND_SOC_DAPM_MUX("DACR Mux", SND_SOC_NOPM, 0, 0, + &nau8825_dacr_mux), + + SND_SOC_DAPM_PGA("HP amp L", NAU8825_REG_CLASSG_CTRL, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("HP amp R", NAU8825_REG_CLASSG_CTRL, 2, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("HP amp power", NAU8825_REG_CLASSG_CTRL, 0, 0, NULL, + 0), + + SND_SOC_DAPM_SUPPLY("Change Pump", NAU8825_REG_CHARGE_PUMP, 5, 0, + nau8825_charge_pump_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + SND_SOC_DAPM_SUPPLY("DACL Power", NAU8825_REG_CHARGE_PUMP, 8, 1, NULL, + 0), + SND_SOC_DAPM_SUPPLY("DACR Power", NAU8825_REG_CHARGE_PUMP, 9, 1, NULL, + 0), + + SND_SOC_DAPM_OUT_DRV_E("Output Driver", SND_SOC_NOPM, 0, 0, NULL, 0, + nau8825_output_driver_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), +}; + +static const struct snd_soc_dapm_route nau8825_dapm_routes[] = { + {"Frontend PGA", NULL, "MIC"}, + {"ADC", NULL, "Frontend PGA"}, + {"ADC", NULL, "ADC Clock"}, + {"Capture", NULL, "ADC"}, + + {"DDACL", NULL, "Playback"}, + {"DDACR", NULL, "Playback"}, + {"DDACL", NULL, "DDAC Clock"}, + {"DDACR", NULL, "DDAC Clock"}, + {"DDACL", NULL, "DACL Power"}, + {"DDACR", NULL, "DACR Power"}, + {"DACL Mux", "DACL", "DDACL"}, + {"DACL Mux", "DACR", "DDACR"}, + {"DACR Mux", "DACL", "DDACL"}, + {"DACR Mux", "DACR", "DDACR"}, + {"HP amp L", NULL, "Change Pump"}, + {"HP amp R", NULL, "Change Pump"}, + {"HP amp L", NULL, "DACL Mux"}, + {"HP amp R", NULL, "DACR Mux"}, + {"HP amp L", NULL, "HP amp power"}, + {"HP amp R", NULL, "HP amp power"}, + {"ADACL", NULL, "HP amp L"}, + {"ADACR", NULL, "HP amp R"}, + {"ADACL", NULL, "ADACL Clock"}, + {"ADACR", NULL, "ADACR Clock"}, + {"Output Driver", NULL, "ADACL"}, + {"Output Driver", NULL, "ADACR"}, + {"HPOL", NULL, "Output Driver"}, + {"HPOR", NULL, "Output Driver"}, +}; + +static int nau8825_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + unsigned int val_len = 0; + + switch (params_width(params)) { + case 16: + val_len |= NAU8825_I2S_DL_16; + break; + case 20: + val_len |= NAU8825_I2S_DL_20; + break; + case 24: + val_len |= NAU8825_I2S_DL_24; + break; + case 32: + val_len |= NAU8825_I2S_DL_32; + break; + default: + return -EINVAL; + } + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL1, + NAU8825_I2S_DL_MASK, val_len); + + return 0; +} + +static int nau8825_mute(struct snd_soc_dai *codec_dai, int mute) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + int reg_val; + + reg_val = mute ? NAU8825_HP_MUTE : 0; + regmap_update_bits(nau8825->regmap, NAU8825_REG_HSVOL_CTRL, + NAU8825_HP_MUTE, reg_val); + + return 0; +} + +static int nau8825_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + unsigned int ctrl1_val = 0, ctrl2_val = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + ctrl2_val |= NAU8825_I2S_MS_MASTER; + break; + case SND_SOC_DAIFMT_CBS_CFS: + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + ctrl1_val |= NAU8825_I2S_BP_INV; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + ctrl1_val |= NAU8825_I2S_DF_I2S; + break; + case SND_SOC_DAIFMT_LEFT_J: + ctrl1_val |= NAU8825_I2S_DF_LEFT; + break; + case SND_SOC_DAIFMT_RIGHT_J: + ctrl1_val |= NAU8825_I2S_DF_RIGTH; + break; + case SND_SOC_DAIFMT_DSP_A: + ctrl1_val |= NAU8825_I2S_DF_PCM_AB; + break; + case SND_SOC_DAIFMT_DSP_B: + ctrl1_val |= NAU8825_I2S_DF_PCM_AB; + ctrl1_val |= NAU8825_I2S_PCMB_EN; + break; + default: + return -EINVAL; + } + + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL1, + NAU8825_I2S_DL_MASK | NAU8825_I2S_DF_MASK | + NAU8825_I2S_BP_MASK | NAU8825_I2S_PCMB_MASK, + ctrl1_val); + regmap_update_bits(nau8825->regmap, NAU8825_REG_I2S_PCM_CTRL2, + NAU8825_I2S_MS_MASK, ctrl2_val); + + return 0; +} + +static const struct snd_soc_dai_ops nau8825_dai_ops = { + .hw_params = nau8825_hw_params, + .digital_mute = nau8825_mute, + .set_fmt = nau8825_set_dai_fmt, +}; + +#define NAU8825_RATES SNDRV_PCM_RATE_8000_192000 +#define NAU8825_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE \ + | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S32_LE) + +static struct snd_soc_dai_driver nau8825_dai = { + .name = "nau8825-hifi", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = NAU8825_RATES, + .formats = NAU8825_FORMATS, + }, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 1, + .rates = NAU8825_RATES, + .formats = NAU8825_FORMATS, + }, + .ops = &nau8825_dai_ops, +}; + +/** + * nau8825_enable_jack_detect - Specify a jack for event reporting + * + * @component: component to register the jack with + * @jack: jack to use to report headset and button events on + * + * After this function has been called the headset insert/remove and button + * events will be routed to the given jack. Jack can be null to stop + * reporting. + */ +int nau8825_enable_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + + snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); + snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); + snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); + snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); + + nau8825->jack = jack; + + return 0; +} +EXPORT_SYMBOL_GPL(nau8825_enable_jack_detect); + + +static bool nau8825_is_jack_inserted(struct regmap *regmap) +{ + int status; + + regmap_read(regmap, NAU8825_REG_I2C_DEVICE_ID, &status); + return !(status & NAU8825_GPIO2JD1); +} + +static void nau8825_restart_jack_detection(struct regmap *regmap) +{ + /* this will restart the entire jack detection process including MIC/GND + * switching and create interrupts. We have to go from 0 to 1 and back + * to 0 to restart. + */ + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_RESTART, NAU8825_JACK_DET_RESTART); + regmap_update_bits(regmap, NAU8825_REG_JACK_DET_CTRL, + NAU8825_JACK_DET_RESTART, 0); +} + +static void nau8825_eject_jack(struct nau8825 *nau8825) +{ + struct snd_soc_dapm_context *dapm = nau8825->dapm; + struct regmap *regmap = nau8825->regmap; + + snd_soc_dapm_disable_pin(dapm, "SAR"); + snd_soc_dapm_disable_pin(dapm, "MICBIAS"); + /* Detach 2kOhm Resistors from MICBIAS to MICGND1/2 */ + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, 0); + /* ground HPL/HPR, MICGRND1/2 */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0xf, 0xf); + + snd_soc_dapm_sync(dapm); +} + +static int nau8825_button_decode(int value) +{ + int buttons; + + if (value & BIT(0)) + buttons |= SND_JACK_BTN_0; + if (value & BIT(1)) + buttons |= SND_JACK_BTN_1; + if (value & BIT(2)) + buttons |= SND_JACK_BTN_2; + if (value & BIT(3)) + buttons |= SND_JACK_BTN_3; + + return buttons; +} + +static int nau8825_jack_insert(struct nau8825 *nau8825) +{ + struct regmap *regmap = nau8825->regmap; + struct snd_soc_dapm_context *dapm = nau8825->dapm; + int jack_status_reg, mic_detected; + int type; + + regmap_read(regmap, NAU8825_REG_GENERAL_STATUS, &jack_status_reg); + mic_detected = (jack_status_reg >> 10) & 3; + + switch (mic_detected) { + case 0: + /* no mic */ + type = SND_JACK_HEADPHONE; + break; + case 1: + dev_info(nau8825->dev, "OMTP (micgnd1) mic connected\n"); + type = SND_JACK_HEADSET; + + /* Unground MICGND1 */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 3 << 2, + 1 << 2); + /* Unground HPL/R */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0x3, 0); + /* Attach 2kOhm Resistor from MICBIAS to MICGND1 */ + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, + NAU8825_MICBIAS_JKR2); + /* Attach SARADC to MICGND1 */ + regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_INPUT_MASK, + NAU8825_SAR_INPUT_JKR2); + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); + snd_soc_dapm_force_enable_pin(dapm, "SAR"); + snd_soc_dapm_sync(dapm); + break; + case 2: + case 3: + dev_info(nau8825->dev, "CTIA (micgnd2) mic connected\n"); + type = SND_JACK_HEADSET; + + /* Unground MICGND2 */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 3 << 2, + 2 << 2); + /* Unground HPL/R */ + regmap_update_bits(regmap, NAU8825_REG_HSD_CTRL, 0x3, 0); + /* Attach 2kOhm Resistor from MICBIAS to MICGND2 */ + regmap_update_bits(regmap, NAU8825_REG_MIC_BIAS, + NAU8825_MICBIAS_JKSLV | NAU8825_MICBIAS_JKR2, + NAU8825_MICBIAS_JKSLV); + /* Attach SARADC to MICGND2 */ + regmap_update_bits(regmap, NAU8825_REG_SAR_CTRL, + NAU8825_SAR_INPUT_MASK, + NAU8825_SAR_INPUT_JKSLV); + + snd_soc_dapm_force_enable_pin(dapm, "MICBIAS"); + snd_soc_dapm_force_enable_pin(dapm, "SAR"); + snd_soc_dapm_sync(dapm); + break; + } + + return type; +} + +#define NAU8825_BUTTONS (SND_JACK_BTN_0 | SND_JACK_BTN_1 | \ + SND_JACK_BTN_2 | SND_JACK_BTN_3) + +static irqreturn_t nau8825_interrupt(int irq, void *data) +{ + struct nau8825 *nau8825 = (struct nau8825 *)data; + struct regmap *regmap = nau8825->regmap; + int active_irq, clear_irq = 0, event = 0, event_mask = 0; + + regmap_read(regmap, NAU8825_REG_IRQ_STATUS, &active_irq); + + if ((active_irq & NAU8825_JACK_EJECTION_IRQ_MASK) == + NAU8825_JACK_EJECTION_DETECTED) { + + nau8825_eject_jack(nau8825); + event_mask |= SND_JACK_HEADSET; + clear_irq = NAU8825_JACK_EJECTION_IRQ_MASK; + } else if (active_irq & NAU8825_KEY_SHORT_PRESS_IRQ) { + int key_status; + + regmap_read(regmap, NAU8825_REG_INT_CLR_KEY_STATUS, + &key_status); + + /* upper 8 bits of the register are for short pressed keys, + lower 8 bits - for long pressed buttons */ + nau8825->button_pressed = nau8825_button_decode( + key_status >> 8); + + event |= nau8825->button_pressed; + event_mask |= NAU8825_BUTTONS; + clear_irq = NAU8825_KEY_SHORT_PRESS_IRQ; + } else if (active_irq & NAU8825_KEY_RELEASE_IRQ) { + event_mask = NAU8825_BUTTONS; + clear_irq = NAU8825_KEY_RELEASE_IRQ; + } else if (active_irq & NAU8825_HEADSET_COMPLETION_IRQ) { + if (nau8825_is_jack_inserted(regmap)) { + event |= nau8825_jack_insert(nau8825); + } else { + dev_warn(nau8825->dev, "Headset completion IRQ fired but no headset connected\n"); + nau8825_eject_jack(nau8825); + } + + event_mask |= SND_JACK_HEADSET; + clear_irq = NAU8825_HEADSET_COMPLETION_IRQ; + } + + if (!clear_irq) + clear_irq = active_irq; + /* clears the rightmost interruption */ + regmap_write(regmap, NAU8825_REG_INT_CLR_KEY_STATUS, clear_irq); + + if (event_mask) + snd_soc_jack_report(nau8825->jack, event, event_mask); + + return IRQ_HANDLED; +} + +static void nau8825_setup_buttons(struct regmap *regmap) +{ + /* Setup Button Detect (Debounce, number of buttons, and Hysteresis) */ + regmap_write(regmap, NAU8825_REG_KEYDET_CTRL, 0x7311); + + /* Setup 4 buttons impedane according to Android specification + * https://source.android.com/accessories/headset-spec.html + * Button 0 - 0-70 Ohm + * Button 1 - 110-180 Ohm + * Button 2 - 210-290 Ohm + * Button 3 - 360-680 Ohm + */ + regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_1, 0x0f1f); + regmap_write(regmap, NAU8825_REG_VDET_THRESHOLD_2, 0x325f); +} + +static void nau8825_init_regs(struct regmap *regmap) +{ + /* Power down DACs for power savings */ + regmap_write(regmap, NAU8825_REG_CHARGE_PUMP, 0x0300); + /* DAC Zero Crossing Enable */ + regmap_write(regmap, NAU8825_REG_MUTE_CTRL, 0x1000); + /* VMID Enable and Tieoff */ + regmap_write(regmap, NAU8825_REG_BIAS_ADJ, 0x0060); + /* Analog Bias Enable, Disable Boost Driver, + Automatic Short circuit protection enable */ + regmap_write(regmap, NAU8825_REG_BOOST, 0x3140); + /* Ground HP Outputs[1:0], + Enable Automatic Mic/Gnd switching reading on insert interrupt[6] */ + regmap_write(regmap, NAU8825_REG_HSD_CTRL, 0x004f); + + regmap_write(regmap, NAU8825_REG_JACK_DET_CTRL, 0x01e0); + /* IRQ Output Enable, Mask Insert Interrupt */ + regmap_write(regmap, NAU8825_REG_INTERRUPT_MASK, 0x0801); + /* Jack Detect pull up (High=eject, Low=insert) */ + regmap_write(regmap, NAU8825_REG_GPIO12_CTRL, 0x0800); + /* Setup SAR ADC */ + regmap_write(regmap, NAU8825_REG_SAR_CTRL, 0x0280); + nau8825_setup_buttons(regmap); + + /* Setup ADC x128 OSR */ + regmap_write(regmap, NAU8825_REG_ADC_RATE, 0x0002); + /* Setup DAC x128 OSR */ + regmap_write(regmap, NAU8825_REG_DAC_CTRL1, 0x0082); + /* DAC bias settings */ + regmap_write(regmap, NAU8825_REG_ANALOG_CONTROL_2, 0x1003); + /* ADC Bias Settings */ + regmap_write(regmap, NAU8825_REG_ANALOG_ADC_2, 0x0260); + /* Maximize Jack Insert Debounce */ + regmap_write(regmap, NAU8825_REG_JACK_DET_CTRL, 0x00e0); + + /* Setup clocks */ + regmap_write(regmap, NAU8825_REG_ENA_CTRL, 0x0416); + + /* Chip needs one FSCLK cycle in order to generate interrupts, + as we cannot guarantee one will be provided by the system. Turning + master mode on then off enables us to generate that FSCLK cycle + with a minimum of contention on the clock bus. + Enables master mode and correctly divides BCLK and FSCLK from MCLK. + */ + regmap_write(regmap, NAU8825_REG_I2S_PCM_CTRL2, 0x2019); + regmap_write(regmap, NAU8825_REG_I2S_PCM_CTRL2, 0x0010); + + /* Mask jack insert interruption */ + regmap_write(regmap, NAU8825_REG_INTERRUPT_DIS_CTRL, 0x0011); +} + +static const struct regmap_config nau8825_regmap_config = { + .val_bits = 16, + .reg_bits = 16, + + .max_register = NAU8825_REG_MAX, + .readable_reg = nau8825_readable_reg, + .writeable_reg = nau8825_writeable_reg, + .volatile_reg = nau8825_volatile_reg, + + .cache_type = REGCACHE_RBTREE, + .reg_defaults = nau8825_reg_defaults, + .num_reg_defaults = ARRAY_SIZE(nau8825_reg_defaults), +}; + +static int nau8825_codec_probe(struct snd_soc_codec *codec) +{ + struct nau8825 *nau8825 = snd_soc_codec_get_drvdata(codec); + struct device *dev = codec->dev; + + nau8825->dapm = &codec->dapm; + + nau8825_init_regs(nau8825->regmap); + /* The interrupt clock is gated by x1[10:8], + * one of them needs to be enabled all the time for + * interrupts to happen. */ + snd_soc_dapm_force_enable_pin(&codec->dapm, "DDACR"); + snd_soc_dapm_sync(&codec->dapm); + + if (nau8825->irq) { + int ret = devm_request_threaded_irq(dev, nau8825->irq, NULL, + nau8825_interrupt, IRQF_TRIGGER_LOW | IRQF_ONESHOT, + "nau8825", nau8825); + + if (ret) { + dev_err(dev, "Cannot request irq %d (%d)\n", + nau8825->irq, ret); + return ret; + } + } + nau8825_restart_jack_detection(nau8825->regmap); + + return 0; +} + +static struct snd_soc_codec_driver nau8825_codec_driver = { + .probe = nau8825_codec_probe, + + .controls = nau8825_snd_controls, + .num_controls = ARRAY_SIZE(nau8825_snd_controls), + .dapm_widgets = nau8825_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(nau8825_dapm_widgets), + .dapm_routes = nau8825_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(nau8825_dapm_routes), +}; + +static void nau8825_reset_chip(struct regmap *regmap) +{ + regmap_write(regmap, NAU8825_REG_RESET, 0x00); + regmap_write(regmap, NAU8825_REG_RESET, 0x00); +} + +static int nau8825_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct device *dev = &i2c->dev; + struct nau8825 *nau8825; + int ret, value; + + nau8825 = devm_kzalloc(dev, sizeof(*nau8825), GFP_KERNEL); + if (!nau8825) + return -ENOMEM; + + i2c_set_clientdata(i2c, nau8825); + + nau8825->regmap = devm_regmap_init_i2c(i2c, &nau8825_regmap_config); + if (IS_ERR(nau8825->regmap)) + return PTR_ERR(nau8825->regmap); + nau8825->irq = i2c->irq; + nau8825->dev = dev; + + nau8825_reset_chip(nau8825->regmap); + ret = regmap_read(nau8825->regmap, NAU8825_REG_I2C_DEVICE_ID, &value); + if (ret < 0) { + dev_err(dev, "Failed to read device id from the NAU8825: %d\n", + ret); + return ret; + } + if ((value & NAU8825_SOFTWARE_ID_MASK) != + NAU8825_SOFTWARE_ID_NAU8825) { + dev_err(dev, "Not a NAU8825 chip\n"); + return -ENODEV; + } + + return snd_soc_register_codec(&i2c->dev, &nau8825_codec_driver, + &nau8825_dai, 1); +} + +static int nau8825_i2c_remove(struct i2c_client *client) +{ + snd_soc_unregister_codec(&client->dev); + return 0; +} + +static const struct i2c_device_id nau8825_i2c_ids[] = { + { "nau8825", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, nau8825_i2c_ids); + +static struct i2c_driver nau8825_driver = { + .driver = { + .name = "nau8825", + .owner = THIS_MODULE, + }, + .probe = nau8825_i2c_probe, + .remove = nau8825_i2c_remove, + .id_table = nau8825_i2c_ids, +}; +module_i2c_driver(nau8825_driver); + +MODULE_DESCRIPTION("ASoC nau8825 driver"); +MODULE_AUTHOR("Anatol Pomozov <anatol(a)chromium.org>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/nau8825.h b/sound/soc/codecs/nau8825.h new file mode 100644 index 0000000..e1eb769 --- /dev/null +++ b/sound/soc/codecs/nau8825.h @@ -0,0 +1,258 @@ +/* + * NAU8825 ALSA SoC audio driver + * + * Copyright 2015 Google Inc. + * Author: Anatol Pomozov <anatol.pomozov(a)chrominium.org> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __NAU8825_H__ +#define __NAU8825_H__ + +#define NAU8825_REG_RESET 0x00 +#define NAU8825_REG_ENA_CTRL 0x01 +#define NAU8825_REG_CLK_DIVIDER 0x03 +#define NAU8825_REG_FLL1 0x04 +#define NAU8825_REG_FLL2 0x05 +#define NAU8825_REG_FLL3 0x06 +#define NAU8825_REG_FLL4 0x07 +#define NAU8825_REG_FLL5 0x08 +#define NAU8825_REG_FLL6 0x09 +#define NAU8825_REG_FLL_VCO_RSV 0x0a +#define NAU8825_REG_HSD_CTRL 0x0c +#define NAU8825_REG_JACK_DET_CTRL 0x0d +#define NAU8825_REG_INTERRUPT_MASK 0x0f +#define NAU8825_REG_IRQ_STATUS 0x10 +#define NAU8825_REG_INT_CLR_KEY_STATUS 0x11 +#define NAU8825_REG_INTERRUPT_DIS_CTRL 0x12 +#define NAU8825_REG_SAR_CTRL 0x13 +#define NAU8825_REG_KEYDET_CTRL 0x14 +#define NAU8825_REG_VDET_THRESHOLD_1 0x15 +#define NAU8825_REG_VDET_THRESHOLD_2 0x16 +#define NAU8825_REG_VDET_THRESHOLD_3 0x17 +#define NAU8825_REG_VDET_THRESHOLD_4 0x18 +#define NAU8825_REG_GPIO34_CTRL 0x19 +#define NAU8825_REG_GPIO12_CTRL 0x1a +#define NAU8825_REG_TDM_CTRL 0x1b +#define NAU8825_REG_I2S_PCM_CTRL1 0x1c +#define NAU8825_REG_I2S_PCM_CTRL2 0x1d +#define NAU8825_REG_LEFT_TIME_SLOT 0x1e +#define NAU8825_REG_RIGHT_TIME_SLOT 0x1f +#define NAU8825_REG_BIQ_CTRL 0x20 +#define NAU8825_REG_BIQ_COF1 0x21 +#define NAU8825_REG_BIQ_COF2 0x22 +#define NAU8825_REG_BIQ_COF3 0x23 +#define NAU8825_REG_BIQ_COF4 0x24 +#define NAU8825_REG_BIQ_COF5 0x25 +#define NAU8825_REG_BIQ_COF6 0x26 +#define NAU8825_REG_BIQ_COF7 0x27 +#define NAU8825_REG_BIQ_COF8 0x28 +#define NAU8825_REG_BIQ_COF9 0x29 +#define NAU8825_REG_BIQ_COF10 0x2a +#define NAU8825_REG_ADC_RATE 0x2b +#define NAU8825_REG_DAC_CTRL1 0x2c +#define NAU8825_REG_DAC_CTRL2 0x2d +#define NAU8825_REG_DAC_DGAIN_CTRL 0x2f +#define NAU8825_REG_ADC_DGAIN_CTRL 0x30 +#define NAU8825_REG_MUTE_CTRL 0x31 +#define NAU8825_REG_HSVOL_CTRL 0x32 +#define NAU8825_REG_DACL_CTRL 0x33 +#define NAU8825_REG_DACR_CTRL 0x34 +#define NAU8825_REG_ADC_DRC_KNEE_IP12 0x38 +#define NAU8825_REG_ADC_DRC_KNEE_IP34 0x39 +#define NAU8825_REG_ADC_DRC_SLOPES 0x3a +#define NAU8825_REG_ADC_DRC_ATKDCY 0x3b +#define NAU8825_REG_DAC_DRC_KNEE_IP12 0x45 +#define NAU8825_REG_DAC_DRC_KNEE_IP34 0x46 +#define NAU8825_REG_DAC_DRC_SLOPES 0x47 +#define NAU8825_REG_DAC_DRC_ATKDCY 0x48 +#define NAU8825_REG_IMM_MODE_CTRL 0x4c +#define NAU8825_REG_IMM_RMS_L 0x4d +#define NAU8825_REG_IMM_RMS_R 0x4e +#define NAU8825_REG_CLASSG_CTRL 0x50 +#define NAU8825_REG_OPT_EFUSE_CTRL 0x51 +#define NAU8825_REG_MISC_CTRL 0x55 +#define NAU8825_REG_I2C_DEVICE_ID 0x58 +#define NAU8825_REG_SARDOUT_RAM_STATUS 0x59 +#define NAU8825_REG_BIAS_ADJ 0x66 +#define NAU8825_REG_TRIM_SETTINGS 0x68 +#define NAU8825_REG_ANALOG_CONTROL_1 0x69 +#define NAU8825_REG_ANALOG_CONTROL_2 0x6a +#define NAU8825_REG_ANALOG_ADC_1 0x71 +#define NAU8825_REG_ANALOG_ADC_2 0x72 +#define NAU8825_REG_RDAC 0x73 +#define NAU8825_REG_MIC_BIAS 0x74 +#define NAU8825_REG_BOOST 0x76 +#define NAU8825_REG_FEPGA 0x77 +#define NAU8825_REG_POWER_UP_CONTROL 0x7f +#define NAU8825_REG_CHARGE_PUMP 0x80 +#define NAU8825_REG_CHARGE_PUMP_INPUT_READ 0x81 +#define NAU8825_REG_GENERAL_STATUS 0x82 +#define NAU8825_REG_MAX NAU8825_REG_GENERAL_STATUS + +/* ENA_CTRL (0x1) */ +#define NAU8825_ENABLE_DACR_SFT 10 +#define NAU8825_ENABLE_DACL_SFT 9 +#define NAU8825_ENABLE_ADC_SFT 8 +#define NAU8825_ENABLE_SAR_SFT 1 + +/* CLK_DIVIDER (0x3) */ +#define NAU8825_CLK_SRC_SFT 15 +#define NAU8825_CLK_SRC_MASK (1 << NAU8825_CLK_SRC_SFT) +#define NAU8825_CLK_SRC_VCO (1 << NAU8825_CLK_SRC_SFT) +#define NAU8825_CLK_SRC_MCLK (0 << NAU8825_CLK_SRC_SFT) + +/* HSD_CTRL (0xc) */ +#define NAU8825_HSD_AUTO_MODE (1 << 6) +#define NAU8825_SPKR_DWN1R_SHORT (1 << 1) +#define NAU8825_SPKR_DWN1L_SHORT (1 << 0) + +/* JACK_DET_CTRL (0xd) */ +#define NAU8825_JACK_DET_RESTART (1 << 9) + +/* IRQ_STATUS (0x10) */ +#define NAU8825_HEADSET_COMPLETION_IRQ (1 << 10) +#define NAU8825_SHORT_CIRCUIT_IRQ (1 << 9) +#define NAU8825_IMPEDANCE_MEAS_IRQ (1 << 8) +#define NAU8825_KEY_IRQ_MASK (0x7 << 5) +#define NAU8825_KEY_RELEASE_IRQ (1 << 7) +#define NAU8825_KEY_LONG_PRESS_IRQ (1 << 6) +#define NAU8825_KEY_SHORT_PRESS_IRQ (1 << 5) +#define NAU8825_MIC_DETECTION_IRQ (1 << 4) +#define NAU8825_JACK_EJECTION_IRQ_MASK (3 << 2) +#define NAU8825_JACK_EJECTION_DETECTED (1 << 2) +#define NAU8825_JACK_INSERTION_IRQ_MASK (3 << 0) +#define NAU8825_JACK_INSERTION_DETECTED (1 << 0) + +/* SAR_CTRL (0x13) */ +#define NAU8825_SAR_ADC_EN_SFT 12 +#define NAU8825_SAR_ADC_EN (1 << NAU8825_SAR_ADC_EN_SFT) +#define NAU8825_SAR_INPUT_MASK (1 << 11) +#define NAU8825_SAR_INPUT_JKSLV (1 << 11) +#define NAU8825_SAR_INPUT_JKR2 (0 << 11) + +/* KEYDET_CTRL (0x14) */ +#define NAU8825_KEYDET_LEVELS_NR_SFT 8 +#define NAU8825_KEYDET_LEVELS_NR_MASK (0x7 << 8) + +/* I2S_PCM_CTRL1 (0x1c) */ +#define NAU8825_I2S_BP_SFT 7 +#define NAU8825_I2S_BP_MASK (1 << NAU8825_I2S_BP_SFT) +#define NAU8825_I2S_BP_INV (1 << NAU8825_I2S_BP_SFT) +#define NAU8825_I2S_PCMB_SFT 6 +#define NAU8825_I2S_PCMB_MASK (1 << NAU8825_I2S_PCMB_SFT) +#define NAU8825_I2S_PCMB_EN (1 << NAU8825_I2S_PCMB_SFT) +#define NAU8825_I2S_DL_SFT 2 +#define NAU8825_I2S_DL_MASK (0x3 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_16 (0 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_20 (1 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_24 (2 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DL_32 (3 << NAU8825_I2S_DL_SFT) +#define NAU8825_I2S_DF_SFT 0 +#define NAU8825_I2S_DF_MASK (0x3 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_RIGTH (0 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_LEFT (1 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_I2S (2 << NAU8825_I2S_DF_SFT) +#define NAU8825_I2S_DF_PCM_AB (3 << NAU8825_I2S_DF_SFT) + +/* I2S_PCM_CTRL2 (0x1d) */ +#define NAU8825_I2S_MS_SFT 3 +#define NAU8825_I2S_MS_MASK (1 << NAU8825_I2S_MS_SFT) +#define NAU8825_I2S_MS_MASTER (1 << NAU8825_I2S_MS_SFT) +#define NAU8825_I2S_MS_SLAVE (0 << NAU8825_I2S_MS_SFT) + +/* ADC_RATE (0x2b) */ +#define NAU8825_ADC_SINC4 (1 << 4) +#define NAU8825_ADC_SINC_DOWN_MASK 3 +#define NAU8825_ADC_SINC_DOWN_32 0 +#define NAU8825_ADC_SINC_DOWN_64 1 +#define NAU8825_ADC_SINC_DOWN_128 2 +#define NAU8825_ADC_SINC_DOWN_256 3 + +/* DAC_CTRL1 (0x2c) */ +#define NAU8825_DAC_CLIP_OFF (1 << 7) +#define NAU8825_DAC_OVERSAMPLE_MASK 7 +#define NAU8825_DAC_OVERSAMPLE_64 0 +#define NAU8825_DAC_OVERSAMPLE_256 1 +#define NAU8825_DAC_OVERSAMPLE_128 2 +#define NAU8825_DAC_OVERSAMPLE_32 4 + +/* MUTE_CTRL (0x31) */ +#define NAU8825_DAC_ZERO_CROSSING_EN (1 << 9) +#define NAU8825_DAC_SOFT_MUTE (1 << 9) + +/* HSVOL_CTRL (0x32) */ +#define NAU8825_HP_MUTE (1 << 15) + +/* DACL_CTRL (0x33) */ +#define NAU8825_DACL_CH_SEL_SFT 9 + +/* DACR_CTRL (0x34) */ +#define NAU8825_DACR_CH_SEL_SFT 9 + +/* I2C_DEVICE_ID (0x58) */ +#define NAU8825_GPIO2JD1 (1 << 7) +#define NAU8825_SOFTWARE_ID_MASK 0x3 +#define NAU8825_SOFTWARE_ID_NAU8825 0x0 + +/* BIAS_ADJ (0x66) */ +#define NAU8825_BIAS_VMID (1 << 6) +#define NAU8825_BIAS_VMID_SEL_MASK (3 << 4) +#define NAU8825_BIAS_VMID_SEL_OPEN (0 << 4) +#define NAU8825_BIAS_VMID_SEL_25KOHM (1 << 4) +#define NAU8825_BIAS_VMID_SEL_125KOHM (2 << 4) +#define NAU8825_BIAS_VMID_SEL_2_5KOHM (3 << 4) + +/* ANALOG_CONTROL_2 (0x6a) */ +#define NAU8825_HP_NON_CLASSG_CURRENT_2xADJ (1 << 12) +#define NAU8825_DAC_CAPACITOR_MSB (1 << 1) +#define NAU8825_DAC_CAPACITOR_LSB (1 << 0) + +/* ANALOG_ADC_2 (0x72) */ +#define NAU8825_ADC_VREFSEL_MASK (0x3 << 8) +#define NAU8825_ADC_VREFSEL_ANALOG (0 << 8) +#define NAU8825_ADC_VREFSEL_VMID (1 << 8) +#define NAU8825_ADC_VREFSEL_VMID_PLUS_0_5DB (2 << 8) +#define NAU8825_ADC_VREFSEL_VMID_PLUS_1DB (3 << 8) +#define NAU8825_POWERUP_ADCL (1 << 6) + +/* MIC_BIAS (0x74) */ +#define NAU8825_MICBIAS_JKSLV (1 << 14) +#define NAU8825_MICBIAS_JKR2 (1 << 12) +#define NAU8825_MICBIAS_POWERUP_SFT 8 + +/* BOOST (0x76) */ +#define NAU8825_PRECHARGE_DIS (1 << 13) +#define NAU8825_GLOBAL_BIAS_EN (1 << 12) +#define NAU8825_HP_BOOST_DIS (1 << 12) +#define NAU8825_SHORT_SHUTDOWN_EN (1 << 6) + +/* POWER_UP_CONTROL (0x7f) */ +#define NAU8825_POWERUP_HP_DRV_L (1 << 0) +#define NAU8825_POWERUP_HP_DRV_R (1 << 1) +#define NAU8825_POWERUP_DRV_IN_L (1 << 2) +#define NAU8825_POWERUP_DRV_IN_R (1 << 3) +#define NAU8825_POWERUP_INTEGR_L (1 << 4) +#define NAU8825_POWERUP_INTEGR_R (1 << 5) + +/* CHARGE_PUMP_AND_POWER_DOWN_CONTROL (0x80) */ +#define NAU8825_JAMNODCLOW (1 << 10) +#define NAU8825_POWER_DOWN_DACR (1 << 9) +#define NAU8825_POWER_DOWN_DACL (1 << 8) +#define NAU8825_CHANRGE_PUMP_EN (1 << 5) + + +/* System Clock Source */ +enum { + NAU8825_CLK_MCLK = 0, + NAU8825_CLK_INTERNAL, +}; + +int nau8825_enable_jack_detect(struct snd_soc_codec *codec, + struct snd_soc_jack *jack); + + +#endif /* __NAU8825_H__ */ -- 2.5.0.rc2.392.g76e840b

2 12

[alsa-devel] [PATCH] ASoC: qcom: Constify asoc_qcom_lpass_cpu_dai_ops
by Axel Lin 28 Aug '15

28 Aug '15

asoc_qcom_lpass_cpu_dai_ops is exported and used by multiple drivers, make it const to prevent modifying it at run time. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/qcom/lpass-cpu.c | 2 +- sound/soc/qcom/lpass.h | 2 +- 2 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/qcom/lpass-cpu.c b/sound/soc/qcom/lpass-cpu.c index 23f3d59..97bc202 100644 --- a/sound/soc/qcom/lpass-cpu.c +++ b/sound/soc/qcom/lpass-cpu.c @@ -235,7 +235,7 @@ static int lpass_cpu_daiops_trigger(struct snd_pcm_substream *substream, return ret; } -struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { +const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops = { .set_sysclk = lpass_cpu_daiops_set_sysclk, .startup = lpass_cpu_daiops_startup, .shutdown = lpass_cpu_daiops_shutdown, diff --git a/sound/soc/qcom/lpass.h b/sound/soc/qcom/lpass.h index d6e86c1..0b63e2e 100644 --- a/sound/soc/qcom/lpass.h +++ b/sound/soc/qcom/lpass.h @@ -93,6 +93,6 @@ int asoc_qcom_lpass_platform_register(struct platform_device *); int asoc_qcom_lpass_cpu_platform_remove(struct platform_device *pdev); int asoc_qcom_lpass_cpu_platform_probe(struct platform_device *pdev); int asoc_qcom_lpass_cpu_dai_probe(struct snd_soc_dai *dai); -extern struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops; +extern const struct snd_soc_dai_ops asoc_qcom_lpass_cpu_dai_ops; #endif /* __LPASS_H__ */ -- 2.1.0

2 1

[alsa-devel] Applied "sound: Clean up docbook warnings" to the asoc tree
by Mark Brown 28 Aug '15

28 Aug '15

The patch sound: Clean up docbook warnings has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 55b1faa33d485ace99a5a9e58bb1007782ae8ec9 Mon Sep 17 00:00:00 2001 From: Jonathan Corbet <corbet(a)lwn.net> Date: Tue, 25 Aug 2015 01:14:48 -0600 Subject: [PATCH] sound: Clean up docbook warnings A number of functions and structures in the sound subsystem had incomplete and/or obsolete DocBook comments, leading to warnings when the docs were built. Correct those comments so that we can enjoy our audio in the absence of warning noise. Signed-off-by: Jonathan Corbet <corbet(a)lwn.net> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/DocBook/alsa-driver-api.tmpl | 2 +- include/sound/soc.h | 11 +++++++---- sound/soc/soc-core.c | 13 +++++++++---- sound/soc/soc-dapm.c | 2 +- 4 files changed, 18 insertions(+), 10 deletions(-) diff --git a/Documentation/DocBook/alsa-driver-api.tmpl b/Documentation/DocBook/alsa-driver-api.tmpl index 71f9246..e94a10b 100644 --- a/Documentation/DocBook/alsa-driver-api.tmpl +++ b/Documentation/DocBook/alsa-driver-api.tmpl @@ -108,7 +108,7 @@ <sect1><title>ASoC Core API</title> !Iinclude/sound/soc.h !Esound/soc/soc-core.c -!Esound/soc/soc-cache.c + !Esound/soc/soc-devres.c !Esound/soc/soc-io.c !Esound/soc/soc-pcm.c diff --git a/include/sound/soc.h b/include/sound/soc.h index 93df8bf..4537e81 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -619,6 +619,7 @@ int snd_soc_put_strobe(struct snd_kcontrol *kcontrol, * @pin: name of the pin to update * @mask: bits to check for in reported jack status * @invert: if non-zero then pin is enabled when status is not reported + * @list: internal list entry */ struct snd_soc_jack_pin { struct list_head list; @@ -635,7 +636,7 @@ struct snd_soc_jack_pin { * @jack_type: type of jack that is expected for this voltage * @debounce_time: debounce_time for jack, codec driver should wait for this * duration before reading the adc for voltages - * @:list: list container + * @list: internal list entry */ struct snd_soc_jack_zone { unsigned int min_mv; @@ -651,12 +652,12 @@ struct snd_soc_jack_zone { * @gpio: legacy gpio number * @idx: gpio descriptor index within the function of the GPIO * consumer device - * @gpiod_dev GPIO consumer device + * @gpiod_dev: GPIO consumer device * @name: gpio name. Also as connection ID for the GPIO consumer * device function name lookup * @report: value to report when jack detected * @invert: report presence in low state - * @debouce_time: debouce time in ms + * @debounce_time: debounce time in ms * @wake: enable as wake source * @jack_status_check: callback function which overrides the detection * to provide more complex checks (eg, reading an @@ -672,11 +673,13 @@ struct snd_soc_jack_gpio { int debounce_time; bool wake; + /* private: */ struct snd_soc_jack *jack; struct delayed_work work; struct gpio_desc *desc; void *data; + /* public: */ int (*jack_status_check)(void *data); }; @@ -1319,7 +1322,7 @@ static inline struct snd_soc_dapm_context *snd_soc_codec_get_dapm( /** * snd_soc_dapm_init_bias_level() - Initialize CODEC DAPM bias level - * @dapm: The CODEC for which to initialize the DAPM bias level + * @codec: The CODEC for which to initialize the DAPM bias level * @level: The DAPM level to initialize to * * Initializes the CODEC DAPM bias level. See snd_soc_dapm_init_bias_level(). diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 90d6335..3224251 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2798,6 +2798,7 @@ EXPORT_SYMBOL_GPL(snd_soc_register_component); /** * snd_soc_unregister_component - Unregister a component from the ASoC core * + * @dev: The device to unregister */ void snd_soc_unregister_component(struct device *dev) { @@ -2877,7 +2878,8 @@ EXPORT_SYMBOL_GPL(snd_soc_add_platform); /** * snd_soc_register_platform - Register a platform with the ASoC core * - * @platform: platform to register + * @dev: The device for the platform + * @platform_drv: The driver for the platform */ int snd_soc_register_platform(struct device *dev, const struct snd_soc_platform_driver *platform_drv) @@ -2938,7 +2940,7 @@ EXPORT_SYMBOL_GPL(snd_soc_lookup_platform); /** * snd_soc_unregister_platform - Unregister a platform from the ASoC core * - * @platform: platform to unregister + * @dev: platform to unregister */ void snd_soc_unregister_platform(struct device *dev) { @@ -3029,7 +3031,10 @@ static int snd_soc_codec_set_bias_level(struct snd_soc_dapm_context *dapm, /** * snd_soc_register_codec - Register a codec with the ASoC core * - * @codec: codec to register + * @dev: The parent device for this codec + * @codec_drv: Codec driver + * @dai_drv: The associated DAI driver + * @num_dai: Number of DAIs */ int snd_soc_register_codec(struct device *dev, const struct snd_soc_codec_driver *codec_drv, @@ -3128,7 +3133,7 @@ EXPORT_SYMBOL_GPL(snd_soc_register_codec); /** * snd_soc_unregister_codec - Unregister a codec from the ASoC core * - * @codec: codec to unregister + * @dev: codec to unregister */ void snd_soc_unregister_codec(struct device *dev) { diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index aa327c9..c4e3720 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2911,7 +2911,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_weak_routes); /** * snd_soc_dapm_new_widgets - add new dapm widgets - * @dapm: DAPM context + * @card: card to be checked for new dapm widgets * * Checks the codec for any new dapm widgets and creates them if found. * -- 2.5.0

1 0

[alsa-devel] [PATCH] sound: Clean up docbook warnings
by Jonathan Corbet 28 Aug '15

28 Aug '15

A number of functions and structures in the sound subsystem had incomplete and/or obsolete DocBook comments, leading to warnings when the docs were built. Correct those comments so that we can enjoy our audio in the absence of warning noise. Signed-off-by: Jonathan Corbet <corbet(a)lwn.net> --- NOTE that it would be good if somebody more knowledgeable than me checked my work, since I had to fill in a couple of things. Documentation/DocBook/alsa-driver-api.tmpl | 2 +- include/sound/soc.h | 11 +++++++---- sound/soc/soc-core.c | 17 +++++++++++------ sound/soc/soc-dapm.c | 2 +- 4 files changed, 20 insertions(+), 12 deletions(-) diff --git a/Documentation/DocBook/alsa-driver-api.tmpl b/Documentation/DocBook/alsa-driver-api.tmpl index 71f9246127ec..e94a10bb4a9e 100644 --- a/Documentation/DocBook/alsa-driver-api.tmpl +++ b/Documentation/DocBook/alsa-driver-api.tmpl @@ -108,7 +108,7 @@ <sect1><title>ASoC Core API</title> !Iinclude/sound/soc.h !Esound/soc/soc-core.c -!Esound/soc/soc-cache.c + !Esound/soc/soc-devres.c !Esound/soc/soc-io.c !Esound/soc/soc-pcm.c diff --git a/include/sound/soc.h b/include/sound/soc.h index 93df8bf9d54a..4537e81eeeda 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -619,6 +619,7 @@ int snd_soc_put_strobe(struct snd_kcontrol *kcontrol, * @pin: name of the pin to update * @mask: bits to check for in reported jack status * @invert: if non-zero then pin is enabled when status is not reported + * @list: internal list entry */ struct snd_soc_jack_pin { struct list_head list; @@ -635,7 +636,7 @@ struct snd_soc_jack_pin { * @jack_type: type of jack that is expected for this voltage * @debounce_time: debounce_time for jack, codec driver should wait for this * duration before reading the adc for voltages - * @:list: list container + * @list: internal list entry */ struct snd_soc_jack_zone { unsigned int min_mv; @@ -651,12 +652,12 @@ struct snd_soc_jack_zone { * @gpio: legacy gpio number * @idx: gpio descriptor index within the function of the GPIO * consumer device - * @gpiod_dev GPIO consumer device + * @gpiod_dev: GPIO consumer device * @name: gpio name. Also as connection ID for the GPIO consumer * device function name lookup * @report: value to report when jack detected * @invert: report presence in low state - * @debouce_time: debouce time in ms + * @debounce_time: debounce time in ms * @wake: enable as wake source * @jack_status_check: callback function which overrides the detection * to provide more complex checks (eg, reading an @@ -672,11 +673,13 @@ struct snd_soc_jack_gpio { int debounce_time; bool wake; + /* private: */ struct snd_soc_jack *jack; struct delayed_work work; struct gpio_desc *desc; void *data; + /* public: */ int (*jack_status_check)(void *data); }; @@ -1319,7 +1322,7 @@ static inline struct snd_soc_dapm_context *snd_soc_codec_get_dapm( /** * snd_soc_dapm_init_bias_level() - Initialize CODEC DAPM bias level - * @dapm: The CODEC for which to initialize the DAPM bias level + * @codec: The CODEC for which to initialize the DAPM bias level * @level: The DAPM level to initialize to * * Initializes the CODEC DAPM bias level. See snd_soc_dapm_init_bias_level(). diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 3a4a5c0e3f97..32242512d828 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2127,7 +2127,7 @@ EXPORT_SYMBOL_GPL(snd_soc_codec_set_pll); /** * snd_soc_dai_set_bclk_ratio - configure BCLK to sample rate ratio. * @dai: DAI - * @ratio Ratio of BCLK to Sample rate. + * @ratio: Ratio of BCLK to Sample rate. * * Configures the DAI for a preset BCLK to sample rate ratio. */ @@ -2798,6 +2798,7 @@ EXPORT_SYMBOL_GPL(snd_soc_register_component); /** * snd_soc_unregister_component - Unregister a component from the ASoC core * + * @dev: The device to unregister */ void snd_soc_unregister_component(struct device *dev) { @@ -2838,7 +2839,7 @@ static void snd_soc_platform_drv_remove(struct snd_soc_component *component) * snd_soc_add_platform - Add a platform to the ASoC core * @dev: The parent device for the platform * @platform: The platform to add - * @platform_driver: The driver for the platform + * @platform_drv: The driver for the platform */ int snd_soc_add_platform(struct device *dev, struct snd_soc_platform *platform, const struct snd_soc_platform_driver *platform_drv) @@ -2877,7 +2878,8 @@ EXPORT_SYMBOL_GPL(snd_soc_add_platform); /** * snd_soc_register_platform - Register a platform with the ASoC core * - * @platform: platform to register + * @dev: The device for the platform + * @platform_drv: The driver for the platform */ int snd_soc_register_platform(struct device *dev, const struct snd_soc_platform_driver *platform_drv) @@ -2938,7 +2940,7 @@ EXPORT_SYMBOL_GPL(snd_soc_lookup_platform); /** * snd_soc_unregister_platform - Unregister a platform from the ASoC core * - * @platform: platform to unregister + * @dev: platform to unregister */ void snd_soc_unregister_platform(struct device *dev) { @@ -3029,7 +3031,10 @@ static int snd_soc_codec_set_bias_level(struct snd_soc_dapm_context *dapm, /** * snd_soc_register_codec - Register a codec with the ASoC core * - * @codec: codec to register + * @dev: The parent device for this codec + * @codec_drv: Codec driver + * @dai_drv: The associated DAI driver + * @num_dai: Number of DAIs */ int snd_soc_register_codec(struct device *dev, const struct snd_soc_codec_driver *codec_drv, @@ -3128,7 +3133,7 @@ EXPORT_SYMBOL_GPL(snd_soc_register_codec); /** * snd_soc_unregister_codec - Unregister a codec from the ASoC core * - * @codec: codec to unregister + * @dev: codec to unregister */ void snd_soc_unregister_codec(struct device *dev) { diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index aa327c92480c..c4e3720bea41 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -2911,7 +2911,7 @@ EXPORT_SYMBOL_GPL(snd_soc_dapm_weak_routes); /** * snd_soc_dapm_new_widgets - add new dapm widgets - * @dapm: DAPM context + * @card: card to be checked for new dapm widgets * * Checks the codec for any new dapm widgets and creates them if found. * -- 2.4.3

3 8

[alsa-devel] Applied "ASoC: ics43432: fix platform_no_drv_owner.cocci warnings" to the asoc tree
by Mark Brown 28 Aug '15

28 Aug '15

The patch ASoC: ics43432: fix platform_no_drv_owner.cocci warnings has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 11d932416ecada27ca7576dca7e74e23477df7ef Mon Sep 17 00:00:00 2001 From: kbuild test robot <fengguang.wu(a)intel.com> Date: Fri, 28 Aug 2015 10:04:03 +0800 Subject: [PATCH] ASoC: ics43432: fix platform_no_drv_owner.cocci warnings sound/soc/codecs/ics43432.c:66:3-8: No need to set .owner here. The core will do it. Remove .owner field if calls are used which set it automatically Generated by: scripts/coccinelle/api/platform_no_drv_owner.cocci CC: Ricard Wanderlof <ricard.wanderlof(a)axis.com> Signed-off-by: Fengguang Wu <fengguang.wu(a)intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/ics43432.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/ics43432.c b/sound/soc/codecs/ics43432.c index 4f202c1..d217fed 100644 --- a/sound/soc/codecs/ics43432.c +++ b/sound/soc/codecs/ics43432.c @@ -63,7 +63,6 @@ MODULE_DEVICE_TABLE(of, ics43432_dt_ids); static struct platform_driver ics43432_driver = { .driver = { .name = "ics43432", - .owner = THIS_MODULE, .of_match_table = of_match_ptr(ics43432_ids), }, .probe = ics43432_probe, -- 2.5.0

1 0

[alsa-devel] [PATCH] ASoC: ics43432: Fix module license
by Axel Lin 28 Aug '15

28 Aug '15

Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- sound/soc/codecs/ics43432.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/ics43432.c b/sound/soc/codecs/ics43432.c index 4f202c1..382a004 100644 --- a/sound/soc/codecs/ics43432.c +++ b/sound/soc/codecs/ics43432.c @@ -6,7 +6,7 @@ * * Copyright (c) 2015 Axis Communications AB * - * Licensed under GPL2. + * Licensed under GPL v2. */ #include <linux/module.h> @@ -74,4 +74,4 @@ module_platform_driver(ics43432_driver); MODULE_DESCRIPTION("ASoC ICS43432 driver"); MODULE_AUTHOR("Ricard Wanderlof <ricardw(a)axis.com>"); -MODULE_LICENSE("GPLv2"); +MODULE_LICENSE("GPL v2"); -- 2.1.0

2 1

[alsa-devel] Applied "ASoC: ics43432: Fix build errors" to the asoc tree
by Mark Brown 28 Aug '15

28 Aug '15

The patch ASoC: ics43432: Fix build errors has been applied to the asoc tree at git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From b37bfdaac914f4886b8be803703b12177eb873f6 Mon Sep 17 00:00:00 2001 From: Ricard Wanderlof <ricard.wanderlof(a)axis.com> Date: Fri, 28 Aug 2015 12:11:48 +0200 Subject: [PATCH] ASoC: ics43432: Fix build errors Fix build errors Signed-off-by: Ricard Wanderlof <ricardw(a)axis.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/ics43432.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/ics43432.c b/sound/soc/codecs/ics43432.c index 7c35c98..dd850b9 100644 --- a/sound/soc/codecs/ics43432.c +++ b/sound/soc/codecs/ics43432.c @@ -57,7 +57,7 @@ static const struct of_device_id ics43432_ids[] = { { .compatible = "invensense,ics43432", }, { } }; -MODULE_DEVICE_TABLE(of, ics43432_dt_ids); +MODULE_DEVICE_TABLE(of, ics43432_ids); #endif static struct platform_driver ics43432_driver = { -- 2.5.0

1 0

[alsa-devel] [PATCH v3] OMAPDSS: hdmi: Reconfigure and restart audio when display is enabled
by Jyri Sarha 28 Aug '15

28 Aug '15

Reconfigure and restart audio when display is enabled, if audio playback was active before. This is needed in a situation when an audio+video stream application opens the audio stream before the video. When video stream is opened the display mode may change and that would aborted audio playback, because the display is momentarily turned off. The audio configuration is stored when it is successfully applied and a boolean is set when the audio playback is started and unset when stopped. This data is used to reconfigure the audio when display is re-enabled. The audio playback is aborted if the reconfiguration fails. A new spin lock is introduced in order to protect state variables related to audio playback status. This is needed for the transition from display enabled state (when audio start/stop commands can be written to HW) to display disabled state (when audio start/stop commands update only the hdmi.audio_playing variable) to always serialize correctly with the start/stop audio commands. The already existing mutex can not be used, because the audio start and stop commands are executed in atomic context. For example: when display is turned back on we take the spinlock and we can be sure that the audio start/stop status will not change while we update the HW according to hdmi.audio_playing state and set hdmi.display_enabled to true. After releasing the lock hdmi.display_enabled is true and all audio_start and audio_stop commands write their stuff directly to HW. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- drivers/video/fbdev/omap2/dss/hdmi.h | 9 ++++- drivers/video/fbdev/omap2/dss/hdmi4.c | 66 +++++++++++++++++++++++++----- drivers/video/fbdev/omap2/dss/hdmi5.c | 76 ++++++++++++++++++++++++++++------- 3 files changed, 124 insertions(+), 27 deletions(-) diff --git a/drivers/video/fbdev/omap2/dss/hdmi.h b/drivers/video/fbdev/omap2/dss/hdmi.h index e4a32fe..53616b0 100644 --- a/drivers/video/fbdev/omap2/dss/hdmi.h +++ b/drivers/video/fbdev/omap2/dss/hdmi.h @@ -351,13 +351,20 @@ struct omap_hdmi { struct regulator *vdda_reg; bool core_enabled; - bool display_enabled; struct omap_dss_device output; struct platform_device *audio_pdev; void (*audio_abort_cb)(struct device *dev); int wp_idlemode; + + bool audio_configured; + struct omap_dss_audio audio_config; + + /* This lock should be taken when booleans bellow are touched. */ + spinlock_t audio_playing_lock; + bool audio_playing; + bool display_enabled; }; #endif diff --git a/drivers/video/fbdev/omap2/dss/hdmi4.c b/drivers/video/fbdev/omap2/dss/hdmi4.c index 6d3aa3f..94c8d55 100644 --- a/drivers/video/fbdev/omap2/dss/hdmi4.c +++ b/drivers/video/fbdev/omap2/dss/hdmi4.c @@ -321,9 +321,22 @@ static int read_edid(u8 *buf, int len) return r; } +static void hdmi_start_audio_stream(struct omap_hdmi *hd) +{ + hdmi_wp_audio_enable(&hd->wp, true); + hdmi4_audio_start(&hd->core, &hd->wp); +} + +static void hdmi_stop_audio_stream(struct omap_hdmi *hd) +{ + hdmi4_audio_stop(&hd->core, &hd->wp); + hdmi_wp_audio_enable(&hd->wp, false); +} + static int hdmi_display_enable(struct omap_dss_device *dssdev) { struct omap_dss_device *out = &hdmi.output; + unsigned long flags; int r = 0; DSSDBG("ENTER hdmi_display_enable\n"); @@ -342,7 +355,21 @@ static int hdmi_display_enable(struct omap_dss_device *dssdev) goto err0; } + if (hdmi.audio_configured) { + r = hdmi4_audio_config(&hdmi.core, &hdmi.wp, &hdmi.audio_config, + hdmi.cfg.timings.pixelclock); + if (r) { + DSSERR("Error restoring audio configuration: %d", r); + hdmi.audio_abort_cb(&hdmi.pdev->dev); + hdmi.audio_configured = false; + } + } + + spin_lock_irqsave(&hdmi.audio_playing_lock, flags); + if (hdmi.audio_configured && hdmi.audio_playing) + hdmi_start_audio_stream(&hdmi); hdmi.display_enabled = true; + spin_unlock_irqrestore(&hdmi.audio_playing_lock, flags); mutex_unlock(&hdmi.lock); return 0; @@ -354,17 +381,19 @@ err0: static void hdmi_display_disable(struct omap_dss_device *dssdev) { + unsigned long flags; + DSSDBG("Enter hdmi_display_disable\n"); mutex_lock(&hdmi.lock); - if (hdmi.audio_pdev && hdmi.audio_abort_cb) - hdmi.audio_abort_cb(&hdmi.audio_pdev->dev); + spin_lock_irqsave(&hdmi.audio_playing_lock, flags); + hdmi_stop_audio_stream(&hdmi); + hdmi.display_enabled = false; + spin_unlock_irqrestore(&hdmi.audio_playing_lock, flags); hdmi_power_off_full(dssdev); - hdmi.display_enabled = false; - mutex_unlock(&hdmi.lock); } @@ -568,6 +597,8 @@ static int hdmi_audio_shutdown(struct device *dev) mutex_lock(&hd->lock); hd->audio_abort_cb = NULL; + hd->audio_configured = false; + hd->audio_playing = false; mutex_unlock(&hd->lock); return 0; @@ -576,25 +607,34 @@ static int hdmi_audio_shutdown(struct device *dev) static int hdmi_audio_start(struct device *dev) { struct omap_hdmi *hd = dev_get_drvdata(dev); + unsigned long flags; WARN_ON(!hdmi_mode_has_audio(&hd->cfg)); - WARN_ON(!hd->display_enabled); - hdmi_wp_audio_enable(&hd->wp, true); - hdmi4_audio_start(&hd->core, &hd->wp); + spin_lock_irqsave(&hd->audio_playing_lock, flags); + + if (hd->display_enabled) + hdmi_start_audio_stream(hd); + hd->audio_playing = true; + spin_unlock_irqrestore(&hd->audio_playing_lock, flags); return 0; } static void hdmi_audio_stop(struct device *dev) { struct omap_hdmi *hd = dev_get_drvdata(dev); + unsigned long flags; WARN_ON(!hdmi_mode_has_audio(&hd->cfg)); - WARN_ON(!hd->display_enabled); - hdmi4_audio_stop(&hd->core, &hd->wp); - hdmi_wp_audio_enable(&hd->wp, false); + spin_lock_irqsave(&hd->audio_playing_lock, flags); + + if (hd->display_enabled) + hdmi_stop_audio_stream(hd); + hd->audio_playing = false; + + spin_unlock_irqrestore(&hd->audio_playing_lock, flags); } static int hdmi_audio_config(struct device *dev, @@ -612,7 +652,10 @@ static int hdmi_audio_config(struct device *dev, ret = hdmi4_audio_config(&hd->core, &hd->wp, dss_audio, hd->cfg.timings.pixelclock); - + if (!ret) { + hd->audio_configured = true; + hd->audio_config = *dss_audio; + } out: mutex_unlock(&hd->lock); @@ -657,6 +700,7 @@ static int hdmi4_bind(struct device *dev, struct device *master, void *data) dev_set_drvdata(&pdev->dev, &hdmi); mutex_init(&hdmi.lock); + spin_lock_init(&hdmi.audio_playing_lock); if (pdev->dev.of_node) { r = hdmi_probe_of(pdev); diff --git a/drivers/video/fbdev/omap2/dss/hdmi5.c b/drivers/video/fbdev/omap2/dss/hdmi5.c index 7f87578..b6c767b 100644 --- a/drivers/video/fbdev/omap2/dss/hdmi5.c +++ b/drivers/video/fbdev/omap2/dss/hdmi5.c @@ -349,9 +349,24 @@ static int read_edid(u8 *buf, int len) return r; } +static void hdmi_start_audio_stream(struct omap_hdmi *hd) +{ + REG_FLD_MOD(hdmi.wp.base, HDMI_WP_SYSCONFIG, 1, 3, 2); + hdmi_wp_audio_enable(&hd->wp, true); + hdmi_wp_audio_core_req_enable(&hd->wp, true); +} + +static void hdmi_stop_audio_stream(struct omap_hdmi *hd) +{ + hdmi_wp_audio_core_req_enable(&hd->wp, false); + hdmi_wp_audio_enable(&hd->wp, false); + REG_FLD_MOD(hd->wp.base, HDMI_WP_SYSCONFIG, hd->wp_idlemode, 3, 2); +} + static int hdmi_display_enable(struct omap_dss_device *dssdev) { struct omap_dss_device *out = &hdmi.output; + unsigned long flags; int r = 0; DSSDBG("ENTER hdmi_display_enable\n"); @@ -370,7 +385,21 @@ static int hdmi_display_enable(struct omap_dss_device *dssdev) goto err0; } + if (hdmi.audio_configured) { + r = hdmi5_audio_config(&hdmi.core, &hdmi.wp, &hdmi.audio_config, + hdmi.cfg.timings.pixelclock); + if (r) { + DSSERR("Error restoring audio configuration: %d", r); + hdmi.audio_abort_cb(&hdmi.pdev->dev); + hdmi.audio_configured = false; + } + } + + spin_lock_irqsave(&hdmi.audio_playing_lock, flags); + if (hdmi.audio_configured && hdmi.audio_playing) + hdmi_start_audio_stream(&hdmi); hdmi.display_enabled = true; + spin_unlock_irqrestore(&hdmi.audio_playing_lock, flags); mutex_unlock(&hdmi.lock); return 0; @@ -382,17 +411,19 @@ err0: static void hdmi_display_disable(struct omap_dss_device *dssdev) { + unsigned long flags; + DSSDBG("Enter hdmi_display_disable\n"); mutex_lock(&hdmi.lock); - if (hdmi.audio_pdev && hdmi.audio_abort_cb) - hdmi.audio_abort_cb(&hdmi.audio_pdev->dev); + spin_lock_irqsave(&hdmi.audio_playing_lock, flags); + hdmi_stop_audio_stream(&hdmi); + hdmi.display_enabled = false; + spin_unlock_irqrestore(&hdmi.audio_playing_lock, flags); hdmi_power_off_full(dssdev); - hdmi.display_enabled = false; - mutex_unlock(&hdmi.lock); } @@ -596,6 +627,8 @@ static int hdmi_audio_shutdown(struct device *dev) mutex_lock(&hd->lock); hd->audio_abort_cb = NULL; + hd->audio_configured = false; + hd->audio_playing = false; mutex_unlock(&hd->lock); return 0; @@ -604,32 +637,35 @@ static int hdmi_audio_shutdown(struct device *dev) static int hdmi_audio_start(struct device *dev) { struct omap_hdmi *hd = dev_get_drvdata(dev); + unsigned long flags; WARN_ON(!hdmi_mode_has_audio(&hd->cfg)); - WARN_ON(!hd->display_enabled); - /* No-idle while playing audio, store the old value */ - hd->wp_idlemode = REG_GET(hdmi.wp.base, HDMI_WP_SYSCONFIG, 3, 2); - REG_FLD_MOD(hdmi.wp.base, HDMI_WP_SYSCONFIG, 1, 3, 2); + spin_lock_irqsave(&hd->audio_playing_lock, flags); - hdmi_wp_audio_enable(&hd->wp, true); - hdmi_wp_audio_core_req_enable(&hd->wp, true); + if (hd->display_enabled) { + hdmi_start_audio_stream(hd); + } + hd->audio_playing = true; + spin_unlock_irqrestore(&hd->audio_playing_lock, flags); return 0; } static void hdmi_audio_stop(struct device *dev) { struct omap_hdmi *hd = dev_get_drvdata(dev); + unsigned long flags; WARN_ON(!hdmi_mode_has_audio(&hd->cfg)); - WARN_ON(!hd->display_enabled); - hdmi_wp_audio_core_req_enable(&hd->wp, false); - hdmi_wp_audio_enable(&hd->wp, false); + spin_lock_irqsave(&hd->audio_playing_lock, flags); + + if (hd->display_enabled) + hdmi_stop_audio_stream(hd); + hd->audio_playing = false; - /* Playback stopped, restore original idlemode */ - REG_FLD_MOD(hdmi.wp.base, HDMI_WP_SYSCONFIG, hd->wp_idlemode, 3, 2); + spin_unlock_irqrestore(&hd->audio_playing_lock, flags); } static int hdmi_audio_config(struct device *dev, @@ -648,6 +684,10 @@ static int hdmi_audio_config(struct device *dev, ret = hdmi5_audio_config(&hd->core, &hd->wp, dss_audio, hd->cfg.timings.pixelclock); + if (!ret) { + hd->audio_configured = true; + hd->audio_config = *dss_audio; + } out: mutex_unlock(&hd->lock); @@ -678,6 +718,11 @@ static int hdmi_audio_register(struct device *dev) if (IS_ERR(hdmi.audio_pdev)) return PTR_ERR(hdmi.audio_pdev); + hdmi_runtime_get(); + hdmi.wp_idlemode = + REG_GET(hdmi.wp.base, HDMI_WP_SYSCONFIG, 3, 2); + hdmi_runtime_put(); + return 0; } @@ -692,6 +737,7 @@ static int hdmi5_bind(struct device *dev, struct device *master, void *data) dev_set_drvdata(&pdev->dev, &hdmi); mutex_init(&hdmi.lock); + spin_lock_init(&hdmi.audio_playing_lock); if (pdev->dev.of_node) { r = hdmi_probe_of(pdev); -- 1.9.1

1 0

[alsa-devel] [PATCH v2] OMAPDSS: hdmi: Reconfigure and restart audio when display is enabled
by Jyri Sarha 28 Aug '15

28 Aug '15

Reconfigure and restart audio when display is enabled, if audio playback was active before. The audio configuration is stored when is is successfully applied and a boolean is set when playback has started and unset when stopped. This data is used to reconfigure the audio when display is re-enabled. Abort audio playback if reconfiguration fails. A new spin lock is introduced in order to protect state variables related to audio playback status. This is needed for the transitions from display enabled state (when audio start/stop commands can be written to HW) to display disabled state (when audio start/stop commands update only the hdmi.audio_playing variable) to always serialize correctly with the start/stop audio commands. For example: when display is turned back on we take the spinlock and we can be sure that the audio start/stop status won't change while we update the HW according to hdmi.audio_playing state and set hdmi.display_enabled to true. After releasing the lock hdmi.display_enabled is true and all audio_start and audio_stop commands write their stuff directly to HW. Signed-off-by: Jyri Sarha <jsarha(a)ti.com> --- I dropped the ASoC maintainers from the recipient list as this patch hardly concerns them. drivers/video/fbdev/omap2/dss/hdmi.h | 9 +++- drivers/video/fbdev/omap2/dss/hdmi4.c | 69 +++++++++++++++++++++++++----- drivers/video/fbdev/omap2/dss/hdmi5.c | 79 ++++++++++++++++++++++++++++------- 3 files changed, 130 insertions(+), 27 deletions(-) diff --git a/drivers/video/fbdev/omap2/dss/hdmi.h b/drivers/video/fbdev/omap2/dss/hdmi.h index e4a32fe..e48aefd 100644 --- a/drivers/video/fbdev/omap2/dss/hdmi.h +++ b/drivers/video/fbdev/omap2/dss/hdmi.h @@ -351,13 +351,20 @@ struct omap_hdmi { struct regulator *vdda_reg; bool core_enabled; - bool display_enabled; struct omap_dss_device output; struct platform_device *audio_pdev; void (*audio_abort_cb)(struct device *dev); int wp_idlemode; + + bool audio_configured; + struct omap_dss_audio audio_config; + + /* This lock should be taken when booleas bellow is touched. */ + spinlock_t audio_playing_lock; + bool audio_playing; + bool display_enabled; }; #endif diff --git a/drivers/video/fbdev/omap2/dss/hdmi4.c b/drivers/video/fbdev/omap2/dss/hdmi4.c index 6d3aa3f..6b9817a 100644 --- a/drivers/video/fbdev/omap2/dss/hdmi4.c +++ b/drivers/video/fbdev/omap2/dss/hdmi4.c @@ -321,9 +321,22 @@ static int read_edid(u8 *buf, int len) return r; } +static void hdmi_start_audio_stream(struct omap_hdmi *hd) +{ + hdmi_wp_audio_enable(&hd->wp, true); + hdmi4_audio_start(&hd->core, &hd->wp); +} + +static void hdmi_stop_audio_stream(struct omap_hdmi *hd) +{ + hdmi4_audio_stop(&hd->core, &hd->wp); + hdmi_wp_audio_enable(&hd->wp, false); +} + static int hdmi_display_enable(struct omap_dss_device *dssdev) { struct omap_dss_device *out = &hdmi.output; + unsigned long flags; int r = 0; DSSDBG("ENTER hdmi_display_enable\n"); @@ -342,7 +355,21 @@ static int hdmi_display_enable(struct omap_dss_device *dssdev) goto err0; } + if (hdmi.audio_configured) { + r = hdmi4_audio_config(&hdmi.core, &hdmi.wp, &hdmi.audio_config, + hdmi.cfg.timings.pixelclock); + if (r) { + DSSERR("Error restoring audio configuration: %d", r); + hdmi.audio_abort_cb(&hdmi.pdev->dev); + hdmi.audio_configured = false; + } + } + + spin_lock_irqsave(&hdmi.audio_playing_lock, flags); + if (hdmi.audio_configured && hdmi.audio_playing) + hdmi_start_audio_stream(&hdmi); hdmi.display_enabled = true; + spin_unlock_irqrestore(&hdmi.audio_playing_lock, flags); mutex_unlock(&hdmi.lock); return 0; @@ -354,17 +381,19 @@ err0: static void hdmi_display_disable(struct omap_dss_device *dssdev) { + unsigned long flags; + DSSDBG("Enter hdmi_display_disable\n"); mutex_lock(&hdmi.lock); - if (hdmi.audio_pdev && hdmi.audio_abort_cb) - hdmi.audio_abort_cb(&hdmi.audio_pdev->dev); + spin_lock_irqsave(&hdmi.audio_playing_lock, flags); + hdmi_stop_audio_stream(&hdmi); + hdmi.display_enabled = false; + spin_unlock_irqrestore(&hdmi.audio_playing_lock, flags); hdmi_power_off_full(dssdev); - hdmi.display_enabled = false; - mutex_unlock(&hdmi.lock); } @@ -565,9 +594,14 @@ out: static int hdmi_audio_shutdown(struct device *dev) { struct omap_hdmi *hd = dev_get_drvdata(dev); + unsigned long flags; mutex_lock(&hd->lock); hd->audio_abort_cb = NULL; + hd->audio_configured = false; + spin_lock_irqsave(&hd->audio_playing_lock, flags); + hd->audio_playing = false; + spin_unlock_irqrestore(&hd->audio_playing_lock, flags); mutex_unlock(&hd->lock); return 0; @@ -576,25 +610,34 @@ static int hdmi_audio_shutdown(struct device *dev) static int hdmi_audio_start(struct device *dev) { struct omap_hdmi *hd = dev_get_drvdata(dev); + unsigned long flags; WARN_ON(!hdmi_mode_has_audio(&hd->cfg)); - WARN_ON(!hd->display_enabled); - hdmi_wp_audio_enable(&hd->wp, true); - hdmi4_audio_start(&hd->core, &hd->wp); + spin_lock_irqsave(&hd->audio_playing_lock, flags); + + if (hd->display_enabled) + hdmi_start_audio_stream(hd); + hd->audio_playing = true; + spin_unlock_irqrestore(&hd->audio_playing_lock, flags); return 0; } static void hdmi_audio_stop(struct device *dev) { struct omap_hdmi *hd = dev_get_drvdata(dev); + unsigned long flags; WARN_ON(!hdmi_mode_has_audio(&hd->cfg)); - WARN_ON(!hd->display_enabled); - hdmi4_audio_stop(&hd->core, &hd->wp); - hdmi_wp_audio_enable(&hd->wp, false); + spin_lock_irqsave(&hd->audio_playing_lock, flags); + + if (hd->display_enabled) + hdmi_stop_audio_stream(hd); + hd->audio_playing = false; + + spin_unlock_irqrestore(&hd->audio_playing_lock, flags); } static int hdmi_audio_config(struct device *dev, @@ -612,7 +655,10 @@ static int hdmi_audio_config(struct device *dev, ret = hdmi4_audio_config(&hd->core, &hd->wp, dss_audio, hd->cfg.timings.pixelclock); - + if (!ret) { + hd->audio_configured = true; + hd->audio_config = *dss_audio; + } out: mutex_unlock(&hd->lock); @@ -657,6 +703,7 @@ static int hdmi4_bind(struct device *dev, struct device *master, void *data) dev_set_drvdata(&pdev->dev, &hdmi); mutex_init(&hdmi.lock); + spin_lock_init(&hdmi.audio_playing_lock); if (pdev->dev.of_node) { r = hdmi_probe_of(pdev); diff --git a/drivers/video/fbdev/omap2/dss/hdmi5.c b/drivers/video/fbdev/omap2/dss/hdmi5.c index 7f87578..85e70af 100644 --- a/drivers/video/fbdev/omap2/dss/hdmi5.c +++ b/drivers/video/fbdev/omap2/dss/hdmi5.c @@ -349,9 +349,24 @@ static int read_edid(u8 *buf, int len) return r; } +static void hdmi_start_audio_stream(struct omap_hdmi *hd) +{ + REG_FLD_MOD(hdmi.wp.base, HDMI_WP_SYSCONFIG, 1, 3, 2); + hdmi_wp_audio_enable(&hd->wp, true); + hdmi_wp_audio_core_req_enable(&hd->wp, true); +} + +static void hdmi_stop_audio_stream(struct omap_hdmi *hd) +{ + hdmi_wp_audio_core_req_enable(&hd->wp, false); + hdmi_wp_audio_enable(&hd->wp, false); + REG_FLD_MOD(hd->wp.base, HDMI_WP_SYSCONFIG, hd->wp_idlemode, 3, 2); +} + static int hdmi_display_enable(struct omap_dss_device *dssdev) { struct omap_dss_device *out = &hdmi.output; + unsigned long flags; int r = 0; DSSDBG("ENTER hdmi_display_enable\n"); @@ -370,7 +385,21 @@ static int hdmi_display_enable(struct omap_dss_device *dssdev) goto err0; } + if (hdmi.audio_configured) { + r = hdmi5_audio_config(&hdmi.core, &hdmi.wp, &hdmi.audio_config, + hdmi.cfg.timings.pixelclock); + if (r) { + DSSERR("Error restoring audio configuration: %d", r); + hdmi.audio_abort_cb(&hdmi.pdev->dev); + hdmi.audio_configured = false; + } + } + + spin_lock_irqsave(&hdmi.audio_playing_lock, flags); + if (hdmi.audio_configured && hdmi.audio_playing) + hdmi_start_audio_stream(&hdmi); hdmi.display_enabled = true; + spin_unlock_irqrestore(&hdmi.audio_playing_lock, flags); mutex_unlock(&hdmi.lock); return 0; @@ -382,17 +411,19 @@ err0: static void hdmi_display_disable(struct omap_dss_device *dssdev) { + unsigned long flags; + DSSDBG("Enter hdmi_display_disable\n"); mutex_lock(&hdmi.lock); - if (hdmi.audio_pdev && hdmi.audio_abort_cb) - hdmi.audio_abort_cb(&hdmi.audio_pdev->dev); + spin_lock_irqsave(&hdmi.audio_playing_lock, flags); + hdmi_stop_audio_stream(&hdmi); + hdmi.display_enabled = false; + spin_unlock_irqrestore(&hdmi.audio_playing_lock, flags); hdmi_power_off_full(dssdev); - hdmi.display_enabled = false; - mutex_unlock(&hdmi.lock); } @@ -593,9 +624,14 @@ out: static int hdmi_audio_shutdown(struct device *dev) { struct omap_hdmi *hd = dev_get_drvdata(dev); + unsigned long flags; mutex_lock(&hd->lock); hd->audio_abort_cb = NULL; + hd->audio_configured = false; + spin_lock_irqsave(&hd->audio_playing_lock, flags); + hd->audio_playing = false; + spin_unlock_irqrestore(&hd->audio_playing_lock, flags); mutex_unlock(&hd->lock); return 0; @@ -604,32 +640,35 @@ static int hdmi_audio_shutdown(struct device *dev) static int hdmi_audio_start(struct device *dev) { struct omap_hdmi *hd = dev_get_drvdata(dev); + unsigned long flags; WARN_ON(!hdmi_mode_has_audio(&hd->cfg)); - WARN_ON(!hd->display_enabled); - /* No-idle while playing audio, store the old value */ - hd->wp_idlemode = REG_GET(hdmi.wp.base, HDMI_WP_SYSCONFIG, 3, 2); - REG_FLD_MOD(hdmi.wp.base, HDMI_WP_SYSCONFIG, 1, 3, 2); + spin_lock_irqsave(&hd->audio_playing_lock, flags); - hdmi_wp_audio_enable(&hd->wp, true); - hdmi_wp_audio_core_req_enable(&hd->wp, true); + if (hd->display_enabled) { + hdmi_start_audio_stream(hd); + } + hd->audio_playing = true; + spin_unlock_irqrestore(&hd->audio_playing_lock, flags); return 0; } static void hdmi_audio_stop(struct device *dev) { struct omap_hdmi *hd = dev_get_drvdata(dev); + unsigned long flags; WARN_ON(!hdmi_mode_has_audio(&hd->cfg)); - WARN_ON(!hd->display_enabled); - hdmi_wp_audio_core_req_enable(&hd->wp, false); - hdmi_wp_audio_enable(&hd->wp, false); + spin_lock_irqsave(&hd->audio_playing_lock, flags); + + if (hd->display_enabled) + hdmi_stop_audio_stream(hd); + hd->audio_playing = false; - /* Playback stopped, restore original idlemode */ - REG_FLD_MOD(hdmi.wp.base, HDMI_WP_SYSCONFIG, hd->wp_idlemode, 3, 2); + spin_unlock_irqrestore(&hd->audio_playing_lock, flags); } static int hdmi_audio_config(struct device *dev, @@ -648,6 +687,10 @@ static int hdmi_audio_config(struct device *dev, ret = hdmi5_audio_config(&hd->core, &hd->wp, dss_audio, hd->cfg.timings.pixelclock); + if (!ret) { + hd->audio_configured = true; + hd->audio_config = *dss_audio; + } out: mutex_unlock(&hd->lock); @@ -678,6 +721,11 @@ static int hdmi_audio_register(struct device *dev) if (IS_ERR(hdmi.audio_pdev)) return PTR_ERR(hdmi.audio_pdev); + hdmi_runtime_get(); + hdmi.wp_idlemode = + REG_GET(hdmi.wp.base, HDMI_WP_SYSCONFIG, 3, 2); + hdmi_runtime_put(); + return 0; } @@ -692,6 +740,7 @@ static int hdmi5_bind(struct device *dev, struct device *master, void *data) dev_set_drvdata(&pdev->dev, &hdmi); mutex_init(&hdmi.lock); + spin_lock_init(&hdmi.audio_playing_lock); if (pdev->dev.of_node) { r = hdmi_probe_of(pdev); -- 1.9.1

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