blob: 9515505af2c2458f04e25b0ef4aae5f0abe74b5d [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 "webrtc/media/engine/fakewebrtccall.h"
#include <algorithm>
#include <utility>
#include "webrtc/api/call/audio_sink.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/platform_file.h"
#include "webrtc/base/gunit.h"
#include "webrtc/media/base/rtputils.h"
namespace cricket {
FakeAudioSendStream::FakeAudioSendStream(
const webrtc::AudioSendStream::Config& config) : config_(config) {
RTC_DCHECK(config.voe_channel_id != -1);
}
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 event,
int duration_ms) {
latest_telephone_event_.payload_type = payload_type;
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_;
}
FakeAudioReceiveStream::FakeAudioReceiveStream(
const webrtc::AudioReceiveStream::Config& config)
: config_(config) {
RTC_DCHECK(config.voe_channel_id != -1);
}
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,
const webrtc::PacketTime& packet_time) {
++received_packets_;
last_packet_.SetData(packet, length);
return true;
}
webrtc::AudioReceiveStream::Stats FakeAudioReceiveStream::GetStats() const {
return stats_;
}
void FakeAudioReceiveStream::SetSink(
std::unique_ptr<webrtc::AudioSinkInterface> sink) {
sink_ = std::move(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),
source_(nullptr),
num_swapped_frames_(0) {
RTC_DCHECK(config.encoder_settings.encoder != NULL);
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_;
}
std::vector<webrtc::VideoStream> FakeVideoSendStream::GetVideoStreams() {
return encoder_config_.streams;
}
bool FakeVideoSendStream::IsSending() const {
return sending_;
}
bool FakeVideoSendStream::GetVp8Settings(
webrtc::VideoCodecVP8* settings) const {
if (!codec_settings_set_) {
return false;
}
*settings = vpx_settings_.vp8;
return true;
}
bool FakeVideoSendStream::GetVp9Settings(
webrtc::VideoCodecVP9* settings) const {
if (!codec_settings_set_) {
return false;
}
*settings = vpx_settings_.vp9;
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_;
last_frame_.ShallowCopy(frame);
}
void FakeVideoSendStream::SetStats(
const webrtc::VideoSendStream::Stats& stats) {
stats_ = stats;
}
webrtc::VideoSendStream::Stats FakeVideoSendStream::GetStats() {
return stats_;
}
void FakeVideoSendStream::EnableEncodedFrameRecording(
const std::vector<rtc::PlatformFile>& files,
size_t byte_limit) {
for (rtc::PlatformFile file : files)
rtc::ClosePlatformFile(file);
}
void FakeVideoSendStream::ReconfigureVideoEncoder(
webrtc::VideoEncoderConfig config) {
if (config.encoder_specific_settings != NULL) {
if (config_.encoder_settings.payload_name == "VP8") {
config.encoder_specific_settings->FillVideoCodecVp8(&vpx_settings_.vp8);
if (!config.streams.empty()) {
vpx_settings_.vp8.numberOfTemporalLayers = static_cast<unsigned char>(
config.streams.back().temporal_layer_thresholds_bps.size() + 1);
}
} else if (config_.encoder_settings.payload_name == "VP9") {
config.encoder_specific_settings->FillVideoCodecVp9(&vpx_settings_.vp9);
if (!config.streams.empty()) {
vpx_settings_.vp9.numberOfTemporalLayers = static_cast<unsigned char>(
config.streams.back().temporal_layer_thresholds_bps.size() + 1);
}
} else {
ADD_FAILURE() << "Unsupported encoder payload: "
<< config_.encoder_settings.payload_name;
}
}
codec_settings_set_ = config.encoder_specific_settings != NULL;
encoder_config_ = std::move(config);
++num_encoder_reconfigurations_;
}
void FakeVideoSendStream::Start() {
sending_ = true;
}
void FakeVideoSendStream::Stop() {
sending_ = false;
}
void FakeVideoSendStream::SetSource(
rtc::VideoSourceInterface<webrtc::VideoFrame>* source) {
RTC_DCHECK(source != source_);
if (source_)
source_->RemoveSink(this);
source_ = source;
if (source)
source->AddOrUpdateSink(this, rtc::VideoSinkWants());
}
FakeVideoReceiveStream::FakeVideoReceiveStream(
webrtc::VideoReceiveStream::Config config)
: config_(std::move(config)), receiving_(false) {}
const webrtc::VideoReceiveStream::Config& FakeVideoReceiveStream::GetConfig() {
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::EnableEncodedFrameRecording(rtc::PlatformFile file,
size_t byte_limit) {
rtc::ClosePlatformFile(file);
}
FakeCall::FakeCall(const webrtc::Call::Config& config)
: config_(config),
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());
}
webrtc::Call::Config FakeCall::GetConfig() const {
return config_;
}
const std::vector<FakeVideoSendStream*>& FakeCall::GetVideoSendStreams() {
return video_send_streams_;
}
const std::vector<FakeVideoReceiveStream*>& FakeCall::GetVideoReceiveStreams() {
return video_receive_streams_;
}
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;
}
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(config);
audio_send_streams_.push_back(fake_stream);
++num_created_send_streams_;
return fake_stream;
}
void FakeCall::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
auto it = std::find(audio_send_streams_.begin(),
audio_send_streams_.end(),
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(config));
++num_created_receive_streams_;
return audio_receive_streams_.back();
}
void FakeCall::DestroyAudioReceiveStream(
webrtc::AudioReceiveStream* receive_stream) {
auto it = std::find(audio_receive_streams_.begin(),
audio_receive_streams_.end(),
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 = std::find(video_send_streams_.begin(),
video_send_streams_.end(),
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 = std::find(video_receive_streams_.begin(),
video_receive_streams_.end(),
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::PacketReceiver* FakeCall::Receiver() {
return this;
}
FakeCall::DeliveryStatus FakeCall::DeliverPacket(
webrtc::MediaType media_type,
const uint8_t* packet,
size_t length,
const webrtc::PacketTime& packet_time) {
EXPECT_GE(length, 12u);
uint32_t ssrc;
if (!GetRtpSsrc(packet, length, &ssrc))
return DELIVERY_PACKET_ERROR;
if (media_type == webrtc::MediaType::ANY ||
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::ANY ||
media_type == webrtc::MediaType::AUDIO) {
for (auto receiver : audio_receive_streams_) {
if (receiver->GetConfig().rtp.remote_ssrc == ssrc) {
receiver->DeliverRtp(packet, length, packet_time);
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::SetBitrateConfig(
const webrtc::Call::Config::BitrateConfig& bitrate_config) {
config_.bitrate_config = bitrate_config;
}
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::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;
}
}
bool FakeCall::StartEventLog(rtc::PlatformFile log_file,
int64_t max_size_bytes) {
return false;
}
void FakeCall::StopEventLog() {}
} // namespace cricket