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[alsa-devel] [PATCH] alsa-plugins/pulse: Implement 'pause'.
by Troy Moure 18 Jun '09

18 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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[alsa-devel] [PATCH 1/1] ALC889A chip identification
by Torben Schulz 17 Jun '09

17 Jun '09

With this patch the ALC889A chip will be identified correctly. The functions patch_alc882() and patch_alc883() have been altered to set the stream names correctly. Without this patch 'aplay -l' would identify an ALC889A chip as ALC885 for example. Apart from that a kernel info message will be printed if ALC885 or ALC889A chips are handled by the wrong patch function. Signed-off by: Torben Schulz <public(a)letorbi.de> --- alsa-driver/alsa-kernel/pci/hda/patch_realtek.c 2009-05-14 18:23:17.000000000 +0200 +++ alsa-driver-mod/alsa-kernel/pci/hda/patch_realtek.c 2009-05-15 01:08:16.000000000 +0200 @@ -7338,8 +7338,17 @@ setup_preset(spec, &alc882_presets[board_config]); if (codec->vendor_id == 0x10ec0885) { - spec->stream_name_analog = "ALC885 Analog"; - spec->stream_name_digital = "ALC885 Digital"; + if ((codec->revision_id == 0x100101) || + (codec->revision_id == 0x100103)) { + spec->stream_name_analog = "ALC889A Analog"; + spec->stream_name_digital = "ALC889A Digital"; + printk(KERN_INFO + "hda_codec: ALC889A handled by patch_alc882()," + " should be handled by patch_alc883()\n"); + } else { + spec->stream_name_analog = "ALC885 Analog"; + spec->stream_name_digital = "ALC885 Digital"; + } } else { spec->stream_name_analog = "ALC882 Analog"; spec->stream_name_digital = "ALC882 Digital"; @@ -9231,6 +9240,27 @@ setup_preset(spec, &alc883_presets[board_config]); switch (codec->vendor_id) { + case 0x10ec0885: + if ((codec->revision_id == 0x100101) || + (codec->revision_id == 0x100103)) { + spec->stream_name_analog = "ALC889A Analog"; + spec->stream_name_digital = "ALC889A Digital"; + if (!spec->num_adc_nids) { + spec->num_adc_nids = ARRAY_SIZE(alc883_adc_nids); + spec->adc_nids = alc883_adc_nids; + } + if (!spec->capsrc_nids) + spec->capsrc_nids = alc883_capsrc_nids; + spec->capture_style = CAPT_MIX; /* matrix-style capture */ + spec->init_amp = ALC_INIT_DEFAULT; /* always initialize */ + } else { + spec->stream_name_analog = "ALC885 Analog"; + spec->stream_name_digital = "ALC885 Digital"; + printk(KERN_INFO + "hda_codec: ALC885 handled by patch_alc883()," + " should be handled by patch_alc882()\n"); + } + break; case 0x10ec0888: if (codec->revision_id == 0x100101) { spec->stream_name_analog = "ALC1200 Analog";

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[alsa-devel] Playback open error: Device or resource busy
by Guilherme 17 Jun '09

17 Jun '09

Mates... I am running a simple app that I found called pcm.c and I can't get through with that. It creates and send a simple sine wave to the speakers! When I run the app I get the following error: 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 Every application I start playbacks the sound without a problem. Is there any special conf I should do to free the Device? Here is the code: #include <stdio.h> #include <stdlib.h> #include <string.h> #include <sched.h> #include <errno.h> #include <getopt.h> #include <alsa/asoundlib.h> #include <sys/time.h> #include <math.h> static char *device = "plughw:0,0"; //playback device aplay -l static snd_pcm_format_t format = SND_PCM_FORMAT_S16; //sample format unsigned 16 bit endian static unsigned int rate = 44100; //stream rate static unsigned int channels = 1; //contagem dos canais static unsigned int buffer_time = 500000; //ring buffer length on us static unsigned int period_time = 100000; //period time in us static double freq = 400; // frequencia da senóide em Hz static int verbose = 0; //verbose flag static int resample = 1; //abilitar alsa-lib resampling static int period_event = 0; // produz um event pool a cada periodo static snd_pcm_sframes_t buffer_size; //quantidade de buffers frames static snd_pcm_sframes_t period_size; //quantidade de periods frames static snd_output_t *output = NULL; //ALSA usa esse ponteiro para lidar com output objects static void generate_sine(const snd_pcm_channel_area_t *areas, snd_pcm_uframes_t offset, int count, double *_phase) { static double max_phase = 2. * M_PI; double phase = *_phase; double step = max_phase*freq / (double)rate; double res; unsigned char *samples[channels], *tmp; int steps[channels]; unsigned int chn, byte; union { int i; unsigned char c[4]; } ires; unsigned int maxval = (1 << (snd_pcm_format_width(format) - 1)) - 1; int bps = snd_pcm_format_width(format) / 8; //bits per sample /*verificar e preparar o conteúdo das areas*/ for (chn = 0; chn < channels; chn++) { if ((areas[chn].first % 8) != 0) { printf("areas[%i].first == %i, abortando...\n", chn, areas[chn].first); exit(EXIT_FAILURE); } //pega endereço e offset da area samples[chn] = /*(signed short *)*/(((unsigned char *)areas[chn].addr) + (areas[chn].first / 8)); if ((areas[chn].step % 16) != 0) { printf("areas[%i].step == %i, aborting...\n", chn, areas[chn].step); exit(EXIT_FAILURE); } steps[chn] = areas[chn].step / 8; samples[chn] += offset * steps[chn]; } //completa os channels da area while(count-- > 0) { res = sin(phase) * maxval; ires.i = res; tmp = ires.c; for (chn = 0; chn < channels; chn++) { for (byte = 0; byte < (unsigned int)bps; byte++) *(samples[chn] + byte) = tmp[byte]; samples[chn] += steps[chn]; } phase += step; if (phase >= max_phase) phase -= max_phase; } *_phase = phase; } static int set_hwparams(snd_pcm_t *handle, snd_pcm_hw_params_t *params, snd_pcm_access_t access) { unsigned int rrate; snd_pcm_uframes_t size; int err, dir; /*escolha todos os parametros*/ err = snd_pcm_hw_params_any(handle, params); if (err < 0) { printf("Broken configuration for playback: no configuration available: %s\n", snd_strerror(err)); return err; } /*set hardware resampling*/ err = snd_pcm_hw_params_set_rate_resample(handle, params, resample); if (err < 0) { printf("Resampling setup failed for playback: %s\n", snd_strerror(err)); return err; } /* set the interleaved read/write format */ err = snd_pcm_hw_params_set_access(handle, params, access); if (err < 0) { printf("Access type not available for playback: %s\n", snd_strerror(err)); return err; } /* set the sample format SND_PCM_FORMAT_S16 */ err = snd_pcm_hw_params_set_format(handle, params, format); if (err < 0) { printf("Sample format not available for playback: %s\n", snd_strerror(err)); return err; } /* set the count of channels 1 */ err = snd_pcm_hw_params_set_channels(handle, params, channels); if (err < 0) { printf("Channels count (%i) not available for playbacks: %s\n", channels, snd_strerror(err)); return err; } /* set the stream rate */ rrate = rate; err = snd_pcm_hw_params_set_rate_near(handle, params, &rrate, 0); if (err < 0) { printf("Rate %iHz not available for playback: %s\n", rate, snd_strerror(err)); return err; } if (rrate != rate) { printf("Rate doesn't match (requested %iHz, get %iHz)\n", rate, err); return -EINVAL; } /* set the buffer time */ err = snd_pcm_hw_params_set_buffer_time_near(handle, params, &buffer_time, &dir); if (err < 0) { printf("Unable to set buffer time %i for playback: %s\n", buffer_time, snd_strerror(err)); return err; } err = snd_pcm_hw_params_get_buffer_size(params, &size); if (err < 0) { printf("Unable to get buffer size for playback: %s\n", snd_strerror(err)); return err; } buffer_size = size; /* set the period time */ err = snd_pcm_hw_params_set_period_time_near(handle, params, &period_time, &dir); if (err < 0) { printf("Unable to set period time %i for playback: %s\n", period_time, snd_strerror(err)); return err; } err = snd_pcm_hw_params_get_period_size(params, &size, &dir); if (err < 0) { printf("Unable to get period size for playback: %s\n", snd_strerror(err)); return err; } period_size = size; /* write the parameters to device */ err = snd_pcm_hw_params(handle, params); if (err < 0) { printf("Unable to set hw params for playback: %s\n", snd_strerror(err)); return err; } return 0; } static int set_swparams(snd_pcm_t *handle, snd_pcm_sw_params_t *swparams) { int err; /*pegar os parametros atuais swparams*/ err = snd_pcm_sw_params_current(handle, swparams); if (err < 0) { printf("Unable to determine current swparams for playback: %s\n", snd_strerror(err)); return err; } /*Começa a transferência quando o buffer está quase cheio*/ /*(buffer_size / avail_min) * avail_min */ err = snd_pcm_sw_params_set_avail_min(handle, swparams, (buffer_size / period_size) * period_size); if (err < 0) { printf("Unable to set start threshold mode for playback: %s\n", snd_strerror(err)); return err; } /* allow the transfer when at least period_size samples can be processed */ /* or disable this mechanism when period event is enabled (aka interrupt like style processing) */ err = snd_pcm_sw_params_set_avail_min(handle, swparams, period_event ? buffer_size : period_size); if (err < 0) { printf("Unable to set avail min for playback: %s\n", snd_strerror(err)); return err; } /* enable period events when requested */ if (period_event) { err = snd_pcm_sw_params_set_period_event(handle, swparams, 1); if (err < 0) { printf("Unable to set period event: %s\n", snd_strerror(err)); return err; } } /* write the parameters to the playback device */ err = snd_pcm_sw_params(handle, swparams); if (err < 0) { printf("Unable to set sw params for playback: %s\n", snd_strerror(err)); return err; } return 0; } /* Tenta recuperar estados de underrun e suspend */ static int xrun_recovery(snd_pcm_t *handle, int err) { if (verbose) printf("stream recovery\n"); if (err == -EPIPE) { //under-run*/ err = snd_pcm_prepare(handle); if (err < 0) printf("Can't recovery from underrun, prepare failed: %s\n", snd_strerror(err)); return 0; } else if (err == -ESTRPIPE) { while ((err = snd_pcm_resume(handle)) == -EAGAIN) sleep(1); /*espera pela release flag*/ if (err < 0) { err = snd_pcm_prepare(handle); if (err < 0) printf("Can't recovery from suspend, prepare failed: %s\n", snd_strerror(err)); } return 0; } return err; } /* Metodo de transferência *WRITE-ONLY* */ static int write_loop(snd_pcm_t *handle, signed short *samples, snd_pcm_channel_area_t *areas) { double phase = 0; signed short *ptr; int err, cptr; while(1) { generate_sine(areas, 0, period_size, &phase); ptr = samples; cptr = period_size; while (cptr > 0) { err = snd_pcm_writei(handle, ptr, cptr); if (err == -EAGAIN) continue; if (err < 0) { if (xrun_recovery(handle, err) < 0) { printf("Write error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); }//if ptr += err * channels; cptr -= err; }//if }//if }//while } /* * Método de transferência - write and wait por espaço no buffer */ static int wait_for_poll(snd_pcm_t *handle, struct pollfd *ufds, unsigned int count) { unsigned short revents; while (1) { poll(ufds, count, -1); snd_pcm_poll_descriptors_revents(handle, ufds, count, &revents); if (revents & POLLERR) return -EIO; if (revents & POLLOUT) return 0; } } static int write_and_poll_loop(snd_pcm_t *handle, signed short *samples, snd_pcm_channel_area_t *areas) { struct pollfd *ufds; double phase = 0; signed short *ptr; int err, count, cptr, init; count = snd_pcm_poll_descriptors_count (handle); if (count <= 0) { printf("Invalid poll descriptors count\n"); return count; } ufds = malloc(sizeof(struct pollfd) * count); if (ufds == NULL) { printf("No enough memory\n"); return -ENOMEM; } if ((err = snd_pcm_poll_descriptors(handle, ufds, count)) < 0) { printf("Unable to obtain poll descriptors for playback: %s\n", snd_strerror(err)); return err; } init = 1; while (1) { if (!init) { err = wait_for_poll(handle, ufds, count); if (err < 0) { if (snd_pcm_state(handle) == SND_PCM_STATE_XRUN || snd_pcm_state(handle) == SND_PCM_STATE_SUSPENDED) { err = snd_pcm_state(handle) == SND_PCM_STATE_XRUN ? -EPIPE : -ESTRPIPE; if (xrun_recovery(handle, err) < 0) { printf("Write error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } init = 1; } else { printf("Wait for poll failed\n"); return err; } } } generate_sine(areas, 0, period_size, &phase); ptr = samples; cptr = period_size; while (cptr > 0) { err = snd_pcm_writei(handle, ptr, cptr); if (err < 0) { if (xrun_recovery(handle, err) < 0) { printf("Write error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } init = 1; break; /* skip one period */ } if (snd_pcm_state(handle) == SND_PCM_STATE_RUNNING) init = 0; ptr += err * channels; cptr -= err; if (cptr == 0) break; /* it is possible, that the initial buffer cannot store */ /* all data from the last period, so wait awhile */ err = wait_for_poll(handle, ufds, count); if (err < 0) { if (snd_pcm_state(handle) == SND_PCM_STATE_XRUN || snd_pcm_state(handle) == SND_PCM_STATE_SUSPENDED) { err = snd_pcm_state(handle) == SND_PCM_STATE_XRUN ? -EPIPE : -ESTRPIPE; if (xrun_recovery(handle, err) < 0) { printf("Write error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } init = 1; } else { printf("Wait for poll failed\n"); return err; } } } } } /* * Metodo de transferencia - Notificação assincrona */ struct async_private_data { signed short *samples; snd_pcm_channel_area_t *areas; double phase; }; static void async_callback(snd_async_handler_t *ahandler) { snd_pcm_t *handle = snd_async_handler_get_pcm(ahandler); struct async_private_data *data = snd_async_handler_get_callback_private(ahandler); signed short *samples = data->samples; snd_pcm_channel_area_t *areas = data->areas; snd_pcm_sframes_t avail; int err; avail = snd_pcm_avail_update(handle); while (avail >= period_size) { generate_sine(areas, 0, period_size, &data->phase); err = snd_pcm_writei(handle, samples, period_size); if (err < 0) { printf("Write error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } if (err != period_size) { printf("Write error: written %i expected %li\n", err, period_size); exit(EXIT_FAILURE); } avail = snd_pcm_avail_update(handle); } } static int async_loop(snd_pcm_t *handle, signed short *samples, snd_pcm_channel_area_t *areas) { struct async_private_data data; snd_async_handler_t *ahandler; int err, count; data.samples = samples; data.areas = areas; data.phase = 0; err = snd_async_add_pcm_handler(&ahandler, handle, async_callback, &data); if (err < 0) { printf("Unable to register async handler\n"); exit(EXIT_FAILURE); } for (count = 0; count < 2; count++) { generate_sine(areas, 0, period_size, &data.phase); err = snd_pcm_writei(handle, samples, period_size); if (err < 0) { printf("Initial write error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } if (err != period_size) { printf("Initial write error: written %i expected %li\n", err, period_size); exit(EXIT_FAILURE); } } if (snd_pcm_state(handle) == SND_PCM_STATE_PREPARED) { err = snd_pcm_start(handle); if (err < 0) { printf("Start error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } } /* because all other work is done in the signal handler, suspend the process */ while (1) { sleep(1); } } /* * Transfer method - asynchronous notification + direct write */ static void async_direct_callback(snd_async_handler_t *ahandler) { snd_pcm_t *handle = snd_async_handler_get_pcm(ahandler); struct async_private_data *data = snd_async_handler_get_callback_private(ahandler); const snd_pcm_channel_area_t *my_areas; snd_pcm_uframes_t offset, frames, size; snd_pcm_sframes_t avail, commitres; snd_pcm_state_t state; int first = 0, err; while (1) { state = snd_pcm_state(handle); if (state == SND_PCM_STATE_XRUN) { err = xrun_recovery(handle, -EPIPE); if (err < 0) { printf("XRUN recovery failed: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } first = 1; } else if (state == SND_PCM_STATE_SUSPENDED) { err = xrun_recovery(handle, -ESTRPIPE); if (err < 0) { printf("SUSPEND recovery failed: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } } avail = snd_pcm_avail_update(handle); if (avail < 0) { err = xrun_recovery(handle, avail); if (err < 0) { printf("avail update failed: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } first = 1; continue; } if (avail < period_size) { if (first) { first = 0; err = snd_pcm_start(handle); if (err < 0) { printf("Start error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } } else { break; } continue; } size = period_size; while (size > 0) { frames = size; err = snd_pcm_mmap_begin(handle, &my_areas, &offset, &frames); if (err < 0) { if ((err = xrun_recovery(handle, err)) < 0) { printf("MMAP begin avail error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } first = 1; } generate_sine(my_areas, offset, frames, &data->phase); commitres = snd_pcm_mmap_commit(handle, offset, frames); if (commitres < 0 || (snd_pcm_uframes_t)commitres != frames) { if ((err = xrun_recovery(handle, commitres >= 0 ? -EPIPE : commitres)) < 0) { printf("MMAP commit error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } first = 1; } size -= frames; } } } static int async_direct_loop(snd_pcm_t *handle, signed short *samples ATTRIBUTE_UNUSED, snd_pcm_channel_area_t *areas ATTRIBUTE_UNUSED) { struct async_private_data data; snd_async_handler_t *ahandler; const snd_pcm_channel_area_t *my_areas; snd_pcm_uframes_t offset, frames, size; snd_pcm_sframes_t commitres; int err, count; data.samples = NULL; /* we do not require the global sample area for direct write */ data.areas = NULL; /* we do not require the global areas for direct write */ data.phase = 0; err = snd_async_add_pcm_handler(&ahandler, handle, async_direct_callback, &data); if (err < 0) { printf("Unable to register async handler\n"); exit(EXIT_FAILURE); } for (count = 0; count < 2; count++) { size = period_size; while (size > 0) { frames = size; err = snd_pcm_mmap_begin(handle, &my_areas, &offset, &frames); if (err < 0) { if ((err = xrun_recovery(handle, err)) < 0) { printf("MMAP begin avail error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } } generate_sine(my_areas, offset, frames, &data.phase); commitres = snd_pcm_mmap_commit(handle, offset, frames); if (commitres < 0 || (snd_pcm_uframes_t)commitres != frames) { if ((err = xrun_recovery(handle, commitres >= 0 ? -EPIPE : commitres)) < 0) { printf("MMAP commit error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } } size -= frames; } } err = snd_pcm_start(handle); if (err < 0) { printf("Start error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } /* because all other work is done in the signal handler, suspend the process */ while (1) { sleep(1); } } /* * Transfer method - direct write only */ static int direct_loop(snd_pcm_t *handle, signed short *samples ATTRIBUTE_UNUSED, snd_pcm_channel_area_t *areas ATTRIBUTE_UNUSED) { double phase = 0; const snd_pcm_channel_area_t *my_areas; snd_pcm_uframes_t offset, frames, size; snd_pcm_sframes_t avail, commitres; snd_pcm_state_t state; int err, first = 1; while (1) { state = snd_pcm_state(handle); if (state == SND_PCM_STATE_XRUN) { err = xrun_recovery(handle, -EPIPE); if (err < 0) { printf("XRUN recovery failed: %s\n", snd_strerror(err)); return err; } first = 1; } else if (state == SND_PCM_STATE_SUSPENDED) { err = xrun_recovery(handle, -ESTRPIPE); if (err < 0) { printf("SUSPEND recovery failed: %s\n", snd_strerror(err)); return err; } } avail = snd_pcm_avail_update(handle); if (avail < 0) { err = xrun_recovery(handle, avail); if (err < 0) { printf("avail update failed: %s\n", snd_strerror(err)); return err; } first = 1; continue; } if (avail < period_size) { if (first) { first = 0; err = snd_pcm_start(handle); if (err < 0) { printf("Start error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } } else { err = snd_pcm_wait(handle, -1); if (err < 0) { if ((err = xrun_recovery(handle, err)) < 0) { printf("snd_pcm_wait error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } first = 1; } } continue; } size = period_size; while (size > 0) { frames = size; err = snd_pcm_mmap_begin(handle, &my_areas, &offset, &frames); if (err < 0) { if ((err = xrun_recovery(handle, err)) < 0) { printf("MMAP begin avail error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } first = 1; } generate_sine(my_areas, offset, frames, &phase); commitres = snd_pcm_mmap_commit(handle, offset, frames); if (commitres < 0 || (snd_pcm_uframes_t)commitres != frames) { if ((err = xrun_recovery(handle, commitres >= 0 ? -EPIPE : commitres)) < 0) { printf("MMAP commit error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } first = 1; } size -= frames; } } } /* * Transfer method - direct write only using mmap_write functions */ static int direct_write_loop(snd_pcm_t *handle, signed short *samples, snd_pcm_channel_area_t *areas) { double phase = 0; signed short *ptr; int err, cptr; while (1) { generate_sine(areas, 0, period_size, &phase); ptr = samples; cptr = period_size; while (cptr > 0) { err = snd_pcm_mmap_writei(handle, ptr, cptr); if (err == -EAGAIN) continue; if (err < 0) { if (xrun_recovery(handle, err) < 0) { printf("Write error: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } break; /* skip one period */ } ptr += err * channels; cptr -= err; } } } struct transfer_method { const char *name; snd_pcm_access_t access; int (*transfer_loop)(snd_pcm_t *handle, signed short *samples, snd_pcm_channel_area_t *areas); }; static struct transfer_method transfer_methods[] = { { "write", SND_PCM_ACCESS_RW_INTERLEAVED, write_loop }, { "write_and_poll", SND_PCM_ACCESS_RW_INTERLEAVED, write_and_poll_loop }, { "async", SND_PCM_ACCESS_RW_INTERLEAVED, async_loop }, { "async_direct", SND_PCM_ACCESS_MMAP_INTERLEAVED, async_direct_loop }, { "direct_interleaved", SND_PCM_ACCESS_MMAP_INTERLEAVED, direct_loop }, { "direct_noninterleaved", SND_PCM_ACCESS_MMAP_NONINTERLEAVED, direct_loop }, { "direct_write", SND_PCM_ACCESS_MMAP_INTERLEAVED, direct_write_loop }, { NULL, SND_PCM_ACCESS_RW_INTERLEAVED, NULL } }; static void help(void) { int k; printf( "Usage: pcm [OPTION]... [FILE]...\n" "-h,--help help\n" "-D,--device playback device\n" "-r,--rate stream rate in Hz\n" "-c,--channels count of channels in stream\n" "-f,--frequency sine wave frequency in Hz\n" "-b,--buffer ring buffer size in us\n" "-p,--period period size in us\n" "-m,--method transfer method\n" "-o,--format sample format\n" "-v,--verbose show the PCM setup parameters\n" "-n,--noresample do not resample\n" "-e,--pevent enable poll event after each period\n" "\n"); printf("Recognized sample formats are:"); for (k = 0; k < SND_PCM_FORMAT_LAST; ++k) { const char *s = snd_pcm_format_name(k); if (s) printf(" %s", s); } printf("\n"); printf("Recognized transfer methods are:"); for (k = 0; transfer_methods[k].name; k++) printf(" %s", transfer_methods[k].name); printf("\n"); } int main (int argc, char *argv[]) { struct option long_option[] = { {"help", 0, NULL, 'h'}, {"device", 1, NULL, 'd'}, {"rate", 1, NULL, 'r'}, {"channels", 1, NULL, 'c'}, {"frequency", 1, NULL, 'f'}, {"buffer", 1, NULL, 'b'}, {"period", 1, NULL, 'p'}, {"method", 1, NULL, 'm'}, {"format", 1, NULL, 'o'}, {"verbose", 1, NULL, 'v'}, {"noresample", 1, NULL, 'n'}, {"pevent", 1, NULL, 'e'}, {NULL, 0, NULL, 0}, }; snd_pcm_t *handle; //ponteiro para pcm handle int err, morehelp; snd_pcm_hw_params_t *hwparams; //parametros de hardware snd_pcm_sw_params_t *swparams; //parametros de software int method = 0; signed short *samples; unsigned int chn; snd_pcm_channel_area_t *areas; //snd_pcm_hw_params_alloca(&hwparams); //snd_pcm_sw_params_alloca(&swparams); morehelp = 0; while (1) { int c; if ((c = getopt_long(argc, argv, "hd:r:c:f:b:p:m:o:vne", long_option, NULL)) < 0) break; switch(c) { case 'h': morehelp++; break; case 'd': device = strdup(optarg); break; case 'r': rate = atoi(optarg); rate = rate < 4000 ? 4000 : rate; rate = rate > 196000 ? 19600 : rate; break; case 'c': channels = atoi(optarg); channels = channels < 1 ? 1 : channels; channels = channels > 1024 ? 1024 : channels; break; case 'f': freq = atoi(optarg); freq = freq < 50 ? 50 : freq; freq = freq > 5000 ? 5000 : freq; break; case 'b': buffer_time = atoi(optarg); buffer_time = buffer_time < 1000 ? 1000 : buffer_time; buffer_time = buffer_time > 1000000 ? 1000000 : buffer_time; break; case 'p': period_time = atoi(optarg); period_time = period_time < 1000 ? 1000 : period_time; period_time = period_time > 1000000 ? 1000000 : period_time; break; case 'm': for (method = 0; transfer_methods[method].name; method++) if (!strcasecmp(transfer_methods[method].name, optarg)) break; if (transfer_methods[method].name == NULL) method = 0; break; case 'o': for (format = 0; format < SND_PCM_FORMAT_LAST; format++) { const char *format_name = snd_pcm_format_name(format); if (format_name) if (!strcasecmp(format_name, optarg)) break; } if (format == SND_PCM_FORMAT_LAST) format = SND_PCM_FORMAT_S16; break; case 'v': verbose = 1; break; case 'n': resample = 0; break; case 'e': period_event = 1; break; }//switch }//while if(morehelp) { help(); return 0; } //cria um novo output objeto err = snd_output_stdio_attach(&output, &stdout, 0); if (err < 0) { printf("Output failed: %s\n", snd_strerror(err)); return 0; } printf("Playback device is %s\n", device); printf("Stream parameters are %iHz, %s, %i channels\n", rate, snd_pcm_format_name(format), channels); printf("sine wave rate is %i %.4fHz\n", freq); printf("Using transfer method: %s\n", transfer_methods[method].name); if ((err = snd_pcm_open(&handle, device, SND_PCM_STREAM_PLAYBACK, 0)) < 0) { printf("Playback open error: %s\n", snd_strerror(err)); return 0; } if ((err = set_hwparams(handle, hwparams, transfer_methods[method].access)) < 0) { printf("Setting of hwparams failed: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } if ((err = set_swparams(handle, swparams)) < 0) { printf("Setting of swparams failed: %s\n", snd_strerror(err)); exit(EXIT_FAILURE); } if (verbose > 0) snd_pcm_dump(handle, output); samples = malloc((period_size * channels * snd_pcm_format_physical_width(format)) / 8); if (samples == NULL) { printf("No enough memory\n"); exit(EXIT_FAILURE); } areas = calloc(channels, sizeof(snd_pcm_channel_area_t)); if (areas == NULL) { printf("No enough memory\n"); exit(EXIT_FAILURE); } for (chn = 0; chn < channels; chn++) { areas[chn].addr = samples; areas[chn].first = chn * snd_pcm_format_physical_width(format); areas[chn].step = channels * snd_pcm_format_physical_width(format); } err = transfer_methods[method].transfer_loop(handle, samples, areas); if (err < 0) printf("Transfer failed: %s\n", snd_strerror(err)); free(areas); free(samples); snd_pcm_close(handle); return 0; } Thanks in advance; -- Tks! ------------------- Guilherme Longo Dept. Eng. da Computação Unaerp Linux User - #484927 *Before Asking http://www.istf.com.br/?page=perguntas !- I'd rather die on my feet than live on my knees -!

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