February 2019 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 1/5] ASoC: wm_adsp: Factor out stripping padding from ADSP data
by Charles Keepax 26 Feb '19

26 Feb '19

In preparation for more refactoring add a helper function to strip the padding from ADSP data. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> --- sound/soc/codecs/wm_adsp.c | 27 +++++++++++++++++---------- 1 file changed, 17 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index 8d16a14cd2266..ae7e1f5d8d10a 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -3231,6 +3231,21 @@ static inline int wm_adsp_buffer_write(struct wm_adsp_compr_buf *buf, buf->host_buf_ptr + field_offset, data); } +static void wm_adsp_remove_padding(u32 *buf, int nwords, int data_word_size) +{ + u8 *pack_in = (u8 *)buf; + u8 *pack_out = (u8 *)buf; + int i, j; + + /* Remove the padding bytes from the data read from the DSP */ + for (i = 0; i < nwords; i++) { + for (j = 0; j < data_word_size; j++) + *pack_out++ = *pack_in++; + + pack_in += sizeof(*buf) - data_word_size; + } +} + static int wm_adsp_legacy_host_buf_addr(struct wm_adsp_compr_buf *buf) { struct wm_adsp_alg_region *alg_region; @@ -3670,11 +3685,9 @@ EXPORT_SYMBOL_GPL(wm_adsp_compr_pointer); static int wm_adsp_buffer_capture_block(struct wm_adsp_compr *compr, int target) { struct wm_adsp_compr_buf *buf = compr->buf; - u8 *pack_in = (u8 *)compr->raw_buf; - u8 *pack_out = (u8 *)compr->raw_buf; unsigned int adsp_addr; int mem_type, nwords, max_read; - int i, j, ret; + int i, ret; /* Calculate read parameters */ for (i = 0; i < wm_adsp_fw[buf->dsp->fw].caps->num_regions; ++i) @@ -3706,13 +3719,7 @@ static int wm_adsp_buffer_capture_block(struct wm_adsp_compr *compr, int target) if (ret < 0) return ret; - /* Remove the padding bytes from the data read from the DSP */ - for (i = 0; i < nwords; i++) { - for (j = 0; j < WM_ADSP_DATA_WORD_SIZE; j++) - *pack_out++ = *pack_in++; - - pack_in += sizeof(*(compr->raw_buf)) - WM_ADSP_DATA_WORD_SIZE; - } + wm_adsp_remove_padding(compr->raw_buf, nwords, WM_ADSP_DATA_WORD_SIZE); /* update read index to account for words read */ buf->read_index += nwords; -- 2.11.0

2 8

[alsa-devel] Applied "ASoC: wm_adsp: Improve logging messages" to the asoc tree
by Mark Brown 26 Feb '19

26 Feb '19

The patch ASoC: wm_adsp: Improve logging messages 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 0d3fba3e7a566917f4286dd42b83c780c47dcbf7 Mon Sep 17 00:00:00 2001 From: Charles Keepax <ckeepax(a)opensource.cirrus.com> Date: Fri, 22 Feb 2019 10:04:21 +0000 Subject: [PATCH] ASoC: wm_adsp: Improve logging messages As the compressed stream implementation has acquired support for multiple DAI links and compressed streams it has become harder to interpret messages in the kernel log. Add additional macros to include the compressed DAI name in the log messages, allowing different streams to be easily disambiguated. Signed-off-by: Charles Keepax <ckeepax(a)opensource.cirrus.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/codecs/wm_adsp.c | 97 +++++++++++++++++++++----------------- 1 file changed, 53 insertions(+), 44 deletions(-) diff --git a/sound/soc/codecs/wm_adsp.c b/sound/soc/codecs/wm_adsp.c index fe802fc331c5..8077c18cbcdf 100644 --- a/sound/soc/codecs/wm_adsp.c +++ b/sound/soc/codecs/wm_adsp.c @@ -46,6 +46,13 @@ #define adsp_dbg(_dsp, fmt, ...) \ dev_dbg(_dsp->dev, "%s: " fmt, _dsp->name, ##__VA_ARGS__) +#define compr_err(_obj, fmt, ...) \ + adsp_err(_obj->dsp, "%s: " fmt, _obj->name ? _obj->name : "legacy", \ + ##__VA_ARGS__) +#define compr_dbg(_obj, fmt, ...) \ + adsp_dbg(_obj->dsp, "%s: " fmt, _obj->name ? _obj->name : "legacy", \ + ##__VA_ARGS__) + #define ADSP1_CONTROL_1 0x00 #define ADSP1_CONTROL_2 0x02 #define ADSP1_CONTROL_3 0x03 @@ -3033,20 +3040,23 @@ int wm_adsp_compr_open(struct wm_adsp *dsp, struct snd_compr_stream *stream) mutex_lock(&dsp->pwr_lock); if (wm_adsp_fw[dsp->fw].num_caps == 0) { - adsp_err(dsp, "Firmware does not support compressed API\n"); + adsp_err(dsp, "%s: Firmware does not support compressed API\n", + rtd->codec_dai->name); ret = -ENXIO; goto out; } if (wm_adsp_fw[dsp->fw].compr_direction != stream->direction) { - adsp_err(dsp, "Firmware does not support stream direction\n"); + adsp_err(dsp, "%s: Firmware does not support stream direction\n", + rtd->codec_dai->name); ret = -EINVAL; goto out; } list_for_each_entry(tmp, &dsp->compr_list, list) { if (!strcmp(tmp->name, rtd->codec_dai->name)) { - adsp_err(dsp, "Only a single stream supported per dai\n"); + adsp_err(dsp, "%s: Only a single stream supported per dai\n", + rtd->codec_dai->name); ret = -EBUSY; goto out; } @@ -3106,9 +3116,9 @@ static int wm_adsp_compr_check_params(struct snd_compr_stream *stream, params->buffer.fragments < WM_ADSP_MIN_FRAGMENTS || params->buffer.fragments > WM_ADSP_MAX_FRAGMENTS || params->buffer.fragment_size % WM_ADSP_DATA_WORD_SIZE) { - adsp_err(dsp, "Invalid buffer fragsize=%d fragments=%d\n", - params->buffer.fragment_size, - params->buffer.fragments); + compr_err(compr, "Invalid buffer fragsize=%d fragments=%d\n", + params->buffer.fragment_size, + params->buffer.fragments); return -EINVAL; } @@ -3136,9 +3146,9 @@ static int wm_adsp_compr_check_params(struct snd_compr_stream *stream, return 0; } - adsp_err(dsp, "Invalid params id=%u ch=%u,%u rate=%u fmt=%u\n", - params->codec.id, params->codec.ch_in, params->codec.ch_out, - params->codec.sample_rate, params->codec.format); + compr_err(compr, "Invalid params id=%u ch=%u,%u rate=%u fmt=%u\n", + params->codec.id, params->codec.ch_in, params->codec.ch_out, + params->codec.sample_rate, params->codec.format); return -EINVAL; } @@ -3160,8 +3170,8 @@ int wm_adsp_compr_set_params(struct snd_compr_stream *stream, compr->size = params->buffer; - adsp_dbg(compr->dsp, "fragment_size=%d fragments=%d\n", - compr->size.fragment_size, compr->size.fragments); + compr_dbg(compr, "fragment_size=%d fragments=%d\n", + compr->size.fragment_size, compr->size.fragments); size = wm_adsp_compr_frag_words(compr) * sizeof(*compr->raw_buf); compr->raw_buf = kmalloc(size, GFP_DMA | GFP_KERNEL); @@ -3303,10 +3313,10 @@ static int wm_adsp_buffer_populate(struct wm_adsp_compr_buf *buf) region->cumulative_size = offset; - adsp_dbg(buf->dsp, - "region=%d type=%d base=%08x off=%08x size=%08x\n", - i, region->mem_type, region->base_addr, - region->offset, region->cumulative_size); + compr_dbg(buf, + "region=%d type=%d base=%08x off=%08x size=%08x\n", + i, region->mem_type, region->base_addr, + region->offset, region->cumulative_size); } return 0; @@ -3380,7 +3390,7 @@ static int wm_adsp_buffer_parse_legacy(struct wm_adsp *dsp) if (ret < 0) return ret; - adsp_dbg(dsp, "legacy host_buf_ptr=%x\n", buf->host_buf_ptr); + compr_dbg(buf, "legacy host_buf_ptr=%x\n", buf->host_buf_ptr); return 0; } @@ -3428,7 +3438,7 @@ static int wm_adsp_buffer_parse_coeff(struct wm_coeff_ctl *ctl) * control is one word, assume version 0. */ if (ctl->len == 4) { - adsp_dbg(ctl->dsp, "host_buf_ptr=%x\n", buf->host_buf_ptr); + compr_dbg(buf, "host_buf_ptr=%x\n", buf->host_buf_ptr); return 0; } @@ -3458,8 +3468,8 @@ static int wm_adsp_buffer_parse_coeff(struct wm_coeff_ctl *ctl) buf->name = kasprintf(GFP_KERNEL, "%s-dsp-%s", ctl->dsp->part, (char *)&coeff_v1.name); - adsp_dbg(ctl->dsp, "host_buf_ptr=%x coeff version %u\n", - buf->host_buf_ptr, val); + compr_dbg(buf, "host_buf_ptr=%x coeff version %u\n", + buf->host_buf_ptr, val); return val; } @@ -3525,7 +3535,7 @@ int wm_adsp_compr_trigger(struct snd_compr_stream *stream, int cmd) struct wm_adsp *dsp = compr->dsp; int ret = 0; - adsp_dbg(dsp, "Trigger: %d\n", cmd); + compr_dbg(compr, "Trigger: %d\n", cmd); mutex_lock(&dsp->pwr_lock); @@ -3534,8 +3544,8 @@ int wm_adsp_compr_trigger(struct snd_compr_stream *stream, int cmd) 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); + compr_err(compr, "Failed to link buffer and stream: %d\n", + ret); break; } } @@ -3547,8 +3557,8 @@ int wm_adsp_compr_trigger(struct snd_compr_stream *stream, int cmd) HOST_BUFFER_FIELD(high_water_mark), wm_adsp_compr_frag_words(compr)); if (ret < 0) { - adsp_err(dsp, "Failed to set high water mark: %d\n", - ret); + compr_err(compr, "Failed to set high water mark: %d\n", + ret); break; } break; @@ -3589,7 +3599,7 @@ static int wm_adsp_buffer_update_avail(struct wm_adsp_compr_buf *buf) read_index = sign_extend32(next_read_index, 23); if (read_index < 0) { - adsp_dbg(buf->dsp, "Avail check on unstarted stream\n"); + compr_dbg(buf, "Avail check on unstarted stream\n"); return 0; } @@ -3607,8 +3617,8 @@ static int wm_adsp_buffer_update_avail(struct wm_adsp_compr_buf *buf) if (avail < 0) avail += wm_adsp_buffer_size(buf); - adsp_dbg(buf->dsp, "readindex=0x%x, writeindex=0x%x, avail=%d\n", - buf->read_index, write_index, avail * WM_ADSP_DATA_WORD_SIZE); + compr_dbg(buf, "readindex=0x%x, writeindex=0x%x, avail=%d\n", + buf->read_index, write_index, avail * WM_ADSP_DATA_WORD_SIZE); buf->avail = avail; @@ -3621,11 +3631,11 @@ static int wm_adsp_buffer_get_error(struct wm_adsp_compr_buf *buf) ret = wm_adsp_buffer_read(buf, HOST_BUFFER_FIELD(error), &buf->error); if (ret < 0) { - adsp_err(buf->dsp, "Failed to check buffer error: %d\n", ret); + compr_err(buf, "Failed to check buffer error: %d\n", ret); return ret; } if (buf->error != 0) { - adsp_err(buf->dsp, "Buffer error occurred: %d\n", buf->error); + compr_err(buf, "Buffer error occurred: %d\n", buf->error); return -EIO; } @@ -3644,6 +3654,7 @@ int wm_adsp_compr_handle_irq(struct wm_adsp *dsp) ret = -ENODEV; goto out; } + adsp_dbg(dsp, "Handling buffer IRQ\n"); list_for_each_entry(buf, &dsp->buffer_list, list) { @@ -3656,13 +3667,13 @@ int wm_adsp_compr_handle_irq(struct wm_adsp *dsp) ret = wm_adsp_buffer_read(buf, HOST_BUFFER_FIELD(irq_count), &buf->irq_count); if (ret < 0) { - adsp_err(dsp, "Failed to get irq_count: %d\n", ret); + compr_err(buf, "Failed to get irq_count: %d\n", ret); goto out; } ret = wm_adsp_buffer_update_avail(buf); if (ret < 0) { - adsp_err(dsp, "Error reading avail: %d\n", ret); + compr_err(buf, "Error reading avail: %d\n", ret); goto out; } @@ -3686,8 +3697,7 @@ static int wm_adsp_buffer_reenable_irq(struct wm_adsp_compr_buf *buf) if (buf->irq_count & 0x01) return 0; - adsp_dbg(buf->dsp, "Enable IRQ(0x%x) for next fragment\n", - buf->irq_count); + compr_dbg(buf, "Enable IRQ(0x%x) for next fragment\n", buf->irq_count); buf->irq_count |= 0x01; @@ -3703,7 +3713,7 @@ int wm_adsp_compr_pointer(struct snd_compr_stream *stream, struct wm_adsp_compr_buf *buf; int ret = 0; - adsp_dbg(dsp, "Pointer request\n"); + compr_dbg(compr, "Pointer request\n"); mutex_lock(&dsp->pwr_lock); @@ -3718,7 +3728,7 @@ int wm_adsp_compr_pointer(struct snd_compr_stream *stream, if (buf->avail < wm_adsp_compr_frag_words(compr)) { ret = wm_adsp_buffer_update_avail(buf); if (ret < 0) { - adsp_err(dsp, "Error reading avail: %d\n", ret); + compr_err(compr, "Error reading avail: %d\n", ret); goto out; } @@ -3737,9 +3747,8 @@ int wm_adsp_compr_pointer(struct snd_compr_stream *stream, ret = wm_adsp_buffer_reenable_irq(buf); if (ret < 0) { - adsp_err(dsp, - "Failed to re-enable buffer IRQ: %d\n", - ret); + compr_err(compr, "Failed to re-enable buffer IRQ: %d\n", + ret); goto out; } } @@ -3814,11 +3823,10 @@ static int wm_adsp_buffer_capture_block(struct wm_adsp_compr *compr, int target) static int wm_adsp_compr_read(struct wm_adsp_compr *compr, char __user *buf, size_t count) { - struct wm_adsp *dsp = compr->dsp; int ntotal = 0; int nwords, nbytes; - adsp_dbg(dsp, "Requested read of %zu bytes\n", count); + compr_dbg(compr, "Requested read of %zu bytes\n", count); if (!compr->buf || compr->buf->error) { snd_compr_stop_error(compr->stream, SNDRV_PCM_STATE_XRUN); @@ -3830,17 +3838,18 @@ static int wm_adsp_compr_read(struct wm_adsp_compr *compr, do { nwords = wm_adsp_buffer_capture_block(compr, count); if (nwords < 0) { - adsp_err(dsp, "Failed to capture block: %d\n", nwords); + compr_err(compr, "Failed to capture block: %d\n", + nwords); return nwords; } nbytes = nwords * WM_ADSP_DATA_WORD_SIZE; - adsp_dbg(dsp, "Read %d bytes\n", nbytes); + compr_dbg(compr, "Read %d bytes\n", nbytes); if (copy_to_user(buf + ntotal, compr->raw_buf, nbytes)) { - adsp_err(dsp, "Failed to copy data to user: %d, %d\n", - ntotal, nbytes); + compr_err(compr, "Failed to copy data to user: %d, %d\n", + ntotal, nbytes); return -EFAULT; } -- 2.20.1

1 0

[alsa-devel] [PATCH v2] ASoC: rsnd: gen: fix SSI9 4/5/6/7 busif related register address
by twischer＠de.adit-jv.com 26 Feb '19

26 Feb '19

From: Jiada Wang <jiada_wang(a)mentor.com> Currently each SSI unit 's busif mode/adinr/dalign address is registered by: (in busif4 case) RSND_GEN_M_REG(SSI_BUSIF4_MODE, 0x500, 0x80) RSND_GEN_M_REG(SSI_BUSIF4_ADINR,0x504, 0x80) RSND_GEN_M_REG(SSI_BUSIF4_DALIGN, 0x508, 0x80) But according to user manual 41.1.4 Register Configuration ssi9 4/5/6/7 busif mode/adinr/dalign register address ( SSI9-[4/5/6/7]_BUSIF_[MODE/ADINR/DALIGN] ) are out of this rule. This patch registers ssi9 4/5/6/7 mode/adinr/dalign register as single register, and access these registers in case of SSI9 BUSIF 4/5/6/7. Fixes: commit 8c9d75033340 ("ASoC: rsnd: ssiu: Support BUSIF other than BUSIF0") Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- Hi all, On 2/25/19 02:29, Kuninori Morimoto wrote: > I think > - and access these these registers > + and access these registers Done. On 2/25/19 02:29, Kuninori Morimoto wrote: > Necessary register on rsnd_mod_write() is just number today. > So how about this ? Code will be more simple/readable > > if ((id == 9) && (busif >= 4)) { > adinr = SSI9_BUSIF_ADINR(); > mode = SSI9_BUSIF_MODE(); > daligh = SSI9_BUSIF_DALIGN(); > } else { > adinr = SSI_BUSIF_ADINR(); > mode = SSI_BUSIF_MODE(); > daligh = SSI_BUSIF_DALIGN(); > } > > rsnd_mod_write(mod, adinr, > rsnd_get_adinr_bit(mod, io) | chnl); > rsnd_mod_write(mod, mode, > rsnd_get_busif_shift(io, mod) | 1); > rsnd_mod_write(mod, dalign, > rsnd_get_dalign(mod, io)); Done. Best regards Timo sound/soc/sh/rcar/gen.c | 24 ++++++++++++++++++++++++ sound/soc/sh/rcar/rsnd.h | 27 +++++++++++++++++++++++++++ sound/soc/sh/rcar/ssiu.c | 24 +++++++++++------------- 3 files changed, 62 insertions(+), 13 deletions(-) diff --git a/sound/soc/sh/rcar/gen.c b/sound/soc/sh/rcar/gen.c index 7cda601..af19010 100644 --- a/sound/soc/sh/rcar/gen.c +++ b/sound/soc/sh/rcar/gen.c @@ -255,6 +255,30 @@ static int rsnd_gen2_probe(struct rsnd_priv *priv) RSND_GEN_M_REG(SSI_MODE, 0xc, 0x80), RSND_GEN_M_REG(SSI_CTRL, 0x10, 0x80), RSND_GEN_M_REG(SSI_INT_ENABLE, 0x18, 0x80), + RSND_GEN_S_REG(SSI9_BUSIF0_MODE, 0x48c), + RSND_GEN_S_REG(SSI9_BUSIF0_ADINR, 0x484), + RSND_GEN_S_REG(SSI9_BUSIF0_DALIGN, 0x488), + RSND_GEN_S_REG(SSI9_BUSIF1_MODE, 0x4a0), + RSND_GEN_S_REG(SSI9_BUSIF1_ADINR, 0x4a4), + RSND_GEN_S_REG(SSI9_BUSIF1_DALIGN, 0x4a8), + RSND_GEN_S_REG(SSI9_BUSIF2_MODE, 0x4c0), + RSND_GEN_S_REG(SSI9_BUSIF2_ADINR, 0x4c4), + RSND_GEN_S_REG(SSI9_BUSIF2_DALIGN, 0x4c8), + RSND_GEN_S_REG(SSI9_BUSIF3_MODE, 0x4e0), + RSND_GEN_S_REG(SSI9_BUSIF3_ADINR, 0x4e4), + RSND_GEN_S_REG(SSI9_BUSIF3_DALIGN, 0x4e8), + RSND_GEN_S_REG(SSI9_BUSIF4_MODE, 0xd80), + RSND_GEN_S_REG(SSI9_BUSIF4_ADINR, 0xd84), + RSND_GEN_S_REG(SSI9_BUSIF4_DALIGN, 0xd88), + RSND_GEN_S_REG(SSI9_BUSIF5_MODE, 0xda0), + RSND_GEN_S_REG(SSI9_BUSIF5_ADINR, 0xda4), + RSND_GEN_S_REG(SSI9_BUSIF5_DALIGN, 0xda8), + RSND_GEN_S_REG(SSI9_BUSIF6_MODE, 0xdc0), + RSND_GEN_S_REG(SSI9_BUSIF6_ADINR, 0xdc4), + RSND_GEN_S_REG(SSI9_BUSIF6_DALIGN, 0xdc8), + RSND_GEN_S_REG(SSI9_BUSIF7_MODE, 0xde0), + RSND_GEN_S_REG(SSI9_BUSIF7_ADINR, 0xde4), + RSND_GEN_S_REG(SSI9_BUSIF7_DALIGN, 0xde8), }; static const struct rsnd_regmap_field_conf conf_scu[] = { diff --git a/sound/soc/sh/rcar/rsnd.h b/sound/soc/sh/rcar/rsnd.h index 605e4b9..90625c5 100644 --- a/sound/soc/sh/rcar/rsnd.h +++ b/sound/soc/sh/rcar/rsnd.h @@ -191,6 +191,30 @@ enum rsnd_reg { SSI_SYS_STATUS7, HDMI0_SEL, HDMI1_SEL, + SSI9_BUSIF0_MODE, + SSI9_BUSIF1_MODE, + SSI9_BUSIF2_MODE, + SSI9_BUSIF3_MODE, + SSI9_BUSIF4_MODE, + SSI9_BUSIF5_MODE, + SSI9_BUSIF6_MODE, + SSI9_BUSIF7_MODE, + SSI9_BUSIF0_ADINR, + SSI9_BUSIF1_ADINR, + SSI9_BUSIF2_ADINR, + SSI9_BUSIF3_ADINR, + SSI9_BUSIF4_ADINR, + SSI9_BUSIF5_ADINR, + SSI9_BUSIF6_ADINR, + SSI9_BUSIF7_ADINR, + SSI9_BUSIF0_DALIGN, + SSI9_BUSIF1_DALIGN, + SSI9_BUSIF2_DALIGN, + SSI9_BUSIF3_DALIGN, + SSI9_BUSIF4_DALIGN, + SSI9_BUSIF5_DALIGN, + SSI9_BUSIF6_DALIGN, + SSI9_BUSIF7_DALIGN, /* SSI */ SSICR, @@ -209,6 +233,9 @@ enum rsnd_reg { #define SSI_BUSIF_MODE(i) (SSI_BUSIF0_MODE + (i)) #define SSI_BUSIF_ADINR(i) (SSI_BUSIF0_ADINR + (i)) #define SSI_BUSIF_DALIGN(i) (SSI_BUSIF0_DALIGN + (i)) +#define SSI9_BUSIF_MODE(i) (SSI9_BUSIF0_MODE + (i)) +#define SSI9_BUSIF_ADINR(i) (SSI9_BUSIF0_ADINR + (i)) +#define SSI9_BUSIF_DALIGN(i) (SSI9_BUSIF0_DALIGN + (i)) #define SSI_SYS_STATUS(i) (SSI_SYS_STATUS0 + (i)) diff --git a/sound/soc/sh/rcar/ssiu.c b/sound/soc/sh/rcar/ssiu.c index c74991d..2347f34 100644 --- a/sound/soc/sh/rcar/ssiu.c +++ b/sound/soc/sh/rcar/ssiu.c @@ -181,28 +181,26 @@ static int rsnd_ssiu_init_gen2(struct rsnd_mod *mod, if (rsnd_ssi_use_busif(io)) { int id = rsnd_mod_id(mod); int busif = rsnd_mod_id_sub(mod); + enum rsnd_reg adinr_reg, mode_reg, dalign_reg; - /* - * FIXME - * - * We can't support SSI9-4/5/6/7, because its address is - * out of calculation rule - */ if ((id == 9) && (busif >= 4)) { - struct device *dev = rsnd_priv_to_dev(priv); - - dev_err(dev, "This driver doesn't support SSI%d-%d, so far", - id, busif); + adinr_reg = SSI9_BUSIF_ADINR(busif); + mode_reg = SSI9_BUSIF_MODE(busif); + dalign_reg = SSI9_BUSIF_DALIGN(busif); + } else { + adinr_reg = SSI_BUSIF_ADINR(busif); + mode_reg = SSI_BUSIF_MODE(busif); + dalign_reg = SSI_BUSIF_DALIGN(busif); } - rsnd_mod_write(mod, SSI_BUSIF_ADINR(busif), + rsnd_mod_write(mod, adinr_reg, rsnd_get_adinr_bit(mod, io) | (rsnd_io_is_play(io) ? rsnd_runtime_channel_after_ctu(io) : rsnd_runtime_channel_original(io))); - rsnd_mod_write(mod, SSI_BUSIF_MODE(busif), + rsnd_mod_write(mod, mode_reg, rsnd_get_busif_shift(io, mod) | 1); - rsnd_mod_write(mod, SSI_BUSIF_DALIGN(busif), + rsnd_mod_write(mod, dalign_reg, rsnd_get_dalign(mod, io)); } -- 2.7.4

3 2

[alsa-devel] Applied "ASoC: Intel: Boards: Add Maxim98373 support" to the asoc tree
by Mark Brown 26 Feb '19

26 Feb '19

The patch ASoC: Intel: Boards: Add Maxim98373 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 716d53cc7837aec7f439ce2a20fc2597a89dae53 Mon Sep 17 00:00:00 2001 From: Jenny TC <jenny.tc(a)intel.com> Date: Mon, 25 Feb 2019 22:17:31 +0530 Subject: [PATCH] ASoC: Intel: Boards: Add Maxim98373 support This patch enables the reuse of kbl_da7219_max98927 machine driver to support max98373. The same machine driver is modified for cases where one amplifier is swapped out with another. Most of the changes are about renaming the codec and codec_dai names, with minor differences due to support for 24 bits in one case and 16 in the other. Signed-off-by: Jenny TC <jenny.tc(a)intel.com> Acked-by: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/intel/boards/Kconfig | 1 + sound/soc/intel/boards/kbl_da7219_max98927.c | 203 ++++++++++++++++-- .../intel/common/soc-acpi-intel-kbl-match.c | 19 ++ 3 files changed, 200 insertions(+), 23 deletions(-) diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 0a7e40d06395..12d6b73e9531 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -293,6 +293,7 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98927_MACH depends on MFD_INTEL_LPSS && I2C && ACPI select SND_SOC_DA7219 select SND_SOC_MAX98927 + select SND_SOC_MAX98373 select SND_SOC_DMIC select SND_SOC_HDAC_HDMI help diff --git a/sound/soc/intel/boards/kbl_da7219_max98927.c b/sound/soc/intel/boards/kbl_da7219_max98927.c index 6dd5c69671b3..2768a572d065 100644 --- a/sound/soc/intel/boards/kbl_da7219_max98927.c +++ b/sound/soc/intel/boards/kbl_da7219_max98927.c @@ -2,7 +2,7 @@ // Copyright(c) 2018 Intel Corporation. /* - * Intel Kabylake I2S Machine Driver with MAX98927 & DA7219 Codecs + * Intel Kabylake I2S Machine Driver with MAX98927, MAX98373 & DA7219 Codecs * * Modified from: * Intel Kabylake I2S Machine driver supporting MAX98927 and @@ -24,8 +24,14 @@ #define KBL_DIALOG_CODEC_DAI "da7219-hifi" #define MAX98927_CODEC_DAI "max98927-aif1" -#define MAXIM_DEV0_NAME "i2c-MX98927:00" -#define MAXIM_DEV1_NAME "i2c-MX98927:01" +#define MAX98927_DEV0_NAME "i2c-MX98927:00" +#define MAX98927_DEV1_NAME "i2c-MX98927:01" + +#define MAX98373_CODEC_DAI "max98373-aif1" +#define MAX98373_DEV0_NAME "i2c-MX98373:00" +#define MAX98373_DEV1_NAME "i2c-MX98373:01" + + #define DUAL_CHANNEL 2 #define QUAD_CHANNEL 4 #define NAME_SIZE 32 @@ -176,20 +182,38 @@ static int kabylake_ssp0_hw_params(struct snd_pcm_substream *substream, for (j = 0; j < runtime->num_codecs; j++) { struct snd_soc_dai *codec_dai = runtime->codec_dais[j]; - if (!strcmp(codec_dai->component->name, MAXIM_DEV0_NAME)) { + if (!strcmp(codec_dai->component->name, MAX98927_DEV0_NAME)) { ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x30, 3, 8, 16); if (ret < 0) { dev_err(runtime->dev, "DEV0 TDM slot err:%d\n", ret); return ret; } } - if (!strcmp(codec_dai->component->name, MAXIM_DEV1_NAME)) { + if (!strcmp(codec_dai->component->name, MAX98927_DEV1_NAME)) { ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xC0, 3, 8, 16); if (ret < 0) { dev_err(runtime->dev, "DEV1 TDM slot err:%d\n", ret); return ret; } } + if (!strcmp(codec_dai->component->name, MAX98373_DEV0_NAME)) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, + 0x03, 3, 8, 24); + if (ret < 0) { + dev_err(runtime->dev, + "DEV0 TDM slot err:%d\n", ret); + return ret; + } + } + if (!strcmp(codec_dai->component->name, MAX98373_DEV1_NAME)) { + ret = snd_soc_dai_set_tdm_slot(codec_dai, + 0x0C, 3, 8, 24); + if (ret < 0) { + dev_err(runtime->dev, + "DEV0 TDM slot err:%d\n", ret); + return ret; + } + } } return 0; @@ -212,6 +236,25 @@ static int kabylake_ssp_fixup(struct snd_soc_pcm_runtime *rtd, struct snd_soc_dai_link *fe_dai_link = dpcm->fe->dai_link; struct snd_soc_dai_link *be_dai_link = dpcm->be->dai_link; + /* + * Topology for kblda7219m98373 & kblmax98373 supports only S24_LE, + * where as kblda7219m98927 & kblmax98927 supports S16_LE by default. + * Skipping the port wise FE and BE configuration for kblda7219m98373 & + * kblmax98373 as the topology (FE & BE) supports S24_LE only. + */ + + if (!strcmp(rtd->card->name, "kblda7219m98373") || + !strcmp(rtd->card->name, "kblmax98373")) { + /* The ADSP will convert the FE rate to 48k, stereo */ + rate->min = rate->max = 48000; + channels->min = channels->max = DUAL_CHANNEL; + + /* set SSP to 24 bit */ + snd_mask_none(fmt); + snd_mask_set_format(fmt, SNDRV_PCM_FORMAT_S24_LE); + return 0; + } + /* * The ADSP will convert the FE rate to 48k, stereo, 24 bit */ @@ -352,20 +395,31 @@ static struct snd_pcm_hw_constraint_list constraints_channels_quad = { static int kbl_fe_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *soc_rt = substream->private_data; /* * On this platform for PCM device we support, * 48Khz * stereo - * 16 bit audio */ runtime->hw.channels_max = DUAL_CHANNEL; snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &constraints_channels); - - runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; - snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); + /* + * Setup S24_LE (32 bit container and 24 bit valid data) for + * kblda7219m98373 & kblmax98373. For kblda7219m98927 & + * kblmax98927 keeping it as 16/16 due to topology FW dependency. + */ + if (!strcmp(soc_rt->card->name, "kblda7219m98373") || + !strcmp(soc_rt->card->name, "kblmax98373")) { + runtime->hw.formats = SNDRV_PCM_FMTBIT_S24_LE; + snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); + + } else { + runtime->hw.formats = SNDRV_PCM_FMTBIT_S16_LE; + snd_pcm_hw_constraint_msbits(runtime, 0, 16, 16); + } snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); @@ -398,11 +452,23 @@ static int kabylake_dmic_fixup(struct snd_soc_pcm_runtime *rtd, static int kabylake_dmic_startup(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *soc_rt = substream->private_data; runtime->hw.channels_min = runtime->hw.channels_max = QUAD_CHANNEL; snd_pcm_hw_constraint_list(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, &constraints_channels_quad); + /* + * Topology for kblda7219m98373 & kblmax98373 supports only S24_LE. + * The DMIC also configured for S24_LE. Forcing the DMIC format to + * S24_LE due to the topology FW dependency. + */ + if (!strcmp(soc_rt->card->name, "kblda7219m98373") || + !strcmp(soc_rt->card->name, "kblmax98373")) { + runtime->hw.formats = SNDRV_PCM_FMTBIT_S24_LE; + snd_pcm_hw_constraint_msbits(runtime, 0, 32, 24); + } + return snd_pcm_hw_constraint_list(substream->runtime, 0, SNDRV_PCM_HW_PARAM_RATE, &constraints_rates); } @@ -448,29 +514,55 @@ static struct snd_soc_ops skylake_refcap_ops = { static struct snd_soc_codec_conf max98927_codec_conf[] = { { - .dev_name = MAXIM_DEV0_NAME, + .dev_name = MAX98927_DEV0_NAME, + .name_prefix = "Right", + }, + + { + .dev_name = MAX98927_DEV1_NAME, + .name_prefix = "Left", + }, +}; + +static struct snd_soc_codec_conf max98373_codec_conf[] = { + + { + .dev_name = MAX98373_DEV0_NAME, .name_prefix = "Right", }, { - .dev_name = MAXIM_DEV1_NAME, + .dev_name = MAX98373_DEV1_NAME, .name_prefix = "Left", }, }; -static struct snd_soc_dai_link_component ssp0_codec_components[] = { +static struct snd_soc_dai_link_component max98927_ssp0_codec_components[] = { { /* Left */ - .name = MAXIM_DEV0_NAME, + .name = MAX98927_DEV0_NAME, .dai_name = MAX98927_CODEC_DAI, }, { /* For Right */ - .name = MAXIM_DEV1_NAME, + .name = MAX98927_DEV1_NAME, .dai_name = MAX98927_CODEC_DAI, }, }; +static struct snd_soc_dai_link_component max98373_ssp0_codec_components[] = { + { /* Left */ + .name = MAX98373_DEV0_NAME, + .dai_name = MAX98373_CODEC_DAI, + }, + + { /* For Right */ + .name = MAX98373_DEV1_NAME, + .dai_name = MAX98373_CODEC_DAI, + }, + +}; + /* kabylake digital audio interface glue - connects codec <--> CPU */ static struct snd_soc_dai_link kabylake_dais[] = { /* Front End DAI links */ @@ -607,8 +699,8 @@ static struct snd_soc_dai_link kabylake_dais[] = { .cpu_dai_name = "SSP0 Pin", .platform_name = "0000:00:1f.3", .no_pcm = 1, - .codecs = ssp0_codec_components, - .num_codecs = ARRAY_SIZE(ssp0_codec_components), + .codecs = max98927_ssp0_codec_components, + .num_codecs = ARRAY_SIZE(max98927_ssp0_codec_components), .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, @@ -683,7 +775,7 @@ static struct snd_soc_dai_link kabylake_dais[] = { }; /* kabylake digital audio interface glue - connects codec <--> CPU */ -static struct snd_soc_dai_link kabylake_max98927_dais[] = { +static struct snd_soc_dai_link kabylake_max98_927_373_dais[] = { /* Front End DAI links */ [KBL_DPCM_AUDIO_PB] = { .name = "Kbl Audio Port", @@ -802,8 +894,8 @@ static struct snd_soc_dai_link kabylake_max98927_dais[] = { .cpu_dai_name = "SSP0 Pin", .platform_name = "0000:00:1f.3", .no_pcm = 1, - .codecs = ssp0_codec_components, - .num_codecs = ARRAY_SIZE(ssp0_codec_components), + .codecs = max98927_ssp0_codec_components, + .num_codecs = ARRAY_SIZE(max98927_ssp0_codec_components), .dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, @@ -917,8 +1009,8 @@ static struct snd_soc_card kbl_audio_card_da7219_m98927 = { static struct snd_soc_card kbl_audio_card_max98927 = { .name = "kblmax98927", .owner = THIS_MODULE, - .dai_link = kabylake_max98927_dais, - .num_links = ARRAY_SIZE(kabylake_max98927_dais), + .dai_link = kabylake_max98_927_373_dais, + .num_links = ARRAY_SIZE(kabylake_max98_927_373_dais), .controls = kabylake_controls, .num_controls = ARRAY_SIZE(kabylake_controls), .dapm_widgets = kabylake_widgets, @@ -931,9 +1023,46 @@ static struct snd_soc_card kbl_audio_card_max98927 = { .late_probe = kabylake_card_late_probe, }; +static struct snd_soc_card kbl_audio_card_da7219_m98373 = { + .name = "kblda7219m98373", + .owner = THIS_MODULE, + .dai_link = kabylake_dais, + .num_links = ARRAY_SIZE(kabylake_dais), + .controls = kabylake_controls, + .num_controls = ARRAY_SIZE(kabylake_controls), + .dapm_widgets = kabylake_widgets, + .num_dapm_widgets = ARRAY_SIZE(kabylake_widgets), + .dapm_routes = kabylake_map, + .num_dapm_routes = ARRAY_SIZE(kabylake_map), + .codec_conf = max98373_codec_conf, + .num_configs = ARRAY_SIZE(max98373_codec_conf), + .fully_routed = true, + .late_probe = kabylake_card_late_probe, +}; + +static struct snd_soc_card kbl_audio_card_max98373 = { + .name = "kblmax98373", + .owner = THIS_MODULE, + .dai_link = kabylake_max98_927_373_dais, + .num_links = ARRAY_SIZE(kabylake_max98_927_373_dais), + .controls = kabylake_controls, + .num_controls = ARRAY_SIZE(kabylake_controls), + .dapm_widgets = kabylake_widgets, + .num_dapm_widgets = ARRAY_SIZE(kabylake_widgets), + .dapm_routes = kabylake_map, + .num_dapm_routes = ARRAY_SIZE(kabylake_map), + .codec_conf = max98373_codec_conf, + .num_configs = ARRAY_SIZE(max98373_codec_conf), + .fully_routed = true, + .late_probe = kabylake_card_late_probe, +}; + static int kabylake_audio_probe(struct platform_device *pdev) { struct kbl_codec_private *ctx; + struct snd_soc_dai_link *kbl_dai_link; + struct snd_soc_dai_link_component **codecs; + int i = 0; ctx = devm_kzalloc(&pdev->dev, sizeof(*ctx), GFP_KERNEL); if (!ctx) @@ -944,6 +1073,22 @@ static int kabylake_audio_probe(struct platform_device *pdev) kabylake_audio_card = (struct snd_soc_card *)pdev->id_entry->driver_data; + kbl_dai_link = kabylake_audio_card->dai_link; + + /* Update codecs for SSP0 with max98373 codec info */ + if (!strcmp(pdev->name, "kbl_da7219_max98373") || + (!strcmp(pdev->name, "kbl_max98373"))) { + for (i = 0; i < kabylake_audio_card->num_links; ++i) { + if (strcmp(kbl_dai_link[i].name, "SSP0-Codec")) + continue; + + codecs = &(kbl_dai_link[i].codecs); + *codecs = max98373_ssp0_codec_components; + kbl_dai_link[i].num_codecs = + ARRAY_SIZE(max98373_ssp0_codec_components); + break; + } + } kabylake_audio_card->dev = &pdev->dev; snd_soc_card_set_drvdata(kabylake_audio_card, ctx); @@ -961,13 +1106,23 @@ static const struct platform_device_id kbl_board_ids[] = { .driver_data = (kernel_ulong_t)&kbl_audio_card_max98927, }, + { + .name = "kbl_da7219_max98373", + .driver_data = + (kernel_ulong_t)&kbl_audio_card_da7219_m98373, + }, + { + .name = "kbl_max98373", + .driver_data = + (kernel_ulong_t)&kbl_audio_card_max98373, + }, { } }; static struct platform_driver kabylake_audio = { .probe = kabylake_audio_probe, .driver = { - .name = "kbl_da7219_max98927", + .name = "kbl_da7219_max98_927_373", .pm = &snd_soc_pm_ops, }, .id_table = kbl_board_ids, @@ -976,8 +1131,10 @@ static struct platform_driver kabylake_audio = { module_platform_driver(kabylake_audio) /* Module information */ -MODULE_DESCRIPTION("Audio KabyLake Machine driver for MAX98927 & DA7219"); +MODULE_DESCRIPTION("Audio KabyLake Machine driver for MAX98927/MAX98373 & DA7219"); MODULE_AUTHOR("Mac Chiang <mac.chiang(a)intel.com>"); MODULE_LICENSE("GPL v2"); MODULE_ALIAS("platform:kbl_da7219_max98927"); MODULE_ALIAS("platform:kbl_max98927"); +MODULE_ALIAS("platform:kbl_da7219_max98373"); +MODULE_ALIAS("platform:kbl_max98373"); diff --git a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c index e6fa6f470526..4b331058e807 100644 --- a/sound/soc/intel/common/soc-acpi-intel-kbl-match.c +++ b/sound/soc/intel/common/soc-acpi-intel-kbl-match.c @@ -37,6 +37,11 @@ static struct snd_soc_acpi_codecs kbl_7219_98927_codecs = { .codecs = {"MX98927"} }; +static struct snd_soc_acpi_codecs kbl_7219_98373_codecs = { + .num_codecs = 1, + .codecs = {"MX98373"} +}; + struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { { .id = "INT343A", @@ -106,6 +111,20 @@ struct snd_soc_acpi_mach snd_soc_acpi_intel_kbl_machines[] = { .drv_name = "kbl_rt5660", .fw_filename = "intel/dsp_fw_kbl.bin", }, + { + .id = "DLGS7219", + .drv_name = "kbl_da7219_max98373", + .fw_filename = "intel/dsp_fw_kbl.bin", + .machine_quirk = snd_soc_acpi_codec_list, + .quirk_data = &kbl_7219_98373_codecs, + .pdata = &skl_dmic_data + }, + { + .id = "MX98373", + .drv_name = "kbl_max98373", + .fw_filename = "intel/dsp_fw_kbl.bin", + .pdata = &skl_dmic_data + }, {}, }; EXPORT_SYMBOL_GPL(snd_soc_acpi_intel_kbl_machines); -- 2.20.1

1 0

[alsa-devel] [PATCH] ASoC: qcom: Kconfig: fix dependency for sdm845
by Cheng-Yi Chiang 26 Feb '19

26 Feb '19

SND_SOC_CROS_EC_CODEC depends on MFD_CROS_EC. Add that dependency to SND_SOC_SDM845 to fix unmet direct dependencies warning. Fixes: 74c6ecf4194e (ASoC: qcom: Kconfig: select dmic for sdm845) Signed-off-by: Cheng-Yi Chiang <cychiang(a)chromium.org> --- sound/soc/qcom/Kconfig | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/qcom/Kconfig b/sound/soc/qcom/Kconfig index 8f206cb4fcc0..75ceb04d8bf0 100644 --- a/sound/soc/qcom/Kconfig +++ b/sound/soc/qcom/Kconfig @@ -98,7 +98,7 @@ config SND_SOC_MSM8996 config SND_SOC_SDM845 tristate "SoC Machine driver for SDM845 boards" - depends on QCOM_APR + depends on QCOM_APR && MFD_CROS_EC select SND_SOC_QDSP6 select SND_SOC_QCOM_COMMON select SND_SOC_RT5663 -- 2.21.0.rc0.258.g878e2cd30e-goog

4 3

[alsa-devel] [PATCH v2] arm64: dts: renesas: r8a77965: add SSIU support for sound
by twischer＠de.adit-jv.com 26 Feb '19

26 Feb '19

From: Jiada Wang <jiada_wang(a)mentor.com> rsnd driver supports SSIU now, let's use it. Then, BUSIF DMA settings on rcar_sound,ssi (= rxu, txu) are no longer needed. Applies commit 8d14bfa074db ("arm64: dts: renesas: r8a7796: add SSIU support for sound") and commit 10bd03fa896e ("arm64: dts: renesas: r8a7796: remove BUSIF0 settings from rcar_sound,ssi") for r8a77965. Signed-off-by: Jiada Wang <jiada_wang(a)mentor.com> Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- Hi all, On 2/25/19 02:41, Kuninori Morimoto wrote: > I think ARM maintainer doesn't want duplicate/copied patches. > So, we want/need to merge into this patch, too > > 10bd03fa896ee5e25fa3548a3d1112b9beb76943 > ("arm64: dts: renesas: r8a7796: remove BUSIF0 settings from rcar_sound,ssi") Done. > And I guess this patch is related, so, we need to merge it too ? > > arm64: dts: renesas: r8a7796: use extended audio dmac register There are some ongoing discussions. I would expect some more delay for this patch. This patch is already helpful because BUSIF0-4 can be used without: arm64: dts: renesas: r8a7796: use extended audio dmac register Therefore I would like to not block this patch by arm64: dts: renesas: r8a7796: use extended audio dmac register Best regards Timo arch/arm64/boot/dts/renesas/r8a77965.dtsi | 251 +++++++++++++++++++++++++++--- 1 file changed, 231 insertions(+), 20 deletions(-) diff --git a/arch/arm64/boot/dts/renesas/r8a77965.dtsi b/arch/arm64/boot/dts/renesas/r8a77965.dtsi index 4b3730f..7c081d6 100644 --- a/arch/arm64/boot/dts/renesas/r8a77965.dtsi +++ b/arch/arm64/boot/dts/renesas/r8a77965.dtsi @@ -1585,56 +1585,267 @@ }; }; + rcar_sound,ssiu { + ssiu00: ssiu-0 { + dmas = <&audma0 0x15>, <&audma1 0x16>; + dma-names = "rx", "tx"; + }; + ssiu01: ssiu-1 { + dmas = <&audma0 0x35>, <&audma1 0x36>; + dma-names = "rx", "tx"; + }; + ssiu02: ssiu-2 { + dmas = <&audma0 0x37>, <&audma1 0x38>; + dma-names = "rx", "tx"; + }; + ssiu03: ssiu-3 { + dmas = <&audma0 0x47>, <&audma1 0x48>; + dma-names = "rx", "tx"; + }; + ssiu04: ssiu-4 { + dmas = <&audma0 0x3F>, <&audma1 0x40>; + dma-names = "rx", "tx"; + }; + ssiu05: ssiu-5 { + dmas = <&audma0 0x43>, <&audma1 0x44>; + dma-names = "rx", "tx"; + }; + ssiu06: ssiu-6 { + dmas = <&audma0 0x4F>, <&audma1 0x50>; + dma-names = "rx", "tx"; + }; + ssiu07: ssiu-7 { + dmas = <&audma0 0x53>, <&audma1 0x54>; + dma-names = "rx", "tx"; + }; + ssiu10: ssiu-8 { + dmas = <&audma0 0x49>, <&audma1 0x4a>; + dma-names = "rx", "tx"; + }; + ssiu11: ssiu-9 { + dmas = <&audma0 0x4B>, <&audma1 0x4C>; + dma-names = "rx", "tx"; + }; + ssiu12: ssiu-10 { + dmas = <&audma0 0x57>, <&audma1 0x58>; + dma-names = "rx", "tx"; + }; + ssiu13: ssiu-11 { + dmas = <&audma0 0x59>, <&audma1 0x5A>; + dma-names = "rx", "tx"; + }; + ssiu14: ssiu-12 { + dmas = <&audma0 0x5F>, <&audma1 0x60>; + dma-names = "rx", "tx"; + }; + ssiu15: ssiu-13 { + dmas = <&audma0 0xC3>, <&audma1 0xC4>; + dma-names = "rx", "tx"; + }; + ssiu16: ssiu-14 { + dmas = <&audma0 0xC7>, <&audma1 0xC8>; + dma-names = "rx", "tx"; + }; + ssiu17: ssiu-15 { + dmas = <&audma0 0xCB>, <&audma1 0xCC>; + dma-names = "rx", "tx"; + }; + ssiu20: ssiu-16 { + dmas = <&audma0 0x63>, <&audma1 0x64>; + dma-names = "rx", "tx"; + }; + ssiu21: ssiu-17 { + dmas = <&audma0 0x67>, <&audma1 0x68>; + dma-names = "rx", "tx"; + }; + ssiu22: ssiu-18 { + dmas = <&audma0 0x6B>, <&audma1 0x6C>; + dma-names = "rx", "tx"; + }; + ssiu23: ssiu-19 { + dmas = <&audma0 0x6D>, <&audma1 0x6E>; + dma-names = "rx", "tx"; + }; + ssiu24: ssiu-20 { + dmas = <&audma0 0xCF>, <&audma1 0xCE>; + dma-names = "rx", "tx"; + }; + ssiu25: ssiu-21 { + dmas = <&audma0 0xEB>, <&audma1 0xEC>; + dma-names = "rx", "tx"; + }; + ssiu26: ssiu-22 { + dmas = <&audma0 0xED>, <&audma1 0xEE>; + dma-names = "rx", "tx"; + }; + ssiu27: ssiu-23 { + dmas = <&audma0 0xEF>, <&audma1 0xF0>; + dma-names = "rx", "tx"; + }; + ssiu30: ssiu-24 { + dmas = <&audma0 0x6f>, <&audma1 0x70>; + dma-names = "rx", "tx"; + }; + ssiu31: ssiu-25 { + dmas = <&audma0 0x21>, <&audma1 0x22>; + dma-names = "rx", "tx"; + }; + ssiu32: ssiu-26 { + dmas = <&audma0 0x23>, <&audma1 0x24>; + dma-names = "rx", "tx"; + }; + ssiu33: ssiu-27 { + dmas = <&audma0 0x25>, <&audma1 0x26>; + dma-names = "rx", "tx"; + }; + ssiu34: ssiu-28 { + dmas = <&audma0 0x27>, <&audma1 0x28>; + dma-names = "rx", "tx"; + }; + ssiu35: ssiu-29 { + dmas = <&audma0 0x29>, <&audma1 0x2A>; + dma-names = "rx", "tx"; + }; + ssiu36: ssiu-30 { + dmas = <&audma0 0x2B>, <&audma1 0x2C>; + dma-names = "rx", "tx"; + }; + ssiu37: ssiu-31 { + dmas = <&audma0 0x2D>, <&audma1 0x2E>; + dma-names = "rx", "tx"; + }; + ssiu40: ssiu-32 { + dmas = <&audma0 0x71>, <&audma1 0x72>; + dma-names = "rx", "tx"; + }; + ssiu41: ssiu-33 { + dmas = <&audma0 0x17>, <&audma1 0x18>; + dma-names = "rx", "tx"; + }; + ssiu42: ssiu-34 { + dmas = <&audma0 0x19>, <&audma1 0x1A>; + dma-names = "rx", "tx"; + }; + ssiu43: ssiu-35 { + dmas = <&audma0 0x1B>, <&audma1 0x1C>; + dma-names = "rx", "tx"; + }; + ssiu44: ssiu-36 { + dmas = <&audma0 0x1D>, <&audma1 0x1E>; + dma-names = "rx", "tx"; + }; + ssiu45: ssiu-37 { + dmas = <&audma0 0x1F>, <&audma1 0x20>; + dma-names = "rx", "tx"; + }; + ssiu46: ssiu-38 { + dmas = <&audma0 0x31>, <&audma1 0x32>; + dma-names = "rx", "tx"; + }; + ssiu47: ssiu-39 { + dmas = <&audma0 0x33>, <&audma1 0x34>; + dma-names = "rx", "tx"; + }; + ssiu50: ssiu-40 { + dmas = <&audma0 0x73>, <&audma1 0x74>; + dma-names = "rx", "tx"; + }; + ssiu60: ssiu-41 { + dmas = <&audma0 0x75>, <&audma1 0x76>; + dma-names = "rx", "tx"; + }; + ssiu70: ssiu-42 { + dmas = <&audma0 0x79>, <&audma1 0x7a>; + dma-names = "rx", "tx"; + }; + ssiu80: ssiu-43 { + dmas = <&audma0 0x7b>, <&audma1 0x7c>; + dma-names = "rx", "tx"; + }; + ssiu90: ssiu-44 { + dmas = <&audma0 0x7d>, <&audma1 0x7e>; + dma-names = "rx", "tx"; + }; + ssiu91: ssiu-45 { + dmas = <&audma0 0x7F>, <&audma1 0x80>; + dma-names = "rx", "tx"; + }; + ssiu92: ssiu-46 { + dmas = <&audma0 0x81>, <&audma1 0x82>; + dma-names = "rx", "tx"; + }; + ssiu93: ssiu-47 { + dmas = <&audma0 0x83>, <&audma1 0x84>; + dma-names = "rx", "tx"; + }; + ssiu94: ssiu-48 { + dmas = <&audma0 0xA3>, <&audma1 0xA4>; + dma-names = "rx", "tx"; + }; + ssiu95: ssiu-49 { + dmas = <&audma0 0xA5>, <&audma1 0xA6>; + dma-names = "rx", "tx"; + }; + ssiu96: ssiu-50 { + dmas = <&audma0 0xA7>, <&audma1 0xA8>; + dma-names = "rx", "tx"; + }; + ssiu97: ssiu-51 { + dmas = <&audma0 0xA9>, <&audma1 0xAA>; + dma-names = "rx", "tx"; + }; + }; + rcar_sound,ssi { ssi0: ssi-0 { interrupts = <GIC_SPI 370 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x01>, <&audma1 0x02>, <&audma0 0x15>, <&audma1 0x16>; - dma-names = "rx", "tx", "rxu", "txu"; + dmas = <&audma0 0x01>, <&audma1 0x02>; + dma-names = "rx", "tx"; }; ssi1: ssi-1 { interrupts = <GIC_SPI 371 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x03>, <&audma1 0x04>, <&audma0 0x49>, <&audma1 0x4a>; - dma-names = "rx", "tx", "rxu", "txu"; + dmas = <&audma0 0x03>, <&audma1 0x04>; + dma-names = "rx", "tx"; }; ssi2: ssi-2 { interrupts = <GIC_SPI 372 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x05>, <&audma1 0x06>, <&audma0 0x63>, <&audma1 0x64>; - dma-names = "rx", "tx", "rxu", "txu"; + dmas = <&audma0 0x05>, <&audma1 0x06>; + dma-names = "rx", "tx"; }; ssi3: ssi-3 { interrupts = <GIC_SPI 373 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x07>, <&audma1 0x08>, <&audma0 0x6f>, <&audma1 0x70>; - dma-names = "rx", "tx", "rxu", "txu"; + dmas = <&audma0 0x07>, <&audma1 0x08>; + dma-names = "rx", "tx"; }; ssi4: ssi-4 { interrupts = <GIC_SPI 374 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x09>, <&audma1 0x0a>, <&audma0 0x71>, <&audma1 0x72>; - dma-names = "rx", "tx", "rxu", "txu"; + dmas = <&audma0 0x09>, <&audma1 0x0a>; + dma-names = "rx", "tx"; }; ssi5: ssi-5 { interrupts = <GIC_SPI 375 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x0b>, <&audma1 0x0c>, <&audma0 0x73>, <&audma1 0x74>; - dma-names = "rx", "tx", "rxu", "txu"; + dmas = <&audma0 0x0b>, <&audma1 0x0c>; + dma-names = "rx", "tx"; }; ssi6: ssi-6 { interrupts = <GIC_SPI 376 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x0d>, <&audma1 0x0e>, <&audma0 0x75>, <&audma1 0x76>; - dma-names = "rx", "tx", "rxu", "txu"; + dmas = <&audma0 0x0d>, <&audma1 0x0e>; + dma-names = "rx", "tx"; }; ssi7: ssi-7 { interrupts = <GIC_SPI 377 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x0f>, <&audma1 0x10>, <&audma0 0x79>, <&audma1 0x7a>; - dma-names = "rx", "tx", "rxu", "txu"; + dmas = <&audma0 0x0f>, <&audma1 0x10>; + dma-names = "rx", "tx"; }; ssi8: ssi-8 { interrupts = <GIC_SPI 378 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x11>, <&audma1 0x12>, <&audma0 0x7b>, <&audma1 0x7c>; - dma-names = "rx", "tx", "rxu", "txu"; + dmas = <&audma0 0x11>, <&audma1 0x12>; + dma-names = "rx", "tx"; }; ssi9: ssi-9 { interrupts = <GIC_SPI 379 IRQ_TYPE_LEVEL_HIGH>; - dmas = <&audma0 0x13>, <&audma1 0x14>, <&audma0 0x7d>, <&audma1 0x7e>; - dma-names = "rx", "tx", "rxu", "txu"; + dmas = <&audma0 0x13>, <&audma1 0x14>; + dma-names = "rx", "tx"; }; }; }; -- 2.7.4

3 2

[alsa-devel] [PATCH] ALSA: firewire-motu: fix construction of PCM frame for capture direction
by Takashi Sakamoto 26 Feb '19

26 Feb '19

In data blocks of common isochronous packet for MOTU devices, PCM frames are multiplexed in a shape of '24 bit * 4 Audio Pack', described in IEC 61883-6. The frames are not aligned to quadlet. For capture PCM substream, ALSA firewire-motu driver constructs PCM frames by reading data blocks byte-by-byte. However this operation includes bug for lower byte of the PCM sample. This brings invalid content of the PCM samples. This commit fixes the bug. Reported-by: Peter Sjöberg <autopeter(a)gmail.com> Cc: <stable(a)vger.kernel.org> # v4.12+ Fixes: 4641c9394010 ("ALSA: firewire-motu: add MOTU specific protocol layer") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/motu/amdtp-motu.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/firewire/motu/amdtp-motu.c b/sound/firewire/motu/amdtp-motu.c index f0555a24d90e..6c9b743ea74b 100644 --- a/sound/firewire/motu/amdtp-motu.c +++ b/sound/firewire/motu/amdtp-motu.c @@ -136,7 +136,9 @@ static void read_pcm_s32(struct amdtp_stream *s, byte = (u8 *)buffer + p->pcm_byte_offset; for (c = 0; c < channels; ++c) { - *dst = (byte[0] << 24) | (byte[1] << 16) | byte[2]; + *dst = (byte[0] << 24) | + (byte[1] << 16) | + (byte[2] << 8); byte += 3; dst++; } -- 2.19.1

2 1

[alsa-devel] [PATCH] ALSA: bebob: use more identical mod_alias for Saffire Pro 10 I/O against Liquid Saffire 56
by Takashi Sakamoto 26 Feb '19

26 Feb '19

ALSA bebob driver has an entry for Focusrite Saffire Pro 10 I/O. The entry matches vendor_id in root directory and model_id in unit directory of configuration ROM for IEEE 1394 bus. On the other hand, configuration ROM of Focusrite Liquid Saffire 56 has the same vendor_id and model_id. This device is an application of TCAT Dice (TCD2220 a.k.a Dice Jr.) however ALSA bebob driver can be bound to it randomly instead of ALSA dice driver. At present, drivers in ALSA firewire stack can not handle this situation appropriately. This commit uses more identical mod_alias for Focusrite Saffire Pro 10 I/O in ALSA bebob driver. $ python2 crpp < /sys/bus/firewire/devices/fw1/config_rom ROM header and bus information block ----------------------------------------------------------------- 400 042a829d bus_info_length 4, crc_length 42, crc 33437 404 31333934 bus_name "1394" 408 f0649222 irmc 1, cmc 1, isc 1, bmc 1, pmc 0, cyc_clk_acc 100, max_rec 9 (1024), max_rom 2, gen 2, spd 2 (S400) 40c 00130e01 company_id 00130e | 410 000606e0 device_id 01000606e0 | EUI-64 00130e01000606e0 root directory ----------------------------------------------------------------- 414 0009d31c directory_length 9, crc 54044 418 04000014 hardware version 41c 0c0083c0 node capabilities per IEEE 1394 420 0300130e vendor 424 81000012 --> descriptor leaf at 46c 428 17000006 model 42c 81000016 --> descriptor leaf at 484 430 130120c2 version 434 d1000002 --> unit directory at 43c 438 d4000006 --> dependent info directory at 450 unit directory at 43c ----------------------------------------------------------------- 43c 0004707c directory_length 4, crc 28796 440 1200a02d specifier id: 1394 TA 444 13010001 version: AV/C 448 17000006 model 44c 81000013 --> descriptor leaf at 498 dependent info directory at 450 ----------------------------------------------------------------- 450 000637c7 directory_length 6, crc 14279 454 120007f5 specifier id 458 13000001 version 45c 3affffc7 (immediate value) 460 3b100000 (immediate value) 464 3cffffc7 (immediate value) 468 3d600000 (immediate value) descriptor leaf at 46c ----------------------------------------------------------------- 46c 00056f3b leaf_length 5, crc 28475 470 00000000 textual descriptor 474 00000000 minimal ASCII 478 466f6375 "Focu" 47c 73726974 "srit" 480 65000000 "e" descriptor leaf at 484 ----------------------------------------------------------------- 484 0004a165 leaf_length 4, crc 41317 488 00000000 textual descriptor 48c 00000000 minimal ASCII 490 50726f31 "Pro1" 494 30494f00 "0IO" descriptor leaf at 498 ----------------------------------------------------------------- 498 0004a165 leaf_length 4, crc 41317 49c 00000000 textual descriptor 4a0 00000000 minimal ASCII 4a4 50726f31 "Pro1" 4a8 30494f00 "0IO" $ python2 crpp < /sys/bus/firewire/devices/fw1/config_rom ROM header and bus information block ----------------------------------------------------------------- 400 040442e4 bus_info_length 4, crc_length 4, crc 17124 404 31333934 bus_name "1394" 408 e0ff8112 irmc 1, cmc 1, isc 1, bmc 0, pmc 0, cyc_clk_acc 255, max_rec 8 (512), max_rom 1, gen 1, spd 2 (S400) 40c 00130e04 company_id 00130e | 410 018001e9 device_id 04018001e9 | EUI-64 00130e04018001e9 root directory ----------------------------------------------------------------- 414 00065612 directory_length 6, crc 22034 418 0300130e vendor 41c 8100000a --> descriptor leaf at 444 420 17000006 model 424 8100000e --> descriptor leaf at 45c 428 0c0087c0 node capabilities per IEEE 1394 42c d1000001 --> unit directory at 430 unit directory at 430 ----------------------------------------------------------------- 430 000418a0 directory_length 4, crc 6304 434 1200130e specifier id 438 13000001 version 43c 17000006 model 440 8100000f --> descriptor leaf at 47c descriptor leaf at 444 ----------------------------------------------------------------- 444 00056f3b leaf_length 5, crc 28475 448 00000000 textual descriptor 44c 00000000 minimal ASCII 450 466f6375 "Focu" 454 73726974 "srit" 458 65000000 "e" descriptor leaf at 45c ----------------------------------------------------------------- 45c 000762c6 leaf_length 7, crc 25286 460 00000000 textual descriptor 464 00000000 minimal ASCII 468 4c495155 "LIQU" 46c 49445f53 "ID_S" 470 41464649 "AFFI" 474 52455f35 "RE_5" 478 36000000 "6" descriptor leaf at 47c ----------------------------------------------------------------- 47c 000762c6 leaf_length 7, crc 25286 480 00000000 textual descriptor 484 00000000 minimal ASCII 488 4c495155 "LIQU" 48c 49445f53 "ID_S" 490 41464649 "AFFI" 494 52455f35 "RE_5" 498 36000000 "6" Cc: <stable(a)vger.kernel.org> # v3.16+ Fixes: 25784ec2d034 ("ALSA: bebob: Add support for Focusrite Saffire/SaffirePro series") Signed-off-by: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/firewire/bebob/bebob.c | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) diff --git a/sound/firewire/bebob/bebob.c b/sound/firewire/bebob/bebob.c index d91874275d2c..5b46e8dcc2dd 100644 --- a/sound/firewire/bebob/bebob.c +++ b/sound/firewire/bebob/bebob.c @@ -448,7 +448,19 @@ static const struct ieee1394_device_id bebob_id_table[] = { /* Focusrite, SaffirePro 26 I/O */ SND_BEBOB_DEV_ENTRY(VEN_FOCUSRITE, 0x00000003, &saffirepro_26_spec), /* Focusrite, SaffirePro 10 I/O */ - SND_BEBOB_DEV_ENTRY(VEN_FOCUSRITE, 0x00000006, &saffirepro_10_spec), + { + // The combination of vendor_id and model_id is the same as the + // same as the one of Liquid Saffire 56. + .match_flags = IEEE1394_MATCH_VENDOR_ID | + IEEE1394_MATCH_MODEL_ID | + IEEE1394_MATCH_SPECIFIER_ID | + IEEE1394_MATCH_VERSION, + .vendor_id = VEN_FOCUSRITE, + .model_id = 0x000006, + .specifier_id = 0x00a02d, + .version = 0x010001, + .driver_data = (kernel_ulong_t)&saffirepro_10_spec, + }, /* Focusrite, Saffire(no label and LE) */ SND_BEBOB_DEV_ENTRY(VEN_FOCUSRITE, MODEL_FOCUSRITE_SAFFIRE_BOTH, &saffire_spec), -- 2.19.1

2 1

[alsa-devel] [PATCH V2] ASoC: fsl_esai: fix channel swap issue when stream starts
by S.j. Wang 26 Feb '19

26 Feb '19

There is very low possibility ( < 0.1% ) that channel swap happened in beginning when multi output/input pin is enabled. The issue is that hardware can't send data to correct pin in the beginning with the normal enable flow. This is hardware issue, but there is no errata, the workaround flow is that: Each time playback/recording, firstly clear the xSMA/xSMB, then enable TE/RE, then enable xSMB and xSMA (xSMB must be enabled before xSMA). Which is to use the xSMA as the trigger start register, previously the xCR_TE or xCR_RE is the bit for starting. Fixes 43d24e76b698 ("ASoC: fsl_esai: Add ESAI CPU DAI driver") Cc: <stable(a)vger.kernel.org> Reviewed-by: Fabio Estevam <festevam(a)gmail.com> Signed-off-by: Shengjiu Wang <shengjiu.wang(a)nxp.com> --- Changes in v2 - update commit comments. sound/soc/fsl/fsl_esai.c | 32 ++++++++++++++++++++++---------- 1 file changed, 22 insertions(+), 10 deletions(-) diff --git a/sound/soc/fsl/fsl_esai.c b/sound/soc/fsl/fsl_esai.c index afe67c865330..23bd0ad4ac31 100644 --- a/sound/soc/fsl/fsl_esai.c +++ b/sound/soc/fsl/fsl_esai.c @@ -54,6 +54,8 @@ struct fsl_esai { u32 fifo_depth; u32 slot_width; u32 slots; + u32 tx_mask; + u32 rx_mask; u32 hck_rate[2]; u32 sck_rate[2]; bool hck_dir[2]; @@ -361,21 +363,13 @@ static int fsl_esai_set_dai_tdm_slot(struct snd_soc_dai *dai, u32 tx_mask, regmap_update_bits(esai_priv->regmap, REG_ESAI_TCCR, ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots)); - regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMA, - ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(tx_mask)); - regmap_update_bits(esai_priv->regmap, REG_ESAI_TSMB, - ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(tx_mask)); - regmap_update_bits(esai_priv->regmap, REG_ESAI_RCCR, ESAI_xCCR_xDC_MASK, ESAI_xCCR_xDC(slots)); - regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMA, - ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(rx_mask)); - regmap_update_bits(esai_priv->regmap, REG_ESAI_RSMB, - ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(rx_mask)); - esai_priv->slot_width = slot_width; esai_priv->slots = slots; + esai_priv->tx_mask = tx_mask; + esai_priv->rx_mask = rx_mask; return 0; } @@ -596,6 +590,7 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, bool tx = substream->stream == SNDRV_PCM_STREAM_PLAYBACK; u8 i, channels = substream->runtime->channels; u32 pins = DIV_ROUND_UP(channels, esai_priv->slots); + u32 mask; switch (cmd) { case SNDRV_PCM_TRIGGER_START: @@ -611,12 +606,23 @@ static int fsl_esai_trigger(struct snd_pcm_substream *substream, int cmd, regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK, tx ? ESAI_xCR_TE(pins) : ESAI_xCR_RE(pins)); + mask = tx ? esai_priv->tx_mask : esai_priv->rx_mask; + + regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMB(tx), + ESAI_xSMB_xS_MASK, ESAI_xSMB_xS(mask)); + regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMA(tx), + ESAI_xSMA_xS_MASK, ESAI_xSMA_xS(mask)); + break; case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: regmap_update_bits(esai_priv->regmap, REG_ESAI_xCR(tx), tx ? ESAI_xCR_TE_MASK : ESAI_xCR_RE_MASK, 0); + regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMA(tx), + ESAI_xSMA_xS_MASK, 0); + regmap_update_bits(esai_priv->regmap, REG_ESAI_xSMB(tx), + ESAI_xSMB_xS_MASK, 0); /* Disable and reset FIFO */ regmap_update_bits(esai_priv->regmap, REG_ESAI_xFCR(tx), @@ -906,6 +912,12 @@ static int fsl_esai_probe(struct platform_device *pdev) return ret; } + /* Clear the TSMA, TSMB, RSMA, RSMB */ + regmap_write(esai_priv->regmap, REG_ESAI_TSMA, 0); + regmap_write(esai_priv->regmap, REG_ESAI_TSMB, 0); + regmap_write(esai_priv->regmap, REG_ESAI_RSMA, 0); + regmap_write(esai_priv->regmap, REG_ESAI_RSMB, 0); + ret = devm_snd_soc_register_component(&pdev->dev, &fsl_esai_component, &fsl_esai_dai, 1); if (ret) { -- 1.9.1

2 4

[alsa-devel] [PATCH] ASoC:hdac_hda: use correct format to setup hda codec
by Rander Wang 25 Feb '19

25 Feb '19

The current implementation of the hdac_hda codec results in zero-valued samples on capture and noise with headset playback when SOF is used on platforms with an on-board HDaudio codec. This is root-caused to SOF using be_hw_params_fixup, and the prepare() call using invalid runtime fields to determine the format. This patch moves the format handling to the hw_params() callback, as done already for hdac_hdmi, to make sure the fixed-up information is taken into account but keeps the codec initialization in prepare() as the stream_tag is only available at that time. Moving everything in the prepare() callback is possible but the code is less elegant so this two-step solution was chosen. The solution was tested with the SST driver with no regressions, and all the issues with SOF playback and capture are solved. Signed-off-by: Rander Wang <rander.wang(a)linux.intel.com> --- sound/soc/codecs/hdac_hda.c | 53 +++++++++++++++++++++++++++++++++------------ sound/soc/codecs/hdac_hda.h | 1 + 2 files changed, 40 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/hdac_hda.c b/sound/soc/codecs/hdac_hda.c index ffecdaaa8cf2..f889d94c8e3c 100644 --- a/sound/soc/codecs/hdac_hda.c +++ b/sound/soc/codecs/hdac_hda.c @@ -38,6 +38,9 @@ static void hdac_hda_dai_close(struct snd_pcm_substream *substream, struct snd_soc_dai *dai); static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai); +static int hdac_hda_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai); static int hdac_hda_dai_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai); static int hdac_hda_dai_set_tdm_slot(struct snd_soc_dai *dai, @@ -50,6 +53,7 @@ static const struct snd_soc_dai_ops hdac_hda_dai_ops = { .startup = hdac_hda_dai_open, .shutdown = hdac_hda_dai_close, .prepare = hdac_hda_dai_prepare, + .hw_params = hdac_hda_dai_hw_params, .hw_free = hdac_hda_dai_hw_free, .set_tdm_slot = hdac_hda_dai_set_tdm_slot, }; @@ -139,6 +143,39 @@ static int hdac_hda_dai_set_tdm_slot(struct snd_soc_dai *dai, return 0; } +static int hdac_hda_dai_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_component *component = dai->component; + struct hdac_hda_priv *hda_pvt; + unsigned int format_val; + unsigned int maxbps; + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + maxbps = dai->driver->playback.sig_bits; + else + maxbps = dai->driver->capture.sig_bits; + + hda_pvt = snd_soc_component_get_drvdata(component); + format_val = snd_hdac_calc_stream_format(params_rate(params), + params_channels(params), + params_format(params), + maxbps, + 0); + if (!format_val) { + dev_err(dai->dev, + "invalid format_val, rate=%d, ch=%d, format=%d, maxbps=%d\n", + params_rate(params), params_channels(params), + params_format(params), maxbps); + + return -EINVAL; + } + + hda_pvt->pcm[dai->id].format_val[substream->stream] = format_val; + return 0; +} + static int hdac_hda_dai_hw_free(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { @@ -162,10 +199,9 @@ static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_component *component = dai->component; + struct hda_pcm_stream *hda_stream; struct hdac_hda_priv *hda_pvt; - struct snd_pcm_runtime *runtime = substream->runtime; struct hdac_device *hdev; - struct hda_pcm_stream *hda_stream; unsigned int format_val; struct hda_pcm *pcm; unsigned int stream; @@ -179,19 +215,8 @@ static int hdac_hda_dai_prepare(struct snd_pcm_substream *substream, hda_stream = &pcm->stream[substream->stream]; - format_val = snd_hdac_calc_stream_format(runtime->rate, - runtime->channels, - runtime->format, - hda_stream->maxbps, - 0); - if (!format_val) { - dev_err(&hdev->dev, - "invalid format_val, rate=%d, ch=%d, format=%d\n", - runtime->rate, runtime->channels, runtime->format); - return -EINVAL; - } - stream = hda_pvt->pcm[dai->id].stream_tag[substream->stream]; + format_val = hda_pvt->pcm[dai->id].format_val[substream->stream]; ret = snd_hda_codec_prepare(&hda_pvt->codec, hda_stream, stream, format_val, substream); diff --git a/sound/soc/codecs/hdac_hda.h b/sound/soc/codecs/hdac_hda.h index e444ef593360..6b1bd4f428e7 100644 --- a/sound/soc/codecs/hdac_hda.h +++ b/sound/soc/codecs/hdac_hda.h @@ -8,6 +8,7 @@ struct hdac_hda_pcm { int stream_tag[2]; + unsigned int format_val[2]; }; struct hdac_hda_priv { -- 2.14.1

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