November 2013 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] (no subject)
by Xiubo Li 20 Nov '13

20 Nov '13

Hello, This patch series is mostly Freescale's SAI SoC Digital Audio Interface driver implementation. And the implementation is only compatible with device tree definition. This patch series is based on linux-next and has been tested on Vybrid VF610 Tower board using device tree. Changed in v2: - Use default settings for the generic dmaengine PCM driver. - Separate receive and transmit setting in most functions, but some couldn't for the HW limitation. - Drop some not reduntant code. - Use devm_snd_soc_register_component() instead of snd_soc_register_component(). - Use devm_snd_soc_register_card() instead of devm_snd_soc_register_card(). - Adjust the code sentences sequence. - Make the namespacing consistent. - Rename CONFIG_SND_SOC_FSL_SGTL5000 to CONFIG_SND_SOC_FSL_SGTL5000_VF610. - Drop some meaningless lines. - Rename the binding document file. Added in v1: - Add SAI SoC Digital Audio Interface driver. - Add Freescale SAI ALSA SoC Digital Audio Interface node for VF610. - Enables SAI ALSA SoC DAI device for Vybrid VF610 TOWER board. - Add device tree bindings for Freescale SAI. - Revise the bugs about the sgt15000 codec. - Add SGT15000 based audio machine driver. - Enable SGT15000 codec based audio driver node for VF610. - Add device tree bindings for Freescale VF610 sound.

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[alsa-devel] [PATCH] ASoC: soc-pcm: fix incorrect channels and sample_bits wiping off
by Nicolin Chen 20 Nov '13

20 Nov '13

This patch wipes off the corresponding channels and sample_bits for codec_dai. Signed-off-by: Nicolin Chen <b42378(a)freescale.com> --- @Mark Brown Sir, I find the previous symmetry patch hasn't been merged into for-next. So are you willing to apply this one or let me refine the previous PATCH and resend that one? sound/soc/soc-pcm.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index e1c5a00..60d9cfb 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -452,8 +452,8 @@ static int soc_pcm_close(struct snd_pcm_substream *substream) if (!codec_dai->active) { codec_dai->rate = 0; - cpu_dai->channels = 0; - cpu_dai->sample_bits = 0; + codec_dai->channels = 0; + codec_dai->sample_bits = 0; } /* Muting the DAC suppresses artifacts caused during digital -- 1.8.4

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[alsa-devel] [PATCH 08/10] ASoC: ux500_pcm: Differentiate between pdata and DT initialisation
by Lee Jones 19 Nov '13

19 Nov '13

If booting with full DT support (i.e. DMA too, the last piece of the puzzle), then we don't need to use the compatible request channel call back and, due to the work we laid down earlier in this patch-set, we can use core function calls to populate the DMA slave_config. We also require slightly different flags to inform the core that we 'are' booting with DT. Cc: alsa-devel(a)alsa-project.org Cc: Mark Brown <broonie(a)kernel.org> Acked-by: Linus Walleij <linus.walleij(a)linaro.org> Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- sound/soc/ux500/ux500_pcm.c | 28 +++++++++++++++++++++++----- 1 file changed, 23 insertions(+), 5 deletions(-) diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c index ce554de..acfec98 100644 --- a/sound/soc/ux500/ux500_pcm.c +++ b/sound/soc/ux500/ux500_pcm.c @@ -139,15 +139,33 @@ static const struct snd_dmaengine_pcm_config ux500_dmaengine_pcm_config = { .prepare_slave_config = ux500_pcm_prepare_slave_config, }; +static const struct snd_dmaengine_pcm_config ux500_dmaengine_of_pcm_config = { + .pcm_hardware = &ux500_pcm_hw, + .prealloc_buffer_size = 128 * 1024, + .prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config, +}; + int ux500_pcm_register_platform(struct platform_device *pdev) { + const struct snd_dmaengine_pcm_config *pcm_config; + struct device_node *np = pdev->dev.of_node; + unsigned int pcm_flags; int ret; - ret = snd_dmaengine_pcm_register(&pdev->dev, - &ux500_dmaengine_pcm_config, - SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | - SND_DMAENGINE_PCM_FLAG_COMPAT | - SND_DMAENGINE_PCM_FLAG_NO_DT); + if (np) { + pcm_config = &ux500_dmaengine_of_pcm_config; + + pcm_flags = SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | + SND_DMAENGINE_PCM_FLAG_COMPAT; + } else { + pcm_config = &ux500_dmaengine_pcm_config; + + pcm_flags = SND_DMAENGINE_PCM_FLAG_NO_RESIDUE | + SND_DMAENGINE_PCM_FLAG_NO_DT | + SND_DMAENGINE_PCM_FLAG_COMPAT; + } + + ret = snd_dmaengine_pcm_register(&pdev->dev, pcm_config, pcm_flags); if (ret < 0) { dev_err(&pdev->dev, "%s: ERROR: Failed to register platform '%s' (%d)!\n", -- 1.8.1.2

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[alsa-devel] [PATCH v2 1/2] ASOC: tegra: fix matching of AC97 components
by Lucas Stach 19 Nov '13

19 Nov '13

Matching works completely based on the cpu of_node. Signed-off-by: Lucas Stach <dev(a)lynxeye.de> Acked-by: Stephen Warren <swarren(a)nvidia.com> --- v2: no changes CC: <alsa-devel(a)alsa-project.org> CC: Mark Brown <broonie(a)linaro.org> CC: Stephen Warren <swarren(a)wwwdotorg.org> --- sound/soc/tegra/tegra_wm9712.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/tegra/tegra_wm9712.c b/sound/soc/tegra/tegra_wm9712.c index 5e11963..45b5789 100644 --- a/sound/soc/tegra/tegra_wm9712.c +++ b/sound/soc/tegra/tegra_wm9712.c @@ -55,7 +55,6 @@ static int tegra_wm9712_init(struct snd_soc_pcm_runtime *rtd) static struct snd_soc_dai_link tegra_wm9712_dai = { .name = "AC97 HiFi", .stream_name = "AC97 HiFi", - .cpu_dai_name = "tegra20-ac97", .codec_dai_name = "wm9712-hifi", .codec_name = "wm9712-codec", .init = tegra_wm9712_init, -- 1.8.3.1

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[alsa-devel] [PATCH] ASoC: rcar: fixup dma_async_issue_pending() timing
by Kuninori Morimoto 19 Nov '13

19 Nov '13

DMAEngine will stall without this patch Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/sh/rcar/core.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 78c35b4..f57e519 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -200,9 +200,8 @@ static void rsnd_dma_do_work(struct work_struct *work) return; } + dma_async_issue_pending(dma->chan); } - - dma_async_issue_pending(dma->chan); } int rsnd_dma_available(struct rsnd_dma *dma) -- 1.7.9.5

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[alsa-devel] [PATCH] ASoC: wm5110: Add post SYSCLK register patch for rev D chip
by Charles Keepax 19 Nov '13

19 Nov '13

Certain registers require patching after the SYSCLK has been brought up add support for this into the CODEC driver. Signed-off-by: Charles Keepax <ckeepax(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm5110.c | 43 ++++++++++++++++++++++++++++++++++++++++++- 1 files changed, 42 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index f2d1094..c3c7396 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -37,6 +37,47 @@ struct wm5110_priv { struct arizona_fll fll[2]; }; +static const struct reg_default wm5110_sysclk_revd_patch[] = { + { 0x3093, 0x1001 }, + { 0x30E3, 0x1301 }, + { 0x3133, 0x1201 }, + { 0x3183, 0x1501 }, + { 0x31D3, 0x1401 }, +}; + +static int wm5110_sysclk_ev(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + struct arizona *arizona = dev_get_drvdata(codec->dev->parent); + struct regmap *regmap = codec->control_data; + const struct reg_default *patch = NULL; + int i, patch_size; + + switch (arizona->rev) { + case 3: + patch = wm5110_sysclk_revd_patch; + patch_size = ARRAY_SIZE(wm5110_sysclk_revd_patch); + break; + default: + return 0; + } + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + if (patch) + for (i = 0; i < patch_size; i++) + regmap_write(regmap, patch[i].reg, + patch[i].def); + break; + + default: + break; + } + + return 0; +} + static DECLARE_TLV_DB_SCALE(ana_tlv, 0, 100, 0); static DECLARE_TLV_DB_SCALE(eq_tlv, -1200, 100, 0); static DECLARE_TLV_DB_SCALE(digital_tlv, -6400, 50, 0); @@ -400,7 +441,7 @@ static const struct snd_kcontrol_new wm5110_aec_loopback_mux = static const struct snd_soc_dapm_widget wm5110_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("SYSCLK", ARIZONA_SYSTEM_CLOCK_1, ARIZONA_SYSCLK_ENA_SHIFT, - 0, NULL, 0), + 0, wm5110_sysclk_ev, SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_SUPPLY("ASYNCCLK", ARIZONA_ASYNC_CLOCK_1, ARIZONA_ASYNC_CLK_ENA_SHIFT, 0, NULL, 0), SND_SOC_DAPM_SUPPLY("OPCLK", ARIZONA_OUTPUT_SYSTEM_CLOCK, -- 1.7.2.5

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[alsa-devel] [PATCH 09/10] ASoC: ux500_pcm: Take out pointless dev_dbg() call
by Lee Jones 19 Nov '13

19 Nov '13

Cc: alsa-devel(a)alsa-project.org Cc: Mark Brown <broonie(a)kernel.org> Acked-by: Linus Walleij <linus.walleij(a)linaro.org> Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- sound/soc/ux500/ux500_pcm.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/ux500/ux500_pcm.c b/sound/soc/ux500/ux500_pcm.c index acfec98..fd7f12e 100644 --- a/sound/soc/ux500/ux500_pcm.c +++ b/sound/soc/ux500/ux500_pcm.c @@ -65,14 +65,10 @@ static struct dma_chan *ux500_pcm_request_chan(struct snd_soc_pcm_runtime *rtd, struct snd_pcm_substream *substream) { struct snd_soc_dai *dai = rtd->cpu_dai; - struct device *dev = dai->dev; u16 per_data_width, mem_data_width; struct stedma40_chan_cfg *dma_cfg; struct ux500_msp_dma_params *dma_params; - dev_dbg(dev, "%s: MSP %d (%s): Enter.\n", __func__, dai->id, - snd_pcm_stream_str(substream)); - dma_params = snd_soc_dai_get_dma_data(dai, substream); dma_cfg = dma_params->dma_cfg; -- 1.8.1.2

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[alsa-devel] [PATCH 06/10] ASoC: ux500_pcm: Extend Device Tree support to deal with DMA data
by Lee Jones 19 Nov '13

19 Nov '13

Soon we will strip out pdata support from the Ux500 set of ASoC drivers. When this happens it will have to supply a DMA slave_config to the dmaengine. At the moment a great deal of this comes from pdata via AUXDATA. We need to become independent of this soon. This patch starts the process by allocating memory for the associated data structures and fetches the MSP id used for const struct indexing. Cc: alsa-devel(a)alsa-project.org Cc: Mark Brown <broonie(a)kernel.org> Acked-by: Linus Walleij <linus.walleij(a)linaro.org> Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- sound/soc/ux500/ux500_msp_i2s.c | 56 ++++++++++++++++++++++++++++++----------- 1 file changed, 41 insertions(+), 15 deletions(-) diff --git a/sound/soc/ux500/ux500_msp_i2s.c b/sound/soc/ux500/ux500_msp_i2s.c index 1ca8b08..7f2a4ac 100644 --- a/sound/soc/ux500/ux500_msp_i2s.c +++ b/sound/soc/ux500/ux500_msp_i2s.c @@ -646,6 +646,34 @@ int ux500_msp_i2s_close(struct ux500_msp *msp, unsigned int dir) } +int ux500_msp_i2s_of_init_msp(struct platform_device *pdev, + struct ux500_msp *msp, + struct msp_i2s_platform_data **platform_data) +{ + struct msp_i2s_platform_data *pdata; + + *platform_data = devm_kzalloc(&pdev->dev, + sizeof(struct msp_i2s_platform_data), + GFP_KERNEL); + pdata = *platform_data; + if (!pdata) + return -ENOMEM; + + msp->playback_dma_data.dma_cfg = devm_kzalloc(&pdev->dev, + sizeof(struct stedma40_chan_cfg), + GFP_KERNEL); + if (!msp->playback_dma_data.dma_cfg) + return -ENOMEM; + + msp->capture_dma_data.dma_cfg = devm_kzalloc(&pdev->dev, + sizeof(struct stedma40_chan_cfg), + GFP_KERNEL); + if (!msp->capture_dma_data.dma_cfg) + return -ENOMEM; + + return 0; +} + int ux500_msp_i2s_init_msp(struct platform_device *pdev, struct ux500_msp **msp_p, struct msp_i2s_platform_data *platform_data) @@ -653,30 +681,28 @@ int ux500_msp_i2s_init_msp(struct platform_device *pdev, struct resource *res = NULL; struct device_node *np = pdev->dev.of_node; struct ux500_msp *msp; + int ret; *msp_p = devm_kzalloc(&pdev->dev, sizeof(struct ux500_msp), GFP_KERNEL); msp = *msp_p; if (!msp) return -ENOMEM; - if (np) { - if (!platform_data) { - platform_data = devm_kzalloc(&pdev->dev, - sizeof(struct msp_i2s_platform_data), GFP_KERNEL); - if (!platform_data) - return -ENOMEM; - } - } else - if (!platform_data) + if (!platform_data) { + if (np) { + ret = ux500_msp_i2s_of_init_msp(pdev, msp, + &platform_data); + if (ret) + return ret; + } else return -EINVAL; + } else { + msp->playback_dma_data.dma_cfg = platform_data->msp_i2s_dma_tx; + msp->capture_dma_data.dma_cfg = platform_data->msp_i2s_dma_rx; + msp->id = platform_data->id; + } - dev_dbg(&pdev->dev, "%s: Enter (name: %s, id: %d).\n", __func__, - pdev->name, platform_data->id); - - msp->id = platform_data->id; msp->dev = &pdev->dev; - msp->playback_dma_data.dma_cfg = platform_data->msp_i2s_dma_tx; - msp->capture_dma_data.dma_cfg = platform_data->msp_i2s_dma_rx; res = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (res == NULL) { -- 1.8.1.2

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[alsa-devel] [PATCH 05/10] ASoC: Ux500: Match platform by device node when booting Device Tree
by Lee Jones 19 Nov '13

19 Nov '13

We're getting closer to fully enabling the Ux500 ASoC driver for Device Tree. When we switch over from using AUXDATA we'll need to match platform by only Device Tree nodes. In this patch we NULL out the platform_name, and supply nodes for each platform device. Cc: alsa-devel(a)alsa-project.org Cc: Mark Brown <broonie(a)kernel.org> Acked-by: Linus Walleij <linus.walleij(a)linaro.org> Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- sound/soc/ux500/mop500.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/ux500/mop500.c b/sound/soc/ux500/mop500.c index 178d1ba..b3b66aa 100644 --- a/sound/soc/ux500/mop500.c +++ b/sound/soc/ux500/mop500.c @@ -91,6 +91,8 @@ static int mop500_of_probe(struct platform_device *pdev, for (i = 0; i < 2; i++) { mop500_dai_links[i].cpu_of_node = msp_np[i]; mop500_dai_links[i].cpu_dai_name = NULL; + mop500_dai_links[i].platform_of_node = msp_np[i]; + mop500_dai_links[i].platform_name = NULL; mop500_dai_links[i].codec_of_node = codec_np; mop500_dai_links[i].codec_name = NULL; } -- 1.8.1.2

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[alsa-devel] [PATCH 1/2] ASoC: ab8500: Revert "ASoC: ab8500: Explicitly set I/O up"
by Lee Jones 19 Nov '13

19 Nov '13

This reverts commit 2245e3c31c15c2d2a26926c4b734f4d3a37ae252, as it is one of two patches which stop audio from working on Snowball. Cc: alsa-devel(a)alsa-project.org Cc: Mark Brown <broonie(a)kernel.org> Signed-off-by: Lee Jones <lee.jones(a)linaro.org> --- sound/soc/codecs/ab8500-codec.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/codecs/ab8500-codec.c b/sound/soc/codecs/ab8500-codec.c index a0394a8..ca500fd 100644 --- a/sound/soc/codecs/ab8500-codec.c +++ b/sound/soc/codecs/ab8500-codec.c @@ -2473,8 +2473,6 @@ static int ab8500_codec_probe(struct snd_soc_codec *codec) dev_dbg(dev, "%s: Enter.\n", __func__); - snd_soc_codec_set_cache_io(codec, 0, 0, SND_SOC_REGMAP); - /* Setup AB8500 according to board-settings */ pdata = dev_get_platdata(dev->parent); -- 1.8.1.2

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