April 2009 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] module snd-atiixp issue with suspend/hibernate/resume
by Ryan Dunn 05 May '09

05 May '09

I have a Compaq V2000 laptop with the ATI IXP chipset and make use of the snd-atiixp module. The laptop needs to have the ac97_quirk option set to 7 to enable the mute LED. After setting this option in the /etc/modprobe.d/options.conf file, the mute LED works after a reboot. However, after a resume from suspend or hibernate, the mute LED does not work. The alsa-info.sh script reports that the quirk is still set with 7. At this point a reboot will NOT fix the problem. The only way to fix it is to remove and reinsert the module with modprobe. This is on a fresh Ubuntu 9.04 install. Any ideas? I wasn't sure if this should be reported here or on the kernel list. I saw a similar issue with the ac97_quirk on the kernel list and they were referred here. I'm a software developer, so I'd be willing to try ideas/possible solutions if you have them. Thanks, Ryan

3 9

Re: [alsa-devel] realtek ALC272 support
by Andres Salomon 04 May '09

04 May '09

On Tue, 28 Apr 2009 15:41:08 -0400 Andres Salomon <dilinger(a)queued.net> wrote: > On Mon, 27 Apr 2009 18:08:21 +0200 > Takashi Iwai <tiwai(a)suse.de> wrote: > [...] > > > > > > > > At Fri, 24 Apr 2009 15:24:15 -0400, > > > > > > > > Andres Salomon wrote: > > > > > > > > > > > > > > > > > > Hi Kailang, > > > > > > > > > > > > > > > > > > I noticed that your name was on the ALC272 support > > > > > > > > > patch for ALSA intel-hda. This patch basically sets > > > > > > > > > ALC272 to use (mostly) the same code as ALC662. I > > > > > > > > > have two machines that have ALC272, and both of them > > > > > > > > > need verb tables in order to function. I'm wondering > > > > > > > > > if ALC662 should really be used.. > > > > > > > > > > > > > > > > > > Here's one - > > > > > > > > > > > > > > > > > > http://kernel.ubuntu.com/git?p=ubuntu/ubuntu-hardy-lum.git;a=commitdiff;h=7… > > > > > > > > > > > > > > > > > > This isn't the final version of the patch (there are > > > > > > > > > further commits I made in order to support headphone > > > > > > > > > mic stuff), but it gives you an idea of the codec > > > > > > > > > values. The other is: > > > > > > > > > > > > > > > > > > http://kernel.ubuntu.com/git?p=ubuntu/ubuntu-hardy-lum.git;a=commitdiff;h=7… > > > > > > > > > > > > > > > > > > All of these leave me wonder if there's a specific > > > > > > > > > patch_alc272 function that could be written to rid > > > > > > > > > ourselves of these specific quirks. Are there > > > > > > > > > machines with ALC272 that are functional with the > > > > > > > > > current ALSA code? > > > > > > > > [...] > > > > Could you try sound-unstable tree a bit later again? > > I found a bug in my patch, and fixed and updated GIT tree now. > > At least, the headphone plugging should work now. > > > > The mic auto-detection still doesn't work with model=auto, though. > > So, I'm going to take your patch anyway later. But I just wanted > > to be sure that the current tree could work somehow better... > > > > Hi, > > I just updated and tried the current tree; still no sound/headphone > output. :( Ok, I believe I've made some progress on this. The problem appears to be related to the autoconfig handling of the line out nids. The current code ends up with something like the following: ALSA sound/pci/hda/hda_codec.c:3833: autoconfig: line_outs=1 (0x17/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:3837: speaker_outs=0 (0x0/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:3841: hp_outs=1 (0x21/0x0/0x0/0x0/0x0) ALSA sound/pci/hda/hda_codec.c:3842: mono: mono_out=0x0 ALSA sound/pci/hda/hda_codec.c:3853: inputs: mic=0x18, fmic=0x19, line=0x0, fline=0x0, cd=0x0, aux=0x0 However, NID 0x17 is actually a speaker_out. The code that checks for line_outs in snd_hda_parse_pin_def_config() unsets the speaker_out and uses that NID for a line_out. For whatever reason, this breaks things (no sound output, no headphone out). According to the ALC272 data sheet (the block diagram on page 5), there are actually 2 line_out NIDs; 0x14 and 0x15. The default pin configs for these two NIDs (0x411111f0 for both) have the Port Connectivity field set to AC_JACK_PORT_NONE, so the auto-config skips those NIDs. If I provide an override to a) set Port Connectivity to AC_JACK_PORT_COMPLEX, and b) set Default Device to AC_JACK_LINE_OUT (from AC_JACK_SPEAKER), I get usable audio (speakers work, headphone works, headphone plugging mutes the speaker properly, and even the mic works). The patch which I've used is below. Clearly, this isn't a very nice solution, so the question is: is there a way to provide an override/quirk for pincfgs (I see one for HWDEP, but that's configuration-specific) within the current hda_codec framework for this specific codec/laptop? Alternatively, I can try and figure out what's wrong with the code in the case where line_out_pins[0] is set to the speaker_out_pins[0] (ie, NID 0x17 in this case). I can check the other ALC272 machine tomorrow (the DELL_ZM1) to see if the line_out hack enables it to work without the verb table. diff --git a/sound/pci/hda/hda_codec.c b/sound/pci/hda/hda_codec.c index b91f6ed..5aa7a0c 100644 --- a/sound/pci/hda/hda_codec.c +++ b/sound/pci/hda/hda_codec.c @@ -3644,6 +3644,15 @@ int snd_hda_parse_pin_def_config(struct hda_codec *codec, continue; def_conf = snd_hda_codec_get_pincfg(codec, nid); + snd_printd("nid %x: pincfg def_conf=%x\n", nid, def_conf); + if (codec->vendor_id == 0x10ec0272 && + (nid == 0x14 || nid == 0x15)) { + def_conf &= ~(0x3 << AC_DEFCFG_PORT_CONN_SHIFT); + def_conf &= ~(0xf << AC_DEFCFG_DEVICE_SHIFT); + snd_printd("nid %x: overriding def_conf, now %x\n", nid, + def_conf); + } + if (get_defcfg_connect(def_conf) == AC_JACK_PORT_NONE) continue; loc = get_defcfg_location(def_conf);

2 12

[alsa-devel] Adding MMAP support to alsa driver
by Gabriel Johnson 04 May '09

04 May '09

What structures need to be filled out in the hardware driver to support MMAP calls?

2 1

[alsa-devel] New codec for STAC9766 used on Efika
by Jon Smirl 02 May '09

02 May '09

Could you please review this new STAC9766 codec implementation. Some of the controls aren't where they need to be. For example the mutes on the mixer inputs. I'm not sure where they are supposed to go. How should SPDIF support work? The hardware supports simultaneous use of analog and SPDIF. Do I want one DAI or two? If I have two DAI then ALSA needs to interleave the samples into a single DMA stream. So if one stream isn't active it needs to be filled with silence. Other options... A single DAI with a switch A single DAI with the same output stream on both analog and SPDIF. I've tried to add the DAPM routes. I don't know of anyone using this chip on a low power design. Datasheet is here: http://www.idt.com/products/getDoc.cfm?docID=13134007 I get a few errors from the current config: Failed to add route All Analog Mixer->Pop Bypass Mux dapm: STAC9766: configuring unknown pin OUT3 dapm: STAC9766: configuring unknown pin MONOOUT Failed to add route HPOUTL->Headphone Jack dapm: STAC9766: configuring unknown pin OUT3 dapm: STAC9766: configuring unknown pin MONOOUT Failed to add route HPOUTL->Headphone Jack /* * stac9766.c -- ALSA Soc STAC9766 codec support * * Copyright 2008 Jon Smirl, Digispeaker * Author: Jon Smirl <jonsmirl(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 as published by the * Free Software Foundation; either version 2 of the License, or (at your * option) any later version. * * Features:- * * o Support for AC97 Codec, S/PDIF * o Support for DAPM */ #include <linux/init.h> #include <linux/module.h> #include <linux/device.h> #include <sound/core.h> #include <sound/pcm.h> #include <sound/ac97_codec.h> #include <sound/initval.h> #include <sound/pcm_params.h> #include <sound/soc.h> #include <sound/soc-dapm.h> #include <sound/soc-of-simple.h> #include "stac9766.h" #define STAC9766_VERSION "0.10" /* * STAC9766 register cache */ static const u16 stac9766_reg[] = { 0x6A90, 0x8000, 0x8000, 0x8000, // 6 0x0000, 0x0000, 0x8008, 0x8008, // e 0x8808, 0x8808, 0x8808, 0x8808, // 16 0x8808, 0x0000, 0x8000, 0x0000, // 1e 0x0000, 0x0000, 0x0000, 0x000f, // 26 0x0a05, 0x0400, 0xbb80, 0x0000, // 2e 0x0000, 0xbb80, 0x0000, 0x0000, // 36 0x0000, 0x2000, 0x0000, 0x0100, // 3e 0x0000, 0x0000, 0x0080, 0x0000, // 46 0x0000, 0x0000, 0x0003, 0xffff, // 4e 0x0000, 0x0000, 0x0000, 0x0000, // 56 0x4000, 0x0000, 0x0000, 0x0000, // 5e 0x1201, 0xFFFF, 0xFFFF, 0x0000, // 66 0x0000, 0x0000, 0x0000, 0x0000, // 6e 0x0000, 0x0000, 0x0000, 0x0006, // 76 0x0000, 0x0000, 0x0000, 0x0000, // 7e }; static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX", "Line", "Stereo Mix", "Mono Mix", "Phone"}; static const char *stac9766_mono_mux[] = {"Mix", "Mic"}; static const char *stac9766_mic_mux[] = {"Mic1", "Mic2"}; static const char *stac9766_SPDIF_mux[] = {"PCM", "ADC Record"}; static const char *stac9766_popbypass_mux[] = {"Normal", "Bypass Mixer"}; static const char *stac9766_record_all_mux[] = {"All analog", "Analog plus DAC"}; static const char *stac9766_3D_separation[] = {"Off", "Low", "Medium", "High"}; static const struct soc_enum stac9766_enum[] = { SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 8, stac9766_record_mux), /* Record Mux 0 */ SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 9, 2, stac9766_mono_mux), /* Mono Mux 1 */ SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, stac9766_mic_mux), /* Mic1/2 Mux 2 */ SOC_ENUM_SINGLE(AC97_SENSE_INFO, 1, 2, stac9766_SPDIF_mux), /* SPDIF Mux 3 */ SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, stac9766_popbypass_mux), /* Pop Bypass Mux 4 */ SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2, stac9766_record_all_mux), /* Record All Mux 5 */ SOC_ENUM_SINGLE(AC97_3D_CONTROL, 2, 4, stac9766_3D_separation), /* 3D Separation 7 */ }; static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = { SOC_DOUBLE("Speaker Playback Volume", AC97_MASTER, 8, 0, 31, 1), SOC_SINGLE("Speaker Playback Switch", AC97_MASTER, 15, 1, 1), SOC_DOUBLE("Headphone Playback Volume", AC97_HEADPHONE, 8, 0, 31, 1), SOC_SINGLE("Headphone Playback Switch", AC97_HEADPHONE, 15, 1, 1), SOC_SINGLE("Mono Playback Volume", AC97_MASTER_MONO, 0, 31, 1), SOC_SINGLE("Mono Playback Switch", AC97_MASTER_MONO, 15, 1, 1), SOC_SINGLE("Mixer PC Beep Volume", AC97_PC_BEEP, 1, 15, 1), SOC_SINGLE("PC Beep Frequency", AC97_PC_BEEP, 5, 127, 1), SOC_SINGLE("Mixer Phone Volume", AC97_PHONE, 0, 31, 1), SOC_SINGLE("Mixer Mic Volume", AC97_MIC, 0, 31, 1), SOC_SINGLE("Mic Boost", AC97_MIC, 6, 1, 1), SOC_SINGLE("Mic Gain", AC97_STAC_ANALOG_SPECIAL, 2, 1, 1), SOC_SINGLE("Stereo Mic", AC97_STAC_STEREO_MIC, 0, 1, 1), SOC_DOUBLE("Mixer Line Volume", AC97_LINE, 8, 0, 31, 1), SOC_DOUBLE("Mixer CD Volume", AC97_CD, 8, 0, 31, 1), SOC_DOUBLE("Mixer Video Volume", AC97_VIDEO, 8, 0, 31, 1), SOC_DOUBLE("Mixer AUX Volume", AC97_AUX, 8, 0, 31, 1), SOC_DOUBLE("All Analog PCM Volume", AC97_PCM, 8, 0, 31, 1), SOC_DOUBLE("Record Gain", AC97_REC_GAIN, 8, 0, 31, 1), SOC_SINGLE("Record Gain Switch", AC97_REC_GAIN, 15, 1, 1), SOC_SINGLE("3D Effect Switch", AC97_GENERAL_PURPOSE, 13, 1, 0), SOC_SINGLE("Loopback Test Switch", AC97_GENERAL_PURPOSE, 7, 1, 0), SOC_ENUM("3D Separation", stac9766_enum[6]), }; /* Mixer */ static const struct snd_kcontrol_new stac9766_main_mixer_controls[] = { SOC_DAPM_SINGLE("PC Beep Switch", AC97_PC_BEEP, 15, 1, 1), SOC_DAPM_SINGLE("Phone Switch", AC97_PHONE, 15, 1, 1), SOC_DAPM_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1), SOC_DAPM_SINGLE("Line Switch", AC97_LINE, 15, 1, 1), SOC_DAPM_SINGLE("CD Switch", AC97_CD, 15, 1, 1), SOC_DAPM_SINGLE("AUX Switch", AC97_AUX, 15, 1, 1), SOC_DAPM_SINGLE("Video Switch", AC97_VIDEO, 15, 1, 1), }; /* Record All Mixer */ static const struct snd_kcontrol_new stac9766_record_all_mixer_controls[] = { SOC_DAPM_SINGLE("PCM Switch", AC97_PCM, 15, 1, 1), SOC_DAPM_SINGLE("Mixer", 0, 0, 0, 0), }; /* Record Mux 0 */ static const struct snd_kcontrol_new stac9766_record_mux_controls = SOC_DAPM_ENUM("Route", stac9766_enum[0]); /* Mono Mux 1 */ static const struct snd_kcontrol_new stac9766_mono_mux_controls = SOC_DAPM_ENUM("Route", stac9766_enum[1]); /* Mic1/2 Mux 2 */ static const struct snd_kcontrol_new stac9766_mic_mux_controls = SOC_DAPM_ENUM("Route", stac9766_enum[2]); /* SPDIF Mux 3 */ static const struct snd_kcontrol_new stac9766_spdif_mux_controls = SOC_DAPM_ENUM("Route", stac9766_enum[3]); /* Pop Bypass 4 */ static const struct snd_kcontrol_new stac9766_popbypass_mux_controls = SOC_DAPM_ENUM("Route", stac9766_enum[4]); /* Record All 5 */ static const struct snd_kcontrol_new stac9766_record_all_mux_controls = SOC_DAPM_ENUM("Route", stac9766_enum[5]); static const struct snd_soc_dapm_widget stac9766_dapm_widgets[] = { SND_SOC_DAPM_INPUT("PCMOut"), SND_SOC_DAPM_INPUT("PCBEEP"), SND_SOC_DAPM_INPUT("Phone"), SND_SOC_DAPM_INPUT("MIC1"), SND_SOC_DAPM_INPUT("MIC2"), SND_SOC_DAPM_INPUT("Line"), SND_SOC_DAPM_INPUT("CD"), SND_SOC_DAPM_INPUT("AUX"), SND_SOC_DAPM_INPUT("Video"), SND_SOC_DAPM_DAC("DAC", "Analog Playback", AC97_POWERDOWN, 9, 1), SND_SOC_DAPM_MUX("SPDIF Mux", SND_SOC_NOPM, 0, 0, &stac9766_spdif_mux_controls), SND_SOC_DAPM_MUX("Mic1/2 Mux", SND_SOC_NOPM, 0, 0, &stac9766_mic_mux_controls), SND_SOC_DAPM_MIXER("Mixer", SND_SOC_NOPM, 0, 1, &stac9766_main_mixer_controls[0], ARRAY_SIZE(stac9766_main_mixer_controls)), SND_SOC_DAPM_MUX("Record All Mux", SND_SOC_NOPM, 0, 0, &stac9766_record_all_mux_controls), SND_SOC_DAPM_MUX("Record Mux", SND_SOC_NOPM, 0, 0, &stac9766_record_mux_controls), SND_SOC_DAPM_MUX("Mono Mux", SND_SOC_NOPM, 0, 0, &stac9766_mono_mux_controls), SND_SOC_DAPM_MUX("Pop Bypass Mux", SND_SOC_NOPM, 0, 0, &stac9766_popbypass_mux_controls), SND_SOC_DAPM_MIXER("All Analog", SND_SOC_NOPM, 0, 1, &stac9766_record_all_mixer_controls[0], ARRAY_SIZE(stac9766_record_all_mixer_controls)), SND_SOC_DAPM_ADC("ADC", "Analog Capture", AC97_POWERDOWN, 8, 1), SND_SOC_DAPM_OUTPUT("HP"), SND_SOC_DAPM_OUTPUT("LINE"), SND_SOC_DAPM_OUTPUT("MONO"), SND_SOC_DAPM_OUTPUT("PCMIn"), SND_SOC_DAPM_VMID("VMID"), }; static const struct snd_soc_dapm_route audio_map[] = { {"HP", NULL, "Pop Bypass Mux"}, {"LINE", NULL, "Pop Bypass Mux"}, /* Mono Mux */ {"MONO", NULL, "Pop Bypass Mux"}, {"MONO", NULL, "Mic1/2 Mux"}, /* Pop Bypass Mux */ {"Pop Bypass Mux", NULL, "DAC"}, {"Pop Bypass Mux", NULL, "All Analog Mixer"}, /* Record Mux */ {"ADC", NULL, "Mic1/2 Mux"}, {"ADC", NULL, "CD"}, {"ADC", NULL, "Video"}, {"ADC", NULL, "AUX"}, {"ADC", NULL, "Line"}, {"ADC", NULL, "Record All Mux"}, {"ADC", NULL, "Phone"}, {"PCMIn", NULL, "ADC"}, /* Record All Mux */ {"Record All Mux", NULL, "Mixer"}, {"Record All Mux", NULL, "All Analog Mixer"}, /* All Analog Mixer */ {"All Analog", NULL, "Mixer"}, {"All Analog", "PCM Switch", "DAC"}, /* Mixer */ {"Mixer", "PC Beep Switch", "PCBEEP"}, {"Mixer", "Phone Switch", "Phone"}, {"Mixer", "Mic Switch", "Mic1/2 Mux"}, {"Mixer", "Line Switch", "Line"}, {"Mixer", "CD Switch", "CD"}, {"Mixer", "AUX Switch", "AUX"}, {"Mixer", "Video Switch", "Video"}, {"DAC", NULL, "PCMOut"}, /* Mic1/2 Mux */ {"Mic1/2 Mux", NULL, "MIC1"}, {"Mic1/2 Mux", NULL, "MIC2"}, /* SPDIF Mux */ {"SPDIF Mux", NULL, "PCMOut"}, {"SPDIF Mux", NULL, "ADC"}, {NULL, NULL, NULL}, }; static int stac9766_add_widgets(struct snd_soc_codec *codec) { snd_soc_dapm_new_controls(codec, stac9766_dapm_widgets, ARRAY_SIZE(stac9766_dapm_widgets)); snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map)); snd_soc_dapm_new_widgets(codec); return 0; } unsigned int stac9766_ac97_read(struct snd_soc_codec *codec, unsigned int reg) { u16 val = 0, *cache = codec->reg_cache; if (reg / 2 > ARRAY_SIZE(stac9766_reg)) return -EIO; if (reg == AC97_RESET || reg == AC97_GPIO_STATUS || AC97_INT_PAGING || reg == AC97_VENDOR_ID1 || reg == AC97_VENDOR_ID2) { val = soc_ac97_ops.read(codec->ac97, reg); return val; } return cache[reg / 2]; } int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg, unsigned int val) { u16 *cache = codec->reg_cache; if (reg / 2 > ARRAY_SIZE(stac9766_reg)) return -EIO; soc_ac97_ops.write(codec->ac97, reg, val); cache[reg / 2] = val; return 0; } static int ac97_analog_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { struct snd_soc_codec *codec = dai->codec; struct snd_pcm_runtime *runtime = substream->runtime; unsigned short reg, vra; vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS); //vra |= 0x4; stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra | 0x1); printk("AC97_EXTENDED_STATUS %x\n", vra); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) reg = AC97_PCM_FRONT_DAC_RATE; else reg = AC97_PCM_LR_ADC_RATE; return stac9766_ac97_write(codec, reg, runtime->rate); } static int ac97_digital_prepare(struct snd_pcm_substream *substream, struct snd_soc_dai *dai) { printk("stac9766: ac97_digital_prepare\n"); return 0; } static int stac9766_set_bias_level(struct snd_soc_codec *codec, enum snd_soc_bias_level level) { switch (level) { case SND_SOC_BIAS_ON: /* full On */ stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000); break; case SND_SOC_BIAS_PREPARE: /* partial On */ stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000); break; case SND_SOC_BIAS_STANDBY: /* Off, with power */ stac9766_ac97_write(codec, AC97_POWERDOWN, 0x0000); break; case SND_SOC_BIAS_OFF: /* Off, without power */ /* disable everything including AC link */ stac9766_ac97_write(codec, AC97_POWERDOWN, 0xffff); break; } codec->bias_level = level; return 0; } static int stac9766_codec_suspend(struct platform_device *pdev, pm_message_t state) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; stac9766_set_bias_level(codec, SND_SOC_BIAS_OFF); return 0; } static int stac9766_codec_resume(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; u16 id; /* give the codec an AC97 warm reset to start the link */ codec->ac97->bus->ops->warm_reset(codec->ac97); id = soc_ac97_ops.read(codec->ac97, AC97_VENDOR_ID2); if (id != 0x4c13) { printk(KERN_ERR "stac9766 failed to resume"); return -EIO; } stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY); if (codec->suspend_bias_level == SND_SOC_BIAS_ON) stac9766_set_bias_level(codec, SND_SOC_BIAS_ON); return 0; } static struct snd_soc_dai_ops stac9766_dai_ops_analog = { .prepare = ac97_analog_prepare, }; static struct snd_soc_dai_ops stac9766_dai_ops_digital = { .prepare = ac97_digital_prepare, }; struct snd_soc_dai stac9766_dai[] = { { .name = "stac9766 analog", .id = 0, .ac97_control = 1, /* stream cababilities */ .playback = { .stream_name = "stac9766 analog", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S32_BE, }, .capture = { .stream_name = "stac9766 analog", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_8000_48000, .formats = SNDRV_PCM_FMTBIT_S32_BE, }, /* alsa ops */ .ops = &stac9766_dai_ops_analog, }, { .name = "stac9766 digital", .id = 1, .ac97_control = 1, /* stream cababilities */ .playback = { .stream_name = "stac9766 digital", .channels_min = 1, .channels_max = 2, .rates = SNDRV_PCM_RATE_32000 | \ SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000, .formats = SNDRV_PCM_FMTBIT_IEC958_SUBFRAME, }, /* alsa ops */ .ops = &stac9766_dai_ops_digital, }}; EXPORT_SYMBOL_GPL(stac9766_dai); int stac9766_reset(struct snd_soc_codec *codec, int try_warm) { if (try_warm && soc_ac97_ops.warm_reset) { soc_ac97_ops.warm_reset(codec->ac97); if (stac9766_ac97_read(codec, 0) == stac9766_reg[0]) return 1; } soc_ac97_ops.reset(codec->ac97); if (soc_ac97_ops.warm_reset) soc_ac97_ops.warm_reset(codec->ac97); if (stac9766_ac97_read(codec, 0) != stac9766_reg[0]) return -EIO; return 0; } static int stac9766_codec_probe(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec; int ret = 0; printk(KERN_INFO "STAC9766 SoC Audio Codec %s\n", STAC9766_VERSION); socdev->card->codec = kzalloc(sizeof(struct snd_soc_codec), GFP_KERNEL); if (socdev->card->codec == NULL) return -ENOMEM; codec = socdev->card->codec; mutex_init(&codec->mutex); codec->reg_cache = kmemdup(stac9766_reg, sizeof(stac9766_reg), GFP_KERNEL); if (codec->reg_cache == NULL) { ret = -ENOMEM; goto cache_err; } codec->reg_cache_size = sizeof(stac9766_reg); codec->reg_cache_step = 2; codec->name = "STAC9766"; codec->owner = THIS_MODULE; codec->dai = stac9766_dai; codec->num_dai = ARRAY_SIZE(stac9766_dai); codec->write = stac9766_ac97_write; codec->read = stac9766_ac97_read; codec->set_bias_level = stac9766_set_bias_level; INIT_LIST_HEAD(&codec->dapm_widgets); INIT_LIST_HEAD(&codec->dapm_paths); ret = snd_soc_new_ac97_codec(codec, &soc_ac97_ops, 0); if (ret < 0) goto codec_err; /* register pcms */ ret = snd_soc_new_pcms(socdev, SNDRV_DEFAULT_IDX1, SNDRV_DEFAULT_STR1); if (ret < 0) goto pcm_err; /* do a cold reset for the controller and then try * a warm reset followed by an optional cold reset for codec */ stac9766_reset(codec, 0); ret = stac9766_reset(codec, 1); if (ret < 0) { printk(KERN_ERR "Failed to reset STAC9766: AC97 link error\n"); goto reset_err; } stac9766_set_bias_level(codec, SND_SOC_BIAS_STANDBY); snd_soc_add_controls(codec, stac9766_snd_ac97_controls, ARRAY_SIZE(stac9766_snd_ac97_controls)); stac9766_add_widgets(codec); ret = snd_soc_init_card(socdev); if (ret < 0) goto reset_err; return 0; reset_err: snd_soc_free_pcms(socdev); pcm_err: snd_soc_free_ac97_codec(codec); codec_err: kfree(codec->private_data); cache_err: kfree(socdev->card->codec); socdev->card->codec = NULL; return ret; } static int stac9766_codec_remove(struct platform_device *pdev) { struct snd_soc_device *socdev = platform_get_drvdata(pdev); struct snd_soc_codec *codec = socdev->card->codec; if (codec == NULL) return 0; snd_soc_dapm_free(socdev); snd_soc_free_pcms(socdev); snd_soc_free_ac97_codec(codec); kfree(codec->reg_cache); kfree(codec); return 0; } struct snd_soc_codec_device soc_codec_dev_stac9766 = { .probe = stac9766_codec_probe, .remove = stac9766_codec_remove, .suspend = stac9766_codec_suspend, .resume = stac9766_codec_resume, }; EXPORT_SYMBOL_GPL(soc_codec_dev_stac9766); static int __init stac9766_probe(struct platform_device *pdev) { #if defined(CONFIG_SND_SOC_OF_SIMPLE) /* Tell the of_soc helper about this codec */ of_snd_soc_register_codec(&soc_codec_dev_stac9766, pdev->dev.archdata.of_node, stac9766_dai, ARRAY_SIZE(stac9766_dai), pdev->dev.archdata.of_node); #endif return 0; } static struct platform_driver stac9766_driver = { .probe = stac9766_probe, .driver = { .name = "stac9766", }, }; static __init int stac9766_driver_init(void) { snd_soc_register_dais(stac9766_dai, ARRAY_SIZE(stac9766_dai)); return platform_driver_register(&stac9766_driver); } static __exit void stac9766_driver_exit(void) { } module_init(stac9766_driver_init); module_exit(stac9766_driver_exit); MODULE_DESCRIPTION("ASoC stac9766 driver"); MODULE_AUTHOR("Jon Smirl <jonsmirl(a)gmail.com>"); MODULE_LICENSE("GPL"); -- Jon Smirl jonsmirl(a)gmail.com

2 8

[alsa-devel] [PATCH] ASoC: TWL4030: Disable DACs in analog loopback
by Lopez Cruz, Misael 30 Apr '09

30 Apr '09

In analog loopback mode, DACs should not be enabled. For that reason, DAC widgets (DAC Right1, DAC Left1, DAC Right2, DAC Left2) will power DACs only during playback. Analog loopback requires to set a master enable bit when any of the analog bypass switches are enabled. Signed-off-by: Misael Lopez Cruz <x0052729(a)ti.com> --- sound/soc/codecs/twl4030.c | 34 +++++++++++++++++++++------------- 1 files changed, 21 insertions(+), 13 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index efa1a80..d3184a0 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -64,7 +64,7 @@ static const u8 twl4030_reg[TWL4030_CACHEREGNUM] = { 0x00, /* REG_VRXPGA (0x14) */ 0x00, /* REG_VSTPGA (0x15) */ 0x00, /* REG_VRX2ARXPGA (0x16) */ - 0x0c, /* REG_AVDAC_CTL (0x17) */ + 0x00, /* REG_AVDAC_CTL (0x17) */ 0x00, /* REG_ARX2VTXPGA (0x18) */ 0x00, /* REG_ARXL1_APGA_CTL (0x19) */ 0x00, /* REG_ARXR1_APGA_CTL (0x1A) */ @@ -585,7 +585,7 @@ static int bypass_event(struct snd_soc_dapm_widget *w, struct soc_mixer_control *m = (struct soc_mixer_control *)w->kcontrols->private_value; struct twl4030_priv *twl4030 = w->codec->private_data; - unsigned char reg; + unsigned char reg, misc; reg = twl4030_read_reg_cache(w->codec, m->reg); @@ -605,6 +605,14 @@ static int bypass_event(struct snd_soc_dapm_widget *w, twl4030->bypass_state &= ~(1 << (m->shift ? 5 : 4)); } + /* Enable master analog loopback mode if any analog switch is enabled*/ + misc = twl4030_read_reg_cache(w->codec, TWL4030_REG_MISC_SET_1); + if (twl4030->bypass_state & 0xF) + misc |= TWL4030_FMLOOP_EN; + else + misc &= ~TWL4030_FMLOOP_EN; + twl4030_write(w->codec, TWL4030_REG_MISC_SET_1, misc); + if (w->codec->bias_level == SND_SOC_BIAS_STANDBY) { if (twl4030->bypass_state) twl4030_codec_mute(w->codec, 0); @@ -927,13 +935,13 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { /* DACs */ SND_SOC_DAPM_DAC("DAC Right1", "Right Front Playback", - SND_SOC_NOPM, 0, 0), + TWL4030_REG_AVDAC_CTL, 0, 0), SND_SOC_DAPM_DAC("DAC Left1", "Left Front Playback", - SND_SOC_NOPM, 0, 0), + TWL4030_REG_AVDAC_CTL, 1, 0), SND_SOC_DAPM_DAC("DAC Right2", "Right Rear Playback", - SND_SOC_NOPM, 0, 0), + TWL4030_REG_AVDAC_CTL, 2, 0), SND_SOC_DAPM_DAC("DAC Left2", "Left Rear Playback", - SND_SOC_NOPM, 0, 0), + TWL4030_REG_AVDAC_CTL, 3, 0), SND_SOC_DAPM_DAC("DAC Voice", "Voice Playback", TWL4030_REG_AVDAC_CTL, 4, 0), @@ -971,14 +979,14 @@ static const struct snd_soc_dapm_widget twl4030_dapm_widgets[] = { &twl4030_dapm_dbypassr_control, bypass_event, SND_SOC_DAPM_POST_REG), - SND_SOC_DAPM_MIXER("Analog R1 Playback Mixer", TWL4030_REG_AVDAC_CTL, + SND_SOC_DAPM_MIXER("Analog R1 Playback Mixer", SND_SOC_NOPM, + 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Analog L1 Playback Mixer", SND_SOC_NOPM, + 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Analog R2 Playback Mixer", SND_SOC_NOPM, + 0, 0, NULL, 0), + SND_SOC_DAPM_MIXER("Analog L2 Playback Mixer", SND_SOC_NOPM, 0, 0, NULL, 0), - SND_SOC_DAPM_MIXER("Analog L1 Playback Mixer", TWL4030_REG_AVDAC_CTL, - 1, 0, NULL, 0), - SND_SOC_DAPM_MIXER("Analog R2 Playback Mixer", TWL4030_REG_AVDAC_CTL, - 2, 0, NULL, 0), - SND_SOC_DAPM_MIXER("Analog L2 Playback Mixer", TWL4030_REG_AVDAC_CTL, - 3, 0, NULL, 0), /* Output MIXER controls */ /* Earpiece */ -- 1.5.4.3

2 5

[alsa-devel] errors in wm8990 codec driver
by Jinyoung Park 30 Apr '09

30 Apr '09

Hello, I found a number of typing errors in wm8990 codec driver. The mis-typing exist in dapm controller definitions and dapm route definitions, so happen mis-matched error when snd_soc_dapm_add_routes(). Belows are my modified code. Please check... Thanks. diff --git a/sound/soc/codecs/wm8990.c b/sound/soc/codecs/wm8990.c index cedf3f9..ddc328e 100644 --- a/sound/soc/codecs/wm8990.c +++ b/sound/soc/codecs/wm8990.c @@ -729,7 +729,7 @@ SND_SOC_DAPM_MIXER_E("INMIXL", WM8990_INTDRIVBITS, WM8990_INMIXL_P inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), /* AINLMUX */ -SND_SOC_DAPM_MUX_E("AILNMUX", WM8990_INTDRIVBITS, WM8990_AINLMUX_PWR_BIT, 0, +SND_SOC_DAPM_MUX_E("AINLMUX", WM8990_INTDRIVBITS, WM8990_AINLMUX_PWR_BIT, 0, &wm8990_dapm_ainlmux_controls, inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), @@ -740,7 +740,7 @@ SND_SOC_DAPM_MIXER_E("INMIXR", WM8990_INTDRIVBITS, WM8990_INMIXR_P inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), /* AINRMUX */ -SND_SOC_DAPM_MUX_E("AIRNMUX", WM8990_INTDRIVBITS, WM8990_AINRMUX_PWR_BIT, 0, +SND_SOC_DAPM_MUX_E("AINRMUX", WM8990_INTDRIVBITS, WM8990_AINRMUX_PWR_BIT, 0, &wm8990_dapm_ainrmux_controls, inmixer_event, SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD), @@ -848,40 +848,40 @@ static const struct snd_soc_dapm_route audio_map[] = { {"LIN12 PGA", "LIN2 Switch", "LIN2"}, /* LIN34 PGA */ {"LIN34 PGA", "LIN3 Switch", "LIN3"}, - {"LIN34 PGA", "LIN4 Switch", "LIN4"}, + {"LIN34 PGA", "LIN4 Switch", "LIN4/RXN"}, /* INMIXL */ {"INMIXL", "Record Left Volume", "LOMIX"}, {"INMIXL", "LIN2 Volume", "LIN2"}, {"INMIXL", "LINPGA12 Switch", "LIN12 PGA"}, {"INMIXL", "LINPGA34 Switch", "LIN34 PGA"}, - /* AILNMUX */ - {"AILNMUX", "INMIXL Mix", "INMIXL"}, - {"AILNMUX", "DIFFINL Mix", "LIN12PGA"}, - {"AILNMUX", "DIFFINL Mix", "LIN34PGA"}, - {"AILNMUX", "RXVOICE Mix", "LIN4/RXN"}, - {"AILNMUX", "RXVOICE Mix", "RIN4/RXP"}, + /* AINLMUX */ + {"AINLMUX", "INMIXL Mix", "INMIXL"}, + {"AINLMUX", "DIFFINL Mix", "LIN12 PGA"}, + {"AINLMUX", "DIFFINL Mix", "LIN34 PGA"}, + {"AINLMUX", "RXVOICE Mix", "LIN4/RXN"}, + {"AINLMUX", "RXVOICE Mix", "RIN4/RXP"}, /* ADC */ - {"Left ADC", NULL, "AILNMUX"}, + {"Left ADC", NULL, "AINLMUX"}, /* RIN12 PGA */ {"RIN12 PGA", "RIN1 Switch", "RIN1"}, {"RIN12 PGA", "RIN2 Switch", "RIN2"}, /* RIN34 PGA */ {"RIN34 PGA", "RIN3 Switch", "RIN3"}, - {"RIN34 PGA", "RIN4 Switch", "RIN4"}, + {"RIN34 PGA", "RIN4 Switch", "RIN4/RXP"}, /* INMIXL */ {"INMIXR", "Record Right Volume", "ROMIX"}, {"INMIXR", "RIN2 Volume", "RIN2"}, {"INMIXR", "RINPGA12 Switch", "RIN12 PGA"}, {"INMIXR", "RINPGA34 Switch", "RIN34 PGA"}, - /* AIRNMUX */ - {"AIRNMUX", "INMIXR Mix", "INMIXR"}, - {"AIRNMUX", "DIFFINR Mix", "RIN12PGA"}, - {"AIRNMUX", "DIFFINR Mix", "RIN34PGA"}, - {"AIRNMUX", "RXVOICE Mix", "RIN4/RXN"}, - {"AIRNMUX", "RXVOICE Mix", "RIN4/RXP"}, + /* AINRMUX */ + {"AINRMUX", "INMIXR Mix", "INMIXR"}, + {"AINRMUX", "DIFFINR Mix", "RIN12 PGA"}, + {"AINRMUX", "DIFFINR Mix", "RIN34 PGA"}, + {"AINRMUX", "RXVOICE Mix", "LIN4/RXN"}, + {"AINRMUX", "RXVOICE Mix", "RIN4/RXP"}, /* ADC */ - {"Right ADC", NULL, "AIRNMUX"}, + {"Right ADC", NULL, "AINRMUX"}, /* LOMIX */ {"LOMIX", "LOMIX RIN3 Bypass Switch", "RIN3"}, @@ -922,7 +922,7 @@ static const struct snd_soc_dapm_route audio_map[] = { {"LOPMIX", "LOPMIX Left Mixer PGA Switch", "LOPGA"}, /* OUT3MIX */ - {"OUT3MIX", "OUT3MIX LIN4/RXP Bypass Switch", "LIN4/RXP"}, + {"OUT3MIX", "OUT3MIX LIN4/RXP Bypass Switch", "LIN4/RXN"}, {"OUT3MIX", "OUT3MIX Left Out PGA Switch", "LOPGA"}, /* OUT4MIX */ @@ -949,7 +949,7 @@ static const struct snd_soc_dapm_route audio_map[] = { /* Output Pins */ {"LON", NULL, "LONMIX"}, {"LOP", NULL, "LOPMIX"}, - {"OUT", NULL, "OUT3MIX"}, + {"OUT3", NULL, "OUT3MIX"}, {"LOUT", NULL, "LOUT PGA"}, {"SPKN", NULL, "SPKMIX"}, {"ROUT", NULL, "ROUT PGA"},

2 1

[alsa-devel] [RFC PATCH 3/5] pxa: clean up the legacy SSP API
by Eric Miao 30 Apr '09

30 Apr '09

Actually, due to the number of possible SSP configurations, it's quite a difficult job to maintain an API that can be well re-used. The guideline here is to keep only a list of SSP devices and their basic information, and it's up to the user driver (like SPI or Audio SSP) to configure the SSP in a desired way. Now that the legacy 'struct ssp_dev' based APIs are removed, and only the functions below are kept, modified a little bit to make them aware of the new 'struct ssp_device': ssp_enable() ssp_disable() ssp_save_state() ssp_restore_state() And finally, ssp_{save,restore}_state() can be merged into the suspend() and resume() interface of the SSP's platform_driver, thus saving the user driver from doing some similar things in common. Signed-off-by: Eric Miao <eric.miao(a)marvell.com> --- arch/arm/mach-pxa/include/mach/ssp.h | 32 +---- arch/arm/mach-pxa/ssp.c | 240 +--------------------------------- sound/soc/pxa/pxa-ssp.c | 65 +++++----- 3 files changed, 45 insertions(+), 292 deletions(-) diff --git a/arch/arm/mach-pxa/include/mach/ssp.h b/arch/arm/mach-pxa/include/mach/ssp.h index cb5cb76..e431685 100644 --- a/arch/arm/mach-pxa/include/mach/ssp.h +++ b/arch/arm/mach-pxa/include/mach/ssp.h @@ -58,26 +58,10 @@ struct ssp_state { u32 psp; }; -struct ssp_dev { - struct ssp_device *ssp; - u32 port; - u32 mode; - u32 flags; - u32 psp_flags; - u32 speed; - int irq; -}; - -int ssp_write_word(struct ssp_dev *dev, u32 data); -int ssp_read_word(struct ssp_dev *dev, u32 *data); -int ssp_flush(struct ssp_dev *dev); -void ssp_enable(struct ssp_dev *dev); -void ssp_disable(struct ssp_dev *dev); -void ssp_save_state(struct ssp_dev *dev, struct ssp_state *ssp); -void ssp_restore_state(struct ssp_dev *dev, struct ssp_state *ssp); -int ssp_init(struct ssp_dev *dev, u32 port, u32 init_flags); -int ssp_config(struct ssp_dev *dev, u32 mode, u32 flags, u32 psp_flags, u32 speed); -void ssp_exit(struct ssp_dev *dev); +void ssp_enable(struct ssp_device *ssp); +void ssp_disable(struct ssp_device *ssp); +void ssp_save_state(struct ssp_device *ssp, struct ssp_state *); +void ssp_restore_state(struct ssp_device *ssp, struct ssp_state *); /** * ssp_write_reg - Write to a SSP register @@ -86,9 +70,9 @@ void ssp_exit(struct ssp_dev *dev); * @reg: Register to write to * @val: Value to be written. */ -static inline void ssp_write_reg(struct ssp_device *dev, u32 reg, u32 val) +static inline void ssp_write_reg(struct ssp_device *ssp, u32 reg, u32 val) { - __raw_writel(val, dev->mmio_base + reg); + __raw_writel(val, ssp->mmio_base + reg); } /** @@ -97,9 +81,9 @@ static inline void ssp_write_reg(struct ssp_device *dev, u32 reg, u32 val) * @dev: SSP device to access * @reg: Register to read from */ -static inline u32 ssp_read_reg(struct ssp_device *dev, u32 reg) +static inline u32 ssp_read_reg(struct ssp_device *ssp, u32 reg) { - return __raw_readl(dev->mmio_base + reg); + return __raw_readl(ssp->mmio_base + reg); } struct ssp_device *ssp_request(int port, const char *label); diff --git a/arch/arm/mach-pxa/ssp.c b/arch/arm/mach-pxa/ssp.c index 965e38c..f48807a 100644 --- a/arch/arm/mach-pxa/ssp.c +++ b/arch/arm/mach-pxa/ssp.c @@ -35,153 +35,25 @@ #include <mach/ssp.h> #include <mach/regs-ssp.h> -#define TIMEOUT 100000 - -static irqreturn_t ssp_interrupt(int irq, void *dev_id) +void ssp_enable(struct ssp_device *ssp) { - struct ssp_dev *dev = dev_id; - struct ssp_device *ssp = dev->ssp; - unsigned int status; - - status = __raw_readl(ssp->mmio_base + SSSR); - __raw_writel(status, ssp->mmio_base + SSSR); - - if (status & SSSR_ROR) - printk(KERN_WARNING "SSP(%d): receiver overrun\n", dev->port); - - if (status & SSSR_TUR) - printk(KERN_WARNING "SSP(%d): transmitter underrun\n", dev->port); - - if (status & SSSR_BCE) - printk(KERN_WARNING "SSP(%d): bit count error\n", dev->port); - - return IRQ_HANDLED; -} - -/** - * ssp_write_word - write a word to the SSP port - * @data: 32-bit, MSB justified data to write. - * - * Wait for a free entry in the SSP transmit FIFO, and write a data - * word to the SSP port. - * - * The caller is expected to perform the necessary locking. - * - * Returns: - * %-ETIMEDOUT timeout occurred - * 0 success - */ -int ssp_write_word(struct ssp_dev *dev, u32 data) -{ - struct ssp_device *ssp = dev->ssp; - int timeout = TIMEOUT; - - while (!(__raw_readl(ssp->mmio_base + SSSR) & SSSR_TNF)) { - if (!--timeout) - return -ETIMEDOUT; - cpu_relax(); - } - - __raw_writel(data, ssp->mmio_base + SSDR); - - return 0; -} - -/** - * ssp_read_word - read a word from the SSP port - * - * Wait for a data word in the SSP receive FIFO, and return the - * received data. Data is LSB justified. - * - * Note: Currently, if data is not expected to be received, this - * function will wait for ever. - * - * The caller is expected to perform the necessary locking. - * - * Returns: - * %-ETIMEDOUT timeout occurred - * 32-bit data success - */ -int ssp_read_word(struct ssp_dev *dev, u32 *data) -{ - struct ssp_device *ssp = dev->ssp; - int timeout = TIMEOUT; - - while (!(__raw_readl(ssp->mmio_base + SSSR) & SSSR_RNE)) { - if (!--timeout) - return -ETIMEDOUT; - cpu_relax(); - } - - *data = __raw_readl(ssp->mmio_base + SSDR); - return 0; -} - -/** - * ssp_flush - flush the transmit and receive FIFOs - * - * Wait for the SSP to idle, and ensure that the receive FIFO - * is empty. - * - * The caller is expected to perform the necessary locking. - */ -int ssp_flush(struct ssp_dev *dev) -{ - struct ssp_device *ssp = dev->ssp; - int timeout = TIMEOUT * 2; - - /* ensure TX FIFO is empty instead of not full */ - if (cpu_is_pxa3xx()) { - while (__raw_readl(ssp->mmio_base + SSSR) & 0xf00) { - if (!--timeout) - return -ETIMEDOUT; - cpu_relax(); - } - timeout = TIMEOUT * 2; - } - - do { - while (__raw_readl(ssp->mmio_base + SSSR) & SSSR_RNE) { - if (!--timeout) - return -ETIMEDOUT; - (void)__raw_readl(ssp->mmio_base + SSDR); - } - if (!--timeout) - return -ETIMEDOUT; - } while (__raw_readl(ssp->mmio_base + SSSR) & SSSR_BSY); - - return 0; -} - -/** - * ssp_enable - enable the SSP port - * - * Turn on the SSP port. - */ -void ssp_enable(struct ssp_dev *dev) -{ - struct ssp_device *ssp = dev->ssp; uint32_t sscr0; sscr0 = __raw_readl(ssp->mmio_base + SSCR0); sscr0 |= SSCR0_SSE; __raw_writel(sscr0, ssp->mmio_base + SSCR0); } +EXPORT_SYMBOL(ssp_enable); -/** - * ssp_disable - shut down the SSP port - * - * Turn off the SSP port, optionally powering it down. - */ -void ssp_disable(struct ssp_dev *dev) +void ssp_disable(struct ssp_device *ssp) { - struct ssp_device *ssp = dev->ssp; uint32_t sscr0; sscr0 = __raw_readl(ssp->mmio_base + SSCR0); sscr0 &= ~SSCR0_SSE; __raw_writel(sscr0, ssp->mmio_base + SSCR0); } +EXPORT_SYMBOL(ssp_disable); /** * ssp_save_state - save the SSP configuration @@ -189,16 +61,12 @@ void ssp_disable(struct ssp_dev *dev) * * Save the configured SSP state for suspend. */ -void ssp_save_state(struct ssp_dev *dev, struct ssp_state *state) +void ssp_save_state(struct ssp_device *ssp, struct ssp_state *state) { - struct ssp_device *ssp = dev->ssp; - state->cr0 = __raw_readl(ssp->mmio_base + SSCR0); state->cr1 = __raw_readl(ssp->mmio_base + SSCR1); state->to = __raw_readl(ssp->mmio_base + SSTO); state->psp = __raw_readl(ssp->mmio_base + SSPSP); - - ssp_disable(dev); } /** @@ -207,9 +75,8 @@ void ssp_save_state(struct ssp_dev *dev, struct ssp_state *state) * * Restore the SSP configuration saved previously by ssp_save_state. */ -void ssp_restore_state(struct ssp_dev *dev, struct ssp_state *state) +void ssp_restore_state(struct ssp_device *ssp, struct ssp_state *state) { - struct ssp_device *ssp = dev->ssp; uint32_t sssr = SSSR_ROR | SSSR_TUR | SSSR_BCE; __raw_writel(sssr, ssp->mmio_base + SSSR); @@ -221,89 +88,6 @@ void ssp_restore_state(struct ssp_dev *dev, struct ssp_state *state) __raw_writel(state->cr0, ssp->mmio_base + SSCR0); } -/** - * ssp_config - configure SSP port settings - * @mode: port operating mode - * @flags: port config flags - * @psp_flags: port PSP config flags - * @speed: port speed - * - * Port MUST be disabled by ssp_disable before making any config changes. - */ -int ssp_config(struct ssp_dev *dev, u32 mode, u32 flags, u32 psp_flags, u32 speed) -{ - struct ssp_device *ssp = dev->ssp; - - dev->mode = mode; - dev->flags = flags; - dev->psp_flags = psp_flags; - dev->speed = speed; - - /* set up port type, speed, port settings */ - __raw_writel((dev->speed | dev->mode), ssp->mmio_base + SSCR0); - __raw_writel(dev->flags, ssp->mmio_base + SSCR1); - __raw_writel(dev->psp_flags, ssp->mmio_base + SSPSP); - - return 0; -} - -/** - * ssp_init - setup the SSP port - * - * initialise and claim resources for the SSP port. - * - * Returns: - * %-ENODEV if the SSP port is unavailable - * %-EBUSY if the resources are already in use - * %0 on success - */ -int ssp_init(struct ssp_dev *dev, u32 port, u32 init_flags) -{ - struct ssp_device *ssp; - int ret; - - ssp = ssp_request(port, "SSP"); - if (ssp == NULL) - return -ENODEV; - - dev->ssp = ssp; - dev->port = port; - - /* do we need to get irq */ - if (!(init_flags & SSP_NO_IRQ)) { - ret = request_irq(ssp->irq, ssp_interrupt, - 0, "SSP", dev); - if (ret) - goto out_region; - dev->irq = ssp->irq; - } else - dev->irq = NO_IRQ; - - /* turn on SSP port clock */ - clk_enable(ssp->clk); - return 0; - -out_region: - ssp_free(ssp); - return ret; -} - -/** - * ssp_exit - undo the effects of ssp_init - * - * release and free resources for the SSP port. - */ -void ssp_exit(struct ssp_dev *dev) -{ - struct ssp_device *ssp = dev->ssp; - - ssp_disable(dev); - if (dev->irq != NO_IRQ) - free_irq(dev->irq, dev); - clk_disable(ssp->clk); - ssp_free(ssp); -} - static DEFINE_MUTEX(ssp_lock); static LIST_HEAD(ssp_list); @@ -531,18 +315,6 @@ static void __exit pxa_ssp_exit(void) arch_initcall(pxa_ssp_init); module_exit(pxa_ssp_exit); -EXPORT_SYMBOL(ssp_write_word); -EXPORT_SYMBOL(ssp_read_word); -EXPORT_SYMBOL(ssp_flush); -EXPORT_SYMBOL(ssp_enable); -EXPORT_SYMBOL(ssp_disable); -EXPORT_SYMBOL(ssp_save_state); -EXPORT_SYMBOL(ssp_restore_state); -EXPORT_SYMBOL(ssp_init); -EXPORT_SYMBOL(ssp_exit); -EXPORT_SYMBOL(ssp_config); - MODULE_DESCRIPTION("PXA SSP driver"); MODULE_AUTHOR("Liam Girdwood"); MODULE_LICENSE("GPL"); - diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index 308a657..a65993c 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -42,7 +42,7 @@ * SSP audio private data */ struct ssp_priv { - struct ssp_dev dev; + struct ssp_device *ssp; unsigned int sysclk; int dai_fmt; #ifdef CONFIG_PM @@ -222,10 +222,8 @@ static int pxa_ssp_startup(struct snd_pcm_substream *substream, int ret = 0; if (!cpu_dai->active) { - priv->dev.port = cpu_dai->id + 1; - priv->dev.irq = NO_IRQ; - clk_enable(priv->dev.ssp->clk); - ssp_disable(&priv->dev); + clk_enable(priv->ssp->clk); + ssp_disable(priv->ssp); } return ret; } @@ -238,8 +236,8 @@ static void pxa_ssp_shutdown(struct snd_pcm_substream *substream, struct ssp_priv *priv = cpu_dai->private_data; if (!cpu_dai->active) { - ssp_disable(&priv->dev); - clk_disable(priv->dev.ssp->clk); + ssp_disable(priv->ssp); + clk_disable(priv->ssp->clk); } } @@ -252,8 +250,9 @@ static int pxa_ssp_suspend(struct snd_soc_dai *cpu_dai) if (!cpu_dai->active) return 0; - ssp_save_state(&priv->dev, &priv->state); - clk_disable(priv->dev.ssp->clk); + ssp_save_state(priv->ssp, &priv->state); + ssp_disable(priv->ssp); + clk_disable(priv->ssp->clk); return 0; } @@ -264,10 +263,9 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) if (!cpu_dai->active) return 0; - clk_enable(priv->dev.ssp->clk); - ssp_restore_state(&priv->dev, &priv->state); - ssp_enable(&priv->dev); - + clk_enable(priv->ssp->clk); + ssp_restore_state(priv->ssp, &priv->state); + ssp_enable(priv->ssp); return 0; } @@ -280,10 +278,9 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai) * ssp_set_clkdiv - set SSP clock divider * @div: serial clock rate divider */ -static void ssp_set_scr(struct ssp_dev *dev, u32 div) +static void ssp_set_scr(struct ssp_device *ssp, u32 div) { - struct ssp_device *ssp = dev->ssp; - u32 sscr0 = ssp_read_reg(dev->ssp, SSCR0) & ~SSCR0_SCR; + u32 sscr0 = ssp_read_reg(ssp, SSCR0) & ~SSCR0_SCR; ssp_write_reg(ssp, SSCR0, (sscr0 | SSCR0_SerClkDiv(div))); } @@ -295,7 +292,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, int clk_id, unsigned int freq, int dir) { struct ssp_priv *priv = cpu_dai->private_data; - struct ssp_device *ssp = priv->dev.ssp; + struct ssp_device *ssp = priv->ssp; int val; u32 sscr0 = ssp_read_reg(ssp, SSCR0) & @@ -326,7 +323,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, break; case PXA_SSP_CLK_AUDIO: priv->sysclk = 0; - ssp_set_scr(&priv->dev, 1); + ssp_set_scr(priv->ssp, 1); sscr0 |= SSCR0_ACS; break; default: @@ -336,11 +333,11 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai, /* The SSP clock must be disabled when changing SSP clock mode * on PXA2xx. On PXA3xx it must be enabled when doing so. */ if (!cpu_is_pxa3xx()) - clk_disable(priv->dev.ssp->clk); + clk_disable(priv->ssp->clk); val = ssp_read_reg(ssp, SSCR0) | sscr0; ssp_write_reg(ssp, SSCR0, val); if (!cpu_is_pxa3xx()) - clk_enable(priv->dev.ssp->clk); + clk_enable(priv->ssp->clk); return 0; } @@ -352,7 +349,7 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, int div_id, int div) { struct ssp_priv *priv = cpu_dai->private_data; - struct ssp_device *ssp = priv->dev.ssp; + struct ssp_device *ssp = priv->ssp; int val; switch (div_id) { @@ -387,7 +384,7 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, ssp_write_reg(ssp, SSACD, val); break; case PXA_SSP_DIV_SCR: - ssp_set_scr(&priv->dev, div); + ssp_set_scr(ssp, div); break; default: return -ENODEV; @@ -403,7 +400,7 @@ static int pxa_ssp_set_dai_pll(struct snd_soc_dai *cpu_dai, int pll_id, unsigned int freq_in, unsigned int freq_out) { struct ssp_priv *priv = cpu_dai->private_data; - struct ssp_device *ssp = priv->dev.ssp; + struct ssp_device *ssp = priv->ssp; u32 ssacd = ssp_read_reg(ssp, SSACD) & ~0x70; #if defined(CONFIG_PXA3xx) @@ -472,7 +469,7 @@ static int pxa_ssp_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, unsigned int mask, int slots) { struct ssp_priv *priv = cpu_dai->private_data; - struct ssp_device *ssp = priv->dev.ssp; + struct ssp_device *ssp = priv->ssp; u32 sscr0; sscr0 = ssp_read_reg(ssp, SSCR0) & ~SSCR0_SlotsPerFrm(7); @@ -494,7 +491,7 @@ static int pxa_ssp_set_dai_tristate(struct snd_soc_dai *cpu_dai, int tristate) { struct ssp_priv *priv = cpu_dai->private_data; - struct ssp_device *ssp = priv->dev.ssp; + struct ssp_device *ssp = priv->ssp; u32 sscr1; sscr1 = ssp_read_reg(ssp, SSCR1); @@ -516,7 +513,7 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) { struct ssp_priv *priv = cpu_dai->private_data; - struct ssp_device *ssp = priv->dev.ssp; + struct ssp_device *ssp = priv->ssp; u32 sscr0; u32 sscr1; u32 sspsp; @@ -622,7 +619,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; struct ssp_priv *priv = cpu_dai->private_data; - struct ssp_device *ssp = priv->dev.ssp; + struct ssp_device *ssp = priv->ssp; int dma = 0, chn = params_channels(params); u32 sscr0; u32 sspsp; @@ -735,12 +732,12 @@ static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd, struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; int ret = 0; struct ssp_priv *priv = cpu_dai->private_data; - struct ssp_device *ssp = priv->dev.ssp; + struct ssp_device *ssp = priv->ssp; int val; switch (cmd) { case SNDRV_PCM_TRIGGER_RESUME: - ssp_enable(&priv->dev); + ssp_enable(ssp); break; case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: val = ssp_read_reg(ssp, SSCR1); @@ -759,7 +756,7 @@ static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd, else val |= SSCR1_RSRE; ssp_write_reg(ssp, SSCR1, val); - ssp_enable(&priv->dev); + ssp_enable(ssp); break; case SNDRV_PCM_TRIGGER_STOP: val = ssp_read_reg(ssp, SSCR1); @@ -770,7 +767,7 @@ static int pxa_ssp_trigger(struct snd_pcm_substream *substream, int cmd, ssp_write_reg(ssp, SSCR1, val); break; case SNDRV_PCM_TRIGGER_SUSPEND: - ssp_disable(&priv->dev); + ssp_disable(ssp); break; case SNDRV_PCM_TRIGGER_PAUSE_PUSH: val = ssp_read_reg(ssp, SSCR1); @@ -800,8 +797,8 @@ static int pxa_ssp_probe(struct platform_device *pdev, if (!priv) return -ENOMEM; - priv->dev.ssp = ssp_request(dai->id + 1, "SoC audio"); - if (priv->dev.ssp == NULL) { + priv->ssp = ssp_request(dai->id + 1, "SoC audio"); + if (priv->ssp == NULL) { ret = -ENODEV; goto err_priv; } @@ -819,7 +816,7 @@ static void pxa_ssp_remove(struct platform_device *pdev, struct snd_soc_dai *dai) { struct ssp_priv *priv = dai->private_data; - ssp_free(priv->dev.ssp); + ssp_free(priv->ssp); } #define PXA_SSP_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\ -- 1.6.0.4

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[alsa-devel] 6-channel output and i2s
by Jon Smirl 29 Apr '09

29 Apr '09

My hardware was has three stereo i2s lines. These three lines are all tied to the same external clock so they are synchronized. Because there are three i2s lines, each with it's own DMA queue and FIFO, I need three independent output buffers. What is the correct way to set this up in ALSA? -- Jon Smirl jonsmirl(a)gmail.com

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[alsa-devel] Detect when sound card is in use
by Claus Ilginnis 29 Apr '09

29 Apr '09

Hi everyone, I have a problem and I dont't know where to ask my questions anymore, so I hope this mailing list can help me make progress with my problem.... I have a speaker set connected to an USB-Power-Plug (by which I can turn the speakers on and off ) Now I want to detect, when the soundcard is in use and when not. (to turn the speakers on when needed...) (I am a C++ programmer so I can mess some things up) So my question is: Which way is best to detect wheather my sound card is in use or not ? Is there a general overview/documentation of the sound kernel modules which describes the general relationship between them ? What would you do ? PLEASE help ! - Mit freundlichen Grüßen / Best regards Claus Ilginnis

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[alsa-devel] Getting the correct capture mixer
by Ludovico Cavedon 29 Apr '09

29 Apr '09

Hi, I am working on fixing some audio related bugs of a VoIP application (qutecom) which, like most VoIP applications, allows the user to change the playback and recording volume. For playback volume I am opening the "Master" mixer element. However I am not able to figure out how the application should get the correct mixer for playback. Is there any rule? For example, my sound card has two capture volumes, one "Int Mic" and one "Ext Mic". I guess I should choose on the basis of what is the user-selected mic. In fact there are also two mutually exclusive switches "IntMic" and "ExtMic". But how should the application understand that? Here part of the output of amixer, my sound card is "Intel [HDA Intel], device 0: CONEXANT Analog [CONEXANT Analog]" ---------------------------- Simple mixer control 'Ext Mic',0 Capabilities: pvolume cvolume pswitch cswitch Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 20 Capture 0 - 23 Front Left: Playback 0 [0%] [-30.00dB] [off] Capture 0 [0%] [0.00dB] [off] Front Right: Playback 0 [0%] [-30.00dB] [off] Capture 0 [0%] [0.00dB] [off] Simple mixer control 'ExtMic',0 Capabilities: cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Capture channels: Mono Mono: Capture [off] Simple mixer control 'Int Mic',0 Capabilities: pvolume cvolume pswitch cswitch Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: Playback 0 - 20 Capture 0 - 23 Front Left: Playback 0 [0%] [-30.00dB] [off] Capture 0 [0%] [0.00dB] [off] Front Right: Playback 0 [0%] [-30.00dB] [off] Capture 0 [0%] [0.00dB] [off] Simple mixer control 'IntMic',0 Capabilities: cswitch cswitch-joined cswitch-exclusive Capture exclusive group: 0 Capture channels: Mono Mono: Capture [on] ------------------------ As an aside, it looks like that the capture volume has no effect, as well all as the cswitches and pswitches of "Int Mic" and "Ext Mic". Thanks, Ludovico

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