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webrtc_m130/webrtc/media/base/fakemediaengine.h
deadbeef e814a0dee0 Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc. 
This CL adds the following interfaces:
* RtpTransportController
* RtpTransport
* RtpSender
* RtpReceiver

They're implemented on top of the "BaseChannel" object, which is normally used
in a PeerConnection, and roughly corresponds to an SDP "m=" section. As a result
of this, there are several limitations:

* You can only have one of each type of sender and receiver (audio/video) on top
  of the same transport controller.
* The sender/receiver with the same media type must use the same RTP transport.
* You can't change the transport after creating the sender or receiver.
* Some of the parameters aren't supported.

Later, these "adapter" objects will be gradually replaced by real objects that don't
have these limitations, as "BaseChannel", "MediaChannel" and related code is
restructured. In this CL, we essentially have:

ORTC adapter objects -> BaseChannel -> Media engine
PeerConnection -> BaseChannel -> Media engine

And later we hope to have simply:

PeerConnection -> "Real" ORTC objects -> Media engine

See the linked bug for more context.

BUG=webrtc:7013
TBR=stefan@webrtc.org

Review-Url: https://codereview.webrtc.org/2675173003
Cr-Commit-Position: refs/heads/master@{#16842}
2017-02-26 02:15:09 +00:00

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/*
* Copyright (c) 2004 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.
*/
#ifndef WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_
#define WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_
#include <list>
#include <map>
#include <memory>
#include <set>
#include <string>
#include <vector>
#include "webrtc/api/call/audio_sink.h"
#include "webrtc/base/checks.h"
#include "webrtc/base/copyonwritebuffer.h"
#include "webrtc/base/networkroute.h"
#include "webrtc/base/stringutils.h"
#include "webrtc/media/base/audiosource.h"
#include "webrtc/media/base/mediaengine.h"
#include "webrtc/media/base/rtputils.h"
#include "webrtc/media/base/streamparams.h"
#include "webrtc/p2p/base/sessiondescription.h"
using webrtc::RtpExtension;
namespace cricket {
class FakeMediaEngine;
class FakeVideoEngine;
class FakeVoiceEngine;
// A common helper class that handles sending and receiving RTP/RTCP packets.
template <class Base> class RtpHelper : public Base {
public:
RtpHelper()
: sending_(false),
playout_(false),
fail_set_send_codecs_(false),
fail_set_recv_codecs_(false),
send_ssrc_(0),
ready_to_send_(false) {}
const std::vector<RtpExtension>& recv_extensions() {
return recv_extensions_;
}
const std::vector<RtpExtension>& send_extensions() {
return send_extensions_;
}
bool sending() const { return sending_; }
bool playout() const { return playout_; }
const std::list<std::string>& rtp_packets() const { return rtp_packets_; }
const std::list<std::string>& rtcp_packets() const { return rtcp_packets_; }
bool SendRtp(const void* data,
size_t len,
const rtc::PacketOptions& options) {
if (!sending_) {
return false;
}
rtc::CopyOnWriteBuffer packet(reinterpret_cast<const uint8_t*>(data), len,
kMaxRtpPacketLen);
return Base::SendPacket(&packet, options);
}
bool SendRtcp(const void* data, size_t len) {
rtc::CopyOnWriteBuffer packet(reinterpret_cast<const uint8_t*>(data), len,
kMaxRtpPacketLen);
return Base::SendRtcp(&packet, rtc::PacketOptions());
}
bool CheckRtp(const void* data, size_t len) {
bool success = !rtp_packets_.empty();
if (success) {
std::string packet = rtp_packets_.front();
rtp_packets_.pop_front();
success = (packet == std::string(static_cast<const char*>(data), len));
}
return success;
}
bool CheckRtcp(const void* data, size_t len) {
bool success = !rtcp_packets_.empty();
if (success) {
std::string packet = rtcp_packets_.front();
rtcp_packets_.pop_front();
success = (packet == std::string(static_cast<const char*>(data), len));
}
return success;
}
bool CheckNoRtp() { return rtp_packets_.empty(); }
bool CheckNoRtcp() { return rtcp_packets_.empty(); }
void set_fail_set_send_codecs(bool fail) { fail_set_send_codecs_ = fail; }
void set_fail_set_recv_codecs(bool fail) { fail_set_recv_codecs_ = fail; }
virtual bool AddSendStream(const StreamParams& sp) {
if (std::find(send_streams_.begin(), send_streams_.end(), sp) !=
send_streams_.end()) {
return false;
}
send_streams_.push_back(sp);
rtp_send_parameters_[sp.first_ssrc()] =
CreateRtpParametersWithOneEncoding();
return true;
}
virtual bool RemoveSendStream(uint32_t ssrc) {
auto parameters_iterator = rtp_send_parameters_.find(ssrc);
if (parameters_iterator != rtp_send_parameters_.end()) {
rtp_send_parameters_.erase(parameters_iterator);
}
return RemoveStreamBySsrc(&send_streams_, ssrc);
}
virtual bool AddRecvStream(const StreamParams& sp) {
if (std::find(receive_streams_.begin(), receive_streams_.end(), sp) !=
receive_streams_.end()) {
return false;
}
receive_streams_.push_back(sp);
rtp_receive_parameters_[sp.first_ssrc()] =
CreateRtpParametersWithOneEncoding();
return true;
}
virtual bool RemoveRecvStream(uint32_t ssrc) {
auto parameters_iterator = rtp_receive_parameters_.find(ssrc);
if (parameters_iterator != rtp_receive_parameters_.end()) {
rtp_receive_parameters_.erase(parameters_iterator);
}
return RemoveStreamBySsrc(&receive_streams_, ssrc);
}
virtual webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const {
auto parameters_iterator = rtp_send_parameters_.find(ssrc);
if (parameters_iterator != rtp_send_parameters_.end()) {
return parameters_iterator->second;
}
return webrtc::RtpParameters();
}
virtual bool SetRtpSendParameters(uint32_t ssrc,
const webrtc::RtpParameters& parameters) {
auto parameters_iterator = rtp_send_parameters_.find(ssrc);
if (parameters_iterator != rtp_send_parameters_.end()) {
parameters_iterator->second = parameters;
return true;
}
// Replicate the behavior of the real media channel: return false
// when setting parameters for unknown SSRCs.
return false;
}
virtual webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const {
auto parameters_iterator = rtp_receive_parameters_.find(ssrc);
if (parameters_iterator != rtp_receive_parameters_.end()) {
return parameters_iterator->second;
}
return webrtc::RtpParameters();
}
virtual bool SetRtpReceiveParameters(
uint32_t ssrc,
const webrtc::RtpParameters& parameters) {
auto parameters_iterator = rtp_receive_parameters_.find(ssrc);
if (parameters_iterator != rtp_receive_parameters_.end()) {
parameters_iterator->second = parameters;
return true;
}
// Replicate the behavior of the real media channel: return false
// when setting parameters for unknown SSRCs.
return false;
}
bool IsStreamMuted(uint32_t ssrc) const {
bool ret = muted_streams_.find(ssrc) != muted_streams_.end();
// If |ssrc = 0| check if the first send stream is muted.
if (!ret && ssrc == 0 && !send_streams_.empty()) {
return muted_streams_.find(send_streams_[0].first_ssrc()) !=
muted_streams_.end();
}
return ret;
}
const std::vector<StreamParams>& send_streams() const {
return send_streams_;
}
const std::vector<StreamParams>& recv_streams() const {
return receive_streams_;
}
bool HasRecvStream(uint32_t ssrc) const {
return GetStreamBySsrc(receive_streams_, ssrc) != nullptr;
}
bool HasSendStream(uint32_t ssrc) const {
return GetStreamBySsrc(send_streams_, ssrc) != nullptr;
}
// TODO(perkj): This is to support legacy unit test that only check one
// sending stream.
uint32_t send_ssrc() const {
if (send_streams_.empty())
return 0;
return send_streams_[0].first_ssrc();
}
// TODO(perkj): This is to support legacy unit test that only check one
// sending stream.
const std::string rtcp_cname() {
if (send_streams_.empty())
return "";
return send_streams_[0].cname;
}
const RtcpParameters& send_rtcp_parameters() { return send_rtcp_parameters_; }
const RtcpParameters& recv_rtcp_parameters() { return recv_rtcp_parameters_; }
bool ready_to_send() const {
return ready_to_send_;
}
int transport_overhead_per_packet() const {
return transport_overhead_per_packet_;
}
rtc::NetworkRoute last_network_route() const { return last_network_route_; }
int num_network_route_changes() const { return num_network_route_changes_; }
void set_num_network_route_changes(int changes) {
num_network_route_changes_ = changes;
}
protected:
bool MuteStream(uint32_t ssrc, bool mute) {
if (!HasSendStream(ssrc) && ssrc != 0) {
return false;
}
if (mute) {
muted_streams_.insert(ssrc);
} else {
muted_streams_.erase(ssrc);
}
return true;
}
bool set_sending(bool send) {
sending_ = send;
return true;
}
void set_playout(bool playout) { playout_ = playout; }
bool SetRecvRtpHeaderExtensions(const std::vector<RtpExtension>& extensions) {
recv_extensions_ = extensions;
return true;
}
bool SetSendRtpHeaderExtensions(const std::vector<RtpExtension>& extensions) {
send_extensions_ = extensions;
return true;
}
void set_send_rtcp_parameters(const RtcpParameters& params) {
send_rtcp_parameters_ = params;
}
void set_recv_rtcp_parameters(const RtcpParameters& params) {
recv_rtcp_parameters_ = params;
}
virtual void OnPacketReceived(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) {
rtp_packets_.push_back(std::string(packet->data<char>(), packet->size()));
}
virtual void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
const rtc::PacketTime& packet_time) {
rtcp_packets_.push_back(std::string(packet->data<char>(), packet->size()));
}
virtual void OnReadyToSend(bool ready) {
ready_to_send_ = ready;
}
virtual void OnTransportOverheadChanged(int transport_overhead_per_packet) {
transport_overhead_per_packet_ = transport_overhead_per_packet;
}
virtual void OnNetworkRouteChanged(const std::string& transport_name,
const rtc::NetworkRoute& network_route) {
last_network_route_ = network_route;
++num_network_route_changes_;
}
bool fail_set_send_codecs() const { return fail_set_send_codecs_; }
bool fail_set_recv_codecs() const { return fail_set_recv_codecs_; }
private:
bool sending_;
bool playout_;
std::vector<RtpExtension> recv_extensions_;
std::vector<RtpExtension> send_extensions_;
std::list<std::string> rtp_packets_;
std::list<std::string> rtcp_packets_;
std::vector<StreamParams> send_streams_;
std::vector<StreamParams> receive_streams_;
RtcpParameters send_rtcp_parameters_;
RtcpParameters recv_rtcp_parameters_;
std::set<uint32_t> muted_streams_;
std::map<uint32_t, webrtc::RtpParameters> rtp_send_parameters_;
std::map<uint32_t, webrtc::RtpParameters> rtp_receive_parameters_;
bool fail_set_send_codecs_;
bool fail_set_recv_codecs_;
uint32_t send_ssrc_;
std::string rtcp_cname_;
bool ready_to_send_;
int transport_overhead_per_packet_;
rtc::NetworkRoute last_network_route_;
int num_network_route_changes_ = 0;
};
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) {}
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);
}
~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));
}
virtual bool SetRecvParameters(const AudioRecvParameters& params) {
set_recv_rtcp_parameters(params.rtcp);
return (SetRecvCodecs(params.codecs) &&
SetRecvRtpHeaderExtensions(params.extensions));
}
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;
}
bool HasSource(uint32_t ssrc) const {
return local_sinks_.find(ssrc) != local_sinks_.end();
}
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;
}
virtual bool GetActiveStreams(AudioInfo::StreamList* streams) { return true; }
virtual int GetOutputLevel() { return 0; }
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;
}
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;
}
virtual bool GetStats(VoiceMediaInfo* info) { return false; }
virtual void SetRawAudioSink(
uint32_t ssrc,
std::unique_ptr<webrtc::AudioSinkInterface> sink) {
sink_ = std::move(sink);
}
private:
class VoiceChannelAudioSink : public AudioSource::Sink {
public:
explicit VoiceChannelAudioSink(AudioSource* source) : source_(source) {
source_->SetSink(this);
}
virtual ~VoiceChannelAudioSink() {
if (source_) {
source_->SetSink(nullptr);
}
}
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_; }
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, new VoiceChannelAudioSink(source)));
}
} else {
if (it != local_sinks_.end()) {
delete it->second;
local_sinks_.erase(it);
}
}
return true;
}
FakeVoiceEngine* engine_;
std::vector<AudioCodec> recv_codecs_;
std::vector<AudioCodec> send_codecs_;
std::map<uint32_t, double> output_scalings_;
std::vector<DtmfInfo> dtmf_info_queue_;
AudioOptions options_;
std::map<uint32_t, VoiceChannelAudioSink*> local_sinks_;
std::unique_ptr<webrtc::AudioSinkInterface> sink_;
int max_bps_;
};
// 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);
}
class FakeVideoMediaChannel : public RtpHelper<VideoMediaChannel> {
public:
explicit FakeVideoMediaChannel(FakeVideoEngine* engine,
const VideoOptions& options)
: engine_(engine), max_bps_(-1) {
SetOptions(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::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);
}
bool GetSendCodec(VideoCodec* send_codec) override {
if (send_codecs_.empty()) {
return false;
}
*send_codec = send_codecs_[0];
return true;
}
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;
}
bool SetSend(bool send) override { return set_sending(send); }
bool SetVideoSend(
uint32_t ssrc,
bool enable,
const VideoOptions* options,
rtc::VideoSourceInterface<webrtc::VideoFrame>* source) override {
if (!RtpHelper<VideoMediaChannel>::MuteStream(ssrc, !enable)) {
return false;
}
if (enable && options) {
if (!SetOptions(*options)) {
return false;
}
}
sources_[ssrc] = source;
return true;
}
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 GetStats(VideoMediaInfo* info) override { return false; }
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;
}
FakeVideoEngine* engine_;
std::vector<VideoCodec> recv_codecs_;
std::vector<VideoCodec> send_codecs_;
std::map<uint32_t, rtc::VideoSinkInterface<webrtc::VideoFrame>*> sinks_;
std::map<uint32_t, rtc::VideoSourceInterface<webrtc::VideoFrame>*> sources_;
VideoOptions options_;
int max_bps_;
};
// Dummy option class, needed for the DataTraits abstraction in
// channel_unittest.c.
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_; }
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;
}
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;
}
}
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; }
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;
}
std::vector<DataCodec> recv_codecs_;
std::vector<DataCodec> send_codecs_;
SendDataParams last_sent_data_params_;
std::string last_sent_data_;
bool send_blocked_;
int max_bps_;
};
// A base class for all of the shared parts between FakeVoiceEngine
// 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;
}
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);
}
}
protected:
// Flag used by optionsmessagehandler_unittest for checking whether any
// relevant setting has been updated.
// TODO(thaloun): Replace with explicit checks of before & after values.
bool options_changed_;
bool fail_create_channel_;
RtpCapabilities capabilities_;
};
class FakeVoiceEngine : public FakeBaseEngine {
public:
FakeVoiceEngine(webrtc::AudioDeviceModule* adm,
const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
audio_decoder_factory,
rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer) {
// 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));
}
rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const {
return rtc::scoped_refptr<webrtc::AudioState>();
}
VoiceMediaChannel* CreateChannel(webrtc::Call* call,
const MediaConfig& config,
const AudioOptions& 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));
}
// 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() {}
private:
std::vector<FakeVoiceMediaChannel*> channels_;
std::vector<AudioCodec> codecs_;
friend class FakeMediaEngine;
};
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"));
}
void Init() {}
bool SetOptions(const VideoOptions& options) {
options_ = options;
options_changed_ = true;
return true;
}
VideoMediaChannel* CreateChannel(webrtc::Call* call,
const MediaConfig& config,
const VideoOptions& options) {
if (fail_create_channel_) {
return NULL;
}
FakeVideoMediaChannel* ch = new FakeVideoMediaChannel(this, options);
channels_.push_back(ch);
return ch;
}
FakeVideoMediaChannel* GetChannel(size_t index) {
return (channels_.size() > index) ? channels_[index] : NULL;
}
void UnregisterChannel(VideoMediaChannel* channel) {
channels_.erase(std::find(channels_.begin(), channels_.end(), channel));
}
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;
}
private:
std::vector<FakeVideoMediaChannel*> channels_;
std::vector<VideoCodec> codecs_;
bool capture_;
VideoOptions options_;
friend class FakeMediaEngine;
};
class FakeMediaEngine :
public CompositeMediaEngine<FakeVoiceEngine, FakeVideoEngine> {
public:
FakeMediaEngine()
: CompositeMediaEngine<FakeVoiceEngine, FakeVideoEngine>(nullptr,
nullptr,
nullptr) {}
virtual ~FakeMediaEngine() {}
void SetAudioCodecs(const std::vector<AudioCodec>& codecs) {
voice_.SetCodecs(codecs);
}
void SetVideoCodecs(const std::vector<VideoCodec>& codecs) {
video_.SetCodecs(codecs);
}
void SetAudioRtpHeaderExtensions(
const std::vector<RtpExtension>& extensions) {
voice_.set_rtp_header_extensions(extensions);
}
void SetVideoRtpHeaderExtensions(
const std::vector<RtpExtension>& extensions) {
video_.set_rtp_header_extensions(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) {
return voice_.GetChannel(index);
}
FakeVideoMediaChannel* GetVideoChannel(size_t index) {
return video_.GetChannel(index);
}
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);
}
};
// 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:
FakeDataEngine(){};
virtual DataMediaChannel* CreateChannel(const MediaConfig& config) {
FakeDataMediaChannel* ch = new FakeDataMediaChannel(this, DataOptions());
channels_.push_back(ch);
return ch;
}
FakeDataMediaChannel* GetChannel(size_t index) {
return (channels_.size() > index) ? channels_[index] : NULL;
}
void UnregisterChannel(DataMediaChannel* channel) {
channels_.erase(std::find(channels_.begin(), channels_.end(), channel));
}
virtual void SetDataCodecs(const std::vector<DataCodec>& data_codecs) {
data_codecs_ = data_codecs;
}
virtual const std::vector<DataCodec>& data_codecs() { return data_codecs_; }
private:
std::vector<FakeDataMediaChannel*> channels_;
std::vector<DataCodec> data_codecs_;
};
} // namespace cricket
#endif // WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_
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