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Re: [alsa-devel] [PATCH] pcm_native: Remove VLA usage
by Kees Cook 29 Mar '18

29 Mar '18

On Wed, Mar 28, 2018 at 3:11 PM, Kyle Spiers <kyle(a)spiers.me> wrote: > As part of the effort to remove VLAs from the kernel[1], this changes > the allocation of the rstamps array from being on the stack to being > kcalloc()ed. This also allows for the removal of the explicit zeroing > loop. > > [1] https://lkml.org/lkml/2018/3/7/621 > > Signed-off-by: Kyle Spiers <kyle(a)spiers.me> > --- > sound/core/pcm_native.c | 8 +++++--- > 1 file changed, 5 insertions(+), 3 deletions(-) > > diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c > index 77ba50d..57240b8 100644 > --- a/sound/core/pcm_native.c > +++ b/sound/core/pcm_native.c > @@ -323,7 +323,7 @@ static int constrain_params_by_rules(struct snd_pcm_substream *substream, > struct snd_pcm_hw_constraints *constrs = > &substream->runtime->hw_constraints; > unsigned int k; > - unsigned int rstamps[constrs->rules_num]; > + unsigned int *rstamps; > unsigned int vstamps[SNDRV_PCM_HW_PARAM_LAST_INTERVAL + 1]; > unsigned int stamp; > struct snd_pcm_hw_rule *r; > @@ -339,8 +339,8 @@ static int constrain_params_by_rules(struct snd_pcm_substream *substream, > * Each member of 'rstamps' array represents the sequence number of > * recent application of corresponding rule. > */ > - for (k = 0; k < constrs->rules_num; k++) > - rstamps[k] = 0; > + > + rstamps = kcalloc(constrs->rules_num, sizeof(*rstamps), GFP_KERNEL); > > /* > * Each member of 'vstamps' array represents the sequence number of > @@ -399,6 +399,7 @@ static int constrain_params_by_rules(struct snd_pcm_substream *substream, > > changed = r->func(params, r); > if (changed < 0) > + kfree(rstamps); > return changed; Whoops, this needs { } s: if (changed < 0) { kfree(rstamps); return changed; } Otherwise, looks good! Once that's fixed: Reviewed-by: Kees Cook <keescook(a)chromium.org> -Kees > > /* > @@ -430,6 +431,7 @@ static int constrain_params_by_rules(struct snd_pcm_substream *substream, > if (again) > goto retry; > > + kfree(rstamps); > return 0; > } > > -- > 2.7.4 > -- Kees Cook Pixel Security

1 0

[alsa-devel] [PATCH][next] ASoC: topology: fix memory leak on se when kstrdup fails
by Colin King 28 Mar '18

28 Mar '18

From: Colin Ian King <colin.king(a)canonical.com> In the case when kstrdup fails to allocation memory for kc[i].name the error exit path does not free up 'se' leading to a small leak. Fix this by kfree'ing 'se' before jumping to err_se and since kc[i].private_value is zero we end up avoiding kfree'ing the rest of the unallocated objects in se. Detected by CoverityScan, CID#1466991 ("Reseource leak") Fixes: 267e2c6fd7ca ("ASoC: topology: Fix kcontrol name string handling") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/soc-topology.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index fa27d0fca6dc..71b20d795b51 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1325,8 +1325,10 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( ec->hdr.name); kc[i].name = kstrdup(ec->hdr.name, GFP_KERNEL); - if (kc[i].name == NULL) + if (kc[i].name == NULL) { + kfree(se); goto err_se; + } kc[i].private_value = (long)se; kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; kc[i].access = ec->hdr.access; -- 2.15.1

1 0

[alsa-devel] [PATCH 2/3 v3] ASoC: TSCS42xx: Cleanup private data members
by Steven Eckhoff 28 Mar '18

28 Mar '18

Remove blrcm from private data Remove dev from private data Signed-off-by: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> --- sound/soc/codecs/tscs42xx.c | 16 ++++++---------- 1 file changed, 6 insertions(+), 10 deletions(-) diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c index 37636d6efe60..d41852a37ac6 100644 --- a/sound/soc/codecs/tscs42xx.c +++ b/sound/soc/codecs/tscs42xx.c @@ -31,7 +31,6 @@ struct tscs42xx { int bclk_ratio; int samplerate; - unsigned int blrcm; struct mutex audio_params_lock; u8 coeff_ram[COEFF_RAM_SIZE]; @@ -41,8 +40,6 @@ struct tscs42xx { struct mutex pll_lock; struct regmap *regmap; - - struct device *dev; }; struct coeff_ram_ctl { @@ -1404,12 +1401,11 @@ static int tscs42xx_i2c_probe(struct i2c_client *i2c, return ret; } i2c_set_clientdata(i2c, tscs42xx); - tscs42xx->dev = &i2c->dev; tscs42xx->regmap = devm_regmap_init_i2c(i2c, &tscs42xx_regmap); if (IS_ERR(tscs42xx->regmap)) { ret = PTR_ERR(tscs42xx->regmap); - dev_err(tscs42xx->dev, "Failed to allocate regmap (%d)\n", ret); + dev_err(&i2c->dev, "Failed to allocate regmap (%d)\n", ret); return ret; } @@ -1417,21 +1413,21 @@ static int tscs42xx_i2c_probe(struct i2c_client *i2c, ret = part_is_valid(tscs42xx); if (ret <= 0) { - dev_err(tscs42xx->dev, "No valid part (%d)\n", ret); + dev_err(&i2c->dev, "No valid part (%d)\n", ret); ret = -ENODEV; return ret; } ret = regmap_write(tscs42xx->regmap, R_RESET, RV_RESET_ENABLE); if (ret < 0) { - dev_err(tscs42xx->dev, "Failed to reset device (%d)\n", ret); + dev_err(&i2c->dev, "Failed to reset device (%d)\n", ret); return ret; } ret = regmap_register_patch(tscs42xx->regmap, tscs42xx_patch, ARRAY_SIZE(tscs42xx_patch)); if (ret < 0) { - dev_err(tscs42xx->dev, "Failed to apply patch (%d)\n", ret); + dev_err(&i2c->dev, "Failed to apply patch (%d)\n", ret); return ret; } @@ -1439,10 +1435,10 @@ static int tscs42xx_i2c_probe(struct i2c_client *i2c, mutex_init(&tscs42xx->coeff_ram_lock); mutex_init(&tscs42xx->pll_lock); - ret = devm_snd_soc_register_component(tscs42xx->dev, + ret = devm_snd_soc_register_component(&i2c->dev, &soc_codec_dev_tscs42xx, &tscs42xx_dai, 1); if (ret) { - dev_err(tscs42xx->dev, "Failed to register codec (%d)\n", ret); + dev_err(&i2c->dev, "Failed to register codec (%d)\n", ret); return ret; } -- 2.15.1

1 0

[alsa-devel] [PATCH v2] ASoC: topology: Fix kcontrol name string handling.
by Liam Girdwood 28 Mar '18

28 Mar '18

Fix the topology kcontrol string handling so that string pointer references are strdup()ed instead of being copied. This fixes issues with kcontrol templates on the stack or ones that are freed. Remember and free the strings too when topology is unloaded. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/soc-topology.c | 23 ++++++++++++++++++----- 1 file changed, 18 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 87f3d9568180..d3ce9aa972f6 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -523,6 +523,7 @@ static void remove_widget(struct snd_soc_component *comp, kfree(se->dobj.control.dtexts[j]); kfree(se); + kfree(w->kcontrol_news[i].name); } kfree(w->kcontrol_news); } else { @@ -540,6 +541,7 @@ static void remove_widget(struct snd_soc_component *comp, */ kfree((void *)kcontrol->private_value); snd_ctl_remove(card, kcontrol); + kfree(w->kcontrol_news[i].name); } kfree(w->kcontrol_news); } @@ -1233,7 +1235,9 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create( dev_dbg(tplg->dev, " adding DAPM widget mixer control %s at %d\n", mc->hdr.name, i); - kc[i].name = mc->hdr.name; + kc[i].name = kstrdup(mc->hdr.name, GFP_KERNEL); + if (kc[i].name == NULL) + goto err_str; kc[i].private_value = (long)sm; kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; kc[i].access = mc->hdr.access; @@ -1278,8 +1282,10 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create( err_str: kfree(sm); err: - for (--i; i >= 0; i--) + for (--i; i >= 0; i--) { kfree((void *)kc[i].private_value); + kfree(kc[i].name); + } kfree(kc); return NULL; } @@ -1310,7 +1316,9 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( dev_dbg(tplg->dev, " adding DAPM widget enum control %s\n", ec->hdr.name); - kc[i].name = ec->hdr.name; + kc[i].name = kstrdup(ec->hdr.name, GFP_KERNEL); + if (kc[i].name == NULL) + goto err_se; kc[i].private_value = (long)se; kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; kc[i].access = ec->hdr.access; @@ -1386,6 +1394,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( kfree(se->dobj.control.dtexts[j]); kfree(se); + kfree(kc[i].name); } err: kfree(kc); @@ -1424,7 +1433,9 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dbytes_create( "ASoC: adding bytes kcontrol %s with access 0x%x\n", be->hdr.name, be->hdr.access); - kc[i].name = be->hdr.name; + kc[i].name = kstrdup(be->hdr.name, GFP_KERNEL); + if (kc[i].name == NULL) + goto err; kc[i].private_value = (long)sbe; kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; kc[i].access = be->hdr.access; @@ -1454,8 +1465,10 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dbytes_create( return kc; err: - for (--i; i >= 0; i--) + for (--i; i >= 0; i--) { kfree((void *)kc[i].private_value); + kfree(kc[i].name); + } kfree(kc); return NULL; -- 2.14.1

2 1

[alsa-devel] [PATCH] ASoC: amd: don't use codec anymore
by Kuninori Morimoto 28 Mar '18

28 Mar '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit c88d31153356 ("ASoC: amd: Enable da7219 master clock using common clock framework") is using rtd->codec, but codec is replaced to component. Let's use component Fixes: c88d31153356 ("ASoC: amd: Enable da7219 master clock using common clock framework") Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/amd/acp-da7219-max98357a.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 99c6b5c..b205c78 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -49,7 +49,6 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) { int ret; struct snd_soc_card *card = rtd->card; - struct snd_soc_codec *codec = rtd->codec; struct snd_soc_dai *codec_dai = rtd->codec_dai; struct snd_soc_component *component = codec_dai->component; @@ -69,7 +68,7 @@ static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) return ret; } - da7219_dai_clk = clk_get(codec->dev, "da7219-dai-clks"); + da7219_dai_clk = clk_get(component->dev, "da7219-dai-clks"); ret = snd_soc_card_jack_new(card, "Headset Jack", SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: Intel: bytcr_rt5651: don't use codec anymore
by Kuninori Morimoto 28 Mar '18

28 Mar '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> commit aeec6cc08215 ("ASoC: Intel: bytcr_rt5651: Configure PLL1 before using it") is using codec->dev, but codec is replaced to component. Let's use component Fixes: aeec6cc08215 ("ASoC: Intel: bytcr_rt5651: Configure PLL1 before using it") Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/intel/boards/bytcr_rt5651.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/boards/bytcr_rt5651.c b/sound/soc/intel/boards/bytcr_rt5651.c index 53ce313..1b1997f 100644 --- a/sound/soc/intel/boards/bytcr_rt5651.c +++ b/sound/soc/intel/boards/bytcr_rt5651.c @@ -147,14 +147,14 @@ static int byt_rt5651_prepare_and_enable_pll1(struct snd_soc_dai *codec_dai, } ret = snd_soc_dai_set_pll(codec_dai, 0, clk_id, clk_freq, rate * 512); if (ret < 0) { - dev_err(codec_dai->codec->dev, "can't set pll: %d\n", ret); + dev_err(codec_dai->component->dev, "can't set pll: %d\n", ret); return ret; } ret = snd_soc_dai_set_sysclk(codec_dai, RT5651_SCLK_S_PLL1, rate * 512, SND_SOC_CLOCK_IN); if (ret < 0) { - dev_err(codec_dai->codec->dev, "can't set clock %d\n", ret); + dev_err(codec_dai->component->dev, "can't set clock %d\n", ret); return ret; } -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: cpcap: replace codec to component
by Kuninori Morimoto 28 Mar '18

28 Mar '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Now we can replace Codec to Component. Let's do it. Note: xxx_codec_xxx() -> xxx_component_xxx() .idle_bias_off = 0 -> .idle_bias_on = 1 .ignore_pmdown_time = 0 -> .use_pmdown_time = 1 - -> .endianness = 1 - -> .non_legacy_dai_naming = 1 Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/cpcap.c | 152 +++++++++++++++++++++++------------------------ 1 file changed, 73 insertions(+), 79 deletions(-) diff --git a/sound/soc/codecs/cpcap.c b/sound/soc/codecs/cpcap.c index aedb267..d7f05b3 100644 --- a/sound/soc/codecs/cpcap.c +++ b/sound/soc/codecs/cpcap.c @@ -243,7 +243,7 @@ enum cpcap_dai { }; struct cpcap_audio { - struct snd_soc_codec *codec; + struct snd_soc_component *component; struct regmap *regmap; u16 vendor; @@ -256,8 +256,8 @@ struct cpcap_audio { static int cpcap_st_workaround(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event) { - struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm); - struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); int err = 0; /* Only CPCAP from ST requires workaround */ @@ -357,8 +357,8 @@ static SOC_ENUM_SINGLE_EXT_DECL(cpcap_input_right_mux_enum, static int cpcap_output_mux_get_enum(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); - struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = snd_soc_dapm_kcontrol_component(kcontrol); + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; unsigned int shift = e->shift_l; int reg_voice, reg_hifi, reg_ext, status; @@ -400,8 +400,8 @@ static int cpcap_output_mux_get_enum(struct snd_kcontrol *kcontrol, static int cpcap_output_mux_put_enum(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); - struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = snd_soc_dapm_kcontrol_component(kcontrol); + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; @@ -445,8 +445,8 @@ static int cpcap_output_mux_put_enum(struct snd_kcontrol *kcontrol, static int cpcap_input_right_mux_get_enum(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); - struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = snd_soc_dapm_kcontrol_component(kcontrol); + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); int regval, mask; int err; @@ -484,8 +484,8 @@ static int cpcap_input_right_mux_get_enum(struct snd_kcontrol *kcontrol, static int cpcap_input_right_mux_put_enum(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); - struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = snd_soc_dapm_kcontrol_component(kcontrol); + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; @@ -529,8 +529,8 @@ static int cpcap_input_right_mux_put_enum(struct snd_kcontrol *kcontrol, static int cpcap_input_left_mux_get_enum(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); - struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = snd_soc_dapm_kcontrol_component(kcontrol); + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); int regval, mask; int err; @@ -560,8 +560,8 @@ static int cpcap_input_left_mux_get_enum(struct snd_kcontrol *kcontrol, static int cpcap_input_left_mux_put_enum(struct snd_kcontrol *kcontrol, struct snd_ctl_elem_value *ucontrol) { - struct snd_soc_codec *codec = snd_soc_dapm_kcontrol_codec(kcontrol); - struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = snd_soc_dapm_kcontrol_component(kcontrol); + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); struct snd_soc_dapm_context *dapm = snd_soc_dapm_kcontrol_dapm(kcontrol); struct soc_enum *e = (struct soc_enum *)kcontrol->private_value; @@ -969,13 +969,13 @@ static int cpcap_set_sysclk(struct cpcap_audio *cpcap, enum cpcap_dai dai, clkidshift = CPCAP_BIT_CLK_IN_SEL; break; default: - dev_err(cpcap->codec->dev, "invalid DAI: %d", dai); + dev_err(cpcap->component->dev, "invalid DAI: %d", dai); return -EINVAL; } /* setup clk id */ if (clk_id < 0 || clk_id > 1) { - dev_err(cpcap->codec->dev, "invalid clk id %d", clk_id); + dev_err(cpcap->component->dev, "invalid clk id %d", clk_id); return -EINVAL; } err = regmap_update_bits(cpcap->regmap, clkidreg, BIT(clkidshift), @@ -1015,7 +1015,7 @@ static int cpcap_set_sysclk(struct cpcap_audio *cpcap, enum cpcap_dai dai, clkfreqval = 0x06 << clkfreqshift; break; default: - dev_err(cpcap->codec->dev, "unsupported freq %u", freq); + dev_err(cpcap->component->dev, "unsupported freq %u", freq); return -EINVAL; } @@ -1035,7 +1035,7 @@ static int cpcap_set_sysclk(struct cpcap_audio *cpcap, enum cpcap_dai dai, static int cpcap_set_samprate(struct cpcap_audio *cpcap, enum cpcap_dai dai, int samplerate) { - struct snd_soc_codec *codec = cpcap->codec; + struct snd_soc_component *component = cpcap->component; u16 sampreg, sampmask, sampshift, sampval, sampreset; int err, sampreadval; @@ -1053,7 +1053,7 @@ static int cpcap_set_samprate(struct cpcap_audio *cpcap, enum cpcap_dai dai, BIT(CPCAP_BIT_CDC_CLOCK_TREE_RESET); break; default: - dev_err(codec->dev, "invalid DAI: %d", dai); + dev_err(component->dev, "invalid DAI: %d", dai); return -EINVAL; } @@ -1087,7 +1087,7 @@ static int cpcap_set_samprate(struct cpcap_audio *cpcap, enum cpcap_dai dai, sampval = 0x0 << sampshift; break; default: - dev_err(codec->dev, "unsupported samplerate %d", samplerate); + dev_err(component->dev, "unsupported samplerate %d", samplerate); return -EINVAL; } err = regmap_update_bits(cpcap->regmap, sampreg, @@ -1103,7 +1103,7 @@ static int cpcap_set_samprate(struct cpcap_audio *cpcap, enum cpcap_dai dai, return err; if (sampreadval & sampreset) { - dev_err(codec->dev, "reset self-clear failed: %04x", + dev_err(component->dev, "reset self-clear failed: %04x", sampreadval); return -EIO; } @@ -1115,20 +1115,20 @@ static int cpcap_hifi_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_codec *codec = dai->codec; - struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = dai->component; + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); int rate = params_rate(params); - dev_dbg(codec->dev, "HiFi setup HW params: rate=%d", rate); + dev_dbg(component->dev, "HiFi setup HW params: rate=%d", rate); return cpcap_set_samprate(cpcap, CPCAP_DAI_HIFI, rate); } static int cpcap_hifi_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { - struct snd_soc_codec *codec = codec_dai->codec; - struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec); - struct device *dev = codec->dev; + struct snd_soc_component *component = codec_dai->component; + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); + struct device *dev = component->dev; dev_dbg(dev, "HiFi setup sysclk: clk_id=%u, freq=%u", clk_id, freq); return cpcap_set_sysclk(cpcap, CPCAP_DAI_HIFI, clk_id, freq); @@ -1137,9 +1137,9 @@ static int cpcap_hifi_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, static int cpcap_hifi_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { - struct snd_soc_codec *codec = codec_dai->codec; - struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec); - struct device *dev = codec->dev; + struct snd_soc_component *component = codec_dai->component; + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); + struct device *dev = component->dev; static const u16 reg = CPCAP_REG_SDACDI; static const u16 mask = BIT(CPCAP_BIT_SMB_ST_DAC) | @@ -1218,8 +1218,8 @@ static int cpcap_hifi_set_dai_fmt(struct snd_soc_dai *codec_dai, static int cpcap_hifi_set_mute(struct snd_soc_dai *dai, int mute) { - struct snd_soc_codec *codec = dai->codec; - struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = dai->component; + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); static const u16 reg = CPCAP_REG_RXSDOA; static const u16 mask = BIT(CPCAP_BIT_ST_DAC_SW); u16 val; @@ -1229,7 +1229,7 @@ static int cpcap_hifi_set_mute(struct snd_soc_dai *dai, int mute) else val = BIT(CPCAP_BIT_ST_DAC_SW); - dev_dbg(codec->dev, "HiFi mute: %d", mute); + dev_dbg(component->dev, "HiFi mute: %d", mute); return regmap_update_bits(cpcap->regmap, reg, mask, val); } @@ -1244,9 +1244,9 @@ static int cpcap_voice_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { - struct snd_soc_codec *codec = dai->codec; - struct device *dev = codec->dev; - struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = dai->component; + struct device *dev = component->dev; + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); static const u16 reg_cdi = CPCAP_REG_CDI; int rate = params_rate(params); int channels = params_channels(params); @@ -1283,10 +1283,10 @@ static int cpcap_voice_hw_params(struct snd_pcm_substream *substream, static int cpcap_voice_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, unsigned int freq, int dir) { - struct snd_soc_codec *codec = codec_dai->codec; - struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = codec_dai->component; + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); - dev_dbg(codec->dev, "Voice setup sysclk: clk_id=%u, freq=%u", + dev_dbg(component->dev, "Voice setup sysclk: clk_id=%u, freq=%u", clk_id, freq); return cpcap_set_sysclk(cpcap, CPCAP_DAI_VOICE, clk_id, freq); } @@ -1294,8 +1294,8 @@ static int cpcap_voice_set_dai_sysclk(struct snd_soc_dai *codec_dai, int clk_id, static int cpcap_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, unsigned int fmt) { - struct snd_soc_codec *codec = codec_dai->codec; - struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = codec_dai->component; + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); static const u16 mask = BIT(CPCAP_BIT_SMB_CDC) | BIT(CPCAP_BIT_CLK_INV) | BIT(CPCAP_BIT_FS_INV) | @@ -1304,7 +1304,7 @@ static int cpcap_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, u16 val = 0x0000; int err; - dev_dbg(codec->dev, "Voice setup dai format (%08x)", fmt); + dev_dbg(component->dev, "Voice setup dai format (%08x)", fmt); /* * "Voice Playback" and "Voice Capture" should always be @@ -1316,7 +1316,7 @@ static int cpcap_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, val &= ~BIT(CPCAP_BIT_SMB_CDC); break; default: - dev_err(codec->dev, "Voice dai fmt failed: CPCAP should be the master"); + dev_err(component->dev, "Voice dai fmt failed: CPCAP should be the master"); val &= ~BIT(CPCAP_BIT_SMB_CDC); break; } @@ -1339,7 +1339,7 @@ static int cpcap_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, val &= ~BIT(CPCAP_BIT_FS_INV); break; default: - dev_err(codec->dev, "Voice dai fmt failed: unsupported clock invert mode"); + dev_err(component->dev, "Voice dai fmt failed: unsupported clock invert mode"); break; } @@ -1361,7 +1361,7 @@ static int cpcap_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, break; } - dev_dbg(codec->dev, "Voice dai format: val=%04x", val); + dev_dbg(component->dev, "Voice dai format: val=%04x", val); err = regmap_update_bits(cpcap->regmap, CPCAP_REG_CDI, mask, val); if (err) return err; @@ -1372,8 +1372,8 @@ static int cpcap_voice_set_dai_fmt(struct snd_soc_dai *codec_dai, static int cpcap_voice_set_mute(struct snd_soc_dai *dai, int mute) { - struct snd_soc_codec *codec = dai->codec; - struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec); + struct snd_soc_component *component = dai->component; + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); static const u16 reg = CPCAP_REG_RXCOA; static const u16 mask = BIT(CPCAP_BIT_CDC_SW); u16 val; @@ -1383,7 +1383,7 @@ static int cpcap_voice_set_mute(struct snd_soc_dai *dai, int mute) else val = BIT(CPCAP_BIT_CDC_SW); - dev_dbg(codec->dev, "Voice mute: %d", mute); + dev_dbg(component->dev, "Voice mute: %d", mute); return regmap_update_bits(cpcap->regmap, reg, mask, val); }; @@ -1460,13 +1460,13 @@ static int cpcap_dai_mux(struct cpcap_audio *cpcap, bool swap_dai_configuration) return 0; } -static int cpcap_audio_reset(struct snd_soc_codec *codec, +static int cpcap_audio_reset(struct snd_soc_component *component, bool swap_dai_configuration) { - struct cpcap_audio *cpcap = snd_soc_codec_get_drvdata(codec); + struct cpcap_audio *cpcap = snd_soc_component_get_drvdata(component); int i, err = 0; - dev_dbg(codec->dev, "init audio codec"); + dev_dbg(component->dev, "init audio codec"); for (i = 0; i < ARRAY_SIZE(cpcap_default_regs); i++) { err = regmap_update_bits(cpcap->regmap, @@ -1500,40 +1500,41 @@ static int cpcap_audio_reset(struct snd_soc_codec *codec, return 0; } -static int cpcap_soc_probe(struct snd_soc_codec *codec) +static int cpcap_soc_probe(struct snd_soc_component *component) { struct cpcap_audio *cpcap; int err; - cpcap = devm_kzalloc(codec->dev, sizeof(*cpcap), GFP_KERNEL); + cpcap = devm_kzalloc(component->dev, sizeof(*cpcap), GFP_KERNEL); if (!cpcap) return -ENOMEM; - snd_soc_codec_set_drvdata(codec, cpcap); - cpcap->codec = codec; + snd_soc_component_set_drvdata(component, cpcap); + cpcap->component = component; - cpcap->regmap = dev_get_regmap(codec->dev->parent, NULL); + cpcap->regmap = dev_get_regmap(component->dev->parent, NULL); if (!cpcap->regmap) return -ENODEV; - snd_soc_codec_init_regmap(codec, cpcap->regmap); + snd_soc_component_init_regmap(component, cpcap->regmap); - err = cpcap_get_vendor(codec->dev, cpcap->regmap, &cpcap->vendor); + err = cpcap_get_vendor(component->dev, cpcap->regmap, &cpcap->vendor); if (err) return err; - return cpcap_audio_reset(codec, false); + return cpcap_audio_reset(component, false); } -static struct snd_soc_codec_driver soc_codec_dev_cpcap = { - .probe = cpcap_soc_probe, - - .component_driver = { - .controls = cpcap_snd_controls, - .num_controls = ARRAY_SIZE(cpcap_snd_controls), - .dapm_widgets = cpcap_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(cpcap_dapm_widgets), - .dapm_routes = intercon, - .num_dapm_routes = ARRAY_SIZE(intercon), - }, +static struct snd_soc_component_driver soc_codec_dev_cpcap = { + .probe = cpcap_soc_probe, + .controls = cpcap_snd_controls, + .num_controls = ARRAY_SIZE(cpcap_snd_controls), + .dapm_widgets = cpcap_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(cpcap_dapm_widgets), + .dapm_routes = intercon, + .num_dapm_routes = ARRAY_SIZE(intercon), + .idle_bias_on = 1, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, }; static int cpcap_codec_probe(struct platform_device *pdev) @@ -1543,19 +1544,12 @@ static int cpcap_codec_probe(struct platform_device *pdev) pdev->dev.of_node = codec_node; - return snd_soc_register_codec(&pdev->dev, &soc_codec_dev_cpcap, + return devm_snd_soc_register_component(&pdev->dev, &soc_codec_dev_cpcap, cpcap_dai, ARRAY_SIZE(cpcap_dai)); } -static int cpcap_codec_remove(struct platform_device *pdev) -{ - snd_soc_unregister_codec(&pdev->dev); - return 0; -} - static struct platform_driver cpcap_codec_driver = { .probe = cpcap_codec_probe, - .remove = cpcap_codec_remove, .driver = { .name = "cpcap-codec", }, -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: mediatek: remove "simple-mfd" in the example
by Ryder Lee 28 Mar '18

28 Mar '18

Remove compatible string "simple-mfd" in the example as we have already added devm_of_platform_populate() in the parent driver. Signed-off-by: Ryder Lee <ryder.lee(a)mediatek.com> --- Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt b/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt index 6df87b9..e2f7f49 100644 --- a/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt +++ b/Documentation/devicetree/bindings/sound/mt2701-afe-pcm.txt @@ -53,7 +53,7 @@ See ../arm/mediatek/mediatek,audsys.txt for details about the parent node. Example: audsys: audio-subsystem@11220000 { - compatible = "mediatek,mt2701-audsys", "syscon", "simple-mfd"; + compatible = "mediatek,mt2701-audsys", "syscon"; ... afe: audio-controller { -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: tscs42xx: Remove owner assignment from i2c_driver
by Fabio Estevam 28 Mar '18

28 Mar '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> Structure i2c_driver does not need to set the owner field, as this will be populated by the driver core. Generated by scripts/coccinelle/api/platform_no_drv_owner.cocci. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- sound/soc/codecs/tscs42xx.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c index cfc71b5..bbfc73a 100644 --- a/sound/soc/codecs/tscs42xx.c +++ b/sound/soc/codecs/tscs42xx.c @@ -1443,7 +1443,6 @@ MODULE_DEVICE_TABLE(of, tscs42xx_of_match); static struct i2c_driver tscs42xx_i2c_driver = { .driver = { .name = "tscs42xx", - .owner = THIS_MODULE, .of_match_table = tscs42xx_of_match, }, .probe = tscs42xx_i2c_probe, -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: Add accessor macros for TLV items
by Ranjani Sridharan 28 Mar '18

28 Mar '18

This patch adds macros for accessing the type, length, min, mute and step items in TLV data. These will be used by drivers to extract the TLV data while loading topology. Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- include/sound/tlv.h | 5 +++++ include/uapi/sound/tlv.h | 7 +++++++ 2 files changed, 12 insertions(+) diff --git a/include/sound/tlv.h b/include/sound/tlv.h index 3677ebb928d5..8fa379e8a568 100644 --- a/include/sound/tlv.h +++ b/include/sound/tlv.h @@ -49,6 +49,11 @@ #define TLV_DB_GAIN_MUTE SNDRV_CTL_TLVD_DB_GAIN_MUTE +#define TLV_ITEM_TYPE SNDRV_CTL_TLVD_ITEM_TYPE +#define TLV_ITEM_LEN SNDRV_CTL_TLVD_ITEM_LEN +#define TLV_DB_MIN SNDRV_CTL_TLVD_DB_SCALE_MIN +#define TLV_DB_MUTE_AND_STEP SNDRV_CTL_TLVD_DB_SCALE_MUTE_AND_STEP + /* * The below assumes that each item TLV is 4 words like DB_SCALE or LINEAR. * This is an old fasion and obsoleted by commit bf1d1c9b6179("ALSA: tlv: add diff --git a/include/uapi/sound/tlv.h b/include/uapi/sound/tlv.h index be5371f09a62..f3fba49ef6e8 100644 --- a/include/uapi/sound/tlv.h +++ b/include/uapi/sound/tlv.h @@ -60,6 +60,13 @@ unsigned int name[] = { \ SNDRV_CTL_TLVD_DB_SCALE_ITEM(min, step, mute) \ } +/* Accessor macros for TLV type and len */ +#define SNDRV_CTL_TLVD_ITEM_TYPE 0 +#define SNDRV_CTL_TLVD_ITEM_LEN 1 + +/* Accessor macros for min, mute and step values from dB scale type TLV */ +#define SNDRV_CTL_TLVD_DB_SCALE_MIN 2 +#define SNDRV_CTL_TLVD_DB_SCALE_MUTE_AND_STEP 3 /* dB scale specified with min/max values instead of step */ #define SNDRV_CTL_TLVD_DB_MINMAX_ITEM(min_dB, max_dB) \ -- 2.14.1

3 3

[alsa-devel] [RFC, alsa-utils] RFC: alsamixer: Improve description of playback switches
by Kirill Marinushkin 27 Mar '18

27 Mar '18

Currently, all playback switches are described as mute. It is shown as: [OO] == sound is on (left and right channels) [MM] == sound is muted (left and right channels) But cards can have different kinds of switch controls, not only mute. At least usb-audio provides boolean controls as playback switches for: * clock source validity (read only switch) * processing features on/off (rw switch) * potentially other similar use-cases It becames confusing, because it is shown as: [OO] == clock source is valid [MM] == clock source is invalid [OO] == processing feature is on [MM] == processing feature is off And from the help, it is not clear how to toggle such switches. In this RFC, I suggest to show switches more generic: [yy] == sound on / clock source is valid / processing feature is on [nn] == sound muted / clock source is invalid / processing feature is off And give a more generic description in the help. Signed-off-by: Kirill Marinushkin <k.marinushkin(a)gmail.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: alsa-devel(a)alsa-project.org --- alsamixer/mixer_display.c | 8 ++++---- alsamixer/mixer_widget.c | 4 ++-- 2 files changed, 6 insertions(+), 6 deletions(-) diff --git a/alsamixer/mixer_display.c b/alsamixer/mixer_display.c index b1f79d0..f73b11a 100644 --- a/alsamixer/mixer_display.c +++ b/alsamixer/mixer_display.c @@ -513,13 +513,13 @@ static void display_control(unsigned int control_index) mvwaddch(mixer_widget.window, base_y + 1, frame_left + 1, switches[0] /* TRANSLATORS: playback on; one character */ - ? _("O")[0] | (control->flags & IS_ACTIVE ? attr_ctl_nomute : 0) + ? _("y")[0] | (control->flags & IS_ACTIVE ? attr_ctl_nomute : 0) /* TRANSLATORS: playback muted; one character */ - : _("M")[0] | (control->flags & IS_ACTIVE ? attr_ctl_mute : 0)); + : _("n")[0] | (control->flags & IS_ACTIVE ? attr_ctl_mute : 0)); waddch(mixer_widget.window, switches[1] - ? _("O")[0] | (control->flags & IS_ACTIVE ? attr_ctl_nomute : 0) - : _("M")[0] | (control->flags & IS_ACTIVE ? attr_ctl_mute : 0)); + ? _("y")[0] | (control->flags & IS_ACTIVE ? attr_ctl_nomute : 0) + : _("n")[0] | (control->flags & IS_ACTIVE ? attr_ctl_mute : 0)); } if (control->flags & TYPE_CSWITCH) { diff --git a/alsamixer/mixer_widget.c b/alsamixer/mixer_widget.c index 0c2b9e0..93d1923 100644 --- a/alsamixer/mixer_widget.c +++ b/alsamixer/mixer_widget.c @@ -210,9 +210,9 @@ static void show_help(void) _("Z X C Decrease left/both/right volumes"), _("B Balance left and right volumes"), "", - _("M Toggle mute"), + _("M Toggle mute / playback switch"), /* TRANSLATORS: or , . */ - _("< > Toggle left/right mute"), + _("< > Toggle left/right mute / playback switch"), "", _("Space Toggle capture"), /* TRANSLATORS: or Insert Delete */ -- 2.13.6

4 12

[alsa-devel] [PATCH v2 1/3] ASoC: AMD: Use single dai for da7219 playback and capture
by Akshu Agrawal 27 Mar '18

27 Mar '18

BT I2S is a bi-directional dai, we will use the same cpu dai for playback and capture. V2: reworded the commit message. TEST=aplay -D hw:0,0 -vv <file> arecord -D hw:0,0 -f dat -d 5 -vv <file> Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-da7219-max98357a.c | 14 ++------------ 1 file changed, 2 insertions(+), 12 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index bbe0f10..d07c2a1 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -124,8 +124,8 @@ static int cz_fe_startup(struct snd_pcm_substream *substream) static struct snd_soc_dai_link cz_dai_7219_98357[] = { { - .name = "amd-da7219-play", - .stream_name = "Playback", + .name = "amd-da7219-play-cap", + .stream_name = "Playback and Capture", .platform_name = "acp_audio_dma.0.auto", .cpu_dai_name = "designware-i2s.3.auto", .codec_dai_name = "da7219-hifi", @@ -134,16 +134,6 @@ static int cz_fe_startup(struct snd_pcm_substream *substream) | SND_SOC_DAIFMT_CBM_CFM, .init = cz_da7219_init, .dpcm_playback = 1, - }, - { - .name = "amd-da7219-cap", - .stream_name = "Capture", - .platform_name = "acp_audio_dma.0.auto", - .cpu_dai_name = "designware-i2s.4.auto", - .codec_dai_name = "da7219-hifi", - .codec_name = "i2c-DLGS7219:00", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, .dpcm_capture = 1, .ops = &cz_da7219_cap_ops, }, -- 1.9.1

2 4

[alsa-devel] [PATCH v2] ASoC: topology: Fix bclk and fsync inversion in set_link_hw_format()
by Kirill Marinushkin 27 Mar '18

27 Mar '18

The values of bclk and fsync are inverted WRT the codec. But the existing solution already works for Broadwell, see the alsa-lib config: `alsa-lib/src/conf/topology/broadwell/broadwell.conf` This commit provides the backwards-compatible solution to fix this misuse. This commit goes in pair with the corresponding patch for alsa-lib. Signed-off-by: Kirill Marinushkin <k.marinushkin(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Takashi Iwai <tiwai(a)suse.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Pierre-Louis Bossart <pierre-louis.bossart(a)linux.intel.com> Cc: Pan Xiuli <xiuli.pan(a)linux.intel.com> Cc: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> Cc: linux-kernel(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org --- include/uapi/sound/asoc.h | 16 ++++++++++++++-- sound/soc/soc-topology.c | 12 +++++++----- 2 files changed, 21 insertions(+), 7 deletions(-) diff --git a/include/uapi/sound/asoc.h b/include/uapi/sound/asoc.h index 69c37ecbff7e..f0e5e21efa54 100644 --- a/include/uapi/sound/asoc.h +++ b/include/uapi/sound/asoc.h @@ -160,6 +160,18 @@ #define SND_SOC_TPLG_LNK_FLGBIT_SYMMETRIC_SAMPLEBITS (1 << 2) #define SND_SOC_TPLG_LNK_FLGBIT_VOICE_WAKEUP (1 << 3) +/* DAI topology BCLK parameter + * For the backwards capability, by default codec is bclk master + */ +#define SND_SOC_TPLG_BCLK_CM 0 /* codec is bclk master */ +#define SND_SOC_TPLG_BCLK_CS 1 /* codec is bclk slave */ + +/* DAI topology FSYNC parameter + * For the backwards capability, by default codec is fsync master + */ +#define SND_SOC_TPLG_FSYNC_CM 0 /* codec is fsync master */ +#define SND_SOC_TPLG_FSYNC_CS 1 /* codec is fsync slave */ + /* * Block Header. * This header precedes all object and object arrays below. @@ -315,8 +327,8 @@ struct snd_soc_tplg_hw_config { __u8 clock_gated; /* 1 if clock can be gated to save power */ __u8 invert_bclk; /* 1 for inverted BCLK, 0 for normal */ __u8 invert_fsync; /* 1 for inverted frame clock, 0 for normal */ - __u8 bclk_master; /* 1 for master of BCLK, 0 for slave */ - __u8 fsync_master; /* 1 for master of FSYNC, 0 for slave */ + __u8 bclk_master; /* SND_SOC_TPLG_BCLK_ value */ + __u8 fsync_master; /* SND_SOC_TPLG_FSYNC_ value */ __u8 mclk_direction; /* 0 for input, 1 for output */ __le16 reserved; /* for 32bit alignment */ __le32 mclk_rate; /* MCLK or SYSCLK freqency in Hz */ diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 01a50413c66f..c5bdc673b195 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1994,13 +1994,15 @@ static void set_link_hw_format(struct snd_soc_dai_link *link, link->dai_fmt |= SND_SOC_DAIFMT_IB_IF; /* clock masters */ - bclk_master = hw_config->bclk_master; - fsync_master = hw_config->fsync_master; - if (!bclk_master && !fsync_master) + bclk_master = (hw_config->bclk_master == + SND_SOC_TPLG_BCLK_CM); + fsync_master = (hw_config->fsync_master == + SND_SOC_TPLG_FSYNC_CM); + if (bclk_master && fsync_master) link->dai_fmt |= SND_SOC_DAIFMT_CBM_CFM; - else if (bclk_master && !fsync_master) - link->dai_fmt |= SND_SOC_DAIFMT_CBS_CFM; else if (!bclk_master && fsync_master) + link->dai_fmt |= SND_SOC_DAIFMT_CBS_CFM; + else if (bclk_master && !fsync_master) link->dai_fmt |= SND_SOC_DAIFMT_CBM_CFS; else link->dai_fmt |= SND_SOC_DAIFMT_CBS_CFS; -- 2.13.6

3 13

[alsa-devel] [PATCH 3/3 v2] ASoC: TSCS42xx: Add CCF support to get sysclk
by Steven Eckhoff 27 Mar '18

27 Mar '18

The TSCS42xx relies on set_sysclk to get a unique clock id and rate, which prevents it from being used with the simple-card. Remove set_sysclk callback Add CCF support to get clock id and rate Add clocks and clock-names to device tree binding Signed-off-by: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> --- .../devicetree/bindings/sound/tscs42xx.txt | 6 ++ sound/soc/codecs/tscs42xx.c | 104 ++++++++++++++------- sound/soc/codecs/tscs42xx.h | 2 +- 3 files changed, 75 insertions(+), 37 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/tscs42xx.txt b/Documentation/devicetree/bindings/sound/tscs42xx.txt index 2ac2f0996697..7eea32e9d078 100644 --- a/Documentation/devicetree/bindings/sound/tscs42xx.txt +++ b/Documentation/devicetree/bindings/sound/tscs42xx.txt @@ -8,9 +8,15 @@ Required Properties: - reg : <0x71> for analog mic <0x69> for digital mic + - clock-names: Must one of the following "mclk1", "xtal", "mclk2" + + - clocks: phandle of the clock that provides the codec sysclk + Example: wookie: codec@69 { compatible = "tempo,tscs42A2"; reg = <0x69>; + clock-names = "xtal"; + clocks = <&audio_xtal>; }; diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c index d41852a37ac6..69cd4c1ed1d6 100644 --- a/sound/soc/codecs/tscs42xx.c +++ b/sound/soc/codecs/tscs42xx.c @@ -12,6 +12,7 @@ #include <linux/module.h> #include <linux/delay.h> #include <linux/mutex.h> +#include <linux/clk.h> #include <sound/tlv.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -40,6 +41,9 @@ struct tscs42xx { struct mutex pll_lock; struct regmap *regmap; + + struct clk *sysclk; + int sysclk_src_id; }; struct coeff_ram_ctl { @@ -1251,13 +1255,46 @@ static int tscs42xx_set_dai_bclk_ratio(struct snd_soc_dai *codec_dai, return 0; } -static int tscs42xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, - int clk_id, unsigned int freq, int dir) +static const struct snd_soc_dai_ops tscs42xx_dai_ops = { + .hw_params = tscs42xx_hw_params, + .mute_stream = tscs42xx_mute_stream, + .set_fmt = tscs42xx_set_dai_fmt, + .set_bclk_ratio = tscs42xx_set_dai_bclk_ratio, +}; + +static int part_is_valid(struct tscs42xx *tscs42xx) { - struct snd_soc_component *component = codec_dai->component; + int val; int ret; + unsigned int reg; + + ret = regmap_read(tscs42xx->regmap, R_DEVIDH, &reg); + if (ret < 0) + return ret; - switch (clk_id) { + val = reg << 8; + ret = regmap_read(tscs42xx->regmap, R_DEVIDL, &reg); + if (ret < 0) + return ret; + + val |= reg; + + switch (val) { + case 0x4A74: + case 0x4A73: + return true; + default: + return false; + }; +} + +static int set_sysclk(struct snd_soc_component *component) +{ + struct tscs42xx *tscs42xx = snd_soc_component_get_drvdata(component); + unsigned long freq; + int ret; + + switch (tscs42xx->sysclk_src_id) { case TSCS42XX_PLL_SRC_XTAL: case TSCS42XX_PLL_SRC_MCLK1: ret = snd_soc_component_write(component, R_PLLREFSEL, @@ -1285,6 +1322,7 @@ static int tscs42xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, return -EINVAL; } + freq = clk_get_rate(tscs42xx->sysclk); ret = set_pll_ctl_from_input_freq(component, freq); if (ret < 0) { dev_err(component->dev, @@ -1295,41 +1333,13 @@ static int tscs42xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, return 0; } -static const struct snd_soc_dai_ops tscs42xx_dai_ops = { - .hw_params = tscs42xx_hw_params, - .mute_stream = tscs42xx_mute_stream, - .set_fmt = tscs42xx_set_dai_fmt, - .set_bclk_ratio = tscs42xx_set_dai_bclk_ratio, - .set_sysclk = tscs42xx_set_dai_sysclk, -}; - -static int part_is_valid(struct tscs42xx *tscs42xx) +static int tscs42xx_probe(struct snd_soc_component *component) { - int val; - int ret; - unsigned int reg; - - ret = regmap_read(tscs42xx->regmap, R_DEVIDH, &reg); - if (ret < 0) - return ret; - - val = reg << 8; - ret = regmap_read(tscs42xx->regmap, R_DEVIDL, &reg); - if (ret < 0) - return ret; - - val |= reg; - - switch (val) { - case 0x4A74: - case 0x4A73: - return true; - default: - return false; - }; + return set_sysclk(component); } static const struct snd_soc_component_driver soc_codec_dev_tscs42xx = { + .probe = tscs42xx_probe, .dapm_widgets = tscs42xx_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tscs42xx_dapm_widgets), .dapm_routes = tscs42xx_intercon, @@ -1387,11 +1397,15 @@ static const struct reg_sequence tscs42xx_patch[] = { { R_AIC2, RV_AIC2_BLRCM_DAC_BCLK_LRCLK_SHARED }, }; +static char const * const src_names[TSCS42XX_PLL_SRC_CNT] = { + "xtal", "mclk1", "mclk2"}; + static int tscs42xx_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct tscs42xx *tscs42xx; - int ret = 0; + int src; + int ret; tscs42xx = devm_kzalloc(&i2c->dev, sizeof(*tscs42xx), GFP_KERNEL); if (!tscs42xx) { @@ -1402,6 +1416,24 @@ static int tscs42xx_i2c_probe(struct i2c_client *i2c, } i2c_set_clientdata(i2c, tscs42xx); + for (src = TSCS42XX_PLL_SRC_XTAL; src < TSCS42XX_PLL_SRC_CNT; src++) { + tscs42xx->sysclk = devm_clk_get(&i2c->dev, src_names[src]); + if (!IS_ERR(tscs42xx->sysclk)) { + break; + } else if (PTR_ERR(tscs42xx->sysclk) != -ENOENT) { + ret = PTR_ERR(tscs42xx->sysclk); + dev_err(&i2c->dev, "Failed to get sysclk (%d)\n", ret); + return ret; + } + } + if (src > TSCS42XX_PLL_SRC_CNT) { + ret = -EINVAL; + dev_err(&i2c->dev, "Failed to get a valid clock name (%d)\n", + ret); + return ret; + } + tscs42xx->sysclk_src_id = src; + tscs42xx->regmap = devm_regmap_init_i2c(i2c, &tscs42xx_regmap); if (IS_ERR(tscs42xx->regmap)) { ret = PTR_ERR(tscs42xx->regmap); diff --git a/sound/soc/codecs/tscs42xx.h b/sound/soc/codecs/tscs42xx.h index d4a30bcbf64b..814c8f3c4a68 100644 --- a/sound/soc/codecs/tscs42xx.h +++ b/sound/soc/codecs/tscs42xx.h @@ -7,10 +7,10 @@ #define __WOOKIE_H__ enum { - TSCS42XX_PLL_SRC_NONE, TSCS42XX_PLL_SRC_XTAL, TSCS42XX_PLL_SRC_MCLK1, TSCS42XX_PLL_SRC_MCLK2, + TSCS42XX_PLL_SRC_CNT, }; #define R_HPVOLL 0x0 -- 2.15.1

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[alsa-devel] No audio on Lenovo Miix 320
by Freaky 27 Mar '18

27 Mar '18

Hi there, we're having some issues getting the audio working on a Lenovo Miix 320. On some forums they say it should work after blacklisting snd_hdmi_lpe_audio. Have blacklisted this, and there's one audio card less now, but still no go unfortunately. Currently running with 4.15.10 from Fedora 27. It logs a lot of: "Audio Port: ASoC: no backend DAIs enabled for Audio Port" lines. Any ideas on how to troubleshoot this? Saw similar problems in https://bugzilla.kernel.org/show_bug.cgi?id=96691 but this seems to be for another codec/DAI. Also, their system seems to be a Baytrail Atom, this is a Cherrytrail. Not sure if any of the patches are applicable. dmesg is attached. I can patch, build and test kernels. Coding is another story however :). Any help is appreciated. Kind regards,

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[alsa-devel] [PATCH] ALSA: usb-audio: fix memory leak on cval
by Colin King 27 Mar '18

27 Mar '18

From: Colin Ian King <colin.king(a)canonical.com> With the current exit return path of the ctl_info allocation failure cval is not being freed resulting in a memory leak. Fix this by kfree'ing it on the return. Detected by CoverityScan, CID#1466878 ("Resource Leak") Fixes: 21e9b3e931f7 ("ALSA: usb-audio: fix uac control query argument") Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/usb/mixer.c | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/sound/usb/mixer.c b/sound/usb/mixer.c index 04dab6f65535..301ad61ed426 100644 --- a/sound/usb/mixer.c +++ b/sound/usb/mixer.c @@ -1365,8 +1365,10 @@ static void build_feature_ctl(struct mixer_build *state, void *raw_desc, cval->cmask = ctl_mask; ctl_info = get_feature_control_info(control); - if (!ctl_info) + if (!ctl_info) { + kfree(cval); return; + } if (state->mixer->protocol == UAC_VERSION_1) cval->val_type = ctl_info->type; else /* UAC_VERSION_2 */ -- 2.15.1

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Re: [alsa-devel] [PATCH] [sound] hdac-codec runtime suspended at PM:Suspend.
by Subhransu S. Prusty 27 Mar '18

27 Mar '18

On Thu, Mar 15, 2018 at 09:02:30PM +0530, Anshuman Gupta wrote: > On Thu, Mar 15, 2018 at 04:12:44PM +0530, Subhransu S. Prusty wrote: > > On Wed, Mar 14, 2018 at 09:07:14PM +0530, Anshuman Gupta wrote: > > > On Wed, Mar 14, 2018 at 11:53:58AM +0100, Rafael J. Wysocki wrote: > > > > On Wed, Mar 14, 2018 at 11:38 AM, Anshuman Gupta > > > > <anshuman.gupta(a)intel.com> wrote: > > > > > On Mon, Mar 12, 2018 at 12:26:53PM +0100, Rafael J. Wysocki wrote: > > > > >> On Mon, Mar 12, 2018 at 12:17 PM, Anshuman Gupta > > > > >> <anshuman.gupta(a)intel.com> wrote: > > > > >> > > > > > >> > + if (pm_runtime_status_suspended(dev)) > > > > >> > + return; > > > > >> > > > > >> That, again, is somewhat fragile from the concurrency perspective. > > > > >> > > > > > > > > And here you want to avoid the below if the device is still suspended. > > > Yes, if we do not avoid the code below, complete callback takes about > > > 3 seconds due to snd_hdac_codec_read timed out because hdac controller > > > would be in runtime suspend state. > > > > > > > > Why is the below code located in the ->complete callback anyway? > > > > Shouldn't it be there in the ->resume one? > > > > > > > Yes even i am also having same doubt, why these power down and power up > > > sequences are part of prepare and complete callback. > > > Adding driver author "Subhransu S. Prusty" to provide more inputs on this. > > > > This driver needs a late resume as it receives a jack notification from the > > i915 driver and the skl controller driver resume may not have happened and > > in turn hda controller may not ready. This ensures a synchronization for > > jack event during resume from S3. > Let me give you insight of the issue, this driver blocks the direct complete > of hda controller PCI 00:1f.3, sometimes it takes 280ms to resume hda controller > from S3 and s2idle. So it does not make sense to suspend/resume hda controller > when it is already in runtime suspend. I get it now. I think this patch needs rework to address both the issues. > > > > I think this patch defeats the purpose. > Here in this case PCI driver may kick the direct complete for hda controller > https://elixir.bootlin.com/linux/latest/source/drivers/pci/pci-driver.c#L691 > But hdac hdmi codec driver is blocking it. > So i think it will be ok to keep this codec and hda controller in runtime > suspend while entering S3 or s2idle, both can resume by runtime PM as well, > will it brake any audio functionality? There was some PM and jack detection issues (I don't recollect now) because of which this was added. This was due to the gfx display power and hda controller synchronization. The rework may be required in both hdac_hdmi and skylake drivers. Regards, Subhransu > > > > Regards, > > Subhransu > > > > > > >> > /* Power up afg */ > > > > >> > snd_hdac_codec_read(hdac, hdac->afg, 0, AC_VERB_SET_POWER_STATE, > > > > >> > AC_PWRST_D0); > > > > >> > -- > > > > >> > 2.7.4 > > > > > > -- > > > Thanks, > > > Anshuman > > > > -- > > -- > Thanks, > Anshuman --

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[alsa-devel] [bug report] ALSA: pcm: Return -EBUSY for OSS ioctls changing busy streams
by Dan Carpenter 27 Mar '18

27 Mar '18

Hello Takashi Iwai, The patch 40cab6e88cb0: "ALSA: pcm: Return -EBUSY for OSS ioctls changing busy streams" from Mar 23, 2018, leads to the following static checker warning: sound/core/oss/pcm_oss.c:1741 snd_pcm_oss_set_rate() warn: inconsistent returns 'mutex:&runtime->oss.params_lock'. Locked on: line 1734 Unlocked on: line 1732 line 1741 sound/core/oss/pcm_oss.c 1717 static int snd_pcm_oss_set_rate(struct snd_pcm_oss_file *pcm_oss_file, int rate) 1718 { 1719 int idx; 1720 1721 for (idx = 1; idx >= 0; --idx) { 1722 struct snd_pcm_substream *substream = pcm_oss_file->streams[idx]; 1723 struct snd_pcm_runtime *runtime; 1724 if (substream == NULL) 1725 continue; 1726 runtime = substream->runtime; 1727 if (rate < 1000) 1728 rate = 1000; 1729 else if (rate > 192000) 1730 rate = 192000; 1731 if (mutex_lock_interruptible(&runtime->oss.params_lock)) 1732 return -ERESTARTSYS; 1733 if (atomic_read(&runtime->oss.rw_ref)) 1734 return -EBUSY; ^^^^^^^^^^^^^ Unlock before returning? 1735 if (runtime->oss.rate != rate) { 1736 runtime->oss.params = 1; 1737 runtime->oss.rate = rate; 1738 } 1739 mutex_unlock(&runtime->oss.params_lock); 1740 } 1741 return snd_pcm_oss_get_rate(pcm_oss_file); 1742 } regards, dan carpenter

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[alsa-devel] [PATCH] ALSA: pcm: potential uninitialized return values
by Dan Carpenter 27 Mar '18

27 Mar '18

Smatch complains that "tmp" can be uninitialized if we do a zero size write. Fixes: 02a5d6925cd3 ("ALSA: pcm: Avoid potential races between OSS ioctls and read/write") Signed-off-by: Dan Carpenter <dan.carpenter(a)oracle.com> diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index a9082f219561..40852c595453 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1364,7 +1364,7 @@ static ssize_t snd_pcm_oss_write2(struct snd_pcm_substream *substream, const cha static ssize_t snd_pcm_oss_write1(struct snd_pcm_substream *substream, const char __user *buf, size_t bytes) { size_t xfer = 0; - ssize_t tmp; + ssize_t tmp = 0; struct snd_pcm_runtime *runtime = substream->runtime; if (atomic_read(&substream->mmap_count)) @@ -1474,7 +1474,7 @@ static ssize_t snd_pcm_oss_read2(struct snd_pcm_substream *substream, char *buf, static ssize_t snd_pcm_oss_read1(struct snd_pcm_substream *substream, char __user *buf, size_t bytes) { size_t xfer = 0; - ssize_t tmp; + ssize_t tmp = 0; struct snd_pcm_runtime *runtime = substream->runtime; if (atomic_read(&substream->mmap_count))

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[alsa-devel] Disable DAPM for click avoidance ?
by Ricard Wanderlof 27 Mar '18

27 Mar '18

I'm having an issue with a codec where the normal DAPM procedure of disabling most of the codec when there is no stream ongoing results in large clicks when a playback stream starts and stops, because the output jumps from 0V when the output is disabled to over a volt when the output is enabled. There appears to be a 'slow stop/start' feature in the codec, however, it takes about one second to ramp up and down the output voltage, and that is too long a delay when starting a stream. In our usecase the power consumtion of the codec is negligable compared to the rest of the electronics, so leaving the codec running at all times is perfectly feasible. Is there any way from a machine driver to disable the DAPM switching, i.e. leave parts of the codec running at all times? In this specific case, there are two internal PM controls which enable the output. One of them is controlled by an output mixer element (SND_SOC_DAPM_MIXER_E), and the other one from set_bias_level(). I've had a look in the core code, but it seems that the whole business of changing the bias level when a stream is started is always performed and that there is no way to say "Set the bias level to ON and leave it there". An alternative would be to augment the codec driver with an 'always on' feature, which effectively always enables PM for certain parts of the chip. But it seems like the wrong place to do this type of thing. Another use for this feature is that we have had a lot of problems over the years with buggy I2S interfaces misbehaving when streams are started and stopped, and there would be an advantage to letting the AIF run the whole time after startup. /Ricard -- Ricard Wolf Wanderlöf ricardw(at)axis.com Axis Communications AB, Lund, Sweden www.axis.com Phone +46 46 272 2016 Fax +46 46 13 61 30

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[alsa-devel] [PATCH v2 0/3] ASoC: Add Multi CPU DAI support
by Shreyas NC 27 Mar '18

27 Mar '18

As discussed in [1], ASoC core supports multi codec DAIs on a DAI link. However it does not do so for CPU DAIs. So, add support for multi CPU DAIs on a DAI Link by adding multi CPU DAI in Card instantiation, suspend and resume functions, PCM ops, stream handling functions and DAPM. [1]: https://www.spinics.net/lists/alsa-devel/msg71369.html Changes in v2: - Addressed review comments by Charles - Rebased on latest for-next branch Shreyas NC (3): ASoC: Add initial support for multiple CPU DAIs ASoC: Add Multi CPU DAI support for PCM ops ASoC: Add Multi CPU DAI support in DAPM include/sound/soc.h | 5 + sound/soc/soc-core.c | 283 ++++++++++++++++++++---------- sound/soc/soc-dapm.c | 71 +++++--- sound/soc/soc-pcm.c | 483 +++++++++++++++++++++++++++++++++------------------ 4 files changed, 564 insertions(+), 278 deletions(-) -- 2.7.4

1 3

[alsa-devel] [PATCH] ALSA: aloop: Mark paused device as inactive
by Robert Rosengren 27 Mar '18

27 Mar '18

Show paused ALSA aloop device as inactive, i.e. the control "PCM Slave Active" set as false. Notification sent upon state change. This makes it possible for client capturing from aloop device to know if data is expected. Without it the client expects data even if playback is paused. Signed-off-by: Robert Rosengren <robertr(a)axis.com> --- sound/drivers/aloop.c | 12 +++++++++--- 1 file changed, 9 insertions(+), 3 deletions(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 0333143a1fa7..5404ab11132d 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -291,6 +291,8 @@ static int loopback_trigger(struct snd_pcm_substream *substream, int cmd) cable->pause |= stream; loopback_timer_stop(dpcm); spin_unlock(&cable->lock); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + loopback_active_notify(dpcm); break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: @@ -299,6 +301,8 @@ static int loopback_trigger(struct snd_pcm_substream *substream, int cmd) cable->pause &= ~stream; loopback_timer_start(dpcm); spin_unlock(&cable->lock); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + loopback_active_notify(dpcm); break; default: return -EINVAL; @@ -879,9 +883,11 @@ static int loopback_active_get(struct snd_kcontrol *kcontrol, [kcontrol->id.subdevice][kcontrol->id.device ^ 1]; unsigned int val = 0; - if (cable != NULL) - val = (cable->running & (1 << SNDRV_PCM_STREAM_PLAYBACK)) ? - 1 : 0; + if (cable != NULL) { + unsigned int running = cable->running ^ cable->pause; + + val = (running & (1 << SNDRV_PCM_STREAM_PLAYBACK)) ? 1 : 0; + } ucontrol->value.integer.value[0] = val; return 0; } -- 2.11.0

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[alsa-devel] Question about cpu_dai/codec_dai unbalance
by Kuninori Morimoto 27 Mar '18

27 Mar '18

Hi Mark, ALSA ML Now, I'm investigating cpu_dai/codec_dai, and I noticed there are many unbalance. For example, snd_soc_suspend(), it calls .digital_mute() only for codec_dai, not for cpc_dai, and, calls .suspend() only for cpu_dai, not for codec_dai. It seems there are many unbalance like this on ALSA SoC. I know it is case-by-case, but I think we can have more balance. For example above snd_soc_suspend() case, .digital_mute() is very common for both cpu/codec I think, and, codec can have .suspend() ? But what do you think ? I think it doesn't breaks current system, because each cpu_dai doesn't have .digital_mute(), and codec_dai doesn't have .suspend() for now. Thus, nothing happen if I added it. The reason why I'm asking is that now, I'm thinking to merge cpu_dai/codec_dai into one dai list somehow. This means, we can have non-categorized DAI, this is same as previous codec/platform replacement into component. Then, this kind of DAI unbalance might be problem. Best regards --- Kuninori Morimoto

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[alsa-devel] two cards as one [frederik@ofb.net: [Alsa-user] the mythical "el-cheapo"]
by frederik＠ofb.net 27 Mar '18

27 Mar '18

Dear ALSA devel, I'm forwarding a message I sent to ALSA-user a week ago; it didn't get any replies. It asks some questions about combining stereo devices into a single multi-channel device. Is this the right place to ask? I'll summarize because maybe the original message was a bit verbose: 1. Is this document still correct (i.e. saying that ALSA doesn't do any resampling when you combine two devices in a "multi" configuration): https://github.com/opensrc/alsa/blob/master/lib/md/TwoCardsAsOne.md 2. What is the best way to combine output devices? Pulseaudio? 3. Is there a way to configure an ALSA pcm which routes to a non-default Pulseaudio sink? Thanks, Frederick ----- Forwarded message from frederik(a)ofb.net ----- Date: Mon, 19 Mar 2018 19:10:47 -0700 From: frederik(a)ofb.net To: alsa-user(a)lists.sourceforge.net Subject: [Alsa-user] the mythical "el-cheapo" X-Spam-Status: No, score=-4.9 required=5.0 tests=BAYES_00,DKIM_SIGNED, HEADER_FROM_DIFFERENT_DOMAINS,RCVD_IN_DNSWL_NONE,RP_MATCHES_RCVD, SPF_HELO_PASS,SPF_PASS,T_DKIM_INVALID autolearn=unavailable autolearn_force=no version=3.4.1 X-Spam-Level: User-Agent: Mutt/1.9.4 (2018-02-28) X-My-Message-ID: <20180320021047.GA4304(a)ofb.net>#2 X-My-Tags: inbox alsa-user Dear ALSA user, I am trying to develop some software to use for a custom sound-masking system where I live (to e.g. mask the sound of construction beeping going on outside). It is crucial to have more than two speakers, because if you can tell where the noise is coming from then it becomes distracting. But I would want this to be an inexpensive solution for students and such and so I don't want to depend on a surround sound system being present. So currently I'm using three $7 USB stereo sound adapters, which control six speakers. For various reasons it would be good for me to be able to combine the USB cards into one audio device. One reason is that users might want to play music on the same speakers that they're using for sound-masking. Another is that having one device with one sample rate would simplify the coding of the noise generation, which might be based on some adaptive algorithm that e.g. listens to a microphone signal, and which might be written in a high-level language where threads are inefficient. However, there are a few documents discouraging me from trying to combine devices in ALSA, saying "you will drift out of sync over time" because of differences in the crystals of the various sound cards. https://www.alsa-project.org/main/index.php/Asoundrc#Virtual_multi_channel_… https://github.com/opensrc/alsa/blob/master/lib/md/TwoCardsAsOne.md But they describe a "route/multi" configuration using ALSA which often seems to work OK for me. For example, I can generate six independent pink noise streams with SoX and it doesn't usually sound glitchy: AUDIODEV=noise play -c 6 -r 44100 -n synth pinknoise gain -10 channels 6 Music also seems to play OK. I've experienced glitches with movie audio, which resolve when I pause/unpause, but sometimes these issues seemed to persist even when I routed the movie audio to a single pair of speakers. I never tracked down the problem but I thought it might be a sample rate issue. Pulseaudio advertises the ability to combine devices correctly using module-combine-sink which tracks the actual frequencies of the various sink crystals: https://www.freedesktop.org/wiki/Software/PulseAudio/Documentation/User/Mod… However, the pulseaudio latency is very high and it makes movie audio out of sync with the picture. Furthermore, it seems difficult to use with ALSA because the "pulse" ALSA module doesn't seem to take a parameter which would let me specify which pulse sink or source I am interested in. Thus it seems impossible to use ALSA programs like SoX's play command (above) to output to a non-default Pulseaudio sink. Is this correct? I want to be able to write software which other people can easily install and which can coexist alongside other software they have installed, including audio software. For example I don't want my system to force them to adopt a certain Pulseaudio sink as "default". Even depending on Pulseaudio seems somewhat undesirable. But since I lack familiarity with audio software, I am hoping someone on this list can point me to useful projects or sources of information so that I can solve these problems in a standard way, and produce software that might actually end up being useful to others. Thank you, Frederick Eaton pcm.mix1 { type dmix ipc_key 2852 # must be unique slave { pcm "hw:3" </home/frederik/hw-settings> } } pcm.mix2 { type dmix ipc_key 2853 slave { pcm "hw:1" </home/frederik/hw-settings> } } pcm.mix3 { type dmix ipc_key 2842 # must be unique slave { pcm "hw:0" </home/frederik/hw-settings> } } pcm.multi_mix { type multi; slaves.a.pcm "mix1"; slaves.b.pcm "mix2"; slaves.c.pcm "mix3"; slaves.a.channels 2; slaves.b.channels 2; slaves.c.channels 2; bindings.0.slave a; bindings.0.channel 0; bindings.1.slave a; bindings.1.channel 1; bindings.2.slave b; bindings.2.channel 0; bindings.3.slave b; bindings.3.channel 1; bindings.4.slave c; bindings.4.channel 0; bindings.5.slave c; bindings.5.channel 1; } # pcm.multi_mix { # type multi; # slaves.a.pcm "mix2"; # slaves.b.pcm "mix1"; # slaves.a.channels 2; # slaves.b.channels 2; # bindings.0.slave a; # bindings.0.channel 0; # bindings.1.slave a; # bindings.1.channel 1; # bindings.2.slave b; # bindings.2.channel 0; # bindings.3.slave b; # bindings.3.channel 1; # } pcm.upmix { type route; slave.pcm "multi_mix"; ttable.0.0 1; ttable.1.1 1; ttable.0.2 1; ttable.1.3 1; ttable.0.4 1; ttable.1.5 1; } # pcm.upmix { # type route; # slave.pcm "multi_mix"; # ttable.0.0 1; # ttable.1.1 1; # ttable.0.2 1; # ttable.1.3 1; # } pcm.noise { type softvol ## FHE 01 Mar 2018 # why is plug: needed here? if i put plug:multi_mix in the upmix # device it breaks slave.pcm "plug:multi_mix" control { name "Noise" card 1 } } ctl.noise { type hw card 1 } pcm.music { type softvol # slave.pcm "plug:mix1" slave.pcm "plug:upmix" ## the following gives "invalid argument", why? # slave { # pcm "upmix"; # channels 2; # } control { name "Music" card 1 } } ctl.music { type hw card 1 } pcm.music_no_bass { type plug; slave.pcm { type ladspa # slave.pcm "plughw:0,0"; slave.pcm "plug:music"; path "/usr/lib/ladspa/"; plugins [{ label hpf input { controls [ 1000 ] } }] hint { show on description "Highpass filter for music" } } } # pcm.!default { # type asym # #playback.pcm "music" # playback.pcm "plug:music_no_bass" # #playback.pcm "plug:hw0mix" # capture.pcm "hw:0" # } # FHE 10 Mar 2018 ## we need an "asym" with capture and playback to make pulseaudio happy pcm.!default { type asym playback.pcm { ## from https://wiki.archlinux.org/index.php/Advanced_Linux_Sound_Architecture/Exam… # set the default device according to the environment # variable ALSA_DEFAULT_PCM and default to plug:music_no_bass @func refer name { @func concat strings [ "pcm." { @func getenv vars [ ALSA_DEFAULT_PCM ] default "music_no_bass" } ] } } capture.pcm "hw:0" } # FHE 10 Mar 2018 old version ## FHE 01 Mar 2018 # pcm.!default { # @func refer # name { @func concat # strings [ "pcm." # { @func getenv # vars [ ALSA_DEFAULT_PCM ] # default "music_no_bass" # } # ] # } # } rate 44100 format S16_LE ## FHE 01 Mar 2018 # Eshelman's settings period_time 0 period_size 2048 buffer_size 65536 buffer_time 0 periods 128 ## overrides # https://www.alsa-project.org/main/index.php/FramesPeriods # buffer size must be >= 2*period_size*(channels*bytes) # = 8*period_size # interrupt frequency is period_size * rate # we seem to get errors with period_size <= 256 # these settings seem to increase CPU usage only slightly, e.g. from # 5.3% (for above settings) to 7.3% (for below) with mpv # period_size 512 # buffer_size 4096 # period_size 1024 # buffer_size 8192 # This will cause the device to interrupt 86 times per second #!/bin/zsh #front-left,front-right,rear-left,rear-right,front-center,lfe pacmd load-module module-alsa-sink device=mix1 sink_name=mix1 channels=2 channel_map=front-left,front-right pacmd load-module module-alsa-sink device=mix2 sink_name=mix2 channels=2 channel_map=rear-left,rear-right pacmd load-module module-alsa-sink device=mix3 sink_name=mix3 channels=2 channel_map=front-center,lfe pacmd load-module module-combine-sink sink_name=mix123 slaves=mix1,mix2,mix3 channels=6 channel_map=front-left,front-right,rear-left,rear-right,front-center,lfe ------------------------------------------------------------------------------ Check out the vibrant tech community on one of the world's most engaging tech sites, Slashdot.org! http://sdm.link/slashdot _______________________________________________ Alsa-user mailing list Alsa-user(a)lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/alsa-user ----- End forwarded message -----

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[alsa-devel] [PATCH v2] ASoC: max98357a: add a delay parameter to avoid speaker pop/crack noise
by Ezequiel Garcia 27 Mar '18

27 Mar '18

From: Mac Chiang <mac.chiang(a)intel.com> Currently, speaker pop/crack noise can happen during PCM playback. This is due to the LRCLK being removed while the BCLK being present, which as per the datasheet can cause large DC output voltage. While on some platforms simply controlling the SD_MODE pin is enough to remove the noise, others have a significant delay between the BCLK and the LRCLK activation. Therefore, this commit adds a delay in the codec driver to guarantee clocks are both active by the time SD_MODE is asserted. Link: https://github.com/raspberrypi/linux/issues/2212 Signed-off-by: Mac Chiang <mac.chiang(a)intel.com> Signed-off-by: Stephen Barber <smbarber(a)chromium.org> [Ezequiel: make the delay a module parameter] Signed-off-by: Ezequiel Garcia <ezequiel(a)collabora.com> --- Changes on v2: * Got rid of spinlock around gpio set. * Make the delay a module parameter, and not a device tree property. sound/soc/codecs/max98357a.c | 65 ++++++++++++++++++++++++++++++++++++++------ 1 file changed, 56 insertions(+), 9 deletions(-) diff --git a/sound/soc/codecs/max98357a.c b/sound/soc/codecs/max98357a.c index d469576b5a7b..7d9a0ee71487 100644 --- a/sound/soc/codecs/max98357a.c +++ b/sound/soc/codecs/max98357a.c @@ -27,27 +27,58 @@ #include <sound/soc-dai.h> #include <sound/soc-dapm.h> +/* According to the DAC datasheet, if LRCLK is removed while BCLK is present, + * the DAC output can cause loud pop/crack noises. This parameter specifies + * a delay for the SD_MODE pin assert, required to eliminate the noise. + */ +static unsigned int sdmode_delay; +module_param(sdmode_delay, uint, 0444); +MODULE_PARM_DESC(sdmode_delay, "SD_MODE pin delay."); + +struct max98357a_priv { + struct delayed_work enable_sdmode_work; + struct gpio_desc *sdmode; + int sdmode_enabled; +}; + +static void max98357a_enable_sdmode_work(struct work_struct *work) +{ + struct max98357a_priv *priv = container_of(work, + struct max98357a_priv, enable_sdmode_work.work); + + gpiod_set_value(priv->sdmode, priv->sdmode_enabled); +} + static int max98357a_daiops_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *dai) { - struct gpio_desc *sdmode = snd_soc_dai_get_drvdata(dai); + struct snd_soc_component *component = dai->component; + struct max98357a_priv *priv = snd_soc_component_get_drvdata(component); - if (!sdmode) + if (!priv->sdmode) return 0; switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - gpiod_set_value(sdmode, 1); + priv->sdmode_enabled = 1; + if (sdmode_delay) { + queue_delayed_work(system_power_efficient_wq, + &priv->enable_sdmode_work, + msecs_to_jiffies(sdmode_delay)); + return 0; + } break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - gpiod_set_value(sdmode, 0); + priv->sdmode_enabled = 0; + cancel_delayed_work(&priv->enable_sdmode_work); break; } + gpiod_set_value(priv->sdmode, priv->sdmode_enabled); return 0; } @@ -61,19 +92,35 @@ static const struct snd_soc_dapm_route max98357a_dapm_routes[] = { static int max98357a_component_probe(struct snd_soc_component *component) { - struct gpio_desc *sdmode; + struct max98357a_priv *priv; - sdmode = devm_gpiod_get_optional(component->dev, "sdmode", GPIOD_OUT_LOW); - if (IS_ERR(sdmode)) - return PTR_ERR(sdmode); + priv = kzalloc(sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; - snd_soc_component_set_drvdata(component, sdmode); + priv->sdmode = gpiod_get_optional(component->dev, + "sdmode", GPIOD_OUT_LOW); + if (IS_ERR(priv->sdmode)) + return PTR_ERR(priv->sdmode); + INIT_DELAYED_WORK(&priv->enable_sdmode_work, + max98357a_enable_sdmode_work); + snd_soc_component_set_drvdata(component, priv); return 0; } +static void max98357a_component_remove(struct snd_soc_component *component) +{ + struct max98357a_priv *priv = snd_soc_component_get_drvdata(component); + + cancel_delayed_work_sync(&priv->enable_sdmode_work); + gpiod_put(priv->sdmode); + kfree(priv); +} + static const struct snd_soc_component_driver max98357a_component_driver = { .probe = max98357a_component_probe, + .remove = max98357a_component_remove, .dapm_widgets = max98357a_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(max98357a_dapm_widgets), .dapm_routes = max98357a_dapm_routes, -- 2.16.2

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[alsa-devel] [PATCH v2] ASoC: topology: Fix bugs of freeing soc topology.
by Liam Girdwood 26 Mar '18

26 Mar '18

From: Yan Wang <yan.wang(a)linux.intel.com> In snd_soc_tplg_component_remove(), it should compare index and not dobj->index with SND_SOC_TPLG_INDEX_ALL for removing all topology objects. Signed-off-by: Yan Wang <yan.wang(a)linux.intel.com> Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- Notes: Sign-off added since V1. 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 b363f2409f72..e5ee7a5a2b02 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -2563,7 +2563,7 @@ int snd_soc_tplg_component_remove(struct snd_soc_component *comp, u32 index) /* match index */ if (dobj->index != index && - dobj->index != SND_SOC_TPLG_INDEX_ALL) + index != SND_SOC_TPLG_INDEX_ALL) continue; switch (dobj->type) { -- 2.14.1

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[alsa-devel] [PATCH] ALSA: pcm: Use dma_bytes as size parameter in dma_mmap_coherent()
by Stefan Roese 26 Mar '18

26 Mar '18

When trying to use the driver (e.g. aplay *.wav), the 4MiB DMA buffer will get mmapp'ed in 16KiB chunks. But this fails with the 2nd 16KiB area, as the page offset is outside of the VMA range (size), which is currently used as size parameter in snd_pcm_lib_default_mmap(). By using the DMA buffer size (dma_bytes) instead, the complete DMA buffer can be mmapp'ed and the issue is fixed. This issue was detected on an ARM platform (TI AM57xx) using the RME HDSP MADI PCIe soundcard. Signed-off-by: Stefan Roese <sr(a)denx.de> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Takashi Sakamoto <o-takashi(a)sakamocchi.jp> --- sound/core/pcm_native.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 77ba50ddcf9e..d18b3982548b 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3422,7 +3422,7 @@ int snd_pcm_lib_default_mmap(struct snd_pcm_substream *substream, area, substream->runtime->dma_area, substream->runtime->dma_addr, - area->vm_end - area->vm_start); + substream->runtime->dma_bytes); #endif /* CONFIG_X86 */ /* mmap with fault handler */ area->vm_ops = &snd_pcm_vm_ops_data_fault; -- 2.16.3

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[alsa-devel] [PATCH 1/2] ASoC: mediatek: remove redundant MODULE_ALIAS entries
by sean.wang＠mediatek.com 26 Mar '18

26 Mar '18

From: Sean Wang <sean.wang(a)mediatek.com> MODULE_ALIAS exports information to allow the module to be auto-loaded at boot for the drivers registered using legacy platform registration. However, currently the driver is always used by DT-only platform, MODULE_ALIAS is redundant and should be removed properly. Signed-off-by: Sean Wang <sean.wang(a)mediatek.com> --- sound/soc/mediatek/mt2701/mt2701-cs42448.c | 1 - sound/soc/mediatek/mt2701/mt2701-wm8960.c | 1 - sound/soc/mediatek/mt8173/mt8173-max98090.c | 1 - sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c | 1 - sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c | 1 - sound/soc/mediatek/mt8173/mt8173-rt5650.c | 1 - 6 files changed, 6 deletions(-) diff --git a/sound/soc/mediatek/mt2701/mt2701-cs42448.c b/sound/soc/mediatek/mt2701/mt2701-cs42448.c index 70f61d5..f664b9d 100644 --- a/sound/soc/mediatek/mt2701/mt2701-cs42448.c +++ b/sound/soc/mediatek/mt2701/mt2701-cs42448.c @@ -409,4 +409,3 @@ module_platform_driver(mt2701_cs42448_machine); MODULE_DESCRIPTION("MT2701 CS42448 ALSA SoC machine driver"); MODULE_AUTHOR("Ir Lian <ir.lian(a)mediatek.com>"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("mt2701 cs42448 soc card"); diff --git a/sound/soc/mediatek/mt2701/mt2701-wm8960.c b/sound/soc/mediatek/mt2701/mt2701-wm8960.c index a08ce23..a137057 100644 --- a/sound/soc/mediatek/mt2701/mt2701-wm8960.c +++ b/sound/soc/mediatek/mt2701/mt2701-wm8960.c @@ -172,5 +172,4 @@ module_platform_driver(mt2701_wm8960_machine); MODULE_DESCRIPTION("MT2701 WM8960 ALSA SoC machine driver"); MODULE_AUTHOR("Ryder Lee <ryder.lee(a)mediatek.com>"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("mt2701 wm8960 soc card"); diff --git a/sound/soc/mediatek/mt8173/mt8173-max98090.c b/sound/soc/mediatek/mt8173/mt8173-max98090.c index e0c2b23..6e58af6 100644 --- a/sound/soc/mediatek/mt8173/mt8173-max98090.c +++ b/sound/soc/mediatek/mt8173/mt8173-max98090.c @@ -203,5 +203,4 @@ module_platform_driver(mt8173_max98090_driver); MODULE_DESCRIPTION("MT8173 MAX98090 ALSA SoC machine driver"); MODULE_AUTHOR("Koro Chen <koro.chen(a)mediatek.com>"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:mt8173-max98090"); diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c index 5a9a548..addbbb3 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5514.c @@ -251,5 +251,4 @@ module_platform_driver(mt8173_rt5650_rt5514_driver); MODULE_DESCRIPTION("MT8173 RT5650 and RT5514 SoC machine driver"); MODULE_AUTHOR("Koro Chen <koro.chen(a)mediatek.com>"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:mtk-rt5650-rt5514"); diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c index b724808..f46e4c8 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650-rt5676.c @@ -308,5 +308,4 @@ module_platform_driver(mt8173_rt5650_rt5676_driver); MODULE_DESCRIPTION("MT8173 RT5650 and RT5676 SoC machine driver"); MODULE_AUTHOR("Koro Chen <koro.chen(a)mediatek.com>"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:mtk-rt5650-rt5676"); diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index 679fc8b..e2e42e6 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -335,5 +335,4 @@ module_platform_driver(mt8173_rt5650_driver); MODULE_DESCRIPTION("MT8173 RT5650 SoC machine driver"); MODULE_AUTHOR("Koro Chen <koro.chen(a)mediatek.com>"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:mtk-rt5650"); -- 2.7.4

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[alsa-devel] Roland SD50 issues
by Keith A. Milner 24 Mar '18

24 Mar '18

In my occasional, but ongoing, quest to try to debug some of the non- functioning audio on newer Roland/Boss devices, I've managed to get hold of a Roland SD-50 to do some comparisons. The lsusb -v and the aplay -L output for the SD-50 are attached. This device works with jack if I use a sample rate of 44100, buffer size 256, 3 periods/buffer. I can record and playback with no obvious issues. However, I can't seem to get this device working directly with ALSA. If I try playing back via audacity, the meter bars move, but there's no audio output. If I try aplay as follows: aplay -Dplughw:CARD=SD50 /usr/share/sounds/alsa/Front_Center.wav It pauses for a second as if it's playing, but I get no audio output. If I use the -v flag for aplay I get the following: Playing WAVE '/usr/share/sounds/alsa/Front_Center.wav' : Signed 16 bit Little Endian, Rate 48000 Hz, Mono Plug PCM: Rate conversion PCM (44100, sformat=S16_LE) Converter: linear-interpolation Protocol version: 10002 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 48000 exact rate : 48000 (48000/1) msbits : 16 buffer_size : 24000 period_size : 6000 period_time : 125011 tstamp_mode : NONE period_step : 1 avail_min : 6000 period_event : 0 start_threshold : 24000 stop_threshold : 24000 silence_threshold: 0 silence_size : 0 boundary : 6755399441055744000 Slave: Route conversion PCM (sformat=S24_3LE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 22051 period_size : 5513 period_time : 125011 tstamp_mode : NONE period_step : 1 avail_min : 5513 period_event : 0 start_threshold : 16539 stop_threshold : 22051 silence_threshold: 0 silence_size : 0 boundary : 6206804711446675456 Slave: Hardware PCM card 2 'SD-50' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S24_3LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 24 buffer_size : 22051 period_size : 5513 period_time : 125011 tstamp_mode : NONE period_step : 1 avail_min : 5513 period_event : 0 start_threshold : 16539 stop_threshold : 22051 silence_threshold: 0 silence_size : 0 boundary : 6206804711446675456 appl_ptr : 0 hw_ptr : 0 I'm trying to work out why I'm unable to get any audio output with ALSA directly, whilst it happily works with Jack. Any pointers are gratefully received. Cheers, Keith

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[alsa-devel] [PATCH 0/2] ALSA: usb-audio: UAC2 jack detection patchset
by Andrew Chant 24 Mar '18

24 Mar '18

These patches implement UAC2 jack detection. This allows USB-to-3.5mm adapters and similar USB devices to report jack state via alsa mixers, using the reporting provided by the UAC2 connector control. The first patch fixes up the function used to query UAC controls. The long lines in the table are unfortunate but keep the table readable. The long lines elsewhere in the patch were in line with existing long lines due to current code structure. Andrew Chant (2): ALSA: usb-audio: fix uac control query argument ALSA: usb-audio: UAC2 jack detection include/linux/usb/audio-v2.h | 4 +- sound/usb/clock.c | 5 +- sound/usb/mixer.c | 168 +++++++++++++++++++++++++++++------ 3 files changed, 146 insertions(+), 31 deletions(-) -- 2.17.0.rc0.231.g781580f067-goog

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[alsa-devel] [PATCH v3 0/2] UAC2 jack detection and clock validity
by Andrew Chant 24 Mar '18

24 Mar '18

These patches implement UAC2 jack detection. This allows USB-to-3.5mm adapters and similar USB devices to report jack state via alsa control, using the reporting provided by the UAC2 connector control. The second patch updates the clock validity control so it doesn't show up in mixer UIs. I declined to mark the controls as VOLATILE since clients should always be notified of changes, since UAC devices are expected to generate interrupt messages when a control value changes. v3 changes: - move channel count correction to first patch. - fix incorrect comment formatting and spacing. v2 changes: - use SNDRV_CTL_ELEM_IFACE_CARD instead of _MIXER - correct channel count to 1, cmask to 0. - add validity patch. Andrew Chant (2): ALSA: usb-audio: UAC2 jack detection ALSA: usb-audio: update clock valid control sound/usb/mixer.c | 117 ++++++++++++++++++++++++++++++++++++++++++---- 1 file changed, 109 insertions(+), 8 deletions(-) -- 2.17.0.rc0.231.g781580f067-goog

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[alsa-devel] [PATCH v2 0/2] UAC2 jack detection and clock validity
by Andrew Chant 24 Mar '18

24 Mar '18

These patches implement UAC2 jack detection. This allows USB-to-3.5mm adapters and similar USB devices to report jack state via alsa control, using the reporting provided by the UAC2 connector control. The second patch updates the clock validity control so it doesn't show up in mixer UIs. I declined to mark the controls as VOLATILE since clients should always be notified of changes, since UAC devices are expected to generate interrupt messages when a control value changes. v2 changes: - use SNDRV_CTL_ELEM_IFACE_CARD instead of _MIXER - correct channel count to 1, cmask to 0. - add validity patch. Andrew Chant (2): ALSA: usb-audio: UAC2 jack detection ALSA: usb-audio: update clock valid control sound/usb/mixer.c | 117 ++++++++++++++++++++++++++++++++++++++++++---- 1 file changed, 109 insertions(+), 8 deletions(-) -- 2.17.0.rc0.231.g781580f067-goog

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[alsa-devel] [PATCH 3/3] ASoC: TSCS42xx: Add CCF support to get sysclk
by Steven Eckhoff 24 Mar '18

24 Mar '18

The TSCS42xx relies on set_sysclk to get a unique clock id and rate, which prevents it from being used with the simple-card. Remove set_sysclk callback Add CCF support to get clock id and rate Add clocks and clock-names to device tree binding Signed-off-by: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> --- .../devicetree/bindings/sound/tscs42xx.txt | 6 ++ sound/soc/codecs/tscs42xx.c | 98 ++++++++++++++-------- sound/soc/codecs/tscs42xx.h | 2 +- 3 files changed, 69 insertions(+), 37 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/tscs42xx.txt b/Documentation/devicetree/bindings/sound/tscs42xx.txt index 2ac2f0996697..7eea32e9d078 100644 --- a/Documentation/devicetree/bindings/sound/tscs42xx.txt +++ b/Documentation/devicetree/bindings/sound/tscs42xx.txt @@ -8,9 +8,15 @@ Required Properties: - reg : <0x71> for analog mic <0x69> for digital mic + - clock-names: Must one of the following "mclk1", "xtal", "mclk2" + + - clocks: phandle of the clock that provides the codec sysclk + Example: wookie: codec@69 { compatible = "tempo,tscs42A2"; reg = <0x69>; + clock-names = "xtal"; + clocks = <&audio_xtal>; }; diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c index d41852a37ac6..081af0e05d33 100644 --- a/sound/soc/codecs/tscs42xx.c +++ b/sound/soc/codecs/tscs42xx.c @@ -12,6 +12,7 @@ #include <linux/module.h> #include <linux/delay.h> #include <linux/mutex.h> +#include <linux/clk.h> #include <sound/tlv.h> #include <sound/pcm_params.h> #include <sound/soc.h> @@ -40,6 +41,9 @@ struct tscs42xx { struct mutex pll_lock; struct regmap *regmap; + + struct clk *sysclk; + int sysclk_src_id; }; struct coeff_ram_ctl { @@ -1251,13 +1255,46 @@ static int tscs42xx_set_dai_bclk_ratio(struct snd_soc_dai *codec_dai, return 0; } -static int tscs42xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, - int clk_id, unsigned int freq, int dir) +static const struct snd_soc_dai_ops tscs42xx_dai_ops = { + .hw_params = tscs42xx_hw_params, + .mute_stream = tscs42xx_mute_stream, + .set_fmt = tscs42xx_set_dai_fmt, + .set_bclk_ratio = tscs42xx_set_dai_bclk_ratio, +}; + +static int part_is_valid(struct tscs42xx *tscs42xx) { - struct snd_soc_component *component = codec_dai->component; + int val; int ret; + unsigned int reg; - switch (clk_id) { + ret = regmap_read(tscs42xx->regmap, R_DEVIDH, &reg); + if (ret < 0) + return ret; + + val = reg << 8; + ret = regmap_read(tscs42xx->regmap, R_DEVIDL, &reg); + if (ret < 0) + return ret; + + val |= reg; + + switch (val) { + case 0x4A74: + case 0x4A73: + return true; + default: + return false; + }; +} + +static int set_sysclk(struct snd_soc_component *component) +{ + struct tscs42xx *tscs42xx = snd_soc_component_get_drvdata(component); + unsigned long freq; + int ret; + + switch (tscs42xx->sysclk_src_id) { case TSCS42XX_PLL_SRC_XTAL: case TSCS42XX_PLL_SRC_MCLK1: ret = snd_soc_component_write(component, R_PLLREFSEL, @@ -1285,6 +1322,7 @@ static int tscs42xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, return -EINVAL; } + freq = clk_get_rate(tscs42xx->sysclk); ret = set_pll_ctl_from_input_freq(component, freq); if (ret < 0) { dev_err(component->dev, @@ -1295,41 +1333,13 @@ static int tscs42xx_set_dai_sysclk(struct snd_soc_dai *codec_dai, return 0; } -static const struct snd_soc_dai_ops tscs42xx_dai_ops = { - .hw_params = tscs42xx_hw_params, - .mute_stream = tscs42xx_mute_stream, - .set_fmt = tscs42xx_set_dai_fmt, - .set_bclk_ratio = tscs42xx_set_dai_bclk_ratio, - .set_sysclk = tscs42xx_set_dai_sysclk, -}; - -static int part_is_valid(struct tscs42xx *tscs42xx) +static int tscs42xx_probe(struct snd_soc_component *component) { - int val; - int ret; - unsigned int reg; - - ret = regmap_read(tscs42xx->regmap, R_DEVIDH, &reg); - if (ret < 0) - return ret; - - val = reg << 8; - ret = regmap_read(tscs42xx->regmap, R_DEVIDL, &reg); - if (ret < 0) - return ret; - - val |= reg; - - switch (val) { - case 0x4A74: - case 0x4A73: - return true; - default: - return false; - }; + return set_sysclk(component); } static const struct snd_soc_component_driver soc_codec_dev_tscs42xx = { + .probe = tscs42xx_probe, .dapm_widgets = tscs42xx_dapm_widgets, .num_dapm_widgets = ARRAY_SIZE(tscs42xx_dapm_widgets), .dapm_routes = tscs42xx_intercon, @@ -1387,11 +1397,15 @@ static const struct reg_sequence tscs42xx_patch[] = { { R_AIC2, RV_AIC2_BLRCM_DAC_BCLK_LRCLK_SHARED }, }; +static char const * const src_names[TSCS42XX_PLL_SRC_CNT] = { + "xtal", "mclk1", "mclk2"}; + static int tscs42xx_i2c_probe(struct i2c_client *i2c, const struct i2c_device_id *id) { struct tscs42xx *tscs42xx; - int ret = 0; + int src; + int ret; tscs42xx = devm_kzalloc(&i2c->dev, sizeof(*tscs42xx), GFP_KERNEL); if (!tscs42xx) { @@ -1402,6 +1416,18 @@ static int tscs42xx_i2c_probe(struct i2c_client *i2c, } i2c_set_clientdata(i2c, tscs42xx); + for (src = TSCS42XX_PLL_SRC_XTAL; src < TSCS42XX_PLL_SRC_CNT; src++) { + tscs42xx->sysclk = devm_clk_get(&i2c->dev, src_names[src]); + if (!IS_ERR(tscs42xx->sysclk)) { + break; + } else if (PTR_ERR(tscs42xx->sysclk) != -ENOENT) { + ret = PTR_ERR(tscs42xx->sysclk); + dev_err(&i2c->dev, "Failed to get sysclk (%d)\n", ret); + return ret; + } + } + tscs42xx->sysclk_src_id = src; + tscs42xx->regmap = devm_regmap_init_i2c(i2c, &tscs42xx_regmap); if (IS_ERR(tscs42xx->regmap)) { ret = PTR_ERR(tscs42xx->regmap); diff --git a/sound/soc/codecs/tscs42xx.h b/sound/soc/codecs/tscs42xx.h index d4a30bcbf64b..814c8f3c4a68 100644 --- a/sound/soc/codecs/tscs42xx.h +++ b/sound/soc/codecs/tscs42xx.h @@ -7,10 +7,10 @@ #define __WOOKIE_H__ enum { - TSCS42XX_PLL_SRC_NONE, TSCS42XX_PLL_SRC_XTAL, TSCS42XX_PLL_SRC_MCLK1, TSCS42XX_PLL_SRC_MCLK2, + TSCS42XX_PLL_SRC_CNT, }; #define R_HPVOLL 0x0 -- 2.15.1

1 0

[alsa-devel] [PATCH 1/3] ALSA: usb-audio: Add native DSD support for Luxman DA-06
by Nobutaka Okabe 23 Mar '18

23 Mar '18

Add native DSD support quirk for Luxman DA-06 DAC, by adding the PID/VID 1852:5065. Rename "is_marantz_denon_dac()" function to "is_itf_usb_dsd_2alts_dac()" to cover broader device family sharing the same USB audio implementation(*). For the same reason, rename "is_teac_dsd_dac()" function to "is_itf_usb_dsd_3alts_dac()". (*) These devices have the same USB controller "ITF-USB DSD", supplied by INTERFACE Co., Ltd. "ITF-USB DSD" USB controller has two patterns, Pattern 1. (2 altsets version) - Altset 0: for control - Altset 1: for stream (S32) - Altset 2: for stream (S32, DSD_U32) Pattern 2. (3 altsets version) - Altset 0: for control - Altset 1: for stream (S16) - Altset 2: for stream (S32) - Altset 3: for stream (S32, DSD_U32) "is_itf_usb_dsd_2alts_dac()" returns true, if the DAC has "Pattern 1" USB controller, and "is_itf_usb_dsd_3alts_dac()" returns true, if "Pattern2". Signed-off-by: Nobutaka Okabe <nob77413(a)gmail.com> --- sound/usb/quirks.c | 29 ++++++++++++++++------------- 1 file changed, 16 insertions(+), 13 deletions(-) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index 794224e1d6df..006da37ad0d9 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1149,24 +1149,27 @@ bool snd_usb_get_sample_rate_quirk(struct snd_usb_audio *chip) return false; } -/* Marantz/Denon USB DACs need a vendor cmd to switch +/* ITF-USB DSD based DACs need a vendor cmd to switch * between PCM and native DSD mode + * (2 altsets version) */ -static bool is_marantz_denon_dac(unsigned int id) +static bool is_itf_usb_dsd_2alts_dac(unsigned int id) { switch (id) { case USB_ID(0x154e, 0x1003): /* Denon DA-300USB */ case USB_ID(0x154e, 0x3005): /* Marantz HD-DAC1 */ case USB_ID(0x154e, 0x3006): /* Marantz SA-14S1 */ + case USB_ID(0x1852, 0x5065): /* Luxman DA-06 */ return true; } return false; } -/* TEAC UD-501/UD-503/NT-503 USB DACs need a vendor cmd to switch - * between PCM/DOP and native DSD mode +/* ITF-USB DSD based DACs need a vendor cmd to switch + * between PCM and native DSD mode + * (3 altsets version) */ -static bool is_teac_dsd_dac(unsigned int id) +static bool is_itf_usb_dsd_3alts_dac(unsigned int id) { switch (id) { case USB_ID(0x0644, 0x8043): /* TEAC UD-501/UD-503/NT-503 */ @@ -1183,7 +1186,7 @@ int snd_usb_select_mode_quirk(struct snd_usb_substream *subs, struct usb_device *dev = subs->dev; int err; - if (is_marantz_denon_dac(subs->stream->chip->usb_id)) { + if (is_itf_usb_dsd_2alts_dac(subs->stream->chip->usb_id)) { /* First switch to alt set 0, otherwise the mode switch cmd * will not be accepted by the DAC */ @@ -1204,7 +1207,7 @@ int snd_usb_select_mode_quirk(struct snd_usb_substream *subs, break; } mdelay(20); - } else if (is_teac_dsd_dac(subs->stream->chip->usb_id)) { + } else if (is_itf_usb_dsd_3alts_dac(subs->stream->chip->usb_id)) { /* Vendor mode switch cmd is required. */ switch (fmt->altsetting) { case 3: /* DSD mode (DSD_U32) requested */ @@ -1300,10 +1303,10 @@ void snd_usb_ctl_msg_quirk(struct usb_device *dev, unsigned int pipe, (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) mdelay(20); - /* Marantz/Denon devices with USB DAC functionality need a delay + /* ITF-USB DSD based DACs functionality need a delay * after each class compliant request */ - if (is_marantz_denon_dac(chip->usb_id) + if (is_itf_usb_dsd_2alts_dac(chip->usb_id) && (requesttype & USB_TYPE_MASK) == USB_TYPE_CLASS) mdelay(20); @@ -1390,14 +1393,14 @@ u64 snd_usb_interface_dsd_format_quirks(struct snd_usb_audio *chip, break; } - /* Denon/Marantz devices with USB DAC functionality */ - if (is_marantz_denon_dac(chip->usb_id)) { + /* ITF-USB DSD based DACs (2 altsets version) */ + if (is_itf_usb_dsd_2alts_dac(chip->usb_id)) { if (fp->altsetting == 2) return SNDRV_PCM_FMTBIT_DSD_U32_BE; } - /* TEAC devices with USB DAC functionality */ - if (is_teac_dsd_dac(chip->usb_id)) { + /* ITF-USB DSD based DACs (3 altsets version) */ + if (is_itf_usb_dsd_3alts_dac(chip->usb_id)) { if (fp->altsetting == 3) return SNDRV_PCM_FMTBIT_DSD_U32_BE; } -- 2.11.1

2 5

[alsa-devel] [PATCH] ALSA: usb-audio: Add native DSD support for TEAC UD-301
by Nobutaka Okabe 23 Mar '18

23 Mar '18

Add native DSD support quirk for TEAC UD-301 DAC, by adding the PID/VID 0644:804a. Signed-off-by: Nobutaka Okabe <nob77413(a)gmail.com> --- Annotation: This patch should be applied before the "Luxman DA-06" patch set. Previously, I sent this patch to alsa-dev ML, but the status is "pending". So, I send again. sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index ea8f3de92fa4..794224e1d6df 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1171,6 +1171,7 @@ static bool is_teac_dsd_dac(unsigned int id) switch (id) { case USB_ID(0x0644, 0x8043): /* TEAC UD-501/UD-503/NT-503 */ case USB_ID(0x0644, 0x8044): /* Esoteric D-05X */ + case USB_ID(0x0644, 0x804a): /* TEAC UD-301 */ return true; } return false; -- 2.11.1

2 1

[alsa-devel] [PATCH 1/2] ALSA: pcm: Avoid potential races between OSS ioctls and read/write
by Takashi Iwai 23 Mar '18

23 Mar '18

Although we apply the params_lock mutex to the whole read and write operations as well as snd_pcm_oss_change_params(), we may still face some races. First off, the params_lock is taken inside the read and write loop. This is intentional for avoiding the too long locking, but it allows the in-between parameter change, which might lead to invalid pointers. We check the readiness of the stream and set up via snd_pcm_oss_make_ready() at the beginning of read and write, but it's called only once, by assuming that it remains ready in the rest. Second, many ioctls that may change the actual parameters (i.e. setting runtime->oss.params=1) aren't protected, hence they can be processed in a half-baked state. This patch is an attempt to plug these holes. The stream readiness check is moved inside the read/write inner loop, so that the stream is always set up in a proper state before further processing. Also, each ioctl that may change the parameter is wrapped with the params_lock for avoiding the races. The issues were triggered by syzkaller in a few different scenarios, particularly the one below appearing as GPF in loopback_pos_update. Reported-by: syzbot+c4227aec125487ec3efa(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/oss/pcm_oss.c | 134 +++++++++++++++++++++++++++++++++++++---------- 1 file changed, 106 insertions(+), 28 deletions(-) diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index 02298c9c6020..f8bfee8022e0 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -823,8 +823,8 @@ static int choose_rate(struct snd_pcm_substream *substream, return snd_pcm_hw_param_near(substream, params, SNDRV_PCM_HW_PARAM_RATE, best_rate, NULL); } -static int snd_pcm_oss_change_params(struct snd_pcm_substream *substream, - bool trylock) +/* call with params_lock held */ +static int snd_pcm_oss_change_params_locked(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; struct snd_pcm_hw_params *params, *sparams; @@ -838,11 +838,8 @@ static int snd_pcm_oss_change_params(struct snd_pcm_substream *substream, const struct snd_mask *sformat_mask; struct snd_mask mask; - if (trylock) { - if (!(mutex_trylock(&runtime->oss.params_lock))) - return -EAGAIN; - } else if (mutex_lock_interruptible(&runtime->oss.params_lock)) - return -ERESTARTSYS; + if (!runtime->oss.params) + return 0; sw_params = kzalloc(sizeof(*sw_params), GFP_KERNEL); params = kmalloc(sizeof(*params), GFP_KERNEL); sparams = kmalloc(sizeof(*sparams), GFP_KERNEL); @@ -1068,6 +1065,23 @@ static int snd_pcm_oss_change_params(struct snd_pcm_substream *substream, kfree(sw_params); kfree(params); kfree(sparams); + return err; +} + +/* this one takes the lock by itself */ +static int snd_pcm_oss_change_params(struct snd_pcm_substream *substream, + bool trylock) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + int err; + + if (trylock) { + if (!(mutex_trylock(&runtime->oss.params_lock))) + return -EAGAIN; + } else if (mutex_lock_interruptible(&runtime->oss.params_lock)) + return -ERESTARTSYS; + + err = snd_pcm_oss_change_params_locked(substream); mutex_unlock(&runtime->oss.params_lock); return err; } @@ -1096,11 +1110,14 @@ static int snd_pcm_oss_get_active_substream(struct snd_pcm_oss_file *pcm_oss_fil return 0; } +/* call with params_lock held */ static int snd_pcm_oss_prepare(struct snd_pcm_substream *substream) { int err; struct snd_pcm_runtime *runtime = substream->runtime; + if (!runtime->oss.prepare) + return 0; err = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_PREPARE, NULL); if (err < 0) { pcm_dbg(substream->pcm, @@ -1120,14 +1137,35 @@ static int snd_pcm_oss_make_ready(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime; int err; - if (substream == NULL) - return 0; runtime = substream->runtime; if (runtime->oss.params) { err = snd_pcm_oss_change_params(substream, false); if (err < 0) return err; } + if (runtime->oss.prepare) { + if (mutex_lock_interruptible(&runtime->oss.params_lock)) + return -ERESTARTSYS; + err = snd_pcm_oss_prepare(substream); + mutex_unlock(&runtime->oss.params_lock); + if (err < 0) + return err; + } + return 0; +} + +/* call with params_lock held */ +static int snd_pcm_oss_make_ready_locked(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime; + int err; + + runtime = substream->runtime; + if (runtime->oss.params) { + err = snd_pcm_oss_change_params_locked(substream); + if (err < 0) + return err; + } if (runtime->oss.prepare) { err = snd_pcm_oss_prepare(substream); if (err < 0) @@ -1332,13 +1370,14 @@ static ssize_t snd_pcm_oss_write1(struct snd_pcm_substream *substream, const cha if (atomic_read(&substream->mmap_count)) return -ENXIO; - if ((tmp = snd_pcm_oss_make_ready(substream)) < 0) - return tmp; while (bytes > 0) { if (mutex_lock_interruptible(&runtime->oss.params_lock)) { tmp = -ERESTARTSYS; break; } + tmp = snd_pcm_oss_make_ready_locked(substream); + if (tmp < 0) + goto err; if (bytes < runtime->oss.period_bytes || runtime->oss.buffer_used > 0) { tmp = bytes; if (tmp + runtime->oss.buffer_used > runtime->oss.period_bytes) @@ -1439,13 +1478,14 @@ static ssize_t snd_pcm_oss_read1(struct snd_pcm_substream *substream, char __use if (atomic_read(&substream->mmap_count)) return -ENXIO; - if ((tmp = snd_pcm_oss_make_ready(substream)) < 0) - return tmp; while (bytes > 0) { if (mutex_lock_interruptible(&runtime->oss.params_lock)) { tmp = -ERESTARTSYS; break; } + tmp = snd_pcm_oss_make_ready_locked(substream); + if (tmp < 0) + goto err; if (bytes < runtime->oss.period_bytes || runtime->oss.buffer_used > 0) { if (runtime->oss.buffer_used == 0) { tmp = snd_pcm_oss_read2(substream, runtime->oss.buffer, runtime->oss.period_bytes, 1); @@ -1501,10 +1541,12 @@ static int snd_pcm_oss_reset(struct snd_pcm_oss_file *pcm_oss_file) continue; runtime = substream->runtime; snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DROP, NULL); + mutex_lock(&runtime->oss.params_lock); runtime->oss.prepare = 1; runtime->oss.buffer_used = 0; runtime->oss.prev_hw_ptr_period = 0; runtime->oss.period_ptr = 0; + mutex_unlock(&runtime->oss.params_lock); } return 0; } @@ -1590,9 +1632,10 @@ static int snd_pcm_oss_sync(struct snd_pcm_oss_file *pcm_oss_file) goto __direct; if ((err = snd_pcm_oss_make_ready(substream)) < 0) return err; + if (mutex_lock_interruptible(&runtime->oss.params_lock)) + return -ERESTARTSYS; format = snd_pcm_oss_format_from(runtime->oss.format); width = snd_pcm_format_physical_width(format); - mutex_lock(&runtime->oss.params_lock); if (runtime->oss.buffer_used > 0) { #ifdef OSS_DEBUG pcm_dbg(substream->pcm, "sync: buffer_used\n"); @@ -1643,7 +1686,9 @@ static int snd_pcm_oss_sync(struct snd_pcm_oss_file *pcm_oss_file) substream->f_flags = saved_f_flags; if (err < 0) return err; + mutex_lock(&runtime->oss.params_lock); runtime->oss.prepare = 1; + mutex_unlock(&runtime->oss.params_lock); } substream = pcm_oss_file->streams[SNDRV_PCM_STREAM_CAPTURE]; @@ -1654,8 +1699,10 @@ static int snd_pcm_oss_sync(struct snd_pcm_oss_file *pcm_oss_file) err = snd_pcm_kernel_ioctl(substream, SNDRV_PCM_IOCTL_DROP, NULL); if (err < 0) return err; + mutex_lock(&runtime->oss.params_lock); runtime->oss.buffer_used = 0; runtime->oss.prepare = 1; + mutex_unlock(&runtime->oss.params_lock); } return 0; } @@ -1674,10 +1721,13 @@ static int snd_pcm_oss_set_rate(struct snd_pcm_oss_file *pcm_oss_file, int rate) rate = 1000; else if (rate > 192000) rate = 192000; + if (mutex_lock_interruptible(&runtime->oss.params_lock)) + return -ERESTARTSYS; if (runtime->oss.rate != rate) { runtime->oss.params = 1; runtime->oss.rate = rate; } + mutex_unlock(&runtime->oss.params_lock); } return snd_pcm_oss_get_rate(pcm_oss_file); } @@ -1705,10 +1755,13 @@ static int snd_pcm_oss_set_channels(struct snd_pcm_oss_file *pcm_oss_file, unsig if (substream == NULL) continue; runtime = substream->runtime; + if (mutex_lock_interruptible(&runtime->oss.params_lock)) + return -ERESTARTSYS; if (runtime->oss.channels != channels) { runtime->oss.params = 1; runtime->oss.channels = channels; } + mutex_unlock(&runtime->oss.params_lock); } return snd_pcm_oss_get_channels(pcm_oss_file); } @@ -1794,10 +1847,13 @@ static int snd_pcm_oss_set_format(struct snd_pcm_oss_file *pcm_oss_file, int for if (substream == NULL) continue; runtime = substream->runtime; + if (mutex_lock_interruptible(&runtime->oss.params_lock)) + return -ERESTARTSYS; if (runtime->oss.format != format) { runtime->oss.params = 1; runtime->oss.format = format; } + mutex_unlock(&runtime->oss.params_lock); } } return snd_pcm_oss_get_format(pcm_oss_file); @@ -1817,8 +1873,6 @@ static int snd_pcm_oss_set_subdivide1(struct snd_pcm_substream *substream, int s { struct snd_pcm_runtime *runtime; - if (substream == NULL) - return 0; runtime = substream->runtime; if (subdivide == 0) { subdivide = runtime->oss.subdivision; @@ -1842,9 +1896,16 @@ static int snd_pcm_oss_set_subdivide(struct snd_pcm_oss_file *pcm_oss_file, int for (idx = 1; idx >= 0; --idx) { struct snd_pcm_substream *substream = pcm_oss_file->streams[idx]; + struct snd_pcm_runtime *runtime; + if (substream == NULL) continue; - if ((err = snd_pcm_oss_set_subdivide1(substream, subdivide)) < 0) + runtime = substream->runtime; + if (mutex_lock_interruptible(&runtime->oss.params_lock)) + return -ERESTARTSYS; + err = snd_pcm_oss_set_subdivide1(substream, subdivide); + mutex_unlock(&runtime->oss.params_lock); + if (err < 0) return err; } return err; @@ -1854,8 +1915,6 @@ static int snd_pcm_oss_set_fragment1(struct snd_pcm_substream *substream, unsign { struct snd_pcm_runtime *runtime; - if (substream == NULL) - return 0; runtime = substream->runtime; if (runtime->oss.subdivision || runtime->oss.fragshift) return -EINVAL; @@ -1875,9 +1934,16 @@ static int snd_pcm_oss_set_fragment(struct snd_pcm_oss_file *pcm_oss_file, unsig for (idx = 1; idx >= 0; --idx) { struct snd_pcm_substream *substream = pcm_oss_file->streams[idx]; + struct snd_pcm_runtime *runtime; + if (substream == NULL) continue; - if ((err = snd_pcm_oss_set_fragment1(substream, val)) < 0) + runtime = substream->runtime; + if (mutex_lock_interruptible(&runtime->oss.params_lock)) + return -ERESTARTSYS; + err = snd_pcm_oss_set_fragment1(substream, val); + mutex_unlock(&runtime->oss.params_lock); + if (err < 0) return err; } return err; @@ -1961,6 +2027,9 @@ static int snd_pcm_oss_set_trigger(struct snd_pcm_oss_file *pcm_oss_file, int tr } if (psubstream) { runtime = psubstream->runtime; + cmd = 0; + if (mutex_lock_interruptible(&runtime->oss.params_lock)) + return -ERESTARTSYS; if (trigger & PCM_ENABLE_OUTPUT) { if (runtime->oss.trigger) goto _skip1; @@ -1978,13 +2047,19 @@ static int snd_pcm_oss_set_trigger(struct snd_pcm_oss_file *pcm_oss_file, int tr cmd = SNDRV_PCM_IOCTL_DROP; runtime->oss.prepare = 1; } - err = snd_pcm_kernel_ioctl(psubstream, cmd, NULL); - if (err < 0) - return err; - } _skip1: + mutex_unlock(&runtime->oss.params_lock); + if (cmd) { + err = snd_pcm_kernel_ioctl(psubstream, cmd, NULL); + if (err < 0) + return err; + } + } if (csubstream) { runtime = csubstream->runtime; + cmd = 0; + if (mutex_lock_interruptible(&runtime->oss.params_lock)) + return -ERESTARTSYS; if (trigger & PCM_ENABLE_INPUT) { if (runtime->oss.trigger) goto _skip2; @@ -1999,11 +2074,14 @@ static int snd_pcm_oss_set_trigger(struct snd_pcm_oss_file *pcm_oss_file, int tr cmd = SNDRV_PCM_IOCTL_DROP; runtime->oss.prepare = 1; } - err = snd_pcm_kernel_ioctl(csubstream, cmd, NULL); - if (err < 0) - return err; - } _skip2: + mutex_unlock(&runtime->oss.params_lock); + if (cmd) { + err = snd_pcm_kernel_ioctl(csubstream, cmd, NULL); + if (err < 0) + return err; + } + } return 0; } -- 2.16.2

1 1

[alsa-devel] [PATCH v3] ASoC: AMD: Enable/Disable auxiliary clock via common clock framework
by Akshu Agrawal 23 Mar '18

23 Mar '18

This enables/disables and sets auxiliary clock at 25Mhz. It uses common clock framework for proper ref counting. This approach will save power in comparison to keeping it always On in firmware. V2: Correcting the pin to OSCOUT1 from OSCOUT2 V3: Fix error/warnings from kbuild test TEST= aplay -vv <file> check register to see clock enabled kill aplay check register to see clock disabled Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-da7219-max98357a.c | 135 ++++++++++++++++++++++++++++++++++- 1 file changed, 133 insertions(+), 2 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 99c6b5c..baea370 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -30,10 +30,13 @@ #include <sound/soc-dapm.h> #include <sound/jack.h> #include <linux/clk.h> +#include <linux/clkdev.h> +#include <linux/clk-provider.h> #include <linux/gpio.h> #include <linux/module.h> #include <linux/i2c.h> #include <linux/acpi.h> +#include <linux/types.h> #include "../codecs/da7219.h" #include "../codecs/da7219-aad.h" @@ -41,6 +44,18 @@ #define CZ_PLAT_CLK 24000000 #define MCLK_RATE 24576000 #define DUAL_CHANNEL 2 +#define CLKDRVSTR2 0x28 +#define MISCCLKCNTL1 0x40 +#define OSCCLKENB 2 +#define OSCOUT1CLK25MHZ 16 + +struct cz_clock { + const char* acp_clks_name; + struct clk_hw acp_clks_hw; + struct clk_lookup *acp_clks_lookup; + struct clk *acp_clks; + void __iomem *res_base; +}; static struct snd_soc_jack cz_jack; struct clk *da7219_dai_clk; @@ -91,6 +106,8 @@ static int cz_da7219_hw_params(struct snd_pcm_substream *substream, { int ret = 0; struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct clk *acpd7219_clk; ret = clk_prepare_enable(da7219_dai_clk); if (ret < 0) { @@ -98,13 +115,27 @@ static int cz_da7219_hw_params(struct snd_pcm_substream *substream, return ret; } + acpd7219_clk = clk_get(card->dev, "acpd7219-clks"); + ret = clk_prepare_enable(acpd7219_clk); + if (ret < 0) { + dev_err(rtd->dev, "can't enable oscillator clock %d\n", ret); + return ret; + } + return ret; } static int cz_da7219_hw_free(struct snd_pcm_substream *substream) { + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_card *card = rtd->card; + struct clk *acpd7219_clk; + clk_disable_unprepare(da7219_dai_clk); + acpd7219_clk = clk_get(card->dev, "acpd7219-clks"); + clk_disable_unprepare(acpd7219_clk); + return 0; } @@ -237,9 +268,106 @@ static int cz_fe_startup(struct snd_pcm_substream *substream) .num_controls = ARRAY_SIZE(cz_mc_controls), }; +static int acpd7219_clks_prepare(struct clk_hw *hw) +{ + u32 reg_val; + struct cz_clock *cz_clock_obj = + container_of(hw, struct cz_clock, acp_clks_hw); + void __iomem *base = cz_clock_obj->res_base; + + reg_val = readl(base + MISCCLKCNTL1); + reg_val &= ~(0x1 << OSCCLKENB); + writel(reg_val, base + MISCCLKCNTL1); + reg_val = readl(base + CLKDRVSTR2); + reg_val |= (0x1 << OSCOUT1CLK25MHZ); + writel(reg_val, base + CLKDRVSTR2); + + return 0; +} + +static void acpd7219_clks_unprepare(struct clk_hw *hw) +{ + u32 reg_val; + struct cz_clock *cz_clock_obj = + container_of(hw, struct cz_clock, acp_clks_hw); + void __iomem *base = cz_clock_obj->res_base; + + reg_val = readl(base + MISCCLKCNTL1); + reg_val |= (0x1 << OSCCLKENB); + writel(reg_val, base + MISCCLKCNTL1); +} + +static int acpd7219_clks_is_enabled(struct clk_hw *hw) +{ + u32 reg_val; + struct cz_clock *cz_clock_obj = + container_of(hw, struct cz_clock, acp_clks_hw); + void __iomem *base = cz_clock_obj->res_base; + + reg_val = readl(base + MISCCLKCNTL1); + + return !(reg_val & OSCCLKENB); +} + +const struct clk_ops acpd7219_clks_ops = { + .prepare = acpd7219_clks_prepare, + .unprepare = acpd7219_clks_unprepare, + .is_enabled = acpd7219_clks_is_enabled, +}; + +static int register_acpd7219_clocks(struct platform_device *pdev) +{ + struct clk_init_data init = {}; + struct clk *acp_clks; + struct clk_lookup *acp_clks_lookup; + struct cz_clock *cz_clock_obj; + struct resource *res; + struct device dev = pdev->dev; + + cz_clock_obj = kzalloc(sizeof(struct cz_clock), GFP_KERNEL); + if (!cz_clock_obj) + return -ENOMEM; + + cz_clock_obj->acp_clks_name = "acpd7219-clks"; + + init.parent_names = NULL; + init.num_parents = 0; + init.name = cz_clock_obj->acp_clks_name; + init.ops = &acpd7219_clks_ops; + cz_clock_obj->acp_clks_hw.init = &init; + + acp_clks = devm_clk_register(&dev, &cz_clock_obj->acp_clks_hw); + if (IS_ERR(acp_clks)) + { + dev_err(&dev, "Failed to register DAI clocks: %ld\n", + PTR_ERR(acp_clks)); + return -EINVAL; + } + cz_clock_obj->acp_clks = acp_clks; + + acp_clks_lookup = clkdev_create(acp_clks, cz_clock_obj->acp_clks_name, + "%s", dev_name(&dev)); + if (!acp_clks_lookup) + dev_warn(&dev, "Failed to create DAI clkdev"); + else + cz_clock_obj->acp_clks_lookup = acp_clks_lookup; + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + dev_err(&pdev->dev, "Failed to get misc io resource.\n"); + return -EINVAL; + } + cz_clock_obj->res_base = devm_ioremap_nocache(&pdev->dev, res->start, + resource_size(res)); + if (!cz_clock_obj->res_base) + return -ENOMEM; + + return 0; +} + static int cz_probe(struct platform_device *pdev) { - int ret; + int ret = 0; struct snd_soc_card *card; card = &cz_card; @@ -252,7 +380,10 @@ static int cz_probe(struct platform_device *pdev) cz_card.name, ret); return ret; } - return 0; + + ret = register_acpd7219_clocks(pdev); + + return ret; } static const struct acpi_device_id cz_audio_acpi_match[] = { -- 1.9.1

2 3

[alsa-devel] [PATCH - PCM HW APPEND 1/1] pcm: hw: Keep control data from kernel when SND_PCM_APPEND
by twischer＠de.adit-jv.com 23 Mar '18

23 Mar '18

From: Timo Wischer <twischer(a)de.adit-jv.com> Without this fix the application pointer would be reseted whenever an application opens a device with SND_PCM_APPEND. This would result in an Xrun if the device is already opened and in running state and the appl_ptr is use. Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index b93926c..65b198c 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -1004,16 +1004,28 @@ static int map_status_and_control_data(snd_pcm_t *pcm, bool force_fallback) hw->sync_ptr = NULL; } - /* Initialize the data. */ - hw->mmap_control->appl_ptr = 0; - hw->mmap_control->avail_min = 1; + /* do not initialize in case of append and keep the values from the + * kernel + */ + if (!(pcm->mode & SND_PCM_APPEND)) { + /* Initialize the data. */ + hw->mmap_control->appl_ptr = 0; + hw->mmap_control->avail_min = 1; + } snd_pcm_set_hw_ptr(pcm, &hw->mmap_status->hw_ptr, hw->fd, SNDRV_PCM_MMAP_OFFSET_STATUS + offsetof(struct snd_pcm_mmap_status, hw_ptr)); snd_pcm_set_appl_ptr(pcm, &hw->mmap_control->appl_ptr, hw->fd, SNDRV_PCM_MMAP_OFFSET_CONTROL); if (hw->mmap_control_fallbacked) { - err = sync_ptr1(hw, 0); + unsigned int flags; + /* read appl_ptr and avail_min from kernel when device opened + * with SND_PCM_APPEND flag + */ + if (pcm->mode & SND_PCM_APPEND) + flags = SNDRV_PCM_SYNC_PTR_APPL | + SNDRV_PCM_SYNC_PTR_AVAIL_MIN; + err = sync_ptr1(hw, flags); if (err < 0) return err; } @@ -1539,6 +1551,8 @@ int snd_pcm_hw_open_fd(snd_pcm_t **pcmp, const char *name, int fd, mode |= SND_PCM_NONBLOCK; if (fmode & O_ASYNC) mode |= SND_PCM_ASYNC; + if (fmode & O_APPEND) + mode |= SND_PCM_APPEND; if (ioctl(fd, SNDRV_PCM_IOCTL_PVERSION, &ver) < 0) { ret = -errno; -- 2.7.4

2 1

[alsa-devel] [PATCH] ALSA: usb-audio: Add native DSD support for TEAC UD-301
by Nobutaka Okabe 23 Mar '18

23 Mar '18

Add native DSD support quirk for TEAC UD-301 DAC, by adding the PID/VID 0644:804a. Signed-off-by: Nobutaka Okabe <nob77413(a)gmail.com> --- sound/usb/quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/quirks.c b/sound/usb/quirks.c index ea8f3de92fa4..794224e1d6df 100644 --- a/sound/usb/quirks.c +++ b/sound/usb/quirks.c @@ -1171,6 +1171,7 @@ static bool is_teac_dsd_dac(unsigned int id) switch (id) { case USB_ID(0x0644, 0x8043): /* TEAC UD-501/UD-503/NT-503 */ case USB_ID(0x0644, 0x8044): /* Esoteric D-05X */ + case USB_ID(0x0644, 0x804a): /* TEAC UD-301 */ return true; } return false; -- 2.11.1

1 0

[alsa-devel] [PATCH][next] ASoC: amd: make pointer da7219_dai_clk static
by Colin King 23 Mar '18

23 Mar '18

From: Colin Ian King <colin.king(a)canonical.com> The pointer da7219_dai_clk is local to the source and does not need to be in global scope, so make it static. Cleans up sparse warning: sound/soc/amd/acp-da7219-max98357a.c:46:12: warning: symbol 'da7219_dai_clk' was not declared. Should it be static? Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/amd/acp-da7219-max98357a.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index 99c6b5ce5c6c..af458bd36c72 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -43,7 +43,7 @@ #define DUAL_CHANNEL 2 static struct snd_soc_jack cz_jack; -struct clk *da7219_dai_clk; +static struct clk *da7219_dai_clk; static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) { -- 2.15.1

1 0

[alsa-devel] [PATCH V2] ASoC: dwc: I2S Controller instance param added
by Vijendar Mukunda 23 Mar '18

23 Mar '18

When multiple I2S controller instances created, i2s_instance parameter refers to i2s controller instance value. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Reviewed-by: Alex Deucher <alexander.deucher(a)amd.com> --- v1->v2: defined I2S instance macros in designware header file include/sound/designware_i2s.h | 6 ++++++ sound/soc/dwc/dwc-i2s.c | 1 + sound/soc/dwc/local.h | 1 + 3 files changed, 8 insertions(+) diff --git a/include/sound/designware_i2s.h b/include/sound/designware_i2s.h index 830f5ca..8113759 100644 --- a/include/sound/designware_i2s.h +++ b/include/sound/designware_i2s.h @@ -44,6 +44,10 @@ struct i2s_platform_data { int channel; u32 snd_fmts; u32 snd_rates; + /* i2s_instance parameter returns I2S controller instance value + * when multiple I2S controllers instantiated + */ + u32 i2s_instance; #define DW_I2S_QUIRK_COMP_REG_OFFSET (1 << 0) #define DW_I2S_QUIRK_COMP_PARAM1 (1 << 1) @@ -74,5 +78,7 @@ struct i2s_dma_data { #define FOUR_CHANNEL_SUPPORT 4 /* up to 3.1 */ #define SIX_CHANNEL_SUPPORT 6 /* up to 5.1 */ #define EIGHT_CHANNEL_SUPPORT 8 /* up to 7.1 */ +#define I2S_SP_INSTANCE 1 +#define I2S_BT_INSTANCE 2 #endif /* __SOUND_DESIGNWARE_I2S_H */ diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index e27e21f..20145c0 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -549,6 +549,7 @@ static int dw_configure_dai_by_pd(struct dw_i2s_dev *dev, if (dev->quirks & DW_I2S_QUIRK_16BIT_IDX_OVERRIDE) idx = 1; + dev->i2s_instance = pdata->i2s_instance; /* Set DMA slaves info */ dev->play_dma_data.pd.data = pdata->play_dma_data; dev->capture_dma_data.pd.data = pdata->capture_dma_data; diff --git a/sound/soc/dwc/local.h b/sound/soc/dwc/local.h index 91dc70a..e89e464 100644 --- a/sound/soc/dwc/local.h +++ b/sound/soc/dwc/local.h @@ -98,6 +98,7 @@ struct dw_i2s_dev { u32 ccr; u32 xfer_resolution; u32 fifo_th; + u32 i2s_instance; /* data related to DMA transfers b/w i2s and DMAC */ union dw_i2s_snd_dma_data play_dma_data; -- 2.7.4

2 1

[alsa-devel] pulseaudio\aplay\arecord
by Muni Sekhar 22 Mar '18

22 Mar '18

[ Please keep me in CC as I'm not subscribed to the list] Hi All, We are using the Ubuntu 16.04.3 LTS (4.4.0-28-generic) kernel. I’ve a sound card hardware which supports analog loopback. During analog loopback test, aplay fails with “aplay: pcm_write:1940: write error: Input/output error”. In our system, I noticed that once the sound driver is loaded, PulseAudio daemon autospawns and taking over control of the sound device. Does the ‘pulseaudio’ causes ‘aplay’ failure? Can we use aplay\arecord if already pulseaudio daemon is running? -- Thanks, Sekhar

2 1

[alsa-devel] [trivial PATCH V2] treewide: Align function definition open/close braces
by Joe Perches 22 Mar '18

22 Mar '18

Some functions definitions have either the initial open brace and/or the closing brace outside of column 1. Move those braces to column 1. This allows various function analyzers like gnu complexity to work properly for these modified functions. Signed-off-by: Joe Perches <joe(a)perches.com> Acked-by: Andy Shevchenko <andy.shevchenko(a)gmail.com> Acked-by: Paul Moore <paul(a)paul-moore.com> Acked-by: Alex Deucher <alexander.deucher(a)amd.com> Acked-by: Dave Chinner <dchinner(a)redhat.com> Reviewed-by: Darrick J. Wong <darrick.wong(a)oracle.com> Acked-by: Alexandre Belloni <alexandre.belloni(a)free-electrons.com> Acked-by: Martin K. Petersen <martin.petersen(a)oracle.com> Acked-by: Takashi Iwai <tiwai(a)suse.de> Acked-by: Mauro Carvalho Chehab <mchehab(a)s-opensource.com> --- git diff -w still shows no difference. This patch was sent but December and not applied. As the trivial maintainer seems not active, it'd be nice if Andrew Morton picks this up. V2: Remove fs/xfs/libxfs/xfs_alloc.c as it's updated and remerge the rest arch/x86/include/asm/atomic64_32.h | 2 +- drivers/acpi/custom_method.c | 2 +- drivers/acpi/fan.c | 2 +- drivers/gpu/drm/amd/display/dc/core/dc.c | 2 +- drivers/media/i2c/msp3400-kthreads.c | 2 +- drivers/message/fusion/mptsas.c | 2 +- drivers/net/ethernet/qlogic/netxen/netxen_nic_init.c | 2 +- drivers/net/wireless/ath/ath9k/xmit.c | 2 +- drivers/platform/x86/eeepc-laptop.c | 2 +- drivers/rtc/rtc-ab-b5ze-s3.c | 2 +- drivers/scsi/dpt_i2o.c | 2 +- drivers/scsi/sym53c8xx_2/sym_glue.c | 2 +- fs/locks.c | 2 +- fs/ocfs2/stack_user.c | 2 +- fs/xfs/xfs_export.c | 2 +- kernel/audit.c | 6 +++--- kernel/trace/trace_printk.c | 4 ++-- lib/raid6/sse2.c | 14 +++++++------- sound/soc/fsl/fsl_dma.c | 2 +- 19 files changed, 28 insertions(+), 28 deletions(-) diff --git a/arch/x86/include/asm/atomic64_32.h b/arch/x86/include/asm/atomic64_32.h index 46e1ef17d92d..92212bf0484f 100644 --- a/arch/x86/include/asm/atomic64_32.h +++ b/arch/x86/include/asm/atomic64_32.h @@ -123,7 +123,7 @@ static inline long long arch_atomic64_read(const atomic64_t *v) long long r; alternative_atomic64(read, "=&A" (r), "c" (v) : "memory"); return r; - } +} /** * arch_atomic64_add_return - add and return diff --git a/drivers/acpi/custom_method.c b/drivers/acpi/custom_method.c index b33fba70ec51..a07fbe999eb6 100644 --- a/drivers/acpi/custom_method.c +++ b/drivers/acpi/custom_method.c @@ -97,7 +97,7 @@ static void __exit acpi_custom_method_exit(void) { if (cm_dentry) debugfs_remove(cm_dentry); - } +} module_init(acpi_custom_method_init); module_exit(acpi_custom_method_exit); diff --git a/drivers/acpi/fan.c b/drivers/acpi/fan.c index 6cf4988206f2..3563103590c6 100644 --- a/drivers/acpi/fan.c +++ b/drivers/acpi/fan.c @@ -219,7 +219,7 @@ fan_set_cur_state(struct thermal_cooling_device *cdev, unsigned long state) return fan_set_state_acpi4(device, state); else return fan_set_state(device, state); - } +} static const struct thermal_cooling_device_ops fan_cooling_ops = { .get_max_state = fan_get_max_state, diff --git a/drivers/gpu/drm/amd/display/dc/core/dc.c b/drivers/gpu/drm/amd/display/dc/core/dc.c index 8394d69b963f..e934326a95d3 100644 --- a/drivers/gpu/drm/amd/display/dc/core/dc.c +++ b/drivers/gpu/drm/amd/display/dc/core/dc.c @@ -588,7 +588,7 @@ static void disable_dangling_plane(struct dc *dc, struct dc_state *context) ******************************************************************************/ struct dc *dc_create(const struct dc_init_data *init_params) - { +{ struct dc *dc = kzalloc(sizeof(*dc), GFP_KERNEL); unsigned int full_pipe_count; diff --git a/drivers/media/i2c/msp3400-kthreads.c b/drivers/media/i2c/msp3400-kthreads.c index 4dd01e9f553b..dc6cb8d475b3 100644 --- a/drivers/media/i2c/msp3400-kthreads.c +++ b/drivers/media/i2c/msp3400-kthreads.c @@ -885,7 +885,7 @@ static int msp34xxg_modus(struct i2c_client *client) } static void msp34xxg_set_source(struct i2c_client *client, u16 reg, int in) - { +{ struct msp_state *state = to_state(i2c_get_clientdata(client)); int source, matrix; diff --git a/drivers/message/fusion/mptsas.c b/drivers/message/fusion/mptsas.c index 439ee9c5f535..231f3a1e27bf 100644 --- a/drivers/message/fusion/mptsas.c +++ b/drivers/message/fusion/mptsas.c @@ -2967,7 +2967,7 @@ mptsas_exp_repmanufacture_info(MPT_ADAPTER *ioc, mutex_unlock(&ioc->sas_mgmt.mutex); out: return ret; - } +} static void mptsas_parse_device_info(struct sas_identify *identify, diff --git a/drivers/net/ethernet/qlogic/netxen/netxen_nic_init.c b/drivers/net/ethernet/qlogic/netxen/netxen_nic_init.c index 3dd973475125..0ea141ece19e 100644 --- a/drivers/net/ethernet/qlogic/netxen/netxen_nic_init.c +++ b/drivers/net/ethernet/qlogic/netxen/netxen_nic_init.c @@ -603,7 +603,7 @@ static struct uni_table_desc *nx_get_table_desc(const u8 *unirom, int section) static int netxen_nic_validate_header(struct netxen_adapter *adapter) - { +{ const u8 *unirom = adapter->fw->data; struct uni_table_desc *directory = (struct uni_table_desc *) &unirom[0]; u32 fw_file_size = adapter->fw->size; diff --git a/drivers/net/wireless/ath/ath9k/xmit.c b/drivers/net/wireless/ath/ath9k/xmit.c index 396bf05c6bf6..88be55ed5b4d 100644 --- a/drivers/net/wireless/ath/ath9k/xmit.c +++ b/drivers/net/wireless/ath/ath9k/xmit.c @@ -252,7 +252,7 @@ ath_tid_pull(struct ath_atx_tid *tid) } return skb; - } +} static bool ath_tid_has_buffered(struct ath_atx_tid *tid) diff --git a/drivers/platform/x86/eeepc-laptop.c b/drivers/platform/x86/eeepc-laptop.c index 5a681962899c..4c38904a8a32 100644 --- a/drivers/platform/x86/eeepc-laptop.c +++ b/drivers/platform/x86/eeepc-laptop.c @@ -492,7 +492,7 @@ static void eeepc_platform_exit(struct eeepc_laptop *eeepc) * potentially bad time, such as a timer interrupt. */ static void tpd_led_update(struct work_struct *work) - { +{ struct eeepc_laptop *eeepc; eeepc = container_of(work, struct eeepc_laptop, tpd_led_work); diff --git a/drivers/rtc/rtc-ab-b5ze-s3.c b/drivers/rtc/rtc-ab-b5ze-s3.c index e55f35fa0b58..8dc451932446 100644 --- a/drivers/rtc/rtc-ab-b5ze-s3.c +++ b/drivers/rtc/rtc-ab-b5ze-s3.c @@ -646,7 +646,7 @@ static int abb5zes3_rtc_set_alarm(struct device *dev, struct rtc_wkalrm *alarm) ret); return ret; - } +} /* Enable or disable battery low irq generation */ static inline int _abb5zes3_rtc_battery_low_irq_enable(struct regmap *regmap, diff --git a/drivers/scsi/dpt_i2o.c b/drivers/scsi/dpt_i2o.c index 0f30792d74c4..cc5fa99a6530 100644 --- a/drivers/scsi/dpt_i2o.c +++ b/drivers/scsi/dpt_i2o.c @@ -3521,7 +3521,7 @@ static int adpt_i2o_systab_send(adpt_hba* pHba) #endif return ret; - } +} /*============================================================================ diff --git a/drivers/scsi/sym53c8xx_2/sym_glue.c b/drivers/scsi/sym53c8xx_2/sym_glue.c index 791a2182de53..7320d5fe4cbc 100644 --- a/drivers/scsi/sym53c8xx_2/sym_glue.c +++ b/drivers/scsi/sym53c8xx_2/sym_glue.c @@ -1393,7 +1393,7 @@ static struct Scsi_Host *sym_attach(struct scsi_host_template *tpnt, int unit, scsi_host_put(shost); return NULL; - } +} /* diff --git a/fs/locks.c b/fs/locks.c index d56a14894fb2..0feaed9f589b 100644 --- a/fs/locks.c +++ b/fs/locks.c @@ -559,7 +559,7 @@ static const struct lock_manager_operations lease_manager_ops = { * Initialize a lease, use the default lock manager operations */ static int lease_init(struct file *filp, long type, struct file_lock *fl) - { +{ if (assign_type(fl, type) != 0) return -EINVAL; diff --git a/fs/ocfs2/stack_user.c b/fs/ocfs2/stack_user.c index dae9eb7c441e..d2fb97b173da 100644 --- a/fs/ocfs2/stack_user.c +++ b/fs/ocfs2/stack_user.c @@ -398,7 +398,7 @@ static int ocfs2_control_do_setnode_msg(struct file *file, static int ocfs2_control_do_setversion_msg(struct file *file, struct ocfs2_control_message_setv *msg) - { +{ long major, minor; char *ptr = NULL; struct ocfs2_control_private *p = file->private_data; diff --git a/fs/xfs/xfs_export.c b/fs/xfs/xfs_export.c index 761f3189eff2..eed698aa9f16 100644 --- a/fs/xfs/xfs_export.c +++ b/fs/xfs/xfs_export.c @@ -122,7 +122,7 @@ xfs_nfs_get_inode( struct super_block *sb, u64 ino, u32 generation) - { +{ xfs_mount_t *mp = XFS_M(sb); xfs_inode_t *ip; int error; diff --git a/kernel/audit.c b/kernel/audit.c index 8fe6dfb67a94..a97d004375e3 100644 --- a/kernel/audit.c +++ b/kernel/audit.c @@ -486,15 +486,15 @@ static int audit_set_failure(u32 state) * Drop any references inside the auditd connection tracking struct and free * the memory. */ - static void auditd_conn_free(struct rcu_head *rcu) - { +static void auditd_conn_free(struct rcu_head *rcu) +{ struct auditd_connection *ac; ac = container_of(rcu, struct auditd_connection, rcu); put_pid(ac->pid); put_net(ac->net); kfree(ac); - } +} /** * auditd_set - Set/Reset the auditd connection state diff --git a/kernel/trace/trace_printk.c b/kernel/trace/trace_printk.c index ad1d6164e946..50f44b7b2b32 100644 --- a/kernel/trace/trace_printk.c +++ b/kernel/trace/trace_printk.c @@ -196,7 +196,7 @@ struct notifier_block module_trace_bprintk_format_nb = { }; int __trace_bprintk(unsigned long ip, const char *fmt, ...) - { +{ int ret; va_list ap; @@ -214,7 +214,7 @@ int __trace_bprintk(unsigned long ip, const char *fmt, ...) EXPORT_SYMBOL_GPL(__trace_bprintk); int __ftrace_vbprintk(unsigned long ip, const char *fmt, va_list ap) - { +{ if (unlikely(!fmt)) return 0; diff --git a/lib/raid6/sse2.c b/lib/raid6/sse2.c index 1d2276b007ee..8191e1d0d2fb 100644 --- a/lib/raid6/sse2.c +++ b/lib/raid6/sse2.c @@ -91,7 +91,7 @@ static void raid6_sse21_gen_syndrome(int disks, size_t bytes, void **ptrs) static void raid6_sse21_xor_syndrome(int disks, int start, int stop, size_t bytes, void **ptrs) - { +{ u8 **dptr = (u8 **)ptrs; u8 *p, *q; int d, z, z0; @@ -200,9 +200,9 @@ static void raid6_sse22_gen_syndrome(int disks, size_t bytes, void **ptrs) kernel_fpu_end(); } - static void raid6_sse22_xor_syndrome(int disks, int start, int stop, +static void raid6_sse22_xor_syndrome(int disks, int start, int stop, size_t bytes, void **ptrs) - { +{ u8 **dptr = (u8 **)ptrs; u8 *p, *q; int d, z, z0; @@ -265,7 +265,7 @@ static void raid6_sse22_gen_syndrome(int disks, size_t bytes, void **ptrs) asm volatile("sfence" : : : "memory"); kernel_fpu_end(); - } +} const struct raid6_calls raid6_sse2x2 = { raid6_sse22_gen_syndrome, @@ -366,9 +366,9 @@ static void raid6_sse24_gen_syndrome(int disks, size_t bytes, void **ptrs) kernel_fpu_end(); } - static void raid6_sse24_xor_syndrome(int disks, int start, int stop, +static void raid6_sse24_xor_syndrome(int disks, int start, int stop, size_t bytes, void **ptrs) - { +{ u8 **dptr = (u8 **)ptrs; u8 *p, *q; int d, z, z0; @@ -471,7 +471,7 @@ static void raid6_sse24_gen_syndrome(int disks, size_t bytes, void **ptrs) } asm volatile("sfence" : : : "memory"); kernel_fpu_end(); - } +} const struct raid6_calls raid6_sse2x4 = { diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index fce2010d3c53..78871de35086 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -886,7 +886,7 @@ static const struct snd_pcm_ops fsl_dma_ops = { }; static int fsl_soc_dma_probe(struct platform_device *pdev) - { +{ struct dma_object *dma; struct device_node *np = pdev->dev.of_node; struct device_node *ssi_np; -- 2.15.0

5 4

[alsa-devel] [PATCH] ALSA: aloop: Fix access to not-yet-ready substream via cable
by Takashi Iwai 22 Mar '18

22 Mar '18

In loopback_open() and loopback_close(), we assign and release the substream object to the corresponding cable in a racy way. It's neither locked nor done in the right position. The open callback assigns the substream before its preparation finishes, hence the other side of the cable may pick it up, which may lead to the invalid memory access. This patch addresses these: move the assignment to the end of the open callback, and wrap with cable->lock for avoiding concurrent accesses. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/drivers/aloop.c | 8 +++++++- 1 file changed, 7 insertions(+), 1 deletion(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 0a08e63e9c66..1063a4377502 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -666,7 +666,9 @@ static void free_cable(struct snd_pcm_substream *substream) return; if (cable->streams[!substream->stream]) { /* other stream is still alive */ + spin_lock_irq(&cable->lock); cable->streams[substream->stream] = NULL; + spin_unlock_irq(&cable->lock); } else { /* free the cable */ loopback->cables[substream->number][dev] = NULL; @@ -705,7 +707,6 @@ static int loopback_open(struct snd_pcm_substream *substream) loopback->cables[substream->number][dev] = cable; } dpcm->cable = cable; - cable->streams[substream->stream] = dpcm; snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); @@ -737,6 +738,11 @@ static int loopback_open(struct snd_pcm_substream *substream) runtime->hw = loopback_pcm_hardware; else runtime->hw = cable->hw; + + spin_lock_irq(&cable->lock); + cable->streams[substream->stream] = dpcm; + spin_unlock_irq(&cable->lock); + unlock: if (err < 0) { free_cable(substream); -- 2.16.2

1 0

[alsa-devel] [ThinkPad] speaker no sound after system resume
by Kailang 22 Mar '18

22 Mar '18

Hi Takashi, Attach patch will fix ThinkPad T580 family for resume issue. It will have a chance speaker no sound after system resume. To toggle NID 0x53 index 0x2 bit 15 will solve this issue. This usage will also suitable with ALC256. This patch also tested by Ubuntu QA. Many Thanks. BR, Kailang

2 1

[alsa-devel] Dell desktop headset support
by Kailang 22 Mar '18

22 Mar '18

Hi Takashi, Attach patch was fixup for Dell desktop. This platform was hardware fixed type for CTIA type for headset port. Assigned 0x19 verb will fix can't record issue. This patch was tested by Ubuntu QA. Thanks. BR, Kailang

2 1

[alsa-devel] [PATCH] ALSA: aloop: Sync stale timer before release
by Takashi Iwai 22 Mar '18

22 Mar '18

The aloop driver tries to stop the pending timer via timer_del() in the trigger callback and in the close callback. The former is correct, as it's an atomic operation, while the latter expects that the timer gets really removed and proceeds the resource releases after that. But timer_del() doesn't synchronize, hence the running timer may still access the released resources. A similar situation can be also seen in the prepare callback after trigger(STOP) where the prepare tries to re-initialize the things while a timer is still running. The problems like the above are seen indirectly in some syzkaller reports (although it's not 100% clear whether this is the only cause, as the race condition is quite narrow and not always easy to trigger). For addressing these issues, this patch adds the explicit alls of timer_del_sync() in some places, so that the pending timer is properly killed / synced. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/drivers/aloop.c | 9 ++++++++- 1 file changed, 8 insertions(+), 1 deletion(-) diff --git a/sound/drivers/aloop.c b/sound/drivers/aloop.c index 0333143a1fa7..0a08e63e9c66 100644 --- a/sound/drivers/aloop.c +++ b/sound/drivers/aloop.c @@ -192,6 +192,11 @@ static inline void loopback_timer_stop(struct loopback_pcm *dpcm) dpcm->timer.expires = 0; } +static inline void loopback_timer_stop_sync(struct loopback_pcm *dpcm) +{ + del_timer_sync(&dpcm->timer); +} + #define CABLE_VALID_PLAYBACK (1 << SNDRV_PCM_STREAM_PLAYBACK) #define CABLE_VALID_CAPTURE (1 << SNDRV_PCM_STREAM_CAPTURE) #define CABLE_VALID_BOTH (CABLE_VALID_PLAYBACK|CABLE_VALID_CAPTURE) @@ -326,6 +331,8 @@ static int loopback_prepare(struct snd_pcm_substream *substream) struct loopback_cable *cable = dpcm->cable; int bps, salign; + loopback_timer_stop_sync(dpcm); + salign = (snd_pcm_format_width(runtime->format) * runtime->channels) / 8; bps = salign * runtime->rate; @@ -744,7 +751,7 @@ static int loopback_close(struct snd_pcm_substream *substream) struct loopback *loopback = substream->private_data; struct loopback_pcm *dpcm = substream->runtime->private_data; - loopback_timer_stop(dpcm); + loopback_timer_stop_sync(dpcm); mutex_lock(&loopback->cable_lock); free_cable(substream); mutex_unlock(&loopback->cable_lock); -- 2.16.2

1 0

[alsa-devel] [PATCH] ALSA: hda - Force polling mode on CFL for fixing codec communication
by Takashi Iwai 22 Mar '18

22 Mar '18

We've observed too long probe time with Coffee Lake machines, and the likely cause is some communication problem between the HD-audio controller and the codec chips. While the controller expects an IRQ wakeup for each codec response, it seems sometimes missing, and it takes one second for the controller driver to time out and read the response in the polling mode. Although we aren't sure about the real culprit yet, in this patch, we put a workaround by forcing the polling mode as default for CFL machines; the polling mode itself isn't too heavy, and much better than other workarounds initially suggested (e.g. disabling power-save), at least. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=199007 Fixes: e79b0006c45c ("ALSA: hda - Add Coffelake PCI ID") Reported-and-tested-by: Hui Wang <hui.wang(a)canonical.com> Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 5 +++++ 1 file changed, 5 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index d5017adf9feb..c507c69029e3 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -375,6 +375,7 @@ enum { ((pci)->device == 0x160c)) #define IS_BXT(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0x5a98) +#define IS_CFL(pci) ((pci)->vendor == 0x8086 && (pci)->device == 0xa348) static char *driver_short_names[] = { [AZX_DRIVER_ICH] = "HDA Intel", @@ -1744,6 +1745,10 @@ static int azx_create(struct snd_card *card, struct pci_dev *pci, else chip->bdl_pos_adj = bdl_pos_adj[dev]; + /* Workaround for a communication error on CFL (bko#199007) */ + if (IS_CFL(pci)) + chip->polling_mode = 1; + err = azx_bus_init(chip, model[dev], &pci_hda_io_ops); if (err < 0) { kfree(hda); -- 2.16.2

3 5

[alsa-devel] [PATCH] ASoC: rt5668: add rt5668B codec driver
by Bard Liao 22 Mar '18

22 Mar '18

This is the initial codec driver for rt5668b. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- Documentation/devicetree/bindings/sound/rt5668.txt | 50 + include/sound/rt5668.h | 40 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/rt5668.c | 2639 ++++++++++++++++++++ sound/soc/codecs/rt5668.h | 1318 ++++++++++ 6 files changed, 4055 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/rt5668.txt create mode 100644 include/sound/rt5668.h create mode 100644 sound/soc/codecs/rt5668.c create mode 100644 sound/soc/codecs/rt5668.h diff --git a/Documentation/devicetree/bindings/sound/rt5668.txt b/Documentation/devicetree/bindings/sound/rt5668.txt new file mode 100644 index 0000000..c88b96e --- /dev/null +++ b/Documentation/devicetree/bindings/sound/rt5668.txt @@ -0,0 +1,50 @@ +RT5668B audio CODEC + +This device supports I2C only. + +Required properties: + +- compatible : "realtek,rt5668b" + +- reg : The I2C address of the device. + +Optional properties: + +- interrupts : The CODEC's interrupt output. + +- realtek,dmic1-data-pin + 0: dmic1 is not used + 1: using GPIO2 pin as dmic1 data pin + 2: using GPIO5 pin as dmic1 data pin + +- realtek,dmic1-clk-pin + 0: using GPIO1 pin as dmic1 clock pin + 1: using GPIO3 pin as dmic1 clock pin + +- realtek,jd-src + 0: No JD is used + 1: using JD1 as JD source + +- realtek,ldo1-en-gpios : The GPIO that controls the CODEC's LDO1_EN pin. + +Pins on the device (for linking into audio routes) for RT5668B: + + * DMIC L1 + * DMIC R1 + * IN1P + * HPOL + * HPOR + +Example: + +rt5668 { + compatible = "realtek,rt5668b"; + reg = <0x1a>; + interrupt-parent = <&gpio>; + interrupts = <TEGRA_GPIO(U, 6) GPIO_ACTIVE_HIGH>; + realtek,ldo1-en-gpios = + <&gpio TEGRA_GPIO(R, 2) GPIO_ACTIVE_HIGH>; + realtek,dmic1-data-pin = <1>; + realtek,dmic1-clk-pin = <1>; + realtek,jd-src = <1>; +}; diff --git a/include/sound/rt5668.h b/include/sound/rt5668.h new file mode 100644 index 0000000..f907b78 --- /dev/null +++ b/include/sound/rt5668.h @@ -0,0 +1,40 @@ +/* + * linux/sound/rt5668.h -- Platform data for RT5668 + * + * Copyright 2018 Realtek Microelectronics + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __LINUX_SND_RT5668_H +#define __LINUX_SND_RT5668_H + +enum rt5668_dmic1_data_pin { + RT5668_DMIC1_NULL, + RT5668_DMIC1_DATA_GPIO2, + RT5668_DMIC1_DATA_GPIO5, +}; + +enum rt5668_dmic1_clk_pin { + RT5668_DMIC1_CLK_GPIO1, + RT5668_DMIC1_CLK_GPIO3, +}; + +enum rt5668_jd_src { + RT5668_JD_NULL, + RT5668_JD1, +}; + +struct rt5668_platform_data { + + int ldo1_en; /* GPIO for LDO1_EN */ + + enum rt5668_dmic1_data_pin dmic1_data_pin; + enum rt5668_dmic1_clk_pin dmic1_clk_pin; + enum rt5668_jd_src jd_src; +}; + +#endif + diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 822df8d..faeb4aa 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -135,6 +135,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_RT5660 if I2C select SND_SOC_RT5663 if I2C select SND_SOC_RT5665 if I2C + select SND_SOC_RT5668 if I2C select SND_SOC_RT5670 if I2C select SND_SOC_RT5677 if I2C && SPI_MASTER select SND_SOC_SGTL5000 if I2C @@ -757,6 +758,7 @@ config SND_SOC_RL6231 default y if SND_SOC_RT5660=y default y if SND_SOC_RT5663=y default y if SND_SOC_RT5665=y + default y if SND_SOC_RT5668=y default y if SND_SOC_RT5670=y default y if SND_SOC_RT5677=y default m if SND_SOC_RT5514=m @@ -768,6 +770,7 @@ config SND_SOC_RL6231 default m if SND_SOC_RT5660=m default m if SND_SOC_RT5663=m default m if SND_SOC_RT5665=m + default m if SND_SOC_RT5668=m default m if SND_SOC_RT5670=m default m if SND_SOC_RT5677=m @@ -831,6 +834,9 @@ config SND_SOC_RT5663 config SND_SOC_RT5665 tristate +config SND_SOC_RT5668 + tristate + config SND_SOC_RT5670 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index bc9425b..2bdf8e4 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -138,6 +138,7 @@ snd-soc-rt5659-objs := rt5659.o snd-soc-rt5660-objs := rt5660.o snd-soc-rt5663-objs := rt5663.o snd-soc-rt5665-objs := rt5665.o +snd-soc-rt5668-objs := rt5668.o snd-soc-rt5670-objs := rt5670.o snd-soc-rt5677-objs := rt5677.o snd-soc-rt5677-spi-objs := rt5677-spi.o @@ -389,6 +390,7 @@ obj-$(CONFIG_SND_SOC_RT5659) += snd-soc-rt5659.o obj-$(CONFIG_SND_SOC_RT5660) += snd-soc-rt5660.o obj-$(CONFIG_SND_SOC_RT5663) += snd-soc-rt5663.o obj-$(CONFIG_SND_SOC_RT5665) += snd-soc-rt5665.o +obj-$(CONFIG_SND_SOC_RT5668) += snd-soc-rt5668.o obj-$(CONFIG_SND_SOC_RT5670) += snd-soc-rt5670.o obj-$(CONFIG_SND_SOC_RT5677) += snd-soc-rt5677.o obj-$(CONFIG_SND_SOC_RT5677_SPI) += snd-soc-rt5677-spi.o diff --git a/sound/soc/codecs/rt5668.c b/sound/soc/codecs/rt5668.c new file mode 100644 index 0000000..52a343f --- /dev/null +++ b/sound/soc/codecs/rt5668.c @@ -0,0 +1,2639 @@ +/* + * rt5668.c -- RT5668B ALSA SoC audio component driver + * + * Copyright 2018 Realtek Semiconductor Corp. + * Author: Bard Liao <bardliao(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/init.h> +#include <linux/delay.h> +#include <linux/pm.h> +#include <linux/i2c.h> +#include <linux/platform_device.h> +#include <linux/spi/spi.h> +#include <linux/acpi.h> +#include <linux/gpio.h> +#include <linux/of_gpio.h> +#include <linux/regulator/consumer.h> +#include <linux/mutex.h> +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/jack.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> +#include <sound/rt5668.h> + +#include "rl6231.h" +#include "rt5668.h" + +#define RT5668_NUM_SUPPLIES 3 + +static const char *rt5668_supply_names[RT5668_NUM_SUPPLIES] = { + "AVDD", + "MICVDD", + "VBAT", +}; + +struct rt5668_priv { + struct snd_soc_component *component; + struct rt5668_platform_data pdata; + struct regmap *regmap; + struct snd_soc_jack *hs_jack; + struct regulator_bulk_data supplies[RT5668_NUM_SUPPLIES]; + struct delayed_work jack_detect_work; + struct delayed_work jd_check_work; + struct mutex calibrate_mutex; + + int sysclk; + int sysclk_src; + int lrck[RT5668_AIFS]; + int bclk[RT5668_AIFS]; + int master[RT5668_AIFS]; + + int pll_src; + int pll_in; + int pll_out; + + int jack_type; +}; + +static const struct reg_default rt5668_reg[] = { + {0x0002, 0x8080}, + {0x0003, 0x8000}, + {0x0005, 0x0000}, + {0x0006, 0x0000}, + {0x0008, 0x800f}, + {0x000b, 0x0000}, + {0x0010, 0x4040}, + {0x0011, 0x0000}, + {0x0012, 0x1404}, + {0x0013, 0x1000}, + {0x0014, 0xa00a}, + {0x0015, 0x0404}, + {0x0016, 0x0404}, + {0x0019, 0xafaf}, + {0x001c, 0x2f2f}, + {0x001f, 0x0000}, + {0x0022, 0x5757}, + {0x0023, 0x0039}, + {0x0024, 0x000b}, + {0x0026, 0xc0c4}, + {0x0029, 0x8080}, + {0x002a, 0xa0a0}, + {0x002b, 0x0300}, + {0x0030, 0x0000}, + {0x003c, 0x0080}, + {0x0044, 0x0c0c}, + {0x0049, 0x0000}, + {0x0061, 0x0000}, + {0x0062, 0x0000}, + {0x0063, 0x003f}, + {0x0064, 0x0000}, + {0x0065, 0x0000}, + {0x0066, 0x0030}, + {0x0067, 0x0000}, + {0x006b, 0x0000}, + {0x006c, 0x0000}, + {0x006d, 0x2200}, + {0x006e, 0x0a10}, + {0x0070, 0x8000}, + {0x0071, 0x8000}, + {0x0073, 0x0000}, + {0x0074, 0x0000}, + {0x0075, 0x0002}, + {0x0076, 0x0001}, + {0x0079, 0x0000}, + {0x007a, 0x0000}, + {0x007b, 0x0000}, + {0x007c, 0x0100}, + {0x007e, 0x0000}, + {0x0080, 0x0000}, + {0x0081, 0x0000}, + {0x0082, 0x0000}, + {0x0083, 0x0000}, + {0x0084, 0x0000}, + {0x0085, 0x0000}, + {0x0086, 0x0005}, + {0x0087, 0x0000}, + {0x0088, 0x0000}, + {0x008c, 0x0003}, + {0x008d, 0x0000}, + {0x008e, 0x0060}, + {0x008f, 0x1000}, + {0x0091, 0x0c26}, + {0x0092, 0x0073}, + {0x0093, 0x0000}, + {0x0094, 0x0080}, + {0x0098, 0x0000}, + {0x009a, 0x0000}, + {0x009b, 0x0000}, + {0x009c, 0x0000}, + {0x009d, 0x0000}, + {0x009e, 0x100c}, + {0x009f, 0x0000}, + {0x00a0, 0x0000}, + {0x00a3, 0x0002}, + {0x00a4, 0x0001}, + {0x00ae, 0x2040}, + {0x00af, 0x0000}, + {0x00b6, 0x0000}, + {0x00b7, 0x0000}, + {0x00b8, 0x0000}, + {0x00b9, 0x0002}, + {0x00be, 0x0000}, + {0x00c0, 0x0160}, + {0x00c1, 0x82a0}, + {0x00c2, 0x0000}, + {0x00d0, 0x0000}, + {0x00d1, 0x2244}, + {0x00d2, 0x3300}, + {0x00d3, 0x2200}, + {0x00d4, 0x0000}, + {0x00d9, 0x0009}, + {0x00da, 0x0000}, + {0x00db, 0x0000}, + {0x00dc, 0x00c0}, + {0x00dd, 0x2220}, + {0x00de, 0x3131}, + {0x00df, 0x3131}, + {0x00e0, 0x3131}, + {0x00e2, 0x0000}, + {0x00e3, 0x4000}, + {0x00e4, 0x0aa0}, + {0x00e5, 0x3131}, + {0x00e6, 0x3131}, + {0x00e7, 0x3131}, + {0x00e8, 0x3131}, + {0x00ea, 0xb320}, + {0x00eb, 0x0000}, + {0x00f0, 0x0000}, + {0x00f1, 0x00d0}, + {0x00f2, 0x00d0}, + {0x00f6, 0x0000}, + {0x00fa, 0x0000}, + {0x00fb, 0x0000}, + {0x00fc, 0x0000}, + {0x00fd, 0x0000}, + {0x00fe, 0x10ec}, + {0x00ff, 0x6530}, + {0x0100, 0xa0a0}, + {0x010b, 0x0000}, + {0x010c, 0xae00}, + {0x010d, 0xaaa0}, + {0x010e, 0x8aa2}, + {0x010f, 0x02a2}, + {0x0110, 0xc000}, + {0x0111, 0x04a2}, + {0x0112, 0x2800}, + {0x0113, 0x0000}, + {0x0117, 0x0100}, + {0x0125, 0x0410}, + {0x0132, 0x6026}, + {0x0136, 0x5555}, + {0x0138, 0x3700}, + {0x013a, 0x2000}, + {0x013b, 0x2000}, + {0x013c, 0x2005}, + {0x013f, 0x0000}, + {0x0142, 0x0000}, + {0x0145, 0x0002}, + {0x0146, 0x0000}, + {0x0147, 0x0000}, + {0x0148, 0x0000}, + {0x0149, 0x0000}, + {0x0150, 0x79a1}, + {0x0151, 0x0000}, + {0x0160, 0x4ec0}, + {0x0161, 0x0080}, + {0x0162, 0x0200}, + {0x0163, 0x0800}, + {0x0164, 0x0000}, + {0x0165, 0x0000}, + {0x0166, 0x0000}, + {0x0167, 0x000f}, + {0x0168, 0x000f}, + {0x0169, 0x0021}, + {0x0190, 0x413d}, + {0x0194, 0x0000}, + {0x0195, 0x0000}, + {0x0197, 0x0022}, + {0x0198, 0x0000}, + {0x0199, 0x0000}, + {0x01af, 0x0000}, + {0x01b0, 0x0400}, + {0x01b1, 0x0000}, + {0x01b2, 0x0000}, + {0x01b3, 0x0000}, + {0x01b4, 0x0000}, + {0x01b5, 0x0000}, + {0x01b6, 0x01c3}, + {0x01b7, 0x02a0}, + {0x01b8, 0x03e9}, + {0x01b9, 0x1389}, + {0x01ba, 0xc351}, + {0x01bb, 0x0009}, + {0x01bc, 0x0018}, + {0x01bd, 0x002a}, + {0x01be, 0x004c}, + {0x01bf, 0x0097}, + {0x01c0, 0x433d}, + {0x01c1, 0x2800}, + {0x01c2, 0x0000}, + {0x01c3, 0x0000}, + {0x01c4, 0x0000}, + {0x01c5, 0x0000}, + {0x01c6, 0x0000}, + {0x01c7, 0x0000}, + {0x01c8, 0x40af}, + {0x01c9, 0x0702}, + {0x01ca, 0x0000}, + {0x01cb, 0x0000}, + {0x01cc, 0x5757}, + {0x01cd, 0x5757}, + {0x01ce, 0x5757}, + {0x01cf, 0x5757}, + {0x01d0, 0x5757}, + {0x01d1, 0x5757}, + {0x01d2, 0x5757}, + {0x01d3, 0x5757}, + {0x01d4, 0x5757}, + {0x01d5, 0x5757}, + {0x01d6, 0x0000}, + {0x01d7, 0x0008}, + {0x01d8, 0x0029}, + {0x01d9, 0x3333}, + {0x01da, 0x0000}, + {0x01db, 0x0004}, + {0x01dc, 0x0000}, + {0x01de, 0x7c00}, + {0x01df, 0x0320}, + {0x01e0, 0x06a1}, + {0x01e1, 0x0000}, + {0x01e2, 0x0000}, + {0x01e3, 0x0000}, + {0x01e4, 0x0000}, + {0x01e6, 0x0001}, + {0x01e7, 0x0000}, + {0x01e8, 0x0000}, + {0x01ea, 0x0000}, + {0x01eb, 0x0000}, + {0x01ec, 0x0000}, + {0x01ed, 0x0000}, + {0x01ee, 0x0000}, + {0x01ef, 0x0000}, + {0x01f0, 0x0000}, + {0x01f1, 0x0000}, + {0x01f2, 0x0000}, + {0x01f3, 0x0000}, + {0x01f4, 0x0000}, + {0x0210, 0x6297}, + {0x0211, 0xa005}, + {0x0212, 0x824c}, + {0x0213, 0xf7ff}, + {0x0214, 0xf24c}, + {0x0215, 0x0102}, + {0x0216, 0x00a3}, + {0x0217, 0x0048}, + {0x0218, 0xa2c0}, + {0x0219, 0x0400}, + {0x021a, 0x00c8}, + {0x021b, 0x00c0}, + {0x021c, 0x0000}, + {0x0250, 0x4500}, + {0x0251, 0x40b3}, + {0x0252, 0x0000}, + {0x0253, 0x0000}, + {0x0254, 0x0000}, + {0x0255, 0x0000}, + {0x0256, 0x0000}, + {0x0257, 0x0000}, + {0x0258, 0x0000}, + {0x0259, 0x0000}, + {0x025a, 0x0005}, + {0x0270, 0x0000}, + {0x02ff, 0x0110}, + {0x0300, 0x001f}, + {0x0301, 0x032c}, + {0x0302, 0x5f21}, + {0x0303, 0x4000}, + {0x0304, 0x4000}, + {0x0305, 0x06d5}, + {0x0306, 0x8000}, + {0x0307, 0x0700}, + {0x0310, 0x4560}, + {0x0311, 0xa4a8}, + {0x0312, 0x7418}, + {0x0313, 0x0000}, + {0x0314, 0x0006}, + {0x0315, 0xffff}, + {0x0316, 0xc400}, + {0x0317, 0x0000}, + {0x03c0, 0x7e00}, + {0x03c1, 0x8000}, + {0x03c2, 0x8000}, + {0x03c3, 0x8000}, + {0x03c4, 0x8000}, + {0x03c5, 0x8000}, + {0x03c6, 0x8000}, + {0x03c7, 0x8000}, + {0x03c8, 0x8000}, + {0x03c9, 0x8000}, + {0x03ca, 0x8000}, + {0x03cb, 0x8000}, + {0x03cc, 0x8000}, + {0x03d0, 0x0000}, + {0x03d1, 0x0000}, + {0x03d2, 0x0000}, + {0x03d3, 0x0000}, + {0x03d4, 0x2000}, + {0x03d5, 0x2000}, + {0x03d6, 0x0000}, + {0x03d7, 0x0000}, + {0x03d8, 0x2000}, + {0x03d9, 0x2000}, + {0x03da, 0x2000}, + {0x03db, 0x2000}, + {0x03dc, 0x0000}, + {0x03dd, 0x0000}, + {0x03de, 0x0000}, + {0x03df, 0x2000}, + {0x03e0, 0x0000}, + {0x03e1, 0x0000}, + {0x03e2, 0x0000}, + {0x03e3, 0x0000}, + {0x03e4, 0x0000}, + {0x03e5, 0x0000}, + {0x03e6, 0x0000}, + {0x03e7, 0x0000}, + {0x03e8, 0x0000}, + {0x03e9, 0x0000}, + {0x03ea, 0x0000}, + {0x03eb, 0x0000}, + {0x03ec, 0x0000}, + {0x03ed, 0x0000}, + {0x03ee, 0x0000}, + {0x03ef, 0x0000}, + {0x03f0, 0x0800}, + {0x03f1, 0x0800}, + {0x03f2, 0x0800}, + {0x03f3, 0x0800}, +}; + +static bool rt5668_volatile_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT5668_RESET: + case RT5668_CBJ_CTRL_2: + case RT5668_INT_ST_1: + case RT5668_4BTN_IL_CMD_1: + case RT5668_AJD1_CTRL: + case RT5668_HP_CALIB_CTRL_1: + case RT5668_DEVICE_ID: + case RT5668_I2C_MODE: + case RT5668_HP_CALIB_CTRL_10: + case RT5668_EFUSE_CTRL_2: + case RT5668_JD_TOP_VC_VTRL: + case RT5668_HP_IMP_SENS_CTRL_19: + case RT5668_IL_CMD_1: + case RT5668_SAR_IL_CMD_2: + case RT5668_SAR_IL_CMD_4: + case RT5668_SAR_IL_CMD_10: + case RT5668_SAR_IL_CMD_11: + case RT5668_EFUSE_CTRL_6...RT5668_EFUSE_CTRL_11: + case RT5668_HP_CALIB_STA_1...RT5668_HP_CALIB_STA_11: + return true; + default: + return false; + } +} + +static bool rt5668_readable_register(struct device *dev, unsigned int reg) +{ + switch (reg) { + case RT5668_RESET: + case RT5668_VERSION_ID: + case RT5668_VENDOR_ID: + case RT5668_DEVICE_ID: + case RT5668_HP_CTRL_1: + case RT5668_HP_CTRL_2: + case RT5668_HPL_GAIN: + case RT5668_HPR_GAIN: + case RT5668_I2C_CTRL: + case RT5668_CBJ_BST_CTRL: + case RT5668_CBJ_CTRL_1: + case RT5668_CBJ_CTRL_2: + case RT5668_CBJ_CTRL_3: + case RT5668_CBJ_CTRL_4: + case RT5668_CBJ_CTRL_5: + case RT5668_CBJ_CTRL_6: + case RT5668_CBJ_CTRL_7: + case RT5668_DAC1_DIG_VOL: + case RT5668_STO1_ADC_DIG_VOL: + case RT5668_STO1_ADC_BOOST: + case RT5668_HP_IMP_GAIN_1: + case RT5668_HP_IMP_GAIN_2: + case RT5668_SIDETONE_CTRL: + case RT5668_STO1_ADC_MIXER: + case RT5668_AD_DA_MIXER: + case RT5668_STO1_DAC_MIXER: + case RT5668_A_DAC1_MUX: + case RT5668_DIG_INF2_DATA: + case RT5668_REC_MIXER: + case RT5668_CAL_REC: + case RT5668_ALC_BACK_GAIN: + case RT5668_PWR_DIG_1: + case RT5668_PWR_DIG_2: + case RT5668_PWR_ANLG_1: + case RT5668_PWR_ANLG_2: + case RT5668_PWR_ANLG_3: + case RT5668_PWR_MIXER: + case RT5668_PWR_VOL: + case RT5668_CLK_DET: + case RT5668_RESET_LPF_CTRL: + case RT5668_RESET_HPF_CTRL: + case RT5668_DMIC_CTRL_1: + case RT5668_I2S1_SDP: + case RT5668_I2S2_SDP: + case RT5668_ADDA_CLK_1: + case RT5668_ADDA_CLK_2: + case RT5668_I2S1_F_DIV_CTRL_1: + case RT5668_I2S1_F_DIV_CTRL_2: + case RT5668_TDM_CTRL: + case RT5668_TDM_ADDA_CTRL_1: + case RT5668_TDM_ADDA_CTRL_2: + case RT5668_DATA_SEL_CTRL_1: + case RT5668_TDM_TCON_CTRL: + case RT5668_GLB_CLK: + case RT5668_PLL_CTRL_1: + case RT5668_PLL_CTRL_2: + case RT5668_PLL_TRACK_1: + case RT5668_PLL_TRACK_2: + case RT5668_PLL_TRACK_3: + case RT5668_PLL_TRACK_4: + case RT5668_PLL_TRACK_5: + case RT5668_PLL_TRACK_6: + case RT5668_PLL_TRACK_11: + case RT5668_SDW_REF_CLK: + case RT5668_DEPOP_1: + case RT5668_DEPOP_2: + case RT5668_HP_CHARGE_PUMP_1: + case RT5668_HP_CHARGE_PUMP_2: + case RT5668_MICBIAS_1: + case RT5668_MICBIAS_2: + case RT5668_PLL_TRACK_12: + case RT5668_PLL_TRACK_14: + case RT5668_PLL2_CTRL_1: + case RT5668_PLL2_CTRL_2: + case RT5668_PLL2_CTRL_3: + case RT5668_PLL2_CTRL_4: + case RT5668_RC_CLK_CTRL: + case RT5668_I2S_M_CLK_CTRL_1: + case RT5668_I2S2_F_DIV_CTRL_1: + case RT5668_I2S2_F_DIV_CTRL_2: + case RT5668_EQ_CTRL_1: + case RT5668_EQ_CTRL_2: + case RT5668_IRQ_CTRL_1: + case RT5668_IRQ_CTRL_2: + case RT5668_IRQ_CTRL_3: + case RT5668_IRQ_CTRL_4: + case RT5668_INT_ST_1: + case RT5668_GPIO_CTRL_1: + case RT5668_GPIO_CTRL_2: + case RT5668_GPIO_CTRL_3: + case RT5668_HP_AMP_DET_CTRL_1: + case RT5668_HP_AMP_DET_CTRL_2: + case RT5668_MID_HP_AMP_DET: + case RT5668_LOW_HP_AMP_DET: + case RT5668_DELAY_BUF_CTRL: + case RT5668_SV_ZCD_1: + case RT5668_SV_ZCD_2: + case RT5668_IL_CMD_1: + case RT5668_IL_CMD_2: + case RT5668_IL_CMD_3: + case RT5668_IL_CMD_4: + case RT5668_IL_CMD_5: + case RT5668_IL_CMD_6: + case RT5668_4BTN_IL_CMD_1: + case RT5668_4BTN_IL_CMD_2: + case RT5668_4BTN_IL_CMD_3: + case RT5668_4BTN_IL_CMD_4: + case RT5668_4BTN_IL_CMD_5: + case RT5668_4BTN_IL_CMD_6: + case RT5668_4BTN_IL_CMD_7: + case RT5668_ADC_STO1_HP_CTRL_1: + case RT5668_ADC_STO1_HP_CTRL_2: + case RT5668_AJD1_CTRL: + case RT5668_JD1_THD: + case RT5668_JD2_THD: + case RT5668_JD_CTRL_1: + case RT5668_DUMMY_1: + case RT5668_DUMMY_2: + case RT5668_DUMMY_3: + case RT5668_DAC_ADC_DIG_VOL1: + case RT5668_BIAS_CUR_CTRL_2: + case RT5668_BIAS_CUR_CTRL_3: + case RT5668_BIAS_CUR_CTRL_4: + case RT5668_BIAS_CUR_CTRL_5: + case RT5668_BIAS_CUR_CTRL_6: + case RT5668_BIAS_CUR_CTRL_7: + case RT5668_BIAS_CUR_CTRL_8: + case RT5668_BIAS_CUR_CTRL_9: + case RT5668_BIAS_CUR_CTRL_10: + case RT5668_VREF_REC_OP_FB_CAP_CTRL: + case RT5668_CHARGE_PUMP_1: + case RT5668_DIG_IN_CTRL_1: + case RT5668_PAD_DRIVING_CTRL: + case RT5668_SOFT_RAMP_DEPOP: + case RT5668_CHOP_DAC: + case RT5668_CHOP_ADC: + case RT5668_CALIB_ADC_CTRL: + case RT5668_VOL_TEST: + case RT5668_SPKVDD_DET_STA: + case RT5668_TEST_MODE_CTRL_1: + case RT5668_TEST_MODE_CTRL_2: + case RT5668_TEST_MODE_CTRL_3: + case RT5668_TEST_MODE_CTRL_4: + case RT5668_TEST_MODE_CTRL_5: + case RT5668_PLL1_INTERNAL: + case RT5668_PLL2_INTERNAL: + case RT5668_STO_NG2_CTRL_1: + case RT5668_STO_NG2_CTRL_2: + case RT5668_STO_NG2_CTRL_3: + case RT5668_STO_NG2_CTRL_4: + case RT5668_STO_NG2_CTRL_5: + case RT5668_STO_NG2_CTRL_6: + case RT5668_STO_NG2_CTRL_7: + case RT5668_STO_NG2_CTRL_8: + case RT5668_STO_NG2_CTRL_9: + case RT5668_STO_NG2_CTRL_10: + case RT5668_STO1_DAC_SIL_DET: + case RT5668_SIL_PSV_CTRL1: + case RT5668_SIL_PSV_CTRL2: + case RT5668_SIL_PSV_CTRL3: + case RT5668_SIL_PSV_CTRL4: + case RT5668_SIL_PSV_CTRL5: + case RT5668_HP_IMP_SENS_CTRL_01: + case RT5668_HP_IMP_SENS_CTRL_02: + case RT5668_HP_IMP_SENS_CTRL_03: + case RT5668_HP_IMP_SENS_CTRL_04: + case RT5668_HP_IMP_SENS_CTRL_05: + case RT5668_HP_IMP_SENS_CTRL_06: + case RT5668_HP_IMP_SENS_CTRL_07: + case RT5668_HP_IMP_SENS_CTRL_08: + case RT5668_HP_IMP_SENS_CTRL_09: + case RT5668_HP_IMP_SENS_CTRL_10: + case RT5668_HP_IMP_SENS_CTRL_11: + case RT5668_HP_IMP_SENS_CTRL_12: + case RT5668_HP_IMP_SENS_CTRL_13: + case RT5668_HP_IMP_SENS_CTRL_14: + case RT5668_HP_IMP_SENS_CTRL_15: + case RT5668_HP_IMP_SENS_CTRL_16: + case RT5668_HP_IMP_SENS_CTRL_17: + case RT5668_HP_IMP_SENS_CTRL_18: + case RT5668_HP_IMP_SENS_CTRL_19: + case RT5668_HP_IMP_SENS_CTRL_20: + case RT5668_HP_IMP_SENS_CTRL_21: + case RT5668_HP_IMP_SENS_CTRL_22: + case RT5668_HP_IMP_SENS_CTRL_23: + case RT5668_HP_IMP_SENS_CTRL_24: + case RT5668_HP_IMP_SENS_CTRL_25: + case RT5668_HP_IMP_SENS_CTRL_26: + case RT5668_HP_IMP_SENS_CTRL_27: + case RT5668_HP_IMP_SENS_CTRL_28: + case RT5668_HP_IMP_SENS_CTRL_29: + case RT5668_HP_IMP_SENS_CTRL_30: + case RT5668_HP_IMP_SENS_CTRL_31: + case RT5668_HP_IMP_SENS_CTRL_32: + case RT5668_HP_IMP_SENS_CTRL_33: + case RT5668_HP_IMP_SENS_CTRL_34: + case RT5668_HP_IMP_SENS_CTRL_35: + case RT5668_HP_IMP_SENS_CTRL_36: + case RT5668_HP_IMP_SENS_CTRL_37: + case RT5668_HP_IMP_SENS_CTRL_38: + case RT5668_HP_IMP_SENS_CTRL_39: + case RT5668_HP_IMP_SENS_CTRL_40: + case RT5668_HP_IMP_SENS_CTRL_41: + case RT5668_HP_IMP_SENS_CTRL_42: + case RT5668_HP_IMP_SENS_CTRL_43: + case RT5668_HP_LOGIC_CTRL_1: + case RT5668_HP_LOGIC_CTRL_2: + case RT5668_HP_LOGIC_CTRL_3: + case RT5668_HP_CALIB_CTRL_1: + case RT5668_HP_CALIB_CTRL_2: + case RT5668_HP_CALIB_CTRL_3: + case RT5668_HP_CALIB_CTRL_4: + case RT5668_HP_CALIB_CTRL_5: + case RT5668_HP_CALIB_CTRL_6: + case RT5668_HP_CALIB_CTRL_7: + case RT5668_HP_CALIB_CTRL_9: + case RT5668_HP_CALIB_CTRL_10: + case RT5668_HP_CALIB_CTRL_11: + case RT5668_HP_CALIB_STA_1: + case RT5668_HP_CALIB_STA_2: + case RT5668_HP_CALIB_STA_3: + case RT5668_HP_CALIB_STA_4: + case RT5668_HP_CALIB_STA_5: + case RT5668_HP_CALIB_STA_6: + case RT5668_HP_CALIB_STA_7: + case RT5668_HP_CALIB_STA_8: + case RT5668_HP_CALIB_STA_9: + case RT5668_HP_CALIB_STA_10: + case RT5668_HP_CALIB_STA_11: + case RT5668_SAR_IL_CMD_1: + case RT5668_SAR_IL_CMD_2: + case RT5668_SAR_IL_CMD_3: + case RT5668_SAR_IL_CMD_4: + case RT5668_SAR_IL_CMD_5: + case RT5668_SAR_IL_CMD_6: + case RT5668_SAR_IL_CMD_7: + case RT5668_SAR_IL_CMD_8: + case RT5668_SAR_IL_CMD_9: + case RT5668_SAR_IL_CMD_10: + case RT5668_SAR_IL_CMD_11: + case RT5668_SAR_IL_CMD_12: + case RT5668_SAR_IL_CMD_13: + case RT5668_EFUSE_CTRL_1: + case RT5668_EFUSE_CTRL_2: + case RT5668_EFUSE_CTRL_3: + case RT5668_EFUSE_CTRL_4: + case RT5668_EFUSE_CTRL_5: + case RT5668_EFUSE_CTRL_6: + case RT5668_EFUSE_CTRL_7: + case RT5668_EFUSE_CTRL_8: + case RT5668_EFUSE_CTRL_9: + case RT5668_EFUSE_CTRL_10: + case RT5668_EFUSE_CTRL_11: + case RT5668_JD_TOP_VC_VTRL: + case RT5668_DRC1_CTRL_0: + case RT5668_DRC1_CTRL_1: + case RT5668_DRC1_CTRL_2: + case RT5668_DRC1_CTRL_3: + case RT5668_DRC1_CTRL_4: + case RT5668_DRC1_CTRL_5: + case RT5668_DRC1_CTRL_6: + case RT5668_DRC1_HARD_LMT_CTRL_1: + case RT5668_DRC1_HARD_LMT_CTRL_2: + case RT5668_DRC1_PRIV_1: + case RT5668_DRC1_PRIV_2: + case RT5668_DRC1_PRIV_3: + case RT5668_DRC1_PRIV_4: + case RT5668_DRC1_PRIV_5: + case RT5668_DRC1_PRIV_6: + case RT5668_DRC1_PRIV_7: + case RT5668_DRC1_PRIV_8: + case RT5668_EQ_AUTO_RCV_CTRL1: + case RT5668_EQ_AUTO_RCV_CTRL2: + case RT5668_EQ_AUTO_RCV_CTRL3: + case RT5668_EQ_AUTO_RCV_CTRL4: + case RT5668_EQ_AUTO_RCV_CTRL5: + case RT5668_EQ_AUTO_RCV_CTRL6: + case RT5668_EQ_AUTO_RCV_CTRL7: + case RT5668_EQ_AUTO_RCV_CTRL8: + case RT5668_EQ_AUTO_RCV_CTRL9: + case RT5668_EQ_AUTO_RCV_CTRL10: + case RT5668_EQ_AUTO_RCV_CTRL11: + case RT5668_EQ_AUTO_RCV_CTRL12: + case RT5668_EQ_AUTO_RCV_CTRL13: + case RT5668_ADC_L_EQ_LPF1_A1: + case RT5668_R_EQ_LPF1_A1: + case RT5668_L_EQ_LPF1_H0: + case RT5668_R_EQ_LPF1_H0: + case RT5668_L_EQ_BPF1_A1: + case RT5668_R_EQ_BPF1_A1: + case RT5668_L_EQ_BPF1_A2: + case RT5668_R_EQ_BPF1_A2: + case RT5668_L_EQ_BPF1_H0: + case RT5668_R_EQ_BPF1_H0: + case RT5668_L_EQ_BPF2_A1: + case RT5668_R_EQ_BPF2_A1: + case RT5668_L_EQ_BPF2_A2: + case RT5668_R_EQ_BPF2_A2: + case RT5668_L_EQ_BPF2_H0: + case RT5668_R_EQ_BPF2_H0: + case RT5668_L_EQ_BPF3_A1: + case RT5668_R_EQ_BPF3_A1: + case RT5668_L_EQ_BPF3_A2: + case RT5668_R_EQ_BPF3_A2: + case RT5668_L_EQ_BPF3_H0: + case RT5668_R_EQ_BPF3_H0: + case RT5668_L_EQ_BPF4_A1: + case RT5668_R_EQ_BPF4_A1: + case RT5668_L_EQ_BPF4_A2: + case RT5668_R_EQ_BPF4_A2: + case RT5668_L_EQ_BPF4_H0: + case RT5668_R_EQ_BPF4_H0: + case RT5668_L_EQ_HPF1_A1: + case RT5668_R_EQ_HPF1_A1: + case RT5668_L_EQ_HPF1_H0: + case RT5668_R_EQ_HPF1_H0: + case RT5668_L_EQ_PRE_VOL: + case RT5668_R_EQ_PRE_VOL: + case RT5668_L_EQ_POST_VOL: + case RT5668_R_EQ_POST_VOL: + case RT5668_I2C_MODE: + return true; + default: + return false; + } +} + +static const DECLARE_TLV_DB_SCALE(hp_vol_tlv, -2250, 150, 0); +static const DECLARE_TLV_DB_SCALE(dac_vol_tlv, -65625, 375, 0); +static const DECLARE_TLV_DB_SCALE(adc_vol_tlv, -17625, 375, 0); +static const DECLARE_TLV_DB_SCALE(adc_bst_tlv, 0, 1200, 0); + +/* {0, +20, +24, +30, +35, +40, +44, +50, +52} dB */ +static const DECLARE_TLV_DB_RANGE(bst_tlv, + 0, 0, TLV_DB_SCALE_ITEM(0, 0, 0), + 1, 1, TLV_DB_SCALE_ITEM(2000, 0, 0), + 2, 2, TLV_DB_SCALE_ITEM(2400, 0, 0), + 3, 5, TLV_DB_SCALE_ITEM(3000, 500, 0), + 6, 6, TLV_DB_SCALE_ITEM(4400, 0, 0), + 7, 7, TLV_DB_SCALE_ITEM(5000, 0, 0), + 8, 8, TLV_DB_SCALE_ITEM(5200, 0, 0) +); + +/* Interface data select */ +static const char * const rt5668_data_select[] = { + "L/R", "R/L", "L/L", "R/R" +}; + +static SOC_ENUM_SINGLE_DECL(rt5668_if2_adc_enum, + RT5668_DIG_INF2_DATA, RT5668_IF2_ADC_SEL_SFT, rt5668_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5668_if1_01_adc_enum, + RT5668_TDM_ADDA_CTRL_1, RT5668_IF1_ADC1_SEL_SFT, rt5668_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5668_if1_23_adc_enum, + RT5668_TDM_ADDA_CTRL_1, RT5668_IF1_ADC2_SEL_SFT, rt5668_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5668_if1_45_adc_enum, + RT5668_TDM_ADDA_CTRL_1, RT5668_IF1_ADC3_SEL_SFT, rt5668_data_select); + +static SOC_ENUM_SINGLE_DECL(rt5668_if1_67_adc_enum, + RT5668_TDM_ADDA_CTRL_1, RT5668_IF1_ADC4_SEL_SFT, rt5668_data_select); + +static const struct snd_kcontrol_new rt5668_if2_adc_swap_mux = + SOC_DAPM_ENUM("IF2 ADC Swap Mux", rt5668_if2_adc_enum); + +static const struct snd_kcontrol_new rt5668_if1_01_adc_swap_mux = + SOC_DAPM_ENUM("IF1 01 ADC Swap Mux", rt5668_if1_01_adc_enum); + +static const struct snd_kcontrol_new rt5668_if1_23_adc_swap_mux = + SOC_DAPM_ENUM("IF1 23 ADC Swap Mux", rt5668_if1_23_adc_enum); + +static const struct snd_kcontrol_new rt5668_if1_45_adc_swap_mux = + SOC_DAPM_ENUM("IF1 45 ADC Swap Mux", rt5668_if1_45_adc_enum); + +static const struct snd_kcontrol_new rt5668_if1_67_adc_swap_mux = + SOC_DAPM_ENUM("IF1 67 ADC Swap Mux", rt5668_if1_67_adc_enum); + +static void rt5668_reset(struct regmap *regmap) +{ + regmap_write(regmap, RT5668_RESET, 0); + regmap_write(regmap, RT5668_I2C_MODE, 1); +} +/** + * rt5668_sel_asrc_clk_src - select ASRC clock source for a set of filters + * @component: SoC audio component device. + * @filter_mask: mask of filters. + * @clk_src: clock source + * + * The ASRC function is for asynchronous MCLK and LRCK. Also, since RT5668 can + * only support standard 32fs or 64fs i2s format, ASRC should be enabled to + * support special i2s clock format such as Intel's 100fs(100 * sampling rate). + * ASRC function will track i2s clock and generate a corresponding system clock + * for codec. This function provides an API to select the clock source for a + * set of filters specified by the mask. And the component driver will turn on + * ASRC for these filters if ASRC is selected as their clock source. + */ +int rt5668_sel_asrc_clk_src(struct snd_soc_component *component, + unsigned int filter_mask, unsigned int clk_src) +{ + + switch (clk_src) { + case RT5668_CLK_SEL_SYS: + case RT5668_CLK_SEL_I2S1_ASRC: + case RT5668_CLK_SEL_I2S2_ASRC: + break; + + default: + return -EINVAL; + } + + if (filter_mask & RT5668_DA_STEREO1_FILTER) { + snd_soc_component_update_bits(component, RT5668_PLL_TRACK_2, + RT5668_FILTER_CLK_SEL_MASK, + clk_src << RT5668_FILTER_CLK_SEL_SFT); + } + + if (filter_mask & RT5668_AD_STEREO1_FILTER) { + snd_soc_component_update_bits(component, RT5668_PLL_TRACK_3, + RT5668_FILTER_CLK_SEL_MASK, + clk_src << RT5668_FILTER_CLK_SEL_SFT); + } + + return 0; +} +EXPORT_SYMBOL_GPL(rt5668_sel_asrc_clk_src); + +static int rt5668_button_detect(struct snd_soc_component *component) +{ + int btn_type, val; + + val = snd_soc_component_read32(component, RT5668_4BTN_IL_CMD_1); + btn_type = val & 0xfff0; + snd_soc_component_write(component, RT5668_4BTN_IL_CMD_1, val); + pr_debug("%s btn_type=%x\n", __func__, btn_type); + + return btn_type; +} + +static void rt5668_enable_push_button_irq(struct snd_soc_component *component, + bool enable) +{ + if (enable) { + snd_soc_component_update_bits(component, RT5668_SAR_IL_CMD_1, + RT5668_SAR_BUTT_DET_MASK, RT5668_SAR_BUTT_DET_EN); + snd_soc_component_update_bits(component, RT5668_SAR_IL_CMD_13, + RT5668_SAR_SOUR_MASK, RT5668_SAR_SOUR_BTN); + snd_soc_component_write(component, RT5668_IL_CMD_1, 0x0040); + snd_soc_component_update_bits(component, RT5668_4BTN_IL_CMD_2, + RT5668_4BTN_IL_MASK | RT5668_4BTN_IL_RST_MASK, + RT5668_4BTN_IL_EN | RT5668_4BTN_IL_NOR); + snd_soc_component_update_bits(component, RT5668_IRQ_CTRL_3, + RT5668_IL_IRQ_MASK, RT5668_IL_IRQ_EN); + } else { + snd_soc_component_update_bits(component, RT5668_IRQ_CTRL_3, + RT5668_IL_IRQ_MASK, RT5668_IL_IRQ_DIS); + snd_soc_component_update_bits(component, RT5668_SAR_IL_CMD_1, + RT5668_SAR_BUTT_DET_MASK, RT5668_SAR_BUTT_DET_DIS); + snd_soc_component_update_bits(component, RT5668_4BTN_IL_CMD_2, + RT5668_4BTN_IL_MASK, RT5668_4BTN_IL_DIS); + snd_soc_component_update_bits(component, RT5668_4BTN_IL_CMD_2, + RT5668_4BTN_IL_RST_MASK, RT5668_4BTN_IL_RST); + snd_soc_component_update_bits(component, RT5668_SAR_IL_CMD_13, + RT5668_SAR_SOUR_MASK, RT5668_SAR_SOUR_TYPE); + } +} + +/** + * rt5668_headset_detect - Detect headset. + * @component: SoC audio component device. + * @jack_insert: Jack insert or not. + * + * Detect whether is headset or not when jack inserted. + * + * Returns detect status. + */ +static int rt5668_headset_detect(struct snd_soc_component *component, + int jack_insert) +{ + struct rt5668_priv *rt5668 = snd_soc_component_get_drvdata(component); + struct snd_soc_dapm_context *dapm = + snd_soc_component_get_dapm(component); + unsigned int val, count; + + if (jack_insert) { + snd_soc_dapm_force_enable_pin(dapm, "CBJ Power"); + snd_soc_dapm_sync(dapm); + snd_soc_component_update_bits(component, RT5668_CBJ_CTRL_1, + RT5668_TRIG_JD_MASK, RT5668_TRIG_JD_HIGH); + + count = 0; + val = snd_soc_component_read32(component, RT5668_CBJ_CTRL_2) + & RT5668_JACK_TYPE_MASK; + while (val == 0 && count < 50) { + usleep_range(10000, 15000); + val = snd_soc_component_read32(component, + RT5668_CBJ_CTRL_2) & RT5668_JACK_TYPE_MASK; + count++; + } + + switch (val) { + case 0x1: + case 0x2: + rt5668->jack_type = SND_JACK_HEADSET; + rt5668_enable_push_button_irq(component, true); + break; + default: + rt5668->jack_type = SND_JACK_HEADPHONE; + } + + } else { + rt5668_enable_push_button_irq(component, false); + snd_soc_component_update_bits(component, RT5668_CBJ_CTRL_1, + RT5668_TRIG_JD_MASK, RT5668_TRIG_JD_LOW); + snd_soc_dapm_disable_pin(dapm, "CBJ Power"); + snd_soc_dapm_sync(dapm); + + rt5668->jack_type = 0; + } + + dev_dbg(component->dev, "jack_type = %d\n", rt5668->jack_type); + return rt5668->jack_type; +} + +static irqreturn_t rt5668_irq(int irq, void *data) +{ + struct rt5668_priv *rt5668 = data; + + mod_delayed_work(system_power_efficient_wq, + &rt5668->jack_detect_work, msecs_to_jiffies(250)); + + return IRQ_HANDLED; +} + +static void rt5668_jd_check_handler(struct work_struct *work) +{ + struct rt5668_priv *rt5668 = container_of(work, struct rt5668_priv, + jd_check_work.work); + + if (snd_soc_component_read32(rt5668->component, RT5668_AJD1_CTRL) + & RT5668_JDH_RS_MASK) { + /* jack out */ + rt5668->jack_type = rt5668_headset_detect(rt5668->component, 0); + + snd_soc_jack_report(rt5668->hs_jack, rt5668->jack_type, + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + } else { + schedule_delayed_work(&rt5668->jd_check_work, 500); + } +} + +static int rt5668_set_jack_detect(struct snd_soc_component *component, + struct snd_soc_jack *hs_jack, void *data) +{ + struct rt5668_priv *rt5668 = snd_soc_component_get_drvdata(component); + + switch (rt5668->pdata.jd_src) { + case RT5668_JD1: + snd_soc_component_update_bits(component, RT5668_CBJ_CTRL_2, + RT5668_EXT_JD_SRC, RT5668_EXT_JD_SRC_MANUAL); + snd_soc_component_write(component, RT5668_CBJ_CTRL_1, 0xd002); + snd_soc_component_update_bits(component, RT5668_CBJ_CTRL_3, + RT5668_CBJ_IN_BUF_EN, RT5668_CBJ_IN_BUF_EN); + snd_soc_component_update_bits(component, RT5668_SAR_IL_CMD_1, + RT5668_SAR_POW_MASK, RT5668_SAR_POW_EN); + regmap_update_bits(rt5668->regmap, RT5668_GPIO_CTRL_1, + RT5668_GP1_PIN_MASK, RT5668_GP1_PIN_IRQ); + regmap_update_bits(rt5668->regmap, RT5668_RC_CLK_CTRL, + RT5668_POW_IRQ | RT5668_POW_JDH | + RT5668_POW_ANA, RT5668_POW_IRQ | + RT5668_POW_JDH | RT5668_POW_ANA); + regmap_update_bits(rt5668->regmap, RT5668_PWR_ANLG_2, + RT5668_PWR_JDH | RT5668_PWR_JDL, + RT5668_PWR_JDH | RT5668_PWR_JDL); + regmap_update_bits(rt5668->regmap, RT5668_IRQ_CTRL_2, + RT5668_JD1_EN_MASK | RT5668_JD1_POL_MASK, + RT5668_JD1_EN | RT5668_JD1_POL_NOR); + mod_delayed_work(system_power_efficient_wq, + &rt5668->jack_detect_work, msecs_to_jiffies(250)); + break; + + case RT5668_JD_NULL: + regmap_update_bits(rt5668->regmap, RT5668_IRQ_CTRL_2, + RT5668_JD1_EN_MASK, RT5668_JD1_DIS); + regmap_update_bits(rt5668->regmap, RT5668_RC_CLK_CTRL, + RT5668_POW_JDH | RT5668_POW_JDL, 0); + break; + + default: + dev_warn(component->dev, "Wrong JD source\n"); + break; + } + + rt5668->hs_jack = hs_jack; + + return 0; +} + +static void rt5668_jack_detect_handler(struct work_struct *work) +{ + struct rt5668_priv *rt5668 = + container_of(work, struct rt5668_priv, jack_detect_work.work); + int val, btn_type; + + while (!rt5668->component) + usleep_range(10000, 15000); + + while (!rt5668->component->card->instantiated) + usleep_range(10000, 15000); + + mutex_lock(&rt5668->calibrate_mutex); + + val = snd_soc_component_read32(rt5668->component, RT5668_AJD1_CTRL) + & RT5668_JDH_RS_MASK; + if (!val) { + /* jack in */ + if (rt5668->jack_type == 0) { + /* jack was out, report jack type */ + rt5668->jack_type = + rt5668_headset_detect(rt5668->component, 1); + } else { + /* jack is already in, report button event */ + rt5668->jack_type = SND_JACK_HEADSET; + btn_type = rt5668_button_detect(rt5668->component); + /** + * rt5668 can report three kinds of button behavior, + * one click, double click and hold. However, + * currently we will report button pressed/released + * event. So all the three button behaviors are + * treated as button pressed. + */ + switch (btn_type) { + case 0x8000: + case 0x4000: + case 0x2000: + rt5668->jack_type |= SND_JACK_BTN_0; + break; + case 0x1000: + case 0x0800: + case 0x0400: + rt5668->jack_type |= SND_JACK_BTN_1; + break; + case 0x0200: + case 0x0100: + case 0x0080: + rt5668->jack_type |= SND_JACK_BTN_2; + break; + case 0x0040: + case 0x0020: + case 0x0010: + rt5668->jack_type |= SND_JACK_BTN_3; + break; + case 0x0000: /* unpressed */ + break; + default: + btn_type = 0; + dev_err(rt5668->component->dev, + "Unexpected button code 0x%04x\n", + btn_type); + break; + } + } + } else { + /* jack out */ + rt5668->jack_type = rt5668_headset_detect(rt5668->component, 0); + } + + snd_soc_jack_report(rt5668->hs_jack, rt5668->jack_type, + SND_JACK_HEADSET | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3); + + if (rt5668->jack_type & (SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3)) + schedule_delayed_work(&rt5668->jd_check_work, 0); + else + cancel_delayed_work_sync(&rt5668->jd_check_work); + + mutex_unlock(&rt5668->calibrate_mutex); +} + +static const struct snd_kcontrol_new rt5668_snd_controls[] = { + /* Headphone Output Volume */ + SOC_DOUBLE_R_TLV("Headphone Playback Volume", RT5668_HPL_GAIN, + RT5668_HPR_GAIN, RT5668_G_HP_SFT, 15, 1, hp_vol_tlv), + + /* DAC Digital Volume */ + SOC_DOUBLE_TLV("DAC1 Playback Volume", RT5668_DAC1_DIG_VOL, + RT5668_L_VOL_SFT, RT5668_R_VOL_SFT, 175, 0, dac_vol_tlv), + + /* IN Boost Volume */ + SOC_SINGLE_TLV("CBJ Boost Volume", RT5668_CBJ_BST_CTRL, + RT5668_BST_CBJ_SFT, 8, 0, bst_tlv), + + /* ADC Digital Volume Control */ + SOC_DOUBLE("STO1 ADC Capture Switch", RT5668_STO1_ADC_DIG_VOL, + RT5668_L_MUTE_SFT, RT5668_R_MUTE_SFT, 1, 1), + SOC_DOUBLE_TLV("STO1 ADC Capture Volume", RT5668_STO1_ADC_DIG_VOL, + RT5668_L_VOL_SFT, RT5668_R_VOL_SFT, 127, 0, adc_vol_tlv), + + /* ADC Boost Volume Control */ + SOC_DOUBLE_TLV("STO1 ADC Boost Gain Volume", RT5668_STO1_ADC_BOOST, + RT5668_STO1_ADC_L_BST_SFT, RT5668_STO1_ADC_R_BST_SFT, + 3, 0, adc_bst_tlv), +}; + + +static int rt5668_div_sel(struct rt5668_priv *rt5668, + int target, const int div[], int size) +{ + int i; + + if (rt5668->sysclk < target) { + pr_err("sysclk rate %d is too low\n", + rt5668->sysclk); + return 0; + } + + for (i = 0; i < size - 1; i++) { + pr_info("div[%d]=%d\n", i, div[i]); + if (target * div[i] == rt5668->sysclk) + return i; + if (target * div[i + 1] > rt5668->sysclk) { + pr_err("can't find div for sysclk %d\n", + rt5668->sysclk); + return i; + } + } + + if (target * div[i] < rt5668->sysclk) + pr_err("sysclk rate %d is too high\n", + rt5668->sysclk); + + return size - 1; + +} + +/** + * set_dmic_clk - Set parameter of dmic. + * + * @w: DAPM widget. + * @kcontrol: The kcontrol of this widget. + * @event: Event id. + * + * Choose dmic clock between 1MHz and 3MHz. + * It is better for clock to approximate 3MHz. + */ +static int set_dmic_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt5668_priv *rt5668 = snd_soc_component_get_drvdata(component); + int idx = -EINVAL; + static const int div[] = {2, 4, 6, 8, 12, 16, 24, 32, 48, 64, 96, 128}; + + idx = rt5668_div_sel(rt5668, 1500000, div, ARRAY_SIZE(div)); + + snd_soc_component_update_bits(component, RT5668_DMIC_CTRL_1, + RT5668_DMIC_CLK_MASK, idx << RT5668_DMIC_CLK_SFT); + + return 0; +} + +static int set_filter_clk(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + struct rt5668_priv *rt5668 = snd_soc_component_get_drvdata(component); + int ref, val, reg, idx = -EINVAL; + static const int div[] = {1, 2, 3, 4, 6, 8, 12, 16, 24, 32, 48}; + + val = snd_soc_component_read32(component, RT5668_GPIO_CTRL_1) && + RT5668_GP4_PIN_MASK; + if (w->shift == RT5668_PWR_ADC_S1F_BIT && + val == RT5668_GP4_PIN_ADCDAT2) + ref = 256 * rt5668->lrck[RT5668_AIF2]; + else + ref = 256 * rt5668->lrck[RT5668_AIF1]; + + idx = rt5668_div_sel(rt5668, ref, div, ARRAY_SIZE(div)); + + if (w->shift == RT5668_PWR_ADC_S1F_BIT) + reg = RT5668_PLL_TRACK_3; + else + reg = RT5668_PLL_TRACK_2; + + snd_soc_component_update_bits(component, reg, + RT5668_FILTER_CLK_SEL_MASK, idx << RT5668_FILTER_CLK_SEL_SFT); + + return 0; +} + +static int is_sys_clk_from_pll1(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + unsigned int val; + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + val = snd_soc_component_read32(component, RT5668_GLB_CLK); + val &= RT5668_SCLK_SRC_MASK; + if (val == RT5668_SCLK_SRC_PLL1) + return 1; + else + return 0; +} + +static int is_using_asrc(struct snd_soc_dapm_widget *w, + struct snd_soc_dapm_widget *sink) +{ + unsigned int reg, shift, val; + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + switch (w->shift) { + case RT5668_ADC_STO1_ASRC_SFT: + reg = RT5668_PLL_TRACK_3; + shift = RT5668_FILTER_CLK_SEL_SFT; + break; + case RT5668_DAC_STO1_ASRC_SFT: + reg = RT5668_PLL_TRACK_2; + shift = RT5668_FILTER_CLK_SEL_SFT; + break; + default: + return 0; + } + + val = (snd_soc_component_read32(component, reg) >> shift) & 0xf; + switch (val) { + case RT5668_CLK_SEL_I2S1_ASRC: + case RT5668_CLK_SEL_I2S2_ASRC: + return 1; + default: + return 0; + } + +} + +/* Digital Mixer */ +static const struct snd_kcontrol_new rt5668_sto1_adc_l_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5668_STO1_ADC_MIXER, + RT5668_M_STO1_ADC_L1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5668_STO1_ADC_MIXER, + RT5668_M_STO1_ADC_L2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5668_sto1_adc_r_mix[] = { + SOC_DAPM_SINGLE("ADC1 Switch", RT5668_STO1_ADC_MIXER, + RT5668_M_STO1_ADC_R1_SFT, 1, 1), + SOC_DAPM_SINGLE("ADC2 Switch", RT5668_STO1_ADC_MIXER, + RT5668_M_STO1_ADC_R2_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5668_dac_l_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5668_AD_DA_MIXER, + RT5668_M_ADCMIX_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5668_AD_DA_MIXER, + RT5668_M_DAC1_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5668_dac_r_mix[] = { + SOC_DAPM_SINGLE("Stereo ADC Switch", RT5668_AD_DA_MIXER, + RT5668_M_ADCMIX_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC1 Switch", RT5668_AD_DA_MIXER, + RT5668_M_DAC1_R_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5668_sto1_dac_l_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5668_STO1_DAC_MIXER, + RT5668_M_DAC_L1_STO_L_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5668_STO1_DAC_MIXER, + RT5668_M_DAC_R1_STO_L_SFT, 1, 1), +}; + +static const struct snd_kcontrol_new rt5668_sto1_dac_r_mix[] = { + SOC_DAPM_SINGLE("DAC L1 Switch", RT5668_STO1_DAC_MIXER, + RT5668_M_DAC_L1_STO_R_SFT, 1, 1), + SOC_DAPM_SINGLE("DAC R1 Switch", RT5668_STO1_DAC_MIXER, + RT5668_M_DAC_R1_STO_R_SFT, 1, 1), +}; + +/* Analog Input Mixer */ +static const struct snd_kcontrol_new rt5668_rec1_l_mix[] = { + SOC_DAPM_SINGLE("CBJ Switch", RT5668_REC_MIXER, + RT5668_M_CBJ_RM1_L_SFT, 1, 1), +}; + +/* STO1 ADC1 Source */ +/* MX-26 [13] [5] */ +static const char * const rt5668_sto1_adc1_src[] = { + "DAC MIX", "ADC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5668_sto1_adc1l_enum, RT5668_STO1_ADC_MIXER, + RT5668_STO1_ADC1L_SRC_SFT, rt5668_sto1_adc1_src); + +static const struct snd_kcontrol_new rt5668_sto1_adc1l_mux = + SOC_DAPM_ENUM("Stereo1 ADC1L Source", rt5668_sto1_adc1l_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5668_sto1_adc1r_enum, RT5668_STO1_ADC_MIXER, + RT5668_STO1_ADC1R_SRC_SFT, rt5668_sto1_adc1_src); + +static const struct snd_kcontrol_new rt5668_sto1_adc1r_mux = + SOC_DAPM_ENUM("Stereo1 ADC1L Source", rt5668_sto1_adc1r_enum); + +/* STO1 ADC Source */ +/* MX-26 [11:10] [3:2] */ +static const char * const rt5668_sto1_adc_src[] = { + "ADC1 L", "ADC1 R" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5668_sto1_adcl_enum, RT5668_STO1_ADC_MIXER, + RT5668_STO1_ADCL_SRC_SFT, rt5668_sto1_adc_src); + +static const struct snd_kcontrol_new rt5668_sto1_adcl_mux = + SOC_DAPM_ENUM("Stereo1 ADCL Source", rt5668_sto1_adcl_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5668_sto1_adcr_enum, RT5668_STO1_ADC_MIXER, + RT5668_STO1_ADCR_SRC_SFT, rt5668_sto1_adc_src); + +static const struct snd_kcontrol_new rt5668_sto1_adcr_mux = + SOC_DAPM_ENUM("Stereo1 ADCR Source", rt5668_sto1_adcr_enum); + +/* STO1 ADC2 Source */ +/* MX-26 [12] [4] */ +static const char * const rt5668_sto1_adc2_src[] = { + "DAC MIX", "DMIC" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5668_sto1_adc2l_enum, RT5668_STO1_ADC_MIXER, + RT5668_STO1_ADC2L_SRC_SFT, rt5668_sto1_adc2_src); + +static const struct snd_kcontrol_new rt5668_sto1_adc2l_mux = + SOC_DAPM_ENUM("Stereo1 ADC2L Source", rt5668_sto1_adc2l_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5668_sto1_adc2r_enum, RT5668_STO1_ADC_MIXER, + RT5668_STO1_ADC2R_SRC_SFT, rt5668_sto1_adc2_src); + +static const struct snd_kcontrol_new rt5668_sto1_adc2r_mux = + SOC_DAPM_ENUM("Stereo1 ADC2R Source", rt5668_sto1_adc2r_enum); + +/* MX-79 [6:4] I2S1 ADC data location */ +static const unsigned int rt5668_if1_adc_slot_values[] = { + 0, + 2, + 4, + 6, +}; + +static const char * const rt5668_if1_adc_slot_src[] = { + "Slot 0", "Slot 2", "Slot 4", "Slot 6" +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(rt5668_if1_adc_slot_enum, + RT5668_TDM_CTRL, RT5668_TDM_ADC_LCA_SFT, RT5668_TDM_ADC_LCA_MASK, + rt5668_if1_adc_slot_src, rt5668_if1_adc_slot_values); + +static const struct snd_kcontrol_new rt5668_if1_adc_slot_mux = + SOC_DAPM_ENUM("IF1 ADC Slot location", rt5668_if1_adc_slot_enum); + +/* Analog DAC L1 Source, Analog DAC R1 Source*/ +/* MX-2B [4], MX-2B [0]*/ +static const char * const rt5668_alg_dac1_src[] = { + "Stereo1 DAC Mixer", "DAC1" +}; + +static SOC_ENUM_SINGLE_DECL( + rt5668_alg_dac_l1_enum, RT5668_A_DAC1_MUX, + RT5668_A_DACL1_SFT, rt5668_alg_dac1_src); + +static const struct snd_kcontrol_new rt5668_alg_dac_l1_mux = + SOC_DAPM_ENUM("Analog DAC L1 Source", rt5668_alg_dac_l1_enum); + +static SOC_ENUM_SINGLE_DECL( + rt5668_alg_dac_r1_enum, RT5668_A_DAC1_MUX, + RT5668_A_DACR1_SFT, rt5668_alg_dac1_src); + +static const struct snd_kcontrol_new rt5668_alg_dac_r1_mux = + SOC_DAPM_ENUM("Analog DAC R1 Source", rt5668_alg_dac_r1_enum); + +/* Out Switch */ +static const struct snd_kcontrol_new hpol_switch = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5668_HP_CTRL_1, + RT5668_L_MUTE_SFT, 1, 1); +static const struct snd_kcontrol_new hpor_switch = + SOC_DAPM_SINGLE_AUTODISABLE("Switch", RT5668_HP_CTRL_1, + RT5668_R_MUTE_SFT, 1, 1); + +static int rt5668_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + snd_soc_component_write(component, + RT5668_HP_LOGIC_CTRL_2, 0x0012); + snd_soc_component_write(component, + RT5668_HP_CTRL_2, 0x6000); + snd_soc_component_update_bits(component, RT5668_STO_NG2_CTRL_1, + RT5668_NG2_EN_MASK, RT5668_NG2_EN); + snd_soc_component_update_bits(component, + RT5668_DEPOP_1, 0x60, 0x60); + break; + + case SND_SOC_DAPM_POST_PMD: + snd_soc_component_update_bits(component, + RT5668_DEPOP_1, 0x60, 0x0); + snd_soc_component_write(component, + RT5668_HP_CTRL_2, 0x0000); + break; + + default: + return 0; + } + + return 0; + +} + +static int set_dmic_power(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + switch (event) { + case SND_SOC_DAPM_POST_PMU: + /*Add delay to avoid pop noise*/ + msleep(150); + break; + + default: + return 0; + } + + return 0; +} + +static int rt5655_set_verf(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_component *component = + snd_soc_dapm_to_component(w->dapm); + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + switch (w->shift) { + case RT5668_PWR_VREF1_BIT: + snd_soc_component_update_bits(component, + RT5668_PWR_ANLG_1, RT5668_PWR_FV1, 0); + break; + + case RT5668_PWR_VREF2_BIT: + snd_soc_component_update_bits(component, + RT5668_PWR_ANLG_1, RT5668_PWR_FV2, 0); + break; + + default: + break; + } + break; + + case SND_SOC_DAPM_POST_PMU: + usleep_range(15000, 20000); + switch (w->shift) { + case RT5668_PWR_VREF1_BIT: + snd_soc_component_update_bits(component, + RT5668_PWR_ANLG_1, RT5668_PWR_FV1, + RT5668_PWR_FV1); + break; + + case RT5668_PWR_VREF2_BIT: + snd_soc_component_update_bits(component, + RT5668_PWR_ANLG_1, RT5668_PWR_FV2, + RT5668_PWR_FV2); + break; + + default: + break; + } + break; + + default: + return 0; + } + + return 0; +} + +static const unsigned int rt5668_adcdat_pin_values[] = { + 1, + 3, +}; + +static const char * const rt5668_adcdat_pin_select[] = { + "ADCDAT1", + "ADCDAT2", +}; + +static SOC_VALUE_ENUM_SINGLE_DECL(rt5668_adcdat_pin_enum, + RT5668_GPIO_CTRL_1, RT5668_GP4_PIN_SFT, RT5668_GP4_PIN_MASK, + rt5668_adcdat_pin_select, rt5668_adcdat_pin_values); + +static const struct snd_kcontrol_new rt5668_adcdat_pin_ctrl = + SOC_DAPM_ENUM("ADCDAT", rt5668_adcdat_pin_enum); + +static const struct snd_soc_dapm_widget rt5668_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("LDO2", RT5668_PWR_ANLG_3, RT5668_PWR_LDO2_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL1", RT5668_PWR_ANLG_3, RT5668_PWR_PLL_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL2B", RT5668_PWR_ANLG_3, RT5668_PWR_PLL2B_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("PLL2F", RT5668_PWR_ANLG_3, RT5668_PWR_PLL2F_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("Vref1", RT5668_PWR_ANLG_1, RT5668_PWR_VREF1_BIT, 0, + rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + SND_SOC_DAPM_SUPPLY("Vref2", RT5668_PWR_ANLG_1, RT5668_PWR_VREF2_BIT, 0, + rt5655_set_verf, SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMU), + + /* ASRC */ + SND_SOC_DAPM_SUPPLY_S("DAC STO1 ASRC", 1, RT5668_PLL_TRACK_1, + RT5668_DAC_STO1_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("ADC STO1 ASRC", 1, RT5668_PLL_TRACK_1, + RT5668_ADC_STO1_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("AD ASRC", 1, RT5668_PLL_TRACK_1, + RT5668_AD_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DA ASRC", 1, RT5668_PLL_TRACK_1, + RT5668_DA_ASRC_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("DMIC ASRC", 1, RT5668_PLL_TRACK_1, + RT5668_DMIC_ASRC_SFT, 0, NULL, 0), + + /* Input Side */ + SND_SOC_DAPM_SUPPLY("MICBIAS1", RT5668_PWR_ANLG_2, RT5668_PWR_MB1_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("MICBIAS2", RT5668_PWR_ANLG_2, RT5668_PWR_MB2_BIT, + 0, NULL, 0), + + /* Input Lines */ + SND_SOC_DAPM_INPUT("DMIC L1"), + SND_SOC_DAPM_INPUT("DMIC R1"), + + SND_SOC_DAPM_INPUT("IN1P"), + + SND_SOC_DAPM_SUPPLY("DMIC CLK", SND_SOC_NOPM, 0, 0, + set_dmic_clk, SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_SUPPLY("DMIC1 Power", RT5668_DMIC_CTRL_1, + RT5668_DMIC_1_EN_SFT, 0, set_dmic_power, SND_SOC_DAPM_POST_PMU), + + /* Boost */ + SND_SOC_DAPM_PGA("BST1 CBJ", SND_SOC_NOPM, + 0, 0, NULL, 0), + + SND_SOC_DAPM_SUPPLY("CBJ Power", RT5668_PWR_ANLG_3, + RT5668_PWR_CBJ_BIT, 0, NULL, 0), + + /* REC Mixer */ + SND_SOC_DAPM_MIXER("RECMIX1L", SND_SOC_NOPM, 0, 0, rt5668_rec1_l_mix, + ARRAY_SIZE(rt5668_rec1_l_mix)), + SND_SOC_DAPM_SUPPLY("RECMIX1L Power", RT5668_PWR_ANLG_2, + RT5668_PWR_RM1_L_BIT, 0, NULL, 0), + + /* ADCs */ + SND_SOC_DAPM_ADC("ADC1 L", NULL, SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_ADC("ADC1 R", NULL, SND_SOC_NOPM, 0, 0), + + SND_SOC_DAPM_SUPPLY("ADC1 L Power", RT5668_PWR_DIG_1, + RT5668_PWR_ADC_L1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC1 R Power", RT5668_PWR_DIG_1, + RT5668_PWR_ADC_R1_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("ADC1 clock", RT5668_CHOP_ADC, + RT5668_CKGEN_ADC1_SFT, 0, NULL, 0), + + /* ADC Mux */ + SND_SOC_DAPM_MUX("Stereo1 ADC L1 Mux", SND_SOC_NOPM, 0, 0, + &rt5668_sto1_adc1l_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R1 Mux", SND_SOC_NOPM, 0, 0, + &rt5668_sto1_adc1r_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC L2 Mux", SND_SOC_NOPM, 0, 0, + &rt5668_sto1_adc2l_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R2 Mux", SND_SOC_NOPM, 0, 0, + &rt5668_sto1_adc2r_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC L Mux", SND_SOC_NOPM, 0, 0, + &rt5668_sto1_adcl_mux), + SND_SOC_DAPM_MUX("Stereo1 ADC R Mux", SND_SOC_NOPM, 0, 0, + &rt5668_sto1_adcr_mux), + SND_SOC_DAPM_MUX("IF1_ADC Mux", SND_SOC_NOPM, 0, 0, + &rt5668_if1_adc_slot_mux), + + /* ADC Mixer */ + SND_SOC_DAPM_SUPPLY("ADC Stereo1 Filter", RT5668_PWR_DIG_2, + RT5668_PWR_ADC_S1F_BIT, 0, set_filter_clk, + SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_MIXER("Stereo1 ADC MIXL", RT5668_STO1_ADC_DIG_VOL, + RT5668_L_MUTE_SFT, 1, rt5668_sto1_adc_l_mix, + ARRAY_SIZE(rt5668_sto1_adc_l_mix)), + SND_SOC_DAPM_MIXER("Stereo1 ADC MIXR", RT5668_STO1_ADC_DIG_VOL, + RT5668_R_MUTE_SFT, 1, rt5668_sto1_adc_r_mix, + ARRAY_SIZE(rt5668_sto1_adc_r_mix)), + + /* ADC PGA */ + SND_SOC_DAPM_PGA("Stereo1 ADC MIX", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Digital Interface */ + SND_SOC_DAPM_SUPPLY("I2S1", RT5668_PWR_DIG_1, RT5668_PWR_I2S1_BIT, + 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("I2S2", RT5668_PWR_DIG_1, RT5668_PWR_I2S2_BIT, + 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1 L", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("IF1 DAC1 R", SND_SOC_NOPM, 0, 0, NULL, 0), + + /* Digital Interface Select */ + SND_SOC_DAPM_MUX("IF1 01 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5668_if1_01_adc_swap_mux), + SND_SOC_DAPM_MUX("IF1 23 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5668_if1_23_adc_swap_mux), + SND_SOC_DAPM_MUX("IF1 45 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5668_if1_45_adc_swap_mux), + SND_SOC_DAPM_MUX("IF1 67 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5668_if1_67_adc_swap_mux), + SND_SOC_DAPM_MUX("IF2 ADC Swap Mux", SND_SOC_NOPM, 0, 0, + &rt5668_if2_adc_swap_mux), + + SND_SOC_DAPM_MUX("ADCDAT Mux", SND_SOC_NOPM, 0, 0, + &rt5668_adcdat_pin_ctrl), + + /* Audio Interface */ + SND_SOC_DAPM_AIF_OUT("AIF1TX", "AIF1 Capture", 0, + RT5668_I2S1_SDP, RT5668_SEL_ADCDAT_SFT, 1), + SND_SOC_DAPM_AIF_OUT("AIF2TX", "AIF2 Capture", 0, + RT5668_I2S2_SDP, RT5668_I2S2_PIN_CFG_SFT, 1), + SND_SOC_DAPM_AIF_IN("AIF1RX", "AIF1 Playback", 0, SND_SOC_NOPM, 0, 0), + + /* Output Side */ + /* DAC mixer before sound effect */ + SND_SOC_DAPM_MIXER("DAC1 MIXL", SND_SOC_NOPM, 0, 0, + rt5668_dac_l_mix, ARRAY_SIZE(rt5668_dac_l_mix)), + SND_SOC_DAPM_MIXER("DAC1 MIXR", SND_SOC_NOPM, 0, 0, + rt5668_dac_r_mix, ARRAY_SIZE(rt5668_dac_r_mix)), + + /* DAC channel Mux */ + SND_SOC_DAPM_MUX("DAC L1 Source", SND_SOC_NOPM, 0, 0, + &rt5668_alg_dac_l1_mux), + SND_SOC_DAPM_MUX("DAC R1 Source", SND_SOC_NOPM, 0, 0, + &rt5668_alg_dac_r1_mux), + + /* DAC Mixer */ + SND_SOC_DAPM_SUPPLY("DAC Stereo1 Filter", RT5668_PWR_DIG_2, + RT5668_PWR_DAC_S1F_BIT, 0, set_filter_clk, + SND_SOC_DAPM_PRE_PMU), + SND_SOC_DAPM_MIXER("Stereo1 DAC MIXL", SND_SOC_NOPM, 0, 0, + rt5668_sto1_dac_l_mix, ARRAY_SIZE(rt5668_sto1_dac_l_mix)), + SND_SOC_DAPM_MIXER("Stereo1 DAC MIXR", SND_SOC_NOPM, 0, 0, + rt5668_sto1_dac_r_mix, ARRAY_SIZE(rt5668_sto1_dac_r_mix)), + + /* DACs */ + SND_SOC_DAPM_DAC("DAC L1", NULL, RT5668_PWR_DIG_1, + RT5668_PWR_DAC_L1_BIT, 0), + SND_SOC_DAPM_DAC("DAC R1", NULL, RT5668_PWR_DIG_1, + RT5668_PWR_DAC_R1_BIT, 0), + SND_SOC_DAPM_SUPPLY_S("DAC 1 Clock", 3, RT5668_CHOP_DAC, + RT5668_CKGEN_DAC1_SFT, 0, NULL, 0), + + /* HPO */ + SND_SOC_DAPM_PGA_S("HP Amp", 1, SND_SOC_NOPM, 0, 0, rt5668_hp_event, + SND_SOC_DAPM_POST_PMD | SND_SOC_DAPM_PRE_PMU), + + SND_SOC_DAPM_SUPPLY("HP Amp L", RT5668_PWR_ANLG_1, + RT5668_PWR_HA_L_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("HP Amp R", RT5668_PWR_ANLG_1, + RT5668_PWR_HA_R_BIT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("Charge Pump", 1, RT5668_DEPOP_1, + RT5668_PUMP_EN_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY_S("Capless", 2, RT5668_DEPOP_1, + RT5668_CAPLESS_EN_SFT, 0, NULL, 0), + + SND_SOC_DAPM_SWITCH("HPOL Playback", SND_SOC_NOPM, 0, 0, + &hpol_switch), + SND_SOC_DAPM_SWITCH("HPOR Playback", SND_SOC_NOPM, 0, 0, + &hpor_switch), + + /* CLK DET */ + SND_SOC_DAPM_SUPPLY("CLKDET SYS", RT5668_CLK_DET, + RT5668_SYS_CLK_DET_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("CLKDET PLL1", RT5668_CLK_DET, + RT5668_PLL1_CLK_DET_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("CLKDET PLL2", RT5668_CLK_DET, + RT5668_PLL2_CLK_DET_SFT, 0, NULL, 0), + SND_SOC_DAPM_SUPPLY("CLKDET", RT5668_CLK_DET, + RT5668_POW_CLK_DET_SFT, 0, NULL, 0), + + /* Output Lines */ + SND_SOC_DAPM_OUTPUT("HPOL"), + SND_SOC_DAPM_OUTPUT("HPOR"), + +}; + +static const struct snd_soc_dapm_route rt5668_dapm_routes[] = { + /*PLL*/ + {"ADC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll1}, + {"DAC Stereo1 Filter", NULL, "PLL1", is_sys_clk_from_pll1}, + + /*ASRC*/ + {"ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc}, + {"DAC Stereo1 Filter", NULL, "DAC STO1 ASRC", is_using_asrc}, + {"ADC STO1 ASRC", NULL, "AD ASRC"}, + {"DAC STO1 ASRC", NULL, "DA ASRC"}, + + /*Vref*/ + {"MICBIAS1", NULL, "Vref1"}, + {"MICBIAS1", NULL, "Vref2"}, + {"MICBIAS2", NULL, "Vref1"}, + {"MICBIAS2", NULL, "Vref2"}, + + {"CLKDET SYS", NULL, "CLKDET"}, + + {"IN1P", NULL, "LDO2"}, + + {"BST1 CBJ", NULL, "IN1P"}, + {"BST1 CBJ", NULL, "CBJ Power"}, + {"CBJ Power", NULL, "Vref2"}, + + {"RECMIX1L", "CBJ Switch", "BST1 CBJ"}, + {"RECMIX1L", NULL, "RECMIX1L Power"}, + + {"ADC1 L", NULL, "RECMIX1L"}, + {"ADC1 L", NULL, "ADC1 L Power"}, + {"ADC1 L", NULL, "ADC1 clock"}, + + {"DMIC L1", NULL, "DMIC CLK"}, + {"DMIC L1", NULL, "DMIC1 Power"}, + {"DMIC R1", NULL, "DMIC CLK"}, + {"DMIC R1", NULL, "DMIC1 Power"}, + {"DMIC CLK", NULL, "DMIC ASRC"}, + + {"Stereo1 ADC L Mux", "ADC1 L", "ADC1 L"}, + {"Stereo1 ADC L Mux", "ADC1 R", "ADC1 R"}, + {"Stereo1 ADC R Mux", "ADC1 L", "ADC1 L"}, + {"Stereo1 ADC R Mux", "ADC1 R", "ADC1 R"}, + + {"Stereo1 ADC L1 Mux", "ADC", "Stereo1 ADC L Mux"}, + {"Stereo1 ADC L1 Mux", "DAC MIX", "Stereo1 DAC MIXL"}, + {"Stereo1 ADC L2 Mux", "DMIC", "DMIC L1"}, + {"Stereo1 ADC L2 Mux", "DAC MIX", "Stereo1 DAC MIXL"}, + + {"Stereo1 ADC R1 Mux", "ADC", "Stereo1 ADC R Mux"}, + {"Stereo1 ADC R1 Mux", "DAC MIX", "Stereo1 DAC MIXR"}, + {"Stereo1 ADC R2 Mux", "DMIC", "DMIC R1"}, + {"Stereo1 ADC R2 Mux", "DAC MIX", "Stereo1 DAC MIXR"}, + + {"Stereo1 ADC MIXL", "ADC1 Switch", "Stereo1 ADC L1 Mux"}, + {"Stereo1 ADC MIXL", "ADC2 Switch", "Stereo1 ADC L2 Mux"}, + {"Stereo1 ADC MIXL", NULL, "ADC Stereo1 Filter"}, + + {"Stereo1 ADC MIXR", "ADC1 Switch", "Stereo1 ADC R1 Mux"}, + {"Stereo1 ADC MIXR", "ADC2 Switch", "Stereo1 ADC R2 Mux"}, + {"Stereo1 ADC MIXR", NULL, "ADC Stereo1 Filter"}, + + {"Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXL"}, + {"Stereo1 ADC MIX", NULL, "Stereo1 ADC MIXR"}, + + {"IF1 01 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF1 01 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF1 01 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF1 01 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + {"IF1 23 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF1 23 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF1 23 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF1 23 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + {"IF1 45 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF1 45 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF1 45 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF1 45 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + {"IF1 67 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF1 67 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF1 67 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF1 67 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + + {"IF1_ADC Mux", "Slot 0", "IF1 01 ADC Swap Mux"}, + {"IF1_ADC Mux", "Slot 2", "IF1 23 ADC Swap Mux"}, + {"IF1_ADC Mux", "Slot 4", "IF1 45 ADC Swap Mux"}, + {"IF1_ADC Mux", "Slot 6", "IF1 67 ADC Swap Mux"}, + {"IF1_ADC Mux", NULL, "I2S1"}, + {"ADCDAT Mux", "ADCDAT1", "IF1_ADC Mux"}, + {"AIF1TX", NULL, "ADCDAT Mux"}, + {"IF2 ADC Swap Mux", "L/R", "Stereo1 ADC MIX"}, + {"IF2 ADC Swap Mux", "R/L", "Stereo1 ADC MIX"}, + {"IF2 ADC Swap Mux", "L/L", "Stereo1 ADC MIX"}, + {"IF2 ADC Swap Mux", "R/R", "Stereo1 ADC MIX"}, + {"ADCDAT Mux", "ADCDAT2", "IF2 ADC Swap Mux"}, + {"AIF2TX", NULL, "ADCDAT Mux"}, + + {"IF1 DAC1 L", NULL, "AIF1RX"}, + {"IF1 DAC1 L", NULL, "I2S1"}, + {"IF1 DAC1 L", NULL, "DAC Stereo1 Filter"}, + {"IF1 DAC1 R", NULL, "AIF1RX"}, + {"IF1 DAC1 R", NULL, "I2S1"}, + {"IF1 DAC1 R", NULL, "DAC Stereo1 Filter"}, + + {"DAC1 MIXL", "Stereo ADC Switch", "Stereo1 ADC MIXL"}, + {"DAC1 MIXL", "DAC1 Switch", "IF1 DAC1 L"}, + {"DAC1 MIXR", "Stereo ADC Switch", "Stereo1 ADC MIXR"}, + {"DAC1 MIXR", "DAC1 Switch", "IF1 DAC1 R"}, + + {"Stereo1 DAC MIXL", "DAC L1 Switch", "DAC1 MIXL"}, + {"Stereo1 DAC MIXL", "DAC R1 Switch", "DAC1 MIXR"}, + + {"Stereo1 DAC MIXR", "DAC R1 Switch", "DAC1 MIXR"}, + {"Stereo1 DAC MIXR", "DAC L1 Switch", "DAC1 MIXL"}, + + {"DAC L1 Source", "DAC1", "DAC1 MIXL"}, + {"DAC L1 Source", "Stereo1 DAC Mixer", "Stereo1 DAC MIXL"}, + {"DAC R1 Source", "DAC1", "DAC1 MIXR"}, + {"DAC R1 Source", "Stereo1 DAC Mixer", "Stereo1 DAC MIXR"}, + + {"DAC L1", NULL, "DAC L1 Source"}, + {"DAC R1", NULL, "DAC R1 Source"}, + + {"DAC L1", NULL, "DAC 1 Clock"}, + {"DAC R1", NULL, "DAC 1 Clock"}, + + {"HP Amp", NULL, "DAC L1"}, + {"HP Amp", NULL, "DAC R1"}, + {"HP Amp", NULL, "HP Amp L"}, + {"HP Amp", NULL, "HP Amp R"}, + {"HP Amp", NULL, "Capless"}, + {"HP Amp", NULL, "Charge Pump"}, + {"HP Amp", NULL, "CLKDET SYS"}, + {"HP Amp", NULL, "CBJ Power"}, + {"HP Amp", NULL, "Vref2"}, + {"HPOL Playback", "Switch", "HP Amp"}, + {"HPOR Playback", "Switch", "HP Amp"}, + {"HPOL", NULL, "HPOL Playback"}, + {"HPOR", NULL, "HPOR Playback"}, +}; + +static int rt5668_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, + unsigned int rx_mask, int slots, int slot_width) +{ + struct snd_soc_component *component = dai->component; + unsigned int val = 0; + + switch (slots) { + case 4: + val |= RT5668_TDM_TX_CH_4; + val |= RT5668_TDM_RX_CH_4; + break; + case 6: + val |= RT5668_TDM_TX_CH_6; + val |= RT5668_TDM_RX_CH_6; + break; + case 8: + val |= RT5668_TDM_TX_CH_8; + val |= RT5668_TDM_RX_CH_8; + break; + case 2: + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, RT5668_TDM_CTRL, + RT5668_TDM_TX_CH_MASK | RT5668_TDM_RX_CH_MASK, val); + + switch (slot_width) { + case 16: + val = RT5668_TDM_CL_16; + break; + case 20: + val = RT5668_TDM_CL_20; + break; + case 24: + val = RT5668_TDM_CL_24; + break; + case 32: + val = RT5668_TDM_CL_32; + break; + default: + return -EINVAL; + } + + snd_soc_component_update_bits(component, RT5668_TDM_TCON_CTRL, + RT5668_TDM_CL_MASK, val); + + return 0; +} + + +static int rt5668_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 rt5668_priv *rt5668 = snd_soc_component_get_drvdata(component); + unsigned int len_1 = 0, len_2 = 0; + int pre_div, frame_size; + + rt5668->lrck[dai->id] = params_rate(params); + pre_div = rl6231_get_clk_info(rt5668->sysclk, rt5668->lrck[dai->id]); + + frame_size = snd_soc_params_to_frame_size(params); + if (frame_size < 0) { + dev_err(component->dev, "Unsupported frame size: %d\n", + frame_size); + return -EINVAL; + } + + dev_dbg(dai->dev, "lrck is %dHz and pre_div is %d for iis %d\n", + rt5668->lrck[dai->id], pre_div, dai->id); + + switch (params_width(params)) { + case 16: + break; + case 20: + len_1 |= RT5668_I2S1_DL_20; + len_2 |= RT5668_I2S2_DL_20; + break; + case 24: + len_1 |= RT5668_I2S1_DL_24; + len_2 |= RT5668_I2S2_DL_24; + break; + case 32: + len_1 |= RT5668_I2S1_DL_32; + len_2 |= RT5668_I2S2_DL_24; + break; + case 8: + len_1 |= RT5668_I2S2_DL_8; + len_2 |= RT5668_I2S2_DL_8; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5668_AIF1: + snd_soc_component_update_bits(component, RT5668_I2S1_SDP, + RT5668_I2S1_DL_MASK, len_1); + if (rt5668->master[RT5668_AIF1]) { + snd_soc_component_update_bits(component, + RT5668_ADDA_CLK_1, RT5668_I2S_M_DIV_MASK, + pre_div << RT5668_I2S_M_DIV_SFT); + } + if (params_channels(params) == 1) /* mono mode */ + snd_soc_component_update_bits(component, + RT5668_I2S1_SDP, RT5668_I2S1_MONO_MASK, + RT5668_I2S1_MONO_EN); + else + snd_soc_component_update_bits(component, + RT5668_I2S1_SDP, RT5668_I2S1_MONO_MASK, + RT5668_I2S1_MONO_DIS); + break; + case RT5668_AIF2: + snd_soc_component_update_bits(component, RT5668_I2S2_SDP, + RT5668_I2S2_DL_MASK, len_2); + if (rt5668->master[RT5668_AIF2]) { + snd_soc_component_update_bits(component, + RT5668_I2S_M_CLK_CTRL_1, RT5668_I2S2_M_PD_MASK, + pre_div << RT5668_I2S2_M_PD_SFT); + } + if (params_channels(params) == 1) /* mono mode */ + snd_soc_component_update_bits(component, + RT5668_I2S2_SDP, RT5668_I2S2_MONO_MASK, + RT5668_I2S2_MONO_EN); + else + snd_soc_component_update_bits(component, + RT5668_I2S2_SDP, RT5668_I2S2_MONO_MASK, + RT5668_I2S2_MONO_DIS); + break; + default: + dev_err(component->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + + return 0; +} + +static int rt5668_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_component *component = dai->component; + struct rt5668_priv *rt5668 = snd_soc_component_get_drvdata(component); + unsigned int reg_val = 0, tdm_ctrl = 0; + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + rt5668->master[dai->id] = 1; + break; + case SND_SOC_DAIFMT_CBS_CFS: + rt5668->master[dai->id] = 0; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_NB_NF: + break; + case SND_SOC_DAIFMT_IB_NF: + reg_val |= RT5668_I2S_BP_INV; + tdm_ctrl |= RT5668_TDM_S_BP_INV; + break; + case SND_SOC_DAIFMT_NB_IF: + if (dai->id == RT5668_AIF1) + tdm_ctrl |= RT5668_TDM_S_LP_INV | RT5668_TDM_M_BP_INV; + else + return -EINVAL; + break; + case SND_SOC_DAIFMT_IB_IF: + if (dai->id == RT5668_AIF1) + tdm_ctrl |= RT5668_TDM_S_BP_INV | RT5668_TDM_S_LP_INV | + RT5668_TDM_M_BP_INV | RT5668_TDM_M_LP_INV; + else + return -EINVAL; + break; + default: + return -EINVAL; + } + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + break; + case SND_SOC_DAIFMT_LEFT_J: + reg_val |= RT5668_I2S_DF_LEFT; + tdm_ctrl |= RT5668_TDM_DF_LEFT; + break; + case SND_SOC_DAIFMT_DSP_A: + reg_val |= RT5668_I2S_DF_PCM_A; + tdm_ctrl |= RT5668_TDM_DF_PCM_A; + break; + case SND_SOC_DAIFMT_DSP_B: + reg_val |= RT5668_I2S_DF_PCM_B; + tdm_ctrl |= RT5668_TDM_DF_PCM_B; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case RT5668_AIF1: + snd_soc_component_update_bits(component, RT5668_I2S1_SDP, + RT5668_I2S_DF_MASK, reg_val); + snd_soc_component_update_bits(component, RT5668_TDM_TCON_CTRL, + RT5668_TDM_MS_MASK | RT5668_TDM_S_BP_MASK | + RT5668_TDM_DF_MASK | RT5668_TDM_M_BP_MASK | + RT5668_TDM_M_LP_MASK | RT5668_TDM_S_LP_MASK, + tdm_ctrl | rt5668->master[dai->id]); + break; + case RT5668_AIF2: + if (rt5668->master[dai->id] == 0) + reg_val |= RT5668_I2S2_MS_S; + snd_soc_component_update_bits(component, RT5668_I2S2_SDP, + RT5668_I2S2_MS_MASK | RT5668_I2S_BP_MASK | + RT5668_I2S_DF_MASK, reg_val); + break; + default: + dev_err(component->dev, "Invalid dai->id: %d\n", dai->id); + return -EINVAL; + } + return 0; +} + +static int rt5668_set_component_sysclk(struct snd_soc_component *component, + int clk_id, int source, unsigned int freq, int dir) +{ + struct rt5668_priv *rt5668 = snd_soc_component_get_drvdata(component); + unsigned int reg_val = 0, src = 0; + + if (freq == rt5668->sysclk && clk_id == rt5668->sysclk_src) + return 0; + + switch (clk_id) { + case RT5668_SCLK_S_MCLK: + reg_val |= RT5668_SCLK_SRC_MCLK; + src = RT5668_CLK_SRC_MCLK; + break; + case RT5668_SCLK_S_PLL1: + reg_val |= RT5668_SCLK_SRC_PLL1; + src = RT5668_CLK_SRC_PLL1; + break; + case RT5668_SCLK_S_PLL2: + reg_val |= RT5668_SCLK_SRC_PLL2; + src = RT5668_CLK_SRC_PLL2; + break; + case RT5668_SCLK_S_RCCLK: + reg_val |= RT5668_SCLK_SRC_RCCLK; + src = RT5668_CLK_SRC_RCCLK; + break; + default: + dev_err(component->dev, "Invalid clock id (%d)\n", clk_id); + return -EINVAL; + } + snd_soc_component_update_bits(component, RT5668_GLB_CLK, + RT5668_SCLK_SRC_MASK, reg_val); + + if (rt5668->master[RT5668_AIF2]) { + snd_soc_component_update_bits(component, + RT5668_I2S_M_CLK_CTRL_1, RT5668_I2S2_SRC_MASK, + src << RT5668_I2S2_SRC_SFT); + } + + rt5668->sysclk = freq; + rt5668->sysclk_src = clk_id; + + dev_dbg(component->dev, "Sysclk is %dHz and clock id is %d\n", + freq, clk_id); + + return 0; +} + +static int rt5668_set_component_pll(struct snd_soc_component *component, + int pll_id, int source, unsigned int freq_in, + unsigned int freq_out) +{ + struct rt5668_priv *rt5668 = snd_soc_component_get_drvdata(component); + struct rl6231_pll_code pll_code; + int ret; + + if (source == rt5668->pll_src && freq_in == rt5668->pll_in && + freq_out == rt5668->pll_out) + return 0; + + if (!freq_in || !freq_out) { + dev_dbg(component->dev, "PLL disabled\n"); + + rt5668->pll_in = 0; + rt5668->pll_out = 0; + snd_soc_component_update_bits(component, RT5668_GLB_CLK, + RT5668_SCLK_SRC_MASK, RT5668_SCLK_SRC_MCLK); + return 0; + } + + switch (source) { + case RT5668_PLL1_S_MCLK: + snd_soc_component_update_bits(component, RT5668_GLB_CLK, + RT5668_PLL1_SRC_MASK, RT5668_PLL1_SRC_MCLK); + break; + case RT5668_PLL1_S_BCLK1: + snd_soc_component_update_bits(component, RT5668_GLB_CLK, + RT5668_PLL1_SRC_MASK, RT5668_PLL1_SRC_BCLK1); + break; + default: + dev_err(component->dev, "Unknown PLL Source %d\n", source); + return -EINVAL; + } + + ret = rl6231_pll_calc(freq_in, freq_out, &pll_code); + if (ret < 0) { + dev_err(component->dev, "Unsupport input clock %d\n", freq_in); + return ret; + } + + dev_dbg(component->dev, "bypass=%d m=%d n=%d k=%d\n", + pll_code.m_bp, (pll_code.m_bp ? 0 : pll_code.m_code), + pll_code.n_code, pll_code.k_code); + + snd_soc_component_write(component, RT5668_PLL_CTRL_1, + pll_code.n_code << RT5668_PLL_N_SFT | pll_code.k_code); + snd_soc_component_write(component, RT5668_PLL_CTRL_2, + (pll_code.m_bp ? 0 : pll_code.m_code) << RT5668_PLL_M_SFT | + pll_code.m_bp << RT5668_PLL_M_BP_SFT); + + rt5668->pll_in = freq_in; + rt5668->pll_out = freq_out; + rt5668->pll_src = source; + + return 0; +} + +static int rt5668_set_bclk_ratio(struct snd_soc_dai *dai, unsigned int ratio) +{ + struct snd_soc_component *component = dai->component; + struct rt5668_priv *rt5668 = snd_soc_component_get_drvdata(component); + + rt5668->bclk[dai->id] = ratio; + + switch (ratio) { + case 64: + snd_soc_component_update_bits(component, RT5668_ADDA_CLK_2, + RT5668_I2S2_BCLK_MS2_MASK, + RT5668_I2S2_BCLK_MS2_64); + break; + case 32: + snd_soc_component_update_bits(component, RT5668_ADDA_CLK_2, + RT5668_I2S2_BCLK_MS2_MASK, + RT5668_I2S2_BCLK_MS2_32); + break; + default: + dev_err(dai->dev, "Invalid bclk ratio %d\n", ratio); + return -EINVAL; + } + + return 0; +} + +static int rt5668_set_bias_level(struct snd_soc_component *component, + enum snd_soc_bias_level level) +{ + struct rt5668_priv *rt5668 = snd_soc_component_get_drvdata(component); + + switch (level) { + case SND_SOC_BIAS_PREPARE: + regmap_update_bits(rt5668->regmap, RT5668_PWR_ANLG_1, + RT5668_PWR_MB | RT5668_PWR_BG, + RT5668_PWR_MB | RT5668_PWR_BG); + regmap_update_bits(rt5668->regmap, RT5668_PWR_DIG_1, + RT5668_DIG_GATE_CTRL | RT5668_PWR_LDO, + RT5668_DIG_GATE_CTRL | RT5668_PWR_LDO); + break; + + case SND_SOC_BIAS_STANDBY: + regmap_update_bits(rt5668->regmap, RT5668_PWR_ANLG_1, + RT5668_PWR_MB, RT5668_PWR_MB); + regmap_update_bits(rt5668->regmap, RT5668_PWR_DIG_1, + RT5668_DIG_GATE_CTRL, RT5668_DIG_GATE_CTRL); + break; + case SND_SOC_BIAS_OFF: + regmap_update_bits(rt5668->regmap, RT5668_PWR_DIG_1, + RT5668_DIG_GATE_CTRL | RT5668_PWR_LDO, 0); + regmap_update_bits(rt5668->regmap, RT5668_PWR_ANLG_1, + RT5668_PWR_MB | RT5668_PWR_BG, 0); + break; + + default: + break; + } + + return 0; +} + +static int rt5668_probe(struct snd_soc_component *component) +{ + struct rt5668_priv *rt5668 = snd_soc_component_get_drvdata(component); + + rt5668->component = component; + + return 0; +} + +static void rt5668_remove(struct snd_soc_component *component) +{ + struct rt5668_priv *rt5668 = snd_soc_component_get_drvdata(component); + + rt5668_reset(rt5668->regmap); +} + +#ifdef CONFIG_PM +static int rt5668_suspend(struct snd_soc_component *component) +{ + struct rt5668_priv *rt5668 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(rt5668->regmap, true); + regcache_mark_dirty(rt5668->regmap); + return 0; +} + +static int rt5668_resume(struct snd_soc_component *component) +{ + struct rt5668_priv *rt5668 = snd_soc_component_get_drvdata(component); + + regcache_cache_only(rt5668->regmap, false); + regcache_sync(rt5668->regmap); + + return 0; +} +#else +#define rt5668_suspend NULL +#define rt5668_resume NULL +#endif + +#define RT5668_STEREO_RATES SNDRV_PCM_RATE_8000_192000 +#define RT5668_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE | \ + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S8) + +static const struct snd_soc_dai_ops rt5668_aif1_dai_ops = { + .hw_params = rt5668_hw_params, + .set_fmt = rt5668_set_dai_fmt, + .set_tdm_slot = rt5668_set_tdm_slot, +}; + +static const struct snd_soc_dai_ops rt5668_aif2_dai_ops = { + .hw_params = rt5668_hw_params, + .set_fmt = rt5668_set_dai_fmt, + .set_bclk_ratio = rt5668_set_bclk_ratio, +}; + +static struct snd_soc_dai_driver rt5668_dai[] = { + { + .name = "rt5668-aif1", + .id = RT5668_AIF1, + .playback = { + .stream_name = "AIF1 Playback", + .channels_min = 1, + .channels_max = 2, + .rates = RT5668_STEREO_RATES, + .formats = RT5668_FORMATS, + }, + .capture = { + .stream_name = "AIF1 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5668_STEREO_RATES, + .formats = RT5668_FORMATS, + }, + .ops = &rt5668_aif1_dai_ops, + }, + { + .name = "rt5668-aif2", + .id = RT5668_AIF2, + .capture = { + .stream_name = "AIF2 Capture", + .channels_min = 1, + .channels_max = 2, + .rates = RT5668_STEREO_RATES, + .formats = RT5668_FORMATS, + }, + .ops = &rt5668_aif2_dai_ops, + }, +}; + +static const struct snd_soc_component_driver soc_component_dev_rt5668 = { + .probe = rt5668_probe, + .remove = rt5668_remove, + .suspend = rt5668_suspend, + .resume = rt5668_resume, + .set_bias_level = rt5668_set_bias_level, + .controls = rt5668_snd_controls, + .num_controls = ARRAY_SIZE(rt5668_snd_controls), + .dapm_widgets = rt5668_dapm_widgets, + .num_dapm_widgets = ARRAY_SIZE(rt5668_dapm_widgets), + .dapm_routes = rt5668_dapm_routes, + .num_dapm_routes = ARRAY_SIZE(rt5668_dapm_routes), + .set_sysclk = rt5668_set_component_sysclk, + .set_pll = rt5668_set_component_pll, + .set_jack = rt5668_set_jack_detect, + .use_pmdown_time = 1, + .endianness = 1, + .non_legacy_dai_naming = 1, +}; + +static const struct regmap_config rt5668_regmap = { + .reg_bits = 16, + .val_bits = 16, + .max_register = RT5668_I2C_MODE, + .volatile_reg = rt5668_volatile_register, + .readable_reg = rt5668_readable_register, + .cache_type = REGCACHE_RBTREE, + .reg_defaults = rt5668_reg, + .num_reg_defaults = ARRAY_SIZE(rt5668_reg), + .use_single_rw = true, +}; + +static const struct i2c_device_id rt5668_i2c_id[] = { + {"rt5668b", 0}, + {} +}; +MODULE_DEVICE_TABLE(i2c, rt5668_i2c_id); + +static int rt5668_parse_dt(struct rt5668_priv *rt5668, struct device *dev) +{ + + of_property_read_u32(dev->of_node, "realtek,dmic1-data-pin", + &rt5668->pdata.dmic1_data_pin); + of_property_read_u32(dev->of_node, "realtek,dmic1-clk-pin", + &rt5668->pdata.dmic1_clk_pin); + of_property_read_u32(dev->of_node, "realtek,jd-src", + &rt5668->pdata.jd_src); + + rt5668->pdata.ldo1_en = of_get_named_gpio(dev->of_node, + "realtek,ldo1-en-gpios", 0); + + return 0; +} + +static void rt5668_calibrate(struct rt5668_priv *rt5668) +{ + int value, count; + + mutex_lock(&rt5668->calibrate_mutex); + + rt5668_reset(rt5668->regmap); + regmap_write(rt5668->regmap, RT5668_PWR_ANLG_1, 0xa2bf); + usleep_range(15000, 20000); + regmap_write(rt5668->regmap, RT5668_PWR_ANLG_1, 0xf2bf); + regmap_write(rt5668->regmap, RT5668_MICBIAS_2, 0x0380); + regmap_write(rt5668->regmap, RT5668_PWR_DIG_1, 0x8001); + regmap_write(rt5668->regmap, RT5668_TEST_MODE_CTRL_1, 0x0000); + regmap_write(rt5668->regmap, RT5668_STO1_DAC_MIXER, 0x2080); + regmap_write(rt5668->regmap, RT5668_STO1_ADC_MIXER, 0x4040); + regmap_write(rt5668->regmap, RT5668_DEPOP_1, 0x0069); + regmap_write(rt5668->regmap, RT5668_CHOP_DAC, 0x3000); + regmap_write(rt5668->regmap, RT5668_HP_CTRL_2, 0x6000); + regmap_write(rt5668->regmap, RT5668_HP_CHARGE_PUMP_1, 0x0f26); + regmap_write(rt5668->regmap, RT5668_CALIB_ADC_CTRL, 0x7f05); + regmap_write(rt5668->regmap, RT5668_STO1_ADC_MIXER, 0x686c); + regmap_write(rt5668->regmap, RT5668_CAL_REC, 0x0d0d); + regmap_write(rt5668->regmap, RT5668_HP_CALIB_CTRL_9, 0x000f); + regmap_write(rt5668->regmap, RT5668_PWR_DIG_1, 0x8d01); + regmap_write(rt5668->regmap, RT5668_HP_CALIB_CTRL_2, 0x0321); + regmap_write(rt5668->regmap, RT5668_HP_LOGIC_CTRL_2, 0x0004); + regmap_write(rt5668->regmap, RT5668_HP_CALIB_CTRL_1, 0x7c00); + regmap_write(rt5668->regmap, RT5668_HP_CALIB_CTRL_3, 0x06a1); + regmap_write(rt5668->regmap, RT5668_A_DAC1_MUX, 0x0311); + regmap_write(rt5668->regmap, RT5668_RESET_HPF_CTRL, 0x0000); + regmap_write(rt5668->regmap, RT5668_ADC_STO1_HP_CTRL_1, 0x3320); + + regmap_write(rt5668->regmap, RT5668_HP_CALIB_CTRL_1, 0xfc00); + + for (count = 0; count < 60; count++) { + regmap_read(rt5668->regmap, RT5668_HP_CALIB_STA_1, &value); + if (!(value & 0x8000)) + break; + + usleep_range(10000, 10005); + } + + if (count >= 60) + pr_err("HP Calibration Failure\n"); + + /* restore settings */ + regmap_write(rt5668->regmap, RT5668_STO1_ADC_MIXER, 0xc0c4); + regmap_write(rt5668->regmap, RT5668_PWR_DIG_1, 0x0000); + + mutex_unlock(&rt5668->calibrate_mutex); + +} + +static int rt5668_i2c_probe(struct i2c_client *i2c, + const struct i2c_device_id *id) +{ + struct rt5668_platform_data *pdata = dev_get_platdata(&i2c->dev); + struct rt5668_priv *rt5668; + int i, ret; + unsigned int val; + + rt5668 = devm_kzalloc(&i2c->dev, sizeof(struct rt5668_priv), + GFP_KERNEL); + + if (rt5668 == NULL) + return -ENOMEM; + + i2c_set_clientdata(i2c, rt5668); + + if (pdata) + rt5668->pdata = *pdata; + else + rt5668_parse_dt(rt5668, &i2c->dev); + + rt5668->regmap = devm_regmap_init_i2c(i2c, &rt5668_regmap); + if (IS_ERR(rt5668->regmap)) { + ret = PTR_ERR(rt5668->regmap); + dev_err(&i2c->dev, "Failed to allocate register map: %d\n", + ret); + return ret; + } + + for (i = 0; i < ARRAY_SIZE(rt5668->supplies); i++) + rt5668->supplies[i].supply = rt5668_supply_names[i]; + + ret = devm_regulator_bulk_get(&i2c->dev, ARRAY_SIZE(rt5668->supplies), + rt5668->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to request supplies: %d\n", ret); + return ret; + } + + ret = regulator_bulk_enable(ARRAY_SIZE(rt5668->supplies), + rt5668->supplies); + if (ret != 0) { + dev_err(&i2c->dev, "Failed to enable supplies: %d\n", ret); + return ret; + } + + if (gpio_is_valid(rt5668->pdata.ldo1_en)) { + if (devm_gpio_request_one(&i2c->dev, rt5668->pdata.ldo1_en, + GPIOF_OUT_INIT_HIGH, "rt5668")) + dev_err(&i2c->dev, "Fail gpio_request gpio_ldo\n"); + } + + /* Sleep for 300 ms miniumum */ + usleep_range(300000, 350000); + + regmap_write(rt5668->regmap, RT5668_I2C_MODE, 0x1); + usleep_range(10000, 15000); + + regmap_read(rt5668->regmap, RT5668_DEVICE_ID, &val); + if (val != DEVICE_ID) { + pr_err("Device with ID register %x is not rt5668\n", val); + return -ENODEV; + } + + rt5668_reset(rt5668->regmap); + + rt5668_calibrate(rt5668); + + regmap_write(rt5668->regmap, RT5668_DEPOP_1, 0x0000); + + /* DMIC pin*/ + if (rt5668->pdata.dmic1_data_pin != RT5668_DMIC1_NULL) { + switch (rt5668->pdata.dmic1_data_pin) { + case RT5668_DMIC1_DATA_GPIO2: /* share with LRCK2 */ + regmap_update_bits(rt5668->regmap, RT5668_DMIC_CTRL_1, + RT5668_DMIC_1_DP_MASK, RT5668_DMIC_1_DP_GPIO2); + regmap_update_bits(rt5668->regmap, RT5668_GPIO_CTRL_1, + RT5668_GP2_PIN_MASK, RT5668_GP2_PIN_DMIC_SDA); + break; + + case RT5668_DMIC1_DATA_GPIO5: /* share with DACDAT1 */ + regmap_update_bits(rt5668->regmap, RT5668_DMIC_CTRL_1, + RT5668_DMIC_1_DP_MASK, RT5668_DMIC_1_DP_GPIO5); + regmap_update_bits(rt5668->regmap, RT5668_GPIO_CTRL_1, + RT5668_GP5_PIN_MASK, RT5668_GP5_PIN_DMIC_SDA); + break; + + default: + dev_dbg(&i2c->dev, "invalid DMIC_DAT pin\n"); + break; + } + + switch (rt5668->pdata.dmic1_clk_pin) { + case RT5668_DMIC1_CLK_GPIO1: /* share with IRQ */ + regmap_update_bits(rt5668->regmap, RT5668_GPIO_CTRL_1, + RT5668_GP1_PIN_MASK, RT5668_GP1_PIN_DMIC_CLK); + break; + + case RT5668_DMIC1_CLK_GPIO3: /* share with BCLK2 */ + regmap_update_bits(rt5668->regmap, RT5668_GPIO_CTRL_1, + RT5668_GP3_PIN_MASK, RT5668_GP3_PIN_DMIC_CLK); + break; + + default: + dev_dbg(&i2c->dev, "invalid DMIC_CLK pin\n"); + break; + } + } + + regmap_update_bits(rt5668->regmap, RT5668_PWR_ANLG_1, + RT5668_LDO1_DVO_MASK | RT5668_HP_DRIVER_MASK, + RT5668_LDO1_DVO_14 | RT5668_HP_DRIVER_5X); + regmap_write(rt5668->regmap, RT5668_MICBIAS_2, 0x0380); + regmap_update_bits(rt5668->regmap, RT5668_GPIO_CTRL_1, + RT5668_GP4_PIN_MASK | RT5668_GP5_PIN_MASK, + RT5668_GP4_PIN_ADCDAT1 | RT5668_GP5_PIN_DACDAT1); + regmap_write(rt5668->regmap, RT5668_TEST_MODE_CTRL_1, 0x0000); + + INIT_DELAYED_WORK(&rt5668->jack_detect_work, + rt5668_jack_detect_handler); + INIT_DELAYED_WORK(&rt5668->jd_check_work, + rt5668_jd_check_handler); + + mutex_init(&rt5668->calibrate_mutex); + + if (i2c->irq) { + ret = devm_request_threaded_irq(&i2c->dev, i2c->irq, NULL, + rt5668_irq, IRQF_TRIGGER_RISING | IRQF_TRIGGER_FALLING + | IRQF_ONESHOT, "rt5668", rt5668); + if (ret) + dev_err(&i2c->dev, "Failed to reguest IRQ: %d\n", ret); + + } + + return snd_soc_register_component(&i2c->dev, &soc_component_dev_rt5668, + rt5668_dai, ARRAY_SIZE(rt5668_dai)); +} + +static int rt5668_i2c_remove(struct i2c_client *i2c) +{ + snd_soc_unregister_component(&i2c->dev); + + return 0; +} + +static void rt5668_i2c_shutdown(struct i2c_client *client) +{ + struct rt5668_priv *rt5668 = i2c_get_clientdata(client); + + rt5668_reset(rt5668->regmap); +} + +#ifdef CONFIG_OF +static const struct of_device_id rt5668_of_match[] = { + {.compatible = "realtek,rt5668b"}, + {}, +}; +MODULE_DEVICE_TABLE(of, rt5668_of_match); +#endif + +#ifdef CONFIG_ACPI +static const struct acpi_device_id rt5668_acpi_match[] = { + {"10EC5668", 0,}, + {}, +}; +MODULE_DEVICE_TABLE(acpi, rt5668_acpi_match); +#endif + +static struct i2c_driver rt5668_i2c_driver = { + .driver = { + .name = "rt5668b", + .of_match_table = of_match_ptr(rt5668_of_match), + .acpi_match_table = ACPI_PTR(rt5668_acpi_match), + }, + .probe = rt5668_i2c_probe, + .remove = rt5668_i2c_remove, + .shutdown = rt5668_i2c_shutdown, + .id_table = rt5668_i2c_id, +}; +module_i2c_driver(rt5668_i2c_driver); + +MODULE_DESCRIPTION("ASoC RT5668B driver"); +MODULE_AUTHOR("Bard Liao <bardliao(a)realtek.com>"); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/codecs/rt5668.h b/sound/soc/codecs/rt5668.h new file mode 100644 index 0000000..3e7bcfd --- /dev/null +++ b/sound/soc/codecs/rt5668.h @@ -0,0 +1,1318 @@ +/* + * rt5668.h -- RT5668/RT5658 ALSA SoC audio driver + * + * Copyright 2018 Realtek Microelectronics + * Author: Bard Liao <bardliao(a)realtek.com> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef __RT5668_H__ +#define __RT5668_H__ + +#include <sound/rt5668.h> + +#define DEVICE_ID 0x6530 + +/* Info */ +#define RT5668_RESET 0x0000 +#define RT5668_VERSION_ID 0x00fd +#define RT5668_VENDOR_ID 0x00fe +#define RT5668_DEVICE_ID 0x00ff +/* I/O - Output */ +#define RT5668_HP_CTRL_1 0x0002 +#define RT5668_HP_CTRL_2 0x0003 +#define RT5668_HPL_GAIN 0x0005 +#define RT5668_HPR_GAIN 0x0006 + +#define RT5668_I2C_CTRL 0x0008 + +/* I/O - Input */ +#define RT5668_CBJ_BST_CTRL 0x000b +#define RT5668_CBJ_CTRL_1 0x0010 +#define RT5668_CBJ_CTRL_2 0x0011 +#define RT5668_CBJ_CTRL_3 0x0012 +#define RT5668_CBJ_CTRL_4 0x0013 +#define RT5668_CBJ_CTRL_5 0x0014 +#define RT5668_CBJ_CTRL_6 0x0015 +#define RT5668_CBJ_CTRL_7 0x0016 +/* I/O - ADC/DAC/DMIC */ +#define RT5668_DAC1_DIG_VOL 0x0019 +#define RT5668_STO1_ADC_DIG_VOL 0x001c +#define RT5668_STO1_ADC_BOOST 0x001f +#define RT5668_HP_IMP_GAIN_1 0x0022 +#define RT5668_HP_IMP_GAIN_2 0x0023 +/* Mixer - D-D */ +#define RT5668_SIDETONE_CTRL 0x0024 +#define RT5668_STO1_ADC_MIXER 0x0026 +#define RT5668_AD_DA_MIXER 0x0029 +#define RT5668_STO1_DAC_MIXER 0x002a +#define RT5668_A_DAC1_MUX 0x002b +#define RT5668_DIG_INF2_DATA 0x0030 +/* Mixer - ADC */ +#define RT5668_REC_MIXER 0x003c +#define RT5668_CAL_REC 0x0044 +#define RT5668_ALC_BACK_GAIN 0x0049 +/* Power */ +#define RT5668_PWR_DIG_1 0x0061 +#define RT5668_PWR_DIG_2 0x0062 +#define RT5668_PWR_ANLG_1 0x0063 +#define RT5668_PWR_ANLG_2 0x0064 +#define RT5668_PWR_ANLG_3 0x0065 +#define RT5668_PWR_MIXER 0x0066 +#define RT5668_PWR_VOL 0x0067 +/* Clock Detect */ +#define RT5668_CLK_DET 0x006b +/* Filter Auto Reset */ +#define RT5668_RESET_LPF_CTRL 0x006c +#define RT5668_RESET_HPF_CTRL 0x006d +/* DMIC */ +#define RT5668_DMIC_CTRL_1 0x006e +/* Format - ADC/DAC */ +#define RT5668_I2S1_SDP 0x0070 +#define RT5668_I2S2_SDP 0x0071 +#define RT5668_ADDA_CLK_1 0x0073 +#define RT5668_ADDA_CLK_2 0x0074 +#define RT5668_I2S1_F_DIV_CTRL_1 0x0075 +#define RT5668_I2S1_F_DIV_CTRL_2 0x0076 +/* Format - TDM Control */ +#define RT5668_TDM_CTRL 0x0079 +#define RT5668_TDM_ADDA_CTRL_1 0x007a +#define RT5668_TDM_ADDA_CTRL_2 0x007b +#define RT5668_DATA_SEL_CTRL_1 0x007c +#define RT5668_TDM_TCON_CTRL 0x007e +/* Function - Analog */ +#define RT5668_GLB_CLK 0x0080 +#define RT5668_PLL_CTRL_1 0x0081 +#define RT5668_PLL_CTRL_2 0x0082 +#define RT5668_PLL_TRACK_1 0x0083 +#define RT5668_PLL_TRACK_2 0x0084 +#define RT5668_PLL_TRACK_3 0x0085 +#define RT5668_PLL_TRACK_4 0x0086 +#define RT5668_PLL_TRACK_5 0x0087 +#define RT5668_PLL_TRACK_6 0x0088 +#define RT5668_PLL_TRACK_11 0x008c +#define RT5668_SDW_REF_CLK 0x008d +#define RT5668_DEPOP_1 0x008e +#define RT5668_DEPOP_2 0x008f +#define RT5668_HP_CHARGE_PUMP_1 0x0091 +#define RT5668_HP_CHARGE_PUMP_2 0x0092 +#define RT5668_MICBIAS_1 0x0093 +#define RT5668_MICBIAS_2 0x0094 +#define RT5668_PLL_TRACK_12 0x0098 +#define RT5668_PLL_TRACK_14 0x009a +#define RT5668_PLL2_CTRL_1 0x009b +#define RT5668_PLL2_CTRL_2 0x009c +#define RT5668_PLL2_CTRL_3 0x009d +#define RT5668_PLL2_CTRL_4 0x009e +#define RT5668_RC_CLK_CTRL 0x009f +#define RT5668_I2S_M_CLK_CTRL_1 0x00a0 +#define RT5668_I2S2_F_DIV_CTRL_1 0x00a3 +#define RT5668_I2S2_F_DIV_CTRL_2 0x00a4 +/* Function - Digital */ +#define RT5668_EQ_CTRL_1 0x00ae +#define RT5668_EQ_CTRL_2 0x00af +#define RT5668_IRQ_CTRL_1 0x00b6 +#define RT5668_IRQ_CTRL_2 0x00b7 +#define RT5668_IRQ_CTRL_3 0x00b8 +#define RT5668_IRQ_CTRL_4 0x00b9 +#define RT5668_INT_ST_1 0x00be +#define RT5668_GPIO_CTRL_1 0x00c0 +#define RT5668_GPIO_CTRL_2 0x00c1 +#define RT5668_GPIO_CTRL_3 0x00c2 +#define RT5668_HP_AMP_DET_CTRL_1 0x00d0 +#define RT5668_HP_AMP_DET_CTRL_2 0x00d1 +#define RT5668_MID_HP_AMP_DET 0x00d2 +#define RT5668_LOW_HP_AMP_DET 0x00d3 +#define RT5668_DELAY_BUF_CTRL 0x00d4 +#define RT5668_SV_ZCD_1 0x00d9 +#define RT5668_SV_ZCD_2 0x00da +#define RT5668_IL_CMD_1 0x00db +#define RT5668_IL_CMD_2 0x00dc +#define RT5668_IL_CMD_3 0x00dd +#define RT5668_IL_CMD_4 0x00de +#define RT5668_IL_CMD_5 0x00df +#define RT5668_IL_CMD_6 0x00e0 +#define RT5668_4BTN_IL_CMD_1 0x00e2 +#define RT5668_4BTN_IL_CMD_2 0x00e3 +#define RT5668_4BTN_IL_CMD_3 0x00e4 +#define RT5668_4BTN_IL_CMD_4 0x00e5 +#define RT5668_4BTN_IL_CMD_5 0x00e6 +#define RT5668_4BTN_IL_CMD_6 0x00e7 +#define RT5668_4BTN_IL_CMD_7 0x00e8 + +#define RT5668_ADC_STO1_HP_CTRL_1 0x00ea +#define RT5668_ADC_STO1_HP_CTRL_2 0x00eb +#define RT5668_AJD1_CTRL 0x00f0 +#define RT5668_JD1_THD 0x00f1 +#define RT5668_JD2_THD 0x00f2 +#define RT5668_JD_CTRL_1 0x00f6 +/* General Control */ +#define RT5668_DUMMY_1 0x00fa +#define RT5668_DUMMY_2 0x00fb +#define RT5668_DUMMY_3 0x00fc + +#define RT5668_DAC_ADC_DIG_VOL1 0x0100 +#define RT5668_BIAS_CUR_CTRL_2 0x010b +#define RT5668_BIAS_CUR_CTRL_3 0x010c +#define RT5668_BIAS_CUR_CTRL_4 0x010d +#define RT5668_BIAS_CUR_CTRL_5 0x010e +#define RT5668_BIAS_CUR_CTRL_6 0x010f +#define RT5668_BIAS_CUR_CTRL_7 0x0110 +#define RT5668_BIAS_CUR_CTRL_8 0x0111 +#define RT5668_BIAS_CUR_CTRL_9 0x0112 +#define RT5668_BIAS_CUR_CTRL_10 0x0113 +#define RT5668_VREF_REC_OP_FB_CAP_CTRL 0x0117 +#define RT5668_CHARGE_PUMP_1 0x0125 +#define RT5668_DIG_IN_CTRL_1 0x0132 +#define RT5668_PAD_DRIVING_CTRL 0x0136 +#define RT5668_SOFT_RAMP_DEPOP 0x0138 +#define RT5668_CHOP_DAC 0x013a +#define RT5668_CHOP_ADC 0x013b +#define RT5668_CALIB_ADC_CTRL 0x013c +#define RT5668_VOL_TEST 0x013f +#define RT5668_SPKVDD_DET_STA 0x0142 +#define RT5668_TEST_MODE_CTRL_1 0x0145 +#define RT5668_TEST_MODE_CTRL_2 0x0146 +#define RT5668_TEST_MODE_CTRL_3 0x0147 +#define RT5668_TEST_MODE_CTRL_4 0x0148 +#define RT5668_TEST_MODE_CTRL_5 0x0149 +#define RT5668_PLL1_INTERNAL 0x0150 +#define RT5668_PLL2_INTERNAL 0x0151 +#define RT5668_STO_NG2_CTRL_1 0x0160 +#define RT5668_STO_NG2_CTRL_2 0x0161 +#define RT5668_STO_NG2_CTRL_3 0x0162 +#define RT5668_STO_NG2_CTRL_4 0x0163 +#define RT5668_STO_NG2_CTRL_5 0x0164 +#define RT5668_STO_NG2_CTRL_6 0x0165 +#define RT5668_STO_NG2_CTRL_7 0x0166 +#define RT5668_STO_NG2_CTRL_8 0x0167 +#define RT5668_STO_NG2_CTRL_9 0x0168 +#define RT5668_STO_NG2_CTRL_10 0x0169 +#define RT5668_STO1_DAC_SIL_DET 0x0190 +#define RT5668_SIL_PSV_CTRL1 0x0194 +#define RT5668_SIL_PSV_CTRL2 0x0195 +#define RT5668_SIL_PSV_CTRL3 0x0197 +#define RT5668_SIL_PSV_CTRL4 0x0198 +#define RT5668_SIL_PSV_CTRL5 0x0199 +#define RT5668_HP_IMP_SENS_CTRL_01 0x01af +#define RT5668_HP_IMP_SENS_CTRL_02 0x01b0 +#define RT5668_HP_IMP_SENS_CTRL_03 0x01b1 +#define RT5668_HP_IMP_SENS_CTRL_04 0x01b2 +#define RT5668_HP_IMP_SENS_CTRL_05 0x01b3 +#define RT5668_HP_IMP_SENS_CTRL_06 0x01b4 +#define RT5668_HP_IMP_SENS_CTRL_07 0x01b5 +#define RT5668_HP_IMP_SENS_CTRL_08 0x01b6 +#define RT5668_HP_IMP_SENS_CTRL_09 0x01b7 +#define RT5668_HP_IMP_SENS_CTRL_10 0x01b8 +#define RT5668_HP_IMP_SENS_CTRL_11 0x01b9 +#define RT5668_HP_IMP_SENS_CTRL_12 0x01ba +#define RT5668_HP_IMP_SENS_CTRL_13 0x01bb +#define RT5668_HP_IMP_SENS_CTRL_14 0x01bc +#define RT5668_HP_IMP_SENS_CTRL_15 0x01bd +#define RT5668_HP_IMP_SENS_CTRL_16 0x01be +#define RT5668_HP_IMP_SENS_CTRL_17 0x01bf +#define RT5668_HP_IMP_SENS_CTRL_18 0x01c0 +#define RT5668_HP_IMP_SENS_CTRL_19 0x01c1 +#define RT5668_HP_IMP_SENS_CTRL_20 0x01c2 +#define RT5668_HP_IMP_SENS_CTRL_21 0x01c3 +#define RT5668_HP_IMP_SENS_CTRL_22 0x01c4 +#define RT5668_HP_IMP_SENS_CTRL_23 0x01c5 +#define RT5668_HP_IMP_SENS_CTRL_24 0x01c6 +#define RT5668_HP_IMP_SENS_CTRL_25 0x01c7 +#define RT5668_HP_IMP_SENS_CTRL_26 0x01c8 +#define RT5668_HP_IMP_SENS_CTRL_27 0x01c9 +#define RT5668_HP_IMP_SENS_CTRL_28 0x01ca +#define RT5668_HP_IMP_SENS_CTRL_29 0x01cb +#define RT5668_HP_IMP_SENS_CTRL_30 0x01cc +#define RT5668_HP_IMP_SENS_CTRL_31 0x01cd +#define RT5668_HP_IMP_SENS_CTRL_32 0x01ce +#define RT5668_HP_IMP_SENS_CTRL_33 0x01cf +#define RT5668_HP_IMP_SENS_CTRL_34 0x01d0 +#define RT5668_HP_IMP_SENS_CTRL_35 0x01d1 +#define RT5668_HP_IMP_SENS_CTRL_36 0x01d2 +#define RT5668_HP_IMP_SENS_CTRL_37 0x01d3 +#define RT5668_HP_IMP_SENS_CTRL_38 0x01d4 +#define RT5668_HP_IMP_SENS_CTRL_39 0x01d5 +#define RT5668_HP_IMP_SENS_CTRL_40 0x01d6 +#define RT5668_HP_IMP_SENS_CTRL_41 0x01d7 +#define RT5668_HP_IMP_SENS_CTRL_42 0x01d8 +#define RT5668_HP_IMP_SENS_CTRL_43 0x01d9 +#define RT5668_HP_LOGIC_CTRL_1 0x01da +#define RT5668_HP_LOGIC_CTRL_2 0x01db +#define RT5668_HP_LOGIC_CTRL_3 0x01dc +#define RT5668_HP_CALIB_CTRL_1 0x01de +#define RT5668_HP_CALIB_CTRL_2 0x01df +#define RT5668_HP_CALIB_CTRL_3 0x01e0 +#define RT5668_HP_CALIB_CTRL_4 0x01e1 +#define RT5668_HP_CALIB_CTRL_5 0x01e2 +#define RT5668_HP_CALIB_CTRL_6 0x01e3 +#define RT5668_HP_CALIB_CTRL_7 0x01e4 +#define RT5668_HP_CALIB_CTRL_9 0x01e6 +#define RT5668_HP_CALIB_CTRL_10 0x01e7 +#define RT5668_HP_CALIB_CTRL_11 0x01e8 +#define RT5668_HP_CALIB_STA_1 0x01ea +#define RT5668_HP_CALIB_STA_2 0x01eb +#define RT5668_HP_CALIB_STA_3 0x01ec +#define RT5668_HP_CALIB_STA_4 0x01ed +#define RT5668_HP_CALIB_STA_5 0x01ee +#define RT5668_HP_CALIB_STA_6 0x01ef +#define RT5668_HP_CALIB_STA_7 0x01f0 +#define RT5668_HP_CALIB_STA_8 0x01f1 +#define RT5668_HP_CALIB_STA_9 0x01f2 +#define RT5668_HP_CALIB_STA_10 0x01f3 +#define RT5668_HP_CALIB_STA_11 0x01f4 +#define RT5668_SAR_IL_CMD_1 0x0210 +#define RT5668_SAR_IL_CMD_2 0x0211 +#define RT5668_SAR_IL_CMD_3 0x0212 +#define RT5668_SAR_IL_CMD_4 0x0213 +#define RT5668_SAR_IL_CMD_5 0x0214 +#define RT5668_SAR_IL_CMD_6 0x0215 +#define RT5668_SAR_IL_CMD_7 0x0216 +#define RT5668_SAR_IL_CMD_8 0x0217 +#define RT5668_SAR_IL_CMD_9 0x0218 +#define RT5668_SAR_IL_CMD_10 0x0219 +#define RT5668_SAR_IL_CMD_11 0x021a +#define RT5668_SAR_IL_CMD_12 0x021b +#define RT5668_SAR_IL_CMD_13 0x021c +#define RT5668_EFUSE_CTRL_1 0x0250 +#define RT5668_EFUSE_CTRL_2 0x0251 +#define RT5668_EFUSE_CTRL_3 0x0252 +#define RT5668_EFUSE_CTRL_4 0x0253 +#define RT5668_EFUSE_CTRL_5 0x0254 +#define RT5668_EFUSE_CTRL_6 0x0255 +#define RT5668_EFUSE_CTRL_7 0x0256 +#define RT5668_EFUSE_CTRL_8 0x0257 +#define RT5668_EFUSE_CTRL_9 0x0258 +#define RT5668_EFUSE_CTRL_10 0x0259 +#define RT5668_EFUSE_CTRL_11 0x025a +#define RT5668_JD_TOP_VC_VTRL 0x0270 +#define RT5668_DRC1_CTRL_0 0x02ff +#define RT5668_DRC1_CTRL_1 0x0300 +#define RT5668_DRC1_CTRL_2 0x0301 +#define RT5668_DRC1_CTRL_3 0x0302 +#define RT5668_DRC1_CTRL_4 0x0303 +#define RT5668_DRC1_CTRL_5 0x0304 +#define RT5668_DRC1_CTRL_6 0x0305 +#define RT5668_DRC1_HARD_LMT_CTRL_1 0x0306 +#define RT5668_DRC1_HARD_LMT_CTRL_2 0x0307 +#define RT5668_DRC1_PRIV_1 0x0310 +#define RT5668_DRC1_PRIV_2 0x0311 +#define RT5668_DRC1_PRIV_3 0x0312 +#define RT5668_DRC1_PRIV_4 0x0313 +#define RT5668_DRC1_PRIV_5 0x0314 +#define RT5668_DRC1_PRIV_6 0x0315 +#define RT5668_DRC1_PRIV_7 0x0316 +#define RT5668_DRC1_PRIV_8 0x0317 +#define RT5668_EQ_AUTO_RCV_CTRL1 0x03c0 +#define RT5668_EQ_AUTO_RCV_CTRL2 0x03c1 +#define RT5668_EQ_AUTO_RCV_CTRL3 0x03c2 +#define RT5668_EQ_AUTO_RCV_CTRL4 0x03c3 +#define RT5668_EQ_AUTO_RCV_CTRL5 0x03c4 +#define RT5668_EQ_AUTO_RCV_CTRL6 0x03c5 +#define RT5668_EQ_AUTO_RCV_CTRL7 0x03c6 +#define RT5668_EQ_AUTO_RCV_CTRL8 0x03c7 +#define RT5668_EQ_AUTO_RCV_CTRL9 0x03c8 +#define RT5668_EQ_AUTO_RCV_CTRL10 0x03c9 +#define RT5668_EQ_AUTO_RCV_CTRL11 0x03ca +#define RT5668_EQ_AUTO_RCV_CTRL12 0x03cb +#define RT5668_EQ_AUTO_RCV_CTRL13 0x03cc +#define RT5668_ADC_L_EQ_LPF1_A1 0x03d0 +#define RT5668_R_EQ_LPF1_A1 0x03d1 +#define RT5668_L_EQ_LPF1_H0 0x03d2 +#define RT5668_R_EQ_LPF1_H0 0x03d3 +#define RT5668_L_EQ_BPF1_A1 0x03d4 +#define RT5668_R_EQ_BPF1_A1 0x03d5 +#define RT5668_L_EQ_BPF1_A2 0x03d6 +#define RT5668_R_EQ_BPF1_A2 0x03d7 +#define RT5668_L_EQ_BPF1_H0 0x03d8 +#define RT5668_R_EQ_BPF1_H0 0x03d9 +#define RT5668_L_EQ_BPF2_A1 0x03da +#define RT5668_R_EQ_BPF2_A1 0x03db +#define RT5668_L_EQ_BPF2_A2 0x03dc +#define RT5668_R_EQ_BPF2_A2 0x03dd +#define RT5668_L_EQ_BPF2_H0 0x03de +#define RT5668_R_EQ_BPF2_H0 0x03df +#define RT5668_L_EQ_BPF3_A1 0x03e0 +#define RT5668_R_EQ_BPF3_A1 0x03e1 +#define RT5668_L_EQ_BPF3_A2 0x03e2 +#define RT5668_R_EQ_BPF3_A2 0x03e3 +#define RT5668_L_EQ_BPF3_H0 0x03e4 +#define RT5668_R_EQ_BPF3_H0 0x03e5 +#define RT5668_L_EQ_BPF4_A1 0x03e6 +#define RT5668_R_EQ_BPF4_A1 0x03e7 +#define RT5668_L_EQ_BPF4_A2 0x03e8 +#define RT5668_R_EQ_BPF4_A2 0x03e9 +#define RT5668_L_EQ_BPF4_H0 0x03ea +#define RT5668_R_EQ_BPF4_H0 0x03eb +#define RT5668_L_EQ_HPF1_A1 0x03ec +#define RT5668_R_EQ_HPF1_A1 0x03ed +#define RT5668_L_EQ_HPF1_H0 0x03ee +#define RT5668_R_EQ_HPF1_H0 0x03ef +#define RT5668_L_EQ_PRE_VOL 0x03f0 +#define RT5668_R_EQ_PRE_VOL 0x03f1 +#define RT5668_L_EQ_POST_VOL 0x03f2 +#define RT5668_R_EQ_POST_VOL 0x03f3 +#define RT5668_I2C_MODE 0xffff + + +/* global definition */ +#define RT5668_L_MUTE (0x1 << 15) +#define RT5668_L_MUTE_SFT 15 +#define RT5668_VOL_L_MUTE (0x1 << 14) +#define RT5668_VOL_L_SFT 14 +#define RT5668_R_MUTE (0x1 << 7) +#define RT5668_R_MUTE_SFT 7 +#define RT5668_VOL_R_MUTE (0x1 << 6) +#define RT5668_VOL_R_SFT 6 +#define RT5668_L_VOL_MASK (0x3f << 8) +#define RT5668_L_VOL_SFT 8 +#define RT5668_R_VOL_MASK (0x3f) +#define RT5668_R_VOL_SFT 0 + +/*Headphone Amp L/R Analog Gain and Digital NG2 Gain Control (0x0005 0x0006)*/ +#define RT5668_G_HP (0xf << 8) +#define RT5668_G_HP_SFT 8 +#define RT5668_G_STO_DA_DMIX (0xf) +#define RT5668_G_STO_DA_SFT 0 + +/* CBJ Control (0x000b) */ +#define RT5668_BST_CBJ_MASK (0xf << 8) +#define RT5668_BST_CBJ_SFT 8 + +/* Embeeded Jack and Type Detection Control 1 (0x0010) */ +#define RT5668_EMB_JD_EN (0x1 << 15) +#define RT5668_EMB_JD_EN_SFT 15 +#define RT5668_EMB_JD_RST (0x1 << 14) +#define RT5668_JD_MODE (0x1 << 13) +#define RT5668_JD_MODE_SFT 13 +#define RT5668_DET_TYPE (0x1 << 12) +#define RT5668_DET_TYPE_SFT 12 +#define RT5668_POLA_EXT_JD_MASK (0x1 << 11) +#define RT5668_POLA_EXT_JD_LOW (0x1 << 11) +#define RT5668_POLA_EXT_JD_HIGH (0x0 << 11) +#define RT5668_EXT_JD_DIG (0x1 << 9) +#define RT5668_POL_FAST_OFF_MASK (0x1 << 8) +#define RT5668_POL_FAST_OFF_HIGH (0x1 << 8) +#define RT5668_POL_FAST_OFF_LOW (0x0 << 8) +#define RT5668_FAST_OFF_MASK (0x1 << 7) +#define RT5668_FAST_OFF_EN (0x1 << 7) +#define RT5668_FAST_OFF_DIS (0x0 << 7) +#define RT5668_VREF_POW_MASK (0x1 << 6) +#define RT5668_VREF_POW_FSM (0x0 << 6) +#define RT5668_VREF_POW_REG (0x1 << 6) +#define RT5668_MB1_PATH_MASK (0x1 << 5) +#define RT5668_CTRL_MB1_REG (0x1 << 5) +#define RT5668_CTRL_MB1_FSM (0x0 << 5) +#define RT5668_MB2_PATH_MASK (0x1 << 4) +#define RT5668_CTRL_MB2_REG (0x1 << 4) +#define RT5668_CTRL_MB2_FSM (0x0 << 4) +#define RT5668_TRIG_JD_MASK (0x1 << 3) +#define RT5668_TRIG_JD_HIGH (0x1 << 3) +#define RT5668_TRIG_JD_LOW (0x0 << 3) +#define RT5668_MIC_CAP_MASK (0x1 << 1) +#define RT5668_MIC_CAP_HS (0x1 << 1) +#define RT5668_MIC_CAP_HP (0x0 << 1) +#define RT5668_MIC_CAP_SRC_MASK (0x1) +#define RT5668_MIC_CAP_SRC_REG (0x1) +#define RT5668_MIC_CAP_SRC_ANA (0x0) + +/* Embeeded Jack and Type Detection Control 2 (0x0011) */ +#define RT5668_EXT_JD_SRC (0x7 << 4) +#define RT5668_EXT_JD_SRC_SFT 4 +#define RT5668_EXT_JD_SRC_GPIO_JD1 (0x0 << 4) +#define RT5668_EXT_JD_SRC_GPIO_JD2 (0x1 << 4) +#define RT5668_EXT_JD_SRC_JDH (0x2 << 4) +#define RT5668_EXT_JD_SRC_JDL (0x3 << 4) +#define RT5668_EXT_JD_SRC_MANUAL (0x4 << 4) +#define RT5668_JACK_TYPE_MASK (0x3) + +/* Combo Jack and Type Detection Control 3 (0x0012) */ +#define RT5668_CBJ_IN_BUF_EN (0x1 << 7) + +/* Combo Jack and Type Detection Control 4 (0x0013) */ +#define RT5668_SEL_SHT_MID_TON_MASK (0x3 << 12) +#define RT5668_SEL_SHT_MID_TON_2 (0x0 << 12) +#define RT5668_SEL_SHT_MID_TON_3 (0x1 << 12) +#define RT5668_CBJ_JD_TEST_MASK (0x1 << 6) +#define RT5668_CBJ_JD_TEST_NORM (0x0 << 6) +#define RT5668_CBJ_JD_TEST_MODE (0x1 << 6) + +/* DAC1 Digital Volume (0x0019) */ +#define RT5668_DAC_L1_VOL_MASK (0xff << 8) +#define RT5668_DAC_L1_VOL_SFT 8 +#define RT5668_DAC_R1_VOL_MASK (0xff) +#define RT5668_DAC_R1_VOL_SFT 0 + +/* ADC Digital Volume Control (0x001c) */ +#define RT5668_ADC_L_VOL_MASK (0x7f << 8) +#define RT5668_ADC_L_VOL_SFT 8 +#define RT5668_ADC_R_VOL_MASK (0x7f) +#define RT5668_ADC_R_VOL_SFT 0 + +/* Stereo1 ADC Boost Gain Control (0x001f) */ +#define RT5668_STO1_ADC_L_BST_MASK (0x3 << 14) +#define RT5668_STO1_ADC_L_BST_SFT 14 +#define RT5668_STO1_ADC_R_BST_MASK (0x3 << 12) +#define RT5668_STO1_ADC_R_BST_SFT 12 + +/* Sidetone Control (0x0024) */ +#define RT5668_ST_SRC_SEL (0x1 << 8) +#define RT5668_ST_SRC_SFT 8 +#define RT5668_ST_EN_MASK (0x1 << 6) +#define RT5668_ST_DIS (0x0 << 6) +#define RT5668_ST_EN (0x1 << 6) +#define RT5668_ST_EN_SFT 6 + +/* Stereo1 ADC Mixer Control (0x0026) */ +#define RT5668_M_STO1_ADC_L1 (0x1 << 15) +#define RT5668_M_STO1_ADC_L1_SFT 15 +#define RT5668_M_STO1_ADC_L2 (0x1 << 14) +#define RT5668_M_STO1_ADC_L2_SFT 14 +#define RT5668_STO1_ADC1L_SRC_MASK (0x1 << 13) +#define RT5668_STO1_ADC1L_SRC_SFT 13 +#define RT5668_STO1_ADC1_SRC_ADC (0x1 << 13) +#define RT5668_STO1_ADC1_SRC_DACMIX (0x0 << 13) +#define RT5668_STO1_ADC2L_SRC_MASK (0x1 << 12) +#define RT5668_STO1_ADC2L_SRC_SFT 12 +#define RT5668_STO1_ADCL_SRC_MASK (0x3 << 10) +#define RT5668_STO1_ADCL_SRC_SFT 10 +#define RT5668_STO1_DD_L_SRC_MASK (0x1 << 9) +#define RT5668_STO1_DD_L_SRC_SFT 9 +#define RT5668_STO1_DMIC_SRC_MASK (0x1 << 8) +#define RT5668_STO1_DMIC_SRC_SFT 8 +#define RT5668_STO1_DMIC_SRC_DMIC2 (0x1 << 8) +#define RT5668_STO1_DMIC_SRC_DMIC1 (0x0 << 8) +#define RT5668_M_STO1_ADC_R1 (0x1 << 7) +#define RT5668_M_STO1_ADC_R1_SFT 7 +#define RT5668_M_STO1_ADC_R2 (0x1 << 6) +#define RT5668_M_STO1_ADC_R2_SFT 6 +#define RT5668_STO1_ADC1R_SRC_MASK (0x1 << 5) +#define RT5668_STO1_ADC1R_SRC_SFT 5 +#define RT5668_STO1_ADC2R_SRC_MASK (0x1 << 4) +#define RT5668_STO1_ADC2R_SRC_SFT 4 +#define RT5668_STO1_ADCR_SRC_MASK (0x3 << 2) +#define RT5668_STO1_ADCR_SRC_SFT 2 + +/* ADC Mixer to DAC Mixer Control (0x0029) */ +#define RT5668_M_ADCMIX_L (0x1 << 15) +#define RT5668_M_ADCMIX_L_SFT 15 +#define RT5668_M_DAC1_L (0x1 << 14) +#define RT5668_M_DAC1_L_SFT 14 +#define RT5668_DAC1_R_SEL_MASK (0x1 << 10) +#define RT5668_DAC1_R_SEL_SFT 10 +#define RT5668_DAC1_L_SEL_MASK (0x1 << 8) +#define RT5668_DAC1_L_SEL_SFT 8 +#define RT5668_M_ADCMIX_R (0x1 << 7) +#define RT5668_M_ADCMIX_R_SFT 7 +#define RT5668_M_DAC1_R (0x1 << 6) +#define RT5668_M_DAC1_R_SFT 6 + +/* Stereo1 DAC Mixer Control (0x002a) */ +#define RT5668_M_DAC_L1_STO_L (0x1 << 15) +#define RT5668_M_DAC_L1_STO_L_SFT 15 +#define RT5668_G_DAC_L1_STO_L_MASK (0x1 << 14) +#define RT5668_G_DAC_L1_STO_L_SFT 14 +#define RT5668_M_DAC_R1_STO_L (0x1 << 13) +#define RT5668_M_DAC_R1_STO_L_SFT 13 +#define RT5668_G_DAC_R1_STO_L_MASK (0x1 << 12) +#define RT5668_G_DAC_R1_STO_L_SFT 12 +#define RT5668_M_DAC_L1_STO_R (0x1 << 7) +#define RT5668_M_DAC_L1_STO_R_SFT 7 +#define RT5668_G_DAC_L1_STO_R_MASK (0x1 << 6) +#define RT5668_G_DAC_L1_STO_R_SFT 6 +#define RT5668_M_DAC_R1_STO_R (0x1 << 5) +#define RT5668_M_DAC_R1_STO_R_SFT 5 +#define RT5668_G_DAC_R1_STO_R_MASK (0x1 << 4) +#define RT5668_G_DAC_R1_STO_R_SFT 4 + +/* Analog DAC1 Input Source Control (0x002b) */ +#define RT5668_M_ST_STO_L (0x1 << 9) +#define RT5668_M_ST_STO_L_SFT 9 +#define RT5668_M_ST_STO_R (0x1 << 8) +#define RT5668_M_ST_STO_R_SFT 8 +#define RT5668_DAC_L1_SRC_MASK (0x3 << 4) +#define RT5668_A_DACL1_SFT 4 +#define RT5668_DAC_R1_SRC_MASK (0x3) +#define RT5668_A_DACR1_SFT 0 + +/* Digital Interface Data Control (0x0030) */ +#define RT5668_IF2_ADC_SEL_MASK (0x3 << 0) +#define RT5668_IF2_ADC_SEL_SFT 0 + +/* REC Left Mixer Control 2 (0x003c) */ +#define RT5668_G_CBJ_RM1_L (0x7 << 10) +#define RT5668_G_CBJ_RM1_L_SFT 10 +#define RT5668_M_CBJ_RM1_L (0x1 << 7) +#define RT5668_M_CBJ_RM1_L_SFT 7 + +/* Power Management for Digital 1 (0x0061) */ +#define RT5668_PWR_I2S1 (0x1 << 15) +#define RT5668_PWR_I2S1_BIT 15 +#define RT5668_PWR_I2S2 (0x1 << 14) +#define RT5668_PWR_I2S2_BIT 14 +#define RT5668_PWR_DAC_L1 (0x1 << 11) +#define RT5668_PWR_DAC_L1_BIT 11 +#define RT5668_PWR_DAC_R1 (0x1 << 10) +#define RT5668_PWR_DAC_R1_BIT 10 +#define RT5668_PWR_LDO (0x1 << 8) +#define RT5668_PWR_LDO_BIT 8 +#define RT5668_PWR_ADC_L1 (0x1 << 4) +#define RT5668_PWR_ADC_L1_BIT 4 +#define RT5668_PWR_ADC_R1 (0x1 << 3) +#define RT5668_PWR_ADC_R1_BIT 3 +#define RT5668_DIG_GATE_CTRL (0x1 << 0) +#define RT5668_DIG_GATE_CTRL_SFT 0 + + +/* Power Management for Digital 2 (0x0062) */ +#define RT5668_PWR_ADC_S1F (0x1 << 15) +#define RT5668_PWR_ADC_S1F_BIT 15 +#define RT5668_PWR_DAC_S1F (0x1 << 10) +#define RT5668_PWR_DAC_S1F_BIT 10 + +/* Power Management for Analog 1 (0x0063) */ +#define RT5668_PWR_VREF1 (0x1 << 15) +#define RT5668_PWR_VREF1_BIT 15 +#define RT5668_PWR_FV1 (0x1 << 14) +#define RT5668_PWR_FV1_BIT 14 +#define RT5668_PWR_VREF2 (0x1 << 13) +#define RT5668_PWR_VREF2_BIT 13 +#define RT5668_PWR_FV2 (0x1 << 12) +#define RT5668_PWR_FV2_BIT 12 +#define RT5668_LDO1_DBG_MASK (0x3 << 10) +#define RT5668_PWR_MB (0x1 << 9) +#define RT5668_PWR_MB_BIT 9 +#define RT5668_PWR_BG (0x1 << 7) +#define RT5668_PWR_BG_BIT 7 +#define RT5668_LDO1_BYPASS_MASK (0x1 << 6) +#define RT5668_LDO1_BYPASS (0x1 << 6) +#define RT5668_LDO1_NOT_BYPASS (0x0 << 6) +#define RT5668_PWR_MA_BIT 6 +#define RT5668_LDO1_DVO_MASK (0x3 << 4) +#define RT5668_LDO1_DVO_09 (0x0 << 4) +#define RT5668_LDO1_DVO_10 (0x1 << 4) +#define RT5668_LDO1_DVO_12 (0x2 << 4) +#define RT5668_LDO1_DVO_14 (0x3 << 4) +#define RT5668_HP_DRIVER_MASK (0x3 << 2) +#define RT5668_HP_DRIVER_1X (0x0 << 2) +#define RT5668_HP_DRIVER_3X (0x1 << 2) +#define RT5668_HP_DRIVER_5X (0x3 << 2) +#define RT5668_PWR_HA_L (0x1 << 1) +#define RT5668_PWR_HA_L_BIT 1 +#define RT5668_PWR_HA_R (0x1 << 0) +#define RT5668_PWR_HA_R_BIT 0 + +/* Power Management for Analog 2 (0x0064) */ +#define RT5668_PWR_MB1 (0x1 << 11) +#define RT5668_PWR_MB1_PWR_DOWN (0x0 << 11) +#define RT5668_PWR_MB1_BIT 11 +#define RT5668_PWR_MB2 (0x1 << 10) +#define RT5668_PWR_MB2_PWR_DOWN (0x0 << 10) +#define RT5668_PWR_MB2_BIT 10 +#define RT5668_PWR_JDH (0x1 << 3) +#define RT5668_PWR_JDH_BIT 3 +#define RT5668_PWR_JDL (0x1 << 2) +#define RT5668_PWR_JDL_BIT 2 +#define RT5668_PWR_RM1_L (0x1 << 1) +#define RT5668_PWR_RM1_L_BIT 1 + +/* Power Management for Analog 3 (0x0065) */ +#define RT5668_PWR_CBJ (0x1 << 9) +#define RT5668_PWR_CBJ_BIT 9 +#define RT5668_PWR_PLL (0x1 << 6) +#define RT5668_PWR_PLL_BIT 6 +#define RT5668_PWR_PLL2B (0x1 << 5) +#define RT5668_PWR_PLL2B_BIT 5 +#define RT5668_PWR_PLL2F (0x1 << 4) +#define RT5668_PWR_PLL2F_BIT 4 +#define RT5668_PWR_LDO2 (0x1 << 2) +#define RT5668_PWR_LDO2_BIT 2 +#define RT5668_PWR_DET_SPKVDD (0x1 << 1) +#define RT5668_PWR_DET_SPKVDD_BIT 1 + +/* Power Management for Mixer (0x0066) */ +#define RT5668_PWR_STO1_DAC_L (0x1 << 5) +#define RT5668_PWR_STO1_DAC_L_BIT 5 +#define RT5668_PWR_STO1_DAC_R (0x1 << 4) +#define RT5668_PWR_STO1_DAC_R_BIT 4 + +/* MCLK and System Clock Detection Control (0x006b) */ +#define RT5668_SYS_CLK_DET (0x1 << 15) +#define RT5668_SYS_CLK_DET_SFT 15 +#define RT5668_PLL1_CLK_DET (0x1 << 14) +#define RT5668_PLL1_CLK_DET_SFT 14 +#define RT5668_PLL2_CLK_DET (0x1 << 13) +#define RT5668_PLL2_CLK_DET_SFT 13 +#define RT5668_POW_CLK_DET2_SFT 8 +#define RT5668_POW_CLK_DET_SFT 0 + +/* Digital Microphone Control 1 (0x006e) */ +#define RT5668_DMIC_1_EN_MASK (0x1 << 15) +#define RT5668_DMIC_1_EN_SFT 15 +#define RT5668_DMIC_1_DIS (0x0 << 15) +#define RT5668_DMIC_1_EN (0x1 << 15) +#define RT5668_DMIC_1_DP_MASK (0x3 << 4) +#define RT5668_DMIC_1_DP_SFT 4 +#define RT5668_DMIC_1_DP_GPIO2 (0x0 << 4) +#define RT5668_DMIC_1_DP_GPIO5 (0x1 << 4) +#define RT5668_DMIC_CLK_MASK (0xf << 0) +#define RT5668_DMIC_CLK_SFT 0 + +/* I2S1 Audio Serial Data Port Control (0x0070) */ +#define RT5668_SEL_ADCDAT_MASK (0x1 << 15) +#define RT5668_SEL_ADCDAT_OUT (0x0 << 15) +#define RT5668_SEL_ADCDAT_IN (0x1 << 15) +#define RT5668_SEL_ADCDAT_SFT 15 +#define RT5668_I2S1_TX_CHL_MASK (0x7 << 12) +#define RT5668_I2S1_TX_CHL_SFT 12 +#define RT5668_I2S1_TX_CHL_16 (0x0 << 12) +#define RT5668_I2S1_TX_CHL_20 (0x1 << 12) +#define RT5668_I2S1_TX_CHL_24 (0x2 << 12) +#define RT5668_I2S1_TX_CHL_32 (0x3 << 12) +#define RT5668_I2S1_TX_CHL_8 (0x4 << 12) +#define RT5668_I2S1_RX_CHL_MASK (0x7 << 8) +#define RT5668_I2S1_RX_CHL_SFT 8 +#define RT5668_I2S1_RX_CHL_16 (0x0 << 8) +#define RT5668_I2S1_RX_CHL_20 (0x1 << 8) +#define RT5668_I2S1_RX_CHL_24 (0x2 << 8) +#define RT5668_I2S1_RX_CHL_32 (0x3 << 8) +#define RT5668_I2S1_RX_CHL_8 (0x4 << 8) +#define RT5668_I2S1_MONO_MASK (0x1 << 7) +#define RT5668_I2S1_MONO_EN (0x1 << 7) +#define RT5668_I2S1_MONO_DIS (0x0 << 7) +#define RT5668_I2S2_MONO_MASK (0x1 << 6) +#define RT5668_I2S2_MONO_EN (0x1 << 6) +#define RT5668_I2S2_MONO_DIS (0x0 << 6) +#define RT5668_I2S1_DL_MASK (0x7 << 4) +#define RT5668_I2S1_DL_SFT 4 +#define RT5668_I2S1_DL_16 (0x0 << 4) +#define RT5668_I2S1_DL_20 (0x1 << 4) +#define RT5668_I2S1_DL_24 (0x2 << 4) +#define RT5668_I2S1_DL_32 (0x3 << 4) +#define RT5668_I2S1_DL_8 (0x4 << 4) + +/* I2S1/2 Audio Serial Data Port Control (0x0070)(0x0071) */ +#define RT5668_I2S2_MS_MASK (0x1 << 15) +#define RT5668_I2S2_MS_SFT 15 +#define RT5668_I2S2_MS_M (0x0 << 15) +#define RT5668_I2S2_MS_S (0x1 << 15) +#define RT5668_I2S2_PIN_CFG_MASK (0x1 << 14) +#define RT5668_I2S2_PIN_CFG_SFT 14 +#define RT5668_I2S2_CLK_SEL_MASK (0x1 << 11) +#define RT5668_I2S2_CLK_SEL_SFT 11 +#define RT5668_I2S2_OUT_MASK (0x1 << 9) +#define RT5668_I2S2_OUT_SFT 9 +#define RT5668_I2S2_OUT_UM (0x0 << 9) +#define RT5668_I2S2_OUT_M (0x1 << 9) +#define RT5668_I2S_BP_MASK (0x1 << 8) +#define RT5668_I2S_BP_SFT 8 +#define RT5668_I2S_BP_NOR (0x0 << 8) +#define RT5668_I2S_BP_INV (0x1 << 8) +#define RT5668_I2S2_MONO_EN (0x1 << 6) +#define RT5668_I2S2_MONO_DIS (0x0 << 6) +#define RT5668_I2S2_DL_MASK (0x3 << 4) +#define RT5668_I2S2_DL_SFT 4 +#define RT5668_I2S2_DL_16 (0x0 << 4) +#define RT5668_I2S2_DL_20 (0x1 << 4) +#define RT5668_I2S2_DL_24 (0x2 << 4) +#define RT5668_I2S2_DL_8 (0x3 << 4) +#define RT5668_I2S_DF_MASK (0x7) +#define RT5668_I2S_DF_SFT 0 +#define RT5668_I2S_DF_I2S (0x0) +#define RT5668_I2S_DF_LEFT (0x1) +#define RT5668_I2S_DF_PCM_A (0x2) +#define RT5668_I2S_DF_PCM_B (0x3) +#define RT5668_I2S_DF_PCM_A_N (0x6) +#define RT5668_I2S_DF_PCM_B_N (0x7) + +/* ADC/DAC Clock Control 1 (0x0073) */ +#define RT5668_ADC_OSR_MASK (0xf << 12) +#define RT5668_ADC_OSR_SFT 12 +#define RT5668_ADC_OSR_D_1 (0x0 << 12) +#define RT5668_ADC_OSR_D_2 (0x1 << 12) +#define RT5668_ADC_OSR_D_4 (0x2 << 12) +#define RT5668_ADC_OSR_D_6 (0x3 << 12) +#define RT5668_ADC_OSR_D_8 (0x4 << 12) +#define RT5668_ADC_OSR_D_12 (0x5 << 12) +#define RT5668_ADC_OSR_D_16 (0x6 << 12) +#define RT5668_ADC_OSR_D_24 (0x7 << 12) +#define RT5668_ADC_OSR_D_32 (0x8 << 12) +#define RT5668_ADC_OSR_D_48 (0x9 << 12) +#define RT5668_I2S_M_DIV_MASK (0xf << 12) +#define RT5668_I2S_M_DIV_SFT 8 +#define RT5668_I2S_M_D_1 (0x0 << 8) +#define RT5668_I2S_M_D_2 (0x1 << 8) +#define RT5668_I2S_M_D_3 (0x2 << 8) +#define RT5668_I2S_M_D_4 (0x3 << 8) +#define RT5668_I2S_M_D_6 (0x4 << 8) +#define RT5668_I2S_M_D_8 (0x5 << 8) +#define RT5668_I2S_M_D_12 (0x6 << 8) +#define RT5668_I2S_M_D_16 (0x7 << 8) +#define RT5668_I2S_M_D_24 (0x8 << 8) +#define RT5668_I2S_M_D_32 (0x9 << 8) +#define RT5668_I2S_M_D_48 (0x10 << 8) +#define RT5668_I2S_CLK_SRC_MASK (0x7 << 4) +#define RT5668_I2S_CLK_SRC_SFT 4 +#define RT5668_I2S_CLK_SRC_MCLK (0x0 << 4) +#define RT5668_I2S_CLK_SRC_PLL1 (0x1 << 4) +#define RT5668_I2S_CLK_SRC_PLL2 (0x2 << 4) +#define RT5668_I2S_CLK_SRC_SDW (0x3 << 4) +#define RT5668_I2S_CLK_SRC_RCCLK (0x4 << 4) /* 25M */ +#define RT5668_DAC_OSR_MASK (0xf << 0) +#define RT5668_DAC_OSR_SFT 0 +#define RT5668_DAC_OSR_D_1 (0x0 << 0) +#define RT5668_DAC_OSR_D_2 (0x1 << 0) +#define RT5668_DAC_OSR_D_4 (0x2 << 0) +#define RT5668_DAC_OSR_D_6 (0x3 << 0) +#define RT5668_DAC_OSR_D_8 (0x4 << 0) +#define RT5668_DAC_OSR_D_12 (0x5 << 0) +#define RT5668_DAC_OSR_D_16 (0x6 << 0) +#define RT5668_DAC_OSR_D_24 (0x7 << 0) +#define RT5668_DAC_OSR_D_32 (0x8 << 0) +#define RT5668_DAC_OSR_D_48 (0x9 << 0) + +/* ADC/DAC Clock Control 2 (0x0074) */ +#define RT5668_I2S2_BCLK_MS2_MASK (0x1 << 11) +#define RT5668_I2S2_BCLK_MS2_SFT 11 +#define RT5668_I2S2_BCLK_MS2_32 (0x0 << 11) +#define RT5668_I2S2_BCLK_MS2_64 (0x1 << 11) + + +/* TDM control 1 (0x0079) */ +#define RT5668_TDM_TX_CH_MASK (0x3 << 12) +#define RT5668_TDM_TX_CH_2 (0x0 << 12) +#define RT5668_TDM_TX_CH_4 (0x1 << 12) +#define RT5668_TDM_TX_CH_6 (0x2 << 12) +#define RT5668_TDM_TX_CH_8 (0x3 << 12) +#define RT5668_TDM_RX_CH_MASK (0x3 << 8) +#define RT5668_TDM_RX_CH_2 (0x0 << 8) +#define RT5668_TDM_RX_CH_4 (0x1 << 8) +#define RT5668_TDM_RX_CH_6 (0x2 << 8) +#define RT5668_TDM_RX_CH_8 (0x3 << 8) +#define RT5668_TDM_ADC_LCA_MASK (0xf << 4) +#define RT5668_TDM_ADC_LCA_SFT 4 +#define RT5668_TDM_ADC_DL_SFT 0 + +/* TDM control 3 (0x007a) */ +#define RT5668_IF1_ADC1_SEL_SFT 14 +#define RT5668_IF1_ADC2_SEL_SFT 12 +#define RT5668_IF1_ADC3_SEL_SFT 10 +#define RT5668_IF1_ADC4_SEL_SFT 8 +#define RT5668_TDM_ADC_SEL_SFT 4 + +/* TDM/I2S control (0x007e) */ +#define RT5668_TDM_S_BP_MASK (0x1 << 15) +#define RT5668_TDM_S_BP_SFT 15 +#define RT5668_TDM_S_BP_NOR (0x0 << 15) +#define RT5668_TDM_S_BP_INV (0x1 << 15) +#define RT5668_TDM_S_LP_MASK (0x1 << 14) +#define RT5668_TDM_S_LP_SFT 14 +#define RT5668_TDM_S_LP_NOR (0x0 << 14) +#define RT5668_TDM_S_LP_INV (0x1 << 14) +#define RT5668_TDM_DF_MASK (0x7 << 11) +#define RT5668_TDM_DF_SFT 11 +#define RT5668_TDM_DF_I2S (0x0 << 11) +#define RT5668_TDM_DF_LEFT (0x1 << 11) +#define RT5668_TDM_DF_PCM_A (0x2 << 11) +#define RT5668_TDM_DF_PCM_B (0x3 << 11) +#define RT5668_TDM_DF_PCM_A_N (0x6 << 11) +#define RT5668_TDM_DF_PCM_B_N (0x7 << 11) +#define RT5668_TDM_CL_MASK (0x3 << 4) +#define RT5668_TDM_CL_16 (0x0 << 4) +#define RT5668_TDM_CL_20 (0x1 << 4) +#define RT5668_TDM_CL_24 (0x2 << 4) +#define RT5668_TDM_CL_32 (0x3 << 4) +#define RT5668_TDM_M_BP_MASK (0x1 << 2) +#define RT5668_TDM_M_BP_SFT 2 +#define RT5668_TDM_M_BP_NOR (0x0 << 2) +#define RT5668_TDM_M_BP_INV (0x1 << 2) +#define RT5668_TDM_M_LP_MASK (0x1 << 1) +#define RT5668_TDM_M_LP_SFT 1 +#define RT5668_TDM_M_LP_NOR (0x0 << 1) +#define RT5668_TDM_M_LP_INV (0x1 << 1) +#define RT5668_TDM_MS_MASK (0x1 << 0) +#define RT5668_TDM_MS_SFT 0 +#define RT5668_TDM_MS_M (0x0 << 0) +#define RT5668_TDM_MS_S (0x1 << 0) + +/* Global Clock Control (0x0080) */ +#define RT5668_SCLK_SRC_MASK (0x7 << 13) +#define RT5668_SCLK_SRC_SFT 13 +#define RT5668_SCLK_SRC_MCLK (0x0 << 13) +#define RT5668_SCLK_SRC_PLL1 (0x1 << 13) +#define RT5668_SCLK_SRC_PLL2 (0x2 << 13) +#define RT5668_SCLK_SRC_SDW (0x3 << 13) +#define RT5668_SCLK_SRC_RCCLK (0x4 << 13) +#define RT5668_PLL1_SRC_MASK (0x3 << 10) +#define RT5668_PLL1_SRC_SFT 10 +#define RT5668_PLL1_SRC_MCLK (0x0 << 10) +#define RT5668_PLL1_SRC_BCLK1 (0x1 << 10) +#define RT5668_PLL1_SRC_SDW (0x2 << 10) +#define RT5668_PLL1_SRC_RC (0x3 << 10) +#define RT5668_PLL2_SRC_MASK (0x3 << 8) +#define RT5668_PLL2_SRC_SFT 8 +#define RT5668_PLL2_SRC_MCLK (0x0 << 8) +#define RT5668_PLL2_SRC_BCLK1 (0x1 << 8) +#define RT5668_PLL2_SRC_SDW (0x2 << 8) +#define RT5668_PLL2_SRC_RC (0x3 << 8) + + + +#define RT5668_PLL_INP_MAX 40000000 +#define RT5668_PLL_INP_MIN 256000 +/* PLL M/N/K Code Control 1 (0x0081) */ +#define RT5668_PLL_N_MAX 0x001ff +#define RT5668_PLL_N_MASK (RT5668_PLL_N_MAX << 7) +#define RT5668_PLL_N_SFT 7 +#define RT5668_PLL_K_MAX 0x001f +#define RT5668_PLL_K_MASK (RT5668_PLL_K_MAX) +#define RT5668_PLL_K_SFT 0 + +/* PLL M/N/K Code Control 2 (0x0082) */ +#define RT5668_PLL_M_MAX 0x00f +#define RT5668_PLL_M_MASK (RT5668_PLL_M_MAX << 12) +#define RT5668_PLL_M_SFT 12 +#define RT5668_PLL_M_BP (0x1 << 11) +#define RT5668_PLL_M_BP_SFT 11 +#define RT5668_PLL_K_BP (0x1 << 10) +#define RT5668_PLL_K_BP_SFT 10 + +/* PLL tracking mode 1 (0x0083) */ +#define RT5668_DA_ASRC_MASK (0x1 << 13) +#define RT5668_DA_ASRC_SFT 13 +#define RT5668_DAC_STO1_ASRC_MASK (0x1 << 12) +#define RT5668_DAC_STO1_ASRC_SFT 12 +#define RT5668_AD_ASRC_MASK (0x1 << 8) +#define RT5668_AD_ASRC_SFT 8 +#define RT5668_AD_ASRC_SEL_MASK (0x1 << 4) +#define RT5668_AD_ASRC_SEL_SFT 4 +#define RT5668_DMIC_ASRC_MASK (0x1 << 3) +#define RT5668_DMIC_ASRC_SFT 3 +#define RT5668_ADC_STO1_ASRC_MASK (0x1 << 2) +#define RT5668_ADC_STO1_ASRC_SFT 2 +#define RT5668_DA_ASRC_SEL_MASK (0x1 << 0) +#define RT5668_DA_ASRC_SEL_SFT 0 + +/* PLL tracking mode 2 3 (0x0084)(0x0085)*/ +#define RT5668_FILTER_CLK_SEL_MASK (0x7 << 12) +#define RT5668_FILTER_CLK_SEL_SFT 12 + +/* ASRC Control 4 (0x0086) */ +#define RT5668_ASRCIN_FTK_N1_MASK (0x3 << 14) +#define RT5668_ASRCIN_FTK_N1_SFT 14 +#define RT5668_ASRCIN_FTK_N2_MASK (0x3 << 12) +#define RT5668_ASRCIN_FTK_N2_SFT 12 +#define RT5668_ASRCIN_FTK_M1_MASK (0x7 << 8) +#define RT5668_ASRCIN_FTK_M1_SFT 8 +#define RT5668_ASRCIN_FTK_M2_MASK (0x7 << 4) +#define RT5668_ASRCIN_FTK_M2_SFT 4 + +/* SoundWire reference clk (0x008d) */ +#define RT5668_PLL2_OUT_MASK (0x1 << 8) +#define RT5668_PLL2_OUT_98M (0x0 << 8) +#define RT5668_PLL2_OUT_49M (0x1 << 8) +#define RT5668_SDW_REF_2_MASK (0xf << 4) +#define RT5668_SDW_REF_2_SFT 4 +#define RT5668_SDW_REF_2_48K (0x0 << 4) +#define RT5668_SDW_REF_2_96K (0x1 << 4) +#define RT5668_SDW_REF_2_192K (0x2 << 4) +#define RT5668_SDW_REF_2_32K (0x3 << 4) +#define RT5668_SDW_REF_2_24K (0x4 << 4) +#define RT5668_SDW_REF_2_16K (0x5 << 4) +#define RT5668_SDW_REF_2_12K (0x6 << 4) +#define RT5668_SDW_REF_2_8K (0x7 << 4) +#define RT5668_SDW_REF_2_44K (0x8 << 4) +#define RT5668_SDW_REF_2_88K (0x9 << 4) +#define RT5668_SDW_REF_2_176K (0xa << 4) +#define RT5668_SDW_REF_2_353K (0xb << 4) +#define RT5668_SDW_REF_2_22K (0xc << 4) +#define RT5668_SDW_REF_2_384K (0xd << 4) +#define RT5668_SDW_REF_2_11K (0xe << 4) +#define RT5668_SDW_REF_1_MASK (0xf << 0) +#define RT5668_SDW_REF_1_SFT 0 +#define RT5668_SDW_REF_1_48K (0x0 << 0) +#define RT5668_SDW_REF_1_96K (0x1 << 0) +#define RT5668_SDW_REF_1_192K (0x2 << 0) +#define RT5668_SDW_REF_1_32K (0x3 << 0) +#define RT5668_SDW_REF_1_24K (0x4 << 0) +#define RT5668_SDW_REF_1_16K (0x5 << 0) +#define RT5668_SDW_REF_1_12K (0x6 << 0) +#define RT5668_SDW_REF_1_8K (0x7 << 0) +#define RT5668_SDW_REF_1_44K (0x8 << 0) +#define RT5668_SDW_REF_1_88K (0x9 << 0) +#define RT5668_SDW_REF_1_176K (0xa << 0) +#define RT5668_SDW_REF_1_353K (0xb << 0) +#define RT5668_SDW_REF_1_22K (0xc << 0) +#define RT5668_SDW_REF_1_384K (0xd << 0) +#define RT5668_SDW_REF_1_11K (0xe << 0) + +/* Depop Mode Control 1 (0x008e) */ +#define RT5668_PUMP_EN (0x1 << 3) +#define RT5668_PUMP_EN_SFT 3 +#define RT5668_CAPLESS_EN (0x1 << 0) +#define RT5668_CAPLESS_EN_SFT 0 + +/* Depop Mode Control 2 (0x8f) */ +#define RT5668_RAMP_MASK (0x1 << 12) +#define RT5668_RAMP_SFT 12 +#define RT5668_RAMP_DIS (0x0 << 12) +#define RT5668_RAMP_EN (0x1 << 12) +#define RT5668_BPS_MASK (0x1 << 11) +#define RT5668_BPS_SFT 11 +#define RT5668_BPS_DIS (0x0 << 11) +#define RT5668_BPS_EN (0x1 << 11) +#define RT5668_FAST_UPDN_MASK (0x1 << 10) +#define RT5668_FAST_UPDN_SFT 10 +#define RT5668_FAST_UPDN_DIS (0x0 << 10) +#define RT5668_FAST_UPDN_EN (0x1 << 10) +#define RT5668_VLO_MASK (0x1 << 7) +#define RT5668_VLO_SFT 7 +#define RT5668_VLO_3V (0x0 << 7) +#define RT5668_VLO_33V (0x1 << 7) + +/* HPOUT charge pump 1 (0x0091) */ +#define RT5668_OSW_L_MASK (0x1 << 11) +#define RT5668_OSW_L_SFT 11 +#define RT5668_OSW_L_DIS (0x0 << 11) +#define RT5668_OSW_L_EN (0x1 << 11) +#define RT5668_OSW_R_MASK (0x1 << 10) +#define RT5668_OSW_R_SFT 10 +#define RT5668_OSW_R_DIS (0x0 << 10) +#define RT5668_OSW_R_EN (0x1 << 10) +#define RT5668_PM_HP_MASK (0x3 << 8) +#define RT5668_PM_HP_SFT 8 +#define RT5668_PM_HP_LV (0x0 << 8) +#define RT5668_PM_HP_MV (0x1 << 8) +#define RT5668_PM_HP_HV (0x2 << 8) +#define RT5668_IB_HP_MASK (0x3 << 6) +#define RT5668_IB_HP_SFT 6 +#define RT5668_IB_HP_125IL (0x0 << 6) +#define RT5668_IB_HP_25IL (0x1 << 6) +#define RT5668_IB_HP_5IL (0x2 << 6) +#define RT5668_IB_HP_1IL (0x3 << 6) + +/* Micbias Control1 (0x93) */ +#define RT5668_MIC1_OV_MASK (0x3 << 14) +#define RT5668_MIC1_OV_SFT 14 +#define RT5668_MIC1_OV_2V7 (0x0 << 14) +#define RT5668_MIC1_OV_2V4 (0x1 << 14) +#define RT5668_MIC1_OV_2V25 (0x3 << 14) +#define RT5668_MIC1_OV_1V8 (0x4 << 14) +#define RT5668_MIC1_CLK_MASK (0x1 << 13) +#define RT5668_MIC1_CLK_SFT 13 +#define RT5668_MIC1_CLK_DIS (0x0 << 13) +#define RT5668_MIC1_CLK_EN (0x1 << 13) +#define RT5668_MIC1_OVCD_MASK (0x1 << 12) +#define RT5668_MIC1_OVCD_SFT 12 +#define RT5668_MIC1_OVCD_DIS (0x0 << 12) +#define RT5668_MIC1_OVCD_EN (0x1 << 12) +#define RT5668_MIC1_OVTH_MASK (0x3 << 10) +#define RT5668_MIC1_OVTH_SFT 10 +#define RT5668_MIC1_OVTH_768UA (0x0 << 10) +#define RT5668_MIC1_OVTH_960UA (0x1 << 10) +#define RT5668_MIC1_OVTH_1152UA (0x2 << 10) +#define RT5668_MIC1_OVTH_1960UA (0x3 << 10) +#define RT5668_MIC2_OV_MASK (0x3 << 8) +#define RT5668_MIC2_OV_SFT 8 +#define RT5668_MIC2_OV_2V7 (0x0 << 8) +#define RT5668_MIC2_OV_2V4 (0x1 << 8) +#define RT5668_MIC2_OV_2V25 (0x3 << 8) +#define RT5668_MIC2_OV_1V8 (0x4 << 8) +#define RT5668_MIC2_CLK_MASK (0x1 << 7) +#define RT5668_MIC2_CLK_SFT 7 +#define RT5668_MIC2_CLK_DIS (0x0 << 7) +#define RT5668_MIC2_CLK_EN (0x1 << 7) +#define RT5668_MIC2_OVTH_MASK (0x3 << 4) +#define RT5668_MIC2_OVTH_SFT 4 +#define RT5668_MIC2_OVTH_768UA (0x0 << 4) +#define RT5668_MIC2_OVTH_960UA (0x1 << 4) +#define RT5668_MIC2_OVTH_1152UA (0x2 << 4) +#define RT5668_MIC2_OVTH_1960UA (0x3 << 4) +#define RT5668_PWR_MB_MASK (0x1 << 3) +#define RT5668_PWR_MB_SFT 3 +#define RT5668_PWR_MB_PD (0x0 << 3) +#define RT5668_PWR_MB_PU (0x1 << 3) + +/* Micbias Control2 (0x0094) */ +#define RT5668_PWR_CLK25M_MASK (0x1 << 9) +#define RT5668_PWR_CLK25M_SFT 9 +#define RT5668_PWR_CLK25M_PD (0x0 << 9) +#define RT5668_PWR_CLK25M_PU (0x1 << 9) +#define RT5668_PWR_CLK1M_MASK (0x1 << 8) +#define RT5668_PWR_CLK1M_SFT 8 +#define RT5668_PWR_CLK1M_PD (0x0 << 8) +#define RT5668_PWR_CLK1M_PU (0x1 << 8) + +/* RC Clock Control (0x009f) */ +#define RT5668_POW_IRQ (0x1 << 15) +#define RT5668_POW_JDH (0x1 << 14) +#define RT5668_POW_JDL (0x1 << 13) +#define RT5668_POW_ANA (0x1 << 12) + +/* I2S Master Mode Clock Control 1 (0x00a0) */ +#define RT5668_CLK_SRC_MCLK (0x0) +#define RT5668_CLK_SRC_PLL1 (0x1) +#define RT5668_CLK_SRC_PLL2 (0x2) +#define RT5668_CLK_SRC_SDW (0x3) +#define RT5668_CLK_SRC_RCCLK (0x4) +#define RT5668_I2S_PD_1 (0x0) +#define RT5668_I2S_PD_2 (0x1) +#define RT5668_I2S_PD_3 (0x2) +#define RT5668_I2S_PD_4 (0x3) +#define RT5668_I2S_PD_6 (0x4) +#define RT5668_I2S_PD_8 (0x5) +#define RT5668_I2S_PD_12 (0x6) +#define RT5668_I2S_PD_16 (0x7) +#define RT5668_I2S_PD_24 (0x8) +#define RT5668_I2S_PD_32 (0x9) +#define RT5668_I2S_PD_48 (0xa) +#define RT5668_I2S2_SRC_MASK (0x3 << 4) +#define RT5668_I2S2_SRC_SFT 4 +#define RT5668_I2S2_M_PD_MASK (0xf << 0) +#define RT5668_I2S2_M_PD_SFT 0 + +/* IRQ Control 1 (0x00b6) */ +#define RT5668_JD1_PULSE_EN_MASK (0x1 << 10) +#define RT5668_JD1_PULSE_EN_SFT 10 +#define RT5668_JD1_PULSE_DIS (0x0 << 10) +#define RT5668_JD1_PULSE_EN (0x1 << 10) + +/* IRQ Control 2 (0x00b7) */ +#define RT5668_JD1_EN_MASK (0x1 << 15) +#define RT5668_JD1_EN_SFT 15 +#define RT5668_JD1_DIS (0x0 << 15) +#define RT5668_JD1_EN (0x1 << 15) +#define RT5668_JD1_POL_MASK (0x1 << 13) +#define RT5668_JD1_POL_NOR (0x0 << 13) +#define RT5668_JD1_POL_INV (0x1 << 13) + +/* IRQ Control 3 (0x00b8) */ +#define RT5668_IL_IRQ_MASK (0x1 << 7) +#define RT5668_IL_IRQ_DIS (0x0 << 7) +#define RT5668_IL_IRQ_EN (0x1 << 7) + +/* GPIO Control 1 (0x00c0) */ +#define RT5668_GP1_PIN_MASK (0x3 << 14) +#define RT5668_GP1_PIN_SFT 14 +#define RT5668_GP1_PIN_GPIO1 (0x0 << 14) +#define RT5668_GP1_PIN_IRQ (0x1 << 14) +#define RT5668_GP1_PIN_DMIC_CLK (0x2 << 14) +#define RT5668_GP2_PIN_MASK (0x3 << 12) +#define RT5668_GP2_PIN_SFT 12 +#define RT5668_GP2_PIN_GPIO2 (0x0 << 12) +#define RT5668_GP2_PIN_LRCK2 (0x1 << 12) +#define RT5668_GP2_PIN_DMIC_SDA (0x2 << 12) +#define RT5668_GP3_PIN_MASK (0x3 << 10) +#define RT5668_GP3_PIN_SFT 10 +#define RT5668_GP3_PIN_GPIO3 (0x0 << 10) +#define RT5668_GP3_PIN_BCLK2 (0x1 << 10) +#define RT5668_GP3_PIN_DMIC_CLK (0x2 << 10) +#define RT5668_GP4_PIN_MASK (0x3 << 8) +#define RT5668_GP4_PIN_SFT 8 +#define RT5668_GP4_PIN_GPIO4 (0x0 << 8) +#define RT5668_GP4_PIN_ADCDAT1 (0x1 << 8) +#define RT5668_GP4_PIN_DMIC_CLK (0x2 << 8) +#define RT5668_GP4_PIN_ADCDAT2 (0x3 << 8) +#define RT5668_GP5_PIN_MASK (0x3 << 6) +#define RT5668_GP5_PIN_SFT 6 +#define RT5668_GP5_PIN_GPIO5 (0x0 << 6) +#define RT5668_GP5_PIN_DACDAT1 (0x1 << 6) +#define RT5668_GP5_PIN_DMIC_SDA (0x2 << 6) +#define RT5668_GP6_PIN_MASK (0x1 << 5) +#define RT5668_GP6_PIN_SFT 5 +#define RT5668_GP6_PIN_GPIO6 (0x0 << 5) +#define RT5668_GP6_PIN_LRCK1 (0x1 << 5) + +/* GPIO Control 2 (0x00c1)*/ +#define RT5668_GP1_PF_MASK (0x1 << 15) +#define RT5668_GP1_PF_IN (0x0 << 15) +#define RT5668_GP1_PF_OUT (0x1 << 15) +#define RT5668_GP1_OUT_MASK (0x1 << 14) +#define RT5668_GP1_OUT_L (0x0 << 14) +#define RT5668_GP1_OUT_H (0x1 << 14) +#define RT5668_GP2_PF_MASK (0x1 << 13) +#define RT5668_GP2_PF_IN (0x0 << 13) +#define RT5668_GP2_PF_OUT (0x1 << 13) +#define RT5668_GP2_OUT_MASK (0x1 << 12) +#define RT5668_GP2_OUT_L (0x0 << 12) +#define RT5668_GP2_OUT_H (0x1 << 12) +#define RT5668_GP3_PF_MASK (0x1 << 11) +#define RT5668_GP3_PF_IN (0x0 << 11) +#define RT5668_GP3_PF_OUT (0x1 << 11) +#define RT5668_GP3_OUT_MASK (0x1 << 10) +#define RT5668_GP3_OUT_L (0x0 << 10) +#define RT5668_GP3_OUT_H (0x1 << 10) +#define RT5668_GP4_PF_MASK (0x1 << 9) +#define RT5668_GP4_PF_IN (0x0 << 9) +#define RT5668_GP4_PF_OUT (0x1 << 9) +#define RT5668_GP4_OUT_MASK (0x1 << 8) +#define RT5668_GP4_OUT_L (0x0 << 8) +#define RT5668_GP4_OUT_H (0x1 << 8) +#define RT5668_GP5_PF_MASK (0x1 << 7) +#define RT5668_GP5_PF_IN (0x0 << 7) +#define RT5668_GP5_PF_OUT (0x1 << 7) +#define RT5668_GP5_OUT_MASK (0x1 << 6) +#define RT5668_GP5_OUT_L (0x0 << 6) +#define RT5668_GP5_OUT_H (0x1 << 6) +#define RT5668_GP6_PF_MASK (0x1 << 5) +#define RT5668_GP6_PF_IN (0x0 << 5) +#define RT5668_GP6_PF_OUT (0x1 << 5) +#define RT5668_GP6_OUT_MASK (0x1 << 4) +#define RT5668_GP6_OUT_L (0x0 << 4) +#define RT5668_GP6_OUT_H (0x1 << 4) + + +/* GPIO Status (0x00c2) */ +#define RT5668_GP6_STA (0x1 << 6) +#define RT5668_GP5_STA (0x1 << 5) +#define RT5668_GP4_STA (0x1 << 4) +#define RT5668_GP3_STA (0x1 << 3) +#define RT5668_GP2_STA (0x1 << 2) +#define RT5668_GP1_STA (0x1 << 1) + +/* Soft volume and zero cross control 1 (0x00d9) */ +#define RT5668_SV_MASK (0x1 << 15) +#define RT5668_SV_SFT 15 +#define RT5668_SV_DIS (0x0 << 15) +#define RT5668_SV_EN (0x1 << 15) +#define RT5668_ZCD_MASK (0x1 << 10) +#define RT5668_ZCD_SFT 10 +#define RT5668_ZCD_PD (0x0 << 10) +#define RT5668_ZCD_PU (0x1 << 10) +#define RT5668_SV_DLY_MASK (0xf) +#define RT5668_SV_DLY_SFT 0 + +/* Soft volume and zero cross control 2 (0x00da) */ +#define RT5668_ZCD_BST1_CBJ_MASK (0x1 << 7) +#define RT5668_ZCD_BST1_CBJ_SFT 7 +#define RT5668_ZCD_BST1_CBJ_DIS (0x0 << 7) +#define RT5668_ZCD_BST1_CBJ_EN (0x1 << 7) +#define RT5668_ZCD_RECMIX_MASK (0x1) +#define RT5668_ZCD_RECMIX_SFT 0 +#define RT5668_ZCD_RECMIX_DIS (0x0) +#define RT5668_ZCD_RECMIX_EN (0x1) + +/* 4 Button Inline Command Control 2 (0x00e3) */ +#define RT5668_4BTN_IL_MASK (0x1 << 15) +#define RT5668_4BTN_IL_EN (0x1 << 15) +#define RT5668_4BTN_IL_DIS (0x0 << 15) +#define RT5668_4BTN_IL_RST_MASK (0x1 << 14) +#define RT5668_4BTN_IL_NOR (0x1 << 14) +#define RT5668_4BTN_IL_RST (0x0 << 14) + +/* Analog JD Control (0x00f0) */ +#define RT5668_JDH_RS_MASK (0x1 << 4) +#define RT5668_JDH_NO_PLUG (0x1 << 4) +#define RT5668_JDH_PLUG (0x0 << 4) + +/* Chopper and Clock control for DAC (0x013a)*/ +#define RT5668_CKXEN_DAC1_MASK (0x1 << 13) +#define RT5668_CKXEN_DAC1_SFT 13 +#define RT5668_CKGEN_DAC1_MASK (0x1 << 12) +#define RT5668_CKGEN_DAC1_SFT 12 + +/* Chopper and Clock control for ADC (0x013b)*/ +#define RT5668_CKXEN_ADC1_MASK (0x1 << 13) +#define RT5668_CKXEN_ADC1_SFT 13 +#define RT5668_CKGEN_ADC1_MASK (0x1 << 12) +#define RT5668_CKGEN_ADC1_SFT 12 + +/* Volume test (0x013f)*/ +#define RT5668_SEL_CLK_VOL_MASK (0x1 << 15) +#define RT5668_SEL_CLK_VOL_EN (0x1 << 15) +#define RT5668_SEL_CLK_VOL_DIS (0x0 << 15) + +/* Test Mode Control 1 (0x0145) */ +#define RT5668_AD2DA_LB_MASK (0x1 << 10) +#define RT5668_AD2DA_LB_SFT 10 + +/* Stereo Noise Gate Control 1 (0x0160) */ +#define RT5668_NG2_EN_MASK (0x1 << 15) +#define RT5668_NG2_EN (0x1 << 15) +#define RT5668_NG2_DIS (0x0 << 15) + +/* Stereo1 DAC Silence Detection Control (0x0190) */ +#define RT5668_DEB_STO_DAC_MASK (0x7 << 4) +#define RT5668_DEB_80_MS (0x0 << 4) + +/* SAR ADC Inline Command Control 1 (0x0210) */ +#define RT5668_SAR_BUTT_DET_MASK (0x1 << 15) +#define RT5668_SAR_BUTT_DET_EN (0x1 << 15) +#define RT5668_SAR_BUTT_DET_DIS (0x0 << 15) +#define RT5668_SAR_BUTDET_MODE_MASK (0x1 << 14) +#define RT5668_SAR_BUTDET_POW_SAV (0x1 << 14) +#define RT5668_SAR_BUTDET_POW_NORM (0x0 << 14) +#define RT5668_SAR_BUTDET_RST_MASK (0x1 << 13) +#define RT5668_SAR_BUTDET_RST_NORMAL (0x1 << 13) +#define RT5668_SAR_BUTDET_RST (0x0 << 13) +#define RT5668_SAR_POW_MASK (0x1 << 12) +#define RT5668_SAR_POW_EN (0x1 << 12) +#define RT5668_SAR_POW_DIS (0x0 << 12) +#define RT5668_SAR_RST_MASK (0x1 << 11) +#define RT5668_SAR_RST_NORMAL (0x1 << 11) +#define RT5668_SAR_RST (0x0 << 11) +#define RT5668_SAR_BYPASS_MASK (0x1 << 10) +#define RT5668_SAR_BYPASS_EN (0x1 << 10) +#define RT5668_SAR_BYPASS_DIS (0x0 << 10) +#define RT5668_SAR_SEL_MB1_MASK (0x1 << 9) +#define RT5668_SAR_SEL_MB1_SEL (0x1 << 9) +#define RT5668_SAR_SEL_MB1_NOSEL (0x0 << 9) +#define RT5668_SAR_SEL_MB2_MASK (0x1 << 8) +#define RT5668_SAR_SEL_MB2_SEL (0x1 << 8) +#define RT5668_SAR_SEL_MB2_NOSEL (0x0 << 8) +#define RT5668_SAR_SEL_MODE_MASK (0x1 << 7) +#define RT5668_SAR_SEL_MODE_CMP (0x1 << 7) +#define RT5668_SAR_SEL_MODE_ADC (0x0 << 7) +#define RT5668_SAR_SEL_MB1_MB2_MASK (0x1 << 5) +#define RT5668_SAR_SEL_MB1_MB2_AUTO (0x1 << 5) +#define RT5668_SAR_SEL_MB1_MB2_MANU (0x0 << 5) +#define RT5668_SAR_SEL_SIGNAL_MASK (0x1 << 4) +#define RT5668_SAR_SEL_SIGNAL_AUTO (0x1 << 4) +#define RT5668_SAR_SEL_SIGNAL_MANU (0x0 << 4) + +/* SAR ADC Inline Command Control 13 (0x021c) */ +#define RT5668_SAR_SOUR_MASK (0x3f) +#define RT5668_SAR_SOUR_BTN (0x3f) +#define RT5668_SAR_SOUR_TYPE (0x0) + + +/* System Clock Source */ +enum { + RT5668_SCLK_S_MCLK, + RT5668_SCLK_S_PLL1, + RT5668_SCLK_S_PLL2, + RT5668_SCLK_S_RCCLK, +}; + +/* PLL Source */ +enum { + RT5668_PLL1_S_MCLK, + RT5668_PLL1_S_BCLK1, + RT5668_PLL1_S_RCCLK, +}; + +enum { + RT5668_AIF1, + RT5668_AIF2, + RT5668_AIFS +}; + +/* filter mask */ +enum { + RT5668_DA_STEREO1_FILTER = 0x1, + RT5668_AD_STEREO1_FILTER = (0x1 << 1), +}; + +enum { + RT5668_CLK_SEL_SYS, + RT5668_CLK_SEL_I2S1_ASRC, + RT5668_CLK_SEL_I2S2_ASRC, +}; + +int rt5668_sel_asrc_clk_src(struct snd_soc_component *component, + unsigned int filter_mask, unsigned int clk_src); + +#endif /* __RT5668_H__ */ -- 2.7.4

1 0

[alsa-devel] [PATCH] ASoC: nau8824: recover system clock when device changes
by John Hsu 22 Mar '18

22 Mar '18

User reports an issue in Ubuntu about the device switch upon playback. We find the FLL will disalbe when switching headphone to speaker. The pulseaudio will stop the headphone and close its power. Then, it just opens the speaker and turn on its power. Therefore, the supply of system clock does the OFF event and disables FLL. But the FLL doesn't enable again when the speaker powers on. The patch adds the recovery of system clock to enable FLL again for this case. And it covers the case that system clock from MCLK. Signed-off-by: John Hsu <KCHSU0(a)nuvoton.com> --- sound/soc/codecs/nau8824.c | 40 +++++++++++++++++++++++++++++++++++++++- 1 file changed, 39 insertions(+), 1 deletion(-) diff --git a/sound/soc/codecs/nau8824.c b/sound/soc/codecs/nau8824.c index 4d9148a..637e952 100644 --- a/sound/soc/codecs/nau8824.c +++ b/sound/soc/codecs/nau8824.c @@ -489,8 +489,12 @@ static int system_clock_control(struct snd_soc_dapm_widget *w, { struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct nau8824 *nau8824 = snd_soc_component_get_drvdata(component); + struct regmap *regmap = nau8824->regmap; + unsigned int value; + bool clk_fll, error; if (SND_SOC_DAPM_EVENT_OFF(event)) { + dev_dbg(nau8824->dev, "system clock control : POWER OFF\n"); /* Set clock source to disable or internal clock before the * playback or capture end. Codec needs clock for Jack * detection and button press if jack inserted; otherwise, @@ -502,7 +506,40 @@ static int system_clock_control(struct snd_soc_dapm_widget *w, } else { nau8824_config_sysclk(nau8824, NAU8824_CLK_DIS, 0); } + } else { + dev_dbg(nau8824->dev, "system clock control : POWER ON\n"); + /* Check the clock source setting is proper or not + * no matter the source is from FLL or MCLK. + */ + regmap_read(regmap, NAU8824_REG_FLL1, &value); + clk_fll = value & NAU8824_FLL_RATIO_MASK; + /* It's error to use internal clock when playback */ + regmap_read(regmap, NAU8824_REG_FLL6, &value); + error = value & NAU8824_DCO_EN; + if (!error) { + /* Check error depending on source is FLL or MCLK. */ + regmap_read(regmap, NAU8824_REG_CLK_DIVIDER, &value); + if (clk_fll) + error = !(value & NAU8824_CLK_SRC_VCO); + else + error = value & NAU8824_CLK_SRC_VCO; + } + /* Recover the clock source setting if error. */ + if (error) { + if (clk_fll) { + regmap_update_bits(regmap, + NAU8824_REG_FLL6, NAU8824_DCO_EN, 0); + regmap_update_bits(regmap, + NAU8824_REG_CLK_DIVIDER, + NAU8824_CLK_SRC_MASK, + NAU8824_CLK_SRC_VCO); + } else { + nau8824_config_sysclk(nau8824, + NAU8824_CLK_MCLK, 0); + } + } } + return 0; } @@ -591,7 +628,8 @@ static const struct snd_kcontrol_new nau8824_dacr_mux = static const struct snd_soc_dapm_widget nau8824_dapm_widgets[] = { SND_SOC_DAPM_SUPPLY("System Clock", SND_SOC_NOPM, 0, 0, - system_clock_control, SND_SOC_DAPM_POST_PMD), + system_clock_control, SND_SOC_DAPM_POST_PMD | + SND_SOC_DAPM_POST_PMU), SND_SOC_DAPM_INPUT("HSMIC1"), SND_SOC_DAPM_INPUT("HSMIC2"), -- 2.6.4

2 1

[alsa-devel] [PATCH RESEND, alsa-lib, v2] topology: Fix parsing config with multiple hw_configs
by Kirill Marinushkin 21 Mar '18

21 Mar '18

Currently, if the config file includes several hw_configs sections, parse_hw_config_refs() returns after parsing only the first section. For example, the following config, based on alsa-lib/src/conf/topology/broadwell/broadwell.conf, is parsed incorrectly: ~~~~ SectionHWConfig."CodecHWConfig" { id "1" format "I2S" # physical audio format. bclk "master" # Platform is master of bit clock fsync "master" # platform is master of fsync } SectionHWConfig."CodecHWConfig2" { id "2" format "AC97" } SectionLink."Codec" { # used for binding to the physical link id "0" hw_configs [ "CodecHWConfig" "CodecHWConfig2" ] default_hw_conf_id "2" } ~~~~ Signed-off-by: Kirill Marinushkin <k.marinushkin(a)gmail.com> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: alsa-devel(a)alsa-project.org --- src/topology/pcm.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/src/topology/pcm.c b/src/topology/pcm.c index 58cee31d..d3836677 100644 --- a/src/topology/pcm.c +++ b/src/topology/pcm.c @@ -882,6 +882,7 @@ static int parse_hw_config_refs(snd_tplg_t *tplg, snd_config_t *cfg, /* refer to a list of HW configs */ snd_config_for_each(i, next, cfg) { const char *val; + int err; n = snd_config_iterator_entry(i); if (snd_config_get_string(n, &val) < 0) @@ -893,7 +894,9 @@ static int parse_hw_config_refs(snd_tplg_t *tplg, snd_config_t *cfg, } link->num_hw_configs++; - return tplg_ref_add(elem, SND_TPLG_TYPE_HW_CONFIG, val); + err = tplg_ref_add(elem, SND_TPLG_TYPE_HW_CONFIG, val); + if (err < 0) + return err; } return 0; -- 2.13.6

2 1

[alsa-devel] [PATCH - JACK PCM plugin] jack: Do not Xrun the ALSA buffer
by twischer＠de.adit-jv.com 21 Mar '18

21 Mar '18

From: Timo Wischer <twischer(a)de.adit-jv.com> when the JACK thread is requesting too many audio frames Playback: Without this commit the ALSA audio buffer will be played with endless repeats as long as the user application has not provided new audio data. Therefore this garbage will be played as long as the user application has not called snd_pcm_stop() after an Xrun. With this fix the rest of the JACK buffer will be filled with silence. Capture: Without this commit the audio data in the ALSA buffer would be overwritten. With this commit the new data from the JACK buffer will not be copied. Therefore the existing data in the ALSA buffer will not be overwritten. Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> --- This patch is depending on http://mailman.alsa-project.org/pipermail/alsa-devel/2018-January/130986.ht… diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c index c22a5d0..a4f35f4 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -54,6 +54,68 @@ typedef struct { static int snd_pcm_jack_stop(snd_pcm_ioplug_t *io); + +static inline snd_pcm_uframes_t snd_pcm_jack_playback_avail(const snd_pcm_ioplug_t* const io, + const snd_pcm_uframes_t hw_ptr, + const snd_pcm_uframes_t appl_ptr) +{ + const snd_pcm_jack_t* const jack = io->private_data; + + /* cannot use io->hw_ptr without calling snd_pcm_avail_update() + * because it is not guranteed that snd_pcm_jack_pointer() was already + * called + */ + snd_pcm_sframes_t avail; + avail = hw_ptr + io->buffer_size - appl_ptr; + if (avail < 0) + avail += jack->boundary; + else if ((snd_pcm_uframes_t) avail >= jack->boundary) + avail -= jack->boundary; + return avail; +} + +static inline snd_pcm_uframes_t snd_pcm_jack_capture_avail(const snd_pcm_ioplug_t* const io, + const snd_pcm_uframes_t hw_ptr, + const snd_pcm_uframes_t appl_ptr) +{ + const snd_pcm_jack_t* const jack = io->private_data; + + /* cannot use io->hw_ptr without calling snd_pcm_avail_update() + * because it is not guranteed that snd_pcm_jack_pointer() was already + * called + */ + snd_pcm_sframes_t avail; + avail = hw_ptr - appl_ptr; + if (avail < 0) + avail += jack->boundary; + return avail; +} + +static inline snd_pcm_uframes_t snd_pcm_jack_avail(const snd_pcm_ioplug_t* const io, + const snd_pcm_uframes_t hw_ptr, + const snd_pcm_uframes_t appl_ptr) +{ + return (io->stream == SND_PCM_STREAM_PLAYBACK) ? + snd_pcm_jack_playback_avail(io, hw_ptr, appl_ptr) : + snd_pcm_jack_capture_avail(io, hw_ptr, appl_ptr); +} + +static inline snd_pcm_uframes_t snd_pcm_jack_hw_avail(const snd_pcm_ioplug_t* const io, + const snd_pcm_uframes_t hw_ptr, + const snd_pcm_uframes_t appl_ptr) +{ + /* available data/space which can be transfered by the user application */ + const snd_pcm_uframes_t user_avail = snd_pcm_jack_avail(io, hw_ptr, + appl_ptr); + + if (user_avail > io->buffer_size) { + /* there was an Xrun */ + return 0; + } + /* available data/space which can be transfered by the DMA */ + return io->buffer_size - user_avail; +} + static int pcm_poll_block_check(snd_pcm_ioplug_t *io) { static char buf[32]; @@ -154,7 +216,6 @@ snd_pcm_jack_process_cb(jack_nframes_t nframes, snd_pcm_ioplug_t *io) { snd_pcm_jack_t *jack = io->private_data; snd_pcm_uframes_t hw_ptr; - const snd_pcm_channel_area_t *areas; snd_pcm_uframes_t xfer = 0; unsigned int channel; @@ -164,40 +225,57 @@ snd_pcm_jack_process_cb(jack_nframes_t nframes, snd_pcm_ioplug_t *io) jack->areas[channel].first = 0; jack->areas[channel].step = jack->sample_bits; } - - if (io->state != SND_PCM_STATE_RUNNING) { - if (io->stream == SND_PCM_STREAM_PLAYBACK) { - for (channel = 0; channel < io->channels; channel++) - snd_pcm_area_silence(&jack->areas[channel], 0, nframes, io->format); - return 0; - } - } hw_ptr = jack->hw_ptr; - areas = snd_pcm_ioplug_mmap_areas(io); - - while (xfer < nframes) { - snd_pcm_uframes_t frames = nframes - xfer; - snd_pcm_uframes_t offset = hw_ptr % io->buffer_size; - snd_pcm_uframes_t cont = io->buffer_size - offset; - - if (cont < frames) - frames = cont; + if (io->state == SND_PCM_STATE_RUNNING) { + const snd_pcm_channel_area_t *areas = snd_pcm_ioplug_mmap_areas(io); + + while (xfer < nframes) { + snd_pcm_uframes_t frames = nframes - xfer; + snd_pcm_uframes_t offset = hw_ptr % io->buffer_size; + snd_pcm_uframes_t cont = io->buffer_size - offset; + snd_pcm_uframes_t hw_avail = + snd_pcm_jack_hw_avail(io, hw_ptr, + io->appl_ptr); + + /* stop copying if there is no more data available */ + if (hw_avail <= 0) + break; + + /* split the snd_pcm_area_copy() function into two parts + * if the data to copy passes the buffer wrap around + */ + if (cont < frames) + frames = cont; + + if (hw_avail < frames) + frames = hw_avail; + + for (channel = 0; channel < io->channels; channel++) { + if (io->stream == SND_PCM_STREAM_PLAYBACK) + snd_pcm_area_copy(&jack->areas[channel], xfer, &areas[channel], offset, frames, io->format); + else + snd_pcm_area_copy(&areas[channel], offset, &jack->areas[channel], xfer, frames, io->format); + } - for (channel = 0; channel < io->channels; channel++) { - if (io->stream == SND_PCM_STREAM_PLAYBACK) - snd_pcm_area_copy(&jack->areas[channel], xfer, &areas[channel], offset, frames, io->format); - else - snd_pcm_area_copy(&areas[channel], offset, &jack->areas[channel], xfer, frames, io->format); + hw_ptr += frames; + if (hw_ptr >= jack->boundary) + hw_ptr -= jack->boundary; + xfer += frames; } - - hw_ptr += frames; - if (hw_ptr >= jack->boundary) - hw_ptr -= jack->boundary; - xfer += frames; } jack->hw_ptr = hw_ptr; + /* check if requested frames were copied */ + if (xfer < nframes) { + /* always fill the not yet written JACK buffer with silence */ + if (io->stream == SND_PCM_STREAM_PLAYBACK) { + const snd_pcm_uframes_t samples = nframes - xfer; + for (channel = 0; channel < io->channels; channel++) + snd_pcm_area_silence(&jack->areas[channel], xfer, samples, io->format); + } + } + pcm_poll_unblock_check(io); /* unblock socket for polling if needed */ return 0; -- 2.7.4

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[alsa-devel] [PATCH - JACK PCM 1/1] jack: Use correct parameter order for snd_pcm_areas_silence()
by twischer＠de.adit-jv.com 21 Mar '18

21 Mar '18

From: Timo Wischer <twischer(a)de.adit-jv.com> Without this fix it will sometimes fail with a segmentation fault. Signed-off-by: Timo Wischer <twischer(a)de.adit-jv.com> diff --git a/jack/pcm_jack.c b/jack/pcm_jack.c index a655668..e3df4d2 100644 --- a/jack/pcm_jack.c +++ b/jack/pcm_jack.c @@ -200,7 +200,7 @@ snd_pcm_jack_process_cb(jack_nframes_t nframes, snd_pcm_ioplug_t *io) if (io->stream == SND_PCM_STREAM_PLAYBACK) { const snd_pcm_uframes_t frames = nframes - xfer; - snd_pcm_areas_silence(jack->areas, io->channels, xfer, + snd_pcm_areas_silence(jack->areas, xfer, io->channels, frames, io->format); } -- 2.7.4

3 3

[alsa-devel] [PATCH v3] USB Audio Device Class 3.0 support
by Ruslan Bilovol 21 Mar '18

21 Mar '18

Hi Takashi, This patch adds initial USB Audio Device Class 3.0 [1] support to the ALSA that we discussed at ELCE. The patch was tested with UAC3 gadget [2] that I posted to USB mailing list before. It is good for working with BADD device which implements such topologies like BAIF (Basic Audio Input Functions), BAOF (Basic Audio Output Functions) or compbination of both. UAC3 spec has changed descriptors laout and many other codes comparing to UAC2 spec, thus making reuse of existing sources impossible or quite complex. There are still many areas of improvement, as this patch doesn't have UAC3 Mixer Unit support nor some new features like Power Management I tested this with BeagleBone Black as UAC3 gadget device. [1] http://www.usb.org/developers/docs/devclass_docs/USB_Audio_v3.0.zip [2] http://www.spinics.net/lists/linux-usb/msg162482.html v3 changes: - dropped audioformat quirks patch (already accepted) - addressed coding style requested changes v2 changes: - moved audioformat quirks to quirks.c - addressed many comments from Pierre-Louis Bossart - reorganized audio-v3.h - added more UAC3 channel relationship definitions Ruslan Bilovol (1): ALSA: usb: initial USB Audio Device Class 3.0 support include/linux/usb/audio-v2.h | 4 +- include/linux/usb/audio-v3.h | 395 +++++++++++++++++++++++++++++++++++++++++ include/uapi/linux/usb/audio.h | 1 + sound/usb/card.c | 7 +- sound/usb/card.h | 2 +- sound/usb/clock.c | 228 +++++++++++++++++++++--- sound/usb/clock.h | 4 +- sound/usb/format.c | 91 ++++++++-- sound/usb/format.h | 6 +- sound/usb/mixer.c | 337 +++++++++++++++++++++++------------ sound/usb/stream.c | 365 +++++++++++++++++++++++++++++++++---- 11 files changed, 1246 insertions(+), 194 deletions(-) create mode 100644 include/linux/usb/audio-v3.h -- 1.9.1

3 3

[alsa-devel] [PATCH V3] ASoC: amd: dma driver changes for BT I2S controller instance
by Vijendar Mukunda 21 Mar '18

21 Mar '18

With in ACP, There are three I2S controllers can be configured/enabled ( I2S SP, I2S MICSP, I2S BT). Default enabled I2S controller instance is I2S SP. This patch provides required changes to support I2S BT controller Instance. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> Reviewed-by: Alex Deucher <alexander.deucher(a)amd.com> --- v1-->v2: removed cpu_dai name comparisons to get the current i2s instance value. Modified code to get the current I2S controller instance value based on dwc i2s_instance v2->v3: moved I2S instance macro definitions to designware header file sound/soc/amd/acp-pcm-dma.c | 106 +++++++++++++++++++++++++++++++++++--------- sound/soc/amd/acp.h | 7 +++ 2 files changed, 91 insertions(+), 22 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 540088d..a61c4e0 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -697,6 +697,7 @@ static int acp_dma_open(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *prtd = substream->private_data; struct snd_soc_component *component = snd_soc_rtdcom_lookup(prtd, DRV_NAME); struct audio_drv_data *intr_data = dev_get_drvdata(component->dev); + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(prtd->cpu_dai); struct audio_substream_data *adata = kzalloc(sizeof(struct audio_substream_data), GFP_KERNEL); if (adata == NULL) @@ -710,7 +711,21 @@ static int acp_dma_open(struct snd_pcm_substream *substream) default: runtime->hw = acp_pcm_hardware_playback; } + adata->i2s_play_instance = dev->i2s_instance; + if (adata->i2s_play_instance == I2S_SP_INSTANCE) + adata->i2ssp_renderbytescount = 0; + else if (adata->i2s_play_instance == I2S_BT_INSTANCE) + adata->i2sbt_renderbytescount = 0; + else + return -EINVAL; } else { + adata->i2s_capture_instance = dev->i2s_instance; + if (adata->i2s_capture_instance == I2S_SP_INSTANCE) + adata->i2ssp_capturebytescount = 0; + else if (adata->i2s_capture_instance == I2S_BT_INSTANCE) + adata->i2sbt_capturebytescount = 0; + else + return -EINVAL; switch (intr_data->asic_type) { case CHIP_STONEY: runtime->hw = acp_st_pcm_hardware_capture; @@ -736,11 +751,20 @@ static int acp_dma_open(struct snd_pcm_substream *substream) * This enablement is not required for another stream, if current * stream is not closed */ - if (!intr_data->play_i2ssp_stream && !intr_data->capture_i2ssp_stream) + if (!intr_data->play_i2ssp_stream && !intr_data->capture_i2ssp_stream && + !intr_data->play_i2sbt_stream && + !intr_data->capture_i2sbt_stream) acp_reg_write(1, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - intr_data->play_i2ssp_stream = substream; + switch (adata->i2s_play_instance) { + case I2S_BT_INSTANCE: + intr_data->play_i2sbt_stream = substream; + break; + case I2S_SP_INSTANCE: + default: + intr_data->play_i2ssp_stream = substream; + } /* For Stoney, Memory gating is disabled,i.e SRAM Banks * won't be turned off. The default state for SRAM banks is ON. * Setting SRAM bank state code skipped for STONEY platform. @@ -751,7 +775,14 @@ static int acp_dma_open(struct snd_pcm_substream *substream) bank, true); } } else { - intr_data->capture_i2ssp_stream = substream; + switch (adata->i2s_capture_instance) { + case I2S_BT_INSTANCE: + intr_data->capture_i2sbt_stream = substream; + break; + case I2S_SP_INSTANCE: + default: + intr_data->capture_i2ssp_stream = substream; + } if (intr_data->asic_type != CHIP_STONEY) { for (bank = 5; bank <= 8; bank++) acp_set_sram_bank_state(intr_data->acp_mmio, @@ -1010,34 +1041,49 @@ static int acp_dma_close(struct snd_pcm_substream *substream) struct snd_soc_component *component = snd_soc_rtdcom_lookup(prtd, DRV_NAME); struct audio_drv_data *adata = dev_get_drvdata(component->dev); - kfree(rtd); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - adata->play_i2ssp_stream = NULL; - /* For Stoney, Memory gating is disabled,i.e SRAM Banks - * won't be turned off. The default state for SRAM banks is ON. - * Setting SRAM bank state code skipped for STONEY platform. - * added condition checks for Carrizo platform only - */ - if (adata->asic_type != CHIP_STONEY) { - for (bank = 1; bank <= 4; bank++) - acp_set_sram_bank_state(adata->acp_mmio, bank, - false); + switch (rtd->i2s_play_instance) { + case I2S_BT_INSTANCE: + adata->play_i2sbt_stream = NULL; + break; + case I2S_SP_INSTANCE: + default: + adata->play_i2ssp_stream = NULL; + /* For Stoney, Memory gating is disabled,i.e SRAM Banks + * won't be turned off. The default state for SRAM banks + * is ON.Setting SRAM bank state code skipped for STONEY + * platform.Added condition checks for Carrizo platform + * only. + */ + if (adata->asic_type != CHIP_STONEY) { + for (bank = 1; bank <= 4; bank++) + acp_set_sram_bank_state(adata->acp_mmio, + bank, false); + } } } else { - adata->capture_i2ssp_stream = NULL; - if (adata->asic_type != CHIP_STONEY) { - for (bank = 5; bank <= 8; bank++) - acp_set_sram_bank_state(adata->acp_mmio, bank, - false); + switch (rtd->i2s_capture_instance) { + case I2S_BT_INSTANCE: + adata->capture_i2sbt_stream = NULL; + break; + case I2S_SP_INSTANCE: + default: + adata->capture_i2ssp_stream = NULL; + if (adata->asic_type != CHIP_STONEY) { + for (bank = 5; bank <= 8; bank++) + acp_set_sram_bank_state(adata->acp_mmio, + bank, false); + } } } /* Disable ACP irq, when the current stream is being closed and * another stream is also not active. - */ - if (!adata->play_i2ssp_stream && !adata->capture_i2ssp_stream) + */ + if (!adata->play_i2ssp_stream && !adata->capture_i2ssp_stream && + !adata->play_i2sbt_stream && !adata->capture_i2sbt_stream) acp_reg_write(0, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); + kfree(rtd); return 0; } @@ -1089,6 +1135,8 @@ static int acp_audio_probe(struct platform_device *pdev) audio_drv_data->play_i2ssp_stream = NULL; audio_drv_data->capture_i2ssp_stream = NULL; + audio_drv_data->play_i2sbt_stream = NULL; + audio_drv_data->capture_i2sbt_stream = NULL; audio_drv_data->asic_type = *pdata; @@ -1177,6 +1225,20 @@ static int acp_pcm_resume(struct device *dev) adata->capture_i2ssp_stream->runtime->private_data, adata->asic_type); } + if (adata->asic_type != CHIP_CARRIZO) { + if (adata->play_i2sbt_stream && + adata->play_i2sbt_stream->runtime) { + config_acp_dma(adata->acp_mmio, + adata->play_i2sbt_stream->runtime->private_data, + adata->asic_type); + } + if (adata->capture_i2sbt_stream && + adata->capture_i2sbt_stream->runtime) { + config_acp_dma(adata->acp_mmio, + adata->capture_i2sbt_stream->runtime->private_data, + adata->asic_type); + } + } acp_reg_write(1, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); return 0; } diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index ba01510..c86bef4 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -4,6 +4,7 @@ #include "include/acp_2_2_d.h" #include "include/acp_2_2_sh_mask.h" +#include "../dwc/local.h" #define ACP_PAGE_SIZE_4K_ENABLE 0x02 @@ -88,12 +89,18 @@ struct audio_substream_data { uint64_t size; u64 i2ssp_renderbytescount; u64 i2ssp_capturebytescount; + u64 i2sbt_renderbytescount; + u64 i2sbt_capturebytescount; void __iomem *acp_mmio; + u16 i2s_play_instance; + u16 i2s_capture_instance; }; struct audio_drv_data { struct snd_pcm_substream *play_i2ssp_stream; struct snd_pcm_substream *capture_i2ssp_stream; + struct snd_pcm_substream *play_i2sbt_stream; + struct snd_pcm_substream *capture_i2sbt_stream; void __iomem *acp_mmio; u32 asic_type; }; -- 2.7.4

2 1

[alsa-devel] [PATCH 0/3] Fix audio regression on rk3288
by Ezequiel Garcia 21 Mar '18

21 Mar '18

From: Ezequiel Garcia <ezequiel(a)collabora.co.uk> Currently on rk3288 boards the ASoC machine driver fails to load, breaking audio support. Apparently, the regression was introduced by commit 24069b589b02 ("ASoC: hdmi-codec: remove multi detection support") which changed the codec name. If there is a lesson here, it would be about binding drivers using strings and how fragile this is. Or maybe it's about regression testing. Or maybe there isn't a lesson. Anyway, the first patch fixes the regression; the second patch makes the machine driver selects all the codecs it; finally the third patch is just a binding documentation fix. Tested on a Radxa Rock2 Square board. Ezequiel Garcia (3): ASoC: rockchip: Fix dai_name for HDMI codec ASoC: rockchip: rk3288-hdmi-analog: Select needed codecs dt-bindings: sound: rockchip Fix compatible string in example .../devicetree/bindings/sound/rockchip,rk3288-hdmi-analog.txt | 2 +- sound/soc/rockchip/Kconfig | 3 +++ sound/soc/rockchip/rk3288_hdmi_analog.c | 2 +- 3 files changed, 5 insertions(+), 2 deletions(-) -- 2.16.2

2 7

[alsa-devel] [asoc:topic/amd 2/7] sound/soc//amd/acp-da7219-max98357a.c:79:22: error: passing argument 1 of 'da7219_aad_jack_det' from incompatible pointer type
by kbuild test robot 21 Mar '18

21 Mar '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/amd head: c88d311533563e6dadc244930a36125d592f5ec5 commit: 9a60cde2dd63e10b421e87e1397641304dd1ef13 [2/7] ASoC: amd: acp-da7219-max98357: replace codec to component config: x86_64-allmodconfig (attached as .config) compiler: gcc-7 (Debian 7.3.0-1) 7.3.0 reproduce: git checkout 9a60cde2dd63e10b421e87e1397641304dd1ef13 # save the attached .config to linux build tree make ARCH=x86_64 All errors (new ones prefixed by >>): sound/soc//amd/acp-da7219-max98357a.c: In function 'cz_da7219_init': >> sound/soc//amd/acp-da7219-max98357a.c:79:22: error: passing argument 1 of 'da7219_aad_jack_det' from incompatible pointer type [-Werror=incompatible-pointer-types] da7219_aad_jack_det(component, &cz_jack); ^~~~~~~~~ In file included from sound/soc//amd/acp-da7219-max98357a.c:38:0: sound/soc//amd/../codecs/da7219-aad.h:209:6: note: expected 'struct snd_soc_codec *' but argument is of type 'struct snd_soc_component *' void da7219_aad_jack_det(struct snd_soc_codec *codec, struct snd_soc_jack *jack); ^~~~~~~~~~~~~~~~~~~ cc1: some warnings being treated as errors sparse warnings: (new ones prefixed by >>) >> sound/soc/amd/acp-da7219-max98357a.c:79:29: sparse: incorrect type in argument 1 (different base types) @@ expected struct snd_soc_codec *codec @@ got strustruct snd_soc_codec *codec @@ sound/soc/amd/acp-da7219-max98357a.c:79:29: expected struct snd_soc_codec *codec sound/soc/amd/acp-da7219-max98357a.c:79:29: got struct snd_soc_component *component sound/soc/amd/acp-da7219-max98357a.c: In function 'cz_da7219_init': sound/soc/amd/acp-da7219-max98357a.c:79:22: error: passing argument 1 of 'da7219_aad_jack_det' from incompatible pointer type [-Werror=incompatible-pointer-types] da7219_aad_jack_det(component, &cz_jack); ^~~~~~~~~ In file included from sound/soc/amd/acp-da7219-max98357a.c:38:0: sound/soc/amd/../codecs/da7219-aad.h:209:6: note: expected 'struct snd_soc_codec *' but argument is of type 'struct snd_soc_component *' void da7219_aad_jack_det(struct snd_soc_codec *codec, struct snd_soc_jack *jack); ^~~~~~~~~~~~~~~~~~~ cc1: some warnings being treated as errors vim +/da7219_aad_jack_det +79 sound/soc//amd/acp-da7219-max98357a.c 45 46 static int cz_da7219_init(struct snd_soc_pcm_runtime *rtd) 47 { 48 int ret; 49 struct snd_soc_card *card = rtd->card; 50 struct snd_soc_dai *codec_dai = rtd->codec_dai; 51 struct snd_soc_component *component = codec_dai->component; 52 53 dev_info(rtd->dev, "codec dai name = %s\n", codec_dai->name); 54 55 ret = snd_soc_dai_set_sysclk(codec_dai, DA7219_CLKSRC_MCLK, 56 CZ_PLAT_CLK, SND_SOC_CLOCK_IN); 57 if (ret < 0) { 58 dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret); 59 return ret; 60 } 61 62 ret = snd_soc_dai_set_pll(codec_dai, 0, DA7219_SYSCLK_PLL, 63 CZ_PLAT_CLK, MCLK_RATE); 64 if (ret < 0) { 65 dev_err(rtd->dev, "can't set codec pll: %d\n", ret); 66 return ret; 67 } 68 69 ret = snd_soc_card_jack_new(card, "Headset Jack", 70 SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | 71 SND_JACK_BTN_0 | SND_JACK_BTN_1 | 72 SND_JACK_BTN_2 | SND_JACK_BTN_3, 73 &cz_jack, NULL, 0); 74 if (ret) { 75 dev_err(card->dev, "HP jack creation failed %d\n", ret); 76 return ret; 77 } 78 > 79 da7219_aad_jack_det(component, &cz_jack); 80 81 return 0; 82 } 83 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

3 2

[alsa-devel] [PATCH] ASoC: rsnd: set pm_ops in hibernate-compatible way
by Nikita Yushchenko 21 Mar '18

21 Mar '18

Use SET_SYSTEM_SLEEP_PM_OPS() macro instead of direct assignment to .suspend and .resume fields. This makes driver working after restore from hibernation. Signed-off-by: Nikita Yushchenko <nikita.yoush(a)cogentembedded.com> --- sound/soc/sh/rcar/core.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/soc/sh/rcar/core.c b/sound/soc/sh/rcar/core.c index 64d5ecb86528..dabf9a35c25c 100644 --- a/sound/soc/sh/rcar/core.c +++ b/sound/soc/sh/rcar/core.c @@ -1560,8 +1560,7 @@ static int rsnd_resume(struct device *dev) } static const struct dev_pm_ops rsnd_pm_ops = { - .suspend = rsnd_suspend, - .resume = rsnd_resume, + SET_SYSTEM_SLEEP_PM_OPS(rsnd_suspend, rsnd_resume) }; static struct platform_driver rsnd_driver = { -- 2.11.0

2 1

[alsa-devel] [PATCH] linux-next: ASoC: mediatek: mt2701: Fix a typo in printk
by Masanari Iida 21 Mar '18

21 Mar '18

This patch fixes a spelling typo in printk Signed-off-by: Masanari Iida <standby24x7(a)gmail.com> --- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c index f7e0702d92f8..f2eb4c7f94e8 100644 --- a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c +++ b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c @@ -343,7 +343,7 @@ static int mt2701_btmrg_hw_params(struct snd_pcm_substream *substream, stream_fs = params_rate(params); if ((stream_fs != 8000) && (stream_fs != 16000)) { - dev_err(afe->dev, "%s() btmgr not supprt this stream_fs %d\n", + dev_err(afe->dev, "%s() btmgr not support this stream_fs %d\n", __func__, stream_fs); return -EINVAL; } -- 2.16.2.520.gd0db9edba005

2 2

[alsa-devel] [PATCH v2 00/21] Allow compile-testing NO_DMA (drivers)
by Geert Uytterhoeven 20 Mar '18

20 Mar '18

Hi all, If NO_DMA=y, get_dma_ops() returns a reference to the non-existing symbol bad_dma_ops, thus causing a link failure if it is ever used. The intention of this is twofold: 1. To catch users of the DMA API on systems that do no support the DMA mapping API, 2. To avoid building drivers that cannot work on such systems anyway. However, the disadvantage is that we have to keep on adding dependencies on HAS_DMA all over the place. Thanks to the COMPILE_TEST symbol, lots of drivers now depend on one or more platform dependencies (that imply HAS_DMA) || COMPILE_TEST, thus already covering intention #2. Having to add an explicit dependency on HAS_DMA here is cumbersome, and hinders compile-testing. Hence I think the time is ripe to reconsider the link failure. Patch series "[PATCH v2 0/5] Allow compile-testing NO_DMA (core)" (https://lkml.org/lkml/2018/3/16/435) already: - Changed get_dma_ops() to return NULL instead, - Added a few more dummies to enable compile-testing. This patch series: - Removes dependencies on HAS_DMA for symbols that already have platform dependencies implying HAS_DMA. To avoid allmodconfig/allyesconfig regressions on NO_DMA=y platforms, this (drivers) series should be applied after the previous (core) series (but not many people may notice/care ;-) Changes compared to v1: - Add Reviewed-by, Acked-by, - Drop dependency of SND_SOC_LPASS_IPQ806X on HAS_DMA, - Drop dependency of VIDEOBUF{,2}_DMA_{CONTIG,SG} on HAS_DMA, - Drop new dependencies of VIDEO_IPU3_CIO2, DVB_C8SECTPFE, and MTD_NAND_MARVELL on HAS_DMA, - Split in per-subsystem patches, - Split-off the core part in a separate series. This series is against v4.16-rc5. It can also be found at https://git.kernel.org/pub/scm/linux/kernel/git/geert/linux-m68k.git/log/?h… It has been compile-tested with allmodconfig and allyesconfig for m68k/sun3, and has received attention from the kbuild test robot. Thanks! Geert Uytterhoeven (21): ASoC: Remove depends on HAS_DMA in case of platform dependency ata: Remove depends on HAS_DMA in case of platform dependency crypto: Remove depends on HAS_DMA in case of platform dependency fbdev: Remove depends on HAS_DMA in case of platform dependency firewire: Remove depends on HAS_DMA in case of platform dependency fpga: Remove depends on HAS_DMA in case of platform dependency i2c: Remove depends on HAS_DMA in case of platform dependency iio: adc: Remove depends on HAS_DMA in case of platform dependency iommu: Remove depends on HAS_DMA in case of platform dependency lightnvm: Remove depends on HAS_DMA in case of platform dependency mailbox: Remove depends on HAS_DMA in case of platform dependency media: Remove depends on HAS_DMA in case of platform dependency mmc: Remove depends on HAS_DMA in case of platform dependency mtd: Remove depends on HAS_DMA in case of platform dependency net: Remove depends on HAS_DMA in case of platform dependency remoteproc: Remove depends on HAS_DMA in case of platform dependency scsi: hisi_sas: Remove depends on HAS_DMA in case of platform dependency serial: Remove depends on HAS_DMA in case of platform dependency spi: Remove depends on HAS_DMA in case of platform dependency staging: vc04_services: Remove depends on HAS_DMA in case of platform dependency usb: Remove depends on HAS_DMA in case of platform dependency drivers/ata/Kconfig | 2 -- drivers/crypto/Kconfig | 14 +++------ drivers/firewire/Kconfig | 1 - drivers/fpga/Kconfig | 1 - drivers/i2c/busses/Kconfig | 3 -- drivers/iio/adc/Kconfig | 2 -- drivers/iommu/Kconfig | 5 ++-- drivers/lightnvm/Kconfig | 2 +- drivers/mailbox/Kconfig | 2 -- drivers/media/common/videobuf2/Kconfig | 2 -- drivers/media/pci/dt3155/Kconfig | 1 - drivers/media/pci/intel/ipu3/Kconfig | 1 - drivers/media/pci/solo6x10/Kconfig | 1 - drivers/media/pci/sta2x11/Kconfig | 1 - drivers/media/pci/tw5864/Kconfig | 1 - drivers/media/pci/tw686x/Kconfig | 1 - drivers/media/platform/Kconfig | 40 ++++++++----------------- drivers/media/platform/am437x/Kconfig | 2 +- drivers/media/platform/atmel/Kconfig | 4 +-- drivers/media/platform/blackfin/Kconfig | 1 - drivers/media/platform/davinci/Kconfig | 6 ---- drivers/media/platform/marvell-ccic/Kconfig | 3 +- drivers/media/platform/rcar-vin/Kconfig | 2 +- drivers/media/platform/soc_camera/Kconfig | 3 +- drivers/media/platform/sti/c8sectpfe/Kconfig | 2 +- drivers/media/v4l2-core/Kconfig | 2 -- drivers/mmc/host/Kconfig | 10 ++----- drivers/mtd/nand/Kconfig | 8 ++--- drivers/mtd/spi-nor/Kconfig | 2 +- drivers/net/ethernet/amd/Kconfig | 2 +- drivers/net/ethernet/apm/xgene-v2/Kconfig | 1 - drivers/net/ethernet/apm/xgene/Kconfig | 1 - drivers/net/ethernet/arc/Kconfig | 6 ++-- drivers/net/ethernet/broadcom/Kconfig | 2 -- drivers/net/ethernet/calxeda/Kconfig | 2 +- drivers/net/ethernet/hisilicon/Kconfig | 2 +- drivers/net/ethernet/marvell/Kconfig | 8 ++--- drivers/net/ethernet/mellanox/mlxsw/Kconfig | 2 +- drivers/net/ethernet/renesas/Kconfig | 2 -- drivers/net/wireless/broadcom/brcm80211/Kconfig | 1 - drivers/net/wireless/quantenna/qtnfmac/Kconfig | 2 +- drivers/remoteproc/Kconfig | 1 - drivers/scsi/hisi_sas/Kconfig | 2 +- drivers/spi/Kconfig | 12 ++------ drivers/staging/media/davinci_vpfe/Kconfig | 1 - drivers/staging/media/omap4iss/Kconfig | 1 - drivers/staging/vc04_services/Kconfig | 1 - drivers/tty/serial/Kconfig | 4 --- drivers/usb/gadget/udc/Kconfig | 4 +-- drivers/usb/mtu3/Kconfig | 2 +- drivers/video/fbdev/Kconfig | 3 +- sound/soc/bcm/Kconfig | 3 +- sound/soc/kirkwood/Kconfig | 1 - sound/soc/pxa/Kconfig | 1 - sound/soc/qcom/Kconfig | 7 ++--- 55 files changed, 56 insertions(+), 143 deletions(-) -- 2.7.4 Gr{oetje,eeting}s, Geert -- Geert Uytterhoeven -- There's lots of Linux beyond ia32 -- geert(a)linux-m68k.org In personal conversations with technical people, I call myself a hacker. But when I'm talking to journalists I just say "programmer" or something like that. -- Linus Torvalds

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[alsa-devel] [PATCH v2] ASoC: samsung: Mark unused Odroid compatibles as deprecated
by Krzysztof Kozlowski 20 Mar '18

20 Mar '18

Compatible for XU4 audio is not being used. Instead the board uses the same compatible as XU3. The devices are now just compatible so they should use the same value. Mark "hardkernel,odroid-xu4-audio" as being deprecated so in this future could be removed to limit useless properties. Additionally add older compatibles appearing in the bindings. Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- Changes since v1: 1. Correct typo XU4->XU3 for one deprecated compatible. --- Documentation/devicetree/bindings/sound/samsung,odroid.txt | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/samsung,odroid.txt b/Documentation/devicetree/bindings/sound/samsung,odroid.txt index f35a2694eb04..e9da2200e173 100644 --- a/Documentation/devicetree/bindings/sound/samsung,odroid.txt +++ b/Documentation/devicetree/bindings/sound/samsung,odroid.txt @@ -3,7 +3,9 @@ Samsung Exynos Odroid XU3/XU4 audio complex with MAX98090 codec Required properties: - compatible - "hardkernel,odroid-xu3-audio" - for Odroid XU3 board, - "hardkernel,odroid-xu4-audio" - for Odroid XU4 board + "hardkernel,odroid-xu4-audio" - for Odroid XU4 board (deprecated), + "samsung,odroid-xu3-audio" - for Odroid XU3 board (deprecated), + "samsung,odroid-xu4-audio" - for Odroid XU4 board (deprecated) - model - the user-visible name of this sound complex - clocks - should contain entries matching clock names in the clock-names property -- 2.14.1

3 7

[alsa-devel] [alsa-lib][RESEND][PATCH] pcm: add set and get methods for no-rewind flag
by Sriram Periyasamy 20 Mar '18

20 Mar '18

From: Ramesh Babu <ramesh.babu(a)intel.com> Add functions to set/get no-rewind flag Signed-off-by: Ramesh Babu <ramesh.babu(a)intel.com> Signed-off-by: Sriram Periyasamy <sriramx.periyasamy(a)intel.com> --- Please find the reference for previous discussion at [1]. [1] https://patchwork.kernel.org/patch/10191481/ include/pcm.h | 2 ++ include/sound/asound.h | 1 + src/pcm/pcm.c | 40 ++++++++++++++++++++++++++++++++++++++++ src/pcm/pcm_local.h | 1 + 4 files changed, 44 insertions(+) diff --git a/include/pcm.h b/include/pcm.h index e2a53435caab..c6af884607ae 100644 --- a/include/pcm.h +++ b/include/pcm.h @@ -805,6 +805,8 @@ int snd_pcm_hw_params_set_export_buffer(snd_pcm_t *pcm, snd_pcm_hw_params_t *par int snd_pcm_hw_params_get_export_buffer(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int *val); int snd_pcm_hw_params_set_period_wakeup(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int val); int snd_pcm_hw_params_get_period_wakeup(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int *val); +int snd_pcm_hw_params_set_no_rewind(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int val); +int snd_pcm_hw_params_get_no_rewind(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int *val); int snd_pcm_hw_params_get_period_time(const snd_pcm_hw_params_t *params, unsigned int *val, int *dir); int snd_pcm_hw_params_get_period_time_min(const snd_pcm_hw_params_t *params, unsigned int *val, int *dir); diff --git a/include/sound/asound.h b/include/sound/asound.h index 6041cab27a23..35fd50ec5395 100644 --- a/include/sound/asound.h +++ b/include/sound/asound.h @@ -375,6 +375,7 @@ typedef int snd_pcm_hw_param_t; #define SNDRV_PCM_HW_PARAMS_NORESAMPLE (1<<0) /* avoid rate resampling */ #define SNDRV_PCM_HW_PARAMS_EXPORT_BUFFER (1<<1) /* export buffer */ #define SNDRV_PCM_HW_PARAMS_NO_PERIOD_WAKEUP (1<<2) /* disable period wakeups */ +#define SNDRV_PCM_HW_PARAMS_NO_REWINDS (1<<3) /* disable rewinds */ struct snd_interval { unsigned int min, max; diff --git a/src/pcm/pcm.c b/src/pcm/pcm.c index ed47cb516c73..3346609fdf6d 100644 --- a/src/pcm/pcm.c +++ b/src/pcm/pcm.c @@ -4826,6 +4826,46 @@ int snd_pcm_hw_params_get_period_wakeup(snd_pcm_t *pcm, snd_pcm_hw_params_t *par } /** + * \brief Configure no-rewind flag through configuration space + * \param pcm PCM handle + * \param params Configuration space + * \param val 0 = disable (default), 1 = enable (default) no-rewind flag + * \return Zero on success, otherwise a negative error code. + * + * This function can be called to inform driver about application uses rewind or not + * + * Based on no-rewind flag, underlying driver could take decision on buffering scheme. + */ + +/* TODO: Invert the logic with function name set_rewind */ +int snd_pcm_hw_params_set_no_rewind(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int val) +{ + assert(pcm && params); + + if (val) { + params->flags |= SND_PCM_HW_PARAMS_NO_REWINDS; + } else + params->flags &= ~SND_PCM_HW_PARAMS_NO_REWINDS; + params->rmask = ~0; + + return snd_pcm_hw_refine(pcm, params); +} + +/** + * \brief Extract no-rewind flag from a configuration space + * \param pcm PCM handle + * \param params Configuration space + * \param val 0 = disabled, 1 = enabled no-rewind flag + * \return Zero on success, otherwise a negative error code. + */ +int snd_pcm_hw_params_get_no_rewind(snd_pcm_t *pcm, snd_pcm_hw_params_t *params, unsigned int *val) +{ + assert(pcm && params && val); + *val = params->flags & SND_PCM_HW_PARAMS_NO_REWINDS ? 0 : 1; + return 0; +} + +/** * \brief Extract period time from a configuration space * \param params Configuration space * \param val Returned approximate period duration in us diff --git a/src/pcm/pcm_local.h b/src/pcm/pcm_local.h index d52229d8ddea..2b3cae5b93f5 100644 --- a/src/pcm/pcm_local.h +++ b/src/pcm/pcm_local.h @@ -103,6 +103,7 @@ #define SND_PCM_HW_PARAMS_NORESAMPLE SNDRV_PCM_HW_PARAMS_NORESAMPLE #define SND_PCM_HW_PARAMS_EXPORT_BUFFER SNDRV_PCM_HW_PARAMS_EXPORT_BUFFER #define SND_PCM_HW_PARAMS_NO_PERIOD_WAKEUP SNDRV_PCM_HW_PARAMS_NO_PERIOD_WAKEUP +#define SND_PCM_HW_PARAMS_NO_REWINDS SNDRV_PCM_HW_PARAMS_NO_REWINDS #define SND_PCM_INFO_MONOTONIC 0x80000000 -- 2.7.4

1 0

[alsa-devel] [alsa-utils][RESEND][PATCH] Add support for no-rewind flag in aplay utility
by Sriram Periyasamy 20 Mar '18

20 Mar '18

From: Ramesh Babu <ramesh.babu(a)intel.com> The "no-rewind" flag is introduced in alsa. Application can use this flag to assure kernel driver that rewind won't be exercised. Driver could use this flag to alter it's buffering scheme to optimize power and performance. Signed-off-by: Ramesh Babu <ramesh.babu(a)intel.com> Signed-off-by: Sriram Periyasamy <sriramx.periyasamy(a)intel.com> --- Please find the reference for previous discussion at [1]. [1] https://patchwork.kernel.org/patch/10191483/ aplay/aplay.c | 14 +++++++++++++- 1 file changed, 13 insertions(+), 1 deletion(-) diff --git a/aplay/aplay.c b/aplay/aplay.c index bbd7fffa04fc..6eb9d3b04d92 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -108,6 +108,7 @@ static snd_pcm_stream_t stream = SND_PCM_STREAM_PLAYBACK; static int mmap_flag = 0; static int interleaved = 1; static int nonblock = 0; +static int no_rewind = 0; static volatile sig_atomic_t in_aborting = 0; static u_char *audiobuf = NULL; static snd_pcm_uframes_t chunk_size = 0; @@ -229,6 +230,7 @@ _("Usage: %s [OPTION]... [FILE]...\n" "-I, --separate-channels one file for each channel\n" "-i, --interactive allow interactive operation from stdin\n" "-m, --chmap=ch1,ch2,.. Give the channel map to override or follow\n" +"-w, --no-rewind enables no rewind mode\n" " --disable-resample disable automatic rate resample\n" " --disable-channels disable automatic channel conversions\n" " --disable-format disable automatic format conversions\n" @@ -470,7 +472,7 @@ int main(int argc, char *argv[]) { int duration_or_sample = 0; int option_index; - static const char short_options[] = "hnlLD:qt:c:f:r:d:s:MNF:A:R:T:B:vV:IPCi" + static const char short_options[] = "hnlLD:qt:c:f:r:d:s:w:MNF:A:R:T:B:vV:IPCi" #ifdef CONFIG_SUPPORT_CHMAP "m:" #endif @@ -514,6 +516,7 @@ int main(int argc, char *argv[]) {"process-id-file", 1, 0, OPT_PROCESS_ID_FILE}, {"use-strftime", 0, 0, OPT_USE_STRFTIME}, {"interactive", 0, 0, 'i'}, + {"no-rewind", 0, 0, 'w'}, {"dump-hw-params", 0, 0, OPT_DUMP_HWPARAMS}, {"fatal-errors", 0, 0, OPT_FATAL_ERRORS}, #ifdef CONFIG_SUPPORT_CHMAP @@ -753,6 +756,9 @@ int main(int argc, char *argv[]) case 'i': interactive = 1; break; + case 'w': + no_rewind = 1; + break; case OPT_DISABLE_RESAMPLE: open_mode |= SND_PCM_NO_AUTO_RESAMPLE; break; @@ -1396,6 +1402,12 @@ static void set_params(void) &buffer_frames); } assert(err >= 0); + + if (no_rewind) { + err = snd_pcm_hw_params_set_no_rewind(handle, params, 1); + assert(err >= 0); + } + monotonic = snd_pcm_hw_params_is_monotonic(params); can_pause = snd_pcm_hw_params_can_pause(params); err = snd_pcm_hw_params(handle, params); -- 2.7.4

1 0

[alsa-devel] [PATCH v2 1/2] ASoC: AMD: Use single dai for da7219 playback and capture
by Akshu Agrawal 20 Mar '18

20 Mar '18

BT I2S is a bi-directional dai, we will use the same cpu dai for playback and capture. V2: reworded the commit message. TEST=aplay -D hw:0,0 -vv <file> arecord -D hw:0,0 -f dat -d 5 -vv <file> Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-da7219-max98357a.c | 14 ++------------ 1 file changed, 2 insertions(+), 12 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index bbe0f10..d07c2a1 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -124,8 +124,8 @@ static int cz_fe_startup(struct snd_pcm_substream *substream) static struct snd_soc_dai_link cz_dai_7219_98357[] = { { - .name = "amd-da7219-play", - .stream_name = "Playback", + .name = "amd-da7219-play-cap", + .stream_name = "Playback and Capture", .platform_name = "acp_audio_dma.0.auto", .cpu_dai_name = "designware-i2s.3.auto", .codec_dai_name = "da7219-hifi", @@ -134,16 +134,6 @@ static int cz_fe_startup(struct snd_pcm_substream *substream) | SND_SOC_DAIFMT_CBM_CFM, .init = cz_da7219_init, .dpcm_playback = 1, - }, - { - .name = "amd-da7219-cap", - .stream_name = "Capture", - .platform_name = "acp_audio_dma.0.auto", - .cpu_dai_name = "designware-i2s.4.auto", - .codec_dai_name = "da7219-hifi", - .codec_name = "i2c-DLGS7219:00", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, .dpcm_capture = 1, .ops = &cz_da7219_cap_ops, }, -- 1.9.1

2 2

[alsa-devel] [PATCH v2 1/3] ASoC: AMD: Use single dai for da7219 playback and capture
by Akshu Agrawal 20 Mar '18

20 Mar '18

BT I2S is a bi-directional dai, we will use the same cpu dai for playback and capture. V2: reworded the commit message. TEST=aplay -D hw:0,0 -vv <file> arecord -D hw:0,0 -f dat -d 5 -vv <file> Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-da7219-max98357a.c | 14 ++------------ 1 file changed, 2 insertions(+), 12 deletions(-) diff --git a/sound/soc/amd/acp-da7219-max98357a.c b/sound/soc/amd/acp-da7219-max98357a.c index bbe0f10..d07c2a1 100644 --- a/sound/soc/amd/acp-da7219-max98357a.c +++ b/sound/soc/amd/acp-da7219-max98357a.c @@ -124,8 +124,8 @@ static int cz_fe_startup(struct snd_pcm_substream *substream) static struct snd_soc_dai_link cz_dai_7219_98357[] = { { - .name = "amd-da7219-play", - .stream_name = "Playback", + .name = "amd-da7219-play-cap", + .stream_name = "Playback and Capture", .platform_name = "acp_audio_dma.0.auto", .cpu_dai_name = "designware-i2s.3.auto", .codec_dai_name = "da7219-hifi", @@ -134,16 +134,6 @@ static int cz_fe_startup(struct snd_pcm_substream *substream) | SND_SOC_DAIFMT_CBM_CFM, .init = cz_da7219_init, .dpcm_playback = 1, - }, - { - .name = "amd-da7219-cap", - .stream_name = "Capture", - .platform_name = "acp_audio_dma.0.auto", - .cpu_dai_name = "designware-i2s.4.auto", - .codec_dai_name = "da7219-hifi", - .codec_name = "i2c-DLGS7219:00", - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF - | SND_SOC_DAIFMT_CBM_CFM, .dpcm_capture = 1, .ops = &cz_da7219_cap_ops, }, -- 1.9.1

2 3

Re: [alsa-devel] [PATCH] ASoC: uniphier: evea: add switch for changing source of line-in
by Mark Brown 20 Mar '18

20 Mar '18

On Mon, Mar 19, 2018 at 01:19:10PM +0900, Katsuhiro Suzuki wrote: > > I'll apply for now but this should really be a DAPM control so that we > > can power down things connected to the disconnected line inputs when > > recording. > Thanks a lot for your suggestion. I tried to change the implementation to DAPM > control as follows: > I can see the value of ALINSW1 register at 'Line In 1 Mux',0 using > amixer get 'Line In 1 Mux',0 > But I can't change the value. > amixer set 'Line In 1 Mux',0 LIN2 > Simple mixer control 'Line In 1 Mux',0 > Capabilities: enum > Items: 'LIN1' 'LIN2' 'LIN3' > Item0: 'LIN1' > Would you tell me what is wrong... Ugh, I *have* run into that before but I can't remember what triggers it and your code doesn't have any mistakes I can spot. Unfortunately I'm at Linaro Connect this week and don't have a test system I can poke at with me to remind myself, and I'm still travelling next week unfortunately. I'd add some trace to the set code path to make sure everything is being called as expected. It's somemthing really small that's hard to make a warning for in the code IIRC.

1 0

[alsa-devel] [PATCH] ASoC: ssm2602: Replace reg_default_raw with reg_default
by James Kelly 20 Mar '18

20 Mar '18

SSM2602 driver is broken on recent kernels (at least since 4.9). User space applications such as amixer or alsamixer get EIO when attempting to access codec controls via the relevant IOCTLs. Root cause of these failures is the regcache_hw_init function in drivers/base/regmap/regcache.c, which prevents regmap cache initalization from the reg_defaults_raw element of the regmap_config structure when registers are write only. It also disables the regmap cache entirely when all registers are write only or volatile as is the case for the SSM2602 driver. Using the reg_defaults element of the regmap_config structure rather than the reg_defaults_raw element to initalize the regmap cache avoids the logic in the regcache_hw_init function entirely. It also makes this driver consistent with other ASoC codec drivers, as this driver was the ONLY codec driver that used the reg_defaults_raw element to initalize the cache. Tested on Digilent Zybo Z7 development board which has a SSM2603 codec chip connected to a Xilinx Zynq SoC. --- sound/soc/codecs/ssm2602.c | 19 +++++++++++++------ 1 file changed, 13 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 031e0eabcc1f..501a4e73b185 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -54,10 +54,17 @@ struct ssm2602_priv { * using 2 wire for device control, so we cache them instead. * There is no point in caching the reset register */ -static const u16 ssm2602_reg[SSM2602_CACHEREGNUM] = { - 0x0097, 0x0097, 0x0079, 0x0079, - 0x000a, 0x0008, 0x009f, 0x000a, - 0x0000, 0x0000 +static const struct reg_default ssm2602_reg[SSM2602_CACHEREGNUM] = { + { .reg = 0x00, .def = 0x0097 }, + { .reg = 0x01, .def = 0x0097 }, + { .reg = 0x02, .def = 0x0079 }, + { .reg = 0x03, .def = 0x0079 }, + { .reg = 0x04, .def = 0x000a }, + { .reg = 0x05, .def = 0x0008 }, + { .reg = 0x06, .def = 0x009f }, + { .reg = 0x07, .def = 0x000a }, + { .reg = 0x08, .def = 0x0000 }, + { .reg = 0x09, .def = 0x0000 } }; @@ -621,8 +628,8 @@ const struct regmap_config ssm2602_regmap_config = { .volatile_reg = ssm2602_register_volatile, .cache_type = REGCACHE_RBTREE, - .reg_defaults_raw = ssm2602_reg, - .num_reg_defaults_raw = ARRAY_SIZE(ssm2602_reg), + .reg_defaults = ssm2602_reg, + .num_reg_defaults = ARRAY_SIZE(ssm2602_reg), }; EXPORT_SYMBOL_GPL(ssm2602_regmap_config); -- 2.14.3 (Apple Git-98)

2 2

[alsa-devel] [PATCH] ASoC: rt286: don't turn off HV and VREF if headset is detected
by Bard Liao 20 Mar '18

20 Mar '18

"HV" and "VREF" will be powered up when jack type detection process is started and will be powered off when jack type detection process is finished. It will generate an unexpected interrupt signal when they are powered up during the capture process. Codec driver will do the jack type detection process and we can't capture properly before the jack detection process is finished. This patch will not power off the "HV" and "VREF" widgets if headset is detected and it will solve the unexpected interrupt issue. As a result, it will also solve the silence data captured at the beginning in headset mic recording issue. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt286.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt286.c b/sound/soc/codecs/rt286.c index e164357..0b0f748 100644 --- a/sound/soc/codecs/rt286.c +++ b/sound/soc/codecs/rt286.c @@ -275,9 +275,10 @@ static int rt286_jack_detect(struct rt286_priv *rt286, bool *hp, bool *mic) regmap_read(rt286->regmap, RT286_GET_MIC1_SENSE, &buf); *mic = buf & 0x80000000; } - - snd_soc_dapm_disable_pin(dapm, "HV"); - snd_soc_dapm_disable_pin(dapm, "VREF"); + if (!*mic) { + snd_soc_dapm_disable_pin(dapm, "HV"); + snd_soc_dapm_disable_pin(dapm, "VREF"); + } if (!*hp) snd_soc_dapm_disable_pin(dapm, "LDO1"); snd_soc_dapm_sync(dapm); -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: rt298: don't turn off HV and VREF if headset is detected
by Bard Liao 20 Mar '18

20 Mar '18

"HV" and "VREF" will be powered up when jack type detection process is started and will be powered off when jack type detection process is finished. It will generate an unexpected interrupt signal when they are powered up during the capture process. Codec driver will do the jack type detection process and we can't capture properly before the jack detection process is finished. This patch will not power off the "HV" and "VREF" widgets if headset is detected and it will solve the unexpected interrupt issue. As a result, it will also solve the silence data captured at the beginning in headset mic recording issue. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt298.c | 7 ++++--- 1 file changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/rt298.c b/sound/soc/codecs/rt298.c index 7bf4b31..06cdba4 100644 --- a/sound/soc/codecs/rt298.c +++ b/sound/soc/codecs/rt298.c @@ -290,9 +290,10 @@ static int rt298_jack_detect(struct rt298_priv *rt298, bool *hp, bool *mic) regmap_read(rt298->regmap, RT298_GET_MIC1_SENSE, &buf); *mic = buf & 0x80000000; } - - snd_soc_dapm_disable_pin(dapm, "HV"); - snd_soc_dapm_disable_pin(dapm, "VREF"); + if (!*mic) { + snd_soc_dapm_disable_pin(dapm, "HV"); + snd_soc_dapm_disable_pin(dapm, "VREF"); + } if (!*hp) snd_soc_dapm_disable_pin(dapm, "LDO1"); snd_soc_dapm_sync(dapm); -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: twl6040: remove duplicated remove callback
by Kuninori Morimoto 20 Mar '18

20 Mar '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> We don't need 2 .remove callback Fixes: 7480389fb0d8 ("ASoC: twl6040: replace codec to component") Reported-by: Julia Lawall <julia.lawall(a)lip6.fr> Reported-by: kbuild test robot <fengguang.wu(a)intel.com> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> --- sound/soc/codecs/twl6040.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/codecs/twl6040.c b/sound/soc/codecs/twl6040.c index b56a647..bfd1abd 100644 --- a/sound/soc/codecs/twl6040.c +++ b/sound/soc/codecs/twl6040.c @@ -1158,7 +1158,6 @@ static void twl6040_remove(struct snd_soc_component *component) .remove = twl6040_remove, .read = twl6040_read, .write = twl6040_write, - .remove = twl6040_remove, .set_bias_level = twl6040_set_bias_level, .controls = twl6040_snd_controls, .num_controls = ARRAY_SIZE(twl6040_snd_controls), -- 1.9.1

3 3

[alsa-devel] [PATCH] ASoC: topology: Fix kcontrol name string handling.
by Liam Girdwood 20 Mar '18

20 Mar '18

Fix the topology kcontrol string handling so that string pointer references are strdup()ed instead of being copied. This fixes issues with kcontrol templates on the stack or ones that are freed. Remember and free the strings too when topology is unloaded. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/soc-topology.c | 23 ++++++++++++++++++----- 1 file changed, 18 insertions(+), 5 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 78197e82fc4f..dd2b118a35b3 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -530,6 +530,7 @@ static void remove_widget(struct snd_soc_component *comp, kfree(se->dobj.control.dtexts[j]); kfree(se); + kfree(w->kcontrol_news[i].name); } } else { /* volume mixer or bytes controls */ @@ -546,6 +547,7 @@ static void remove_widget(struct snd_soc_component *comp, */ kfree((void *)kcontrol->private_value); snd_ctl_remove(card, kcontrol); + kfree(w->kcontrol_news[i].name); } } @@ -1256,7 +1258,9 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create( dev_dbg(tplg->dev, " adding DAPM widget mixer control %s at %d\n", mc->hdr.name, i); - kc[i].name = mc->hdr.name; + kc[i].name = kstrdup(mc->hdr.name, GFP_KERNEL); + if (kc[i].name == NULL) + goto err_str; kc[i].private_value = (long)sm; kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; kc[i].access = mc->hdr.access; @@ -1304,8 +1308,10 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create( err_str: kfree(sm); err: - for (--i; i >= 0; i--) + for (--i; i >= 0; i--) { kfree((void *)kc[i].private_value); + kfree(kc[i].name); + } kfree(kc); return NULL; } @@ -1336,7 +1342,9 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( dev_dbg(tplg->dev, " adding DAPM widget enum control %s\n", ec->hdr.name); - kc[i].name = ec->hdr.name; + kc[i].name = kstrdup(ec->hdr.name, GFP_KERNEL); + if (kc[i].name == NULL) + goto err_se; kc[i].private_value = (long)se; kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; kc[i].access = ec->hdr.access; @@ -1412,6 +1420,7 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_denum_create( kfree(se->dobj.control.dtexts[j]); kfree(se); + kfree(kc[i].name); } err: kfree(kc); @@ -1450,7 +1459,9 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dbytes_create( "ASoC: adding bytes kcontrol %s with access 0x%x\n", be->hdr.name, be->hdr.access); - kc[i].name = be->hdr.name; + kc[i].name = kstrdup(be->hdr.name, GFP_KERNEL); + if (kc[i].name == NULL) + goto err; kc[i].private_value = (long)sbe; kc[i].iface = SNDRV_CTL_ELEM_IFACE_MIXER; kc[i].access = be->hdr.access; @@ -1480,8 +1491,10 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dbytes_create( return kc; err: - for (--i; i >= 0; i--) + for (--i; i >= 0; i--) { kfree((void *)kc[i].private_value); + kfree(kc[i].name); + } kfree(kc); return NULL; -- 2.14.1

2 3

[alsa-devel] [PATCH v2 0/2] USB Audio Device Class 3.0 support
by Ruslan Bilovol 20 Mar '18

20 Mar '18

Hi Takashi, This patch adds initial USB Audio Device Class 3.0 [1] support to the ALSA that we discussed at ELCE. The patch was tested with UAC3 gadget [2] that I posted to USB mailing list before. It is good for working with BADD device which implements such topologies like BAIF (Basic Audio Input Functions), BAOF (Basic Audio Output Functions) or compbination of both. UAC3 spec has changed descriptors laout and many other codes comparing to UAC2 spec, thus making reuse of existing sources impossible or quite complex. There are still many areas of improvement, as this patch doesn't have UAC3 Mixer Unit support nor some new features like Power Management I tested this with BeagleBone Black as UAC3 gadget device. [1] http://www.usb.org/developers/docs/devclass_docs/USB_Audio_v3.0.zip [2] http://www.spinics.net/lists/linux-usb/msg162482.html v2 changes: - moved audioformat quirks to quirks.c - addressed many comments from Pierre-Louis Bossart - reorganized audio-v3.h - added more UAC3 channel relationship definitions Ruslan Bilovol (2): ALSA: usb-audio: move audioformat quirks to quirks.c ALSA: usb: initial USB Audio Device Class 3.0 support include/linux/usb/audio-v2.h | 4 +- include/linux/usb/audio-v3.h | 395 +++++++++++++++++++++++++++++++++++++++++ include/uapi/linux/usb/audio.h | 1 + sound/usb/card.c | 7 +- sound/usb/card.h | 2 +- sound/usb/clock.c | 228 +++++++++++++++++++++--- sound/usb/clock.h | 4 +- sound/usb/format.c | 91 ++++++++-- sound/usb/format.h | 6 +- sound/usb/mixer.c | 337 +++++++++++++++++++++++------------ sound/usb/quirks.c | 34 ++++ sound/usb/quirks.h | 4 + sound/usb/stream.c | 393 +++++++++++++++++++++++++++++++++------- 13 files changed, 1284 insertions(+), 222 deletions(-) create mode 100644 include/linux/usb/audio-v3.h -- 1.9.1

4 9

[alsa-devel] [PATCH] ALSA: usb-audio: Fix parsing descriptor of UAC2 processing unit
by Kirill Marinushkin 20 Mar '18

20 Mar '18

Currently, the offsets in the UAC2 processing unit descriptor are calculated incorrectly. It causes an issue when connecting the device which provides such a feature: ~~~~ [84126.724420] usb 1-1.3.1: invalid Processing Unit descriptor (id 18) ~~~~ After this patch is applied, the UAC2 processing unit inits w/o this error. Fixes: 5e1ddb481776 ("UAPI: (Scripted) Disintegrate include/linux/usb") Signed-off-by: Kirill Marinushkin <k.marinushkin(a)gmail.com> Cc: Mark Brown <broonie(a)kernel.org> Cc: Jaroslav Kysela <perex(a)perex.cz> Cc: Takashi Iwai <tiwai(a)suse.com> Cc: Thomas Gleixner <tglx(a)linutronix.de> Cc: Felipe Balbi <felipe.balbi(a)linux.intel.com> Cc: Philippe Ombredanne <pombredanne(a)nexb.com> Cc: Greg Kroah-Hartman <gregkh(a)linuxfoundation.org> Cc: Ruslan Bilovol <ruslan.bilovol(a)gmail.com> Cc: linux-kernel(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org --- include/uapi/linux/usb/audio.h | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/include/uapi/linux/usb/audio.h b/include/uapi/linux/usb/audio.h index 17a022c5b414..da3315ed1bcd 100644 --- a/include/uapi/linux/usb/audio.h +++ b/include/uapi/linux/usb/audio.h @@ -370,7 +370,7 @@ static inline __u8 uac_processing_unit_bControlSize(struct uac_processing_unit_d { return (protocol == UAC_VERSION_1) ? desc->baSourceID[desc->bNrInPins + 4] : - desc->baSourceID[desc->bNrInPins + 6]; + 2; /* in UAC2, this value is constant */ } static inline __u8 *uac_processing_unit_bmControls(struct uac_processing_unit_descriptor *desc, @@ -378,7 +378,7 @@ static inline __u8 *uac_processing_unit_bmControls(struct uac_processing_unit_de { return (protocol == UAC_VERSION_1) ? &desc->baSourceID[desc->bNrInPins + 5] : - &desc->baSourceID[desc->bNrInPins + 7]; + &desc->baSourceID[desc->bNrInPins + 6]; } static inline __u8 uac_processing_unit_iProcessing(struct uac_processing_unit_descriptor *desc, -- 2.13.6

4 4

[alsa-devel] [PATCH V2] ASoC: amd: dma driver changes for BT I2S controller instance
by Vijendar Mukunda 19 Mar '18

19 Mar '18

With in ACP, There are three I2S controllers can be configured/enabled ( I2S SP, I2S MICSP, I2S BT). Default enabled I2S controller instance is I2S SP. This patch provides required changes to support I2S BT controller Instance. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Signed-off-by: Akshu Agrawal <akshu.agrawal(a)amd.com> --- sound/soc/amd/acp-pcm-dma.c | 106 +++++++++++++++++++++++++++++++++++--------- sound/soc/amd/acp.h | 10 +++++ 2 files changed, 94 insertions(+), 22 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 540088d..a61c4e0 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -697,6 +697,7 @@ static int acp_dma_open(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *prtd = substream->private_data; struct snd_soc_component *component = snd_soc_rtdcom_lookup(prtd, DRV_NAME); struct audio_drv_data *intr_data = dev_get_drvdata(component->dev); + struct dw_i2s_dev *dev = snd_soc_dai_get_drvdata(prtd->cpu_dai); struct audio_substream_data *adata = kzalloc(sizeof(struct audio_substream_data), GFP_KERNEL); if (adata == NULL) @@ -710,7 +711,21 @@ static int acp_dma_open(struct snd_pcm_substream *substream) default: runtime->hw = acp_pcm_hardware_playback; } + adata->i2s_play_instance = dev->i2s_instance; + if (adata->i2s_play_instance == I2S_SP_INSTANCE) + adata->i2ssp_renderbytescount = 0; + else if (adata->i2s_play_instance == I2S_BT_INSTANCE) + adata->i2sbt_renderbytescount = 0; + else + return -EINVAL; } else { + adata->i2s_capture_instance = dev->i2s_instance; + if (adata->i2s_capture_instance == I2S_SP_INSTANCE) + adata->i2ssp_capturebytescount = 0; + else if (adata->i2s_capture_instance == I2S_BT_INSTANCE) + adata->i2sbt_capturebytescount = 0; + else + return -EINVAL; switch (intr_data->asic_type) { case CHIP_STONEY: runtime->hw = acp_st_pcm_hardware_capture; @@ -736,11 +751,20 @@ static int acp_dma_open(struct snd_pcm_substream *substream) * This enablement is not required for another stream, if current * stream is not closed */ - if (!intr_data->play_i2ssp_stream && !intr_data->capture_i2ssp_stream) + if (!intr_data->play_i2ssp_stream && !intr_data->capture_i2ssp_stream && + !intr_data->play_i2sbt_stream && + !intr_data->capture_i2sbt_stream) acp_reg_write(1, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - intr_data->play_i2ssp_stream = substream; + switch (adata->i2s_play_instance) { + case I2S_BT_INSTANCE: + intr_data->play_i2sbt_stream = substream; + break; + case I2S_SP_INSTANCE: + default: + intr_data->play_i2ssp_stream = substream; + } /* For Stoney, Memory gating is disabled,i.e SRAM Banks * won't be turned off. The default state for SRAM banks is ON. * Setting SRAM bank state code skipped for STONEY platform. @@ -751,7 +775,14 @@ static int acp_dma_open(struct snd_pcm_substream *substream) bank, true); } } else { - intr_data->capture_i2ssp_stream = substream; + switch (adata->i2s_capture_instance) { + case I2S_BT_INSTANCE: + intr_data->capture_i2sbt_stream = substream; + break; + case I2S_SP_INSTANCE: + default: + intr_data->capture_i2ssp_stream = substream; + } if (intr_data->asic_type != CHIP_STONEY) { for (bank = 5; bank <= 8; bank++) acp_set_sram_bank_state(intr_data->acp_mmio, @@ -1010,34 +1041,49 @@ static int acp_dma_close(struct snd_pcm_substream *substream) struct snd_soc_component *component = snd_soc_rtdcom_lookup(prtd, DRV_NAME); struct audio_drv_data *adata = dev_get_drvdata(component->dev); - kfree(rtd); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - adata->play_i2ssp_stream = NULL; - /* For Stoney, Memory gating is disabled,i.e SRAM Banks - * won't be turned off. The default state for SRAM banks is ON. - * Setting SRAM bank state code skipped for STONEY platform. - * added condition checks for Carrizo platform only - */ - if (adata->asic_type != CHIP_STONEY) { - for (bank = 1; bank <= 4; bank++) - acp_set_sram_bank_state(adata->acp_mmio, bank, - false); + switch (rtd->i2s_play_instance) { + case I2S_BT_INSTANCE: + adata->play_i2sbt_stream = NULL; + break; + case I2S_SP_INSTANCE: + default: + adata->play_i2ssp_stream = NULL; + /* For Stoney, Memory gating is disabled,i.e SRAM Banks + * won't be turned off. The default state for SRAM banks + * is ON.Setting SRAM bank state code skipped for STONEY + * platform.Added condition checks for Carrizo platform + * only. + */ + if (adata->asic_type != CHIP_STONEY) { + for (bank = 1; bank <= 4; bank++) + acp_set_sram_bank_state(adata->acp_mmio, + bank, false); + } } } else { - adata->capture_i2ssp_stream = NULL; - if (adata->asic_type != CHIP_STONEY) { - for (bank = 5; bank <= 8; bank++) - acp_set_sram_bank_state(adata->acp_mmio, bank, - false); + switch (rtd->i2s_capture_instance) { + case I2S_BT_INSTANCE: + adata->capture_i2sbt_stream = NULL; + break; + case I2S_SP_INSTANCE: + default: + adata->capture_i2ssp_stream = NULL; + if (adata->asic_type != CHIP_STONEY) { + for (bank = 5; bank <= 8; bank++) + acp_set_sram_bank_state(adata->acp_mmio, + bank, false); + } } } /* Disable ACP irq, when the current stream is being closed and * another stream is also not active. - */ - if (!adata->play_i2ssp_stream && !adata->capture_i2ssp_stream) + */ + if (!adata->play_i2ssp_stream && !adata->capture_i2ssp_stream && + !adata->play_i2sbt_stream && !adata->capture_i2sbt_stream) acp_reg_write(0, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); + kfree(rtd); return 0; } @@ -1089,6 +1135,8 @@ static int acp_audio_probe(struct platform_device *pdev) audio_drv_data->play_i2ssp_stream = NULL; audio_drv_data->capture_i2ssp_stream = NULL; + audio_drv_data->play_i2sbt_stream = NULL; + audio_drv_data->capture_i2sbt_stream = NULL; audio_drv_data->asic_type = *pdata; @@ -1177,6 +1225,20 @@ static int acp_pcm_resume(struct device *dev) adata->capture_i2ssp_stream->runtime->private_data, adata->asic_type); } + if (adata->asic_type != CHIP_CARRIZO) { + if (adata->play_i2sbt_stream && + adata->play_i2sbt_stream->runtime) { + config_acp_dma(adata->acp_mmio, + adata->play_i2sbt_stream->runtime->private_data, + adata->asic_type); + } + if (adata->capture_i2sbt_stream && + adata->capture_i2sbt_stream->runtime) { + config_acp_dma(adata->acp_mmio, + adata->capture_i2sbt_stream->runtime->private_data, + adata->asic_type); + } + } acp_reg_write(1, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); return 0; } diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index ba01510..8bc78cc 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -4,6 +4,8 @@ #include "include/acp_2_2_d.h" #include "include/acp_2_2_sh_mask.h" +#include <sound/designware_i2s.h> +#include "../dwc/local.h" #define ACP_PAGE_SIZE_4K_ENABLE 0x02 @@ -72,6 +74,8 @@ #define mmACP_I2S_16BIT_RESOLUTION_EN 0x5209 #define ACP_I2S_MIC_16BIT_RESOLUTION_EN 0x01 #define ACP_I2S_SP_16BIT_RESOLUTION_EN 0x02 +#define I2S_SP_INSTANCE 1 +#define I2S_BT_INSTANCE 2 enum acp_dma_priority_level { /* 0x0 Specifies the DMA channel is given normal priority */ ACP_DMA_PRIORITY_LEVEL_NORMAL = 0x0, @@ -88,12 +92,18 @@ struct audio_substream_data { uint64_t size; u64 i2ssp_renderbytescount; u64 i2ssp_capturebytescount; + u64 i2sbt_renderbytescount; + u64 i2sbt_capturebytescount; void __iomem *acp_mmio; + u16 i2s_play_instance; + u16 i2s_capture_instance; }; struct audio_drv_data { struct snd_pcm_substream *play_i2ssp_stream; struct snd_pcm_substream *capture_i2ssp_stream; + struct snd_pcm_substream *play_i2sbt_stream; + struct snd_pcm_substream *capture_i2sbt_stream; void __iomem *acp_mmio; u32 asic_type; }; -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: dwc: I2S Controller instance param added
by Vijendar Mukunda 19 Mar '18

19 Mar '18

When multiple I2S controller instances created, i2s_instance parameter refers to i2s controller instance value. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- include/sound/designware_i2s.h | 4 ++++ sound/soc/dwc/dwc-i2s.c | 1 + sound/soc/dwc/local.h | 1 + 3 files changed, 6 insertions(+) diff --git a/include/sound/designware_i2s.h b/include/sound/designware_i2s.h index 830f5ca..2a94e7c 100644 --- a/include/sound/designware_i2s.h +++ b/include/sound/designware_i2s.h @@ -44,6 +44,10 @@ struct i2s_platform_data { int channel; u32 snd_fmts; u32 snd_rates; + /* i2s_instance parameter returns I2S controller instance value + * when multiple I2S controllers instantiated + */ + u32 i2s_instance; #define DW_I2S_QUIRK_COMP_REG_OFFSET (1 << 0) #define DW_I2S_QUIRK_COMP_PARAM1 (1 << 1) diff --git a/sound/soc/dwc/dwc-i2s.c b/sound/soc/dwc/dwc-i2s.c index e27e21f..20145c0 100644 --- a/sound/soc/dwc/dwc-i2s.c +++ b/sound/soc/dwc/dwc-i2s.c @@ -549,6 +549,7 @@ static int dw_configure_dai_by_pd(struct dw_i2s_dev *dev, if (dev->quirks & DW_I2S_QUIRK_16BIT_IDX_OVERRIDE) idx = 1; + dev->i2s_instance = pdata->i2s_instance; /* Set DMA slaves info */ dev->play_dma_data.pd.data = pdata->play_dma_data; dev->capture_dma_data.pd.data = pdata->capture_dma_data; diff --git a/sound/soc/dwc/local.h b/sound/soc/dwc/local.h index 91dc70a..e89e464 100644 --- a/sound/soc/dwc/local.h +++ b/sound/soc/dwc/local.h @@ -98,6 +98,7 @@ struct dw_i2s_dev { u32 ccr; u32 xfer_resolution; u32 fifo_th; + u32 i2s_instance; /* data related to DMA transfers b/w i2s and DMAC */ union dw_i2s_snd_dma_data play_dma_data; -- 2.7.4

2 1

[alsa-devel] [PATCH v2] ASoC: Intel: Boards: Add CNL RT274 I2S machine driver
by Guneshwor Singh 19 Mar '18

19 Mar '18

Add CNL I2S machine driver using Realtek RT274 codec in I2S mode configured to ssp0. Signed-off-by: Guneshwor Singh <guneshwor.o.singh(a)intel.com> --- sound/soc/intel/boards/Kconfig | 11 ++ sound/soc/intel/boards/Makefile | 2 + sound/soc/intel/boards/cnl_rt274.c | 232 +++++++++++++++++++++++++++++++++++++ 3 files changed, 245 insertions(+) create mode 100644 sound/soc/intel/boards/cnl_rt274.c diff --git a/sound/soc/intel/boards/Kconfig b/sound/soc/intel/boards/Kconfig index 24797482a3d2..822b9143dcb1 100644 --- a/sound/soc/intel/boards/Kconfig +++ b/sound/soc/intel/boards/Kconfig @@ -281,6 +281,17 @@ config SND_SOC_INTEL_KBL_DA7219_MAX98357A_MACH Say Y if you have such a device. If unsure select "N". +config SND_SOC_INTEL_CNL_RT274_MACH + tristate "ASoC Audio driver for Cannonlake with RT274 I2S mode" + depends on MFD_INTEL_LPSS && ACPI && I2C + select SND_SOC_RT274 + select SND_SOC_DMIC + help + This adds support for ASoC machine driver for Cannonlake platform + with RT274 I2S audio codec. + Say Y if you have such a device. + If unsure select "N". + endif ## SND_SOC_INTEL_SKYLAKE endif ## SND_SOC_INTEL_MACH diff --git a/sound/soc/intel/boards/Makefile b/sound/soc/intel/boards/Makefile index 92b5507291af..2b49cb6aa84a 100644 --- a/sound/soc/intel/boards/Makefile +++ b/sound/soc/intel/boards/Makefile @@ -21,6 +21,7 @@ snd-soc-kbl_rt5663_rt5514_max98927-objs := kbl_rt5663_rt5514_max98927.o snd-soc-skl_rt286-objs := skl_rt286.o snd-skl_nau88l25_max98357a-objs := skl_nau88l25_max98357a.o snd-soc-skl_nau88l25_ssm4567-objs := skl_nau88l25_ssm4567.o +snd-soc-cnl-rt274-objs := cnl_rt274.o obj-$(CONFIG_SND_SOC_INTEL_HASWELL_MACH) += snd-soc-sst-haswell.o obj-$(CONFIG_SND_SOC_INTEL_BYT_RT5640_MACH) += snd-soc-sst-byt-rt5640-mach.o @@ -44,3 +45,4 @@ obj-$(CONFIG_SND_SOC_INTEL_KBL_RT5663_RT5514_MAX98927_MACH) += snd-soc-kbl_rt566 obj-$(CONFIG_SND_SOC_INTEL_SKL_RT286_MACH) += snd-soc-skl_rt286.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_MAX98357A_MACH) += snd-skl_nau88l25_max98357a.o obj-$(CONFIG_SND_SOC_INTEL_SKL_NAU88L25_SSM4567_MACH) += snd-soc-skl_nau88l25_ssm4567.o +obj-$(CONFIG_SND_SOC_INTEL_CNL_RT274_MACH) += snd-soc-cnl-rt274.o diff --git a/sound/soc/intel/boards/cnl_rt274.c b/sound/soc/intel/boards/cnl_rt274.c new file mode 100644 index 000000000000..04986c015f0f --- /dev/null +++ b/sound/soc/intel/boards/cnl_rt274.c @@ -0,0 +1,232 @@ +// SPDX-License-Identifier: GPL-2.0 +// Copyright(c) 2016-18 Intel Corporation. + +#include <linux/module.h> +#include <sound/jack.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> +#include "../../codecs/rt274.h" + +#define CNL_FREQ_OUT 24000000 +#define CNL_BE_FIXUP_RATE 48000 +#define RT274_CODEC_DAI "rt274-aif1" + +static int cnl_rt274_clock_control(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + struct snd_soc_dapm_context *dapm = w->dapm; + struct snd_soc_card *card = dapm->card; + struct snd_soc_dai *codec_dai = + snd_soc_card_get_codec_dai(card, RT274_CODEC_DAI); + int ret, ratio = 100; + + if (!codec_dai) + return -EINVAL; + + /* Codec needs clock for Jack detection and button press */ + ret = snd_soc_dai_set_sysclk(codec_dai, RT274_SCLK_S_PLL2, + CNL_FREQ_OUT, SND_SOC_CLOCK_IN); + if (ret < 0) { + dev_err(codec_dai->dev, "set codec sysclk failed: %d\n", ret); + return ret; + } + + if (SND_SOC_DAPM_EVENT_ON(event)) { + ret = snd_soc_dai_set_bclk_ratio(codec_dai, ratio); + if (ret) { + dev_err(codec_dai->dev, + "set bclk ratio failed: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_pll(codec_dai, 0, RT274_PLL2_S_BCLK, + CNL_BE_FIXUP_RATE * ratio, + CNL_FREQ_OUT); + if (ret) { + dev_err(codec_dai->dev, + "enable PLL2 failed: %d\n", ret); + return ret; + } + } + + return 0; +} + +static const struct snd_kcontrol_new cnl_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphone Jack"), + SOC_DAPM_PIN_SWITCH("Mic Jack"), +}; + +static const struct snd_soc_dapm_widget cnl_rt274_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_MIC("Mic Jack", NULL), + SND_SOC_DAPM_MIC("SoC DMIC", NULL), + SND_SOC_DAPM_SUPPLY("Platform Clock", SND_SOC_NOPM, 0, 0, + cnl_rt274_clock_control, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), +}; + +static int cnl_dmic_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *channels = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + + if (params_channels(params) == 2) + channels->min = channels->max = 2; + else + channels->min = channels->max = 4; + + return 0; +} + +static const struct snd_soc_dapm_route cnl_map[] = { + {"Headphone Jack", NULL, "HPO Pin"}, + {"MIC", NULL, "Mic Jack"}, + {"DMic", NULL, "SoC DMIC"}, + {"DMIC01 Rx", NULL, "Capture"}, + {"dmic01_hifi", NULL, "DMIC01 Rx"}, + + {"AIF1 Playback", NULL, "ssp0 Tx"}, + {"ssp0 Tx", NULL, "codec1_out"}, + {"ssp0 Tx", NULL, "codec0_out"}, + + {"ssp0 Rx", NULL, "AIF1 Capture"}, + {"codec0_in", NULL, "ssp0 Rx"}, + + {"Headphone Jack", NULL, "Platform Clock"}, + {"Mic Jack", NULL, "Platform Clock"}, +}; + +static struct snd_soc_jack_pin cnl_headset_pins[] = { + { + .pin = "Mic Jack", + .mask = SND_JACK_MICROPHONE, + }, + { + .pin = "Headphone Jack", + .mask = SND_JACK_HEADPHONE, + }, +}; + +static struct snd_soc_jack cnl_headset; + +static int cnl_rt274_init(struct snd_soc_pcm_runtime *runtime) +{ + struct snd_soc_card *card = runtime->card; + struct snd_soc_dai *codec_dai = runtime->codec_dai; + struct snd_soc_component *component = codec_dai->component; + int ret; + + ret = snd_soc_card_jack_new(runtime->card, "Headset", + SND_JACK_HEADSET, &cnl_headset, + cnl_headset_pins, ARRAY_SIZE(cnl_headset_pins)); + if (ret) + return ret; + + ret = snd_soc_component_set_jack(component, &cnl_headset, NULL); + if (ret) + return ret; + + /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xf, 0xf, 4, 24); + if (ret < 0) { + dev_err(runtime->dev, "can't set codec pcm format %d\n", ret); + return ret; + } + + card->dapm.idle_bias_off = true; + + return 0; +} + +static int cnl_be_fixup(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params) +{ + struct snd_interval *rate = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE); + struct snd_interval *channels = + hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS); + + rate->min = rate->max = CNL_BE_FIXUP_RATE; + channels->min = channels->max = 2; + snd_mask_none(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT)); + snd_mask_set(hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT), + SNDRV_PCM_FORMAT_S24_LE); + + return 0; +} + +static struct snd_soc_dai_link cnl_rt274_dailink[] = { + { + .name = "SSP0-Codec", + .cpu_dai_name = "SSP0 Pin", + .codec_name = "i2c-INT34C2:00", + .codec_dai_name = "rt274-aif1", + .platform_name = "0000:00:1f.3", + .be_hw_params_fixup = cnl_be_fixup, + .no_pcm = 1, + .ignore_pmdown_time = 1, + .dai_fmt = SND_SOC_DAIFMT_DSP_A | + SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS, + .dpcm_playback = 1, + .dpcm_capture = 1, + .init = cnl_rt274_init, + }, + { + .name = "dmic01", + .cpu_dai_name = "DMIC01 Pin", + .codec_name = "dmic-codec", + .codec_dai_name = "dmic-hifi", + .platform_name = "0000:00:1f.3", + .be_hw_params_fixup = cnl_dmic_fixup, + .no_pcm = 1, + .ignore_suspend = 1, + .dpcm_capture = 1, + }, +}; + +static int +cnl_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *link) +{ + link->platform_name = "0000:00:1f.3"; + link->nonatomic = 1; + + return 0; +} + +/* SoC card */ +static struct snd_soc_card snd_soc_card_cnl = { + .name = KBUILD_MODNAME, + .dai_link = cnl_rt274_dailink, + .num_links = ARRAY_SIZE(cnl_rt274_dailink), + .dapm_widgets = cnl_rt274_widgets, + .num_dapm_widgets = ARRAY_SIZE(cnl_rt274_widgets), + .dapm_routes = cnl_map, + .num_dapm_routes = ARRAY_SIZE(cnl_map), + .controls = cnl_controls, + .num_controls = ARRAY_SIZE(cnl_controls), + .add_dai_link = cnl_add_dai_link, + .fully_routed = true, +}; + +static int cnl_rt274_probe(struct platform_device *pdev) +{ + snd_soc_card_cnl.dev = &pdev->dev; + + return devm_snd_soc_register_card(&pdev->dev, &snd_soc_card_cnl); +} + +static struct platform_driver cnl_rt274_driver = { + .driver = { + .name = "cnl_rt274", + .pm = &snd_soc_pm_ops, + }, + .probe = cnl_rt274_probe, +}; + +module_platform_driver(cnl_rt274_driver); + +MODULE_AUTHOR("Guneshwor Singh <guneshwor.o.singh(a)intel.com>"); +MODULE_LICENSE("GPL v2"); +MODULE_ALIAS("platform:cnl_rt274"); -- 2.16.2

2 5

[alsa-devel] [PATCH] ASoC: hdac_hdmi: Program pin-port-mux during jack report event
by Guneshwor Singh 19 Mar '18

19 Mar '18

From: Sriram Periyasamy <sriramx.periyasamy(a)intel.com> During d3/d0 cycle, the connection selection index of all pins points to the default value. This needs to be restored to ensure audio is restored after d3/d0 cycle. So store the connection selection index and program it during jack report event which gets invoked in cases like d3/d0 cycle, hot plug detection when multiple displays are connected. Signed-off-by: Sriram Periyasamy <sriramx.periyasamy(a)intel.com> Signed-off-by: Sanyog Kale <sanyog.r.kale(a)intel.com> Signed-off-by: Jeeja KP <jeeja.kp(a)intel.com> Signed-off-by: Guneshwor Singh <guneshwor.o.singh(a)intel.com> --- sound/soc/codecs/hdac_hdmi.c | 32 ++++++++++++++++++++++++++++++++ 1 file changed, 32 insertions(+) diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c index dba6f4c5074a..cabcb75bf4fc 100644 --- a/sound/soc/codecs/hdac_hdmi.c +++ b/sound/soc/codecs/hdac_hdmi.c @@ -32,6 +32,7 @@ #include <sound/hda_chmap.h> #include "../../hda/local.h" #include "hdac_hdmi.h" +#include <sound/hda_regmap.h> #define NAME_SIZE 32 @@ -83,6 +84,7 @@ struct hdac_hdmi_pin { struct list_head head; hda_nid_t nid; bool mst_capable; + int conn_index; struct hdac_hdmi_port *ports; int num_ports; struct hdac_ext_device *edev; @@ -141,6 +143,9 @@ struct hdac_hdmi_priv { #define hdev_to_hdmi_priv(_hdev) ((to_ehdac_device(_hdev))->private_data) +static int hdac_hdmi_port_select_set(struct hdac_ext_device *edev, + struct hdac_hdmi_port *port); + static struct hdac_hdmi_pcm * hdac_hdmi_get_pcm_from_cvt(struct hdac_hdmi_priv *hdmi, struct hdac_hdmi_cvt *cvt) @@ -159,6 +164,7 @@ static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm, struct hdac_hdmi_port *port, bool is_connect) { struct hdac_ext_device *edev = port->pin->edev; + int cmd, err; if (is_connect) snd_soc_dapm_enable_pin(port->dapm, port->jack_pin); @@ -166,6 +172,29 @@ static void hdac_hdmi_jack_report(struct hdac_hdmi_pcm *pcm, snd_soc_dapm_disable_pin(port->dapm, port->jack_pin); if (is_connect) { + /* set the device if pin is mst_capable */ + if (hdac_hdmi_port_select_set(edev, port) < 0) { + dev_err(&edev->hdev.dev, + "port %d device select fail\n", port->id); + return; + } + /* + * Restore the connection selection index of the + * respective pin. + */ + if (port->pin->conn_index > 0) { + cmd = snd_hdac_regmap_encode_verb(port->pin->nid, + AC_VERB_SET_CONNECT_SEL); + err = snd_hdac_regmap_write_raw(&edev->hdev, cmd, + port->pin->conn_index - 1); + if (err < 0) { + dev_err(&edev->hdev.dev, + "pin %d conn select index fail %d\n", + port->pin->nid, err); + return; + } + } + /* * Report Jack connect event when a device is connected * for the first time where same PCM is attached to multiple @@ -891,6 +920,9 @@ static int hdac_hdmi_set_pin_port_mux(struct snd_kcontrol *kcontrol, } } + if (ucontrol->value.enumerated.item[0] > 0) + port->pin->conn_index = ucontrol->value.enumerated.item[0]; + /* * Jack status is not reported during device probe as the * PCMs are not registered by then. So report it here. -- 2.16.1

2 1

[alsa-devel] [PATCH 4/5] ASoC: samsung: Document compatible strings for Odroid X/X2, Odroid XU3
by Sylwester Nawrocki 19 Mar '18

19 Mar '18

There are no significant differences across Odroid X, Odroid-X2, Odroid-U3 and Odroid-XU3 with regards to the audio subsystem so add more entries to the compatible strings list. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- Documentation/devicetree/bindings/sound/samsung,odroid.txt | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/samsung,odroid.txt b/Documentation/devicetree/bindings/sound/samsung,odroid.txt index f35a2694eb04..a50a0c6a354b 100644 --- a/Documentation/devicetree/bindings/sound/samsung,odroid.txt +++ b/Documentation/devicetree/bindings/sound/samsung,odroid.txt @@ -1,8 +1,10 @@ -Samsung Exynos Odroid XU3/XU4 audio complex with MAX98090 codec +Hardkernel Odroid X/X2/U3/XU3/XU4 audio complex with MAX98090 codec Required properties: - - compatible - "hardkernel,odroid-xu3-audio" - for Odroid XU3 board, + - compatible - "hardkernel,odroid-u3-audio" - for Odroid U3 board, + "hardkernel,odroid-x-audio" - for Odroid X, Odroid X2 boards, + "hardkernel,odroid-xu3-audio" - for Odroid XU3 board, "hardkernel,odroid-xu4-audio" - for Odroid XU4 board - model - the user-visible name of this sound complex - clocks - should contain entries matching clock names in the clock-names -- 2.14.2

3 5

[alsa-devel] [PATCH v2] ASoC: ssm2602: Replace reg_default_raw with reg_default
by James Kelly 19 Mar '18

19 Mar '18

SSM2602 driver is broken on recent kernels (at least since 4.9). User space applications such as amixer or alsamixer get EIO when attempting to access codec controls via the relevant IOCTLs. Root cause of these failures is the regcache_hw_init function in drivers/base/regmap/regcache.c, which prevents regmap cache initalization from the reg_defaults_raw element of the regmap_config structure when registers are write only. It also disables the regmap cache entirely when all registers are write only or volatile as is the case for the SSM2602 driver. Using the reg_defaults element of the regmap_config structure rather than the reg_defaults_raw element to initalize the regmap cache avoids the logic in the regcache_hw_init function entirely. It also makes this driver consistent with other ASoC codec drivers, as this driver was the ONLY codec driver that used the reg_defaults_raw element to initalize the cache. Tested on Digilent Zybo Z7 development board which has a SSM2603 codec chip connected to a Xilinx Zynq SoC. Signed-off-by: James Kelly <jamespeterkelly(a)gmail.com> --- Sorry I forgot the signature in v1. sound/soc/codecs/ssm2602.c | 19 +++++++++++++------ 1 file changed, 13 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 031e0eabcc1f..501a4e73b185 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -54,10 +54,17 @@ struct ssm2602_priv { * using 2 wire for device control, so we cache them instead. * There is no point in caching the reset register */ -static const u16 ssm2602_reg[SSM2602_CACHEREGNUM] = { - 0x0097, 0x0097, 0x0079, 0x0079, - 0x000a, 0x0008, 0x009f, 0x000a, - 0x0000, 0x0000 +static const struct reg_default ssm2602_reg[SSM2602_CACHEREGNUM] = { + { .reg = 0x00, .def = 0x0097 }, + { .reg = 0x01, .def = 0x0097 }, + { .reg = 0x02, .def = 0x0079 }, + { .reg = 0x03, .def = 0x0079 }, + { .reg = 0x04, .def = 0x000a }, + { .reg = 0x05, .def = 0x0008 }, + { .reg = 0x06, .def = 0x009f }, + { .reg = 0x07, .def = 0x000a }, + { .reg = 0x08, .def = 0x0000 }, + { .reg = 0x09, .def = 0x0000 } }; @@ -621,8 +628,8 @@ const struct regmap_config ssm2602_regmap_config = { .volatile_reg = ssm2602_register_volatile, .cache_type = REGCACHE_RBTREE, - .reg_defaults_raw = ssm2602_reg, - .num_reg_defaults_raw = ARRAY_SIZE(ssm2602_reg), + .reg_defaults = ssm2602_reg, + .num_reg_defaults = ARRAY_SIZE(ssm2602_reg), }; EXPORT_SYMBOL_GPL(ssm2602_regmap_config); -- 2.14.3 (Apple Git-98)

1 0

[alsa-devel] [PATCH] ASoC: pcm: improve debug output for DPCM BE searching.
by Liam Girdwood 19 Mar '18

19 Mar '18

Improve the DPCM BE search debug output to make it easier to debug issues in topologies. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/soc-pcm.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index ea36de977c93..4ce489165a6d 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -1404,12 +1404,18 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, struct snd_soc_pcm_runtime *be; int i; + dev_dbg(card->dev, "ASoC: find BE for widget %s\n", widget->name); + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { list_for_each_entry(be, &card->rtd_list, list) { if (!be->dai_link->no_pcm) continue; + dev_dbg(card->dev, "ASoC: try BE : %s\n", + be->cpu_dai->playback_widget ? + be->cpu_dai->playback_widget->name : "(not set)"); + if (be->cpu_dai->playback_widget == widget) return be; @@ -1426,6 +1432,10 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, if (!be->dai_link->no_pcm) continue; + dev_dbg(card->dev, "ASoC: try BE %s\n", + be->cpu_dai->capture_widget ? + be->cpu_dai->capture_widget->name : "(not set)"); + if (be->cpu_dai->capture_widget == widget) return be; @@ -1437,6 +1447,7 @@ static struct snd_soc_pcm_runtime *dpcm_get_be(struct snd_soc_card *card, } } + /* dai link name and stream name set correctly ? */ dev_err(card->dev, "ASoC: can't get %s BE for %s\n", stream ? "capture" : "playback", widget->name); return NULL; -- 2.14.1

2 1

[alsa-devel] [PATCH] ASoC: topology: Check widget kcontrols before deref.
by Liam Girdwood 19 Mar '18

19 Mar '18

Validate the topology input before we dereference the pointer. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- sound/soc/soc-topology.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index 4616286869a8..78197e82fc4f 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -508,6 +508,9 @@ static void remove_widget(struct snd_soc_component *comp, if (dobj->ops && dobj->ops->widget_unload) dobj->ops->widget_unload(comp, dobj); + if (!w->kcontrols) + goto free_news; + /* * Dynamic Widgets either have 1..N enum kcontrols or mixers. * The enum may either have an array of values or strings. @@ -528,7 +531,6 @@ static void remove_widget(struct snd_soc_component *comp, kfree(se); } - kfree(w->kcontrol_news); } else { /* volume mixer or bytes controls */ for (i = 0; i < w->num_kcontrols; i++) { @@ -545,8 +547,11 @@ static void remove_widget(struct snd_soc_component *comp, kfree((void *)kcontrol->private_value); snd_ctl_remove(card, kcontrol); } - kfree(w->kcontrol_news); } + +free_news: + kfree(w->kcontrol_news); + /* widget w is freed by soc-dapm.c */ } -- 2.14.1

2 1

[alsa-devel] [BUG] Kernel crash on Allwinner H3 due to sound core changes
by Jernej Škrabec 19 Mar '18

19 Mar '18

Hi all, with todays linux-next (next-20180228), kernel on Allwinner H3 SoC crashes with dmesg like that: https://pastebin.com/raw/0D5JeaJ8 I bisected the kernel and first offending commit is: be7ee5f32a9a ("ASoC: soc-generic-dmaengine-pcm: replace platform to component") I know that crash message is completely unrelated to sound subsystem, but it turns out that if I disable CONFIG_SND_SUN4I_CODEC kernel works ok, but this way I lose analog audio output. Any suggestions what can be the issue? Best regards, Jernej

3 12

[alsa-devel] [PATCH] soc-core: don't call kfree() for component
by Kuninori Morimoto 19 Mar '18

19 Mar '18

From: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> When driver register its component to ALSA SoC, almost all drivers are using snd_soc_register_component(), but soc-generic-dmaengine-pcm is using snd_soc_add_component(). Existing component function had been assumed that registered component was allocated, and it calling kfree() for it. But, the user who used snd_soc_add_component() doesn't. This patch uses devm_kzalloc() instead of kzalloc() for component, and doesn't call kree() anymore. This patch fixes commit be7ee5f32a9a ("ASoC: soc-generic-dmaengine-pcm: replace platform to component"). Allwinner H3 SoC will crash without this patch. Thanks Jernej report. Reported-by: Jernej Skrabec <jernej.skrabec(a)siol.net> Signed-off-by: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> Tested-by: Jernej Skrabec <jernej.skrabec(a)siol.net> --- sound/soc/soc-core.c | 4 +--- 1 file changed, 1 insertion(+), 3 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index c0edac8..4a8de23 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -3476,7 +3476,6 @@ int snd_soc_add_component(struct device *dev, err_cleanup: snd_soc_component_cleanup(component); err_free: - kfree(component); return ret; } EXPORT_SYMBOL_GPL(snd_soc_add_component); @@ -3488,7 +3487,7 @@ int snd_soc_register_component(struct device *dev, { struct snd_soc_component *component; - component = kzalloc(sizeof(*component), GFP_KERNEL); + component = devm_kzalloc(dev, sizeof(*component), GFP_KERNEL); if (!component) return -ENOMEM; @@ -3523,7 +3522,6 @@ static int __snd_soc_unregister_component(struct device *dev) if (found) { snd_soc_component_cleanup(component); - kfree(component); } return found; -- 1.9.1

1 0

[alsa-devel] [PATCH v1 0/6] ALSA: usb: UAC3. Add support for Basic Audio Device (BADD)
by Jorge Sanjuan 19 Mar '18

19 Mar '18

This adds functionality for the Basic Audio Device (BADD) subset that is defined in the USB Audio Class 3 (UAC3). In a BADD compliant driver, all Class Specific descriptors are inferred and so they should all be generated in the *host* as the device wont expose those. This also applies to the new HIGH CAPABILITY descriptor as the GET request is not to be supported in a BADD compliant driver. The new class requires the device to have 3 usb configurations as follows: 1: Legacy Mode: UAC1 or UAC2. 2: BADD device with a prefined topology. (Minimum). 3: UAC3 device for more detailed description or more complex devices that can't be covered by the BADD profile. This patch series also includes some minor fixes to sound/usb. It has been tested with an actual UAC3 compliant device in different host machines and all configs (UAC1/2, BADD and UAC3) work. Also, this has been implemented on top of the the patch which adds UAC3 support to the usb sound card driver: commit ddd452d7b04b86fb5f9285a19ac54deca9264ac1 Author: Ruslan Bilovol <ruslan.bilovol(a)gmail.com> Date: Tue Nov 7 04:01:20 2017 +0200 Jorge Sanjuan (6): usb: audio: Add bi-directional terminal types. usb: audio: Fix variable length field to be variable. ALSA: usb: UAC3. Add initial mixer support. ALSA: usb: Only get AudioControl header for UAC1 class. ALSA: usb: Use Class Specific EP for UAC3 devices. ALSA: usb: ADC3: Add initial BADD spec support include/linux/usb/audio-v3.h | 2 +- include/uapi/linux/usb/audio.h | 21 +- sound/usb/Makefile | 3 +- sound/usb/badd.c | 518 +++++++++++++++++++++++++++++++++++++++++ sound/usb/badd.h | 14 ++ sound/usb/card.c | 25 +- sound/usb/mixer.c | 24 ++ sound/usb/stream.c | 165 ++++++++----- sound/usb/usbaudio.h | 1 + 9 files changed, 702 insertions(+), 71 deletions(-) create mode 100644 sound/usb/badd.c create mode 100644 sound/usb/badd.h -- 2.11.0

4 11

[alsa-devel] [PATCH -next] ASoC: wm8400: Use devm_snd_soc_register_component()
by Wei Yongjun 19 Mar '18

19 Mar '18

Since the remove callback is removed, the snd_soc_unregister_component() is missing when remove device. Using devm_snd_soc_register_component() instead of snd_soc_register_component(). Fixes: 10dc44c6462d ("ASoC: wm8400: replace codec to component") Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/codecs/wm8400.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/wm8400.c b/sound/soc/codecs/wm8400.c index c237365..57b2206 100644 --- a/sound/soc/codecs/wm8400.c +++ b/sound/soc/codecs/wm8400.c @@ -1343,7 +1343,7 @@ static void wm8400_component_remove(struct snd_soc_component *component) static int wm8400_probe(struct platform_device *pdev) { - return snd_soc_register_component(&pdev->dev, + return devm_snd_soc_register_component(&pdev->dev, &soc_component_dev_wm8400, &wm8400_dai, 1); }

4 3

[alsa-devel] [PATCH -next] ASoC: mediatek: mt2701: drop unnessary snd_soc_unregister_component()
by Wei Yongjun 19 Mar '18

19 Mar '18

It's not necessary to unregister a component registered with devm_snd_soc_register_component(). Fixes: f1b5bf07365d ("ASoC: mt2701/mt8173: replace platform to componen") Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/mediatek/mt2701/mt2701-afe-pcm.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c index f7e0702..43837a3 100644 --- a/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c +++ b/sound/soc/mediatek/mt2701/mt2701-afe-pcm.c @@ -1561,8 +1561,6 @@ static int mt2701_afe_pcm_dev_remove(struct platform_device *pdev) if (!pm_runtime_status_suspended(&pdev->dev)) mt2701_afe_runtime_suspend(&pdev->dev); - snd_soc_unregister_component(&pdev->dev); - return 0; }

4 3

[alsa-devel] [PATCH 1/5] ASoC: samsung: odroid: Fix 32000 sample rate handling
by Sylwester Nawrocki 19 Mar '18

19 Mar '18

In case of sample rates lower than 44100 currently there is too low MCLK frequency set for the CODEC. Playback fails with following errors: $ speaker-test -c2 -t sine -f 1500 -l2 -r 32000 Sine wave rate is 1500.0000Hz Rate set to 32000Hz (requested 32000Hz) Buffer size range from 128 to 131072 Period size range from 64 to 65536 Using max buffer size 131072 Periods = 4 Unable to set hw params for playback: Invalid argument Setting of hwparams failed: Invalid argument [ 497.883700] max98090 1-0010: Invalid master clock frequency To fix this the I2S root clock's frequency is increased, depending on sampling rate. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/odroid.c | 11 +++++++---- 1 file changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c index 92d750806d1d..e698e67f5f84 100644 --- a/sound/soc/samsung/odroid.c +++ b/sound/soc/samsung/odroid.c @@ -36,23 +36,26 @@ static int odroid_card_hw_params(struct snd_pcm_substream *substream, { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct odroid_priv *priv = snd_soc_card_get_drvdata(rtd->card); - unsigned int pll_freq, rclk_freq; + unsigned int pll_freq, rclk_freq, rfs; int ret; switch (params_rate(params)) { - case 32000: case 64000: - pll_freq = 131072006U; + pll_freq = 196608001U; + rfs = 384; break; case 44100: case 88200: case 176400: pll_freq = 180633609U; + rfs = 512; break; + case 32000: case 48000: case 96000: case 192000: pll_freq = 196608001U; + rfs = 512; break; default: return -EINVAL; @@ -67,7 +70,7 @@ static int odroid_card_hw_params(struct snd_pcm_substream *substream, * frequency values due to the EPLL output frequency not being exact * multiple of the audio sampling rate. */ - rclk_freq = params_rate(params) * 256 + 1; + rclk_freq = params_rate(params) * rfs + 1; ret = clk_set_rate(priv->sclk_i2s, rclk_freq); if (ret < 0) -- 2.14.2

3 8

[alsa-devel] [PATCH v2 0/2] TDA7419 audio processor driver
by Matt Porter 19 Mar '18

19 Mar '18

This series adds an ASoC component driver for the ST TDA7419 audio processor which is commonly used in automotive audio applications. The datasheet can be found at http://www.st.com/resource/en/datasheet/tda7419.pdf Changes since v1: * add supply and mute gpios to binding * remove writeable register, already the default * add "Switch" to all switch controls * move register default reset to probe and comment * rework part-specific volume handling to handle register power-on default values * remove custom register init values * add audio route pins to binding * add DAPM support Matt Porter (2): ASoC: add tda7419 audio processor binding ASoC: add tda7419 audio processor driver .../devicetree/bindings/sound/tda7419.txt | 38 ++ sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tda7419.c | 654 +++++++++++++++++++++ 4 files changed, 700 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tda7419.txt create mode 100644 sound/soc/codecs/tda7419.c -- 2.11.0

2 4

[alsa-devel] [PATCH v3 0/2] ASoC: Add support for DAC PCM1789
by Mylène Josserand 19 Mar '18

19 Mar '18

Hello everyone, The current series is the 3rd version to add the support of Texas Instrument's DAC PCM1789. This DAC is very minimalist and does not have many registers. It is important to notice that this DAC needs to always have clocks enabled (even without any data) otherwise it will be in a "desynchronized" state and can not send data correctly. This issue has been solved by performing a reset each time a sound is played. This reset can produce a "pop" noise. Depending on your DAI, you will need to provide and enable the MCLK to be able to communicate with this codec throught i2c. Changes since v2: - Update SPDX headers according to Mark Brown's review. - Update the dt bindings according to Rob Herring's review. Changes since v1: - Create a new file to support pcm1789 instead of converting the existing pcm179x driver. All the patches are merged into one patch. - Update the code to use gpiod for the reset. - Add some fixes according to Thomas Petazzoni's reviews - Create a new patch to add device-tree bindings for this new DAC. Thank you in advance for any review. Best regards, Mylène Mylène Josserand (2): ASoC: codecs: Add support for PCM1789 ASoC: Add bindings for PCM1789 .../devicetree/bindings/sound/pcm1789.txt | 22 ++ sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/pcm1789-i2c.c | 62 +++++ sound/soc/codecs/pcm1789.c | 274 +++++++++++++++++++++ sound/soc/codecs/pcm1789.h | 17 ++ 6 files changed, 391 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/pcm1789.txt create mode 100644 sound/soc/codecs/pcm1789-i2c.c create mode 100644 sound/soc/codecs/pcm1789.c create mode 100644 sound/soc/codecs/pcm1789.h -- 2.11.0

4 5

[alsa-devel] [PATCH 0/2] add syscon property to binding document
by Katsuhiro Suzuki 19 Mar '18

19 Mar '18

This patch adds syscon property for specifying soc-glue core. The soc-glue core is used for changing the state of S/PDIF signal output pin to signal output state or Hi-Z state. After resetting of SoC Hi-Z state is selected. Currently, we set this mode in local changed U-Boot or some bootloaders fixedly. This patch removes dependencies of bootloader from UniPhier audio system driver. Katsuhiro Suzuki (2): ASoC: uniphier: add syscon property to binding document ASoC: uniphier: add syscon property for UniPhier sound system .../devicetree/bindings/sound/uniphier,aio.txt | 8 ++++++++ sound/soc/uniphier/aio-core.c | 21 +++++++++++++++++++++ sound/soc/uniphier/aio-cpu.c | 11 +++++++++++ sound/soc/uniphier/aio-reg.h | 3 +++ sound/soc/uniphier/aio.h | 2 ++ 5 files changed, 45 insertions(+) -- 2.16.2

2 4

[alsa-devel] [PATCH -next] ASoC: da7210: Use devm_snd_soc_register_component()
by Wei Yongjun 19 Mar '18

19 Mar '18

Since the remove callback is removed, the snd_soc_unregister_component() is missing when remove device. Using devm_snd_soc_register_component() instead of snd_soc_register_component(). Fixes: d06f33aed85c ("ASoC: da7210: replace codec to component") Signed-off-by: Wei Yongjun <weiyongjun1(a)huawei.com> --- sound/soc/codecs/da7210.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/da7210.c b/sound/soc/codecs/da7210.c index 07153be..a664111 100644 --- a/sound/soc/codecs/da7210.c +++ b/sound/soc/codecs/da7210.c @@ -1233,7 +1233,7 @@ static int da7210_i2c_probe(struct i2c_client *i2c, if (ret != 0) dev_warn(&i2c->dev, "Failed to apply regmap patch: %d\n", ret); - ret = snd_soc_register_component(&i2c->dev, + ret = devm_snd_soc_register_component(&i2c->dev, &soc_component_dev_da7210, &da7210_dai, 1); if (ret < 0) dev_err(&i2c->dev, "Failed to register component: %d\n", ret);

2 1

[alsa-devel] [PATCH] ASoC: rsnd: Document R-Car M3-W support
by Geert Uytterhoeven 19 Mar '18

19 Mar '18

Document support for the sound modules in the Renesas M3-W (r8a7796) SoC. No driver update is needed. Signed-off-by: Geert Uytterhoeven <geert+renesas(a)glider.be> --- "renesas,rcar_sound-r8a7796" has been in use since quite a while, but the binding update was still missing. --- Documentation/devicetree/bindings/sound/renesas,rsnd.txt | 1 + 1 file changed, 1 insertion(+) diff --git a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt index 5bed9a5957729453..b86d790f630ff79a 100644 --- a/Documentation/devicetree/bindings/sound/renesas,rsnd.txt +++ b/Documentation/devicetree/bindings/sound/renesas,rsnd.txt @@ -351,6 +351,7 @@ Required properties: - "renesas,rcar_sound-r8a7793" (R-Car M2-N) - "renesas,rcar_sound-r8a7794" (R-Car E2) - "renesas,rcar_sound-r8a7795" (R-Car H3) + - "renesas,rcar_sound-r8a7796" (R-Car M3-W) - reg : Should contain the register physical address. required register is SRU/ADG/SSI if generation1 -- 2.7.4

3 2

[alsa-devel] [PATCH 0/2] TDA7419 audio processor driver
by Matt Porter 18 Mar '18

18 Mar '18

This series adds an ASoC component driver for the ST TDA7419 audio processor which is commonly used in automotive audio applications. The datasheet can be found at http://www.st.com/resource/en/datasheet/tda7419.pdf Matt Porter (2): ASoC: add tda7419 audio processor binding ASoC: add tda7419 audio processor driver .../devicetree/bindings/sound/tda7419.txt | 15 + sound/soc/codecs/Kconfig | 6 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/tda7419.c | 571 +++++++++++++++++++++ 4 files changed, 594 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/tda7419.txt create mode 100644 sound/soc/codecs/tda7419.c -- 2.11.0

4 10

[alsa-devel] [PATCH 1/3] sound: usb: line6: Replace mdelay with msleep in line6_read_data
by Jia-Ju Bai 18 Mar '18

18 Mar '18

line6_read_data() is never called in atomic context. The call chains ending up at line6_read_data() are: [1] line6_read_data() <- line6_read_serial_number() -> pod_startup4() [2] line6_read_data() <- line6_read_serial_number() <- podhd_startup_workqueue() [3] line6_read_data() <- podhd_dev_start() <- podhd_startup_workqueue() [4] line6_read_data() <- line6_read_serial_number() <- toneport_init() <- line6_probe() [5] line6_read_data() <- toneport_init() <- line6_probe() pod_startup4() and podhd_startup_workqueue() are set in INIT_WORK(). line6_probe() is called in toneport_probe(), pod_probe(), podhd_probe() and variax_probe(), which are set as ".probe" in usb_driver structure. These functions are not called in atomic context. Despite never getting called from atomic context, line6_read_data() calls mdelay for busy wait. That is not necessary and can be replaced with msleep to avoid busy waiting. This is found by a static analysis tool named DCNS written by myself. Signed-off-by: Jia-Ju Bai <baijiaju1990(a)gmail.com> --- sound/usb/line6/driver.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index 0ff5a7d..af99e73 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -356,7 +356,7 @@ int line6_read_data(struct usb_line6 *line6, unsigned address, void *data, /* Wait for data length. We'll get 0xff until length arrives. */ for (count = 0; count < LINE6_READ_WRITE_MAX_RETRIES; count++) { - mdelay(LINE6_READ_WRITE_STATUS_DELAY); + msleep(LINE6_READ_WRITE_STATUS_DELAY); ret = usb_control_msg(usbdev, usb_rcvctrlpipe(usbdev, 0), 0x67, USB_TYPE_VENDOR | USB_RECIP_DEVICE | -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: samsung: Mark unused Odroid compatibles as deprecated
by Krzysztof Kozlowski 18 Mar '18

18 Mar '18

Compatible for XU4 audio is not being used. Instead the board uses the same compatible as XU3. The devices are now just compatible so they should use the same value. Mark hardkernel,odroid-xu4-audio as being deprecated so in future could be removed to limit useless compatibles. Additionally add older compatibles appearing in the bindings. Signed-off-by: Krzysztof Kozlowski <krzk(a)kernel.org> --- Documentation/devicetree/bindings/sound/samsung,odroid.txt | 4 +++- 1 file changed, 3 insertions(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/samsung,odroid.txt b/Documentation/devicetree/bindings/sound/samsung,odroid.txt index f35a2694eb04..4a18d83456c3 100644 --- a/Documentation/devicetree/bindings/sound/samsung,odroid.txt +++ b/Documentation/devicetree/bindings/sound/samsung,odroid.txt @@ -3,7 +3,9 @@ Samsung Exynos Odroid XU3/XU4 audio complex with MAX98090 codec Required properties: - compatible - "hardkernel,odroid-xu3-audio" - for Odroid XU3 board, - "hardkernel,odroid-xu4-audio" - for Odroid XU4 board + "hardkernel,odroid-xu4-audio" - for Odroid XU4 board (deprecated) + "samsung,odroid-xu3-audio" - for Odroid XU4 board (deprecated) + "samsung,odroid-xu4-audio" - for Odroid XU4 board (deprecated) - model - the user-visible name of this sound complex - clocks - should contain entries matching clock names in the clock-names property -- 2.14.1

1 0

[alsa-devel] [PATCH 3/3] sound: usb: line6: Replace GFP_ATOMIC with GFP_KERNEL in line6_alloc_sysex_buffer
by Jia-Ju Bai 18 Mar '18

18 Mar '18

line6_alloc_sysex_buffer() is never called in atomic context. The call chain ending up at line6_alloc_sysex_buffer() are: [1] line6_alloc_sysex_buffer() <- pod_alloc_sysex_buffer() <- pod_set_system_param_int() <- snd_pod_control_monitor_put() snd_pod_control_monitor_put() is only set as ".put" in snd_kcontrol_new structure "pod_control_monitor". And ->put() is called in snd_ctl_elem_write() (sound/core/control.c) snd_ctl_elem_write() are called in ctl_elem_write_user() and snd_ctl_elem_write_user(). These functions are not called in atomic context. Despite never getting called from atomic context, line6_alloc_sysex_buffer() calls kmalloc with GFP_ATOMIC, which waits busily for allocation. GFP_ATOMIC is not necessary and can be replaced with GFP_KERNEL to avoid busy waiting. This is found by a static analysis tool named DCNS written by myself. Signed-off-by: Jia-Ju Bai <baijiaju1990(a)gmail.com> --- sound/usb/line6/driver.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index 0ff5a7d..70a509e 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -265,7 +265,7 @@ int line6_send_sysex_message(struct usb_line6 *line6, const char *buffer, char *line6_alloc_sysex_buffer(struct usb_line6 *line6, int code1, int code2, int size) { - char *buffer = kmalloc(size + SYSEX_EXTRA_SIZE, GFP_ATOMIC); + char *buffer = kmalloc(size + SYSEX_EXTRA_SIZE, GFP_KERNEL); if (!buffer) return NULL; -- 1.9.1

1 0

[alsa-devel] [PATCH 2/3] sound: usb: line6: Replace mdelay with msleep in line6_write_data
by Jia-Ju Bai 18 Mar '18

18 Mar '18

line6_write_data() is never called in atomic context. The call chains ending up at line6_write_data() are: [1] line6_write_data() <- toneport_setup() <- toneport_init() <- line6_probe() [2] line6_write_data() <- toneport_setup() <- toneport_reset_resume() line6_probe() is called in toneport_probe(), pod_probe(), podhd_probe() and variax_probe(), which are set as ".probe" in usb_driver structure. toneport_reset_resume() is set as ".reset_resume" in usb_driver structure. These functions are not called in atomic context. Despite never getting called from atomic context, line6_write_data() calls mdelay for busy wait. That is not necessary and can be replaced with msleep to avoid busy waiting. This is found by a static analysis tool named DCNS written by myself. Signed-off-by: Jia-Ju Bai <baijiaju1990(a)gmail.com> --- sound/usb/line6/driver.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/line6/driver.c b/sound/usb/line6/driver.c index 0ff5a7d..29acd86 100644 --- a/sound/usb/line6/driver.c +++ b/sound/usb/line6/driver.c @@ -426,7 +426,7 @@ int line6_write_data(struct usb_line6 *line6, unsigned address, void *data, } for (count = 0; count < LINE6_READ_WRITE_MAX_RETRIES; count++) { - mdelay(LINE6_READ_WRITE_STATUS_DELAY); + msleep(LINE6_READ_WRITE_STATUS_DELAY); ret = usb_control_msg(usbdev, usb_rcvctrlpipe(usbdev, 0), 0x67, -- 1.9.1

1 0

[alsa-devel] unstable channel assignments for 8 channel codec
by Daniel Loranger 18 Mar '18

18 Mar '18

Hello all, I don't know if anyone can help me, or if this is even the right place to track down an issue I am facing. I have tried the developer, and filed a bug in github against the c code, but not getting any responses. I am using an audio injector octo sound board on a raspberry pi 3. this sound board has 6 inputs and 8 outputs. The issue I am facing is the audio channels appear to be randomly assigned each time speaker-test utility is ran. My best guess at this point is there is a missing configuration in either the C driver or in the alsa configuration. I am hoping the latter hence my reaching out here. My research has lead me to think that the missing information is related to the channel assignments are missing from the config, but I am out of my league in understanding the right way to go about this. here is the driver if that is relevant at all https://github.com/raspberrypi/linux/blob/rpi-4.9.y/sound/soc/bcm/audioinje… I have configured per the device manufacturers instructions as follow here is the asoundrc contents pcm.!default { # type hw # card 0 type plug slave.pcm "anyChannelCount" } ctl.!default { type hw card 0 } Here is the asound.conf contents pcm.!default { # type hw # card 0 type plug slave.pcm "anyChannelCount" } ctl.!default { type hw card 0 } pcm.anyChannelCount { type route slave.pcm "hw:0" slave.channels 8; ttable { 0.0 1 1.1 1 2.2 1 3.3 1 4.4 1 5.5 1 6.6 1 7.7 1 } } ctl.anyChannelCount { type hw; card 0; } -- Daniel Loranger

1 0

[alsa-devel] [PATCH] ALSA: hda/realtek - Always immediately update mute LED with pin VREF
by Takashi Iwai 17 Mar '18

17 Mar '18

Some HP laptops have a mute mute LED controlled by a pin VREF. The Realtek codec driver updates the VREF via vmaster hook by calling snd_hda_set_pin_ctl_cache(). This works fine as long as the driver is running in a normal mode. However, when the VREF change happens during the codec being in runtime PM suspend, the regmap access will skip and postpone the actual register change. This ends up with the unchanged LED status until the next runtime PM resume even if you change the Master mute switch. (Interestingly, the machine keeps the LED status even after the codec goes into D3 -- but it's another story.) For improving this usability, let the driver temporarily powering up / down only during the pin VREF change. This can be achieved easily by wrapping the call with snd_hda_power_up_pm() / *_down_pm(). Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=199073 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 9af301c6bba2..5ef056f2aecc 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -3596,8 +3596,12 @@ static void alc269_fixup_mic_mute_hook(void *private_data, int enabled) pinval = snd_hda_codec_get_pin_target(codec, spec->mute_led_nid); pinval &= ~AC_PINCTL_VREFEN; pinval |= enabled ? AC_PINCTL_VREF_HIZ : AC_PINCTL_VREF_80; - if (spec->mute_led_nid) + if (spec->mute_led_nid) { + /* temporarily power up/down for setting VREF */ + snd_hda_power_up_pm(codec); snd_hda_set_pin_ctl_cache(codec, spec->mute_led_nid, pinval); + snd_hda_power_down_pm(codec); + } } /* Make sure the led works even in runtime suspend */ -- 2.16.2

1 0

[alsa-devel] alsalib and snd_pcm_hw_params_set_rate_minmax
by Oleksandr Andrushchenko 16 Mar '18

16 Mar '18

Hi, Is it possible for user-space to reduce configuration space with snd_pcm_hw_params_set_rate_minmax and then change it with another snd_pcm_hw_params_set_rate_minmax with values out of the reduced config? For example, the initial min/max is 44100/48000 and I set 44100 first, e.g. snd_pcm_hw_params_set_rate_minmax(handle, hw_params, 44100, 0, 44100, 0) and then want snd_pcm_hw_params_set_rate_minmax(handle, hw_params, 48000, 0, 48000, 0) Obviously, the last call fails as we have already a reduced space of [44100; 44100]. Is there a way I can still set the range to [48000; 48000]? Thank you, Oleksandr P.S. This is in context of work done for [1] [1] https://www.spinics.net/lists/alsa-devel/msg75382.html

4 12

[alsa-devel] [PATCH 0/7] Enablement of BT I2S controller instance for AMD APUs
by Vijendar Mukunda 16 Mar '18

16 Mar '18

This patch set updates Audio CoProcessor (ACP) audio drivers to enable BT I2S controller instance. In Audio Coprocessor, There are three I2S controllers can be configured/enabled.(I2S SP, I2S MICSP, BT I2S) Default enabled I2S controller instance is I2S SP instance. There is a requirement to enable BT I2S controller Instance along with I2S SP controller instance. This patch set bring up functionality to render & capture on BT I2S controller instance. The current code is based on asoc-next, but I'm happy to rebase on whatever tree this ends up going through if there are any problems applying. Vijendar Mukunda (7): ASoC: amd: dma driver changes for BT I2S controller instance ASoC: amd: dma descriptor changes for BT I2S Instance ASoC: amd: Interrupt handler changes for BT I2S instance ASoC: amd: prepare callback modifications for bt i2s instance ASoC: amd: pointer and trigger callback modifications for bt i2s ASoC: amd: 16bit resolution support for bt i2s instance ASoC: amd: enabling bt i2s config after acp reset sound/soc/amd/acp-pcm-dma.c | 521 +++++++++++++++++++++++++++++++++++--------- sound/soc/amd/acp.h | 52 ++++- 2 files changed, 460 insertions(+), 113 deletions(-) -- 2.7.4

3 11

[alsa-devel] [PATCH] ALSA: core: Allow drivers to set R/W wait time.
by Liam Girdwood 16 Mar '18

16 Mar '18

Currently ALSA core blocks userspace for about 10 seconds for PCM R/W IO. This needs to be configurable for modern hardware like DSPs where no pointer update in milliseconds can indicate terminal DSP errors. Add a substream variable to set the wait time in ms. This allows userspace and drivers to recover more quickly from terminal DSP errors. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/sound/pcm.h | 6 ++++++ sound/core/pcm_lib.c | 15 ++++++++++----- 2 files changed, 16 insertions(+), 5 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index e054c583d3b3..e4694684c524 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -462,6 +462,7 @@ struct snd_pcm_substream { /* -- timer section -- */ struct snd_timer *timer; /* timer */ unsigned timer_running: 1; /* time is running */ + unsigned wait_time; /* time in ms for R/W to wait for avail */ /* -- next substream -- */ struct snd_pcm_substream *next; /* -- linked substreams -- */ @@ -579,6 +580,11 @@ int snd_pcm_start(struct snd_pcm_substream *substream); int snd_pcm_stop(struct snd_pcm_substream *substream, snd_pcm_state_t status); int snd_pcm_drain_done(struct snd_pcm_substream *substream); int snd_pcm_stop_xrun(struct snd_pcm_substream *substream); +static inline void snd_pcm_wait_time(struct snd_pcm_substream *substream, + unsigned wait_time) +{ + substream->wait_time = wait_time; +} #ifdef CONFIG_PM int snd_pcm_suspend(struct snd_pcm_substream *substream); int snd_pcm_suspend_all(struct snd_pcm *pcm); diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index a83152e7d387..2ee76c70f55f 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1839,12 +1839,17 @@ static int wait_for_avail(struct snd_pcm_substream *substream, if (runtime->no_period_wakeup) wait_time = MAX_SCHEDULE_TIMEOUT; else { - wait_time = 10; - if (runtime->rate) { - long t = runtime->period_size * 2 / runtime->rate; - wait_time = max(t, wait_time); + /* use wait time from substream if available */ + if (substream->wait_time) { + wait_time = msecs_to_jiffies(substream->wait_time); + } else { + wait_time = 10; + if (runtime->rate) { + long t = runtime->period_size * 2 / runtime->rate; + wait_time = max(t, wait_time); + } + wait_time = msecs_to_jiffies(wait_time * 1000); } - wait_time = msecs_to_jiffies(wait_time * 1000); } for (;;) { -- 2.14.1

3 7

[alsa-devel] [Xen-devel][PATCH v2 0/3] sndif: add explicit back and front synchronization
by Oleksandr Andrushchenko 16 Mar '18

16 Mar '18

From: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> Hello, all! In order to provide explicit synchronization between backend and frontend the following changes are introduced in the protocol: - bump protocol version to 2 - add new ring buffer for sending asynchronous events from backend to frontend to report number of bytes played by the frontend (XENSND_EVT_CUR_POS) - introduce trigger events for playback control: start/stop/pause/resume - add "req-" prefix to event-channel and ring-ref to unify naming of the Xen event channels for requests and events Changes since v1: 1. Changed protocol version definition from string to integer, so it can easily be used in comparisons. Konrad, I have removed your r-b tag for the reason of this change. 2. In order to provide explicit stream parameter negotiation between backend and frontend the following changes are introduced in the protocol: add XENSND_OP_HW_PARAM_QUERY request to read/update configuration space for the parameter given: request passes desired parameter interval (mask) and the response to this request returns min/max interval (mask) for the parameter to be used. Parameters supported by this request/response: - format mask - sample rate interval - number of channels interval - buffer size, interval, frames - period size, interval, frames Thank you, Oleksandr Andrushchenko Oleksandr Andrushchenko (3): sndif: Introduce protocol version sndif: Add explicit back and front synchronization sndif: Add explicit back and front parameter negotiation xen/include/public/io/sndif.h | 295 ++++++++++++++++++++++++++++++++++++++++-- 1 file changed, 286 insertions(+), 9 deletions(-) -- 2.7.4

2 6

Re: [alsa-devel] [PATCH] [sound] hdac-codec runtime suspended at PM:Suspend.
by Subhransu S. Prusty 15 Mar '18

15 Mar '18

On Wed, Mar 14, 2018 at 09:07:14PM +0530, Anshuman Gupta wrote: > On Wed, Mar 14, 2018 at 11:53:58AM +0100, Rafael J. Wysocki wrote: > > On Wed, Mar 14, 2018 at 11:38 AM, Anshuman Gupta > > <anshuman.gupta(a)intel.com> wrote: > > > On Mon, Mar 12, 2018 at 12:26:53PM +0100, Rafael J. Wysocki wrote: > > >> On Mon, Mar 12, 2018 at 12:17 PM, Anshuman Gupta > > >> <anshuman.gupta(a)intel.com> wrote: > > >> > > > >> > + if (pm_runtime_status_suspended(dev)) > > >> > + return; > > >> > > >> That, again, is somewhat fragile from the concurrency perspective. > > >> > > > > And here you want to avoid the below if the device is still suspended. > Yes, if we do not avoid the code below, complete callback takes about > 3 seconds due to snd_hdac_codec_read timed out because hdac controller > would be in runtime suspend state. > > > > Why is the below code located in the ->complete callback anyway? > > Shouldn't it be there in the ->resume one? > > > Yes even i am also having same doubt, why these power down and power up > sequences are part of prepare and complete callback. > Adding driver author "Subhransu S. Prusty" to provide more inputs on this. This driver needs a late resume as it receives a jack notification from the i915 driver and the skl controller driver resume may not have happened and in turn hda controller may not ready. This ensures a synchronization for jack event during resume from S3. I think this patch defeats the purpose. Regards, Subhransu > > >> > /* Power up afg */ > > >> > snd_hdac_codec_read(hdac, hdac->afg, 0, AC_VERB_SET_POWER_STATE, > > >> > AC_PWRST_D0); > > >> > -- > > >> > 2.7.4 > > -- > Thanks, > Anshuman --

2 1

[alsa-devel] [PATCH 00/47] arch-removal: device drivers
by Arnd Bergmann 15 Mar '18

15 Mar '18

Hi driver maintainers, I just posted one series with the removal of eight architectures, see https://lkml.org/lkml/2018/3/14/505 for details, or https://lwn.net/Articles/748074/ for more background. These are the device drivers that go along with them. I have already picked up the drivers for arch/metag/ into my tree, they were reviewed earlier. Please let me know if you have any concerns with the patch, or if you prefer to pick up the patches in your respective trees. I created the patches with 'git format-patch -D', so they will not apply without manually removing those files. For anything else, I'd keep the removal patches in my asm-generic tree and will send a pull request for 4.17 along with the actual arch removal. Arnd Arnd Bergmann edac: remove tile driver net: tile: remove ethernet drivers net: adi: remove blackfin ethernet drivers net: 8390: remove m32r specific bits net: remove cris etrax ethernet driver net: smsc: remove m32r specific smc91x configuration raid: remove tile specific raid6 implementation rtc: remove tile driver rtc: remove bfin driver char: remove obsolete ds1302 rtc driver char: remove tile-srom.c char: remove blackfin OTP driver pcmcia: remove m32r drivers pcmcia: remove blackfin driver ASoC: remove blackfin drivers video/logo: remove obsolete logo files fbdev: remove blackfin drivers fbdev: s1d13xxxfb: remove m32r specific hacks crypto: remove blackfin CRC driver media: platform: remove blackfin capture driver media: platform: remove m32r specific arv driver cpufreq: remove blackfin driver cpufreq: remove cris specific drivers gpio: remove etraxfs driver pinctrl: remove adi2/blackfin drivers ata: remove bf54x driver input: keyboard: remove bf54x driver input: misc: remove blackfin rotary driver mmc: remove bfin_sdh driver can: remove bfin_can driver watchdog: remove bfin_wdt driver mtd: maps: remove bfin-async-flash driver mtd: nand: remove bf5xx_nand driver spi: remove blackfin related host drivers i2c: remove bfin-twi driver pwm: remobe pwm-bfin driver usb: host: remove tilegx platform glue usb: musb: remove blackfin port usb: isp1362: remove blackfin arch glue serial: remove cris/etrax uart drivers serial: remove blackfin drivers serial: remove m32r_sio driver serial: remove tile uart driver tty: remove bfin_jtag_comm and hvc_bfin_jtag drivers tty: hvc: remove tile driver staging: irda: remove bfin_sir driver staging: iio: remove iio-trig-bfin-timer driver .../devicetree/bindings/gpio/gpio-etraxfs.txt | 22 - .../bindings/serial/axis,etraxfs-uart.txt | 22 - Documentation/watchdog/watchdog-parameters.txt | 5 - MAINTAINERS | 8 - drivers/ata/Kconfig | 9 - drivers/ata/Makefile | 1 - drivers/ata/pata_bf54x.c | 1703 -------- drivers/char/Kconfig | 48 - drivers/char/Makefile | 3 - drivers/char/bfin-otp.c | 237 -- drivers/char/ds1302.c | 357 -- drivers/char/tile-srom.c | 475 --- drivers/cpufreq/Makefile | 3 - drivers/cpufreq/blackfin-cpufreq.c | 217 - drivers/cpufreq/cris-artpec3-cpufreq.c | 93 - drivers/cpufreq/cris-etraxfs-cpufreq.c | 92 - drivers/crypto/Kconfig | 7 - drivers/crypto/Makefile | 1 - drivers/crypto/bfin_crc.c | 753 ---- drivers/crypto/bfin_crc.h | 124 - drivers/edac/Kconfig | 8 - drivers/edac/Makefile | 2 - drivers/edac/tile_edac.c | 265 -- drivers/gpio/Kconfig | 9 - drivers/gpio/Makefile | 1 - drivers/gpio/gpio-etraxfs.c | 475 --- drivers/i2c/busses/Kconfig | 18 - drivers/i2c/busses/Makefile | 1 - drivers/i2c/busses/i2c-bfin-twi.c | 737 ---- drivers/input/keyboard/Kconfig | 9 - drivers/input/keyboard/Makefile | 1 - drivers/input/keyboard/bf54x-keys.c | 396 -- drivers/input/misc/Kconfig | 9 - drivers/input/misc/Makefile | 1 - drivers/input/misc/bfin_rotary.c | 294 -- drivers/media/platform/Kconfig | 22 - drivers/media/platform/Makefile | 4 - drivers/media/platform/arv.c | 884 ---- drivers/media/platform/blackfin/Kconfig | 16 - drivers/media/platform/blackfin/Makefile | 2 - drivers/media/platform/blackfin/bfin_capture.c | 983 ----- drivers/media/platform/blackfin/ppi.c | 361 -- drivers/mmc/host/Kconfig | 19 - drivers/mmc/host/Makefile | 1 - drivers/mmc/host/bfin_sdh.c | 679 ---- drivers/mtd/maps/Kconfig | 10 - drivers/mtd/maps/Makefile | 1 - drivers/mtd/maps/bfin-async-flash.c | 196 - drivers/mtd/nand/raw/Kconfig | 32 - drivers/mtd/nand/raw/Makefile | 1 - drivers/mtd/nand/raw/bf5xx_nand.c | 861 ---- drivers/net/Makefile | 1 - drivers/net/can/Kconfig | 9 - drivers/net/can/Makefile | 1 - drivers/net/can/bfin_can.c | 784 ---- drivers/net/cris/Makefile | 1 - drivers/net/cris/eth_v10.c | 1742 -------- drivers/net/ethernet/8390/Kconfig | 3 +- drivers/net/ethernet/8390/ne.c | 23 +- drivers/net/ethernet/Kconfig | 2 - drivers/net/ethernet/Makefile | 2 - drivers/net/ethernet/adi/Kconfig | 66 - drivers/net/ethernet/adi/Makefile | 5 - drivers/net/ethernet/adi/bfin_mac.c | 1881 --------- drivers/net/ethernet/adi/bfin_mac.h | 104 - drivers/net/ethernet/smsc/Kconfig | 4 +- drivers/net/ethernet/smsc/smc91x.h | 26 - drivers/net/ethernet/tile/Kconfig | 18 - drivers/net/ethernet/tile/Makefile | 11 - drivers/net/ethernet/tile/tilegx.c | 2279 ----------- drivers/net/ethernet/tile/tilepro.c | 2397 ----------- drivers/pcmcia/Kconfig | 26 - drivers/pcmcia/Makefile | 3 - drivers/pcmcia/bfin_cf_pcmcia.c | 316 -- drivers/pcmcia/m32r_cfc.c | 786 ---- drivers/pcmcia/m32r_cfc.h | 88 - drivers/pcmcia/m32r_pcc.c | 763 ---- drivers/pcmcia/m32r_pcc.h | 66 - drivers/pinctrl/Kconfig | 19 - drivers/pinctrl/Makefile | 3 - drivers/pinctrl/pinctrl-adi2-bf54x.c | 588 --- drivers/pinctrl/pinctrl-adi2-bf60x.c | 517 --- drivers/pinctrl/pinctrl-adi2.c | 1114 ----- drivers/pinctrl/pinctrl-adi2.h | 75 - drivers/pwm/Kconfig | 9 - drivers/pwm/Makefile | 1 - drivers/pwm/pwm-bfin.c | 157 - drivers/rtc/Kconfig | 17 - drivers/rtc/Makefile | 2 - drivers/rtc/rtc-bfin.c | 448 --- drivers/rtc/rtc-tile.c | 143 - drivers/spi/Kconfig | 19 - drivers/spi/Makefile | 3 - drivers/spi/spi-adi-v3.c | 984 ----- drivers/spi/spi-bfin-sport.c | 919 ----- drivers/spi/spi-bfin5xx.c | 1462 ------- drivers/staging/iio/Kconfig | 1 - drivers/staging/iio/Makefile | 1 - drivers/staging/iio/trigger/Kconfig | 19 - drivers/staging/iio/trigger/Makefile | 5 - drivers/staging/iio/trigger/iio-trig-bfin-timer.c | 292 -- drivers/staging/iio/trigger/iio-trig-bfin-timer.h | 25 - drivers/staging/irda/drivers/Kconfig | 45 - drivers/staging/irda/drivers/Makefile | 1 - drivers/staging/irda/drivers/bfin_sir.c | 819 ---- drivers/staging/irda/drivers/bfin_sir.h | 93 - drivers/tty/Kconfig | 13 - drivers/tty/Makefile | 1 - drivers/tty/bfin_jtag_comm.c | 353 -- drivers/tty/hvc/Kconfig | 9 - drivers/tty/hvc/Makefile | 2 - drivers/tty/hvc/hvc_bfin_jtag.c | 104 - drivers/tty/hvc/hvc_tile.c | 196 - drivers/tty/serial/Kconfig | 198 - drivers/tty/serial/Makefile | 6 - drivers/tty/serial/bfin_sport_uart.c | 937 ----- drivers/tty/serial/bfin_sport_uart.h | 86 - drivers/tty/serial/bfin_uart.c | 1551 ------- drivers/tty/serial/crisv10.c | 4248 -------------------- drivers/tty/serial/crisv10.h | 133 - drivers/tty/serial/etraxfs-uart.c | 960 ----- drivers/tty/serial/m32r_sio.c | 1053 ----- drivers/tty/serial/m32r_sio_reg.h | 150 - drivers/tty/serial/tilegx.c | 689 ---- drivers/usb/host/Kconfig | 1 + drivers/usb/host/ehci-hcd.c | 5 - drivers/usb/host/ehci-tilegx.c | 207 - drivers/usb/host/isp1362.h | 44 - drivers/usb/host/ohci-hcd.c | 18 - drivers/usb/host/ohci-tilegx.c | 196 - drivers/usb/musb/Kconfig | 10 +- drivers/usb/musb/Makefile | 1 - drivers/usb/musb/blackfin.c | 623 --- drivers/usb/musb/blackfin.h | 81 - drivers/usb/musb/musb_core.c | 5 - drivers/usb/musb/musb_core.h | 30 - drivers/usb/musb/musb_debugfs.c | 2 - drivers/usb/musb/musb_dma.h | 11 - drivers/usb/musb/musb_gadget.c | 35 - drivers/usb/musb/musb_regs.h | 182 - drivers/usb/musb/musbhsdma.c | 5 - drivers/usb/musb/musbhsdma.h | 64 - drivers/video/fbdev/Kconfig | 103 - drivers/video/fbdev/Makefile | 5 - drivers/video/fbdev/bf537-lq035.c | 891 ---- drivers/video/fbdev/bf54x-lq043fb.c | 764 ---- drivers/video/fbdev/bfin-lq035q1-fb.c | 864 ---- drivers/video/fbdev/bfin-t350mcqb-fb.c | 669 --- drivers/video/fbdev/bfin_adv7393fb.c | 828 ---- drivers/video/fbdev/bfin_adv7393fb.h | 319 -- drivers/video/fbdev/s1d13xxxfb.c | 10 - drivers/video/logo/Kconfig | 15 - drivers/video/logo/Makefile | 3 - drivers/video/logo/logo.c | 12 - drivers/video/logo/logo_blackfin_clut224.ppm | 1127 ------ drivers/video/logo/logo_blackfin_vga16.ppm | 1127 ------ drivers/video/logo/logo_m32r_clut224.ppm | 1292 ------ drivers/watchdog/Kconfig | 17 - drivers/watchdog/Makefile | 7 - drivers/watchdog/bfin_wdt.c | 476 --- include/linux/bfin_mac.h | 30 - include/linux/linux_logo.h | 3 - include/linux/platform_data/bfin_rotary.h | 117 - include/linux/platform_data/pinctrl-adi2.h | 40 - include/linux/raid/pq.h | 1 - include/linux/usb/musb.h | 7 - include/linux/usb/tilegx.h | 35 - include/media/blackfin/bfin_capture.h | 39 - include/media/blackfin/ppi.h | 94 - lib/raid6/Makefile | 6 - lib/raid6/algos.c | 3 - lib/raid6/test/Makefile | 7 - lib/raid6/tilegx.uc | 87 - sound/soc/Kconfig | 1 - sound/soc/Makefile | 1 - sound/soc/blackfin/Kconfig | 205 - sound/soc/blackfin/Makefile | 40 - sound/soc/blackfin/bf5xx-ac97-pcm.c | 480 --- sound/soc/blackfin/bf5xx-ac97.c | 388 -- sound/soc/blackfin/bf5xx-ac97.h | 57 - sound/soc/blackfin/bf5xx-ad1836.c | 109 - sound/soc/blackfin/bf5xx-ad193x.c | 131 - sound/soc/blackfin/bf5xx-ad1980.c | 109 - sound/soc/blackfin/bf5xx-ad73311.c | 212 - sound/soc/blackfin/bf5xx-i2s-pcm.c | 373 -- sound/soc/blackfin/bf5xx-i2s-pcm.h | 17 - sound/soc/blackfin/bf5xx-i2s.c | 391 -- sound/soc/blackfin/bf5xx-sport.c | 1102 ----- sound/soc/blackfin/bf5xx-sport.h | 174 - sound/soc/blackfin/bf5xx-ssm2602.c | 126 - sound/soc/blackfin/bf6xx-i2s.c | 239 -- sound/soc/blackfin/bf6xx-sport.c | 425 -- sound/soc/blackfin/bf6xx-sport.h | 82 - sound/soc/blackfin/bfin-eval-adau1373.c | 173 - sound/soc/blackfin/bfin-eval-adau1701.c | 113 - sound/soc/blackfin/bfin-eval-adau1x61.c | 142 - sound/soc/blackfin/bfin-eval-adau1x81.c | 129 - sound/soc/blackfin/bfin-eval-adav80x.c | 145 - 198 files changed, 7 insertions(+), 56230 deletions(-) delete mode 100644 Documentation/devicetree/bindings/gpio/gpio-etraxfs.txt delete mode 100644 Documentation/devicetree/bindings/serial/axis,etraxfs-uart.txt delete mode 100644 drivers/ata/pata_bf54x.c delete mode 100644 drivers/char/bfin-otp.c delete mode 100644 drivers/char/ds1302.c delete mode 100644 drivers/char/tile-srom.c delete mode 100644 drivers/cpufreq/blackfin-cpufreq.c delete mode 100644 drivers/cpufreq/cris-artpec3-cpufreq.c delete mode 100644 drivers/cpufreq/cris-etraxfs-cpufreq.c delete mode 100644 drivers/crypto/bfin_crc.c delete mode 100644 drivers/crypto/bfin_crc.h delete mode 100644 drivers/edac/tile_edac.c delete mode 100644 drivers/gpio/gpio-etraxfs.c delete mode 100644 drivers/i2c/busses/i2c-bfin-twi.c delete mode 100644 drivers/input/keyboard/bf54x-keys.c delete mode 100644 drivers/input/misc/bfin_rotary.c delete mode 100644 drivers/media/platform/arv.c delete mode 100644 drivers/media/platform/blackfin/Kconfig delete mode 100644 drivers/media/platform/blackfin/Makefile delete mode 100644 drivers/media/platform/blackfin/bfin_capture.c delete mode 100644 drivers/media/platform/blackfin/ppi.c delete mode 100644 drivers/mmc/host/bfin_sdh.c delete mode 100644 drivers/mtd/maps/bfin-async-flash.c delete mode 100644 drivers/mtd/nand/raw/bf5xx_nand.c delete mode 100644 drivers/net/can/bfin_can.c delete mode 100644 drivers/net/cris/Makefile delete mode 100644 drivers/net/cris/eth_v10.c delete mode 100644 drivers/net/ethernet/adi/Kconfig delete mode 100644 drivers/net/ethernet/adi/Makefile delete mode 100644 drivers/net/ethernet/adi/bfin_mac.c delete mode 100644 drivers/net/ethernet/adi/bfin_mac.h delete mode 100644 drivers/net/ethernet/tile/Kconfig delete mode 100644 drivers/net/ethernet/tile/Makefile delete mode 100644 drivers/net/ethernet/tile/tilegx.c delete mode 100644 drivers/net/ethernet/tile/tilepro.c delete mode 100644 drivers/pcmcia/bfin_cf_pcmcia.c delete mode 100644 drivers/pcmcia/m32r_cfc.c delete mode 100644 drivers/pcmcia/m32r_cfc.h delete mode 100644 drivers/pcmcia/m32r_pcc.c delete mode 100644 drivers/pcmcia/m32r_pcc.h delete mode 100644 drivers/pinctrl/pinctrl-adi2-bf54x.c delete mode 100644 drivers/pinctrl/pinctrl-adi2-bf60x.c delete mode 100644 drivers/pinctrl/pinctrl-adi2.c delete mode 100644 drivers/pinctrl/pinctrl-adi2.h delete mode 100644 drivers/pwm/pwm-bfin.c delete mode 100644 drivers/rtc/rtc-bfin.c delete mode 100644 drivers/rtc/rtc-tile.c delete mode 100644 drivers/spi/spi-adi-v3.c delete mode 100644 drivers/spi/spi-bfin-sport.c delete mode 100644 drivers/spi/spi-bfin5xx.c delete mode 100644 drivers/staging/iio/trigger/Kconfig delete mode 100644 drivers/staging/iio/trigger/Makefile delete mode 100644 drivers/staging/iio/trigger/iio-trig-bfin-timer.c delete mode 100644 drivers/staging/iio/trigger/iio-trig-bfin-timer.h delete mode 100644 drivers/staging/irda/drivers/bfin_sir.c delete mode 100644 drivers/staging/irda/drivers/bfin_sir.h delete mode 100644 drivers/tty/bfin_jtag_comm.c delete mode 100644 drivers/tty/hvc/hvc_bfin_jtag.c delete mode 100644 drivers/tty/hvc/hvc_tile.c delete mode 100644 drivers/tty/serial/bfin_sport_uart.c delete mode 100644 drivers/tty/serial/bfin_sport_uart.h delete mode 100644 drivers/tty/serial/bfin_uart.c delete mode 100644 drivers/tty/serial/crisv10.c delete mode 100644 drivers/tty/serial/crisv10.h delete mode 100644 drivers/tty/serial/etraxfs-uart.c delete mode 100644 drivers/tty/serial/m32r_sio.c delete mode 100644 drivers/tty/serial/m32r_sio_reg.h delete mode 100644 drivers/tty/serial/tilegx.c delete mode 100644 drivers/usb/host/ehci-tilegx.c delete mode 100644 drivers/usb/host/ohci-tilegx.c delete mode 100644 drivers/usb/musb/blackfin.c delete mode 100644 drivers/usb/musb/blackfin.h delete mode 100644 drivers/video/fbdev/bf537-lq035.c delete mode 100644 drivers/video/fbdev/bf54x-lq043fb.c delete mode 100644 drivers/video/fbdev/bfin-lq035q1-fb.c delete mode 100644 drivers/video/fbdev/bfin-t350mcqb-fb.c delete mode 100644 drivers/video/fbdev/bfin_adv7393fb.c delete mode 100644 drivers/video/fbdev/bfin_adv7393fb.h delete mode 100644 drivers/video/logo/logo_blackfin_clut224.ppm delete mode 100644 drivers/video/logo/logo_blackfin_vga16.ppm delete mode 100644 drivers/video/logo/logo_m32r_clut224.ppm delete mode 100644 drivers/watchdog/bfin_wdt.c delete mode 100644 include/linux/bfin_mac.h delete mode 100644 include/linux/platform_data/bfin_rotary.h delete mode 100644 include/linux/platform_data/pinctrl-adi2.h delete mode 100644 include/linux/usb/tilegx.h delete mode 100644 include/media/blackfin/bfin_capture.h delete mode 100644 include/media/blackfin/ppi.h delete mode 100644 lib/raid6/tilegx.uc delete mode 100644 sound/soc/blackfin/Kconfig delete mode 100644 sound/soc/blackfin/Makefile delete mode 100644 sound/soc/blackfin/bf5xx-ac97-pcm.c delete mode 100644 sound/soc/blackfin/bf5xx-ac97.c delete mode 100644 sound/soc/blackfin/bf5xx-ac97.h delete mode 100644 sound/soc/blackfin/bf5xx-ad1836.c delete mode 100644 sound/soc/blackfin/bf5xx-ad193x.c delete mode 100644 sound/soc/blackfin/bf5xx-ad1980.c delete mode 100644 sound/soc/blackfin/bf5xx-ad73311.c delete mode 100644 sound/soc/blackfin/bf5xx-i2s-pcm.c delete mode 100644 sound/soc/blackfin/bf5xx-i2s-pcm.h delete mode 100644 sound/soc/blackfin/bf5xx-i2s.c delete mode 100644 sound/soc/blackfin/bf5xx-sport.c delete mode 100644 sound/soc/blackfin/bf5xx-sport.h delete mode 100644 sound/soc/blackfin/bf5xx-ssm2602.c delete mode 100644 sound/soc/blackfin/bf6xx-i2s.c delete mode 100644 sound/soc/blackfin/bf6xx-sport.c delete mode 100644 sound/soc/blackfin/bf6xx-sport.h delete mode 100644 sound/soc/blackfin/bfin-eval-adau1373.c delete mode 100644 sound/soc/blackfin/bfin-eval-adau1701.c delete mode 100644 sound/soc/blackfin/bfin-eval-adau1x61.c delete mode 100644 sound/soc/blackfin/bfin-eval-adau1x81.c delete mode 100644 sound/soc/blackfin/bfin-eval-adav80x.c -- Cc: linux(a)roeck-us.net Cc: corbet(a)lwn.net Cc: tj(a)kernel.org Cc: gregkh(a)linuxfoundation.org Cc: rjw(a)rjwysocki.net Cc: viresh.kumar(a)linaro.org Cc: herbert(a)gondor.apana.org.au Cc: davem(a)davemloft.net Cc: bp(a)alien8.de Cc: mchehab(a)kernel.org Cc: linus.walleij(a)linaro.org Cc: wsa(a)the-dreams.de Cc: dmitry.torokhov(a)gmail.com Cc: ulf.hansson(a)linaro.org Cc: boris.brezillon(a)bootlin.com Cc: cyrille.pitchen(a)wedev4u.fr Cc: wg(a)grandegger.com Cc: mkl(a)pengutronix.de Cc: alexandre.belloni(a)bootlin.com Cc: broonie(a)kernel.org Cc: jic23(a)kernel.org Cc: lars(a)metafoo.de Cc: jslaby(a)suse.com Cc: stern(a)rowland.harvard.edu Cc: b.zolnierkie(a)samsung.com Cc: shli(a)kernel.org Cc: linux-watchdog(a)vger.kernel.org Cc: linux-doc(a)vger.kernel.org Cc: linux-ide(a)vger.kernel.org Cc: linux-pm(a)vger.kernel.org Cc: linux-crypto(a)vger.kernel.org Cc: linux-edac(a)vger.kernel.org Cc: linux-gpio(a)vger.kernel.org Cc: linux-i2c(a)vger.kernel.org Cc: linux-input(a)vger.kernel.org Cc: linux-media(a)vger.kernel.org Cc: linux-mmc(a)vger.kernel.org Cc: linux-mtd(a)lists.infradead.org Cc: netdev(a)vger.kernel.org Cc: linux-can(a)vger.kernel.org Cc: linux-pwm(a)vger.kernel.org Cc: linux-rtc(a)vger.kernel.org Cc: linux-spi(a)vger.kernel.org Cc: linux-iio(a)vger.kernel.org Cc: linux-serial(a)vger.kernel.org Cc: linux-usb(a)vger.kernel.org Cc: linux-fbdev(a)vger.kernel.org Cc: linux-raid(a)vger.kernel.org Cc: alsa-devel(a)alsa-project.org

4 7

[alsa-devel] [PATCH] ASoC: topology: Give more data to clients via callbacks
by Liam Girdwood 14 Mar '18

14 Mar '18

Give topology clients more access to the topology data by passing index, pcm, link_config and dai_driver to clients. This allows clients to fully instantiate and track topology objects. The SOF driver is the first user of these new APIs and needs them to build component topology driver and FW objects. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/sound/soc-topology.h | 23 ++++++++++++++--------- sound/soc/intel/skylake/skl-pcm.c | 7 ++++--- sound/soc/intel/skylake/skl-topology.c | 5 +++-- sound/soc/intel/skylake/skl-topology.h | 7 +++---- sound/soc/soc-topology.c | 31 ++++++++++++++++++------------- 5 files changed, 42 insertions(+), 31 deletions(-) diff --git a/include/sound/soc-topology.h b/include/sound/soc-topology.h index f552c3f56368..e1f265e21ee1 100644 --- a/include/sound/soc-topology.h +++ b/include/sound/soc-topology.h @@ -30,6 +30,8 @@ struct snd_soc_dapm_context; struct snd_soc_card; struct snd_kcontrol_new; struct snd_soc_dai_link; +struct snd_soc_dai_driver; +struct snd_soc_dai; /* object scan be loaded and unloaded in groups with identfying indexes */ #define SND_SOC_TPLG_INDEX_ALL 0 /* ID that matches all FW objects */ @@ -109,35 +111,38 @@ struct snd_soc_tplg_widget_events { struct snd_soc_tplg_ops { /* external kcontrol init - used for any driver specific init */ - int (*control_load)(struct snd_soc_component *, + int (*control_load)(struct snd_soc_component *, int index, struct snd_kcontrol_new *, struct snd_soc_tplg_ctl_hdr *); int (*control_unload)(struct snd_soc_component *, struct snd_soc_dobj *); /* external widget init - used for any driver specific init */ - int (*widget_load)(struct snd_soc_component *, + int (*widget_load)(struct snd_soc_component *, int index, struct snd_soc_dapm_widget *, struct snd_soc_tplg_dapm_widget *); - int (*widget_ready)(struct snd_soc_component *, + int (*widget_ready)(struct snd_soc_component *, int index, struct snd_soc_dapm_widget *, struct snd_soc_tplg_dapm_widget *); int (*widget_unload)(struct snd_soc_component *, struct snd_soc_dobj *); /* FE DAI - used for any driver specific init */ - int (*dai_load)(struct snd_soc_component *, - struct snd_soc_dai_driver *dai_drv); + int (*dai_load)(struct snd_soc_component *, int index, + struct snd_soc_dai_driver *dai_drv, + struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai); + int (*dai_unload)(struct snd_soc_component *, struct snd_soc_dobj *); /* DAI link - used for any driver specific init */ - int (*link_load)(struct snd_soc_component *, - struct snd_soc_dai_link *link); + int (*link_load)(struct snd_soc_component *, int index, + struct snd_soc_dai_link *link, + struct snd_soc_tplg_link_config *cfg); int (*link_unload)(struct snd_soc_component *, struct snd_soc_dobj *); /* callback to handle vendor bespoke data */ - int (*vendor_load)(struct snd_soc_component *, + int (*vendor_load)(struct snd_soc_component *, int index, struct snd_soc_tplg_hdr *); int (*vendor_unload)(struct snd_soc_component *, struct snd_soc_tplg_hdr *); @@ -146,7 +151,7 @@ struct snd_soc_tplg_ops { void (*complete)(struct snd_soc_component *); /* manifest - optional to inform component of manifest */ - int (*manifest)(struct snd_soc_component *, + int (*manifest)(struct snd_soc_component *, int index, struct snd_soc_tplg_manifest *); /* vendor specific kcontrol handlers available for binding */ diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 15cb8ac3e374..c466265a2fbb 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -989,10 +989,11 @@ static struct snd_soc_dai_driver skl_platform_dai[] = { }, }; -int skl_dai_load(struct snd_soc_component *cmp, - struct snd_soc_dai_driver *pcm_dai) +int skl_dai_load(struct snd_soc_component *cmp, int index, + struct snd_soc_dai_driver *dai_drv, + struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai) { - pcm_dai->ops = &skl_pcm_dai_ops; + dai_drv->ops = &skl_pcm_dai_ops; return 0; } diff --git a/sound/soc/intel/skylake/skl-topology.c b/sound/soc/intel/skylake/skl-topology.c index 3b1dca419883..6ac081f1f215 100644 --- a/sound/soc/intel/skylake/skl-topology.c +++ b/sound/soc/intel/skylake/skl-topology.c @@ -2851,7 +2851,7 @@ void skl_cleanup_resources(struct skl *skl) * information to the driver about module and pipeline parameters which DSP * FW expects like ids, resource values, formats etc */ -static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, +static int skl_tplg_widget_load(struct snd_soc_component *cmpnt, int index, struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w) { @@ -2958,6 +2958,7 @@ static int skl_init_enum_data(struct device *dev, struct soc_enum *se, } static int skl_tplg_control_load(struct snd_soc_component *cmpnt, + int index, struct snd_kcontrol_new *kctl, struct snd_soc_tplg_ctl_hdr *hdr) { @@ -3446,7 +3447,7 @@ static int skl_tplg_get_manifest_data(struct snd_soc_tplg_manifest *manifest, return 0; } -static int skl_manifest_load(struct snd_soc_component *cmpnt, +static int skl_manifest_load(struct snd_soc_component *cmpnt, int index, struct snd_soc_tplg_manifest *manifest) { struct hdac_ext_bus *ebus = snd_soc_component_get_drvdata(cmpnt); diff --git a/sound/soc/intel/skylake/skl-topology.h b/sound/soc/intel/skylake/skl-topology.h index b1e0667c0ae0..da301d899431 100644 --- a/sound/soc/intel/skylake/skl-topology.h +++ b/sound/soc/intel/skylake/skl-topology.h @@ -512,8 +512,7 @@ int skl_pcm_host_dma_prepare(struct device *dev, int skl_pcm_link_dma_prepare(struct device *dev, struct skl_pipe_params *params); -int skl_dai_load(struct snd_soc_component *cmp, - struct snd_soc_dai_driver *pcm_dai); -void skl_tplg_add_moduleid_in_bind_params(struct skl *skl, - struct snd_soc_dapm_widget *w); +int skl_dai_load(struct snd_soc_component *cmp, int index, + struct snd_soc_dai_driver *dai_drv, + struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai); #endif diff --git a/sound/soc/soc-topology.c b/sound/soc/soc-topology.c index e81b52600d58..9cf98259bb58 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -315,7 +315,7 @@ static int soc_tplg_vendor_load_(struct soc_tplg *tplg, int ret = 0; if (tplg->comp && tplg->ops && tplg->ops->vendor_load) - ret = tplg->ops->vendor_load(tplg->comp, hdr); + ret = tplg->ops->vendor_load(tplg->comp, tplg->index, hdr); else { dev_err(tplg->dev, "ASoC: no vendor load callback for ID %d\n", hdr->vendor_type); @@ -347,7 +347,8 @@ static int soc_tplg_widget_load(struct soc_tplg *tplg, struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w) { if (tplg->comp && tplg->ops && tplg->ops->widget_load) - return tplg->ops->widget_load(tplg->comp, w, tplg_w); + return tplg->ops->widget_load(tplg->comp, tplg->index, w, + tplg_w); return 0; } @@ -358,27 +359,30 @@ static int soc_tplg_widget_ready(struct soc_tplg *tplg, struct snd_soc_dapm_widget *w, struct snd_soc_tplg_dapm_widget *tplg_w) { if (tplg->comp && tplg->ops && tplg->ops->widget_ready) - return tplg->ops->widget_ready(tplg->comp, w, tplg_w); + return tplg->ops->widget_ready(tplg->comp, tplg->index, w, + tplg_w); return 0; } /* pass DAI configurations to component driver for extra initialization */ static int soc_tplg_dai_load(struct soc_tplg *tplg, - struct snd_soc_dai_driver *dai_drv) + struct snd_soc_dai_driver *dai_drv, + struct snd_soc_tplg_pcm *pcm, struct snd_soc_dai *dai) { if (tplg->comp && tplg->ops && tplg->ops->dai_load) - return tplg->ops->dai_load(tplg->comp, dai_drv); + return tplg->ops->dai_load(tplg->comp, tplg->index, dai_drv, + pcm, dai); return 0; } /* pass link configurations to component driver for extra initialization */ static int soc_tplg_dai_link_load(struct soc_tplg *tplg, - struct snd_soc_dai_link *link) + struct snd_soc_dai_link *link, struct snd_soc_tplg_link_config *cfg) { if (tplg->comp && tplg->ops && tplg->ops->link_load) - return tplg->ops->link_load(tplg->comp, link); + return tplg->ops->link_load(tplg->comp, tplg->index, link, cfg); return 0; } @@ -692,7 +696,8 @@ static int soc_tplg_init_kcontrol(struct soc_tplg *tplg, struct snd_kcontrol_new *k, struct snd_soc_tplg_ctl_hdr *hdr) { if (tplg->comp && tplg->ops && tplg->ops->control_load) - return tplg->ops->control_load(tplg->comp, k, hdr); + return tplg->ops->control_load(tplg->comp, tplg->index, k, + hdr); return 0; } @@ -1733,7 +1738,7 @@ static int soc_tplg_dai_create(struct soc_tplg *tplg, } /* pass control to component driver for optional further init */ - ret = soc_tplg_dai_load(tplg, dai_drv); + ret = soc_tplg_dai_load(tplg, dai_drv, pcm, NULL); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n"); kfree(dai_drv); @@ -1803,7 +1808,7 @@ static int soc_tplg_fe_link_create(struct soc_tplg *tplg, set_link_flags(link, pcm->flag_mask, pcm->flags); /* pass control to component driver for optional further init */ - ret = soc_tplg_dai_link_load(tplg, link); + ret = soc_tplg_dai_link_load(tplg, link, NULL); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: FE link loading failed\n"); kfree(link); @@ -2094,7 +2099,7 @@ static int soc_tplg_link_config(struct soc_tplg *tplg, set_link_flags(link, cfg->flag_mask, cfg->flags); /* pass control to component driver for optional further init */ - ret = soc_tplg_dai_link_load(tplg, link); + ret = soc_tplg_dai_link_load(tplg, link, cfg); if (ret < 0) { dev_err(tplg->dev, "ASoC: physical link loading failed\n"); return ret; @@ -2216,7 +2221,7 @@ static int soc_tplg_dai_config(struct soc_tplg *tplg, set_dai_flags(dai_drv, d->flag_mask, d->flags); /* pass control to component driver for optional further init */ - ret = soc_tplg_dai_load(tplg, dai_drv); + ret = soc_tplg_dai_load(tplg, dai_drv, NULL, dai); if (ret < 0) { dev_err(tplg->comp->dev, "ASoC: DAI loading failed\n"); return ret; @@ -2322,7 +2327,7 @@ static int soc_tplg_manifest_load(struct soc_tplg *tplg, /* pass control to component driver for optional further init */ if (tplg->comp && tplg->ops && tplg->ops->manifest) - return tplg->ops->manifest(tplg->comp, _manifest); + return tplg->ops->manifest(tplg->comp, tplg->index, _manifest); if (!abi_match) /* free the duplicated one */ kfree(_manifest); -- 2.14.1

1 0

[alsa-devel] [PATCH] ASoC: core: Allow topology to override machine driver FE DAI link config.
by Liam Girdwood 14 Mar '18

14 Mar '18

Machine drivers statically define a number of DAI links that currently cannot be changed or removed by topology. This means PCMs and platform components cannot be changed by topology at runtime AND machine drivers are tightly coupled to topology. This patch allows topology to override the machine driver DAI link config in order to reuse machine drivers with different topologies and platform components. The patch supports :- 1) create new FE PCMs with a topology defined PCM ID. 2) destroy existing static FE PCMs 3) change the platform component driver. 4) assign any new HW params fixups. The patch requires no changes to the machine drivers, but does add some platform component flags that the platform component driver can assign before loading topologies. The initial user for this patch will be SOF. Signed-off-by: Liam Girdwood <liam.r.girdwood(a)linux.intel.com> --- include/sound/soc.h | 10 +++++++ sound/soc/soc-core.c | 80 ++++++++++++++++++++++++++++++++++++++++++++++++++-- sound/soc/soc-pcm.c | 12 ++++++++ 3 files changed, 99 insertions(+), 3 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index ad266d7e9553..4e368c308f93 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1012,6 +1012,13 @@ struct snd_soc_platform_driver { /* platform stream compress ops */ const struct snd_compr_ops *compr_ops; + + /* this platform uses topology and ignore machine driver FEs */ + const char *ignore_machine; + int (*be_hw_params_fixup)(struct snd_soc_pcm_runtime *rtd, + struct snd_pcm_hw_params *params); + bool use_dai_pcm_id; /* use the DAI link PCM ID as PCM device number */ + int be_pcm_base; /* base device ID for all BE PCMs */ }; struct snd_soc_dai_link_component { @@ -1118,6 +1125,9 @@ struct snd_soc_dai_link { /* pmdown_time is ignored at stop */ unsigned int ignore_pmdown_time:1; + /* Do not create a PCM for this DAI link */ + unsigned int ignore:1; + struct list_head list; /* DAI link list of the soc card */ struct snd_soc_dobj dobj; /* For topology */ }; diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 0bd8613c52ab..33840e5bc24e 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -1050,6 +1050,9 @@ static int soc_bind_dai_link(struct snd_soc_card *card, const char *platform_name; int i; + if (dai_link->ignore) + return 0; + dev_dbg(card->dev, "ASoC: binding %s\n", dai_link->name); if (soc_is_dai_link_bound(card, dai_link)) { @@ -1672,7 +1675,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card, { struct snd_soc_dai_link *dai_link = rtd->dai_link; struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - int i, ret; + int i, ret, num; dev_dbg(card->dev, "ASoC: probe %s dai link %d late %d\n", card->name, rtd->num, order); @@ -1718,9 +1721,23 @@ static int soc_probe_link_dais(struct snd_soc_card *card, soc_dpcm_debugfs_add(rtd); #endif + /* + * most drivers will register their PCMs using DAI link ordering but + * topology based drivers can use the DAI link id field to set PCM + * device number and then use rtd + a base offset of the BEs. + */ + if (rtd->platform->driver->use_dai_pcm_id) { + if (rtd->dai_link->no_pcm) + num = rtd->platform->driver->be_pcm_base + rtd->num; + else + num = rtd->dai_link->id; + } else { + num = rtd->num; + } + if (cpu_dai->driver->compress_new) { /*create compress_device"*/ - ret = cpu_dai->driver->compress_new(rtd, rtd->num); + ret = cpu_dai->driver->compress_new(rtd, num); if (ret < 0) { dev_err(card->dev, "ASoC: can't create compress %s\n", dai_link->stream_name); @@ -1730,7 +1747,7 @@ static int soc_probe_link_dais(struct snd_soc_card *card, if (!dai_link->params) { /* create the pcm */ - ret = soc_new_pcm(rtd, rtd->num); + ret = soc_new_pcm(rtd, num); if (ret < 0) { dev_err(card->dev, "ASoC: can't create pcm %s :%d\n", dai_link->stream_name, ret); @@ -2076,6 +2093,60 @@ int snd_soc_set_dmi_name(struct snd_soc_card *card, const char *flavour) EXPORT_SYMBOL_GPL(snd_soc_set_dmi_name); #endif /* CONFIG_DMI */ +/* check if we need to ignore or override the FEs for this card */ +static void soc_check_tplg_fes(struct snd_soc_card *card) +{ + struct snd_soc_platform *platform; + struct snd_soc_dai_link *dai_link; + int i; + + list_for_each_entry(platform, &platform_list, list) { + + /* does this platform override FEs ? */ + if (!platform->driver->ignore_machine) + continue; + + /* for this machine ? */ + if (strcmp(platform->driver->ignore_machine, + card->dev->driver->name)) + continue; + + /* machine matches, so override the rtd data */ + for (i = 0; i < card->num_links; i++) { + + dai_link = &card->dai_link[i]; + + /* ignore this FE */ + if (dai_link->dynamic) { + dai_link->ignore = true; + continue; + } + + dev_info(card->dev, "info: override FE DAI link %s\n", + card->dai_link[i].name); + + /* override platform */ + dai_link->platform_name = platform->component.name; + dai_link->cpu_dai_name = platform->component.name; + + /* convert non BE into BE */ + dai_link->no_pcm = 1; + dai_link->dpcm_playback = 1; + dai_link->dpcm_capture = 1; + + /* override any BE fixups */ + dai_link->be_hw_params_fixup = + platform->driver->be_hw_params_fixup; + + /* most BE links dont set stream name, so set it to + * dai link name if it's NULL to help bind widgets. + */ + if (!dai_link->stream_name) + dai_link->stream_name = dai_link->name; + } + } +} + static int snd_soc_instantiate_card(struct snd_soc_card *card) { struct snd_soc_codec *codec; @@ -2086,6 +2157,9 @@ static int snd_soc_instantiate_card(struct snd_soc_card *card) mutex_lock(&client_mutex); mutex_lock_nested(&card->mutex, SND_SOC_CARD_CLASS_INIT); + /* check whether any platform is ignore machine FE and using topology */ + soc_check_tplg_fes(card); + /* bind DAIs */ for (i = 0; i < card->num_links; i++) { ret = soc_bind_dai_link(card, &card->dai_link[i]); diff --git a/sound/soc/soc-pcm.c b/sound/soc/soc-pcm.c index 084125463d10..ea36de977c93 100644 --- a/sound/soc/soc-pcm.c +++ b/sound/soc/soc-pcm.c @@ -909,8 +909,20 @@ int soc_dai_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { + struct snd_soc_pcm_runtime *rtd = substream->private_data; int ret; + /* perform any topology hw_params fixups before DAI */ + if (rtd->dai_link->be_hw_params_fixup) { + ret = rtd->dai_link->be_hw_params_fixup(rtd, params); + if (ret < 0) { + dev_err(rtd->dev, + "ASoC: hw_params topology fixup failed %d\n", + ret); + return ret; + } + } + if (dai->driver->ops->hw_params) { ret = dai->driver->ops->hw_params(substream, params, dai); if (ret < 0) { -- 2.14.1

1 0

[alsa-devel] [PATCH][next] ASoC: samsung: remove redundant pointer cpu_dai
by Colin King 14 Mar '18

14 Mar '18

From: Colin Ian King <colin.king(a)canonical.com> The pointer cpu_dai is assigned a value but the pointer is never read, hence it is redundant and can be removed. Cleans up clang warning: sound/soc/samsung/snow.c:45:2: warning: Value stored to 'cpu_dai' is never read Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/samsung/snow.c | 2 -- 1 file changed, 2 deletions(-) diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c index fc62110f500f..47b904e2b8c3 100644 --- a/sound/soc/samsung/snow.c +++ b/sound/soc/samsung/snow.c @@ -37,12 +37,10 @@ static int snow_late_probe(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd; struct snd_soc_dai *codec_dai; - struct snd_soc_dai *cpu_dai; int ret; rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); codec_dai = rtd->codec_dai; - cpu_dai = rtd->cpu_dai; /* Set the MCLK rate for the codec */ ret = snd_soc_dai_set_sysclk(codec_dai, 0, -- 2.15.1

4 4

[alsa-devel] [BUG] snd_pcm_drop() does not stop the PCM immediately
by Martin Schreiber 14 Mar '18

14 Mar '18

Hi, The documentation states: " This function stops the PCM immediately. The pending samples on the buffer are ignored. " This does not work for the device "sysdefault" where a piece of the old sound and the new sound overlap, see attached alsadrop1.c testprogram. It is OK with the "default" device which is wired over Pulseaudio, uncomment alsadrop1.c:55 in order to test. The project is here: https://gitlab.com/mseide-msegui/mseuniverse/tree/master/testcase/audio/c/a… Environment: openSUSE Leap 42.3, see attached alsa-info.tar.gz. Martin

2 6

[alsa-devel] [PATCH] ASoC: uniphier: evea: add switch for changing source of line-in
by Katsuhiro Suzuki 14 Mar '18

14 Mar '18

This patch adds mixer switch for changing audio source of line-in. We can choose one of LIN1, 2, 3, default is LIN1. Signed-off-by: Katsuhiro Suzuki <suzuki.katsuhiro(a)socionext.com> --- sound/soc/uniphier/evea.c | 11 +++++++++++ 1 file changed, 11 insertions(+) diff --git a/sound/soc/uniphier/evea.c b/sound/soc/uniphier/evea.c index 439f14f91b23..73fd6730095c 100644 --- a/sound/soc/uniphier/evea.c +++ b/sound/soc/uniphier/evea.c @@ -18,6 +18,8 @@ #define AADCPOW(n) (0x0078 + 0x04 * (n)) #define AADCPOW_AADC_POWD BIT(0) +#define ALINSW1 0x0088 +#define ALINSW1_SEL1_SHIFT 3 #define AHPOUTPOW 0x0098 #define AHPOUTPOW_HP_ON BIT(4) #define ALINEPOW 0x009c @@ -278,7 +280,16 @@ static int evea_set_switch_hp(struct snd_kcontrol *kcontrol, return evea_update_switch_hp(evea); } +static const char * const linsw1_sel1_text[] = { + "LIN1", "LIN2", "LIN3" +}; + +static SOC_ENUM_SINGLE_DECL(linsw1_sel1_enum, + ALINSW1, ALINSW1_SEL1_SHIFT, + linsw1_sel1_text); + static const struct snd_kcontrol_new evea_controls[] = { + SOC_ENUM("Line Capture Source", linsw1_sel1_enum), SOC_SINGLE_BOOL_EXT("Line Capture Switch", 0, evea_get_switch_lin, evea_set_switch_lin), SOC_SINGLE_BOOL_EXT("Line Playback Switch", 0, -- 2.16.1

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[alsa-devel] [asoc:for-next 462/535] sound/soc/codecs/da7219.c:1777:22: sparse: symbol 'da7219_dai_clks_ops' was not declared. Should it be static?
by kbuild test robot 14 Mar '18

14 Mar '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git for-next head: 27f6213507962a2c754d7dff7c25515e09e31752 commit: fc8f7ea2d6c074baaad202c9187962bfa493ef13 [462/535] ASoC: da7219: Add common clock usage for providing DAI clks reproduce: # apt-get install sparse git checkout fc8f7ea2d6c074baaad202c9187962bfa493ef13 make ARCH=x86_64 allmodconfig make C=1 CF=-D__CHECK_ENDIAN__ sparse warnings: (new ones prefixed by >>) sound/soc/codecs/da7219.c:440:44: sparse: cast to restricted __le16 sound/soc/codecs/da7219.c:461:13: sparse: incorrect type in assignment (different base types) @@ expected unsigned short [unsigned] [usertype] val @@ got short [unsigned] [usertype] val @@ sound/soc/codecs/da7219.c:461:13: expected unsigned short [unsigned] [usertype] val sound/soc/codecs/da7219.c:461:13: got restricted __le16 [usertype] <noident> sound/soc/codecs/da7219.c:817:57: sparse: dubious: x & !y sound/soc/codecs/da7219.c:1427:16: sparse: incorrect type in assignment (different base types) @@ expected unsigned short [unsigned] [usertype] offset @@ got short [unsigned] [usertype] offset @@ sound/soc/codecs/da7219.c:1427:16: expected unsigned short [unsigned] [usertype] offset sound/soc/codecs/da7219.c:1427:16: got restricted __le16 [usertype] <noident> >> sound/soc/codecs/da7219.c:1777:22: sparse: symbol 'da7219_dai_clks_ops' was not declared. Should it be static? Please review and possibly fold the followup patch. --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

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[alsa-devel] [PATCH v3] ASoC: remove blackfin drivers
by Arnd Bergmann 14 Mar '18

14 Mar '18

The blackfin architecture is getting removed, so the ASoC drivers are all obsolete as well. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- v2: full patch v3: rebased against asoc #for-next MAINTAINERS | 7 - sound/soc/Kconfig | 1 - sound/soc/Makefile | 1 - sound/soc/blackfin/Kconfig | 205 ------ sound/soc/blackfin/Makefile | 40 -- sound/soc/blackfin/bf5xx-ac97-pcm.c | 480 -------------- sound/soc/blackfin/bf5xx-ac97.c | 388 ----------- sound/soc/blackfin/bf5xx-ac97.h | 57 -- sound/soc/blackfin/bf5xx-ad1836.c | 109 --- sound/soc/blackfin/bf5xx-ad193x.c | 131 ---- sound/soc/blackfin/bf5xx-ad1980.c | 109 --- sound/soc/blackfin/bf5xx-ad73311.c | 212 ------ sound/soc/blackfin/bf5xx-i2s-pcm.c | 373 ----------- sound/soc/blackfin/bf5xx-i2s-pcm.h | 17 - sound/soc/blackfin/bf5xx-i2s.c | 391 ----------- sound/soc/blackfin/bf5xx-sport.c | 1102 ------------------------------- sound/soc/blackfin/bf5xx-sport.h | 174 ----- sound/soc/blackfin/bf5xx-ssm2602.c | 126 ---- sound/soc/blackfin/bf6xx-i2s.c | 239 ------- sound/soc/blackfin/bf6xx-sport.c | 425 ------------ sound/soc/blackfin/bf6xx-sport.h | 82 --- sound/soc/blackfin/bfin-eval-adau1373.c | 173 ----- sound/soc/blackfin/bfin-eval-adau1701.c | 113 ---- sound/soc/blackfin/bfin-eval-adau1x61.c | 142 ---- sound/soc/blackfin/bfin-eval-adau1x81.c | 129 ---- sound/soc/blackfin/bfin-eval-adav80x.c | 145 ---- 26 files changed, 5371 deletions(-) delete mode 100644 sound/soc/blackfin/Kconfig delete mode 100644 sound/soc/blackfin/Makefile delete mode 100644 sound/soc/blackfin/bf5xx-ac97-pcm.c delete mode 100644 sound/soc/blackfin/bf5xx-ac97.c delete mode 100644 sound/soc/blackfin/bf5xx-ac97.h delete mode 100644 sound/soc/blackfin/bf5xx-ad1836.c delete mode 100644 sound/soc/blackfin/bf5xx-ad193x.c delete mode 100644 sound/soc/blackfin/bf5xx-ad1980.c delete mode 100644 sound/soc/blackfin/bf5xx-ad73311.c delete mode 100644 sound/soc/blackfin/bf5xx-i2s-pcm.c delete mode 100644 sound/soc/blackfin/bf5xx-i2s-pcm.h delete mode 100644 sound/soc/blackfin/bf5xx-i2s.c delete mode 100644 sound/soc/blackfin/bf5xx-sport.c delete mode 100644 sound/soc/blackfin/bf5xx-sport.h delete mode 100644 sound/soc/blackfin/bf5xx-ssm2602.c delete mode 100644 sound/soc/blackfin/bf6xx-i2s.c delete mode 100644 sound/soc/blackfin/bf6xx-sport.c delete mode 100644 sound/soc/blackfin/bf6xx-sport.h delete mode 100644 sound/soc/blackfin/bfin-eval-adau1373.c delete mode 100644 sound/soc/blackfin/bfin-eval-adau1701.c delete mode 100644 sound/soc/blackfin/bfin-eval-adau1x61.c delete mode 100644 sound/soc/blackfin/bfin-eval-adau1x81.c delete mode 100644 sound/soc/blackfin/bfin-eval-adav80x.c diff --git a/MAINTAINERS b/MAINTAINERS index 205c8fc12a9c..508b9e402abb 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -841,13 +841,6 @@ F: sound/soc/codecs/ad7* F: sound/soc/codecs/ssm* F: sound/soc/codecs/sigmadsp.* -ANALOG DEVICES INC ASOC DRIVERS -L: adi-buildroot-devel(a)lists.sourceforge.net (moderated for non-subscribers) -L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) -W: http://blackfin.uclinux.org/ -S: Supported -F: sound/soc/blackfin/* - ANALOG DEVICES INC DMA DRIVERS M: Lars-Peter Clausen <lars(a)metafoo.de> W: http://ez.analog.com/community/linux-device-drivers diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 84c3582f3982..41af6b9cc350 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -45,7 +45,6 @@ source "sound/soc/amd/Kconfig" source "sound/soc/atmel/Kconfig" source "sound/soc/au1x/Kconfig" source "sound/soc/bcm/Kconfig" -source "sound/soc/blackfin/Kconfig" source "sound/soc/cirrus/Kconfig" source "sound/soc/davinci/Kconfig" source "sound/soc/dwc/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 74cd1858d38b..8d92492183d2 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -29,7 +29,6 @@ obj-$(CONFIG_SND_SOC) += amd/ obj-$(CONFIG_SND_SOC) += atmel/ obj-$(CONFIG_SND_SOC) += au1x/ obj-$(CONFIG_SND_SOC) += bcm/ -obj-$(CONFIG_SND_SOC) += blackfin/ obj-$(CONFIG_SND_SOC) += cirrus/ obj-$(CONFIG_SND_SOC) += davinci/ obj-$(CONFIG_SND_SOC) += dwc/ diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig deleted file mode 100644 index 6410aa2cc2cf..000000000000 --- a/sound/soc/blackfin/Kconfig +++ /dev/null @@ -1,205 +0,0 @@ -config SND_BF5XX_I2S - tristate "SoC I2S Audio for the ADI Blackfin chip" - depends on BLACKFIN - select SND_BF5XX_SOC_SPORT if !BF60x - select SND_BF6XX_SOC_SPORT if BF60x - help - Say Y or M if you want to add support for codecs attached to - the Blackfin SPORT (synchronous serial ports) interface in I2S - mode (supports single stereo In/Out). - You will also need to select the audio interfaces to support below. - -config SND_BF5XX_SOC_SSM2602 - tristate "SoC SSM2602 Audio Codec Add-On Card support" - depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI - select SND_BF5XX_SOC_I2S if !BF60x - select SND_BF6XX_SOC_I2S if BF60x - select SND_SOC_SSM2602_SPI if SPI_MASTER - select SND_SOC_SSM2602_I2C if I2C - help - Say Y if you want to add support for the Analog Devices - SSM2602 Audio Codec Add-On Card. - -config SND_SOC_BFIN_EVAL_ADAU1701 - tristate "Support for the EVAL-ADAU1701MINIZ board on Blackfin eval boards" - depends on SND_BF5XX_I2S && I2C - select SND_BF5XX_SOC_I2S - select SND_SOC_ADAU1701 - help - Say Y if you want to add support for the Analog Devices EVAL-ADAU1701MINIZ - board connected to one of the Blackfin evaluation boards like the - BF5XX-STAMP or BF5XX-EZKIT. - -config SND_SOC_BFIN_EVAL_ADAU1373 - tristate "Support for the EVAL-ADAU1373 board on Blackfin eval boards" - depends on SND_BF5XX_I2S && I2C - select SND_BF5XX_SOC_I2S - select SND_SOC_ADAU1373 - help - Say Y if you want to add support for the Analog Devices EVAL-ADAU1373 - board connected to one of the Blackfin evaluation boards like the - BF5XX-STAMP or BF5XX-EZKIT. - - Note: This driver assumes that first ADAU1373 DAI is connected to the - first SPORT port on the BF5XX board. - -config SND_SOC_BFIN_EVAL_ADAU1X61 - tristate "Support for the EVAL-ADAU1X61 board on Blackfin eval boards" - depends on SND_BF5XX_I2S && I2C - select SND_BF5XX_SOC_I2S - select SND_SOC_ADAU1761_I2C - help - Say Y if you want to add support for the Analog Devices EVAL-ADAU1X61 - board connected to one of the Blackfin evaluation boards like the - BF5XX-STAMP or BF5XX-EZKIT. - - Note: This driver assumes that the ADAU1X61 is connected to the - first SPORT port on the BF5XX board. - -config SND_SOC_BFIN_EVAL_ADAU1X81 - tristate "Support for the EVAL-ADAU1X81 boards on Blackfin eval boards" - depends on SND_BF5XX_I2S && I2C - select SND_BF5XX_SOC_I2S - select SND_SOC_ADAU1781_I2C - help - Say Y if you want to add support for the Analog Devices EVAL-ADAU1X81 - board connected to one of the Blackfin evaluation boards like the - BF5XX-STAMP or BF5XX-EZKIT. - - Note: This driver assumes that the ADAU1X81 is connected to the - first SPORT port on the BF5XX board. - -config SND_SOC_BFIN_EVAL_ADAV80X - tristate "Support for the EVAL-ADAV80X boards on Blackfin eval boards" - depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI - select SND_BF5XX_SOC_I2S - select SND_SOC_ADAV801 if SPI_MASTER - select SND_SOC_ADAV803 if I2C - help - Say Y if you want to add support for the Analog Devices EVAL-ADAV801 or - EVAL-ADAV803 board connected to one of the Blackfin evaluation boards - like the BF5XX-STAMP or BF5XX-EZKIT. - - Note: This driver assumes that the ADAV80X digital record and playback - interfaces are connected to the first SPORT port on the BF5XX board. - -config SND_BF5XX_SOC_AD1836 - tristate "SoC AD1836 Audio support for BF5xx" - depends on SND_BF5XX_I2S && SPI_MASTER - select SND_BF5XX_SOC_I2S - select SND_SOC_AD1836 - help - Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT. - -config SND_BF5XX_SOC_AD193X - tristate "SoC AD193X Audio support for Blackfin" - depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI - select SND_BF5XX_SOC_I2S - select SND_SOC_AD193X_I2C if I2C - select SND_SOC_AD193X_SPI if SPI_MASTER - help - Say Y if you want to add support for AD193X codec on Blackfin. - This driver supports AD1936, AD1937, AD1938 and AD1939. - -config SND_BF5XX_SOC_AD73311 - tristate "SoC AD73311 Audio support for Blackfin" - depends on SND_BF5XX_I2S - select SND_BF5XX_SOC_I2S - select SND_SOC_AD73311 - help - Say Y if you want to add support for AD73311 codec on Blackfin. - -config SND_BFIN_AD73311_SE - int "PF pin for AD73311L Chip Select" - depends on SND_BF5XX_SOC_AD73311 - default 4 - help - Enter the GPIO used to control AD73311's SE pin. Acceptable - values are 0 to 7 - -config SND_BF5XX_AC97 - tristate "SoC AC97 Audio for the ADI BF5xx chip" - depends on BLACKFIN - select AC97_BUS - select SND_SOC_AC97_BUS - select SND_BF5XX_SOC_SPORT - select SND_BF5XX_SOC_AC97 - help - Say Y or M if you want to add support for codecs attached to - the Blackfin SPORT (synchronous serial ports) interface in slot 16 - mode (pseudo AC97 interface). - You will also need to select the audio interfaces to support below. - - Note: - AC97 codecs which do not implement the slot-16 mode will not function - properly with this driver. This driver is known to work with the - Analog Devices line of AC97 codecs. - -config SND_BF5XX_MMAP_SUPPORT - bool "Enable MMAP Support" - depends on SND_BF5XX_AC97 - default y - help - Say y if you want AC97 driver to support mmap mode. - We introduce an intermediate buffer to simulate mmap. - -config SND_BF5XX_MULTICHAN_SUPPORT - bool "Enable Multichannel Support" - depends on SND_BF5XX_AC97 - default n - help - Say y if you want AC97 driver to support up to 5.1 channel audio. - this mode will consume much more memory for DMA. - -config SND_BF5XX_HAVE_COLD_RESET - bool "BOARD has COLD Reset GPIO" - depends on SND_BF5XX_AC97 - default y if BFIN548_EZKIT - default n if !BFIN548_EZKIT - -config SND_BF5XX_RESET_GPIO_NUM - int "Set a GPIO for cold reset" - depends on SND_BF5XX_HAVE_COLD_RESET - range 0 159 - default 19 if BFIN548_EZKIT - default 5 if BFIN537_STAMP - default 0 - help - Set the correct GPIO for RESET the sound chip. - -config SND_BF5XX_SOC_AD1980 - tristate "SoC AD1980/1 Audio support for BF5xx (Obsolete)" - depends on SND_BF5XX_AC97 - select SND_BF5XX_SOC_AC97 - select SND_SOC_AD1980 - help - Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT. - - Warning: - Because Analog Devices Inc. discontinued the ad1980 sound chip since - Sep. 2009, this ad1980 driver is not maintained, tested and supported - by ADI now. - -config SND_BF5XX_SOC_SPORT - tristate - -config SND_BF6XX_SOC_SPORT - tristate - -config SND_BF5XX_SOC_I2S - tristate - -config SND_BF6XX_SOC_I2S - tristate - -config SND_BF5XX_SOC_AC97 - tristate - -config SND_BF5XX_SPORT_NUM - int "Set a SPORT for Sound chip" - depends on (SND_BF5XX_SOC_SPORT || SND_BF6XX_SOC_SPORT) - range 0 3 if BF54x - range 0 1 if !BF54x - default 0 - help - Set the correct SPORT for sound chip. diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile deleted file mode 100644 index ebeb6a9cedd2..000000000000 --- a/sound/soc/blackfin/Makefile +++ /dev/null @@ -1,40 +0,0 @@ -# SPDX-License-Identifier: GPL-2.0 -# Blackfin Platform Support -snd-bf5xx-ac97-objs := bf5xx-ac97-pcm.o -snd-bf5xx-i2s-objs := bf5xx-i2s-pcm.o -snd-soc-bf5xx-sport-objs := bf5xx-sport.o -snd-soc-bf6xx-sport-objs := bf6xx-sport.o -snd-soc-bf5xx-ac97-objs := bf5xx-ac97.o -snd-soc-bf5xx-i2s-objs := bf5xx-i2s.o -snd-soc-bf6xx-i2s-objs := bf6xx-i2s.o - -obj-$(CONFIG_SND_BF5XX_AC97) += snd-bf5xx-ac97.o -obj-$(CONFIG_SND_BF5XX_I2S) += snd-bf5xx-i2s.o -obj-$(CONFIG_SND_BF5XX_SOC_SPORT) += snd-soc-bf5xx-sport.o -obj-$(CONFIG_SND_BF6XX_SOC_SPORT) += snd-soc-bf6xx-sport.o -obj-$(CONFIG_SND_BF5XX_SOC_AC97) += snd-soc-bf5xx-ac97.o -obj-$(CONFIG_SND_BF5XX_SOC_I2S) += snd-soc-bf5xx-i2s.o -obj-$(CONFIG_SND_BF6XX_SOC_I2S) += snd-soc-bf6xx-i2s.o - -# Blackfin Machine Support -snd-ad1836-objs := bf5xx-ad1836.o -snd-ad1980-objs := bf5xx-ad1980.o -snd-ssm2602-objs := bf5xx-ssm2602.o -snd-ad73311-objs := bf5xx-ad73311.o -snd-ad193x-objs := bf5xx-ad193x.o -snd-soc-bfin-eval-adau1373-objs := bfin-eval-adau1373.o -snd-soc-bfin-eval-adau1x61-objs := bfin-eval-adau1x61.o -snd-soc-bfin-eval-adau1x81-objs := bfin-eval-adau1x81.o -snd-soc-bfin-eval-adau1701-objs := bfin-eval-adau1701.o -snd-soc-bfin-eval-adav80x-objs := bfin-eval-adav80x.o - -obj-$(CONFIG_SND_BF5XX_SOC_AD1836) += snd-ad1836.o -obj-$(CONFIG_SND_BF5XX_SOC_AD1980) += snd-ad1980.o -obj-$(CONFIG_SND_BF5XX_SOC_SSM2602) += snd-ssm2602.o -obj-$(CONFIG_SND_BF5XX_SOC_AD73311) += snd-ad73311.o -obj-$(CONFIG_SND_BF5XX_SOC_AD193X) += snd-ad193x.o -obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAU1373) += snd-soc-bfin-eval-adau1373.o -obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAU1X61) += snd-soc-bfin-eval-adau1x61.o -obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAU1X81) += snd-soc-bfin-eval-adau1x81.o -obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAU1701) += snd-soc-bfin-eval-adau1701.o -obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAV80X) += snd-soc-bfin-eval-adav80x.o diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c deleted file mode 100644 index 08bceb519c29..000000000000 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ /dev/null @@ -1,480 +0,0 @@ -/* - * File: sound/soc/blackfin/bf5xx-ac97-pcm.c - * Author: Cliff Cai <Cliff.Cai(a)analog.com> - * - * Created: Tue June 06 2008 - * Description: DMA Driver for AC97 sound chip - * - * Modified: - * Copyright 2008 Analog Devices Inc. - * - * Bugs: Enter bugs at http://blackfin.uclinux.org/ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, see the file COPYING, or write - * to the Free Software Foundation, Inc., - * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA - */ - -#include <linux/module.h> -#include <linux/init.h> -#include <linux/platform_device.h> -#include <linux/dma-mapping.h> -#include <linux/gfp.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include <asm/dma.h> - -#include "bf5xx-ac97.h" -#include "bf5xx-sport.h" - -static unsigned int ac97_chan_mask[] = { - SP_FL, /* Mono */ - SP_STEREO, /* Stereo */ - SP_2DOT1, /* 2.1*/ - SP_QUAD,/*Quadraquic*/ - SP_FL | SP_FR | SP_FC | SP_SL | SP_SR,/*5 channels */ - SP_5DOT1, /* 5.1 */ -}; - -#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) -static void bf5xx_mmap_copy(struct snd_pcm_substream *substream, - snd_pcm_uframes_t count) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct sport_device *sport = runtime->private_data; - unsigned int chan_mask = ac97_chan_mask[runtime->channels - 1]; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - bf5xx_pcm_to_ac97((struct ac97_frame *)sport->tx_dma_buf + - sport->tx_pos, (__u16 *)runtime->dma_area + sport->tx_pos * - runtime->channels, count, chan_mask); - sport->tx_pos += runtime->period_size; - if (sport->tx_pos >= runtime->buffer_size) - sport->tx_pos %= runtime->buffer_size; - sport->tx_delay_pos = sport->tx_pos; - } else { - bf5xx_ac97_to_pcm((struct ac97_frame *)sport->rx_dma_buf + - sport->rx_pos, (__u16 *)runtime->dma_area + sport->rx_pos * - runtime->channels, count); - sport->rx_pos += runtime->period_size; - if (sport->rx_pos >= runtime->buffer_size) - sport->rx_pos %= runtime->buffer_size; - } -} -#endif - -static void bf5xx_dma_irq(void *data) -{ - struct snd_pcm_substream *pcm = data; -#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) - struct snd_pcm_runtime *runtime = pcm->runtime; - struct sport_device *sport = runtime->private_data; - bf5xx_mmap_copy(pcm, runtime->period_size); - if (pcm->stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (sport->once == 0) { - snd_pcm_period_elapsed(pcm); - bf5xx_mmap_copy(pcm, runtime->period_size); - sport->once = 1; - } - } -#endif - snd_pcm_period_elapsed(pcm); -} - -/* The memory size for pure pcm data is 128*1024 = 0x20000 bytes. - * The total rx/tx buffer is for ac97 frame to hold all pcm data - * is 0x20000 * sizeof(struct ac97_frame) / 4. - */ -static const struct snd_pcm_hardware bf5xx_pcm_hardware = { - .info = SNDRV_PCM_INFO_INTERLEAVED | -#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | -#endif - SNDRV_PCM_INFO_BLOCK_TRANSFER, - - .period_bytes_min = 32, - .period_bytes_max = 0x10000, - .periods_min = 1, - .periods_max = PAGE_SIZE/32, - .buffer_bytes_max = 0x20000, /* 128 kbytes */ - .fifo_size = 16, -}; - -static int bf5xx_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - size_t size = bf5xx_pcm_hardware.buffer_bytes_max - * sizeof(struct ac97_frame) / 4; - - snd_pcm_lib_malloc_pages(substream, size); - - return 0; -} - -static int bf5xx_pcm_hw_free(struct snd_pcm_substream *substream) -{ -#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) - struct snd_pcm_runtime *runtime = substream->runtime; - struct sport_device *sport = runtime->private_data; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - sport->once = 0; - if (runtime->dma_area) - memset(runtime->dma_area, 0, runtime->buffer_size); - memset(sport->tx_dma_buf, 0, runtime->buffer_size * - sizeof(struct ac97_frame)); - } else - memset(sport->rx_dma_buf, 0, runtime->buffer_size * - sizeof(struct ac97_frame)); -#endif - snd_pcm_lib_free_pages(substream); - return 0; -} - -static int bf5xx_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct sport_device *sport = runtime->private_data; - - /* An intermediate buffer is introduced for implementing mmap for - * SPORT working in TMD mode(include AC97). - */ -#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - sport_set_tx_callback(sport, bf5xx_dma_irq, substream); - sport_config_tx_dma(sport, sport->tx_dma_buf, runtime->periods, - runtime->period_size * sizeof(struct ac97_frame)); - } else { - sport_set_rx_callback(sport, bf5xx_dma_irq, substream); - sport_config_rx_dma(sport, sport->rx_dma_buf, runtime->periods, - runtime->period_size * sizeof(struct ac97_frame)); - } -#else - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - sport_set_tx_callback(sport, bf5xx_dma_irq, substream); - sport_config_tx_dma(sport, runtime->dma_area, runtime->periods, - runtime->period_size * sizeof(struct ac97_frame)); - } else { - sport_set_rx_callback(sport, bf5xx_dma_irq, substream); - sport_config_rx_dma(sport, runtime->dma_area, runtime->periods, - runtime->period_size * sizeof(struct ac97_frame)); - } -#endif - return 0; -} - -static int bf5xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct sport_device *sport = runtime->private_data; - int ret = 0; - - pr_debug("%s enter\n", __func__); - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { -#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) - bf5xx_mmap_copy(substream, runtime->period_size); - sport->tx_delay_pos = 0; -#endif - sport_tx_start(sport); - } else - sport_rx_start(sport); - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { -#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) - sport->tx_pos = 0; -#endif - sport_tx_stop(sport); - } else { -#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) - sport->rx_pos = 0; -#endif - sport_rx_stop(sport); - } - break; - default: - ret = -EINVAL; - } - return ret; -} - -static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct sport_device *sport = runtime->private_data; - unsigned int curr; - -#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - curr = sport->tx_delay_pos; - else - curr = sport->rx_pos; -#else - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - curr = sport_curr_offset_tx(sport) / sizeof(struct ac97_frame); - else - curr = sport_curr_offset_rx(sport) / sizeof(struct ac97_frame); - -#endif - return curr; -} - -static int bf5xx_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct sport_device *sport_handle = snd_soc_dai_get_drvdata(cpu_dai); - struct snd_pcm_runtime *runtime = substream->runtime; - int ret; - - pr_debug("%s enter\n", __func__); - snd_soc_set_runtime_hwparams(substream, &bf5xx_pcm_hardware); - - ret = snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) - goto out; - - if (sport_handle != NULL) - runtime->private_data = sport_handle; - else { - pr_err("sport_handle is NULL\n"); - return -1; - } - return 0; - - out: - return ret; -} - -#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) -static int bf5xx_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - size_t size = vma->vm_end - vma->vm_start; - vma->vm_start = (unsigned long)runtime->dma_area; - vma->vm_end = vma->vm_start + size; - vma->vm_flags |= VM_SHARED; - return 0 ; -} -#else -static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, - int channel, unsigned long pos, - void *buf, unsigned long count) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int chan_mask = ac97_chan_mask[runtime->channels - 1]; - struct ac97_frame *dst; - - pr_debug("%s copy pos:0x%lx count:0x%lx\n", - substream->stream ? "Capture" : "Playback", pos, count); - dst = (struct ac97_frame *)runtime->dma_area + - bytes_to_frames(runtime, pos); - count = bytes_to_frames(runtime, count); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - bf5xx_pcm_to_ac97(dst, buf, count, chan_mask); - else - bf5xx_ac97_to_pcm(dst, buf, count); - return 0; -} - -static int bf5xx_pcm_copy_user(struct snd_pcm_substream *substream, - int channel, unsigned long pos, - void __user *buf, unsigned long count) -{ - return bf5xx_pcm_copy(substream, channel, pos, (void *)buf, count); -} -#endif - -static const struct snd_pcm_ops bf5xx_pcm_ac97_ops = { - .open = bf5xx_pcm_open, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = bf5xx_pcm_hw_params, - .hw_free = bf5xx_pcm_hw_free, - .prepare = bf5xx_pcm_prepare, - .trigger = bf5xx_pcm_trigger, - .pointer = bf5xx_pcm_pointer, -#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) - .mmap = bf5xx_pcm_mmap, -#else - .copy_user = bf5xx_pcm_copy_user, - .copy_kernel = bf5xx_pcm_copy, -#endif -}; - -static int bf5xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) -{ - struct snd_soc_pcm_runtime *rtd = pcm->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct sport_device *sport_handle = snd_soc_dai_get_drvdata(cpu_dai); - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = bf5xx_pcm_hardware.buffer_bytes_max - * sizeof(struct ac97_frame) / 4; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_coherent(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - if (!buf->area) { - pr_err("Failed to allocate dma memory\n"); - pr_err("Please increase uncached DMA memory region\n"); - return -ENOMEM; - } - buf->bytes = size; - - pr_debug("%s, area:%p, size:0x%08lx\n", __func__, - buf->area, buf->bytes); - - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - sport_handle->tx_buf = buf->area; - else - sport_handle->rx_buf = buf->area; - -/* - * Need to allocate local buffer when enable - * MMAP for SPORT working in TMD mode (include AC97). - */ -#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (!sport_handle->tx_dma_buf) { - sport_handle->tx_dma_buf = dma_alloc_coherent(NULL, \ - size, &sport_handle->tx_dma_phy, GFP_KERNEL); - if (!sport_handle->tx_dma_buf) { - pr_err("Failed to allocate memory for tx dma buf - Please increase uncached DMA memory region\n"); - return -ENOMEM; - } else - memset(sport_handle->tx_dma_buf, 0, size); - } else - memset(sport_handle->tx_dma_buf, 0, size); - } else { - if (!sport_handle->rx_dma_buf) { - sport_handle->rx_dma_buf = dma_alloc_coherent(NULL, \ - size, &sport_handle->rx_dma_phy, GFP_KERNEL); - if (!sport_handle->rx_dma_buf) { - pr_err("Failed to allocate memory for rx dma buf - Please increase uncached DMA memory region\n"); - return -ENOMEM; - } else - memset(sport_handle->rx_dma_buf, 0, size); - } else - memset(sport_handle->rx_dma_buf, 0, size); - } -#endif - return 0; -} - -static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; -#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) - struct snd_soc_pcm_runtime *rtd = pcm->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct sport_device *sport_handle = snd_soc_dai_get_drvdata(cpu_dai); - size_t size = bf5xx_pcm_hardware.buffer_bytes_max * - sizeof(struct ac97_frame) / 4; -#endif - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - dma_free_coherent(NULL, buf->bytes, buf->area, 0); - buf->area = NULL; -#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (sport_handle->tx_dma_buf) - dma_free_coherent(NULL, size, \ - sport_handle->tx_dma_buf, 0); - sport_handle->tx_dma_buf = NULL; - } else { - - if (sport_handle->rx_dma_buf) - dma_free_coherent(NULL, size, \ - sport_handle->rx_dma_buf, 0); - sport_handle->rx_dma_buf = NULL; - } -#endif - } -} - -static int bf5xx_pcm_ac97_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_pcm *pcm = rtd->pcm; - int ret; - - pr_debug("%s enter\n", __func__); - ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); - if (ret) - return ret; - - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - ret = bf5xx_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - ret = bf5xx_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } - out: - return ret; -} - -static struct snd_soc_component_driver bf5xx_ac97_soc_component = { - .ops = &bf5xx_pcm_ac97_ops, - .pcm_new = bf5xx_pcm_ac97_new, - .pcm_free = bf5xx_pcm_free_dma_buffers, -}; - -static int bf5xx_soc_platform_probe(struct platform_device *pdev) -{ - return devm_snd_soc_register_component(&pdev->dev, - &bf5xx_ac97_soc_component, NULL, 0); -} - -static struct platform_driver bf5xx_pcm_driver = { - .driver = { - .name = "bfin-ac97-pcm-audio", - }, - - .probe = bf5xx_soc_platform_probe, -}; - -module_platform_driver(bf5xx_pcm_driver); - -MODULE_AUTHOR("Cliff Cai"); -MODULE_DESCRIPTION("ADI Blackfin AC97 PCM DMA module"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c deleted file mode 100644 index a040cfe29fc0..000000000000 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ /dev/null @@ -1,388 +0,0 @@ -/* - * bf5xx-ac97.c -- AC97 support for the ADI blackfin chip. - * - * Author: Roy Huang - * Created: 11th. June 2007 - * Copyright: Analog Device Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include <linux/init.h> -#include <linux/module.h> -#include <linux/platform_device.h> -#include <linux/interrupt.h> -#include <linux/wait.h> -#include <linux/delay.h> -#include <linux/slab.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/ac97_codec.h> -#include <sound/initval.h> -#include <sound/soc.h> - -#include <asm/irq.h> -#include <asm/portmux.h> -#include <linux/mutex.h> -#include <linux/gpio.h> - -#include "bf5xx-sport.h" -#include "bf5xx-ac97.h" - -/* Anomaly notes: - * 05000250 - AD1980 is running in TDM mode and RFS/TFS are generated by SPORT - * contrtoller. But, RFSDIV and TFSDIV are always set to 16*16-1, - * while the max AC97 data size is 13*16. The DIV is always larger - * than data size. AD73311 and ad2602 are not running in TDM mode. - * AD1836 and AD73322 depend on external RFS/TFS only. So, this - * anomaly does not affect blackfin sound drivers. -*/ - -static struct sport_device *ac97_sport_handle; - -void bf5xx_pcm_to_ac97(struct ac97_frame *dst, const __u16 *src, - size_t count, unsigned int chan_mask) -{ - while (count--) { - dst->ac97_tag = TAG_VALID; - if (chan_mask & SP_FL) { - dst->ac97_pcm_r = *src++; - dst->ac97_tag |= TAG_PCM_RIGHT; - } - if (chan_mask & SP_FR) { - dst->ac97_pcm_l = *src++; - dst->ac97_tag |= TAG_PCM_LEFT; - - } -#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT) - if (chan_mask & SP_SR) { - dst->ac97_sl = *src++; - dst->ac97_tag |= TAG_PCM_SL; - } - if (chan_mask & SP_SL) { - dst->ac97_sr = *src++; - dst->ac97_tag |= TAG_PCM_SR; - } - if (chan_mask & SP_LFE) { - dst->ac97_lfe = *src++; - dst->ac97_tag |= TAG_PCM_LFE; - } - if (chan_mask & SP_FC) { - dst->ac97_center = *src++; - dst->ac97_tag |= TAG_PCM_CENTER; - } -#endif - dst++; - } -} -EXPORT_SYMBOL(bf5xx_pcm_to_ac97); - -void bf5xx_ac97_to_pcm(const struct ac97_frame *src, __u16 *dst, - size_t count) -{ - while (count--) { - *(dst++) = src->ac97_pcm_l; - *(dst++) = src->ac97_pcm_r; - src++; - } -} -EXPORT_SYMBOL(bf5xx_ac97_to_pcm); - -static unsigned int sport_tx_curr_frag(struct sport_device *sport) -{ - return sport->tx_curr_frag = sport_curr_offset_tx(sport) / - sport->tx_fragsize; -} - -static void enqueue_cmd(struct snd_ac97 *ac97, __u16 addr, __u16 data) -{ - struct sport_device *sport = ac97_sport_handle; - int *cmd_count = sport->private_data; - int nextfrag = sport_tx_curr_frag(sport); - struct ac97_frame *nextwrite; - - sport_incfrag(sport, &nextfrag, 1); - - nextwrite = (struct ac97_frame *)(sport->tx_buf + - nextfrag * sport->tx_fragsize); - pr_debug("sport->tx_buf:%p, nextfrag:0x%x nextwrite:%p, cmd_count:%d\n", - sport->tx_buf, nextfrag, nextwrite, cmd_count[nextfrag]); - nextwrite[cmd_count[nextfrag]].ac97_tag |= TAG_CMD; - nextwrite[cmd_count[nextfrag]].ac97_addr = addr; - nextwrite[cmd_count[nextfrag]].ac97_data = data; - ++cmd_count[nextfrag]; - pr_debug("ac97_sport: Inserting %02x/%04x into fragment %d\n", - addr >> 8, data, nextfrag); -} - -static unsigned short bf5xx_ac97_read(struct snd_ac97 *ac97, - unsigned short reg) -{ - struct sport_device *sport_handle = ac97_sport_handle; - struct ac97_frame out_frame[2], in_frame[2]; - - pr_debug("%s enter 0x%x\n", __func__, reg); - - /* When dma descriptor is enabled, the register should not be read */ - if (sport_handle->tx_run || sport_handle->rx_run) { - pr_err("Could you send a mail to cliff.cai(a)analog.com " - "to report this?\n"); - return -EFAULT; - } - - memset(&out_frame, 0, 2 * sizeof(struct ac97_frame)); - memset(&in_frame, 0, 2 * sizeof(struct ac97_frame)); - out_frame[0].ac97_tag = TAG_VALID | TAG_CMD; - out_frame[0].ac97_addr = ((reg << 8) | 0x8000); - sport_send_and_recv(sport_handle, (unsigned char *)&out_frame, - (unsigned char *)&in_frame, - 2 * sizeof(struct ac97_frame)); - return in_frame[1].ac97_data; -} - -void bf5xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg, - unsigned short val) -{ - struct sport_device *sport_handle = ac97_sport_handle; - - pr_debug("%s enter 0x%x:0x%04x\n", __func__, reg, val); - - if (sport_handle->tx_run) { - enqueue_cmd(ac97, (reg << 8), val); /* write */ - enqueue_cmd(ac97, (reg << 8) | 0x8000, 0); /* read back */ - } else { - struct ac97_frame frame; - memset(&frame, 0, sizeof(struct ac97_frame)); - frame.ac97_tag = TAG_VALID | TAG_CMD; - frame.ac97_addr = (reg << 8); - frame.ac97_data = val; - sport_send_and_recv(sport_handle, (unsigned char *)&frame, \ - NULL, sizeof(struct ac97_frame)); - } -} - -static void bf5xx_ac97_warm_reset(struct snd_ac97 *ac97) -{ - struct sport_device *sport_handle = ac97_sport_handle; - u16 gpio = P_IDENT(sport_handle->pin_req[3]); - - pr_debug("%s enter\n", __func__); - - peripheral_free_list(sport_handle->pin_req); - gpio_request(gpio, "bf5xx-ac97"); - gpio_direction_output(gpio, 1); - udelay(2); - gpio_set_value(gpio, 0); - udelay(1); - gpio_free(gpio); - peripheral_request_list(sport_handle->pin_req, "soc-audio"); -} - -static void bf5xx_ac97_cold_reset(struct snd_ac97 *ac97) -{ -#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET - pr_debug("%s enter\n", __func__); - - /* It is specified for bf548-ezkit */ - gpio_set_value(CONFIG_SND_BF5XX_RESET_GPIO_NUM, 0); - /* Keep reset pin low for 1 ms */ - mdelay(1); - gpio_set_value(CONFIG_SND_BF5XX_RESET_GPIO_NUM, 1); - /* Wait for bit clock recover */ - mdelay(1); -#else - pr_info("%s: Not implemented\n", __func__); -#endif -} - -static struct snd_ac97_bus_ops bf5xx_ac97_ops = { - .read = bf5xx_ac97_read, - .write = bf5xx_ac97_write, - .warm_reset = bf5xx_ac97_warm_reset, - .reset = bf5xx_ac97_cold_reset, -}; - -#ifdef CONFIG_PM -static int bf5xx_ac97_suspend(struct snd_soc_dai *dai) -{ - struct sport_device *sport = snd_soc_dai_get_drvdata(dai); - - pr_debug("%s : sport %d\n", __func__, dai->id); - if (!dai->active) - return 0; - if (dai->capture_active) - sport_rx_stop(sport); - if (dai->playback_active) - sport_tx_stop(sport); - return 0; -} - -static int bf5xx_ac97_resume(struct snd_soc_dai *dai) -{ - int ret; - struct sport_device *sport = snd_soc_dai_get_drvdata(dai); - - pr_debug("%s : sport %d\n", __func__, dai->id); - if (!dai->active) - return 0; - -#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT) - ret = sport_set_multichannel(sport, 16, 0x3FF, 0x3FF, 1); -#else - ret = sport_set_multichannel(sport, 16, 0x1F, 0x1F, 1); -#endif - if (ret) { - pr_err("SPORT is busy!\n"); - return -EBUSY; - } - - ret = sport_config_rx(sport, IRFS, 0xF, 0, (16*16-1)); - if (ret) { - pr_err("SPORT is busy!\n"); - return -EBUSY; - } - - ret = sport_config_tx(sport, ITFS, 0xF, 0, (16*16-1)); - if (ret) { - pr_err("SPORT is busy!\n"); - return -EBUSY; - } - - return 0; -} - -#else -#define bf5xx_ac97_suspend NULL -#define bf5xx_ac97_resume NULL -#endif - -static struct snd_soc_dai_driver bfin_ac97_dai = { - .bus_control = true, - .suspend = bf5xx_ac97_suspend, - .resume = bf5xx_ac97_resume, - .playback = { - .stream_name = "AC97 Playback", - .channels_min = 2, -#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT) - .channels_max = 6, -#else - .channels_max = 2, -#endif - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, }, - .capture = { - .stream_name = "AC97 Capture", - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, }, -}; - -static const struct snd_soc_component_driver bfin_ac97_component = { - .name = "bfin-ac97", -}; - -static int asoc_bfin_ac97_probe(struct platform_device *pdev) -{ - struct sport_device *sport_handle; - int ret; - -#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET - /* Request PB3 as reset pin */ - ret = devm_gpio_request_one(&pdev->dev, - CONFIG_SND_BF5XX_RESET_GPIO_NUM, - GPIOF_OUT_INIT_HIGH, "SND_AD198x RESET"); - if (ret) { - dev_err(&pdev->dev, - "Failed to request GPIO_%d for reset: %d\n", - CONFIG_SND_BF5XX_RESET_GPIO_NUM, ret); - return ret; - } -#endif - - sport_handle = sport_init(pdev, 2, sizeof(struct ac97_frame), - PAGE_SIZE); - if (!sport_handle) { - ret = -ENODEV; - goto sport_err; - } - - /*SPORT works in TDM mode to simulate AC97 transfers*/ -#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT) - ret = sport_set_multichannel(sport_handle, 16, 0x3FF, 0x3FF, 1); -#else - ret = sport_set_multichannel(sport_handle, 16, 0x1F, 0x1F, 1); -#endif - if (ret) { - pr_err("SPORT is busy!\n"); - ret = -EBUSY; - goto sport_config_err; - } - - ret = sport_config_rx(sport_handle, IRFS, 0xF, 0, (16*16-1)); - if (ret) { - pr_err("SPORT is busy!\n"); - ret = -EBUSY; - goto sport_config_err; - } - - ret = sport_config_tx(sport_handle, ITFS, 0xF, 0, (16*16-1)); - if (ret) { - pr_err("SPORT is busy!\n"); - ret = -EBUSY; - goto sport_config_err; - } - - ret = snd_soc_set_ac97_ops(&bf5xx_ac97_ops); - if (ret != 0) { - dev_err(&pdev->dev, "Failed to set AC'97 ops: %d\n", ret); - goto sport_config_err; - } - - ret = snd_soc_register_component(&pdev->dev, &bfin_ac97_component, - &bfin_ac97_dai, 1); - if (ret) { - pr_err("Failed to register DAI: %d\n", ret); - goto sport_config_err; - } - - ac97_sport_handle = sport_handle; - - return 0; - -sport_config_err: - sport_done(sport_handle); -sport_err: - snd_soc_set_ac97_ops(NULL); - - return ret; -} - -static int asoc_bfin_ac97_remove(struct platform_device *pdev) -{ - struct sport_device *sport_handle = platform_get_drvdata(pdev); - - snd_soc_unregister_component(&pdev->dev); - sport_done(sport_handle); - snd_soc_set_ac97_ops(NULL); - - return 0; -} - -static struct platform_driver asoc_bfin_ac97_driver = { - .driver = { - .name = "bfin-ac97", - }, - - .probe = asoc_bfin_ac97_probe, - .remove = asoc_bfin_ac97_remove, -}; - -module_platform_driver(asoc_bfin_ac97_driver); - -MODULE_AUTHOR("Roy Huang"); -MODULE_DESCRIPTION("AC97 driver for ADI Blackfin"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/blackfin/bf5xx-ac97.h b/sound/soc/blackfin/bf5xx-ac97.h deleted file mode 100644 index a680fdc9bb42..000000000000 --- a/sound/soc/blackfin/bf5xx-ac97.h +++ /dev/null @@ -1,57 +0,0 @@ -/* - * sound/soc/blackfin/bf5xx-ac97.h - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _BF5XX_AC97_H -#define _BF5XX_AC97_H - -/* Frame format in memory, only support stereo currently */ -struct ac97_frame { - u16 ac97_tag; /* slot 0 */ - u16 ac97_addr; /* slot 1 */ - u16 ac97_data; /* slot 2 */ - u16 ac97_pcm_l; /*slot 3:front left*/ - u16 ac97_pcm_r; /*slot 4:front left*/ -#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT) - u16 ac97_mdm_l1; - u16 ac97_center; /*slot 6:center*/ - u16 ac97_sl; /*slot 7:surround left*/ - u16 ac97_sr; /*slot 8:surround right*/ - u16 ac97_lfe; /*slot 9:lfe*/ -#endif -} __attribute__ ((packed)); - -/* Speaker location */ -#define SP_FL 0x0001 -#define SP_FR 0x0010 -#define SP_FC 0x0002 -#define SP_LFE 0x0020 -#define SP_SL 0x0004 -#define SP_SR 0x0040 - -#define SP_STEREO (SP_FL | SP_FR) -#define SP_2DOT1 (SP_FL | SP_FR | SP_LFE) -#define SP_QUAD (SP_FL | SP_FR | SP_SL | SP_SR) -#define SP_5DOT1 (SP_FL | SP_FR | SP_FC | SP_LFE | SP_SL | SP_SR) - -#define TAG_VALID 0x8000 -#define TAG_CMD 0x6000 -#define TAG_PCM_LEFT 0x1000 -#define TAG_PCM_RIGHT 0x0800 -#define TAG_PCM_MDM_L1 0x0400 -#define TAG_PCM_CENTER 0x0200 -#define TAG_PCM_SL 0x0100 -#define TAG_PCM_SR 0x0080 -#define TAG_PCM_LFE 0x0040 - -void bf5xx_pcm_to_ac97(struct ac97_frame *dst, const __u16 *src, \ - size_t count, unsigned int chan_mask); - -void bf5xx_ac97_to_pcm(const struct ac97_frame *src, __u16 *dst, \ - size_t count); - -#endif diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c deleted file mode 100644 index 864df2616e10..000000000000 --- a/sound/soc/blackfin/bf5xx-ad1836.c +++ /dev/null @@ -1,109 +0,0 @@ -/* - * File: sound/soc/blackfin/bf5xx-ad1836.c - * Author: Barry Song <Barry.Song(a)analog.com> - * - * Created: Aug 4 2009 - * Description: Board driver for ad1836 sound chip - * - * Bugs: Enter bugs at http://blackfin.uclinux.org/ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/device.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/pcm_params.h> - -#include <asm/blackfin.h> -#include <asm/cacheflush.h> -#include <asm/irq.h> -#include <asm/dma.h> -#include <asm/portmux.h> - -#include "../codecs/ad1836.h" - -static struct snd_soc_card bf5xx_ad1836; - -static int bf5xx_ad1836_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - unsigned int channel_map[] = {0, 4, 1, 5, 2, 6, 3, 7}; - int ret = 0; - - /* set cpu DAI channel mapping */ - ret = snd_soc_dai_set_channel_map(cpu_dai, ARRAY_SIZE(channel_map), - channel_map, ARRAY_SIZE(channel_map), channel_map); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0xFF, 0xFF, 8, 32); - if (ret < 0) - return ret; - - return 0; -} - -#define BF5XX_AD1836_DAIFMT (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_IF | \ - SND_SOC_DAIFMT_CBM_CFM) - -static struct snd_soc_dai_link bf5xx_ad1836_dai = { - .name = "ad1836", - .stream_name = "AD1836", - .codec_dai_name = "ad1836-hifi", - .platform_name = "bfin-i2s-pcm-audio", - .dai_fmt = BF5XX_AD1836_DAIFMT, - .init = bf5xx_ad1836_init, -}; - -static struct snd_soc_card bf5xx_ad1836 = { - .name = "bfin-ad1836", - .owner = THIS_MODULE, - .dai_link = &bf5xx_ad1836_dai, - .num_links = 1, -}; - -static int bf5xx_ad1836_driver_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &bf5xx_ad1836; - const char **link_name; - int ret; - - link_name = pdev->dev.platform_data; - if (!link_name) { - dev_err(&pdev->dev, "No platform data supplied\n"); - return -EINVAL; - } - bf5xx_ad1836_dai.cpu_dai_name = link_name[0]; - bf5xx_ad1836_dai.codec_name = link_name[1]; - - card->dev = &pdev->dev; - platform_set_drvdata(pdev, card); - - ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) - dev_err(&pdev->dev, "Failed to register card\n"); - return ret; -} - -static struct platform_driver bf5xx_ad1836_driver = { - .driver = { - .name = "bfin-snd-ad1836", - .pm = &snd_soc_pm_ops, - }, - .probe = bf5xx_ad1836_driver_probe, -}; -module_platform_driver(bf5xx_ad1836_driver); - -/* Module information */ -MODULE_AUTHOR("Barry Song"); -MODULE_DESCRIPTION("ALSA SoC AD1836 board driver"); -MODULE_LICENSE("GPL"); - diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c deleted file mode 100644 index 603ad1f2b9b9..000000000000 --- a/sound/soc/blackfin/bf5xx-ad193x.c +++ /dev/null @@ -1,131 +0,0 @@ -/* - * File: sound/soc/blackfin/bf5xx-ad193x.c - * Author: Barry Song <Barry.Song(a)analog.com> - * - * Created: Thur June 4 2009 - * Description: Board driver for ad193x sound chip - * - * Bugs: Enter bugs at http://blackfin.uclinux.org/ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, see the file COPYING, or write - * to the Free Software Foundation, Inc., - * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/device.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/pcm_params.h> - -#include <asm/blackfin.h> -#include <asm/cacheflush.h> -#include <asm/irq.h> -#include <asm/dma.h> -#include <asm/portmux.h> - -#include "../codecs/ad193x.h" - -static struct snd_soc_card bf5xx_ad193x; - -static int bf5xx_ad193x_link_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - /* set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, 0, 24576000, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* set codec DAI slots, 8 channels, all channels are enabled */ - ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xFF, 0xFF, 8, 32); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0xFF, 0xFF, 8, 32); - if (ret < 0) - return ret; - - return 0; -} - -#define BF5XX_AD193X_DAIFMT (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_IF | \ - SND_SOC_DAIFMT_CBM_CFM) - -static struct snd_soc_dai_link bf5xx_ad193x_dai[] = { - { - .name = "ad193x", - .stream_name = "AD193X", - .cpu_dai_name = "bfin-i2s.0", - .codec_dai_name ="ad193x-hifi", - .platform_name = "bfin-i2s-pcm-audio", - .codec_name = "spi0.5", - .dai_fmt = BF5XX_AD193X_DAIFMT, - .init = bf5xx_ad193x_link_init, - }, - { - .name = "ad193x", - .stream_name = "AD193X", - .cpu_dai_name = "bfin-i2s.1", - .codec_dai_name ="ad193x-hifi", - .platform_name = "bfin-i2s-pcm-audio", - .codec_name = "spi0.5", - .dai_fmt = BF5XX_AD193X_DAIFMT, - .init = bf5xx_ad193x_link_init, - }, -}; - -static struct snd_soc_card bf5xx_ad193x = { - .name = "bfin-ad193x", - .owner = THIS_MODULE, - .dai_link = &bf5xx_ad193x_dai[CONFIG_SND_BF5XX_SPORT_NUM], - .num_links = 1, -}; - -static struct platform_device *bfxx_ad193x_snd_device; - -static int __init bf5xx_ad193x_init(void) -{ - int ret; - - bfxx_ad193x_snd_device = platform_device_alloc("soc-audio", -1); - if (!bfxx_ad193x_snd_device) - return -ENOMEM; - - platform_set_drvdata(bfxx_ad193x_snd_device, &bf5xx_ad193x); - ret = platform_device_add(bfxx_ad193x_snd_device); - - if (ret) - platform_device_put(bfxx_ad193x_snd_device); - - return ret; -} - -static void __exit bf5xx_ad193x_exit(void) -{ - platform_device_unregister(bfxx_ad193x_snd_device); -} - -module_init(bf5xx_ad193x_init); -module_exit(bf5xx_ad193x_exit); - -/* Module information */ -MODULE_AUTHOR("Barry Song"); -MODULE_DESCRIPTION("ALSA SoC AD193X board driver"); -MODULE_LICENSE("GPL"); - diff --git a/sound/soc/blackfin/bf5xx-ad1980.c b/sound/soc/blackfin/bf5xx-ad1980.c deleted file mode 100644 index 0fa81a523b8a..000000000000 --- a/sound/soc/blackfin/bf5xx-ad1980.c +++ /dev/null @@ -1,109 +0,0 @@ -/* - * File: sound/soc/blackfin/bf5xx-ad1980.c - * Author: Cliff Cai <Cliff.Cai(a)analog.com> - * - * Created: Tue June 06 2008 - * Description: Board driver for AD1980/1 audio codec - * - * Modified: - * Copyright 2008 Analog Devices Inc. - * - * Bugs: Enter bugs at http://blackfin.uclinux.org/ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, see the file COPYING, or write - * to the Free Software Foundation, Inc., - * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA - */ - -/* - * WARNING: - * - * Because Analog Devices Inc. discontinued the ad1980 sound chip since - * Sep. 2009, this ad1980 driver is not maintained, tested and supported - * by ADI now. - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/device.h> -#include <asm/dma.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> - -#include <linux/gpio.h> -#include <asm/portmux.h> - -#include "bf5xx-ac97.h" - -static struct snd_soc_card bf5xx_board; - -static struct snd_soc_dai_link bf5xx_board_dai[] = { - { - .name = "AC97", - .stream_name = "AC97 HiFi", - .cpu_dai_name = "bfin-ac97.0", - .codec_dai_name = "ad1980-hifi", - .platform_name = "bfin-ac97-pcm-audio", - .codec_name = "ad1980", - }, - { - .name = "AC97", - .stream_name = "AC97 HiFi", - .cpu_dai_name = "bfin-ac97.1", - .codec_dai_name = "ad1980-hifi", - .platform_name = "bfin-ac97-pcm-audio", - .codec_name = "ad1980", - }, -}; - -static struct snd_soc_card bf5xx_board = { - .name = "bfin-ad1980", - .owner = THIS_MODULE, - .dai_link = &bf5xx_board_dai[CONFIG_SND_BF5XX_SPORT_NUM], - .num_links = 1, -}; - -static struct platform_device *bf5xx_board_snd_device; - -static int __init bf5xx_board_init(void) -{ - int ret; - - bf5xx_board_snd_device = platform_device_alloc("soc-audio", -1); - if (!bf5xx_board_snd_device) - return -ENOMEM; - - platform_set_drvdata(bf5xx_board_snd_device, &bf5xx_board); - ret = platform_device_add(bf5xx_board_snd_device); - - if (ret) - platform_device_put(bf5xx_board_snd_device); - - return ret; -} - -static void __exit bf5xx_board_exit(void) -{ - platform_device_unregister(bf5xx_board_snd_device); -} - -module_init(bf5xx_board_init); -module_exit(bf5xx_board_exit); - -/* Module information */ -MODULE_AUTHOR("Cliff Cai"); -MODULE_DESCRIPTION("ALSA SoC AD1980/1 BF5xx board (Obsolete)"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c deleted file mode 100644 index 786bbdd96e7c..000000000000 --- a/sound/soc/blackfin/bf5xx-ad73311.c +++ /dev/null @@ -1,212 +0,0 @@ -/* - * File: sound/soc/blackfin/bf5xx-ad73311.c - * Author: Cliff Cai <Cliff.Cai(a)analog.com> - * - * Created: Thur Sep 25 2008 - * Description: Board driver for ad73311 sound chip - * - * Modified: - * Copyright 2008 Analog Devices Inc. - * - * Bugs: Enter bugs at http://blackfin.uclinux.org/ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, see the file COPYING, or write - * to the Free Software Foundation, Inc., - * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/device.h> -#include <linux/delay.h> -#include <linux/gpio.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/pcm_params.h> - -#include <asm/blackfin.h> -#include <asm/cacheflush.h> -#include <asm/irq.h> -#include <asm/dma.h> -#include <asm/portmux.h> - -#include "../codecs/ad73311.h" -#include "bf5xx-sport.h" - -#if CONFIG_SND_BF5XX_SPORT_NUM == 0 -#define bfin_write_SPORT_TCR1 bfin_write_SPORT0_TCR1 -#define bfin_read_SPORT_TCR1 bfin_read_SPORT0_TCR1 -#define bfin_write_SPORT_TCR2 bfin_write_SPORT0_TCR2 -#define bfin_write_SPORT_TX16 bfin_write_SPORT0_TX16 -#define bfin_read_SPORT_STAT bfin_read_SPORT0_STAT -#else -#define bfin_write_SPORT_TCR1 bfin_write_SPORT1_TCR1 -#define bfin_read_SPORT_TCR1 bfin_read_SPORT1_TCR1 -#define bfin_write_SPORT_TCR2 bfin_write_SPORT1_TCR2 -#define bfin_write_SPORT_TX16 bfin_write_SPORT1_TX16 -#define bfin_read_SPORT_STAT bfin_read_SPORT1_STAT -#endif - -#define GPIO_SE CONFIG_SND_BFIN_AD73311_SE - -static struct snd_soc_card bf5xx_ad73311; - -static int snd_ad73311_startup(void) -{ - pr_debug("%s enter\n", __func__); - - /* Pull up SE pin on AD73311L */ - gpio_set_value(GPIO_SE, 1); - return 0; -} - -static int snd_ad73311_configure(void) -{ - unsigned short ctrl_regs[6]; - unsigned short status = 0; - int count = 0; - - /* DMCLK = MCLK = 16.384 MHz - * SCLK = DMCLK/8 = 2.048 MHz - * Sample Rate = DMCLK/2048 = 8 KHz - */ - ctrl_regs[0] = AD_CONTROL | AD_WRITE | CTRL_REG_B | REGB_MCDIV(0) | \ - REGB_SCDIV(0) | REGB_DIRATE(0); - ctrl_regs[1] = AD_CONTROL | AD_WRITE | CTRL_REG_C | REGC_PUDEV | \ - REGC_PUADC | REGC_PUDAC | REGC_PUREF | REGC_REFUSE ; - ctrl_regs[2] = AD_CONTROL | AD_WRITE | CTRL_REG_D | REGD_OGS(2) | \ - REGD_IGS(2); - ctrl_regs[3] = AD_CONTROL | AD_WRITE | CTRL_REG_E | REGE_DA(0x1f); - ctrl_regs[4] = AD_CONTROL | AD_WRITE | CTRL_REG_F | REGF_SEEN ; - ctrl_regs[5] = AD_CONTROL | AD_WRITE | CTRL_REG_A | REGA_MODE_DATA; - - local_irq_disable(); - snd_ad73311_startup(); - udelay(1); - - bfin_write_SPORT_TCR1(TFSR); - bfin_write_SPORT_TCR2(0xF); - SSYNC(); - - /* SPORT Tx Register is a 8 x 16 FIFO, all the data can be put to - * FIFO before enable SPORT to transfer the data - */ - for (count = 0; count < 6; count++) - bfin_write_SPORT_TX16(ctrl_regs[count]); - SSYNC(); - bfin_write_SPORT_TCR1(bfin_read_SPORT_TCR1() | TSPEN); - SSYNC(); - - /* When TUVF is set, the data is already send out */ - while (!(status & TUVF) && ++count < 10000) { - udelay(1); - status = bfin_read_SPORT_STAT(); - SSYNC(); - } - bfin_write_SPORT_TCR1(bfin_read_SPORT_TCR1() & ~TSPEN); - SSYNC(); - local_irq_enable(); - - if (count >= 10000) { - printk(KERN_ERR "ad73311: failed to configure codec\n"); - return -1; - } - return 0; -} - -static int bf5xx_probe(struct snd_soc_card *card) -{ - int err; - if (gpio_request(GPIO_SE, "AD73311_SE")) { - printk(KERN_ERR "%s: Failed ro request GPIO_%d\n", __func__, GPIO_SE); - return -EBUSY; - } - - gpio_direction_output(GPIO_SE, 0); - - err = snd_ad73311_configure(); - if (err < 0) - return -EFAULT; - - return 0; -} - -#define BF5XX_AD7311_DAI_FMT (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF | \ - SND_SOC_DAIFMT_CBM_CFM) - -static struct snd_soc_dai_link bf5xx_ad73311_dai[] = { - { - .name = "ad73311", - .stream_name = "AD73311", - .cpu_dai_name = "bfin-i2s.0", - .codec_dai_name = "ad73311-hifi", - .platform_name = "bfin-i2s-pcm-audio", - .codec_name = "ad73311", - .dai_fmt = BF5XX_AD7311_DAI_FMT, - }, - { - .name = "ad73311", - .stream_name = "AD73311", - .cpu_dai_name = "bfin-i2s.1", - .codec_dai_name = "ad73311-hifi", - .platform_name = "bfin-i2s-pcm-audio", - .codec_name = "ad73311", - .dai_fmt = BF5XX_AD7311_DAI_FMT, - }, -}; - -static struct snd_soc_card bf5xx_ad73311 = { - .name = "bfin-ad73311", - .owner = THIS_MODULE, - .probe = bf5xx_probe, - .dai_link = &bf5xx_ad73311_dai[CONFIG_SND_BF5XX_SPORT_NUM], - .num_links = 1, -}; - -static struct platform_device *bf5xx_ad73311_snd_device; - -static int __init bf5xx_ad73311_init(void) -{ - int ret; - - pr_debug("%s enter\n", __func__); - bf5xx_ad73311_snd_device = platform_device_alloc("soc-audio", -1); - if (!bf5xx_ad73311_snd_device) - return -ENOMEM; - - platform_set_drvdata(bf5xx_ad73311_snd_device, &bf5xx_ad73311); - ret = platform_device_add(bf5xx_ad73311_snd_device); - - if (ret) - platform_device_put(bf5xx_ad73311_snd_device); - - return ret; -} - -static void __exit bf5xx_ad73311_exit(void) -{ - pr_debug("%s enter\n", __func__); - platform_device_unregister(bf5xx_ad73311_snd_device); -} - -module_init(bf5xx_ad73311_init); -module_exit(bf5xx_ad73311_exit); - -/* Module information */ -MODULE_AUTHOR("Cliff Cai"); -MODULE_DESCRIPTION("ALSA SoC AD73311 Blackfin"); -MODULE_LICENSE("GPL"); - diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c deleted file mode 100644 index 6712b01dabd1..000000000000 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ /dev/null @@ -1,373 +0,0 @@ -/* - * File: sound/soc/blackfin/bf5xx-i2s-pcm.c - * Author: Cliff Cai <Cliff.Cai(a)analog.com> - * - * Created: Tue June 06 2008 - * Description: DMA driver for i2s codec - * - * Modified: - * Copyright 2008 Analog Devices Inc. - * - * Bugs: Enter bugs at http://blackfin.uclinux.org/ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, see the file COPYING, or write - * to the Free Software Foundation, Inc., - * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA - */ - -#include <linux/module.h> -#include <linux/init.h> -#include <linux/platform_device.h> -#include <linux/dma-mapping.h> -#include <linux/gfp.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include <asm/dma.h> - -#include "bf5xx-sport.h" -#include "bf5xx-i2s-pcm.h" - -static void bf5xx_dma_irq(void *data) -{ - struct snd_pcm_substream *pcm = data; - snd_pcm_period_elapsed(pcm); -} - -static const struct snd_pcm_hardware bf5xx_pcm_hardware = { - .info = SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_BLOCK_TRANSFER, - .period_bytes_min = 32, - .period_bytes_max = 0x10000, - .periods_min = 1, - .periods_max = PAGE_SIZE/32, - .buffer_bytes_max = 0x20000, /* 128 kbytes */ - .fifo_size = 16, -}; - -static int bf5xx_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - unsigned int buffer_size = params_buffer_bytes(params); - struct bf5xx_i2s_pcm_data *dma_data; - - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - if (dma_data->tdm_mode) - buffer_size = buffer_size / params_channels(params) * 8; - - return snd_pcm_lib_malloc_pages(substream, buffer_size); -} - -static int bf5xx_pcm_hw_free(struct snd_pcm_substream *substream) -{ - snd_pcm_lib_free_pages(substream); - - return 0; -} - -static int bf5xx_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - struct sport_device *sport = runtime->private_data; - int period_bytes = frames_to_bytes(runtime, runtime->period_size); - struct bf5xx_i2s_pcm_data *dma_data; - - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - if (dma_data->tdm_mode) - period_bytes = period_bytes / runtime->channels * 8; - - pr_debug("%s enter\n", __func__); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - sport_set_tx_callback(sport, bf5xx_dma_irq, substream); - sport_config_tx_dma(sport, runtime->dma_area, - runtime->periods, period_bytes); - } else { - sport_set_rx_callback(sport, bf5xx_dma_irq, substream); - sport_config_rx_dma(sport, runtime->dma_area, - runtime->periods, period_bytes); - } - - return 0; -} - -static int bf5xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct sport_device *sport = runtime->private_data; - int ret = 0; - - pr_debug("%s enter\n", __func__); - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - sport_tx_start(sport); - else - sport_rx_start(sport); - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - sport_tx_stop(sport); - else - sport_rx_stop(sport); - break; - default: - ret = -EINVAL; - } - - return ret; -} - -static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - struct sport_device *sport = runtime->private_data; - unsigned int diff; - snd_pcm_uframes_t frames; - struct bf5xx_i2s_pcm_data *dma_data; - - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - pr_debug("%s enter\n", __func__); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - diff = sport_curr_offset_tx(sport); - } else { - diff = sport_curr_offset_rx(sport); - } - - /* - * TX at least can report one frame beyond the end of the - * buffer if we hit the wraparound case - clamp to within the - * buffer as the ALSA APIs require. - */ - if (diff == snd_pcm_lib_buffer_bytes(substream)) - diff = 0; - - frames = bytes_to_frames(substream->runtime, diff); - if (dma_data->tdm_mode) - frames = frames * runtime->channels / 8; - - return frames; -} - -static int bf5xx_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct sport_device *sport_handle = snd_soc_dai_get_drvdata(cpu_dai); - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_dma_buffer *buf = &substream->dma_buffer; - struct bf5xx_i2s_pcm_data *dma_data; - int ret; - - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - pr_debug("%s enter\n", __func__); - - snd_soc_set_runtime_hwparams(substream, &bf5xx_pcm_hardware); - if (dma_data->tdm_mode) - runtime->hw.buffer_bytes_max /= 4; - else - runtime->hw.info |= SNDRV_PCM_INFO_MMAP; - - ret = snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) - goto out; - - if (sport_handle != NULL) { - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - sport_handle->tx_buf = buf->area; - else - sport_handle->rx_buf = buf->area; - - runtime->private_data = sport_handle; - } else { - pr_err("sport_handle is NULL\n"); - return -1; - } - return 0; - - out: - return ret; -} - -static int bf5xx_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - size_t size = vma->vm_end - vma->vm_start; - vma->vm_start = (unsigned long)runtime->dma_area; - vma->vm_end = vma->vm_start + size; - vma->vm_flags |= VM_SHARED; - - return 0 ; -} - -static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, - int channel, unsigned long pos, - void *buf, unsigned long count) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int sample_size = runtime->sample_bits / 8; - struct bf5xx_i2s_pcm_data *dma_data; - unsigned int i; - void *src, *dst; - - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - if (dma_data->tdm_mode) { - pos = bytes_to_frames(runtime, pos); - count = bytes_to_frames(runtime, count); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - src = buf; - dst = runtime->dma_area; - dst += pos * sample_size * 8; - - while (count--) { - for (i = 0; i < runtime->channels; i++) { - memcpy(dst + dma_data->map[i] * - sample_size, src, sample_size); - src += sample_size; - } - dst += 8 * sample_size; - } - } else { - src = runtime->dma_area; - src += pos * sample_size * 8; - dst = buf; - - while (count--) { - for (i = 0; i < runtime->channels; i++) { - memcpy(dst, src + dma_data->map[i] * - sample_size, sample_size); - dst += sample_size; - } - src += 8 * sample_size; - } - } - } else { - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - src = buf; - dst = runtime->dma_area; - dst += pos; - } else { - src = runtime->dma_area; - src += pos; - dst = buf; - } - - memcpy(dst, src, count); - } - - return 0; -} - -static int bf5xx_pcm_copy_user(struct snd_pcm_substream *substream, - int channel, unsigned long pos, - void __user *buf, unsigned long count) -{ - return bf5xx_pcm_copy(substream, channel, pos, (void *)buf, count); -} - -static int bf5xx_pcm_silence(struct snd_pcm_substream *substream, - int channel, unsigned long pos, - unsigned long count) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int sample_size = runtime->sample_bits / 8; - void *buf = runtime->dma_area; - struct bf5xx_i2s_pcm_data *dma_data; - unsigned int offset, samples; - - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - if (dma_data->tdm_mode) { - offset = bytes_to_frames(runtime, pos) * 8 * sample_size; - samples = bytes_to_frames(runtime, count) * 8; - } else { - offset = pos; - samples = bytes_to_samples(runtime, count); - } - - snd_pcm_format_set_silence(runtime->format, buf + offset, samples); - - return 0; -} - -static const struct snd_pcm_ops bf5xx_pcm_i2s_ops = { - .open = bf5xx_pcm_open, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = bf5xx_pcm_hw_params, - .hw_free = bf5xx_pcm_hw_free, - .prepare = bf5xx_pcm_prepare, - .trigger = bf5xx_pcm_trigger, - .pointer = bf5xx_pcm_pointer, - .mmap = bf5xx_pcm_mmap, - .copy_user = bf5xx_pcm_copy_user, - .copy_kernel = bf5xx_pcm_copy, - .fill_silence = bf5xx_pcm_silence, -}; - -static int bf5xx_pcm_i2s_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - size_t size = bf5xx_pcm_hardware.buffer_bytes_max; - int ret; - - pr_debug("%s enter\n", __func__); - ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); - if (ret) - return ret; - - return snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, - SNDRV_DMA_TYPE_DEV, card->dev, size, size); -} - -static struct snd_soc_component_driver bf5xx_i2s_soc_component = { - .ops = &bf5xx_pcm_i2s_ops, - .pcm_new = bf5xx_pcm_i2s_new, -}; - -static int bfin_i2s_soc_platform_probe(struct platform_device *pdev) -{ - return devm_snd_soc_register_component(&pdev->dev, - &bf5xx_i2s_soc_component, NULL, 0); -} - -static struct platform_driver bfin_i2s_pcm_driver = { - .driver = { - .name = "bfin-i2s-pcm-audio", - }, - - .probe = bfin_i2s_soc_platform_probe, -}; - -module_platform_driver(bfin_i2s_pcm_driver); - -MODULE_AUTHOR("Cliff Cai"); -MODULE_DESCRIPTION("ADI Blackfin I2S PCM DMA module"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.h b/sound/soc/blackfin/bf5xx-i2s-pcm.h deleted file mode 100644 index 1f0435249f88..000000000000 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.h +++ /dev/null @@ -1,17 +0,0 @@ -/* - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _BF5XX_TDM_PCM_H -#define _BF5XX_TDM_PCM_H - -#define BFIN_TDM_DAI_MAX_SLOTS 8 - -struct bf5xx_i2s_pcm_data { - unsigned int map[BFIN_TDM_DAI_MAX_SLOTS]; - bool tdm_mode; -}; - -#endif diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c deleted file mode 100644 index b69aeef6418e..000000000000 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ /dev/null @@ -1,391 +0,0 @@ -/* - * File: sound/soc/blackfin/bf5xx-i2s.c - * Author: Cliff Cai <Cliff.Cai(a)analog.com> - * - * Created: Tue June 06 2008 - * Description: Blackfin I2S CPU DAI driver - * - * Modified: - * Copyright 2008 Analog Devices Inc. - * - * Bugs: Enter bugs at http://blackfin.uclinux.org/ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, see the file COPYING, or write - * to the Free Software Foundation, Inc., - * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA - */ - -#include <linux/init.h> -#include <linux/module.h> -#include <linux/device.h> -#include <linux/delay.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/initval.h> -#include <sound/soc.h> - -#include <asm/irq.h> -#include <asm/portmux.h> -#include <linux/mutex.h> -#include <linux/gpio.h> - -#include "bf5xx-sport.h" -#include "bf5xx-i2s-pcm.h" - -struct bf5xx_i2s_port { - u16 tcr1; - u16 rcr1; - u16 tcr2; - u16 rcr2; - int configured; - - unsigned int slots; - unsigned int tx_mask; - unsigned int rx_mask; - - struct bf5xx_i2s_pcm_data tx_dma_data; - struct bf5xx_i2s_pcm_data rx_dma_data; -}; - -static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) -{ - struct sport_device *sport_handle = snd_soc_dai_get_drvdata(cpu_dai); - struct bf5xx_i2s_port *bf5xx_i2s = sport_handle->private_data; - int ret = 0; - - /* interface format:support I2S,slave mode */ - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: - bf5xx_i2s->tcr1 |= TFSR | TCKFE; - bf5xx_i2s->rcr1 |= RFSR | RCKFE; - bf5xx_i2s->tcr2 |= TSFSE; - bf5xx_i2s->rcr2 |= RSFSE; - break; - case SND_SOC_DAIFMT_DSP_A: - bf5xx_i2s->tcr1 |= TFSR; - bf5xx_i2s->rcr1 |= RFSR; - break; - case SND_SOC_DAIFMT_LEFT_J: - ret = -EINVAL; - break; - default: - dev_err(cpu_dai->dev, "%s: Unknown DAI format type\n", - __func__); - ret = -EINVAL; - break; - } - - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - break; - case SND_SOC_DAIFMT_CBS_CFS: - case SND_SOC_DAIFMT_CBM_CFS: - case SND_SOC_DAIFMT_CBS_CFM: - ret = -EINVAL; - break; - default: - dev_err(cpu_dai->dev, "%s: Unknown DAI master type\n", - __func__); - ret = -EINVAL; - break; - } - - return ret; -} - -static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct sport_device *sport_handle = snd_soc_dai_get_drvdata(dai); - struct bf5xx_i2s_port *bf5xx_i2s = sport_handle->private_data; - int ret = 0; - - bf5xx_i2s->tcr2 &= ~0x1f; - bf5xx_i2s->rcr2 &= ~0x1f; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - bf5xx_i2s->tcr2 |= 7; - bf5xx_i2s->rcr2 |= 7; - sport_handle->wdsize = 1; - break; - case SNDRV_PCM_FORMAT_S16_LE: - bf5xx_i2s->tcr2 |= 15; - bf5xx_i2s->rcr2 |= 15; - sport_handle->wdsize = 2; - break; - case SNDRV_PCM_FORMAT_S24_LE: - bf5xx_i2s->tcr2 |= 23; - bf5xx_i2s->rcr2 |= 23; - sport_handle->wdsize = 3; - break; - case SNDRV_PCM_FORMAT_S32_LE: - bf5xx_i2s->tcr2 |= 31; - bf5xx_i2s->rcr2 |= 31; - sport_handle->wdsize = 4; - break; - } - - if (!bf5xx_i2s->configured) { - /* - * TX and RX are not independent,they are enabled at the - * same time, even if only one side is running. So, we - * need to configure both of them at the time when the first - * stream is opened. - * - * CPU DAI:slave mode. - */ - bf5xx_i2s->configured = 1; - ret = sport_config_rx(sport_handle, bf5xx_i2s->rcr1, - bf5xx_i2s->rcr2, 0, 0); - if (ret) { - dev_err(dai->dev, "SPORT is busy!\n"); - return -EBUSY; - } - - ret = sport_config_tx(sport_handle, bf5xx_i2s->tcr1, - bf5xx_i2s->tcr2, 0, 0); - if (ret) { - dev_err(dai->dev, "SPORT is busy!\n"); - return -EBUSY; - } - } - - return 0; -} - -static void bf5xx_i2s_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct sport_device *sport_handle = snd_soc_dai_get_drvdata(dai); - struct bf5xx_i2s_port *bf5xx_i2s = sport_handle->private_data; - - dev_dbg(dai->dev, "%s enter\n", __func__); - /* No active stream, SPORT is allowed to be configured again. */ - if (!dai->active) - bf5xx_i2s->configured = 0; -} - -static int bf5xx_i2s_set_channel_map(struct snd_soc_dai *dai, - unsigned int tx_num, unsigned int *tx_slot, - unsigned int rx_num, unsigned int *rx_slot) -{ - struct sport_device *sport_handle = snd_soc_dai_get_drvdata(dai); - struct bf5xx_i2s_port *bf5xx_i2s = sport_handle->private_data; - unsigned int tx_mapped = 0, rx_mapped = 0; - unsigned int slot; - int i; - - if ((tx_num > BFIN_TDM_DAI_MAX_SLOTS) || - (rx_num > BFIN_TDM_DAI_MAX_SLOTS)) - return -EINVAL; - - for (i = 0; i < tx_num; i++) { - slot = tx_slot[i]; - if ((slot < BFIN_TDM_DAI_MAX_SLOTS) && - (!(tx_mapped & (1 << slot)))) { - bf5xx_i2s->tx_dma_data.map[i] = slot; - tx_mapped |= 1 << slot; - } else - return -EINVAL; - } - for (i = 0; i < rx_num; i++) { - slot = rx_slot[i]; - if ((slot < BFIN_TDM_DAI_MAX_SLOTS) && - (!(rx_mapped & (1 << slot)))) { - bf5xx_i2s->rx_dma_data.map[i] = slot; - rx_mapped |= 1 << slot; - } else - return -EINVAL; - } - - return 0; -} - -static int bf5xx_i2s_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, - unsigned int rx_mask, int slots, int width) -{ - struct sport_device *sport_handle = snd_soc_dai_get_drvdata(dai); - struct bf5xx_i2s_port *bf5xx_i2s = sport_handle->private_data; - - if (slots % 8 != 0 || slots > 8) - return -EINVAL; - - if (width != 32) - return -EINVAL; - - bf5xx_i2s->slots = slots; - bf5xx_i2s->tx_mask = tx_mask; - bf5xx_i2s->rx_mask = rx_mask; - - bf5xx_i2s->tx_dma_data.tdm_mode = slots != 0; - bf5xx_i2s->rx_dma_data.tdm_mode = slots != 0; - - return sport_set_multichannel(sport_handle, slots, tx_mask, rx_mask, 0); -} - -#ifdef CONFIG_PM -static int bf5xx_i2s_suspend(struct snd_soc_dai *dai) -{ - struct sport_device *sport_handle = snd_soc_dai_get_drvdata(dai); - - dev_dbg(dai->dev, "%s : sport %d\n", __func__, dai->id); - - if (dai->capture_active) - sport_rx_stop(sport_handle); - if (dai->playback_active) - sport_tx_stop(sport_handle); - return 0; -} - -static int bf5xx_i2s_resume(struct snd_soc_dai *dai) -{ - struct sport_device *sport_handle = snd_soc_dai_get_drvdata(dai); - struct bf5xx_i2s_port *bf5xx_i2s = sport_handle->private_data; - int ret; - - dev_dbg(dai->dev, "%s : sport %d\n", __func__, dai->id); - - ret = sport_config_rx(sport_handle, bf5xx_i2s->rcr1, - bf5xx_i2s->rcr2, 0, 0); - if (ret) { - dev_err(dai->dev, "SPORT is busy!\n"); - return -EBUSY; - } - - ret = sport_config_tx(sport_handle, bf5xx_i2s->tcr1, - bf5xx_i2s->tcr2, 0, 0); - if (ret) { - dev_err(dai->dev, "SPORT is busy!\n"); - return -EBUSY; - } - - return sport_set_multichannel(sport_handle, bf5xx_i2s->slots, - bf5xx_i2s->tx_mask, bf5xx_i2s->rx_mask, 0); -} - -#else -#define bf5xx_i2s_suspend NULL -#define bf5xx_i2s_resume NULL -#endif - -static int bf5xx_i2s_dai_probe(struct snd_soc_dai *dai) -{ - struct sport_device *sport_handle = snd_soc_dai_get_drvdata(dai); - struct bf5xx_i2s_port *bf5xx_i2s = sport_handle->private_data; - unsigned int i; - - for (i = 0; i < BFIN_TDM_DAI_MAX_SLOTS; i++) { - bf5xx_i2s->tx_dma_data.map[i] = i; - bf5xx_i2s->rx_dma_data.map[i] = i; - } - - dai->playback_dma_data = &bf5xx_i2s->tx_dma_data; - dai->capture_dma_data = &bf5xx_i2s->rx_dma_data; - - return 0; -} - -#define BF5XX_I2S_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ - SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \ - SNDRV_PCM_RATE_96000) - -#define BF5XX_I2S_FORMATS \ - (SNDRV_PCM_FMTBIT_S8 | \ - SNDRV_PCM_FMTBIT_S16_LE | \ - SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S32_LE) - -static const struct snd_soc_dai_ops bf5xx_i2s_dai_ops = { - .shutdown = bf5xx_i2s_shutdown, - .hw_params = bf5xx_i2s_hw_params, - .set_fmt = bf5xx_i2s_set_dai_fmt, - .set_tdm_slot = bf5xx_i2s_set_tdm_slot, - .set_channel_map = bf5xx_i2s_set_channel_map, -}; - -static struct snd_soc_dai_driver bf5xx_i2s_dai = { - .probe = bf5xx_i2s_dai_probe, - .suspend = bf5xx_i2s_suspend, - .resume = bf5xx_i2s_resume, - .playback = { - .channels_min = 2, - .channels_max = 8, - .rates = BF5XX_I2S_RATES, - .formats = BF5XX_I2S_FORMATS,}, - .capture = { - .channels_min = 2, - .channels_max = 8, - .rates = BF5XX_I2S_RATES, - .formats = BF5XX_I2S_FORMATS,}, - .ops = &bf5xx_i2s_dai_ops, -}; - -static const struct snd_soc_component_driver bf5xx_i2s_component = { - .name = "bf5xx-i2s", -}; - -static int bf5xx_i2s_probe(struct platform_device *pdev) -{ - struct sport_device *sport_handle; - int ret; - - /* configure SPORT for I2S */ - sport_handle = sport_init(pdev, 4, 8 * sizeof(u32), - sizeof(struct bf5xx_i2s_port)); - if (!sport_handle) - return -ENODEV; - - /* register with the ASoC layers */ - ret = snd_soc_register_component(&pdev->dev, &bf5xx_i2s_component, - &bf5xx_i2s_dai, 1); - if (ret) { - dev_err(&pdev->dev, "Failed to register DAI: %d\n", ret); - sport_done(sport_handle); - return ret; - } - - return 0; -} - -static int bf5xx_i2s_remove(struct platform_device *pdev) -{ - struct sport_device *sport_handle = platform_get_drvdata(pdev); - - dev_dbg(&pdev->dev, "%s enter\n", __func__); - - snd_soc_unregister_component(&pdev->dev); - sport_done(sport_handle); - - return 0; -} - -static struct platform_driver bfin_i2s_driver = { - .probe = bf5xx_i2s_probe, - .remove = bf5xx_i2s_remove, - .driver = { - .name = "bfin-i2s", - }, -}; - -module_platform_driver(bfin_i2s_driver); - -/* Module information */ -MODULE_AUTHOR("Cliff Cai"); -MODULE_DESCRIPTION("I2S driver for ADI Blackfin"); -MODULE_LICENSE("GPL"); - diff --git a/sound/soc/blackfin/bf5xx-sport.c b/sound/soc/blackfin/bf5xx-sport.c deleted file mode 100644 index 9dfa1241ea66..000000000000 --- a/sound/soc/blackfin/bf5xx-sport.c +++ /dev/null @@ -1,1102 +0,0 @@ -/* - * File: bf5xx_sport.c - * Based on: - * Author: Roy Huang <roy.huang(a)analog.com> - * - * Created: Tue Sep 21 10:52:42 CEST 2004 - * Description: - * Blackfin SPORT Driver - * - * Copyright 2004-2007 Analog Devices Inc. - * - * Bugs: Enter bugs at http://blackfin.uclinux.org/ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, see the file COPYING, or write - * to the Free Software Foundation, Inc., - * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA - */ - -#include <linux/kernel.h> -#include <linux/slab.h> -#include <linux/delay.h> -#include <linux/dma-mapping.h> -#include <linux/gpio.h> -#include <linux/bug.h> -#include <linux/module.h> -#include <asm/portmux.h> -#include <asm/dma.h> -#include <asm/blackfin.h> -#include <asm/cacheflush.h> - -#include "bf5xx-sport.h" -/* delay between frame sync pulse and first data bit in multichannel mode */ -#define FRAME_DELAY (1<<12) - -/* note: multichannel is in units of 8 channels, - * tdm_count is # channels NOT / 8 ! */ -int sport_set_multichannel(struct sport_device *sport, - int tdm_count, u32 tx_mask, u32 rx_mask, int packed) -{ - pr_debug("%s tdm_count=%d tx_mask:0x%08x rx_mask:0x%08x packed=%d\n", - __func__, tdm_count, tx_mask, rx_mask, packed); - - if ((sport->regs->tcr1 & TSPEN) || (sport->regs->rcr1 & RSPEN)) - return -EBUSY; - - if (tdm_count & 0x7) - return -EINVAL; - - if (tdm_count > 32) - return -EINVAL; /* Only support less than 32 channels now */ - - if (tdm_count) { - sport->regs->mcmc1 = ((tdm_count>>3)-1) << 12; - sport->regs->mcmc2 = FRAME_DELAY | MCMEN | \ - (packed ? (MCDTXPE|MCDRXPE) : 0); - - sport->regs->mtcs0 = tx_mask; - sport->regs->mrcs0 = rx_mask; - sport->regs->mtcs1 = 0; - sport->regs->mrcs1 = 0; - sport->regs->mtcs2 = 0; - sport->regs->mrcs2 = 0; - sport->regs->mtcs3 = 0; - sport->regs->mrcs3 = 0; - } else { - sport->regs->mcmc1 = 0; - sport->regs->mcmc2 = 0; - - sport->regs->mtcs0 = 0; - sport->regs->mrcs0 = 0; - } - - sport->regs->mtcs1 = 0; sport->regs->mtcs2 = 0; sport->regs->mtcs3 = 0; - sport->regs->mrcs1 = 0; sport->regs->mrcs2 = 0; sport->regs->mrcs3 = 0; - - SSYNC(); - - return 0; -} -EXPORT_SYMBOL(sport_set_multichannel); - -int sport_config_rx(struct sport_device *sport, unsigned int rcr1, - unsigned int rcr2, unsigned int clkdiv, unsigned int fsdiv) -{ - if ((sport->regs->tcr1 & TSPEN) || (sport->regs->rcr1 & RSPEN)) - return -EBUSY; - - sport->regs->rcr1 = rcr1; - sport->regs->rcr2 = rcr2; - sport->regs->rclkdiv = clkdiv; - sport->regs->rfsdiv = fsdiv; - - SSYNC(); - - return 0; -} -EXPORT_SYMBOL(sport_config_rx); - -int sport_config_tx(struct sport_device *sport, unsigned int tcr1, - unsigned int tcr2, unsigned int clkdiv, unsigned int fsdiv) -{ - if ((sport->regs->tcr1 & TSPEN) || (sport->regs->rcr1 & RSPEN)) - return -EBUSY; - - sport->regs->tcr1 = tcr1; - sport->regs->tcr2 = tcr2; - sport->regs->tclkdiv = clkdiv; - sport->regs->tfsdiv = fsdiv; - - SSYNC(); - - return 0; -} -EXPORT_SYMBOL(sport_config_tx); - -static void setup_desc(struct dmasg *desc, void *buf, int fragcount, - size_t fragsize, unsigned int cfg, - unsigned int x_count, unsigned int ycount, size_t wdsize) -{ - - int i; - - for (i = 0; i < fragcount; ++i) { - desc[i].next_desc_addr = &(desc[i + 1]); - desc[i].start_addr = (unsigned long)buf + i*fragsize; - desc[i].cfg = cfg; - desc[i].x_count = x_count; - desc[i].x_modify = wdsize; - desc[i].y_count = ycount; - desc[i].y_modify = wdsize; - } - - /* make circular */ - desc[fragcount-1].next_desc_addr = desc; - - pr_debug("setup desc: desc0=%p, next0=%p, desc1=%p," - "next1=%p\nx_count=%x,y_count=%x,addr=0x%lx,cfs=0x%x\n", - desc, desc[0].next_desc_addr, - desc+1, desc[1].next_desc_addr, - desc[0].x_count, desc[0].y_count, - desc[0].start_addr, desc[0].cfg); -} - -static int sport_start(struct sport_device *sport) -{ - enable_dma(sport->dma_rx_chan); - enable_dma(sport->dma_tx_chan); - sport->regs->rcr1 |= RSPEN; - sport->regs->tcr1 |= TSPEN; - SSYNC(); - - return 0; -} - -static int sport_stop(struct sport_device *sport) -{ - sport->regs->tcr1 &= ~TSPEN; - sport->regs->rcr1 &= ~RSPEN; - SSYNC(); - - disable_dma(sport->dma_rx_chan); - disable_dma(sport->dma_tx_chan); - return 0; -} - -static inline int sport_hook_rx_dummy(struct sport_device *sport) -{ - struct dmasg *desc, temp_desc; - unsigned long flags; - - if (WARN_ON(!sport->dummy_rx_desc) || - WARN_ON(sport->curr_rx_desc == sport->dummy_rx_desc)) - return -EINVAL; - - /* Maybe the dummy buffer descriptor ring is damaged */ - sport->dummy_rx_desc->next_desc_addr = sport->dummy_rx_desc + 1; - - local_irq_save(flags); - desc = get_dma_next_desc_ptr(sport->dma_rx_chan); - /* Copy the descriptor which will be damaged to backup */ - temp_desc = *desc; - desc->x_count = sport->dummy_count / 2; - desc->y_count = 0; - desc->next_desc_addr = sport->dummy_rx_desc; - local_irq_restore(flags); - /* Waiting for dummy buffer descriptor is already hooked*/ - while ((get_dma_curr_desc_ptr(sport->dma_rx_chan) - - sizeof(struct dmasg)) != sport->dummy_rx_desc) - continue; - sport->curr_rx_desc = sport->dummy_rx_desc; - /* Restore the damaged descriptor */ - *desc = temp_desc; - - return 0; -} - -static inline int sport_rx_dma_start(struct sport_device *sport, int dummy) -{ - if (dummy) { - sport->dummy_rx_desc->next_desc_addr = sport->dummy_rx_desc; - sport->curr_rx_desc = sport->dummy_rx_desc; - } else - sport->curr_rx_desc = sport->dma_rx_desc; - - set_dma_next_desc_addr(sport->dma_rx_chan, sport->curr_rx_desc); - set_dma_x_count(sport->dma_rx_chan, 0); - set_dma_x_modify(sport->dma_rx_chan, 0); - set_dma_config(sport->dma_rx_chan, (DMAFLOW_LARGE | NDSIZE_9 | \ - WDSIZE_32 | WNR)); - set_dma_curr_addr(sport->dma_rx_chan, sport->curr_rx_desc->start_addr); - SSYNC(); - - return 0; -} - -static inline int sport_tx_dma_start(struct sport_device *sport, int dummy) -{ - if (dummy) { - sport->dummy_tx_desc->next_desc_addr = sport->dummy_tx_desc; - sport->curr_tx_desc = sport->dummy_tx_desc; - } else - sport->curr_tx_desc = sport->dma_tx_desc; - - set_dma_next_desc_addr(sport->dma_tx_chan, sport->curr_tx_desc); - set_dma_x_count(sport->dma_tx_chan, 0); - set_dma_x_modify(sport->dma_tx_chan, 0); - set_dma_config(sport->dma_tx_chan, - (DMAFLOW_LARGE | NDSIZE_9 | WDSIZE_32)); - set_dma_curr_addr(sport->dma_tx_chan, sport->curr_tx_desc->start_addr); - SSYNC(); - - return 0; -} - -int sport_rx_start(struct sport_device *sport) -{ - unsigned long flags; - pr_debug("%s enter\n", __func__); - if (sport->rx_run) - return -EBUSY; - if (sport->tx_run) { - /* tx is running, rx is not running */ - if (WARN_ON(!sport->dma_rx_desc) || - WARN_ON(sport->curr_rx_desc != sport->dummy_rx_desc)) - return -EINVAL; - local_irq_save(flags); - while ((get_dma_curr_desc_ptr(sport->dma_rx_chan) - - sizeof(struct dmasg)) != sport->dummy_rx_desc) - continue; - sport->dummy_rx_desc->next_desc_addr = sport->dma_rx_desc; - local_irq_restore(flags); - sport->curr_rx_desc = sport->dma_rx_desc; - } else { - sport_tx_dma_start(sport, 1); - sport_rx_dma_start(sport, 0); - sport_start(sport); - } - - sport->rx_run = 1; - - return 0; -} -EXPORT_SYMBOL(sport_rx_start); - -int sport_rx_stop(struct sport_device *sport) -{ - pr_debug("%s enter\n", __func__); - - if (!sport->rx_run) - return 0; - if (sport->tx_run) { - /* TX dma is still running, hook the dummy buffer */ - sport_hook_rx_dummy(sport); - } else { - /* Both rx and tx dma will be stopped */ - sport_stop(sport); - sport->curr_rx_desc = NULL; - sport->curr_tx_desc = NULL; - } - - sport->rx_run = 0; - - return 0; -} -EXPORT_SYMBOL(sport_rx_stop); - -static inline int sport_hook_tx_dummy(struct sport_device *sport) -{ - struct dmasg *desc, temp_desc; - unsigned long flags; - - if (WARN_ON(!sport->dummy_tx_desc) || - WARN_ON(sport->curr_tx_desc == sport->dummy_tx_desc)) - return -EINVAL; - - sport->dummy_tx_desc->next_desc_addr = sport->dummy_tx_desc + 1; - - /* Shorten the time on last normal descriptor */ - local_irq_save(flags); - desc = get_dma_next_desc_ptr(sport->dma_tx_chan); - /* Store the descriptor which will be damaged */ - temp_desc = *desc; - desc->x_count = sport->dummy_count / 2; - desc->y_count = 0; - desc->next_desc_addr = sport->dummy_tx_desc; - local_irq_restore(flags); - /* Waiting for dummy buffer descriptor is already hooked*/ - while ((get_dma_curr_desc_ptr(sport->dma_tx_chan) - \ - sizeof(struct dmasg)) != sport->dummy_tx_desc) - continue; - sport->curr_tx_desc = sport->dummy_tx_desc; - /* Restore the damaged descriptor */ - *desc = temp_desc; - - return 0; -} - -int sport_tx_start(struct sport_device *sport) -{ - unsigned long flags; - pr_debug("%s: tx_run:%d, rx_run:%d\n", __func__, - sport->tx_run, sport->rx_run); - if (sport->tx_run) - return -EBUSY; - if (sport->rx_run) { - if (WARN_ON(!sport->dma_tx_desc) || - WARN_ON(sport->curr_tx_desc != sport->dummy_tx_desc)) - return -EINVAL; - /* Hook the normal buffer descriptor */ - local_irq_save(flags); - while ((get_dma_curr_desc_ptr(sport->dma_tx_chan) - - sizeof(struct dmasg)) != sport->dummy_tx_desc) - continue; - sport->dummy_tx_desc->next_desc_addr = sport->dma_tx_desc; - local_irq_restore(flags); - sport->curr_tx_desc = sport->dma_tx_desc; - } else { - - sport_tx_dma_start(sport, 0); - /* Let rx dma run the dummy buffer */ - sport_rx_dma_start(sport, 1); - sport_start(sport); - } - sport->tx_run = 1; - return 0; -} -EXPORT_SYMBOL(sport_tx_start); - -int sport_tx_stop(struct sport_device *sport) -{ - if (!sport->tx_run) - return 0; - if (sport->rx_run) { - /* RX is still running, hook the dummy buffer */ - sport_hook_tx_dummy(sport); - } else { - /* Both rx and tx dma stopped */ - sport_stop(sport); - sport->curr_rx_desc = NULL; - sport->curr_tx_desc = NULL; - } - - sport->tx_run = 0; - - return 0; -} -EXPORT_SYMBOL(sport_tx_stop); - -static inline int compute_wdsize(size_t wdsize) -{ - switch (wdsize) { - case 1: - return WDSIZE_8; - case 2: - return WDSIZE_16; - case 4: - default: - return WDSIZE_32; - } -} - -int sport_config_rx_dma(struct sport_device *sport, void *buf, - int fragcount, size_t fragsize) -{ - unsigned int x_count; - unsigned int y_count; - unsigned int cfg; - dma_addr_t addr; - - pr_debug("%s buf:%p, frag:%d, fragsize:0x%lx\n", __func__, \ - buf, fragcount, fragsize); - - x_count = fragsize / sport->wdsize; - y_count = 0; - - /* for fragments larger than 64k words we use 2d dma, - * denote fragecount as two numbers' mutliply and both of them - * are less than 64k.*/ - if (x_count >= 0x10000) { - int i, count = x_count; - - for (i = 16; i > 0; i--) { - x_count = 1 << i; - if ((count & (x_count - 1)) == 0) { - y_count = count >> i; - if (y_count < 0x10000) - break; - } - } - if (i == 0) - return -EINVAL; - } - pr_debug("%s(x_count:0x%x, y_count:0x%x)\n", __func__, - x_count, y_count); - - if (sport->dma_rx_desc) - dma_free_coherent(NULL, sport->rx_desc_bytes, - sport->dma_rx_desc, 0); - - /* Allocate a new descritor ring as current one. */ - sport->dma_rx_desc = dma_alloc_coherent(NULL, \ - fragcount * sizeof(struct dmasg), &addr, 0); - sport->rx_desc_bytes = fragcount * sizeof(struct dmasg); - - if (!sport->dma_rx_desc) { - pr_err("Failed to allocate memory for rx desc\n"); - return -ENOMEM; - } - - sport->rx_buf = buf; - sport->rx_fragsize = fragsize; - sport->rx_frags = fragcount; - - cfg = 0x7000 | DI_EN | compute_wdsize(sport->wdsize) | WNR | \ - (DESC_ELEMENT_COUNT << 8); /* large descriptor mode */ - - if (y_count != 0) - cfg |= DMA2D; - - setup_desc(sport->dma_rx_desc, buf, fragcount, fragsize, - cfg|DMAEN, x_count, y_count, sport->wdsize); - - return 0; -} -EXPORT_SYMBOL(sport_config_rx_dma); - -int sport_config_tx_dma(struct sport_device *sport, void *buf, \ - int fragcount, size_t fragsize) -{ - unsigned int x_count; - unsigned int y_count; - unsigned int cfg; - dma_addr_t addr; - - pr_debug("%s buf:%p, fragcount:%d, fragsize:0x%lx\n", - __func__, buf, fragcount, fragsize); - - x_count = fragsize/sport->wdsize; - y_count = 0; - - /* for fragments larger than 64k words we use 2d dma, - * denote fragecount as two numbers' mutliply and both of them - * are less than 64k.*/ - if (x_count >= 0x10000) { - int i, count = x_count; - - for (i = 16; i > 0; i--) { - x_count = 1 << i; - if ((count & (x_count - 1)) == 0) { - y_count = count >> i; - if (y_count < 0x10000) - break; - } - } - if (i == 0) - return -EINVAL; - } - pr_debug("%s x_count:0x%x, y_count:0x%x\n", __func__, - x_count, y_count); - - - if (sport->dma_tx_desc) { - dma_free_coherent(NULL, sport->tx_desc_bytes, \ - sport->dma_tx_desc, 0); - } - - sport->dma_tx_desc = dma_alloc_coherent(NULL, \ - fragcount * sizeof(struct dmasg), &addr, 0); - sport->tx_desc_bytes = fragcount * sizeof(struct dmasg); - if (!sport->dma_tx_desc) { - pr_err("Failed to allocate memory for tx desc\n"); - return -ENOMEM; - } - - sport->tx_buf = buf; - sport->tx_fragsize = fragsize; - sport->tx_frags = fragcount; - cfg = 0x7000 | DI_EN | compute_wdsize(sport->wdsize) | \ - (DESC_ELEMENT_COUNT << 8); /* large descriptor mode */ - - if (y_count != 0) - cfg |= DMA2D; - - setup_desc(sport->dma_tx_desc, buf, fragcount, fragsize, - cfg|DMAEN, x_count, y_count, sport->wdsize); - - return 0; -} -EXPORT_SYMBOL(sport_config_tx_dma); - -/* setup dummy dma descriptor ring, which don't generate interrupts, - * the x_modify is set to 0 */ -static int sport_config_rx_dummy(struct sport_device *sport) -{ - struct dmasg *desc; - unsigned config; - - pr_debug("%s entered\n", __func__); - if (L1_DATA_A_LENGTH) - desc = l1_data_sram_zalloc(2 * sizeof(*desc)); - else { - dma_addr_t addr; - desc = dma_alloc_coherent(NULL, 2 * sizeof(*desc), &addr, 0); - memset(desc, 0, 2 * sizeof(*desc)); - } - if (desc == NULL) { - pr_err("Failed to allocate memory for dummy rx desc\n"); - return -ENOMEM; - } - sport->dummy_rx_desc = desc; - desc->start_addr = (unsigned long)sport->dummy_buf; - config = DMAFLOW_LARGE | NDSIZE_9 | compute_wdsize(sport->wdsize) - | WNR | DMAEN; - desc->cfg = config; - desc->x_count = sport->dummy_count/sport->wdsize; - desc->x_modify = sport->wdsize; - desc->y_count = 0; - desc->y_modify = 0; - memcpy(desc+1, desc, sizeof(*desc)); - desc->next_desc_addr = desc + 1; - desc[1].next_desc_addr = desc; - return 0; -} - -static int sport_config_tx_dummy(struct sport_device *sport) -{ - struct dmasg *desc; - unsigned int config; - - pr_debug("%s entered\n", __func__); - - if (L1_DATA_A_LENGTH) - desc = l1_data_sram_zalloc(2 * sizeof(*desc)); - else { - dma_addr_t addr; - desc = dma_alloc_coherent(NULL, 2 * sizeof(*desc), &addr, 0); - memset(desc, 0, 2 * sizeof(*desc)); - } - if (!desc) { - pr_err("Failed to allocate memory for dummy tx desc\n"); - return -ENOMEM; - } - sport->dummy_tx_desc = desc; - desc->start_addr = (unsigned long)sport->dummy_buf + \ - sport->dummy_count; - config = DMAFLOW_LARGE | NDSIZE_9 | - compute_wdsize(sport->wdsize) | DMAEN; - desc->cfg = config; - desc->x_count = sport->dummy_count/sport->wdsize; - desc->x_modify = sport->wdsize; - desc->y_count = 0; - desc->y_modify = 0; - memcpy(desc+1, desc, sizeof(*desc)); - desc->next_desc_addr = desc + 1; - desc[1].next_desc_addr = desc; - return 0; -} - -unsigned long sport_curr_offset_rx(struct sport_device *sport) -{ - unsigned long curr = get_dma_curr_addr(sport->dma_rx_chan); - - return (unsigned char *)curr - sport->rx_buf; -} -EXPORT_SYMBOL(sport_curr_offset_rx); - -unsigned long sport_curr_offset_tx(struct sport_device *sport) -{ - unsigned long curr = get_dma_curr_addr(sport->dma_tx_chan); - - return (unsigned char *)curr - sport->tx_buf; -} -EXPORT_SYMBOL(sport_curr_offset_tx); - -void sport_incfrag(struct sport_device *sport, int *frag, int tx) -{ - ++(*frag); - if (tx == 1 && *frag == sport->tx_frags) - *frag = 0; - - if (tx == 0 && *frag == sport->rx_frags) - *frag = 0; -} -EXPORT_SYMBOL(sport_incfrag); - -void sport_decfrag(struct sport_device *sport, int *frag, int tx) -{ - --(*frag); - if (tx == 1 && *frag == 0) - *frag = sport->tx_frags; - - if (tx == 0 && *frag == 0) - *frag = sport->rx_frags; -} -EXPORT_SYMBOL(sport_decfrag); - -static int sport_check_status(struct sport_device *sport, - unsigned int *sport_stat, - unsigned int *rx_stat, - unsigned int *tx_stat) -{ - int status = 0; - - if (sport_stat) { - SSYNC(); - status = sport->regs->stat; - if (status & (TOVF|TUVF|ROVF|RUVF)) - sport->regs->stat = (status & (TOVF|TUVF|ROVF|RUVF)); - SSYNC(); - *sport_stat = status; - } - - if (rx_stat) { - SSYNC(); - status = get_dma_curr_irqstat(sport->dma_rx_chan); - if (status & (DMA_DONE|DMA_ERR)) - clear_dma_irqstat(sport->dma_rx_chan); - SSYNC(); - *rx_stat = status; - } - - if (tx_stat) { - SSYNC(); - status = get_dma_curr_irqstat(sport->dma_tx_chan); - if (status & (DMA_DONE|DMA_ERR)) - clear_dma_irqstat(sport->dma_tx_chan); - SSYNC(); - *tx_stat = status; - } - - return 0; -} - -int sport_dump_stat(struct sport_device *sport, char *buf, size_t len) -{ - int ret; - - ret = snprintf(buf, len, - "sts: 0x%04x\n" - "rx dma %d sts: 0x%04x tx dma %d sts: 0x%04x\n", - sport->regs->stat, - sport->dma_rx_chan, - get_dma_curr_irqstat(sport->dma_rx_chan), - sport->dma_tx_chan, - get_dma_curr_irqstat(sport->dma_tx_chan)); - buf += ret; - len -= ret; - - ret += snprintf(buf, len, - "curr_rx_desc:0x%p, curr_tx_desc:0x%p\n" - "dma_rx_desc:0x%p, dma_tx_desc:0x%p\n" - "dummy_rx_desc:0x%p, dummy_tx_desc:0x%p\n", - sport->curr_rx_desc, sport->curr_tx_desc, - sport->dma_rx_desc, sport->dma_tx_desc, - sport->dummy_rx_desc, sport->dummy_tx_desc); - - return ret; -} - -static irqreturn_t rx_handler(int irq, void *dev_id) -{ - unsigned int rx_stat; - struct sport_device *sport = dev_id; - - pr_debug("%s enter\n", __func__); - sport_check_status(sport, NULL, &rx_stat, NULL); - if (!(rx_stat & DMA_DONE)) - pr_err("rx dma is already stopped\n"); - - if (sport->rx_callback) { - sport->rx_callback(sport->rx_data); - return IRQ_HANDLED; - } - - return IRQ_NONE; -} - -static irqreturn_t tx_handler(int irq, void *dev_id) -{ - unsigned int tx_stat; - struct sport_device *sport = dev_id; - pr_debug("%s enter\n", __func__); - sport_check_status(sport, NULL, NULL, &tx_stat); - if (!(tx_stat & DMA_DONE)) { - pr_err("tx dma is already stopped\n"); - return IRQ_HANDLED; - } - if (sport->tx_callback) { - sport->tx_callback(sport->tx_data); - return IRQ_HANDLED; - } - - return IRQ_NONE; -} - -static irqreturn_t err_handler(int irq, void *dev_id) -{ - unsigned int status = 0; - struct sport_device *sport = dev_id; - - pr_debug("%s\n", __func__); - if (sport_check_status(sport, &status, NULL, NULL)) { - pr_err("error checking status ??"); - return IRQ_NONE; - } - - if (status & (TOVF|TUVF|ROVF|RUVF)) { - pr_info("sport status error:%s%s%s%s\n", - status & TOVF ? " TOVF" : "", - status & TUVF ? " TUVF" : "", - status & ROVF ? " ROVF" : "", - status & RUVF ? " RUVF" : ""); - if (status & TOVF || status & TUVF) { - disable_dma(sport->dma_tx_chan); - if (sport->tx_run) - sport_tx_dma_start(sport, 0); - else - sport_tx_dma_start(sport, 1); - enable_dma(sport->dma_tx_chan); - } else { - disable_dma(sport->dma_rx_chan); - if (sport->rx_run) - sport_rx_dma_start(sport, 0); - else - sport_rx_dma_start(sport, 1); - enable_dma(sport->dma_rx_chan); - } - } - status = sport->regs->stat; - if (status & (TOVF|TUVF|ROVF|RUVF)) - sport->regs->stat = (status & (TOVF|TUVF|ROVF|RUVF)); - SSYNC(); - - if (sport->err_callback) - sport->err_callback(sport->err_data); - - return IRQ_HANDLED; -} - -int sport_set_rx_callback(struct sport_device *sport, - void (*rx_callback)(void *), void *rx_data) -{ - if (WARN_ON(!rx_callback)) - return -EINVAL; - sport->rx_callback = rx_callback; - sport->rx_data = rx_data; - - return 0; -} -EXPORT_SYMBOL(sport_set_rx_callback); - -int sport_set_tx_callback(struct sport_device *sport, - void (*tx_callback)(void *), void *tx_data) -{ - if (WARN_ON(!tx_callback)) - return -EINVAL; - sport->tx_callback = tx_callback; - sport->tx_data = tx_data; - - return 0; -} -EXPORT_SYMBOL(sport_set_tx_callback); - -int sport_set_err_callback(struct sport_device *sport, - void (*err_callback)(void *), void *err_data) -{ - if (WARN_ON(!err_callback)) - return -EINVAL; - sport->err_callback = err_callback; - sport->err_data = err_data; - - return 0; -} -EXPORT_SYMBOL(sport_set_err_callback); - -static int sport_config_pdev(struct platform_device *pdev, struct sport_param *param) -{ - /* Extract settings from platform data */ - struct device *dev = &pdev->dev; - struct bfin_snd_platform_data *pdata = dev->platform_data; - struct resource *res; - - param->num = pdev->id; - - if (!pdata) { - dev_err(dev, "no platform_data\n"); - return -ENODEV; - } - param->pin_req = pdata->pin_req; - - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - dev_err(dev, "no MEM resource\n"); - return -ENODEV; - } - param->regs = (struct sport_register *)res->start; - - /* first RX, then TX */ - res = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!res) { - dev_err(dev, "no rx DMA resource\n"); - return -ENODEV; - } - param->dma_rx_chan = res->start; - - res = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!res) { - dev_err(dev, "no tx DMA resource\n"); - return -ENODEV; - } - param->dma_tx_chan = res->start; - - res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); - if (!res) { - dev_err(dev, "no irq resource\n"); - return -ENODEV; - } - param->err_irq = res->start; - - return 0; -} - -struct sport_device *sport_init(struct platform_device *pdev, - unsigned int wdsize, unsigned int dummy_count, size_t priv_size) -{ - struct device *dev = &pdev->dev; - struct sport_param param; - struct sport_device *sport; - int ret; - - dev_dbg(dev, "%s enter\n", __func__); - - param.wdsize = wdsize; - param.dummy_count = dummy_count; - if (WARN_ON(param.wdsize == 0 || param.dummy_count == 0)) - return NULL; - - ret = sport_config_pdev(pdev, &param); - if (ret) - return NULL; - - if (peripheral_request_list(param.pin_req, "soc-audio")) { - dev_err(dev, "requesting Peripherals failed\n"); - return NULL; - } - - sport = kzalloc(sizeof(*sport), GFP_KERNEL); - if (!sport) { - dev_err(dev, "failed to allocate for sport device\n"); - goto __init_err0; - } - - sport->num = param.num; - sport->dma_rx_chan = param.dma_rx_chan; - sport->dma_tx_chan = param.dma_tx_chan; - sport->err_irq = param.err_irq; - sport->regs = param.regs; - sport->pin_req = param.pin_req; - - if (request_dma(sport->dma_rx_chan, "SPORT RX Data") == -EBUSY) { - dev_err(dev, "failed to request RX dma %d\n", sport->dma_rx_chan); - goto __init_err1; - } - if (set_dma_callback(sport->dma_rx_chan, rx_handler, sport) != 0) { - dev_err(dev, "failed to request RX irq %d\n", sport->dma_rx_chan); - goto __init_err2; - } - - if (request_dma(sport->dma_tx_chan, "SPORT TX Data") == -EBUSY) { - dev_err(dev, "failed to request TX dma %d\n", sport->dma_tx_chan); - goto __init_err2; - } - - if (set_dma_callback(sport->dma_tx_chan, tx_handler, sport) != 0) { - dev_err(dev, "failed to request TX irq %d\n", sport->dma_tx_chan); - goto __init_err3; - } - - if (request_irq(sport->err_irq, err_handler, IRQF_SHARED, "SPORT err", - sport) < 0) { - dev_err(dev, "failed to request err irq %d\n", sport->err_irq); - goto __init_err3; - } - - dev_info(dev, "dma rx:%d tx:%d, err irq:%d, regs:%p\n", - sport->dma_rx_chan, sport->dma_tx_chan, - sport->err_irq, sport->regs); - - sport->wdsize = param.wdsize; - sport->dummy_count = param.dummy_count; - - sport->private_data = kzalloc(priv_size, GFP_KERNEL); - if (!sport->private_data) { - dev_err(dev, "could not alloc priv data %zu bytes\n", priv_size); - goto __init_err4; - } - - if (L1_DATA_A_LENGTH) - sport->dummy_buf = l1_data_sram_zalloc(param.dummy_count * 2); - else - sport->dummy_buf = kzalloc(param.dummy_count * 2, GFP_KERNEL); - if (sport->dummy_buf == NULL) { - dev_err(dev, "failed to allocate dummy buffer\n"); - goto __error1; - } - - ret = sport_config_rx_dummy(sport); - if (ret) { - dev_err(dev, "failed to config rx dummy ring\n"); - goto __error2; - } - ret = sport_config_tx_dummy(sport); - if (ret) { - dev_err(dev, "failed to config tx dummy ring\n"); - goto __error3; - } - - platform_set_drvdata(pdev, sport); - - return sport; -__error3: - if (L1_DATA_A_LENGTH) - l1_data_sram_free(sport->dummy_rx_desc); - else - dma_free_coherent(NULL, 2*sizeof(struct dmasg), - sport->dummy_rx_desc, 0); -__error2: - if (L1_DATA_A_LENGTH) - l1_data_sram_free(sport->dummy_buf); - else - kfree(sport->dummy_buf); -__error1: - kfree(sport->private_data); -__init_err4: - free_irq(sport->err_irq, sport); -__init_err3: - free_dma(sport->dma_tx_chan); -__init_err2: - free_dma(sport->dma_rx_chan); -__init_err1: - kfree(sport); -__init_err0: - peripheral_free_list(param.pin_req); - return NULL; -} -EXPORT_SYMBOL(sport_init); - -void sport_done(struct sport_device *sport) -{ - if (sport == NULL) - return; - - sport_stop(sport); - if (sport->dma_rx_desc) - dma_free_coherent(NULL, sport->rx_desc_bytes, - sport->dma_rx_desc, 0); - if (sport->dma_tx_desc) - dma_free_coherent(NULL, sport->tx_desc_bytes, - sport->dma_tx_desc, 0); - -#if L1_DATA_A_LENGTH != 0 - l1_data_sram_free(sport->dummy_rx_desc); - l1_data_sram_free(sport->dummy_tx_desc); - l1_data_sram_free(sport->dummy_buf); -#else - dma_free_coherent(NULL, 2*sizeof(struct dmasg), - sport->dummy_rx_desc, 0); - dma_free_coherent(NULL, 2*sizeof(struct dmasg), - sport->dummy_tx_desc, 0); - kfree(sport->dummy_buf); -#endif - free_dma(sport->dma_rx_chan); - free_dma(sport->dma_tx_chan); - free_irq(sport->err_irq, sport); - - kfree(sport->private_data); - peripheral_free_list(sport->pin_req); - kfree(sport); -} -EXPORT_SYMBOL(sport_done); - -/* -* It is only used to send several bytes when dma is not enabled - * sport controller is configured but not enabled. - * Multichannel cannot works with pio mode */ -/* Used by ac97 to write and read codec register */ -int sport_send_and_recv(struct sport_device *sport, u8 *out_data, \ - u8 *in_data, int len) -{ - unsigned short dma_config; - unsigned short status; - unsigned long flags; - unsigned long wait = 0; - - pr_debug("%s enter, out_data:%p, in_data:%p len:%d\n", \ - __func__, out_data, in_data, len); - pr_debug("tcr1:0x%04x, tcr2:0x%04x, tclkdiv:0x%04x, tfsdiv:0x%04x\n" - "mcmc1:0x%04x, mcmc2:0x%04x\n", - sport->regs->tcr1, sport->regs->tcr2, - sport->regs->tclkdiv, sport->regs->tfsdiv, - sport->regs->mcmc1, sport->regs->mcmc2); - flush_dcache_range((unsigned)out_data, (unsigned)(out_data + len)); - - /* Enable tx dma */ - dma_config = (RESTART | WDSIZE_16 | DI_EN); - set_dma_start_addr(sport->dma_tx_chan, (unsigned long)out_data); - set_dma_x_count(sport->dma_tx_chan, len/2); - set_dma_x_modify(sport->dma_tx_chan, 2); - set_dma_config(sport->dma_tx_chan, dma_config); - enable_dma(sport->dma_tx_chan); - - if (in_data != NULL) { - invalidate_dcache_range((unsigned)in_data, \ - (unsigned)(in_data + len)); - /* Enable rx dma */ - dma_config = (RESTART | WDSIZE_16 | WNR | DI_EN); - set_dma_start_addr(sport->dma_rx_chan, (unsigned long)in_data); - set_dma_x_count(sport->dma_rx_chan, len/2); - set_dma_x_modify(sport->dma_rx_chan, 2); - set_dma_config(sport->dma_rx_chan, dma_config); - enable_dma(sport->dma_rx_chan); - } - - local_irq_save(flags); - sport->regs->tcr1 |= TSPEN; - sport->regs->rcr1 |= RSPEN; - SSYNC(); - - status = get_dma_curr_irqstat(sport->dma_tx_chan); - while (status & DMA_RUN) { - udelay(1); - status = get_dma_curr_irqstat(sport->dma_tx_chan); - pr_debug("DMA status:0x%04x\n", status); - if (wait++ > 100) - goto __over; - } - status = sport->regs->stat; - wait = 0; - - while (!(status & TXHRE)) { - pr_debug("sport status:0x%04x\n", status); - udelay(1); - status = *(unsigned short *)&sport->regs->stat; - if (wait++ > 1000) - goto __over; - } - /* Wait for the last byte sent out */ - udelay(20); - pr_debug("sport status:0x%04x\n", status); - -__over: - sport->regs->tcr1 &= ~TSPEN; - sport->regs->rcr1 &= ~RSPEN; - SSYNC(); - disable_dma(sport->dma_tx_chan); - /* Clear the status */ - clear_dma_irqstat(sport->dma_tx_chan); - if (in_data != NULL) { - disable_dma(sport->dma_rx_chan); - clear_dma_irqstat(sport->dma_rx_chan); - } - SSYNC(); - local_irq_restore(flags); - - return 0; -} -EXPORT_SYMBOL(sport_send_and_recv); - -MODULE_AUTHOR("Roy Huang"); -MODULE_DESCRIPTION("SPORT driver for ADI Blackfin"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/blackfin/bf5xx-sport.h b/sound/soc/blackfin/bf5xx-sport.h deleted file mode 100644 index 9fc2192feb3b..000000000000 --- a/sound/soc/blackfin/bf5xx-sport.h +++ /dev/null @@ -1,174 +0,0 @@ -/* - * File: bf5xx_sport.h - * Based on: - * Author: Roy Huang <roy.huang(a)analog.com> - * - * Created: - * Description: - * - * Copyright 2004-2007 Analog Devices Inc. - * - * Bugs: Enter bugs at http://blackfin.uclinux.org/ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, see the file COPYING, or write - * to the Free Software Foundation, Inc., - * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA - */ - - -#ifndef __BF5XX_SPORT_H__ -#define __BF5XX_SPORT_H__ - -#include <linux/types.h> -#include <linux/wait.h> -#include <linux/workqueue.h> -#include <linux/platform_device.h> -#include <asm/dma.h> -#include <asm/bfin_sport.h> - -#define DESC_ELEMENT_COUNT 9 - -struct sport_device { - int num; - int dma_rx_chan; - int dma_tx_chan; - int err_irq; - const unsigned short *pin_req; - struct sport_register *regs; - - unsigned char *rx_buf; - unsigned char *tx_buf; - unsigned int rx_fragsize; - unsigned int tx_fragsize; - unsigned int rx_frags; - unsigned int tx_frags; - unsigned int wdsize; - - /* for dummy dma transfer */ - void *dummy_buf; - unsigned int dummy_count; - - /* DMA descriptor ring head of current audio stream*/ - struct dmasg *dma_rx_desc; - struct dmasg *dma_tx_desc; - unsigned int rx_desc_bytes; - unsigned int tx_desc_bytes; - - unsigned int rx_run:1; /* rx is running */ - unsigned int tx_run:1; /* tx is running */ - - struct dmasg *dummy_rx_desc; - struct dmasg *dummy_tx_desc; - - struct dmasg *curr_rx_desc; - struct dmasg *curr_tx_desc; - - int rx_curr_frag; - int tx_curr_frag; - - unsigned int rcr1; - unsigned int rcr2; - int rx_tdm_count; - - unsigned int tcr1; - unsigned int tcr2; - int tx_tdm_count; - - void (*rx_callback)(void *data); - void *rx_data; - void (*tx_callback)(void *data); - void *tx_data; - void (*err_callback)(void *data); - void *err_data; - unsigned char *tx_dma_buf; - unsigned char *rx_dma_buf; -#ifdef CONFIG_SND_BF5XX_MMAP_SUPPORT - dma_addr_t tx_dma_phy; - dma_addr_t rx_dma_phy; - int tx_pos;/*pcm sample count*/ - int rx_pos; - unsigned int tx_buffer_size; - unsigned int rx_buffer_size; - int tx_delay_pos; - int once; -#endif - void *private_data; -}; - -struct sport_param { - int num; - int dma_rx_chan; - int dma_tx_chan; - int err_irq; - const unsigned short *pin_req; - struct sport_register *regs; - unsigned int wdsize; - unsigned int dummy_count; - void *private_data; -}; - -struct sport_device *sport_init(struct platform_device *pdev, - unsigned int wdsize, unsigned int dummy_count, size_t priv_size); - -void sport_done(struct sport_device *sport); - -/* first use these ...*/ - -/* note: multichannel is in units of 8 channels, tdm_count is number of channels - * NOT / 8 ! all channels are enabled by default */ -int sport_set_multichannel(struct sport_device *sport, int tdm_count, - u32 tx_mask, u32 rx_mask, int packed); - -int sport_config_rx(struct sport_device *sport, - unsigned int rcr1, unsigned int rcr2, - unsigned int clkdiv, unsigned int fsdiv); - -int sport_config_tx(struct sport_device *sport, - unsigned int tcr1, unsigned int tcr2, - unsigned int clkdiv, unsigned int fsdiv); - -/* ... then these: */ - -/* buffer size (in bytes) == fragcount * fragsize_bytes */ - -/* this is not a very general api, it sets the dma to 2d autobuffer mode */ - -int sport_config_rx_dma(struct sport_device *sport, void *buf, - int fragcount, size_t fragsize_bytes); - -int sport_config_tx_dma(struct sport_device *sport, void *buf, - int fragcount, size_t fragsize_bytes); - -int sport_tx_start(struct sport_device *sport); -int sport_tx_stop(struct sport_device *sport); -int sport_rx_start(struct sport_device *sport); -int sport_rx_stop(struct sport_device *sport); - -/* for use in interrupt handler */ -unsigned long sport_curr_offset_rx(struct sport_device *sport); -unsigned long sport_curr_offset_tx(struct sport_device *sport); - -void sport_incfrag(struct sport_device *sport, int *frag, int tx); -void sport_decfrag(struct sport_device *sport, int *frag, int tx); - -int sport_set_rx_callback(struct sport_device *sport, - void (*rx_callback)(void *), void *rx_data); -int sport_set_tx_callback(struct sport_device *sport, - void (*tx_callback)(void *), void *tx_data); -int sport_set_err_callback(struct sport_device *sport, - void (*err_callback)(void *), void *err_data); - -int sport_send_and_recv(struct sport_device *sport, u8 *out_data, \ - u8 *in_data, int len); -#endif /* BF53X_SPORT_H */ diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c deleted file mode 100644 index 9c19ccc936e2..000000000000 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ /dev/null @@ -1,126 +0,0 @@ -/* - * File: sound/soc/blackfin/bf5xx-ssm2602.c - * Author: Cliff Cai <Cliff.Cai(a)analog.com> - * - * Created: Tue June 06 2008 - * Description: board driver for SSM2602 sound chip - * - * Modified: - * Copyright 2008 Analog Devices Inc. - * - * Bugs: Enter bugs at http://blackfin.uclinux.org/ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, see the file COPYING, or write - * to the Free Software Foundation, Inc., - * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/device.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/pcm_params.h> - -#include <asm/dma.h> -#include <asm/portmux.h> -#include <linux/gpio.h> -#include "../codecs/ssm2602.h" -#include "bf5xx-sport.h" - -static struct snd_soc_card bf5xx_ssm2602; - -static int bf5xx_ssm2602_dai_init(struct snd_soc_pcm_runtime *rtd) -{ - /* - * If you are using a crystal source which frequency is not 12MHz - * then modify the below case statement with frequency of the crystal. - * - * If you are using the SPORT to generate clocking then this is - * where to do it. - */ - return snd_soc_dai_set_sysclk(rtd->codec_dai, SSM2602_SYSCLK, 12000000, - SND_SOC_CLOCK_IN); -} - -/* CODEC is master for BCLK and LRC in this configuration. */ -#define BF5XX_SSM2602_DAIFMT (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | \ - SND_SOC_DAIFMT_CBM_CFM) - -static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = { - { - .name = "ssm2602", - .stream_name = "SSM2602", - .cpu_dai_name = "bfin-i2s.0", - .codec_dai_name = "ssm2602-hifi", - .platform_name = "bfin-i2s-pcm-audio", - .codec_name = "ssm2602.0-001b", - .init = bf5xx_ssm2602_dai_init, - .dai_fmt = BF5XX_SSM2602_DAIFMT, - }, - { - .name = "ssm2602", - .stream_name = "SSM2602", - .cpu_dai_name = "bfin-i2s.1", - .codec_dai_name = "ssm2602-hifi", - .platform_name = "bfin-i2s-pcm-audio", - .codec_name = "ssm2602.0-001b", - .init = bf5xx_ssm2602_dai_init, - .dai_fmt = BF5XX_SSM2602_DAIFMT, - }, -}; - -static struct snd_soc_card bf5xx_ssm2602 = { - .name = "bfin-ssm2602", - .owner = THIS_MODULE, - .dai_link = &bf5xx_ssm2602_dai[CONFIG_SND_BF5XX_SPORT_NUM], - .num_links = 1, -}; - -static struct platform_device *bf5xx_ssm2602_snd_device; - -static int __init bf5xx_ssm2602_init(void) -{ - int ret; - - pr_debug("%s enter\n", __func__); - bf5xx_ssm2602_snd_device = platform_device_alloc("soc-audio", -1); - if (!bf5xx_ssm2602_snd_device) - return -ENOMEM; - - platform_set_drvdata(bf5xx_ssm2602_snd_device, &bf5xx_ssm2602); - ret = platform_device_add(bf5xx_ssm2602_snd_device); - - if (ret) - platform_device_put(bf5xx_ssm2602_snd_device); - - return ret; -} - -static void __exit bf5xx_ssm2602_exit(void) -{ - pr_debug("%s enter\n", __func__); - platform_device_unregister(bf5xx_ssm2602_snd_device); -} - -module_init(bf5xx_ssm2602_init); -module_exit(bf5xx_ssm2602_exit); - -/* Module information */ -MODULE_AUTHOR("Cliff Cai"); -MODULE_DESCRIPTION("ALSA SoC SSM2602 BF527-EZKIT"); -MODULE_LICENSE("GPL"); - diff --git a/sound/soc/blackfin/bf6xx-i2s.c b/sound/soc/blackfin/bf6xx-i2s.c deleted file mode 100644 index 819cff149a25..000000000000 --- a/sound/soc/blackfin/bf6xx-i2s.c +++ /dev/null @@ -1,239 +0,0 @@ -/* - * bf6xx-i2s.c - Analog Devices BF6XX i2s interface driver - * - * Copyright (c) 2012 Analog Devices Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. - */ - -#include <linux/device.h> -#include <linux/init.h> -#include <linux/module.h> -#include <linux/platform_device.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> -#include <sound/soc-dai.h> - -#include "bf6xx-sport.h" - -struct sport_params param; - -static int bfin_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) -{ - struct sport_device *sport = snd_soc_dai_get_drvdata(cpu_dai); - struct device *dev = &sport->pdev->dev; - int ret = 0; - - param.spctl &= ~(SPORT_CTL_OPMODE | SPORT_CTL_CKRE | SPORT_CTL_FSR - | SPORT_CTL_LFS | SPORT_CTL_LAFS); - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: - param.spctl |= SPORT_CTL_OPMODE | SPORT_CTL_CKRE - | SPORT_CTL_LFS; - break; - case SND_SOC_DAIFMT_DSP_A: - param.spctl |= SPORT_CTL_FSR; - break; - case SND_SOC_DAIFMT_LEFT_J: - param.spctl |= SPORT_CTL_OPMODE | SPORT_CTL_LFS - | SPORT_CTL_LAFS; - break; - default: - dev_err(dev, "%s: Unknown DAI format type\n", __func__); - ret = -EINVAL; - break; - } - - param.spctl &= ~(SPORT_CTL_ICLK | SPORT_CTL_IFS); - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - break; - case SND_SOC_DAIFMT_CBS_CFS: - case SND_SOC_DAIFMT_CBM_CFS: - case SND_SOC_DAIFMT_CBS_CFM: - ret = -EINVAL; - break; - default: - dev_err(dev, "%s: Unknown DAI master type\n", __func__); - ret = -EINVAL; - break; - } - - return ret; -} - -static int bfin_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct sport_device *sport = snd_soc_dai_get_drvdata(dai); - struct device *dev = &sport->pdev->dev; - int ret = 0; - - param.spctl &= ~SPORT_CTL_SLEN; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - param.spctl |= 0x70; - sport->wdsize = 1; - break; - case SNDRV_PCM_FORMAT_S16_LE: - param.spctl |= 0xf0; - sport->wdsize = 2; - break; - case SNDRV_PCM_FORMAT_S24_LE: - param.spctl |= 0x170; - sport->wdsize = 3; - break; - case SNDRV_PCM_FORMAT_S32_LE: - param.spctl |= 0x1f0; - sport->wdsize = 4; - break; - } - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - ret = sport_set_tx_params(sport, &param); - if (ret) { - dev_err(dev, "SPORT tx is busy!\n"); - return ret; - } - } else { - ret = sport_set_rx_params(sport, &param); - if (ret) { - dev_err(dev, "SPORT rx is busy!\n"); - return ret; - } - } - return 0; -} - -#ifdef CONFIG_PM -static int bfin_i2s_suspend(struct snd_soc_dai *dai) -{ - struct sport_device *sport = snd_soc_dai_get_drvdata(dai); - - if (dai->capture_active) - sport_rx_stop(sport); - if (dai->playback_active) - sport_tx_stop(sport); - return 0; -} - -static int bfin_i2s_resume(struct snd_soc_dai *dai) -{ - struct sport_device *sport = snd_soc_dai_get_drvdata(dai); - struct device *dev = &sport->pdev->dev; - int ret; - - ret = sport_set_tx_params(sport, &param); - if (ret) { - dev_err(dev, "SPORT tx is busy!\n"); - return ret; - } - ret = sport_set_rx_params(sport, &param); - if (ret) { - dev_err(dev, "SPORT rx is busy!\n"); - return ret; - } - - return 0; -} - -#else -#define bfin_i2s_suspend NULL -#define bfin_i2s_resume NULL -#endif - -#define BFIN_I2S_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ - SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \ - SNDRV_PCM_RATE_96000) - -#define BFIN_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ - SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) - -static const struct snd_soc_dai_ops bfin_i2s_dai_ops = { - .hw_params = bfin_i2s_hw_params, - .set_fmt = bfin_i2s_set_dai_fmt, -}; - -static struct snd_soc_dai_driver bfin_i2s_dai = { - .suspend = bfin_i2s_suspend, - .resume = bfin_i2s_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = BFIN_I2S_RATES, - .formats = BFIN_I2S_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = BFIN_I2S_RATES, - .formats = BFIN_I2S_FORMATS, - }, - .ops = &bfin_i2s_dai_ops, -}; - -static const struct snd_soc_component_driver bfin_i2s_component = { - .name = "bfin-i2s", -}; - -static int bfin_i2s_probe(struct platform_device *pdev) -{ - struct sport_device *sport; - struct device *dev = &pdev->dev; - int ret; - - sport = sport_create(pdev); - if (!sport) - return -ENODEV; - - /* register with the ASoC layers */ - ret = snd_soc_register_component(dev, &bfin_i2s_component, - &bfin_i2s_dai, 1); - if (ret) { - dev_err(dev, "Failed to register DAI: %d\n", ret); - sport_delete(sport); - return ret; - } - platform_set_drvdata(pdev, sport); - - return 0; -} - -static int bfin_i2s_remove(struct platform_device *pdev) -{ - struct sport_device *sport = platform_get_drvdata(pdev); - - snd_soc_unregister_component(&pdev->dev); - sport_delete(sport); - - return 0; -} - -static struct platform_driver bfin_i2s_driver = { - .probe = bfin_i2s_probe, - .remove = bfin_i2s_remove, - .driver = { - .name = "bfin-i2s", - }, -}; - -module_platform_driver(bfin_i2s_driver); - -MODULE_DESCRIPTION("Analog Devices BF6XX i2s interface driver"); -MODULE_AUTHOR("Scott Jiang <Scott.Jiang.Linux(a)gmail.com>"); -MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/blackfin/bf6xx-sport.c b/sound/soc/blackfin/bf6xx-sport.c deleted file mode 100644 index d2caadfe7b6d..000000000000 --- a/sound/soc/blackfin/bf6xx-sport.c +++ /dev/null @@ -1,425 +0,0 @@ -/* - * bf6xx_sport.c Analog Devices BF6XX SPORT driver - * - * Copyright (c) 2012 Analog Devices Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. - */ - -#include <linux/device.h> -#include <linux/dma-mapping.h> -#include <linux/interrupt.h> -#include <linux/module.h> -#include <linux/platform_device.h> -#include <linux/slab.h> - -#include <asm/blackfin.h> -#include <asm/dma.h> -#include <asm/portmux.h> - -#include "bf6xx-sport.h" - -int sport_set_tx_params(struct sport_device *sport, - struct sport_params *params) -{ - if (sport->tx_regs->spctl & SPORT_CTL_SPENPRI) - return -EBUSY; - sport->tx_regs->spctl = params->spctl | SPORT_CTL_SPTRAN; - sport->tx_regs->div = params->div; - SSYNC(); - return 0; -} -EXPORT_SYMBOL(sport_set_tx_params); - -int sport_set_rx_params(struct sport_device *sport, - struct sport_params *params) -{ - if (sport->rx_regs->spctl & SPORT_CTL_SPENPRI) - return -EBUSY; - sport->rx_regs->spctl = params->spctl & ~SPORT_CTL_SPTRAN; - sport->rx_regs->div = params->div; - SSYNC(); - return 0; -} -EXPORT_SYMBOL(sport_set_rx_params); - -static int compute_wdsize(size_t wdsize) -{ - switch (wdsize) { - case 1: - return WDSIZE_8 | PSIZE_8; - case 2: - return WDSIZE_16 | PSIZE_16; - default: - return WDSIZE_32 | PSIZE_32; - } -} - -void sport_tx_start(struct sport_device *sport) -{ - set_dma_next_desc_addr(sport->tx_dma_chan, sport->tx_desc); - set_dma_config(sport->tx_dma_chan, DMAFLOW_LIST | DI_EN - | compute_wdsize(sport->wdsize) | NDSIZE_6); - enable_dma(sport->tx_dma_chan); - sport->tx_regs->spctl |= SPORT_CTL_SPENPRI; - SSYNC(); -} -EXPORT_SYMBOL(sport_tx_start); - -void sport_rx_start(struct sport_device *sport) -{ - set_dma_next_desc_addr(sport->rx_dma_chan, sport->rx_desc); - set_dma_config(sport->rx_dma_chan, DMAFLOW_LIST | DI_EN | WNR - | compute_wdsize(sport->wdsize) | NDSIZE_6); - enable_dma(sport->rx_dma_chan); - sport->rx_regs->spctl |= SPORT_CTL_SPENPRI; - SSYNC(); -} -EXPORT_SYMBOL(sport_rx_start); - -void sport_tx_stop(struct sport_device *sport) -{ - sport->tx_regs->spctl &= ~SPORT_CTL_SPENPRI; - SSYNC(); - disable_dma(sport->tx_dma_chan); -} -EXPORT_SYMBOL(sport_tx_stop); - -void sport_rx_stop(struct sport_device *sport) -{ - sport->rx_regs->spctl &= ~SPORT_CTL_SPENPRI; - SSYNC(); - disable_dma(sport->rx_dma_chan); -} -EXPORT_SYMBOL(sport_rx_stop); - -void sport_set_tx_callback(struct sport_device *sport, - void (*tx_callback)(void *), void *tx_data) -{ - sport->tx_callback = tx_callback; - sport->tx_data = tx_data; -} -EXPORT_SYMBOL(sport_set_tx_callback); - -void sport_set_rx_callback(struct sport_device *sport, - void (*rx_callback)(void *), void *rx_data) -{ - sport->rx_callback = rx_callback; - sport->rx_data = rx_data; -} -EXPORT_SYMBOL(sport_set_rx_callback); - -static void setup_desc(struct dmasg *desc, void *buf, int fragcount, - size_t fragsize, unsigned int cfg, - unsigned int count, size_t wdsize) -{ - - int i; - - for (i = 0; i < fragcount; ++i) { - desc[i].next_desc_addr = &(desc[i + 1]); - desc[i].start_addr = (unsigned long)buf + i * fragsize; - desc[i].cfg = cfg; - desc[i].x_count = count; - desc[i].x_modify = wdsize; - desc[i].y_count = 0; - desc[i].y_modify = 0; - } - - /* make circular */ - desc[fragcount - 1].next_desc_addr = desc; -} - -int sport_config_tx_dma(struct sport_device *sport, void *buf, - int fragcount, size_t fragsize) -{ - unsigned int count; - unsigned int cfg; - dma_addr_t addr; - - count = fragsize / sport->wdsize; - - if (sport->tx_desc) - dma_free_coherent(NULL, sport->tx_desc_size, - sport->tx_desc, 0); - - sport->tx_desc = dma_alloc_coherent(NULL, - fragcount * sizeof(struct dmasg), &addr, 0); - sport->tx_desc_size = fragcount * sizeof(struct dmasg); - if (!sport->tx_desc) - return -ENOMEM; - - sport->tx_buf = buf; - sport->tx_fragsize = fragsize; - sport->tx_frags = fragcount; - cfg = DMAFLOW_LIST | DI_EN | compute_wdsize(sport->wdsize) | NDSIZE_6; - - setup_desc(sport->tx_desc, buf, fragcount, fragsize, - cfg | DMAEN, count, sport->wdsize); - return 0; -} -EXPORT_SYMBOL(sport_config_tx_dma); - -int sport_config_rx_dma(struct sport_device *sport, void *buf, - int fragcount, size_t fragsize) -{ - unsigned int count; - unsigned int cfg; - dma_addr_t addr; - - count = fragsize / sport->wdsize; - - if (sport->rx_desc) - dma_free_coherent(NULL, sport->rx_desc_size, - sport->rx_desc, 0); - - sport->rx_desc = dma_alloc_coherent(NULL, - fragcount * sizeof(struct dmasg), &addr, 0); - sport->rx_desc_size = fragcount * sizeof(struct dmasg); - if (!sport->rx_desc) - return -ENOMEM; - - sport->rx_buf = buf; - sport->rx_fragsize = fragsize; - sport->rx_frags = fragcount; - cfg = DMAFLOW_LIST | DI_EN | compute_wdsize(sport->wdsize) - | WNR | NDSIZE_6; - - setup_desc(sport->rx_desc, buf, fragcount, fragsize, - cfg | DMAEN, count, sport->wdsize); - return 0; -} -EXPORT_SYMBOL(sport_config_rx_dma); - -unsigned long sport_curr_offset_tx(struct sport_device *sport) -{ - unsigned long curr = get_dma_curr_addr(sport->tx_dma_chan); - - return (unsigned char *)curr - sport->tx_buf; -} -EXPORT_SYMBOL(sport_curr_offset_tx); - -unsigned long sport_curr_offset_rx(struct sport_device *sport) -{ - unsigned long curr = get_dma_curr_addr(sport->rx_dma_chan); - - return (unsigned char *)curr - sport->rx_buf; -} -EXPORT_SYMBOL(sport_curr_offset_rx); - -static irqreturn_t sport_tx_irq(int irq, void *dev_id) -{ - struct sport_device *sport = dev_id; - static unsigned long status; - - status = get_dma_curr_irqstat(sport->tx_dma_chan); - if (status & (DMA_DONE | DMA_ERR)) { - clear_dma_irqstat(sport->tx_dma_chan); - SSYNC(); - } - if (sport->tx_callback) - sport->tx_callback(sport->tx_data); - return IRQ_HANDLED; -} - -static irqreturn_t sport_rx_irq(int irq, void *dev_id) -{ - struct sport_device *sport = dev_id; - unsigned long status; - - status = get_dma_curr_irqstat(sport->rx_dma_chan); - if (status & (DMA_DONE | DMA_ERR)) { - clear_dma_irqstat(sport->rx_dma_chan); - SSYNC(); - } - if (sport->rx_callback) - sport->rx_callback(sport->rx_data); - return IRQ_HANDLED; -} - -static irqreturn_t sport_err_irq(int irq, void *dev_id) -{ - struct sport_device *sport = dev_id; - struct device *dev = &sport->pdev->dev; - - if (sport->tx_regs->spctl & SPORT_CTL_DERRPRI) - dev_err(dev, "sport error: TUVF\n"); - if (sport->rx_regs->spctl & SPORT_CTL_DERRPRI) - dev_err(dev, "sport error: ROVF\n"); - - return IRQ_HANDLED; -} - -static int sport_get_resource(struct sport_device *sport) -{ - struct platform_device *pdev = sport->pdev; - struct device *dev = &pdev->dev; - struct bfin_snd_platform_data *pdata = dev->platform_data; - struct resource *res; - - if (!pdata) { - dev_err(dev, "No platform data\n"); - return -ENODEV; - } - sport->pin_req = pdata->pin_req; - - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - dev_err(dev, "No tx MEM resource\n"); - return -ENODEV; - } - sport->tx_regs = (struct sport_register *)res->start; - - res = platform_get_resource(pdev, IORESOURCE_MEM, 1); - if (!res) { - dev_err(dev, "No rx MEM resource\n"); - return -ENODEV; - } - sport->rx_regs = (struct sport_register *)res->start; - - res = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!res) { - dev_err(dev, "No tx DMA resource\n"); - return -ENODEV; - } - sport->tx_dma_chan = res->start; - - res = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!res) { - dev_err(dev, "No rx DMA resource\n"); - return -ENODEV; - } - sport->rx_dma_chan = res->start; - - res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); - if (!res) { - dev_err(dev, "No tx error irq resource\n"); - return -ENODEV; - } - sport->tx_err_irq = res->start; - - res = platform_get_resource(pdev, IORESOURCE_IRQ, 1); - if (!res) { - dev_err(dev, "No rx error irq resource\n"); - return -ENODEV; - } - sport->rx_err_irq = res->start; - - return 0; -} - -static int sport_request_resource(struct sport_device *sport) -{ - struct device *dev = &sport->pdev->dev; - int ret; - - ret = peripheral_request_list(sport->pin_req, "soc-audio"); - if (ret) { - dev_err(dev, "Unable to request sport pin\n"); - return ret; - } - - ret = request_dma(sport->tx_dma_chan, "SPORT TX Data"); - if (ret) { - dev_err(dev, "Unable to allocate DMA channel for sport tx\n"); - goto err_tx_dma; - } - set_dma_callback(sport->tx_dma_chan, sport_tx_irq, sport); - - ret = request_dma(sport->rx_dma_chan, "SPORT RX Data"); - if (ret) { - dev_err(dev, "Unable to allocate DMA channel for sport rx\n"); - goto err_rx_dma; - } - set_dma_callback(sport->rx_dma_chan, sport_rx_irq, sport); - - ret = request_irq(sport->tx_err_irq, sport_err_irq, - 0, "SPORT TX ERROR", sport); - if (ret) { - dev_err(dev, "Unable to allocate tx error IRQ for sport\n"); - goto err_tx_irq; - } - - ret = request_irq(sport->rx_err_irq, sport_err_irq, - 0, "SPORT RX ERROR", sport); - if (ret) { - dev_err(dev, "Unable to allocate rx error IRQ for sport\n"); - goto err_rx_irq; - } - - return 0; -err_rx_irq: - free_irq(sport->tx_err_irq, sport); -err_tx_irq: - free_dma(sport->rx_dma_chan); -err_rx_dma: - free_dma(sport->tx_dma_chan); -err_tx_dma: - peripheral_free_list(sport->pin_req); - return ret; -} - -static void sport_free_resource(struct sport_device *sport) -{ - free_irq(sport->rx_err_irq, sport); - free_irq(sport->tx_err_irq, sport); - free_dma(sport->rx_dma_chan); - free_dma(sport->tx_dma_chan); - peripheral_free_list(sport->pin_req); -} - -struct sport_device *sport_create(struct platform_device *pdev) -{ - struct device *dev = &pdev->dev; - struct sport_device *sport; - int ret; - - sport = kzalloc(sizeof(*sport), GFP_KERNEL); - if (!sport) - return NULL; - - sport->pdev = pdev; - - ret = sport_get_resource(sport); - if (ret) - goto free_data; - - ret = sport_request_resource(sport); - if (ret) - goto free_data; - - dev_dbg(dev, "SPORT create success\n"); - return sport; -free_data: - kfree(sport); - return NULL; -} -EXPORT_SYMBOL(sport_create); - -void sport_delete(struct sport_device *sport) -{ - if (sport->tx_desc) - dma_free_coherent(NULL, sport->tx_desc_size, - sport->tx_desc, 0); - if (sport->rx_desc) - dma_free_coherent(NULL, sport->rx_desc_size, - sport->rx_desc, 0); - sport_free_resource(sport); - kfree(sport); -} -EXPORT_SYMBOL(sport_delete); - -MODULE_DESCRIPTION("Analog Devices BF6XX SPORT driver"); -MODULE_AUTHOR("Scott Jiang <Scott.Jiang.Linux(a)gmail.com>"); -MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/blackfin/bf6xx-sport.h b/sound/soc/blackfin/bf6xx-sport.h deleted file mode 100644 index 307d193cfcef..000000000000 --- a/sound/soc/blackfin/bf6xx-sport.h +++ /dev/null @@ -1,82 +0,0 @@ -/* - * bf6xx_sport - Analog Devices BF6XX SPORT driver - * - * Copyright (c) 2012 Analog Devices Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. - */ - -#ifndef _BF6XX_SPORT_H_ -#define _BF6XX_SPORT_H_ - -#include <linux/platform_device.h> -#include <asm/bfin_sport3.h> - -struct sport_device { - struct platform_device *pdev; - const unsigned short *pin_req; - struct sport_register *tx_regs; - struct sport_register *rx_regs; - int tx_dma_chan; - int rx_dma_chan; - int tx_err_irq; - int rx_err_irq; - - void (*tx_callback)(void *data); - void *tx_data; - void (*rx_callback)(void *data); - void *rx_data; - - struct dmasg *tx_desc; - struct dmasg *rx_desc; - unsigned int tx_desc_size; - unsigned int rx_desc_size; - unsigned char *tx_buf; - unsigned char *rx_buf; - unsigned int tx_fragsize; - unsigned int rx_fragsize; - unsigned int tx_frags; - unsigned int rx_frags; - unsigned int wdsize; -}; - -struct sport_params { - u32 spctl; - u32 div; -}; - -struct sport_device *sport_create(struct platform_device *pdev); -void sport_delete(struct sport_device *sport); -int sport_set_tx_params(struct sport_device *sport, - struct sport_params *params); -int sport_set_rx_params(struct sport_device *sport, - struct sport_params *params); -void sport_tx_start(struct sport_device *sport); -void sport_rx_start(struct sport_device *sport); -void sport_tx_stop(struct sport_device *sport); -void sport_rx_stop(struct sport_device *sport); -void sport_set_tx_callback(struct sport_device *sport, - void (*tx_callback)(void *), void *tx_data); -void sport_set_rx_callback(struct sport_device *sport, - void (*rx_callback)(void *), void *rx_data); -int sport_config_tx_dma(struct sport_device *sport, void *buf, - int fragcount, size_t fragsize); -int sport_config_rx_dma(struct sport_device *sport, void *buf, - int fragcount, size_t fragsize); -unsigned long sport_curr_offset_tx(struct sport_device *sport); -unsigned long sport_curr_offset_rx(struct sport_device *sport); - - - -#endif diff --git a/sound/soc/blackfin/bfin-eval-adau1373.c b/sound/soc/blackfin/bfin-eval-adau1373.c deleted file mode 100644 index 64b88fdc1f6c..000000000000 --- a/sound/soc/blackfin/bfin-eval-adau1373.c +++ /dev/null @@ -1,173 +0,0 @@ -/* - * Machine driver for EVAL-ADAU1373 on Analog Devices bfin - * evaluation boards. - * - * Copyright 2011 Analog Devices Inc. - * Author: Lars-Peter Clausen <lars(a)metafoo.de> - * - * Licensed under the GPL-2 or later. - */ - -#include <linux/module.h> -#include <linux/device.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/pcm_params.h> - -#include "../codecs/adau1373.h" - -static const struct snd_soc_dapm_widget bfin_eval_adau1373_dapm_widgets[] = { - SND_SOC_DAPM_LINE("Line In1", NULL), - SND_SOC_DAPM_LINE("Line In2", NULL), - SND_SOC_DAPM_LINE("Line In3", NULL), - SND_SOC_DAPM_LINE("Line In4", NULL), - - SND_SOC_DAPM_LINE("Line Out1", NULL), - SND_SOC_DAPM_LINE("Line Out2", NULL), - SND_SOC_DAPM_LINE("Stereo Out", NULL), - SND_SOC_DAPM_HP("Headphone", NULL), - SND_SOC_DAPM_HP("Earpiece", NULL), - SND_SOC_DAPM_SPK("Speaker", NULL), -}; - -static const struct snd_soc_dapm_route bfin_eval_adau1373_dapm_routes[] = { - { "AIN1L", NULL, "Line In1" }, - { "AIN1R", NULL, "Line In1" }, - { "AIN2L", NULL, "Line In2" }, - { "AIN2R", NULL, "Line In2" }, - { "AIN3L", NULL, "Line In3" }, - { "AIN3R", NULL, "Line In3" }, - { "AIN4L", NULL, "Line In4" }, - { "AIN4R", NULL, "Line In4" }, - - /* MICBIAS can be connected via a jumper to the line-in jack, since w - don't know which one is going to be used, just power both. */ - { "Line In1", NULL, "MICBIAS1" }, - { "Line In2", NULL, "MICBIAS1" }, - { "Line In3", NULL, "MICBIAS1" }, - { "Line In4", NULL, "MICBIAS1" }, - { "Line In1", NULL, "MICBIAS2" }, - { "Line In2", NULL, "MICBIAS2" }, - { "Line In3", NULL, "MICBIAS2" }, - { "Line In4", NULL, "MICBIAS2" }, - - { "Line Out1", NULL, "LOUT1L" }, - { "Line Out1", NULL, "LOUT1R" }, - { "Line Out2", NULL, "LOUT2L" }, - { "Line Out2", NULL, "LOUT2R" }, - { "Headphone", NULL, "HPL" }, - { "Headphone", NULL, "HPR" }, - { "Earpiece", NULL, "EP" }, - { "Speaker", NULL, "SPKL" }, - { "Stereo Out", NULL, "SPKR" }, -}; - -static int bfin_eval_adau1373_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - int pll_rate; - - switch (params_rate(params)) { - case 48000: - case 8000: - case 12000: - case 16000: - case 24000: - case 32000: - pll_rate = 48000 * 1024; - break; - case 44100: - case 7350: - case 11025: - case 14700: - case 22050: - case 29400: - pll_rate = 44100 * 1024; - break; - default: - return -EINVAL; - } - - ret = snd_soc_dai_set_pll(codec_dai, ADAU1373_PLL1, - ADAU1373_PLL_SRC_MCLK1, 12288000, pll_rate); - if (ret) - return ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, ADAU1373_CLK_SRC_PLL1, pll_rate, - SND_SOC_CLOCK_IN); - - return ret; -} - -static int bfin_eval_adau1373_codec_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_dai *codec_dai = rtd->codec_dai; - unsigned int pll_rate = 48000 * 1024; - int ret; - - ret = snd_soc_dai_set_pll(codec_dai, ADAU1373_PLL1, - ADAU1373_PLL_SRC_MCLK1, 12288000, pll_rate); - if (ret) - return ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, ADAU1373_CLK_SRC_PLL1, pll_rate, - SND_SOC_CLOCK_IN); - - return ret; -} -static const struct snd_soc_ops bfin_eval_adau1373_ops = { - .hw_params = bfin_eval_adau1373_hw_params, -}; - -static struct snd_soc_dai_link bfin_eval_adau1373_dai = { - .name = "adau1373", - .stream_name = "adau1373", - .cpu_dai_name = "bfin-i2s.0", - .codec_dai_name = "adau1373-aif1", - .platform_name = "bfin-i2s-pcm-audio", - .codec_name = "adau1373.0-001a", - .ops = &bfin_eval_adau1373_ops, - .init = bfin_eval_adau1373_codec_init, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, -}; - -static struct snd_soc_card bfin_eval_adau1373 = { - .name = "bfin-eval-adau1373", - .owner = THIS_MODULE, - .dai_link = &bfin_eval_adau1373_dai, - .num_links = 1, - - .dapm_widgets = bfin_eval_adau1373_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(bfin_eval_adau1373_dapm_widgets), - .dapm_routes = bfin_eval_adau1373_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(bfin_eval_adau1373_dapm_routes), -}; - -static int bfin_eval_adau1373_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &bfin_eval_adau1373; - - card->dev = &pdev->dev; - - return devm_snd_soc_register_card(&pdev->dev, &bfin_eval_adau1373); -} - -static struct platform_driver bfin_eval_adau1373_driver = { - .driver = { - .name = "bfin-eval-adau1373", - .pm = &snd_soc_pm_ops, - }, - .probe = bfin_eval_adau1373_probe, -}; - -module_platform_driver(bfin_eval_adau1373_driver); - -MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); -MODULE_DESCRIPTION("ALSA SoC bfin adau1373 driver"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:bfin-eval-adau1373"); diff --git a/sound/soc/blackfin/bfin-eval-adau1701.c b/sound/soc/blackfin/bfin-eval-adau1701.c deleted file mode 100644 index 5c67f72cf9a9..000000000000 --- a/sound/soc/blackfin/bfin-eval-adau1701.c +++ /dev/null @@ -1,113 +0,0 @@ -/* - * Machine driver for EVAL-ADAU1701MINIZ on Analog Devices bfin - * evaluation boards. - * - * Copyright 2011 Analog Devices Inc. - * Author: Lars-Peter Clausen <lars(a)metafoo.de> - * - * Licensed under the GPL-2 or later. - */ - -#include <linux/module.h> -#include <linux/device.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/pcm_params.h> - -#include "../codecs/adau1701.h" - -static const struct snd_soc_dapm_widget bfin_eval_adau1701_dapm_widgets[] = { - SND_SOC_DAPM_SPK("Speaker", NULL), - SND_SOC_DAPM_LINE("Line Out", NULL), - SND_SOC_DAPM_LINE("Line In", NULL), -}; - -static const struct snd_soc_dapm_route bfin_eval_adau1701_dapm_routes[] = { - { "Speaker", NULL, "OUT0" }, - { "Speaker", NULL, "OUT1" }, - { "Line Out", NULL, "OUT2" }, - { "Line Out", NULL, "OUT3" }, - - { "IN0", NULL, "Line In" }, - { "IN1", NULL, "Line In" }, -}; - -static int bfin_eval_adau1701_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, ADAU1701_CLK_SRC_OSC, 12288000, - SND_SOC_CLOCK_IN); - - return ret; -} - -static struct snd_soc_ops bfin_eval_adau1701_ops = { - .hw_params = bfin_eval_adau1701_hw_params, -}; - -#define BFIN_EVAL_ADAU1701_DAI_FMT (SND_SOC_DAIFMT_I2S | \ - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM) - -static struct snd_soc_dai_link bfin_eval_adau1701_dai[] = { - { - .name = "adau1701", - .stream_name = "adau1701", - .cpu_dai_name = "bfin-i2s.0", - .codec_dai_name = "adau1701", - .platform_name = "bfin-i2s-pcm-audio", - .codec_name = "adau1701.0-0034", - .ops = &bfin_eval_adau1701_ops, - .dai_fmt = BFIN_EVAL_ADAU1701_DAI_FMT, - }, - { - .name = "adau1701", - .stream_name = "adau1701", - .cpu_dai_name = "bfin-i2s.1", - .codec_dai_name = "adau1701", - .platform_name = "bfin-i2s-pcm-audio", - .codec_name = "adau1701.0-0034", - .ops = &bfin_eval_adau1701_ops, - .dai_fmt = BFIN_EVAL_ADAU1701_DAI_FMT, - }, -}; - -static struct snd_soc_card bfin_eval_adau1701 = { - .name = "bfin-eval-adau1701", - .owner = THIS_MODULE, - .dai_link = &bfin_eval_adau1701_dai[CONFIG_SND_BF5XX_SPORT_NUM], - .num_links = 1, - - .dapm_widgets = bfin_eval_adau1701_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(bfin_eval_adau1701_dapm_widgets), - .dapm_routes = bfin_eval_adau1701_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(bfin_eval_adau1701_dapm_routes), -}; - -static int bfin_eval_adau1701_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &bfin_eval_adau1701; - - card->dev = &pdev->dev; - - return devm_snd_soc_register_card(&pdev->dev, &bfin_eval_adau1701); -} - -static struct platform_driver bfin_eval_adau1701_driver = { - .driver = { - .name = "bfin-eval-adau1701", - .pm = &snd_soc_pm_ops, - }, - .probe = bfin_eval_adau1701_probe, -}; - -module_platform_driver(bfin_eval_adau1701_driver); - -MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); -MODULE_DESCRIPTION("ALSA SoC bfin ADAU1701 driver"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:bfin-eval-adau1701"); diff --git a/sound/soc/blackfin/bfin-eval-adau1x61.c b/sound/soc/blackfin/bfin-eval-adau1x61.c deleted file mode 100644 index fddfe00c9d69..000000000000 --- a/sound/soc/blackfin/bfin-eval-adau1x61.c +++ /dev/null @@ -1,142 +0,0 @@ -/* - * Machine driver for EVAL-ADAU1x61MINIZ on Analog Devices bfin - * evaluation boards. - * - * Copyright 2011-2014 Analog Devices Inc. - * Author: Lars-Peter Clausen <lars(a)metafoo.de> - * - * Licensed under the GPL-2 or later. - */ - -#include <linux/module.h> -#include <linux/device.h> -#include <linux/slab.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/pcm_params.h> - -#include "../codecs/adau17x1.h" - -static const struct snd_soc_dapm_widget bfin_eval_adau1x61_dapm_widgets[] = { - SND_SOC_DAPM_LINE("In 1", NULL), - SND_SOC_DAPM_LINE("In 2", NULL), - SND_SOC_DAPM_LINE("In 3-4", NULL), - - SND_SOC_DAPM_LINE("Diff Out L", NULL), - SND_SOC_DAPM_LINE("Diff Out R", NULL), - SND_SOC_DAPM_LINE("Stereo Out", NULL), - SND_SOC_DAPM_HP("Capless HP Out", NULL), -}; - -static const struct snd_soc_dapm_route bfin_eval_adau1x61_dapm_routes[] = { - { "LAUX", NULL, "In 3-4" }, - { "RAUX", NULL, "In 3-4" }, - { "LINP", NULL, "In 1" }, - { "LINN", NULL, "In 1"}, - { "RINP", NULL, "In 2" }, - { "RINN", NULL, "In 2" }, - - { "In 1", NULL, "MICBIAS" }, - { "In 2", NULL, "MICBIAS" }, - - { "Capless HP Out", NULL, "LHP" }, - { "Capless HP Out", NULL, "RHP" }, - { "Diff Out L", NULL, "LOUT" }, - { "Diff Out R", NULL, "ROUT" }, - { "Stereo Out", NULL, "LOUT" }, - { "Stereo Out", NULL, "ROUT" }, -}; - -static int bfin_eval_adau1x61_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int pll_rate; - int ret; - - switch (params_rate(params)) { - case 48000: - case 8000: - case 12000: - case 16000: - case 24000: - case 32000: - case 96000: - pll_rate = 48000 * 1024; - break; - case 44100: - case 7350: - case 11025: - case 14700: - case 22050: - case 29400: - case 88200: - pll_rate = 44100 * 1024; - break; - default: - return -EINVAL; - } - - ret = snd_soc_dai_set_pll(codec_dai, ADAU17X1_PLL, - ADAU17X1_PLL_SRC_MCLK, 12288000, pll_rate); - if (ret) - return ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, ADAU17X1_CLK_SRC_PLL, pll_rate, - SND_SOC_CLOCK_IN); - - return ret; -} - -static const struct snd_soc_ops bfin_eval_adau1x61_ops = { - .hw_params = bfin_eval_adau1x61_hw_params, -}; - -static struct snd_soc_dai_link bfin_eval_adau1x61_dai = { - .name = "adau1x61", - .stream_name = "adau1x61", - .cpu_dai_name = "bfin-i2s.0", - .codec_dai_name = "adau-hifi", - .platform_name = "bfin-i2s-pcm-audio", - .codec_name = "adau1761.0-0038", - .ops = &bfin_eval_adau1x61_ops, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, -}; - -static struct snd_soc_card bfin_eval_adau1x61 = { - .name = "bfin-eval-adau1x61", - .owner = THIS_MODULE, - .driver_name = "eval-adau1x61", - .dai_link = &bfin_eval_adau1x61_dai, - .num_links = 1, - - .dapm_widgets = bfin_eval_adau1x61_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(bfin_eval_adau1x61_dapm_widgets), - .dapm_routes = bfin_eval_adau1x61_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(bfin_eval_adau1x61_dapm_routes), - .fully_routed = true, -}; - -static int bfin_eval_adau1x61_probe(struct platform_device *pdev) -{ - bfin_eval_adau1x61.dev = &pdev->dev; - - return devm_snd_soc_register_card(&pdev->dev, &bfin_eval_adau1x61); -} - -static struct platform_driver bfin_eval_adau1x61_driver = { - .driver = { - .name = "bfin-eval-adau1x61", - .pm = &snd_soc_pm_ops, - }, - .probe = bfin_eval_adau1x61_probe, -}; -module_platform_driver(bfin_eval_adau1x61_driver); - -MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); -MODULE_DESCRIPTION("ALSA SoC bfin adau1x61 driver"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:bfin-eval-adau1x61"); diff --git a/sound/soc/blackfin/bfin-eval-adau1x81.c b/sound/soc/blackfin/bfin-eval-adau1x81.c deleted file mode 100644 index 3e01cbe53fc7..000000000000 --- a/sound/soc/blackfin/bfin-eval-adau1x81.c +++ /dev/null @@ -1,129 +0,0 @@ -/* - * Machine driver for EVAL-ADAU1x81 on Analog Devices bfin - * evaluation boards. - * - * Copyright 2011-2014 Analog Devices Inc. - * Author: Lars-Peter Clausen <lars(a)metafoo.de> - * - * Licensed under the GPL-2 or later. - */ - -#include <linux/module.h> -#include <linux/device.h> -#include <linux/slab.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/pcm_params.h> - -#include "../codecs/adau17x1.h" - -static const struct snd_soc_dapm_widget bfin_eval_adau1x81_dapm_widgets[] = { - SND_SOC_DAPM_LINE("Stereo In", NULL), - SND_SOC_DAPM_LINE("Beep", NULL), - - SND_SOC_DAPM_SPK("Speaker", NULL), - SND_SOC_DAPM_HP("Headphone", NULL), -}; - -static const struct snd_soc_dapm_route bfin_eval_adau1x81_dapm_routes[] = { - { "BEEP", NULL, "Beep" }, - { "LMIC", NULL, "Stereo In" }, - { "LMIC", NULL, "Stereo In" }, - - { "Headphone", NULL, "AOUTL" }, - { "Headphone", NULL, "AOUTR" }, - { "Speaker", NULL, "SP" }, -}; - -static int bfin_eval_adau1x81_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int pll_rate; - int ret; - - switch (params_rate(params)) { - case 48000: - case 8000: - case 12000: - case 16000: - case 24000: - case 32000: - case 96000: - pll_rate = 48000 * 1024; - break; - case 44100: - case 7350: - case 11025: - case 14700: - case 22050: - case 29400: - case 88200: - pll_rate = 44100 * 1024; - break; - default: - return -EINVAL; - } - - ret = snd_soc_dai_set_pll(codec_dai, ADAU17X1_PLL, - ADAU17X1_PLL_SRC_MCLK, 12288000, pll_rate); - if (ret) - return ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, ADAU17X1_CLK_SRC_PLL, pll_rate, - SND_SOC_CLOCK_IN); - - return ret; -} - -static const struct snd_soc_ops bfin_eval_adau1x81_ops = { - .hw_params = bfin_eval_adau1x81_hw_params, -}; - -static struct snd_soc_dai_link bfin_eval_adau1x81_dai = { - .name = "adau1x81", - .stream_name = "adau1x81", - .cpu_dai_name = "bfin-i2s.0", - .codec_dai_name = "adau-hifi", - .platform_name = "bfin-i2s-pcm-audio", - .codec_name = "adau1781.0-0038", - .ops = &bfin_eval_adau1x81_ops, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, -}; - -static struct snd_soc_card bfin_eval_adau1x81 = { - .name = "bfin-eval-adau1x81", - .driver_name = "eval-adau1x81", - .dai_link = &bfin_eval_adau1x81_dai, - .num_links = 1, - - .dapm_widgets = bfin_eval_adau1x81_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(bfin_eval_adau1x81_dapm_widgets), - .dapm_routes = bfin_eval_adau1x81_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(bfin_eval_adau1x81_dapm_routes), - .fully_routed = true, -}; - -static int bfin_eval_adau1x81_probe(struct platform_device *pdev) -{ - bfin_eval_adau1x81.dev = &pdev->dev; - - return devm_snd_soc_register_card(&pdev->dev, &bfin_eval_adau1x81); -} - -static struct platform_driver bfin_eval_adau1x81_driver = { - .driver = { - .name = "bfin-eval-adau1x81", - .pm = &snd_soc_pm_ops, - }, - .probe = bfin_eval_adau1x81_probe, -}; -module_platform_driver(bfin_eval_adau1x81_driver); - -MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); -MODULE_DESCRIPTION("ALSA SoC bfin adau1x81 driver"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:bfin-eval-adau1x81"); diff --git a/sound/soc/blackfin/bfin-eval-adav80x.c b/sound/soc/blackfin/bfin-eval-adav80x.c deleted file mode 100644 index 99e5ecabdcda..000000000000 --- a/sound/soc/blackfin/bfin-eval-adav80x.c +++ /dev/null @@ -1,145 +0,0 @@ -/* - * Machine driver for EVAL-ADAV801 and EVAL-ADAV803 on Analog Devices bfin - * evaluation boards. - * - * Copyright 2011 Analog Devices Inc. - * Author: Lars-Peter Clausen <lars(a)metafoo.de> - * - * Licensed under the GPL-2 or later. - */ - -#include <linux/init.h> -#include <linux/platform_device.h> -#include <linux/module.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> - -#include "../codecs/adav80x.h" - -static const struct snd_soc_dapm_widget bfin_eval_adav80x_dapm_widgets[] = { - SND_SOC_DAPM_LINE("Line Out", NULL), - SND_SOC_DAPM_LINE("Line In", NULL), -}; - -static const struct snd_soc_dapm_route bfin_eval_adav80x_dapm_routes[] = { - { "Line Out", NULL, "VOUTL" }, - { "Line Out", NULL, "VOUTR" }, - - { "VINL", NULL, "Line In" }, - { "VINR", NULL, "Line In" }, -}; - -static int bfin_eval_adav80x_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - ret = snd_soc_dai_set_pll(codec_dai, ADAV80X_PLL1, ADAV80X_PLL_SRC_XTAL, - 27000000, params_rate(params) * 256); - if (ret) - return ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_PLL1, - params_rate(params) * 256, SND_SOC_CLOCK_IN); - - return ret; -} - -static int bfin_eval_adav80x_codec_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_dai *codec_dai = rtd->codec_dai; - - snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_SYSCLK1, 0, - SND_SOC_CLOCK_OUT); - snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_SYSCLK2, 0, - SND_SOC_CLOCK_OUT); - snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_SYSCLK3, 0, - SND_SOC_CLOCK_OUT); - - snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_XTAL, 2700000, 0); - - return 0; -} - -static const struct snd_soc_ops bfin_eval_adav80x_ops = { - .hw_params = bfin_eval_adav80x_hw_params, -}; - -static struct snd_soc_dai_link bfin_eval_adav80x_dais[] = { - { - .name = "adav80x", - .stream_name = "ADAV80x HiFi", - .cpu_dai_name = "bfin-i2s.0", - .codec_dai_name = "adav80x-hifi", - .platform_name = "bfin-i2s-pcm-audio", - .init = bfin_eval_adav80x_codec_init, - .ops = &bfin_eval_adav80x_ops, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, - }, -}; - -static struct snd_soc_card bfin_eval_adav80x = { - .name = "bfin-eval-adav80x", - .owner = THIS_MODULE, - .dai_link = bfin_eval_adav80x_dais, - .num_links = ARRAY_SIZE(bfin_eval_adav80x_dais), - - .dapm_widgets = bfin_eval_adav80x_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(bfin_eval_adav80x_dapm_widgets), - .dapm_routes = bfin_eval_adav80x_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(bfin_eval_adav80x_dapm_routes), -}; - -enum bfin_eval_adav80x_type { - BFIN_EVAL_ADAV801, - BFIN_EVAL_ADAV803, -}; - -static int bfin_eval_adav80x_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &bfin_eval_adav80x; - const char *codec_name; - - switch (platform_get_device_id(pdev)->driver_data) { - case BFIN_EVAL_ADAV801: - codec_name = "spi0.1"; - break; - case BFIN_EVAL_ADAV803: - codec_name = "adav803.0-0034"; - break; - default: - return -EINVAL; - } - - bfin_eval_adav80x_dais[0].codec_name = codec_name; - - card->dev = &pdev->dev; - - return devm_snd_soc_register_card(&pdev->dev, &bfin_eval_adav80x); -} - -static const struct platform_device_id bfin_eval_adav80x_ids[] = { - { "bfin-eval-adav801", BFIN_EVAL_ADAV801 }, - { "bfin-eval-adav803", BFIN_EVAL_ADAV803 }, - { }, -}; -MODULE_DEVICE_TABLE(platform, bfin_eval_adav80x_ids); - -static struct platform_driver bfin_eval_adav80x_driver = { - .driver = { - .name = "bfin-eval-adav80x", - .pm = &snd_soc_pm_ops, - }, - .probe = bfin_eval_adav80x_probe, - .id_table = bfin_eval_adav80x_ids, -}; - -module_platform_driver(bfin_eval_adav80x_driver); - -MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); -MODULE_DESCRIPTION("ALSA SoC bfin adav80x driver"); -MODULE_LICENSE("GPL"); -- 2.9.0

2 1

[alsa-devel] [PATCH] ASoC: remove blackfin drivers
by Arnd Bergmann 14 Mar '18

14 Mar '18

The blackfin architecture is getting removed, so the ASoC drivers are all obsolete as well. Signed-off-by: Arnd Bergmann <arnd(a)arndb.de> --- MAINTAINERS | 7 - sound/soc/Kconfig | 1 - sound/soc/Makefile | 1 - sound/soc/blackfin/Kconfig | 205 ------ sound/soc/blackfin/Makefile | 40 -- sound/soc/blackfin/bf5xx-ac97-pcm.c | 480 -------------- sound/soc/blackfin/bf5xx-ac97.c | 388 ----------- sound/soc/blackfin/bf5xx-ac97.h | 57 -- sound/soc/blackfin/bf5xx-ad1836.c | 109 --- sound/soc/blackfin/bf5xx-ad193x.c | 131 ---- sound/soc/blackfin/bf5xx-ad1980.c | 109 --- sound/soc/blackfin/bf5xx-ad73311.c | 212 ------ sound/soc/blackfin/bf5xx-i2s-pcm.c | 373 ----------- sound/soc/blackfin/bf5xx-i2s-pcm.h | 17 - sound/soc/blackfin/bf5xx-i2s.c | 391 ----------- sound/soc/blackfin/bf5xx-sport.c | 1102 ------------------------------- sound/soc/blackfin/bf5xx-sport.h | 174 ----- sound/soc/blackfin/bf5xx-ssm2602.c | 126 ---- sound/soc/blackfin/bf6xx-i2s.c | 239 ------- sound/soc/blackfin/bf6xx-sport.c | 425 ------------ sound/soc/blackfin/bf6xx-sport.h | 82 --- sound/soc/blackfin/bfin-eval-adau1373.c | 173 ----- sound/soc/blackfin/bfin-eval-adau1701.c | 113 ---- sound/soc/blackfin/bfin-eval-adau1x61.c | 142 ---- sound/soc/blackfin/bfin-eval-adau1x81.c | 129 ---- sound/soc/blackfin/bfin-eval-adav80x.c | 145 ---- 26 files changed, 5371 deletions(-) delete mode 100644 sound/soc/blackfin/Kconfig delete mode 100644 sound/soc/blackfin/Makefile delete mode 100644 sound/soc/blackfin/bf5xx-ac97-pcm.c delete mode 100644 sound/soc/blackfin/bf5xx-ac97.c delete mode 100644 sound/soc/blackfin/bf5xx-ac97.h delete mode 100644 sound/soc/blackfin/bf5xx-ad1836.c delete mode 100644 sound/soc/blackfin/bf5xx-ad193x.c delete mode 100644 sound/soc/blackfin/bf5xx-ad1980.c delete mode 100644 sound/soc/blackfin/bf5xx-ad73311.c delete mode 100644 sound/soc/blackfin/bf5xx-i2s-pcm.c delete mode 100644 sound/soc/blackfin/bf5xx-i2s-pcm.h delete mode 100644 sound/soc/blackfin/bf5xx-i2s.c delete mode 100644 sound/soc/blackfin/bf5xx-sport.c delete mode 100644 sound/soc/blackfin/bf5xx-sport.h delete mode 100644 sound/soc/blackfin/bf5xx-ssm2602.c delete mode 100644 sound/soc/blackfin/bf6xx-i2s.c delete mode 100644 sound/soc/blackfin/bf6xx-sport.c delete mode 100644 sound/soc/blackfin/bf6xx-sport.h delete mode 100644 sound/soc/blackfin/bfin-eval-adau1373.c delete mode 100644 sound/soc/blackfin/bfin-eval-adau1701.c delete mode 100644 sound/soc/blackfin/bfin-eval-adau1x61.c delete mode 100644 sound/soc/blackfin/bfin-eval-adau1x81.c delete mode 100644 sound/soc/blackfin/bfin-eval-adav80x.c diff --git a/MAINTAINERS b/MAINTAINERS index c06fcd0e1514..d485d45ad690 100644 --- a/MAINTAINERS +++ b/MAINTAINERS @@ -841,13 +841,6 @@ F: sound/soc/codecs/ad7* F: sound/soc/codecs/ssm* F: sound/soc/codecs/sigmadsp.* -ANALOG DEVICES INC ASOC DRIVERS -L: adi-buildroot-devel(a)lists.sourceforge.net (moderated for non-subscribers) -L: alsa-devel(a)alsa-project.org (moderated for non-subscribers) -W: http://blackfin.uclinux.org/ -S: Supported -F: sound/soc/blackfin/* - ANALOG DEVICES INC DMA DRIVERS M: Lars-Peter Clausen <lars(a)metafoo.de> W: http://ez.analog.com/community/linux-device-drivers diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig index 84c3582f3982..41af6b9cc350 100644 --- a/sound/soc/Kconfig +++ b/sound/soc/Kconfig @@ -45,7 +45,6 @@ source "sound/soc/amd/Kconfig" source "sound/soc/atmel/Kconfig" source "sound/soc/au1x/Kconfig" source "sound/soc/bcm/Kconfig" -source "sound/soc/blackfin/Kconfig" source "sound/soc/cirrus/Kconfig" source "sound/soc/davinci/Kconfig" source "sound/soc/dwc/Kconfig" diff --git a/sound/soc/Makefile b/sound/soc/Makefile index 74cd1858d38b..8d92492183d2 100644 --- a/sound/soc/Makefile +++ b/sound/soc/Makefile @@ -29,7 +29,6 @@ obj-$(CONFIG_SND_SOC) += amd/ obj-$(CONFIG_SND_SOC) += atmel/ obj-$(CONFIG_SND_SOC) += au1x/ obj-$(CONFIG_SND_SOC) += bcm/ -obj-$(CONFIG_SND_SOC) += blackfin/ obj-$(CONFIG_SND_SOC) += cirrus/ obj-$(CONFIG_SND_SOC) += davinci/ obj-$(CONFIG_SND_SOC) += dwc/ diff --git a/sound/soc/blackfin/Kconfig b/sound/soc/blackfin/Kconfig deleted file mode 100644 index 6410aa2cc2cf..000000000000 --- a/sound/soc/blackfin/Kconfig +++ /dev/null @@ -1,205 +0,0 @@ -config SND_BF5XX_I2S - tristate "SoC I2S Audio for the ADI Blackfin chip" - depends on BLACKFIN - select SND_BF5XX_SOC_SPORT if !BF60x - select SND_BF6XX_SOC_SPORT if BF60x - help - Say Y or M if you want to add support for codecs attached to - the Blackfin SPORT (synchronous serial ports) interface in I2S - mode (supports single stereo In/Out). - You will also need to select the audio interfaces to support below. - -config SND_BF5XX_SOC_SSM2602 - tristate "SoC SSM2602 Audio Codec Add-On Card support" - depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI - select SND_BF5XX_SOC_I2S if !BF60x - select SND_BF6XX_SOC_I2S if BF60x - select SND_SOC_SSM2602_SPI if SPI_MASTER - select SND_SOC_SSM2602_I2C if I2C - help - Say Y if you want to add support for the Analog Devices - SSM2602 Audio Codec Add-On Card. - -config SND_SOC_BFIN_EVAL_ADAU1701 - tristate "Support for the EVAL-ADAU1701MINIZ board on Blackfin eval boards" - depends on SND_BF5XX_I2S && I2C - select SND_BF5XX_SOC_I2S - select SND_SOC_ADAU1701 - help - Say Y if you want to add support for the Analog Devices EVAL-ADAU1701MINIZ - board connected to one of the Blackfin evaluation boards like the - BF5XX-STAMP or BF5XX-EZKIT. - -config SND_SOC_BFIN_EVAL_ADAU1373 - tristate "Support for the EVAL-ADAU1373 board on Blackfin eval boards" - depends on SND_BF5XX_I2S && I2C - select SND_BF5XX_SOC_I2S - select SND_SOC_ADAU1373 - help - Say Y if you want to add support for the Analog Devices EVAL-ADAU1373 - board connected to one of the Blackfin evaluation boards like the - BF5XX-STAMP or BF5XX-EZKIT. - - Note: This driver assumes that first ADAU1373 DAI is connected to the - first SPORT port on the BF5XX board. - -config SND_SOC_BFIN_EVAL_ADAU1X61 - tristate "Support for the EVAL-ADAU1X61 board on Blackfin eval boards" - depends on SND_BF5XX_I2S && I2C - select SND_BF5XX_SOC_I2S - select SND_SOC_ADAU1761_I2C - help - Say Y if you want to add support for the Analog Devices EVAL-ADAU1X61 - board connected to one of the Blackfin evaluation boards like the - BF5XX-STAMP or BF5XX-EZKIT. - - Note: This driver assumes that the ADAU1X61 is connected to the - first SPORT port on the BF5XX board. - -config SND_SOC_BFIN_EVAL_ADAU1X81 - tristate "Support for the EVAL-ADAU1X81 boards on Blackfin eval boards" - depends on SND_BF5XX_I2S && I2C - select SND_BF5XX_SOC_I2S - select SND_SOC_ADAU1781_I2C - help - Say Y if you want to add support for the Analog Devices EVAL-ADAU1X81 - board connected to one of the Blackfin evaluation boards like the - BF5XX-STAMP or BF5XX-EZKIT. - - Note: This driver assumes that the ADAU1X81 is connected to the - first SPORT port on the BF5XX board. - -config SND_SOC_BFIN_EVAL_ADAV80X - tristate "Support for the EVAL-ADAV80X boards on Blackfin eval boards" - depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI - select SND_BF5XX_SOC_I2S - select SND_SOC_ADAV801 if SPI_MASTER - select SND_SOC_ADAV803 if I2C - help - Say Y if you want to add support for the Analog Devices EVAL-ADAV801 or - EVAL-ADAV803 board connected to one of the Blackfin evaluation boards - like the BF5XX-STAMP or BF5XX-EZKIT. - - Note: This driver assumes that the ADAV80X digital record and playback - interfaces are connected to the first SPORT port on the BF5XX board. - -config SND_BF5XX_SOC_AD1836 - tristate "SoC AD1836 Audio support for BF5xx" - depends on SND_BF5XX_I2S && SPI_MASTER - select SND_BF5XX_SOC_I2S - select SND_SOC_AD1836 - help - Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT. - -config SND_BF5XX_SOC_AD193X - tristate "SoC AD193X Audio support for Blackfin" - depends on SND_BF5XX_I2S && SND_SOC_I2C_AND_SPI - select SND_BF5XX_SOC_I2S - select SND_SOC_AD193X_I2C if I2C - select SND_SOC_AD193X_SPI if SPI_MASTER - help - Say Y if you want to add support for AD193X codec on Blackfin. - This driver supports AD1936, AD1937, AD1938 and AD1939. - -config SND_BF5XX_SOC_AD73311 - tristate "SoC AD73311 Audio support for Blackfin" - depends on SND_BF5XX_I2S - select SND_BF5XX_SOC_I2S - select SND_SOC_AD73311 - help - Say Y if you want to add support for AD73311 codec on Blackfin. - -config SND_BFIN_AD73311_SE - int "PF pin for AD73311L Chip Select" - depends on SND_BF5XX_SOC_AD73311 - default 4 - help - Enter the GPIO used to control AD73311's SE pin. Acceptable - values are 0 to 7 - -config SND_BF5XX_AC97 - tristate "SoC AC97 Audio for the ADI BF5xx chip" - depends on BLACKFIN - select AC97_BUS - select SND_SOC_AC97_BUS - select SND_BF5XX_SOC_SPORT - select SND_BF5XX_SOC_AC97 - help - Say Y or M if you want to add support for codecs attached to - the Blackfin SPORT (synchronous serial ports) interface in slot 16 - mode (pseudo AC97 interface). - You will also need to select the audio interfaces to support below. - - Note: - AC97 codecs which do not implement the slot-16 mode will not function - properly with this driver. This driver is known to work with the - Analog Devices line of AC97 codecs. - -config SND_BF5XX_MMAP_SUPPORT - bool "Enable MMAP Support" - depends on SND_BF5XX_AC97 - default y - help - Say y if you want AC97 driver to support mmap mode. - We introduce an intermediate buffer to simulate mmap. - -config SND_BF5XX_MULTICHAN_SUPPORT - bool "Enable Multichannel Support" - depends on SND_BF5XX_AC97 - default n - help - Say y if you want AC97 driver to support up to 5.1 channel audio. - this mode will consume much more memory for DMA. - -config SND_BF5XX_HAVE_COLD_RESET - bool "BOARD has COLD Reset GPIO" - depends on SND_BF5XX_AC97 - default y if BFIN548_EZKIT - default n if !BFIN548_EZKIT - -config SND_BF5XX_RESET_GPIO_NUM - int "Set a GPIO for cold reset" - depends on SND_BF5XX_HAVE_COLD_RESET - range 0 159 - default 19 if BFIN548_EZKIT - default 5 if BFIN537_STAMP - default 0 - help - Set the correct GPIO for RESET the sound chip. - -config SND_BF5XX_SOC_AD1980 - tristate "SoC AD1980/1 Audio support for BF5xx (Obsolete)" - depends on SND_BF5XX_AC97 - select SND_BF5XX_SOC_AC97 - select SND_SOC_AD1980 - help - Say Y if you want to add support for SoC audio on BF5xx STAMP/EZKIT. - - Warning: - Because Analog Devices Inc. discontinued the ad1980 sound chip since - Sep. 2009, this ad1980 driver is not maintained, tested and supported - by ADI now. - -config SND_BF5XX_SOC_SPORT - tristate - -config SND_BF6XX_SOC_SPORT - tristate - -config SND_BF5XX_SOC_I2S - tristate - -config SND_BF6XX_SOC_I2S - tristate - -config SND_BF5XX_SOC_AC97 - tristate - -config SND_BF5XX_SPORT_NUM - int "Set a SPORT for Sound chip" - depends on (SND_BF5XX_SOC_SPORT || SND_BF6XX_SOC_SPORT) - range 0 3 if BF54x - range 0 1 if !BF54x - default 0 - help - Set the correct SPORT for sound chip. diff --git a/sound/soc/blackfin/Makefile b/sound/soc/blackfin/Makefile deleted file mode 100644 index ebeb6a9cedd2..000000000000 --- a/sound/soc/blackfin/Makefile +++ /dev/null @@ -1,40 +0,0 @@ -# SPDX-License-Identifier: GPL-2.0 -# Blackfin Platform Support -snd-bf5xx-ac97-objs := bf5xx-ac97-pcm.o -snd-bf5xx-i2s-objs := bf5xx-i2s-pcm.o -snd-soc-bf5xx-sport-objs := bf5xx-sport.o -snd-soc-bf6xx-sport-objs := bf6xx-sport.o -snd-soc-bf5xx-ac97-objs := bf5xx-ac97.o -snd-soc-bf5xx-i2s-objs := bf5xx-i2s.o -snd-soc-bf6xx-i2s-objs := bf6xx-i2s.o - -obj-$(CONFIG_SND_BF5XX_AC97) += snd-bf5xx-ac97.o -obj-$(CONFIG_SND_BF5XX_I2S) += snd-bf5xx-i2s.o -obj-$(CONFIG_SND_BF5XX_SOC_SPORT) += snd-soc-bf5xx-sport.o -obj-$(CONFIG_SND_BF6XX_SOC_SPORT) += snd-soc-bf6xx-sport.o -obj-$(CONFIG_SND_BF5XX_SOC_AC97) += snd-soc-bf5xx-ac97.o -obj-$(CONFIG_SND_BF5XX_SOC_I2S) += snd-soc-bf5xx-i2s.o -obj-$(CONFIG_SND_BF6XX_SOC_I2S) += snd-soc-bf6xx-i2s.o - -# Blackfin Machine Support -snd-ad1836-objs := bf5xx-ad1836.o -snd-ad1980-objs := bf5xx-ad1980.o -snd-ssm2602-objs := bf5xx-ssm2602.o -snd-ad73311-objs := bf5xx-ad73311.o -snd-ad193x-objs := bf5xx-ad193x.o -snd-soc-bfin-eval-adau1373-objs := bfin-eval-adau1373.o -snd-soc-bfin-eval-adau1x61-objs := bfin-eval-adau1x61.o -snd-soc-bfin-eval-adau1x81-objs := bfin-eval-adau1x81.o -snd-soc-bfin-eval-adau1701-objs := bfin-eval-adau1701.o -snd-soc-bfin-eval-adav80x-objs := bfin-eval-adav80x.o - -obj-$(CONFIG_SND_BF5XX_SOC_AD1836) += snd-ad1836.o -obj-$(CONFIG_SND_BF5XX_SOC_AD1980) += snd-ad1980.o -obj-$(CONFIG_SND_BF5XX_SOC_SSM2602) += snd-ssm2602.o -obj-$(CONFIG_SND_BF5XX_SOC_AD73311) += snd-ad73311.o -obj-$(CONFIG_SND_BF5XX_SOC_AD193X) += snd-ad193x.o -obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAU1373) += snd-soc-bfin-eval-adau1373.o -obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAU1X61) += snd-soc-bfin-eval-adau1x61.o -obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAU1X81) += snd-soc-bfin-eval-adau1x81.o -obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAU1701) += snd-soc-bfin-eval-adau1701.o -obj-$(CONFIG_SND_SOC_BFIN_EVAL_ADAV80X) += snd-soc-bfin-eval-adav80x.o diff --git a/sound/soc/blackfin/bf5xx-ac97-pcm.c b/sound/soc/blackfin/bf5xx-ac97-pcm.c deleted file mode 100644 index 8c1d1983b8f9..000000000000 --- a/sound/soc/blackfin/bf5xx-ac97-pcm.c +++ /dev/null @@ -1,480 +0,0 @@ -/* - * File: sound/soc/blackfin/bf5xx-ac97-pcm.c - * Author: Cliff Cai <Cliff.Cai(a)analog.com> - * - * Created: Tue June 06 2008 - * Description: DMA Driver for AC97 sound chip - * - * Modified: - * Copyright 2008 Analog Devices Inc. - * - * Bugs: Enter bugs at http://blackfin.uclinux.org/ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, see the file COPYING, or write - * to the Free Software Foundation, Inc., - * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA - */ - -#include <linux/module.h> -#include <linux/init.h> -#include <linux/platform_device.h> -#include <linux/dma-mapping.h> -#include <linux/gfp.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include <asm/dma.h> - -#include "bf5xx-ac97.h" -#include "bf5xx-sport.h" - -static unsigned int ac97_chan_mask[] = { - SP_FL, /* Mono */ - SP_STEREO, /* Stereo */ - SP_2DOT1, /* 2.1*/ - SP_QUAD,/*Quadraquic*/ - SP_FL | SP_FR | SP_FC | SP_SL | SP_SR,/*5 channels */ - SP_5DOT1, /* 5.1 */ -}; - -#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) -static void bf5xx_mmap_copy(struct snd_pcm_substream *substream, - snd_pcm_uframes_t count) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct sport_device *sport = runtime->private_data; - unsigned int chan_mask = ac97_chan_mask[runtime->channels - 1]; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - bf5xx_pcm_to_ac97((struct ac97_frame *)sport->tx_dma_buf + - sport->tx_pos, (__u16 *)runtime->dma_area + sport->tx_pos * - runtime->channels, count, chan_mask); - sport->tx_pos += runtime->period_size; - if (sport->tx_pos >= runtime->buffer_size) - sport->tx_pos %= runtime->buffer_size; - sport->tx_delay_pos = sport->tx_pos; - } else { - bf5xx_ac97_to_pcm((struct ac97_frame *)sport->rx_dma_buf + - sport->rx_pos, (__u16 *)runtime->dma_area + sport->rx_pos * - runtime->channels, count); - sport->rx_pos += runtime->period_size; - if (sport->rx_pos >= runtime->buffer_size) - sport->rx_pos %= runtime->buffer_size; - } -} -#endif - -static void bf5xx_dma_irq(void *data) -{ - struct snd_pcm_substream *pcm = data; -#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) - struct snd_pcm_runtime *runtime = pcm->runtime; - struct sport_device *sport = runtime->private_data; - bf5xx_mmap_copy(pcm, runtime->period_size); - if (pcm->stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (sport->once == 0) { - snd_pcm_period_elapsed(pcm); - bf5xx_mmap_copy(pcm, runtime->period_size); - sport->once = 1; - } - } -#endif - snd_pcm_period_elapsed(pcm); -} - -/* The memory size for pure pcm data is 128*1024 = 0x20000 bytes. - * The total rx/tx buffer is for ac97 frame to hold all pcm data - * is 0x20000 * sizeof(struct ac97_frame) / 4. - */ -static const struct snd_pcm_hardware bf5xx_pcm_hardware = { - .info = SNDRV_PCM_INFO_INTERLEAVED | -#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) - SNDRV_PCM_INFO_MMAP | - SNDRV_PCM_INFO_MMAP_VALID | -#endif - SNDRV_PCM_INFO_BLOCK_TRANSFER, - - .period_bytes_min = 32, - .period_bytes_max = 0x10000, - .periods_min = 1, - .periods_max = PAGE_SIZE/32, - .buffer_bytes_max = 0x20000, /* 128 kbytes */ - .fifo_size = 16, -}; - -static int bf5xx_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - size_t size = bf5xx_pcm_hardware.buffer_bytes_max - * sizeof(struct ac97_frame) / 4; - - snd_pcm_lib_malloc_pages(substream, size); - - return 0; -} - -static int bf5xx_pcm_hw_free(struct snd_pcm_substream *substream) -{ -#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) - struct snd_pcm_runtime *runtime = substream->runtime; - struct sport_device *sport = runtime->private_data; - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - sport->once = 0; - if (runtime->dma_area) - memset(runtime->dma_area, 0, runtime->buffer_size); - memset(sport->tx_dma_buf, 0, runtime->buffer_size * - sizeof(struct ac97_frame)); - } else - memset(sport->rx_dma_buf, 0, runtime->buffer_size * - sizeof(struct ac97_frame)); -#endif - snd_pcm_lib_free_pages(substream); - return 0; -} - -static int bf5xx_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct sport_device *sport = runtime->private_data; - - /* An intermediate buffer is introduced for implementing mmap for - * SPORT working in TMD mode(include AC97). - */ -#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - sport_set_tx_callback(sport, bf5xx_dma_irq, substream); - sport_config_tx_dma(sport, sport->tx_dma_buf, runtime->periods, - runtime->period_size * sizeof(struct ac97_frame)); - } else { - sport_set_rx_callback(sport, bf5xx_dma_irq, substream); - sport_config_rx_dma(sport, sport->rx_dma_buf, runtime->periods, - runtime->period_size * sizeof(struct ac97_frame)); - } -#else - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - sport_set_tx_callback(sport, bf5xx_dma_irq, substream); - sport_config_tx_dma(sport, runtime->dma_area, runtime->periods, - runtime->period_size * sizeof(struct ac97_frame)); - } else { - sport_set_rx_callback(sport, bf5xx_dma_irq, substream); - sport_config_rx_dma(sport, runtime->dma_area, runtime->periods, - runtime->period_size * sizeof(struct ac97_frame)); - } -#endif - return 0; -} - -static int bf5xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct sport_device *sport = runtime->private_data; - int ret = 0; - - pr_debug("%s enter\n", __func__); - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { -#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) - bf5xx_mmap_copy(substream, runtime->period_size); - sport->tx_delay_pos = 0; -#endif - sport_tx_start(sport); - } else - sport_rx_start(sport); - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { -#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) - sport->tx_pos = 0; -#endif - sport_tx_stop(sport); - } else { -#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) - sport->rx_pos = 0; -#endif - sport_rx_stop(sport); - } - break; - default: - ret = -EINVAL; - } - return ret; -} - -static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct sport_device *sport = runtime->private_data; - unsigned int curr; - -#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - curr = sport->tx_delay_pos; - else - curr = sport->rx_pos; -#else - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - curr = sport_curr_offset_tx(sport) / sizeof(struct ac97_frame); - else - curr = sport_curr_offset_rx(sport) / sizeof(struct ac97_frame); - -#endif - return curr; -} - -static int bf5xx_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct sport_device *sport_handle = snd_soc_dai_get_drvdata(cpu_dai); - struct snd_pcm_runtime *runtime = substream->runtime; - int ret; - - pr_debug("%s enter\n", __func__); - snd_soc_set_runtime_hwparams(substream, &bf5xx_pcm_hardware); - - ret = snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) - goto out; - - if (sport_handle != NULL) - runtime->private_data = sport_handle; - else { - pr_err("sport_handle is NULL\n"); - return -1; - } - return 0; - - out: - return ret; -} - -#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) -static int bf5xx_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - size_t size = vma->vm_end - vma->vm_start; - vma->vm_start = (unsigned long)runtime->dma_area; - vma->vm_end = vma->vm_start + size; - vma->vm_flags |= VM_SHARED; - return 0 ; -} -#else -static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, - int channel, unsigned long pos, - void *buf, unsigned long count) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int chan_mask = ac97_chan_mask[runtime->channels - 1]; - struct ac97_frame *dst; - - pr_debug("%s copy pos:0x%lx count:0x%lx\n", - substream->stream ? "Capture" : "Playback", pos, count); - dst = (struct ac97_frame *)runtime->dma_area + - bytes_to_frames(runtime, pos); - count = bytes_to_frames(runtime, count); - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - bf5xx_pcm_to_ac97(dst, buf, count, chan_mask); - else - bf5xx_ac97_to_pcm(dst, buf, count); - return 0; -} - -static int bf5xx_pcm_copy_user(struct snd_pcm_substream *substream, - int channel, unsigned long pos, - void __user *buf, unsigned long count) -{ - return bf5xx_pcm_copy(substream, channel, pos, (void *)buf, count); -} -#endif - -static const struct snd_pcm_ops bf5xx_pcm_ac97_ops = { - .open = bf5xx_pcm_open, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = bf5xx_pcm_hw_params, - .hw_free = bf5xx_pcm_hw_free, - .prepare = bf5xx_pcm_prepare, - .trigger = bf5xx_pcm_trigger, - .pointer = bf5xx_pcm_pointer, -#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) - .mmap = bf5xx_pcm_mmap, -#else - .copy_user = bf5xx_pcm_copy_user, - .copy_kernel = bf5xx_pcm_copy, -#endif -}; - -static int bf5xx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) -{ - struct snd_soc_pcm_runtime *rtd = pcm->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct sport_device *sport_handle = snd_soc_dai_get_drvdata(cpu_dai); - struct snd_pcm_substream *substream = pcm->streams[stream].substream; - struct snd_dma_buffer *buf = &substream->dma_buffer; - size_t size = bf5xx_pcm_hardware.buffer_bytes_max - * sizeof(struct ac97_frame) / 4; - - buf->dev.type = SNDRV_DMA_TYPE_DEV; - buf->dev.dev = pcm->card->dev; - buf->private_data = NULL; - buf->area = dma_alloc_coherent(pcm->card->dev, size, - &buf->addr, GFP_KERNEL); - if (!buf->area) { - pr_err("Failed to allocate dma memory\n"); - pr_err("Please increase uncached DMA memory region\n"); - return -ENOMEM; - } - buf->bytes = size; - - pr_debug("%s, area:%p, size:0x%08lx\n", __func__, - buf->area, buf->bytes); - - if (stream == SNDRV_PCM_STREAM_PLAYBACK) - sport_handle->tx_buf = buf->area; - else - sport_handle->rx_buf = buf->area; - -/* - * Need to allocate local buffer when enable - * MMAP for SPORT working in TMD mode (include AC97). - */ -#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (!sport_handle->tx_dma_buf) { - sport_handle->tx_dma_buf = dma_alloc_coherent(NULL, \ - size, &sport_handle->tx_dma_phy, GFP_KERNEL); - if (!sport_handle->tx_dma_buf) { - pr_err("Failed to allocate memory for tx dma buf - Please increase uncached DMA memory region\n"); - return -ENOMEM; - } else - memset(sport_handle->tx_dma_buf, 0, size); - } else - memset(sport_handle->tx_dma_buf, 0, size); - } else { - if (!sport_handle->rx_dma_buf) { - sport_handle->rx_dma_buf = dma_alloc_coherent(NULL, \ - size, &sport_handle->rx_dma_phy, GFP_KERNEL); - if (!sport_handle->rx_dma_buf) { - pr_err("Failed to allocate memory for rx dma buf - Please increase uncached DMA memory region\n"); - return -ENOMEM; - } else - memset(sport_handle->rx_dma_buf, 0, size); - } else - memset(sport_handle->rx_dma_buf, 0, size); - } -#endif - return 0; -} - -static void bf5xx_pcm_free_dma_buffers(struct snd_pcm *pcm) -{ - struct snd_pcm_substream *substream; - struct snd_dma_buffer *buf; - int stream; -#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) - struct snd_soc_pcm_runtime *rtd = pcm->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct sport_device *sport_handle = snd_soc_dai_get_drvdata(cpu_dai); - size_t size = bf5xx_pcm_hardware.buffer_bytes_max * - sizeof(struct ac97_frame) / 4; -#endif - for (stream = 0; stream < 2; stream++) { - substream = pcm->streams[stream].substream; - if (!substream) - continue; - - buf = &substream->dma_buffer; - if (!buf->area) - continue; - dma_free_coherent(NULL, buf->bytes, buf->area, 0); - buf->area = NULL; -#if defined(CONFIG_SND_BF5XX_MMAP_SUPPORT) - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - if (sport_handle->tx_dma_buf) - dma_free_coherent(NULL, size, \ - sport_handle->tx_dma_buf, 0); - sport_handle->tx_dma_buf = NULL; - } else { - - if (sport_handle->rx_dma_buf) - dma_free_coherent(NULL, size, \ - sport_handle->rx_dma_buf, 0); - sport_handle->rx_dma_buf = NULL; - } -#endif - } -} - -static int bf5xx_pcm_ac97_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - struct snd_pcm *pcm = rtd->pcm; - int ret; - - pr_debug("%s enter\n", __func__); - ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); - if (ret) - return ret; - - if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) { - ret = bf5xx_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_PLAYBACK); - if (ret) - goto out; - } - - if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) { - ret = bf5xx_pcm_preallocate_dma_buffer(pcm, - SNDRV_PCM_STREAM_CAPTURE); - if (ret) - goto out; - } - out: - return ret; -} - -static struct snd_soc_platform_driver bf5xx_ac97_soc_platform = { - .ops = &bf5xx_pcm_ac97_ops, - .pcm_new = bf5xx_pcm_ac97_new, - .pcm_free = bf5xx_pcm_free_dma_buffers, -}; - -static int bf5xx_soc_platform_probe(struct platform_device *pdev) -{ - return devm_snd_soc_register_platform(&pdev->dev, - &bf5xx_ac97_soc_platform); -} - -static struct platform_driver bf5xx_pcm_driver = { - .driver = { - .name = "bfin-ac97-pcm-audio", - }, - - .probe = bf5xx_soc_platform_probe, -}; - -module_platform_driver(bf5xx_pcm_driver); - -MODULE_AUTHOR("Cliff Cai"); -MODULE_DESCRIPTION("ADI Blackfin AC97 PCM DMA module"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/blackfin/bf5xx-ac97.c b/sound/soc/blackfin/bf5xx-ac97.c deleted file mode 100644 index a040cfe29fc0..000000000000 --- a/sound/soc/blackfin/bf5xx-ac97.c +++ /dev/null @@ -1,388 +0,0 @@ -/* - * bf5xx-ac97.c -- AC97 support for the ADI blackfin chip. - * - * Author: Roy Huang - * Created: 11th. June 2007 - * Copyright: Analog Device Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#include <linux/init.h> -#include <linux/module.h> -#include <linux/platform_device.h> -#include <linux/interrupt.h> -#include <linux/wait.h> -#include <linux/delay.h> -#include <linux/slab.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/ac97_codec.h> -#include <sound/initval.h> -#include <sound/soc.h> - -#include <asm/irq.h> -#include <asm/portmux.h> -#include <linux/mutex.h> -#include <linux/gpio.h> - -#include "bf5xx-sport.h" -#include "bf5xx-ac97.h" - -/* Anomaly notes: - * 05000250 - AD1980 is running in TDM mode and RFS/TFS are generated by SPORT - * contrtoller. But, RFSDIV and TFSDIV are always set to 16*16-1, - * while the max AC97 data size is 13*16. The DIV is always larger - * than data size. AD73311 and ad2602 are not running in TDM mode. - * AD1836 and AD73322 depend on external RFS/TFS only. So, this - * anomaly does not affect blackfin sound drivers. -*/ - -static struct sport_device *ac97_sport_handle; - -void bf5xx_pcm_to_ac97(struct ac97_frame *dst, const __u16 *src, - size_t count, unsigned int chan_mask) -{ - while (count--) { - dst->ac97_tag = TAG_VALID; - if (chan_mask & SP_FL) { - dst->ac97_pcm_r = *src++; - dst->ac97_tag |= TAG_PCM_RIGHT; - } - if (chan_mask & SP_FR) { - dst->ac97_pcm_l = *src++; - dst->ac97_tag |= TAG_PCM_LEFT; - - } -#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT) - if (chan_mask & SP_SR) { - dst->ac97_sl = *src++; - dst->ac97_tag |= TAG_PCM_SL; - } - if (chan_mask & SP_SL) { - dst->ac97_sr = *src++; - dst->ac97_tag |= TAG_PCM_SR; - } - if (chan_mask & SP_LFE) { - dst->ac97_lfe = *src++; - dst->ac97_tag |= TAG_PCM_LFE; - } - if (chan_mask & SP_FC) { - dst->ac97_center = *src++; - dst->ac97_tag |= TAG_PCM_CENTER; - } -#endif - dst++; - } -} -EXPORT_SYMBOL(bf5xx_pcm_to_ac97); - -void bf5xx_ac97_to_pcm(const struct ac97_frame *src, __u16 *dst, - size_t count) -{ - while (count--) { - *(dst++) = src->ac97_pcm_l; - *(dst++) = src->ac97_pcm_r; - src++; - } -} -EXPORT_SYMBOL(bf5xx_ac97_to_pcm); - -static unsigned int sport_tx_curr_frag(struct sport_device *sport) -{ - return sport->tx_curr_frag = sport_curr_offset_tx(sport) / - sport->tx_fragsize; -} - -static void enqueue_cmd(struct snd_ac97 *ac97, __u16 addr, __u16 data) -{ - struct sport_device *sport = ac97_sport_handle; - int *cmd_count = sport->private_data; - int nextfrag = sport_tx_curr_frag(sport); - struct ac97_frame *nextwrite; - - sport_incfrag(sport, &nextfrag, 1); - - nextwrite = (struct ac97_frame *)(sport->tx_buf + - nextfrag * sport->tx_fragsize); - pr_debug("sport->tx_buf:%p, nextfrag:0x%x nextwrite:%p, cmd_count:%d\n", - sport->tx_buf, nextfrag, nextwrite, cmd_count[nextfrag]); - nextwrite[cmd_count[nextfrag]].ac97_tag |= TAG_CMD; - nextwrite[cmd_count[nextfrag]].ac97_addr = addr; - nextwrite[cmd_count[nextfrag]].ac97_data = data; - ++cmd_count[nextfrag]; - pr_debug("ac97_sport: Inserting %02x/%04x into fragment %d\n", - addr >> 8, data, nextfrag); -} - -static unsigned short bf5xx_ac97_read(struct snd_ac97 *ac97, - unsigned short reg) -{ - struct sport_device *sport_handle = ac97_sport_handle; - struct ac97_frame out_frame[2], in_frame[2]; - - pr_debug("%s enter 0x%x\n", __func__, reg); - - /* When dma descriptor is enabled, the register should not be read */ - if (sport_handle->tx_run || sport_handle->rx_run) { - pr_err("Could you send a mail to cliff.cai(a)analog.com " - "to report this?\n"); - return -EFAULT; - } - - memset(&out_frame, 0, 2 * sizeof(struct ac97_frame)); - memset(&in_frame, 0, 2 * sizeof(struct ac97_frame)); - out_frame[0].ac97_tag = TAG_VALID | TAG_CMD; - out_frame[0].ac97_addr = ((reg << 8) | 0x8000); - sport_send_and_recv(sport_handle, (unsigned char *)&out_frame, - (unsigned char *)&in_frame, - 2 * sizeof(struct ac97_frame)); - return in_frame[1].ac97_data; -} - -void bf5xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg, - unsigned short val) -{ - struct sport_device *sport_handle = ac97_sport_handle; - - pr_debug("%s enter 0x%x:0x%04x\n", __func__, reg, val); - - if (sport_handle->tx_run) { - enqueue_cmd(ac97, (reg << 8), val); /* write */ - enqueue_cmd(ac97, (reg << 8) | 0x8000, 0); /* read back */ - } else { - struct ac97_frame frame; - memset(&frame, 0, sizeof(struct ac97_frame)); - frame.ac97_tag = TAG_VALID | TAG_CMD; - frame.ac97_addr = (reg << 8); - frame.ac97_data = val; - sport_send_and_recv(sport_handle, (unsigned char *)&frame, \ - NULL, sizeof(struct ac97_frame)); - } -} - -static void bf5xx_ac97_warm_reset(struct snd_ac97 *ac97) -{ - struct sport_device *sport_handle = ac97_sport_handle; - u16 gpio = P_IDENT(sport_handle->pin_req[3]); - - pr_debug("%s enter\n", __func__); - - peripheral_free_list(sport_handle->pin_req); - gpio_request(gpio, "bf5xx-ac97"); - gpio_direction_output(gpio, 1); - udelay(2); - gpio_set_value(gpio, 0); - udelay(1); - gpio_free(gpio); - peripheral_request_list(sport_handle->pin_req, "soc-audio"); -} - -static void bf5xx_ac97_cold_reset(struct snd_ac97 *ac97) -{ -#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET - pr_debug("%s enter\n", __func__); - - /* It is specified for bf548-ezkit */ - gpio_set_value(CONFIG_SND_BF5XX_RESET_GPIO_NUM, 0); - /* Keep reset pin low for 1 ms */ - mdelay(1); - gpio_set_value(CONFIG_SND_BF5XX_RESET_GPIO_NUM, 1); - /* Wait for bit clock recover */ - mdelay(1); -#else - pr_info("%s: Not implemented\n", __func__); -#endif -} - -static struct snd_ac97_bus_ops bf5xx_ac97_ops = { - .read = bf5xx_ac97_read, - .write = bf5xx_ac97_write, - .warm_reset = bf5xx_ac97_warm_reset, - .reset = bf5xx_ac97_cold_reset, -}; - -#ifdef CONFIG_PM -static int bf5xx_ac97_suspend(struct snd_soc_dai *dai) -{ - struct sport_device *sport = snd_soc_dai_get_drvdata(dai); - - pr_debug("%s : sport %d\n", __func__, dai->id); - if (!dai->active) - return 0; - if (dai->capture_active) - sport_rx_stop(sport); - if (dai->playback_active) - sport_tx_stop(sport); - return 0; -} - -static int bf5xx_ac97_resume(struct snd_soc_dai *dai) -{ - int ret; - struct sport_device *sport = snd_soc_dai_get_drvdata(dai); - - pr_debug("%s : sport %d\n", __func__, dai->id); - if (!dai->active) - return 0; - -#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT) - ret = sport_set_multichannel(sport, 16, 0x3FF, 0x3FF, 1); -#else - ret = sport_set_multichannel(sport, 16, 0x1F, 0x1F, 1); -#endif - if (ret) { - pr_err("SPORT is busy!\n"); - return -EBUSY; - } - - ret = sport_config_rx(sport, IRFS, 0xF, 0, (16*16-1)); - if (ret) { - pr_err("SPORT is busy!\n"); - return -EBUSY; - } - - ret = sport_config_tx(sport, ITFS, 0xF, 0, (16*16-1)); - if (ret) { - pr_err("SPORT is busy!\n"); - return -EBUSY; - } - - return 0; -} - -#else -#define bf5xx_ac97_suspend NULL -#define bf5xx_ac97_resume NULL -#endif - -static struct snd_soc_dai_driver bfin_ac97_dai = { - .bus_control = true, - .suspend = bf5xx_ac97_suspend, - .resume = bf5xx_ac97_resume, - .playback = { - .stream_name = "AC97 Playback", - .channels_min = 2, -#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT) - .channels_max = 6, -#else - .channels_max = 2, -#endif - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, }, - .capture = { - .stream_name = "AC97 Capture", - .channels_min = 2, - .channels_max = 2, - .rates = SNDRV_PCM_RATE_48000, - .formats = SNDRV_PCM_FMTBIT_S16_LE, }, -}; - -static const struct snd_soc_component_driver bfin_ac97_component = { - .name = "bfin-ac97", -}; - -static int asoc_bfin_ac97_probe(struct platform_device *pdev) -{ - struct sport_device *sport_handle; - int ret; - -#ifdef CONFIG_SND_BF5XX_HAVE_COLD_RESET - /* Request PB3 as reset pin */ - ret = devm_gpio_request_one(&pdev->dev, - CONFIG_SND_BF5XX_RESET_GPIO_NUM, - GPIOF_OUT_INIT_HIGH, "SND_AD198x RESET"); - if (ret) { - dev_err(&pdev->dev, - "Failed to request GPIO_%d for reset: %d\n", - CONFIG_SND_BF5XX_RESET_GPIO_NUM, ret); - return ret; - } -#endif - - sport_handle = sport_init(pdev, 2, sizeof(struct ac97_frame), - PAGE_SIZE); - if (!sport_handle) { - ret = -ENODEV; - goto sport_err; - } - - /*SPORT works in TDM mode to simulate AC97 transfers*/ -#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT) - ret = sport_set_multichannel(sport_handle, 16, 0x3FF, 0x3FF, 1); -#else - ret = sport_set_multichannel(sport_handle, 16, 0x1F, 0x1F, 1); -#endif - if (ret) { - pr_err("SPORT is busy!\n"); - ret = -EBUSY; - goto sport_config_err; - } - - ret = sport_config_rx(sport_handle, IRFS, 0xF, 0, (16*16-1)); - if (ret) { - pr_err("SPORT is busy!\n"); - ret = -EBUSY; - goto sport_config_err; - } - - ret = sport_config_tx(sport_handle, ITFS, 0xF, 0, (16*16-1)); - if (ret) { - pr_err("SPORT is busy!\n"); - ret = -EBUSY; - goto sport_config_err; - } - - ret = snd_soc_set_ac97_ops(&bf5xx_ac97_ops); - if (ret != 0) { - dev_err(&pdev->dev, "Failed to set AC'97 ops: %d\n", ret); - goto sport_config_err; - } - - ret = snd_soc_register_component(&pdev->dev, &bfin_ac97_component, - &bfin_ac97_dai, 1); - if (ret) { - pr_err("Failed to register DAI: %d\n", ret); - goto sport_config_err; - } - - ac97_sport_handle = sport_handle; - - return 0; - -sport_config_err: - sport_done(sport_handle); -sport_err: - snd_soc_set_ac97_ops(NULL); - - return ret; -} - -static int asoc_bfin_ac97_remove(struct platform_device *pdev) -{ - struct sport_device *sport_handle = platform_get_drvdata(pdev); - - snd_soc_unregister_component(&pdev->dev); - sport_done(sport_handle); - snd_soc_set_ac97_ops(NULL); - - return 0; -} - -static struct platform_driver asoc_bfin_ac97_driver = { - .driver = { - .name = "bfin-ac97", - }, - - .probe = asoc_bfin_ac97_probe, - .remove = asoc_bfin_ac97_remove, -}; - -module_platform_driver(asoc_bfin_ac97_driver); - -MODULE_AUTHOR("Roy Huang"); -MODULE_DESCRIPTION("AC97 driver for ADI Blackfin"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/blackfin/bf5xx-ac97.h b/sound/soc/blackfin/bf5xx-ac97.h deleted file mode 100644 index a680fdc9bb42..000000000000 --- a/sound/soc/blackfin/bf5xx-ac97.h +++ /dev/null @@ -1,57 +0,0 @@ -/* - * sound/soc/blackfin/bf5xx-ac97.h - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _BF5XX_AC97_H -#define _BF5XX_AC97_H - -/* Frame format in memory, only support stereo currently */ -struct ac97_frame { - u16 ac97_tag; /* slot 0 */ - u16 ac97_addr; /* slot 1 */ - u16 ac97_data; /* slot 2 */ - u16 ac97_pcm_l; /*slot 3:front left*/ - u16 ac97_pcm_r; /*slot 4:front left*/ -#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT) - u16 ac97_mdm_l1; - u16 ac97_center; /*slot 6:center*/ - u16 ac97_sl; /*slot 7:surround left*/ - u16 ac97_sr; /*slot 8:surround right*/ - u16 ac97_lfe; /*slot 9:lfe*/ -#endif -} __attribute__ ((packed)); - -/* Speaker location */ -#define SP_FL 0x0001 -#define SP_FR 0x0010 -#define SP_FC 0x0002 -#define SP_LFE 0x0020 -#define SP_SL 0x0004 -#define SP_SR 0x0040 - -#define SP_STEREO (SP_FL | SP_FR) -#define SP_2DOT1 (SP_FL | SP_FR | SP_LFE) -#define SP_QUAD (SP_FL | SP_FR | SP_SL | SP_SR) -#define SP_5DOT1 (SP_FL | SP_FR | SP_FC | SP_LFE | SP_SL | SP_SR) - -#define TAG_VALID 0x8000 -#define TAG_CMD 0x6000 -#define TAG_PCM_LEFT 0x1000 -#define TAG_PCM_RIGHT 0x0800 -#define TAG_PCM_MDM_L1 0x0400 -#define TAG_PCM_CENTER 0x0200 -#define TAG_PCM_SL 0x0100 -#define TAG_PCM_SR 0x0080 -#define TAG_PCM_LFE 0x0040 - -void bf5xx_pcm_to_ac97(struct ac97_frame *dst, const __u16 *src, \ - size_t count, unsigned int chan_mask); - -void bf5xx_ac97_to_pcm(const struct ac97_frame *src, __u16 *dst, \ - size_t count); - -#endif diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c deleted file mode 100644 index 864df2616e10..000000000000 --- a/sound/soc/blackfin/bf5xx-ad1836.c +++ /dev/null @@ -1,109 +0,0 @@ -/* - * File: sound/soc/blackfin/bf5xx-ad1836.c - * Author: Barry Song <Barry.Song(a)analog.com> - * - * Created: Aug 4 2009 - * Description: Board driver for ad1836 sound chip - * - * Bugs: Enter bugs at http://blackfin.uclinux.org/ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/device.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/pcm_params.h> - -#include <asm/blackfin.h> -#include <asm/cacheflush.h> -#include <asm/irq.h> -#include <asm/dma.h> -#include <asm/portmux.h> - -#include "../codecs/ad1836.h" - -static struct snd_soc_card bf5xx_ad1836; - -static int bf5xx_ad1836_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - unsigned int channel_map[] = {0, 4, 1, 5, 2, 6, 3, 7}; - int ret = 0; - - /* set cpu DAI channel mapping */ - ret = snd_soc_dai_set_channel_map(cpu_dai, ARRAY_SIZE(channel_map), - channel_map, ARRAY_SIZE(channel_map), channel_map); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0xFF, 0xFF, 8, 32); - if (ret < 0) - return ret; - - return 0; -} - -#define BF5XX_AD1836_DAIFMT (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_IF | \ - SND_SOC_DAIFMT_CBM_CFM) - -static struct snd_soc_dai_link bf5xx_ad1836_dai = { - .name = "ad1836", - .stream_name = "AD1836", - .codec_dai_name = "ad1836-hifi", - .platform_name = "bfin-i2s-pcm-audio", - .dai_fmt = BF5XX_AD1836_DAIFMT, - .init = bf5xx_ad1836_init, -}; - -static struct snd_soc_card bf5xx_ad1836 = { - .name = "bfin-ad1836", - .owner = THIS_MODULE, - .dai_link = &bf5xx_ad1836_dai, - .num_links = 1, -}; - -static int bf5xx_ad1836_driver_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &bf5xx_ad1836; - const char **link_name; - int ret; - - link_name = pdev->dev.platform_data; - if (!link_name) { - dev_err(&pdev->dev, "No platform data supplied\n"); - return -EINVAL; - } - bf5xx_ad1836_dai.cpu_dai_name = link_name[0]; - bf5xx_ad1836_dai.codec_name = link_name[1]; - - card->dev = &pdev->dev; - platform_set_drvdata(pdev, card); - - ret = devm_snd_soc_register_card(&pdev->dev, card); - if (ret) - dev_err(&pdev->dev, "Failed to register card\n"); - return ret; -} - -static struct platform_driver bf5xx_ad1836_driver = { - .driver = { - .name = "bfin-snd-ad1836", - .pm = &snd_soc_pm_ops, - }, - .probe = bf5xx_ad1836_driver_probe, -}; -module_platform_driver(bf5xx_ad1836_driver); - -/* Module information */ -MODULE_AUTHOR("Barry Song"); -MODULE_DESCRIPTION("ALSA SoC AD1836 board driver"); -MODULE_LICENSE("GPL"); - diff --git a/sound/soc/blackfin/bf5xx-ad193x.c b/sound/soc/blackfin/bf5xx-ad193x.c deleted file mode 100644 index 603ad1f2b9b9..000000000000 --- a/sound/soc/blackfin/bf5xx-ad193x.c +++ /dev/null @@ -1,131 +0,0 @@ -/* - * File: sound/soc/blackfin/bf5xx-ad193x.c - * Author: Barry Song <Barry.Song(a)analog.com> - * - * Created: Thur June 4 2009 - * Description: Board driver for ad193x sound chip - * - * Bugs: Enter bugs at http://blackfin.uclinux.org/ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, see the file COPYING, or write - * to the Free Software Foundation, Inc., - * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/device.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/pcm_params.h> - -#include <asm/blackfin.h> -#include <asm/cacheflush.h> -#include <asm/irq.h> -#include <asm/dma.h> -#include <asm/portmux.h> - -#include "../codecs/ad193x.h" - -static struct snd_soc_card bf5xx_ad193x; - -static int bf5xx_ad193x_link_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - /* set the codec system clock for DAC and ADC */ - ret = snd_soc_dai_set_sysclk(codec_dai, 0, 24576000, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; - - /* set codec DAI slots, 8 channels, all channels are enabled */ - ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xFF, 0xFF, 8, 32); - if (ret < 0) - return ret; - - ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0xFF, 0xFF, 8, 32); - if (ret < 0) - return ret; - - return 0; -} - -#define BF5XX_AD193X_DAIFMT (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_IF | \ - SND_SOC_DAIFMT_CBM_CFM) - -static struct snd_soc_dai_link bf5xx_ad193x_dai[] = { - { - .name = "ad193x", - .stream_name = "AD193X", - .cpu_dai_name = "bfin-i2s.0", - .codec_dai_name ="ad193x-hifi", - .platform_name = "bfin-i2s-pcm-audio", - .codec_name = "spi0.5", - .dai_fmt = BF5XX_AD193X_DAIFMT, - .init = bf5xx_ad193x_link_init, - }, - { - .name = "ad193x", - .stream_name = "AD193X", - .cpu_dai_name = "bfin-i2s.1", - .codec_dai_name ="ad193x-hifi", - .platform_name = "bfin-i2s-pcm-audio", - .codec_name = "spi0.5", - .dai_fmt = BF5XX_AD193X_DAIFMT, - .init = bf5xx_ad193x_link_init, - }, -}; - -static struct snd_soc_card bf5xx_ad193x = { - .name = "bfin-ad193x", - .owner = THIS_MODULE, - .dai_link = &bf5xx_ad193x_dai[CONFIG_SND_BF5XX_SPORT_NUM], - .num_links = 1, -}; - -static struct platform_device *bfxx_ad193x_snd_device; - -static int __init bf5xx_ad193x_init(void) -{ - int ret; - - bfxx_ad193x_snd_device = platform_device_alloc("soc-audio", -1); - if (!bfxx_ad193x_snd_device) - return -ENOMEM; - - platform_set_drvdata(bfxx_ad193x_snd_device, &bf5xx_ad193x); - ret = platform_device_add(bfxx_ad193x_snd_device); - - if (ret) - platform_device_put(bfxx_ad193x_snd_device); - - return ret; -} - -static void __exit bf5xx_ad193x_exit(void) -{ - platform_device_unregister(bfxx_ad193x_snd_device); -} - -module_init(bf5xx_ad193x_init); -module_exit(bf5xx_ad193x_exit); - -/* Module information */ -MODULE_AUTHOR("Barry Song"); -MODULE_DESCRIPTION("ALSA SoC AD193X board driver"); -MODULE_LICENSE("GPL"); - diff --git a/sound/soc/blackfin/bf5xx-ad1980.c b/sound/soc/blackfin/bf5xx-ad1980.c deleted file mode 100644 index 0fa81a523b8a..000000000000 --- a/sound/soc/blackfin/bf5xx-ad1980.c +++ /dev/null @@ -1,109 +0,0 @@ -/* - * File: sound/soc/blackfin/bf5xx-ad1980.c - * Author: Cliff Cai <Cliff.Cai(a)analog.com> - * - * Created: Tue June 06 2008 - * Description: Board driver for AD1980/1 audio codec - * - * Modified: - * Copyright 2008 Analog Devices Inc. - * - * Bugs: Enter bugs at http://blackfin.uclinux.org/ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, see the file COPYING, or write - * to the Free Software Foundation, Inc., - * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA - */ - -/* - * WARNING: - * - * Because Analog Devices Inc. discontinued the ad1980 sound chip since - * Sep. 2009, this ad1980 driver is not maintained, tested and supported - * by ADI now. - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/device.h> -#include <asm/dma.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> - -#include <linux/gpio.h> -#include <asm/portmux.h> - -#include "bf5xx-ac97.h" - -static struct snd_soc_card bf5xx_board; - -static struct snd_soc_dai_link bf5xx_board_dai[] = { - { - .name = "AC97", - .stream_name = "AC97 HiFi", - .cpu_dai_name = "bfin-ac97.0", - .codec_dai_name = "ad1980-hifi", - .platform_name = "bfin-ac97-pcm-audio", - .codec_name = "ad1980", - }, - { - .name = "AC97", - .stream_name = "AC97 HiFi", - .cpu_dai_name = "bfin-ac97.1", - .codec_dai_name = "ad1980-hifi", - .platform_name = "bfin-ac97-pcm-audio", - .codec_name = "ad1980", - }, -}; - -static struct snd_soc_card bf5xx_board = { - .name = "bfin-ad1980", - .owner = THIS_MODULE, - .dai_link = &bf5xx_board_dai[CONFIG_SND_BF5XX_SPORT_NUM], - .num_links = 1, -}; - -static struct platform_device *bf5xx_board_snd_device; - -static int __init bf5xx_board_init(void) -{ - int ret; - - bf5xx_board_snd_device = platform_device_alloc("soc-audio", -1); - if (!bf5xx_board_snd_device) - return -ENOMEM; - - platform_set_drvdata(bf5xx_board_snd_device, &bf5xx_board); - ret = platform_device_add(bf5xx_board_snd_device); - - if (ret) - platform_device_put(bf5xx_board_snd_device); - - return ret; -} - -static void __exit bf5xx_board_exit(void) -{ - platform_device_unregister(bf5xx_board_snd_device); -} - -module_init(bf5xx_board_init); -module_exit(bf5xx_board_exit); - -/* Module information */ -MODULE_AUTHOR("Cliff Cai"); -MODULE_DESCRIPTION("ALSA SoC AD1980/1 BF5xx board (Obsolete)"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/blackfin/bf5xx-ad73311.c b/sound/soc/blackfin/bf5xx-ad73311.c deleted file mode 100644 index 786bbdd96e7c..000000000000 --- a/sound/soc/blackfin/bf5xx-ad73311.c +++ /dev/null @@ -1,212 +0,0 @@ -/* - * File: sound/soc/blackfin/bf5xx-ad73311.c - * Author: Cliff Cai <Cliff.Cai(a)analog.com> - * - * Created: Thur Sep 25 2008 - * Description: Board driver for ad73311 sound chip - * - * Modified: - * Copyright 2008 Analog Devices Inc. - * - * Bugs: Enter bugs at http://blackfin.uclinux.org/ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, see the file COPYING, or write - * to the Free Software Foundation, Inc., - * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/device.h> -#include <linux/delay.h> -#include <linux/gpio.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/pcm_params.h> - -#include <asm/blackfin.h> -#include <asm/cacheflush.h> -#include <asm/irq.h> -#include <asm/dma.h> -#include <asm/portmux.h> - -#include "../codecs/ad73311.h" -#include "bf5xx-sport.h" - -#if CONFIG_SND_BF5XX_SPORT_NUM == 0 -#define bfin_write_SPORT_TCR1 bfin_write_SPORT0_TCR1 -#define bfin_read_SPORT_TCR1 bfin_read_SPORT0_TCR1 -#define bfin_write_SPORT_TCR2 bfin_write_SPORT0_TCR2 -#define bfin_write_SPORT_TX16 bfin_write_SPORT0_TX16 -#define bfin_read_SPORT_STAT bfin_read_SPORT0_STAT -#else -#define bfin_write_SPORT_TCR1 bfin_write_SPORT1_TCR1 -#define bfin_read_SPORT_TCR1 bfin_read_SPORT1_TCR1 -#define bfin_write_SPORT_TCR2 bfin_write_SPORT1_TCR2 -#define bfin_write_SPORT_TX16 bfin_write_SPORT1_TX16 -#define bfin_read_SPORT_STAT bfin_read_SPORT1_STAT -#endif - -#define GPIO_SE CONFIG_SND_BFIN_AD73311_SE - -static struct snd_soc_card bf5xx_ad73311; - -static int snd_ad73311_startup(void) -{ - pr_debug("%s enter\n", __func__); - - /* Pull up SE pin on AD73311L */ - gpio_set_value(GPIO_SE, 1); - return 0; -} - -static int snd_ad73311_configure(void) -{ - unsigned short ctrl_regs[6]; - unsigned short status = 0; - int count = 0; - - /* DMCLK = MCLK = 16.384 MHz - * SCLK = DMCLK/8 = 2.048 MHz - * Sample Rate = DMCLK/2048 = 8 KHz - */ - ctrl_regs[0] = AD_CONTROL | AD_WRITE | CTRL_REG_B | REGB_MCDIV(0) | \ - REGB_SCDIV(0) | REGB_DIRATE(0); - ctrl_regs[1] = AD_CONTROL | AD_WRITE | CTRL_REG_C | REGC_PUDEV | \ - REGC_PUADC | REGC_PUDAC | REGC_PUREF | REGC_REFUSE ; - ctrl_regs[2] = AD_CONTROL | AD_WRITE | CTRL_REG_D | REGD_OGS(2) | \ - REGD_IGS(2); - ctrl_regs[3] = AD_CONTROL | AD_WRITE | CTRL_REG_E | REGE_DA(0x1f); - ctrl_regs[4] = AD_CONTROL | AD_WRITE | CTRL_REG_F | REGF_SEEN ; - ctrl_regs[5] = AD_CONTROL | AD_WRITE | CTRL_REG_A | REGA_MODE_DATA; - - local_irq_disable(); - snd_ad73311_startup(); - udelay(1); - - bfin_write_SPORT_TCR1(TFSR); - bfin_write_SPORT_TCR2(0xF); - SSYNC(); - - /* SPORT Tx Register is a 8 x 16 FIFO, all the data can be put to - * FIFO before enable SPORT to transfer the data - */ - for (count = 0; count < 6; count++) - bfin_write_SPORT_TX16(ctrl_regs[count]); - SSYNC(); - bfin_write_SPORT_TCR1(bfin_read_SPORT_TCR1() | TSPEN); - SSYNC(); - - /* When TUVF is set, the data is already send out */ - while (!(status & TUVF) && ++count < 10000) { - udelay(1); - status = bfin_read_SPORT_STAT(); - SSYNC(); - } - bfin_write_SPORT_TCR1(bfin_read_SPORT_TCR1() & ~TSPEN); - SSYNC(); - local_irq_enable(); - - if (count >= 10000) { - printk(KERN_ERR "ad73311: failed to configure codec\n"); - return -1; - } - return 0; -} - -static int bf5xx_probe(struct snd_soc_card *card) -{ - int err; - if (gpio_request(GPIO_SE, "AD73311_SE")) { - printk(KERN_ERR "%s: Failed ro request GPIO_%d\n", __func__, GPIO_SE); - return -EBUSY; - } - - gpio_direction_output(GPIO_SE, 0); - - err = snd_ad73311_configure(); - if (err < 0) - return -EFAULT; - - return 0; -} - -#define BF5XX_AD7311_DAI_FMT (SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF | \ - SND_SOC_DAIFMT_CBM_CFM) - -static struct snd_soc_dai_link bf5xx_ad73311_dai[] = { - { - .name = "ad73311", - .stream_name = "AD73311", - .cpu_dai_name = "bfin-i2s.0", - .codec_dai_name = "ad73311-hifi", - .platform_name = "bfin-i2s-pcm-audio", - .codec_name = "ad73311", - .dai_fmt = BF5XX_AD7311_DAI_FMT, - }, - { - .name = "ad73311", - .stream_name = "AD73311", - .cpu_dai_name = "bfin-i2s.1", - .codec_dai_name = "ad73311-hifi", - .platform_name = "bfin-i2s-pcm-audio", - .codec_name = "ad73311", - .dai_fmt = BF5XX_AD7311_DAI_FMT, - }, -}; - -static struct snd_soc_card bf5xx_ad73311 = { - .name = "bfin-ad73311", - .owner = THIS_MODULE, - .probe = bf5xx_probe, - .dai_link = &bf5xx_ad73311_dai[CONFIG_SND_BF5XX_SPORT_NUM], - .num_links = 1, -}; - -static struct platform_device *bf5xx_ad73311_snd_device; - -static int __init bf5xx_ad73311_init(void) -{ - int ret; - - pr_debug("%s enter\n", __func__); - bf5xx_ad73311_snd_device = platform_device_alloc("soc-audio", -1); - if (!bf5xx_ad73311_snd_device) - return -ENOMEM; - - platform_set_drvdata(bf5xx_ad73311_snd_device, &bf5xx_ad73311); - ret = platform_device_add(bf5xx_ad73311_snd_device); - - if (ret) - platform_device_put(bf5xx_ad73311_snd_device); - - return ret; -} - -static void __exit bf5xx_ad73311_exit(void) -{ - pr_debug("%s enter\n", __func__); - platform_device_unregister(bf5xx_ad73311_snd_device); -} - -module_init(bf5xx_ad73311_init); -module_exit(bf5xx_ad73311_exit); - -/* Module information */ -MODULE_AUTHOR("Cliff Cai"); -MODULE_DESCRIPTION("ALSA SoC AD73311 Blackfin"); -MODULE_LICENSE("GPL"); - diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.c b/sound/soc/blackfin/bf5xx-i2s-pcm.c deleted file mode 100644 index 51cae76f14e6..000000000000 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.c +++ /dev/null @@ -1,373 +0,0 @@ -/* - * File: sound/soc/blackfin/bf5xx-i2s-pcm.c - * Author: Cliff Cai <Cliff.Cai(a)analog.com> - * - * Created: Tue June 06 2008 - * Description: DMA driver for i2s codec - * - * Modified: - * Copyright 2008 Analog Devices Inc. - * - * Bugs: Enter bugs at http://blackfin.uclinux.org/ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, see the file COPYING, or write - * to the Free Software Foundation, Inc., - * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA - */ - -#include <linux/module.h> -#include <linux/init.h> -#include <linux/platform_device.h> -#include <linux/dma-mapping.h> -#include <linux/gfp.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> - -#include <asm/dma.h> - -#include "bf5xx-sport.h" -#include "bf5xx-i2s-pcm.h" - -static void bf5xx_dma_irq(void *data) -{ - struct snd_pcm_substream *pcm = data; - snd_pcm_period_elapsed(pcm); -} - -static const struct snd_pcm_hardware bf5xx_pcm_hardware = { - .info = SNDRV_PCM_INFO_INTERLEAVED | - SNDRV_PCM_INFO_MMAP_VALID | - SNDRV_PCM_INFO_BLOCK_TRANSFER, - .period_bytes_min = 32, - .period_bytes_max = 0x10000, - .periods_min = 1, - .periods_max = PAGE_SIZE/32, - .buffer_bytes_max = 0x20000, /* 128 kbytes */ - .fifo_size = 16, -}; - -static int bf5xx_pcm_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - unsigned int buffer_size = params_buffer_bytes(params); - struct bf5xx_i2s_pcm_data *dma_data; - - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - if (dma_data->tdm_mode) - buffer_size = buffer_size / params_channels(params) * 8; - - return snd_pcm_lib_malloc_pages(substream, buffer_size); -} - -static int bf5xx_pcm_hw_free(struct snd_pcm_substream *substream) -{ - snd_pcm_lib_free_pages(substream); - - return 0; -} - -static int bf5xx_pcm_prepare(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - struct sport_device *sport = runtime->private_data; - int period_bytes = frames_to_bytes(runtime, runtime->period_size); - struct bf5xx_i2s_pcm_data *dma_data; - - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - if (dma_data->tdm_mode) - period_bytes = period_bytes / runtime->channels * 8; - - pr_debug("%s enter\n", __func__); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - sport_set_tx_callback(sport, bf5xx_dma_irq, substream); - sport_config_tx_dma(sport, runtime->dma_area, - runtime->periods, period_bytes); - } else { - sport_set_rx_callback(sport, bf5xx_dma_irq, substream); - sport_config_rx_dma(sport, runtime->dma_area, - runtime->periods, period_bytes); - } - - return 0; -} - -static int bf5xx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - struct sport_device *sport = runtime->private_data; - int ret = 0; - - pr_debug("%s enter\n", __func__); - switch (cmd) { - case SNDRV_PCM_TRIGGER_START: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - sport_tx_start(sport); - else - sport_rx_start(sport); - break; - case SNDRV_PCM_TRIGGER_STOP: - case SNDRV_PCM_TRIGGER_SUSPEND: - case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - sport_tx_stop(sport); - else - sport_rx_stop(sport); - break; - default: - ret = -EINVAL; - } - - return ret; -} - -static snd_pcm_uframes_t bf5xx_pcm_pointer(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - struct sport_device *sport = runtime->private_data; - unsigned int diff; - snd_pcm_uframes_t frames; - struct bf5xx_i2s_pcm_data *dma_data; - - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - pr_debug("%s enter\n", __func__); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - diff = sport_curr_offset_tx(sport); - } else { - diff = sport_curr_offset_rx(sport); - } - - /* - * TX at least can report one frame beyond the end of the - * buffer if we hit the wraparound case - clamp to within the - * buffer as the ALSA APIs require. - */ - if (diff == snd_pcm_lib_buffer_bytes(substream)) - diff = 0; - - frames = bytes_to_frames(substream->runtime, diff); - if (dma_data->tdm_mode) - frames = frames * runtime->channels / 8; - - return frames; -} - -static int bf5xx_pcm_open(struct snd_pcm_substream *substream) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *cpu_dai = rtd->cpu_dai; - struct sport_device *sport_handle = snd_soc_dai_get_drvdata(cpu_dai); - struct snd_pcm_runtime *runtime = substream->runtime; - struct snd_dma_buffer *buf = &substream->dma_buffer; - struct bf5xx_i2s_pcm_data *dma_data; - int ret; - - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - pr_debug("%s enter\n", __func__); - - snd_soc_set_runtime_hwparams(substream, &bf5xx_pcm_hardware); - if (dma_data->tdm_mode) - runtime->hw.buffer_bytes_max /= 4; - else - runtime->hw.info |= SNDRV_PCM_INFO_MMAP; - - ret = snd_pcm_hw_constraint_integer(runtime, - SNDRV_PCM_HW_PARAM_PERIODS); - if (ret < 0) - goto out; - - if (sport_handle != NULL) { - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - sport_handle->tx_buf = buf->area; - else - sport_handle->rx_buf = buf->area; - - runtime->private_data = sport_handle; - } else { - pr_err("sport_handle is NULL\n"); - return -1; - } - return 0; - - out: - return ret; -} - -static int bf5xx_pcm_mmap(struct snd_pcm_substream *substream, - struct vm_area_struct *vma) -{ - struct snd_pcm_runtime *runtime = substream->runtime; - size_t size = vma->vm_end - vma->vm_start; - vma->vm_start = (unsigned long)runtime->dma_area; - vma->vm_end = vma->vm_start + size; - vma->vm_flags |= VM_SHARED; - - return 0 ; -} - -static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, - int channel, unsigned long pos, - void *buf, unsigned long count) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int sample_size = runtime->sample_bits / 8; - struct bf5xx_i2s_pcm_data *dma_data; - unsigned int i; - void *src, *dst; - - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - if (dma_data->tdm_mode) { - pos = bytes_to_frames(runtime, pos); - count = bytes_to_frames(runtime, count); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - src = buf; - dst = runtime->dma_area; - dst += pos * sample_size * 8; - - while (count--) { - for (i = 0; i < runtime->channels; i++) { - memcpy(dst + dma_data->map[i] * - sample_size, src, sample_size); - src += sample_size; - } - dst += 8 * sample_size; - } - } else { - src = runtime->dma_area; - src += pos * sample_size * 8; - dst = buf; - - while (count--) { - for (i = 0; i < runtime->channels; i++) { - memcpy(dst, src + dma_data->map[i] * - sample_size, sample_size); - dst += sample_size; - } - src += 8 * sample_size; - } - } - } else { - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - src = buf; - dst = runtime->dma_area; - dst += pos; - } else { - src = runtime->dma_area; - src += pos; - dst = buf; - } - - memcpy(dst, src, count); - } - - return 0; -} - -static int bf5xx_pcm_copy_user(struct snd_pcm_substream *substream, - int channel, unsigned long pos, - void __user *buf, unsigned long count) -{ - return bf5xx_pcm_copy(substream, channel, pos, (void *)buf, count); -} - -static int bf5xx_pcm_silence(struct snd_pcm_substream *substream, - int channel, unsigned long pos, - unsigned long count) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_pcm_runtime *runtime = substream->runtime; - unsigned int sample_size = runtime->sample_bits / 8; - void *buf = runtime->dma_area; - struct bf5xx_i2s_pcm_data *dma_data; - unsigned int offset, samples; - - dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream); - - if (dma_data->tdm_mode) { - offset = bytes_to_frames(runtime, pos) * 8 * sample_size; - samples = bytes_to_frames(runtime, count) * 8; - } else { - offset = pos; - samples = bytes_to_samples(runtime, count); - } - - snd_pcm_format_set_silence(runtime->format, buf + offset, samples); - - return 0; -} - -static const struct snd_pcm_ops bf5xx_pcm_i2s_ops = { - .open = bf5xx_pcm_open, - .ioctl = snd_pcm_lib_ioctl, - .hw_params = bf5xx_pcm_hw_params, - .hw_free = bf5xx_pcm_hw_free, - .prepare = bf5xx_pcm_prepare, - .trigger = bf5xx_pcm_trigger, - .pointer = bf5xx_pcm_pointer, - .mmap = bf5xx_pcm_mmap, - .copy_user = bf5xx_pcm_copy_user, - .copy_kernel = bf5xx_pcm_copy, - .fill_silence = bf5xx_pcm_silence, -}; - -static int bf5xx_pcm_i2s_new(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_card *card = rtd->card->snd_card; - size_t size = bf5xx_pcm_hardware.buffer_bytes_max; - int ret; - - pr_debug("%s enter\n", __func__); - ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32)); - if (ret) - return ret; - - return snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, - SNDRV_DMA_TYPE_DEV, card->dev, size, size); -} - -static struct snd_soc_platform_driver bf5xx_i2s_soc_platform = { - .ops = &bf5xx_pcm_i2s_ops, - .pcm_new = bf5xx_pcm_i2s_new, -}; - -static int bfin_i2s_soc_platform_probe(struct platform_device *pdev) -{ - return devm_snd_soc_register_platform(&pdev->dev, - &bf5xx_i2s_soc_platform); -} - -static struct platform_driver bfin_i2s_pcm_driver = { - .driver = { - .name = "bfin-i2s-pcm-audio", - }, - - .probe = bfin_i2s_soc_platform_probe, -}; - -module_platform_driver(bfin_i2s_pcm_driver); - -MODULE_AUTHOR("Cliff Cai"); -MODULE_DESCRIPTION("ADI Blackfin I2S PCM DMA module"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/blackfin/bf5xx-i2s-pcm.h b/sound/soc/blackfin/bf5xx-i2s-pcm.h deleted file mode 100644 index 1f0435249f88..000000000000 --- a/sound/soc/blackfin/bf5xx-i2s-pcm.h +++ /dev/null @@ -1,17 +0,0 @@ -/* - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef _BF5XX_TDM_PCM_H -#define _BF5XX_TDM_PCM_H - -#define BFIN_TDM_DAI_MAX_SLOTS 8 - -struct bf5xx_i2s_pcm_data { - unsigned int map[BFIN_TDM_DAI_MAX_SLOTS]; - bool tdm_mode; -}; - -#endif diff --git a/sound/soc/blackfin/bf5xx-i2s.c b/sound/soc/blackfin/bf5xx-i2s.c deleted file mode 100644 index b69aeef6418e..000000000000 --- a/sound/soc/blackfin/bf5xx-i2s.c +++ /dev/null @@ -1,391 +0,0 @@ -/* - * File: sound/soc/blackfin/bf5xx-i2s.c - * Author: Cliff Cai <Cliff.Cai(a)analog.com> - * - * Created: Tue June 06 2008 - * Description: Blackfin I2S CPU DAI driver - * - * Modified: - * Copyright 2008 Analog Devices Inc. - * - * Bugs: Enter bugs at http://blackfin.uclinux.org/ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, see the file COPYING, or write - * to the Free Software Foundation, Inc., - * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA - */ - -#include <linux/init.h> -#include <linux/module.h> -#include <linux/device.h> -#include <linux/delay.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/initval.h> -#include <sound/soc.h> - -#include <asm/irq.h> -#include <asm/portmux.h> -#include <linux/mutex.h> -#include <linux/gpio.h> - -#include "bf5xx-sport.h" -#include "bf5xx-i2s-pcm.h" - -struct bf5xx_i2s_port { - u16 tcr1; - u16 rcr1; - u16 tcr2; - u16 rcr2; - int configured; - - unsigned int slots; - unsigned int tx_mask; - unsigned int rx_mask; - - struct bf5xx_i2s_pcm_data tx_dma_data; - struct bf5xx_i2s_pcm_data rx_dma_data; -}; - -static int bf5xx_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) -{ - struct sport_device *sport_handle = snd_soc_dai_get_drvdata(cpu_dai); - struct bf5xx_i2s_port *bf5xx_i2s = sport_handle->private_data; - int ret = 0; - - /* interface format:support I2S,slave mode */ - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: - bf5xx_i2s->tcr1 |= TFSR | TCKFE; - bf5xx_i2s->rcr1 |= RFSR | RCKFE; - bf5xx_i2s->tcr2 |= TSFSE; - bf5xx_i2s->rcr2 |= RSFSE; - break; - case SND_SOC_DAIFMT_DSP_A: - bf5xx_i2s->tcr1 |= TFSR; - bf5xx_i2s->rcr1 |= RFSR; - break; - case SND_SOC_DAIFMT_LEFT_J: - ret = -EINVAL; - break; - default: - dev_err(cpu_dai->dev, "%s: Unknown DAI format type\n", - __func__); - ret = -EINVAL; - break; - } - - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - break; - case SND_SOC_DAIFMT_CBS_CFS: - case SND_SOC_DAIFMT_CBM_CFS: - case SND_SOC_DAIFMT_CBS_CFM: - ret = -EINVAL; - break; - default: - dev_err(cpu_dai->dev, "%s: Unknown DAI master type\n", - __func__); - ret = -EINVAL; - break; - } - - return ret; -} - -static int bf5xx_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct sport_device *sport_handle = snd_soc_dai_get_drvdata(dai); - struct bf5xx_i2s_port *bf5xx_i2s = sport_handle->private_data; - int ret = 0; - - bf5xx_i2s->tcr2 &= ~0x1f; - bf5xx_i2s->rcr2 &= ~0x1f; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - bf5xx_i2s->tcr2 |= 7; - bf5xx_i2s->rcr2 |= 7; - sport_handle->wdsize = 1; - break; - case SNDRV_PCM_FORMAT_S16_LE: - bf5xx_i2s->tcr2 |= 15; - bf5xx_i2s->rcr2 |= 15; - sport_handle->wdsize = 2; - break; - case SNDRV_PCM_FORMAT_S24_LE: - bf5xx_i2s->tcr2 |= 23; - bf5xx_i2s->rcr2 |= 23; - sport_handle->wdsize = 3; - break; - case SNDRV_PCM_FORMAT_S32_LE: - bf5xx_i2s->tcr2 |= 31; - bf5xx_i2s->rcr2 |= 31; - sport_handle->wdsize = 4; - break; - } - - if (!bf5xx_i2s->configured) { - /* - * TX and RX are not independent,they are enabled at the - * same time, even if only one side is running. So, we - * need to configure both of them at the time when the first - * stream is opened. - * - * CPU DAI:slave mode. - */ - bf5xx_i2s->configured = 1; - ret = sport_config_rx(sport_handle, bf5xx_i2s->rcr1, - bf5xx_i2s->rcr2, 0, 0); - if (ret) { - dev_err(dai->dev, "SPORT is busy!\n"); - return -EBUSY; - } - - ret = sport_config_tx(sport_handle, bf5xx_i2s->tcr1, - bf5xx_i2s->tcr2, 0, 0); - if (ret) { - dev_err(dai->dev, "SPORT is busy!\n"); - return -EBUSY; - } - } - - return 0; -} - -static void bf5xx_i2s_shutdown(struct snd_pcm_substream *substream, - struct snd_soc_dai *dai) -{ - struct sport_device *sport_handle = snd_soc_dai_get_drvdata(dai); - struct bf5xx_i2s_port *bf5xx_i2s = sport_handle->private_data; - - dev_dbg(dai->dev, "%s enter\n", __func__); - /* No active stream, SPORT is allowed to be configured again. */ - if (!dai->active) - bf5xx_i2s->configured = 0; -} - -static int bf5xx_i2s_set_channel_map(struct snd_soc_dai *dai, - unsigned int tx_num, unsigned int *tx_slot, - unsigned int rx_num, unsigned int *rx_slot) -{ - struct sport_device *sport_handle = snd_soc_dai_get_drvdata(dai); - struct bf5xx_i2s_port *bf5xx_i2s = sport_handle->private_data; - unsigned int tx_mapped = 0, rx_mapped = 0; - unsigned int slot; - int i; - - if ((tx_num > BFIN_TDM_DAI_MAX_SLOTS) || - (rx_num > BFIN_TDM_DAI_MAX_SLOTS)) - return -EINVAL; - - for (i = 0; i < tx_num; i++) { - slot = tx_slot[i]; - if ((slot < BFIN_TDM_DAI_MAX_SLOTS) && - (!(tx_mapped & (1 << slot)))) { - bf5xx_i2s->tx_dma_data.map[i] = slot; - tx_mapped |= 1 << slot; - } else - return -EINVAL; - } - for (i = 0; i < rx_num; i++) { - slot = rx_slot[i]; - if ((slot < BFIN_TDM_DAI_MAX_SLOTS) && - (!(rx_mapped & (1 << slot)))) { - bf5xx_i2s->rx_dma_data.map[i] = slot; - rx_mapped |= 1 << slot; - } else - return -EINVAL; - } - - return 0; -} - -static int bf5xx_i2s_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, - unsigned int rx_mask, int slots, int width) -{ - struct sport_device *sport_handle = snd_soc_dai_get_drvdata(dai); - struct bf5xx_i2s_port *bf5xx_i2s = sport_handle->private_data; - - if (slots % 8 != 0 || slots > 8) - return -EINVAL; - - if (width != 32) - return -EINVAL; - - bf5xx_i2s->slots = slots; - bf5xx_i2s->tx_mask = tx_mask; - bf5xx_i2s->rx_mask = rx_mask; - - bf5xx_i2s->tx_dma_data.tdm_mode = slots != 0; - bf5xx_i2s->rx_dma_data.tdm_mode = slots != 0; - - return sport_set_multichannel(sport_handle, slots, tx_mask, rx_mask, 0); -} - -#ifdef CONFIG_PM -static int bf5xx_i2s_suspend(struct snd_soc_dai *dai) -{ - struct sport_device *sport_handle = snd_soc_dai_get_drvdata(dai); - - dev_dbg(dai->dev, "%s : sport %d\n", __func__, dai->id); - - if (dai->capture_active) - sport_rx_stop(sport_handle); - if (dai->playback_active) - sport_tx_stop(sport_handle); - return 0; -} - -static int bf5xx_i2s_resume(struct snd_soc_dai *dai) -{ - struct sport_device *sport_handle = snd_soc_dai_get_drvdata(dai); - struct bf5xx_i2s_port *bf5xx_i2s = sport_handle->private_data; - int ret; - - dev_dbg(dai->dev, "%s : sport %d\n", __func__, dai->id); - - ret = sport_config_rx(sport_handle, bf5xx_i2s->rcr1, - bf5xx_i2s->rcr2, 0, 0); - if (ret) { - dev_err(dai->dev, "SPORT is busy!\n"); - return -EBUSY; - } - - ret = sport_config_tx(sport_handle, bf5xx_i2s->tcr1, - bf5xx_i2s->tcr2, 0, 0); - if (ret) { - dev_err(dai->dev, "SPORT is busy!\n"); - return -EBUSY; - } - - return sport_set_multichannel(sport_handle, bf5xx_i2s->slots, - bf5xx_i2s->tx_mask, bf5xx_i2s->rx_mask, 0); -} - -#else -#define bf5xx_i2s_suspend NULL -#define bf5xx_i2s_resume NULL -#endif - -static int bf5xx_i2s_dai_probe(struct snd_soc_dai *dai) -{ - struct sport_device *sport_handle = snd_soc_dai_get_drvdata(dai); - struct bf5xx_i2s_port *bf5xx_i2s = sport_handle->private_data; - unsigned int i; - - for (i = 0; i < BFIN_TDM_DAI_MAX_SLOTS; i++) { - bf5xx_i2s->tx_dma_data.map[i] = i; - bf5xx_i2s->rx_dma_data.map[i] = i; - } - - dai->playback_dma_data = &bf5xx_i2s->tx_dma_data; - dai->capture_dma_data = &bf5xx_i2s->rx_dma_data; - - return 0; -} - -#define BF5XX_I2S_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ - SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \ - SNDRV_PCM_RATE_96000) - -#define BF5XX_I2S_FORMATS \ - (SNDRV_PCM_FMTBIT_S8 | \ - SNDRV_PCM_FMTBIT_S16_LE | \ - SNDRV_PCM_FMTBIT_S24_LE | \ - SNDRV_PCM_FMTBIT_S32_LE) - -static const struct snd_soc_dai_ops bf5xx_i2s_dai_ops = { - .shutdown = bf5xx_i2s_shutdown, - .hw_params = bf5xx_i2s_hw_params, - .set_fmt = bf5xx_i2s_set_dai_fmt, - .set_tdm_slot = bf5xx_i2s_set_tdm_slot, - .set_channel_map = bf5xx_i2s_set_channel_map, -}; - -static struct snd_soc_dai_driver bf5xx_i2s_dai = { - .probe = bf5xx_i2s_dai_probe, - .suspend = bf5xx_i2s_suspend, - .resume = bf5xx_i2s_resume, - .playback = { - .channels_min = 2, - .channels_max = 8, - .rates = BF5XX_I2S_RATES, - .formats = BF5XX_I2S_FORMATS,}, - .capture = { - .channels_min = 2, - .channels_max = 8, - .rates = BF5XX_I2S_RATES, - .formats = BF5XX_I2S_FORMATS,}, - .ops = &bf5xx_i2s_dai_ops, -}; - -static const struct snd_soc_component_driver bf5xx_i2s_component = { - .name = "bf5xx-i2s", -}; - -static int bf5xx_i2s_probe(struct platform_device *pdev) -{ - struct sport_device *sport_handle; - int ret; - - /* configure SPORT for I2S */ - sport_handle = sport_init(pdev, 4, 8 * sizeof(u32), - sizeof(struct bf5xx_i2s_port)); - if (!sport_handle) - return -ENODEV; - - /* register with the ASoC layers */ - ret = snd_soc_register_component(&pdev->dev, &bf5xx_i2s_component, - &bf5xx_i2s_dai, 1); - if (ret) { - dev_err(&pdev->dev, "Failed to register DAI: %d\n", ret); - sport_done(sport_handle); - return ret; - } - - return 0; -} - -static int bf5xx_i2s_remove(struct platform_device *pdev) -{ - struct sport_device *sport_handle = platform_get_drvdata(pdev); - - dev_dbg(&pdev->dev, "%s enter\n", __func__); - - snd_soc_unregister_component(&pdev->dev); - sport_done(sport_handle); - - return 0; -} - -static struct platform_driver bfin_i2s_driver = { - .probe = bf5xx_i2s_probe, - .remove = bf5xx_i2s_remove, - .driver = { - .name = "bfin-i2s", - }, -}; - -module_platform_driver(bfin_i2s_driver); - -/* Module information */ -MODULE_AUTHOR("Cliff Cai"); -MODULE_DESCRIPTION("I2S driver for ADI Blackfin"); -MODULE_LICENSE("GPL"); - diff --git a/sound/soc/blackfin/bf5xx-sport.c b/sound/soc/blackfin/bf5xx-sport.c deleted file mode 100644 index 9dfa1241ea66..000000000000 --- a/sound/soc/blackfin/bf5xx-sport.c +++ /dev/null @@ -1,1102 +0,0 @@ -/* - * File: bf5xx_sport.c - * Based on: - * Author: Roy Huang <roy.huang(a)analog.com> - * - * Created: Tue Sep 21 10:52:42 CEST 2004 - * Description: - * Blackfin SPORT Driver - * - * Copyright 2004-2007 Analog Devices Inc. - * - * Bugs: Enter bugs at http://blackfin.uclinux.org/ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, see the file COPYING, or write - * to the Free Software Foundation, Inc., - * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA - */ - -#include <linux/kernel.h> -#include <linux/slab.h> -#include <linux/delay.h> -#include <linux/dma-mapping.h> -#include <linux/gpio.h> -#include <linux/bug.h> -#include <linux/module.h> -#include <asm/portmux.h> -#include <asm/dma.h> -#include <asm/blackfin.h> -#include <asm/cacheflush.h> - -#include "bf5xx-sport.h" -/* delay between frame sync pulse and first data bit in multichannel mode */ -#define FRAME_DELAY (1<<12) - -/* note: multichannel is in units of 8 channels, - * tdm_count is # channels NOT / 8 ! */ -int sport_set_multichannel(struct sport_device *sport, - int tdm_count, u32 tx_mask, u32 rx_mask, int packed) -{ - pr_debug("%s tdm_count=%d tx_mask:0x%08x rx_mask:0x%08x packed=%d\n", - __func__, tdm_count, tx_mask, rx_mask, packed); - - if ((sport->regs->tcr1 & TSPEN) || (sport->regs->rcr1 & RSPEN)) - return -EBUSY; - - if (tdm_count & 0x7) - return -EINVAL; - - if (tdm_count > 32) - return -EINVAL; /* Only support less than 32 channels now */ - - if (tdm_count) { - sport->regs->mcmc1 = ((tdm_count>>3)-1) << 12; - sport->regs->mcmc2 = FRAME_DELAY | MCMEN | \ - (packed ? (MCDTXPE|MCDRXPE) : 0); - - sport->regs->mtcs0 = tx_mask; - sport->regs->mrcs0 = rx_mask; - sport->regs->mtcs1 = 0; - sport->regs->mrcs1 = 0; - sport->regs->mtcs2 = 0; - sport->regs->mrcs2 = 0; - sport->regs->mtcs3 = 0; - sport->regs->mrcs3 = 0; - } else { - sport->regs->mcmc1 = 0; - sport->regs->mcmc2 = 0; - - sport->regs->mtcs0 = 0; - sport->regs->mrcs0 = 0; - } - - sport->regs->mtcs1 = 0; sport->regs->mtcs2 = 0; sport->regs->mtcs3 = 0; - sport->regs->mrcs1 = 0; sport->regs->mrcs2 = 0; sport->regs->mrcs3 = 0; - - SSYNC(); - - return 0; -} -EXPORT_SYMBOL(sport_set_multichannel); - -int sport_config_rx(struct sport_device *sport, unsigned int rcr1, - unsigned int rcr2, unsigned int clkdiv, unsigned int fsdiv) -{ - if ((sport->regs->tcr1 & TSPEN) || (sport->regs->rcr1 & RSPEN)) - return -EBUSY; - - sport->regs->rcr1 = rcr1; - sport->regs->rcr2 = rcr2; - sport->regs->rclkdiv = clkdiv; - sport->regs->rfsdiv = fsdiv; - - SSYNC(); - - return 0; -} -EXPORT_SYMBOL(sport_config_rx); - -int sport_config_tx(struct sport_device *sport, unsigned int tcr1, - unsigned int tcr2, unsigned int clkdiv, unsigned int fsdiv) -{ - if ((sport->regs->tcr1 & TSPEN) || (sport->regs->rcr1 & RSPEN)) - return -EBUSY; - - sport->regs->tcr1 = tcr1; - sport->regs->tcr2 = tcr2; - sport->regs->tclkdiv = clkdiv; - sport->regs->tfsdiv = fsdiv; - - SSYNC(); - - return 0; -} -EXPORT_SYMBOL(sport_config_tx); - -static void setup_desc(struct dmasg *desc, void *buf, int fragcount, - size_t fragsize, unsigned int cfg, - unsigned int x_count, unsigned int ycount, size_t wdsize) -{ - - int i; - - for (i = 0; i < fragcount; ++i) { - desc[i].next_desc_addr = &(desc[i + 1]); - desc[i].start_addr = (unsigned long)buf + i*fragsize; - desc[i].cfg = cfg; - desc[i].x_count = x_count; - desc[i].x_modify = wdsize; - desc[i].y_count = ycount; - desc[i].y_modify = wdsize; - } - - /* make circular */ - desc[fragcount-1].next_desc_addr = desc; - - pr_debug("setup desc: desc0=%p, next0=%p, desc1=%p," - "next1=%p\nx_count=%x,y_count=%x,addr=0x%lx,cfs=0x%x\n", - desc, desc[0].next_desc_addr, - desc+1, desc[1].next_desc_addr, - desc[0].x_count, desc[0].y_count, - desc[0].start_addr, desc[0].cfg); -} - -static int sport_start(struct sport_device *sport) -{ - enable_dma(sport->dma_rx_chan); - enable_dma(sport->dma_tx_chan); - sport->regs->rcr1 |= RSPEN; - sport->regs->tcr1 |= TSPEN; - SSYNC(); - - return 0; -} - -static int sport_stop(struct sport_device *sport) -{ - sport->regs->tcr1 &= ~TSPEN; - sport->regs->rcr1 &= ~RSPEN; - SSYNC(); - - disable_dma(sport->dma_rx_chan); - disable_dma(sport->dma_tx_chan); - return 0; -} - -static inline int sport_hook_rx_dummy(struct sport_device *sport) -{ - struct dmasg *desc, temp_desc; - unsigned long flags; - - if (WARN_ON(!sport->dummy_rx_desc) || - WARN_ON(sport->curr_rx_desc == sport->dummy_rx_desc)) - return -EINVAL; - - /* Maybe the dummy buffer descriptor ring is damaged */ - sport->dummy_rx_desc->next_desc_addr = sport->dummy_rx_desc + 1; - - local_irq_save(flags); - desc = get_dma_next_desc_ptr(sport->dma_rx_chan); - /* Copy the descriptor which will be damaged to backup */ - temp_desc = *desc; - desc->x_count = sport->dummy_count / 2; - desc->y_count = 0; - desc->next_desc_addr = sport->dummy_rx_desc; - local_irq_restore(flags); - /* Waiting for dummy buffer descriptor is already hooked*/ - while ((get_dma_curr_desc_ptr(sport->dma_rx_chan) - - sizeof(struct dmasg)) != sport->dummy_rx_desc) - continue; - sport->curr_rx_desc = sport->dummy_rx_desc; - /* Restore the damaged descriptor */ - *desc = temp_desc; - - return 0; -} - -static inline int sport_rx_dma_start(struct sport_device *sport, int dummy) -{ - if (dummy) { - sport->dummy_rx_desc->next_desc_addr = sport->dummy_rx_desc; - sport->curr_rx_desc = sport->dummy_rx_desc; - } else - sport->curr_rx_desc = sport->dma_rx_desc; - - set_dma_next_desc_addr(sport->dma_rx_chan, sport->curr_rx_desc); - set_dma_x_count(sport->dma_rx_chan, 0); - set_dma_x_modify(sport->dma_rx_chan, 0); - set_dma_config(sport->dma_rx_chan, (DMAFLOW_LARGE | NDSIZE_9 | \ - WDSIZE_32 | WNR)); - set_dma_curr_addr(sport->dma_rx_chan, sport->curr_rx_desc->start_addr); - SSYNC(); - - return 0; -} - -static inline int sport_tx_dma_start(struct sport_device *sport, int dummy) -{ - if (dummy) { - sport->dummy_tx_desc->next_desc_addr = sport->dummy_tx_desc; - sport->curr_tx_desc = sport->dummy_tx_desc; - } else - sport->curr_tx_desc = sport->dma_tx_desc; - - set_dma_next_desc_addr(sport->dma_tx_chan, sport->curr_tx_desc); - set_dma_x_count(sport->dma_tx_chan, 0); - set_dma_x_modify(sport->dma_tx_chan, 0); - set_dma_config(sport->dma_tx_chan, - (DMAFLOW_LARGE | NDSIZE_9 | WDSIZE_32)); - set_dma_curr_addr(sport->dma_tx_chan, sport->curr_tx_desc->start_addr); - SSYNC(); - - return 0; -} - -int sport_rx_start(struct sport_device *sport) -{ - unsigned long flags; - pr_debug("%s enter\n", __func__); - if (sport->rx_run) - return -EBUSY; - if (sport->tx_run) { - /* tx is running, rx is not running */ - if (WARN_ON(!sport->dma_rx_desc) || - WARN_ON(sport->curr_rx_desc != sport->dummy_rx_desc)) - return -EINVAL; - local_irq_save(flags); - while ((get_dma_curr_desc_ptr(sport->dma_rx_chan) - - sizeof(struct dmasg)) != sport->dummy_rx_desc) - continue; - sport->dummy_rx_desc->next_desc_addr = sport->dma_rx_desc; - local_irq_restore(flags); - sport->curr_rx_desc = sport->dma_rx_desc; - } else { - sport_tx_dma_start(sport, 1); - sport_rx_dma_start(sport, 0); - sport_start(sport); - } - - sport->rx_run = 1; - - return 0; -} -EXPORT_SYMBOL(sport_rx_start); - -int sport_rx_stop(struct sport_device *sport) -{ - pr_debug("%s enter\n", __func__); - - if (!sport->rx_run) - return 0; - if (sport->tx_run) { - /* TX dma is still running, hook the dummy buffer */ - sport_hook_rx_dummy(sport); - } else { - /* Both rx and tx dma will be stopped */ - sport_stop(sport); - sport->curr_rx_desc = NULL; - sport->curr_tx_desc = NULL; - } - - sport->rx_run = 0; - - return 0; -} -EXPORT_SYMBOL(sport_rx_stop); - -static inline int sport_hook_tx_dummy(struct sport_device *sport) -{ - struct dmasg *desc, temp_desc; - unsigned long flags; - - if (WARN_ON(!sport->dummy_tx_desc) || - WARN_ON(sport->curr_tx_desc == sport->dummy_tx_desc)) - return -EINVAL; - - sport->dummy_tx_desc->next_desc_addr = sport->dummy_tx_desc + 1; - - /* Shorten the time on last normal descriptor */ - local_irq_save(flags); - desc = get_dma_next_desc_ptr(sport->dma_tx_chan); - /* Store the descriptor which will be damaged */ - temp_desc = *desc; - desc->x_count = sport->dummy_count / 2; - desc->y_count = 0; - desc->next_desc_addr = sport->dummy_tx_desc; - local_irq_restore(flags); - /* Waiting for dummy buffer descriptor is already hooked*/ - while ((get_dma_curr_desc_ptr(sport->dma_tx_chan) - \ - sizeof(struct dmasg)) != sport->dummy_tx_desc) - continue; - sport->curr_tx_desc = sport->dummy_tx_desc; - /* Restore the damaged descriptor */ - *desc = temp_desc; - - return 0; -} - -int sport_tx_start(struct sport_device *sport) -{ - unsigned long flags; - pr_debug("%s: tx_run:%d, rx_run:%d\n", __func__, - sport->tx_run, sport->rx_run); - if (sport->tx_run) - return -EBUSY; - if (sport->rx_run) { - if (WARN_ON(!sport->dma_tx_desc) || - WARN_ON(sport->curr_tx_desc != sport->dummy_tx_desc)) - return -EINVAL; - /* Hook the normal buffer descriptor */ - local_irq_save(flags); - while ((get_dma_curr_desc_ptr(sport->dma_tx_chan) - - sizeof(struct dmasg)) != sport->dummy_tx_desc) - continue; - sport->dummy_tx_desc->next_desc_addr = sport->dma_tx_desc; - local_irq_restore(flags); - sport->curr_tx_desc = sport->dma_tx_desc; - } else { - - sport_tx_dma_start(sport, 0); - /* Let rx dma run the dummy buffer */ - sport_rx_dma_start(sport, 1); - sport_start(sport); - } - sport->tx_run = 1; - return 0; -} -EXPORT_SYMBOL(sport_tx_start); - -int sport_tx_stop(struct sport_device *sport) -{ - if (!sport->tx_run) - return 0; - if (sport->rx_run) { - /* RX is still running, hook the dummy buffer */ - sport_hook_tx_dummy(sport); - } else { - /* Both rx and tx dma stopped */ - sport_stop(sport); - sport->curr_rx_desc = NULL; - sport->curr_tx_desc = NULL; - } - - sport->tx_run = 0; - - return 0; -} -EXPORT_SYMBOL(sport_tx_stop); - -static inline int compute_wdsize(size_t wdsize) -{ - switch (wdsize) { - case 1: - return WDSIZE_8; - case 2: - return WDSIZE_16; - case 4: - default: - return WDSIZE_32; - } -} - -int sport_config_rx_dma(struct sport_device *sport, void *buf, - int fragcount, size_t fragsize) -{ - unsigned int x_count; - unsigned int y_count; - unsigned int cfg; - dma_addr_t addr; - - pr_debug("%s buf:%p, frag:%d, fragsize:0x%lx\n", __func__, \ - buf, fragcount, fragsize); - - x_count = fragsize / sport->wdsize; - y_count = 0; - - /* for fragments larger than 64k words we use 2d dma, - * denote fragecount as two numbers' mutliply and both of them - * are less than 64k.*/ - if (x_count >= 0x10000) { - int i, count = x_count; - - for (i = 16; i > 0; i--) { - x_count = 1 << i; - if ((count & (x_count - 1)) == 0) { - y_count = count >> i; - if (y_count < 0x10000) - break; - } - } - if (i == 0) - return -EINVAL; - } - pr_debug("%s(x_count:0x%x, y_count:0x%x)\n", __func__, - x_count, y_count); - - if (sport->dma_rx_desc) - dma_free_coherent(NULL, sport->rx_desc_bytes, - sport->dma_rx_desc, 0); - - /* Allocate a new descritor ring as current one. */ - sport->dma_rx_desc = dma_alloc_coherent(NULL, \ - fragcount * sizeof(struct dmasg), &addr, 0); - sport->rx_desc_bytes = fragcount * sizeof(struct dmasg); - - if (!sport->dma_rx_desc) { - pr_err("Failed to allocate memory for rx desc\n"); - return -ENOMEM; - } - - sport->rx_buf = buf; - sport->rx_fragsize = fragsize; - sport->rx_frags = fragcount; - - cfg = 0x7000 | DI_EN | compute_wdsize(sport->wdsize) | WNR | \ - (DESC_ELEMENT_COUNT << 8); /* large descriptor mode */ - - if (y_count != 0) - cfg |= DMA2D; - - setup_desc(sport->dma_rx_desc, buf, fragcount, fragsize, - cfg|DMAEN, x_count, y_count, sport->wdsize); - - return 0; -} -EXPORT_SYMBOL(sport_config_rx_dma); - -int sport_config_tx_dma(struct sport_device *sport, void *buf, \ - int fragcount, size_t fragsize) -{ - unsigned int x_count; - unsigned int y_count; - unsigned int cfg; - dma_addr_t addr; - - pr_debug("%s buf:%p, fragcount:%d, fragsize:0x%lx\n", - __func__, buf, fragcount, fragsize); - - x_count = fragsize/sport->wdsize; - y_count = 0; - - /* for fragments larger than 64k words we use 2d dma, - * denote fragecount as two numbers' mutliply and both of them - * are less than 64k.*/ - if (x_count >= 0x10000) { - int i, count = x_count; - - for (i = 16; i > 0; i--) { - x_count = 1 << i; - if ((count & (x_count - 1)) == 0) { - y_count = count >> i; - if (y_count < 0x10000) - break; - } - } - if (i == 0) - return -EINVAL; - } - pr_debug("%s x_count:0x%x, y_count:0x%x\n", __func__, - x_count, y_count); - - - if (sport->dma_tx_desc) { - dma_free_coherent(NULL, sport->tx_desc_bytes, \ - sport->dma_tx_desc, 0); - } - - sport->dma_tx_desc = dma_alloc_coherent(NULL, \ - fragcount * sizeof(struct dmasg), &addr, 0); - sport->tx_desc_bytes = fragcount * sizeof(struct dmasg); - if (!sport->dma_tx_desc) { - pr_err("Failed to allocate memory for tx desc\n"); - return -ENOMEM; - } - - sport->tx_buf = buf; - sport->tx_fragsize = fragsize; - sport->tx_frags = fragcount; - cfg = 0x7000 | DI_EN | compute_wdsize(sport->wdsize) | \ - (DESC_ELEMENT_COUNT << 8); /* large descriptor mode */ - - if (y_count != 0) - cfg |= DMA2D; - - setup_desc(sport->dma_tx_desc, buf, fragcount, fragsize, - cfg|DMAEN, x_count, y_count, sport->wdsize); - - return 0; -} -EXPORT_SYMBOL(sport_config_tx_dma); - -/* setup dummy dma descriptor ring, which don't generate interrupts, - * the x_modify is set to 0 */ -static int sport_config_rx_dummy(struct sport_device *sport) -{ - struct dmasg *desc; - unsigned config; - - pr_debug("%s entered\n", __func__); - if (L1_DATA_A_LENGTH) - desc = l1_data_sram_zalloc(2 * sizeof(*desc)); - else { - dma_addr_t addr; - desc = dma_alloc_coherent(NULL, 2 * sizeof(*desc), &addr, 0); - memset(desc, 0, 2 * sizeof(*desc)); - } - if (desc == NULL) { - pr_err("Failed to allocate memory for dummy rx desc\n"); - return -ENOMEM; - } - sport->dummy_rx_desc = desc; - desc->start_addr = (unsigned long)sport->dummy_buf; - config = DMAFLOW_LARGE | NDSIZE_9 | compute_wdsize(sport->wdsize) - | WNR | DMAEN; - desc->cfg = config; - desc->x_count = sport->dummy_count/sport->wdsize; - desc->x_modify = sport->wdsize; - desc->y_count = 0; - desc->y_modify = 0; - memcpy(desc+1, desc, sizeof(*desc)); - desc->next_desc_addr = desc + 1; - desc[1].next_desc_addr = desc; - return 0; -} - -static int sport_config_tx_dummy(struct sport_device *sport) -{ - struct dmasg *desc; - unsigned int config; - - pr_debug("%s entered\n", __func__); - - if (L1_DATA_A_LENGTH) - desc = l1_data_sram_zalloc(2 * sizeof(*desc)); - else { - dma_addr_t addr; - desc = dma_alloc_coherent(NULL, 2 * sizeof(*desc), &addr, 0); - memset(desc, 0, 2 * sizeof(*desc)); - } - if (!desc) { - pr_err("Failed to allocate memory for dummy tx desc\n"); - return -ENOMEM; - } - sport->dummy_tx_desc = desc; - desc->start_addr = (unsigned long)sport->dummy_buf + \ - sport->dummy_count; - config = DMAFLOW_LARGE | NDSIZE_9 | - compute_wdsize(sport->wdsize) | DMAEN; - desc->cfg = config; - desc->x_count = sport->dummy_count/sport->wdsize; - desc->x_modify = sport->wdsize; - desc->y_count = 0; - desc->y_modify = 0; - memcpy(desc+1, desc, sizeof(*desc)); - desc->next_desc_addr = desc + 1; - desc[1].next_desc_addr = desc; - return 0; -} - -unsigned long sport_curr_offset_rx(struct sport_device *sport) -{ - unsigned long curr = get_dma_curr_addr(sport->dma_rx_chan); - - return (unsigned char *)curr - sport->rx_buf; -} -EXPORT_SYMBOL(sport_curr_offset_rx); - -unsigned long sport_curr_offset_tx(struct sport_device *sport) -{ - unsigned long curr = get_dma_curr_addr(sport->dma_tx_chan); - - return (unsigned char *)curr - sport->tx_buf; -} -EXPORT_SYMBOL(sport_curr_offset_tx); - -void sport_incfrag(struct sport_device *sport, int *frag, int tx) -{ - ++(*frag); - if (tx == 1 && *frag == sport->tx_frags) - *frag = 0; - - if (tx == 0 && *frag == sport->rx_frags) - *frag = 0; -} -EXPORT_SYMBOL(sport_incfrag); - -void sport_decfrag(struct sport_device *sport, int *frag, int tx) -{ - --(*frag); - if (tx == 1 && *frag == 0) - *frag = sport->tx_frags; - - if (tx == 0 && *frag == 0) - *frag = sport->rx_frags; -} -EXPORT_SYMBOL(sport_decfrag); - -static int sport_check_status(struct sport_device *sport, - unsigned int *sport_stat, - unsigned int *rx_stat, - unsigned int *tx_stat) -{ - int status = 0; - - if (sport_stat) { - SSYNC(); - status = sport->regs->stat; - if (status & (TOVF|TUVF|ROVF|RUVF)) - sport->regs->stat = (status & (TOVF|TUVF|ROVF|RUVF)); - SSYNC(); - *sport_stat = status; - } - - if (rx_stat) { - SSYNC(); - status = get_dma_curr_irqstat(sport->dma_rx_chan); - if (status & (DMA_DONE|DMA_ERR)) - clear_dma_irqstat(sport->dma_rx_chan); - SSYNC(); - *rx_stat = status; - } - - if (tx_stat) { - SSYNC(); - status = get_dma_curr_irqstat(sport->dma_tx_chan); - if (status & (DMA_DONE|DMA_ERR)) - clear_dma_irqstat(sport->dma_tx_chan); - SSYNC(); - *tx_stat = status; - } - - return 0; -} - -int sport_dump_stat(struct sport_device *sport, char *buf, size_t len) -{ - int ret; - - ret = snprintf(buf, len, - "sts: 0x%04x\n" - "rx dma %d sts: 0x%04x tx dma %d sts: 0x%04x\n", - sport->regs->stat, - sport->dma_rx_chan, - get_dma_curr_irqstat(sport->dma_rx_chan), - sport->dma_tx_chan, - get_dma_curr_irqstat(sport->dma_tx_chan)); - buf += ret; - len -= ret; - - ret += snprintf(buf, len, - "curr_rx_desc:0x%p, curr_tx_desc:0x%p\n" - "dma_rx_desc:0x%p, dma_tx_desc:0x%p\n" - "dummy_rx_desc:0x%p, dummy_tx_desc:0x%p\n", - sport->curr_rx_desc, sport->curr_tx_desc, - sport->dma_rx_desc, sport->dma_tx_desc, - sport->dummy_rx_desc, sport->dummy_tx_desc); - - return ret; -} - -static irqreturn_t rx_handler(int irq, void *dev_id) -{ - unsigned int rx_stat; - struct sport_device *sport = dev_id; - - pr_debug("%s enter\n", __func__); - sport_check_status(sport, NULL, &rx_stat, NULL); - if (!(rx_stat & DMA_DONE)) - pr_err("rx dma is already stopped\n"); - - if (sport->rx_callback) { - sport->rx_callback(sport->rx_data); - return IRQ_HANDLED; - } - - return IRQ_NONE; -} - -static irqreturn_t tx_handler(int irq, void *dev_id) -{ - unsigned int tx_stat; - struct sport_device *sport = dev_id; - pr_debug("%s enter\n", __func__); - sport_check_status(sport, NULL, NULL, &tx_stat); - if (!(tx_stat & DMA_DONE)) { - pr_err("tx dma is already stopped\n"); - return IRQ_HANDLED; - } - if (sport->tx_callback) { - sport->tx_callback(sport->tx_data); - return IRQ_HANDLED; - } - - return IRQ_NONE; -} - -static irqreturn_t err_handler(int irq, void *dev_id) -{ - unsigned int status = 0; - struct sport_device *sport = dev_id; - - pr_debug("%s\n", __func__); - if (sport_check_status(sport, &status, NULL, NULL)) { - pr_err("error checking status ??"); - return IRQ_NONE; - } - - if (status & (TOVF|TUVF|ROVF|RUVF)) { - pr_info("sport status error:%s%s%s%s\n", - status & TOVF ? " TOVF" : "", - status & TUVF ? " TUVF" : "", - status & ROVF ? " ROVF" : "", - status & RUVF ? " RUVF" : ""); - if (status & TOVF || status & TUVF) { - disable_dma(sport->dma_tx_chan); - if (sport->tx_run) - sport_tx_dma_start(sport, 0); - else - sport_tx_dma_start(sport, 1); - enable_dma(sport->dma_tx_chan); - } else { - disable_dma(sport->dma_rx_chan); - if (sport->rx_run) - sport_rx_dma_start(sport, 0); - else - sport_rx_dma_start(sport, 1); - enable_dma(sport->dma_rx_chan); - } - } - status = sport->regs->stat; - if (status & (TOVF|TUVF|ROVF|RUVF)) - sport->regs->stat = (status & (TOVF|TUVF|ROVF|RUVF)); - SSYNC(); - - if (sport->err_callback) - sport->err_callback(sport->err_data); - - return IRQ_HANDLED; -} - -int sport_set_rx_callback(struct sport_device *sport, - void (*rx_callback)(void *), void *rx_data) -{ - if (WARN_ON(!rx_callback)) - return -EINVAL; - sport->rx_callback = rx_callback; - sport->rx_data = rx_data; - - return 0; -} -EXPORT_SYMBOL(sport_set_rx_callback); - -int sport_set_tx_callback(struct sport_device *sport, - void (*tx_callback)(void *), void *tx_data) -{ - if (WARN_ON(!tx_callback)) - return -EINVAL; - sport->tx_callback = tx_callback; - sport->tx_data = tx_data; - - return 0; -} -EXPORT_SYMBOL(sport_set_tx_callback); - -int sport_set_err_callback(struct sport_device *sport, - void (*err_callback)(void *), void *err_data) -{ - if (WARN_ON(!err_callback)) - return -EINVAL; - sport->err_callback = err_callback; - sport->err_data = err_data; - - return 0; -} -EXPORT_SYMBOL(sport_set_err_callback); - -static int sport_config_pdev(struct platform_device *pdev, struct sport_param *param) -{ - /* Extract settings from platform data */ - struct device *dev = &pdev->dev; - struct bfin_snd_platform_data *pdata = dev->platform_data; - struct resource *res; - - param->num = pdev->id; - - if (!pdata) { - dev_err(dev, "no platform_data\n"); - return -ENODEV; - } - param->pin_req = pdata->pin_req; - - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - dev_err(dev, "no MEM resource\n"); - return -ENODEV; - } - param->regs = (struct sport_register *)res->start; - - /* first RX, then TX */ - res = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!res) { - dev_err(dev, "no rx DMA resource\n"); - return -ENODEV; - } - param->dma_rx_chan = res->start; - - res = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!res) { - dev_err(dev, "no tx DMA resource\n"); - return -ENODEV; - } - param->dma_tx_chan = res->start; - - res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); - if (!res) { - dev_err(dev, "no irq resource\n"); - return -ENODEV; - } - param->err_irq = res->start; - - return 0; -} - -struct sport_device *sport_init(struct platform_device *pdev, - unsigned int wdsize, unsigned int dummy_count, size_t priv_size) -{ - struct device *dev = &pdev->dev; - struct sport_param param; - struct sport_device *sport; - int ret; - - dev_dbg(dev, "%s enter\n", __func__); - - param.wdsize = wdsize; - param.dummy_count = dummy_count; - if (WARN_ON(param.wdsize == 0 || param.dummy_count == 0)) - return NULL; - - ret = sport_config_pdev(pdev, &param); - if (ret) - return NULL; - - if (peripheral_request_list(param.pin_req, "soc-audio")) { - dev_err(dev, "requesting Peripherals failed\n"); - return NULL; - } - - sport = kzalloc(sizeof(*sport), GFP_KERNEL); - if (!sport) { - dev_err(dev, "failed to allocate for sport device\n"); - goto __init_err0; - } - - sport->num = param.num; - sport->dma_rx_chan = param.dma_rx_chan; - sport->dma_tx_chan = param.dma_tx_chan; - sport->err_irq = param.err_irq; - sport->regs = param.regs; - sport->pin_req = param.pin_req; - - if (request_dma(sport->dma_rx_chan, "SPORT RX Data") == -EBUSY) { - dev_err(dev, "failed to request RX dma %d\n", sport->dma_rx_chan); - goto __init_err1; - } - if (set_dma_callback(sport->dma_rx_chan, rx_handler, sport) != 0) { - dev_err(dev, "failed to request RX irq %d\n", sport->dma_rx_chan); - goto __init_err2; - } - - if (request_dma(sport->dma_tx_chan, "SPORT TX Data") == -EBUSY) { - dev_err(dev, "failed to request TX dma %d\n", sport->dma_tx_chan); - goto __init_err2; - } - - if (set_dma_callback(sport->dma_tx_chan, tx_handler, sport) != 0) { - dev_err(dev, "failed to request TX irq %d\n", sport->dma_tx_chan); - goto __init_err3; - } - - if (request_irq(sport->err_irq, err_handler, IRQF_SHARED, "SPORT err", - sport) < 0) { - dev_err(dev, "failed to request err irq %d\n", sport->err_irq); - goto __init_err3; - } - - dev_info(dev, "dma rx:%d tx:%d, err irq:%d, regs:%p\n", - sport->dma_rx_chan, sport->dma_tx_chan, - sport->err_irq, sport->regs); - - sport->wdsize = param.wdsize; - sport->dummy_count = param.dummy_count; - - sport->private_data = kzalloc(priv_size, GFP_KERNEL); - if (!sport->private_data) { - dev_err(dev, "could not alloc priv data %zu bytes\n", priv_size); - goto __init_err4; - } - - if (L1_DATA_A_LENGTH) - sport->dummy_buf = l1_data_sram_zalloc(param.dummy_count * 2); - else - sport->dummy_buf = kzalloc(param.dummy_count * 2, GFP_KERNEL); - if (sport->dummy_buf == NULL) { - dev_err(dev, "failed to allocate dummy buffer\n"); - goto __error1; - } - - ret = sport_config_rx_dummy(sport); - if (ret) { - dev_err(dev, "failed to config rx dummy ring\n"); - goto __error2; - } - ret = sport_config_tx_dummy(sport); - if (ret) { - dev_err(dev, "failed to config tx dummy ring\n"); - goto __error3; - } - - platform_set_drvdata(pdev, sport); - - return sport; -__error3: - if (L1_DATA_A_LENGTH) - l1_data_sram_free(sport->dummy_rx_desc); - else - dma_free_coherent(NULL, 2*sizeof(struct dmasg), - sport->dummy_rx_desc, 0); -__error2: - if (L1_DATA_A_LENGTH) - l1_data_sram_free(sport->dummy_buf); - else - kfree(sport->dummy_buf); -__error1: - kfree(sport->private_data); -__init_err4: - free_irq(sport->err_irq, sport); -__init_err3: - free_dma(sport->dma_tx_chan); -__init_err2: - free_dma(sport->dma_rx_chan); -__init_err1: - kfree(sport); -__init_err0: - peripheral_free_list(param.pin_req); - return NULL; -} -EXPORT_SYMBOL(sport_init); - -void sport_done(struct sport_device *sport) -{ - if (sport == NULL) - return; - - sport_stop(sport); - if (sport->dma_rx_desc) - dma_free_coherent(NULL, sport->rx_desc_bytes, - sport->dma_rx_desc, 0); - if (sport->dma_tx_desc) - dma_free_coherent(NULL, sport->tx_desc_bytes, - sport->dma_tx_desc, 0); - -#if L1_DATA_A_LENGTH != 0 - l1_data_sram_free(sport->dummy_rx_desc); - l1_data_sram_free(sport->dummy_tx_desc); - l1_data_sram_free(sport->dummy_buf); -#else - dma_free_coherent(NULL, 2*sizeof(struct dmasg), - sport->dummy_rx_desc, 0); - dma_free_coherent(NULL, 2*sizeof(struct dmasg), - sport->dummy_tx_desc, 0); - kfree(sport->dummy_buf); -#endif - free_dma(sport->dma_rx_chan); - free_dma(sport->dma_tx_chan); - free_irq(sport->err_irq, sport); - - kfree(sport->private_data); - peripheral_free_list(sport->pin_req); - kfree(sport); -} -EXPORT_SYMBOL(sport_done); - -/* -* It is only used to send several bytes when dma is not enabled - * sport controller is configured but not enabled. - * Multichannel cannot works with pio mode */ -/* Used by ac97 to write and read codec register */ -int sport_send_and_recv(struct sport_device *sport, u8 *out_data, \ - u8 *in_data, int len) -{ - unsigned short dma_config; - unsigned short status; - unsigned long flags; - unsigned long wait = 0; - - pr_debug("%s enter, out_data:%p, in_data:%p len:%d\n", \ - __func__, out_data, in_data, len); - pr_debug("tcr1:0x%04x, tcr2:0x%04x, tclkdiv:0x%04x, tfsdiv:0x%04x\n" - "mcmc1:0x%04x, mcmc2:0x%04x\n", - sport->regs->tcr1, sport->regs->tcr2, - sport->regs->tclkdiv, sport->regs->tfsdiv, - sport->regs->mcmc1, sport->regs->mcmc2); - flush_dcache_range((unsigned)out_data, (unsigned)(out_data + len)); - - /* Enable tx dma */ - dma_config = (RESTART | WDSIZE_16 | DI_EN); - set_dma_start_addr(sport->dma_tx_chan, (unsigned long)out_data); - set_dma_x_count(sport->dma_tx_chan, len/2); - set_dma_x_modify(sport->dma_tx_chan, 2); - set_dma_config(sport->dma_tx_chan, dma_config); - enable_dma(sport->dma_tx_chan); - - if (in_data != NULL) { - invalidate_dcache_range((unsigned)in_data, \ - (unsigned)(in_data + len)); - /* Enable rx dma */ - dma_config = (RESTART | WDSIZE_16 | WNR | DI_EN); - set_dma_start_addr(sport->dma_rx_chan, (unsigned long)in_data); - set_dma_x_count(sport->dma_rx_chan, len/2); - set_dma_x_modify(sport->dma_rx_chan, 2); - set_dma_config(sport->dma_rx_chan, dma_config); - enable_dma(sport->dma_rx_chan); - } - - local_irq_save(flags); - sport->regs->tcr1 |= TSPEN; - sport->regs->rcr1 |= RSPEN; - SSYNC(); - - status = get_dma_curr_irqstat(sport->dma_tx_chan); - while (status & DMA_RUN) { - udelay(1); - status = get_dma_curr_irqstat(sport->dma_tx_chan); - pr_debug("DMA status:0x%04x\n", status); - if (wait++ > 100) - goto __over; - } - status = sport->regs->stat; - wait = 0; - - while (!(status & TXHRE)) { - pr_debug("sport status:0x%04x\n", status); - udelay(1); - status = *(unsigned short *)&sport->regs->stat; - if (wait++ > 1000) - goto __over; - } - /* Wait for the last byte sent out */ - udelay(20); - pr_debug("sport status:0x%04x\n", status); - -__over: - sport->regs->tcr1 &= ~TSPEN; - sport->regs->rcr1 &= ~RSPEN; - SSYNC(); - disable_dma(sport->dma_tx_chan); - /* Clear the status */ - clear_dma_irqstat(sport->dma_tx_chan); - if (in_data != NULL) { - disable_dma(sport->dma_rx_chan); - clear_dma_irqstat(sport->dma_rx_chan); - } - SSYNC(); - local_irq_restore(flags); - - return 0; -} -EXPORT_SYMBOL(sport_send_and_recv); - -MODULE_AUTHOR("Roy Huang"); -MODULE_DESCRIPTION("SPORT driver for ADI Blackfin"); -MODULE_LICENSE("GPL"); diff --git a/sound/soc/blackfin/bf5xx-sport.h b/sound/soc/blackfin/bf5xx-sport.h deleted file mode 100644 index 9fc2192feb3b..000000000000 --- a/sound/soc/blackfin/bf5xx-sport.h +++ /dev/null @@ -1,174 +0,0 @@ -/* - * File: bf5xx_sport.h - * Based on: - * Author: Roy Huang <roy.huang(a)analog.com> - * - * Created: - * Description: - * - * Copyright 2004-2007 Analog Devices Inc. - * - * Bugs: Enter bugs at http://blackfin.uclinux.org/ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, see the file COPYING, or write - * to the Free Software Foundation, Inc., - * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA - */ - - -#ifndef __BF5XX_SPORT_H__ -#define __BF5XX_SPORT_H__ - -#include <linux/types.h> -#include <linux/wait.h> -#include <linux/workqueue.h> -#include <linux/platform_device.h> -#include <asm/dma.h> -#include <asm/bfin_sport.h> - -#define DESC_ELEMENT_COUNT 9 - -struct sport_device { - int num; - int dma_rx_chan; - int dma_tx_chan; - int err_irq; - const unsigned short *pin_req; - struct sport_register *regs; - - unsigned char *rx_buf; - unsigned char *tx_buf; - unsigned int rx_fragsize; - unsigned int tx_fragsize; - unsigned int rx_frags; - unsigned int tx_frags; - unsigned int wdsize; - - /* for dummy dma transfer */ - void *dummy_buf; - unsigned int dummy_count; - - /* DMA descriptor ring head of current audio stream*/ - struct dmasg *dma_rx_desc; - struct dmasg *dma_tx_desc; - unsigned int rx_desc_bytes; - unsigned int tx_desc_bytes; - - unsigned int rx_run:1; /* rx is running */ - unsigned int tx_run:1; /* tx is running */ - - struct dmasg *dummy_rx_desc; - struct dmasg *dummy_tx_desc; - - struct dmasg *curr_rx_desc; - struct dmasg *curr_tx_desc; - - int rx_curr_frag; - int tx_curr_frag; - - unsigned int rcr1; - unsigned int rcr2; - int rx_tdm_count; - - unsigned int tcr1; - unsigned int tcr2; - int tx_tdm_count; - - void (*rx_callback)(void *data); - void *rx_data; - void (*tx_callback)(void *data); - void *tx_data; - void (*err_callback)(void *data); - void *err_data; - unsigned char *tx_dma_buf; - unsigned char *rx_dma_buf; -#ifdef CONFIG_SND_BF5XX_MMAP_SUPPORT - dma_addr_t tx_dma_phy; - dma_addr_t rx_dma_phy; - int tx_pos;/*pcm sample count*/ - int rx_pos; - unsigned int tx_buffer_size; - unsigned int rx_buffer_size; - int tx_delay_pos; - int once; -#endif - void *private_data; -}; - -struct sport_param { - int num; - int dma_rx_chan; - int dma_tx_chan; - int err_irq; - const unsigned short *pin_req; - struct sport_register *regs; - unsigned int wdsize; - unsigned int dummy_count; - void *private_data; -}; - -struct sport_device *sport_init(struct platform_device *pdev, - unsigned int wdsize, unsigned int dummy_count, size_t priv_size); - -void sport_done(struct sport_device *sport); - -/* first use these ...*/ - -/* note: multichannel is in units of 8 channels, tdm_count is number of channels - * NOT / 8 ! all channels are enabled by default */ -int sport_set_multichannel(struct sport_device *sport, int tdm_count, - u32 tx_mask, u32 rx_mask, int packed); - -int sport_config_rx(struct sport_device *sport, - unsigned int rcr1, unsigned int rcr2, - unsigned int clkdiv, unsigned int fsdiv); - -int sport_config_tx(struct sport_device *sport, - unsigned int tcr1, unsigned int tcr2, - unsigned int clkdiv, unsigned int fsdiv); - -/* ... then these: */ - -/* buffer size (in bytes) == fragcount * fragsize_bytes */ - -/* this is not a very general api, it sets the dma to 2d autobuffer mode */ - -int sport_config_rx_dma(struct sport_device *sport, void *buf, - int fragcount, size_t fragsize_bytes); - -int sport_config_tx_dma(struct sport_device *sport, void *buf, - int fragcount, size_t fragsize_bytes); - -int sport_tx_start(struct sport_device *sport); -int sport_tx_stop(struct sport_device *sport); -int sport_rx_start(struct sport_device *sport); -int sport_rx_stop(struct sport_device *sport); - -/* for use in interrupt handler */ -unsigned long sport_curr_offset_rx(struct sport_device *sport); -unsigned long sport_curr_offset_tx(struct sport_device *sport); - -void sport_incfrag(struct sport_device *sport, int *frag, int tx); -void sport_decfrag(struct sport_device *sport, int *frag, int tx); - -int sport_set_rx_callback(struct sport_device *sport, - void (*rx_callback)(void *), void *rx_data); -int sport_set_tx_callback(struct sport_device *sport, - void (*tx_callback)(void *), void *tx_data); -int sport_set_err_callback(struct sport_device *sport, - void (*err_callback)(void *), void *err_data); - -int sport_send_and_recv(struct sport_device *sport, u8 *out_data, \ - u8 *in_data, int len); -#endif /* BF53X_SPORT_H */ diff --git a/sound/soc/blackfin/bf5xx-ssm2602.c b/sound/soc/blackfin/bf5xx-ssm2602.c deleted file mode 100644 index 9c19ccc936e2..000000000000 --- a/sound/soc/blackfin/bf5xx-ssm2602.c +++ /dev/null @@ -1,126 +0,0 @@ -/* - * File: sound/soc/blackfin/bf5xx-ssm2602.c - * Author: Cliff Cai <Cliff.Cai(a)analog.com> - * - * Created: Tue June 06 2008 - * Description: board driver for SSM2602 sound chip - * - * Modified: - * Copyright 2008 Analog Devices Inc. - * - * Bugs: Enter bugs at http://blackfin.uclinux.org/ - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License as published by - * the Free Software Foundation; either version 2 of the License, or - * (at your option) any later version. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, see the file COPYING, or write - * to the Free Software Foundation, Inc., - * 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA - */ - -#include <linux/module.h> -#include <linux/moduleparam.h> -#include <linux/device.h> - -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/pcm_params.h> - -#include <asm/dma.h> -#include <asm/portmux.h> -#include <linux/gpio.h> -#include "../codecs/ssm2602.h" -#include "bf5xx-sport.h" - -static struct snd_soc_card bf5xx_ssm2602; - -static int bf5xx_ssm2602_dai_init(struct snd_soc_pcm_runtime *rtd) -{ - /* - * If you are using a crystal source which frequency is not 12MHz - * then modify the below case statement with frequency of the crystal. - * - * If you are using the SPORT to generate clocking then this is - * where to do it. - */ - return snd_soc_dai_set_sysclk(rtd->codec_dai, SSM2602_SYSCLK, 12000000, - SND_SOC_CLOCK_IN); -} - -/* CODEC is master for BCLK and LRC in this configuration. */ -#define BF5XX_SSM2602_DAIFMT (SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | \ - SND_SOC_DAIFMT_CBM_CFM) - -static struct snd_soc_dai_link bf5xx_ssm2602_dai[] = { - { - .name = "ssm2602", - .stream_name = "SSM2602", - .cpu_dai_name = "bfin-i2s.0", - .codec_dai_name = "ssm2602-hifi", - .platform_name = "bfin-i2s-pcm-audio", - .codec_name = "ssm2602.0-001b", - .init = bf5xx_ssm2602_dai_init, - .dai_fmt = BF5XX_SSM2602_DAIFMT, - }, - { - .name = "ssm2602", - .stream_name = "SSM2602", - .cpu_dai_name = "bfin-i2s.1", - .codec_dai_name = "ssm2602-hifi", - .platform_name = "bfin-i2s-pcm-audio", - .codec_name = "ssm2602.0-001b", - .init = bf5xx_ssm2602_dai_init, - .dai_fmt = BF5XX_SSM2602_DAIFMT, - }, -}; - -static struct snd_soc_card bf5xx_ssm2602 = { - .name = "bfin-ssm2602", - .owner = THIS_MODULE, - .dai_link = &bf5xx_ssm2602_dai[CONFIG_SND_BF5XX_SPORT_NUM], - .num_links = 1, -}; - -static struct platform_device *bf5xx_ssm2602_snd_device; - -static int __init bf5xx_ssm2602_init(void) -{ - int ret; - - pr_debug("%s enter\n", __func__); - bf5xx_ssm2602_snd_device = platform_device_alloc("soc-audio", -1); - if (!bf5xx_ssm2602_snd_device) - return -ENOMEM; - - platform_set_drvdata(bf5xx_ssm2602_snd_device, &bf5xx_ssm2602); - ret = platform_device_add(bf5xx_ssm2602_snd_device); - - if (ret) - platform_device_put(bf5xx_ssm2602_snd_device); - - return ret; -} - -static void __exit bf5xx_ssm2602_exit(void) -{ - pr_debug("%s enter\n", __func__); - platform_device_unregister(bf5xx_ssm2602_snd_device); -} - -module_init(bf5xx_ssm2602_init); -module_exit(bf5xx_ssm2602_exit); - -/* Module information */ -MODULE_AUTHOR("Cliff Cai"); -MODULE_DESCRIPTION("ALSA SoC SSM2602 BF527-EZKIT"); -MODULE_LICENSE("GPL"); - diff --git a/sound/soc/blackfin/bf6xx-i2s.c b/sound/soc/blackfin/bf6xx-i2s.c deleted file mode 100644 index 819cff149a25..000000000000 --- a/sound/soc/blackfin/bf6xx-i2s.c +++ /dev/null @@ -1,239 +0,0 @@ -/* - * bf6xx-i2s.c - Analog Devices BF6XX i2s interface driver - * - * Copyright (c) 2012 Analog Devices Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. - */ - -#include <linux/device.h> -#include <linux/init.h> -#include <linux/module.h> -#include <linux/platform_device.h> -#include <sound/pcm.h> -#include <sound/pcm_params.h> -#include <sound/soc.h> -#include <sound/soc-dai.h> - -#include "bf6xx-sport.h" - -struct sport_params param; - -static int bfin_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, - unsigned int fmt) -{ - struct sport_device *sport = snd_soc_dai_get_drvdata(cpu_dai); - struct device *dev = &sport->pdev->dev; - int ret = 0; - - param.spctl &= ~(SPORT_CTL_OPMODE | SPORT_CTL_CKRE | SPORT_CTL_FSR - | SPORT_CTL_LFS | SPORT_CTL_LAFS); - switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: - param.spctl |= SPORT_CTL_OPMODE | SPORT_CTL_CKRE - | SPORT_CTL_LFS; - break; - case SND_SOC_DAIFMT_DSP_A: - param.spctl |= SPORT_CTL_FSR; - break; - case SND_SOC_DAIFMT_LEFT_J: - param.spctl |= SPORT_CTL_OPMODE | SPORT_CTL_LFS - | SPORT_CTL_LAFS; - break; - default: - dev_err(dev, "%s: Unknown DAI format type\n", __func__); - ret = -EINVAL; - break; - } - - param.spctl &= ~(SPORT_CTL_ICLK | SPORT_CTL_IFS); - switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { - case SND_SOC_DAIFMT_CBM_CFM: - break; - case SND_SOC_DAIFMT_CBS_CFS: - case SND_SOC_DAIFMT_CBM_CFS: - case SND_SOC_DAIFMT_CBS_CFM: - ret = -EINVAL; - break; - default: - dev_err(dev, "%s: Unknown DAI master type\n", __func__); - ret = -EINVAL; - break; - } - - return ret; -} - -static int bfin_i2s_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params, - struct snd_soc_dai *dai) -{ - struct sport_device *sport = snd_soc_dai_get_drvdata(dai); - struct device *dev = &sport->pdev->dev; - int ret = 0; - - param.spctl &= ~SPORT_CTL_SLEN; - switch (params_format(params)) { - case SNDRV_PCM_FORMAT_S8: - param.spctl |= 0x70; - sport->wdsize = 1; - break; - case SNDRV_PCM_FORMAT_S16_LE: - param.spctl |= 0xf0; - sport->wdsize = 2; - break; - case SNDRV_PCM_FORMAT_S24_LE: - param.spctl |= 0x170; - sport->wdsize = 3; - break; - case SNDRV_PCM_FORMAT_S32_LE: - param.spctl |= 0x1f0; - sport->wdsize = 4; - break; - } - - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - ret = sport_set_tx_params(sport, &param); - if (ret) { - dev_err(dev, "SPORT tx is busy!\n"); - return ret; - } - } else { - ret = sport_set_rx_params(sport, &param); - if (ret) { - dev_err(dev, "SPORT rx is busy!\n"); - return ret; - } - } - return 0; -} - -#ifdef CONFIG_PM -static int bfin_i2s_suspend(struct snd_soc_dai *dai) -{ - struct sport_device *sport = snd_soc_dai_get_drvdata(dai); - - if (dai->capture_active) - sport_rx_stop(sport); - if (dai->playback_active) - sport_tx_stop(sport); - return 0; -} - -static int bfin_i2s_resume(struct snd_soc_dai *dai) -{ - struct sport_device *sport = snd_soc_dai_get_drvdata(dai); - struct device *dev = &sport->pdev->dev; - int ret; - - ret = sport_set_tx_params(sport, &param); - if (ret) { - dev_err(dev, "SPORT tx is busy!\n"); - return ret; - } - ret = sport_set_rx_params(sport, &param); - if (ret) { - dev_err(dev, "SPORT rx is busy!\n"); - return ret; - } - - return 0; -} - -#else -#define bfin_i2s_suspend NULL -#define bfin_i2s_resume NULL -#endif - -#define BFIN_I2S_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ - SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 | \ - SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \ - SNDRV_PCM_RATE_96000) - -#define BFIN_I2S_FORMATS (SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_LE | \ - SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE) - -static const struct snd_soc_dai_ops bfin_i2s_dai_ops = { - .hw_params = bfin_i2s_hw_params, - .set_fmt = bfin_i2s_set_dai_fmt, -}; - -static struct snd_soc_dai_driver bfin_i2s_dai = { - .suspend = bfin_i2s_suspend, - .resume = bfin_i2s_resume, - .playback = { - .channels_min = 1, - .channels_max = 2, - .rates = BFIN_I2S_RATES, - .formats = BFIN_I2S_FORMATS, - }, - .capture = { - .channels_min = 1, - .channels_max = 2, - .rates = BFIN_I2S_RATES, - .formats = BFIN_I2S_FORMATS, - }, - .ops = &bfin_i2s_dai_ops, -}; - -static const struct snd_soc_component_driver bfin_i2s_component = { - .name = "bfin-i2s", -}; - -static int bfin_i2s_probe(struct platform_device *pdev) -{ - struct sport_device *sport; - struct device *dev = &pdev->dev; - int ret; - - sport = sport_create(pdev); - if (!sport) - return -ENODEV; - - /* register with the ASoC layers */ - ret = snd_soc_register_component(dev, &bfin_i2s_component, - &bfin_i2s_dai, 1); - if (ret) { - dev_err(dev, "Failed to register DAI: %d\n", ret); - sport_delete(sport); - return ret; - } - platform_set_drvdata(pdev, sport); - - return 0; -} - -static int bfin_i2s_remove(struct platform_device *pdev) -{ - struct sport_device *sport = platform_get_drvdata(pdev); - - snd_soc_unregister_component(&pdev->dev); - sport_delete(sport); - - return 0; -} - -static struct platform_driver bfin_i2s_driver = { - .probe = bfin_i2s_probe, - .remove = bfin_i2s_remove, - .driver = { - .name = "bfin-i2s", - }, -}; - -module_platform_driver(bfin_i2s_driver); - -MODULE_DESCRIPTION("Analog Devices BF6XX i2s interface driver"); -MODULE_AUTHOR("Scott Jiang <Scott.Jiang.Linux(a)gmail.com>"); -MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/blackfin/bf6xx-sport.c b/sound/soc/blackfin/bf6xx-sport.c deleted file mode 100644 index d2caadfe7b6d..000000000000 --- a/sound/soc/blackfin/bf6xx-sport.c +++ /dev/null @@ -1,425 +0,0 @@ -/* - * bf6xx_sport.c Analog Devices BF6XX SPORT driver - * - * Copyright (c) 2012 Analog Devices Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. - */ - -#include <linux/device.h> -#include <linux/dma-mapping.h> -#include <linux/interrupt.h> -#include <linux/module.h> -#include <linux/platform_device.h> -#include <linux/slab.h> - -#include <asm/blackfin.h> -#include <asm/dma.h> -#include <asm/portmux.h> - -#include "bf6xx-sport.h" - -int sport_set_tx_params(struct sport_device *sport, - struct sport_params *params) -{ - if (sport->tx_regs->spctl & SPORT_CTL_SPENPRI) - return -EBUSY; - sport->tx_regs->spctl = params->spctl | SPORT_CTL_SPTRAN; - sport->tx_regs->div = params->div; - SSYNC(); - return 0; -} -EXPORT_SYMBOL(sport_set_tx_params); - -int sport_set_rx_params(struct sport_device *sport, - struct sport_params *params) -{ - if (sport->rx_regs->spctl & SPORT_CTL_SPENPRI) - return -EBUSY; - sport->rx_regs->spctl = params->spctl & ~SPORT_CTL_SPTRAN; - sport->rx_regs->div = params->div; - SSYNC(); - return 0; -} -EXPORT_SYMBOL(sport_set_rx_params); - -static int compute_wdsize(size_t wdsize) -{ - switch (wdsize) { - case 1: - return WDSIZE_8 | PSIZE_8; - case 2: - return WDSIZE_16 | PSIZE_16; - default: - return WDSIZE_32 | PSIZE_32; - } -} - -void sport_tx_start(struct sport_device *sport) -{ - set_dma_next_desc_addr(sport->tx_dma_chan, sport->tx_desc); - set_dma_config(sport->tx_dma_chan, DMAFLOW_LIST | DI_EN - | compute_wdsize(sport->wdsize) | NDSIZE_6); - enable_dma(sport->tx_dma_chan); - sport->tx_regs->spctl |= SPORT_CTL_SPENPRI; - SSYNC(); -} -EXPORT_SYMBOL(sport_tx_start); - -void sport_rx_start(struct sport_device *sport) -{ - set_dma_next_desc_addr(sport->rx_dma_chan, sport->rx_desc); - set_dma_config(sport->rx_dma_chan, DMAFLOW_LIST | DI_EN | WNR - | compute_wdsize(sport->wdsize) | NDSIZE_6); - enable_dma(sport->rx_dma_chan); - sport->rx_regs->spctl |= SPORT_CTL_SPENPRI; - SSYNC(); -} -EXPORT_SYMBOL(sport_rx_start); - -void sport_tx_stop(struct sport_device *sport) -{ - sport->tx_regs->spctl &= ~SPORT_CTL_SPENPRI; - SSYNC(); - disable_dma(sport->tx_dma_chan); -} -EXPORT_SYMBOL(sport_tx_stop); - -void sport_rx_stop(struct sport_device *sport) -{ - sport->rx_regs->spctl &= ~SPORT_CTL_SPENPRI; - SSYNC(); - disable_dma(sport->rx_dma_chan); -} -EXPORT_SYMBOL(sport_rx_stop); - -void sport_set_tx_callback(struct sport_device *sport, - void (*tx_callback)(void *), void *tx_data) -{ - sport->tx_callback = tx_callback; - sport->tx_data = tx_data; -} -EXPORT_SYMBOL(sport_set_tx_callback); - -void sport_set_rx_callback(struct sport_device *sport, - void (*rx_callback)(void *), void *rx_data) -{ - sport->rx_callback = rx_callback; - sport->rx_data = rx_data; -} -EXPORT_SYMBOL(sport_set_rx_callback); - -static void setup_desc(struct dmasg *desc, void *buf, int fragcount, - size_t fragsize, unsigned int cfg, - unsigned int count, size_t wdsize) -{ - - int i; - - for (i = 0; i < fragcount; ++i) { - desc[i].next_desc_addr = &(desc[i + 1]); - desc[i].start_addr = (unsigned long)buf + i * fragsize; - desc[i].cfg = cfg; - desc[i].x_count = count; - desc[i].x_modify = wdsize; - desc[i].y_count = 0; - desc[i].y_modify = 0; - } - - /* make circular */ - desc[fragcount - 1].next_desc_addr = desc; -} - -int sport_config_tx_dma(struct sport_device *sport, void *buf, - int fragcount, size_t fragsize) -{ - unsigned int count; - unsigned int cfg; - dma_addr_t addr; - - count = fragsize / sport->wdsize; - - if (sport->tx_desc) - dma_free_coherent(NULL, sport->tx_desc_size, - sport->tx_desc, 0); - - sport->tx_desc = dma_alloc_coherent(NULL, - fragcount * sizeof(struct dmasg), &addr, 0); - sport->tx_desc_size = fragcount * sizeof(struct dmasg); - if (!sport->tx_desc) - return -ENOMEM; - - sport->tx_buf = buf; - sport->tx_fragsize = fragsize; - sport->tx_frags = fragcount; - cfg = DMAFLOW_LIST | DI_EN | compute_wdsize(sport->wdsize) | NDSIZE_6; - - setup_desc(sport->tx_desc, buf, fragcount, fragsize, - cfg | DMAEN, count, sport->wdsize); - return 0; -} -EXPORT_SYMBOL(sport_config_tx_dma); - -int sport_config_rx_dma(struct sport_device *sport, void *buf, - int fragcount, size_t fragsize) -{ - unsigned int count; - unsigned int cfg; - dma_addr_t addr; - - count = fragsize / sport->wdsize; - - if (sport->rx_desc) - dma_free_coherent(NULL, sport->rx_desc_size, - sport->rx_desc, 0); - - sport->rx_desc = dma_alloc_coherent(NULL, - fragcount * sizeof(struct dmasg), &addr, 0); - sport->rx_desc_size = fragcount * sizeof(struct dmasg); - if (!sport->rx_desc) - return -ENOMEM; - - sport->rx_buf = buf; - sport->rx_fragsize = fragsize; - sport->rx_frags = fragcount; - cfg = DMAFLOW_LIST | DI_EN | compute_wdsize(sport->wdsize) - | WNR | NDSIZE_6; - - setup_desc(sport->rx_desc, buf, fragcount, fragsize, - cfg | DMAEN, count, sport->wdsize); - return 0; -} -EXPORT_SYMBOL(sport_config_rx_dma); - -unsigned long sport_curr_offset_tx(struct sport_device *sport) -{ - unsigned long curr = get_dma_curr_addr(sport->tx_dma_chan); - - return (unsigned char *)curr - sport->tx_buf; -} -EXPORT_SYMBOL(sport_curr_offset_tx); - -unsigned long sport_curr_offset_rx(struct sport_device *sport) -{ - unsigned long curr = get_dma_curr_addr(sport->rx_dma_chan); - - return (unsigned char *)curr - sport->rx_buf; -} -EXPORT_SYMBOL(sport_curr_offset_rx); - -static irqreturn_t sport_tx_irq(int irq, void *dev_id) -{ - struct sport_device *sport = dev_id; - static unsigned long status; - - status = get_dma_curr_irqstat(sport->tx_dma_chan); - if (status & (DMA_DONE | DMA_ERR)) { - clear_dma_irqstat(sport->tx_dma_chan); - SSYNC(); - } - if (sport->tx_callback) - sport->tx_callback(sport->tx_data); - return IRQ_HANDLED; -} - -static irqreturn_t sport_rx_irq(int irq, void *dev_id) -{ - struct sport_device *sport = dev_id; - unsigned long status; - - status = get_dma_curr_irqstat(sport->rx_dma_chan); - if (status & (DMA_DONE | DMA_ERR)) { - clear_dma_irqstat(sport->rx_dma_chan); - SSYNC(); - } - if (sport->rx_callback) - sport->rx_callback(sport->rx_data); - return IRQ_HANDLED; -} - -static irqreturn_t sport_err_irq(int irq, void *dev_id) -{ - struct sport_device *sport = dev_id; - struct device *dev = &sport->pdev->dev; - - if (sport->tx_regs->spctl & SPORT_CTL_DERRPRI) - dev_err(dev, "sport error: TUVF\n"); - if (sport->rx_regs->spctl & SPORT_CTL_DERRPRI) - dev_err(dev, "sport error: ROVF\n"); - - return IRQ_HANDLED; -} - -static int sport_get_resource(struct sport_device *sport) -{ - struct platform_device *pdev = sport->pdev; - struct device *dev = &pdev->dev; - struct bfin_snd_platform_data *pdata = dev->platform_data; - struct resource *res; - - if (!pdata) { - dev_err(dev, "No platform data\n"); - return -ENODEV; - } - sport->pin_req = pdata->pin_req; - - res = platform_get_resource(pdev, IORESOURCE_MEM, 0); - if (!res) { - dev_err(dev, "No tx MEM resource\n"); - return -ENODEV; - } - sport->tx_regs = (struct sport_register *)res->start; - - res = platform_get_resource(pdev, IORESOURCE_MEM, 1); - if (!res) { - dev_err(dev, "No rx MEM resource\n"); - return -ENODEV; - } - sport->rx_regs = (struct sport_register *)res->start; - - res = platform_get_resource(pdev, IORESOURCE_DMA, 0); - if (!res) { - dev_err(dev, "No tx DMA resource\n"); - return -ENODEV; - } - sport->tx_dma_chan = res->start; - - res = platform_get_resource(pdev, IORESOURCE_DMA, 1); - if (!res) { - dev_err(dev, "No rx DMA resource\n"); - return -ENODEV; - } - sport->rx_dma_chan = res->start; - - res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); - if (!res) { - dev_err(dev, "No tx error irq resource\n"); - return -ENODEV; - } - sport->tx_err_irq = res->start; - - res = platform_get_resource(pdev, IORESOURCE_IRQ, 1); - if (!res) { - dev_err(dev, "No rx error irq resource\n"); - return -ENODEV; - } - sport->rx_err_irq = res->start; - - return 0; -} - -static int sport_request_resource(struct sport_device *sport) -{ - struct device *dev = &sport->pdev->dev; - int ret; - - ret = peripheral_request_list(sport->pin_req, "soc-audio"); - if (ret) { - dev_err(dev, "Unable to request sport pin\n"); - return ret; - } - - ret = request_dma(sport->tx_dma_chan, "SPORT TX Data"); - if (ret) { - dev_err(dev, "Unable to allocate DMA channel for sport tx\n"); - goto err_tx_dma; - } - set_dma_callback(sport->tx_dma_chan, sport_tx_irq, sport); - - ret = request_dma(sport->rx_dma_chan, "SPORT RX Data"); - if (ret) { - dev_err(dev, "Unable to allocate DMA channel for sport rx\n"); - goto err_rx_dma; - } - set_dma_callback(sport->rx_dma_chan, sport_rx_irq, sport); - - ret = request_irq(sport->tx_err_irq, sport_err_irq, - 0, "SPORT TX ERROR", sport); - if (ret) { - dev_err(dev, "Unable to allocate tx error IRQ for sport\n"); - goto err_tx_irq; - } - - ret = request_irq(sport->rx_err_irq, sport_err_irq, - 0, "SPORT RX ERROR", sport); - if (ret) { - dev_err(dev, "Unable to allocate rx error IRQ for sport\n"); - goto err_rx_irq; - } - - return 0; -err_rx_irq: - free_irq(sport->tx_err_irq, sport); -err_tx_irq: - free_dma(sport->rx_dma_chan); -err_rx_dma: - free_dma(sport->tx_dma_chan); -err_tx_dma: - peripheral_free_list(sport->pin_req); - return ret; -} - -static void sport_free_resource(struct sport_device *sport) -{ - free_irq(sport->rx_err_irq, sport); - free_irq(sport->tx_err_irq, sport); - free_dma(sport->rx_dma_chan); - free_dma(sport->tx_dma_chan); - peripheral_free_list(sport->pin_req); -} - -struct sport_device *sport_create(struct platform_device *pdev) -{ - struct device *dev = &pdev->dev; - struct sport_device *sport; - int ret; - - sport = kzalloc(sizeof(*sport), GFP_KERNEL); - if (!sport) - return NULL; - - sport->pdev = pdev; - - ret = sport_get_resource(sport); - if (ret) - goto free_data; - - ret = sport_request_resource(sport); - if (ret) - goto free_data; - - dev_dbg(dev, "SPORT create success\n"); - return sport; -free_data: - kfree(sport); - return NULL; -} -EXPORT_SYMBOL(sport_create); - -void sport_delete(struct sport_device *sport) -{ - if (sport->tx_desc) - dma_free_coherent(NULL, sport->tx_desc_size, - sport->tx_desc, 0); - if (sport->rx_desc) - dma_free_coherent(NULL, sport->rx_desc_size, - sport->rx_desc, 0); - sport_free_resource(sport); - kfree(sport); -} -EXPORT_SYMBOL(sport_delete); - -MODULE_DESCRIPTION("Analog Devices BF6XX SPORT driver"); -MODULE_AUTHOR("Scott Jiang <Scott.Jiang.Linux(a)gmail.com>"); -MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/blackfin/bf6xx-sport.h b/sound/soc/blackfin/bf6xx-sport.h deleted file mode 100644 index 307d193cfcef..000000000000 --- a/sound/soc/blackfin/bf6xx-sport.h +++ /dev/null @@ -1,82 +0,0 @@ -/* - * bf6xx_sport - Analog Devices BF6XX SPORT driver - * - * Copyright (c) 2012 Analog Devices Inc. - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - * - * This program is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the - * GNU General Public License for more details. - * - * You should have received a copy of the GNU General Public License - * along with this program; if not, write to the Free Software - * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. - */ - -#ifndef _BF6XX_SPORT_H_ -#define _BF6XX_SPORT_H_ - -#include <linux/platform_device.h> -#include <asm/bfin_sport3.h> - -struct sport_device { - struct platform_device *pdev; - const unsigned short *pin_req; - struct sport_register *tx_regs; - struct sport_register *rx_regs; - int tx_dma_chan; - int rx_dma_chan; - int tx_err_irq; - int rx_err_irq; - - void (*tx_callback)(void *data); - void *tx_data; - void (*rx_callback)(void *data); - void *rx_data; - - struct dmasg *tx_desc; - struct dmasg *rx_desc; - unsigned int tx_desc_size; - unsigned int rx_desc_size; - unsigned char *tx_buf; - unsigned char *rx_buf; - unsigned int tx_fragsize; - unsigned int rx_fragsize; - unsigned int tx_frags; - unsigned int rx_frags; - unsigned int wdsize; -}; - -struct sport_params { - u32 spctl; - u32 div; -}; - -struct sport_device *sport_create(struct platform_device *pdev); -void sport_delete(struct sport_device *sport); -int sport_set_tx_params(struct sport_device *sport, - struct sport_params *params); -int sport_set_rx_params(struct sport_device *sport, - struct sport_params *params); -void sport_tx_start(struct sport_device *sport); -void sport_rx_start(struct sport_device *sport); -void sport_tx_stop(struct sport_device *sport); -void sport_rx_stop(struct sport_device *sport); -void sport_set_tx_callback(struct sport_device *sport, - void (*tx_callback)(void *), void *tx_data); -void sport_set_rx_callback(struct sport_device *sport, - void (*rx_callback)(void *), void *rx_data); -int sport_config_tx_dma(struct sport_device *sport, void *buf, - int fragcount, size_t fragsize); -int sport_config_rx_dma(struct sport_device *sport, void *buf, - int fragcount, size_t fragsize); -unsigned long sport_curr_offset_tx(struct sport_device *sport); -unsigned long sport_curr_offset_rx(struct sport_device *sport); - - - -#endif diff --git a/sound/soc/blackfin/bfin-eval-adau1373.c b/sound/soc/blackfin/bfin-eval-adau1373.c deleted file mode 100644 index 64b88fdc1f6c..000000000000 --- a/sound/soc/blackfin/bfin-eval-adau1373.c +++ /dev/null @@ -1,173 +0,0 @@ -/* - * Machine driver for EVAL-ADAU1373 on Analog Devices bfin - * evaluation boards. - * - * Copyright 2011 Analog Devices Inc. - * Author: Lars-Peter Clausen <lars(a)metafoo.de> - * - * Licensed under the GPL-2 or later. - */ - -#include <linux/module.h> -#include <linux/device.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/pcm_params.h> - -#include "../codecs/adau1373.h" - -static const struct snd_soc_dapm_widget bfin_eval_adau1373_dapm_widgets[] = { - SND_SOC_DAPM_LINE("Line In1", NULL), - SND_SOC_DAPM_LINE("Line In2", NULL), - SND_SOC_DAPM_LINE("Line In3", NULL), - SND_SOC_DAPM_LINE("Line In4", NULL), - - SND_SOC_DAPM_LINE("Line Out1", NULL), - SND_SOC_DAPM_LINE("Line Out2", NULL), - SND_SOC_DAPM_LINE("Stereo Out", NULL), - SND_SOC_DAPM_HP("Headphone", NULL), - SND_SOC_DAPM_HP("Earpiece", NULL), - SND_SOC_DAPM_SPK("Speaker", NULL), -}; - -static const struct snd_soc_dapm_route bfin_eval_adau1373_dapm_routes[] = { - { "AIN1L", NULL, "Line In1" }, - { "AIN1R", NULL, "Line In1" }, - { "AIN2L", NULL, "Line In2" }, - { "AIN2R", NULL, "Line In2" }, - { "AIN3L", NULL, "Line In3" }, - { "AIN3R", NULL, "Line In3" }, - { "AIN4L", NULL, "Line In4" }, - { "AIN4R", NULL, "Line In4" }, - - /* MICBIAS can be connected via a jumper to the line-in jack, since w - don't know which one is going to be used, just power both. */ - { "Line In1", NULL, "MICBIAS1" }, - { "Line In2", NULL, "MICBIAS1" }, - { "Line In3", NULL, "MICBIAS1" }, - { "Line In4", NULL, "MICBIAS1" }, - { "Line In1", NULL, "MICBIAS2" }, - { "Line In2", NULL, "MICBIAS2" }, - { "Line In3", NULL, "MICBIAS2" }, - { "Line In4", NULL, "MICBIAS2" }, - - { "Line Out1", NULL, "LOUT1L" }, - { "Line Out1", NULL, "LOUT1R" }, - { "Line Out2", NULL, "LOUT2L" }, - { "Line Out2", NULL, "LOUT2R" }, - { "Headphone", NULL, "HPL" }, - { "Headphone", NULL, "HPR" }, - { "Earpiece", NULL, "EP" }, - { "Speaker", NULL, "SPKL" }, - { "Stereo Out", NULL, "SPKR" }, -}; - -static int bfin_eval_adau1373_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - int pll_rate; - - switch (params_rate(params)) { - case 48000: - case 8000: - case 12000: - case 16000: - case 24000: - case 32000: - pll_rate = 48000 * 1024; - break; - case 44100: - case 7350: - case 11025: - case 14700: - case 22050: - case 29400: - pll_rate = 44100 * 1024; - break; - default: - return -EINVAL; - } - - ret = snd_soc_dai_set_pll(codec_dai, ADAU1373_PLL1, - ADAU1373_PLL_SRC_MCLK1, 12288000, pll_rate); - if (ret) - return ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, ADAU1373_CLK_SRC_PLL1, pll_rate, - SND_SOC_CLOCK_IN); - - return ret; -} - -static int bfin_eval_adau1373_codec_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_dai *codec_dai = rtd->codec_dai; - unsigned int pll_rate = 48000 * 1024; - int ret; - - ret = snd_soc_dai_set_pll(codec_dai, ADAU1373_PLL1, - ADAU1373_PLL_SRC_MCLK1, 12288000, pll_rate); - if (ret) - return ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, ADAU1373_CLK_SRC_PLL1, pll_rate, - SND_SOC_CLOCK_IN); - - return ret; -} -static const struct snd_soc_ops bfin_eval_adau1373_ops = { - .hw_params = bfin_eval_adau1373_hw_params, -}; - -static struct snd_soc_dai_link bfin_eval_adau1373_dai = { - .name = "adau1373", - .stream_name = "adau1373", - .cpu_dai_name = "bfin-i2s.0", - .codec_dai_name = "adau1373-aif1", - .platform_name = "bfin-i2s-pcm-audio", - .codec_name = "adau1373.0-001a", - .ops = &bfin_eval_adau1373_ops, - .init = bfin_eval_adau1373_codec_init, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, -}; - -static struct snd_soc_card bfin_eval_adau1373 = { - .name = "bfin-eval-adau1373", - .owner = THIS_MODULE, - .dai_link = &bfin_eval_adau1373_dai, - .num_links = 1, - - .dapm_widgets = bfin_eval_adau1373_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(bfin_eval_adau1373_dapm_widgets), - .dapm_routes = bfin_eval_adau1373_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(bfin_eval_adau1373_dapm_routes), -}; - -static int bfin_eval_adau1373_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &bfin_eval_adau1373; - - card->dev = &pdev->dev; - - return devm_snd_soc_register_card(&pdev->dev, &bfin_eval_adau1373); -} - -static struct platform_driver bfin_eval_adau1373_driver = { - .driver = { - .name = "bfin-eval-adau1373", - .pm = &snd_soc_pm_ops, - }, - .probe = bfin_eval_adau1373_probe, -}; - -module_platform_driver(bfin_eval_adau1373_driver); - -MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); -MODULE_DESCRIPTION("ALSA SoC bfin adau1373 driver"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:bfin-eval-adau1373"); diff --git a/sound/soc/blackfin/bfin-eval-adau1701.c b/sound/soc/blackfin/bfin-eval-adau1701.c deleted file mode 100644 index 5c67f72cf9a9..000000000000 --- a/sound/soc/blackfin/bfin-eval-adau1701.c +++ /dev/null @@ -1,113 +0,0 @@ -/* - * Machine driver for EVAL-ADAU1701MINIZ on Analog Devices bfin - * evaluation boards. - * - * Copyright 2011 Analog Devices Inc. - * Author: Lars-Peter Clausen <lars(a)metafoo.de> - * - * Licensed under the GPL-2 or later. - */ - -#include <linux/module.h> -#include <linux/device.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/pcm_params.h> - -#include "../codecs/adau1701.h" - -static const struct snd_soc_dapm_widget bfin_eval_adau1701_dapm_widgets[] = { - SND_SOC_DAPM_SPK("Speaker", NULL), - SND_SOC_DAPM_LINE("Line Out", NULL), - SND_SOC_DAPM_LINE("Line In", NULL), -}; - -static const struct snd_soc_dapm_route bfin_eval_adau1701_dapm_routes[] = { - { "Speaker", NULL, "OUT0" }, - { "Speaker", NULL, "OUT1" }, - { "Line Out", NULL, "OUT2" }, - { "Line Out", NULL, "OUT3" }, - - { "IN0", NULL, "Line In" }, - { "IN1", NULL, "Line In" }, -}; - -static int bfin_eval_adau1701_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, ADAU1701_CLK_SRC_OSC, 12288000, - SND_SOC_CLOCK_IN); - - return ret; -} - -static struct snd_soc_ops bfin_eval_adau1701_ops = { - .hw_params = bfin_eval_adau1701_hw_params, -}; - -#define BFIN_EVAL_ADAU1701_DAI_FMT (SND_SOC_DAIFMT_I2S | \ - SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM) - -static struct snd_soc_dai_link bfin_eval_adau1701_dai[] = { - { - .name = "adau1701", - .stream_name = "adau1701", - .cpu_dai_name = "bfin-i2s.0", - .codec_dai_name = "adau1701", - .platform_name = "bfin-i2s-pcm-audio", - .codec_name = "adau1701.0-0034", - .ops = &bfin_eval_adau1701_ops, - .dai_fmt = BFIN_EVAL_ADAU1701_DAI_FMT, - }, - { - .name = "adau1701", - .stream_name = "adau1701", - .cpu_dai_name = "bfin-i2s.1", - .codec_dai_name = "adau1701", - .platform_name = "bfin-i2s-pcm-audio", - .codec_name = "adau1701.0-0034", - .ops = &bfin_eval_adau1701_ops, - .dai_fmt = BFIN_EVAL_ADAU1701_DAI_FMT, - }, -}; - -static struct snd_soc_card bfin_eval_adau1701 = { - .name = "bfin-eval-adau1701", - .owner = THIS_MODULE, - .dai_link = &bfin_eval_adau1701_dai[CONFIG_SND_BF5XX_SPORT_NUM], - .num_links = 1, - - .dapm_widgets = bfin_eval_adau1701_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(bfin_eval_adau1701_dapm_widgets), - .dapm_routes = bfin_eval_adau1701_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(bfin_eval_adau1701_dapm_routes), -}; - -static int bfin_eval_adau1701_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &bfin_eval_adau1701; - - card->dev = &pdev->dev; - - return devm_snd_soc_register_card(&pdev->dev, &bfin_eval_adau1701); -} - -static struct platform_driver bfin_eval_adau1701_driver = { - .driver = { - .name = "bfin-eval-adau1701", - .pm = &snd_soc_pm_ops, - }, - .probe = bfin_eval_adau1701_probe, -}; - -module_platform_driver(bfin_eval_adau1701_driver); - -MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); -MODULE_DESCRIPTION("ALSA SoC bfin ADAU1701 driver"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:bfin-eval-adau1701"); diff --git a/sound/soc/blackfin/bfin-eval-adau1x61.c b/sound/soc/blackfin/bfin-eval-adau1x61.c deleted file mode 100644 index fddfe00c9d69..000000000000 --- a/sound/soc/blackfin/bfin-eval-adau1x61.c +++ /dev/null @@ -1,142 +0,0 @@ -/* - * Machine driver for EVAL-ADAU1x61MINIZ on Analog Devices bfin - * evaluation boards. - * - * Copyright 2011-2014 Analog Devices Inc. - * Author: Lars-Peter Clausen <lars(a)metafoo.de> - * - * Licensed under the GPL-2 or later. - */ - -#include <linux/module.h> -#include <linux/device.h> -#include <linux/slab.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/pcm_params.h> - -#include "../codecs/adau17x1.h" - -static const struct snd_soc_dapm_widget bfin_eval_adau1x61_dapm_widgets[] = { - SND_SOC_DAPM_LINE("In 1", NULL), - SND_SOC_DAPM_LINE("In 2", NULL), - SND_SOC_DAPM_LINE("In 3-4", NULL), - - SND_SOC_DAPM_LINE("Diff Out L", NULL), - SND_SOC_DAPM_LINE("Diff Out R", NULL), - SND_SOC_DAPM_LINE("Stereo Out", NULL), - SND_SOC_DAPM_HP("Capless HP Out", NULL), -}; - -static const struct snd_soc_dapm_route bfin_eval_adau1x61_dapm_routes[] = { - { "LAUX", NULL, "In 3-4" }, - { "RAUX", NULL, "In 3-4" }, - { "LINP", NULL, "In 1" }, - { "LINN", NULL, "In 1"}, - { "RINP", NULL, "In 2" }, - { "RINN", NULL, "In 2" }, - - { "In 1", NULL, "MICBIAS" }, - { "In 2", NULL, "MICBIAS" }, - - { "Capless HP Out", NULL, "LHP" }, - { "Capless HP Out", NULL, "RHP" }, - { "Diff Out L", NULL, "LOUT" }, - { "Diff Out R", NULL, "ROUT" }, - { "Stereo Out", NULL, "LOUT" }, - { "Stereo Out", NULL, "ROUT" }, -}; - -static int bfin_eval_adau1x61_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int pll_rate; - int ret; - - switch (params_rate(params)) { - case 48000: - case 8000: - case 12000: - case 16000: - case 24000: - case 32000: - case 96000: - pll_rate = 48000 * 1024; - break; - case 44100: - case 7350: - case 11025: - case 14700: - case 22050: - case 29400: - case 88200: - pll_rate = 44100 * 1024; - break; - default: - return -EINVAL; - } - - ret = snd_soc_dai_set_pll(codec_dai, ADAU17X1_PLL, - ADAU17X1_PLL_SRC_MCLK, 12288000, pll_rate); - if (ret) - return ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, ADAU17X1_CLK_SRC_PLL, pll_rate, - SND_SOC_CLOCK_IN); - - return ret; -} - -static const struct snd_soc_ops bfin_eval_adau1x61_ops = { - .hw_params = bfin_eval_adau1x61_hw_params, -}; - -static struct snd_soc_dai_link bfin_eval_adau1x61_dai = { - .name = "adau1x61", - .stream_name = "adau1x61", - .cpu_dai_name = "bfin-i2s.0", - .codec_dai_name = "adau-hifi", - .platform_name = "bfin-i2s-pcm-audio", - .codec_name = "adau1761.0-0038", - .ops = &bfin_eval_adau1x61_ops, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, -}; - -static struct snd_soc_card bfin_eval_adau1x61 = { - .name = "bfin-eval-adau1x61", - .owner = THIS_MODULE, - .driver_name = "eval-adau1x61", - .dai_link = &bfin_eval_adau1x61_dai, - .num_links = 1, - - .dapm_widgets = bfin_eval_adau1x61_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(bfin_eval_adau1x61_dapm_widgets), - .dapm_routes = bfin_eval_adau1x61_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(bfin_eval_adau1x61_dapm_routes), - .fully_routed = true, -}; - -static int bfin_eval_adau1x61_probe(struct platform_device *pdev) -{ - bfin_eval_adau1x61.dev = &pdev->dev; - - return devm_snd_soc_register_card(&pdev->dev, &bfin_eval_adau1x61); -} - -static struct platform_driver bfin_eval_adau1x61_driver = { - .driver = { - .name = "bfin-eval-adau1x61", - .pm = &snd_soc_pm_ops, - }, - .probe = bfin_eval_adau1x61_probe, -}; -module_platform_driver(bfin_eval_adau1x61_driver); - -MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); -MODULE_DESCRIPTION("ALSA SoC bfin adau1x61 driver"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:bfin-eval-adau1x61"); diff --git a/sound/soc/blackfin/bfin-eval-adau1x81.c b/sound/soc/blackfin/bfin-eval-adau1x81.c deleted file mode 100644 index 3e01cbe53fc7..000000000000 --- a/sound/soc/blackfin/bfin-eval-adau1x81.c +++ /dev/null @@ -1,129 +0,0 @@ -/* - * Machine driver for EVAL-ADAU1x81 on Analog Devices bfin - * evaluation boards. - * - * Copyright 2011-2014 Analog Devices Inc. - * Author: Lars-Peter Clausen <lars(a)metafoo.de> - * - * Licensed under the GPL-2 or later. - */ - -#include <linux/module.h> -#include <linux/device.h> -#include <linux/slab.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> -#include <sound/pcm_params.h> - -#include "../codecs/adau17x1.h" - -static const struct snd_soc_dapm_widget bfin_eval_adau1x81_dapm_widgets[] = { - SND_SOC_DAPM_LINE("Stereo In", NULL), - SND_SOC_DAPM_LINE("Beep", NULL), - - SND_SOC_DAPM_SPK("Speaker", NULL), - SND_SOC_DAPM_HP("Headphone", NULL), -}; - -static const struct snd_soc_dapm_route bfin_eval_adau1x81_dapm_routes[] = { - { "BEEP", NULL, "Beep" }, - { "LMIC", NULL, "Stereo In" }, - { "LMIC", NULL, "Stereo In" }, - - { "Headphone", NULL, "AOUTL" }, - { "Headphone", NULL, "AOUTR" }, - { "Speaker", NULL, "SP" }, -}; - -static int bfin_eval_adau1x81_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int pll_rate; - int ret; - - switch (params_rate(params)) { - case 48000: - case 8000: - case 12000: - case 16000: - case 24000: - case 32000: - case 96000: - pll_rate = 48000 * 1024; - break; - case 44100: - case 7350: - case 11025: - case 14700: - case 22050: - case 29400: - case 88200: - pll_rate = 44100 * 1024; - break; - default: - return -EINVAL; - } - - ret = snd_soc_dai_set_pll(codec_dai, ADAU17X1_PLL, - ADAU17X1_PLL_SRC_MCLK, 12288000, pll_rate); - if (ret) - return ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, ADAU17X1_CLK_SRC_PLL, pll_rate, - SND_SOC_CLOCK_IN); - - return ret; -} - -static const struct snd_soc_ops bfin_eval_adau1x81_ops = { - .hw_params = bfin_eval_adau1x81_hw_params, -}; - -static struct snd_soc_dai_link bfin_eval_adau1x81_dai = { - .name = "adau1x81", - .stream_name = "adau1x81", - .cpu_dai_name = "bfin-i2s.0", - .codec_dai_name = "adau-hifi", - .platform_name = "bfin-i2s-pcm-audio", - .codec_name = "adau1781.0-0038", - .ops = &bfin_eval_adau1x81_ops, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, -}; - -static struct snd_soc_card bfin_eval_adau1x81 = { - .name = "bfin-eval-adau1x81", - .driver_name = "eval-adau1x81", - .dai_link = &bfin_eval_adau1x81_dai, - .num_links = 1, - - .dapm_widgets = bfin_eval_adau1x81_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(bfin_eval_adau1x81_dapm_widgets), - .dapm_routes = bfin_eval_adau1x81_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(bfin_eval_adau1x81_dapm_routes), - .fully_routed = true, -}; - -static int bfin_eval_adau1x81_probe(struct platform_device *pdev) -{ - bfin_eval_adau1x81.dev = &pdev->dev; - - return devm_snd_soc_register_card(&pdev->dev, &bfin_eval_adau1x81); -} - -static struct platform_driver bfin_eval_adau1x81_driver = { - .driver = { - .name = "bfin-eval-adau1x81", - .pm = &snd_soc_pm_ops, - }, - .probe = bfin_eval_adau1x81_probe, -}; -module_platform_driver(bfin_eval_adau1x81_driver); - -MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); -MODULE_DESCRIPTION("ALSA SoC bfin adau1x81 driver"); -MODULE_LICENSE("GPL"); -MODULE_ALIAS("platform:bfin-eval-adau1x81"); diff --git a/sound/soc/blackfin/bfin-eval-adav80x.c b/sound/soc/blackfin/bfin-eval-adav80x.c deleted file mode 100644 index 99e5ecabdcda..000000000000 --- a/sound/soc/blackfin/bfin-eval-adav80x.c +++ /dev/null @@ -1,145 +0,0 @@ -/* - * Machine driver for EVAL-ADAV801 and EVAL-ADAV803 on Analog Devices bfin - * evaluation boards. - * - * Copyright 2011 Analog Devices Inc. - * Author: Lars-Peter Clausen <lars(a)metafoo.de> - * - * Licensed under the GPL-2 or later. - */ - -#include <linux/init.h> -#include <linux/platform_device.h> -#include <linux/module.h> -#include <sound/core.h> -#include <sound/pcm.h> -#include <sound/soc.h> - -#include "../codecs/adav80x.h" - -static const struct snd_soc_dapm_widget bfin_eval_adav80x_dapm_widgets[] = { - SND_SOC_DAPM_LINE("Line Out", NULL), - SND_SOC_DAPM_LINE("Line In", NULL), -}; - -static const struct snd_soc_dapm_route bfin_eval_adav80x_dapm_routes[] = { - { "Line Out", NULL, "VOUTL" }, - { "Line Out", NULL, "VOUTR" }, - - { "VINL", NULL, "Line In" }, - { "VINR", NULL, "Line In" }, -}; - -static int bfin_eval_adav80x_hw_params(struct snd_pcm_substream *substream, - struct snd_pcm_hw_params *params) -{ - struct snd_soc_pcm_runtime *rtd = substream->private_data; - struct snd_soc_dai *codec_dai = rtd->codec_dai; - int ret; - - ret = snd_soc_dai_set_pll(codec_dai, ADAV80X_PLL1, ADAV80X_PLL_SRC_XTAL, - 27000000, params_rate(params) * 256); - if (ret) - return ret; - - ret = snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_PLL1, - params_rate(params) * 256, SND_SOC_CLOCK_IN); - - return ret; -} - -static int bfin_eval_adav80x_codec_init(struct snd_soc_pcm_runtime *rtd) -{ - struct snd_soc_dai *codec_dai = rtd->codec_dai; - - snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_SYSCLK1, 0, - SND_SOC_CLOCK_OUT); - snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_SYSCLK2, 0, - SND_SOC_CLOCK_OUT); - snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_SYSCLK3, 0, - SND_SOC_CLOCK_OUT); - - snd_soc_dai_set_sysclk(codec_dai, ADAV80X_CLK_XTAL, 2700000, 0); - - return 0; -} - -static const struct snd_soc_ops bfin_eval_adav80x_ops = { - .hw_params = bfin_eval_adav80x_hw_params, -}; - -static struct snd_soc_dai_link bfin_eval_adav80x_dais[] = { - { - .name = "adav80x", - .stream_name = "ADAV80x HiFi", - .cpu_dai_name = "bfin-i2s.0", - .codec_dai_name = "adav80x-hifi", - .platform_name = "bfin-i2s-pcm-audio", - .init = bfin_eval_adav80x_codec_init, - .ops = &bfin_eval_adav80x_ops, - .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBM_CFM, - }, -}; - -static struct snd_soc_card bfin_eval_adav80x = { - .name = "bfin-eval-adav80x", - .owner = THIS_MODULE, - .dai_link = bfin_eval_adav80x_dais, - .num_links = ARRAY_SIZE(bfin_eval_adav80x_dais), - - .dapm_widgets = bfin_eval_adav80x_dapm_widgets, - .num_dapm_widgets = ARRAY_SIZE(bfin_eval_adav80x_dapm_widgets), - .dapm_routes = bfin_eval_adav80x_dapm_routes, - .num_dapm_routes = ARRAY_SIZE(bfin_eval_adav80x_dapm_routes), -}; - -enum bfin_eval_adav80x_type { - BFIN_EVAL_ADAV801, - BFIN_EVAL_ADAV803, -}; - -static int bfin_eval_adav80x_probe(struct platform_device *pdev) -{ - struct snd_soc_card *card = &bfin_eval_adav80x; - const char *codec_name; - - switch (platform_get_device_id(pdev)->driver_data) { - case BFIN_EVAL_ADAV801: - codec_name = "spi0.1"; - break; - case BFIN_EVAL_ADAV803: - codec_name = "adav803.0-0034"; - break; - default: - return -EINVAL; - } - - bfin_eval_adav80x_dais[0].codec_name = codec_name; - - card->dev = &pdev->dev; - - return devm_snd_soc_register_card(&pdev->dev, &bfin_eval_adav80x); -} - -static const struct platform_device_id bfin_eval_adav80x_ids[] = { - { "bfin-eval-adav801", BFIN_EVAL_ADAV801 }, - { "bfin-eval-adav803", BFIN_EVAL_ADAV803 }, - { }, -}; -MODULE_DEVICE_TABLE(platform, bfin_eval_adav80x_ids); - -static struct platform_driver bfin_eval_adav80x_driver = { - .driver = { - .name = "bfin-eval-adav80x", - .pm = &snd_soc_pm_ops, - }, - .probe = bfin_eval_adav80x_probe, - .id_table = bfin_eval_adav80x_ids, -}; - -module_platform_driver(bfin_eval_adav80x_driver); - -MODULE_AUTHOR("Lars-Peter Clausen <lars(a)metafoo.de>"); -MODULE_DESCRIPTION("ALSA SoC bfin adav80x driver"); -MODULE_LICENSE("GPL"); -- 2.9.0

2 1

Re: [alsa-devel] [PATCH] [sound] hdac-codec runtime suspended at PM:Suspend.
by Rafael J. Wysocki 14 Mar '18

14 Mar '18

On Wed, Mar 14, 2018 at 11:38 AM, Anshuman Gupta <anshuman.gupta(a)intel.com> wrote: > On Mon, Mar 12, 2018 at 12:26:53PM +0100, Rafael J. Wysocki wrote: >> On Mon, Mar 12, 2018 at 12:17 PM, Anshuman Gupta >> <anshuman.gupta(a)intel.com> wrote: >> > Keep hdac-codec to be in runtime suspended while entering to suspend. >> > If hdac-codec is already in runtime suspend state skip its power down >> > sequence in prepare, power up sequence in complete phase. >> > >> > Avoid resuming hdac controller PCI device 00:1f.3 from runtime suspend >> > state in case hdac-codec already in runtime-suspend state, this is >> > unnecessary and block the direct complete even for hdac controller >> > PCI device 00:1f.3. >> > >> > This enabled direct complete path for hdac-codec and PCI device 00:1f.3. >> > >> > Signed-off-by: Anshuman Gupta <anshuman.gupta(a)intel.com> >> > --- >> > sound/soc/codecs/hdac_hdmi.c | 4 ++++ >> > 1 file changed, 4 insertions(+) >> > >> > diff --git a/sound/soc/codecs/hdac_hdmi.c b/sound/soc/codecs/hdac_hdmi.c >> > index f3b4f4d..810a8a6 100644 >> > --- a/sound/soc/codecs/hdac_hdmi.c >> > +++ b/sound/soc/codecs/hdac_hdmi.c >> > @@ -1852,6 +1852,8 @@ static int hdmi_codec_prepare(struct device *dev) >> > struct hdac_ext_device *edev = to_hda_ext_device(dev); >> > struct hdac_device *hdac = &edev->hdac; >> > >> > + if (pm_runtime_status_suspended(dev)) >> > + return 1; >> >> This is racy in principle, because the runtime PM status can still >> change after you've checked here. > Will using pm_runtime_disable/pm_runtime_enable for safe check of runtime > status avoids this race? It might help, but is this a leaf device or does it have children? >> But even if it isn't racy in practice, the following >> pm_runtime_get_sync() becomes redundant after it, doesn't it? > I have no idea but if there can be a case that PCI 00:1f.3 (hdac controller) > is runtime suspended and hdac hdmi codec is active, in that case it may be > required to use pm_runtime_get_sync() for hdac controller as it is parent of > hdac hdmi codec. >> >> > pm_runtime_get_sync(&edev->hdac.dev); >> > So it sounds like you need the "get" part, but you don't need the "sync" one, because you've just checked that the device is not suspended. I guess the idea is to return 1 for the direct-complete mechanism to kick in if the device is already suspended, but otherwise bump up its reference counter to prevent it from suspending going forward, right? >> > /* >> > @@ -1873,6 +1875,8 @@ static void hdmi_codec_complete(struct device *dev) >> > struct hdac_hdmi_priv *hdmi = edev->private_data; >> > struct hdac_device *hdac = &edev->hdac; >> > >> > + if (pm_runtime_status_suspended(dev)) >> > + return; >> >> That, again, is somewhat fragile from the concurrency perspective. >> And here you want to avoid the below if the device is still suspended. Why is the below code located in the ->complete callback anyway? Shouldn't it be there in the ->resume one? >> > /* Power up afg */ >> > snd_hdac_codec_read(hdac, hdac->afg, 0, AC_VERB_SET_POWER_STATE, >> > AC_PWRST_D0); >> > -- >> > 2.7.4

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[alsa-devel] [PATCH][next] ASoC: da7219: make structure da7219_dai_clks_ops static
by Colin King 14 Mar '18

14 Mar '18

From: Colin Ian King <colin.king(a)canonical.com> structure da7219_dai_clks_ops is local to the source and does not need to be in global scope, so make it static. Cleans up sparse warning: sound/soc/codecs/da7219.c:1777:22: warning: symbol 'da7219_dai_clks_ops' was not declared. Should it be static? Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/da7219.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 08f55d7a11ba..980a6a8bf56d 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1774,7 +1774,7 @@ static int da7219_dai_clks_is_prepared(struct clk_hw *hw) return !!(clk_reg & DA7219_DAI_CLK_EN_MASK); } -const struct clk_ops da7219_dai_clks_ops = { +static const struct clk_ops da7219_dai_clks_ops = { .prepare = da7219_dai_clks_prepare, .unprepare = da7219_dai_clks_unprepare, .is_prepared = da7219_dai_clks_is_prepared, -- 2.15.1

1 0

[alsa-devel] [GIT PULL]: kbl audio firmware update
by Sanyog Kale 14 Mar '18

14 Mar '18

Hi, Please PULL to receive the update for KBL audio DSP firmware. The following changes since commit 9cf0ec08c8fa93ff3aa1cb5f16189735909c3ac6: Mellanox: Add new mlxsw_spectrum firmware 13.1620.192 (2018-03-12 09:59:13 -0400) are available in the git repository at: https://github.com/sanyogrk/firmware.git kbl for you to fetch changes up to 4c0bf113a55975d702673e57c5542f150807ad66: linux-firmware: intel: Update Kabylake audio firmware (2018-03-14 16:23:08 +0530) ---------------------------------------------------------------- Sanyog Kale (1): linux-firmware: intel: Update Kabylake audio firmware WHENCE | 4 +++- intel/dsp_fw_kbl.bin | 2 +- intel/dsp_fw_kbl_v3420.bin | Bin 0 -> 243016 bytes 3 files changed, 4 insertions(+), 2 deletions(-) create mode 100644 intel/dsp_fw_kbl_v3420.bin Thanks, Sanyog --

2 1

[alsa-devel] [PATCH v2] sound: omap-twl4030: do not complain on -EPROBE_DEFER when initializing
by Ladislav Michl 14 Mar '18

14 Mar '18

Don't complain on -EPROBE_DEFER when initializing, the driver probe will be retried later. Signed-off-by: Ladislav Michl <ladis(a)linux-mips.org> Acked-by: Jarkko Nikula <jarkko.nikula(a)bitmer.com> --- Changes: -v2: fixed typo in commit log, added Jarkko Nikula's ack sound/soc/omap/omap-twl4030.c | 6 ++++-- 1 file changed, 4 insertions(+), 2 deletions(-) diff --git a/sound/soc/omap/omap-twl4030.c b/sound/soc/omap/omap-twl4030.c index cccc316743fa..7c1f71884768 100644 --- a/sound/soc/omap/omap-twl4030.c +++ b/sound/soc/omap/omap-twl4030.c @@ -322,8 +322,10 @@ static int omap_twl4030_probe(struct platform_device *pdev) snd_soc_card_set_drvdata(card, priv); ret = devm_snd_soc_register_card(&pdev->dev, card); if (ret) { - dev_err(&pdev->dev, "devm_snd_soc_register_card() failed: %d\n", - ret); + if (ret != -EPROBE_DEFER) + dev_err(&pdev->dev, + "devm_snd_soc_register_card() failed: %d\n", + ret); return ret; } -- 2.16.1

2 1

[alsa-devel] linux-next: merge conflict between drm-misc and sound trees
by Lukas Wunner 14 Mar '18

14 Mar '18

Dear Stephen, when rebuilding linux-next you're going to see a merge conflict in: sound/pci/hda/hda_intel.c betwen commit: 1ba8f9d30817 ("ALSA: hda: Add a power_save blacklist") from the sound tree and commit: 07f4f97d7b4b ("vga_switcheroo: Use device link for HDA controller") from the drm-misc tree. I recommend fixing it up like below. -- >8 -- @@@ -2284,16 -2298,22 +2303,30 @@@ chip->running = 1; azx_add_card_list(chip); + /* + * The discrete GPU cannot power down unless the HDA controller runtime + * suspends, so activate runtime PM on codecs even if power_save == 0. + */ + if (use_vga_switcheroo(hda)) + list_for_each_codec(codec, &chip->bus) + codec->auto_runtime_pm = 1; + - snd_hda_set_power_save(&chip->bus, power_save * 1000); + val = power_save; + #ifdef CONFIG_PM + if (val == -1) { + const struct snd_pci_quirk *q; + + val = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; + q = snd_pci_quirk_lookup(chip->pci, power_save_blacklist); + if (q && val) { + dev_info(chip->card->dev, "device %04x:%04x is on the power_save blacklist, forcing power_save to 0\n", + q->subvendor, q->subdevice); + val = 0; + } + } + #endif /* CONFIG_PM */ + snd_hda_set_power_save(&chip->bus, val * 1000); - if (azx_has_pm_runtime(chip) || hda->use_vga_switcheroo) + if (azx_has_pm_runtime(chip)) pm_runtime_put_autosuspend(&pci->dev); out_free:

3 3

[alsa-devel] [Xen-devel][PATCH 0/2] sndif: add explicit back and front synchronization
by Oleksandr Andrushchenko 14 Mar '18

14 Mar '18

From: Oleksandr Andrushchenko <oleksandr_andrushchenko(a)epam.com> Hi, all! Foreword ======== This change is aimed to add support for explicit back and front synchronization during playback and capture in response to comments raised during upstream attempt of the para-virtualized sound frontend driver for Xen [1], [2] and gather opinions from the relevant communities (ALSA, Xen) on the change. The relevant backend is implemented as a user-space application [3] and uses accompanying helper library [4]. Both frontend driver and backend were tested on real HW running Xen hypervisor (Renesas R-Car ARM based H3/M3 boards, x86) to make sure the proposed solution does work. Rationale ========= During the first attempt to upstream the Linux front driver [5] number of comments and concerns were raised, one of the biggest flaws in the design were questioned by both Clemens Ladisch [6] and Takashi Sakamoto [7]: the absence of synchronization between frontend and backend during capture/playback. Two options were discussed: “In design of ALSA PCM core, drivers are expected to synchronize to actual hardwares for semi-realtime data transmission. The synchronization is done by two points: 1) Interrupts to respond events from actual hardwares. 2) Positions of actual data transmission in any serial sound interfaces of actual hardwares. “ and finally a change to the existing protocol was suggested: “In 'include/xen/interface/io/sndif.h', there's no functionalities I described the above: 1. notifications from DomU to Dom0 about the size of period for interrupts from actual hardwares. Or no way from Dom0 to DomU about the configured size of the period. 2. notifications of the interrupts from actual hardwares to DomU.” This is implemented as a change to the sndif protocol and allows removing period emulation: 1. Introduced a new event channel from back to front 2. New event with number of bytes played/captured (XENSND_EVT_CUR_POS, to be used for sending snd_pcm_period_elapsed at frontend (in Linux implementation). Sent in bytes, not frames to make the protocol generic and consistent) 3. New request for playback/capture control (XENSND_OP_TRIGGER) with start/pause/stop/resume sub-ops 4. Playback/capture buffer size is set on the backend side via XENSND_FIELD_BUFFER_SIZE XenStore entry Waiting for your valuable comments, Thank you, Oleksandr [1] https://github.com/andr2000/linux/commits/snd_upstream_v1 [2] https://git.kernel.org/pub/scm/linux/kernel/git/torvalds/linux.git/tree/inc… [3] https://github.com/xen-troops/snd_be [4] https://github.com/xen-troops/libxenbe [5] https://lkml.org/lkml/2017/8/7/363 [6] http://mailman.alsa-project.org/pipermail/alsa-devel/2017-August/123617.html [7] http://mailman.alsa-project.org/pipermail/alsa-devel/2017-August/123744.html Oleksandr Andrushchenko (2): sndif: introduce protocol version sndif: add explicit back and front synchronization xen/include/public/io/sndif.h | 173 +++++++++++++++++++++++++++++++++++++++++- 1 file changed, 170 insertions(+), 3 deletions(-) -- 2.7.4

4 29

Re: [alsa-devel] [PATCH v2 0/7] Modernize vga_switcheroo by using device link for HDA
by Lukas Wunner 14 Mar '18

14 Mar '18

On Tue, Mar 06, 2018 at 11:29:40AM +0100, Daniel Vetter wrote: > On Sat, Mar 03, 2018 at 10:53:24AM +0100, Lukas Wunner wrote: > > Modernize vga_switcheroo by using a device link to enforce a runtime PM > > dependency from an HDA controller to the GPU it's integrated into, v2. > > > > https://github.com/l1k/linux/commits/switcheroo_devlink_v2 > > This all looks really reasonable and like a good cleanup, but it's a bit > too much detail so I'll punt review to someone else with more clue. Patches [3/7] to [7/7] were reviewed by Peter Wu, the HDA bits in patch [5/7] additionally by Takashi. Patch [2/7] was acked by Bjorn. There was no ack for patch [1/7] (authored by Rafael), but it adressed the objection Bjorn raised against my original patch, so I'm assuming Bjorn is okay with it. (Bjorn, please let me know if that isn't the case.) The series has been tested on 5 systems, which raises the confidence: 2x AMD PowerXpress (Mike Lothian, Kai Heng Feng) 2x Nvidia Optimus (Denis Lisov, Peter Wu) 1x MacBook Pro The issues found during Peter Wu's thorough testing appear to all be unrelated to this series, as per my e-mail yesterday. If there are no objections, I plan to push the series to drm-misc-next by the middle of the coming week so that it would still catch the last train to 4.17. Thanks, Lukas

3 3

[alsa-devel] [PATCH - alsa-utils 1/1] aplay: Fix wav file not being split on 32 bit platforms
by erwin 13 Mar '18

13 Mar '18

On my 32 bit armhf board arecord exits because of write() returning EFBIG when the output file size reaches 2147483647 bytes. To fix this, include generated header file before system header files so that _FILE_OFFSET_BITS=64 is used properly, as required in documentation "man feature_test_macros". Signed-off-by: erwin <nebelbank(a)posteo.de> diff --git a/aplay/aplay.c b/aplay/aplay.c index 6b740c2..bbd7fff 100644 --- a/aplay/aplay.c +++ b/aplay/aplay.c @@ -27,6 +27,7 @@ */ #define _GNU_SOURCE +#include "aconfig.h" #include <stdio.h> #include <malloc.h> #include <unistd.h> @@ -49,7 +50,6 @@ #include <sys/stat.h> #include <sys/types.h> #include <endian.h> -#include "aconfig.h" #include "gettext.h" #include "formats.h" #include "version.h" -- 2.14.1

2 2

[alsa-devel] [asoc:topic/intel 22/38] sound/soc/intel/boards/cht_bsw_nau8824.c:137:29: error: passing argument 1 of 'nau8824_enable_jack_detect' from incompatible pointer type
by kbuild test robot 13 Mar '18

13 Mar '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/intel head: fc9fdd61c4809b14faa9b84fe3d8f4167a836326 commit: 3bf045d15f52529e8abe26543dbe22af44efb41a [22/38] ASoC: Intel: fix argument error in nau8824 machine config: i386-allmodconfig (attached as .config) compiler: gcc-7 (Debian 7.3.0-1) 7.3.0 reproduce: git checkout 3bf045d15f52529e8abe26543dbe22af44efb41a # save the attached .config to linux build tree make ARCH=i386 All errors (new ones prefixed by >>): sound/soc/intel/boards/cht_bsw_nau8824.c: In function 'cht_codec_init': >> sound/soc/intel/boards/cht_bsw_nau8824.c:137:29: error: passing argument 1 of 'nau8824_enable_jack_detect' from incompatible pointer type [-Werror=incompatible-pointer-types] nau8824_enable_jack_detect(component, jack); ^~~~~~~~~ In file included from sound/soc/intel/boards/cht_bsw_nau8824.c:31:0: sound/soc/intel/boards/../../codecs/nau8824.h:474:5: note: expected 'struct snd_soc_codec *' but argument is of type 'struct snd_soc_component *' int nau8824_enable_jack_detect(struct snd_soc_codec *codec, ^~~~~~~~~~~~~~~~~~~~~~~~~~ cc1: some warnings being treated as errors vim +/nau8824_enable_jack_detect +137 sound/soc/intel/boards/cht_bsw_nau8824.c 101 102 static int cht_codec_init(struct snd_soc_pcm_runtime *runtime) 103 { 104 struct cht_mc_private *ctx = snd_soc_card_get_drvdata(runtime->card); 105 struct snd_soc_jack *jack = &ctx->jack; 106 struct snd_soc_dai *codec_dai = runtime->codec_dai; 107 struct snd_soc_component *component = codec_dai->component; 108 int ret, jack_type; 109 110 /* TDM 4 slots 24 bit, set Rx & Tx bitmask to 4 active slots */ 111 ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xf, 0x1, 4, 24); 112 if (ret < 0) { 113 dev_err(runtime->dev, "can't set codec TDM slot %d\n", ret); 114 return ret; 115 } 116 117 /* NAU88L24 supports 4 butons headset detection 118 * KEY_MEDIA 119 * KEY_VOICECOMMAND 120 * KEY_VOLUMEUP 121 * KEY_VOLUMEDOWN 122 */ 123 jack_type = SND_JACK_HEADPHONE | SND_JACK_BTN_0 | SND_JACK_BTN_1 | 124 SND_JACK_BTN_2 | SND_JACK_BTN_3; 125 ret = snd_soc_card_jack_new(runtime->card, "Headset", jack_type, jack, 126 cht_bsw_jack_pins, ARRAY_SIZE(cht_bsw_jack_pins)); 127 if (ret) { 128 dev_err(runtime->dev, 129 "Headset Jack creation failed %d\n", ret); 130 return ret; 131 } 132 snd_jack_set_key(jack->jack, SND_JACK_BTN_0, KEY_MEDIA); 133 snd_jack_set_key(jack->jack, SND_JACK_BTN_1, KEY_VOICECOMMAND); 134 snd_jack_set_key(jack->jack, SND_JACK_BTN_2, KEY_VOLUMEUP); 135 snd_jack_set_key(jack->jack, SND_JACK_BTN_3, KEY_VOLUMEDOWN); 136 > 137 nau8824_enable_jack_detect(component, jack); 138 139 return ret; 140 } 141 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH v2] ASoC: dapm: add support to pinctrl dapm
by srinivas.kandagatla＠linaro.org 13 Mar '18

13 Mar '18

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Purpose of having pinctrl dapm is to dynamically put the pins in low power state when they are not actively used by the audio and saving power. Without this each driver has to set the pinctrl states, either during probe or dynamically depending on the callbacks received from ASoC core. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Changes since v1: - removed synchronize with the DAPM async queue include/sound/soc-dapm.h | 16 ++++++++++++++++ sound/soc/soc-dapm.c | 42 ++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 58 insertions(+) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 344b96c206a3..a6ce2de4e20a 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -269,6 +269,13 @@ struct device; .reg = SND_SOC_NOPM, .shift = wdelay, .event = dapm_regulator_event, \ .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD, \ .on_val = wflags} +#define SND_SOC_DAPM_PINCTRL(wname, active, sleep) \ +{ .id = snd_soc_dapm_pinctrl, .name = wname, \ + .priv = (&(struct snd_soc_dapm_pinctrl_priv) \ + { .active_state = active, .sleep_state = sleep,}), \ + .reg = SND_SOC_NOPM, .event = dapm_pinctrl_event, \ + .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD } + /* dapm kcontrol types */ @@ -374,6 +381,8 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); int dapm_clock_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); +int dapm_pinctrl_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event); /* dapm controls */ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, @@ -500,6 +509,7 @@ enum snd_soc_dapm_type { snd_soc_dapm_pre, /* machine specific pre widget - exec first */ snd_soc_dapm_post, /* machine specific post widget - exec last */ snd_soc_dapm_supply, /* power/clock supply */ + snd_soc_dapm_pinctrl, /* pinctrl */ snd_soc_dapm_regulator_supply, /* external regulator */ snd_soc_dapm_clock_supply, /* external clock */ snd_soc_dapm_aif_in, /* audio interface input */ @@ -581,6 +591,7 @@ struct snd_soc_dapm_widget { void *priv; /* widget specific data */ struct regulator *regulator; /* attached regulator */ + struct pinctrl *pinctrl; /* attached pinctrl */ const struct snd_soc_pcm_stream *params; /* params for dai links */ unsigned int num_params; /* number of params for dai links */ unsigned int params_select; /* currently selected param for dai link */ @@ -683,6 +694,11 @@ struct snd_soc_dapm_stats { int neighbour_checks; }; +struct snd_soc_dapm_pinctrl_priv { + const char *active_state; + const char *sleep_state; +}; + /** * snd_soc_dapm_init_bias_level() - Initialize DAPM bias level * @dapm: The DAPM context to initialize diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 40085a4c8f92..ac4f13230faf 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -35,6 +35,7 @@ #include <linux/debugfs.h> #include <linux/pm_runtime.h> #include <linux/regulator/consumer.h> +#include <linux/pinctrl/consumer.h> #include <linux/clk.h> #include <linux/slab.h> #include <sound/core.h> @@ -72,6 +73,7 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, static int dapm_up_seq[] = { [snd_soc_dapm_pre] = 0, [snd_soc_dapm_regulator_supply] = 1, + [snd_soc_dapm_pinctrl] = 1, [snd_soc_dapm_clock_supply] = 1, [snd_soc_dapm_supply] = 2, [snd_soc_dapm_micbias] = 3, @@ -119,6 +121,7 @@ static int dapm_down_seq[] = { [snd_soc_dapm_dai_link] = 11, [snd_soc_dapm_supply] = 12, [snd_soc_dapm_clock_supply] = 13, + [snd_soc_dapm_pinctrl] = 13, [snd_soc_dapm_regulator_supply] = 13, [snd_soc_dapm_post] = 14, }; @@ -1287,6 +1290,31 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, } EXPORT_SYMBOL_GPL(dapm_regulator_event); +/* + * Handler for pinctrl widget. + */ +int dapm_pinctrl_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_dapm_pinctrl_priv *priv = w->priv; + struct pinctrl *p = w->pinctrl; + struct pinctrl_state *s; + + if (!p || !priv) + return -EIO; + + if (SND_SOC_DAPM_EVENT_ON(event)) + s = pinctrl_lookup_state(p, priv->active_state); + else + s = pinctrl_lookup_state(p, priv->sleep_state); + + if (IS_ERR(s)) + return PTR_ERR(s); + + return pinctrl_select_state(p, s); +} +EXPORT_SYMBOL_GPL(dapm_pinctrl_event); + /* * Handler for clock supply widget. */ @@ -1900,6 +1928,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) break; case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_pinctrl: case snd_soc_dapm_clock_supply: case snd_soc_dapm_micbias: if (d->target_bias_level < SND_SOC_BIAS_STANDBY) @@ -2313,6 +2342,7 @@ static ssize_t dapm_widget_show_component(struct snd_soc_component *cmpnt, case snd_soc_dapm_mixer_named_ctl: case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_pinctrl: case snd_soc_dapm_clock_supply: if (w->name) count += sprintf(buf + count, "%s: %s\n", @@ -3462,6 +3492,17 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, w->name, ret); } break; + case snd_soc_dapm_pinctrl: + w->pinctrl = devm_pinctrl_get(dapm->dev); + if (IS_ERR_OR_NULL(w->pinctrl)) { + ret = PTR_ERR(w->pinctrl); + if (ret == -EPROBE_DEFER) + return ERR_PTR(ret); + dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n", + w->name, ret); + return NULL; + } + break; case snd_soc_dapm_clock_supply: #ifdef CONFIG_CLKDEV_LOOKUP w->clk = devm_clk_get(dapm->dev, w->name); @@ -3541,6 +3582,7 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, break; case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_pinctrl: case snd_soc_dapm_clock_supply: case snd_soc_dapm_kcontrol: w->is_supply = 1; -- 2.15.1

2 1

[alsa-devel] [PATCH] ASoC: da7219: clkdev_drop usage depends on CONFIG_COMMON_CLK
by Adam Thomson 13 Mar '18

13 Mar '18

Fixes: ASoC: da7219: Add common clock usage for providing DAI clks clkdev_drop usage in the codec remove function should be dependent on if CONFIG_COMMON_CLK is defined for the platform, otherwise it can cause build failures for platforms that do not support this. The clkdev_* functions are still defined for those platforms, in headers and source but the functions are not linked in. This patch resolves this issue, so clkdev_drop is only used if CONFIG_COMMON_CLK is defined. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- sound/soc/codecs/da7219.c | 2 ++ 1 file changed, 2 insertions(+) diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 4412159..08f55d7 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -1966,8 +1966,10 @@ static void da7219_remove(struct snd_soc_component *component) da7219_aad_exit(component); +#ifdef CONFIG_COMMON_CLK if (da7219->dai_clks_lookup) clkdev_drop(da7219->dai_clks_lookup); +#endif /* Supplies */ regulator_bulk_disable(DA7219_NUM_SUPPLIES, da7219->supplies); -- 1.9.1

2 1

[alsa-devel] [PATCH 4/6] ASoC: Snow Chromebook DT bindings update
by Sylwester Nawrocki 13 Mar '18

13 Mar '18

The new DT bindings are introduced so sound on the HDMI interface is also supported. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- Documentation/devicetree/bindings/sound/snow.txt | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/snow.txt b/Documentation/devicetree/bindings/sound/snow.txt index 6df74f15687f..80fd9a87bb3f 100644 --- a/Documentation/devicetree/bindings/sound/snow.txt +++ b/Documentation/devicetree/bindings/sound/snow.txt @@ -5,8 +5,17 @@ Required properties: "google,snow-audio-max98090" or "google,snow-audio-max98091" or "google,snow-audio-max98095" -- samsung,i2s-controller: The phandle of the Samsung I2S controller -- samsung,audio-codec: The phandle of the audio codec +- samsung,i2s-controller (deprecated): The phandle of the Samsung I2S controller +- samsung,audio-codec (deprecated): The phandle of the audio codec + +Required sub-nodes: + + - 'cpu' subnode with a 'sound-dai' property containing the phandle of the I2S + controller + - 'codec' subnode with a 'sound-dai' property containing list of phandles + to the CODEC nodes, first entry must be the phandle of the MAX98090, + MAX98091 or MAX98095 CODEC (exact device type is indicated by the compatible + string) and the second entry must be the phandle of the HDMI IP block node Optional: - samsung,model: The name of the sound-card -- 2.14.2

3 2

[alsa-devel] [PATCH v3 1/2] ASoC: samsung: Snow DT bindings update
by Sylwester Nawrocki 13 Mar '18

13 Mar '18

The new DT bindings are introduced so sound on the HDMI interface can also be supported. The exisiting CPU and CODEC DAI phandle properties are marked as deprecated. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Reviewed-by: Rob Herring <robh(a)kernel.org> --- Changes since v2: - edited summary line. --- Documentation/devicetree/bindings/sound/snow.txt | 13 +++++++++++-- 1 file changed, 11 insertions(+), 2 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/snow.txt b/Documentation/devicetree/bindings/sound/snow.txt index 6df74f15687f..80fd9a87bb3f 100644 --- a/Documentation/devicetree/bindings/sound/snow.txt +++ b/Documentation/devicetree/bindings/sound/snow.txt @@ -5,8 +5,17 @@ Required properties: "google,snow-audio-max98090" or "google,snow-audio-max98091" or "google,snow-audio-max98095" -- samsung,i2s-controller: The phandle of the Samsung I2S controller -- samsung,audio-codec: The phandle of the audio codec +- samsung,i2s-controller (deprecated): The phandle of the Samsung I2S controller +- samsung,audio-codec (deprecated): The phandle of the audio codec + +Required sub-nodes: + + - 'cpu' subnode with a 'sound-dai' property containing the phandle of the I2S + controller + - 'codec' subnode with a 'sound-dai' property containing list of phandles + to the CODEC nodes, first entry must be the phandle of the MAX98090, + MAX98091 or MAX98095 CODEC (exact device type is indicated by the compatible + string) and the second entry must be the phandle of the HDMI IP block node Optional: - samsung,model: The name of the sound-card -- 2.14.2

3 4

[alsa-devel] [PATCH v2] ASoC: Intel: Skylake: Disable clock and power gating during FW/LIB download
by Guneshwor Singh 13 Mar '18

13 Mar '18

From: Sanyog Kale <sanyog.r.kale(a)intel.com> In order to achieve better DMA performance and reduce download time for firmware and library, it is recommended to disable dynamic clock and power gating. In some scenarios, DMA may wait to accumulate more data and last chunk of data never gets completed if dynamic clock and power gating is kept enabled. This patch adds support to disable/enable dynamic clock and power gating and use it during firmware and library download. Signed-off-by: Rakesh Ughreja <rakesh.a.ughreja(a)intel.com> Signed-off-by: Sanyog Kale <sanyog.r.kale(a)intel.com> Signed-off-by: Guneshwor Singh <guneshwor.o.singh(a)intel.com> --- Changes since v1: - Rebased on for-next. -- sound/soc/intel/skylake/skl-messages.c | 7 ++++++- sound/soc/intel/skylake/skl-pcm.c | 7 ++++++- sound/soc/intel/skylake/skl-sst-ipc.h | 3 +++ sound/soc/intel/skylake/skl.c | 27 +++++++++++++++++++++++++++ sound/soc/intel/skylake/skl.h | 4 ++++ 5 files changed, 46 insertions(+), 2 deletions(-) diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index f161907859f3..57d4a58522a6 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -417,11 +417,16 @@ int skl_resume_dsp(struct skl *skl) if (skl->skl_sst->is_first_boot == true) return 0; - /* disable dynamic clock gating during fw and lib download */ + /* + * Disable dynamic clock and power gating during firmware + * and library download + */ ctx->enable_miscbdcge(ctx->dev, false); + ctx->clock_power_gating(ctx->dev, false); ret = skl_dsp_wake(ctx->dsp); ctx->enable_miscbdcge(ctx->dev, true); + ctx->clock_power_gating(ctx->dev, true); if (ret < 0) return ret; diff --git a/sound/soc/intel/skylake/skl-pcm.c b/sound/soc/intel/skylake/skl-pcm.c index 84e40faeb869..15cb8ac3e374 100644 --- a/sound/soc/intel/skylake/skl-pcm.c +++ b/sound/soc/intel/skylake/skl-pcm.c @@ -1356,11 +1356,16 @@ static int skl_platform_soc_probe(struct snd_soc_component *component) return -EIO; } - /* disable dynamic clock gating during fw and lib download */ + /* + * Disable dynamic clock and power gating during firmware + * and library download + */ skl->skl_sst->enable_miscbdcge(component->dev, false); + skl->skl_sst->clock_power_gating(component->dev, false); ret = ops->init_fw(component->dev, skl->skl_sst); skl->skl_sst->enable_miscbdcge(component->dev, true); + skl->skl_sst->clock_power_gating(component->dev, true); if (ret < 0) { dev_err(component->dev, "Failed to boot first fw: %d\n", ret); return ret; diff --git a/sound/soc/intel/skylake/skl-sst-ipc.h b/sound/soc/intel/skylake/skl-sst-ipc.h index 55f2d2ce09df..f74f040dfd83 100644 --- a/sound/soc/intel/skylake/skl-sst-ipc.h +++ b/sound/soc/intel/skylake/skl-sst-ipc.h @@ -118,6 +118,9 @@ struct skl_sst { struct skl_d0i3_data d0i3; const struct skl_dsp_ops *dsp_ops; + + /* Callback to update dynamic clock and power gating registers */ + void (*clock_power_gating)(struct device *dev, bool enable); }; struct skl_ipc_init_instance_msg { diff --git a/sound/soc/intel/skylake/skl.c b/sound/soc/intel/skylake/skl.c index aa9edc84eefb..abf324747b29 100644 --- a/sound/soc/intel/skylake/skl.c +++ b/sound/soc/intel/skylake/skl.c @@ -94,6 +94,32 @@ static void skl_enable_miscbdcge(struct device *dev, bool enable) update_pci_dword(pci, AZX_PCIREG_CGCTL, AZX_CGCTL_MISCBDCGE_MASK, val); } +/** + * skl_clock_power_gating: Enable/Disable clock and power gating + * + * @dev: Device pointer + * @enable: Enable/Disable flag + */ +static void skl_clock_power_gating(struct device *dev, bool enable) +{ + struct pci_dev *pci = to_pci_dev(dev); + struct hdac_ext_bus *ebus = pci_get_drvdata(pci); + struct hdac_bus *bus = ebus_to_hbus(ebus); + u32 val; + + /* Update PDCGE bit of CGCTL register */ + val = enable ? AZX_CGCTL_ADSPDCGE : 0; + update_pci_dword(pci, AZX_PCIREG_CGCTL, AZX_CGCTL_ADSPDCGE, val); + + /* Update L1SEN bit of EM2 register */ + val = enable ? AZX_REG_VS_EM2_L1SEN : 0; + snd_hdac_chip_updatel(bus, VS_EM2, AZX_REG_VS_EM2_L1SEN, val); + + /* Update ADSPPGD bit of PGCTL register */ + val = enable ? 0 : AZX_PGCTL_ADSPPGD; + update_pci_dword(pci, AZX_PCIREG_PGCTL, AZX_PGCTL_ADSPPGD, val); +} + /* * While performing reset, controller may not come back properly causing * issues, so recommendation is to set CGCTL.MISCBDCGE to 0 then do reset @@ -916,6 +942,7 @@ static int skl_probe(struct pci_dev *pci, goto out_nhlt_free; } skl->skl_sst->enable_miscbdcge = skl_enable_miscbdcge; + skl->skl_sst->clock_power_gating = skl_clock_power_gating; } if (bus->mlcap) snd_hdac_ext_bus_get_ml_capabilities(ebus); diff --git a/sound/soc/intel/skylake/skl.h b/sound/soc/intel/skylake/skl.h index ca46ad1d0a08..0d5375cbcf6e 100644 --- a/sound/soc/intel/skylake/skl.h +++ b/sound/soc/intel/skylake/skl.h @@ -33,8 +33,10 @@ #define AZX_PCIREG_PGCTL 0x44 #define AZX_PGCTL_LSRMD_MASK (1 << 4) +#define AZX_PGCTL_ADSPPGD BIT(2) #define AZX_PCIREG_CGCTL 0x48 #define AZX_CGCTL_MISCBDCGE_MASK (1 << 6) +#define AZX_CGCTL_ADSPDCGE BIT(1) /* D0I3C Register fields */ #define AZX_REG_VS_D0I3C_CIP 0x1 /* Command in progress */ #define AZX_REG_VS_D0I3C_I3 0x4 /* D0i3 enable */ @@ -43,6 +45,8 @@ #define DMA_TRANSMITION_START 2 #define DMA_TRANSMITION_STOP 3 +#define AZX_REG_VS_EM2_L1SEN BIT(13) + struct skl_dsp_resource { u32 max_mcps; u32 max_mem; -- 2.16.2

2 1

[alsa-devel] [PATCH v2 0/6] ASoC: sun4i-i2s: Updates to the driver
by codekipper＠gmail.com 13 Mar '18

13 Mar '18

From: Marcus Cooper <codekipper(a)gmail.com> Hi All, here is a patch series to add some improvements to the sun4i-i2s driver found whilst getting slave clocking and hdmi audio working on the newer SoCs. The original attempt had some changes related to the slave work but I've left them out for this release as I would still like to do some more investigations. The functionality included with the new patch set has been extended to cover more sample resolutions, multi-lane data output for HDMI audio and loopback so block testing can be made without an external codec. I've also moved away from using tdm to set the slot width and now use a dedicated property. This has been tested on a Pine64 using the ES9023 audio POT board (https://github.com/codekipper/linux-sunxi/commits/upstream) and HDMI audio (https://github.com/codekipper/linux-sunxi/commits/sunxi-wip-a64) BR, CK --- v2 changes compared to v1 are: - removed slave mode changes which didn't set mclk and bclk div. - removed use of tdm and now use a dedicated property. - fix commit message to better explain reason for sign extending - add divider calculations for newer SoCs. - add support for multi-lane i2s data output. - add support for 20, 24 and 32 bit samples. - add loopback property so blocks can be tested without a codec. --- Marcus Cooper (6): ASoC: sun4i-i2s: Add slot width override ASoC: sun4i-i2s: Add regmap field to sign extend sample ASoC: sun4i-i2s: Correct divider calculations ASoC: sun4i-i2s: Add multi-lane functionality ASoc: sun4i-i2s: Add 20, 24 and 32 bit support ASoC: sun4i-i2s: Add support for loopback .../devicetree/bindings/sound/sun4i-i2s.txt | 11 ++ sound/soc/sunxi/sun4i-i2s.c | 199 ++++++++++++++++----- 2 files changed, 170 insertions(+), 40 deletions(-) -- 2.16.2

6 17

[alsa-devel] [PATCH] ALSA: pcm: Use krealloc() for resizing the rules array
by Takashi Iwai 13 Mar '18

13 Mar '18

Just a minor simplification. Change from kcalloc() shouldn't matter as each array element is fully initialized. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_lib.c | 8 ++------ 1 file changed, 2 insertions(+), 6 deletions(-) diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index a83152e7d387..f4a19509cccf 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -1129,16 +1129,12 @@ int snd_pcm_hw_rule_add(struct snd_pcm_runtime *runtime, unsigned int cond, if (constrs->rules_num >= constrs->rules_all) { struct snd_pcm_hw_rule *new; unsigned int new_rules = constrs->rules_all + 16; - new = kcalloc(new_rules, sizeof(*c), GFP_KERNEL); + new = krealloc(constrs->rules, new_rules * sizeof(*c), + GFP_KERNEL); if (!new) { va_end(args); return -ENOMEM; } - if (constrs->rules) { - memcpy(new, constrs->rules, - constrs->rules_num * sizeof(*c)); - kfree(constrs->rules); - } constrs->rules = new; constrs->rules_all = new_rules; } -- 2.16.2

2 1

[alsa-devel] [PATCH] ALSA: pcm: Remove VLA usage
by Takashi Iwai 13 Mar '18

13 Mar '18

A helper function used by snd_pcm_hw_refine() still keeps using VLA for timestamps of hw constraint rules that are non-fixed size. Let's replace the VLA with a simple kmalloc() array. Reference: https://lkml.org/lkml/2018/3/7/621 Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/pcm_native.c | 19 ++++++++++++------- 1 file changed, 12 insertions(+), 7 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 77ba50ddcf9e..756a9a3884a5 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -323,7 +323,7 @@ static int constrain_params_by_rules(struct snd_pcm_substream *substream, struct snd_pcm_hw_constraints *constrs = &substream->runtime->hw_constraints; unsigned int k; - unsigned int rstamps[constrs->rules_num]; + unsigned int *rstamps; unsigned int vstamps[SNDRV_PCM_HW_PARAM_LAST_INTERVAL + 1]; unsigned int stamp; struct snd_pcm_hw_rule *r; @@ -331,7 +331,7 @@ static int constrain_params_by_rules(struct snd_pcm_substream *substream, struct snd_mask old_mask; struct snd_interval old_interval; bool again; - int changed; + int changed, err = 0; /* * Each application of rule has own sequence number. @@ -339,8 +339,9 @@ static int constrain_params_by_rules(struct snd_pcm_substream *substream, * Each member of 'rstamps' array represents the sequence number of * recent application of corresponding rule. */ - for (k = 0; k < constrs->rules_num; k++) - rstamps[k] = 0; + rstamps = kcalloc(constrs->rules_num, sizeof(unsigned int), GFP_KERNEL); + if (!rstamps) + return -ENOMEM; /* * Each member of 'vstamps' array represents the sequence number of @@ -398,8 +399,10 @@ static int constrain_params_by_rules(struct snd_pcm_substream *substream, } changed = r->func(params, r); - if (changed < 0) - return changed; + if (changed < 0) { + err = changed; + goto out; + } /* * When the parameter is changed, notify it to the caller @@ -430,7 +433,9 @@ static int constrain_params_by_rules(struct snd_pcm_substream *substream, if (again) goto retry; - return 0; + out: + kfree(rstamps); + return err; } static int fixup_unreferenced_params(struct snd_pcm_substream *substream, -- 2.16.2

2 1

[alsa-devel] [PATCH] ALSA: hda: Add Icelake PCI ID
by Guneshwor Singh 13 Mar '18

13 Mar '18

Icelake is a next generation Intel platform. Add PCI ID for it. Signed-off-by: Guneshwor Singh <guneshwor.o.singh(a)intel.com> --- sound/pci/hda/hda_intel.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 96143df19b21..32acab080781 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -2426,6 +2426,9 @@ static const struct pci_device_id azx_ids[] = { /* Cannonlake */ { PCI_DEVICE(0x8086, 0x9dc8), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, + /* Icelake */ + { PCI_DEVICE(0x8086, 0x34c8), + .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_SKYLAKE}, /* Broxton-P(Apollolake) */ { PCI_DEVICE(0x8086, 0x5a98), .driver_data = AZX_DRIVER_SKL | AZX_DCAPS_INTEL_BROXTON }, -- 2.16.2

2 1

[alsa-devel] [PATCH v3 00/22] ASoC: rt5651: jack-detect fixes and improvements
by Hans de Goede 13 Mar '18

13 Mar '18

Hi All, Here is v3 of my rt5651 codec and Intel machine-driver series which mainly consists of various jack-detect related fixes and improvements. Changes in v3: -Drop patches of which the v2 version has been merged (Thanks Mark) -Split out dt-bindings documentation into a separate patch -Rework how we deal with the machine-driver adding properties, so that there no longer is a need for calling the ugly private rt5651_apply_properties function from the machine-driver -Split the patch for moving the configuration of the jack-detect source to device-properties into separate codec and machine-driver patches -Split the patch making the OVCD threshold-current and scale-factor configurable into 2 separate patches Changes in v2; -Get rid of platform_data and use device-properties instead -Split up a couple of the micbias / OVCD / jack-detect patches into smaller patches, as they were doing too much in a single patch -Improved commit messages -Concentrate the knowledge that SSP0 is 16 bits on BYTCR in the fixup function and extract the sample-format from the hw_params in other places Regards, Hans

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[alsa-devel] [asoc:topic/twl6040 1/2] sound/soc/codecs/twl6040.c:1156:71-72: remove: first occurrence line 1158, second occurrence line 1160 (fwd)
by Julia Lawall 13 Mar '18

13 Mar '18

There are two initializations of the remove field. Both give it the same value, so one can just be dropped. julia ---------- Forwarded message ---------- Date: Tue, 13 Mar 2018 05:55:05 +0800 From: kbuild test robot <fengguang.wu(a)intel.com> To: kbuild(a)01.org Cc: Julia Lawall <julia.lawall(a)lip6.fr> Subject: [asoc:topic/twl6040 1/2] sound/soc/codecs/twl6040.c:1156:71-72: remove: first occurrence line 1158, second occurrence line 1160 CC: kbuild-all(a)01.org CC: alsa-devel(a)alsa-project.org TO: Kuninori Morimoto <kuninori.morimoto.gx(a)renesas.com> CC: Mark Brown <broonie(a)kernel.org> tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/twl6040 head: c9d066650cd5adf4ed4bf5734386699e765222b7 commit: 7480389fb0d873ed78619542bf5d2717a7ad7786 [1/2] ASoC: twl6040: replace codec to component :::::: branch date: 5 hours ago :::::: commit date: 4 weeks ago >> sound/soc/codecs/twl6040.c:1156:71-72: remove: first occurrence line 1158, second occurrence line 1160 # https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git/commit/?i… git remote add asoc https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git git remote update asoc git checkout 7480389fb0d873ed78619542bf5d2717a7ad7786 vim +1156 sound/soc/codecs/twl6040.c 8ecbabd9 Misael Lopez Cruz 2010-03-19 1155 7480389f Kuninori Morimoto 2018-01-29 @1156 static const struct snd_soc_component_driver soc_component_dev_twl6040 = { f0fba2ad Liam Girdwood 2010-03-17 1157 .probe = twl6040_probe, f0fba2ad Liam Girdwood 2010-03-17 @1158 .remove = twl6040_remove, 8146acff Tony Lindgren 2018-01-25 1159 .read = twl6040_read, 7480389f Kuninori Morimoto 2018-01-29 @1160 .remove = twl6040_remove, f0fba2ad Liam Girdwood 2010-03-17 1161 .set_bias_level = twl6040_set_bias_level, a175fce0 Peter Ujfalusi 2011-10-11 1162 .controls = twl6040_snd_controls, a175fce0 Peter Ujfalusi 2011-10-11 1163 .num_controls = ARRAY_SIZE(twl6040_snd_controls), a175fce0 Peter Ujfalusi 2011-10-11 1164 .dapm_widgets = twl6040_dapm_widgets, a175fce0 Peter Ujfalusi 2011-10-11 1165 .num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets), a175fce0 Peter Ujfalusi 2011-10-11 1166 .dapm_routes = intercon, a175fce0 Peter Ujfalusi 2011-10-11 1167 .num_dapm_routes = ARRAY_SIZE(intercon), 7480389f Kuninori Morimoto 2018-01-29 1168 .suspend_bias_off = 1, 7480389f Kuninori Morimoto 2018-01-29 1169 .idle_bias_on = 1, 7480389f Kuninori Morimoto 2018-01-29 1170 .endianness = 1, 7480389f Kuninori Morimoto 2018-01-29 1171 .non_legacy_dai_naming = 1, f0fba2ad Liam Girdwood 2010-03-17 1172 }; f0fba2ad Liam Girdwood 2010-03-17 1173 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

1 0

[alsa-devel] [PATCH] ASoC: sgtl5000: Pass the required '#sound-dai-cells'
by Fabio Estevam 12 Mar '18

12 Mar '18

From: Fabio Estevam <fabio.estevam(a)nxp.com> DTC now warns about missing #sound-dai-cells: arch/arm/boot/dts/imx6ul-geam.dtb: Warning (sound_dai_property): /sound/simple-audio-card,codec: Missing property '#sound-dai-cells' in node /soc/aips-bus@2100000/i2c@21a0000/codec@a or bad phandle (referred from sound-dai[0]) Pass the required '#sound-dai-cells' property to improve the documentation. Signed-off-by: Fabio Estevam <fabio.estevam(a)nxp.com> --- Documentation/devicetree/bindings/sound/sgtl5000.txt | 3 +++ 1 file changed, 3 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/sgtl5000.txt b/Documentation/devicetree/bindings/sound/sgtl5000.txt index 060cb4a..9a36c7e 100644 --- a/Documentation/devicetree/bindings/sound/sgtl5000.txt +++ b/Documentation/devicetree/bindings/sound/sgtl5000.txt @@ -5,6 +5,8 @@ Required properties: - reg : the I2C address of the device +- #sound-dai-cells: must be equal to 0 + - clocks : the clock provider of SYS_MCLK - VDDA-supply : the regulator provider of VDDA @@ -40,6 +42,7 @@ Example: codec: sgtl5000@a { compatible = "fsl,sgtl5000"; reg = <0x0a>; + #sound-dai-cells = <0>; clocks = <&clks 150>; micbias-resistor-k-ohms = <2>; micbias-voltage-m-volts = <2250>; -- 2.7.4

2 1

[alsa-devel] [PATCH 1/2] ASoC: rt5651: move definitions of dt-binding constants to include/dt-bindings
by Hans de Goede 12 Mar '18

12 Mar '18

Move the definitions of constants used in the dt-bindings from include/sound/rt5651.h to include/dt-bindings/sound/rt5651.h. As dt-bindings headers may also be parsed by the dt-compiler, they cannot use enums, only defines, so this commit also changes the code declaring the constants to use defines. Signed-off-by: Hans de Goede <hdegoede(a)redhat.com> --- include/dt-bindings/sound/rt5651.h | 15 +++++++++++++++ include/sound/rt5651.h | 36 ------------------------------------ sound/soc/codecs/rt5651.h | 4 ++-- 3 files changed, 17 insertions(+), 38 deletions(-) create mode 100644 include/dt-bindings/sound/rt5651.h delete mode 100644 include/sound/rt5651.h diff --git a/include/dt-bindings/sound/rt5651.h b/include/dt-bindings/sound/rt5651.h new file mode 100644 index 000000000000..2f2dac915168 --- /dev/null +++ b/include/dt-bindings/sound/rt5651.h @@ -0,0 +1,15 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +#ifndef __DT_RT5651_H +#define __DT_RT5651_H + +#define RT5651_JD_NULL 0 +#define RT5651_JD1_1 1 +#define RT5651_JD1_2 2 +#define RT5651_JD2 3 + +#define RT5651_OVCD_SF_0P5 0 +#define RT5651_OVCD_SF_0P75 1 +#define RT5651_OVCD_SF_1P0 2 +#define RT5651_OVCD_SF_1P5 3 + +#endif /* __DT_RT5651_H */ diff --git a/include/sound/rt5651.h b/include/sound/rt5651.h deleted file mode 100644 index 6403b862fb9a..000000000000 --- a/include/sound/rt5651.h +++ /dev/null @@ -1,36 +0,0 @@ -/* - * linux/sound/rt286.h -- Platform data for RT286 - * - * Copyright 2013 Realtek Microelectronics - * - * This program is free software; you can redistribute it and/or modify - * it under the terms of the GNU General Public License version 2 as - * published by the Free Software Foundation. - */ - -#ifndef __LINUX_SND_RT5651_H -#define __LINUX_SND_RT5651_H - -/* - * Note these MUST match the values from the DT binding: - * Documentation/devicetree/bindings/sound/rt5651.txt - */ -enum rt5651_jd_src { - RT5651_JD_NULL, - RT5651_JD1_1, - RT5651_JD1_2, - RT5651_JD2, -}; - -/* - * Note these MUST match the values from the DT binding: - * Documentation/devicetree/bindings/sound/rt5651.txt - */ -enum rt5651_ovcd_sf { - RT5651_OVCD_SF_0P5, - RT5651_OVCD_SF_0P75, - RT5651_OVCD_SF_1P0, - RT5651_OVCD_SF_1P5, -}; - -#endif diff --git a/sound/soc/codecs/rt5651.h b/sound/soc/codecs/rt5651.h index f20c9be94fb2..3a0968c53fde 100644 --- a/sound/soc/codecs/rt5651.h +++ b/sound/soc/codecs/rt5651.h @@ -12,7 +12,7 @@ #ifndef __RT5651_H__ #define __RT5651_H__ -#include <sound/rt5651.h> +#include <dt-bindings/sound/rt5651.h> /* Info */ #define RT5651_RESET 0x00 @@ -2073,7 +2073,7 @@ struct rt5651_priv { struct regmap *regmap; struct snd_soc_jack *hp_jack; struct work_struct jack_detect_work; - enum rt5651_jd_src jd_src; + unsigned int jd_src; unsigned int ovcd_th; unsigned int ovcd_sf; -- 2.14.3

2 3

[alsa-devel] [PATCH] ASoC: topology: create TLV data for dapm widgets
by Ranjani Sridharan 12 Mar '18

12 Mar '18

This patch adds the change required to create the TLV data for dapm widget kcontrols from topology. This also fixes the following TLV read error shown in amixer while showing the card control contents. "amixer: Control hw:1 element TLV read error: No such device or address" Signed-off-by: Ranjani Sridharan <ranjani.sridharan(a)linux.intel.com> --- 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 293124855cf6..dc6f06e853d0 100644 --- a/sound/soc/soc-topology.c +++ b/sound/soc/soc-topology.c @@ -1294,6 +1294,9 @@ static struct snd_kcontrol_new *soc_tplg_dapm_widget_dmixer_create( kfree(sm); continue; } + + /* create any TLV data */ + soc_tplg_create_tlv(tplg, &kc[i], &mc->hdr); } return kc; -- 2.14.1

2 1

[alsa-devel] [PATCH 2/6] ASoC: samsung: i2s: Fix rclk_srcrate handling
by Sylwester Nawrocki 12 Mar '18

12 Mar '18

As the RCLK clock may be updated through the common clk API before each snd_soc_dai_ops::trigger call, it is not enough to update i2s->rclk_srcrate only once after it has been initially set to 0. To avoid wrong PSR values we always get the RCLK frequency from the CLK_I2S_RCLK_SRC, when this clock is available. Fixes: e1417fdf3011 "ASoC: samsung: i2s: Ensure the RCLK rate is properly determined" Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/i2s.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/samsung/i2s.c b/sound/soc/samsung/i2s.c index b6407fbabdd1..f914ed45db7d 100644 --- a/sound/soc/samsung/i2s.c +++ b/sound/soc/samsung/i2s.c @@ -887,7 +887,7 @@ static int config_setup(struct i2s_dai *i2s) if (!(i2s->quirks & QUIRK_NO_MUXPSR)) { struct clk *rclksrc = i2s->clk_table[CLK_I2S_RCLK_SRC]; - if (i2s->rclk_srcrate == 0 && rclksrc && !IS_ERR(rclksrc)) + if (rclksrc && !IS_ERR(rclksrc)) i2s->rclk_srcrate = clk_get_rate(rclksrc); psr = i2s->rclk_srcrate / i2s->frmclk / rfs; -- 2.14.2

2 1

[alsa-devel] [PATCH 5/6] ASoC: samsung: Drop uneeded RCLKSRC setting in the Snow driver
by Sylwester Nawrocki 12 Mar '18

12 Mar '18

The SAMSUNG_I2S_RCLKSRC_0 RCLKSRC mux is a default configuration after reset so there is no need for this explicit snd_soc_dai_set_sysclk() call. Also, this static mux clock configuration can be specified in DT. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/snow.c | 5 ----- 1 file changed, 5 deletions(-) diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c index d8ac907bbb0d..fc62110f500f 100644 --- a/sound/soc/samsung/snow.c +++ b/sound/soc/samsung/snow.c @@ -50,11 +50,6 @@ static int snow_late_probe(struct snd_soc_card *card) if (ret < 0) return ret; - /* Select I2S Bus clock to set RCLK and BCLK */ - ret = snd_soc_dai_set_sysclk(cpu_dai, SAMSUNG_I2S_RCLKSRC_0, - 0, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; return 0; } -- 2.14.2

2 1

[alsa-devel] [PATCH v2 1/7] ASoC: max98090: Add #sound-dai-cells property documentation
by Sylwester Nawrocki 12 Mar '18

12 Mar '18

MAX98090 CODEC supports 1 input DAI, adding the #sound-dai-cells property helps to reference the DAI in DT in a standard way. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- Changes since v1: - new patch. Documentation/devicetree/bindings/sound/max98090.txt | 2 ++ 1 file changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/max98090.txt b/Documentation/devicetree/bindings/sound/max98090.txt index 4e3be6682c98..7e1bbd5c27fd 100644 --- a/Documentation/devicetree/bindings/sound/max98090.txt +++ b/Documentation/devicetree/bindings/sound/max98090.txt @@ -16,6 +16,8 @@ Optional properties: - clock-names: Should be "mclk" +- #sound-dai-cells : should be 0. + - maxim,dmic-freq: Frequency at which to clock DMIC - maxim,micbias: Micbias voltage applies to the analog mic, valid voltages value are: -- 2.14.2

3 11

[alsa-devel] Applied "ASoC: Add snd_soc_of_put_dai_link_codecs() helper function" to the asoc tree
by Mark Brown 12 Mar '18

12 Mar '18

The patch ASoC: Add snd_soc_of_put_dai_link_codecs() helper function 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 946857636d279486eaa47c930930185631f28fce Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Date: Fri, 9 Mar 2018 18:48:54 +0100 Subject: [PATCH] ASoC: Add snd_soc_of_put_dai_link_codecs() helper function The code for dereferencing device nodes in the 'codecs' array is moved to a separate function so we can avoid open coding that in drivers. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- include/sound/soc.h | 1 + sound/soc/soc-core.c | 32 +++++++++++++++++++++++--------- 2 files changed, 24 insertions(+), 9 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 747fd583b9dc..4a387f0b3d56 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1807,6 +1807,7 @@ int snd_soc_of_get_dai_name(struct device_node *of_node, int snd_soc_of_get_dai_link_codecs(struct device *dev, struct device_node *of_node, struct snd_soc_dai_link *dai_link); +void snd_soc_of_put_dai_link_codecs(struct snd_soc_dai_link *dai_link); int snd_soc_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 96c44f6576c9..a1f86e85bff9 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4397,6 +4397,26 @@ int snd_soc_of_get_dai_name(struct device_node *of_node, } EXPORT_SYMBOL_GPL(snd_soc_of_get_dai_name); +/* + * snd_soc_of_put_dai_link_codecs - Dereference device nodes in the codecs array + * @dai_link: DAI link + * + * Dereference device nodes acquired by snd_soc_of_get_dai_link_codecs(). + */ +void snd_soc_of_put_dai_link_codecs(struct snd_soc_dai_link *dai_link) +{ + struct snd_soc_dai_link_component *component = dai_link->codecs; + int index; + + for (index = 0; index < dai_link->num_codecs; index++, component++) { + if (!component->of_node) + break; + of_node_put(component->of_node); + component->of_node = NULL; + } +} +EXPORT_SYMBOL_GPL(snd_soc_of_put_dai_link_codecs); + /* * snd_soc_of_get_dai_link_codecs - Parse a list of CODECs in the devicetree * @dev: Card device @@ -4406,7 +4426,8 @@ EXPORT_SYMBOL_GPL(snd_soc_of_get_dai_name); * Builds an array of CODEC DAI components from the DAI link property * 'sound-dai'. * The array is set in the DAI link and the number of DAIs is set accordingly. - * The device nodes in the array (of_node) must be dereferenced by the caller. + * The device nodes in the array (of_node) must be dereferenced by calling + * snd_soc_of_put_dai_link_codecs() on @dai_link. * * Returns 0 for success */ @@ -4454,14 +4475,7 @@ int snd_soc_of_get_dai_link_codecs(struct device *dev, } return 0; err: - for (index = 0, component = dai_link->codecs; - index < dai_link->num_codecs; - index++, component++) { - if (!component->of_node) - break; - of_node_put(component->of_node); - component->of_node = NULL; - } + snd_soc_of_put_dai_link_codecs(dai_link); dai_link->codecs = NULL; dai_link->num_codecs = 0; return ret; -- 2.16.2

1 0

[alsa-devel] [PATCH 0/2] ASoC: soc-io, twl6040: Fix regression in linux-next
by Peter Ujfalusi 12 Mar '18

12 Mar '18

Hi, I just noticed in linux-next that after the codec -> component conversion the twl6040 codec is not working (and most likely the twl4030 neither). I have fixed it with the two patch in the series. Regards, Peter --- Peter Ujfalusi (2): ASoC: twl6040: Add back missing write callback ASoC: soc-io: Fix snd_soc_component_update_bits_legacy sound/soc/codecs/twl6040.c | 1 + sound/soc/soc-io.c | 7 ++----- 2 files changed, 3 insertions(+), 5 deletions(-) -- Peter Texas Instruments Finland Oy, Porkkalankatu 22, 00180 Helsinki. Y-tunnus/Business ID: 0615521-4. Kotipaikka/Domicile: Helsinki

2 6

[alsa-devel] [PATCH 1/3] ASoC: rt5659: fix wrong bit define for ADC L2 power.
by Bard Liao 12 Mar '18

12 Mar '18

RT5659_PWR_ADC_L2_BIT should be 2. Signed-off-by: Zhong An <zhongan(a)pinecone.net> Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5659.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/rt5659.h b/sound/soc/codecs/rt5659.h index 73969ef..3b51c8a 100644 --- a/sound/soc/codecs/rt5659.h +++ b/sound/soc/codecs/rt5659.h @@ -1008,7 +1008,7 @@ #define RT5659_PWR_ADC_R1 (0x1 << 3) #define RT5659_PWR_ADC_R1_BIT 3 #define RT5659_PWR_ADC_L2 (0x1 << 2) -#define RT5659_PWR_ADC_L2_BIT 4 +#define RT5659_PWR_ADC_L2_BIT 2 #define RT5659_PWR_ADC_R2 (0x1 << 1) #define RT5659_PWR_ADC_R2_BIT 1 #define RT5659_PWR_CLS_D (0x1) -- 2.7.4

2 5

[alsa-devel] Applied "ASoC: max98090: Add #sound-dai-cells property documentation" to the asoc tree
by Mark Brown 12 Mar '18

12 Mar '18

The patch ASoC: max98090: Add #sound-dai-cells property documentation 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 b50c32bd28e399cd4672a1e43645399e5bd1eb85 Mon Sep 17 00:00:00 2001 From: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Date: Fri, 9 Mar 2018 18:48:52 +0100 Subject: [PATCH] ASoC: max98090: Add #sound-dai-cells property documentation MAX98090 CODEC supports 1 input DAI, adding the #sound-dai-cells property helps to reference the DAI in DT in a standard way. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- Documentation/devicetree/bindings/sound/max98090.txt | 2 ++ 1 file changed, 2 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/max98090.txt b/Documentation/devicetree/bindings/sound/max98090.txt index 4e3be6682c98..7e1bbd5c27fd 100644 --- a/Documentation/devicetree/bindings/sound/max98090.txt +++ b/Documentation/devicetree/bindings/sound/max98090.txt @@ -16,6 +16,8 @@ Optional properties: - clock-names: Should be "mclk" +- #sound-dai-cells : should be 0. + - maxim,dmic-freq: Frequency at which to clock DMIC - maxim,micbias: Micbias voltage applies to the analog mic, valid voltages value are: -- 2.16.2

1 0

[alsa-devel] [PATCH] ASoC: mt8173-rt5650: fix child-node lookup
by Johan Hovold 12 Mar '18

12 Mar '18

This driver used the wrong OF-helper when looking up the optional capture-codec child node during probe. Instead of searching just children of the sound node, a tree-wide depth-first search starting at the unrelated platform node was done. Not only could this end up matching an unrelated node or no node at all; the platform node could also be prematurely freed since of_find_node_by_name() drops a reference to its first argument. This particular pattern has been observed leading to crashes after probe deferrals in other drivers. Fix this by dropping the broken call to of_find_node_by_name() and keeping only the second, correct lookup using of_get_child_by_name() while taking care not to bail out if the optional node is missing. Note that this also addresses two capture-codec node-reference leaks (one for each of the original helper calls). Compile tested only. Fixes: d349caeb0510 ("ASoC: mediatek: Add second I2S on mt8173-rt5650 machine driver") Cc: PC Liao <pc.liao(a)mediatek.com> Signed-off-by: Johan Hovold <johan(a)kernel.org> --- sound/soc/mediatek/mt8173/mt8173-rt5650.c | 11 +++-------- 1 file changed, 3 insertions(+), 8 deletions(-) diff --git a/sound/soc/mediatek/mt8173/mt8173-rt5650.c b/sound/soc/mediatek/mt8173/mt8173-rt5650.c index 40ebefd625c1..679fc8bea0a3 100644 --- a/sound/soc/mediatek/mt8173/mt8173-rt5650.c +++ b/sound/soc/mediatek/mt8173/mt8173-rt5650.c @@ -272,15 +272,10 @@ static int mt8173_rt5650_dev_probe(struct platform_device *pdev) } mt8173_rt5650_codecs[1].of_node = mt8173_rt5650_codecs[0].of_node; - if (of_find_node_by_name(platform_node, "codec-capture")) { - np = of_get_child_by_name(pdev->dev.of_node, "codec-capture"); - if (!np) { - dev_err(&pdev->dev, - "%s: Can't find codec-capture DT node\n", - __func__); - return -EINVAL; - } + np = of_get_child_by_name(pdev->dev.of_node, "codec-capture"); + if (np) { ret = snd_soc_of_get_dai_name(np, &codec_capture_dai); + of_node_put(np); if (ret < 0) { dev_err(&pdev->dev, "%s codec_capture_dai name fail %d\n", -- 2.13.6

2 3

[alsa-devel] [GIT PULL] ASoC fixes for v4.16
by Mark Brown 12 Mar '18

12 Mar '18

The following changes since commit 0c8efd610b58cb23cefdfa12015799079aef94ae: Linux 4.16-rc5 (2018-03-11 17:25:09 -0700) are available in the Git repository at: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git tags/asoc-fix-v4.16-rc5 for you to fetch changes up to a3e39ed1f40e14f24bec0f91c4fa0408fb6c4d02: Merge remote-tracking branches 'asoc/fix/amd', 'asoc/fix/hdmi-codec', 'asoc/fix/rt5651', 'asoc/fix/samsung', 'asoc/fix/sgtl5000', 'asoc/fix/sunxi' and 'asoc/fix/wm-adsp' into asoc-linus (2018-03-12 09:14:07 -0700) ---------------------------------------------------------------- ASoC: Fixes for v4.16 This is a fairly standard collection of fixes, there's no changes to the core here just a bunch of small device specific changes for single drivers plus an update to the MAINTAINERS file for the sgl5000. ---------------------------------------------------------------- Fabio Estevam (2): MAINTAINERS: Add myself as sgtl5000 maintainer ASoC: sgtl5000: Fix suspend/resume Hans de Goede (1): ASoC: rt5651: Fix regcache sync errors on resume Mark Brown (1): Merge remote-tracking branches 'asoc/fix/amd', 'asoc/fix/hdmi-codec', 'asoc/fix/rt5651', 'asoc/fix/samsung', 'asoc/fix/sgtl5000', 'asoc/fix/sunxi' and 'asoc/fix/wm-adsp' into asoc-linus Michal Oleszczyk (1): sgtl5000: change digital_mute policy Peter Ujfalusi (1): ASoC: hdmi-codec: Fix module unloading caused kernel crash Richard Fitzgerald (1): ASoC: wm_adsp: For TLV controls only register TLV get/set Sylwester Nawrocki (1): ASoC: samsung: Add the DT binding files entry to MAINTAINERS Vijendar Mukunda (1): ASoC: amd: 16bit resolution support for i2s sp instance Yong Deng (1): ASoC: sun4i-i2s: Fix RX slot number of SUN8I MAINTAINERS | 8 ++++++++ sound/soc/amd/acp-pcm-dma.c | 16 +++++++++------- sound/soc/amd/acp.h | 2 ++ sound/soc/codecs/hdmi-codec.c | 7 +------ sound/soc/codecs/rt5651.c | 1 + sound/soc/codecs/sgtl5000.c | 26 +++++++++++++++++++++++--- sound/soc/codecs/wm_adsp.c | 14 ++++++++------ sound/soc/sunxi/sun4i-i2s.c | 2 +- 8 files changed, 53 insertions(+), 23 deletions(-)

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[alsa-devel] [RESEND PATCH 0/2] ASoC: Add support for max9759 Amplifier Driver
by Neil Armstrong 12 Mar '18

12 Mar '18

The MAX9759 is a gpio controlled speaker amplifier, this patchset add the DT Bindings and the corresponding ASoC component driver. Changes with resend : - Add rob ack on bindings Neil Armstrong (2): ASoC: Add DT bindings documentation for max9759 amplifier ASoC: max9759: Add Amplifier Driver .../devicetree/bindings/sound/maxim,max9759.txt | 18 ++ sound/soc/codecs/Kconfig | 5 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/max9759.c | 207 +++++++++++++++++++++ 4 files changed, 232 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/maxim,max9759.txt create mode 100644 sound/soc/codecs/max9759.c -- 2.7.4

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[alsa-devel] [PATCH] ASoC: soc-core: Add missing NULL check
by Kees Cook 12 Mar '18

12 Mar '18

If a codec is not attached to the sound soc, a NULL deref is possible as a regular user in /sys. [ 2278.331878] DSS: context saved [ 2278.820343] Unable to handle kernel NULL pointer dereference at virtual address 00000004 [ 2278.828948] pgd = c36040a2 [ 2278.831787] [00000004] *pgd=876c4831, *pte=00000000, *ppte=00000000 [ 2278.838439] Internal error: Oops: 17 [#1] ARM [ 2278.843017] Modules linked in: [ 2278.846221] CPU: 0 PID: 16337 Comm: grep Tainted: G W 4.16.0-rc4-next-20180308 #71 [ 2278.855529] Hardware name: Nokia RX-51 board [ 2278.860015] PC is at soc_codec_reg_show+0x8/0x19c [ 2278.864959] LR is at codec_reg_show+0x28/0x30 Reported-by: Pavel Machek <pavel(a)ucw.cz> Signed-off-by: Kees Cook <keescook(a)chromium.org> --- No idea if this is the _right_ fix, but it should stop the crash from unprivileged userspace. --- sound/soc/soc-core.c | 3 +++ 1 file changed, 3 insertions(+) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 96c44f6576c9..78ad165ad424 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -137,6 +137,9 @@ static ssize_t soc_codec_reg_show(struct snd_soc_codec *codec, char *buf, size_t total = 0; loff_t p = 0; + if (!codec || !codec->driver) + return 0; + wordsize = min_bytes_needed(codec->driver->reg_cache_size) * 2; regsize = codec->driver->reg_word_size * 2; -- 2.7.4 -- Kees Cook Pixel Security

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[alsa-devel] [PATCH] echoaudio: remove redundant initialization of pointer 'pipe'
by Colin King 12 Mar '18

12 Mar '18

From: Colin Ian King <colin.king(a)canonical.com> The pointer 'pipe' is being initialized with a value that is never read and it is re-assigned later, hence the initialization is redundant and can be removed. Also remove pointer 'runtime' as it is no longer required. Cleans up clang warning: sound/pci/echoaudio/echoaudio.c:740:20: warning: Value stored to 'pipe' during its initialization is never read Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/pci/echoaudio/echoaudio.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) diff --git a/sound/pci/echoaudio/echoaudio.c b/sound/pci/echoaudio/echoaudio.c index d68f99e076a8..0935a5c8741f 100644 --- a/sound/pci/echoaudio/echoaudio.c +++ b/sound/pci/echoaudio/echoaudio.c @@ -736,8 +736,7 @@ static int pcm_prepare(struct snd_pcm_substream *substream) static int pcm_trigger(struct snd_pcm_substream *substream, int cmd) { struct echoaudio *chip = snd_pcm_substream_chip(substream); - struct snd_pcm_runtime *runtime = substream->runtime; - struct audiopipe *pipe = runtime->private_data; + struct audiopipe *pipe; int i, err; u32 channelmask = 0; struct snd_pcm_substream *s; -- 2.15.1

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[alsa-devel] [PATCH] ALSA: hda - Revert power_save option default value
by Takashi Iwai 12 Mar '18

12 Mar '18

With the commit 1ba8f9d30817 ("ALSA: hda: Add a power_save blacklist"), we changed the default value of power_save option to -1 for processing the power-save blacklist. Unfortunately, this seems breaking user-space applications that actually read the power_save parameter value via sysfs and judge / adjust the power-saving status. They see the value -1 as if the power-save is turned off, although the actual value is taken from CONFIG_SND_HDA_POWER_SAVE_DEFAULT and it can be a positive. So, overall, passing -1 there was no good idea. Let's partially revert it -- at least for power_save option default value is restored again to CONFIG_SND_HDA_POWER_SAVE_DEFAULT. Meanwhile, in this patch, we keep the blacklist behavior and make is adjustable via the new option, pm_blacklist. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=199073 Fixes: 1ba8f9d30817 ("ALSA: hda: Add a power_save blacklist") Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/hda_intel.c | 9 ++++++--- 1 file changed, 6 insertions(+), 3 deletions(-) diff --git a/sound/pci/hda/hda_intel.c b/sound/pci/hda/hda_intel.c index 96143df19b21..d5017adf9feb 100644 --- a/sound/pci/hda/hda_intel.c +++ b/sound/pci/hda/hda_intel.c @@ -181,11 +181,15 @@ static const struct kernel_param_ops param_ops_xint = { }; #define param_check_xint param_check_int -static int power_save = -1; +static int power_save = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; module_param(power_save, xint, 0644); MODULE_PARM_DESC(power_save, "Automatic power-saving timeout " "(in second, 0 = disable)."); +static bool pm_blacklist = true; +module_param(pm_blacklist, bool, 0644); +MODULE_PARM_DESC(pm_blacklist, "Enable power-management blacklist"); + /* reset the HD-audio controller in power save mode. * this may give more power-saving, but will take longer time to * wake up. @@ -2300,10 +2304,9 @@ static int azx_probe_continue(struct azx *chip) val = power_save; #ifdef CONFIG_PM - if (val == -1) { + if (pm_blacklist) { const struct snd_pci_quirk *q; - val = CONFIG_SND_HDA_POWER_SAVE_DEFAULT; q = snd_pci_quirk_lookup(chip->pci, power_save_blacklist); if (q && val) { dev_info(chip->card->dev, "device %04x:%04x is on the power_save blacklist, forcing power_save to 0\n", -- 2.16.2

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Re: [alsa-devel] [RFC] hda/realtek filtered headphone output
by Takashi Iwai 12 Mar '18

12 Mar '18

Resending with the correct Cc to alsa-devel ML... On Mon, 05 Mar 2018 12:55:30 +0100, Takashi Iwai wrote: > > On Sat, 03 Mar 2018 13:06:49 +0100, > René Rebe wrote: > > > > Hi all, > > > > testing some new laptops, I noticed the headphone frequency response was lacking > > bass. First I noticed this in the Lenovo T470s, and later I found this also be the case > > on Surface Pro 2 and 3. The Dell XPS13 may also be affected, but I had to return it > > before I found out the solution on the other machines: > > > > https://www.youtube.com/watch?v=a1YkPtfC4LI > > > > I had not yet sent the patch because a) I was too busy and b) I wanted to review this further. > > > > As the other day I got a Lenovo ideapad 720s to test with exactly the same issue, here is the > > current hacked patch: > > > > https://svn.exactcode.de/t2/trunk/package/base/linux/surface-audio.patch > > > > However, given that all newer realtek codecs that came on my desk are affected, > > maybe there is something wrong with the hda_auto_parser.c - I did not yet fully grasp > > all the logic of that. > > > > As you can see in the patch and video (…) the problem is the signal routing of NID 0x15 > > or 0x21 (depending whether it is an older, or newer codec). This is the headphone out, > > and as far as I can see the auto_parser assigned the index 0 pcm output (NID 0x0c) > > which result in this filtered frequency response, likely intended for the laptop’s speakers? > > Using the next NID (I guess that was 0xd?) results in a full frequency response and at > > least to my ears is immediately hearable and obvious. > > > > Without this fixed routing the headphone sounds like a PC speaker in a coke can. > > > > So the question for the authors and HDA experts: Can this be fixed in the auto_parser, > > or is my patch (which btw. has to small unrelated cleanups) modulo potentially some > > renaming of the enum and function, now that more than the Surface is affected, the > > way to go? If this can be fixed more generally I would be happy to hear! > > Could you give the output of alsa-info.sh? We need to check the > details of the codec topology. > > In general, the right approach is to fix at parsing the routing, not > fiddling with the codec routing dynamically at initialization. > > Currently the easy fix would be to provide the fixed DAC connection > list. Maybe we can enhance the generic parser to receive some hints > for DAC selection. The routes are determined by calculating "badness" > values per route, so in theory, we can give some badness values > depending on the target pin type (headphone, speaker, etc). > > > thanks, > > Takashi

1 0

[alsa-devel] [PATCH 0/3] ASoC: Add Multi CPU DAI support
by Shreyas NC 12 Mar '18

12 Mar '18

As discussed in [1], ASoC core supports multi codec DAIs on a DAI link. However it does not do so for CPU DAIs. So, add support for multi CPU DAIs on a DAI Link by adding multi CPU DAI in Card instantiation, suspend and resume functions, PCM ops, stream handling functions and DAPM. [1]: https://www.spinics.net/lists/alsa-devel/msg71369.html Shreyas NC (3): ASoC: Add initial support for multiple CPU DAIs ASoC: Add Multi CPU DAI support for PCM ops ASoC: Add Multi CPU DAI support in DAPM include/sound/soc.h | 5 + sound/soc/soc-core.c | 195 ++++++++++++++++------ sound/soc/soc-dapm.c | 71 +++++--- sound/soc/soc-pcm.c | 460 +++++++++++++++++++++++++++++++++------------------ 4 files changed, 495 insertions(+), 236 deletions(-) -- 2.7.4

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[alsa-devel] [PATCH] ALSA: pcm: Fix UAF in snd_pcm_oss_get_formats()
by Takashi Iwai 11 Mar '18

11 Mar '18

snd_pcm_oss_get_formats() has an obvious UAF around snd_mask_test() calls, as spotted by syzbot. The passed format_mask argument is a pointer to the hw_params object that is freed before the loop. What a surprise that it has been present since the original code of decades ago... Reported-by: syzbot+4090700a4f13fccaf648(a)syzkaller.appspotmail.com Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/oss/pcm_oss.c | 10 ++++++---- 1 file changed, 6 insertions(+), 4 deletions(-) diff --git a/sound/core/oss/pcm_oss.c b/sound/core/oss/pcm_oss.c index b044c0a5a674..02298c9c6020 100644 --- a/sound/core/oss/pcm_oss.c +++ b/sound/core/oss/pcm_oss.c @@ -1762,10 +1762,9 @@ static int snd_pcm_oss_get_formats(struct snd_pcm_oss_file *pcm_oss_file) return -ENOMEM; _snd_pcm_hw_params_any(params); err = snd_pcm_hw_refine(substream, params); - format_mask = hw_param_mask_c(params, SNDRV_PCM_HW_PARAM_FORMAT); - kfree(params); if (err < 0) - return err; + goto error; + format_mask = hw_param_mask_c(params, SNDRV_PCM_HW_PARAM_FORMAT); for (fmt = 0; fmt < 32; ++fmt) { if (snd_mask_test(format_mask, fmt)) { int f = snd_pcm_oss_format_to(fmt); @@ -1773,7 +1772,10 @@ static int snd_pcm_oss_get_formats(struct snd_pcm_oss_file *pcm_oss_file) formats |= f; } } - return formats; + + error: + kfree(params); + return err < 0 ? err : formats; } static int snd_pcm_oss_set_format(struct snd_pcm_oss_file *pcm_oss_file, int format) -- 2.16.2

1 0

[alsa-devel] KASAN: use-after-free Read in snd_pcm_oss_get_formats
by syzbot 11 Mar '18

11 Mar '18

Hello, syzbot hit the following crash on upstream commit 1b88accf6a659c46d5c8e68912896f112bf882bb (Thu Mar 8 01:49:33 2018 +0000) Merge tag 'for_linus' of git://git.kernel.org/pub/scm/linux/kernel/git/mst/vhost So far this crash happened 3 times on upstream. C reproducer is attached. syzkaller reproducer is attached. Raw console output is attached. compiler: gcc (GCC) 7.1.1 20170620 .config is attached. IMPORTANT: if you fix the bug, please add the following tag to the commit: Reported-by: syzbot+4090700a4f13fccaf648(a)syzkaller.appspotmail.com It will help syzbot understand when the bug is fixed. See footer for details. If you forward the report, please keep this part and the footer. IPVS: ftp: loaded support on port[0] = 21 audit: type=1400 audit(1520612143.836:8): avc: denied { map } for pid=4259 comm="syzkaller191035" path="/dev/dsp1" dev="devtmpfs" ino=1250 scontext=unconfined_u:system_r:insmod_t:s0-s0:c0.c1023 tcontext=system_u:object_r:sound_device_t:s0 tclass=chr_file permissive=1 IPVS: ftp: loaded support on port[0] = 21 IPVS: ftp: loaded support on port[0] = 21 ================================================================== BUG: KASAN: use-after-free in snd_mask_test include/sound/pcm_params.h:141 [inline] BUG: KASAN: use-after-free in snd_pcm_oss_get_formats+0x320/0x380 sound/core/oss/pcm_oss.c:1770 Read of size 4 at addr ffff8801bace4024 by task syzkaller191035/4271 CPU: 0 PID: 4271 Comm: syzkaller191035 Not tainted 4.16.0-rc4+ #346 IPVS: ftp: loaded support on port[0] = 21 Hardware name: Google Google Compute Engine/Google Compute Engine, BIOS Google 01/01/2011 Call Trace: __dump_stack lib/dump_stack.c:17 [inline] dump_stack+0x194/0x24d lib/dump_stack.c:53 IPVS: ftp: loaded support on port[0] = 21 print_address_description+0x73/0x250 mm/kasan/report.c:256 kasan_report_error mm/kasan/report.c:354 [inline] kasan_report+0x23c/0x360 mm/kasan/report.c:412 __asan_report_load4_noabort+0x14/0x20 mm/kasan/report.c:432 snd_mask_test include/sound/pcm_params.h:141 [inline] snd_pcm_oss_get_formats+0x320/0x380 sound/core/oss/pcm_oss.c:1770 snd_pcm_oss_set_format sound/core/oss/pcm_oss.c:1784 [inline] snd_pcm_oss_ioctl+0x1e2f/0x39f0 sound/core/oss/pcm_oss.c:2515 IPVS: ftp: loaded support on port[0] = 21 IPVS: ftp: loaded support on port[0] = 21 vfs_ioctl fs/ioctl.c:46 [inline] do_vfs_ioctl+0x1b1/0x1520 fs/ioctl.c:686 SYSC_ioctl fs/ioctl.c:701 [inline] SyS_ioctl+0x8f/0xc0 fs/ioctl.c:692 do_syscall_64+0x281/0x940 arch/x86/entry/common.c:287 entry_SYSCALL_64_after_hwframe+0x42/0xb7 RIP: 0033:0x4499e9 RSP: 002b:00007fa1872a5da8 EFLAGS: 00000246 ORIG_RAX: 0000000000000010 RAX: ffffffffffffffda RBX: 00000000006dbc24 RCX: 00000000004499e9 RDX: 0000000020000080 RSI: 00800000c0045005 RDI: 0000000000000003 RBP: 0000000000000000 R08: 0000000000000000 R09: 0000000000000000 R10: 0000000000000000 R11: 0000000000000246 R12: 00000000006dbc20 R13: 7073642f7665642f R14: 00800000c0045005 R15: 0000000000000003 Allocated by task 4271: save_stack+0x43/0xd0 mm/kasan/kasan.c:447 set_track mm/kasan/kasan.c:459 [inline] kasan_kmalloc+0xad/0xe0 mm/kasan/kasan.c:552 kmem_cache_alloc_trace+0x136/0x740 mm/slab.c:3607 kmalloc include/linux/slab.h:512 [inline] snd_pcm_oss_get_formats+0x1c7/0x380 sound/core/oss/pcm_oss.c:1760 snd_pcm_oss_set_format sound/core/oss/pcm_oss.c:1784 [inline] snd_pcm_oss_ioctl+0x1e2f/0x39f0 sound/core/oss/pcm_oss.c:2515 vfs_ioctl fs/ioctl.c:46 [inline] do_vfs_ioctl+0x1b1/0x1520 fs/ioctl.c:686 SYSC_ioctl fs/ioctl.c:701 [inline] SyS_ioctl+0x8f/0xc0 fs/ioctl.c:692 do_syscall_64+0x281/0x940 arch/x86/entry/common.c:287 entry_SYSCALL_64_after_hwframe+0x42/0xb7 Freed by task 4271: save_stack+0x43/0xd0 mm/kasan/kasan.c:447 set_track mm/kasan/kasan.c:459 [inline] __kasan_slab_free+0x11a/0x170 mm/kasan/kasan.c:520 kasan_slab_free+0xe/0x10 mm/kasan/kasan.c:527 __cache_free mm/slab.c:3485 [inline] kfree+0xd9/0x260 mm/slab.c:3800 snd_pcm_oss_get_formats+0x216/0x380 sound/core/oss/pcm_oss.c:1766 snd_pcm_oss_set_format sound/core/oss/pcm_oss.c:1784 [inline] snd_pcm_oss_ioctl+0x1e2f/0x39f0 sound/core/oss/pcm_oss.c:2515 vfs_ioctl fs/ioctl.c:46 [inline] do_vfs_ioctl+0x1b1/0x1520 fs/ioctl.c:686 SYSC_ioctl fs/ioctl.c:701 [inline] SyS_ioctl+0x8f/0xc0 fs/ioctl.c:692 do_syscall_64+0x281/0x940 arch/x86/entry/common.c:287 entry_SYSCALL_64_after_hwframe+0x42/0xb7 The buggy address belongs to the object at ffff8801bace4000 which belongs to the cache kmalloc-1024 of size 1024 The buggy address is located 36 bytes inside of 1024-byte region [ffff8801bace4000, ffff8801bace4400) The buggy address belongs to the page: page:ffffea0006eb3900 count:1 mapcount:0 mapping:ffff8801bace4000 index:0x0 compound_mapcount: 0 flags: 0x2fffc0000008100(slab|head) raw: 02fffc0000008100 ffff8801bace4000 0000000000000000 0000000100000007 raw: ffffea000761d7a0 ffffea0006eb39a0 ffff8801dac00ac0 0000000000000000 page dumped because: kasan: bad access detected Memory state around the buggy address: ffff8801bace3f00: fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc ffff8801bace3f80: fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc fc > ffff8801bace4000: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ^ ffff8801bace4080: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ffff8801bace4100: fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb fb ================================================================== --- This bug is generated by a dumb bot. It may contain errors. See https://goo.gl/tpsmEJ for details. Direct all questions to syzkaller(a)googlegroups.com. syzbot will keep track of this bug report. If you forgot to add the Reported-by tag, once the fix for this bug is merged into any tree, please reply to this email with: #syz fix: exact-commit-title If you want to test a patch for this bug, please reply with: #syz test: git://repo/address.git branch and provide the patch inline or as an attachment. To mark this as a duplicate of another syzbot report, please reply with: #syz dup: exact-subject-of-another-report If it's a one-off invalid bug report, please reply with: #syz invalid Note: if the crash happens again, it will cause creation of a new bug report. Note: all commands must start from beginning of the line in the email body.

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[alsa-devel] [PATCH 1/2] ALSA: seq: Fix possible UAF in snd_seq_check_queue()
by Takashi Iwai 10 Mar '18

10 Mar '18

Although we've covered the races between concurrent write() and ioctl() in the previous patch series, there is still a possible UAF in the following scenario: A: user client closed B: timer irq -> snd_seq_release() -> snd_seq_timer_interrupt() -> snd_seq_free_client() -> snd_seq_check_queue() -> cell = snd_seq_prioq_cell_peek() -> snd_seq_prioq_leave() .... removing all cells -> snd_seq_pool_done() .... vfree() -> snd_seq_compare_tick_time(cell) ... Oops So the problem is that a cell is peeked and accessed without any protection until it's retrieved from the queue again via snd_seq_prioq_cell_out(). This patch tries to address it, also cleans up the code by a slight refactoring. snd_seq_prioq_cell_out() now receives an extra pointer argument. When it's non-NULL, the function checks the event timestamp with the given pointer. The caller needs to pass the right reference either to snd_seq_tick or snd_seq_realtime depending on the event timestamp type. A good news is that the above change allows us to remove the snd_seq_prioq_cell_peek(), too, thus the patch actually reduces the code size. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/seq_prioq.c | 28 ++++++++++++++-------------- sound/core/seq/seq_prioq.h | 6 ++---- sound/core/seq/seq_queue.c | 28 +++++++++------------------- 3 files changed, 25 insertions(+), 37 deletions(-) diff --git a/sound/core/seq/seq_prioq.c b/sound/core/seq/seq_prioq.c index bc1c8488fc2a..2bc6759e4adc 100644 --- a/sound/core/seq/seq_prioq.c +++ b/sound/core/seq/seq_prioq.c @@ -87,7 +87,7 @@ void snd_seq_prioq_delete(struct snd_seq_prioq **fifo) if (f->cells > 0) { /* drain prioQ */ while (f->cells > 0) - snd_seq_cell_free(snd_seq_prioq_cell_out(f)); + snd_seq_cell_free(snd_seq_prioq_cell_out(f, NULL)); } kfree(f); @@ -214,8 +214,18 @@ int snd_seq_prioq_cell_in(struct snd_seq_prioq * f, return 0; } +/* return 1 if the current time >= event timestamp */ +static int event_is_ready(struct snd_seq_event *ev, void *current_time) +{ + if ((ev->flags & SNDRV_SEQ_TIME_STAMP_MASK) == SNDRV_SEQ_TIME_STAMP_TICK) + return snd_seq_compare_tick_time(current_time, &ev->time.tick); + else + return snd_seq_compare_real_time(current_time, &ev->time.time); +} + /* dequeue cell from prioq */ -struct snd_seq_event_cell *snd_seq_prioq_cell_out(struct snd_seq_prioq *f) +struct snd_seq_event_cell *snd_seq_prioq_cell_out(struct snd_seq_prioq *f, + void *current_time) { struct snd_seq_event_cell *cell; unsigned long flags; @@ -227,6 +237,8 @@ struct snd_seq_event_cell *snd_seq_prioq_cell_out(struct snd_seq_prioq *f) spin_lock_irqsave(&f->lock, flags); cell = f->head; + if (cell && current_time && !event_is_ready(&cell->event, current_time)) + cell = NULL; if (cell) { f->head = cell->next; @@ -252,18 +264,6 @@ int snd_seq_prioq_avail(struct snd_seq_prioq * f) return f->cells; } - -/* peek at cell at the head of the prioq */ -struct snd_seq_event_cell *snd_seq_prioq_cell_peek(struct snd_seq_prioq * f) -{ - if (f == NULL) { - pr_debug("ALSA: seq: snd_seq_prioq_cell_in() called with NULL prioq\n"); - return NULL; - } - return f->head; -} - - static inline int prioq_match(struct snd_seq_event_cell *cell, int client, int timestamp) { diff --git a/sound/core/seq/seq_prioq.h b/sound/core/seq/seq_prioq.h index d38bb78d9345..2c315ca10fc4 100644 --- a/sound/core/seq/seq_prioq.h +++ b/sound/core/seq/seq_prioq.h @@ -44,14 +44,12 @@ void snd_seq_prioq_delete(struct snd_seq_prioq **fifo); int snd_seq_prioq_cell_in(struct snd_seq_prioq *f, struct snd_seq_event_cell *cell); /* dequeue cell from prioq */ -struct snd_seq_event_cell *snd_seq_prioq_cell_out(struct snd_seq_prioq *f); +struct snd_seq_event_cell *snd_seq_prioq_cell_out(struct snd_seq_prioq *f, + void *current_time); /* return number of events available in prioq */ int snd_seq_prioq_avail(struct snd_seq_prioq *f); -/* peek at cell at the head of the prioq */ -struct snd_seq_event_cell *snd_seq_prioq_cell_peek(struct snd_seq_prioq *f); - /* client left queue */ void snd_seq_prioq_leave(struct snd_seq_prioq *f, int client, int timestamp); diff --git a/sound/core/seq/seq_queue.c b/sound/core/seq/seq_queue.c index 0428e9061b47..b377f5048352 100644 --- a/sound/core/seq/seq_queue.c +++ b/sound/core/seq/seq_queue.c @@ -277,30 +277,20 @@ void snd_seq_check_queue(struct snd_seq_queue *q, int atomic, int hop) __again: /* Process tick queue... */ - while ((cell = snd_seq_prioq_cell_peek(q->tickq)) != NULL) { - if (snd_seq_compare_tick_time(&q->timer->tick.cur_tick, - &cell->event.time.tick)) { - cell = snd_seq_prioq_cell_out(q->tickq); - if (cell) - snd_seq_dispatch_event(cell, atomic, hop); - } else { - /* event remains in the queue */ + for (;;) { + cell = snd_seq_prioq_cell_out(q->tickq, + &q->timer->tick.cur_tick); + if (!cell) break; - } + snd_seq_dispatch_event(cell, atomic, hop); } - /* Process time queue... */ - while ((cell = snd_seq_prioq_cell_peek(q->timeq)) != NULL) { - if (snd_seq_compare_real_time(&q->timer->cur_time, - &cell->event.time.time)) { - cell = snd_seq_prioq_cell_out(q->timeq); - if (cell) - snd_seq_dispatch_event(cell, atomic, hop); - } else { - /* event remains in the queue */ + for (;;) { + cell = snd_seq_prioq_cell_out(q->timeq, &q->timer->cur_time); + if (!cell) break; - } + snd_seq_dispatch_event(cell, atomic, hop); } /* free lock */ -- 2.15.0

1 1

[alsa-devel] Applied "ASoC: core: Fix typo roup->group" to the asoc tree
by Mark Brown 09 Mar '18

09 Mar '18

The patch ASoC: core: Fix typo roup->group 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 f7e73b26aea58003931e2f3925d9c01f6b6c4c4d Mon Sep 17 00:00:00 2001 From: Mark Brown <broonie(a)kernel.org> Date: Fri, 9 Mar 2018 12:46:27 +0000 Subject: [PATCH] ASoC: core: Fix typo roup->group Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/soc-core.c | 4 ++-- 1 file changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index e1f047de599e..4cd32f1bceb4 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -221,14 +221,14 @@ static const struct attribute_group soc_dapm_dev_group = { .is_visible = soc_dev_attr_is_visible, }; -static const struct attribute_group soc_dev_roup = { +static const struct attribute_group soc_dev_group = { .attrs = soc_dev_attrs, .is_visible = soc_dev_attr_is_visible, }; static const struct attribute_group *soc_dev_attr_groups[] = { &soc_dapm_dev_group, - &soc_dev_roup, + &soc_dev_group, NULL }; -- 2.16.2

1 0

[alsa-devel] [PATCH v1] ASoC: TSCS42xx: Add missing headers
by Steven Eckhoff 09 Mar '18

09 Mar '18

Add missing headers Signed-off-by: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> --- sound/soc/codecs/tscs42xx.c | 9 +++++++-- 1 file changed, 7 insertions(+), 2 deletions(-) diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c index e915261fa1cb..cfc71b5411ee 100644 --- a/sound/soc/codecs/tscs42xx.c +++ b/sound/soc/codecs/tscs42xx.c @@ -3,14 +3,19 @@ // Copyright 2017 Tempo Semiconductor, Inc. // Author: Steven Eckhoff <steven.eckhoff.opensource(a)gmail.com> +#include <linux/kernel.h> +#include <linux/device.h> +#include <linux/regmap.h> #include <linux/i2c.h> +#include <linux/err.h> +#include <linux/string.h> #include <linux/module.h> +#include <linux/delay.h> #include <linux/mutex.h> -#include <linux/regmap.h> +#include <sound/tlv.h> #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/soc-dapm.h> -#include <sound/tlv.h> #include "tscs42xx.h" -- 2.15.1

2 1

[alsa-devel] [PATCH] ASoC: amd: 16bit resolution support for i2s sp instance
by Vijendar Mukunda 09 Mar '18

09 Mar '18

Moved 16bit resolution condition check for stoney platform to acp_hw_params.Depending upon substream required register value need to be programmed rather than enabling 16bit resolution support all time in acp init. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/acp-pcm-dma.c | 16 +++++++++------- sound/soc/amd/acp.h | 2 ++ 2 files changed, 11 insertions(+), 7 deletions(-) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 9d7ab01..540088d 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -579,13 +579,6 @@ static int acp_init(void __iomem *acp_mmio, u32 asic_type) for (bank = 1; bank < 48; bank++) acp_set_sram_bank_state(acp_mmio, bank, false); } - - /* Stoney supports 16bit resolution */ - if (asic_type == CHIP_STONEY) { - val = acp_reg_read(acp_mmio, mmACP_I2S_16BIT_RESOLUTION_EN); - val |= 0x03; - acp_reg_write(val, acp_mmio, mmACP_I2S_16BIT_RESOLUTION_EN); - } return 0; } @@ -774,6 +767,7 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, { int status; uint64_t size; + u32 val = 0; struct page *pg; struct snd_pcm_runtime *runtime; struct audio_substream_data *rtd; @@ -787,6 +781,14 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, if (WARN_ON(!rtd)) return -EINVAL; + if (adata->asic_type == CHIP_STONEY) { + val = acp_reg_read(adata->acp_mmio, mmACP_I2S_16BIT_RESOLUTION_EN); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + val |= ACP_I2S_SP_16BIT_RESOLUTION_EN; + else + val |= ACP_I2S_MIC_16BIT_RESOLUTION_EN; + acp_reg_write(val, adata->acp_mmio, mmACP_I2S_16BIT_RESOLUTION_EN); + } size = params_buffer_bytes(params); status = snd_pcm_lib_malloc_pages(substream, size); if (status < 0) diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index 27803b2..ba01510 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -70,6 +70,8 @@ #define CAPTURE_END_DMA_DESCR_CH15 7 #define mmACP_I2S_16BIT_RESOLUTION_EN 0x5209 +#define ACP_I2S_MIC_16BIT_RESOLUTION_EN 0x01 +#define ACP_I2S_SP_16BIT_RESOLUTION_EN 0x02 enum acp_dma_priority_level { /* 0x0 Specifies the DMA channel is given normal priority */ ACP_DMA_PRIORITY_LEVEL_NORMAL = 0x0, -- 2.7.4

2 1

[alsa-devel] Applied "ASoC: amd: modifications in dma stop sequence" to the asoc tree
by Mark Brown 09 Mar '18

09 Mar '18

The patch ASoC: amd: modifications in dma stop sequence 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 17aa9521d0c2856808c54ca1144d5c708e748928 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Date: Fri, 9 Mar 2018 20:51:32 +0530 Subject: [PATCH] ASoC: amd: modifications in dma stop sequence As per design, non-circular dma also need to be stopped explicitly for both playback and capture scenarios. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> Signed-off-by: Mark Brown <broonie(a)kernel.org> --- sound/soc/amd/acp-pcm-dma.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 935c8fd42a7a..a1c6534455e6 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -949,12 +949,16 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) * completes : SYSRAM_TO_ACP_CH_NUM / ACP_TO_SYSRAM_CH_NUM */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + ret = acp_dma_stop(rtd->acp_mmio, + SYSRAM_TO_ACP_CH_NUM); ret = acp_dma_stop(rtd->acp_mmio, ACP_TO_I2S_DMA_CH_NUM); rtd->i2ssp_renderbytescount = 0; } else { ret = acp_dma_stop(rtd->acp_mmio, I2S_TO_ACP_DMA_CH_NUM); + ret = acp_dma_stop(rtd->acp_mmio, + ACP_TO_SYSRAM_CH_NUM); rtd->i2ssp_capturebytescount = 0; } break; -- 2.16.2

1 0

[alsa-devel] [PATCH] ASoC: uniphier: add support for UniPhier PXs2 AIO
by Katsuhiro Suzuki 09 Mar '18

09 Mar '18

This patch adds support for UniPhier AIO sound driver which is included in UniPhier PXs2 SoCs. Signed-off-by: Katsuhiro Suzuki <suzuki.katsuhiro(a)socionext.com> --- .../devicetree/bindings/sound/uniphier,aio.txt | 1 + sound/soc/uniphier/Kconfig | 11 + sound/soc/uniphier/Makefile | 2 + sound/soc/uniphier/aio-ld11.c | 2 +- sound/soc/uniphier/aio-pxs2.c | 320 +++++++++++++++++++++ sound/soc/uniphier/aio.h | 2 + 6 files changed, 337 insertions(+), 1 deletion(-) create mode 100644 sound/soc/uniphier/aio-pxs2.c diff --git a/Documentation/devicetree/bindings/sound/uniphier,aio.txt b/Documentation/devicetree/bindings/sound/uniphier,aio.txt index 73f6c27ae4f7..65d71cf6ef0f 100644 --- a/Documentation/devicetree/bindings/sound/uniphier,aio.txt +++ b/Documentation/devicetree/bindings/sound/uniphier,aio.txt @@ -7,6 +7,7 @@ Required properties: - compatible : should be one of the following: "socionext,uniphier-ld11-aio" "socionext,uniphier-ld20-aio" + "socionext,uniphier-pxs2-aio" - reg : offset and length of the register set for the device. - interrupts : should contain I2S or S/PDIF interrupt. - pinctrl-names : should be "default". diff --git a/sound/soc/uniphier/Kconfig b/sound/soc/uniphier/Kconfig index 5da545b9bf2a..aa3592ee1358 100644 --- a/sound/soc/uniphier/Kconfig +++ b/sound/soc/uniphier/Kconfig @@ -30,6 +30,17 @@ config SND_SOC_UNIPHIER_LD11 Select Y if you use such device. If unsure select "N". +config SND_SOC_UNIPHIER_PXS2 + tristate "UniPhier PXs2 Device Driver" + depends on SND_SOC_UNIPHIER + select SND_SOC_UNIPHIER_AIO + select SND_SOC_UNIPHIER_AIO_DMA + help + This adds ASoC driver for Socionext UniPhier PXs2 + input and output that can be used with other codecs. + Select Y if you use such device. + If unsure select "N". + config SND_SOC_UNIPHIER_EVEA_CODEC tristate "UniPhier SoC internal audio codec" depends on SND_SOC_UNIPHIER diff --git a/sound/soc/uniphier/Makefile b/sound/soc/uniphier/Makefile index 587a89700950..88169395f68a 100644 --- a/sound/soc/uniphier/Makefile +++ b/sound/soc/uniphier/Makefile @@ -1,9 +1,11 @@ # SPDX-License-Identifier: GPL-2.0 snd-soc-uniphier-aio-cpu-objs := aio-core.o aio-dma.o aio-cpu.o aio-compress.o snd-soc-uniphier-aio-ld11-objs := aio-ld11.o +snd-soc-uniphier-aio-pxs2-objs := aio-pxs2.o obj-$(CONFIG_SND_SOC_UNIPHIER_AIO) += snd-soc-uniphier-aio-cpu.o obj-$(CONFIG_SND_SOC_UNIPHIER_LD11) += snd-soc-uniphier-aio-ld11.o +obj-$(CONFIG_SND_SOC_UNIPHIER_PXS2) += snd-soc-uniphier-aio-pxs2.o snd-soc-uniphier-evea-objs := evea.o obj-$(CONFIG_SND_SOC_UNIPHIER_EVEA_CODEC) += snd-soc-uniphier-evea.o diff --git a/sound/soc/uniphier/aio-ld11.c b/sound/soc/uniphier/aio-ld11.c index 8e40e6c2e42f..4c4dd3dd4dee 100644 --- a/sound/soc/uniphier/aio-ld11.c +++ b/sound/soc/uniphier/aio-ld11.c @@ -418,7 +418,7 @@ MODULE_DEVICE_TABLE(of, uniphier_aio_of_match); static struct platform_driver uniphier_aio_driver = { .driver = { - .name = "snd-uniphier-aio", + .name = "snd-uniphier-aio-ld11", .of_match_table = of_match_ptr(uniphier_aio_of_match), }, .probe = uniphier_aio_probe, diff --git a/sound/soc/uniphier/aio-pxs2.c b/sound/soc/uniphier/aio-pxs2.c new file mode 100644 index 000000000000..69cd5b0af948 --- /dev/null +++ b/sound/soc/uniphier/aio-pxs2.c @@ -0,0 +1,320 @@ +// SPDX-License-Identifier: GPL-2.0 +// +// Socionext UniPhier AIO ALSA driver for PXs2. +// +// Copyright (c) 2018 Socionext Inc. + +#include <linux/module.h> + +#include "aio.h" + +static const struct uniphier_aio_spec uniphier_aio_pxs2[] = { + /* for Line PCM In, Pin:AI1Dx */ + { + .name = AUD_NAME_PCMIN1, + .gname = AUD_GNAME_LINE, + .swm = { + .type = PORT_TYPE_I2S, + .dir = PORT_DIR_INPUT, + .rb = { 16, 11, }, + .ch = { 16, 11, }, + .iif = { 0, 0, }, + .iport = { 0, AUD_HW_PCMIN1, }, + }, + }, + + /* for Speaker/Headphone/Mic PCM In, Pin:AI2Dx */ + { + .name = AUD_NAME_PCMIN2, + .gname = AUD_GNAME_AUX, + .swm = { + .type = PORT_TYPE_I2S, + .dir = PORT_DIR_INPUT, + .rb = { 17, 12, }, + .ch = { 17, 12, }, + .iif = { 1, 1, }, + .iport = { 1, AUD_HW_PCMIN2, }, + }, + }, + + /* for HDMI PCM Out, Pin:AO1Dx (inner) */ + { + .name = AUD_NAME_HPCMOUT1, + .gname = AUD_GNAME_HDMI, + .swm = { + .type = PORT_TYPE_I2S, + .dir = PORT_DIR_OUTPUT, + .rb = { 0, 0, }, + .ch = { 0, 0, }, + .oif = { 0, 0, }, + .oport = { 3, AUD_HW_HPCMOUT1, }, + }, + }, + + /* for Line PCM Out, Pin:AO2Dx */ + { + .name = AUD_NAME_PCMOUT1, + .gname = AUD_GNAME_LINE, + .swm = { + .type = PORT_TYPE_I2S, + .dir = PORT_DIR_OUTPUT, + .rb = { 1, 1, }, + .ch = { 1, 1, }, + .oif = { 1, 1, }, + .oport = { 0, AUD_HW_PCMOUT1, }, + }, + }, + + /* for Speaker/Headphone/Mic PCM Out, Pin:AO3Dx */ + { + .name = AUD_NAME_PCMOUT2, + .gname = AUD_GNAME_AUX, + .swm = { + .type = PORT_TYPE_I2S, + .dir = PORT_DIR_OUTPUT, + .rb = { 2, 2, }, + .ch = { 2, 2, }, + .oif = { 2, 2, }, + .oport = { 1, AUD_HW_PCMOUT2, }, + }, + }, + + /* for HDMI Out, Pin:AO1IEC */ + { + .name = AUD_NAME_HIECOUT1, + .swm = { + .type = PORT_TYPE_SPDIF, + .dir = PORT_DIR_OUTPUT, + .rb = { 6, 4, }, + .ch = { 6, 4, }, + .oif = { 6, 4, }, + .oport = { 12, AUD_HW_HIECOUT1, }, + }, + }, + + /* for HDMI Out, Pin:AO1IEC, Compress */ + { + .name = AUD_NAME_HIECCOMPOUT1, + .swm = { + .type = PORT_TYPE_SPDIF, + .dir = PORT_DIR_OUTPUT, + .rb = { 6, 4, }, + .ch = { 6, 4, }, + .oif = { 6, 4, }, + .oport = { 12, AUD_HW_HIECOUT1, }, + }, + }, + + /* for S/PDIF Out, Pin:AO2IEC */ + { + .name = AUD_NAME_IECOUT1, + .swm = { + .type = PORT_TYPE_SPDIF, + .dir = PORT_DIR_OUTPUT, + .rb = { 7, 5, }, + .ch = { 7, 5, }, + .oif = { 7, 5, }, + .oport = { 13, AUD_HW_IECOUT1, }, + }, + }, + + /* for S/PDIF Out, Pin:AO2IEC */ + { + .name = AUD_NAME_IECCOMPOUT1, + .swm = { + .type = PORT_TYPE_SPDIF, + .dir = PORT_DIR_OUTPUT, + .rb = { 7, 5, }, + .ch = { 7, 5, }, + .oif = { 7, 5, }, + .oport = { 13, AUD_HW_IECOUT1, }, + }, + }, +}; + +static const struct uniphier_aio_pll uniphier_aio_pll_pxs2[] = { + [AUD_PLL_A1] = { .enable = true, }, + [AUD_PLL_F1] = { .enable = true, }, + [AUD_PLL_A2] = { .enable = true, }, + [AUD_PLL_F2] = { .enable = true, }, + [AUD_PLL_APLL] = { .enable = true, }, + [AUD_PLL_HSC0] = { .enable = true, }, +}; + +static int uniphier_aio_pxs2_probe(struct snd_soc_dai *dai) +{ + int ret; + + ret = uniphier_aio_dai_probe(dai); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_pll(dai, AUD_PLL_A1, 0, 0, 36864000); + if (ret < 0) + return ret; + ret = snd_soc_dai_set_pll(dai, AUD_PLL_F1, 0, 0, 36864000); + if (ret < 0) + return ret; + + ret = snd_soc_dai_set_pll(dai, AUD_PLL_A2, 0, 0, 33868800); + if (ret < 0) + return ret; + ret = snd_soc_dai_set_pll(dai, AUD_PLL_F2, 0, 0, 33868800); + if (ret < 0) + return ret; + + return 0; +} + +static struct snd_soc_dai_driver uniphier_aio_dai_pxs2[] = { + { + .name = AUD_GNAME_HDMI, + .probe = uniphier_aio_pxs2_probe, + .remove = uniphier_aio_dai_remove, + .suspend = uniphier_aio_dai_suspend, + .resume = uniphier_aio_dai_resume, + .playback = { + .stream_name = AUD_NAME_HPCMOUT1, + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .rates = SNDRV_PCM_RATE_48000, + .channels_min = 2, + .channels_max = 2, + }, + .ops = &uniphier_aio_i2s_ops, + }, + { + .name = AUD_GNAME_LINE, + .probe = uniphier_aio_pxs2_probe, + .remove = uniphier_aio_dai_remove, + .suspend = uniphier_aio_dai_suspend, + .resume = uniphier_aio_dai_resume, + .playback = { + .stream_name = AUD_NAME_PCMOUT1, + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .rates = SNDRV_PCM_RATE_48000, + .channels_min = 2, + .channels_max = 2, + }, + .capture = { + .stream_name = AUD_NAME_PCMIN1, + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .rates = SNDRV_PCM_RATE_48000, + .channels_min = 2, + .channels_max = 2, + }, + .ops = &uniphier_aio_i2s_ops, + }, + { + .name = AUD_GNAME_AUX, + .probe = uniphier_aio_pxs2_probe, + .remove = uniphier_aio_dai_remove, + .suspend = uniphier_aio_dai_suspend, + .resume = uniphier_aio_dai_resume, + .playback = { + .stream_name = AUD_NAME_PCMOUT2, + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .rates = SNDRV_PCM_RATE_48000, + .channels_min = 2, + .channels_max = 2, + }, + .capture = { + .stream_name = AUD_NAME_PCMIN2, + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .rates = SNDRV_PCM_RATE_48000, + .channels_min = 2, + .channels_max = 2, + }, + .ops = &uniphier_aio_i2s_ops, + }, + { + .name = AUD_NAME_HIECOUT1, + .probe = uniphier_aio_pxs2_probe, + .remove = uniphier_aio_dai_remove, + .suspend = uniphier_aio_dai_suspend, + .resume = uniphier_aio_dai_resume, + .playback = { + .stream_name = AUD_NAME_HIECOUT1, + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .rates = SNDRV_PCM_RATE_48000, + .channels_min = 2, + .channels_max = 2, + }, + .ops = &uniphier_aio_spdif_ops, + }, + { + .name = AUD_NAME_IECOUT1, + .probe = uniphier_aio_pxs2_probe, + .remove = uniphier_aio_dai_remove, + .suspend = uniphier_aio_dai_suspend, + .resume = uniphier_aio_dai_resume, + .playback = { + .stream_name = AUD_NAME_IECOUT1, + .formats = SNDRV_PCM_FMTBIT_S32_LE, + .rates = SNDRV_PCM_RATE_48000, + .channels_min = 2, + .channels_max = 2, + }, + .ops = &uniphier_aio_spdif_ops, + }, + { + .name = AUD_NAME_HIECCOMPOUT1, + .probe = uniphier_aio_pxs2_probe, + .remove = uniphier_aio_dai_remove, + .suspend = uniphier_aio_dai_suspend, + .resume = uniphier_aio_dai_resume, + .compress_new = snd_soc_new_compress, + .playback = { + .stream_name = AUD_NAME_HIECCOMPOUT1, + .channels_min = 1, + .channels_max = 1, + }, + .ops = &uniphier_aio_spdif_ops, + }, + { + .name = AUD_NAME_IECCOMPOUT1, + .probe = uniphier_aio_pxs2_probe, + .remove = uniphier_aio_dai_remove, + .suspend = uniphier_aio_dai_suspend, + .resume = uniphier_aio_dai_resume, + .compress_new = snd_soc_new_compress, + .playback = { + .stream_name = AUD_NAME_IECCOMPOUT1, + .channels_min = 1, + .channels_max = 1, + }, + .ops = &uniphier_aio_spdif_ops, + }, +}; + +static const struct uniphier_aio_chip_spec uniphier_aio_pxs2_spec = { + .specs = uniphier_aio_pxs2, + .num_specs = ARRAY_SIZE(uniphier_aio_pxs2), + .dais = uniphier_aio_dai_pxs2, + .num_dais = ARRAY_SIZE(uniphier_aio_dai_pxs2), + .plls = uniphier_aio_pll_pxs2, + .num_plls = ARRAY_SIZE(uniphier_aio_pll_pxs2), + .addr_ext = 0, +}; + +static const struct of_device_id uniphier_aio_of_match[] = { + { + .compatible = "socionext,uniphier-pxs2-aio", + .data = &uniphier_aio_pxs2_spec, + }, + {}, +}; +MODULE_DEVICE_TABLE(of, uniphier_aio_of_match); + +static struct platform_driver uniphier_aio_driver = { + .driver = { + .name = "snd-uniphier-aio-pxs2", + .of_match_table = of_match_ptr(uniphier_aio_of_match), + }, + .probe = uniphier_aio_probe, + .remove = uniphier_aio_remove, +}; +module_platform_driver(uniphier_aio_driver); + +MODULE_AUTHOR("Katsuhiro Suzuki <suzuki.katsuhiro(a)socionext.com>"); +MODULE_DESCRIPTION("UniPhier PXs2 AIO driver."); +MODULE_LICENSE("GPL v2"); diff --git a/sound/soc/uniphier/aio.h b/sound/soc/uniphier/aio.h index 2cd64273fb5b..793334675cb3 100644 --- a/sound/soc/uniphier/aio.h +++ b/sound/soc/uniphier/aio.h @@ -110,9 +110,11 @@ enum IEC61937_PC { #define AUD_NAME_IECOUT1 "aio-iecout1" #define AUD_NAME_CMASTER "aio-cmaster" #define AUD_NAME_HIECCOMPOUT1 "aio-hieccompout1" +#define AUD_NAME_IECCOMPOUT1 "aio-ieccompout1" #define AUD_GNAME_HDMI "aio-hdmi" #define AUD_GNAME_LINE "aio-line" +#define AUD_GNAME_AUX "aio-aux" #define AUD_GNAME_IEC "aio-iec" #define AUD_CLK_IO 0 -- 2.16.1

2 1

[alsa-devel] [PATCH] ASoC: da7219: Add common clock usage for providing DAI clks
by Adam Thomson 09 Mar '18

09 Mar '18

There is a need to use DA7219 as DAI clock master for other codecs within a system, which means that the DAI clocks are required to remain, regardless of whether the codec is actually running playback/capture. To be able to expose control of the DAI clocking the common clock framework has been employed. The current implementation adds a simple clock gate for enabling and disabling the DAI clocks, with no rate control supported (this is still handled through standard hw_params() functions as before). If DT is enabled then the clock is added to the OF providers list, otherwise a clkdev lookup is used. Signed-off-by: Adam Thomson <Adam.Thomson.Opensource(a)diasemi.com> --- Documentation/devicetree/bindings/sound/da7219.txt | 6 + include/sound/da7219.h | 2 + sound/soc/codecs/da7219.c | 129 +++++++++++++++++++-- sound/soc/codecs/da7219.h | 9 ++ 4 files changed, 138 insertions(+), 8 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/da7219.txt b/Documentation/devicetree/bindings/sound/da7219.txt index 5b54d2d..c3df92d 100644 --- a/Documentation/devicetree/bindings/sound/da7219.txt +++ b/Documentation/devicetree/bindings/sound/da7219.txt @@ -25,6 +25,9 @@ Optional properties: interrupt is to be used to wake system, otherwise "irq" should be used. - wakeup-source: Flag to indicate this device can wake system (suspend/resume). +- #clock-cells : Should be set to '<0>', only one clock source provided; +- clock-output-names : Name given for DAI clocks output; + - clocks : phandle and clock specifier for codec MCLK. - clock-names : Clock name string for 'clocks' attribute, should be "mclk". @@ -83,6 +86,9 @@ Example: VDDMIC-supply = <&reg_audio>; VDDIO-supply = <&reg_audio>; + #clock-cells = <0>; + clock-output-names = "dai-clks"; + clocks = <&clks 201>; clock-names = "mclk"; diff --git a/include/sound/da7219.h b/include/sound/da7219.h index 409ef139..1bfcb16 100644 --- a/include/sound/da7219.h +++ b/include/sound/da7219.h @@ -36,6 +36,8 @@ enum da7219_mic_amp_in_sel { struct da7219_pdata { bool wakeup_source; + const char *dai_clks_name; + /* Mic */ enum da7219_micbias_voltage micbias_lvl; enum da7219_mic_amp_in_sel mic_amp_in_sel; diff --git a/sound/soc/codecs/da7219.c b/sound/soc/codecs/da7219.c index 5e043d0..4412159 100644 --- a/sound/soc/codecs/da7219.c +++ b/sound/soc/codecs/da7219.c @@ -13,6 +13,8 @@ #include <linux/acpi.h> #include <linux/clk.h> +#include <linux/clkdev.h> +#include <linux/clk-provider.h> #include <linux/i2c.h> #include <linux/of_device.h> #include <linux/property.h> @@ -772,16 +774,27 @@ static int da7219_dai_event(struct snd_soc_dapm_widget *w, struct snd_soc_component *component = snd_soc_dapm_to_component(w->dapm); struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); u8 pll_ctrl, pll_status; - int i = 0; + int i = 0, ret; bool srm_lock = false; switch (event) { case SND_SOC_DAPM_PRE_PMU: - if (da7219->master) + if (da7219->master) { /* Enable DAI clks for master mode */ - snd_soc_component_update_bits(component, DA7219_DAI_CLK_MODE, - DA7219_DAI_CLK_EN_MASK, - DA7219_DAI_CLK_EN_MASK); + if (da7219->dai_clks) { + ret = clk_prepare_enable(da7219->dai_clks); + if (ret) { + dev_err(component->dev, + "Failed to enable dai_clks\n"); + return ret; + } + } else { + snd_soc_component_update_bits(component, + DA7219_DAI_CLK_MODE, + DA7219_DAI_CLK_EN_MASK, + DA7219_DAI_CLK_EN_MASK); + } + } /* PC synchronised to DAI */ snd_soc_component_update_bits(component, DA7219_PC_COUNT, @@ -814,9 +827,16 @@ static int da7219_dai_event(struct snd_soc_dapm_widget *w, DA7219_PC_FREERUN_MASK); /* Disable DAI clks if in master mode */ - if (da7219->master) - snd_soc_component_update_bits(component, DA7219_DAI_CLK_MODE, - DA7219_DAI_CLK_EN_MASK, 0); + if (da7219->master) { + if (da7219->dai_clks) + clk_disable_unprepare(da7219->dai_clks); + else + snd_soc_component_update_bits(component, + DA7219_DAI_CLK_MODE, + DA7219_DAI_CLK_EN_MASK, + 0); + } + return 0; default: return -EINVAL; @@ -1598,6 +1618,12 @@ static struct da7219_pdata *da7219_fw_to_pdata(struct snd_soc_component *compone pdata->wakeup_source = device_property_read_bool(dev, "wakeup-source"); + pdata->dai_clks_name = "da7219-dai-clks"; + if (device_property_read_string(dev, "clock-output-names", + &pdata->dai_clks_name)) + dev_warn(dev, "Using default clk name: %s\n", + pdata->dai_clks_name); + if (device_property_read_u32(dev, "dlg,micbias-lvl", &of_val32) >= 0) pdata->micbias_lvl = da7219_fw_micbias_lvl(dev, of_val32); else @@ -1712,6 +1738,88 @@ static int da7219_handle_supplies(struct snd_soc_component *component) return 0; } +#ifdef CONFIG_COMMON_CLK +static int da7219_dai_clks_prepare(struct clk_hw *hw) +{ + struct da7219_priv *da7219 = + container_of(hw, struct da7219_priv, dai_clks_hw); + struct snd_soc_component *component = da7219->aad->component; + + snd_soc_component_update_bits(component, DA7219_DAI_CLK_MODE, + DA7219_DAI_CLK_EN_MASK, + DA7219_DAI_CLK_EN_MASK); + + return 0; +} + +static void da7219_dai_clks_unprepare(struct clk_hw *hw) +{ + struct da7219_priv *da7219 = + container_of(hw, struct da7219_priv, dai_clks_hw); + struct snd_soc_component *component = da7219->aad->component; + + snd_soc_component_update_bits(component, DA7219_DAI_CLK_MODE, + DA7219_DAI_CLK_EN_MASK, 0); +} + +static int da7219_dai_clks_is_prepared(struct clk_hw *hw) +{ + struct da7219_priv *da7219 = + container_of(hw, struct da7219_priv, dai_clks_hw); + struct snd_soc_component *component = da7219->aad->component; + u8 clk_reg; + + clk_reg = snd_soc_component_read32(component, DA7219_DAI_CLK_MODE); + + return !!(clk_reg & DA7219_DAI_CLK_EN_MASK); +} + +const struct clk_ops da7219_dai_clks_ops = { + .prepare = da7219_dai_clks_prepare, + .unprepare = da7219_dai_clks_unprepare, + .is_prepared = da7219_dai_clks_is_prepared, +}; + +static void da7219_register_dai_clks(struct snd_soc_component *component) +{ + struct device *dev = component->dev; + struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); + struct da7219_pdata *pdata = da7219->pdata; + struct clk_init_data init = {}; + struct clk *dai_clks; + struct clk_lookup *dai_clks_lookup; + + init.parent_names = NULL; + init.num_parents = 0; + init.name = pdata->dai_clks_name; + init.ops = &da7219_dai_clks_ops; + da7219->dai_clks_hw.init = &init; + + dai_clks = devm_clk_register(dev, &da7219->dai_clks_hw); + if (IS_ERR(dai_clks)) { + dev_warn(dev, "Failed to register DAI clocks: %ld\n", + PTR_ERR(dai_clks)); + return; + } + da7219->dai_clks = dai_clks; + + /* If we're using DT, then register as provider accordingly */ + if (dev->of_node) { + devm_of_clk_add_hw_provider(dev, of_clk_hw_simple_get, + &da7219->dai_clks_hw); + } else { + dai_clks_lookup = clkdev_create(dai_clks, pdata->dai_clks_name, + "%s", dev_name(dev)); + if (!dai_clks_lookup) + dev_warn(dev, "Failed to create DAI clkdev"); + else + da7219->dai_clks_lookup = dai_clks_lookup; + } +} +#else +static inline void da7219_register_dai_clks(struct snd_soc_component *component) {} +#endif /* CONFIG_COMMON_CLK */ + static void da7219_handle_pdata(struct snd_soc_component *component) { struct da7219_priv *da7219 = snd_soc_component_get_drvdata(component); @@ -1722,6 +1830,8 @@ static void da7219_handle_pdata(struct snd_soc_component *component) da7219->wakeup_source = pdata->wakeup_source; + da7219_register_dai_clks(component); + /* Mic Bias voltages */ switch (pdata->micbias_lvl) { case DA7219_MICBIAS_1_6V: @@ -1856,6 +1966,9 @@ static void da7219_remove(struct snd_soc_component *component) da7219_aad_exit(component); + if (da7219->dai_clks_lookup) + clkdev_drop(da7219->dai_clks_lookup); + /* Supplies */ regulator_bulk_disable(DA7219_NUM_SUPPLIES, da7219->supplies); } diff --git a/sound/soc/codecs/da7219.h b/sound/soc/codecs/da7219.h index 1acb34c..1b00023 100644 --- a/sound/soc/codecs/da7219.h +++ b/sound/soc/codecs/da7219.h @@ -14,6 +14,9 @@ #ifndef __DA7219_H #define __DA7219_H +#include <linux/clk.h> +#include <linux/clkdev.h> +#include <linux/clk-provider.h> #include <linux/regmap.h> #include <linux/regulator/consumer.h> #include <sound/da7219.h> @@ -813,6 +816,12 @@ struct da7219_priv { struct mutex ctrl_lock; struct mutex pll_lock; +#ifdef CONFIG_COMMON_CLK + struct clk_hw dai_clks_hw; +#endif + struct clk_lookup *dai_clks_lookup; + struct clk *dai_clks; + struct clk *mclk; unsigned int mclk_rate; int clk_src; -- 1.9.1

2 1

[alsa-devel] [PATCH] ASoC: amd: modifications in dma stop sequence
by Vijendar Mukunda 09 Mar '18

09 Mar '18

As per design, non-circular dma also need to be stopped explicitly for both playback and capture scenarios. Signed-off-by: Vijendar Mukunda <Vijendar.Mukunda(a)amd.com> --- sound/soc/amd/acp-pcm-dma.c | 4 ++++ 1 file changed, 4 insertions(+) diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 91c4775..9d7ab01 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -954,11 +954,15 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) */ if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = acp_dma_stop(rtd->acp_mmio, + SYSRAM_TO_ACP_CH_NUM); + ret = acp_dma_stop(rtd->acp_mmio, ACP_TO_I2S_DMA_CH_NUM); rtd->i2ssp_renderbytescount = 0; } else { ret = acp_dma_stop(rtd->acp_mmio, I2S_TO_ACP_DMA_CH_NUM); + ret = acp_dma_stop(rtd->acp_mmio, + ACP_TO_SYSRAM_CH_NUM); rtd->i2ssp_capturebytescount = 0; } break; -- 2.7.4

1 0

[alsa-devel] Cirrus CS4207 not working on 4.14.24
by Jorge Fernandez Monteagudo 09 Mar '18

09 Mar '18

Hi all! I've updated my system from a 3.16.7 kernel to a 4.14.24 kernel and sound stops working. I have a Cirrus logic CS4207 codec. I've tracked down the problem using the '/proc/asound/card0/codec\#0' info. With the 3.16.7 kernel all nodes power states are: Node 0x03: Power: setting=D0, actual=D0 Node 0x0a: Power: setting=D0, actual=D0 and with 4.14.24 they are on Node 0x03: Power: setting=D0, actual=D3 Node 0x0a: Power: setting=D3, actual=D3 Once in this situation I'm able to go to D0 again and make the sound works. I've used hda-verb 0.4 to SET_POWER_STATE to 0 with no luck. Finally, I've patched the 4.14.24 to avoid changing to D3 and now the sound is working again. I know it's not the way to go but I need the sound... --- a/linux-4.14.24/sound/hda/hdac_device.c 2018-03-08 14:12:44.263575493 +0100 +++ b/linux-4.14.24/sound/hda/hdac_device.c 2018-03-08 14:12:44.263575493 +0100 @@ -1061,6 +1061,9 @@ int snd_hdac_codec_write(struct hdac_device *hdac, hda_nid_t nid, int flags, unsigned int verb, unsigned int parm) { + /* Ugly patch to avoid going to low power state!! */ + if (verb == 0x705 && parm == 0x3) + return 0; return codec_write(hdac, nid, flags, verb, parm); } EXPORT_SYMBOL_GPL(snd_hdac_codec_write); I've patched both kernels to show all the codec writes to show all the sequence in both kernels. Maybe it can help to fix the problem in the correct way. kernel 3.16.7 [ 28.685789] snd_hda_intel: probe of 0000:00:01.1 failed with error -2 [ 28.697622] -- snd_hda_codec_write: nid 0x2, verb:0x705, parm:0x0 [ 28.697681] -- snd_hda_codec_write: nid 0x3, verb:0x705, parm:0x0 [ 28.697726] -- snd_hda_codec_write: nid 0x4, verb:0x705, parm:0x0 [ 28.697762] -- snd_hda_codec_write: nid 0x5, verb:0x705, parm:0x0 [ 28.697807] -- snd_hda_codec_write: nid 0x6, verb:0x705, parm:0x0 [ 28.697851] -- snd_hda_codec_write: nid 0x7, verb:0x705, parm:0x0 [ 28.697887] -- snd_hda_codec_write: nid 0x8, verb:0x705, parm:0x0 [ 28.697932] -- snd_hda_codec_write: nid 0x9, verb:0x705, parm:0x0 [ 28.697977] -- snd_hda_codec_write: nid 0xA, verb:0x705, parm:0x0 [ 28.698013] -- snd_hda_codec_write: nid 0xC, verb:0x705, parm:0x0 [ 28.698058] -- snd_hda_codec_write: nid 0xD, verb:0x705, parm:0x0 [ 28.698102] -- snd_hda_codec_write: nid 0xF, verb:0x705, parm:0x0 [ 28.698138] -- snd_hda_codec_write: nid 0x14, verb:0x705, parm:0x0 [ 28.698669] sound hdaudioC0D0: autoconfig: line_outs=1 (0xa/0x0/0x0/0x0/0x0) type:line [ 28.698673] sound hdaudioC0D0: speaker_outs=1 (0xb/0x0/0x0/0x0/0x0) [ 28.698676] sound hdaudioC0D0: hp_outs=1 (0x9/0x0/0x0/0x0/0x0) [ 28.698678] sound hdaudioC0D0: mono: mono_out=0x0 [ 28.698681] sound hdaudioC0D0: dig-out=0x10/0x15 [ 28.698683] sound hdaudioC0D0: inputs: [ 28.698687] sound hdaudioC0D0: Mic=0xd [ 28.698689] sound hdaudioC0D0: Line=0xc [ 28.698691] sound hdaudioC0D0: dig-in=0x12 [ 28.699332] -- snd_hda_codec_write: nid 0x9, verb:0x708, parm:0x81 [ 28.699370] -- snd_hda_codec_write: nid 0xA, verb:0x708, parm:0x82 [ 28.700249] -- snd_hda_codec_write: nid 0x1, verb:0x705, parm:0x0 [ 28.700285] -- snd_hda_codec_write: nid 0x11, verb:0x703, parm:0x1 [ 28.700330] -- snd_hda_codec_write: nid 0x11, verb:0x500, parm:0x8 [ 28.700376] -- snd_hda_codec_write: nid 0x11, verb:0x400, parm:0x9999 [ 28.700411] -- snd_hda_codec_write: nid 0x11, verb:0x500, parm:0x17 [ 28.700456] -- snd_hda_codec_write: nid 0x11, verb:0x400, parm:0xA412 [ 28.700501] -- snd_hda_codec_write: nid 0x11, verb:0x500, parm:0x1 [ 28.700537] -- snd_hda_codec_write: nid 0x11, verb:0x400, parm:0x9 [ 28.700582] -- snd_hda_codec_write: nid 0x7, verb:0x705, parm:0x0 [ 28.700626] -- snd_hda_codec_write: nid 0x8, verb:0x705, parm:0x0 [ 28.700662] -- snd_hda_codec_write: nid 0x11, verb:0x500, parm:0x17 [ 28.700706] -- snd_hda_codec_write: nid 0x11, verb:0x400, parm:0x2412 [ 28.700751] -- snd_hda_codec_write: nid 0x11, verb:0x500, parm:0x8 [ 28.700785] -- snd_hda_codec_write: nid 0x11, verb:0x400, parm:0x0 [ 28.700830] -- snd_hda_codec_write: nid 0x11, verb:0x500, parm:0x1 [ 28.700874] -- snd_hda_codec_write: nid 0x11, verb:0x400, parm:0x8 [ 28.700911] -- snd_hda_codec_write: nid 0x11, verb:0x703, parm:0x0 [ 28.700956] -- snd_hda_codec_write: nid 0x11, verb:0x703, parm:0x1 [ 28.701001] -- snd_hda_codec_write: nid 0x11, verb:0x500, parm:0x3 [ 28.701038] -- snd_hda_codec_write: nid 0x11, verb:0x400, parm:0x146A [ 28.701083] -- snd_hda_codec_write: nid 0x11, verb:0x500, parm:0x2 [ 28.701124] -- snd_hda_codec_write: nid 0x11, verb:0x400, parm:0xA [ 28.701169] -- snd_hda_codec_write: nid 0x11, verb:0x500, parm:0x4 [ 28.701226] -- snd_hda_codec_write: nid 0x11, verb:0x400, parm:0x7 [ 28.702188] -- snd_hda_codec_write: nid 0x3, verb:0x300, parm:0xA080 [ 28.702235] -- snd_hda_codec_write: nid 0x3, verb:0x300, parm:0x9080 [ 28.702295] -- snd_hda_codec_write: nid 0x2, verb:0x300, parm:0xA080 [ 28.702330] -- snd_hda_codec_write: nid 0x2, verb:0x300, parm:0x9080 [ 28.702379] -- snd_hda_codec_write: nid 0x4, verb:0x300, parm:0xA080 [ 28.702414] -- snd_hda_codec_write: nid 0x4, verb:0x300, parm:0x9080 [ 28.702459] -- snd_hda_codec_write: nid 0x6, verb:0x300, parm:0x6080 [ 28.702497] -- snd_hda_codec_write: nid 0x6, verb:0x300, parm:0x5080 [ 28.702542] -- snd_hda_codec_write: nid 0x5, verb:0x300, parm:0x6080 [ 28.702586] -- snd_hda_codec_write: nid 0x5, verb:0x300, parm:0x5080 [ 28.702621] -- snd_hda_codec_write: nid 0x6, verb:0x300, parm:0x6180 [ 28.702666] -- snd_hda_codec_write: nid 0x6, verb:0x300, parm:0x5180 [ 28.702710] -- snd_hda_codec_write: nid 0x5, verb:0x300, parm:0x6180 [ 28.702745] -- snd_hda_codec_write: nid 0x5, verb:0x300, parm:0x5180 [ 28.702791] -- snd_hda_codec_write: nid 0xA, verb:0x707, parm:0x40 [ 28.702836] -- snd_hda_codec_write: nid 0x9, verb:0x707, parm:0xC0 [ 28.702872] -- snd_hda_codec_write: nid 0xB, verb:0x707, parm:0x40 [ 28.702917] -- snd_hda_codec_write: nid 0xD, verb:0x707, parm:0x24 [ 28.702962] -- snd_hda_codec_write: nid 0xC, verb:0x707, parm:0x20 [ 28.702998] -- snd_hda_codec_write: nid 0x6, verb:0x701, parm:0x0 [ 28.703044] -- snd_hda_codec_write: nid 0x10, verb:0x707, parm:0x40 [ 28.703081] -- snd_hda_codec_write: nid 0x15, verb:0x707, parm:0x40 [ 28.703130] -- snd_hda_codec_write: nid 0x12, verb:0x707, parm:0x0 [ 28.703185] -- snd_hda_codec_write: nid 0xC, verb:0x708, parm:0x0 [ 28.703230] -- snd_hda_codec_write: nid 0xD, verb:0x708, parm:0x0 [ 28.703275] -- snd_hda_codec_write: nid 0xF, verb:0x708, parm:0x0 [ 28.703311] -- snd_hda_codec_write: nid 0x11, verb:0x500, parm:0x4 [ 28.703398] -- snd_hda_codec_write: nid 0x11, verb:0x500, parm:0x4 [ 28.703434] -- snd_hda_codec_write: nid 0x11, verb:0x400, parm:0x1F [ 28.703479] -- snd_hda_codec_write: nid 0x11, verb:0x500, parm:0x1 [ 28.703517] -- snd_hda_codec_write: nid 0x11, verb:0x400, parm:0x400A [ 28.704045] -- snd_hda_codec_write: nid 0xD, verb:0x708, parm:0x83 [ 28.704148] -- snd_hda_codec_write: nid 0xC, verb:0x708, parm:0x84 [ 28.917726] -- snd_hda_codec_write: nid 0xB, verb:0x707, parm:0x0 [ 28.917797] -- snd_hda_codec_write: nid 0xA, verb:0x707, parm:0x0 [ 29.746841] -- snd_hda_codec_write: nid 0x3, verb:0x300, parm:0xA000 [ 29.746893] -- snd_hda_codec_write: nid 0x3, verb:0x300, parm:0x9000 [ 29.746940] -- snd_hda_codec_write: nid 0x2, verb:0x300, parm:0xA000 [ 29.746991] -- snd_hda_codec_write: nid 0x2, verb:0x300, parm:0x9000 [ 29.747037] -- snd_hda_codec_write: nid 0x4, verb:0x300, parm:0xA000 [ 29.747074] -- snd_hda_codec_write: nid 0x4, verb:0x300, parm:0x9000 [ 29.761263] -- snd_hda_codec_write: nid 0x8, verb:0x70D, parm:0x1 [ 29.761322] -- snd_hda_codec_write: nid 0x14, verb:0x70D, parm:0x1 [ 29.773546] -- snd_hda_codec_write: nid 0x3, verb:0x300, parm:0xA073 [ 29.773601] -- snd_hda_codec_write: nid 0x3, verb:0x300, parm:0x9073 [ 29.773648] -- snd_hda_codec_write: nid 0x2, verb:0x300, parm:0xA073 [ 29.773693] -- snd_hda_codec_write: nid 0x2, verb:0x300, parm:0x9073 [ 29.773749] -- snd_hda_codec_write: nid 0x4, verb:0x300, parm:0xA073 [ 29.773793] -- snd_hda_codec_write: nid 0x4, verb:0x300, parm:0x9073 [ 29.779742] -- snd_hda_codec_write: nid 0x3, verb:0x300, parm:0xA062 [ 29.779829] -- snd_hda_codec_write: nid 0x3, verb:0x300, parm:0x9062 [ 29.794831] -- snd_hda_codec_write: nid 0x8, verb:0x706, parm:0x50 [ 29.796917] -- snd_hda_codec_write: nid 0x8, verb:0x200, parm:0x31 [ 29.797021] -- snd_hda_codec_write: nid 0x14, verb:0x706, parm:0x50 [ 29.798923] -- snd_hda_codec_write: nid 0x14, verb:0x200, parm:0x31 [ 29.799030] -- snd_hda_codec_write: nid 0x3, verb:0x706, parm:0x50 [ 29.800919] -- snd_hda_codec_write: nid 0x3, verb:0x200, parm:0x31 [ 29.801005] -- snd_hda_codec_write: nid 0x2, verb:0x706, parm:0x50 [ 29.802960] -- snd_hda_codec_write: nid 0x2, verb:0x200, parm:0x31 [ 29.803053] -- snd_hda_codec_write: nid 0x4, verb:0x706, parm:0x50 [ 29.804932] -- snd_hda_codec_write: nid 0x4, verb:0x200, parm:0x31 [ 33.323243] systemd-udevd[1441]: starting version 215 kernel 4.14.24 [ 33.942455] snd_hda_intel: probe of 0000:00:01.1 failed with error -2 [ 33.949069] -- snd_hda_codec_write: nid 0x2, verb:0x705, parm:0x0 [ 33.949105] -- snd_hda_codec_write: nid 0x3, verb:0x705, parm:0x0 [ 33.949149] -- snd_hda_codec_write: nid 0x4, verb:0x705, parm:0x0 [ 33.949196] -- snd_hda_codec_write: nid 0x5, verb:0x705, parm:0x0 [ 33.949231] -- snd_hda_codec_write: nid 0x6, verb:0x705, parm:0x0 [ 33.949275] -- snd_hda_codec_write: nid 0x7, verb:0x705, parm:0x0 [ 33.949318] -- snd_hda_codec_write: nid 0x8, verb:0x705, parm:0x0 [ 33.949363] -- snd_hda_codec_write: nid 0x9, verb:0x705, parm:0x0 [ 33.949397] -- snd_hda_codec_write: nid 0xA, verb:0x705, parm:0x0 [ 33.949441] -- snd_hda_codec_write: nid 0xC, verb:0x705, parm:0x0 [ 33.949484] -- snd_hda_codec_write: nid 0xD, verb:0x705, parm:0x0 [ 33.949522] -- snd_hda_codec_write: nid 0xF, verb:0x705, parm:0x0 [ 33.949566] -- snd_hda_codec_write: nid 0x14, verb:0x705, parm:0x0 [ 33.950243] snd_hda_codec_cirrus hdaudioC0D0: autoconfig for CS4207: line_outs=1 (0xa/0x0/0x0/0x0/0x0) type:line [ 33.950245] snd_hda_codec_cirrus hdaudioC0D0: speaker_outs=1 (0xb/0x0/0x0/0x0/0x0) [ 33.950251] snd_hda_codec_cirrus hdaudioC0D0: hp_outs=1 (0x9/0x0/0x0/0x0/0x0) [ 33.950252] snd_hda_codec_cirrus hdaudioC0D0: mono: mono_out=0x0 [ 33.950253] snd_hda_codec_cirrus hdaudioC0D0: dig-out=0x10/0x15 [ 33.950254] snd_hda_codec_cirrus hdaudioC0D0: inputs: [ 33.950256] snd_hda_codec_cirrus hdaudioC0D0: Mic=0xd [ 33.950257] snd_hda_codec_cirrus hdaudioC0D0: Line=0xc [ 33.950258] snd_hda_codec_cirrus hdaudioC0D0: dig-in=0x12 [ 33.951210] -- snd_hda_codec_write: nid 0x4, verb:0x705, parm:0x3 [ 33.952301] -- snd_hda_codec_write: nid 0x1, verb:0x705, parm:0x0 [ 33.952336] -- snd_hda_codec_write: nid 0x11, verb:0x703, parm:0x1 [ 33.952381] -- snd_hda_codec_write: nid 0x11, verb:0x500, parm:0x8 [ 33.952420] -- snd_hda_codec_write: nid 0x11, verb:0x400, parm:0x9999 [ 33.952464] -- snd_hda_codec_write: nid 0x11, verb:0x500, parm:0x17 [ 33.952502] -- snd_hda_codec_write: nid 0x11, verb:0x400, parm:0xA412 [ 33.952547] -- snd_hda_codec_write: nid 0x11, verb:0x500, parm:0x1 [ 33.952586] -- snd_hda_codec_write: nid 0x11, verb:0x400, parm:0x9 [ 33.952629] -- snd_hda_codec_write: nid 0x7, verb:0x705, parm:0x0 [ 33.952669] -- snd_hda_codec_write: nid 0x8, verb:0x705, parm:0x0 [ 33.952713] -- snd_hda_codec_write: nid 0x11, verb:0x500, parm:0x17 [ 33.952752] -- snd_hda_codec_write: nid 0x11, verb:0x400, parm:0x2412 [ 33.952796] -- snd_hda_codec_write: nid 0x11, verb:0x500, parm:0x8 [ 33.952835] -- snd_hda_codec_write: nid 0x11, verb:0x400, parm:0x0 [ 33.952879] -- snd_hda_codec_write: nid 0x11, verb:0x500, parm:0x1 [ 33.952918] -- snd_hda_codec_write: nid 0x11, verb:0x400, parm:0x8 [ 33.952962] -- snd_hda_codec_write: nid 0x11, verb:0x703, parm:0x0 [ 33.953006] -- snd_hda_codec_write: nid 0x11, verb:0x703, parm:0x1 [ 33.953045] -- snd_hda_codec_write: nid 0x11, verb:0x500, parm:0x3 [ 33.953087] -- snd_hda_codec_write: nid 0x11, verb:0x400, parm:0x146A [ 33.953132] -- snd_hda_codec_write: nid 0x11, verb:0x500, parm:0x2 [ 33.953178] -- snd_hda_codec_write: nid 0x11, verb:0x400, parm:0xA [ 33.953234] -- snd_hda_codec_write: nid 0x11, verb:0x500, parm:0x4 [ 33.953273] -- snd_hda_codec_write: nid 0x11, verb:0x400, parm:0x7 [ 33.953421] -- snd_hda_codec_write: nid 0x3, verb:0x705, parm:0x3 [ 33.953504] -- snd_hda_codec_write: nid 0xA, verb:0x705, parm:0x3 [ 33.953796] -- snd_hda_codec_write: nid 0x2, verb:0x705, parm:0x3 [ 33.953879] -- snd_hda_codec_write: nid 0x9, verb:0x705, parm:0x3 [ 33.954468] -- snd_hda_codec_write: nid 0xD, verb:0x705, parm:0x3 [ 33.955253] -- snd_hda_codec_write: nid 0x8, verb:0x705, parm:0x3 [ 33.955378] -- snd_hda_codec_write: nid 0x14, verb:0x705, parm:0x3 [ 33.955924] -- snd_hda_codec_write: nid 0x11, verb:0x500, parm:0x4 [ 33.956026] -- snd_hda_codec_write: nid 0x11, verb:0x500, parm:0x4 [ 33.956065] -- snd_hda_codec_write: nid 0x11, verb:0x400, parm:0x1F [ 33.956109] -- snd_hda_codec_write: nid 0x11, verb:0x500, parm:0x1 [ 33.956149] -- snd_hda_codec_write: nid 0x11, verb:0x400, parm:0x400A [ 33.956587] -- snd_hda_codec_write: nid 0x7, verb:0x705, parm:0x3 [ 33.956796] -- snd_hda_codec_write: nid 0xC, verb:0x705, parm:0x3 [ 33.956919] -- snd_hda_codec_write: nid 0xF, verb:0x705, parm:0x3 [ 33.957382] input: HD-Audio Generic Mic as /devices/pci0000:00/0000:00:09.2/sound/card0/input5 [ 33.957430] input: HD-Audio Generic Line as /devices/pci0000:00/0000:00:09.2/sound/card0/input6 [ 33.957476] input: HD-Audio Generic Line Out as /devices/pci0000:00/0000:00:09.2/sound/card0/input7 [ 33.957518] input: HD-Audio Generic Front Headphone as /devices/pci0000:00/0000:00:09.2/sound/card0/input8 [ 33.970031] -- snd_hda_codec_write: nid 0x2, verb:0x705, parm:0x3 [ 33.970091] -- snd_hda_codec_write: nid 0x3, verb:0x705, parm:0x3 [ 33.970127] -- snd_hda_codec_write: nid 0x4, verb:0x705, parm:0x3 [ 33.970170] -- snd_hda_codec_write: nid 0x5, verb:0x705, parm:0x3 [ 33.970214] -- snd_hda_codec_write: nid 0x6, verb:0x705, parm:0x3 [ 33.970253] -- snd_hda_codec_write: nid 0x7, verb:0x705, parm:0x3 [ 33.970297] -- snd_hda_codec_write: nid 0x8, verb:0x705, parm:0x3 [ 33.970336] -- snd_hda_codec_write: nid 0x9, verb:0x705, parm:0x3 [ 33.970380] -- snd_hda_codec_write: nid 0xA, verb:0x705, parm:0x3 [ 33.970420] -- snd_hda_codec_write: nid 0xC, verb:0x705, parm:0x3 [ 33.970463] -- snd_hda_codec_write: nid 0xD, verb:0x705, parm:0x3 [ 33.970502] -- snd_hda_codec_write: nid 0xF, verb:0x705, parm:0x3 [ 33.970546] -- snd_hda_codec_write: nid 0x14, verb:0x705, parm:0x3 [ 33.970693] -- snd_hda_codec_write: nid 0x2, verb:0x705, parm:0x3 [ 33.970732] -- snd_hda_codec_write: nid 0x3, verb:0x705, parm:0x3 [ 33.970775] -- snd_hda_codec_write: nid 0x4, verb:0x705, parm:0x3 [ 33.970815] -- snd_hda_codec_write: nid 0x5, verb:0x705, parm:0x0 [ 33.970858] -- snd_hda_codec_write: nid 0x6, verb:0x705, parm:0x0 [ 33.970903] -- snd_hda_codec_write: nid 0x7, verb:0x705, parm:0x3 [ 33.970942] -- snd_hda_codec_write: nid 0x8, verb:0x705, parm:0x3 [ 33.970987] -- snd_hda_codec_write: nid 0x9, verb:0x705, parm:0x3 [ 33.971025] -- snd_hda_codec_write: nid 0xA, verb:0x705, parm:0x3 [ 33.971065] -- snd_hda_codec_write: nid 0xC, verb:0x705, parm:0x3 [ 33.971109] -- snd_hda_codec_write: nid 0xD, verb:0x705, parm:0x3 [ 33.971153] -- snd_hda_codec_write: nid 0xF, verb:0x705, parm:0x3 [ 33.971191] -- snd_hda_codec_write: nid 0x14, verb:0x705, parm:0x3 [ 33.971276] -- snd_hda_codec_write: nid 0x1, verb:0x705, parm:0x0 [ 33.971314] -- snd_hda_codec_write: nid 0x11, verb:0x703, parm:0x1 [ 33.971358] -- snd_hda_codec_write: nid 0x11, verb:0x500, parm:0x8 [ 33.971402] -- snd_hda_codec_write: nid 0x11, verb:0x400, parm:0x9999 [ 33.971441] -- snd_hda_codec_write: nid 0x11, verb:0x500, parm:0x17 [ 33.971485] -- snd_hda_codec_write: nid 0x11, verb:0x400, parm:0xA412 [ 33.971524] -- snd_hda_codec_write: nid 0x11, verb:0x500, parm:0x1 [ 33.971569] -- snd_hda_codec_write: nid 0x11, verb:0x400, parm:0x9 [ 33.971608] -- snd_hda_codec_write: nid 0x7, verb:0x705, parm:0x0 [ 33.971652] -- snd_hda_codec_write: nid 0x8, verb:0x705, parm:0x0 [ 33.971691] -- snd_hda_codec_write: nid 0x11, verb:0x500, parm:0x17 [ 33.971735] -- snd_hda_codec_write: nid 0x11, verb:0x400, parm:0x2412 [ 33.971781] -- snd_hda_codec_write: nid 0x11, verb:0x500, parm:0x8 [ 33.971815] -- snd_hda_codec_write: nid 0x11, verb:0x400, parm:0x0 [ 33.971864] -- snd_hda_codec_write: nid 0x11, verb:0x500, parm:0x1 [ 33.971898] -- snd_hda_codec_write: nid 0x11, verb:0x400, parm:0x8 [ 33.971942] -- snd_hda_codec_write: nid 0x11, verb:0x703, parm:0x0 [ 33.971987] -- snd_hda_codec_write: nid 0x11, verb:0x703, parm:0x1 [ 33.972023] -- snd_hda_codec_write: nid 0x11, verb:0x500, parm:0x3 [ 33.972068] -- snd_hda_codec_write: nid 0x11, verb:0x400, parm:0x146A [ 33.972107] -- snd_hda_codec_write: nid 0x11, verb:0x500, parm:0x2 [ 33.972151] -- snd_hda_codec_write: nid 0x11, verb:0x400, parm:0xA [ 33.972189] -- snd_hda_codec_write: nid 0x11, verb:0x500, parm:0x4 [ 33.972234] -- snd_hda_codec_write: nid 0x11, verb:0x400, parm:0x7 [ 33.972815] -- snd_hda_codec_write: nid 0x6, verb:0x705, parm:0x0 [ 33.973004] -- snd_hda_codec_write: nid 0x8, verb:0x705, parm:0x3 [ 33.974881] -- snd_hda_codec_write: nid 0x11, verb:0x500, parm:0x4 [ 33.974962] -- snd_hda_codec_write: nid 0x11, verb:0x500, parm:0x4 [ 33.975002] -- snd_hda_codec_write: nid 0x11, verb:0x400, parm:0x1F [ 33.975046] -- snd_hda_codec_write: nid 0x11, verb:0x500, parm:0x1 [ 33.975086] -- snd_hda_codec_write: nid 0x11, verb:0x400, parm:0x400A [ 35.029721] -- snd_hda_codec_write: nid 0x8, verb:0x705, parm:0x0 [ 35.029816] -- snd_hda_codec_write: nid 0x8, verb:0x706, parm:0x50 [ 35.032018] -- snd_hda_codec_write: nid 0x8, verb:0x200, parm:0x31 [ 35.032108] -- snd_hda_codec_write: nid 0x14, verb:0x705, parm:0x0 [ 35.032192] -- snd_hda_codec_write: nid 0x14, verb:0x706, parm:0x50 [ 35.036014] -- snd_hda_codec_write: nid 0x14, verb:0x200, parm:0x31 [ 35.036130] -- snd_hda_codec_write: nid 0x3, verb:0x705, parm:0x0 [ 35.036253] -- snd_hda_codec_write: nid 0x3, verb:0x706, parm:0x50 [ 35.039063] -- snd_hda_codec_write: nid 0x3, verb:0x200, parm:0x31 [ 35.039155] -- snd_hda_codec_write: nid 0x2, verb:0x705, parm:0x0 [ 35.039275] -- snd_hda_codec_write: nid 0x2, verb:0x706, parm:0x50 [ 35.042023] -- snd_hda_codec_write: nid 0x2, verb:0x200, parm:0x31 [ 35.042109] -- snd_hda_codec_write: nid 0x4, verb:0x705, parm:0x0 [ 35.042192] -- snd_hda_codec_write: nid 0x4, verb:0x706, parm:0x50 [ 35.045006] -- snd_hda_codec_write: nid 0x4, verb:0x200, parm:0x31 [ 38.553348] systemd-udevd[1465]: starting version 215

2 8

Re: [alsa-devel] [PATCH] dt-bindings: exynos: Document #sound-dai-cells property of the HDMI node
by Sylwester Nawrocki 09 Mar '18

09 Mar '18

Hi Inki, Cc: alsa-devel(a)alsa-project.org On 03/08/2018 09:15 AM, Inki Dae wrote: > By the way, it seems 'sound-dai-cells' property never affect Exynos4210/4212> 5420/5433. It seems that even through ALSA TM2 audio driver(tm2_wm5110.c) > exists the driver never check the property. > > However, this patch adds below description. > > "Optional properties for Exynos 4210, 4212, 5420 and 5433" > > Is there a possibility for other boards based on Exynos4210/4212/5420/5433 > SoC to use this property later? All these SoCs have the HDMI IP block which has one input DAI, connected internally over I2S bus with the IS2 controller. I think there is no advantage in limiting ourselves now only to SoC's for which we currently rely on that DT property in current kernel code, just to update this documentation later when we actually put the property in dts files. In case of exynos5420 we already require #sound-dai-cells for Odroid and I have also a patch for exynos5420-peach-pit board which will need it as well. As far as exynos4210 and exynos4212 are concerned it's a matter of adding support for Odroid-U3, then we will also need this property because we are going to use "multi-codec" (HDMI and external I2S0 pins for CODEC are wired in parallel). In case of exynos5433 it just happens that the code in current driver doesn't require #sound-dai-cells property - one of the reasons I made it this way was to avoid dependency on dts, but it doesn't imply we should describe the HW in DT incompletely. Once the property is in dtbs we can update the driver to use more generic code, instead of open coding things. Actually I have forgotten to add also exynos5250 to the list. -- Regards, Sylwester

2 2

[alsa-devel] [PATCH 0/9] Enablement of BT I2S controller instance for AMD APUs
by Vijendar Mukunda 09 Mar '18

09 Mar '18

This patch set updates Audio CoProcessor (ACP) audio drivers to enable BT I2S controller instance. Default enabled I2S controller instance is I2S SP instance. There is a requirement to enable BT I2S controller Instance along with I2S SP controller instance. This patch set bring up functionality to render & capture on BT I2S controller instance. The current code is based on asoc-next, but I'm happy to rebase on whatever tree this ends up going through if there are any problems applying. Vijendar Mukunda (9): ASoC: amd: renaming pcm substream names and bytescount params ASoC: amd: Coding style changes for acp dma driver ASoC: amd: dma driver changes for BT I2S controller instance ASoC: amd: dma descriptor changes for BT I2S Instance ASoC: amd: Interrupt handler changes for BT I2S instance ASoC: amd: pcm callbacks modifications for bt i2s instance ASoC: amd: modifications in dma stop sequence ASoC: amd: 16bit resolution support for bt i2s instance ASoC: amd: enabling bt i2s config after acp reset sound/soc/amd/acp-pcm-dma.c | 623 ++++++++++++++++++++++++++++++++------------ sound/soc/amd/acp.h | 61 ++++- 2 files changed, 514 insertions(+), 170 deletions(-) -- 2.7.4

3 16

[alsa-devel] [PATCH] ASoC: TSCS42xx: make const array norm_addrs static, reduces object code size
by Colin King 09 Mar '18

09 Mar '18

From: Colin Ian King <colin.king(a)canonical.com> Don't populate the const array norm_addrs on the stack, instead make it static. Makes the object code smaller by over 230 bytes. Also re-format array data as the insertion of the static keywork made the first line overly long. Before: text data bss dec hex filename 53780 34752 256 88788 15ad4 linux/sound/soc/codecs/tscs42xx.o After: text data bss dec hex filename 53461 34840 256 88557 159ed linux/sound/soc/codecs/tscs42xx.o (gcc version 7.2.0 x86_64) Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/codecs/tscs42xx.c | 12 +++++++----- 1 file changed, 7 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/tscs42xx.c b/sound/soc/codecs/tscs42xx.c index e7661d0315e6..205a55a0329e 100644 --- a/sound/soc/codecs/tscs42xx.c +++ b/sound/soc/codecs/tscs42xx.c @@ -1317,11 +1317,13 @@ static struct snd_soc_codec_driver soc_codec_dev_tscs42xx = { static inline void init_coeff_ram_cache(struct tscs42xx *tscs42xx) { - const u8 norm_addrs[] = { 0x00, 0x05, 0x0a, 0x0f, 0x14, 0x19, 0x1f, - 0x20, 0x25, 0x2a, 0x2f, 0x34, 0x39, 0x3f, 0x40, 0x45, 0x4a, - 0x4f, 0x54, 0x59, 0x5f, 0x60, 0x65, 0x6a, 0x6f, 0x74, 0x79, - 0x7f, 0x80, 0x85, 0x8c, 0x91, 0x96, 0x97, 0x9c, 0xa3, 0xa8, - 0xad, 0xaf, 0xb0, 0xb5, 0xba, 0xbf, 0xc4, 0xc9, }; + static const u8 norm_addrs[] = { + 0x00, 0x05, 0x0a, 0x0f, 0x14, 0x19, 0x1f, 0x20, 0x25, 0x2a, + 0x2f, 0x34, 0x39, 0x3f, 0x40, 0x45, 0x4a, 0x4f, 0x54, 0x59, + 0x5f, 0x60, 0x65, 0x6a, 0x6f, 0x74, 0x79, 0x7f, 0x80, 0x85, + 0x8c, 0x91, 0x96, 0x97, 0x9c, 0xa3, 0xa8, 0xad, 0xaf, 0xb0, + 0xb5, 0xba, 0xbf, 0xc4, 0xc9, + }; u8 *coeff_ram = tscs42xx->coeff_ram; int i; -- 2.15.1

3 2

[alsa-devel] [PATCH] ASoC: cygnus: remove redundant assignment to pointer 'res'
by Colin King 08 Mar '18

08 Mar '18

From: Colin Ian King <colin.king(a)canonical.com> The pointer res is being initialized with a value that is never read and re-assigned immediately after, hence the initialization is redundant and can be removed. Cleans up clang warning: sound/soc/bcm/cygnus-ssp.c:1284:19: warning: Value stored to 'res' during its initialization is never read Signed-off-by: Colin Ian King <colin.king(a)canonical.com> --- sound/soc/bcm/cygnus-ssp.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/bcm/cygnus-ssp.c b/sound/soc/bcm/cygnus-ssp.c index abafadc0b534..b733f1446353 100644 --- a/sound/soc/bcm/cygnus-ssp.c +++ b/sound/soc/bcm/cygnus-ssp.c @@ -1281,7 +1281,7 @@ static int cygnus_ssp_probe(struct platform_device *pdev) { struct device *dev = &pdev->dev; struct device_node *child_node; - struct resource *res = pdev->resource; + struct resource *res; struct cygnus_audio *cygaud; int err = -EINVAL; int node_count; -- 2.15.1

3 2

[alsa-devel] linux-next on n900: oops in codec_reg_show() when grepping sysfs
by Pavel Machek 08 Mar '18

08 Mar '18

Hi! pavel@n900:/sys$ uname -a Linux n900 4.16.0-rc4-next-20180308 #71 Thu Mar 8 15:27:49 CET 2018 armv7l GNU/Linux pavel@n900:/sys$ grep -ri fasdfasdf . leads to an oops: [ 2278.331878] DSS: context saved [ 2278.820343] Unable to handle kernel NULL pointer dereference at virtual address 00000004 [ 2278.828948] pgd = c36040a2 [ 2278.831787] [00000004] *pgd=876c4831, *pte=00000000, *ppte=00000000 [ 2278.838439] Internal error: Oops: 17 [#1] ARM [ 2278.843017] Modules linked in: [ 2278.846221] CPU: 0 PID: 16337 Comm: grep Tainted: G W 4.16.0-rc4-next-20180308 #71 [ 2278.855529] Hardware name: Nokia RX-51 board [ 2278.860015] PC is at soc_codec_reg_show+0x8/0x19c [ 2278.864959] LR is at codec_reg_show+0x28/0x30 [ 2278.869506] pc : [<c0598f84>] lr : [<c0599140>] psr: a0000013 [ 2278.876098] sp : ce063e10 ip : cd7a9000 fp : 00000001 [ 2278.881561] r10: cd7a8000 r9 : c0842a04 r8 : 00001000 [ 2278.887023] r7 : 00000fff r6 : ce69fd80 r5 : 00000000 r4 : 00000000 [ 2278.893890] r3 : ce5d2000 r2 : 00001000 r1 : cd7a8000 r0 : 00000000 [ 2278.900726] Flags: NzCv IRQs on FIQs on Mode SVC_32 ISA ARM Segment none [ 2278.908203] Control: 10c5387d Table: 87664019 DAC: 00000051 [ 2278.914215] Process grep (pid: 16337, stack limit = 0x2c91d92b) [ 2278.920410] Stack: (0xce063e10 to 0xce064000) [ 2278.924987] 3e00: ce063ed0 00000000 00000000 ce063ed0 [ 2278.933563] 3e20: 00000000 00000000 00000000 c01e113c 00000000 ce063ed0 ce063f74 c01e5068 [ 2278.942138] 3e40: c059c388 00000000 00000000 ce69fd80 00000fff 00001000 c0842a04 cd7a8000 [ 2278.950714] 3e60: 00000001 c0599140 00000000 00000000 00000040 c7699200 c0c4a0c8 c0412f14 [ 2278.959289] 3e80: c0412ef8 c7699200 ce597008 c023d598 ce063ec0 00008000 cd7a6000 ce063f80 [ 2278.967895] 3ea0: c7699200 c7699230 00000001 c023bf90 c023bf6c c01fab30 00042000 00000000 [ 2278.976470] 3ec0: 00000000 00000000 00000000 00000000 00000000 c080f7c4 00042000 cd7a6000 [ 2278.985046] 3ee0: ce063f80 00008000 ce062000 00042000 00041f90 c01d82e8 5aa1523b 34732c66 [ 2278.993621] 3f00: 5aa1523b 34732c66 00004024 00000000 00000000 00000000 0003520c c718eaf0 [ 2279.002197] 3f20: 00020000 cd7a6008 00000004 ce063f80 ce062000 00042000 00041f90 c02fad9c [ 2279.010772] 3f40: 00000000 00008000 cd7a6000 00042000 00000000 ce063f80 00042000 c01d8464 [ 2279.019348] 3f60: 00000010 cd7a6000 00000000 00000000 cd7a6000 cd7a6000 00008000 c01d88ec [ 2279.027923] 3f80: 00000000 00000000 beaa2744 00035330 00042000 00008000 00000003 c01011e4 [ 2279.036499] 3fa0: 00000000 c0101000 00035330 00042000 00000004 00042000 00008000 00008000 [ 2279.045074] 3fc0: 00035330 00042000 00008000 00000003 00001000 00000000 7fffffff 00041f90 [ 2279.053649] 3fe0: 00000000 beaa2748 0000d350 b6eb78dc 40000010 00000004 00000000 00000000 [ 2279.062255] [<c0598f84>] (soc_codec_reg_show) from [<c0599140>] (codec_reg_show+0x28/0x30) [ 2279.070922] [<c0599140>] (codec_reg_show) from [<c0412f14>] (dev_attr_show+0x1c/0x4c) [ 2279.079162] [<c0412f14>] (dev_attr_show) from [<c023d598>] (sysfs_kf_seq_show+0x90/0x108) [ 2279.087738] [<c023d598>] (sysfs_kf_seq_show) from [<c023bf90>] (kernfs_seq_show+0x24/0x28) [ 2279.096435] [<c023bf90>] (kernfs_seq_show) from [<c01fab30>] (seq_read+0x1dc/0x500) [ 2279.104461] [<c01fab30>] (seq_read) from [<c01d82e8>] (__vfs_read+0x2c/0x120) [ 2279.111968] [<c01d82e8>] (__vfs_read) from [<c01d8464>] (vfs_read+0x88/0x114) [ 2279.119445] [<c01d8464>] (vfs_read) from [<c01d88ec>] (SyS_read+0x40/0x8c) [ 2279.126678] [<c01d88ec>] (SyS_read) from [<c0101000>] (ret_fast_syscall+0x0/0x54) [ 2279.134521] Exception stack(0xce063fa8 to 0xce063ff0) [ 2279.139801] 3fa0: 00035330 00042000 00000004 00042000 00008000 00008000 [ 2279.148376] 3fc0: 00035330 00042000 00008000 00000003 00001000 00000000 7fffffff 00041f90 [ 2279.156951] 3fe0: 00000000 beaa2748 0000d350 b6eb78dc [ 2279.162261] Code: ea05f08a c09c66ec e92d4ff0 e24dd034 (e5905004) [ 2279.168792] ---[ end trace cbbe0b4a3061a0e7 ]--- -- (english) http://www.livejournal.com/~pavelmachek (cesky, pictures) http://atrey.karlin.mff.cuni.cz/~pavel/picture/horses/blog.html

2 1

[alsa-devel] [PATCH] ASoC: uniphier: remove superfluous <linux/mfd/syscon.h> inclusion
by Masahiro Yamada 08 Mar '18

08 Mar '18

None of aio-compress.c depends on the syscon header. Signed-off-by: Masahiro Yamada <yamada.masahiro(a)socionext.com> Acked-by: Katsuhiro Suzuki <suzuki.katsuhiro(a)socionext.com> --- sound/soc/uniphier/aio-compress.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/soc/uniphier/aio-compress.c b/sound/soc/uniphier/aio-compress.c index 4751c8b..4c1027a 100644 --- a/sound/soc/uniphier/aio-compress.c +++ b/sound/soc/uniphier/aio-compress.c @@ -22,7 +22,6 @@ #include <linux/dma-mapping.h> #include <linux/errno.h> #include <linux/kernel.h> -#include <linux/mfd/syscon.h> #include <linux/module.h> #include <sound/core.h> #include <sound/pcm.h> -- 2.7.4

2 1

[alsa-devel] [PATCH] ASoC: Use proper DT compatible string for Hardkernel Odroid boards
by Sylwester Nawrocki 08 Mar '18

08 Mar '18

The Odroid boards are manufactured by Hardkernel, not Samsung. New compatible string entries are added, with "hardkernel," instead of "samsung," vendor prefix. Support for the old compatible strings is going to be removed after some time. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- Documentation/devicetree/bindings/sound/samsung,odroid.txt | 6 +++--- sound/soc/samsung/odroid.c | 4 +++- 2 files changed, 6 insertions(+), 4 deletions(-) diff --git a/Documentation/devicetree/bindings/sound/samsung,odroid.txt b/Documentation/devicetree/bindings/sound/samsung,odroid.txt index 625b1b18fd02..f35a2694eb04 100644 --- a/Documentation/devicetree/bindings/sound/samsung,odroid.txt +++ b/Documentation/devicetree/bindings/sound/samsung,odroid.txt @@ -2,8 +2,8 @@ Samsung Exynos Odroid XU3/XU4 audio complex with MAX98090 codec Required properties: - - compatible - "samsung,odroidxu3-audio" - for Odroid XU3 board, - "samsung,odroidxu4-audio" - for Odroid XU4 board + - compatible - "hardkernel,odroid-xu3-audio" - for Odroid XU3 board, + "hardkernel,odroid-xu4-audio" - for Odroid XU4 board - model - the user-visible name of this sound complex - clocks - should contain entries matching clock names in the clock-names property @@ -35,7 +35,7 @@ Required sub-nodes: Example: sound { - compatible = "samsung,odroidxu3-audio"; + compatible = "hardkernel,odroid-xu3-audio"; model = "Odroid-XU3"; samsung,audio-routing = "Headphone Jack", "HPL", diff --git a/sound/soc/samsung/odroid.c b/sound/soc/samsung/odroid.c index 44b6de5a331a..92d750806d1d 100644 --- a/sound/soc/samsung/odroid.c +++ b/sound/soc/samsung/odroid.c @@ -213,8 +213,10 @@ static int odroid_audio_remove(struct platform_device *pdev) } static const struct of_device_id odroid_audio_of_match[] = { + { .compatible = "hardkernel,odroid-xu3-audio" }, + { .compatible = "hardkernel,odroid-xu4-audio" }, { .compatible = "samsung,odroid-xu3-audio" }, - { .compatible = "samsung,odroid-xu4-audio"}, + { .compatible = "samsung,odroid-xu4-audio" }, { }, }; MODULE_DEVICE_TABLE(of, odroid_audio_of_match); -- 2.14.2

4 5

[alsa-devel] [PATCH 2/2] ALSA: hda: add dock and led support for HP ProBook 640 G2
by Dennis Wassenberg 08 Mar '18

08 Mar '18

This patch adds missing initialisation for HP 2013 UltraSlim Dock Line-In/Out PINs and activates keyboard mute/micmute leds for HP ProBook 640 G2 Signed-off-by: Dennis Wassenberg <dennis.wassenberg(a)secunet.com> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index d5c9b88..5b4dbce 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -958,6 +958,7 @@ static void cxt_fixup_mute_led_gpio(struct hda_codec *codec, SND_PCI_QUIRK(0x1025, 0x054f, "Acer Aspire 4830T", CXT_FIXUP_ASPIRE_DMIC), SND_PCI_QUIRK(0x103c, 0x8079, "HP EliteBook 840 G3", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x807C, "HP EliteBook 820 G3", CXT_FIXUP_HP_DOCK), + SND_PCI_QUIRK(0x103c, 0x80FD, "HP ProBook 640 G2", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE), SND_PCI_QUIRK(0x103c, 0x8115, "HP Z1 Gen3", CXT_FIXUP_HP_GATE_MIC), SND_PCI_QUIRK(0x103c, 0x814f, "HP ZBook 15u G3", CXT_FIXUP_MUTE_LED_GPIO), -- 1.9.1

2 1

[alsa-devel] [PATCH 1/2] ALSA: hda: add dock and led support for HP EliteBook 820 G3
by Dennis Wassenberg 08 Mar '18

08 Mar '18

This patch adds missing initialisation for HP 2013 UltraSlim Dock Line-In/Out PINs and activates keyboard mute/micmute leds for HP EliteBook 820 G3 Signed-off-by: Dennis Wassenberg <dennis.wassenberg(a)secunet.com> --- sound/pci/hda/patch_conexant.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_conexant.c b/sound/pci/hda/patch_conexant.c index 37e1cf8..d5c9b88 100644 --- a/sound/pci/hda/patch_conexant.c +++ b/sound/pci/hda/patch_conexant.c @@ -957,6 +957,7 @@ static void cxt_fixup_mute_led_gpio(struct hda_codec *codec, SND_PCI_QUIRK(0x1025, 0x054c, "Acer Aspire 3830TG", CXT_FIXUP_ASPIRE_DMIC), SND_PCI_QUIRK(0x1025, 0x054f, "Acer Aspire 4830T", CXT_FIXUP_ASPIRE_DMIC), SND_PCI_QUIRK(0x103c, 0x8079, "HP EliteBook 840 G3", CXT_FIXUP_HP_DOCK), + SND_PCI_QUIRK(0x103c, 0x807C, "HP EliteBook 820 G3", CXT_FIXUP_HP_DOCK), SND_PCI_QUIRK(0x103c, 0x8174, "HP Spectre x360", CXT_FIXUP_HP_SPECTRE), SND_PCI_QUIRK(0x103c, 0x8115, "HP Z1 Gen3", CXT_FIXUP_HP_GATE_MIC), SND_PCI_QUIRK(0x103c, 0x814f, "HP ZBook 15u G3", CXT_FIXUP_MUTE_LED_GPIO), -- 1.9.1

2 1

[alsa-devel] [PATCH] hda: realtek: Make dock sound work on ThinkPad L570
by Dennis Wassenberg 08 Mar '18

08 Mar '18

One version of Lenovo Thinkpad T570 did not use ALC298 (like other Kaby Lake devices). Instead it uses ALC292. In order to make the Lenovo dock working with that codec the dock quirk for ALC292 will be used. Signed-off-by: Dennis Wassenberg <dennis.wassenberg(a)secunet.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index b9c93fa..b5d0f8c 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6508,6 +6508,7 @@ enum { SND_PCI_QUIRK(0x17aa, 0x2245, "Thinkpad T470", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x2246, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x2247, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), + SND_PCI_QUIRK(0x17aa, 0x2249, "Thinkpad", ALC292_FIXUP_TPT460), SND_PCI_QUIRK(0x17aa, 0x224b, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x224c, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x224d, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), -- 1.9.1

2 3

[alsa-devel] [PATCH 6/6] ASoC: samsung: Add HDMI audio support for Chromebook Snow
by Sylwester Nawrocki 08 Mar '18

08 Mar '18

This patch updates the driver so, in addition to current DT bindings, it also handles the new DT bindings with cpu, codec subnodes which allow for sound support on the HDMI interface. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- sound/soc/samsung/snow.c | 225 +++++++++++++++++++++++++++++++++++++---------- 1 file changed, 177 insertions(+), 48 deletions(-) diff --git a/sound/soc/samsung/snow.c b/sound/soc/samsung/snow.c index fc62110f500f..d10fd578b60d 100644 --- a/sound/soc/samsung/snow.c +++ b/sound/soc/samsung/snow.c @@ -11,92 +11,207 @@ * General Public License for more details. */ +#include <linux/clk.h> #include <linux/module.h> #include <linux/platform_device.h> #include <linux/of.h> #include <linux/of_device.h> - +#include <sound/pcm_params.h> #include <sound/soc.h> #include "i2s.h" -#define FIN_PLL_RATE 24000000 +#define FIN_PLL_RATE 24000000UL -static struct snd_soc_dai_link snow_dai[] = { - { - .name = "Primary", - .stream_name = "Primary", - .codec_dai_name = "HiFi", - .dai_fmt = SND_SOC_DAIFMT_I2S | - SND_SOC_DAIFMT_NB_NF | - SND_SOC_DAIFMT_CBS_CFS, - }, +struct snow_priv { + struct snd_soc_dai_link dai_link; + struct clk *clk_i2s_bus; +}; + +static int snow_card_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + static const unsigned int pll_rate[] = { + 73728000U, 67737602U, 49152000U, 45158401U, 32768001U + }; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snow_priv *priv = snd_soc_card_get_drvdata(rtd->card); + int bfs, psr, rfs, bitwidth; + unsigned long int rclk; + long int freq = -EINVAL; + int ret, i; + + bitwidth = snd_pcm_format_width(params_format(params)); + if (bitwidth < 0) { + dev_err(rtd->card->dev, "Invalid bit-width: %d\n", bitwidth); + return bitwidth; + } + + if (bitwidth != 16 && bitwidth != 24) { + dev_err(rtd->card->dev, "Unsupported bit-width: %d\n", bitwidth); + return -EINVAL; + } + + bfs = 2 * bitwidth; + + switch (params_rate(params)) { + case 16000: + case 22050: + case 24000: + case 32000: + case 44100: + case 48000: + case 88200: + case 96000: + rfs = 8 * bfs; + break; + case 64000: + rfs = 384; + break; + case 8000: + case 11025: + case 12000: + rfs = 16 * bfs; + break; + default: + return -EINVAL; + } + + rclk = params_rate(params) * rfs; + + for (psr = 8; psr > 0; psr /= 2) { + for (i = 0; i < ARRAY_SIZE(pll_rate); i++) { + if ((pll_rate[i] - rclk * psr) <= 2) { + freq = pll_rate[i]; + break; + } + } + } + if (freq < 0) { + dev_err(rtd->card->dev, "Unsupported RCLK rate: %lu\n", rclk); + return -EINVAL; + } + + ret = clk_set_rate(priv->clk_i2s_bus, freq); + if (ret < 0) { + dev_err(rtd->card->dev, "I2S bus clock rate set failed\n"); + return ret; + } + + return 0; +} + +static const struct snd_soc_ops snow_card_ops = { + .hw_params = snow_card_hw_params, }; static int snow_late_probe(struct snd_soc_card *card) { struct snd_soc_pcm_runtime *rtd; struct snd_soc_dai *codec_dai; - struct snd_soc_dai *cpu_dai; - int ret; rtd = snd_soc_get_pcm_runtime(card, card->dai_link[0].name); - codec_dai = rtd->codec_dai; - cpu_dai = rtd->cpu_dai; - - /* Set the MCLK rate for the codec */ - ret = snd_soc_dai_set_sysclk(codec_dai, 0, - FIN_PLL_RATE, SND_SOC_CLOCK_IN); - if (ret < 0) - return ret; + /* In the multi-codec case codec_dai 0 is HDMI and 1 is MAX98095. */ + if (rtd->num_codecs > 1) + codec_dai = rtd->codec_dais[1]; + else + codec_dai = rtd->codec_dai; - return 0; + /* Set the MCLK rate for the codec */ + return snd_soc_dai_set_sysclk(codec_dai, 0, + FIN_PLL_RATE, SND_SOC_CLOCK_IN); } static struct snd_soc_card snow_snd = { .name = "Snow-I2S", .owner = THIS_MODULE, - .dai_link = snow_dai, - .num_links = ARRAY_SIZE(snow_dai), - .late_probe = snow_late_probe, }; static int snow_probe(struct platform_device *pdev) { + struct device *dev = &pdev->dev; struct snd_soc_card *card = &snow_snd; - struct device_node *i2s_node, *codec_node; - int i, ret; - - i2s_node = of_parse_phandle(pdev->dev.of_node, - "samsung,i2s-controller", 0); - if (!i2s_node) { - dev_err(&pdev->dev, - "Property 'i2s-controller' missing or invalid\n"); - return -EINVAL; - } + struct device_node *cpu, *codec; + struct snd_soc_dai_link *link; + struct snow_priv *priv; + int ret; - codec_node = of_parse_phandle(pdev->dev.of_node, - "samsung,audio-codec", 0); - if (!codec_node) { - dev_err(&pdev->dev, - "Property 'audio-codec' missing or invalid\n"); - return -EINVAL; - } + priv = devm_kzalloc(dev, sizeof(*priv), GFP_KERNEL); + if (!priv) + return -ENOMEM; + + link = &priv->dai_link; + + link->dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS; + + link->name = "Primary"; + link->stream_name = link->name; + + card->dai_link = link; + card->num_links = 1; + card->dev = dev; + + /* Try new DT bindings with HDMI support first. */ + cpu = of_get_child_by_name(dev->of_node, "cpu"); + + if (cpu) { + link->ops = &snow_card_ops; - for (i = 0; i < ARRAY_SIZE(snow_dai); i++) { - snow_dai[i].codec_of_node = codec_node; - snow_dai[i].cpu_of_node = i2s_node; - snow_dai[i].platform_of_node = i2s_node; + link->cpu_of_node = of_parse_phandle(cpu, "sound-dai", 0); + of_node_put(cpu); + + if (!link->cpu_of_node) { + dev_err(dev, "Failed parsing cpu/sound-dai property\n"); + return -EINVAL; + } + + codec = of_get_child_by_name(dev->of_node, "codec"); + ret = snd_soc_of_get_dai_link_codecs(dev, codec, link); + of_node_put(codec); + + if (ret < 0) { + of_node_put(link->cpu_of_node); + dev_err(dev, "Failed parsing codec node\n"); + return ret; + } + + priv->clk_i2s_bus = of_clk_get_by_name(link->cpu_of_node, + "i2s_opclk0"); + if (IS_ERR(priv->clk_i2s_bus)) { + snd_soc_of_put_dai_link_codecs(link); + of_node_put(link->cpu_of_node); + return PTR_ERR(priv->clk_i2s_bus); + } + } else { + link->codec_dai_name = "HiFi", + + link->cpu_of_node = of_parse_phandle(dev->of_node, + "samsung,i2s-controller", 0); + if (!link->cpu_of_node) { + dev_err(dev, "i2s-controller property parse error\n"); + return -EINVAL; + } + + link->codec_of_node = of_parse_phandle(dev->of_node, + "samsung,audio-codec", 0); + if (!link->codec_of_node) { + of_node_put(link->cpu_of_node); + dev_err(dev, "audio-codec property parse error\n"); + return -EINVAL; + } } - card->dev = &pdev->dev; + link->platform_of_node = link->cpu_of_node; /* Update card-name if provided through DT, else use default name */ snd_soc_of_parse_card_name(card, "samsung,model"); - ret = devm_snd_soc_register_card(&pdev->dev, card); + snd_soc_card_set_drvdata(card, priv); + + ret = devm_snd_soc_register_card(dev, card); if (ret) { dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret); return ret; @@ -105,6 +220,19 @@ static int snow_probe(struct platform_device *pdev) return ret; } +static int snow_remove(struct platform_device *pdev) +{ + struct snow_priv *priv = platform_get_drvdata(pdev); + + of_node_put(priv->dai_link.cpu_of_node); + of_node_put(link->codec_of_node); + snd_soc_of_put_dai_link_codecs(&priv->dai_link); + + clk_put(priv->clk_i2s_bus); + + return 0; +} + static const struct of_device_id snow_of_match[] = { { .compatible = "google,snow-audio-max98090", }, { .compatible = "google,snow-audio-max98091", }, @@ -120,6 +248,7 @@ static struct platform_driver snow_driver = { .of_match_table = snow_of_match, }, .probe = snow_probe, + .remove = snow_remove, }; module_platform_driver(snow_driver); -- 2.14.2

1 1

[alsa-devel] [PATCH] ALSA: seq: Remove superfluous snd_seq_queue_client_leave_cells() call
by Takashi Iwai 08 Mar '18

08 Mar '18

With the previous two fixes for the write / ioctl races: ALSA: seq: Don't allow resizing pool in use ALSA: seq: More protection for concurrent write and ioctl races the cells aren't any longer in queues at the point calling snd_seq_pool_done() in snd_seq_ioctl_set_client_pool(). Hence the function call snd_seq_queue_client_leave_cells() can be dropped safely from there. Suggested-by: Nicolai Stange <nstange(a)suse.de> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/seq/seq_clientmgr.c | 1 - 1 file changed, 1 deletion(-) diff --git a/sound/core/seq/seq_clientmgr.c b/sound/core/seq/seq_clientmgr.c index 1b62421dadd1..918338dea5b9 100644 --- a/sound/core/seq/seq_clientmgr.c +++ b/sound/core/seq/seq_clientmgr.c @@ -1846,7 +1846,6 @@ static int snd_seq_ioctl_set_client_pool(struct snd_seq_client *client, return -EBUSY; /* remove all existing cells */ snd_seq_pool_mark_closing(client->pool); - snd_seq_queue_client_leave_cells(client->number); snd_seq_pool_done(client->pool); } client->pool->size = info->output_pool; -- 2.16.2

1 0

[alsa-devel] [PATCH 0/2] More hardening for ALSA sequencer write/ioctl races
by Takashi Iwai 08 Mar '18

08 Mar '18

Hi, here is two patches to paper over the still remaining races in ALSA sequencer write and ioctl that have been leaked in the previous fix for CVE-2018-1000004. The reports came up totally individually, so both people are put as Reported-by tag here. thanks, Takashi === Takashi Iwai (2): ALSA: seq: Don't allow resizing pool in use ALSA: seq: More protection for concurrent write and ioctl races sound/core/seq/seq_clientmgr.c | 21 ++++++++++++++------- sound/core/seq/seq_fifo.c | 2 +- sound/core/seq/seq_memory.c | 14 ++++++++++---- sound/core/seq/seq_memory.h | 3 ++- 4 files changed, 27 insertions(+), 13 deletions(-) -- 2.16.2

2 5

[alsa-devel] [PATCH v2 0/2] ASoC: Add support for DAC PCM1789
by Mylène Josserand 08 Mar '18

08 Mar '18

Hello everyone, The current series is the second version to add the support of Texas Instrument's DAC PCM1789. This DAC is very minimalist and does not have many registers. It is important to notice that this DAC needs to always have clocks enabled (even without any data) otherwise it will be in a "desynchronized" state and can not send data correctly. This issue has been solved by performing a reset each time a sound is played. This reset can produce a "pop" noise. Depending on your DAI, you will need to provide and enable the MCLK to be able to communicate with this codec throught i2c. Changes since v1: - Create a new file to support pcm1789 instead of converting the existing pcm179x driver. All the patches are merged into one patch. - Update the code to use gpiod for the reset. - Add some fixes according to Thomas Petazzoni's reviews - Create a new patch to add device-tree bindings for this new DAC. Thank you in advance for any review. Best regards, Mylène Mylène Josserand (2): ASoC: codecs: Add support for PCM1789 ASoC: Add bindings for PCM1789 .../devicetree/bindings/sound/pcm1789.txt | 21 ++ sound/soc/codecs/Kconfig | 12 + sound/soc/codecs/Makefile | 4 + sound/soc/codecs/pcm1789-i2c.c | 76 ++++++ sound/soc/codecs/pcm1789.c | 288 +++++++++++++++++++++ sound/soc/codecs/pcm1789.h | 28 ++ 6 files changed, 429 insertions(+) create mode 100644 Documentation/devicetree/bindings/sound/pcm1789.txt create mode 100644 sound/soc/codecs/pcm1789-i2c.c create mode 100644 sound/soc/codecs/pcm1789.c create mode 100644 sound/soc/codecs/pcm1789.h -- 2.11.0

4 8

[alsa-devel] [PATCH 1/2] dell-led: Change dell-led.h to dell-common.h
by Kai-Heng Feng 08 Mar '18

08 Mar '18

This header will be used for more than just led. Change it to a more generic name. Signed-off-by: Kai-Heng Feng <kai.heng.feng(a)canonical.com> --- drivers/platform/x86/dell-laptop.c | 2 +- include/linux/{dell-led.h => dell-common.h} | 4 ++-- sound/pci/hda/dell_wmi_helper.c | 2 +- 3 files changed, 4 insertions(+), 4 deletions(-) rename include/linux/{dell-led.h => dell-common.h} (61%) diff --git a/drivers/platform/x86/dell-laptop.c b/drivers/platform/x86/dell-laptop.c index c52c6723374b..8ba820e6c3d0 100644 --- a/drivers/platform/x86/dell-laptop.c +++ b/drivers/platform/x86/dell-laptop.c @@ -29,7 +29,7 @@ #include <linux/mm.h> #include <linux/i8042.h> #include <linux/debugfs.h> -#include <linux/dell-led.h> +#include <linux/dell-common.h> #include <linux/seq_file.h> #include <acpi/video.h> #include "dell-rbtn.h" diff --git a/include/linux/dell-led.h b/include/linux/dell-common.h similarity index 61% rename from include/linux/dell-led.h rename to include/linux/dell-common.h index 92521471517f..37e4b614dd74 100644 --- a/include/linux/dell-led.h +++ b/include/linux/dell-common.h @@ -1,6 +1,6 @@ /* SPDX-License-Identifier: GPL-2.0 */ -#ifndef __DELL_LED_H__ -#define __DELL_LED_H__ +#ifndef __DELL_COMMON_H__ +#define __DELL_COMMON_H__ int dell_micmute_led_set(int on); diff --git a/sound/pci/hda/dell_wmi_helper.c b/sound/pci/hda/dell_wmi_helper.c index 1b48a8c19d28..56050cc3c0ee 100644 --- a/sound/pci/hda/dell_wmi_helper.c +++ b/sound/pci/hda/dell_wmi_helper.c @@ -4,7 +4,7 @@ */ #if IS_ENABLED(CONFIG_DELL_LAPTOP) -#include <linux/dell-led.h> +#include <linux/dell-common.h> enum { MICMUTE_LED_ON, -- 2.15.1

4 6

[alsa-devel] [PATCH 1/2] ALSA: vmaster: Propagate slave error
by Takashi Iwai 08 Mar '18

08 Mar '18

In slave_update() of vmaster code ignores the error from the slave get() callback and copies the values. It's not only about the missing error code but also that this may potentially lead to a leak of uninitialized variables when the slave get() don't clear them. This patch fixes slave_update() not to copy the potentially uninitialized values when an error is returned from the slave get() callback, and to propagate the error value properly. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/core/vmaster.c | 5 ++++- 1 file changed, 4 insertions(+), 1 deletion(-) diff --git a/sound/core/vmaster.c b/sound/core/vmaster.c index 8632301489fa..b67de2bb06a2 100644 --- a/sound/core/vmaster.c +++ b/sound/core/vmaster.c @@ -68,10 +68,13 @@ static int slave_update(struct link_slave *slave) return -ENOMEM; uctl->id = slave->slave.id; err = slave->slave.get(&slave->slave, uctl); + if (err < 0) + goto error; for (ch = 0; ch < slave->info.count; ch++) slave->vals[ch] = uctl->value.integer.value[ch]; + error: kfree(uctl); - return 0; + return err < 0 ? err : 0; } /* get the slave ctl info and save the initial values */ -- 2.16.2

1 1

[alsa-devel] [PATCH 3/6] ASoC: max98095: Add #sound-dai-cells property documentation
by Sylwester Nawrocki 08 Mar '18

08 Mar '18

MAX98095 CODEC supports 3 input DAIs, adding the #sound-dai-cells property allows to reference selected DAI in DT. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- Documentation/devicetree/bindings/sound/max98095.txt | 3 +++ 1 file changed, 3 insertions(+) diff --git a/Documentation/devicetree/bindings/sound/max98095.txt b/Documentation/devicetree/bindings/sound/max98095.txt index 318a4c82f17f..944efa06487b 100644 --- a/Documentation/devicetree/bindings/sound/max98095.txt +++ b/Documentation/devicetree/bindings/sound/max98095.txt @@ -14,6 +14,9 @@ Optional properties: - clock-names: Should be "mclk" +- #sound-dai-cells : should be 1. + + Example: max98095: codec@11 { -- 2.14.2

2 1

[alsa-devel] [PATCH 1/6] ASoC: Add snd_soc_of_put_dai_link_codecs() helper function
by Sylwester Nawrocki 07 Mar '18

07 Mar '18

The code for dereferencing device nodes in the 'codecs' array is moved to a separate function so we can avoid repeating the code in drivers. Signed-off-by: Sylwester Nawrocki <s.nawrocki(a)samsung.com> --- include/sound/soc.h | 1 + sound/soc/soc-core.c | 32 +++++++++++++++++++++++--------- 2 files changed, 24 insertions(+), 9 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index 6a11b0239f74..ad266d7e9553 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -1814,6 +1814,7 @@ int snd_soc_of_get_dai_name(struct device_node *of_node, int snd_soc_of_get_dai_link_codecs(struct device *dev, struct device_node *of_node, struct snd_soc_dai_link *dai_link); +void snd_soc_of_put_dai_link_codecs(struct snd_soc_dai_link *dai_link); int snd_soc_add_dai_link(struct snd_soc_card *card, struct snd_soc_dai_link *dai_link); diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 39876737c336..e815be0d5863 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -4358,6 +4358,26 @@ int snd_soc_of_get_dai_name(struct device_node *of_node, } EXPORT_SYMBOL_GPL(snd_soc_of_get_dai_name); +/* + * snd_soc_of_put_dai_link_codecs - Dereference device nodes in the codecs array + * @dai_link: DAI link + * + * Dereference device nodes acquired by snd_soc_of_get_dai_link_codecs(). + */ +void snd_soc_of_put_dai_link_codecs(struct snd_soc_dai_link *dai_link) +{ + struct snd_soc_dai_link_component *component = dai_link->codecs; + int index; + + for (index = 0; index < dai_link->num_codecs; index++, component++) { + if (!component->of_node) + break; + of_node_put(component->of_node); + component->of_node = NULL; + } +} +EXPORT_SYMBOL_GPL(snd_soc_of_put_dai_link_codecs); + /* * snd_soc_of_get_dai_link_codecs - Parse a list of CODECs in the devicetree * @dev: Card device @@ -4367,7 +4387,8 @@ EXPORT_SYMBOL_GPL(snd_soc_of_get_dai_name); * Builds an array of CODEC DAI components from the DAI link property * 'sound-dai'. * The array is set in the DAI link and the number of DAIs is set accordingly. - * The device nodes in the array (of_node) must be dereferenced by the caller. + * The device nodes in the array (of_node) must be dereferenced by calling + * snd_soc_of_put_dai_link_codecs() on @dai_link. * * Returns 0 for success */ @@ -4415,14 +4436,7 @@ int snd_soc_of_get_dai_link_codecs(struct device *dev, } return 0; err: - for (index = 0, component = dai_link->codecs; - index < dai_link->num_codecs; - index++, component++) { - if (!component->of_node) - break; - of_node_put(component->of_node); - component->of_node = NULL; - } + snd_soc_of_put_dai_link_codecs(dai_link); dai_link->codecs = NULL; dai_link->num_codecs = 0; return ret; -- 2.14.2

1 0

[alsa-devel] [RFC PATCH] ASoC: dapm: add support to pinctrl dapm
by srinivas.kandagatla＠linaro.org 07 Mar '18

07 Mar '18

From: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> Purpose of having pinctrl dapm is to dynamically put the pins in low power state when they are not actively used by the audio and saving power. Without this each driver has to set the pinctrl states, either during probe or dynamically depending on the callbacks received from ASoC core. This patch adds a single macro SND_SOC_DAPM_PINCTRL which takes, widget, active and sleep state names, and the dapm would use active and sleep state pinctrl names to put the pins in active and sleep state respectively. Signed-off-by: Srinivas Kandagatla <srinivas.kandagatla(a)linaro.org> --- Hi All, Purpose of this patch is to discuss the idea of having pinctrl part of dapm. As an example on Qualcomm SoCs there are multiple I2S Interfaces, each of these interfaces have there own pinctrls setting, It does not make sense for the machine/platform/componet driver to put them in default/active state durning probe/init as it would waste power. If we make these pinctrl part of the dapm sequence, these would only become active when the stream is active which would save power. This patch adds a single macro SND_SOC_DAPM_PINCTRL which takes, widget, active and sleep state names, and the dapm would use active and sleep state pinctrl names to put the pins in active and sleep state respectively. Thanks, Srini include/sound/soc-dapm.h | 16 ++++++++++++++++ sound/soc/soc-dapm.c | 44 ++++++++++++++++++++++++++++++++++++++++++++ 2 files changed, 60 insertions(+) diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index 344b96c206a3..a6ce2de4e20a 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -269,6 +269,13 @@ struct device; .reg = SND_SOC_NOPM, .shift = wdelay, .event = dapm_regulator_event, \ .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD, \ .on_val = wflags} +#define SND_SOC_DAPM_PINCTRL(wname, active, sleep) \ +{ .id = snd_soc_dapm_pinctrl, .name = wname, \ + .priv = (&(struct snd_soc_dapm_pinctrl_priv) \ + { .active_state = active, .sleep_state = sleep,}), \ + .reg = SND_SOC_NOPM, .event = dapm_pinctrl_event, \ + .event_flags = SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD } + /* dapm kcontrol types */ @@ -374,6 +381,8 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); int dapm_clock_event(struct snd_soc_dapm_widget *w, struct snd_kcontrol *kcontrol, int event); +int dapm_pinctrl_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event); /* dapm controls */ int snd_soc_dapm_put_volsw(struct snd_kcontrol *kcontrol, @@ -500,6 +509,7 @@ enum snd_soc_dapm_type { snd_soc_dapm_pre, /* machine specific pre widget - exec first */ snd_soc_dapm_post, /* machine specific post widget - exec last */ snd_soc_dapm_supply, /* power/clock supply */ + snd_soc_dapm_pinctrl, /* pinctrl */ snd_soc_dapm_regulator_supply, /* external regulator */ snd_soc_dapm_clock_supply, /* external clock */ snd_soc_dapm_aif_in, /* audio interface input */ @@ -581,6 +591,7 @@ struct snd_soc_dapm_widget { void *priv; /* widget specific data */ struct regulator *regulator; /* attached regulator */ + struct pinctrl *pinctrl; /* attached pinctrl */ const struct snd_soc_pcm_stream *params; /* params for dai links */ unsigned int num_params; /* number of params for dai links */ unsigned int params_select; /* currently selected param for dai link */ @@ -683,6 +694,11 @@ struct snd_soc_dapm_stats { int neighbour_checks; }; +struct snd_soc_dapm_pinctrl_priv { + const char *active_state; + const char *sleep_state; +}; + /** * snd_soc_dapm_init_bias_level() - Initialize DAPM bias level * @dapm: The DAPM context to initialize diff --git a/sound/soc/soc-dapm.c b/sound/soc/soc-dapm.c index 40085a4c8f92..babff1cd2849 100644 --- a/sound/soc/soc-dapm.c +++ b/sound/soc/soc-dapm.c @@ -35,6 +35,7 @@ #include <linux/debugfs.h> #include <linux/pm_runtime.h> #include <linux/regulator/consumer.h> +#include <linux/pinctrl/consumer.h> #include <linux/clk.h> #include <linux/slab.h> #include <sound/core.h> @@ -72,6 +73,7 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, static int dapm_up_seq[] = { [snd_soc_dapm_pre] = 0, [snd_soc_dapm_regulator_supply] = 1, + [snd_soc_dapm_pinctrl] = 1, [snd_soc_dapm_clock_supply] = 1, [snd_soc_dapm_supply] = 2, [snd_soc_dapm_micbias] = 3, @@ -119,6 +121,7 @@ static int dapm_down_seq[] = { [snd_soc_dapm_dai_link] = 11, [snd_soc_dapm_supply] = 12, [snd_soc_dapm_clock_supply] = 13, + [snd_soc_dapm_pinctrl] = 13, [snd_soc_dapm_regulator_supply] = 13, [snd_soc_dapm_post] = 14, }; @@ -1287,6 +1290,33 @@ int dapm_regulator_event(struct snd_soc_dapm_widget *w, } EXPORT_SYMBOL_GPL(dapm_regulator_event); +/* + * Handler for pinctrl widget. + */ +int dapm_pinctrl_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_dapm_pinctrl_priv *priv = w->priv; + struct pinctrl *p = w->pinctrl; + struct pinctrl_state *s; + + if (!p || !priv) + return -EIO; + + soc_dapm_async_complete(w->dapm); + + if (SND_SOC_DAPM_EVENT_ON(event)) + s = pinctrl_lookup_state(p, priv->active_state); + else + s = pinctrl_lookup_state(p, priv->sleep_state); + + if (IS_ERR(s)) + return PTR_ERR(s); + + return pinctrl_select_state(p, s); +} +EXPORT_SYMBOL_GPL(dapm_pinctrl_event); + /* * Handler for clock supply widget. */ @@ -1900,6 +1930,7 @@ static int dapm_power_widgets(struct snd_soc_card *card, int event) break; case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_pinctrl: case snd_soc_dapm_clock_supply: case snd_soc_dapm_micbias: if (d->target_bias_level < SND_SOC_BIAS_STANDBY) @@ -2313,6 +2344,7 @@ static ssize_t dapm_widget_show_component(struct snd_soc_component *cmpnt, case snd_soc_dapm_mixer_named_ctl: case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_pinctrl: case snd_soc_dapm_clock_supply: if (w->name) count += sprintf(buf + count, "%s: %s\n", @@ -3462,6 +3494,17 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, w->name, ret); } break; + case snd_soc_dapm_pinctrl: + w->pinctrl = devm_pinctrl_get(dapm->dev); + if (IS_ERR_OR_NULL(w->pinctrl)) { + ret = PTR_ERR(w->pinctrl); + if (ret == -EPROBE_DEFER) + return ERR_PTR(ret); + dev_err(dapm->dev, "ASoC: Failed to request %s: %d\n", + w->name, ret); + return NULL; + } + break; case snd_soc_dapm_clock_supply: #ifdef CONFIG_CLKDEV_LOOKUP w->clk = devm_clk_get(dapm->dev, w->name); @@ -3541,6 +3584,7 @@ snd_soc_dapm_new_control_unlocked(struct snd_soc_dapm_context *dapm, break; case snd_soc_dapm_supply: case snd_soc_dapm_regulator_supply: + case snd_soc_dapm_pinctrl: case snd_soc_dapm_clock_supply: case snd_soc_dapm_kcontrol: w->is_supply = 1; -- 2.15.1

3 4

[alsa-devel] [PATCH] ASoC: dt-bindings: wm8524: Remove unit address
by Fabio Estevam 07 Mar '18

07 Mar '18

WM8524 does not use I2C nor SPI, hence no 'reg' property is used, so remove the unit address from the bindings example to conform. This avoids users getting dtc warnings when they copy the binding example to a real dts file. Signed-off-by: Fabio Estevam <festevam(a)gmail.com> --- Documentation/devicetree/bindings/sound/wm8524.txt | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/Documentation/devicetree/bindings/sound/wm8524.txt b/Documentation/devicetree/bindings/sound/wm8524.txt index 20c6200..0f05535 100644 --- a/Documentation/devicetree/bindings/sound/wm8524.txt +++ b/Documentation/devicetree/bindings/sound/wm8524.txt @@ -10,7 +10,7 @@ Required properties: Example: -codec: wm8524@0 { +codec: wm8524 { compatible = "wlf,wm8524"; wlf,mute-gpios = <&gpio1 8 GPIO_ACTIVE_LOW>; }; -- 2.7.4

2 1

[alsa-devel] [PATCH v2 00/32] ASoC: rt5651: jack-detect fixes and improvements
by Hans de Goede 07 Mar '18

07 Mar '18

Hi All, Here is v2 of my rt5651 codec and Intel machine-driver series which mainly consists of various jack-detect related fixes and improvements. As discussed in the review of v1 I've made the following changes: 1) Get rid of platform_data and use device-properties instead 2) Split up a couple of the micbias / OVCD / jack-detect patches into smaller patches, as they were doing too much in a single patch 3) Improved commit messages 4) Concentrate the knowledge that SSP0 is 16 bits on BYTCR in the fixup function and extract the sample-format from the hw_params in other places Regards, Hans

4 64

[alsa-devel] [PATCH] slimbus: core: use put_device() instead of kfree()
by Arvind Yadav 07 Mar '18

07 Mar '18

Never directly free @dev after calling device_register(), even if it returned an error! Always use put_device() to give up the reference initialized. Signed-off-by: Arvind Yadav <arvind.yadav.cs(a)gmail.com> --- drivers/slimbus/core.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/slimbus/core.c b/drivers/slimbus/core.c index 4988a8f..7ddfc67 100644 --- a/drivers/slimbus/core.c +++ b/drivers/slimbus/core.c @@ -141,7 +141,7 @@ static struct slim_device *slim_alloc_device(struct slim_controller *ctrl, sbdev->e_addr = *eaddr; ret = slim_add_device(ctrl, sbdev, node); if (ret) { - kfree(sbdev); + put_device(&sbdev->dev); return NULL; } -- 1.9.1

2 1

Re: [alsa-devel] Dell new AIO platform
by Hui Wang 07 Mar '18

07 Mar '18

This is the alsa-info.txt for the machine of zuma-p (the machine kailang sent the patch for): https://pastebin.ubuntu.com/p/V4QgWWY5Nj/ And we have a similar bug with this one, but it is not for output, it is for input. On one of Dell desktops, there is no internal microphone, the only input devices are "Headphone Mic=0x1a" and "Headset Mic=0x19", so the imux_pins[spec->gen.cur_mux[0]] is headphone mic or headset mic, let us suppose it is headset mic, and we are unlucky that the auto-mute is disabled, and then unplug a headset from the jack, the audio is now in unplugged mode (alc_headset_mode_unplugged(codec);), then next time we plug the headset into the jack, the "what-did-you-plug" dialogue will pop up, we select headset-mic, since the imux_pins[spec->gen.cur_mux[0]] is already the headset mic (not internal microphone since there is no internal microphone on this machine), the alc_update_headset_mode() will not be called and then alc_determine_headset_type(), alc_headset_mode_ctia() and alc_headset_mode_omtp() will not be called, as a result the headset mic can't record sound. So far, I have no idea how to fix this problem. And the alsa-info.txt for my machine is as below shown: https://pastebin.ubuntu.com/p/SNMNjbRgnT/ Regards, Hui. On 2018年03月06日 17:13, Kailang wrote: > > >> -----Original Message----- >> From: Takashi Iwai [mailto:tiwai@suse.de] >> Sent: Tuesday, March 06, 2018 5:10 PM >> To: Kailang >> Cc: Hui Wang; (alsa-devel(a)alsa-project.org)" >> Subject: Re: Dell new AIO platform >> >> On Tue, 06 Mar 2018 10:03:38 +0100, >> Kailang wrote: >>> >>> >>>> -----Original Message----- >>>> From: Takashi Iwai [mailto:tiwai@suse.de] >>>> Sent: Tuesday, March 06, 2018 4:49 PM >>>> To: Kailang >>>> Cc: (alsa-devel(a)alsa-project.org) >>>> Subject: Re: Dell new AIO platform >>>> >>>> On Tue, 06 Mar 2018 08:54:39 +0100, >>>> Kailang wrote: >>>>> Hi Takashi, >>>>> >>>>> This platform was only one phone Jack. >>>>> Add dummy lineout verb to fix automute mode disable. >>>>> This just the workarround. >>>> I'm not sure why this phantom line pin is needed. >>>> What happens without that? >>> Looks like I find the root cause for this issue, since >> there is only one output device (headphone）， so the Auto-mute >> is not supported on this machine, as a result, when users >> plug headphone to the jack, audomute_hook or hp_automute_hook >> will not be called, then the alc_update_headset_mode() will >> not be called, and alc_headset_mode_default() will not be >> called. the headset jack is always in the >> alc_headset_mode_unplugged() state. >> >> OK, now understood. >> >> I can take the patch as a tentative workaround. >> But for the proper fix, could you give alsa-info.sh output? > I had no machine in my site. > > Hi Hui Wang, > > Could you help to send alsa-info.sh output to Takashi? > >> >> thanks, >> >> Takashi >> >> ------Please consider the environment before printing this e-mail.

1 1

[alsa-devel] [PATCH] ALSA: hda/realtek - Fix dock line-out volume on Dell Precision 7520
by Takashi Iwai 06 Mar '18

06 Mar '18

The dock line-out pin (NID 0x17 of ALC3254 codec) on Dell Precision 7520 may route to three different DACs, 0x02, 0x03 and 0x06. The first two DACS have the volume amp controls while the last one doesn't. And unfortunately, the auto-parser assigns this pin to DAC3, resulting in the non-working volume control for the line out. Fix it by disabling the routing to DAC3 on the corresponding pin. Bugzilla: https://bugzilla.kernel.org/show_bug.cgi?id=199029 Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_realtek.c | 16 ++++++++++++++++ 1 file changed, 16 insertions(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index 85d9efe343d2..60f1b9524957 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -5274,6 +5274,16 @@ static void alc298_fixup_speaker_volume(struct hda_codec *codec, } } +/* disable DAC3 (0x06) selection on NID 0x17 as it has no volume amp control */ +static void alc295_fixup_disable_dac3(struct hda_codec *codec, + const struct hda_fixup *fix, int action) +{ + if (action == HDA_FIXUP_ACT_PRE_PROBE) { + hda_nid_t conn[2] = { 0x02, 0x03 }; + snd_hda_override_conn_list(codec, 0x17, 2, conn); + } +} + /* Hook to update amp GPIO4 for automute */ static void alc280_hp_gpio4_automute_hook(struct hda_codec *codec, struct hda_jack_callback *jack) @@ -5466,6 +5476,7 @@ enum { ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY, ALC255_FIXUP_DELL_SPK_NOISE, ALC225_FIXUP_DELL1_MIC_NO_PRESENCE, + ALC295_FIXUP_DISABLE_DAC3, ALC280_FIXUP_HP_HEADSET_MIC, ALC221_FIXUP_HP_FRONT_MIC, ALC292_FIXUP_TPT460, @@ -6200,6 +6211,10 @@ static const struct hda_fixup alc269_fixups[] = { .chained = true, .chain_id = ALC298_FIXUP_DELL_AIO_MIC_NO_PRESENCE, }, + [ALC295_FIXUP_DISABLE_DAC3] = { + .type = HDA_FIXUP_FUNC, + .v.func = alc295_fixup_disable_dac3, + }, [ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER] = { .type = HDA_FIXUP_PINS, .v.pins = (const struct hda_pintbl[]) { @@ -6390,6 +6405,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x1028, 0x0725, "Dell Inspiron 3162", ALC255_FIXUP_DELL_SPK_NOISE), SND_PCI_QUIRK(0x1028, 0x075b, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), SND_PCI_QUIRK(0x1028, 0x075d, "Dell AIO", ALC298_FIXUP_SPK_VOLUME), + SND_PCI_QUIRK(0x1028, 0x07b0, "Dell Precision 7520", ALC295_FIXUP_DISABLE_DAC3), SND_PCI_QUIRK(0x1028, 0x0798, "Dell Inspiron 17 7000 Gaming", ALC256_FIXUP_DELL_INSPIRON_7559_SUBWOOFER), SND_PCI_QUIRK(0x1028, 0x080c, "Dell WYSE", ALC225_FIXUP_DELL_WYSE_MIC_NO_PRESENCE), SND_PCI_QUIRK(0x1028, 0x082a, "Dell XPS 13 9360", ALC256_FIXUP_DELL_XPS_13_HEADPHONE_NOISE), -- 2.16.2

1 0

[alsa-devel] [PATCH] ALSA: hda/realtek: Limit mic boost on T480
by Benjamin Berg 06 Mar '18

06 Mar '18

The internal mic boost on the T480 is too high. Fix this by applying the ALC269_FIXUP_LIMIT_INT_MIC_BOOST fixup to the machine to limit the gain. Signed-off-by: Benjamin Berg <bberg(a)redhat.com> Tested-by: Benjamin Berg <bberg(a)redhat.com> --- sound/pci/hda/patch_realtek.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index ce28f7ce64e6..8467ce3db767 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -6510,6 +6510,7 @@ static const struct snd_pci_quirk alc269_fixup_tbl[] = { SND_PCI_QUIRK(0x17aa, 0x224b, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x224c, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), SND_PCI_QUIRK(0x17aa, 0x224d, "Thinkpad", ALC298_FIXUP_TPT470_DOCK), + SND_PCI_QUIRK(0x17aa, 0x225d, "Thinkpad T480", ALC269_FIXUP_LIMIT_INT_MIC_BOOST), SND_PCI_QUIRK(0x17aa, 0x30bb, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x30e2, "ThinkCentre AIO", ALC233_FIXUP_LENOVO_LINE2_MIC_HOTKEY), SND_PCI_QUIRK(0x17aa, 0x310c, "ThinkCentre Station", ALC294_FIXUP_LENOVO_MIC_LOCATION), -- 2.14.3

3 8

Re: [alsa-devel] Dell new AIO platform
by Hui Wang 06 Mar '18

06 Mar '18

On 2018年03月06日 17:13, Kailang wrote: > > >> -----Original Message----- >> From: Takashi Iwai [mailto:tiwai@suse.de] >> Sent: Tuesday, March 06, 2018 5:10 PM >> To: Kailang >> Cc: Hui Wang; (alsa-devel(a)alsa-project.org) >> Subject: Re: Dell new AIO platform >> >> On Tue, 06 Mar 2018 10:03:38 +0100, >> Kailang wrote: >>> >>> >>>> -----Original Message----- >>>> From: Takashi Iwai [mailto:tiwai@suse.de] >>>> Sent: Tuesday, March 06, 2018 4:49 PM >>>> To: Kailang >>>> Cc: (alsa-devel(a)alsa-project.org) >>>> Subject: Re: Dell new AIO platform >>>> >>>> On Tue, 06 Mar 2018 08:54:39 +0100, >>>> Kailang wrote: >>>>> Hi Takashi, >>>>> >>>>> This platform was only one phone Jack. >>>>> Add dummy lineout verb to fix automute mode disable. >>>>> This just the workarround. >>>> I'm not sure why this phantom line pin is needed. >>>> What happens without that? >>> Looks like I find the root cause for this issue, since >> there is only one output device (headphone）， so the Auto-mute >> is not supported on this machine, as a result, when users >> plug headphone to the jack, audomute_hook or hp_automute_hook >> will not be called, then the alc_update_headset_mode() will >> not be called, and alc_headset_mode_default() will not be >> called. the headset jack is always in the >> alc_headset_mode_unplugged() state. >> >> OK, now understood. >> >> I can take the patch as a tentative workaround. >> But for the proper fix, could you give alsa-info.sh output? > I had no machine in my site. > > Hi Hui Wang, > > Could you help to send alsa-info.sh output to Takashi? OK, I will collect the alsa-info.txt tomorrow. Regards, Hui. >> >> thanks, >> >> Takashi >> >> ------Please consider the environment before printing this e-mail.

1 0

Re: [alsa-devel] Dell new AIO platform
by Takashi Iwai 06 Mar '18

06 Mar '18

On Tue, 06 Mar 2018 10:24:55 +0100, Kailang wrote: > > > > > -----Original Message----- > > From: Takashi Iwai [mailto:tiwai@suse.de] > > Sent: Tuesday, March 06, 2018 5:16 PM > > To: Kailang > > Cc: Hui Wang; (alsa-devel(a)alsa-project.org) > > Subject: Re: Dell new AIO platform > > > > On Tue, 06 Mar 2018 10:13:16 +0100, > > Kailang wrote: > > > > > > > > > > > > > -----Original Message----- > > > > From: Takashi Iwai [mailto:tiwai@suse.de] > > > > Sent: Tuesday, March 06, 2018 5:10 PM > > > > To: Kailang > > > > Cc: Hui Wang; (alsa-devel(a)alsa-project.org) > > > > Subject: Re: Dell new AIO platform > > > > > > > > On Tue, 06 Mar 2018 10:03:38 +0100, > > > > Kailang wrote: > > > > > > > > > > > > > > > > > > > > > -----Original Message----- > > > > > > From: Takashi Iwai [mailto:tiwai@suse.de] > > > > > > Sent: Tuesday, March 06, 2018 4:49 PM > > > > > > To: Kailang > > > > > > Cc: (alsa-devel(a)alsa-project.org) > > > > > > Subject: Re: Dell new AIO platform > > > > > > > > > > > > On Tue, 06 Mar 2018 08:54:39 +0100, Kailang wrote: > > > > > > > > > > > > > > Hi Takashi, > > > > > > > > > > > > > > This platform was only one phone Jack. > > > > > > > Add dummy lineout verb to fix automute mode disable. > > > > > > > This just the workarround. > > > > > > > > > > > > I'm not sure why this phantom line pin is needed. > > > > > > What happens without that? > > > > > > > > > > Looks like I find the root cause for this issue, since > > > > there is only one output device (headphone）， so the > > Auto-mute is not > > > > supported on this machine, as a result, when users plug > > headphone to > > > > the jack, audomute_hook or hp_automute_hook will not be > > called, then > > > > the alc_update_headset_mode() will not be called, and > > > > alc_headset_mode_default() will not be called. the > > headset jack is > > > > always in the > > > > alc_headset_mode_unplugged() state. > > > > > > > > OK, now understood. > > > > > > > > I can take the patch as a tentative workaround. > > > > But for the proper fix, could you give alsa-info.sh output? > > > > > > I had no machine in my site. > > > > > > Hi Hui Wang, > > > > > > Could you help to send alsa-info.sh output to Takashi? > > > > Meanwhile, I found there is another quirk > > ALC274_FIXUP_DELL_AIO_LINEOUT_VERB, which looks like serving > > for a similar purpose, but with more stuff (e.g. bind-dac). > > > No, this issue was not the same. > ALC274_FIXUP_DELL_AIO_LINEOUT_VERB ==> this lineout was true pin. > Bios verb table can't fill this verb. So, it need driver to fill it. OK, I'll add more information about that in the changelog. thanks, Takashi

1 0

Re: [alsa-devel] Dell new AIO platform
by Takashi Iwai 06 Mar '18

06 Mar '18

On Tue, 06 Mar 2018 10:13:16 +0100, Kailang wrote: > > > > > -----Original Message----- > > From: Takashi Iwai [mailto:tiwai@suse.de] > > Sent: Tuesday, March 06, 2018 5:10 PM > > To: Kailang > > Cc: Hui Wang; (alsa-devel(a)alsa-project.org) > > Subject: Re: Dell new AIO platform > > > > On Tue, 06 Mar 2018 10:03:38 +0100, > > Kailang wrote: > > > > > > > > > > > > > -----Original Message----- > > > > From: Takashi Iwai [mailto:tiwai@suse.de] > > > > Sent: Tuesday, March 06, 2018 4:49 PM > > > > To: Kailang > > > > Cc: (alsa-devel(a)alsa-project.org) > > > > Subject: Re: Dell new AIO platform > > > > > > > > On Tue, 06 Mar 2018 08:54:39 +0100, > > > > Kailang wrote: > > > > > > > > > > Hi Takashi, > > > > > > > > > > This platform was only one phone Jack. > > > > > Add dummy lineout verb to fix automute mode disable. > > > > > This just the workarround. > > > > > > > > I'm not sure why this phantom line pin is needed. > > > > What happens without that? > > > > > > Looks like I find the root cause for this issue, since > > there is only one output device (headphone）， so the Auto-mute > > is not supported on this machine, as a result, when users > > plug headphone to the jack, audomute_hook or hp_automute_hook > > will not be called, then the alc_update_headset_mode() will > > not be called, and alc_headset_mode_default() will not be > > called. the headset jack is always in the > > alc_headset_mode_unplugged() state. > > > > OK, now understood. > > > > I can take the patch as a tentative workaround. > > But for the proper fix, could you give alsa-info.sh output? > > I had no machine in my site. > > Hi Hui Wang, > > Could you help to send alsa-info.sh output to Takashi? Meanwhile, I found there is another quirk ALC274_FIXUP_DELL_AIO_LINEOUT_VERB, which looks like serving for a similar purpose, but with more stuff (e.g. bind-dac). thanks, Takashi

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Re: [alsa-devel] Dell new AIO platform
by Takashi Iwai 06 Mar '18

06 Mar '18

On Tue, 06 Mar 2018 10:03:38 +0100, Kailang wrote: > > > > > -----Original Message----- > > From: Takashi Iwai [mailto:tiwai@suse.de] > > Sent: Tuesday, March 06, 2018 4:49 PM > > To: Kailang > > Cc: (alsa-devel(a)alsa-project.org) > > Subject: Re: Dell new AIO platform > > > > On Tue, 06 Mar 2018 08:54:39 +0100, > > Kailang wrote: > > > > > > Hi Takashi, > > > > > > This platform was only one phone Jack. > > > Add dummy lineout verb to fix automute mode disable. > > > This just the workarround. > > > > I'm not sure why this phantom line pin is needed. > > What happens without that? > > Looks like I find the root cause for this issue, since there is only one output device (headphone）， so the Auto-mute is not supported on this machine, as a result, when users plug headphone to the jack, audomute_hook or hp_automute_hook will not be called, then the alc_update_headset_mode() will not be called, and alc_headset_mode_default() will not be called. the headset jack is always in the alc_headset_mode_unplugged() state. OK, now understood. I can take the patch as a tentative workaround. But for the proper fix, could you give alsa-info.sh output? thanks, Takashi

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[alsa-devel] Dell New WYSE enable headset mode
by Kailang 06 Mar '18

06 Mar '18

Hi Takashi, Attach was new Dell WYSE platform. This patch will enable headset mode for it. BR, Kailang

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Re: [alsa-devel] Dell new AIO platform
by Takashi Iwai 06 Mar '18

06 Mar '18

On Tue, 06 Mar 2018 08:54:39 +0100, Kailang wrote: > > Hi Takashi, > > This platform was only one phone Jack. > Add dummy lineout verb to fix automute mode disable. > This just the workarround. I'm not sure why this phantom line pin is needed. What happens without that? thanks, Takashi > > This patch was test by Hui Wang@canonical > > Many Thanks. > > BR, > Kailang > [1.2 <text/html; us-ascii (quoted-printable)>] > > [2 0000-dell-0x0873-headset-mode-support.patch <application/octet-stream (base64)>] >

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[alsa-devel] FW: Dell new AIO platform
by Kailang 06 Mar '18

06 Mar '18

Hi Takashi, This platform was only one phone Jack. Add dummy lineout verb to fix automute mode disable. This just the workarround. This patch was test by Hui Wang@canonical Many Thanks. BR, Kailang

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[alsa-devel] [PATCH] ALSA: core: Whitespace cleanup
by Ladislav Michl 05 Mar '18

05 Mar '18

Remove whitespace on blank line, whitespace before newline and space before tab. Signed-off-by: Ladislav Michl <ladis(a)linux-mips.org> --- Hi, this is just a side product of reading ALSA code to get better clue how core handles codecs. As I enable 'let c_space_errors=1' in vimrc, here are 'red spots' corrected. Feel free to ignore, apply when you find it usefull :) No functionsl changes. regards, ladis include/sound/pcm.h | 22 ++++++------ sound/core/control.c | 6 ++-- sound/core/control_compat.c | 2 +- sound/core/device.c | 4 +-- sound/core/hwdep.c | 16 ++++----- sound/core/init.c | 17 +++++---- sound/core/memalloc.c | 2 +- sound/core/memory.c | 2 +- sound/core/pcm.c | 22 ++++++------ sound/core/pcm_compat.c | 2 +- sound/core/pcm_lib.c | 40 ++++++++++----------- sound/core/pcm_misc.c | 2 +- sound/core/pcm_native.c | 88 ++++++++++++++++++++++----------------------- sound/core/rawmidi.c | 18 +++++----- sound/core/seq_device.c | 2 +- sound/core/sgbuf.c | 2 +- sound/core/sound_oss.c | 10 +++--- sound/core/timer.c | 4 +-- sound/core/timer_compat.c | 2 +- 19 files changed, 131 insertions(+), 132 deletions(-) diff --git a/include/sound/pcm.h b/include/sound/pcm.h index e054c583d3b3..c5a3f8fa7e14 100644 --- a/include/sound/pcm.h +++ b/include/sound/pcm.h @@ -243,7 +243,7 @@ struct snd_pcm_hw_rule { }; struct snd_pcm_hw_constraints { - struct snd_mask masks[SNDRV_PCM_HW_PARAM_LAST_MASK - + struct snd_mask masks[SNDRV_PCM_HW_PARAM_LAST_MASK - SNDRV_PCM_HW_PARAM_FIRST_MASK + 1]; struct snd_interval intervals[SNDRV_PCM_HW_PARAM_LAST_INTERVAL - SNDRV_PCM_HW_PARAM_FIRST_INTERVAL + 1]; @@ -382,7 +382,7 @@ struct snd_pcm_runtime { /* -- SW params -- */ int tstamp_mode; /* mmap timestamp is updated */ - unsigned int period_step; + unsigned int period_step; snd_pcm_uframes_t start_threshold; snd_pcm_uframes_t stop_threshold; snd_pcm_uframes_t silence_threshold; /* Silence filling happens when @@ -400,7 +400,7 @@ struct snd_pcm_runtime { struct snd_pcm_mmap_control *control; /* -- locking / scheduling -- */ - snd_pcm_uframes_t twake; /* do transfer (!poll) wakeup if non-zero */ + snd_pcm_uframes_t twake; /* do transfer (!poll) wakeup if non-zero */ wait_queue_head_t sleep; /* poll sleep */ wait_queue_head_t tsleep; /* transfer sleep */ struct fasync_struct *fasync; @@ -417,7 +417,7 @@ struct snd_pcm_runtime { unsigned int timer_resolution; /* timer resolution */ int tstamp_type; /* timestamp type */ - /* -- DMA -- */ + /* -- DMA -- */ unsigned char *dma_area; /* DMA area */ dma_addr_t dma_addr; /* physical bus address (not accessible from main CPU) */ size_t dma_bytes; /* size of DMA area */ @@ -835,7 +835,7 @@ static inline int snd_pcm_capture_ready(struct snd_pcm_substream *substream) static inline int snd_pcm_playback_data(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - + if (runtime->stop_threshold >= runtime->boundary) return 1; return snd_pcm_playback_avail(runtime) < runtime->buffer_size; @@ -883,7 +883,7 @@ static inline int snd_pcm_capture_empty(struct snd_pcm_substream *substream) * The trigger_master mark is cleared at timestamp updates at the end * of trigger operations. */ -static inline void snd_pcm_trigger_done(struct snd_pcm_substream *substream, +static inline void snd_pcm_trigger_done(struct snd_pcm_substream *substream, struct snd_pcm_substream *master) { substream->runtime->trigger_master = master; @@ -998,7 +998,7 @@ int snd_pcm_hw_constraint_mask64(struct snd_pcm_runtime *runtime, snd_pcm_hw_par int snd_pcm_hw_constraint_minmax(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var, unsigned int min, unsigned int max); int snd_pcm_hw_constraint_integer(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var); -int snd_pcm_hw_constraint_list(struct snd_pcm_runtime *runtime, +int snd_pcm_hw_constraint_list(struct snd_pcm_runtime *runtime, unsigned int cond, snd_pcm_hw_param_t var, const struct snd_pcm_hw_constraint_list *l); @@ -1006,15 +1006,15 @@ int snd_pcm_hw_constraint_ranges(struct snd_pcm_runtime *runtime, unsigned int cond, snd_pcm_hw_param_t var, const struct snd_pcm_hw_constraint_ranges *r); -int snd_pcm_hw_constraint_ratnums(struct snd_pcm_runtime *runtime, +int snd_pcm_hw_constraint_ratnums(struct snd_pcm_runtime *runtime, unsigned int cond, snd_pcm_hw_param_t var, const struct snd_pcm_hw_constraint_ratnums *r); -int snd_pcm_hw_constraint_ratdens(struct snd_pcm_runtime *runtime, +int snd_pcm_hw_constraint_ratdens(struct snd_pcm_runtime *runtime, unsigned int cond, snd_pcm_hw_param_t var, const struct snd_pcm_hw_constraint_ratdens *r); -int snd_pcm_hw_constraint_msbits(struct snd_pcm_runtime *runtime, +int snd_pcm_hw_constraint_msbits(struct snd_pcm_runtime *runtime, unsigned int cond, unsigned int width, unsigned int msbits); @@ -1079,7 +1079,7 @@ void snd_pcm_set_ops(struct snd_pcm * pcm, int direction, const struct snd_pcm_ops *ops); void snd_pcm_set_sync(struct snd_pcm_substream *substream); int snd_pcm_lib_ioctl(struct snd_pcm_substream *substream, - unsigned int cmd, void *arg); + unsigned int cmd, void *arg); void snd_pcm_period_elapsed(struct snd_pcm_substream *substream); snd_pcm_sframes_t __snd_pcm_lib_xfer(struct snd_pcm_substream *substream, void *buf, bool interleaved, diff --git a/sound/core/control.c b/sound/core/control.c index 69734b0eafd0..a7929ae2a37b 100644 --- a/sound/core/control.c +++ b/sound/core/control.c @@ -98,7 +98,7 @@ static int snd_ctl_open(struct inode *inode, struct file *file) __error1: if (card) snd_card_unref(card); - return err; + return err; } static void snd_ctl_empty_read_queue(struct snd_ctl_file * ctl) @@ -327,7 +327,7 @@ static bool snd_ctl_remove_numid_conflict(struct snd_card *card, list_for_each_entry(kctl, &card->controls, list) { if (kctl->id.numid < card->last_numid + 1 + count && kctl->id.numid + kctl->count > card->last_numid + 1) { - card->last_numid = kctl->id.numid + kctl->count - 1; + card->last_numid = kctl->id.numid + kctl->count - 1; return true; } } @@ -1664,7 +1664,7 @@ static ssize_t snd_ctl_read(struct file *file, char __user *buffer, __end_lock: spin_unlock_irq(&ctl->read_lock); __end: - return result > 0 ? result : err; + return result > 0 ? result : err; } static __poll_t snd_ctl_poll(struct file *file, poll_table * wait) diff --git a/sound/core/control_compat.c b/sound/core/control_compat.c index a848836a5de0..738f4b7894f9 100644 --- a/sound/core/control_compat.c +++ b/sound/core/control_compat.c @@ -428,7 +428,7 @@ static int snd_ctl_elem_add_compat(struct snd_ctl_file *file, error: kfree(data); return err; -} +} enum { SNDRV_CTL_IOCTL_ELEM_LIST32 = _IOWR('U', 0x10, struct snd_ctl_elem_list32), diff --git a/sound/core/device.c b/sound/core/device.c index cb0e46f66cc9..69b4ff8324b1 100644 --- a/sound/core/device.c +++ b/sound/core/device.c @@ -145,7 +145,7 @@ EXPORT_SYMBOL_GPL(snd_device_disconnect); void snd_device_free(struct snd_card *card, void *device_data) { struct snd_device *dev; - + if (snd_BUG_ON(!card || !device_data)) return; dev = look_for_dev(card, device_data); @@ -205,7 +205,7 @@ int snd_device_register_all(struct snd_card *card) { struct snd_device *dev; int err; - + if (snd_BUG_ON(!card)) return -ENXIO; list_for_each_entry(dev, &card->devices, list) { diff --git a/sound/core/hwdep.c b/sound/core/hwdep.c index 26e71cf05f1e..6fc28fffb310 100644 --- a/sound/core/hwdep.c +++ b/sound/core/hwdep.c @@ -68,7 +68,7 @@ static ssize_t snd_hwdep_read(struct file * file, char __user *buf, struct snd_hwdep *hw = file->private_data; if (hw->ops.read) return hw->ops.read(hw, buf, count, offset); - return -ENXIO; + return -ENXIO; } static ssize_t snd_hwdep_write(struct file * file, const char __user *buf, @@ -77,7 +77,7 @@ static ssize_t snd_hwdep_write(struct file * file, const char __user *buf, struct snd_hwdep *hw = file->private_data; if (hw->ops.write) return hw->ops.write(hw, buf, count, offset); - return -ENXIO; + return -ENXIO; } static int snd_hwdep_open(struct inode *inode, struct file * file) @@ -189,10 +189,10 @@ static int snd_hwdep_info(struct snd_hwdep *hw, struct snd_hwdep_info __user *_info) { struct snd_hwdep_info info; - + memset(&info, 0, sizeof(info)); info.card = hw->card->number; - strlcpy(info.id, hw->id, sizeof(info.id)); + strlcpy(info.id, hw->id, sizeof(info.id)); strlcpy(info.name, hw->name, sizeof(info.name)); info.iface = hw->iface; if (copy_to_user(_info, &info, sizeof(info))) @@ -205,7 +205,7 @@ static int snd_hwdep_dsp_status(struct snd_hwdep *hw, { struct snd_hwdep_dsp_status info; int err; - + if (! hw->ops.dsp_status) return -ENXIO; memset(&info, 0, sizeof(info)); @@ -222,7 +222,7 @@ static int snd_hwdep_dsp_load(struct snd_hwdep *hw, { struct snd_hwdep_dsp_image info; int err; - + if (! hw->ops.dsp_load) return -ENXIO; memset(&info, 0, sizeof(info)); @@ -333,9 +333,9 @@ static int snd_hwdep_control_ioctl(struct snd_card *card, static const struct file_operations snd_hwdep_f_ops = { - .owner = THIS_MODULE, + .owner = THIS_MODULE, .llseek = snd_hwdep_llseek, - .read = snd_hwdep_read, + .read = snd_hwdep_read, .write = snd_hwdep_write, .open = snd_hwdep_open, .release = snd_hwdep_release, diff --git a/sound/core/init.c b/sound/core/init.c index 79b4df1c1dc7..f11c9909f9ff 100644 --- a/sound/core/init.c +++ b/sound/core/init.c @@ -289,7 +289,7 @@ int snd_card_new(struct device *parent, int idx, const char *xid, snd_device_free_all(card); __error: put_device(&card->card_dev); - return err; + return err; } EXPORT_SYMBOL(snd_card_new); @@ -367,9 +367,9 @@ static int snd_disconnect_fasync(int fd, struct file *file, int on) static const struct file_operations snd_shutdown_f_ops = { - .owner = THIS_MODULE, + .owner = THIS_MODULE, .llseek = snd_disconnect_llseek, - .read = snd_disconnect_read, + .read = snd_disconnect_read, .write = snd_disconnect_write, .release = snd_disconnect_release, .poll = snd_disconnect_poll, @@ -412,9 +412,8 @@ int snd_card_disconnect(struct snd_card *card) snd_cards[card->number] = NULL; clear_bit(card->number, snd_cards_lock); mutex_unlock(&snd_card_mutex); - + /* phase 2: replace file->f_op with special dummy operations */ - spin_lock(&card->files_lock); list_for_each_entry(mfile, &card->files_list, list) { /* it's critical part, use endless loop */ @@ -428,7 +427,7 @@ int snd_card_disconnect(struct snd_card *card) mfile->file->f_op = &snd_shutdown_f_ops; fops_get(mfile->file->f_op); } - spin_unlock(&card->files_lock); + spin_unlock(&card->files_lock); /* phase 3: notify all connected devices about disconnection */ /* at this point, they cannot respond to any calls except release() */ @@ -449,7 +448,7 @@ int snd_card_disconnect(struct snd_card *card) #ifdef CONFIG_PM wake_up(&card->power_sleep); #endif - return 0; + return 0; } EXPORT_SYMBOL(snd_card_disconnect); @@ -604,7 +603,7 @@ static void snd_card_set_id_no_lock(struct snd_card *card, const char *src, int len, loops; bool is_default = false; char *id; - + copy_valid_id_string(card, src, nid); id = card->id; @@ -901,7 +900,7 @@ int __init snd_card_info_init(void) * * Return: Zero otherwise a negative error code. */ - + int snd_component_add(struct snd_card *card, const char *component) { char *ptr; diff --git a/sound/core/memalloc.c b/sound/core/memalloc.c index 7f89d3c79a4b..f347b0310abb 100644 --- a/sound/core/memalloc.c +++ b/sound/core/memalloc.c @@ -1,7 +1,7 @@ /* * Copyright (c) by Jaroslav Kysela <perex(a)perex.cz> * Takashi Iwai <tiwai(a)suse.de> - * + * * Generic memory allocators * * diff --git a/sound/core/memory.c b/sound/core/memory.c index 19c9ea90d9bf..b6b36df9d921 100644 --- a/sound/core/memory.c +++ b/sound/core/memory.c @@ -1,6 +1,6 @@ /* * Copyright (c) by Jaroslav Kysela <perex(a)perex.cz> - * + * * Misc memory accessors * * diff --git a/sound/core/pcm.c b/sound/core/pcm.c index 09ee8c6b9f75..38141a0e815a 100644 --- a/sound/core/pcm.c +++ b/sound/core/pcm.c @@ -163,7 +163,7 @@ static int snd_pcm_control_ioctl(struct snd_card *card, case SNDRV_CTL_IOCTL_PCM_PREFER_SUBDEVICE: { int val; - + if (get_user(val, (int __user *)arg)) return -EFAULT; control->preferred_subdevice[SND_CTL_SUBDEV_PCM] = val; @@ -247,7 +247,7 @@ EXPORT_SYMBOL_GPL(snd_pcm_format_name); #define TSTAMP(v) [SNDRV_PCM_TSTAMP_##v] = #v #define ACCESS(v) [SNDRV_PCM_ACCESS_##v] = #v #define START(v) [SNDRV_PCM_START_##v] = #v -#define SUBFORMAT(v) [SNDRV_PCM_SUBFORMAT_##v] = #v +#define SUBFORMAT(v) [SNDRV_PCM_SUBFORMAT_##v] = #v static char *snd_pcm_stream_names[] = { STREAM(PLAYBACK), @@ -266,7 +266,7 @@ static char *snd_pcm_state_names[] = { }; static char *snd_pcm_access_names[] = { - ACCESS(MMAP_INTERLEAVED), + ACCESS(MMAP_INTERLEAVED), ACCESS(MMAP_NONINTERLEAVED), ACCESS(MMAP_COMPLEX), ACCESS(RW_INTERLEAVED), @@ -274,7 +274,7 @@ static char *snd_pcm_access_names[] = { }; static char *snd_pcm_subformat_names[] = { - SUBFORMAT(STD), + SUBFORMAT(STD), }; static char *snd_pcm_tstamp_mode_names[] = { @@ -404,14 +404,14 @@ static void snd_pcm_substream_proc_hw_params_read(struct snd_info_entry *entry, snd_iprintf(buffer, "access: %s\n", snd_pcm_access_name(runtime->access)); snd_iprintf(buffer, "format: %s\n", snd_pcm_format_name(runtime->format)); snd_iprintf(buffer, "subformat: %s\n", snd_pcm_subformat_name(runtime->subformat)); - snd_iprintf(buffer, "channels: %u\n", runtime->channels); - snd_iprintf(buffer, "rate: %u (%u/%u)\n", runtime->rate, runtime->rate_num, runtime->rate_den); - snd_iprintf(buffer, "period_size: %lu\n", runtime->period_size); - snd_iprintf(buffer, "buffer_size: %lu\n", runtime->buffer_size); + snd_iprintf(buffer, "channels: %u\n", runtime->channels); + snd_iprintf(buffer, "rate: %u (%u/%u)\n", runtime->rate, runtime->rate_num, runtime->rate_den); + snd_iprintf(buffer, "period_size: %lu\n", runtime->period_size); + snd_iprintf(buffer, "buffer_size: %lu\n", runtime->buffer_size); #if IS_ENABLED(CONFIG_SND_PCM_OSS) if (substream->oss.oss) { snd_iprintf(buffer, "OSS format: %s\n", snd_pcm_oss_format_name(runtime->oss.format)); - snd_iprintf(buffer, "OSS channels: %u\n", runtime->oss.channels); + snd_iprintf(buffer, "OSS channels: %u\n", runtime->oss.channels); snd_iprintf(buffer, "OSS rate: %u\n", runtime->oss.rate); snd_iprintf(buffer, "OSS period bytes: %lu\n", (unsigned long)runtime->oss.period_bytes); snd_iprintf(buffer, "OSS periods: %u\n", runtime->oss.periods); @@ -523,7 +523,7 @@ static int snd_pcm_stream_proc_init(struct snd_pcm_str *pstr) struct snd_info_entry *entry; char name[16]; - sprintf(name, "pcm%i%c", pcm->device, + sprintf(name, "pcm%i%c", pcm->device, pstr->stream == SNDRV_PCM_STREAM_PLAYBACK ? 'p' : 'c'); entry = snd_info_create_card_entry(pcm->card, name, pcm->card->proc_root); @@ -763,7 +763,7 @@ int snd_pcm_new_stream(struct snd_pcm *pcm, int stream, int substream_count) prev = substream; } return 0; -} +} EXPORT_SYMBOL(snd_pcm_new_stream); static int _snd_pcm_new(struct snd_card *card, const char *id, int device, diff --git a/sound/core/pcm_compat.c b/sound/core/pcm_compat.c index b719d0bd833e..fce376c74a49 100644 --- a/sound/core/pcm_compat.c +++ b/sound/core/pcm_compat.c @@ -314,7 +314,7 @@ static int snd_pcm_status_user_x32(struct snd_pcm_substream *substream, /* both for HW_PARAMS and HW_REFINE */ static int snd_pcm_ioctl_hw_params_compat(struct snd_pcm_substream *substream, - int refine, + int refine, struct snd_pcm_hw_params32 __user *data32) { struct snd_pcm_hw_params *data; diff --git a/sound/core/pcm_lib.c b/sound/core/pcm_lib.c index a83152e7d387..be9efc98d6b7 100644 --- a/sound/core/pcm_lib.c +++ b/sound/core/pcm_lib.c @@ -489,7 +489,7 @@ void snd_pcm_set_ops(struct snd_pcm *pcm, int direction, { struct snd_pcm_str *stream = &pcm->streams[direction]; struct snd_pcm_substream *substream; - + for (substream = stream->substream; substream != NULL; substream = substream->next) substream->ops = ops; } @@ -504,7 +504,7 @@ EXPORT_SYMBOL(snd_pcm_set_ops); void snd_pcm_set_sync(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; - + runtime->sync.id32[0] = substream->pcm->card->number; runtime->sync.id32[1] = -1; runtime->sync.id32[2] = -1; @@ -516,7 +516,7 @@ EXPORT_SYMBOL(snd_pcm_set_sync); * Standard ioctl routine */ -static inline unsigned int div32(unsigned int a, unsigned int b, +static inline unsigned int div32(unsigned int a, unsigned int b, unsigned int *r) { if (b == 0) { @@ -774,7 +774,7 @@ void snd_interval_mulkdiv(const struct snd_interval *a, unsigned int k, /** * snd_interval_ratnum - refine the interval value * @i: interval to refine - * @rats_count: number of ratnum_t + * @rats_count: number of ratnum_t * @rats: ratnum_t array * @nump: pointer to store the resultant numerator * @denp: pointer to store the resultant denominator @@ -829,7 +829,7 @@ int snd_interval_ratnum(struct snd_interval *i, } t.min = div_down(best_num, best_den); t.openmin = !!(best_num % best_den); - + result_num = best_num; result_diff = best_diff; result_den = best_den; @@ -941,7 +941,7 @@ static int snd_interval_ratden(struct snd_interval *i, } t.min = div_down(best_num, best_den); t.openmin = !!(best_num % best_den); - + best_num = best_den = best_diff = 0; for (k = 0; k < rats_count; ++k) { unsigned int num; @@ -1233,7 +1233,7 @@ EXPORT_SYMBOL(snd_pcm_hw_constraint_integer); * @var: hw_params variable to apply the range * @min: the minimal value * @max: the maximal value - * + * * Apply the min/max range constraint to an interval parameter. * * Return: Positive if the value is changed, zero if it's not changed, or a @@ -1257,7 +1257,7 @@ static int snd_pcm_hw_rule_list(struct snd_pcm_hw_params *params, { struct snd_pcm_hw_constraint_list *list = rule->private; return snd_interval_list(hw_param_interval(params, rule->var), list->count, list->list, list->mask); -} +} /** @@ -1266,7 +1266,7 @@ static int snd_pcm_hw_rule_list(struct snd_pcm_hw_params *params, * @cond: condition bits * @var: hw_params variable to apply the list constraint * @l: list - * + * * Apply the list of constraints to an interval parameter. * * Return: Zero if successful, or a negative error code on failure. @@ -1337,7 +1337,7 @@ static int snd_pcm_hw_rule_ratnums(struct snd_pcm_hw_params *params, * * Return: Zero if successful, or a negative error code on failure. */ -int snd_pcm_hw_constraint_ratnums(struct snd_pcm_runtime *runtime, +int snd_pcm_hw_constraint_ratnums(struct snd_pcm_runtime *runtime, unsigned int cond, snd_pcm_hw_param_t var, const struct snd_pcm_hw_constraint_ratnums *r) @@ -1371,7 +1371,7 @@ static int snd_pcm_hw_rule_ratdens(struct snd_pcm_hw_params *params, * * Return: Zero if successful, or a negative error code on failure. */ -int snd_pcm_hw_constraint_ratdens(struct snd_pcm_runtime *runtime, +int snd_pcm_hw_constraint_ratdens(struct snd_pcm_runtime *runtime, unsigned int cond, snd_pcm_hw_param_t var, const struct snd_pcm_hw_constraint_ratdens *r) @@ -1415,7 +1415,7 @@ static int snd_pcm_hw_rule_msbits(struct snd_pcm_hw_params *params, * * Return: Zero if successful, or a negative error code on failure. */ -int snd_pcm_hw_constraint_msbits(struct snd_pcm_runtime *runtime, +int snd_pcm_hw_constraint_msbits(struct snd_pcm_runtime *runtime, unsigned int cond, unsigned int width, unsigned int msbits) @@ -1449,7 +1449,7 @@ int snd_pcm_hw_constraint_step(struct snd_pcm_runtime *runtime, snd_pcm_hw_param_t var, unsigned long step) { - return snd_pcm_hw_rule_add(runtime, cond, var, + return snd_pcm_hw_rule_add(runtime, cond, var, snd_pcm_hw_rule_step, (void *) step, var, -1); } @@ -1465,7 +1465,7 @@ static int snd_pcm_hw_rule_pow2(struct snd_pcm_hw_params *params, struct snd_pcm }; return snd_interval_list(hw_param_interval(params, rule->var), ARRAY_SIZE(pow2_sizes), pow2_sizes, 0); -} +} /** * snd_pcm_hw_constraint_pow2 - add a hw constraint power-of-2 rule @@ -1479,7 +1479,7 @@ int snd_pcm_hw_constraint_pow2(struct snd_pcm_runtime *runtime, unsigned int cond, snd_pcm_hw_param_t var) { - return snd_pcm_hw_rule_add(runtime, cond, var, + return snd_pcm_hw_rule_add(runtime, cond, var, snd_pcm_hw_rule_pow2, NULL, var, -1); } @@ -1622,8 +1622,8 @@ static int _snd_pcm_hw_param_first(struct snd_pcm_hw_params *params, * * Return: The minimum, or a negative error code on failure. */ -int snd_pcm_hw_param_first(struct snd_pcm_substream *pcm, - struct snd_pcm_hw_params *params, +int snd_pcm_hw_param_first(struct snd_pcm_substream *pcm, + struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var, int *dir) { int changed = _snd_pcm_hw_param_first(params, var); @@ -1668,7 +1668,7 @@ static int _snd_pcm_hw_param_last(struct snd_pcm_hw_params *params, * * Return: The maximum, or a negative error code on failure. */ -int snd_pcm_hw_param_last(struct snd_pcm_substream *pcm, +int snd_pcm_hw_param_last(struct snd_pcm_substream *pcm, struct snd_pcm_hw_params *params, snd_pcm_hw_param_t var, int *dir) { @@ -1882,7 +1882,7 @@ static int wait_for_avail(struct snd_pcm_substream *substream, case SNDRV_PCM_STATE_DRAINING: if (is_playback) err = -EPIPE; - else + else avail = 0; /* indicate draining */ goto _endloop; case SNDRV_PCM_STATE_OPEN: @@ -1907,7 +1907,7 @@ static int wait_for_avail(struct snd_pcm_substream *substream, *availp = avail; return err; } - + typedef int (*pcm_transfer_f)(struct snd_pcm_substream *substream, int channel, unsigned long hwoff, void *buf, unsigned long bytes); diff --git a/sound/core/pcm_misc.c b/sound/core/pcm_misc.c index c4eb561d2008..29f5cad1ee28 100644 --- a/sound/core/pcm_misc.c +++ b/sound/core/pcm_misc.c @@ -18,7 +18,7 @@ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA * */ - + #include <linux/time.h> #include <linux/export.h> #include <sound/core.h> diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 77ba50ddcf9e..6fb1b6bfd585 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -923,7 +923,7 @@ int snd_pcm_status(struct snd_pcm_substream *substream, runtime->avail_max = 0; runtime->overrange = 0; _end: - snd_pcm_stream_unlock_irq(substream); + snd_pcm_stream_unlock_irq(substream); return 0; } @@ -956,7 +956,7 @@ static int snd_pcm_channel_info(struct snd_pcm_substream *substream, { struct snd_pcm_runtime *runtime; unsigned int channel; - + channel = info->channel; runtime = substream->runtime; snd_pcm_stream_lock_irq(substream); @@ -977,7 +977,7 @@ static int snd_pcm_channel_info_user(struct snd_pcm_substream *substream, { struct snd_pcm_channel_info info; int res; - + if (copy_from_user(&info, _info, sizeof(info))) return -EFAULT; res = snd_pcm_channel_info(substream, &info); @@ -1077,7 +1077,7 @@ static int snd_pcm_action_single(const struct action_ops *ops, int state) { int res; - + res = ops->pre_action(substream, state); if (res < 0) return res; @@ -1187,7 +1187,7 @@ static void snd_pcm_post_start(struct snd_pcm_substream *substream, int state) struct snd_pcm_runtime *runtime = substream->runtime; snd_pcm_trigger_tstamp(substream); runtime->hw_ptr_jiffies = jiffies; - runtime->hw_ptr_buffer_jiffies = (runtime->buffer_size * HZ) / + runtime->hw_ptr_buffer_jiffies = (runtime->buffer_size * HZ) / runtime->rate; runtime->status->state = state; if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK && @@ -1875,7 +1875,7 @@ static int snd_pcm_drop(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime; int result = 0; - + if (PCM_RUNTIME_CHECK(substream)) return -ENXIO; runtime = substream->runtime; @@ -2111,7 +2111,7 @@ static int snd_pcm_hw_rule_rate(struct snd_pcm_hw_params *params, return snd_interval_list(hw_param_interval(params, rule->var), snd_pcm_known_rates.count, snd_pcm_known_rates.list, hw->rates); -} +} static int snd_pcm_hw_rule_buffer_bytes_max(struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule) @@ -2124,7 +2124,7 @@ static int snd_pcm_hw_rule_buffer_bytes_max(struct snd_pcm_hw_params *params, t.openmax = 0; t.integer = 1; return snd_interval_refine(hw_param_interval(params, rule->var), &t); -} +} int snd_pcm_hw_constraints_init(struct snd_pcm_substream *substream) { @@ -2151,98 +2151,98 @@ int snd_pcm_hw_constraints_init(struct snd_pcm_substream *substream) SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, snd_pcm_hw_rule_sample_bits, NULL, - SNDRV_PCM_HW_PARAM_FORMAT, + SNDRV_PCM_HW_PARAM_FORMAT, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, snd_pcm_hw_rule_div, NULL, SNDRV_PCM_HW_PARAM_FRAME_BITS, SNDRV_PCM_HW_PARAM_CHANNELS, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FRAME_BITS, + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FRAME_BITS, snd_pcm_hw_rule_mul, NULL, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, SNDRV_PCM_HW_PARAM_CHANNELS, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FRAME_BITS, + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FRAME_BITS, snd_pcm_hw_rule_mulkdiv, (void*) 8, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FRAME_BITS, + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_FRAME_BITS, snd_pcm_hw_rule_mulkdiv, (void*) 8, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS, snd_pcm_hw_rule_div, NULL, SNDRV_PCM_HW_PARAM_FRAME_BITS, SNDRV_PCM_HW_PARAM_SAMPLE_BITS, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, snd_pcm_hw_rule_mulkdiv, (void*) 1000000, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, SNDRV_PCM_HW_PARAM_PERIOD_TIME, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, snd_pcm_hw_rule_mulkdiv, (void*) 1000000, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, SNDRV_PCM_HW_PARAM_BUFFER_TIME, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIODS, + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIODS, snd_pcm_hw_rule_div, NULL, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, snd_pcm_hw_rule_div, NULL, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, SNDRV_PCM_HW_PARAM_PERIODS, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, snd_pcm_hw_rule_mulkdiv, (void*) 8, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, SNDRV_PCM_HW_PARAM_FRAME_BITS, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, snd_pcm_hw_rule_muldivk, (void*) 1000000, SNDRV_PCM_HW_PARAM_PERIOD_TIME, SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, snd_pcm_hw_rule_mul, NULL, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, SNDRV_PCM_HW_PARAM_PERIODS, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, snd_pcm_hw_rule_mulkdiv, (void*) 8, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, SNDRV_PCM_HW_PARAM_FRAME_BITS, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, snd_pcm_hw_rule_muldivk, (void*) 1000000, SNDRV_PCM_HW_PARAM_BUFFER_TIME, SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_BYTES, snd_pcm_hw_rule_muldivk, (void*) 8, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, SNDRV_PCM_HW_PARAM_FRAME_BITS, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, snd_pcm_hw_rule_muldivk, (void*) 8, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, SNDRV_PCM_HW_PARAM_FRAME_BITS, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_TIME, + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_PERIOD_TIME, snd_pcm_hw_rule_mulkdiv, (void*) 1000000, SNDRV_PCM_HW_PARAM_PERIOD_SIZE, SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_TIME, + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_TIME, snd_pcm_hw_rule_mulkdiv, (void*) 1000000, SNDRV_PCM_HW_PARAM_BUFFER_SIZE, SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) @@ -2306,7 +2306,7 @@ int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream) if (err < 0) return err; - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, snd_pcm_hw_rule_buffer_bytes_max, substream, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, -1); if (err < 0) @@ -2320,7 +2320,7 @@ int snd_pcm_hw_constraints_complete(struct snd_pcm_substream *substream) } if (!(hw->rates & (SNDRV_PCM_RATE_KNOT | SNDRV_PCM_RATE_CONTINUOUS))) { - err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, + err = snd_pcm_hw_rule_add(runtime, 0, SNDRV_PCM_HW_PARAM_RATE, snd_pcm_hw_rule_rate, hw, SNDRV_PCM_HW_PARAM_RATE, -1); if (err < 0) @@ -2510,9 +2510,9 @@ static int snd_pcm_open(struct file *file, struct snd_pcm *pcm, int stream) __error: module_put(pcm->card->module); __error2: - snd_card_file_remove(pcm->card, file); + snd_card_file_remove(pcm->card, file); __error1: - return err; + return err; } static int snd_pcm_release(struct inode *inode, struct file *file) @@ -2686,7 +2686,7 @@ static int snd_pcm_hwsync(struct snd_pcm_substream *substream) snd_pcm_stream_unlock_irq(substream); return err; } - + static snd_pcm_sframes_t snd_pcm_delay(struct snd_pcm_substream *substream) { struct snd_pcm_runtime *runtime = substream->runtime; @@ -2705,7 +2705,7 @@ static snd_pcm_sframes_t snd_pcm_delay(struct snd_pcm_substream *substream) snd_pcm_stream_unlock_irq(substream); return err < 0 ? err : n; } - + static int snd_pcm_sync_ptr(struct snd_pcm_substream *substream, struct snd_pcm_sync_ptr __user *_sync_ptr) { @@ -2719,7 +2719,7 @@ static int snd_pcm_sync_ptr(struct snd_pcm_substream *substream, if (get_user(sync_ptr.flags, (unsigned __user *)&(_sync_ptr->flags))) return -EFAULT; if (copy_from_user(&sync_ptr.c.control, &(_sync_ptr->c.control), sizeof(struct snd_pcm_mmap_control))) - return -EFAULT; + return -EFAULT; status = runtime->status; control = runtime->control; if (sync_ptr.flags & SNDRV_PCM_SYNC_PTR_HWSYNC) { @@ -2756,7 +2756,7 @@ static int snd_pcm_tstamp(struct snd_pcm_substream *substream, int __user *_arg) { struct snd_pcm_runtime *runtime = substream->runtime; int arg; - + if (get_user(arg, _arg)) return -EFAULT; if (arg < 0 || arg > SNDRV_PCM_TSTAMP_TYPE_LAST) @@ -2980,7 +2980,7 @@ int snd_pcm_kernel_ioctl(struct snd_pcm_substream *substream, { snd_pcm_uframes_t *frames = arg; snd_pcm_sframes_t result; - + switch (cmd) { case SNDRV_PCM_IOCTL_FORWARD: { @@ -3233,7 +3233,7 @@ static int snd_pcm_mmap_status_fault(struct vm_fault *vmf) { struct snd_pcm_substream *substream = vmf->vma->vm_private_data; struct snd_pcm_runtime *runtime; - + if (substream == NULL) return VM_FAULT_SIGBUS; runtime = substream->runtime; @@ -3269,7 +3269,7 @@ static int snd_pcm_mmap_control_fault(struct vm_fault *vmf) { struct snd_pcm_substream *substream = vmf->vma->vm_private_data; struct snd_pcm_runtime *runtime; - + if (substream == NULL) return VM_FAULT_SIGBUS; runtime = substream->runtime; @@ -3361,7 +3361,7 @@ static int snd_pcm_mmap_data_fault(struct vm_fault *vmf) unsigned long offset; struct page * page; size_t dma_bytes; - + if (substream == NULL) return VM_FAULT_SIGBUS; runtime = substream->runtime; @@ -3504,9 +3504,9 @@ EXPORT_SYMBOL(snd_pcm_mmap_data); static int snd_pcm_mmap(struct file *file, struct vm_area_struct *area) { struct snd_pcm_file * pcm_file; - struct snd_pcm_substream *substream; + struct snd_pcm_substream *substream; unsigned long offset; - + pcm_file = file->private_data; substream = pcm_file->substream; if (PCM_RUNTIME_CHECK(substream)) @@ -3704,7 +3704,7 @@ const struct file_operations snd_pcm_f_ops[2] = { .llseek = no_llseek, .poll = snd_pcm_playback_poll, .unlocked_ioctl = snd_pcm_ioctl, - .compat_ioctl = snd_pcm_ioctl_compat, + .compat_ioctl = snd_pcm_ioctl_compat, .mmap = snd_pcm_mmap, .fasync = snd_pcm_fasync, .get_unmapped_area = snd_pcm_get_unmapped_area, @@ -3718,7 +3718,7 @@ const struct file_operations snd_pcm_f_ops[2] = { .llseek = no_llseek, .poll = snd_pcm_capture_poll, .unlocked_ioctl = snd_pcm_ioctl, - .compat_ioctl = snd_pcm_ioctl_compat, + .compat_ioctl = snd_pcm_ioctl_compat, .mmap = snd_pcm_mmap, .fasync = snd_pcm_fasync, .get_unmapped_area = snd_pcm_get_unmapped_area, diff --git a/sound/core/rawmidi.c b/sound/core/rawmidi.c index 69616d00481c..e1903a978ffa 100644 --- a/sound/core/rawmidi.c +++ b/sound/core/rawmidi.c @@ -370,7 +370,7 @@ static int snd_rawmidi_open(struct inode *inode, struct file *file) struct snd_rawmidi_file *rawmidi_file = NULL; wait_queue_entry_t wait; - if ((file->f_flags & O_APPEND) && !(file->f_flags & O_NONBLOCK)) + if ((file->f_flags & O_APPEND) && !(file->f_flags & O_NONBLOCK)) return -EINVAL; /* invalid combination */ err = nonseekable_open(inode, file); @@ -520,7 +520,7 @@ int snd_rawmidi_kernel_release(struct snd_rawmidi_file *rfile) if (snd_BUG_ON(!rfile)) return -ENXIO; - + rmidi = rfile->rmidi; rawmidi_release_priv(rfile); module_put(rmidi->card->module); @@ -548,7 +548,7 @@ static int snd_rawmidi_info(struct snd_rawmidi_substream *substream, struct snd_rawmidi_info *info) { struct snd_rawmidi *rmidi; - + if (substream == NULL) return -ENODEV; rmidi = substream->rmidi; @@ -637,7 +637,7 @@ int snd_rawmidi_output_params(struct snd_rawmidi_substream *substream, { char *newbuf; struct snd_rawmidi_runtime *runtime = substream->runtime; - + if (substream->append && substream->use_count > 1) return -EBUSY; snd_rawmidi_drain_output(substream); @@ -836,7 +836,7 @@ static int snd_rawmidi_control_ioctl(struct snd_card *card, case SNDRV_CTL_IOCTL_RAWMIDI_NEXT_DEVICE: { int device; - + if (get_user(device, (int __user *)argp)) return -EFAULT; if (device >= SNDRV_RAWMIDI_DEVICES) /* next device is -1 */ @@ -858,7 +858,7 @@ static int snd_rawmidi_control_ioctl(struct snd_card *card, case SNDRV_CTL_IOCTL_RAWMIDI_PREFER_SUBDEVICE: { int val; - + if (get_user(val, (int __user *)argp)) return -EFAULT; control->preferred_subdevice[SND_CTL_SUBDEV_RAWMIDI] = val; @@ -1064,7 +1064,7 @@ int snd_rawmidi_transmit_empty(struct snd_rawmidi_substream *substream) spin_lock_irqsave(&runtime->lock, flags); result = runtime->avail >= runtime->buffer_size; spin_unlock_irqrestore(&runtime->lock, flags); - return result; + return result; } EXPORT_SYMBOL(snd_rawmidi_transmit_empty); @@ -1202,7 +1202,7 @@ EXPORT_SYMBOL(snd_rawmidi_transmit_ack); * @substream: the rawmidi substream * @buffer: the buffer pointer * @count: the data size to transfer - * + * * Copies data from the buffer to the device and advances the pointer. * * Return: The copied size if successful, or a negative error code on failure. @@ -1761,7 +1761,7 @@ void snd_rawmidi_set_ops(struct snd_rawmidi *rmidi, int stream, const struct snd_rawmidi_ops *ops) { struct snd_rawmidi_substream *substream; - + list_for_each_entry(substream, &rmidi->streams[stream].substreams, list) substream->ops = ops; } diff --git a/sound/core/seq_device.c b/sound/core/seq_device.c index e40a2cba5002..9515bb3632b9 100644 --- a/sound/core/seq_device.c +++ b/sound/core/seq_device.c @@ -242,7 +242,7 @@ int snd_seq_device_new(struct snd_card *card, int device, const char *id, put_device(&dev->dev); return err; } - + if (result) *result = dev; diff --git a/sound/core/sgbuf.c b/sound/core/sgbuf.c index 84fffabdd129..78100aaeb385 100644 --- a/sound/core/sgbuf.c +++ b/sound/core/sgbuf.c @@ -57,7 +57,7 @@ int snd_free_sgbuf_pages(struct snd_dma_buffer *dmab) kfree(sgbuf->page_table); kfree(sgbuf); dmab->private_data = NULL; - + return 0; } diff --git a/sound/core/sound_oss.c b/sound/core/sound_oss.c index 0a5c66229a22..74a275034235 100644 --- a/sound/core/sound_oss.c +++ b/sound/core/sound_oss.c @@ -149,14 +149,14 @@ int snd_register_oss_device(int type, struct snd_card *card, int dev, return 0; __end: - if (register2 >= 0) - unregister_sound_special(register2); - if (register1 >= 0) - unregister_sound_special(register1); + if (register2 >= 0) + unregister_sound_special(register2); + if (register1 >= 0) + unregister_sound_special(register1); snd_oss_minors[minor] = NULL; mutex_unlock(&sound_oss_mutex); kfree(preg); - return -EBUSY; + return -EBUSY; } EXPORT_SYMBOL(snd_register_oss_device); diff --git a/sound/core/timer.c b/sound/core/timer.c index dc87728c5b74..225686dac072 100644 --- a/sound/core/timer.c +++ b/sound/core/timer.c @@ -951,7 +951,7 @@ static int snd_timer_dev_register(struct snd_device *dev) return -ENXIO; if (!(timer->hw.flags & SNDRV_TIMER_HW_SLAVE) && !timer->hw.resolution && timer->hw.c_resolution == NULL) - return -EINVAL; + return -EINVAL; mutex_lock(&register_mutex); list_for_each_entry(timer1, &snd_timer_list, device_list) { @@ -2108,7 +2108,7 @@ static const struct file_operations snd_timer_f_ops = .poll = snd_timer_user_poll, .unlocked_ioctl = snd_timer_user_ioctl, .compat_ioctl = snd_timer_user_ioctl_compat, - .fasync = snd_timer_user_fasync, + .fasync = snd_timer_user_fasync, }; /* unregister the system timer */ diff --git a/sound/core/timer_compat.c b/sound/core/timer_compat.c index e00f7e399e46..e53563e808fc 100644 --- a/sound/core/timer_compat.c +++ b/sound/core/timer_compat.c @@ -97,7 +97,7 @@ static int snd_timer_user_status_compat(struct file *file, { struct snd_timer_user *tu; struct snd_timer_status32 status; - + tu = file->private_data; if (!tu->timeri) return -EBADFD; -- 2.16.2

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[alsa-devel] [PATCH] ASoC: Intel: Skylake: Fix typo
by abhijeet.kumar＠intel.com 05 Mar '18

05 Mar '18

From: Abhijeet Kumar <abhijeet.kumar(a)intel.com> Spelling mistake in state. So fixing it. Signed-off-by: Abhijeet Kumar <abhijeet.kumar(a)intel.com> --- sound/soc/intel/skylake/skl-messages.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/intel/skylake/skl-messages.c b/sound/soc/intel/skylake/skl-messages.c index 60d76adade43..f161907859f3 100644 --- a/sound/soc/intel/skylake/skl-messages.c +++ b/sound/soc/intel/skylake/skl-messages.c @@ -1210,7 +1210,7 @@ int skl_bind_modules(struct skl_sst *ctx, static int skl_set_pipe_state(struct skl_sst *ctx, struct skl_pipe *pipe, enum skl_ipc_pipeline_state state) { - dev_dbg(ctx->dev, "%s: pipe_satate = %d\n", __func__, state); + dev_dbg(ctx->dev, "%s: pipe_state = %d\n", __func__, state); return skl_ipc_set_pipeline_state(&ctx->ipc, pipe->ppl_id, state); } -- 1.9.1

2 1

[alsa-devel] [PATCH] speaker-test: Support S24_3LE sample format
by Julian Scheel 05 Mar '18

05 Mar '18

Implement support signed 24 bit samples, packed in 3 bytes. Signed-off-by: Julian Scheel <julian(a)jusst.de> --- speaker-test/speaker-test.c | 38 ++++++++++++++++++++++++++++++++++++++ 1 file changed, 38 insertions(+) diff --git a/speaker-test/speaker-test.c b/speaker-test/speaker-test.c index 46b0589..dbee63f 100644 --- a/speaker-test/speaker-test.c +++ b/speaker-test/speaker-test.c @@ -284,6 +284,7 @@ static const int supported_formats[] = { SND_PCM_FORMAT_S16_LE, SND_PCM_FORMAT_S16_BE, SND_PCM_FORMAT_FLOAT_LE, + SND_PCM_FORMAT_S24_3LE, SND_PCM_FORMAT_S32_LE, SND_PCM_FORMAT_S32_BE, -1 @@ -341,6 +342,19 @@ static void generate_sine(uint8_t *frames, int channel, int count, double *_phas *samp_f++ = 0.0; } break; + case SND_PCM_FORMAT_S24_3LE: + if (chn==channel) { + res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * generator_scale * 0x7fffff; + ires = res; + *samp8++ = LE_INT(ires); + *samp8++ = LE_INT(ires) >> 8; + *samp8++ = LE_INT(ires) >> 16; + } else { + *samp8++ = 0; + *samp8++ = 0; + *samp8++ = 0; + } + break; case SND_PCM_FORMAT_S32_LE: if (chn==channel) { res = (sin((phase * 2 * M_PI) / max_phase - M_PI)) * generator_scale * 0x7fffffff; @@ -415,6 +429,19 @@ static void generate_pink_noise( uint8_t *frames, int channel, int count) { *samp16++ = 0; } break; + case SND_PCM_FORMAT_S24_3LE: + if (chn==channel) { + res = generate_pink_noise_sample(&pink) * generator_scale * 0x07fffff; + ires = res; + *samp8++ = LE_INT(ires); + *samp8++ = LE_INT(ires) >> 8; + *samp8++ = LE_INT(ires) >> 16; + } else { + *samp8++ = 0; + *samp8++ = 0; + *samp8++ = 0; + } + break; case SND_PCM_FORMAT_S32_LE: if (chn==channel) { res = generate_pink_noise_sample(&pink) * generator_scale * 0x07fffffff; @@ -482,6 +509,17 @@ static void generate_pattern(uint8_t *frames, int channel, int count, int *_patt *samp_f++ = 0.0; } break; + case SND_PCM_FORMAT_S24_3LE: + if (chn==channel) { + *samp8++ = LE_INT(pattern); + *samp8++ = LE_INT(pattern) >> 8; + *samp8++ = LE_INT(pattern) >> 16; + } else { + *samp8++ = 0; + *samp8++ = 0; + *samp8++ = 0; + } + break; case SND_PCM_FORMAT_S32_LE: if (chn==channel) { *samp32++ = LE_INT(pattern); -- 2.16.1

2 1

[alsa-devel] [PATCH] sound/tlv320dac33: Add device tree support
by Pavel Machek 05 Mar '18

05 Mar '18

This adds device tree support to tlv320dac33.c. Signed-off-by: Pavel Machek <pavel(a)ucw.cz> diff --git a/Documentation/devicetree/bindings/sound/tlv320dac33.txt b/Documentation/devicetree/bindings/sound/tlv320dac33.txt new file mode 100644 index 0000000..6cbd311 --- /dev/null +++ b/Documentation/devicetree/bindings/sound/tlv320dac33.txt @@ -0,0 +1,32 @@ +Texas Instruments - tlv320dac33 Codec module + +The tlv320dac33 serial control bus communicates through I2C protocols. + +Required properties: + +- compatible - "ti,tlv320dac33" +- reg - I2C slave address + +Optional properties: + +- power-gpios - gpio pin to power the device, active high + +- interrupts - The interrupt output from the device. +- interrupt-parent - The parent interrupt controller. + +- ti,keep-bclk - Keep the BCLK running in FIFO modes +- ti,burst-bclkdiv - BCLK divider value in burst mode + +Example: + +tlv320dac33: audio-codec@19 { + compatible = "ti,tlv320dac33"; + reg = <0x19>; + + interrupt-parent = <&gpio2>; + interrupts = <21 1>; /* gpio_53, IRQF_TRIGGER_RISING */ + power-gpio = <&gpio2 28 0>; /* gpio_60 */ + + ti,keep-bclk; + ti,burst-bclkdiv = <3>; +}; diff --git a/sound/soc/codecs/tlv320dac33.c b/sound/soc/codecs/tlv320dac33.c index 5b94a15..2a0fe06 100644 --- a/sound/soc/codecs/tlv320dac33.c +++ b/sound/soc/codecs/tlv320dac33.c @@ -30,6 +32,7 @@ #include <linux/interrupt.h> #include <linux/gpio.h> #include <linux/regulator/consumer.h> +#include <linux/of_gpio.h> #include <linux/slab.h> #include <sound/core.h> #include <sound/pcm.h> @@ -105,7 +108,7 @@ struct tlv320dac33_priv { unsigned int nsample; /* burst read amount from host */ int mode1_latency; /* latency caused by the i2c writes in * us */ - u8 burst_bclkdiv; /* BCLK divider value in burst mode */ + u32 burst_bclkdiv; /* BCLK divider value in burst mode */ unsigned int burst_rate; /* Interface speed in Burst modes */ int keep_bclk; /* Keep the BCLK continuously running @@ -1484,16 +1487,11 @@ static struct snd_soc_dai_driver dac33_dai = { static int dac33_i2c_probe(struct i2c_client *client, const struct i2c_device_id *id) { - struct tlv320dac33_platform_data *pdata; + struct tlv320dac33_platform_data *pdata = client->dev.platform_data; struct tlv320dac33_priv *dac33; + struct device_node *np = client->dev.of_node; int ret, i; - if (client->dev.platform_data == NULL) { - dev_err(&client->dev, "Platform data not set\n"); - return -ENODEV; - } - pdata = client->dev.platform_data; - dac33 = devm_kzalloc(&client->dev, sizeof(struct tlv320dac33_priv), GFP_KERNEL); if (dac33 == NULL) @@ -1505,10 +1503,26 @@ static int dac33_i2c_probe(struct i2c_client *client, i2c_set_clientdata(client, dac33); - dac33->power_gpio = pdata->power_gpio; - dac33->burst_bclkdiv = pdata->burst_bclkdiv; - dac33->keep_bclk = pdata->keep_bclk; - dac33->mode1_latency = pdata->mode1_latency; + if (pdata) { + dac33->power_gpio = pdata->power_gpio; + dac33->burst_bclkdiv = pdata->burst_bclkdiv; + dac33->keep_bclk = pdata->keep_bclk; + dac33->mode1_latency = pdata->mode1_latency; + } else if (np) { + ret = of_get_named_gpio(np, "power-gpios", 0); + if (ret >= 0) + dac33->power_gpio = ret; + else + dac33->power_gpio = -1; + + if (of_property_read_bool(np, "ti,keep-bclk")) + dac33->keep_bclk = true; + + of_property_read_u32(np, "ti,burst-bclkdiv", &dac33->burst_bclkdiv); + } else { + dev_err(&client->dev, "Platform data not set\n"); + return -ENODEV; + } if (!dac33->mode1_latency) dac33->mode1_latency = 10000; /* 10ms */ dac33->irq = client->irq; @@ -1574,9 +1588,16 @@ static const struct i2c_device_id tlv320dac33_i2c_id[] = { }; MODULE_DEVICE_TABLE(i2c, tlv320dac33_i2c_id); +static const struct of_device_id tlv320dac33_of_match[] = { + { .compatible = "ti,tlv320dac33", }, + {}, +}; +MODULE_DEVICE_TABLE(i2c, tlv320dac33_of_match); + static struct i2c_driver tlv320dac33_i2c_driver = { .driver = { .name = "tlv320dac33-codec", + .of_match_table = of_match_ptr(tlv320dac33_of_match), }, .probe = dac33_i2c_probe, .remove = dac33_i2c_remove, -- (english) http://www.livejournal.com/~pavelmachek (cesky, pictures) http://atrey.karlin.mff.cuni.cz/~pavel/picture/horses/blog.html

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[alsa-devel] [asoc:topic/rt5514 1/3] sound/soc/codecs/rt5514-spi.c:477:70: error: 'DRV_NAME' undeclared
by kbuild test robot 05 Mar '18

05 Mar '18

tree: https://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound.git topic/rt5514 head: 890861aa85043cc9d99a01e1bd26883722e95243 commit: 04e8262ac8c3f59599ea6dfe5cd1a039828ea4ad [1/3] ASoC: rt5514: replace codec to component config: x86_64-randconfig-s1-03021352 (attached as .config) compiler: gcc-6 (Debian 6.4.0-9) 6.4.0 20171026 reproduce: git checkout 04e8262ac8c3f59599ea6dfe5cd1a039828ea4ad # save the attached .config to linux build tree make ARCH=x86_64 All errors (new ones prefixed by >>): sound/soc/codecs/rt5514-spi.c: In function 'rt5514_resume': >> sound/soc/codecs/rt5514-spi.c:477:70: error: 'DRV_NAME' undeclared (first use in this function) struct snd_soc_component *component = snd_soc_lookup_component(dev, DRV_NAME); ^~~~~~~~ sound/soc/codecs/rt5514-spi.c:477:70: note: each undeclared identifier is reported only once for each function it appears in vim +/DRV_NAME +477 sound/soc/codecs/rt5514-spi.c 474 475 static int __maybe_unused rt5514_resume(struct device *dev) 476 { > 477 struct snd_soc_component *component = snd_soc_lookup_component(dev, DRV_NAME); 478 struct rt5514_dsp *rt5514_dsp = 479 snd_soc_component_get_drvdata(component); 480 int irq = to_spi_device(dev)->irq; 481 u8 buf[8]; 482 483 if (device_may_wakeup(dev)) 484 disable_irq_wake(irq); 485 486 if (rt5514_dsp) { 487 if (rt5514_dsp->substream) { 488 rt5514_spi_burst_read(RT5514_IRQ_CTRL, (u8 *)&buf, 489 sizeof(buf)); 490 if (buf[0] & RT5514_IRQ_STATUS_BIT) 491 rt5514_schedule_copy(rt5514_dsp); 492 } 493 } 494 495 return 0; 496 } 497 --- 0-DAY kernel test infrastructure Open Source Technology Center https://lists.01.org/pipermail/kbuild-all Intel Corporation

2 1

[alsa-devel] [RFC] hda/realtek filtered headphone output
by René Rebe 04 Mar '18

04 Mar '18

Hi all, testing some new laptops, I noticed the headphone frequency response was lacking bass. First I noticed this in the Lenovo T470s, and later I found this also be the case on Surface Pro 2 and 3. The Dell XPS13 may also be affected, but I had to return it before I found out the solution on the other machines: https://www.youtube.com/watch?v=a1YkPtfC4LI I had not yet sent the patch because a) I was too busy and b) I wanted to review this further. As the other day I got a Lenovo ideapad 720s to test with exactly the same issue, here is the current hacked patch: https://svn.exactcode.de/t2/trunk/package/base/linux/surface-audio.patch However, given that all newer realtek codecs that came on my desk are affected, maybe there is something wrong with the hda_auto_parser.c - I did not yet fully grasp all the logic of that. As you can see in the patch and video (…) the problem is the signal routing of NID 0x15 or 0x21 (depending whether it is an older, or newer codec). This is the headphone out, and as far as I can see the auto_parser assigned the index 0 pcm output (NID 0x0c) which result in this filtered frequency response, likely intended for the laptop’s speakers? Using the next NID (I guess that was 0xd?) results in a full frequency response and at least to my ears is immediately hearable and obvious. Without this fixed routing the headphone sounds like a PC speaker in a coke can. So the question for the authors and HDA experts: Can this be fixed in the auto_parser, or is my patch (which btw. has to small unrelated cleanups) modulo potentially some renaming of the enum and function, now that more than the Surface is affected, the way to go? If this can be fixed more generally I would be happy to hear! Greetings from Berlin, René Rebe -- ExactCODE GmbH, Lietzenburger Str. 42, DE-10789 Berlin http://exactcode.com | http://exactscan.com | http://ocrkit.com | http://t2-project.org | http://rene.rebe.de

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