C++ snd_pcm_delay函数代码示例

static guint
gst_alsasink_delay (GstAudioSink * asink)
{
    GstAlsaSink *alsa;
    snd_pcm_sframes_t delay;
    int res;

    alsa = GST_ALSA_SINK (asink);

    GST_DELAY_SINK_LOCK (asink);
    res = snd_pcm_delay (alsa->handle, &delay);
    GST_DELAY_SINK_UNLOCK (asink);
    if (G_UNLIKELY (res < 0)) {
        /* on errors, report 0 delay */
        GST_DEBUG_OBJECT (alsa, "snd_pcm_delay returned %d", res);
        delay = 0;
    }
    if (G_UNLIKELY (delay < 0)) {
        /* make sure we never return a negative delay */
        GST_WARNING_OBJECT (alsa, "snd_pcm_delay returned negative delay");
        delay = 0;
    }

    return delay;
}

int audioStreamer_ALSA::Read(char *buf, int len) // returns 0 if blocked, < 0 if error, > 0 if data
{
	int ret;
	if (m_sleep >= 0)
	{
		struct pollfd pfds[32];
		int cnt=snd_pcm_poll_descriptors(pcm_handle,pfds,32);
		if (cnt>0) poll(pfds,cnt,m_sleep);
	}

	ret=snd_pcm_readi(pcm_handle, buf, len/(m_nch*(m_bps/8)));

	if (ret < 0) 
	{
		if (ret != -EAGAIN) { snd_pcm_prepare(pcm_handle);  }
		return 0;
	}
#if 0
	snd_pcm_sframes_t del=0;
	if (!snd_pcm_delay(pcm_handle,&del) && del > m_bufsize/2 /* JF>used to be /1 */)
	{
		audiostream_onover();
		for (;;) if (snd_pcm_readi(pcm_handle, buf, len/(m_nch*(m_bps/8)))<0) break;
		// we have too many samples, eat some
	}
#endif

	return ret*m_nch*(m_bps/8);
}

int audioStreamer_ALSA::Write(char *buf, int len) // returns 0 on success
{
	snd_pcm_sframes_t del=0;
	if (!len) return 0;

	int cnt=1;
	if (!m_started || !snd_pcm_delay(pcm_handle,&del) && del<1)
	{
		if (m_started) audiostream_onunder();
		else m_started=1;
		cnt=m_nfrags;
    memset(buf,0,len); // reduce noise

	} 

	while (cnt-->0)
	{
		int ret=snd_pcm_writei(pcm_handle, buf, len/(m_nch*(m_bps/8)));
		if (ret < 0)
		{
			if (ret == -EPIPE) snd_pcm_prepare(pcm_handle);
			return 0;
		}
	}

	return 0;
}

static uint64_t
laudio_alsa_get_pos(void)
{
  snd_pcm_sframes_t delay;
  int ret;

  if (pcm_pos == 0)
    return 0;

  if (pcm_last_error != 0)
    return pcm_pos;

  if (snd_pcm_state(hdl) != SND_PCM_STATE_RUNNING)
    return pcm_pos;

  ret = snd_pcm_delay(hdl, &delay);
  if (ret < 0)
    {
      DPRINTF(E_WARN, L_LAUDIO, "Could not obtain PCM delay: %s\n", snd_strerror(ret));

      return pcm_pos;
    }

  return pcm_pos - delay;
}

static int
alsa_stream_get_position(cubeb_stream * stm, uint64_t * position)
{
  snd_pcm_sframes_t delay;

  assert(stm && position);

  pthread_mutex_lock(&stm->mutex);

  delay = -1;
  if (snd_pcm_state(stm->pcm) != SND_PCM_STATE_RUNNING ||
      snd_pcm_delay(stm->pcm, &delay) != 0) {
    *position = stm->last_position;
    pthread_mutex_unlock(&stm->mutex);
    return CUBEB_OK;
  }

  assert(delay >= 0);

  *position = 0;
  if (stm->write_position >= (snd_pcm_uframes_t) delay) {
    *position = stm->write_position - delay;
  }

  stm->last_position = *position;

  pthread_mutex_unlock(&stm->mutex);
  return CUBEB_OK;
}

size_t
audio_offset()
{
  snd_pcm_delay(playback_handle, &delay);

  return written - delay;
}

static void CheckXRUN(IDsDriverBufferImpl* This)
{
    snd_pcm_state_t state = snd_pcm_state(This->pcm);
    snd_pcm_sframes_t delay;
    int err;

    snd_pcm_hwsync(This->pcm);
    snd_pcm_delay(This->pcm, &delay);
    if ( state == SND_PCM_STATE_XRUN )
    {
        err = snd_pcm_prepare(This->pcm);
        CommitAll(This);
        snd_pcm_start(This->pcm);
        WARN("xrun occurred\n");
        if ( err < 0 )
            ERR("recovery from xrun failed, prepare failed: %s\n", snd_strerror(err));
    }
    else if ( state == SND_PCM_STATE_SUSPENDED )
    {
        int err = snd_pcm_resume(This->pcm);
        TRACE("recovery from suspension occurred\n");
        if (err < 0 && err != -EAGAIN){
            err = snd_pcm_prepare(This->pcm);
            if (err < 0)
                ERR("recovery from suspend failed, prepare failed: %s\n", snd_strerror(err));
        }
    } else if ( state != SND_PCM_STATE_RUNNING ) {
        FIXME("Unhandled state: %d\n", state);
    }
}

void alsa_pause(struct codec *c, int p)
{
	struct alsa_state *state = (struct alsa_state *) c->data;
	if (snd_pcm_state(state->dev) != SND_PCM_STATE_PAUSED)
		snd_pcm_delay(state->dev, &state->delay);
	snd_pcm_pause(state->dev, p);
}

static void alsa_close ()
{
	snd_pcm_sframes_t delay;

	assert (handle != NULL);

	/* play what remained in the buffer */
	if (alsa_buf_fill) {
		assert (alsa_buf_fill < chunk_size);

		snd_pcm_format_set_silence (params.format,
				alsa_buf + alsa_buf_fill,
				(chunk_size - alsa_buf_fill) / bytes_per_frame
				* params.channels);
		alsa_buf_fill = chunk_size;
		play_buf_chunks ();
	}

	/* Wait for ALSA buffers to empty.
	 * Do not be tempted to use snd_pcm_nonblock() and snd_pcm_drain()
	 * here; there are two bugs in ALSA which make it a bad idea (see
	 * the SVN commit log for r2550).  Instead we sleep for the duration
	 * of the still unplayed samples. */
	if (snd_pcm_delay (handle, &delay) == 0)
		usleep ((uint64_t) delay * 1000000 / params.rate);
	snd_pcm_close (handle);
	logit ("ALSA device closed");

	params.format = 0;
	params.rate = 0;
	params.channels = 0;
	handle = NULL;
}

ssize_t alsa_delay(struct codec *c)
{
	struct alsa_state *state = (struct alsa_state *) c->data;
	if (snd_pcm_state(state->dev) == SND_PCM_STATE_PAUSED)
		return state->delay;
	snd_pcm_delay(state->dev, &state->delay);
	return state->delay;
}

static void
audio_get_output_timestamp(AVFormatContext *s1, int stream,
    int64_t *dts, int64_t *wall)
{
    AlsaData *s  = s1->priv_data;
    snd_pcm_sframes_t delay = 0;
    *wall = av_gettime();
    snd_pcm_delay(s->h, &delay);
    *dts = s1->streams[0]->cur_dts - delay;
}

            float delay()
            {
               if (!runnable)
                  return 0.0;

               snd_pcm_sframes_t delay;
               snd_pcm_delay(pcm, &delay);

               return (float)delay / fps;
            }

qint64 QAudioInputPrivate::bytesAvailable() const
{
    if ( !handle )
        return 0;

    snd_pcm_sframes_t frames;
    if ( snd_pcm_delay( handle, &frames ) < 0 )
        return 0;

    return snd_pcm_frames_to_bytes( handle, frames );
}

static int
alsa_audio_deliver(audio_decoder_t *ad, int samples, int64_t pts, int epoch)
{
  decoder_t *d = (decoder_t *)ad;
  media_pipe_t *mp = ad->ad_mp;
  int c;

 retry:
  c = snd_pcm_wait(d->h, 100);
  if(c >= 0) {
    c = snd_pcm_avail_update(d->h);
  }
  if(c == -EPIPE) {
    snd_pcm_prepare(d->h);
    usleep(100000);
    TRACE(TRACE_DEBUG, "ALSA", "Audio underrun");
    d->samples = 0;
    goto retry;
  }

  c = MIN(d->max_frames_per_write, c);

  uint8_t *planes[8] = {0};
  planes[0] = d->tmp;
  c = avresample_read(ad->ad_avr, planes, c);
  
  snd_pcm_status_t *status;
  int err;
  snd_pcm_status_alloca(&status);
  if ((err = snd_pcm_status(d->h, status)) >= 0) {

    if(pts != AV_NOPTS_VALUE) {
      snd_htimestamp_t hts;
      snd_pcm_status_get_trigger_htstamp(status, &hts);
      int64_t ts = hts.tv_sec * 1000000LL + hts.tv_nsec / 1000;
      ts += d->samples * 1000000LL / ad->ad_out_sample_rate;

      hts_mutex_lock(&mp->mp_clock_mutex);
      mp->mp_audio_clock_avtime = ts;
      mp->mp_audio_clock = pts;
      mp->mp_audio_clock_epoch = epoch;
      hts_mutex_unlock(&mp->mp_clock_mutex);
    }
  }

  snd_pcm_sframes_t fr;
  if(!snd_pcm_delay(d->h, &fr))
    ad->ad_delay = 1000000L * fr / ad->ad_out_sample_rate;

  c = snd_pcm_writei(d->h, d->tmp, c);
  d->samples += c;
  return 0;
}

static snd_pcm_sframes_t
alsaspdifsink_delay (AlsaSPDIFSink * sink)
{
  snd_pcm_sframes_t delay;
  int err;

  err = snd_pcm_delay (sink->pcm, &delay);
  if (err < 0 || delay < 0) {
    return 0;
  }

  return delay;
}