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Update the remote bitrate estimator before passing the packet to the RTP module.

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This solves the problem of reconstructed packets biasing the bandwidth estimate.

TEST=vie_auto_test --automated, trybots
R=mflodman@webrtc.org, solenberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1594005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4171 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
stefan@webrtc.org 2013-06-04 12:15:40 +00:00
parent 6998c8ef7a
commit de98478965
6 changed files with 19 additions and 22 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h
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@ -55,7 +55,7 @@ class RemoteBitrateEstimator : public CallStatsObserver, public Module {
// the WebRtcRTPHeader to be initialized.
virtual void IncomingPacket(int64_t arrival_time_ms,
int payload_size,
const WebRtcRTPHeader& header) = 0;
const RTPHeader& header) = 0;
// Removes all data for |ssrc|.
virtual void RemoveStream(unsigned int ssrc) = 0;

10
webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_multi_stream.cc
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@ -44,7 +44,7 @@ class RemoteBitrateEstimatorMultiStream : public RemoteBitrateEstimator {
// Note that |payload_size| is the packet size excluding headers.
virtual void IncomingPacket(int64_t arrival_time_ms,
int payload_size,
const WebRtcRTPHeader& header);
const RTPHeader& header);
// Triggers a new estimate calculation.
// Implements the Module interface.
@ -139,10 +139,10 @@ void RemoteBitrateEstimatorMultiStream::IncomingRtcp(unsigned int ssrc,
void RemoteBitrateEstimatorMultiStream::IncomingPacket(
int64_t arrival_time_ms,
int payload_size,
const WebRtcRTPHeader& header) {
uint32_t ssrc = header.header.ssrc;
uint32_t rtp_timestamp = header.header.timestamp +
header.header.extension.transmissionTimeOffset;
const RTPHeader& header) {
uint32_t ssrc = header.ssrc;
uint32_t rtp_timestamp = header.timestamp +
header.extension.transmissionTimeOffset;
CriticalSectionScoped cs(crit_sect_.get());
incoming_bitrate_.Update(payload_size, arrival_time_ms);
// Add this stream to the map of streams if it doesn't already exist.

10
webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
View File

@ -37,7 +37,7 @@ class RemoteBitrateEstimatorSingleStream : public RemoteBitrateEstimator {
// packet size excluding headers.
virtual void IncomingPacket(int64_t arrival_time_ms,
int payload_size,
const WebRtcRTPHeader& header);
const RTPHeader& header);
// Triggers a new estimate calculation.
// Implements the Module interface.
@ -86,10 +86,10 @@ RemoteBitrateEstimatorSingleStream::RemoteBitrateEstimatorSingleStream(
void RemoteBitrateEstimatorSingleStream::IncomingPacket(
int64_t arrival_time_ms,
int payload_size,
const WebRtcRTPHeader& header) {
uint32_t ssrc = header.header.ssrc;
uint32_t rtp_timestamp = header.header.timestamp +
header.header.extension.transmissionTimeOffset;
const RTPHeader& header) {
uint32_t ssrc = header.ssrc;
uint32_t rtp_timestamp = header.timestamp +
header.extension.transmissionTimeOffset;
CriticalSectionScoped cs(crit_sect_.get());
SsrcOveruseDetectorMap::iterator it = overuse_detectors_.find(ssrc);
if (it == overuse_detectors_.end()) {

8
webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc
View File

@ -223,11 +223,11 @@ void RemoteBitrateEstimatorTest::IncomingPacket(uint32_t ssrc,
int64_t arrival_time,
uint32_t rtp_timestamp,
uint32_t absolute_send_time) {
WebRtcRTPHeader header;
RTPHeader header;
memset(&header, 0, sizeof(header));
header.header.ssrc = ssrc;
header.header.timestamp = rtp_timestamp;
header.header.extension.absoluteSendTime = absolute_send_time;
header.ssrc = ssrc;
header.timestamp = rtp_timestamp;
header.extension.absoluteSendTime = absolute_send_time;
bitrate_estimator_->IncomingPacket(arrival_time, payload_size, header);
}

2
webrtc/video_engine/vie_channel_group.cc
View File

@ -68,7 +68,7 @@ class WrappingBitrateEstimator : public RemoteBitrateEstimator {
virtual void IncomingPacket(int64_t arrival_time_ms,
int payload_size,
const WebRtcRTPHeader& header) {
const RTPHeader& header) {
CriticalSectionScoped cs(crit_sect_.get());
rbe_->IncomingPacket(arrival_time_ms, payload_size, header);
}

9
webrtc/video_engine/vie_receiver.cc
View File

@ -133,12 +133,6 @@ int32_t ViEReceiver::OnReceivedPayloadData(
if (rtp_header == NULL) {
return 0;
}
// TODO(holmer): Make sure packets reconstructed using FEC are not passed to
// the bandwidth estimator.
const int packet_size = payload_size + rtp_header->header.paddingLength;
remote_bitrate_estimator_->IncomingPacket(TickTime::MillisecondTimestamp(),
packet_size, *rtp_header);
if (vcm_->IncomingPacket(payload_data, payload_size, *rtp_header) != 0) {
// Check this...
return -1;
@ -197,6 +191,9 @@ int ViEReceiver::InsertRTPPacket(const int8_t* rtp_packet,
"IncomingPacket invalid RTP header");
return -1;
}
const int payload_size = received_packet_length - header.headerLength;
remote_bitrate_estimator_->IncomingPacket(TickTime::MillisecondTimestamp(),
payload_size, header);
assert(rtp_rtcp_); // Should be set by owner at construction time.
return rtp_rtcp_->IncomingRtpPacket(received_packet, received_packet_length,
header);