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/*
* 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 <utility>
#include "absl/algorithm/container.h"
#include "api/call/audio_sink.h"
#include "media/base/rtp_utils.h"
#include "rtc_base/checks.h"
#include "rtc_base/gunit.h"
namespace cricket {
FakeAudioSendStream::FakeAudioSendStream(
int id, const webrtc::AudioSendStream::Config& config)
: id_(id), config_(config) {
}
void FakeAudioSendStream::Reconfigure(
const webrtc::AudioSendStream::Config& config) {
config_ = config;
}
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::AudioReceiveStream::Config& config)
: id_(id), config_(config) {
}
const webrtc::AudioReceiveStream::Config&
FakeAudioReceiveStream::GetConfig() const {
return config_;
}
void FakeAudioReceiveStream::SetStats(
const webrtc::AudioReceiveStream::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::Reconfigure(
const webrtc::AudioReceiveStream::Config& config) {
config_ = config;
}
webrtc::AudioReceiveStream::Stats FakeAudioReceiveStream::GetStats() const {
return stats_;
}
void FakeAudioReceiveStream::SetSink(webrtc::AudioSinkInterface* sink) {
sink_ = sink;
}
void FakeAudioReceiveStream::SetGain(float gain) {
gain_ = gain;
}
FakeVideoSendStream::FakeVideoSendStream(
webrtc::VideoSendStream::Config config,
webrtc::VideoEncoderConfig encoder_config)
: 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;
}
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()) {
video_streams_ = encoder_config_.video_stream_factory->CreateEncoderStreams(
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) {
int width, height;
if (last_frame_) {
width = last_frame_->width();
height = last_frame_->height();
} else {
width = height = 0;
}
video_streams_ =
config.video_stream_factory->CreateEncoderStreams(width, height, config);
if (config.encoder_specific_settings != NULL) {
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") {
config.encoder_specific_settings->FillVideoCodecH264(
&codec_specific_settings_.h264);
codec_specific_settings_.h264.numberOfTemporalLayers =
num_temporal_layers;
} else {
ADD_FAILURE() << "Unsupported encoder payload: "
<< config_.rtp.payload_name;
}
}
codec_settings_set_ = config.encoder_specific_settings != NULL;
encoder_config_ = std::move(config);
++num_encoder_reconfigurations_;
}
void FakeVideoSendStream::UpdateActiveSimulcastLayers(
const std::vector<bool> active_layers) {
sending_ = false;
for (const bool active_layer : active_layers) {
if (active_layer) {
sending_ = true;
break;
}
}
}
void FakeVideoSendStream::Start() {
sending_ = true;
}
void FakeVideoSendStream::Stop() {
sending_ = false;
}
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::InjectVideoSinkWants(
const rtc::VideoSinkWants& wants) {
sink_wants_ = wants;
source_->AddOrUpdateSink(this, wants);
}
FakeVideoReceiveStream::FakeVideoReceiveStream(
webrtc::VideoReceiveStream::Config config)
: config_(std::move(config)),
receiving_(false),
num_added_secondary_sinks_(0),
num_removed_secondary_sinks_(0) {}
const webrtc::VideoReceiveStream::Config& FakeVideoReceiveStream::GetConfig()
const {
return config_;
}
bool FakeVideoReceiveStream::IsReceiving() const {
return receiving_;
}
void FakeVideoReceiveStream::InjectFrame(const webrtc::VideoFrame& frame) {
config_.renderer->OnFrame(frame);
}
webrtc::VideoReceiveStream::Stats FakeVideoReceiveStream::GetStats() const {
return stats_;
}
void FakeVideoReceiveStream::Start() {
receiving_ = true;
}
void FakeVideoReceiveStream::Stop() {
receiving_ = false;
}
void FakeVideoReceiveStream::SetStats(
const webrtc::VideoReceiveStream::Stats& stats) {
stats_ = stats;
}
void FakeVideoReceiveStream::AddSecondarySink(
webrtc::RtpPacketSinkInterface* sink) {
++num_added_secondary_sinks_;
}
void FakeVideoReceiveStream::RemoveSecondarySink(
const webrtc::RtpPacketSinkInterface* sink) {
++num_removed_secondary_sinks_;
}
int FakeVideoReceiveStream::GetNumAddedSecondarySinks() const {
return num_added_secondary_sinks_;
}
int FakeVideoReceiveStream::GetNumRemovedSecondarySinks() const {
return num_removed_secondary_sinks_;
}
FakeFlexfecReceiveStream::FakeFlexfecReceiveStream(
const webrtc::FlexfecReceiveStream::Config& config)
: config_(config) {}
const webrtc::FlexfecReceiveStream::Config&
FakeFlexfecReceiveStream::GetConfig() const {
return config_;
}
// TODO(brandtr): Implement when the stats have been designed.
webrtc::FlexfecReceiveStream::Stats FakeFlexfecReceiveStream::GetStats() const {
return webrtc::FlexfecReceiveStream::Stats();
}
void FakeFlexfecReceiveStream::OnRtpPacket(const webrtc::RtpPacketReceived&) {
RTC_NOTREACHED() << "Not implemented.";
}
FakeCall::FakeCall()
: 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::AudioReceiveStream* FakeCall::CreateAudioReceiveStream(
const webrtc::AudioReceiveStream::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::AudioReceiveStream* 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(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::VideoReceiveStream* FakeCall::CreateVideoReceiveStream(
webrtc::VideoReceiveStream::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::VideoReceiveStream* 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(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);
}
}
webrtc::PacketReceiver* FakeCall::Receiver() {
return this;
}
FakeCall::DeliveryStatus FakeCall::DeliverPacket(webrtc::MediaType media_type,
rtc::CopyOnWriteBuffer packet,
int64_t packet_time_us) {
EXPECT_GE(packet.size(), 12u);
RTC_DCHECK(media_type == webrtc::MediaType::AUDIO ||
media_type == webrtc::MediaType::VIDEO);
uint32_t ssrc;
if (!GetRtpSsrc(packet.cdata(), packet.size(), &ssrc))
return DELIVERY_PACKET_ERROR;
if (media_type == webrtc::MediaType::VIDEO) {
for (auto receiver : video_receive_streams_) {
if (receiver->GetConfig().rtp.remote_ssrc == ssrc)
return DELIVERY_OK;
}
}
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(), packet_time_us);
return DELIVERY_OK;
}
}
}
return DELIVERY_UNKNOWN_SSRC;
}
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_;
}
void FakeCall::SetBitrateAllocationStrategy(
std::unique_ptr<rtc::BitrateAllocationStrategy>
bitrate_allocation_strategy) {
// TODO(alexnarest): not implemented
}
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::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;
}
}
void FakeCall::MediaTransportChange(
webrtc::MediaTransportInterface* media_transport_interface) {}
} // namespace cricket