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Have AudioSendStream register CNG payload types with the RtpRtcpModule.

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TBR=kwiberg@webrtc.org # Turn perf-bots green again

BUG=webrtc:5806

Review-Url: https://codereview.webrtc.org/2844803003
Cr-Commit-Position: refs/heads/master@{#17911}
This commit is contained in:
ossu
2017-04-27 08:03:42 -07:00
committed by Commit bot
parent 9078d8cf05
commit 3b9ff38d8a
3 changed files with 39 additions and 1 deletions
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30
webrtc/audio/audio_send_stream.cc
View File

@ -28,7 +28,7 @@
#include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
#include "webrtc/modules/congestion_controller/include/send_side_congestion_controller.h"
#include "webrtc/modules/pacing/paced_sender.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/voice_engine/channel_proxy.h"
#include "webrtc/voice_engine/include/voe_base.h"
#include "webrtc/voice_engine/transmit_mixer.h"
@ -421,6 +421,10 @@ bool AudioSendStream::SetupSendCodec(AudioSendStream* stream,
cng_config.speech_encoder = std::move(encoder);
cng_config.vad_mode = Vad::kVadNormal;
encoder.reset(new AudioEncoderCng(std::move(cng_config)));
stream->RegisterCngPayloadType(
*spec.cng_payload_type,
new_config.send_codec_spec->format.clockrate_hz);
}
stream->channel_proxy_->SetEncoder(new_config.send_codec_spec->payload_type,
@ -498,6 +502,14 @@ void AudioSendStream::ReconfigureCNG(AudioSendStream* stream,
return;
}
// Register the CNG payload type if it's been added, don't do anything if CNG
// is removed. Payload types must not be redefined.
if (new_config.send_codec_spec->cng_payload_type) {
stream->RegisterCngPayloadType(
*new_config.send_codec_spec->cng_payload_type,
new_config.send_codec_spec->format.clockrate_hz);
}
// Wrap or unwrap the encoder in an AudioEncoderCNG.
stream->channel_proxy_->ModifyEncoder(
[&](std::unique_ptr<AudioEncoder>* encoder_ptr) {
@ -572,5 +584,21 @@ void AudioSendStream::RemoveBitrateObserver() {
thread_sync_event.Wait(rtc::Event::kForever);
}
void AudioSendStream::RegisterCngPayloadType(int payload_type,
int clockrate_hz) {
RtpRtcp* rtpRtcpModule = nullptr;
RtpReceiver* rtpReceiver = nullptr; // Unused, but required for call.
channel_proxy_->GetRtpRtcp(&rtpRtcpModule, &rtpReceiver);
const CodecInst codec = {payload_type, "CN", clockrate_hz, 0, 1, 0};
if (rtpRtcpModule->RegisterSendPayload(codec) != 0) {
rtpRtcpModule->DeRegisterSendPayload(codec.pltype);
if (rtpRtcpModule->RegisterSendPayload(codec) != 0) {
LOG(LS_ERROR) << "RegisterCngPayloadType() failed to register CN to "
"RTP/RTCP module";
}
}
}
} // namespace internal
} // namespace webrtc

2
webrtc/audio/audio_send_stream.h
View File

@ -93,6 +93,8 @@ class AudioSendStream final : public webrtc::AudioSendStream,
void ConfigureBitrateObserver(int min_bitrate_bps, int max_bitrate_bps);
void RemoveBitrateObserver();
void RegisterCngPayloadType(int payload_type, int clockrate_hz);
rtc::ThreadChecker worker_thread_checker_;
rtc::ThreadChecker pacer_thread_checker_;
rtc::TaskQueue* worker_queue_;

8
webrtc/audio/audio_send_stream_unittest.cc
View File

@ -25,6 +25,7 @@
#include "webrtc/modules/congestion_controller/include/mock/mock_congestion_observer.h"
#include "webrtc/modules/congestion_controller/include/send_side_congestion_controller.h"
#include "webrtc/modules/pacing/paced_sender.h"
#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/mock_voe_channel_proxy.h"
@ -217,6 +218,12 @@ struct ConfigHelper {
void SetupDefaultChannelProxy(bool audio_bwe_enabled) {
using testing::StrEq;
channel_proxy_ = new testing::StrictMock<MockVoEChannelProxy>();
EXPECT_CALL(*channel_proxy_, GetRtpRtcp(_, _))
.WillRepeatedly(Invoke(
[this](RtpRtcp** rtp_rtcp_module, RtpReceiver** rtp_receiver) {
*rtp_rtcp_module = &this->rtp_rtcp_;
*rtp_receiver = nullptr; // Not deemed necessary for tests yet.
}));
EXPECT_CALL(*channel_proxy_, SetRTCPStatus(true)).Times(1);
EXPECT_CALL(*channel_proxy_, SetLocalSSRC(kSsrc)).Times(1);
EXPECT_CALL(*channel_proxy_, SetRTCP_CNAME(StrEq(kCName))).Times(1);
@ -321,6 +328,7 @@ struct ConfigHelper {
AudioProcessing::AudioProcessingStatistics audio_processing_stats_;
FakeRtpTransportController fake_transport_;
MockRtcEventLog event_log_;
MockRtpRtcp rtp_rtcp_;
MockRtcpRttStats rtcp_rtt_stats_;
testing::NiceMock<MockLimitObserver> limit_observer_;
BitrateAllocator bitrate_allocator_;