961 lines
30 KiB
C++
961 lines
30 KiB
C++
/*
|
|
* 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/audio_sink.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;
|
|
}
|
|
|
|
bool ready_to_send() const {
|
|
return ready_to_send_;
|
|
}
|
|
|
|
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;
|
|
}
|
|
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 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_;
|
|
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_;
|
|
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), time_since_last_typing_(-1), 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) {
|
|
return (SetSendCodecs(params.codecs) &&
|
|
SetSendRtpHeaderExtensions(params.extensions) &&
|
|
SetMaxSendBandwidth(params.max_bandwidth_bps) &&
|
|
SetOptions(params.options));
|
|
}
|
|
|
|
virtual bool SetRecvParameters(const AudioRecvParameters& params) {
|
|
return (SetRecvCodecs(params.codecs) &&
|
|
SetRecvRtpHeaderExtensions(params.extensions));
|
|
}
|
|
|
|
virtual bool SetPlayout(bool playout) {
|
|
set_playout(playout);
|
|
return true;
|
|
}
|
|
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;
|
|
}
|
|
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; }
|
|
void set_time_since_last_typing(int ms) { time_since_last_typing_ = ms; }
|
|
virtual int GetTimeSinceLastTyping() { return time_since_last_typing_; }
|
|
virtual void SetTypingDetectionParameters(
|
|
int time_window, int cost_per_typing, int reporting_threshold,
|
|
int penalty_decay, int type_event_delay) {}
|
|
|
|
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()) {
|
|
ASSERT(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_;
|
|
int time_since_last_typing_;
|
|
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<VideoFrame>*>& sinks()
|
|
const {
|
|
return sinks_;
|
|
}
|
|
int max_bps() const { return max_bps_; }
|
|
bool SetSendParameters(const VideoSendParameters& params) override {
|
|
return (SetSendCodecs(params.codecs) &&
|
|
SetSendRtpHeaderExtensions(params.extensions) &&
|
|
SetMaxSendBandwidth(params.max_bandwidth_bps));
|
|
}
|
|
bool SetRecvParameters(const VideoRecvParameters& params) override {
|
|
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<cricket::VideoFrame>* sink) override {
|
|
if (ssrc != 0 && sinks_.find(ssrc) == sinks_.end()) {
|
|
return false;
|
|
}
|
|
if (ssrc != 0) {
|
|
sinks_[ssrc] = sink;
|
|
}
|
|
return true;
|
|
}
|
|
|
|
bool SetSend(bool send) override { return set_sending(send); }
|
|
bool SetVideoSend(
|
|
uint32_t ssrc,
|
|
bool enable,
|
|
const VideoOptions* options,
|
|
rtc::VideoSourceInterface<cricket::VideoFrame>* source) override {
|
|
if (!RtpHelper<VideoMediaChannel>::MuteStream(ssrc, !enable)) {
|
|
return false;
|
|
}
|
|
if (enable && options) {
|
|
return SetOptions(*options);
|
|
}
|
|
sources_[ssrc] = source;
|
|
return true;
|
|
}
|
|
|
|
bool HasSource(uint32_t ssrc) const {
|
|
return sources_.find(ssrc) != sources_.end();
|
|
}
|
|
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<VideoFrame>*> sinks_;
|
|
std::map<uint32_t, rtc::VideoSourceInterface<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) {
|
|
return (SetSendCodecs(params.codecs) &&
|
|
SetMaxSendBandwidth(params.max_bandwidth_bps));
|
|
}
|
|
virtual bool SetRecvParameters(const DataRecvParameters& params) {
|
|
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) {
|
|
// 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", 0, 0, 0));
|
|
}
|
|
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) {}
|
|
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);
|
|
}
|
|
};
|
|
|
|
// CompositeMediaEngine with FakeVoiceEngine to expose SetAudioCodecs to
|
|
// establish a media connectionwith minimum set of audio codes required
|
|
template <class VIDEO>
|
|
class CompositeMediaEngineWithFakeVoiceEngine :
|
|
public CompositeMediaEngine<FakeVoiceEngine, VIDEO> {
|
|
public:
|
|
CompositeMediaEngineWithFakeVoiceEngine() {}
|
|
virtual ~CompositeMediaEngineWithFakeVoiceEngine() {}
|
|
|
|
virtual void SetAudioCodecs(const std::vector<AudioCodec>& codecs) {
|
|
CompositeMediaEngine<FakeVoiceEngine, VIDEO>::voice_.SetCodecs(codecs);
|
|
}
|
|
};
|
|
|
|
// 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() : last_channel_type_(DCT_NONE) {}
|
|
|
|
virtual DataMediaChannel* CreateChannel(DataChannelType data_channel_type) {
|
|
last_channel_type_ = data_channel_type;
|
|
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_; }
|
|
|
|
DataChannelType last_channel_type() const { return last_channel_type_; }
|
|
|
|
private:
|
|
std::vector<FakeDataMediaChannel*> channels_;
|
|
std::vector<DataCodec> data_codecs_;
|
|
DataChannelType last_channel_type_;
|
|
};
|
|
|
|
} // namespace cricket
|
|
|
|
#endif // WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_
|