From 2dc0f16b83b43fd1f86a2358d46f46488230c6c8 Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 21 Apr 2015 07:02:34 +0000 Subject: [PATCH 1/5] ASoC: soc.h: tidyup struct snd_soc_dai_link definition order Current struct snd_soc_dai_link has many members, but definition order was random. Especially, bool / bit field are defined randomly. This patch tidyups these definition order to calculate data alignment easy. Signed-off-by: Kuninori Morimoto Signed-off-by: Mark Brown --- include/sound/soc.h | 48 ++++++++++++++++++++++----------------------- 1 file changed, 24 insertions(+), 24 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index fcb312b3f258..38757fe7a3d8 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -949,30 +949,6 @@ struct snd_soc_dai_link { enum snd_soc_dpcm_trigger trigger[2]; /* trigger type for DPCM */ - /* Keep DAI active over suspend */ - unsigned int ignore_suspend:1; - - /* Symmetry requirements */ - unsigned int symmetric_rates:1; - unsigned int symmetric_channels:1; - unsigned int symmetric_samplebits:1; - - /* Mark this pcm with non atomic ops */ - bool nonatomic; - - /* Do not create a PCM for this DAI link (Backend link) */ - unsigned int no_pcm:1; - - /* This DAI link can route to other DAI links at runtime (Frontend)*/ - unsigned int dynamic:1; - - /* DPCM capture and Playback support */ - unsigned int dpcm_capture:1; - unsigned int dpcm_playback:1; - - /* pmdown_time is ignored at stop */ - unsigned int ignore_pmdown_time:1; - /* codec/machine specific init - e.g. add machine controls */ int (*init)(struct snd_soc_pcm_runtime *rtd); @@ -987,6 +963,30 @@ struct snd_soc_dai_link { /* For unidirectional dai links */ bool playback_only; bool capture_only; + + /* Mark this pcm with non atomic ops */ + bool nonatomic; + + /* Keep DAI active over suspend */ + unsigned int ignore_suspend:1; + + /* Symmetry requirements */ + unsigned int symmetric_rates:1; + unsigned int symmetric_channels:1; + unsigned int symmetric_samplebits:1; + + /* Do not create a PCM for this DAI link (Backend link) */ + unsigned int no_pcm:1; + + /* This DAI link can route to other DAI links at runtime (Frontend)*/ + unsigned int dynamic:1; + + /* DPCM capture and Playback support */ + unsigned int dpcm_capture:1; + unsigned int dpcm_playback:1; + + /* pmdown_time is ignored at stop */ + unsigned int ignore_pmdown_time:1; }; struct snd_soc_codec_conf { From 1f114f772ade64bca1c477322a18da8ed3bb8e6b Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Thu, 23 Apr 2015 16:16:04 +0300 Subject: [PATCH 2/5] ASoC: davinci-mcasp: Calculate BCLK using TDM slots and remove channels rule The McASP driver currently always sends as many slots or channels to a i2s-wire as there are configured tdm_slots (see mcasp_i2s_hw_param()). Thus the BLCK rate does not depend on the amount of channels, just the configure amount of tdm-slots. Reported-by: Misael Lopez Cruz Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 82 +++++-------------------------- 1 file changed, 12 insertions(+), 70 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index bb4b78eada58..a01c6db6017b 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -915,15 +915,12 @@ static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream, * the machine driver, we need to calculate the ratio. */ if (mcasp->bclk_master && mcasp->bclk_div == 0 && mcasp->sysclk_freq) { - int channels = params_channels(params); + int slots = mcasp->tdm_slots; int rate = params_rate(params); int sbits = params_width(params); int ppm, div; - if (channels > mcasp->tdm_slots) - channels = mcasp->tdm_slots; - - div = davinci_mcasp_calc_clk_div(mcasp, rate*sbits*channels, + div = davinci_mcasp_calc_clk_div(mcasp, rate*sbits*slots, &ppm); if (ppm) dev_info(mcasp->dev, "Sample-rate is off by %d PPM\n", @@ -1024,17 +1021,14 @@ static int davinci_mcasp_hw_rule_rate(struct snd_pcm_hw_params *params, struct snd_interval *ri = hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); int sbits = params_width(params); - int channels = params_channels(params); + int slots = rd->mcasp->tdm_slots; unsigned int list[ARRAY_SIZE(davinci_mcasp_dai_rates)]; int i, count = 0; - if (channels > rd->mcasp->tdm_slots) - channels = rd->mcasp->tdm_slots; - for (i = 0; i < ARRAY_SIZE(davinci_mcasp_dai_rates); i++) { if (ri->min <= davinci_mcasp_dai_rates[i] && ri->max >= davinci_mcasp_dai_rates[i]) { - uint bclk_freq = sbits*channels* + uint bclk_freq = sbits*slots* davinci_mcasp_dai_rates[i]; int ppm; @@ -1044,8 +1038,8 @@ static int davinci_mcasp_hw_rule_rate(struct snd_pcm_hw_params *params, } } dev_dbg(rd->mcasp->dev, - "%d frequencies (%d-%d) for %d sbits and %d channels\n", - count, ri->min, ri->max, sbits, channels); + "%d frequencies (%d-%d) for %d sbits and %d tdm slots\n", + count, ri->min, ri->max, sbits, slots); return snd_interval_list(hw_param_interval(params, rule->var), count, list, 0); @@ -1058,17 +1052,14 @@ static int davinci_mcasp_hw_rule_format(struct snd_pcm_hw_params *params, struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT); struct snd_mask nfmt; int rate = params_rate(params); - int channels = params_channels(params); + int slots = rd->mcasp->tdm_slots; int i, count = 0; snd_mask_none(&nfmt); - if (channels > rd->mcasp->tdm_slots) - channels = rd->mcasp->tdm_slots; - for (i = 0; i < SNDRV_PCM_FORMAT_LAST; i++) { if (snd_mask_test(fmt, i)) { - uint bclk_freq = snd_pcm_format_width(i)*channels*rate; + uint bclk_freq = snd_pcm_format_width(i)*slots*rate; int ppm; davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); @@ -1079,51 +1070,12 @@ static int davinci_mcasp_hw_rule_format(struct snd_pcm_hw_params *params, } } dev_dbg(rd->mcasp->dev, - "%d possible sample format for %d Hz and %d channels\n", - count, rate, channels); + "%d possible sample format for %d Hz and %d tdm slots\n", + count, rate, slots); return snd_mask_refine(fmt, &nfmt); } -static int davinci_mcasp_hw_rule_channels(struct snd_pcm_hw_params *params, - struct snd_pcm_hw_rule *rule) -{ - struct davinci_mcasp_ruledata *rd = rule->private; - struct snd_interval *ci = - hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); - int sbits = params_width(params); - int rate = params_rate(params); - int max_chan_per_wire = rd->mcasp->tdm_slots < ci->max ? - rd->mcasp->tdm_slots : ci->max; - unsigned int list[ci->max - ci->min + 1]; - int c1, c, count = 0; - - for (c1 = ci->min; c1 <= max_chan_per_wire; c1++) { - uint bclk_freq = c1*sbits*rate; - int ppm; - - davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); - if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) { - /* If we can use all tdm_slots, we can put any - amount of channels to remaining wires as - long as they fit in. */ - if (c1 == rd->mcasp->tdm_slots) { - for (c = c1; c <= rd->serializers*c1 && - c <= ci->max; c++) - list[count++] = c; - } else { - list[count++] = c1; - } - } - } - dev_dbg(rd->mcasp->dev, - "%d possible channel counts (%d-%d) for %d Hz and %d sbits\n", - count, ci->min, ci->max, rate, sbits); - - return snd_interval_list(hw_param_interval(params, rule->var), - count, list, 0); -} - static int davinci_mcasp_startup(struct snd_pcm_substream *substream, struct snd_soc_dai *cpu_dai) { @@ -1180,24 +1132,14 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, SNDRV_PCM_HW_PARAM_RATE, davinci_mcasp_hw_rule_rate, ruledata, - SNDRV_PCM_HW_PARAM_FORMAT, - SNDRV_PCM_HW_PARAM_CHANNELS, -1); + SNDRV_PCM_HW_PARAM_FORMAT, -1); if (ret) return ret; ret = snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT, davinci_mcasp_hw_rule_format, ruledata, - SNDRV_PCM_HW_PARAM_RATE, - SNDRV_PCM_HW_PARAM_CHANNELS, -1); - if (ret) - return ret; - ret = snd_pcm_hw_rule_add(substream->runtime, 0, - SNDRV_PCM_HW_PARAM_CHANNELS, - davinci_mcasp_hw_rule_channels, - ruledata, - SNDRV_PCM_HW_PARAM_RATE, - SNDRV_PCM_HW_PARAM_FORMAT, -1); + SNDRV_PCM_HW_PARAM_RATE, -1); if (ret) return ret; } From 5935a05626bc84810175e5f7b03b355a90769368 Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Thu, 23 Apr 2015 16:16:05 +0300 Subject: [PATCH 3/5] ASoC: davinci-mcasp: Channel count constraints for multi-serializer case Set channel count constraints for multiple serializers case. On McASP the active channels mask is the same for all the serializers. With the current implementation this means that if more than one serializers is used, all TDM slots have to be active on all serializers. The patch sets the channel count constraints according to number of RX and TX serializers. Reported-by: Misael Lopez Cruz Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 63 +++++++++++++++++++++++++++++++ 1 file changed, 63 insertions(+) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index a01c6db6017b..f8417072c66b 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -107,6 +107,7 @@ struct davinci_mcasp { #endif struct davinci_mcasp_ruledata ruledata[2]; + struct snd_pcm_hw_constraint_list chconstr[2]; }; static inline void mcasp_set_bits(struct davinci_mcasp *mcasp, u32 offset, @@ -1119,6 +1120,11 @@ static int davinci_mcasp_startup(struct snd_pcm_substream *substream, SNDRV_PCM_HW_PARAM_CHANNELS, 2, max_channels); + if (mcasp->chconstr[substream->stream].count) + snd_pcm_hw_constraint_list(substream->runtime, + 0, SNDRV_PCM_HW_PARAM_CHANNELS, + &mcasp->chconstr[substream->stream]); + /* * If we rely on implicit BCLK divider setting we should * set constraints based on what we can provide. @@ -1498,6 +1504,59 @@ nodata: return pdata; } +/* All serializers must have equal number of channels */ +static int davinci_mcasp_ch_constraint(struct davinci_mcasp *mcasp, + struct snd_pcm_hw_constraint_list *cl, + int serializers) +{ + unsigned int *list; + int i, count = 0; + + if (serializers <= 1) + return 0; + + list = devm_kzalloc(mcasp->dev, sizeof(unsigned int) * + (mcasp->tdm_slots + serializers - 2), + GFP_KERNEL); + if (!list) + return -ENOMEM; + + for (i = 2; i <= mcasp->tdm_slots; i++) + list[count++] = i; + + for (i = 2; i <= serializers; i++) + list[count++] = i*mcasp->tdm_slots; + + cl->count = count; + cl->list = list; + + return 0; +} + + +static int davinci_mcasp_init_ch_constraints(struct davinci_mcasp *mcasp) +{ + int rx_serializers = 0, tx_serializers = 0, ret, i; + + for (i = 0; i < mcasp->num_serializer; i++) + if (mcasp->serial_dir[i] == TX_MODE) + tx_serializers++; + else if (mcasp->serial_dir[i] == RX_MODE) + rx_serializers++; + + ret = davinci_mcasp_ch_constraint(mcasp, &mcasp->chconstr[ + SNDRV_PCM_STREAM_PLAYBACK], + tx_serializers); + if (ret) + return ret; + + ret = davinci_mcasp_ch_constraint(mcasp, &mcasp->chconstr[ + SNDRV_PCM_STREAM_CAPTURE], + rx_serializers); + + return ret; +} + static int davinci_mcasp_probe(struct platform_device *pdev) { struct snd_dmaengine_dai_dma_data *dma_data; @@ -1681,6 +1740,10 @@ static int davinci_mcasp_probe(struct platform_device *pdev) mcasp->fifo_base = DAVINCI_MCASP_V3_AFIFO_BASE; } + ret = davinci_mcasp_init_ch_constraints(mcasp); + if (ret) + goto err; + dev_set_drvdata(&pdev->dev, mcasp); mcasp_reparent_fck(pdev); From 518f6bab13842a5f25bd8f89b1cae32aa8adf91f Mon Sep 17 00:00:00 2001 From: Jyri Sarha Date: Thu, 23 Apr 2015 16:16:06 +0300 Subject: [PATCH 4/5] ASoC: davinci-macsp: Optimize implicit BLCK sample-rate rule There is no need to copy the list of all supported sample-rates. Finding the supported endpoints within the current range is enough (see snd_interval_list()). Signed-off-by: Jyri Sarha Signed-off-by: Mark Brown --- sound/soc/davinci/davinci-mcasp.c | 30 +++++++++++++++++++----------- 1 file changed, 19 insertions(+), 11 deletions(-) diff --git a/sound/soc/davinci/davinci-mcasp.c b/sound/soc/davinci/davinci-mcasp.c index f8417072c66b..56da8ce1faf3 100644 --- a/sound/soc/davinci/davinci-mcasp.c +++ b/sound/soc/davinci/davinci-mcasp.c @@ -1023,27 +1023,35 @@ static int davinci_mcasp_hw_rule_rate(struct snd_pcm_hw_params *params, hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); int sbits = params_width(params); int slots = rd->mcasp->tdm_slots; - unsigned int list[ARRAY_SIZE(davinci_mcasp_dai_rates)]; - int i, count = 0; + struct snd_interval range; + int i; + + snd_interval_any(&range); + range.empty = 1; for (i = 0; i < ARRAY_SIZE(davinci_mcasp_dai_rates); i++) { - if (ri->min <= davinci_mcasp_dai_rates[i] && - ri->max >= davinci_mcasp_dai_rates[i]) { + if (snd_interval_test(ri, davinci_mcasp_dai_rates[i])) { uint bclk_freq = sbits*slots* davinci_mcasp_dai_rates[i]; int ppm; davinci_mcasp_calc_clk_div(rd->mcasp, bclk_freq, &ppm); - if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) - list[count++] = davinci_mcasp_dai_rates[i]; + if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) { + if (range.empty) { + range.min = davinci_mcasp_dai_rates[i]; + range.empty = 0; + } + range.max = davinci_mcasp_dai_rates[i]; + } } } - dev_dbg(rd->mcasp->dev, - "%d frequencies (%d-%d) for %d sbits and %d tdm slots\n", - count, ri->min, ri->max, sbits, slots); - return snd_interval_list(hw_param_interval(params, rule->var), - count, list, 0); + dev_dbg(rd->mcasp->dev, + "Frequencies %d-%d -> %d-%d for %d sbits and %d tdm slots\n", + ri->min, ri->max, range.min, range.max, sbits, slots); + + return snd_interval_refine(hw_param_interval(params, rule->var), + &range); } static int davinci_mcasp_hw_rule_format(struct snd_pcm_hw_params *params, From b073ed4e21268da59c40a4fc5d56e3af808ecc4d Mon Sep 17 00:00:00 2001 From: Kuninori Morimoto Date: Tue, 12 May 2015 02:03:33 +0000 Subject: [PATCH 5/5] ASoC: soc-pcm: DPCM cares BE format Current DPCM is caring only FE format. but it will be no sound if FE/BE was below style, and user selects S24_LE format. FE: S16_LE/S24_LE BE: S16_LE DPCM can rewrite the format, so basically we don't want to constrain with the BE constraints. But sometimes it will be trouble. This patch adds new .dpcm_merged_format on struct snd_soc_dai_link. DPCM will use FE / BE merged format if .struct snd_soc_dai_link has it. We can have other .dpcm_merged_xxx in the future .dpcm_merged_foramt .dpcm_merged_rate .dpcm_merged_chan Signed-off-by: Kuninori Morimoto Tested-by: Keita Kobayashi Signed-off-by: Mark Brown --- include/sound/soc.h | 3 +++ sound/soc/soc-pcm.c | 47 ++++++++++++++++++++++++++++++++++++++++----- 2 files changed, 45 insertions(+), 5 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 38757fe7a3d8..cf63ac1c8968 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -985,6 +985,9 @@ struct snd_soc_dai_link { unsigned int dpcm_capture:1; unsigned int dpcm_playback:1; + /* DPCM used FE & BE merged format */ + unsigned int dpcm_merged_format:1; + /* pmdown_time is ignored at stop */ unsigned int ignore_pmdown_time:1; }; diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 35fe58f4fa86..256b9c91aa94 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1485,30 +1485,67 @@ unwind: } static void dpcm_init_runtime_hw(struct snd_pcm_runtime *runtime, - struct snd_soc_pcm_stream *stream) + struct snd_soc_pcm_stream *stream, + u64 formats) { runtime->hw.rate_min = stream->rate_min; runtime->hw.rate_max = stream->rate_max; runtime->hw.channels_min = stream->channels_min; runtime->hw.channels_max = stream->channels_max; if (runtime->hw.formats) - runtime->hw.formats &= stream->formats; + runtime->hw.formats &= formats & stream->formats; else - runtime->hw.formats = stream->formats; + runtime->hw.formats = formats & stream->formats; runtime->hw.rates = stream->rates; } +static u64 dpcm_runtime_base_format(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *fe = substream->private_data; + struct snd_soc_dpcm *dpcm; + u64 formats = ULLONG_MAX; + int stream = substream->stream; + + if (!fe->dai_link->dpcm_merged_format) + return formats; + + /* + * It returns merged BE codec format + * if FE want to use it (= dpcm_merged_format) + */ + + list_for_each_entry(dpcm, &fe->dpcm[stream].be_clients, list_be) { + struct snd_soc_pcm_runtime *be = dpcm->be; + struct snd_soc_dai_driver *codec_dai_drv; + struct snd_soc_pcm_stream *codec_stream; + int i; + + for (i = 0; i < be->num_codecs; i++) { + codec_dai_drv = be->codec_dais[i]->driver; + if (stream == SNDRV_PCM_STREAM_PLAYBACK) + codec_stream = &codec_dai_drv->playback; + else + codec_stream = &codec_dai_drv->capture; + + formats &= codec_stream->formats; + } + } + + return formats; +} + static void dpcm_set_fe_runtime(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; struct snd_soc_dai_driver *cpu_dai_drv = cpu_dai->driver; + u64 format = dpcm_runtime_base_format(substream); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - dpcm_init_runtime_hw(runtime, &cpu_dai_drv->playback); + dpcm_init_runtime_hw(runtime, &cpu_dai_drv->playback, format); else - dpcm_init_runtime_hw(runtime, &cpu_dai_drv->capture); + dpcm_init_runtime_hw(runtime, &cpu_dai_drv->capture, format); } static int dpcm_fe_dai_do_trigger(struct snd_pcm_substream *substream, int cmd);