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henrike@webrtc.org28e20752013-07-10 00:45:361/*
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
buildbot@webrtc.orgd4e598d2014-07-29 17:36:5237#include "webrtc/base/buffer.h"
38#include "webrtc/base/stringutils.h"
henrike@webrtc.org1e09a712013-07-26 19:17:5939#include "talk/media/base/audiorenderer.h"
henrike@webrtc.org28e20752013-07-10 00:45:3640#include "talk/media/base/mediaengine.h"
41#include "talk/media/base/rtputils.h"
42#include "talk/media/base/streamparams.h"
43#include "talk/p2p/base/sessiondescription.h"
44
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 19:17:5973 if (!sending_) {
henrike@webrtc.org28e20752013-07-10 00:45:3674 return false;
75 }
buildbot@webrtc.orgd4e598d2014-07-29 17:36:5276 rtc::Buffer packet(data, len, kMaxRtpPacketLen);
henrike@webrtc.org1e09a712013-07-26 19:17:5977 return Base::SendPacket(&packet);
henrike@webrtc.org28e20752013-07-10 00:45:3678 }
79 bool SendRtcp(const void* data, int len) {
buildbot@webrtc.orgd4e598d2014-07-29 17:36:5280 rtc::Buffer packet(data, len, kMaxRtpPacketLen);
henrike@webrtc.org1e09a712013-07-26 19:17:5981 return Base::SendRtcp(&packet);
henrike@webrtc.org28e20752013-07-10 00:45:3682 }
83
84 bool CheckRtp(const void* data, int len) {
85 bool success = !rtp_packets_.empty();
86 if (success) {
87 std::string packet = rtp_packets_.front();
88 rtp_packets_.pop_front();
89 success = (packet == std::string(static_cast<const char*>(data), len));
90 }
91 return success;
92 }
93 bool CheckRtcp(const void* data, int len) {
94 bool success = !rtcp_packets_.empty();
95 if (success) {
96 std::string packet = rtcp_packets_.front();
97 rtcp_packets_.pop_front();
98 success = (packet == std::string(static_cast<const char*>(data), len));
99 }
100 return success;
101 }
102 bool CheckNoRtp() { return rtp_packets_.empty(); }
103 bool CheckNoRtcp() { return rtcp_packets_.empty(); }
104 virtual bool SetRecvRtpHeaderExtensions(
105 const std::vector<RtpHeaderExtension>& extensions) {
106 recv_extensions_ = extensions;
107 return true;
108 }
109 virtual bool SetSendRtpHeaderExtensions(
110 const std::vector<RtpHeaderExtension>& extensions) {
111 send_extensions_ = extensions;
112 return true;
113 }
114 void set_fail_set_send_codecs(bool fail) { fail_set_send_codecs_ = fail; }
115 void set_fail_set_recv_codecs(bool fail) { fail_set_recv_codecs_ = fail; }
116 virtual bool AddSendStream(const StreamParams& sp) {
117 if (std::find(send_streams_.begin(), send_streams_.end(), sp) !=
118 send_streams_.end()) {
119 return false;
120 }
121 send_streams_.push_back(sp);
122 return true;
123 }
124 virtual bool RemoveSendStream(uint32 ssrc) {
125 return RemoveStreamBySsrc(&send_streams_, ssrc);
126 }
127 virtual bool AddRecvStream(const StreamParams& sp) {
128 if (std::find(receive_streams_.begin(), receive_streams_.end(), sp) !=
129 receive_streams_.end()) {
130 return false;
131 }
132 receive_streams_.push_back(sp);
133 return true;
134 }
135 virtual bool RemoveRecvStream(uint32 ssrc) {
136 return RemoveStreamBySsrc(&receive_streams_, ssrc);
137 }
138 virtual bool MuteStream(uint32 ssrc, bool on) {
139 if (!HasSendStream(ssrc) && ssrc != 0)
140 return false;
141 if (on)
142 muted_streams_.insert(ssrc);
143 else
144 muted_streams_.erase(ssrc);
145 return true;
146 }
147 bool IsStreamMuted(uint32 ssrc) const {
148 bool ret = muted_streams_.find(ssrc) != muted_streams_.end();
149 // If |ssrc = 0| check if the first send stream is muted.
150 if (!ret && ssrc == 0 && !send_streams_.empty()) {
151 return muted_streams_.find(send_streams_[0].first_ssrc()) !=
152 muted_streams_.end();
153 }
154 return ret;
155 }
156 const std::vector<StreamParams>& send_streams() const {
157 return send_streams_;
158 }
159 const std::vector<StreamParams>& recv_streams() const {
160 return receive_streams_;
161 }
162 bool HasRecvStream(uint32 ssrc) const {
163 return GetStreamBySsrc(receive_streams_, ssrc, NULL);
164 }
165 bool HasSendStream(uint32 ssrc) const {
166 return GetStreamBySsrc(send_streams_, ssrc, NULL);
167 }
168 // TODO(perkj): This is to support legacy unit test that only check one
169 // sending stream.
170 const uint32 send_ssrc() {
171 if (send_streams_.empty())
172 return 0;
173 return send_streams_[0].first_ssrc();
174 }
175
176 // TODO(perkj): This is to support legacy unit test that only check one
177 // sending stream.
178 const std::string rtcp_cname() {
179 if (send_streams_.empty())
180 return "";
181 return send_streams_[0].cname;
182 }
183
184 bool ready_to_send() const {
185 return ready_to_send_;
186 }
187
188 protected:
189 bool set_sending(bool send) {
190 sending_ = send;
191 return true;
192 }
193 void set_playout(bool playout) { playout_ = playout; }
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52194 virtual void OnPacketReceived(rtc::Buffer* packet,
195 const rtc::PacketTime& packet_time) {
henrike@webrtc.org28e20752013-07-10 00:45:36196 rtp_packets_.push_back(std::string(packet->data(), packet->length()));
197 }
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52198 virtual void OnRtcpReceived(rtc::Buffer* packet,
199 const rtc::PacketTime& packet_time) {
henrike@webrtc.org28e20752013-07-10 00:45:36200 rtcp_packets_.push_back(std::string(packet->data(), packet->length()));
201 }
202 virtual void OnReadyToSend(bool ready) {
203 ready_to_send_ = ready;
204 }
205 bool fail_set_send_codecs() const { return fail_set_send_codecs_; }
206 bool fail_set_recv_codecs() const { return fail_set_recv_codecs_; }
207
208 private:
209 bool sending_;
210 bool playout_;
211 std::vector<RtpHeaderExtension> recv_extensions_;
212 std::vector<RtpHeaderExtension> send_extensions_;
213 std::list<std::string> rtp_packets_;
214 std::list<std::string> rtcp_packets_;
215 std::vector<StreamParams> send_streams_;
216 std::vector<StreamParams> receive_streams_;
217 std::set<uint32> muted_streams_;
218 bool fail_set_send_codecs_;
219 bool fail_set_recv_codecs_;
220 uint32 send_ssrc_;
221 std::string rtcp_cname_;
222 bool ready_to_send_;
223};
224
225class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
226 public:
227 struct DtmfInfo {
228 DtmfInfo(uint32 ssrc, int event_code, int duration, int flags)
229 : ssrc(ssrc), event_code(event_code), duration(duration), flags(flags) {
230 }
231 uint32 ssrc;
232 int event_code;
233 int duration;
234 int flags;
235 };
236 explicit FakeVoiceMediaChannel(FakeVoiceEngine* engine)
237 : engine_(engine),
238 fail_set_send_(false),
239 ringback_tone_ssrc_(0),
240 ringback_tone_play_(false),
241 ringback_tone_loop_(false),
242 time_since_last_typing_(-1) {
243 output_scalings_[0] = OutputScaling(); // For default channel.
244 }
245 ~FakeVoiceMediaChannel();
246 const std::vector<AudioCodec>& recv_codecs() const { return recv_codecs_; }
247 const std::vector<AudioCodec>& send_codecs() const { return send_codecs_; }
248 const std::vector<AudioCodec>& codecs() const { return send_codecs(); }
249 const std::vector<DtmfInfo>& dtmf_info_queue() const {
250 return dtmf_info_queue_;
251 }
252 const AudioOptions& options() const { return options_; }
253
254 uint32 ringback_tone_ssrc() const { return ringback_tone_ssrc_; }
255 bool ringback_tone_play() const { return ringback_tone_play_; }
256 bool ringback_tone_loop() const { return ringback_tone_loop_; }
257
258 virtual bool SetRecvCodecs(const std::vector<AudioCodec>& codecs) {
259 if (fail_set_recv_codecs()) {
260 // Fake the failure in SetRecvCodecs.
261 return false;
262 }
263 recv_codecs_ = codecs;
264 return true;
265 }
266 virtual bool SetSendCodecs(const std::vector<AudioCodec>& codecs) {
267 if (fail_set_send_codecs()) {
268 // Fake the failure in SetSendCodecs.
269 return false;
270 }
271 send_codecs_ = codecs;
272 return true;
273 }
274 virtual bool SetPlayout(bool playout) {
275 set_playout(playout);
276 return true;
277 }
278 virtual bool SetSend(SendFlags flag) {
279 if (fail_set_send_) {
280 return false;
281 }
282 return set_sending(flag != SEND_NOTHING);
283 }
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54284 virtual bool SetStartSendBandwidth(int bps) { return true; }
285 virtual bool SetMaxSendBandwidth(int bps) { return true; }
henrike@webrtc.org28e20752013-07-10 00:45:36286 virtual bool AddRecvStream(const StreamParams& sp) {
287 if (!RtpHelper<VoiceMediaChannel>::AddRecvStream(sp))
288 return false;
289 output_scalings_[sp.first_ssrc()] = OutputScaling();
290 return true;
291 }
292 virtual bool RemoveRecvStream(uint32 ssrc) {
293 if (!RtpHelper<VoiceMediaChannel>::RemoveRecvStream(ssrc))
294 return false;
295 output_scalings_.erase(ssrc);
296 return true;
297 }
henrike@webrtc.org1e09a712013-07-26 19:17:59298 virtual bool SetRemoteRenderer(uint32 ssrc, AudioRenderer* renderer) {
299 std::map<uint32, AudioRenderer*>::iterator it =
300 remote_renderers_.find(ssrc);
301 if (renderer) {
302 if (it != remote_renderers_.end()) {
303 ASSERT(it->second == renderer);
304 } else {
305 remote_renderers_.insert(std::make_pair(ssrc, renderer));
306 renderer->AddChannel(0);
307 }
308 } else {
309 if (it != remote_renderers_.end()) {
310 it->second->RemoveChannel(0);
311 remote_renderers_.erase(it);
312 } else {
313 return false;
314 }
315 }
316 return true;
317 }
318 virtual bool SetLocalRenderer(uint32 ssrc, AudioRenderer* renderer) {
henrike@webrtc.orga7b98182014-02-21 15:51:43319 std::map<uint32, VoiceChannelAudioSink*>::iterator it =
320 local_renderers_.find(ssrc);
henrike@webrtc.org1e09a712013-07-26 19:17:59321 if (renderer) {
322 if (it != local_renderers_.end()) {
henrike@webrtc.orga7b98182014-02-21 15:51:43323 ASSERT(it->second->renderer() == renderer);
henrike@webrtc.org1e09a712013-07-26 19:17:59324 } else {
henrike@webrtc.orga7b98182014-02-21 15:51:43325 local_renderers_.insert(std::make_pair(
326 ssrc, new VoiceChannelAudioSink(renderer)));
henrike@webrtc.org1e09a712013-07-26 19:17:59327 }
328 } else {
329 if (it != local_renderers_.end()) {
henrike@webrtc.orga7b98182014-02-21 15:51:43330 delete it->second;
henrike@webrtc.org1e09a712013-07-26 19:17:59331 local_renderers_.erase(it);
332 } else {
333 return false;
334 }
335 }
336 return true;
henrike@webrtc.org28e20752013-07-10 00:45:36337 }
338
339 virtual bool GetActiveStreams(AudioInfo::StreamList* streams) { return true; }
340 virtual int GetOutputLevel() { return 0; }
341 void set_time_since_last_typing(int ms) { time_since_last_typing_ = ms; }
342 virtual int GetTimeSinceLastTyping() { return time_since_last_typing_; }
343 virtual void SetTypingDetectionParameters(
344 int time_window, int cost_per_typing, int reporting_threshold,
345 int penalty_decay, int type_event_delay) {}
346
347 virtual bool SetRingbackTone(const char* buf, int len) { return true; }
348 virtual bool PlayRingbackTone(uint32 ssrc, bool play, bool loop) {
349 ringback_tone_ssrc_ = ssrc;
350 ringback_tone_play_ = play;
351 ringback_tone_loop_ = loop;
352 return true;
353 }
354
355 virtual bool CanInsertDtmf() {
356 for (std::vector<AudioCodec>::const_iterator it = send_codecs_.begin();
357 it != send_codecs_.end(); ++it) {
358 // Find the DTMF telephone event "codec".
359 if (_stricmp(it->name.c_str(), "telephone-event") == 0) {
360 return true;
361 }
362 }
363 return false;
364 }
365 virtual bool InsertDtmf(uint32 ssrc, int event_code, int duration,
366 int flags) {
367 dtmf_info_queue_.push_back(DtmfInfo(ssrc, event_code, duration, flags));
368 return true;
369 }
370
371 virtual bool SetOutputScaling(uint32 ssrc, double left, double right) {
372 if (0 == ssrc) {
373 std::map<uint32, OutputScaling>::iterator it;
374 for (it = output_scalings_.begin(); it != output_scalings_.end(); ++it) {
375 it->second.left = left;
376 it->second.right = right;
377 }
378 return true;
379 } else if (output_scalings_.find(ssrc) != output_scalings_.end()) {
380 output_scalings_[ssrc].left = left;
381 output_scalings_[ssrc].right = right;
382 return true;
383 }
384 return false;
385 }
386 virtual bool GetOutputScaling(uint32 ssrc, double* left, double* right) {
387 if (output_scalings_.find(ssrc) == output_scalings_.end())
388 return false;
389 *left = output_scalings_[ssrc].left;
390 *right = output_scalings_[ssrc].right;
391 return true;
392 }
393
394 virtual bool GetStats(VoiceMediaInfo* info) { return false; }
395 virtual void GetLastMediaError(uint32* ssrc,
396 VoiceMediaChannel::Error* error) {
397 *ssrc = 0;
398 *error = fail_set_send_ ? VoiceMediaChannel::ERROR_REC_DEVICE_OPEN_FAILED
399 : VoiceMediaChannel::ERROR_NONE;
400 }
401
402 void set_fail_set_send(bool fail) { fail_set_send_ = fail; }
403 void TriggerError(uint32 ssrc, VoiceMediaChannel::Error error) {
404 VoiceMediaChannel::SignalMediaError(ssrc, error);
405 }
406
407 virtual bool SetOptions(const AudioOptions& options) {
408 // Does a "merge" of current options and set options.
409 options_.SetAll(options);
410 return true;
411 }
412 virtual bool GetOptions(AudioOptions* options) const {
413 *options = options_;
414 return true;
415 }
416
417 private:
418 struct OutputScaling {
419 OutputScaling() : left(1.0), right(1.0) {}
420 double left, right;
421 };
422
henrike@webrtc.orga7b98182014-02-21 15:51:43423 class VoiceChannelAudioSink : public AudioRenderer::Sink {
424 public:
425 explicit VoiceChannelAudioSink(AudioRenderer* renderer)
426 : renderer_(renderer) {
427 renderer_->AddChannel(0);
428 renderer_->SetSink(this);
429 }
430 virtual ~VoiceChannelAudioSink() {
431 if (renderer_) {
432 renderer_->RemoveChannel(0);
433 renderer_->SetSink(NULL);
434 }
435 }
436 virtual void OnData(const void* audio_data,
437 int bits_per_sample,
438 int sample_rate,
439 int number_of_channels,
440 int number_of_frames) OVERRIDE {}
441 virtual void OnClose() OVERRIDE {
442 renderer_ = NULL;
443 }
444 AudioRenderer* renderer() const { return renderer_; }
445
446 private:
447 AudioRenderer* renderer_;
448 };
449
450
henrike@webrtc.org28e20752013-07-10 00:45:36451 FakeVoiceEngine* engine_;
452 std::vector<AudioCodec> recv_codecs_;
453 std::vector<AudioCodec> send_codecs_;
454 std::map<uint32, OutputScaling> output_scalings_;
455 std::vector<DtmfInfo> dtmf_info_queue_;
456 bool fail_set_send_;
457 uint32 ringback_tone_ssrc_;
458 bool ringback_tone_play_;
459 bool ringback_tone_loop_;
460 int time_since_last_typing_;
461 AudioOptions options_;
henrike@webrtc.orga7b98182014-02-21 15:51:43462 std::map<uint32, VoiceChannelAudioSink*> local_renderers_;
henrike@webrtc.org1e09a712013-07-26 19:17:59463 std::map<uint32, AudioRenderer*> remote_renderers_;
henrike@webrtc.org28e20752013-07-10 00:45:36464};
465
466// A helper function to compare the FakeVoiceMediaChannel::DtmfInfo.
467inline bool CompareDtmfInfo(const FakeVoiceMediaChannel::DtmfInfo& info,
468 uint32 ssrc, int event_code, int duration,
469 int flags) {
470 return (info.duration == duration && info.event_code == event_code &&
471 info.flags == flags && info.ssrc == ssrc);
472}
473
474class FakeVideoMediaChannel : public RtpHelper<VideoMediaChannel> {
475 public:
476 explicit FakeVideoMediaChannel(FakeVideoEngine* engine)
477 : engine_(engine),
478 sent_intra_frame_(false),
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54479 requested_intra_frame_(false),
480 start_bps_(-1),
481 max_bps_(-1) {}
henrike@webrtc.org28e20752013-07-10 00:45:36482 ~FakeVideoMediaChannel();
483
484 const std::vector<VideoCodec>& recv_codecs() const { return recv_codecs_; }
485 const std::vector<VideoCodec>& send_codecs() const { return send_codecs_; }
486 const std::vector<VideoCodec>& codecs() const { return send_codecs(); }
487 bool rendering() const { return playout(); }
488 const VideoOptions& options() const { return options_; }
489 const std::map<uint32, VideoRenderer*>& renderers() const {
490 return renderers_;
491 }
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54492 int start_bps() const { return start_bps_; }
493 int max_bps() const { return max_bps_; }
henrike@webrtc.org28e20752013-07-10 00:45:36494 bool GetSendStreamFormat(uint32 ssrc, VideoFormat* format) {
495 if (send_formats_.find(ssrc) == send_formats_.end()) {
496 return false;
497 }
498 *format = send_formats_[ssrc];
499 return true;
500 }
501 virtual bool SetSendStreamFormat(uint32 ssrc, const VideoFormat& format) {
502 if (send_formats_.find(ssrc) == send_formats_.end()) {
503 return false;
504 }
505 send_formats_[ssrc] = format;
506 return true;
507 }
508
509 virtual bool AddSendStream(const StreamParams& sp) {
510 if (!RtpHelper<VideoMediaChannel>::AddSendStream(sp)) {
511 return false;
512 }
513 SetSendStreamDefaultFormat(sp.first_ssrc());
514 return true;
515 }
516 virtual bool RemoveSendStream(uint32 ssrc) {
517 send_formats_.erase(ssrc);
518 return RtpHelper<VideoMediaChannel>::RemoveSendStream(ssrc);
519 }
520
521 virtual bool SetRecvCodecs(const std::vector<VideoCodec>& codecs) {
522 if (fail_set_recv_codecs()) {
523 // Fake the failure in SetRecvCodecs.
524 return false;
525 }
526 recv_codecs_ = codecs;
527 return true;
528 }
529 virtual bool SetSendCodecs(const std::vector<VideoCodec>& codecs) {
530 if (fail_set_send_codecs()) {
531 // Fake the failure in SetSendCodecs.
532 return false;
533 }
534 send_codecs_ = codecs;
535
536 for (std::vector<StreamParams>::const_iterator it = send_streams().begin();
537 it != send_streams().end(); ++it) {
538 SetSendStreamDefaultFormat(it->first_ssrc());
539 }
540 return true;
541 }
542 virtual bool GetSendCodec(VideoCodec* send_codec) {
543 if (send_codecs_.empty()) {
544 return false;
545 }
546 *send_codec = send_codecs_[0];
547 return true;
548 }
549 virtual bool SetRender(bool render) {
550 set_playout(render);
551 return true;
552 }
553 virtual bool SetRenderer(uint32 ssrc, VideoRenderer* r) {
554 if (ssrc != 0 && renderers_.find(ssrc) == renderers_.end()) {
555 return false;
556 }
557 if (ssrc != 0) {
558 renderers_[ssrc] = r;
559 }
560 return true;
561 }
562
563 virtual bool SetSend(bool send) { return set_sending(send); }
564 virtual bool SetCapturer(uint32 ssrc, VideoCapturer* capturer) {
565 capturers_[ssrc] = capturer;
566 return true;
567 }
568 bool HasCapturer(uint32 ssrc) const {
569 return capturers_.find(ssrc) != capturers_.end();
570 }
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54571 virtual bool SetStartSendBandwidth(int bps) {
572 start_bps_ = bps;
573 return true;
574 }
575 virtual bool SetMaxSendBandwidth(int bps) {
576 max_bps_ = bps;
577 return true;
578 }
henrike@webrtc.org28e20752013-07-10 00:45:36579 virtual bool AddRecvStream(const StreamParams& sp) {
580 if (!RtpHelper<VideoMediaChannel>::AddRecvStream(sp))
581 return false;
582 renderers_[sp.first_ssrc()] = NULL;
583 return true;
584 }
585 virtual bool RemoveRecvStream(uint32 ssrc) {
586 if (!RtpHelper<VideoMediaChannel>::RemoveRecvStream(ssrc))
587 return false;
588 renderers_.erase(ssrc);
589 return true;
590 }
591
wu@webrtc.orgb9a088b2014-02-13 23:18:49592 virtual bool GetStats(const StatsOptions& options,
593 VideoMediaInfo* info) { return false; }
henrike@webrtc.org28e20752013-07-10 00:45:36594 virtual bool SendIntraFrame() {
595 sent_intra_frame_ = true;
596 return true;
597 }
598 virtual bool RequestIntraFrame() {
599 requested_intra_frame_ = true;
600 return true;
601 }
602 virtual bool SetOptions(const VideoOptions& options) {
603 options_ = options;
604 return true;
605 }
606 virtual bool GetOptions(VideoOptions* options) const {
607 *options = options_;
608 return true;
609 }
610 virtual void UpdateAspectRatio(int ratio_w, int ratio_h) {}
611 void set_sent_intra_frame(bool v) { sent_intra_frame_ = v; }
612 bool sent_intra_frame() const { return sent_intra_frame_; }
613 void set_requested_intra_frame(bool v) { requested_intra_frame_ = v; }
614 bool requested_intra_frame() const { return requested_intra_frame_; }
615
616 private:
617 // Be default, each send stream uses the first send codec format.
618 void SetSendStreamDefaultFormat(uint32 ssrc) {
619 if (!send_codecs_.empty()) {
620 send_formats_[ssrc] = VideoFormat(
621 send_codecs_[0].width, send_codecs_[0].height,
622 cricket::VideoFormat::FpsToInterval(send_codecs_[0].framerate),
623 cricket::FOURCC_I420);
624 }
625 }
626
627 FakeVideoEngine* engine_;
628 std::vector<VideoCodec> recv_codecs_;
629 std::vector<VideoCodec> send_codecs_;
630 std::map<uint32, VideoRenderer*> renderers_;
631 std::map<uint32, VideoFormat> send_formats_;
632 std::map<uint32, VideoCapturer*> capturers_;
633 bool sent_intra_frame_;
634 bool requested_intra_frame_;
635 VideoOptions options_;
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54636 int start_bps_;
637 int max_bps_;
henrike@webrtc.org28e20752013-07-10 00:45:36638};
639
640class FakeSoundclipMedia : public SoundclipMedia {
641 public:
642 virtual bool PlaySound(const char* buf, int len, int flags) { return true; }
643};
644
645class FakeDataMediaChannel : public RtpHelper<DataMediaChannel> {
646 public:
647 explicit FakeDataMediaChannel(void* unused)
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54648 : send_blocked_(false), max_bps_(-1) {}
henrike@webrtc.org28e20752013-07-10 00:45:36649 ~FakeDataMediaChannel() {}
650 const std::vector<DataCodec>& recv_codecs() const { return recv_codecs_; }
651 const std::vector<DataCodec>& send_codecs() const { return send_codecs_; }
652 const std::vector<DataCodec>& codecs() const { return send_codecs(); }
henrike@webrtc.org28e20752013-07-10 00:45:36653 int max_bps() const { return max_bps_; }
654
655 virtual bool SetRecvCodecs(const std::vector<DataCodec>& codecs) {
656 if (fail_set_recv_codecs()) {
657 // Fake the failure in SetRecvCodecs.
658 return false;
659 }
660 recv_codecs_ = codecs;
661 return true;
662 }
663 virtual bool SetSendCodecs(const std::vector<DataCodec>& codecs) {
664 if (fail_set_send_codecs()) {
665 // Fake the failure in SetSendCodecs.
666 return false;
667 }
668 send_codecs_ = codecs;
669 return true;
670 }
671 virtual bool SetSend(bool send) { return set_sending(send); }
672 virtual bool SetReceive(bool receive) {
673 set_playout(receive);
674 return true;
675 }
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54676 virtual bool SetStartSendBandwidth(int bps) { return true; }
677 virtual bool SetMaxSendBandwidth(int bps) {
henrike@webrtc.org28e20752013-07-10 00:45:36678 max_bps_ = bps;
679 return true;
680 }
681 virtual bool AddRecvStream(const StreamParams& sp) {
682 if (!RtpHelper<DataMediaChannel>::AddRecvStream(sp))
683 return false;
684 return true;
685 }
686 virtual bool RemoveRecvStream(uint32 ssrc) {
687 if (!RtpHelper<DataMediaChannel>::RemoveRecvStream(ssrc))
688 return false;
689 return true;
690 }
691
692 virtual bool SendData(const SendDataParams& params,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52693 const rtc::Buffer& payload,
henrike@webrtc.org28e20752013-07-10 00:45:36694 SendDataResult* result) {
wu@webrtc.orgd64719d2013-08-01 00:00:07695 if (send_blocked_) {
696 *result = SDR_BLOCK;
697 return false;
698 } else {
699 last_sent_data_params_ = params;
700 last_sent_data_ = std::string(payload.data(), payload.length());
701 return true;
702 }
henrike@webrtc.org28e20752013-07-10 00:45:36703 }
704
705 SendDataParams last_sent_data_params() { return last_sent_data_params_; }
706 std::string last_sent_data() { return last_sent_data_; }
wu@webrtc.orgd64719d2013-08-01 00:00:07707 bool is_send_blocked() { return send_blocked_; }
708 void set_send_blocked(bool blocked) { send_blocked_ = blocked; }
henrike@webrtc.org28e20752013-07-10 00:45:36709
710 private:
711 std::vector<DataCodec> recv_codecs_;
712 std::vector<DataCodec> send_codecs_;
713 SendDataParams last_sent_data_params_;
714 std::string last_sent_data_;
wu@webrtc.orgd64719d2013-08-01 00:00:07715 bool send_blocked_;
henrike@webrtc.org28e20752013-07-10 00:45:36716 int max_bps_;
717};
718
719// A base class for all of the shared parts between FakeVoiceEngine
720// and FakeVideoEngine.
721class FakeBaseEngine {
722 public:
723 FakeBaseEngine()
724 : loglevel_(-1),
henrike@webrtc.org28e20752013-07-10 00:45:36725 options_changed_(false),
726 fail_create_channel_(false) {}
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52727 bool Init(rtc::Thread* worker_thread) { return true; }
henrike@webrtc.org28e20752013-07-10 00:45:36728 void Terminate() {}
729
henrike@webrtc.org28e20752013-07-10 00:45:36730 void SetLogging(int level, const char* filter) {
731 loglevel_ = level;
732 logfilter_ = filter;
733 }
734
735 void set_fail_create_channel(bool fail) { fail_create_channel_ = fail; }
736
737 const std::vector<RtpHeaderExtension>& rtp_header_extensions() const {
738 return rtp_header_extensions_;
739 }
henrike@webrtc.org79047f92014-03-06 23:46:59740 void set_rtp_header_extensions(
741 const std::vector<RtpHeaderExtension>& extensions) {
742 rtp_header_extensions_ = extensions;
743 }
henrike@webrtc.org28e20752013-07-10 00:45:36744
745 protected:
746 int loglevel_;
747 std::string logfilter_;
henrike@webrtc.org28e20752013-07-10 00:45:36748 // Flag used by optionsmessagehandler_unittest for checking whether any
749 // relevant setting has been updated.
750 // TODO(thaloun): Replace with explicit checks of before & after values.
751 bool options_changed_;
752 bool fail_create_channel_;
753 std::vector<RtpHeaderExtension> rtp_header_extensions_;
754};
755
756class FakeVoiceEngine : public FakeBaseEngine {
757 public:
758 FakeVoiceEngine()
759 : output_volume_(-1),
760 delay_offset_(0),
761 rx_processor_(NULL),
762 tx_processor_(NULL) {
763 // Add a fake audio codec. Note that the name must not be "" as there are
764 // sanity checks against that.
765 codecs_.push_back(AudioCodec(101, "fake_audio_codec", 0, 0, 1, 0));
766 }
767 int GetCapabilities() { return AUDIO_SEND | AUDIO_RECV; }
mallinath@webrtc.orga27be8e2013-09-27 23:04:10768 AudioOptions GetAudioOptions() const {
769 return options_;
770 }
771 AudioOptions GetOptions() const {
772 return options_;
773 }
774 bool SetOptions(const AudioOptions& options) {
775 options_ = options;
776 options_changed_ = true;
777 return true;
778 }
henrike@webrtc.org28e20752013-07-10 00:45:36779
780 VoiceMediaChannel* CreateChannel() {
781 if (fail_create_channel_) {
782 return NULL;
783 }
784
785 FakeVoiceMediaChannel* ch = new FakeVoiceMediaChannel(this);
786 channels_.push_back(ch);
787 return ch;
788 }
789 FakeVoiceMediaChannel* GetChannel(size_t index) {
790 return (channels_.size() > index) ? channels_[index] : NULL;
791 }
792 void UnregisterChannel(VoiceMediaChannel* channel) {
793 channels_.erase(std::find(channels_.begin(), channels_.end(), channel));
794 }
795 SoundclipMedia* CreateSoundclip() { return new FakeSoundclipMedia(); }
796
797 const std::vector<AudioCodec>& codecs() { return codecs_; }
798 void SetCodecs(const std::vector<AudioCodec> codecs) { codecs_ = codecs; }
799
800 bool SetDelayOffset(int offset) {
801 delay_offset_ = offset;
802 return true;
803 }
804
805 bool SetDevices(const Device* in_device, const Device* out_device) {
806 in_device_ = (in_device) ? in_device->name : "";
807 out_device_ = (out_device) ? out_device->name : "";
808 options_changed_ = true;
809 return true;
810 }
811
812 bool GetOutputVolume(int* level) {
813 *level = output_volume_;
814 return true;
815 }
816
817 bool SetOutputVolume(int level) {
818 output_volume_ = level;
819 options_changed_ = true;
820 return true;
821 }
822
823 int GetInputLevel() { return 0; }
824
825 bool SetLocalMonitor(bool enable) { return true; }
826
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52827 bool StartAecDump(rtc::PlatformFile file) { return false; }
wu@webrtc.orga9890802013-12-13 00:21:03828
henrike@webrtc.org28e20752013-07-10 00:45:36829 bool RegisterProcessor(uint32 ssrc, VoiceProcessor* voice_processor,
830 MediaProcessorDirection direction) {
831 if (direction == MPD_RX) {
832 rx_processor_ = voice_processor;
833 return true;
834 } else if (direction == MPD_TX) {
835 tx_processor_ = voice_processor;
836 return true;
837 }
838 return false;
839 }
840
841 bool UnregisterProcessor(uint32 ssrc, VoiceProcessor* voice_processor,
842 MediaProcessorDirection direction) {
843 bool unregistered = false;
844 if (direction & MPD_RX) {
845 rx_processor_ = NULL;
846 unregistered = true;
847 }
848 if (direction & MPD_TX) {
849 tx_processor_ = NULL;
850 unregistered = true;
851 }
852 return unregistered;
853 }
854
855 private:
856 std::vector<FakeVoiceMediaChannel*> channels_;
857 std::vector<AudioCodec> codecs_;
858 int output_volume_;
859 int delay_offset_;
860 std::string in_device_;
861 std::string out_device_;
862 VoiceProcessor* rx_processor_;
863 VoiceProcessor* tx_processor_;
mallinath@webrtc.orga27be8e2013-09-27 23:04:10864 AudioOptions options_;
henrike@webrtc.org28e20752013-07-10 00:45:36865
866 friend class FakeMediaEngine;
867};
868
869class FakeVideoEngine : public FakeBaseEngine {
870 public:
871 FakeVideoEngine() : renderer_(NULL), capture_(false), processor_(NULL) {
872 // Add a fake video codec. Note that the name must not be "" as there are
873 // sanity checks against that.
874 codecs_.push_back(VideoCodec(0, "fake_video_codec", 0, 0, 0, 0));
875 }
mallinath@webrtc.orga27be8e2013-09-27 23:04:10876 bool GetOptions(VideoOptions* options) const {
877 *options = options_;
878 return true;
879 }
880 bool SetOptions(const VideoOptions& options) {
881 options_ = options;
882 options_changed_ = true;
883 return true;
884 }
henrike@webrtc.org28e20752013-07-10 00:45:36885 int GetCapabilities() { return VIDEO_SEND | VIDEO_RECV; }
886 bool SetDefaultEncoderConfig(const VideoEncoderConfig& config) {
887 default_encoder_config_ = config;
888 return true;
889 }
wu@webrtc.org78187522013-10-07 23:32:02890 VideoEncoderConfig GetDefaultEncoderConfig() const {
891 return default_encoder_config_;
892 }
henrike@webrtc.org28e20752013-07-10 00:45:36893 const VideoEncoderConfig& default_encoder_config() const {
894 return default_encoder_config_;
895 }
896
897 VideoMediaChannel* CreateChannel(VoiceMediaChannel* channel) {
898 if (fail_create_channel_) {
899 return NULL;
900 }
901
902 FakeVideoMediaChannel* ch = new FakeVideoMediaChannel(this);
903 channels_.push_back(ch);
904 return ch;
905 }
906 FakeVideoMediaChannel* GetChannel(size_t index) {
907 return (channels_.size() > index) ? channels_[index] : NULL;
908 }
909 void UnregisterChannel(VideoMediaChannel* channel) {
910 channels_.erase(std::find(channels_.begin(), channels_.end(), channel));
911 }
912
913 const std::vector<VideoCodec>& codecs() const { return codecs_; }
914 bool FindCodec(const VideoCodec& in) {
915 for (size_t i = 0; i < codecs_.size(); ++i) {
916 if (codecs_[i].Matches(in)) {
917 return true;
918 }
919 }
920 return false;
921 }
922 void SetCodecs(const std::vector<VideoCodec> codecs) { codecs_ = codecs; }
923
924 bool SetCaptureDevice(const Device* device) {
925 in_device_ = (device) ? device->name : "";
926 options_changed_ = true;
927 return true;
928 }
929 bool SetLocalRenderer(VideoRenderer* r) {
930 renderer_ = r;
931 return true;
932 }
henrike@webrtc.org28e20752013-07-10 00:45:36933 bool SetCapture(bool capture) {
934 capture_ = capture;
935 return true;
936 }
937 VideoFormat GetStartCaptureFormat() const {
938 return VideoFormat(640, 480, cricket::VideoFormat::FpsToInterval(30),
939 FOURCC_I420);
940 }
941
942 sigslot::repeater2<VideoCapturer*, CaptureState> SignalCaptureStateChange;
943
944 private:
945 std::vector<FakeVideoMediaChannel*> channels_;
946 std::vector<VideoCodec> codecs_;
947 VideoEncoderConfig default_encoder_config_;
948 std::string in_device_;
949 VideoRenderer* renderer_;
950 bool capture_;
951 VideoProcessor* processor_;
mallinath@webrtc.orga27be8e2013-09-27 23:04:10952 VideoOptions options_;
henrike@webrtc.org28e20752013-07-10 00:45:36953
954 friend class FakeMediaEngine;
955};
956
957class FakeMediaEngine :
958 public CompositeMediaEngine<FakeVoiceEngine, FakeVideoEngine> {
959 public:
960 FakeMediaEngine() {
961 voice_ = FakeVoiceEngine();
962 video_ = FakeVideoEngine();
963 }
964 virtual ~FakeMediaEngine() {}
965
henrike@webrtc.org79047f92014-03-06 23:46:59966 void SetAudioCodecs(const std::vector<AudioCodec>& codecs) {
henrike@webrtc.org28e20752013-07-10 00:45:36967 voice_.SetCodecs(codecs);
968 }
henrike@webrtc.org79047f92014-03-06 23:46:59969 void SetVideoCodecs(const std::vector<VideoCodec>& codecs) {
henrike@webrtc.org28e20752013-07-10 00:45:36970 video_.SetCodecs(codecs);
971 }
972
henrike@webrtc.org79047f92014-03-06 23:46:59973 void SetAudioRtpHeaderExtensions(
974 const std::vector<RtpHeaderExtension>& extensions) {
975 voice_.set_rtp_header_extensions(extensions);
976 }
977 void SetVideoRtpHeaderExtensions(
978 const std::vector<RtpHeaderExtension>& extensions) {
979 video_.set_rtp_header_extensions(extensions);
980 }
981
henrike@webrtc.org28e20752013-07-10 00:45:36982 FakeVoiceMediaChannel* GetVoiceChannel(size_t index) {
983 return voice_.GetChannel(index);
984 }
henrike@webrtc.org28e20752013-07-10 00:45:36985 FakeVideoMediaChannel* GetVideoChannel(size_t index) {
986 return video_.GetChannel(index);
987 }
988
mallinath@webrtc.orga27be8e2013-09-27 23:04:10989 AudioOptions audio_options() const { return voice_.options_; }
henrike@webrtc.org28e20752013-07-10 00:45:36990 int audio_delay_offset() const { return voice_.delay_offset_; }
991 int output_volume() const { return voice_.output_volume_; }
992 const VideoEncoderConfig& default_video_encoder_config() const {
993 return video_.default_encoder_config_;
994 }
995 const std::string& audio_in_device() const { return voice_.in_device_; }
996 const std::string& audio_out_device() const { return voice_.out_device_; }
997 VideoRenderer* local_renderer() { return video_.renderer_; }
998 int voice_loglevel() const { return voice_.loglevel_; }
999 const std::string& voice_logfilter() const { return voice_.logfilter_; }
1000 int video_loglevel() const { return video_.loglevel_; }
1001 const std::string& video_logfilter() const { return video_.logfilter_; }
1002 bool capture() const { return video_.capture_; }
1003 bool options_changed() const {
1004 return voice_.options_changed_ || video_.options_changed_;
1005 }
1006 void clear_options_changed() {
1007 video_.options_changed_ = false;
1008 voice_.options_changed_ = false;
1009 }
1010 void set_fail_create_channel(bool fail) {
1011 voice_.set_fail_create_channel(fail);
1012 video_.set_fail_create_channel(fail);
1013 }
1014 bool voice_processor_registered(MediaProcessorDirection direction) const {
1015 if (direction == MPD_RX) {
1016 return voice_.rx_processor_ != NULL;
1017 } else if (direction == MPD_TX) {
1018 return voice_.tx_processor_ != NULL;
1019 }
1020 return false;
1021 }
1022};
1023
1024// CompositeMediaEngine with FakeVoiceEngine to expose SetAudioCodecs to
1025// establish a media connectionwith minimum set of audio codes required
1026template <class VIDEO>
1027class CompositeMediaEngineWithFakeVoiceEngine :
1028 public CompositeMediaEngine<FakeVoiceEngine, VIDEO> {
1029 public:
1030 CompositeMediaEngineWithFakeVoiceEngine() {}
1031 virtual ~CompositeMediaEngineWithFakeVoiceEngine() {}
1032
1033 virtual void SetAudioCodecs(const std::vector<AudioCodec>& codecs) {
1034 CompositeMediaEngine<FakeVoiceEngine, VIDEO>::voice_.SetCodecs(codecs);
1035 }
1036};
1037
1038// Have to come afterwards due to declaration order
1039inline FakeVoiceMediaChannel::~FakeVoiceMediaChannel() {
1040 if (engine_) {
1041 engine_->UnregisterChannel(this);
1042 }
1043}
1044
1045inline FakeVideoMediaChannel::~FakeVideoMediaChannel() {
1046 if (engine_) {
1047 engine_->UnregisterChannel(this);
1048 }
1049}
1050
1051class FakeDataEngine : public DataEngineInterface {
1052 public:
1053 FakeDataEngine() : last_channel_type_(DCT_NONE) {}
1054
1055 virtual DataMediaChannel* CreateChannel(DataChannelType data_channel_type) {
1056 last_channel_type_ = data_channel_type;
1057 FakeDataMediaChannel* ch = new FakeDataMediaChannel(this);
1058 channels_.push_back(ch);
1059 return ch;
1060 }
1061
1062 FakeDataMediaChannel* GetChannel(size_t index) {
1063 return (channels_.size() > index) ? channels_[index] : NULL;
1064 }
1065
1066 void UnregisterChannel(DataMediaChannel* channel) {
1067 channels_.erase(std::find(channels_.begin(), channels_.end(), channel));
1068 }
1069
1070 virtual void SetDataCodecs(const std::vector<DataCodec>& data_codecs) {
1071 data_codecs_ = data_codecs;
1072 }
1073
1074 virtual const std::vector<DataCodec>& data_codecs() { return data_codecs_; }
1075
1076 DataChannelType last_channel_type() const { return last_channel_type_; }
1077
1078 private:
1079 std::vector<FakeDataMediaChannel*> channels_;
1080 std::vector<DataCodec> data_codecs_;
1081 DataChannelType last_channel_type_;
1082};
1083
1084} // namespace cricket
1085
1086#endif // TALK_MEDIA_BASE_FAKEMEDIAENGINE_H_