C++ snd_pcm_hw_params_set_channels函数代码示例

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本文整理汇总了C++中snd_pcm_hw_params_set_channels函数的典型用法代码示例。如果您正苦于以下问题：C++ snd_pcm_hw_params_set_channels函数的具体用法？C++ snd_pcm_hw_params_set_channels怎么用？C++ snd_pcm_hw_params_set_channels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_hw_params_set_channels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: snd_pcm_hw_params_malloc

bool PokeLaunchApplication::sound() {
#if JUCE_LINUX
  int err;
  int freq = 44100, channels = 2;
  snd_pcm_hw_params_t *hw_params;
  snd_pcm_sw_params_t *sw_params;

  snd_pcm_hw_params_malloc( &hw_params );
  snd_pcm_sw_params_malloc( &sw_params );
  err = snd_pcm_open( &g_alsa_playback_handle, "default", SND_PCM_STREAM_PLAYBACK, 0 );
  if( err < 0 )
  {
  	 DBG( "ALSA ERROR: Can't open audio device: " << snd_strerror( err ) );
	   return false;
  }
  DBG("Opened Audio Device");
  err = snd_pcm_hw_params_any( g_alsa_playback_handle, hw_params );
  if( err < 0 )
  {
	    DBG( "ALSA ERROR: Can't initialize hardware parameter structure: " << snd_strerror( err ) );
	    return false;
  }
  err = snd_pcm_hw_params_set_access( g_alsa_playback_handle, hw_params, SND_PCM_ACCESS_RW_INTERLEAVED );
  if( err < 0 )
  {
      DBG( "ALSA ERROR: Can't set access type: " << snd_strerror( err ) );
      return false;
  }
  //const UTF8_CHAR *sample_format = "";
  err = snd_pcm_hw_params_set_format( g_alsa_playback_handle, hw_params, SND_PCM_FORMAT_S16_LE );
  if( err < 0 )
  {
	    DBG( "ALSA ERROR: Can't set sample format :" << snd_strerror( err ) );
	    return false;
  }
  err = snd_pcm_hw_params_set_rate_near( g_alsa_playback_handle, hw_params, (unsigned int*)&freq, 0 );
  if( err < 0 )
  {
      DBG( "ALSA ERROR: Can't set sample rate: " << snd_strerror( err ) );
	    return false;
  }
  DBG( "ALSA Sample rate: "<< freq );
  err = snd_pcm_hw_params_set_channels( g_alsa_playback_handle, hw_params, channels );
  if( err < 0 )
  {
	    DBG( "ALSA ERROR: Can't set channel count: " << snd_strerror( err ) );
	    return false;
  }
  snd_pcm_uframes_t frames;
	frames = DEFAULT_BUFFER_SIZE;
  err = snd_pcm_hw_params_set_buffer_size_near( g_alsa_playback_handle, hw_params, &frames );
  if( err < 0 )
  {
	    DBG( "ALSA ERROR: Can't set buffer size: " << snd_strerror( err ) );
	    return false;
  }
  snd_pcm_hw_params_get_buffer_size( hw_params, &frames );
  DBG( "ALSA Buffer size: 4096 samples" );
  err = snd_pcm_hw_params( g_alsa_playback_handle, hw_params );
  if( err < 0 )
  {
	   DBG( "ALSA ERROR: Can't set parameters: " << snd_strerror( err ) );
	    return false;
  }
  snd_pcm_hw_params_free( hw_params );
  snd_pcm_sw_params_free( sw_params );

  err = snd_pcm_prepare( g_alsa_playback_handle );
  if( err < 0 )
  {
	   DBG( "ALSA ERROR: Can't prepare audio interface for use: " << snd_strerror( err ) );
	  return false;
  }

  /* Stop PCM device and drop pending frames */
  snd_pcm_drain(g_alsa_playback_handle);
#endif

  return true;
}

开发者ID:brandonlanky，项目名称:PocketCHIP-pocket-home，代码行数:80，代码来源:Main.cpp

示例2: main

int 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;
	char *buffer;

	/* 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_S16_LE);

	/* Two channels (stereo) */
	snd_pcm_hw_params_set_channels(handle, params, 2);

	/* 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 = 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 5 seconds */
	snd_pcm_hw_params_get_period_time(params,
			&val, &dir);
	loops = 5000000 / val;

	while (loops > 0) {
		loops--;
		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",
					snd_strerror(rc));
		} else if (rc != (int)frames) {
			fprintf(stderr, "short read, read %d frames\n", rc);
		}
		rc = write(1, buffer, size);
		if (rc != size)
			fprintf(stderr,
					"short write: wrote %d bytes\n", rc);
	}

	snd_pcm_drain(handle);
	snd_pcm_close(handle);
	free(buffer);

	return 0;
}

开发者ID:windleos，项目名称:sound-card，代码行数:96，代码来源:demo5.c

示例3: pcm_open

static int pcm_open(struct alsa_pcm *alsa, const char *device_name,
                    snd_pcm_stream_t stream, int rate, int buffer_time)
{
    int r, dir;
    unsigned int p;
    size_t bytes;
    snd_pcm_hw_params_t *hw_params;
    
    r = snd_pcm_open(&alsa->pcm, device_name, stream, SND_PCM_NONBLOCK);
    if (r < 0) {
        alsa_error("open", r);
        return -1;
    }

    snd_pcm_hw_params_alloca(&hw_params);

    r = snd_pcm_hw_params_any(alsa->pcm, hw_params);
    if (r < 0) {
        alsa_error("hw_params_any", r);
        return -1;
    }
    
    r = snd_pcm_hw_params_set_access(alsa->pcm, hw_params,
                                     SND_PCM_ACCESS_RW_INTERLEAVED);
    if (r < 0) {
        alsa_error("hw_params_set_access", r);
        return -1;
    }
    
    r = snd_pcm_hw_params_set_format(alsa->pcm, hw_params, SND_PCM_FORMAT_S16);
    if (r < 0) {
        alsa_error("hw_params_set_format", r);
        fprintf(stderr, "16-bit signed format is not available. "
                "You may need to use a 'plughw' device.\n");
        return -1;
    }

    r = snd_pcm_hw_params_set_rate(alsa->pcm, hw_params, rate, 0);
    if (r < 0) {
        alsa_error("hw_params_set_rate", r);
        fprintf(stderr, "%dHz sample rate not available. You may need to use "
                "a 'plughw' device.\n", rate);
        return -1;
    }
    alsa->rate = rate;

    r = snd_pcm_hw_params_set_channels(alsa->pcm, hw_params, DEVICE_CHANNELS);
    if (r < 0) {
        alsa_error("hw_params_set_channels", r);
        fprintf(stderr, "%d channel audio not available on this device.\n",
                DEVICE_CHANNELS);
        return -1;
    }

    p = buffer_time * 1000; /* microseconds */
    dir = -1;
    r = snd_pcm_hw_params_set_buffer_time_max(alsa->pcm, hw_params, &p, &dir);
    if (r < 0) {
        alsa_error("hw_params_set_buffer_time_max", r);
        fprintf(stderr, "Buffer of %dms may be too small for this hardware.\n",
                buffer_time);
        return -1;
    }

    p = 2; /* double buffering */
    dir = 1;
    r = snd_pcm_hw_params_set_periods_min(alsa->pcm, hw_params, &p, &dir);
    if (r < 0) {
        alsa_error("hw_params_set_periods_min", r);
        fprintf(stderr, "Buffer of %dms may be too small for this hardware.\n",
                buffer_time);
        return -1;
    }

    r = snd_pcm_hw_params(alsa->pcm, hw_params);
    if (r < 0) {
        alsa_error("hw_params", r);
        return -1;
    }
    
    r = snd_pcm_hw_params_get_period_size(hw_params, &alsa->period, &dir);
    if (r < 0) {
        alsa_error("get_period_size", r);
        return -1;
    }

    bytes = alsa->period * DEVICE_CHANNELS * sizeof(signed short);
    alsa->buf = malloc(bytes);
    if (!alsa->buf) {
        perror("malloc");
        return -1;
    }

    /* snd_pcm_readi() returns uninitialised memory on first call,
     * possibly caused by premature POLLIN. Keep valgrind happy. */

    memset(alsa->buf, 0, bytes);

    return 0;
}

开发者ID:ewanuno，项目名称:xwax，代码行数:100，代码来源:alsa.c

示例4: pcm_open

static int pcm_open(pcm_handle_t* pcm, const pcm_desc_t* desc)
{
  const snd_pcm_format_t fmt = SND_PCM_FORMAT_S16_LE;
  snd_pcm_stream_t stm;
  int err;

  if (desc->flags & PCM_FLAG_IN) stm = SND_PCM_STREAM_CAPTURE;
  else stm = SND_PCM_STREAM_PLAYBACK;

  err = snd_pcm_open
    (&pcm->pcm, desc->name, stm, SND_PCM_NONBLOCK);
  if (err) PERROR_GOTO(snd_strerror(err), on_error_0);

  err = snd_pcm_hw_params_malloc(&pcm->hw_params);
  if (err) PERROR_GOTO(snd_strerror(err), on_error_1);

  err = snd_pcm_hw_params_any(pcm->pcm, pcm->hw_params);
  if (err) PERROR_GOTO(snd_strerror(err), on_error_2);

  err = snd_pcm_hw_params_set_access
    (pcm->pcm, pcm->hw_params, SND_PCM_ACCESS_RW_INTERLEAVED);
  if (err) PERROR_GOTO(snd_strerror(err), on_error_2);

  err = snd_pcm_hw_params_set_format(pcm->pcm, pcm->hw_params, fmt);
  if (err) PERROR_GOTO(snd_strerror(err), on_error_2);

  err = snd_pcm_hw_params_set_rate
    (pcm->pcm, pcm->hw_params, desc->fsampl, 0);
  if (err) PERROR_GOTO(snd_strerror(err), on_error_2);

  pcm->nchan = desc->nchan;
  pcm->wchan = (size_t)snd_pcm_format_physical_width(fmt) / 8;
  pcm->scale = pcm->nchan * pcm->wchan;

  err = snd_pcm_hw_params_set_channels
    (pcm->pcm, pcm->hw_params, desc->nchan);
  if (err) PERROR_GOTO(snd_strerror(err), on_error_2);
  
  err = snd_pcm_hw_params(pcm->pcm, pcm->hw_params);
  if (err) PERROR_GOTO(snd_strerror(err), on_error_2);

  err = snd_pcm_sw_params_malloc(&pcm->sw_params);
  if (err) PERROR_GOTO(snd_strerror(err), on_error_2);

  err = snd_pcm_sw_params_current(pcm->pcm, pcm->sw_params);
  if (err) PERROR_GOTO(snd_strerror(err), on_error_3);

#if 1
  err = snd_pcm_sw_params_set_avail_min
    (pcm->pcm, pcm->sw_params, 1024);
  if (err) PERROR_GOTO(snd_strerror(err), on_error_3);
#endif

#if 1
  err = snd_pcm_sw_params_set_start_threshold
    (pcm->pcm, pcm->sw_params, 0U);
  if (err) PERROR_GOTO(snd_strerror(err), on_error_3);
#endif

  err = snd_pcm_sw_params(pcm->pcm, pcm->sw_params);
  if (err) PERROR_GOTO(snd_strerror(err), on_error_3);
  
  err = snd_pcm_prepare(pcm->pcm);
  if (err) PERROR_GOTO(snd_strerror(err), on_error_3);

  pcm->rpos = 0;
  pcm->wpos = 0;
  pcm->nsampl = (size_t)desc->fsampl * 10;
  pcm->buf = malloc(pcm->nsampl * pcm->scale);
  if (pcm->buf == NULL) goto on_error_3;

  return 0;

 on_error_3:
  snd_pcm_sw_params_free(pcm->sw_params);
 on_error_2:
  snd_pcm_hw_params_free(pcm->hw_params);
 on_error_1:
  snd_pcm_close(pcm->pcm);
 on_error_0:
  return -1;
  
}

开发者ID:texane，项目名称:aspect，代码行数:83，代码来源:main.c

示例5: SetupSound

void SetupSound(void)
{
 snd_pcm_hw_params_t *hwparams;
 snd_pcm_sw_params_t *swparams;
 snd_pcm_status_t *status;
 int pspeed;
 int pchannels;
 int format;
 int buffer_time;
 int period_time;
 int err;

 if(iDisStereo) pchannels=1;
 else pchannels=2;

 pspeed=48000;
 format=SND_PCM_FORMAT_S16_LE;
 buffer_time=500000;
 period_time=buffer_time/4;

 if((err=snd_pcm_open(&handle, "default", 
                      SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK))<0)
  {
   printf("Audio open error: %s\n", snd_strerror(err));
   return;
  }

 if((err=snd_pcm_nonblock(handle, 0))<0)
  {
   printf("Can't set blocking moded: %s\n", snd_strerror(err));
   return;
  }

 snd_pcm_hw_params_alloca(&hwparams);
 snd_pcm_sw_params_alloca(&swparams);
 if((err=snd_pcm_hw_params_any(handle, hwparams))<0)
  {
   printf("Broken configuration for this PCM: %s\n", snd_strerror(err));
   return;
  }

 if((err=snd_pcm_hw_params_set_access(handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED))<0)
  {
   printf("Access type not available: %s\n", snd_strerror(err));
   return;
  }

 if((err=snd_pcm_hw_params_set_format(handle, hwparams, format))<0)
  {
   printf("Sample format not available: %s\n", snd_strerror(err));
   return;
  }

 if((err=snd_pcm_hw_params_set_channels(handle, hwparams, pchannels))<0)
  {
   printf("Channels count not available: %s\n", snd_strerror(err));
   return;
  }

 if((err=snd_pcm_hw_params_set_rate_near(handle, hwparams, &pspeed, 0))<0)
  {
   printf("Rate not available: %s\n", snd_strerror(err));
   return;
  }

 if((err=snd_pcm_hw_params_set_buffer_time_near(handle, hwparams, &buffer_time, 0))<0)
  {
   printf("Buffer time error: %s\n", snd_strerror(err));
   return;
  }

 if((err=snd_pcm_hw_params_set_period_time_near(handle, hwparams, &period_time, 0))<0)
  {
   printf("Period time error: %s\n", snd_strerror(err));
   return;
  }

 if((err=snd_pcm_hw_params(handle, hwparams))<0)
  {
   printf("Unable to install hw params: %s\n", snd_strerror(err));
   return;
  }

 snd_pcm_status_alloca(&status);
 if((err=snd_pcm_status(handle, status))<0)
  {
   printf("Unable to get status: %s\n", snd_strerror(err));
   return;
  }

 buffer_size=snd_pcm_status_get_avail(status);
}

开发者ID:0xZERO3，项目名称:PCSX2-rr-lua，代码行数:92，代码来源:alsa.c

示例6: set_params_raw

static int set_params_raw(alsa_param_t *alsa_params)
{
    snd_pcm_hw_params_t *hwparams;
    snd_pcm_sw_params_t *swparams;
    //  snd_pcm_uframes_t buffer_size;
    //  snd_pcm_uframes_t boundary;
    //  unsigned int period_time = 0;
    //  unsigned int buffer_time = 0;
    snd_pcm_uframes_t bufsize;
    int err;
    unsigned int rate;
    snd_pcm_uframes_t start_threshold, stop_threshold;
    snd_pcm_hw_params_alloca(&hwparams);
    snd_pcm_sw_params_alloca(&swparams);

    err = snd_pcm_hw_params_any(alsa_params->handle, hwparams);
    if (err < 0) {
        adec_print("Broken configuration for this PCM: no configurations available");
        return err;
    }

    err = snd_pcm_hw_params_set_access(alsa_params->handle, hwparams,
                                       SND_PCM_ACCESS_RW_INTERLEAVED);
    if (err < 0) {
        adec_print("Access type not available");
        return err;
    }

    err = snd_pcm_hw_params_set_format(alsa_params->handle, hwparams, alsa_params->format);
    if (err < 0) {
        adec_print("Sample format non available");
        return err;
    }

    err = snd_pcm_hw_params_set_channels(alsa_params->handle, hwparams, alsa_params->channelcount);
    if (err < 0) {
        adec_print("Channels count non available");
        return err;
    }

    rate = alsa_params->rate;
    err = snd_pcm_hw_params_set_rate_near(alsa_params->handle, hwparams, &alsa_params->rate, 0);
    assert(err >= 0);
#if 0
    err = snd_pcm_hw_params_get_buffer_time_max(hwparams,  &buffer_time, 0);
    assert(err >= 0);
    if (buffer_time > 500000) {
        buffer_time = 500000;
    }

    period_time = buffer_time / 4;

    err = snd_pcm_hw_params_set_period_time_near(handle, hwparams,
            &period_time, 0);
    assert(err >= 0);

    err = snd_pcm_hw_params_set_buffer_time_near(handle, hwparams,
            &buffer_time, 0);
    assert(err >= 0);

#endif
    alsa_params->bits_per_sample = snd_pcm_format_physical_width(alsa_params->format);
    //bits_per_frame = bits_per_sample * hwparams.realchanl;
    alsa_params->bits_per_frame = alsa_params->bits_per_sample * alsa_params->channelcount;

    bufsize = 	PERIOD_NUM*PERIOD_SIZE*4;

    err = snd_pcm_hw_params_set_buffer_size_near(alsa_params->handle, hwparams,&bufsize);
    if (err < 0) {
        adec_print("Unable to set  buffer  size \n");
        return err;
    }

    err = snd_pcm_hw_params_set_period_size_near(alsa_params->handle, hwparams, &chunk_size, NULL);
    if (err < 0) {
        adec_print("Unable to set period size \n");
        return err;
    }
#if 0
    err = snd_pcm_hw_params_set_periods_near(alsa_params->handle, hwparams, &fragcount, NULL);
    if (err < 0) {
      adec_print("Unable to set periods \n");
      return err;
    }
#endif
    err = snd_pcm_hw_params(alsa_params->handle, hwparams);
    if (err < 0) {
        adec_print("Unable to install hw params:");
        return err;
    }

    err = snd_pcm_hw_params_get_buffer_size(hwparams, &bufsize);
    if (err < 0) {
        adec_print("Unable to get buffersize \n");
        return err;
    }
    adec_print("[%s::%d]--[alsa raw buffer frame size:%d]\n", __FUNCTION__, __LINE__,bufsize);
    alsa_params->buffer_size = bufsize * alsa_params->bits_per_frame / 8;

#if 1
//.........这里部分代码省略.........

开发者ID:mdrjr，项目名称:c2_aml_libs，代码行数:101，代码来源:alsa-out-raw.c

示例7: snd_pcm_close

/**
* @brief Setup the ALSA handle to the audio device
*/
void AlsaPlayback::setupHandle()
{
	if(alsa_handle)
	{
		snd_pcm_close(alsa_handle);
		alsa_handle = NULL;
	}

	if(snd_pcm_open(&alsa_handle, "default", SND_PCM_STREAM_PLAYBACK, 0) < 0)
	{
		TRACE("Unable to open playback device!!\n");
		alsa_handle = NULL;
	}
	else
	{
		int rc = -1;
		snd_pcm_hw_params_t *params = NULL;

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

		if(NULL != params)
		{
			int dir = 0;

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

			/* Set the desired hardware parameters. */
			/* Interleaved mode */
			snd_pcm_hw_params_set_access(alsa_handle, params, SND_PCM_ACCESS_RW_INTERLEAVED);

			/* Signed 16-bit little-endian format */
			snd_pcm_hw_params_set_format(alsa_handle, params, SND_PCM_FORMAT_S16_LE);

			/* One channel (mono) */
			snd_pcm_hw_params_set_channels(alsa_handle, params, 1);

			/* set sampling rate */
			snd_pcm_hw_params_set_rate_near(alsa_handle, params, &AlsaPlayback::_sample_rate, &dir);

			/* Set period size to 128 frames (samples) */
			frames = AlsaPlayback::FRAME_PERIOD;
			snd_pcm_hw_params_set_period_size_near(alsa_handle, params, &frames, &dir);

			snd_pcm_uframes_t buf_size = MINIMUM_SAMPLE_SET_SIZE;
			snd_pcm_hw_params_set_buffer_size_near(alsa_handle, params, &buf_size);

			/* Write the parameters to the driver */
			rc = snd_pcm_hw_params(alsa_handle, params);

			if(rc >= 0)
			{
				TRACE("AlsaPlayback init completed successfully..\n");
			}
			else
			{
				snd_pcm_close(alsa_handle);
				alsa_handle = NULL;
				TRACE("AlsaPlayback init failed\n");
			}

			snd_pcm_hw_params_free(params);
			params = NULL;
		}
		else
		{
			TRACE("playAudio - snd_pcm_hw_params_alloc() failed\n");
		}
	}
}

开发者ID:AnXi-TieGuanYin-Tea，项目名称:Multi-Channel-Audio-Mixer，代码行数:74，代码来源:AlsaAudioPlayback.cpp

示例8: create_recorder

void create_recorder(int samp_rate,char* dev_name,int nchan,void **capture_handle_f)
{
  int err;
  snd_pcm_t *capture_handle;
  snd_pcm_hw_params_t *hw_params;
  printf("From C: *capture_handle_f=%p\n",*capture_handle_f);
  if ((err = snd_pcm_open ((snd_pcm_t**)capture_handle_f, dev_name,
			   SND_PCM_STREAM_CAPTURE, 0)) < 0) {
    fprintf (stderr, "cannot open audio device %s (%s)\n", 
	     dev_name,
	     snd_strerror (err));
    fprintf(stderr, "Hint: Use \"arecord -l\" to list recording devices.\n");
    exit (1);
  }
  capture_handle = (snd_pcm_t*)(*capture_handle_f);
  printf("made handle %p (&=%p) to %s at %d\n",capture_handle,&capture_handle,dev_name,samp_rate);
     
  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 (capture_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 (capture_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 (capture_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_resample(capture_handle,hw_params, 0)) < 0) {
    fprintf(stderr, "failed attempting to prevent ALSA resampling. (%s)",
	    snd_strerror(err));
    exit (1);
  }
  
  int dir = 0;
  unsigned int val = samp_rate;
  if ((err = snd_pcm_hw_params_set_rate_near (capture_handle,
       hw_params, &val, &dir)) < 0) {
    fprintf (stderr, "cannot set sample rate (%s)\n",
	     snd_strerror (err));
    exit (1);
  }

  if ((err = snd_pcm_hw_params_set_channels (capture_handle, hw_params, nchan)) < 0) {
    fprintf (stderr, "cannot set channel count (%s)\n",
	     snd_strerror (err));
    exit (1);
  }
  
  if ((err = snd_pcm_hw_params (capture_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 (capture_handle)) < 0) {
    fprintf (stderr, "cannot prepare audio interface for use (%s)\n",
	     snd_strerror (err));
    exit (1);
  }

  /*
  int nbuf = 8000;
  short *buf = (short*)malloc(sizeof(short)*nbuf);
  int i;
  
  for (i=0; i<2; ++i) {
    if ((err = snd_pcm_readi(capture_handle, buf, nbuf)) != nbuf) {
      fprintf (stderr, "read from audio interface failed (%s)\n",
	       snd_strerror(err));
      exit (1);
    }
    else {
      printf("read from C using %p into %p.\n",capture_handle,buf);
      }
  }
  */

}

开发者ID:lurobi，项目名称:pecanpi，代码行数:96，代码来源:alsa_pcm_read_simple.c

示例9: main

int main()
{
    long loops;
    int rc,i = 0;
    int size;
    FILE *fp ;
    snd_pcm_t *handle;
    snd_pcm_hw_params_t *params;
    unsigned int val,val2;
    int dir;
    snd_pcm_uframes_t frames;
    char *buffer;
    if(  (fp =fopen("sound.wav","w")) < 0)
        printf("open sound.wav fial\n");
    /* 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_S16_LE);
    /* Two channels (stereo) */
    snd_pcm_hw_params_set_channels(handle, params, 2);
    /* 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 = 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 */
    printf("size = %d\n",size);
    buffer = (char *) malloc(size);
    /* We want to loop for 5 seconds */
    snd_pcm_hw_params_get_period_time(params,  &val, &dir);
    loops = 10000000 / val;
    while (loops > 0)
    {
        loops--;
        rc = snd_pcm_readi(handle, buffer, frames);
        printf("%d\n",i++);
        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 = fwrite( buffer,1, size,fp);
        rc = write(1,buffer,size);
        if (rc != size)
            fprintf(stderr,  "short write: wrote %d bytes/n", rc);
        else printf("fwrite buffer success\n");
    }
    /******************打印参数*********************/
    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);
//.........这里部分代码省略.........

开发者ID:renhardly，项目名称:learngit，代码行数:101，代码来源:capture.c

示例10: main

int main(int argc, char *argv[])
{
	ros::init(argc, argv, "cmd_control");
	ros::NodeHandle n;
	ros::Publisher motor_pub = n.advertise<std_msgs::Char>("motor_chatter", 1000);
	std::string port;
	ros::param::param<std::string>("~port", port, "/dev/ttyACM0");
	int baud;
	ros::param::param<int>("~baud", baud, 57600);
	int byte_per_sample = bits_per_sample >> 3;
	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;
	char *speechbuf;

	/* 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_S16_LE);

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

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

	/* Set period size to 32 frames. */
	frames = frames_per_period;
	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 * channels * byte_per_sample; /* 2 bytes/sample, 1 channels */
	speechbuf = (char *) malloc(size);
	//printf("buffer size  %d\n", size);
 
	/* We want to loop for 5 seconds */
	snd_pcm_hw_params_get_period_time(params,
			&val, &dir);
	
	int client_sockfd;  
	struct sockaddr_in remote_addr; //服务器端网络地址结构体  
	memset(&remote_addr,0,sizeof(remote_addr)); //数据初始化--清零  
	remote_addr.sin_family=AF_INET; //设置为IP通信  
	remote_addr.sin_addr.s_addr=inet_addr("10.0.1.0");//服务器IP地址  
	remote_addr.sin_port=htons(port); //服务器端口号  

	/*创建客户端套接字--IPv4协议，面向连接通信，TCP协议*/  
	if((client_sockfd=socket(PF_INET,SOCK_STREAM,0))<0)  
	{  
		perror("socket failed\n");  
		return 1;  
	}  

	/*将套接字绑定到服务器的网络地址上*/  
	if(connect(client_sockfd,(struct sockaddr *)&remote_addr,sizeof(struct sockaddr))<0)  
	{  
		perror("connect failed\n");  
		return 1;  
//.........这里部分代码省略.........

开发者ID:Yvaine，项目名称:speech-robot，代码行数:101，代码来源:recvcmd.cpp

示例11: AudioOutputDevice

    /**
     * Create and initialize Alsa audio output device with given parameters.
     *
     * @param Parameters - optional parameters
     * @throws AudioOutputException  if output device cannot be opened
     */
    AudioOutputDeviceAlsa::AudioOutputDeviceAlsa(std::map<String,DeviceCreationParameter*> Parameters) : AudioOutputDevice(Parameters), Thread(true, true, 1, 0) {
        pcm_handle           = NULL;
        stream               = SND_PCM_STREAM_PLAYBACK;
        this->uiAlsaChannels = ((DeviceCreationParameterInt*)Parameters["CHANNELS"])->ValueAsInt();
        this->uiSamplerate   = ((DeviceCreationParameterInt*)Parameters["SAMPLERATE"])->ValueAsInt();
        this->FragmentSize   = ((DeviceCreationParameterInt*)Parameters["FRAGMENTSIZE"])->ValueAsInt();
        uint Fragments       = ((DeviceCreationParameterInt*)Parameters["FRAGMENTS"])->ValueAsInt();
        String Card          = ((DeviceCreationParameterString*)Parameters["CARD"])->ValueAsString();

        dmsg(2,("Checking if hw parameters supported...\n"));
        if (HardwareParametersSupported(Card, uiAlsaChannels, uiSamplerate, Fragments, FragmentSize)) {
            pcm_name = "hw:" + Card;
        }
        else {
            fprintf(stderr, "Warning: your soundcard doesn't support chosen hardware parameters; ");
            fprintf(stderr, "trying to compensate support lack with plughw...");
            fflush(stdout);
            pcm_name = "plughw:" + Card;
        }
        dmsg(2,("HW check completed.\n"));

        int err;

        snd_pcm_hw_params_alloca(&hwparams);  // Allocate the snd_pcm_hw_params_t structure on the stack.

        /* Open PCM. The last parameter of this function is the mode. */
        /* If this is set to 0, the standard mode is used. Possible   */
        /* other values are SND_PCM_NONBLOCK and SND_PCM_ASYNC.       */
        /* If SND_PCM_NONBLOCK is used, read / write access to the    */
        /* PCM device will return immediately. If SND_PCM_ASYNC is    */
        /* specified, SIGIO will be emitted whenever a period has     */
        /* been completely processed by the soundcard.                */
        if ((err = snd_pcm_open(&pcm_handle, pcm_name.c_str(), stream, 0)) < 0) {
            throw AudioOutputException(String("Error opening PCM device ") + pcm_name + ": " + snd_strerror(err));
        }

        if ((err = snd_pcm_hw_params_any(pcm_handle, hwparams)) < 0) {
            throw AudioOutputException(String("Error, cannot initialize hardware parameter structure: ") + snd_strerror(err));
        }

        /* Set access type. This can be either    */
        /* SND_PCM_ACCESS_RW_INTERLEAVED or       */
        /* SND_PCM_ACCESS_RW_NONINTERLEAVED.      */
        if ((err = snd_pcm_hw_params_set_access(pcm_handle, hwparams, SND_PCM_ACCESS_RW_INTERLEAVED)) < 0) {
            throw AudioOutputException(String("Error snd_pcm_hw_params_set_access: ") + snd_strerror(err));
        }

        /* Set sample format */
        #if WORDS_BIGENDIAN
        if ((err = snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_BE)) < 0)
        #else // little endian
        if ((err = snd_pcm_hw_params_set_format(pcm_handle, hwparams, SND_PCM_FORMAT_S16_LE)) < 0)
        #endif
        {
            throw AudioOutputException(String("Error setting sample format: ") + snd_strerror(err));
        }

        int dir = 0;

        /* Set sample rate. If the exact rate is not supported */
        /* by the hardware, use nearest possible rate.         */
        #if ALSA_MAJOR > 0
        if((err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, &uiSamplerate, &dir)) < 0)
        #else
        if((err = snd_pcm_hw_params_set_rate_near(pcm_handle, hwparams, uiSamplerate, &dir)) < 0)
        #endif
        {
            throw AudioOutputException(String("Error setting sample rate: ") + snd_strerror(err));
        }

        if ((err = snd_pcm_hw_params_set_channels(pcm_handle, hwparams, uiAlsaChannels)) < 0) {
            throw AudioOutputException(String("Error setting number of channels: ") + snd_strerror(err));
        }

        /* Set number of periods. Periods used to be called fragments. */
        if ((err = snd_pcm_hw_params_set_periods(pcm_handle, hwparams, Fragments, dir)) < 0) {
            throw AudioOutputException(String("Error setting number of ") + ToString(Fragments) + " periods: " + snd_strerror(err));
        }

        /* Set buffer size (in frames). The resulting latency is given by */
        /* latency = periodsize * periods / (rate * bytes_per_frame)     */
        if ((err = snd_pcm_hw_params_set_buffer_size(pcm_handle, hwparams, (FragmentSize * Fragments))) < 0) {
            throw AudioOutputException(String("Error setting buffersize: ") + snd_strerror(err));
        }

        /* Apply HW parameter settings to */
        /* PCM device and prepare device  */
        if ((err = snd_pcm_hw_params(pcm_handle, hwparams)) < 0) {
            throw AudioOutputException(String("Error setting HW params: ") + snd_strerror(err));
        }

        if (snd_pcm_sw_params_malloc(&swparams) != 0) {
            throw AudioOutputException(String("Error in snd_pcm_sw_params_malloc: ") + snd_strerror(err));
        }
//.........这里部分代码省略.........

开发者ID:svn2github，项目名称:linuxsampler，代码行数:101，代码来源:AudioOutputDeviceAlsa.cpp

示例12: printf

int AudioAlsa::setHWParams( const ch_cnt_t _channels, snd_pcm_access_t _access )
{
	int err, dir;

	// choose all parameters
	if( ( err = snd_pcm_hw_params_any( m_handle, m_hwParams ) ) < 0 )
	{
		printf( "Broken configuration for playback: no configurations "
				"available: %s\n", snd_strerror( err ) );
		return err;
	}

	// set the interleaved read/write format
	if( ( err = snd_pcm_hw_params_set_access( m_handle, m_hwParams,
							_access ) ) < 0 )
	{
		printf( "Access type not available for playback: %s\n",
							snd_strerror( err ) );
		return err;
	}

	// set the sample format
	if( ( snd_pcm_hw_params_set_format( m_handle, m_hwParams,
						SND_PCM_FORMAT_S16_LE ) ) < 0 )
	{
		if( ( snd_pcm_hw_params_set_format( m_handle, m_hwParams,
						SND_PCM_FORMAT_S16_BE ) ) < 0 )
		{
			printf( "Neither little- nor big-endian available for "
					"playback: %s\n", snd_strerror( err ) );
			return err;
		}
		m_convertEndian = isLittleEndian();
	}
	else
	{
		m_convertEndian = !isLittleEndian();
	}

	// set the count of channels
	if( ( err = snd_pcm_hw_params_set_channels( m_handle, m_hwParams,
							_channels ) ) < 0 )
	{
		printf( "Channel count (%i) not available for playbacks: %s\n"
				"(Does your soundcard not support surround?)\n",
					_channels, snd_strerror( err ) );
		return err;
	}

	// set the sample rate
	if( ( err = snd_pcm_hw_params_set_rate( m_handle, m_hwParams,
						sampleRate(), 0 ) ) < 0 )
	{
		if( ( err = snd_pcm_hw_params_set_rate( m_handle, m_hwParams,
				mixer()->baseSampleRate(), 0 ) ) < 0 )
		{
			printf( "Could not set sample rate: %s\n",
							snd_strerror( err ) );
			return err;
		}
	}

	m_periodSize = mixer()->framesPerPeriod();
	m_bufferSize = m_periodSize * 8;
	dir = 0;
	err = snd_pcm_hw_params_set_period_size_near( m_handle, m_hwParams,
							&m_periodSize, &dir );
	if( err < 0 )
	{
		printf( "Unable to set period size %lu for playback: %s\n",
					m_periodSize, snd_strerror( err ) );
		return err;
	}
	dir = 0;
	err = snd_pcm_hw_params_get_period_size( m_hwParams, &m_periodSize,
									&dir );
	if( err < 0 )
	{
		printf( "Unable to get period size for playback: %s\n",
							snd_strerror( err ) );
	}

	dir = 0;
	err = snd_pcm_hw_params_set_buffer_size_near( m_handle, m_hwParams,
								&m_bufferSize );
	if( err < 0 )
	{
		printf( "Unable to set buffer size %lu for playback: %s\n",
					m_bufferSize, snd_strerror( err ) );
		return ( err );
	}
	err = snd_pcm_hw_params_get_buffer_size( m_hwParams, &m_bufferSize );

	if( 2 * m_periodSize > m_bufferSize )
	{
		printf( "buffer to small, could not use\n" );
		return ( err );
	}


//.........这里部分代码省略.........

开发者ID:AHudon，项目名称:lmms，代码行数:101，代码来源:AudioAlsa.cpp

示例13: pcm_init

void pcm_init()
{
	int n, m, err;
	snd_pcm_hw_params_t *hw_params;

	if (!sound)
	{
		pcm.hz = 11025;
		pcm.len = 4096;
		pcm.buf = malloc(pcm.len);
		pcm.pos = 0;
		playback_handle = NULL;
		return;
	}

	if (!dsp_device) dsp_device = strdup(DSP_DEVICE);

	if ((err = snd_pcm_open (&playback_handle, dsp_device, SND_PCM_STREAM_PLAYBACK, 0)) < 0) {
		fprintf (stderr, "cannot open audio device %s (%s)\n", 
			 dsp_device,
			 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_U8)) < 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, samplerate, 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, stereo ? 2 : 1)) < 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);
	}

	pcm.stereo = stereo;
	pcm.hz = samplerate;
	pcm.len = 4096;
	pcm.buf = malloc(pcm.len);
}

开发者ID:geekmaster，项目名称:fbgnuboy，代码行数:79，代码来源:alsa.c

示例14: init_alsa

int init_alsa(unsigned int channels, unsigned sample_rate,
				 snd_pcm_format_t format)
{
	int	ret;
	snd_pcm_hw_params_t *hw_params;

	playback_handle = NULL;
	ret = snd_pcm_open(&playback_handle, "default", SND_PCM_STREAM_PLAYBACK,
			0);
	if (ret < 0) {
		fprintf(stderr, "can NOT open soundcard\n");
		goto fail;
	}

	ret = snd_pcm_hw_params_malloc(&hw_params);
	if (ret < 0) {
		fprintf(stderr, "can NOT allocate hardware paramter structure (%s)\n",
				snd_strerror(ret));
		goto fail;
	}

	ret = snd_pcm_hw_params_any(playback_handle, hw_params);
	if (ret < 0) {
		fprintf(stderr, "can NOT initialize hardware paramter structure (%s)\n",
				snd_strerror(ret));
		goto fail;
	}

	ret = snd_pcm_hw_params_set_access(playback_handle, hw_params,
									   SND_PCM_ACCESS_RW_INTERLEAVED);
	if (ret < 0) {
		fprintf(stderr, "can NOT set access type (%s)\n", snd_strerror(ret));
		goto fail;
	}

	ret = snd_pcm_hw_params_set_format(playback_handle, hw_params, format);
	if (ret < 0) {
		fprintf(stderr, "can NOT set sample format (%s)\n", snd_strerror(ret));
		goto fail;
	}

	ret = snd_pcm_hw_params_set_rate_near(playback_handle, hw_params,
										  &sample_rate, 0);
	if (ret < 0) {
		fprintf(stderr, "can NOT set sample rate (%s)\n", snd_strerror(ret));
		goto fail;
	}

	ret = snd_pcm_hw_params_set_channels(playback_handle, hw_params, channels);
	if (ret < 0) {
		fprintf(stderr, "can NOT set channels (%s)\n", snd_strerror(ret));
		goto fail;
	}

	snd_pcm_hw_params_get_buffer_size_max(hw_params, &buffer_size);
	buffer_size = buffer_size < ALSA_BUFFER_SIZE_MAX ?
				  buffer_size : ALSA_BUFFER_SIZE_MAX;
	ret = snd_pcm_hw_params_set_buffer_size_near(playback_handle, hw_params,
												 &buffer_size);
	if (ret < 0) {
		fprintf(stderr, "can NOT set alsa buffer size (%s)\n", snd_strerror(ret));
		goto fail;
	}

	snd_pcm_hw_params_get_period_size_min(hw_params, &period_size, NULL);
	if (!period_size)
		period_size = buffer_size / 4;
	ret = snd_pcm_hw_params_set_period_size_near(playback_handle, hw_params,
												 &period_size, NULL);
	if (ret < 0) {
		fprintf(stderr, "can NOT set alsa period size (%s)\n", snd_strerror(ret));
		goto fail;
	}

	ret = snd_pcm_hw_params(playback_handle, hw_params);
	if (ret < 0) {
		fprintf(stderr, "can NOT set parameters (%s)\n", snd_strerror(ret));
		goto fail;
	}
	
	snd_pcm_hw_params_free(hw_params);

	audio_channels = channels;
	audio_sample_rate = sample_rate;
	audio_format = format;

	ret = 0;

	return ret;
fail:
	if (playback_handle) {
		snd_pcm_close(playback_handle);
		playback_handle = NULL;
	}
	return ret;
}

开发者ID:fedoracdu，项目名称:tyl_player，代码行数:96，代码来源:alsa.c

示例15: sizeof

static void *alsa_init(const char *device, unsigned rate, unsigned latency)
{
   alsa_t *alsa = (alsa_t*)calloc(1, sizeof(alsa_t));
   if (!alsa)
      return NULL;

   snd_pcm_hw_params_t *params = NULL;
   snd_pcm_sw_params_t *sw_params = NULL;

   unsigned latency_usec = latency * 1000;
   unsigned channels = 2;
   unsigned periods = 4;
   snd_pcm_format_t format;

   const char *alsa_dev = "default";
   if (device)
      alsa_dev = device;

   snd_pcm_uframes_t buffer_size;

   TRY_ALSA(snd_pcm_open(&alsa->pcm, alsa_dev, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK));
   TRY_ALSA(snd_pcm_hw_params_malloc(&params));
   alsa->has_float = find_float_format(alsa->pcm, params);
   format = alsa->has_float ? SND_PCM_FORMAT_FLOAT : SND_PCM_FORMAT_S16;

   TRY_ALSA(snd_pcm_hw_params_any(alsa->pcm, params));
   TRY_ALSA(snd_pcm_hw_params_set_access(alsa

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

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

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C++ snd_pcm_hw_params_set_channels函数代码示例

示例1: snd_pcm_hw_params_malloc

示例2: main

示例3: pcm_open

示例4: pcm_open

示例5: SetupSound

示例6: set_params_raw

示例7: snd_pcm_close

示例8: create_recorder

示例9: main

示例10: main

示例11: AudioOutputDevice

示例12: printf

示例13: pcm_init

示例14: init_alsa

示例15: sizeof

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