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

[alsa-devel] [PATCH] ASoC: Intel: update stream only on stream IPC msgs
by Jie Yang 01 Aug '14

01 Aug '14

From: Paweł Piskorski <pawel.piskorski(a)intel.com> Only update the stream when the IPC message type matches stream type. Signed-off-by: Paweł Piskorski <pawel.piskorski(a)intel.com> Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-haswell-ipc.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 0fec686..da27f90 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -780,7 +780,8 @@ static int hsw_process_reply(struct sst_hsw *hsw, u32 header) } /* update any stream states */ - hsw_stream_update(hsw, msg); + if (msg_get_global_type(header) == IPC_GLB_STREAM_MESSAGE) + hsw_stream_update(hsw, msg); /* wake up and return the error if we have waiters on this message ? */ list_del(&msg->list); -- 1.8.3.2

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[alsa-devel] [PATCH] ASoC: Intel: Don't issue ipc when processing response
by Jie Yang 01 Aug '14

01 Aug '14

From: Paweł Piskorski <pawel.piskorski(a)intel.com> Make sure we dont issue IPC when we are processing a response. Signed-off-by: Paweł Piskorski <pawel.piskorski(a)intel.com> Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-haswell-ipc.c | 5 +++-- 1 file changed, 3 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 5a607ab..0fec686 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -457,9 +457,10 @@ static void ipc_tx_msgs(struct kthread_work *work) return; } - /* if the DSP is busy we will TX messages after IRQ */ + /* if the DSP is busy, we will TX messages after IRQ. + * also postpone if we are in the middle of procesing completion irq*/ ipcx = sst_dsp_shim_read_unlocked(hsw->dsp, SST_IPCX); - if (ipcx & SST_IPCX_BUSY) { + if (ipcx & (SST_IPCX_BUSY | SST_IPCX_DONE)) { spin_unlock_irqrestore(&hsw->dsp->spinlock, flags); return; } -- 1.8.3.2

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[alsa-devel] [PATCH] ASoC: Intel: Delete message when IPC timeout occurs
by Jie Yang 01 Aug '14

01 Aug '14

From: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> This fixes a bug where we dont delete the current message when an IPC message timeout occurs. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/sst-haswell-ipc.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 96373ab..5a607ab 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -502,6 +502,7 @@ static int tx_wait_done(struct sst_hsw *hsw, struct ipc_message *msg, ipc_shim_dbg(hsw, "message timeout"); trace_ipc_error("error message timeout for", msg->header); + list_del(&msg->list); ret = -ETIMEDOUT; } else { -- 1.8.3.2

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[alsa-devel] [PATCH] ASoC: Intel: Add dependency to DW_DMAC for BDW platform
by Jie Yang 01 Aug '14

01 Aug '14

Add dependency to DW_DMAC for broadwell machine, which have built in DW dma engines. Signed-off-by: Jie Yang <yang.jie(a)intel.com> --- sound/soc/intel/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/Kconfig b/sound/soc/intel/Kconfig index 0b305f9..f5b4a9c7 100644 --- a/sound/soc/intel/Kconfig +++ b/sound/soc/intel/Kconfig @@ -61,7 +61,7 @@ config SND_SOC_INTEL_BYT_MAX98090_MACH config SND_SOC_INTEL_BROADWELL_MACH tristate "ASoC Audio DSP support for Intel Broadwell Wildcatpoint" - depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS + depends on SND_SOC_INTEL_SST && X86_INTEL_LPSS && DW_DMAC select SND_SOC_INTEL_HASWELL select SND_COMPRESS_OFFLOAD select SND_SOC_RT286 -- 1.8.3.2

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[alsa-devel] [PATCH 0/2] ASoC: rsnd: adds Mute support
by Kuninori Morimoto 01 Aug '14

01 Aug '14

Hi Mark These patch adds Mute support to Renesas R-Car sound Kuninori Morimoto (2): ASoC: rsnd: tidyup DVC control method ASoC: rsnd: enable Mute control on DVC sound/soc/sh/rcar/dvc.c | 93 ++++++++++++++++++++++++++++++++--------------- 1 file changed, 63 insertions(+), 30 deletions(-) Best regards --- Kuninori Morimoto

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[alsa-devel] [PATCH] ASoC: imx-audmux: Use uintptr_t for port numbers
by Mark Brown 01 Aug '14

01 Aug '14

From: Mark Brown <broonie(a)linaro.org> Since we pass the port number through file private data for debugfs we cast it to and from a pointer so use uintptr_t in order to ensure that the types are compatible, avoiding warnings on 64 bit platforms where pointers are 64 bit and unsigned integers 32 bit. Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/fsl/imx-audmux.c | 8 ++++---- 1 file changed, 4 insertions(+), 4 deletions(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index 267717a..46f9beb 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -67,7 +67,7 @@ static ssize_t audmux_read_file(struct file *file, char __user *user_buf, { ssize_t ret; char *buf; - int port = (int)file->private_data; + uintptr_t port = (uintptr_t)file->private_data; u32 pdcr, ptcr; if (audmux_clk) { @@ -147,7 +147,7 @@ static const struct file_operations audmux_debugfs_fops = { static void audmux_debugfs_init(void) { - int i; + uintptr_t i; char buf[20]; audmux_debugfs_root = debugfs_create_dir("audmux", NULL); @@ -157,10 +157,10 @@ static void audmux_debugfs_init(void) } for (i = 0; i < MX31_AUDMUX_PORT7_SSI_PINS_7 + 1; i++) { - snprintf(buf, sizeof(buf), "ssi%d", i); + snprintf(buf, sizeof(buf), "ssi%lu", i); if (!debugfs_create_file(buf, 0444, audmux_debugfs_root, (void *)i, &audmux_debugfs_fops)) - pr_warning("Failed to create AUDMUX port %d debugfs file\n", + pr_warning("Failed to create AUDMUX port %lu debugfs file\n", i); } } -- 2.0.0

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[alsa-devel] [PATCH 1/2] ASoC: rt5670: Staticise non-exported symbols
by Mark Brown 01 Aug '14

01 Aug '14

From: Mark Brown <broonie(a)linaro.org> Signed-off-by: Mark Brown <broonie(a)linaro.org> --- sound/soc/codecs/rt5670.c | 6 +++--- 1 file changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 879d42e..31f664f 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2407,7 +2407,7 @@ static int rt5670_resume(struct snd_soc_codec *codec) #define RT5670_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) -struct snd_soc_dai_ops rt5670_aif_dai_ops = { +static struct snd_soc_dai_ops rt5670_aif_dai_ops = { .hw_params = rt5670_hw_params, .set_fmt = rt5670_set_dai_fmt, .set_sysclk = rt5670_set_dai_sysclk, @@ -2415,7 +2415,7 @@ struct snd_soc_dai_ops rt5670_aif_dai_ops = { .set_pll = rt5670_set_dai_pll, }; -struct snd_soc_dai_driver rt5670_dai[] = { +static struct snd_soc_dai_driver rt5670_dai[] = { { .name = "rt5670-aif1", .id = RT5670_AIF1, @@ -2675,7 +2675,7 @@ static int rt5670_i2c_remove(struct i2c_client *i2c) return 0; } -struct i2c_driver rt5670_i2c_driver = { +static struct i2c_driver rt5670_i2c_driver = { .driver = { .name = "rt5670", .owner = THIS_MODULE, -- 2.0.0

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[alsa-devel] [PATCH] ARM: sun7i: Fix HDMI Audio driver's Alsa interface and do 8CH IEC60958 output
by thomas schorpp 01 Aug '14

01 Aug '14

First shot. HDMI backend seems to work with speaker-test but stream samples wrong order /mapping issues and the Alsa frontend is a stub, Pulseaudio needs more, still defaults to 2CH and "Analog". Please test if it switches Your receivers correctly into multichannel pcm mode, strange people with those overexpensed high end 7.1 receivers ;-) Signed-off-by: Not Yet ----------- https://github.com/linux-sunxi/linux-sunxi * stage/sunxi-3.4 diff --git a/drivers/video/sunxi/hdmi/drv_hdmi.c b/drivers/video/sunxi/hdmi/drv_hdmi.c index 283d2ae..9cf2cc2 100644 --- a/drivers/video/sunxi/hdmi/drv_hdmi.c +++ b/drivers/video/sunxi/hdmi/drv_hdmi.c @@ -314,7 +314,7 @@ __s32 Hdmi_init(struct platform_device *dev) if (err) return err; - audio_info.channel_num = 2; + audio_info.channel_num = 8; #if 0 { /* for audio test */ hdmi_audio_t audio_para; diff --git a/sound/soc/sunxi/hdmiaudio/sndhdmi.c b/sound/soc/sunxi/hdmiaudio/sndhdmi.c index 1c306d2..4563010 100644 --- a/sound/soc/sunxi/hdmiaudio/sndhdmi.c +++ b/sound/soc/sunxi/hdmiaudio/sndhdmi.c @@ -85,7 +85,9 @@ static int sndhdmi_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - if (4 == hdmi_para.channel_num) + if (2 < hdmi_para.channel_num) + hdmi_para.channel_num = 8; + if (2 > hdmi_para.channel_num) hdmi_para.channel_num = 2; g_hdmi_func.hdmi_set_audio_para(&hdmi_para); @@ -129,11 +131,11 @@ struct snd_soc_dai_ops sndhdmi_dai_ops = { /* codec dai */ struct snd_soc_dai_driver sndhdmi_dai = { .name = "sndhdmi", - /* playback capabilities */ + /* playback capabilities, hdmi_core::audio_config() claims 8ch cap */ .playback = { .stream_name = "Playback", - .channels_min = 1, - .channels_max = 2, + .channels_min = 2, + .channels_max = 8, .rates = SNDHDMI_RATES, .formats = SNDHDMI_FORMATS, }, diff --git a/sound/soc/sunxi/hdmiaudio/sunxi-hdmiaudio.c b/sound/soc/sunxi/hdmiaudio/sunxi-hdmiaudio.c index 12da1a6..80dd3b9 100644 --- a/sound/soc/sunxi/hdmiaudio/sunxi-hdmiaudio.c +++ b/sound/soc/sunxi/hdmiaudio/sunxi-hdmiaudio.c @@ -34,8 +34,8 @@ #include <plat/sys_config.h> #include <plat/dma_compat.h> -static struct sunxi_dma_params sunxi_hdmiaudio_pcm_stereo_out = { - .client.name = "HDMIAUDIO PCM Stereo out", +static struct sunxi_dma_params sunxi_hdmiaudio_pcm_iec60958_out = { + .client.name = "HDMIAUDIO PCM IEC60958 out", #if defined CONFIG_ARCH_SUN4I || defined CONFIG_ARCH_SUN5I .channel = DMACH_HDMIAUDIO, #endif @@ -67,7 +67,7 @@ static int sunxi_hdmiaudio_hw_params(struct snd_pcm_substream *substream, rtd = substream->private_data; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &sunxi_hdmiaudio_pcm_stereo_out; + dma_data = &sunxi_hdmiaudio_pcm_iec60958_out; else printk("error:hdmiaudio can't support capture:%s,line:%d\n", __func__, __LINE__); @@ -182,8 +182,8 @@ static struct snd_soc_dai_driver sunxi_hdmiaudio_dai = { .resume = sunxi_hdmiaudio_resume, .remove = sunxi_hdmiaudio_dai_remove, .playback = { - .channels_min = 1, - .channels_max = 2, + .channels_min = 2, + .channels_max = 8, .rates = SUNXI_HDMI_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, }, diff --git a/sound/soc/sunxi/hdmiaudio/sunxi-hdmipcm.c b/sound/soc/sunxi/hdmiaudio/sunxi-hdmipcm.c index 4916b51..9ab01ee 100644 --- a/sound/soc/sunxi/hdmiaudio/sunxi-hdmipcm.c +++ b/sound/soc/sunxi/hdmiaudio/sunxi-hdmipcm.c @@ -45,8 +45,8 @@ static const struct snd_pcm_hardware sunxi_pcm_hardware = { SNDRV_PCM_RATE_KNOT, .rate_min = 8000, .rate_max = 192000, - .channels_min = 1, - .channels_max = 2, + .channels_min = 2, + .channels_max = 8, .buffer_bytes_max = 128*1024, /* value must be (2^n)Kbyte size */ .period_bytes_min = 1024*4, .period_bytes_max = 1024*32,

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[alsa-devel] [PATCH] ARM: sun7i: Fix HDMI Audio driver's Alsa interface and do 8CH IEC60958 output
by thomas schorpp 01 Aug '14

01 Aug '14

First shot. HDMI backend seems to work with speaker-test but stream samples wrong order /mapping issues and the Alsa frontend is a stub, Pulseaudio needs more, still defaults to 2CH and "Analog". Please test if it switches Your receivers correctly into multichannel pcm mode, strange people with those overexpensed high end 7.1 receivers ;-) Signed-off-by: Not Yet ----------- https://github.com/linux-sunxi/linux-sunxi * stage/sunxi-3.4 diff --git a/drivers/video/sunxi/hdmi/drv_hdmi.c b/drivers/video/sunxi/hdmi/drv_hdmi.c index 283d2ae..9cf2cc2 100644 --- a/drivers/video/sunxi/hdmi/drv_hdmi.c +++ b/drivers/video/sunxi/hdmi/drv_hdmi.c @@ -314,7 +314,7 @@ __s32 Hdmi_init(struct platform_device *dev) if (err) return err; - audio_info.channel_num = 2; + audio_info.channel_num = 8; #if 0 { /* for audio test */ hdmi_audio_t audio_para; diff --git a/sound/soc/sunxi/hdmiaudio/sndhdmi.c b/sound/soc/sunxi/hdmiaudio/sndhdmi.c index 1c306d2..4563010 100644 --- a/sound/soc/sunxi/hdmiaudio/sndhdmi.c +++ b/sound/soc/sunxi/hdmiaudio/sndhdmi.c @@ -85,7 +85,9 @@ static int sndhdmi_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } - if (4 == hdmi_para.channel_num) + if (2 < hdmi_para.channel_num) + hdmi_para.channel_num = 8; + if (2 > hdmi_para.channel_num) hdmi_para.channel_num = 2; g_hdmi_func.hdmi_set_audio_para(&hdmi_para); @@ -129,11 +131,11 @@ struct snd_soc_dai_ops sndhdmi_dai_ops = { /* codec dai */ struct snd_soc_dai_driver sndhdmi_dai = { .name = "sndhdmi", - /* playback capabilities */ + /* playback capabilities, hdmi_core::audio_config() claims 8ch cap */ .playback = { .stream_name = "Playback", - .channels_min = 1, - .channels_max = 2, + .channels_min = 2, + .channels_max = 8, .rates = SNDHDMI_RATES, .formats = SNDHDMI_FORMATS, }, diff --git a/sound/soc/sunxi/hdmiaudio/sunxi-hdmiaudio.c b/sound/soc/sunxi/hdmiaudio/sunxi-hdmiaudio.c index 12da1a6..80dd3b9 100644 --- a/sound/soc/sunxi/hdmiaudio/sunxi-hdmiaudio.c +++ b/sound/soc/sunxi/hdmiaudio/sunxi-hdmiaudio.c @@ -34,8 +34,8 @@ #include <plat/sys_config.h> #include <plat/dma_compat.h> -static struct sunxi_dma_params sunxi_hdmiaudio_pcm_stereo_out = { - .client.name = "HDMIAUDIO PCM Stereo out", +static struct sunxi_dma_params sunxi_hdmiaudio_pcm_iec60958_out = { + .client.name = "HDMIAUDIO PCM IEC60958 out", #if defined CONFIG_ARCH_SUN4I || defined CONFIG_ARCH_SUN5I .channel = DMACH_HDMIAUDIO, #endif @@ -67,7 +67,7 @@ static int sunxi_hdmiaudio_hw_params(struct snd_pcm_substream *substream, rtd = substream->private_data; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &sunxi_hdmiaudio_pcm_stereo_out; + dma_data = &sunxi_hdmiaudio_pcm_iec60958_out; else printk("error:hdmiaudio can't support capture:%s,line:%d\n", __func__, __LINE__); @@ -182,8 +182,8 @@ static struct snd_soc_dai_driver sunxi_hdmiaudio_dai = { .resume = sunxi_hdmiaudio_resume, .remove = sunxi_hdmiaudio_dai_remove, .playback = { - .channels_min = 1, - .channels_max = 2, + .channels_min = 2, + .channels_max = 8, .rates = SUNXI_HDMI_RATES, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE, }, diff --git a/sound/soc/sunxi/hdmiaudio/sunxi-hdmipcm.c b/sound/soc/sunxi/hdmiaudio/sunxi-hdmipcm.c index 4916b51..9ab01ee 100644 --- a/sound/soc/sunxi/hdmiaudio/sunxi-hdmipcm.c +++ b/sound/soc/sunxi/hdmiaudio/sunxi-hdmipcm.c @@ -45,8 +45,8 @@ static const struct snd_pcm_hardware sunxi_pcm_hardware = { SNDRV_PCM_RATE_KNOT, .rate_min = 8000, .rate_max = 192000, - .channels_min = 1, - .channels_max = 2, + .channels_min = 2, + .channels_max = 8, .buffer_bytes_max = 128*1024, /* value must be (2^n)Kbyte size */ .period_bytes_min = 1024*4, .period_bytes_max = 1024*32,

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[alsa-devel] [PATCH v3 0/2] Add Freescale ASRC driver
by Nicolin Chen 01 Aug '14

01 Aug '14

This series of patches add Freescale ASRC module driver support along with the extra request in imx-sdma structure required by SDMA Device to Device script. The previous version has been in the maillist for nearly six months without any comment and reply. So I decide to drop the SDMA part's change since this series focus on ASRC support (Later, we can send SDMA part separately.) so that ASRC driver may get upstream first. Changelog v3: * PATCH-1: Added an Acked-by from Shawn Guo * PATCH-2: Dropped THIS_MODULE as Tobias suggests * PATCH-2: Use dma_coerce_mask_and_coherent() in fsl_asrc_dma_pcm_new() * PATCH-2: Added substream check in fsl_asrc_dma_pcm_new() v2: * PATCH-1: Dropped the change to SDMA driver. * PATCH-2: Dropped useless member in private data structures. * PATCH-2: Refined some comments in fsl_asrc_dma.c driver. * PATCH-2: Refined commit comments. Nicolin Chen (2): ARM: imx: Add the secondary request into the structure for imx-sdma ASoC: fsl_asrc: Add ASRC ASoC CPU DAI and platform drivers .../devicetree/bindings/sound/fsl,asrc.txt | 62 ++ include/linux/platform_data/dma-imx.h | 1 + sound/soc/fsl/Kconfig | 9 + sound/soc/fsl/Makefile | 2 + sound/soc/fsl/fsl_asrc.c | 1030 ++++++++++++++++++++ sound/soc/fsl/fsl_asrc.h | 461 +++++++++ sound/soc/fsl/fsl_asrc_dma.c | 386 ++++++++ 7 files changed, 1951 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/fsl,asrc.txt create mode 100644 sound/soc/fsl/fsl_asrc.c create mode 100644 sound/soc/fsl/fsl_asrc.h create mode 100644 sound/soc/fsl/fsl_asrc_dma.c -- 1.8.4

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