June 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: core - Optimise and refactor pcm_new() to pass only rtd
by Liam Girdwood 07 Jun '11

07 Jun '11

Currently pcm_new() passes in 3 arguments :- card, pcm and DAI. Refactor this to only pass in 1 argument (i.e. the rtd) since struct rtd contains card, pcm and DAI along with other members too that are useful too. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc.h | 3 +-- sound/soc/atmel/atmel-pcm.c | 6 ++++-- sound/soc/au1x/dbdma2.c | 7 ++++--- sound/soc/blackfin/bf5xx-ac97-pcm.c | 6 ++++-- sound/soc/blackfin/bf5xx-i2s-pcm.c | 6 ++++-- sound/soc/blackfin/bf5xx-tdm-pcm.c | 6 ++++-- sound/soc/davinci/davinci-pcm.c | 6 ++++-- sound/soc/ep93xx/ep93xx-pcm.c | 6 ++++-- sound/soc/fsl/fsl_dma.c | 6 ++++-- sound/soc/fsl/mpc5200_dma.c | 7 ++++--- sound/soc/imx/imx-pcm-fiq.c | 8 +++++--- sound/soc/imx/imx-ssi.c | 7 ++++--- sound/soc/imx/imx-ssi.h | 3 +-- sound/soc/jz4740/jz4740-pcm.c | 6 ++++-- sound/soc/kirkwood/kirkwood-dma.c | 6 ++++-- sound/soc/mid-x86/sst_platform.c | 5 +++-- sound/soc/nuc900/nuc900-pcm.c | 7 +++++-- sound/soc/omap/omap-pcm.c | 6 ++++-- sound/soc/pxa/pxa2xx-pcm.c | 6 ++++-- sound/soc/s6000/s6000-pcm.c | 7 ++++--- sound/soc/samsung/dma.c | 6 ++++-- sound/soc/sh/dma-sh7760.c | 6 +++--- sound/soc/sh/fsi.c | 6 +++--- sound/soc/sh/siu_pcm.c | 5 +++-- sound/soc/soc-core.c | 3 +-- sound/soc/tegra/tegra_pcm.c | 6 ++++-- sound/soc/txx9/txx9aclc.c | 5 +++-- 27 files changed, 96 insertions(+), 61 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index baaba6a..d5db87e 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -638,8 +638,7 @@ struct snd_soc_platform_driver { int (*resume)(struct snd_soc_dai *dai); /* pcm creation and destruction */ - int (*pcm_new)(struct snd_card *, struct snd_soc_dai *, - struct snd_pcm *); + int (*pcm_new)(struct snd_soc_pcm_runtime *); void (*pcm_free)(struct snd_pcm *); /* diff --git a/sound/soc/atmel/atmel-pcm.c b/sound/soc/atmel/atmel-pcm.c index 51dde4e..f81d4c3 100644 --- a/sound/soc/atmel/atmel-pcm.c +++ b/sound/soc/atmel/atmel-pcm.c @@ -364,9 +364,11 @@ static struct snd_pcm_ops atmel_pcm_ops = { \*--------------------------------------------------------------------------*/ static u64 atmel_pcm_dmamask = 0xffffffff; -static int atmel_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) +static int atmel_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret = 0; if (!card->dev->dma_mask) diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c index 10fdd28..20bb53a 100644 --- a/sound/soc/au1x/dbdma2.c +++ b/sound/soc/au1x/dbdma2.c @@ -319,10 +319,11 @@ static void au1xpsc_pcm_free_dma_buffers(struct snd_pcm *pcm) snd_pcm_lib_preallocate_free_for_all(pcm); } -static int au1xpsc_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int au1xpsc_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_pcm *pcm = rtd->pcm; + snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV, card->dev, AU1XPSC_BUFFER_MIN_BYTES, (4096 * 1024) - 1); diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c index 98b44b3..9e59f68 100644 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ b/sound/soc/blackfin/bf5xx-ac97-pcm.c @@ -418,9 +418,11 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); -int bf5xx_pcm_ac97_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +int bf5xx_pcm_ac97_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret = 0; pr_debug("%s enter\n", __func__); diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c index b5101ef..96d0d90 100644 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ b/sound/soc/blackfin/bf5xx-i2s-pcm.c @@ -248,9 +248,11 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); -int bf5xx_pcm_i2s_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +int bf5xx_pcm_i2s_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret = 0; pr_debug("%s enter\n", __func__); diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c index 07cfc7a..c95cc03 100644 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.c +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c @@ -283,9 +283,11 @@ static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) static u64 bf5xx_pcm_dmamask = DMA_BIT_MASK(32); -static int bf5xx_pcm_tdm_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int bf5xx_pcm_tdm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret = 0; if (!card->dev->dma_mask) diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index c9e0320..a49e667 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -833,9 +833,11 @@ static void davinci_pcm_free(struct snd_pcm *pcm) static u64 davinci_pcm_dmamask = 0xffffffff; -static int davinci_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) +static int davinci_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret; if (!card->dev->dma_mask) diff --git a/sound/soc/ep93xx/ep93xx-pcm.c b/sound/soc/ep93xx/ep93xx-pcm.c index a456e49..e27c417 100644 --- a/sound/soc/ep93xx/ep93xx-pcm.c +++ b/sound/soc/ep93xx/ep93xx-pcm.c @@ -266,9 +266,11 @@ static void ep93xx_pcm_free_dma_buffers(struct snd_pcm *pcm) static u64 ep93xx_pcm_dmamask = 0xffffffff; -static int ep93xx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int ep93xx_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret = 0; if (!card->dev->dma_mask) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index 15dac0f..50b5df8 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -294,9 +294,11 @@ static irqreturn_t fsl_dma_isr(int irq, void *dev_id) * Regardless of where the memory is actually allocated, since the device can * technically DMA to any 36-bit address, we do need to set the DMA mask to 36. */ -static int fsl_dma_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int fsl_dma_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; static u64 fsl_dma_dmamask = DMA_BIT_MASK(36); int ret; diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index fff695c..19ad0c1 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -299,10 +299,11 @@ static struct snd_pcm_ops psc_dma_ops = { }; static u64 psc_dma_dmamask = 0xffffffff; -static int psc_dma_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int psc_dma_new(struct snd_soc_pcm_runtime *rtd) { - struct snd_soc_pcm_runtime *rtd = pcm->private_data; + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; struct psc_dma *psc_dma = snd_soc_dai_get_drvdata(rtd->cpu_dai); size_t size = psc_dma_hardware.buffer_bytes_max; int rc = 0; diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c index 413b78d..309c59e 100644 --- a/sound/soc/imx/imx-pcm-fiq.c +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -238,12 +238,14 @@ static struct snd_pcm_ops imx_pcm_ops = { static int ssi_irq = 0; -static int imx_pcm_fiq_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int imx_pcm_fiq_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret; - ret = imx_pcm_new(card, dai, pcm); + ret = imx_pcm_new(rtd); if (ret) return ret; diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c index 5b13fec..158a91c 100644 --- a/sound/soc/imx/imx-ssi.c +++ b/sound/soc/imx/imx-ssi.c @@ -388,10 +388,11 @@ static int imx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) static u64 imx_pcm_dmamask = DMA_BIT_MASK(32); -int imx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +int imx_pcm_new(struct snd_soc_pcm_runtime *rtd) { - + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret = 0; if (!card->dev->dma_mask) diff --git a/sound/soc/imx/imx-ssi.h b/sound/soc/imx/imx-ssi.h index dc8a875..0a84cec 100644 --- a/sound/soc/imx/imx-ssi.h +++ b/sound/soc/imx/imx-ssi.h @@ -225,8 +225,7 @@ struct snd_soc_platform *imx_ssi_dma_mx2_init(struct platform_device *pdev, struct imx_ssi *ssi); int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma); -int imx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm); +int imx_pcm_new(struct snd_soc_pcm_runtime *rtd); void imx_pcm_free(struct snd_pcm *pcm); /* diff --git a/sound/soc/jz4740/jz4740-pcm.c b/sound/soc/jz4740/jz4740-pcm.c index fb1483f..a7c9578 100644 --- a/sound/soc/jz4740/jz4740-pcm.c +++ b/sound/soc/jz4740/jz4740-pcm.c @@ -299,9 +299,11 @@ static void jz4740_pcm_free(struct snd_pcm *pcm) static u64 jz4740_pcm_dmamask = DMA_BIT_MASK(32); -int jz4740_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +int jz4740_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret = 0; if (!card->dev->dma_mask) diff --git a/sound/soc/kirkwood/kirkwood-dma.c b/sound/soc/kirkwood/kirkwood-dma.c index e13c6ce..cd33de1 100644 --- a/sound/soc/kirkwood/kirkwood-dma.c +++ b/sound/soc/kirkwood/kirkwood-dma.c @@ -312,9 +312,11 @@ static int kirkwood_dma_preallocate_dma_buffer(struct snd_pcm *pcm, return 0; } -static int kirkwood_dma_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) +static int kirkwood_dma_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret; if (!card->dev->dma_mask) diff --git a/sound/soc/mid-x86/sst_platform.c b/sound/soc/mid-x86/sst_platform.c index 5a946b4..3e78260 100644 --- a/sound/soc/mid-x86/sst_platform.c +++ b/sound/soc/mid-x86/sst_platform.c @@ -402,9 +402,10 @@ static void sst_pcm_free(struct snd_pcm *pcm) snd_pcm_lib_preallocate_free_for_all(pcm); } -int sst_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +int sst_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int retval = 0; pr_debug("sst_pcm_new called\n"); diff --git a/sound/soc/nuc900/nuc900-pcm.c b/sound/soc/nuc900/nuc900-pcm.c index 8263f56..d589ef1 100644 --- a/sound/soc/nuc900/nuc900-pcm.c +++ b/sound/soc/nuc900/nuc900-pcm.c @@ -315,9 +315,12 @@ static void nuc900_dma_free_dma_buffers(struct snd_pcm *pcm) } static u64 nuc900_pcm_dmamask = DMA_BIT_MASK(32); -static int nuc900_dma_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) +static int nuc900_dma_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; + if (!card->dev->dma_mask) card->dev->dma_mask = &nuc900_pcm_dmamask; if (!card->dev->coherent_dma_mask) diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c index e6a6b99..b2f5751 100644 --- a/sound/soc/omap/omap-pcm.c +++ b/sound/soc/omap/omap-pcm.c @@ -366,9 +366,11 @@ static void omap_pcm_free_dma_buffers(struct snd_pcm *pcm) } } -static int omap_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret = 0; if (!card->dev->dma_mask) diff --git a/sound/soc/pxa/pxa2xx-pcm.c b/sound/soc/pxa/pxa2xx-pcm.c index 2ce0b2d..d73d6f6 100644 --- a/sound/soc/pxa/pxa2xx-pcm.c +++ b/sound/soc/pxa/pxa2xx-pcm.c @@ -85,9 +85,11 @@ static struct snd_pcm_ops pxa2xx_pcm_ops = { static u64 pxa2xx_pcm_dmamask = DMA_BIT_MASK(32); -static int pxa2xx_soc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int pxa2xx_soc_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret = 0; if (!card->dev->dma_mask) diff --git a/sound/soc/s6000/s6000-pcm.c b/sound/soc/s6000/s6000-pcm.c index ab3ccae..80c85fd 100644 --- a/sound/soc/s6000/s6000-pcm.c +++ b/sound/soc/s6000/s6000-pcm.c @@ -443,10 +443,11 @@ static void s6000_pcm_free(struct snd_pcm *pcm) static u64 s6000_pcm_dmamask = DMA_BIT_MASK(32); -static int s6000_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) +static int s6000_pcm_new(struct snd_soc_pcm_runtime *runtime) { - struct snd_soc_pcm_runtime *runtime = pcm->private_data; + struct snd_card *card = runtime->card->snd_card; + struct snd_soc_dai *dai = runtime->cpu_dai; + struct snd_pcm *pcm = runtime->pcm; struct s6000_pcm_dma_params *params; int res; diff --git a/sound/soc/samsung/dma.c b/sound/soc/samsung/dma.c index 5cb3b88..9465588 100644 --- a/sound/soc/samsung/dma.c +++ b/sound/soc/samsung/dma.c @@ -425,9 +425,11 @@ static void dma_free_dma_buffers(struct snd_pcm *pcm) static u64 dma_mask = DMA_BIT_MASK(32); -static int dma_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) +static int dma_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret = 0; pr_debug("Entered %s\n", __func__); diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c index c326d29..db74005 100644 --- a/sound/soc/sh/dma-sh7760.c +++ b/sound/soc/sh/dma-sh7760.c @@ -327,10 +327,10 @@ static void camelot_pcm_free(struct snd_pcm *pcm) snd_pcm_lib_preallocate_free_for_all(pcm); } -static int camelot_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int camelot_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_pcm *pcm = rtd->pcm; + /* dont use SNDRV_DMA_TYPE_DEV, since it will oops the SH kernel * in MMAP mode (i.e. aplay -M) */ diff --git a/sound/soc/sh/fsi.c b/sound/soc/sh/fsi.c index d2f17ce..8e112cc 100644 --- a/sound/soc/sh/fsi.c +++ b/sound/soc/sh/fsi.c @@ -1168,10 +1168,10 @@ static void fsi_pcm_free(struct snd_pcm *pcm) snd_pcm_lib_preallocate_free_for_all(pcm); } -static int fsi_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int fsi_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_pcm *pcm = rtd->pcm; + /* * dont use SNDRV_DMA_TYPE_DEV, since it will oops the SH kernel * in MMAP mode (i.e. aplay -M) diff --git a/sound/soc/sh/siu_pcm.c b/sound/soc/sh/siu_pcm.c index a423bab..f8f6816 100644 --- a/sound/soc/sh/siu_pcm.c +++ b/sound/soc/sh/siu_pcm.c @@ -527,10 +527,11 @@ static snd_pcm_uframes_t siu_pcm_pointer_dma(struct snd_pcm_substream *ss) return bytes_to_frames(ss->runtime, ptr); } -static int siu_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int siu_pcm_new(struct snd_soc_pcm_runtime *rtd) { /* card->dev == socdev->dev, see snd_soc_new_pcms() */ + struct snd_card *card = rtd->card->snd_card; + struct snd_pcm *pcm = rtd->pcm; struct siu_info *info = siu_i2s_data; struct platform_device *pdev = to_platform_device(card->dev); int ret; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index ae4c5e4..df5608d 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2161,8 +2161,7 @@ static int soc_new_pcm(struct snd_soc_pcm_runtime *rtd, int num) snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, &soc_pcm_ops); if (platform->driver->pcm_new) { - ret = platform->driver->pcm_new(rtd->card->snd_card, - codec_dai, pcm); + ret = platform->driver->pcm_new(rtd); if (ret < 0) { pr_err("asoc: platform pcm constructor failed\n"); return ret; diff --git a/sound/soc/tegra/tegra_pcm.c b/sound/soc/tegra/tegra_pcm.c index 3c271f9..ff86e5e 100644 --- a/sound/soc/tegra/tegra_pcm.c +++ b/sound/soc/tegra/tegra_pcm.c @@ -322,9 +322,11 @@ static void tegra_pcm_deallocate_dma_buffer(struct snd_pcm *pcm, int stream) static u64 tegra_dma_mask = DMA_BIT_MASK(32); -static int tegra_pcm_new(struct snd_card *card, - struct snd_soc_dai *dai, struct snd_pcm *pcm) +static int tegra_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_card *card = rtd->card->snd_card; + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; int ret = 0; if (!card->dev->dma_mask) diff --git a/sound/soc/txx9/txx9aclc.c b/sound/soc/txx9/txx9aclc.c index f4aa4e0..34aa972 100644 --- a/sound/soc/txx9/txx9aclc.c +++ b/sound/soc/txx9/txx9aclc.c @@ -288,9 +288,10 @@ static void txx9aclc_pcm_free_dma_buffers(struct snd_pcm *pcm) snd_pcm_lib_preallocate_free_for_all(pcm); } -static int txx9aclc_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, - struct snd_pcm *pcm) +static int txx9aclc_pcm_new(struct snd_soc_pcm_runtime *rtd) { + struct snd_soc_dai *dai = rtd->cpu_dai; + struct snd_pcm *pcm = rtd->pcm; struct platform_device *pdev = to_platform_device(dai->platform->dev); struct txx9aclc_soc_device *dev; struct resource *r; -- 1.7.4.1

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[alsa-devel] [PATCH] ASoC: core - Allow components to probe/remove in sequence.
by Liam Girdwood 07 Jun '11

07 Jun '11

Some ASoC components depend on other ASoC components to provide clocks and power resources in order to probe() and vice versa for remove(). Allow components to be ordered so that components can be probed() and removed() in sequences that conform to their dependencies. Signed-off-by: Liam Girdwood <lrg(a)ti.com> --- include/sound/soc-dai.h | 4 +++ include/sound/soc.h | 18 ++++++++++++++ sound/soc/soc-core.c | 60 ++++++++++++++++++++++++++++++---------------- 3 files changed, 61 insertions(+), 21 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 1bafe95..5ad5f3a 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -209,6 +209,10 @@ struct snd_soc_dai_driver { struct snd_soc_pcm_stream capture; struct snd_soc_pcm_stream playback; unsigned int symmetric_rates:1; + + /* probe ordering - for components with runtime dependencies */ + int probe_order; + int remove_order; }; /* diff --git a/include/sound/soc.h b/include/sound/soc.h index f55efc1..baaba6a 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -203,6 +203,16 @@ SOC_VALUE_ENUM_DOUBLE_DECL(name, xreg, xshift, xshift, xmask, xtexts, xvalues) /* + * Component probe and remove ordering levels for components with runtime + * dependencies. + */ +#define SND_SOC_COMP_ORDER_FIRST -2 +#define SND_SOC_COMP_ORDER_EARLY -1 +#define SND_SOC_COMP_ORDER_NORMAL 0 +#define SND_SOC_COMP_ORDER_LATE 1 +#define SND_SOC_COMP_ORDER_LAST 2 + +/* * Bias levels * * @ON: Bias is fully on for audio playback and capture operations. @@ -613,6 +623,10 @@ struct snd_soc_codec_driver { void (*seq_notifier)(struct snd_soc_dapm_context *, enum snd_soc_dapm_type, int); + + /* probe ordering - for components with runtime dependencies */ + int probe_order; + int remove_order; }; /* SoC platform interface */ @@ -637,6 +651,10 @@ struct snd_soc_platform_driver { /* platform stream ops */ struct snd_pcm_ops *ops; + + /* probe ordering - for components with runtime dependencies */ + int probe_order; + int remove_order; }; struct snd_soc_platform { diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c53f5d5..ae4c5e4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1394,7 +1394,7 @@ static void soc_remove_codec(struct snd_soc_codec *codec) module_put(codec->dev->driver->owner); } -static void soc_remove_dai_link(struct snd_soc_card *card, int num) +static void soc_remove_dai_link(struct snd_soc_card *card, int num, int order) { struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; struct snd_soc_codec *codec = rtd->codec; @@ -1411,7 +1411,8 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num) } /* remove the CODEC DAI */ - if (codec_dai && codec_dai->probed) { + if (codec_dai && codec_dai->probed && + codec_dai->driver->remove_order == order) { if (codec_dai->driver->remove) { err = codec_dai->driver->remove(codec_dai); if (err < 0) @@ -1422,7 +1423,8 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num) } /* remove the platform */ - if (platform && platform->probed) { + if (platform && platform->probed && + platform->driver->remove_order == order) { if (platform->driver->remove) { err = platform->driver->remove(platform); if (err < 0) @@ -1434,11 +1436,13 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num) } /* remove the CODEC */ - if (codec && codec->probed) + if (codec && codec->probed && + codec->driver->remove_order == order) soc_remove_codec(codec); /* remove the cpu_dai */ - if (cpu_dai && cpu_dai->probed) { + if (cpu_dai && cpu_dai->probed && + cpu_dai->driver->remove_order == order) { if (cpu_dai->driver->remove) { err = cpu_dai->driver->remove(cpu_dai); if (err < 0) @@ -1452,11 +1456,13 @@ static void soc_remove_dai_link(struct snd_soc_card *card, int num) static void soc_remove_dai_links(struct snd_soc_card *card) { - int i; - - for (i = 0; i < card->num_rtd; i++) - soc_remove_dai_link(card, i); + int dai, order; + for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; + order++) { + for (dai = 0; dai < card->num_rtd; dai++) + soc_remove_dai_link(card, dai, order); + } card->num_rtd = 0; } @@ -1597,7 +1603,7 @@ static int soc_post_component_init(struct snd_soc_card *card, return 0; } -static int soc_probe_dai_link(struct snd_soc_card *card, int num) +static int soc_probe_dai_link(struct snd_soc_card *card, int num, int order) { struct snd_soc_dai_link *dai_link = &card->dai_link[num]; struct snd_soc_pcm_runtime *rtd = &card->rtd[num]; @@ -1606,7 +1612,8 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) struct snd_soc_dai *codec_dai = rtd->codec_dai, *cpu_dai = rtd->cpu_dai; int ret; - dev_dbg(card->dev, "probe %s dai link %d\n", card->name, num); + dev_dbg(card->dev, "probe %s dai link %d late %d\n", + card->name, num, order); /* config components */ codec_dai->codec = codec; @@ -1618,7 +1625,8 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) rtd->pmdown_time = pmdown_time; /* probe the cpu_dai */ - if (!cpu_dai->probed) { + if (!cpu_dai->probed && + cpu_dai->driver->probe_order == order) { if (!try_module_get(cpu_dai->dev->driver->owner)) return -ENODEV; @@ -1637,14 +1645,16 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) } /* probe the CODEC */ - if (!codec->probed) { + if (!codec->probed && + codec->driver->probe_order == order) { ret = soc_probe_codec(card, codec); if (ret < 0) return ret; } /* probe the platform */ - if (!platform->probed) { + if (!platform->probed && + platform->driver->probe_order == order) { if (!try_module_get(platform->dev->driver->owner)) return -ENODEV; @@ -1663,7 +1673,7 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) } /* probe the CODEC DAI */ - if (!codec_dai->probed) { + if (!codec_dai->probed && codec_dai->driver->probe_order == order) { if (codec_dai->driver->probe) { ret = codec_dai->driver->probe(codec_dai); if (ret < 0) { @@ -1678,6 +1688,10 @@ static int soc_probe_dai_link(struct snd_soc_card *card, int num) list_add(&codec_dai->card_list, &card->dai_dev_list); } + /* complete DAI probe during last probe */ + if (order != SND_SOC_COMP_ORDER_LAST) + return 0; + /* DAPM dai link stream work */ INIT_DELAYED_WORK(&rtd->delayed_work, close_delayed_work); @@ -1818,7 +1832,7 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) struct snd_soc_codec *codec; struct snd_soc_codec_conf *codec_conf; enum snd_soc_compress_type compress_type; - int ret, i; + int ret, i, order; mutex_lock(&card->mutex); @@ -1896,12 +1910,16 @@ static void snd_soc_instantiate_card(struct snd_soc_card *card) goto card_probe_error; } - for (i = 0; i < card->num_links; i++) { - ret = soc_probe_dai_link(card, i); - if (ret < 0) { - pr_err("asoc: failed to instantiate card %s: %d\n", + /* early DAI link probe */ + for (order = SND_SOC_COMP_ORDER_FIRST; order <= SND_SOC_COMP_ORDER_LAST; + order++) { + for (i = 0; i < card->num_links; i++) { + ret = soc_probe_dai_link(card, i, order); + if (ret < 0) { + pr_err("asoc: failed to instantiate card %s: %d\n", card->name, ret); - goto probe_dai_err; + goto probe_dai_err; + } } } -- 1.7.4.1

2 1

[alsa-devel] [PATCH] ASoC: twl6040 - According to TWL6040 specification, gain start at 6dB and not -6dB.
by Liam Girdwood 07 Jun '11

07 Jun '11

Signed-off-by: Ricardo Neri <ricardo.neri(a)ti.com> --- sound/soc/codecs/twl6040.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index 4c33663..cd63bba 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -954,9 +954,9 @@ static DECLARE_TLV_DB_SCALE(mic_preamp_tlv, -600, 600, 0); /* * MICGAIN volume control: - * from -6 to 30 dB in 6 dB steps + * from 6 to 30 dB in 6 dB steps */ -static DECLARE_TLV_DB_SCALE(mic_amp_tlv, -600, 600, 0); +static DECLARE_TLV_DB_SCALE(mic_amp_tlv, 600, 600, 0); /* * AFMGAIN volume control: -- 1.7.4.1

2 2

[alsa-devel] MIDI issue with RME HDSPM MADI
by Jörn Nettingsmeier 07 Jun '11

07 Jun '11

Hi *! The HDSPM MADI driver seems to have an issue with its MIDI implementation. It is rock-solid for audio under Jack at 64 frames/period, but as soon as I start using the MIDI over MADI interface (to remote-control my Micstasy preamps), I'm getting xruns every ten seconds or so. I've tried to debug the issue with the help of faberman, and he came up with a midi drain() implementation for the driver, but it didn't help with the xrun issue. It did help with busy MIDI programs in that they are no longer stuck in D state almost all the time, but sleeping, like any good program should. This is a 2.6.39-rc7 kernel, and faberman has confirmed the issue with .39. The attached test program rawmiditest.c (build with -lasound) will reproducibly cause xruns as soon as there is a mictasy on the MADI ring so that there's actual traffic. With no-one listening, there will be no xruns. (Might that be a hint that the problem is at the receiving end?) This is really totally over my head, but I found one place in the driver (snd_hdspm_midi_input_trigger) where a midi routine locks the entire hdspm structure. it's flushing the midi input with the hdspm spinlock held - could this be taking so long as to cause xruns? I'm waiting for a MIDI cable to arrive so that I can test if the same problem occurs when using the normal MIDI interfaces of the card (i.e. not the MIDI over MADI one)... For the record, faberman suspected that there is a bit too much locking going on wrt the hmidi struct. OTOH, when we removed all hmidi spinlocks, I was able to lock up the machine twice... maybe there is some middle ground? Meanwhile, any insights would be appreciated. Best, Jörn -- Jörn Nettingsmeier Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487 Meister für Veranstaltungstechnik (Bühne/Studio) Tonmeister (VDT) http://stackingdwarves.net

2 5

[alsa-devel] [RFC PATCH] Add combined clock support to Atmel SoC audio
by Ryan Mallon 07 Jun '11

07 Jun '11

The following patch is one that has been floating around in various forms in our own internal trees for a while. The Atmel SSC peripheral has seperate TX and RX clocks which use separate pins from the the micro. TF (frame) and TK (clock) for transmit and RF and RK for receive. Not all codecs have separate frame and bit clocks for transmit and receive so we want to be able to do both playback and capture using a single set of pins. This patch introduces a combined clock mode for the Atmel SSC peripheral. Which allows playback and capture to use a single set of pins. Currently combined clock is only supported on the TF/TK pins (some incomplete support exists for using RF/RK). I have tested this patch on our AT91SAM9G45 + TLV320AIC26 platform. Playback and capture work individually. Simultaneous playback and capture have been tested by connecting a loopback cable on the linein and lineout jacks and then doing: arecord -c 2 -f S16_LE -r 44100 > recording.wav & aplay 500hz_sine.wav This patch is posted as RFC since the approach is incomplete and a bit hackish. I am mostly interested in knowing if this is a sensible approach, and could be cleaned up for mainline inclusion, or if there is a better way to do this. Signed-off-by: Ryan Mallon <ryan(a)bluewatersys.com> --- diff --git a/sound/soc/atmel/atmel_ssc_dai.c b/sound/soc/atmel/atmel_ssc_dai.c index 5d230ce..ee00172 100644 --- a/sound/soc/atmel/atmel_ssc_dai.c +++ b/sound/soc/atmel/atmel_ssc_dai.c @@ -43,6 +43,7 @@ #include <sound/soc.h> #include <mach/hardware.h> +#include <asm/atomic.h> #include "atmel-pcm.h" #include "atmel_ssc_dai.h" @@ -246,7 +247,18 @@ static void atmel_ssc_shutdown(struct snd_pcm_substream *substream, dma_params = ssc_p->dma_params[dir]; if (dma_params != NULL) { - ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_disable); + if (ssc_p->combined_clock) { + /* + * When using a combined clock we only disable the + * clock once all substreams have completed + */ + if (atomic_dec_and_test(&ssc_p->substreams_running)) + ssc_writel(ssc_p->ssc->regs, CR, + dma_params->mask->ssc_disable); + } else { + ssc_writel(ssc_p->ssc->regs, CR, + dma_params->mask->ssc_disable); + } pr_debug("atmel_ssc_shutdown: %s disabled SSC_SR=0x%08x\n", (dir ? "receive" : "transmit"), ssc_readl(ssc_p->ssc->regs, SR)); @@ -328,6 +340,21 @@ static int atmel_ssc_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, return 0; } +void atmel_ssc_setup_combined_clock(struct atmel_ssc_info *ssc_p, + int combined_clock) +{ + int i; + + ssc_p->combined_clock = combined_clock; + for (i = 0; i < ARRAY_SIZE(ssc_dma_params); i++) { + if (ssc_p->combined_clock == ATMEL_SSC_CLOCK_TX_ON_RX) + ssc_dma_params[i][0].mask->ssc_enable |= SSC_BIT(CR_RXEN); + else if (ssc_p->combined_clock == ATMEL_SSC_CLOCK_RX_ON_TX) + ssc_dma_params[i][1].mask->ssc_enable |= SSC_BIT(CR_TXEN); + } +} +EXPORT_SYMBOL(atmel_ssc_setup_combined_clock); + /* * Configure the SSC. */ @@ -548,6 +575,34 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, ssc_p->daifmt); return -EINVAL; } + + if (ssc_p->combined_clock == ATMEL_SSC_CLOCK_RX_ON_TX) { + /* RX clock is always running */ + rcmr &= ~SSC_BF(RCMR_CKO, 0x7); + rcmr |= SSC_BF(RCMR_CKO, SSC_CKO_CONTINUOUS); + + /* TX clock is always running */ + tcmr &= ~SSC_BF(TCMR_CKO, 0x7); + tcmr |= SSC_BF(TCMR_CKO, SSC_CKO_CONTINUOUS); + + /* RX clock is sourced from TK pin */ + rcmr &= ~SSC_BF(RCMR_CKS, 0x7); + rcmr |= SSC_BF(RCMR_CKS, SSC_CKS_CLOCK); + + /* Start RX on TX start */ + rcmr &= ~SSC_BF(RCMR_START, 0xf); + rcmr |= SSC_BF(RCMR_START, SSC_START_TX_RX); + + if (dir == 1) { + /* + * Set the TX clock period to the RX clock period + * FIXME - Is this okay if we are already doing TX? + */ + tcmr &= 0x00ffffff; + tcmr |= rcmr & 0xff000000; + } + } + pr_debug("atmel_ssc_hw_params: " "RCMR=%08x RFMR=%08x TCMR=%08x TFMR=%08x\n", rcmr, rfmr, tcmr, tfmr); @@ -581,6 +636,9 @@ static int atmel_ssc_hw_params(struct snd_pcm_substream *substream, return ret; } + if (ssc_p->combined_clock) + atomic_set(&ssc_p->substreams_running, 0); + ssc_p->initialized = 1; } @@ -616,6 +674,21 @@ static int atmel_ssc_prepare(struct snd_pcm_substream *substream, ssc_writel(ssc_p->ssc->regs, CR, dma_params->mask->ssc_enable); + if (ssc_p->combined_clock) { + atomic_inc(&ssc_p->substreams_running); + + if (ssc_p->combined_clock == ATMEL_SSC_CLOCK_RX_ON_TX && + dir == 1) { + /* + * Jump start the dma for capture by loading the + * transmit holding register with a dummy value. + */ + pr_debug("Jump starting SSC RX DMA\n"); + if (!(ssc_readl(ssc_p->ssc->regs, SR) & (1 << 16))) + ssc_writel(ssc_p->ssc->regs, THR, 0); + } + } + pr_debug("%s enabled SSC_SR=0x%08x\n", dir ? "receive" : "transmit", ssc_readl(ssc_p->ssc->regs, SR)); diff --git a/sound/soc/atmel/atmel_ssc_dai.h b/sound/soc/atmel/atmel_ssc_dai.h index 5d4f0f9..6b20ee6 100644 --- a/sound/soc/atmel/atmel_ssc_dai.h +++ b/sound/soc/atmel/atmel_ssc_dai.h @@ -102,6 +102,8 @@ struct atmel_ssc_state { u32 ssc_imr; }; +#define ATMEL_SSC_CLOCK_RX_ON_TX 1 +#define ATMEL_SSC_CLOCK_TX_ON_RX 2 struct atmel_ssc_info { char *name; @@ -115,8 +117,14 @@ struct atmel_ssc_info { unsigned short rcmr_period; struct atmel_pcm_dma_params *dma_params[2]; struct atmel_ssc_state ssc_state; + + /* For combined clocks */ + int combined_clock; + atomic_t substreams_running; }; int atmel_ssc_set_audio(int ssc); +extern void atmel_ssc_setup_combined_clock(struct atmel_ssc_info *ssc_p, + int combined_clock); #endif /* _AT91_SSC_DAI_H */

3 6

[alsa-devel] [PATCH] RME HDSPM MADI lock fix
by Jörn Nettingsmeier 07 Jun '11

07 Jun '11

hi *! while we were trying to debug the midi-related xruns in the current driver, i came across what looks like an unrelated locking bug to me: static int snd_hdspm_midi_input_read (struct hdspm_midi *hmidi) ... spin_lock_irqsave (&hmidi->lock, flags); <--------------- n_pending = snd_hdspm_midi_input_available (hmidi->hdspm, hmidi->id); if (n_pending > 0) { if (hmidi->input) { if (n_pending > (int)sizeof (buf)) n_pending = sizeof (buf); for (i = 0; i < n_pending; ++i) buf[i] = snd_hdspm_midi_read_byte (hmidi->hdspm, hmidi->id); if (n_pending) snd_rawmidi_receive (hmidi->input, buf, n_pending); } else { /* flush the MIDI input FIFO */ while (n_pending--) snd_hdspm_midi_read_byte (hmidi->hdspm, hmidi->id); } } hmidi->pending = 0; hmidi->hdspm->control_register |= hmidi->ie;<------- !!! hdspm_write(hmidi->hdspm, HDSPM_controlRegister, hmidi->hdspm->control_register); spin_unlock_irqrestore (&hmidi->lock, flags);<--------- return snd_hdspm_midi_output_write (hmidi); } if i understand this whole business correctly, we should be holding the hdspm lock here, not the hmidi one. please review the attached patch and apply if you think it's ok. best, jörn -- Jörn Nettingsmeier Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487 Meister für Veranstaltungstechnik (Bühne/Studio) Tonmeister (VDT) http://stackingdwarves.net

3 7

[alsa-devel] [PATCH] ASoC: Blackfin: bf5xx-ad1836: Fix codec device name
by Lars-Peter Clausen 07 Jun '11

07 Jun '11

Fix the codec_name field of the dai_link to match the actual device name of the codec. Otherwise the card won't be instantiated. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/blackfin/bf5xx-ad1836.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c index ea4951c..f79d165 100644 --- a/sound/soc/blackfin/bf5xx-ad1836.c +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -75,7 +75,7 @@ static struct snd_soc_dai_link bf5xx_ad1836_dai[] = { .cpu_dai_name = "bfin-tdm.0", .codec_dai_name = "ad1836-hifi", .platform_name = "bfin-tdm-pcm-audio", - .codec_name = "ad1836.0", + .codec_name = "spi0.4", .ops = &bf5xx_ad1836_ops, }, { @@ -84,7 +84,7 @@ static struct snd_soc_dai_link bf5xx_ad1836_dai[] = { .cpu_dai_name = "bfin-tdm.1", .codec_dai_name = "ad1836-hifi", .platform_name = "bfin-tdm-pcm-audio", - .codec_name = "ad1836.0", + .codec_name = "spi0.4", .ops = &bf5xx_ad1836_ops, }, }; -- 1.7.2.5

3 2

[alsa-devel] [PATCH] ASoC: AD1836: Fix build error
by Lars-Peter Clausen 07 Jun '11

07 Jun '11

Commit f97d0c6d5f94 ("ASoC: AD1836: Add input gain control for ADC2") contained a typo in the register name, causing a build error. This patch fixes it. Signed-off-by: Lars-Peter Clausen <lars(a)metafoo.de> --- sound/soc/codecs/ad1836.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/ad1836.c b/sound/soc/codecs/ad1836.c index 1e9a801..e3a9493 100644 --- a/sound/soc/codecs/ad1836.c +++ b/sound/soc/codecs/ad1836.c @@ -127,7 +127,7 @@ static const struct snd_soc_dapm_route ad183x_dapm_routes[] = { static const DECLARE_TLV_DB_SCALE(ad1836_in_tlv, 0, 300, 0); static const struct snd_kcontrol_new ad1836_controls[] = { - SOC_DOUBLE_TLV("ADC2 Capture Volume", AD183X_ADC_CTRL1, 3, 0, 4, 0, + SOC_DOUBLE_TLV("ADC2 Capture Volume", AD1836_ADC_CTRL1, 3, 0, 4, 0, ad1836_in_tlv), }; -- 1.7.2.5

3 2

Re: [alsa-devel] [PATCH] wm8940: remove unecessary if statement
by Jonathan Cameron 07 Jun '11

07 Jun '11

On 06/06/11 01:47, Greg Dietsche wrote: > the code always returns ret regardless, so if(ret) check is unecessary. Good point, though please spell check your commit messages. unecessary -> unnecessary Also if you want to do this sort of cleanup, please also fix the equivalent in wm8940_resume and wm8940_add_widgets. Ack is for what is here, plus those if you do them. Just as an aside, there is no earthly point in cc'ing lkml for a simple cleanup like this. Just adds to already huge amount of noise! > > Signed-off-by: Greg Dietsche <Gregory.Dietsche(a)cuw.edu> Acked-by: Jonathan Cameron <jic23(a)cam.ac.uk> > --- > sound/soc/codecs/wm8940.c | 3 --- > 1 files changed, 0 insertions(+), 3 deletions(-) > > diff --git a/sound/soc/codecs/wm8940.c b/sound/soc/codecs/wm8940.c > index 25580e3..1ada6d3 100644 > --- a/sound/soc/codecs/wm8940.c > +++ b/sound/soc/codecs/wm8940.c > @@ -730,9 +730,6 @@ static int wm8940_probe(struct snd_soc_codec *codec) > if (ret) > return ret; > ret = wm8940_add_widgets(codec); > - if (ret) > - return ret; > - > return ret; > } >

4 8

[alsa-devel] [PATCH] hdspm.c: make driver name generic to avoid user confusion
by Jörn Nettingsmeier 07 Jun '11

07 Jun '11

hi adi, *! here's a trivial patch to change the driver name of hdspm.c to avoid irritating users who don't have the MADI version (just happened on the LAU list). iiuc, no external software should make any assumptions about this string, so there should be no side effects, but i have been wrong before :) please review and apply. best, jörn -- Jörn Nettingsmeier Lortzingstr. 11, 45128 Essen, Tel. +49 177 7937487 Meister für Veranstaltungstechnik (Bühne/Studio) Tonmeister (VDT) http://stackingdwarves.net

1 0