482 lines
14 KiB
C++
482 lines
14 KiB
C++
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/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/media/engine/fakewebrtccall.h"
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#include <algorithm>
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#include <utility>
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#include "webrtc/audio_sink.h"
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#include "webrtc/base/checks.h"
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#include "webrtc/base/gunit.h"
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#include "webrtc/media/base/rtputils.h"
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namespace cricket {
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FakeAudioSendStream::FakeAudioSendStream(
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const webrtc::AudioSendStream::Config& config) : config_(config) {
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RTC_DCHECK(config.voe_channel_id != -1);
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}
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const webrtc::AudioSendStream::Config&
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FakeAudioSendStream::GetConfig() const {
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return config_;
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}
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void FakeAudioSendStream::SetStats(
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const webrtc::AudioSendStream::Stats& stats) {
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stats_ = stats;
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}
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FakeAudioSendStream::TelephoneEvent
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FakeAudioSendStream::GetLatestTelephoneEvent() const {
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return latest_telephone_event_;
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}
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bool FakeAudioSendStream::SendTelephoneEvent(int payload_type, int event,
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int duration_ms) {
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latest_telephone_event_.payload_type = payload_type;
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latest_telephone_event_.event_code = event;
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latest_telephone_event_.duration_ms = duration_ms;
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return true;
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}
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void FakeAudioSendStream::SetMuted(bool muted) {
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muted_ = muted;
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}
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webrtc::AudioSendStream::Stats FakeAudioSendStream::GetStats() const {
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return stats_;
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}
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FakeAudioReceiveStream::FakeAudioReceiveStream(
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const webrtc::AudioReceiveStream::Config& config)
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: config_(config) {
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RTC_DCHECK(config.voe_channel_id != -1);
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}
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const webrtc::AudioReceiveStream::Config&
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FakeAudioReceiveStream::GetConfig() const {
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return config_;
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}
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void FakeAudioReceiveStream::SetStats(
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const webrtc::AudioReceiveStream::Stats& stats) {
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stats_ = stats;
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}
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bool FakeAudioReceiveStream::VerifyLastPacket(const uint8_t* data,
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size_t length) const {
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return last_packet_ == rtc::Buffer(data, length);
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}
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bool FakeAudioReceiveStream::DeliverRtp(const uint8_t* packet,
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size_t length,
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const webrtc::PacketTime& packet_time) {
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++received_packets_;
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last_packet_.SetData(packet, length);
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return true;
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}
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webrtc::AudioReceiveStream::Stats FakeAudioReceiveStream::GetStats() const {
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return stats_;
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}
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void FakeAudioReceiveStream::SetSink(
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std::unique_ptr<webrtc::AudioSinkInterface> sink) {
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sink_ = std::move(sink);
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}
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void FakeAudioReceiveStream::SetGain(float gain) {
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gain_ = gain;
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}
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FakeVideoSendStream::FakeVideoSendStream(
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const webrtc::VideoSendStream::Config& config,
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const webrtc::VideoEncoderConfig& encoder_config)
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: sending_(false),
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config_(config),
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codec_settings_set_(false),
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num_swapped_frames_(0) {
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RTC_DCHECK(config.encoder_settings.encoder != NULL);
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ReconfigureVideoEncoder(encoder_config);
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}
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webrtc::VideoSendStream::Config FakeVideoSendStream::GetConfig() const {
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return config_;
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}
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webrtc::VideoEncoderConfig FakeVideoSendStream::GetEncoderConfig() const {
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return encoder_config_;
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}
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std::vector<webrtc::VideoStream> FakeVideoSendStream::GetVideoStreams() {
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return encoder_config_.streams;
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}
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bool FakeVideoSendStream::IsSending() const {
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return sending_;
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}
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bool FakeVideoSendStream::GetVp8Settings(
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webrtc::VideoCodecVP8* settings) const {
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if (!codec_settings_set_) {
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return false;
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}
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*settings = vpx_settings_.vp8;
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return true;
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}
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bool FakeVideoSendStream::GetVp9Settings(
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webrtc::VideoCodecVP9* settings) const {
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if (!codec_settings_set_) {
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return false;
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}
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*settings = vpx_settings_.vp9;
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return true;
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}
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int FakeVideoSendStream::GetNumberOfSwappedFrames() const {
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return num_swapped_frames_;
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}
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int FakeVideoSendStream::GetLastWidth() const {
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return last_frame_.width();
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}
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int FakeVideoSendStream::GetLastHeight() const {
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return last_frame_.height();
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}
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int64_t FakeVideoSendStream::GetLastTimestamp() const {
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RTC_DCHECK(last_frame_.ntp_time_ms() == 0);
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return last_frame_.render_time_ms();
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}
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void FakeVideoSendStream::IncomingCapturedFrame(
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const webrtc::VideoFrame& frame) {
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++num_swapped_frames_;
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last_frame_.ShallowCopy(frame);
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}
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void FakeVideoSendStream::SetStats(
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const webrtc::VideoSendStream::Stats& stats) {
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stats_ = stats;
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}
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webrtc::VideoSendStream::Stats FakeVideoSendStream::GetStats() {
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return stats_;
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}
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void FakeVideoSendStream::ReconfigureVideoEncoder(
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const webrtc::VideoEncoderConfig& config) {
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encoder_config_ = config;
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if (config.encoder_specific_settings != NULL) {
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if (config_.encoder_settings.payload_name == "VP8") {
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vpx_settings_.vp8 = *reinterpret_cast<const webrtc::VideoCodecVP8*>(
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config.encoder_specific_settings);
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if (!config.streams.empty()) {
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vpx_settings_.vp8.numberOfTemporalLayers = static_cast<unsigned char>(
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config.streams.back().temporal_layer_thresholds_bps.size() + 1);
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}
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} else if (config_.encoder_settings.payload_name == "VP9") {
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vpx_settings_.vp9 = *reinterpret_cast<const webrtc::VideoCodecVP9*>(
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config.encoder_specific_settings);
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if (!config.streams.empty()) {
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vpx_settings_.vp9.numberOfTemporalLayers = static_cast<unsigned char>(
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config.streams.back().temporal_layer_thresholds_bps.size() + 1);
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}
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} else {
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ADD_FAILURE() << "Unsupported encoder payload: "
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<< config_.encoder_settings.payload_name;
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}
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}
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codec_settings_set_ = config.encoder_specific_settings != NULL;
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++num_encoder_reconfigurations_;
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}
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webrtc::VideoCaptureInput* FakeVideoSendStream::Input() {
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return this;
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}
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void FakeVideoSendStream::Start() {
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sending_ = true;
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}
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void FakeVideoSendStream::Stop() {
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sending_ = false;
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}
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FakeVideoReceiveStream::FakeVideoReceiveStream(
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webrtc::VideoReceiveStream::Config config)
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: config_(std::move(config)), receiving_(false) {}
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const webrtc::VideoReceiveStream::Config& FakeVideoReceiveStream::GetConfig() {
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return config_;
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}
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bool FakeVideoReceiveStream::IsReceiving() const {
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return receiving_;
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}
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void FakeVideoReceiveStream::InjectFrame(const webrtc::VideoFrame& frame) {
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config_.renderer->OnFrame(frame);
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}
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webrtc::VideoReceiveStream::Stats FakeVideoReceiveStream::GetStats() const {
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return stats_;
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}
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void FakeVideoReceiveStream::Start() {
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receiving_ = true;
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}
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void FakeVideoReceiveStream::Stop() {
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receiving_ = false;
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}
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void FakeVideoReceiveStream::SetStats(
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const webrtc::VideoReceiveStream::Stats& stats) {
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stats_ = stats;
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}
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FakeCall::FakeCall(const webrtc::Call::Config& config)
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: config_(config),
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audio_network_state_(webrtc::kNetworkUp),
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video_network_state_(webrtc::kNetworkUp),
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num_created_send_streams_(0),
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num_created_receive_streams_(0) {}
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FakeCall::~FakeCall() {
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EXPECT_EQ(0u, video_send_streams_.size());
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EXPECT_EQ(0u, audio_send_streams_.size());
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EXPECT_EQ(0u, video_receive_streams_.size());
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EXPECT_EQ(0u, audio_receive_streams_.size());
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}
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webrtc::Call::Config FakeCall::GetConfig() const {
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return config_;
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}
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const std::vector<FakeVideoSendStream*>& FakeCall::GetVideoSendStreams() {
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return video_send_streams_;
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}
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const std::vector<FakeVideoReceiveStream*>& FakeCall::GetVideoReceiveStreams() {
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return video_receive_streams_;
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}
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const std::vector<FakeAudioSendStream*>& FakeCall::GetAudioSendStreams() {
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return audio_send_streams_;
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}
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const FakeAudioSendStream* FakeCall::GetAudioSendStream(uint32_t ssrc) {
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for (const auto* p : GetAudioSendStreams()) {
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if (p->GetConfig().rtp.ssrc == ssrc) {
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return p;
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}
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}
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return nullptr;
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}
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const std::vector<FakeAudioReceiveStream*>& FakeCall::GetAudioReceiveStreams() {
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return audio_receive_streams_;
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}
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const FakeAudioReceiveStream* FakeCall::GetAudioReceiveStream(uint32_t ssrc) {
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for (const auto* p : GetAudioReceiveStreams()) {
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if (p->GetConfig().rtp.remote_ssrc == ssrc) {
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return p;
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}
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}
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return nullptr;
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}
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webrtc::NetworkState FakeCall::GetNetworkState(webrtc::MediaType media) const {
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switch (media) {
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case webrtc::MediaType::AUDIO:
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return audio_network_state_;
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case webrtc::MediaType::VIDEO:
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return video_network_state_;
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case webrtc::MediaType::DATA:
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case webrtc::MediaType::ANY:
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ADD_FAILURE() << "GetNetworkState called with unknown parameter.";
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return webrtc::kNetworkDown;
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}
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// Even though all the values for the enum class are listed above,the compiler
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// will emit a warning as the method may be called with a value outside of the
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// valid enum range, unless this case is also handled.
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ADD_FAILURE() << "GetNetworkState called with unknown parameter.";
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return webrtc::kNetworkDown;
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}
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webrtc::AudioSendStream* FakeCall::CreateAudioSendStream(
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const webrtc::AudioSendStream::Config& config) {
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FakeAudioSendStream* fake_stream = new FakeAudioSendStream(config);
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audio_send_streams_.push_back(fake_stream);
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++num_created_send_streams_;
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return fake_stream;
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}
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void FakeCall::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
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auto it = std::find(audio_send_streams_.begin(),
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audio_send_streams_.end(),
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static_cast<FakeAudioSendStream*>(send_stream));
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if (it == audio_send_streams_.end()) {
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ADD_FAILURE() << "DestroyAudioSendStream called with unknown parameter.";
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} else {
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delete *it;
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audio_send_streams_.erase(it);
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}
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}
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webrtc::AudioReceiveStream* FakeCall::CreateAudioReceiveStream(
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const webrtc::AudioReceiveStream::Config& config) {
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audio_receive_streams_.push_back(new FakeAudioReceiveStream(config));
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++num_created_receive_streams_;
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return audio_receive_streams_.back();
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}
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void FakeCall::DestroyAudioReceiveStream(
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webrtc::AudioReceiveStream* receive_stream) {
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auto it = std::find(audio_receive_streams_.begin(),
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audio_receive_streams_.end(),
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static_cast<FakeAudioReceiveStream*>(receive_stream));
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if (it == audio_receive_streams_.end()) {
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ADD_FAILURE() << "DestroyAudioReceiveStream called with unknown parameter.";
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} else {
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delete *it;
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audio_receive_streams_.erase(it);
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}
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}
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webrtc::VideoSendStream* FakeCall::CreateVideoSendStream(
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const webrtc::VideoSendStream::Config& config,
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const webrtc::VideoEncoderConfig& encoder_config) {
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FakeVideoSendStream* fake_stream =
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new FakeVideoSendStream(config, encoder_config);
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video_send_streams_.push_back(fake_stream);
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++num_created_send_streams_;
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return fake_stream;
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}
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void FakeCall::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) {
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auto it = std::find(video_send_streams_.begin(),
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video_send_streams_.end(),
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static_cast<FakeVideoSendStream*>(send_stream));
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if (it == video_send_streams_.end()) {
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ADD_FAILURE() << "DestroyVideoSendStream called with unknown parameter.";
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} else {
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delete *it;
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video_send_streams_.erase(it);
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|
}
|
||
|
|
}
|
||
|
|
|
||
|
|
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;
|
||
|
|
}
|
||
|
|
}
|
||
|
|
|
||
|
|
} // namespace cricket
|