January 2010 - Alsa-devel - mailman.alsa-project.org

[alsa-devel] [PATCH] add atomic operations for Blackfin parts
by Mike Frysinger 18 Jan '10

18 Jan '10

Signed-off-by: Mike Frysinger <vapier(a)gentoo.org> --- include/iatomic.h | 23 +++++++++++++++++++++++ 1 files changed, 23 insertions(+), 0 deletions(-) diff --git a/include/iatomic.h b/include/iatomic.h index 8f6ec22..e92dbfd 100644 --- a/include/iatomic.h +++ b/include/iatomic.h @@ -1079,6 +1079,29 @@ static __inline__ int atomic_sub_return(int i, volatile atomic_t *v) #endif /* __sh__ */ +#ifdef __bfin__ + +#include <bfin_fixed_code.h> + +typedef struct { volatile int counter; } atomic_t; + +#define ATOMIC_INIT(i) { (i) } + +#define atomic_read(v) ((v)->counter) +#define atomic_set(v,i) (((v)->counter) = (i)) +#define atomic_add(i,v) bfin_atomic_add32(&(v)->counter, i) +#define atomic_sub(i,v) bfin_atomic_sub32(&(v)->counter, i) +#define atomic_inc(v) bfin_atomic_inc32(&(v)->counter); +#define atomic_dec(v) bfin_atomic_dec32(&(v)->counter); + +#define mb() __asm__ __volatile__ ("" : : : "memory") +#define rmb() mb() +#define wmb() mb() + +#define IATOMIC_DEFINED 1 + +#endif /* __bfin__ */ + #ifndef IATOMIC_DEFINED /* * non supported architecture. -- 1.6.6

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[alsa-devel] ASoC updates for 2.6.34
by Mark Brown 18 Jan '10

18 Jan '10

The following changes since commit 735fe4cfbc3cedea41bd0ed31955054dae6beb46: Mark Brown (1): ASoC: Add missing __devexit and __devinit annotations are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.34 Daniel Mack (3): ASoC: ak4104: allow more sample rates ASoC: cs4270: allow passing freq=0 in set_dai_sysclk() ASoC: support more sample rates on raumfeld devices Mark Brown (6): ASoC: Fix i.MX audio build for i.MX3x ASoC: Convert new i.MX SSI driver to use static DAI array ASoC: i.MX SSI driver does not yet support master mode ASoC: Remove old i.MX driver code ASoC: Mark new i.MX drivers as BROKEN until arch/arm merged Merge branch 'mxc-audio' into for-2.6.34 Peter Ujfalusi (1): ASoC: TWL4030: Replace comma with semicolon in probe function Sascha Hauer (1): ASoC: Add a new imx-ssi sound driver Thomas Weber (1): ASoC: TWL4030: Fix typo in comment in header file arch/arm/plat-mxc/Makefile | 4 + arch/arm/plat-mxc/ssi-fiq-ksym.c | 20 + arch/arm/plat-mxc/ssi-fiq.S | 134 ++++++ sound/soc/codecs/ak4104.c | 4 +- sound/soc/codecs/cs4270.c | 38 ++- sound/soc/codecs/twl4030.c | 2 +- sound/soc/codecs/twl4030.h | 2 +- sound/soc/imx/Kconfig | 20 +- sound/soc/imx/Makefile | 12 +- sound/soc/imx/imx-pcm-dma-mx2.c | 313 ++++++++++++++ sound/soc/imx/imx-pcm-fiq.c | 277 ++++++++++++ sound/soc/imx/imx-ssi.c | 758 +++++++++++++++++++++++++++++++++ sound/soc/imx/imx-ssi.h | 237 +++++++++++ sound/soc/imx/mx1_mx2-pcm.c | 488 --------------------- sound/soc/imx/mx1_mx2-pcm.h | 26 -- sound/soc/imx/mx27vis_wm8974.c | 318 -------------- sound/soc/imx/mxc-ssi.c | 860 -------------------------------------- sound/soc/imx/mxc-ssi.h | 238 ----------- sound/soc/pxa/raumfeld.c | 61 ++- 19 files changed, 1823 insertions(+), 1989 deletions(-) create mode 100644 arch/arm/plat-mxc/ssi-fiq-ksym.c create mode 100644 arch/arm/plat-mxc/ssi-fiq.S create mode 100644 sound/soc/imx/imx-pcm-dma-mx2.c create mode 100644 sound/soc/imx/imx-pcm-fiq.c create mode 100644 sound/soc/imx/imx-ssi.c create mode 100644 sound/soc/imx/imx-ssi.h delete mode 100644 sound/soc/imx/mx1_mx2-pcm.c delete mode 100644 sound/soc/imx/mx1_mx2-pcm.h delete mode 100644 sound/soc/imx/mx27vis_wm8974.c delete mode 100644 sound/soc/imx/mxc-ssi.c delete mode 100644 sound/soc/imx/mxc-ssi.h

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[alsa-devel] MSM DSP
by GNUtoo 17 Jan '10

17 Jan '10

hi, I've bought an HTC dream,before buying I looked at the feasibility of porting GNU/Linux on it(so not Android). There is a patch(for maing the driver standard) or a standard(everything is non-standard in android) driver for most of the components. The biggest problem seems the sound. So knowing that codeaurora( https://www.codeaurora.org/gitweb/quic/le/?p=kernel/msm.git ) had alsa in its kenrel I first tried to compile that kernel...but due to numerous bugs(It didn't compile and needed too much fixing) in it android-msm-2.6.29 branch I decided to import alsa in the linuxtogo kernel,and I pushed the sources at a temporary htc-alsa branch at git://gitorious.org/replicant/gnulinuxkernel.git (it's called gnulinux in order to differentiate from an android kernel,I didn't find a good name for it) from the commit message available here : http://gitorious.org/replicant/gnulinuxkernel/commit/92945ccd0921e940fc2675… I had to change snd_rpc_ids.vers = 0x00020001 into snd_rpc_ids.vers = 0xaa2b1a44 ,and to make minor build system changes and I had aplay working rapidely... but only aplay... mplayer was waiting indefinitely for the following system call: ioctl(6, 0x400c4150 ls -l /proc/$(pido mplayer)/fd/6 gave /dev/snd/pcmC0D0p and 0x400c4150 gave that: SNDRV_PCM_IOCTL_WRITEI_FRAMES = _IOW('A', 0x50, struct snd_xferi) Then with printks(I had no serial cable(I've looked for a craddle/docking station with the special HTC connector for easy soldering...but I didn't found one) so I couldn't use kgdb) I went until that point: rc = wait_event_interruptible(the_locks.write_wait, (frame->used == 0)|| (prtd->stopped)); That's in the alsa_send_buffer function in /sound/soc/msm/msm-pcm.c Then I continued the tracing puting printks everywhere...and I found that: with aplay: msm_pcm_playback_copy in the /sound/soc/msm/msm7k-pcm.c was called rc = alsa_send_buffer(prtd, buf, fbytes, NULL); was called once inside alsa_send_buffer (in /sound/soc/msm/msm-pcm.c) the count loop was only executed once because count was exhausted on the first run then copy_count is incremented and the audio is configured then at second call of the alsa_send_buffer the audio is configured and works. here's an output of the aplay parameters: $ aplay -v 05\ -\ Jet\ Set.wav Playing WAVE '05 - Jet Set.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo Plug PCM: Hardware PCM card 0 'msm-audio' device 0 subdevice 0 Its setup is: stream : PLAYBACK access : RW_INTERLEAVED format : S16_LE subformat : STD channels : 2 rate : 44100 exact rate : 44100 (44100/1) msbits : 16 buffer_size : 4800 period_size : 1200 period_time : 27210 tick_time : 0 tstamp_mode : NONE period_step : 1 sleep_min : 0 avail_min : 1200 xfer_align : 1200 start_threshold : 4800 stop_threshold : 4800 silence_threshold: 0 silence_size : 0 boundary : 1258291200 note that frame = prtd->out + prtd->out_tail; makes frame->used have a bad value the first time it is run but as the count is only run once it doesn't import. the next runs prtd->out + prtd->out_tail have a good value with mplayer the story was quite different,even if the beginning was the same : msm_pcm_playback_copy in the /sound/soc/msm/msm7k-pcm.c was also called,rc = alsa_send_buffer(prtd, buf, fbytes, NULL); was also called inside alsa_send_buffer (in /sound/soc/msm/msm-pcm.c) but once in alsa_send_buffer,more precisely under the count loop the story changes: count is not exhausted the first time,because in "xfer = count > frame->size ? frame->size : count;" the smallest between count and frame->size is assigned,and so after it does "count -= xfer" and so count is not 0,because the buffer size detected by mplayer is 19200 insterad of 4800 so here what it does in detail: it passes 19200 as count here: fbytes = frames_to_bytes(runtime, frames); rc = alsa_send_buffer(prtd, buf, fbytes, NULL); that goes into that: ssize_t alsa_send_buffer(struct msm_audio *prtd, const char __user *buf, size_t count, loff_t *pos) and then the count variable is 19200 in alsa_send_buffer then there is xfer = count > frame->size ? frame->size : count; so it assigns the minimum between count and frame->size to xfer count is still 19200 frame->size is 4800 so xfer becomes 4800 then count -= xfer decrement count but only of 4800 so count is not exhausted and runs another time then : frame = prtd->out + prtd->out_tail; makes frame->used have a bad value,and then it blocks here: rc = wait_event_interruptible(the_locks.write_wait,(frame->used == 0) || (prtd->stopped)); because frame->used is not 0. here an mplayer output: Forced audio codec: mad Opening audio decoder: [pcm] Uncompressed PCM audio decoder dec_audio: Allocating 2048 + 65536 = 67584 bytes for output buffer. AUDIO: 44100 Hz, 2 ch, s16le, 1411.2 kbit/100.00% (ratio: 176400->176400) Selected audio codec: [pcm] afm: pcm (Uncompressed PCM) ========================================================================== Building audio filter chain for 44100Hz/2ch/s16le -> 0Hz/0ch/??... [libaf] Adding filter dummy [dummy] Was reinitialized: 44100Hz/2ch/s16le [dummy] Was reinitialized: 44100Hz/2ch/s16le Trying preferred audio driver 'alsa', options '[none]' alsa-init: requested format: 44100 Hz, 2 channels, 9 alsa-init: using ALSA 1.0.15 alsa-init: setup for 1/2 channel(s) alsa-init: using device default alsa-init: pcm opened in blocking mode alsa-init: got buffersize=19200 alsa-init: got period size 300 alsa: 44100 Hz/2 channels/4 bpf/19200 bytes buffer/Signed 16 bit Little Endian AO: [alsa] 44100Hz 2ch s16le (2 bytes per sample) AO: Description: ALSA-0.9.x-1.x audio output AO: Author: Alex Beregszaszi, Zsolt Barat <joy(a)streamminister.de> AO: Comment: under developement Building audio filter chain for 44100Hz/2ch/s16le -> 44100Hz/2ch/s16le... [dummy] Was reinitialized: 44100Hz/2ch/s16le [dummy] Was reinitialized: 44100Hz/2ch/s16le Video: no video Freeing 0 unused video chunks. Starting playback... Increasing filtered audio buffer size from 0 to 21248 so the quick solution was to force mplayer to use 4800 as buffer size with something like that: diff --git a/sound/soc/msm/msm7k-pcm.c b/sound/soc/msm/msm7k-pcm.c index 38e8283..bfbc66d 100644 --- a/sound/soc/msm/msm7k-pcm.c +++ b/sound/soc/msm/msm7k-pcm.c @@ -115,7 +115,7 @@ static struct snd_pcm_hardware msm_pcm_playback_hardware = { .rate_max = USE_RATE_MAX, .channels_min = USE_CHANNELS_MIN, .channels_max = USE_CHANNELS_MAX, - .buffer_bytes_max = MAX_BUFFER_PLAYBACK_SIZE, + .buffer_bytes_max = 4800, .period_bytes_min = 64, .period_bytes_max = MAX_PERIOD_SIZE, .periods_min = USE_PERIODS_MIN, but then mplayer seem to loop on an ioctl: mplayer is frozen here and doesn't respond to other input than ctrl+c: 0.0 (00.0) of 206.0 (03:26.0) ??,?% strace shows a lot of : nanosleep followed by SNDRV_PCM_IOCTL_STATUS (ioctl(6, 0x806c4120, 0xbe8bc9c8) = 0) an analyse of the status shows a constant : status->delay is 1200,and status->avail is 0,they don't vary over time and are printed just after that code in snd_pcm_status in sound/core/pcm_native.c if (runtime->status->state == SNDRV_PCM_STATE_RUNNING || runtime->status->state == SNDRV_PCM_STATE_DRAINING) status->delay = runtime->buffer_size - status->avail; else status->delay = 0; I was told in #mplayerdev that it was because the DSP didn't play the frames...indeed I hear no sound.... So how should I fix that mess? *did the quick and dirty fix cause the second problem? *should I use the DSP directly and make a pulseaudio plugin that talks to the android driver, instead(the driver seems very buggy) The good would be support for mp3 and similar things within the DSP,and that it should be more stable. The bad is that I already spent *a lot* of time with the alsa kernel driver...and that writing that will also require some userspaces changes such as: *modification of the phone routing in the "FSO" ( http://www.freesmartphone.org/index.php/Main_Page ) dbus framework to use pulseaudio *would require changes in the build system for using pulseaudio(openembedded(http://wiki.openembedded.net/index.php/Main_Page )) *may be slow...openmoko stopped to use pulseaudio because of speed issues. Denis

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[alsa-devel] Audio buffer mmap-ing on sh arch vs. cache aliasing problem
by Pawel MOLL 16 Jan '10

16 Jan '10

Hi Guys, Recently I was investigating an issue with capturing audio using USB Audio Class device on a sh4-based board. "Bad voice quality" was reported... Finally I have traced the problem to something which is (unfortunately) well known to sh developers as a D-cache aliasing (or synonym) problem. Briefly speaking: due to some MMU design decisions, one can have two different virtual address pointing to the same physical location, which is fine, but going via different cache slots! So if there was a value of "0" in the memory and user "A" will write "1" there, user "B" will still read "0"... The solution is to ensure all TLB entries (so virtual memory areas) are beginning from a 16kB-aligned virtual address. Otherwise it is necessary to flush the cache between accesses from "A" and "B" sides. And now. The USB Audio Class driver (sound/usb/usbaudio.c) is allocating the sound buffer like this... static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t size) { [...] runtime->dma_area = vmalloc(size); [...] } ... and vmalloc will return a page(4k)-aligned pointer, possibly not 16k-aligned one. This is the source of all evil ;-) When using RW transfers everything is fine, as data is memcopied between two independent buffers. In a case of MMAP mode we will end up with such a situation: 1. The driver will memcpy data from an URB to the buffer. It will populate several cache lines. Let's call it a "kernel cache". 2. As the library has mapped the non-16k-aligned address via different cache line ("user cache"), it will read some rubbish from the physical memory, populating the "user cache" by the way. 3. Some time later the "kernel cache" is flushed, writing the data to the memory. 4. Some new data from URB is entering "kernel cache". 5. The library will access mmap-ed area again, going via the "user cache", which may or may not reflect the correct data (depending on the fact was the "user cache" flushed or not) etc. Of course this cycle is completely not deterministic, so some of the "kernel cache" lines will be flushed before being accessed from user space, other not... The final effect is... hmm... bizarre :-) First, of course, you will get a (probably loud) glitch (rubbish from buffer's underlying memory, before the first valid data is written back), and then something that could be described as an "echo" ;-) I mean - you are capturing "1 2 3 4 5 6 7..." and the result is (G stands for Glitch ;-) "G 1 23 3 4 4 6 67..." As a quick-and-dirty work-around I have modified snd_pcm_alloc_vmalloc_buffer() to allocate always 12kB more and then use 16k-aligned pointer: static int snd_pcm_alloc_vmalloc_buffer(struct snd_pcm_substream *subs, size_t size) { struct snd_pcm_runtime *runtime = subs->runtime; if (runtime->dma_addr) { if (runtime->dma_bytes >= size) return 0; /* already large enough */ vfree((void *)runtime->dma_addr); } runtime->dma_addr = (unsigned long)vmalloc(size + 12 * 1024); if (!runtime->dma_addr) return -ENOMEM; runtime->dma_area = (void *)ALIGN(runtime->dma_addr, 16 * 1024); runtime->dma_bytes = size; return 0; } Of course it cannot be regarded as anything more as a hack. And definitely not as a solution... So my question is: Any idea how the "proper solution" should look like? I see no obvious method to get an arch-independent "mmap-compliant" buffer. This problem was found on a sh arch, using an USB Audio Class device, however out there some other architectures are suffering from this MMU "feature" as well (ie. see http://docs.hp.com/en/B3906-90006/ch07s09.html) and possibly other drivers could behave "wrongly" (quoted as the driver is actually innocent...) To be honest I just have absolutely no idea what to do with this all! :-O I hope I was clear enough in the description... Any feedback, advise, idea etc. will be more than appreciated. Cheers Paweł

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[alsa-devel] [PATCH] - fix speaker output for toshiba nb205 netbooks
by Nickolas 16 Jan '10

16 Jan '10

this is a quick fix i wrote up a couple days ago to fix speaker output on the Realtek ALC272 codec present in toshiba nb205 netbooks. presently, speaker output doesn't work at all on this model, regardless of settings in the mixer. this is because the current driver thinks the speaker is at nid 0x0c instead of the correct 0x17. i'm not too familiar with this hardware, so i'm not sure that this patch won't have any negative effects for other hardware. it also seems like a messy way of fixing the problem. any comments or suggestions are welcome, as is testing for those who have this board. diffed against the latest git. Thanks, Nick -- Signed-off-by: Nickolas Lloyd <ultrageek.lloyd(a)gmail.com> --- alsa-driver/sound/pci/hda/patch_realtek.c 2010-01-12 08:03:28.000000000 -0500 +++ alsa-driver-b/sound/pci/hda/patch_realtek.c 2010-01-12 16:27:48.000000000 -0500 @@ -192,6 +192,7 @@ enum { ALC663_ASUS_MODE8, ALC272_DELL, ALC272_DELL_ZM1, + ALC272_TOSHIBA_N200, ALC272_SAMSUNG_NC10, ALC662_AUTO, ALC662_MODEL_LAST, @@ -16725,6 +16726,11 @@ static struct hda_verb alc272_dell_init_ {} }; +static struct hda_verb alc272_toshiba_init_verbs[] = { + {0x17, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, + {} +}; + static struct hda_verb alc663_mode7_init_verbs[] = { {0x15, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, {0x16, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_IN}, @@ -17255,6 +17261,36 @@ static void alc663_g50v_inithook(struct alc_mic_automute(codec); } +static void alc272_toshiba_speaker_automute(struct hda_codec *codec) +{ + unsigned int present; + unsigned char bits; + + present = snd_hda_jack_detect(codec, 0x21); + bits = present ? HDA_AMP_MUTE : 0; + snd_hda_codec_amp_stereo(codec, 0x17, HDA_OUTPUT, 0, + HDA_AMP_MUTE, bits); +} + +static void alc272_toshiba_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + switch (res >> 26) { + case ALC880_HP_EVENT: + alc272_toshiba_speaker_automute(codec); + break; + case ALC880_MIC_EVENT: + alc_mic_automute(codec); + break; + } +} + +static void alc272_toshiba_init_hook(struct hda_codec *codec) +{ + alc272_toshiba_speaker_automute(codec); + alc_mic_automute(codec); +} + static struct snd_kcontrol_new alc662_ecs_mixer[] = { HDA_CODEC_VOLUME("Master Playback Volume", 0x02, 0x0, HDA_OUTPUT), ALC262_HIPPO_MASTER_SWITCH, @@ -17391,7 +17427,7 @@ static struct snd_pci_quirk alc662_cfg_t SND_PCI_QUIRK(0x105b, 0x0cd6, "Foxconn", ALC662_ECS), SND_PCI_QUIRK(0x105b, 0x0d47, "Foxconn 45CMX/45GMX/45CMX-K", ALC662_3ST_6ch_DIG), - SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB200", ALC663_ASUS_MODE4), + SND_PCI_QUIRK(0x1179, 0xff6e, "Toshiba NB200", ALC272_TOSHIBA_N200), SND_PCI_QUIRK(0x144d, 0xca00, "Samsung NC10", ALC272_SAMSUNG_NC10), SND_PCI_QUIRK(0x1458, 0xa002, "Gigabyte 945GCM-S2L", ALC662_3ST_6ch_DIG), @@ -17696,6 +17732,22 @@ static struct alc_config_preset alc662_p .setup = alc663_m51va_setup, .init_hook = alc663_m51va_inithook, }, + [ALC272_TOSHIBA_N200] = { + .mixers = { alc663_asus_21jd_clfe_mixer }, + .cap_mixer = alc662_auto_capture_mixer, + .init_verbs = { alc662_init_verbs, + alc663_21jd_amic_init_verbs, + alc272_toshiba_init_verbs }, + .num_dacs = ARRAY_SIZE(alc662_dac_nids), + .hp_nid = 0x03, + .dac_nids = alc662_dac_nids, + .dig_out_nid = ALC662_DIGOUT_NID, + .num_channel_mode = ARRAY_SIZE(alc662_3ST_2ch_modes), + .channel_mode = alc662_3ST_2ch_modes, + .unsol_event = alc272_toshiba_unsol_event, + .setup = alc663_mode4_setup, + .init_hook = alc272_toshiba_init_hook, + }, [ALC272_SAMSUNG_NC10] = { .mixers = { alc272_nc10_mixer }, .init_verbs = { alc662_init_verbs,

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[alsa-devel] Audio suspend/resume status on OMAP3 based platforms
by Aggarwal, Anuj 15 Jan '10

15 Jan '10

Hi, I want to check the status of suspend/resume functionality when audio subsystem is being used. Specifically, has some one tried the following tests on their omap3 based platforms: a) echo mem > /sys/power/state, when audio playback is running in background b) same command, when audio capture is running in background b) same command, when audio loopback is running in background I am facing some issues with (b) & (c) so wanted to confirm whether someone else has faced similar problem or not. Regards, Anuj Aggarwal

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[alsa-devel] [PATCH] ALSA: ASoC: support more sample rates on raumfeld devices
by Daniel Mack 15 Jan '10

15 Jan '10

Add support for sample rates other than 44100Khz on raumfeld audio devices. Force the codec's supported sample rates as the table of supported rates which is calculated at run-time does not suffice: if we call snd_soc_dai_set_sysclk() at stream startup, half of the possible rates are missing. Use the external audio clock generator to provide double data rate clocks as the PXA's internal baud generator does anything but what's described in the datasheets. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> Cc: Timur Tabi <timur(a)freescale.com> Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/soc/pxa/raumfeld.c | 61 ++++++++++++++++++++++++++++++---------------- 1 files changed, 40 insertions(+), 21 deletions(-) diff --git a/sound/soc/pxa/raumfeld.c b/sound/soc/pxa/raumfeld.c index e713181..8d65a49 100644 --- a/sound/soc/pxa/raumfeld.c +++ b/sound/soc/pxa/raumfeld.c @@ -41,7 +41,9 @@ static struct i2c_board_info max9486_hwmon_info = { }; #define MAX9485_MCLK_FREQ_112896 0x22 -#define MAX9485_MCLK_FREQ_122880 0x23 +#define MAX9485_MCLK_FREQ_122880 0x23 +#define MAX9485_MCLK_FREQ_225792 0x32 +#define MAX9485_MCLK_FREQ_245760 0x33 static void set_max9485_clk(char clk) { @@ -71,9 +73,18 @@ static int raumfeld_cs4270_startup(struct snd_pcm_substream *substream) struct snd_soc_pcm_runtime *rtd = substream->private_data; struct snd_soc_dai *codec_dai = rtd->dai->codec_dai; - set_max9485_clk(MAX9485_MCLK_FREQ_112896); + codec_dai->playback.rates = SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000; + codec_dai->playback.rate_min = 44100; + codec_dai->playback.rate_max = 48000; - return snd_soc_dai_set_sysclk(codec_dai, 0, 11289600, 0); + codec_dai->capture.rates = codec_dai->playback.rates; + codec_dai->capture.rate_min = codec_dai->playback.rate_min; + codec_dai->capture.rate_max = codec_dai->playback.rate_max; + + return 0; } static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream, @@ -86,20 +97,24 @@ static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream, int ret = 0; switch (params_rate(params)) { - case 8000: - case 16000: + case 44100: + set_max9485_clk(MAX9485_MCLK_FREQ_112896); + clk = 11289600; + break; case 48000: - case 96000: set_max9485_clk(MAX9485_MCLK_FREQ_122880); clk = 12288000; break; - case 11025: - case 22050: - case 44100: case 88200: - set_max9485_clk(MAX9485_MCLK_FREQ_112896); - clk = 11289600; + set_max9485_clk(MAX9485_MCLK_FREQ_225792); + clk = 22579200; + break; + case 96000: + set_max9485_clk(MAX9485_MCLK_FREQ_245760); + clk = 24576000; break; + default: + return -EINVAL; } fmt = SND_SOC_DAIFMT_I2S | @@ -128,7 +143,7 @@ static int raumfeld_cs4270_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, 0, 1); + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1); if (ret < 0) return ret; @@ -181,20 +196,24 @@ static int raumfeld_ak4104_hw_params(struct snd_pcm_substream *substream, int fmt, ret = 0, clk = 0; switch (params_rate(params)) { - case 8000: - case 16000: + case 44100: + set_max9485_clk(MAX9485_MCLK_FREQ_112896); + clk = 11289600; + break; case 48000: - case 96000: set_max9485_clk(MAX9485_MCLK_FREQ_122880); clk = 12288000; break; - case 11025: - case 22050: - case 44100: case 88200: - set_max9485_clk(MAX9485_MCLK_FREQ_112896); - clk = 11289600; + set_max9485_clk(MAX9485_MCLK_FREQ_225792); + clk = 22579200; + break; + case 96000: + set_max9485_clk(MAX9485_MCLK_FREQ_245760); + clk = 24576000; break; + default: + return -EINVAL; } fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF; @@ -217,7 +236,7 @@ static int raumfeld_ak4104_hw_params(struct snd_pcm_substream *substream, if (ret < 0) return ret; - ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, 0, 1); + ret = snd_soc_dai_set_sysclk(cpu_dai, PXA_SSP_CLK_EXT, clk, 1); if (ret < 0) return ret; -- 1.6.5.2

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[alsa-devel] [PATCH] ALSA: Fix mute led GPIO on HP dv-series notebooks.
by Kunal Gangakhedkar 15 Jan '10

15 Jan '10

On my laptop (HP dv6-1110ax), there are no OEM strings in SMBIOS of type "HP_Mute_LED*". Hence, the GPIO for the mute button LED doesn't get set properly. I didn't find the strings in my cousin's laptop (HP dv9500t CTO) either. As per the documentation of find_mute_led_gpio(), these strings occur in HP B-series systems - so, before scanning the SMBIOS strings, we need to check if we're dealing with a B-series system. Need to get confirmation from HP if this logic takes care of all the systems. I'm trying to poke a friend there. Please let me know if this looks OK or needs changes. Signed-off-by: Kunal Gangakhedkar <kunal.gangakhedkar(a)gmail.com> --- sound/pci/hda/patch_sigmatel.c | 82 ++++++++++++++++++++++++++++++---------- 1 files changed, 62 insertions(+), 20 deletions(-) diff --git a/sound/pci/hda/patch_sigmatel.c b/sound/pci/hda/patch_sigmatel.c index 117919a..d421b3b 100644 --- a/sound/pci/hda/patch_sigmatel.c +++ b/sound/pci/hda/patch_sigmatel.c @@ -4737,6 +4737,8 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) } } +static int hp_blike_system(u32 subsystem_id); + /* * This method searches for the mute LED GPIO configuration * provided as OEM string in SMBIOS. The format of that string @@ -4748,6 +4750,19 @@ static void stac92xx_unsol_event(struct hda_codec *codec, unsigned int res) * * So, HP B-series like systems may have HP_Mute_LED_0 (current models) * or HP_Mute_LED_0_3 (future models) OEM SMBIOS strings + * + * + * The dv-series laptops don't seem to have the HP_Mute_LED* strings in + * SMBIOS - at least the ones I have seen do not have them - which include + * my own system (HP Pavilion dv6-1110ax) and my cousin's + * HP Pavilion dv9500t CTO. + * Need more information on whether it is true across the entire series. + * + * The "else" could be converted into a conditional check for dv-series - + * say, hp_dv_series_system() - just like hp_blike_system(); however, + * I don't have all the PCI ids to identify the dv-series machines - maybe, + * we can add that later. + * -- kunal */ static int find_mute_led_gpio(struct hda_codec *codec) { @@ -4755,29 +4770,54 @@ static int find_mute_led_gpio(struct hda_codec *codec) const struct dmi_device *dev = NULL; if ((codec->subsystem_id >> 16) == PCI_VENDOR_ID_HP) { - while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING, + if (hp_blike_system(codec->subsystem_id)) { + while ((dev = dmi_find_device(DMI_DEV_TYPE_OEM_STRING, NULL, dev))) { - if (sscanf(dev->name, "HP_Mute_LED_%d_%d", - &spec->gpio_led_polarity, - &spec->gpio_led) == 2) { - spec->gpio_led = 1 << spec->gpio_led; - return 1; - } - if (sscanf(dev->name, "HP_Mute_LED_%d", - &spec->gpio_led_polarity) == 1) { - switch (codec->vendor_id) { - case 0x111d7608: - /* GPIO 0 */ - spec->gpio_led = 0x01; - return 1; - case 0x111d7600: - case 0x111d7601: - case 0x111d7602: - case 0x111d7603: - /* GPIO 3 */ - spec->gpio_led = 0x08; + if (sscanf(dev->name, "HP_Mute_LED_%d_%d", + &spec->gpio_led_polarity, + &spec->gpio_led) == 2) { + spec->gpio_led = 1 << spec->gpio_led; return 1; } + if (sscanf(dev->name, "HP_Mute_LED_%d", + &spec->gpio_led_polarity) == 1) { + switch (codec->vendor_id) { + case 0x111d7608: + /* GPIO 0 */ + spec->gpio_led = 0x01; + return 1; + case 0x111d7600: + case 0x111d7601: + case 0x111d7602: + case 0x111d7603: + /* GPIO 3 */ + spec->gpio_led = 0x08; + return 1; + } + } + } + } else { + /* + * Pavilion dv series notebooks do not seem to have + * HP_Mute_LED* in SMBIOS. + * If HP decides to add it in dv series, + * we'll need to fix this part + * - or merge it with the B-like check above. + */ + switch (codec->vendor_id) { + case 0x111d7608: + /* GPIO 0 */ + spec->gpio_led = 0x01; + spec->gpio_led_polarity |= spec->gpio_led; + return 1; + case 0x111d7600: + case 0x111d7601: + case 0x111d7602: + case 0x111d7603: + /* GPIO 3 */ + spec->gpio_led = 0x08; + spec->gpio_led_polarity |= spec->gpio_led; + return 1; } } } @@ -5603,6 +5643,8 @@ again: spec->num_dmuxes = ARRAY_SIZE(stac92hd71bxx_dmux_nids); spec->num_smuxes = stac92hd71bxx_connected_smuxes(codec, 0x1e); + snd_printd("Found board config: %d\n", spec->board_config); + switch (spec->board_config) { case STAC_HP_M4: /* enable internal microphone */ -- 1.6.6.60.gc2ff1

2 8

[alsa-devel] [PATCH] alsa: use subsys_initcall for sound core instead of module_init
by Thadeu Lima de Souza Cascardo 15 Jan '10

15 Jan '10

This is needed for built-in drivers which are built before the sound directory, like thinkpad_acpi. Otherwise, registering a card fails. Signed-off-by: Thadeu Lima de Souza Cascardo <cascardo(a)holoscopio.com> --- sound/core/sound.c | 4 ++-- sound/sound_core.c | 2 +- 2 files changed, 3 insertions(+), 3 deletions(-) diff --git a/sound/core/sound.c b/sound/core/sound.c index 7872a02..563d196 100644 --- a/sound/core/sound.c +++ b/sound/core/sound.c @@ -468,5 +468,5 @@ static void __exit alsa_sound_exit(void) unregister_chrdev(major, "alsa"); } -module_init(alsa_sound_init) -module_exit(alsa_sound_exit) +subsys_initcall(alsa_sound_init); +module_exit(alsa_sound_exit); diff --git a/sound/sound_core.c b/sound/sound_core.c index dbca7c9..7c2d677 100644 --- a/sound/sound_core.c +++ b/sound/sound_core.c @@ -61,7 +61,7 @@ static void __exit cleanup_soundcore(void) class_destroy(sound_class); } -module_init(init_soundcore); +subsys_initcall(init_soundcore); module_exit(cleanup_soundcore); -- 1.6.6

2 6

Re: [alsa-devel] ALSA DMIX support without MMAP ?
by ANISH KUMAR 15 Jan '10

15 Jan '10

>For dmix, the stop_threshold should be equal to boundary value, thus XRUN >check should never be true. If you got xrun with dmix, something is wrong. DMIX is working now. > but when i am > playing with Plug PCM there is some noise in the song why????Is rate > conversion algoritm not able to resample it properly? >The default linear resampler is not very good quality. You may use some >sox resamplers from the alsa-plugins package, but they're more CPU >intensive. Check google. Actually i am working on soc platform(android-S3C6410) and wanted to use DMIX and some good resampler for my software mixing instead of current software mixing and resampling in upper layer(Above alsa-lib).DMIX is working fine but this resampling is causing some jitter.During rate conversion it looks for libasound_module_rate_speexrate.so which i don't have so its using default one causing the jitter. Can i build this .so file (as current way of alsa-plugin downloading will not work for me)?For sox resampler i guess i need to use differnet .so and corresponding changes in asound.conf file?? Can i get speex or sox resampler code so that i can build it for my system????? This code(speex or sox) i guess would have been integrated with alsa-lib so that alsa-lib can call corresponding functions of resampler source code. OR simply point me to url as googling is not helping.Everywhere i am getting this "you need to download plugins".This will not work in my case.

2 1

[alsa-devel] soc-cache problem
by Cliff Cai 15 Jan '10

15 Jan '10

Hi all, Nowadays, soc tries to set up standard I/O functions with same addr_bits and data_bits for a codec. My problem is some codec might have registers with different data length.i.e.ADAU1761, Most of its registers are 8 bits long except PLL register,which is a 8 bytes long one.In order to set/read PLL , a block of data(8 bytes) needs to be transfer to/from the PLL register at a time. So, if soc-cache needs to consider this kind of case, or just keep it in local driver? Thanks Cliff

2 1

Re: [alsa-devel] VGN-P11Z_G (hda-intel ALC262) internal microphone not working
by Daniel Chen 15 Jan '10

15 Jan '10

Hi, Takashi, On Wed, Jan 13, 2010 at 12:28 PM, Christian Stöveken <christian.stoeveken(a)gmail.com> wrote: > sorry to get back to you so late - I tried again 5 minutes ago and it is now > working with the false parameter removed (/etc/modules:9: snd-hda-intel > model=vaio) > > working "model=auto": > http://www.alsa-project.org/db/?f=67f6a692f0ccd70d6893665d36c8b90a51f4a577 > > also with "power_save=10 power_save_controller=N model=auto": > http://www.alsa-project.org/db/?f=618f4d4ae530957b4b6d2b66751871d4073df685 > > I hope this is how you expected it to look like. > Also the headphones now work correctly. Attached is a patch against current sound-2.6 master HEAD that: * adds Christian's SSID to the relevant quirk table, * prefers BIOS autoprobing for existing Sony VAIO models matching 0x104d90xx. The second change was made to ease backing out the change in case of regressions. Best, -Dan

3 6

Re: [alsa-devel] [PATCH 0/5] PCM mmap (temporary) fixes for non-coherent architectures
by Takashi Iwai 15 Jan '10

15 Jan '10

At Fri, 1 Jan 2010 20:31:30 +0100, Andreas Mohr wrote: > > Hi, > > I've tried this patch set (with the typo-corrected part 4) on my ASUS > WL-500gP v2 MIPSEL via a backport to 2.6.31.9, but all I get is a > small blip of the sound I wanted to play, and then the system is fubar > (I believe just the same thing as what happened without having this patch > applied). As I mentioned in the previous followup, if your device is a USB-audio, the patch doesn't help because it's for devices with buffers using dma_alloc_coherent(). For USB-audio, it uses vmalloc for an intermediate buffer. Maybe this should be changed to dma_*() stuff for such architectures. Nevertheless, I don't know whether the crash is related with the audio part... Takashi

3 8

Re: [alsa-devel] hda_intel.c patch requested for Teradici
by Maher Fahmi 15 Jan '10

15 Jan '10

ALSA Team, Sorry to bother you with this again. I haven't yet received confirmation (or rejection) from the moderator, and just wondering if there's anything else I can/should do. Best regards & Happy New Year, Maher. ________________________________ From: Maher Fahmi Sent: Friday, December 04, 2009 1:41 PM To: 'alsa-devel(a)alsa-project.org' Subject: hda_intel.c patch requested for Teradici Alsa Team, Enclosed is a patch to the hda_intel.c file from the alsa-driver-1.0.21 package. I tried to submit this using git, but ran into some problems that exceed my limited understanding of Linux. I hope this will suffice for you. Summary: Fix to a HW issue in which the 64OK register bit of the GCAP register is incorrectly asserted. Signed-off-by: Maher Fahmi mfahmi(a)teradici.com Thanks very much, Maher.

2 1

[alsa-devel] snd_pcm_avail_update returns 0
by Singaravelan Nallasellan 14 Jan '10

14 Jan '10

Hi, Our audio driver uses the indirect PCM data transfer mechanism to transfer the buffer pointers and the size of the buffer to the Audio engine which DMAs the PCM data to the sound card. The driver uses the .ack callback to receive the buffer pointer and the size. It also sends it to the audio engine. We are facing a problem when we play long wav file using our driver. The application has written the PCM data and the ring buffer is full. snd_pcm_avail_update returns 0 sporadically. But the .ack callback was not called. The playback is stuck. It is not progressing. I have tried with pulseaudio (0.9.15) and with my own player. Both gives the same result. I would appreciate if somebody could provide some pointer to identify the cause of this issue. Let me know if anybody needs more info. Thanks Sing

2 5

Re: [alsa-devel] snd-hdsp driver error?
by Raymond Yau 14 Jan '10

14 Jan '10

AFAIK , alsa-lib use the driver name to find the name of corresponding conf file from /usr/share/alsa/cards/aliases.conf In your case , H-DSP.conf does not exist and there is no aliases in /usr/share/alsa/cards/aliases.conf You have to ask Takashi or Jaroslav to fix this issue for your card open("/usr/share/alsa/cards/aliases.conf", O_RDONLY) = 3 access("/usr/share/alsa/cards/H-DSP.conf", R_OK) = -1 ENOENT (No such file or directory) open("/dev/snd/controlC1", O_RDONLY) = 3 close(3) = 0 write(2, "ALSA lib confmisc.c:1286:(snd_fu"..., 42ALSA lib confmisc.c:1286:(snd_func_refer) ) = 42 write(2, "Unable to find definition 'cards"..., 58Unable to find definition 'cards.H-DSP.pcm.front.0:CARD=0') = 58 write(2, "\n", 1 2009/11/7 Lars Zobbe <fedesensfar(a)gmail.com> > It is a clean install. > lars@lars-desktop:~$ lsb_release -a && uname -a > No LSB modules are available. > Distributor ID: Ubuntu > Description: Ubuntu 9.10 > Release: 9.10 > Codename: karmic > Linux lars-desktop 2.6.31-14-generic #48-Ubuntu SMP Fri Oct 16 14:05:01 UTC > 2009 x86_64 GNU/Linux > > It didn't work with pulseaudio in any previos version (of ubuntu or any > other distro I've tried) either. > And the standard answer from people "out there" is "forget about pulse it > doesn't work with the HDSP" f.x. this guy > http://ubuntuforums.org/showpost.php?p=7578331&postcount=10 > And that's why I contacted pulseaudio developers first when I found out the > it was because snd_hdsp didn't load with the preferred udev but only with > module-detect (se my first post). They said that I should inform Alsa > developers as there seems to be a problem with the alsa driver. > > I've attached the strace output. > > Thanks. > Lars. > > > > > On Fri, Nov 6, 2009 at 2:53 AM, Raymond Yau <superquad.vortex2(a)gmail.com>wrote: > >> Is it a clean install of Ubuntu 9.10 or upgrade from previous Ubuntu ? >> >> you may need to find out why aplay -v -Dfront 32bit.wav fail ? >> >> strace aplay -v -Dfront 32bit.wav >> >> >> >> >> >> !!Linux Distribution >> >> !!------------------ >> >> Ubuntu 9.10 \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 9.10" >> >> !!Sound Servers on this system >> !!---------------------------- >> >> Pulseaudio: >> Installed - Yes (/usr/bin/pulseaudio) >> >> >> Running - Yes >> >> ESound Daemon: >> Installed - Yes (/usr/bin/esd) >> Running - No >> >> Jack: >> Installed - Yes (/usr/bin/jackd) >> Running - No >> >> >> !!ALSA configuration files >> >> >> !!------------------------ >> >> !!User specific config file (~/.asoundrc) >> >> pcm.!default { >> type plug >> slave { pcm "jack" } >> } >> >> pcm.jack { >> type jack >> playback_ports { >> >> 0 alsa_pcm:playback_1 >> >> 1 alsa_pcm:playback_2 >> } >> capture_ports { >> 0 alsa_pcm:capture_1 >> 1 alsa_pcm:capture_2 >> } >> } >> >> ctl.mixer0 { >> type hw >> card 0 >> } >> >> >> lars@lars-desktop:~$ aplay -v -Dfront:0 32bit.wav >> ALSA lib confmisc.c:1286:(snd_func_refer) Unable to find definition >> 'cards.H-DSP.pcm.front.0:CARD=0' >> ALSA lib conf.c:3513:(_snd_config_evaluate) function snd_func_refer >> returned error: No such file or directory >> ALSA lib conf.c:3985:(snd_config_expand) Evaluate error: No such file or >> directory >> ALSA lib pcm.c:2211:(snd_pcm_open_noupdate) Unknown PCM front:0 >> aplay: main:608: audio open error: No such file or directory >> >> >> Did PA developer tell you not to load oss module ? >> >> >> >> D: alsa-util.c: Checking for subset analog-mono (hw:%f) >> D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ... >> D: alsa-util.c: Managed to open hw:0 >> D: alsa-util.c: snd_pcm_hw_params_set_access() failed: Invalid argument >> D: alsa-util.c: Trying hw:0 without SND_PCM_NO_AUTO_FORMAT ... >> D: alsa-util.c: Managed to open hw:0 >> D: alsa-util.c: snd_pcm_hw_params_set_access() failed: Invalid argument >> D: alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ... >> D: alsa-util.c: Managed to open plug:hw:0 >> D: alsa-util.c: snd_pcm_hw_params_set_format(Signed 16 bit Little Endian) >> failed: Invalid argument >> D: alsa-util.c: snd_pcm_hw_params_set_format(Signed 16 bit Big Endian) >> failed: Invalid argument >> D: alsa-util.c: snd_pcm_hw_params_set_format(Float 32 bit Little Endian) >> failed: Invalid argument >> D: alsa-util.c: snd_pcm_hw_params_set_format(Float 32 bit Big Endian) >> failed: Invalid argument >> D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument >> D: alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ... >> D: alsa-util.c: Managed to open plug:hw:0 >> D: alsa-util.c: snd_pcm_hw_params_set_channels(1) failed: Invalid argument >> I: alsa-util.c: Failed to set hardware parameters on plug:hw:0: Invalid >> argument >> D: alsa-util.c: Trying hw:0 as last resort... >> D: alsa-util.c: Trying hw:0 with SND_PCM_NO_AUTO_FORMAT ... >> D: alsa-util.c: Managed to open hw:0 >> D: alsa-util.c: snd_pcm_hw_params_set_access() failed: Invalid argument >> D: alsa-util.c: Trying hw:0 without SND_PCM_NO_AUTO_FORMAT ... >> D: alsa-util.c: Managed to open hw:0 >> D: alsa-util.c: snd_pcm_hw_params_set_access() failed: Invalid argument >> D: alsa-util.c: Trying plug:hw:0 with SND_PCM_NO_AUTO_FORMAT ... >> D: alsa-util.c: Managed to open plug:hw:0 >> D: alsa-util.c: snd_pcm_hw_params_set_format(Signed 16 bit Little Endian) >> failed: Invalid argument >> D: alsa-util.c: snd_pcm_hw_params_set_format(Signed 16 bit Big Endian) >> failed: Invalid argument >> D: alsa-util.c: snd_pcm_hw_params_set_format(Float 32 bit Little Endian) >> failed: Invalid argument >> D: alsa-util.c: snd_pcm_hw_params_set_format(Float 32 bit Big Endian) >> failed: Invalid argument >> D: alsa-util.c: Maximum hw buffer size is 371 ms >> I: (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed >> I: (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed >> I: (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed >> I: (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed >> D: alsa-util.c: Set neither period nor buffer size. >> I: (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed >> I: alsa-util.c: snd_pcm_hw_params failed: Device or resource busy >> D: alsa-util.c: Trying plug:hw:0 without SND_PCM_NO_AUTO_FORMAT ... >> D: alsa-util.c: Managed to open plug:hw:0 >> D: alsa-util.c: Maximum hw buffer size is 371 ms >> I: (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed >> I: (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed >> I: (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed >> I: (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed >> D: alsa-util.c: Set neither period nor buffer size. >> I: (alsa-lib)pcm_hw.c: SNDRV_PCM_IOCTL_HW_PARAMS failed >> I: alsa-util.c: snd_pcm_hw_params failed: Device or resource busy >> I: alsa-util.c: Failed to set hardware parameters on plug:hw:0: Device or >> resource busy >> E: module.c: Failed to load module "module-alsa-source" (argument: >> "device_id=0"): initialization failed. >> W: module.c: module-oss is deprecated: Please use module-alsa-card instead >> of module-oss! >> D: oss-util.c: capabilities: MMAP REALTIME TRIGGER >> I: module-oss.c: Device opened in O_RDONLY mode. >> D: oss-util.c: Asking for 4 fragments of size 512 (requested 880) >> E: oss-util.c: SNDCTL_DSP_SETFMT: Device or resource busy >> E: module.c: Failed to load module "module-oss" (argument: >> "device=/dev/dsp"): initialization failed. >> W: module-detect.c: failed to detect any sound hardware. >> >> >> 2009/11/6 Lars Zobbe <fedesensfar(a)gmail.com> >> >> Sorry. I didn't realise that it was a bash script. Here is the link to my >>> alsa-info >>> >>> http://www.alsa-project.org/db/?f=d3728557be71067e2ed0920747118ae7efff2829 >>> >>> Thanks again. >>> Lars. >>> >>> >>> On Thu, Nov 5, 2009 at 5:29 AM, Raymond Yau <superquad.vortex2(a)gmail.com >>> > wrote: >>> >>>> Please provide the full pulseaudio log and output of alsa-info.sh >>>> >>>> http://www.pulseaudio.org/wiki/Troubleshooting >>>> >>>> pulseaudio -k; pulseaudio -vvv >>>> >>>> and >>>> >>>> aplay -v -Dfront:0 32bit.wav >>>> >>>> aplay -v -M -Dfront:0 32bit.wav >>>> >>>> aplay -v -Dplughw:0 any.wav >>>> >>>> aplay -v -M -Dplughw:0 any.wav >>>> >>>> >>>> 2009/11/4 Lars Zobbe <fedesensfar(a)gmail.com> >>>> >>>> > Hi Raymond. >>>> > >>>> > I so sorry Raymond. But I'm not sure what to do with this info. Is >>>> there >>>> > something more I have to tell you? Is there anything I can do to make >>>> > pulseaudio able to detect via udev? Can I copy-paste that bit of code >>>> you >>>> > sent me somewhere? >>>> > Please tell me if there's more you need to know. >>>> > Greetings. >>>> > Lars. >>>> > >>>> > >>>> > >>>> > >>>> > On Tue, Nov 3, 2009 at 7:48 AM, Raymond Yau < >>>> superquad.vortex2(a)gmail.com>wrote: >>>> > >>>> >> I guess route plugin used by RME96*.conf does not support mmap when >>>> there >>>> >> is >>>> >> channel conversion since minimum channels supported by hdsp is 14 >>>> >> >>>> >> >>>> >> static struct snd_pcm_hardware snd_hdsp_playback_subinfo = >>>> >> { >>>> >> .info = (SNDRV_PCM_INFO_MMAP | >>>> >> SNDRV_PCM_INFO_MMAP_VALID | >>>> >> SNDRV_PCM_INFO_NONINTERLEAVED | >>>> >> SNDRV_PCM_INFO_SYNC_START | >>>> >> SNDRV_PCM_INFO_DOUBLE), >>>> >> #ifdef SNDRV_BIG_ENDIAN >>>> >> .formats = SNDRV_PCM_FMTBIT_S32_BE, >>>> >> #else >>>> >> .formats = SNDRV_PCM_FMTBIT_S32_LE, >>>> >> #endif >>>> >> .rates = (SNDRV_PCM_RATE_32000 | >>>> >> SNDRV_PCM_RATE_44100 | >>>> >> SNDRV_PCM_RATE_48000 | >>>> >> SNDRV_PCM_RATE_64000 | >>>> >> SNDRV_PCM_RATE_88200 | >>>> >> SNDRV_PCM_RATE_96000), >>>> >> .rate_min = 32000, >>>> >> .rate_max = 96000, >>>> >> .channels_min = 14, >>>> >> .channels_max = HDSP_MAX_CHANNELS, >>>> >> .buffer_bytes_max = HDSP_CHANNEL_BUFFER_BYTES * HDSP_MAX_CHANNELS, >>>> >> .period_bytes_min = (64 * 4) * 10, >>>> >> .period_bytes_max = (8192 * 4) * HDSP_MAX_CHANNELS, >>>> >> .periods_min = 2, >>>> >> .periods_max = 2, >>>> >> .fifo_size = 0 >>>> >> }; >>>> >> >>>> >> 2009/11/2 Lars Zobbe <fedesensfar(a)gmail.com> >>>> >> >>>> >> > Hi. >>>> >> > >>>> >> > Talking to one of the pulseaudio developers (*Colin Guthrie*) about >>>> a >>>> >> > problem with pulseaudio loading modules and my sound card (RME HDSP >>>> >> 9632), >>>> >> > I >>>> >> > was adviced to inform you about the problem. >>>> >> > >>>> >> > I'm on ubuntu 9.10, 64bit, kernel 2.6.31 with an amd64 x2 3800+ >>>> >> processor. >>>> >> > >>>> >> > >>>> >> > Symptoms are: >>>> >> > Can only make pulseaudio load modules with the deprecated >>>> load-module >>>> >> > module-detect command, module-udev-detect doesn't detect anything >>>> (no >>>> >> sound >>>> >> > - only null sink output in padevchooser). >>>> >> > I get terrible stuttering. >>>> >> > I can remove pulseaudio and make things work, more or less. >>>> However, >>>> >> that >>>> >> > leaves me without software mixing and apps fighting for the rights >>>> to >>>> >> make >>>> >> > sound. And more inconveniences (fx. no volume-control). >>>> >> > I can play sound from audacity without stuttering by choosing the >>>> >> cardname >>>> >> > directly in output HDSP, not ALSA, not pulseaudio. >>>> >> > >>>> >> > >>>> >> > When trying to run alsa-info.sh I get a lot of "not found" errors. >>>> So >>>> >> > you'll >>>> >> > get some manual info (they might have moved som files around in >>>> ubuntu >>>> >> > 9.10). >>>> >> > >>>> >> > This is part of my syslog during a boot. (I did the formatting to >>>> bold). >>>> >> > >>>> >> > Nov 2 10:57:32 lars-desktop pulseaudio[1524]: module.c: >>>> module-detect >>>> >> is >>>> >> > deprecated: Please use module-udev-detect instead of module-detect! >>>> >> > Nov 2 10:57:37 lars-desktop pulseaudio[1524]: ratelimit.c: 27 >>>> events >>>> >> > suppressed >>>> >> > Nov 2 10:57:39 lars-desktop pulseaudio[1612]: module.c: >>>> module-detect >>>> >> is >>>> >> > deprecated: Please use module-udev-detect instead of module-detect! >>>> >> > Nov 2 10:57:44 lars-desktop pulseaudio[1612]: ratelimit.c: 23 >>>> events >>>> >> > suppressed >>>> >> > Nov 2 10:57:44 lars-desktop pulseaudio[1612]: alsa-sink.c: ALSA >>>> woke us >>>> >> up >>>> >> > to write new data to the device, but there was actually nothing to >>>> >> write! >>>> >> > Nov 2 10:57:44 lars-desktop pulseaudio[1612]: alsa-sink.c: *Most >>>> likely >>>> >> > this is a bug in the ALSA driver 'snd_hdsp'. Please report this >>>> issue to >>>> >> > the >>>> >> > ALSA developers.* >>>> >> > Nov 2 10:57:44 lars-desktop pulseaudio[1612]: alsa-sink.c: We were >>>> >> woken >>>> >> > up >>>> >> > with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 >>>> or >>>> >> > another value < min_avail. >>>> >> > Nov 2 10:57:49 lars-desktop pulseaudio[1612]: ratelimit.c: 99 >>>> events >>>> >> > suppressed >>>> >> > >>>> >> > Later if I try to play audio I get terrible skipping or stuttering >>>> and >>>> >> my >>>> >> > syslog (in less than a minute) writes: >>>> >> > Nov 2 11:55:16 lars-desktop pulseaudio[1612]: ratelimit.c: 16 >>>> events >>>> >> > suppressed >>>> >> > Nov 2 11:55:21 lars-desktop pulseaudio[1612]: ratelimit.c: 34 >>>> events >>>> >> > suppressed >>>> >> > Nov 2 11:55:26 lars-desktop pulseaudio[1612]: ratelimit.c: 70 >>>> events >>>> >> > suppressed >>>> >> > Nov 2 11:55:31 lars-desktop pulseaudio[1612]: ratelimit.c: 29 >>>> events >>>> >> > suppressed >>>> >> > Nov 2 11:55:36 lars-desktop pulseaudio[1612]: ratelimit.c: 43 >>>> events >>>> >> > suppressed >>>> >> > Nov 2 11:55:41 lars-desktop pulseaudio[1612]: ratelimit.c: 161 >>>> events >>>> >> > suppressed >>>> >> > Nov 2 11:55:46 lars-desktop pulseaudio[1612]: ratelimit.c: 28 >>>> events >>>> >> > suppressed >>>> >> > Nov 2 11:55:51 lars-desktop pulseaudio[1612]: ratelimit.c: 18 >>>> events >>>> >> > suppressed >>>> >> > Nov 2 11:55:56 lars-desktop pulseaudio[1612]: ratelimit.c: 16 >>>> events >>>> >> > suppressed >>>> >> > >>>> >> > Here's the relevant part (concerning my sound card) of lspci -vvv >>>> >> > >>>> >> > 02:09.0 Multimedia audio controller: Xilinx Corporation RME >>>> Hammerfall >>>> >> DSP >>>> >> > (rev 9a) >>>> >> > Control: I/O- Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- >>>> ParErr- >>>> >> > Stepping- SERR- FastB2B- DisINTx- >>>> >> > Status: Cap- 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- >>>> >> > <TAbort- <MAbort- >SERR- <PERR- INTx- >>>> >> > Latency: 255 >>>> >> > Interrupt: pin A routed to IRQ 19 >>>> >> > Region 0: Memory at fdff0000 (32-bit, non-prefetchable) >>>> [size=64K] >>>> >> > Kernel driver in use: RME Hammerfall DSP >>>> >> > Kernel modules: snd-hdsp >>>> >> > >>>> >> > And: >>>> >> > lars@lars-desktop:~$ aplay -l >>>> >> > **** List of PLAYBACK Hardware Devices **** >>>> >> > card 0: DSP [Hammerfall DSP], device 0: RME Hammerfall HDSP 9632 >>>> [RME >>>> >> > Hammerfall HDSP 9632] >>>> >> > Subdevices: 1/1 >>>> >> > Subdevice #0: subdevice #0 >>>> >> > >>>> >> > >>>> >> > lars@lars-desktop:~$ cat /proc/asound/version >>>> >> > Advanced Linux Sound Architecture Driver Version 1.0.21. >>>> >> > Compiled on Oct 15 2009 for kernel 2.6.31-14-generic (SMP). >>>> >> > >>>> >> > lars@lars-desktop:~$ cat /proc/asound/cards >>>> >> > 0 [DSP ]: H-DSP - Hammerfall DSP >>>> >> > RME Hammerfall HDSP 9632 at 0xfdff0000, irq 19 >>>> >> > >>>> >> > >>>> >> > lars@lars-desktop:~$ cat /proc/asound/modules >>>> >> > 0 snd_hdsp >>>> >> > >>>> >> > I don't have .asound.conf files. >>>> >> > >>>> >> > lars@lars-desktop:~$ hdsploader >>>> >> > hdsploader - firmware loader for RME Hammerfall DSP cards >>>> >> > Looking for HDSP + Multiface or Digiface cards : >>>> >> > Card 0 : RME Hammerfall HDSP 9632 at 0xfdff0000, irq 19 >>>> >> > >>>> >> > And of course. My computer makes no sound if I don't load the >>>> hdspmixer >>>> >> at >>>> >> > startup. I guess that's just normal for this card though. >>>> >> > >>>> >> > Please let me know, if you need more info. I really hope you can >>>> help >>>> >> out >>>> >> > here. And that this might help you to. >>>> >> > >>>> >> > Thanks for the great work you do. Greetings. >>>> >> > Lars Zobbe. >>>> >> > _______________________________________________ >>>> >> > Alsa-devel mailing list >>>> >> > Alsa-devel(a)alsa-project.org >>>> >> > http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >>>> >> > >>>> >> _______________________________________________ >>>> >> Alsa-devel mailing list >>>> >> Alsa-devel(a)alsa-project.org >>>> >> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >>>> >> >>>> > >>>> > >>>> _______________________________________________ >>>> Alsa-devel mailing list >>>> Alsa-devel(a)alsa-project.org >>>> http://mailman.alsa-project.org/mailman/listinfo/alsa-devel >>>> >>> >>> >> >

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[alsa-devel] [PATCH] ASoC: TWL4030: Fix typo in comment in header file
by Thomas Weber 14 Jan '10

14 Jan '10

Signed-off-by: Thomas Weber <weber(a)corscience.de> --- sound/soc/codecs/twl4030.h | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/twl4030.h b/sound/soc/codecs/twl4030.h index dd6396e..f206d24 100644 --- a/sound/soc/codecs/twl4030.h +++ b/sound/soc/codecs/twl4030.h @@ -25,7 +25,7 @@ /* Register descriptions are here */ #include <linux/mfd/twl4030-codec.h> -/* Sgadow register used by the audio driver */ +/* Shadow register used by the audio driver */ #define TWL4030_REG_SW_SHADOW 0x4A #define TWL4030_CACHEREGNUM (TWL4030_REG_SW_SHADOW + 1) -- 1.6.4.4

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Re: [alsa-devel] ALSA DMIX support without MMAP ?
by ANISH KUMAR 14 Jan '10

14 Jan '10

>mmap is supported by the Samsung drivers. Thanks for this information which i overlooked. >This looks like the dmix data rates are getting out of sync with what >the hardware is doing. Enabling CONFIG_SND_PCM_XRUN_DEBUG will produce >more diagnostic information from the driver side I got the problem and after doing the changes DMIX plugin is working very fine. As you know XRUN was happening and this happening in snd_pcm_update_hw_ptr_post function. This happened because stop_threshold is increased suddenly(Is it done by dmix? If yes then why ?) ?????? In normal condtions this would have been fine but in my code i was multiplying it by some amount causing it to wrap around(stop_threshold=0 );Causing XRUN.So i just removed this multiplying factor. One more Rate conversion query: ./alsa_aplay -v -D"plug:'dmix:RATE=44100'" bond.wav Playing WAVE 'bond.wav' : Unsigned 8 bit, Rate 11025 Hz, Mono Plug PCM: Rate conversion PCM (44100, sformat=U8) As you can see bond.wav is sampled at 11025k but when i am playing with Plug PCM there is some noise in the song why????Is rate conversion algoritm not able to resample it properly? If i play with ./alsa_aplay -v -D"plug:'dmix:RATE=11025'" bond.wav then no noise.Probably because no resampling happening? Mark Brown Heartfelt Thanks for your time.

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Re: [alsa-devel] ALSA DMIX support without MMAP ?
by ANISH KUMAR 14 Jan '10

14 Jan '10

2 1

[alsa-devel] Linux kernel 2.6.32.3 bug report
by d binderman 14 Jan '10

14 Jan '10

Hello there, I just ran the sourceforge tool cppcheck over the source code of the Linux kernel 2.6.32.3 It said [./sound/soc/s3c24xx/s3c24xx-pcm.c:128]: (error) Possible null pointer dereference: substream - otherwise it is redundant to check if substream is null at line 130 I checked the source code and I agree with the compiler. Suggest code rework. Regards David Binderman _________________________________________________________________ Do you have a story that started on Hotmail? Tell us now http://clk.atdmt.com/UKM/go/195013117/direct/01/

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[alsa-devel] capturing data from the microphone
by Riccardo Magliocchetti 14 Jan '10

14 Jan '10

[the same message is waiting in moderator queue, sorry if you receive it two times] Hello, i'm trying to capture audio data from the microphone, but i'm not able to read any frames. The frames are always 0 and delay is fixed to 2730. I've tried calling snd_pcm_avail_delay, snd_pcm_avail_update but does not make a difference. The soundcard is driven by snd_hda_intel. The code is more or less the same as Paul Davis' tutorial on using the alsa api. This is the code for init and the loop that tries to read the data: #define MIC_BUFSIZE 4096 static gpointer snd_pcm_read(gpointer data) { int error; snd_pcm_sframes_t frames = MIC_BUFSIZE; while (TRUE) { if ((error = snd_pcm_wait (pcm_handle, 1000)) < 0) { g_printerr("Failed to poll: %s\n", snd_strerror(error)); continue; } if ((error = snd_pcm_avail(pcm_handle)) < 0) { if (error == -EPIPE) { g_printerr("xrun! %s\n", snd_strerror(error)); continue; //return GINT_TO_POINTER(FALSE); } else { g_printerr("alsa_pcm_avail_update error %s\n", snd_strerror(error)); continue; //return GINT_TO_POINTER(FALSE); } } frames = error; if (frames == 0) continue; frames = frames > MIC_BUFSIZE ? MIC_BUFSIZE : frames; g_printerr ("frames: %ld\n", frames); g_static_mutex_lock(&mutex); error = snd_pcm_readi(pcm_handle, Mic_Buffer[Mic_WriteBuf], frames); g_static_mutex_unlock(&mutex); if (error < 0) error = snd_pcm_recover(pcm_handle, error, 0); if (error < 0) { LOG("snd_pcm_readi FAIL!: %s\n", snd_strerror(error)); } } return GINT_TO_POINTER(TRUE); } BOOL Mic_Init() { snd_pcm_hw_params_t *hwparams; snd_pcm_sw_params_t *swparams; int err; if (Mic_Inited) return TRUE; // Open the default sound card in capture if ((err = snd_pcm_open(&pcm_handle, "default", SND_PCM_STREAM_CAPTURE, /*SND_PCM_NONBLOCK*/ 0)) < 0) { g_printerr("Failed to open device: %s\n", snd_strerror(err)); return FALSE; } // Allocate the snd_pcm_hw_params_t structure and fill it. snd_pcm_hw_params_alloca(&hwparams); if ((err = snd_pcm_hw_params_any(pcm_handle, hwparams)) < 0) { g_printerr("Failed to setup hw parameters: %s\n", snd_strerror(err)); return FALSE; } //Set the access if ((err = snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) { g_printerr("Failed to set access: %s\n", snd_strerror(err)); return FALSE; } //dir 0 == exacte (Rate = 16K exacte) if ((err = snd_pcm_hw_params_set_rate(pcm_handle, hwparams, 16000, 0)) < 0) { g_printerr("Failed to set rate: %s\n", snd_strerror(err)); return FALSE; } /* Set sample format */ if ((err = snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S8)) < 0) { g_printerr("Failed to set format: %s\n", snd_strerror(err)); return FALSE; } // Set one channel (mono) if ((err = snd_pcm_hw_params_set_channels(pcm_handle, hwparams, 1)) < 0) { g_printerr("Failed to set channels: %s\n", snd_strerror(err)); return FALSE; } //Set the params if ((err = snd_pcm_hw_params(pcm_handle, hwparams)) < 0) { g_printerr("Failed to set hw parameters: %s\n", snd_strerror(err)); return FALSE; } snd_pcm_sw_params_alloca(&swparams); if ((err = snd_pcm_sw_params_current (pcm_handle, swparams)) < 0) { g_printerr("Failed to set current sw parameters: %s\n", snd_strerror(err)); return FALSE; } if ((err = snd_pcm_sw_params_set_avail_min (pcm_handle, swparams, MIC_BUFSIZE)) < 0) { g_printerr("Failed to set minimum available count: %s\n", snd_strerror(err)); return FALSE; } if ((err = snd_pcm_sw_params_set_start_threshold (pcm_handle, swparams, 0U)) < 0) { g_printerr("Failed to set start mode: %s\n", snd_strerror(err)); return FALSE; } if ((err = snd_pcm_sw_params (pcm_handle, swparams)) < 0) { g_printerr("Failed to set sw parameters: %s\n", snd_strerror(err)); return FALSE; } if ((err = snd_pcm_prepare (pcm_handle)) < 0) { g_printerr("Failed to prepare audio interface to use: %s\n", snd_strerror(err)); return FALSE; } Mic_Inited = TRUE; Mic_Reset(); mic_reader = g_thread_create(snd_pcm_read, NULL, TRUE, NULL); return TRUE; } thanks, Riccardo

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[alsa-devel] alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write!
by Konrad Schlichtherle 14 Jan '10

14 Jan '10

Hello, sometimes it occure that my speakers do not play sound although the speakers should play music with VLC. Then i took a look at /var/log/messages and found this: Jan 13 17:01:39 riker pulseaudio[3335]: alsa-sink.c: ALSA woke us up to write new data to the device, but there was actually nothing to write! Jan 13 17:01:39 riker pulseaudio[3335]: alsa-sink.c: Most likely this is a bug in the ALSA driver 'snd_hda_intel'. Please report this issue to the ALSA developers. Jan 13 17:01:39 riker pulseaudio[3335]: alsa-sink.c: We were woken up with POLLOUT set -- however a subsequent snd_pcm_avail() returned 0 or another value < min_avail. The output of utils_alsa-info.sh is there: http://www.alsa-project.org/db/?f=1767e9bc745724a67c908549d20c137375d0605e I didn't know wether this is a bug or a misconfiguration of my openSUSE 11.2 Konrad

2 1

Re: [alsa-devel] [PATCH] OMAP: Exporting functions doing common register access
by Aggarwal, Anuj 13 Jan '10

13 Jan '10

> -----Original Message----- > From: Paul Walmsley [mailto:paul@pwsan.com] > Sent: Tuesday, November 17, 2009 2:46 AM > To: Aggarwal, Anuj > Cc: linux-omap(a)vger.kernel.org > Subject: Re: [PATCH] OMAP: Exporting functions doing common register > access > > Hello Anuj, > > On Mon, 16 Nov 2009, Anuj Aggarwal wrote: > > > These functions need to be exported so that drivers (e.g. McBSP) can > > be configured as modules. > > > > McBSP driver gets built as a module when ASoC driver for OMAP3 EVM > > is configured as module. McBSP driver uses functions like > > omap_ctrl_readl/omap_ctrl_writel, which are defined in control.c > > file but not exported. Without that, McBSP driver fails to build as > > a module. > > > > Signed-off-by: Anuj Aggarwal <anuj.aggarwal(a)ti.com> > > This will prevent the McBSP driver from being usable across chips. For > example, if a future OMAP comes out with McBSP clock source switching > implemented in a different way, or if a DaVinci chip comes out with a > McBSP block but no SCM block, this either won't build or won't run. > > A better approach would be to move the OMAP2/3-specific functions from the > McBSP driver into the OMAP McBSP integration code, > arch/arm/*omap*/mcbsp.c, and to define function pointers in > struct platform_data for these functions, and to pass them from the > integration code into the device driver code. That will make these > drivers independent of chip integration details. Sorry to confuse you, I was talking about the platform specific ASoC-McBSP code for OMAP (file - sound/soc/omap/omap-mcbsp.c) platform. The McBSP driver - both platform and machine specific - doesn't use the functions exported by the patch. Including alsa-devel list too... > > > - Paul

3 7

[alsa-devel] Newbie question: Sending untimed MIDI data to ALSA seq
by Josh Lehan 13 Jan '10

13 Jan '10

Hello. Newbie question: I'm wondering about sending untimed MIDI data to the ALSA sequencer subsystem. The application would be for a "beep"-like program, from the command line, that instead of making beeps, would play MIDI instruments to make better-sounding beeps. I have TiMidity hooked up into ALSA, as a softsynth, and it works just fine for playing MIDI data sent to it by ALSA. The "aplaymidi" command works for .MID files, and the "vkeybd" program works for simulating a keyboard and sending in untimed MIDI data. The "aseqview" program also works great for watching the notes being played, as they go by. So, my MIDI software stack appears to be in order. I don't have a hardware synthesizer hooked up at the moment (my Kawai K1 is showing its age, and a few keys are broken). I also looked at "amidi", which features a -S option to send in untimed MIDI bytes, but unfortunately, that addresses an entirely different namespace. Evidently, it only speaks to "RawMIDI" devices, of which TiMidity is not. So, I can't generate untimed MIDI data on my own. I was thinking of writing a small "midibeep" program that would do this, similar to what vkeybd does, but without requiring any user interaction. Does such a program already exist? Thanks! Josh

3 4

[alsa-devel] Mic Boost control problem with snd-hda-intel codec AL662 rev 1
by Anisse Astier 13 Jan '10

13 Jan '10

Hi, The Mic Boost control on a laptop seems to behave incorrectly. The control has the flag "volume" instead of "cvolume", causing it to appear in both the "Playback" and "Capture" tabs of alsamixer. This wouldn't be a problem if it would just work in the Capture tab: it doesn't. When I modify it in Capture, it doesn't retain its value: it's gone after i close alsamixer. But if i modify it in "Playback" it will retain its value and the value will be mirrored by the Mic Boost in Capture (both Mic Boost seem to reflect the same control, except one is RO, the other is RW). It's a minor a annoyance, but I'd like to fix it and have "Mic Boost" appear only as a Capture control and retaining its value. Maybe I should create a new mixer configuration for this hardware in patch_realtek.c, but I don't know where to start. Bios autoprobing works for everything but just what I mentioned. My alsa-info.sh : http://www.alsa-project.org/db/?f=fc17b509e532fc366eb2d5fe5ec9e6730c96c837 Any ideas? Where should I start? Regards, Anisse

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[alsa-devel] [PATCH] ASoC: TWL4030: Replace comma with semicolon in probe function
by Peter Ujfalusi 13 Jan '10

13 Jan '10

The codec structure initialization statements should be separated by semicolons. Signed-off-by: Peter Ujfalusi <peter.ujfalusi(a)nokia.com> --- sound/soc/codecs/twl4030.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/sound/soc/codecs/twl4030.c b/sound/soc/codecs/twl4030.c index 2a27f7b..74f0d65 100644 --- a/sound/soc/codecs/twl4030.c +++ b/sound/soc/codecs/twl4030.c @@ -2192,7 +2192,7 @@ static int __devinit twl4030_codec_probe(struct platform_device *pdev) codec->write = twl4030_write; codec->set_bias_level = twl4030_set_bias_level; codec->dai = twl4030_dai; - codec->num_dai = ARRAY_SIZE(twl4030_dai), + codec->num_dai = ARRAY_SIZE(twl4030_dai); codec->reg_cache_size = sizeof(twl4030_reg); codec->reg_cache = kmemdup(twl4030_reg, sizeof(twl4030_reg), GFP_KERNEL); -- 1.6.6

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[alsa-devel] [PATCH] ALSA: ASoC: ak4104: allow more sample rates
by Daniel Mack 13 Jan '10

13 Jan '10

The transmitter supports all sample rates up to 192KHz, so the driver should not give a limit. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> Cc: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- sound/soc/codecs/ak4104.c | 4 +--- 1 files changed, 1 insertions(+), 3 deletions(-) diff --git a/sound/soc/codecs/ak4104.c b/sound/soc/codecs/ak4104.c index 3a14c6f..b9ef7e4 100644 --- a/sound/soc/codecs/ak4104.c +++ b/sound/soc/codecs/ak4104.c @@ -185,9 +185,7 @@ struct snd_soc_dai ak4104_dai = { .stream_name = "Playback", .channels_min = 2, .channels_max = 2, - .rates = SNDRV_PCM_RATE_44100 | - SNDRV_PCM_RATE_48000 | - SNDRV_PCM_RATE_32000, + .rates = SNDRV_PCM_RATE_8000_192000, .formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_3LE | SNDRV_PCM_FMTBIT_S24_LE -- 1.6.5.2

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[alsa-devel] Personal invitation from Kunal Narkhede
by Kunal Narkhede 13 Jan '10

13 Jan '10

Hi, On 12/24/2009 9:02:23 AM, you were invited to join Kunal Narkhede's UNYK address book so he/she would always have access to your contact info and you to his/hers. To accept his/her request, Click here. http://www.unyk.com/ml/65/5/?i=C7B72DD0179E4FD7A6DED23A2976A060&s=17267503 UNYK is a smart and simple way to manage your contacts so you never lose anyones contact information again. No more worrying about your contacts' info. From now on, let them do it for you. By changing their information in UNYK.com, you address book will be automatically updated. By changing your information in UNYK.com, their address book will be automatically updated. Its so simple and its free... Already 25 million users. Please do not reply to this email. Questions ? Contact us - http://unyk.com/en/Contact UNYK Inc., 1010 de Serigny, suite 820, Longueuil, QC, J4K 5G7, Canada. I no longer wish to receive invitations to join UNYK from Kunal Narkhede: Unsubscribe http://www.unyk.com/ml/245/74/?mid=9E98A4AA807983A0&email=alsa-devel@alsa-p… I never again wish to receive invitation to join UNYK: Unsubscribe http://www.unyk.com/ml/65/6/?i=C7B72DD0179E4FD7A6DED23A2976A060&s=17267503 UNYK, the first smart address book that updates itself!

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[alsa-devel] [PATCH - hda 1/1] ALSA: hda - Improved MacBook (Pro) 5, 1 / 5, 2 support
by murray.alex＠gmail.com 13 Jan '10

13 Jan '10

From: Alex Murray <murray.alex(a)gmail.com> This patch adds support for automatically muting the speakers when headphones are inserted, as well as relabelling the headphone widgets from the non-standard "HP" to the standard "Headphone" for the mb5 model. Signed-off-by: Alex Murray <murray.alex(a)gmail.com> --- pci/hda/patch_realtek.c | 28 ++++++++++++++++++++++++++-- 1 files changed, 26 insertions(+), 2 deletions(-) diff --git a/pci/hda/patch_realtek.c b/pci/hda/patch_realtek.c index f858dcd..bd65e47 100644 --- a/pci/hda/patch_realtek.c +++ b/pci/hda/patch_realtek.c @@ -7201,8 +7201,8 @@ static struct snd_kcontrol_new alc885_mb5_mixer[] = { HDA_BIND_MUTE ("Surround Playback Switch", 0x0d, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("LFE Playback Volume", 0x0e, 0x00, HDA_OUTPUT), HDA_BIND_MUTE ("LFE Playback Switch", 0x0e, 0x02, HDA_INPUT), - HDA_CODEC_VOLUME("HP Playback Volume", 0x0f, 0x00, HDA_OUTPUT), - HDA_BIND_MUTE ("HP Playback Switch", 0x0f, 0x02, HDA_INPUT), + HDA_CODEC_VOLUME("Headphone Playback Volume", 0x0f, 0x00, HDA_OUTPUT), + HDA_BIND_MUTE ("Headphone Playback Switch", 0x0f, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Line Playback Volume", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_MUTE ("Line Playback Switch", 0x0b, 0x02, HDA_INPUT), HDA_CODEC_VOLUME("Mic Playback Volume", 0x0b, 0x01, HDA_INPUT), @@ -7603,6 +7603,7 @@ static struct hda_verb alc885_mb5_init_verbs[] = { {0x14, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_OUT}, {0x14, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_UNMUTE}, {0x14, AC_VERB_SET_CONNECT_SEL, 0x03}, + {0x14, AC_VERB_SET_UNSOLICITED_ENABLE, ALC880_HP_EVENT | AC_USRSP_EN}, /* Front Mic pin: input vref at 80% */ {0x19, AC_VERB_SET_PIN_WIDGET_CONTROL, PIN_VREF80}, {0x19, AC_VERB_SET_AMP_GAIN_MUTE, AMP_OUT_MUTE}, @@ -7787,6 +7788,27 @@ static void alc885_mbp3_setup(struct hda_codec *codec) spec->autocfg.speaker_pins[0] = 0x14; } +static void alc885_mb5_automute(struct hda_codec *codec) +{ + unsigned int present; + + present = snd_hda_codec_read(codec, 0x14, 0, + AC_VERB_GET_PIN_SENSE, 0) & 0x80000000; + snd_hda_codec_amp_stereo(codec, 0x18, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + snd_hda_codec_amp_stereo(codec, 0x1a, HDA_OUTPUT, 0, + HDA_AMP_MUTE, present ? HDA_AMP_MUTE : 0); + +} + +static void alc885_mb5_unsol_event(struct hda_codec *codec, + unsigned int res) +{ + /* Headphone insertion or removal. */ + if ((res >> 26) == ALC880_HP_EVENT) + alc885_mb5_automute(codec); +} + static void alc885_imac91_automute(struct hda_codec *codec) { unsigned int present; @@ -9233,6 +9255,8 @@ static struct alc_config_preset alc882_presets[] = { .input_mux = &mb5_capture_source, .dig_out_nid = ALC882_DIGOUT_NID, .dig_in_nid = ALC882_DIGIN_NID, + .unsol_event = alc885_mb5_unsol_event, + .init_hook = alc885_mb5_automute, }, [ALC885_MACPRO] = { .mixers = { alc882_macpro_mixer }, -- 1.6.3.3

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[alsa-devel] ALSA DMIX support without MMAP ?
by ANISH KUMAR 13 Jan '10

13 Jan '10

Hi, I am working on android and and in our device we use ALSA. I have enabled dmix support for andoid as by default android doesn't have this plugin enabled and soon will post it in android forum however i have run into some problems.Hoping takashi can help me out on this. Problem is when i am trying to use dmix plugin i am not getting any sound (logs shown below).Is it because i don't have mmap mode support on my device??? If yes then can't dmix work without mmap mode??? Logs shown below: Entered s3c24xx_pcm_hw_params, Entered s3c24xx_pcm_hw_params, line 330 Entered s3c24xx_pcm_mmap Entered s3c24xx_pcm_prepare Entered s3c24xx_pcm_enqueue Entered s3c24xx_pcm_trigger Entered s3c24xx_pcm_pointer Entered s3c24xx_pcm_pointer res_value[0] snd_pcm_update_hw_ptr_pos xun: [0]occured XRUN! Normal playback and recording is fine if i dont use dmix plugin.During my normal playback and recording i don't use mmap as i preallocate DMA memory and use them. Regards, anish

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[alsa-devel] [2.6.33-rc3] HDA works in polling mode
by Tvrtko Ursulin 13 Jan '10

13 Jan '10

Hi, I have a AMD 785G motherboard where there is a worrying situation logged by ALSA drivers. Sound nevertheless works. lspci: 00:14.2 Audio device: ATI Technologies Inc SBx00 Azalia (Intel HDA) dmesg: HDA Intel 0000:01:05.1: PCI INT B disabled HDA Intel 0000:00:14.2: PCI INT A disabled HDA Intel 0000:00:14.2: PCI INT A -> GSI 16 (level, low) -> IRQ 16 HDA Intel 0000:01:05.1: PCI INT B -> GSI 19 (level, low) -> IRQ 19 HDA Intel 0000:01:05.1: irq 28 for MSI/MSI-X HDA Intel 0000:01:05.1: setting latency timer to 64 hda-intel: azx_get_response timeout, switching to polling mode: last cmd=0x000f0000 I tried position_fix=0, 1 and 2 with no difference. I also tried enable_msi=0 to make it use IO-APIC interrupt. Then this error is not displayed, sound works, but again no interrupts are generated at least count is not growing in /proc/interrupts. Please tell me what else I can try and what debugging info you need. Thanks, Tvrtko

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[alsa-devel] No CD audio with kernel 2.6.32.3
by Chris Clayton 12 Jan '10

12 Jan '10

Hi, Please cc me on any reply - I'm not subscribed. Except for CD audio, sound is working fine on my new HP G61 laptop with kernel 2.6.32.3. I can play CD audio fine in Windows. IIRC, on my (very) old Dell laptop, alsamixer showed a CD audio device, but that's not the case on the G61. I have asla built as modules. Diagnostics: [chris:~]$ lsmod | grep snd snd_hda_codec_intelhdmi 10250 1 snd_hda_codec_idt 40146 1 snd_hda_intel 17040 1 snd_hda_codec 46119 3 snd_hda_codec_intelhdmi,snd_hda_codec_idt,snd_hda_intel snd_pcm 51424 2 snd_hda_intel,snd_hda_codec snd_timer 15133 1 snd_pcm snd 37459 8 snd_hda_codec_intelhdmi,snd_hda_codec_idt,snd_hda_intel,snd_hda_codec,snd_pcm,snd_timer snd_page_alloc 5657 2 snd_hda_intel,snd_pcm [chris:~]$ aplay -l **** List of PLAYBACK Hardware Devices **** card 0: Intel [HDA Intel], device 0: STAC92xx Analog [STAC92xx Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: Intel [HDA Intel], device 3: INTEL HDMI [INTEL HDMI] Subdevices: 1/1 Subdevice #0: subdevice #0 [chris:~]$ amixer controls numid=10,iface=MIXER,name='Master Playback Switch' numid=9,iface=MIXER,name='Master Playback Volume' numid=5,iface=MIXER,name='Headphone Playback Switch' numid=4,iface=MIXER,name='Headphone Playback Volume' numid=15,iface=MIXER,name='PCM Playback Volume' numid=3,iface=MIXER,name='Mic Jack Mode' numid=7,iface=MIXER,name='Capture Switch' numid=6,iface=MIXER,name='Capture Volume' numid=11,iface=MIXER,name='IEC958 Playback Con Mask' numid=12,iface=MIXER,name='IEC958 Playback Pro Mask' numid=13,iface=MIXER,name='IEC958 Playback Default' numid=14,iface=MIXER,name='IEC958 Playback Switch' numid=8,iface=MIXER,name='Mux Capture Volume' numid=2,iface=MIXER,name='Speaker Playback Switch' numid=1,iface=MIXER,name='Speaker Playback Volume' [chris:~]$ amixer scontrols Simple mixer control 'Master',0 Simple mixer control 'Headphone',0 Simple mixer control 'Speaker',0 Simple mixer control 'PCM',0 Simple mixer control 'Mic Jack Mode',0 Simple mixer control 'IEC958',0 Simple mixer control 'Capture',0 Simple mixer control 'Mux',0 Extract from dmesg: HDA Intel 0000:00:1b.0: PCI INT A -> GSI 22 (level, low) -> IRQ 22 HDA Intel 0000:00:1b.0: setting latency timer to 64 Let me know if there is any other information I can provide. Thanks Chris -- No, Sir; there is nothing which has yet been contrived by man, by which so much happiness is produced as by a good tavern or inn - Doctor Samuel Johnson

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[alsa-devel] X-Fi notebook support
by The Source 12 Jan '10

12 Jan '10

Hello. I have some info on that card, but not much. I have discovered that this card is detected connected via USB rather than PCI-E (the slot is ExpressCard). snd_ctxfi uses PCI interfaces right? Can X-Fi support be also added to snd_usb_audio? Or can snd_ctxfi support USB? lsusb -v output on that card is attached (sorry for duplicate mail, forgot to attach it first time). I now have this card and willing to help to test (or perhaps create) proper driver.

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[alsa-devel] ASoC updates for 2.6.33
by Mark Brown 12 Jan '10

12 Jan '10

The following changes since commit 7f50548abb5454bd82c25aae15f0a3bf6a530f46: Mark Brown (1): Merge commit 'v2.6.33-rc2' into for-2.6.33 are available in the git repository at: git://git.kernel.org/pub/scm/linux/kernel/git/broonie/sound-2.6.git for-2.6.33 Manuel Lauss (1): ASoC: fixup oops in generic AC97 codec glue Mark Brown (1): ASoC: Fix WM8350 DSP mode B configuration sound/soc/codecs/ac97.c | 6 ++++++ sound/soc/codecs/wm8350.c | 2 +- 2 files changed, 7 insertions(+), 1 deletions(-)

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[alsa-devel] hda-intel: no codecs found
by Ozan Çağlayan 12 Jan '10

12 Jan '10

Hi, Does that warning can be interpreted as a hardware failure? I'm reverting to older alsa-driver snapshots that I'm sure was working OK but still having this warning. I'm suspecting the power outage that the system suffered on Friday. Is there a way to understand whether it's driver related or HW related? Thanks, Ozan

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[alsa-devel] [PATCH 1/4] ASoC: Add a new imx-ssi sound driver
by Mark Brown 11 Jan '10

11 Jan '10

From: Sascha Hauer <s.hauer(a)pengutronix.de> The old driver has the number of SSI units in the system hardcoded, does not make use of the device model and works only on i.MX21/27. This driver replaces it. It works in DMA mode on i.MX21/27 and using an FIQ handler on other systems. It also supports AC97 mode of the SSI units. Signed-off-by: Sascha Hauer <s.hauer(a)pengutronix.de> Signed-off-by: Mark Brown <broonie(a)opensource.wolfsonmicro.com> --- arch/arm/plat-mxc/Makefile | 6 + arch/arm/plat-mxc/ssi-fiq-ksym.c | 20 + arch/arm/plat-mxc/ssi-fiq.S | 134 +++++++ sound/soc/imx/Kconfig | 20 +- sound/soc/imx/Makefile | 12 +- sound/soc/imx/imx-pcm-dma-mx2.c | 313 ++++++++++++++++ sound/soc/imx/imx-pcm-fiq.c | 277 ++++++++++++++ sound/soc/imx/imx-ssi.c | 762 ++++++++++++++++++++++++++++++++++++++ sound/soc/imx/imx-ssi.h | 238 ++++++++++++ 9 files changed, 1762 insertions(+), 20 deletions(-) create mode 100644 arch/arm/plat-mxc/ssi-fiq-ksym.c create mode 100644 arch/arm/plat-mxc/ssi-fiq.S create mode 100644 sound/soc/imx/imx-pcm-dma-mx2.c create mode 100644 sound/soc/imx/imx-pcm-fiq.c create mode 100644 sound/soc/imx/imx-ssi.c create mode 100644 sound/soc/imx/imx-ssi.h diff --git a/arch/arm/plat-mxc/Makefile b/arch/arm/plat-mxc/Makefile index e3212c8..b0b9fc3 100644 --- a/arch/arm/plat-mxc/Makefile +++ b/arch/arm/plat-mxc/Makefile @@ -9,3 +9,9 @@ obj-$(CONFIG_ARCH_MX1) += iomux-mx1-mx2.o dma-mx1-mx2.o obj-$(CONFIG_ARCH_MX2) += iomux-mx1-mx2.o dma-mx1-mx2.o obj-$(CONFIG_ARCH_MXC_IOMUX_V3) += iomux-v3.o obj-$(CONFIG_MXC_PWM) += pwm.o +obj-$(CONFIG_ARCH_MXC_AUDMUX_V1) += audmux-v1.o +obj-$(CONFIG_ARCH_MXC_AUDMUX_V2) += audmux-v2.o +ifdef CONFIG_SND_IMX_SOC +obj-y += ssi-fiq.o +obj-y += ssi-fiq-ksym.o +endif diff --git a/arch/arm/plat-mxc/ssi-fiq-ksym.c b/arch/arm/plat-mxc/ssi-fiq-ksym.c new file mode 100644 index 0000000..b5fad45 --- /dev/null +++ b/arch/arm/plat-mxc/ssi-fiq-ksym.c @@ -0,0 +1,20 @@ +/* + * Exported ksyms for the SSI FIQ handler + * + * Copyright (C) 2009, Sascha Hauer <s.hauer(a)pengutronix.de> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/module.h> + +#include <mach/ssi.h> + +EXPORT_SYMBOL(imx_ssi_fiq_tx_buffer); +EXPORT_SYMBOL(imx_ssi_fiq_rx_buffer); +EXPORT_SYMBOL(imx_ssi_fiq_start); +EXPORT_SYMBOL(imx_ssi_fiq_end); +EXPORT_SYMBOL(imx_ssi_fiq_base); + diff --git a/arch/arm/plat-mxc/ssi-fiq.S b/arch/arm/plat-mxc/ssi-fiq.S new file mode 100644 index 0000000..4ddce56 --- /dev/null +++ b/arch/arm/plat-mxc/ssi-fiq.S @@ -0,0 +1,134 @@ +/* + * Copyright (C) 2009 Sascha Hauer <s.hauer(a)pengutronix.de> + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include <linux/linkage.h> +#include <asm/assembler.h> + +/* + * r8 = bit 0-15: tx offset, bit 16-31: tx buffer size + * r9 = bit 0-15: rx offset, bit 16-31: rx buffer size + */ + +#define SSI_STX0 0x00 +#define SSI_SRX0 0x08 +#define SSI_SISR 0x14 +#define SSI_SIER 0x18 +#define SSI_SACNT 0x38 + +#define SSI_SACNT_AC97EN (1 << 0) + +#define SSI_SIER_TFE0_EN (1 << 0) +#define SSI_SISR_TFE0 (1 << 0) +#define SSI_SISR_RFF0 (1 << 2) +#define SSI_SIER_RFF0_EN (1 << 2) + + .text + .global imx_ssi_fiq_start + .global imx_ssi_fiq_end + .global imx_ssi_fiq_base + .global imx_ssi_fiq_rx_buffer + .global imx_ssi_fiq_tx_buffer + +imx_ssi_fiq_start: + ldr r12, imx_ssi_fiq_base + + /* TX */ + ldr r11, imx_ssi_fiq_tx_buffer + + /* shall we send? */ + ldr r13, [r12, #SSI_SIER] + tst r13, #SSI_SIER_TFE0_EN + beq 1f + + /* TX FIFO empty? */ + ldr r13, [r12, #SSI_SISR] + tst r13, #SSI_SISR_TFE0 + beq 1f + + mov r10, #0x10000 + sub r10, #1 + and r10, r10, r8 /* r10: current buffer offset */ + + add r11, r11, r10 + + ldrh r13, [r11] + strh r13, [r12, #SSI_STX0] + + ldrh r13, [r11, #2] + strh r13, [r12, #SSI_STX0] + + ldrh r13, [r11, #4] + strh r13, [r12, #SSI_STX0] + + ldrh r13, [r11, #6] + strh r13, [r12, #SSI_STX0] + + add r10, #8 + lsr r13, r8, #16 /* r13: buffer size */ + cmp r10, r13 + lslgt r8, r13, #16 + addle r8, #8 +1: + /* RX */ + + /* shall we receive? */ + ldr r13, [r12, #SSI_SIER] + tst r13, #SSI_SIER_RFF0_EN + beq 1f + + /* RX FIFO full? */ + ldr r13, [r12, #SSI_SISR] + tst r13, #SSI_SISR_RFF0 + beq 1f + + ldr r11, imx_ssi_fiq_rx_buffer + + mov r10, #0x10000 + sub r10, #1 + and r10, r10, r9 /* r10: current buffer offset */ + + add r11, r11, r10 + + ldr r13, [r12, #SSI_SACNT] + tst r13, #SSI_SACNT_AC97EN + + ldr r13, [r12, #SSI_SRX0] + strh r13, [r11] + + ldr r13, [r12, #SSI_SRX0] + strh r13, [r11, #2] + + /* dummy read to skip slot 12 */ + ldrne r13, [r12, #SSI_SRX0] + + ldr r13, [r12, #SSI_SRX0] + strh r13, [r11, #4] + + ldr r13, [r12, #SSI_SRX0] + strh r13, [r11, #6] + + /* dummy read to skip slot 12 */ + ldrne r13, [r12, #SSI_SRX0] + + add r10, #8 + lsr r13, r9, #16 /* r13: buffer size */ + cmp r10, r13 + lslgt r9, r13, #16 + addle r9, #8 + +1: + @ return from FIQ + subs pc, lr, #4 +imx_ssi_fiq_base: + .word 0x0 +imx_ssi_fiq_rx_buffer: + .word 0x0 +imx_ssi_fiq_tx_buffer: + .word 0x0 +imx_ssi_fiq_end: + diff --git a/sound/soc/imx/Kconfig b/sound/soc/imx/Kconfig index a700562..84a25e6 100644 --- a/sound/soc/imx/Kconfig +++ b/sound/soc/imx/Kconfig @@ -1,21 +1,13 @@ -config SND_MX1_MX2_SOC - tristate "SoC Audio for Freecale i.MX1x i.MX2x CPUs" - depends on ARCH_MX2 || ARCH_MX1 +config SND_IMX_SOC + tristate "SoC Audio for Freecale i.MX CPUs" + depends on ARCH_MXC select SND_PCM + select FIQ + select SND_SOC_AC97_BUS help Say Y or M if you want to add support for codecs attached to - the MX1 or MX2 SSI interface. + the i.MX SSI interface. config SND_MXC_SOC_SSI tristate -config SND_SOC_MX27VIS_WM8974 - tristate "SoC Audio support for MX27 - WM8974 Visstrim_sm10 board" - depends on SND_MX1_MX2_SOC && MACH_MX27 && MACH_IMX27_VISSTRIM_M10 - select SND_MXC_SOC_SSI - select SND_SOC_WM8974 - help - Say Y if you want to add support for SoC audio on Visstrim SM10 - board with WM8974. - - diff --git a/sound/soc/imx/Makefile b/sound/soc/imx/Makefile index c2ffd2c..4bde34a 100644 --- a/sound/soc/imx/Makefile +++ b/sound/soc/imx/Makefile @@ -1,10 +1,10 @@ # i.MX Platform Support -snd-soc-mx1_mx2-objs := mx1_mx2-pcm.o -snd-soc-mxc-ssi-objs := mxc-ssi.o +snd-soc-imx-objs := imx-ssi.o imx-pcm-fiq.o imx-pcm-dma-mx2.o -obj-$(CONFIG_SND_MX1_MX2_SOC) += snd-soc-mx1_mx2.o -obj-$(CONFIG_SND_MXC_SOC_SSI) += snd-soc-mxc-ssi.o +ifdef CONFIG_MACH_MX27 +snd-soc-imx-objs += imx-pcm-dma-mx2.o +endif + +obj-$(CONFIG_SND_IMX_SOC) += snd-soc-imx.o # i.MX Machine Support -snd-soc-mx27vis-wm8974-objs := mx27vis_wm8974.o -obj-$(CONFIG_SND_SOC_MX27VIS_WM8974) += snd-soc-mx27vis-wm8974.o diff --git a/sound/soc/imx/imx-pcm-dma-mx2.c b/sound/soc/imx/imx-pcm-dma-mx2.c new file mode 100644 index 0000000..19452e4 --- /dev/null +++ b/sound/soc/imx/imx-pcm-dma-mx2.c @@ -0,0 +1,313 @@ +/* + * imx-pcm-dma-mx2.c -- ALSA Soc Audio Layer + * + * Copyright 2009 Sascha Hauer <s.hauer(a)pengutronix.de> + * + * This code is based on code copyrighted by Freescale, + * Liam Girdwood, Javier Martin and probably others. + * + * 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. + */ +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/dma-mapping.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <mach/dma-mx1-mx2.h> + +#include "imx-ssi.h" + +struct imx_pcm_runtime_data { + int sg_count; + struct scatterlist *sg_list; + int period; + int periods; + unsigned long dma_addr; + int dma; + struct snd_pcm_substream *substream; + unsigned long offset; + unsigned long size; + unsigned long period_cnt; + void *buf; + int period_time; +}; + +/* Called by the DMA framework when a period has elapsed */ +static void imx_ssi_dma_progression(int channel, void *data, + struct scatterlist *sg) +{ + struct snd_pcm_substream *substream = data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + + if (!sg) + return; + + runtime = iprtd->substream->runtime; + + iprtd->offset = sg->dma_address - runtime->dma_addr; + + snd_pcm_period_elapsed(iprtd->substream); +} + +static void imx_ssi_dma_callback(int channel, void *data) +{ + pr_err("%s shouldn't be called\n", __func__); +} + +static void snd_imx_dma_err_callback(int channel, void *data, int err) +{ + pr_err("DMA error callback called\n"); + + pr_err("DMA timeout on channel %d -%s%s%s%s\n", + channel, + err & IMX_DMA_ERR_BURST ? " burst" : "", + err & IMX_DMA_ERR_REQUEST ? " request" : "", + err & IMX_DMA_ERR_TRANSFER ? " transfer" : "", + err & IMX_DMA_ERR_BUFFER ? " buffer" : ""); +} + +static int imx_ssi_dma_alloc(struct snd_pcm_substream *substream) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct imx_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + int ret; + + iprtd->dma = imx_dma_request_by_prio(DRV_NAME, DMA_PRIO_HIGH); + if (iprtd->dma < 0) { + pr_err("Failed to claim the audio DMA\n"); + return -ENODEV; + } + + ret = imx_dma_setup_handlers(iprtd->dma, + imx_ssi_dma_callback, + snd_imx_dma_err_callback, substream); + if (ret) + goto out; + + ret = imx_dma_setup_progression_handler(iprtd->dma, + imx_ssi_dma_progression); + if (ret) { + pr_err("Failed to setup the DMA handler\n"); + goto out; + } + + ret = imx_dma_config_channel(iprtd->dma, + IMX_DMA_MEMSIZE_16 | IMX_DMA_TYPE_FIFO, + IMX_DMA_MEMSIZE_32 | IMX_DMA_TYPE_LINEAR, + dma_params->dma, 1); + if (ret < 0) { + pr_err("Cannot configure DMA channel: %d\n", ret); + goto out; + } + + imx_dma_config_burstlen(iprtd->dma, dma_params->burstsize * 2); + + return 0; +out: + imx_dma_free(iprtd->dma); + return ret; +} + +static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + int i; + unsigned long dma_addr; + + imx_ssi_dma_alloc(substream); + + iprtd->size = params_buffer_bytes(params); + iprtd->periods = params_periods(params); + iprtd->period = params_period_bytes(params); + iprtd->offset = 0; + iprtd->period_time = HZ / (params_rate(params) / + params_period_size(params)); + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + if (iprtd->sg_count != iprtd->periods) { + kfree(iprtd->sg_list); + + iprtd->sg_list = kcalloc(iprtd->periods + 1, + sizeof(struct scatterlist), GFP_KERNEL); + if (!iprtd->sg_list) + return -ENOMEM; + iprtd->sg_count = iprtd->periods + 1; + } + + sg_init_table(iprtd->sg_list, iprtd->sg_count); + dma_addr = runtime->dma_addr; + + for (i = 0; i < iprtd->periods; i++) { + iprtd->sg_list[i].page_link = 0; + iprtd->sg_list[i].offset = 0; + iprtd->sg_list[i].dma_address = dma_addr; + iprtd->sg_list[i].length = iprtd->period; + dma_addr += iprtd->period; + } + + /* close the loop */ + iprtd->sg_list[iprtd->sg_count - 1].offset = 0; + iprtd->sg_list[iprtd->sg_count - 1].length = 0; + iprtd->sg_list[iprtd->sg_count - 1].page_link = + ((unsigned long) iprtd->sg_list | 0x01) & ~0x02; + return 0; +} + +static int snd_imx_pcm_hw_free(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + + if (iprtd->dma >= 0) { + imx_dma_free(iprtd->dma); + iprtd->dma = -EINVAL; + } + + kfree(iprtd->sg_list); + iprtd->sg_list = NULL; + + return 0; +} + +static int snd_imx_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct imx_pcm_dma_params *dma_params = rtd->dai->cpu_dai->dma_data; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + int err; + + iprtd->substream = substream; + iprtd->buf = (unsigned int *)substream->dma_buffer.area; + iprtd->period_cnt = 0; + + pr_debug("%s: buf: %p period: %d periods: %d\n", + __func__, iprtd->buf, iprtd->period, iprtd->periods); + + err = imx_dma_setup_sg(iprtd->dma, iprtd->sg_list, iprtd->sg_count, + IMX_DMA_LENGTH_LOOP, dma_params->dma_addr, + substream->stream == SNDRV_PCM_STREAM_PLAYBACK ? + DMA_MODE_WRITE : DMA_MODE_READ); + if (err) + return err; + + return 0; +} + +static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + imx_dma_enable(iprtd->dma); + + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + imx_dma_disable(iprtd->dma); + + break; + default: + return -EINVAL; + } + + return 0; +} + +static snd_pcm_uframes_t snd_imx_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + + return bytes_to_frames(substream->runtime, iprtd->offset); +} + +static struct snd_pcm_hardware snd_imx_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rate_min = 8000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = IMX_SSI_DMABUF_SIZE, + .period_bytes_min = 128, + .period_bytes_max = 16 * 1024, + .periods_min = 2, + .periods_max = 255, + .fifo_size = 0, +}; + +static int snd_imx_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd; + int ret; + + iprtd = kzalloc(sizeof(*iprtd), GFP_KERNEL); + runtime->private_data = iprtd; + + ret = snd_pcm_hw_constraint_integer(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + return ret; + + snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware); + return 0; +} + +static struct snd_pcm_ops imx_pcm_ops = { + .open = snd_imx_open, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_imx_pcm_hw_params, + .hw_free = snd_imx_pcm_hw_free, + .prepare = snd_imx_pcm_prepare, + .trigger = snd_imx_pcm_trigger, + .pointer = snd_imx_pcm_pointer, + .mmap = snd_imx_pcm_mmap, +}; + +static struct snd_soc_platform imx_soc_platform_dma = { + .name = "imx-audio", + .pcm_ops = &imx_pcm_ops, + .pcm_new = imx_pcm_new, + .pcm_free = imx_pcm_free, +}; + +struct snd_soc_platform *imx_ssi_dma_mx2_init(struct platform_device *pdev, + struct imx_ssi *ssi) +{ + ssi->dma_params_tx.burstsize = DMA_TXFIFO_BURST; + ssi->dma_params_rx.burstsize = DMA_RXFIFO_BURST; + + return &imx_soc_platform_dma; +} + diff --git a/sound/soc/imx/imx-pcm-fiq.c b/sound/soc/imx/imx-pcm-fiq.c new file mode 100644 index 0000000..5532579 --- /dev/null +++ b/sound/soc/imx/imx-pcm-fiq.c @@ -0,0 +1,277 @@ +/* + * imx-pcm-fiq.c -- ALSA Soc Audio Layer + * + * Copyright 2009 Sascha Hauer <s.hauer(a)pengutronix.de> + * + * This code is based on code copyrighted by Freescale, + * Liam Girdwood, Javier Martin and probably others. + * + * 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. + */ +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/dma-mapping.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <asm/fiq.h> + +#include <mach/ssi.h> + +#include "imx-ssi.h" + +struct imx_pcm_runtime_data { + int period; + int periods; + unsigned long dma_addr; + int dma; + unsigned long offset; + unsigned long size; + unsigned long period_cnt; + void *buf; + struct timer_list timer; + int period_time; +}; + +static void imx_ssi_timer_callback(unsigned long data) +{ + struct snd_pcm_substream *substream = (void *)data; + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + struct pt_regs regs; + + get_fiq_regs(&regs); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + iprtd->offset = regs.ARM_r8 & 0xffff; + else + iprtd->offset = regs.ARM_r9 & 0xffff; + + iprtd->timer.expires = jiffies + iprtd->period_time; + add_timer(&iprtd->timer); + snd_pcm_period_elapsed(substream); +} + +static struct fiq_handler fh = { + .name = DRV_NAME, +}; + +static int snd_imx_pcm_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + + iprtd->size = params_buffer_bytes(params); + iprtd->periods = params_periods(params); + iprtd->period = params_period_bytes(params); + iprtd->offset = 0; + iprtd->period_time = HZ / (params_rate(params) / params_period_size(params)); + + snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer); + + return 0; +} + +static int snd_imx_pcm_prepare(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + struct pt_regs regs; + + get_fiq_regs(&regs); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + regs.ARM_r8 = (iprtd->period * iprtd->periods - 1) << 16; + else + regs.ARM_r9 = (iprtd->period * iprtd->periods - 1) << 16; + + set_fiq_regs(&regs); + + return 0; +} + +static int fiq_enable; +static int imx_pcm_fiq; + +static int snd_imx_pcm_trigger(struct snd_pcm_substream *substream, int cmd) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + iprtd->timer.expires = jiffies + iprtd->period_time; + add_timer(&iprtd->timer); + if (++fiq_enable == 1) + enable_fiq(imx_pcm_fiq); + + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + del_timer(&iprtd->timer); + if (--fiq_enable == 0) + disable_fiq(imx_pcm_fiq); + + + break; + default: + return -EINVAL; + } + + return 0; +} + +static snd_pcm_uframes_t snd_imx_pcm_pointer(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + + return bytes_to_frames(substream->runtime, iprtd->offset); +} + +static struct snd_pcm_hardware snd_imx_hardware = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | + SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_PAUSE | + SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + .rate_min = 8000, + .channels_min = 2, + .channels_max = 2, + .buffer_bytes_max = IMX_SSI_DMABUF_SIZE, + .period_bytes_min = 128, + .period_bytes_max = 16 * 1024, + .periods_min = 2, + .periods_max = 255, + .fifo_size = 0, +}; + +static int snd_imx_open(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd; + int ret; + + iprtd = kzalloc(sizeof(*iprtd), GFP_KERNEL); + runtime->private_data = iprtd; + + init_timer(&iprtd->timer); + iprtd->timer.data = (unsigned long)substream; + iprtd->timer.function = imx_ssi_timer_callback; + + ret = snd_pcm_hw_constraint_integer(substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (ret < 0) + return ret; + + snd_soc_set_runtime_hwparams(substream, &snd_imx_hardware); + return 0; +} + +static int snd_imx_close(struct snd_pcm_substream *substream) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + struct imx_pcm_runtime_data *iprtd = runtime->private_data; + + del_timer_sync(&iprtd->timer); + kfree(iprtd); + + return 0; +} + +static struct snd_pcm_ops imx_pcm_ops = { + .open = snd_imx_open, + .close = snd_imx_close, + .ioctl = snd_pcm_lib_ioctl, + .hw_params = snd_imx_pcm_hw_params, + .prepare = snd_imx_pcm_prepare, + .trigger = snd_imx_pcm_trigger, + .pointer = snd_imx_pcm_pointer, + .mmap = snd_imx_pcm_mmap, +}; + +static int imx_pcm_fiq_new(struct snd_card *card, struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + int ret; + + ret = imx_pcm_new(card, dai, pcm); + if (ret) + return ret; + + if (dai->playback.channels_min) { + struct snd_pcm_substream *substream = + pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + + imx_ssi_fiq_tx_buffer = (unsigned long)buf->area; + } + + if (dai->capture.channels_min) { + struct snd_pcm_substream *substream = + pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + + imx_ssi_fiq_rx_buffer = (unsigned long)buf->area; + } + + set_fiq_handler(&imx_ssi_fiq_start, + &imx_ssi_fiq_end - &imx_ssi_fiq_start); + + return 0; +} + +static struct snd_soc_platform imx_soc_platform_fiq = { + .pcm_ops = &imx_pcm_ops, + .pcm_new = imx_pcm_fiq_new, + .pcm_free = imx_pcm_free, +}; + +struct snd_soc_platform *imx_ssi_fiq_init(struct platform_device *pdev, + struct imx_ssi *ssi) +{ + int ret = 0; + + ret = claim_fiq(&fh); + if (ret) { + dev_err(&pdev->dev, "failed to claim fiq: %d", ret); + return ERR_PTR(ret); + } + + mxc_set_irq_fiq(ssi->irq, 1); + + imx_pcm_fiq = ssi->irq; + + imx_ssi_fiq_base = (unsigned long)ssi->base; + + ssi->dma_params_tx.burstsize = 4; + ssi->dma_params_rx.burstsize = 6; + + return &imx_soc_platform_fiq; +} + +void imx_ssi_fiq_exit(struct platform_device *pdev, + struct imx_ssi *ssi) +{ + mxc_set_irq_fiq(ssi->irq, 0); + release_fiq(&fh); +} + diff --git a/sound/soc/imx/imx-ssi.c b/sound/soc/imx/imx-ssi.c new file mode 100644 index 0000000..c57a11f --- /dev/null +++ b/sound/soc/imx/imx-ssi.c @@ -0,0 +1,762 @@ +/* + * imx-ssi.c -- ALSA Soc Audio Layer + * + * Copyright 2009 Sascha Hauer <s.hauer(a)pengutronix.de> + * + * This code is based on code copyrighted by Freescale, + * Liam Girdwood, Javier Martin and probably others. + * + * 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. + * + * + * The i.MX SSI core has some nasty limitations in AC97 mode. While most + * sane processor vendors have a FIFO per AC97 slot, the i.MX has only + * one FIFO which combines all valid receive slots. We cannot even select + * which slots we want to receive. The WM9712 with which this driver + * was developped with always sends GPIO status data in slot 12 which + * we receive in our (PCM-) data stream. The only chance we have is to + * manually skip this data in the FIQ handler. With sampling rates different + * from 48000Hz not every frame has valid receive data, so the ratio + * between pcm data and GPIO status data changes. Our FIQ handler is not + * able to handle this, hence this driver only works with 48000Hz sampling + * rate. + * Reading and writing AC97 registers is another challange. The core + * provides us status bits when the read register is updated with *another* + * value. When we read the same register two times (and the register still + * contains the same value) these status bits are not set. We work + * around this by not polling these bits but only wait a fixed delay. + * + */ + +#include <linux/clk.h> +#include <linux/delay.h> +#include <linux/device.h> +#include <linux/dma-mapping.h> +#include <linux/init.h> +#include <linux/interrupt.h> +#include <linux/module.h> +#include <linux/platform_device.h> + +#include <sound/core.h> +#include <sound/initval.h> +#include <sound/pcm.h> +#include <sound/pcm_params.h> +#include <sound/soc.h> + +#include <mach/ssi.h> +#include <mach/hardware.h> + +#include "imx-ssi.h" + +#define SSI_SACNT_DEFAULT (SSI_SACNT_AC97EN | SSI_SACNT_FV) + +/* + * SSI Network Mode or TDM slots configuration. + * Should only be called when port is inactive (i.e. SSIEN = 0). + */ +static int imx_ssi_set_dai_tdm_slot(struct snd_soc_dai *cpu_dai, + unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width) +{ + struct imx_ssi *ssi = container_of(cpu_dai, struct imx_ssi, dai); + u32 sccr; + + sccr = readl(ssi->base + SSI_STCCR); + sccr &= ~SSI_STCCR_DC_MASK; + sccr |= SSI_STCCR_DC(slots - 1); + writel(sccr, ssi->base + SSI_STCCR); + + sccr = readl(ssi->base + SSI_SRCCR); + sccr &= ~SSI_STCCR_DC_MASK; + sccr |= SSI_STCCR_DC(slots - 1); + writel(sccr, ssi->base + SSI_SRCCR); + + writel(tx_mask, ssi->base + SSI_STMSK); + writel(rx_mask, ssi->base + SSI_SRMSK); + + return 0; +} + +/* + * SSI DAI format configuration. + * Should only be called when port is inactive (i.e. SSIEN = 0). + * Note: We don't use the I2S modes but instead manually configure the + * SSI for I2S because the I2S mode is only a register preset. + */ +static int imx_ssi_set_dai_fmt(struct snd_soc_dai *cpu_dai, unsigned int fmt) +{ + struct imx_ssi *ssi = container_of(cpu_dai, struct imx_ssi, dai); + u32 strcr = 0, scr; + + scr = readl(ssi->base + SSI_SCR) & ~(SSI_SCR_SYN | SSI_SCR_NET); + + /* DAI mode */ + switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + /* data on rising edge of bclk, frame low 1clk before data */ + strcr |= SSI_STCR_TFSI | SSI_STCR_TEFS | SSI_STCR_TXBIT0; + scr |= SSI_SCR_NET; + break; + case SND_SOC_DAIFMT_LEFT_J: + /* data on rising edge of bclk, frame high with data */ + strcr |= SSI_STCR_TXBIT0; + break; + case SND_SOC_DAIFMT_DSP_B: + /* data on rising edge of bclk, frame high with data */ + strcr |= SSI_STCR_TFSL; + break; + case SND_SOC_DAIFMT_DSP_A: + /* data on rising edge of bclk, frame high 1clk before data */ + strcr |= SSI_STCR_TFSL | SSI_STCR_TEFS; + break; + } + + /* DAI clock inversion */ + switch (fmt & SND_SOC_DAIFMT_INV_MASK) { + case SND_SOC_DAIFMT_IB_IF: + strcr |= SSI_STCR_TFSI; + strcr &= ~SSI_STCR_TSCKP; + break; + case SND_SOC_DAIFMT_IB_NF: + strcr &= ~(SSI_STCR_TSCKP | SSI_STCR_TFSI); + break; + case SND_SOC_DAIFMT_NB_IF: + strcr |= SSI_STCR_TFSI | SSI_STCR_TSCKP; + break; + case SND_SOC_DAIFMT_NB_NF: + strcr &= ~SSI_STCR_TFSI; + strcr |= SSI_STCR_TSCKP; + break; + } + + /* DAI clock master masks */ + switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) { + case SND_SOC_DAIFMT_CBS_CFS: + strcr |= SSI_STCR_TFDIR | SSI_STCR_TXDIR; + break; + case SND_SOC_DAIFMT_CBM_CFS: + strcr |= SSI_STCR_TFDIR; + break; + case SND_SOC_DAIFMT_CBS_CFM: + strcr |= SSI_STCR_TXDIR; + break; + } + + strcr |= SSI_STCR_TFEN0; + + writel(strcr, ssi->base + SSI_STCR); + writel(strcr, ssi->base + SSI_SRCR); + writel(scr, ssi->base + SSI_SCR); + + return 0; +} + +/* + * SSI system clock configuration. + * Should only be called when port is inactive (i.e. SSIEN = 0). + */ +static int imx_ssi_set_dai_sysclk(struct snd_soc_dai *cpu_dai, + int clk_id, unsigned int freq, int dir) +{ + struct imx_ssi *ssi = container_of(cpu_dai, struct imx_ssi, dai); + u32 scr; + + scr = readl(ssi->base + SSI_SCR); + + switch (clk_id) { + case IMX_SSP_SYS_CLK: + if (dir == SND_SOC_CLOCK_OUT) + scr |= SSI_SCR_SYS_CLK_EN; + else + scr &= ~SSI_SCR_SYS_CLK_EN; + break; + default: + return -EINVAL; + } + + writel(scr, ssi->base + SSI_SCR); + + return 0; +} + +/* + * SSI Clock dividers + * Should only be called when port is inactive (i.e. SSIEN = 0). + */ +static int imx_ssi_set_dai_clkdiv(struct snd_soc_dai *cpu_dai, + int div_id, int div) +{ + struct imx_ssi *ssi = container_of(cpu_dai, struct imx_ssi, dai); + u32 stccr, srccr; + + stccr = readl(ssi->base + SSI_STCCR); + srccr = readl(ssi->base + SSI_SRCCR); + + switch (div_id) { + case IMX_SSI_TX_DIV_2: + stccr &= ~SSI_STCCR_DIV2; + stccr |= div; + break; + case IMX_SSI_TX_DIV_PSR: + stccr &= ~SSI_STCCR_PSR; + stccr |= div; + break; + case IMX_SSI_TX_DIV_PM: + stccr &= ~0xff; + stccr |= SSI_STCCR_PM(div); + break; + case IMX_SSI_RX_DIV_2: + stccr &= ~SSI_STCCR_DIV2; + stccr |= div; + break; + case IMX_SSI_RX_DIV_PSR: + stccr &= ~SSI_STCCR_PSR; + stccr |= div; + break; + case IMX_SSI_RX_DIV_PM: + stccr &= ~0xff; + stccr |= SSI_STCCR_PM(div); + break; + default: + return -EINVAL; + } + + writel(stccr, ssi->base + SSI_STCCR); + writel(srccr, ssi->base + SSI_SRCCR); + + return 0; +} + +/* + * Should only be called when port is inactive (i.e. SSIEN = 0), + * although can be called multiple times by upper layers. + */ +static int imx_ssi_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params, + struct snd_soc_dai *cpu_dai) +{ + struct imx_ssi *ssi = container_of(cpu_dai, struct imx_ssi, dai); + u32 reg, sccr; + + /* Tx/Rx config */ + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + reg = SSI_STCCR; + cpu_dai->dma_data = &ssi->dma_params_tx; + } else { + reg = SSI_SRCCR; + cpu_dai->dma_data = &ssi->dma_params_rx; + } + + sccr = readl(ssi->base + reg) & ~SSI_STCCR_WL_MASK; + + /* DAI data (word) size */ + switch (params_format(params)) { + case SNDRV_PCM_FORMAT_S16_LE: + sccr |= SSI_SRCCR_WL(16); + break; + case SNDRV_PCM_FORMAT_S20_3LE: + sccr |= SSI_SRCCR_WL(20); + break; + case SNDRV_PCM_FORMAT_S24_LE: + sccr |= SSI_SRCCR_WL(24); + break; + } + + writel(sccr, ssi->base + reg); + + return 0; +} + +static int imx_ssi_trigger(struct snd_pcm_substream *substream, int cmd, + struct snd_soc_dai *dai) +{ + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai; + struct imx_ssi *ssi = container_of(cpu_dai, struct imx_ssi, dai); + unsigned int sier_bits, sier; + unsigned int scr; + + scr = readl(ssi->base + SSI_SCR); + sier = readl(ssi->base + SSI_SIER); + + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (ssi->flags & IMX_SSI_DMA) + sier_bits = SSI_SIER_TDMAE; + else + sier_bits = SSI_SIER_TIE | SSI_SIER_TFE0_EN; + } else { + if (ssi->flags & IMX_SSI_DMA) + sier_bits = SSI_SIER_RDMAE; + else + sier_bits = SSI_SIER_RIE | SSI_SIER_RFF0_EN; + } + + switch (cmd) { + case SNDRV_PCM_TRIGGER_START: + case SNDRV_PCM_TRIGGER_RESUME: + case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + scr |= SSI_SCR_TE; + else + scr |= SSI_SCR_RE; + sier |= sier_bits; + + if (++ssi->enabled == 1) + scr |= SSI_SCR_SSIEN; + + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + case SNDRV_PCM_TRIGGER_PAUSE_PUSH: + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + scr &= ~SSI_SCR_TE; + else + scr &= ~SSI_SCR_RE; + sier &= ~sier_bits; + + if (--ssi->enabled == 0) + scr &= ~SSI_SCR_SSIEN; + + break; + default: + return -EINVAL; + } + + if (!(ssi->flags & IMX_SSI_USE_AC97)) + /* rx/tx are always enabled to access ac97 registers */ + writel(scr, ssi->base + SSI_SCR); + + writel(sier, ssi->base + SSI_SIER); + + return 0; +} + +static struct snd_soc_dai_ops imx_ssi_pcm_dai_ops = { + .hw_params = imx_ssi_hw_params, + .set_fmt = imx_ssi_set_dai_fmt, + .set_clkdiv = imx_ssi_set_dai_clkdiv, + .set_sysclk = imx_ssi_set_dai_sysclk, + .set_tdm_slot = imx_ssi_set_dai_tdm_slot, + .trigger = imx_ssi_trigger, +}; + +static struct snd_soc_dai imx_ssi_dai = { + .playback = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_96000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &imx_ssi_pcm_dai_ops, +}; + +int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, + struct vm_area_struct *vma) +{ + struct snd_pcm_runtime *runtime = substream->runtime; + int ret; + + ret = dma_mmap_coherent(NULL, vma, runtime->dma_area, + runtime->dma_addr, runtime->dma_bytes); + + pr_debug("%s: ret: %d %p 0x%08x 0x%08x\n", __func__, ret, + runtime->dma_area, + runtime->dma_addr, + runtime->dma_bytes); + return ret; +} + +static int imx_pcm_preallocate_dma_buffer(struct snd_pcm *pcm, int stream) +{ + struct snd_pcm_substream *substream = pcm->streams[stream].substream; + struct snd_dma_buffer *buf = &substream->dma_buffer; + size_t size = IMX_SSI_DMABUF_SIZE; + + buf->dev.type = SNDRV_DMA_TYPE_DEV; + buf->dev.dev = pcm->card->dev; + buf->private_data = NULL; + buf->area = dma_alloc_writecombine(pcm->card->dev, size, + &buf->addr, GFP_KERNEL); + if (!buf->area) + return -ENOMEM; + buf->bytes = size; + + return 0; +} + +static u64 imx_pcm_dmamask = DMA_BIT_MASK(32); + +int imx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, + struct snd_pcm *pcm) +{ + + int ret = 0; + + if (!card->dev->dma_mask) + card->dev->dma_mask = &imx_pcm_dmamask; + if (!card->dev->coherent_dma_mask) + card->dev->coherent_dma_mask = DMA_BIT_MASK(32); + if (dai->playback.channels_min) { + ret = imx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_PLAYBACK); + if (ret) + goto out; + } + + if (dai->capture.channels_min) { + ret = imx_pcm_preallocate_dma_buffer(pcm, + SNDRV_PCM_STREAM_CAPTURE); + if (ret) + goto out; + } + +out: + return ret; +} + +void imx_pcm_free(struct snd_pcm *pcm) +{ + struct snd_pcm_substream *substream; + struct snd_dma_buffer *buf; + int stream; + + for (stream = 0; stream < 2; stream++) { + substream = pcm->streams[stream].substream; + if (!substream) + continue; + + buf = &substream->dma_buffer; + if (!buf->area) + continue; + + dma_free_writecombine(pcm->card->dev, buf->bytes, + buf->area, buf->addr); + buf->area = NULL; + } +} + +struct snd_soc_platform imx_soc_platform = { + .name = "imx-audio", +}; +EXPORT_SYMBOL_GPL(imx_soc_platform); + +static struct snd_soc_dai imx_ac97_dai = { + .name = "AC97", + .ac97_control = 1, + .playback = { + .stream_name = "AC97 Playback", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .capture = { + .stream_name = "AC97 Capture", + .channels_min = 2, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_48000, + .formats = SNDRV_PCM_FMTBIT_S16_LE, + }, + .ops = &imx_ssi_pcm_dai_ops, +}; + +static void setup_channel_to_ac97(struct imx_ssi *imx_ssi) +{ + void __iomem *base = imx_ssi->base; + + writel(0x0, base + SSI_SCR); + writel(0x0, base + SSI_STCR); + writel(0x0, base + SSI_SRCR); + + writel(SSI_SCR_SYN | SSI_SCR_NET, base + SSI_SCR); + + writel(SSI_SFCSR_RFWM0(8) | + SSI_SFCSR_TFWM0(8) | + SSI_SFCSR_RFWM1(8) | + SSI_SFCSR_TFWM1(8), base + SSI_SFCSR); + + writel(SSI_STCCR_WL(16) | SSI_STCCR_DC(12), base + SSI_STCCR); + writel(SSI_STCCR_WL(16) | SSI_STCCR_DC(12), base + SSI_SRCCR); + + writel(SSI_SCR_SYN | SSI_SCR_NET | SSI_SCR_SSIEN, base + SSI_SCR); + writel(SSI_SOR_WAIT(3), base + SSI_SOR); + + writel(SSI_SCR_SYN | SSI_SCR_NET | SSI_SCR_SSIEN | + SSI_SCR_TE | SSI_SCR_RE, + base + SSI_SCR); + + writel(SSI_SACNT_DEFAULT, base + SSI_SACNT); + writel(0xff, base + SSI_SACCDIS); + writel(0x300, base + SSI_SACCEN); +} + +static struct imx_ssi *ac97_ssi; + +static void imx_ssi_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + struct imx_ssi *imx_ssi = ac97_ssi; + void __iomem *base = imx_ssi->base; + unsigned int lreg; + unsigned int lval; + + if (reg > 0x7f) + return; + + pr_debug("%s: 0x%02x 0x%04x\n", __func__, reg, val); + + lreg = reg << 12; + writel(lreg, base + SSI_SACADD); + + lval = val << 4; + writel(lval , base + SSI_SACDAT); + + writel(SSI_SACNT_DEFAULT | SSI_SACNT_WR, base + SSI_SACNT); + udelay(100); +} + +static unsigned short imx_ssi_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + struct imx_ssi *imx_ssi = ac97_ssi; + void __iomem *base = imx_ssi->base; + + unsigned short val = -1; + unsigned int lreg; + + lreg = (reg & 0x7f) << 12 ; + writel(lreg, base + SSI_SACADD); + writel(SSI_SACNT_DEFAULT | SSI_SACNT_RD, base + SSI_SACNT); + + udelay(100); + + val = (readl(base + SSI_SACDAT) >> 4) & 0xffff; + + pr_debug("%s: 0x%02x 0x%04x\n", __func__, reg, val); + + return val; +} + +static void imx_ssi_ac97_reset(struct snd_ac97 *ac97) +{ + struct imx_ssi *imx_ssi = ac97_ssi; + + if (imx_ssi->ac97_reset) + imx_ssi->ac97_reset(ac97); +} + +static void imx_ssi_ac97_warm_reset(struct snd_ac97 *ac97) +{ + struct imx_ssi *imx_ssi = ac97_ssi; + + if (imx_ssi->ac97_warm_reset) + imx_ssi->ac97_warm_reset(ac97); +} + +struct snd_ac97_bus_ops soc_ac97_ops = { + .read = imx_ssi_ac97_read, + .write = imx_ssi_ac97_write, + .reset = imx_ssi_ac97_reset, + .warm_reset = imx_ssi_ac97_warm_reset +}; +EXPORT_SYMBOL_GPL(soc_ac97_ops); + +struct snd_soc_dai *imx_ssi_pcm_dai[2]; +EXPORT_SYMBOL_GPL(imx_ssi_pcm_dai); + +static int imx_ssi_probe(struct platform_device *pdev) +{ + struct resource *res; + struct imx_ssi *ssi; + struct imx_ssi_platform_data *pdata = pdev->dev.platform_data; + struct snd_soc_platform *platform; + int ret = 0; + unsigned int val; + + ssi = kzalloc(sizeof(*ssi), GFP_KERNEL); + if (!ssi) + return -ENOMEM; + + if (pdata) { + ssi->ac97_reset = pdata->ac97_reset; + ssi->ac97_warm_reset = pdata->ac97_warm_reset; + ssi->flags = pdata->flags; + } + + imx_ssi_pcm_dai[pdev->id] = &ssi->dai; + + ssi->irq = platform_get_irq(pdev, 0); + + ssi->clk = clk_get(&pdev->dev, NULL); + if (IS_ERR(ssi->clk)) { + ret = PTR_ERR(ssi->clk); + dev_err(&pdev->dev, "Cannot get the clock: %d\n", + ret); + goto failed_clk; + } + clk_enable(ssi->clk); + + res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + if (!res) { + ret = -ENODEV; + goto failed_get_resource; + } + + if (!request_mem_region(res->start, resource_size(res), DRV_NAME)) { + dev_err(&pdev->dev, "request_mem_region failed\n"); + ret = -EBUSY; + goto failed_get_resource; + } + + ssi->base = ioremap(res->start, resource_size(res)); + if (!ssi->base) { + dev_err(&pdev->dev, "ioremap failed\n"); + ret = -ENODEV; + goto failed_ioremap; + } + + if (ssi->flags & IMX_SSI_USE_AC97) { + if (ac97_ssi) { + ret = -EBUSY; + goto failed_ac97; + } + ac97_ssi = ssi; + setup_channel_to_ac97(ssi); + memcpy(&ssi->dai, &imx_ac97_dai, sizeof(imx_ac97_dai)); + } else + memcpy(&ssi->dai, &imx_ssi_dai, sizeof(imx_ssi_dai)); + + ssi->dai.id = pdev->id; + ssi->dai.dev = &pdev->dev; + ssi->dai.name = kasprintf(GFP_KERNEL, "imx-ssi.%d", pdev->id); + + writel(0x0, ssi->base + SSI_SIER); + + ssi->dma_params_rx.dma_addr = res->start + SSI_SRX0; + ssi->dma_params_tx.dma_addr = res->start + SSI_STX0; + + res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx0"); + if (res) + ssi->dma_params_tx.dma = res->start; + + res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx0"); + if (res) + ssi->dma_params_rx.dma = res->start; + + ssi->dai.id = pdev->id; + ssi->dai.dev = &pdev->dev; + ssi->dai.name = kasprintf(GFP_KERNEL, "imx-ssi.%d", pdev->id); + + if ((cpu_is_mx27() || cpu_is_mx21()) && + !(ssi->flags & IMX_SSI_USE_AC97)) { + ssi->flags |= IMX_SSI_DMA; + platform = imx_ssi_dma_mx2_init(pdev, ssi); + } else + platform = imx_ssi_fiq_init(pdev, ssi); + + imx_soc_platform.pcm_ops = platform->pcm_ops; + imx_soc_platform.pcm_new = platform->pcm_new; + imx_soc_platform.pcm_free = platform->pcm_free; + + val = SSI_SFCSR_TFWM0(ssi->dma_params_tx.burstsize) | + SSI_SFCSR_RFWM0(ssi->dma_params_rx.burstsize); + writel(val, ssi->base + SSI_SFCSR); + + ret = snd_soc_register_dai(&ssi->dai); + if (ret) { + dev_err(&pdev->dev, "register DAI failed\n"); + goto failed_register; + } + + platform_set_drvdata(pdev, ssi); + + return 0; + +failed_register: +failed_ac97: + iounmap(ssi->base); +failed_ioremap: + release_mem_region(res->start, resource_size(res)); +failed_get_resource: + clk_disable(ssi->clk); + clk_put(ssi->clk); +failed_clk: + kfree(ssi); + + return ret; +} + +static int __devexit imx_ssi_remove(struct platform_device *pdev) +{ + struct resource *res = platform_get_resource(pdev, IORESOURCE_MEM, 0); + struct imx_ssi *ssi = platform_get_drvdata(pdev); + + snd_soc_unregister_dai(&ssi->dai); + + if (ssi->flags & IMX_SSI_USE_AC97) + ac97_ssi = NULL; + + if (!(ssi->flags & IMX_SSI_DMA)) + imx_ssi_fiq_exit(pdev, ssi); + + iounmap(ssi->base); + release_mem_region(res->start, resource_size(res)); + clk_disable(ssi->clk); + clk_put(ssi->clk); + kfree(ssi); + + return 0; +} + +static struct platform_driver imx_ssi_driver = { + .probe = imx_ssi_probe, + .remove = __devexit_p(imx_ssi_remove), + + .driver = { + .name = DRV_NAME, + .owner = THIS_MODULE, + }, +}; + +static int __init imx_ssi_init(void) +{ + int ret; + + ret = snd_soc_register_platform(&imx_soc_platform); + if (ret) { + pr_err("failed to register soc platform: %d\n", ret); + return ret; + } + + ret = platform_driver_register(&imx_ssi_driver); + if (ret) { + snd_soc_unregister_platform(&imx_soc_platform); + return ret; + } + + return 0; +} + +static void __exit imx_ssi_exit(void) +{ + platform_driver_unregister(&imx_ssi_driver); + snd_soc_unregister_platform(&imx_soc_platform); +} + +module_init(imx_ssi_init); +module_exit(imx_ssi_exit); + +/* Module information */ +MODULE_AUTHOR("Sascha Hauer, <s.hauer(a)pengutronix.de>"); +MODULE_DESCRIPTION("i.MX I2S/ac97 SoC Interface"); +MODULE_LICENSE("GPL"); + diff --git a/sound/soc/imx/imx-ssi.h b/sound/soc/imx/imx-ssi.h new file mode 100644 index 0000000..cb2c81f --- /dev/null +++ b/sound/soc/imx/imx-ssi.h @@ -0,0 +1,238 @@ +/* + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#ifndef _IMX_SSI_H +#define _IMX_SSI_H + +#define SSI_STX0 0x00 +#define SSI_STX1 0x04 +#define SSI_SRX0 0x08 +#define SSI_SRX1 0x0c + +#define SSI_SCR 0x10 +#define SSI_SCR_CLK_IST (1 << 9) +#define SSI_SCR_CLK_IST_SHIFT 9 +#define SSI_SCR_TCH_EN (1 << 8) +#define SSI_SCR_SYS_CLK_EN (1 << 7) +#define SSI_SCR_I2S_MODE_NORM (0 << 5) +#define SSI_SCR_I2S_MODE_MSTR (1 << 5) +#define SSI_SCR_I2S_MODE_SLAVE (2 << 5) +#define SSI_I2S_MODE_MASK (3 << 5) +#define SSI_SCR_SYN (1 << 4) +#define SSI_SCR_NET (1 << 3) +#define SSI_SCR_RE (1 << 2) +#define SSI_SCR_TE (1 << 1) +#define SSI_SCR_SSIEN (1 << 0) + +#define SSI_SISR 0x14 +#define SSI_SISR_MASK ((1 << 19) - 1) +#define SSI_SISR_CMDAU (1 << 18) +#define SSI_SISR_CMDDU (1 << 17) +#define SSI_SISR_RXT (1 << 16) +#define SSI_SISR_RDR1 (1 << 15) +#define SSI_SISR_RDR0 (1 << 14) +#define SSI_SISR_TDE1 (1 << 13) +#define SSI_SISR_TDE0 (1 << 12) +#define SSI_SISR_ROE1 (1 << 11) +#define SSI_SISR_ROE0 (1 << 10) +#define SSI_SISR_TUE1 (1 << 9) +#define SSI_SISR_TUE0 (1 << 8) +#define SSI_SISR_TFS (1 << 7) +#define SSI_SISR_RFS (1 << 6) +#define SSI_SISR_TLS (1 << 5) +#define SSI_SISR_RLS (1 << 4) +#define SSI_SISR_RFF1 (1 << 3) +#define SSI_SISR_RFF0 (1 << 2) +#define SSI_SISR_TFE1 (1 << 1) +#define SSI_SISR_TFE0 (1 << 0) + +#define SSI_SIER 0x18 +#define SSI_SIER_RDMAE (1 << 22) +#define SSI_SIER_RIE (1 << 21) +#define SSI_SIER_TDMAE (1 << 20) +#define SSI_SIER_TIE (1 << 19) +#define SSI_SIER_CMDAU_EN (1 << 18) +#define SSI_SIER_CMDDU_EN (1 << 17) +#define SSI_SIER_RXT_EN (1 << 16) +#define SSI_SIER_RDR1_EN (1 << 15) +#define SSI_SIER_RDR0_EN (1 << 14) +#define SSI_SIER_TDE1_EN (1 << 13) +#define SSI_SIER_TDE0_EN (1 << 12) +#define SSI_SIER_ROE1_EN (1 << 11) +#define SSI_SIER_ROE0_EN (1 << 10) +#define SSI_SIER_TUE1_EN (1 << 9) +#define SSI_SIER_TUE0_EN (1 << 8) +#define SSI_SIER_TFS_EN (1 << 7) +#define SSI_SIER_RFS_EN (1 << 6) +#define SSI_SIER_TLS_EN (1 << 5) +#define SSI_SIER_RLS_EN (1 << 4) +#define SSI_SIER_RFF1_EN (1 << 3) +#define SSI_SIER_RFF0_EN (1 << 2) +#define SSI_SIER_TFE1_EN (1 << 1) +#define SSI_SIER_TFE0_EN (1 << 0) + +#define SSI_STCR 0x1c +#define SSI_STCR_TXBIT0 (1 << 9) +#define SSI_STCR_TFEN1 (1 << 8) +#define SSI_STCR_TFEN0 (1 << 7) +#define SSI_FIFO_ENABLE_0_SHIFT 7 +#define SSI_STCR_TFDIR (1 << 6) +#define SSI_STCR_TXDIR (1 << 5) +#define SSI_STCR_TSHFD (1 << 4) +#define SSI_STCR_TSCKP (1 << 3) +#define SSI_STCR_TFSI (1 << 2) +#define SSI_STCR_TFSL (1 << 1) +#define SSI_STCR_TEFS (1 << 0) + +#define SSI_SRCR 0x20 +#define SSI_SRCR_RXBIT0 (1 << 9) +#define SSI_SRCR_RFEN1 (1 << 8) +#define SSI_SRCR_RFEN0 (1 << 7) +#define SSI_FIFO_ENABLE_0_SHIFT 7 +#define SSI_SRCR_RFDIR (1 << 6) +#define SSI_SRCR_RXDIR (1 << 5) +#define SSI_SRCR_RSHFD (1 << 4) +#define SSI_SRCR_RSCKP (1 << 3) +#define SSI_SRCR_RFSI (1 << 2) +#define SSI_SRCR_RFSL (1 << 1) +#define SSI_SRCR_REFS (1 << 0) + +#define SSI_SRCCR 0x28 +#define SSI_SRCCR_DIV2 (1 << 18) +#define SSI_SRCCR_PSR (1 << 17) +#define SSI_SRCCR_WL(x) ((((x) - 2) >> 1) << 13) +#define SSI_SRCCR_DC(x) (((x) & 0x1f) << 8) +#define SSI_SRCCR_PM(x) (((x) & 0xff) << 0) +#define SSI_SRCCR_WL_MASK (0xf << 13) +#define SSI_SRCCR_DC_MASK (0x1f << 8) +#define SSI_SRCCR_PM_MASK (0xff << 0) + +#define SSI_STCCR 0x24 +#define SSI_STCCR_DIV2 (1 << 18) +#define SSI_STCCR_PSR (1 << 17) +#define SSI_STCCR_WL(x) ((((x) - 2) >> 1) << 13) +#define SSI_STCCR_DC(x) (((x) & 0x1f) << 8) +#define SSI_STCCR_PM(x) (((x) & 0xff) << 0) +#define SSI_STCCR_WL_MASK (0xf << 13) +#define SSI_STCCR_DC_MASK (0x1f << 8) +#define SSI_STCCR_PM_MASK (0xff << 0) + +#define SSI_SFCSR 0x2c +#define SSI_SFCSR_RFCNT1(x) (((x) & 0xf) << 28) +#define SSI_RX_FIFO_1_COUNT_SHIFT 28 +#define SSI_SFCSR_TFCNT1(x) (((x) & 0xf) << 24) +#define SSI_TX_FIFO_1_COUNT_SHIFT 24 +#define SSI_SFCSR_RFWM1(x) (((x) & 0xf) << 20) +#define SSI_SFCSR_TFWM1(x) (((x) & 0xf) << 16) +#define SSI_SFCSR_RFCNT0(x) (((x) & 0xf) << 12) +#define SSI_RX_FIFO_0_COUNT_SHIFT 12 +#define SSI_SFCSR_TFCNT0(x) (((x) & 0xf) << 8) +#define SSI_TX_FIFO_0_COUNT_SHIFT 8 +#define SSI_SFCSR_RFWM0(x) (((x) & 0xf) << 4) +#define SSI_SFCSR_TFWM0(x) (((x) & 0xf) << 0) +#define SSI_SFCSR_RFWM0_MASK (0xf << 4) +#define SSI_SFCSR_TFWM0_MASK (0xf << 0) + +#define SSI_STR 0x30 +#define SSI_STR_TEST (1 << 15) +#define SSI_STR_RCK2TCK (1 << 14) +#define SSI_STR_RFS2TFS (1 << 13) +#define SSI_STR_RXSTATE(x) (((x) & 0xf) << 8) +#define SSI_STR_TXD2RXD (1 << 7) +#define SSI_STR_TCK2RCK (1 << 6) +#define SSI_STR_TFS2RFS (1 << 5) +#define SSI_STR_TXSTATE(x) (((x) & 0xf) << 0) + +#define SSI_SOR 0x34 +#define SSI_SOR_CLKOFF (1 << 6) +#define SSI_SOR_RX_CLR (1 << 5) +#define SSI_SOR_TX_CLR (1 << 4) +#define SSI_SOR_INIT (1 << 3) +#define SSI_SOR_WAIT(x) (((x) & 0x3) << 1) +#define SSI_SOR_WAIT_MASK (0x3 << 1) +#define SSI_SOR_SYNRST (1 << 0) + +#define SSI_SACNT 0x38 +#define SSI_SACNT_FRDIV(x) (((x) & 0x3f) << 5) +#define SSI_SACNT_WR (1 << 4) +#define SSI_SACNT_RD (1 << 3) +#define SSI_SACNT_TIF (1 << 2) +#define SSI_SACNT_FV (1 << 1) +#define SSI_SACNT_AC97EN (1 << 0) + +#define SSI_SACADD 0x3c +#define SSI_SACDAT 0x40 +#define SSI_SATAG 0x44 +#define SSI_STMSK 0x48 +#define SSI_SRMSK 0x4c +#define SSI_SACCST 0x50 +#define SSI_SACCEN 0x54 +#define SSI_SACCDIS 0x58 + +/* SSI clock sources */ +#define IMX_SSP_SYS_CLK 0 + +/* SSI audio dividers */ +#define IMX_SSI_TX_DIV_2 0 +#define IMX_SSI_TX_DIV_PSR 1 +#define IMX_SSI_TX_DIV_PM 2 +#define IMX_SSI_RX_DIV_2 3 +#define IMX_SSI_RX_DIV_PSR 4 +#define IMX_SSI_RX_DIV_PM 5 + +extern struct snd_soc_dai *imx_ssi_pcm_dai[2]; +extern struct snd_soc_platform imx_soc_platform; + +#define DRV_NAME "imx-ssi" + +struct imx_pcm_dma_params { + int dma; + unsigned long dma_addr; + int burstsize; +}; + +struct imx_ssi { + struct snd_soc_dai dai; + struct platform_device *ac97_dev; + + struct snd_soc_device imx_ac97; + struct clk *clk; + void __iomem *base; + int irq; + int fiq_enable; + unsigned int offset; + + unsigned int flags; + + void (*ac97_reset) (struct snd_ac97 *ac97); + void (*ac97_warm_reset)(struct snd_ac97 *ac97); + + struct imx_pcm_dma_params dma_params_rx; + struct imx_pcm_dma_params dma_params_tx; + + int enabled; +}; + +struct snd_soc_platform *imx_ssi_fiq_init(struct platform_device *pdev, + struct imx_ssi *ssi); +void imx_ssi_fiq_exit(struct platform_device *pdev, struct imx_ssi *ssi); +struct snd_soc_platform *imx_ssi_dma_mx2_init(struct platform_device *pdev, + struct imx_ssi *ssi); + +int snd_imx_pcm_mmap(struct snd_pcm_substream *substream, struct vm_area_struct *vma); +int imx_pcm_new(struct snd_card *card, struct snd_soc_dai *dai, + struct snd_pcm *pcm); +void imx_pcm_free(struct snd_pcm *pcm); + +/* + * Do not change this as the FIQ handler depends on this size + */ +#define IMX_SSI_DMABUF_SIZE (64 * 1024) + +#define DMA_RXFIFO_BURST 0x4 +#define DMA_TXFIFO_BURST 0x6 + +#endif /* _IMX_SSI_H */ -- 1.6.6

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[alsa-devel] [PATCH] ASoC: tpa6130a2: Support for tpa6140's regulators
by Ilkka Koskinen 11 Jan '10

11 Jan '10

tpa6140a2 uses different names for the regulators. Signed-off-by: Ilkka Koskinen <ilkka.koskinen(a)nokia.com> --- include/sound/tpa6130a2-plat.h | 6 ++++++ sound/soc/codecs/tpa6130a2.c | 22 ++++++++++++++++++++-- 2 files changed, 26 insertions(+), 2 deletions(-) diff --git a/include/sound/tpa6130a2-plat.h b/include/sound/tpa6130a2-plat.h index e8c901e..e29fde6 100644 --- a/include/sound/tpa6130a2-plat.h +++ b/include/sound/tpa6130a2-plat.h @@ -23,7 +23,13 @@ #ifndef TPA6130A2_PLAT_H #define TPA6130A2_PLAT_H +enum tpa_model { + TPA6130A2, + TPA6140A2, +}; + struct tpa6130a2_platform_data { + enum tpa_model id; int power_gpio; }; diff --git a/sound/soc/codecs/tpa6130a2.c b/sound/soc/codecs/tpa6130a2.c index 8e98ccf..8b27281 100644 --- a/sound/soc/codecs/tpa6130a2.c +++ b/sound/soc/codecs/tpa6130a2.c @@ -41,6 +41,11 @@ static const char *tpa6130a2_supply_names[TPA6130A2_NUM_SUPPLIES] = { "Vdd", }; +static const char *tpa6140a2_supply_names[TPA6130A2_NUM_SUPPLIES] = { + "HPVdd", + "AVdd", +}; + /* This struct is used to save the context */ struct tpa6130a2_data { struct mutex mutex; @@ -420,8 +425,21 @@ static int tpa6130a2_probe(struct i2c_client *client, gpio_direction_output(data->power_gpio, 0); } - for (i = 0; i < ARRAY_SIZE(data->supplies); i++) - data->supplies[i].supply = tpa6130a2_supply_names[i]; + switch (pdata->id) { + case TPA6130A2: + for (i = 0; i < ARRAY_SIZE(data->supplies); i++) + data->supplies[i].supply = tpa6130a2_supply_names[i]; + break; + case TPA6140A2: + for (i = 0; i < ARRAY_SIZE(data->supplies); i++) + data->supplies[i].supply = tpa6140a2_supply_names[i];; + break; + default: + dev_warn(dev, "Unknown TPA model (%d). Assuming 6130A2\n", + pdata->id); + for (i = 0; i < ARRAY_SIZE(data->supplies); i++) + data->supplies[i].supply = tpa6130a2_supply_names[i]; + } ret = regulator_bulk_get(dev, ARRAY_SIZE(data->supplies), data->supplies); -- 1.6.0.4

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[alsa-devel] Problems with snd_pcm_wait(..)
by Markus Luttenberger 11 Jan '10

11 Jan '10

Hey, I tried to implement a callback functionality with ALSA based on the example given at http://www.cs.utk.edu/~vose/linux/equalarea.com/alsa-audio.html#interruptex. Especially the while-loop is basically the same in my program. However, it doesn't work. I can trace the program to snd_pcm_wait(..) which just waits endlessly. The whole thing is called within a thread. The thread is still accessible from elsewhere in the pogram. It's just the snd_pcm_wait(..)-function which does not return although a time-out of 1000 ms is given. If I don't wait for the device I get the following assertion: pcm.c:1079: snd_pcm_writei: Assertion `size == 0 || buffer' failed. Any ideas what could be wrong? greetings.. ..Markus

2 2

[alsa-devel] alsa-util.c: snd_pcm_avail() returned a value that is exceptionally large: 349696 bytes (1821 ms).
by Chris 11 Jan '10

11 Jan '10

Mandriva 2010.0, Gnome 2.28, pulseaudio 0.9.19-7mdv2010.0. With pulseaudio enabled in MCC periodically the hd light will constantly be on for a period of 4 or 5 minutes and system will slow down to a crawl. Also these comments will be in syslog: pulseaudio[4331]: alsa-util.c: snd_pcm_avail() returned a value that is exceptionally large: 349696 bytes (1821 ms). Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: Most likely this is a bug in the ALSA driver 'snd_via82xx'. Please report this issue to the ALSA developers. Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: snd_pcm_dump(): Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: Hardware PCM card 1 'VIA 8237' device 0 subdevice 0 Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: Its setup is: Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: stream : PLAYBACK Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: access : MMAP_INTERLEAVED Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: format : S16_LE Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: subformat : STD Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: channels : 2 Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: rate : 48000 Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: exact rate : 48000 (48000/1) Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: msbits : 16 Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: buffer_size : 16384 Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: period_size : 8192 Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: period_time : 170666 Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: tstamp_mode : ENABLE Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: period_step : 1 Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: avail_min : 9665 Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: period_event : 0 Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: start_threshold : -1 Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: stop_threshold : 1073741824 Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: silence_threshold: 0 Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: silence_size : 0 Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: boundary : 1073741824 Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: appl_ptr : 767919576 Dec 14 20:26:29 localhost pulseaudio[4331]: alsa-util.c: hw_ptr : 767990616 Dec 14 20:26:29 localhost pulseaudio[4331]: ratelimit.c: 7 events suppressed Dec 14 20:26:31 localhost pulseaudio[4331]: asyncq.c: q overrun, queuing locally I also see this quite frequently: Dec 14 20:15:29 localhost pulseaudio[4331]: asyncq.c: q overrun, queuing locally Dec 14 20:15:34 localhost pulseaudio[4331]: ratelimit.c: 657 events suppressed Dec 14 20:15:34 localhost pulseaudio[4331]: asyncq.c: q overrun, queuing locally Dec 14 20:15:39 localhost pulseaudio[4331]: ratelimit.c: 2508 events suppressed Dec 14 20:15:39 localhost pulseaudio[4331]: asyncq.c: q overrun, queuing locally Dec 14 20:15:44 localhost pulseaudio[4331]: ratelimit.c: 3530 events suppressed Dec 14 20:15:44 localhost pulseaudio[4331]: asyncq.c: q overrun, queuing locally Dec 14 20:15:44 localhost pulseaudio[4331]: asyncq.c: q overrun, queuing locally Dec 14 20:15:49 localhost pulseaudio[4331]: ratelimit.c: 2672 events suppressed Dec 14 20:15:49 localhost pulseaudio[4331]: asyncq.c: q overrun, queuing locally Dec 14 20:16:05 localhost pulseaudio[4331]: ratelimit.c: 2244 events suppressed Dec 14 20:16:18 localhost pulseaudio[4331]: ratelimit.c: 1 events suppressed Dec 14 20:16:23 localhost pulseaudio[4331]: ratelimit.c: 1 events suppressed Dec 14 20:16:34 localhost pulseaudio[4331]: ratelimit.c: 5 events suppressed Dec 14 20:16:39 localhost pulseaudio[4331]: ratelimit.c: 9 events suppressed Dec 14 20:16:45 localhost pulseaudio[4331]: ratelimit.c: 11 events suppressed Dec 14 20:16:50 localhost pulseaudio[4331]: ratelimit.c: 13 events suppressed Dec 14 20:16:55 localhost pulseaudio[4331]: ratelimit.c: 16 events suppressed Dec 14 20:17:01 localhost pulseaudio[4331]: ratelimit.c: 17 events suppressed Dec 14 20:17:06 localhost pulseaudio[4331]: ratelimit.c: 23 events suppressed Dec 14 20:17:11 localhost pulseaudio[4331]: ratelimit.c: 19 events suppressed Dec 14 20:17:31 localhost pulseaudio[4331]: ratelimit.c: 23 events suppressed Dec 14 20:17:36 localhost pulseaudio[4331]: ratelimit.c: 19 events suppressed Dec 14 20:17:42 localhost pulseaudio[4331]: ratelimit.c: 21 events suppressed Dec 14 20:17:47 localhost pulseaudio[4331]: ratelimit.c: 19 events suppressed I have pulseaudio enabled but have glitch-free mode disabled. Any assistance in troubleshooting this would be appreciated. Chris -- KeyID 0xE372A7DA98E6705C

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[alsa-devel] [PATCH] ac97: add AC97 STMicroelectronics' codecs
by Krzysztof Helt 10 Jan '10

10 Jan '10

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> Add the STMicroelectronics ST7597 codec and an unknown codec from the same manufacturer found on the Creative SB 128 card (CT4810). Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- Datasheet of the ST7597 can be found on the web (various datasheet sites have copies). sound/pci/ac97/ac97_codec.c | 10 ++++++++++ sound/pci/ac97/ac97_id.h | 2 ++ 2 files changed, 12 insertions(+), 0 deletions(-) diff --git a/sound/pci/ac97/ac97_codec.c b/sound/pci/ac97/ac97_codec.c index c119206..a7630e9 100644 --- a/sound/pci/ac97/ac97_codec.c +++ b/sound/pci/ac97/ac97_codec.c @@ -83,6 +83,7 @@ static const struct ac97_codec_id snd_ac97_codec_id_vendors[] = { { 0x4e534300, 0xffffff00, "National Semiconductor", NULL, NULL }, { 0x50534300, 0xffffff00, "Philips", NULL, NULL }, { 0x53494c00, 0xffffff00, "Silicon Laboratory", NULL, NULL }, +{ 0x53544d00, 0xffffff00, "STMicroelectronics", NULL, NULL }, { 0x54524100, 0xffffff00, "TriTech", NULL, NULL }, { 0x54584e00, 0xffffff00, "Texas Instruments", NULL, NULL }, { 0x56494100, 0xffffff00, "VIA Technologies", NULL, NULL }, @@ -161,6 +162,7 @@ static const struct ac97_codec_id snd_ac97_codec_ids[] = { { 0x4e534350, 0xffffffff, "LM4550", patch_lm4550, NULL }, // volume wrap fix { 0x50534304, 0xffffffff, "UCB1400", patch_ucb1400, NULL }, { 0x53494c20, 0xffffffe0, "Si3036,8", mpatch_si3036, mpatch_si3036, AC97_MODEM_PATCH }, +{ 0x53544d02, 0xffffffff, "ST7597", NULL, NULL }, { 0x54524102, 0xffffffff, "TR28022", NULL, NULL }, { 0x54524103, 0xffffffff, "TR28023", NULL, NULL }, { 0x54524106, 0xffffffff, "TR28026", NULL, NULL }, @@ -213,6 +215,14 @@ static int snd_ac97_valid_reg(struct snd_ac97 *ac97, unsigned short reg) { /* filter some registers for buggy codecs */ switch (ac97->id) { + case AC97_ID_ST_AC97_ID4: + if (reg == 0x08) + return 0; + /* fall through */ + case AC97_ID_ST7597: + if (reg == 0x22 || reg == 0x7a) + return 1; + /* fall through */ case AC97_ID_AK4540: case AC97_ID_AK4542: if (reg <= 0x1c || reg == 0x20 || reg == 0x26 || reg >= 0x7c) diff --git a/sound/pci/ac97/ac97_id.h b/sound/pci/ac97/ac97_id.h index c129492..d603147 100644 --- a/sound/pci/ac97/ac97_id.h +++ b/sound/pci/ac97/ac97_id.h @@ -62,3 +62,5 @@ #define AC97_ID_CM9761_78 0x434d4978 #define AC97_ID_CM9761_82 0x434d4982 #define AC97_ID_CM9761_83 0x434d4983 +#define AC97_ID_ST7597 0x53544d02 +#define AC97_ID_ST_AC97_ID4 0x53544d04 -- 1.6.4

2 1

[alsa-devel] [PATCH] ALSA: ac97: Add Dell Dimension 2400 to Headphone/Line Jack Sense blacklist
by Daniel Chen 10 Jan '10

10 Jan '10

Hi, Attached is a patch against current sound-2.6 master HEAD fixing inaudible playback by default. Thanks, -Dan

2 1

[alsa-devel] aplay - Audio Surveilance
by John Sauter 10 Jan '10

10 Jan '10

I have added some features to aplay to make audio surveillance easier: --process-id-file writes the PID to a file so another program can signal aplay to close its capture file and open another one without missing any sound. --max-file-time causes the close and re-open to happen automatically. --use-strftime allows capture files to be named for their start times. Details are in the man file. For ease of review, this patch is divided into three parts. Part 1 is just --process-id-file, which is intrusive because it must intercept the exit points to delete the file. Part 2 is the remainder of the new features in aplay.c. Part 3 updates the man file. John Sauter (John_Sauter(a)systemeyescomputerstore.com)

2 1

[alsa-devel] VGN-P11Z_G (hda-intel ALC262) internal microphone not working
by Christian Stöveken 09 Jan '10

09 Jan '10

I'm running Ubuntu Karmic on a Vaio P11z and wanted to ask if there's anything I can do to help get the not working microphone going for me and all P11z users ;-) Manufacturer: Sony Corporation Product Name: VGN-P11Z_G Soundcards recognised by ALSA: HDA-Intel - HDA Intel MID Codec: Realtek ALC262 I already upgraded to alsa 1.0.21 using that http://www.webupd8.org/2009/11/ubuntu-karmic-upgrade-alsa-to-1021-from.html… and am atm on 2.6.33-020633rc3-generic. alsa-info: -> http://www.alsa-project.org/db/?f=35205f955e08e7e16b5274862f6b376b890731e3 -> toshiba-s06 (somewhat working) http://www.alsa-project.org/db/?f=8ae0ab7b6672e41550651dbda63587438c93756f the issues: - Microphone isn't working with toshiba-s06 it is sometimes working right after powering up, sometimes I have to force-reload alsa. - Sound also comes from the speakers with plugged-in headphones sony-assamd / toshiba-s06 (and multiple others fix that issue) Regards, Chris <http://mailman.alsa-project.org/mailman/listinfo/alsa-devel>

2 3

[PATCH] pulse: Fix invalid buffer pointer return value
by David Henningsson 09 Jan '10

09 Jan '10

This patch improves recovering from underruns, and prevents hangs inside snd_pcm_write* and snd_pcm_read* due to snd_pcm_avail* returning too low values. It especially helps low latency situations. Signed-off-by: David Henningsson <diwic(a)ubuntu.com> --- pulse/pcm_pulse.c | 15 ++++++++++++++- 1 files changed, 14 insertions(+), 1 deletions(-) Index: alsa-plugins-1.0.22/pulse/pcm_pulse.c =================================================================== --- alsa-plugins-1.0.22.orig/pulse/pcm_pulse.c 2010-02-28 17:12:02.000000000 -0500 +++ alsa-plugins-1.0.22/pulse/pcm_pulse.c 2010-02-28 17:12:25.000000000 -0500 @@ -90,6 +90,10 @@ if (pcm->io.stream == SND_PCM_STREAM_CAPTURE) size -= pcm->offset; + /* Prevent accidental overrun of the fake ringbuffer */ + if (size >= pcm->buffer_attr.tlength) + size = pcm->buffer_attr.tlength-1; + if (size > pcm->last_size) { pcm->ptr += size - pcm->last_size; pcm->ptr %= pcm->buffer_attr.tlength; @@ -424,6 +428,7 @@ snd_pcm_pulse_t *pcm = io->private_data; const char *buf; snd_pcm_sframes_t ret = 0; + size_t writebytes; assert(pcm); @@ -445,13 +450,15 @@ (char *) areas->addr + (areas->first + areas->step * offset) / 8; - ret = pa_stream_write(pcm->stream, buf, size * pcm->frame_size, NULL, 0, 0); + writebytes = size * pcm->frame_size; + ret = pa_stream_write(pcm->stream, buf, writebytes, NULL, 0, 0); if (ret < 0) { ret = -EIO; goto finish; } /* Make sure the buffer pointer is in sync */ + pcm->last_size -= writebytes; ret = update_ptr(pcm); if (ret < 0) goto finish; @@ -528,6 +535,7 @@ dst_buf = (char *) dst_buf + frag_length; remain_size -= frag_length; + pcm->last_size -= frag_length; } /* Make sure the buffer pointer is in sync */ @@ -730,6 +738,11 @@ pcm->offset = 0; /*pcm->underrun = 0;*/ + /* Reset fake ringbuffer */ + pcm->last_size = 0; + pcm->ptr = 0; + update_ptr(pcm); + finish: pa_threaded_mainloop_unlock(pcm->p->mainloop); --------------040006090607070407080605--

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[alsa-devel] [RFC] audio surveilance
by John Sauter 08 Jan '10

08 Jan '10

When doing audio surveillance, you want to record all the sound, but occasionally close the old output file and open a new one, so you can listen to a captured event without disturbing the recording of future sounds. I have some ideas for additions to aplay to meet the needs of audio surveillance, and would appreciate feedback from the alsa developers. My proposed patches are attached. They have been tested only on GNU/Linux 64-bit. John Sauter (John_Sauter(a)systemeyescomputerstore.com)

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Re: [alsa-devel] alsa support for HDA ALC889A nVidia Corporation MCP55 High Definition Audio EVGA 122-CK-NF68
by Oliver Stieber 08 Jan '10

08 Jan '10

just quit x and tried mb3, no sound. module snd-hda-intel did unload ok though (often the module is still showen as used when it's not) i wrote a little bash script to try every model listed in the source code, and mb5 was the only one that workedish and i had lots of kernel mess along the way. On Jan 8, 2010 3:46 PM, "Justin P. Mattock" <justinmattock(a)gmail.com> wrote: On 01/07/10 16:39, Oliver Stieber wrote: > > Hi, > > I'm having some issues, (and think I've discove... using model=mb3 gives no effect? Justin P. Mattock

2 1

[alsa-devel] [PATCH V2] Add i.MX sound support
by Sascha Hauer 08 Jan '10

08 Jan '10

Hi All, Here is the second version of the patches to add i.MX sound support. This time I haven't included ARM related patches, so this series should be safe to apply. I skipped the mc13783 driver since I'm not ready integrating the comments from Mark. The main change in the driver is that I splitted out the transport from the driver, so it should be easy to add another transport layer to the driver when SDMA support is merged. I hopefully also addressed the other comments from Mark. Sascha

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[alsa-devel] Speakers won't turn off, when headphones are plugged in
by Joern Bernhardt 08 Jan '10

08 Jan '10

Hi there, my speakers won't turn off, when I plug in headphones. I've tried all of these models: ref Reference board ref-no-jd Reference board without HP/Mic jack detection 3stack D965 3stack 5stack D965 5stack SPDIF 5stack-no-fp D965 5stack without front panel dell-3stack Dell Dimension E520 dell-bios Fixes with Dell BIOS setup auto BIOS setup (default) Some of these only had sound when headphones were plugged in (speakers didn't work at all) or when they were plugged in the microphone jack. My laptop is a HP HDX 9000 and my alsa-info script output is here: http://www.alsa-project.org/db/?f=b5a4386553266384382868ddef56553d643e7b4a Thanks for helping, Joern

2 1

[alsa-devel] alsa support for HDA ALC889A nVidia Corporation MCP55 High Definition Audio EVGA 122-CK-NF68
by Oliver Stieber 08 Jan '10

08 Jan '10

Hi, I'm having some issues, (and think I've discovered some bugs/ performance features) in alsa with my card. Certainly sound used to work on front stereo speakers with mode=auto in ubuntu 9, but had to change to model=mb5 and have severe stability issues (alsa no longer works, kernel gets trashed) when I do things like try to change volume, and kde volume changer doesn't work unless I use the mixer etc... Also, this is the only model where I get sound, other models trash kernel space etc... With this model I can't unload the module as it says it's still in use even with the sound app has terminated (didn't have this problem under ubuntu 9) Have some dmesg dumps in a bug I filed (will dig it out). Anyhow, I'm more than happy to have a crack at getting my sound card working perfectly. Oliver. For starters.... alsa-info.sh is here. http://www.alsa-project.org/db/?f=a65b159863be66dc66863ec82fb65d616bd2fd52 !!################################ !!ALSA Information Script v 0.4.58 !!################################ !!Script ran on: Fri Jan 8 00:30:41 UTC 2010 !!Linux Distribution !!------------------ Ubuntu 9.10 \n \l DISTRIB_ID=Ubuntu DISTRIB_DESCRIPTION="Ubuntu 9.10" !!DMI Information !!--------------- Manufacturer: EVGA Product Name: 122-CK-NF68 !!Kernel Information !!------------------ Kernel release: 2.6.32-0.dmz.2-liquorix-amd64 Operating System: GNU/Linux Architecture: x86_64 Processor: unknown SMP Enabled: Yes !!ALSA Version !!------------ Driver version: 1.0.22 Library version: 1.0.20 Utilities version: 1.0.22 !!Loaded ALSA modules !!------------------- snd_hda_intel !!Sound Servers on this system !!---------------------------- aRts: Installed - Yes (/usr/bin/artsd) Running - No Jack: Installed - Yes (/usr/bin/jackd) Running - No !!Soundcards recognised by ALSA !!----------------------------- 0 [NVidia ]: HDA-Intel - HDA NVidia HDA NVidia at 0xcfff0000 irq 25 !!PCI Soundcards installed in the system !!-------------------------------------- 00:0f.1 Audio device: nVidia Corporation MCP55 High Definition Audio (rev a2) !!Advanced information - PCI Vendor/Device/Susbsystem ID's !!-------------------------------------------------------- 00:0f.1 0403: 10de:0371 (rev a2) Subsystem: 10de:c55e -- Subsystem: 10b0:0401 Control: I/O Mem BusMaster SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- !!Modprobe options (Sound related) !!-------------------------------- snd-atiixp-modem: index=-2 snd-intel8x0m: index=-2 snd-via82xx-modem: index=-2 snd-usb-audio: index=-2 snd-usb-us122l: index=-2 snd-usb-usx2y: index=-2 snd-usb-caiaq: index=-2 snd-cmipci: mpu_port=0x330 fm_port=0x388 snd-pcsp: index=-2 snd-hda-intel: power_save=0 model=mb5 !!Loaded sound module options !!-------------------------- !!Module: snd_hda_intel bdl_pos_adj : 32,-1,-1,-1,-1,-1,-1,-1 beep_mode : 1,1,1,1,1,1,1,1 enable : Y,Y,Y,Y,Y,Y,Y,Y enable_msi : -1 id : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> index : -1,-1,-1,-1,-1,-1,-1,-1 model : mb5,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> patch : <NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL>,<NULL> position_fix : 0,0,0,0,0,0,0,0 power_save : 0 power_save_controller : Y probe_mask : -1,-1,-1,-1,-1,-1,-1,-1 probe_only : N,N,N,N,N,N,N,N single_cmd : N !!HDA-Intel Codec information !!--------------------------- --startcollapse-- Codec: Realtek ALC889A Address: 0 Function Id: 0x1 Vendor Id: 0x10ec0885 Subsystem Id: 0x10ece601 Revision Id: 0x100101 No Modem Function Group found Default PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Default Amp-In caps: N/A Default Amp-Out caps: N/A GPIO: io=2, o=0, i=0, unsolicited=1, wake=0 IO[0]: enable=1, dir=1, wake=0, sticky=0, data=1, unsol=0 IO[1]: enable=0, dir=0, wake=0, sticky=0, data=0, unsol=0 Node 0x02 [Audio Output] wcaps 0x11: Stereo Device: name="ALC889A Analog", type="Audio", device=0 Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x03 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x04 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x05 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x06 [Audio Output] wcaps 0x211: Stereo Digital Control: name="IEC958 Playback Con Mask", index=0, device=0 Control: name="IEC958 Playback Pro Mask", index=0, device=0 Control: name="IEC958 Playback Default", index=0, device=0 Control: name="IEC958 Playback Switch", index=0, device=0 Control: name="IEC958 Default PCM Playback Switch", index=0, device=0 Device: name="ALC889A Digital", type="SPDIF", device=1 Converter: stream=0, channel=0 Digital: Digital category: 0x0 PCM: rates [0x5e0]: 44100 48000 88200 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Node 0x07 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Capture Switch", index=0, device=0 Control: name="Capture Volume", index=0, device=0 Device: name="ALC889A Analog", type="Audio", device=0 Amp-In caps: ofs=0x10, nsteps=0x2e, stepsize=0x03, mute=1 Amp-In vals: [0x1c 0x1c] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Connection: 1 0x24 Node 0x08 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Capture Switch", index=1, device=0 Control: name="Capture Volume", index=1, device=0 Device: name="ALC889A Analog", type="Audio", device=2 Amp-In caps: ofs=0x10, nsteps=0x2e, stepsize=0x03, mute=1 Amp-In vals: [0x80 0x80] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Connection: 1 0x23 Node 0x09 [Audio Input] wcaps 0x10011b: Stereo Amp-In Control: name="Capture Switch", index=2, device=0 Control: name="Capture Volume", index=2, device=0 Amp-In caps: ofs=0x10, nsteps=0x2e, stepsize=0x03, mute=1 Amp-In vals: [0x80 0x80] Converter: stream=0, channel=0 SDI-Select: 0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Connection: 1 0x22 Node 0x0a [Audio Input] wcaps 0x100391: Stereo Digital Control: name="IEC958 Capture Switch", index=0, device=0 Control: name="IEC958 Capture Default", index=0, device=0 Device: name="ALC889A Digital", type="SPDIF", device=1 Converter: stream=0, channel=0 SDI-Select: 0 Digital: Digital category: 0x0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0x1e]: 16 20 24 32 formats [0x1]: PCM Unsolicited: tag=00, enabled=0 Connection: 1 0x1f Node 0x0b [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Line Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Line Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=2, ofs=0 Control: name="Mic Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Mic Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=1, ofs=0 Control: name="Beep Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=In, idx=5, ofs=0 Control: name="Beep Playback Switch", index=0, device=0 ControlAmp: chs=3, dir=In, idx=5, ofs=0 Amp-In caps: ofs=0x17, nsteps=0x1f, stepsize=0x05, mute=1 Amp-In vals: [0x97 0x97] [0x80 0x80] [0x80 0x80] [0x97 0x97] [0x97 0x97] [0x00 0x00] [0x97 0x97] [0x97 0x97] [0x97 0x97] [0x97 0x97] Connection: 10 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 Node 0x0c [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Front Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x2c 0x2c] Connection: 2 0x02 0x0b Node 0x0d [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="Surround Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x1d 0x1d] Connection: 2 0x03 0x0b Node 0x0e [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="LFE Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x19 0x00] Connection: 2 0x04 0x0b Node 0x0f [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Control: name="HP Playback Volume", index=0, device=0 ControlAmp: chs=3, dir=Out, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x00 0x00] Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x00 0x00] Connection: 2 0x05 0x0b Node 0x10 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x11 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x12 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x13 [Vendor Defined Widget] wcaps 0xf00000: Mono Node 0x14 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x01014010: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0x0 Pin-ctls: 0x40: OUT VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e 0x0f* 0x26 Node 0x15 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Line Boost", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x01011012: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0x1, Sequence = 0x2 Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d* 0x0e 0x0f 0x26 Node 0x16 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000003c: IN OUT HP Detect Pin Default 0x01016011: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Orange DefAssociation = 0x1, Sequence = 0x1 Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e* 0x0f 0x26 Node 0x17 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000003c: IN OUT HP Detect Pin Default 0x01012014: [Jack] Line Out at Ext Rear Conn = 1/8, Color = Grey DefAssociation = 0x1, Sequence = 0x4 Pin-ctls: 0x20: IN Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e 0x0f* 0x26 Node 0x18 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x01a19830: [Jack] Mic at Ext Rear Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0x0 Pin-ctls: 0x41: OUT VREF_50 Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x19 [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Control: name="Mic Boost", index=0, device=0 ControlAmp: chs=3, dir=In, idx=0, ofs=0 Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x02a1983f: [Jack] Mic at Ext Front Conn = 1/8, Color = Pink DefAssociation = 0x3, Sequence = 0xf Pin-ctls: 0x24: IN VREF_80 Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1a [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x00 0x00] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x01813031: [Jack] Line In at Ext Rear Conn = 1/8, Color = Blue DefAssociation = 0x3, Sequence = 0x1 Pin-ctls: 0x41: OUT VREF_50 Unsolicited: tag=00, enabled=0 Connection: 5 0x0c 0x0d 0x0e* 0x0f 0x26 Node 0x1b [Pin Complex] wcaps 0x40018f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x03, stepsize=0x27, mute=0 Amp-In vals: [0x00 0x00] Amp-Out caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-Out vals: [0x80 0x80] Pincap 0x0000373c: IN OUT HP Detect Vref caps: HIZ 50 GRD 80 100 Pin Default 0x02214c1f: [Jack] HP Out at Ext Front Conn = 1/8, Color = Green DefAssociation = 0x1, Sequence = 0xf Pin-ctls: 0x20: IN VREF_HIZ Unsolicited: tag=00, enabled=0 Connection: 5 0x0c* 0x0d 0x0e 0x0f 0x26 Node 0x1c [Pin Complex] wcaps 0x400001: Stereo Pincap 0x00000020: IN Pin Default 0x411111f0: [N/A] Speaker at Ext Rear Conn = 1/8, Color = Black DefAssociation = 0xf, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x1d [Pin Complex] wcaps 0x400000: Mono Pincap 0x00000020: IN Pin Default 0x99830132: [Fixed] Line In at Int ATAPI Conn = ATAPI, Color = Unknown DefAssociation = 0x3, Sequence = 0x2 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x1e [Pin Complex] wcaps 0x400300: Mono Digital Pincap 0x00000010: OUT Pin Default 0x01441120: [Jack] SPDIF Out at Ext Rear Conn = RCA, Color = Black DefAssociation = 0x2, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x40: OUT Connection: 1 0x06 Node 0x1f [Pin Complex] wcaps 0x400200: Mono Digital Pincap 0x00000020: IN Pin Default 0x01c46140: [Jack] SPDIF In at Ext Rear Conn = RCA, Color = Orange DefAssociation = 0x4, Sequence = 0x0 Misc = NO_PRESENCE Pin-ctls: 0x20: IN Node 0x20 [Vendor Defined Widget] wcaps 0xf00040: Mono Processing caps: benign=0, ncoeff=17 Processing Coefficient: 0xc128 Coefficient Index: 0x02 Node 0x21 [Volume Knob Widget] wcaps 0x600080: Mono Volume-Knob: delta=0, steps=32, direct=0, val=64 Unsolicited: tag=00, enabled=0 Connection: 0 Node 0x22 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Input Source", index=0, device=0 Control: name="Input Source", index=1, device=0 Control: name="Input Source", index=2, device=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x23 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Input Source", index=0, device=0 Control: name="Input Source", index=1, device=0 Control: name="Input Source", index=2, device=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x24 [Audio Mixer] wcaps 0x20010b: Stereo Amp-In Control: name="Input Source", index=0, device=0 Control: name="Input Source", index=1, device=0 Control: name="Input Source", index=2, device=0 Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x80 0x80] [0x00 0x00] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] [0x80 0x80] Connection: 11 0x18 0x19 0x1a 0x1b 0x1c 0x1d 0x14 0x15 0x16 0x17 0x0b Node 0x25 [Audio Output] wcaps 0x11: Stereo Converter: stream=0, channel=0 PCM: rates [0x560]: 44100 48000 96000 192000 bits [0xe]: 16 20 24 formats [0x1]: PCM Node 0x26 [Audio Mixer] wcaps 0x20010f: Stereo Amp-In Amp-Out Amp-In caps: ofs=0x00, nsteps=0x00, stepsize=0x00, mute=1 Amp-In vals: [0x00 0x00] [0x80 0x80] Amp-Out caps: ofs=0x40, nsteps=0x40, stepsize=0x03, mute=0 Amp-Out vals: [0x40 0x40] Connection: 2 0x25 0x0b --endcollapse-- !!ALSA Device nodes !!----------------- crw-rw---- 1 root audio 116, 0 Jan 7 22:48 /dev/snd/controlC0 crw-rw---- 1 root audio 116, 4 Jan 7 22:48 /dev/snd/hwC0D0 crw-rw---- 1 root audio 116, 24 Jan 7 22:48 /dev/snd/pcmC0D0c crw-rw---- 1 root audio 116, 16 Jan 7 23:01 /dev/snd/pcmC0D0p crw-rw---- 1 root audio 116, 25 Jan 7 22:48 /dev/snd/pcmC0D1c crw-rw---- 1 root audio 116, 17 Jan 7 22:48 /dev/snd/pcmC0D1p crw-rw---- 1 root audio 116, 26 Jan 7 22:48 /dev/snd/pcmC0D2c crw-rw---- 1 root audio 116, 1 Jan 7 22:47 /dev/snd/seq crw-rw---- 1 root audio 116, 33 Jan 7 22:47 /dev/snd/timer /dev/snd/by-path: total 0 drwxr-xr-x 2 root root 60 Jan 7 22:48 . drwxr-xr-x 3 root root 240 Jan 7 22:48 .. lrwxrwxrwx 1 root root 12 Jan 7 22:48 pci-0000:00:0f.1 -> ../controlC0 !!Aplay/Arecord output !!------------ APLAY **** List of PLAYBACK Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: ALC889A Analog [ALC889A Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: NVidia [HDA NVidia], device 1: ALC889A Digital [ALC889A Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 ARECORD **** List of CAPTURE Hardware Devices **** card 0: NVidia [HDA NVidia], device 0: ALC889A Analog [ALC889A Analog] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: NVidia [HDA NVidia], device 1: ALC889A Digital [ALC889A Digital] Subdevices: 1/1 Subdevice #0: subdevice #0 card 0: NVidia [HDA NVidia], device 2: ALC889A Analog [ALC889A Analog] Subdevices: 2/2 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 !!Amixer output !!------------- !!-------Mixer controls for card 0 [NVidia] Card hw:0 'NVidia'/'HDA NVidia at 0xcfff0000 irq 25' Mixer name : 'Realtek ALC889A' Components : 'HDA:10ec0885,10ece601,00100101' Controls : 36 Simple ctrls : 20 Simple mixer control 'Master',0 Capabilities: pvolume pvolume-joined pswitch pswitch-joined Playback channels: Mono Limits: Playback 0 - 64 Mono: Playback 44 [69%] [-20.00dB] [on] Simple mixer control 'PCM',0 Capabilities: pvolume Playback channels: Front Left - Front Right Limits: Playback 0 - 255 Mono: Front Left: Playback 255 [100%] [0.00dB] Front Right: Playback 255 [100%] [0.00dB] Simple mixer control 'Front',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 64 [100%] [0.00dB] [on] Front Right: Playback 64 [100%] [0.00dB] [on] Simple mixer control 'Surround',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 49 [77%] [-15.00dB] [on] Front Right: Playback 49 [77%] [-15.00dB] [on] Simple mixer control 'LFE',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 45 [70%] [-19.00dB] [on] Front Right: Playback 0 [0%] [-64.00dB] [on] Simple mixer control 'Line',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Line Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] Front Right: 0 [0%] Simple mixer control 'Mic',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [off] Front Right: Playback 0 [0%] [-34.50dB] [off] Simple mixer control 'Mic Boost',0 Capabilities: volume Playback channels: Front Left - Front Right Capture channels: Front Left - Front Right Limits: 0 - 3 Front Left: 0 [0%] Front Right: 0 [0%] Simple mixer control 'IEC958',0 Capabilities: pswitch pswitch-joined cswitch cswitch-joined Playback channels: Mono Capture channels: Mono Mono: Playback [off] Capture [off] Simple mixer control 'IEC958 Default PCM',0 Capabilities: pswitch pswitch-joined Playback channels: Mono Mono: Playback [on] Simple mixer control 'Capture',0 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 46 Front Left: Capture 28 [61%] [12.00dB] [on] Front Right: Capture 28 [61%] [12.00dB] [on] Simple mixer control 'Capture',1 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 46 Front Left: Capture 0 [0%] [-16.00dB] [off] Front Right: Capture 0 [0%] [-16.00dB] [off] Simple mixer control 'Capture',2 Capabilities: cvolume cswitch Capture channels: Front Left - Front Right Limits: Capture 0 - 46 Front Left: Capture 0 [0%] [-16.00dB] [off] Front Right: Capture 0 [0%] [-16.00dB] [off] Simple mixer control 'Beep',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 31 Mono: Front Left: Playback 0 [0%] [-34.50dB] [on] Front Right: Playback 0 [0%] [-34.50dB] [on] Simple mixer control 'Channel Mode',0 Capabilities: enum Items: '2ch' '6ch' Item0: '2ch' Simple mixer control 'HP',0 Capabilities: pvolume pswitch Playback channels: Front Left - Front Right Limits: Playback 0 - 64 Mono: Front Left: Playback 0 [0%] [-64.00dB] [on] Front Right: Playback 0 [0%] [-64.00dB] [on] Simple mixer control 'Input Source',0 Capabilities: cenum Items: 'Mic' 'Line' 'CD' Item0: 'Mic' Simple mixer control 'Input Source',1 Capabilities: cenum Items: 'Mic' 'Line' 'CD' Item0: 'Mic' Simple mixer control 'Input Source',2 Capabilities: cenum Items: 'Mic' 'Line' 'CD' Item0: 'Mic' !!Alsactl output !!------------- --startcollapse-- state.NVidia { control.1 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 64' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Front Playback Volume' value.0 64 value.1 64 } control.2 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Front Playback Switch' value.0 true value.1 true } control.3 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 64' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Surround Playback Volume' value.0 49 value.1 49 } control.4 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Surround Playback Switch' value.0 true value.1 true } control.5 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 64' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'LFE Playback Volume' value.0 45 value.1 0 } control.6 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'LFE Playback Switch' value.0 true value.1 true } control.7 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 64' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'HP Playback Volume' value.0 0 value.1 0 } control.8 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'HP Playback Switch' value.0 true value.1 true } control.9 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Line Playback Volume' value.0 0 value.1 0 } control.10 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Line Playback Switch' value.0 false value.1 false } control.11 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Mic Playback Volume' value.0 0 value.1 0 } control.12 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Mic Playback Switch' value.0 false value.1 false } control.13 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 3' comment.dbmin 0 comment.dbmax 3000 iface MIXER name 'Line Boost' value.0 0 value.1 0 } control.14 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 3' comment.dbmin 0 comment.dbmax 3000 iface MIXER name 'Mic Boost' value.0 0 value.1 0 } control.15 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 '2ch' comment.item.1 '6ch' iface MIXER name 'Channel Mode' value '2ch' } control.16 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Capture Switch' value.0 true value.1 true } control.17 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Capture Switch' index 1 value.0 false value.1 false } control.18 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Capture Switch' index 2 value.0 false value.1 false } control.19 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 46' comment.dbmin -1600 comment.dbmax 3000 iface MIXER name 'Capture Volume' value.0 28 value.1 28 } control.20 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 46' comment.dbmin -1600 comment.dbmax 3000 iface MIXER name 'Capture Volume' index 1 value.0 0 value.1 0 } control.21 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 46' comment.dbmin -1600 comment.dbmax 3000 iface MIXER name 'Capture Volume' index 2 value.0 0 value.1 0 } control.22 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mic comment.item.1 Line comment.item.2 CD iface MIXER name 'Input Source' value Mic } control.23 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mic comment.item.1 Line comment.item.2 CD iface MIXER name 'Input Source' index 1 value Mic } control.24 { comment.access 'read write' comment.type ENUMERATED comment.count 1 comment.item.0 Mic comment.item.1 Line comment.item.2 CD iface MIXER name 'Input Source' index 2 value Mic } control.25 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Con Mask' value '0fff000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.26 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Pro Mask' value '0f00000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.27 { comment.access 'read write' comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Playback Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.28 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'IEC958 Playback Switch' value false } control.29 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'IEC958 Default PCM Playback Switch' value true } control.30 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'IEC958 Capture Switch' value false } control.31 { comment.access read comment.type IEC958 comment.count 1 iface MIXER name 'IEC958 Capture Default' value '0400000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000000' } control.32 { comment.access 'read write' comment.type INTEGER comment.count 2 comment.range '0 - 31' comment.dbmin -3450 comment.dbmax 1200 iface MIXER name 'Beep Playback Volume' value.0 0 value.1 0 } control.33 { comment.access 'read write' comment.type BOOLEAN comment.count 2 iface MIXER name 'Beep Playback Switch' value.0 true value.1 true } control.34 { comment.access 'read write' comment.type INTEGER comment.count 1 comment.range '0 - 64' comment.dbmin -6400 comment.dbmax 0 iface MIXER name 'Master Playback Volume' value 44 } control.35 { comment.access 'read write' comment.type BOOLEAN comment.count 1 iface MIXER name 'Master Playback Switch' value true } control.36 { comment.access 'read write user' comment.type INTEGER comment.count 2 comment.range '0 - 255' comment.tlv '0000000100000008ffffec1400000014' comment.dbmin -5100 comment.dbmax 0 iface MIXER name 'PCM Playback Volume' value.0 255 value.1 255 } } --endcollapse-- !!All Loaded Modules !!------------------ Module binfmt_misc ppdev rfcomm sco bridge stp bnep l2cap crc16 ipt_ULOG battery cpufreq_powersave cpufreq_performance af_packet usblp ipv6 snd_hda_codec_realtek snd_hda_intel arc4 snd_hda_codec ecb snd_hwdep snd_pcm_oss snd_mixer_oss iptable_filter ip_tables x_tables b43 snd_pcm rng_core snd_seq_oss snd_seq_midi_event snd_seq mac80211 snd_timer snd_seq_device btusb snd cfg80211 soundcore i2c_nforce2 snd_page_alloc led_class bluetooth rfkill lp evdev rtc_cmos processor parport ext3 jbd mbcache sr_mod usbhid hid sg cdrom sd_mod crc_t10dif ata_generic usb_storage fan ohci_hcd pata_amd fuse ssb pata_acpi ehci_hcd sata_nv floppy nvidia mmc_core pcmcia forcedeth pcmcia_core usbcore nls_base i2c_core libata thermal scsi_mod button !!Sysfs Files !!----------- /sys/class/sound/hwC0D0/init_pin_configs: 0x14 0x01014010 0x15 0x01011012 0x16 0x01016011 0x17 0x01012014 0x18 0x01a19830 0x19 0x02a1983f 0x1a 0x01813031 0x1b 0x02214c1f 0x1c 0x411111f0 0x1d 0x99830132 0x1e 0x01441120 0x1f 0x01c46140 /sys/class/sound/hwC0D0/driver_pin_configs: /sys/class/sound/hwC0D0/user_pin_configs: /sys/class/sound/hwC0D0/init_verbs: !!ALSA/HDA dmesg !!------------------ ACPI: PCI Interrupt Link [AAZA] enabled at IRQ 20 HDA Intel 0000:00:0f.1: PCI INT B -> Link[AAZA] -> GSI 20 (level, low) -> IRQ 20 alloc irq_desc for 25 on node -1 alloc kstat_irqs on node -1 HDA Intel 0000:00:0f.1: irq 25 for MSI/MSI-X HDA Intel 0000:00:0f.1: setting latency timer to 64 input: HDA Digital PCBeep as /devices/pci0000:00/0000:00:0f.1/input/input5 EXT3 FS on sda3, internal journal

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[alsa-devel] [PATCH] sbawe: fix memory detection part 2
by Krzysztof Helt 08 Jan '10

08 Jan '10

From: Krzysztof Helt <krzysztof.h1(a)wp.pl> The patch "sbawe: fix memory detection" fixed detection for memoryless SB32 cards but broke detection of memory above 512KB. This patch fixes the regression. The patch has been tested on the SB32 card (CT3670) with 0MB, 2MB and 8MB memory installed. Signed-off-by: Krzysztof Helt <krzysztof.h1(a)wp.pl> --- sound/isa/sb/emu8000.c | 11 +++++++---- 1 files changed, 7 insertions(+), 4 deletions(-) diff --git a/sound/isa/sb/emu8000.c b/sound/isa/sb/emu8000.c index 751762f..0c40951 100644 --- a/sound/isa/sb/emu8000.c +++ b/sound/isa/sb/emu8000.c @@ -377,12 +377,13 @@ init_arrays(struct snd_emu8000 *emu) static void __devinit size_dram(struct snd_emu8000 *emu) { - int i, size; + int i, size, detected_size; if (emu->dram_checked) return; size = 0; + detected_size = 0; /* write out a magic number */ snd_emu8000_dma_chan(emu, 0, EMU8000_RAM_WRITE); @@ -393,6 +394,8 @@ size_dram(struct snd_emu8000 *emu) while (size < EMU8000_MAX_DRAM) { + size += 512 * 1024; /* increment 512kbytes */ + /* Write a unique data on the test address. * if the address is out of range, the data is written on * 0x200000(=EMU8000_DRAM_OFFSET). Then the id word is @@ -414,7 +417,7 @@ size_dram(struct snd_emu8000 *emu) if (EMU8000_SMLD_READ(emu) != UNIQUE_ID2) break; /* no memory at this address */ - size += 512 * 1024; /* increment 512kbytes */ + detected_size = size; snd_emu8000_read_wait(emu); @@ -442,9 +445,9 @@ size_dram(struct snd_emu8000 *emu) snd_emu8000_dma_chan(emu, 1, EMU8000_RAM_CLOSE); snd_printdd("EMU8000 [0x%lx]: %d Kb on-board memory detected\n", - emu->port1, size/1024); + emu->port1, detected_size/1024); - emu->mem_size = size; + emu->mem_size = detected_size; emu->dram_checked = 1; } -- 1.6.4 ---------------------------------------------------------------------- Sypie, sypie, ale czy sypac bedzie nadal? Sprawdz pogode >> http://link.interia.pl/f2530

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