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Add ability to set bitrate of DegradedCall via PeerConnection::SetBitrate

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Bug: None
Change-Id: Iac8970c95a01c1322fa65a19ab11ffd8f94412e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279200
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38442}
This commit is contained in:
Byoungchan Lee
2022-10-14 12:06:30 +09:00
committed by WebRTC LUCI CQ
parent 09da10e24f
commit 3f519e0c89
3 changed files with 18 additions and 3 deletions
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5
call/degraded_call.cc
View File

@ -414,6 +414,11 @@ PacketReceiver::DeliveryStatus DegradedCall::DeliverPacket(
return status;
}
void DegradedCall::SetClientBitratePreferences(
const webrtc::BitrateSettings& preferences) {
call_->SetClientBitratePreferences(preferences);
}
void DegradedCall::UpdateSendNetworkConfig() {
send_config_index_ = (send_config_index_ + 1) % send_configs_.size();
send_simulated_network_->SetConfig(send_configs_[send_config_index_]);

2
call/degraded_call.h
View File

@ -191,7 +191,7 @@ class DegradedCall : public Call, private PacketReceiver {
};
void SetClientBitratePreferences(
const webrtc::BitrateSettings& preferences) override {}
const webrtc::BitrateSettings& preferences) override;
void UpdateSendNetworkConfig();
void UpdateReceiveNetworkConfig();

14
pc/peer_connection_field_trial_tests.cc
View File

@ -17,6 +17,7 @@
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/create_peerconnection_factory.h"
#include "api/peer_connection_interface.h"
#include "api/stats/rtcstats_objects.h"
#include "api/task_queue/default_task_queue_factory.h"
#include "api/video_codecs/builtin_video_decoder_factory.h"
#include "api/video_codecs/builtin_video_encoder_factory.h"
@ -228,6 +229,10 @@ TEST_F(PeerConnectionFieldTrialTest, ApplyFakeNetworkConfig) {
CreatePCFactory(std::move(field_trials));
WrapperPtr caller = CreatePeerConnection();
BitrateSettings bitrate_settings;
bitrate_settings.start_bitrate_bps = 1'000'000;
bitrate_settings.max_bitrate_bps = 1'000'000;
caller->pc()->SetBitrate(bitrate_settings);
FrameGeneratorCapturerVideoTrackSource::Config config;
auto video_track_source =
rtc::make_ref_counted<FrameGeneratorCapturerVideoTrackSource>(
@ -259,9 +264,14 @@ TEST_F(PeerConnectionFieldTrialTest, ApplyFakeNetworkConfig) {
ASSERT_TRUE_WAIT(caller->IsIceConnected(), kDefaultTimeoutMs);
// Send packets for kDefaultTimeoutMs
// For now, whether this field trial works or not is checked by
// whether a crash occurs. Additional validation can be added later.
WAIT(false, kDefaultTimeoutMs);
std::vector<const RTCOutboundRTPStreamStats*> outbound_rtp_stats =
caller->GetStats()->GetStatsOfType<RTCOutboundRTPStreamStats>();
ASSERT_GE(outbound_rtp_stats.size(), 1u);
ASSERT_TRUE(outbound_rtp_stats[0]->target_bitrate.is_defined());
// Link capacity is limited to 500k, so BWE is expected to be close to 500k.
ASSERT_LE(*outbound_rtp_stats[0]->target_bitrate, 500'000 * 1.1);
}
} // namespace webrtc