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Consolidate GetFrame and InsertPacket and move NACK list processing to after a packet has been successfully inserted.

Browse Source 
TEST=trybots
BUG=1799
R=mikhal@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1509004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4080 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
stefan@webrtc.org
2013-05-21 15:25:53 +00:00
parent 956aa7e087
commit 3417eb49f6
9 changed files with 311 additions and 394 deletions
Download Patch File Download Diff File Expand all files Collapse all files

86
webrtc/modules/video_coding/main/source/receiver.cc
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@ -77,66 +77,36 @@ void VCMReceiver::UpdateRtt(uint32_t rtt) {
int32_t VCMReceiver::InsertPacket(const VCMPacket& packet,
uint16_t frame_width,
uint16_t frame_height) {
// Find an empty frame.
VCMEncodedFrame* buffer = NULL;
const int32_t error = jitter_buffer_.GetFrame(packet, buffer);
if (error == VCM_OLD_PACKET_ERROR) {
// Insert the packet into the jitter buffer. The packet can either be empty or
// contain media at this point.
bool retransmitted = false;
const VCMFrameBufferEnum ret = jitter_buffer_.InsertPacket(packet,
&retransmitted);
if (ret == kOldPacket) {
return VCM_OK;
} else if (error != VCM_OK) {
return error;
} else if (ret == kFlushIndicator) {
return VCM_FLUSH_INDICATOR;
} else if (ret < 0) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCoding,
VCMId(vcm_id_, receiver_id_),
"Error inserting packet seqnum=%u, timestamp=%u",
packet.seqNum, packet.timestamp);
return VCM_JITTER_BUFFER_ERROR;
}
assert(buffer);
{
CriticalSectionScoped cs(crit_sect_);
if (frame_width && frame_height) {
buffer->SetEncodedSize(static_cast<uint32_t>(frame_width),
static_cast<uint32_t>(frame_height));
}
if (master_) {
// Only trace the primary receiver to make it possible to parse and plot
// the trace file.
WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding,
VCMId(vcm_id_, receiver_id_),
"Packet seq_no %u of frame %u at %u",
packet.seqNum, packet.timestamp,
MaskWord64ToUWord32(clock_->TimeInMilliseconds()));
}
// First packet received belonging to this frame.
if (buffer->Length() == 0 && master_) {
const int64_t now_ms = clock_->TimeInMilliseconds();
// Only trace the primary receiver to make it possible to parse and plot
// the trace file.
WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding,
VCMId(vcm_id_, receiver_id_),
"First packet of frame %u at %u", packet.timestamp,
MaskWord64ToUWord32(now_ms));
}
// Insert packet into the jitter buffer both media and empty packets.
const VCMFrameBufferEnum
ret = jitter_buffer_.InsertPacket(buffer, packet);
if (ret == kCompleteSession) {
bool retransmitted = false;
const int64_t last_packet_time_ms =
jitter_buffer_.LastPacketTime(buffer, &retransmitted);
if (last_packet_time_ms >= 0 && !retransmitted) {
// We don't want to include timestamps which have suffered from
// retransmission here, since we compensate with extra retransmission
// delay within the jitter estimate.
timing_->IncomingTimestamp(packet.timestamp, last_packet_time_ms);
}
}
if (ret == kFlushIndicator) {
return VCM_FLUSH_INDICATOR;
} else if (ret < 0) {
WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCoding,
VCMId(vcm_id_, receiver_id_),
"Error inserting packet seq_no=%u, time_stamp=%u",
packet.seqNum, packet.timestamp);
return VCM_JITTER_BUFFER_ERROR;
}
if (ret == kCompleteSession && !retransmitted) {
// We don't want to include timestamps which have suffered from
// retransmission here, since we compensate with extra retransmission
// delay within the jitter estimate.
timing_->IncomingTimestamp(packet.timestamp, clock_->TimeInMilliseconds());
}
if (master_) {
// Only trace the primary receiver to make it possible to parse and plot
// the trace file.
WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding,
VCMId(vcm_id_, receiver_id_),
"Packet seqnum=%u timestamp=%u inserted at %u",
packet.seqNum, packet.timestamp,
MaskWord64ToUWord32(clock_->TimeInMilliseconds()));
}
return VCM_OK;
}