January 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 0/4] ASoC: tlv320aic32x4: DT support
by Markus Pargmann 17 Jan '14

17 Jan '14

Hi, Here are some more updates for the tlv320aic32x4 codec. This series fixes the mixer controls to operate on the full range instead of only the positive range. This depends on my other series: "ASoC: core: volume control using signed register values" Patch 3 adds DT support. Patch 4 adds support to control the master clock connected to the codec. Regards, Markus Markus Pargmann (4): ASoC: tlv320aic32x4: Use signed int mixer controls ASoC: tlv320aic32x4: Use gpio_is_valid ASoC: tlv320aic32x4: DT support ASoC: tlv320aic32x4: Support for master clock .../devicetree/bindings/sound/tlv320aic32x4.txt | 25 +++++++ sound/soc/codecs/tlv320aic32x4.c | 83 ++++++++++++++++++---- 2 files changed, 96 insertions(+), 12 deletions(-) create mode 100644 Documentation/devicetree/bindings/sound/tlv320aic32x4.txt -- 1.8.5.2

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[alsa-devel] [PATCH v2 0/2] Dynamic routing for compressed streams.
by Liam Girdwood 17 Jan '14

17 Jan '14

Changes for V2 * We no longer copy the compressed params into the BE hw_params. This was bogus, but didn't show up on Haswell testing as the BE params were never used. The DSP ran the BE is a fixed configuration so all BE params were ignored. * The BE hw_params are now zeroed to force the machine driver to fixup the BE configuration. i.e. machine driver can query the DSP encodec/decoder for correct hw_params given the use case. * No longer need to allocate memory for hw_params. Regards Liam

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[alsa-devel] [PATCH v2 1/2] ASoC: DPCM: make some DPCM API calls non static for compressed usage
by Liam Girdwood 17 Jan '14

17 Jan '14

The ASoC compressed code needs to call the internal DPCM APIs in order to dynamically route compressed data to different DAIs. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/sound/soc-dpcm.h | 22 ++++++++++++++++++++++ sound/soc/soc-pcm.c | 29 ++++++++++++----------------- 2 files changed, 34 insertions(+), 17 deletions(-) diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index 047d657..2883a7a 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -11,6 +11,7 @@ #ifndef __LINUX_SND_SOC_DPCM_H #define __LINUX_SND_SOC_DPCM_H +#include <linux/slab.h> #include <linux/list.h> #include <sound/pcm.h> @@ -135,4 +136,25 @@ int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute); int soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd); int soc_dpcm_runtime_update(struct snd_soc_card *); +int dpcm_path_get(struct snd_soc_pcm_runtime *fe, + int stream, struct snd_soc_dapm_widget_list **list_); +int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, + int stream, struct snd_soc_dapm_widget_list **list, int new); +int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream); +int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream); +void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream); +void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream); +int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream); +int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int tream); +int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd); +int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream); +int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, + int event); + +static inline void dpcm_path_put(struct snd_soc_dapm_widget_list **list) +{ + kfree(*list); +} + + #endif diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 1534115..7023f7e 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -58,7 +58,7 @@ int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams); /* DPCM stream event, send event to FE and all active BEs. */ -static int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, +int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, int event) { struct snd_soc_dpcm *dpcm; @@ -877,7 +877,7 @@ static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe, } /* disconnect a BE and FE */ -static void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) +void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm, *d; @@ -973,7 +973,7 @@ static int widget_in_list(struct snd_soc_dapm_widget_list *list, return 0; } -static int dpcm_path_get(struct snd_soc_pcm_runtime *fe, +int dpcm_path_get(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dapm_widget_list **list_) { struct snd_soc_dai *cpu_dai = fe->cpu_dai; @@ -995,11 +995,6 @@ static int dpcm_path_get(struct snd_soc_pcm_runtime *fe, return paths; } -static inline void dpcm_path_put(struct snd_soc_dapm_widget_list **list) -{ - kfree(*list); -} - static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dapm_widget_list **list_) { @@ -1069,7 +1064,7 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, continue; /* don't connect if FE is not running */ - if (!fe->dpcm[stream].runtime) + if (!fe->dpcm[stream].runtime && !fe->fe_compr) continue; /* newly connected FE and BE */ @@ -1094,7 +1089,7 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, * Find the corresponding BE DAIs that source or sink audio to this * FE substream. */ -static int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, +int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dapm_widget_list **list, int new) { if (new) @@ -1103,7 +1098,7 @@ static int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, return dpcm_prune_paths(fe, stream, list); } -static void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream) +void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; @@ -1141,7 +1136,7 @@ static void dpcm_be_dai_startup_unwind(struct snd_soc_pcm_runtime *fe, } } -static int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; int err, count = 0; @@ -1293,7 +1288,7 @@ be_err: return ret; } -static int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; @@ -1354,7 +1349,7 @@ static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream) return 0; } -static int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; @@ -1419,7 +1414,7 @@ static int dpcm_fe_dai_hw_free(struct snd_pcm_substream *substream) return 0; } -static int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; int ret; @@ -1549,7 +1544,7 @@ static int dpcm_do_trigger(struct snd_soc_dpcm *dpcm, return ret; } -static int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, +int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd) { struct snd_soc_dpcm *dpcm; @@ -1717,7 +1712,7 @@ out: return ret; } -static int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; int ret = 0; -- 1.8.3.2

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[alsa-devel] ProVideo PV 918A - audi o device created but cannot capture sound
by siddharth 17 Jan '14

17 Jan '14

Hi, I have installed a ProVideo PV 981. I was able to get the sound device detected by putting modprobe snd-bt87x load_all=1 digital_rate=48000 enable=1 but am unable to capture sound. I am using Fedora 19 on an intel i5 machine. In dmesg i get the error: [ 235.145932] ALSA sound/pci/bt87x.c:860 unknown card 0x878-0xaa00:0x1460 [ 235.145935] ALSA sound/pci/bt87x.c:863 please mail id, board name, and, if it works, the correct digital_rate option to <alsa-devel(a)alsa-project.org> [ 235.146061] ALSA sound/pci/bt87x.c:931 bt87x0: Using board 0, analog, digital (rate 48000 Hz) Any help will be appreciated. regards Siddharth

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Re: [alsa-devel] [pulseaudio-discuss] Need help with audio gain on an Intel HDA equipt ASUS M2N-SLI Deluxe
by Raymond Yau 17 Jan '14

17 Jan '14

2014/1/16 Axel Braun <axel.braun(a)gmx.de> > Am Donnerstag, 16. Januar 2014, 16:23:19 schrieb Raymond Yau: > > Any hint you could give me? > > > > post the output of alsa-info.sh , there is a bug in alsa-driver which > treat > > dock headphone and headphone to support surround 4.0 > > /sys/class/sound/hwC0D0/init_pin_configs: 0x19 0x04211040 0x1a 0x61a19050 0x1b 0x04a11060 0x1c 0x6121401f 0x1d 0x40f001f0 0x1e 0x40f001f0 0x1f 0x90170110 0x20 0x40f001f0 0x22 0x40f001f0 0x23 0x90a60170 /sys/class/sound/hwC0D0/driver_pin_configs: 0x19 0x042110ff 0x1a 0x21a190f0 0x1c 0x212140ff https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… it was using auto but now the driver modify the dock headphone , dock mic and headphone pin default. there are also changes in the thinkpad_acpi module too. https://git.kernel.org/cgit/linux/kernel/git/tiwai/sound.git/commit/sound/p… do this thinkpad support headset or headphone ? it is strange that speaker volume is at max but amp out vals at node 0x11 is 0x3d instead of 0x4a (0dB) Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 74 Mono: Playback 61 [82%] [-13.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 74 Mono: Front Left: Playback 0 [0%] [-74.00dB] [off] Front Right: Playback 0 [0%] [-74.00dB] [off] Simple mixer control 'Speaker',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 74 Mono: Front Left: Playback 74 [100%] [0.00dB] [on] Front Right: Playback 74 [100%] [0.00dB] [on] Node 0x10 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Control: name="Speaker Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Speaker Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Device: name="CX20590 Analog", type="Audio", device=0 Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-Out vals: [0x3d 0x3d] Converter: stream=8, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0 Node 0x11 [Audio Output] wcaps 0xc1d: Stereo Amp-Out R/L Control: name="Headphone Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Control: name="Headphone Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-Out caps: ofs=0x4a, nsteps=0x4a, stepsize=0x03, mute=1 Amp-Out vals: [0x80 0x80] Converter: stream=8, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Power states: D0 D1 D2 D3 D3cold EPSS Power: setting=D0, actual=D0

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[alsa-devel] [PATCH 1/2] ASoC: DPCM: make some DPCM API calls non static for compressed usage
by Liam Girdwood 17 Jan '14

17 Jan '14

The ASoC compressed code needs to call the internal DPCM APIs in order to dynamically route compressed data to different DAIs. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/sound/soc-dpcm.h | 22 ++++++++++++++++++++++ sound/soc/soc-pcm.c | 29 ++++++++++++----------------- 2 files changed, 34 insertions(+), 17 deletions(-) diff --git a/include/sound/soc-dpcm.h b/include/sound/soc-dpcm.h index 047d657..2883a7a 100644 --- a/include/sound/soc-dpcm.h +++ b/include/sound/soc-dpcm.h @@ -11,6 +11,7 @@ #ifndef __LINUX_SND_SOC_DPCM_H #define __LINUX_SND_SOC_DPCM_H +#include <linux/slab.h> #include <linux/list.h> #include <sound/pcm.h> @@ -135,4 +136,25 @@ int soc_dpcm_be_digital_mute(struct snd_soc_pcm_runtime *fe, int mute); int soc_dpcm_debugfs_add(struct snd_soc_pcm_runtime *rtd); int soc_dpcm_runtime_update(struct snd_soc_card *); +int dpcm_path_get(struct snd_soc_pcm_runtime *fe, + int stream, struct snd_soc_dapm_widget_list **list_); +int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, + int stream, struct snd_soc_dapm_widget_list **list, int new); +int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream); +int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream); +void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream); +void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream); +int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream); +int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int tream); +int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd); +int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream); +int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, + int event); + +static inline void dpcm_path_put(struct snd_soc_dapm_widget_list **list) +{ + kfree(*list); +} + + #endif diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 1534115..7023f7e 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -58,7 +58,7 @@ int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream, EXPORT_SYMBOL_GPL(snd_soc_set_runtime_hwparams); /* DPCM stream event, send event to FE and all active BEs. */ -static int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, +int dpcm_dapm_stream_event(struct snd_soc_pcm_runtime *fe, int dir, int event) { struct snd_soc_dpcm *dpcm; @@ -877,7 +877,7 @@ static void dpcm_be_reparent(struct snd_soc_pcm_runtime *fe, } /* disconnect a BE and FE */ -static void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) +void dpcm_be_disconnect(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm, *d; @@ -973,7 +973,7 @@ static int widget_in_list(struct snd_soc_dapm_widget_list *list, return 0; } -static int dpcm_path_get(struct snd_soc_pcm_runtime *fe, +int dpcm_path_get(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dapm_widget_list **list_) { struct snd_soc_dai *cpu_dai = fe->cpu_dai; @@ -995,11 +995,6 @@ static int dpcm_path_get(struct snd_soc_pcm_runtime *fe, return paths; } -static inline void dpcm_path_put(struct snd_soc_dapm_widget_list **list) -{ - kfree(*list); -} - static int dpcm_prune_paths(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dapm_widget_list **list_) { @@ -1069,7 +1064,7 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, continue; /* don't connect if FE is not running */ - if (!fe->dpcm[stream].runtime) + if (!fe->dpcm[stream].runtime && !fe->fe_compr) continue; /* newly connected FE and BE */ @@ -1094,7 +1089,7 @@ static int dpcm_add_paths(struct snd_soc_pcm_runtime *fe, int stream, * Find the corresponding BE DAIs that source or sink audio to this * FE substream. */ -static int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, +int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, int stream, struct snd_soc_dapm_widget_list **list, int new) { if (new) @@ -1103,7 +1098,7 @@ static int dpcm_process_paths(struct snd_soc_pcm_runtime *fe, return dpcm_prune_paths(fe, stream, list); } -static void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream) +void dpcm_clear_pending_state(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; @@ -1141,7 +1136,7 @@ static void dpcm_be_dai_startup_unwind(struct snd_soc_pcm_runtime *fe, } } -static int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_startup(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; int err, count = 0; @@ -1293,7 +1288,7 @@ be_err: return ret; } -static int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_shutdown(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; @@ -1354,7 +1349,7 @@ static int dpcm_fe_dai_shutdown(struct snd_pcm_substream *substream) return 0; } -static int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_hw_free(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; @@ -1419,7 +1414,7 @@ static int dpcm_fe_dai_hw_free(struct snd_pcm_substream *substream) return 0; } -static int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_hw_params(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; int ret; @@ -1549,7 +1544,7 @@ static int dpcm_do_trigger(struct snd_soc_dpcm *dpcm, return ret; } -static int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, +int dpcm_be_dai_trigger(struct snd_soc_pcm_runtime *fe, int stream, int cmd) { struct snd_soc_dpcm *dpcm; @@ -1717,7 +1712,7 @@ out: return ret; } -static int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) +int dpcm_be_dai_prepare(struct snd_soc_pcm_runtime *fe, int stream) { struct snd_soc_dpcm *dpcm; int ret = 0; -- 1.8.3.2

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[alsa-devel] [PATCH v4 0/9] ASoC: fsl-ssi: offline/online configuration and cleanups
by Markus Pargmann 17 Jan '14

17 Jan '14

Hi, This patch series fixes some fsl-ssi code that does not act exactly as it is described in the reference manuals. The reference manuals before imx51 do mention that some register bits, including TDMAE/RDMAE, should not be changed while the SSI unit is enabled (SSIEN). At the same time the SDMA unit has a undefined request handling if there are DMA requests before the SDMA engine and its channel is configured. The reference manual states that SSI DMA requests are sent as soon as the FIFO and the DMAE bits are enabled. It does not mention a dependency to SCR's TE/RE bit. To avoid sending DMA requests before a channel is configured, we have to seperate the fsl-ssi behavior into offline and online configuration. SoCs before imx51 have to use offline configuration while later SoC versions can use online reconfiguration. This series adds support for online configuration and cleans up the configuration register code especially in fsl_ssi_trigger. At the end we have a seperation between enable/disable logic and the actual configuration register values used to enable/disable TX/RX. v4 is another rebase onto latest topic/fsl branch. It includes the DAI master mode changes from Nicolin. @Nicolin: Could you please test this series with your hardware to verify that your patch still works? I tested this series on mx53. The series is based on Mark Brown's topic/fsl branch. Regards, Markus Changes in v4: - Support for irqstats on all imx SoCs - Fix offline_config and use switch statement - Introduce ssi_private->irq_stats bool Changes in v3: - Rebase on latest topic/fsl - Added patch to cleanup probe function error handling - Fix interrupt stats for imx53 - Replace imx50-ssi by imx51-ssi Markus Pargmann (9): ASoC: fsl-ssi: Fix probe error handling ASoC: fsl-ssi: Move sysfs stats to debugfs ASoC: fsl-ssi: Add imx51-ssi and of_device_id matching ASoC: fsl-ssi: Fix interrupt stats for imx ASoC: fsl-ssi: Add offline_config flag ASoC: fsl-ssi: Add configuration helper functions ASoC: fsl-ssi: Move RX/TX configuration to seperate functions ASoC: fsl-ssi: Drop ac97 specific trigger function ARM: DTS: imx5* imx6*, use imx51-ssi arch/arm/boot/dts/imx53.dtsi | 10 +- arch/arm/boot/dts/imx6qdl.dtsi | 12 +- arch/arm/boot/dts/imx6sl.dtsi | 12 +- sound/soc/fsl/fsl_ssi.c | 567 ++++++++++++++++++++++++++++------------- 4 files changed, 411 insertions(+), 190 deletions(-) -- 1.8.5.1

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[alsa-devel] [PATCH] ASoC: core: Fix possible NULL pointer dereference of pcm->config
by Xiubo Li 16 Jan '14

16 Jan '14

Since the soc generic dmaengine pcm driver allows using the defualt settings, so the pcm->config maybe NULL. Signed-off-by: Xiubo Li <Li.Xiubo(a)freescale.com> --- sound/soc/soc-generic-dmaengine-pcm.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 560a778..5bace12 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -176,17 +176,20 @@ static struct dma_chan *dmaengine_pcm_compat_request_channel( { struct dmaengine_pcm *pcm = soc_platform_to_pcm(rtd->platform); struct snd_dmaengine_dai_dma_data *dma_data; + dma_filter_fn fn = NULL; dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); if ((pcm->flags & SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX) && pcm->chan[0]) return pcm->chan[0]; - if (pcm->config->compat_request_channel) + if (pcm->config && pcm->config->compat_request_channel) return pcm->config->compat_request_channel(rtd, substream); - return snd_dmaengine_pcm_request_channel(pcm->config->compat_filter_fn, - dma_data->filter_data); + if (pcm->config) + fn = pcm->config->compat_filter_fn; + + return snd_dmaengine_pcm_request_channel(fn, dma_data->filter_data); } static bool dmaengine_pcm_can_report_residue(struct dma_chan *chan) -- 1.8.4

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[alsa-devel] ASoC fixes for v3.13
by Mark Brown 16 Jan '14

16 Jan '14

The following changes since commit 7e22e91102c6b9df7c4ae2168910e19d2bb14cd6: Linux 3.13-rc8 (2014-01-12 17:04:18 +0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.13-rc8 for you to fetch changes up to c5281227c90578c7b7beffca56e8c3a68d4607de: Merge remote-tracking branches 'asoc/fix/adau1701', 'asoc/fix/fsl' and 'asoc/fix/tlv320aic32x4' into asoc-linus (2014-01-15 23:17:15 +0000) ---------------------------------------------------------------- ASoC: Fixes for v3.13 A few small fixes in drivers, nothing too remarkable here but all good to have - mainly these are fixes for things that were introduced in the last merge window but only just got useful testing. ---------------------------------------------------------------- Lars-Peter Clausen (1): ASoC: adau1701: Fix ADAU1701_SEROCTL_WORD_LEN_16 constant Mark Brown (1): Merge remote-tracking branches 'asoc/fix/adau1701', 'asoc/fix/fsl' and 'asoc/fix/tlv320aic32x4' into asoc-linus Markus Pargmann (2): ASoC: codec: tlv320aic32x4: Fix regmap range config ASoC: tlv320aic32x4: Fix regmap range_min Nicolin Chen (1): ASoC: imx-spdif: Use snd-soc-dummy CODEC driver to link card sound/soc/codecs/adau1701.c | 2 +- sound/soc/codecs/tlv320aic32x4.c | 4 +- sound/soc/fsl/Kconfig | 1 - sound/soc/fsl/imx-spdif.c | 84 ++++++++++------------------------------ 4 files changed, 24 insertions(+), 67 deletions(-)

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[alsa-devel] [GIT PULL] ASoC updates for v3.13
by Mark Brown 16 Jan '14

16 Jan '14

The following changes since commit 7e22e91102c6b9df7c4ae2168910e19d2bb14cd6: Linux 3.13-rc8 (2014-01-12 17:04:18 +0700) are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-v3.13-rc8-2 for you to fetch changes up to c6affc0dba45bb2074ff56b21386d22eeaf682f8: Merge remote-tracking branches 'asoc/fix/adau1701' and 'asoc/fix/tlv320aic32x4' into asoc-linus (2014-01-16 14:02:52 +0000) ---------------------------------------------------------------- ASoC: Fixes for v3.13 A few small fixes in drivers, nothing too remarkable here but all good to have - mainly these are fixes for things that were introduced in the last merge window but only just got useful testing. ---------------------------------------------------------------- Lars-Peter Clausen (1): ASoC: adau1701: Fix ADAU1701_SEROCTL_WORD_LEN_16 constant Mark Brown (1): Merge remote-tracking branches 'asoc/fix/adau1701' and 'asoc/fix/tlv320aic32x4' into asoc-linus Markus Pargmann (2): ASoC: codec: tlv320aic32x4: Fix regmap range config ASoC: tlv320aic32x4: Fix regmap range_min sound/soc/codecs/adau1701.c | 2 +- sound/soc/codecs/tlv320aic32x4.c | 4 ++-- 2 files changed, 3 insertions(+), 3 deletions(-)

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