blob: f5a1f693e4eba5e030a5ebb2a48682301d3ed83e [file] [log] [blame]
/*
* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "media/engine/fake_webrtc_call.h"
#include <cstdint>
#include <utility>
#include "absl/algorithm/container.h"
#include "absl/strings/string_view.h"
#include "api/call/audio_sink.h"
#include "api/environment/environment.h"
#include "api/units/timestamp.h"
#include "call/packet_receiver.h"
#include "media/base/media_channel.h"
#include "modules/rtp_rtcp/source/rtp_util.h"
#include "rtc_base/checks.h"
#include "rtc_base/gunit.h"
#include "rtc_base/thread.h"
#include "video/config/encoder_stream_factory.h"
namespace cricket {
using ::webrtc::Environment;
using ::webrtc::ParseRtpSsrc;
FakeAudioSendStream::FakeAudioSendStream(
int id,
const webrtc::AudioSendStream::Config& config)
: id_(id), config_(config) {}
void FakeAudioSendStream::Reconfigure(
const webrtc::AudioSendStream::Config& config,
webrtc::SetParametersCallback callback) {
config_ = config;
webrtc::InvokeSetParametersCallback(callback, webrtc::RTCError::OK());
}
const webrtc::AudioSendStream::Config& FakeAudioSendStream::GetConfig() const {
return config_;
}
void FakeAudioSendStream::SetStats(
const webrtc::AudioSendStream::Stats& stats) {
stats_ = stats;
}
FakeAudioSendStream::TelephoneEvent
FakeAudioSendStream::GetLatestTelephoneEvent() const {
return latest_telephone_event_;
}
bool FakeAudioSendStream::SendTelephoneEvent(int payload_type,
int payload_frequency,
int event,
int duration_ms) {
latest_telephone_event_.payload_type = payload_type;
latest_telephone_event_.payload_frequency = payload_frequency;
latest_telephone_event_.event_code = event;
latest_telephone_event_.duration_ms = duration_ms;
return true;
}
void FakeAudioSendStream::SetMuted(bool muted) {
muted_ = muted;
}
webrtc::AudioSendStream::Stats FakeAudioSendStream::GetStats() const {
return stats_;
}
webrtc::AudioSendStream::Stats FakeAudioSendStream::GetStats(
bool /*has_remote_tracks*/) const {
return stats_;
}
FakeAudioReceiveStream::FakeAudioReceiveStream(
int id,
const webrtc::AudioReceiveStreamInterface::Config& config)
: id_(id), config_(config) {}
const webrtc::AudioReceiveStreamInterface::Config&
FakeAudioReceiveStream::GetConfig() const {
return config_;
}
void FakeAudioReceiveStream::SetStats(
const webrtc::AudioReceiveStreamInterface::Stats& stats) {
stats_ = stats;
}
bool FakeAudioReceiveStream::VerifyLastPacket(const uint8_t* data,
size_t length) const {
return last_packet_ == rtc::Buffer(data, length);
}
bool FakeAudioReceiveStream::DeliverRtp(const uint8_t* packet,
size_t length,
int64_t /* packet_time_us */) {
++received_packets_;
last_packet_.SetData(packet, length);
return true;
}
void FakeAudioReceiveStream::SetDepacketizerToDecoderFrameTransformer(
rtc::scoped_refptr<webrtc::FrameTransformerInterface> frame_transformer) {
config_.frame_transformer = std::move(frame_transformer);
}
void FakeAudioReceiveStream::SetDecoderMap(
std::map<int, webrtc::SdpAudioFormat> decoder_map) {
config_.decoder_map = std::move(decoder_map);
}
void FakeAudioReceiveStream::SetNackHistory(int history_ms) {
config_.rtp.nack.rtp_history_ms = history_ms;
}
void FakeAudioReceiveStream::SetRtcpMode(webrtc::RtcpMode mode) {
config_.rtp.rtcp_mode = mode;
}
void FakeAudioReceiveStream::SetNonSenderRttMeasurement(bool enabled) {
config_.enable_non_sender_rtt = enabled;
}
void FakeAudioReceiveStream::SetFrameDecryptor(
rtc::scoped_refptr<webrtc::FrameDecryptorInterface> frame_decryptor) {
config_.frame_decryptor = std::move(frame_decryptor);
}
webrtc::AudioReceiveStreamInterface::Stats FakeAudioReceiveStream::GetStats(
bool get_and_clear_legacy_stats) const {
return stats_;
}
void FakeAudioReceiveStream::SetSink(webrtc::AudioSinkInterface* sink) {
sink_ = sink;
}
void FakeAudioReceiveStream::SetGain(float gain) {
gain_ = gain;
}
FakeVideoSendStream::FakeVideoSendStream(
const Environment& env,
webrtc::VideoSendStream::Config config,
webrtc::VideoEncoderConfig encoder_config)
: env_(env),
sending_(false),
config_(std::move(config)),
codec_settings_set_(false),
resolution_scaling_enabled_(false),
framerate_scaling_enabled_(false),
source_(nullptr),
num_swapped_frames_(0) {
RTC_DCHECK(config.encoder_settings.encoder_factory != nullptr);
RTC_DCHECK(config.encoder_settings.bitrate_allocator_factory != nullptr);
ReconfigureVideoEncoder(std::move(encoder_config));
}
FakeVideoSendStream::~FakeVideoSendStream() {
if (source_)
source_->RemoveSink(this);
}
const webrtc::VideoSendStream::Config& FakeVideoSendStream::GetConfig() const {
return config_;
}
const webrtc::VideoEncoderConfig& FakeVideoSendStream::GetEncoderConfig()
const {
return encoder_config_;
}
const std::vector<webrtc::VideoStream>& FakeVideoSendStream::GetVideoStreams()
const {
return video_streams_;
}
bool FakeVideoSendStream::IsSending() const {
return sending_;
}
bool FakeVideoSendStream::GetVp8Settings(
webrtc::VideoCodecVP8* settings) const {
if (!codec_settings_set_) {
return false;
}
*settings = codec_specific_settings_.vp8;
return true;
}
bool FakeVideoSendStream::GetVp9Settings(
webrtc::VideoCodecVP9* settings) const {
if (!codec_settings_set_) {
return false;
}
*settings = codec_specific_settings_.vp9;
return true;
}
bool FakeVideoSendStream::GetH264Settings(
webrtc::VideoCodecH264* settings) const {
if (!codec_settings_set_) {
return false;
}
*settings = codec_specific_settings_.h264;
return true;
}
bool FakeVideoSendStream::GetAv1Settings(
webrtc::VideoCodecAV1* settings) const {
if (!codec_settings_set_) {
return false;
}
*settings = codec_specific_settings_.av1;
return true;
}
int FakeVideoSendStream::GetNumberOfSwappedFrames() const {
return num_swapped_frames_;
}
int FakeVideoSendStream::GetLastWidth() const {
return last_frame_->width();
}
int FakeVideoSendStream::GetLastHeight() const {
return last_frame_->height();
}
int64_t FakeVideoSendStream::GetLastTimestamp() const {
RTC_DCHECK(last_frame_->ntp_time_ms() == 0);
return last_frame_->render_time_ms();
}
void FakeVideoSendStream::OnFrame(const webrtc::VideoFrame& frame) {
++num_swapped_frames_;
if (!last_frame_ || frame.width() != last_frame_->width() ||
frame.height() != last_frame_->height() ||
frame.rotation() != last_frame_->rotation()) {
if (encoder_config_.video_stream_factory) {
// Note: only tests set their own EncoderStreamFactory...
video_streams_ =
encoder_config_.video_stream_factory->CreateEncoderStreams(
env_.field_trials(), frame.width(), frame.height(),
encoder_config_);
} else {
webrtc::VideoEncoder::EncoderInfo encoder_info;
auto factory = rtc::make_ref_counted<cricket::EncoderStreamFactory>(
encoder_config_.video_format.name, encoder_config_.max_qp,
encoder_config_.content_type ==
webrtc::VideoEncoderConfig::ContentType::kScreen,
encoder_config_.legacy_conference_mode, encoder_info);
video_streams_ = factory->CreateEncoderStreams(
env_.field_trials(), frame.width(), frame.height(), encoder_config_);
}
}
last_frame_ = frame;
}
void FakeVideoSendStream::SetStats(
const webrtc::VideoSendStream::Stats& stats) {
stats_ = stats;
}
webrtc::VideoSendStream::Stats FakeVideoSendStream::GetStats() {
return stats_;
}
void FakeVideoSendStream::ReconfigureVideoEncoder(
webrtc::VideoEncoderConfig config) {
ReconfigureVideoEncoder(std::move(config), nullptr);
}
void FakeVideoSendStream::ReconfigureVideoEncoder(
webrtc::VideoEncoderConfig config,
webrtc::SetParametersCallback callback) {
int width, height;
if (last_frame_) {
width = last_frame_->width();
height = last_frame_->height();
} else {
width = height = 0;
}
if (config.video_stream_factory) {
// Note: only tests set their own EncoderStreamFactory...
video_streams_ = config.video_stream_factory->CreateEncoderStreams(
env_.field_trials(), width, height, config);
} else {
webrtc::VideoEncoder::EncoderInfo encoder_info;
auto factory = rtc::make_ref_counted<cricket::EncoderStreamFactory>(
config.video_format.name, config.max_qp,
config.content_type == webrtc::VideoEncoderConfig::ContentType::kScreen,
config.legacy_conference_mode, encoder_info);
video_streams_ = factory->CreateEncoderStreams(env_.field_trials(), width,
height, config);
}
if (config.encoder_specific_settings != nullptr) {
const unsigned char num_temporal_layers = static_cast<unsigned char>(
video_streams_.back().num_temporal_layers.value_or(1));
if (config_.rtp.payload_name == "VP8") {
config.encoder_specific_settings->FillVideoCodecVp8(
&codec_specific_settings_.vp8);
if (!video_streams_.empty()) {
codec_specific_settings_.vp8.numberOfTemporalLayers =
num_temporal_layers;
}
} else if (config_.rtp.payload_name == "VP9") {
config.encoder_specific_settings->FillVideoCodecVp9(
&codec_specific_settings_.vp9);
if (!video_streams_.empty()) {
codec_specific_settings_.vp9.numberOfTemporalLayers =
num_temporal_layers;
}
} else if (config_.rtp.payload_name == "H264") {
codec_specific_settings_.h264.numberOfTemporalLayers =
num_temporal_layers;
} else if (config_.rtp.payload_name == "AV1") {
config.encoder_specific_settings->FillVideoCodecAv1(
&codec_specific_settings_.av1);
} else {
ADD_FAILURE() << "Unsupported encoder payload: "
<< config_.rtp.payload_name;
}
}
codec_settings_set_ = config.encoder_specific_settings != nullptr;
encoder_config_ = std::move(config);
++num_encoder_reconfigurations_;
webrtc::InvokeSetParametersCallback(callback, webrtc::RTCError::OK());
}
void FakeVideoSendStream::Start() {
sending_ = true;
}
void FakeVideoSendStream::Stop() {
sending_ = false;
}
void FakeVideoSendStream::AddAdaptationResource(
rtc::scoped_refptr<webrtc::Resource> resource) {}
std::vector<rtc::scoped_refptr<webrtc::Resource>>
FakeVideoSendStream::GetAdaptationResources() {
return {};
}
void FakeVideoSendStream::SetSource(
rtc::VideoSourceInterface<webrtc::VideoFrame>* source,
const webrtc::DegradationPreference& degradation_preference) {
if (source_)
source_->RemoveSink(this);
source_ = source;
switch (degradation_preference) {
case webrtc::DegradationPreference::MAINTAIN_FRAMERATE:
resolution_scaling_enabled_ = true;
framerate_scaling_enabled_ = false;
break;
case webrtc::DegradationPreference::MAINTAIN_RESOLUTION:
resolution_scaling_enabled_ = false;
framerate_scaling_enabled_ = true;
break;
case webrtc::DegradationPreference::BALANCED:
resolution_scaling_enabled_ = true;
framerate_scaling_enabled_ = true;
break;
case webrtc::DegradationPreference::DISABLED:
resolution_scaling_enabled_ = false;
framerate_scaling_enabled_ = false;
break;
}
if (source)
source->AddOrUpdateSink(this, resolution_scaling_enabled_
? sink_wants_
: rtc::VideoSinkWants());
}
void FakeVideoSendStream::GenerateKeyFrame(
const std::vector<std::string>& rids) {
keyframes_requested_by_rid_ = rids;
}
void FakeVideoSendStream::InjectVideoSinkWants(
const rtc::VideoSinkWants& wants) {
sink_wants_ = wants;
source_->AddOrUpdateSink(this, wants);
}
FakeVideoReceiveStream::FakeVideoReceiveStream(
webrtc::VideoReceiveStreamInterface::Config config)
: config_(std::move(config)), receiving_(false) {}
const webrtc::VideoReceiveStreamInterface::Config&
FakeVideoReceiveStream::GetConfig() const {
return config_;
}
bool FakeVideoReceiveStream::IsReceiving() const {
return receiving_;
}
void FakeVideoReceiveStream::InjectFrame(const webrtc::VideoFrame& frame) {
config_.renderer->OnFrame(frame);
}
webrtc::VideoReceiveStreamInterface::Stats FakeVideoReceiveStream::GetStats()
const {
return stats_;
}
void FakeVideoReceiveStream::Start() {
receiving_ = true;
}
void FakeVideoReceiveStream::Stop() {
receiving_ = false;
}
void FakeVideoReceiveStream::SetStats(
const webrtc::VideoReceiveStreamInterface::Stats& stats) {
stats_ = stats;
}
FakeFlexfecReceiveStream::FakeFlexfecReceiveStream(
const webrtc::FlexfecReceiveStream::Config config)
: config_(std::move(config)) {}
const webrtc::FlexfecReceiveStream::Config&
FakeFlexfecReceiveStream::GetConfig() const {
return config_;
}
void FakeFlexfecReceiveStream::OnRtpPacket(const webrtc::RtpPacketReceived&) {
RTC_DCHECK_NOTREACHED() << "Not implemented.";
}
FakeCall::FakeCall(const Environment& env)
: FakeCall(env, rtc::Thread::Current(), rtc::Thread::Current()) {}
FakeCall::FakeCall(const Environment& env,
webrtc::TaskQueueBase* worker_thread,
webrtc::TaskQueueBase* network_thread)
: env_(env),
network_thread_(network_thread),
worker_thread_(worker_thread),
audio_network_state_(webrtc::kNetworkUp),
video_network_state_(webrtc::kNetworkUp),
num_created_send_streams_(0),
num_created_receive_streams_(0) {}
FakeCall::~FakeCall() {
EXPECT_EQ(0u, video_send_streams_.size());
EXPECT_EQ(0u, audio_send_streams_.size());
EXPECT_EQ(0u, video_receive_streams_.size());
EXPECT_EQ(0u, audio_receive_streams_.size());
}
const std::vector<FakeVideoSendStream*>& FakeCall::GetVideoSendStreams() {
return video_send_streams_;
}
const std::vector<FakeVideoReceiveStream*>& FakeCall::GetVideoReceiveStreams() {
return video_receive_streams_;
}
const FakeVideoReceiveStream* FakeCall::GetVideoReceiveStream(uint32_t ssrc) {
for (const auto* p : GetVideoReceiveStreams()) {
if (p->GetConfig().rtp.remote_ssrc == ssrc) {
return p;
}
}
return nullptr;
}
const std::vector<FakeAudioSendStream*>& FakeCall::GetAudioSendStreams() {
return audio_send_streams_;
}
const FakeAudioSendStream* FakeCall::GetAudioSendStream(uint32_t ssrc) {
for (const auto* p : GetAudioSendStreams()) {
if (p->GetConfig().rtp.ssrc == ssrc) {
return p;
}
}
return nullptr;
}
const std::vector<FakeAudioReceiveStream*>& FakeCall::GetAudioReceiveStreams() {
return audio_receive_streams_;
}
const FakeAudioReceiveStream* FakeCall::GetAudioReceiveStream(uint32_t ssrc) {
for (const auto* p : GetAudioReceiveStreams()) {
if (p->GetConfig().rtp.remote_ssrc == ssrc) {
return p;
}
}
return nullptr;
}
const std::vector<FakeFlexfecReceiveStream*>&
FakeCall::GetFlexfecReceiveStreams() {
return flexfec_receive_streams_;
}
webrtc::NetworkState FakeCall::GetNetworkState(webrtc::MediaType media) const {
switch (media) {
case webrtc::MediaType::AUDIO:
return audio_network_state_;
case webrtc::MediaType::VIDEO:
return video_network_state_;
case webrtc::MediaType::DATA:
case webrtc::MediaType::ANY:
ADD_FAILURE() << "GetNetworkState called with unknown parameter.";
return webrtc::kNetworkDown;
}
// Even though all the values for the enum class are listed above,the compiler
// will emit a warning as the method may be called with a value outside of the
// valid enum range, unless this case is also handled.
ADD_FAILURE() << "GetNetworkState called with unknown parameter.";
return webrtc::kNetworkDown;
}
webrtc::AudioSendStream* FakeCall::CreateAudioSendStream(
const webrtc::AudioSendStream::Config& config) {
FakeAudioSendStream* fake_stream =
new FakeAudioSendStream(next_stream_id_++, config);
audio_send_streams_.push_back(fake_stream);
++num_created_send_streams_;
return fake_stream;
}
void FakeCall::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
auto it = absl::c_find(audio_send_streams_,
static_cast<FakeAudioSendStream*>(send_stream));
if (it == audio_send_streams_.end()) {
ADD_FAILURE() << "DestroyAudioSendStream called with unknown parameter.";
} else {
delete *it;
audio_send_streams_.erase(it);
}
}
webrtc::AudioReceiveStreamInterface* FakeCall::CreateAudioReceiveStream(
const webrtc::AudioReceiveStreamInterface::Config& config) {
audio_receive_streams_.push_back(
new FakeAudioReceiveStream(next_stream_id_++, config));
++num_created_receive_streams_;
return audio_receive_streams_.back();
}
void FakeCall::DestroyAudioReceiveStream(
webrtc::AudioReceiveStreamInterface* receive_stream) {
auto it = absl::c_find(audio_receive_streams_,
static_cast<FakeAudioReceiveStream*>(receive_stream));
if (it == audio_receive_streams_.end()) {
ADD_FAILURE() << "DestroyAudioReceiveStream called with unknown parameter.";
} else {
delete *it;
audio_receive_streams_.erase(it);
}
}
webrtc::VideoSendStream* FakeCall::CreateVideoSendStream(
webrtc::VideoSendStream::Config config,
webrtc::VideoEncoderConfig encoder_config) {
FakeVideoSendStream* fake_stream = new FakeVideoSendStream(
env_, std::move(config), std::move(encoder_config));
video_send_streams_.push_back(fake_stream);
++num_created_send_streams_;
return fake_stream;
}
void FakeCall::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) {
auto it = absl::c_find(video_send_streams_,
static_cast<FakeVideoSendStream*>(send_stream));
if (it == video_send_streams_.end()) {
ADD_FAILURE() << "DestroyVideoSendStream called with unknown parameter.";
} else {
delete *it;
video_send_streams_.erase(it);
}
}
webrtc::VideoReceiveStreamInterface* FakeCall::CreateVideoReceiveStream(
webrtc::VideoReceiveStreamInterface::Config config) {
video_receive_streams_.push_back(
new FakeVideoReceiveStream(std::move(config)));
++num_created_receive_streams_;
return video_receive_streams_.back();
}
void FakeCall::DestroyVideoReceiveStream(
webrtc::VideoReceiveStreamInterface* receive_stream) {
auto it = absl::c_find(video_receive_streams_,
static_cast<FakeVideoReceiveStream*>(receive_stream));
if (it == video_receive_streams_.end()) {
ADD_FAILURE() << "DestroyVideoReceiveStream called with unknown parameter.";
} else {
delete *it;
video_receive_streams_.erase(it);
}
}
webrtc::FlexfecReceiveStream* FakeCall::CreateFlexfecReceiveStream(
const webrtc::FlexfecReceiveStream::Config config) {
FakeFlexfecReceiveStream* fake_stream =
new FakeFlexfecReceiveStream(std::move(config));
flexfec_receive_streams_.push_back(fake_stream);
++num_created_receive_streams_;
return fake_stream;
}
void FakeCall::DestroyFlexfecReceiveStream(
webrtc::FlexfecReceiveStream* receive_stream) {
auto it =
absl::c_find(flexfec_receive_streams_,
static_cast<FakeFlexfecReceiveStream*>(receive_stream));
if (it == flexfec_receive_streams_.end()) {
ADD_FAILURE()
<< "DestroyFlexfecReceiveStream called with unknown parameter.";
} else {
delete *it;
flexfec_receive_streams_.erase(it);
}
}
void FakeCall::AddAdaptationResource(
rtc::scoped_refptr<webrtc::Resource> resource) {}
webrtc::PacketReceiver* FakeCall::Receiver() {
return this;
}
void FakeCall::DeliverRtpPacket(
webrtc::MediaType media_type,
webrtc::RtpPacketReceived packet,
OnUndemuxablePacketHandler undemuxable_packet_handler) {
if (!DeliverPacketInternal(media_type, packet.Ssrc(), packet.Buffer(),
packet.arrival_time())) {
if (undemuxable_packet_handler(packet)) {
DeliverPacketInternal(media_type, packet.Ssrc(), packet.Buffer(),
packet.arrival_time());
}
}
last_received_rtp_packet_ = packet;
}
bool FakeCall::DeliverPacketInternal(webrtc::MediaType media_type,
uint32_t ssrc,
const rtc::CopyOnWriteBuffer& packet,
webrtc::Timestamp arrival_time) {
EXPECT_GE(packet.size(), 12u);
RTC_DCHECK(arrival_time.IsFinite());
RTC_DCHECK(media_type == webrtc::MediaType::AUDIO ||
media_type == webrtc::MediaType::VIDEO);
if (media_type == webrtc::MediaType::VIDEO) {
for (auto receiver : video_receive_streams_) {
if (receiver->GetConfig().rtp.remote_ssrc == ssrc ||
receiver->GetConfig().rtp.rtx_ssrc == ssrc) {
++delivered_packets_by_ssrc_[ssrc];
return true;
}
}
}
if (media_type == webrtc::MediaType::AUDIO) {
for (auto receiver : audio_receive_streams_) {
if (receiver->GetConfig().rtp.remote_ssrc == ssrc) {
receiver->DeliverRtp(packet.cdata(), packet.size(), arrival_time.us());
++delivered_packets_by_ssrc_[ssrc];
return true;
}
}
}
return false;
}
void FakeCall::SetStats(const webrtc::Call::Stats& stats) {
stats_ = stats;
}
int FakeCall::GetNumCreatedSendStreams() const {
return num_created_send_streams_;
}
int FakeCall::GetNumCreatedReceiveStreams() const {
return num_created_receive_streams_;
}
webrtc::Call::Stats FakeCall::GetStats() const {
return stats_;
}
webrtc::TaskQueueBase* FakeCall::network_thread() const {
return network_thread_;
}
webrtc::TaskQueueBase* FakeCall::worker_thread() const {
return worker_thread_;
}
void FakeCall::SignalChannelNetworkState(webrtc::MediaType media,
webrtc::NetworkState state) {
switch (media) {
case webrtc::MediaType::AUDIO:
audio_network_state_ = state;
break;
case webrtc::MediaType::VIDEO:
video_network_state_ = state;
break;
case webrtc::MediaType::DATA:
case webrtc::MediaType::ANY:
ADD_FAILURE()
<< "SignalChannelNetworkState called with unknown parameter.";
}
}
void FakeCall::OnAudioTransportOverheadChanged(
int transport_overhead_per_packet) {}
void FakeCall::OnLocalSsrcUpdated(webrtc::AudioReceiveStreamInterface& stream,
uint32_t local_ssrc) {
auto& fake_stream = static_cast<FakeAudioReceiveStream&>(stream);
fake_stream.SetLocalSsrc(local_ssrc);
}
void FakeCall::OnLocalSsrcUpdated(webrtc::VideoReceiveStreamInterface& stream,
uint32_t local_ssrc) {
auto& fake_stream = static_cast<FakeVideoReceiveStream&>(stream);
fake_stream.SetLocalSsrc(local_ssrc);
}
void FakeCall::OnLocalSsrcUpdated(webrtc::FlexfecReceiveStream& stream,
uint32_t local_ssrc) {
auto& fake_stream = static_cast<FakeFlexfecReceiveStream&>(stream);
fake_stream.SetLocalSsrc(local_ssrc);
}
void FakeCall::OnUpdateSyncGroup(webrtc::AudioReceiveStreamInterface& stream,
absl::string_view sync_group) {
auto& fake_stream = static_cast<FakeAudioReceiveStream&>(stream);
fake_stream.SetSyncGroup(sync_group);
}
void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) {
last_sent_packet_ = sent_packet;
if (sent_packet.packet_id >= 0) {
last_sent_nonnegative_packet_id_ = sent_packet.packet_id;
}
}
} // namespace cricket