ASoC updates
Subsystems (6)
cover ASoC updates
The following changes since commit 0ee4663617fb0f78cec4cc6558a096ccbd8c3ffc:
Takashi Iwai (1):
ALSA: ASoC - Remove unnecessary inclusion of linux/version.h
are available in the git repository at:
git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai
David Anders (1):
ASoC: Add new parameter to s3c24xx_pcm_enqueue
Marek Vasut (1):
ASoC: Add Palm/PXA27x unified ASoC audio driver
Mark Brown (1):
ASoC: Remove core version number
Troy Kisky (1):
ASoC: TLV320AIC23B Support more sample rates
arch/arm/mach-pxa/include/mach/palmasoc.h | 13 ++ include/sound/soc.h | 2 - sound/soc/codecs/tlv320aic23.c | 227 +++++++++++++++++++----- sound/soc/pxa/Kconfig | 9 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/palm27x.c | 269 +++++++++++++++++++++++++++++ sound/soc/s3c24xx/s3c24xx-pcm.c | 12 +- sound/soc/soc-core.c | 1 - 8 files changed, 483 insertions(+), 52 deletions(-) create mode 100644 arch/arm/mach-pxa/include/mach/palmasoc.h create mode 100644 sound/soc/pxa/palm27x.c
PATCH 1/4 ASoC: TLV320AIC23B Support more sample rates
From: Troy Kisky <[email protected]>
Add support for more sample rates, different crystals and split playback/capture rates.
--- sound/soc/codecs/tlv320aic23.c | 227 ++++++++++++++++++++++++++++++++-------- 1 files changed, 182 insertions(+), 45 deletions(-)
sound/soc/codecs/tlv320aic23.c +182 -45
@@ -37,12 +37,6 @@#define AIC23_VERSION "0.1" -struct tlv320aic23_srate_reg_info { -u32 sample_rate; -u8 control;/* SR3, SR2, SR1, SR0 and BOSR */ -u8 divider;/* if 0 CLKIN = MCLK, if 1 CLKIN = MCLK/2 */ -}; -/* * AIC23 register cache */
@@ -261,19 +255,155 @@ static const struct snd_soc_dapm_route intercon[] = { }; -/* tlv320aic23 related */ -static const struct tlv320aic23_srate_reg_info srate_reg_info[] = { - {4000 ,0x06 ,1 },/* 4000 */ - {8000 ,0x06 ,0 },/* 8000 */ - {16000 ,0x0C ,1 },/* 16000 */ - {22050 ,0x11 ,1 },/* 22050 */ - {24000 ,0x00 ,1 },/* 24000 */ - {32000 ,0x0C ,0 },/* 32000 */ - {44100 ,0x11 ,0 },/* 44100 */ - {48000 ,0x00 ,0 },/* 48000 */ - {88200 ,0x1F ,0 },/* 88200 */ - {96000 ,0x0E ,0 },/* 96000 */ +/* AIC23 driver data */ +struct aic23 { +struct snd_soc_codec codec; +int mclk; +int requested_adc; +int requested_dac; +}; + +/* +* Common Crystals used +* 11.2896 Mhz /128 = *88.2k /192 = 58.8k +* 12.0000 Mhz /125 = *96k /136 = 88.235K +* 12.2880 Mhz /128 = *96k /192 = 64k +* 16.9344 Mhz /128 = 132.3k /192 = *88.2k +* 18.4320 Mhz /128 = 144k /192 = *96k +*/ + +/* +* Normal BOSR 0-256/2 = 128, 1-384/2 = 192 +* USB BOSR 0-250/2 = 125, 1-272/2 = 136 +*/ +static const int bosr_usb_divisor_table[] = { +128 ,125 ,192 ,136 +}; +#define LOWER_GROUP ((1<<0) | (1<<1) | (1<<2) | (1<<3) | (1<<6) | (1<<7)) +#define UPPER_GROUP ((1<<8) | (1<<9) | (1<<10) | (1<<11) | (1<<15)) +static const unsigned short sr_valid_mask[] = { +LOWER_GROUP |UPPER_GROUP ,/* Normal, bosr - 0*/ +LOWER_GROUP |UPPER_GROUP ,/* Normal, bosr - 1*/ +LOWER_GROUP ,/* Usb, bosr - 0*/ +UPPER_GROUP ,/* Usb, bosr - 1*/ +}; +/* +* Every divisor is a factor of 11*12 +*/ +#define SR_MULT (11*12) +#define A (x) (x) ? (SR_MULT/x) : 0 +static const unsigned char sr_adc_mult_table[] = { +A (2 ),A (2 ),A (12 ),A (12 ),A (0 ),A (0 ),A (3 ),A (1 ), +A (2 ),A (2 ),A (11 ),A (11 ),A (0 ),A (0 ),A (0 ),A (1 ) }; +static const unsigned char sr_dac_mult_table[] = { +A (2 ),A (12 ),A (2 ),A (12 ),A (0 ),A (0 ),A (3 ),A (1 ), +A (2 ),A (11 ),A (2 ),A (11 ),A (0 ),A (0 ),A (0 ),A (1 ) +}; + +unsigned get_score (int adc,int adc_l,int adc_h,int need_adc, +int dac,int dac_l,int dac_h,int need_dac) +{ +if ((adc >= adc_l) && (adc <= adc_h) && + (dac >= dac_l) && (dac <= dac_h)) { +unsigned diff_adc = need_adc - adc; +unsigned diff_dac = need_dac - dac; +unsigned score; +if (((int )diff_adc) <0 ) + diff_adc = -diff_adc; +if (((int )diff_dac) <0 ) + diff_dac = -diff_dac; + score = diff_adc + diff_dac; +return score; + } +return 0xffffffff ; +} +int find_rate (int mclk,u32 need_adc,u32 need_dac) +{ +int i, j; +int best_i =-1 ; +int best_j =-1 ; +int best_div =0 ; +unsigned best_score =0xffffffff ; +int adc_l, adc_h, dac_l, dac_h; + + need_adc *=SR_MULT ; + need_dac *=SR_MULT ; +/* +* rates given are +/- 1/32 +*/ + adc_l = need_adc - (need_adc >>5 ); + adc_h = need_adc + (need_adc >>5 ); + dac_l = need_dac - (need_dac >>5 ); + dac_h = need_dac + (need_dac >>5 ); +for (i =0 ; i <4 ; i++) { +int base = mclk / bosr_usb_divisor_table[i]; +int mask = sr_valid_mask[i]; +for (j =0 ; j <16 ; j++, mask >>=1 ) { +int adc; +int dac; +int score; +if ((mask &1 ) ==0 ) +continue ; + adc = base * sr_adc_mult_table[j]; + dac = base * sr_dac_mult_table[j]; + score =get_score (adc, adc_l, adc_h, need_adc, + dac, dac_l, dac_h, need_dac); +if (best_score > score) { + best_score = score; + best_i = i; + best_j = j; + best_div =0 ; + } + score =get_score ((adc >>1 ), adc_l, adc_h, need_adc, + (dac >>1 ), dac_l, dac_h, need_dac); +/* prefer to have a /2 */ +if ((score !=0xffffffff ) && (best_score >= score)) { + best_score = score; + best_i = i; + best_j = j; + best_div =1 ; + } + } + } +return (best_j <<2 ) | best_i | (best_div <<TLV320AIC23_CLKIN_SHIFT ); +} + +static void get_current_sample_rates (struct snd_soc_codec *codec,int mclk, +u32 *sample_rate_adc,u32 *sample_rate_dac) +{ +int src =tlv320aic23_read_reg_cache (codec,TLV320AIC23_SRATE ); +int sr = (src >>2 ) &0x0f ; +int val = (mclk / bosr_usb_divisor_table[src &3 ]); +int adc = (val * sr_adc_mult_table[sr]) /SR_MULT ; +int dac = (val * sr_dac_mult_table[sr]) /SR_MULT ; +if (src &TLV320AIC23_CLKIN_HALF ) { + adc >>=1 ; + dac >>=1 ; + } + *sample_rate_adc = adc; + *sample_rate_dac = dac; +} + +static int set_sample_rate_control (struct snd_soc_codec *codec,int mclk, +u32 sample_rate_adc,u32 sample_rate_dac) +{ +/* Search for the right sample rate */ +int data =find_rate (mclk, sample_rate_adc, sample_rate_dac); +if (data <0 ) { +printk (KERN_ERR "%s:Invalid rate %u,%u requested\n" , + __func__, sample_rate_adc, sample_rate_dac); +return -EINVAL ; + } +tlv320aic23_write (codec,TLV320AIC23_SRATE , data); +if (1 ) { +int adc, dac; +get_current_sample_rates (codec, mclk, &adc, &dac); +printk (KERN_DEBUG "actual samplerate = %u,%u reg=%x\n" , + adc, dac, data); + } +return 0 ; +}static int tlv320aic23_add_widgets (struct snd_soc_codec *codec) {
@@ -293,27 +423,30 @@ static int tlv320aic23_hw_params(struct snd_pcm_substream *substream,struct snd_soc_pcm_runtime *rtd = substream->private_data;struct snd_soc_device *socdev = rtd->socdev;struct snd_soc_codec *codec = socdev->codec; -u16 iface_reg, data; -u8 count =0 ; +u16 iface_reg; +int ret; + struct aic23 *aic23 = container_of(codec, struct aic23, codec); +u32 sample_rate_adc = aic23->requested_adc; +u32 sample_rate_dac = aic23->requested_dac; +u32 sample_rate =params_rate (params); + +if (substream->stream ==SNDRV_PCM_STREAM_PLAYBACK ) { + aic23->requested_dac = sample_rate_dac = sample_rate; +if (!sample_rate_adc) + sample_rate_adc = sample_rate; + }else { + aic23->requested_adc = sample_rate_adc = sample_rate; +if (!sample_rate_dac) + sample_rate_dac = sample_rate; + } + ret =set_sample_rate_control (codec, aic23->mclk, sample_rate_adc, + sample_rate_dac); +if (ret <0 ) +return ret; iface_reg =tlv320aic23_read_reg_cache (codec,TLV320AIC23_DIGT_FMT ) & ~(0x03 <<2 ); - -/* Search for the right sample rate */ -/* Verify what happens if the rate is not supported -* now it goes to 96Khz */ -while ((srate_reg_info[count].sample_rate !=params_rate (params)) && - (count <ARRAY_SIZE (srate_reg_info))) { - count++; - } - - data = (srate_reg_info[count].divider <<TLV320AIC23_CLKIN_SHIFT ) | - (srate_reg_info[count]. control <<TLV320AIC23_BOSR_SHIFT ) | -TLV320AIC23_USB_CLK_ON ; - -tlv320aic23_write (codec,TLV320AIC23_SRATE , data); -switch (params_format (params)) {case SNDRV_PCM_FORMAT_S16_LE :break ;
@@ -349,12 +482,17 @@ static void tlv320aic23_shutdown(struct snd_pcm_substream *substream)struct snd_soc_pcm_runtime *rtd = substream->private_data;struct snd_soc_device *socdev = rtd->socdev;struct snd_soc_codec *codec = socdev->codec; + struct aic23 *aic23 = container_of(codec, struct aic23, codec);/* deactivate */ if (!codec->active) {udelay (50 );tlv320aic23_write (codec,TLV320AIC23_ACTIVE ,0x0 ); } +if (substream->stream ==SNDRV_PCM_STREAM_PLAYBACK ) + aic23->requested_dac =0 ; +else + aic23->requested_adc =0 ; }static int tlv320aic23_mute (struct snd_soc_dai *dai,int mute)
@@ -422,12 +560,9 @@ static int tlv320aic23_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; - -switch (freq) { -case 12000000 : -return 0 ; - } -return -EINVAL ; + struct aic23 *aic23 = container_of(codec, struct aic23, codec); + aic23->mclk = freq; +return 0 ; } static int tlv320aic23_set_bias_level(struct snd_soc_codec *codec,
@@ -659,14 +794,15 @@ static int tlv320aic23_probe(struct platform_device *pdev) {struct snd_soc_device *socdev =platform_get_drvdata (pdev);struct snd_soc_codec *codec; +struct aic23 *aic23;int ret =0 ;printk (KERN_INFO "AIC23 Audio Codec %s\n" ,AIC23_VERSION ); - codec =kzalloc (sizeof (struct snd_soc_codec ),GFP_KERNEL ); -if (codec ==NULL ) + aic23 =kzalloc (sizeof (struct aic23 ),GFP_KERNEL ); +if (aic23 ==NULL )return -ENOMEM ; - + codec = &aic23->codec; socdev->codec = codec;mutex_init (&codec->mutex);INIT_LIST_HEAD (&codec->dapm_widgets);
@@ -687,6 +823,7 @@ static int tlv320aic23_remove(struct platform_device *pdev) {struct snd_soc_device *socdev =platform_get_drvdata (pdev);struct snd_soc_codec *codec = socdev->codec; + struct aic23 *aic23 = container_of(codec, struct aic23, codec);if (codec->control_data)tlv320aic23_set_bias_level (codec,SND_SOC_BIAS_OFF );
@@ -697,7 +834,7 @@ static int tlv320aic23_remove(struct platform_device *pdev)i2c_del_driver (&tlv320aic23_i2c_driver);#endif kfree (codec->reg_cache); -kfree (codec); +kfree (aic23);return 0 ; }
Signature · 1 line
1.5.6.5
PATCH 2/4 ASoC: Add Palm/PXA27x unified ASoC audio driver
From: Marek Vasut <[email protected]>
this patch adds asoc audio driver for pxa27x based Palm PDAs. I tested it for palmtx, t5 and ld, it should work with palmz72 as well (slapin, please test). I sent it here some time ago, but now I got to fixing bugs in it. It should be somehow mostly ok and ready for applying.
[Converted to use snd_soc_dapm_nc_pin() -- broonie]
--- arch/arm/mach-pxa/include/mach/palmasoc.h | 13 ++ sound/soc/pxa/Kconfig | 9 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/palm27x.c | 269 +++++++++++++++++++++++++++++ 4 files changed, 293 insertions(+), 0 deletions(-) create mode 100644 arch/arm/mach-pxa/include/mach/palmasoc.h create mode 100644 sound/soc/pxa/palm27x.c
arch/arm/mach-pxa/include/mach/palmasoc.h new file +13 -0
@@ -0,0 +1,13 @@ +#ifndef _INCLUDE_PALMASOC_H_ +#define _INCLUDE_PALMASOC_H_ +struct palm27x_asoc_info { +int jack_gpio; +}; + +#ifdef CONFIG_SND_PXA2XX_SOC_PALM27X +void __init palm27x_asoc_set_pdata (struct palm27x_asoc_info *data); +#else +static inline void palm27x_asoc_set_pdata (struct palm27x_asoc_info *data) {} +#endif + +#endif
sound/soc/pxa/Kconfig +9 -0
@@ -79,6 +79,15 @@ config SND_PXA2XX_SOC_EM_X270 Say Y if you want to add support for SoC audio on CompuLab EM-x270. +config SND_PXA2XX_SOC_PALM27X + tristate "SoC Audio support for Palm T|X, T5 and LifeDrive" + depends on SND_PXA2XX_SOC && (MACH_PALMLD || MACH_PALMTX || MACH_PALMT5) + select SND_PXA2XX_SOC_AC97 + select SND_SOC_WM9712 + help + Say Y if you want to add support for SoC audio on + Palm T|X, T5 or LifeDrive handheld computer. + config SND_SOC_ZYLONITE tristate "SoC Audio support for Marvell Zylonite" depends on SND_PXA2XX_SOC && MACH_ZYLONITE
sound/soc/pxa/Makefile +2 -0
@@ -16,6 +16,7 @@ snd-soc-tosa-objs := tosa.o snd-soc-e800-objs := e800_wm9712.o snd-soc-spitz-objs := spitz.o snd-soc-em-x270-objs := em-x270.o +snd-soc-palm27x-objs := palm27x.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o
@@ -23,3 +24,4 @@ obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o +obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.o
sound/soc/pxa/palm27x.c new file +269 -0
269 lines not on this page · raw message
Signature · 1 line
1.5.6.5
PATCH 3/4 ASoC: Remove core version number
Rather than try to remember to keep the core version number updated (which hasn't been happening) just remove it. It was much more useful when ASoC was out of tree.
--- include/sound/soc.h | 2 -- sound/soc/soc-core.c | 1 - 2 files changed, 0 insertions(+), 3 deletions(-)
include/sound/soc.h +0 -2
@@ -21,8 +21,6 @@#include <sound/control.h> #include <sound/ac97_codec.h> -#define SND_SOC_VERSION "0.13.2" -/* * Convenience kcontrol builders */
sound/soc/soc-core.c +0 -1
@@ -1965,7 +1965,6 @@ EXPORT_SYMBOL_GPL(snd_soc_dai_digital_mute);static int __devinit snd_soc_init (void ) { -printk (KERN_INFO "ASoC version %s\n" ,SND_SOC_VERSION );return platform_driver_register (&soc_driver); }
Signature · 1 line
1.5.6.5
PATCH 4/4 ASoC: Add new parameter to s3c24xx_pcm_enqueue
From: David Anders <[email protected]>
The S3C24xx dma does not allow more than one buffer to be enqueue prior to the dma transfers starting. This patch adds an additional parameter to s3c24xx_pcm_enqueue() to allow for passing an initial dma maximum load value.
--- sound/soc/s3c24xx/s3c24xx-pcm.c | 12 ++++++++---- 1 files changed, 8 insertions(+), 4 deletions(-)
sound/soc/s3c24xx/s3c24xx-pcm.c +8 -4
@@ -78,7 +78,8 @@ struct s3c24xx_runtime_data {* place a dma buffer onto the queue for the dma system * to handle. */ -static void s3c24xx_pcm_enqueue (struct snd_pcm_substream *substream) +static void s3c24xx_pcm_enqueue (struct snd_pcm_substream *substream, +int dma_max) {struct s3c24xx_runtime_data *prtd = substream->runtime->private_data;dma_addr_t pos = prtd->dma_pos;
@@ -86,7 +87,10 @@ static void s3c24xx_pcm_enqueue(struct snd_pcm_substream *substream)DBG ("Entered %s\n" , __func__); -while (prtd->dma_loaded < prtd->dma_limit) { +if (!dma_max) + dma_max = prtd->dma_limit; + +while (prtd->dma_loaded < dma_max) {unsigned long len = prtd->dma_period;DBG ("dma_loaded: %d\n" , prtd->dma_loaded);
@@ -132,7 +136,7 @@ static void s3c24xx_audio_buffdone(struct s3c2410_dma_chan *channel,spin_lock (&prtd->lock);if (prtd->state &ST_RUNNING ) { prtd->dma_loaded--; -s3c24xx_pcm_enqueue (substream); +s3c24xx_pcm_enqueue (substream,0 ); }spin_unlock (&prtd->lock);
@@ -249,7 +253,7 @@ static int s3c24xx_pcm_prepare(struct snd_pcm_substream *substream) prtd->dma_pos = prtd->dma_start;/* enqueue dma buffers */ -s3c24xx_pcm_enqueue (substream); +s3c24xx_pcm_enqueue (substream,1 );return ret; }
Signature · 1 line
1.5.6.5
Re: [PATCH 2/4] ASoC: Add Palm/PXA27x unified ASoC audio driver
In reply to Mark BrownQuoted 8 lines · On Wednesday 05 of November 2008 19:53:29 Mark Brown wrote:
From: Marek Vasut <[email protected]>
this patch adds asoc audio driver for pxa27x based Palm PDAs. I tested it for palmtx, t5 and ld, it should work with palmz72 as well (slapin, please test). I sent it here some time ago, but now I got to fixing bugs in it. It should be somehow mostly ok and ready for applying.
[Converted to use
snd_soc_dapm_nc_pin()-- broonie]
Mark, thanks a lot for fixing it, I'm really overloaded with school recently and I dont see it ending any soon.
Quoted 344 lines
Signed-off-by Marek Vasut <[email protected]>Signed-off-by Mark Brown <[email protected]>--- arch/arm/mach-pxa/include/mach/palmasoc.h | 13 ++ sound/soc/pxa/Kconfig | 9 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/palm27x.c | 269 +++++++++++++++++++++++++++++ 4 files changed, 293 insertions(+), 0 deletions(-) create mode 100644 arch/arm/mach-pxa/include/mach/palmasoc.h create mode 100644 sound/soc/pxa/palm27x.c(quoted diff) +0
-0
b/arch/arm/mach-pxa/include/mach/palmasoc.hnew file mode 100644 index 0000000..6c4b1f7arch/arm/mach-pxa/include/mach/palmasoc.h new file +10
-0@@ -0,0 +1,13 @@ +#ifndef _INCLUDE_PALMASOC_H_ +#define _INCLUDE_PALMASOC_H_ +struct palm27x_asoc_info { +int jack_gpio; +}; + +#ifdef CONFIG_SND_PXA2XX_SOC_PALM27X +void __init palm27x_asoc_set_pdata (struct palm27x_asoc_info *data); +#else +static inline void palm27x_asoc_set_pdata (struct palm27x_asoc_info *data){} +#endif
arch/arm/mach-pxa/include/mach/palmasoc.h new file +2
-0+ +#endif sound/soc/pxa/Kconfig +9
-0@@ -79,6 +79,15 @@ config SND_PXA2XX_SOC_EM_X270 Say Y if you want to add support for SoC audio on CompuLab EM-x270. +config SND_PXA2XX_SOC_PALM27X + tristate "SoC Audio support for Palm T|X, T5 and LifeDrive" + depends on SND_PXA2XX_SOC && (MACH_PALMLD || MACH_PALMTX || MACH_PALMT5) + select SND_PXA2XX_SOC_AC97 + select SND_SOC_WM9712 + help + Say Y if you want to add support for SoC audio on + Palm T|X, T5 or LifeDrive handheld computer. + config SND_SOC_ZYLONITE tristate "SoC Audio support for Marvell Zylonite" depends on SND_PXA2XX_SOC && MACH_ZYLONITEsound/soc/pxa/Makefile +2
-0@@ -16,6 +16,7 @@ snd-soc-tosa-objs := tosa.o snd-soc-e800-objs := e800_wm9712.o snd-soc-spitz-objs := spitz.o snd-soc-em-x270-objs := em-x270.o +snd-soc-palm27x-objs := palm27x.o obj-$(CONFIG_SND_PXA2XX_SOC_CORGI) += snd-soc-corgi.o obj-$(CONFIG_SND_PXA2XX_SOC_POODLE) += snd-soc-poodle.o@@ -23,3 +24,4 @@ obj-$(CONFIG_SND_PXA2XX_SOC_TOSA) += snd-soc-tosa.o obj-$(CONFIG_SND_PXA2XX_SOC_E800) += snd-soc-e800.o obj-$(CONFIG_SND_PXA2XX_SOC_SPITZ) += snd-soc-spitz.o obj-$(CONFIG_SND_PXA2XX_SOC_EM_X270) += snd-soc-em-x270.o +obj-$(CONFIG_SND_PXA2XX_SOC_PALM27X) += snd-soc-palm27x.osound/soc/pxa/palm27x.c new file +269
-0@@ -0,0 +1,269 @@ +/* +* linux/sound/soc/pxa/palm27x.c +* +* SoC Audio driver for Palm T|X, T5 and LifeDrive +* +* based on tosa.c +* +* Copyright (C) 2008 Marek Vasut <[email protected]> +* +* This program is free software; you can redistribute it and/or modify +* it under the terms of the GNU General Public License version 2 as +* published by the Free Software Foundation. +* +*/ + +#include <linux/module.h> +#include <linux/moduleparam.h> +#include <linux/device.h> +#include <linux/gpio.h> +#include <linux/interrupt.h> +#include <linux/irq.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> +#include <mach/audio.h> +#include <mach/palmasoc.h> + +#include "../codecs/wm9712.h" +#include "pxa2xx-pcm.h" +#include "pxa2xx-ac97.h" + +static int palm27x_jack_func =1 ; +static int palm27x_spk_func =1 ; +static int palm27x_ep_gpio =-1 ; + +static void palm27x_ext_control (struct snd_soc_codec *codec) +{ +if (!palm27x_spk_func) +snd_soc_dapm_enable_pin (codec,"Speaker" ); +else +snd_soc_dapm_disable_pin (codec,"Speaker" ); + +if (!palm27x_jack_func) +snd_soc_dapm_enable_pin (codec,"Headphone Jack" ); +else +snd_soc_dapm_disable_pin (codec,"Headphone Jack" ); + +snd_soc_dapm_sync (codec); +} + +static int palm27x_startup (struct snd_pcm_substream *substream) +{ +struct snd_soc_pcm_runtime *rtd = substream->private_data; +struct snd_soc_codec *codec = rtd->socdev->codec; + +/* check the jack status at stream startup */ +palm27x_ext_control (codec); +return 0 ; +} + +static struct snd_soc_ops palm27x_ops = { + .startup = palm27x_startup, +}; + +static irqreturn_t palm27x_interrupt (int irq,void *v) +{ + palm27x_spk_func =gpio_get_value (palm27x_ep_gpio); + palm27x_jack_func = !palm27x_spk_func; +return IRQ_HANDLED ; +} + +static int palm27x_get_jack (struct snd_kcontrol *kcontrol, +struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0 ] = palm27x_jack_func; +return 0 ; +} + +static int palm27x_set_jack (struct snd_kcontrol *kcontrol, +struct snd_ctl_elem_value *ucontrol) +{ +struct snd_soc_codec *codec =snd_kcontrol_chip (kcontrol); + +if (palm27x_jack_func == ucontrol->value.integer.value[0 ]) +return 0 ; + + palm27x_jack_func = ucontrol->value.integer.value[0 ]; +palm27x_ext_control (codec); +return 1 ; +} + +static int palm27x_get_spk (struct snd_kcontrol *kcontrol, +struct snd_ctl_elem_value *ucontrol) +{ + ucontrol->value.integer.value[0 ] = palm27x_spk_func; +return 0 ; +} + +static int palm27x_set_spk (struct snd_kcontrol *kcontrol, +struct snd_ctl_elem_value *ucontrol) +{ +struct snd_soc_codec *codec =snd_kcontrol_chip (kcontrol); + +if (palm27x_spk_func == ucontrol->value.integer.value[0 ]) +return 0 ; + + palm27x_spk_func = ucontrol->value.integer.value[0 ]; +palm27x_ext_control (codec); +return 1 ; +} + +/* PalmTX machine dapm widgets */ +static const struct snd_soc_dapm_widget palm27x_dapm_widgets[] = { +SND_SOC_DAPM_HP ("Headphone Jack" ,NULL ), +SND_SOC_DAPM_SPK ("Speaker" ,NULL ), +}; + +/* PalmTX audio map */ +static const struct snd_soc_dapm_route audio_map[] = { +/* headphone connected to HPOUTL, HPOUTR */ + {"Headphone Jack" ,NULL ,"HPOUTL" }, + {"Headphone Jack" ,NULL ,"HPOUTR" }, + +/* ext speaker connected to ROUT2, LOUT2 */ + {"Speaker" ,NULL ,"LOUT2" }, + {"Speaker" ,NULL ,"ROUT2" }, +}; + +static const char *jack_function[] = {"Headphone" ,"Off" }; +static const char *spk_function[] = {"On" ,"Off" }; +static const struct soc_enum palm27x_enum[] = { +SOC_ENUM_SINGLE_EXT (2 , jack_function), +SOC_ENUM_SINGLE_EXT (2 , spk_function), +}; + +static const struct snd_kcontrol_new palm27x_controls[] = { +SOC_ENUM_EXT ("Jack Function" , palm27x_enum[0 ], palm27x_get_jack, + palm27x_set_jack), +SOC_ENUM_EXT ("Speaker Function" , palm27x_enum[1 ], palm27x_get_spk, + palm27x_set_spk), +}; + +static int palm27x_ac97_init (struct snd_soc_codec *codec) +{ +int i, err; + +snd_soc_dapm_nc_pin (codec,"OUT3" ); +snd_soc_dapm_nc_pin (codec,"MONOOUT" ); + +/* add palm27x specific controls */ +for (i =0 ; i <ARRAY_SIZE (palm27x_controls); i++) { + err =snd_ctl_add (codec->card, +snd_soc_cnew (&palm27x_controls[i], + codec,NULL )); +if (err <0 ) +return err; + } + +/* add palm27x specific widgets */ +snd_soc_dapm_new_controls (codec, palm27x_dapm_widgets, +ARRAY_SIZE (palm27x_dapm_widgets)); + +/* set up palm27x specific audio path audio_map */ +snd_soc_dapm_add_routes (codec, audio_map,ARRAY_SIZE (audio_map)); + +snd_soc_dapm_sync (codec); +return 0 ; +} + +static struct snd_soc_dai_link palm27x_dai[] = { +{ + .name ="AC97 HiFi" , + .stream_name ="AC97 HiFi" , + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_HIFI ], + .codec_dai = &wm9712_dai[WM9712_DAI_AC97_HIFI ], + .init = palm27x_ac97_init, + .ops = &palm27x_ops, +}, +{ + .name ="AC97 Aux" , + .stream_name ="AC97 Aux" , + .cpu_dai = &pxa_ac97_dai[PXA2XX_DAI_AC97_AUX ], + .codec_dai = &wm9712_dai[WM9712_DAI_AC97_AUX ], + .ops = &palm27x_ops, +}, +}; + +static struct snd_soc_machine palm27x_asoc = { + .name ="Palm/PXA27x" , + .dai_link = palm27x_dai, + .num_links =ARRAY_SIZE (palm27x_dai), +}; + +static struct snd_soc_device palm27x_snd_devdata = { + .machine = &palm27x_asoc, + .platform = &pxa2xx_soc_platform, + .codec_dev = &soc_codec_dev_wm9712, +}; + +static struct platform_device *palm27x_snd_device; + +static int __init palm27x_asoc_init (void ) +{ +int ret; + +if (!(machine_is_palmtx () ||machine_is_palmt5 () || +machine_is_palmld ())) +return -ENODEV ; + + ret =gpio_request (palm27x_ep_gpio,"Headphone Jack" ); +if (ret) +return ret; + ret =gpio_direction_input (palm27x_ep_gpio); +if (ret) + goto err_alloc; + +if (request_irq (gpio_to_irq (palm27x_ep_gpio), palm27x_interrupt, +IRQF_TRIGGER_RISING |IRQF_TRIGGER_FALLING , +"Headphone jack" ,NULL )) + goto err_alloc; + + palm27x_snd_device =platform_device_alloc ("soc-audio" ,-1 ); +if (!palm27x_snd_device) { + ret = -ENOMEM ; + goto err_dev; + } + +platform_set_drvdata (palm27x_snd_device, &palm27x_snd_devdata); + palm27x_snd_devdata.dev = &palm27x_snd_device->dev; + ret =platform_device_add (palm27x_snd_device); + +if (ret !=0 ) + goto put_device; + +return 0 ; + +put_device: +platform_device_put (palm27x_snd_device); +err_dev: +free_irq (gpio_to_irq (palm27x_ep_gpio),NULL ); +err_alloc: +gpio_free (palm27x_ep_gpio); + +return ret; +} + +static void __exit palm27x_asoc_exit(void) +{ +free_irq (gpio_to_irq (palm27x_ep_gpio),NULL ); +gpio_free (palm27x_ep_gpio); +platform_device_unregister (palm27x_snd_device); +} + +void __init palm27x_asoc_set_pdata (struct palm27x_asoc_info *data) +{ + palm27x_ep_gpio = data->jack_gpio; +} + +module_init (palm27x_asoc_init); +module_exit (palm27x_asoc_exit); + +/* Module information */ +MODULE_AUTHOR ("Marek Vasut <[email protected]>" ); +MODULE_DESCRIPTION ("ALSA SoC Palm T|X, T5 and LifeDrive" ); +MODULE_LICENSE ("GPL" );
Re: [PATCH 2/4] ASoC: Add Palm/PXA27x unified ASoC audio driver
In reply to Mark BrownAt Wed, 5 Nov 2008 18:53:29 +0000,
Quoted 27 lines · Mark Brown
From: Marek Vasut <[email protected]>
this patch adds asoc audio driver for pxa27x based Palm PDAs. I tested it for palmtx, t5 and ld, it should work with palmz72 as well (slapin, please test). I sent it here some time ago, but now I got to fixing bugs in it. It should be somehow mostly ok and ready for applying.
[Converted to use
snd_soc_dapm_nc_pin()-- broonie]Signed-off-by Marek Vasut <[email protected]>Signed-off-by Mark Brown <[email protected]>--- arch/arm/mach-pxa/include/mach/palmasoc.h | 13 ++ sound/soc/pxa/Kconfig | 9 + sound/soc/pxa/Makefile | 2 + sound/soc/pxa/palm27x.c | 269 +++++++++++++++++++++++++++++ 4 files changed, 293 insertions(+), 0 deletions(-) create mode 100644 arch/arm/mach-pxa/include/mach/palmasoc.h create mode 100644 sound/soc/pxa/palm27x.carch/arm/mach-pxa/include/mach/palmasoc.h new file +0
-0
arch/arm/mach-pxa/include/mach/palmasoc.h line 8
Quoted 8 lines
arch/arm/mach-pxa/include/mach/palmasoc.h new file +8
-0@@ -0,0 +1,13 @@ +#ifndef _INCLUDE_PALMASOC_H_ +#define _INCLUDE_PALMASOC_H_ +struct palm27x_asoc_info { +int jack_gpio; +}; + +#ifdef CONFIG_SND_PXA2XX_SOC_PALM27X +void __init palm27x_asoc_set_pdata (struct palm27x_asoc_info *data);
I wonder whether it's really __init.
Note that the driver can be a module in theory.
Takashi
Re: [PATCH 2/4] ASoC: Add Palm/PXA27x unified ASoC audio driver
Quoted 4 lines · Takashi Iwai
arch/arm/mach-pxa/include/mach/palmasoc.h line 8
Quoted 2 lines
arch/arm/mach-pxa/include/mach/palmasoc.h new file +2
-0+#ifdef CONFIG_SND_PXA2XX_SOC_PALM27X +void __init palm27x_asoc_set_pdata (struct palm27x_asoc_info *data);I wonder whether it's really
__init.
Yes. This will be set up by the machine init code which has to be built in. It should be platform data but we can't support that yet.
Quoted 1 line
Note that the driver can be a module in theory.
Yeah, that practice won't actually build so should probably be masked out in Kconfig.
Re: [PATCH 1/4] ASoC: TLV320AIC23B Support more sample rates
In reply to Mark BrownAt Wed, 5 Nov 2008 18:53:28 +0000,
Quoted 17 lines · Mark Brown
From: Troy Kisky <[email protected]>
Add support for more sample rates, different crystals and split playback/capture rates.
Signed-off-by Troy Kisky <[email protected]>Acked-by Arun KS <[email protected]>Signed-off-by Mark Brown <[email protected]>--- sound/soc/codecs/tlv320aic23.c | 227 ++++++++++++++++++++++++++++++++-------- 1 files changed, 182 insertions(+), 45 deletions(-)sound/soc/codecs/tlv320aic23.c +0
-0
sound/soc/codecs/tlv320aic23.c static const struct snd_soc_dapm_route intercon[] = { · line 305
Quoted 2 lines
sound/soc/codecs/tlv320aic23.c +2
-0+unsigned get_score (int adc,int adc_l,int adc_h,int need_adc, +int dac,int dac_l,int dac_h,int need_dac)
Should be static.
Quoted 4 lines
sound/soc/codecs/tlv320aic23.c +4
-0+{ +if ((adc >= adc_l) && (adc <= adc_h) && + (dac >= dac_l) && (dac <= dac_h)) { +unsigned diff_adc = need_adc - adc;
sound/soc/codecs/tlv320aic23.c static const struct snd_soc_dapm_route intercon[] = { · line 319
Quoted 10 lines
sound/soc/codecs/tlv320aic23.c +10
-0+unsigned diff_dac = need_dac - dac; +unsigned score; +if (((int )diff_adc) <0 ) + diff_adc = -diff_adc; +if (((int )diff_dac) <0 ) + diff_dac = -diff_dac; + score = diff_adc + diff_dac; +return score; + } +return 0xffffffff ;
-1 would be simpler and safer (for possible missing f's).
sound/soc/codecs/tlv320aic23.c
Quoted 2 lines
sound/soc/codecs/tlv320aic23.c +2
-0+} +int find_rate (int mclk,u32 need_adc,u32 need_dac)
Use static.
sound/soc/codecs/tlv320aic23.c static const struct snd_soc_dapm_route intercon[] = { · line 327
Quoted 6 lines
sound/soc/codecs/tlv320aic23.c +6
-0+{ +int i, j; +int best_i =-1 ; +int best_j =-1 ; +int best_div =0 ; +unsigned best_score =0xffffffff ;
-1 would be safer.
Quoted 62 lines
sound/soc/codecs/tlv320aic23.c +62
-0+int adc_l, adc_h, dac_l, dac_h; + + need_adc *=SR_MULT ; + need_dac *=SR_MULT ; +/* +* rates given are +/- 1/32 +*/ + adc_l = need_adc - (need_adc >>5 ); + adc_h = need_adc + (need_adc >>5 ); + dac_l = need_dac - (need_dac >>5 ); + dac_h = need_dac + (need_dac >>5 ); +for (i =0 ; i <4 ; i++) { +int base = mclk / bosr_usb_divisor_table[i]; +int mask = sr_valid_mask[i]; +for (j =0 ; j <16 ; j++, mask >>=1 ) { +int adc; +int dac; +int score; +if ((mask &1 ) ==0 ) +continue ; + adc = base * sr_adc_mult_table[j]; + dac = base * sr_dac_mult_table[j]; + score =get_score (adc, adc_l, adc_h, need_adc, + dac, dac_l, dac_h, need_dac); +if (best_score > score) { + best_score = score; + best_i = i; + best_j = j; + best_div =0 ; + } + score =get_score ((adc >>1 ), adc_l, adc_h, need_adc, + (dac >>1 ), dac_l, dac_h, need_dac); +/* prefer to have a /2 */ +if ((score !=0xffffffff ) && (best_score >= score)) { + best_score = score; + best_i = i; + best_j = j; + best_div =1 ; + } + } + } +return (best_j <<2 ) | best_i | (best_div <<TLV320AIC23_CLKIN_SHIFT ); +} + +static void get_current_sample_rates (struct snd_soc_codec *codec,int mclk, +u32 *sample_rate_adc,u32 *sample_rate_dac) +{ +int src =tlv320aic23_read_reg_cache (codec,TLV320AIC23_SRATE ); +int sr = (src >>2 ) &0x0f ; +int val = (mclk / bosr_usb_divisor_table[src &3 ]); +int adc = (val * sr_adc_mult_table[sr]) /SR_MULT ; +int dac = (val * sr_dac_mult_table[sr]) /SR_MULT ; +if (src &TLV320AIC23_CLKIN_HALF ) { + adc >>=1 ; + dac >>=1 ; + } + *sample_rate_adc = adc; + *sample_rate_dac = dac; +} + +static int set_sample_rate_control (struct snd_soc_codec *codec,int mclk, +u32 sample_rate_adc,u32 sample_rate_dac)
sound/soc/codecs/tlv320aic23.c static const struct snd_soc_dapm_route intercon[] = { · line 399
Quoted 10 lines
sound/soc/codecs/tlv320aic23.c +10
-0+{ +/* Search for the right sample rate */ +int data =find_rate (mclk, sample_rate_adc, sample_rate_dac); +if (data <0 ) { +printk (KERN_ERR "%s:Invalid rate %u,%u requested\n" , + __func__, sample_rate_adc, sample_rate_dac); +return -EINVAL ; + } +tlv320aic23_write (codec,TLV320AIC23_SRATE , data); +if (1 ) {
For what?
Takashi
Re: [PATCH 1/4] ASoC: TLV320AIC23B Support more sample rates
Quoted 5 lines · Takashi Iwai
sound/soc/codecs/tlv320aic23.c static const struct snd_soc_dapm_route intercon[] = { · line 399
Quoted 1 line · Mark Brown
sound/soc/codecs/tlv320aic23.c +1
-0+if (1 ) {For what?
That should be an ifdef DEBUG - sorry, looks like I've sent an old version of the patch. My bad, sorry. Troy, when addressing Takashi's other comments please send an incremental patch.
PATCH 1/1 ASoC: TLV320AIC23B Support more sample rates
Add support for more sample rates, different crystals and split playback/capture rates.
Mark, this is an incremental patch to the #ifdef DEBUG version
sound/soc/codecs/tlv320aic23.c +9 -14
@@ -301,30 +301,25 @@ static const unsigned char sr_dac_mult_table[] = {A (2 ),A (11 ),A (2 ),A (11 ),A (0 ),A (0 ),A (0 ),A (1 ) }; -unsigned get_score (int adc,int adc_l,int adc_h,int need_adc, +static unsigned get_score (int adc,int adc_l,int adc_h,int need_adc,int dac,int dac_l,int dac_h,int need_dac) {if ((adc >= adc_l) && (adc <= adc_h) && (dac >= dac_l) && (dac <= dac_h)) { -unsigned diff_adc = need_adc - adc; -unsigned diff_dac = need_dac - dac; -unsigned score; -if (((int )diff_adc) <0 ) - diff_adc = -diff_adc; -if (((int )diff_dac) <0 ) - diff_dac = -diff_dac; - score = diff_adc + diff_dac; -return score; +int diff_adc = need_adc - adc; +int diff_dac = need_dac - dac; +return abs (diff_adc) +abs (diff_dac); } -return 0xffffffff ; +return UINT_MAX ; } -int find_rate (int mclk,u32 need_adc,u32 need_dac) + +static int find_rate (int mclk,u32 need_adc,u32 need_dac) {int i, j;int best_i =-1 ;int best_j =-1 ;int best_div =0 ; -unsigned best_score =0xffffffff ; +unsigned best_score =UINT_MAX ;int adc_l, adc_h, dac_l, dac_h; need_adc *=SR_MULT ;
@@ -359,7 +354,7 @@ int find_rate(int mclk, u32 need_adc, u32 need_dac) score =get_score ((adc >>1 ), adc_l, adc_h, need_adc, (dac >>1 ), dac_l, dac_h, need_dac);/* prefer to have a /2 */ -if ((score !=0xffffffff ) && (best_score >= score)) { +if ((score !=UINT_MAX ) && (best_score >= score)) { best_score = score; best_i = i; best_j = j;
Signature · 1 line
1.5.4.3
Re: [PATCH 1/1] [ALSA] ASoC: TLV320AIC23B Support more sample rates
Quoted 6 lines · Troy Kisky
Add support for more sample rates, different crystals and split playback/capture rates.
Signed-off-by Troy Kisky <[email protected]>Mark, this is an incremental patch to the #ifdef DEBUG version
I'll squash this down and push it today or Monday.
Re: [PATCH 1/4] ASoC: TLV320AIC23B Support more sample rates
In reply to Takashi IwaiQuoted 7 lines · Takashi Iwai
sound/soc/codecs/tlv320aic23.c static const struct snd_soc_dapm_route intercon[] = { · line 319
Quoted 3 lines · Mark Brown
sound/soc/codecs/tlv320aic23.c +2
-0+return score; +return 0xffffffff ;-1 would be simpler and safer (for possible missing f's).
I'm not sure that'd add to the clarity - the goal isn't to have all bits set, the goal is to have a much larger score than could ever otherwise be achieved so putting in a very large constant expresses that much more directly.
Re: [PATCH 1/4] ASoC: TLV320AIC23B Support more sample rates
At Thu, 6 Nov 2008 10:49:06 +0000,
Quoted 6 lines · Mark Brown
Quoted 3 lines · Takashi Iwai
Quoted 1 line · Mark Brown
sound/soc/codecs/tlv320aic23.c +1
-0+return score;
Quoted 8 lines
Quoted 3 lines
sound/soc/codecs/tlv320aic23.c static const struct snd_soc_dapm_route intercon[] = { · line 319
Quoted 1 line
sound/soc/codecs/tlv320aic23.c +1
-0+return 0xffffffff ;-1 would be simpler and safer (for possible missing f's).
I'm not sure that'd add to the clarity - the goal isn't to have all bits set, the goal is to have a much larger score than could ever otherwise be achieved so putting in a very large constant expresses that much more directly.
In that case, the best form is UINT_MAX.
Takashi
Re: [PATCH 1/4] ASoC: TLV320AIC23B Support more sample rates
In reply to Mark BrownAt Thu, 6 Nov 2008 10:49:06 +0000,
Quoted 6 lines · Mark Brown
Quoted 3 lines · Takashi Iwai
Quoted 1 line · Mark Brown
sound/soc/codecs/tlv320aic23.c +1
-0+return score;
Quoted 8 lines
Quoted 3 lines
sound/soc/codecs/tlv320aic23.c static const struct snd_soc_dapm_route intercon[] = { · line 319
Quoted 1 line
sound/soc/codecs/tlv320aic23.c +1
-0+return 0xffffffff ;-1 would be simpler and safer (for possible missing f's).
I'm not sure that'd add to the clarity - the goal isn't to have all bits set, the goal is to have a much larger score than could ever otherwise be achieved so putting in a very large constant expresses that much more directly.
Also, regarding unsigned integer:
Quoted 4 lines
sound/soc/codecs/tlv320aic23.c +4
-0+unsigned get_score (int adc,int adc_l,int adc_h,int need_adc, +int dac,int dac_l,int dac_h,int need_dac) +{ + if ((adc >= adc_l) && (adc <= adc_h) &&
sound/soc/codecs/tlv320aic23.c static const struct snd_soc_dapm_route intercon[] = { · line 317
Quoted 10 lines
sound/soc/codecs/tlv320aic23.c +10
-0+ (dac >= dac_l) && (dac <= dac_h)) { +unsigned diff_adc = need_adc - adc; +unsigned diff_dac = need_dac - dac; +unsigned score; +if (((int )diff_adc) <0 ) + diff_adc = -diff_adc; +if (((int )diff_dac) <0 ) + diff_dac = -diff_dac; + score = diff_adc + diff_dac; +return score;
It's more readable when diff_adc, diff_dac are int so that you have no ugly cast:
int diff_adc = need_adc - adc; int diff_dac = need_dac - dac; unsigned score; if (diff_adc < 0) diff_adc = -diff_adc; if (diff_dac < 0) diff_dac = -diff_dac; score = diff_adc + diff_dac; return score;
And, even more readable with abs() (not surprisingly defined in linux/kernel.h):
int diff_adc = need_adc - adc; int diff_dac = need_dac - dac; return abs(diff_adc) + abs(diff_dac);
Takashi