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C++ snd_pcm_prepare函数代码示例

原作者: [db:作者] 来自: [db:来源] 收藏 邀请

本文整理汇总了C++中snd_pcm_prepare函数的典型用法代码示例。如果您正苦于以下问题:C++ snd_pcm_prepare函数的具体用法?C++ snd_pcm_prepare怎么用?C++ snd_pcm_prepare使用的例子?那么恭喜您, 这里精选的函数代码示例或许可以为您提供帮助。



在下文中一共展示了snd_pcm_prepare函数的20个代码示例,这些例子默认根据受欢迎程度排序。您可以为喜欢或者感觉有用的代码点赞,您的评价将有助于我们的系统推荐出更棒的C++代码示例。

示例1: an_configure_capture_card


//.........这里部分代码省略.........

    // Set the cropping, ignore any errors
    cropcap.type = V4L2_BUF_TYPE_VIDEO_CAPTURE;

    if (ioctl(m_i_fd, VIDIOC_CROPCAP, &cropcap)==0) {
            crop.type = V4L2_BUF_TYPE_VIDEO_CAPTURE;
            crop.c    = cropcap.defrect; /* reset to default */

            ioctl(m_i_fd, VIDIOC_S_CROP, &crop);
    }
    //
    // Analogue Video capture
    //
    pix_fmt = V4L2_PIX_FMT_YUV420;
    m_sws   = NULL;

    fmt.type                = V4L2_BUF_TYPE_VIDEO_CAPTURE;
    fmt.fmt.pix.width       = m_width;
    fmt.fmt.pix.height      = m_height;
    fmt.fmt.pix.pixelformat = pix_fmt;
    fmt.fmt.pix.field       = V4L2_FIELD_INTERLACED;

    if (ioctl(m_i_fd, VIDIOC_S_FMT, &fmt) == 0)
    {
        an_set_image_size( AV_PIX_FMT_YUV420P );
    }
    else
    {
        pix_fmt = V4L2_PIX_FMT_YUYV; // capture format
        fmt.type                = V4L2_BUF_TYPE_VIDEO_CAPTURE;
        fmt.fmt.pix.width       = m_width;
        fmt.fmt.pix.height      = m_height;
        fmt.fmt.pix.pixelformat = pix_fmt;
        fmt.fmt.pix.field       = V4L2_FIELD_INTERLACED;
        if (ioctl(m_i_fd, VIDIOC_S_FMT, &fmt) == 0)
        {
            // Format conversion will be required
            m_sws = sws_getContext( m_width, m_height, AV_PIX_FMT_YUYV422,
                                    m_width, m_height, AV_PIX_FMT_YUV420P,
                                    SWS_BICUBIC, NULL,NULL, NULL);
            an_set_image_size( AV_PIX_FMT_YUYV422 );
        }
        else
        {
            logger("CAP ANALOGUE FORMAT NOT SUPPORTED");
        }
    }

    if(ioctl(m_i_fd, VIDIOC_G_FMT, &fmt)<0 )
        logger("can't get format");

    //	input = V4L2_INPUT_TYPE_CAMERA;
    input = info.video_capture_device_input;

    if( ioctl( m_i_fd, VIDIOC_S_INPUT, &input) < 0 )
    {
        loggerf("CAP Error VIDIOC_S_INPUT %d",input);
    }

/*
    v4l2_streamparm parm;
    memset(&parm,0,sizeof(v4l2_streamparm));
    parm.type = V4L2_BUF_TYPE_VIDEO_CAPTURE;
    parm.parm.capture.capturemode = 0;
    parm.parm.capture.readbuffers = 0;

    if( ioctl( m_i_fd, VIDIOC_S_PARM, &parm) < 0 )
    {
        loggerf("CAP Error VIDIOC_S_PARM");
    }
*/
    info.video_bitrate = calculate_video_bitrate();

    //
    // Analogue sound capture
    //
    snd_pcm_hw_params_t *hw_params;

    if(snd_pcm_open(&m_audio_handle, "pulse", SND_PCM_STREAM_CAPTURE, 0)< 0 )
    {
        loggerf("Unable to open sound device");
        return;
    }
    unsigned int rate = 48000;
    int r;
    r = snd_pcm_hw_params_malloc(&hw_params);
    r = snd_pcm_hw_params_any(m_audio_handle, hw_params);
    r = snd_pcm_hw_params_set_access(m_audio_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
    r = snd_pcm_hw_params_set_format(m_audio_handle, hw_params, SND_PCM_FORMAT_S16_LE);
    r = snd_pcm_hw_params_set_rate_near(m_audio_handle, hw_params, &rate, 0);
    r = snd_pcm_hw_params_set_channels(m_audio_handle, hw_params, 2);
    r = snd_pcm_hw_params(m_audio_handle, hw_params);
    snd_pcm_hw_params_free(hw_params);
    r = snd_pcm_prepare(m_audio_handle);

    an_init_codecs();
    m_capturing = true;
    an_setup_video_capturing( m_i_fd );
    an_setup_audio_capturing();
}
开发者ID:Elitalan,项目名称:datvexpress_gui,代码行数:101,代码来源:an_capture.cpp


示例2: alsa_set_format

static RD_BOOL
alsa_set_format(snd_pcm_t * pcm, RD_WAVEFORMATEX * pwfx)
{
	snd_pcm_hw_params_t *hwparams = NULL;
	int err;
	unsigned int buffertime;
	short samplewidth;
	int audiochannels;
	unsigned int rate;

	samplewidth = pwfx->wBitsPerSample / 8;

	if ((err = snd_pcm_hw_params_malloc(&hwparams)) < 0)
	{
		error("snd_pcm_hw_params_malloc: %s\n", snd_strerror(err));
		return False;
	}

	if ((err = snd_pcm_hw_params_any(pcm, hwparams)) < 0)
	{
		error("snd_pcm_hw_params_any: %s\n", snd_strerror(err));
		return False;
	}

	if ((err = snd_pcm_hw_params_set_access(pcm, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0)
	{
		error("snd_pcm_hw_params_set_access: %s\n", snd_strerror(err));
		return False;
	}

	if (pwfx->wBitsPerSample == 16)
	{
		if ((err = snd_pcm_hw_params_set_format(pcm, hwparams, SND_PCM_FORMAT_S16_LE)) < 0)
		{
			error("snd_pcm_hw_params_set_format: %s\n", snd_strerror(err));
			return False;
		}
	}
	else
	{
		if ((err = snd_pcm_hw_params_set_format(pcm, hwparams, SND_PCM_FORMAT_S8)) < 0)
		{
			error("snd_pcm_hw_params_set_format: %s\n", snd_strerror(err));
			return False;
		}
	}

#if 0
	if ((err = snd_pcm_hw_params_set_rate_resample(pcm, hwparams, 1)) < 0)
	{
		error("snd_pcm_hw_params_set_rate_resample: %s\n", snd_strerror(err));
		return False;
	}
#endif

	rate = pwfx->nSamplesPerSec;
	if ((err = snd_pcm_hw_params_set_rate_near(pcm, hwparams, &rate, 0)) < 0)
	{
		error("snd_pcm_hw_params_set_rate_near: %s\n", snd_strerror(err));
		return False;
	}

	audiochannels = pwfx->nChannels;
	if ((err = snd_pcm_hw_params_set_channels(pcm, hwparams, pwfx->nChannels)) < 0)
	{
		error("snd_pcm_hw_params_set_channels: %s\n", snd_strerror(err));
		return False;
	}


	buffertime = 500000;	/* microseconds */
	if ((err = snd_pcm_hw_params_set_buffer_time_near(pcm, hwparams, &buffertime, 0)) < 0)
	{
		error("snd_pcm_hw_params_set_buffer_time_near: %s\n", snd_strerror(err));
		return False;
	}

	if ((err = snd_pcm_hw_params(pcm, hwparams)) < 0)
	{
		error("snd_pcm_hw_params: %s\n", snd_strerror(err));
		return False;
	}

	snd_pcm_hw_params_free(hwparams);

	if ((err = snd_pcm_prepare(pcm)) < 0)
	{
		error("snd_pcm_prepare: %s\n", snd_strerror(err));
		return False;
	}

	reopened = True;

	return True;
}
开发者ID:AmesianX,项目名称:rdesktop-fuzzer,代码行数:95,代码来源:rdpsnd_alsa.c


示例3: main

int main()
{
	int fp;
  unsigned int pcm, tmp, dir;
  int buff_size;
	
  long loops;
  int rc;
  int size;
  snd_pcm_t *handle;
  snd_pcm_hw_params_t *params;
  unsigned int val;
  
  snd_pcm_uframes_t frames;
  char *buff;
  int rate, channels, seconds;

  /* Open PCM device for recording (capture). */
  rc = snd_pcm_open(&handle, "default", SND_PCM_STREAM_CAPTURE, 0);

  if (rc < 0) 
  {
    fprintf(stderr,"unable to open pcm device: %s\n",   snd_strerror(rc));
    exit(1);
  }

  /* Allocate a hardware parameters object. */
  snd_pcm_hw_params_alloca(&params);

  /* Fill it in with default values. */
  snd_pcm_hw_params_any(handle, params);

  /* Set the desired hardware parameters. */

  /* Interleaved mode */
  snd_pcm_hw_params_set_access(handle, params,  SND_PCM_ACCESS_RW_INTERLEAVED);

  /* Signed 16-bit little-endian format */
  snd_pcm_hw_params_set_format(handle, params,SND_PCM_FORMAT_U8);

  /* Two channels (stereo) */
  snd_pcm_hw_params_set_channels(handle, params, 1);

  /* 44100 bits/second sampling rate (CD quality) */
  val = 8000;
  
  snd_pcm_hw_params_set_rate_near(handle, params,  &val, &dir);

  /* Set period size to 32 frames. */
  frames = 32;
  snd_pcm_hw_params_set_period_size_near(handle,params, &frames, &dir);

  /* Write the parameters to the driver */
  rc = snd_pcm_hw_params(handle, params);
  if (rc < 0) 
  {
    fprintf(stderr,"unable to set hw parameters: %s\n", snd_strerror(rc));
    exit(1);
  }

  /* Use a buff large enough to hold one period */

  snd_pcm_hw_params_get_period_size(params,  &frames, &dir);
  size = frames * 4; /* 2 bytes/sample, 2 channels */
  buff = (char *) malloc(size);

  /* We want to loop for 5 seconds */

  snd_pcm_hw_params_get_period_time(params, &val, &dir);
  loops = 50000000 / val;

	fp=open("222.wav",O_WRONLY);
  while (loops > 0) 
  {
		loops--;
		bzero(buff,sizeof(buff));
		rc = snd_pcm_readi(handle, buff, frames);
		write(fp,buff,128);
		if (rc == -EPIPE) 
		{
		  /* EPIPE means overrun */
		  fprintf(stderr, "overrun occurred\n");
		  snd_pcm_prepare(handle);
		} else if (rc < 0) 
		{
		  fprintf(stderr,"error from read: %s\n",  snd_strerror(rc));
		} else if (rc != (int)frames) 
		{
		  fprintf(stderr, "short read, read %d frames\n", rc);
		}
		
		/*rc = write(1, buff, size);
		if (rc != size)
		  fprintf(stderr,"short write: wrote %d bytes\n", rc);
	*/
  }

  snd_pcm_drain(handle);
  snd_pcm_close(handle);
  
//.........这里部分代码省略.........
开发者ID:Adityaypatil,项目名称:my-codes,代码行数:101,代码来源:arecord.c


示例4: main

// alsa	      
	main (int argc, char *argv[])
	{
		int i;
		int err;
		short buf[128];
		snd_pcm_t *playback_handle;
		snd_pcm_hw_params_t *hw_params;
	
		if ((err = snd_pcm_open (&playback_handle, argv[1], SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
			fprintf (stderr, "cannot open audio device %s (%s)\n", 
				 argv[1],
				 snd_strerror (err));
			exit (1);
		}
		   
		if ((err = snd_pcm_hw_params_malloc (&hw_params)) < 0) {
			fprintf (stderr, "cannot allocate hardware parameter structure (%s)\n",
				 snd_strerror (err));
			exit (1);
		}
				 
		if ((err = snd_pcm_hw_params_any (playback_handle, hw_params)) < 0) {
			fprintf (stderr, "cannot initialize hardware parameter structure (%s)\n",
				 snd_strerror (err));
			exit (1);
		}
	
		if ((err = snd_pcm_hw_params_set_access (playback_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
			fprintf (stderr, "cannot set access type (%s)\n",
				 snd_strerror (err));
			exit (1);
		}
	
		if ((err = snd_pcm_hw_params_set_format (playback_handle, hw_params, SND_PCM_FORMAT_S16_LE)) < 0) {
			fprintf (stderr, "cannot set sample format (%s)\n",
				 snd_strerror (err));
			exit (1);
		}
	
		if ((err = snd_pcm_hw_params_set_rate_near (playback_handle, hw_params, 44100, 0)) < 0) {
			fprintf (stderr, "cannot set sample rate (%s)\n",
				 snd_strerror (err));
			exit (1);
		}
	
		if ((err = snd_pcm_hw_params_set_channels (playback_handle, hw_params, 2)) < 0) {
			fprintf (stderr, "cannot set channel count (%s)\n",
				 snd_strerror (err));
			exit (1);
		}
	
		if ((err = snd_pcm_hw_params (playback_handle, hw_params)) < 0) {
			fprintf (stderr, "cannot set parameters (%s)\n",
				 snd_strerror (err));
			exit (1);
		}
	
		snd_pcm_hw_params_free (hw_params);
	
		if ((err = snd_pcm_prepare (playback_handle)) < 0) {
			fprintf (stderr, "cannot prepare audio interface for use (%s)\n",
				 snd_strerror (err));
			exit (1);
		}
	
		for (i = 0; i < 10; ++i) {
			if ((err = snd_pcm_writei (playback_handle, buf, 128)) != 128) {
				fprintf (stderr, "write to audio interface failed (%s)\n",
					 snd_strerror (err));
				exit (1);
			}
		}
	
		snd_pcm_close (playback_handle);
		exit (0);
	}
开发者ID:1090310214,项目名称:sound-card,代码行数:77,代码来源:alsa练习1.c


示例5: sound_thread

static void* sound_thread(void* context)
{
    struct SPU* c = (struct SPU*)context;
    int div;
    snd_pcm_t* snd;
    snd_pcm_hw_params_t* hwp;
    int rate = c->sampling;
    int periods = 3;
    snd_pcm_hw_params_alloca(&hwp);
    if (snd_pcm_open(&snd, "default", SND_PCM_STREAM_PLAYBACK, 0) < 0) {
        return NULL;
    }
    if (snd_pcm_hw_params_any(snd, hwp) < 0) {
        return NULL;
    }
    div = c->bit / 8;
    switch (c->bit) {
        case 8:
            if (snd_pcm_hw_params_set_format(snd, hwp, SND_PCM_FORMAT_U8) < 0) {
                return NULL;
            }
            break;
        case 16:
            if (snd_pcm_hw_params_set_format(snd, hwp, SND_PCM_FORMAT_S16_LE) < 0) {
                return NULL;
            }
            break;
        case 24:
            if (snd_pcm_hw_params_set_format(snd, hwp, SND_PCM_FORMAT_S24_LE) < 0) {
                return NULL;
            }
            break;
        case 32:
            if (snd_pcm_hw_params_set_format(snd, hwp, SND_PCM_FORMAT_S32_LE) < 0) {
                return NULL;
            }
            break;
        default:
            return NULL;
    }
    if (snd_pcm_hw_params_set_rate_near(snd, hwp, &rate, 0) < 0) {
        return NULL;
    }
    if (rate != SAMPLE_RATE) {
        return NULL;
    }
    if (snd_pcm_hw_params_set_channels(snd, hwp, c->ch) < 0) {
        return NULL;
    }
    if (snd_pcm_hw_params_set_periods(snd, hwp, periods, 0) < 0) {
        return NULL;
    }
    if (snd_pcm_hw_params_set_buffer_size(snd, hwp, (periods * c->size) / 4) < 0) {
        return NULL;
    }
    if (snd_pcm_hw_params(snd, hwp) < 0) {
        return NULL;
    }
    while (c->alive) {
        c->callback(c->buffer, c->size);
        while (c->alive) {
            if (snd_pcm_writei(snd, c->buffer, c->size / div) < 0) {
                snd_pcm_prepare(snd);
            } else {
                break;
            }
        }
    }
    snd_pcm_drain(snd);
    snd_pcm_close(snd);
    return NULL;
}
开发者ID:suzukiplan,项目名称:vgs-spu,代码行数:72,代码来源:vgsspu_alsa.c


示例6: main

 main() {
  long loops;
  int rc;
  int size;
  snd_pcm_t *handle;
  snd_pcm_hw_params_t *params;
  unsigned int val;
  int dir;
  snd_pcm_uframes_t frames;
 unsigned char *buffer;
  int i=0;

/* socket setting */
   int sd;
   struct sockaddr_in s_addr;
   int n, n_send, status;

   int on = 1;


   sd = socket (AF_INET, SOCK_DGRAM, 0);

   bzero(&s_addr, sizeof(s_addr));
   s_addr.sin_family = AF_INET;
   s_addr.sin_addr.s_addr = inet_addr("192.168.42.255");
   s_addr.sin_port = htons(2007);

   if((status = setsockopt(sd, SOL_SOCKET, SO_BROADCAST, &on, sizeof(on))) != 0 )
   {
      printf("setsockopt error\n");
      exit(-1);
   }


  /* Open PCM device for recording (capture). */
  rc = snd_pcm_open(&handle, "plughw:1,0",
                    SND_PCM_STREAM_CAPTURE, 0);
  if (rc < 0) {
    fprintf(stderr,
            "unable to open pcm device: %s\n",
            snd_strerror(rc));
    exit(1);
  }

  /* Allocate a hardware parameters object. */
  snd_pcm_hw_params_alloca(&params);

  /* Fill it in with default values. */
  snd_pcm_hw_params_any(handle, params);

  /* Set the desired hardware parameters. */

  /* Interleaved mode */
  snd_pcm_hw_params_set_access(handle, params,
                      SND_PCM_ACCESS_RW_INTERLEAVED);

  /* Signed 16-bit little-endian format */
  snd_pcm_hw_params_set_format(handle, params,SND_PCM_FORMAT_U8);

  /* Two channels (stereo) */
  snd_pcm_hw_params_set_channels(handle, params, 1);

  /* 44100 bits/second sampling rate (CD quality) */
  val = 44100;
  snd_pcm_hw_params_set_rate_near(handle, params,
                                  &val, &dir);

  /* Set period size to 32 frames. */
  frames =4800;
  snd_pcm_hw_params_set_period_size_near(handle,
                              params, &frames, &dir);

  /* Write the parameters to the driver */
  rc = snd_pcm_hw_params(handle, params);
  if (rc < 0) {
    fprintf(stderr,
            "unable to set hw parameters: %s\n",
            snd_strerror(rc));
    exit(1);
  }

  /* Use a buffer large enough to hold one period */
  snd_pcm_hw_params_get_period_size(params,
                                      &frames, &dir);
  size = frames *1; /* 2 bytes/sample, 2 channels */

  buffer = (unsigned char*) malloc(size);
	
  /* We want to loop for 5 seconds */
  snd_pcm_hw_params_get_period_time(params,
                                         &val, &dir);
  while (1) {
    rc = snd_pcm_readi(handle, buffer, frames);
    if (rc == -EPIPE) {
      /* EPIPE means overrun */
      fprintf(stderr, "overrun occurred\n");
      snd_pcm_prepare(handle);
    } else if (rc < 0) {
      fprintf(stderr,
              "error from read: %s\n",
//.........这里部分代码省略.........
开发者ID:DegreeTeam,项目名称:HamProject,代码行数:101,代码来源:broadcast.c


示例7: play

void play(int sec, int freq, char *file)
{
	long loops;
	int rc;
	int size;
	snd_pcm_t *handle;
	snd_pcm_hw_params_t *params;
	unsigned int val;
	int dir;
	snd_pcm_uframes_t frames;
	char *buffer;
	int fd;

	/* Check ranges */
	if (sec < 1 || sec > 4000) {
		printf("WARNING: Incorrect time to play range [1,4000] s\n");
		printf("\tSetting time to play to 5s...\n");
		sec = 5;
	}
	if (freq < 1000 || freq > 100000) {
		printf("ERROR: Incorrect frequency range [1000,100000] Hz\n");
		printf("\tSetting frequency to 44.1 kHz...\n");
		freq = 44100;
	}

	/* Open file */
	fd = open(file, O_RDONLY);
	if (fd < 0) {
		/* There was an error opening the file */
		printf("ERROR: Couldn't open file to play\n");
		printf("\tPlease make sure file exists\n");
	} else {
		/* Print that the file is playing with its parameters */
		printf("Playing file %s for %d seconds", file, sec);
		printf(" and frequency %d...\n", freq);

		/* Open PCM device for playback */
		rc = snd_pcm_open(&handle, "default", SND_PCM_STREAM_PLAYBACK,
			0);
		if (rc < 0) {
			fprintf(stderr, "unable to open pcm device: %s\n",
				snd_strerror(rc));
			exit(1);
		}

		/* Allocate a hardware parameters object */
		snd_pcm_hw_params_alloca(&params);
		/* Fill it in with default values */
		snd_pcm_hw_params_any(handle, params);

		/* Set hardware parameters */

		/* Interleaved mode */
		snd_pcm_hw_params_set_access(handle, params,
		SND_PCM_ACCESS_RW_INTERLEAVED);
		/* Signed 16-bit little-endian format */
		snd_pcm_hw_params_set_format(handle, params,
			SND_PCM_FORMAT_S16_LE);
		/* Two channels (stereo) */
		snd_pcm_hw_params_set_channels(handle, params, 2);
		/* freq bits/second sampling rate (CD quality) */
		val = freq;
		snd_pcm_hw_params_set_rate_near(handle, params, &val, &dir);
		/* Set period size to 32 frames */
		frames = 32;
		snd_pcm_hw_params_set_period_size_near(handle, params,
			&frames, &dir);
		/* Write the parameters to the driver */
		rc = snd_pcm_hw_params(handle, params);
		if (rc < 0) {
			fprintf(stderr, "unable to set hw parameters: %s\n",
				snd_strerror(rc));
			exit(1);
		}

		/* Use a buffer large enough to hold one period */
		snd_pcm_hw_params_get_period_size(params, &frames, &dir);
		size = frames * 4; /* 2 bytes/sample, 2 channels */
		buffer = (char *) malloc(size);
		/* We want to loop for sec seconds */
		snd_pcm_hw_params_get_period_time(params, &val, &dir);
		/* sec seconds in microseconds divided by period time */
		loops = sec*1000000 / val;

		while (loops > 0) {
			loops--;
			rc = read(fd, buffer, size);
			if (rc == 0) {
				fprintf(stderr, "end of file on input\n");
				break;
			} else if (rc != size) {
				fprintf(stderr, "short read: read %d bytes\n",
					rc);
			}
			rc = snd_pcm_writei(handle, buffer, frames);
			if (rc == -EPIPE) {
				/* EPIPE means underrun */
				fprintf(stderr, "underrun occurred\n");
				snd_pcm_prepare(handle);
			} else if (rc < 0) {
//.........这里部分代码省略.........
开发者ID:ivan-gomez,项目名称:drivers-gdl,代码行数:101,代码来源:player.c


示例8: AlsaOutput_Write

/*----------------------------------------------------------------------
|    AlsaOutput_Write
+---------------------------------------------------------------------*/
static BLT_Result
AlsaOutput_Write(AlsaOutput* self, void* buffer, BLT_Size size)
{
    int          watchdog = 5;
    int          io_result;
    unsigned int sample_count;
    unsigned int sample_size;
    BLT_Result   result;

    /* ensure that the device is prepared */
    result = AlsaOutput_Prepare(self);
    if (BLT_FAILED(result)) return result;

    /* compute the number of samples */
    sample_size  = self->media_type.channel_count*self->media_type.bits_per_sample/8;
    sample_count = size / sample_size;
                           
    /* write samples to the device and handle underruns */       
    do {
        while (sample_count) {
            io_result = snd_pcm_writei(self->device_handle, 
                                       buffer, sample_count);
            if (io_result > 0) {
                buffer = (void*)((char*)buffer + io_result*sample_size);
                if ((unsigned int)io_result <= sample_count) {
                    sample_count -= io_result;
                } else {
                    /* strange, snd_pcm_writei returned more than we wrote */
                    sample_count = 0;
                }
            } else {
                break;
            }
        }        
        if (sample_count == 0) return BLT_SUCCESS;

        /* we reach this point if the first write failed */
        if (io_result < 0) {
            snd_pcm_status_t* status;
            snd_pcm_state_t   state;
            snd_pcm_status_alloca_no_assert(&status);

            io_result = snd_pcm_status(self->device_handle, status);
            if (io_result != 0) {
                return BLT_FAILURE;
            }
            state = snd_pcm_status_get_state(status);
            if (state == SND_PCM_STATE_XRUN) {
                ATX_LOG_FINE("**** UNDERRUN *****");
            
                /* re-prepare the channel */
                io_result = snd_pcm_prepare(self->device_handle);
                if (io_result != 0) {
                    return BLT_FAILURE;
                }
            } else {
               ATX_LOG_WARNING_1("**** STATE = %d ****", state);
            }
        } else {
            ATX_LOG_WARNING_1("snd_pcm_writei() returned %d", io_result); 
        }
        
        ATX_LOG_FINE("**** RETRY *****");

    } while(watchdog--);

    ATX_LOG_SEVERE("**** THE WATCHDOG BIT US ****");
    return BLT_FAILURE;
}
开发者ID:axiomatic-systems,项目名称:BlueTune,代码行数:72,代码来源:BltAlsaOutput.c


示例9: pthread_getschedparam

void *audiothread(void *v) {
    int						policy;
    sched_param		sched;
    int						err;
    alsaaudio 		*dev;

    pthread_getschedparam(pthread_self(),&policy,&sched);
    sched.sched_priority++;//policy=SCHED_RR;
    pthread_setschedparam(pthread_self(),policy,&sched);
    dev=(alsaaudio*)v;

    unsigned int val;
    snd_pcm_t *fd;
    snd_pcm_uframes_t periodsize;
    snd_pcm_hw_params_t *hwparams;
    snd_pcm_hw_params_alloca(&hwparams);
    int output_rate;
    int channels;
    int fragment_size;
    int fragment_count;

    err=snd_pcm_open(&fd, strdup("default"), SND_PCM_STREAM_PLAYBACK, 0);
    if (err<0) return -1;

    fragment_size=LINUXFRAG;  //overall buffer size
    fragment_count=2; //2 - 16 fragment count - 2 minimum, the lower it is potentially the lower the latency

//configure device
    if (snd_pcm_hw_params_any(fd, hwparams) < 0) {
        //printf("linuxaudio failed at params any\n");
        return -1;
    }
    if (snd_pcm_hw_params_set_access(fd, hwparams,SND_PCM_ACCESS_RW_INTERLEAVED) < 0) {
        //printf("linuxaudio failed at set access\n");
        return -1;
    }

    if (snd_pcm_hw_params_set_format(fd, hwparams,SND_PCM_FORMAT_S16_LE) < 0) {
        //printf("linuxaudio failed at set format\n");
        return -1;
    }
    val = 44100;
    if (snd_pcm_hw_params_set_rate_near(fd, hwparams,&val, 0) < 0) {
        // Try 48KHZ too
        //printf("linuxaudio - %d HZ not available, trying 48000HZ\n", output_rate);
        val = 48000;
        if (snd_pcm_hw_params_set_rate_near(fd, hwparams,&val, 0) < 0) {
            //printf("linuxaudio failed at setting output rate (%d)\n", output_rate);
            return -1;
        }
        dev->mix->freq=val;
    }
    channels=2;
    if (snd_pcm_hw_params_set_channels(fd, hwparams, channels) < 0) {
        //printf("linuxaudio failed at set channels (%d)\n", channels);
        return -1;
    }
    periodsize = (fragment_size) / 4; // bytes -> frames for 16-bit,stereo - should be a minimum of 512
    if (snd_pcm_hw_params_set_period_size_near(fd, hwparams,&periodsize, 0) < 0) {
        //printf("linuxaudio failed at set period size (%d)\n", (int)periodsize);
        return -1;
    }
    val = fragment_count;
    if (snd_pcm_hw_params_set_periods_near(fd, hwparams,&val, 0) < 0) {
        //printf("linuxaudio failed at set periods (%d)\n", val);
        //should attempt a one by one period increase up to 16?
        return -1;
    }
    if (snd_pcm_hw_params(fd, hwparams) < 0) {
        //printf("linuxaudio failed at installing hw params\n");
        return -1;
    }
    //loop while playing sound
    dev->playing=1;
    while (dev->playing)
    {
        dev->mix->mix16(dev->buffer);
        if ((snd_pcm_writei (fd, dev->buffer,LINUXFRAG/2)) < 0) {	//Half buffer for two channels?
            //printf ("linuxaudio warning: buffer underrun occurred\n");
            if (snd_pcm_prepare(fd) < 0) {
                //printf ("linuxaudio failed at preparing pcm\n");
                dev->playing=0; //die gracefully
            }
        }
    }
    snd_pcm_drop(fd);
    snd_pcm_close (fd);
    return 0;
}
开发者ID:baatar,项目名称:blitzmax,代码行数:89,代码来源:alsadevice.cpp


示例10: alsa_open


//.........这里部分代码省略.........
		error ("Can't initialize hardware parameters structure: %s",
				snd_strerror(err));
		snd_pcm_hw_params_free (hw_params);
		return 0;
	}

	if ((err = snd_pcm_hw_params_set_access (handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
		error ("Can't set alsa access type: %s", snd_strerror(err));
		snd_pcm_hw_params_free (hw_params);
		return 0;
	}

	if ((err = snd_pcm_hw_params_set_format (handle, hw_params,
					params.format)) < 0) {
		error ("Can't set sample format: %s", snd_strerror(err));
		snd_pcm_hw_params_free (hw_params);
		return 0;
	}

	params.rate = sound_params->rate;
	if ((err = snd_pcm_hw_params_set_rate_near (handle, hw_params,
					&params.rate, 0)) < 0) {
		error ("Can't set sample rate: %s", snd_strerror(err));
		snd_pcm_hw_params_free (hw_params);
		return 0;
	}

	logit ("Set rate to %d", params.rate);
	
	if ((err = snd_pcm_hw_params_set_channels (handle, hw_params,
					sound_params->channels)) < 0) {
		error ("Can't set number of channels: %s", snd_strerror(err));
		snd_pcm_hw_params_free (hw_params);
		return 0;
	}

	if ((err = snd_pcm_hw_params_get_buffer_time_max(hw_params,
					&buffer_time, 0)) < 0) {
		error ("Can't get maximum buffer time: %s", snd_strerror(err));
		snd_pcm_hw_params_free (hw_params);
		return 0;
	}

	if (buffer_time > BUFFER_MAX_USEC)
		buffer_time = BUFFER_MAX_USEC;
	period_time = buffer_time / 4;

	if ((err = snd_pcm_hw_params_set_period_time_near(handle, hw_params, 
					&period_time, 0)) < 0) {
		error ("Can't set period time: %s", snd_strerror(err));
		snd_pcm_hw_params_free (hw_params);
		return 0;
	}

	if ((err = snd_pcm_hw_params_set_buffer_time_near(handle, hw_params, 
					&buffer_time, 0)) < 0) {
		error ("Can't set buffer time: %s", snd_strerror(err));
		snd_pcm_hw_params_free (hw_params);
		return 0;
	}

	if ((err = snd_pcm_hw_params (handle, hw_params)) < 0) {
		error ("Can't set audio parameters: %s", snd_strerror(err));
		snd_pcm_hw_params_free (hw_params);
		return 0;
	}

	snd_pcm_hw_params_get_period_size (hw_params, &chunk_frames, 0);
	snd_pcm_hw_params_get_buffer_size (hw_params, &buffer_frames);

	bytes_per_frame = sound_params->channels
		* sfmt_Bps(sound_params->fmt);

	logit ("Buffer time: %ldus", buffer_frames * bytes_per_frame);

	if (chunk_frames == buffer_frames) {
		error ("Can't use period equal to buffer size (%lu == %lu)",
				chunk_frames, buffer_frames);
		snd_pcm_hw_params_free (hw_params);
		return 0;
	}

	chunk_size = chunk_frames * bytes_per_frame;

	debug ("Chunk size: %d", chunk_size);
	
	snd_pcm_hw_params_free (hw_params);
	
	if ((err = snd_pcm_prepare(handle)) < 0) {
		error ("Can't prepare audio interface for use: %s",
				snd_strerror(err));
		return 0;
	}

	debug ("ALSA device initialized");
	
	params.channels = sound_params->channels;
	alsa_buf_fill = 0;
	return 1;
}
开发者ID:ecthiender,项目名称:mocp-git,代码行数:101,代码来源:alsa.c


示例11: ca_thread_func

static int ca_thread_func (void *arg)
{
    struct alsa_stream* stream = (struct alsa_stream*) arg;
    snd_pcm_t* pcm             = stream->ca_pcm;
    int size                   = stream->ca_buf_size;
    snd_pcm_uframes_t nframes  = stream->ca_frames;
    void* user_data            = stream->user_data;
    char* buf 		       = stream->ca_buf;
    pj_timestamp tstamp;
    int result;
    struct sched_param param;
    pthread_t* thid;

    thid = (pthread_t*) pj_thread_get_os_handle (pj_thread_this());
    param.sched_priority = sched_get_priority_max (SCHED_RR);
    PJ_LOG (5,(THIS_FILE, "ca_thread_func(%u): Set thread priority "
		          "for audio capture thread.",
		          (unsigned)syscall(SYS_gettid)));
    result = pthread_setschedparam (*thid, SCHED_RR, &param);
    if (result) {
	if (result == EPERM)
	    PJ_LOG (5,(THIS_FILE, "Unable to increase thread priority, "
				  "root access needed."));
	else
	    PJ_LOG (5,(THIS_FILE, "Unable to increase thread priority, "
				  "error: %d",
				  result));
    }

    pj_bzero (buf, size);
    tstamp.u64 = 0;

    TRACE_((THIS_FILE, "ca_thread_func(%u): Started",
	    (unsigned)syscall(SYS_gettid)));

    snd_pcm_prepare (pcm);

    while (!stream->quit) {
	pjmedia_frame frame;

	pj_bzero (buf, size);
	result = snd_pcm_readi (pcm, buf, nframes);
	if (result == -EPIPE) {
	    PJ_LOG (4,(THIS_FILE, "ca_thread_func: overrun!"));
	    snd_pcm_prepare (pcm);
	    continue;
	} else if (result < 0) {
	    PJ_LOG (4,(THIS_FILE, "ca_thread_func: error reading data!"));
	}
	if (stream->quit)
	    break;

	frame.type = PJMEDIA_FRAME_TYPE_AUDIO;
	frame.buf = (void*) buf;
	frame.size = size;
	frame.timestamp.u64 = tstamp.u64;
	frame.bit_info = 0;

	result = stream->ca_cb (user_data, &frame);
	if (result != PJ_SUCCESS || stream->quit)
	    break;

	tstamp.u64 += nframes;
    }
    snd_pcm_drain (pcm);
    TRACE_((THIS_FILE, "ca_thread_func: Stopped"));

    return PJ_SUCCESS;
}
开发者ID:0x0B501E7E,项目名称:pjproject,代码行数:69,代码来源:alsa_dev.c


示例12: main


//.........这里部分代码省略.........
         snd_pcm_subformat_name((snd_pcm_subformat_t)val),
         snd_pcm_subformat_description(
            (snd_pcm_subformat_t)val));

   snd_pcm_hw_params_get_channels(params, &val);
   printf("channels = %d\n", val);

   snd_pcm_hw_params_get_rate(params, &val, &dir);
   printf("rate = %d bps\n", val);

   snd_pcm_hw_params_get_period_time(params,
         &val, &dir);
   printf("period time = %d us\n", val);

   snd_pcm_hw_params_get_period_size(params,
         &frames, &dir);
   printf("period size = %d frames\n", (int)frames);

   snd_pcm_hw_params_get_buffer_time(params,
         &val, &dir);
   printf("buffer time = %d us\n", val);

   snd_pcm_hw_params_get_buffer_size(params,
         (snd_pcm_uframes_t *) &val);
   printf("buffer size = %d frames\n", val);

   snd_pcm_hw_params_get_periods(params, &val, &dir);
   printf("periods per buffer = %d frames\n", val);

   snd_pcm_hw_params_get_rate_numden(params,
         &val, &val2);
   printf("exact rate = %d/%d bps\n", val, val2);

   val = snd_pcm_hw_params_get_sbits(params);
   printf("significant bits = %d\n", val);

   val = snd_pcm_hw_params_is_batch(params);
   printf("is batch = %d\n", val);

   val = snd_pcm_hw_params_is_block_transfer(params);
   printf("is block transfer = %d\n", val);

   val = snd_pcm_hw_params_is_double(params);
   printf("is double = %d\n", val);

   val = snd_pcm_hw_params_is_half_duplex(params);
   printf("is half duplex = %d\n", val);

   val = snd_pcm_hw_params_is_joint_duplex(params);
   printf("is joint duplex = %d\n", val);

   val = snd_pcm_hw_params_can_overrange(params);
   printf("can overrange = %d\n", val);

   val = snd_pcm_hw_params_can_mmap_sample_resolution(params);
   printf("can mmap = %d\n", val);

   val = snd_pcm_hw_params_can_pause(params);
   printf("can pause = %d\n", val);

   val = snd_pcm_hw_params_can_resume(params);
   printf("can resume = %d\n", val);

   val = snd_pcm_hw_params_can_sync_start(params);
   printf("can sync start = %d\n", val);

   settings = open_pitch_tracker();

   while(1){
      rc = snd_pcm_readi(handle, buf, frames);
      if (rc == -EPIPE) {
         /* EPIPE means overrun */
         fprintf(stderr, "overrun occurred\n");
         snd_pcm_prepare(handle);
      } else if (rc < 0) {
         fprintf(stderr,
               "error from read: %s\n",
               snd_strerror(rc));
      } else if (rc != (int)frames) {
         fprintf(stderr, "short read, read %d frames\n", rc);
      }


      for( i = 0 ; i< BUFSIZE ; i++){
         d_buffer[i] = (double) buf[i];
      }
      pitch = compute_pitch(d_buffer, BUFSIZE, S16, settings ,ACCURACY);
      if( pitch != 0.0 )
         printf("frequency -> %f\n",pitch);

      memset(buf,0,BUFSIZE);
   }


   close_pitch_tracker(&settings);
   snd_pcm_drain(handle);
   snd_pcm_close(handle);

   return 0;
}
开发者ID:ichigo663,项目名称:WaveTrack,代码行数:101,代码来源:pitchTracker_Example.c


示例13: snd_pcm_drop

void WaveRecorder::stop()
{
    snd_pcm_drop(_pcm);
    snd_pcm_prepare(_pcm);
}
开发者ID:SPICEorg,项目名称:spice,代码行数:5,代码来源:record.cpp


示例14: GetChannelLayout

bool CAESinkALSA::Initialize(AEAudioFormat &format, std::string &device)
{
  m_initDevice = device;
  m_initFormat = format;

  /* if we are raw, correct the data format */
  if (AE_IS_RAW(format.m_dataFormat))
  {
    m_channelLayout     = GetChannelLayout(format);
    format.m_dataFormat = AE_FMT_S16NE;
    m_passthrough       = true;
  }
  else
  {
    m_channelLayout = GetChannelLayout(format);
    m_passthrough   = false;
  }
#if defined(HAS_AMLPLAYER) || defined(HAS_LIBAMCODEC)
  if (aml_present())
  {
    aml_set_audio_passthrough(m_passthrough);
    device = "default";
  }
#endif

  if (m_channelLayout.Count() == 0)
  {
    CLog::Log(LOGERROR, "CAESinkALSA::Initialize - Unable to open the requested channel layout");
    return false;
  }

  format.m_channelLayout = m_channelLayout;

  AEDeviceType devType = AEDeviceTypeFromName(device);

  std::string AESParams;
  /* digital interfaces should have AESx set, though in practice most
   * receivers don't care */
  if (m_passthrough || devType == AE_DEVTYPE_HDMI || devType == AE_DEVTYPE_IEC958)
    GetAESParams(format, AESParams);

  CLog::Log(LOGINFO, "CAESinkALSA::Initialize - Attempting to open device \"%s\"", device.c_str());

  /* get the sound config */
  snd_config_t *config;
  snd_config_copy(&config, snd_config);

  if (!OpenPCMDevice(device, AESParams, m_channelLayout.Count(), &m_pcm, config))
  {
    CLog::Log(LOGERROR, "CAESinkALSA::Initialize - failed to initialize device \"%s\"", device.c_str());
    snd_config_delete(config);
    return false;
  }

  /* get the actual device name that was used */
  device = snd_pcm_name(m_pcm);
  m_device = device;

  CLog::Log(LOGINFO, "CAESinkALSA::Initialize - Opened device \"%s\"", device.c_str());

  /* free the sound config */
  snd_config_delete(config);

  if (!InitializeHW(format) || !InitializeSW(format))
    return false;

  snd_pcm_nonblock(m_pcm, 1);
  snd_pcm_prepare (m_pcm);

  m_format              = format;
  m_formatSampleRateMul = 1.0 / (double)m_format.m_sampleRate;

  return true;
}
开发者ID:Ilia,项目名称:xbmc,代码行数:74,代码来源:AESinkALSA.cpp


示例15: main


//.........这里部分代码省略.........
		return -1;
	}
*/


/*
avail=snd_pcm_avail_update(pcm_handle);
printf("\nAvailable frames = %d \n\n",(int)avail);
struct timeval tvbefore,tvafter;
gettimeofday(&tvbefore,NULL);
*/
unsigned char *ptr[2];



int i;
//unsigned int steps=0;

for(i=0;i<2;i++)
 ptr[i] = (unsigned char*)areas[i].addr  + (areas[i].first/8) + offset*(areas[i].step/8);

printf("\nOffset of the first sample : %u \n",areas[0].first);
printf("\n\nOffset of the memory map is : %u \n",areas[0].step/8);
printf("\nAreas start address is : %u \n",areas[0].addr);
printf("\nNumber of frames ; %u \n",frames);

printf("\n\nPointer before ; %u \n",ptr[0]);


long pl;
for(pl=0;pl<100000;pl++);

if(ptr[1]==NULL)
printf("\nNull pointer 1  allocated\n");
if(ptr[0]==NULL)
printf("\nNull pointer 0 allocated \n");




int check2=1;

snd_pcm_sframes_t size2 =30;
while(1)
	{
		for(i=0;i<2;i++)
		{
		//ptr[i] = (unsigned char*)areas[i].addr + (areas[i].first/8) + offset*(areas[i].step/8);
		
		if(read(dest,ptr[i],120)!=0)
			{
				
				
					if(ptr[i]!=NULL)
					{
						 	
						if(pcmreturn = snd_pcm_mmap_writei(pcm_handle,ptr[i],size2)<0)
							{
								snd_pcm_prepare(pcm_handle);
								printf("\n<<<<<<<Buffer Underrun>>>>>>>>\n");
								break;
							}
					
			

						//	printf("\n%d\n",pcmreturn);
						ptr[i]+=0;
						printf("\n%lu\n",ptr[i]);


					}
											

			}
			else {
				check2=0;
				break;
				}
		}
		
		

		if(check2==0)
		break;
		

			
		
	//	l:break;

	}

	commitres = snd_pcm_mmap_commit(pcm_handle,offset,frames);
	 if(commitres<0)
	         {
	                 printf("\nFrames not committed to memory\n");
	         }

	
}
开发者ID:siddharthcs94,项目名称:test,代码行数:101,代码来源:alsatest.c


示例16: SetFormat


//.........这里部分代码省略.........
    if (err < 0)
    {
        if (errno != EBUSY || !This->pcm)
        {
            WARN("Cannot open sound device: %s\n", snd_strerror(err));
            return DSERR_GENERIC;
        }
        snd_pcm_drop(This->pcm);
        snd_pcm_close(This->pcm);
        This->pcm = NULL;
        err = snd_pcm_open(&pcm, WOutDev[This->drv->wDevID].pcmname, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
        if (err < 0)
        {
            WARN("Cannot open sound device: %s\n", snd_strerror(err));
            return DSERR_BUFFERLOST;
        }
    }

    /* Set some defaults */
    snd_pcm_hw_params_any(pcm, hw_params);
    err = snd_pcm_hw_params_set_channels(pcm, hw_params, pwfx->nChannels);
    if (err < 0) { WARN("Could not set channels to %d\n", pwfx->nChannels); goto err; }

    err = snd_pcm_hw_params_set_format(pcm, hw_params, format);
    if (err < 0) { WARN("Could not set format to %d bpp\n", pwfx->wBitsPerSample); goto err; }

    /* Alsa's rate resampling is only used if the application specifically requests
     * a buffer at a certain frequency, else it is better to disable it due to unwanted
     * side effects, which may include: Less granular pointer, changing buffer sizes, etc
     */
#if SND_LIB_VERSION >= 0x010009
    snd_pcm_hw_params_set_rate_resample(pcm, hw_params, 0);
#endif

    err = snd_pcm_hw_params_set_rate_near(pcm, hw_params, &rate, NULL);
    if (err < 0) { rate = pwfx->nSamplesPerSec; WARN("Could not set rate\n"); goto err; }

    if (!ALSA_NearMatch(rate, pwfx->nSamplesPerSec))
    {
        WARN("Could not set sound rate to %d, but instead to %d\n", pwfx->nSamplesPerSec, rate);
        pwfx->nSamplesPerSec = rate;
        pwfx->nAvgBytesPerSec = rate * pwfx->nBlockAlign;
        /* Let DirectSound detect this */
    }

    snd_pcm_hw_params_set_periods_integer(pcm, hw_params);
    snd_pcm_hw_params_set_buffer_time_near(pcm, hw_params, &buffer_time, NULL);
    buffer_time = 10000;
    snd_pcm_hw_params_set_period_time_near(pcm, hw_params, &buffer_time, NULL);

    err = snd_pcm_hw_params_get_period_size(hw_params, &psize, NULL);
    buffer_time = 16;
    snd_pcm_hw_params_set_periods_near(pcm, hw_params, &buffer_time, NULL);

    if (!This->mmap)
    {
        HeapFree(GetProcessHeap(), 0, This->mmap_buffer);
        This->mmap_buffer = NULL;
    }

    err = snd_pcm_hw_params_set_access (pcm, hw_params, SND_PCM_ACCESS_MMAP_INTERLEAVED);
    if (err >= 0)
        This->mmap = 1;
    else
    {
        This->mmap = 0;
        err = snd_pcm_hw_params_set_access (pcm, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
    }

    err = snd_pcm_hw_params(pcm, hw_params);

    /* ALSA needs at least 3 buffers to work successfully */
    This->mmap_commitahead = 3 * psize;
    while (This->mmap_commitahead <= 512)
        This->mmap_commitahead += psize;

    if (This->pcm)
    {
        snd_pcm_drop(This->pcm);
        snd_pcm_close(This->pcm);
    }
    This->pcm = pcm;
    snd_pcm_prepare(This->pcm);
    DSDB_CreateMMAP(This);
    return S_OK;

    err:
    if (err < 0)
        WARN("Failed to apply changes: %s\n", snd_strerror(err));

    if (!This->pcm)
        This->pcm = pcm;
    else
        snd_pcm_close(pcm);

    if (This->pcm)
        snd_pcm_hw_params_current(This->pcm, This->hw_params);

    return DSERR_BADFORMAT;
}
开发者ID:r6144,项目名称:wine,代码行数:101,代码来源:dsoutput.c


示例17: rdpsnd_alsa_set_params

static void rdpsnd_alsa_set_params(rdpsndAlsaPlugin* alsa)
{
    snd_pcm_hw_params_t* hw_p 

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C++ snd_pcm_recover函数代码示例发布时间:2022-05-30
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