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

[alsa-devel] [PATCH v2] ALSA: core api: define offsets for TLV items
by Ranjani Sridharan 29 Mar '18

29 Mar '18

This patch defines accessor offsets for 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> --- V2: removed redundant redefinitions and updated prefix --- include/uapi/sound/tlv.h | 8 ++++++++ 1 file changed, 8 insertions(+) diff --git a/include/uapi/sound/tlv.h b/include/uapi/sound/tlv.h index be5371f09a62..1fd786a5e57b 100644 --- a/include/uapi/sound/tlv.h +++ b/include/uapi/sound/tlv.h @@ -98,4 +98,12 @@ #define SNDRV_CTL_TLVD_DB_GAIN_MUTE -9999999 +/* Accessor offsets for TLV data items */ +#define SNDRV_CTL_TLV_OFFSET_TYPE 0 +#define SNDRV_CTL_TLV_OFFSET_LEN 1 + +/* Accessor offsets for min, mute and step items in dB scale type TLV */ +#define SNDRV_CTL_TLV_OFFSET_DB_MIN 2 +#define SNDRV_CTL_TLV_OFFSET_DB_MUTE_AND_STEP 3 + #endif -- 2.14.1

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[alsa-devel] [PATCH -next] ASoC: amd: acp-da7219-max98357: Make symbol da7219_dai_clk static
by Wei Yongjun 29 Mar '18

29 Mar '18

Fixes the following 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: Wei Yongjun <weiyongjun1(a)huawei.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 b205c78..f41560e 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) {

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Re: [alsa-devel] [PATCH v2] pcm_native: Remove VLA usage
by Takashi Sakamoto 29 Mar '18

29 Mar '18

Hi, On Mar 29 2018 07:24, Kyle Spiers 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. > > Signed-off-by: Kyle Spiers <kyle(a)spiers.me> > --- > sound/core/pcm_native.c | 11 +++++++---- > 1 file changed, 7 insertions(+), 4 deletions(-) This bug was already fixed at a commit 5730f9f744cf (' ALSA: pcm: Remove VLA usage')[1][2]. [1] https://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound.git/commit/soun… [2] http://mailman.alsa-project.org/pipermail/alsa-devel/2018-March/133268.html Thanks Takashi Sakamoto

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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

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[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

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[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

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[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

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[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

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[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

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[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

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