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Re: [alsa-devel] trace PCM data from usb bus to alsa lib
by Takashi Iwai 18 Jun '09

18 Jun '09

At Wed, 17 Jun 2009 16:10:26 +0800, Zero Luo wrote: > > Dear Takashi, > > Data from usb dongle is 16K*16 bit PCM data. And what is the format and channels you are using in alsa-lib? alsa-lib itself can convert formats and channels automatically if needed. Takashi > 2009/6/17, Takashi Iwai <tiwai(a)suse.de>: > > At Wed, 17 Jun 2009 07:25:25 +0000 (UTC), > ZeroCluo wrote: > > > > Hi > > > > I am trying to trace the Audio data from usb bus. > > in kernel: > > dump data in retire_capture_urb() ,found data format is like: > > AABB 0000 CCDD 0000 ; > > > > but in alsa library , > > dump data in snd_pcm_mmap_begin(),the "0000" is cut,only "AABB CCDD", > > > > Do you know when and where those "0000" is cut? > > Are you using the native format that the hardware runs? > > Takashi > >

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Re: [alsa-devel] snd-ctxfi oops
by The Source 18 Jun '09

18 Jun '09

On 15.06.2009 17:01, Takashi Iwai wrote: > At Mon, 15 Jun 2009 15:00:03 +0200, > I wrote: > >> At Mon, 15 Jun 2009 16:59:53 +0400, >> The Source wrote: >> >>> On 15.06.2009 16:56, Takashi Iwai wrote: >>> >>>> At Sat, 13 Jun 2009 10:07:49 +0200, >>>> I wrote: >>>> >>>> >>>>> >>>>> >>>>>> Tested the patch more. Really no oopses. Instead of them I get rare >>>>>> system lockups. >>>>>> >>>>>> >>>>> In which situation, exactly? >>>>> Could you check whether it happens with use_system_timer=1 module >>>>> option, too? >>>>> >>>>> >>>> Or try the very latest alsa-driver-snapshot (not daily tarball). >>>> I fixed a deadlock issue in cttimer.c now. >>>> >>>> >>>> thanks, >>>> >>>> Takashi >>>> >>>> >>>> >>> Should I apply that patch to this snapshot? >>> >> Which patch do you mean? >> Just grab the latest alsa-driver-snapshot.tar.gz from >> ftp://ftp.kernel.org/pub/linux/kernel/people/tiwai/snapshot/ >> > Also, check alsa-driver/alsa-kernel/HEAD whether the commit > 8dca419721d188bfee5f19fad45275856c619a5c > ALSA: ctxfi - Fix deadlock with xfi-timer > is included. > > > Takashi > > Tested the driver and it's fine so far. No craches with any application including Doom 3, which caused system lockups often with earlier versions of the driver.

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[alsa-devel] [PATCH] alsa-plugins/pulse: Implement 'pause'.
by Troy Moure 17 Jun '09

17 Jun '09

Just cork or uncork the stream to pause or unpause it. Signed-off-by: Troy Moure <twmoure(a)szypr.net> --- I'm not very familiar with alsa or pulseaudio, so this might not be right. If it is, feel free to apply it, though. It *does* fix my problem, which was that pause and unpause didn't work in 'cmus' after I upgraded to Fedora 11. Since the pulse plug-in didn't support snd_pcm_pause, pausing in cmus would just cause cmus to stop writing to the buffer, leading to buffer underruns in pulseaudio. With this patch, pausing in cmus works perfectly for me. If this should be fixed another way, please let me know what it is! Thanks. pulse/pcm_pulse.c | 20 ++++++++++++++++++++ 1 files changed, 20 insertions(+), 0 deletions(-) diff --git a/pulse/pcm_pulse.c b/pulse/pcm_pulse.c index db8d1e1..c322884 100644 --- a/pulse/pcm_pulse.c +++ b/pulse/pcm_pulse.c @@ -739,6 +739,25 @@ static int pulse_close(snd_pcm_ioplug_t * io) return 0; } +static int pulse_pause(snd_pcm_ioplug_t * io, int enable) +{ + snd_pcm_pulse_t *pcm = io->private_data; + int err = 0; + + assert (pcm); + assert (pcm->p); + + pa_threaded_mainloop_lock(pcm->p->mainloop); + + if (pcm->stream) + if (!pa_stream_cork(pcm->stream, enable, NULL, NULL)) + err = -EIO; + + pa_threaded_mainloop_unlock(pcm->p->mainloop); + + return err; +} + static const snd_pcm_ioplug_callback_t pulse_playback_callback = { .start = pulse_start, .stop = pulse_stop, @@ -750,6 +769,7 @@ static const snd_pcm_ioplug_callback_t pulse_playback_callback = { .prepare = pulse_prepare, .hw_params = pulse_hw_params, .close = pulse_close, + .pause = pulse_pause }; -- 1.6.2.2

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[alsa-devel] ICE1712 softdevices fixes
by Stefan Kost 17 Jun '09

17 Jun '09

hi, > The following patches fixes the default device as they appear for > ICE1712-based cards. Without this cards, 'aplay -L' will show a series of > devices like surround40 and surround71 that are not meaningful for this card, > and a front device that would not work. > > Diego E. 'Flameeyes' Pettenò (2): Remove definition for surround sound > devices on ICE1712. Make the front:idx device work for ICE1712. thanks for making those. this was exactly what I needed to get pulseaudio working with my card again. I also agree that even though that my card (Hoontech DSP 24) has 8/10 physical in/outs, those are essentially mono channels. The surround mapping seem to be more harm that use to me. If a musician indeed wires 6 3.5 jack outs somehow to his 5.1 amp, then its probably okay to uncomment such mappings. So what about making diegos patches the default? Stefan

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[alsa-devel] [PATCH] ALSA: snd_usb_caiaq: fix legacy input streaming
by Daniel Mack 17 Jun '09

17 Jun '09

Seems that nobody recently tried the input on the very first supported sound card model, RK2. This patch fixes the byte offset to make it running again. Signed-off-by: Daniel Mack <daniel(a)caiaq.de> --- sound/usb/caiaq/audio.c | 5 +++-- sound/usb/caiaq/device.c | 2 +- 2 files changed, 4 insertions(+), 3 deletions(-) diff --git a/sound/usb/caiaq/audio.c b/sound/usb/caiaq/audio.c index b144513..8f9b60c 100644 --- a/sound/usb/caiaq/audio.c +++ b/sound/usb/caiaq/audio.c @@ -199,8 +199,9 @@ static int snd_usb_caiaq_pcm_prepare(struct snd_pcm_substream *substream) dev->period_out_count[index] = BYTES_PER_SAMPLE + 1; dev->audio_out_buf_pos[index] = BYTES_PER_SAMPLE + 1; } else { - dev->period_in_count[index] = BYTES_PER_SAMPLE; - dev->audio_in_buf_pos[index] = BYTES_PER_SAMPLE; + int in_pos = (dev->spec.data_alignment == 2) ? 0 : 2; + dev->period_in_count[index] = BYTES_PER_SAMPLE + in_pos; + dev->audio_in_buf_pos[index] = BYTES_PER_SAMPLE + in_pos; } if (dev->streaming) diff --git a/sound/usb/caiaq/device.c b/sound/usb/caiaq/device.c index 2240624..0e5db71 100644 --- a/sound/usb/caiaq/device.c +++ b/sound/usb/caiaq/device.c @@ -35,7 +35,7 @@ #include "input.h" MODULE_AUTHOR("Daniel Mack <daniel(a)caiaq.de>"); -MODULE_DESCRIPTION("caiaq USB audio, version 1.3.16"); +MODULE_DESCRIPTION("caiaq USB audio, version 1.3.17"); MODULE_LICENSE("GPL"); MODULE_SUPPORTED_DEVICE("{{Native Instruments, RigKontrol2}," "{Native Instruments, RigKontrol3}," -- 1.6.3.1

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[alsa-devel] sound device is buzy to run pcm.c
by Guilherme Raymo Longo 17 Jun '09

17 Jun '09

I am getting this problem when running the pcm.c demo program at http://alsa-project.org/alsa-doc/alsa-lib/_2test_2pcm_8c-example.html This is the execution of the program: guilherme@lap:/home/guilherme/Experimentos# ./PCM Playback device is plughw:0,0 Stream parameters are 44100Hz, S16_LE, 1 channels sine wave rate is 0 0.0000Hz Using transfer method: write Playback open error: Device or resource busy I tried to check if the device is really buzy: root@lap:/home/guilherme/Experimentos# lsof |grep snd lsof: WARNING: can't stat() fuse.gvfs-fuse-daemon file system */ home/guilherme/*.gvfs Output information may be incomplete. esd 5433 guilherme mem CHR 116,16 6947 /dev/snd/pcmC0D0p esd 5433 guilherme 5r CHR 116,33 6787 /dev/snd/timer esd 5433 guilherme 6w CHR 116,16 6947 /dev/snd/pcmC0D0p esd 5433 guilherme 7r CHR 116,0 6987 /dev/snd/controlC0 and it seems to be ok. That's my card root@lap:/home/guilherme/Experimentos# aplay -l **** List of PLAYBACK Hardware Devices **** card 0: SB [HDA ATI SB], device 0: ALC883 Analog [ALC883 Analog] Subdevices: 0/1 Subdevice #0: subdevice #0 Is there a way to free the device or smth is wrong with my code?? I am using slackware 12.2 kernel 2.6 Great Regads;

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[alsa-devel] OSS support
by The Source 17 Jun '09

17 Jun '09

Is oss emulation still supported in latest alsa-driver snapshot? I installed that driver in Fedora 11 on both my static machine (X-Fi platinum fatality champion) and laptop (HDA Intel) and I don't have /dev/dsp or /dev/mixer. Should I compile the driver with some specific flag?

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[alsa-devel] [PATCH 3/3] ASoC: pxa-ssp.c handle the I2S and LEFT_J cases.
by Daniel Ribeiro 17 Jun '09

17 Jun '09

Handles I2S and LEFT_J cases, if the user does _not_ call set_tdm_slot we assume that the codec is I2S compliant, and that we can use frame_width = sample_width * channels. If the user calls set_tdm_slot then we assume that the wire is _not_ networked, and that the user wants to set a non-standard frame format instead, in this case we set the frame duration (the LRCLK falling edge) to be half of the wire frame width. eg. * standard 2*[16|24|32|..] bits I2S/LEFT_J doesn't need to call set_tdm_slot * to use 2*16bits I2S/LEFTJ on 64bits frames, with the LRCLK falling edge at the 32nd bitclk (+1 for I2S) needs to set_tdm_slot(5, 5, 4, 16) (4 slots of 16 bits, first and third RX/TX slots active). This should work even if pxa is slave of SFRM. Signed-off-by: Daniel Ribeiro <drwyrm(a)gmail.com> --- arch/arm/mach-pxa/include/mach/regs-ssp.h | 14 ++-- sound/soc/pxa/pxa-ssp.c | 121 +++++++++++++--------------- 2 files changed, 63 insertions(+), 72 deletions(-) diff --git a/arch/arm/mach-pxa/include/mach/regs-ssp.h b/arch/arm/mach-pxa/include/mach/regs-ssp.h index 6a2ed35..060e23b 100644 --- a/arch/arm/mach-pxa/include/mach/regs-ssp.h +++ b/arch/arm/mach-pxa/include/mach/regs-ssp.h @@ -108,21 +108,21 @@ #define SSSR_TINT (1 << 19) /* Receiver Time-out Interrupt */ #define SSSR_PINT (1 << 18) /* Peripheral Trailing Byte Interrupt */ -#if defined(CONFIG_PXA3xx) -#define SSPSP_EDMYSTOP(x) ((x) << 28) /* Extended Dummy Stop */ -#define SSPSP_EDMYSTRT(x) ((x) << 26) /* Extended Dummy Start */ -#endif - #define SSPSP_FSRT (1 << 25) /* Frame Sync Relative Timing */ -#define SSPSP_DMYSTOP(x) ((x) << 23) /* Dummy Stop */ #define SSPSP_SFRMWDTH(x) ((x) << 16) /* Serial Frame Width */ #define SSPSP_SFRMDLY(x) ((x) << 9) /* Serial Frame Delay */ -#define SSPSP_DMYSTRT(x) ((x) << 7) /* Dummy Start */ #define SSPSP_STRTDLY(x) ((x) << 4) /* Start Delay */ #define SSPSP_ETDS (1 << 3) /* End of Transfer data State */ #define SSPSP_SFRMP (1 << 2) /* Serial Frame Polarity */ #define SSPSP_SCMODE(x) ((x) << 0) /* Serial Bit Rate Clock Mode */ +/* NOTE: PXA3xx extends the bit number of dummy start and stop, the macros + * below are compatible with PXA25x/27x as long as the parameter is within + * the correct limits, driver code has to take care of this. + */ +#define SSPSP_DMYSTRT(x) ((((x) & 3) << 7) | ((((x) >> 2) & 3) << 26)) +#define SSPSP_DMYSTOP(x) ((((x) & 3) << 23) | ((((x) >> 2) & 7) << 28)) + #define SSACD_SCDB (1 << 3) /* SSPSYSCLK Divider Bypass */ #define SSACD_ACPS(x) ((x) << 4) /* Audio clock PLL select */ #define SSACD_ACDS(x) ((x) << 0) /* Audio clock divider select */ diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c index d60492e..aac85fe 100644 --- a/sound/soc/pxa/pxa-ssp.c +++ b/sound/soc/pxa/pxa-ssp.c @@ -179,21 +179,6 @@ static void ssp_set_scr(struct ssp_device *ssp, u32 div) ssp_write_reg(ssp, SSCR0, sscr0); } -/** - * ssp_get_clkdiv - get SSP clock divider - */ -static u32 ssp_get_scr(struct ssp_device *ssp) -{ - u32 sscr0 = ssp_read_reg(ssp, SSCR0); - u32 div; - - if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) - div = ((sscr0 >> 8) & 0xff) * 2 + 2; - else - div = ((sscr0 >> 8) & 0xfff) + 1; - return div; -} - /* * Set the SSP ports SYSCLK. */ @@ -487,17 +472,14 @@ static int pxa_ssp_set_dai_fmt(struct snd_soc_dai *cpu_dai, } switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: - sscr0 |= SSCR0_PSP; - sscr1 |= SSCR1_RWOT | SSCR1_TRAIL; - /* See hw_params() */ - break; - case SND_SOC_DAIFMT_DSP_A: sspsp |= SSPSP_FSRT; case SND_SOC_DAIFMT_DSP_B: + case SND_SOC_DAIFMT_LEFT_J: + case SND_SOC_DAIFMT_I2S: sscr0 |= SSCR0_PSP; sscr1 |= SSCR1_TRAIL | SSCR1_RWOT; + /* See hw_params() for I2S and LEFT_J */ break; default: @@ -561,6 +543,59 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, sscr0 |= SSCR0_FPCKE; #endif + sspsp = ssp_read_reg(ssp, SSPSP); + switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { + case SND_SOC_DAIFMT_I2S: + /* + * I2S and LEFT_J are stereo only, we have to send data for + * both channels. + */ + if (chn == 1) + frame_width *= 2; + + /* + * If the user did not use network mode, we assume the codec + * is I2S compliant. + */ + if (frame_width > 0) { + sspsp |= SSPSP_SFRMWDTH(frame_width / 2); + sspsp |= SSPSP_FSRT; + } else { + /* + * Otherwise we assume that it is a single TDM slot, and + * the user is abusing set_tdm_slot to support an + * out of spec codec. + */ + int slots = ((sscr0 & SSCR0_SlotsPerFrm(8)) >> 24) + 1; + + /* PXA2XX doesn't support DMYSTOP > 3 */ + if (slot_width != (width * 2) && !cpu_is_pxa3xx()) + return -EINVAL; + + sspsp |= SSPSP_DMYSTRT(1); + sspsp |= SSPSP_DMYSTOP( + (slot_width * slots) / 2 - width - 1); + sspsp |= SSPSP_SFRMWDTH((slot_width * slots) / 2); + } + break; + + case SND_SOC_DAIFMT_LEFT_J: + if (chn == 1) + frame_width *= 2; + + if (frame_width > 0) { + sspsp |= SSPSP_SFRMWDTH(frame_width / 2); + } else { + int slots = ((sscr0 & SSCR0_SlotsPerFrm(8)) >> 24) + 1; + + sspsp |= SSPSP_SFRMWDTH((slot_width * slots) / 2); + } + break; + default: + break; + } + ssp_write_reg(ssp, SSPSP, sspsp); + if (frame_width > 0) { /* Not using network mode */ if (frame_width > 16) @@ -598,50 +633,6 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream, ssp_write_reg(ssp, SSCR0, sscr0); - switch (priv->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) { - case SND_SOC_DAIFMT_I2S: - sspsp = ssp_read_reg(ssp, SSPSP); - - if ((ssp_get_scr(ssp) == 4) && (width == 16)) { - /* This is a special case where the bitclk is 64fs - * and we're not dealing with 2*32 bits of audio - * samples. - * - * The SSP values used for that are all found out by - * trying and failing a lot; some of the registers - * needed for that mode are only available on PXA3xx. - */ - -#ifdef CONFIG_PXA3xx - if (!cpu_is_pxa3xx()) - return -EINVAL; - - sspsp |= SSPSP_SFRMWDTH(width * 2); - sspsp |= SSPSP_SFRMDLY(width * 4); - sspsp |= SSPSP_EDMYSTOP(3); - sspsp |= SSPSP_DMYSTOP(3); - sspsp |= SSPSP_DMYSTRT(1); -#else - return -EINVAL; -#endif - } else { - /* The frame width is the width the LRCLK is - * asserted for; the delay is expressed in - * half cycle units. We need the extra cycle - * because the data starts clocking out one BCLK - * after LRCLK changes polarity. - */ - sspsp |= SSPSP_SFRMWDTH(width + 1); - sspsp |= SSPSP_SFRMDLY((width + 1) * 2); - sspsp |= SSPSP_DMYSTRT(1); - } - - ssp_write_reg(ssp, SSPSP, sspsp); - break; - default: - break; - } - dump_registers(ssp); return 0; -- tg: (119677b..) asoc/leftj-and-i2s (depends on: asoc/ssp-internals) -- Daniel Ribeiro

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[alsa-devel] trace PCM data from usb bus to alsa lib
by ZeroCluo 17 Jun '09

17 Jun '09

Hi I am trying to trace the Audio data from usb bus. in kernel: dump data in retire_capture_urb() ,found data format is like: AABB 0000 CCDD 0000 ; but in alsa library , dump data in snd_pcm_mmap_begin(),the "0000" is cut,only "AABB CCDD", Do you know when and where those "0000" is cut? btw,use mmap method map dma_area to user space; and use snd_pcm_readi to get audio data; Thank you very very much. Best Regards, Zero

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Re: [alsa-devel] No jack sense on Intel 82801I / IDT 92HD71B7X in a HP dv4
by Wael Nasreddine (a.k.a eMxyzptlk) 17 Jun '09

17 Jun '09

All the information you need are here -> http://www.alsa-project.org/db/?f=6b588ea53af5cf2794fb8b4239311722809b3c34 2009/6/15 Jean-Pierre André <jean-pierre.andre(a)wanadoo.fr>: > Hi Wael, > > Wael Nasreddine (a.k.a eMxyzptlk) wrote: >> >> I don't know if this concerns my situation ( >> http://mailman.alsa-project.org/pipermail/alsa-devel/2009-June/018262.html >> ) or not.. >> >> headphones work for me, front speaker work but LFE doesn't and the >> Irq22 stays on 0 too >> >> # cat /proc/interrupts | grep '22:' >> 22: 0 0 IO-APIC-fasteoi uhci_hcd:usb6, HDA Intel >> >> Takashi told me to play with hda-analyser and hda-verb though I don't >> what to do with hda-verb and I don't if what hda-analyser showing me >> is the current situation or the desired one.. >> >> Thanks > > So it looks like this a general situation on this hardware. > > If you get sound, at least from the front speakers, > without the IRQ22 being triggered, what are your > kernel and alsa versions, your boot parameters and > alsa (modprobe) parameters ? > > I have no knowledge about the hda-verb, etc. > so I am not to help you much in this area. > > Regards > > Jean-Pierre > > > > -- Wael Nasreddine Blog : http://wael.nasreddine.com E-mail : wael.nasreddine(a)gmail.com gTalk : wael.nasreddine(a)gmail.com Tel : +33.6.32.94.70.13 Skype : eMxyzptlk Twitter : @eMxyzptlk Sabayon Linux Chief Development Officer - http://www.sabayonlinux.org PGP: 1024D/C8DD18A2 06F6 1622 4BC8 4CEB D724 DE12 5565 3945 C8DD 18A2 .: An infinite number of monkeys typing into GNU emacs, would never make a good program. (L. Torvalds 1995) :.

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