March 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH 2.6.30] ASoC: remove trigger delay in Freescale MPC8610 sound driver
by Timur Tabi 26 Mar '09

26 Mar '09

Remove the delay from the trigger function in the Freescale MPC8610 sound driver when capture is started. This delay was used to ensure that the DMA controller was active when ALSA call the .pointer function to request a DMA transfer status. A better approach is for the .pointer function to detect that DMA has not started, and return zero instead. This change eliminates the need for the delay. Also add some related code to check for a DMA programming error, and report XRUN if it occurs. Signed-off-by: Timur Tabi <timur(a)freescale.com> --- sound/soc/fsl/fsl_dma.c | 17 +++++++++++++++++ sound/soc/fsl/fsl_ssi.c | 20 ++------------------ 2 files changed, 19 insertions(+), 18 deletions(-) diff --git a/sound/soc/fsl/fsl_dma.c b/sound/soc/fsl/fsl_dma.c index b3eb857..2c4892c 100644 --- a/sound/soc/fsl/fsl_dma.c +++ b/sound/soc/fsl/fsl_dma.c @@ -697,6 +697,23 @@ static snd_pcm_uframes_t fsl_dma_pointer(struct snd_pcm_substream *substream) else position = in_be32(&dma_channel->dar); + /* + * When capture is started, the SSI immediately starts to fill its FIFO. + * This means that the DMA controller is not started until the FIFO is + * full. However, ALSA calls this function before that happens, when + * MR.DAR is still zero. In this case, just return zero to indicate + * that nothing has been received yet. + */ + if (!position) + return 0; + + if ((position < dma_private->dma_buf_phys) || + (position > dma_private->dma_buf_end)) { + dev_err(substream->pcm->card->dev, + "dma pointer is out of range, halting stream\n"); + return SNDRV_PCM_POS_XRUN; + } + frames = bytes_to_frames(runtime, position - dma_private->dma_buf_phys); /* diff --git a/sound/soc/fsl/fsl_ssi.c b/sound/soc/fsl/fsl_ssi.c index 169bca2..72823a2 100644 --- a/sound/soc/fsl/fsl_ssi.c +++ b/sound/soc/fsl/fsl_ssi.c @@ -466,28 +466,12 @@ static int fsl_ssi_trigger(struct snd_pcm_substream *substream, int cmd, case SNDRV_PCM_TRIGGER_START: clrbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN); case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_TE); - } else { - long timeout = jiffies + 10; - + else setbits32(&ssi->scr, CCSR_SSI_SCR_SSIEN | CCSR_SSI_SCR_RE); - - /* Wait until the SSI has filled its FIFO. Without this - * delay, ALSA complains about overruns. When the FIFO - * is full, the DMA controller initiates its first - * transfer. Until then, however, the DMA's DAR - * register is zero, which translates to an - * out-of-bounds pointer. This makes ALSA think an - * overrun has occurred. - */ - while (!(in_be32(&ssi->sisr) & CCSR_SSI_SISR_RFF0) && - (jiffies < timeout)); - if (!(in_be32(&ssi->sisr) & CCSR_SSI_SISR_RFF0)) - return -EIO; - } break; case SNDRV_PCM_TRIGGER_STOP: -- 1.6.0.6

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[alsa-devel] [RESEND] [PATCH] pxa2xx-ac97: fix displaying GSR after reset timeout
by Luotao Fu 26 Mar '09

26 Mar '09

the variable gsr_bit is set in isr. It is however set to 0 and interrupts are disabled prior to reset on pxa27x and pxa3xx. Hence it doesn't make a lot of sense to show the content of gsr_bit in case of a reset timeout. Signed-off-by: Luotao Fu <l.fu(a)pengutronix.de> --- sound/arm/pxa2xx-ac97-lib.c | 4 ++-- 1 files changed, 2 insertions(+), 2 deletions(-) diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 35afd0c..25933ba 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -218,7 +218,7 @@ bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97) if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) { printk(KERN_INFO "%s: warm reset timeout (GSR=%#lx)\n", - __func__, gsr_bits); + __func__, GSR | gsr_bits); return false; } @@ -248,7 +248,7 @@ bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97) if (!((GSR | gsr_bits) & (GSR_PCR | GSR_SCR))) { printk(KERN_INFO "%s: cold reset timeout (GSR=%#lx)\n", - __func__, gsr_bits); + __func__, GSR | gsr_bits); return false; } -- 1.5.6.5

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[alsa-devel] [patch] alsa: provide a more useful get_unmapped_area handler for pcm
by Daniel Glöckner 26 Mar '09

26 Mar '09

Shared memory mappings on nommu machines require a get_unmapped_area file operation that suggests an address for the mapping. The current implementation returns 0 and thus forces the driver to implement an mmap handler that fixes up the start and end address of the vma. This patch returns the address of the dma buffer, so it should work out of the box for all drivers that use the snd_pcm_runtime->dma_area pointer. Addresses for mapping the status and control pages are returned as well, but to make those work the conditional compilation of snd_pcm_mmap_{status,control} would need to be revised. Signed-off-by: Daniel Glöckner <dg(a)emlix.com> --- sound/core/pcm_native.c | 30 ++++++++++++++++++++++-------- 1 files changed, 22 insertions(+), 8 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index a789efc..09ec8d7 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -3377,14 +3377,28 @@ out: #endif /* CONFIG_SND_SUPPORT_OLD_API */ #ifndef CONFIG_MMU -unsigned long dummy_get_unmapped_area(struct file *file, unsigned long addr, - unsigned long len, unsigned long pgoff, - unsigned long flags) -{ - return 0; +static unsigned long snd_pcm_get_unmapped_area(struct file *file, + unsigned long addr, + unsigned long len, + unsigned long pgoff, + unsigned long flags) +{ + struct snd_pcm_file *pcm_file = file->private_data; + struct snd_pcm_substream *substream = pcm_file->substream; + struct snd_pcm_runtime *runtime = substream->runtime; + unsigned long offset = pgoff << PAGE_SHIFT; + + switch (offset) { + case SNDRV_PCM_MMAP_OFFSET_STATUS: + return (unsigned long)runtime->status; + case SNDRV_PCM_MMAP_OFFSET_CONTROL: + return (unsigned long)runtime->control; + default: + return (unsigned long)runtime->dma_area + offset; + } } #else -# define dummy_get_unmapped_area NULL +# define snd_pcm_get_unmapped_area NULL #endif /* @@ -3403,7 +3417,7 @@ const struct file_operations snd_pcm_f_ops[2] = { .compat_ioctl = snd_pcm_ioctl_compat, .mmap = snd_pcm_mmap, .fasync = snd_pcm_fasync, - .get_unmapped_area = dummy_get_unmapped_area, + .get_unmapped_area = snd_pcm_get_unmapped_area, }, { .owner = THIS_MODULE, @@ -3416,6 +3430,6 @@ const struct file_operations snd_pcm_f_ops[2] = { .compat_ioctl = snd_pcm_ioctl_compat, .mmap = snd_pcm_mmap, .fasync = snd_pcm_fasync, - .get_unmapped_area = dummy_get_unmapped_area, + .get_unmapped_area = snd_pcm_get_unmapped_area, } }; -- 1.6.2.107.ge47ee

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[alsa-devel] CA0106 mic not working, logs attached
by Steven Edwards 26 Mar '09

26 Mar '09

Dear ALSA, I installed Ubuntu 8.10 yesterday, updating ALSA to 1.0.19 to mute my speakers when the headset is plugged in. I unchecked the mute below the Microphone slider, maximized the volume, clicked Close. Sound Recorder detected nothing, so I returned to Volume Control->Recording and noticed that Microphone stayed checked. Repeat x3, same results. I then tried the PulseAudio setup outlined by markbuntu at: http://ubuntuforums.org/showthread.php?t=843012 That failed, so I tried soundrecorder's ALSA upgrade script at: http://ubuntuforums.org/showthread.php?t=1046137 This (sudo bash ./AlsaUpgrade-1.0.x-rev-1.16.sh -di) killed all sound and aplay -l detected no sound cards. Restore (sudo bash ./ AlsaUpgrade-1.0.x-rev-1.16.sh -r) didn't restore my speakers, so I ran alsa-info.sh: http://www.alsa-project.org/db/?f=5a408b2a5328279dbac25a54ce34f6ae707e7cf6 Next, retrying the update script didn't restore my speakers either. alsa-info.sh: http://www.alsa-project.org/db/?f=bd33dc839806cd151e21bdfe4c32bf02a0a9aa57 Finally, (sudo ./configure && sudo make && sudo make install) restored my speakers' ability to output sound, but the mic still isn't recording sound. The original problem--Volume Recorder not accepting/remembering/allowing my unmuted Microphone--remains. alsa-info.sh: http://www.alsa-project.org/db/?f=5e4bcbc453960ece52e3bf21fb475c768b50f979 I don't know where to go next. I hope the reports help and that you will be able to help me. Best, Steven

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[alsa-devel] [PATCH 1/2] ASoC: dapm - Add DAPM for always active DAC or ADC
by Joonyoung Shim 25 Mar '09

25 Mar '09

This patch supports DAPM widget for always active DAC or ADC regardless of stream. This widgets will be used for voice call at TWL4030 codec. TWL4030 codec use the digital signal for voice call, so it needs DAC and ADC to change signal to digital or analog, however currently DAC and ADC widgets cannot support voice call because they are activated only when playback or capture is executed. Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- Documentation/sound/alsa/soc/dapm.txt | 3 +++ include/sound/soc-dapm.h | 6 ++++++ 2 files changed, 9 insertions(+), 0 deletions(-) diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt index 9e67632..50a1e21 100644 --- a/Documentation/sound/alsa/soc/dapm.txt +++ b/Documentation/sound/alsa/soc/dapm.txt @@ -93,6 +93,9 @@ e.g. stream widgets for HiFi playback and capture SND_SOC_DAPM_DAC("HiFi DAC", "HiFi Playback", REG, 3, 1), SND_SOC_DAPM_ADC("HiFi ADC", "HiFi Capture", REG, 2, 1), +If you want always active DAC or ADC widget regardless of stream, you can use +SND_SOC_DAPM_DAC_ACTIVE or SND_SOC_DAPM_ADC_ACTIVE instead, and stream name +parameter isn't used. 2.2 Path Domain Widgets ----------------------- diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h index a7def6a..5e26f81 100644 --- a/include/sound/soc-dapm.h +++ b/include/sound/soc-dapm.h @@ -143,6 +143,12 @@ #define SND_SOC_DAPM_ADC(wname, stname, wreg, wshift, winvert) \ { .id = snd_soc_dapm_adc, .name = wname, .sname = stname, .reg = wreg, \ .shift = wshift, .invert = winvert} +#define SND_SOC_DAPM_DAC_ACTIVE(wname, wreg, wshift, winvert) \ +{ .id = snd_soc_dapm_dac, .name = wname, .reg = wreg, \ + .shift = wshift, .invert = winvert, .active = 1} +#define SND_SOC_DAPM_ADC_ACTIVE(wname, wreg, wshift, winvert) \ +{ .id = snd_soc_dapm_adc, .name = wname, .reg = wreg, \ + .shift = wshift, .invert = winvert, .active = 1} /* generic register modifier widget */ #define SND_SOC_DAPM_REG(wid, wname, wreg, wshift, wmask, won_val, woff_val) \ -- 1.5.6.3

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[alsa-devel] [PATCH 2/2] ASoC: twl4030 - Add VDL path support
by Joonyoung Shim 25 Mar '09

25 Mar '09

Adds DAPMs for VDL(voice down link) path. To support VDL path, we have to change DAPM of outputs(Earpiece, PreDrive Left/Right, Headset Left/Right, Carkit Left/Right) from mux to mixer, also codec mode from option1 to option2. Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- sound/soc/codecs/twl4030.c | 259 +++++++++++++++++++++----------------------- 1 files changed, 126 insertions(+), 133 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 97738e2..967048a 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -42,8 +42,8 @@ */ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* this register not used */ - 0x91, /* REG_CODEC_MODE (0x1) */ - 0xc3, /* REG_OPTION (0x2) */ + 0x90, /* REG_CODEC_MODE (0x1) */ + 0xd3, /* REG_OPTION (0x2) */ 0x00, /* REG_UNKNOWN (0x3) */ 0x00, /* REG_MICBIAS_CTL (0x4) */ 0x20, /* REG_ANAMICL (0x5) */ @@ -56,7 +56,7 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* REG_AVTXL2PGA (0xC) */ 0x00, /* REG_AVTXR2PGA (0xD) */ 0x01, /* REG_AUDIO_IF (0xE) */ - 0x00, /* REG_VOICE_IF (0xF) */ + 0xe3, /* REG_VOICE_IF (0xF) */ 0x00, /* REG_ARXR1PGA (0x10) */ 0x00, /* REG_ARXL1PGA (0x11) */ 0x6c, /* REG_ARXR2PGA (0x12) */ @@ -313,104 +313,60 @@ static void twl4030_power_down(struct snd_soc_codec *codec) } /* Earpiece */ -static const char *twl4030_earpiece_texts[] = - {"Off", "DACL1", "DACL2", "DACR1"}; - -static const unsigned int twl4030_earpiece_values[] = - {0x0, 0x1, 0x2, 0x4}; - -static const struct soc_enum twl4030_earpiece_enum = - SOC_VALUE_ENUM_SINGLE(TWL4030_REG_EAR_CTL, 1, 0x7, - ARRAY_SIZE(twl4030_earpiece_texts), - twl4030_earpiece_texts, - twl4030_earpiece_values); - -static const struct snd_kcontrol_new twl4030_dapm_earpiece_control = -SOC_DAPM_VALUE_ENUM("Route", twl4030_earpiece_enum); +static const struct snd_kcontrol_new twl4030_dapm_earpiece_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_EAR_CTL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_EAR_CTL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_EAR_CTL, 2, 1, 0), + SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_EAR_CTL, 3, 1, 0), +}; /* PreDrive Left */ -static const char *twl4030_predrivel_texts[] = - {"Off", "DACL1", "DACL2", "DACR2"}; - -static const unsigned int twl4030_predrivel_values[] = - {0x0, 0x1, 0x2, 0x4}; - -static const struct soc_enum twl4030_predrivel_enum = - SOC_VALUE_ENUM_SINGLE(TWL4030_REG_PREDL_CTL, 1, 0x7, - ARRAY_SIZE(twl4030_predrivel_texts), - twl4030_predrivel_texts, - twl4030_predrivel_values); - -static const struct snd_kcontrol_new twl4030_dapm_predrivel_control = -SOC_DAPM_VALUE_ENUM("Route", twl4030_predrivel_enum); +static const struct snd_kcontrol_new twl4030_dapm_predrivel_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_PREDL_CTL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_PREDL_CTL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_PREDL_CTL, 2, 1, 0), + SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_PREDL_CTL, 3, 1, 0), +}; /* PreDrive Right */ -static const char *twl4030_predriver_texts[] = - {"Off", "DACR1", "DACR2", "DACL2"}; - -static const unsigned int twl4030_predriver_values[] = - {0x0, 0x1, 0x2, 0x4}; - -static const struct soc_enum twl4030_predriver_enum = - SOC_VALUE_ENUM_SINGLE(TWL4030_REG_PREDR_CTL, 1, 0x7, - ARRAY_SIZE(twl4030_predriver_texts), - twl4030_predriver_texts, - twl4030_predriver_values); - -static const struct snd_kcontrol_new twl4030_dapm_predriver_control = -SOC_DAPM_VALUE_ENUM("Route", twl4030_predriver_enum); +static const struct snd_kcontrol_new twl4030_dapm_predriver_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_PREDR_CTL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_PREDR_CTL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_PREDR_CTL, 2, 1, 0), + SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_PREDR_CTL, 3, 1, 0), +}; /* Headset Left */ -static const char *twl4030_hsol_texts[] = - {"Off", "DACL1", "DACL2"}; - -static const struct soc_enum twl4030_hsol_enum = - SOC_ENUM_SINGLE(TWL4030_REG_HS_SEL, 1, - ARRAY_SIZE(twl4030_hsol_texts), - twl4030_hsol_texts); - -static const struct snd_kcontrol_new twl4030_dapm_hsol_control = -SOC_DAPM_ENUM("Route", twl4030_hsol_enum); +static const struct snd_kcontrol_new twl4030_dapm_hsol_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_HS_SEL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_HS_SEL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_HS_SEL, 2, 1, 0), +}; /* Headset Right */ -static const char *twl4030_hsor_texts[] = - {"Off", "DACR1", "DACR2"}; - -static const struct soc_enum twl4030_hsor_enum = - SOC_ENUM_SINGLE(TWL4030_REG_HS_SEL, 4, - ARRAY_SIZE(twl4030_hsor_texts), - twl4030_hsor_texts); - -static const struct snd_kcontrol_new twl4030_dapm_hsor_control = -SOC_DAPM_ENUM("Route", twl4030_hsor_enum); +static const struct snd_kcontrol_new twl4030_dapm_hsor_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_HS_SEL, 3, 1, 0), + SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_HS_SEL, 4, 1, 0), + SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_HS_SEL, 5, 1, 0), +}; /* Carkit Left */ -static const char *twl4030_carkitl_texts[] = - {"Off", "DACL1", "DACL2"}; - -static const struct soc_enum twl4030_carkitl_enum = - SOC_ENUM_SINGLE(TWL4030_REG_PRECKL_CTL, 1, - ARRAY_SIZE(twl4030_carkitl_texts), - twl4030_carkitl_texts); - -static const struct snd_kcontrol_new twl4030_dapm_carkitl_control = -SOC_DAPM_ENUM("Route", twl4030_carkitl_enum); +static const struct snd_kcontrol_new twl4030_dapm_carkitl_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_PRECKL_CTL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_PRECKL_CTL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_PRECKL_CTL, 2, 1, 0), +}; /* Carkit Right */ -static const char *twl4030_carkitr_texts[] = - {"Off", "DACR1", "DACR2"}; - -static const struct soc_enum twl4030_carkitr_enum = - SOC_ENUM_SINGLE(TWL4030_REG_PRECKR_CTL, 1, - ARRAY_SIZE(twl4030_carkitr_texts), - twl4030_carkitr_texts); - -static const struct snd_kcontrol_new twl4030_dapm_carkitr_control = -SOC_DAPM_ENUM("Route", twl4030_carkitr_enum); +static const struct snd_kcontrol_new twl4030_dapm_carkitr_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_PRECKR_CTL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_PRECKR_CTL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_PRECKR_CTL, 2, 1, 0), +}; /* Handsfree Left */ static const char *twl4030_handsfreel_texts[] = - {"Voice", "DACL1", "DACL2", "DACR2"}; + {"Voice", "AudioL1", "AudioL2", "AudioR2"}; static const struct soc_enum twl4030_handsfreel_enum = SOC_ENUM_SINGLE(TWL4030_REG_HFL_CTL, 0, @@ -422,7 +378,7 @@ SOC_DAPM_ENUM("Route", twl4030_handsfreel_enum); /* Handsfree Right */ static const char *twl4030_handsfreer_texts[] = - {"Voice", "DACR1", "DACR2", "DACL2"}; + {"Voice", "AudioR1", "AudioR2", "AudioL2"}; static const struct soc_enum twl4030_handsfreer_enum = SOC_ENUM_SINGLE(TWL4030_REG_HFR_CTL, 0, @@ -808,6 +764,12 @@ static int snd_soc_put_volsw_r2_twl4030(struct snd_kcontrol *kcontrol, } /* + * Voice RX GAIN volume control: + * from -37 to 12 dB in 1 dB steps (mute instead of -37 dB) + */ +static DECLARE_TLV_DB_SCALE(voice_rx_tlv, -3700, 100, 1); + +/* * FGAIN volume control: * from -62 to 0 dB in 1 dB steps (mute instead of -63 dB) */ @@ -856,6 +818,9 @@ static const struct soc_enum twl4030_rampdelay_enum = twl4030_rampdelay_texts); static const struct snd_kcontrol_new twl4030_snd_controls[] = { + SOC_SINGLE_TLV("DAC Voice RX Volume", + TWL4030_REG_VRXPGA, 0, 0x31, 0, voice_rx_tlv), + /* Common playback gain controls */ SOC_DOUBLE_R_TLV("DAC1 Digital Fine Playback Volume", TWL4030_REG_ARXL1PGA, TWL4030_REG_ARXR1PGA, @@ -871,6 +836,11 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { TWL4030_REG_ARXL2PGA, TWL4030_REG_ARXR2PGA, 6, 0x2, 0, digital_coarse_tlv), + SOC_SINGLE_TLV("DAC Voice Down Link Analog Volume", + TWL4030_REG_VDL_APGA_CTL, 3, 0x12, 1, analog_tlv), + SOC_SINGLE("DAC Voice Down Link Switch", + TWL4030_REG_VDL_APGA_CTL, 1, 1, 0), + SOC_DOUBLE_R_TLV("DAC1 Analog Playback Volume", TWL4030_REG_ARXL1_APGA_CTL, TWL4030_REG_ARXR1_APGA_CTL, 3, 0x12, 1, analog_tlv), @@ -940,6 +910,7 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HFR"), /* DACs */ + SND_SOC_DAPM_DAC_ACTIVE("DAC Voice", TWL4030_REG_AVDAC_CTL, 4, 0), SND_SOC_DAPM_DAC("DAC Right1", "Right Front Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DAC Left1", "Left Front Playback", @@ -950,6 +921,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_NOPM, 0, 0), /* Analog PGAs */ + SND_SOC_DAPM_PGA("VDL_APGA", TWL4030_REG_VDL_APGA_CTL, + 0, 0, NULL, 0), SND_SOC_DAPM_PGA("ARXR1_APGA", TWL4030_REG_ARXR1_APGA_CTL, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("ARXL1_APGA", TWL4030_REG_ARXL1_APGA_CTL, @@ -990,26 +963,35 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Analog L2 Playback Mixer", TWL4030_REG_AVDAC_CTL, 3, 0, NULL, 0), - /* Output MUX controls */ + /* Output MIXER controls */ /* Earpiece */ - SND_SOC_DAPM_VALUE_MUX("Earpiece Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_earpiece_control), + SND_SOC_DAPM_MIXER("Earpiece Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_earpiece_controls[0], + ARRAY_SIZE(twl4030_dapm_earpiece_controls)), /* PreDrivL/R */ - SND_SOC_DAPM_VALUE_MUX("PredriveL Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_predrivel_control), - SND_SOC_DAPM_VALUE_MUX("PredriveR Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_predriver_control), + SND_SOC_DAPM_MIXER("PredriveL Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_predrivel_controls[0], + ARRAY_SIZE(twl4030_dapm_predrivel_controls)), + SND_SOC_DAPM_MIXER("PredriveR Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_predriver_controls[0], + ARRAY_SIZE(twl4030_dapm_predriver_controls)), /* HeadsetL/R */ - SND_SOC_DAPM_MUX_E("HeadsetL Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_hsol_control, headsetl_event, - SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_MUX("HeadsetR Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_hsor_control), + SND_SOC_DAPM_MIXER_E("HeadsetL Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_hsol_controls[0], + ARRAY_SIZE(twl4030_dapm_hsol_controls), headsetl_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MIXER("HeadsetR Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_hsor_controls[0], + ARRAY_SIZE(twl4030_dapm_hsor_controls)), /* CarkitL/R */ - SND_SOC_DAPM_MUX("CarkitL Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_carkitl_control), - SND_SOC_DAPM_MUX("CarkitR Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_carkitr_control), + SND_SOC_DAPM_MIXER("CarkitL Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_carkitl_controls[0], + ARRAY_SIZE(twl4030_dapm_carkitl_controls)), + SND_SOC_DAPM_MIXER("CarkitR Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_carkitr_controls[0], + ARRAY_SIZE(twl4030_dapm_carkitr_controls)), + + /* Output MUX controls */ /* HandsfreeL/R */ SND_SOC_DAPM_MUX_E("HandsfreeL Mux", TWL4030_REG_HFL_CTL, 5, 0, &twl4030_dapm_handsfreel_control, handsfree_event, @@ -1064,6 +1046,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { }; static const struct snd_soc_dapm_route intercon[] = { + {"VDL_APGA", NULL, "DAC Voice"}, + {"Analog L1 Playback Mixer", NULL, "DAC Left1"}, {"Analog R1 Playback Mixer", NULL, "DAC Right1"}, {"Analog L2 Playback Mixer", NULL, "DAC Left2"}, @@ -1076,48 +1060,57 @@ static const struct snd_soc_dapm_route intercon[] = { /* Internal playback routings */ /* Earpiece */ - {"Earpiece Mux", "DACL1", "ARXL1_APGA"}, - {"Earpiece Mux", "DACL2", "ARXL2_APGA"}, - {"Earpiece Mux", "DACR1", "ARXR1_APGA"}, + {"Earpiece Mixer", "Voice", "VDL_APGA"}, + {"Earpiece Mixer", "AudioL1", "ARXL1_APGA"}, + {"Earpiece Mixer", "AudioL2", "ARXL2_APGA"}, + {"Earpiece Mixer", "AudioR1", "ARXR1_APGA"}, /* PreDrivL */ - {"PredriveL Mux", "DACL1", "ARXL1_APGA"}, - {"PredriveL Mux", "DACL2", "ARXL2_APGA"}, - {"PredriveL Mux", "DACR2", "ARXR2_APGA"}, + {"PredriveL Mixer", "Voice", "VDL_APGA"}, + {"PredriveL Mixer", "AudioL1", "ARXL1_APGA"}, + {"PredriveL Mixer", "AudioL2", "ARXL2_APGA"}, + {"PredriveL Mixer", "AudioR2", "ARXR2_APGA"}, /* PreDrivR */ - {"PredriveR Mux", "DACR1", "ARXR1_APGA"}, - {"PredriveR Mux", "DACR2", "ARXR2_APGA"}, - {"PredriveR Mux", "DACL2", "ARXL2_APGA"}, + {"PredriveR Mixer", "Voice", "VDL_APGA"}, + {"PredriveR Mixer", "AudioR1", "ARXR1_APGA"}, + {"PredriveR Mixer", "AudioR2", "ARXR2_APGA"}, + {"PredriveR Mixer", "AudioL2", "ARXL2_APGA"}, /* HeadsetL */ - {"HeadsetL Mux", "DACL1", "ARXL1_APGA"}, - {"HeadsetL Mux", "DACL2", "ARXL2_APGA"}, + {"HeadsetL Mixer", "Voice", "VDL_APGA"}, + {"HeadsetL Mixer", "AudioL1", "ARXL1_APGA"}, + {"HeadsetL Mixer", "AudioL2", "ARXL2_APGA"}, /* HeadsetR */ - {"HeadsetR Mux", "DACR1", "ARXR1_APGA"}, - {"HeadsetR Mux", "DACR2", "ARXR2_APGA"}, + {"HeadsetR Mixer", "Voice", "VDL_APGA"}, + {"HeadsetR Mixer", "AudioR1", "ARXR1_APGA"}, + {"HeadsetR Mixer", "AudioR2", "ARXR2_APGA"}, /* CarkitL */ - {"CarkitL Mux", "DACL1", "ARXL1_APGA"}, - {"CarkitL Mux", "DACL2", "ARXL2_APGA"}, + {"CarkitL Mixer", "Voice", "VDL_APGA"}, + {"CarkitL Mixer", "AudioL1", "ARXL1_APGA"}, + {"CarkitL Mixer", "AudioL2", "ARXL2_APGA"}, /* CarkitR */ - {"CarkitR Mux", "DACR1", "ARXR1_APGA"}, - {"CarkitR Mux", "DACR2", "ARXR2_APGA"}, + {"CarkitR Mixer", "Voice", "VDL_APGA"}, + {"CarkitR Mixer", "AudioR1", "ARXR1_APGA"}, + {"CarkitR Mixer", "AudioR2", "ARXR2_APGA"}, /* HandsfreeL */ - {"HandsfreeL Mux", "DACL1", "ARXL1_APGA"}, - {"HandsfreeL Mux", "DACL2", "ARXL2_APGA"}, - {"HandsfreeL Mux", "DACR2", "ARXR2_APGA"}, + {"HandsfreeL Mux", "Voice", "VDL_APGA"}, + {"HandsfreeL Mux", "AudioL1", "ARXL1_APGA"}, + {"HandsfreeL Mux", "AudioL2", "ARXL2_APGA"}, + {"HandsfreeL Mux", "AudioR2", "ARXR2_APGA"}, /* HandsfreeR */ - {"HandsfreeR Mux", "DACR1", "ARXR1_APGA"}, - {"HandsfreeR Mux", "DACR2", "ARXR2_APGA"}, - {"HandsfreeR Mux", "DACL2", "ARXL2_APGA"}, + {"HandsfreeR Mux", "Voice", "VDL_APGA"}, + {"HandsfreeR Mux", "AudioR1", "ARXR1_APGA"}, + {"HandsfreeR Mux", "AudioR2", "ARXR2_APGA"}, + {"HandsfreeR Mux", "AudioL2", "ARXL2_APGA"}, /* outputs */ {"OUTL", NULL, "ARXL2_APGA"}, {"OUTR", NULL, "ARXR2_APGA"}, - {"EARPIECE", NULL, "Earpiece Mux"}, - {"PREDRIVEL", NULL, "PredriveL Mux"}, - {"PREDRIVER", NULL, "PredriveR Mux"}, - {"HSOL", NULL, "HeadsetL Mux"}, - {"HSOR", NULL, "HeadsetR Mux"}, - {"CARKITL", NULL, "CarkitL Mux"}, - {"CARKITR", NULL, "CarkitR Mux"}, + {"EARPIECE", NULL, "Earpiece Mixer"}, + {"PREDRIVEL", NULL, "PredriveL Mixer"}, + {"PREDRIVER", NULL, "PredriveR Mixer"}, + {"HSOL", NULL, "HeadsetL Mixer"}, + {"HSOR", NULL, "HeadsetR Mixer"}, + {"CARKITL", NULL, "CarkitL Mixer"}, + {"CARKITR", NULL, "CarkitR Mixer"}, {"HFL", NULL, "HandsfreeL Mux"}, {"HFR", NULL, "HandsfreeR Mux"}, -- 1.5.6.3

3 4

Re: [alsa-devel] Acer Ferrari 5000
by Takashi Iwai 25 Mar '09

25 Mar '09

At Wed, 25 Mar 2009 01:47:16 -0700, Russ Dill wrote: > > The patch is causing breakage for me (no audio): > > [37998.770034] HDA Intel 0000:00:14.2: PCI INT A disabled > [38019.228311] HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, > low) -> IRQ 16 > [38019.360994] input: HDA Digital PCBeep as > /devices/pci0000:00/0000:00:14.2/input/input9 > [38019.389307] hda_codec: formats == 0 (nid=0x2, val=0xe0560, ovrd=1, > streams=0xe0560) > [38019.389318] hda_codec: cannot attach PCM stream 0 for codec #1 > [38019.389367] hda_codec: formats == 0 (nid=0x6, val=0x1e0560, ovrd=1, > streams=0x1e0560) > [38019.389373] hda_codec: cannot attach PCM stream 1 for codec #1 > [38019.389435] hda_codec: formats == 0 (nid=0x9, val=0x60160, ovrd=1, > streams=0x60160) > [38019.389440] hda_codec: cannot attach PCM stream 2 for codec #1 Looks like something wrong in my caching code, likely a wrong hash key generation or so. Will take a look after returning from the vacation. thanks, Takashi > > > On Mon, Mar 23, 2009 at 4:27 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > > At Mon, 23 Mar 2009 16:12:38 -0700, > > Russ Dill wrote: > >> > >> On Mon, Mar 23, 2009 at 3:55 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > >> > At Mon, 23 Mar 2009 15:48:19 -0700, > >> > Russ Dill wrote: > >> >> > >> >> On Mon, Mar 23, 2009 at 3:45 PM, Takashi Iwai <tiwai(a)suse.de> wrote: > >> >> > At Mon, 23 Mar 2009 15:28:57 -0700, > >> >> > Russ Dill wrote: > >> >> >> > >> >> >> >> > BTW, I found why model=auto doesn't work well on your machine. > >> >> >> >> > It's because (again) of BIOS. It doesn't set the codec SSID properly > >> >> >> >> > so the driver doesn't accept the HP auto-toggle and other features. > >> >> >> >> > > >> >> >> >> > So, in your case, using model=acer-aspire would be likely the best > >> >> >> >> > choice (supposing it matches with most functionality). If you have no > >> >> >> >> > problem (at least regressions) with this model, I'll add the entry to > >> >> >> >> > point to model=acer-aspire > >> >> >> >> > >> >> >> >> Add Acer Ferrar 5000 Quirk to Intel HDA driver > >> >> >> > > >> >> >> > Thanks, a similar fix was already merged now. > >> >> >> > >> >> >> It all works for a while, but eventually I'll get a "azx_get_response > >> >> >> timeout, switching to polling mode: last cmd=0x10bb2001" and it no > >> >> >> longer responds to headphone plug events without an unload/reload of > >> >> >> the module. > >> >> > > >> >> > Did you try my test patch? > >> >> > >> >> It is with the patch: > >> >> > >> >> [16877.337044] hda_intel: azx_get_response timeout, switching to > >> >> polling mode: last cmd=0x106f000a > >> >> [16878.341056] hda_intel: azx_get_response timeout (ERROR): last cmd=0x106f000a > >> >> [16879.345036] hda_intel: azx_get_response timeout (ERROR): last cmd=0x106f000a > >> >> [16880.380016] hda_intel: azx_get_response timeout (ERROR): last cmd=0x106f000a > >> >> [16881.384022] hda_intel: azx_get_response timeout (ERROR): last cmd=0x106f000a > >> > > >> > Interesting, this is again the same verb. > >> > So, there seems something unstable regarding the digital PCM... > >> > > >> > And do you still have the problem with HP plugging? > >> > > >> > >> There seems to be some confusion. All I'm doing right now is playing > >> audio through the analog PCM. After a while, I get that dmesg output > >> and HP plugging stops working. > > > > Yes, that's what I understood, too. The driver still queries the PCM > > parameter for the digital output widget because you turn on "IEC958 > > Default PCM" switch, thus the signal is still routed to the digital, > > too. > > > > > > The below is another test patch to reduce this problem (hopefully). > > It basically caches the parameter values to avoid superfluous > > queries. It's against the very latest alsa-driver snapshot. > > Apply it in addition to the previous test patch. > > > > > > thanks, > > > > Takashi > > > > --- > > > > diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c > > index a4e5e59..1d88e2f 100644 > > --- a/sound/pci/hda/hda_codec.c > > +++ b/sound/pci/hda/hda_codec.c > > @@ -1053,6 +1053,8 @@ EXPORT_SYMBOL_HDA(snd_hda_codec_cleanup_stream); > > /* FIXME: more better hash key? */ > > #define HDA_HASH_KEY(nid,dir,idx) (u32)((nid) + ((idx) << 16) + ((dir) << 24)) > > #define HDA_HASH_PINCAP_KEY(nid) (u32)((nid) + (0x02 << 24)) > > +#define HDA_HASH_PARPCM_KEY(nid) (u32)((nid) + (0x03 << 24)) > > +#define HDA_HASH_PARSTR_KEY(nid) (u32)((nid) + (0x04 << 24)) > > #define INFO_AMP_CAPS (1<<0) > > #define INFO_AMP_VOL(ch) (1 << (1 + (ch))) > > > > @@ -1143,7 +1145,9 @@ int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, > > } > > EXPORT_SYMBOL_HDA(snd_hda_override_amp_caps); > > > > -u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid) > > +static unsigned int > > +query_caps_hash(struct hda_codec *codec, hda_nid_t nid, u32 key, > > + unsigned int (*func)(struct hda_codec *, hda_nid_t)) > > { > > struct hda_amp_info *info; > > > > @@ -1151,11 +1155,22 @@ u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid) > > if (!info) > > return 0; > > if (!info->head.val) { > > - info->amp_caps = snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); > > info->head.val |= INFO_AMP_CAPS; > > + info->amp_caps = func(codec, nid); > > } > > return info->amp_caps; > > } > > + > > +static unsigned int read_pin_cap(struct hda_codec *codec, hda_nid_t nid) > > +{ > > + return snd_hda_param_read(codec, nid, AC_PAR_PIN_CAP); > > +} > > + > > +u32 snd_hda_query_pin_caps(struct hda_codec *codec, hda_nid_t nid) > > +{ > > + return query_caps_hash(codec, nid, HDA_HASH_PINCAP_KEY(nid), > > + read_pin_cap); > > +} > > EXPORT_SYMBOL_HDA(snd_hda_query_pin_caps); > > > > /* > > @@ -2538,6 +2553,41 @@ unsigned int snd_hda_calc_stream_format(unsigned int rate, > > } > > EXPORT_SYMBOL_HDA(snd_hda_calc_stream_format); > > > > +static unsigned int get_pcm_param(struct hda_codec *codec, hda_nid_t nid) > > +{ > > + unsigned int val = 0; > > + if (nid != codec->afg && > > + (get_wcaps(codec, nid) & AC_WCAP_FORMAT_OVRD)) > > + val = snd_hda_param_read(codec, nid, AC_PAR_PCM); > > + if (!val || val == -1) > > + val = snd_hda_param_read(codec, codec->afg, AC_PAR_PCM); > > + if (!val || val == -1) > > + return 0; > > + return val; > > +} > > + > > +static unsigned int query_pcm_param(struct hda_codec *codec, hda_nid_t nid) > > +{ > > + return query_caps_hash(codec, nid, HDA_HASH_PARPCM_KEY(nid), > > + get_pcm_param); > > +} > > + > > +static unsigned int get_stream_param(struct hda_codec *codec, hda_nid_t nid) > > +{ > > + unsigned int streams = snd_hda_param_read(codec, nid, AC_PAR_STREAM); > > + if (!streams || streams == -1) > > + streams = snd_hda_param_read(codec, codec->afg, AC_PAR_STREAM); > > + if (!streams || streams == -1) > > + return 0; > > + return streams; > > +} > > + > > +static unsigned int query_stream_param(struct hda_codec *codec, hda_nid_t nid) > > +{ > > + return query_caps_hash(codec, nid, HDA_HASH_PARSTR_KEY(nid), > > + get_stream_param); > > +} > > + > > /** > > * snd_hda_query_supported_pcm - query the supported PCM rates and formats > > * @codec: the HDA codec > > @@ -2556,15 +2606,8 @@ static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, > > { > > unsigned int i, val, wcaps; > > > > - val = 0; > > wcaps = get_wcaps(codec, nid); > > - if (nid != codec->afg && (wcaps & AC_WCAP_FORMAT_OVRD)) { > > - val = snd_hda_param_read(codec, nid, AC_PAR_PCM); > > - if (val == -1) > > - return -EIO; > > - } > > - if (!val) > > - val = snd_hda_param_read(codec, codec->afg, AC_PAR_PCM); > > + val = query_pcm_param(codec, nid); > > > > if (ratesp) { > > u32 rates = 0; > > @@ -2586,15 +2629,9 @@ static int snd_hda_query_supported_pcm(struct hda_codec *codec, hda_nid_t nid, > > u64 formats = 0; > > unsigned int streams, bps; > > > > - streams = snd_hda_param_read(codec, nid, AC_PAR_STREAM); > > - if (streams == -1) > > + streams = query_stream_param(codec, nid); > > + if (!streams) > > return -EIO; > > - if (!streams) { > > - streams = snd_hda_param_read(codec, codec->afg, > > - AC_PAR_STREAM); > > - if (streams == -1) > > - return -EIO; > > - } > > > > bps = 0; > > if (streams & AC_SUPFMT_PCM) { > > @@ -2668,17 +2705,9 @@ int snd_hda_is_supported_format(struct hda_codec *codec, hda_nid_t nid, > > int i; > > unsigned int val = 0, rate, stream; > > > > - if (nid != codec->afg && > > - (get_wcaps(codec, nid) & AC_WCAP_FORMAT_OVRD)) { > > - val = snd_hda_param_read(codec, nid, AC_PAR_PCM); > > - if (val == -1) > > - return 0; > > - } > > - if (!val) { > > - val = snd_hda_param_read(codec, codec->afg, AC_PAR_PCM); > > - if (val == -1) > > - return 0; > > - } > > + val = query_pcm_param(codec, nid); > > + if (!val) > > + return 0; > > > > rate = format & 0xff00; > > for (i = 0; i < AC_PAR_PCM_RATE_BITS; i++) > > @@ -2690,12 +2719,8 @@ int snd_hda_is_supported_format(struct hda_codec *codec, hda_nid_t nid, > > if (i >= AC_PAR_PCM_RATE_BITS) > > return 0; > > > > - stream = snd_hda_param_read(codec, nid, AC_PAR_STREAM); > > - if (stream == -1) > > - return 0; > > - if (!stream && nid != codec->afg) > > - stream = snd_hda_param_read(codec, codec->afg, AC_PAR_STREAM); > > - if (!stream || stream == -1) > > + stream = query_stream_param(codec, nid); > > + if (!stream) > > return 0; > > > > if (stream & AC_SUPFMT_PCM) { > > >

1 0

[alsa-devel] building soc module drivers
by Geiger Ho 25 Mar '09

25 Mar '09

Hi all, I would like to build the WM8510 SOC sound driver for my ARM platform. After downloading alsa-driver-1.0.19, "./configure --with-cross=arm-linux- --with-kernel=/home/source/linux-2.6.22.19 --with-build=/home/source/linux-2.6.22.19 --with-cards=all", and then make. I get no wm8510.c compiled. What other stuff should I do to make it compiled? Many thanks. Best regards, Geiger -- Geiger Ho <geiger.ho(a)fmp.fujitsu.com>, armed with Agile and Lean. Mission: To deliver quality of work. Phone : (852) 2736-3232 Direct : (852) 2737-9681 Fax : (852) 2314-4207 Fujitsu Microelectronics Pacific Asia Ltd, Design Centre

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Re: [alsa-devel] How to find sampling rate supported
by reetesh chandran 24 Mar '09

24 Mar '09

Thanks Takashi. If the lib does not provide the API to read the device capabilitiess, is there a way for applications to directly read it from the Alsa kernel driver ? --- On Mon, 3/23/09, Takashi Iwai <tiwai(a)suse.de> wrote: From: Takashi Iwai <tiwai(a)suse.de> Subject: Re: [alsa-devel] How to find sampling rate supported To: "reetesh chandran" <reeteshchandran(a)yahoo.com> Cc: alsa-devel(a)alsa-project.org Date: Monday, March 23, 2009, 3:03 AM At Sun, 22 Mar 2009 09:00:00 -0700 (PDT), reetesh chandran wrote: > > Hi all, > > I am new to the group and to Alsa. > > For our project, we have to find the max sampling rate and other pcm hardware parameters that the connected USB headset supports. I looked at the alsa library, but could not find an API which returned this information. > > I found snd_pcm_hw_params_test_* function to test a specific value is supported, but I was wondering if there was a way to query the device to get the values. The min / max can be obtained via *_get_rate_min() and *_get_rate_max(), at least. The other detailed information can't be get except for testing the explicit value via *_test_rate(), so far. HTH, Takashi _______________________________________________ Alsa-devel mailing list Alsa-devel(a)alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel

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[alsa-devel] [PATCH] ASoC: twl4030 - Add VDL path support
by Joonyoung Shim 24 Mar '09

24 Mar '09

Adds DAPMs for VDL(voice down link) path. To support VDL path, we have to change DAPM of outputs(Earpiece, PreDrive Left/Right, Headset Left/Right, Carkit Left/Right) from mux to mixer. Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- sound/soc/codecs/twl4030.c | 226 ++++++++++++++++++++------------------------ 1 files changed, 103 insertions(+), 123 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 5ed5f1b..317d710 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -313,104 +313,60 @@ static void twl4030_power_down(struct snd_soc_codec *codec) } /* Earpiece */ -static const char *twl4030_earpiece_texts[] = - {"Off", "DACL1", "DACL2", "DACR1"}; - -static const unsigned int twl4030_earpiece_values[] = - {0x0, 0x1, 0x2, 0x4}; - -static const struct soc_enum twl4030_earpiece_enum = - SOC_VALUE_ENUM_SINGLE(TWL4030_REG_EAR_CTL, 1, 0x7, - ARRAY_SIZE(twl4030_earpiece_texts), - twl4030_earpiece_texts, - twl4030_earpiece_values); - -static const struct snd_kcontrol_new twl4030_dapm_earpiece_control = -SOC_DAPM_VALUE_ENUM("Route", twl4030_earpiece_enum); +static const struct snd_kcontrol_new twl4030_dapm_earpiece_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_EAR_CTL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_EAR_CTL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_EAR_CTL, 2, 1, 0), + SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_EAR_CTL, 3, 1, 0), +}; /* PreDrive Left */ -static const char *twl4030_predrivel_texts[] = - {"Off", "DACL1", "DACL2", "DACR2"}; - -static const unsigned int twl4030_predrivel_values[] = - {0x0, 0x1, 0x2, 0x4}; - -static const struct soc_enum twl4030_predrivel_enum = - SOC_VALUE_ENUM_SINGLE(TWL4030_REG_PREDL_CTL, 1, 0x7, - ARRAY_SIZE(twl4030_predrivel_texts), - twl4030_predrivel_texts, - twl4030_predrivel_values); - -static const struct snd_kcontrol_new twl4030_dapm_predrivel_control = -SOC_DAPM_VALUE_ENUM("Route", twl4030_predrivel_enum); +static const struct snd_kcontrol_new twl4030_dapm_predrivel_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_PREDL_CTL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_PREDL_CTL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_PREDL_CTL, 2, 1, 0), + SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_PREDL_CTL, 3, 1, 0), +}; /* PreDrive Right */ -static const char *twl4030_predriver_texts[] = - {"Off", "DACR1", "DACR2", "DACL2"}; - -static const unsigned int twl4030_predriver_values[] = - {0x0, 0x1, 0x2, 0x4}; - -static const struct soc_enum twl4030_predriver_enum = - SOC_VALUE_ENUM_SINGLE(TWL4030_REG_PREDR_CTL, 1, 0x7, - ARRAY_SIZE(twl4030_predriver_texts), - twl4030_predriver_texts, - twl4030_predriver_values); - -static const struct snd_kcontrol_new twl4030_dapm_predriver_control = -SOC_DAPM_VALUE_ENUM("Route", twl4030_predriver_enum); +static const struct snd_kcontrol_new twl4030_dapm_predriver_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_PREDR_CTL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_PREDR_CTL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_PREDR_CTL, 2, 1, 0), + SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_PREDR_CTL, 3, 1, 0), +}; /* Headset Left */ -static const char *twl4030_hsol_texts[] = - {"Off", "DACL1", "DACL2"}; - -static const struct soc_enum twl4030_hsol_enum = - SOC_ENUM_SINGLE(TWL4030_REG_HS_SEL, 1, - ARRAY_SIZE(twl4030_hsol_texts), - twl4030_hsol_texts); - -static const struct snd_kcontrol_new twl4030_dapm_hsol_control = -SOC_DAPM_ENUM("Route", twl4030_hsol_enum); +static const struct snd_kcontrol_new twl4030_dapm_hsol_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_HS_SEL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_HS_SEL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_HS_SEL, 2, 1, 0), +}; /* Headset Right */ -static const char *twl4030_hsor_texts[] = - {"Off", "DACR1", "DACR2"}; - -static const struct soc_enum twl4030_hsor_enum = - SOC_ENUM_SINGLE(TWL4030_REG_HS_SEL, 4, - ARRAY_SIZE(twl4030_hsor_texts), - twl4030_hsor_texts); - -static const struct snd_kcontrol_new twl4030_dapm_hsor_control = -SOC_DAPM_ENUM("Route", twl4030_hsor_enum); +static const struct snd_kcontrol_new twl4030_dapm_hsor_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_HS_SEL, 3, 1, 0), + SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_HS_SEL, 4, 1, 0), + SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_HS_SEL, 5, 1, 0), +}; /* Carkit Left */ -static const char *twl4030_carkitl_texts[] = - {"Off", "DACL1", "DACL2"}; - -static const struct soc_enum twl4030_carkitl_enum = - SOC_ENUM_SINGLE(TWL4030_REG_PRECKL_CTL, 1, - ARRAY_SIZE(twl4030_carkitl_texts), - twl4030_carkitl_texts); - -static const struct snd_kcontrol_new twl4030_dapm_carkitl_control = -SOC_DAPM_ENUM("Route", twl4030_carkitl_enum); +static const struct snd_kcontrol_new twl4030_dapm_carkitl_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_PRECKL_CTL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioL1", TWL4030_REG_PRECKL_CTL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioL2", TWL4030_REG_PRECKL_CTL, 2, 1, 0), +}; /* Carkit Right */ -static const char *twl4030_carkitr_texts[] = - {"Off", "DACR1", "DACR2"}; - -static const struct soc_enum twl4030_carkitr_enum = - SOC_ENUM_SINGLE(TWL4030_REG_PRECKR_CTL, 1, - ARRAY_SIZE(twl4030_carkitr_texts), - twl4030_carkitr_texts); - -static const struct snd_kcontrol_new twl4030_dapm_carkitr_control = -SOC_DAPM_ENUM("Route", twl4030_carkitr_enum); +static const struct snd_kcontrol_new twl4030_dapm_carkitr_controls[] = { + SOC_DAPM_SINGLE("Voice", TWL4030_REG_PRECKR_CTL, 0, 1, 0), + SOC_DAPM_SINGLE("AudioR1", TWL4030_REG_PRECKR_CTL, 1, 1, 0), + SOC_DAPM_SINGLE("AudioR2", TWL4030_REG_PRECKR_CTL, 2, 1, 0), +}; /* Handsfree Left */ static const char *twl4030_handsfreel_texts[] = - {"Voice", "DACL1", "DACL2", "DACR2"}; + {"Voice", "AudioL1", "AudioL2", "AudioR2"}; static const struct soc_enum twl4030_handsfreel_enum = SOC_ENUM_SINGLE(TWL4030_REG_HFL_CTL, 0, @@ -422,7 +378,7 @@ SOC_DAPM_ENUM("Route", twl4030_handsfreel_enum); /* Handsfree Right */ static const char *twl4030_handsfreer_texts[] = - {"Voice", "DACR1", "DACR2", "DACL2"}; + {"Voice", "AudioR1", "AudioR2", "AudioL2"}; static const struct soc_enum twl4030_handsfreer_enum = SOC_ENUM_SINGLE(TWL4030_REG_HFR_CTL, 0, @@ -940,6 +896,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("HFR"), /* DACs */ + SND_SOC_DAPM_DAC("DAC Voice", "Voice Down Link", + TWL4030_REG_AVDAC_CTL, 4, 0), SND_SOC_DAPM_DAC("DAC Right1", "Right Front Playback", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_DAC("DAC Left1", "Left Front Playback", @@ -950,6 +908,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_NOPM, 0, 0), /* Analog PGAs */ + SND_SOC_DAPM_PGA("VDL_APGA", TWL4030_REG_VDL_APGA_CTL, + 0, 0, NULL, 0), SND_SOC_DAPM_PGA("ARXR1_APGA", TWL4030_REG_ARXR1_APGA_CTL, 0, 0, NULL, 0), SND_SOC_DAPM_PGA("ARXL1_APGA", TWL4030_REG_ARXL1_APGA_CTL, @@ -990,26 +950,35 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_MIXER("Analog L2 Playback Mixer", TWL4030_REG_AVDAC_CTL, 3, 0, NULL, 0), - /* Output MUX controls */ + /* Output MIXER controls */ /* Earpiece */ - SND_SOC_DAPM_VALUE_MUX("Earpiece Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_earpiece_control), + SND_SOC_DAPM_MIXER("Earpiece Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_earpiece_controls[0], + ARRAY_SIZE(twl4030_dapm_earpiece_controls)), /* PreDrivL/R */ - SND_SOC_DAPM_VALUE_MUX("PredriveL Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_predrivel_control), - SND_SOC_DAPM_VALUE_MUX("PredriveR Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_predriver_control), + SND_SOC_DAPM_MIXER("PredriveL Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_predrivel_controls[0], + ARRAY_SIZE(twl4030_dapm_predrivel_controls)), + SND_SOC_DAPM_MIXER("PredriveR Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_predriver_controls[0], + ARRAY_SIZE(twl4030_dapm_predriver_controls)), /* HeadsetL/R */ - SND_SOC_DAPM_MUX_E("HeadsetL Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_hsol_control, headsetl_event, - SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), - SND_SOC_DAPM_MUX("HeadsetR Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_hsor_control), + SND_SOC_DAPM_MIXER_E("HeadsetL Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_hsol_controls[0], + ARRAY_SIZE(twl4030_dapm_hsol_controls), headsetl_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MIXER("HeadsetR Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_hsor_controls[0], + ARRAY_SIZE(twl4030_dapm_hsor_controls)), /* CarkitL/R */ - SND_SOC_DAPM_MUX("CarkitL Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_carkitl_control), - SND_SOC_DAPM_MUX("CarkitR Mux", SND_SOC_NOPM, 0, 0, - &twl4030_dapm_carkitr_control), + SND_SOC_DAPM_MIXER("CarkitL Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_carkitl_controls[0], + ARRAY_SIZE(twl4030_dapm_carkitl_controls)), + SND_SOC_DAPM_MIXER("CarkitR Mixer", SND_SOC_NOPM, 0, 0, + &twl4030_dapm_carkitr_controls[0], + ARRAY_SIZE(twl4030_dapm_carkitr_controls)), + + /* Output MUX controls */ /* HandsfreeL/R */ SND_SOC_DAPM_MUX_E("HandsfreeL Mux", TWL4030_REG_HFL_CTL, 5, 0, &twl4030_dapm_handsfreel_control, handsfree_event, @@ -1064,6 +1033,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { }; static const struct snd_soc_dapm_route intercon[] = { + {"VDL_APGA", NULL, "DAC Voice"}, + {"Analog L1 Playback Mixer", NULL, "DAC Left1"}, {"Analog R1 Playback Mixer", NULL, "DAC Right1"}, {"Analog L2 Playback Mixer", NULL, "DAC Left2"}, @@ -1076,37 +1047,46 @@ static const struct snd_soc_dapm_route intercon[] = { /* Internal playback routings */ /* Earpiece */ - {"Earpiece Mux", "DACL1", "ARXL1_APGA"}, - {"Earpiece Mux", "DACL2", "ARXL2_APGA"}, - {"Earpiece Mux", "DACR1", "ARXR1_APGA"}, + {"Earpiece Mixer", "Voice", "VDL_APGA"}, + {"Earpiece Mixer", "AudioL1", "ARXL1_APGA"}, + {"Earpiece Mixer", "AudioL2", "ARXL2_APGA"}, + {"Earpiece Mixer", "AudioR1", "ARXR1_APGA"}, /* PreDrivL */ - {"PredriveL Mux", "DACL1", "ARXL1_APGA"}, - {"PredriveL Mux", "DACL2", "ARXL2_APGA"}, - {"PredriveL Mux", "DACR2", "ARXR2_APGA"}, + {"PredriveL Mixer", "Voice", "VDL_APGA"}, + {"PredriveL Mixer", "AudioL1", "ARXL1_APGA"}, + {"PredriveL Mixer", "AudioL2", "ARXL2_APGA"}, + {"PredriveL Mixer", "AudioR2", "ARXR2_APGA"}, /* PreDrivR */ - {"PredriveR Mux", "DACR1", "ARXR1_APGA"}, - {"PredriveR Mux", "DACR2", "ARXR2_APGA"}, - {"PredriveR Mux", "DACL2", "ARXL2_APGA"}, + {"PredriveR Mixer", "Voice", "VDL_APGA"}, + {"PredriveR Mixer", "AudioR1", "ARXR1_APGA"}, + {"PredriveR Mixer", "AudioR2", "ARXR2_APGA"}, + {"PredriveR Mixer", "AudioL2", "ARXL2_APGA"}, /* HeadsetL */ - {"HeadsetL Mux", "DACL1", "ARXL1_APGA"}, - {"HeadsetL Mux", "DACL2", "ARXL2_APGA"}, + {"HeadsetL Mixer", "Voice", "VDL_APGA"}, + {"HeadsetL Mixer", "AudioL1", "ARXL1_APGA"}, + {"HeadsetL Mixer", "AudioL2", "ARXL2_APGA"}, /* HeadsetR */ - {"HeadsetR Mux", "DACR1", "ARXR1_APGA"}, - {"HeadsetR Mux", "DACR2", "ARXR2_APGA"}, + {"HeadsetR Mixer", "Voice", "VDL_APGA"}, + {"HeadsetR Mixer", "AudioR1", "ARXR1_APGA"}, + {"HeadsetR Mixer", "AudioR2", "ARXR2_APGA"}, /* CarkitL */ - {"CarkitL Mux", "DACL1", "ARXL1_APGA"}, - {"CarkitL Mux", "DACL2", "ARXL2_APGA"}, + {"CarkitL Mixer", "Voice", "VDL_APGA"}, + {"CarkitL Mixer", "AudioL1", "ARXL1_APGA"}, + {"CarkitL Mixer", "AudioL2", "ARXL2_APGA"}, /* CarkitR */ - {"CarkitR Mux", "DACR1", "ARXR1_APGA"}, - {"CarkitR Mux", "DACR2", "ARXR2_APGA"}, + {"CarkitR Mixer", "Voice", "VDL_APGA"}, + {"CarkitR Mixer", "AudioR1", "ARXR1_APGA"}, + {"CarkitR Mixer", "AudioR2", "ARXR2_APGA"}, /* HandsfreeL */ - {"HandsfreeL Mux", "DACL1", "ARXL1_APGA"}, - {"HandsfreeL Mux", "DACL2", "ARXL2_APGA"}, - {"HandsfreeL Mux", "DACR2", "ARXR2_APGA"}, + {"HandsfreeL Mux", "Voice", "VDL_APGA"}, + {"HandsfreeL Mux", "AudioL1", "ARXL1_APGA"}, + {"HandsfreeL Mux", "AudioL2", "ARXL2_APGA"}, + {"HandsfreeL Mux", "AudioR2", "ARXR2_APGA"}, /* HandsfreeR */ - {"HandsfreeR Mux", "DACR1", "ARXR1_APGA"}, - {"HandsfreeR Mux", "DACR2", "ARXR2_APGA"}, - {"HandsfreeR Mux", "DACL2", "ARXL2_APGA"}, + {"HandsfreeR Mux", "Voice", "VDL_APGA"}, + {"HandsfreeR Mux", "AudioR1", "ARXR1_APGA"}, + {"HandsfreeR Mux", "AudioR2", "ARXR2_APGA"}, + {"HandsfreeR Mux", "AudioL2", "ARXL2_APGA"}, /* outputs */ {"OUTL", NULL, "ARXL2_APGA"}, -- 1.5.6.3

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