/** * gst_rtp_base_audio_payload_push: * @baseaudiopayload: a #GstRTPBasePayload * @data: data to set as payload * @payload_len: length of payload * @timestamp: a #GstClockTime * * Create an RTP buffer and store @payload_len bytes of @data as the * payload. Set the timestamp on the new buffer to @timestamp before pushing * the buffer downstream. * * Returns: a #GstFlowReturn */ GstFlowReturn gst_rtp_base_audio_payload_push (GstRTPBaseAudioPayload * baseaudiopayload, const guint8 * data, guint payload_len, GstClockTime timestamp) { GstRTPBasePayload *basepayload; GstBuffer *outbuf; guint8 *payload; GstFlowReturn ret; GstRTPBuffer rtp = { NULL }; basepayload = GST_RTP_BASE_PAYLOAD (baseaudiopayload); GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT, payload_len, GST_TIME_ARGS (timestamp)); /* create buffer to hold the payload */ outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload, payload_len, 0, 0); /* copy payload */ gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp); payload = gst_rtp_buffer_get_payload (&rtp); memcpy (payload, data, payload_len); gst_rtp_buffer_unmap (&rtp); /* set metadata */ gst_rtp_base_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len, timestamp); ret = gst_rtp_base_payload_push (basepayload, outbuf); return ret; }
static GstFlowReturn gst_rtp_base_audio_payload_push_buffer (GstRTPBaseAudioPayload * baseaudiopayload, GstBuffer * buffer, GstClockTime timestamp) { GstRTPBasePayload *basepayload; GstRTPBaseAudioPayloadPrivate *priv; GstBuffer *outbuf; guint payload_len; GstFlowReturn ret; priv = baseaudiopayload->priv; basepayload = GST_RTP_BASE_PAYLOAD (baseaudiopayload); payload_len = gst_buffer_get_size (buffer); GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT, payload_len, GST_TIME_ARGS (timestamp)); /* create just the RTP header buffer */ outbuf = gst_rtp_base_payload_allocate_output_buffer (basepayload, 0, 0, 0); /* set metadata */ gst_rtp_base_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len, timestamp); if (priv->buffer_list) { GstBufferList *list; guint i, len; list = gst_buffer_list_new (); len = gst_buffer_list_length (list); for (i = 0; i < len; i++) { /* FIXME */ g_warning ("bufferlist not implemented"); gst_buffer_list_add (list, outbuf); gst_buffer_list_add (list, buffer); } GST_DEBUG_OBJECT (baseaudiopayload, "Pushing list %p", list); ret = gst_rtp_base_payload_push_list (basepayload, list); } else { CopyMetaData data; /* copy payload */ data.pay = baseaudiopayload; data.outbuf = outbuf; gst_buffer_foreach_meta (buffer, foreach_metadata, &data); outbuf = gst_buffer_append (outbuf, buffer); GST_DEBUG_OBJECT (baseaudiopayload, "Pushing buffer %p", outbuf); ret = gst_rtp_base_payload_push (basepayload, outbuf); } return ret; }
/** * 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; }
static GstFlowReturn gst_rtp_base_audio_payload_push_buffer (GstRTPBaseAudioPayload * baseaudiopayload, GstBuffer * buffer, GstClockTime timestamp) { GstRTPBasePayload *basepayload; GstRTPBaseAudioPayloadPrivate *priv; GstBuffer *outbuf; guint8 *payload; guint payload_len; GstFlowReturn ret; priv = baseaudiopayload->priv; basepayload = GST_RTP_BASE_PAYLOAD (baseaudiopayload); payload_len = gst_buffer_get_size (buffer); GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT, payload_len, GST_TIME_ARGS (timestamp)); if (priv->buffer_list) { /* create just the RTP header buffer */ outbuf = gst_rtp_buffer_new_allocate (0, 0, 0); } else { /* create buffer to hold the payload */ outbuf = gst_rtp_buffer_new_allocate (payload_len, 0, 0); } /* set metadata */ gst_rtp_base_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len, timestamp); if (priv->buffer_list) { GstBufferList *list; guint i, len; list = gst_buffer_list_new (); len = gst_buffer_list_length (list); for (i = 0; i < len; i++) { /* FIXME */ g_warning ("bufferlist not implemented"); gst_buffer_list_add (list, outbuf); gst_buffer_list_add (list, buffer); } GST_DEBUG_OBJECT (baseaudiopayload, "Pushing list %p", list); ret = gst_rtp_base_payload_push_list (basepayload, list); } else { GstRTPBuffer rtp = { NULL }; /* copy payload */ gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp); payload = gst_rtp_buffer_get_payload (&rtp); gst_buffer_extract (buffer, 0, payload, payload_len); gst_rtp_buffer_unmap (&rtp); gst_buffer_unref (buffer); GST_DEBUG_OBJECT (baseaudiopayload, "Pushing buffer %p", outbuf); ret = gst_rtp_base_payload_push (basepayload, outbuf); } return ret; }