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henrike@webrtc.org28e20752013-07-10 00:45:36 +00001/*
2 * libjingle
3 * Copyright 2004 Google Inc.
4 *
5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met:
7 *
8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice,
11 * this list of conditions and the following disclaimer in the documentation
12 * and/or other materials provided with the distribution.
13 * 3. The name of the author may not be used to endorse or promote products
14 * derived from this software without specific prior written permission.
15 *
16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26 */
27
28#ifndef TALK_MEDIA_BASE_FAKEMEDIAENGINE_H_
29#define TALK_MEDIA_BASE_FAKEMEDIAENGINE_H_
30
31#include <list>
32#include <map>
33#include <set>
34#include <string>
35#include <vector>
36
henrike@webrtc.org1e09a712013-07-26 19:17:59 +000037#include "talk/media/base/audiorenderer.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000038#include "talk/media/base/mediaengine.h"
39#include "talk/media/base/rtputils.h"
40#include "talk/media/base/streamparams.h"
henrike@webrtc.org269fb4b2014-10-28 22:20:11 +000041#include "webrtc/p2p/base/sessiondescription.h"
buildbot@webrtc.orga09a9992014-08-13 17:26:08 +000042#include "webrtc/base/buffer.h"
43#include "webrtc/base/stringutils.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000044
45namespace cricket {
46
47class FakeMediaEngine;
48class FakeVideoEngine;
49class FakeVoiceEngine;
50
51// A common helper class that handles sending and receiving RTP/RTCP packets.
52template <class Base> class RtpHelper : public Base {
53 public:
54 RtpHelper()
55 : sending_(false),
56 playout_(false),
57 fail_set_send_codecs_(false),
58 fail_set_recv_codecs_(false),
59 send_ssrc_(0),
60 ready_to_send_(false) {}
61 const std::vector<RtpHeaderExtension>& recv_extensions() {
62 return recv_extensions_;
63 }
64 const std::vector<RtpHeaderExtension>& send_extensions() {
65 return send_extensions_;
66 }
67 bool sending() const { return sending_; }
68 bool playout() const { return playout_; }
69 const std::list<std::string>& rtp_packets() const { return rtp_packets_; }
70 const std::list<std::string>& rtcp_packets() const { return rtcp_packets_; }
71
72 bool SendRtp(const void* data, int len) {
henrike@webrtc.org1e09a712013-07-26 19:17:59 +000073 if (!sending_) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +000074 return false;
75 }
Karl Wiberg94784372015-04-20 14:03:07 +020076 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
77 kMaxRtpPacketLen);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +000078 return Base::SendPacket(&packet);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000079 }
80 bool SendRtcp(const void* data, int len) {
Karl Wiberg94784372015-04-20 14:03:07 +020081 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
82 kMaxRtpPacketLen);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +000083 return Base::SendRtcp(&packet);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000084 }
85
86 bool CheckRtp(const void* data, int len) {
87 bool success = !rtp_packets_.empty();
88 if (success) {
89 std::string packet = rtp_packets_.front();
90 rtp_packets_.pop_front();
91 success = (packet == std::string(static_cast<const char*>(data), len));
92 }
93 return success;
94 }
95 bool CheckRtcp(const void* data, int len) {
96 bool success = !rtcp_packets_.empty();
97 if (success) {
98 std::string packet = rtcp_packets_.front();
99 rtcp_packets_.pop_front();
100 success = (packet == std::string(static_cast<const char*>(data), len));
101 }
102 return success;
103 }
104 bool CheckNoRtp() { return rtp_packets_.empty(); }
105 bool CheckNoRtcp() { return rtcp_packets_.empty(); }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000106 void set_fail_set_send_codecs(bool fail) { fail_set_send_codecs_ = fail; }
107 void set_fail_set_recv_codecs(bool fail) { fail_set_recv_codecs_ = fail; }
108 virtual bool AddSendStream(const StreamParams& sp) {
109 if (std::find(send_streams_.begin(), send_streams_.end(), sp) !=
110 send_streams_.end()) {
111 return false;
112 }
113 send_streams_.push_back(sp);
114 return true;
115 }
116 virtual bool RemoveSendStream(uint32 ssrc) {
117 return RemoveStreamBySsrc(&send_streams_, ssrc);
118 }
119 virtual bool AddRecvStream(const StreamParams& sp) {
120 if (std::find(receive_streams_.begin(), receive_streams_.end(), sp) !=
121 receive_streams_.end()) {
122 return false;
123 }
124 receive_streams_.push_back(sp);
125 return true;
126 }
127 virtual bool RemoveRecvStream(uint32 ssrc) {
128 return RemoveStreamBySsrc(&receive_streams_, ssrc);
129 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000130 bool IsStreamMuted(uint32 ssrc) const {
131 bool ret = muted_streams_.find(ssrc) != muted_streams_.end();
132 // If |ssrc = 0| check if the first send stream is muted.
133 if (!ret && ssrc == 0 && !send_streams_.empty()) {
134 return muted_streams_.find(send_streams_[0].first_ssrc()) !=
135 muted_streams_.end();
136 }
137 return ret;
138 }
139 const std::vector<StreamParams>& send_streams() const {
140 return send_streams_;
141 }
142 const std::vector<StreamParams>& recv_streams() const {
143 return receive_streams_;
144 }
145 bool HasRecvStream(uint32 ssrc) const {
tommi@webrtc.org586f2ed2015-01-22 23:00:41 +0000146 return GetStreamBySsrc(receive_streams_, ssrc) != nullptr;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000147 }
148 bool HasSendStream(uint32 ssrc) const {
tommi@webrtc.org586f2ed2015-01-22 23:00:41 +0000149 return GetStreamBySsrc(send_streams_, ssrc) != nullptr;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000150 }
151 // TODO(perkj): This is to support legacy unit test that only check one
152 // sending stream.
pthatcher@webrtc.org40b276e2014-12-12 02:44:30 +0000153 uint32 send_ssrc() const {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000154 if (send_streams_.empty())
155 return 0;
156 return send_streams_[0].first_ssrc();
157 }
158
159 // TODO(perkj): This is to support legacy unit test that only check one
160 // sending stream.
161 const std::string rtcp_cname() {
162 if (send_streams_.empty())
163 return "";
164 return send_streams_[0].cname;
165 }
166
167 bool ready_to_send() const {
168 return ready_to_send_;
169 }
170
171 protected:
solenberg1dd98f32015-09-10 01:57:14 -0700172 bool MuteStream(uint32 ssrc, bool mute) {
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200173 if (!HasSendStream(ssrc) && ssrc != 0) {
solenberg1dd98f32015-09-10 01:57:14 -0700174 return false;
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200175 }
176 if (mute) {
solenberg1dd98f32015-09-10 01:57:14 -0700177 muted_streams_.insert(ssrc);
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200178 } else {
solenberg1dd98f32015-09-10 01:57:14 -0700179 muted_streams_.erase(ssrc);
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200180 }
solenberg1dd98f32015-09-10 01:57:14 -0700181 return true;
182 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000183 bool set_sending(bool send) {
184 sending_ = send;
185 return true;
186 }
187 void set_playout(bool playout) { playout_ = playout; }
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200188 bool SetRecvRtpHeaderExtensions(
189 const std::vector<RtpHeaderExtension>& extensions) {
190 recv_extensions_ = extensions;
191 return true;
192 }
193 bool SetSendRtpHeaderExtensions(
194 const std::vector<RtpHeaderExtension>& extensions) {
195 send_extensions_ = extensions;
196 return true;
197 }
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000198 virtual void OnPacketReceived(rtc::Buffer* packet,
199 const rtc::PacketTime& packet_time) {
Karl Wiberg94784372015-04-20 14:03:07 +0200200 rtp_packets_.push_back(std::string(packet->data<char>(), packet->size()));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000201 }
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000202 virtual void OnRtcpReceived(rtc::Buffer* packet,
203 const rtc::PacketTime& packet_time) {
Karl Wiberg94784372015-04-20 14:03:07 +0200204 rtcp_packets_.push_back(std::string(packet->data<char>(), packet->size()));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000205 }
206 virtual void OnReadyToSend(bool ready) {
207 ready_to_send_ = ready;
208 }
209 bool fail_set_send_codecs() const { return fail_set_send_codecs_; }
210 bool fail_set_recv_codecs() const { return fail_set_recv_codecs_; }
211
212 private:
213 bool sending_;
214 bool playout_;
215 std::vector<RtpHeaderExtension> recv_extensions_;
216 std::vector<RtpHeaderExtension> send_extensions_;
217 std::list<std::string> rtp_packets_;
218 std::list<std::string> rtcp_packets_;
219 std::vector<StreamParams> send_streams_;
220 std::vector<StreamParams> receive_streams_;
221 std::set<uint32> muted_streams_;
222 bool fail_set_send_codecs_;
223 bool fail_set_recv_codecs_;
224 uint32 send_ssrc_;
225 std::string rtcp_cname_;
226 bool ready_to_send_;
227};
228
229class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
230 public:
231 struct DtmfInfo {
232 DtmfInfo(uint32 ssrc, int event_code, int duration, int flags)
233 : ssrc(ssrc), event_code(event_code), duration(duration), flags(flags) {
234 }
235 uint32 ssrc;
236 int event_code;
237 int duration;
238 int flags;
239 };
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200240 explicit FakeVoiceMediaChannel(FakeVoiceEngine* engine,
241 const AudioOptions& options)
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000242 : engine_(engine),
243 fail_set_send_(false),
244 ringback_tone_ssrc_(0),
245 ringback_tone_play_(false),
246 ringback_tone_loop_(false),
247 time_since_last_typing_(-1) {
248 output_scalings_[0] = OutputScaling(); // For default channel.
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200249 SetOptions(options);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000250 }
251 ~FakeVoiceMediaChannel();
252 const std::vector<AudioCodec>& recv_codecs() const { return recv_codecs_; }
253 const std::vector<AudioCodec>& send_codecs() const { return send_codecs_; }
254 const std::vector<AudioCodec>& codecs() const { return send_codecs(); }
255 const std::vector<DtmfInfo>& dtmf_info_queue() const {
256 return dtmf_info_queue_;
257 }
258 const AudioOptions& options() const { return options_; }
259
260 uint32 ringback_tone_ssrc() const { return ringback_tone_ssrc_; }
261 bool ringback_tone_play() const { return ringback_tone_play_; }
262 bool ringback_tone_loop() const { return ringback_tone_loop_; }
263
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200264 virtual bool SetSendParameters(const AudioSendParameters& params) {
265 return (SetSendCodecs(params.codecs) &&
266 SetSendRtpHeaderExtensions(params.extensions) &&
267 SetMaxSendBandwidth(params.max_bandwidth_bps) &&
268 SetOptions(params.options));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000269 }
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200270
271 virtual bool SetRecvParameters(const AudioRecvParameters& params) {
272 return (SetRecvCodecs(params.codecs) &&
273 SetRecvRtpHeaderExtensions(params.extensions));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000274 }
275 virtual bool SetPlayout(bool playout) {
276 set_playout(playout);
277 return true;
278 }
279 virtual bool SetSend(SendFlags flag) {
280 if (fail_set_send_) {
281 return false;
282 }
283 return set_sending(flag != SEND_NOTHING);
284 }
solenberg1dd98f32015-09-10 01:57:14 -0700285 virtual bool SetAudioSend(uint32 ssrc, bool mute,
286 const AudioOptions* options,
287 AudioRenderer* renderer) {
288 if (!SetLocalRenderer(ssrc, renderer)) {
289 return false;
290 }
291 if (!RtpHelper<VoiceMediaChannel>::MuteStream(ssrc, mute)) {
292 return false;
293 }
294 if (!mute && options) {
295 return SetOptions(*options);
296 }
297 return true;
298 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000299 virtual bool AddRecvStream(const StreamParams& sp) {
300 if (!RtpHelper<VoiceMediaChannel>::AddRecvStream(sp))
301 return false;
302 output_scalings_[sp.first_ssrc()] = OutputScaling();
303 return true;
304 }
305 virtual bool RemoveRecvStream(uint32 ssrc) {
306 if (!RtpHelper<VoiceMediaChannel>::RemoveRecvStream(ssrc))
307 return false;
308 output_scalings_.erase(ssrc);
309 return true;
310 }
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000311 virtual bool SetRemoteRenderer(uint32 ssrc, AudioRenderer* renderer) {
312 std::map<uint32, AudioRenderer*>::iterator it =
313 remote_renderers_.find(ssrc);
314 if (renderer) {
315 if (it != remote_renderers_.end()) {
316 ASSERT(it->second == renderer);
317 } else {
318 remote_renderers_.insert(std::make_pair(ssrc, renderer));
319 renderer->AddChannel(0);
320 }
321 } else {
322 if (it != remote_renderers_.end()) {
323 it->second->RemoveChannel(0);
324 remote_renderers_.erase(it);
325 } else {
326 return false;
327 }
328 }
329 return true;
330 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000331
332 virtual bool GetActiveStreams(AudioInfo::StreamList* streams) { return true; }
333 virtual int GetOutputLevel() { return 0; }
334 void set_time_since_last_typing(int ms) { time_since_last_typing_ = ms; }
335 virtual int GetTimeSinceLastTyping() { return time_since_last_typing_; }
336 virtual void SetTypingDetectionParameters(
337 int time_window, int cost_per_typing, int reporting_threshold,
338 int penalty_decay, int type_event_delay) {}
339
340 virtual bool SetRingbackTone(const char* buf, int len) { return true; }
341 virtual bool PlayRingbackTone(uint32 ssrc, bool play, bool loop) {
342 ringback_tone_ssrc_ = ssrc;
343 ringback_tone_play_ = play;
344 ringback_tone_loop_ = loop;
345 return true;
346 }
347
348 virtual bool CanInsertDtmf() {
349 for (std::vector<AudioCodec>::const_iterator it = send_codecs_.begin();
350 it != send_codecs_.end(); ++it) {
351 // Find the DTMF telephone event "codec".
352 if (_stricmp(it->name.c_str(), "telephone-event") == 0) {
353 return true;
354 }
355 }
356 return false;
357 }
358 virtual bool InsertDtmf(uint32 ssrc, int event_code, int duration,
359 int flags) {
360 dtmf_info_queue_.push_back(DtmfInfo(ssrc, event_code, duration, flags));
361 return true;
362 }
363
364 virtual bool SetOutputScaling(uint32 ssrc, double left, double right) {
365 if (0 == ssrc) {
366 std::map<uint32, OutputScaling>::iterator it;
367 for (it = output_scalings_.begin(); it != output_scalings_.end(); ++it) {
368 it->second.left = left;
369 it->second.right = right;
370 }
371 return true;
372 } else if (output_scalings_.find(ssrc) != output_scalings_.end()) {
373 output_scalings_[ssrc].left = left;
374 output_scalings_[ssrc].right = right;
375 return true;
376 }
377 return false;
378 }
solenbergbb741b32015-09-07 03:56:38 -0700379 bool GetOutputScaling(uint32 ssrc, double* left, double* right) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000380 if (output_scalings_.find(ssrc) == output_scalings_.end())
381 return false;
382 *left = output_scalings_[ssrc].left;
383 *right = output_scalings_[ssrc].right;
384 return true;
385 }
386
387 virtual bool GetStats(VoiceMediaInfo* info) { return false; }
388 virtual void GetLastMediaError(uint32* ssrc,
389 VoiceMediaChannel::Error* error) {
390 *ssrc = 0;
391 *error = fail_set_send_ ? VoiceMediaChannel::ERROR_REC_DEVICE_OPEN_FAILED
392 : VoiceMediaChannel::ERROR_NONE;
393 }
394
395 void set_fail_set_send(bool fail) { fail_set_send_ = fail; }
396 void TriggerError(uint32 ssrc, VoiceMediaChannel::Error error) {
397 VoiceMediaChannel::SignalMediaError(ssrc, error);
398 }
399
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000400 private:
401 struct OutputScaling {
402 OutputScaling() : left(1.0), right(1.0) {}
403 double left, right;
404 };
405
henrike@webrtc.orga7b98182014-02-21 15:51:43 +0000406 class VoiceChannelAudioSink : public AudioRenderer::Sink {
407 public:
408 explicit VoiceChannelAudioSink(AudioRenderer* renderer)
409 : renderer_(renderer) {
410 renderer_->AddChannel(0);
411 renderer_->SetSink(this);
412 }
413 virtual ~VoiceChannelAudioSink() {
414 if (renderer_) {
415 renderer_->RemoveChannel(0);
416 renderer_->SetSink(NULL);
417 }
418 }
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000419 void OnData(const void* audio_data,
420 int bits_per_sample,
421 int sample_rate,
422 int number_of_channels,
Peter Kastingdce40cf2015-08-24 14:52:23 -0700423 size_t number_of_frames) override {}
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000424 void OnClose() override { renderer_ = NULL; }
henrike@webrtc.orga7b98182014-02-21 15:51:43 +0000425 AudioRenderer* renderer() const { return renderer_; }
426
427 private:
428 AudioRenderer* renderer_;
429 };
430
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200431 bool SetRecvCodecs(const std::vector<AudioCodec>& codecs) {
432 if (fail_set_recv_codecs()) {
433 // Fake the failure in SetRecvCodecs.
434 return false;
435 }
436 recv_codecs_ = codecs;
437 return true;
438 }
439 bool SetSendCodecs(const std::vector<AudioCodec>& codecs) {
440 if (fail_set_send_codecs()) {
441 // Fake the failure in SetSendCodecs.
442 return false;
443 }
444 send_codecs_ = codecs;
445 return true;
446 }
447 bool SetMaxSendBandwidth(int bps) { return true; }
448 bool SetOptions(const AudioOptions& options) {
449 // Does a "merge" of current options and set options.
450 options_.SetAll(options);
451 return true;
452 }
solenberg1dd98f32015-09-10 01:57:14 -0700453 bool SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer) {
454 auto it = local_renderers_.find(ssrc);
455 if (renderer) {
456 if (it != local_renderers_.end()) {
457 ASSERT(it->second->renderer() == renderer);
458 } else {
459 local_renderers_.insert(std::make_pair(
460 ssrc, new VoiceChannelAudioSink(renderer)));
461 }
462 } else {
463 if (it != local_renderers_.end()) {
464 delete it->second;
465 local_renderers_.erase(it);
466 }
467 }
468 return true;
469 }
henrike@webrtc.orga7b98182014-02-21 15:51:43 +0000470
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000471 FakeVoiceEngine* engine_;
472 std::vector<AudioCodec> recv_codecs_;
473 std::vector<AudioCodec> send_codecs_;
474 std::map<uint32, OutputScaling> output_scalings_;
475 std::vector<DtmfInfo> dtmf_info_queue_;
476 bool fail_set_send_;
477 uint32 ringback_tone_ssrc_;
478 bool ringback_tone_play_;
479 bool ringback_tone_loop_;
480 int time_since_last_typing_;
481 AudioOptions options_;
henrike@webrtc.orga7b98182014-02-21 15:51:43 +0000482 std::map<uint32, VoiceChannelAudioSink*> local_renderers_;
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000483 std::map<uint32, AudioRenderer*> remote_renderers_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000484};
485
486// A helper function to compare the FakeVoiceMediaChannel::DtmfInfo.
487inline bool CompareDtmfInfo(const FakeVoiceMediaChannel::DtmfInfo& info,
488 uint32 ssrc, int event_code, int duration,
489 int flags) {
490 return (info.duration == duration && info.event_code == event_code &&
491 info.flags == flags && info.ssrc == ssrc);
492}
493
494class FakeVideoMediaChannel : public RtpHelper<VideoMediaChannel> {
495 public:
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200496 explicit FakeVideoMediaChannel(FakeVideoEngine* engine,
497 const VideoOptions& options)
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000498 : engine_(engine),
499 sent_intra_frame_(false),
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000500 requested_intra_frame_(false),
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200501 max_bps_(-1) {
502 SetOptions(options);
503 }
buildbot@webrtc.org1ecbe452014-10-14 20:29:28 +0000504
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000505 ~FakeVideoMediaChannel();
506
507 const std::vector<VideoCodec>& recv_codecs() const { return recv_codecs_; }
508 const std::vector<VideoCodec>& send_codecs() const { return send_codecs_; }
509 const std::vector<VideoCodec>& codecs() const { return send_codecs(); }
510 bool rendering() const { return playout(); }
511 const VideoOptions& options() const { return options_; }
512 const std::map<uint32, VideoRenderer*>& renderers() const {
513 return renderers_;
514 }
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000515 int max_bps() const { return max_bps_; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000516 bool GetSendStreamFormat(uint32 ssrc, VideoFormat* format) {
517 if (send_formats_.find(ssrc) == send_formats_.end()) {
518 return false;
519 }
520 *format = send_formats_[ssrc];
521 return true;
522 }
523 virtual bool SetSendStreamFormat(uint32 ssrc, const VideoFormat& format) {
524 if (send_formats_.find(ssrc) == send_formats_.end()) {
525 return false;
526 }
527 send_formats_[ssrc] = format;
528 return true;
529 }
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200530 virtual bool SetSendParameters(const VideoSendParameters& params) {
531 return (SetSendCodecs(params.codecs) &&
532 SetSendRtpHeaderExtensions(params.extensions) &&
533 SetMaxSendBandwidth(params.max_bandwidth_bps) &&
534 SetOptions(params.options));
535 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000536
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200537 virtual bool SetRecvParameters(const VideoRecvParameters& params) {
538 return (SetRecvCodecs(params.codecs) &&
539 SetRecvRtpHeaderExtensions(params.extensions));
540 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000541 virtual bool AddSendStream(const StreamParams& sp) {
542 if (!RtpHelper<VideoMediaChannel>::AddSendStream(sp)) {
543 return false;
544 }
545 SetSendStreamDefaultFormat(sp.first_ssrc());
546 return true;
547 }
548 virtual bool RemoveSendStream(uint32 ssrc) {
549 send_formats_.erase(ssrc);
550 return RtpHelper<VideoMediaChannel>::RemoveSendStream(ssrc);
551 }
552
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000553 virtual bool GetSendCodec(VideoCodec* send_codec) {
554 if (send_codecs_.empty()) {
555 return false;
556 }
557 *send_codec = send_codecs_[0];
558 return true;
559 }
560 virtual bool SetRender(bool render) {
561 set_playout(render);
562 return true;
563 }
564 virtual bool SetRenderer(uint32 ssrc, VideoRenderer* r) {
565 if (ssrc != 0 && renderers_.find(ssrc) == renderers_.end()) {
566 return false;
567 }
568 if (ssrc != 0) {
569 renderers_[ssrc] = r;
570 }
571 return true;
572 }
573
574 virtual bool SetSend(bool send) { return set_sending(send); }
solenberg1dd98f32015-09-10 01:57:14 -0700575 virtual bool SetVideoSend(uint32 ssrc, bool mute,
576 const VideoOptions* options) {
577 if (!RtpHelper<VideoMediaChannel>::MuteStream(ssrc, mute)) {
578 return false;
579 }
580 if (!mute && options) {
581 return SetOptions(*options);
solenberg1dd98f32015-09-10 01:57:14 -0700582 }
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200583 return true;
solenberg1dd98f32015-09-10 01:57:14 -0700584 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000585 virtual bool SetCapturer(uint32 ssrc, VideoCapturer* capturer) {
586 capturers_[ssrc] = capturer;
587 return true;
588 }
589 bool HasCapturer(uint32 ssrc) const {
590 return capturers_.find(ssrc) != capturers_.end();
591 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000592 virtual bool AddRecvStream(const StreamParams& sp) {
593 if (!RtpHelper<VideoMediaChannel>::AddRecvStream(sp))
594 return false;
595 renderers_[sp.first_ssrc()] = NULL;
596 return true;
597 }
598 virtual bool RemoveRecvStream(uint32 ssrc) {
599 if (!RtpHelper<VideoMediaChannel>::RemoveRecvStream(ssrc))
600 return false;
601 renderers_.erase(ssrc);
602 return true;
603 }
604
pbos@webrtc.org058b1f12015-03-04 08:54:32 +0000605 virtual bool GetStats(VideoMediaInfo* info) { return false; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000606 virtual bool SendIntraFrame() {
607 sent_intra_frame_ = true;
608 return true;
609 }
610 virtual bool RequestIntraFrame() {
611 requested_intra_frame_ = true;
612 return true;
613 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000614 virtual void UpdateAspectRatio(int ratio_w, int ratio_h) {}
615 void set_sent_intra_frame(bool v) { sent_intra_frame_ = v; }
616 bool sent_intra_frame() const { return sent_intra_frame_; }
617 void set_requested_intra_frame(bool v) { requested_intra_frame_ = v; }
618 bool requested_intra_frame() const { return requested_intra_frame_; }
619
620 private:
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200621 bool SetRecvCodecs(const std::vector<VideoCodec>& codecs) {
622 if (fail_set_recv_codecs()) {
623 // Fake the failure in SetRecvCodecs.
624 return false;
625 }
626 recv_codecs_ = codecs;
627 return true;
628 }
629 bool SetSendCodecs(const std::vector<VideoCodec>& codecs) {
630 if (fail_set_send_codecs()) {
631 // Fake the failure in SetSendCodecs.
632 return false;
633 }
634 send_codecs_ = codecs;
635
636 for (std::vector<StreamParams>::const_iterator it = send_streams().begin();
637 it != send_streams().end(); ++it) {
638 SetSendStreamDefaultFormat(it->first_ssrc());
639 }
640 return true;
641 }
642 bool SetOptions(const VideoOptions& options) {
643 options_ = options;
644 return true;
645 }
646 bool SetMaxSendBandwidth(int bps) {
647 max_bps_ = bps;
648 return true;
649 }
650
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000651 // Be default, each send stream uses the first send codec format.
652 void SetSendStreamDefaultFormat(uint32 ssrc) {
653 if (!send_codecs_.empty()) {
654 send_formats_[ssrc] = VideoFormat(
655 send_codecs_[0].width, send_codecs_[0].height,
656 cricket::VideoFormat::FpsToInterval(send_codecs_[0].framerate),
657 cricket::FOURCC_I420);
658 }
659 }
660
661 FakeVideoEngine* engine_;
662 std::vector<VideoCodec> recv_codecs_;
663 std::vector<VideoCodec> send_codecs_;
664 std::map<uint32, VideoRenderer*> renderers_;
665 std::map<uint32, VideoFormat> send_formats_;
666 std::map<uint32, VideoCapturer*> capturers_;
667 bool sent_intra_frame_;
668 bool requested_intra_frame_;
669 VideoOptions options_;
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000670 int max_bps_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000671};
672
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000673class FakeDataMediaChannel : public RtpHelper<DataMediaChannel> {
674 public:
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200675 explicit FakeDataMediaChannel(void* unused, const DataOptions& options)
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000676 : send_blocked_(false), max_bps_(-1) {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000677 ~FakeDataMediaChannel() {}
678 const std::vector<DataCodec>& recv_codecs() const { return recv_codecs_; }
679 const std::vector<DataCodec>& send_codecs() const { return send_codecs_; }
680 const std::vector<DataCodec>& codecs() const { return send_codecs(); }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000681 int max_bps() const { return max_bps_; }
682
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200683 virtual bool SetSendParameters(const DataSendParameters& params) {
684 return (SetSendCodecs(params.codecs) &&
685 SetMaxSendBandwidth(params.max_bandwidth_bps));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000686 }
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200687 virtual bool SetRecvParameters(const DataRecvParameters& params) {
688 return SetRecvCodecs(params.codecs);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000689 }
690 virtual bool SetSend(bool send) { return set_sending(send); }
691 virtual bool SetReceive(bool receive) {
692 set_playout(receive);
693 return true;
694 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000695 virtual bool AddRecvStream(const StreamParams& sp) {
696 if (!RtpHelper<DataMediaChannel>::AddRecvStream(sp))
697 return false;
698 return true;
699 }
700 virtual bool RemoveRecvStream(uint32 ssrc) {
701 if (!RtpHelper<DataMediaChannel>::RemoveRecvStream(ssrc))
702 return false;
703 return true;
704 }
705
706 virtual bool SendData(const SendDataParams& params,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000707 const rtc::Buffer& payload,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000708 SendDataResult* result) {
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000709 if (send_blocked_) {
710 *result = SDR_BLOCK;
711 return false;
712 } else {
713 last_sent_data_params_ = params;
Karl Wiberg94784372015-04-20 14:03:07 +0200714 last_sent_data_ = std::string(payload.data<char>(), payload.size());
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000715 return true;
716 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000717 }
718
719 SendDataParams last_sent_data_params() { return last_sent_data_params_; }
720 std::string last_sent_data() { return last_sent_data_; }
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000721 bool is_send_blocked() { return send_blocked_; }
722 void set_send_blocked(bool blocked) { send_blocked_ = blocked; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000723
724 private:
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200725 bool SetRecvCodecs(const std::vector<DataCodec>& codecs) {
726 if (fail_set_recv_codecs()) {
727 // Fake the failure in SetRecvCodecs.
728 return false;
729 }
730 recv_codecs_ = codecs;
731 return true;
732 }
733 bool SetSendCodecs(const std::vector<DataCodec>& codecs) {
734 if (fail_set_send_codecs()) {
735 // Fake the failure in SetSendCodecs.
736 return false;
737 }
738 send_codecs_ = codecs;
739 return true;
740 }
741 bool SetMaxSendBandwidth(int bps) {
742 max_bps_ = bps;
743 return true;
744 }
745
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000746 std::vector<DataCodec> recv_codecs_;
747 std::vector<DataCodec> send_codecs_;
748 SendDataParams last_sent_data_params_;
749 std::string last_sent_data_;
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000750 bool send_blocked_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000751 int max_bps_;
752};
753
754// A base class for all of the shared parts between FakeVoiceEngine
755// and FakeVideoEngine.
756class FakeBaseEngine {
757 public:
758 FakeBaseEngine()
759 : loglevel_(-1),
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000760 options_changed_(false),
761 fail_create_channel_(false) {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000762 void SetLogging(int level, const char* filter) {
763 loglevel_ = level;
764 logfilter_ = filter;
765 }
766
767 void set_fail_create_channel(bool fail) { fail_create_channel_ = fail; }
768
769 const std::vector<RtpHeaderExtension>& rtp_header_extensions() const {
770 return rtp_header_extensions_;
771 }
henrike@webrtc.org79047f92014-03-06 23:46:59 +0000772 void set_rtp_header_extensions(
773 const std::vector<RtpHeaderExtension>& extensions) {
774 rtp_header_extensions_ = extensions;
775 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000776
777 protected:
778 int loglevel_;
779 std::string logfilter_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000780 // Flag used by optionsmessagehandler_unittest for checking whether any
781 // relevant setting has been updated.
782 // TODO(thaloun): Replace with explicit checks of before & after values.
783 bool options_changed_;
784 bool fail_create_channel_;
785 std::vector<RtpHeaderExtension> rtp_header_extensions_;
786};
787
788class FakeVoiceEngine : public FakeBaseEngine {
789 public:
790 FakeVoiceEngine()
791 : output_volume_(-1),
792 delay_offset_(0),
793 rx_processor_(NULL),
794 tx_processor_(NULL) {
795 // Add a fake audio codec. Note that the name must not be "" as there are
796 // sanity checks against that.
797 codecs_.push_back(AudioCodec(101, "fake_audio_codec", 0, 0, 1, 0));
798 }
Fredrik Solenberg9a416bd2015-05-22 09:04:09 +0200799 bool Init(rtc::Thread* worker_thread) { return true; }
800 void Terminate() {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000801 int GetCapabilities() { return AUDIO_SEND | AUDIO_RECV; }
Fredrik Solenberg709ed672015-09-15 12:26:33 +0200802 webrtc::VoiceEngine* GetVoE() { return nullptr; }
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000803 AudioOptions GetOptions() const {
804 return options_;
805 }
806 bool SetOptions(const AudioOptions& options) {
807 options_ = options;
808 options_changed_ = true;
809 return true;
810 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000811
Fredrik Solenberg709ed672015-09-15 12:26:33 +0200812 VoiceMediaChannel* CreateChannel(webrtc::Call* call,
813 const AudioOptions& options) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000814 if (fail_create_channel_) {
Jelena Marusicc28a8962015-05-29 15:05:44 +0200815 return nullptr;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000816 }
817
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200818 FakeVoiceMediaChannel* ch = new FakeVoiceMediaChannel(this, options);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000819 channels_.push_back(ch);
820 return ch;
821 }
822 FakeVoiceMediaChannel* GetChannel(size_t index) {
823 return (channels_.size() > index) ? channels_[index] : NULL;
824 }
825 void UnregisterChannel(VoiceMediaChannel* channel) {
826 channels_.erase(std::find(channels_.begin(), channels_.end(), channel));
827 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000828
829 const std::vector<AudioCodec>& codecs() { return codecs_; }
830 void SetCodecs(const std::vector<AudioCodec> codecs) { codecs_ = codecs; }
831
832 bool SetDelayOffset(int offset) {
833 delay_offset_ = offset;
834 return true;
835 }
836
837 bool SetDevices(const Device* in_device, const Device* out_device) {
838 in_device_ = (in_device) ? in_device->name : "";
839 out_device_ = (out_device) ? out_device->name : "";
840 options_changed_ = true;
841 return true;
842 }
843
844 bool GetOutputVolume(int* level) {
845 *level = output_volume_;
846 return true;
847 }
848
849 bool SetOutputVolume(int level) {
850 output_volume_ = level;
851 options_changed_ = true;
852 return true;
853 }
854
855 int GetInputLevel() { return 0; }
856
857 bool SetLocalMonitor(bool enable) { return true; }
858
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000859 bool StartAecDump(rtc::PlatformFile file) { return false; }
wu@webrtc.orga9890802013-12-13 00:21:03 +0000860
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000861 bool RegisterProcessor(uint32 ssrc, VoiceProcessor* voice_processor,
862 MediaProcessorDirection direction) {
863 if (direction == MPD_RX) {
864 rx_processor_ = voice_processor;
865 return true;
866 } else if (direction == MPD_TX) {
867 tx_processor_ = voice_processor;
868 return true;
869 }
870 return false;
871 }
872
873 bool UnregisterProcessor(uint32 ssrc, VoiceProcessor* voice_processor,
874 MediaProcessorDirection direction) {
875 bool unregistered = false;
876 if (direction & MPD_RX) {
877 rx_processor_ = NULL;
878 unregistered = true;
879 }
880 if (direction & MPD_TX) {
881 tx_processor_ = NULL;
882 unregistered = true;
883 }
884 return unregistered;
885 }
886
887 private:
888 std::vector<FakeVoiceMediaChannel*> channels_;
889 std::vector<AudioCodec> codecs_;
890 int output_volume_;
891 int delay_offset_;
892 std::string in_device_;
893 std::string out_device_;
894 VoiceProcessor* rx_processor_;
895 VoiceProcessor* tx_processor_;
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000896 AudioOptions options_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000897
898 friend class FakeMediaEngine;
899};
900
901class FakeVideoEngine : public FakeBaseEngine {
902 public:
Fredrik Solenberg709ed672015-09-15 12:26:33 +0200903 FakeVideoEngine() : capture_(false) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000904 // Add a fake video codec. Note that the name must not be "" as there are
905 // sanity checks against that.
906 codecs_.push_back(VideoCodec(0, "fake_video_codec", 0, 0, 0, 0));
907 }
Fredrik Solenberg9a416bd2015-05-22 09:04:09 +0200908 void Init() {}
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000909 bool SetOptions(const VideoOptions& options) {
910 options_ = options;
911 options_changed_ = true;
912 return true;
913 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000914 int GetCapabilities() { return VIDEO_SEND | VIDEO_RECV; }
915 bool SetDefaultEncoderConfig(const VideoEncoderConfig& config) {
916 default_encoder_config_ = config;
917 return true;
918 }
919 const VideoEncoderConfig& default_encoder_config() const {
920 return default_encoder_config_;
921 }
922
Fredrik Solenberg709ed672015-09-15 12:26:33 +0200923 VideoMediaChannel* CreateChannel(webrtc::Call* call,
924 const VideoOptions& options) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000925 if (fail_create_channel_) {
926 return NULL;
927 }
928
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200929 FakeVideoMediaChannel* ch = new FakeVideoMediaChannel(this, options);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000930 channels_.push_back(ch);
931 return ch;
932 }
933 FakeVideoMediaChannel* GetChannel(size_t index) {
934 return (channels_.size() > index) ? channels_[index] : NULL;
935 }
936 void UnregisterChannel(VideoMediaChannel* channel) {
937 channels_.erase(std::find(channels_.begin(), channels_.end(), channel));
938 }
939
940 const std::vector<VideoCodec>& codecs() const { return codecs_; }
941 bool FindCodec(const VideoCodec& in) {
942 for (size_t i = 0; i < codecs_.size(); ++i) {
943 if (codecs_[i].Matches(in)) {
944 return true;
945 }
946 }
947 return false;
948 }
949 void SetCodecs(const std::vector<VideoCodec> codecs) { codecs_ = codecs; }
950
951 bool SetCaptureDevice(const Device* device) {
952 in_device_ = (device) ? device->name : "";
953 options_changed_ = true;
954 return true;
955 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000956 bool SetCapture(bool capture) {
957 capture_ = capture;
958 return true;
959 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000960
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000961 private:
962 std::vector<FakeVideoMediaChannel*> channels_;
963 std::vector<VideoCodec> codecs_;
964 VideoEncoderConfig default_encoder_config_;
965 std::string in_device_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000966 bool capture_;
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000967 VideoOptions options_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000968
969 friend class FakeMediaEngine;
970};
971
972class FakeMediaEngine :
973 public CompositeMediaEngine<FakeVoiceEngine, FakeVideoEngine> {
974 public:
975 FakeMediaEngine() {
976 voice_ = FakeVoiceEngine();
977 video_ = FakeVideoEngine();
978 }
979 virtual ~FakeMediaEngine() {}
980
henrike@webrtc.org79047f92014-03-06 23:46:59 +0000981 void SetAudioCodecs(const std::vector<AudioCodec>& codecs) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000982 voice_.SetCodecs(codecs);
983 }
henrike@webrtc.org79047f92014-03-06 23:46:59 +0000984 void SetVideoCodecs(const std::vector<VideoCodec>& codecs) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000985 video_.SetCodecs(codecs);
986 }
987
henrike@webrtc.org79047f92014-03-06 23:46:59 +0000988 void SetAudioRtpHeaderExtensions(
989 const std::vector<RtpHeaderExtension>& extensions) {
990 voice_.set_rtp_header_extensions(extensions);
991 }
992 void SetVideoRtpHeaderExtensions(
993 const std::vector<RtpHeaderExtension>& extensions) {
994 video_.set_rtp_header_extensions(extensions);
995 }
996
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000997 FakeVoiceMediaChannel* GetVoiceChannel(size_t index) {
998 return voice_.GetChannel(index);
999 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001000 FakeVideoMediaChannel* GetVideoChannel(size_t index) {
1001 return video_.GetChannel(index);
1002 }
1003
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +00001004 AudioOptions audio_options() const { return voice_.options_; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001005 int audio_delay_offset() const { return voice_.delay_offset_; }
1006 int output_volume() const { return voice_.output_volume_; }
1007 const VideoEncoderConfig& default_video_encoder_config() const {
1008 return video_.default_encoder_config_;
1009 }
1010 const std::string& audio_in_device() const { return voice_.in_device_; }
1011 const std::string& audio_out_device() const { return voice_.out_device_; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001012 int voice_loglevel() const { return voice_.loglevel_; }
1013 const std::string& voice_logfilter() const { return voice_.logfilter_; }
1014 int video_loglevel() const { return video_.loglevel_; }
1015 const std::string& video_logfilter() const { return video_.logfilter_; }
1016 bool capture() const { return video_.capture_; }
1017 bool options_changed() const {
1018 return voice_.options_changed_ || video_.options_changed_;
1019 }
1020 void clear_options_changed() {
1021 video_.options_changed_ = false;
1022 voice_.options_changed_ = false;
1023 }
1024 void set_fail_create_channel(bool fail) {
1025 voice_.set_fail_create_channel(fail);
1026 video_.set_fail_create_channel(fail);
1027 }
1028 bool voice_processor_registered(MediaProcessorDirection direction) const {
1029 if (direction == MPD_RX) {
1030 return voice_.rx_processor_ != NULL;
1031 } else if (direction == MPD_TX) {
1032 return voice_.tx_processor_ != NULL;
1033 }
1034 return false;
1035 }
1036};
1037
1038// CompositeMediaEngine with FakeVoiceEngine to expose SetAudioCodecs to
1039// establish a media connectionwith minimum set of audio codes required
1040template <class VIDEO>
1041class CompositeMediaEngineWithFakeVoiceEngine :
1042 public CompositeMediaEngine<FakeVoiceEngine, VIDEO> {
1043 public:
1044 CompositeMediaEngineWithFakeVoiceEngine() {}
1045 virtual ~CompositeMediaEngineWithFakeVoiceEngine() {}
1046
1047 virtual void SetAudioCodecs(const std::vector<AudioCodec>& codecs) {
1048 CompositeMediaEngine<FakeVoiceEngine, VIDEO>::voice_.SetCodecs(codecs);
1049 }
1050};
1051
1052// Have to come afterwards due to declaration order
1053inline FakeVoiceMediaChannel::~FakeVoiceMediaChannel() {
1054 if (engine_) {
1055 engine_->UnregisterChannel(this);
1056 }
1057}
1058
1059inline FakeVideoMediaChannel::~FakeVideoMediaChannel() {
1060 if (engine_) {
1061 engine_->UnregisterChannel(this);
1062 }
1063}
1064
1065class FakeDataEngine : public DataEngineInterface {
1066 public:
1067 FakeDataEngine() : last_channel_type_(DCT_NONE) {}
1068
1069 virtual DataMediaChannel* CreateChannel(DataChannelType data_channel_type) {
1070 last_channel_type_ = data_channel_type;
Fredrik Solenbergb071a192015-09-17 16:42:56 +02001071 FakeDataMediaChannel* ch = new FakeDataMediaChannel(this, DataOptions());
henrike@webrtc.org28e20752013-07-10 00:45:36 +00001072 channels_.push_back(ch);
1073 return ch;
1074 }
1075
1076 FakeDataMediaChannel* GetChannel(size_t index) {
1077 return (channels_.size() > index) ? channels_[index] : NULL;
1078 }
1079
1080 void UnregisterChannel(DataMediaChannel* channel) {
1081 channels_.erase(std::find(channels_.begin(), channels_.end(), channel));
1082 }
1083
1084 virtual void SetDataCodecs(const std::vector<DataCodec>& data_codecs) {
1085 data_codecs_ = data_codecs;
1086 }
1087
1088 virtual const std::vector<DataCodec>& data_codecs() { return data_codecs_; }
1089
1090 DataChannelType last_channel_type() const { return last_channel_type_; }
1091
1092 private:
1093 std::vector<FakeDataMediaChannel*> channels_;
1094 std::vector<DataCodec> data_codecs_;
1095 DataChannelType last_channel_type_;
1096};
1097
1098} // namespace cricket
1099
1100#endif // TALK_MEDIA_BASE_FAKEMEDIAENGINE_H_