[alsa-devel] [PATCH 0/4] ASoC updates
The following changes since commit e18c94d20224f3df584531a48d944d8cccfda46d: Grazvydas Ignotas (1): ALSA: ASoC: TWL4030 codec - fix 256*Fs clock
are available in the git repository at:
git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai
Christian Pellegrin (1): ASoC: s3c24xx 8 bit sound fix
Hugo Villeneuve (1): ASoC: Add Right-Justified mode and Codec clock master to davinci-i2s
Naresh Medisetty (1): ASoC: DaVinci: Audio: Fix swapping of channels at start of stereo playback
Troy Kisky (1): ASoC: TLV320AIC23B Support more sample rates
sound/soc/codecs/tlv320aic23.c | 222 +++++++++++++++++++++++++++++++-------- sound/soc/davinci/davinci-i2s.c | 89 ++++++++++++++-- sound/soc/s3c24xx/s3c24xx-i2s.c | 7 ++ 3 files changed, 263 insertions(+), 55 deletions(-)
From: Troy Kisky troy.kisky@boundarydevices.com
Add support for more sample rates, different crystals and split playback/capture rates.
Signed-off-by: Troy Kisky troy.kisky@boundarydevices.com Acked-by: Arun KS arunks@mistralsolutions.com Signed-off-by: Mark Brown broonie@opensource.wolfsonmicro.com --- sound/soc/codecs/tlv320aic23.c | 222 ++++++++++++++++++++++++++++++++-------- 1 files changed, 177 insertions(+), 45 deletions(-)
diff --git a/sound/soc/codecs/tlv320aic23.c b/sound/soc/codecs/tlv320aic23.c index 44308da..a95b538 100644 --- a/sound/soc/codecs/tlv320aic23.c +++ b/sound/soc/codecs/tlv320aic23.c @@ -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 */ @@ -261,20 +255,151 @@ 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) };
+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)) { + int diff_adc = need_adc - adc; + int diff_dac = need_dac - dac; + return abs(diff_adc) + abs(diff_dac); + } + return UINT_MAX; +} + +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 = UINT_MAX; + 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 != UINT_MAX) && (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) { snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets, @@ -293,27 +418,30 @@ 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 +477,17 @@ 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 +555,9 @@ 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 +789,15 @@ 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 +818,7 @@ 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 +829,7 @@ 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; }
From: Christian Pellegrin chripell@gmail.com
fixes playing/recording of 8 bit audio files.
Generated on 20081108 against v2.6.27
Signed-off-by: Christian Pellegrin chripell@fsfe.org Signed-off-by: Mark Brown broonie@opensource.wolfsonmicro.com --- sound/soc/s3c24xx/s3c24xx-i2s.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-)
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index ba4476b..c18977b 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -261,10 +261,17 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream,
switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8: + iismod &= ~S3C2410_IISMOD_16BIT; + ((struct s3c24xx_pcm_dma_params *) + rtd->dai->cpu_dai->dma_data)->dma_size = 1; break; case SNDRV_PCM_FORMAT_S16_LE: iismod |= S3C2410_IISMOD_16BIT; + ((struct s3c24xx_pcm_dma_params *) + rtd->dai->cpu_dai->dma_data)->dma_size = 2; break; + default: + return -EINVAL; }
writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD);
From: Hugo Villeneuve hugo.villeneuve@lyrtech.com
The TI DVEVM board uses the SND_SOC_DAIFMT_CBM_CFM & I2S formats, but the Lyrtech SFFSDR board uses the SND_SOC_DAIFMT_CBM_CFS & RIGHT-JUSTIFIED formats.
Signed-off-by: Hugo Villeneuve hugo.villeneuve@lyrtech.com Signed-off-by: Mark Brown broonie@opensource.wolfsonmicro.com --- sound/soc/davinci/davinci-i2s.c | 40 ++++++++++++++++++++++++++++++++------ 1 files changed, 33 insertions(+), 7 deletions(-)
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index abb5fed..d814ec8 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -59,6 +59,7 @@ #define DAVINCI_MCBSP_PCR_CLKXP (1 << 1) #define DAVINCI_MCBSP_PCR_FSRP (1 << 2) #define DAVINCI_MCBSP_PCR_FSXP (1 << 3) +#define DAVINCI_MCBSP_PCR_SCLKME (1 << 7) #define DAVINCI_MCBSP_PCR_CLKRM (1 << 8) #define DAVINCI_MCBSP_PCR_CLKXM (1 << 9) #define DAVINCI_MCBSP_PCR_FSRM (1 << 10) @@ -171,6 +172,16 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, DAVINCI_MCBSP_SRGR_FSGM); break; + case SND_SOC_DAIFMT_CBM_CFS: + /* McBSP CLKR pin is the input for the Sample Rate Generator. + * McBSP FSR and FSX are driven by the Sample Rate Generator. */ + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, + DAVINCI_MCBSP_PCR_SCLKME | + DAVINCI_MCBSP_PCR_FSXM | + DAVINCI_MCBSP_PCR_FSRM); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, + DAVINCI_MCBSP_SRGR_FSGM); + break; case SND_SOC_DAIFMT_CBM_CFM: davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, 0); break; @@ -205,6 +216,28 @@ static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai, return -EINVAL; }
+ switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_RIGHT_J: + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, + DAVINCI_MCBSP_RCR_RFRLEN1(1) | + DAVINCI_MCBSP_RCR_RDATDLY(0)); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, + DAVINCI_MCBSP_XCR_XFRLEN1(1) | + DAVINCI_MCBSP_XCR_XDATDLY(0) | + DAVINCI_MCBSP_XCR_XFIG); + break; + case SND_SOC_DAIFMT_I2S: + default: + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, + DAVINCI_MCBSP_RCR_RFRLEN1(1) | + DAVINCI_MCBSP_RCR_RDATDLY(1)); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, + DAVINCI_MCBSP_XCR_XFRLEN1(1) | + DAVINCI_MCBSP_XCR_XDATDLY(1) | + DAVINCI_MCBSP_XCR_XFIG); + break; + } + return 0; }
@@ -223,13 +256,6 @@ static int davinci_i2s_hw_params(struct snd_pcm_substream *substream, DAVINCI_MCBSP_SPCR_RINTM(3) | DAVINCI_MCBSP_SPCR_XINTM(3) | DAVINCI_MCBSP_SPCR_FREE); - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, - DAVINCI_MCBSP_RCR_RFRLEN1(1) | - DAVINCI_MCBSP_RCR_RDATDLY(1)); - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, - DAVINCI_MCBSP_XCR_XFRLEN1(1) | - DAVINCI_MCBSP_XCR_XDATDLY(1) | - DAVINCI_MCBSP_XCR_XFIG);
i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS); w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SRGR_REG);
From: Naresh Medisetty naresh@ti.com
Fixes swapping of channels at start of stereo playback.
Channel swap can be observed while playing left-only or right-only audio data. The channel swap is fixed by handling the XSYNCERR condition.
Signed-off-by: Naresh Medisetty naresh@ti.com Signed-off-by: Mark Brown broonie@opensource.wolfsonmicro.com --- sound/soc/davinci/davinci-i2s.c | 49 ++++++++++++++++++++++++++++++++++++-- 1 files changed, 46 insertions(+), 3 deletions(-)
diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c index d814ec8..8c1bf87 100644 --- a/sound/soc/davinci/davinci-i2s.c +++ b/sound/soc/davinci/davinci-i2s.c @@ -111,16 +111,59 @@ static void davinci_mcbsp_start(struct snd_pcm_substream *substream) { struct snd_soc_pcm_runtime *rtd = substream->private_data; struct davinci_mcbsp_dev *dev = rtd->dai->cpu_dai->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_platform *platform = socdev->platform; u32 w; + int ret;
/* Start the sample generator and enable transmitter/receiver */ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_GRST, 1); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + /* Stop the DMA to avoid data loss */ + /* while the transmitter is out of reset to handle XSYNCERR */ + if (platform->pcm_ops->trigger) { + ret = platform->pcm_ops->trigger(substream, + SNDRV_PCM_TRIGGER_STOP); + if (ret < 0) + printk(KERN_DEBUG "Playback DMA stop failed\n"); + } + + /* Enable the transmitter */ + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 1); - else + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + + /* wait for any unexpected frame sync error to occur */ + udelay(100); + + /* Disable the transmitter to clear any outstanding XSYNCERR */ + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 0); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + + /* Restart the DMA */ + if (platform->pcm_ops->trigger) { + ret = platform->pcm_ops->trigger(substream, + SNDRV_PCM_TRIGGER_START); + if (ret < 0) + printk(KERN_DEBUG "Playback DMA start failed\n"); + } + /* Enable the transmitter */ + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); + MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_XRST, 1); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + + } else { + + /* Enable the reciever */ + w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG); MOD_REG_BIT(w, DAVINCI_MCBSP_SPCR_RRST, 1); - davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, w); + } +
/* Start frame sync */ w = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
At Wed, 12 Nov 2008 11:55:47 +0000, Mark Brown wrote:
From: Christian Pellegrin chripell@gmail.com
fixes playing/recording of 8 bit audio files.
Generated on 20081108 against v2.6.27
Signed-off-by: Christian Pellegrin chripell@fsfe.org Signed-off-by: Mark Brown broonie@opensource.wolfsonmicro.com
sound/soc/s3c24xx/s3c24xx-i2s.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-)
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index ba4476b..c18977b 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -261,10 +261,17 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream,
switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8:
iismod &= ~S3C2410_IISMOD_16BIT;((struct s3c24xx_pcm_dma_params *)rtd->dai->cpu_dai->dma_data)->dma_size = 1;((struct s3c24xx_pcm_dma_params *)rtd->dai->cpu_dai->dma_data)->dma_size = 2;
I still feel a bit uncomfortable with this as it changes the global variables. Maybe better to assign the corresponding fields in the local struct.
Takashi
On Wed, Nov 12, 2008 at 01:33:34PM +0100, Takashi Iwai wrote:
I still feel a bit uncomfortable with this as it changes the global variables. Maybe better to assign the corresponding fields in the local struct.
The global variable is used to tell the DMA driver how to set the hardware up for the port. Since the variable is tied to the underlying hardware channel it's used for there's no real win from duplicating it into a variable - the hardware it is used to configure can't be shared.
At Wed, 12 Nov 2008 15:10:38 +0000, Mark Brown wrote:
On Wed, Nov 12, 2008 at 01:33:34PM +0100, Takashi Iwai wrote:
I still feel a bit uncomfortable with this as it changes the global variables. Maybe better to assign the corresponding fields in the local struct.
The global variable is used to tell the DMA driver how to set the hardware up for the port. Since the variable is tied to the underlying hardware channel it's used for there's no real win from duplicating it into a variable - the hardware it is used to configure can't be shared.
I know it, but it's still uncomfortable that such secret mythes and legends are underlying without explanation. Could you add some more comments around that code to avoid misunderstanding by readers?
thanks,
Takashi
On Wed, Nov 12, 2008 at 04:16:08PM +0100, Takashi Iwai wrote:
Mark Brown wrote:
The global variable is used to tell the DMA driver how to set the hardware up for the port. Since the variable is tied to the underlying hardware channel it's used for there's no real win from duplicating it into a variable - the hardware it is used to configure can't be shared.
I know it, but it's still uncomfortable that such secret mythes and legends are underlying without explanation. Could you add some more comments around that code to avoid misunderstanding by readers?
Hrm, that's a fairly standard idiom for ASoC DMA/DAI driver interaction so probably more of a Documentation/ thing but doing it there is not too helpful for addressing your concern. I'll have a think.
On Wed, Nov 12, 2008 at 11:55:47AM +0000, Mark Brown wrote:
From: Christian Pellegrin chripell@gmail.com
fixes playing/recording of 8 bit audio files.
Generated on 20081108 against v2.6.27
Signed-off-by: Christian Pellegrin chripell@fsfe.org Signed-off-by: Mark Brown broonie@opensource.wolfsonmicro.com
sound/soc/s3c24xx/s3c24xx-i2s.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-)
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index ba4476b..c18977b 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -261,10 +261,17 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream,
switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8:
iismod &= ~S3C2410_IISMOD_16BIT;((struct s3c24xx_pcm_dma_params *)rtd->dai->cpu_dai->dma_data)->dma_size = 1;((struct s3c24xx_pcm_dma_params *)rtd->dai->cpu_dai->dma_data)->dma_size = 2;- default:
return -EINVAL;
Is playing with the dma parameters like that the correct way?
writel(iismod, s3c24xx_i2s.regs + S3C2410_IISMOD);
1.5.6.5
Alsa-devel mailing list Alsa-devel@alsa-project.org http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
At Wed, 12 Nov 2008 14:19:02 +0000, Ben Dooks wrote:
On Wed, Nov 12, 2008 at 11:55:47AM +0000, Mark Brown wrote:
From: Christian Pellegrin chripell@gmail.com
fixes playing/recording of 8 bit audio files.
Generated on 20081108 against v2.6.27
Signed-off-by: Christian Pellegrin chripell@fsfe.org Signed-off-by: Mark Brown broonie@opensource.wolfsonmicro.com
sound/soc/s3c24xx/s3c24xx-i2s.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-)
diff --git a/sound/soc/s3c24xx/s3c24xx-i2s.c b/sound/soc/s3c24xx/s3c24xx-i2s.c index ba4476b..c18977b 100644 --- a/sound/soc/s3c24xx/s3c24xx-i2s.c +++ b/sound/soc/s3c24xx/s3c24xx-i2s.c @@ -261,10 +261,17 @@ static int s3c24xx_i2s_hw_params(struct snd_pcm_substream *substream,
switch (params_format(params)) { case SNDRV_PCM_FORMAT_S8:
iismod &= ~S3C2410_IISMOD_16BIT;((struct s3c24xx_pcm_dma_params *)rtd->dai->cpu_dai->dma_data)->dma_size = 1;((struct s3c24xx_pcm_dma_params *)rtd->dai->cpu_dai->dma_data)->dma_size = 2;- default:
return -EINVAL;Is playing with the dma parameters like that the correct way?
It will work, I guess. But I don't say it's entirely correct :)
Takashi
On Wed, Nov 12, 2008 at 02:19:02PM +0000, Ben Dooks wrote:
On Wed, Nov 12, 2008 at 11:55:47AM +0000, Mark Brown wrote:
case SNDRV_PCM_FORMAT_S8:
iismod &= ~S3C2410_IISMOD_16BIT;((struct s3c24xx_pcm_dma_params *)rtd->dai->cpu_dai->dma_data)->dma_size = 1;((struct s3c24xx_pcm_dma_params *)rtd->dai->cpu_dai->dma_data)->dma_size = 2;
Is playing with the dma parameters like that the correct way?
It makes the DMA size match the input data size which doesn't seem totally unreasonable.
This is all internal to the s3c24xx I2S driver - what this does is change the parameter passed by the DMA driver to s3c2410_dma_config() for the I2S DMA channels which is just read blindly. This parameter is not exposed to user space.
On Wed, Nov 12, 2008 at 3:19 PM, Ben Dooks ben-linux@fluff.org wrote:
Is playing with the dma parameters like that the correct way?
Well this worked for me. I'm opened to better solutions and to resend a patch but it's no clear to me how to proceed.
Out of curiosity I did a search in Openmoko if someone noticed the same problem and found some people that had 8 bit ringtones sounding awful on the neo/forerunner. But they just resampled them to 16bit 44khz and lived happy. Unfortunately for consultants there is always the customer that wants to play that old 8-bit 11khz wav with the vocal announcement from 10 year ago.
At Wed, 12 Nov 2008 11:55:07 +0000, Mark Brown wrote:
The following changes since commit e18c94d20224f3df584531a48d944d8cccfda46d: Grazvydas Ignotas (1): ALSA: ASoC: TWL4030 codec - fix 256*Fs clock
are available in the git repository at:
git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai
Thanks, pulled in, and pushed out.
Takashi
participants (5)
-
Ben Dooks -
christian pellegrin -
Mark Brown -
Mark Brown -
Takashi Iwai