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

[alsa-devel] [PATCH 00/18] ARM: s3c64xx: Let amba-pl08x driver handle DMA
by Tomasz Figa 02 Sep '13

02 Sep '13

This is first non-RFC version of my patches extending support of amba-pl08x DMA engine driver to PL080S DMA engine (PL080 modified by Samsung) found in Samsung S3C64xx SoCs. Due to changes scattered across different areas of kernel, patches are based on merged 3 branches: - for-next of Kgene's Samsung tree, - clk-next of Mike's clock tree, - next of Vinod's slave DMA tree. To ease testing I have prepared a branch in my private tree for anyone willing to check the patches out: git://github.com/tom3q/linux.git v3.12-pl080 Dependencies (already applied in my branch): - for patches 14 and 16 - CCF-based clock driver for s3c64xx. Some of the patches not related to the amba-pl08x driver itself can be likely applied into appropriate trees separately, namely: - 09/18 - ASoC: Samsung: Do not queue cyclic buffers multiple times, - 14/18 - clk: samsung: s3c64xx: Add aliases for DMA clocks. After patch 14/18, both old and new DMA drivers can be supported on S3C64xx, depending on Kconfig options. Patches 15-18 remove the old driver leaving support only for the generic pl08x driver. Feel free to drop those patches for now if we want more testing, but I don't suspect any problems. On S3C64xx-based Mini6410 and SMDK6410 boards, with I2S audio playback and capture (including full duplex operation) and also SPI using spidev: Tested-by: Tomasz Figa <tomasz.figa(a)gmail.com> Changes since RFC v2: - Added clkdev lookups to old clock driver. - Added patches removing the old DMA driver and any remaining code needed by it. - Fixed DMA support for SPI. - Added a word about PL080S to amba-pl08x.c file header. - Changed definition of LLI words from enums to macros. - Extended debugging messages to handle PL080S variant as well. - Little cleanup of LLI dumping code. - Added check for peripheral flow control, which is unsupported by PL080S to dma_set_runtime_config. - Corrected transfer size mask of PL080S. Changes since RFC v1: - Returned to original way of storing quirks as booleans, as suggested by Russell, Linus and Arnd. - Added reg_config field to pl08x_phy_chan struct, which stores variant-specific address of channel config register, as suggested by Russell. - Simplified handling of extended maximum transfer size of PL080S (no more conditional passing of 0 as length to pl08x_cctl_bits()). - Reworked LLI handling in the driver to stop casting arbitrary memory to a struct and allow different word count of LLI entry, as suggested by Linus. - Removed AMBA ID override from S3C64xx PL080 initialization code. - Fixed brokenness of Samsung DMA wrapper API, which caused cyclic buffers to be queued multiple times when DMA engine is used. - Included patch adding clock aliases for DMA engines (depends on Common Clock Framework driver for S3C64xx). - Fixed several minor stylistic issues. For reference, here is the original description of the series: One of the biggest roadblocks on the way of S3C64xx to DeviceTree support is its DMA driver, which is completely platform-specific and provides private API (s3c-dma), not even saying that its design is completely against multiplatform-awareness. The DMA controller present on this SoC series is a custom variant of ARM PrimeCell PL080 modified by Samsung to add some extra features. It is mostly compatible with original PL080, except: - CH_CONTROL2 register is added between CH_CONTROL and CH_CONFIG, - offset of CH_CONFIG register is different, - transfer size field is moved from CH_CONTROL to CH_CONTROL2, - transfer size field is extended to 24 bits, allowing much bigger single transfer, - LLI consists of one more word, to account for CH_CONTROL2 register. Since all the rest is fully compatible with standard PL080 there is no point in having separate driver just for this single variant, so I decided to look into adding support for it to the amba-pl08x driver. There was already some attempt to achieve this before, but this was before Russel's big rework of the driver to use virtual channels, making the old patches being not much of use. This RFC series is a proof of concept that I managed to make during last days of hacking. Except one patch adding clkdev lookup to clock driver (which is being replaced with a CCF-compliant driver ATM), this is enough to get memcpy and slave transfers to work on S3C64xx. I have tested this on Mini6410 and SMDK6410 boards using dmatest for memcpy and Samsung I2S with madplay/aplay for slave transfers. Unfortunately I do not have access to other platforms with PL08x so I could not test for any regressions introduced on them. Credits for two patches go to Alban Bedel, who made a series fixing this driver to make it usable with audio drivers. I rebased his patches on top of mine and corrected coding style a bit. OK, that's all. Any comments are welcome. Feel free to start throwing eggs and tomatoes if you find this awful, but I won't be upset if I get some Tested-by or Acked-by as well. ;) Alban Bedel (2): dmaengine: PL08x: Fix reading the byte count in cctl dmaengine: PL08x: Add cyclic transfer support Tomasz Figa (16): dmaengine: PL08x: Refactor pl08x_getbytes_chan() to lower indentation dmaengine: PL08x: Add support for different offset of CONFIG register dmaengine: PL08x: Rework LLI handling to be less fragile dmaengine: PL08x: Move LLI dumping code into separate function dmaengine: PL08x: Add support for PL080S variant dmaengine: PL08x: Add support for different maximum transfer size ASoC: Samsung: Do not queue cyclic buffers multiple times spi: s3c64xx: Do not require legacy DMA API in case of S3C64XX ASoC: Samsung: Do not require legacy DMA API in case of S3C64XX ARM: s3c64xx: Add support for DMA using generic amba-pl08x driver ARM: s3c64xx: clock: Add clkdev lookup for DMA clocks clk: samsung: s3c64xx: Add aliases for DMA clocks ARM: s3c64xx: Remove legacy DMA driver clk: samsung: s3c64xx: Remove clock aliases of old DMA driver spi: s3c64xx: Always select S3C64XX_PL080 when ARCH_S3C64XX is enabled ASoC: Samsung: Always select S3C64XX_PL080 when ARCH_S3C64XX is enabled arch/arm/Kconfig | 1 + arch/arm/mach-s3c64xx/Kconfig | 7 +- arch/arm/mach-s3c64xx/Makefile | 2 +- arch/arm/mach-s3c64xx/clock.c | 28 +- arch/arm/mach-s3c64xx/common.h | 5 + arch/arm/mach-s3c64xx/dma.c | 753 ------------------------------- arch/arm/mach-s3c64xx/include/mach/dma.h | 144 ++---- arch/arm/mach-s3c64xx/pl080.c | 244 ++++++++++ arch/arm/plat-samsung/devs.c | 6 +- arch/arm/plat-samsung/s3c-dma-ops.c | 13 +- drivers/clk/samsung/clk-s3c64xx.c | 4 +- drivers/dma/amba-pl08x.c | 501 ++++++++++++++------ drivers/spi/Kconfig | 2 +- include/linux/amba/pl080.h | 1 + sound/soc/samsung/Kconfig | 2 +- sound/soc/samsung/dma.c | 7 + 16 files changed, 705 insertions(+), 1015 deletions(-) delete mode 100644 arch/arm/mach-s3c64xx/dma.c create mode 100644 arch/arm/mach-s3c64xx/pl080.c -- 1.8.3.2

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[alsa-devel] [PATCH 1/2] ASoC: soc-pcm: Allow to specify unidirectional dai_link
by Fabio Estevam 01 Sep '13

01 Sep '13

Add 'playback_only' and 'capture_only' fields that can be used for specifying that a dai_link has a unidirectional capability. The motivation for this is for the cases of systems, such as Freescale MX28, that has two unidirectional DAIs. Signed-off-by: Fabio Estevam <fabio.estevam(a)freescale.com> --- include/sound/soc.h | 4 ++++ sound/soc/soc-pcm.c | 10 ++++++++++ 2 files changed, 14 insertions(+) diff --git a/include/sound/soc.h b/include/sound/soc.h index 8e2ad52..774d3be 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -934,6 +934,10 @@ struct snd_soc_dai_link { /* machine stream operations */ const struct snd_soc_ops *ops; const struct snd_soc_compr_ops *compr_ops; + + /* For unidirectional dai links */ + bool playback_only; + bool capture_only; }; struct snd_soc_codec_conf { diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index fb70fbe..5e44863 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -2020,6 +2020,16 @@ int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) capture = 1; } + if (rtd->dai_link->playback_only) { + playback = 1; + capture = 0; + } + + if (rtd->dai_link->capture_only) { + playback = 0; + capture = 1; + } + /* create the PCM */ if (rtd->dai_link->no_pcm) { snprintf(new_name, sizeof(new_name), "(%s)", -- 1.8.1.2

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[alsa-devel] [PATCH v3 00/11] ALSA: usb-audio: fix playback/capture concurrent usage
by Eldad Zack 01 Sep '13

01 Sep '13

Hi, This patch series attempts to fix a long standing problem with the concurrent usage of playback and capture on implicit feedback devices. Applies against current for-next (HEAD 68538bf2bce557c3b5fe8c59b034d45352500db1 ). v2: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-August/065649.html v1: http://mailman.alsa-project.org/pipermail/alsa-devel/2013-August/065517.html * Changed "altset_idx" to "altsetting" thanks to Clemens' clarification. #3: Thanks to Daniel's test, fixed a NULL dereference caused by userspace calling snd_pcm_open() and then snd_pcm_close() immediately. #8: subs->altset_idx was left intact, since it corresponds to the array index. #10: this patch now removed subs->altset_idx, but also changes the check to be performed against fmt->altsetting. #11: New. This changes the name of snd_usb_endpoint member alt_idx to altsetting for the same reason Clemens described. Clemens and Daniel - thank you very much for your help! Cheers, Eldad Eldad Zack (11): ALSA: usb-audio: remove unused parameter from sync_ep_set_params ALSA: usb-audio: remove deactivate_endpoints() ALSA: usb-audio: prevent NULL dereference on stop trigger ALSA: usb-audio: don't deactivate URBs on in-use EP ALSA: usb-audio: void return type of snd_usb_endpoint_deactivate() ALSA: usb-audio: clear SUBSTREAM_FLAG_SYNC_EP_STARTED on error ALSA: usb-audio: correct ep use_count semantics (add set_param flag) ALSA: usb-audio: conditional interface altsetting ALSA: usb-audio: conditional concurrent usage of endpoint ALSA: usb-audio: remove altset_idx from snd_usb_substream ALSA: usb-audio: rename alt_idx to altsetting sound/usb/card.h | 4 +- sound/usb/endpoint.c | 82 ++++++++++++++++++------- sound/usb/endpoint.h | 11 +++- sound/usb/pcm.c | 168 ++++++++++++++++++++++++++++++++++++++------------- sound/usb/pcm.h | 2 + sound/usb/proc.c | 4 +- 6 files changed, 201 insertions(+), 70 deletions(-) -- 1.8.1.5

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[alsa-devel] [PATCH] ASoC: fsl_spdif: Staticse non-exported symbols
by Mark Brown 01 Sep '13

01 Sep '13

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/fsl/fsl_spdif.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index e93dc0d..c5c228a 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -411,8 +411,8 @@ static int spdif_set_sample_rate(struct snd_pcm_substream *substream, return 0; } -int fsl_spdif_startup(struct snd_pcm_substream *substream, - struct snd_soc_dai *cpu_dai) +static int fsl_spdif_startup(struct snd_pcm_substream *substream, + struct snd_soc_dai *cpu_dai) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct fsl_spdif_priv *spdif_priv = snd_soc_dai_get_drvdata(rtd->cpu_dai); @@ -546,7 +546,7 @@ static int fsl_spdif_trigger(struct snd_pcm_substream *substream, return 0; } -struct snd_soc_dai_ops fsl_spdif_dai_ops = { +static struct snd_soc_dai_ops fsl_spdif_dai_ops = { .startup = fsl_spdif_startup, .hw_params = fsl_spdif_hw_params, .trigger = fsl_spdif_trigger, @@ -919,7 +919,7 @@ static int fsl_spdif_dai_probe(struct snd_soc_dai *dai) return 0; } -struct snd_soc_dai_driver fsl_spdif_dai = { +static struct snd_soc_dai_driver fsl_spdif_dai = { .probe = &fsl_spdif_dai_probe, .playback = { .channels_min = 2, -- 1.8.4.rc3

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[alsa-devel] [PATCH] ASoC: ssm2602: Fix cache sync
by Lars-Peter Clausen 31 Aug '13

31 Aug '13

The ssm2602 uses regmap for caching not soc-cache, so we need to use regcache_sync() instead of snd_soc_cache_sync(). Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ssm2602.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index f8d30e5..492644e 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -561,8 +561,9 @@ static int ssm2602_suspend(struct snd_soc_codec *codec) static int ssm2602_resume(struct snd_soc_codec *codec) { - snd_soc_cache_sync(codec); + struct ssm2602_priv *ssm2602 = snd_soc_codec_get_drvdata(codec); + regcache_sync(ssm2602->regmap); ssm2602_set_bias_level(codec, SND_SOC_BIAS_STANDBY); return 0; -- 1.8.0

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[alsa-devel] [PATCH] ASoC: cq93vc: Use table based control registration
by Mark Brown 31 Aug '13

31 Aug '13

From: Mark Brown <broonie(a)linaro.org> Saves a little code. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/cq93vc.c | 6 ++---- 1 file changed, 2 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/cq93vc.c b/sound/soc/codecs/cq93vc.c index 3ce535b..43737a27 100644 --- a/sound/soc/codecs/cq93vc.c +++ b/sound/soc/codecs/cq93vc.c @@ -142,10 +142,6 @@ static int cq93vc_probe(struct snd_soc_codec *codec) snd_soc_codec_set_cache_io(codec, 32, 32, SND_SOC_REGMAP); - /* Set controls */ - snd_soc_add_codec_controls(codec, cq93vc_snd_controls, - ARRAY_SIZE(cq93vc_snd_controls)); - /* Off, with power on */ cq93vc_set_bias_level(codec, SND_SOC_BIAS_STANDBY); @@ -164,6 +160,8 @@ static struct snd_soc_codec_driver soc_codec_dev_cq93vc = { .probe = cq93vc_probe, .remove = cq93vc_remove, .resume = cq93vc_resume, + .controls = cq93vc_snd_controls, + .num_controls = ARRAY_SIZE(cq93vc_snd_controls), }; static int cq93vc_platform_probe(struct platform_device *pdev) -- 1.8.4.rc3

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[alsa-devel] [PATCH 1/2] ASoC: wm8400: Use supplies to manage input power
by Mark Brown 31 Aug '13

31 Aug '13

From: Mark Brown <broonie(a)linaro.org> Rather than using a fake register to manage input power create some supply widgets and use those. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/wm8400.c | 73 ++++++++++++----------------------------------- 1 file changed, 18 insertions(+), 55 deletions(-) diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index d2a0928..95c33d1 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -32,13 +32,6 @@ #include "wm8400.h" -/* Fake register for internal state */ -#define WM8400_INTDRIVBITS (WM8400_REGISTER_COUNT + 1) -#define WM8400_INMIXL_PWR 0 -#define WM8400_AINLMUX_PWR 1 -#define WM8400_INMIXR_PWR 2 -#define WM8400_AINRMUX_PWR 3 - static struct regulator_bulk_data power[] = { { .supply = "I2S1VDD", @@ -79,10 +72,7 @@ static inline unsigned int wm8400_read(struct snd_soc_codec *codec, { struct wm8400_priv *wm8400 = snd_soc_codec_get_drvdata(codec); - if (reg == WM8400_INTDRIVBITS) - return wm8400->fake_register; - else - return wm8400_reg_read(wm8400->wm8400, reg); + return wm8400_reg_read(wm8400->wm8400, reg); } /* @@ -93,11 +83,7 @@ static int wm8400_write(struct snd_soc_codec *codec, unsigned int reg, { struct wm8400_priv *wm8400 = snd_soc_codec_get_drvdata(codec); - if (reg == WM8400_INTDRIVBITS) { - wm8400->fake_register = value; - return 0; - } else - return wm8400_set_bits(wm8400->wm8400, reg, 0xffff, value); + return wm8400_set_bits(wm8400->wm8400, reg, 0xffff, value); } static void wm8400_codec_reset(struct snd_soc_codec *codec) @@ -352,32 +338,6 @@ SOC_SINGLE("RIN34 Mute Switch", WM8400_RIGHT_LINE_INPUT_3_4_VOLUME, * _DAPM_ Controls */ -static int inmixer_event (struct snd_soc_dapm_widget *w, - struct snd_kcontrol *kcontrol, int event) -{ - u16 reg, fakepower; - - reg = snd_soc_read(w->codec, WM8400_POWER_MANAGEMENT_2); - fakepower = snd_soc_read(w->codec, WM8400_INTDRIVBITS); - - if (fakepower & ((1 << WM8400_INMIXL_PWR) | - (1 << WM8400_AINLMUX_PWR))) { - reg |= WM8400_AINL_ENA; - } else { - reg &= ~WM8400_AINL_ENA; - } - - if (fakepower & ((1 << WM8400_INMIXR_PWR) | - (1 << WM8400_AINRMUX_PWR))) { - reg |= WM8400_AINR_ENA; - } else { - reg &= ~WM8400_AINR_ENA; - } - snd_soc_write(w->codec, WM8400_POWER_MANAGEMENT_2, reg); - - return 0; -} - static int outmixer_event (struct snd_soc_dapm_widget *w, struct snd_kcontrol * kcontrol, int event) { @@ -658,27 +618,26 @@ SND_SOC_DAPM_MIXER("RIN34 PGA", WM8400_POWER_MANAGEMENT_2, 0, &wm8400_dapm_rin34_pga_controls[0], ARRAY_SIZE(wm8400_dapm_rin34_pga_controls)), +SND_SOC_DAPM_SUPPLY("INL", WM8400_POWER_MANAGEMENT_2, WM8400_AINL_ENA_SHIFT, + 0, NULL, 0), +SND_SOC_DAPM_SUPPLY("INR", WM8400_POWER_MANAGEMENT_2, WM8400_AINR_ENA_SHIFT, + 0, NULL, 0), + /* INMIXL */ -SND_SOC_DAPM_MIXER_E("INMIXL", WM8400_INTDRIVBITS, WM8400_INMIXL_PWR, 0, +SND_SOC_DAPM_MIXER("INMIXL", SND_SOC_NOPM, 0, 0, &wm8400_dapm_inmixl_controls[0], - ARRAY_SIZE(wm8400_dapm_inmixl_controls), - inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + ARRAY_SIZE(wm8400_dapm_inmixl_controls)), /* AINLMUX */ -SND_SOC_DAPM_MUX_E("AILNMUX", WM8400_INTDRIVBITS, WM8400_AINLMUX_PWR, 0, - &wm8400_dapm_ainlmux_controls, inmixer_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_MUX("AILNMUX", SND_SOC_NOPM, 0, 0, &wm8400_dapm_ainlmux_controls), /* INMIXR */ -SND_SOC_DAPM_MIXER_E("INMIXR", WM8400_INTDRIVBITS, WM8400_INMIXR_PWR, 0, +SND_SOC_DAPM_MIXER("INMIXR", SND_SOC_NOPM, 0, 0, &wm8400_dapm_inmixr_controls[0], - ARRAY_SIZE(wm8400_dapm_inmixr_controls), - inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), + ARRAY_SIZE(wm8400_dapm_inmixr_controls)), /* AINRMUX */ -SND_SOC_DAPM_MUX_E("AIRNMUX", WM8400_INTDRIVBITS, WM8400_AINRMUX_PWR, 0, - &wm8400_dapm_ainrmux_controls, inmixer_event, - SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), +SND_SOC_DAPM_MUX("AIRNMUX", SND_SOC_NOPM, 0, 0, &wm8400_dapm_ainrmux_controls), /* Output Side */ /* DACs */ @@ -789,11 +748,13 @@ static const struct snd_soc_dapm_route wm8400_dapm_routes[] = { {"LIN34 PGA", "LIN3 Switch", "LIN3"}, {"LIN34 PGA", "LIN4 Switch", "LIN4/RXN"}, /* INMIXL */ + {"INMIXL", NULL, "INL"}, {"INMIXL", "Record Left Volume", "LOMIX"}, {"INMIXL", "LIN2 Volume", "LIN2"}, {"INMIXL", "LINPGA12 Switch", "LIN12 PGA"}, {"INMIXL", "LINPGA34 Switch", "LIN34 PGA"}, /* AILNMUX */ + {"AILNMUX", NULL, "INL"}, {"AILNMUX", "INMIXL Mix", "INMIXL"}, {"AILNMUX", "DIFFINL Mix", "LIN12 PGA"}, {"AILNMUX", "DIFFINL Mix", "LIN34 PGA"}, @@ -808,12 +769,14 @@ static const struct snd_soc_dapm_route wm8400_dapm_routes[] = { /* RIN34 PGA */ {"RIN34 PGA", "RIN3 Switch", "RIN3"}, {"RIN34 PGA", "RIN4 Switch", "RIN4/RXP"}, - /* INMIXL */ + /* INMIXR */ + {"INMIXR", NULL, "INR"}, {"INMIXR", "Record Right Volume", "ROMIX"}, {"INMIXR", "RIN2 Volume", "RIN2"}, {"INMIXR", "RINPGA12 Switch", "RIN12 PGA"}, {"INMIXR", "RINPGA34 Switch", "RIN34 PGA"}, /* AIRNMUX */ + {"AIRNMUX", NULL, "INR"}, {"AIRNMUX", "INMIXR Mix", "INMIXR"}, {"AIRNMUX", "DIFFINR Mix", "RIN12 PGA"}, {"AIRNMUX", "DIFFINR Mix", "RIN34 PGA"}, -- 1.8.4.rc3

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[alsa-devel] [PATCH 1/3] ASoC: Remove unused control_type field from snd_soc_codec struct
by Lars-Peter Clausen 31 Aug '13

31 Aug '13

The control_type field was used by the core to track which raw IO methods to use, but when switching to regmap this was no longer necessary and so the last user of the field was removed in commit be3ea3b9 ("ASoC: Use new register map API for ASoC generic physical I/O"). The field is now completely unused and can be removed. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- include/sound/soc.h | 1 - 1 file changed, 1 deletion(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 8e2ad52..9d4c2ab 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -705,7 +705,6 @@ struct snd_soc_codec { /* codec IO */ void *control_data; /* codec control (i2c/3wire) data */ - enum snd_soc_control_type control_type; hw_write_t hw_write; unsigned int (*hw_read)(struct snd_soc_codec *, unsigned int); unsigned int (*read)(struct snd_soc_codec *, unsigned int); -- 1.8.0

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[alsa-devel] [PATCH v11 0/7] ASoC: fsl-ssi: ac97-slave support
by Markus Pargmann 31 Aug '13

31 Aug '13

Hi, this series implements ac97-slave support for fsl-ssi and adds a simple generic DT sound card. Regards, Markus Changes in v11: - Dropped phycard specific sound patches - Added a generic simple DT sound card with support for ac97 using ac97-codec - Some new bugfix patches Markus Pargmann (5): ASoC: imx-pcm-dma: Add missing include ASoC: core: Generic ac97 link reset functions ASoC: generic: simple DT sound card ASoC: fsl: Kconfig, visible config items for simple card ASoC: fsl-ssi: ac97-slave support Michael Grzeschik (2): ASoC: fsl-ssi: imx-pcm-fiq bugfix ASoC: fsl: disable ssi irq for imx Steffen Trumtrar (1): ASoC: fsl-ssi: add SSIEN errata work around .../devicetree/bindings/sound/fsl,ssi.txt | 4 + .../bindings/sound/generic,simple-dt-card.txt | 28 ++ .../devicetree/bindings/sound/soc-ac97link.txt | 28 ++ include/sound/soc.h | 2 + sound/soc/fsl/Kconfig | 23 +- sound/soc/fsl/fsl_ssi.c | 369 ++++++++++++++++----- sound/soc/fsl/imx-pcm-dma.c | 1 + sound/soc/generic/Kconfig | 1 + sound/soc/generic/Makefile | 3 +- sound/soc/generic/simple-dt-card.c | 186 +++++++++++ sound/soc/soc-core.c | 153 +++++++++ 11 files changed, 711 insertions(+), 87 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/generic,simple-dt-card.txt create mode 100644 Documentation/devicetree/bindings/sound/soc-ac97link.txt create mode 100644 sound/soc/generic/simple-dt-card.c

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[alsa-devel] [PATCH 1/2] ASoC: fsl: Add wrapping for dev_dbg() in fsl_spdif.c
by Nicolin Chen 30 Aug '13

30 Aug '13

Add wrapping '\n' for dev_dbg() in fsl_spdif.c Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- sound/soc/fsl/fsl_spdif.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index e93dc0d..98741e9 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -1071,9 +1071,9 @@ static int fsl_spdif_probe_txclk(struct fsl_spdif_priv *spdif_priv, break; } - dev_dbg(&pdev->dev, "use rxtx%d as tx clock source for %dHz sample rate", + dev_dbg(&pdev->dev, "use rxtx%d as tx clock source for %dHz sample rate\n", spdif_priv->txclk_src[index], rate[index]); - dev_dbg(&pdev->dev, "use divisor %d for %dHz sample rate", + dev_dbg(&pdev->dev, "use divisor %d for %dHz sample rate\n", spdif_priv->txclk_div[index], rate[index]); return 0; -- 1.7.1

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