November 2014 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH v2] ALSA: usb-audio: Fix memory leak in FTU quirk
by Takashi Iwai 11 Nov '14

11 Nov '14

M-audio FastTrack Ultra quirk doesn't release the kzalloc'ed memory. This patch adds the private_free callback to release it properly. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- v1->v2: free the correct field sound/usb/mixer_quirks.c | 6 ++++++ 1 file changed, 6 insertions(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index f119a41ed9a9..7c83bab69dee 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -885,6 +885,11 @@ static int snd_ftu_eff_switch_put(struct snd_kcontrol *kctl, return changed; } +static void kctl_private_value_free(struct snd_kcontrol *kctl) +{ + kfree((void *)kctl->private_value); +} + static int snd_ftu_create_effect_switch(struct usb_mixer_interface *mixer, int validx, int bUnitID) { @@ -919,6 +924,7 @@ static int snd_ftu_create_effect_switch(struct usb_mixer_interface *mixer, return -ENOMEM; } + kctl->private_free = kctl_private_value_free; err = snd_ctl_add(mixer->chip->card, kctl); if (err < 0) return err; -- 2.1.3

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[alsa-devel] [PATCH] ALSA: usb-audio: Fix memory leak in FTU quirk
by Takashi Iwai 11 Nov '14

11 Nov '14

M-audio FastTrack Ultra quirk doesn't release the kzalloc'ed memory. This patch adds the private_free callback to release it properly. Cc: <stable(a)vger.kernel.org> Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/usb/mixer_quirks.c | 1 + 1 file changed, 1 insertion(+) diff --git a/sound/usb/mixer_quirks.c b/sound/usb/mixer_quirks.c index 45203168ac6d..f6f8acf0446b 100644 --- a/sound/usb/mixer_quirks.c +++ b/sound/usb/mixer_quirks.c @@ -1021,6 +1021,7 @@ static int snd_ftu_create_effect_switch(struct usb_mixer_interface *mixer, return -ENOMEM; } + kctl->private_free = usb_mixer_elem_free; err = snd_ctl_add(mixer->chip->card, kctl); if (err < 0) return err; -- 2.1.3

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[alsa-devel] [PATCH v6 0/3] Add Phytec pbab01 audio support
by Dmitry Lavnikevich 11 Nov '14

11 Nov '14

This version of patchset contains following style changes since v5: - nodes sorted in alphabetical order; - pinctrl entries sorted alphabetically; - new line inserted between nodes.

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[alsa-devel] [PATCH 0/2] Subject: [PATCH 0/2] ASoC: tlv320aic3x: soc_enum array conversion and pop reduction
by Peter Ujfalusi 11 Nov '14

11 Nov '14

Hi, Convert the soc_enum arrays to individual items and add the pop reduction patch on top of this change. Regards, Peter --- Misael Lopez Cruz (1): ASoC: tlv320aic3x: Add output driver pop reduction controls Peter Ujfalusi (1): ASoC: tlv320aic3x: Convert SOC_ENUM_SINGLE/DOUBLE arrays to individual sound/soc/codecs/tlv320aic3x.c | 165 +++++++++++++++++++++++------------------ 1 file changed, 93 insertions(+), 72 deletions(-) -- 2.1.3

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[alsa-devel] Realtek HDA SKU/assembly parsing
by Daniel Drake 11 Nov '14

11 Nov '14

Hi, I see that there is a function that parses the subsystem ID for various realtek codecs, alc_auto_parse_customize_define Is there a public specification that documents the specific meaning of each bit here? Thanks Daniel -- To unsubscribe from this list: send the line "unsubscribe alsa-devel" in the body of a message to majordomo(a)vger.kernel.org More majordomo info at http://vger.kernel.org/majordomo-info.html

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[alsa-devel] [PATCH] ASoC: rt5645: remove unused rt5645_clk_sel_put
by Bard Liao 11 Nov '14

11 Nov '14

Remove rt5645_clk_sel_put function since it is never used. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- sound/soc/codecs/rt5645.c | 59 ----------------------------------------------- 1 file changed, 59 deletions(-) diff --git a/sound/soc/codecs/rt5645.c b/sound/soc/codecs/rt5645.c index 1adab38..d6298b3 100644 --- a/sound/soc/codecs/rt5645.c +++ b/sound/soc/codecs/rt5645.c @@ -443,65 +443,6 @@ static SOC_ENUM_SINGLE_DECL(rt5645_tdm_adc_sel_enum, RT5645_TDM_CTRL_1, 8, rt5645_tdm_adc_data_select); -static int rt5645_clk_sel_put(struct snd_kcontrol *kcontrol, - struct snd_ctl_elem_value *ucontrol) -{ - struct snd_soc_codec *codec = snd_soc_kcontrol_codec(kcontrol); - unsigned int u_bit = 0, p_bit = 0; - struct soc_enum *em = - (struct soc_enum *)kcontrol->private_value; - - switch (em->reg) { - case RT5645_ASRC_2: - switch (em->shift_l) { - case 0: - u_bit = 0x8; - p_bit = RT5645_PWR_ADC_S1F; - break; - case 4: - u_bit = 0x100; - p_bit = RT5645_PWR_DAC_MF_R; - break; - case 8: - u_bit = 0x200; - p_bit = RT5645_PWR_DAC_MF_L; - break; - case 12: - u_bit = 0x400; - p_bit = RT5645_PWR_DAC_S1F; - break; - } - break; - case RT5645_ASRC_3: - switch (em->shift_l) { - case 0: - u_bit = 0x1; - p_bit = RT5645_PWR_ADC_MF_R; - break; - case 4: - u_bit = 0x2; - p_bit = RT5645_PWR_ADC_MF_L; - break; - } - break; - } - - if (u_bit || p_bit) { - switch (ucontrol->value.integer.value[0]) { - case 1 ... 4: /*enable*/ - if (snd_soc_read(codec, RT5645_PWR_DIG2) & p_bit) - snd_soc_update_bits(codec, - RT5645_ASRC_1, u_bit, u_bit); - break; - default: /*disable*/ - snd_soc_update_bits(codec, RT5645_ASRC_1, u_bit, 0); - break; - } - } - - return snd_soc_put_enum_double(kcontrol, ucontrol); -} - static const struct snd_kcontrol_new rt5645_snd_controls[] = { /* Speaker Output Volume */ SOC_DOUBLE("Speaker Channel Switch", RT5645_SPK_VOL, -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH v2] ASoC: rt5670: add rt5672 codec support
by bardliao＠realtek.com 11 Nov '14

11 Nov '14

From: Bard Liao <bardliao(a)realtek.com> rt5672 is very similar to rt5670. Therefore we use one codec driver to support both codecs. The difference between rt5670 and rt5672 is there is some difference in their dapm routing table. Signed-off-by: Bard Liao <bardliao(a)realtek.com> --- * Add { "rt5671", 0 }, in rt5670_i2c_id[] --- sound/soc/codecs/rt5670.c | 71 +++++++++++++++++++++++++++++++++++++++-------- sound/soc/codecs/rt5670.h | 6 ++++ 2 files changed, 65 insertions(+), 12 deletions(-) diff --git a/sound/soc/codecs/rt5670.c b/sound/soc/codecs/rt5670.c index 9bd8b4f..c4c42d5 100644 --- a/sound/soc/codecs/rt5670.c +++ b/sound/soc/codecs/rt5670.c @@ -1595,29 +1595,40 @@ static const struct snd_soc_dapm_widget rt5670_dapm_widgets[] = { /* PDM */ SND_SOC_DAPM_SUPPLY("PDM1 Power", RT5670_PWR_DIG2, RT5670_PWR_PDM1_BIT, 0, NULL, 0), - SND_SOC_DAPM_SUPPLY("PDM2 Power", RT5670_PWR_DIG2, - RT5670_PWR_PDM2_BIT, 0, NULL, 0), SND_SOC_DAPM_MUX("PDM1 L Mux", RT5670_PDM_OUT_CTRL, RT5670_M_PDM1_L_SFT, 1, &rt5670_pdm1_l_mux), SND_SOC_DAPM_MUX("PDM1 R Mux", RT5670_PDM_OUT_CTRL, RT5670_M_PDM1_R_SFT, 1, &rt5670_pdm1_r_mux), - SND_SOC_DAPM_MUX("PDM2 L Mux", RT5670_PDM_OUT_CTRL, - RT5670_M_PDM2_L_SFT, 1, &rt5670_pdm2_l_mux), - SND_SOC_DAPM_MUX("PDM2 R Mux", RT5670_PDM_OUT_CTRL, - RT5670_M_PDM2_R_SFT, 1, &rt5670_pdm2_r_mux), /* Output Lines */ SND_SOC_DAPM_OUTPUT("HPOL"), SND_SOC_DAPM_OUTPUT("HPOR"), SND_SOC_DAPM_OUTPUT("LOUTL"), SND_SOC_DAPM_OUTPUT("LOUTR"), +}; + +static const struct snd_soc_dapm_widget rt5670_specific_dapm_widgets[] = { + SND_SOC_DAPM_SUPPLY("PDM2 Power", RT5670_PWR_DIG2, + RT5670_PWR_PDM2_BIT, 0, NULL, 0), + SND_SOC_DAPM_MUX("PDM2 L Mux", RT5670_PDM_OUT_CTRL, + RT5670_M_PDM2_L_SFT, 1, &rt5670_pdm2_l_mux), + SND_SOC_DAPM_MUX("PDM2 R Mux", RT5670_PDM_OUT_CTRL, + RT5670_M_PDM2_R_SFT, 1, &rt5670_pdm2_r_mux), SND_SOC_DAPM_OUTPUT("PDM1L"), SND_SOC_DAPM_OUTPUT("PDM1R"), SND_SOC_DAPM_OUTPUT("PDM2L"), SND_SOC_DAPM_OUTPUT("PDM2R"), }; +static const struct snd_soc_dapm_widget rt5672_specific_dapm_widgets[] = { + SND_SOC_DAPM_PGA("SPO Amp", SND_SOC_NOPM, 0, 0, NULL, 0), + SND_SOC_DAPM_OUTPUT("SPOLP"), + SND_SOC_DAPM_OUTPUT("SPOLN"), + SND_SOC_DAPM_OUTPUT("SPORP"), + SND_SOC_DAPM_OUTPUT("SPORN"), +}; + static const struct snd_soc_dapm_route rt5670_dapm_routes[] = { { "ADC Stereo1 Filter", NULL, "ADC STO1 ASRC", is_using_asrc }, { "ADC Stereo2 Filter", NULL, "ADC STO2 ASRC", is_using_asrc }, @@ -1970,12 +1981,6 @@ static const struct snd_soc_dapm_route rt5670_dapm_routes[] = { { "PDM1 R Mux", "Stereo DAC", "Stereo DAC MIXR" }, { "PDM1 R Mux", "Mono DAC", "Mono DAC MIXR" }, { "PDM1 R Mux", NULL, "PDM1 Power" }, - { "PDM2 L Mux", "Stereo DAC", "Stereo DAC MIXL" }, - { "PDM2 L Mux", "Mono DAC", "Mono DAC MIXL" }, - { "PDM2 L Mux", NULL, "PDM2 Power" }, - { "PDM2 R Mux", "Stereo DAC", "Stereo DAC MIXR" }, - { "PDM2 R Mux", "Mono DAC", "Mono DAC MIXR" }, - { "PDM2 R Mux", NULL, "PDM2 Power" }, { "HP Amp", NULL, "HPO MIX" }, { "HP Amp", NULL, "Mic Det Power" }, @@ -1993,13 +1998,30 @@ static const struct snd_soc_dapm_route rt5670_dapm_routes[] = { { "LOUTR", NULL, "LOUT R Playback" }, { "LOUTL", NULL, "Improve HP Amp Drv" }, { "LOUTR", NULL, "Improve HP Amp Drv" }, +}; +static const struct snd_soc_dapm_route rt5670_specific_dapm_routes[] = { + { "PDM2 L Mux", "Stereo DAC", "Stereo DAC MIXL" }, + { "PDM2 L Mux", "Mono DAC", "Mono DAC MIXL" }, + { "PDM2 L Mux", NULL, "PDM2 Power" }, + { "PDM2 R Mux", "Stereo DAC", "Stereo DAC MIXR" }, + { "PDM2 R Mux", "Mono DAC", "Mono DAC MIXR" }, + { "PDM2 R Mux", NULL, "PDM2 Power" }, { "PDM1L", NULL, "PDM1 L Mux" }, { "PDM1R", NULL, "PDM1 R Mux" }, { "PDM2L", NULL, "PDM2 L Mux" }, { "PDM2R", NULL, "PDM2 R Mux" }, }; +static const struct snd_soc_dapm_route rt5672_specific_dapm_routes[] = { + { "SPO Amp", NULL, "PDM1 L Mux" }, + { "SPO Amp", NULL, "PDM1 R Mux" }, + { "SPOLP", NULL, "SPO Amp" }, + { "SPOLN", NULL, "SPO Amp" }, + { "SPORP", NULL, "SPO Amp" }, + { "SPORN", NULL, "SPO Amp" }, +}; + static int rt5670_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { @@ -2331,6 +2353,29 @@ static int rt5670_probe(struct snd_soc_codec *codec) { struct rt5670_priv *rt5670 = snd_soc_codec_get_drvdata(codec); + switch (snd_soc_read(codec, RT5670_RESET) & RT5670_ID_MASK) { + case RT5670_ID_5670: + case RT5670_ID_5671: + snd_soc_dapm_new_controls(&codec->dapm, + rt5670_specific_dapm_widgets, + ARRAY_SIZE(rt5670_specific_dapm_widgets)); + snd_soc_dapm_add_routes(&codec->dapm, + rt5670_specific_dapm_routes, + ARRAY_SIZE(rt5670_specific_dapm_routes)); + break; + case RT5670_ID_5672: + snd_soc_dapm_new_controls(&codec->dapm, + rt5672_specific_dapm_widgets, + ARRAY_SIZE(rt5672_specific_dapm_widgets)); + snd_soc_dapm_add_routes(&codec->dapm, + rt5672_specific_dapm_routes, + ARRAY_SIZE(rt5672_specific_dapm_routes)); + break; + default: + dev_err(codec->dev, + "The driver is for RT5670 RT5671 or RT5672 only\n"); + return -ENODEV; + } rt5670->codec = codec; return 0; @@ -2452,6 +2497,8 @@ static const struct regmap_config rt5670_regmap = { static const struct i2c_device_id rt5670_i2c_id[] = { { "rt5670", 0 }, + { "rt5671", 0 }, + { "rt5672", 0 }, { } }; MODULE_DEVICE_TABLE(i2c, rt5670_i2c_id); diff --git a/sound/soc/codecs/rt5670.h b/sound/soc/codecs/rt5670.h index a0b5c85..d11b9c2 100644 --- a/sound/soc/codecs/rt5670.h +++ b/sound/soc/codecs/rt5670.h @@ -228,6 +228,12 @@ #define RT5670_R_VOL_MASK (0x3f) #define RT5670_R_VOL_SFT 0 +/* SW Reset & Device ID (0x00) */ +#define RT5670_ID_MASK (0x3 << 1) +#define RT5670_ID_5670 (0x0 << 1) +#define RT5670_ID_5672 (0x1 << 1) +#define RT5670_ID_5671 (0x2 << 1) + /* Combo Jack Control 1 (0x0a) */ #define RT5670_CBJ_BST1_MASK (0xf << 12) #define RT5670_CBJ_BST1_SFT (12) -- 1.8.1.1.439.g50a6b54

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[alsa-devel] [PATCH 0/6] ASoC: davinci-mcasp: fixes and over/underrun handling
by Peter Ujfalusi 11 Nov '14

11 Nov '14

Hi, Some fixes for the mcasp code: - rx formating fix - constraint for number of channels and sample bits handling of over/underrun: When such event happens (in McASP vs DMA level) we force restart the stream in order to get McASP and the DMA in sync again. Regards, Peter --- Misael Lopez Cruz (2): ASoC: davinci-mcasp: Active slots depend on active serializers ASoC: davinci-mcasp: Add overrun/underrun event handling Peter Ujfalusi (4): ASoC: davinci-mcasp: Validate tdm_slots parameter at probe time ASoC: davinci-mcasp: Place constraint on number of channels ASoC: davinci-mcasp: Symmetric sample bits for IIS mode ASoC: davinci-mcasp: Fix rx format when more bclk is used on the bus .../bindings/sound/davinci-mcasp-audio.txt | 2 +- sound/soc/davinci/davinci-mcasp.c | 212 +++++++++++++++++++-- sound/soc/davinci/davinci-mcasp.h | 10 + 3 files changed, 210 insertions(+), 14 deletions(-) -- 2.1.3

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[alsa-devel] [PATCH 1/2] ASoC: tlv320aic3x: Add output driver pop reduction controls
by Peter Ujfalusi 11 Nov '14

11 Nov '14

From: Misael Lopez Cruz <misael.lopez(a)ti.com> Output driver has two parameters that can be configured to reduce pop noise: power-on delay and ramp-up step time. Two new kcontrols have been added to set these parameters. Signed-off-by: Misael Lopez Cruz <misael.lopez(a)ti.com> Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- sound/soc/codecs/tlv320aic3x.c | 13 +++++++++++++ 1 file changed, 13 insertions(+) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index f7c2a575a892..70f8b8be9173 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -270,6 +270,15 @@ static const struct soc_enum aic3x_agc_decay_enum[] = { SOC_ENUM_SINGLE(RAGC_CTRL_A, 0, 4, aic3x_agc_decay), }; +static const char * const aic3x_poweron_time[] = { + "0us", "10us", "100us", "1ms", "10ms", "50ms", + "100ms", "200ms", "400ms", "800ms", "2s", "4s" }; +static const char * const aic3x_rampup_step[] = { "0ms", "1ms", "2ms", "4ms" }; +static const struct soc_enum aic3x_pop_reduction_enum[] = { + SOC_ENUM_SINGLE(HPOUT_POP_REDUCTION, 4, 12, aic3x_poweron_time), + SOC_ENUM_SINGLE(HPOUT_POP_REDUCTION, 2, 4, aic3x_rampup_step), +}; + /* * DAC digital volumes. From -63.5 to 0 dB in 0.5 dB steps */ @@ -399,6 +408,10 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1), SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]), + + /* Pop reduction */ + SOC_ENUM("Output Driver Power-On time", aic3x_pop_reduction_enum[0]), + SOC_ENUM("Output Driver Ramp-up step", aic3x_pop_reduction_enum[1]), }; static const struct snd_kcontrol_new aic3x_mono_controls[] = { -- 2.1.3

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[alsa-devel] [PATCH 0/2 v6] Digidesign Mbox 1 enhancements
by Damien Zammit 11 Nov '14

11 Nov '14

Revision 6. Damien Zammit (2): ALSA: usb-audio: Add mixer control for Digidesign Mbox 1 clock source ALSA: usb-audio: Add duplex mode for Digidesign Mbox 1 and enable mixer sound/usb/mixer_maps.c | 9 ++++ sound/usb/mixer_quirks.c | 129 +++++++++++++++++++++++++++++++++++++++++++++++ sound/usb/quirks-table.h | 43 ++++++++++++---- 3 files changed, 171 insertions(+), 10 deletions(-) -- 1.9.1

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