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

[alsa-devel] [PATCH] ASoC: fsl_spdif: Add core clock control for DMA access
by Nicolin Chen 24 Apr '14

24 Apr '14

Regmap is able to enable/disable the core clock automatically each time it's going to access the registers. But for DMA cases during playback or recording, it's totally beyong control of regmap. So we have to open the clock manually. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/fsl/fsl_spdif.c | 22 +++++++++++++++++++++- 1 file changed, 21 insertions(+), 1 deletion(-) diff --git a/sound/soc/fsl/fsl_spdif.c b/sound/soc/fsl/fsl_spdif.c index daa6198..e2836b3 100644 --- a/sound/soc/fsl/fsl_spdif.c +++ b/sound/soc/fsl/fsl_spdif.c @@ -80,6 +80,7 @@ struct fsl_spdif_priv { u8 rxclk_src; struct clk *txclk[SPDIF_TXRATE_MAX]; struct clk *rxclk; + struct clk *coreclk; struct snd_dmaengine_dai_dma_data dma_params_tx; struct snd_dmaengine_dai_dma_data dma_params_rx; @@ -423,10 +424,16 @@ static int fsl_spdif_startup(struct snd_pcm_substream *substream, /* Reset module and interrupts only for first initialization */ if (!cpu_dai->active) { + ret = clk_prepare_enable(spdif_priv->coreclk); + if (ret) { + dev_err(&pdev->dev, "failed to enable core clock\n"); + return ret; + } + ret = spdif_softreset(spdif_priv); if (ret) { dev_err(&pdev->dev, "failed to soft reset\n"); - return ret; + goto err; } /* Disable all the interrupts */ @@ -454,6 +461,11 @@ static int fsl_spdif_startup(struct snd_pcm_substream *substream, regmap_update_bits(regmap, REG_SPDIF_SCR, SCR_LOW_POWER, 0); return 0; + +err: + clk_disable_unprepare(spdif_priv->coreclk); + + return ret; } static void fsl_spdif_shutdown(struct snd_pcm_substream *substream, @@ -484,6 +496,7 @@ static void fsl_spdif_shutdown(struct snd_pcm_substream *substream, spdif_intr_status_clear(spdif_priv); regmap_update_bits(regmap, REG_SPDIF_SCR, SCR_LOW_POWER, SCR_LOW_POWER); + clk_disable_unprepare(spdif_priv->coreclk); } } @@ -1134,6 +1147,13 @@ static int fsl_spdif_probe(struct platform_device *pdev) return ret; } + /* Get core clock for data register access via DMA */ + spdif_priv->coreclk = devm_clk_get(&pdev->dev, "core"); + if (IS_ERR(spdif_priv->coreclk)) { + dev_err(&pdev->dev, "no core clock in devicetree\n"); + return PTR_ERR(spdif_priv->coreclk); + } + /* Select clock source for rx/tx clock */ spdif_priv->rxclk = devm_clk_get(&pdev->dev, "rxtx1"); if (IS_ERR(spdif_priv->rxclk)) { -- 1.8.4

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[alsa-devel] [PATCH] ASoC: Intel: Cancel hsw_notification_work before freeing the stream
by Jarkko Nikula 24 Apr '14

24 Apr '14

I suppose there is a possibility that hsw_notification_work() may run after sst_hsw_stream_free() which can lead to a kernel crash since struct sst_hsw_stream is freed at that point and stream = container_of(work, struct sst_hsw_stream, notify_work) is not valid when hsw_notification_work() is run. Reported-by: Derek Basehore <dbasehore(a)chromium.org> Reported-by: Wenkai Du <wenkai.du(a)intel.com> Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- Untested, I don't have at the moment Haswell to test this and Liam hasn't seen this patch yet so ack/tested-by from him might be good to have :-) --- 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 6c0b4f247a86..5bcf5963a0ba 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -1207,6 +1207,7 @@ int sst_hsw_stream_free(struct sst_hsw *hsw, struct sst_hsw_stream *stream) trace_hsw_stream_free_req(stream, &stream->free_req); out: + cancel_work_sync(&stream->notify_work); spin_lock_irqsave(&sst->spinlock, flags); list_del(&stream->node); kfree(stream); -- 1.9.2

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[alsa-devel] [PATCH] ASoC: imx-audmux: Fix section mismatch
by Lars-Peter Clausen 24 Apr '14

24 Apr '14

audmux_debugfs_init() is marked as __init, but is called from imx_audmux_probe() which is not marked as __init. This creates a section mismatch and a potential runtime crash (if imx_audmux_probe() is called after the .init section was dropped). This patch removes the __init annotation from audmux_debugfs_init(), which fixes the following warning: WARNING: sound/soc/built-in.o(.text+0x86960): Section mismatch in reference from the function imx_audmux_probe() to the function .init.text:audmux_debugfs_init() Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/fsl/imx-audmux.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/fsl/imx-audmux.c b/sound/soc/fsl/imx-audmux.c index ac86993..267717a 100644 --- a/sound/soc/fsl/imx-audmux.c +++ b/sound/soc/fsl/imx-audmux.c @@ -145,7 +145,7 @@ static const struct file_operations audmux_debugfs_fops = { .llseek = default_llseek, }; -static void __init audmux_debugfs_init(void) +static void audmux_debugfs_init(void) { int i; char buf[20]; -- 1.8.0

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Re: [alsa-devel] [PATCH] ASoC: imx-sgtl5000.c: Add support for specifying PDCR_TXRXEN
by Shawn Guo 24 Apr '14

24 Apr '14

On Wed, Apr 23, 2014 at 05:50:18PM +0200, David Jander wrote: > This enables effectively swapping RX/TX pins on the external audio port > > Signed-off-by: David Jander <david(a)protonic.nl> You should copy Mark Brown <broonie(a)kernel.org> and devicetree(a)vger.kernel.org on this patch. Shawn > --- > Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt | 4 ++++ > sound/soc/fsl/imx-sgtl5000.c | 5 ++++- > 2 files changed, 8 insertions(+), 1 deletion(-) > > diff --git a/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt b/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt > index e4acdd8..3afea85 100644 > --- a/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt > +++ b/Documentation/devicetree/bindings/sound/imx-audio-sgtl5000.txt > @@ -29,6 +29,10 @@ Required properties: > - mux-int-port : The internal port of the i.MX audio muxer (AUDMUX) > - mux-ext-port : The external port of the i.MX audio muxer > > +Optional properties: > +- ext-port-txrxen : Flag. Set to swap RX/TX signals on external port (set the > + IMX_AUDMUX_V2_PDCR_TXRXEN bit in the PDC register of the external port) > + > Note: The AUDMUX port numbering should start at 1, which is consistent with > hardware manual. > > diff --git a/sound/soc/fsl/imx-sgtl5000.c b/sound/soc/fsl/imx-sgtl5000.c > index 1cb22dd..b543fa6 100644 > --- a/sound/soc/fsl/imx-sgtl5000.c > +++ b/sound/soc/fsl/imx-sgtl5000.c > @@ -63,6 +63,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) > struct i2c_client *codec_dev; > struct imx_sgtl5000_data *data = NULL; > int int_port, ext_port; > + unsigned int ext_txrxen = 0; > int ret; > > ret = of_property_read_u32(np, "mux-int-port", &int_port); > @@ -75,6 +76,8 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) > dev_err(&pdev->dev, "mux-ext-port missing or invalid\n"); > return ret; > } > + if (of_property_read_bool(np, "ext-port-txrxen")) > + ext_txrxen = IMX_AUDMUX_V2_PDCR_TXRXEN; > > /* > * The port numbering in the hardware manual starts at 1, while > @@ -95,7 +98,7 @@ static int imx_sgtl5000_probe(struct platform_device *pdev) > } > ret = imx_audmux_v2_configure_port(ext_port, > IMX_AUDMUX_V2_PTCR_SYN, > - IMX_AUDMUX_V2_PDCR_RXDSEL(int_port)); > + IMX_AUDMUX_V2_PDCR_RXDSEL(int_port) | ext_txrxen); > if (ret) { > dev_err(&pdev->dev, "audmux external port setup failed\n"); > return ret; > -- > 1.9.1 >

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[alsa-devel] [PATCH] ASoC: core: Don't break component searching if both id and num_dai are 0
by Nicolin Chen 23 Apr '14

23 Apr '14

The commit e41975ed (ASoC: core: Fix the DAI name getting) added a break within the "if (id < 0 || id >= pos->num_dai)" while the original design of the search didn't break the loop if that condition contented but only mark the ret error and let it go on to search the next component. In a case like dmaengine which's not a dai but as a component sharing an identical name with a dai, both the id and pos->num_dai here could be 0. If we break the search, we may never find the dai we want as it might be placed behind its dmaengine in the component list. So this patch fixes the issue above by following the original design to let the search carry on. Signed-off-by: Nicolin Chen <Guangyu.Chen(a)freescale.com> --- sound/soc/soc-core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index a17c047..82e9fdf 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4591,7 +4591,7 @@ int snd_soc_of_get_dai_name(struct device_node *of_node, if (id < 0 || id >= pos->num_dai) { ret = -EINVAL; - break; + continue; } ret = 0; -- 1.8.4

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[alsa-devel] [PATCH] ASoC: SAMSUNG: Don't clear clock setting during i2s_startup
by Tushar Behera 23 Apr '14

23 Apr '14

In exiting kernel, if DAIFMT flags are set in dai_link and I2S is set to run in master mode, the I2S clocks are not getting configured resulting in no output. Existing code clears the current I2S clock settings during i2s_startup and requires that the clocks are reconfigured. It then assumes that sound-card driver would call snd_soc_dai_{set_sysclk/set_fmt} to configure the root clock. 1. Since I2S clock settings remain fixed for a board, it would be better to set the clocks once during sound-card probe. 2. Also if the DAIFMT flags are set in dai_link, snd_soc_dai_set_fmt is called during DAI probe. If both these conditions are true, then I2S clock remains unconfigured during audio playback. Fix this by removing the code to clear rclk_srcrate in i2s_startup. Instead, reset this during DAI probe. Signed-off-by: Tushar Behera <tushar.behera(a)linaro.org> --- The patch is based on v3.15-rc2. sound/soc/samsung/i2s.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index 048ead9..6e61db7 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -724,9 +724,6 @@ static int i2s_startup(struct snd_pcm_substream *substream, else i2s->mode |= DAI_MANAGER; - /* Enforce set_sysclk in Master mode */ - i2s->rclk_srcrate = 0; - if (!any_active(i2s) && (i2s->quirks & QUIRK_NEED_RSTCLR)) writel(CON_RSTCLR, i2s->addr + I2SCON); @@ -984,6 +981,7 @@ probe_exit: /* Reset any constraint on RFS and BFS */ i2s->rfs = 0; i2s->bfs = 0; + i2s->rclk_srcrate = 0; i2s_txctrl(i2s, 0); i2s_rxctrl(i2s, 0); i2s_fifo(i2s, FIC_TXFLUSH); -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: Intel: Fix audio crash due to race condition in stream deletion
by Jarkko Nikula 23 Apr '14

23 Apr '14

From: Wenkai Du <wenkai.du(a)intel.com> There is a race between sst_byt_stream_free() and sst_byt_get_stream() if sst_byt_get_stream() called from sst_byt_irq_thread() context is accessing the byt->stream_list while a stream is deleted from the list. A stream is added to byt->stream_list in sst_byt_stream_new() and deleted in sst_byt_stream_free(). sst_byt_get_stream() is always protected by sst->spinlock, but the stream addition and deletion are not protected. The patch adds spinlock to both stream addition and deletion. [Jarkko: Same fix added to sst-haswell-ipc.c too] Signed-off-by: Wenkai Du <wenkai.du(a)intel.com> Signed-off-by: Jarkko Nikula <jarkko.nikula(a)linux.intel.com> --- sound/soc/intel/sst-baytrail-ipc.c | 8 ++++++++ sound/soc/intel/sst-haswell-ipc.c | 8 ++++++++ 2 files changed, 16 insertions(+) diff --git a/sound/soc/intel/sst-baytrail-ipc.c b/sound/soc/intel/sst-baytrail-ipc.c index d0eaeee21be4..0d31dbbf4806 100644 --- a/sound/soc/intel/sst-baytrail-ipc.c +++ b/sound/soc/intel/sst-baytrail-ipc.c @@ -542,16 +542,20 @@ struct sst_byt_stream *sst_byt_stream_new(struct sst_byt *byt, int id, void *data) { struct sst_byt_stream *stream; + struct sst_dsp *sst = byt->dsp; + unsigned long flags; stream = kzalloc(sizeof(*stream), GFP_KERNEL); if (stream == NULL) return NULL; + spin_lock_irqsave(&sst->spinlock, flags); list_add(&stream->node, &byt->stream_list); stream->notify_position = notify_position; stream->pdata = data; stream->byt = byt; stream->str_id = id; + spin_unlock_irqrestore(&sst->spinlock, flags); return stream; } @@ -630,6 +634,8 @@ int sst_byt_stream_free(struct sst_byt *byt, struct sst_byt_stream *stream) { u64 header; int ret = 0; + struct sst_dsp *sst = byt->dsp; + unsigned long flags; if (!stream->commited) goto out; @@ -644,8 +650,10 @@ int sst_byt_stream_free(struct sst_byt *byt, struct sst_byt_stream *stream) stream->commited = false; out: + spin_lock_irqsave(&sst->spinlock, flags); list_del(&stream->node); kfree(stream); + spin_unlock_irqrestore(&sst->spinlock, flags); return ret; } diff --git a/sound/soc/intel/sst-haswell-ipc.c b/sound/soc/intel/sst-haswell-ipc.c index 50e4246d4b57..6c0b4f247a86 100644 --- a/sound/soc/intel/sst-haswell-ipc.c +++ b/sound/soc/intel/sst-haswell-ipc.c @@ -1159,11 +1159,14 @@ struct sst_hsw_stream *sst_hsw_stream_new(struct sst_hsw *hsw, int id, void *data) { struct sst_hsw_stream *stream; + struct sst_dsp *sst = hsw->dsp; + unsigned long flags; stream = kzalloc(sizeof(*stream), GFP_KERNEL); if (stream == NULL) return NULL; + spin_lock_irqsave(&sst->spinlock, flags); list_add(&stream->node, &hsw->stream_list); stream->notify_position = notify_position; stream->pdata = data; @@ -1172,6 +1175,7 @@ struct sst_hsw_stream *sst_hsw_stream_new(struct sst_hsw *hsw, int id, /* work to process notification messages */ INIT_WORK(&stream->notify_work, hsw_notification_work); + spin_unlock_irqrestore(&sst->spinlock, flags); return stream; } @@ -1180,6 +1184,8 @@ int sst_hsw_stream_free(struct sst_hsw *hsw, struct sst_hsw_stream *stream) { u32 header; int ret = 0; + struct sst_dsp *sst = hsw->dsp; + unsigned long flags; /* dont free DSP streams that are not commited */ if (!stream->commited) @@ -1201,8 +1207,10 @@ int sst_hsw_stream_free(struct sst_hsw *hsw, struct sst_hsw_stream *stream) trace_hsw_stream_free_req(stream, &stream->free_req); out: + spin_lock_irqsave(&sst->spinlock, flags); list_del(&stream->node); kfree(stream); + spin_unlock_irqrestore(&sst->spinlock, flags); return ret; } -- 1.9.2

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[alsa-devel] [PATCH 1/6] ASoC: jz4740: Remove Makefile entry for removed file
by Lars-Peter Clausen 23 Apr '14

23 Apr '14

Commit 0406a40a0 ("ASoC: jz4740: Use the generic dmaengine PCM driver") jz4740-pcm.c file, but neglected to remove the Makefile entries. Fixes: 0406a40a0 ("ASoC: jz4740: Use the generic dmaengine PCM driver") Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Reported-by: Ralf Baechle <ralf(a)linux-mips.org> Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/jz4740/Makefile | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/jz4740/Makefile b/sound/soc/jz4740/Makefile index be873c1..d32c540 100644 --- a/sound/soc/jz4740/Makefile +++ b/sound/soc/jz4740/Makefile @@ -1,10 +1,8 @@ # # Jz4740 Platform Support # -snd-soc-jz4740-objs := jz4740-pcm.o snd-soc-jz4740-i2s-objs := jz4740-i2s.o -obj-$(CONFIG_SND_JZ4740_SOC) += snd-soc-jz4740.o obj-$(CONFIG_SND_JZ4740_SOC_I2S) += snd-soc-jz4740-i2s.o # Jz4740 Machine Support -- 1.8.0

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[alsa-devel] [PATCH] ASoC: rsnd: call rsnd_dai_pointer_update() from outside of lock
by Kuninori Morimoto 23 Apr '14

23 Apr '14

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> rsnd_soc_dai_trigger() will be called after rsnd_dai_pointer_update() function which is using rsnd_lock(). Thus, it should be called from outside of rsnd_lock(). Kernel will be hangup without this patch. Special thanks to Kataoka-san Reported-by: Ryo Kataoka <ryo.kataoka.wt(a)renesas.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 5 ++--- 1 file changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 1f6981a..964463d 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -197,13 +197,12 @@ static void rsnd_dma_complete(void *data) * rsnd_dai_pointer_update() will be called twice, * ant it will breaks io->byte_pos */ - - rsnd_dai_pointer_update(io, io->byte_per_period); - if (dma->submit_loop) rsnd_dma_continue(dma); rsnd_unlock(priv, flags); + + rsnd_dai_pointer_update(io, io->byte_per_period); } static void __rsnd_dma_start(struct rsnd_dma *dma) -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: max98095: Add an explicit of_match_table
by Tushar Behera 23 Apr '14

23 Apr '14

Create an explicit of_match_table entry for MAX98095 codec. Also add a binding Documentation for this compatible string. Signed-off-by: Tushar Behera <tushar.behera(a)linaro.org> --- .../devicetree/bindings/sound/max98095.txt | 16 ++++++++++++++++ sound/soc/codecs/max98095.c | 7 +++++++ 2 files changed, 23 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/max98095.txt diff --git a/Documentation/devicetree/bindings/sound/max98095.txt b/Documentation/devicetree/bindings/sound/max98095.txt new file mode 100644 index 0000000..bacbeaa --- /dev/null +++ b/Documentation/devicetree/bindings/sound/max98095.txt @@ -0,0 +1,16 @@ +MAX98095 audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "maxim,max98095". + +- reg : The I2C address of the device. + +Example: + +max98095: codec@11 { + compatible = "maxim,max98095"; + reg = <0x11>; +}; diff --git a/sound/soc/codecs/max98095.c b/sound/soc/codecs/max98095.c index 03f0536..8e548af 100644 --- a/sound/soc/codecs/max98095.c +++ b/sound/soc/codecs/max98095.c @@ -2399,10 +2399,17 @@ static const struct i2c_device_id max98095_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, max98095_i2c_id); +static const struct of_device_id max98095_of_match[] = { + { .compatible = "maxim,max98095", }, + { } +}; +MODULE_DEVICE_TABLE(of, max98095_of_match); + static struct i2c_driver max98095_i2c_driver = { .driver = { .name = "max98095", .owner = THIS_MODULE, + .of_match_table = of_match_ptr(max98095_of_match), }, .probe = max98095_i2c_probe, .remove = max98095_i2c_remove, -- 1.7.9.5

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