May 2011 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] ASoC: Disable WM8994/58 microphone detection over suspend
by Mark Brown 11 May '11

11 May '11

It will be non-functional with the basises and clocks off anyway, if the system needs microphone detection enabled over suspend then it should be causing the CODEC to ignore suspend using the APIs for that to prevent the biases being disabled. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8994.c | 25 +++++++++++++++++++++++++ 1 files changed, 25 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/wm8994.c b/sound/soc/codecs/wm8994.c index b6d47e7..e6dfa10 100644 --- a/sound/soc/codecs/wm8994.c +++ b/sound/soc/codecs/wm8994.c @@ -2416,8 +2416,19 @@ static struct snd_soc_dai_driver wm8994_dai[] = { static int wm8994_suspend(struct snd_soc_codec *codec, pm_message_t state) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + struct wm8994 *control = codec->control_data; int i, ret; + switch (control->type) { + case WM8994: + snd_soc_update_bits(codec, WM8994_MICBIAS, WM8994_MICD_ENA, 0); + break; + case WM8958: + snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, + WM8958_MICD_ENA, 0); + break; + } + for (i = 0; i < ARRAY_SIZE(wm8994->fll); i++) { memcpy(&wm8994->fll_suspend[i], &wm8994->fll[i], sizeof(struct wm8994_fll_config)); @@ -2435,6 +2446,7 @@ static int wm8994_suspend(struct snd_soc_codec *codec, pm_message_t state) static int wm8994_resume(struct snd_soc_codec *codec) { struct wm8994_priv *wm8994 = snd_soc_codec_get_drvdata(codec); + struct wm8994 *control = codec->control_data; int i, ret; unsigned int val, mask; @@ -2473,6 +2485,19 @@ static int wm8994_resume(struct snd_soc_codec *codec) i + 1, ret); } + switch (control->type) { + case WM8994: + if (wm8994->micdet[0].jack || wm8994->micdet[1].jack) + snd_soc_update_bits(codec, WM8994_MICBIAS, + WM8994_MICD_ENA, WM8994_MICD_ENA); + break; + case WM8958: + if (wm8994->jack_cb) + snd_soc_update_bits(codec, WM8958_MIC_DETECT_1, + WM8958_MICD_ENA, WM8958_MICD_ENA); + break; + } + return 0; } #else -- 1.7.5.1

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[alsa-devel] [PATCH] ASoC: Don't squash 16x8 registers down to 8 bits
by Mark Brown 11 May '11

11 May '11

Currently we'll force all registers to fit in 8 bits before passing down to the I/O function. Looks like a cut'n'paste bug. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/soc-cache.c | 1 - 1 files changed, 0 insertions(+), 1 deletions(-) diff --git a/sound/soc/soc-cache.c b/sound/soc/soc-cache.c index bb8e446..40548ce 100644 --- a/sound/soc/soc-cache.c +++ b/sound/soc/soc-cache.c @@ -232,7 +232,6 @@ static int snd_soc_16_8_write(struct snd_soc_codec *codec, unsigned int reg, data[0] = (reg >> 8) & 0xff; data[1] = reg & 0xff; data[2] = value; - reg &= 0xff; return do_hw_write(codec, reg, value, data, 3); } -- 1.7.5.1

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[alsa-devel] [PATCH] ASoC: Fix NULL vs. 0 warning in SSM2602
by Mark Brown 11 May '11

11 May '11

sparse complains if 0 is used as a NULL pointer constant. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/ssm2602.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c index 70099c9..84f4ad5 100644 --- a/sound/soc/codecs/ssm2602.c +++ b/sound/soc/codecs/ssm2602.c @@ -137,7 +137,7 @@ SND_SOC_DAPM_DAC("DAC", "HiFi Playback", SSM2602_PWR, 3, 1), SND_SOC_DAPM_ADC("ADC", "HiFi Capture", SSM2602_PWR, 2, 1), SND_SOC_DAPM_PGA("Line Input", SSM2602_PWR, 0, 1, NULL, 0), -SND_SOC_DAPM_SUPPLY("Digital Core Power", SSM2602_ACTIVE, 0, 0, 0, 0), +SND_SOC_DAPM_SUPPLY("Digital Core Power", SSM2602_ACTIVE, 0, 0, NULL, 0), SND_SOC_DAPM_OUTPUT("LOUT"), SND_SOC_DAPM_OUTPUT("ROUT"), -- 1.7.5.1

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[alsa-devel] a xrun() query
by Santosh Sivaraj 11 May '11

11 May '11

Hi, I am getting glitches while playing audio with aplay. With little debugging I figured out that the state changes from RUNNING to XRUN at which time the trigger gets called. I also figured out that XRUN state is set in the function: static int snd_pcm_update_hw_ptr_post(struct snd_pcm_substream *substream, struct snd_pcm_runtime *runtime) { <snip> } else { if (avail >= runtime->stop_threshold) { xrun(substream); return -EPIPE; } } </snip> } I am not able to find out why the condition (avail > runtime->stop_threshold) becomes true. Can the experts here tell during what use cases/scenarios this happens? Thanks, Santosh -- http://fossix.org, http://dot.slashb.in Luck is a matter of preparation meeting the opportunity. -- Anon

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[alsa-devel] [PATCH 1/7] ASoC: Add control_type in snd_soc_codec
by Dimitris Papastamos 11 May '11

11 May '11

This is mainly used by the soc-cache code to easily determine the currently used underlying serial bus. Set SND_SOC_CUSTOM to 1 so we can distinguish it if it is not initialized or set. Signed-off-by: Dimitris Papastamos <dp(a)opensource.wolfsonmicro.com> --- include/sound/soc.h | 3 ++- 1 files changed, 2 insertions(+), 1 deletions(-) diff --git a/include/sound/soc.h b/include/sound/soc.h index bfa4836..abb56ef 100644 --- a/include/sound/soc.h +++ b/include/sound/soc.h @@ -248,7 +248,7 @@ typedef int (*hw_write_t)(void *,const char* ,int); extern struct snd_ac97_bus_ops soc_ac97_ops; enum snd_soc_control_type { - SND_SOC_CUSTOM, + SND_SOC_CUSTOM = 1, SND_SOC_I2C, SND_SOC_SPI, }; @@ -539,6 +539,7 @@ struct snd_soc_codec { /* codec IO */ void *control_data; /* codec control (i2c/3wire) data */ + enum snd_soc_control_type control_type; hw_write_t hw_write; unsigned int (*hw_read)(struct snd_soc_codec *, unsigned int); unsigned int (*read)(struct snd_soc_codec *, unsigned int); -- 1.7.4.1

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[alsa-devel] fm801: implement TEA575x tuner autodetection
by Ondrej Zary 11 May '11

11 May '11

Autodetect TEA575x tuner connection type during init. This allows tuner to work out-of-the box. tea575x_tuner module parameter remains functional to force tuner type. Tested with SF256-PCP and SF64-PCR. Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- linux-2.6.39-rc2-/sound/pci/fm801.c 2011-05-10 22:31:45.000000000 +0200 +++ linux-2.6.39-rc2/sound/pci/fm801.c 2011-05-10 23:21:42.000000000 +0200 @@ -53,7 +53,7 @@ static int enable[SNDRV_CARDS] = SNDRV_D /* * Enable TEA575x tuner * 1 = MediaForte 256-PCS - * 2 = MediaForte 256-PCPR + * 2 = MediaForte 256-PCP * 3 = MediaForte 64-PCR * 16 = setup tuner only (this is additional bit), i.e. SF64-PCR FM card * High 16-bits are video (radio) device number + 1 @@ -67,7 +67,7 @@ MODULE_PARM_DESC(id, "ID string for the module_param_array(enable, bool, NULL, 0444); MODULE_PARM_DESC(enable, "Enable FM801 soundcard."); module_param_array(tea575x_tuner, int, NULL, 0444); -MODULE_PARM_DESC(tea575x_tuner, "TEA575x tuner access method (1 = SF256-PCS, 2=SF256-PCPR, 3=SF64-PCR, +16=tuner-only)."); +MODULE_PARM_DESC(tea575x_tuner, "TEA575x tuner access method (0 = auto, 1 = SF256-PCS, 2=SF256-PCP, 3=SF64-PCR, 8=disable, +16=tuner-only)."); #define TUNER_ONLY (1<<4) #define TUNER_TYPE_MASK (~TUNER_ONLY & 0xFFFF) @@ -720,12 +720,13 @@ static int __devinit snd_fm801_pcm(struc /* GPIO to TEA575x maps */ struct snd_fm801_tea575x_gpio { u8 data, clk, wren, most; + char *name; }; static struct snd_fm801_tea575x_gpio snd_fm801_tea575x_gpios[] = { - { .data = 1, .clk = 3, .wren = 2, .most = 0 }, /* SF256-PCS */ - { .data = 1, .clk = 0, .wren = 2, .most = 3 }, /* SF256-PCP */ - { .data = 2, .clk = 0, .wren = 1, .most = 3 }, /* SF64-PCR */ + { .data = 1, .clk = 3, .wren = 2, .most = 0, .name = "SF256-PCS" }, + { .data = 1, .clk = 0, .wren = 2, .most = 3, .name = "SF256-PCP" }, + { .data = 2, .clk = 0, .wren = 1, .most = 3, .name = "SF64-PCR" }, }; static void snd_fm801_tea575x_set_pins(struct snd_tea575x *tea, u8 pins) @@ -1229,14 +1230,24 @@ static int __devinit snd_fm801_create(st snd_card_set_dev(card, &pci->dev); #ifdef TEA575X_RADIO + chip->tea.card = card; + chip->tea.freq_fixup = 10700; + chip->tea.private_data = chip; + chip->tea.ops = &snd_fm801_tea_ops; if ((tea575x_tuner & TUNER_TYPE_MASK) > 0 && (tea575x_tuner & TUNER_TYPE_MASK) < 4) { - chip->tea.card = card; - chip->tea.freq_fixup = 10700; - chip->tea.private_data = chip; - chip->tea.ops = &snd_fm801_tea_ops; - snd_tea575x_init(&chip->tea); - } + if (snd_tea575x_init(&chip->tea)) + snd_printk(KERN_ERR "TEA575x radio not found\n"); + } else if ((tea575x_tuner & TUNER_TYPE_MASK) == 0) + /* autodetect tuner connection */ + for (tea575x_tuner = 1; tea575x_tuner <= 3; tea575x_tuner++) { + chip->tea575x_tuner = tea575x_tuner; + if (!snd_tea575x_init(&chip->tea)) { + snd_printk(KERN_INFO "detected TEA575x radio type %s\n", + snd_fm801_tea575x_gpios[tea575x_tuner - 1].name); + break; + } + } #endif *rchip = chip; -- Ondrej Zary

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[alsa-devel] [PATCH] ASoC: Update cx20442 for TTY API change
by Mark Brown 11 May '11

11 May '11

receive_buf() was recently changed to return the number of bytes received but the cx20442 driver wasn't updated to match the new API. I don't have any hardware but since we don't actually appears to be listening to the data at all just report that we accepted all the data that was offered to us. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/cx20442.c | 8 +++++--- 1 files changed, 5 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/cx20442.c b/sound/soc/codecs/cx20442.c index d68ea53..f8c663d 100644 --- a/sound/soc/codecs/cx20442.c +++ b/sound/soc/codecs/cx20442.c @@ -262,14 +262,14 @@ static int v253_hangup(struct tty_struct *tty) } /* Line discipline .receive_buf() */ -static void v253_receive(struct tty_struct *tty, - const unsigned char *cp, char *fp, int count) +static unsigned int v253_receive(struct tty_struct *tty, + const unsigned char *cp, char *fp, int count) { struct snd_soc_codec *codec = tty->disc_data; struct cx20442_priv *cx20442; if (!codec) - return; + return count; cx20442 = snd_soc_codec_get_drvdata(codec); @@ -281,6 +281,8 @@ static void v253_receive(struct tty_struct *tty, codec->hw_write = (hw_write_t)tty->ops->write; codec->card->pop_time = 1; } + + return count; } /* Line discipline .write_wakeup() */ -- 1.7.5.1

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[alsa-devel] ASoC: codecs: max98088: bug fixes and changes
by Jinyoung Park 11 May '11

11 May '11

Dear Mark, Recently, I working on max98088 and changed something in max98088 asoc driver. I included three changes for max98088 in this mail. - 1/3 Fixed invalid register definitions in mixer controls * This change was confirmed by Peter Hsiang who author of max98088 driver and maxim engineer. - 2/3 Moved the EX Limiter Mode from dapm widget to control - 3/3 Added digital mute function in DAI1 and DAI2 Please review below my changes. ------------------------------------------------------------------------------------------------------------------------------

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[alsa-devel] [PATCH v2 0/5] ASoC: Update e-mail address
by Peter Ujfalusi 11 May '11

11 May '11

Hello, Update my e-mail address due to change in employment. Changes since v1: - Replace the no longer valid Nokia mail address Peter --- *** BLURB HERE *** Peter Ujfalusi (5): ASoC: tlv320dac33: Update e-mail address ASoC: tpa6130a2: Update e-mail address ASoC: omap-mcbsp: Update e-mail address ASoC: omap-pcm: Update e-mail address ASoC: RX51: Update e-mail address include/sound/tlv320dac33-plat.h | 2 +- include/sound/tpa6130a2-plat.h | 2 +- sound/soc/codecs/tlv320dac33.c | 4 ++-- sound/soc/codecs/tlv320dac33.h | 2 +- sound/soc/codecs/tpa6130a2.c | 4 ++-- sound/soc/codecs/tpa6130a2.h | 2 +- sound/soc/omap/omap-mcbsp.c | 2 +- sound/soc/omap/omap-mcbsp.h | 2 +- sound/soc/omap/omap-pcm.c | 2 +- sound/soc/omap/omap-pcm.h | 2 +- sound/soc/omap/rx51.c | 2 +- 11 files changed, 13 insertions(+), 13 deletions(-) -- 1.7.5.rc3

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[PATCH 3/3] ASoC: codecs: max98088: Added digital mute function =
by Jin Park 11 May '11

11 May '11

in DAI1 and DAI2 Added digital mute function in DAI1 and DAI2. Signed-off-by: Jin Park <jinyoungp(a)nvidia.com<mailto:jinyoungp@nvidia.com= >> --- sound/soc/codecs/max98088.c | 32 ++++++++++++++++++++++++++++++++ sound/soc/codecs/max98088.h | 13 +++++++++++++ 2 files changed, 45 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/max98088.c b/sound/soc/codecs/max98088.c index c520093..b5ccf37 100644 --- a/sound/soc/codecs/max98088.c +++ b/sound/soc/codecs/max98088.c @@ -1582,6 +1582,36 @@ static int max98088_dai2_set_fmt(struct snd_soc_da= i *codec_dai, =20 return 0; } +static int max98088_dai1_digital_mute(struct snd_soc_dai *codec_dai, int= =20mute) +{ + struct snd_soc_codec *codec =3D codec_dai->codec; + int reg; + + if (mute) + reg =3D M98088_DAI_MUTE; + else + reg =3D 0; + + snd_soc_update_bits(codec, M98088_REG_2F_LVL_DAI1_PLAY, + M98088_DAI_MUTE_MASK, reg); + return 0; +} + +static int max98088_dai2_digital_mute(struct snd_soc_dai *codec_dai, int= =20mute) +{ + struct snd_soc_codec *codec =3D codec_dai->codec; + int reg; + + if (mute) + reg =3D M98088_DAI_MUTE; + else + reg =3D 0; + + snd_soc_update_bits(codec, M98088_REG_31_LVL_DAI2_PLAY, + M98088_DAI_MUTE_MASK, reg); + return 0; +} + static void max98088_sync_cache(struct snd_soc_codec *codec) { =20 u16 *reg_cache =3D codec->reg_cache; @@ -1643,12 +1673,14 @@ static struct snd_soc_dai_ops max98088_dai1_ops =3D= =20{ =20 .set_sysclk =3D max98088_dai_set_sysclk, =20 .set_fmt =3D max98088_dai1_set_fmt, =20 .hw_params =3D max98088_dai1_hw_params, + .digital_mute =3D max98088_dai1_digital_mute, }; =20static struct snd_soc_dai_ops max98088_dai2_ops =3D { =20 .set_sysclk =3D max98088_dai_set_sysclk, =20 .set_fmt =3D max98088_dai2_set_fmt, =20 .hw_params =3D max98088_dai2_hw_params, + .digital_mute =3D max98088_dai2_digital_mute, }; =20static struct snd_soc_dai_driver max98088_dai[] =3D { diff --git a/sound/soc/codecs/max98088.h b/sound/soc/codecs/max98088.h index 56554c7..be89a4f 100644 --- a/sound/soc/codecs/max98088.h +++ b/sound/soc/codecs/max98088.h @@ -133,6 +133,19 @@ =20 #define M98088_REC_LINEMODE (1<<7) =20 #define M98088_REC_LINEMODE_MASK (1<<7) +/* M98088_REG_2D_MIX_SPK_CNTL */ + #define M98088_MIX_SPKR_GAIN_MASK (3<<2) + #define M98088_MIX_SPKR_GAIN_SHIFT 2 + #define M98088_MIX_SPKL_GAIN_MASK (3<<0) + #define M98088_MIX_SPKL_GAIN_SHIFT 0 + +/* M98088_REG_2F_LVL_DAI1_PLAY, M98088_REG_31_LVL_DAI2_PLAY */ + #define M98088_DAI_MUTE (1<<7) + #define M98088_DAI_MUTE_MASK (1<<7) + #define M98088_DAI_VOICE_GAIN_MASK (3<<4) + #define M98088_DAI_ATTENUATION_MASK (0xF<<0) + #define M98088_DAI_ATTENUATION_SHIFT 0 + /* M98088_REG_35_LVL_MIC1, M98088_REG_36_LVL_MIC2 */ =20 #define M98088_MICPRE_MASK (3<<5) =20 #define M98088_MICPRE_SHIFT 5 -- 1.7.1 Best regards, Jinyoung. -------------------------------------------------------------------------= ---------- This email message is for the sole use of the intended recipient(s) and m= ay contain confidential information. Any unauthorized review, use, disclosure or di= stribution is prohibited. If you are not the intended recipient, please contact the= =20sender by reply email and destroy all copies of the original message. -------------------------------------------------------------------------= ----------

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