Moves fake media engine implementation to cc file.

This CL moves the implementations of the fake media engine from
fakemediaengine.h to fakemediaengine.cc.

Bug: webrtc:9883
Change-Id: I0f91ef63a366abe9638fc885bc14aba7dd5436aa
Reviewed-on: https://webrtc-review.googlesource.com/c/106923
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25260}
This commit is contained in:
Sebastian Jansson 2018-10-18 17:03:30 +02:00 committed by Commit Bot
parent 7dc97740ea
commit cb06cac5b4
3 changed files with 782 additions and 500 deletions

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@ -414,6 +414,7 @@ if (rtc_include_tests) {
sources = [
"base/fakeframesource.cc",
"base/fakeframesource.h",
"base/fakemediaengine.cc",
"base/fakemediaengine.h",
"base/fakenetworkinterface.h",
"base/fakertp.cc",

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@ -0,0 +1,637 @@
/*
* Copyright 2018 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/base/fakemediaengine.h"
#include <utility>
#include "rtc_base/checks.h"
namespace cricket {
FakeVoiceMediaChannel::DtmfInfo::DtmfInfo(uint32_t ssrc,
int event_code,
int duration)
: ssrc(ssrc), event_code(event_code), duration(duration) {}
FakeVoiceMediaChannel::VoiceChannelAudioSink::VoiceChannelAudioSink(
AudioSource* source)
: source_(source) {
source_->SetSink(this);
}
FakeVoiceMediaChannel::VoiceChannelAudioSink::~VoiceChannelAudioSink() {
if (source_) {
source_->SetSink(nullptr);
}
}
void FakeVoiceMediaChannel::VoiceChannelAudioSink::OnData(
const void* audio_data,
int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames) {}
void FakeVoiceMediaChannel::VoiceChannelAudioSink::OnClose() {
source_ = nullptr;
}
AudioSource* FakeVoiceMediaChannel::VoiceChannelAudioSink::source() const {
return source_;
}
FakeVoiceMediaChannel::FakeVoiceMediaChannel(FakeVoiceEngine* engine,
const AudioOptions& options)
: engine_(engine), max_bps_(-1) {
output_scalings_[0] = 1.0; // For default channel.
SetOptions(options);
}
FakeVoiceMediaChannel::~FakeVoiceMediaChannel() {
if (engine_) {
engine_->UnregisterChannel(this);
}
}
const std::vector<AudioCodec>& FakeVoiceMediaChannel::recv_codecs() const {
return recv_codecs_;
}
const std::vector<AudioCodec>& FakeVoiceMediaChannel::send_codecs() const {
return send_codecs_;
}
const std::vector<AudioCodec>& FakeVoiceMediaChannel::codecs() const {
return send_codecs();
}
const std::vector<FakeVoiceMediaChannel::DtmfInfo>&
FakeVoiceMediaChannel::dtmf_info_queue() const {
return dtmf_info_queue_;
}
const AudioOptions& FakeVoiceMediaChannel::options() const {
return options_;
}
int FakeVoiceMediaChannel::max_bps() const {
return max_bps_;
}
bool FakeVoiceMediaChannel::SetSendParameters(
const AudioSendParameters& params) {
set_send_rtcp_parameters(params.rtcp);
return (SetSendCodecs(params.codecs) &&
SetSendRtpHeaderExtensions(params.extensions) &&
SetMaxSendBandwidth(params.max_bandwidth_bps) &&
SetOptions(params.options));
}
bool FakeVoiceMediaChannel::SetRecvParameters(
const AudioRecvParameters& params) {
set_recv_rtcp_parameters(params.rtcp);
return (SetRecvCodecs(params.codecs) &&
SetRecvRtpHeaderExtensions(params.extensions));
}
void FakeVoiceMediaChannel::SetPlayout(bool playout) {
set_playout(playout);
}
void FakeVoiceMediaChannel::SetSend(bool send) {
set_sending(send);
}
bool FakeVoiceMediaChannel::SetAudioSend(uint32_t ssrc,
bool enable,
const AudioOptions* options,
AudioSource* source) {
if (!SetLocalSource(ssrc, source)) {
return false;
}
if (!RtpHelper<VoiceMediaChannel>::MuteStream(ssrc, !enable)) {
return false;
}
if (enable && options) {
return SetOptions(*options);
}
return true;
}
bool FakeVoiceMediaChannel::HasSource(uint32_t ssrc) const {
return local_sinks_.find(ssrc) != local_sinks_.end();
}
bool FakeVoiceMediaChannel::AddRecvStream(const StreamParams& sp) {
if (!RtpHelper<VoiceMediaChannel>::AddRecvStream(sp))
return false;
output_scalings_[sp.first_ssrc()] = 1.0;
return true;
}
bool FakeVoiceMediaChannel::RemoveRecvStream(uint32_t ssrc) {
if (!RtpHelper<VoiceMediaChannel>::RemoveRecvStream(ssrc))
return false;
output_scalings_.erase(ssrc);
return true;
}
bool FakeVoiceMediaChannel::CanInsertDtmf() {
for (std::vector<AudioCodec>::const_iterator it = send_codecs_.begin();
it != send_codecs_.end(); ++it) {
// Find the DTMF telephone event "codec".
if (_stricmp(it->name.c_str(), "telephone-event") == 0) {
return true;
}
}
return false;
}
bool FakeVoiceMediaChannel::InsertDtmf(uint32_t ssrc,
int event_code,
int duration) {
dtmf_info_queue_.push_back(DtmfInfo(ssrc, event_code, duration));
return true;
}
bool FakeVoiceMediaChannel::SetOutputVolume(uint32_t ssrc, double volume) {
if (0 == ssrc) {
std::map<uint32_t, double>::iterator it;
for (it = output_scalings_.begin(); it != output_scalings_.end(); ++it) {
it->second = volume;
}
return true;
} else if (output_scalings_.find(ssrc) != output_scalings_.end()) {
output_scalings_[ssrc] = volume;
return true;
}
return false;
}
bool FakeVoiceMediaChannel::GetOutputVolume(uint32_t ssrc, double* volume) {
if (output_scalings_.find(ssrc) == output_scalings_.end())
return false;
*volume = output_scalings_[ssrc];
return true;
}
bool FakeVoiceMediaChannel::GetStats(VoiceMediaInfo* info) {
return false;
}
void FakeVoiceMediaChannel::SetRawAudioSink(
uint32_t ssrc,
std::unique_ptr<webrtc::AudioSinkInterface> sink) {
sink_ = std::move(sink);
}
std::vector<webrtc::RtpSource> FakeVoiceMediaChannel::GetSources(
uint32_t ssrc) const {
return std::vector<webrtc::RtpSource>();
}
bool FakeVoiceMediaChannel::SetRecvCodecs(
const std::vector<AudioCodec>& codecs) {
if (fail_set_recv_codecs()) {
// Fake the failure in SetRecvCodecs.
return false;
}
recv_codecs_ = codecs;
return true;
}
bool FakeVoiceMediaChannel::SetSendCodecs(
const std::vector<AudioCodec>& codecs) {
if (fail_set_send_codecs()) {
// Fake the failure in SetSendCodecs.
return false;
}
send_codecs_ = codecs;
return true;
}
bool FakeVoiceMediaChannel::SetMaxSendBandwidth(int bps) {
max_bps_ = bps;
return true;
}
bool FakeVoiceMediaChannel::SetOptions(const AudioOptions& options) {
// Does a "merge" of current options and set options.
options_.SetAll(options);
return true;
}
bool FakeVoiceMediaChannel::SetLocalSource(uint32_t ssrc, AudioSource* source) {
auto it = local_sinks_.find(ssrc);
if (source) {
if (it != local_sinks_.end()) {
RTC_CHECK(it->second->source() == source);
} else {
local_sinks_.insert(std::make_pair(
ssrc, absl::make_unique<VoiceChannelAudioSink>(source)));
}
} else {
if (it != local_sinks_.end()) {
local_sinks_.erase(it);
}
}
return true;
}
bool CompareDtmfInfo(const FakeVoiceMediaChannel::DtmfInfo& info,
uint32_t ssrc,
int event_code,
int duration) {
return (info.duration == duration && info.event_code == event_code &&
info.ssrc == ssrc);
}
FakeVideoMediaChannel::FakeVideoMediaChannel(FakeVideoEngine* engine,
const VideoOptions& options)
: engine_(engine), max_bps_(-1) {
SetOptions(options);
}
FakeVideoMediaChannel::~FakeVideoMediaChannel() {
if (engine_) {
engine_->UnregisterChannel(this);
}
}
const std::vector<VideoCodec>& FakeVideoMediaChannel::recv_codecs() const {
return recv_codecs_;
}
const std::vector<VideoCodec>& FakeVideoMediaChannel::send_codecs() const {
return send_codecs_;
}
const std::vector<VideoCodec>& FakeVideoMediaChannel::codecs() const {
return send_codecs();
}
bool FakeVideoMediaChannel::rendering() const {
return playout();
}
const VideoOptions& FakeVideoMediaChannel::options() const {
return options_;
}
const std::map<uint32_t, rtc::VideoSinkInterface<webrtc::VideoFrame>*>&
FakeVideoMediaChannel::sinks() const {
return sinks_;
}
int FakeVideoMediaChannel::max_bps() const {
return max_bps_;
}
bool FakeVideoMediaChannel::SetSendParameters(
const VideoSendParameters& params) {
set_send_rtcp_parameters(params.rtcp);
return (SetSendCodecs(params.codecs) &&
SetSendRtpHeaderExtensions(params.extensions) &&
SetMaxSendBandwidth(params.max_bandwidth_bps));
}
bool FakeVideoMediaChannel::SetRecvParameters(
const VideoRecvParameters& params) {
set_recv_rtcp_parameters(params.rtcp);
return (SetRecvCodecs(params.codecs) &&
SetRecvRtpHeaderExtensions(params.extensions));
}
bool FakeVideoMediaChannel::AddSendStream(const StreamParams& sp) {
return RtpHelper<VideoMediaChannel>::AddSendStream(sp);
}
bool FakeVideoMediaChannel::RemoveSendStream(uint32_t ssrc) {
return RtpHelper<VideoMediaChannel>::RemoveSendStream(ssrc);
}
bool FakeVideoMediaChannel::GetSendCodec(VideoCodec* send_codec) {
if (send_codecs_.empty()) {
return false;
}
*send_codec = send_codecs_[0];
return true;
}
bool FakeVideoMediaChannel::SetSink(
uint32_t ssrc,
rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) {
if (ssrc != 0 && sinks_.find(ssrc) == sinks_.end()) {
return false;
}
if (ssrc != 0) {
sinks_[ssrc] = sink;
}
return true;
}
bool FakeVideoMediaChannel::HasSink(uint32_t ssrc) const {
return sinks_.find(ssrc) != sinks_.end() && sinks_.at(ssrc) != nullptr;
}
bool FakeVideoMediaChannel::SetSend(bool send) {
return set_sending(send);
}
bool FakeVideoMediaChannel::SetVideoSend(
uint32_t ssrc,
const VideoOptions* options,
rtc::VideoSourceInterface<webrtc::VideoFrame>* source) {
if (options) {
if (!SetOptions(*options)) {
return false;
}
}
sources_[ssrc] = source;
return true;
}
bool FakeVideoMediaChannel::HasSource(uint32_t ssrc) const {
return sources_.find(ssrc) != sources_.end() && sources_.at(ssrc) != nullptr;
}
bool FakeVideoMediaChannel::AddRecvStream(const StreamParams& sp) {
if (!RtpHelper<VideoMediaChannel>::AddRecvStream(sp))
return false;
sinks_[sp.first_ssrc()] = NULL;
return true;
}
bool FakeVideoMediaChannel::RemoveRecvStream(uint32_t ssrc) {
if (!RtpHelper<VideoMediaChannel>::RemoveRecvStream(ssrc))
return false;
sinks_.erase(ssrc);
return true;
}
void FakeVideoMediaChannel::FillBitrateInfo(BandwidthEstimationInfo* bwe_info) {
}
bool FakeVideoMediaChannel::GetStats(VideoMediaInfo* info) {
return false;
}
std::vector<webrtc::RtpSource> FakeVideoMediaChannel::GetSources(
uint32_t ssrc) const {
return {};
}
bool FakeVideoMediaChannel::SetRecvCodecs(
const std::vector<VideoCodec>& codecs) {
if (fail_set_recv_codecs()) {
// Fake the failure in SetRecvCodecs.
return false;
}
recv_codecs_ = codecs;
return true;
}
bool FakeVideoMediaChannel::SetSendCodecs(
const std::vector<VideoCodec>& codecs) {
if (fail_set_send_codecs()) {
// Fake the failure in SetSendCodecs.
return false;
}
send_codecs_ = codecs;
return true;
}
bool FakeVideoMediaChannel::SetOptions(const VideoOptions& options) {
options_ = options;
return true;
}
bool FakeVideoMediaChannel::SetMaxSendBandwidth(int bps) {
max_bps_ = bps;
return true;
}
FakeDataMediaChannel::FakeDataMediaChannel(void* unused,
const DataOptions& options)
: send_blocked_(false), max_bps_(-1) {}
FakeDataMediaChannel::~FakeDataMediaChannel() {}
const std::vector<DataCodec>& FakeDataMediaChannel::recv_codecs() const {
return recv_codecs_;
}
const std::vector<DataCodec>& FakeDataMediaChannel::send_codecs() const {
return send_codecs_;
}
const std::vector<DataCodec>& FakeDataMediaChannel::codecs() const {
return send_codecs();
}
int FakeDataMediaChannel::max_bps() const {
return max_bps_;
}
bool FakeDataMediaChannel::SetSendParameters(const DataSendParameters& params) {
set_send_rtcp_parameters(params.rtcp);
return (SetSendCodecs(params.codecs) &&
SetMaxSendBandwidth(params.max_bandwidth_bps));
}
bool FakeDataMediaChannel::SetRecvParameters(const DataRecvParameters& params) {
set_recv_rtcp_parameters(params.rtcp);
return SetRecvCodecs(params.codecs);
}
bool FakeDataMediaChannel::SetSend(bool send) {
return set_sending(send);
}
bool FakeDataMediaChannel::SetReceive(bool receive) {
set_playout(receive);
return true;
}
bool FakeDataMediaChannel::AddRecvStream(const StreamParams& sp) {
if (!RtpHelper<DataMediaChannel>::AddRecvStream(sp))
return false;
return true;
}
bool FakeDataMediaChannel::RemoveRecvStream(uint32_t ssrc) {
if (!RtpHelper<DataMediaChannel>::RemoveRecvStream(ssrc))
return false;
return true;
}
bool FakeDataMediaChannel::SendData(const SendDataParams& params,
const rtc::CopyOnWriteBuffer& payload,
SendDataResult* result) {
if (send_blocked_) {
*result = SDR_BLOCK;
return false;
} else {
last_sent_data_params_ = params;
last_sent_data_ = std::string(payload.data<char>(), payload.size());
return true;
}
}
SendDataParams FakeDataMediaChannel::last_sent_data_params() {
return last_sent_data_params_;
}
std::string FakeDataMediaChannel::last_sent_data() {
return last_sent_data_;
}
bool FakeDataMediaChannel::is_send_blocked() {
return send_blocked_;
}
void FakeDataMediaChannel::set_send_blocked(bool blocked) {
send_blocked_ = blocked;
}
bool FakeDataMediaChannel::SetRecvCodecs(const std::vector<DataCodec>& codecs) {
if (fail_set_recv_codecs()) {
// Fake the failure in SetRecvCodecs.
return false;
}
recv_codecs_ = codecs;
return true;
}
bool FakeDataMediaChannel::SetSendCodecs(const std::vector<DataCodec>& codecs) {
if (fail_set_send_codecs()) {
// Fake the failure in SetSendCodecs.
return false;
}
send_codecs_ = codecs;
return true;
}
bool FakeDataMediaChannel::SetMaxSendBandwidth(int bps) {
max_bps_ = bps;
return true;
}
FakeBaseEngine::FakeBaseEngine()
: options_changed_(false), fail_create_channel_(false) {}
void FakeBaseEngine::set_fail_create_channel(bool fail) {
fail_create_channel_ = fail;
}
void FakeBaseEngine::set_rtp_header_extensions(
const std::vector<webrtc::RtpExtension>& extensions) {
capabilities_.header_extensions = extensions;
}
void FakeBaseEngine::set_rtp_header_extensions(
const std::vector<RtpHeaderExtension>& extensions) {
for (const cricket::RtpHeaderExtension& ext : extensions) {
RtpExtension webrtc_ext;
webrtc_ext.uri = ext.uri;
webrtc_ext.id = ext.id;
capabilities_.header_extensions.push_back(webrtc_ext);
}
}
FakeVoiceEngine::FakeVoiceEngine() {
// Add a fake audio codec. Note that the name must not be "" as there are
// sanity checks against that.
codecs_.push_back(AudioCodec(101, "fake_audio_codec", 0, 0, 1));
}
RtpCapabilities FakeVoiceEngine::GetCapabilities() const {
return capabilities_;
}
void FakeVoiceEngine::Init() {}
rtc::scoped_refptr<webrtc::AudioState> FakeVoiceEngine::GetAudioState() const {
return rtc::scoped_refptr<webrtc::AudioState>();
}
VoiceMediaChannel* FakeVoiceEngine::CreateChannel(
webrtc::Call* call,
const MediaConfig& config,
const AudioOptions& options,
const webrtc::CryptoOptions& crypto_options) {
if (fail_create_channel_) {
return nullptr;
}
FakeVoiceMediaChannel* ch = new FakeVoiceMediaChannel(this, options);
channels_.push_back(ch);
return ch;
}
FakeVoiceMediaChannel* FakeVoiceEngine::GetChannel(size_t index) {
return (channels_.size() > index) ? channels_[index] : NULL;
}
void FakeVoiceEngine::UnregisterChannel(VoiceMediaChannel* channel) {
channels_.erase(std::find(channels_.begin(), channels_.end(), channel));
}
const std::vector<AudioCodec>& FakeVoiceEngine::send_codecs() const {
return codecs_;
}
const std::vector<AudioCodec>& FakeVoiceEngine::recv_codecs() const {
return codecs_;
}
void FakeVoiceEngine::SetCodecs(const std::vector<AudioCodec>& codecs) {
codecs_ = codecs;
}
int FakeVoiceEngine::GetInputLevel() {
return 0;
}
bool FakeVoiceEngine::StartAecDump(rtc::PlatformFile file,
int64_t max_size_bytes) {
return false;
}
void FakeVoiceEngine::StopAecDump() {}
bool FakeVoiceEngine::StartRtcEventLog(rtc::PlatformFile file,
int64_t max_size_bytes) {
return false;
}
void FakeVoiceEngine::StopRtcEventLog() {}
FakeVideoEngine::FakeVideoEngine() : capture_(false) {
// Add a fake video codec. Note that the name must not be "" as there are
// sanity checks against that.
codecs_.push_back(VideoCodec(0, "fake_video_codec"));
}
RtpCapabilities FakeVideoEngine::GetCapabilities() const {
return capabilities_;
}
bool FakeVideoEngine::SetOptions(const VideoOptions& options) {
options_ = options;
options_changed_ = true;
return true;
}
VideoMediaChannel* FakeVideoEngine::CreateChannel(
webrtc::Call* call,
const MediaConfig& config,
const VideoOptions& options,
const webrtc::CryptoOptions& crypto_options) {
if (fail_create_channel_) {
return nullptr;
}
FakeVideoMediaChannel* ch = new FakeVideoMediaChannel(this, options);
channels_.emplace_back(ch);
return ch;
}
FakeVideoMediaChannel* FakeVideoEngine::GetChannel(size_t index) {
return (channels_.size() > index) ? channels_[index] : nullptr;
}
void FakeVideoEngine::UnregisterChannel(VideoMediaChannel* channel) {
auto it = std::find(channels_.begin(), channels_.end(), channel);
RTC_DCHECK(it != channels_.end());
channels_.erase(it);
}
std::vector<VideoCodec> FakeVideoEngine::codecs() const {
return codecs_;
}
void FakeVideoEngine::SetCodecs(const std::vector<VideoCodec> codecs) {
codecs_ = codecs;
}
bool FakeVideoEngine::SetCapture(bool capture) {
capture_ = capture;
return true;
}
FakeMediaEngine::FakeMediaEngine()
: CompositeMediaEngine<FakeVoiceEngine, FakeVideoEngine>(std::tuple<>(),
std::tuple<>()),
voice_(&voice()),
video_(&video()) {}
FakeMediaEngine::~FakeMediaEngine() {}
void FakeMediaEngine::SetAudioCodecs(const std::vector<AudioCodec>& codecs) {
voice_->SetCodecs(codecs);
}
void FakeMediaEngine::SetVideoCodecs(const std::vector<VideoCodec>& codecs) {
video_->SetCodecs(codecs);
}
void FakeMediaEngine::SetAudioRtpHeaderExtensions(
const std::vector<webrtc::RtpExtension>& extensions) {
voice_->set_rtp_header_extensions(extensions);
}
void FakeMediaEngine::SetVideoRtpHeaderExtensions(
const std::vector<webrtc::RtpExtension>& extensions) {
video_->set_rtp_header_extensions(extensions);
}
void FakeMediaEngine::SetAudioRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) {
voice_->set_rtp_header_extensions(extensions);
}
void FakeMediaEngine::SetVideoRtpHeaderExtensions(
const std::vector<RtpHeaderExtension>& extensions) {
video_->set_rtp_header_extensions(extensions);
}
FakeVoiceMediaChannel* FakeMediaEngine::GetVoiceChannel(size_t index) {
return voice_->GetChannel(index);
}
FakeVideoMediaChannel* FakeMediaEngine::GetVideoChannel(size_t index) {
return video_->GetChannel(index);
}
bool FakeMediaEngine::capture() const {
return video_->capture_;
}
bool FakeMediaEngine::options_changed() const {
return video_->options_changed_;
}
void FakeMediaEngine::clear_options_changed() {
video_->options_changed_ = false;
}
void FakeMediaEngine::set_fail_create_channel(bool fail) {
voice_->set_fail_create_channel(fail);
video_->set_fail_create_channel(fail);
}
DataMediaChannel* FakeDataEngine::CreateChannel(const MediaConfig& config) {
FakeDataMediaChannel* ch = new FakeDataMediaChannel(this, DataOptions());
channels_.push_back(ch);
return ch;
}
FakeDataMediaChannel* FakeDataEngine::GetChannel(size_t index) {
return (channels_.size() > index) ? channels_[index] : NULL;
}
void FakeDataEngine::UnregisterChannel(DataMediaChannel* channel) {
channels_.erase(std::find(channels_.begin(), channels_.end(), channel));
}
void FakeDataEngine::SetDataCodecs(const std::vector<DataCodec>& data_codecs) {
data_codecs_ = data_codecs;
}
const std::vector<DataCodec>& FakeDataEngine::data_codecs() {
return data_codecs_;
}
} // namespace cricket

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@ -17,10 +17,8 @@
#include <set>
#include <string>
#include <tuple>
#include <utility>
#include <vector>
#include "absl/memory/memory.h"
#include "api/call/audio_sink.h"
#include "media/base/audiosource.h"
#include "media/base/mediaengine.h"
@ -28,10 +26,8 @@
#include "media/base/streamparams.h"
#include "media/engine/webrtcvideoengine.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "rtc_base/checks.h"
#include "rtc_base/copyonwritebuffer.h"
#include "rtc_base/networkroute.h"
#include "rtc_base/stringutils.h"
using webrtc::RtpExtension;
@ -312,187 +308,72 @@ class RtpHelper : public Base {
class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
public:
struct DtmfInfo {
DtmfInfo(uint32_t ssrc, int event_code, int duration)
: ssrc(ssrc), event_code(event_code), duration(duration) {}
DtmfInfo(uint32_t ssrc, int event_code, int duration);
uint32_t ssrc;
int event_code;
int duration;
};
explicit FakeVoiceMediaChannel(FakeVoiceEngine* engine,
const AudioOptions& options)
: engine_(engine), max_bps_(-1) {
output_scalings_[0] = 1.0; // For default channel.
SetOptions(options);
}
const AudioOptions& options);
~FakeVoiceMediaChannel();
const std::vector<AudioCodec>& recv_codecs() const { return recv_codecs_; }
const std::vector<AudioCodec>& send_codecs() const { return send_codecs_; }
const std::vector<AudioCodec>& codecs() const { return send_codecs(); }
const std::vector<DtmfInfo>& dtmf_info_queue() const {
return dtmf_info_queue_;
}
const AudioOptions& options() const { return options_; }
int max_bps() const { return max_bps_; }
virtual bool SetSendParameters(const AudioSendParameters& params) {
set_send_rtcp_parameters(params.rtcp);
return (SetSendCodecs(params.codecs) &&
SetSendRtpHeaderExtensions(params.extensions) &&
SetMaxSendBandwidth(params.max_bandwidth_bps) &&
SetOptions(params.options));
}
const std::vector<AudioCodec>& recv_codecs() const;
const std::vector<AudioCodec>& send_codecs() const;
const std::vector<AudioCodec>& codecs() const;
const std::vector<DtmfInfo>& dtmf_info_queue() const;
const AudioOptions& options() const;
int max_bps() const;
bool SetSendParameters(const AudioSendParameters& params) override;
virtual bool SetRecvParameters(const AudioRecvParameters& params) {
set_recv_rtcp_parameters(params.rtcp);
return (SetRecvCodecs(params.codecs) &&
SetRecvRtpHeaderExtensions(params.extensions));
}
bool SetRecvParameters(const AudioRecvParameters& params) override;
virtual void SetPlayout(bool playout) { set_playout(playout); }
virtual void SetSend(bool send) { set_sending(send); }
virtual bool SetAudioSend(uint32_t ssrc,
bool enable,
const AudioOptions* options,
AudioSource* source) {
if (!SetLocalSource(ssrc, source)) {
return false;
}
if (!RtpHelper<VoiceMediaChannel>::MuteStream(ssrc, !enable)) {
return false;
}
if (enable && options) {
return SetOptions(*options);
}
return true;
}
void SetPlayout(bool playout) override;
void SetSend(bool send) override;
bool SetAudioSend(uint32_t ssrc,
bool enable,
const AudioOptions* options,
AudioSource* source) override;
bool HasSource(uint32_t ssrc) const {
return local_sinks_.find(ssrc) != local_sinks_.end();
}
bool HasSource(uint32_t ssrc) const;
virtual bool AddRecvStream(const StreamParams& sp) {
if (!RtpHelper<VoiceMediaChannel>::AddRecvStream(sp))
return false;
output_scalings_[sp.first_ssrc()] = 1.0;
return true;
}
virtual bool RemoveRecvStream(uint32_t ssrc) {
if (!RtpHelper<VoiceMediaChannel>::RemoveRecvStream(ssrc))
return false;
output_scalings_.erase(ssrc);
return true;
}
bool AddRecvStream(const StreamParams& sp) override;
bool RemoveRecvStream(uint32_t ssrc) override;
virtual bool CanInsertDtmf() {
for (std::vector<AudioCodec>::const_iterator it = send_codecs_.begin();
it != send_codecs_.end(); ++it) {
// Find the DTMF telephone event "codec".
if (_stricmp(it->name.c_str(), "telephone-event") == 0) {
return true;
}
}
return false;
}
virtual bool InsertDtmf(uint32_t ssrc, int event_code, int duration) {
dtmf_info_queue_.push_back(DtmfInfo(ssrc, event_code, duration));
return true;
}
bool CanInsertDtmf() override;
bool InsertDtmf(uint32_t ssrc, int event_code, int duration) override;
virtual bool SetOutputVolume(uint32_t ssrc, double volume) {
if (0 == ssrc) {
std::map<uint32_t, double>::iterator it;
for (it = output_scalings_.begin(); it != output_scalings_.end(); ++it) {
it->second = volume;
}
return true;
} else if (output_scalings_.find(ssrc) != output_scalings_.end()) {
output_scalings_[ssrc] = volume;
return true;
}
return false;
}
bool GetOutputVolume(uint32_t ssrc, double* volume) {
if (output_scalings_.find(ssrc) == output_scalings_.end())
return false;
*volume = output_scalings_[ssrc];
return true;
}
bool SetOutputVolume(uint32_t ssrc, double volume) override;
bool GetOutputVolume(uint32_t ssrc, double* volume);
virtual bool GetStats(VoiceMediaInfo* info) { return false; }
bool GetStats(VoiceMediaInfo* info) override;
virtual void SetRawAudioSink(
void SetRawAudioSink(
uint32_t ssrc,
std::unique_ptr<webrtc::AudioSinkInterface> sink) {
sink_ = std::move(sink);
}
std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
virtual std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const {
return std::vector<webrtc::RtpSource>();
}
std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override;
private:
class VoiceChannelAudioSink : public AudioSource::Sink {
public:
explicit VoiceChannelAudioSink(AudioSource* source) : source_(source) {
source_->SetSink(this);
}
virtual ~VoiceChannelAudioSink() {
if (source_) {
source_->SetSink(nullptr);
}
}
explicit VoiceChannelAudioSink(AudioSource* source);
~VoiceChannelAudioSink() override;
void OnData(const void* audio_data,
int bits_per_sample,
int sample_rate,
size_t number_of_channels,
size_t number_of_frames) override {}
void OnClose() override { source_ = nullptr; }
AudioSource* source() const { return source_; }
size_t number_of_frames) override;
void OnClose() override;
AudioSource* source() const;
private:
AudioSource* source_;
};
bool SetRecvCodecs(const std::vector<AudioCodec>& codecs) {
if (fail_set_recv_codecs()) {
// Fake the failure in SetRecvCodecs.
return false;
}
recv_codecs_ = codecs;
return true;
}
bool SetSendCodecs(const std::vector<AudioCodec>& codecs) {
if (fail_set_send_codecs()) {
// Fake the failure in SetSendCodecs.
return false;
}
send_codecs_ = codecs;
return true;
}
bool SetMaxSendBandwidth(int bps) {
max_bps_ = bps;
return true;
}
bool SetOptions(const AudioOptions& options) {
// Does a "merge" of current options and set options.
options_.SetAll(options);
return true;
}
bool SetLocalSource(uint32_t ssrc, AudioSource* source) {
auto it = local_sinks_.find(ssrc);
if (source) {
if (it != local_sinks_.end()) {
RTC_CHECK(it->second->source() == source);
} else {
local_sinks_.insert(std::make_pair(
ssrc, absl::make_unique<VoiceChannelAudioSink>(source)));
}
} else {
if (it != local_sinks_.end()) {
local_sinks_.erase(it);
}
}
return true;
}
bool SetRecvCodecs(const std::vector<AudioCodec>& codecs);
bool SetSendCodecs(const std::vector<AudioCodec>& codecs);
bool SetMaxSendBandwidth(int bps);
bool SetOptions(const AudioOptions& options);
bool SetLocalSource(uint32_t ssrc, AudioSource* source);
FakeVoiceEngine* engine_;
std::vector<AudioCodec> recv_codecs_;
@ -506,136 +387,55 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
};
// A helper function to compare the FakeVoiceMediaChannel::DtmfInfo.
inline bool CompareDtmfInfo(const FakeVoiceMediaChannel::DtmfInfo& info,
uint32_t ssrc,
int event_code,
int duration) {
return (info.duration == duration && info.event_code == event_code &&
info.ssrc == ssrc);
}
bool CompareDtmfInfo(const FakeVoiceMediaChannel::DtmfInfo& info,
uint32_t ssrc,
int event_code,
int duration);
class FakeVideoMediaChannel : public RtpHelper<VideoMediaChannel> {
public:
FakeVideoMediaChannel(FakeVideoEngine* engine, const VideoOptions& options)
: engine_(engine), max_bps_(-1) {
SetOptions(options);
}
FakeVideoMediaChannel(FakeVideoEngine* engine, const VideoOptions& options);
~FakeVideoMediaChannel();
const std::vector<VideoCodec>& recv_codecs() const { return recv_codecs_; }
const std::vector<VideoCodec>& send_codecs() const { return send_codecs_; }
const std::vector<VideoCodec>& codecs() const { return send_codecs(); }
bool rendering() const { return playout(); }
const VideoOptions& options() const { return options_; }
const std::vector<VideoCodec>& recv_codecs() const;
const std::vector<VideoCodec>& send_codecs() const;
const std::vector<VideoCodec>& codecs() const;
bool rendering() const;
const VideoOptions& options() const;
const std::map<uint32_t, rtc::VideoSinkInterface<webrtc::VideoFrame>*>&
sinks() const {
return sinks_;
}
int max_bps() const { return max_bps_; }
bool SetSendParameters(const VideoSendParameters& params) override {
set_send_rtcp_parameters(params.rtcp);
return (SetSendCodecs(params.codecs) &&
SetSendRtpHeaderExtensions(params.extensions) &&
SetMaxSendBandwidth(params.max_bandwidth_bps));
}
bool SetRecvParameters(const VideoRecvParameters& params) override {
set_recv_rtcp_parameters(params.rtcp);
return (SetRecvCodecs(params.codecs) &&
SetRecvRtpHeaderExtensions(params.extensions));
}
bool AddSendStream(const StreamParams& sp) override {
return RtpHelper<VideoMediaChannel>::AddSendStream(sp);
}
bool RemoveSendStream(uint32_t ssrc) override {
return RtpHelper<VideoMediaChannel>::RemoveSendStream(ssrc);
}
sinks() const;
int max_bps() const;
bool SetSendParameters(const VideoSendParameters& params) override;
bool SetRecvParameters(const VideoRecvParameters& params) override;
bool AddSendStream(const StreamParams& sp) override;
bool RemoveSendStream(uint32_t ssrc) override;
bool GetSendCodec(VideoCodec* send_codec) override {
if (send_codecs_.empty()) {
return false;
}
*send_codec = send_codecs_[0];
return true;
}
bool GetSendCodec(VideoCodec* send_codec) override;
bool SetSink(uint32_t ssrc,
rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override {
if (ssrc != 0 && sinks_.find(ssrc) == sinks_.end()) {
return false;
}
if (ssrc != 0) {
sinks_[ssrc] = sink;
}
return true;
}
bool HasSink(uint32_t ssrc) const {
return sinks_.find(ssrc) != sinks_.end() && sinks_.at(ssrc) != nullptr;
}
rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override;
bool HasSink(uint32_t ssrc) const;
bool SetSend(bool send) override { return set_sending(send); }
bool SetSend(bool send) override;
bool SetVideoSend(
uint32_t ssrc,
const VideoOptions* options,
rtc::VideoSourceInterface<webrtc::VideoFrame>* source) override {
if (options) {
if (!SetOptions(*options)) {
return false;
}
}
sources_[ssrc] = source;
return true;
}
rtc::VideoSourceInterface<webrtc::VideoFrame>* source) override;
bool HasSource(uint32_t ssrc) const {
return sources_.find(ssrc) != sources_.end() &&
sources_.at(ssrc) != nullptr;
}
bool AddRecvStream(const StreamParams& sp) override {
if (!RtpHelper<VideoMediaChannel>::AddRecvStream(sp))
return false;
sinks_[sp.first_ssrc()] = NULL;
return true;
}
bool RemoveRecvStream(uint32_t ssrc) override {
if (!RtpHelper<VideoMediaChannel>::RemoveRecvStream(ssrc))
return false;
sinks_.erase(ssrc);
return true;
}
bool HasSource(uint32_t ssrc) const;
bool AddRecvStream(const StreamParams& sp) override;
bool RemoveRecvStream(uint32_t ssrc) override;
void FillBitrateInfo(BandwidthEstimationInfo* bwe_info) override {}
bool GetStats(VideoMediaInfo* info) override { return false; }
void FillBitrateInfo(BandwidthEstimationInfo* bwe_info) override;
bool GetStats(VideoMediaInfo* info) override;
std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override {
return {};
}
std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override;
private:
bool SetRecvCodecs(const std::vector<VideoCodec>& codecs) {
if (fail_set_recv_codecs()) {
// Fake the failure in SetRecvCodecs.
return false;
}
recv_codecs_ = codecs;
return true;
}
bool SetSendCodecs(const std::vector<VideoCodec>& codecs) {
if (fail_set_send_codecs()) {
// Fake the failure in SetSendCodecs.
return false;
}
send_codecs_ = codecs;
return true;
}
bool SetOptions(const VideoOptions& options) {
options_ = options;
return true;
}
bool SetMaxSendBandwidth(int bps) {
max_bps_ = bps;
return true;
}
bool SetRecvCodecs(const std::vector<VideoCodec>& codecs);
bool SetSendCodecs(const std::vector<VideoCodec>& codecs);
bool SetOptions(const VideoOptions& options);
bool SetMaxSendBandwidth(int bps);
FakeVideoEngine* engine_;
std::vector<VideoCodec> recv_codecs_;
@ -652,78 +452,33 @@ class DataOptions {};
class FakeDataMediaChannel : public RtpHelper<DataMediaChannel> {
public:
explicit FakeDataMediaChannel(void* unused, const DataOptions& options)
: send_blocked_(false), max_bps_(-1) {}
~FakeDataMediaChannel() {}
const std::vector<DataCodec>& recv_codecs() const { return recv_codecs_; }
const std::vector<DataCodec>& send_codecs() const { return send_codecs_; }
const std::vector<DataCodec>& codecs() const { return send_codecs(); }
int max_bps() const { return max_bps_; }
explicit FakeDataMediaChannel(void* unused, const DataOptions& options);
~FakeDataMediaChannel();
const std::vector<DataCodec>& recv_codecs() const;
const std::vector<DataCodec>& send_codecs() const;
const std::vector<DataCodec>& codecs() const;
int max_bps() const;
virtual bool SetSendParameters(const DataSendParameters& params) {
set_send_rtcp_parameters(params.rtcp);
return (SetSendCodecs(params.codecs) &&
SetMaxSendBandwidth(params.max_bandwidth_bps));
}
virtual bool SetRecvParameters(const DataRecvParameters& params) {
set_recv_rtcp_parameters(params.rtcp);
return SetRecvCodecs(params.codecs);
}
virtual bool SetSend(bool send) { return set_sending(send); }
virtual bool SetReceive(bool receive) {
set_playout(receive);
return true;
}
virtual bool AddRecvStream(const StreamParams& sp) {
if (!RtpHelper<DataMediaChannel>::AddRecvStream(sp))
return false;
return true;
}
virtual bool RemoveRecvStream(uint32_t ssrc) {
if (!RtpHelper<DataMediaChannel>::RemoveRecvStream(ssrc))
return false;
return true;
}
bool SetSendParameters(const DataSendParameters& params) override;
bool SetRecvParameters(const DataRecvParameters& params) override;
bool SetSend(bool send) override;
bool SetReceive(bool receive) override;
bool AddRecvStream(const StreamParams& sp) override;
bool RemoveRecvStream(uint32_t ssrc) override;
virtual bool SendData(const SendDataParams& params,
const rtc::CopyOnWriteBuffer& payload,
SendDataResult* result) {
if (send_blocked_) {
*result = SDR_BLOCK;
return false;
} else {
last_sent_data_params_ = params;
last_sent_data_ = std::string(payload.data<char>(), payload.size());
return true;
}
}
bool SendData(const SendDataParams& params,
const rtc::CopyOnWriteBuffer& payload,
SendDataResult* result) override;
SendDataParams last_sent_data_params() { return last_sent_data_params_; }
std::string last_sent_data() { return last_sent_data_; }
bool is_send_blocked() { return send_blocked_; }
void set_send_blocked(bool blocked) { send_blocked_ = blocked; }
SendDataParams last_sent_data_params();
std::string last_sent_data();
bool is_send_blocked();
void set_send_blocked(bool blocked);
private:
bool SetRecvCodecs(const std::vector<DataCodec>& codecs) {
if (fail_set_recv_codecs()) {
// Fake the failure in SetRecvCodecs.
return false;
}
recv_codecs_ = codecs;
return true;
}
bool SetSendCodecs(const std::vector<DataCodec>& codecs) {
if (fail_set_send_codecs()) {
// Fake the failure in SetSendCodecs.
return false;
}
send_codecs_ = codecs;
return true;
}
bool SetMaxSendBandwidth(int bps) {
max_bps_ = bps;
return true;
}
bool SetRecvCodecs(const std::vector<DataCodec>& codecs);
bool SetSendCodecs(const std::vector<DataCodec>& codecs);
bool SetMaxSendBandwidth(int bps);
std::vector<DataCodec> recv_codecs_;
std::vector<DataCodec> send_codecs_;
@ -737,23 +492,11 @@ class FakeDataMediaChannel : public RtpHelper<DataMediaChannel> {
// and FakeVideoEngine.
class FakeBaseEngine {
public:
FakeBaseEngine() : options_changed_(false), fail_create_channel_(false) {}
void set_fail_create_channel(bool fail) { fail_create_channel_ = fail; }
RtpCapabilities GetCapabilities() const { return capabilities_; }
void set_rtp_header_extensions(const std::vector<RtpExtension>& extensions) {
capabilities_.header_extensions = extensions;
}
FakeBaseEngine();
void set_fail_create_channel(bool fail);
void set_rtp_header_extensions(const std::vector<RtpExtension>& extensions);
void set_rtp_header_extensions(
const std::vector<cricket::RtpHeaderExtension>& extensions) {
for (const cricket::RtpHeaderExtension& ext : extensions) {
RtpExtension webrtc_ext;
webrtc_ext.uri = ext.uri;
webrtc_ext.id = ext.id;
capabilities_.header_extensions.push_back(webrtc_ext);
}
}
const std::vector<cricket::RtpHeaderExtension>& extensions);
protected:
// Flag used by optionsmessagehandler_unittest for checking whether any
@ -766,55 +509,28 @@ class FakeBaseEngine {
class FakeVoiceEngine : public FakeBaseEngine {
public:
FakeVoiceEngine() {
// Add a fake audio codec. Note that the name must not be "" as there are
// sanity checks against that.
codecs_.push_back(AudioCodec(101, "fake_audio_codec", 0, 0, 1));
}
void Init() {}
rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const {
return rtc::scoped_refptr<webrtc::AudioState>();
}
FakeVoiceEngine();
RtpCapabilities GetCapabilities() const;
void Init();
rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const;
VoiceMediaChannel* CreateChannel(
webrtc::Call* call,
const MediaConfig& config,
const AudioOptions& options,
const webrtc::CryptoOptions& crypto_options) {
if (fail_create_channel_) {
return nullptr;
}
FakeVoiceMediaChannel* ch = new FakeVoiceMediaChannel(this, options);
channels_.push_back(ch);
return ch;
}
FakeVoiceMediaChannel* GetChannel(size_t index) {
return (channels_.size() > index) ? channels_[index] : NULL;
}
void UnregisterChannel(VoiceMediaChannel* channel) {
channels_.erase(std::find(channels_.begin(), channels_.end(), channel));
}
VoiceMediaChannel* CreateChannel(webrtc::Call* call,
const MediaConfig& config,
const AudioOptions& options,
const webrtc::CryptoOptions& crypto_options);
FakeVoiceMediaChannel* GetChannel(size_t index);
void UnregisterChannel(VoiceMediaChannel* channel);
// TODO(ossu): For proper testing, These should either individually settable
// or the voice engine should reference mockable factories.
const std::vector<AudioCodec>& send_codecs() { return codecs_; }
const std::vector<AudioCodec>& recv_codecs() { return codecs_; }
void SetCodecs(const std::vector<AudioCodec>& codecs) { codecs_ = codecs; }
int GetInputLevel() { return 0; }
bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) {
return false;
}
void StopAecDump() {}
bool StartRtcEventLog(rtc::PlatformFile file, int64_t max_size_bytes) {
return false;
}
void StopRtcEventLog() {}
const std::vector<AudioCodec>& send_codecs() const;
const std::vector<AudioCodec>& recv_codecs() const;
void SetCodecs(const std::vector<AudioCodec>& codecs);
int GetInputLevel();
bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes);
void StopAecDump();
bool StartRtcEventLog(rtc::PlatformFile file, int64_t max_size_bytes);
void StopRtcEventLog();
private:
std::vector<FakeVoiceMediaChannel*> channels_;
@ -825,49 +541,18 @@ class FakeVoiceEngine : public FakeBaseEngine {
class FakeVideoEngine : public FakeBaseEngine {
public:
FakeVideoEngine() : capture_(false) {
// Add a fake video codec. Note that the name must not be "" as there are
// sanity checks against that.
codecs_.push_back(VideoCodec(0, "fake_video_codec"));
}
bool SetOptions(const VideoOptions& options) {
options_ = options;
options_changed_ = true;
return true;
}
FakeVideoEngine();
RtpCapabilities GetCapabilities() const;
bool SetOptions(const VideoOptions& options);
VideoMediaChannel* CreateChannel(webrtc::Call* call,
const MediaConfig& config,
const VideoOptions& options,
const webrtc::CryptoOptions crypto_options) {
if (fail_create_channel_) {
return nullptr;
}
FakeVideoMediaChannel* ch = new FakeVideoMediaChannel(this, options);
channels_.emplace_back(ch);
return ch;
}
FakeVideoMediaChannel* GetChannel(size_t index) {
return (channels_.size() > index) ? channels_[index] : nullptr;
}
void UnregisterChannel(VideoMediaChannel* channel) {
auto it = std::find(channels_.begin(), channels_.end(), channel);
RTC_DCHECK(it != channels_.end());
channels_.erase(it);
}
const std::vector<VideoCodec>& codecs() const { return codecs_; }
void SetCodecs(const std::vector<VideoCodec> codecs) { codecs_ = codecs; }
bool SetCapture(bool capture) {
capture_ = capture;
return true;
}
const webrtc::CryptoOptions& crypto_options);
FakeVideoMediaChannel* GetChannel(size_t index);
void UnregisterChannel(VideoMediaChannel* channel);
std::vector<VideoCodec> codecs() const;
void SetCodecs(const std::vector<VideoCodec> codecs);
bool SetCapture(bool capture);
private:
std::vector<FakeVideoMediaChannel*> channels_;
@ -881,88 +566,47 @@ class FakeVideoEngine : public FakeBaseEngine {
class FakeMediaEngine
: public CompositeMediaEngine<FakeVoiceEngine, FakeVideoEngine> {
public:
FakeMediaEngine()
: CompositeMediaEngine<FakeVoiceEngine, FakeVideoEngine>(std::tuple<>(),
std::tuple<>()) {
}
FakeMediaEngine();
virtual ~FakeMediaEngine() {}
~FakeMediaEngine() override;
void SetAudioCodecs(const std::vector<AudioCodec>& codecs) {
voice().SetCodecs(codecs);
}
void SetVideoCodecs(const std::vector<VideoCodec>& codecs) {
video().SetCodecs(codecs);
}
void SetAudioCodecs(const std::vector<AudioCodec>& codecs);
void SetVideoCodecs(const std::vector<VideoCodec>& codecs);
void SetAudioRtpHeaderExtensions(const std::vector<RtpExtension>& extensions);
void SetVideoRtpHeaderExtensions(const std::vector<RtpExtension>& extensions);
void SetAudioRtpHeaderExtensions(
const std::vector<RtpExtension>& extensions) {
voice().set_rtp_header_extensions(extensions);
}
const std::vector<cricket::RtpHeaderExtension>& extensions);
void SetVideoRtpHeaderExtensions(
const std::vector<RtpExtension>& extensions) {
video().set_rtp_header_extensions(extensions);
}
const std::vector<cricket::RtpHeaderExtension>& extensions);
void SetAudioRtpHeaderExtensions(
const std::vector<cricket::RtpHeaderExtension>& extensions) {
voice().set_rtp_header_extensions(extensions);
}
void SetVideoRtpHeaderExtensions(
const std::vector<cricket::RtpHeaderExtension>& extensions) {
video().set_rtp_header_extensions(extensions);
}
FakeVoiceMediaChannel* GetVoiceChannel(size_t index);
FakeVideoMediaChannel* GetVideoChannel(size_t index);
FakeVoiceMediaChannel* GetVoiceChannel(size_t index) {
return voice().GetChannel(index);
}
FakeVideoMediaChannel* GetVideoChannel(size_t index) {
return video().GetChannel(index);
}
bool capture() const;
bool options_changed() const;
void clear_options_changed();
void set_fail_create_channel(bool fail);
bool capture() const { return video().capture_; }
bool options_changed() const { return video().options_changed_; }
void clear_options_changed() { video().options_changed_ = false; }
void set_fail_create_channel(bool fail) {
voice().set_fail_create_channel(fail);
video().set_fail_create_channel(fail);
}
private:
FakeVoiceEngine* const voice_;
FakeVideoEngine* const video_;
};
// Have to come afterwards due to declaration order
inline FakeVoiceMediaChannel::~FakeVoiceMediaChannel() {
if (engine_) {
engine_->UnregisterChannel(this);
}
}
inline FakeVideoMediaChannel::~FakeVideoMediaChannel() {
if (engine_) {
engine_->UnregisterChannel(this);
}
}
class FakeDataEngine : public DataEngineInterface {
public:
virtual DataMediaChannel* CreateChannel(const MediaConfig& config) {
FakeDataMediaChannel* ch = new FakeDataMediaChannel(this, DataOptions());
channels_.push_back(ch);
return ch;
}
DataMediaChannel* CreateChannel(const MediaConfig& config) override;
FakeDataMediaChannel* GetChannel(size_t index) {
return (channels_.size() > index) ? channels_[index] : NULL;
}
FakeDataMediaChannel* GetChannel(size_t index);
void UnregisterChannel(DataMediaChannel* channel) {
channels_.erase(std::find(channels_.begin(), channels_.end(), channel));
}
void UnregisterChannel(DataMediaChannel* channel);
virtual void SetDataCodecs(const std::vector<DataCodec>& data_codecs) {
data_codecs_ = data_codecs;
}
void SetDataCodecs(const std::vector<DataCodec>& data_codecs);
virtual const std::vector<DataCodec>& data_codecs() { return data_codecs_; }
const std::vector<DataCodec>& data_codecs() override;
private:
std::vector<FakeDataMediaChannel*> channels_;