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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 MEDIA_BASE_FAKEMEDIAENGINE_H_
#define MEDIA_BASE_FAKEMEDIAENGINE_H_
#include <list>
#include <map>
#include <memory>
#include <set>
#include <string>
#include <tuple>
#include <vector>
#include "api/call/audio_sink.h"
#include "media/base/audiosource.h"
#include "media/base/mediaengine.h"
#include "media/base/rtputils.h"
#include "media/base/streamparams.h"
#include "media/engine/webrtcvideoengine.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "rtc_base/copyonwritebuffer.h"
#include "rtc_base/networkroute.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),
transport_overhead_per_packet_(0),
num_network_route_changes_(0) {}
virtual ~RtpHelper() = default;
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()] =
CreateRtpParametersWithEncodings(sp);
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 webrtc::RTCError SetRtpSendParameters(
uint32_t ssrc,
const webrtc::RtpParameters& parameters) {
auto parameters_iterator = rtp_send_parameters_.find(ssrc);
if (parameters_iterator != rtp_send_parameters_.end()) {
auto result =
ValidateRtpParameters(parameters_iterator->second, parameters);
if (!result.ok())
return result;
parameters_iterator->second = parameters;
return webrtc::RTCError::OK();
}
// Replicate the behavior of the real media channel: return false
// when setting parameters for unknown SSRCs.
return webrtc::RTCError(webrtc::RTCErrorType::INTERNAL_ERROR);
}
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 SetSendExtmapAllowMixed(bool extmap_allow_mixed) {
if (Base::ExtmapAllowMixed() != extmap_allow_mixed) {
Base::SetExtmapAllowMixed(extmap_allow_mixed);
}
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,
int64_t packet_time_us) {
rtp_packets_.push_back(std::string(packet->data<char>(), packet->size()));
}
virtual void OnRtcpReceived(rtc::CopyOnWriteBuffer* packet,
int64_t packet_time_us) {
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_;
transport_overhead_per_packet_ = network_route.packet_overhead;
}
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_;
};
class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
public:
struct DtmfInfo {
DtmfInfo(uint32_t ssrc, int event_code, int duration);
uint32_t ssrc;
int event_code;
int duration;
};
explicit FakeVoiceMediaChannel(FakeVoiceEngine* engine,
const AudioOptions& options);
~FakeVoiceMediaChannel();
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;
bool SetRecvParameters(const AudioRecvParameters& params) override;
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;
bool AddRecvStream(const StreamParams& sp) override;
bool RemoveRecvStream(uint32_t ssrc) override;
bool CanInsertDtmf() override;
bool InsertDtmf(uint32_t ssrc, int event_code, int duration) override;
bool SetOutputVolume(uint32_t ssrc, double volume) override;
bool GetOutputVolume(uint32_t ssrc, double* volume);
bool GetStats(VoiceMediaInfo* info) override;
void SetRawAudioSink(
uint32_t ssrc,
std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override;
private:
class VoiceChannelAudioSink : public AudioSource::Sink {
public:
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;
AudioSource* source() const;
private:
AudioSource* source_;
};
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_;
std::vector<AudioCodec> send_codecs_;
std::map<uint32_t, double> output_scalings_;
std::vector<DtmfInfo> dtmf_info_queue_;
AudioOptions options_;
std::map<uint32_t, std::unique_ptr<VoiceChannelAudioSink>> local_sinks_;
std::unique_ptr<webrtc::AudioSinkInterface> sink_;
int max_bps_;
};
// A helper function to compare the FakeVoiceMediaChannel::DtmfInfo.
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);
~FakeVideoMediaChannel();
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;
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;
bool SetSink(uint32_t ssrc,
rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override;
bool HasSink(uint32_t ssrc) const;
bool SetSend(bool send) override;
bool SetVideoSend(
uint32_t ssrc,
const VideoOptions* options,
rtc::VideoSourceInterface<webrtc::VideoFrame>* source) override;
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;
std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override;
private:
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_;
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);
~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;
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;
bool SendData(const SendDataParams& params,
const rtc::CopyOnWriteBuffer& payload,
SendDataResult* result) override;
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);
bool SetSendCodecs(const std::vector<DataCodec>& codecs);
bool SetMaxSendBandwidth(int bps);
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_;
};
class FakeVoiceEngine {
public:
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);
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() 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_;
std::vector<AudioCodec> codecs_;
bool fail_create_channel_;
friend class FakeMediaEngine;
};
class FakeVideoEngine {
public:
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);
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_;
std::vector<VideoCodec> codecs_;
bool capture_;
VideoOptions options_;
bool fail_create_channel_;
friend class FakeMediaEngine;
};
class FakeMediaEngine
: public CompositeMediaEngine<FakeVoiceEngine, FakeVideoEngine> {
public:
FakeMediaEngine();
~FakeMediaEngine() override;
void SetAudioCodecs(const std::vector<AudioCodec>& codecs);
void SetVideoCodecs(const std::vector<VideoCodec>& codecs);
FakeVoiceMediaChannel* GetVoiceChannel(size_t index);
FakeVideoMediaChannel* GetVideoChannel(size_t index);
void set_fail_create_channel(bool fail);
private:
FakeVoiceEngine* const voice_;
FakeVideoEngine* const video_;
};
// Have to come afterwards due to declaration order
class FakeDataEngine : public DataEngineInterface {
public:
DataMediaChannel* CreateChannel(const MediaConfig& config) override;
FakeDataMediaChannel* GetChannel(size_t index);
void UnregisterChannel(DataMediaChannel* channel);
void SetDataCodecs(const std::vector<DataCodec>& data_codecs);
const std::vector<DataCodec>& data_codecs() override;
private:
std::vector<FakeDataMediaChannel*> channels_;
std::vector<DataCodec> data_codecs_;
};
} // namespace cricket
#endif // MEDIA_BASE_FAKEMEDIAENGINE_H_