February 2012 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] question on Asoc tlv320aic3104
by rohith R 28 Feb '12

28 Feb '12

Hi, I'm modifying Asoc tlv320aic3104 codec driver to support MIC3L/R inputs(currently not supported in the code.) I'm a newbie to ALSA programming and had these questions: 1.Is there any documentation for ASOC apart from the one in kernel sources? 2.ALSA api reference guide link on the website is giving a 404 error.Any updated link? 3.Trying to program Page0 register17/18 to add as a "route".Which is the best way to do it.Still new to the range of SOC_** #defines available. --Rohith

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[alsa-devel] [PATCH] OMAP4: panda: Correct cpu version check for 4430
by Peter Ujfalusi 28 Feb '12

28 Feb '12

The cpu_is_omap4430() macro always return with 0. Use the correct cpu_is_omap443x() to check for Panda revision. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)ti.com> --- Hi, Since the depending patches are going via Liam's tree it would be best if we can queue this via audio as well. Without this patch the audio on Panda 4430 is not working correctly. Regards, Peter arch/arm/mach-omap2/board-omap4panda.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/arch/arm/mach-omap2/board-omap4panda.c b/arch/arm/mach-omap2/board-omap4panda.c index eada955..74db647 100644 --- a/arch/arm/mach-omap2/board-omap4panda.c +++ b/arch/arm/mach-omap2/board-omap4panda.c @@ -520,7 +520,7 @@ void omap4_panda_display_init(void) static void omap4_panda_init_rev(void) { - if (cpu_is_omap4430()) { + if (cpu_is_omap443x()) { /* PandaBoard 4430 */ /* ASoC audio configuration */ panda_abe_audio_data.card_name = "PandaBoard"; -- 1.7.8.4

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Re: [alsa-devel] [PATCH] Sony Vaio with stac9872
by Takashi Iwai 28 Feb '12

28 Feb '12

At Tue, 28 Feb 2012 07:26:25 +0300, 107(a)mail.ru wrote: > > 2012/2/27 Takashi Iwai <tiwai(a)suse.de>: > > At Fri, 24 Feb 2012 16:41:43 +0400, > > 107 wrote: > >> > >> Hello, I have no sound on my Sony Vaio SZ4XRN laptop with newest kernels. > >> Sound: hda-intel, sigmatel - STAC9872 > >> > >> To solve this issue I make this patch for sound/pci/hda/patch_sigmatel.c: > >> > >> --- patch_sigmatel.c.bak 2012-02-23 21:01:28.577515317 +0400 > >> +++ patch_sigmatel.c 2012-02-23 21:53:21.035605216 +0400 > >> @@ -6357,7 +6357,16 @@ > >> */ > >> > >> static const struct hda_verb stac9872_core_init[] = { > >> - {0x15, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mic-sel: 0a,0d,14,02 */ > >> + {0x0a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP <- 0x2 */ > >> + {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Speaker <- 0x5 */ > >> + {0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? (<- 0x2) */ > >> + {0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD */ > >> + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? */ > >> + {0x15, AC_VERB_SET_CONNECT_SEL, 0x2 /*0x1*/ }, /* mic-sel: 0a,0d,14,02 */ > >> + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* HP */ > >> + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Speaker */ > >> + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* capture sw/vol -> 0x8 */ > >> + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* CD-in -> 0x6 */ > >> {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Mic-in -> 0x9 */ > >> {} > >> }; > >> > >> > >> Constants got from old kernel (2.6.22). > >> With this patch sound works fine. Thank you. > > > > Most of verbs above must be superfluous, but some of them might be > > needed in your case. Could you give alsa-info.sh outputs (run with > > --no-upload option) on both working and non-working cases? > > > > > > thanks, > > > > Takashi > > > > Yes of course, here is my alsa-info-nosound.txt on default kernel and > alsa-info-work.txt with patch (see attach). Thanks. Could you check whether the one-liner below fixes your problem? Takashi --- diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 8c346ac..5988dbd 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4638,7 +4638,7 @@ static void stac92xx_hp_detect(struct hda_codec *codec) unsigned int val = AC_PINCTL_OUT_EN | AC_PINCTL_HP_EN; if (no_hp_sensing(spec, i)) continue; - if (presence) + if (1 /*presence*/) stac92xx_set_pinctl(codec, cfg->hp_pins[i], val); #if 0 /* FIXME */ /* Resetting the pinctl like below may lead to (a sort of) regressions

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[alsa-devel] [PATCH] tea575x: fix HW seek
by Ondrej Zary 27 Feb '12

27 Feb '12

Fix HW seek in TEA575x to work properly: - a delay must be present after search start and before first register read or the seek does weird things - when the search stops, the new frequency is not available immediately, we must wait until it appears in the register (fortunately, we can clear the frequency bits when starting the search as it starts at the frequency currently set, not from the value written) - sometimes, seek remains on the current frequency (or moves only a little), so repeat it until it moves by at least 50 kHz Signed-off-by: Ondrej Zary <linux(a)rainbow-software.org> --- a/sound/i2c/other/tea575x-tuner.c +++ b/sound/i2c/other/tea575x-tuner.c @@ -89,7 +89,7 @@ static void snd_tea575x_write(struct snd_tea575x *tea, unsigned int val) tea->ops->set_pins(tea, 0); } -static unsigned int snd_tea575x_read(struct snd_tea575x *tea) +static u32 snd_tea575x_read(struct snd_tea575x *tea) { u16 l, rdata; u32 data = 0; @@ -120,6 +120,27 @@ static unsigned int snd_tea575x_read(struct snd_tea575x *tea) return data; } +static void snd_tea575x_get_freq(struct snd_tea575x *tea) +{ + u32 freq = snd_tea575x_read(tea) & TEA575X_BIT_FREQ_MASK; + + if (freq == 0) { + tea->freq = 0; + return; + } + + /* freq *= 12.5 */ + freq *= 125; + freq /= 10; + /* crystal fixup */ + if (tea->tea5759) + freq += TEA575X_FMIF; + else + freq -= TEA575X_FMIF; + + tea->freq = clamp(freq * 16, FREQ_LO, FREQ_HI); /* from kHz */ +} + static void snd_tea575x_set_freq(struct snd_tea575x *tea) { u32 freq = tea->freq; @@ -203,6 +224,8 @@ static int vidioc_g_frequency(struct file *file, void *priv, if (f->tuner != 0) return -EINVAL; f->type = V4L2_TUNER_RADIO; + if (!tea->cannot_read_data) + snd_tea575x_get_freq(tea); f->frequency = tea->freq; return 0; } @@ -225,36 +248,50 @@ static int vidioc_s_hw_freq_seek(struct file *file, void *fh, struct v4l2_hw_freq_seek *a) { struct snd_tea575x *tea = video_drvdata(file); + int i, old_freq; + unsigned long timeout; if (tea->cannot_read_data) return -ENOTTY; + + snd_tea575x_get_freq(tea); + old_freq = tea->freq; + /* clear the frequency, HW will fill it in */ + tea->val &= ~TEA575X_BIT_FREQ_MASK; tea->val |= TEA575X_BIT_SEARCH; - tea->val &= ~TEA575X_BIT_UPDOWN; if (a->seek_upward) tea->val |= TEA575X_BIT_UPDOWN; + else + tea->val &= ~TEA575X_BIT_UPDOWN; snd_tea575x_write(tea, tea->val); + timeout = jiffies + msecs_to_jiffies(10000); for (;;) { - unsigned val = snd_tea575x_read(tea); - - if (!(val & TEA575X_BIT_SEARCH)) { - /* Found a frequency */ - val &= TEA575X_BIT_FREQ_MASK; - val = (val * 10) / 125; - if (tea->tea5759) - val += TEA575X_FMIF; - else - val -= TEA575X_FMIF; - tea->freq = clamp(val * 16, FREQ_LO, FREQ_HI); - return 0; - } + if (time_after(jiffies, timeout)) + break; if (schedule_timeout_interruptible(msecs_to_jiffies(10))) { /* some signal arrived, stop search */ tea->val &= ~TEA575X_BIT_SEARCH; snd_tea575x_write(tea, tea->val); return -ERESTARTSYS; } + if (!(snd_tea575x_read(tea) & TEA575X_BIT_SEARCH)) { + /* Found a frequency, wait until it can be read */ + for (i = 0; i < 100; i++) { + msleep(10); + snd_tea575x_get_freq(tea); + if (tea->freq != 0) /* available */ + break; + } + /* if we moved by less than 50 kHz, continue seeking */ + if (abs(old_freq - tea->freq) < 16 * 500) { + snd_tea575x_write(tea, tea->val); + continue; + } + tea->val &= ~TEA575X_BIT_SEARCH; + return 0; + } } - return 0; + return -ETIMEDOUT; } static int tea575x_s_ctrl(struct v4l2_ctrl *ctrl) @@ -318,7 +355,7 @@ int snd_tea575x_init(struct snd_tea575x *tea) return -ENODEV; } - tea->val = TEA575X_BIT_BAND_FM | TEA575X_BIT_SEARCH_10_40; + tea->val = TEA575X_BIT_BAND_FM | TEA575X_BIT_SEARCH_5_28; tea->freq = 90500 * 16; /* 90.5Mhz default */ snd_tea575x_set_freq(tea); -- Ondrej Zary

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[alsa-devel] [PATCH 1/2] ALSA: Use a define for the number of jack switch types
by Mark Brown 27 Feb '12

27 Feb '12

This is intended to facilitate the merge of the two jack detection mechanisms. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/sound/jack.h | 3 +++ sound/core/jack.c | 4 ++-- 2 files changed, 5 insertions(+), 2 deletions(-) diff --git a/include/sound/jack.h b/include/sound/jack.h index 63c7907..5891657 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -53,6 +53,9 @@ enum snd_jack_types { SND_JACK_BTN_5 = 0x0200, }; +/* Keep in sync with definitions above */ +#define SND_JACK_SWITCH_TYPES 6 + struct snd_jack { struct input_dev *input_dev; int registered; diff --git a/sound/core/jack.c b/sound/core/jack.c index 26edf63..471e1e3 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -25,7 +25,7 @@ #include <sound/jack.h> #include <sound/core.h> -static int jack_switch_types[] = { +static int jack_switch_types[SND_JACK_SWITCH_TYPES] = { SW_HEADPHONE_INSERT, SW_MICROPHONE_INSERT, SW_LINEOUT_INSERT, @@ -128,7 +128,7 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, jack->type = type; - for (i = 0; i < ARRAY_SIZE(jack_switch_types); i++) + for (i = 0; i < SND_JACK_SWITCH_TYPES; i++) if (type & (1 << i)) input_set_capability(jack->input_dev, EV_SW, jack_switch_types[i]); -- 1.7.9.rc1

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[alsa-devel] [PATCH] Add default config for tegra-alc5632 (as found on paz00)
by Paul Fertser 27 Feb '12

27 Feb '12

This makes paz00 (aka Toshiba ac100, a popular arm-based netbook) use dmix/dsnoop by default since it doesn't support hw mixing. Signed-off-by: Paul Fertser <fercerpav(a)gmail.com> --- Hi, I haven't seen any other ASoC configs here so probably i'm missing something obvious. My reasoning is that when a person uses a regular notebook with HDA he gets DMix by default so it should be the case for this less common netbook too. Rate is set to 44100 because it's unlikely a person will listen to Hi-Fi music and hence it avoids resampling for the most common usecase. Feel free to drop if that's inappropriate, the device works fine with 48000 too. src/conf/cards/Makefile.am | 1 + src/conf/cards/tegra-alc5632.conf | 23 +++++++++++++++++++++++ 2 files changed, 24 insertions(+), 0 deletions(-) create mode 100644 src/conf/cards/tegra-alc5632.conf diff --git a/src/conf/cards/Makefile.am b/src/conf/cards/Makefile.am index b7190e7..efed590 100644 --- a/src/conf/cards/Makefile.am +++ b/src/conf/cards/Makefile.am @@ -44,6 +44,7 @@ cfg_files = aliases.conf \ RME9652.conf \ SI7018.conf \ SB-XFi.conf \ + tegra-alc5632.conf \ TRID4DWAVENX.conf \ USB-Audio.conf \ YMF744.conf \ diff --git a/src/conf/cards/tegra-alc5632.conf b/src/conf/cards/tegra-alc5632.conf new file mode 100644 index 0000000..371fe2a --- /dev/null +++ b/src/conf/cards/tegra-alc5632.conf @@ -0,0 +1,23 @@ +# default with dmix/dsnoop +tegra-alc5632.pcm.default { + @args [ CARD ] + @args.CARD { + type string + } + type asym + playback.pcm { + type plug + slave.pcm { + @func concat + strings [ "dmix:CARD=" $CARD ",RATE=44100" ] + } + } + capture.pcm { + type plug + slave.pcm { + @func concat + strings [ "dsnoop:CARD=" $CARD ",RATE=44100" ] + } + } +} + -- 1.7.3.4

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[alsa-devel] [PATCH] ASoC: wm8753: Convert to devm_kzalloc()
by Mark Brown 27 Feb '12

27 Feb '12

Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/wm8753.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/wm8753.c b/sound/soc/codecs/wm8753.c index 21ed75d..59c28db 100644 --- a/sound/soc/codecs/wm8753.c +++ b/sound/soc/codecs/wm8753.c @@ -1557,7 +1557,8 @@ static int __devinit wm8753_spi_probe(struct spi_device *spi) struct wm8753_priv *wm8753; int ret; - wm8753 = kzalloc(sizeof(struct wm8753_priv), GFP_KERNEL); + wm8753 = devm_kzalloc(&spi->dev, sizeof(struct wm8753_priv), + GFP_KERNEL); if (wm8753 == NULL) return -ENOMEM; @@ -1580,7 +1581,6 @@ static int __devinit wm8753_spi_probe(struct spi_device *spi) err_regmap: regmap_exit(wm8753->regmap); err: - kfree(wm8753); return ret; } @@ -1612,7 +1612,8 @@ static __devinit int wm8753_i2c_probe(struct i2c_client *i2c, struct wm8753_priv *wm8753; int ret; - wm8753 = kzalloc(sizeof(struct wm8753_priv), GFP_KERNEL); + wm8753 = devm_kzalloc(&i2c->dev, sizeof(struct wm8753_priv), + GFP_KERNEL); if (wm8753 == NULL) return -ENOMEM; @@ -1632,10 +1633,10 @@ static __devinit int wm8753_i2c_probe(struct i2c_client *i2c, dev_err(&i2c->dev, "Failed to register CODEC: %d\n", ret); goto err_regmap; } + err_regmap: regmap_exit(wm8753->regmap); err: - kfree(wm8753); return ret; } @@ -1645,7 +1646,6 @@ static __devexit int wm8753_i2c_remove(struct i2c_client *client) snd_soc_unregister_codec(&client->dev); regmap_exit(wm8753->regmap); - kfree(wm8753); return 0; } -- 1.7.9.1

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[alsa-devel] [PATCH] ASoC: imx: let SND_MXC_SOC_FIQ select FIQ
by Shawn Guo 27 Feb '12

27 Feb '12

CONFIG_FIQ is only needed when CONFIG_SND_MXC_SOC_FIQ is selected to build imx-pcm-fiq.c, so let SND_MXC_SOC_FIQ select FIQ. Signed-off-by: Shawn Guo <shawn.guo(a)linaro.org> --- sound/soc/imx/Kconfig | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index 91b66ef..e89139f 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -1,7 +1,6 @@ menuconfig SND_IMX_SOC tristate "SoC Audio for Freescale i.MX CPUs" depends on ARCH_MXC - select FIQ select SND_SOC_AC97_BUS help Say Y or M if you want to add support for codecs attached to @@ -11,6 +10,7 @@ menuconfig SND_IMX_SOC if SND_IMX_SOC config SND_MXC_SOC_FIQ + select FIQ tristate config SND_MXC_SOC_MX2 -- 1.7.5.4

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Re: [alsa-devel] Support new HDA codec
by Mark Brown 27 Feb '12

27 Feb '12

On Mon, Feb 27, 2012 at 03:01:11PM +0100, Jorge Fernandez Monteagudo wrote: > > Mark Brown wrote > > It should be very standards confirming and not require any magic Wolfson > > specific verbs to use the functionality, a generic HDA driver should do > > fine with an appropriate setup from the BIOS. > With the generic HDA the only enabled path is DAC1 and PORT-A, the first > it founds. In our carrier design we only use the stereo differential and > SPIDF outputs. I've implemented a 'patch_wolfson.c' in order to enable these > outputs and share the stream through both outputs, because I've no access > to the BIOS code. The inputs are not used in our carrier, then I've not > implemented anything. Please share this code. Whatever is being done to work around the BIOS probably ought to be board specific, even if the BIOS is not configurable hopefully there's enough information been added to distinguish the board.

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[alsa-devel] [PATCH] Sony Vaio with stac9872
by 107 27 Feb '12

27 Feb '12

Hello, I have no sound on my Sony Vaio SZ4XRN laptop with newest kernels. Sound: hda-intel, sigmatel - STAC9872 To solve this issue I make this patch for sound/pci/hda/patch_sigmatel.c: --- patch_sigmatel.c.bak 2012-02-23 21:01:28.577515317 +0400 +++ patch_sigmatel.c 2012-02-23 21:53:21.035605216 +0400 @@ -6357,7 +6357,16 @@ */ static const struct hda_verb stac9872_core_init[] = { - {0x15, AC_VERB_SET_CONNECT_SEL, 0x1}, /* mic-sel: 0a,0d,14,02 */ + {0x0a, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_HP }, /* HP <- 0x2 */ + {0x0f, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT }, /* Speaker <- 0x5 */ + {0x0d, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? (<- 0x2) */ + {0x0e, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN }, /* CD */ + {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80 }, /* Mic? */ + {0x15, AC_VERB_SET_CONNECT_SEL, 0x2 /*0x1*/ }, /* mic-sel: 0a,0d,14,02 */ + {0x02, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* HP */ + {0x05, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, /* Speaker */ + {0x09, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0)}, /* capture sw/vol -> 0x8 */ + {0x07, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_UNMUTE(0)}, /* CD-in -> 0x6 */ {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, /* Mic-in -> 0x9 */ {} }; Constants got from old kernel (2.6.22). With this patch sound works fine. Thank you.

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