June 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH V3 1/1] ASoC: tlv320aic3x: Fix codec pll configure bug
by Hebbar, Gururaja 27 Jun '12

27 Jun '12

In sound/soc/codecs/tlv320aic3x.c data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); snd_soc_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT)); In the above code, pll-p value is OR'ed with previous value without clearing it. Bug is not seen if pll-p value doesn't change across Sampling frequency. However on some platforms (like AM335x EVM-SK), pll-p may have different values across different sampling frequencies. In such case, above code configures the pll with a wrong value. Because of this bug, when a audio stream is played with pll value different from previous stream, audio is heard as differently(like its stretched). Signed-off-by: Hebbar, Gururaja <gururaja.hebbar(a)ti.com> --- chnages in V2: modify subject to indicate ASOC and codec as tlv320aic3x changes in V3: use snd_soc_update_bits() instead of read-mask-write sound/soc/codecs/tlv320aic3x.c | 4 +--- sound/soc/codecs/tlv320aic3x.h | 1 + 2 files changed, 2 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 64d2a4f..e9b62b5 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -935,9 +935,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, } found: - data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); - snd_soc_write(codec, AIC3X_PLL_PROGA_REG, - data | (pll_p << PLLP_SHIFT)); + snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG, PLLP_MASK, pll_p); snd_soc_write(codec, AIC3X_OVRF_STATUS_AND_PLLR_REG, pll_r << PLLR_SHIFT); snd_soc_write(codec, AIC3X_PLL_PROGB_REG, pll_j << PLLJ_SHIFT); diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 6f097fb..08c7f66 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -166,6 +166,7 @@ /* PLL registers bitfields */ #define PLLP_SHIFT 0 +#define PLLP_MASK 7 #define PLLQ_SHIFT 3 #define PLLR_SHIFT 0 #define PLLJ_SHIFT 2 -- 1.7.1

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[alsa-devel] [PATCH] ALSA: ASoC: Remove const attribute from lm49453's snd_soc_dai_driver
by Daniel Mack 27 Jun '12

27 Jun '12

Fixes the following build warning. CC sound/soc/codecs/lm49453.o sound/soc/codecs/lm49453.c: In function ‘lm49453_i2c_probe’: sound/soc/codecs/lm49453.c:1511:24: warning: passing argument 3 of ‘snd_soc_register_codec’ discards qualifiers from pointer target type include/sound/soc.h:333:5: note: expected ‘struct snd_soc_dai_driver *’ but argument is of type ‘const struct snd_soc_dai_driver *’ Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/soc/codecs/lm49453.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/soc/codecs/lm49453.c b/sound/soc/codecs/lm49453.c index 802b9f1..5bd1294 100644 --- a/sound/soc/codecs/lm49453.c +++ b/sound/soc/codecs/lm49453.c @@ -1358,7 +1358,7 @@ static struct snd_soc_dai_ops lm49453_lineout_dai_ops = { }; /* LM49453 dai structure. */ -static const struct snd_soc_dai_driver lm49453_dai[] = { +static struct snd_soc_dai_driver lm49453_dai[] = { { .name = "LM49453 Headset", .playback = { -- 1.7.10.2

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[alsa-devel] [PATCH] ASoC: tlv320aic3x: add AGC, MIC_BIAS
by Prchal Jiří 27 Jun '12

27 Jun '12

This patch adds settings for TLV320AIC3106 of AGC, MIC_BIAS, De-emphasis. Also added default settings of output pop-up reduction, short circuit protection and mix both channels to mono_out to get maximum level. Signed-off-by: Jiri Prchal <jiri.prchal(a)aksignal.cz> --- /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.c +++ /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.c.new @@ -194,6 +194,11 @@ static const char *aic3x_linein_mode_mux[] = { "single-ended", "differential" }; static const char *aic3x_adc_hpf[] = { "Disabled", "0.0045xFs", "0.0125xFs", "0.025xFs" }; +static const char *aic3x_agc_level[] = + { "-5.5dB", "-8dB", "-10dB", "-12dB", "-14dB", "-17dB", "-20dB", "-24dB" }; +static const char *aic3x_agc_attack[] = { "8ms", "11ms", "16ms", "20ms" }; +static const char *aic3x_agc_decay[] = { "100ms", "200ms", "400ms", "500ms" }; +static const char *aic3x_mic_bias[] = { "off", "2V", "2.5V", "AVDD" }; #define LDAC_ENUM 0 #define RDAC_ENUM 1 @@ -206,6 +211,13 @@ #define LINE2L_ENUM 8 #define LINE2R_ENUM 9 #define ADC_HPF_ENUM 10 +#define LAGC_LEV_ENUM 11 +#define RAGC_LEV_ENUM 12 +#define LAGC_ATT_ENUM 13 +#define RAGC_ATT_ENUM 14 +#define LAGC_DEC_ENUM 15 +#define RAGC_DEC_ENUM 16 +#define MIC_BIAS_ENUM 17 static const struct soc_enum aic3x_enum[] = { SOC_ENUM_SINGLE(DAC_LINE_MUX, 6, 3, aic3x_left_dac_mux), @@ -219,6 +231,13 @@ SOC_ENUM_SINGLE(LINE2L_2_LADC_CTRL, 7, 2, aic3x_linein_mode_mux), SOC_ENUM_SINGLE(LINE2R_2_RADC_CTRL, 7, 2, aic3x_linein_mode_mux), SOC_ENUM_DOUBLE(AIC3X_CODEC_DFILT_CTRL, 6, 4, 4, aic3x_adc_hpf), + SOC_ENUM_SINGLE(LAGC_CTRL_A, 4, 8, aic3x_agc_level), + SOC_ENUM_SINGLE(RAGC_CTRL_A, 4, 8, aic3x_agc_level), + SOC_ENUM_SINGLE(LAGC_CTRL_A, 2, 4, aic3x_agc_attack), + SOC_ENUM_SINGLE(RAGC_CTRL_A, 2, 4, aic3x_agc_attack), + SOC_ENUM_SINGLE(LAGC_CTRL_A, 0, 4, aic3x_agc_decay), + SOC_ENUM_SINGLE(RAGC_CTRL_A, 0, 4, aic3x_agc_decay), + SOC_ENUM_SINGLE(MICBIAS_CTRL, 6, 4, aic3x_mic_bias), }; /* @@ -345,6 +364,15 @@ * adjust PGA to max value when ADC is on and will never go back. */ SOC_DOUBLE_R("AGC Switch", LAGC_CTRL_A, RAGC_CTRL_A, 7, 0x01, 0), + SOC_ENUM("Left AGC Target level", aic3x_enum[LAGC_LEV_ENUM]), + SOC_ENUM("Right AGC Target level", aic3x_enum[RAGC_LEV_ENUM]), + SOC_ENUM("Left AGC Attack time", aic3x_enum[LAGC_ATT_ENUM]), + SOC_ENUM("Right AGC Attack time", aic3x_enum[RAGC_ATT_ENUM]), + SOC_ENUM("Left AGC Decay time", aic3x_enum[LAGC_DEC_ENUM]), + SOC_ENUM("Right AGC Decay time", aic3x_enum[RAGC_DEC_ENUM]), + + /* De-emphasis */ + SOC_DOUBLE("De-emphasis Switch", AIC3X_CODEC_DFILT_CTRL, 2, 0, 0x01, 0), /* Input */ SOC_DOUBLE_R_TLV("PGA Capture Volume", LADC_VOL, RADC_VOL, @@ -352,6 +380,8 @@ SOC_DOUBLE_R("PGA Capture Switch", LADC_VOL, RADC_VOL, 7, 0x01, 1), SOC_ENUM("ADC HPF Cut-off", aic3x_enum[ADC_HPF_ENUM]), + + SOC_ENUM("Mic Bias", aic3x_enum[MIC_BIAS_ENUM]), }; /* @@ -1261,6 +1291,13 @@ snd_soc_write(codec, AIC3X_PAGE_SELECT, PAGE0_SELECT); snd_soc_write(codec, AIC3X_RESET, SOFT_RESET); + /* set to avoid artifacts on the audio output during power-on/off */ + snd_soc_write(codec, AIC3X_HEADSET_DETECT_CTRL_B, 0x80); /*ac-coupled*/ + snd_soc_write(codec, HPOUT_POP_REDUCTION, 0x4e); /* 10 + 4 ms, reference*/ + + /* short circuit protection */ + snd_soc_write(codec, HPRCOM_CFG, 0x04); + /* DAC default volume and mute */ snd_soc_write(codec, LDAC_VOL, DEFAULT_VOL | MUTE_ON); snd_soc_write(codec, RDAC_VOL, DEFAULT_VOL | MUTE_ON); @@ -1274,8 +1311,9 @@ snd_soc_write(codec, DACL1_2_LLOPM_VOL, DEFAULT_VOL | ROUTE_ON); snd_soc_write(codec, DACR1_2_RLOPM_VOL, DEFAULT_VOL | ROUTE_ON); /* DAC to Mono Line Out default volume and route to Output mixer */ - snd_soc_write(codec, DACL1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); - snd_soc_write(codec, DACR1_2_MONOLOPM_VOL, DEFAULT_VOL | ROUTE_ON); + /* mix both channels with -6dB level */ + snd_soc_write(codec, DACL1_2_MONOLOPM_VOL, (DEFAULT_VOL + 12) | ROUTE_ON); + snd_soc_write(codec, DACR1_2_MONOLOPM_VOL, (DEFAULT_VOL + 12) | ROUTE_ON); /* unmute all outputs */ snd_soc_update_bits(codec, LLOPM_CTRL, UNMUTE, UNMUTE); @@ -1292,6 +1330,12 @@ /* By default route Line1 to ADC PGA mixer */ snd_soc_write(codec, LINE1L_2_LADC_CTRL, 0x0); snd_soc_write(codec, LINE1R_2_RADC_CTRL, 0x0); + + /* AGC to -10dB, 20 / 500ms, no clip stepping, noise gate -90dB, hysteresis 3dB*/ + snd_soc_write(codec, LAGC_CTRL_A, 0x2f); + snd_soc_write(codec, RAGC_CTRL_A, 0x2f); + snd_soc_write(codec, LAGC_CTRL_C, 0xbe); + snd_soc_write(codec, RAGC_CTRL_C, 0xbe); /* PGA to HP Bypass default volume, disconnect from Output Mixer */ snd_soc_write(codec, PGAL_2_HPLOUT_VOL, DEFAULT_VOL);

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[alsa-devel] [PATCH V2 0/9] Add ASoC drivers for SPEAr platform & I2S driver for Designware module
by Rajeev Kumar 27 Jun '12

27 Jun '12

Changes since V1: 1. Move Synopsys i2s designware controller from 'spear' folder to a new folder 'dwc' 2. Implemented regmap in sta529 codec. 3. In-corporated review comments received on V1. Rajeev Kumar (6): sound:asoc: Add support for STA529 Audio Codec sound:asoc: Add support for synopsys i2s controller as per ASoC framework. sound:asoc: Add support for SPEAr ASoC pcm layer. sound:asoc: Add support for SPEAr ASoC machine driver. sound:asoc: Add Kconfig and Makefile to support SPEAr audio driver sound:asoc: Update Kconfig and Makefile(sound/soc/) to support SPEAr audio Vipin Kumar (3): sound:asoc: Add support for spdif in Audio Codec sound:asoc:spdif_in: Add spdif IN support sound:asoc:spdif_out: Add spdif out support include/sound/designware_i2s.h | 69 ++++++ include/sound/spear_dma.h | 35 +++ include/sound/spear_spdif.h | 29 +++ sound/soc/Kconfig | 2 + sound/soc/Makefile | 2 + sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 4 +- sound/soc/codecs/spdif_receiver.c | 76 ++++++ sound/soc/codecs/sta529.c | 477 +++++++++++++++++++++++++++++++++++++ sound/soc/dwc/Kconfig | 8 + sound/soc/dwc/Makefile | 3 + sound/soc/dwc/designware_i2s.c | 454 +++++++++++++++++++++++++++++++++++ sound/soc/spear/Kconfig | 40 +++ sound/soc/spear/Makefile | 7 + sound/soc/spear/spdif_in.c | 297 +++++++++++++++++++++++ sound/soc/spear/spdif_in_regs.h | 60 +++++ sound/soc/spear/spdif_out.c | 397 ++++++++++++++++++++++++++++++ sound/soc/spear/spdif_out_regs.h | 79 ++++++ sound/soc/spear/spear_evb.c | 206 ++++++++++++++++ sound/soc/spear/spear_pcm.c | 214 +++++++++++++++++ 20 files changed, 2462 insertions(+), 1 deletions(-) create mode 100644 include/sound/designware_i2s.h create mode 100644 include/sound/spear_dma.h create mode 100644 include/sound/spear_spdif.h create mode 100644 sound/soc/codecs/spdif_receiver.c create mode 100644 sound/soc/codecs/sta529.c create mode 100644 sound/soc/dwc/Kconfig create mode 100644 sound/soc/dwc/Makefile create mode 100644 sound/soc/dwc/designware_i2s.c create mode 100644 sound/soc/spear/Kconfig create mode 100644 sound/soc/spear/Makefile create mode 100644 sound/soc/spear/spdif_in.c create mode 100644 sound/soc/spear/spdif_in_regs.h create mode 100644 sound/soc/spear/spdif_out.c create mode 100644 sound/soc/spear/spdif_out_regs.h create mode 100644 sound/soc/spear/spear_evb.c create mode 100644 sound/soc/spear/spear_pcm.c

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[alsa-devel] [PATCH] ALSA: snd-usb-caiaq: initialize card pointer
by Daniel Mack 27 Jun '12

27 Jun '12

Fixes the following warning: CC [M] sound/usb/caiaq/device.o sound/usb/caiaq/device.c: In function ‘snd_probe’: sound/usb/caiaq/device.c:500:16: warning: ‘card’ may be used uninitialized in this function [-Wmaybe-uninitialized] Signed-off-by: Daniel Mack <zonque(a)gmail.com> --- sound/usb/caiaq/device.c | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 64aed43..7da0d0a 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -485,7 +485,7 @@ static int __devinit snd_probe(struct usb_interface *intf, const struct usb_device_id *id) { int ret; - struct snd_card *card; + struct snd_card *card = NULL; struct usb_device *device = interface_to_usbdev(intf); ret = create_card(device, intf, &card); -- 1.7.10.2

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[alsa-devel] [Regression] No sound on HP Pavilion dv6, STAC92xx
by Luis Henriques 27 Jun '12

27 Jun '12

Hi, After commit b0791dda813c179e539b0fc1ecd3f5f30f2571e2 ("ALSA: hda/idt - Fix power-map for speaker-pins with some HP laptops") we have several bug reports about not having sound on HP Pavilion laptops. Here's a link to one of these bug reports: http://bugs.launchpad.net/bugs/1013183 After reverting the referred commit, everything seems to be working fine again. Any ideas/fixes? Cheers, -- Luis

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[alsa-devel] [PATCH] ASoC: tlv320aic3x: add missing registers and bits
by Prchal Jiří 26 Jun '12

26 Jun '12

This patch adds missing registers and bits for TLV320AIC3106. Signed-off-by: Jiri Prchal <jiri.prchal(a)aksignal.cz> --- /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.c +++ /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.c.new @@ -97,32 +97,20 @@ * There is no point in caching the reset register */ static const u8 aic3x_reg[AIC3X_CACHEREGNUM] = { - 0x00, 0x00, 0x00, 0x10, /* 0 */ - 0x04, 0x00, 0x00, 0x00, /* 4 */ - 0x00, 0x00, 0x00, 0x01, /* 8 */ - 0x00, 0x00, 0x00, 0x80, /* 12 */ - 0x80, 0xff, 0xff, 0x78, /* 16 */ - 0x78, 0x78, 0x78, 0x78, /* 20 */ - 0x78, 0x00, 0x00, 0xfe, /* 24 */ - 0x00, 0x00, 0xfe, 0x00, /* 28 */ - 0x18, 0x18, 0x00, 0x00, /* 32 */ - 0x00, 0x00, 0x00, 0x00, /* 36 */ - 0x00, 0x00, 0x00, 0x80, /* 40 */ - 0x80, 0x00, 0x00, 0x00, /* 44 */ - 0x00, 0x00, 0x00, 0x04, /* 48 */ - 0x00, 0x00, 0x00, 0x00, /* 52 */ - 0x00, 0x00, 0x04, 0x00, /* 56 */ - 0x00, 0x00, 0x00, 0x00, /* 60 */ - 0x00, 0x04, 0x00, 0x00, /* 64 */ - 0x00, 0x00, 0x00, 0x00, /* 68 */ - 0x04, 0x00, 0x00, 0x00, /* 72 */ - 0x00, 0x00, 0x00, 0x00, /* 76 */ - 0x00, 0x00, 0x00, 0x00, /* 80 */ - 0x00, 0x00, 0x00, 0x00, /* 84 */ - 0x00, 0x00, 0x00, 0x00, /* 88 */ - 0x00, 0x00, 0x00, 0x00, /* 92 */ - 0x00, 0x00, 0x00, 0x00, /* 96 */ - 0x00, 0x00, 0x02, /* 100 */ + 0x00, 0x00, 0x00, 0x10, 0x04, 0x00, 0x00, 0x00, /* 0 */ + 0x00, 0x00, 0x00, 0x01, 0x00, 0x00, 0x00, 0x80, /* 8 */ + 0x80, 0xff, 0xff, 0x78, 0x78, 0x78, 0x78, 0x78, /* 16 */ + 0x78, 0x00, 0x00, 0xfe, 0x00, 0x00, 0xfe, 0x00, /* 24 */ + 0x18, 0x18, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 32 */ + 0x00, 0x00, 0x00, 0x80, 0x80, 0x00, 0x00, 0x00, /* 40 */ + 0x00, 0x00, 0x00, 0x04, 0x00, 0x00, 0x00, 0x00, /* 48 */ + 0x00, 0x00, 0x04, 0x00, 0x00, 0x00, 0x00, 0x00, /* 56 */ + 0x00, 0x04, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 64 */ + 0x04, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 72 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 80 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 88 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, 0x02, 0x00, /* 96 */ + 0x00, 0x00, 0x00, 0x00, 0x00, 0x00, /* 104 */ }; #define SOC_DAPM_SINGLE_AIC3X(xname, reg, shift, mask, invert) \ --- /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.h +++ /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.h.new @@ -13,7 +13,7 @@ #define _AIC3X_H /* AIC3X register space */ -#define AIC3X_CACHEREGNUM 103 +#define AIC3X_CACHEREGNUM 110 /* Page select register */ #define AIC3X_PAGE_SELECT 0 @@ -74,6 +74,8 @@ #define HPLCOM_CFG 37 /* Right High Power Output control registers */ #define HPRCOM_CFG 38 +/* High Power Output Stage Control Register */ +#define HPOUT_SC 40 /* DAC Output Switching control registers */ #define DAC_LINE_MUX 41 /* High Power Output Driver Pop Reduction registers */ @@ -148,6 +150,17 @@ #define AIC3X_GPIOB_REG 101 /* Clock generation control register */ #define AIC3X_CLKGEN_CTRL_REG 102 +/* New AGC registers */ +#define LAGCN_ATTACK 103 +#define LAGCN_DECAY 104 +#define RAGCN_ATTACK 105 +#define RAGCN_DECAY 106 +/* New Programmable ADC Digital Path and I2C Bus Condition Register */ +#define NEW_ADC_DIGITALPATH 107 +/* Passive Analog Signal Bypass Selection During Powerdown Register */ +#define PASSIVE_BYPASS 108 +/* DAC Quiescent Current Adjustment Register */ +#define DAC_ICC_ADJ 109 /* Page select register bits */ #define PAGE0_SELECT 0 @@ -163,6 +176,10 @@ #define DUAL_RATE_MODE ((1 << 5) | (1 << 6)) #define LDAC2LCH (0x1 << 3) #define RDAC2RCH (0x1 << 1) +#define LDAC2RCH (0x2 << 3) +#define RDAC2LCH (0x2 << 1) +#define LDAC2MONOMIX (0x3 << 3) +#define RDAC2MONOMIX (0x3 << 1) /* PLL registers bitfields */ #define PLLP_SHIFT 0

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Re: [alsa-devel] [PATCH V2] ASoC: tlv320aic3x: Fix codec pll configure bug
by Hebbar, Gururaja 26 Jun '12

26 Jun '12

On Tue, Jun 26, 2012 at 14:31:42, Prchal Jiří wrote: > Hi Gururaja, > shouldn't be better to use: > > snd_soc_update_bits(codec, AIC3X_PLL_PROGA_REG, PLLP_MASK, pll_p); > > instead of "read mask write" ? Sure will resend the patch. Thanks for the review > Even at this place you don't need to keep rest of register (PLL_Q and PLL_enable), so yust writing is OK. > > snd_soc_write(codec, AIC3X_PLL_PROGA_REG, pll_p << PLLP_SHIFT); > > Dne 26.6.2012 08:03, Hebbar, Gururaja napsal(a): > > In sound/soc/codecs/tlv320aic3x.c > > > > data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); > > snd_soc_write(codec, AIC3X_PLL_PROGA_REG, > > data | (pll_p<< PLLP_SHIFT)); > > > > In the above code, pll-p value is OR'ed with previous value without > > clearing it. Bug is not seen if pll-p value doesn't change across > > Sampling frequency. > > > > However on some platforms (like AM335x EVM-SK), pll-p may have different > > values across different sampling frequencies. In such case, above code > > configures the pll with a wrong value. > > Because of this bug, when a audio stream is played with pll value > > different from previous stream, audio is heard as differently. > > > > Signed-off-by: Hebbar, Gururaja<gururaja.hebbar(a)ti.com> > > --- > > chnages in V2: > > modify subject to indicate ASOC and codec as tlv320aic3x > > > > sound/soc/codecs/tlv320aic3x.c | 2 +- > > sound/soc/codecs/tlv320aic3x.h | 1 + > > 2 files changed, 2 insertions(+), 1 deletions(-) > > > > diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c > > index d0dbac1..8c1977b 100644 > > --- a/sound/soc/codecs/tlv320aic3x.c > > +++ b/sound/soc/codecs/tlv320aic3x.c > > @@ -963,7 +963,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, > > } > > > > found: > > - data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); > > + data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG)& ~PLLP_MASK; > > snd_soc_write(codec, AIC3X_PLL_PROGA_REG, > > data | (pll_p<< PLLP_SHIFT)); > > snd_soc_write(codec, AIC3X_OVRF_STATUS_AND_PLLR_REG, > > diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h > > index 06a1978..16d9999 100644 > > --- a/sound/soc/codecs/tlv320aic3x.h > > +++ b/sound/soc/codecs/tlv320aic3x.h > > @@ -166,6 +166,7 @@ > > > > /* PLL registers bitfields */ > > #define PLLP_SHIFT 0 > > +#define PLLP_MASK 7 > > #define PLLQ_SHIFT 3 > > #define PLLR_SHIFT 0 > > #define PLLJ_SHIFT 2 > > Regards, Gururaja

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[alsa-devel] [PATCH V2] ASoC: tlv320aic3x: Fix codec pll configure bug
by Hebbar, Gururaja 26 Jun '12

26 Jun '12

In sound/soc/codecs/tlv320aic3x.c data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); snd_soc_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT)); In the above code, pll-p value is OR'ed with previous value without clearing it. Bug is not seen if pll-p value doesn't change across Sampling frequency. However on some platforms (like AM335x EVM-SK), pll-p may have different values across different sampling frequencies. In such case, above code configures the pll with a wrong value. Because of this bug, when a audio stream is played with pll value different from previous stream, audio is heard as differently. Signed-off-by: Hebbar, Gururaja <gururaja.hebbar(a)ti.com> --- chnages in V2: modify subject to indicate ASOC and codec as tlv320aic3x sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/tlv320aic3x.h | 1 + 2 files changed, 2 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index d0dbac1..8c1977b 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -963,7 +963,7 @@ static int aic3x_hw_params(struct snd_pcm_substream *substream, } found: - data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG); + data = snd_soc_read(codec, AIC3X_PLL_PROGA_REG) & ~PLLP_MASK; snd_soc_write(codec, AIC3X_PLL_PROGA_REG, data | (pll_p << PLLP_SHIFT)); snd_soc_write(codec, AIC3X_OVRF_STATUS_AND_PLLR_REG, diff --git a/sound/soc/codecs/tlv320aic3x.h b/sound/soc/codecs/tlv320aic3x.h index 06a1978..16d9999 100644 --- a/sound/soc/codecs/tlv320aic3x.h +++ b/sound/soc/codecs/tlv320aic3x.h @@ -166,6 +166,7 @@ /* PLL registers bitfields */ #define PLLP_SHIFT 0 +#define PLLP_MASK 7 #define PLLQ_SHIFT 3 #define PLLR_SHIFT 0 #define PLLJ_SHIFT 2 -- 1.7.1

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[alsa-devel] [PATCH] ASoC: tlv320aic3x: add input clock selection
by Prchal Jiří 26 Jun '12

26 Jun '12

This patch adds input selection of main codec clock - from what pin. Tested with TLV320AIC3106 on BCLK and MCLK. Signed-off-by: Jiri Prchal <jiri.prchal(a)aksignal.cz> --- /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.c.orig +++ /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.c @@ -972,6 +972,10 @@ struct snd_soc_codec *codec = codec_dai->codec; struct aic3x_priv *aic3x = snd_soc_codec_get_drvdata(codec); + /* set clock on MCLK or GPIO2 or BCLK */ + snd_soc_update_bits(codec, AIC3X_CLKGEN_CTRL_REG, PLLCLK_IN_MASK, clk_id); + snd_soc_update_bits(codec, AIC3X_CLKGEN_CTRL_REG, CLKDIV_IN_MASK, clk_id); + aic3x->sysclk = freq; return 0; } --- /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.h.orig +++ /home/prchal/arm/fw-cdu/linux/linux-3.5-rc3/sound/soc/codecs/tlv320aic3x.h @@ -178,6 +178,13 @@ #define PLL_CLKIN_SHIFT 4 #define MCLK_SOURCE 0x0 #define PLL_CLKDIV_SHIFT 0 +#define PLLCLK_IN_MASK 0x30 +#define CLKDIV_IN_MASK 0xc0 +/* clock in source */ +#define CLKIN_MCLK 0 +#define CLKIN_GPIO2 1 +#define CLKIN_BCLK 2 + /* Software reset register bits */ #define SOFT_RESET 0x80

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