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[PATCH] ASoC: Make WM8580 DAI clocking more standard
by Mark Brown 14 Sep '09

14 Sep '09

Move the WM8580 driver over to more of a standard clocking setup method, specifying the master clock rate for the DAI using the set_sysclk() API and configuring the LRCLK automatically using that. This is an initial update only to present external APIs, further work that needs doing includes enforcing the constraints flowing from the SYSCLK selection, automatic tie in with PLL configuration and BCLK configuration (the current BCLK configuration is compatible with all sample rates). Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8580.c | 121 +++++++++++++++++++++++++++++++-------------- sound/soc/codecs/wm8580.h | 4 +- 2 files changed, 85 insertions(+), 40 deletions(-) diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c index 6bded8c..dffc093 100644 --- a/sound/soc/codecs/wm8580.c +++ b/sound/soc/codecs/wm8580.c @@ -105,19 +105,18 @@ #define WM8580_PLLB4_CLKOUTSRC_OSCCLK 0x180 /* CLKSEL (register 8h) */ -#define WM8580_CLKSEL_DAC_CLKSEL_MASK 0x03 -#define WM8580_CLKSEL_DAC_CLKSEL_PLLA 0x01 -#define WM8580_CLKSEL_DAC_CLKSEL_PLLB 0x02 +#define WM8580_CLKSEL_DAC_CLKSEL_MASK 0x03 +#define WM8580_CLKSEL_DAC_CLKSEL_SHIFT 0 +#define WM8580_CLKSEL_ADC_CLKSEL_MASK 0x0c +#define WM8580_CLKSEL_ADC_CLKSEL_SHIFT 2 + +#define WM8580_CLKSEL_PLLA 0x01 +#define WM8580_CLKSEL_PLLB 0x02 +#define WM8580_CLKSEL_MCLK 0x03 /* AIF control 1 (registers 9h-bh) */ #define WM8580_AIF_RATE_MASK 0x7 -#define WM8580_AIF_RATE_128 0x0 -#define WM8580_AIF_RATE_192 0x1 -#define WM8580_AIF_RATE_256 0x2 -#define WM8580_AIF_RATE_384 0x3 -#define WM8580_AIF_RATE_512 0x4 -#define WM8580_AIF_RATE_768 0x5 -#define WM8580_AIF_RATE_1152 0x6 +#define WM8580_AIF_RATE_SHIFT 0x2 #define WM8580_AIF_BCLKSEL_MASK 0x18 #define WM8580_AIF_BCLKSEL_64 0x00 @@ -203,6 +202,7 @@ struct wm8580_priv { u16 reg_cache[WM8580_MAX_REGISTER + 1]; struct pll_state a; struct pll_state b; + int sysclk[WM8580_NUM_DAI]; }; static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1); @@ -476,6 +476,10 @@ static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai, return 0; } +static int lrclk_ratios[] = { + 128, 192, 256, 384, 512, 768, 1152 +}; + /* * Set PCM DAI bit size and sample rate. */ @@ -483,12 +487,31 @@ static int wm8580_paif_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 snd_soc_codec *codec = dai->codec; + struct wm8580_priv *priv = codec->private_data; u16 paifb = snd_soc_read(codec, WM8580_PAIF3 + dai->id); + int sysclk, i; + + /* SYSCLK needs to be configured prior to this being called. */ + if (!priv->sysclk[dai->id]) { + dev_err(dai->dev, "SYSCLK not configured\n"); + return -EINVAL; + } + sysclk = priv->sysclk[dai->id]; + dev_dbg(dai->dev, "SYSCLK is %dHz\n", sysclk); + + paifb &= ~(WM8580_AIF_LENGTH_MASK | WM8580_AIF_RATE_MASK); + + /* Look for an exact match; the steps are too big otherwise */ + for (i = 0; i < ARRAY_SIZE(lrclk_ratios); i++) { + if (params_rate(params) == sysclk / lrclk_ratios[i]) + break; + } + if (i == ARRAY_SIZE(lrclk_ratios)) + return -EINVAL; + dev_dbg(dai->dev, "LRCLK ratio %d\n", lrclk_ratios[i]); + paifb |= i << WM8580_AIF_RATE_SHIFT; - paifb &= ~WM8580_AIF_LENGTH_MASK; /* bit size */ switch (params_format(params)) { case SNDRV_PCM_FORMAT_S16_LE: @@ -624,28 +647,6 @@ static int wm8580_set_dai_clkdiv(struct snd_soc_dai *codec_dai, snd_soc_write(codec, WM8580_PLLB4, reg); break; - case WM8580_DAC_CLKSEL: - reg = snd_soc_read(codec, WM8580_CLKSEL); - reg &= ~WM8580_CLKSEL_DAC_CLKSEL_MASK; - - switch (div) { - case WM8580_CLKSRC_MCLK: - break; - - case WM8580_CLKSRC_PLLA: - reg |= WM8580_CLKSEL_DAC_CLKSEL_PLLA; - break; - - case WM8580_CLKSRC_PLLB: - reg |= WM8580_CLKSEL_DAC_CLKSEL_PLLB; - break; - - default: - return -EINVAL; - } - snd_soc_write(codec, WM8580_CLKSEL, reg); - break; - case WM8580_CLKOUTSRC: reg = snd_soc_read(codec, WM8580_PLLB4); reg &= ~WM8580_PLLB4_CLKOUTSRC_MASK; @@ -679,6 +680,48 @@ static int wm8580_set_dai_clkdiv(struct snd_soc_dai *codec_dai, return 0; } +static int wm8580_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id, + unsigned int freq, int dir) +{ + struct snd_soc_codec *codec = dai->codec; + struct wm8580_priv *priv = codec->private_data; + int mask, val, ret; + + switch (clk_id) { + case WM8580_CLKSRC_MCLK: + val = WM8580_CLKSEL_MCLK; + break; + case WM8580_CLKSRC_PLLA: + val = WM8580_CLKSEL_PLLA; + break; + case WM8580_CLKSRC_PLLB: + val = WM8580_CLKSEL_PLLB; + break; + default: + return -EINVAL; + } + + switch (dai->id) { + case WM8580_DAI_PAIFRX: + mask = WM8580_CLKSEL_DAC_CLKSEL_MASK; + break; + + case WM8580_DAI_PAIFTX: + mask = WM8580_CLKSEL_ADC_CLKSEL_MASK; + val <<= WM8580_CLKSEL_ADC_CLKSEL_SHIFT; + break; + + default: + return -EINVAL; + } + + ret = snd_soc_update_bits(codec, WM8580_CLKSEL, mask, val); + if (ret == 0) + priv->sysclk[dai->id] = freq; + + return ret; +} + static int wm8580_digital_mute(struct snd_soc_dai *codec_dai, int mute) { struct snd_soc_codec *codec = codec_dai->codec; @@ -736,6 +779,7 @@ static struct snd_soc_dai_ops wm8580_dai_ops_playback = { .set_fmt = wm8580_set_paif_dai_fmt, .set_clkdiv = wm8580_set_dai_clkdiv, .set_pll = wm8580_set_dai_pll, + .set_sysclk = wm8580_set_dai_sysclk, .digital_mute = wm8580_digital_mute, }; @@ -743,13 +787,14 @@ static struct snd_soc_dai_ops wm8580_dai_ops_capture = { .hw_params = wm8580_paif_hw_params, .set_fmt = wm8580_set_paif_dai_fmt, .set_clkdiv = wm8580_set_dai_clkdiv, + .set_sysclk = wm8580_set_dai_sysclk, .set_pll = wm8580_set_dai_pll, }; struct snd_soc_dai wm8580_dai[] = { { .name = "WM8580 PAIFRX", - .id = 0, + .id = WM8580_DAI_PAIFRX, .playback = { .stream_name = "Playback", .channels_min = 1, @@ -761,7 +806,7 @@ struct snd_soc_dai wm8580_dai[] = { }, { .name = "WM8580 PAIFTX", - .id = 1, + .id = WM8580_DAI_PAIFTX, .capture = { .stream_name = "Capture", .channels_min = 2, diff --git a/sound/soc/codecs/wm8580.h b/sound/soc/codecs/wm8580.h index 0dfb5dd..f03ae9e 100644 --- a/sound/soc/codecs/wm8580.h +++ b/sound/soc/codecs/wm8580.h @@ -19,8 +19,7 @@ #define WM8580_PLLB 2 #define WM8580_MCLK 1 -#define WM8580_DAC_CLKSEL 2 -#define WM8580_CLKOUTSRC 3 +#define WM8580_CLKOUTSRC 2 #define WM8580_CLKSRC_MCLK 1 #define WM8580_CLKSRC_PLLA 2 @@ -30,6 +29,7 @@ #define WM8580_DAI_PAIFRX 0 #define WM8580_DAI_PAIFTX 1 +#define WM8580_NUM_DAI 2 extern struct snd_soc_dai wm8580_dai[]; extern struct snd_soc_codec_device soc_codec_dev_wm8580; -- 1.6.3.3

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[alsa-devel] [PATCH 3/3] ASoC: TWL6030: Handle power-up seq completion thru audio interrupt
by Lopez Cruz, Misael 14 Sep '09

14 Sep '09

NAUDINT interrupt line is provided by the TWL6030 codec to signal externally events like headset plug/unplug, hook, power-up sequence completion, etc. When the codec is powered-up through external AUDPWRON line it will start its power-up sequence. The completion of the sequence is signaled through the audio interrupt, and then codec is operational. CODEC driver starts a wait_for_completion just after calling external power-up callback. It's signaled as complete when servicing READYINT interrupt. MACHINE drivers should request IRQ line used in corresponding hardware platform and initialize workqueue and completion structs of twl6030_setup_data as well. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl6030.c | 103 +++++++++++++++++++++++++++++++++++++++++--- sound/soc/codecs/twl6030.h | 17 +++++++ 2 files changed, 113 insertions(+), 7 deletions(-) diff --git a/sound/soc/codecs/twl6030.c b/sound/soc/codecs/twl6030.c index 0f2269e..dc85399 100644 --- a/sound/soc/codecs/twl6030.c +++ b/sound/soc/codecs/twl6030.c @@ -43,6 +43,7 @@ /* codec private data */ struct twl6030_priv_data { + int codec_powered; unsigned int sysclk; }; @@ -54,7 +55,7 @@ static const u8 twl6030_reg[TWL6030_CACHEREGNUM] = { 0x4B, /* TWL6030_ASICID (ro) 0x01 */ 0x00, /* TWL6030_ASICREV (ro) 0x02 */ 0x00, /* TWL6030_INTID 0x03 */ - 0x41, /* TWL6030_INTMR 0x04 */ + 0x00, /* TWL6030_INTMR 0x04 */ 0x00, /* TWL6030_NCPCTRL 0x05 */ 0x00, /* TWL6030_LDOCTL 0x06 */ 0x00, /* TWL6030_HPPLLCTL 0x07 */ @@ -127,6 +128,23 @@ static inline void twl6030_write_reg_cache(struct snd_soc_codec *codec, } /* + * read from twl6030 hardware register + */ +static int twl6030_read(struct snd_soc_codec *codec, + unsigned int reg) +{ + u8 value; + + if (reg > TWL6030_CACHEREGNUM) + return -EIO; + + twl_i2c_read_u8(TWL6030_MODULE_AUDIO, &value, reg); + twl6030_write_reg_cache(codec, reg, value); + + return value; +} + +/* * write to the twl6030 register space */ static int twl6030_write(struct snd_soc_codec *codec, @@ -159,27 +177,89 @@ static void twl6030_init_chip(struct snd_soc_codec *codec) static void twl6030_power_up(struct snd_soc_codec *codec) { struct snd_soc_device *socdev = codec->socdev; + struct twl6030_priv_data *priv = codec->private_data; struct twl6030_setup_data *setup = socdev->codec_data; + if (priv->codec_powered) + return; + setup->codec_enable(1); + /* wait for ready interrupt */ + wait_for_completion(&setup->ready_completion); + /* sync registers updated during power-up sequence */ - twl6030_write_reg_cache(codec, TWL6030_REG_NCPCTL, 0x81); - twl6030_write_reg_cache(codec, TWL6030_REG_LDOCTL, 0x45); - twl6030_write_reg_cache(codec, TWL6030_REG_LPPLLCTL, 0x01); + twl6030_read(codec, TWL6030_REG_NCPCTL); + twl6030_read(codec, TWL6030_REG_LDOCTL); + twl6030_read(codec, TWL6030_REG_LPPLLCTL); } static void twl6030_power_down(struct snd_soc_codec *codec) { struct snd_soc_device *socdev = codec->socdev; + struct twl6030_priv_data *priv = codec->private_data; struct twl6030_setup_data *setup = socdev->codec_data; setup->codec_enable(0); + udelay(500); /* sync registers updated during power-down sequence */ - twl6030_write_reg_cache(codec, TWL6030_REG_NCPCTL, 0x00); - twl6030_write_reg_cache(codec, TWL6030_REG_LDOCTL, 0x00); - twl6030_write_reg_cache(codec, TWL6030_REG_LPPLLCTL, 0x00); + twl6030_read(codec, TWL6030_REG_NCPCTL); + twl6030_read(codec, TWL6030_REG_LDOCTL); + twl6030_read(codec, TWL6030_REG_LPPLLCTL); + + priv->codec_powered = 0; +} + +/* audio interrupt handler */ +irqreturn_t twl6030_naudint_handler(int irq, void *data) +{ + struct twl6030_setup_data *setup = data; + + schedule_work(&setup->audint_work); + + /* disable audint irq to let workqueue to execute */ + disable_irq_nosync(irq); + + return IRQ_HANDLED; +} + +void twl6030_naudint_work(struct work_struct *work) +{ + struct snd_soc_codec *codec; + struct twl6030_setup_data *setup; + struct twl6030_priv_data *priv; + u8 intid; + + setup = container_of(work, struct twl6030_setup_data, audint_work); + codec = setup->codec; + priv = codec->private_data; + + twl_i2c_read_u8(TWL6030_MODULE_AUDIO, &intid, TWL6030_REG_INTID); + + switch (intid) { + case TWL6030_THINT: + dev_alert(codec->dev, "die temp over-limit detection\n"); + break; + case TWL6030_PLUGINT: + case TWL6030_UNPLUGINT: + case TWL6030_HOOKINT: + break; + case TWL6030_HFINT: + dev_alert(codec->dev, "hf drivers over current detection\n"); + break; + case TWL6030_VIBINT: + dev_alert(codec->dev, "vib drivers over current detection\n"); + break; + case TWL6030_READYINT: + priv->codec_powered = 1; + complete(&setup->ready_completion); + break; + default: + dev_err(codec->dev, "unknown twl6030 audio interrupt\n"); + } + + enable_irq(setup->irq); } /* set headset dac and driver power mode */ @@ -679,6 +759,7 @@ static int twl6030_resume(struct platform_device *pdev) static int twl6030_init(struct snd_soc_device *socdev) { struct snd_soc_codec *codec = socdev->card->codec; + struct twl6030_setup_data *setup = socdev->codec_data; int ret = 0; dev_info(codec->dev, "TWL6030 Audio Codec init\n"); @@ -703,6 +784,14 @@ static int twl6030_init(struct snd_soc_device *socdev) goto pcm_err; } + /* platform setup data is required for power-off/off the device */ + if (setup == NULL) { + printk(KERN_ERR "twl6030: platform setup data missing\n"); + goto pcm_err; + } + + setup->codec = codec; + /* power on device */ twl6030_set_bias_level(codec, SND_SOC_BIAS_STANDBY); diff --git a/sound/soc/codecs/twl6030.h b/sound/soc/codecs/twl6030.h index 9e49c85..1e1fb3f 100644 --- a/sound/soc/codecs/twl6030.h +++ b/sound/soc/codecs/twl6030.h @@ -64,6 +64,16 @@ #define TWL6030_CACHEREGNUM (TWL6030_REG_STATUS + 1) +/* INTID (0x03) fields */ + +#define TWL6030_THINT 0x01 +#define TWL6030_PLUGINT 0x02 +#define TWL6030_UNPLUGINT 0x04 +#define TWL6030_HOOKINT 0x08 +#define TWL6030_HFINT 0x10 +#define TWL6030_VIBINT 0x20 +#define TWL6030_READYINT 0x40 + /* HPPLLCTL (0x07) fields */ #define TWL6030_HPLLENA 0x01 @@ -105,8 +115,15 @@ extern struct snd_soc_dai twl6030_dai; extern struct snd_soc_codec_device soc_codec_dev_twl6030; +irqreturn_t twl6030_naudint_handler(int irq, void *data); +void twl6030_naudint_work(struct work_struct *work); + struct twl6030_setup_data { void (*codec_enable)(int enable); + int irq; + struct work_struct audint_work; + struct completion ready_completion; + struct snd_soc_codec *codec; }; #endif /* End of __TWL6030_H__ */ -- 1.5.4.3

2 1

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

14 Sep '09

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

2 1

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

14 Sep '09

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

1 0

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

14 Sep '09

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

2 1

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

14 Sep '09

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

2 1

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

14 Sep '09

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

2 1

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

14 Sep '09

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

1 0

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

13 Sep '09

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

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

12 Sep '09

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

3 4