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rtpbaseaudiopayload: Don't copy memory if not needed, just append payload to the RTP buffer

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This commit is contained in:
Sebastian Dröge 2015-06-30 10:37:27 +02:00
parent 49d0083456
commit 353fb82ea1
1 changed files with 3 additions and 16 deletions
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19
gst-libs/gst/rtp/gstrtpbaseaudiopayload.c
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@ -484,7 +484,6 @@ gst_rtp_base_audio_payload_push_buffer (GstRTPBaseAudioPayload *
GstRTPBasePayload *basepayload; GstRTPBasePayload *basepayload;
GstRTPBaseAudioPayloadPrivate *priv; GstRTPBaseAudioPayloadPrivate *priv;
GstBuffer *outbuf; GstBuffer *outbuf;
guint8 *payload;
guint payload_len; guint payload_len;
GstFlowReturn ret; GstFlowReturn ret;
@ -496,13 +495,8 @@ gst_rtp_base_audio_payload_push_buffer (GstRTPBaseAudioPayload *
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT, GST_DEBUG_OBJECT (baseaudiopayload, "Pushing %d bytes ts %" GST_TIME_FORMAT,
payload_len, GST_TIME_ARGS (timestamp)); payload_len, GST_TIME_ARGS (timestamp));
if (priv->buffer_list) { /* create just the RTP header buffer */
/* create just the RTP header buffer */ outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
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 */ /* set metadata */
gst_rtp_base_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len, gst_rtp_base_audio_payload_set_meta (baseaudiopayload, outbuf, payload_len,
@ -525,15 +519,8 @@ gst_rtp_base_audio_payload_push_buffer (GstRTPBaseAudioPayload *
GST_DEBUG_OBJECT (baseaudiopayload, "Pushing list %p", list); GST_DEBUG_OBJECT (baseaudiopayload, "Pushing list %p", list);
ret = gst_rtp_base_payload_push_list (basepayload, list); ret = gst_rtp_base_payload_push_list (basepayload, list);
} else { } else {
GstRTPBuffer rtp = { NULL };
/* copy payload */ /* copy payload */
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp); outbuf = gst_buffer_append (outbuf, buffer);
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); GST_DEBUG_OBJECT (baseaudiopayload, "Pushing buffer %p", outbuf);
ret = gst_rtp_base_payload_push (basepayload, outbuf); ret = gst_rtp_base_payload_push (basepayload, outbuf);