February 2018 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/2] ASoC: soc-generic-dmaengine-pcm: Fix error handling
by Fabio Estevam 26 Feb '18

26 Feb '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> When dmaengine_pcm_request_chan_of() fails it should release the previously acquired resources, which in this case is to call kfree(pcm), so jump to the correct point in the error path. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/soc-generic-dmaengine-pcm.c | 3 ++- 1 file changed, 2 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 768247f..32ea16d 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -450,7 +450,7 @@ int snd_dmaengine_pcm_register(struct device *dev, ret = dmaengine_pcm_request_chan_of(pcm, dev, config); if (ret) - goto err_free_dma; + goto err_free_pcm; ret = snd_soc_add_component(dev, &pcm->component, &dmaengine_pcm_component, NULL, 0); @@ -461,6 +461,7 @@ int snd_dmaengine_pcm_register(struct device *dev, err_free_dma: dmaengine_pcm_release_chan(pcm); +err_free_pcm: kfree(pcm); return ret; } -- 2.7.4

3 5

[alsa-devel] [RFC PATCH 0/3] ASoC: stm32: add S/PDIF support to SAI
by Olivier Moysan 26 Feb '18

26 Feb '18

This patch-set adds S/PDIF IEC6958 protocol support to STM32 SAI. Only S/PDIF playback is supported here, as SAI peripheral does not support S/PDIF capture. SAI peripheral does not take as input full IEC60958 frames as it generates preamble and parity bit. However, it cannot handle raw PCM data, as it is not designed to insert channel status and user data in the generated IEC frames. Moreover, SAI data register format does not match exactly IEC60958 subframe format. (see figure below). Channel status (C) and User data (U) are at 26-25 bits position, in data register, instead of 30-29 bits position. IEC60958 subframe bits: 31-28 27-4 3-0 P C U V data (16|24 bits) preamble SAI data register format: 27-24 23-0 P C U V data (16|24 bits) The impact in software is that the data have to be processed before DMA transfer. These constraints should be managed at driver level to minimize the impact on application. This means that a callback is necessary to perform data processing (shift and channel status insertion) This patch-set introduces a callback to allow processing on samples. The implementation is based on previous discussions available here: https://patchwork.kernel.org/patch/9570255/ BRs Olivier Olivier Moysan (3): ASoC: stm32: Add S/PDIF to SAI bindings ASoC: dmaengine_pcm: add processing support ASoC: stm32: sai: Add support of S/PDIF playback .../devicetree/bindings/sound/st,stm32-sai.txt | 6 + include/sound/dmaengine_pcm.h | 3 + sound/soc/soc-generic-dmaengine-pcm.c | 62 ++++++++- sound/soc/stm/stm32_sai.c | 2 + sound/soc/stm/stm32_sai.h | 2 + sound/soc/stm/stm32_sai_sub.c | 153 +++++++++++++++++---- 6 files changed, 202 insertions(+), 26 deletions(-) -- 1.9.1

2 5

[alsa-devel] [PATCH] ASoC: uniphier: fix broken sound if use SRC in replay
by Katsuhiro Suzuki 26 Feb '18

26 Feb '18

This patch fixes settings for ports with SRC. These ports need to set the fixed audio rate and clock. If not, the SRC outputs broken sound. Signed-off-by: Katsuhiro Suzuki <suzuki.katsuhiro(a)socionext.com> --- sound/soc/uniphier/aio-core.c | 17 +++++++++++++++-- 1 file changed, 15 insertions(+), 2 deletions(-) diff --git a/sound/soc/uniphier/aio-core.c b/sound/soc/uniphier/aio-core.c index 1e5f053d9f6b..1711361fc0c2 100644 --- a/sound/soc/uniphier/aio-core.c +++ b/sound/soc/uniphier/aio-core.c @@ -501,7 +501,7 @@ int aio_port_set_clk(struct uniphier_aio_sub *sub) OPORTMXCTR2_MSSEL_MASTER | OPORTMXCTR2_EXTLSIFSSEL_36 | OPORTMXCTR2_DACCKSEL_1_2; - } else { + } else if (sub->swm->type == PORT_TYPE_SPDIF) { if (sub->aio->pll_out >= ARRAY_SIZE(v_pll)) { dev_err(dev, "PLL(%d) is invalid\n", sub->aio->pll_out); @@ -521,6 +521,11 @@ int aio_port_set_clk(struct uniphier_aio_sub *sub) v |= OPORTMXCTR2_EXTLSIFSSEL_24; break; } + } else { + v = OPORTMXCTR2_ACLKSEL_A1 | + OPORTMXCTR2_MSSEL_MASTER | + OPORTMXCTR2_EXTLSIFSSEL_36 | + OPORTMXCTR2_DACCKSEL_1_2; } regmap_write(r, OPORTMXCTR2(sub->swm->oport.map), v); } else { @@ -550,11 +555,19 @@ int aio_port_set_param(struct uniphier_aio_sub *sub, int pass_through, const struct snd_pcm_hw_params *params) { struct regmap *r = sub->aio->chip->regmap; + unsigned int rate; u32 v; int ret; if (!pass_through) { - ret = aio_port_set_rate(sub, params_rate(params)); + if (sub->swm->type == PORT_TYPE_EVE || + sub->swm->type == PORT_TYPE_CONV) { + rate = 48000; + } else { + rate = params_rate(params); + } + + ret = aio_port_set_rate(sub, rate); if (ret) return ret; -- 2.16.1

2 1

[alsa-devel] [PATCH] ASoC: sirf: remove duplicated bit-wise or of USP_RXFIFO_THD_INT
by Colin King 26 Feb '18

26 Feb '18

From: Colin Ian King <colin.king(a)canonical.com> Bit pattern USP_RXFIFO_THD_INT is being bit-wise or'd twice; remove the redundant 2nd USP_RXFIFO_THD_INT Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/sirf/sirf-usp.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sirf/sirf-usp.h b/sound/soc/sirf/sirf-usp.h index bf0201cb15bc..e22e13a619e8 100644 --- a/sound/soc/sirf/sirf-usp.h +++ b/sound/soc/sirf/sirf-usp.h @@ -154,7 +154,7 @@ USP_RX_IO_DMA_INT|\ USP_RXFIFO_FULL_INT|\ USP_RXFIFO_THD_INT|\ - USP_RXFIFO_THD_INT|USP_RX_TIMEOUT_INT) + USP_RX_TIMEOUT_INT) #define USP_INT_ALL 0x1FFF -- 2.15.1

2 1

[alsa-devel] [PATCH] ASoC: pxa: remove duplicated bit-wise or of SNDRV_PCM_FMTBIT_S24_LE
by Colin King 26 Feb '18

26 Feb '18

From: Colin Ian King <colin.king(a)canonical.com> Bit pattern SNDRV_PCM_FMTBIT_S24_LE is being bit-wise or'd twice; remove the redundant 2nd SNDRV_PCM_FMTBIT_S24_LE Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/pxa/mmp-sspa.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/pxa/mmp-sspa.c b/sound/soc/pxa/mmp-sspa.c index 64b85e30c1f8..7c998ea4ebee 100644 --- a/sound/soc/pxa/mmp-sspa.c +++ b/sound/soc/pxa/mmp-sspa.c @@ -377,7 +377,6 @@ static int mmp_sspa_probe(struct snd_soc_dai *dai) #define MMP_SSPA_FORMATS (SNDRV_PCM_FMTBIT_S8 | \ SNDRV_PCM_FMTBIT_S16_LE | \ SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S24_LE | \ SNDRV_PCM_FMTBIT_S32_LE) static const struct snd_soc_dai_ops mmp_sspa_dai_ops = { -- 2.15.1

2 1

[alsa-devel] Applied "ASoC: sun4i-i2s: Fix RX slot number of SUN8I" to the asoc tree
by Mark Brown 26 Feb '18

26 Feb '18

The patch ASoC: sun4i-i2s: Fix RX slot number of SUN8I has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 5a3386790a172cf738194e1574f631cd43c6140a Mon Sep 17 00:00:00 2001 From: Yong Deng <yong.deng(a)magewell.com> Date: Mon, 26 Feb 2018 10:43:52 +0800 Subject: [PATCH] ASoC: sun4i-i2s: Fix RX slot number of SUN8I I2S's RX slot number of SUN8I should be shifted 4 bit to left. Fixes: 7d2993811a1e ("ASoC: sun4i-i2s: Add support for H3") Signed-off-by: Yong Deng <yong.deng(a)magewell.com> Reviewed-by: Chen-Yu Tsai <wens(a)csie.org> Signed-off-by: Mark Brown <broonie(a)kernel.org> Cc: stable(a)vger.kernel.org --- sound/soc/sunxi/sun4i-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index dca1143c1150..a4aa931ebfae 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -104,7 +104,7 @@ #define SUN8I_I2S_CHAN_CFG_REG 0x30 #define SUN8I_I2S_CHAN_CFG_RX_SLOT_NUM_MASK GENMASK(6, 4) -#define SUN8I_I2S_CHAN_CFG_RX_SLOT_NUM(chan) (chan - 1) +#define SUN8I_I2S_CHAN_CFG_RX_SLOT_NUM(chan) ((chan - 1) << 4) #define SUN8I_I2S_CHAN_CFG_TX_SLOT_NUM_MASK GENMASK(2, 0) #define SUN8I_I2S_CHAN_CFG_TX_SLOT_NUM(chan) (chan - 1) -- 2.16.1

1 0

[alsa-devel] [PATCH] ASoC: wm_adsp: Support streams which can start/stop with DSP active
by Charles Keepax 26 Feb '18

26 Feb '18

Clear the buffer data structure on each trigger start such that the buffer is in a sensible state even if the DSP itself didn't restart. This is necessary to support voice control streams which can trigger multiple times without reloading the firmware. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 28 ++++++++++++++++++---------- 1 file changed, 18 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index b6e7c614301f..9b27390cf4f1 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -3258,6 +3258,13 @@ static int wm_adsp_buffer_populate(struct wm_adsp_compr_buf *buf) return 0; } +static void wm_adsp_buffer_clear(struct wm_adsp_compr_buf *buf) +{ + buf->irq_count = 0xFFFFFFFF; + buf->read_index = -1; + buf->avail = 0; +} + static int wm_adsp_buffer_init(struct wm_adsp *dsp) { struct wm_adsp_compr_buf *buf; @@ -3268,8 +3275,8 @@ static int wm_adsp_buffer_init(struct wm_adsp *dsp) return -ENOMEM; buf->dsp = dsp; - buf->read_index = -1; - buf->irq_count = 0xFFFFFFFF; + + wm_adsp_buffer_clear(buf); ret = wm_adsp_buffer_locate(buf); if (ret < 0) { @@ -3327,16 +3334,17 @@ int wm_adsp_compr_trigger(struct snd_compr_stream *stream, int cmd) switch (cmd) { case SNDRV_PCM_TRIGGER_START: - if (wm_adsp_compr_attached(compr)) - break; - - ret = wm_adsp_compr_attach(compr); - if (ret < 0) { - adsp_err(dsp, "Failed to link buffer and stream: %d\n", - ret); - break; + if (!wm_adsp_compr_attached(compr)) { + ret = wm_adsp_compr_attach(compr); + if (ret < 0) { + adsp_err(dsp, "Failed to link buffer and stream: %d\n", + ret); + break; + } } + wm_adsp_buffer_clear(compr->buf); + /* Trigger the IRQ at one fragment of data */ ret = wm_adsp_buffer_write(compr->buf, HOST_BUFFER_FIELD(high_water_mark), -- 2.11.0

2 1

[alsa-devel] Applied "ASoC: dmaengine_pcm: add processing support" to the asoc tree
by Mark Brown 26 Feb '18

26 Feb '18

The patch ASoC: dmaengine_pcm: add processing support has been applied to the asoc tree at https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git All being well this means that it will be integrated into the linux-next tree (usually sometime in the next 24 hours) and sent to Linus during the next merge window (or sooner if it is a bug fix), however if problems are discovered then the patch may be dropped or reverted. You may get further e-mails resulting from automated or manual testing and review of the tree, please engage with people reporting problems and send followup patches addressing any issues that are reported if needed. If any updates are required or you are submitting further changes they should be sent as incremental updates against current git, existing patches will not be replaced. Please add any relevant lists and maintainers to the CCs when replying to this mail. Thanks, Mark >From 78648092ef46255e6dc6685202164199c86cf930 Mon Sep 17 00:00:00 2001 From: Olivier Moysan <olivier.moysan(a)st.com> Date: Mon, 19 Feb 2018 16:00:36 +0100 Subject: [PATCH] ASoC: dmaengine_pcm: add processing support Allow dmaengine client to optionally register a processing callback. This callback is intended to apply processing on samples in buffer copied from/to user space, before/after DMA transfer. Signed-off-by: Olivier Moysan <olivier.moysan(a)st.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/dmaengine_pcm.h | 3 ++ sound/soc/soc-generic-dmaengine-pcm.c | 62 +++++++++++++++++++++++++++++++++-- 2 files changed, 63 insertions(+), 2 deletions(-) diff --git a/include/sound/dmaengine_pcm.h b/include/sound/dmaengine_pcm.h index 8a5a8404966e..47ef486852ed 100644 --- a/include/sound/dmaengine_pcm.h +++ b/include/sound/dmaengine_pcm.h @@ -140,6 +140,9 @@ struct snd_dmaengine_pcm_config { struct dma_chan *(*compat_request_channel)( struct snd_soc_pcm_runtime *rtd, struct snd_pcm_substream *substream); + int (*process)(struct snd_pcm_substream *substream, + int channel, unsigned long hwoff, + void *buf, unsigned long bytes); dma_filter_fn compat_filter_fn; struct device *dma_dev; const char *chan_names[SNDRV_PCM_STREAM_LAST + 1]; diff --git a/sound/soc/soc-generic-dmaengine-pcm.c b/sound/soc/soc-generic-dmaengine-pcm.c index 785f25ede3e5..567fbdfd1ca9 100644 --- a/sound/soc/soc-generic-dmaengine-pcm.c +++ b/sound/soc/soc-generic-dmaengine-pcm.c @@ -341,6 +341,41 @@ static snd_pcm_uframes_t dmaengine_pcm_pointer( return snd_dmaengine_pcm_pointer(substream); } +static int dmaengine_copy_user(struct snd_pcm_substream *substream, + int channel, unsigned long hwoff, + void *buf, unsigned long bytes) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_component *component = + snd_soc_rtdcom_lookup(rtd, SND_DMAENGINE_PCM_DRV_NAME); + struct snd_pcm_runtime *runtime = substream->runtime; + struct dmaengine_pcm *pcm = soc_component_to_pcm(component); + int (*process)(struct snd_pcm_substream *substream, + int channel, unsigned long hwoff, + void *buf, unsigned long bytes) = pcm->config->process; + bool is_playback = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; + void *dma_ptr = runtime->dma_area + hwoff + + channel * (runtime->dma_bytes / runtime->channels); + int ret; + + if (is_playback) + if (copy_from_user(dma_ptr, (void __user *)buf, bytes)) + return -EFAULT; + + if (process) { + ret = process(substream, channel, hwoff, + (void __user *)buf, bytes); + if (ret < 0) + return ret; + } + + if (!is_playback) + if (copy_to_user((void __user *)buf, dma_ptr, bytes)) + return -EFAULT; + + return 0; +} + static const struct snd_pcm_ops dmaengine_pcm_ops = { .open = dmaengine_pcm_open, .close = snd_dmaengine_pcm_close, @@ -351,6 +386,17 @@ static const struct snd_pcm_ops dmaengine_pcm_ops = { .pointer = dmaengine_pcm_pointer, }; +static const struct snd_pcm_ops dmaengine_pcm_process_ops = { + .open = dmaengine_pcm_open, + .close = snd_dmaengine_pcm_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = dmaengine_pcm_hw_params, + .hw_free = snd_pcm_lib_free_pages, + .trigger = snd_dmaengine_pcm_trigger, + .pointer = dmaengine_pcm_pointer, + .copy_user = dmaengine_copy_user, +}; + static const struct snd_soc_component_driver dmaengine_pcm_component = { .name = SND_DMAENGINE_PCM_DRV_NAME, .probe_order = SND_SOC_COMP_ORDER_LATE, @@ -358,6 +404,13 @@ static const struct snd_soc_component_driver dmaengine_pcm_component = { .pcm_new = dmaengine_pcm_new, }; +static const struct snd_soc_component_driver dmaengine_pcm_component_process = { + .name = SND_DMAENGINE_PCM_DRV_NAME, + .probe_order = SND_SOC_COMP_ORDER_LATE, + .ops = &dmaengine_pcm_process_ops, + .pcm_new = dmaengine_pcm_new, +}; + static const char * const dmaengine_pcm_dma_channel_names[] = { [SNDRV_PCM_STREAM_PLAYBACK] = "tx", [SNDRV_PCM_STREAM_CAPTURE] = "rx", @@ -453,8 +506,13 @@ int snd_dmaengine_pcm_register(struct device *dev, if (ret) goto err_free_pcm; - ret = snd_soc_add_component(dev, &pcm->component, - &dmaengine_pcm_component, NULL, 0); + if (config && config->process) + ret = snd_soc_add_component(dev, &pcm->component, + &dmaengine_pcm_component_process, + NULL, 0); + else + ret = snd_soc_add_component(dev, &pcm->component, + &dmaengine_pcm_component, NULL, 0); if (ret) goto err_free_dma; -- 2.16.1

1 0

[alsa-devel] [PATCH] ASoC: sun4i-i2s: Fix offset tune for SUN8I
by Yong Deng 26 Feb '18

26 Feb '18

SUN8I_I2S_TX_CHAN_OFFSET_MASK should be GENMASK(13, 12). And offset tune of RX should also be updated. Fixes: 7d2993811a1e ("ASoC: sun4i-i2s: Add support for H3") Signed-off-by: Yong Deng <yong.deng(a)magewell.com> --- sound/soc/sunxi/sun4i-i2s.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index a4aa931ebfae..797cafffaac8 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -110,12 +110,15 @@ #define SUN8I_I2S_TX_CHAN_MAP_REG 0x44 #define SUN8I_I2S_TX_CHAN_SEL_REG 0x34 -#define SUN8I_I2S_TX_CHAN_OFFSET_MASK GENMASK(13, 11) +#define SUN8I_I2S_TX_CHAN_OFFSET_MASK GENMASK(13, 12) #define SUN8I_I2S_TX_CHAN_OFFSET(offset) (offset << 12) #define SUN8I_I2S_TX_CHAN_EN_MASK GENMASK(11, 4) #define SUN8I_I2S_TX_CHAN_EN(num_chan) (((1 << num_chan) - 1) << 4) #define SUN8I_I2S_RX_CHAN_SEL_REG 0x54 +#define SUN8I_I2S_RX_CHAN_OFFSET_MASK GENMASK(13, 12) +#define SUN8I_I2S_RX_CHAN_OFFSET(offset) (offset << 12) + #define SUN8I_I2S_RX_CHAN_MAP_REG 0x58 /** @@ -460,6 +463,9 @@ static int sun4i_i2s_set_fmt(struct snd_soc_dai *dai, unsigned int fmt) regmap_update_bits(i2s->regmap, SUN8I_I2S_TX_CHAN_SEL_REG, SUN8I_I2S_TX_CHAN_OFFSET_MASK, SUN8I_I2S_TX_CHAN_OFFSET(offset)); + regmap_update_bits(i2s->regmap, SUN8I_I2S_RX_CHAN_SEL_REG, + SUN8I_I2S_RX_CHAN_OFFSET_MASK, + SUN8I_I2S_RX_CHAN_OFFSET(offset)); } regmap_field_write(i2s->field_fmt_mode, val); -- 1.8.3.1

1 0

[alsa-devel] [PATCH] ASoC: sun4i-i2s: Fix RX slot number of SUN8I
by Yong Deng 26 Feb '18

26 Feb '18

I2S's RX slot number of SUN8I should be shifted 4 bit to left. Fixes: 7d2993811a1e ("ASoC: sun4i-i2s: Add support for H3") Signed-off-by: Yong Deng <yong.deng(a)magewell.com> --- sound/soc/sunxi/sun4i-i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/sunxi/sun4i-i2s.c b/sound/soc/sunxi/sun4i-i2s.c index dca1143c1150..a4aa931ebfae 100644 --- a/sound/soc/sunxi/sun4i-i2s.c +++ b/sound/soc/sunxi/sun4i-i2s.c @@ -104,7 +104,7 @@ #define SUN8I_I2S_CHAN_CFG_REG 0x30 #define SUN8I_I2S_CHAN_CFG_RX_SLOT_NUM_MASK GENMASK(6, 4) -#define SUN8I_I2S_CHAN_CFG_RX_SLOT_NUM(chan) (chan - 1) +#define SUN8I_I2S_CHAN_CFG_RX_SLOT_NUM(chan) ((chan - 1) << 4) #define SUN8I_I2S_CHAN_CFG_TX_SLOT_NUM_MASK GENMASK(2, 0) #define SUN8I_I2S_CHAN_CFG_TX_SLOT_NUM(chan) (chan - 1) -- 1.8.3.1

2 1