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

[alsa-devel] [PATCH] ALSA: hda - Disable AA-loopback on ALC668
by Gabriele Mazzotta 20 May '14

20 May '14

Disable the AA-loopback path to get rid of the constant white noise that can be heard when headphones are used. This patch was tested only on a Dell XPS13 9333. Fix based on d2e92709e88d ("ALSA: hda - Disable AA-loopback on ALC283 Chromebook"). Signed-off-by: Gabriele Mazzotta <gabriele.mzt(a)gmail.com> --- sound/pci/hda/patch_realtek.c | 7 +++++++ 1 file changed, 7 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 5f7c765..21c94e4 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5690,6 +5690,13 @@ static int patch_alc662(struct hda_codec *codec) if (has_cdefine_beep(codec)) spec->gen.beep_nid = 0x01; + switch (codec->vendor_id) { + case 0x10ec0668: + /* Disable AA-loopback as it causes white noise */ + spec->gen.mixer_nid = 0; + break; + } + if ((alc_get_coef0(codec) & (1 << 14)) && codec->bus->pci && codec->bus->pci->subsystem_vendor == 0x1025 && spec->cdefine.platform_type == 1) { -- 2.0.0.rc2

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[alsa-devel] Force an update of the ELD info?
by VDR User 20 May '14

20 May '14

Hi. I was wondering if it's possible to force alsa/drivers to update the ELD info yet? If not, is this something that can be added? Reason being, htpc users may or may not have their receiver/surround sound on at any given time. The ELD info can be used to determine this, and thus audio playback can be appropriately set for surround/no-surround at any given time. As it is now, to even use surround, the receiver must be turned on before the system is booted. That may not always be convenient. Also, leaving a receiver on 24/7 is a waste of power, generates heat, and puts unnecessary wear & tear on equipment that's not being used at the time. I have been told one way to force an ELD update is to completely unload alsa/drivers, and reload them. However, that's very inconvenient as it requires plugging in a keyboard, or ssh'ing into the box, unloading everything, toggling the receivers power state, then reloading everything. That's a pretty disruptive process which, if can be avoided, should be. Also, could someone with the proper knowledge clarify is this is something that alsa needs to support, the sound drivers need to support, and/or the video drivers need to support? I've heard very different answers and would like some clear info on this if anyone has any to offer. In case anyone is wondering, I have multiple linux htpcs connected via hdmi to receivers, that are then connected to tvs. These boxes have nothing more than an IR receiver connected to them for navigating an OSD. All of the boxes are using Nvidia video cards with vdpau decoding, and audio-over-hdmi with the snd_hda_intel. Many thanks for any input! -DK

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[alsa-devel] [RFC] A question regarding open ioctl for user space in ASoC driver.
by Nicolin Chen 20 May '14

20 May '14

Hi all, I have an ASoC driver in my own branch and I'm going to get it upstream in the months ahead. The driver is designed for Freescale ASRC module, Asynchronous Sample Rate Converter. It has the capability to convert audio data without involving software conversion like SRC function in ALSA-lib. But before I try to revise and send the driver, I've a few questions here. We currently have two main functions for ASRC: P2P's and M2M's. 1) P2P, peripheral to peripheral, regards ASRC as a FE (front end link) corresponding to a BE (back end link like ESAI<->CS42888). The P2P function resolves a problem of unsupported sample rates for BE that we can convert the unsupported audio format to a supported one before we send the data into BE. It also has better quality and less CPU loading comparing to ALSA-lib's SRC function. * For this function, I don't see any critical problem here. But... 2) M2M, memory to memory, can simply make ASRC as a sample rate converter without playback via any BE sound card, just like using any software to convert a WAV file in one sample rate to another WAV in a required sample rate. The driver has its self-designed application to compelte this function by sending the audio data from a wave file into a misc device via non-generic ioctl: static long asrc_ioctl(struct file *file, unsigned int cmd, unsigned long arg) { struct asrc_pair_params *params = file->private_data; void __user *user = (void __user *)arg; long ret = 0; switch (cmd) { case ASRC_REQ_PAIR: ret = asrc_ioctl_req_pair(params, user); break; case ASRC_CONFIG_PAIR: ret = asrc_ioctl_config_pair(params, user); break; case ASRC_START_CONV: [...] So I'm wondering if this kind of ioctl would be allowed or there's any alternative approach? Not only having application in user space, but we also have an ASRC plugin for ALSA-lib so that it can involve M2M function by using the plugin before sending data via pcm_write(). And here's the another question -- to share the ioctl and the struct asrc_pair_params with user space, there's a header file need to be put into include/uapi/. Is it possible? There's a reference code (drivers/mxc/asrc/mxc_asrc.c) located in: http://git.freescale.com/git/cgit.cgi/imx/linux-2.6-imx.git/tree/?h=imx_3.1… In our internal branch, the ASRC driver currently is divided into two drivers for M2M and P2P separately. For upstream version, I'm trying to merge them into a single ASoC driver with both M2M and P2P, just not sure how to handle the M2M part appropriately. Thank you, Nicolin Chen

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[alsa-devel] [PATCH 1/2] ASoC: samsung: s3c24{xx, 12}-i2s: port to use generic dmaengine API
by Vasily Khoruzhick 20 May '14

20 May '14

Use dmaengine instead of legacy s3c24xx DMA API for s3c24xx and s3c2412 Signed-off-by: Vasily Khoruzhick <anarsoul(a)gmail.com> --- sound/soc/samsung/Kconfig | 9 ++---- sound/soc/samsung/dmaengine.c | 3 ++ sound/soc/samsung/s3c-i2s-v2.c | 17 +--------- sound/soc/samsung/s3c2412-i2s.c | 63 +++++++++++++++++++++--------------- sound/soc/samsung/s3c24xx-i2s.c | 72 +++++++++++++++++++++++------------------ 5 files changed, 84 insertions(+), 80 deletions(-) diff --git a/sound/soc/samsung/Kconfig b/sound/soc/samsung/Kconfig index 3507574..022c7a4 100644 --- a/sound/soc/samsung/Kconfig +++ b/sound/soc/samsung/Kconfig @@ -1,11 +1,10 @@ config SND_SOC_SAMSUNG tristate "ASoC support for Samsung" depends on PLAT_SAMSUNG - select S3C2410_DMA if ARCH_S3C24XX + select S3C24XX_DMAC if ARCH_S3C24XX select S3C64XX_PL080 if ARCH_S3C64XX - select SND_S3C_DMA if !ARCH_S3C24XX - select SND_S3C_DMA_LEGACY if ARCH_S3C24XX - select SND_SOC_GENERIC_DMAENGINE_PCM if !ARCH_S3C24XX + select SND_S3C_DMA + select SND_SOC_GENERIC_DMAENGINE_PCM help Say Y or M if you want to add support for codecs attached to the Samsung SoCs' Audio interfaces. You will also need to @@ -19,7 +18,6 @@ config SND_S3C_DMA_LEGACY config SND_S3C24XX_I2S tristate - select S3C24XX_DMA config SND_S3C_I2SV2_SOC tristate @@ -27,7 +25,6 @@ config SND_S3C_I2SV2_SOC config SND_S3C2412_SOC_I2S tristate select SND_S3C_I2SV2_SOC - select S3C2410_DMA config SND_SAMSUNG_PCM tristate diff --git a/sound/soc/samsung/dmaengine.c b/sound/soc/samsung/dmaengine.c index 750ce58..72f27d1 100644 --- a/sound/soc/samsung/dmaengine.c +++ b/sound/soc/samsung/dmaengine.c @@ -17,6 +17,7 @@ #include <linux/module.h> #include <linux/amba/pl08x.h> +#include <linux/platform_data/dma-s3c24xx.h> #include <sound/core.h> #include <sound/pcm.h> @@ -29,6 +30,8 @@ #ifdef CONFIG_ARCH_S3C64XX #define filter_fn pl08x_filter_id +#elif defined(CONFIG_ARCH_S3C24XX) +#define filter_fn s3c24xx_dma_filter #else #define filter_fn NULL #endif diff --git a/sound/soc/samsung/s3c-i2s-v2.c b/sound/soc/samsung/s3c-i2s-v2.c index 79e7efb..1a74051 100644 --- a/sound/soc/samsung/s3c-i2s-v2.c +++ b/sound/soc/samsung/s3c-i2s-v2.c @@ -392,8 +392,6 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, int capture = (substream->stream == SNDRV_PCM_STREAM_CAPTURE); unsigned long irqs; int ret = 0; - struct s3c_dma_params *dma_data = - snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); pr_debug("Entered %s\n", __func__); @@ -424,13 +422,6 @@ static int s3c2412_i2s_trigger(struct snd_pcm_substream *substream, int cmd, local_irq_restore(irqs); - /* - * Load the next buffer to DMA to meet the reqirement - * of the auto reload mechanism of S3C24XX. - * This call won't bother S3C64XX. - */ - s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); - break; case SNDRV_PCM_TRIGGER_STOP: @@ -644,12 +635,6 @@ int s3c_i2sv2_probe(struct snd_soc_dai *dai, /* record our i2s structure for later use in the callbacks */ snd_soc_dai_set_drvdata(dai, i2s); - i2s->regs = ioremap(base, 0x100); - if (i2s->regs == NULL) { - dev_err(dev, "cannot ioremap registers\n"); - return -ENXIO; - } - i2s->iis_pclk = clk_get(dev, "iis"); if (IS_ERR(i2s->iis_pclk)) { dev_err(dev, "failed to get iis_clock\n"); @@ -729,7 +714,7 @@ int s3c_i2sv2_register_component(struct device *dev, int id, struct snd_soc_component_driver *cmp_drv, struct snd_soc_dai_driver *dai_drv) { - struct snd_soc_dai_ops *ops = dai_drv->ops; + struct snd_soc_dai_ops *ops = (struct snd_soc_dai_ops *)dai_drv->ops; ops->trigger = s3c2412_i2s_trigger; if (!ops->hw_params) diff --git a/sound/soc/samsung/s3c2412-i2s.c b/sound/soc/samsung/s3c2412-i2s.c index d079445..f672352 100644 --- a/sound/soc/samsung/s3c2412-i2s.c +++ b/sound/soc/samsung/s3c2412-i2s.c @@ -33,25 +33,17 @@ #include "regs-i2s-v2.h" #include "s3c2412-i2s.h" -static struct s3c2410_dma_client s3c2412_dma_client_out = { - .name = "I2S PCM Stereo out" -}; - -static struct s3c2410_dma_client s3c2412_dma_client_in = { - .name = "I2S PCM Stereo in" -}; - static struct s3c_dma_params s3c2412_i2s_pcm_stereo_out = { - .client = &s3c2412_dma_client_out, - .channel = DMACH_I2S_OUT, - .dma_addr = S3C2410_PA_IIS + S3C2412_IISTXD, + .client = + (struct s3c2410_dma_client *)&s3c2412_i2s_pcm_stereo_out, + .ch_name = "tx", .dma_size = 4, }; static struct s3c_dma_params s3c2412_i2s_pcm_stereo_in = { - .client = &s3c2412_dma_client_in, - .channel = DMACH_I2S_IN, - .dma_addr = S3C2410_PA_IIS + S3C2412_IISRXD, + .client = + (struct s3c2410_dma_client *)&s3c2412_i2s_pcm_stereo_in, + .ch_name = "rx", .dma_size = 4, }; @@ -63,6 +55,9 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) pr_debug("Entered %s\n", __func__); + samsung_asoc_init_dma_data(dai, &s3c2412_i2s_pcm_stereo_out, + &s3c2412_i2s_pcm_stereo_in); + ret = s3c_i2sv2_probe(dai, &s3c2412_i2s, S3C2410_PA_IIS); if (ret) return ret; @@ -70,10 +65,9 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) s3c2412_i2s.dma_capture = &s3c2412_i2s_pcm_stereo_in; s3c2412_i2s.dma_playback = &s3c2412_i2s_pcm_stereo_out; - s3c2412_i2s.iis_cclk = clk_get(dai->dev, "i2sclk"); + s3c2412_i2s.iis_cclk = devm_clk_get(dai->dev, "i2sclk"); if (IS_ERR(s3c2412_i2s.iis_cclk)) { pr_err("failed to get i2sclk clock\n"); - iounmap(s3c2412_i2s.regs); return PTR_ERR(s3c2412_i2s.iis_cclk); } @@ -94,8 +88,6 @@ static int s3c2412_i2s_probe(struct snd_soc_dai *dai) static int s3c2412_i2s_remove(struct snd_soc_dai *dai) { clk_disable(s3c2412_i2s.iis_cclk); - clk_put(s3c2412_i2s.iis_cclk); - iounmap(s3c2412_i2s.regs); return 0; } @@ -105,18 +97,10 @@ static int s3c2412_i2s_hw_params(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { struct s3c_i2sv2_info *i2s = snd_soc_dai_get_drvdata(cpu_dai); - struct s3c_dma_params *dma_data; u32 iismod; pr_debug("Entered %s\n", __func__); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = i2s->dma_playback; - else - dma_data = i2s->dma_capture; - - snd_soc_dai_set_dma_data(cpu_dai, substream, dma_data); - iismod = readl(i2s->regs + S3C2412_IISMOD); pr_debug("%s: r: IISMOD: %x\n", __func__, iismod); @@ -169,6 +153,33 @@ static const struct snd_soc_component_driver s3c2412_i2s_component = { static int s3c2412_iis_dev_probe(struct platform_device *pdev) { int ret = 0; + struct resource *res; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "Can't get IO resource.\n"); + return -ENOENT; + } + s3c2412_i2s.regs = devm_ioremap_resource(&pdev->dev, res); + if (s3c2412_i2s.regs == NULL) + return -ENXIO; + + s3c2412_i2s_pcm_stereo_out.dma_addr = res->start + S3C2412_IISTXD; + s3c2412_i2s_pcm_stereo_in.dma_addr = res->start + S3C2412_IISRXD; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res) { + dev_err(&pdev->dev, "Can't get DMA resource for playback.\n"); + return -ENOENT; + } + s3c2412_i2s_pcm_stereo_out.channel = res->start; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!res) { + dev_err(&pdev->dev, "Can't get DMA resource for capture.\n"); + return -ENOENT; + } + s3c2412_i2s_pcm_stereo_in.channel = res->start; ret = s3c_i2sv2_register_component(&pdev->dev, -1, &s3c2412_i2s_component, diff --git a/sound/soc/samsung/s3c24xx-i2s.c b/sound/soc/samsung/s3c24xx-i2s.c index f31e916..f854d45 100644 --- a/sound/soc/samsung/s3c24xx-i2s.c +++ b/sound/soc/samsung/s3c24xx-i2s.c @@ -31,25 +31,17 @@ #include "dma.h" #include "s3c24xx-i2s.h" -static struct s3c2410_dma_client s3c24xx_dma_client_out = { - .name = "I2S PCM Stereo out" -}; - -static struct s3c2410_dma_client s3c24xx_dma_client_in = { - .name = "I2S PCM Stereo in" -}; - static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_out = { - .client = &s3c24xx_dma_client_out, - .channel = DMACH_I2S_OUT, - .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO, + .client = + (struct s3c2410_dma_client *)&s3c24xx_i2s_pcm_stereo_out, + .ch_name = "tx", .dma_size = 2, }; static struct s3c_dma_params s3c24xx_i2s_pcm_stereo_in = { - .client = &s3c24xx_dma_client_in, - .channel = DMACH_I2S_IN, - .dma_addr = S3C2410_PA_IIS + S3C2410_IISFIFO, + .client = + (struct s3c2410_dma_client *)&s3c24xx_i2s_pcm_stereo_in, + .ch_name = "rx", .dma_size = 2, }; @@ -231,18 +223,12 @@ static int s3c24xx_i2s_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 s3c_dma_params *dma_data; + struct snd_dmaengine_dai_dma_data *dma_data; u32 iismod; pr_debug("Entered %s\n", __func__); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dma_data = &s3c24xx_i2s_pcm_stereo_out; - else - dma_data = &s3c24xx_i2s_pcm_stereo_in; - - snd_soc_dai_set_dma_data(rtd->cpu_dai, substream, dma_data); + dma_data = snd_soc_dai_get_dma_data(dai, substream); /* Working copies of register */ iismod = readl(s3c24xx_i2s.regs + S3C2410_IISMOD); @@ -251,11 +237,11 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream, switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: iismod &= ~S3C2410_IISMOD_16BIT; - dma_data->dma_size = 1; + dma_data->addr_width = 1; break; case SNDRV_PCM_FORMAT_S16_LE: iismod |= S3C2410_IISMOD_16BIT; - dma_data->dma_size = 2; + dma_data->addr_width = 2; break; default: return -EINVAL; @@ -270,8 +256,6 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { int ret = 0; - struct s3c_dma_params *dma_data = - snd_soc_dai_get_dma_data(dai, substream); pr_debug("Entered %s\n", __func__); @@ -290,7 +274,6 @@ static int s3c24xx_i2s_trigger(struct snd_pcm_substream *substream, int cmd, else s3c24xx_snd_txctrl(1); - s3c2410_dma_ctrl(dma_data->channel, S3C2410_DMAOP_STARTED); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: @@ -380,14 +363,12 @@ static int s3c24xx_i2s_probe(struct snd_soc_dai *dai) { pr_debug("Entered %s\n", __func__); - s3c24xx_i2s.regs = ioremap(S3C2410_PA_IIS, 0x100); - if (s3c24xx_i2s.regs == NULL) - return -ENXIO; + samsung_asoc_init_dma_data(dai, &s3c24xx_i2s_pcm_stereo_out, + &s3c24xx_i2s_pcm_stereo_in); - s3c24xx_i2s.iis_clk = clk_get(dai->dev, "iis"); + s3c24xx_i2s.iis_clk = devm_clk_get(dai->dev, "iis"); if (IS_ERR(s3c24xx_i2s.iis_clk)) { pr_err("failed to get iis_clock\n"); - iounmap(s3c24xx_i2s.regs); return PTR_ERR(s3c24xx_i2s.iis_clk); } clk_enable(s3c24xx_i2s.iis_clk); @@ -474,6 +455,33 @@ static const struct snd_soc_component_driver s3c24xx_i2s_component = { static int s3c24xx_iis_dev_probe(struct platform_device *pdev) { int ret = 0; + struct resource *res; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "Can't get IO resource.\n"); + return -ENOENT; + } + s3c24xx_i2s.regs = devm_ioremap_resource(&pdev->dev, res); + if (s3c24xx_i2s.regs == NULL) + return -ENXIO; + + s3c24xx_i2s_pcm_stereo_out.dma_addr = res->start + S3C2410_IISFIFO; + s3c24xx_i2s_pcm_stereo_in.dma_addr = res->start + S3C2410_IISFIFO; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 0); + if (!res) { + dev_err(&pdev->dev, "Can't get DMA resource for playback.\n"); + return -ENOENT; + } + s3c24xx_i2s_pcm_stereo_out.channel = res->start; + + res = platform_get_resource(pdev, IORESOURCE_DMA, 1); + if (!res) { + dev_err(&pdev->dev, "Can't get DMA resource for capture.\n"); + return -ENOENT; + } + s3c24xx_i2s_pcm_stereo_in.channel = res->start; ret = snd_soc_register_component(&pdev->dev, &s3c24xx_i2s_component, &s3c24xx_i2s_dai, 1); -- 1.9.3

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[alsa-devel] [PATCHv5 1/2] ALSA: hda/hdmi - Add Nvidia Tegra124 HDMI support
by Dylan Reid 20 May '14

20 May '14

From: Sumit Bhattacharya <sumitb(a)nvidia.com> Add the Tegra12x HDA codec id to patch_hdmi. Signed-off-by: Sumit Bhattacharya <sumitb(a)nvidia.com> Signed-off-by: Dylan Reid <dgreid(a)chromium.org> Acked-by: Stephen Warren <swarren(a)nvidia.com> --- Changes since v4: Commit message only. --- sound/pci/hda/patch_hdmi.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/pci/hda/patch_hdmi.c b/sound/pci/hda/patch_hdmi.c index 2a3050d..be0a9ee 100644 --- a/sound/pci/hda/patch_hdmi.c +++ b/sound/pci/hda/patch_hdmi.c @@ -3320,6 +3320,7 @@ static const struct hda_codec_preset snd_hda_preset_hdmi[] = { { .id = 0x10de001a, .name = "GPU 1a HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de001b, .name = "GPU 1b HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de001c, .name = "GPU 1c HDMI/DP", .patch = patch_nvhdmi }, +{ .id = 0x10de0028, .name = "Tegra12x HDMI", .patch = patch_nvhdmi }, { .id = 0x10de0040, .name = "GPU 40 HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de0041, .name = "GPU 41 HDMI/DP", .patch = patch_nvhdmi }, { .id = 0x10de0042, .name = "GPU 42 HDMI/DP", .patch = patch_nvhdmi }, @@ -3376,6 +3377,7 @@ MODULE_ALIAS("snd-hda-codec-id:10de0019"); MODULE_ALIAS("snd-hda-codec-id:10de001a"); MODULE_ALIAS("snd-hda-codec-id:10de001b"); MODULE_ALIAS("snd-hda-codec-id:10de001c"); +MODULE_ALIAS("snd-hda-codec-id:10de0028"); MODULE_ALIAS("snd-hda-codec-id:10de0040"); MODULE_ALIAS("snd-hda-codec-id:10de0041"); MODULE_ALIAS("snd-hda-codec-id:10de0042"); -- 1.8.1.3.605.g02339dd

2 1

[alsa-devel] [PATCH] sound: soc: samsung: dma.c: Fix for possible null pointer dereference
by Rickard Strandqvist 20 May '14

20 May '14

There is otherwise a risk of a possible null pointer dereference. Was largely found by using a static code analysis program called cppcheck. Signed-off-by: Rickard Strandqvist <rickard_strandqvist(a)spectrumdigital.se> --- sound/soc/samsung/dma.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index dc09b71..1d9bcaa 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -115,17 +115,19 @@ static void dma_enqueue(struct snd_pcm_substream *substream) static void audio_buffdone(void *data) { struct snd_pcm_substream *substream = data; - struct runtime_data *prtd = substream->runtime->private_data; + struct runtime_data *prtd = NULL; pr_debug("Entered %s\n", __func__); - if (prtd->state & ST_RUNNING) { + if (substream) + prtd = substream->runtime->private_data; + + if (prtd && prtd->state & ST_RUNNING) { prtd->dma_pos += prtd->dma_period; if (prtd->dma_pos >= prtd->dma_end) prtd->dma_pos = prtd->dma_start; - if (substream) - snd_pcm_period_elapsed(substream); + snd_pcm_period_elapsed(substream); spin_lock(&prtd->lock); if (!samsung_dma_has_circular()) { -- 1.7.10.4

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[alsa-devel] [PATCH] ASoC: Intel: Fix pcm stream context restore crash
by Jarkko Nikula 19 May '14

19 May '14

From: Kevin Strasser <kevin.strasser(a)intel.com> In some cases the pcm stream is closed while context has been scheduled to be restored, causing a null pointer deref panic. Cancel work to ensure stream does not get freed while work is still active/pending. Also, restoring the pcm context can be safely skipped after the stream has been stopped. Check if pcm stream is still running before restoring stream context to help pending work finish more quickly in stream close path. Signed-off-by: Kevin Strasser <kevin.strasser(a)intel.com> Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- For topic/intel branch since fixes issue in recently added suspend/resume code. --- sound/soc/intel/sst-baytrail-pcm.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/intel/sst-baytrail-pcm.c b/sound/soc/intel/sst-baytrail-pcm.c index 6242ccce2bb7..3af38576e91e 100644 --- a/sound/soc/intel/sst-baytrail-pcm.c +++ b/sound/soc/intel/sst-baytrail-pcm.c @@ -164,7 +164,8 @@ static void sst_byt_pcm_work(struct work_struct *work) struct sst_byt_pcm_data *pcm_data = container_of(work, struct sst_byt_pcm_data, work); - sst_byt_pcm_restore_stream_context(pcm_data->substream); + if (snd_pcm_running(pcm_data->substream)) + sst_byt_pcm_restore_stream_context(pcm_data->substream); } static int sst_byt_pcm_trigger(struct snd_pcm_substream *substream, int cmd) @@ -277,6 +278,7 @@ static int sst_byt_pcm_close(struct snd_pcm_substream *substream) dev_dbg(rtd->dev, "PCM: close\n"); + cancel_work_sync(&pcm_data->work); mutex_lock(&pcm_data->mutex); ret = sst_byt_stream_free(byt, pcm_data->stream); if (ret < 0) { -- 2.0.0.rc2

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[alsa-devel] [PATCH RFT] ASoC: rt5645: Fix updating wrong register for T5645_AIF2 case
by Axel Lin 19 May '14

19 May '14

This looks like a copy-paste bug, fix it. Signed-off-by: Axel Lin <axel.lin(a)ingics.com> --- Hi, I don't have this h/w, so please test if this patch works. Axel sound/soc/codecs/rt5645.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 68923ec..ab97d72 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -1932,8 +1932,8 @@ static int rt5645_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) RT5645_I2S_MS_MASK | RT5645_I2S_BP_MASK | RT5645_I2S_DF_MASK, reg_val); break; - case RT5645_AIF2: - snd_soc_update_bits(codec, RT5645_I2S1_SDP, + case RT5645_AIF2: + snd_soc_update_bits(codec, RT5645_I2S2_SDP, RT5645_I2S_MS_MASK | RT5645_I2S_BP_MASK | RT5645_I2S_DF_MASK, reg_val); break; -- 1.8.3.2

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[alsa-devel] [PATCH 1/2] ASoC: Remove needless snd_soc_dapm_sync() from machine driver inits
by Jarkko Nikula 19 May '14

19 May '14

ALSA SoC core takes care of calling snd_soc_dapm_sync() at the end snd_soc_instantiate_card() so there is no need to call it from machine driver init functions. Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/intel/byt-rt5640.c | 1 - sound/soc/pxa/brownstone.c | 2 -- sound/soc/tegra/tegra_wm9712.c | 4 +--- 3 files changed, 1 insertion(+), 6 deletions(-) diff --git a/sound/soc/intel/byt-rt5640.c b/sound/soc/intel/byt-rt5640.c index dbc63d09f424..eee001f70964 100644 --- a/sound/soc/intel/byt-rt5640.c +++ b/sound/soc/intel/byt-rt5640.c @@ -105,7 +105,6 @@ static int byt_rt5640_init(struct snd_soc_pcm_runtime *runtime) snd_soc_dapm_enable_pin(dapm, "Speaker"); snd_soc_dapm_enable_pin(dapm, "Internal Mic"); - snd_soc_dapm_sync(dapm); return ret; } diff --git a/sound/soc/pxa/brownstone.c b/sound/soc/pxa/brownstone.c index 08acdc236bf8..9c128d4c8cbf 100644 --- a/sound/soc/pxa/brownstone.c +++ b/sound/soc/pxa/brownstone.c @@ -70,8 +70,6 @@ static int brownstone_wm8994_init(struct snd_soc_pcm_runtime *rtd) snd_soc_dapm_nc_pin(dapm, "IN2RP:VXRP"); snd_soc_dapm_nc_pin(dapm, "IN2LN"); - snd_soc_dapm_sync(dapm); - return 0; } diff --git a/sound/soc/tegra/tegra_wm9712.c b/sound/soc/tegra/tegra_wm9712.c index 25a7f8211ecf..de087ee3458a 100644 --- a/sound/soc/tegra/tegra_wm9712.c +++ b/sound/soc/tegra/tegra_wm9712.c @@ -50,9 +50,7 @@ static int tegra_wm9712_init(struct snd_soc_pcm_runtime *rtd) struct snd_soc_codec *codec = codec_dai->codec; struct snd_soc_dapm_context *dapm = &codec->dapm; - snd_soc_dapm_force_enable_pin(dapm, "Mic Bias"); - - return snd_soc_dapm_sync(dapm); + return snd_soc_dapm_force_enable_pin(dapm, "Mic Bias"); } static struct snd_soc_dai_link tegra_wm9712_dai = { -- 2.0.0.rc2

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[alsa-devel] [PATCH 1/2] ASoC: cs42xx8: Do not use rtd->codec
by Lars-Peter Clausen 19 May '14

19 May '14

rtd->codec does not necessarily point to the CODEC instance for which the callback was called (e.g. for CODEC<->CODEC or multi-CODEC links). Use dai->codec instead. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/cs42xx8.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/codecs/cs42xx8.c b/sound/soc/codecs/cs42xx8.c index 8502032..a25bc60 100644 --- a/sound/soc/codecs/cs42xx8.c +++ b/sound/soc/codecs/cs42xx8.c @@ -248,8 +248,7 @@ static int cs42xx8_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 snd_soc_codec *codec = rtd->codec; + struct snd_soc_codec *codec = dai->codec; struct cs42xx8_priv *cs42xx8 = snd_soc_codec_get_drvdata(codec); bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; u32 ratio = cs42xx8->sysclk / params_rate(params); -- 1.8.0

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