September 2017 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2] ASoC: rt5670: add set_bclk_ratio in dai ops
by Bard Liao 15 Sep '17

15 Sep '17

We need to set a specific bit for 50 bclk rate. So add set_bclk_ratio function to set the bit. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- v2: * add define for tdm data format select bit. --- sound/soc/codecs/rt5670.c | 19 +++++++++++++++++++ sound/soc/codecs/rt5670.h | 4 ++++ 2 files changed, 23 insertions(+) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 9545764..da7614c 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -2582,6 +2582,24 @@ static int rt5670_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, return 0; } +static int rt5670_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_codec *codec = dai->codec; + + dev_dbg(codec->dev, "%s ratio=%d\n", __func__, ratio); + if (dai->id != RT5670_AIF1) + return 0; + + if (ratio == 50) + snd_soc_update_bits(codec, RT5670_GEN_CTRL3, + RT5670_TDM_DATA_MODE_SEL, RT5670_TDM_DATA_MODE_50FS); + else + snd_soc_update_bits(codec, RT5670_GEN_CTRL3, + RT5670_TDM_DATA_MODE_SEL, RT5670_TDM_DATA_MODE_NOR); + + return 0; +} + static int rt5670_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { @@ -2712,6 +2730,7 @@ static const struct snd_soc_dai_ops rt5670_aif_dai_ops = { .set_fmt = rt5670_set_dai_fmt, .set_tdm_slot = rt5670_set_tdm_slot, .set_pll = rt5670_set_dai_pll, + .set_bclk_ratio = rt5670_set_bclk_ratio, }; static struct snd_soc_dai_driver rt5670_dai[] = { diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index 5ba485c..265df80 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -1816,6 +1816,10 @@ #define RT5670_ZCD_HP_DIS (0x0 << 15) #define RT5670_ZCD_HP_EN (0x1 << 15) +/* General Control 3 (0xfc) */ +#define RT5670_TDM_DATA_MODE_SEL (0x1 << 11) +#define RT5670_TDM_DATA_MODE_NOR (0x0 << 11) +#define RT5670_TDM_DATA_MODE_50FS (0x1 << 11) /* Codec Private Register definition */ /* 3D Speaker Control (0x63) */ -- 2.7.4

2 6

[alsa-devel] [PATCH] ASoC: remove unneeded dai->driver check
by Kuninori Morimoto 15 Sep '17

15 Sep '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> On soc_add_dai(), it is checking dai->driver->xxx, This means, dai->driver is mandatory, never been NULL. dai->driver check is not needed. This patch removes it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-core.c | 16 ++++++++-------- 1 file changed, 8 insertions(+), 8 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0d170a0..f8696dc 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2634,7 +2634,7 @@ int snd_soc_add_dai_controls(struct snd_soc_dai *dai, int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id, unsigned int freq, int dir) { - if (dai->driver && dai->driver->ops->set_sysclk) + if (dai->driver->ops->set_sysclk) return dai->driver->ops->set_sysclk(dai, clk_id, freq, dir); return snd_soc_component_set_sysclk(dai->component, clk_id, 0, @@ -2702,7 +2702,7 @@ int snd_soc_component_set_sysclk(struct snd_soc_component *component, int clk_id int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, int div_id, int div) { - if (dai->driver && dai->driver->ops->set_clkdiv) + if (dai->driver->ops->set_clkdiv) return dai->driver->ops->set_clkdiv(dai, div_id, div); else return -EINVAL; @@ -2722,7 +2722,7 @@ int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai, int snd_soc_dai_set_pll(struct snd_soc_dai *dai, int pll_id, int source, unsigned int freq_in, unsigned int freq_out) { - if (dai->driver && dai->driver->ops->set_pll) + if (dai->driver->ops->set_pll) return dai->driver->ops->set_pll(dai, pll_id, source, freq_in, freq_out); @@ -2788,7 +2788,7 @@ int snd_soc_component_set_pll(struct snd_soc_component *component, int pll_id, */ int snd_soc_dai_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) { - if (dai->driver && dai->driver->ops->set_bclk_ratio) + if (dai->driver->ops->set_bclk_ratio) return dai->driver->ops->set_bclk_ratio(dai, ratio); else return -EINVAL; @@ -2862,7 +2862,7 @@ static int snd_soc_xlate_tdm_slot_mask(unsigned int slots, int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) { - if (dai->driver && dai->driver->ops->xlate_tdm_slot_mask) + if (dai->driver->ops->xlate_tdm_slot_mask) dai->driver->ops->xlate_tdm_slot_mask(slots, &tx_mask, &rx_mask); else @@ -2871,7 +2871,7 @@ int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, dai->tx_mask = tx_mask; dai->rx_mask = rx_mask; - if (dai->driver && dai->driver->ops->set_tdm_slot) + if (dai->driver->ops->set_tdm_slot) return dai->driver->ops->set_tdm_slot(dai, tx_mask, rx_mask, slots, slot_width); else @@ -2895,7 +2895,7 @@ int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai, unsigned int tx_num, unsigned int *tx_slot, unsigned int rx_num, unsigned int *rx_slot) { - if (dai->driver && dai->driver->ops->set_channel_map) + if (dai->driver->ops->set_channel_map) return dai->driver->ops->set_channel_map(dai, tx_num, tx_slot, rx_num, rx_slot); else @@ -2912,7 +2912,7 @@ int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai, */ int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate) { - if (dai->driver && dai->driver->ops->set_tristate) + if (dai->driver->ops->set_tristate) return dai->driver->ops->set_tristate(dai, tristate); else return -EINVAL; -- 1.9.1

1 0

[alsa-devel] [PATCH] ASoC: remove unneeded dai->driver->ops check
by Kuninori Morimoto 15 Sep '17

15 Sep '17

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> On soc_add_dai(), it uses null_dai_ops if driver doesn't have its own ops. This means, dai->driver->ops never been NULL. dai->driver->ops check is not needed. This patch removes it. Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/soc-dapm.c | 8 ++++---- sound/soc/soc-pcm.c | 31 +++++++++++++++---------------- 2 files changed, 19 insertions(+), 20 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index dcef67a..9d4748e 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3681,7 +3681,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, switch (event) { case SND_SOC_DAPM_PRE_PMU: substream.stream = SNDRV_PCM_STREAM_CAPTURE; - if (source->driver->ops && source->driver->ops->startup) { + if (source->driver->ops->startup) { ret = source->driver->ops->startup(&substream, source); if (ret < 0) { dev_err(source->dev, @@ -3695,7 +3695,7 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, goto out; substream.stream = SNDRV_PCM_STREAM_PLAYBACK; - if (sink->driver->ops && sink->driver->ops->startup) { + if (sink->driver->ops->startup) { ret = sink->driver->ops->startup(&substream, sink); if (ret < 0) { dev_err(sink->dev, @@ -3725,13 +3725,13 @@ static int snd_soc_dai_link_event(struct snd_soc_dapm_widget *w, ret = 0; source->active--; - if (source->driver->ops && source->driver->ops->shutdown) { + if (source->driver->ops->shutdown) { substream.stream = SNDRV_PCM_STREAM_CAPTURE; source->driver->ops->shutdown(&substream, source); } sink->active--; - if (sink->driver->ops && sink->driver->ops->shutdown) { + if (sink->driver->ops->shutdown) { substream.stream = SNDRV_PCM_STREAM_PLAYBACK; sink->driver->ops->shutdown(&substream, sink); } diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index da55e4e..e906b7c 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -483,7 +483,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) mutex_lock_nested(&rtd->pcm_mutex, rtd->pcm_subclass); /* startup the audio subsystem */ - if (cpu_dai->driver->ops && cpu_dai->driver->ops->startup) { + if (cpu_dai->driver->ops->startup) { ret = cpu_dai->driver->ops->startup(substream, cpu_dai); if (ret < 0) { dev_err(cpu_dai->dev, "ASoC: can't open interface" @@ -526,7 +526,7 @@ static int soc_pcm_open(struct snd_pcm_substream *substream) for (i = 0; i < rtd->num_codecs; i++) { codec_dai = rtd->codec_dais[i]; - if (codec_dai->driver->ops && codec_dai->driver->ops->startup) { + if (codec_dai->driver->ops->startup) { ret = codec_dai->driver->ops->startup(substream, codec_dai); if (ret < 0) { @@ -853,7 +853,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) for (i = 0; i < rtd->num_codecs; i++) { codec_dai = rtd->codec_dais[i]; - if (codec_dai->driver->ops && codec_dai->driver->ops->prepare) { + if (codec_dai->driver->ops->prepare) { ret = codec_dai->driver->ops->prepare(substream, codec_dai); if (ret < 0) { @@ -865,7 +865,7 @@ static int soc_pcm_prepare(struct snd_pcm_substream *substream) } } - if (cpu_dai->driver->ops && cpu_dai->driver->ops->prepare) { + if (cpu_dai->driver->ops->prepare) { ret = cpu_dai->driver->ops->prepare(substream, cpu_dai); if (ret < 0) { dev_err(cpu_dai->dev, @@ -911,7 +911,7 @@ int soc_dai_hw_params(struct snd_pcm_substream *substream, { int ret; - if (dai->driver->ops && dai->driver->ops->hw_params) { + if (dai->driver->ops->hw_params) { ret = dai->driver->ops->hw_params(substream, params, dai); if (ret < 0) { dev_err(dai->dev, "ASoC: can't set %s hw params: %d\n", @@ -1060,7 +1060,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, platform->driver->ops->hw_free(substream); platform_err: - if (cpu_dai->driver->ops && cpu_dai->driver->ops->hw_free) + if (cpu_dai->driver->ops->hw_free) cpu_dai->driver->ops->hw_free(substream, cpu_dai); interface_err: @@ -1069,7 +1069,7 @@ static int soc_pcm_hw_params(struct snd_pcm_substream *substream, codec_err: while (--i >= 0) { struct snd_soc_dai *codec_dai = rtd->codec_dais[i]; - if (codec_dai->driver->ops && codec_dai->driver->ops->hw_free) + if (codec_dai->driver->ops->hw_free) codec_dai->driver->ops->hw_free(substream, codec_dai); codec_dai->rate = 0; } @@ -1147,11 +1147,11 @@ static int soc_pcm_hw_free(struct snd_pcm_substream *substream) /* now free hw params for the DAIs */ for (i = 0; i < rtd->num_codecs; i++) { codec_dai = rtd->codec_dais[i]; - if (codec_dai->driver->ops && codec_dai->driver->ops->hw_free) + if (codec_dai->driver->ops->hw_free) codec_dai->driver->ops->hw_free(substream, codec_dai); } - if (cpu_dai->driver->ops && cpu_dai->driver->ops->hw_free) + if (cpu_dai->driver->ops->hw_free) cpu_dai->driver->ops->hw_free(substream, cpu_dai); mutex_unlock(&rtd->pcm_mutex); @@ -1170,7 +1170,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) for (i = 0; i < rtd->num_codecs; i++) { codec_dai = rtd->codec_dais[i]; - if (codec_dai->driver->ops && codec_dai->driver->ops->trigger) { + if (codec_dai->driver->ops->trigger) { ret = codec_dai->driver->ops->trigger(substream, cmd, codec_dai); if (ret < 0) @@ -1200,7 +1200,7 @@ static int soc_pcm_trigger(struct snd_pcm_substream *substream, int cmd) return ret; } - if (cpu_dai->driver->ops && cpu_dai->driver->ops->trigger) { + if (cpu_dai->driver->ops->trigger) { ret = cpu_dai->driver->ops->trigger(substream, cmd, cpu_dai); if (ret < 0) return ret; @@ -1225,8 +1225,7 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, for (i = 0; i < rtd->num_codecs; i++) { codec_dai = rtd->codec_dais[i]; - if (codec_dai->driver->ops && - codec_dai->driver->ops->bespoke_trigger) { + if (codec_dai->driver->ops->bespoke_trigger) { ret = codec_dai->driver->ops->bespoke_trigger(substream, cmd, codec_dai); if (ret < 0) @@ -1234,7 +1233,7 @@ static int soc_pcm_bespoke_trigger(struct snd_pcm_substream *substream, } } - if (cpu_dai->driver->ops && cpu_dai->driver->ops->bespoke_trigger) { + if (cpu_dai->driver->ops->bespoke_trigger) { ret = cpu_dai->driver->ops->bespoke_trigger(substream, cmd, cpu_dai); if (ret < 0) return ret; @@ -1279,12 +1278,12 @@ static snd_pcm_uframes_t soc_pcm_pointer(struct snd_pcm_substream *substream) break; } - if (cpu_dai->driver->ops && cpu_dai->driver->ops->delay) + if (cpu_dai->driver->ops->delay) delay += cpu_dai->driver->ops->delay(substream, cpu_dai); for (i = 0; i < rtd->num_codecs; i++) { codec_dai = rtd->codec_dais[i]; - if (codec_dai->driver->ops && codec_dai->driver->ops->delay) + if (codec_dai->driver->ops->delay) codec_delay = max(codec_delay, codec_dai->driver->ops->delay(substream, codec_dai)); -- 1.9.1

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[alsa-devel] [PATCH 4/4] ASoC: dapm: Remove the redundant check
by anish kumar 15 Sep '17

15 Sep '17

From: anish kumar <yesanishhere(a)gmail.com> Remove this redundant check, the check for creating the control in the case of codec to codec dai link is moved to snd_soc_dapm_new_pcm function. Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- sound/soc/soc-dapm.c | 4 ---- 1 file changed, 4 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 3c493b3..fcad345 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -1032,10 +1032,6 @@ static int dapm_new_dai_link(struct snd_soc_dapm_widget *w) struct snd_soc_dapm_context *dapm = w->dapm; struct snd_card *card = dapm->card->snd_card; - /* create control for links with > 1 config */ - if (w->num_params <= 1) - return 0; - /* add kcontrol */ for (i = 0; i < w->num_kcontrols; i++) { kcontrol = snd_soc_cnew(&w->kcontrol_news[i], w, -- 2.5.4 (Apple Git-61)

1 0

[alsa-devel] [PATCH 3/4] ASoC: dapm: Avoid creating kcontrol for params
by anish kumar 15 Sep '17

15 Sep '17

From: anish kumar <yesanishhere(a)gmail.com> Currently in codec to codec dai link if there are multiple params defined then dapm can use created kcontrol to decide which param to apply at runtime. However, in case there is only single param configuration then there is no point in creating the kcontrol and also there is no point in allocating memory for kcontrol. In the snd_soc_dapm_new_pcm function, there is memory allocation happening for kcontrol which is later used or not used based on num_param. It is better to not allocate memory when there is only a single configuration. This change is to remedy that anomaly. Signed-off-by: anish kumar <yesanishhere(a)gmail.com> --- sound/soc/soc-dapm.c | 114 +++++++++++++++++++++++++++++---------------------- 1 file changed, 66 insertions(+), 48 deletions(-) diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index d55cac6..3c493b3 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -3778,18 +3778,12 @@ static int snd_soc_dapm_dai_link_put(struct snd_kcontrol *kcontrol, return 0; } -int snd_soc_dapm_new_pcm(struct snd_soc_card *card, - const struct snd_soc_pcm_stream *params, - unsigned int num_params, - struct snd_soc_dapm_widget *source, - struct snd_soc_dapm_widget *sink) +static struct snd_kcontrol_new * +snd_soc_dapm_alloc_kcontrol(struct snd_soc_card *card, + char *link_name, + const struct snd_soc_pcm_stream *params, + int num_params) { - struct snd_soc_dapm_widget template; - struct snd_soc_dapm_widget *w; - char *link_name; - int ret, count; - unsigned long private_value; - const char **w_param_text; struct soc_enum w_param_enum[] = { SOC_ENUM_SINGLE(0, 0, 0, NULL), }; @@ -3798,19 +3792,16 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, snd_soc_dapm_dai_link_get, snd_soc_dapm_dai_link_put), }; + struct snd_kcontrol_new *kcontrol_news; const struct snd_soc_pcm_stream *config = params; + const char **w_param_text; + unsigned long private_value; + int count; w_param_text = devm_kcalloc(card->dev, num_params, - sizeof(char *), GFP_KERNEL); + sizeof(char *), GFP_KERNEL); if (!w_param_text) - return -ENOMEM; - - link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s", - source->name, sink->name); - if (!link_name) { - ret = -ENOMEM; - goto outfree_w_param; - } + return NULL; for (count = 0 ; count < num_params; count++) { if (!config->stream_name) { @@ -3827,24 +3818,14 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, strlen(config->stream_name) + 1, GFP_KERNEL); } - if (!w_param_text[count]) { - ret = -ENOMEM; - goto outfree_link_name; - } + if (!w_param_text[count]) + goto outfree_w_param; config++; } + w_param_enum[0].items = num_params; w_param_enum[0].texts = w_param_text; - memset(&template, 0, sizeof(template)); - template.reg = SND_SOC_NOPM; - template.id = snd_soc_dapm_dai_link; - template.name = link_name; - template.event = snd_soc_dai_link_event; - template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | - SND_SOC_DAPM_PRE_PMD; - template.num_kcontrols = 1; - /* duplicate w_param_enum on heap so that memory persists */ private_value = (unsigned long) devm_kmemdup(card->dev, (void *)(kcontrol_dai_link[0].private_value), @@ -3852,22 +3833,66 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, if (!private_value) { dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", link_name); - ret = -ENOMEM; - goto outfree_link_name; + goto outfree_w_param; } kcontrol_dai_link[0].private_value = private_value; /* duplicate kcontrol_dai_link on heap so that memory persists */ - template.kcontrol_news = - devm_kmemdup(card->dev, &kcontrol_dai_link[0], - sizeof(struct snd_kcontrol_new), - GFP_KERNEL); - if (!template.kcontrol_news) { + kcontrol_news = + devm_kmemdup(card->dev, &kcontrol_dai_link[0], + sizeof(struct snd_kcontrol_new), + GFP_KERNEL); + if (!kcontrol_news) { dev_err(card->dev, "ASoC: Failed to create control for %s widget\n", link_name); - ret = -ENOMEM; goto outfree_private_value; } + return kcontrol_news; + +outfree_private_value: + devm_kfree(card->dev, (void *)private_value); +outfree_w_param: + for (count = 0 ; count < num_params; count++) + devm_kfree(card->dev, (void *)w_param_text[count]); + devm_kfree(card->dev, w_param_text); + return NULL; +} + +int snd_soc_dapm_new_pcm(struct snd_soc_card *card, + const struct snd_soc_pcm_stream *params, + unsigned int num_params, + struct snd_soc_dapm_widget *source, + struct snd_soc_dapm_widget *sink) +{ + struct snd_soc_dapm_widget template; + struct snd_soc_dapm_widget *w; + char *link_name; + int ret; + + link_name = devm_kasprintf(card->dev, GFP_KERNEL, "%s-%s", + source->name, sink->name); + if (!link_name) + return -ENOMEM; + memset(&template, 0, sizeof(template)); + template.reg = SND_SOC_NOPM; + template.id = snd_soc_dapm_dai_link; + template.name = link_name; + template.event = snd_soc_dai_link_event; + template.event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU | + SND_SOC_DAPM_PRE_PMD; + template.kcontrol_news = NULL; + + /* allocate memory for control, only in case of multiple configs */ + if (num_params > 1) { + template.num_kcontrols = 1; + template.kcontrol_news = + snd_soc_dapm_alloc_kcontrol(card, + link_name, params, num_params); + if (!template.kcontrol_news) { + ret = -ENOMEM; + goto outfree_link_name; + } + } dev_dbg(card->dev, "ASoC: adding %s widget\n", link_name); w = snd_soc_dapm_new_control_unlocked(&card->dapm, &template); @@ -3899,15 +3924,8 @@ int snd_soc_dapm_new_pcm(struct snd_soc_card *card, devm_kfree(card->dev, w); outfree_kcontrol_news: devm_kfree(card->dev, (void *)template.kcontrol_news); -outfree_private_value: - devm_kfree(card->dev, (void *)private_value); outfree_link_name: devm_kfree(card->dev, link_name); - for (count = 0 ; count < num_params; count++) - devm_kfree(card->dev, (void *)w_param_text[count]); -outfree_w_param: - devm_kfree(card->dev, w_param_text); - return ret; } -- 2.5.4 (Apple Git-61)

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[alsa-devel] Resend: [patch 0/4] ASoC: dapm: Avoid creating kcontrol for params
by anish kumar 15 Sep '17

15 Sep '17

Currently in codec to codec dai link if there are multiple params defined then dapm can use created kcontrol to decide which param to apply at runtime. However, in case there is only single param configuration then there is no point in creating the kcontrol and also there is no point in allocating memory for kcontrol. In the snd_soc_dapm_new_pcm function, there is memory allocation happening for kcontrol which is later used or not used based on num_param. It is better to not allocate memory when there is only a single configuration. This change is to remedy that anomaly. This code change doesn't do much except just refactor snd_soc_dapm_new_pcm function. Now the part of allocating memory for kcontrol is allocated to a new function and it is called only in the case it is needed. anish kumar (4): ASoC: dapm: fix error path in snd_soc_dapm_new_pcm ASoC: dapm: Refactor the code in snd_soc_dapm_new_pcm ASoC: dapm: Avoid creating kcontrol for params ASoC: dapm: Remove the redundant check sound/soc/soc-dapm.c | 123 ++++++++++++++++++++++++++++----------------------- 1 file changed, 67 insertions(+), 56 deletions(-) -- 2.10.1 (Apple Git-78)

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[alsa-devel] [PATCH 1/3] ASoC: max98927: Added support for DSP_A and DSP_B format
by Ryan Lee 15 Sep '17

15 Sep '17

Signed-off-by: Ryan Lee <ryans.lee(a)maximintegrated.com> --- Changelog: Added support for DSP_A and DSP_B format Added 'max98927_dai_tdm_slot' function to set proper slot configuration. Moved max98927->iface out of switch statement to avoid line duplication. Added variable 'tdm_mode' to avoid BCLK overwrite after 'max98927_dai_tdm_slot' configure BCLK value. sound/soc/codecs/max98927.c | 155 +++++++++++++++++++++++++++++++++++--------- sound/soc/codecs/max98927.h | 7 +- 2 files changed, 129 insertions(+), 33 deletions(-) diff --git a/sound/soc/codecs/max98927.c b/sound/soc/codecs/max98927.c index d9dbbe7..a1d3935 100644 --- a/sound/soc/codecs/max98927.c +++ b/sound/soc/codecs/max98927.c @@ -1,7 +1,7 @@ /* * max98927.c -- MAX98927 ALSA Soc Audio driver * - * Copyright (C) 2016 Maxim Integrated Products + * Copyright (C) 2016-2017 Maxim Integrated Products * Author: Ryan Lee <ryans.lee(a)maximintegrated.com> * * This program is free software; you can redistribute it and/or modify it @@ -146,6 +146,7 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); unsigned int mode = 0; unsigned int format = 0; + bool use_pdm = false; unsigned int invert = 0; dev_dbg(codec->dev, "%s: fmt 0x%08X\n", __func__, fmt); @@ -187,22 +188,27 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) /* interface format */ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { case SND_SOC_DAIFMT_I2S: - max98927->iface |= SND_SOC_DAIFMT_I2S; format = MAX98927_PCM_FORMAT_I2S; break; case SND_SOC_DAIFMT_LEFT_J: - max98927->iface |= SND_SOC_DAIFMT_LEFT_J; format = MAX98927_PCM_FORMAT_LJ; break; + case SND_SOC_DAIFMT_DSP_A: + format = MAX98927_PCM_FORMAT_TDM_MODE1; + break; + case SND_SOC_DAIFMT_DSP_B: + format = MAX98927_PCM_FORMAT_TDM_MODE0; + break; case SND_SOC_DAIFMT_PDM: - max98927->iface |= SND_SOC_DAIFMT_PDM; + use_pdm = true; break; default: return -EINVAL; } + max98927->iface = fmt & SND_SOC_DAIFMT_FORMAT_MASK; - /* pcm channel configuration */ - if (max98927->iface & (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_LEFT_J)) { + if (!use_pdm) { + /* pcm channel configuration */ regmap_update_bits(max98927->regmap, MAX98927_R0018_PCM_RX_EN_A, MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, @@ -217,13 +223,11 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) MAX98927_R003B_SPK_SRC_SEL, MAX98927_SPK_SRC_MASK, 0); - } else regmap_update_bits(max98927->regmap, - MAX98927_R0018_PCM_RX_EN_A, - MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, 0); - - /* pdm channel configuration */ - if (max98927->iface & SND_SOC_DAIFMT_PDM) { + MAX98927_R0035_PDM_RX_CTRL, + MAX98927_PDM_RX_EN_MASK, 0); + } else { + /* pdm channel configuration */ regmap_update_bits(max98927->regmap, MAX98927_R0035_PDM_RX_CTRL, MAX98927_PDM_RX_EN_MASK, 1); @@ -231,10 +235,11 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) regmap_update_bits(max98927->regmap, MAX98927_R003B_SPK_SRC_SEL, MAX98927_SPK_SRC_MASK, 3); - } else + regmap_update_bits(max98927->regmap, - MAX98927_R0035_PDM_RX_CTRL, - MAX98927_PDM_RX_EN_MASK, 0); + MAX98927_R0018_PCM_RX_EN_A, + MAX98927_PCM_RX_CH0_EN | MAX98927_PCM_RX_CH1_EN, 0); + } return 0; } @@ -245,6 +250,21 @@ static int max98927_dai_set_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) 13000000, 19200000, }; +/* BCLKs per LRCLK */ +static const int bclk_sel_table[] = { + 32, 48, 64, 96, 128, 192, 256, 384, 512, +}; + +static int max98927_get_bclk_sel(int bclk) +{ + int i; + /* match BCLKs per LRCLK */ + for (i = 0; i < ARRAY_SIZE(bclk_sel_table); i++) { + if (bclk_sel_table[i] == bclk) + return i + 2; + } + return 0; +} static int max98927_set_clock(struct max98927_priv *max98927, struct snd_pcm_hw_params *params) { @@ -270,23 +290,20 @@ static int max98927_set_clock(struct max98927_priv *max98927, i << MAX98927_PCM_MASTER_MODE_MCLK_RATE_SHIFT); } - switch (blr_clk_ratio) { - case 32: - value = 2; - break; - case 48: - value = 3; - break; - case 64: - value = 4; - break; - default: - return -EINVAL; + if (!max98927->tdm_mode) { + /* BCLK configuration */ + value = max98927_get_bclk_sel(blr_clk_ratio); + if (!value) { + dev_err(codec->dev, "format unsupported %d\n", + params_format(params)); + return -EINVAL; + } + + regmap_update_bits(max98927->regmap, + MAX98927_R0022_PCM_CLK_SETUP, + MAX98927_PCM_CLK_SETUP_BSEL_MASK, + value); } - regmap_update_bits(max98927->regmap, - MAX98927_R0022_PCM_CLK_SETUP, - MAX98927_PCM_CLK_SETUP_BSEL_MASK, - value); return 0; } @@ -386,6 +403,78 @@ static int max98927_dai_hw_params(struct snd_pcm_substream *substream, return -EINVAL; } +static int max98927_dai_tdm_slot(struct snd_soc_dai *dai, + unsigned int tx_mask, unsigned int rx_mask, + int slots, int slot_width) +{ + struct snd_soc_codec *codec = dai->codec; + struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); + int bsel = 0; + unsigned int chan_sz = 0; + + max98927->tdm_mode = true; + + /* BCLK configuration */ + bsel = max98927_get_bclk_sel(slots * slot_width); + if (bsel == 0) { + dev_err(codec->dev, "BCLK %d not supported\n", + slots * slot_width); + return -EINVAL; + } + + regmap_update_bits(max98927->regmap, + MAX98927_R0022_PCM_CLK_SETUP, + MAX98927_PCM_CLK_SETUP_BSEL_MASK, + bsel); + + /* Channel size configuration */ + switch (slot_width) { + case 16: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_16; + break; + case 24: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_24; + break; + case 32: + chan_sz = MAX98927_PCM_MODE_CFG_CHANSZ_32; + break; + default: + dev_err(codec->dev, "format unsupported %d\n", + slot_width); + return -EINVAL; + } + + regmap_update_bits(max98927->regmap, + MAX98927_R0020_PCM_MODE_CFG, + MAX98927_PCM_MODE_CFG_CHANSZ_MASK, chan_sz); + + /* Rx slot configuration */ + regmap_write(max98927->regmap, + MAX98927_R0018_PCM_RX_EN_A, + rx_mask & 0xFF); + regmap_write(max98927->regmap, + MAX98927_R0019_PCM_RX_EN_B, + (rx_mask & 0xFF00) >> 8); + + /* Tx slot configuration */ + regmap_write(max98927->regmap, + MAX98927_R001A_PCM_TX_EN_A, + tx_mask & 0xFF); + regmap_write(max98927->regmap, + MAX98927_R001B_PCM_TX_EN_B, + (tx_mask & 0xFF00) >> 8); + + /* Tx slot Hi-Z configuration */ + regmap_write(max98927->regmap, + MAX98927_R001C_PCM_TX_HIZ_CTRL_A, + ~tx_mask & 0xFF); + regmap_write(max98927->regmap, + MAX98927_R001D_PCM_TX_HIZ_CTRL_B, + (~tx_mask & 0xFF00) >> 8); + + return 0; +} + #define MAX98927_RATES SNDRV_PCM_RATE_8000_48000 #define MAX98927_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \ @@ -405,6 +494,7 @@ static int max98927_dai_set_sysclk(struct snd_soc_dai *dai, .set_sysclk = max98927_dai_set_sysclk, .set_fmt = max98927_dai_set_fmt, .hw_params = max98927_dai_hw_params, + .set_tdm_slot = max98927_dai_tdm_slot, }; static int max98927_dac_event(struct snd_soc_dapm_widget *w, @@ -414,6 +504,9 @@ static int max98927_dac_event(struct snd_soc_dapm_widget *w, struct max98927_priv *max98927 = snd_soc_codec_get_drvdata(codec); switch (event) { + case SND_SOC_DAPM_PRE_PMU: + max98927->tdm_mode = 0; + break; case SND_SOC_DAPM_POST_PMU: regmap_update_bits(max98927->regmap, MAX98927_R003A_AMP_EN, diff --git a/sound/soc/codecs/max98927.h b/sound/soc/codecs/max98927.h index ece6a60..9ea8397 100644 --- a/sound/soc/codecs/max98927.h +++ b/sound/soc/codecs/max98927.h @@ -1,7 +1,7 @@ /* * max98927.h -- MAX98927 ALSA Soc Audio driver * - * Copyright 2013-15 Maxim Integrated Products + * Copyright (C) 2016-2017 Maxim Integrated Products * Author: Ryan Lee <ryans.lee(a)maximintegrated.com> * * This program is free software; you can redistribute it and/or modify it @@ -161,7 +161,9 @@ #define MAX98927_PCM_MODE_CFG_FORMAT_SHIFT (3) #define MAX98927_PCM_FORMAT_I2S (0x0 << 0) #define MAX98927_PCM_FORMAT_LJ (0x1 << 0) - +#define MAX98927_PCM_FORMAT_TDM_MODE0 (0x3 << 0) +#define MAX98927_PCM_FORMAT_TDM_MODE1 (0x4 << 0) +#define MAX98927_PCM_FORMAT_TDM_MODE2 (0x5 << 0) #define MAX98927_PCM_MODE_CFG_CHANSZ_MASK (0x3 << 6) #define MAX98927_PCM_MODE_CFG_CHANSZ_16 (0x1 << 6) #define MAX98927_PCM_MODE_CFG_CHANSZ_24 (0x2 << 6) @@ -268,5 +270,6 @@ struct max98927_priv { unsigned int iface; unsigned int master; unsigned int digital_gain; + bool tdm_mode; }; #endif -- 1.9.1

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[alsa-devel] [PATCH 2/2] ASoC: topology: Fix a potential memory leak in 'soc_tplg_dapm_widget_denum_create()'
by Christophe JAILLET 14 Sep '17

14 Sep '17

If this sanity check fails, we must free the memory that has already been allocated. So we must go to 'err' as in the other error handling parth of this function. Fixes: 1a7dd6e2f192 ("ASoC: topology: Allow a widget to have multiple enum controls") Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/soc-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 2499b54a57e8..01a50413c66f 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1301,7 +1301,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( /* validate kcontrol */ if (strnlen(ec->hdr.name, SNDRV_CTL_ELEM_ID_NAME_MAXLEN) == SNDRV_CTL_ELEM_ID_NAME_MAXLEN) - return NULL; + goto err; se = kzalloc(sizeof(*se), GFP_KERNEL); if (se == NULL) -- 2.11.0

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[alsa-devel] [PATCH 1/2] ASoC: topology: Fix a potential NULL pointer dereference in 'soc_tplg_dapm_widget_denum_create()'
by Christophe JAILLET 14 Sep '17

14 Sep '17

if 'se = kzalloc()' fails in the 'for' loop, we will branch to 'err'. But in this case, 'kc[i].private_value' will still be NULL. A NULL pointer dereference will then occur is the error handling path. In such a case, it is safe to just 'continue' in order to skip this entry and free the other ones. Fixes: 1a7dd6e2f192 ("ASoC: topology: Allow a widget to have multiple enum controls") Signed-off-by: Christophe JAILLET <christophe.jaillet(a)wanadoo.fr> --- sound/soc/soc-topology.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index dd471d2c0266..2499b54a57e8 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1378,6 +1378,9 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( for (; i >= 0; i--) { /* free values and texts */ se = (struct soc_enum *)kc[i].private_value; + if (!se) + continue; + kfree(se->dobj.control.dvalues); for (j = 0; j < ec->items; j++) kfree(se->dobj.control.dtexts[j]); -- 2.11.0

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[alsa-devel] [PATCH] ASoC: Intel: Skylake: Decrease loglevel for topology loading
by Chintan Patel 14 Sep '17

14 Sep '17

In case of failure in loading customize topology firmware, dfw_sst.bin gets loaded. However, current log provides this message as error even after successfully falling back to default topology "dfw_sst.bin". Hence to convey proper message, changing log level and message. Signed-off-by: Chintan Patel <chintan.m.patel(a)intel.com> --- sound/soc/intel/skylake/skl-topology.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 22f768ca3c73..14dafa9f5b35 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -3404,7 +3404,7 @@ int skl_tplg_init(struct snd_soc_platform *platform, struct hdac_ext_bus *ebus) ret = request_firmware(&fw, skl->tplg_name, bus->dev); if (ret < 0) { - dev_err(bus->dev, "tplg fw %s load failed with %d\n", + dev_warn(bus->dev, "tplg fw %s load failed with %d, falling back to dfw_sst.bin", skl->tplg_name, ret); ret = request_firmware(&fw, "dfw_sst.bin", bus->dev); if (ret < 0) { -- 1.9.1

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