mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2025-03-30 12:49:40 +00:00
Code Issues 1 Projects Releases Wiki Activity

rtpsbcpay: fix if buffer size exceeds MTU

Browse source 
The plugin queued buffer data if not all buffer data fit
into a single RTP packet. Now RTP packets are pushed as long
as enough data is available.
This commit is contained in:
Jochen Henneberg 2017-08-14 10:36:56 +00:00 committed by Arun Raghavan
parent 36fc2a747a
commit f641ac60e3
1 changed files with 35 additions and 28 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

63
gst/rtp/gstrtpsbcpay.c
View file

@ -178,51 +178,58 @@ gst_rtp_sbc_pay_flush_buffers (GstRtpSBCPay * sbcpay)
guint frame_count;
guint payload_length;
struct rtp_payload *payload;
GstFlowReturn res;
if (sbcpay->frame_length == 0) {
GST_ERROR_OBJECT (sbcpay, "Frame length is 0");
return GST_FLOW_ERROR;
}
available = gst_adapter_available (sbcpay->adapter);
do {
available = gst_adapter_available (sbcpay->adapter);
max_payload =
gst_rtp_buffer_calc_payload_len (GST_RTP_BASE_PAYLOAD_MTU (sbcpay) -
RTP_SBC_PAYLOAD_HEADER_SIZE, 0, 0);
max_payload =
gst_rtp_buffer_calc_payload_len (GST_RTP_BASE_PAYLOAD_MTU (sbcpay) -
RTP_SBC_PAYLOAD_HEADER_SIZE, 0, 0);
max_payload = MIN (max_payload, available);
frame_count = max_payload / sbcpay->frame_length;
payload_length = frame_count * sbcpay->frame_length;
if (payload_length == 0) /* Nothing to send */
return GST_FLOW_OK;
max_payload = MIN (max_payload, available);
frame_count = max_payload / sbcpay->frame_length;
payload_length = frame_count * sbcpay->frame_length;
if (payload_length == 0) /* Nothing to send */
return GST_FLOW_OK;
outbuf = gst_rtp_buffer_new_allocate (RTP_SBC_PAYLOAD_HEADER_SIZE, 0, 0);
outbuf = gst_rtp_buffer_new_allocate (RTP_SBC_PAYLOAD_HEADER_SIZE, 0, 0);
/* get payload */
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
/* get payload */
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
gst_rtp_buffer_set_payload_type (&rtp, GST_RTP_BASE_PAYLOAD_PT (sbcpay));
gst_rtp_buffer_set_payload_type (&rtp, GST_RTP_BASE_PAYLOAD_PT (sbcpay));
/* write header and copy data into payload */
payload_data = gst_rtp_buffer_get_payload (&rtp);
payload = (struct rtp_payload *) payload_data;
memset (payload, 0, sizeof (struct rtp_payload));
payload->frame_count = frame_count;
/* write header and copy data into payload */
payload_data = gst_rtp_buffer_get_payload (&rtp);
payload = (struct rtp_payload *) payload_data;
memset (payload, 0, sizeof (struct rtp_payload));
payload->frame_count = frame_count;
gst_rtp_buffer_unmap (&rtp);
gst_rtp_buffer_unmap (&rtp);
paybuf = gst_adapter_take_buffer_fast (sbcpay->adapter, payload_length);
gst_rtp_copy_audio_meta (sbcpay, outbuf, paybuf);
outbuf = gst_buffer_append (outbuf, paybuf);
paybuf = gst_adapter_take_buffer_fast (sbcpay->adapter, payload_length);
gst_rtp_copy_audio_meta (sbcpay, outbuf, paybuf);
outbuf = gst_buffer_append (outbuf, paybuf);
GST_BUFFER_PTS (outbuf) = sbcpay->last_timestamp;
GST_BUFFER_DURATION (outbuf) = frame_count * sbcpay->frame_duration;
GST_DEBUG_OBJECT (sbcpay, "Pushing %d bytes: %" GST_TIME_FORMAT,
payload_length, GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)));
GST_BUFFER_PTS (outbuf) = sbcpay->last_timestamp;
GST_BUFFER_DURATION (outbuf) = frame_count * sbcpay->frame_duration;
GST_DEBUG_OBJECT (sbcpay, "Pushing %d bytes: %" GST_TIME_FORMAT,
payload_length, GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)));
sbcpay->last_timestamp += frame_count * sbcpay->frame_duration;
sbcpay->last_timestamp += frame_count * sbcpay->frame_duration;
return gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (sbcpay), outbuf);
res = gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (sbcpay), outbuf);
/* try to send another RTP buffer if available data exceeds MTU size */
} while (res == GST_FLOW_OK)
return res;
}
static GstFlowReturn