예제 #1
0
gboolean
gst_pulse_fill_format_info (GstAudioRingBufferSpec * spec, pa_format_info ** f,
    guint * channels)
{
  pa_format_info *format;
  pa_sample_format_t sf = PA_SAMPLE_INVALID;
  GstAudioInfo *ainfo = &spec->info;

  format = pa_format_info_new ();

  if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MU_LAW
      && GST_AUDIO_INFO_WIDTH (ainfo) == 8) {
    format->encoding = PA_ENCODING_PCM;
    sf = PA_SAMPLE_ULAW;
  } else if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_A_LAW
      && GST_AUDIO_INFO_WIDTH (ainfo) == 8) {
    format->encoding = PA_ENCODING_PCM;
    sf = PA_SAMPLE_ALAW;
  } else if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW) {
    format->encoding = PA_ENCODING_PCM;
    if (!gstaudioformat_to_pasampleformat (GST_AUDIO_INFO_FORMAT (ainfo), &sf))
      goto fail;
  } else if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3) {
    format->encoding = PA_ENCODING_AC3_IEC61937;
  } else if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_EAC3) {
    format->encoding = PA_ENCODING_EAC3_IEC61937;
  } else if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_DTS) {
    format->encoding = PA_ENCODING_DTS_IEC61937;
  } else if (spec->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_MPEG) {
    format->encoding = PA_ENCODING_MPEG_IEC61937;
  } else {
    goto fail;
  }

  if (format->encoding == PA_ENCODING_PCM) {
    pa_format_info_set_sample_format (format, sf);
    pa_format_info_set_channels (format, GST_AUDIO_INFO_CHANNELS (ainfo));
  }

  pa_format_info_set_rate (format, GST_AUDIO_INFO_RATE (ainfo));

  if (!pa_format_info_valid (format))
    goto fail;

  *f = format;
  *channels = GST_AUDIO_INFO_CHANNELS (ainfo);

  return TRUE;

fail:
  if (format)
    pa_format_info_free (format);
  return FALSE;
}
예제 #2
0
void
gst_audio_parse_update_frame_size (GstAudioParse * ap)
{
  gint framesize, width;

  switch (ap->format) {
    case GST_AUDIO_PARSE_FORMAT_ALAW:
    case GST_AUDIO_PARSE_FORMAT_MULAW:
      width = 8;
      break;
    case GST_AUDIO_PARSE_FORMAT_RAW:
    default:
    {
      GstAudioInfo info;

      gst_audio_info_init (&info);
      /* rate, etc do not really matter here */
      gst_audio_info_set_format (&info, ap->raw_format, 44100, ap->channels,
          NULL);
      width = GST_AUDIO_INFO_WIDTH (&info);
      break;
    }
  }

  framesize = (width / 8) * ap->channels;

  gst_raw_parse_set_framesize (GST_RAW_PARSE (ap), framesize);
}
예제 #3
0
static GstFlowReturn
gst_deinterleave_chain (GstPad * pad, GstObject * parent, GstBuffer * buffer)
{
  GstDeinterleave *self = GST_DEINTERLEAVE (parent);
  GstFlowReturn ret;

  g_return_val_if_fail (self->func != NULL, GST_FLOW_NOT_NEGOTIATED);
  g_return_val_if_fail (GST_AUDIO_INFO_WIDTH (&self->audio_info) > 0,
      GST_FLOW_NOT_NEGOTIATED);
  g_return_val_if_fail (GST_AUDIO_INFO_CHANNELS (&self->audio_info) > 0,
      GST_FLOW_NOT_NEGOTIATED);

  ret = gst_deinterleave_process (self, buffer);

  if (ret != GST_FLOW_OK)
    GST_DEBUG_OBJECT (self, "flow return: %s", gst_flow_get_name (ret));

  return ret;
}
예제 #4
0
static gboolean
gst_audio_segment_clip_set_caps (GstSegmentClip * base, GstCaps * caps)
{
  GstAudioSegmentClip *self = GST_AUDIO_SEGMENT_CLIP (base);
  gboolean ret;
  GstAudioInfo info;
  gint rate, channels, width;

  gst_audio_info_init (&info);
  ret = gst_audio_info_from_caps (&info, caps);

  if (ret) {
    rate = GST_AUDIO_INFO_RATE (&info);
    channels = GST_AUDIO_INFO_CHANNELS (&info);
    width = GST_AUDIO_INFO_WIDTH (&info);

    GST_DEBUG_OBJECT (self, "Configured: rate %d channels %d width %d",
        rate, channels, width);
    self->rate = rate;
    self->framesize = (width / 8) * channels;
  }

  return ret;
}
예제 #5
0
static gboolean
gst_deinterleave_set_process_function (GstDeinterleave * self)
{
  switch (GST_AUDIO_INFO_WIDTH (&self->audio_info)) {
    case 8:
      self->func = (GstDeinterleaveFunc) deinterleave_8;
      break;
    case 16:
      self->func = (GstDeinterleaveFunc) deinterleave_16;
      break;
    case 24:
      self->func = (GstDeinterleaveFunc) deinterleave_24;
      break;
    case 32:
      self->func = (GstDeinterleaveFunc) deinterleave_32;
      break;
    case 64:
      self->func = (GstDeinterleaveFunc) deinterleave_64;
      break;
    default:
      return FALSE;
  }
  return TRUE;
}
예제 #6
0
static gboolean
gst_wavenc_sink_setcaps (GstPad * pad, GstCaps * caps)
{
  GstWavEnc *wavenc;
  GstStructure *structure;
  const gchar *name;
  gint chans, rate;
  GstCaps *ccaps;

  wavenc = GST_WAVENC (gst_pad_get_parent (pad));

  ccaps = gst_pad_get_current_caps (pad);
  if (wavenc->sent_header && ccaps && !gst_caps_can_intersect (caps, ccaps)) {
    gst_caps_unref (ccaps);
    GST_WARNING_OBJECT (wavenc, "cannot change format in middle of stream");
    goto fail;
  }
  if (ccaps)
    gst_caps_unref (ccaps);

  GST_DEBUG_OBJECT (wavenc, "got caps: %" GST_PTR_FORMAT, caps);

  structure = gst_caps_get_structure (caps, 0);
  name = gst_structure_get_name (structure);

  if (!gst_structure_get_int (structure, "channels", &chans) ||
      !gst_structure_get_int (structure, "rate", &rate)) {
    GST_WARNING_OBJECT (wavenc, "caps incomplete");
    goto fail;
  }

  if (strcmp (name, "audio/x-raw") == 0) {
    GstAudioInfo info;

    if (!gst_audio_info_from_caps (&info, caps))
      goto fail;

    if (GST_AUDIO_INFO_IS_INTEGER (&info))
      wavenc->format = GST_RIFF_WAVE_FORMAT_PCM;
    else if (GST_AUDIO_INFO_IS_FLOAT (&info))
      wavenc->format = GST_RIFF_WAVE_FORMAT_IEEE_FLOAT;
    else
      goto fail;

    wavenc->width = GST_AUDIO_INFO_WIDTH (&info);
  } else if (strcmp (name, "audio/x-alaw") == 0) {
    wavenc->format = GST_RIFF_WAVE_FORMAT_ALAW;
    wavenc->width = 8;
  } else if (strcmp (name, "audio/x-mulaw") == 0) {
    wavenc->format = GST_RIFF_WAVE_FORMAT_MULAW;
    wavenc->width = 8;
  } else {
    GST_WARNING_OBJECT (wavenc, "Unsupported format %s", name);
    goto fail;
  }

  wavenc->channels = chans;
  wavenc->rate = rate;

  GST_LOG_OBJECT (wavenc,
      "accepted caps: format=0x%04x chans=%u width=%u rate=%u",
      wavenc->format, wavenc->channels, wavenc->width, wavenc->rate);

  gst_object_unref (wavenc);
  return TRUE;

fail:
  gst_object_unref (wavenc);
  return FALSE;
}
예제 #7
0
static gboolean
gst_interleave_sink_setcaps (GstInterleave * self, GstPad * pad,
    const GstCaps * caps, const GstAudioInfo * info)
{
  g_return_val_if_fail (GST_IS_INTERLEAVE_PAD (pad), FALSE);

  /* TODO: handle caps changes */
  if (self->sinkcaps && !gst_caps_is_subset (caps, self->sinkcaps)) {
    goto cannot_change_caps;
  } else {
    GstCaps *srccaps;
    GstStructure *s;
    gboolean res;

    self->width = GST_AUDIO_INFO_WIDTH (info);
    self->rate = GST_AUDIO_INFO_RATE (info);

    gst_interleave_set_process_function (self);

    srccaps = gst_caps_copy (caps);
    s = gst_caps_get_structure (srccaps, 0);

    gst_structure_remove_field (s, "channel-mask");

    gst_structure_set (s, "channels", G_TYPE_INT, self->channels, NULL);
    gst_interleave_set_channel_positions (self, s);

    gst_pad_set_active (self->src, TRUE);
    res = gst_pad_set_caps (self->src, srccaps);
    gst_caps_unref (srccaps);

    if (!res)
      goto src_did_not_accept;
  }

  if (!self->sinkcaps) {
    GstCaps *sinkcaps = gst_caps_copy (caps);
    GstStructure *s = gst_caps_get_structure (sinkcaps, 0);

    gst_structure_remove_field (s, "channel-mask");

    GST_DEBUG_OBJECT (self, "setting sinkcaps %" GST_PTR_FORMAT, sinkcaps);

    gst_caps_replace (&self->sinkcaps, sinkcaps);

    gst_caps_unref (sinkcaps);
  }

  return TRUE;

cannot_change_caps:
  {
    GST_WARNING_OBJECT (self, "caps of %" GST_PTR_FORMAT " already set, can't "
        "change", self->sinkcaps);
    return FALSE;
  }
src_did_not_accept:
  {
    GST_WARNING_OBJECT (self, "src did not accept setcaps()");
    return FALSE;
  }
}
예제 #8
0
gboolean
gst_sndio_prepare (struct gstsndio *sio, GstAudioRingBufferSpec *spec)
{
  struct sio_par par, retpar;
  unsigned nchannels;

  GST_DEBUG_OBJECT (sio, "prepare");

  if (spec->type != GST_AUDIO_RING_BUFFER_FORMAT_TYPE_RAW) {
      GST_ELEMENT_ERROR (sio, RESOURCE, OPEN_READ_WRITE,
	("Only raw buffer format supported by sndio"), (NULL));
      return FALSE;
  }
  if (!GST_AUDIO_INFO_IS_INTEGER(&spec->info)) {
      GST_ELEMENT_ERROR (sio, RESOURCE, OPEN_READ_WRITE,
	("Only integer format supported"), (NULL));
      return FALSE;
  }
  if (GST_AUDIO_INFO_DEPTH(&spec->info) % 8) {
      GST_ELEMENT_ERROR (sio, RESOURCE, OPEN_READ_WRITE,
	("Only depths multiple of 8 are supported"), (NULL));
      return FALSE;
  }

  sio_initpar (&par);
  switch (GST_AUDIO_INFO_FORMAT (&spec->info)) {
  case GST_AUDIO_FORMAT_S8:
  case GST_AUDIO_FORMAT_U8:
  case GST_AUDIO_FORMAT_S16LE:
  case GST_AUDIO_FORMAT_S16BE:
  case GST_AUDIO_FORMAT_U16LE:
  case GST_AUDIO_FORMAT_U16BE:
  case GST_AUDIO_FORMAT_S32LE:
  case GST_AUDIO_FORMAT_S32BE:
  case GST_AUDIO_FORMAT_U32LE:
  case GST_AUDIO_FORMAT_U32BE:
  case GST_AUDIO_FORMAT_S24_32LE:
  case GST_AUDIO_FORMAT_S24_32BE:
  case GST_AUDIO_FORMAT_U24_32LE:
  case GST_AUDIO_FORMAT_U24_32BE:
  case GST_AUDIO_FORMAT_S24LE:
  case GST_AUDIO_FORMAT_S24BE:
  case GST_AUDIO_FORMAT_U24LE:
  case GST_AUDIO_FORMAT_U24BE:
      break;
  default:
      GST_ELEMENT_ERROR (sio, RESOURCE, OPEN_READ_WRITE,
	  ("Unsupported audio format"),
	  ("format = %d", GST_AUDIO_INFO_FORMAT (&spec->info)));
      return FALSE;
  }
  par.sig = GST_AUDIO_INFO_IS_SIGNED(&spec->info);
  par.bits = GST_AUDIO_INFO_WIDTH(&spec->info);
  par.bps = GST_AUDIO_INFO_DEPTH(&spec->info) / 8;
  if (par.bps > 1)
      par.le = GST_AUDIO_INFO_IS_LITTLE_ENDIAN(&spec->info);
  if (par.bits < par.bps * 8)
      par.msb = 0;
  par.rate = GST_AUDIO_INFO_RATE(&spec->info);
  if (sio->mode == SIO_PLAY)
      par.pchan = GST_AUDIO_INFO_CHANNELS(&spec->info);
  else
      par.rchan = GST_AUDIO_INFO_CHANNELS(&spec->info);
  par.round = par.rate / 1000000. * spec->latency_time;
  par.appbufsz = par.rate / 1000000. * spec->buffer_time;

  if (!sio_setpar (sio->hdl, &par)) {
      GST_ELEMENT_ERROR (sio, RESOURCE, OPEN_WRITE,
	("Unsupported audio encoding"), (NULL));
      return FALSE;
  }
  if (!sio_getpar (sio->hdl, &retpar)) {
      GST_ELEMENT_ERROR (sio, RESOURCE, OPEN_WRITE,
	("Couldn't get audio device parameters"), (NULL));
      return FALSE;
  }
#if 0
  fprintf(stderr, "format = %s, "
         "requested: sig = %d, bits = %d, bps = %d, le = %d, msb = %d, "
	 "rate = %d, pchan = %d, round = %d, appbufsz = %d; "
	 "returned: sig = %d, bits = %d, bps = %d, le = %d, msb = %d, "
	 "rate = %d, pchan = %d, round = %d, appbufsz = %d, bufsz = %d\n",
	 GST_AUDIO_INFO_NAME(&spec->info),
	 par.sig, par.bits, par.bps, par.le, par.msb,
	 par.rate, par.pchan, par.round, par.appbufsz,
	 retpar.sig, retpar.bits, retpar.bps, retpar.le, retpar.msb,
	 retpar.rate, retpar.pchan, retpar.round, retpar.appbufsz, retpar.bufsz);
#endif
  if (par.bits != retpar.bits ||
      par.bps != retpar.bps ||
      par.rate != retpar.rate ||
      (sio->mode == SIO_PLAY && par.pchan != retpar.pchan) ||
      (sio->mode == SIO_REC && par.rchan != retpar.rchan) ||
      (par.bps > 1 && par.le != retpar.le) ||
      (par.bits < par.bps * 8 && par.msb != retpar.msb)) {
      GST_ELEMENT_ERROR (sio, RESOURCE, OPEN_WRITE,
	("Audio device refused requested parameters"), (NULL));
      return FALSE;
  }

  nchannels = (sio->mode == SIO_PLAY) ? retpar.pchan : retpar.rchan;
  spec->segsize = retpar.round * retpar.bps * nchannels;
  spec->segtotal = retpar.bufsz / retpar.round;
  sio->bpf = retpar.bps * nchannels;
  sio->delay = 0;
  sio_onmove (sio->hdl, gst_sndio_cb, sio);

  if (!sio_start (sio->hdl)) {
    GST_ELEMENT_ERROR (sio->obj, RESOURCE, OPEN_READ_WRITE,
      ("Could not start sndio"), (NULL));
    return FALSE;
  }
  return TRUE;
}
예제 #9
0
static FLAC__StreamDecoderWriteStatus
gst_flac_dec_write (GstFlacDec * flacdec, const FLAC__Frame * frame,
    const FLAC__int32 * const buffer[])
{
  GstFlowReturn ret = GST_FLOW_OK;
  GstBuffer *outbuf;
  guint depth = frame->header.bits_per_sample;
  guint width, gdepth;
  guint sample_rate = frame->header.sample_rate;
  guint channels = frame->header.channels;
  guint samples = frame->header.blocksize;
  guint j, i;
  GstMapInfo map;
  gboolean caps_changed;

  GST_LOG_OBJECT (flacdec, "samples in frame header: %d", samples);

  if (depth == 0) {
    if (flacdec->depth < 4 || flacdec->depth > 32) {
      GST_ERROR_OBJECT (flacdec, "unsupported depth %d from STREAMINFO",
          flacdec->depth);
      ret = GST_FLOW_ERROR;
      goto done;
    }

    depth = flacdec->depth;
  }

  switch (depth) {
    case 8:
      gdepth = width = 8;
      break;
    case 12:
    case 16:
      gdepth = width = 16;
      break;
    case 20:
    case 24:
      gdepth = 24;
      width = 32;
      break;
    case 32:
      gdepth = width = 32;
      break;
    default:
      GST_ERROR_OBJECT (flacdec, "unsupported depth %d", depth);
      ret = GST_FLOW_ERROR;
      goto done;
  }

  if (sample_rate == 0) {
    if (flacdec->info.rate != 0) {
      sample_rate = flacdec->info.rate;
    } else {
      GST_ERROR_OBJECT (flacdec, "unknown sample rate");
      ret = GST_FLOW_ERROR;
      goto done;
    }
  }

  caps_changed = (sample_rate != GST_AUDIO_INFO_RATE (&flacdec->info))
      || (width != GST_AUDIO_INFO_WIDTH (&flacdec->info))
      || (gdepth != GST_AUDIO_INFO_DEPTH (&flacdec->info))
      || (channels != GST_AUDIO_INFO_CHANNELS (&flacdec->info));

  if (caps_changed
      || !gst_pad_has_current_caps (GST_AUDIO_DECODER_SRC_PAD (flacdec))) {
    GST_DEBUG_OBJECT (flacdec, "Negotiating %d Hz @ %d channels", sample_rate,
        channels);

    gst_audio_info_set_format (&flacdec->info,
        gst_audio_format_build_integer (TRUE, G_BYTE_ORDER, width, gdepth),
        sample_rate, channels, NULL);

    memcpy (flacdec->info.position,
        channel_positions[flacdec->info.channels - 1],
        sizeof (GstAudioChannelPosition) * flacdec->info.channels);
    gst_audio_channel_positions_to_valid_order (flacdec->info.position,
        flacdec->info.channels);
    /* Note: we create the inverse reordering map here */
    gst_audio_get_channel_reorder_map (flacdec->info.channels,
        flacdec->info.position, channel_positions[flacdec->info.channels - 1],
        flacdec->channel_reorder_map);

    flacdec->depth = depth;

    gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (flacdec),
        &flacdec->info);
  }

  outbuf =
      gst_buffer_new_allocate (NULL, samples * channels * (width / 8), NULL);

  gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
  if (width == 8) {
    gint8 *outbuffer = (gint8 *) map.data;
    gint *reorder_map = flacdec->channel_reorder_map;

    if (gdepth != depth) {
      for (i = 0; i < samples; i++) {
        for (j = 0; j < channels; j++) {
          *outbuffer++ =
              (gint8) (buffer[reorder_map[j]][i] << (gdepth - depth));
        }
      }
    } else {
      for (i = 0; i < samples; i++) {
        for (j = 0; j < channels; j++) {
          *outbuffer++ = (gint8) buffer[reorder_map[j]][i];
        }
      }
    }
  } else if (width == 16) {
    gint16 *outbuffer = (gint16 *) map.data;
    gint *reorder_map = flacdec->channel_reorder_map;

    if (gdepth != depth) {
      for (i = 0; i < samples; i++) {
        for (j = 0; j < channels; j++) {
          *outbuffer++ =
              (gint16) (buffer[reorder_map[j]][i] << (gdepth - depth));
        }
      }
    } else {
      for (i = 0; i < samples; i++) {
        for (j = 0; j < channels; j++) {
          *outbuffer++ = (gint16) buffer[reorder_map[j]][i];
        }
      }
    }
  } else if (width == 32) {
    gint32 *outbuffer = (gint32 *) map.data;
    gint *reorder_map = flacdec->channel_reorder_map;

    if (gdepth != depth) {
      for (i = 0; i < samples; i++) {
        for (j = 0; j < channels; j++) {
          *outbuffer++ =
              (gint32) (buffer[reorder_map[j]][i] << (gdepth - depth));
        }
      }
    } else {
      for (i = 0; i < samples; i++) {
        for (j = 0; j < channels; j++) {
          *outbuffer++ = (gint32) buffer[reorder_map[j]][i];
        }
      }
    }
  } else {
    g_assert_not_reached ();
  }
  gst_buffer_unmap (outbuf, &map);

  GST_DEBUG_OBJECT (flacdec, "pushing %d samples", samples);
  if (flacdec->error_count)
    flacdec->error_count--;

  ret = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (flacdec), outbuf, 1);

  if (G_UNLIKELY (ret != GST_FLOW_OK)) {
    GST_DEBUG_OBJECT (flacdec, "finish_frame flow %s", gst_flow_get_name (ret));
  }

done:

  /* we act on the flow return value later in the handle_frame function, as we
   * don't want to mess up the internal decoder state by returning ABORT when
   * the error is in fact non-fatal (like a pad in flushing mode) and we want
   * to continue later. So just pretend everything's dandy and act later. */
  flacdec->last_flow = ret;

  return FLAC__STREAM_DECODER_WRITE_STATUS_CONTINUE;
}
예제 #10
0
static GstFlowReturn
gst_deinterleave_process (GstDeinterleave * self, GstBuffer * buf)
{
  GstFlowReturn ret = GST_FLOW_OK;
  guint channels = GST_AUDIO_INFO_CHANNELS (&self->audio_info);
  guint pads_pushed = 0, buffers_allocated = 0;
  guint nframes =
      gst_buffer_get_size (buf) / channels /
      (GST_AUDIO_INFO_WIDTH (&self->audio_info) / 8);
  guint bufsize = nframes * (GST_AUDIO_INFO_WIDTH (&self->audio_info) / 8);
  guint i;
  GList *srcs;
  GstBuffer **buffers_out = g_new0 (GstBuffer *, channels);
  guint8 *in, *out;
  GstMapInfo read_info;
  GList *pending_events, *l;

  /* Send any pending events to all src pads */
  GST_OBJECT_LOCK (self);
  pending_events = self->pending_events;
  self->pending_events = NULL;
  GST_OBJECT_UNLOCK (self);

  if (pending_events) {
    GstEvent *event;

    GST_DEBUG_OBJECT (self, "Sending pending events to all src pads");
    for (l = pending_events; l; l = l->next) {
      event = l->data;
      for (srcs = self->srcpads; srcs != NULL; srcs = srcs->next)
        gst_pad_push_event (GST_PAD (srcs->data), gst_event_ref (event));
      gst_event_unref (event);
    }
    g_list_free (pending_events);
  }

  gst_buffer_map (buf, &read_info, GST_MAP_READ);

  /* Allocate buffers */
  for (srcs = self->srcpads, i = 0; srcs; srcs = srcs->next, i++) {
    buffers_out[i] = gst_buffer_new_allocate (NULL, bufsize, NULL);

    /* Make sure we got a correct buffer. The only other case we allow
     * here is an unliked pad */
    if (!buffers_out[i])
      goto alloc_buffer_failed;
    else if (buffers_out[i] && gst_buffer_get_size (buffers_out[i]) != bufsize)
      goto alloc_buffer_bad_size;

    if (buffers_out[i]) {
      gst_buffer_copy_into (buffers_out[i], buf, GST_BUFFER_COPY_METADATA, 0,
          -1);
      buffers_allocated++;
    }
  }

  /* Return NOT_LINKED if no pad was linked */
  if (!buffers_allocated) {
    GST_WARNING_OBJECT (self,
        "Couldn't allocate any buffers because no pad was linked");
    ret = GST_FLOW_NOT_LINKED;
    goto done;
  }

  /* deinterleave */
  for (srcs = self->srcpads, i = 0; srcs; srcs = srcs->next, i++) {
    GstPad *pad = (GstPad *) srcs->data;
    GstMapInfo write_info;

    in = (guint8 *) read_info.data;
    in += i * (GST_AUDIO_INFO_WIDTH (&self->audio_info) / 8);
    if (buffers_out[i]) {
      gst_buffer_map (buffers_out[i], &write_info, GST_MAP_WRITE);
      out = (guint8 *) write_info.data;
      self->func (out, in, channels, nframes);
      gst_buffer_unmap (buffers_out[i], &write_info);

      ret = gst_pad_push (pad, buffers_out[i]);
      buffers_out[i] = NULL;
      if (ret == GST_FLOW_OK)
        pads_pushed++;
      else if (ret == GST_FLOW_NOT_LINKED)
        ret = GST_FLOW_OK;
      else
        goto push_failed;
    }
  }

  /* Return NOT_LINKED if no pad was linked */
  if (!pads_pushed)
    ret = GST_FLOW_NOT_LINKED;

  GST_DEBUG_OBJECT (self, "Pushed on %d pads", pads_pushed);

done:
  gst_buffer_unmap (buf, &read_info);
  gst_buffer_unref (buf);
  g_free (buffers_out);
  return ret;

alloc_buffer_failed:
  {
    GST_WARNING ("gst_pad_alloc_buffer() returned %s", gst_flow_get_name (ret));
    goto clean_buffers;

  }
alloc_buffer_bad_size:
  {
    GST_WARNING ("called alloc_buffer(), but didn't get requested bytes");
    ret = GST_FLOW_NOT_NEGOTIATED;
    goto clean_buffers;
  }
push_failed:
  {
    GST_DEBUG ("push() failed, flow = %s", gst_flow_get_name (ret));
    goto clean_buffers;
  }
clean_buffers:
  {
    gst_buffer_unmap (buf, &read_info);
    for (i = 0; i < channels; i++) {
      if (buffers_out[i])
        gst_buffer_unref (buffers_out[i]);
    }
    gst_buffer_unref (buf);
    g_free (buffers_out);
    return ret;
  }
}
static gboolean
gst_osx_audio_ring_buffer_acquire (GstAudioRingBuffer * buf,
    GstAudioRingBufferSpec * spec)
{
  gboolean ret = FALSE, is_passthrough = FALSE;
  GstOsxAudioRingBuffer *osxbuf;
  AudioStreamBasicDescription format;

  osxbuf = GST_OSX_AUDIO_RING_BUFFER (buf);

  if (RINGBUFFER_IS_SPDIF (spec->type)) {
    format.mFormatID = kAudioFormat60958AC3;
    format.mSampleRate = (double) GST_AUDIO_INFO_RATE (&spec->info);
    format.mChannelsPerFrame = 2;
    format.mFormatFlags = kAudioFormatFlagIsSignedInteger |
        kAudioFormatFlagIsPacked | kAudioFormatFlagIsNonMixable;
    format.mBytesPerFrame = 0;
    format.mBitsPerChannel = 16;
    format.mBytesPerPacket = 6144;
    format.mFramesPerPacket = 1536;
    format.mReserved = 0;
    spec->segsize = 6144;
    spec->segtotal = 10;
    is_passthrough = TRUE;
  } else {
    int width, depth;
    /* Fill out the audio description we're going to be using */
    format.mFormatID = kAudioFormatLinearPCM;
    format.mSampleRate = (double) GST_AUDIO_INFO_RATE (&spec->info);
    format.mChannelsPerFrame = GST_AUDIO_INFO_CHANNELS (&spec->info);
    if (GST_AUDIO_INFO_IS_FLOAT (&spec->info)) {
      format.mFormatFlags = kAudioFormatFlagsNativeFloatPacked;
      width = depth = GST_AUDIO_INFO_WIDTH (&spec->info);
    } else {
      format.mFormatFlags = kAudioFormatFlagIsSignedInteger;
      width = GST_AUDIO_INFO_WIDTH (&spec->info);
      depth = GST_AUDIO_INFO_DEPTH (&spec->info);
      if (width == depth) {
        format.mFormatFlags |= kAudioFormatFlagIsPacked;
      } else {
        format.mFormatFlags |= kAudioFormatFlagIsAlignedHigh;
      }
    }

    if (GST_AUDIO_INFO_IS_BIG_ENDIAN (&spec->info)) {
      format.mFormatFlags |= kAudioFormatFlagIsBigEndian;
    }

    format.mBytesPerFrame = GST_AUDIO_INFO_BPF (&spec->info);
    format.mBitsPerChannel = depth;
    format.mBytesPerPacket = GST_AUDIO_INFO_BPF (&spec->info);
    format.mFramesPerPacket = 1;
    format.mReserved = 0;
    spec->segsize =
        (spec->latency_time * GST_AUDIO_INFO_RATE (&spec->info) /
        G_USEC_PER_SEC) * GST_AUDIO_INFO_BPF (&spec->info);
    spec->segtotal = spec->buffer_time / spec->latency_time;
    is_passthrough = FALSE;
  }

  GST_DEBUG_OBJECT (osxbuf, "Format: " CORE_AUDIO_FORMAT,
      CORE_AUDIO_FORMAT_ARGS (format));

  /* gst_audio_ring_buffer_set_channel_positions is not called
   * since the AUs perform channel reordering themselves.
   * (see gst_core_audio_set_channel_layout) */

  buf->size = spec->segtotal * spec->segsize;
  buf->memory = g_malloc0 (buf->size);

  ret = gst_core_audio_initialize (osxbuf->core_audio, format, spec->caps,
      is_passthrough);

  if (!ret) {
    g_free (buf->memory);
    buf->memory = NULL;
    buf->size = 0;
  }

  osxbuf->segoffset = 0;

  return ret;
}