1 /*
2  *  Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
3  *
4  *  Use of this source code is governed by a BSD-style license
5  *  that can be found in the LICENSE file in the root of the source
6  *  tree. An additional intellectual property rights grant can be found
7  *  in the file PATENTS.  All contributing project authors may
8  *  be found in the AUTHORS file in the root of the source tree.
9  */
10 
11 #ifndef MEDIA_BASE_FAKE_MEDIA_ENGINE_H_
12 #define MEDIA_BASE_FAKE_MEDIA_ENGINE_H_
13 
14 #include <list>
15 #include <map>
16 #include <memory>
17 #include <set>
18 #include <string>
19 #include <tuple>
20 #include <vector>
21 
22 #include "absl/algorithm/container.h"
23 #include "api/call/audio_sink.h"
24 #include "media/base/audio_source.h"
25 #include "media/base/media_engine.h"
26 #include "media/base/rtp_utils.h"
27 #include "media/base/stream_params.h"
28 #include "media/engine/webrtc_video_engine.h"
29 #include "modules/audio_processing/include/audio_processing.h"
30 #include "rtc_base/copy_on_write_buffer.h"
31 #include "rtc_base/network_route.h"
32 
33 using webrtc::RtpExtension;
34 
35 namespace cricket {
36 
37 class FakeMediaEngine;
38 class FakeVideoEngine;
39 class FakeVoiceEngine;
40 
41 // A common helper class that handles sending and receiving RTP/RTCP packets.
42 template <class Base>
43 class RtpHelper : public Base {
44  public:
RtpHelper()45   RtpHelper()
46       : sending_(false),
47         playout_(false),
48         fail_set_send_codecs_(false),
49         fail_set_recv_codecs_(false),
50         send_ssrc_(0),
51         ready_to_send_(false),
52         transport_overhead_per_packet_(0),
53         num_network_route_changes_(0) {}
54   virtual ~RtpHelper() = default;
recv_extensions()55   const std::vector<RtpExtension>& recv_extensions() {
56     return recv_extensions_;
57   }
send_extensions()58   const std::vector<RtpExtension>& send_extensions() {
59     return send_extensions_;
60   }
sending()61   bool sending() const { return sending_; }
playout()62   bool playout() const { return playout_; }
rtp_packets()63   const std::list<std::string>& rtp_packets() const { return rtp_packets_; }
rtcp_packets()64   const std::list<std::string>& rtcp_packets() const { return rtcp_packets_; }
65 
SendRtp(const void * data,size_t len,const rtc::PacketOptions & options)66   bool SendRtp(const void* data,
67                size_t len,
68                const rtc::PacketOptions& options) {
69     if (!sending_) {
70       return false;
71     }
72     rtc::CopyOnWriteBuffer packet(reinterpret_cast<const uint8_t*>(data), len,
73                                   kMaxRtpPacketLen);
74     return Base::SendPacket(&packet, options);
75   }
SendRtcp(const void * data,size_t len)76   bool SendRtcp(const void* data, size_t len) {
77     rtc::CopyOnWriteBuffer packet(reinterpret_cast<const uint8_t*>(data), len,
78                                   kMaxRtpPacketLen);
79     return Base::SendRtcp(&packet, rtc::PacketOptions());
80   }
81 
CheckRtp(const void * data,size_t len)82   bool CheckRtp(const void* data, size_t len) {
83     bool success = !rtp_packets_.empty();
84     if (success) {
85       std::string packet = rtp_packets_.front();
86       rtp_packets_.pop_front();
87       success = (packet == std::string(static_cast<const char*>(data), len));
88     }
89     return success;
90   }
CheckRtcp(const void * data,size_t len)91   bool CheckRtcp(const void* data, size_t len) {
92     bool success = !rtcp_packets_.empty();
93     if (success) {
94       std::string packet = rtcp_packets_.front();
95       rtcp_packets_.pop_front();
96       success = (packet == std::string(static_cast<const char*>(data), len));
97     }
98     return success;
99   }
CheckNoRtp()100   bool CheckNoRtp() { return rtp_packets_.empty(); }
CheckNoRtcp()101   bool CheckNoRtcp() { return rtcp_packets_.empty(); }
set_fail_set_send_codecs(bool fail)102   void set_fail_set_send_codecs(bool fail) { fail_set_send_codecs_ = fail; }
set_fail_set_recv_codecs(bool fail)103   void set_fail_set_recv_codecs(bool fail) { fail_set_recv_codecs_ = fail; }
AddSendStream(const StreamParams & sp)104   virtual bool AddSendStream(const StreamParams& sp) {
105     if (absl::c_linear_search(send_streams_, sp)) {
106       return false;
107     }
108     send_streams_.push_back(sp);
109     rtp_send_parameters_[sp.first_ssrc()] =
110         CreateRtpParametersWithEncodings(sp);
111     return true;
112   }
RemoveSendStream(uint32_t ssrc)113   virtual bool RemoveSendStream(uint32_t ssrc) {
114     auto parameters_iterator = rtp_send_parameters_.find(ssrc);
115     if (parameters_iterator != rtp_send_parameters_.end()) {
116       rtp_send_parameters_.erase(parameters_iterator);
117     }
118     return RemoveStreamBySsrc(&send_streams_, ssrc);
119   }
ResetUnsignaledRecvStream()120   virtual void ResetUnsignaledRecvStream() {}
OnDemuxerCriteriaUpdatePending()121   virtual void OnDemuxerCriteriaUpdatePending() {}
OnDemuxerCriteriaUpdateComplete()122   virtual void OnDemuxerCriteriaUpdateComplete() {}
123 
AddRecvStream(const StreamParams & sp)124   virtual bool AddRecvStream(const StreamParams& sp) {
125     if (absl::c_linear_search(receive_streams_, sp)) {
126       return false;
127     }
128     receive_streams_.push_back(sp);
129     rtp_receive_parameters_[sp.first_ssrc()] =
130         CreateRtpParametersWithEncodings(sp);
131     return true;
132   }
RemoveRecvStream(uint32_t ssrc)133   virtual bool RemoveRecvStream(uint32_t ssrc) {
134     auto parameters_iterator = rtp_receive_parameters_.find(ssrc);
135     if (parameters_iterator != rtp_receive_parameters_.end()) {
136       rtp_receive_parameters_.erase(parameters_iterator);
137     }
138     return RemoveStreamBySsrc(&receive_streams_, ssrc);
139   }
140 
GetRtpSendParameters(uint32_t ssrc)141   virtual webrtc::RtpParameters GetRtpSendParameters(uint32_t ssrc) const {
142     auto parameters_iterator = rtp_send_parameters_.find(ssrc);
143     if (parameters_iterator != rtp_send_parameters_.end()) {
144       return parameters_iterator->second;
145     }
146     return webrtc::RtpParameters();
147   }
SetRtpSendParameters(uint32_t ssrc,const webrtc::RtpParameters & parameters)148   virtual webrtc::RTCError SetRtpSendParameters(
149       uint32_t ssrc,
150       const webrtc::RtpParameters& parameters) {
151     auto parameters_iterator = rtp_send_parameters_.find(ssrc);
152     if (parameters_iterator != rtp_send_parameters_.end()) {
153       auto result = CheckRtpParametersInvalidModificationAndValues(
154           parameters_iterator->second, parameters);
155       if (!result.ok())
156         return result;
157 
158       parameters_iterator->second = parameters;
159       return webrtc::RTCError::OK();
160     }
161     // Replicate the behavior of the real media channel: return false
162     // when setting parameters for unknown SSRCs.
163     return webrtc::RTCError(webrtc::RTCErrorType::INTERNAL_ERROR);
164   }
165 
GetRtpReceiveParameters(uint32_t ssrc)166   virtual webrtc::RtpParameters GetRtpReceiveParameters(uint32_t ssrc) const {
167     auto parameters_iterator = rtp_receive_parameters_.find(ssrc);
168     if (parameters_iterator != rtp_receive_parameters_.end()) {
169       return parameters_iterator->second;
170     }
171     return webrtc::RtpParameters();
172   }
GetDefaultRtpReceiveParameters()173   virtual webrtc::RtpParameters GetDefaultRtpReceiveParameters() const {
174     return webrtc::RtpParameters();
175   }
176 
IsStreamMuted(uint32_t ssrc)177   bool IsStreamMuted(uint32_t ssrc) const {
178     bool ret = muted_streams_.find(ssrc) != muted_streams_.end();
179     // If |ssrc = 0| check if the first send stream is muted.
180     if (!ret && ssrc == 0 && !send_streams_.empty()) {
181       return muted_streams_.find(send_streams_[0].first_ssrc()) !=
182              muted_streams_.end();
183     }
184     return ret;
185   }
send_streams()186   const std::vector<StreamParams>& send_streams() const {
187     return send_streams_;
188   }
recv_streams()189   const std::vector<StreamParams>& recv_streams() const {
190     return receive_streams_;
191   }
HasRecvStream(uint32_t ssrc)192   bool HasRecvStream(uint32_t ssrc) const {
193     return GetStreamBySsrc(receive_streams_, ssrc) != nullptr;
194   }
HasSendStream(uint32_t ssrc)195   bool HasSendStream(uint32_t ssrc) const {
196     return GetStreamBySsrc(send_streams_, ssrc) != nullptr;
197   }
198   // TODO(perkj): This is to support legacy unit test that only check one
199   // sending stream.
send_ssrc()200   uint32_t send_ssrc() const {
201     if (send_streams_.empty())
202       return 0;
203     return send_streams_[0].first_ssrc();
204   }
205 
206   // TODO(perkj): This is to support legacy unit test that only check one
207   // sending stream.
rtcp_cname()208   const std::string rtcp_cname() {
209     if (send_streams_.empty())
210       return "";
211     return send_streams_[0].cname;
212   }
send_rtcp_parameters()213   const RtcpParameters& send_rtcp_parameters() { return send_rtcp_parameters_; }
recv_rtcp_parameters()214   const RtcpParameters& recv_rtcp_parameters() { return recv_rtcp_parameters_; }
215 
ready_to_send()216   bool ready_to_send() const { return ready_to_send_; }
217 
transport_overhead_per_packet()218   int transport_overhead_per_packet() const {
219     return transport_overhead_per_packet_;
220   }
221 
last_network_route()222   rtc::NetworkRoute last_network_route() const { return last_network_route_; }
num_network_route_changes()223   int num_network_route_changes() const { return num_network_route_changes_; }
set_num_network_route_changes(int changes)224   void set_num_network_route_changes(int changes) {
225     num_network_route_changes_ = changes;
226   }
227 
OnRtcpPacketReceived(rtc::CopyOnWriteBuffer * packet,int64_t packet_time_us)228   void OnRtcpPacketReceived(rtc::CopyOnWriteBuffer* packet,
229                             int64_t packet_time_us) {
230     rtcp_packets_.push_back(std::string(packet->cdata<char>(), packet->size()));
231   }
232 
233  protected:
MuteStream(uint32_t ssrc,bool mute)234   bool MuteStream(uint32_t ssrc, bool mute) {
235     if (!HasSendStream(ssrc) && ssrc != 0) {
236       return false;
237     }
238     if (mute) {
239       muted_streams_.insert(ssrc);
240     } else {
241       muted_streams_.erase(ssrc);
242     }
243     return true;
244   }
set_sending(bool send)245   bool set_sending(bool send) {
246     sending_ = send;
247     return true;
248   }
set_playout(bool playout)249   void set_playout(bool playout) { playout_ = playout; }
SetRecvRtpHeaderExtensions(const std::vector<RtpExtension> & extensions)250   bool SetRecvRtpHeaderExtensions(const std::vector<RtpExtension>& extensions) {
251     recv_extensions_ = extensions;
252     return true;
253   }
SetSendExtmapAllowMixed(bool extmap_allow_mixed)254   bool SetSendExtmapAllowMixed(bool extmap_allow_mixed) {
255     if (Base::ExtmapAllowMixed() != extmap_allow_mixed) {
256       Base::SetExtmapAllowMixed(extmap_allow_mixed);
257     }
258     return true;
259   }
SetSendRtpHeaderExtensions(const std::vector<RtpExtension> & extensions)260   bool SetSendRtpHeaderExtensions(const std::vector<RtpExtension>& extensions) {
261     send_extensions_ = extensions;
262     return true;
263   }
set_send_rtcp_parameters(const RtcpParameters & params)264   void set_send_rtcp_parameters(const RtcpParameters& params) {
265     send_rtcp_parameters_ = params;
266   }
set_recv_rtcp_parameters(const RtcpParameters & params)267   void set_recv_rtcp_parameters(const RtcpParameters& params) {
268     recv_rtcp_parameters_ = params;
269   }
OnPacketReceived(rtc::CopyOnWriteBuffer packet,int64_t packet_time_us)270   virtual void OnPacketReceived(rtc::CopyOnWriteBuffer packet,
271                                 int64_t packet_time_us) {
272     rtp_packets_.push_back(std::string(packet.cdata<char>(), packet.size()));
273   }
OnReadyToSend(bool ready)274   virtual void OnReadyToSend(bool ready) { ready_to_send_ = ready; }
275 
OnNetworkRouteChanged(const std::string & transport_name,const rtc::NetworkRoute & network_route)276   virtual void OnNetworkRouteChanged(const std::string& transport_name,
277                                      const rtc::NetworkRoute& network_route) {
278     last_network_route_ = network_route;
279     ++num_network_route_changes_;
280     transport_overhead_per_packet_ = network_route.packet_overhead;
281   }
fail_set_send_codecs()282   bool fail_set_send_codecs() const { return fail_set_send_codecs_; }
fail_set_recv_codecs()283   bool fail_set_recv_codecs() const { return fail_set_recv_codecs_; }
284 
285  private:
286   bool sending_;
287   bool playout_;
288   std::vector<RtpExtension> recv_extensions_;
289   std::vector<RtpExtension> send_extensions_;
290   std::list<std::string> rtp_packets_;
291   std::list<std::string> rtcp_packets_;
292   std::vector<StreamParams> send_streams_;
293   std::vector<StreamParams> receive_streams_;
294   RtcpParameters send_rtcp_parameters_;
295   RtcpParameters recv_rtcp_parameters_;
296   std::set<uint32_t> muted_streams_;
297   std::map<uint32_t, webrtc::RtpParameters> rtp_send_parameters_;
298   std::map<uint32_t, webrtc::RtpParameters> rtp_receive_parameters_;
299   bool fail_set_send_codecs_;
300   bool fail_set_recv_codecs_;
301   uint32_t send_ssrc_;
302   std::string rtcp_cname_;
303   bool ready_to_send_;
304   int transport_overhead_per_packet_;
305   rtc::NetworkRoute last_network_route_;
306   int num_network_route_changes_;
307 };
308 
309 class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
310  public:
311   struct DtmfInfo {
312     DtmfInfo(uint32_t ssrc, int event_code, int duration);
313     uint32_t ssrc;
314     int event_code;
315     int duration;
316   };
317   explicit FakeVoiceMediaChannel(FakeVoiceEngine* engine,
318                                  const AudioOptions& options);
319   ~FakeVoiceMediaChannel();
320   const std::vector<AudioCodec>& recv_codecs() const;
321   const std::vector<AudioCodec>& send_codecs() const;
322   const std::vector<AudioCodec>& codecs() const;
323   const std::vector<DtmfInfo>& dtmf_info_queue() const;
324   const AudioOptions& options() const;
325   int max_bps() const;
326   bool SetSendParameters(const AudioSendParameters& params) override;
327 
328   bool SetRecvParameters(const AudioRecvParameters& params) override;
329 
330   void SetPlayout(bool playout) override;
331   void SetSend(bool send) override;
332   bool SetAudioSend(uint32_t ssrc,
333                     bool enable,
334                     const AudioOptions* options,
335                     AudioSource* source) override;
336 
337   bool HasSource(uint32_t ssrc) const;
338 
339   bool AddRecvStream(const StreamParams& sp) override;
340   bool RemoveRecvStream(uint32_t ssrc) override;
341 
342   bool CanInsertDtmf() override;
343   bool InsertDtmf(uint32_t ssrc, int event_code, int duration) override;
344 
345   bool SetOutputVolume(uint32_t ssrc, double volume) override;
346   bool SetDefaultOutputVolume(double volume) override;
347 
348   bool GetOutputVolume(uint32_t ssrc, double* volume);
349 
350   bool SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, int delay_ms) override;
351   absl::optional<int> GetBaseMinimumPlayoutDelayMs(
352       uint32_t ssrc) const override;
353 
354   bool GetStats(VoiceMediaInfo* info, bool get_and_clear_legacy_stats) override;
355 
356   void SetRawAudioSink(
357       uint32_t ssrc,
358       std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
359   void SetDefaultRawAudioSink(
360       std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
361 
362   std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override;
363 
364  private:
365   class VoiceChannelAudioSink : public AudioSource::Sink {
366    public:
367     explicit VoiceChannelAudioSink(AudioSource* source);
368     ~VoiceChannelAudioSink() override;
369     void OnData(const void* audio_data,
370                 int bits_per_sample,
371                 int sample_rate,
372                 size_t number_of_channels,
373                 size_t number_of_frames,
374                 absl::optional<int64_t> absolute_capture_timestamp_ms) override;
375     void OnClose() override;
NumPreferredChannels()376     int NumPreferredChannels() const override { return -1; }
377     AudioSource* source() const;
378 
379    private:
380     AudioSource* source_;
381   };
382 
383   bool SetRecvCodecs(const std::vector<AudioCodec>& codecs);
384   bool SetSendCodecs(const std::vector<AudioCodec>& codecs);
385   bool SetMaxSendBandwidth(int bps);
386   bool SetOptions(const AudioOptions& options);
387   bool SetLocalSource(uint32_t ssrc, AudioSource* source);
388 
389   FakeVoiceEngine* engine_;
390   std::vector<AudioCodec> recv_codecs_;
391   std::vector<AudioCodec> send_codecs_;
392   std::map<uint32_t, double> output_scalings_;
393   std::map<uint32_t, int> output_delays_;
394   std::vector<DtmfInfo> dtmf_info_queue_;
395   AudioOptions options_;
396   std::map<uint32_t, std::unique_ptr<VoiceChannelAudioSink>> local_sinks_;
397   std::unique_ptr<webrtc::AudioSinkInterface> sink_;
398   int max_bps_;
399 };
400 
401 // A helper function to compare the FakeVoiceMediaChannel::DtmfInfo.
402 bool CompareDtmfInfo(const FakeVoiceMediaChannel::DtmfInfo& info,
403                      uint32_t ssrc,
404                      int event_code,
405                      int duration);
406 
407 class FakeVideoMediaChannel : public RtpHelper<VideoMediaChannel> {
408  public:
409   FakeVideoMediaChannel(FakeVideoEngine* engine, const VideoOptions& options);
410 
411   ~FakeVideoMediaChannel();
412 
413   const std::vector<VideoCodec>& recv_codecs() const;
414   const std::vector<VideoCodec>& send_codecs() const;
415   const std::vector<VideoCodec>& codecs() const;
416   bool rendering() const;
417   const VideoOptions& options() const;
418   const std::map<uint32_t, rtc::VideoSinkInterface<webrtc::VideoFrame>*>&
419   sinks() const;
420   int max_bps() const;
421   bool SetSendParameters(const VideoSendParameters& params) override;
422   bool SetRecvParameters(const VideoRecvParameters& params) override;
423   bool AddSendStream(const StreamParams& sp) override;
424   bool RemoveSendStream(uint32_t ssrc) override;
425 
426   bool GetSendCodec(VideoCodec* send_codec) override;
427   bool SetSink(uint32_t ssrc,
428                rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override;
429   void SetDefaultSink(
430       rtc::VideoSinkInterface<webrtc::VideoFrame>* sink) override;
431   bool HasSink(uint32_t ssrc) const;
432 
433   bool SetSend(bool send) override;
434   bool SetVideoSend(
435       uint32_t ssrc,
436       const VideoOptions* options,
437       rtc::VideoSourceInterface<webrtc::VideoFrame>* source) override;
438 
439   bool HasSource(uint32_t ssrc) const;
440   bool AddRecvStream(const StreamParams& sp) override;
441   bool RemoveRecvStream(uint32_t ssrc) override;
442 
443   void FillBitrateInfo(BandwidthEstimationInfo* bwe_info) override;
444   bool GetStats(VideoMediaInfo* info) override;
445 
446   std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const override;
447 
448   bool SetBaseMinimumPlayoutDelayMs(uint32_t ssrc, int delay_ms) override;
449   absl::optional<int> GetBaseMinimumPlayoutDelayMs(
450       uint32_t ssrc) const override;
451 
452   void SetRecordableEncodedFrameCallback(
453       uint32_t ssrc,
454       std::function<void(const webrtc::RecordableEncodedFrame&)> callback)
455       override;
456   void ClearRecordableEncodedFrameCallback(uint32_t ssrc) override;
457   void GenerateKeyFrame(uint32_t ssrc) override;
458 
459  private:
460   bool SetRecvCodecs(const std::vector<VideoCodec>& codecs);
461   bool SetSendCodecs(const std::vector<VideoCodec>& codecs);
462   bool SetOptions(const VideoOptions& options);
463   bool SetMaxSendBandwidth(int bps);
464 
465   FakeVideoEngine* engine_;
466   std::vector<VideoCodec> recv_codecs_;
467   std::vector<VideoCodec> send_codecs_;
468   std::map<uint32_t, rtc::VideoSinkInterface<webrtc::VideoFrame>*> sinks_;
469   std::map<uint32_t, rtc::VideoSourceInterface<webrtc::VideoFrame>*> sources_;
470   std::map<uint32_t, int> output_delays_;
471   VideoOptions options_;
472   int max_bps_;
473 };
474 
475 // Dummy option class, needed for the DataTraits abstraction in
476 // channel_unittest.c.
477 class DataOptions {};
478 
479 class FakeDataMediaChannel : public RtpHelper<DataMediaChannel> {
480  public:
481   explicit FakeDataMediaChannel(void* unused, const DataOptions& options);
482   ~FakeDataMediaChannel();
483   const std::vector<DataCodec>& recv_codecs() const;
484   const std::vector<DataCodec>& send_codecs() const;
485   const std::vector<DataCodec>& codecs() const;
486   int max_bps() const;
487 
488   bool SetSendParameters(const DataSendParameters& params) override;
489   bool SetRecvParameters(const DataRecvParameters& params) override;
490   bool SetSend(bool send) override;
491   bool SetReceive(bool receive) override;
492   bool AddRecvStream(const StreamParams& sp) override;
493   bool RemoveRecvStream(uint32_t ssrc) override;
494 
495   bool SendData(const SendDataParams& params,
496                 const rtc::CopyOnWriteBuffer& payload,
497                 SendDataResult* result) override;
498 
499   SendDataParams last_sent_data_params();
500   std::string last_sent_data();
501   bool is_send_blocked();
502   void set_send_blocked(bool blocked);
503 
504  private:
505   bool SetRecvCodecs(const std::vector<DataCodec>& codecs);
506   bool SetSendCodecs(const std::vector<DataCodec>& codecs);
507   bool SetMaxSendBandwidth(int bps);
508 
509   std::vector<DataCodec> recv_codecs_;
510   std::vector<DataCodec> send_codecs_;
511   SendDataParams last_sent_data_params_;
512   std::string last_sent_data_;
513   bool send_blocked_;
514   int max_bps_;
515 };
516 
517 class FakeVoiceEngine : public VoiceEngineInterface {
518  public:
519   FakeVoiceEngine();
520   void Init() override;
521   rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const override;
522 
523   VoiceMediaChannel* CreateMediaChannel(
524       webrtc::Call* call,
525       const MediaConfig& config,
526       const AudioOptions& options,
527       const webrtc::CryptoOptions& crypto_options) override;
528   FakeVoiceMediaChannel* GetChannel(size_t index);
529   void UnregisterChannel(VoiceMediaChannel* channel);
530 
531   // TODO(ossu): For proper testing, These should either individually settable
532   //             or the voice engine should reference mockable factories.
533   const std::vector<AudioCodec>& send_codecs() const override;
534   const std::vector<AudioCodec>& recv_codecs() const override;
535   void SetCodecs(const std::vector<AudioCodec>& codecs);
536   void SetRecvCodecs(const std::vector<AudioCodec>& codecs);
537   void SetSendCodecs(const std::vector<AudioCodec>& codecs);
538   int GetInputLevel();
539   bool StartAecDump(webrtc::FileWrapper file, int64_t max_size_bytes) override;
540   void StopAecDump() override;
541   std::vector<webrtc::RtpHeaderExtensionCapability> GetRtpHeaderExtensions()
542       const override;
543   void SetRtpHeaderExtensions(
544       std::vector<webrtc::RtpHeaderExtensionCapability> header_extensions);
545 
546  private:
547   std::vector<FakeVoiceMediaChannel*> channels_;
548   std::vector<AudioCodec> recv_codecs_;
549   std::vector<AudioCodec> send_codecs_;
550   bool fail_create_channel_;
551   std::vector<webrtc::RtpHeaderExtensionCapability> header_extensions_;
552 
553   friend class FakeMediaEngine;
554 };
555 
556 class FakeVideoEngine : public VideoEngineInterface {
557  public:
558   FakeVideoEngine();
559   bool SetOptions(const VideoOptions& options);
560   VideoMediaChannel* CreateMediaChannel(
561       webrtc::Call* call,
562       const MediaConfig& config,
563       const VideoOptions& options,
564       const webrtc::CryptoOptions& crypto_options,
565       webrtc::VideoBitrateAllocatorFactory* video_bitrate_allocator_factory)
566       override;
567   FakeVideoMediaChannel* GetChannel(size_t index);
568   void UnregisterChannel(VideoMediaChannel* channel);
569   std::vector<VideoCodec> send_codecs() const override;
570   std::vector<VideoCodec> recv_codecs() const override;
571   void SetSendCodecs(const std::vector<VideoCodec>& codecs);
572   void SetRecvCodecs(const std::vector<VideoCodec>& codecs);
573   bool SetCapture(bool capture);
574   std::vector<webrtc::RtpHeaderExtensionCapability> GetRtpHeaderExtensions()
575       const override;
576   void SetRtpHeaderExtensions(
577       std::vector<webrtc::RtpHeaderExtensionCapability> header_extensions);
578 
579  private:
580   std::vector<FakeVideoMediaChannel*> channels_;
581   std::vector<VideoCodec> send_codecs_;
582   std::vector<VideoCodec> recv_codecs_;
583   bool capture_;
584   VideoOptions options_;
585   bool fail_create_channel_;
586   std::vector<webrtc::RtpHeaderExtensionCapability> header_extensions_;
587 
588   friend class FakeMediaEngine;
589 };
590 
591 class FakeMediaEngine : public CompositeMediaEngine {
592  public:
593   FakeMediaEngine();
594 
595   ~FakeMediaEngine() override;
596 
597   void SetAudioCodecs(const std::vector<AudioCodec>& codecs);
598   void SetAudioRecvCodecs(const std::vector<AudioCodec>& codecs);
599   void SetAudioSendCodecs(const std::vector<AudioCodec>& codecs);
600   void SetVideoCodecs(const std::vector<VideoCodec>& codecs);
601 
602   FakeVoiceMediaChannel* GetVoiceChannel(size_t index);
603   FakeVideoMediaChannel* GetVideoChannel(size_t index);
604 
605   void set_fail_create_channel(bool fail);
606 
607  private:
608   FakeVoiceEngine* const voice_;
609   FakeVideoEngine* const video_;
610 };
611 
612 // Have to come afterwards due to declaration order
613 
614 class FakeDataEngine : public DataEngineInterface {
615  public:
616   DataMediaChannel* CreateChannel(const MediaConfig& config) override;
617 
618   FakeDataMediaChannel* GetChannel(size_t index);
619 
620   void UnregisterChannel(DataMediaChannel* channel);
621 
622   void SetDataCodecs(const std::vector<DataCodec>& data_codecs);
623 
624   const std::vector<DataCodec>& data_codecs() override;
625 
626  private:
627   std::vector<FakeDataMediaChannel*> channels_;
628   std::vector<DataCodec> data_codecs_;
629 };
630 
631 }  // namespace cricket
632 
633 #endif  // MEDIA_BASE_FAKE_MEDIA_ENGINE_H_
634