static gboolean
get_next_chunk (GstAdapter *adapter, Chunk *chunk)
{
const guint8 *data;
guint length;
data = gst_adapter_peek (adapter, 8);
if (!data) return FALSE;
length = GST_READ_UINT32_BE (data + 4);
chunk->fourcc = GST_READ_UINT32_LE (data);
chunk->length = length;
gst_adapter_flush (adapter, 8);
chunk->data = gst_adapter_peek (adapter, gst_adapter_available_fast (adapter));
chunk->available = gst_adapter_available_fast (adapter);
return TRUE;
}
/**
* gst_rtp_base_audio_payload_flush:
* @baseaudiopayload: a #GstRTPBasePayload
* @payload_len: length of payload
* @timestamp: a #GstClockTime
*
* Create an RTP buffer and store @payload_len bytes of the adapter as the
* payload. Set the timestamp on the new buffer to @timestamp before pushing
* the buffer downstream.
*
* If @payload_len is -1, all pending bytes will be flushed. If @timestamp is
* -1, the timestamp will be calculated automatically.
*
* Returns: a #GstFlowReturn
*/
GstFlowReturn
gst_rtp_base_audio_payload_flush (GstRTPBaseAudioPayload * baseaudiopayload,
guint payload_len, GstClockTime timestamp)
{
GstRTPBasePayload *basepayload;
GstRTPBaseAudioPayloadPrivate *priv;
GstBuffer *outbuf;
GstFlowReturn ret;
GstAdapter *adapter;
guint64 distance;
priv = baseaudiopayload->priv;
adapter = priv->adapter;
basepayload = GST_RTP_BASE_PAYLOAD (baseaudiopayload);
if (payload_len == -1)
payload_len = gst_adapter_available (adapter);
/* nothing to do, just return */
if (payload_len == 0)
return GST_FLOW_OK;
if (timestamp == -1) {
/* calculate the timestamp */
timestamp = gst_adapter_prev_pts (adapter, &distance);
GST_LOG_OBJECT (baseaudiopayload,
"last timestamp %" GST_TIME_FORMAT ", distance %" G_GUINT64_FORMAT,
GST_TIME_ARGS (timestamp), distance);
if (GST_CLOCK_TIME_IS_VALID (timestamp) && distance > 0) {
/* convert the number of bytes since the last timestamp to time and add to
* the last seen timestamp */
timestamp += priv->bytes_to_time (baseaudiopayload, distance);
}
}
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT,
payload_len, GST_TIME_ARGS (timestamp));
if (priv->buffer_list && gst_adapter_available_fast (adapter) >= payload_len) {
GstBuffer *buffer;
/* we can quickly take a buffer out of the adapter without having to copy
* anything. */
buffer = gst_adapter_take_buffer (adapter, payload_len);
ret =
gst_rtp_base_audio_payload_push_buffer (baseaudiopayload, buffer,
timestamp);
} else {
GstBuffer *paybuf;
CopyMetaData data;
/* create buffer to hold the payload */
outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload, 0, 0, 0);
paybuf = gst_adapter_take_buffer_fast (adapter, payload_len);
data.pay = baseaudiopayload;
data.outbuf = outbuf;
gst_buffer_foreach_meta (paybuf, foreach_metadata, &data);
outbuf = gst_buffer_append (outbuf, paybuf);
/* set metadata */
gst_rtp_base_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
timestamp);
ret = gst_rtp_base_payload_push (basepayload, outbuf);
}
return ret;
}
