May 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v4] ASoC: dfbmcs320: make the driver common for other BT modules
by Barry Song 22 May '13

22 May '13

DFBM-CS320 is only one of bluetooth modules using CSR bluetooth chips, we don't want everyone to have a seperate codec driver. anyway, the feature of Bluetooth SCO is same on all platforms, so this patch makes the DFBM-CS320 driver become a common BT SCO link driver. Cc: Graeme Gregory <graeme(a)openmoko.org> Cc: Lars-Peter Clausen <lars(a)metafoo.de> Signed-off-by: Barry Song <Baohua.Song(a)csr.com> --- v4: change "select SND_SOC_DFBMCS320" to "select SND_SOC_BT_SCO" for OpenMOKO user as v3 missed this sound/soc/codecs/Kconfig | 4 ++-- sound/soc/codecs/Makefile | 4 ++-- sound/soc/codecs/{dfbmcs320.c => bt-sco.c} | 37 +++++++++++++++++++----------- sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/neo1973_wm8753.c | 2 +- 5 files changed, 29 insertions(+), 20 deletions(-) rename sound/soc/codecs/{dfbmcs320.c => bt-sco.c} (53%) diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index d8c4f3d..386292d 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -40,7 +40,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_DA7213 if I2C select SND_SOC_DA732X if I2C select SND_SOC_DA9055 if I2C - select SND_SOC_DFBMCS320 + select SND_SOC_BT_SCO select SND_SOC_ISABELLE if I2C select SND_SOC_JZ4740_CODEC select SND_SOC_LM4857 if I2C @@ -263,7 +263,7 @@ config SND_SOC_DA732X config SND_SOC_DA9055 tristate -config SND_SOC_DFBMCS320 +config SND_SOC_BT_SCO tristate config SND_SOC_DMIC diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index 49ff127..82aa34e 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -27,7 +27,7 @@ snd-soc-da7210-objs := da7210.o snd-soc-da7213-objs := da7213.o snd-soc-da732x-objs := da732x.o snd-soc-da9055-objs := da9055.o -snd-soc-dfbmcs320-objs := dfbmcs320.o +snd-soc-bt-sco-objs := bt-sco.o snd-soc-dmic-objs := dmic.o snd-soc-isabelle-objs := isabelle.o snd-soc-jz4740-codec-objs := jz4740.o @@ -154,7 +154,7 @@ obj-$(CONFIG_SND_SOC_DA7210) += snd-soc-da7210.o obj-$(CONFIG_SND_SOC_DA7213) += snd-soc-da7213.o obj-$(CONFIG_SND_SOC_DA732X) += snd-soc-da732x.o obj-$(CONFIG_SND_SOC_DA9055) += snd-soc-da9055.o -obj-$(CONFIG_SND_SOC_DFBMCS320) += snd-soc-dfbmcs320.o +obj-$(CONFIG_SND_SOC_BT_SCO) += snd-soc-bt-sco.o obj-$(CONFIG_SND_SOC_DMIC) += snd-soc-dmic.o obj-$(CONFIG_SND_SOC_ISABELLE) += snd-soc-isabelle.o obj-$(CONFIG_SND_SOC_JZ4740_CODEC) += snd-soc-jz4740-codec.o diff --git a/sound/soc/codecs/dfbmcs320.c b/sound/soc/codecs/bt-sco.c similarity index 53% rename from sound/soc/codecs/dfbmcs320.c rename to sound/soc/codecs/bt-sco.c index 4f4f7f4..a081d9f 100644 --- a/sound/soc/codecs/dfbmcs320.c +++ b/sound/soc/codecs/bt-sco.c @@ -1,5 +1,5 @@ /* - * Driver for the DFBM-CS320 bluetooth module + * Driver for generic Bluetooth SCO link * Copyright 2011 Lars-Peter Clausen <lars(a)metafoo.de> * * This program is free software; you can redistribute it and/or modify it @@ -15,8 +15,8 @@ #include <sound/soc.h> -static struct snd_soc_dai_driver dfbmcs320_dai = { - .name = "dfbmcs320-pcm", +static struct snd_soc_dai_driver bt_sco_dai = { + .name = "bt-sco-pcm", .playback = { .channels_min = 1, .channels_max = 1, @@ -31,32 +31,41 @@ static struct snd_soc_dai_driver dfbmcs320_dai = { }, }; -static struct snd_soc_codec_driver soc_codec_dev_dfbmcs320; +static struct snd_soc_codec_driver soc_codec_dev_bt_sco; -static int dfbmcs320_probe(struct platform_device *pdev) +static int bt_sco_probe(struct platform_device *pdev) { - return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_dfbmcs320, - &dfbmcs320_dai, 1); + return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_bt_sco, + &bt_sco_dai, 1); } -static int dfbmcs320_remove(struct platform_device *pdev) +static int bt_sco_remove(struct platform_device *pdev) { snd_soc_unregister_codec(&pdev->dev); return 0; } -static struct platform_driver dfmcs320_driver = { +static struct platform_device_id bt_sco_driver_ids[] = { + { + .name = "dfbmcs320", + }, + {}, +}; +MODULE_DEVICE_TABLE(platform, bt_sco_driver_ids); + +static struct platform_driver bt_sco_driver = { .driver = { - .name = "dfbmcs320", + .name = "bt-sco", .owner = THIS_MODULE, }, - .probe = dfbmcs320_probe, - .remove = dfbmcs320_remove, + .probe = bt_sco_probe, + .remove = bt_sco_remove, + .id_table = bt_sco_driver_ids, }; -module_platform_driver(dfmcs320_driver); +module_platform_driver(bt_sco_driver); MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); -MODULE_DESCRIPTION("ASoC DFBM-CS320 bluethooth module driver"); +MODULE_DESCRIPTION("ASoC generic bluethooth sco link driver"); MODULE_LICENSE("GPL"); diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 475fb0d..ae0ea87 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -39,7 +39,7 @@ config SND_SOC_SAMSUNG_NEO1973_WM8753 depends on SND_SOC_SAMSUNG && MACH_NEO1973_GTA02 select SND_S3C24XX_I2S select SND_SOC_WM8753 - select SND_SOC_DFBMCS320 + select SND_SOC_SCO help Say Y here to enable audio support for the Openmoko Neo1973 Smartphones. diff --git a/sound/soc/samsung/neo1973_wm8753.c b/sound/soc/samsung/neo1973_wm8753.c index e591c38..807db41 100644 --- a/sound/soc/samsung/neo1973_wm8753.c +++ b/sound/soc/samsung/neo1973_wm8753.c @@ -373,7 +373,7 @@ static struct snd_soc_dai_link neo1973_dai[] = { { /* Voice via BT */ .name = "Bluetooth", .stream_name = "Voice", - .cpu_dai_name = "dfbmcs320-pcm", + .cpu_dai_name = "bt-sco-pcm", .codec_dai_name = "wm8753-voice", .codec_name = "wm8753.0-001a", .ops = &neo1973_voice_ops, -- 1.8.2.3

3 4

Re: [alsa-devel] [PATCH 3/3] sound/soc/fsl: remove leftover release_mem_region
by Mark Brown 22 May '13

22 May '13

On Wed, May 22, 2013 at 03:21:26PM +0200, Wolfram Sang wrote: > When converting this driver to devm_ioremap_resource, the removal of this now > unneeded function has been forgotten. This doesn't apply against current code, can you please regenerate against topic/fsl or mainline?

1 0

[alsa-devel] [PATCH] ALSA: hda - Enable runtime pm for Haswell
by Wang Xingchao 22 May '13

22 May '13

Haswell doesnot support runtime pm by default. This patch let haswell Display HD-A controller enter runtime suspend, and bring more power saving whith power-well. Signed-off-by: Wang Xingchao <xingchao.wang(a)linux.intel.com> --- sound/pci/hda/hda_intel.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index bf27693..eb25888 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -3755,6 +3755,12 @@ static int azx_probe(struct pci_dev *pci, if (pci_dev_run_wake(pci)) pm_runtime_put_noidle(&pci->dev); + else if (chip->driver_caps + & AZX_DCAPS_I915_POWERWELL) { + /* Haswell doesnot support runtime pm by default */ + pm_runtime_put_noidle(&pci->dev); + pm_runtime_allow(&pci->dev); + } dev++; complete_all(&chip->probe_wait); -- 1.7.9.5

3 4

[alsa-devel] [PATCH] ASoC: davinci: fix sample rotation
by Daniel Mack 21 May '13

21 May '13

McASP serial audio engine needs different rotation values on TX and RX channels. Commit dde109fb462 ("ASoC: McASP: Fix data rotation for playback. Enables 24bit audio playback") changed the calculation to fix the playback format, but broke the capture stream by doing it for both TXFMT and RXFMT. Signed-off-by: Daniel Mack <zonque(a)gmail.com> Cc: Michal Bachraty <michal.bachraty(a)streamunlimited.com> Cc: stable(a)vger.kernel.org [3.9 only] --- sound/soc/davinci/davinci-mcasp.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index 63633e9..8d805ec 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -628,7 +628,8 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, int word_length) { u32 fmt; - u32 rotate = (word_length / 4) & 0x7; + u32 tx_rotate = (word_length / 4) & 0x7; + u32 rx_rotate = (32 - word_length) / 4; u32 mask = (1ULL << word_length) - 1; /* @@ -651,9 +652,9 @@ static int davinci_config_channel_size(struct davinci_audio_dev *dev, mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, TXSSZ(fmt), TXSSZ(0x0F)); mcasp_mod_bits(dev->base + DAVINCI_MCASP_TXFMT_REG, - TXROT(rotate), TXROT(7)); + TXROT(tx_rotate), TXROT(7)); mcasp_mod_bits(dev->base + DAVINCI_MCASP_RXFMT_REG, - RXROT(rotate), RXROT(7)); + RXROT(rx_rotate), RXROT(7)); mcasp_set_reg(dev->base + DAVINCI_MCASP_RXMASK_REG, mask); } -- 1.8.1.4

2 1

[alsa-devel] [PATCH 0/2] Set ab8500-codec dai slots from machine drivers
by Fabio Baltieri 21 May '13

21 May '13

Hi Mark, these two patches rework the slot selection code in ab8500-codec to use the slots masks provided by the machine drivers instead of the hardcoded ones as suggested. The patches replaces some of the previous macros with a parametrized version in the effort of making the actual code compact and readable, and should be applied in order. Also, the second patch drops the hardcoded defintions swapped in -rc2, so you may want to rebase the ux500 topic branch to avoid the conflict, or I can resend the patch based on the topic branch directly. Just let me know. Big thanks to Ola for helping me figuring out the weird ab8500 slot mapping logic. Fabio Fabio Baltieri (2): ASoC: ab8500-codec: Set tx dai slots from tx_mask ASoC: ab8500-codec: Set rx dai slots from rx_mask sound/soc/codecs/ab8500-codec.c | 60 ++++++++++++++++++++++++++--------------- sound/soc/codecs/ab8500-codec.h | 42 +++++++++++++++-------------- 2 files changed, 61 insertions(+), 41 deletions(-) -- 1.8.2

2 5

[alsa-devel] [PATCH 0/2 V4] Power-well API implementation for Haswell
by Wang Xingchao 21 May '13

21 May '13

Hi all, This is V4 and here're some changes notes: change between V3-->V4: - add new structure i915_power_well - initialize drm_device pointer at module init time - change function name change between V2-->V3: - make SND_HDA_I915 selectable - use snd_printdd to output message - add return error code check - use symbol_request to replace symbol_get - release power_well at azx_free - some typo fixes changes between V1-->V2: - use reference count to track power-well usage - remove external module, compiled into snd-hda-intel instead - manage symbols and module loading properly - remove IS_HSW macro, use flag instead - remove audio callback for gfx driver to avoid dependency - split whole patch into two pieces for easy review - more typo fixes Wang Xingchao (2): i915/drm: Add private api for power well usage ALSA: hda - Add power-welll support for haswell HDA drivers/gpu/drm/i915/i915_dma.c | 6 +++ drivers/gpu/drm/i915/i915_drv.h | 12 +++++ drivers/gpu/drm/i915/intel_drv.h | 4 ++ drivers/gpu/drm/i915/intel_pm.c | 98 +++++++++++++++++++++++++++++++++++--- include/drm/i915_powerwell.h | 36 ++++++++++++++ sound/pci/hda/Kconfig | 10 ++++ sound/pci/hda/Makefile | 3 ++ sound/pci/hda/hda_i915.c | 75 +++++++++++++++++++++++++++++ sound/pci/hda/hda_i915.h | 35 ++++++++++++++ sound/pci/hda/hda_intel.c | 41 ++++++++++++++-- 10 files changed, 310 insertions(+), 10 deletions(-) create mode 100644 include/drm/i915_powerwell.h create mode 100644 sound/pci/hda/hda_i915.c create mode 100644 sound/pci/hda/hda_i915.h -- 1.7.9.5

3 7

Re: [alsa-devel] [Alsa-devel] add support for syba 7.1 surround sound card
by Clemens Ladisch 21 May '13

21 May '13

good1.2guy(a)gmail.com wrote: > Summary: add support for syba 7.1 surround sound card > Keywords: syba via VT1723 Tremor > Kernel: 3.8.8-203.fc18.x86_64 What are these three lines for? > add support for syba 7.1 surround sound card > actually the card only has 6 dmas, so it is really only 5.1 The website shows a daughterboard. > added code to force_rate setting to not fail EBUSY when the dmas are running. > suggest using force_rate with pulseaudio to avoid loops with error logging. This sounds as if it has nothing to do with this particular SYBA card but affects all VT1723 cards. Please move it into a separate patch. And there should be no reason for a module parameter; if this is a bug in the driver and/or in PulseAudio, it should be just fixed. > Signed-off-by: psmith2004(a)yahoo.com Real names only. Please read Documentation/SubmittingPatches. > +static void syba_quirk_open(struct snd_ice1712 *ice) > +{ > + unsigned long flags; It's common to have an empty line here. > + spin_lock_irqsave(&ice->reg_lock, flags); > + outb(0xff, ICEREG1724(ice, AC97_CFG)); > + udelay(500); > + outb(ice->eeprom.data[ICE_EEP2_ACLINK], ICEREG1724(ice, AC97_CFG)); > + spin_unlock_irqrestore(&ice->reg_lock, flags); > +} What is the reason for this quirk? Is it possible that this might be needed for other cards using a AC'97 codec? Regards, Clemens

1 0

[alsa-devel] what is the difference between SNDRV_PCM_IOCTL_WRITEN_FRAMES and SNDRV_PCM_IOCTL_WRITEI_FRAMES
by Zhang wei 21 May '13

21 May '13

I know the format likes below when use SNDRV_PCM_IOCTL_WRITEI_FRAMES <http://lxr.free-electrons.com/ident?i=SNDRV_PCM_IOCTL_WRITEN_FRAMES> L+R+L+R+L+R... L:a frame left channel data. R:a frame right channel data. Right? But what is the kind of format when use SNDRV_PCM_IOCTL_WRITEN_FRAMES <http://lxr.free-electrons.com/ident?i=SNDRV_PCM_IOCTL_WRITEN_FRAMES> ? Thank you:-) Br zw

2 4

[alsa-devel] [PATCH] ASoC: wm8994: Handle LRCLK inversion for WM8958 and WM1811A
by Mark Brown 20 May '13

20 May '13

On WM8958 and WM1811A separate control of the LRCLK inversion bit is available for the DAC and ADC LRCLKs which for compatibility reasons is done in a new register bit. Since writes to each scheme have no effect on parts using the other just always write to both for simplicity. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Acked-by: Vinod Koul <vinod.koul(a)intel.com> Tested-by: Samreen Nilofer <samreen.nilofer(a)intel.com> --- include/linux/mfd/wm8994/registers.h | 8 ++++++++ sound/soc/codecs/wm8994.c | 14 ++++++++++++++ 2 files changed, 22 insertions(+) diff --git a/include/linux/mfd/wm8994/registers.h b/include/linux/mfd/wm8994/registers.h index 0535489..db8cef3 100644 --- a/include/linux/mfd/wm8994/registers.h +++ b/include/linux/mfd/wm8994/registers.h @@ -2668,6 +2668,10 @@ /* * R772 (0x304) - AIF1ADC LRCLK */ +#define WM8958_AIF1_LRCLK_INV 0x1000 /* AIF1_LRCLK_INV */ +#define WM8958_AIF1_LRCLK_INV_MASK 0x1000 /* AIF1_LRCLK_INV */ +#define WM8958_AIF1_LRCLK_INV_SHIFT 12 /* AIF1_LRCLK_INV */ +#define WM8958_AIF1_LRCLK_INV_WIDTH 1 /* AIF1_LRCLK_INV */ #define WM8994_AIF1ADC_LRCLK_DIR 0x0800 /* AIF1ADC_LRCLK_DIR */ #define WM8994_AIF1ADC_LRCLK_DIR_MASK 0x0800 /* AIF1ADC_LRCLK_DIR */ #define WM8994_AIF1ADC_LRCLK_DIR_SHIFT 11 /* AIF1ADC_LRCLK_DIR */ @@ -2679,6 +2683,10 @@ /* * R773 (0x305) - AIF1DAC LRCLK */ +#define WM8958_AIF1_LRCLK_INV 0x1000 /* AIF1_LRCLK_INV */ +#define WM8958_AIF1_LRCLK_INV_MASK 0x1000 /* AIF1_LRCLK_INV */ +#define WM8958_AIF1_LRCLK_INV_SHIFT 12 /* AIF1_LRCLK_INV */ +#define WM8958_AIF1_LRCLK_INV_WIDTH 1 /* AIF1_LRCLK_INV */ #define WM8994_AIF1DAC_LRCLK_DIR 0x0800 /* AIF1DAC_LRCLK_DIR */ #define WM8994_AIF1DAC_LRCLK_DIR_MASK 0x0800 /* AIF1DAC_LRCLK_DIR */ #define WM8994_AIF1DAC_LRCLK_DIR_SHIFT 11 /* AIF1DAC_LRCLK_DIR */ diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index cb67603..0bcbd12 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2574,17 +2574,24 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) struct wm8994 *control = wm8994->wm8994; int ms_reg; int aif1_reg; + int dac_reg; + int adc_reg; int ms = 0; int aif1 = 0; + int lrclk = 0; switch (dai->id) { case 1: ms_reg = WM8994_AIF1_MASTER_SLAVE; aif1_reg = WM8994_AIF1_CONTROL_1; + dac_reg = WM8994_AIF1DAC_LRCLK; + adc_reg = WM8994_AIF1ADC_LRCLK; break; case 2: ms_reg = WM8994_AIF2_MASTER_SLAVE; aif1_reg = WM8994_AIF2_CONTROL_1; + dac_reg = WM8994_AIF1DAC_LRCLK; + adc_reg = WM8994_AIF1ADC_LRCLK; break; default: return -EINVAL; @@ -2603,6 +2610,7 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_DSP_B: aif1 |= WM8994_AIF1_LRCLK_INV; + lrclk |= WM8958_AIF1_LRCLK_INV; case SND_SOC_DAIFMT_DSP_A: aif1 |= 0x18; break; @@ -2641,12 +2649,14 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) break; case SND_SOC_DAIFMT_IB_IF: aif1 |= WM8994_AIF1_BCLK_INV | WM8994_AIF1_LRCLK_INV; + lrclk |= WM8958_AIF1_LRCLK_INV; break; case SND_SOC_DAIFMT_IB_NF: aif1 |= WM8994_AIF1_BCLK_INV; break; case SND_SOC_DAIFMT_NB_IF: aif1 |= WM8994_AIF1_LRCLK_INV; + lrclk |= WM8958_AIF1_LRCLK_INV; break; default: return -EINVAL; @@ -2677,6 +2687,10 @@ static int wm8994_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) aif1); snd_soc_update_bits(codec, ms_reg, WM8994_AIF1_MSTR, ms); + snd_soc_update_bits(codec, dac_reg, + WM8958_AIF1_LRCLK_INV, lrclk); + snd_soc_update_bits(codec, adc_reg, + WM8958_AIF1_LRCLK_INV, lrclk); return 0; } -- 1.7.10.4

1 0

[alsa-devel] [PATCH v2] ASoC: soc-compress: Send correct stream event for capture start
by Charles Keepax 20 May '13

20 May '13

Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/soc-compress.c | 8 ++++++-- 1 files changed, 6 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-compress.c b/sound/soc/soc-compress.c index 3853f7e..06a8000 100644 --- a/sound/soc/soc-compress.c +++ b/sound/soc/soc-compress.c @@ -220,8 +220,12 @@ static int soc_compr_set_params(struct snd_compr_stream *cstream, goto err; } - snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, - SND_SOC_DAPM_STREAM_START); + if (cstream->direction == SND_COMPRESS_PLAYBACK) + snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_PLAYBACK, + SND_SOC_DAPM_STREAM_START); + else + snd_soc_dapm_stream_event(rtd, SNDRV_PCM_STREAM_CAPTURE, + SND_SOC_DAPM_STREAM_START); /* cancel any delayed stream shutdown that is pending */ rtd->pop_wait = 0; -- 1.7.2.5

2 1