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[alsa-devel] [PATCH 1/1] intel hda: add model dell-m6 quirk for the Dell Studio 15 (1537)
by Andy Whitcroft 02 Dec '08

02 Dec '08

Sort out the headphone connections for the Dell Studio 15 (1537), by default there is no sound from this socket. This seems to have the same wiring as a dell-m6. Automatically detect and select that model. Signed-off-by: Andy Whitcroft <apw(a)canonical.com> --- sound/pci/hda/patch_sigmatel.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 243f8de..7760636 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -1325,6 +1325,8 @@ static struct snd_pci_quirk stac92hd73xx_cfg_tbl[] = { "unknown Dell", STAC_DELL_M6), SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x0271, "unknown Dell", STAC_DELL_M6), + SND_PCI_QUIRK(PCI_VENDOR_ID_DELL, 0x029f, + "unknown Dell", STAC_DELL_M6), {} /* terminator */ }; -- 1.6.0.4.911.gc990

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[alsa-devel] [PATCH] ASoC: TWL4030: Add capture source selection and mic gain control
by Grazvydas Ignotas 02 Dec '08

02 Dec '08

The TWL4030 codec device has two ADCs. Both of them can have several inputs routed to them, but TRM says that only one source can be selected for every ADC, even though every source has a dedicated bit in the registers. This patch adds controls for capture source controls. When a microphone is selected, handlers automatically enable bias power (disable when deselected). The patch also adds microphone gain control. Microphone and line input recording tested on OMAP3 pandora board. Signed-off-by: Grazvydas Ignotas <notasas(a)gmail.com> --- sound/soc/codecs/twl4030.c | 139 ++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/twl4030.h | 7 ++ 2 files changed, 146 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index ffd5120..755a872 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -347,6 +347,130 @@ static int snd_soc_put_volsw_r2_twl4030(struct snd_kcontrol *kcontrol, return err; } +static int twl4030_get_left_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = kcontrol->private_data; + u8 reg = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL); + int result = 0; + + /* one bit must be set a time */ + reg &= TWL4030_CKMIC_EN | TWL4030_AUXL_EN | TWL4030_HSMIC_EN + | TWL4030_MAINMIC_EN; + if (reg != 0) { + while ((reg & 1) == 0) { + result++; + reg >>= 1; + } + } + + ucontrol->value.integer.value[0] = result; + return 0; +} + +static int twl4030_put_left_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = kcontrol->private_data; + int value = ucontrol->value.integer.value[0]; + u8 anamicl, micbias; + + anamicl = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICL); + anamicl &= ~(TWL4030_CKMIC_EN | TWL4030_AUXL_EN | TWL4030_HSMIC_EN + | TWL4030_MAINMIC_EN); + micbias = twl4030_read_reg_cache(codec, TWL4030_REG_MICBIAS_CTL); + micbias &= ~(TWL4030_HSMICBIAS_EN | TWL4030_MICBIAS1_EN); + + switch (value) { + case 0: + anamicl |= TWL4030_MAINMIC_EN; + micbias |= TWL4030_MICBIAS1_EN; + break; + case 1: + anamicl |= TWL4030_HSMIC_EN; + micbias |= TWL4030_HSMICBIAS_EN; + break; + case 2: + anamicl |= TWL4030_AUXL_EN; + break; + case 3: + anamicl |= TWL4030_CKMIC_EN; + break; + default: + return -EINVAL; + } + + twl4030_write(codec, TWL4030_REG_ANAMICL, anamicl); + twl4030_write(codec, TWL4030_REG_MICBIAS_CTL, micbias); + + return 1; +} + +static int twl4030_get_right_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = kcontrol->private_data; + u8 reg = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICR); + + reg &= TWL4030_SUBMIC_EN|TWL4030_AUXR_EN; + if (reg == TWL4030_SUBMIC_EN) + ucontrol->value.integer.value[0] = 0; + else if (reg == TWL4030_AUXR_EN) + ucontrol->value.integer.value[0] = 1; + + return 0; +} + +static int twl4030_put_right_input(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct snd_soc_codec *codec = kcontrol->private_data; + int value = ucontrol->value.integer.value[0]; + u8 anamicr, micbias; + + anamicr = twl4030_read_reg_cache(codec, TWL4030_REG_ANAMICR); + anamicr &= ~(TWL4030_SUBMIC_EN|TWL4030_AUXR_EN); + micbias = twl4030_read_reg_cache(codec, TWL4030_REG_MICBIAS_CTL); + micbias &= ~TWL4030_MICBIAS2_EN; + + switch (value) { + case 0: + anamicr |= TWL4030_SUBMIC_EN; + micbias |= TWL4030_MICBIAS2_EN; + break; + case 1: + anamicr |= TWL4030_AUXR_EN; + break; + default: + return -EINVAL; + } + + twl4030_write(codec, TWL4030_REG_ANAMICR, anamicr); + twl4030_write(codec, TWL4030_REG_MICBIAS_CTL, micbias); + + return 1; +} + +static const char *twl4030_left_in_sel[] = { + "Main Mic", + "Headset Mic", + "Line In", + "Carkit Mic", +}; + +static const char *twl4030_right_in_sel[] = { + "Sub Mic", + "Line In", +}; + +static const struct soc_enum twl4030_left_input_mux = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(twl4030_left_in_sel), + twl4030_left_in_sel); + +static const struct soc_enum twl4030_right_input_mux = + SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(twl4030_right_in_sel), + twl4030_right_in_sel); + /* * FGAIN volume control: * from -62 to 0 dB in 1 dB steps (mute instead of -63 dB) @@ -378,6 +502,12 @@ static DECLARE_TLV_DB_SCALE(output_tvl, -1200, 600, 1); */ static DECLARE_TLV_DB_SCALE(digital_capture_tlv, 0, 100, 0); +/* + * Gain control for microphone amplifier + * 0 dB to 30 dB in 6 dB steps + */ +static DECLARE_TLV_DB_SCALE(microphone_gain_tlv, 0, 600, 0); + static const struct snd_kcontrol_new twl4030_snd_controls[] = { /* Common playback gain controls */ SOC_DOUBLE_R_TLV("DAC1 Digital Fine Playback Volume", @@ -420,6 +550,15 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = { SOC_DOUBLE_R_TLV("Capture Volume", TWL4030_REG_ATXL1PGA, TWL4030_REG_ATXR1PGA, 0, 0x1f, 0, digital_capture_tlv), + + SOC_DOUBLE_TLV("Microphone Boost", TWL4030_REG_ANAMIC_GAIN, + 0, 3, 5, 0, microphone_gain_tlv), + + /* Capture source controls */ + SOC_ENUM_EXT("Left Capture Source", twl4030_left_input_mux, + twl4030_get_left_input, twl4030_put_left_input), + SOC_ENUM_EXT("Right Capture Source", twl4030_right_input_mux, + twl4030_get_right_input, twl4030_put_right_input), }; /* add non dapm controls */ diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index 09865d9..8dae844 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -113,6 +113,13 @@ #define TWL4030_CODECPDZ 0x02 #define TWL4030_OPT_MODE 0x01 +/* TWL4030_REG_MICBIAS_CTL (0x04) Fields */ +#define TWL4030_MICBIAS2_CTL 0x40 +#define TWL4030_MICBIAS1_CTL 0x20 +#define TWL4030_HSMICBIAS_EN 0x04 +#define TWL4030_MICBIAS2_EN 0x02 +#define TWL4030_MICBIAS1_EN 0x01 + /* ANAMICL (0x05) Fields */ #define TWL4030_CNCL_OFFSET_START 0x80 #define TWL4030_OFFSET_CNCL_SEL 0x60 -- 1.5.4.3

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[alsa-devel] ASoC: TWL4030 codec: Multiple DAC nameing
by Peter Ujfalusi 02 Dec '08

02 Dec '08

Hello, Now I have some working prototype for the TWL4030 codec DAPM playback routing. But before I send the patches I want to clear out something: Since the TWL has four DACs and I want to map all of them, what would be the correct SND_SOC_DAPM_DAC naming for them? At the moment I have these: /* DACs */ SND_SOC_DAPM_DAC("DACR1", "Right Front Playback", TWL4030_REG_AVDAC_CTL, 0, 0), SND_SOC_DAPM_DAC("DACL1", "Left Front Playback", TWL4030_REG_AVDAC_CTL, 1, 0), SND_SOC_DAPM_DAC("DACR2", "Right Rear Playback", TWL4030_REG_AVDAC_CTL, 2, 0), SND_SOC_DAPM_DAC("DACL2", "Left Rear Playback", TWL4030_REG_AVDAC_CTL, 3, 0), Are these are OK, or should I just use "Right Playback" for both DACR1 and DACR2? Btw: the 'dynamic' routing and switching now works (no user visible power control for the DACs and APGAs) in a way, that if the output is configured to take the audio from one of the DAC, than the corresponding chain (APGA, DAC) will be powered on only. Thanks, Péter

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[alsa-devel] problem in set frames and buffer size‏
by Haiping Wang 01 Dec '08

01 Dec '08

hello all,I have write a alsa app from a example from internet.The original code is good running. format 16bitchannel 2rate 44100/* This example reads from the default PCM device and writes to standard output for 5 seconds of data. */ /* Use the newer ALSA API */ #define ALSA_PCM_NEW_HW_PARAMS_API #include <alsa/asoundlib.h> int main() { long loops; int rc; int size; snd_pcm_t *handle; snd_pcm_hw_params_t *params; unsigned int val; int dir; snd_pcm_uframes_t frames; char *buffer; /* Open PCM device for recording (capture). */ rc = snd_pcm_open(&handle, "default", SND_PCM_STREAM_CAPTURE, 0); if (rc < 0) { fprintf(stderr, "unable to open pcm device: %s\n", snd_strerror(rc)); exit(1); } /* Allocate a hardware parameters object. */ snd_pcm_hw_params_alloca(&params); /* Fill it in with default values. */ snd_pcm_hw_params_any(handle, params); /* Set the desired hardware parameters. */ /* Interleaved mode */ snd_pcm_hw_params_set_access(handle, params, SND_PCM_ACCESS_RW_INTERLEAVED); /* Signed 16-bit little-endian format */ snd_pcm_hw_params_set_format(handle, params, SND_PCM_FORMAT_S16_LE); /* Two channels (stereo) */ snd_pcm_hw_params_set_channels(handle, params, 2); /* 44100 bits/second sampling rate (CD quality) */ val = 44100; snd_pcm_hw_params_set_rate_near(handle, params, &val, &dir); /* Set period size to 32 frames. */ frames = 32; snd_pcm_hw_params_set_period_size_near(handle, params, &frames, &dir); /* Write the parameters to the driver */ rc = snd_pcm_hw_params(handle, params); if (rc < 0) { fprintf(stderr, "unable to set hw parameters: %s\n", snd_strerror(rc)); exit(1); } /* Use a buffer large enough to hold one period */ snd_pcm_hw_params_get_period_size(params, &frames, &dir); size = frames * 4; /* 2 bytes/sample, 2 channels */ buffer = (char *) malloc(size); /* We want to loop for 5 seconds */ snd_pcm_hw_params_get_period_time(params, &val, &dir); loops = 5000000 / val; while (loops > 0) { loops--; rc = snd_pcm_readi(handle, buffer, frames); if (rc == -EPIPE) { /* EPIPE means overrun */ fprintf(stderr, "overrun occurred\n"); snd_pcm_prepare(handle); } else if (rc < 0) { fprintf(stderr, "error from read: %s\n", snd_strerror(rc)); } else if (rc != (int)frames) { fprintf(stderr, "short read, read %d frames\n", rc); } rc = write(1, buffer, size); if (rc != size) fprintf(stderr, "short write: wrote %d bytes\n", rc); } snd_pcm_drain(handle); snd_pcm_close(handle); free(buffer); return 0; } now I want to change channel and format to write my own programm.format = 8bit, 24bit and 32bitchannel = 1 or 2how can i change the parameters in frames and buffer size.for each format 8/16/24/32bit how to set frames = ?and alsa channel 1/2 how to set buffersize??? I try many times and failed .I want your help. thanks!! _________________________________________________________________ Spielspaß für Jedermann! Mit tausenden von Spielen für Windows Vista! http://redirect.gimas.net/?n=M0812xSpieleVistaDE

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[alsa-devel] Merged upstream to topic/asoc branch
by Takashi Iwai 01 Dec '08

01 Dec '08

Hi, FYI, I merged the current Linus tree back to topic/asoc branch on sound git tree for clean-up works including the latest commit in sound/soc/fsl. If you have any problems with it, please let me know. thanks, Takashi

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[alsa-devel] (SoC) Does i.MX27's SSI work as I2S clock master?
by ngc＠drvlabo.jp 01 Dec '08

01 Dec '08

Hello, lists. Now I'm porting WM8728 driver with i.MX27ADS, based on Freescale's BSP kernel(2.6.22) and refering to ALSA SoC git repository. I'm connecting the WM8728 to the SSI2 controller via SSI-3 signal set. Then, my ADS board has altered: i.MX27 WM8728 --------------------- SSI3_CLK BCLKIN SSI3_FS LRCIN SSI3_TXDAT DIN TOUT/SSI2_MCLK MCLK Of course, I added some codes to configure GPIO pins as appropriate into board specific sources. Then, I'm trying to play sounds via WM8728 and have not hear any sounds yet. I checked SSI-3 signal set with an oscilloscope, no clock signal appears on SSI3_CLK. I tried the various SSI setting, I2S normal, I2S master, asychronous/synchronous, normal mode/network mode, ... But, I can't see clock signal, yet. I want to know an example setting to act i.MX' SSI as I2S clock master. Regards, -------------- NOGUCHI Hiroshi ngc(a)drvlabo.jp

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Re: [alsa-devel] [ath5k-devel] Bugs on aspire one A150
by Maxim Levitsky 01 Dec '08

01 Dec '08

On Tue, 2008-11-11 at 00:12 +0200, Nick Kossifidis wrote: > 2008/11/10 Maxim Levitsky <maximlevitsky(a)gmail.com>: > > Bob Copeland wrote: > >> On Fri, Nov 7, 2008 at 10:26 PM, Maxim Levitsky <maximlevitsky(a)gmail.com> wrote: > >>> ** 2 - wireless: not to mention the fact that ath5k wasn't installed by > >>> default in ubuntu... > >>> wireless more or less works, but kernel log is full of backtraces. > >> > >> As you've seen, these should at least be gone now... > >> > >>> Was able to connect to my WPA2 access point. > >>> Sometimes wireless fails completely, especially after suspend to ram. > >> > >> Did you get noise calibration failures in dmesg at this point? Felix posted > >> a patch recently that handles them better. > > Yep, but they seems to be gone in latest -git too. > > Wireless work fine now, it seems like my iwl3945. > > > >> > >>> Advanced features like monitor/injection work, but when I changed the card's > >>> mac address it stopped working. > >>> I also noticed that if I then start airodump, then wireless works with new > >>> mac. > > Why it doesn't accept new mac?, can this be fixed? > > > > > >> > >>> ** 4 - wireless led doesn't work. > >>> ath5k devs, can you fix this? > >> > >> Currently all the LED code we have in there is for setting a particular gpio > >> for particular laptops (and they differ between models). We only have quirks > >> for IBM and HP, none for Acer. Also, the legacy-hal seems to have a different > >> strategy for 2425 chips. I can try to hack up some code for you to test. > >> > > > > I have seen on the web that madwifi did support the led with some gpio settings > > > > sysctl -w dev.wifi0.ledpin=3 > > sysctl -w dev.wifi0.softled=1 > > > > from https://help.ubuntu.com/community/AspireOne > > > > Best regards, > > Maxim Levitsky > > i think that some led settings are stored on EEPROM, let me check it out... Hi, Any update on that? Best regards, Maxim Levitsky > >

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[alsa-devel] Unused TWL4030 macros
by Takashi Iwai 01 Dec '08

01 Dec '08

Hi, after the commit b0bd53a7399f65e2d1b37cd44c5003e55b886c1e, we get compile warnings as: sound/soc/codecs/twl4030.c:233: warning: 'snd_soc_get_volsw_twl4030' defined but not used sound/soc/codecs/twl4030.c:262: warning: 'snd_soc_put_volsw_twl4030' defined but not used sound/soc/codecs/twl4030.c:291: warning: 'snd_soc_get_volsw_r2_twl4030' defined but not used sound/soc/codecs/twl4030.c:318: warning: 'snd_soc_put_volsw_r2_twl4030' defined but not used Will the new macros ever used? Takashi

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[alsa-devel] [PATCH] ALSA: Add WM9715 to AC97 IDs
by Mark Brown 01 Dec '08

01 Dec '08

The WM9715 is software compatible with the WM9711 and WM9712. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/pci/ac97/ac97_codec.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index 6704acb..e8375ff 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -175,7 +175,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x574d4C04, 0xffffffff, "WM9704M,WM9704Q", patch_wolfson04, NULL}, { 0x574d4C05, 0xffffffff, "WM9705,WM9710", patch_wolfson05, NULL}, { 0x574d4C09, 0xffffffff, "WM9709", NULL, NULL}, -{ 0x574d4C12, 0xffffffff, "WM9711,WM9712", patch_wolfson11, NULL}, +{ 0x574d4C12, 0xffffffff, "WM9711,WM9712,WM9715", patch_wolfson11, NULL}, { 0x574d4c13, 0xffffffff, "WM9713,WM9714", patch_wolfson13, NULL, AC97_DEFAULT_POWER_OFF}, { 0x594d4800, 0xffffffff, "YMF743", patch_yamaha_ymf743, NULL }, { 0x594d4802, 0xffffffff, "YMF752", NULL, NULL }, -- 1.5.6.5

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[alsa-devel] [PATCH 0/4] ASoC: TWL4030: Gain control updates, adding pre-DAC routing controls
by Peter Ujfalusi 01 Dec '08

01 Dec '08

Hello, This series adds 'proper' gain controls for: Per-DAC common gain controls. There are three gain controls asspciated wit a DAC in TWL, There are four DAC can be used (in OPT_MODE=1). Digital fine gain, Digital coarse gain, Analog gain. Every output, except the HandsFree has their own gain controls per channel (L/R) Pre-DAC routings: Digital input namings: SDRL1: TDM #1 SDRR1: TDM #2 SDRL2: TDM #3, I2S Left SDRR2: TDM #4, I2S Right Mono mixing: SDRM1: mixed SDRL1 and SDRR1 SDRM2: mixed SDRL2 and SDRR2 The possible mux settings for the DAC inputs: DACL1 <- SDRL1, SDRM1, SDRM2, SDRL2 DACR1 <- SDRR1, SDRM1, SDRM2, SDRR2 DACL2 <- SDRL2, SDRM2 DACR2 <- SDRR2, SDRM2 --- Peter Ujfalusi (4): ASoC: TWL4030: Change the capture volume control to TLV ASoC: TWL4030: Change the common playback volume controls ASoC: TWL4030: Add volume controls for outputs ASoC: TWL4030: Add pre-DAC digital mux controls sound/soc/codecs/twl4030.c | 59 ++++++++++++++++++++++++++++++++++++++++++++ 1 files changed, 59 insertions(+), 0 deletions(-)

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