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Allows audio bitrate allocation in video calls without enabling TWCC (Transport Wide Congestion Control as defined at https://tools.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01.html) for audio stream.

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This will allow experimenting with audio bitrate allocation in video calls without increasing transport overhead.

Bug: webrtc:8243
Change-Id: If961780921d53bdce95b68c26641df6875509c1f
Reviewed-on: https://webrtc-review.googlesource.com/84501
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23755}
This commit is contained in:
Alex Narest
2018-06-25 16:08:36 +02:00
committed by Commit Bot
parent 81f5197512
commit bcf91808a2
13 changed files with 46 additions and 3 deletions
Download Patch File Download Diff File Expand all files Collapse all files

9
audio/audio_send_stream.cc
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@ -294,7 +294,8 @@ void AudioSendStream::Start() {
!webrtc::field_trial::IsEnabled("WebRTC-Audio-ForceNoTWCC");
if (config_.min_bitrate_bps != -1 && config_.max_bitrate_bps != -1 &&
(has_transport_sequence_number ||
!webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe"))) {
!webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe") ||
webrtc::field_trial::IsEnabled("WebRTC-Audio-ABWENoTWCC"))) {
// Audio BWE is enabled.
transport_->packet_sender()->SetAccountForAudioPackets(true);
ConfigureBitrateObserver(config_.min_bitrate_bps, config_.max_bitrate_bps,
@ -409,6 +410,12 @@ uint32_t AudioSendStream::OnBitrateUpdated(uint32_t bitrate_bps,
uint8_t fraction_loss,
int64_t rtt,
int64_t bwe_period_ms) {
// Audio transport feedback will not be reported in this mode, instead update
// acknowledged bitrate estimator with the bitrate allocated for audio.
if (webrtc::field_trial::IsEnabled("WebRTC-Audio-ABWENoTWCC")) {
transport_->SetAllocatedBitrateWithoutFeedback(bitrate_bps);
}
// A send stream may be allocated a bitrate of zero if the allocator decides
// to disable it. For now we ignore this decision and keep sending on min
// bitrate.

5
call/rtp_transport_controller_send.cc
View File

@ -264,4 +264,9 @@ void RtpTransportControllerSend::SetClientBitratePreferences(
<< "nothing to update";
}
}
void RtpTransportControllerSend::SetAllocatedBitrateWithoutFeedback(
uint32_t bitrate_bps) {
send_side_cc_->SetAllocatedBitrateWithoutFeedback(bitrate_bps);
}
} // namespace webrtc

2
call/rtp_transport_controller_send.h
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@ -85,6 +85,8 @@ class RtpTransportControllerSend final
void SetSdpBitrateParameters(const BitrateConstraints& constraints) override;
void SetClientBitratePreferences(const BitrateSettings& preferences) override;
void SetAllocatedBitrateWithoutFeedback(uint32_t bitrate_bps) override;
private:
const Clock* const clock_;
PacketRouter packet_router_;

2
call/rtp_transport_controller_send_interface.h
View File

@ -110,6 +110,8 @@ class RtpTransportControllerSendInterface {
const BitrateConstraints& constraints) = 0;
virtual void SetClientBitratePreferences(
const BitrateSettings& preferences) = 0;
virtual void SetAllocatedBitrateWithoutFeedback(uint32_t bitrate_bps) = 0;
};
} // namespace webrtc

1
call/test/mock_rtp_transport_controller_send.h
View File

@ -51,6 +51,7 @@ class MockRtpTransportControllerSend
MOCK_METHOD1(OnSentPacket, void(const rtc::SentPacket&));
MOCK_METHOD1(SetSdpBitrateParameters, void(const BitrateConstraints&));
MOCK_METHOD1(SetClientBitratePreferences, void(const BitrateSettings&));
MOCK_METHOD1(SetAllocatedBitrateWithoutFeedback, void(uint32_t));
};
} // namespace webrtc
#endif // CALL_TEST_MOCK_RTP_TRANSPORT_CONTROLLER_SEND_H_

3
media/engine/webrtcvoiceengine.cc
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@ -596,7 +596,8 @@ RtpCapabilities WebRtcVoiceEngine::GetCapabilities() const {
capabilities.header_extensions.push_back(
webrtc::RtpExtension(webrtc::RtpExtension::kAudioLevelUri,
webrtc::RtpExtension::kAudioLevelDefaultId));
if (webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe")) {
if (webrtc::field_trial::IsEnabled("WebRTC-Audio-SendSideBwe") &&
!webrtc::field_trial::IsEnabled("WebRTC-Audio-ABWENoTWCC")) {
capabilities.header_extensions.push_back(webrtc::RtpExtension(
webrtc::RtpExtension::kTransportSequenceNumberUri,
webrtc::RtpExtension::kTransportSequenceNumberDefaultId));

11
modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.cc
View File

@ -48,7 +48,11 @@ void AcknowledgedBitrateEstimator::IncomingPacketFeedbackVector(
}
absl::optional<uint32_t> AcknowledgedBitrateEstimator::bitrate_bps() const {
return bitrate_estimator_->bitrate_bps();
auto estimated_bitrate = bitrate_estimator_->bitrate_bps();
return estimated_bitrate
? *estimated_bitrate +
allocated_bitrate_without_feedback_bps_.value_or(0)
: estimated_bitrate;
}
void AcknowledgedBitrateEstimator::SetAlrEndedTimeMs(
@ -56,6 +60,11 @@ void AcknowledgedBitrateEstimator::SetAlrEndedTimeMs(
alr_ended_time_ms_.emplace(alr_ended_time_ms);
}
void AcknowledgedBitrateEstimator::SetAllocatedBitrateWithoutFeedback(
uint32_t bitrate_bps) {
allocated_bitrate_without_feedback_bps_.emplace(bitrate_bps);
}
void AcknowledgedBitrateEstimator::MaybeExpectFastRateChange(
int64_t packet_send_time_ms) {
if (alr_ended_time_ms_ && packet_send_time_ms > *alr_ended_time_ms_) {

2
modules/congestion_controller/goog_cc/acknowledged_bitrate_estimator.h
View File

@ -33,11 +33,13 @@ class AcknowledgedBitrateEstimator {
const std::vector<PacketFeedback>& packet_feedback_vector);
absl::optional<uint32_t> bitrate_bps() const;
void SetAlrEndedTimeMs(int64_t alr_ended_time_ms);
void SetAllocatedBitrateWithoutFeedback(uint32_t bitrate_bps);
private:
void MaybeExpectFastRateChange(int64_t packet_arrival_time_ms);
absl::optional<int64_t> alr_ended_time_ms_;
std::unique_ptr<BitrateEstimator> bitrate_estimator_;
absl::optional<uint32_t> allocated_bitrate_without_feedback_bps_;
};
} // namespace webrtc

2
modules/congestion_controller/include/send_side_congestion_controller.h
View File

@ -120,6 +120,8 @@ class SendSideCongestionController
void SetPacingFactor(float pacing_factor) override;
void SetAllocatedBitrateWithoutFeedback(uint32_t bitrate_bps) override;
private:
void MaybeTriggerOnNetworkChanged();

1
modules/congestion_controller/include/send_side_congestion_controller_interface.h
View File

@ -64,6 +64,7 @@ class SendSideCongestionControllerInterface : public CallStatsObserver,
virtual void EnablePeriodicAlrProbing(bool enable) = 0;
virtual void OnSentPacket(const rtc::SentPacket& sent_packet) = 0;
virtual void SetPacingFactor(float pacing_factor) = 0;
virtual void SetAllocatedBitrateWithoutFeedback(uint32_t bitrate_bps) = 0;
RTC_DISALLOW_COPY_AND_ASSIGN(SendSideCongestionControllerInterface);
};

2
modules/congestion_controller/rtp/include/send_side_congestion_controller.h
View File

@ -139,6 +139,8 @@ class SendSideCongestionController
void SetPacingFactor(float pacing_factor) override;
void SetAllocatedBitrateWithoutFeedback(uint32_t bitrate_bps) override;
protected:
// TODO(srte): The tests should be rewritten to not depend on internals and
// these functions should be removed.

3
modules/congestion_controller/rtp/send_side_congestion_controller.cc
View File

@ -773,6 +773,9 @@ void SendSideCongestionController::SetPacingFactor(float pacing_factor) {
});
}
void SendSideCongestionController::SetAllocatedBitrateWithoutFeedback(
uint32_t bitrate_bps) {}
void SendSideCongestionController::DisablePeriodicTasks() {
task_queue_->PostTask([this]() {
RTC_DCHECK_RUN_ON(task_queue_);

6
modules/congestion_controller/send_side_congestion_controller.cc
View File

@ -415,6 +415,12 @@ void SendSideCongestionController::SetPacingFactor(float pacing_factor) {
pacer_->SetPacingFactor(pacing_factor);
}
void SendSideCongestionController::SetAllocatedBitrateWithoutFeedback(
uint32_t bitrate_bps) {
acknowledged_bitrate_estimator_->SetAllocatedBitrateWithoutFeedback(
bitrate_bps);
}
void SendSideCongestionController::MaybeTriggerOnNetworkChanged() {
uint32_t bitrate_bps;
uint8_t fraction_loss;