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

[alsa-devel] [PATCH1/2] hda: Added mono_out_pin to autoconfig
by Matthew Ranostay 24 Jan '08

24 Jan '08

Added a mono_out_pin field to autocfg struct, and code to parse for the mono_out_line. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r 5492010913f0 pci/hda/hda_codec.c --- a/pci/hda/hda_codec.c Tue Jan 22 15:37:13 2008 +0100 +++ b/pci/hda/hda_codec.c Wed Jan 23 10:55:11 2008 -0500 @@ -769,7 +769,7 @@ get_alloc_amp_hash(struct hda_codec *cod /* * query AMP capabilities for the given widget and direction */ -static u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction) +u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction) { struct hda_amp_info *info; @@ -2708,6 +2708,10 @@ int snd_hda_parse_pin_def_config(struct case AC_JACK_LINE_OUT: seq = get_defcfg_sequence(def_conf); assoc = get_defcfg_association(def_conf); + + if (!(wid_caps & AC_WCAP_STEREO)) + if (!cfg->mono_out_pin) + cfg->mono_out_pin = nid; if (!assoc) continue; if (!assoc_line_out) @@ -2856,6 +2860,7 @@ int snd_hda_parse_pin_def_config(struct cfg->hp_outs, cfg->hp_pins[0], cfg->hp_pins[1], cfg->hp_pins[2], cfg->hp_pins[3], cfg->hp_pins[4]); + snd_printd(" mono: mono_out=0x%x\n", cfg->mono_out_pin); snd_printd(" inputs: mic=0x%x, fmic=0x%x, line=0x%x, fline=0x%x," " cd=0x%x, aux=0x%x\n", cfg->input_pins[AUTO_PIN_MIC], diff -r 5492010913f0 pci/hda/hda_local.h --- a/pci/hda/hda_local.h Tue Jan 22 15:37:13 2008 +0100 +++ b/pci/hda/hda_local.h Wed Jan 23 10:56:40 2008 -0500 @@ -332,6 +332,7 @@ struct auto_pin_cfg { hda_nid_t input_pins[AUTO_PIN_LAST]; hda_nid_t dig_out_pin; hda_nid_t dig_in_pin; + hda_nid_t mono_out_pin; }; #define get_defcfg_connect(cfg) \ @@ -374,6 +375,7 @@ static inline u32 get_wcaps(struct hda_c return codec->wcaps[nid - codec->start_nid]; } +u32 query_amp_caps(struct hda_codec *codec, hda_nid_t nid, int direction); int snd_hda_override_amp_caps(struct hda_codec *codec, hda_nid_t nid, int dir, unsigned int caps);

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[alsa-devel] Fixing my sound driver
by Adrian McMenamin 24 Jan '08

24 Jan '08

I am still trying to find out what is wrong with my sound driver - for the Sega Dreamcast (sound/sh/aica.c). This works perfectly with the OSS emulation layer but will not work with the ALSA layer per se. As this shows - there is something wrong with how it interacts with the PCM layer of alsa-lib: $ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: AICA [AICA], device 0: AICA PCM [AICA PCM] Subdevices: 1/1 Subdevice #0: subdevice #0 $ $ $ aplay -L null Discard all samples (playback) or generate zero samples (capture) I have no reason to think that my build of alsa-lib is wrong - both the automagic buildroot build and my own by-hand build behave the same way. So therefore I assume it must be something in the driver?! Some questions: The driver implements nothing for capture (the AICA device is pure playback) - could that have an effect? What information must the driver pass to the PCM layer? What could make the alsa-lib layer detect the card but then report "null" is the only PCM device? Adrian

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[alsa-devel] Specific question about ICE1724 + AK4114
by Pavel Hofman 24 Jan '08

24 Jan '08

Hello, I would like to add support for automatic detection of incoming SPDIF sample rate for ICE cards with AK4114 DIR, specifically ESI Juli and Prodigy192 with MI/IODI/O card. Juli windows drivers support the detection, Prodigy drivers don't (broken in some other way too), but the hardware is basically identical. AK4114 support detection by comparing the recovered clock from SPDIF with another clock of 11, 12 or 24MHz, supplied by a crystal, or externally. In both cards this functionality seems to be provided by Xilinx XC9536 CPLD, together with PLL frequency multiplier MK73-1. It all fits together, the CPLD provides the freqency division for PLL feedback of the multiplier. External reference clock input of AK4114 is connected to CPLD, as well as setup pins XTL0/1. Output frequency of MK73-1 specified in datasheet is in the range of 10-50MHz Here is my question: For the PLL multiplier to work, its datasheet says it must be fed with input pulses of 20-50kHz. This frequency could be easily provided by ICE1724, BUT in master mode only. When in slave mode (which is the case for active SPDIF input - clock provided by AK4114), all the clocks are derived from the external clock. This signal obviously cannot be used as reference for the sample rate detection in AK4114 - i.e. basically of itself. According to the ICE1724 datasheet, only I2C and MIDI clocks are derived from the 48kHz mode crystal in slave mode. The datasheed says I2C clock is output only during I2C read/write activity, MIDI clock is not directly output at all. The cards have no other visible crystals/clock modules apart of those two connected directly to ICE1724. I could not trace any connections from the actual crystals to CPLD. Now here I am, absolutely wondering where the CPLD gets its low frequency stable signal for the PLL multiplier, to provide reference clock for SPDIF rate detection. I am afraid without this knowledge I cannot make the detection work in the alsa driver. Hopefully someone experienced in this area will throw a ray of light upon this mystery. Thanks in advance for any ideas, information, suggestions. Best regards, Pavel Hofman.

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[alsa-devel] [PATCH2/2] hda: Add dynamic mono mixer support for STAC92xx codecs
by Matthew Ranostay 23 Jan '08

23 Jan '08

Allows for dynamically creating mono out mixer controls and well as mono mux controls. Signed-off-by: Matthew Ranostay <mranostay(a)embeddedalley.com> --- diff -r 5492010913f0 pci/hda/patch_sigmatel.c --- a/pci/hda/patch_sigmatel.c Tue Jan 22 15:37:13 2008 +0100 +++ b/pci/hda/patch_sigmatel.c Tue Jan 22 21:22:52 2008 -0500 @@ -756,7 +756,6 @@ static struct snd_kcontrol_new stac92hd7 static struct snd_kcontrol_new stac92hd71bxx_analog_mixer[] = { STAC_INPUT_SOURCE(2), - STAC_MONO_MUX, HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), @@ -768,15 +767,12 @@ static struct snd_kcontrol_new stac92hd7 HDA_CODEC_MUTE("Analog Loopback 1", 0x17, 0x3, HDA_INPUT), HDA_CODEC_MUTE("Analog Loopback 2", 0x17, 0x4, HDA_INPUT), - - HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x14, 0x1, 0, HDA_INPUT), { } /* end */ }; static struct snd_kcontrol_new stac92hd71bxx_mixer[] = { STAC_INPUT_SOURCE(2), STAC_ANALOG_LOOPBACK(0xFA0, 0x7A0, 2), - STAC_MONO_MUX, HDA_CODEC_VOLUME_IDX("Capture Volume", 0x0, 0x1c, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x0, 0x1c, 0x0, HDA_OUTPUT), @@ -785,8 +781,6 @@ static struct snd_kcontrol_new stac92hd7 HDA_CODEC_VOLUME_IDX("Capture Volume", 0x1, 0x1d, 0x0, HDA_OUTPUT), HDA_CODEC_MUTE_IDX("Capture Switch", 0x1, 0x1d, 0x0, HDA_OUTPUT), HDA_CODEC_VOLUME_IDX("Capture Mux Volume", 0x1, 0x1b, 0x0, HDA_OUTPUT), - - HDA_CODEC_MUTE_MONO("Mono Playback Switch", 0x14, 0x1, 0, HDA_INPUT), { } /* end */ }; @@ -1157,7 +1151,7 @@ static struct snd_pci_quirk stac9200_cfg static unsigned int ref925x_pin_configs[8] = { 0x40c003f0, 0x424503f2, 0x01813022, 0x02a19021, - 0x90a70320, 0x02214210, 0x400003f1, 0x9033032e, + 0x90a70320, 0x02214210, 0x01019020, 0x9033032e, }; static unsigned int stac925x_MA6_pin_configs[8] = { @@ -1561,7 +1555,7 @@ static struct snd_pci_quirk stac927x_cfg static unsigned int ref9205_pin_configs[12] = { 0x40000100, 0x40000100, 0x01016011, 0x01014010, - 0x01813122, 0x01a19021, 0x40000100, 0x40000100, + 0x01813122, 0x01a19021, 0x01019020, 0x40000100, 0x90a000f0, 0x90a000f0, 0x01441030, 0x01c41030 }; @@ -2018,6 +2012,7 @@ enum { enum { STAC_CTL_WIDGET_VOL, STAC_CTL_WIDGET_MUTE, + STAC_CTL_WIDGET_MONO_MUX, STAC_CTL_WIDGET_IO_SWITCH, STAC_CTL_WIDGET_CLFE_SWITCH }; @@ -2025,6 +2020,7 @@ static struct snd_kcontrol_new stac92xx_ static struct snd_kcontrol_new stac92xx_control_templates[] = { HDA_CODEC_VOLUME(NULL, 0, 0, 0), HDA_CODEC_MUTE(NULL, 0, 0, 0), + STAC_MONO_MUX, STAC_CODEC_IO_SWITCH(NULL, 0), STAC_CODEC_CLFE_SWITCH(NULL, 0), }; @@ -2388,7 +2384,9 @@ static int stac92xx_auto_create_mono_out mono_mux->items[mono_mux->num_items].index = i; mono_mux->num_items++; } - return 0; + + return stac92xx_add_control(spec, STAC_CTL_WIDGET_MONO_MUX, + "Mono Mux", spec->mono_nid); } /* labels for dmic mux inputs */ @@ -2569,6 +2567,50 @@ static int stac92xx_parse_auto_config(st sizeof(spec->autocfg.hp_pins)); spec->autocfg.line_outs = spec->autocfg.hp_outs; hp_speaker_swap = 1; + } + if (spec->autocfg.mono_out_pin) { + int dir = (get_wcaps(codec, spec->autocfg.mono_out_pin) + & AC_WCAP_OUT_AMP) ? HDA_OUTPUT : HDA_INPUT; + u32 caps = query_amp_caps(codec, + spec->autocfg.mono_out_pin, dir); + hda_nid_t conn_list[1]; + + /* get the mixer node and then the mono mux if it exists */ + if (snd_hda_get_connections(codec, + spec->autocfg.mono_out_pin, conn_list, 1) && + snd_hda_get_connections(codec, conn_list[0], + conn_list, 1)) { + + int wcaps = get_wcaps(codec, conn_list[0]); + int wid_type = (wcaps & AC_WCAP_TYPE) + >> AC_WCAP_TYPE_SHIFT; + /* LR swap check, some stac925x have a mux that + * changes the DACs output path instead of the + * mono-mux path. + */ + if (wid_type == AC_WID_AUD_SEL && + !(wcaps & AC_WCAP_LR_SWAP)) + spec->mono_nid = conn_list[0]; + } + /* all mono outs have a least a mute/unmute switch */ + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_MUTE, + "Mono Playback Switch", + HDA_COMPOSE_AMP_VAL(spec->autocfg.mono_out_pin, + 1, 0, dir)); + if (err < 0) + return err; + /* check to see if there is volume support for the amp */ + if ((caps & AC_AMPCAP_NUM_STEPS) >> AC_AMPCAP_NUM_STEPS_SHIFT) { + err = stac92xx_add_control(spec, STAC_CTL_WIDGET_VOL, + "Mono Playback Volume", + HDA_COMPOSE_AMP_VAL(spec->autocfg.mono_out_pin, + 1, 0, dir)); + if (err < 0) + return err; + } + + stac92xx_auto_set_pinctl(codec, spec->autocfg.mono_out_pin, + AC_PINCTL_OUT_EN); } if ((err = stac92xx_add_dyn_out_pins(codec, &spec->autocfg)) < 0) @@ -3317,7 +3359,6 @@ again: spec->gpio_mask = spec->gpio_data = 0x00000001; /* GPIO0 High = EAPD */ - spec->mono_nid = 0x15; spec->mux_nids = stac92hd71bxx_mux_nids; spec->adc_nids = stac92hd71bxx_adc_nids; spec->dmic_nids = stac92hd71bxx_dmic_nids;

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[alsa-devel] problems with Intel Corporation 82801H (ICH8 Family) HD Audio Controller (rev 03)
by forwind 23 Jan '08

23 Jan '08

Hi folks, I am having difficulty getting a friends on-board speakers working on her laptop. After some reading of forums I'm not sure whether there is a solution for this or not ? The setup and details can be read below. The headphones work fine but the speakers are mute even though nothing is muted. I installed from source alsa version 1.0.15rc3 but to no avail. ALSA Audio Debug v0.1.0 - Thu Jan 17 23:21:23 GMT 2008 http://alsa.opensrc.org/aadebug http://www.gnu.org/licenses/gpl.txt Kernel ---------------------------------------------------- Linux laura-laptop 2.6.22-14-rt #1 SMP PREEMPT RT Tue Dec 18 10:01:34 UTC 2007 i686 GNU/Linux Loaded Modules -------------------------------------------- snd_rtctimer 4384 1 snd_hda_intel 291616 1 snd_pcm_oss 43264 0 snd_mixer_oss 17664 1 snd_pcm_oss snd_pcm 80132 2 snd_hda_intel,snd_pcm_oss snd_page_alloc 11400 2 snd_hda_intel,snd_pcm snd_hwdep 10500 1 snd_hda_intel snd_seq_dummy 4740 0 snd_seq_oss 33792 0 snd_seq_midi 9600 0 snd_rawmidi 25728 1 snd_seq_midi snd_seq_midi_event 8576 2 snd_seq_oss,snd_seq_midi snd_seq 54128 7 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_seq_midi_event snd_timer 24324 3 snd_rtctimer,snd_pcm,snd_seq snd_seq_device 9228 5 snd_seq_dummy,snd_seq_oss,snd_seq_midi,snd_rawmidi,snd_seq snd 55428 13 snd_hda_intel,snd_pcm_oss,snd_mixer_oss,snd_pcm,snd_hwdep,snd_seq_oss,snd_rawmidi,snd_seq,snd_timer,snd_seq_device Modprobe Conf --------------------------------------------- Warning: module config file does not exist This means any kernel modules will not be auto loaded See your linux distro docs on how to create this file Proc Asound ----------------------------------------------- Advanced Linux Sound Architecture Driver Version 1.0.15rc3. 0 [Intel ]: HDA-Intel - HDA Intel HDA Intel at 0xfc200000 irq 22 2: : timer 3: : sequencer 4: [ 0- 2]: digital audio capture 5: [ 0- 1]: digital audio playback 6: [ 0- 0]: digital audio playback 7: [ 0- 0]: digital audio capture 8: [ 0- 1]: hardware dependent 9: [ 0- 0]: hardware dependent 10: [ 0] : control 00-01: HDA Codec 1 00-00: HDA Codec 0 00-02: ALC883 Analog : ALC883 Analog : capture 2 00-01: ALC883 Digital : ALC883 Digital : playback 1 00-00: ALC883 Analog : ALC883 Analog : playback 1 : capture 2 Client info cur clients : 4 peak clients : 4 max clients : 192 Client 0 : "System" [Kernel] Port 0 : "Timer" (Rwe-) Port 1 : "Announce" (R-e-) Connecting To: 15:0 Client 14 : "Midi Through" [Kernel] Port 0 : "Midi Through Port-0" (RWe-) Client 15 : "OSS sequencer" [Kernel] Port 0 : "Receiver" (-we-) Connected From: 0:1 Client 128 : "TiMidity" [User] Port 0 : "TiMidity port 0" (-We-) Port 1 : "TiMidity port 1" (-We-) Port 2 : "TiMidity port 2" (-We-) Port 3 : "TiMidity port 3" (-We-) Output pool : Pool size : 500 Cells in use : 0 Peak cells in use : 0 Alloc success : 0 Alloc failures : 0 Input pool : Pool size : 1000 Cells in use : 0 Peak cells in use : 0 Alloc success : 0 Alloc failures : 0 Dev Snd --------------------------------------------------- controlC0 hwC0D0 hwC0D1 pcmC0D0c pcmC0D0p pcmC0D1p pcmC0D2c seq timer CPU ------------------------------------------------------- model name : Intel(R) Core(TM)2 Duo CPU T7300 @ 2.00GHz cpu MHz : 2001.000 model name : Intel(R) Core(TM)2 Duo CPU T7300 @ 2.00GHz cpu MHz : 2001.000 RAM ------------------------------------------------------- MemTotal: 1028132 kB SwapTotal: 3028212 kB Hardware -------------------------------------------------- 00:00.0 Host bridge: Intel Corporation Mobile PM965/GM965/GL960 Memory Controller Hub (rev 03) does this sound familiar or is everybody blue in the face from hearing about this 'bug' :)? Best regards, Conor

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[alsa-devel] The state of Digigram cards
by Mark Hills 23 Jan '08

23 Jan '08

Hi, I've been trying to use Digigram cards with ALSA using the drivers specific to these cards: * VX222 (detected as vx222/Old) * miXArt 8 Since about 1 year ago, I've tried both cards in various machines with various software configurations, and my experiences have ranged from complete system lockups to problems opening the device. Using more than one CPU seems to be a certain way to get system lockups. Is anybody actually using the Digigram drivers? Searching the various archives shows interest in the cards from a number of years ago. Is it possible that more recent developments to ALSA and the kernel cause the drivers to be unstable? Thanks, Mark

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[alsa-devel] S16_LE + softvol = bad
by Alexander E. Patrakov 23 Jan '08

23 Jan '08

Hello developers, some cards (most notably, Intel HD audio) have the PCM control as softvol, and default to S16_LE as a format for dmix. Since softvol is configured to attenuate signal, and not many people set this control to 100% (especially those who have no other volume controls), this effectively leaves less than 16 bits per sample. The card can surely produce sound of better quality in most cases, if one changes defaults.pcm.dmix.format in ~/.asoundrc. However, one should now search a lot of forums in order to know about this setting. Not everyone does this, and not everyone is even able to diagnose the "noisy sound in Linux" problem properly as "too few bits per sample in the default configuration". A much better situation would be if ALSA defaulted to good quality (i.e.: S24_LE or S32_LE on all cards that require softvol and support this sample format). Could you please change the files in /usr/share/alsa/cards accordingly? -- Alexander E. Patrakov

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Re: [alsa-devel] how do you people debug ALSA?
by mazarick＠bellsouth.net 23 Jan '08

23 Jan '08

"Mihaela Vitalariu" <admiralwolverine(a)gmail.com> wrote: > I meant what tools do you use, what Debuggers? > In my post i wasn't complaining about ALSA, i was just curios about the > debugging tools that you use, from a developer's standpoint... The group that I'm working with (to develop an alsa driver for the Gadget Labs soundcard) would be very interested in any input on this subject. The team is made up of skilled C/C++ coders, but there is very little to no experience with writing a linux hardware driver. It would be fine to say "vi is the development environment, and we sprinkle the code with printk statements and take them out when the soup's done". It would save us a lot of overhead in figuring out the best way to get on the same page with each other since it's a pretty new team. Any other words of wisdom from those who've been there would be very welcomed and appreciated. If you use KDE, Eclipse, or something else, that's fine too. Some particular questions (in no particular order): 1. Do you regularly use the "alsasound" kernel function during development so you can take the driver in and out of the kernel? If it's used, is is used from the beginning or in the last stages of development? (I'm not sure if it's alsasound, sndconfig, etc). 2. Are the process statements given in the 'How-To' (snd_printk, snd_assert, snd_BUG) the main ones or are there others that are 'pre-built' and widely useful? http://www.alsa-project.org/~tiwai/writing-an-alsa-driver/c1696.htm 3. Are there tools that are almost always used when writing a sound card driver like gdb, ddd, etc? 4. Since it's easy to do, is an interface into the proc file system useful from the start? http://www.alsa-project.org/~tiwai/writing-an-alsa-driver/c1493.htm 5. Is the alsa wiki on debugging the sum total of what's needed or is there more we don't know about? (and we don't know that much!!) http://www.alsa-project.org/main/index.php/Driver_Compilation Of course, I could bring up the current Hot and Burning question we have (they are more important now): http://wavepro-driver.svn.sourceforge.net/viewvc/wavepro-driver/ 1. We've having difficulty getting the Altera PLD to load. Any suggestions on how to do this, how people without cards can help, an easy way to know when you are successful, or an incremental approach to the problem? 2. The GL card is powered from an Altera PLD (FPGA) (it has similar function as the Envy ICE chip once it's loaded). For the card to function at all, the Altera PLD has to be loaded when power is applied to the computer. Should the 'load the Altera' function be part of the alsa driver, part of alsa firmware, or should we just load it up when the computer boots (not in alsa)? 3. We're planning on using FC8/PlanetCCRMA as a 'beta test' environment. I've just noticed the new FC8 specific instructions on how to overcome my many bad habits when fooling around with the kernel. I'm putting it here for reference so I don't forget it (like I just did when trying to figure out if it was alsasound, procsound, or whatever to unload the driver). http://fedoraproject.org/wiki/Docs/CustomKernel If you would like to just put your two cents worth in to the very small mailing list (for the next 5 days), send a note to: wavepro-driver-developers(a)lists.sourceforge.net (in 5 days, you'll have to be subscribed to send a note) -Mike Mazarick

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[alsa-devel] alsa-lib(1.0.12) crash
by Cliff Cai 23 Jan '08

23 Jan '08

Hi all, I'm experiencing weird crash of alsa-lib. the sound card has 16/24 bit ADC. The ADC works on 24 bit mode, hw->formats= SNDRV_PCM_FMTBIT_S32_LE,and use 'copy' playback to copy DMA buffer to user space's buffer. I ran: arecord -t wav -c 2 -f U8 -r 8000 -d 5 a.wav then arecord crash with following information: Data access misaligned address violation - Attempted misaligned data memory or data cache access. RETE: /* Maybe null pointer? */ RETN: /* unknown address */ RETX: [ /usr/lib/libasound.so.2 + 0x46da8 ] RETS: [ /usr/lib/libasound.so.2 + 0x46aec ] PC : [ /usr/lib/libasound.so.2 + 0x46da8 ] DCPLB_FAULT_ADDR: [ arecord + 0x2ac ] ICPLB_FAULT_ADDR: [ /usr/lib/libasound.so.2 + 0x46da8 ] Hardware Trace: 0 Target : { _trap_c + 0x0 } Source : { _exception_to_level5 + 0xb4 } 1 Target : { _exception_to_level5 + 0x0 } Source : { _ex_trap_c + 0x5c } 2 Target : { _ex_trap_c + 0x0 } Source : { _trap + 0x28 } 3 Target : { _trap + 0x0 } Source : [ /usr/lib/libasound.so.2 + 0x46da6 ] 4 Target : [ /usr/lib/libasound.so.2 + 0x46d9a ] Source : [ /usr/lib/libasound.so.2 + 0x46b54 ] 5 Target : [ /usr/lib/libasound.so.2 + 0x46b46 ] Source : [ /usr/lib/libasound.so.2 + 0x46b12 ] 6 Target : [ /usr/lib/libasound.so.2 + 0x46aec ] Source : [ /usr/lib/libasound.so.2 + 0x4d16a ] 7 Target : [ /usr/lib/libasound.so.2 + 0x4d166 ] Source : [ /usr/lib/libasound.so.2 + 0x4d17e ] 8 Target : [ /usr/lib/libasound.so.2 + 0x4d17c ] Source : [ /usr/lib/libasound.so.2 + 0x4d0b2 ] 9 Target : [ /usr/lib/libasound.so.2 + 0x4d09c ] Source : [ /usr/lib/libasound.so.2 + 0x278ec ] 10 Target : [ /usr/lib/libasound.so.2 + 0x278e4 ] Source : [ /usr/lib/libasound.so.2 + 0x46ae8 ] 11 Target : [ /usr/lib/libasound.so.2 + 0x46a74 ] Source : [ /usr/lib/libasound.so.2 + 0x28602 ] 12 Target : [ /usr/lib/libasound.so.2 + 0x285fa ] Source : [ /usr/lib/libasound.so.2 + 0x46eac ] 13 Target : [ /usr/lib/libasound.so.2 + 0x46e94 ] Source : [ /usr/lib/libasound.so.2 + 0x46e8a ] 14 Target : [ /usr/lib/libasound.so.2 + 0x46e88 ] Source : [ /usr/lib/libasound.so.2 + 0x46e38 ] 15 Target : [ /usr/lib/libasound.so.2 + 0x46e0e ] Source : [ /usr/lib/libasound.so.2 + 0x4d16a ] Then I modified the driver ,set the ADC to 16bit mode and set hw->formats= SNDRV_PCM_FMTBIT_S16_LE no crash any more. Any idea is appreciated. Thanks a lot in advance! Cliff _________________________________________________________________ 天凉了，添衣了，心动了，“七件”了 http://get.live.cn

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[alsa-devel] [PATCH] ASoC documentation updates
by Mark Brown 23 Jan '08

23 Jan '08

Update the ASoC documentation. Along with minor formatting and grammar cleanups it moves the ASoC overview into the present tense to reflect the fact that it has now been merged. Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- Documentation/sound/alsa/soc/DAI.txt | 6 +- Documentation/sound/alsa/soc/clocking.txt | 10 ++-- Documentation/sound/alsa/soc/codec.txt | 53 +++++++++++++------------- Documentation/sound/alsa/soc/dapm.txt | 51 +++++++++++++------------ Documentation/sound/alsa/soc/machine.txt | 12 +++--- Documentation/sound/alsa/soc/overview.txt | 42 +++++++++++---------- Documentation/sound/alsa/soc/platform.txt | 6 +- Documentation/sound/alsa/soc/pops_clicks.txt | 10 ++-- 8 files changed, 97 insertions(+), 93 deletions(-) diff --git a/Documentation/sound/alsa/soc/DAI.txt b/Documentation/sound/alsa/soc/DAI.txt index 3feeb9e..0ebd7ea 100644 --- a/Documentation/sound/alsa/soc/DAI.txt +++ b/Documentation/sound/alsa/soc/DAI.txt @@ -1,5 +1,5 @@ ASoC currently supports the three main Digital Audio Interfaces (DAI) found on -SoC controllers and portable audio CODECS today, namely AC97, I2S and PCM. +SoC controllers and portable audio CODECs today, namely AC97, I2S and PCM. AC97 @@ -25,7 +25,7 @@ left/right clock (LRC) synchronise the link. I2S is flexible in that either the controller or CODEC can drive (master) the BCLK and LRC clock lines. Bit clock usually varies depending on the sample rate and the master system clock (SYSCLK). LRCLK is the same as the sample rate. A few devices support separate -ADC and DAC LRCLK's, this allows for simultaneous capture and playback at +ADC and DAC LRCLKs, this allows for simultaneous capture and playback at different sample rates. I2S has several different operating modes:- @@ -35,7 +35,7 @@ I2S has several different operating modes:- o Left Justified - MSB is transmitted on transition of LRC. - o Right Justified - MSB is transmitted sample size BCLK's before LRC + o Right Justified - MSB is transmitted sample size BCLKs before LRC transition. PCM diff --git a/Documentation/sound/alsa/soc/clocking.txt b/Documentation/sound/alsa/soc/clocking.txt index 1493088..b130016 100644 --- a/Documentation/sound/alsa/soc/clocking.txt +++ b/Documentation/sound/alsa/soc/clocking.txt @@ -13,7 +13,7 @@ or SYSCLK). This audio master clock can be derived from a number of sources (e.g. crystal, PLL, CPU clock) and is responsible for producing the correct audio playback and capture sample rates. -Some master clocks (e.g. PLL's and CPU based clocks) are configurable in that +Some master clocks (e.g. PLLs and CPU based clocks) are configurable in that their speed can be altered by software (depending on the system use and to save power). Other master clocks are fixed at a set frequency (i.e. crystals). @@ -41,11 +41,11 @@ BCLK = LRC * x BCLK = LRC * Channels * Word Size This relationship depends on the codec or SoC CPU in particular. In general -it's best to configure BCLK to the lowest possible speed (depending on your -rate, number of channels and wordsize) to save on power. +it is best to configure BCLK to the lowest possible speed (depending on your +rate, number of channels and word size) to save on power. -It's also desirable to use the codec (if possible) to drive (or master) the -audio clocks as it's usually gives more accurate sample rates than the CPU. +It is also desirable to use the codec (if possible) to drive (or master) the +audio clocks as it usually gives more accurate sample rates than the CPU. diff --git a/Documentation/sound/alsa/soc/codec.txt b/Documentation/sound/alsa/soc/codec.txt index 1e766ad..1e95342 100644 --- a/Documentation/sound/alsa/soc/codec.txt +++ b/Documentation/sound/alsa/soc/codec.txt @@ -9,7 +9,7 @@ code should be added to the platform and machine drivers respectively. Each codec driver *must* provide the following features:- 1) Codec DAI and PCM configuration - 2) Codec control IO - using I2C, 3 Wire(SPI) or both API's + 2) Codec control IO - using I2C, 3 Wire(SPI) or both APIs 3) Mixers and audio controls 4) Codec audio operations @@ -19,7 +19,7 @@ Optionally, codec drivers can also provide:- 6) DAPM event handler. 7) DAC Digital mute control. -It's probably best to use this guide in conjunction with the existing codec +Its probably best to use this guide in conjunction with the existing codec driver code in sound/soc/codecs/ ASoC Codec driver breakdown @@ -27,8 +27,8 @@ ASoC Codec driver breakdown 1 - Codec DAI and PCM configuration ----------------------------------- -Each codec driver must have a struct snd_soc_codec_dai to define it's DAI and -PCM's capabilities and operations. This struct is exported so that it can be +Each codec driver must have a struct snd_soc_codec_dai to define its DAI and +PCM capabilities and operations. This struct is exported so that it can be registered with the core by your machine driver. e.g. @@ -67,18 +67,18 @@ EXPORT_SYMBOL_GPL(wm8731_dai); 2 - Codec control IO -------------------- -The codec can usually be controlled via an I2C or SPI style interface (AC97 -combines control with data in the DAI). The codec drivers will have to provide -functions to read and write the codec registers along with supplying a register -cache:- +The codec can usually be controlled via an I2C or SPI style interface +(AC97 combines control with data in the DAI). The codec drivers provide +functions to read and write the codec registers along with supplying a +register cache:- /* IO control data and register cache */ - void *control_data; /* codec control (i2c/3wire) data */ - void *reg_cache; + void *control_data; /* codec control (i2c/3wire) data */ + void *reg_cache; -Codec read/write should do any data formatting and call the hardware read write -below to perform the IO. These functions are called by the core and alsa when -performing DAPM or changing the mixer:- +Codec read/write should do any data formatting and call the hardware +read write below to perform the IO. These functions are called by the +core and ALSA when performing DAPM or changing the mixer:- unsigned int (*read)(struct snd_soc_codec *, unsigned int); int (*write)(struct snd_soc_codec *, unsigned int, unsigned int); @@ -131,7 +131,7 @@ Defines a stereo enumerated control 4 - Codec Audio Operations -------------------------- -The codec driver also supports the following alsa operations:- +The codec driver also supports the following ALSA operations:- /* SoC audio ops */ struct snd_soc_ops { @@ -142,15 +142,15 @@ struct snd_soc_ops { int (*prepare)(struct snd_pcm_substream *); }; -Please refer to the alsa driver PCM documentation for details. +Please refer to the ALSA driver PCM documentation for details. http://www.alsa-project.org/~iwai/writing-an-alsa-driver/c436.htm 5 - DAPM description. --------------------- -The Dynamic Audio Power Management description describes the codec's power -components, their relationships and registers to the ASoC core. Please read -dapm.txt for details of building the description. +The Dynamic Audio Power Management description describes the codec power +components and their relationships and registers to the ASoC core. +Please read dapm.txt for details of building the description. Please also see the examples in other codec drivers. @@ -158,8 +158,8 @@ Please also see the examples in other codec drivers. 6 - DAPM event handler ---------------------- This function is a callback that handles codec domain PM calls and system -domain PM calls (e.g. suspend and resume). It's used to put the codec to sleep -when not in use. +domain PM calls (e.g. suspend and resume). It is used to put the codec +to sleep when not in use. Power states:- @@ -175,13 +175,14 @@ Power states:- SNDRV_CTL_POWER_D3cold: /* Everything Off, without power */ -7 - Codec DAC digital mute control. ------------------------------------- -Most codecs have a digital mute before the DAC's that can be used to minimise -any system noise. The mute stops any digital data from entering the DAC. +7 - Codec DAC digital mute control +---------------------------------- +Most codecs have a digital mute before the DACs that can be used to +minimise any system noise. The mute stops any digital data from +entering the DAC. -A callback can be created that is called by the core for each codec DAI when the -mute is applied or freed. +A callback can be created that is called by the core for each codec DAI +when the mute is applied or freed. i.e. diff --git a/Documentation/sound/alsa/soc/dapm.txt b/Documentation/sound/alsa/soc/dapm.txt index ab0766f..c784a18 100644 --- a/Documentation/sound/alsa/soc/dapm.txt +++ b/Documentation/sound/alsa/soc/dapm.txt @@ -4,20 +4,20 @@ Dynamic Audio Power Management for Portable Devices 1. Description ============== -Dynamic Audio Power Management (DAPM) is designed to allow portable Linux devices -to use the minimum amount of power within the audio subsystem at all times. It -is independent of other kernel PM and as such, can easily co-exist with the -other PM systems. +Dynamic Audio Power Management (DAPM) is designed to allow portable +Linux devices to use the minimum amount of power within the audio +subsystem at all times. It is independent of other kernel PM and as +such, can easily co-exist with the other PM systems. -DAPM is also completely transparent to all user space applications as all power -switching is done within the ASoC core. No code changes or recompiling are -required for user space applications. DAPM makes power switching decisions based -upon any audio stream (capture/playback) activity and audio mixer settings -within the device. +DAPM is also completely transparent to all user space applications as +all power switching is done within the ASoC core. No code changes or +recompiling are required for user space applications. DAPM makes power +switching decisions based upon any audio stream (capture/playback) +activity and audio mixer settings within the device. -DAPM spans the whole machine. It covers power control within the entire audio -subsystem, this includes internal codec power blocks and machine level power -systems. +DAPM spans the whole machine. It covers power control within the entire +audio subsystem, this includes internal codec power blocks and machine +level power systems. There are 4 power domains within DAPM @@ -34,7 +34,7 @@ There are 4 power domains within DAPM Automatically set when mixer and mux settings are changed by the user. e.g. alsamixer, amixer. - 4. Stream domain - DAC's and ADC's. + 4. Stream domain - DACs and ADCs. Enabled and disabled when stream playback/capture is started and stopped respectively. e.g. aplay, arecord. @@ -51,7 +51,7 @@ widgets hereafter. Audio DAPM widgets fall into a number of types:- o Mixer - Mixes several analog signals into a single analog signal. - o Mux - An analog switch that outputs only 1 of it's inputs. + o Mux - An analog switch that outputs only one of many inputs. o PGA - A programmable gain amplifier or attenuation widget. o ADC - Analog to Digital Converter o DAC - Digital to Analog Converter @@ -78,14 +78,14 @@ parameters for stream name and kcontrols. 2.1 Stream Domain Widgets ------------------------- -Stream Widgets relate to the stream power domain and only consist of ADC's -(analog to digital converters) and DAC's (digital to analog converters). +Stream Widgets relate to the stream power domain and only consist of ADCs +(analog to digital converters) and DACs (digital to analog converters). Stream widgets have the following format:- SND_SOC_DAPM_DAC(name, stream name, reg, shift, invert), -NOTE: the stream name must match the corresponding stream name in your codecs +NOTE: the stream name must match the corresponding stream name in your codec snd_soc_codec_dai. e.g. stream widgets for HiFi playback and capture @@ -97,7 +97,7 @@ SND_SOC_DAPM_ADC("HiFi ADC", "HiFi Capture", REG, 2, 1), 2.2 Path Domain Widgets ----------------------- -Path domain widgets have a ability to control or effect the audio signal or +Path domain widgets have a ability to control or affect the audio signal or audio paths within the audio subsystem. They have the following form:- SND_SOC_DAPM_PGA(name, reg, shift, invert, controls, num_controls) @@ -149,7 +149,7 @@ SND_SOC_DAPM_MIC("Mic Jack", spitz_mic_bias), 2.4 Codec Domain ---------------- -The Codec power domain has no widgets and is handled by the codecs DAPM event +The codec power domain has no widgets and is handled by the codecs DAPM event handler. This handler is called when the codec powerstate is changed wrt to any stream event or by kernel PM events. @@ -158,8 +158,8 @@ stream event or by kernel PM events. ------------------- Sometimes widgets exist in the codec or machine audio map that don't have any -corresponding register bit for power control. In this case it's necessary to -create a virtual widget - a widget with no control bits e.g. +corresponding soft power control. In this case it is necessary to create +a virtual widget - a widget with no control bits e.g. SND_SOC_DAPM_MIXER("AC97 Mixer", SND_SOC_DAPM_NOPM, 0, 0, NULL, 0), @@ -172,13 +172,14 @@ subsystem individually with a call to snd_soc_dapm_new_control(). 3. Codec Widget Interconnections ================================ -Widgets are connected to each other within the codec and machine by audio -paths (called interconnections). Each interconnection must be defined in order -to create a map of all audio paths between widgets. +Widgets are connected to each other within the codec and machine by audio paths +(called interconnections). Each interconnection must be defined in order to +create a map of all audio paths between widgets. + This is easiest with a diagram of the codec (and schematic of the machine audio system), as it requires joining widgets together via their audio signal paths. -i.e. from the WM8731 codec's output mixer (wm8731.c) +e.g., from the WM8731 output mixer (wm8731.c) The WM8731 output mixer has 3 inputs (sources) diff --git a/Documentation/sound/alsa/soc/machine.txt b/Documentation/sound/alsa/soc/machine.txt index 72bd222..f370e7d 100644 --- a/Documentation/sound/alsa/soc/machine.txt +++ b/Documentation/sound/alsa/soc/machine.txt @@ -16,7 +16,7 @@ struct snd_soc_machine { int (*remove)(struct platform_device *pdev); /* the pre and post PM functions are used to do any PM work before and - * after the codec and DAI's do any PM work. */ + * after the codec and DAIs do any PM work. */ int (*suspend_pre)(struct platform_device *pdev, pm_message_t state); int (*suspend_post)(struct platform_device *pdev, pm_message_t state); int (*resume_pre)(struct platform_device *pdev); @@ -38,7 +38,7 @@ probe/remove are optional. Do any machine specific probe here. suspend()/resume() ------------------ The machine driver has pre and post versions of suspend and resume to take care -of any machine audio tasks that have to be done before or after the codec, DAI's +of any machine audio tasks that have to be done before or after the codec, DAIs and DMA is suspended and resumed. Optional. @@ -49,10 +49,10 @@ The machine specific audio operations can be set here. Again this is optional. Machine DAI Configuration ------------------------- -The machine DAI configuration glues all the codec and CPU DAI's together. It can +The machine DAI configuration glues all the codec and CPU DAIs together. It can also be used to set up the DAI system clock and for any machine related DAI initialisation e.g. the machine audio map can be connected to the codec audio -map, unconnnected codec pins can be set as such. Please see corgi.c, spitz.c +map, unconnected codec pins can be set as such. Please see corgi.c, spitz.c for examples. struct snd_soc_dai_link is used to set up each DAI in your machine. e.g. @@ -67,7 +67,7 @@ static struct snd_soc_dai_link corgi_dai = { .ops = &corgi_ops, }; -struct snd_soc_machine then sets up the machine with it's DAI's. e.g. +struct snd_soc_machine then sets up the machine with it's DAIs. e.g. /* corgi audio machine driver */ static struct snd_soc_machine snd_soc_machine_corgi = { @@ -110,4 +110,4 @@ details. Machine Controls ---------------- -Machine specific audio mixer controls can be added in the dai init function. \ No newline at end of file +Machine specific audio mixer controls can be added in the DAI init function. diff --git a/Documentation/sound/alsa/soc/overview.txt b/Documentation/sound/alsa/soc/overview.txt index c47ce95..1e4c6d3 100644 --- a/Documentation/sound/alsa/soc/overview.txt +++ b/Documentation/sound/alsa/soc/overview.txt @@ -1,25 +1,26 @@ ALSA SoC Layer ============== -The overall project goal of the ALSA System on Chip (ASoC) layer is to provide -better ALSA support for embedded system-on-chip processors (e.g. pxa2xx, au1x00, -iMX, etc) and portable audio codecs. Currently there is some support in the -kernel for SoC audio, however it has some limitations:- +The overall project goal of the ALSA System on Chip (ASoC) layer is to +provide better ALSA support for embedded system-on-chip processors (e.g. +pxa2xx, au1x00, iMX, etc) and portable audio codecs. Prior to the ASoC +subsystem there was some support in the kernel for SoC audio, however it +had some limitations:- - * Currently, codec drivers are often tightly coupled to the underlying SoC - CPU. This is not ideal and leads to code duplication i.e. Linux now has 4 - different wm8731 drivers for 4 different SoC platforms. + * Codec drivers were often tightly coupled to the underlying SoC + CPU. This is not ideal and leads to code duplication - for example, + Linux had different wm8731 drivers for 4 different SoC platforms. - * There is no standard method to signal user initiated audio events (e.g. + * There was no standard method to signal user initiated audio events (e.g. Headphone/Mic insertion, Headphone/Mic detection after an insertion event). These are quite common events on portable devices and often require machine specific code to re-route audio, enable amps, etc., after such an event. - * Current drivers tend to power up the entire codec when playing - (or recording) audio. This is fine for a PC, but tends to waste a lot of - power on portable devices. There is also no support for saving power via - changing codec oversampling rates, bias currents, etc. + * Drivers tended to power up the entire codec when playing (or + recording) audio. This is fine for a PC, but tends to waste a lot of + power on portable devices. There was also no support for saving + power via changing codec oversampling rates, bias currents, etc. ASoC Design @@ -31,12 +32,13 @@ features :- * Codec independence. Allows reuse of codec drivers on other platforms and machines. - * Easy I2S/PCM audio interface setup between codec and SoC. Each SoC interface - and codec registers it's audio interface capabilities with the core and are - subsequently matched and configured when the application hw params are known. + * Easy I2S/PCM audio interface setup between codec and SoC. Each SoC + interface and codec registers it's audio interface capabilities with the + core and are subsequently matched and configured when the application + hardware parameters are known. * Dynamic Audio Power Management (DAPM). DAPM automatically sets the codec to - it's minimum power state at all times. This includes powering up/down + its minimum power state at all times. This includes powering up/down internal power blocks depending on the internal codec audio routing and any active streams. @@ -45,16 +47,16 @@ features :- signals the codec when to change power states. * Machine specific controls: Allow machines to add controls to the sound card - (e.g. volume control for speaker amp). + (e.g. volume control for speaker amplifier). To achieve all this, ASoC basically splits an embedded audio system into 3 components :- * Codec driver: The codec driver is platform independent and contains audio - controls, audio interface capabilities, codec dapm definition and codec IO + controls, audio interface capabilities, codec DAPM definition and codec IO functions. - * Platform driver: The platform driver contains the audio dma engine and audio + * Platform driver: The platform driver contains the audio DMA engine and audio interface drivers (e.g. I2S, AC97, PCM) for that platform. * Machine driver: The machine driver handles any machine specific controls and @@ -81,4 +83,4 @@ machine.txt: Machine driver internals. pop_clicks.txt: How to minimise audio artifacts. -clocking.txt: ASoC clocking for best power performance. \ No newline at end of file +clocking.txt: ASoC clocking for best power performance. diff --git a/Documentation/sound/alsa/soc/platform.txt b/Documentation/sound/alsa/soc/platform.txt index d4678b4..b681d17 100644 --- a/Documentation/sound/alsa/soc/platform.txt +++ b/Documentation/sound/alsa/soc/platform.txt @@ -8,7 +8,7 @@ specific code. Audio DMA ========= -The platform DMA driver optionally supports the following alsa operations:- +The platform DMA driver optionally supports the following ALSA operations:- /* SoC audio ops */ struct snd_soc_ops { @@ -38,7 +38,7 @@ struct snd_soc_platform { struct snd_pcm_ops *pcm_ops; }; -Please refer to the alsa driver documentation for details of audio DMA. +Please refer to the ALSA driver documentation for details of audio DMA. http://www.alsa-project.org/~iwai/writing-an-alsa-driver/c436.htm An example DMA driver is soc/pxa/pxa2xx-pcm.c @@ -52,7 +52,7 @@ Each SoC DAI driver must provide the following features:- 1) Digital audio interface (DAI) description 2) Digital audio interface configuration 3) PCM's description - 4) Sysclk configuration + 4) SYSCLK configuration 5) Suspend and resume (optional) Please see codec.txt for a description of items 1 - 4. diff --git a/Documentation/sound/alsa/soc/pops_clicks.txt b/Documentation/sound/alsa/soc/pops_clicks.txt index 3371bd9..e1e74da 100644 --- a/Documentation/sound/alsa/soc/pops_clicks.txt +++ b/Documentation/sound/alsa/soc/pops_clicks.txt @@ -15,11 +15,11 @@ click every time a component power state is changed. Minimising Playback Pops and Clicks =================================== -Playback pops in portable audio subsystems cannot be completely eliminated atm, -however future audio codec hardware will have better pop and click suppression. -Pops can be reduced within playback by powering the audio components in a -specific order. This order is different for startup and shutdown and follows -some basic rules:- +Playback pops in portable audio subsystems cannot be completely eliminated +currently, however future audio codec hardware will have better pop and click +suppression. Pops can be reduced within playback by powering the audio +components in a specific order. This order is different for startup and +shutdown and follows some basic rules:- Startup Order :- DAC --> Mixers --> Output PGA --> Digital Unmute -- 1.5.3.8

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