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

[alsa-devel] [PATCH 2/3] ASoC: TWL6030: Add support for low-power mode
by Lopez Cruz, Misael 14 Sep '09

14 Sep '09

TWL6030 codec supports two power modes: low-power and high-performance. In low-power mode, headset downlink must be the only path enabled and components in that path (headset DAC and driver) should be in that mode too. In this mode, codec can stream audio at 44.1 and 48 kHz if sys clock is configured to 17.64 and 19.2 MHz from CLK32K using Low-Power PLL, respectively. In high-performance mode, codec can only work at 19.2 MHz from High-Performance PLL. All paths supported in the codec can be used but the audio can be streamed only at 48 kHz. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl6030.c | 192 +++++++++++++++++++++++++++++++++++++------- sound/soc/codecs/twl6030.h | 19 ++++- 2 files changed, 181 insertions(+), 30 deletions(-) diff --git a/sound/soc/codecs/twl6030.c b/sound/soc/codecs/twl6030.c index c5e76fa..0f2269e 100644 --- a/sound/soc/codecs/twl6030.c +++ b/sound/soc/codecs/twl6030.c @@ -41,6 +41,11 @@ #define TWL6030_RATES (SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000) #define TWL6030_FORMATS (SNDRV_PCM_FMTBIT_S32_LE) +/* codec private data */ +struct twl6030_priv_data { + unsigned int sysclk; +}; + /* * twl6030 register cache & default register settings */ @@ -177,6 +182,29 @@ static void twl6030_power_down(struct snd_soc_codec *codec) twl6030_write_reg_cache(codec, TWL6030_REG_LPPLLCTL, 0x00); } +/* set headset dac and driver power mode */ +static int headset_power_mode(struct snd_soc_codec *codec, int high_perf) +{ + int hslctl, hsrctl; + int mask = TWL6030_HSDRVMODEL | TWL6030_HSDACMODEL; + + hslctl = twl6030_read_reg_cache(codec, TWL6030_REG_HSLCTL); + hsrctl = twl6030_read_reg_cache(codec, TWL6030_REG_HSRCTL); + + if (high_perf) { + hslctl &= ~mask; + hsrctl &= ~mask; + } else { + hslctl |= mask; + hsrctl |= mask; + } + + twl6030_write(codec, TWL6030_REG_HSLCTL, hslctl); + twl6030_write(codec, TWL6030_REG_HSRCTL, hsrctl); + + return 0; +} + /* * MICATT volume control: * from -6 to 0 dB in 6 dB steps @@ -419,6 +447,10 @@ static int twl6030_hw_params(struct snd_pcm_substream *substream, struct snd_pcm_hw_params *params, struct snd_soc_dai *dai) { + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_device *socdev = rtd->socdev; + struct snd_soc_codec *codec = socdev->card->codec; + struct twl6030_priv_data *twl6030_priv = codec->private_data; int rate, format; /* hardware dai (McPDM) requires bit stream of twice @@ -428,7 +460,20 @@ static int twl6030_hw_params(struct snd_pcm_substream *substream, rate = params_rate(params); switch (rate) { case 44100: + if (twl6030_priv->sysclk != 17640000) { + dev_err(codec->dev, + "rate %d not supported at current sysclk %d\n", + rate, twl6030_priv->sysclk); + return -EINVAL; + } + break; case 48000: + if (twl6030_priv->sysclk != 19200000) { + dev_err(codec->dev, + "rate %d not supported at current sysclk %d\n", + rate, twl6030_priv->sysclk); + return -EINVAL; + } break; default: dev_err(codec->dev, "unknown rate %d\n", rate); @@ -451,46 +496,120 @@ static int twl6030_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; + struct twl6030_priv_data *twl6030_priv = codec->private_data; u8 hppll, lppll; + lppll = twl6030_read_reg_cache(codec, TWL6030_REG_LPPLLCTL); hppll = twl6030_read_reg_cache(codec, TWL6030_REG_HPPLLCTL); - hppll &= TWL6030_HPLLRST; - - switch (freq) { - case 12000000: - /* MCLK input, PLL enabled */ - hppll = TWL6030_MCLK_12000KHZ - | TWL6030_HPLLSQRBP - | TWL6030_HPLLENA; - break; - case 19200000: - /* MCLK input, PLL disabled */ - hppll = TWL6030_MCLK_19200KHZ - | TWL6030_HPLLSQRBP - | TWL6030_HPLLBP; + + switch (clk_id) { + case TWL6030_SYSCLK_SEL_LPPLL: + if (freq != 32768) { + dev_err(codec->dev, "invalid sysclk freq %d\n", freq); + return -EINVAL; + } + + /* CLK32K input requires low-power PLL */ + lppll |= TWL6030_LPLLENA | TWL6030_LPLLSEL; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + mdelay(5); + lppll &= ~TWL6030_HPLLSEL; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + + /* headset dac and driver must be in low-power mode */ + headset_power_mode(codec, 0); + + hppll &= ~TWL6030_HPLLENA; + twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppll); break; - case 26000000: - /* MCLK input, PLL enabled */ - hppll = TWL6030_MCLK_26000KHZ - | TWL6030_HPLLSQRBP - | TWL6030_HPLLENA; + case TWL6030_SYSCLK_SEL_HPPLL: + switch (freq) { + case 12000000: + hppll = TWL6030_MCLK_12000KHZ; + break; + case 26000000: + hppll = TWL6030_MCLK_26000KHZ; + break; + default: + dev_err(codec->dev, "invalid sysclk freq %d\n", freq); + return -EINVAL; + } + + /* 12 and 26 MHz freqs require high-performance PLL */ + hppll |= TWL6030_HPLLSQRBP | TWL6030_HPLLENA; + twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppll); + udelay(500); + + /* headset dac and driver must be in high-performance mode */ + headset_power_mode(codec, 1); + + lppll |= TWL6030_HPLLSEL; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + lppll &= ~TWL6030_LPLLENA; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + + /* only 19.2 MHz can be generated by HPPLL */ + twl6030_priv->sysclk = 19200000; break; - case 38400000: - /* clk slicer input, PLL disabled */ - hppll = TWL6030_MCLK_38400KHZ - | TWL6030_HPLLSQRENA - | TWL6030_HPLLBP; + case TWL6030_SYSCLK_SEL_MCLK: + switch (freq) { + case 19200000: + hppll = TWL6030_MCLK_19200KHZ | TWL6030_HPLLSQRBP; + break; + case 38400000: + hppll = TWL6030_MCLK_38400KHZ; + break; + default: + dev_err(codec->dev, "invalid sysclk freq %d\n", freq); + return -EINVAL; + } + + /* 19.2 and 38.4 MHz freqs don't require PLL */ + hppll |= TWL6030_HPLLBP; + twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppll); + udelay(500); + + /* headset dac and driver must be in high-performance mode */ + headset_power_mode(codec, 1); + + lppll |= TWL6030_HPLLSEL; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + lppll &= ~TWL6030_LPLLENA; + twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + + /* only 19.2 MHz can be generated by MCLK */ + twl6030_priv->sysclk = 19200000; break; default: - dev_err(codec->dev, "unknown sysclk rate %d\n", freq); + dev_err(codec->dev, "unknown clk_id %d\n", clk_id); return -EINVAL; } - twl6030_write(codec, TWL6030_REG_HPPLLCTL, hppll); + return 0; +} + +static int twl6030_set_dai_pll(struct snd_soc_dai *codec_dai, int pll_id, + unsigned int freq_in, unsigned int freq_out) +{ + struct snd_soc_codec *codec = codec_dai->codec; + struct twl6030_priv_data *twl6030_priv = codec->private_data; + int div; + + if (!freq_in || !freq_out) + return -EINVAL; + + /* div = round(freq_out / freq_in) */ + div = (freq_out + (freq_in >> 2)) / freq_in; - /* Disable LPPLL and select HPPLL */ - lppll = TWL6030_HPLLSEL; - twl6030_write(codec, TWL6030_REG_LPPLLCTL, lppll); + if (pll_id == TWL6030_LPPLL_ID) { + if (div < 512) { + dev_err(codec->dev, "invalid pll divider %d\n", div); + return -EINVAL; + } + + twl6030_write(codec, TWL6030_REG_LPPLLDIV, div - 512); + twl6030_priv->sysclk = freq_out; + } return 0; } @@ -510,6 +629,7 @@ static int twl6030_set_dai_fmt(struct snd_soc_dai *codec_dai, static struct snd_soc_dai_ops twl6030_dai_ops = { .hw_params = twl6030_hw_params, .set_sysclk = twl6030_set_dai_sysclk, + .set_pll = twl6030_set_dai_pll, .set_fmt = twl6030_set_dai_fmt, }; @@ -614,12 +734,21 @@ static int twl6030_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec; + struct twl6030_priv_data *twl6030_priv; + int ret; codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); if (codec == NULL) return -ENOMEM; + twl6030_priv = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); + if (twl6030_priv == NULL) { + ret = -ENOMEM; + goto priv_err; + } + codec->dev = &pdev->dev; + codec->private_data = twl6030_priv; socdev->card->codec = codec; mutex_init(&codec->mutex); INIT_LIST_HEAD(&codec->dapm_widgets); @@ -629,6 +758,10 @@ static int twl6030_probe(struct platform_device *pdev) twl6030_init(socdev); return 0; + +priv_err: + kfree(codec); + return ret; } static int twl6030_remove(struct platform_device *pdev) @@ -640,6 +773,7 @@ static int twl6030_remove(struct platform_device *pdev) twl6030_set_bias_level(codec, SND_SOC_BIAS_OFF); snd_soc_free_pcms(socdev); snd_soc_dapm_free(socdev); + kfree(codec->private_data); kfree(codec); return 0; diff --git a/sound/soc/codecs/twl6030.h b/sound/soc/codecs/twl6030.h index a83f9e7..9e49c85 100644 --- a/sound/soc/codecs/twl6030.h +++ b/sound/soc/codecs/twl6030.h @@ -77,14 +77,31 @@ #define TWL6030_MCLK_38400KHZ (3 << 5) #define TWL6030_MCLK_MSK 0x60 +#define TWL6030_SYSCLK_SEL_LPPLL 1 +#define TWL6030_SYSCLK_SEL_HPPLL 2 +#define TWL6030_SYSCLK_SEL_MCLK 3 + /* LPPLLCTL (0x08) fields */ #define TWL6030_LPLLENA 0x01 #define TWL6030_LPLLRST 0x02 #define TWL6030_LPLLSEL 0x04 -#define TWL6030_FIN 0x08 +#define TWL6030_LPLLFIN 0x08 #define TWL6030_HPLLSEL 0x10 +#define TWL6030_HPPLL_ID 1 +#define TWL6030_LPPLL_ID 2 + +/* HSLCTL (0x10) fields */ + +#define TWL6030_HSDACMODEL 0x02 +#define TWL6030_HSDRVMODEL 0x08 + +/* HSRCTL (0x11) fields */ + +#define TWL6030_HSDACMODER 0x02 +#define TWL6030_HSDRVMODER 0x08 + extern struct snd_soc_dai twl6030_dai; extern struct snd_soc_codec_device soc_codec_dev_twl6030; -- 1.5.4.3

2 1

[PATCH] ALSA: hda - Add support for HP dv6
by Takashi Iwai 14 Sep '09

14 Sep '09

Add the quirk entry for HP dv6. Also add a workaround for the headphone detection by setting hp_detect=1 beforehand. Without this, the driver won't do auto-muting because BIOS doesn't give any HP pin but only a line-out pin. Signed-off-by: Takashi Iwai <tiwai(a)suse.de> --- sound/pci/hda/patch_sigmatel.c | 7 +++++++ 1 files changed, 7 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index fe2ba76..cc0e26c 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1690,6 +1690,8 @@ static struct snd_pci_quirk stac92hd71bxx_cfg_tbl[] = { "HP mini 1000", STAC_HP_M4), SND_PCI_QUIRK(PCI_VENDOR_ID_HP, 0x361b, "HP HDX", STAC_HP_HDX), /* HDX16 */ + SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x3620, + "HP dv6", STAC_HP_DV5), SND_PCI_QUIRK_MASK(PCI_VENDOR_ID_HP, 0xfff0, 0x7010, "HP", STAC_HP_DV5), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0233, @@ -5383,6 +5385,11 @@ again: case STAC_HP_DV5: snd_hda_codec_set_pincfg(codec, 0x0d, 0x90170010); stac92xx_auto_set_pinctl(codec, 0x0d, AC_PINCTL_OUT_EN); + /* HP dv6 gives the headphone pin as a line-out. Thus we + * need to set hp_detect flag here to force to enable HP + * detection. + */ + spec->hp_detect = 1; break; case STAC_HP_HDX: spec->num_dmics = 1; -- 1.6.4.2

1 0

[alsa-devel] ice1724: dmesg filled with IRQ errors
by Vedran Miletić 14 Sep '09

14 Sep '09

I ran winecfg, and it probed around sound stuff a bit, and then dmesg got filled with: [ 6623.654899] ice1724: Too long irq loop, status = 0x30 [ 6623.654901] ice1724: Disabling MPU_TX [ 6623.654982] ice1724: Too long irq loop, status = 0x30 [ 6623.654983] ice1724: Disabling MPU_TX [ 6623.655064] ice1724: Too long irq loop, status = 0x30 [ 6623.655065] ice1724: Disabling MPU_TX [ 6623.655146] ice1724: Too long irq loop, status = 0x30 [ 6623.655147] ice1724: Disabling MPU_TX [ 6623.655228] ice1724: Too long irq loop, status = 0x30 [ 6623.655229] ice1724: Disabling MPU_TX [ 6623.655310] ice1724: Too long irq loop, status = 0x30 [ 6623.655311] ice1724: Disabling MPU_TX [ 6623.655391] ice1724: Too long irq loop, status = 0x30 [ 6623.655393] ice1724: Disabling MPU_TX [ 6623.655473] ice1724: Too long irq loop, status = 0x30 [ 6623.655475] ice1724: Disabling MPU_TX [ 6623.655555] ice1724: Too long irq loop, status = 0x30 [ 6623.655557] ice1724: Disabling MPU_TX [ 6623.655637] ice1724: Too long irq loop, status = 0x30 [ 6623.655639] ice1724: Disabling MPU_TX [ 6623.655719] ice1724: Too long irq loop, status = 0x30 [ 6623.655721] ice1724: Disabling MPU_TX [ 6623.655801] ice1724: Too long irq loop, status = 0x30 [ 6623.655803] ice1724: Disabling MPU_TX [ 6623.655883] ice1724: Too long irq loop, status = 0x30 [ 6623.655885] ice1724: Disabling MPU_TX [ 6623.655966] ice1724: Too long irq loop, status = 0x30 [ 6623.655967] ice1724: Disabling MPU_TX And so on. I had to reset, since it was impossible to shutdown or unload ice1724 module. Is this a known bug? Can I help debugging it? Vedran Miletić

2 1

[alsa-devel] [PATCH] use set_channel_map api to reorder channels for AD1938 and AD1836
by Barry Song 14 Sep '09

14 Sep '09

From: Barry Song <Barry.Song(a)analog.com> --- sound/soc/blackfin/bf5xx-ad1836.c | 7 +++++++ sound/soc/blackfin/bf5xx-ad1938.c | 9 ++++++++- sound/soc/blackfin/bf5xx-tdm-pcm.c | 7 +++++-- sound/soc/blackfin/bf5xx-tdm.c | 28 ++++++++++++++++++++-------- sound/soc/blackfin/bf5xx-tdm.h | 11 +++++++++++ 5 files changed, 51 insertions(+), 11 deletions(-) diff --git a/sound/soc/blackfin/bf5xx-ad1836.c b/sound/soc/blackfin/bf5xx-ad1836.c index cd361e3..d3b41b7 100644 --- a/sound/soc/blackfin/bf5xx-ad1836.c +++ b/sound/soc/blackfin/bf5xx-ad1836.c @@ -52,6 +52,7 @@ static int bf5xx_ad1836_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + unsigned int channel_map[8] = {0, 4, 1, 5, 2, 6, 3, 7}; int ret = 0; /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | @@ -65,6 +66,12 @@ static int bf5xx_ad1836_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; + /* set cpu DAI channel mapping */ + ret = snd_soc_dai_set_channel_map(cpu_dai, 8, channel_map, + 8, channel_map); + if (ret < 0) + return ret; + return 0; } diff --git a/sound/soc/blackfin/bf5xx-ad1938.c b/sound/soc/blackfin/bf5xx-ad1938.c index 08269e9..686e84c 100644 --- a/sound/soc/blackfin/bf5xx-ad1938.c +++ b/sound/soc/blackfin/bf5xx-ad1938.c @@ -61,6 +61,7 @@ static int bf5xx_ad1938_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + unsigned int channel_map[8] = {0, 1, 2, 3, 4, 5, 6, 7}; int ret = 0; /* set cpu DAI configuration */ ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A | @@ -75,7 +76,13 @@ static int bf5xx_ad1938_hw_params(struct snd_pcm_substream *substream, return ret; /* set codec DAI slots, 8 channels, all channels are enabled */ - ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xFF, 8); + ret = snd_soc_dai_set_tdm_slot(codec_dai, 0xFF, 0xFF, 8, 32); + if (ret < 0) + return ret; + + /* set cpu DAI channel mapping */ + ret = snd_soc_dai_set_channel_map(cpu_dai, 8, channel_map, + 8, channel_map); if (ret < 0) return ret; diff --git a/sound/soc/blackfin/bf5xx-tdm-pcm.c b/sound/soc/blackfin/bf5xx-tdm-pcm.c index ccb5e82..881a1d9 100644 --- a/sound/soc/blackfin/bf5xx-tdm-pcm.c +++ b/sound/soc/blackfin/bf5xx-tdm-pcm.c @@ -177,6 +177,9 @@ out: static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, snd_pcm_uframes_t pos, void *buf, snd_pcm_uframes_t count) { + struct snd_pcm_runtime *runtime = substream->runtime; + struct sport_device *sport = runtime->private_data; + struct bf5xx_tdm_port *tdm_port = sport->private_data; unsigned int *src; unsigned int *dst; int i; @@ -188,7 +191,7 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, dst += pos * 8; while (count--) { for (i = 0; i < substream->runtime->channels; i++) - *(dst + i) = *src++; + *(dst + tdm_port->tx_map[i]) = *src++; dst += 8; } } else { @@ -198,7 +201,7 @@ static int bf5xx_pcm_copy(struct snd_pcm_substream *substream, int channel, src += pos * 8; while (count--) { for (i = 0; i < substream->runtime->channels; i++) - *dst++ = *(src+i); + *dst++ = *(src + tdm_port->rx_map[i]); src += 8; } } diff --git a/sound/soc/blackfin/bf5xx-tdm.c b/sound/soc/blackfin/bf5xx-tdm.c index 3096bad..6995d5b 100644 --- a/sound/soc/blackfin/bf5xx-tdm.c +++ b/sound/soc/blackfin/bf5xx-tdm.c @@ -46,14 +46,6 @@ #include "bf5xx-sport.h" #include "bf5xx-tdm.h" -struct bf5xx_tdm_port { - u16 tcr1; - u16 rcr1; - u16 tcr2; - u16 rcr2; - int configured; -}; - static struct bf5xx_tdm_port bf5xx_tdm; static int sport_num = CONFIG_SND_BF5XX_SPORT_NUM; @@ -181,6 +173,23 @@ static void bf5xx_tdm_shutdown(struct snd_pcm_substream *substream, bf5xx_tdm.configured = 0; } +static int bf5xx_tdm_set_channel_map(struct snd_soc_dai *dai, + unsigned int tx_num, unsigned int *tx_slot, + unsigned int rx_num, unsigned int *rx_slot) +{ + int i; + + if ((tx_num > MAX_TDM_SLOTS) || (rx_num > MAX_TDM_SLOTS)) + return -EINVAL; + + for (i = 0; i < tx_num; i++) + bf5xx_tdm.tx_map[i] = tx_slot[i]; + for (i = 0; i < rx_num; i++) + bf5xx_tdm.rx_map[i] = rx_slot[i]; + + return 0; +} + #ifdef CONFIG_PM static int bf5xx_tdm_suspend(struct snd_soc_dai *dai) { @@ -235,6 +244,7 @@ static struct snd_soc_dai_ops bf5xx_tdm_dai_ops = { .hw_params = bf5xx_tdm_hw_params, .set_fmt = bf5xx_tdm_set_dai_fmt, .shutdown = bf5xx_tdm_shutdown, + .set_channel_map = bf5xx_tdm_set_channel_map, }; struct snd_soc_dai bf5xx_tdm_dai = { @@ -300,6 +310,8 @@ static int __devinit bfin_tdm_probe(struct platform_device *pdev) pr_err("Failed to register DAI: %d\n", ret); goto sport_config_err; } + + sport_handle->private_data = &bf5xx_tdm; return 0; sport_config_err: diff --git a/sound/soc/blackfin/bf5xx-tdm.h b/sound/soc/blackfin/bf5xx-tdm.h index 618ec3d..f85790a 100644 --- a/sound/soc/blackfin/bf5xx-tdm.h +++ b/sound/soc/blackfin/bf5xx-tdm.h @@ -9,6 +9,17 @@ #ifndef _BF5XX_TDM_H #define _BF5XX_TDM_H +#define MAX_TDM_SLOTS 8 +struct bf5xx_tdm_port { + u16 tcr1; + u16 rcr1; + u16 tcr2; + u16 rcr2; + unsigned int tx_map[MAX_TDM_SLOTS]; + unsigned int rx_map[MAX_TDM_SLOTS]; + int configured; +}; + extern struct snd_soc_dai bf5xx_tdm_dai; #endif -- 1.5.6.3

2 1

[alsa-devel] [RFC][PATCH] ALSA: pcm_native.c initialize drec and num_drecs before it get used
by Jaswinder Singh Rajput 14 Sep '09

14 Sep '09

[This is untested] drec and num_drecs is used uninitialized for nonblock(goto lock) This also fixed following compilation warnings : CC [M] sound/core/pcm_native.o sound/core/pcm_native.c: In function ‘snd_pcm_drain’: sound/core/pcm_native.c:1413: warning: ‘drec’ may be used uninitialized in this function sound/core/pcm_native.c:1411: warning: ‘num_drecs’ may be used uninitialized in this function Signed-off-by: Jaswinder Singh Rajput <jaswinderrajput(a)gmail.com> --- sound/core/pcm_native.c | 7 ++++--- 1 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/core/pcm_native.c b/sound/core/pcm_native.c index 59e5fbe..8284dc1 100644 --- a/sound/core/pcm_native.c +++ b/sound/core/pcm_native.c @@ -1433,6 +1433,9 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, } else if (substream->f_flags & O_NONBLOCK) nonblock = 1; + drec = &drec_tmp; + num_drecs = 0; + if (nonblock) goto lock; /* no need to allocate waitqueues */ @@ -1445,11 +1448,9 @@ static int snd_pcm_drain(struct snd_pcm_substream *substream, snd_power_unlock(card); return -ENOMEM; } - } else - drec = &drec_tmp; + } /* count only playback streams */ - num_drecs = 0; snd_pcm_group_for_each_entry(s, substream) { runtime = s->runtime; if (s->stream == SNDRV_PCM_STREAM_PLAYBACK) { -- 1.6.4.2

2 1

[alsa-devel] still have popping every few seconds, like metronome, with xfi drivers
by Ted T. Logan 14 Sep '09

14 Sep '09

I just tried latest snapshot with my SB0670/XFI, which works fine with the drivers: XFiDrv_Linux_Public_US_1.00.tar.gz and the same exact problems remain for all the snapshots I've tried... clicking like a metronome every 1 second or so on ALL playback. Can anything be done about this? It's completely unusable with these drivers. Thanks

1 0

[alsa-devel] [PATCH v2] add channel order map interface in asoc core
by Barry Song 13 Sep '09

13 Sep '09

The patch adds an interface to set the relationship between audio channel number and slot number. The interface should be really useful because audio channel n doesn't always use slot n in all platforms. And for some devices, the relationship even can change with sound mode switch in 2.1,3.1,4.1,5.1,6.1,7.1 etc. Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- include/sound/soc-dai.h | 7 +++++++ sound/soc/soc-core.c | 24 ++++++++++++++++++++++++ 2 files changed, 31 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 97ca9af..5c84566 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -114,6 +114,10 @@ int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt); int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); +int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai, + unsigned int tx_num, unsigned int *tx_slot, + unsigned int rx_num, unsigned int *rx_slot); + int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); /* Digital Audio Interface mute */ @@ -148,6 +152,9 @@ struct snd_soc_dai_ops { int (*set_tdm_slot)(struct snd_soc_dai *dai, unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width); + int (*set_channel_map)(struct snd_soc_dai *dai, + unsigned int tx_num, unsigned int *tx_slot, + unsigned int rx_num, unsigned int *rx_slot); int (*set_tristate)(struct snd_soc_dai *dai, int tristate); /* diff --git a/sound/soc/soc-core.c b/sound/soc/soc-core.c index 7ff04ad..ba1f8e5 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2251,6 +2251,30 @@ int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai, EXPORT_SYMBOL_GPL(snd_soc_dai_set_tdm_slot); /** + * snd_soc_dai_set_channel_map - configure DAI audio channel map + * @dai: DAI + * @tx_num: how many TX channels + * @tx_slot: pointer to an array which imply the TX slot number channel + * 0~num-1 uses + * @rx_num: how many RX channels + * @rx_slot: pointer to an array which imply the RX slot number channel + * 0~num-1 uses + * + * configure the relationship between channel number and TDM slot number. + */ +int snd_soc_dai_set_channel_map(struct snd_soc_dai *dai, + unsigned int tx_num, unsigned int *tx_slot, + unsigned int rx_num, unsigned int *rx_slot) +{ + if (dai->ops && dai->ops->set_tdm_slot) + return dai->ops->set_channel_map(dai, tx_num, tx_slot, + rx_num, rx_slot); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_channel_map); + +/** * snd_soc_dai_set_tristate - configure DAI system or master clock. * @dai: DAI * @tristate: tristate enable -- 1.5.6.3

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[alsa-devel] How to test suspend ?
by Giuliano Pochini 12 Sep '09

12 Sep '09

I couldn't make s2ram work on my machine and /sys/pci/.../power/state virtual files were removed one year ago. How can I test suspend/resume ? -- Giuliano.

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[alsa-devel] SPDIF issues
by Ben 12 Sep '09

12 Sep '09

All, I have been happily using a small setup for the purpose of MythTV. I recently decided to upgrade the graphics so I could enjoy VPDAU which has been integrated into the 0.22 branch of MythTV. There is just one snag, I don't have any sound. I am using an Asus P5E-VM HDMI board, it has onboard sound & graphics. The HDMI interface outputs to my TV and it works fine. I then came across an nvidia 9500GT (gainwood) card which has a VGA/DVI/HDMI out and also takes in sound via SPDIF. I connected the cable to my spdif headers and I can't seem to get any sound. I'm running Ubuntu Karmic so I thought I would upgrade the Alsa drivers/lib/utils to the latest, this has not helped matters. I have checked Alsa mixer and nothing is muted. I have tried: speaker-test -Dplughw:0,1 - (I think this is correct). I'm sure it's something pretty trivial I could be doing wrong but someone mentioned this kind of setup can be quite problematic and suggested I posted the details to this list. Details here: http://www.alsa-project.org/db/?f=ffb9e4b76f37a87f21fec2a4db18295479608d24 I'd appreciate your thoughts! Thanks, Ben

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[alsa-devel] [PATCH 8/8, retry] sound/soc: introduce missing kfree
by Julia Lawall 12 Sep '09

12 Sep '09

From: Julia Lawall <julia(a)diku.dk> Error handling code following a kzalloc should free the allocated data. Error handling code following an ioremap should iounmap the allocated data. The semantic match that finds the first problem is as follows: (http://www.emn.fr/x-info/coccinelle/) // <smpl> @r exists@ local idexpression x; statement S; expression E; identifier f,f1,l; position p1,p2; expression *ptr != NULL; @@ x@p1 = \(kmalloc\|kzalloc\|kcalloc\)(...); ... if (x == NULL) S <... when != x when != if (...) { <+...x...+> } ( x->f1 = E | (x->f1 == NULL || ...) | f(...,x->f1,...) ) ...> ( return \(0\|<+...x...+>\|ptr\); | return@p2 ...; ) @script:python@ p1 << r.p1; p2 << r.p2; @@ print "* file: %s kmalloc %s return %s" % (p1[0].file,p1[0].line,p2[0].line) // </smpl> Signed-off-by: Julia Lawall <julia(a)diku.dk> --- sound/soc/fsl/mpc5200_dma.c | 33 ++++++++++++-------- 1 files changed, 20 insertions(+), 13 deletions(-) diff --git a/var/linuxes/linux-next/sound/soc/fsl/mpc5200_dma.c b/var/julia/linuxcopy/sound/soc/fsl/mpc5200_dma.c index 9ff62e3..6096d22 100644 --- a/var/linuxes/linux-next/sound/soc/fsl/mpc5200_dma.c +++ b/var/julia/linuxcopy/sound/soc/fsl/mpc5200_dma.c @@ -447,6 +447,7 @@ int mpc5200_audio_dma_create(struct of_device *op) int size, irq, rc; const __be32 *prop; void __iomem *regs; + int ret; /* Fetch the registers and IRQ of the PSC */ irq = irq_of_parse_and_map(op->node, 0); @@ -463,14 +464,16 @@ int mpc5200_audio_dma_create(struct of_device *op) /* Allocate and initialize the driver private data */ psc_dma = kzalloc(sizeof *psc_dma, GFP_KERNEL); if (!psc_dma) { - iounmap(regs); - return -ENOMEM; + ret = -ENOMEM; + goto out_unmap; } /* Get the PSC ID */ prop = of_get_property(op->node, "cell-index", &size); - if (!prop || size < sizeof *prop) - return -ENODEV; + if (!prop || size < sizeof *prop) { + ret = -ENODEV; + goto out_free; + } spin_lock_init(&psc_dma->lock); mutex_init(&psc_dma->mutex); @@ -493,9 +496,8 @@ int mpc5200_audio_dma_create(struct of_device *op) if (!psc_dma->capture.bcom_task || !psc_dma->playback.bcom_task) { dev_err(&op->dev, "Could not allocate bestcomm tasks\n"); - iounmap(regs); - kfree(psc_dma); - return -ENODEV; + ret = -ENODEV; + goto out_free; } /* Disable all interrupts and reset the PSC */ @@ -537,12 +539,8 @@ int mpc5200_audio_dma_create(struct of_device *op) &psc_dma_bcom_irq_tx, IRQF_SHARED, "psc-dma-playback", &psc_dma->playback); if (rc) { - free_irq(psc_dma->irq, psc_dma); - free_irq(psc_dma->capture.irq, - &psc_dma->capture); - free_irq(psc_dma->playback.irq, - &psc_dma->playback); - return -ENODEV; + ret = -ENODEV; + goto out_irq; } /* Save what we've done so it can be found again later */ @@ -550,6 +548,15 @@ int mpc5200_audio_dma_create(struct of_device *op) /* Tell the ASoC OF helpers about it */ return snd_soc_register_platform(&mpc5200_audio_dma_platform); +out_irq: + free_irq(psc_dma->irq, psc_dma); + free_irq(psc_dma->capture.irq, &psc_dma->capture); + free_irq(psc_dma->playback.irq, &psc_dma->playback); +out_free: + kfree(psc_dma); +out_unmap: + iounmap(regs); + return ret; } EXPORT_SYMBOL_GPL(mpc5200_audio_dma_create);

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