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

2 2

[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

1 0

[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

1 0

[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

2 3

[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(-)

2 10

[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(-)

2 13

[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(-)

2 20

[alsa-devel] ASoC: hook for codec control updates and clock controls
by Daniel Mack 08 Dec '08

08 Dec '08

Hi, Codecs like the tlv320aic3x family have soft volume controls to smoothly drive up and down mixer elements in order to prevent clicks and pops. In fact, most of the controls on this chip are implemented as such. For the mixers to work, however, the I2S input clock is needed which is currently only true when the DAC/ACD elements are active but not when the chip is just operating as an analog mixer. To allow this functionality, the I2S clock must be present for a short period after each volume control update to the codec and disabled again after some milliseconds. It should, however, not be enabled if it's already running and of course, it must not be disabled if the ADC/DAC is still running. Adding a hook to struct snd_soc_codec which is called from snd_soc_update_bits() isn't a big deal, but the problem is the abstraction layers in this case - for good reasons, the cpu_dai is not accessible from the codec dai. Or did I miss a link? Is there a sane way to address this problem? Thanks and best regards, Daniel

3 9

[alsa-devel] [patch] alsa-utils fix for intl support
by Hans-Christian Egtvedt 08 Dec '08

08 Dec '08

Hi, Feel free to comment on the included patch. I had to add it because aseqnet would not link without -lintl. -- Best regards, Hans-Christian Egtvedt

2 5

[alsa-devel] [patch] alsa-utils check if __USE_BSD is defined before compiling "BSD functions"
by Hans-Christian Egtvedt 08 Dec '08

08 Dec '08

Hi, Another bug/issue I tripped over when compiling alsa-utils in an environment using uClibc to supply the C library functions. Here I have enabled some old BSD style functions. The attached patch will honor them if they are enabled. Without this patch I get a redefined error during compile. -- Best regards, Hans-Christian Egtvedt

2 3

[alsa-devel] [PATCH 0/7] ASoC updates
by Mark Brown 08 Dec '08

08 Dec '08

The following changes since commit 682d5874f3d654b5d13d9b8dd56b9e05cfadd01b: Takashi Iwai (1): ALSA: ASoC: Fix old-style trigger callback in s3c2443-ac97.c are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Mark Brown (4): ASoC: Rename snd_soc_register_card() to snd_soc_init_card() ASoC: Annotate core removal function ASoC: Remove in-code changelog from AD73311 driver ASoC: Don't free static data in WM9713 Peter Ujfalusi (3): ASoC: TWL4030: Change the capture volume control to TLV ASoC: TWL4030: Change the common playback volume controls ASoC: TWL4030: Add volume controls for outputs include/sound/soc.h | 2 +- sound/soc/codecs/ac97.c | 2 +- sound/soc/codecs/ad1980.c | 2 +- sound/soc/codecs/ad73311.c | 5 +-- sound/soc/codecs/ak4535.c | 2 +- sound/soc/codecs/cs4270.c | 2 +- sound/soc/codecs/pcm3008.c | 2 +- sound/soc/codecs/ssm2602.c | 2 +- sound/soc/codecs/tlv320aic23.c | 2 +- sound/soc/codecs/tlv320aic26.c | 2 +- sound/soc/codecs/tlv320aic3x.c | 2 +- sound/soc/codecs/twl4030.c | 74 +++++++++++++++++++++++++++++++++------ sound/soc/codecs/uda134x.c | 2 +- sound/soc/codecs/uda1380.c | 2 +- sound/soc/codecs/wm8510.c | 2 +- sound/soc/codecs/wm8580.c | 2 +- sound/soc/codecs/wm8728.c | 2 +- sound/soc/codecs/wm8731.c | 2 +- sound/soc/codecs/wm8750.c | 2 +- sound/soc/codecs/wm8753.c | 2 +- sound/soc/codecs/wm8900.c | 2 +- sound/soc/codecs/wm8903.c | 2 +- sound/soc/codecs/wm8971.c | 2 +- sound/soc/codecs/wm8990.c | 2 +- sound/soc/codecs/wm9712.c | 2 +- sound/soc/codecs/wm9713.c | 3 +- sound/soc/soc-core.c | 8 ++-- 27 files changed, 91 insertions(+), 45 deletions(-)

4 12

[alsa-devel] HDA: CLFE DAC on ALC889 (Acer Aspire 8930G)
by Hector Martin 07 Dec '08

07 Dec '08

I've been trying to get ALSA working with this laptop. So far the attached patch adds rear support to the existing 4930g model, and tries to add CLFE support. However, I haven't been able to get CLFE to work. I don't seem to get any audio through DACs/mixers other than the first two (front and rear). I can assign the CLFE output (pin 0x16) to the front or rear DAC and hear the front audio through the CLFE speakers. I can also force the front DAC (0x02) or rear DAC (0x03) to use the CLFE stream and hear it through the front or rear speakers, so the audio is getting to the codec. But either DACs 0x04 and 0x05 aren't working, or their associated mixers (0x0e and 0x0f) aren't passing through the audio, or the OUT pins don't like selections above 1. I've checked that they're unmuted and that the volumes are turned up on both. I've also tried the 0x25->0x26 DAC/mixer auxiliary pathway, which doesn't work either. Here's what the relevant NIDs correspond to: 0x02 - (normally) Front DAC 0x03 - (normally) Rear DAC 0x04 - (normally) CLFE DAC 0x05 - (presumably) side DAC (no side speakers) 0x0c - (normally) Front mixer 0x0d - (normally) Rear mixer 0x0e - (normally) CLFE mixer 0x0f - (presumably) side mixer 0x14 - Front speakers pin 0x15 - Headphones pin (also EAPD for speaker amp) 0x16 - CLFE speakers pin 0x1b - Rear speakers pin Here's what the codec graph looks like: http://marcansoft.com/transf/codec.png Any ideas? I'm pretty stumped here. Are there any other magic bits to enable the remaining DACs or mixers? -- Hector Martin (hector(a)marcansoft.com) Public Key: http://www.marcansoft.com/marcan.asc

2 3

Re: [alsa-devel] [Swfdec] Can't use Mozilla plugin or play video with 0.8.2 / 0.9.2
by Benjamin Otte 07 Dec '08

07 Dec '08

I think I know what this is. But I spent the best part of my breakfast wondering about it. I think using Pulse Audio might work around this problem, but I think it's a driver, configuration issue on the ALSA side or a misunderstanding of the API on my part, which is why I'm sending this to the ALSA list. So here's the problem: Swfdec uses ALSA in non-blocking mode and polls its file descriptor(s). Whenever the audio queue becomes empty enough that it's useful to render more audio, the file descriptor wakes up from the poll. This setting is apparently configurable using various hw params, however, Swfdec does not configure it and has been relying on the fact that the sound device picks proper defaults, thinking that a sound card with a small buffer will likely want to wake up more often than something like Pulse Audio's new design, which keeps a 2+ second buffer for the audio. It seems though that your sound card wakes up roughly whenever a new sample can be rendered, which is 44100 times per second (or more, as from your profile it looks like ALSA resamples the audio). This causes the whole rendering code inside Swfdec to run for every sample, which is so much overhead that it kills all your performance. My question to you ALSA people now is: Who's responsible for ensuring that the number of wakeups is sane? Should I do some configuring of the audio device and if so, what kind of configuration, or should ALSA be able to pick reasonable defaults for me? Cheers, Benjamin On Sun, Dec 7, 2008 at 2:46 AM, Jesse Zbikowski <embeddedlinuxguy(a)gmail.com> wrote: > Hi Riccardo, > > On Fri, Dec 5, 2008 at 2:32 PM, Riccardo Magliocchetti > <riccardo.magliocchetti(a)gmail.com> wrote: >> Give it a run with oprofile, this is my opreport > > Thanks for this data. It looks like I may have an audio bottleneck. > opreport -t3 shows > > 31.5811 no-vmlinux > 14.4701 libswfdec-0.9.so.3.0.0 > 8.3210 libasound_module_rate_speexrate.so > 6.0088 libc-2.8.90.so > 4.7060 libpixman-1.so.0.12.0 > 3.6615 python2.5 > 3.2831 libcairo.so.2.10800.0 > > If I just look at the libraries I see the top libswfdec routines are > all audio related: > > 14.1497 libswfdec-0.9.so.3.0.0 swfdec_sound_buffer_get_n_samples > 2.1666 libswfdec-0.9.so.3.0.0 swfdec_audio_format_get_channels > 1.9369 libswfdec-0.9.so.3.0.0 swfdec_audio_format_new > 1.7904 libswfdec-0.9.so.3.0.0 swfdec_audio_flv_render > 1.3205 libswfdec-0.9.so.3.0.0 __i686.get_pc_thunk.bx > 1.2073 libswfdec-0.9.so.3.0.0 swfdec_audio_format_get_granularity > > I could try using the PulseAudio backend instead of ALSA. > >> You can try to improve your performance by using latest stable version of >> cairo and pixman > > OK, I am currently using the latest Ubuntu versions which are a bit > behind latest stable. I will see if this makes a difference. > > Package Ibex Latest > xorg-xserver 1.5.2 1.5.3 > cairo 1.8.0 1.8.4 > pixman 0.12 0.13.2 > >> play a bit with your X video driver config ie. exa vs xaa. > > I did not see libexa.so in my oprofile output although it is > installed, so I will make sure this is enabled. > > Thanks again, > Jesse Z. > _______________________________________________ > Swfdec mailing list > Swfdec(a)lists.freedesktop.org > http://lists.freedesktop.org/mailman/listinfo/swfdec >

1 0

Re: [alsa-devel] Status of the aloop loopback driver
by Takashi Iwai 05 Dec '08

05 Dec '08

At Fri, 5 Dec 2008 17:11:07 +0000, Andy Whitcroft wrote: > > I am interesting in finding out the status of, and plans for the aloop > loopback driver. I note that this has been about for a while and has > not been pushed up to the mainline kernel. Are you intending on > pushing that up or is it planning on being in the alsa-drivers tarball > for the forseeable future? It's still buggy, unfortunately, due to the current design. The timing of capture and playback isn't synchronized, so you'll get often noisy outputs. I've worked on this driver yesterday, just did some clean up works. Hopefully I'll be able to fix the issue sometime later... thanks, Takashi

1 0

[alsa-devel] [PATCH] tlv320aic3x: control additions and cleanups
by Daniel Mack 05 Dec '08

05 Dec '08

Some minor updates for sound/soc/codecs/tlv320aic3x.c: - removed "DAC" from name for "Mono playback switch", "HP playback switch" and "HPCOM playback switch". The registers used in the macro don't have anything to do with the DAC but mute the whole output, hence the names were misleading. - split "Line Playback Switch" into "LineL Playback Switch" and "LineR Playback Switch" - split "Line PGA Bypass Playback Volume" into "LineL Left PGA Bypass Playback Volume" and "LineR Right PGA Bypass Playback Volume" - split "Line Line2 Bypass Playback Volume" into "LineL Line2 Bypass Playback Volume" and "LineR Line2 Bypass Playback Volume" - Added "HP Right PGA Bypass Playback Volume" Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Signed-off-by: Uli Luckas <u.luckas(a)road.de> --- sound/soc/codecs/tlv320aic3x.c | 24 ++++++++++++++++-------- 1 files changed, 16 insertions(+), 8 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 537bbbf..7cdaea3 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -253,16 +253,22 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_DOUBLE_R("Line DAC Playback Volume", DACL1_2_LLOPM_VOL, DACR1_2_RLOPM_VOL, 0, 0x7f, 1), - SOC_DOUBLE_R("Line DAC Playback Switch", LLOPM_CTRL, RLOPM_CTRL, 3, - 0x01, 0), - SOC_DOUBLE_R("Line PGA Bypass Playback Volume", PGAL_2_LLOPM_VOL, - PGAR_2_RLOPM_VOL, 0, 0x7f, 1), - SOC_DOUBLE_R("Line Line2 Bypass Playback Volume", LINE2L_2_LLOPM_VOL, + SOC_SINGLE("LineL Playback Switch", LLOPM_CTRL, 3, 0x01, 0), + SOC_SINGLE("LineR Playback Switch", RLOPM_CTRL, 3, 0x01, 0), + SOC_DOUBLE_R("LineL DAC Playback Volume", DACL1_2_LLOPM_VOL, + DACR1_2_LLOPM_VOL, 0, 0x7f, 1), + SOC_SINGLE("LineL Left PGA Bypass Playback Volume", PGAL_2_LLOPM_VOL, + 0, 0x7f, 1), + SOC_SINGLE("LineR Right PGA Bypass Playback Volume", PGAR_2_RLOPM_VOL, + 0, 0x7f, 1), + SOC_DOUBLE_R("LineL Line2 Bypass Playback Volume", LINE2L_2_LLOPM_VOL, + LINE2R_2_LLOPM_VOL, 0, 0x7f, 1), + SOC_DOUBLE_R("LineR Line2 Bypass Playback Volume", LINE2L_2_RLOPM_VOL, LINE2R_2_RLOPM_VOL, 0, 0x7f, 1), SOC_DOUBLE_R("Mono DAC Playback Volume", DACL1_2_MONOLOPM_VOL, DACR1_2_MONOLOPM_VOL, 0, 0x7f, 1), - SOC_SINGLE("Mono DAC Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0), + SOC_SINGLE("Mono Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0), SOC_DOUBLE_R("Mono PGA Bypass Playback Volume", PGAL_2_MONOLOPM_VOL, PGAR_2_MONOLOPM_VOL, 0, 0x7f, 1), SOC_DOUBLE_R("Mono Line2 Bypass Playback Volume", LINE2L_2_MONOLOPM_VOL, @@ -270,8 +276,10 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_DOUBLE_R("HP DAC Playback Volume", DACL1_2_HPLOUT_VOL, DACR1_2_HPROUT_VOL, 0, 0x7f, 1), - SOC_DOUBLE_R("HP DAC Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3, + SOC_DOUBLE_R("HP Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3, 0x01, 0), + SOC_DOUBLE_R("HP Right PGA Bypass Playback Volume", PGAR_2_HPLOUT_VOL, + PGAR_2_HPROUT_VOL, 0, 0x7f, 1), SOC_SINGLE("HPL PGA Bypass Playback Volume", PGAL_2_HPLOUT_VOL, 0, 0x7f, 1), SOC_SINGLE("HPR PGA Bypass Playback Volume", PGAL_2_HPROUT_VOL, @@ -281,7 +289,7 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_DOUBLE_R("HPCOM DAC Playback Volume", DACL1_2_HPLCOM_VOL, DACR1_2_HPRCOM_VOL, 0, 0x7f, 1), - SOC_DOUBLE_R("HPCOM DAC Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3, + SOC_DOUBLE_R("HPCOM Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3, 0x01, 0), SOC_SINGLE("HPLCOM PGA Bypass Playback Volume", PGAL_2_HPLCOM_VOL, 0, 0x7f, 1), -- 1.5.6.5

3 18

[alsa-devel] [PATCH] tlv320aic3x.c: controller naming fixes
by Daniel Mack 05 Dec '08

05 Dec '08

- removed "DAC" from name for "Mono playback switch", "HP playback switch" and "HPCOM playback switch". The registers used in the macro don't have anything to do with the DAC but mute the whole output, hence the names were misleading. - renamed "HPL PGA Bypass Playback Volume" to "HPL Left PGA Bypass Playback Volume" - renamed "HPR PGA Bypass Playback Volume" to "HPR Left PGA Bypass Playback Volume" Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- This is the second part of the patch I sent yesterday, plus the naming fixes recently discussed on the list. sound/soc/codecs/tlv320aic3x.c | 10 +++++----- 1 files changed, 5 insertions(+), 5 deletions(-) diff --git a/sound/soc/codecs/tlv320aic3x.c b/sound/soc/codecs/tlv320aic3x.c index 6a12dc3..081f7c4 100644 --- a/sound/soc/codecs/tlv320aic3x.c +++ b/sound/soc/codecs/tlv320aic3x.c @@ -268,7 +268,7 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_DOUBLE_R("Mono DAC Playback Volume", DACL1_2_MONOLOPM_VOL, DACR1_2_MONOLOPM_VOL, 0, 0x7f, 1), - SOC_SINGLE("Mono DAC Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0), + SOC_SINGLE("Mono Playback Switch", MONOLOPM_CTRL, 3, 0x01, 0), SOC_DOUBLE_R("Mono PGA Bypass Playback Volume", PGAL_2_MONOLOPM_VOL, PGAR_2_MONOLOPM_VOL, 0, 0x7f, 1), SOC_DOUBLE_R("Mono Line2 Bypass Playback Volume", LINE2L_2_MONOLOPM_VOL, @@ -276,20 +276,20 @@ static const struct snd_kcontrol_new aic3x_snd_controls[] = { SOC_DOUBLE_R("HP DAC Playback Volume", DACL1_2_HPLOUT_VOL, DACR1_2_HPROUT_VOL, 0, 0x7f, 1), - SOC_DOUBLE_R("HP DAC Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3, + SOC_DOUBLE_R("HP Playback Switch", HPLOUT_CTRL, HPROUT_CTRL, 3, 0x01, 0), SOC_DOUBLE_R("HP Right PGA Bypass Playback Volume", PGAR_2_HPLOUT_VOL, PGAR_2_HPROUT_VOL, 0, 0x7f, 1), - SOC_SINGLE("HPL PGA Bypass Playback Volume", PGAL_2_HPLOUT_VOL, + SOC_SINGLE("HPL Left PGA Bypass Playback Volume", PGAL_2_HPLOUT_VOL, 0, 0x7f, 1), - SOC_SINGLE("HPR PGA Bypass Playback Volume", PGAL_2_HPROUT_VOL, + SOC_SINGLE("HPR Left PGA Bypass Playback Volume", PGAL_2_HPROUT_VOL, 0, 0x7f, 1), SOC_DOUBLE_R("HP Line2 Bypass Playback Volume", LINE2L_2_HPLOUT_VOL, LINE2R_2_HPROUT_VOL, 0, 0x7f, 1), SOC_DOUBLE_R("HPCOM DAC Playback Volume", DACL1_2_HPLCOM_VOL, DACR1_2_HPRCOM_VOL, 0, 0x7f, 1), - SOC_DOUBLE_R("HPCOM DAC Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3, + SOC_DOUBLE_R("HPCOM Playback Switch", HPLCOM_CTRL, HPRCOM_CTRL, 3, 0x01, 0), SOC_SINGLE("HPLCOM PGA Bypass Playback Volume", PGAL_2_HPLCOM_VOL, 0, 0x7f, 1), -- 1.5.6.5

1 0

[alsa-devel] [PATCH] ASoC: Add support for OMAP3 Pandora
by Grazvydas Ignotas 05 Dec '08

05 Dec '08

This patch adds basic support for OMAP3 Pandora. Signed-off-by: Grazvydas Ignotas <notasas(a)gmail.com> --- sound/soc/omap/Kconfig | 8 + sound/soc/omap/Makefile | 2 + sound/soc/omap/omap3pandora.c | 311 +++++++++++++++++++++++++++++++++++++++++ 3 files changed, 321 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/omap3pandora.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index da39f27..a7b1d77 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -46,3 +46,11 @@ config SND_OMAP_SOC_SDP3430 help Say Y if you want to add support for SoC audio on Texas Instruments SDP3430. + +config SND_OMAP_SOC_OMAP3_PANDORA + tristate "SoC Audio support for OMAP3 Pandora" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_PANDORA + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the OMAP3 Pandora. diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 29cf3a8..76fedd9 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -11,9 +11,11 @@ snd-soc-osk5912-objs := osk5912.o snd-soc-overo-objs := overo.o snd-soc-omap2evm-objs := omap2evm.o snd-soc-sdp3430-objs := sdp3430.o +snd-soc-omap3pandora-objs := omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o +obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c new file mode 100644 index 0000000..bd91594 --- /dev/null +++ b/sound/soc/omap/omap3pandora.c @@ -0,0 +1,311 @@ +/* + * omap3pandora.c -- SoC audio for Pandora Handheld Console + * + * Author: Gražvydas Ignotas <notasas(a)gmail.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <linux/gpio.h> +#include <linux/delay.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +#define OMAP3_PANDORA_DAC_POWER_GPIO 118 +#define OMAP3_PANDORA_AMP_POWER_GPIO 14 + +#define PREFIX "ASoC omap3pandora: " + +static int omap3pandora_cmn_hw_params(struct snd_soc_dai *codec_dai, + struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + pr_err(PREFIX "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) { + pr_err(PREFIX "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err(PREFIX "can't set codec system clock\n"); + return ret; + } + + /* Set McBSP clock to external */ + ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_EXT, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err(PREFIX "can't set cpu system clock\n"); + return ret; + } + + ret = snd_soc_dai_set_clkdiv(cpu_dai, OMAP_MCBSP_CLKGDV, 8); + if (ret < 0) { + pr_err(PREFIX "can't set SRG clock divider\n"); + return ret; + } + + return 0; +} + +static int omap3pandora_out_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + return omap3pandora_cmn_hw_params(codec_dai, cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBS_CFS); +} + +static int omap3pandora_in_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + return omap3pandora_cmn_hw_params(codec_dai, cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); +} + +static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + if (SND_SOC_DAPM_EVENT_ON(event)) { + gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, 1); + gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 1); + } else { + gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 0); + mdelay(1); + gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, 0); + } + + return 0; +} + +/* + * Audio paths on Pandora board: + * + * |O| ---> PCM DAC +-> AMP -> Headphone Jack + * |M| A +--------> Line Out + * |A| <~~clk~~+ + * |P| <--- TWL4030 <--------- Line In and MICs + */ +static const struct snd_soc_dapm_widget omap3pandora_out_dapm_widgets[] = { + SND_SOC_DAPM_DAC("PCM DAC", "Playback", SND_SOC_NOPM, 0, 0), + SND_SOC_DAPM_PGA_E("Headphone Amplifier", SND_SOC_NOPM, + 0, 0, NULL, 0, omap3pandora_hp_event, + SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD), + SND_SOC_DAPM_HP("Headphone Jack", NULL), + SND_SOC_DAPM_LINE("Line Out", NULL), +}; + +static const struct snd_soc_dapm_widget omap3pandora_in_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Mic (Internal)", NULL), + SND_SOC_DAPM_MIC("Mic (external)", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), +}; + +static const struct snd_soc_dapm_route omap3pandora_out_map[] = { + {"Headphone Amplifier", NULL, "PCM DAC"}, + {"Line Out", NULL, "PCM DAC"}, + {"Headphone Jack", NULL, "Headphone Amplifier"}, +}; + +static const struct snd_soc_dapm_route omap3pandora_in_map[] = { + {"INL", NULL, "Line In"}, + {"INR", NULL, "Line In"}, + {"INL", NULL, "Mic (Internal)"}, + {"INR", NULL, "Mic (external)"}, +}; + +static int omap3pandora_out_init(struct snd_soc_codec *codec) +{ + int ret; + + ret = snd_soc_dapm_new_controls(codec, omap3pandora_out_dapm_widgets, + ARRAY_SIZE(omap3pandora_out_dapm_widgets)); + if (ret < 0) + return ret; + + snd_soc_dapm_add_routes(codec, omap3pandora_out_map, + ARRAY_SIZE(omap3pandora_out_map)); + + return snd_soc_dapm_sync(codec); +} + +static int omap3pandora_in_init(struct snd_soc_codec *codec) +{ + int ret; + + ret = snd_soc_dapm_new_controls(codec, omap3pandora_in_dapm_widgets, + ARRAY_SIZE(omap3pandora_in_dapm_widgets)); + if (ret < 0) + return ret; + + snd_soc_dapm_add_routes(codec, omap3pandora_in_map, + ARRAY_SIZE(omap3pandora_in_map)); + + return snd_soc_dapm_sync(codec); +} + +static struct snd_soc_ops omap3pandora_out_ops = { + .hw_params = omap3pandora_out_hw_params, +}; + +static struct snd_soc_ops omap3pandora_in_ops = { + .hw_params = omap3pandora_in_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link omap3pandora_dai[] = { + { + .name = "PCM1773", + .stream_name = "HiFi Out", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai, + .ops = &omap3pandora_out_ops, + .init = omap3pandora_out_init, + }, { + .name = "TWL4030", + .stream_name = "Line/Mic In", + .cpu_dai = &omap_mcbsp_dai[1], + .codec_dai = &twl4030_dai, + .ops = &omap3pandora_in_ops, + .init = omap3pandora_in_init, + } +}; + +/* SoC card */ +static struct snd_soc_card snd_soc_card_omap3pandora = { + .name = "omap3pandora", + .platform = &omap_soc_platform, + .dai_link = omap3pandora_dai, + .num_links = ARRAY_SIZE(omap3pandora_dai), +}; + +/* Audio subsystem */ +static struct snd_soc_device omap3pandora_snd_data = { + .card = &snd_soc_card_omap3pandora, + .codec_dev = &soc_codec_dev_twl4030, +}; + +static struct platform_device *omap3pandora_snd_device; + +static int __init omap3pandora_soc_init(void) +{ + int ret; + + if (!machine_is_omap3_pandora()) { + pr_debug(PREFIX "Not OMAP3 Pandora\n"); + return -ENODEV; + } + pr_info("OMAP3 Pandora SoC init\n"); + + ret = gpio_request(OMAP3_PANDORA_DAC_POWER_GPIO, "dac_power"); + if (ret) { + pr_err(PREFIX "Failed to get DAC power GPIO\n"); + return ret; + } + + ret = gpio_direction_output(OMAP3_PANDORA_DAC_POWER_GPIO, 0); + if (ret) { + pr_err(PREFIX "Failed to set DAC power GPIO direction\n"); + goto fail0; + } + + ret = gpio_request(OMAP3_PANDORA_AMP_POWER_GPIO, "amp_power"); + if (ret) { + pr_err(PREFIX "Failed to get amp power GPIO\n"); + goto fail0; + } + + ret = gpio_direction_output(OMAP3_PANDORA_AMP_POWER_GPIO, 0); + if (ret) { + pr_err(PREFIX "Failed to set amp power GPIO direction\n"); + goto fail1; + } + + omap3pandora_snd_device = platform_device_alloc("soc-audio", -1); + if (omap3pandora_snd_device == NULL) { + pr_err(PREFIX "Platform device allocation failed\n"); + ret = -ENOMEM; + goto fail1; + } + + platform_set_drvdata(omap3pandora_snd_device, &omap3pandora_snd_data); + omap3pandora_snd_data.dev = &omap3pandora_snd_device->dev; + *(unsigned int *)omap_mcbsp_dai[0].private_data = 1; /* McBSP2 */ + *(unsigned int *)omap_mcbsp_dai[1].private_data = 3; /* McBSP4 */ + + ret = platform_device_add(omap3pandora_snd_device); + if (ret) { + pr_err(PREFIX "Unable to add platform device\n"); + goto fail2; + } + + return 0; + +fail2: + platform_device_put(omap3pandora_snd_device); +fail1: + gpio_free(OMAP3_PANDORA_AMP_POWER_GPIO); +fail0: + gpio_free(OMAP3_PANDORA_DAC_POWER_GPIO); + return ret; +} +module_init(omap3pandora_soc_init); + +static void __exit omap3pandora_soc_exit(void) +{ + platform_device_unregister(omap3pandora_snd_device); + gpio_free(OMAP3_PANDORA_AMP_POWER_GPIO); + gpio_free(OMAP3_PANDORA_DAC_POWER_GPIO); +} +module_exit(omap3pandora_soc_exit); + +MODULE_AUTHOR("Grazvydas Ignotas <notasas(a)gmail.com>"); +MODULE_DESCRIPTION("ALSA SoC OMAP3 Pandora"); +MODULE_LICENSE("GPL"); -- 1.5.4.3

2 3

[alsa-devel] [PATCH] ASoC: Add support for OMAP3 Pandora
by Grazvydas Ignotas 04 Dec '08

04 Dec '08

This patch adds basic support for OMAP3 Pandora. Signed-off-by: Grazvydas Ignotas <notasas(a)gmail.com> --- I'm not sure if splitting DAPM and widgets was good idea, I can adjust if needed. The map is also not quite right, it says headphone jack is connected to TWL4030 codec OUTL and OUTR, but that is not really the case. It is connected to external DAC, and DAC is connected to OMAP. However DAC uses clock from TWL4030, so TWL needs to be active. I've basically done that connection to get event callback, where I can switch DAC and headphone amp on/off. sound/soc/omap/Kconfig | 8 + sound/soc/omap/Makefile | 2 + sound/soc/omap/omap3pandora.c | 293 +++++++++++++++++++++++++++++++++++++++++ 3 files changed, 303 insertions(+), 0 deletions(-) create mode 100644 sound/soc/omap/omap3pandora.c diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig index da39f27..a7b1d77 100644 --- a/sound/soc/omap/Kconfig +++ b/sound/soc/omap/Kconfig @@ -46,3 +46,11 @@ config SND_OMAP_SOC_SDP3430 help Say Y if you want to add support for SoC audio on Texas Instruments SDP3430. + +config SND_OMAP_SOC_OMAP3_PANDORA + tristate "SoC Audio support for OMAP3 Pandora" + depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_PANDORA + select SND_OMAP_SOC_MCBSP + select SND_SOC_TWL4030 + help + Say Y if you want to add support for SoC audio on the OMAP3 Pandora. diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile index 29cf3a8..76fedd9 100644 --- a/sound/soc/omap/Makefile +++ b/sound/soc/omap/Makefile @@ -11,9 +11,11 @@ snd-soc-osk5912-objs := osk5912.o snd-soc-overo-objs := overo.o snd-soc-omap2evm-objs := omap2evm.o snd-soc-sdp3430-objs := sdp3430.o +snd-soc-omap3pandora-objs := omap3pandora.o obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o +obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c new file mode 100644 index 0000000..4f92877 --- /dev/null +++ b/sound/soc/omap/omap3pandora.c @@ -0,0 +1,293 @@ +/* + * omap3pandora.c -- SoC audio for Pandora Handheld Console + * + * Author: Gražvydas Ignotas <notasas(a)gmail.com> + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * version 2 as published by the Free Software Foundation. + * + * This program is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA + * 02110-1301 USA + * + */ + +#include <linux/clk.h> +#include <linux/platform_device.h> +#include <linux/gpio.h> + +#include <sound/core.h> +#include <sound/pcm.h> +#include <sound/soc.h> +#include <sound/soc-dapm.h> + +#include <asm/mach-types.h> + +#include "omap-mcbsp.h" +#include "omap-pcm.h" +#include "../codecs/twl4030.h" + +#define OMAP3_PANDORA_DAC_POWER_GPIO 118 +#define OMAP3_PANDORA_AMP_POWER_GPIO 14 + +#define PREFIX "ASoC omap3pandora: " + +static int omap3pandora_cmn_hw_params(struct snd_soc_dai *codec_dai, + struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + int ret; + + /* Set codec DAI configuration */ + ret = snd_soc_dai_set_fmt(codec_dai, fmt); + if (ret < 0) { + pr_err(PREFIX "can't set codec DAI configuration\n"); + return ret; + } + + /* Set cpu DAI configuration */ + ret = snd_soc_dai_set_fmt(cpu_dai, fmt); + if (ret < 0) { + pr_err(PREFIX "can't set cpu DAI configuration\n"); + return ret; + } + + /* Set the codec system clock for DAC and ADC */ + ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err(PREFIX "can't set codec system clock\n"); + return ret; + } + + /* Set McBSP clock to external */ + ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_EXT, 0, + SND_SOC_CLOCK_IN); + if (ret < 0) { + pr_err(PREFIX "can't set cpu system clock\n"); + return ret; + } + + ret = snd_soc_dai_set_clkdiv(cpu_dai, OMAP_MCBSP_CLKGDV, 8); + if (ret < 0) { + pr_err(PREFIX "can't set SRG clock divider\n"); + return ret; + } + + return 0; +} + +static int omap3pandora_out_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + return omap3pandora_cmn_hw_params(codec_dai, cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_IB_NF | + SND_SOC_DAIFMT_CBS_CFS); +} + +static int omap3pandora_in_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + + return omap3pandora_cmn_hw_params(codec_dai, cpu_dai, + SND_SOC_DAIFMT_I2S | + SND_SOC_DAIFMT_NB_NF | + SND_SOC_DAIFMT_CBS_CFS); +} + +static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w, + struct snd_kcontrol *k, int event) +{ + gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, + SND_SOC_DAPM_EVENT_ON(event) ? 1 : 0); + gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, + SND_SOC_DAPM_EVENT_ON(event) ? 1 : 0); + + return 0; +} + +/* Machine DAPM widgets */ +static const struct snd_soc_dapm_widget omap3pandora_out_dapm_widgets[] = { + SND_SOC_DAPM_HP("Headphone Jack", omap3pandora_hp_event), +}; + +static const struct snd_soc_dapm_widget omap3pandora_in_dapm_widgets[] = { + SND_SOC_DAPM_MIC("Mic (Internal)", NULL), + SND_SOC_DAPM_MIC("Mic (external)", NULL), + SND_SOC_DAPM_LINE("Line In", NULL), +}; + +static const struct snd_soc_dapm_route omap3pandora_out_map[] = { + {"Headphone Jack", NULL, "OUTL"}, + {"Headphone Jack", NULL, "OUTR"}, +}; + +static const struct snd_soc_dapm_route omap3pandora_in_map[] = { + {"INL", NULL, "Line In"}, + {"INR", NULL, "Line In"}, + {"INL", NULL, "Mic (Internal)"}, + {"INR", NULL, "Mic (external)"}, +}; + +static int omap3pandora_out_init(struct snd_soc_codec *codec) +{ + int ret; + + ret = snd_soc_dapm_new_controls(codec, omap3pandora_out_dapm_widgets, + ARRAY_SIZE(omap3pandora_out_dapm_widgets)); + if (ret < 0) + return ret; + + snd_soc_dapm_add_routes(codec, omap3pandora_out_map, + ARRAY_SIZE(omap3pandora_out_map)); + + return snd_soc_dapm_sync(codec); +} + +static int omap3pandora_in_init(struct snd_soc_codec *codec) +{ + int ret; + + ret = snd_soc_dapm_new_controls(codec, omap3pandora_in_dapm_widgets, + ARRAY_SIZE(omap3pandora_in_dapm_widgets)); + if (ret < 0) + return ret; + + snd_soc_dapm_add_routes(codec, omap3pandora_in_map, + ARRAY_SIZE(omap3pandora_in_map)); + + return snd_soc_dapm_sync(codec); +} + +static struct snd_soc_ops omap3pandora_out_ops = { + .hw_params = omap3pandora_out_hw_params, +}; + +static struct snd_soc_ops omap3pandora_in_ops = { + .hw_params = omap3pandora_in_hw_params, +}; + +/* Digital audio interface glue - connects codec <--> CPU */ +static struct snd_soc_dai_link omap3pandora_dai[] = { + { + .name = "PCM1773", + .stream_name = "HiFi Out", + .cpu_dai = &omap_mcbsp_dai[0], + .codec_dai = &twl4030_dai, + .ops = &omap3pandora_out_ops, + .init = omap3pandora_out_init, + }, { + .name = "TWL4030", + .stream_name = "Line/Mic In", + .cpu_dai = &omap_mcbsp_dai[1], + .codec_dai = &twl4030_dai, + .ops = &omap3pandora_in_ops, + .init = omap3pandora_in_init, + } +}; + +/* Audio machine driver */ +static struct snd_soc_card snd_soc_card_omap3pandora = { + .name = "omap3pandora", + .dai_link = omap3pandora_dai, + .num_links = ARRAY_SIZE(omap3pandora_dai), +}; + +/* Audio subsystem */ +static struct snd_soc_device omap3pandora_snd_data = { + .card = &snd_soc_card_omap3pandora, + .platform = &omap_soc_platform, + .codec_dev = &soc_codec_dev_twl4030, +}; + +static struct platform_device *omap3pandora_snd_device; + +static int __init omap3pandora_soc_init(void) +{ + int ret; + + if (!machine_is_omap3_pandora()) { + pr_debug(PREFIX "Not OMAP3 Pandora\n"); + return -ENODEV; + } + pr_info("OMAP3 Pandora SoC init\n"); + + ret = gpio_request(OMAP3_PANDORA_DAC_POWER_GPIO, "dac_power"); + if (ret) { + pr_err(PREFIX "Failed to get DAC power GPIO\n"); + return ret; + } + + ret = gpio_direction_output(OMAP3_PANDORA_DAC_POWER_GPIO, 0); + if (ret) { + pr_err(PREFIX "Failed to set DAC power GPIO direction\n"); + goto fail0; + } + + ret = gpio_request(OMAP3_PANDORA_AMP_POWER_GPIO, "amp_power"); + if (ret) { + pr_err(PREFIX "Failed to get amp power GPIO\n"); + goto fail0; + } + + ret = gpio_direction_output(OMAP3_PANDORA_AMP_POWER_GPIO, 0); + if (ret) { + pr_err(PREFIX "Failed to set amp power GPIO direction\n"); + goto fail1; + } + + omap3pandora_snd_device = platform_device_alloc("soc-audio", -1); + if (omap3pandora_snd_device == NULL) { + pr_err(PREFIX "Platform device allocation failed\n"); + ret = -ENOMEM; + goto fail1; + } + + platform_set_drvdata(omap3pandora_snd_device, &omap3pandora_snd_data); + omap3pandora_snd_data.dev = &omap3pandora_snd_device->dev; + *(unsigned int *)omap_mcbsp_dai[0].private_data = 1; /* McBSP2 */ + *(unsigned int *)omap_mcbsp_dai[1].private_data = 3; /* McBSP4 */ + + ret = platform_device_add(omap3pandora_snd_device); + if (ret) { + printk(KERN_ERR "Unable to add platform device\n"); + goto fail2; + } + + return 0; + +fail2: + platform_device_put(omap3pandora_snd_device); +fail1: + gpio_free(OMAP3_PANDORA_AMP_POWER_GPIO); +fail0: + gpio_free(OMAP3_PANDORA_DAC_POWER_GPIO); + return ret; +} +module_init(omap3pandora_soc_init); + +static void __exit omap3pandora_soc_exit(void) +{ + platform_device_unregister(omap3pandora_snd_device); + gpio_free(OMAP3_PANDORA_AMP_POWER_GPIO); + gpio_free(OMAP3_PANDORA_DAC_POWER_GPIO); +} +module_exit(omap3pandora_soc_exit); + +MODULE_AUTHOR("Grazvydas Ignotas <notasas(a)gmail.com>"); +MODULE_DESCRIPTION("ALSA SoC OMAP3 Pandora"); +MODULE_LICENSE("GPL"); -- 1.5.4.3

2 1

[alsa-devel] [PATCH 0/4] ASoC updates
by Mark Brown 04 Dec '08

04 Dec '08

The following changes since commit 6f2a974bfc8d3be7a30674c71e2fef003b39a8d2: Daniel Mack (1): ASoC: tlv320aic3x: headset/button press support are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Mark Brown (4): ASoC: Push debugfs files out of the snd_soc_device structure ASoC: Remove device from platform suspend and resume operations ASoC: Remove platform device from DAI suspend and resume operations ASoC: Remove obsolete declaration of struct snd_soc_clock_info include/sound/soc-dai.h | 6 +-- include/sound/soc.h | 15 +++---- sound/soc/atmel/atmel-pcm.c | 6 +-- sound/soc/atmel/atmel_ssc_dai.c | 6 +-- sound/soc/au1x/psc-ac97.c | 6 +-- sound/soc/au1x/psc-i2s.c | 6 +-- sound/soc/blackfin/bf5xx-ac97.c | 6 +-- sound/soc/blackfin/bf5xx-i2s.c | 6 +-- sound/soc/pxa/pxa-ssp.c | 6 +-- sound/soc/pxa/pxa2xx-ac97.c | 6 +-- sound/soc/pxa/pxa2xx-i2s.c | 6 +-- sound/soc/s3c24xx/s3c2412-i2s.c | 16 +++---- sound/soc/s3c24xx/s3c24xx-i2s.c | 6 +-- sound/soc/soc-core.c | 85 +++++++++++++++++++++----------------- 14 files changed, 83 insertions(+), 99 deletions(-)

2 5

[alsa-devel] question for the all-knowing - firewire audio hardware
by S. Aguinaga 04 Dec '08

04 Dec '08

Hello! I have a couple of machines connected to FireWire Motu devices (running OS X). We want to develop some new apps on linux and the only info I've found is: http://www.olafchrist.de/ieee1394/index.php, but its rather old and single device specific. Are there any other solutions available? Or are there other devices (firewire) that are supported by the ALSA project? Any info on this will be great!! // Sal

2 3

Re: [alsa-devel] [PATCH - patch_nvhdmi.c 1/1] Resubmiited with full name. Added id for MCP67. Please note that I'm not really sure what I'm doing but this works for me :-)
by Takashi Iwai 04 Dec '08

04 Dec '08

At Wed, 3 Dec 2008 21:11:22 +0000, Scott Waye wrote: > > > Signed-off-by: Scott Waye <scott(a)waye.co.uk> Applied now. Thanks. Takashi > > diff --git a/pci/hda/patch_nvhdmi.c b/pci/hda/patch_nvhdmi.c > index 1360d54..b78c8f2 100644 > --- a/pci/hda/patch_nvhdmi.c > +++ b/pci/hda/patch_nvhdmi.c > @@ -162,5 +162,6 @@ static int patch_nvhdmi(struct hda_codec *codec) > struct hda_codec_preset snd_hda_preset_nvhdmi[] = { > { .id = 0x10de0002, .name = "NVIDIA MCP78 HDMI", .patch = patch_nvhdmi }, > { .id = 0x10de0007, .name = "NVIDIA MCP7A HDMI", .patch = patch_nvhdmi }, > + { .id = 0x10de0067, .name = "NVIDIA MCP67 HDMI", .patch = patch_nvhdmi }, > {} /* terminator */ > }; > -- > 1.5.6 >

1 0

[alsa-devel] [PATCH] ALSA: Reduce stall detection timeout in riptide.c
by nokos＠gmx.net 03 Dec '08

03 Dec '08

Reduce the command timeout to 0.5sec. Should be enough to allow a working command interface but removes a RCU stall and slow resume on some revisions where the AC97 revision detection stalls in resume. Signed-off-by: Peter Gruber <nokos(a)gmx.net> --- diff --git a/sound/pci/riptide/riptide.c b/sound/pci/riptide/riptide.c index e9f0706..d0ccfc6 100644 --- a/sound/pci/riptide/riptide.c +++ b/sound/pci/riptide/riptide.c @@ -172,7 +172,7 @@ #define MAX_WRITE_RETRY 10 /* cmd interface limits */ #define MAX_ERROR_COUNT 10 -#define CMDIF_TIMEOUT 500000 +#define CMDIF_TIMEOUT 50000 #define RESET_TRIES 5 #define READ_PORT_ULONG(p) inl((unsigned long)&(p))

2 1

[alsa-devel] [PATCH 0/5] ASoC updates
by Mark Brown 03 Dec '08

03 Dec '08

The following changes since commit 0ecfe7987855d21c2a89ffe003ddf0ee11b42d47: Mark Brown (1): ASoC: Don't free static data in WM9713 are available in the git repository at: git://opensource.wolfsonmicro.com/linux-2.6-asoc for-tiwai Daniel Mack (1): ASoC: tlv320aic3x: headset/button press support Grazvydas Ignotas (1): ASoC: TWL4030: Add input selection and gain controls Mark Brown (3): ASoC: Push workqueue data into snd_soc_card ASoC: Push platform registration down into the card ASoC: Fix WM8903 right mixer bypass path include/sound/soc.h | 9 +- sound/soc/atmel/playpaq_wm8510.c | 2 +- sound/soc/atmel/sam9g20_wm8731.c | 2 +- sound/soc/blackfin/bf5xx-ad1980.c | 2 +- sound/soc/blackfin/bf5xx-ad73311.c | 2 +- sound/soc/blackfin/bf5xx-ssm2602.c | 2 +- sound/soc/codecs/tlv320aic3x.c | 31 ++++++- sound/soc/codecs/tlv320aic3x.h | 46 +++++++++- sound/soc/codecs/twl4030.c | 177 +++++++++++++++++++++++++++++++++- sound/soc/codecs/twl4030.h | 16 +++ sound/soc/codecs/wm8903.c | 8 +- sound/soc/davinci/davinci-evm.c | 2 +- sound/soc/davinci/davinci-i2s.c | 2 +- sound/soc/davinci/davinci-sffsdr.c | 2 +- sound/soc/fsl/mpc8610_hpcd.c | 2 +- sound/soc/fsl/soc-of-simple.c | 2 +- sound/soc/omap/n810.c | 2 +- sound/soc/omap/omap2evm.c | 2 +- sound/soc/omap/omap3beagle.c | 2 +- sound/soc/omap/osk5912.c | 2 +- sound/soc/omap/overo.c | 2 +- sound/soc/omap/sdp3430.c | 2 +- sound/soc/pxa/corgi.c | 2 +- sound/soc/pxa/e800_wm9712.c | 2 +- sound/soc/pxa/em-x270.c | 2 +- sound/soc/pxa/palm27x.c | 2 +- sound/soc/pxa/poodle.c | 2 +- sound/soc/pxa/spitz.c | 2 +- sound/soc/pxa/tosa.c | 2 +- sound/soc/pxa/zylonite.c | 2 +- sound/soc/s3c24xx/ln2440sbc_alc650.c | 2 +- sound/soc/s3c24xx/neo1973_wm8753.c | 2 +- sound/soc/s3c24xx/s3c24xx_uda134x.c | 2 +- sound/soc/s3c24xx/smdk2443_wm9710.c | 2 +- sound/soc/sh/sh7760-ac97.c | 2 +- sound/soc/soc-core.c | 77 +++++++++------ 36 files changed, 348 insertions(+), 74 deletions(-)

3 8

Re: [alsa-devel] Gateway MX3042 Laptop. No sound
by Takashi Iwai 03 Dec '08

03 Dec '08

At Wed, 3 Dec 2008 06:39:24 -0600, Michael Satterwhite wrote: > > On Wednesday 03 December 2008 01:02:07 am you wrote: > > At Wed, 3 Dec 2008 00:12:54 +0100, > > > > Nigel Henry wrote: > > > I'm trying to help someone on the Kubuntu list, using Hardy Heron 8.04, > > > which uses alsa driver 1.0.16. This is with a Gateway MX3042 Laptop, but > > > no sounds. > > > > > > The output from running alsa-info.sh is below. > > > > Try my sound.git tree and sound-unstable.git tree. > > > > git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6.git > > > > git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-unstable-2.6.git > > > > The snapshot tarballs are found in > > ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/ > > > I'll be happy to (and *DO* thank you for your help!). I'm not sure what this > means, however. Are these drivers, a kernel ... how do I try them? It's a linux kernel git tree. Ask other Ubuntu guys how to build your kernel. Or, use the alsa-driver snapshot tarball in the ftp directory above. You can install it just like the normal alsa-driver release tarball. Takashi

1 0

[alsa-devel] [PATCH] tlv320aic3x: headset/button press support
by Daniel Mack 03 Dec '08

03 Dec '08

Hi, This patch adds a function to setup headset and button press detection modes. Also, aic3x_headset_detected() returns the value from the current state's register rather than the interrupt cause flag so it works without an interrupt as well. Best regards, Daniel

3 5

[alsa-devel] [PATCH] ASoC: TWL4030: Add input selection and gain controls
by Grazvydas Ignotas 03 Dec '08

03 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 input source controls. It modifies default register values to have no inputs selected and ADCs disabled. When some input is selected, control handlers enable apropriate input amplifier and ADC. If a microphone is selected, bias power is automatically enabled. When some input is deselected, unused chip parts are disabled. Microphone and line input recording tested on OMAP3 pandora board. Signed-off-by: Grazvydas Ignotas <notasas(a)gmail.com> --- Compared to my previous patch, I've changed "Microphone Boost" name to "Input Boost Volume", as it affects line inputs too. I thought that hiding this control for line-in was not a good idea, as it might still be useful for weak sources. "Capture" was also changed to "Input", as inputs can be also routed to outputs, not only recorded. sound/soc/codecs/twl4030.c | 177 +++++++++++++++++++++++++++++++++++++++++++- sound/soc/codecs/twl4030.h | 16 ++++ 2 files changed, 190 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index ffd5120..3c9fdf2 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -46,9 +46,9 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0xc3, /* REG_OPTION (0x2) */ 0x00, /* REG_UNKNOWN (0x3) */ 0x00, /* REG_MICBIAS_CTL (0x4) */ - 0x24, /* REG_ANAMICL (0x5) */ - 0x04, /* REG_ANAMICR (0x6) */ - 0x0a, /* REG_AVADC_CTL (0x7) */ + 0x20, /* REG_ANAMICL (0x5) */ + 0x00, /* REG_ANAMICR (0x6) */ + 0x00, /* REG_AVADC_CTL (0x7) */ 0x00, /* REG_ADCMICSEL (0x8) */ 0x00, /* REG_DIGMIXING (0x9) */ 0x0c, /* REG_ATXL1PGA (0xA) */ @@ -347,6 +347,162 @@ 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) { + result++; + 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, avadc_ctl; + + 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); + avadc_ctl = twl4030_read_reg_cache(codec, TWL4030_REG_AVADC_CTL); + + switch (value) { + case 1: + anamicl |= TWL4030_MAINMIC_EN; + micbias |= TWL4030_MICBIAS1_EN; + break; + case 2: + anamicl |= TWL4030_HSMIC_EN; + micbias |= TWL4030_HSMICBIAS_EN; + break; + case 3: + anamicl |= TWL4030_AUXL_EN; + break; + case 4: + anamicl |= TWL4030_CKMIC_EN; + break; + default: + break; + } + + /* If some input is selected, enable amp and ADC */ + if (value != 0) { + anamicl |= TWL4030_MICAMPL_EN; + avadc_ctl |= TWL4030_ADCL_EN; + } else { + anamicl &= ~TWL4030_MICAMPL_EN; + avadc_ctl &= ~TWL4030_ADCL_EN; + } + + twl4030_write(codec, TWL4030_REG_ANAMICL, anamicl); + twl4030_write(codec, TWL4030_REG_MICBIAS_CTL, micbias); + twl4030_write(codec, TWL4030_REG_AVADC_CTL, avadc_ctl); + + 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); + int value = 0; + + reg &= TWL4030_SUBMIC_EN|TWL4030_AUXR_EN; + switch (reg) { + case TWL4030_SUBMIC_EN: + value = 1; + break; + case TWL4030_AUXR_EN: + value = 2; + break; + default: + break; + } + + ucontrol->value.integer.value[0] = value; + 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, avadc_ctl; + + 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; + avadc_ctl = twl4030_read_reg_cache(codec, TWL4030_REG_AVADC_CTL); + + switch (value) { + case 1: + anamicr |= TWL4030_SUBMIC_EN; + micbias |= TWL4030_MICBIAS2_EN; + break; + case 2: + anamicr |= TWL4030_AUXR_EN; + break; + default: + break; + } + + if (value != 0) { + anamicr |= TWL4030_MICAMPR_EN; + avadc_ctl |= TWL4030_ADCR_EN; + } else { + anamicr &= ~TWL4030_MICAMPR_EN; + avadc_ctl &= ~TWL4030_ADCR_EN; + } + + twl4030_write(codec, TWL4030_REG_ANAMICR, anamicr); + twl4030_write(codec, TWL4030_REG_MICBIAS_CTL, micbias); + twl4030_write(codec, TWL4030_REG_AVADC_CTL, avadc_ctl); + + return 1; +} + +static const char *twl4030_left_in_sel[] = { + "None", + "Main Mic", + "Headset Mic", + "Line In", + "Carkit Mic", +}; + +static const char *twl4030_right_in_sel[] = { + "None", + "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 +534,12 @@ static DECLARE_TLV_DB_SCALE(output_tvl, -1200, 600, 1); */ static DECLARE_TLV_DB_SCALE(digital_capture_tlv, 0, 100, 0); +/* + * Gain control for input amplifiers + * 0 dB to 30 dB in 6 dB steps + */ +static DECLARE_TLV_DB_SCALE(input_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 +582,15 @@ 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("Input Boost Volume", TWL4030_REG_ANAMIC_GAIN, + 0, 3, 5, 0, input_gain_tlv), + + /* Input source controls */ + SOC_ENUM_EXT("Left Input Source", twl4030_left_input_mux, + twl4030_get_left_input, twl4030_put_left_input), + SOC_ENUM_EXT("Right Input 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..a2065d4 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -113,7 +113,16 @@ #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 #define TWL4030_OFFSET_CNCL_SEL_ARX1 0x00 @@ -127,10 +136,17 @@ #define TWL4030_MAINMIC_EN 0x01 /* ANAMICR (0x06) Fields */ + #define TWL4030_MICAMPR_EN 0x10 #define TWL4030_AUXR_EN 0x04 #define TWL4030_SUBMIC_EN 0x01 +/* AVADC_CTL (0x07) Fields */ + +#define TWL4030_ADCL_EN 0x08 +#define TWL4030_AVADC_CLK_PRIORITY 0x04 +#define TWL4030_ADCR_EN 0x02 + /* AUDIO_IF (0x0E) Fields */ #define TWL4030_AIF_SLAVE_EN 0x80 -- 1.5.4.3

3 3

[alsa-devel] Gateway MX3042 Laptop. No sound
by Nigel Henry 03 Dec '08

03 Dec '08

I'm trying to help someone on the Kubuntu list, using Hardy Heron 8.04, which uses alsa driver 1.0.16. This is with a Gateway MX3042 Laptop, but no sounds. The output from running alsa-info.sh is below. !!################################ !!ALSA Information Script v 0.4.52 !!################################ !!Script ran on: Tue Dec 2 16:30:06 CST 2008 !!Linux Distribution !!------------------ Ubuntu 8.04.1 \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 8.04.1" !!Kernel Information !!------------------ Kernel release: 2.6.24-22-generic Operating System: GNU/Linux Architecture: i686 Processor: unknown SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.16 Library version: Utilities version: 1.0.15 !!Loaded ALSA modules !!------------------- snd_intel8x0 !!Soundcards recognised by ALSA !!----------------------------- 0 [I82801DBICH4 ]: ICH4 - Intel 82801DB-ICH4 Intel 82801DB-ICH4 with AD1981B at irq 17 !!PCI Soundcards installed in the system !!-------------------------------------- 00:1f.5 Multimedia audio controller: Intel Corporation 82801DB/DBL/DBM (ICH4/ICH4-L/ICH4-M) AC'97 Audio Controller (rev 03) !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 00:1f.5 0401: 8086:24c5 (rev 03) Subsystem: 161f:203e !!Modprobe options (Sound related) !!-------------------------------- snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 snd-usb-audio: index=-2 snd-usb-usx2y: index=-2 snd-usb-caiaq: index=-2 snd-cmipci: mpu_port=0x330 fm_port=0x388 !!Loaded sound module options !!-------------------------- !!Module: snd_intel8x0 ac97_clock : 0 ac97_quirk : <NULL> buggy_irq : N buggy_semaphore : N enable : N id : <NULL> index : -1 joystick : 0 spdif_aclink : 0 xbox : N !!AC97 Codec information !!--------------------------- --startcollapse-- 0-0/0: Analog Devices AD1981B PCI Subsys Vendor: 0x161f PCI Subsys Device: 0x203e Capabilities : -headphone out- DAC resolution : 20-bit ADC resolution : 16-bit 3D enhancement : No 3D Stereo Enhancement Current setup Mic gain : 20dB [ 20dB] POP path : pre 3D Sim. stereo : off 3D enhancement : off Loudness : off Mono output : MIX Mic select : Mic1 ADC/DAC loopback : off Extended ID : codec=0 rev=1 AMAP DSA=0 SPDIF VRA Extended status : SPCV SPDIF=10/11 SPDIF VRA PCM front DAC : 48000Hz PCM ADC : 48000Hz SPDIF Control : Consumer PCM Category=0x2 Generation=1 Rate=48kHz AD18XX configuration Unchained : 0x1000,0x0000,0x0000 Chained : 0x0000,0x0000,0x0000 0:00 = 0090 0:02 = 0000 0:04 = 0000 0:06 = 0000 0:08 = 0000 0:0a = 0000 0:0c = 001f 0:0e = 805f 0:10 = 0808 0:12 = 0606 0:14 = 0000 0:16 = 1f1f 0:18 = 0000 0:1a = 0505 0:1c = 0000 0:1e = 0000 0:20 = 0000 0:22 = 0000 0:24 = 0000 0:26 = 800f 0:28 = 0605 0:2a = 0435 0:2c = bb80 0:2e = 0000 0:30 = 0000 0:32 = bb80 0:34 = 0000 0:36 = 0000 0:38 = 0000 0:3a = 2824 0:3c = 0000 0:3e = 0000 0:40 = 0000 0:42 = 0000 0:44 = 0000 0:46 = 0000 0:48 = 0000 0:4a = 0000 0:4c = 0000 0:4e = 0000 0:50 = 0000 0:52 = 0000 0:54 = 0000 0:56 = 0000 0:58 = 0000 0:5a = 0000 0:5c = 4000 0:5e = 0000 0:60 = 8080 0:62 = 0000 0:64 = 8000 0:66 = 0000 0:68 = 0000 0:6a = 0000 0:6c = 0000 0:6e = 0000 0:70 = 0000 0:72 = 1800 0:74 = 1001 0:76 = 2050 0:78 = 0000 0:7a = 0000 0:7c = 4144 0:7e = 5374 --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw---- 1 root audio 116, 0 2008-12-02 11:09 /dev/snd/controlC0 crw-rw---- 1 root audio 116, 24 2008-12-02 11:09 /dev/snd/pcmC0D0c crw-rw---- 1 root audio 116, 16 2008-12-02 14:27 /dev/snd/pcmC0D0p crw-rw---- 1 root audio 116, 25 2008-12-02 11:09 /dev/snd/pcmC0D1c crw-rw---- 1 root audio 116, 26 2008-12-02 11:09 /dev/snd/pcmC0D2c crw-rw---- 1 root audio 116, 27 2008-12-02 11:09 /dev/snd/pcmC0D3c crw-rw---- 1 root audio 116, 20 2008-12-02 11:09 /dev/snd/pcmC0D4p crw-rw---- 1 root audio 116, 1 2008-12-02 11:09 /dev/snd/seq crw-rw---- 1 root audio 116, 33 2008-12-02 11:09 /dev/snd/timer !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 0: Intel ICH [Intel 82801DB-ICH4] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 4: Intel ICH - IEC958 [Intel 82801DB-ICH4 - IEC958] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 0: Intel ICH [Intel 82801DB-ICH4] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 1: Intel ICH - MIC ADC [Intel 82801DB-ICH4 - MIC ADC] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 2: Intel ICH - MIC2 ADC [Intel 82801DB-ICH4 - MIC2 ADC] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: I82801DBICH4 [Intel 82801DB-ICH4], device 3: Intel ICH - ADC2 [Intel 82801DB-ICH4 - ADC2] Subdevices: 1/1 Subdevice #0: subdevice #0 !!Amixer output !!------------- !!-------Mixer controls for card 0 [I82801DBICH4] Card hw:0 'I82801DBICH4'/'Intel 82801DB-ICH4 with AD1981B at irq 17' Mixer name : 'Analog Devices AD1981B' Components : 'AC97a:41445374' Controls : 34 Simple ctrls : 23 Simple mixer control 'Master',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [0.00dB] [on] Front Right: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'Master Mono',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 31 Mono: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'Headphone',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [0.00dB] [on] Front Right: Playback 31 [100%] [0.00dB] [on] Simple mixer control 'Headphone Jack Sense',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'PCM',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 31 [100%] [12.00dB] [on] Front Right: Playback 31 [100%] [12.00dB] [on] Simple mixer control 'Line',0 Capabilities: pvolume pswitch cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 23 [74%] [0.00dB] [on] Capture [off] Front Right: Playback 23 [74%] [0.00dB] [on] Capture [off] Simple mixer control 'Line Jack Sense',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'CD',0 Capabilities: pvolume pswitch cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 25 [81%] [3.00dB] [on] Capture [off] Front Right: Playback 25 [81%] [3.00dB] [on] Capture [off] Simple mixer control 'Mic',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Mono Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Playback 0 [0%] [-34.50dB] [off] Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'Mic Boost ( 20dB)',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'Mic Select',0 Capabilities: enum Items: 'Mic1' 'Mic2' Item0: 'Mic1' Simple mixer control 'Video',0 Capabilities: cswitch cswitch-exclusive Capture exclusive group: 0 Capture channels: Front Left - Front Right Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'Phone',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Mono Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Playback 0 [0%] [-34.50dB] [on] Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'IEC958 Playback AC97-SPSA',0 Capabilities: volume volume-joined Playback channels: Mono Capture channels: Mono Limits: 0 - 3 Mono: 3 [100%] Simple mixer control 'IEC958 Playback Source',0 Capabilities: enum Items: 'AC-Link' 'A/D Converter' Item0: 'AC-Link' Simple mixer control 'Aux',0 Capabilities: pvolume pswitch cswitch cswitch-exclusive Capture exclusive group: 0 Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 31 Front Left: Playback 0 [0%] [-34.50dB] [on] Capture [off] Front Right: Playback 0 [0%] [-34.50dB] [on] Capture [off] Simple mixer control 'Mono Output Select',0 Capabilities: enum Items: 'Mix' 'Mic' Item0: 'Mix' Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 15 Front Left: Capture 0 [0%] [0.00dB] [on] Front Right: Capture 0 [0%] [0.00dB] [on] Simple mixer control 'Mix',0 Capabilities: cswitch cswitch-exclusive Capture exclusive group: 0 Capture channels: Front Left - Front Right Front Left: Capture [on] Front Right: Capture [on] Simple mixer control 'Mix Mono',0 Capabilities: cswitch cswitch-exclusive Capture exclusive group: 0 Capture channels: Front Left - Front Right Front Left: Capture [off] Front Right: Capture [off] Simple mixer control 'External Amplifier',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [off] Simple mixer control 'Stereo Mic',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] !!Alsactl output !!------------- --startcollapse-- state.I82801DBICH4 { control.1 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Master Playback Switch' value.0 true value.1 true } control.2 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' iface MIXER name 'Master Playback Volume' value.0 31 value.1 31 } control.3 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Headphone Playback Switch' value.0 true value.1 true } control.4 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' iface MIXER name 'Headphone Playback Volume' value.0 31 value.1 31 } control.5 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Master Mono Playback Switch' value true } control.6 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 31' iface MIXER name 'Master Mono Playback Volume' value 31 } control.7 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Phone Playback Switch' value true } control.8 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 31' iface MIXER name 'Phone Playback Volume' value 0 } control.9 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Mic Playback Switch' value false } control.10 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 31' iface MIXER name 'Mic Playback Volume' value 0 } control.11 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Mic Boost ( 20dB)' value true } control.12 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Line Playback Switch' value.0 true value.1 true } control.13 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' iface MIXER name 'Line Playback Volume' value.0 23 value.1 23 } control.14 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'CD Playback Switch' value.0 true value.1 true } control.15 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' iface MIXER name 'CD Playback Volume' value.0 25 value.1 25 } control.16 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Aux Playback Switch' value.0 true value.1 true } control.17 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' iface MIXER name 'Aux Playback Volume' value.0 0 value.1 0 } control.18 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'PCM Playback Switch' value.0 true value.1 true } control.19 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' iface MIXER name 'PCM Playback Volume' value.0 31 value.1 31 } control.20 { comment.access 'read write' comment.type ENUMERATED comment.count 2 comment.item.0 Mic comment.item.1 CD comment.item.2 Video comment.item.3 Aux comment.item.4 Line comment.item.5 Mix comment.item.6 'Mix Mono' comment.item.7 Phone iface MIXER name 'Capture Source' value.0 Mix value.1 Mix } control.21 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Capture Switch' value.0 true value.1 true } control.22 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 15' iface MIXER name 'Capture Volume' value.0 0 value.1 0 } control.23 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mix comment.item.1 Mic iface MIXER name 'Mono Output Select' value Mix } control.24 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mic1 comment.item.1 Mic2 iface MIXER name 'Mic Select' value Mic1 } control.25 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Con Mask' value '0fff000f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.26 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Pro Mask' value cf00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000 } control.27 { comment.access 'read write' comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Default' value '0082000200000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.28 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'IEC958 Playback Switch' value true } control.29 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 3' iface MIXER name 'IEC958 Playback AC97-SPSA' value 3 } control.30 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 AC-Link comment.item.1 'A/D Converter' iface MIXER name 'IEC958 Playback Source' value AC-Link } control.31 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Stereo Mic' value true } control.32 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Headphone Jack Sense' value true } control.33 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Line Jack Sense' value true } control.34 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'External Amplifier' value false } } --endcollapse-- !!All Loaded Modules !!------------------ Module nls_iso8859_1 nls_cp437 vfat fat usb_storage libusual ipv6 af_packet rfcomm l2cap bluetooth rfkill_input i915 drm ppdev cpufreq_userspace cpufreq_stats cpufreq_powersave cpufreq_conservative cpufreq_ondemand freq_table dock container sbs sbshc iptable_filter ip_tables x_tables aes_i586 dm_crypt dm_mod sbp2 parport_pc lp parport joydev tifm_sd mmc_core pcmcia arc4 ecb blkcipher b43 rfkill mac80211 cfg80211 led_class input_polldev snd_intel8x0 snd_ac97_codec ac97_bus snd_pcm_oss snd_mixer_oss snd_pcm snd_seq_dummy video output evdev psmouse tifm_7xx1 serio_raw snd_seq_oss tifm_core yenta_socket rsrc_nonstatic snd_seq_midi pcmcia_core snd_rawmidi snd_seq_midi_event snd_seq snd_timer snd_seq_device battery ac button pcspkr snd soundcore iTCO_wdt iTCO_vendor_support snd_page_alloc intel_agp shpchp pci_hotplug agpgart ext3 jbd mbcache sg sr_mod cdrom sd_mod ata_generic usbhid hid b44 ata_piix ohci1394 pata_acpi ssb mii ieee1394 libata scsi_mod ehci_hcd uhci_hcd usbcore thermal processor fan fbcon tileblit font bitblit softcursor fuse Any help in getting Michael's sounds working, greatly appreciated. Nigel.

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[alsa-devel] [PATCH - patch_nvhdmi.c 1/1] Added id for MCP67. Please note that I'm not really sure what I'm doing but this works for me :-)
by scott＠waye.co.uk 03 Dec '08

03 Dec '08

From: root <scott(a)waye.co.uk> Signed-off-by: root <scott(a)waye.co.uk> diff --git a/pci/hda/patch_nvhdmi.c b/pci/hda/patch_nvhdmi.c index 1360d54..b78c8f2 100644 --- a/pci/hda/patch_nvhdmi.c +++ b/pci/hda/patch_nvhdmi.c @@ -162,5 +162,6 @@ static int patch_nvhdmi(struct hda_codec *codec) struct hda_codec_preset snd_hda_preset_nvhdmi[] = { { .id = 0x10de0002, .name = "NVIDIA MCP78 HDMI", .patch = patch_nvhdmi }, { .id = 0x10de0007, .name = "NVIDIA MCP7A HDMI", .patch = patch_nvhdmi }, + { .id = 0x10de0067, .name = "NVIDIA MCP67 HDMI", .patch = patch_nvhdmi }, {} /* terminator */ }; -- 1.5.6

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[PATCH - snd-intel-hda 1/1] Add partial 8930G support
by Hector Martin 03 Dec '08

03 Dec '08

Signed-off-by: Hector Martin <hector(a)marcansoft.com> diff --git a/pci/hda/patch_realtek.c b/pci/hda/patch_realtek.c index 4afa0c9..6d6e06a 100644 --- a/pci/hda/patch_realtek.c +++ b/pci/hda/patch_realtek.c @@ -1302,7 +1302,7 @@ static struct hda_verb alc888_acer_aspire_4930g_verbs[] = { {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)}, /* Enable unsolicited event for HP jack */ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, -/* Connect Internal HP to front */ +/* Connect Front speakers to front */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -1310,6 +1310,16 @@ static struct hda_verb alc888_acer_aspire_4930g_verbs[] = { {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, +/* Connect Rear speakers to rear */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1b, AC_VERB_SET_CONNECT_SEL, 0x01}, +/* Connect CLFE speakers to CLFE */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x16, AC_VERB_SET_CONNECT_SEL, 0x02}, +/* Set EAPD to enable speakers */ + {0x15, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, { } }; @@ -1365,8 +1375,15 @@ static void alc888_acer_aspire_4930g_automute(struct hda_codec *codec) present = snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; bits = present ? HDA_AMP_MUTE : 0; + /* Front */ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, HDA_AMP_MUTE, bits); + /* CLFE */ + snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); + /* Rear */ + snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); } static void alc888_acer_aspire_4930g_unsol_event(struct hda_codec *codec, -- 1.5.6.4 --------------060204020609000202040409--

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[PATCH - snd-intel-hda 1/1] Add partial 8930G support
by Hector Martin 03 Dec '08

03 Dec '08

Signed-off-by: Hector Martin <hector(a)marcansoft.com> diff --git a/pci/hda/patch_realtek.c b/pci/hda/patch_realtek.c index 4afa0c9..6d6e06a 100644 --- a/pci/hda/patch_realtek.c +++ b/pci/hda/patch_realtek.c @@ -1302,7 +1302,7 @@ static struct hda_verb alc888_acer_aspire_4930g_verbs[] = { {0x22, AC_VERB_SET_AMP_GAIN_MUTE, AMP_IN_MUTE(0xb)}, /* Enable unsolicited event for HP jack */ {0x15, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, -/* Connect Internal HP to front */ +/* Connect Front speakers to front */ {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x14, AC_VERB_SET_CONNECT_SEL, 0x00}, @@ -1310,6 +1310,16 @@ static struct hda_verb alc888_acer_aspire_4930g_verbs[] = { {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x15, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x15, AC_VERB_SET_CONNECT_SEL, 0x00}, +/* Connect Rear speakers to rear */ + {0x1b, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x1b, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x1b, AC_VERB_SET_CONNECT_SEL, 0x01}, +/* Connect CLFE speakers to CLFE */ + {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {0x16, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, + {0x16, AC_VERB_SET_CONNECT_SEL, 0x02}, +/* Set EAPD to enable speakers */ + {0x15, AC_VERB_SET_EAPD_BTLENABLE, 0x02}, { } }; @@ -1365,8 +1375,15 @@ static void alc888_acer_aspire_4930g_automute(struct hda_codec *codec) present = snd_hda_codec_read(codec, 0x15, 0, AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; bits = present ? HDA_AMP_MUTE : 0; + /* Front */ snd_hda_codec_amp_stereo(codec, 0x14, HDA_OUTPUT, 0, HDA_AMP_MUTE, bits); + /* CLFE */ + snd_hda_codec_amp_stereo(codec, 0x16, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); + /* Rear */ + snd_hda_codec_amp_stereo(codec, 0x1b, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); } static void alc888_acer_aspire_4930g_unsol_event(struct hda_codec *codec, -- 1.5.6.4 --------------070404090902050101040901--

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[alsa-devel] alsa-driver-1.0.18rc3 fails to compile against linux-2.6.27.7/arch/powerpc
by kordex - 02 Dec '08

02 Dec '08

make -C /lib/modules/2.6.27.7/sourceSUBDIRS=/root/alsa/alsa-driver-1.0.18rc3 CPP="gcc -E" CC="gcc" modules make[1]: Siirrytään hakemistoon "/usr/src/linux-2.6.27.7" CC [M] /root/alsa/alsa-driver-1.0.18rc3/acore/hwdep.o In file included from /usr/src/linux-2.6.27.7/arch/powerpc/include/asm/system.h:528, from include/linux/list.h:7, from include/linux/module.h:9, from /root/alsa/alsa-driver-1.0.18rc3/include/adriver.h:26, from /root/alsa/alsa-driver-1.0.18rc3/acore/hwdep.c:1: include/asm-generic/cmpxchg-local.h: In function '__cmpxchg_local_generic': include/asm-generic/cmpxchg-local.h:23: error: implicit declaration of function 'local_irq_save' include/asm-generic/cmpxchg-local.h:44: error: implicit declaration of function 'local_irq_restore' make[3]: *** [/root/alsa/alsa-driver-1.0.18rc3/acore/hwdep.o] Virhe 1 make[2]: *** [/root/alsa/alsa-driver-1.0.18rc3/acore] Virhe 2 make[1]: *** [_module_/root/alsa/alsa-driver-1.0.18rc3] Virhe 2 make[1]: Poistutaan hakemistosta "/usr/src/linux-2.6.27.7" make: *** [compile] Virhe 2 root@powerline:/root/alsa/alsa-driver-1.0.18rc3# root@powerline:/root/alsa/alsa-driver-1.0.18rc3# I used this configure line (compiling for emu 1212m): ./configure --with-cards=emu10k1 --with-sequencer=yes --with-oss=yes Gcc version: root@powerline:/root/alsa/alsa-driver-1.0.18rc3# gcc -v Using built-in specs. Target: powerpc-linux-gnu Configured with: ../src/configure -v --with-pkgversion='Debian 4.3.2-1' --with-bugurl=file:///usr/share/doc/gcc-4.3/README.Bugs --enable-languages=c,c++,fortran,objc,obj-c++ --prefix=/usr --enable-shared --with-system-zlib --libexecdir=/usr/lib --without-included-gettext --enable-threads=posix --enable-nls --with-gxx-include-dir=/usr/include/c++/4.3 --program-suffix=-4.3 --enable-clocale=gnu --enable-libstdcxx-debug --enable-objc-gc --enable-mpfr --disable-softfloat --enable-secureplt --enable-targets=powerpc-linux,powerpc64-linux --with-cpu=default32 --with-long-double-128 --enable-checking=release --build=powerpc-linux-gnu --host=powerpc-linux-gnu --target=powerpc-linux-gnu Thread model: posix gcc version 4.3.2 (Debian 4.3.2-1)

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[alsa-devel] Where is xfi driver now?
by Nickolay Hodyunya 02 Dec '08

02 Dec '08

Hello, I have updated to the latest git tree of sound-usntable kernel 1 day ago and now I can not find xfi drivers. Where are they go? -- Regards, Nickolay Hodyunya.

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