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

2 3

[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.

3 4

[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

1 0

[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);

2 1

[alsa-devel] hda-intel missing mixer channels
by Orm Finnendahl 12 Sep '09

12 Sep '09

Hi, I sent this to alsa-user a week ago but didn't get any reply. Does anybody know, what is responsible for the missing faders? Did the driver change or is it some module loading mechanism in the distro which prevents the faders from showing up in alsamixer? This is so strange. Please help! -- Orm -------------------------------------------------------------------- Hi everybody, after the last update of Ubuntu (intrepid to jaunty) on my Lenovo Ideapad my microphone input doesn't work anymore (neither internal nor external). I found out that alsamixer lacks quite a lot of faders in the newer kernel/module compared to the older one. Some of the faders now unavailable were crucial for Mic input to work on the older system so I'd very much like to get them back ;-) Some more infos: Soundcard: Intel 82801G (ICH7 Family) Chip: Realtek ALC269 Module: snd_hda_intel (in both kernels) Older kernel: Ubuntu stock 2.6.27-14-generic Newer kernel: Ubuntu stock 2.6.28-13-rt Is there any way to make the missing controls available or did the driver change and I have to compile the older driver for the new kernel? Yours, Orm

1 0

[alsa-devel] [PATCH 8/8] 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 | 5 ++++- 1 files changed, 4 insertions(+), 1 deletions(-) diff --git a/sound/soc/fsl/mpc5200_dma.c b/sound/soc/fsl/mpc5200_dma.c index 9ff62e3..41ffd57 100644 --- a/sound/soc/fsl/mpc5200_dma.c +++ b/sound/soc/fsl/mpc5200_dma.c @@ -469,8 +469,11 @@ int mpc5200_audio_dma_create(struct of_device *op) /* Get the PSC ID */ prop = of_get_property(op->node, "cell-index", &size); - if (!prop || size < sizeof *prop) + if (!prop || size < sizeof *prop) { + iounmap(regs); + kfree(psc_dma); return -ENODEV; + } spin_lock_init(&psc_dma->lock); mutex_init(&psc_dma->mutex);

2 2

[alsa-devel] WM8580 DAI: Debugged
by jassi brar 12 Sep '09

12 Sep '09

Hi, I have been making SMDK6410 I2S work with WM8580 codec. In the process i debugged the wm8580 driver. The patch is:- Last patch(Minor WM8580 enhancements) _minus_ the MCLKRATIO/BCLKRATIO stuff _plus_ the snd_soc_dai definition debugged. The snd_soc_dai debugging is necessary if the Audio Capture is to work with the CODEC driver. Next comes the patch.....

1 0

[alsa-devel] ALSA 1.0.21 release
by Jaroslav Kysela 11 Sep '09

11 Sep '09

Hello all, all ALSA packages except alsa-oss and alsa-firmware (no changes from 1.0.20) are available for download at http://www.alsa-project.org . Changelog is available here: http://www.alsa-project.org/main/index.php/Changes_v1.0.20_v1.0.21 Jaroslav Kysela ----- Jaroslav Kysela <perex(a)perex.cz> Linux Kernel Sound Maintainer ALSA Project, Red Hat, Inc.

3 5

[alsa-devel] hda-intel missing mixer channels
by Orm Finnendahl 11 Sep '09

11 Sep '09

Hi, I sent this to alsa-user a week ago but didn't get any reply. Does anybody know, who is responsible for the missing faders? Did the driver change or is it some module loading mechanism in the distro which prevents the faders from showing up in alsamixer? This is so strange. Please help! -- Orm -------------------------------------------------------------------- Hi everybody, after the last update of Ubuntu (intrepid to jaunty) on my Lenovo Ideapad my microphone input doesn't work anymore (neither internal nor external). I found out that alsamixer lacks quite a lot of faders in the newer kernel/module compared to the older one. Some of the faders now unavailable were crucial for Mic input to work on the older system so I'd very much like to get them back ;-) Some more infos: Soundcard: Intel 82801G (ICH7 Family) Chip: Realtek ALC269 Module: snd_hda_intel (in both kernels) Older kernel: Ubuntu stock 2.6.27-14-generic Newer kernel: Ubuntu stock 2.6.28-13-rt Is there any way to make the missing controls available or did the driver change and I have to compile the older driver for the new kernel? Yours, Orm

1 0

[alsa-devel] [PATCH] add channel order setting interface in asoc core
by Barry Song 11 Sep '09

11 Sep '09

Hi Mark, As discussed, I am adding a patch to add an interface to set the relationship between audio channel No. and slot No. The interface should be really useful because nobody can ensure audio channel n always uses 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. After the patch is applied, I can change bfin stuff based on it. Thanks, Barry Signed-off-by: Barry Song <21cnbao(a)gmail.com> --- include/sound/soc-dai.h | 5 +++++ sound/soc/soc-core.c | 18 ++++++++++++++++++ 2 files changed, 23 insertions(+), 0 deletions(-) diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h index 97ca9af..7439425 100644 --- a/include/sound/soc-dai.h +++ b/include/sound/soc-dai.h @@ -114,6 +114,9 @@ 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_order(struct snd_soc_dai *dai, + unsigned int num, unsigned int *slot); + int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate); /* Digital Audio Interface mute */ @@ -148,6 +151,8 @@ 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_order)(struct snd_soc_dai *dai, + unsigned int num, unsigned int *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..30d0143 100644 --- a/sound/soc/soc-core.c +++ b/sound/soc/soc-core.c @@ -2251,6 +2251,24 @@ 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_order - configure DAI audio channel order + * @dai: DAI + * @num: how many channels + * @slot: pointer to an array which imply the slot number channel 0~num-1 uses + * + * configure the relationship between channel number and TDM slot number. + */ +int snd_soc_dai_set_channel_order(struct snd_soc_dai *dai, + unsigned int num, unsigned int *slot) +{ + if (dai->ops && dai->ops->set_tdm_slot) + return dai->ops->set_channel_order(dai, num, slot); + else + return -EINVAL; +} +EXPORT_SYMBOL_GPL(snd_soc_dai_set_channel_order); + +/** * snd_soc_dai_set_tristate - configure DAI system or master clock. * @dai: DAI * @tristate: tristate enable -- 1.5.6.3

2 1

[alsa-devel] [PATCH] [RFC 8/13] Intel SST sound card driver
by Vinod Koul 11 Sep '09

11 Sep '09

This adds support for Moorestown ALSA Sound card driver. This is an ALSA driver for supporting PCM playback/capture in traditional ALSA way. Anyone who chooses not to use DSP for decoding/encoding can use ALSA path to play/capture (in non low power mode). This driver registers the control interface and PCM interface with the LPE driver which finally sends it to the hardware. This driver allows any subsystem in OS which wants to use the audio-subsystems to be routed through the ALSA Signed-off-by: Vinod Koul <vinod.koul(a)intel.com> Signed-off-by: Harsha Priya <priya.harsha(a)intel.com> new file: sound/pci/sst/intelmid.c --- sound/pci/sst/intelmid.c | 1761 ++++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 1761 insertions(+), 0 deletions(-) create mode 100644 sound/pci/sst/intelmid.c diff --git a/sound/pci/sst/intelmid.c b/sound/pci/sst/intelmid.c new file mode 100644 index 0000000..28a6dcc --- /dev/null +++ b/sound/pci/sst/intelmid.c @@ -0,0 +1,1761 @@ +/* + * intelmid.c - Intel Sound card driver for MID + * + * Copyright (C) 2008-09 Intel Corp + * Authors: Harsha Priya <priya.harsha(a)intel.com> + * Vinod Koul <vinod.koul(a)intel.com> + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License as published by + * the Free Software Foundation; version 2 of the License. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License along + * with this program; if not, write to the Free Software Foundation, Inc., + * 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA. + * + * ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ + * ALSA driver for Intel MID sound card chipset + */ +#include <sound/core.h> +#include <sound/control.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/info.h> +#include <sound/initval.h> +#include <sound/pcm-indirect.h> +#include <sound/intel_sst.h> +#include <sound/intel_sst_ioctl.h> +#include "intelmid_snd_control.h" +#include "intelmid.h" +#include "intelmid_pvt.h" +#include <linux/spi/spi.h> +#include <linux/io.h> +#include <linux/delay.h> +#include <linux/interrupt.h> +#include <linux/init.h> +#include <linux/slab.h> +#include <linux/moduleparam.h> +#include <linux/sched.h> +#include <linux/gpe.h> +#include <net/netlink.h> +#include <net/genetlink.h> + +MODULE_AUTHOR("Harsha Priya <priya.harsha(a)intel.com>"); +MODULE_AUTHOR("Vinod Koul <vinod.koul(a)intel.com>"); +MODULE_DESCRIPTION("Intel MAD Sound card driver"); +MODULE_LICENSE("GPL v2"); +MODULE_SUPPORTED_DEVICE("{Intel,Intel_MAD}"); + +static int card_index = SNDRV_DEFAULT_IDX1;/* Index 0-MAX */ +static char *card_id = SNDRV_DEFAULT_STR1; /* ID for this card */ +static int vendor_id; +static unsigned int audio_event_seqnum; +static int cum_bytes; + + +#ifdef FULL_CTRL +static char *out_names[] = {"Headphones", + "Internal speakers", + "EarPiece"}; +static char *in_names[] = { "DMIC", + "MIC1", + "MIC2", + "Line_in"}; +#endif +static struct genl_family audio_event_genl_family = { + .id = GENL_ID_GENERATE, + .name = "audio events", + .version = 0x01, + .maxattr = 0, +}; + +static struct genl_multicast_group audio_event_mcgrp = { + .name = "audio_group", +}; + +/*control path functionalities*/ +/** +* snd_intelmad_mute_info - provides information about the mute controls +* @kcontrol: pointer to the control +* @uinfo: pointer to the structure where the control's info need +* to be filled +* This function is called when a mixer application requests for control's info +*/ +static int snd_intelmad_mute_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + WARN_ON(!uinfo); + WARN_ON(!kcontrol); + + /*set up the mute as a boolean mono control with min-max values*/ + uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN; + uinfo->count = MONO_CNTL; + uinfo->value.integer.min = MIN_MUTE; + uinfo->value.integer.max = MAX_MUTE; + return 0; +} + +#ifdef FULL_CTRL +/** +* snd_intelmad_volume_line_info - provides information about the volume control +* @kcontrol: pointer to the control +* @uinfo: pointer to the structure where the control's info need +* to be filled +* This function is called when a mixer application requests for control's info +*/ +static int snd_intelmad_line_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) + +{ + snd_intelmad_volume_info(uinfo, STEREO_CNTL, + ctrl_val[vendor_id].linein_vol_max, + ctrl_val[vendor_id].linein_vol_min); + return 0; +} + +/** +* snd_intelmad_mic1_volume_info - provides information about the volume control +* @kcontrol: pointer to the control +* @uinfo: pointer to the structure where the control's info need +* to be filled +* This function is called when a mixer application requests for control's info +*/ +static int snd_intelmad_mic1_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + snd_intelmad_volume_info(uinfo, STEREO_CNTL, + ctrl_val[vendor_id].mic1_vol_max, + ctrl_val[vendor_id].mic1_vol_min); + return 0; +} + +/** +* snd_intelmad_mic2_volume_info - provides information about the volume control +* @kcontrol: pointer to the control +* @uinfo: pointer to the structure where the control's info need +* to be filled +* This function is called when a mixer application requests for control's info +*/ +static int snd_intelmad_mic2_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + snd_intelmad_volume_info(uinfo, STEREO_CNTL, + ctrl_val[vendor_id].mic2_vol_max, + ctrl_val[vendor_id].mic2_vol_min); + return 0; +} + +/** +* snd_intelmad_dmic_volume_info - provides information about the volume control +* @kcontrol: pointer to the control +* @uinfo: pointer to the structure where the control's info need +* to be filled +* This function is called when a mixer application requests for control's info +*/ +static int snd_intelmad_dmic_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + snd_intelmad_volume_info(uinfo, STEREO_CNTL, + ctrl_val[vendor_id].dmic_vol_max, + ctrl_val[vendor_id].dmic_vol_min); + return 0; +} +#endif +/** +* snd_intelmad_hp_volume_info - provides information about the volume control +* @kcontrol: pointer to the control +* @uinfo: pointer to the structure where the control's info need +* to be filled +* This function is called when a mixer application requests for control's info +*/ +static int snd_intelmad_hp_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + snd_intelmad_volume_info(uinfo, STEREO_CNTL, + ctrl_val[vendor_id].hp_vol_max, + ctrl_val[vendor_id].hp_vol_min); + return 0; +} + +#ifdef FULL_CTRL +/** +* snd_intelmad_speaker_volume_info - provides information about the volume control +* @kcontrol: pointer to the control +* @uinfo: pointer to the structure where the control's info need +* to be filled +* This function is called when a mixer application requests for control's info +*/ +static int snd_intelmad_speaker_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + snd_intelmad_volume_info(uinfo, STEREO_CNTL, + ctrl_val[vendor_id].speaker_vol_max, + ctrl_val[vendor_id].speaker_vol_min); + return 0; +} + +/** +* snd_intelmad_earpiece_volume_info - provides information about the volume control +* @kcontrol: pointer to the control +* @uinfo: pointer to the structure where the control's info need +* to be filled +* This function is called when a mixer application requests for control's info +*/ +static int snd_intelmad_earpiece_volume_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + snd_intelmad_volume_info(uinfo, MONO_CNTL, + ctrl_val[vendor_id].earpiece_vol_max, + ctrl_val[vendor_id].earpiece_vol_min); + return 0; +} + +/** +* snd_intelmad_device_info - provides information about the devices available +* @kcontrol: pointer to the control +* @uinfo: pointer to the structure where the devices's info need +* to be filled +* This function is called when a mixer application requests for device's info +*/ +static int snd_intelmad_device_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + WARN_ON(!kcontrol); + WARN_ON(!uinfo); + /*setup device select as drop down controls with different values*/ + if (OUTPUT_SEL == kcontrol->id.numid) + uinfo->value.enumerated.items = ARRAY_SIZE(out_names); + else + uinfo->value.enumerated.items = ARRAY_SIZE(in_names); + uinfo->count = MONO_CNTL; + uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED; + + if (uinfo->value.enumerated.item >= uinfo->value.enumerated.items) + uinfo->value.enumerated.item = 1; + if (OUTPUT_SEL == kcontrol->id.numid) + strncpy(uinfo->value.enumerated.name, + out_names[uinfo->value.enumerated.item], + strlen(out_names[uinfo->value.enumerated.item])); + else + strncpy(uinfo->value.enumerated.name, + in_names[uinfo->value.enumerated.item], + strlen(in_names[uinfo->value.enumerated.item])); + return 0; +} +#endif + +/** +* snd_intelmad_volume_get - gets the current volume for the control +* @kcontrol: pointer to the control +* @uval: pointer to the structure where the control's info need +* to be filled +* This function is called when .get function of a control is invoked from app +*/ +static int snd_intelmad_volume_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uval) +{ + int ret_val = 0, cntl_list[2] = {0,}, value = 0; + struct snd_intelmad *intelmaddata = NULL; + struct snd_pmic_ops *scard_ops = NULL; + + sst_dbg("called\n"); + + WARN_ON(!uval); + WARN_ON(!kcontrol); + + intelmaddata = kcontrol->private_data; + + WARN_ON(!intelmaddata->sstdrv_ops); + + if (PMIC_UNINIT == intelmaddata->pmic_status) { + ret_val = intelmaddata->sstdrv_ops->scard_ops->init_card(); + intelmaddata->pmic_status = PMIC_INIT; + } + + if (0 != ret_val) + return ret_val; + + scard_ops = intelmaddata->sstdrv_ops->scard_ops; + + WARN_ON(!scard_ops); + + switch (kcontrol->id.numid) { + case HEADPHONES_VOL: + cntl_list[0] = PMIC_SND_RIGHT_HP_VOL; + cntl_list[1] = PMIC_SND_LEFT_HP_VOL; + break; +#ifdef FULL_CTRL + case INTERNAL_SPEAKERS_VOL: + cntl_list[0] = PMIC_SND_RIGHT_SPEAKER_VOL; + cntl_list[1] = PMIC_SND_LEFT_SPEAKER_VOL; + break; + case LINE_IN_VOL: + cntl_list[0] = PMIC_SND_INPUT_VOL_RIGHT_LINE_IN; + cntl_list[1] = PMIC_SND_INPUT_VOL_LEFT_LINE_IN; + break; + case EARPIECE_VOL: + cntl_list[0] = PMIC_SND_MONO_EARPIECE_VOL; + break; + case MIC1_VOL: + cntl_list[0] = PMIC_SND_INPUT_VOL_MIC1; + break; + case MIC2_VOL: + cntl_list[0] = PMIC_SND_INPUT_VOL_MIC2; + break; + case DMIC_VOL: + cntl_list[0] = PMIC_SND_INPUT_VOL_DMIC; + break; +#endif + default: + return -EINVAL; + } + + ret_val = scard_ops->get_vol(cntl_list[0], &value); + uval->value.integer.value[0] = value; + + if (0 != ret_val) + return ret_val; +#ifdef FULL_CTRL + if (HEADPHONES_VOL == kcontrol->id.numid + || INTERNAL_SPEAKERS_VOL == kcontrol->id.numid || + LINE_IN_VOL == kcontrol->id.numid) { +#else + if (1) { +#endif + + ret_val = scard_ops->get_vol(cntl_list[1], &value); + uval->value.integer.value[1] = value; + + if (0 != ret_val) + return ret_val; + } + return ret_val; +} + +/** +* snd_intelmad_mute_get - gets the current mute status for the control +* @kcontrol: pointer to the control +* @uval: pointer to the structure where the control's info need +* to be filled +* This function is called when .get function of a control is invoked from app +*/ +static int snd_intelmad_mute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uval) +{ + + int cntl_list = 0, ret_val = 0, value = 0; + struct snd_intelmad *intelmaddata = NULL; + struct snd_pmic_ops *scard_ops = NULL; + + sst_dbg("called\n"); + + WARN_ON(!uval); + WARN_ON(!kcontrol); + + intelmaddata = kcontrol->private_data; + + WARN_ON(!intelmaddata->sstdrv_ops); + + if (PMIC_UNINIT == intelmaddata->pmic_status) { + ret_val = intelmaddata->sstdrv_ops->scard_ops->init_card(); + intelmaddata->pmic_status = PMIC_INIT; + } + + if (0 != ret_val) + return ret_val; + + scard_ops = intelmaddata->sstdrv_ops->scard_ops; + + WARN_ON(!scard_ops); + + switch (kcontrol->id.numid) { + case HEADPHONES_MUTE: + cntl_list = PMIC_SND_LEFT_HP_MUTE; + break; +#ifdef FULL_CTRL + case INTERNAL_SPEAKERS_MUTE: + cntl_list = PMIC_SND_LEFT_SPEAKER_MUTE; + break; + case EARPIECE_MUTE: + cntl_list = PMIC_SND_MONO_EARPIECE_MUTE; + break; + case LINE_IN_MUTE: + cntl_list = PMIC_SND_INPUT_MUTE_LINE_IN; + break; + case MIC1_MUTE: + cntl_list = PMIC_SND_INPUT_MUTE_MIC1; + break; + case MIC2_MUTE: + cntl_list = PMIC_SND_INPUT_MUTE_MIC2; + break; + case DMIC_MUTE: + cntl_list = PMIC_SND_INPUT_MUTE_DMIC; + break; + case MASTER_MUTE: + uval->value.integer.value[0] = kcontrol->private_value; + return 0; +#endif + + default: + return -EINVAL; + } + + ret_val = scard_ops->get_mute(cntl_list, &value); + uval->value.integer.value[0] = value; + return ret_val; +} + +/** +* snd_intelmad_volume_set - sets the volume control's info +* @kcontrol: pointer to the control +* @uval: pointer to the structure where the control's info is +* available to be set +* This function is called when .set function of a control is invoked from app +*/ +static int snd_intelmad_volume_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uval) +{ + + int ret_val = 0, cntl_list[2] = {0,}; + struct snd_intelmad *intelmaddata = NULL; + struct snd_pmic_ops *scard_ops = NULL; + + sst_dbg("volume set called\n:%ld %ld", + uval->value.integer.value[0], + uval->value.integer.value[1]); + + WARN_ON(!uval); + WARN_ON(!kcontrol); + + intelmaddata = kcontrol->private_data; + + WARN_ON(!intelmaddata->sstdrv_ops); + + if (PMIC_UNINIT == intelmaddata->pmic_status) { + ret_val = intelmaddata->sstdrv_ops->scard_ops->init_card(); + intelmaddata->pmic_status = PMIC_INIT; + } + + if (0 != ret_val) + return ret_val; + + scard_ops = intelmaddata->sstdrv_ops->scard_ops; + + WARN_ON(!scard_ops); + + switch (kcontrol->id.numid) { + case HEADPHONES_VOL: + cntl_list[0] = PMIC_SND_RIGHT_HP_VOL; + cntl_list[1] = PMIC_SND_LEFT_HP_VOL; + break; +#ifdef FULL_CTRL + case INTERNAL_SPEAKERS_VOL: + cntl_list[0] = PMIC_SND_RIGHT_SPEAKER_VOL; + cntl_list[1] = PMIC_SND_LEFT_SPEAKER_VOL; + break; + case LINE_IN_VOL: + cntl_list[0] = PMIC_SND_INPUT_VOL_RIGHT_LINE_IN; + cntl_list[1] = PMIC_SND_INPUT_VOL_LEFT_LINE_IN; + break; + case EARPIECE_VOL: + cntl_list[0] = PMIC_SND_MONO_EARPIECE_VOL; + break; + case MIC1_VOL: + cntl_list[0] = PMIC_SND_INPUT_VOL_MIC1; + break; + case MIC2_VOL: + cntl_list[0] = PMIC_SND_INPUT_VOL_MIC2; + break; + case DMIC_VOL: + cntl_list[0] = PMIC_SND_INPUT_VOL_DMIC; + break; +#endif + default: + return -EINVAL; + } + + ret_val = scard_ops->set_vol(cntl_list[0], + uval->value.integer.value[0]); + if (0 != ret_val) + return ret_val; +#ifdef FULL_CTRL + if (HEADPHONES_VOL == kcontrol->id.numid + || INTERNAL_SPEAKERS_VOL == kcontrol->id.numid || + LINE_IN_VOL == kcontrol->id.numid) { +#else + if (1) { +#endif + ret_val = scard_ops->set_vol(cntl_list[1], + uval->value.integer.value[1]); + if (0 != ret_val) + return ret_val; + } + kcontrol->private_value = uval->value.integer.value[0]; + return ret_val; +} + + + +/** +* snd_intelmad_mute_set - sets the mute control's info +* @kcontrol: pointer to the control +* @uval: pointer to the structure where the control's info is +* available to be set +* This function is called when .set function of a control is invoked from app +*/ +static int snd_intelmad_mute_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uval) +{ + int cntl_list[2] = {0,}, ret_val = 0; + struct snd_intelmad *intelmaddata = NULL; + struct snd_pmic_ops *scard_ops = NULL; + + sst_dbg("called\n"); + + WARN_ON(!uval); + WARN_ON(!kcontrol); + + intelmaddata = kcontrol->private_data; + + WARN_ON(!intelmaddata->sstdrv_ops); + + if (PMIC_UNINIT == intelmaddata->pmic_status) { + ret_val = intelmaddata->sstdrv_ops->scard_ops->init_card(); + intelmaddata->pmic_status = PMIC_INIT; + } + + if (0 != ret_val) + return ret_val; + + scard_ops = intelmaddata->sstdrv_ops->scard_ops; + + WARN_ON(!scard_ops); + + kcontrol->private_value = uval->value.integer.value[0]; + + switch (kcontrol->id.numid) { + case HEADPHONES_MUTE: + cntl_list[0] = PMIC_SND_LEFT_HP_MUTE; + cntl_list[1] = PMIC_SND_RIGHT_HP_MUTE; + break; +#ifdef FULL_CTRL + case INTERNAL_SPEAKERS_MUTE: + cntl_list[0] = PMIC_SND_LEFT_SPEAKER_MUTE; + cntl_list[1] = PMIC_SND_RIGHT_SPEAKER_MUTE; + break; + case EARPIECE_MUTE: + cntl_list[0] = PMIC_SND_MONO_EARPIECE_MUTE; + break; + case LINE_IN_MUTE: + cntl_list[0] = PMIC_SND_INPUT_MUTE_LINE_IN; + break; + case MIC1_MUTE: + cntl_list[0] = PMIC_SND_INPUT_MUTE_MIC1; + break; + case MIC2_MUTE: + cntl_list[0] = PMIC_SND_INPUT_MUTE_MIC2; + break; + case DMIC_MUTE: + cntl_list[0] = PMIC_SND_INPUT_MUTE_DMIC; + break; + case MASTER_MUTE: + mute_all(kcontrol, uval); + kcontrol->private_value = uval->value.integer.value[0]; + return 0; +#endif + default: + return -EINVAL; + } + + ret_val = scard_ops->set_mute(cntl_list[0], + uval->value.integer.value[0]); + if (0 != ret_val) + return ret_val; + +#ifdef FULL_CTRL + if (HEADPHONES_MUTE == kcontrol->id.numid + || INTERNAL_SPEAKERS_MUTE == kcontrol->id.numid) +#endif + ret_val = scard_ops->set_mute(cntl_list[1], + uval->value.integer.value[0]); + kcontrol->private_value = uval->value.integer.value[0]; + return ret_val; +} + +#ifdef FULL_CTRL +/** +* snd_intelmad_device_get - get the device select control's info +* @kcontrol: pointer to the control +* @uval: pointer to the structure where the control's info is +* to be filled +* This function is called when .get function of a control is invoked from app +*/ +static int snd_intelmad_device_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uval) +{ + sst_dbg("called\n"); + + WARN_ON(!uval); + WARN_ON(!kcontrol); + + uval->value.enumerated.item[0] = kcontrol->private_value; + return 0; +} + +/** +* snd_intelmad_device_set - set the device select control's info +* @kcontrol: pointer to the control +* @uval: pointer to the structure where the control's info is +* available to be set +* This function is called when .set function of a control is invoked from app +*/ +static int snd_intelmad_device_set(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *uval) +{ + struct snd_intelmad *intelmaddata = NULL; + struct snd_pmic_ops *scard_ops = NULL; + int ret_val = 0; + + sst_dbg("called\n"); + + WARN_ON(!uval); + WARN_ON(!kcontrol); + + intelmaddata = kcontrol->private_data; + + WARN_ON(!intelmaddata->sstdrv_ops); + + if (PMIC_UNINIT == intelmaddata->pmic_status) { + ret_val = intelmaddata->sstdrv_ops->scard_ops->init_card(); + intelmaddata->pmic_status = PMIC_INIT; + } + + if (0 != ret_val) + return ret_val; + + scard_ops = intelmaddata->sstdrv_ops->scard_ops; + + WARN_ON(!scard_ops); + + /*store value with driver*/ + kcontrol->private_value = uval->value.enumerated.item[0]; + + switch (kcontrol->id.numid) { + case OUTPUT_SEL: + ret_val = scard_ops->set_output_dev( + uval->value.enumerated.item[0]); + break; + case INPUT_SEL: + ret_val = scard_ops->set_input_dev( + uval->value.enumerated.item[0]); + break; + default: + return -EINVAL; + } + kcontrol->private_value = uval->value.enumerated.item[0]; + return ret_val; +} +#endif + +/*structure of all the controls of the sound card that is exposed*/ +static struct snd_kcontrol_new snd_intelmad_controls[] __devinitdata = { +#ifdef FULL_CTRL +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_line_volume_info, + .get = snd_intelmad_volume_get, + .put = snd_intelmad_volume_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line Capture Switch", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_mute_info, + .get = snd_intelmad_mute_get, + .put = snd_intelmad_mute_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "DMIC Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_dmic_volume_info, + .get = snd_intelmad_volume_get, + .put = snd_intelmad_volume_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "DMIC Capture Switch", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_mute_info, + .get = snd_intelmad_mute_get, + .put = snd_intelmad_mute_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic1 Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_mic1_volume_info, + .get = snd_intelmad_volume_get, + .put = snd_intelmad_volume_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic1 Capture Switch", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_mute_info, + .get = snd_intelmad_mute_get, + .put = snd_intelmad_mute_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic2 Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_mic2_volume_info, + .get = snd_intelmad_volume_get, + .put = snd_intelmad_volume_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mic2 Capture Switch", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_mute_info, + .get = snd_intelmad_mute_get, + .put = snd_intelmad_mute_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Capture Source", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_device_info, + .get = snd_intelmad_device_get, + .put = snd_intelmad_device_set, + .private_value = 0, +}, +#endif +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Headphone Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_hp_volume_info, + .get = snd_intelmad_volume_get, + .put = snd_intelmad_volume_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Headphone Playback Switch", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_mute_info, + .get = snd_intelmad_mute_get, + .put = snd_intelmad_mute_set, + .private_value = 0, +}, +#ifdef FULL_CTRL +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Internal Speaker Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_speaker_volume_info, + .get = snd_intelmad_volume_get, + .put = snd_intelmad_volume_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Internal Speaker Playback Switch", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_mute_info, + .get = snd_intelmad_mute_get, + .put = snd_intelmad_mute_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mono Earpiece Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_earpiece_volume_info, + .get = snd_intelmad_volume_get, + .put = snd_intelmad_volume_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Mono Earpiece Playback Switch", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_mute_info, + .get = snd_intelmad_mute_get, + .put = snd_intelmad_mute_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Source", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_device_info, + .get = snd_intelmad_device_get, + .put = snd_intelmad_device_set, + .private_value = 0, +}, +{ + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Mute Switch", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = snd_intelmad_mute_info, + .get = snd_intelmad_mute_get, + .put = snd_intelmad_mute_set, + .private_value = 0, +}, +#endif +}; + +/*Data path functionalities*/ +static struct snd_pcm_hardware snd_intelmad_stream = { + .info = (SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_DOUBLE | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_MMAP| + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_SYNC_START), + .formats = (SNDRV_PCM_FMTBIT_S16 | SNDRV_PCM_FMTBIT_U16 | + /*SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_U8 |*/ + SNDRV_PCM_FMTBIT_S24 | SNDRV_PCM_FMTBIT_U24), + .rates = (SNDRV_PCM_RATE_8000| + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000), + .rate_min = MIN_RATE, + + .rate_max = MAX_RATE, + .channels_min = MIN_CHANNEL, + .channels_max = MAX_CHANNEL, + .buffer_bytes_max = MAX_BUFFER, + .period_bytes_min = MIN_PERIOD_BYTES, + .period_bytes_max = MAX_PERIOD_BYTES, + .periods_min = MIN_PERIODS, + .periods_max = MAX_PERIODS, + .fifo_size = FIFO_SIZE, +}; + +static void send_buffer_to_sst(struct snd_pcm_substream *substream, + struct snd_pcm_indirect *rec, size_t bytes) +{ + struct snd_intelmad *intelmaddata = snd_pcm_substream_chip(substream); + struct stream_buffer buffer_to_sst = {0,}; + struct mad_stream_pvt *stream = substream->runtime->private_data; + int ret_val = 0; + /*sends data to SST to be processed*/ + + cum_bytes += bytes; + sst_dbg("bytes = %d \n", bytes); + sst_dbg("cum_bytes = 0x%x, \n", cum_bytes); + buffer_to_sst.length = bytes; + buffer_to_sst.addr = (unsigned long) substream->runtime->dma_area + + rec->sw_data; + /*SST API to actually send the buffer to be played*/ + ret_val = intelmaddata->sstdrv_ops->send_buffer(stream->stream_id, + &buffer_to_sst); + sst_dbg("send_buffer ret_val = 0x%x \n", ret_val); + return; +} + +void start_capture(struct work_struct *work) +{ + int ret_val = 0; + struct mad_capture_wq *stream_work = + container_of(work, struct mad_capture_wq, wq); + struct stream_buffer *buffer_to_sst = &stream_work->buffer_to_sst; + struct snd_pcm_substream *substream = stream_work->substream; + struct mad_stream_pvt *stream = substream->runtime->private_data; + struct snd_intelmad *intelmaddata = snd_pcm_substream_chip(substream); + + ret_val = intelmaddata->sstdrv_ops->send_buffer(stream->stream_id, + buffer_to_sst); + sst_dbg("send_buffer ret_val = 0x%x \n", ret_val); + return; +} + +void snd_card_pcm_wq_function(struct work_struct *work) +{ + struct mad_wq *stream_work = + container_of(work, struct mad_wq, wq); + struct snd_pcm_substream *substream; + struct snd_pcm_indirect *rec = NULL; + struct mad_stream_pvt *stream; + + mutex_lock(&stream_work->wq_lock); + substream = stream_work->substream; + if (substream == NULL || substream->runtime == NULL) { + sst_err("subsream structure is null \n"); + mutex_unlock(&stream_work->wq_lock); + return; + } + stream = substream->runtime->private_data; + sst_dbg("called %d\n", stream->stream_status); + if (stream == NULL) { + sst_err("stream is null in wq function \n"); + mutex_unlock(&stream_work->wq_lock); + return; + } + + if (stream->stream_status != INIT) { + rec = &stream->pcm_indirect; + if (substream->stream == STREAM_OPS_PLAYBACK) + snd_pcm_indirect_playback_transfer(substream, rec, + send_buffer_to_sst); + else if (substream->stream == STREAM_OPS_CAPTURE) { + sst_dbg("calling indirect capture transfer\n"); + snd_pcm_indirect_capture_transfer(substream, rec, + send_buffer_to_sst); + } + stream->stream_status = RUNNING; + } + mutex_unlock(&stream_work->wq_lock); + return; +} + +static int snd_intelmad_pcm_ack(struct snd_pcm_substream *substream) +{ + struct mad_stream_pvt *stream; + + stream = substream->runtime->private_data; + sst_dbg("...called\n"); + stream->mad_msg_wq.substream = substream; + schedule_work(&stream->mad_msg_wq.wq); + return 0; +} + +/** +* snd_intelmad_pcm_trigger - stream activities are handled here +* @substream:substream for which the stream function is called +*@cmd:the stream commamd thats requested from upper layer +* This function is called whenever an a stream activity is invoked +*/ +static int snd_intelmad_pcm_trigger(struct snd_pcm_substream *substream, + int cmd) +{ + int ret_val = 0, stream_type = 0/*, sample_size = 0*/; + struct snd_intelmad *intelmaddata = NULL; + struct mad_stream_pvt *stream = NULL; + + sst_dbg("called\n"); + + WARN_ON(!substream); + + intelmaddata = snd_pcm_substream_chip(substream); + stream_type = substream->stream; + stream = substream->runtime->private_data; + + if (PMIC_UNINIT == intelmaddata->pmic_status) { + ret_val = intelmaddata->sstdrv_ops->scard_ops->init_card(); + intelmaddata->pmic_status = PMIC_INIT; + } + + if (0 != ret_val) + return ret_val; + + WARN_ON(!intelmaddata->sstdrv_ops); + WARN_ON(!intelmaddata->sstdrv_ops->scard_ops); + + switch (cmd) { + case SNDRV_PCM_TRIGGER_RESUME: + /*invoked on a power managment -resume operation*/ + /*stream parameters and context resuming to be done here*/ + /*TBD*/ + sst_dbg("in resume\n"); + break; + case SNDRV_PCM_TRIGGER_START: + stream->substream = substream; + stream->pcm_size = snd_pcm_lib_buffer_bytes(substream); + sst_dbg("pcm_size=%d\n", stream->pcm_size); +#ifdef TIMER_TS + bps = (substream->runtime->sample_bits/8) + * (substream->runtime->channels) + * (substream->runtime->rate); + stream->pcm_jiffie = ((bps * ELAPSED_PERIOD) / HZ); + stream->pcm_buf_pos = 0; + sst_dbg("pcm_jiffie=%d \n", stream->pcm_jiffie); +#endif + stream->stream_status = STARTED; + if (substream->stream == STREAM_OPS_PLAYBACK) + snd_intelmad_pcm_ack(substream); + else if (substream->stream == STREAM_OPS_CAPTURE) { + stream->mad_capture_wq.substream = substream; + stream->mad_capture_wq.buffer_to_sst.length = + frames_to_bytes(substream->runtime, + substream->runtime->period_size); + stream->mad_capture_wq.buffer_to_sst.addr = + (unsigned long) substream->runtime->dma_area; + schedule_work(&stream->mad_capture_wq.wq); + cum_bytes = stream->mad_capture_wq.buffer_to_sst.length; + stream->stream_status = RUNNING; + } + break; + case SNDRV_PCM_TRIGGER_SUSPEND: + /*invoked on suspend - power management event*/ + /*stream parameters and context to be saved*/ + /*TBD*/ + sst_dbg("in suspend\n"); + break; + case SNDRV_PCM_TRIGGER_STOP: + sst_dbg("in stop\n"); + /*ret_val = intelmaddata->sstdrv_ops->control_set(SST_SND_DROP, + &stream->stream_id); + if (0 != ret_val) + return ret_val;*/ + stream->stream_status = DROPPED; + break; + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + sst_dbg("in pause\n"); + ret_val = intelmaddata->sstdrv_ops->control_set(SST_SND_PAUSE, + &stream->stream_id); + if (0 != ret_val) + return ret_val; + stream->stream_status = PAUSED; + break; + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + sst_dbg("in pause release \n"); + ret_val = intelmaddata->sstdrv_ops->control_set(SST_SND_RESUME, + &stream->stream_id); + if (0 != ret_val) + return ret_val; + stream->stream_status = RUNNING; + break; + default: + return -EINVAL; + } + return ret_val; +} + +/** +* snd_intelmad_pcm_prepare- internal preparation before starting a stream +* @substream: substream for which the function is called +* This function is called when a stream is started for internal preparation. +*/ +static int snd_intelmad_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct mad_stream_pvt *stream = NULL; + int ret_val = 0; + + sst_dbg("called \n"); + + WARN_ON(!substream); + + stream = substream->runtime->private_data; + + sst_dbg("format = %d \n", substream->runtime->format); + if (stream->stream_id == 0) { + ret_val = snd_intelmad_alloc_stream(substream); + cum_bytes = 0; + if (ret_val < 0) + return ret_val; + ret_val = snd_intelmad_init_stream(substream); + if (0 != ret_val) + return ret_val; + sst_dbg("period size = %d \n", + (int)substream->runtime->period_size); + sst_dbg("buf size = %d \n", + (int)substream->runtime->buffer_size); + memset(&stream->pcm_indirect, 0, sizeof(stream->pcm_indirect)); + stream->pcm_indirect.hw_buffer_size = + snd_pcm_lib_buffer_bytes(substream); + stream->pcm_indirect.sw_buffer_size = + stream->pcm_indirect.hw_buffer_size; + /*return back the stream id*/ + snprintf(substream->pcm->id, sizeof(substream->pcm->id), + "%d", stream->stream_id); + sst_dbg("stream id to user = %s\n", substream->pcm->id); + } + return ret_val; +} + +static int snd_intelmad_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *hw_params) +{ + int ret_val = 0; + + sst_dbg("called\n"); + ret_val = snd_pcm_lib_malloc_pages(substream, + params_buffer_bytes(hw_params)); + memset(substream->runtime->dma_area, 0, params_buffer_bytes(hw_params)); + return ret_val; +} + +static int snd_intelmad_hw_free(struct snd_pcm_substream *substream) +{ + sst_dbg("called\n"); + return snd_pcm_lib_free_pages(substream); +} + +/** +* snd_intelmad_pcm_pointer- to send the current buffer pointer processed by hw +* @substream: substream for which the function is called +* This function is called by ALSA framework to get the current hw buffer ptr +* when a period is elapsed +*/ +static snd_pcm_uframes_t snd_intelmad_pcm_pointer + (struct snd_pcm_substream *substream) +{ + /*struct snd_pcm_runtime *runtime = substream->runtime;*/ + struct mad_stream_pvt *stream = NULL; + struct snd_intelmad *intelmaddata = NULL; + int ret_val = 0; + unsigned long buf_size = 0; + + WARN_ON(!substream); + + intelmaddata = snd_pcm_substream_chip(substream); + stream = substream->runtime->private_data; + if (stream->stream_status == INIT) + return 0; + + stream->buf_ptr.str_id = stream->stream_id; + +#ifndef TIMER_TS + ret_val = intelmaddata->sstdrv_ops->control_set(SST_SND_BUFFER_POINTER, + &stream->buf_ptr); + if (0 != ret_val) { + sst_err("error code = 0x%x \n", ret_val); + return ret_val; + } + sst_dbg("samples reported out 0x%lx \n", stream->buf_ptr.buffer_ptr); + buf_size = frames_to_bytes(substream->runtime, + stream->buf_ptr.buffer_ptr); + sst_dbg(" bytes reported out = 0x%lx\n", buf_size); + if (buf_size > cum_bytes) + sst_err("excess reported\n"); + + if (substream->stream == STREAM_OPS_PLAYBACK) + stream->period_elapsed_ptr = snd_pcm_indirect_playback_pointer( + substream, &stream->pcm_indirect, buf_size); + else + stream->period_elapsed_ptr = snd_pcm_indirect_capture_pointer( + substream, &stream->pcm_indirect, buf_size); + + sst_dbg("indirect value = 0x%lx\n", stream->period_elapsed_ptr); + + return stream->period_elapsed_ptr; +#else + stream->pcm_buf_pos += stream->pcm_jiffie; + stream->pcm_buf_pos %= stream->pcm_size; + return snd_pcm_indirect_playback_pointer(substream, + &stream->pcm_indirect, stream->pcm_buf_pos); +#endif + +} + +/** +* snd_intelmad_close- to free parameteres when stream is stopped +* @substream: substream for which the function is called +* This function is called by ALSA framework when stream is stopped +*/ +static int snd_intelmad_close(struct snd_pcm_substream *substream) +{ + struct snd_intelmad *intelmaddata = NULL; + struct mad_stream_pvt *stream = NULL; + int ret_val = 0; + + WARN_ON(!substream); + + stream = substream->runtime->private_data; + + sst_dbg("called \n"); + intelmaddata = snd_pcm_substream_chip(substream); + mutex_lock(&stream->mad_msg_wq.wq_lock); + sst_dbg("str id = %d\n", stream->stream_id); + /*SST API to actually stop/free the stream*/ + ret_val = intelmaddata->sstdrv_ops->control_set(SST_SND_FREE, + &stream->stream_id); + kfree(substream->runtime->private_data); + mutex_unlock(&stream->mad_msg_wq.wq_lock); + return ret_val; +} + +/** +* snd_intelmad_open- to set runtime parameters during stream start +* @substream: substream for which the function is called +* This function is called by ALSA framework when stream is started +*/ +static int snd_intelmad_open(struct snd_pcm_substream *substream) +{ + struct snd_intelmad *intelmaddata; + struct snd_pcm_runtime *runtime; + struct mad_stream_pvt *stream; + + WARN_ON(!substream); + + sst_dbg("called \n"); + + intelmaddata = snd_pcm_substream_chip(substream); + runtime = substream->runtime; + /*set the runtime hw parameter with our local snd_pcm_hardware struct*/ + runtime->hw = snd_intelmad_stream; + /*setup the internal datastruture stream pointers based on it being + playback or capture stream*/ + stream = kzalloc(sizeof(*stream), GFP_KERNEL); + if (NULL == stream) + return -ENOMEM; + stream->stream_id = 0; + stream->stream_status = INIT; + INIT_WORK(&stream->mad_msg_wq.wq, snd_card_pcm_wq_function); + INIT_WORK(&stream->mad_capture_wq.wq, start_capture); + mutex_init(&stream->mad_msg_wq.wq_lock); + runtime->private_data = stream; + return 0; +} + +static struct snd_pcm_ops snd_intelmad_playback_ops = { + .open = snd_intelmad_open, + .close = snd_intelmad_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_intelmad_hw_params, + .hw_free = snd_intelmad_hw_free, + .prepare = snd_intelmad_pcm_prepare, + .trigger = snd_intelmad_pcm_trigger, + .pointer = snd_intelmad_pcm_pointer, + .ack = snd_intelmad_pcm_ack, +}; + +static struct snd_pcm_ops snd_intelmad_capture_ops = { + .open = snd_intelmad_open, + .close = snd_intelmad_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_intelmad_hw_params, + .hw_free = snd_intelmad_hw_free, + .prepare = snd_intelmad_pcm_prepare, + .trigger = snd_intelmad_pcm_trigger, + .pointer = snd_intelmad_pcm_pointer, + .ack = snd_intelmad_pcm_ack, +}; + +int snd_intelmad_generate_netlink(u32 orig, enum eaudio_events event) +{ + struct sk_buff *skb = NULL; + struct nlattr *attr = NULL; + struct audio_genl_event *aud_event = NULL; + void *msg_header = NULL; + int size = 0, ret_val = 0; + + /* allocate memory */ + size = nla_total_size(sizeof(struct audio_genl_event)) + \ + nla_total_size(0); + + skb = genlmsg_new(size, GFP_ATOMIC); + if (!skb) + return -ENOMEM; + + /* add the genetlink message header */ + msg_header = genlmsg_put(skb, 0, audio_event_seqnum++, + &audio_event_genl_family, 0, + AUDIO_GENL_CMD_EVENT); + if (!msg_header) { + nlmsg_free(skb); + return -ENOMEM; + } + + /* fill the data */ + attr = nla_reserve(skb, AUDIO_GENL_ATTR_EVENT, \ + sizeof(struct audio_genl_event)); + + if (!attr) { + nlmsg_free(skb); + return -EINVAL; + } + + aud_event = nla_data(attr); + if (!aud_event) { + nlmsg_free(skb); + return -EINVAL; + } + + memset(aud_event, 0, sizeof(struct audio_genl_event)); + + aud_event->orig = orig; + aud_event->event = event; + + /* send multicast genetlink message */ + ret_val = genlmsg_end(skb, msg_header); + if (ret_val < 0) { + nlmsg_free(skb); + return ret_val; + } + + ret_val = genlmsg_multicast(skb, 0, audio_event_mcgrp.id, GFP_ATOMIC); + + if (ret_val) + printk(KERN_INFO "Failed to send a Genetlink message!\n"); + return 0; +} + +#ifdef REG_IRQ +/** +* snd_intelmad_intr_handler- interrupt handler +*@irq : irq number of the interrupt received +*@dev: device context +* This function is called when an interrupt is raised at the sound card +*/ +static irqreturn_t snd_intelmad_intr_handler(int irq, void *dev) +{ + struct snd_intelmad *intelmaddata = + (struct snd_intelmad *)dev; + struct sc_reg_access pmic_reg = {0,}; + int ret_val = 0; + u8 intsts = 0; + memcpy_fromio(&intsts, + ((void *)(intelmaddata->int_base)), + sizeof(u8)); + if (intsts != 0) + sst_dbg("intstatus = 0x%x\n", intsts); + switch (intelmaddata->sstdrv_ops->vendor_id) { + case SND_FS: + if (intsts & 0x1) { + /*send long push*/ + ret_val = snd_intelmad_generate_netlink(1, + AUDIO_EVENT_LONG_PRESS); + } + if (intsts & 0x2) { + /*send short push*/ + ret_val = snd_intelmad_generate_netlink(1, + AUDIO_EVENT_SHORT_PRESS); + } + if (intsts & 0x4) { + /*send headphone detect*/ + ret_val = snd_intelmad_generate_netlink(1, + AUDIO_EVENT_HP_DETECT); + } + if (intsts & 0x8) { + /*send headset detect*/ + ret_val = snd_intelmad_generate_netlink(1, + AUDIO_EVENT_HS_DETECT); + } + break; + case SND_MX: + if (intsts & 0x2) { + pmic_reg.reg_addr = 0x201; + ret_val = sst_sc_reg_access(&pmic_reg, PMIC_READ, 1); + if (0 != ret_val) { + /*pmic communication fails*/ + sst_err("pmic commn failed \n"); + return IRQ_HANDLED; + } + intsts = pmic_reg.value; + if (intsts & 0xc0) + /*send jack detect/undetect*/ + sst_dbg("to handle jack sensing\n"); + } + break; + case SND_NC: + if (intsts & 0x1) { + /*send headset detect/undetect*/ + ret_val = snd_intelmad_generate_netlink(1, + AUDIO_EVENT_HS_DETECT); + } + if (intsts & 0x2) { + /*send headphone detect/undetect*/ + ret_val = snd_intelmad_generate_netlink(1, + AUDIO_EVENT_HP_DETECT); + } + if (intsts & 0x4) { + /*send short push*/ + ret_val = snd_intelmad_generate_netlink(1, + AUDIO_EVENT_SHORT_PRESS); + } + if (intsts & 0x8) { + /*send long push*/ + ret_val = snd_intelmad_generate_netlink(1, + AUDIO_EVENT_LONG_PRESS); + } + break; + } + return IRQ_HANDLED; +} +static int __devinit snd_intelmad_register_irq( + struct snd_intelmad *intelmaddata) +{ + int ret_val = 0; + u32 regbase = AUDINT_BASE, regsize = 8; + + /* interpret irq field */ + sst_dbg("irq = 0x%x\n", intelmaddata->irq); + + if ((intelmaddata->irq) & PMIC_SOUND_IRQ_TYPE_MASK) { + /* irq -> GPE_ID */ + intelmaddata->irq = intelmaddata->irq & + (~PMIC_SOUND_IRQ_TYPE_MASK); + sst_dbg("gpe = 0x%x\n", intelmaddata->irq); + ret_val = request_gpe(intelmaddata->irq, + (gpio_function_t)snd_intelmad_intr_handler, + intelmaddata, 0); + if (ret_val) { + sst_dbg("cannot register gpe\n"); + return ret_val; + } + + } else { + /* irq -> system irq */ + ret_val = request_irq(intelmaddata->irq, + snd_intelmad_intr_handler, + IRQF_SHARED, DRIVER_NAME, + intelmaddata); + if (0 != ret_val) { + sst_dbg("cannot register IRQ\n"); + return ret_val; + } + } + + intelmaddata->int_base = ioremap_nocache(regbase, regsize); + return ret_val; +} +static int __devinit snd_intelmad_register_netlink(void) +{ + int ret_val = 0; + + ret_val = genl_register_family(&audio_event_genl_family); + if (0 != ret_val) { + sst_dbg("netlink registration failed\n"); + return ret_val; + } + ret_val = genl_register_mc_group(&audio_event_genl_family, + &audio_event_mcgrp); + if (ret_val) { + sst_dbg("netlink group registration failed\n"); + genl_unregister_family(&audio_event_genl_family); + return ret_val; + } + return ret_val; +} +#endif + +static int __devinit snd_intelmad_sst_register( + struct snd_intelmad *intelmaddata) +{ + int ret_val = 0; + struct sc_reg_access pmic_reg = {0,}; + + pmic_reg.reg_addr = 0; + ret_val = sst_sc_reg_access(&pmic_reg, PMIC_READ, 1); + + if (0 != ret_val) + return ret_val; + + vendor_id = pmic_reg.value & (MASK2|MASK1|MASK0); + sst_info("orginal reg n extrated vendor id = 0x%x %d\n", + pmic_reg.value, vendor_id); + if (vendor_id < 0 || vendor_id > 2) { + sst_err("vendor card not supported!!\n"); + return -EIO; + } + intelmaddata->sstdrv_ops->module_name = SST_CARD_NAMES; + intelmaddata->sstdrv_ops->vendor_id = vendor_id; + intelmaddata->sstdrv_ops->scard_ops = &v[vendor_id]; + + /*registering with SST driver to get access to SST APIs to use*/ + ret_val = register_sst_card(intelmaddata->sstdrv_ops); + if (0 != ret_val) { + sst_err("sst card registration failed\n"); + return ret_val; + } + + vendor_id = intelmaddata->sstdrv_ops->vendor_id; + intelmaddata->pmic_status = PMIC_UNINIT; + return ret_val; +} + +/*Driver Init/exit functionalities*/ +/** +* snd_intelmad_pcm- to setup pcm for the card +* @card: pointer to the sound card structure +*@intelmaddata: pointer to internal context +* This function is called from probe function to set up pcm params and functions +*/ +static int __devinit snd_intelmad_pcm(struct snd_card *card, + struct snd_intelmad *intelmaddata) +{ + struct snd_pcm *pcm; + int i, ret_val = 0; + char name[32] = INTEL_MAD; + + WARN_ON(!card); + WARN_ON(!intelmaddata); + + for (i = 0; i < MAX_DEVICES; i++) { + ret_val = snd_pcm_new(card, name, i, PLAYBACK_COUNT, + CAPTURE_COUNT, &pcm); + if (0 != ret_val) + break; + /*setup the ops for playback and capture streams*/ + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK, + &snd_intelmad_playback_ops); + snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE, + &snd_intelmad_capture_ops); + /*setup private data which can be retrieved when required*/ + pcm->private_data = intelmaddata; + pcm->info_flags = 0; + strncpy(pcm->name, card->shortname, strlen(card->shortname)); + /*allocate dma pages for ALSA stream operations*/ + snd_pcm_lib_preallocate_pages_for_all(pcm, + SNDRV_DMA_TYPE_CONTINUOUS, + snd_dma_continuous_data(GFP_KERNEL), + MIN_BUFFER, MAX_BUFFER); + } + return ret_val; +} + +/** +* snd_intelmad_mixer- to setup mixer settings of the card +*@intelmaddata: pointer to internal context +* This function is called from probe function to set up mixer controls +*/ +static int __devinit snd_intelmad_mixer(struct snd_intelmad *intelmaddata) +{ + struct snd_card *card = NULL; + unsigned int idx; + int ret_val = 0; + char *mixername = "IntelMAD Controls"; + + WARN_ON(!intelmaddata); + + card = intelmaddata->card; + + strncpy(card->mixername, mixername, strlen(mixername)); + /*add all widget controls and expose the same*/ + for (idx = 0; idx < ARRAY_SIZE(snd_intelmad_controls); idx++) { + ret_val = snd_ctl_add(card, + snd_ctl_new1(&snd_intelmad_controls[idx], + intelmaddata)); + sst_dbg("mixer[idx]=%d added \n", idx); + if (0 != ret_val) + break; + } + return ret_val; +} + +/** +* snd_intelmad_dev_free- to free device +*@device: pointer to the device +* This function is called when driver module is removed +*/ +static int snd_intelmad_dev_free(struct snd_device *device) +{ + struct snd_intelmad *intelmaddata; + + WARN_ON(!device); + + intelmaddata = device->device_data; + + sst_dbg("called\n"); + snd_card_free(intelmaddata->card); + genl_unregister_family(&audio_event_genl_family); + unregister_sst_card(intelmaddata->sstdrv_ops); + + /*free allocated memory for internal context*/ + kfree(intelmaddata->sstdrv_ops); + kfree(intelmaddata); + return 0; +} + +/** +* snd_intelmad_create- called from probe to create a snd device +*@intelmaddata : pointer to the internal context +*@card : pointer to the sound card +* This function is called when driver module is started +*/ +static int __devinit snd_intelmad_create( + struct snd_intelmad *intelmaddata, + struct snd_card *card) +{ + int ret_val = 0; + static struct snd_device_ops ops = { + .dev_free = snd_intelmad_dev_free, + }; + + WARN_ON(!intelmaddata); + WARN_ON(!card); + /*ALSA api to register for the device*/ + ret_val = snd_device_new(card, SNDRV_DEV_LOWLEVEL, intelmaddata, &ops); + return ret_val; +} + +/********************************************************************* + * SPI Functions + *********************************************************************/ + + +/** +* snd_intelmad_probe- function registred for init +*@spi : pointer to the spi device context +* This function is called when the device is initialized +*/ +int __devinit snd_intelmad_probe(struct spi_device *spi) +{ + struct snd_card *card = NULL; + int ret_val = 0; + struct snd_intelmad *intelmaddata = NULL; + + sst_dbg("called \n"); + + /*allocate memory for saving internal context and working*/ + intelmaddata = kzalloc(sizeof(*intelmaddata), GFP_KERNEL); + if (NULL == intelmaddata) + return -ENOMEM; + + /*allocate memory for LPE API set*/ + intelmaddata->sstdrv_ops = kzalloc(sizeof(struct intel_sst_card_ops), + GFP_KERNEL); + if (NULL == intelmaddata->sstdrv_ops) { + sst_err("mem alloctn fail \n"); + kfree(intelmaddata); + return -ENOMEM; + } + + /*create a card instance with ALSA framework*/ + card = snd_card_new(card_index, card_id, THIS_MODULE, 0); + if (NULL == card) + return -ENOMEM; + + intelmaddata->spi = spi; + intelmaddata->irq = spi->irq; + dev_set_drvdata(&spi->dev, intelmaddata); + intelmaddata->card = card; + intelmaddata->card_id = card_id; + intelmaddata->card_index = card_index; + strncpy(card->driver, INTEL_MAD, strlen(INTEL_MAD)); + strncpy(card->shortname, INTEL_MAD, strlen(INTEL_MAD)); + +#ifdef REG_IRQ + ret_val = snd_intelmad_register_irq(intelmaddata); + if (0 != ret_val) { + sst_err("snd_intelmad_register_irq fail\n"); + goto free_allocs; + } + ret_val = snd_intelmad_register_netlink(); + if (0 == ret_val) { + sst_dbg("...complete\n"); + return ret_val; + } +#endif + + /*internal function call to register device with ALSA*/ + ret_val = snd_intelmad_create(intelmaddata, card); + if (0 != ret_val) { + sst_err("snd_intelmad_create failed\n"); + goto free_allocs; + } + + ret_val = snd_intelmad_pcm(card, intelmaddata); + if (0 != ret_val) { + sst_err("snd_intelmad_pcm failed\n"); + goto free_allocs; + } + + ret_val = snd_intelmad_mixer(intelmaddata); + if (0 != ret_val) { + sst_err("snd_intelmad_mixer failed\n"); + goto free_allocs; + } + + card->private_data = &intelmaddata; + snd_card_set_dev(card, &spi->dev); + ret_val = snd_card_register(card); + if (0 != ret_val) + goto free_allocs; + + + intelmaddata->sstdrv_ops->module_name = SST_CARD_NAMES; + /*registering with LPE driver to get access to LPE APIsito use*/ + ret_val = snd_intelmad_sst_register(intelmaddata); + if (0 == ret_val) { + ret_val = intelmaddata->sstdrv_ops->scard_ops->init_card(); + intelmaddata->pmic_status = PMIC_INIT; + sst_dbg("...complete\n"); + return ret_val; + } + + sst_err("registering with LPE failed \n"); + +free_allocs: + /*TODO: unregister IRQ*/ + sst_err("probe failed\n"); + /*snd_card_free(card);*/ + kfree(intelmaddata->sstdrv_ops); + kfree(intelmaddata); + return ret_val; +} + + +/** +* snd_intelmad_remove- function registred for exit +*@spi : pointer to the spi device context +* This function is called when the device is uninitialized +*/ +static int snd_intelmad_remove(struct spi_device *spi) +{ + struct snd_intelmad *intelmaddata = + dev_get_drvdata(&spi->dev); + /* + * TODO:: de-register interrupt handler + */ + + if (intelmaddata) { + snd_card_free(intelmaddata->card); + genl_unregister_family(&audio_event_genl_family); + unregister_sst_card(intelmaddata->sstdrv_ops); + /*free allocated memory for internal context*/ + kfree(intelmaddata->sstdrv_ops); + kfree(intelmaddata); + } + return 0; +} + +/********************************************************************* + * Driver initialization and exit + *********************************************************************/ + +static struct spi_driver snd_intelmad_driver = { + .driver = { + .name = DRIVER_NAME, + .bus = &spi_bus_type, + .owner = THIS_MODULE, + }, + .probe = snd_intelmad_probe, + .remove = __devexit_p(snd_intelmad_remove), +}; + +/* +* alsa_card_intelmad_init- driver init function +* This function is called when driver module is inserted +*/ +static int __init alsa_card_intelmad_init(void) +{ + sst_dbg("called\n"); + return spi_register_driver(&snd_intelmad_driver); +} + +/** +* alsa_card_intelmad_exit- driver exit function +* This function is called when driver module is removed +*/ +static void __exit alsa_card_intelmad_exit(void) +{ + sst_dbg("called\n"); + spi_unregister_driver(&snd_intelmad_driver); +} + +module_init(alsa_card_intelmad_init) +module_exit(alsa_card_intelmad_exit) + -- 1.5.4.5

6 31

[alsa-devel] [PATCH 1/1] pcm_hw: Always use delay ioctl in snd_pcm_delay()
by Kai Vehmanen 11 Sep '09

11 Sep '09

As the result of snd_pcm_delay() is affected not only by hw_ptr and appl_ptr, but also by 'runtime->delay' property, either SNDRV_PCM_IOCTL_DELAY or SNDRV_PCM_IOCTL_STATUS ioctl must be used to get the correct result. Previously 'runtime->delay' was ignored in case 'hw->sync_ptr' was used. Signed-off-by: Kai Vehmanen <kvehmanen(a)eca.cx> --- src/pcm/pcm_hw.c | 22 ---------------------- 1 files changed, 0 insertions(+), 22 deletions(-) diff --git a/src/pcm/pcm_hw.c b/src/pcm/pcm_hw.c index c46d14f..8abb204 100644 --- a/src/pcm/pcm_hw.c +++ b/src/pcm/pcm_hw.c @@ -507,28 +507,6 @@ static int snd_pcm_hw_delay(snd_pcm_t *pcm, snd_pcm_sframes_t *delayp) { snd_pcm_hw_t *hw = pcm->private_data; int fd = hw->fd, err; - if (hw->sync_ptr) { - err = sync_ptr1(hw, SNDRV_PCM_SYNC_PTR_HWSYNC); - if (err < 0) - return err; - switch (FAST_PCM_STATE(hw)) { - case SNDRV_PCM_STATE_RUNNING: - case SNDRV_PCM_STATE_DRAINING: - case SNDRV_PCM_STATE_PAUSED: - case SNDRV_PCM_STATE_PREPARED: - case SNDRV_PCM_STATE_SUSPENDED: - break; - case SNDRV_PCM_STATE_XRUN: - return -EPIPE; - default: - return -EBADFD; - } - if (pcm->stream == SND_PCM_STREAM_PLAYBACK) - *delayp = snd_pcm_mmap_playback_hw_avail(pcm); - else - *delayp = snd_pcm_mmap_capture_avail(pcm); - return 0; - } if (ioctl(fd, SNDRV_PCM_IOCTL_DELAY, delayp) < 0) { err = -errno; SYSMSG("SNDRV_PCM_IOCTL_DELAY failed"); -- 1.5.6.5

1 0

[alsa-devel] How to support mutiple audio interface in alsa
by Feng Ye 10 Sep '09

10 Sep '09

Hello there, This question is for ALSA, not ALSA SOC. I am doing an ALSA driver for our chip (soc chip with cpu inside), inside it has I2S and DAC. The I2S can be feed to DAC, or come out of the chip to an external CODEC chip. I wonder how should I model this in the driver, are they multiple ALSA cards, or devices, or subdevices? Since the I2S is shared, I think the most appropriate is 'subdevice', but I do not see how this is used in linux/sound, this is only used in pci, and seems to tied together with substream number. If the answer is 'device', is there some existing driver that are doing this? Thanks, Feng

2 1

[alsa-devel] i.MX31L + AC97
by Geoffrey Wossum 10 Sep '09

10 Sep '09

Hi all, I'm currently looking at a board which uses a Freescale i.MX31L connected to a Wolfson WM9715L over the AC97 bus. The WM9715L will be used for both sound and a touchscreen. It doesn't look like there are any AC97 SoC drivers for the i.MX31. Is this correct? If so, should I try to get the board changed to use a codec on the I2S bus (the board is being layed out right now), or would creating the AC97 driver simple enough that we should continue on with this hardware and write the driver when needed? TIA, --- Geoffrey

2 1

[alsa-devel] Alsa plugin unable to be loaded by Pulseaudio
by Ng Oon-Ee 10 Sep '09

10 Sep '09

Hello, please find below an issue I had with loading plugins into pulseaudio. The pulse dev has sent this mail on his list concerning alsa's interaction with pulse, indicating that there is something buggy with how it works. Details below. I'd be willing to post something to the bugtracker, if I only knew what the below means... -------- Forwarded Message -------- From: Lennart Poettering <lennart(a)poettering.net> Reply-to: General PulseAudio Discussion <pulseaudio-discuss(a)mail.0pointer.de> To: pulseaudio-discuss(a)mail.0pointer.de Subject: Re: [pulseaudio-discuss] Alsa output devices, able to use alsa plugin? Date: Fri, 4 Sep 2009 01:20:15 +0200 On Wed, 26.08.09 12:27, Tanu Kaskinen (tanuk(a)iki.fi) wrote: > D: alsa-util.c: Trying test with SND_PCM_NO_AUTO_FORMAT ... > D: alsa-util.c: Managed to open test > D: alsa-util.c: Maximum hw buffer size is 371 ms > I: alsa-util.c: snd_pcm_hw_params_set_periods_near() failed: Tiedostoa tai hakemistoa ei ole > I: alsa-util.c: snd_pcm_hw_params_set_buffer_size_near() failed: Tiedostoa tai hakemistoa ei ole > D: alsa-util.c: Trying test without SND_PCM_NO_AUTO_FORMAT ... > D: alsa-util.c: Managed to open test > D: alsa-util.c: Maximum hw buffer size is 371 ms > I: alsa-util.c: snd_pcm_hw_params_set_periods_near() failed: Tiedostoa tai hakemistoa ei ole > I: alsa-util.c: snd_pcm_hw_params_set_buffer_size_near() failed: Tiedostoa tai hakemistoa ei ole > D: alsa-util.c: Trying plug:test with SND_PCM_NO_AUTO_FORMAT ... > D: alsa-util.c: Managed to open plug:test > D: alsa-util.c: Maximum hw buffer size is 371 ms > I: alsa-util.c: snd_pcm_hw_params_set_periods_near() failed: Tiedostoa tai hakemistoa ei ole > I: alsa-util.c: snd_pcm_hw_params_set_buffer_size_near() failed: Tiedostoa tai hakemistoa ei ole > D: alsa-util.c: Trying plug:test without SND_PCM_NO_AUTO_FORMAT ... > D: alsa-util.c: Managed to open plug:test > D: alsa-util.c: Maximum hw buffer size is 371 ms > I: alsa-util.c: snd_pcm_hw_params_set_periods_near() failed: Tiedostoa tai hakemistoa ei ole > I: alsa-util.c: snd_pcm_hw_params_set_buffer_size_near() failed: Tiedostoa tai hakemistoa ei ole > I: alsa-util.c: Failed to set hardware parameters on plug:test: Tiedostoa tai hakemistoa ei ole > E: module.c: Failed to load module "module-alsa-sink" (argument: "device=test"): initialization failed. > > That "Tiedostoa tai hakemistoa ei ole" error means "No such file or > directory". I tried also with device=hw:0,0 and device=plughw:0,0 and > the first worked while the latter gave the same errors. Since the failed > function calls pertain to buffering, I also tried with tsched=0. It > worked. So, maybe this is a bug in the "plug" alsa plugin, ie. it > doesn't implement the *_near() functions properly. > > Lennart, any insights? I am pretty sure that ALSA is broken here. We nowadays first try to set the number of periods, and then set the buffer size in the hwparams struct. If this fails we print a warning message (the ones you see above with all the vowels regarding snd_pcm_hw_params_set_periods_near() resp. snd_pcm_hw_params_set_buffer_size_near()), but we do not consider this fatal. After doing this we call snd_pcm_hw_params() to activate our settings -- and if this fails then we consider this a fatal problem. Now the brokeness in ALSA is that after refusing our parameters it still fails to configure our settings, i.e. calling the two functions that fail breaks the hwparams setup: if we wouldn't call them, no breakage happens at all. The two functions in question should either succeed or fail, but if they fail they should not modify hwparams in a way that it subsequently becomes unusable for snd_pcm_hw_params(). This issue needs to be fixed in ALSA. Lennart

3 9

[alsa-devel] Problems with safe API and snd-cs46xx
by Sophie Hamilton 10 Sep '09

10 Sep '09

Hi there, I was pointed here by Audacious developers, who believe I've found a bug in the snd-cs46xx driver, which I use for my sound card (a Turtle Beach Santa Cruz, whose PCI ID is 1013:6003). Audacious, as of version 2.1, uses the default period time offered by ALSA and limits itself to the "safe API" as per Lennart Poettering. This already uncovered a bug in the Aureal Vortex driver, and I believe, based on the information that the devs gave me, that I have just uncovered another. (I'm also having problems with recent versions of OpenAL. However, I'll explain this further below.) I first started having the problem after having upgraded to Audacious 2.1, from Audacious 1.5.1. (to be more precise, the Gentoo ebuild 1.5.1-r1.) At the time, I was using the Gentoo-patched 2.6.29-series kernel, which I use as my normal kernel. After upgrading, I could no longer play audio directly using ALSA; instead, Audacious would initialise to 00:00, not play anything, and report the following to the owning console every so often: > ERROR: ALSA: alsa-core.c:226 (alsaplug_write_buffer): (write) snd_pcm_recover: Input/output error The thread which was doing this also seemed to be blocking, as when I tried to do anything else in Audacious, the interface would then freeze until the thread was done. OSS output seems to be okay, although as I don't have OSS compiled at all into the kernel (not even as a module), this would be going via ALSA's own OSS emulation. In addition, ALSA output to another audio device from Audacious works fine. I Googled for this and came across another snd-cs46xx user having the exact same problem in bug AUD-53 in the JIRA implementation they were using: http://jira.atheme.org/browse/AUD-53 . Since that user had been asked to upgrade to tip but hadn't, I did so instead (uninstalling the Gentoo ebuilds frst, of course) and reported back the additional debug info: > DEBUG: ALSA: alsa-core.c:226 (alsaplug_write_buffer): snd_pcm_writei error: Input/output error > ERROR: ALSA: alsa-core.c:230 (alsaplug_write_buffer): snd_pcm_recover error: Input/output error I then didn't know how to continue as I wasn't sure whether the bug would be seen as it had previously been closed as unreproducible, so I hopped onto the IRC channel and asked about what should be done. It was there that the probability of it being a driver bug in the snd-cs46xx sound driver was mentioned, and the reasoning - that the use of ALSA's safe API has been exposing driver bugs that were previously hidden. As mentioned previously, I'm also having issues with OpenAL. The version I'm using now is 1.7.411; previously I was using 1.5.304, which I believe worked fine. (I can't easily test this now as the Gentoo ebuild for it has gone, however.) The symptoms I got were similar - I got no sound, and messages were printed to the console every so often (more often than with Audacious, however): > AL lib: alsa.c:194: Wait timeout... buffer size too low? At first I thought this was an issue with Second Life, but on seeing the exact same symptoms using mplayer with the '-ao openal' switch, and also seeing it occur with 'openal-info', it seems to be an OpenAL thing. (Since I don't have many apps that I use with OpenAL, I hadn't known this before.) It seems that this is a symptom of whatever driver bug is causing the Audacious error. For the purposes of making this post, I decided to compile and install a vanilla 2.6.30.5 kernel to boot into, so that I could make sure there were no problems with the latest stable build. (I'd rather not run unstable versions if I can help it, or I would have done so; this is my primary machine. However, I'm willing to apply patches for this specific problem in order to test them.) The results of this were that I no longer get the messages and the threads don't block, but I get entirely new problems now - instead of useable sound, I get what sounds a little like noise, but more crackly, and the seek bar in Audacious seems to be going along as fast as it can - for each wallclock second, the seek bar goes up by about 114 seconds. The same occurs when I use mplayer with the '-ao openal' switch; I get the noise/crackles, and the time position shown on the console zooms by. The description of my sound card given by 'lspci -vv' (under both kernels) is thus: > 05:04.0 Multimedia audio controller: Cirrus Logic CS 4614/22/24/30 [CrystalClear SoundFusion Audio Accelerator] (rev 01) > Subsystem: Voyetra Technologies Santa Cruz > Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- > Status: Cap+ 66MHz- UDF- FastB2B- ParErr- DEVSEL=slow >TAbort- <TAbort- <MAbort- >SERR- <PERR- INTx- > Latency: 64 (1000ns min, 6000ns max) > Interrupt: pin A routed to IRQ 22 > Region 0: Memory at fdafe000 (32-bit, non-prefetchable) [size=4K] > Region 1: Memory at fd900000 (32-bit, non-prefetchable) [size=1M] > Capabilities: [40] Power Management version 2 > Flags: PMEClk- DSI+ D1+ D2+ AuxCurrent=0mA PME(D0-,D1-,D2-,D3hot-,D3cold-) > Status: D0 PME-Enable- DSel=0 DScale=0 PME- > Kernel driver in use: Sound Fusion CS46xx > Kernel modules: snd-cs46xx Most applications that I have seem to work properly with regards to audio, although I've heard that this is probably because ALSA's safe API isn't very well documented and as such most applications will not be using it, thus not exposing the bug in snd-cs46xx. No information is output to dmesg when this happens, on either the Gentoo-patched 2.6.29 kernel, or the vanilla 2.6.30.5 kernel. If you need any further information, please let me know! As stated above, I'm willing to apply patches to the vanilla sources of the latest stable kernel in order to test them. I don't have experience of downloading the ALSA drivers and compiling them separately but I could also do that if it's needed. Thank you, - Sophie.

6 32

Re: [alsa-devel] [PATCH] To avoid the divide by zero error during the first execution, initialize the data type.
by Troy Kisky 10 Sep '09

10 Sep '09

avm(a)ti.com wrote: > From: Arun Mani <a0270733(a)gtcx26221.gt.design.ti.com> > > Signed-off-by: Arun Mani <a0270733(a)gtcx26221.gt.design.ti.com> > --- > sound/soc/davinci/davinci-i2s.c | 2 ++ > 1 files changed, 2 insertions(+), 0 deletions(-) > > diff --git a/sound/soc/davinci/davinci-i2s.c b/sound/soc/davinci/davinci-i2s.c > index b1ea52f..61b1763 100644 > --- a/sound/soc/davinci/davinci-i2s.c > +++ b/sound/soc/davinci/davinci-i2s.c > @@ -104,10 +104,12 @@ enum { > > static struct davinci_pcm_dma_params davinci_i2s_pcm_out = { > .name = "I2S PCM Stereo out", > + .data_type = 2, //Initialize the data type for playback to avoid divide by zero You should have ".acnt = 2" also for consistency. > }; > > static struct davinci_pcm_dma_params davinci_i2s_pcm_in = { > .name = "I2S PCM Stereo in", > + .data_type = 2, //Initialize the data type for playback to avoid divide by zero > }; > > struct davinci_mcbsp_dev { And if the 1st stream is an 8-bit audio stream, will it initialize it incorrectly ??? Of course, we have formats = SNDRV_PCM_FMTBIT_S16_LE, currently so it's not an issue yet... I don't see how data_type is not being set in davinci_i2s_hw_params before being used in davinci_pcm_prepare. Can prepare be called before hw_params ? There is a "return -EINVAL" in davinci_i2s_hw_params with a "printk(KERN_WARNING "davinci-i2s: unsupported PCM format\n")" before it. Do you see this message in your log ? Thanks Troy

2 2

[alsa-devel] [PATCH v3] ASoC: AK4671: add ak4671 codec driver
by Joonyoung Shim 10 Sep '09

10 Sep '09

The AK4671 is a stereo CODEC with a built-in Microphone-Amplifier, Receiver-Amplifier and Headphone-Amplifier. The datasheet for the ak4671 can find at the following url: http://www.asahi-kasei.co.jp/akm/en/product/ak4671/ak4671_f01e.pdf Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- Change log: v2: - use soc-cache - remove unused pll_on variable - add event handler for DAC and ADC - remove unnecessary comments and PM functions - modify error handling in ak4671_register function v3: - use SND_SOC_DAPM_SUPPLY instead of SND_SOC_DAPM_DAC_E and SND_SOC_DAPM_ADC_E sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4671.c | 825 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ak4671.h | 156 +++++++++ 4 files changed, 987 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/ak4671.c create mode 100644 sound/soc/codecs/ak4671.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0edca93..a2bb659 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -19,6 +19,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C select SND_SOC_AK4642 if I2C + select SND_SOC_AK4671 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 @@ -96,6 +97,9 @@ config SND_SOC_AK4535 config SND_SOC_AK4642 tristate +config SND_SOC_AK4671 + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fb4af28..13f7b4f 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -6,6 +6,7 @@ snd-soc-ad73311-objs := ad73311.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o snd-soc-ak4642-objs := ak4642.o +snd-soc-ak4671-objs := ak4671.o snd-soc-cs4270-objs := cs4270.o snd-soc-cx20442-objs := cx20442.o snd-soc-l3-objs := l3.o @@ -56,6 +57,7 @@ obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o +obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c new file mode 100644 index 0000000..b61214d --- /dev/null +++ b/sound/soc/codecs/ak4671.c @@ -0,0 +1,825 @@ +/* + * ak4671.c -- audio driver for AK4671 + * + * Copyright (C) 2009 Samsung Electronics Co.Ltd + * Author: Joonyoung Shim <jy0922.shim(a)samsung.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/delay.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "ak4671.h" + +static struct snd_soc_codec *ak4671_codec; + +/* codec private data */ +struct ak4671_priv { + struct snd_soc_codec codec; + u8 reg_cache[AK4671_CACHEREGNUM]; +}; + +/* ak4671 register cache & default register settings */ +static const u8 ak4671_reg[AK4671_CACHEREGNUM] = { + 0x00, /* AK4671_AD_DA_POWER_MANAGEMENT (0x00) */ + 0xf6, /* AK4671_PLL_MODE_SELECT0 (0x01) */ + 0x00, /* AK4671_PLL_MODE_SELECT1 (0x02) */ + 0x02, /* AK4671_FORMAT_SELECT (0x03) */ + 0x00, /* AK4671_MIC_SIGNAL_SELECT (0x04) */ + 0x55, /* AK4671_MIC_AMP_GAIN (0x05) */ + 0x00, /* AK4671_MIXING_POWER_MANAGEMENT0 (0x06) */ + 0x00, /* AK4671_MIXING_POWER_MANAGEMENT1 (0x07) */ + 0xb5, /* AK4671_OUTPUT_VOLUME_CONTROL (0x08) */ + 0x00, /* AK4671_LOUT1_SIGNAL_SELECT (0x09) */ + 0x00, /* AK4671_ROUT1_SIGNAL_SELECT (0x0a) */ + 0x00, /* AK4671_LOUT2_SIGNAL_SELECT (0x0b) */ + 0x00, /* AK4671_ROUT2_SIGNAL_SELECT (0x0c) */ + 0x00, /* AK4671_LOUT3_SIGNAL_SELECT (0x0d) */ + 0x00, /* AK4671_ROUT3_SIGNAL_SELECT (0x0e) */ + 0x00, /* AK4671_LOUT1_POWER_MANAGERMENT (0x0f) */ + 0x00, /* AK4671_LOUT2_POWER_MANAGERMENT (0x10) */ + 0x80, /* AK4671_LOUT3_POWER_MANAGERMENT (0x11) */ + 0x91, /* AK4671_LCH_INPUT_VOLUME_CONTROL (0x12) */ + 0x91, /* AK4671_RCH_INPUT_VOLUME_CONTROL (0x13) */ + 0xe1, /* AK4671_ALC_REFERENCE_SELECT (0x14) */ + 0x00, /* AK4671_DIGITAL_MIXING_CONTROL (0x15) */ + 0x00, /* AK4671_ALC_TIMER_SELECT (0x16) */ + 0x00, /* AK4671_ALC_MODE_CONTROL (0x17) */ + 0x02, /* AK4671_MODE_CONTROL1 (0x18) */ + 0x01, /* AK4671_MODE_CONTROL2 (0x19) */ + 0x18, /* AK4671_LCH_OUTPUT_VOLUME_CONTROL (0x1a) */ + 0x18, /* AK4671_RCH_OUTPUT_VOLUME_CONTROL (0x1b) */ + 0x00, /* AK4671_SIDETONE_A_CONTROL (0x1c) */ + 0x02, /* AK4671_DIGITAL_FILTER_SELECT (0x1d) */ + 0x00, /* AK4671_FIL3_COEFFICIENT0 (0x1e) */ + 0x00, /* AK4671_FIL3_COEFFICIENT1 (0x1f) */ + 0x00, /* AK4671_FIL3_COEFFICIENT2 (0x20) */ + 0x00, /* AK4671_FIL3_COEFFICIENT3 (0x21) */ + 0x00, /* AK4671_EQ_COEFFICIENT0 (0x22) */ + 0x00, /* AK4671_EQ_COEFFICIENT1 (0x23) */ + 0x00, /* AK4671_EQ_COEFFICIENT2 (0x24) */ + 0x00, /* AK4671_EQ_COEFFICIENT3 (0x25) */ + 0x00, /* AK4671_EQ_COEFFICIENT4 (0x26) */ + 0x00, /* AK4671_EQ_COEFFICIENT5 (0x27) */ + 0xa9, /* AK4671_FIL1_COEFFICIENT0 (0x28) */ + 0x1f, /* AK4671_FIL1_COEFFICIENT1 (0x29) */ + 0xad, /* AK4671_FIL1_COEFFICIENT2 (0x2a) */ + 0x20, /* AK4671_FIL1_COEFFICIENT3 (0x2b) */ + 0x00, /* AK4671_FIL2_COEFFICIENT0 (0x2c) */ + 0x00, /* AK4671_FIL2_COEFFICIENT1 (0x2d) */ + 0x00, /* AK4671_FIL2_COEFFICIENT2 (0x2e) */ + 0x00, /* AK4671_FIL2_COEFFICIENT3 (0x2f) */ + 0x00, /* AK4671_DIGITAL_FILTER_SELECT2 (0x30) */ + 0x00, /* this register not used */ + 0x00, /* AK4671_E1_COEFFICIENT0 (0x32) */ + 0x00, /* AK4671_E1_COEFFICIENT1 (0x33) */ + 0x00, /* AK4671_E1_COEFFICIENT2 (0x34) */ + 0x00, /* AK4671_E1_COEFFICIENT3 (0x35) */ + 0x00, /* AK4671_E1_COEFFICIENT4 (0x36) */ + 0x00, /* AK4671_E1_COEFFICIENT5 (0x37) */ + 0x00, /* AK4671_E2_COEFFICIENT0 (0x38) */ + 0x00, /* AK4671_E2_COEFFICIENT1 (0x39) */ + 0x00, /* AK4671_E2_COEFFICIENT2 (0x3a) */ + 0x00, /* AK4671_E2_COEFFICIENT3 (0x3b) */ + 0x00, /* AK4671_E2_COEFFICIENT4 (0x3c) */ + 0x00, /* AK4671_E2_COEFFICIENT5 (0x3d) */ + 0x00, /* AK4671_E3_COEFFICIENT0 (0x3e) */ + 0x00, /* AK4671_E3_COEFFICIENT1 (0x3f) */ + 0x00, /* AK4671_E3_COEFFICIENT2 (0x40) */ + 0x00, /* AK4671_E3_COEFFICIENT3 (0x41) */ + 0x00, /* AK4671_E3_COEFFICIENT4 (0x42) */ + 0x00, /* AK4671_E3_COEFFICIENT5 (0x43) */ + 0x00, /* AK4671_E4_COEFFICIENT0 (0x44) */ + 0x00, /* AK4671_E4_COEFFICIENT1 (0x45) */ + 0x00, /* AK4671_E4_COEFFICIENT2 (0x46) */ + 0x00, /* AK4671_E4_COEFFICIENT3 (0x47) */ + 0x00, /* AK4671_E4_COEFFICIENT4 (0x48) */ + 0x00, /* AK4671_E4_COEFFICIENT5 (0x49) */ + 0x00, /* AK4671_E5_COEFFICIENT0 (0x4a) */ + 0x00, /* AK4671_E5_COEFFICIENT1 (0x4b) */ + 0x00, /* AK4671_E5_COEFFICIENT2 (0x4c) */ + 0x00, /* AK4671_E5_COEFFICIENT3 (0x4d) */ + 0x00, /* AK4671_E5_COEFFICIENT4 (0x4e) */ + 0x00, /* AK4671_E5_COEFFICIENT5 (0x4f) */ + 0x88, /* AK4671_EQ_CONTROL_250HZ_100HZ (0x50) */ + 0x88, /* AK4671_EQ_CONTROL_3500HZ_1KHZ (0x51) */ + 0x08, /* AK4671_EQ_CONTRO_10KHZ (0x52) */ + 0x00, /* AK4671_PCM_IF_CONTROL0 (0x53) */ + 0x00, /* AK4671_PCM_IF_CONTROL1 (0x54) */ + 0x00, /* AK4671_PCM_IF_CONTROL2 (0x55) */ + 0x18, /* AK4671_DIGITAL_VOLUME_B_CONTROL (0x56) */ + 0x18, /* AK4671_DIGITAL_VOLUME_C_CONTROL (0x57) */ + 0x00, /* AK4671_SIDETONE_VOLUME_CONTROL (0x58) */ + 0x00, /* AK4671_DIGITAL_MIXING_CONTROL2 (0x59) */ + 0x00, /* AK4671_SAR_ADC_CONTROL (0x5a) */ +}; + +/* + * LOUT1/ROUT1 output volume control: + * from -24 to 6 dB in 6 dB steps (mute instead of -30 dB) + */ +static DECLARE_TLV_DB_SCALE(out1_tlv, -3000, 600, 1); + +/* + * LOUT2/ROUT2 output volume control: + * from -33 to 6 dB in 3 dB steps (mute instead of -33 dB) + */ +static DECLARE_TLV_DB_SCALE(out2_tlv, -3300, 300, 1); + +/* + * LOUT3/ROUT3 output volume control: + * from -6 to 3 dB in 3 dB steps + */ +static DECLARE_TLV_DB_SCALE(out3_tlv, -600, 300, 0); + +/* + * Mic amp gain control: + * from -15 to 30 dB in 3 dB steps + * REVISIT: The actual min value(0x01) is -12 dB and the reg value 0x00 is not + * available + */ +static DECLARE_TLV_DB_SCALE(mic_amp_tlv, -1500, 300, 0); + +static const struct snd_kcontrol_new ak4671_snd_controls[] = { + /* Common playback gain controls */ + SOC_SINGLE_TLV("Line Output1 Playback Volume", + AK4671_OUTPUT_VOLUME_CONTROL, 0, 0x6, 0, out1_tlv), + SOC_SINGLE_TLV("Headphone Output2 Playback Volume", + AK4671_OUTPUT_VOLUME_CONTROL, 4, 0xd, 0, out2_tlv), + SOC_SINGLE_TLV("Line Output3 Playback Volume", + AK4671_LOUT3_POWER_MANAGERMENT, 6, 0x3, 0, out3_tlv), + + /* Common capture gain controls */ + SOC_DOUBLE_TLV("Mic Amp Capture Volume", + AK4671_MIC_AMP_GAIN, 0, 4, 0xf, 0, mic_amp_tlv), +}; + +/* event handlers */ +static int ak4671_out2_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + u8 reg; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + reg = snd_soc_read(codec, AK4671_LOUT2_POWER_MANAGERMENT); + reg |= AK4671_MUTEN; + snd_soc_write(codec, AK4671_LOUT2_POWER_MANAGERMENT, reg); + break; + case SND_SOC_DAPM_PRE_PMD: + reg = snd_soc_read(codec, AK4671_LOUT2_POWER_MANAGERMENT); + reg &= ~AK4671_MUTEN; + snd_soc_write(codec, AK4671_LOUT2_POWER_MANAGERMENT, reg); + break; + } + + return 0; +} + +/* Output Mixers */ +static const struct snd_kcontrol_new ak4671_lout1_mixer_controls[] = { + SOC_DAPM_SINGLE("DACL", AK4671_LOUT1_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("LINL1", AK4671_LOUT1_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("LINL2", AK4671_LOUT1_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("LINL3", AK4671_LOUT1_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("LINL4", AK4671_LOUT1_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPL", AK4671_LOUT1_SIGNAL_SELECT, 5, 1, 0), +}; + +static const struct snd_kcontrol_new ak4671_rout1_mixer_controls[] = { + SOC_DAPM_SINGLE("DACR", AK4671_ROUT1_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("RINR1", AK4671_ROUT1_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("RINR2", AK4671_ROUT1_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("RINR3", AK4671_ROUT1_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("RINR4", AK4671_ROUT1_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPR", AK4671_ROUT1_SIGNAL_SELECT, 5, 1, 0), +}; + +static const struct snd_kcontrol_new ak4671_lout2_mixer_controls[] = { + SOC_DAPM_SINGLE("DACHL", AK4671_LOUT2_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("LINH1", AK4671_LOUT2_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("LINH2", AK4671_LOUT2_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("LINH3", AK4671_LOUT2_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("LINH4", AK4671_LOUT2_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPHL", AK4671_LOUT2_SIGNAL_SELECT, 5, 1, 0), +}; + +static const struct snd_kcontrol_new ak4671_rout2_mixer_controls[] = { + SOC_DAPM_SINGLE("DACHR", AK4671_ROUT2_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("RINH1", AK4671_ROUT2_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("RINH2", AK4671_ROUT2_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("RINH3", AK4671_ROUT2_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("RINH4", AK4671_ROUT2_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPHR", AK4671_ROUT2_SIGNAL_SELECT, 5, 1, 0), +}; + +static const struct snd_kcontrol_new ak4671_lout3_mixer_controls[] = { + SOC_DAPM_SINGLE("DACSL", AK4671_LOUT3_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("LINS1", AK4671_LOUT3_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("LINS2", AK4671_LOUT3_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("LINS3", AK4671_LOUT3_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("LINS4", AK4671_LOUT3_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPSL", AK4671_LOUT3_SIGNAL_SELECT, 5, 1, 0), +}; + +static const struct snd_kcontrol_new ak4671_rout3_mixer_controls[] = { + SOC_DAPM_SINGLE("DACSR", AK4671_ROUT3_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("RINS1", AK4671_ROUT3_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("RINS2", AK4671_ROUT3_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("RINS3", AK4671_ROUT3_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("RINS4", AK4671_ROUT3_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPSR", AK4671_ROUT3_SIGNAL_SELECT, 5, 1, 0), +}; + +/* Input MUXs */ +static const char *ak4671_lin_mux_texts[] = + {"LIN1", "LIN2", "LIN3", "LIN4"}; +static const struct soc_enum ak4671_lin_mux_enum = + SOC_ENUM_SINGLE(AK4671_MIC_SIGNAL_SELECT, 0, + ARRAY_SIZE(ak4671_lin_mux_texts), + ak4671_lin_mux_texts); +static const struct snd_kcontrol_new ak4671_lin_mux_control = + SOC_DAPM_ENUM("Route", ak4671_lin_mux_enum); + +static const char *ak4671_rin_mux_texts[] = + {"RIN1", "RIN2", "RIN3", "RIN4"}; +static const struct soc_enum ak4671_rin_mux_enum = + SOC_ENUM_SINGLE(AK4671_MIC_SIGNAL_SELECT, 2, + ARRAY_SIZE(ak4671_rin_mux_texts), + ak4671_rin_mux_texts); +static const struct snd_kcontrol_new ak4671_rin_mux_control = + SOC_DAPM_ENUM("Route", ak4671_rin_mux_enum); + +static const struct snd_soc_dapm_widget ak4671_dapm_widgets[] = { + /* Inputs */ + SND_SOC_DAPM_INPUT("LIN1"), + SND_SOC_DAPM_INPUT("RIN1"), + SND_SOC_DAPM_INPUT("LIN2"), + SND_SOC_DAPM_INPUT("RIN2"), + SND_SOC_DAPM_INPUT("LIN3"), + SND_SOC_DAPM_INPUT("RIN3"), + SND_SOC_DAPM_INPUT("LIN4"), + SND_SOC_DAPM_INPUT("RIN4"), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("LOUT1"), + SND_SOC_DAPM_OUTPUT("ROUT1"), + SND_SOC_DAPM_OUTPUT("LOUT2"), + SND_SOC_DAPM_OUTPUT("ROUT2"), + SND_SOC_DAPM_OUTPUT("LOUT3"), + SND_SOC_DAPM_OUTPUT("ROUT3"), + + /* DAC */ + SND_SOC_DAPM_DAC("DAC Left", "Left HiFi Playback", + AK4671_AD_DA_POWER_MANAGEMENT, 6, 0), + SND_SOC_DAPM_DAC("DAC Right", "Right HiFi Playback", + AK4671_AD_DA_POWER_MANAGEMENT, 7, 0), + + /* ADC */ + SND_SOC_DAPM_ADC("ADC Left", "Left HiFi Capture", + AK4671_AD_DA_POWER_MANAGEMENT, 4, 0), + SND_SOC_DAPM_ADC("ADC Right", "Right HiFi Capture", + AK4671_AD_DA_POWER_MANAGEMENT, 5, 0), + + /* PGA */ + SND_SOC_DAPM_PGA("LOUT2 Mix Amp", + AK4671_LOUT2_POWER_MANAGERMENT, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("ROUT2 Mix Amp", + AK4671_LOUT2_POWER_MANAGERMENT, 6, 0, NULL, 0), + + SND_SOC_DAPM_PGA("LIN1 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("RIN1 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("LIN2 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("RIN2 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("LIN3 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("RIN3 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("LIN4 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("RIN4 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 7, 0, NULL, 0), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("LOUT1 Mixer", AK4671_LOUT1_POWER_MANAGERMENT, 0, 0, + &ak4671_lout1_mixer_controls[0], + ARRAY_SIZE(ak4671_lout1_mixer_controls)), + SND_SOC_DAPM_MIXER("ROUT1 Mixer", AK4671_LOUT1_POWER_MANAGERMENT, 1, 0, + &ak4671_rout1_mixer_controls[0], + ARRAY_SIZE(ak4671_rout1_mixer_controls)), + SND_SOC_DAPM_MIXER_E("LOUT2 Mixer", AK4671_LOUT2_POWER_MANAGERMENT, + 0, 0, &ak4671_lout2_mixer_controls[0], + ARRAY_SIZE(ak4671_lout2_mixer_controls), + ak4671_out2_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_MIXER_E("ROUT2 Mixer", AK4671_LOUT2_POWER_MANAGERMENT, + 1, 0, &ak4671_rout2_mixer_controls[0], + ARRAY_SIZE(ak4671_rout2_mixer_controls), + ak4671_out2_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_MIXER("LOUT3 Mixer", AK4671_LOUT3_POWER_MANAGERMENT, 0, 0, + &ak4671_lout3_mixer_controls[0], + ARRAY_SIZE(ak4671_lout3_mixer_controls)), + SND_SOC_DAPM_MIXER("ROUT3 Mixer", AK4671_LOUT3_POWER_MANAGERMENT, 1, 0, + &ak4671_rout3_mixer_controls[0], + ARRAY_SIZE(ak4671_rout3_mixer_controls)), + + /* Input MUXs */ + SND_SOC_DAPM_MUX("LIN MUX", AK4671_AD_DA_POWER_MANAGEMENT, 2, 0, + &ak4671_lin_mux_control), + SND_SOC_DAPM_MUX("RIN MUX", AK4671_AD_DA_POWER_MANAGEMENT, 3, 0, + &ak4671_rin_mux_control), + + /* Mic Power */ + SND_SOC_DAPM_MICBIAS("Mic Bias", AK4671_AD_DA_POWER_MANAGEMENT, 1, 0), + + /* Supply */ + SND_SOC_DAPM_SUPPLY("PMPLL", AK4671_PLL_MODE_SELECT1, 0, 0, NULL, 0), +}; + +static const struct snd_soc_dapm_route intercon[] = { + {"DAC Left", "NULL", "PMPLL"}, + {"DAC Right", "NULL", "PMPLL"}, + {"ADC Left", "NULL", "PMPLL"}, + {"ADC Right", "NULL", "PMPLL"}, + + /* Outputs */ + {"LOUT1", "NULL", "LOUT1 Mixer"}, + {"ROUT1", "NULL", "ROUT1 Mixer"}, + {"LOUT2", "NULL", "LOUT2 Mix Amp"}, + {"ROUT2", "NULL", "ROUT2 Mix Amp"}, + {"LOUT3", "NULL", "LOUT3 Mixer"}, + {"ROUT3", "NULL", "ROUT3 Mixer"}, + + {"LOUT1 Mixer", "DACL", "DAC Left"}, + {"ROUT1 Mixer", "DACR", "DAC Right"}, + {"LOUT2 Mixer", "DACHL", "DAC Left"}, + {"ROUT2 Mixer", "DACHR", "DAC Right"}, + {"LOUT2 Mix Amp", "NULL", "LOUT2 Mixer"}, + {"ROUT2 Mix Amp", "NULL", "ROUT2 Mixer"}, + {"LOUT3 Mixer", "DACSL", "DAC Left"}, + {"ROUT3 Mixer", "DACSR", "DAC Right"}, + + /* Inputs */ + {"LIN MUX", "LIN1", "LIN1"}, + {"LIN MUX", "LIN2", "LIN2"}, + {"LIN MUX", "LIN3", "LIN3"}, + {"LIN MUX", "LIN4", "LIN4"}, + + {"RIN MUX", "RIN1", "RIN1"}, + {"RIN MUX", "RIN2", "RIN2"}, + {"RIN MUX", "RIN3", "RIN3"}, + {"RIN MUX", "RIN4", "RIN4"}, + + {"LIN1", NULL, "Mic Bias"}, + {"RIN1", NULL, "Mic Bias"}, + {"LIN2", NULL, "Mic Bias"}, + {"RIN2", NULL, "Mic Bias"}, + + {"ADC Left", "NULL", "LIN MUX"}, + {"ADC Right", "NULL", "RIN MUX"}, + + /* Analog Loops */ + {"LIN1 Mixing Circuit", "NULL", "LIN1"}, + {"RIN1 Mixing Circuit", "NULL", "RIN1"}, + {"LIN2 Mixing Circuit", "NULL", "LIN2"}, + {"RIN2 Mixing Circuit", "NULL", "RIN2"}, + {"LIN3 Mixing Circuit", "NULL", "LIN3"}, + {"RIN3 Mixing Circuit", "NULL", "RIN3"}, + {"LIN4 Mixing Circuit", "NULL", "LIN4"}, + {"RIN4 Mixing Circuit", "NULL", "RIN4"}, + + {"LOUT1 Mixer", "LINL1", "LIN1 Mixing Circuit"}, + {"ROUT1 Mixer", "RINR1", "RIN1 Mixing Circuit"}, + {"LOUT2 Mixer", "LINH1", "LIN1 Mixing Circuit"}, + {"ROUT2 Mixer", "RINH1", "RIN1 Mixing Circuit"}, + {"LOUT3 Mixer", "LINS1", "LIN1 Mixing Circuit"}, + {"ROUT3 Mixer", "RINS1", "RIN1 Mixing Circuit"}, + + {"LOUT1 Mixer", "LINL2", "LIN2 Mixing Circuit"}, + {"ROUT1 Mixer", "RINR2", "RIN2 Mixing Circuit"}, + {"LOUT2 Mixer", "LINH2", "LIN2 Mixing Circuit"}, + {"ROUT2 Mixer", "RINH2", "RIN2 Mixing Circuit"}, + {"LOUT3 Mixer", "LINS2", "LIN2 Mixing Circuit"}, + {"ROUT3 Mixer", "RINS2", "RIN2 Mixing Circuit"}, + + {"LOUT1 Mixer", "LINL3", "LIN3 Mixing Circuit"}, + {"ROUT1 Mixer", "RINR3", "RIN3 Mixing Circuit"}, + {"LOUT2 Mixer", "LINH3", "LIN3 Mixing Circuit"}, + {"ROUT2 Mixer", "RINH3", "RIN3 Mixing Circuit"}, + {"LOUT3 Mixer", "LINS3", "LIN3 Mixing Circuit"}, + {"ROUT3 Mixer", "RINS3", "RIN3 Mixing Circuit"}, + + {"LOUT1 Mixer", "LINL4", "LIN4 Mixing Circuit"}, + {"ROUT1 Mixer", "RINR4", "RIN4 Mixing Circuit"}, + {"LOUT2 Mixer", "LINH4", "LIN4 Mixing Circuit"}, + {"ROUT2 Mixer", "RINH4", "RIN4 Mixing Circuit"}, + {"LOUT3 Mixer", "LINS4", "LIN4 Mixing Circuit"}, + {"ROUT3 Mixer", "RINS4", "RIN4 Mixing Circuit"}, +}; + +static int ak4671_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, ak4671_dapm_widgets, + ARRAY_SIZE(ak4671_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static int ak4671_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + u8 fs; + + fs = snd_soc_read(codec, AK4671_PLL_MODE_SELECT0); + fs &= ~AK4671_FS; + + switch (params_rate(params)) { + case 8000: + fs |= AK4671_FS_8KHZ; + break; + case 12000: + fs |= AK4671_FS_12KHZ; + break; + case 16000: + fs |= AK4671_FS_16KHZ; + break; + case 24000: + fs |= AK4671_FS_24KHZ; + break; + case 11025: + fs |= AK4671_FS_11_025KHZ; + break; + case 22050: + fs |= AK4671_FS_22_05KHZ; + break; + case 32000: + fs |= AK4671_FS_32KHZ; + break; + case 44100: + fs |= AK4671_FS_44_1KHZ; + break; + case 48000: + fs |= AK4671_FS_48KHZ; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, AK4671_PLL_MODE_SELECT0, fs); + + return 0; +} + +static int ak4671_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + u8 pll; + + pll = snd_soc_read(codec, AK4671_PLL_MODE_SELECT0); + pll &= ~AK4671_PLL; + + switch (freq) { + case 11289600: + pll |= AK4671_PLL_11_2896MHZ; + break; + case 12000000: + pll |= AK4671_PLL_12MHZ; + break; + case 12288000: + pll |= AK4671_PLL_12_288MHZ; + break; + case 13000000: + pll |= AK4671_PLL_13MHZ; + break; + case 13500000: + pll |= AK4671_PLL_13_5MHZ; + break; + case 19200000: + pll |= AK4671_PLL_19_2MHZ; + break; + case 24000000: + pll |= AK4671_PLL_24MHZ; + break; + case 26000000: + pll |= AK4671_PLL_26MHZ; + break; + case 27000000: + pll |= AK4671_PLL_27MHZ; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, AK4671_PLL_MODE_SELECT0, pll); + + return 0; +} + +static int ak4671_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + u8 mode; + u8 format; + + /* set master/slave audio interface */ + mode = snd_soc_read(codec, AK4671_PLL_MODE_SELECT1); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + mode |= AK4671_M_S; + break; + case SND_SOC_DAIFMT_CBM_CFS: + mode &= ~(AK4671_M_S); + break; + default: + return -EINVAL; + } + + /* interface format */ + format = snd_soc_read(codec, AK4671_FORMAT_SELECT); + format &= ~AK4671_DIF; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + format |= AK4671_DIF_I2S_MODE; + break; + case SND_SOC_DAIFMT_LEFT_J: + format |= AK4671_DIF_MSB_MODE; + break; + case SND_SOC_DAIFMT_DSP_A: + format |= AK4671_DIF_DSP_MODE; + format |= AK4671_BCKP; + format |= AK4671_MSBS; + break; + default: + return -EINVAL; + } + + /* set mode and format */ + snd_soc_write(codec, AK4671_PLL_MODE_SELECT1, mode); + snd_soc_write(codec, AK4671_FORMAT_SELECT, format); + + return 0; +} + +static int ak4671_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u8 reg; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + reg = snd_soc_read(codec, AK4671_AD_DA_POWER_MANAGEMENT); + snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT, + reg | AK4671_PMVCM); + break; + case SND_SOC_BIAS_OFF: + snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT, 0x00); + break; + } + codec->bias_level = level; + return 0; +} + +#define AK4671_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000) + +#define AK4671_FORMATS SNDRV_PCM_FMTBIT_S16_LE + +static struct snd_soc_dai_ops ak4671_dai_ops = { + .hw_params = ak4671_hw_params, + .set_sysclk = ak4671_set_dai_sysclk, + .set_fmt = ak4671_set_dai_fmt, +}; + +struct snd_soc_dai ak4671_dai = { + .name = "AK4671", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = AK4671_RATES, + .formats = AK4671_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = AK4671_RATES, + .formats = AK4671_FORMATS,}, + .ops = &ak4671_dai_ops, +}; +EXPORT_SYMBOL_GPL(ak4671_dai); + +static int ak4671_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (ak4671_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = ak4671_codec; + codec = ak4671_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + snd_soc_add_controls(codec, ak4671_snd_controls, + ARRAY_SIZE(ak4671_snd_controls)); + ak4671_add_widgets(codec); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card: %d\n", ret); + goto card_err; + } + + ak4671_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; +} + +static int ak4671_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ak4671 = { + .probe = ak4671_probe, + .remove = ak4671_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ak4671); + +static int ak4671_register(struct ak4671_priv *ak4671, + enum snd_soc_control_type control) +{ + int ret; + struct snd_soc_codec *codec = &ak4671->codec; + + if (ak4671_codec) { + dev_err(codec->dev, "Another AK4671 is registered\n"); + ret = -EINVAL; + goto err; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = ak4671; + codec->name = "AK4671"; + codec->owner = THIS_MODULE; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = ak4671_set_bias_level; + codec->dai = &ak4671_dai; + codec->num_dai = 1; + codec->reg_cache_size = AK4671_CACHEREGNUM; + codec->reg_cache = &ak4671->reg_cache; + + memcpy(codec->reg_cache, ak4671_reg, sizeof(ak4671_reg)); + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, control); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + + ak4671_dai.dev = codec->dev; + ak4671_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto err; + } + + ret = snd_soc_register_dai(&ak4671_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + goto err_codec; + } + + return 0; + +err_codec: + snd_soc_unregister_codec(codec); +err: + kfree(ak4671); + return ret; +} + +static void ak4671_unregister(struct ak4671_priv *ak4671) +{ + ak4671_set_bias_level(&ak4671->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dai(&ak4671_dai); + snd_soc_unregister_codec(&ak4671->codec); + kfree(ak4671); + ak4671_codec = NULL; +} + +static int __devinit ak4671_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct ak4671_priv *ak4671; + struct snd_soc_codec *codec; + + ak4671 = kzalloc(sizeof(struct ak4671_priv), GFP_KERNEL); + if (ak4671 == NULL) + return -ENOMEM; + + codec = &ak4671->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + + i2c_set_clientdata(client, ak4671); + codec->control_data = client; + + codec->dev = &client->dev; + + return ak4671_register(ak4671, SND_SOC_I2C); +} + +static __devexit int ak4671_i2c_remove(struct i2c_client *client) +{ + struct ak4671_priv *ak4671 = i2c_get_clientdata(client); + + ak4671_unregister(ak4671); + + return 0; +} + +static const struct i2c_device_id ak4671_i2c_id[] = { + { "ak4671", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ak4671_i2c_id); + +static struct i2c_driver ak4671_i2c_driver = { + .driver = { + .name = "ak4671", + .owner = THIS_MODULE, + }, + .probe = ak4671_i2c_probe, + .remove = __devexit_p(ak4671_i2c_remove), + .id_table = ak4671_i2c_id, +}; + +static int __init ak4671_modinit(void) +{ + return i2c_add_driver(&ak4671_i2c_driver); +} +module_init(ak4671_modinit); + +static void __exit ak4671_exit(void) +{ + i2c_del_driver(&ak4671_i2c_driver); +} +module_exit(ak4671_exit); + +MODULE_DESCRIPTION("ASoC AK4671 codec driver"); +MODULE_AUTHOR("Joonyoung Shim <jy0922.shim(a)samsung.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ak4671.h b/sound/soc/codecs/ak4671.h new file mode 100644 index 0000000..e2fad96 --- /dev/null +++ b/sound/soc/codecs/ak4671.h @@ -0,0 +1,156 @@ +/* + * ak4671.h -- audio driver for AK4671 + * + * Copyright (C) 2009 Samsung Electronics Co.Ltd + * Author: Joonyoung Shim <jy0922.shim(a)samsung.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#ifndef _AK4671_H +#define _AK4671_H + +#define AK4671_AD_DA_POWER_MANAGEMENT 0x00 +#define AK4671_PLL_MODE_SELECT0 0x01 +#define AK4671_PLL_MODE_SELECT1 0x02 +#define AK4671_FORMAT_SELECT 0x03 +#define AK4671_MIC_SIGNAL_SELECT 0x04 +#define AK4671_MIC_AMP_GAIN 0x05 +#define AK4671_MIXING_POWER_MANAGEMENT0 0x06 +#define AK4671_MIXING_POWER_MANAGEMENT1 0x07 +#define AK4671_OUTPUT_VOLUME_CONTROL 0x08 +#define AK4671_LOUT1_SIGNAL_SELECT 0x09 +#define AK4671_ROUT1_SIGNAL_SELECT 0x0a +#define AK4671_LOUT2_SIGNAL_SELECT 0x0b +#define AK4671_ROUT2_SIGNAL_SELECT 0x0c +#define AK4671_LOUT3_SIGNAL_SELECT 0x0d +#define AK4671_ROUT3_SIGNAL_SELECT 0x0e +#define AK4671_LOUT1_POWER_MANAGERMENT 0x0f +#define AK4671_LOUT2_POWER_MANAGERMENT 0x10 +#define AK4671_LOUT3_POWER_MANAGERMENT 0x11 +#define AK4671_LCH_INPUT_VOLUME_CONTROL 0x12 +#define AK4671_RCH_INPUT_VOLUME_CONTROL 0x13 +#define AK4671_ALC_REFERENCE_SELECT 0x14 +#define AK4671_DIGITAL_MIXING_CONTROL 0x15 +#define AK4671_ALC_TIMER_SELECT 0x16 +#define AK4671_ALC_MODE_CONTROL 0x17 +#define AK4671_MODE_CONTROL1 0x18 +#define AK4671_MODE_CONTROL2 0x19 +#define AK4671_LCH_OUTPUT_VOLUME_CONTROL 0x1a +#define AK4671_RCH_OUTPUT_VOLUME_CONTROL 0x1b +#define AK4671_SIDETONE_A_CONTROL 0x1c +#define AK4671_DIGITAL_FILTER_SELECT 0x1d +#define AK4671_FIL3_COEFFICIENT0 0x1e +#define AK4671_FIL3_COEFFICIENT1 0x1f +#define AK4671_FIL3_COEFFICIENT2 0x20 +#define AK4671_FIL3_COEFFICIENT3 0x21 +#define AK4671_EQ_COEFFICIENT0 0x22 +#define AK4671_EQ_COEFFICIENT1 0x23 +#define AK4671_EQ_COEFFICIENT2 0x24 +#define AK4671_EQ_COEFFICIENT3 0x25 +#define AK4671_EQ_COEFFICIENT4 0x26 +#define AK4671_EQ_COEFFICIENT5 0x27 +#define AK4671_FIL1_COEFFICIENT0 0x28 +#define AK4671_FIL1_COEFFICIENT1 0x29 +#define AK4671_FIL1_COEFFICIENT2 0x2a +#define AK4671_FIL1_COEFFICIENT3 0x2b +#define AK4671_FIL2_COEFFICIENT0 0x2c +#define AK4671_FIL2_COEFFICIENT1 0x2d +#define AK4671_FIL2_COEFFICIENT2 0x2e +#define AK4671_FIL2_COEFFICIENT3 0x2f +#define AK4671_DIGITAL_FILTER_SELECT2 0x30 +#define AK4671_E1_COEFFICIENT0 0x32 +#define AK4671_E1_COEFFICIENT1 0x33 +#define AK4671_E1_COEFFICIENT2 0x34 +#define AK4671_E1_COEFFICIENT3 0x35 +#define AK4671_E1_COEFFICIENT4 0x36 +#define AK4671_E1_COEFFICIENT5 0x37 +#define AK4671_E2_COEFFICIENT0 0x38 +#define AK4671_E2_COEFFICIENT1 0x39 +#define AK4671_E2_COEFFICIENT2 0x3a +#define AK4671_E2_COEFFICIENT3 0x3b +#define AK4671_E2_COEFFICIENT4 0x3c +#define AK4671_E2_COEFFICIENT5 0x3d +#define AK4671_E3_COEFFICIENT0 0x3e +#define AK4671_E3_COEFFICIENT1 0x3f +#define AK4671_E3_COEFFICIENT2 0x40 +#define AK4671_E3_COEFFICIENT3 0x41 +#define AK4671_E3_COEFFICIENT4 0x42 +#define AK4671_E3_COEFFICIENT5 0x43 +#define AK4671_E4_COEFFICIENT0 0x44 +#define AK4671_E4_COEFFICIENT1 0x45 +#define AK4671_E4_COEFFICIENT2 0x46 +#define AK4671_E4_COEFFICIENT3 0x47 +#define AK4671_E4_COEFFICIENT4 0x48 +#define AK4671_E4_COEFFICIENT5 0x49 +#define AK4671_E5_COEFFICIENT0 0x4a +#define AK4671_E5_COEFFICIENT1 0x4b +#define AK4671_E5_COEFFICIENT2 0x4c +#define AK4671_E5_COEFFICIENT3 0x4d +#define AK4671_E5_COEFFICIENT4 0x4e +#define AK4671_E5_COEFFICIENT5 0x4f +#define AK4671_EQ_CONTROL_250HZ_100HZ 0x50 +#define AK4671_EQ_CONTROL_3500HZ_1KHZ 0x51 +#define AK4671_EQ_CONTRO_10KHZ 0x52 +#define AK4671_PCM_IF_CONTROL0 0x53 +#define AK4671_PCM_IF_CONTROL1 0x54 +#define AK4671_PCM_IF_CONTROL2 0x55 +#define AK4671_DIGITAL_VOLUME_B_CONTROL 0x56 +#define AK4671_DIGITAL_VOLUME_C_CONTROL 0x57 +#define AK4671_SIDETONE_VOLUME_CONTROL 0x58 +#define AK4671_DIGITAL_MIXING_CONTROL2 0x59 +#define AK4671_SAR_ADC_CONTROL 0x5a + +#define AK4671_CACHEREGNUM (AK4671_SAR_ADC_CONTROL + 1) + +/* Bitfield Definitions */ + +/* AK4671_AD_DA_POWER_MANAGEMENT (0x00) Fields */ +#define AK4671_PMVCM 0x01 + +/* AK4671_PLL_MODE_SELECT0 (0x01) Fields */ +#define AK4671_PLL 0x0f +#define AK4671_PLL_11_2896MHZ (4 << 0) +#define AK4671_PLL_12_288MHZ (5 << 0) +#define AK4671_PLL_12MHZ (6 << 0) +#define AK4671_PLL_24MHZ (7 << 0) +#define AK4671_PLL_19_2MHZ (8 << 0) +#define AK4671_PLL_13_5MHZ (12 << 0) +#define AK4671_PLL_27MHZ (13 << 0) +#define AK4671_PLL_13MHZ (14 << 0) +#define AK4671_PLL_26MHZ (15 << 0) +#define AK4671_FS 0xf0 +#define AK4671_FS_8KHZ (0 << 4) +#define AK4671_FS_12KHZ (1 << 4) +#define AK4671_FS_16KHZ (2 << 4) +#define AK4671_FS_24KHZ (3 << 4) +#define AK4671_FS_11_025KHZ (5 << 4) +#define AK4671_FS_22_05KHZ (7 << 4) +#define AK4671_FS_32KHZ (10 << 4) +#define AK4671_FS_48KHZ (11 << 4) +#define AK4671_FS_44_1KHZ (15 << 4) + +/* AK4671_PLL_MODE_SELECT1 (0x02) Fields */ +#define AK4671_PMPLL 0x01 +#define AK4671_M_S 0x02 + +/* AK4671_FORMAT_SELECT (0x03) Fields */ +#define AK4671_DIF 0x03 +#define AK4671_DIF_DSP_MODE (0 << 0) +#define AK4671_DIF_MSB_MODE (2 << 0) +#define AK4671_DIF_I2S_MODE (3 << 0) +#define AK4671_BCKP 0x04 +#define AK4671_MSBS 0x08 +#define AK4671_SDOD 0x10 + +/* AK4671_LOUT2_POWER_MANAGEMENT (0x10) Fields */ +#define AK4671_MUTEN 0x04 + +extern struct snd_soc_dai ak4671_dai; +extern struct snd_soc_codec_device soc_codec_dev_ak4671; + +#endif -- 1.6.0.4

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[alsa-devel] Is there a way to figure out if the clocks of two snd_pcm_t's deviate? (snd_pcm_info_get_sync?)
by Lennart Poettering 09 Sep '09

09 Sep '09

Heya! I am looking for a way to find out whether the clocks of two snd_pcm_t's are perfectly in sync and don't deviate (or only deviate by a constant phase). i.e. for real duplex devices that use the same crystal for input and output this should tell me "Yes" for the playback snd_pcm_t tested against the recording snd_pcm_t; and for all other, unrelated audio devices it should return "No". I found snd_pcm_info_get_sync() in the docs, but its documentation is a bit terse, so I am wondering if this might be what I want. Is it? Does this (untested) code snippet do what I hope it does? <snip> snd_pcm_info(a, info_a); snd_pcm_info(b, info_b); in_sync = memcmp(&snd_pcm_info_get_sync(info_a), &snd_pcm_info_get_sync(info_b)) == 0; </snip> Thanks, Lennart -- Lennart Poettering Red Hat, Inc. lennart [at] poettering [dot] net http://0pointer.net/lennart/ GnuPG 0x1A015CC4

4 6

[alsa-devel] pts value using alsa for audio capture
by Nicolas Krieger 09 Sep '09

09 Sep '09

Hi, I have a question about how pts are computed using alsa while grabbing audio. I take an example to explain : I do grab audio during 10 seconds, the first 2 seconds, there is no problem, so the pts are like that (I take examples values) : 1 2 3 4 5 6 .... x <- x is the value of the last pts (2nd second). Then, for any reason, no capture during 3 seconds. And then no problem until the end. Nicolas Krieger

1 1

Re: [alsa-devel] [PATCH 2/3] ASoC: DaVinci: pcm, rename variables in prep for ping/pong
by Nori, Sekhar 09 Sep '09

09 Sep '09

On Wed, Sep 09, 2009 at 18:03:58, Caglar Akyuz wrote: > On Wednesday 09 September 2009 15:08:42 Nori, Sekhar wrote: [...] > > > > However, with or without your patch, I noticed a segmentation fault for the > > first time the 'arecord | aplay' loop is created. There is no fault the > > second time around. I haven't spent time debugging this yet. > > > > This is due to these lines in davinci_pcm_enqueue_dma: > > data_type = prtd->params->data_type; > count = period_size / data_type; > > First time running data_type is set to 0. This is a long time issue, I > remember modifying my kernels to check for zero before this division.. Ah, thanks! Planning to submit a fix for this? We can work on the patch if you are busy otherwise. Regards, Sekhar

3 2

[alsa-devel] Layla 3G does not recover from ACPI Suspend
by Mark Hills 09 Sep '09

09 Sep '09

I have an Echo Layla 3G in my workstation. It works for audio, but does not recover from ACPI suspend to RAM. On recovery the system is fine, and the Layla exists in /proc/asound/cards. But when the Layla is used it prints this message to dmesg, multiple times: wait_handshake(): Timeout waiting for DSP Here are the relevant dmesg lines after awakening: Echoaudio Echo3G 0000:03:03.0: restoring config space at offset 0xf (was 0x100, writing 0x104) Echoaudio Echo3G 0000:03:03.0: restoring config space at offset 0x4 (was 0x0, writing 0xe7d00000) Echoaudio Echo3G 0000:03:03.0: restoring config space at offset 0x3 (was 0x0, writing 0xc010) Echoaudio Echo3G 0000:03:03.0: restoring config space at offset 0x1 (was 0x2800000, writing 0x2800112) Echoaudio Echo3G 0000:03:03.0: PCI INT A -> GSI 19 (level, low) -> IRQ 19 I couldn't see anything loading the firmware. I thought unloading and loading snd-echo3g after recovery would help. This shows the firmware being loaded, but then no ALSA device is shown in /proc/asound/cards. On loading snd-echo3g: Echoaudio Echo3G 0000:03:03.0: PCI INT A -> GSI 19 (level, low) -> IRQ 19 Echoaudio Echo3G 0000:03:03.0: firmware: requesting ea/echo3g_dsp.fw Echoaudio Echo3G 0000:03:03.0: firmware: requesting ea/loader_dsp.fw Echoaudio Echo3G 0000:03:03.0: PCI INT A disabled Echoaudio Echo3G: probe of 0000:03:03.0 failed with error -5 Is the firmware not being loaded when it should be? Or is there some extra initialisation not being done (eg. init_hw() in echo3g_dsp.c)? All other devices in the system recover fine, but as far as I'm aware no other PCI devices require firmware. It's a Dell x86 system, kernel 2.6.31-rc7. Thanks for any help, -- Mark

3 3

[alsa-devel] [PATCH v2] ASoC: au1x: PSC-AC97 bugfixes
by Manuel Lauss 08 Sep '09

08 Sep '09

This patch fixes the following bugs: - only reprogram bitdepth if it has changed since last call to hw_params. - add locking inside ac97_read/write functions: When reprogramming sample depth, the ac97 unit has to be disabled, which should not be done in the middle of codec register accesses. - retry timed-out codec register accesses. - wait for status bits to set/clear when starting/stopping various functional blocks; very important after reenabling AC97 unit else sound may be distorted (e.g. high-pitch noise in 1kHz sine wave). - clear fifos before/after starting/stopping RX/TX. - longer timeouts waiting for PSC/AC97 ready after cold reset with certain codecs this can take ridiculous amounts of time. Run-tested on various Au1200 platforms with various codecs. Signed-off-by: Manuel Lauss <manuel.lauss(a)gmail.com> --- The previous patch missed a git add for psc.h, sorry! sound/soc/au1x/psc-ac97.c | 129 +++++++++++++++++++++++++++++++++------------ sound/soc/au1x/psc.h | 1 + 2 files changed, 97 insertions(+), 33 deletions(-) diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 479d7bd..a521aa9 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -1,8 +1,8 @@ /* * Au12x0/Au1550 PSC ALSA ASoC audio support. * - * (c) 2007-2008 MSC Vertriebsges.m.b.H., - * Manuel Lauss <mano(a)roarinelk.homelinux.net> + * (c) 2007-2009 MSC Vertriebsges.m.b.H., + * Manuel Lauss <manuel.lauss(a)gmail.com> * * 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 @@ -19,6 +19,7 @@ #include <linux/module.h> #include <linux/device.h> #include <linux/delay.h> +#include <linux/mutex.h> #include <linux/suspend.h> #include <sound/core.h> #include <sound/pcm.h> @@ -29,6 +30,9 @@ #include "psc.h" +/* how often to retry failed codec register reads/writes */ +#define AC97_RW_RETRIES 5 + #define AC97_DIR \ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) @@ -45,6 +49,9 @@ #define AC97PCR_CLRFIFO(stype) \ ((stype) == PCM_TX ? PSC_AC97PCR_TC : PSC_AC97PCR_RC) +#define AC97STAT_BUSY(stype) \ + ((stype) == PCM_TX ? PSC_AC97STAT_TB : PSC_AC97STAT_RB) + /* instance data. There can be only one, MacLeod!!!! */ static struct au1xpsc_audio_data *au1xpsc_ac97_workdata; @@ -54,24 +61,33 @@ static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97, { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; - unsigned short data, tmo; + unsigned short data, retry, tmo; - au_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg), AC97_CDC(pscdata)); + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); au_sync(); - tmo = 1000; - while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo) - udelay(2); + retry = AC97_RW_RETRIES; + do { + mutex_lock(&pscdata->lock); + + au_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg), + AC97_CDC(pscdata)); + au_sync(); + + tmo = 2000; + while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) + && --tmo) + udelay(2); - if (!tmo) - data = 0xffff; - else data = au_readl(AC97_CDC(pscdata)) & 0xffff; - au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); - au_sync(); + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); + au_sync(); + + mutex_unlock(&pscdata->lock); + } while (--retry && !tmo); - return data; + return retry ? data : 0xffff; } /* AC97 controller writes to codec register */ @@ -80,16 +96,29 @@ static void au1xpsc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; - unsigned int tmo; + unsigned int tmo, retry; - au_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff), AC97_CDC(pscdata)); + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); au_sync(); - tmo = 1000; - while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo) + + retry = AC97_RW_RETRIES; + do { + mutex_lock(&pscdata->lock); + + au_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff), + AC97_CDC(pscdata)); au_sync(); - au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); - au_sync(); + tmo = 2000; + while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) + && --tmo) + udelay(2); + + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); + au_sync(); + + mutex_unlock(&pscdata->lock); + } while (--retry && !tmo); } /* AC97 controller asserts a warm reset */ @@ -129,9 +158,9 @@ static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97) au_sync(); /* wait for PSC to indicate it's ready */ - i = 100000; + i = 1000; while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_SR)) && (--i)) - au_sync(); + msleep(1); if (i == 0) { printk(KERN_ERR "au1xpsc-ac97: PSC not ready!\n"); @@ -143,9 +172,9 @@ static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97) au_sync(); /* wait for AC97 core to become ready */ - i = 100000; + i = 1000; while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) && (--i)) - au_sync(); + msleep(1); if (i == 0) printk(KERN_ERR "au1xpsc-ac97: AC97 ctrl not ready\n"); } @@ -165,12 +194,12 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; - unsigned long r, stat; + unsigned long r, ro, stat; int chans, stype = SUBSTREAM_TYPE(substream); chans = params_channels(params); - r = au_readl(AC97_CFG(pscdata)); + r = ro = au_readl(AC97_CFG(pscdata)); stat = au_readl(AC97_STAT(pscdata)); /* already active? */ @@ -180,9 +209,6 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, (pscdata->rate != params_rate(params))) return -EINVAL; } else { - /* disable AC97 device controller first */ - au_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); - au_sync(); /* set sample bitdepth: REG[24:21]=(BITS-2)/2 */ r &= ~PSC_AC97CFG_LEN_MASK; @@ -199,14 +225,40 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, r |= PSC_AC97CFG_RXSLOT_ENA(4); } - /* finally enable the AC97 controller again */ + /* do we need to poke the hardware? */ + if (!(r ^ ro)) + goto out; + + /* ac97 engine is about to be disabled */ + mutex_lock(&pscdata->lock); + + /* disable AC97 device controller first... */ + au_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); + au_sync(); + + /* ...wait for it... */ + while (au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR) + asm volatile ("nop"); + + /* ...write config... */ + au_writel(r, AC97_CFG(pscdata)); + au_sync(); + + /* ...enable the AC97 controller again... */ au_writel(r | PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); au_sync(); + /* ...and wait for ready bit */ + while (!(au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) + asm volatile ("nop"); + + mutex_unlock(&pscdata->lock); + pscdata->cfg = r; pscdata->rate = params_rate(params); } +out: return 0; } @@ -222,6 +274,8 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: + au_writel(AC97PCR_CLRFIFO(stype), AC97_PCR(pscdata)); + au_sync(); au_writel(AC97PCR_START(stype), AC97_PCR(pscdata)); au_sync(); break; @@ -229,6 +283,13 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_SUSPEND: au_writel(AC97PCR_STOP(stype), AC97_PCR(pscdata)); au_sync(); + + while (au_readl(AC97_STAT(pscdata)) & AC97STAT_BUSY(stype)) + asm volatile ("nop"); + + au_writel(AC97PCR_CLRFIFO(stype), AC97_PCR(pscdata)); + au_sync(); + break; default: ret = -EINVAL; @@ -251,6 +312,8 @@ static int au1xpsc_ac97_probe(struct platform_device *pdev, if (!au1xpsc_ac97_workdata) return -ENOMEM; + mutex_init(&au1xpsc_ac97_workdata->lock); + r = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!r) { ret = -ENODEV; @@ -269,9 +332,9 @@ static int au1xpsc_ac97_probe(struct platform_device *pdev, goto out1; /* configuration: max dma trigger threshold, enable ac97 */ - au1xpsc_ac97_workdata->cfg = PSC_AC97CFG_RT_FIFO8 | - PSC_AC97CFG_TT_FIFO8 | - PSC_AC97CFG_DE_ENABLE; + au1xpsc_ac97_workdata->cfg = PSC_AC97CFG_RT_FIFO8 | + PSC_AC97CFG_TT_FIFO8 | + PSC_AC97CFG_DE_ENABLE; /* preserve PSC clock source set up by platform (dev.platform_data * is already occupied by soc layer) @@ -386,4 +449,4 @@ module_exit(au1xpsc_ac97_exit); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC AC97 ALSA ASoC audio driver"); -MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>"); +MODULE_AUTHOR("Manuel Lauss <manuel.lauss(a)gmail.com>"); diff --git a/sound/soc/au1x/psc.h b/sound/soc/au1x/psc.h index 8fdb1a0..3f474e8 100644 --- a/sound/soc/au1x/psc.h +++ b/sound/soc/au1x/psc.h @@ -29,6 +29,7 @@ struct au1xpsc_audio_data { unsigned long pm[2]; struct resource *ioarea; + struct mutex lock; }; #define PCM_TX 0 -- 1.6.4.2

3 3

[alsa-devel] ASoC update for 2.6.32
by Mark Brown 08 Sep '09

08 Sep '09

The following changes since commit 87831cb660954356d68cebdb1406f3be09e784e9: Mark Brown (1): ASoC: Fix WM835x Out4 capture enumeration are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Manuel Lauss (1): ASoC: au1x: PSC-AC97 bugfixes sound/soc/au1x/psc-ac97.c | 129 +++++++++++++++++++++++++++++++++------------ sound/soc/au1x/psc.h | 1 + 2 files changed, 97 insertions(+), 33 deletions(-)

2 1

[alsa-devel] [PATCH] ASoC: au1x: PSC-AC97 bugfixes
by Manuel Lauss 08 Sep '09

08 Sep '09

This patch fixes the following bugs: - only reprogram bitdepth if it has changed since last call to hw_params. - add locking inside ac97_read/write functions: When reprogramming sample depth, the ac97 unit has to be disabled, which should not be done in the middle of codec register accesses. - retry timed-out codec register accesses. - wait for status bits to set/clear when starting/stopping various functional blocks; very important after reenabling AC97 unit else sound may be distorted (e.g. high-pitch noise in 1kHz sine wave). - clear fifos before/after starting/stopping RX/TX. - longer timeouts waiting for PSC/AC97 ready after cold reset with certain codecs this can take ridiculous amounts of time. Run-tested on various Au1200 platforms with various codecs. Signed-off-by: Manuel Lauss <manuel.lauss(a)gmail.com> --- sound/soc/au1x/psc-ac97.c | 129 +++++++++++++++++++++++++++++++++------------ 1 files changed, 96 insertions(+), 33 deletions(-) diff --git a/sound/soc/au1x/psc-ac97.c b/sound/soc/au1x/psc-ac97.c index 479d7bd..a521aa9 100644 --- a/sound/soc/au1x/psc-ac97.c +++ b/sound/soc/au1x/psc-ac97.c @@ -1,8 +1,8 @@ /* * Au12x0/Au1550 PSC ALSA ASoC audio support. * - * (c) 2007-2008 MSC Vertriebsges.m.b.H., - * Manuel Lauss <mano(a)roarinelk.homelinux.net> + * (c) 2007-2009 MSC Vertriebsges.m.b.H., + * Manuel Lauss <manuel.lauss(a)gmail.com> * * 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 @@ -19,6 +19,7 @@ #include <linux/module.h> #include <linux/device.h> #include <linux/delay.h> +#include <linux/mutex.h> #include <linux/suspend.h> #include <sound/core.h> #include <sound/pcm.h> @@ -29,6 +30,9 @@ #include "psc.h" +/* how often to retry failed codec register reads/writes */ +#define AC97_RW_RETRIES 5 + #define AC97_DIR \ (SND_SOC_DAIDIR_PLAYBACK | SND_SOC_DAIDIR_CAPTURE) @@ -45,6 +49,9 @@ #define AC97PCR_CLRFIFO(stype) \ ((stype) == PCM_TX ? PSC_AC97PCR_TC : PSC_AC97PCR_RC) +#define AC97STAT_BUSY(stype) \ + ((stype) == PCM_TX ? PSC_AC97STAT_TB : PSC_AC97STAT_RB) + /* instance data. There can be only one, MacLeod!!!! */ static struct au1xpsc_audio_data *au1xpsc_ac97_workdata; @@ -54,24 +61,33 @@ static unsigned short au1xpsc_ac97_read(struct snd_ac97 *ac97, { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; - unsigned short data, tmo; + unsigned short data, retry, tmo; - au_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg), AC97_CDC(pscdata)); + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); au_sync(); - tmo = 1000; - while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo) - udelay(2); + retry = AC97_RW_RETRIES; + do { + mutex_lock(&pscdata->lock); + + au_writel(PSC_AC97CDC_RD | PSC_AC97CDC_INDX(reg), + AC97_CDC(pscdata)); + au_sync(); + + tmo = 2000; + while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) + && --tmo) + udelay(2); - if (!tmo) - data = 0xffff; - else data = au_readl(AC97_CDC(pscdata)) & 0xffff; - au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); - au_sync(); + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); + au_sync(); + + mutex_unlock(&pscdata->lock); + } while (--retry && !tmo); - return data; + return retry ? data : 0xffff; } /* AC97 controller writes to codec register */ @@ -80,16 +96,29 @@ static void au1xpsc_ac97_write(struct snd_ac97 *ac97, unsigned short reg, { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; - unsigned int tmo; + unsigned int tmo, retry; - au_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff), AC97_CDC(pscdata)); + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); au_sync(); - tmo = 1000; - while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) && --tmo) + + retry = AC97_RW_RETRIES; + do { + mutex_lock(&pscdata->lock); + + au_writel(PSC_AC97CDC_INDX(reg) | (val & 0xffff), + AC97_CDC(pscdata)); au_sync(); - au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); - au_sync(); + tmo = 2000; + while ((!(au_readl(AC97_EVNT(pscdata)) & PSC_AC97EVNT_CD)) + && --tmo) + udelay(2); + + au_writel(PSC_AC97EVNT_CD, AC97_EVNT(pscdata)); + au_sync(); + + mutex_unlock(&pscdata->lock); + } while (--retry && !tmo); } /* AC97 controller asserts a warm reset */ @@ -129,9 +158,9 @@ static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97) au_sync(); /* wait for PSC to indicate it's ready */ - i = 100000; + i = 1000; while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_SR)) && (--i)) - au_sync(); + msleep(1); if (i == 0) { printk(KERN_ERR "au1xpsc-ac97: PSC not ready!\n"); @@ -143,9 +172,9 @@ static void au1xpsc_ac97_cold_reset(struct snd_ac97 *ac97) au_sync(); /* wait for AC97 core to become ready */ - i = 100000; + i = 1000; while (!((au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) && (--i)) - au_sync(); + msleep(1); if (i == 0) printk(KERN_ERR "au1xpsc-ac97: AC97 ctrl not ready\n"); } @@ -165,12 +194,12 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, { /* FIXME */ struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata; - unsigned long r, stat; + unsigned long r, ro, stat; int chans, stype = SUBSTREAM_TYPE(substream); chans = params_channels(params); - r = au_readl(AC97_CFG(pscdata)); + r = ro = au_readl(AC97_CFG(pscdata)); stat = au_readl(AC97_STAT(pscdata)); /* already active? */ @@ -180,9 +209,6 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, (pscdata->rate != params_rate(params))) return -EINVAL; } else { - /* disable AC97 device controller first */ - au_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); - au_sync(); /* set sample bitdepth: REG[24:21]=(BITS-2)/2 */ r &= ~PSC_AC97CFG_LEN_MASK; @@ -199,14 +225,40 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream, r |= PSC_AC97CFG_RXSLOT_ENA(4); } - /* finally enable the AC97 controller again */ + /* do we need to poke the hardware? */ + if (!(r ^ ro)) + goto out; + + /* ac97 engine is about to be disabled */ + mutex_lock(&pscdata->lock); + + /* disable AC97 device controller first... */ + au_writel(r & ~PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); + au_sync(); + + /* ...wait for it... */ + while (au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR) + asm volatile ("nop"); + + /* ...write config... */ + au_writel(r, AC97_CFG(pscdata)); + au_sync(); + + /* ...enable the AC97 controller again... */ au_writel(r | PSC_AC97CFG_DE_ENABLE, AC97_CFG(pscdata)); au_sync(); + /* ...and wait for ready bit */ + while (!(au_readl(AC97_STAT(pscdata)) & PSC_AC97STAT_DR)) + asm volatile ("nop"); + + mutex_unlock(&pscdata->lock); + pscdata->cfg = r; pscdata->rate = params_rate(params); } +out: return 0; } @@ -222,6 +274,8 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, switch (cmd) { case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: + au_writel(AC97PCR_CLRFIFO(stype), AC97_PCR(pscdata)); + au_sync(); au_writel(AC97PCR_START(stype), AC97_PCR(pscdata)); au_sync(); break; @@ -229,6 +283,13 @@ static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream, case SNDRV_PCM_TRIGGER_SUSPEND: au_writel(AC97PCR_STOP(stype), AC97_PCR(pscdata)); au_sync(); + + while (au_readl(AC97_STAT(pscdata)) & AC97STAT_BUSY(stype)) + asm volatile ("nop"); + + au_writel(AC97PCR_CLRFIFO(stype), AC97_PCR(pscdata)); + au_sync(); + break; default: ret = -EINVAL; @@ -251,6 +312,8 @@ static int au1xpsc_ac97_probe(struct platform_device *pdev, if (!au1xpsc_ac97_workdata) return -ENOMEM; + mutex_init(&au1xpsc_ac97_workdata->lock); + r = platform_get_resource(pdev, IORESOURCE_MEM, 0); if (!r) { ret = -ENODEV; @@ -269,9 +332,9 @@ static int au1xpsc_ac97_probe(struct platform_device *pdev, goto out1; /* configuration: max dma trigger threshold, enable ac97 */ - au1xpsc_ac97_workdata->cfg = PSC_AC97CFG_RT_FIFO8 | - PSC_AC97CFG_TT_FIFO8 | - PSC_AC97CFG_DE_ENABLE; + au1xpsc_ac97_workdata->cfg = PSC_AC97CFG_RT_FIFO8 | + PSC_AC97CFG_TT_FIFO8 | + PSC_AC97CFG_DE_ENABLE; /* preserve PSC clock source set up by platform (dev.platform_data * is already occupied by soc layer) @@ -386,4 +449,4 @@ module_exit(au1xpsc_ac97_exit); MODULE_LICENSE("GPL"); MODULE_DESCRIPTION("Au12x0/Au1550 PSC AC97 ALSA ASoC audio driver"); -MODULE_AUTHOR("Manuel Lauss <mano(a)roarinelk.homelinux.net>"); +MODULE_AUTHOR("Manuel Lauss <manuel.lauss(a)gmail.com>"); -- 1.6.4.2

1 0

[alsa-devel] Update to problem with Creative X-fi mixer problems
by Susan Cragin 08 Sep '09

08 Sep '09

I saw that someone was going to try to get all the mixer controls to work properly. I updated and opened alsamixer this morning, and found that both the incoming/recording controls were there! (I should have quit while I was ahead.) Then I re-booted and recompiled wine and opened winecfg and found that oss could find the proper mixer but alsa did not. (The oss mixer is listed as Creative CA0110-IBG) ALSA lib pcm_dmix.c:1008:(snd_pcm_dmix_open) unable to open slave ALSA lib pcm_dsnoop.c:593:(snd_pcm_dsnoop_open) unable to open slave fixme:mixer:ALSA_MixerInit No master control found on HD-Audio Generic, disabling mixer susan@ubuntu:~$ alsamixer No mixer elems found

1 0

Re: [alsa-devel] snd_dummy on Centos - Redhat
by Takashi Iwai 08 Sep '09

08 Sep '09

At Fri, 4 Sep 2009 00:43:09 +0200, Giovanni Maruzzelli wrote: > > On Fri, Sep 4, 2009 at 12:03 AM, Giovanni > Maruzzelli<gmaruzz.lists(a)gmail.com> wrote: > > I'll test it and report back here. > > Still we have big problems.... OK, then upgrade your alsa driver stack to 1.0.21. There have been some changes in PCM core. Then, get dummy.c again from the latest alsa-driver-unstable tarball. It already contains my latest fix. If 1.0.21+dummy.c doesn't work, apply the patch below to get position logs. If this still causes the invalid backward jump to position 0, you should check xrun debug, e.g. # echo 9 > /proc/asound/card0/pcm0p/xrun_debug (assuming snd-dummy is card0). Takashi --- diff --git a/dummy.c b/dummy.c index e8e29bf..f7c21d5 100644 --- a/dummy.c +++ b/dummy.c @@ -239,6 +239,10 @@ static void dummy_systimer_update(struct dummy_systimer_pcm *dpcm) dpcm->frac_period_rest += dpcm->frac_period_size; } dpcm->frac_period_rest -= delta; + printk(KERN_DEBUG "xxx delta=%ld, pos=%d(%d), rest=%d(%d)\n", + delta / dpcm->rate, + dpcm->frac_pos, dpcm->frac_pos / HZ, + dpcm->frac_period_rest, dpcm->frac_period_rest / HZ); } static int dummy_systimer_start(struct snd_pcm_substream *substream) @@ -248,6 +252,10 @@ static int dummy_systimer_start(struct snd_pcm_substream *substream) dpcm->base_time = jiffies; dummy_systimer_rearm(dpcm); spin_unlock(&dpcm->lock); + printk(KERN_DEBUG "*** START: buffer=%d, period=%d, rate=%d\n", + (int)substream->runtime->buffer_size, + (int)substream->runtime->period_size, + substream->runtime->rate); return 0; } @@ -257,6 +265,7 @@ static int dummy_systimer_stop(struct snd_pcm_substream *substream) spin_lock(&dpcm->lock); del_timer(&dpcm->timer); spin_unlock(&dpcm->lock); + printk(KERN_DEBUG "*** STOP\n"); return 0; }

2 24

[alsa-devel] Audio cutoff when playing using dmix
by ruchi sirauthiya 08 Sep '09

08 Sep '09

Hi, I am facing audio clipping issues when dmix is used for playback. For the test we used a full scale sine wave tone generated using software and encoded as "Signed 16 bit Little Endian, Rate 44100 Hz, Mono". The audio is getting cut out at a 2-3 second interval during playback. This happens only for a fraction of second but it is consistent. This does not happen when audio is played back on the device directly but only when going through dmix. We have observed that there were differences in audio cutting off when the sampling rate was set to 8000Hz and 32000 Hz in asound.conf. The contents of asound.conf is pasted below for your reference: pcm.dmixer { type dmix ipc_key 2048 ipc_key_add_uid true slave { pcm hw:0 rate 8000 channels 2 } } pcm.asymed { type asym playback.pcm "dmixer" capture.pcm "hw:0,0" } pcm.rateConvert { type plug slave.pcm "asymed" } The configuration given above works fine with dmix in 2.6.29 kernel, I am using kernel 2.6.22.18. I have compared the configuration differences (buffers and periods) with and without using dmix. I tried applying some parameters (like buffer_size, period_time and period_size) which was in use when audio was played directly to the hardware device by aplay using the following command: aplay -D rateConvert --period-time=92879 --buffer-size=65536 --period-size=4096 /mnt/test.wav -vvv -d 5 There was come improvement in the audio cutting off issue but it was not eliminated completely. Given below are the logs for aplay (-vvv) with and without using dmix (with audio cutoff issue on kernel 2.6.22): Without using dmix: [root@OMAP3 tests]# aplay /mnt/test.wav -vvv -d 5 5 Playing WAVE '/mnt/test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono Plug PCM: Hardware PCM card 0 'TWL4030' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 65536 period_size : 4096 period_time : 92879 tick_time : 7812 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 4096 xfer_align : 4096 start_threshold : 65536 stop_threshold : 65536 silence_threshold: 0 silence_size : 0 boundary : 1073741824 Max peak (4096 samples): 0x00006665 ################ 79% Max peak (4096 samples): 0x00006665 ################ 79% ...Same text repeats 50 times... Max peak (4096 samples): 0x00006665 ################ 79% Max peak (4096 samples): 0x00006665 ################ 79% Using dmix: [root@OMAP3 tests]# aplay -D rateConvert /mnt/test.wav -vvv -d 5 Playing WAVE '/mnt/test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono Plug PCM: Rate conversion PCM (8000, sformat=S16_LE) Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 22050 period_size : 5512 period_time : 125000 tick_time : 0 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 5512 xfer_align : 5512 start_threshold : 22048 stop_threshold : 22050 silence_threshold: 0 silence_size : 0 boundary : 1445068800 Slave: Route conversion PCM (sformat=S16_LE) Transformation table: 0 <- 0 1 <- 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 1 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tick_time : 0 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 1000 xfer_align : 1000 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 Slave: Direct Stream Mixing PCM Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 4000 period_size : 1000 period_time : 125000 tick_time : 0 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 1000 xfer_align : 1000 start_threshold : 4000 stop_threshold : 4000 silence_threshold: 0 silence_size : 0 boundary : 2097152000 Hardware PCM card 0 'TWL4030' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : MMAP_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 8000 exact rate : 8000 (8000/1) msbits : 16 buffer_size : 16000 period_size : 1000 period_time : 125000 tick_time : 7812 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 1000 xfer_align : 1000 start_threshold : 1 stop_threshold : 2097152000 silence_threshold: 0 silence_size : 2097152000 boundary : 2097152000 Max peak (5512 samples): 0x00006665 ################ 79% Max peak (5512 samples): 0x00006665 ################ 79% ...Same text repeats 36 times... Max peak (5512 samples): 0x00006665 ################ 79% Max peak (5512 samples): 0x00006665 ################ 79% Max peak (5512 samples): 0x00005f1a ############### 74% Please help me fix this issue. Does this occur due to non-optimal settings of the parameters (buffers, periods) in the configuration or is it some other issue with the ALSA library or the kernel? Please ask if any more setup/configuration settings are required for reference. Thanks and Regards, Ruchi Sirauthiya

1 0

[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 08 Sep '09

08 Sep '09

The following changes since commit 367da1527aff2a4f789d03082f9fb191cdf332c4: Robert Schwebel (1): ASoC: fix pxa2xx-ac97.c breakage are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Mark Brown (2): ASoC: Remove unuused hw_read_t ASoC: Fix WM835x Out4 capture enumeration include/sound/soc.h | 1 - sound/soc/codecs/wm8350.c | 2 +- 2 files changed, 1 insertions(+), 2 deletions(-)

2 1

Re: [alsa-devel] Patch for au88x0 driver. Fix audacious issue and add auto reset in case of hardware ADB ctrl failure.
by Raymond Yau 08 Sep '09

08 Sep '09

2009/9/7 Manuel Jander <manuel.jander(a)gmail.com> > Hi Raymond, > > On Mon, Sep 7, 2009 at 3:47 AM, Raymond Yau<superquad.vortex2(a)gmail.com> > wrote: > > On Thursday 03 September 2009 21:10, Manuel Jander wrote: > >> does anybody still have a AU88x0 card ? I know they are quite ancient > >> now, but if anyone could give the attached patch a try ? Would be > >> nice. > >> I made the patch against linux-2.6.30.4 > >> http://micro.homelinux.net/~mjander/au88x0.diff.gz<http://micro.homelinux.net/%7Emjander/au88x0.diff.gz>< > http://micro.homelinux.net/%7Emjander/au88x0.diff.gz> > >> > >> Best Regards, > >> Manuel > > > > Which version of audaicous (1.5.x or 2.1) are you testing ? > > (alsa-gapless or alsa-ng plugin) > > > > au88x0 most likely cannot support 1 sample per period because of the > > FIFO as described in patent 6167465. > > Well, that is clear to me now. Using a DMA transfer that is smaller > than one audio sample is not really a good idea. > I would be better off using PIO instead of DMA. > > > if you look at the source code of alsa-ng and alsa-gapless plugin > > both of them use the own way to calculate the buffer size instead of > > using snd_pcm_hw_params_get_buffer_size() > > > > > > > http://hg.atheme.org/audacious-plugins/file/2c0a9bba77d7/src/alsa-ng/alsa-c… > > > > CHECK_FAIL (snd_pcm_hw_params_set_access (pcm_handle, hwparams, > > SND_PCM_ACCESS_RW_INTERLEAVED), "snd_pcm_hw_params_set_access"); > > CHECK_FAIL (snd_pcm_hw_params_set_format (pcm_handle, hwparams, afmt), > > "snd_pcm_hw_params_set_format"); > > CHECK_FAIL (snd_pcm_hw_params_set_channels (pcm_handle, hwparams, nch), > > "snd_pcm_hw_params_set_channels"); > > CHECK_FAIL (snd_pcm_hw_params_set_rate (pcm_handle, hwparams, rate, 0), > > "snd_pcm_hw_params_set_rate"); > > CHECK_FAIL (snd_pcm_hw_params (pcm_handle, hwparams), > "snd_pcm_hw_params"); > > bitwidth = snd_pcm_format_physical_width(afmt); > > bps = (rate * bitwidth * nch) >> 3; > > buf_size = MAX(aud_cfg->output_buffer_size, 100); > > ringbuf_size = buf_size * bps / 1000; > > I increased the minimum period bytes value to 256 in that patch. > Audacious version 2 (Debian SID) > works for me. > well, audacious.i386 1.5.1-9.fc10 cannot play any music (alsa plugin) on my au8830 with your patch, it only play music with (oss plugin) Use xmms instead of audacious since they have similar UI 256 seem much bigger than the figures mentioned in the patent , may be it is the practical limit on your VIA ITX In one embodiment of the present invention, the FIFO buffer is 32 samples or words in depth, which would withstand a maximum bus latency of about 4 microseconds. By way of example, if 48 DMA channels trigger simultaneously, with a trigger level set at half the buffer size (16 samples), each DMA channel can wait 16 samples (20 microseconds per sample), or approximately 320 microseconds before running out of data. By way of example, CPU 210 may update-sub-buffer tables 380(0)-380(x) routinely every 10 milliseconds. The periodic checks may be initiated by audio processing peripheral 230 by way of interrupts, or may be scheduled independently by the driver > > How do you get the "ADB Ctrl failure" error ? > > I do not know how to reproduce that. Sometimes it happens, and its not > possible > to recover without a hardware reset (unload and load module). Somehow > it semes to > happen when my computer gets really hot (watching an MPEG-4 video, or > compiling the kernel...). > Its a old VIA ITX board, with a 1.3GHz processor (on demand cpufreq). > > The patch tries to reset the audio card if all PCM streams are > released once such a > condition was detected. This is the situation were the windows driver > used to crash the > whole machine. > it work quite well on my two motherbroad using intel chipset > > There are two main problem of the au88x0 driver > > 1) alsa is unable to calculate the correct value for the three > > functions snd_pcm_hw_params_set_buffer_time_near() , > > snd_pcm_hw_params_set_period_time_near() and snd_pcm_set_params() > > > > https://bugtrack.alsa-project.org/alsa-bug/view.php?id=1724 > > > > Test case : > > > > alsa-lib/test/pcm_min.c ( change default to hw:0,0 ) > > > > Real world case: > > > > gstreamer and winealsa.drv > > I will try that. I had no idea about that bugs. > A simple workaround is add another constraint snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 4096 ); however winealsa.drv will be using a larger period bytes than wineoss.drv > > > 2) unable to recover when the number of available src is not enough > > (FATAL: ResManager: resource type 1 exhaust) > > > > https://bugtrack.alsa-project.org/alsa-bug/view.php?id=2228 > > > > test case : > > > > running application to use 15 src ( 7 stereo and one mono stream ) and > > open another stereo stream > > Hmm, the good ol' resource manager :) OK, I will try to find a > solution. I have not much > time these days, but maybe it rains again, and I get bored, who knows :D > > Best Regards, > Manuel > simple fix may be adding counts of the resource and check that all resources are enough before allocating them

1 0

[alsa-devel] Duplicate configuration nodes
by Jochem Wichers Hoeth 07 Sep '09

07 Sep '09

Hi, Since I upgraded from alsa-lib 1.0.20 to 1.0.21 my configuration is not parsed correctly anymore: $ aplay test.wav ALSA lib conf.c:4639:(snd_config_expand) Expand error (walk): File exists Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono As a result of this error, ALSA seems to ignore the second half of HDA-Intel.conf, including the iec958 definition. I traced the "File exists" error back to snd_config_add() returning -EEXIST, while evaluating the "strings" node at line 36 of cards/HDA-Intel.conf. I verified that at that point the src node indeed has four children: two duplicates of each child node of the original strings array. I'm not exactly sure what's happening here, but I do see that the HDA-Intel.conf file (and all of its included files) are actually read *twice* by aplay. With alsa-lib 1.0.20 aplay only reads these files once. The problem seems to be related to commit 96da0c842d14b40ce8e37726b259229634b3aa21 ("Fix driver conf parsing in snd_config_hook_load_for_all_cards()"), which subtly changes the logic when the same driver is encountered for the second time. I have two cards using the HDA-Intel config (one analog and one digital). For the first card, the config is loaded without issues. But for the second card, snd_config_search(root, fdriver, &n) succeeds but snd_config_get_string(n, &driver) fails. In 1.0.20 this results in a "goto __err", skipping the actual config load. But in 1.0.21 this results in the else-branch being chosen (driver = fdriver) and next snd_config_hook_load being called. I know too little of the entire config loading process to be sure what exactly is going wrong here, but a assume that the second call to snd_config_hook_load for the same configaration file somehow eventually results in the duplicate node error in snd_config_add(). I've verified that reverting 96da0c842d14b40ce8e37726b259229634b3aa21 fixes the problem, or at least the symptoms. :-) Please let me know if you need extra information, of if you'd like me to do some more testing. Cheers, Jochem Wichers Hoeth

1 0

[alsa-devel] [PATCH] Minor WM8580 enhancements.
by jassi brar 07 Sep '09

07 Sep '09

Add more options to select clock source for various blocks in WM8580. Also, explicitly set defaults for the sake of consistency. Signed-off-by: Jassi <jassi.brar(a)samsung.com> --- sound/soc/codecs/wm8580.c | 133 +++++++++++++++++++++++++++++++++++++++++++-- sound/soc/codecs/wm8580.h | 25 +++++---- 2 files changed, 144 insertions(+), 14 deletions(-) diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 6bded8c..276bc8e 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -95,6 +95,7 @@ /* PLLB4 (register 7h) */ #define WM8580_PLLB4_MCLKOUTSRC_MASK 0x60 +#define WM8580_PLLB4_MCLKOUTSRC_MCLK 0x00 #define WM8580_PLLB4_MCLKOUTSRC_PLLA 0x20 #define WM8580_PLLB4_MCLKOUTSRC_PLLB 0x40 #define WM8580_PLLB4_MCLKOUTSRC_OSC 0x60 @@ -106,9 +107,16 @@ /* CLKSEL (register 8h) */ #define WM8580_CLKSEL_DAC_CLKSEL_MASK 0x03 +#define WM8580_CLKSEL_DAC_CLKSEL_MCLK 0x00 #define WM8580_CLKSEL_DAC_CLKSEL_PLLA 0x01 #define WM8580_CLKSEL_DAC_CLKSEL_PLLB 0x02 +#define WM8580_CLKSEL_ADC_CLKSEL_MASK 0x0c +#define WM8580_CLKSEL_ADC_CLKSEL_ADCMCLK 0x00 +#define WM8580_CLKSEL_ADC_CLKSEL_PLLA 0x04 +#define WM8580_CLKSEL_ADC_CLKSEL_PLLB 0x08 +#define WM8580_CLKSEL_ADC_CLKSEL_MCLK 0x0c + /* AIF control 1 (registers 9h-bh) */ #define WM8580_AIF_RATE_MASK 0x7 #define WM8580_AIF_RATE_128 0x0 @@ -121,15 +129,16 @@ #define WM8580_AIF_BCLKSEL_MASK 0x18 #define WM8580_AIF_BCLKSEL_64 0x00 -#define WM8580_AIF_BCLKSEL_128 0x08 -#define WM8580_AIF_BCLKSEL_256 0x10 +#define WM8580_AIF_BCLKSEL_32 0x08 +#define WM8580_AIF_BCLKSEL_16 0x10 #define WM8580_AIF_BCLKSEL_SYSCLK 0x18 #define WM8580_AIF_MS 0x20 #define WM8580_AIF_CLKSRC_MASK 0xc0 +#define WM8580_AIF_CLKSRC_ADCMCLK 0x00 #define WM8580_AIF_CLKSRC_PLLA 0x40 -#define WM8580_AIF_CLKSRC_PLLB 0x40 +#define WM8580_AIF_CLKSRC_PLLB 0x80 #define WM8580_AIF_CLKSRC_MCLK 0xc0 /* AIF control 2 (registers ch-eh) */ @@ -604,12 +613,13 @@ static int wm8580_set_dai_clkdiv(struct snd_soc_dai *codec_dai, switch (div) { case WM8580_CLKSRC_MCLK: - /* Input */ + reg |= WM8580_PLLB4_MCLKOUTSRC_MCLK; break; case WM8580_CLKSRC_PLLA: reg |= WM8580_PLLB4_MCLKOUTSRC_PLLA; break; + case WM8580_CLKSRC_PLLB: reg |= WM8580_PLLB4_MCLKOUTSRC_PLLB; break; @@ -630,6 +640,7 @@ static int wm8580_set_dai_clkdiv(struct snd_soc_dai *codec_dai, switch (div) { case WM8580_CLKSRC_MCLK: + reg |= WM8580_CLKSEL_DAC_CLKSEL_MCLK; break; case WM8580_CLKSRC_PLLA: @@ -646,6 +657,59 @@ static int wm8580_set_dai_clkdiv(struct snd_soc_dai *codec_dai, snd_soc_write(codec, WM8580_CLKSEL, reg); break; + case WM8580_ADC_CLKSEL: + reg = snd_soc_read(codec, WM8580_CLKSEL); + reg &= ~WM8580_CLKSEL_ADC_CLKSEL_MASK; + + switch (div) { + case WM8580_CLKSRC_ADCMCLK: + reg |= WM8580_CLKSEL_ADC_CLKSEL_ADCMCLK; + break; + + case WM8580_CLKSRC_MCLK: + reg |= WM8580_CLKSEL_ADC_CLKSEL_MCLK; + break; + + case WM8580_CLKSRC_PLLA: + reg |= WM8580_CLKSEL_ADC_CLKSEL_PLLA; + break; + + case WM8580_CLKSRC_PLLB: + reg |= WM8580_CLKSEL_ADC_CLKSEL_PLLB; + break; + + default: + return -EINVAL; + } + snd_soc_write(codec, WM8580_CLKSEL, reg); + break; + + case WM8580_PAIF_CLKSEL: + reg = snd_soc_read(codec, WM8580_PAIF1 + codec_dai->id); + reg &= ~WM8580_AIF_CLKSRC_MASK; + switch (div) { + case WM8580_CLKSRC_ADCMCLK: + reg |= WM8580_AIF_CLKSRC_ADCMCLK; + break; + + case WM8580_CLKSRC_PLLA: + reg |= WM8580_AIF_CLKSRC_PLLA; + break; + + case WM8580_CLKSRC_PLLB: + reg |= WM8580_AIF_CLKSRC_PLLB; + break; + + case WM8580_CLKSRC_MCLK: + reg |= WM8580_AIF_CLKSRC_MCLK; + break; + + default: + return -EINVAL; + } + snd_soc_write(codec, WM8580_PAIF1 + codec_dai->id, reg); + break; + case WM8580_CLKOUTSRC: reg = snd_soc_read(codec, WM8580_PLLB4); reg &= ~WM8580_PLLB4_CLKOUTSRC_MASK; @@ -672,6 +736,67 @@ static int wm8580_set_dai_clkdiv(struct snd_soc_dai *codec_dai, snd_soc_write(codec, WM8580_PLLB4, reg); break; + case WM8580_MCLKRATIO: + reg = snd_soc_read(codec, WM8580_PAIF1 + codec_dai->id); + reg &= ~WM8580_AIF_RATE_MASK; + switch(div) { + case 128: + reg |= WM8580_AIF_RATE_128; + break; + + case 192: + reg |= WM8580_AIF_RATE_192; + break; + + case 256: + reg |= WM8580_AIF_RATE_256; + break; + + case 384: + reg |= WM8580_AIF_RATE_384; + break; + + case 512: + reg |= WM8580_AIF_RATE_512; + break; + + case 768: + reg |= WM8580_AIF_RATE_768; + break; + + case 1152: + reg |= WM8580_AIF_RATE_1152; + break; + + default: + return -EINVAL; + } + snd_soc_write(codec, WM8580_PAIF1 + codec_dai->id, reg); + break; + + case WM8580_BCLKRATIO: + reg = snd_soc_read(codec, WM8580_PAIF1 + codec_dai->id); + reg &= ~WM8580_AIF_BCLKSEL_MASK; + switch(div) { + case 64: + reg |= WM8580_AIF_BCLKSEL_64; + break; + + case 32: + reg |= WM8580_AIF_BCLKSEL_32; + break; + + case 16: + reg |= WM8580_AIF_BCLKSEL_16; + break; + + default: + reg |= WM8580_AIF_BCLKSEL_SYSCLK; + break; + } + snd_soc_write(codec, WM8580_PAIF1 + codec_dai->id, reg); + break; + default: return -EINVAL; } diff --git a/sound/soc/codecs/wm8580.h b/sound/soc/codecs/wm8580.h index 0dfb5dd..0ecdbfa 100644 --- a/sound/soc/codecs/wm8580.h +++ b/sound/soc/codecs/wm8580.h @@ -18,16 +18,21 @@ #define WM8580_PLLA 1 #define WM8580_PLLB 2 -#define WM8580_MCLK 1 -#define WM8580_DAC_CLKSEL 2 -#define WM8580_CLKOUTSRC 3 - -#define WM8580_CLKSRC_MCLK 1 -#define WM8580_CLKSRC_PLLA 2 -#define WM8580_CLKSRC_PLLB 3 -#define WM8580_CLKSRC_OSC 4 -#define WM8580_CLKSRC_NONE 5 - +#define WM8580_MCLK 1 +#define WM8580_DAC_CLKSEL 2 +#define WM8580_ADC_CLKSEL 3 +#define WM8580_PAIF_CLKSEL 4 +#define WM8580_CLKOUTSRC 5 +#define WM8580_MCLKRATIO 6 +#define WM8580_BCLKRATIO 7 + +#define WM8580_CLKSRC_ADCMCLK 0 +#define WM8580_CLKSRC_MCLK 1 +#define WM8580_CLKSRC_PLLA 2 +#define WM8580_CLKSRC_PLLB 3 +#define WM8580_CLKSRC_OSC 4 +#define WM8580_CLKSRC_NONE 5 + #define WM8580_DAI_PAIFRX 0 #define WM8580_DAI_PAIFTX 1 -- 1.6.2.5

2 8

[alsa-devel] [PATCH] ASoC: Factor out I2C 8 bit address 8 bit data I/O
by Joonyoung Shim 07 Sep '09

07 Sep '09

This patch is for the AK4671 codec driver using this format. Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- sound/soc/soc-cache.c | 30 ++++++++++++++++++++++++++++++ 1 files changed, 30 insertions(+), 0 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index c8ceddc..404231e 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -77,6 +77,35 @@ static int snd_soc_7_9_spi_write(void *control_data, const char *data, #define snd_soc_7_9_spi_write NULL #endif +static int snd_soc_8_8_write(struct snd_soc_codec *codec, unsigned int reg, + unsigned int value) +{ + u8 *cache = codec->reg_cache; + u8 data[2]; + + BUG_ON(codec->volatile_register); + + data[0] = reg & 0xff; + data[1] = value & 0xff; + + if (reg < codec->reg_cache_size) + cache[reg] = value; + + if (codec->hw_write(codec->control_data, data, 2) == 2) + return 0; + else + return -EIO; +} + +static unsigned int snd_soc_8_8_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 *cache = codec->reg_cache; + if (reg >= codec->reg_cache_size) + return -1; + return cache[reg]; +} + static int snd_soc_8_16_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int value) { @@ -151,6 +180,7 @@ static struct { unsigned int (*i2c_read)(struct snd_soc_codec *, unsigned int); } io_types[] = { { 7, 9, snd_soc_7_9_write, snd_soc_7_9_spi_write, snd_soc_7_9_read }, + { 8, 8, snd_soc_8_8_write, NULL, snd_soc_8_8_read, NULL }, { 8, 16, snd_soc_8_16_write, NULL, snd_soc_8_16_read, snd_soc_8_16_read_i2c }, }; -- 1.6.0.4

3 5

[alsa-devel] [PATCH v2] ASoC: AK4671: add ak4671 codec driver
by Joonyoung Shim 07 Sep '09

07 Sep '09

The AK4671 is a stereo CODEC with a built-in Microphone-Amplifier, Receiver-Amplifier and Headphone-Amplifier. The datasheet for the ak4671 can find at the following url: http://www.asahi-kasei.co.jp/akm/en/product/ak4671/ak4671_f01e.pdf Signed-off-by: Joonyoung Shim <jy0922.shim(a)samsung.com> --- Change log: v2: - use soc-cache - remove unused pll_on variable - add event handler for DAC and ADC - remove unnecessary comments and PM functions - modify error handling in ak4671_register function sound/soc/codecs/Kconfig | 4 + sound/soc/codecs/Makefile | 2 + sound/soc/codecs/ak4671.c | 845 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/ak4671.h | 156 +++++++++ 4 files changed, 1007 insertions(+), 0 deletions(-) create mode 100644 sound/soc/codecs/ak4671.c create mode 100644 sound/soc/codecs/ak4671.h diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig index 0edca93..a2bb659 100644 --- a/sound/soc/codecs/Kconfig +++ b/sound/soc/codecs/Kconfig @@ -19,6 +19,7 @@ config SND_SOC_ALL_CODECS select SND_SOC_AK4104 if SPI_MASTER select SND_SOC_AK4535 if I2C select SND_SOC_AK4642 if I2C + select SND_SOC_AK4671 if I2C select SND_SOC_CS4270 if I2C select SND_SOC_MAX9877 if I2C select SND_SOC_PCM3008 @@ -96,6 +97,9 @@ config SND_SOC_AK4535 config SND_SOC_AK4642 tristate +config SND_SOC_AK4671 + tristate + # Cirrus Logic CS4270 Codec config SND_SOC_CS4270 tristate diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile index fb4af28..13f7b4f 100644 --- a/sound/soc/codecs/Makefile +++ b/sound/soc/codecs/Makefile @@ -6,6 +6,7 @@ snd-soc-ad73311-objs := ad73311.o snd-soc-ak4104-objs := ak4104.o snd-soc-ak4535-objs := ak4535.o snd-soc-ak4642-objs := ak4642.o +snd-soc-ak4671-objs := ak4671.o snd-soc-cs4270-objs := cs4270.o snd-soc-cx20442-objs := cx20442.o snd-soc-l3-objs := l3.o @@ -56,6 +57,7 @@ obj-$(CONFIG_SND_SOC_AD73311) += snd-soc-ad73311.o obj-$(CONFIG_SND_SOC_AK4104) += snd-soc-ak4104.o obj-$(CONFIG_SND_SOC_AK4535) += snd-soc-ak4535.o obj-$(CONFIG_SND_SOC_AK4642) += snd-soc-ak4642.o +obj-$(CONFIG_SND_SOC_AK4671) += snd-soc-ak4671.o obj-$(CONFIG_SND_SOC_CS4270) += snd-soc-cs4270.o obj-$(CONFIG_SND_SOC_CX20442) += snd-soc-cx20442.o obj-$(CONFIG_SND_SOC_L3) += snd-soc-l3.o diff --git a/sound/soc/codecs/ak4671.c b/sound/soc/codecs/ak4671.c new file mode 100644 index 0000000..cb6735f --- /dev/null +++ b/sound/soc/codecs/ak4671.c @@ -0,0 +1,845 @@ +/* + * ak4671.c -- audio driver for AK4671 + * + * Copyright (C) 2009 Samsung Electronics Co.Ltd + * Author: Joonyoung Shim <jy0922.shim(a)samsung.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#include <linux/module.h> +#include <linux/init.h> +#include <linux/i2c.h> +#include <linux/delay.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> +#include <sound/initval.h> +#include <sound/tlv.h> + +#include "ak4671.h" + +static struct snd_soc_codec *ak4671_codec; + +/* codec private data */ +struct ak4671_priv { + struct snd_soc_codec codec; + u8 reg_cache[AK4671_CACHEREGNUM]; +}; + +/* ak4671 register cache & default register settings */ +static const u8 ak4671_reg[AK4671_CACHEREGNUM] = { + 0x00, /* AK4671_AD_DA_POWER_MANAGEMENT (0x00) */ + 0xf6, /* AK4671_PLL_MODE_SELECT0 (0x01) */ + 0x00, /* AK4671_PLL_MODE_SELECT1 (0x02) */ + 0x02, /* AK4671_FORMAT_SELECT (0x03) */ + 0x00, /* AK4671_MIC_SIGNAL_SELECT (0x04) */ + 0x55, /* AK4671_MIC_AMP_GAIN (0x05) */ + 0x00, /* AK4671_MIXING_POWER_MANAGEMENT0 (0x06) */ + 0x00, /* AK4671_MIXING_POWER_MANAGEMENT1 (0x07) */ + 0xb5, /* AK4671_OUTPUT_VOLUME_CONTROL (0x08) */ + 0x00, /* AK4671_LOUT1_SIGNAL_SELECT (0x09) */ + 0x00, /* AK4671_ROUT1_SIGNAL_SELECT (0x0a) */ + 0x00, /* AK4671_LOUT2_SIGNAL_SELECT (0x0b) */ + 0x00, /* AK4671_ROUT2_SIGNAL_SELECT (0x0c) */ + 0x00, /* AK4671_LOUT3_SIGNAL_SELECT (0x0d) */ + 0x00, /* AK4671_ROUT3_SIGNAL_SELECT (0x0e) */ + 0x00, /* AK4671_LOUT1_POWER_MANAGERMENT (0x0f) */ + 0x00, /* AK4671_LOUT2_POWER_MANAGERMENT (0x10) */ + 0x80, /* AK4671_LOUT3_POWER_MANAGERMENT (0x11) */ + 0x91, /* AK4671_LCH_INPUT_VOLUME_CONTROL (0x12) */ + 0x91, /* AK4671_RCH_INPUT_VOLUME_CONTROL (0x13) */ + 0xe1, /* AK4671_ALC_REFERENCE_SELECT (0x14) */ + 0x00, /* AK4671_DIGITAL_MIXING_CONTROL (0x15) */ + 0x00, /* AK4671_ALC_TIMER_SELECT (0x16) */ + 0x00, /* AK4671_ALC_MODE_CONTROL (0x17) */ + 0x02, /* AK4671_MODE_CONTROL1 (0x18) */ + 0x01, /* AK4671_MODE_CONTROL2 (0x19) */ + 0x18, /* AK4671_LCH_OUTPUT_VOLUME_CONTROL (0x1a) */ + 0x18, /* AK4671_RCH_OUTPUT_VOLUME_CONTROL (0x1b) */ + 0x00, /* AK4671_SIDETONE_A_CONTROL (0x1c) */ + 0x02, /* AK4671_DIGITAL_FILTER_SELECT (0x1d) */ + 0x00, /* AK4671_FIL3_COEFFICIENT0 (0x1e) */ + 0x00, /* AK4671_FIL3_COEFFICIENT1 (0x1f) */ + 0x00, /* AK4671_FIL3_COEFFICIENT2 (0x20) */ + 0x00, /* AK4671_FIL3_COEFFICIENT3 (0x21) */ + 0x00, /* AK4671_EQ_COEFFICIENT0 (0x22) */ + 0x00, /* AK4671_EQ_COEFFICIENT1 (0x23) */ + 0x00, /* AK4671_EQ_COEFFICIENT2 (0x24) */ + 0x00, /* AK4671_EQ_COEFFICIENT3 (0x25) */ + 0x00, /* AK4671_EQ_COEFFICIENT4 (0x26) */ + 0x00, /* AK4671_EQ_COEFFICIENT5 (0x27) */ + 0xa9, /* AK4671_FIL1_COEFFICIENT0 (0x28) */ + 0x1f, /* AK4671_FIL1_COEFFICIENT1 (0x29) */ + 0xad, /* AK4671_FIL1_COEFFICIENT2 (0x2a) */ + 0x20, /* AK4671_FIL1_COEFFICIENT3 (0x2b) */ + 0x00, /* AK4671_FIL2_COEFFICIENT0 (0x2c) */ + 0x00, /* AK4671_FIL2_COEFFICIENT1 (0x2d) */ + 0x00, /* AK4671_FIL2_COEFFICIENT2 (0x2e) */ + 0x00, /* AK4671_FIL2_COEFFICIENT3 (0x2f) */ + 0x00, /* AK4671_DIGITAL_FILTER_SELECT2 (0x30) */ + 0x00, /* this register not used */ + 0x00, /* AK4671_E1_COEFFICIENT0 (0x32) */ + 0x00, /* AK4671_E1_COEFFICIENT1 (0x33) */ + 0x00, /* AK4671_E1_COEFFICIENT2 (0x34) */ + 0x00, /* AK4671_E1_COEFFICIENT3 (0x35) */ + 0x00, /* AK4671_E1_COEFFICIENT4 (0x36) */ + 0x00, /* AK4671_E1_COEFFICIENT5 (0x37) */ + 0x00, /* AK4671_E2_COEFFICIENT0 (0x38) */ + 0x00, /* AK4671_E2_COEFFICIENT1 (0x39) */ + 0x00, /* AK4671_E2_COEFFICIENT2 (0x3a) */ + 0x00, /* AK4671_E2_COEFFICIENT3 (0x3b) */ + 0x00, /* AK4671_E2_COEFFICIENT4 (0x3c) */ + 0x00, /* AK4671_E2_COEFFICIENT5 (0x3d) */ + 0x00, /* AK4671_E3_COEFFICIENT0 (0x3e) */ + 0x00, /* AK4671_E3_COEFFICIENT1 (0x3f) */ + 0x00, /* AK4671_E3_COEFFICIENT2 (0x40) */ + 0x00, /* AK4671_E3_COEFFICIENT3 (0x41) */ + 0x00, /* AK4671_E3_COEFFICIENT4 (0x42) */ + 0x00, /* AK4671_E3_COEFFICIENT5 (0x43) */ + 0x00, /* AK4671_E4_COEFFICIENT0 (0x44) */ + 0x00, /* AK4671_E4_COEFFICIENT1 (0x45) */ + 0x00, /* AK4671_E4_COEFFICIENT2 (0x46) */ + 0x00, /* AK4671_E4_COEFFICIENT3 (0x47) */ + 0x00, /* AK4671_E4_COEFFICIENT4 (0x48) */ + 0x00, /* AK4671_E4_COEFFICIENT5 (0x49) */ + 0x00, /* AK4671_E5_COEFFICIENT0 (0x4a) */ + 0x00, /* AK4671_E5_COEFFICIENT1 (0x4b) */ + 0x00, /* AK4671_E5_COEFFICIENT2 (0x4c) */ + 0x00, /* AK4671_E5_COEFFICIENT3 (0x4d) */ + 0x00, /* AK4671_E5_COEFFICIENT4 (0x4e) */ + 0x00, /* AK4671_E5_COEFFICIENT5 (0x4f) */ + 0x88, /* AK4671_EQ_CONTROL_250HZ_100HZ (0x50) */ + 0x88, /* AK4671_EQ_CONTROL_3500HZ_1KHZ (0x51) */ + 0x08, /* AK4671_EQ_CONTRO_10KHZ (0x52) */ + 0x00, /* AK4671_PCM_IF_CONTROL0 (0x53) */ + 0x00, /* AK4671_PCM_IF_CONTROL1 (0x54) */ + 0x00, /* AK4671_PCM_IF_CONTROL2 (0x55) */ + 0x18, /* AK4671_DIGITAL_VOLUME_B_CONTROL (0x56) */ + 0x18, /* AK4671_DIGITAL_VOLUME_C_CONTROL (0x57) */ + 0x00, /* AK4671_SIDETONE_VOLUME_CONTROL (0x58) */ + 0x00, /* AK4671_DIGITAL_MIXING_CONTROL2 (0x59) */ + 0x00, /* AK4671_SAR_ADC_CONTROL (0x5a) */ +}; + +/* + * LOUT1/ROUT1 output volume control: + * from -24 to 6 dB in 6 dB steps (mute instead of -30 dB) + */ +static DECLARE_TLV_DB_SCALE(out1_tlv, -3000, 600, 1); + +/* + * LOUT2/ROUT2 output volume control: + * from -33 to 6 dB in 3 dB steps (mute instead of -33 dB) + */ +static DECLARE_TLV_DB_SCALE(out2_tlv, -3300, 300, 1); + +/* + * LOUT3/ROUT3 output volume control: + * from -6 to 3 dB in 3 dB steps + */ +static DECLARE_TLV_DB_SCALE(out3_tlv, -600, 300, 0); + +/* + * Mic amp gain control: + * from -15 to 30 dB in 3 dB steps + * REVISIT: The actual min value(0x01) is -12 dB and the reg value 0x00 is not + * available + */ +static DECLARE_TLV_DB_SCALE(mic_amp_tlv, -1500, 300, 0); + +static const struct snd_kcontrol_new ak4671_snd_controls[] = { + /* Common playback gain controls */ + SOC_SINGLE_TLV("Line Output1 Playback Volume", + AK4671_OUTPUT_VOLUME_CONTROL, 0, 0x6, 0, out1_tlv), + SOC_SINGLE_TLV("Headphone Output2 Playback Volume", + AK4671_OUTPUT_VOLUME_CONTROL, 4, 0xd, 0, out2_tlv), + SOC_SINGLE_TLV("Line Output3 Playback Volume", + AK4671_LOUT3_POWER_MANAGERMENT, 6, 0x3, 0, out3_tlv), + + /* Common capture gain controls */ + SOC_DOUBLE_TLV("Mic Amp Capture Volume", + AK4671_MIC_AMP_GAIN, 0, 4, 0xf, 0, mic_amp_tlv), +}; + +/* event handlers */ +static int ak4671_out2_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + u8 reg; + + switch (event) { + case SND_SOC_DAPM_POST_PMU: + reg = snd_soc_read(codec, AK4671_LOUT2_POWER_MANAGERMENT); + reg |= AK4671_MUTEN; + snd_soc_write(codec, AK4671_LOUT2_POWER_MANAGERMENT, reg); + break; + case SND_SOC_DAPM_PRE_PMD: + reg = snd_soc_read(codec, AK4671_LOUT2_POWER_MANAGERMENT); + reg &= ~AK4671_MUTEN; + snd_soc_write(codec, AK4671_LOUT2_POWER_MANAGERMENT, reg); + break; + } + + return 0; +} + +static int ak4671_pll_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct snd_soc_codec *codec = w->codec; + u8 reg; + + switch (event) { + case SND_SOC_DAPM_PRE_PMU: + reg = snd_soc_read(codec, AK4671_PLL_MODE_SELECT1); + reg |= AK4671_PMPLL; + snd_soc_write(codec, AK4671_PLL_MODE_SELECT1, reg); + /* pll lock time: max 40ms */ + mdelay(40); + break; + case SND_SOC_DAPM_POST_PMD: + reg = snd_soc_read(codec, AK4671_PLL_MODE_SELECT1); + reg &= ~AK4671_PMPLL; + snd_soc_write(codec, AK4671_PLL_MODE_SELECT1, reg); + break; + } + + return 0; +} + +/* Output Mixers */ +static const struct snd_kcontrol_new ak4671_lout1_mixer_controls[] = { + SOC_DAPM_SINGLE("DACL", AK4671_LOUT1_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("LINL1", AK4671_LOUT1_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("LINL2", AK4671_LOUT1_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("LINL3", AK4671_LOUT1_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("LINL4", AK4671_LOUT1_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPL", AK4671_LOUT1_SIGNAL_SELECT, 5, 1, 0), +}; + +static const struct snd_kcontrol_new ak4671_rout1_mixer_controls[] = { + SOC_DAPM_SINGLE("DACR", AK4671_ROUT1_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("RINR1", AK4671_ROUT1_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("RINR2", AK4671_ROUT1_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("RINR3", AK4671_ROUT1_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("RINR4", AK4671_ROUT1_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPR", AK4671_ROUT1_SIGNAL_SELECT, 5, 1, 0), +}; + +static const struct snd_kcontrol_new ak4671_lout2_mixer_controls[] = { + SOC_DAPM_SINGLE("DACHL", AK4671_LOUT2_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("LINH1", AK4671_LOUT2_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("LINH2", AK4671_LOUT2_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("LINH3", AK4671_LOUT2_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("LINH4", AK4671_LOUT2_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPHL", AK4671_LOUT2_SIGNAL_SELECT, 5, 1, 0), +}; + +static const struct snd_kcontrol_new ak4671_rout2_mixer_controls[] = { + SOC_DAPM_SINGLE("DACHR", AK4671_ROUT2_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("RINH1", AK4671_ROUT2_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("RINH2", AK4671_ROUT2_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("RINH3", AK4671_ROUT2_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("RINH4", AK4671_ROUT2_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPHR", AK4671_ROUT2_SIGNAL_SELECT, 5, 1, 0), +}; + +static const struct snd_kcontrol_new ak4671_lout3_mixer_controls[] = { + SOC_DAPM_SINGLE("DACSL", AK4671_LOUT3_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("LINS1", AK4671_LOUT3_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("LINS2", AK4671_LOUT3_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("LINS3", AK4671_LOUT3_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("LINS4", AK4671_LOUT3_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPSL", AK4671_LOUT3_SIGNAL_SELECT, 5, 1, 0), +}; + +static const struct snd_kcontrol_new ak4671_rout3_mixer_controls[] = { + SOC_DAPM_SINGLE("DACSR", AK4671_ROUT3_SIGNAL_SELECT, 0, 1, 0), + SOC_DAPM_SINGLE("RINS1", AK4671_ROUT3_SIGNAL_SELECT, 1, 1, 0), + SOC_DAPM_SINGLE("RINS2", AK4671_ROUT3_SIGNAL_SELECT, 2, 1, 0), + SOC_DAPM_SINGLE("RINS3", AK4671_ROUT3_SIGNAL_SELECT, 3, 1, 0), + SOC_DAPM_SINGLE("RINS4", AK4671_ROUT3_SIGNAL_SELECT, 4, 1, 0), + SOC_DAPM_SINGLE("LOOPSR", AK4671_ROUT3_SIGNAL_SELECT, 5, 1, 0), +}; + +/* Input MUXs */ +static const char *ak4671_lin_mux_texts[] = + {"LIN1", "LIN2", "LIN3", "LIN4"}; +static const struct soc_enum ak4671_lin_mux_enum = + SOC_ENUM_SINGLE(AK4671_MIC_SIGNAL_SELECT, 0, + ARRAY_SIZE(ak4671_lin_mux_texts), + ak4671_lin_mux_texts); +static const struct snd_kcontrol_new ak4671_lin_mux_control = + SOC_DAPM_ENUM("Route", ak4671_lin_mux_enum); + +static const char *ak4671_rin_mux_texts[] = + {"RIN1", "RIN2", "RIN3", "RIN4"}; +static const struct soc_enum ak4671_rin_mux_enum = + SOC_ENUM_SINGLE(AK4671_MIC_SIGNAL_SELECT, 2, + ARRAY_SIZE(ak4671_rin_mux_texts), + ak4671_rin_mux_texts); +static const struct snd_kcontrol_new ak4671_rin_mux_control = + SOC_DAPM_ENUM("Route", ak4671_rin_mux_enum); + +static const struct snd_soc_dapm_widget ak4671_dapm_widgets[] = { + /* Inputs */ + SND_SOC_DAPM_INPUT("LIN1"), + SND_SOC_DAPM_INPUT("RIN1"), + SND_SOC_DAPM_INPUT("LIN2"), + SND_SOC_DAPM_INPUT("RIN2"), + SND_SOC_DAPM_INPUT("LIN3"), + SND_SOC_DAPM_INPUT("RIN3"), + SND_SOC_DAPM_INPUT("LIN4"), + SND_SOC_DAPM_INPUT("RIN4"), + + /* Outputs */ + SND_SOC_DAPM_OUTPUT("LOUT1"), + SND_SOC_DAPM_OUTPUT("ROUT1"), + SND_SOC_DAPM_OUTPUT("LOUT2"), + SND_SOC_DAPM_OUTPUT("ROUT2"), + SND_SOC_DAPM_OUTPUT("LOUT3"), + SND_SOC_DAPM_OUTPUT("ROUT3"), + + /* DAC */ + SND_SOC_DAPM_DAC_E("DAC Left", "Left HiFi Playback", + AK4671_AD_DA_POWER_MANAGEMENT, 6, 0, ak4671_pll_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_DAC_E("DAC Right", "Right HiFi Playback", + AK4671_AD_DA_POWER_MANAGEMENT, 7, 0, ak4671_pll_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + /* ADC */ + SND_SOC_DAPM_ADC_E("ADC Left", "Left HiFi Capture", + AK4671_AD_DA_POWER_MANAGEMENT, 4, 0, ak4671_pll_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_ADC_E("ADC Right", "Right HiFi Capture", + AK4671_AD_DA_POWER_MANAGEMENT, 5, 0, ak4671_pll_event, + SND_SOC_DAPM_PRE_PMU | SND_SOC_DAPM_POST_PMD), + + /* PGA */ + SND_SOC_DAPM_PGA("LOUT2 Mix Amp", + AK4671_LOUT2_POWER_MANAGERMENT, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("ROUT2 Mix Amp", + AK4671_LOUT2_POWER_MANAGERMENT, 6, 0, NULL, 0), + + SND_SOC_DAPM_PGA("LIN1 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 0, 0, NULL, 0), + SND_SOC_DAPM_PGA("RIN1 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 1, 0, NULL, 0), + SND_SOC_DAPM_PGA("LIN2 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 2, 0, NULL, 0), + SND_SOC_DAPM_PGA("RIN2 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 3, 0, NULL, 0), + SND_SOC_DAPM_PGA("LIN3 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 4, 0, NULL, 0), + SND_SOC_DAPM_PGA("RIN3 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 5, 0, NULL, 0), + SND_SOC_DAPM_PGA("LIN4 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 6, 0, NULL, 0), + SND_SOC_DAPM_PGA("RIN4 Mixing Circuit", + AK4671_MIXING_POWER_MANAGEMENT1, 7, 0, NULL, 0), + + /* Output Mixers */ + SND_SOC_DAPM_MIXER("LOUT1 Mixer", AK4671_LOUT1_POWER_MANAGERMENT, 0, 0, + &ak4671_lout1_mixer_controls[0], + ARRAY_SIZE(ak4671_lout1_mixer_controls)), + SND_SOC_DAPM_MIXER("ROUT1 Mixer", AK4671_LOUT1_POWER_MANAGERMENT, 1, 0, + &ak4671_rout1_mixer_controls[0], + ARRAY_SIZE(ak4671_rout1_mixer_controls)), + SND_SOC_DAPM_MIXER_E("LOUT2 Mixer", AK4671_LOUT2_POWER_MANAGERMENT, + 0, 0, &ak4671_lout2_mixer_controls[0], + ARRAY_SIZE(ak4671_lout2_mixer_controls), + ak4671_out2_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_MIXER_E("ROUT2 Mixer", AK4671_LOUT2_POWER_MANAGERMENT, + 1, 0, &ak4671_rout2_mixer_controls[0], + ARRAY_SIZE(ak4671_rout2_mixer_controls), + ak4671_out2_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_MIXER("LOUT3 Mixer", AK4671_LOUT3_POWER_MANAGERMENT, 0, 0, + &ak4671_lout3_mixer_controls[0], + ARRAY_SIZE(ak4671_lout3_mixer_controls)), + SND_SOC_DAPM_MIXER("ROUT3 Mixer", AK4671_LOUT3_POWER_MANAGERMENT, 1, 0, + &ak4671_rout3_mixer_controls[0], + ARRAY_SIZE(ak4671_rout3_mixer_controls)), + + /* Input MUXs */ + SND_SOC_DAPM_MUX("LIN MUX", AK4671_AD_DA_POWER_MANAGEMENT, 2, 0, + &ak4671_lin_mux_control), + SND_SOC_DAPM_MUX("RIN MUX", AK4671_AD_DA_POWER_MANAGEMENT, 3, 0, + &ak4671_rin_mux_control), + + /* Mic Power */ + SND_SOC_DAPM_MICBIAS("Mic Bias", AK4671_AD_DA_POWER_MANAGEMENT, 1, 0), +}; + +static const struct snd_soc_dapm_route intercon[] = { + /* Outputs */ + {"LOUT1", "NULL", "LOUT1 Mixer"}, + {"ROUT1", "NULL", "ROUT1 Mixer"}, + {"LOUT2", "NULL", "LOUT2 Mix Amp"}, + {"ROUT2", "NULL", "ROUT2 Mix Amp"}, + {"LOUT3", "NULL", "LOUT3 Mixer"}, + {"ROUT3", "NULL", "ROUT3 Mixer"}, + + {"LOUT1 Mixer", "DACL", "DAC Left"}, + {"ROUT1 Mixer", "DACR", "DAC Right"}, + {"LOUT2 Mixer", "DACHL", "DAC Left"}, + {"ROUT2 Mixer", "DACHR", "DAC Right"}, + {"LOUT2 Mix Amp", "NULL", "LOUT2 Mixer"}, + {"ROUT2 Mix Amp", "NULL", "ROUT2 Mixer"}, + {"LOUT3 Mixer", "DACSL", "DAC Left"}, + {"ROUT3 Mixer", "DACSR", "DAC Right"}, + + /* Inputs */ + {"LIN MUX", "LIN1", "LIN1"}, + {"LIN MUX", "LIN2", "LIN2"}, + {"LIN MUX", "LIN3", "LIN3"}, + {"LIN MUX", "LIN4", "LIN4"}, + + {"RIN MUX", "RIN1", "RIN1"}, + {"RIN MUX", "RIN2", "RIN2"}, + {"RIN MUX", "RIN3", "RIN3"}, + {"RIN MUX", "RIN4", "RIN4"}, + + {"LIN1", NULL, "Mic Bias"}, + {"RIN1", NULL, "Mic Bias"}, + {"LIN2", NULL, "Mic Bias"}, + {"RIN2", NULL, "Mic Bias"}, + + {"ADC Left", "NULL", "LIN MUX"}, + {"ADC Right", "NULL", "RIN MUX"}, + + /* Analog Loops */ + {"LIN1 Mixing Circuit", "NULL", "LIN1"}, + {"RIN1 Mixing Circuit", "NULL", "RIN1"}, + {"LIN2 Mixing Circuit", "NULL", "LIN2"}, + {"RIN2 Mixing Circuit", "NULL", "RIN2"}, + {"LIN3 Mixing Circuit", "NULL", "LIN3"}, + {"RIN3 Mixing Circuit", "NULL", "RIN3"}, + {"LIN4 Mixing Circuit", "NULL", "LIN4"}, + {"RIN4 Mixing Circuit", "NULL", "RIN4"}, + + {"LOUT1 Mixer", "LINL1", "LIN1 Mixing Circuit"}, + {"ROUT1 Mixer", "RINR1", "RIN1 Mixing Circuit"}, + {"LOUT2 Mixer", "LINH1", "LIN1 Mixing Circuit"}, + {"ROUT2 Mixer", "RINH1", "RIN1 Mixing Circuit"}, + {"LOUT3 Mixer", "LINS1", "LIN1 Mixing Circuit"}, + {"ROUT3 Mixer", "RINS1", "RIN1 Mixing Circuit"}, + + {"LOUT1 Mixer", "LINL2", "LIN2 Mixing Circuit"}, + {"ROUT1 Mixer", "RINR2", "RIN2 Mixing Circuit"}, + {"LOUT2 Mixer", "LINH2", "LIN2 Mixing Circuit"}, + {"ROUT2 Mixer", "RINH2", "RIN2 Mixing Circuit"}, + {"LOUT3 Mixer", "LINS2", "LIN2 Mixing Circuit"}, + {"ROUT3 Mixer", "RINS2", "RIN2 Mixing Circuit"}, + + {"LOUT1 Mixer", "LINL3", "LIN3 Mixing Circuit"}, + {"ROUT1 Mixer", "RINR3", "RIN3 Mixing Circuit"}, + {"LOUT2 Mixer", "LINH3", "LIN3 Mixing Circuit"}, + {"ROUT2 Mixer", "RINH3", "RIN3 Mixing Circuit"}, + {"LOUT3 Mixer", "LINS3", "LIN3 Mixing Circuit"}, + {"ROUT3 Mixer", "RINS3", "RIN3 Mixing Circuit"}, + + {"LOUT1 Mixer", "LINL4", "LIN4 Mixing Circuit"}, + {"ROUT1 Mixer", "RINR4", "RIN4 Mixing Circuit"}, + {"LOUT2 Mixer", "LINH4", "LIN4 Mixing Circuit"}, + {"ROUT2 Mixer", "RINH4", "RIN4 Mixing Circuit"}, + {"LOUT3 Mixer", "LINS4", "LIN4 Mixing Circuit"}, + {"ROUT3 Mixer", "RINS4", "RIN4 Mixing Circuit"}, +}; + +static int ak4671_add_widgets(struct snd_soc_codec *codec) +{ + snd_soc_dapm_new_controls(codec, ak4671_dapm_widgets, + ARRAY_SIZE(ak4671_dapm_widgets)); + + snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon)); + + snd_soc_dapm_new_widgets(codec); + return 0; +} + +static int ak4671_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *dai) +{ + struct snd_soc_codec *codec = dai->codec; + u8 fs; + + fs = snd_soc_read(codec, AK4671_PLL_MODE_SELECT0); + fs &= ~AK4671_FS; + + switch (params_rate(params)) { + case 8000: + fs |= AK4671_FS_8KHZ; + break; + case 12000: + fs |= AK4671_FS_12KHZ; + break; + case 16000: + fs |= AK4671_FS_16KHZ; + break; + case 24000: + fs |= AK4671_FS_24KHZ; + break; + case 11025: + fs |= AK4671_FS_11_025KHZ; + break; + case 22050: + fs |= AK4671_FS_22_05KHZ; + break; + case 32000: + fs |= AK4671_FS_32KHZ; + break; + case 44100: + fs |= AK4671_FS_44_1KHZ; + break; + case 48000: + fs |= AK4671_FS_48KHZ; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, AK4671_PLL_MODE_SELECT0, fs); + + return 0; +} + +static int ak4671_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + u8 pll; + + pll = snd_soc_read(codec, AK4671_PLL_MODE_SELECT0); + pll &= ~AK4671_PLL; + + switch (freq) { + case 11289600: + pll |= AK4671_PLL_11_2896MHZ; + break; + case 12000000: + pll |= AK4671_PLL_12MHZ; + break; + case 12288000: + pll |= AK4671_PLL_12_288MHZ; + break; + case 13000000: + pll |= AK4671_PLL_13MHZ; + break; + case 13500000: + pll |= AK4671_PLL_13_5MHZ; + break; + case 19200000: + pll |= AK4671_PLL_19_2MHZ; + break; + case 24000000: + pll |= AK4671_PLL_24MHZ; + break; + case 26000000: + pll |= AK4671_PLL_26MHZ; + break; + case 27000000: + pll |= AK4671_PLL_27MHZ; + break; + default: + return -EINVAL; + } + + snd_soc_write(codec, AK4671_PLL_MODE_SELECT0, pll); + + return 0; +} + +static int ak4671_set_dai_fmt(struct snd_soc_dai *dai, unsigned int fmt) +{ + struct snd_soc_codec *codec = dai->codec; + u8 mode; + u8 format; + + /* set master/slave audio interface */ + mode = snd_soc_read(codec, AK4671_PLL_MODE_SELECT1); + + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBM_CFM: + mode |= AK4671_M_S; + break; + case SND_SOC_DAIFMT_CBM_CFS: + mode &= ~(AK4671_M_S); + break; + default: + return -EINVAL; + } + + /* interface format */ + format = snd_soc_read(codec, AK4671_FORMAT_SELECT); + format &= ~AK4671_DIF; + + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + format |= AK4671_DIF_I2S_MODE; + break; + case SND_SOC_DAIFMT_LEFT_J: + format |= AK4671_DIF_MSB_MODE; + break; + case SND_SOC_DAIFMT_DSP_A: + format |= AK4671_DIF_DSP_MODE; + format |= AK4671_BCKP; + format |= AK4671_MSBS; + break; + default: + return -EINVAL; + } + + /* set mode and format */ + snd_soc_write(codec, AK4671_PLL_MODE_SELECT1, mode); + snd_soc_write(codec, AK4671_FORMAT_SELECT, format); + + return 0; +} + +static int ak4671_set_bias_level(struct snd_soc_codec *codec, + enum snd_soc_bias_level level) +{ + u8 reg; + + switch (level) { + case SND_SOC_BIAS_ON: + case SND_SOC_BIAS_PREPARE: + case SND_SOC_BIAS_STANDBY: + reg = snd_soc_read(codec, AK4671_AD_DA_POWER_MANAGEMENT); + snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT, + reg | AK4671_PMVCM); + break; + case SND_SOC_BIAS_OFF: + snd_soc_write(codec, AK4671_AD_DA_POWER_MANAGEMENT, 0x00); + break; + } + codec->bias_level = level; + return 0; +} + +#define AK4671_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ + SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\ + SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\ + SNDRV_PCM_RATE_48000) + +#define AK4671_FORMATS SNDRV_PCM_FMTBIT_S16_LE + +static struct snd_soc_dai_ops ak4671_dai_ops = { + .hw_params = ak4671_hw_params, + .set_sysclk = ak4671_set_dai_sysclk, + .set_fmt = ak4671_set_dai_fmt, +}; + +struct snd_soc_dai ak4671_dai = { + .name = "AK4671", + .playback = { + .stream_name = "Playback", + .channels_min = 1, + .channels_max = 2, + .rates = AK4671_RATES, + .formats = AK4671_FORMATS,}, + .capture = { + .stream_name = "Capture", + .channels_min = 1, + .channels_max = 2, + .rates = AK4671_RATES, + .formats = AK4671_FORMATS,}, + .ops = &ak4671_dai_ops, +}; +EXPORT_SYMBOL_GPL(ak4671_dai); + +static int ak4671_probe(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + struct snd_soc_codec *codec; + int ret = 0; + + if (ak4671_codec == NULL) { + dev_err(&pdev->dev, "Codec device not registered\n"); + return -ENODEV; + } + + socdev->card->codec = ak4671_codec; + codec = ak4671_codec; + + /* register pcms */ + ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); + if (ret < 0) { + dev_err(codec->dev, "failed to create pcms: %d\n", ret); + goto pcm_err; + } + + snd_soc_add_controls(codec, ak4671_snd_controls, + ARRAY_SIZE(ak4671_snd_controls)); + ak4671_add_widgets(codec); + + ret = snd_soc_init_card(socdev); + if (ret < 0) { + dev_err(codec->dev, "failed to register card: %d\n", ret); + goto card_err; + } + + ak4671_set_bias_level(codec, SND_SOC_BIAS_STANDBY); + + return ret; + +card_err: + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); +pcm_err: + return ret; +} + +static int ak4671_remove(struct platform_device *pdev) +{ + struct snd_soc_device *socdev = platform_get_drvdata(pdev); + + snd_soc_free_pcms(socdev); + snd_soc_dapm_free(socdev); + + return 0; +} + +struct snd_soc_codec_device soc_codec_dev_ak4671 = { + .probe = ak4671_probe, + .remove = ak4671_remove, +}; +EXPORT_SYMBOL_GPL(soc_codec_dev_ak4671); + +static int ak4671_register(struct ak4671_priv *ak4671, + enum snd_soc_control_type control) +{ + int ret; + struct snd_soc_codec *codec = &ak4671->codec; + + if (ak4671_codec) { + dev_err(codec->dev, "Another AK4671 is registered\n"); + ret = -EINVAL; + goto err; + } + + mutex_init(&codec->mutex); + INIT_LIST_HEAD(&codec->dapm_widgets); + INIT_LIST_HEAD(&codec->dapm_paths); + + codec->private_data = ak4671; + codec->name = "AK4671"; + codec->owner = THIS_MODULE; + codec->bias_level = SND_SOC_BIAS_OFF; + codec->set_bias_level = ak4671_set_bias_level; + codec->dai = &ak4671_dai; + codec->num_dai = 1; + codec->reg_cache_size = AK4671_CACHEREGNUM; + codec->reg_cache = &ak4671->reg_cache; + + memcpy(codec->reg_cache, ak4671_reg, sizeof(ak4671_reg)); + + ret = snd_soc_codec_set_cache_io(codec, 8, 8, control); + if (ret < 0) { + dev_err(codec->dev, "Failed to set cache I/O: %d\n", ret); + goto err; + } + + ak4671_dai.dev = codec->dev; + ak4671_codec = codec; + + ret = snd_soc_register_codec(codec); + if (ret != 0) { + dev_err(codec->dev, "Failed to register codec: %d\n", ret); + goto err; + } + + ret = snd_soc_register_dai(&ak4671_dai); + if (ret != 0) { + dev_err(codec->dev, "Failed to register DAI: %d\n", ret); + goto err_codec; + } + + return 0; + +err_codec: + snd_soc_unregister_codec(codec); +err: + kfree(ak4671); + return ret; +} + +static void ak4671_unregister(struct ak4671_priv *ak4671) +{ + ak4671_set_bias_level(&ak4671->codec, SND_SOC_BIAS_OFF); + snd_soc_unregister_dai(&ak4671_dai); + snd_soc_unregister_codec(&ak4671->codec); + kfree(ak4671); + ak4671_codec = NULL; +} + +static int __devinit ak4671_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct ak4671_priv *ak4671; + struct snd_soc_codec *codec; + + ak4671 = kzalloc(sizeof(struct ak4671_priv), GFP_KERNEL); + if (ak4671 == NULL) + return -ENOMEM; + + codec = &ak4671->codec; + codec->hw_write = (hw_write_t)i2c_master_send; + + i2c_set_clientdata(client, ak4671); + codec->control_data = client; + + codec->dev = &client->dev; + + return ak4671_register(ak4671, SND_SOC_I2C); +} + +static __devexit int ak4671_i2c_remove(struct i2c_client *client) +{ + struct ak4671_priv *ak4671 = i2c_get_clientdata(client); + + ak4671_unregister(ak4671); + + return 0; +} + +static const struct i2c_device_id ak4671_i2c_id[] = { + { "ak4671", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c, ak4671_i2c_id); + +static struct i2c_driver ak4671_i2c_driver = { + .driver = { + .name = "ak4671", + .owner = THIS_MODULE, + }, + .probe = ak4671_i2c_probe, + .remove = __devexit_p(ak4671_i2c_remove), + .id_table = ak4671_i2c_id, +}; + +static int __init ak4671_modinit(void) +{ + return i2c_add_driver(&ak4671_i2c_driver); +} +module_init(ak4671_modinit); + +static void __exit ak4671_exit(void) +{ + i2c_del_driver(&ak4671_i2c_driver); +} +module_exit(ak4671_exit); + +MODULE_DESCRIPTION("ASoC AK4671 codec driver"); +MODULE_AUTHOR("Joonyoung Shim <jy0922.shim(a)samsung.com>"); +MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/ak4671.h b/sound/soc/codecs/ak4671.h new file mode 100644 index 0000000..e2fad96 --- /dev/null +++ b/sound/soc/codecs/ak4671.h @@ -0,0 +1,156 @@ +/* + * ak4671.h -- audio driver for AK4671 + * + * Copyright (C) 2009 Samsung Electronics Co.Ltd + * Author: Joonyoung Shim <jy0922.shim(a)samsung.com> + * + * This program is free software; you can redistribute it and/or modify it + * under the terms of the GNU General Public License as published by the + * Free Software Foundation; either version 2 of the License, or (at your + * option) any later version. + * + */ + +#ifndef _AK4671_H +#define _AK4671_H + +#define AK4671_AD_DA_POWER_MANAGEMENT 0x00 +#define AK4671_PLL_MODE_SELECT0 0x01 +#define AK4671_PLL_MODE_SELECT1 0x02 +#define AK4671_FORMAT_SELECT 0x03 +#define AK4671_MIC_SIGNAL_SELECT 0x04 +#define AK4671_MIC_AMP_GAIN 0x05 +#define AK4671_MIXING_POWER_MANAGEMENT0 0x06 +#define AK4671_MIXING_POWER_MANAGEMENT1 0x07 +#define AK4671_OUTPUT_VOLUME_CONTROL 0x08 +#define AK4671_LOUT1_SIGNAL_SELECT 0x09 +#define AK4671_ROUT1_SIGNAL_SELECT 0x0a +#define AK4671_LOUT2_SIGNAL_SELECT 0x0b +#define AK4671_ROUT2_SIGNAL_SELECT 0x0c +#define AK4671_LOUT3_SIGNAL_SELECT 0x0d +#define AK4671_ROUT3_SIGNAL_SELECT 0x0e +#define AK4671_LOUT1_POWER_MANAGERMENT 0x0f +#define AK4671_LOUT2_POWER_MANAGERMENT 0x10 +#define AK4671_LOUT3_POWER_MANAGERMENT 0x11 +#define AK4671_LCH_INPUT_VOLUME_CONTROL 0x12 +#define AK4671_RCH_INPUT_VOLUME_CONTROL 0x13 +#define AK4671_ALC_REFERENCE_SELECT 0x14 +#define AK4671_DIGITAL_MIXING_CONTROL 0x15 +#define AK4671_ALC_TIMER_SELECT 0x16 +#define AK4671_ALC_MODE_CONTROL 0x17 +#define AK4671_MODE_CONTROL1 0x18 +#define AK4671_MODE_CONTROL2 0x19 +#define AK4671_LCH_OUTPUT_VOLUME_CONTROL 0x1a +#define AK4671_RCH_OUTPUT_VOLUME_CONTROL 0x1b +#define AK4671_SIDETONE_A_CONTROL 0x1c +#define AK4671_DIGITAL_FILTER_SELECT 0x1d +#define AK4671_FIL3_COEFFICIENT0 0x1e +#define AK4671_FIL3_COEFFICIENT1 0x1f +#define AK4671_FIL3_COEFFICIENT2 0x20 +#define AK4671_FIL3_COEFFICIENT3 0x21 +#define AK4671_EQ_COEFFICIENT0 0x22 +#define AK4671_EQ_COEFFICIENT1 0x23 +#define AK4671_EQ_COEFFICIENT2 0x24 +#define AK4671_EQ_COEFFICIENT3 0x25 +#define AK4671_EQ_COEFFICIENT4 0x26 +#define AK4671_EQ_COEFFICIENT5 0x27 +#define AK4671_FIL1_COEFFICIENT0 0x28 +#define AK4671_FIL1_COEFFICIENT1 0x29 +#define AK4671_FIL1_COEFFICIENT2 0x2a +#define AK4671_FIL1_COEFFICIENT3 0x2b +#define AK4671_FIL2_COEFFICIENT0 0x2c +#define AK4671_FIL2_COEFFICIENT1 0x2d +#define AK4671_FIL2_COEFFICIENT2 0x2e +#define AK4671_FIL2_COEFFICIENT3 0x2f +#define AK4671_DIGITAL_FILTER_SELECT2 0x30 +#define AK4671_E1_COEFFICIENT0 0x32 +#define AK4671_E1_COEFFICIENT1 0x33 +#define AK4671_E1_COEFFICIENT2 0x34 +#define AK4671_E1_COEFFICIENT3 0x35 +#define AK4671_E1_COEFFICIENT4 0x36 +#define AK4671_E1_COEFFICIENT5 0x37 +#define AK4671_E2_COEFFICIENT0 0x38 +#define AK4671_E2_COEFFICIENT1 0x39 +#define AK4671_E2_COEFFICIENT2 0x3a +#define AK4671_E2_COEFFICIENT3 0x3b +#define AK4671_E2_COEFFICIENT4 0x3c +#define AK4671_E2_COEFFICIENT5 0x3d +#define AK4671_E3_COEFFICIENT0 0x3e +#define AK4671_E3_COEFFICIENT1 0x3f +#define AK4671_E3_COEFFICIENT2 0x40 +#define AK4671_E3_COEFFICIENT3 0x41 +#define AK4671_E3_COEFFICIENT4 0x42 +#define AK4671_E3_COEFFICIENT5 0x43 +#define AK4671_E4_COEFFICIENT0 0x44 +#define AK4671_E4_COEFFICIENT1 0x45 +#define AK4671_E4_COEFFICIENT2 0x46 +#define AK4671_E4_COEFFICIENT3 0x47 +#define AK4671_E4_COEFFICIENT4 0x48 +#define AK4671_E4_COEFFICIENT5 0x49 +#define AK4671_E5_COEFFICIENT0 0x4a +#define AK4671_E5_COEFFICIENT1 0x4b +#define AK4671_E5_COEFFICIENT2 0x4c +#define AK4671_E5_COEFFICIENT3 0x4d +#define AK4671_E5_COEFFICIENT4 0x4e +#define AK4671_E5_COEFFICIENT5 0x4f +#define AK4671_EQ_CONTROL_250HZ_100HZ 0x50 +#define AK4671_EQ_CONTROL_3500HZ_1KHZ 0x51 +#define AK4671_EQ_CONTRO_10KHZ 0x52 +#define AK4671_PCM_IF_CONTROL0 0x53 +#define AK4671_PCM_IF_CONTROL1 0x54 +#define AK4671_PCM_IF_CONTROL2 0x55 +#define AK4671_DIGITAL_VOLUME_B_CONTROL 0x56 +#define AK4671_DIGITAL_VOLUME_C_CONTROL 0x57 +#define AK4671_SIDETONE_VOLUME_CONTROL 0x58 +#define AK4671_DIGITAL_MIXING_CONTROL2 0x59 +#define AK4671_SAR_ADC_CONTROL 0x5a + +#define AK4671_CACHEREGNUM (AK4671_SAR_ADC_CONTROL + 1) + +/* Bitfield Definitions */ + +/* AK4671_AD_DA_POWER_MANAGEMENT (0x00) Fields */ +#define AK4671_PMVCM 0x01 + +/* AK4671_PLL_MODE_SELECT0 (0x01) Fields */ +#define AK4671_PLL 0x0f +#define AK4671_PLL_11_2896MHZ (4 << 0) +#define AK4671_PLL_12_288MHZ (5 << 0) +#define AK4671_PLL_12MHZ (6 << 0) +#define AK4671_PLL_24MHZ (7 << 0) +#define AK4671_PLL_19_2MHZ (8 << 0) +#define AK4671_PLL_13_5MHZ (12 << 0) +#define AK4671_PLL_27MHZ (13 << 0) +#define AK4671_PLL_13MHZ (14 << 0) +#define AK4671_PLL_26MHZ (15 << 0) +#define AK4671_FS 0xf0 +#define AK4671_FS_8KHZ (0 << 4) +#define AK4671_FS_12KHZ (1 << 4) +#define AK4671_FS_16KHZ (2 << 4) +#define AK4671_FS_24KHZ (3 << 4) +#define AK4671_FS_11_025KHZ (5 << 4) +#define AK4671_FS_22_05KHZ (7 << 4) +#define AK4671_FS_32KHZ (10 << 4) +#define AK4671_FS_48KHZ (11 << 4) +#define AK4671_FS_44_1KHZ (15 << 4) + +/* AK4671_PLL_MODE_SELECT1 (0x02) Fields */ +#define AK4671_PMPLL 0x01 +#define AK4671_M_S 0x02 + +/* AK4671_FORMAT_SELECT (0x03) Fields */ +#define AK4671_DIF 0x03 +#define AK4671_DIF_DSP_MODE (0 << 0) +#define AK4671_DIF_MSB_MODE (2 << 0) +#define AK4671_DIF_I2S_MODE (3 << 0) +#define AK4671_BCKP 0x04 +#define AK4671_MSBS 0x08 +#define AK4671_SDOD 0x10 + +/* AK4671_LOUT2_POWER_MANAGEMENT (0x10) Fields */ +#define AK4671_MUTEN 0x04 + +extern struct snd_soc_dai ak4671_dai; +extern struct snd_soc_codec_device soc_codec_dev_ak4671; + +#endif -- 1.6.0.4

2 2

[alsa-devel] [PATCH] AC97 atmel: add support for AT91(common with AVR32).
by Sedji Gaouaou 07 Sep '09

07 Sep '09

This patch add AC97 support for ATMEL AT91 boards, using the AVR32 code. It is based on Takashi git tree(sound-2.6/for-next). Regards, Sedji Signed-off-by: Sedji Gaouaou <sedji.gaouaou(a)atmel.com> --- sound/atmel/Kconfig | 2 +- sound/atmel/ac97c.c | 364 +++++++++++++++++++++++++++++++++++++-------------- 2 files changed, 268 insertions(+), 98 deletions(-) diff --git a/sound/atmel/Kconfig b/sound/atmel/Kconfig index 6c228a9..94de43a 100644 --- a/sound/atmel/Kconfig +++ b/sound/atmel/Kconfig @@ -12,7 +12,7 @@ config SND_ATMEL_AC97C tristate "Atmel AC97 Controller (AC97C) driver" select SND_PCM select SND_AC97_CODEC - depends on DW_DMAC && AVR32 + depends on (DW_DMAC && AVR32) || ARCH_AT91 help ALSA sound driver for the Atmel AC97 controller. diff --git a/sound/atmel/ac97c.c b/sound/atmel/ac97c.c index 0c0f877..379d7ea 100644 --- a/sound/atmel/ac97c.c +++ b/sound/atmel/ac97c.c @@ -13,6 +13,7 @@ #include <linux/device.h> #include <linux/dmaengine.h> #include <linux/dma-mapping.h> +#include <linux/atmel_pdc.h> #include <linux/init.h> #include <linux/interrupt.h> #include <linux/module.h> @@ -31,6 +32,10 @@ #include <linux/dw_dmac.h> +#include <mach/cpu.h> +#include <mach/hardware.h> +#include <mach/gpio.h> + #include "ac97c.h" enum { @@ -63,6 +68,7 @@ struct atmel_ac97c { u64 cur_format; unsigned int cur_rate; unsigned long flags; + int period; /* Serialize access to opened variable */ spinlock_t lock; void __iomem *regs; @@ -167,6 +173,7 @@ static int atmel_ac97c_playback_open(struct snd_pcm_substream *substream) mutex_lock(&opened_mutex); chip->opened++; runtime->hw = atmel_ac97c_hw; + chip->period = 0; if (chip->cur_rate) { runtime->hw.rate_min = chip->cur_rate; runtime->hw.rate_max = chip->cur_rate; @@ -186,6 +193,7 @@ static int atmel_ac97c_capture_open(struct snd_pcm_substream *substream) mutex_lock(&opened_mutex); chip->opened++; runtime->hw = atmel_ac97c_hw; + chip->period = 0; if (chip->cur_rate) { runtime->hw.rate_min = chip->cur_rate; runtime->hw.rate_max = chip->cur_rate; @@ -239,12 +247,14 @@ static int atmel_ac97c_playback_hw_params(struct snd_pcm_substream *substream, retval = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); - if (retval < 0) - return retval; - /* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */ - if (retval == 1) - if (test_and_clear_bit(DMA_TX_READY, &chip->flags)) - dw_dma_cyclic_free(chip->dma.tx_chan); + if(cpu_is_at32ap7000()) { + if (retval < 0) + return retval; + /* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */ + if (retval == 1) + if (test_and_clear_bit(DMA_TX_READY, &chip->flags)) + dw_dma_cyclic_free(chip->dma.tx_chan); + } /* Set restrictions to params. */ mutex_lock(&opened_mutex); @@ -263,12 +273,14 @@ static int atmel_ac97c_capture_hw_params(struct snd_pcm_substream *substream, retval = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params)); - if (retval < 0) - return retval; - /* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */ - if (retval == 1) - if (test_and_clear_bit(DMA_RX_READY, &chip->flags)) - dw_dma_cyclic_free(chip->dma.rx_chan); + if(cpu_is_at32ap7000()) { + if (retval < 0) + return retval; + /* snd_pcm_lib_malloc_pages returns 1 if buffer is changed. */ + if (retval == 1) + if (test_and_clear_bit(DMA_RX_READY, &chip->flags)) + dw_dma_cyclic_free(chip->dma.rx_chan); + } /* Set restrictions to params. */ mutex_lock(&opened_mutex); @@ -282,16 +294,20 @@ static int atmel_ac97c_capture_hw_params(struct snd_pcm_substream *substream, static int atmel_ac97c_playback_hw_free(struct snd_pcm_substream *substream) { struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); - if (test_and_clear_bit(DMA_TX_READY, &chip->flags)) - dw_dma_cyclic_free(chip->dma.tx_chan); + if(cpu_is_at32ap7000()) { + if (test_and_clear_bit(DMA_TX_READY, &chip->flags)) + dw_dma_cyclic_free(chip->dma.tx_chan); + } return snd_pcm_lib_free_pages(substream); } static int atmel_ac97c_capture_hw_free(struct snd_pcm_substream *substream) { struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); - if (test_and_clear_bit(DMA_RX_READY, &chip->flags)) - dw_dma_cyclic_free(chip->dma.rx_chan); + if(cpu_is_at32ap7000()) { + if (test_and_clear_bit(DMA_RX_READY, &chip->flags)) + dw_dma_cyclic_free(chip->dma.rx_chan); + } return snd_pcm_lib_free_pages(substream); } @@ -299,6 +315,7 @@ static int atmel_ac97c_playback_prepare(struct snd_pcm_substream *substream) { struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; + int block_size = frames_to_bytes(runtime, runtime->period_size); unsigned long word = ac97c_readl(chip, OCA); int retval; @@ -324,7 +341,9 @@ static int atmel_ac97c_playback_prepare(struct snd_pcm_substream *substream) switch (runtime->format) { case SNDRV_PCM_FORMAT_S16_LE: - word |= AC97C_CMR_CEM_LITTLE; + if(cpu_is_at32ap7000()) { + word |= AC97C_CMR_CEM_LITTLE; + } break; case SNDRV_PCM_FORMAT_S16_BE: /* fall through */ word &= ~(AC97C_CMR_CEM_LITTLE); @@ -363,9 +382,18 @@ static int atmel_ac97c_playback_prepare(struct snd_pcm_substream *substream) dev_dbg(&chip->pdev->dev, "could not set rate %d Hz\n", runtime->rate); - if (!test_bit(DMA_TX_READY, &chip->flags)) - retval = atmel_ac97c_prepare_dma(chip, substream, - DMA_TO_DEVICE); + if(cpu_is_at32ap7000()) { + if (!test_bit(DMA_TX_READY, &chip->flags)) + retval = atmel_ac97c_prepare_dma(chip, substream, + DMA_TO_DEVICE); + } else { + /* Initialize and start the PDC */ + writel(runtime->dma_addr, chip->regs + ATMEL_PDC_TPR); + writel(block_size / 2, chip->regs + ATMEL_PDC_TCR); + writel(runtime->dma_addr + block_size, + chip->regs + ATMEL_PDC_TNPR); + writel(block_size / 2, chip->regs + ATMEL_PDC_TNCR); + } return retval; } @@ -374,6 +402,7 @@ static int atmel_ac97c_capture_prepare(struct snd_pcm_substream *substream) { struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); struct snd_pcm_runtime *runtime = substream->runtime; + int block_size = frames_to_bytes(runtime, runtime->period_size); unsigned long word = ac97c_readl(chip, ICA); int retval; @@ -415,11 +444,15 @@ static int atmel_ac97c_capture_prepare(struct snd_pcm_substream *substream) word |= AC97C_CSR_OVRUN; ac97c_writel(chip, CAMR, word); - /* Enable channel A event interrupt */ word = ac97c_readl(chip, IMR); word |= AC97C_SR_CAEVT; - ac97c_writel(chip, IER, word); + ac97c_writel(chip, IER, /*word*/AC97C_SR_CAEVT); + + /* Enable channel A event interrupt */ + /*word = ac97c_readl(chip, IMR); + word |= AC97C_SR_CAEVT; + ac97c_writel(chip, IER, word);*/ /* set variable rate if needed */ if (runtime->rate != 48000) { @@ -438,9 +471,18 @@ static int atmel_ac97c_capture_prepare(struct snd_pcm_substream *substream) dev_dbg(&chip->pdev->dev, "could not set rate %d Hz\n", runtime->rate); - if (!test_bit(DMA_RX_READY, &chip->flags)) - retval = atmel_ac97c_prepare_dma(chip, substream, - DMA_FROM_DEVICE); + if(cpu_is_at32ap7000()) { + if (!test_bit(DMA_RX_READY, &chip->flags)) + retval = atmel_ac97c_prepare_dma(chip, substream, + DMA_FROM_DEVICE); + } else { + /* Initialize and start the PDC */ + writel(runtime->dma_addr, chip->regs + ATMEL_PDC_RPR); + writel(block_size / 2, chip->regs + ATMEL_PDC_RCR); + writel(runtime->dma_addr + block_size, + chip->regs + ATMEL_PDC_RNPR); + writel(block_size / 2, chip->regs + ATMEL_PDC_RNCR); + } return retval; } @@ -449,7 +491,7 @@ static int atmel_ac97c_playback_trigger(struct snd_pcm_substream *substream, int cmd) { struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); - unsigned long camr; + unsigned long camr, ptcr = 0; int retval = 0; camr = ac97c_readl(chip, CAMR); @@ -458,15 +500,23 @@ atmel_ac97c_playback_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* fall through */ case SNDRV_PCM_TRIGGER_RESUME: /* fall through */ case SNDRV_PCM_TRIGGER_START: - retval = dw_dma_cyclic_start(chip->dma.tx_chan); - if (retval) - goto out; - camr |= AC97C_CMR_CENA; + if(cpu_is_at32ap7000()) { + retval = dw_dma_cyclic_start(chip->dma.tx_chan); + if (retval) + goto out; + } else { + ptcr = ATMEL_PDC_TXTEN; + } + camr |= AC97C_CMR_CENA | AC97C_CSR_ENDTX; break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* fall through */ case SNDRV_PCM_TRIGGER_SUSPEND: /* fall through */ case SNDRV_PCM_TRIGGER_STOP: - dw_dma_cyclic_stop(chip->dma.tx_chan); + if(cpu_is_at32ap7000()) { + dw_dma_cyclic_stop(chip->dma.tx_chan); + } else { + ptcr = ATMEL_PDC_TXTDIS; + } if (chip->opened <= 1) camr &= ~AC97C_CMR_CENA; break; @@ -476,6 +526,8 @@ atmel_ac97c_playback_trigger(struct snd_pcm_substream *substream, int cmd) } ac97c_writel(chip, CAMR, camr); + if(!cpu_is_at32ap7000()) + writel(ptcr, chip->regs + ATMEL_PDC_PTCR); out: return retval; } @@ -484,7 +536,7 @@ static int atmel_ac97c_capture_trigger(struct snd_pcm_substream *substream, int cmd) { struct atmel_ac97c *chip = snd_pcm_substream_chip(substream); - unsigned long camr; + unsigned long camr, ptcr = 0; int retval = 0; camr = ac97c_readl(chip, CAMR); @@ -493,15 +545,24 @@ atmel_ac97c_capture_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: /* fall through */ case SNDRV_PCM_TRIGGER_RESUME: /* fall through */ case SNDRV_PCM_TRIGGER_START: - retval = dw_dma_cyclic_start(chip->dma.rx_chan); - if (retval) - goto out; + if(cpu_is_at32ap7000()) { + retval = dw_dma_cyclic_start(chip->dma.rx_chan); + if (retval) + goto out; + } else { + ptcr = ATMEL_PDC_RXTEN; + } camr |= AC97C_CMR_CENA; break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: /* fall through */ case SNDRV_PCM_TRIGGER_SUSPEND: /* fall through */ case SNDRV_PCM_TRIGGER_STOP: - dw_dma_cyclic_stop(chip->dma.rx_chan); + if(cpu_is_at32ap7000()) { + dw_dma_cyclic_stop(chip->dma.rx_chan); + } + else { + ptcr = ATMEL_PDC_RXTDIS; + } if (chip->opened <= 1) camr &= ~AC97C_CMR_CENA; break; @@ -511,6 +572,8 @@ atmel_ac97c_capture_trigger(struct snd_pcm_substream *substream, int cmd) } ac97c_writel(chip, CAMR, camr); + if(!cpu_is_at32ap7000()) + writel(ptcr, chip->regs + ATMEL_PDC_PTCR); out: return retval; } @@ -523,7 +586,11 @@ atmel_ac97c_playback_pointer(struct snd_pcm_substream *substream) snd_pcm_uframes_t frames; unsigned long bytes; - bytes = dw_dma_get_src_addr(chip->dma.tx_chan); + if(cpu_is_at32ap7000()) { + bytes = dw_dma_get_src_addr(chip->dma.tx_chan); + } else { + bytes = readl(chip->regs + ATMEL_PDC_TPR); + } bytes -= runtime->dma_addr; frames = bytes_to_frames(runtime, bytes); @@ -540,7 +607,11 @@ atmel_ac97c_capture_pointer(struct snd_pcm_substream *substream) snd_pcm_uframes_t frames; unsigned long bytes; - bytes = dw_dma_get_dst_addr(chip->dma.rx_chan); + if(cpu_is_at32ap7000()) { + bytes = dw_dma_get_dst_addr(chip->dma.rx_chan); + } else { + bytes = readl(chip->regs + ATMEL_PDC_RPR); + } bytes -= runtime->dma_addr; frames = bytes_to_frames(runtime, bytes); @@ -578,20 +649,66 @@ static irqreturn_t atmel_ac97c_interrupt(int irq, void *dev) u32 sr = ac97c_readl(chip, SR); u32 casr = ac97c_readl(chip, CASR); u32 cosr = ac97c_readl(chip, COSR); + u32 camr = ac97c_readl(chip, CAMR); if (sr & AC97C_SR_CAEVT) { - dev_info(&chip->pdev->dev, "channel A event%s%s%s%s%s%s\n", + struct snd_pcm_runtime *runtime; + int offset, next_period, block_size; + dev_dbg(&chip->pdev->dev, "channel A event%s%s%s%s%s%s\n", casr & AC97C_CSR_OVRUN ? " OVRUN" : "", casr & AC97C_CSR_RXRDY ? " RXRDY" : "", casr & AC97C_CSR_UNRUN ? " UNRUN" : "", casr & AC97C_CSR_TXEMPTY ? " TXEMPTY" : "", casr & AC97C_CSR_TXRDY ? " TXRDY" : "", !casr ? " NONE" : ""); + if(!cpu_is_at32ap7000()) { + if ((casr & camr) & AC97C_CSR_ENDTX) { + runtime = chip->playback_substream->runtime; + block_size = frames_to_bytes(runtime, + runtime->period_size); + chip->period++; + + if (chip->period == runtime->periods) + chip->period = 0; + next_period = chip->period + 1; + if (next_period == runtime->periods) + next_period = 0; + + offset = block_size * next_period; + + writel(runtime->dma_addr + offset, + chip->regs + ATMEL_PDC_TNPR); + writel(block_size / 2, + chip->regs + ATMEL_PDC_TNCR); + + snd_pcm_period_elapsed(chip->playback_substream); + } + if ((casr & camr) & AC97C_CSR_ENDRX) { + runtime = chip->capture_substream->runtime; + block_size = frames_to_bytes(runtime, + runtime->period_size); + chip->period++; + + if (chip->period == runtime->periods) + chip->period = 0; + next_period = chip->period + 1; + if (next_period == runtime->periods) + next_period = 0; + + offset = block_size * next_period; + + writel(runtime->dma_addr + offset, + chip->regs + ATMEL_PDC_RNPR); + writel(block_size / 2, + chip->regs + ATMEL_PDC_RNCR); + snd_pcm_period_elapsed(chip->capture_substream); + } + } retval = IRQ_HANDLED; } if (sr & AC97C_SR_COEVT) { - dev_info(&chip->pdev->dev, "codec channel event%s%s%s%s%s\n", + dev_dbg(&chip->pdev->dev, "codec channel event%s%s%s%s%s\n", cosr & AC97C_CSR_OVRUN ? " OVRUN" : "", cosr & AC97C_CSR_RXRDY ? " RXRDY" : "", cosr & AC97C_CSR_TXEMPTY ? " TXEMPTY" : "", @@ -608,15 +725,50 @@ static irqreturn_t atmel_ac97c_interrupt(int irq, void *dev) return retval; } +static struct ac97_pcm at91_ac97_pcm_defs[] __devinitdata = { + /* Playback */ + { + .exclusive = 1, + .r = { { + .slots = ((1 << AC97_SLOT_PCM_LEFT) + | (1 << AC97_SLOT_PCM_RIGHT)), + } }, + }, + /* PCM in */ + { + .stream = 1, + .exclusive = 1, + .r = { { + .slots = ((1 << AC97_SLOT_PCM_LEFT) + | (1 << AC97_SLOT_PCM_RIGHT)), + } } + }, + /* Mic in */ + { + .stream = 1, + .exclusive = 1, + .r = { { + .slots = (1<<AC97_SLOT_MIC), + } } + }, +}; + static int __devinit atmel_ac97c_pcm_new(struct atmel_ac97c *chip) { struct snd_pcm *pcm; struct snd_pcm_hardware hw = atmel_ac97c_hw; - int capture, playback, retval; + int capture, playback, retval, err; capture = test_bit(DMA_RX_CHAN_PRESENT, &chip->flags); playback = test_bit(DMA_TX_CHAN_PRESENT, &chip->flags); + if(!cpu_is_at32ap7000()) { + err = snd_ac97_pcm_assign(chip->ac97_bus, + ARRAY_SIZE(at91_ac97_pcm_defs), + at91_ac97_pcm_defs); + if (err) + return err; + } retval = snd_pcm_new(chip->card, chip->card->shortname, chip->pdev->id, playback, capture, &pcm); if (retval) @@ -775,7 +927,12 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) return -ENXIO; } - pclk = clk_get(&pdev->dev, "pclk"); + if(cpu_is_at32ap7000()) { + pclk = clk_get(&pdev->dev, "pclk"); + } else { + pclk = clk_get(&pdev->dev, "ac97_clk"); + } + if (IS_ERR(pclk)) { dev_dbg(&pdev->dev, "no peripheral clock\n"); return PTR_ERR(pclk); @@ -844,44 +1001,51 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) goto err_ac97_bus; } - if (pdata->rx_dws.dma_dev) { - struct dw_dma_slave *dws = &pdata->rx_dws; - dma_cap_mask_t mask; + if(cpu_is_at32ap7000()) { + if (pdata->rx_dws.dma_dev) { + struct dw_dma_slave *dws = &pdata->rx_dws; + dma_cap_mask_t mask; - dws->rx_reg = regs->start + AC97C_CARHR + 2; + dws->rx_reg = regs->start + AC97C_CARHR + 2; - dma_cap_zero(mask); - dma_cap_set(DMA_SLAVE, mask); + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); - chip->dma.rx_chan = dma_request_channel(mask, filter, dws); + chip->dma.rx_chan = dma_request_channel(mask, filter, dws); - dev_info(&chip->pdev->dev, "using %s for DMA RX\n", + dev_info(&chip->pdev->dev, "using %s for DMA RX\n", dev_name(&chip->dma.rx_chan->dev->device)); - set_bit(DMA_RX_CHAN_PRESENT, &chip->flags); - } + set_bit(DMA_RX_CHAN_PRESENT, &chip->flags); + } - if (pdata->tx_dws.dma_dev) { - struct dw_dma_slave *dws = &pdata->tx_dws; - dma_cap_mask_t mask; + if (pdata->tx_dws.dma_dev) { + struct dw_dma_slave *dws = &pdata->tx_dws; + dma_cap_mask_t mask; - dws->tx_reg = regs->start + AC97C_CATHR + 2; + dws->tx_reg = regs->start + AC97C_CATHR + 2; - dma_cap_zero(mask); - dma_cap_set(DMA_SLAVE, mask); + dma_cap_zero(mask); + dma_cap_set(DMA_SLAVE, mask); - chip->dma.tx_chan = dma_request_channel(mask, filter, dws); + chip->dma.tx_chan = dma_request_channel(mask, filter, dws); - dev_info(&chip->pdev->dev, "using %s for DMA TX\n", - dev_name(&chip->dma.tx_chan->dev->device)); - set_bit(DMA_TX_CHAN_PRESENT, &chip->flags); - } + dev_info(&chip->pdev->dev, "using %s for DMA TX\n", + dev_name(&chip->dma.tx_chan->dev->device)); + set_bit(DMA_TX_CHAN_PRESENT, &chip->flags); + } - if (!test_bit(DMA_RX_CHAN_PRESENT, &chip->flags) && - !test_bit(DMA_TX_CHAN_PRESENT, &chip->flags)) { - dev_dbg(&pdev->dev, "DMA not available\n"); - retval = -ENODEV; - goto err_dma; - } + if (!test_bit(DMA_RX_CHAN_PRESENT, &chip->flags) && + !test_bit(DMA_TX_CHAN_PRESENT, &chip->flags)) { + dev_dbg(&pdev->dev, "DMA not available\n"); + retval = -ENODEV; + goto err_dma; + } + } else { + /* Just pretend that we have DMA channel(for at91 i is actually + * the PDC */ + set_bit(DMA_RX_CHAN_PRESENT, &chip->flags); + set_bit(DMA_TX_CHAN_PRESENT, &chip->flags); + } retval = atmel_ac97c_pcm_new(chip); if (retval) { @@ -897,20 +1061,22 @@ static int __devinit atmel_ac97c_probe(struct platform_device *pdev) platform_set_drvdata(pdev, card); - dev_info(&pdev->dev, "Atmel AC97 controller at 0x%p\n", - chip->regs); + dev_info(&pdev->dev, "Atmel AC97 controller at 0x%p, irq = %d\n", + chip->regs, irq); return 0; err_dma: - if (test_bit(DMA_RX_CHAN_PRESENT, &chip->flags)) - dma_release_channel(chip->dma.rx_chan); - if (test_bit(DMA_TX_CHAN_PRESENT, &chip->flags)) - dma_release_channel(chip->dma.tx_chan); - clear_bit(DMA_RX_CHAN_PRESENT, &chip->flags); - clear_bit(DMA_TX_CHAN_PRESENT, &chip->flags); - chip->dma.rx_chan = NULL; - chip->dma.tx_chan = NULL; + if(cpu_is_at32ap7000()) { + if (test_bit(DMA_RX_CHAN_PRESENT, &chip->flags)) + dma_release_channel(chip->dma.rx_chan); + if (test_bit(DMA_TX_CHAN_PRESENT, &chip->flags)) + dma_release_channel(chip->dma.tx_chan); + clear_bit(DMA_RX_CHAN_PRESENT, &chip->flags); + clear_bit(DMA_TX_CHAN_PRESENT, &chip->flags); + chip->dma.rx_chan = NULL; + chip->dma.tx_chan = NULL; + } err_ac97_bus: snd_card_set_dev(card, NULL); @@ -934,12 +1100,13 @@ static int atmel_ac97c_suspend(struct platform_device *pdev, pm_message_t msg) struct snd_card *card = platform_get_drvdata(pdev); struct atmel_ac97c *chip = card->private_data; - if (test_bit(DMA_RX_READY, &chip->flags)) - dw_dma_cyclic_stop(chip->dma.rx_chan); - if (test_bit(DMA_TX_READY, &chip->flags)) - dw_dma_cyclic_stop(chip->dma.tx_chan); + if(cpu_is_at32ap7000()) { + if (test_bit(DMA_RX_READY, &chip->flags)) + dw_dma_cyclic_stop(chip->dma.rx_chan); + if (test_bit(DMA_TX_READY, &chip->flags)) + dw_dma_cyclic_stop(chip->dma.tx_chan); + } clk_disable(chip->pclk); - return 0; } @@ -949,11 +1116,12 @@ static int atmel_ac97c_resume(struct platform_device *pdev) struct atmel_ac97c *chip = card->private_data; clk_enable(chip->pclk); - if (test_bit(DMA_RX_READY, &chip->flags)) - dw_dma_cyclic_start(chip->dma.rx_chan); - if (test_bit(DMA_TX_READY, &chip->flags)) - dw_dma_cyclic_start(chip->dma.tx_chan); - + if(cpu_is_at32ap7000()) { + if (test_bit(DMA_RX_READY, &chip->flags)) + dw_dma_cyclic_start(chip->dma.rx_chan); + if (test_bit(DMA_TX_READY, &chip->flags)) + dw_dma_cyclic_start(chip->dma.tx_chan); + } return 0; } #else @@ -978,14 +1146,16 @@ static int __devexit atmel_ac97c_remove(struct platform_device *pdev) iounmap(chip->regs); free_irq(chip->irq, chip); - if (test_bit(DMA_RX_CHAN_PRESENT, &chip->flags)) - dma_release_channel(chip->dma.rx_chan); - if (test_bit(DMA_TX_CHAN_PRESENT, &chip->flags)) - dma_release_channel(chip->dma.tx_chan); - clear_bit(DMA_RX_CHAN_PRESENT, &chip->flags); - clear_bit(DMA_TX_CHAN_PRESENT, &chip->flags); - chip->dma.rx_chan = NULL; - chip->dma.tx_chan = NULL; + if(cpu_is_at32ap7000()) { + if (test_bit(DMA_RX_CHAN_PRESENT, &chip->flags)) + dma_release_channel(chip->dma.rx_chan); + if (test_bit(DMA_TX_CHAN_PRESENT, &chip->flags)) + dma_release_channel(chip->dma.tx_chan); + clear_bit(DMA_RX_CHAN_PRESENT, &chip->flags); + clear_bit(DMA_TX_CHAN_PRESENT, &chip->flags); + chip->dma.rx_chan = NULL; + chip->dma.tx_chan = NULL; + } snd_card_set_dev(card, NULL); snd_card_free(card); -- 1.5.3.7

2 9

Re: [alsa-devel] Patch for au88x0 driver. Fix audacious issue and add auto reset in case of hardware ADB ctrl failure.
by Raymond Yau 07 Sep '09

07 Sep '09

On Thursday 03 September 2009 21:10, Manuel Jander wrote: > does anybody still have a AU88x0 card ? I know they are quite ancient > now, but if anyone could give the attached patch a try ? Would be > nice. > I made the patch against linux-2.6.30.4 > http://micro.homelinux.net/~mjander/au88x0.diff.gz <http://micro.homelinux.net/%7Emjander/au88x0.diff.gz> > > Best Regards, > Manuel Which version of audaicous (1.5.x or 2.1) are you testing ? (alsa-gapless or alsa-ng plugin) au88x0 most likely cannot support 1 sample per period because of the FIFO as described in patent 6167465. if you look at the source code of alsa-ng and alsa-gapless plugin both of them use the own way to calculate the buffer size instead of using snd_pcm_hw_params_get_buffer_size() http://hg.atheme.org/audacious-plugins/file/2c0a9bba77d7/src/alsa-ng/alsa-c… CHECK_FAIL (snd_pcm_hw_params_set_access (pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED), "snd_pcm_hw_params_set_access"); CHECK_FAIL (snd_pcm_hw_params_set_format (pcm_handle, hwparams, afmt), "snd_pcm_hw_params_set_format"); CHECK_FAIL (snd_pcm_hw_params_set_channels (pcm_handle, hwparams, nch), "snd_pcm_hw_params_set_channels"); CHECK_FAIL (snd_pcm_hw_params_set_rate (pcm_handle, hwparams, rate, 0), "snd_pcm_hw_params_set_rate"); CHECK_FAIL (snd_pcm_hw_params (pcm_handle, hwparams), "snd_pcm_hw_params"); bitwidth = snd_pcm_format_physical_width(afmt); bps = (rate * bitwidth * nch) >> 3; buf_size = MAX(aud_cfg->output_buffer_size, 100); ringbuf_size = buf_size * bps / 1000; How do you get the "ADB Ctrl failure" error ? There are two main problem of the au88x0 driver 1) alsa is unable to calculate the correct value for the three functions snd_pcm_hw_params_set_buffer_time_near() , snd_pcm_hw_params_set_period_time_near() and snd_pcm_set_params() https://bugtrack.alsa-project.org/alsa-bug/view.php?id=1724 Test case : alsa-lib/test/pcm_min.c ( change default to hw:0,0 ) Real world case: gstreamer and winealsa.drv 2) unable to recover when the number of available src is not enough (FATAL: ResManager: resource type 1 exhaust) https://bugtrack.alsa-project.org/alsa-bug/view.php?id=2228 test case : running application to use 15 src ( 7 stereo and one mono stream ) and open another stereo stream

1 0

[alsa-devel] [PATCH - Realtek HD-audio codec 1/1] ALSA: hda/realtek: Added support for CLEVO M540R subsystem, 6 channel + digital
by ddiaz＠cenditel.gob.ve 06 Sep '09

06 Sep '09

The model clevo-m540r was created with 6-channel and digital support. All functions verified except spdif. Tested with a VIT D2000 laptop which has: [lspci extract] Audio device [0403]: Intel Corporation 82801H (ICH8 Family) HD Audio Controller [8086:284b] (rev 03) Subsystem: CLEVO/KAPOK Computer Device [1558:5409] [/proc/asound/card0/codec\#0 header] Codec: Realtek ALC883 Address: 0 Function Id: 0x1 Vendor Id: 0x10ec0883 Subsystem Id: 0x15585409 Revision Id: 0x100002 Signed-off-by: Dhionel Diaz <ddiaz(a)cenditel.gob.ve> --- sound/pci/hda/patch_realtek.c | 77 +++++++++++++++++++++++++++++++++++++++++ 1 files changed, 77 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_realtek.c b/sound/pci/hda/patch_realtek.c index f3d186a..32faaac 100644 --- a/sound/pci/hda/patch_realtek.c +++ b/sound/pci/hda/patch_realtek.c @@ -232,6 +232,7 @@ enum { ALC888_3ST_HP, ALC888_6ST_DELL, ALC883_MITAC, + ALC883_CLEVO_M540R, ALC883_CLEVO_M720, ALC883_FUJITSU_PI2515, ALC888_FUJITSU_XA3530, @@ -6657,6 +6658,52 @@ static struct hda_channel_mode alc882_3ST_6ch_modes[3] = { #define alc883_3ST_6ch_modes alc882_3ST_6ch_modes /* + * 2ch mode + */ +static struct hda_verb alc883_3ST_ch2_clevo_init[] = { + { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { } /* end */ +}; + +/* + * 4ch mode + */ +static struct hda_verb alc883_3ST_ch4_clevo_init[] = { + { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { } /* end */ +}; + +/* + * 6ch mode + */ +static struct hda_verb alc883_3ST_ch6_clevo_init[] = { + { 0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x18, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x18, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x18, AC_VERB_SET_CONNECT_SEL, 0x02 }, + { 0x1a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, + { 0x1a, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE }, + { 0x1a, AC_VERB_SET_CONNECT_SEL, 0x01 }, + { } /* end */ +}; + +static struct hda_channel_mode alc883_3ST_6ch_clevo_modes[3] = { + { 2, alc883_3ST_ch2_clevo_init }, + { 4, alc883_3ST_ch4_clevo_init }, + { 6, alc883_3ST_ch6_clevo_init }, +}; + + +/* * 6ch mode */ static struct hda_verb alc882_sixstack_ch6_init[] = { @@ -8134,6 +8181,22 @@ static struct hda_verb alc883_mitac_verbs[] = { { } /* end */ }; +static struct hda_verb alc883_clevo_m540r_verbs[] = { + /* HP */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, + {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP}, + /* Int speaker */ + /*{0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP},*/ + + /* enable unsolicited event */ + /* + {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, + {0x18, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_MIC_EVENT | AC_USRSP_EN}, + */ + + { } /* end */ +}; + static struct hda_verb alc883_clevo_m720_verbs[] = { /* HP */ {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -8670,6 +8733,7 @@ static const char *alc882_models[ALC882_MODEL_LAST] = { [ALC888_3ST_HP] = "3stack-hp", [ALC888_6ST_DELL] = "6stack-dell", [ALC883_MITAC] = "mitac", + [ALC883_CLEVO_M540R] = "clevo-m540r", [ALC883_CLEVO_M720] = "clevo-m720", [ALC883_FUJITSU_PI2515] = "fujitsu-pi2515", [ALC888_FUJITSU_XA3530] = "fujitsu-xa3530", @@ -8776,6 +8840,7 @@ static struct snd_pci_quirk alc882_cfg_tbl[] = { SND_PCI_QUIRK(0x147b, 0x1083, "Abit IP35-PRO", ALC883_6ST_DIG), SND_PCI_QUIRK(0x1558, 0x0721, "Clevo laptop M720R", ALC883_CLEVO_M720), SND_PCI_QUIRK(0x1558, 0x0722, "Clevo laptop M720SR", ALC883_CLEVO_M720), + SND_PCI_QUIRK(0x1558, 0x5409, "Clevo laptop M540R", ALC883_CLEVO_M540R), SND_PCI_QUIRK_VENDOR(0x1558, "Clevo laptop", ALC883_LAPTOP_EAPD), SND_PCI_QUIRK(0x15d9, 0x8780, "Supermicro PDSBA", ALC883_3ST_6ch), /* SND_PCI_QUIRK(0x161f, 0x2054, "Arima W820", ALC882_ARIMA), */ @@ -9264,6 +9329,18 @@ static struct alc_config_preset alc882_presets[] = { .channel_mode = alc883_3ST_2ch_modes, .input_mux = &alc883_capture_source, }, + [ALC883_CLEVO_M540R] = { + .mixers = { alc883_3ST_6ch_mixer, alc883_chmode_mixer }, + .init_verbs = { alc883_init_verbs, alc883_clevo_m540r_verbs }, + .num_dacs = ARRAY_SIZE(alc883_dac_nids), + .dac_nids = alc883_dac_nids, + .dig_out_nid = ALC883_DIGOUT_NID, + .dig_in_nid = ALC883_DIGIN_NID, + .num_channel_mode = ARRAY_SIZE(alc883_3ST_6ch_clevo_modes), + .channel_mode = alc883_3ST_6ch_clevo_modes, + .need_dac_fix = 1, + .input_mux = &alc883_capture_source, + }, [ALC883_CLEVO_M720] = { .mixers = { alc883_clevo_m720_mixer }, .init_verbs = { alc883_init_verbs, alc883_clevo_m720_verbs }, -- 1.6.3.3

2 1

[alsa-devel] Bug-Report: IDT 92HD75B3X5 - Headphone jacks not working
by Ronald Schmitt 06 Sep '09

06 Sep '09

Problem: Headphone-Jacks are not working. There is no sound output and the built-in speakers don't shut down when plugging a headphone in. Tried several model options with snd-hda-intel options, with "model=hp-dv5" at least I get all speakers working, but still no headphones. System: Hardware: HP Pavilion DV7 2140 Notebook Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03) card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Codec: IDT 92HD75B3X5 Kernel: 2.6.30 upload=true&script=true&cardinfo= !!################################ !!ALSA Information Script v 0.4.58 !!################################ !!Script ran on: Fri Sep 4 10:58:37 UTC 2009 !!Linux Distribution !!------------------ Arch Linux \r (\n) (\l) !!DMI Information !!--------------- Manufacturer: Hewlett-Packard Product Name: HP Pavilion dv7 Notebook PC !!Kernel Information !!------------------ Kernel release: 2.6.30-ARCH Operating System: GNU/Linux Architecture: x86_64 Processor: Intel(R) Core(TM)2 Quad CPU Q9000 @ 2.00GHz SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.20 Library version: 1.0.20 Utilities version: 1.0.20 !!Loaded ALSA modules !!------------------- snd_hda_intel snd_hda_intel !!Sound Servers on this system !!---------------------------- ESound Daemon: Installed - Yes (/usr/bin/esd) Running - No Jack: Installed - Yes (/usr/bin/jackd) Running - No !!Soundcards recognised by ALSA !!----------------------------- 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xda100000 irq 22 1 [HDMI ]: HDA-Intel - HDA ATI HDMI HDA ATI HDMI at 0xda010000 irq 17 !!PCI Soundcards installed in the system !!-------------------------------------- 00:1b.0 Audio device: Intel Corporation 82801I (ICH9 Family) HD Audio Controller (rev 03) 01:00.1 Audio device: ATI Technologies Inc R700 Audio Device [Radeon HD 4000 Series] !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 00:1b.0 0403: 8086:293e (rev 03) Subsystem: 103c:3624 -- 01:00.1 0403: 1002:aa38 Subsystem: 103c:3624 !!Modprobe options (Sound related) !!-------------------------------- snd-hda-intel: model=hp-dv5 !!Loaded sound module options !!-------------------------- !!Module: snd_hda_intel bdl_pos_adj : 1,32,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : 0 id : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 model : hp-dv5,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> position_fix : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N single_cmd : N !!Module: snd_hda_intel bdl_pos_adj : 1,32,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 enable : Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y,Y enable_msi : 0 id : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> index : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 model : hp-dv5,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> position_fix : 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1,-1 probe_only : N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N,N single_cmd : N !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: IDT 92HD75B3X5 Address: 0 Function Id: 0x1 Vendor Id: 0x111d7603 Subsystem Id: 0x103c3624 Revision Id: 0x100202 No Modem Function Group found Default PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Default Amp-Out caps: ofs=0x7f, nsteps=0x7f, stepsize=0x02, mute=1 GPIO: io=8, o=0, i=0, unsolicited=1, wake=1 IO[0]: enable=1, dir=1, wake=0, sticky=0, data=1, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[2]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[3]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[4]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[5]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[6]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 IO[7]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Power-Map: 0x00 Analog Loopback: 0x00 Node 0x0a [Pin Complex] wcaps 0x400181: Stereo Pincap 0x0000001c: OUT HP Detect Pin Default 0x40f100f0: [N/A] Other at Ext N/A Conn = 1/8, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x00: Unsolicited: tag=00, enabled=0 Connection: 3 0x10 0x11 0x17* Node 0x0b [Pin Complex] wcaps 0x400081: Stereo Pincap 0x00001724: IN Detect Vref caps: HIZ 50 GRD 80 Pin Default 0x02a11020: [Jack] Mic at Ext Front Conn = 1/8, Color = Black DefAssociation = 0x2, Sequence = 0x0 Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=01, enabled=1 Node 0x0c [Pin Complex] wcaps 0x400081: Stereo Pincap 0x00001724: IN Detect Vref caps: HIZ 50 GRD 80 Pin Default 0x40f000f1: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x1 Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Node 0x0d [Pin Complex] wcaps 0x400181: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x90170010: [Fixed] Speaker at Int N/A Conn = Analog, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 3 0x10 0x11* 0x17 Node 0x0e [Pin Complex] wcaps 0x400081: Stereo Pincap 0x00001724: IN Detect Vref caps: HIZ 50 GRD 80 Pin Default 0x40f100f2: [N/A] Other at Ext N/A Conn = 1/8, Color = Unknown DefAssociation = 0xf, Sequence = 0x2 Pin-ctls: 0x00: VREF_HIZ Unsolicited: tag=00, enabled=0 Node 0x0f [Pin Complex] wcaps 0x400181: Stereo Pincap 0x00000014: OUT Detect Pin Default 0x02014040: [Jack] Line Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x4, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 3 0x10 0x11 0x17* Node 0x10 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0x5d 0x5d] Converter: stream=5, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x11 [Audio Output] wcaps 0xd0c05: Stereo Amp-Out R/L Amp-Out caps: N/A Amp-Out vals: [0x63 0x63] Converter: stream=5, channel=0 Power: setting=D0, actual=D0 Delay: 13 samples Node 0x12 [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 SDI-Select: 0 Power: setting=D3, actual=D3 Delay: 13 samples Connection: 1 0x1c Processing caps: benign=0, ncoeff=0 Node 0x13 [Audio Input] wcaps 0x1d0541: Stereo Converter: stream=0, channel=0 SDI-Select: 0 Power: setting=D3, actual=D3 Delay: 13 samples Connection: 1 0x1d Processing caps: benign=0, ncoeff=0 Node 0x14 [Pin Complex] wcaps 0x400100: Mono Pincap 0x00000030: IN OUT Pin Default 0x40f000f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x00: Connection: 1 0x16 Node 0x15 [Audio Selector] wcaps 0x300101: Stereo Connection: 3 0x10 0x11 0x17* Node 0x16 [Audio Mixer] wcaps 0x200100: Mono Connection: 1 0x15 Node 0x17 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x97 0x97] [0x97 0x97] [0x97 0x97] [0x97 0x97] [0x97 0x97] Connection: 5 0x10 0x11 0x14 0x1a 0x1b Node 0x18 [Pin Complex] wcaps 0x40000b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00000020: IN Pin Default 0x90a60350: [Fixed] Mic at Int N/A Conn = Digital, Color = Unknown DefAssociation = 0x5, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x19 [Pin Complex] wcaps 0x40000b: Stereo Amp-In Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Pincap 0x00000020: IN Pin Default 0x40f000f4: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x4 Pin-ctls: 0x00: Node 0x1a [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Connection: 3 0x0b* 0x0c 0x0e Node 0x1b [Audio Selector] wcaps 0x30010d: Stereo Amp-Out Amp-Out caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-Out vals: [0x00 0x00] Connection: 3 0x0b* 0x0c 0x0e Node 0x1c [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 Amp-Out vals: [0x88 0x88] Connection: 4 0x1a* 0x17 0x18 0x19 Node 0x1d [Audio Selector] wcaps 0x30090d: Stereo Amp-Out R/L Amp-Out caps: ofs=0x00, nsteps=0x0f, stepsize=0x05, mute=1 Amp-Out vals: [0x88 0x88] Connection: 4 0x1b* 0x17 0x18 0x19 Node 0x1e [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x024511a0: [Jack] SPDIF Out at Ext Front Conn = Optical, Color = Black DefAssociation = 0xa, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x24 Node 0x1f [Pin Complex] wcaps 0x400701: Stereo Digital Pincap 0x00010010: OUT EAPD EAPD 0x0: Pin Default 0x40f000f5: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x5 Pin-ctls: 0x00: Power: setting=D0, actual=D0 Connection: 2 0x24* 0x25 Node 0x20 [Pin Complex] wcaps 0x400301: Stereo Digital Pincap 0x00000010: OUT Pin Default 0x40f000f0: [N/A] Other at Ext N/A Conn = Unknown, Color = Unknown DefAssociation = 0xf, Sequence = 0x0 Pin-ctls: 0x00: Connection: 1 0x25 Node 0x21 [Audio Output] wcaps 0x40211: Stereo Digital Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Node 0x22 [Audio Output] wcaps 0x40211: Stereo Digital Converter: stream=0, channel=0 Digital: Enabled Digital category: 0x0 PCM: rates [0x7e0]: 44100 48000 88200 96000 176400 192000 bits [0xe]: 16 20 24 formats [0x5]: PCM AC3 Delay: 4 samples Node 0x23 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x24 [Audio Selector] wcaps 0x300101: Stereo Connection: 3 0x21* 0x1c 0x1d Node 0x25 [Audio Selector] wcaps 0x300101: Stereo Connection: 3 0x22* 0x1c 0x1d Node 0x26 [Beep Generator Widget] wcaps 0x70000c: Mono Amp-Out Amp-Out caps: ofs=0x03, nsteps=0x03, stepsize=0x17, mute=1 Amp-Out vals: [0x00] Node 0x27 [Volume Knob Widget] wcaps 0x600000: Mono Volume-Knob: delta=1, steps=127, direct=0, val=127 Connection: 2 0x10 0x11 Codec: ATI R6xx HDMI Address: 0 Function Id: 0x1 Vendor Id: 0x1002aa01 Subsystem Id: 0x00aa0100 Revision Id: 0x100100 No Modem Function Group found Default PCM: rates [0x40]: 48000 bits [0x6]: 16 20 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=0, o=0, i=0, unsolicited=0, wake=0 Node 0x02 [Audio Output] wcaps 0x201: Stereo Digital Converter: stream=0, channel=0 Digital: Digital category: 0x0 Node 0x03 [Pin Complex] wcaps 0x400381: Stereo Digital Pincap 0x00000094: OUT Detect HDMI Pin Default 0x18560010: [Jack] Digital Out at Int HDMI Conn = Digital, Color = Unknown DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT Unsolicited: tag=00, enabled=0 Connection: 1 0x02 --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw----+ 1 root audio 116, 7 Sep 4 11:06 /dev/snd/controlC0 crw-rw----+ 1 root audio 116, 10 Sep 4 11:06 /dev/snd/controlC1 crw-rw----+ 1 root audio 116, 6 Sep 4 11:06 /dev/snd/hwC0D0 crw-rw----+ 1 root audio 116, 9 Sep 4 11:06 /dev/snd/hwC1D0 crw-rw----+ 1 root audio 116, 5 Sep 4 11:06 /dev/snd/pcmC0D0c crw-rw----+ 1 root audio 116, 4 Sep 4 12:12 /dev/snd/pcmC0D0p crw-rw----+ 1 root audio 116, 3 Sep 4 11:06 /dev/snd/pcmC0D1p crw-rw----+ 1 root audio 116, 8 Sep 4 11:06 /dev/snd/pcmC1D3p crw-rw----+ 1 root audio 116, 11 Sep 4 11:06 /dev/snd/seq crw-rw----+ 1 root audio 116, 2 Sep 4 11:06 /dev/snd/timer /dev/snd/by-path: total 0 drwxr-xr-x 2 root root 80 Sep 4 11:06 . drwxr-xr-x 3 root root 260 Sep 4 11:06 .. lrwxrwxrwx 1 root root 12 Sep 4 11:06 pci-0000:00:1b.0 -> ../controlC0 lrwxrwxrwx 1 root root 12 Sep 4 11:06 pci-0000:01:00.1 -> ../controlC1 !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 1: STAC92xx Digital [STAC92xx Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 1: HDMI [HDA ATI HDMI], device 3: ATI HDMI [ATI HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 !!Amixer output !!------------- !!-------Mixer controls for card 0 [Intel] Card hw:0 'Intel'/'HDA Intel at 0xda100000 irq 22' Mixer name : 'IDT 92HD75B3X5' Components : 'HDA:111d7603,103c3624,00100202' Controls : 32 Simple ctrls : 19 Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 49 [77%] [-11.25dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] Simple mixer control 'Front',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 45 [70%] [-14.25dB] [on] Front Right: Playback 45 [70%] [-14.25dB] [on] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958 Default PCM',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958 Playback Source',0 Capabilities: enum Items: 'Digital Playback' 'Analog Mux 1' 'Analog Mux 2' Item0: 'Digital Playback' Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 15 Front Left: Capture 8 [53%] [12.00dB] [off] Front Right: Capture 8 [53%] [12.00dB] [off] Simple mixer control 'Capture',1 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 15 Front Left: Capture 8 [53%] [12.00dB] [off] Front Right: Capture 8 [53%] [12.00dB] [off] Simple mixer control 'DAC0',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 23 [74%] [0.00dB] [off] Front Right: Capture 23 [74%] [0.00dB] [off] Simple mixer control 'DAC1',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 23 [74%] [0.00dB] [off] Front Right: Capture 23 [74%] [0.00dB] [off] Simple mixer control 'Digital',0 Capabilities: cvolume Capture channels: Front Left - Front Right Limits: Capture 0 - 120 Front Left: Capture 60 [50%] [0.00dB] Front Right: Capture 60 [50%] [0.00dB] Simple mixer control 'Digital Input Source',0 Capabilities: enum Items: 'Analog Inputs' 'Mixer' 'Digital Mic 1' Item0: 'Analog Inputs' Simple mixer control 'Digital Input Source',1 Capabilities: enum Items: 'Analog Inputs' 'Mixer' 'Digital Mic 1' Item0: 'Analog Inputs' Simple mixer control 'Import0 Mux',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 23 [74%] [0.00dB] [off] Front Right: Capture 23 [74%] [0.00dB] [off] Simple mixer control 'Import1 Mux',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 31 Front Left: Capture 23 [74%] [0.00dB] [off] Front Right: Capture 23 [74%] [0.00dB] [off] Simple mixer control 'Mux',0 Capabilities: cvolume Capture channels: Front Left - Front Right Limits: Capture 0 - 3 Front Left: Capture 0 [0%] [0.00dB] Front Right: Capture 0 [0%] [0.00dB] Simple mixer control 'Mux',1 Capabilities: cvolume Capture channels: Front Left - Front Right Limits: Capture 0 - 3 Front Left: Capture 0 [0%] [0.00dB] Front Right: Capture 0 [0%] [0.00dB] Simple mixer control 'PC Beep',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 3 Mono: Playback 1 [33%] [-12.00dB] [off] Simple mixer control 'Speaker',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 51 [80%] [-9.75dB] [on] Front Right: Playback 51 [80%] [-9.75dB] [on] !!-------Mixer controls for card 1 [HDMI] Card hw:1 'HDMI'/'HDA ATI HDMI at 0xda010000 irq 17' Mixer name : 'ATI R6xx HDMI' Components : 'HDA:1002aa01,00aa0100,00100100' Controls : 4 Simple ctrls : 1 Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] !!Alsactl output !!------------- --startcollapse-- state.Intel { control.1 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 15' comment.dbmin 0 comment.dbmax 2250 iface MIXER name 'Capture Volume' value.0 8 value.1 8 } control.2 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Capture Switch' value.0 false value.1 false } control.3 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 15' comment.dbmin 0 comment.dbmax 2250 iface MIXER name 'Capture Volume' index 1 value.0 8 value.1 8 } control.4 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Capture Switch' index 1 value.0 false value.1 false } control.5 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Import0 Mux Capture Switch' value.0 false value.1 false } control.6 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Import0 Mux Capture Volume' value.0 23 value.1 23 } control.7 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Import1 Mux Capture Switch' value.0 false value.1 false } control.8 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Import1 Mux Capture Volume' value.0 23 value.1 23 } control.9 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'DAC0 Capture Switch' value.0 false value.1 false } control.10 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'DAC0 Capture Volume' value.0 23 value.1 23 } control.11 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'DAC1 Capture Switch' value.0 false value.1 false } control.12 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'DAC1 Capture Volume' value.0 23 value.1 23 } control.13 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 64' comment.dbmin -4800 comment.dbmax 0 iface MIXER name 'Front Playback Volume' value.0 45 value.1 45 } control.14 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Front Playback Switch' value.0 true value.1 true } control.15 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'PC Beep Playback Switch' value false } control.16 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 3' comment.dbmin -1800 comment.dbmax 0 iface MIXER name 'PC Beep Playback Volume' value 1 } control.17 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 64' comment.dbmin -4800 comment.dbmax 0 iface MIXER name 'Speaker Playback Volume' value.0 51 value.1 51 } control.18 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Speaker Playback Switch' value.0 true value.1 true } control.19 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 3' comment.dbmin 0 comment.dbmax 3000 iface MIXER name 'Mux Capture Volume' value.0 0 value.1 0 } control.20 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 3' comment.dbmin 0 comment.dbmax 3000 iface MIXER name 'Mux Capture Volume' index 1 value.0 0 value.1 0 } control.21 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 'Analog Inputs' comment.item.1 Mixer comment.item.2 'Digital Mic 1' iface MIXER name 'Digital Input Source' value 'Analog Inputs' } control.22 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 'Analog Inputs' comment.item.1 Mixer comment.item.2 'Digital Mic 1' iface MIXER name 'Digital Input Source' index 1 value 'Analog Inputs' } control.23 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 'Digital Playback' comment.item.1 'Analog Mux 1' comment.item.2 'Analog Mux 2' iface MIXER name 'IEC958 Playback Source' value 'Digital Playback' } control.24 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.25 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.26 { comment.access 'read write' comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.27 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'IEC958 Playback Switch' value true } control.28 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'IEC958 Default PCM Playback Switch' value true } control.29 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 64' comment.dbmin -4800 comment.dbmax 0 iface MIXER name 'Master Playback Volume' value 49 } control.30 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Master Playback Switch' value true } control.31 { comment.access 'read write user' comment.type INTEGER comment.count 2 comment.range '0 - 255' comment.tlv '0000000100000008ffffec1400000014' comment.dbmin -5100 comment.dbmax 0 iface MIXER name 'PCM Playback Volume' value.0 255 value.1 255 } control.32 { comment.access 'read write user' comment.type INTEGER comment.count 2 comment.range '0 - 120' comment.tlv '0000000100000008fffff44800000032' comment.dbmin -3000 comment.dbmax 3000 iface MIXER name 'Digital Capture Volume' value.0 60 value.1 60 } } state.HDMI { control.1 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.2 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.3 { comment.access 'read write' comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.4 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'IEC958 Playback Switch' value false } } --endcollapse-- !!All Loaded Modules !!------------------ Module usbhid hid cryptd aes_x86_64 aes_generic nfsd exportfs nfs lockd fscache nfs_acl auth_rpcgss sunrpc ipv6 arc4 ecb uvcvideo videodev v4l1_compat snd_seq_dummy iwlagn v4l2_compat_ioctl32 snd_seq_oss fan iwlcore snd_seq_midi_event snd_seq snd_seq_device snd_hda_codec_atihdmi cpufreq_ondemand rfkill snd_hda_codec_idt snd_pcm_oss snd_mixer_oss mac80211 jmb38x_ms i2c_i801 joydev snd_hda_intel snd_hda_codec acpi_cpufreq sdhci_pci sdhci snd_hwdep snd_pcm snd_timer snd psmouse uhci_hcd ehci_hcd iTCO_wdt freq_table hp_accel lis3lv02d mmc_core cfg80211 soundcore snd_page_alloc serio_raw pcspkr memstick ohci1394 wmi intel_agp i2c_core iTCO_vendor_support usbcore r8169 mii thermal video output sg led_class ac processor evdev battery ieee1394 button coretemp rtc_cmos rtc_core rtc_lib ext4 mbcache jbd2 crc16 sr_mod sd_mod cdrom ahci libata scsi_mod !!Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x0a 0x40f100f0 0x0b 0x02a11020 0x0c 0x40f000f1 0x0d 0x90170110 0x0e 0x40f100f2 0x0f 0x02014040 0x14 0x40f000f0 0x18 0x90a60350 0x19 0x40f000f4 0x1e 0x024511a0 0x1f 0x40f000f5 0x20 0x40f000f0 /sys/class/sound/hwC0D0/driver_pin_configs: 0x0d 0x90170010 /sys/class/sound/hwC0D0/user_pin_configs: /sys/class/sound/hwC0D0/init_verbs: /sys/class/sound/hwC1D0/init_pin_configs: 0x03 0x18560010 /sys/class/sound/hwC1D0/driver_pin_configs: /sys/class/sound/hwC1D0/user_pin_configs: /sys/class/sound/hwC1D0/init_verbs: !!ALSA/HDA dmesg !!------------------ usb 2-4: new high speed USB device using ehci_hcd and address 2 HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 HDA Intel 0000:00:1b.0: setting latency timer to 64 jmb38x_ms 0000:06:00.3: PCI INT A -> GSI 16 (level, low) -> IRQ 16 -- usb 2-4: configuration #1 chosen from 1 choice input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:1b.0/input/input9 input: HDA Intel Mic at Ext Front Jack as /devices/pci0000:00/0000:00:1b.0/input/input10 input: HDA Intel Line Out at Ext Front Jack as /devices/pci0000:00/0000:00:1b.0/input/input11 HDA Intel 0000:01:00.1: PCI INT B -> GSI 17 (level, low) -> IRQ 17 HDA Intel 0000:01:00.1: setting latency timer to 64 i801_smbus 0000:00:1f.3: PCI INT C -> GSI 18 (level, low) -> IRQ 18

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[alsa-devel] Kernel panic with alsa-driver 1.0.21 and RHEL 5
by Paulo Cavalcanti 06 Sep '09

06 Sep '09

Hi, I upgraded to alsa 1.0.21 on Fedora 10/11 with no problem at all. However, I always get a kernel panic when I install alsa-driver 1.0.21 on RHEL 5 (either i386 or x86_63) and reboot: I have no idea on how to debug this situation. The memory dump references: snd_hda_codec .... snd_hda_intel ... RHEL 5 runs just fine with alsa-driver 1.0.20, thought. Thanks. -- Paulo Roma Cavalcanti LCG - UFRJ

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[alsa-devel] Check out my photos on Facebook
by Justin P. Mattock 05 Sep '09

05 Sep '09

I set up a Facebook profile where I can post my pictures, videos and events and I want to add you as a friend so you can see it. First, you need to join Facebook! Once you join, you can also create your own profile. Thanks, Justin To sign up for Facebook, follow the link below: http://www.facebook.com/p.php?i=1358683573&k=Z31T6VT5R4TF6BD1Q142YPT2RT&r alsa-devel(a)alsa-project.org was invited to join Facebook by Justin P. Mattock. If you do not wish to receive this type of email from Facebook in the future, please click on the link below to unsubscribe. http://www.facebook.com/o.php?k=15ec76&u=100000227845811&mid=10cc736G5af31e… Facebook's offices are located at 1601 S. California Ave., Palo Alto, CA 94304.

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Re: [alsa-devel] [Pkg-alsa-devel] Bug#544607: /usr/share/alsa/init/default:52: missing closing brace for format
by Elimar Riesebieter 05 Sep '09

05 Sep '09

forward 544607 alsa-devel(a)alsa-project.org thanks * Philipp Jocham [090901 22:26 +0200] > Package: alsa-utils > Version: 1.0.20-2 > Severity: normal > > > $ alsactl init > Unknown hardware: "HDA-Intel" "Analog Devices AD1884" "HDA:11d41884,103c3029,00100100" "0x103c" "0x3029" > Hardware is initialized using a guess method > /usr/share/alsa/init/default:51: control element not found > /usr/share/alsa/init/default:52: missing closing brace for format > /usr/share/alsa/init/default:52: error parsing CTL attribute > /usr/share/alsa/init/default:52: invalid rule > > It's just a small typo: > diff -r -U1 alsa-utils-1.0.20/alsactl/init/default alsa-utils-1.0.20.1/alsactl/init/default > --- alsa-utils-1.0.20/alsactl/init/default 2009-05-06 09:07:24.000000000 +0200 > +++ alsa-utils-1.0.20.1/alsactl/init/default 2009-09-01 22:13:16.832163534 +0200 > @@ -51,3 +51,3 @@ > ENV{has_pmaster_vol}=="true",CTL{values}="0dB",RESULT=="0",GOTO="headphone0_end" > -ENV{has_pmaster_vol}=="true",CTL{values)="100%",GOTO="headphone0_end" > +ENV{has_pmaster_vol}=="true",CTL{values}="100%",GOTO="headphone0_end" > CTL{values}="$env{pvolume}",RESULT!="0",CTL{values}="$env{ppercent}" > > Sound was very quiet per default. > > Regards, > Philipp Thanks a lot! will be patched at Debian 1.0.21 upload and hopefully recognized by alsa-devel ;-) Elimar -- We all know Linux is great... it does infinite loops in 5 seconds. -- Linus Torvalds

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[alsa-devel] ASoC updates for 2.6.32
by Mark Brown 05 Sep '09

05 Sep '09

The following changes since commit 5c0d38c9478e79ab7deb1b470dd181d2308a608e: jassi brar (1): ASoC: Debugged improper setting of PLL fields in WM8580 driver are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.32 Mark Brown (1): ASoC: Fully specify DC servo bits to update in wm_hubs Robert Schwebel (1): ASoC: fix pxa2xx-ac97.c breakage sound/arm/pxa2xx-ac97.c | 2 +- sound/soc/codecs/wm_hubs.c | 12 ++---------- 2 files changed, 3 insertions(+), 11 deletions(-)

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[alsa-devel] [PATCH 1/2] ASoC: DaVinci: pcm, rename variables in prep for ping/pong
by Troy Kisky 05 Sep '09

05 Sep '09

Rename variable master_lch to asp_channel Rename variable slave_lch to asp_link[0] Rename local variables: lch to link count to asp_count src to asp_src dst to asp_dst Signed-off-by: Troy Kisky <troy.kisky(a)boundarydevices.com> --- This goes on top of Kevin temp/asoc branch plus [PATCH 1/3] ASoC: DaVinci: i2s, reduce underruns by combining into 1 element Changes since last version: removing lch references as suggested by David Brownell sound/soc/davinci/davinci-pcm.c | 64 +++++++++++++++++++------------------- 1 files changed, 32 insertions(+), 32 deletions(-) diff --git a/sound/soc/davinci/davinci-pcm.c b/sound/soc/davinci/davinci-pcm.c index 2f7da49..a0a977e 100644 --- a/sound/soc/davinci/davinci-pcm.c +++ b/sound/soc/davinci/davinci-pcm.c @@ -51,8 +51,8 @@ static struct snd_pcm_hardware davinci_pcm_hardware = { struct davinci_runtime_data { spinlock_t lock; int period; /* current DMA period */ - int master_lch; /* Master DMA channel */ - int slave_lch; /* linked parameter RAM reload slot */ + int asp_channel; /* Master DMA channel */ + int asp_link[2]; /* asp parameter link channel, ping/pong */ struct davinci_pcm_dma_params *params; /* DMA params */ }; @@ -60,7 +60,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) { struct davinci_runtime_data *prtd = substream->runtime->private_data; struct snd_pcm_runtime *runtime = substream->runtime; - int lch = prtd->slave_lch; + int link = prtd->asp_link[0]; unsigned int period_size; unsigned int dma_offset; dma_addr_t dma_pos; @@ -75,7 +75,7 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) dma_pos = runtime->dma_addr + dma_offset; pr_debug("davinci_pcm: audio_set_dma_params_play channel = %d " - "dma_ptr = %x period_size=%x\n", lch, dma_pos, period_size); + "dma_ptr = %x period_size=%x\n", link, dma_pos, period_size); data_type = prtd->params->data_type; count = period_size / data_type; @@ -93,23 +93,23 @@ static void davinci_pcm_enqueue_dma(struct snd_pcm_substream *substream) } acnt = prtd->params->acnt; - edma_set_src(lch, src, INCR, W8BIT); - edma_set_dest(lch, dst, INCR, W8BIT); - edma_set_src_index(lch, src_bidx, 0); - edma_set_dest_index(lch, dst_bidx, 0); - edma_set_transfer_params(lch, acnt, count, 1, 0, ASYNC); + edma_set_src(link, src, INCR, W8BIT); + edma_set_dest(link, dst, INCR, W8BIT); + edma_set_src_index(link, src_bidx, 0); + edma_set_dest_index(link, dst_bidx, 0); + edma_set_transfer_params(link, acnt, count, 1, 0, ASYNC); prtd->period++; if (unlikely(prtd->period >= runtime->periods)) prtd->period = 0; } -static void davinci_pcm_dma_irq(unsigned lch, u16 ch_status, void *data) +static void davinci_pcm_dma_irq(unsigned link, u16 ch_status, void *data) { struct snd_pcm_substream *substream = data; struct davinci_runtime_data *prtd = substream->runtime->private_data; - pr_debug("davinci_pcm: lch=%d, status=0x%x\n", lch, ch_status); + pr_debug("davinci_pcm: link=%d, status=0x%x\n", link, ch_status); if (unlikely(ch_status != DMA_COMPLETE)) return; @@ -142,15 +142,15 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) EVENTQ_0); if (ret < 0) return ret; - prtd->master_lch = ret; + prtd->asp_channel = ret; /* Request parameter RAM reload slot */ - ret = edma_alloc_slot(EDMA_CTLR(prtd->master_lch), EDMA_SLOT_ANY); + ret = edma_alloc_slot(EDMA_CTLR(prtd->asp_channel), EDMA_SLOT_ANY); if (ret < 0) { - edma_free_channel(prtd->master_lch); + edma_free_channel(prtd->asp_channel); return ret; } - prtd->slave_lch = ret; + prtd->asp_link[0] = ret; /* Issue transfer completion IRQ when the channel completes a * transfer, then always reload from the same slot (by a kind @@ -161,10 +161,10 @@ static int davinci_pcm_dma_request(struct snd_pcm_substream *substream) * the buffer and its length (ccnt) ... use it as a template * so davinci_pcm_enqueue_dma() takes less time in IRQ. */ - edma_read_slot(prtd->slave_lch, &p_ram); - p_ram.opt |= TCINTEN | EDMA_TCC(EDMA_CHAN_SLOT(prtd->master_lch)); - p_ram.link_bcntrld = EDMA_CHAN_SLOT(prtd->slave_lch) << 5; - edma_write_slot(prtd->slave_lch, &p_ram); + edma_read_slot(prtd->asp_link[0], &p_ram); + p_ram.opt |= TCINTEN | EDMA_TCC(EDMA_CHAN_SLOT(prtd->asp_channel)); + p_ram.link_bcntrld = EDMA_CHAN_SLOT(prtd->asp_link[0]) << 5; + edma_write_slot(prtd->asp_link[0], &p_ram); return 0; } @@ -180,12 +180,12 @@ static int davinci_pcm_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_RESUME: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: - edma_start(prtd->master_lch); + edma_start(prtd->asp_channel); break; case SNDRV_PCM_TRIGGER_STOP: case SNDRV_PCM_TRIGGER_SUSPEND: case SNDRV_PCM_TRIGGER_PAUSE_PUSH: - edma_stop(prtd->master_lch); + edma_stop(prtd->asp_channel); break; default: ret = -EINVAL; @@ -206,8 +206,8 @@ static int davinci_pcm_prepare(struct snd_pcm_substream *substream) davinci_pcm_enqueue_dma(substream); /* Copy self-linked parameter RAM entry into master channel */ - edma_read_slot(prtd->slave_lch, &temp); - edma_write_slot(prtd->master_lch, &temp); + edma_read_slot(prtd->asp_link[0], &temp); + edma_write_slot(prtd->asp_channel, &temp); davinci_pcm_enqueue_dma(substream); return 0; @@ -219,20 +219,20 @@ davinci_pcm_pointer(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct davinci_runtime_data *prtd = runtime->private_data; unsigned int offset; - dma_addr_t count; - dma_addr_t src, dst; + int asp_count; + dma_addr_t asp_src, asp_dst; spin_lock(&prtd->lock); - edma_get_position(prtd->master_lch, &src, &dst); + edma_get_position(prtd->asp_channel, &asp_src, &asp_dst); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - count = src - runtime->dma_addr; + asp_count = asp_src - runtime->dma_addr; else - count = dst - runtime->dma_addr; + asp_count = asp_dst - runtime->dma_addr; spin_unlock(&prtd->lock); - offset = bytes_to_frames(runtime, count); + offset = bytes_to_frames(runtime, asp_count); if (offset >= runtime->buffer_size) offset = 0; @@ -274,10 +274,10 @@ static int davinci_pcm_close(struct snd_pcm_substream *substream) struct snd_pcm_runtime *runtime = substream->runtime; struct davinci_runtime_data *prtd = runtime->private_data; - edma_unlink(prtd->slave_lch); + edma_unlink(prtd->asp_link[0]); - edma_free_slot(prtd->slave_lch); - edma_free_channel(prtd->master_lch); + edma_free_slot(prtd->asp_link[0]); + edma_free_channel(prtd->asp_channel); kfree(prtd); -- 1.5.6.3

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Re: [alsa-devel] [Pkg-alsa-devel] Bug#545092: amixer -h doesn't show the whole set of commands
by jidanni＠jidanni.org 05 Sep '09

05 Sep '09

>> $ amixer -h >> doesn't show the "info" command. But it is listed on the man page. >> There maybe more missing too. ER> Please send a patch to alsa-devel(a)alsa-project.org. They will apply ER> your patch. Sounds like work :-( I'll just alert them upstream by CCing this message. Hope it doesn't bounce.

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