December 2008 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [Question] Can I open a substream in kernel space without attach to a file pointer?
by Bryan Wu 12 Dec '08

12 Dec '08

Hi Takashi, I am developing a USB gadget driver compliant to USB Audio Class Spec 2.0. So I want to open a PCM substream and do some playback of capture, then close them? I found snd_pcm_open_substream() is for opening a substream and attach it to a file. But in my application, there is no need to open a file before opening a substream. - Is there any interface for me to open a substream in kernel space without attach to a file? - How to playback and capture in kernel space, use snd_pcm_lib_write and snd_pcm_lib_read? - How to get the snd_pcm_hardware struct from low level driver, because I have to get the hardware configuration of the snd pcm device? And I am reading the code of OSS emulator in ALSA. It provides some info about the kernel space sound card programming. Thanks a lot -Bryan

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[alsa-devel] Macbook Air 2.1 - No sound on main speakers
by Klaus Schulz 12 Dec '08

12 Dec '08

Hi there. Fowarded from my "Ubuntu manual upgrade thread": Another problem with Realtek. No sound on mainspeakers on MACBOOK Air 2.1. A known issue?? See also the different aplay and codec IDs! Klaus Quote: ----------------------- No sound on main internal speaker. Headphones do work fine. Macbook Air 2,1 Ubuntu Intrepid server Kernel 2.6.27-9 Alsa Driver version 1.0.18a GIT snapshot from 12/4/08 Sound card nVidia MCP79 (device id 10de:0ac0) aplay -l: ALC885 /proc/asound/card0/codec#0 CODEC ALC889a lspci -nv 00:08.0 0403: 10de:0ac0 (rev b1) Subsystem: 10de:cb79 ------------------------

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[alsa-devel] HP Pavillion TX2000z - Headphonejack not working
by Klaus Schulz 12 Dec '08

12 Dec '08

Hi folks. Another problem now with HP and realtek AC861VD. The headphone-jack is not working. Known issue? THX \Klaus QUOTE: -------------------------- drivers 1.0.18a HP Pavillion tx2000z, Realtek AC861VD, Alsa version 1.0.18a. Running Ubuntu Studio Intrepid 8.10 Kernel 2.6.27-9-generic options snd-hda-intel model=hp kqian@kkqian:~$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: ALC861VD Analog [ALC861VD Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: NVidia [HDA NVidia], device 1: ALC861VD Digital [ALC861VD Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: NVidia [HDA NVidia], device 6: Si3054 Modem [Si3054 Modem] Subdevices: 1/1 Subdevice #0: subdevice #0 kqian@kkqian:~$ cat /proc/asound/card0/codec#* | grep Codec Codec: Realtek ALC861-VD Codec: Motorola Si3054 Oh, I should also add, headphones do not show up in volume control. I have checked the switches, as well as the main interface. The only things that are there are: Master, PCM, Front, Front Mic, Front Mic Boost, Microphone, Mic Boost, Speaker, Capture, IEC958, IEC958 Default PCM, Caller ID, Off-hook, Input Source, All of them are checked, and none of them are muted.

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[alsa-devel] More notes on HD-audio driver
by Takashi Iwai 11 Dec '08

11 Dec '08

Hi, per popular demand, I wrote a brief document more about HD-audio driver. It's now found in sound git tree as Documentation/sound/alsa/HD-Audio.txt. The git tree is: git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git You can get the text from web: http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=blob_plain;… A PDF file is found at: http://ftp.kernel.org/pub/linux/kernel/people/tiwai/docs/HD-Audio.pdf Comments, corrections and additions are appreciated. thanks, Takashi

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[alsa-devel] [PATCH 0/2] ASoC updates
by Mark Brown 11 Dec '08

11 Dec '08

The following changes since commit 6de45d5d776d2a7e7a9adc8ea49d37fe1bd45fb2: Takashi Iwai (1): ALSA: ASoC - Fix DAI registration in s3c2443-ac97.c are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-tiwai Mark Brown (1): ASoC: Fix variable name for Blackfin I2S DAI Stanley.Miao (1): ASoC: TWL4030: hands-free start-up sequence. sound/soc/blackfin/bf5xx-i2s.c | 4 ++-- sound/soc/codecs/twl4030.c | 34 ++++++++++++++++++++++++++++++---- sound/soc/codecs/twl4030.h | 6 ++++++ 3 files changed, 38 insertions(+), 6 deletions(-)

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[alsa-devel] [PATCH] TWL4030: Add event callback to enable hands-free.
by Stanley.Miao 11 Dec '08

11 Dec '08

A special sequence is required to enable hands-free on TWL4030. Signed-off-by: Stanley.Miao <stanley.miao(a)windriver.com> --- sound/soc/codecs/twl4030.c | 33 +++++++++++++++++++++++++++++---- sound/soc/codecs/twl4030.h | 6 ++++++ 2 files changed, 35 insertions(+), 4 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 71ab534..86d6553 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -325,6 +325,29 @@ static int outmixer_event(struct snd_soc_dapm_widget *w, return ret; } +static int handsfree_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *kcontrol, int event) +{ + struct soc_enum *e = (struct soc_enum *)w->kcontrols->private_value; + unsigned char hs_ctl; + + + hs_ctl = twl4030_read_reg_cache(w->codec, e->reg); + + if (hs_ctl & TWL4030_HF_CTL_REF_EN) { + hs_ctl |= TWL4030_HF_CTL_RAMP_EN; + twl4030_write(w->codec, e->reg, hs_ctl); + hs_ctl |= TWL4030_HF_CTL_LOOP_EN; + twl4030_write(w->codec, e->reg, hs_ctl); + hs_ctl |= TWL4030_HF_CTL_HB_EN; + twl4030_write(w->codec, e->reg, hs_ctl); + } else { + twl4030_write(w->codec, e->reg, 0x0); + } + + return 0; +} + /* * Some of the gain controls in TWL (mostly those which are associated with * the outputs) are implemented in an interesting way: @@ -807,10 +830,12 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_MUX("CarkitR Mux", SND_SOC_NOPM, 0, 0, &twl4030_dapm_carkitr_control), /* HandsfreeL/R */ - SND_SOC_DAPM_MUX("HandsfreeL Mux", TWL4030_REG_HFL_CTL, 5, 0, - &twl4030_dapm_handsfreel_control), - SND_SOC_DAPM_MUX("HandsfreeR Mux", TWL4030_REG_HFR_CTL, 5, 0, - &twl4030_dapm_handsfreer_control), + SND_SOC_DAPM_MUX_E("HandsfreeL Mux", TWL4030_REG_HFL_CTL, 5, 0, + &twl4030_dapm_handsfreel_control, handsfree_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), + SND_SOC_DAPM_MUX_E("HandsfreeR Mux", TWL4030_REG_HFR_CTL, 5, 0, + &twl4030_dapm_handsfreer_control, handsfree_event, + SND_SOC_DAPM_POST_PMU|SND_SOC_DAPM_POST_PMD), SND_SOC_DAPM_ADC("ADCL", "Left Capture", SND_SOC_NOPM, 0, 0), SND_SOC_DAPM_ADC("ADCR", "Right Capture", SND_SOC_NOPM, 0, 0), diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index a2065d4..54615c7 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -191,6 +191,12 @@ #define TWL4030_RAMP_DELAY_2581MS 0x1C #define TWL4030_RAMP_EN 0x02 +/* HFL_CTL (0x29, 0x2A) Fields */ +#define TWL4030_HF_CTL_HB_EN 0x04 +#define TWL4030_HF_CTL_LOOP_EN 0x08 +#define TWL4030_HF_CTL_RAMP_EN 0x10 +#define TWL4030_HF_CTL_REF_EN 0x20 + /* APLL_CTL (0x3A) Fields */ #define TWL4030_APLL_EN 0x10 -- 1.5.6.3

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[alsa-devel] ASoC: asound.conf
by naveen krishna ch 11 Dec '08

11 Dec '08

Hi all, I have a custom built ASoC CODEC driver for twl4030. I am integrating my kernel (ASoC driver) with the application. The application expects a /etc/asound.conf file is there any utility to build or restore the asound.conf for my ASoC driver. I have checked the "alsactl" which can give me the state/controls information. Do i need to make a asound.conf file.If so, can any one help my with some information (a sample asound.conf) with details of the driver in it. Sorry, if this the wrong place to raise this issue. -- Thanks, (: Naveen Krishna Ch :)

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[alsa-devel] ASoC: asound.conf
by naveen krishna ch 11 Dec '08

11 Dec '08

Hi all, I have a custom built ASoC CODEC driver for twl4030. I am integrating my kernel (ASoC driver) with the application. The application expects a /etc/asound.conf file is there any utility to build or restore the asound.conf for my ASoC driver. I have checked the "alsactl" which can give me the state/controls information. Do i need to make a asound.conf file.If so, can any one help my with some information (a sample asound.conf) with details of the driver in it. Sorry, if this the wrong place to raise this issue. -- Thanks, (: Naveen Krishna Ch :)

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[alsa-devel] 2.6.26.[6|7]-rt11, alsa rawmidi, seq hang
by Fernando Lopez-Lezcano 11 Dec '08

11 Dec '08

(starting a new thread with the hope of being more precise after a lot of testing and rebooting) I'm seeing a realtime patch related hard hang in the kernel alsa subsystem (MIDI input/output). In a nutshell: - alsa rawmidi works (ie: "rawmidi -v -i hw:0" outputs a stream of messages when pointed to a midi capable card that has an external keyboard connected). - the alsa sequencer interface works (ie: aplaymidi connected to aseqdump transfers data just fine). - BOTH combined do NOT work (ie: use aconnect to connect the port that corresponds to the external midi interface to aseqdump: aseqdump hangs forever after transferring the first message and the only way out is a reboot). Just tested this on 2.6.26.7 + rt11, Gina3G soundcard with an external MIDI keyboard sending messages. This is not specific to the Gina card, I have tested also USB midi interfaces and an RME hdsp card with a Multiface interface. 2.6.24.7 + rt21 works fine on the same hardware combinations. The standard Fedora 9 2.6.26.x kernel works fine as well. I'm attaching the kernel configuration for the kernel I'm currently testing plus a dmesg dump of the whole boot process (rt kernel hackers can find there some additional sleeping in wrong context warning messages there), including the output of a "echo t >/proc/sysrq-trigger" that should show where aseqdump currently hangs (or so I think). Hopefully this will narrow the area affected enough so that a solution can be found or more debugging info can be requested. Currently 2.6.26.x-rtx is unusable for musical purposes (at least when you have to interact with external midi controllers). Thanks for any help... -- Fernando

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Re: [alsa-devel] More notes on HD-audio driver
by kls.schlz＠googlemail.com 11 Dec '08

11 Dec '08

If you prepare a Wiki for the HD-Audio, don't forget to merge somehow ALSA-Configuration.txt. IMO this goes together. On Dec 11, 2008 7:24am, Takashi Iwai <tiwai(a)suse.de> wrote: > At Wed, 10 Dec 2008 14:45:52 -0700, > > stan wrote: > > > > > > Takashi Iwai wrote: > > > > Hi, > > > > > > > > per popular demand, I wrote a brief document more about HD-audio driver. > > > > It's now found in sound git tree as Documentation/sound/alsa/HD-Audio.txt. > > > > The git tree is: > > > > git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git > > > > > > > > You can get the text from web: > > > > http://git.kernel.org/?p=linux/kernel/git/tiwai/sound-2.6.git;a=blob_plain;… > > > > > > > > A PDF file is found at: > > > > http://ftp.kernel.org/pub/linux/kernel/people/tiwai/docs/HD-Audio.pdf > > > > > > Thank you for this. I found it informative and helpful. > > > > > > Maybe it could be placed on the wiki or sent to the > > > alsa-user list once a month as "how to troubleshoot hd audio"? > > > > Putting Wiki is a good idea. I'm not sure whether it'd be better to > > put the text into Wiki or to put a link to the above URL, though... > > > > > > Takashi > > _______________________________________________ > > Alsa-devel mailing list > > Alsa-devel(a)alsa-project.org > > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >

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[alsa-devel] Idea about the laptop speakers
by Alexander E. Patrakov 11 Dec '08

11 Dec '08

Hello developers, some time ago, people complained that the sound came out from both the laptop speakers and headphones. While in many cases this is not desirable, I think this effect (when configurable via opening a different PCM) can be useful in the following situation. Suppose that you have a laptop as your main working machine. The headset is always connected, but you are not always near the laptop (e.g., think better if you walk in the room). Then, you configure (by telling it to open different devices) your SIP softphone to play voice through headphones only, but play ringtones through both headphones and speakers. Thus, you can walk in the room without being afraid to miss a call. Is the above proposal implementable? -- Alexander E. Patrakov

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[alsa-devel] [PATCH 0/8] ASoC updates
by Mark Brown 10 Dec '08

10 Dec '08

The following changes since commit f73f2a6a23e34de9cca9672f727694e5af00e6c7: Takashi Iwai (1): ALSA: ASoC - Fix symbol conflicts in omac-mcbsp.c are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-tiwai Mark Brown (5): ASoC: Work around warnings from some build environments ASoC: Add codec registration API ASoC: Convert WM8900 to do more work at I2C probe time ASoC: Convert WM8903 driver to register at I2C probe time ASoC: Stop WM8903 SYSCLK when suspending Peter Ujfalusi (3): ASoC: TWL4030: Add missing Carkit output ASoC: TWL4030: Small cleanup ASoC: TWL4030: Change the name for the DACs include/sound/soc.h | 5 + sound/soc/codecs/twl4030.c | 27 +++--- sound/soc/codecs/wm8900.c | 159 +++++++++++++++-------------- sound/soc/codecs/wm8900.h | 7 -- sound/soc/codecs/wm8903.c | 241 +++++++++++++++++++------------------------- sound/soc/codecs/wm8903.h | 5 - sound/soc/soc-core.c | 43 ++++++++ 7 files changed, 247 insertions(+), 240 deletions(-)

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[alsa-devel] [PATCH] ALSA: Add support for mechanical jack insertion
by Mark Brown 10 Dec '08

10 Dec '08

Some systems support both mechanical and electrical jack detection, allowing them to report that a jack is physically present but does not have any functioning connections. Add a new jack type for these, allowing user space to report faulty connections. Thanks to Guillem Jover for the suggestion. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- include/linux/input.h | 1 + include/sound/jack.h | 1 + sound/core/jack.c | 6 ++++++ 3 files changed, 8 insertions(+), 0 deletions(-) diff --git a/include/linux/input.h b/include/linux/input.h index 1a958cd..9a6355f 100644 --- a/include/linux/input.h +++ b/include/linux/input.h @@ -660,6 +660,7 @@ struct input_absinfo { #define SW_MICROPHONE_INSERT 0x04 /* set = inserted */ #define SW_DOCK 0x05 /* set = plugged into dock */ #define SW_LINEOUT_INSERT 0x06 /* set = inserted */ +#define SW_JACK_PHYSICAL_INSERT 0x07 /* set = mechanical switch set */ #define SW_MAX 0x0f #define SW_CNT (SW_MAX+1) diff --git a/include/sound/jack.h b/include/sound/jack.h index 7cb25f4..2e0315c 100644 --- a/include/sound/jack.h +++ b/include/sound/jack.h @@ -36,6 +36,7 @@ enum snd_jack_types { SND_JACK_MICROPHONE = 0x0002, SND_JACK_HEADSET = SND_JACK_HEADPHONE | SND_JACK_MICROPHONE, SND_JACK_LINEOUT = 0x0004, + SND_JACK_MECHANICAL = 0x0008, /* If detected separately */ }; struct snd_jack { diff --git a/sound/core/jack.c b/sound/core/jack.c index 4978db1..7ca905c 100644 --- a/sound/core/jack.c +++ b/sound/core/jack.c @@ -108,6 +108,9 @@ int snd_jack_new(struct snd_card *card, const char *id, int type, if (type & SND_JACK_MICROPHONE) input_set_capability(jack->input_dev, EV_SW, SW_MICROPHONE_INSERT); + if (type & SND_JACK_MECHANICAL) + input_set_capability(jack->input_dev, EV_SW, + SW_JACK_PHYSICAL_INSERT); err = snd_device_new(card, SNDRV_DEV_JACK, jack, &ops); if (err < 0) @@ -162,6 +165,9 @@ void snd_jack_report(struct snd_jack *jack, int status) if (jack->type & SND_JACK_MICROPHONE) input_report_switch(jack->input_dev, SW_MICROPHONE_INSERT, status & SND_JACK_MICROPHONE); + if (jack->type & SND_JACK_MECHANICAL) + input_report_switch(jack->input_dev, SW_JACK_PHYSICAL_INSERT, + status & SND_JACK_MECHANICAL); input_sync(jack->input_dev); } -- 1.5.6.5

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[alsa-devel] Improvements on aloop driver
by Takashi Iwai 10 Dec '08

10 Dec '08

Hi, as some people have asked about aloop bugs and developments, I started working on it a bit. The fact is that aloop is buggy per design in two points: - Sharing the same buffer for both playback and capture doesn't work. The playback app may change the "available" buffer area at any time. It can update the last played period very quickly. OTOH, the capture needs to read the last period. So, it obviously conflicts. - We needs to synchronize both playback and capture streams. In the current implementation, both streams are updated individually. For example, the capture can start in the middle of the playback period. This results in the partial available samples. So, I wrote the code and got almost another shape of the driver. The result is found in my alsa-driver-build git tree git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/alsa-driver-build.git and the snapshot tarball ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/alsa-driver-sna… Let me know whether this works for you. Once after confirmed to work well enough, we should push this to the upstream -- or abandon if the driver is conceptually wrong... thanks, Takashi

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[alsa-devel] [PATCH 0/3] ASoC: TWL4030: Fixes and small cleanup
by Peter Ujfalusi 10 Dec '08

10 Dec '08

Hello, The CarkitL/R DAPM output somehow got lost, the first patch adds it. Really small cleanup for line wrap in the mux switch texts. DAC names changed in order to avoid confusion with the mux switch names. --- Peter Ujfalusi (3): ASoC: TWL4030: Add missing Carkit output ASoC: TWL4030: Small cleanup ASoC: TWL4030: Change the name for the DACs sound/soc/codecs/twl4030.c | 27 +++++++++++++-------------- 1 files changed, 13 insertions(+), 14 deletions(-)

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[alsa-devel] Ask a question about WM8976
by peter meng 10 Dec '08

10 Dec '08

Hi, I'm a newbie for alsa and using wm8976 first time . I found the driver of wm8976 comments the contents of wm8976_dapm_widgets . 1. Is this mean all widget in wm8976 does not have power management function ? 2. If wm8976 does not add widget then the process of 'set up audio path map ' will fails , is this correct ? ------------------------------------------------------------------- // TODO Widgets static const struct snd_soc_dapm_widget wm8976_dapm_widgets[] = { #if 0 //SND_SOC_DAPM_MUTE("Mono Mute", WM8976_MONOMIX, 6, 0), //SND_SOC_DAPM_MUTE("Speaker Mute", WM8976_SPKMIX, 6, 0), SND_SOC_DAPM_MIXER("Speaker Mixer", WM8976_POWER3, 2, 0, &wm8976_speaker_mixer_controls[0], ARRAY_SIZE(wm8976_speaker_mixer_controls)), SND_SOC_DAPM_MIXER("Mono Mixer", WM8976_POWER3, 3, 0, &wm8976_mono_mixer_controls[0], ARRAY_SIZE(wm8976_mono_mixer_controls)), SND_SOC_DAPM_DAC("DAC", "HiFi Playback", WM8976_POWER3, 0, 0), SND_SOC_DAPM_ADC("ADC", "HiFi Capture", WM8976_POWER3, 0, 0), SND_SOC_DAPM_PGA("Aux Input", WM8976_POWER1, 6, 0, NULL, 0), SND_SOC_DAPM_PGA("SpkN Out", WM8976_POWER3, 5, 0, NULL, 0), SND_SOC_DAPM_PGA("SpkP Out", WM8976_POWER3, 6, 0, NULL, 0), SND_SOC_DAPM_PGA("Mono Out", WM8976_POWER3, 7, 0, NULL, 0), SND_SOC_DAPM_PGA("Mic PGA", WM8976_POWER2, 2, 0, NULL, 0), SND_SOC_DAPM_PGA("Aux Boost", SND_SOC_NOPM, 0, 0, &wm8976_aux_boost_controls, 1), SND_SOC_DAPM_PGA("Mic Boost", SND_SOC_NOPM, 0, 0, &wm8976_mic_boost_controls, 1), SND_SOC_DAPM_SWITCH("Capture Boost", SND_SOC_NOPM, 0, 0, &wm8976_capture_boost_controls), SND_SOC_DAPM_MIXER("Boost Mixer", WM8976_POWER2, 4, 0, NULL, 0), SND_SOC_DAPM_MICBIAS("Mic Bias", WM8976_POWER1, 4, 0), SND_SOC_DAPM_INPUT("MICN"), SND_SOC_DAPM_INPUT("MICP"), SND_SOC_DAPM_INPUT("AUX"), SND_SOC_DAPM_OUTPUT("MONOOUT"), SND_SOC_DAPM_OUTPUT("SPKOUTP"), SND_SOC_DAPM_OUTPUT("SPKOUTN"), #endif }; static int wm8976_add_widgets(struct snd_soc_codec *codec) { int i; for(i = 0; i < ARRAY_SIZE(wm8976_dapm_widgets); i++) { snd_soc_dapm_new_control(codec, &wm8976_dapm_widgets[i]); } /* set up audio path map */ for(i = 0; audio_map[i][0] != NULL; i++) { snd_soc_dapm_connect_input(codec, audio_map[i][0], audio_map[i][1], audio_map[i][2]); } snd_soc_dapm_new_widgets(codec); return 0; } ----------------------------------------------- Thanks advanced . Best Regards Peter Meng

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Re: [alsa-devel] Warnings in wm8903.c (Re: Merged upstream to topic/asoc branch)
by Mark Brown 10 Dec '08

10 Dec '08

On Wed, Dec 10, 2008 at 08:26:56AM +0100, Takashi Iwai wrote: > Hmm, I got the warnings again on a x86-64 with gcc-4.3.2 randconfig > test build again. > The config is attached below. OK, I've managed to reproduce this with that .config. I'll push a patch later today which adds return statements to silence the warning, meaning that any genuine issues won't be suppressed.

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[alsa-devel] [PATCH 00/18] ASoC updates
by Mark Brown 10 Dec '08

10 Dec '08

The following changes since commit 28a1d869560a49d960ba2a3b450ec965712e5560: Daniel Mack (1): ASoC: tlv320aic3x: control additions and cleanups are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Mark Brown (9): ASoC: Add card registration API ASoC: Add DAI registration API ASoC: Register platform DAIs ASoC: Add platform registration API ASoC: Register platform drivers ASoC: Register non-AC97 codec DAIs ASoC: Initial framework for dynamic card instantiation ASoC: Wait for non-AC97 codec DAIs before instantiating ASoC: Convert WM8900 to allow registration by machine code Peter Ujfalusi (9): ASoC: TWL4030: Correct DAPM_DAC with power control ASoC: TWL4030: Add Analog PGA control switch to DAPM ASoC: TWL4030: Add DAPM event handler for output MUX selection ASoC: TWL4030: DAPM mapping of the Earpiece output ASoC: TWL4030: DAPM mapping of the PreDriv outputs ASoC: TWL4030: DAPM mapping of the Headset outputs ASoC: TWL4030: DAPM mapping of the Carkit outputs ASoC: TWL4030: DAPM mapping of the Handsfree outputs ASoC: TWL4030: Do not alter the Headset output volume on power-up/down include/sound/soc-dai.h | 8 + include/sound/soc.h | 10 ++ sound/soc/atmel/atmel-pcm.c | 12 ++ sound/soc/atmel/atmel_ssc_dai.c | 11 ++ sound/soc/au1x/dbdma2.c | 3 +- sound/soc/au1x/psc-ac97.c | 3 +- sound/soc/au1x/psc-i2s.c | 3 +- sound/soc/blackfin/bf5xx-ac97-pcm.c | 12 ++ sound/soc/blackfin/bf5xx-ac97.c | 12 ++ sound/soc/blackfin/bf5xx-i2s-pcm.c | 12 ++ sound/soc/blackfin/bf5xx-i2s.c | 12 ++ sound/soc/codecs/ad73311.c | 12 ++ sound/soc/codecs/ak4535.c | 12 ++ sound/soc/codecs/cs4270.c | 12 ++ sound/soc/codecs/pcm3008.c | 12 ++ sound/soc/codecs/ssm2602.c | 12 ++ sound/soc/codecs/tlv320aic23.c | 12 ++ sound/soc/codecs/tlv320aic26.c | 15 ++- sound/soc/codecs/tlv320aic3x.c | 12 ++ sound/soc/codecs/twl4030.c | 277 +++++++++++++++++++++++++++++++-- sound/soc/codecs/uda134x.c | 12 ++ sound/soc/codecs/uda1380.c | 12 ++ sound/soc/codecs/wm8510.c | 12 ++ sound/soc/codecs/wm8580.c | 12 ++ sound/soc/codecs/wm8728.c | 12 ++ sound/soc/codecs/wm8731.c | 12 ++ sound/soc/codecs/wm8750.c | 12 ++ sound/soc/codecs/wm8753.c | 12 ++ sound/soc/codecs/wm8900.c | 105 ++++--------- sound/soc/codecs/wm8903.c | 12 ++ sound/soc/codecs/wm8971.c | 12 ++ sound/soc/codecs/wm8990.c | 12 ++ sound/soc/davinci/davinci-i2s.c | 12 ++ sound/soc/davinci/davinci-pcm.c | 12 ++ sound/soc/fsl/fsl_dma.c | 12 ++ sound/soc/fsl/fsl_ssi.c | 10 ++ sound/soc/fsl/mpc5200_psc_i2s.c | 4 + sound/soc/omap/omap-mcbsp.c | 13 ++ sound/soc/omap/omap-pcm.c | 12 ++ sound/soc/pxa/pxa-ssp.c | 12 ++ sound/soc/pxa/pxa2xx-ac97.c | 12 ++ sound/soc/pxa/pxa2xx-i2s.c | 13 ++- sound/soc/pxa/pxa2xx-pcm.c | 12 ++ sound/soc/s3c24xx/s3c2412-i2s.c | 13 ++ sound/soc/s3c24xx/s3c2443-ac97.c | 13 ++ sound/soc/s3c24xx/s3c24xx-i2s.c | 12 ++ sound/soc/s3c24xx/s3c24xx-pcm.c | 12 ++ sound/soc/sh/dma-sh7760.c | 12 ++ sound/soc/sh/hac.c | 12 ++ sound/soc/sh/ssi.c | 12 ++ sound/soc/soc-core.c | 291 +++++++++++++++++++++++++++++++++-- 51 files changed, 1102 insertions(+), 110 deletions(-)

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[alsa-devel] No sound from laptop speakers (Resolved)
by Manilal K M 10 Dec '08

10 Dec '08

Hello everybody, Recently I have installed Fedora 10 on my Compaq CQ45 laptop. But unfortunately there was no sound from the laptop speakers even though the headphones worked perfectly. I did some google search but it didn't helped. Later, on a discussion at #alsa(freenode.net), one of the IRC member(nickname: copper) gave some clue and finally found that I have to upgrade the alsa driver and pass the model id(dell-m4-1) as a parameter in modprobe.conf. Here are the results generated by alsa-info.sh before and after upgrading to the latest alsa(1.0.18a). http://www.alsa-project.org/db/?f=9a29eac2731b3de1c17183d6c43aac3fa11942c9 http://www.alsa-project.org/db/?f=9a29eac2731b3de1c17183d6c43aac3fa11942c9 The option I have passed to /etc/modprobe.conf/alsa-base is : options snd-hda-intel model=dell-m4-1 After reboot, I was able to hear sound from laptop speakers. -- Manilal K M : മണിലാല്‍ കെ എം. http://libregeek.blogspot.com

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[alsa-devel] [PATCH 2/2] ASoC: TWL4030: Small cleanup
by Peter Ujfalusi 10 Dec '08

10 Dec '08

The mux switch related texts fits to on line, no need to wrap them. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/twl4030.c | 9 +++------ 1 files changed, 3 insertions(+), 6 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index df9f2ce..e3a7115 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -192,8 +192,7 @@ static void twl4030_init_chip(struct snd_soc_codec *codec) /* Earpiece */ static const char *twl4030_earpiece_texts[] = - {"Off", "DACL1", "DACL2", "Invalid", - "DACR1"}; + {"Off", "DACL1", "DACL2", "Invalid", "DACR1"}; static const struct soc_enum twl4030_earpiece_enum = SOC_ENUM_SINGLE(TWL4030_REG_EAR_CTL, 1, @@ -205,8 +204,7 @@ SOC_DAPM_ENUM("Route", twl4030_earpiece_enum); /* PreDrive Left */ static const char *twl4030_predrivel_texts[] = - {"Off", "DACL1", "DACL2", "Invalid", - "DACR2"}; + {"Off", "DACL1", "DACL2", "Invalid", "DACR2"}; static const struct soc_enum twl4030_predrivel_enum = SOC_ENUM_SINGLE(TWL4030_REG_PREDL_CTL, 1, @@ -218,8 +216,7 @@ SOC_DAPM_ENUM("Route", twl4030_predrivel_enum); /* PreDrive Right */ static const char *twl4030_predriver_texts[] = - {"Off", "DACR1", "DACR2", "Invalid", - "DACL2"}; + {"Off", "DACR1", "DACR2", "Invalid", "DACL2"}; static const struct soc_enum twl4030_predriver_enum = SOC_ENUM_SINGLE(TWL4030_REG_PREDR_CTL, 1, -- 1.6.0.4

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[alsa-devel] [PATCH 1/2] ASoC: TWL4030: Add missing Carkit output
by Peter Ujfalusi 10 Dec '08

10 Dec '08

SND_SOC_DAPM_OUTPUT definition for carkitL/R was missing. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/twl4030.c | 2 ++ 1 files changed, 2 insertions(+), 0 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 373daa4..df9f2ce 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -761,6 +761,8 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { SND_SOC_DAPM_OUTPUT("PREDRIVER"), SND_SOC_DAPM_OUTPUT("HSOL"), SND_SOC_DAPM_OUTPUT("HSOR"), + SND_SOC_DAPM_OUTPUT("CARKITL"), + SND_SOC_DAPM_OUTPUT("CARKITR"), SND_SOC_DAPM_OUTPUT("HFL"), SND_SOC_DAPM_OUTPUT("HFR"), -- 1.6.0.4

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[alsa-devel] How to write a PXA CPU
by Juan Pablo Gómez 09 Dec '08

09 Dec '08

Hi, I'm looking for documentation that explains me how to write a PXA document. It's for a NIOS 2 processor in an altera De2 board. -- Regards, Juan Pablo Muchas gracias. Atentamente, Juan Pablo Gómez

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[alsa-devel] [PATCH 0/9] ASoC: TWL4030: DAPM implementation for outputs (playback) - try3
by Peter Ujfalusi 09 Dec '08

09 Dec '08

Hello, The whole series again, altrough 1 and 2 has been accepted already. This series of patches (try3) adds DAPM implementation for all outputs found on TWL. The design allows one to configure all outputs independently from each other, since the internal routing of TWL actually treats them individually. The pre-DAC routings can be configured in the board specific files of needed. I have sent a patch which has added this to the codec driver. Based on the output routing selections all the corresponding DAC and analog PGA on the chain will be powered on. Remove the Headset output gain modifications from the twl4030_power_up and twl4030_power_down function, since they are not needed. Changes since try1 (comments addressed): - Removed the " Switch" from the option texts and also from the intercon. --- Peter Ujfalusi (9): ASoC: TWL4030: Correct DAPM_DAC with power control ASoC: TWL4030: Add Analog PGA control switch to DAPM ASoC: TWL4030: Add DAPM event handler for output MUX selection ASoC: TWL4030: DAPM mapping of the Earpiece output ASoC: TWL4030: DAPM mapping of the PreDriv outputs ASoC: TWL4030: DAPM mapping of the Headset outputs ASoC: TWL4030: DAPM mapping of the Carkit outputs ASoC: TWL4030: DAPM mapping of the Handsfree outputs ASoC: TWL4030: Do not alter the Headset output volume on power-up/down sound/soc/codecs/twl4030.c | 263 +++++++++++++++++++++++++++++++++++++++++--- 1 files changed, 248 insertions(+), 15 deletions(-)

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[alsa-devel] [PATCH 0/9] ASoC: TWL4030: DAPM implementation for outputs (playback) - try2
by Peter Ujfalusi 09 Dec '08

09 Dec '08

Hello, This series of patches (try2) adds DAPM implementation for all outputs found on TWL. The design allows one to configure all outputs independently from each other, since the internal routing of TWL actually treats them individually. The pre-DAC routings can be configured in the board specific files of needed. I have sent a patch which has added this to the codec driver. Based on the output routing selections all the corresponding DAC and analog PGA on the chain will be powered on. Remove the Headset output gain modifications from the twl4030_power_up and twl4030_power_down function, since they are not needed. Changes since try1 (comments addressed): - erapiece typo - too fast/slow finger combination ;) - Removed the " Switch" from the option texts - rename the confusing _controls to _control - Analog PGA control reworked: - SND_SOC_DAPM_PGA for power switch - Normal control stereo (user control) for DA enable/disable (mute/unmute): - "DAC1 Analog Playback Volume" + "DAC1 Analog Playback Switch" - "DAC2 Analog Playback Volume" + "DAC2 Analog Playback Switch" --- Peter Ujfalusi (9): ASoC: TWL4030: Correct DAPM_DAC with power control ASoC: TWL4030: Add Analog PGA control switch to DAPM ASoC: TWL4030: Add DAPM event handler for output MUX selection ASoC: TWL4030: DAPM mapping of the Earpiece output ASoC: TWL4030: DAPM mapping of the PreDriv outputs ASoC: TWL4030: DAPM mapping of the Headset outputs ASoC: TWL4030: DAPM mapping of the Carkit outputs ASoC: TWL4030: DAPM mapping of the Handsfree outputs ASoC: TWL4030: Do not alter the Headset output volume on power-up/down sound/soc/codecs/twl4030.c | 263 +++++++++++++++++++++++++++++++++++++++++--- 1 files changed, 248 insertions(+), 15 deletions(-)

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[alsa-devel] [PATCH 0/8] ASoC: TWL4030: DAPM implementation for outputs (playback)
by Peter Ujfalusi 09 Dec '08

09 Dec '08

Hello, This series of patches adds DAPM implementation for all outputs found on TWL. The design allows one to configure all outputs independently from each other, since the internal routing of TWL actually treats them individually. The pre-DAC routings can be configured in the board specific files of needed. I have sent a patch which has added this to the codec driver. Based on the output routing selections all the corresponding DAC and analog PGA on the chain will be powered on. --- Peter Ujfalusi (8): ASoC: TWL4030: Correct DAPM_DAC with power control ASoC: TWL4030: Add Analog PGA control switch to DAPM ASoC: TWL4030: Add DAPM event handler for output MUX selection ASoC: TWL4030: DAPM mapping of the Earpiece output ASoC: TWL4030: DAPM mapping of the PreDriv outputs ASoC: TWL4030: DAPM mapping of the Headset outputs ASoC: TWL4030: DAPM mapping of the Carkit outputs ASoC: TWL4030: DAPM mapping of the Handsfree outputs sound/soc/codecs/twl4030.c | 248 +++++++++++++++++++++++++++++++++++++++++++- 1 files changed, 243 insertions(+), 5 deletions(-)

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