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Re: [alsa-devel] [Alsa-user] looking for advice on buying a card
by Dmitri Seletski 02 May '09

02 May '09

Hi Gary. Well, thank you for the idea, but i somehow not comfortable with it. I have to buy this thing, even if it's inexpensive, i will have to pay good amount of money for delivery. I am looking around 20 euro, not counting price for device itself. Than again, i have to buy new card and pay for delivery there. To make things worse, none xfi sound cards are no longer produced by creative labs and i really don't like idea of second hand sound cards(even though you have to be very skillful to break one of those). So it simply makes sense to pay a bit extra(i was loosing those money with that pcie to pci extentsion and its delivery anyway) but to get that card I was referring before, Asus Xonar DX I know it's supported by alsa, because of this: http://www.alsa-project.org/main/index.php/Matrix:Vendor-Asus I am just double checking, since this specific card is really promising one and i really feel like buying one. But again, i am opened for other options. Thanks a lot for your thoughts. I have kinda got used to USB cards by now myself and all those troubles they cause :) gary wrote: > Having spent an ungodly obscene amount of time trying to get two > different USB audio cards to work in 64 bit opensuse, I suggest not > trying to be a pioneer. I solved my need for multiple sound cards > (exceeding the number of PCI slots I have) by using a PCI bridge. > > For one soundcard, you can try thing: > http://www.siig.com/ViewProduct.aspx?pn=JJ-000111-S1 > It doesn't list linux in support unfortunately. However, the latest > linux kernal might recognize the card. Then pick a soundcard from the > list of supported devices. > > For my particular situation, I bought a Magma PCI expander on ebay. I > plugged it in and it worked with the soundcards I had on hand (Diamond > 5.1 using C-Media chipset). With the shipping, it worked out to be > about $39 a PCI slot. [My time, about 10 minutes, not counting > searching ebay.] Thus the price for the SIIG seems reasonable if you > only need to run one card. However, I can't vouch it will work. The > Magma boxes do work, but are obscenely expensive new and bulky. > > > > Dmitri Seletski wrote: >> Hello Guys >> I am thinking to buy pci express sound card. Nothing fancy, just >> something with dolby surround 7.1 and decent sound quality, rather >> inexpensive, i am not audiophile. >> >> I have noticed that there are not that many sound cards out there which >> have descent support in Linux(unlike many forums claim). I have had >> MAJOR problems with pci xfi card from creative labs(fatality gamers >> edition), my buddy still has a lot of problem with it, like major issues >> with input. No mic - no skype or other voip, you know yourself... >> >> I have found source of asus xonar dx(namely ebay), it is not that >> expensive compared to many other pci express cards out there, still is >> expensive enough, from alsa matrix it appears to be quite descent. >> Question is, is it worth buying? Is it painful to get it >> running?(probably nothing unusual, matter of loading module), does it >> work fully(does microphone work, does surround sound 7.1 work?)? >> Is there something better for similar money?(EUR 73 or £65 or $97) >> Please do not propose incompletely supported cards(while i wouldn't mind >> submitting issues to mail lists if i come across bugs, i wouldn't want >> product that doesn't work) >> >> I appreciate any comments. >> >> Thank you in advance. >> Dmitri >> >> ------------------------------------------------------------------------------ >> >> Register Now & Save for Velocity, the Web Performance & Operations >> Conference from O'Reilly Media. Velocity features a full day of >> expert-led, hands-on workshops and two days of sessions from industry >> leaders in dedicated Performance & Operations tracks. Use code >> vel09scf and Save an extra 15% before 5/3. >> http://p.sf.net/sfu/velocityconf >> _______________________________________________ >> Alsa-user mailing list >> Alsa-user(a)lists.sourceforge.net >> https://lists.sourceforge.net/lists/listinfo/alsa-user >> >> >> ------------------------------------------------------------------------ >> >> >> No virus found in this incoming message. >> Checked by AVG - www.avg.com Version: 8.0.238 / Virus Database: >> 270.12.13/2091 - Release Date: 05/01/09 17:52:00 >> > ------------------------------------------------------------------------ > > > No virus found in this outgoing message. > Checked by AVG - www.avg.com > Version: 8.0.238 / Virus Database: 270.12.15/2093 - Release Date: 05/02/09 14:23:00 > >

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[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

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Re: [alsa-devel] errors in wm8990 codec driver
by Park Jinyoung 01 May '09

01 May '09

Hello, I made wm8990 patch file on your request. The patch file was attached in this. I use my company email, because I have no other method for file attach in my company. Thanks. 2009/4/30 Mark Brown <broonie(a)sirena.org.uk>: > On Thu, Apr 30, 2009 at 02:03:03PM +0900, Jinyoung Park wrote: >> 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... > > This all looks reasonable - could you please supply a Signed-off-by for > your patch? It may also help to use git send-email to send your patch > or attach it, gmail tends to mangle patches so that they can't be > applied and it looks like some of that may have happened here, though > I've not actually tried applying it. >

2 1

[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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[alsa-devel] No sound on Acer Aspire 6935G
by Karthik Ramgopal 29 Apr '09

29 Apr '09

Hi, I am using Ubuntu 9.04 Jaunty Jackalope with the latest alsa-driver snapshot provided by Takashi. The sound system does not work without a hda-verb command manually executed every time the system starts up. The command I execute is: hda-verb /dev/snd/hwC0D0 0x15 SET_EAPD_BTLENABLE 2 If I do this I get audio out. But my internal mic does not work and nor does jack sensing. When I insert a device into the headphone jack the speakers do not get muted and there is also no sound from the headphones themselves. Even an external mic/line-in device does not work. I have attached the output of alsa-info.sh here.. Please go through it and try to solve the problem. Thanks and Regards, Karthik

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