admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Enable transport seq num extension on receive channel to suppress log warning.

Browse Source 
TBR=pbos@webrtc.org

BUG=webrtc:5263

Review URL: https://codereview.webrtc.org/1608563005

Cr-Commit-Position: refs/heads/master@{#11338}
This commit is contained in:
stefan 2016-01-21 06:32:43 -08:00 committed by Commit bot
parent d664836efa
commit 3313ec901f
7 changed files with 30 additions and 9 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

1
webrtc/audio/audio_receive_stream.cc
View File

@ -108,6 +108,7 @@ AudioReceiveStream::AudioReceiveStream(
kRtpExtensionAbsoluteSendTime, extension.id);
RTC_DCHECK(registered);
} else if (extension.name == RtpExtension::kTransportSequenceNumber) {
channel_proxy_->EnableReceiveTransportSequenceNumber(extension.id);
bool registered = rtp_header_parser_->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber, extension.id);
RTC_DCHECK(registered);

8
webrtc/audio/audio_receive_stream_unittest.cc
View File

@ -9,6 +9,7 @@
*/
#include <string>
#include <vector>
#include "testing/gtest/include/gtest/gtest.h"
@ -90,6 +91,9 @@ struct ConfigHelper {
EXPECT_CALL(*channel_proxy_,
SetReceiveAudioLevelIndicationStatus(true, kAudioLevelId))
.Times(1);
EXPECT_CALL(*channel_proxy_, EnableReceiveTransportSequenceNumber(
kTransportSequenceNumberId))
.Times(1);
EXPECT_CALL(*channel_proxy_, SetCongestionControlObjects(
nullptr, nullptr, &packet_router_))
.Times(1);
@ -107,6 +111,8 @@ struct ConfigHelper {
RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
stream_config_.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAudioLevel, kAudioLevelId));
stream_config_.rtp.extensions.push_back(RtpExtension(
RtpExtension::kTransportSequenceNumber, kTransportSequenceNumberId));
}
MockCongestionController* congestion_controller() {
@ -261,8 +267,6 @@ TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweFeedback) {
ConfigHelper helper;
helper.config().combined_audio_video_bwe = true;
helper.config().rtp.transport_cc = true;
helper.config().rtp.extensions.push_back(RtpExtension(
RtpExtension::kTransportSequenceNumber, kTransportSequenceNumberId));
helper.SetupMockForBweFeedback(true);
internal::AudioReceiveStream recv_stream(
helper.congestion_controller(), helper.config(), helper.audio_state());

3
webrtc/test/mock_voe_channel_proxy.h
View File

@ -25,9 +25,10 @@ class MockVoEChannelProxy : public voe::ChannelProxy {
MOCK_METHOD1(SetRTCP_CNAME, void(const std::string& c_name));
MOCK_METHOD2(SetSendAbsoluteSenderTimeStatus, void(bool enable, int id));
MOCK_METHOD2(SetSendAudioLevelIndicationStatus, void(bool enable, int id));
MOCK_METHOD1(EnableSendTransportSequenceNumber, void(int id));
MOCK_METHOD2(SetReceiveAbsoluteSenderTimeStatus, void(bool enable, int id));
MOCK_METHOD2(SetReceiveAudioLevelIndicationStatus, void(bool enable, int id));
MOCK_METHOD1(EnableSendTransportSequenceNumber, void(int id));
MOCK_METHOD1(EnableReceiveTransportSequenceNumber, void(int id));
MOCK_METHOD3(SetCongestionControlObjects,
void(RtpPacketSender* rtp_packet_sender,
TransportFeedbackObserver* transport_feedback_observer,

8
webrtc/voice_engine/channel.cc
View File

@ -2924,6 +2924,14 @@ void Channel::EnableSendTransportSequenceNumber(int id) {
RTC_DCHECK_EQ(0, ret);
}
void Channel::EnableReceiveTransportSequenceNumber(int id) {
rtp_header_parser_->DeregisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber);
bool ret = rtp_header_parser_->RegisterRtpHeaderExtension(
kRtpExtensionTransportSequenceNumber, id);
RTC_DCHECK(ret);
}
void Channel::SetCongestionControlObjects(
RtpPacketSender* rtp_packet_sender,
TransportFeedbackObserver* transport_feedback_observer,

1
webrtc/voice_engine/channel.h
View File

@ -329,6 +329,7 @@ public:
int SetSendAbsoluteSenderTimeStatus(bool enable, unsigned char id);
int SetReceiveAbsoluteSenderTimeStatus(bool enable, unsigned char id);
void EnableSendTransportSequenceNumber(int id);
void EnableReceiveTransportSequenceNumber(int id);
void SetCongestionControlObjects(
RtpPacketSender* rtp_packet_sender,

15
webrtc/voice_engine/channel_proxy.cc
View File

@ -57,11 +57,6 @@ void ChannelProxy::SetSendAudioLevelIndicationStatus(bool enable, int id) {
RTC_DCHECK_EQ(0, error);
}
void ChannelProxy::EnableSendTransportSequenceNumber(int id) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
channel()->EnableSendTransportSequenceNumber(id);
}
void ChannelProxy::SetReceiveAbsoluteSenderTimeStatus(bool enable, int id) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
int error = channel()->SetReceiveAbsoluteSenderTimeStatus(enable, id);
@ -74,6 +69,16 @@ void ChannelProxy::SetReceiveAudioLevelIndicationStatus(bool enable, int id) {
RTC_DCHECK_EQ(0, error);
}
void ChannelProxy::EnableSendTransportSequenceNumber(int id) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
channel()->EnableSendTransportSequenceNumber(id);
}
void ChannelProxy::EnableReceiveTransportSequenceNumber(int id) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
channel()->EnableReceiveTransportSequenceNumber(id);
}
void ChannelProxy::SetCongestionControlObjects(
RtpPacketSender* rtp_packet_sender,
TransportFeedbackObserver* transport_feedback_observer,

3
webrtc/voice_engine/channel_proxy.h
View File

@ -47,9 +47,10 @@ class ChannelProxy {
virtual void SetRTCP_CNAME(const std::string& c_name);
virtual void SetSendAbsoluteSenderTimeStatus(bool enable, int id);
virtual void SetSendAudioLevelIndicationStatus(bool enable, int id);
virtual void EnableSendTransportSequenceNumber(int id);
virtual void SetReceiveAbsoluteSenderTimeStatus(bool enable, int id);
virtual void SetReceiveAudioLevelIndicationStatus(bool enable, int id);
virtual void EnableSendTransportSequenceNumber(int id);
virtual void EnableReceiveTransportSequenceNumber(int id);
virtual void SetCongestionControlObjects(
RtpPacketSender* rtp_packet_sender,
TransportFeedbackObserver* transport_feedback_observer,