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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 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200116 virtual bool RemoveSendStream(uint32_t ssrc) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000117 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 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200127 virtual bool RemoveRecvStream(uint32_t ssrc) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000128 return RemoveStreamBySsrc(&receive_streams_, ssrc);
129 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200130 bool IsStreamMuted(uint32_t ssrc) const {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000131 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 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200145 bool HasRecvStream(uint32_t 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 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200148 bool HasSendStream(uint32_t 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.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200153 uint32_t 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:
Peter Boström0c4e06b2015-10-07 12:23:21 +0200172 bool MuteStream(uint32_t 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_;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200221 std::set<uint32_t> muted_streams_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000222 bool fail_set_send_codecs_;
223 bool fail_set_recv_codecs_;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200224 uint32_t send_ssrc_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000225 std::string rtcp_cname_;
226 bool ready_to_send_;
227};
228
229class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
230 public:
231 struct DtmfInfo {
Peter Boström0c4e06b2015-10-07 12:23:21 +0200232 DtmfInfo(uint32_t ssrc, int event_code, int duration, int flags)
233 : ssrc(ssrc),
234 event_code(event_code),
235 duration(duration),
236 flags(flags) {}
237 uint32_t ssrc;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000238 int event_code;
239 int duration;
240 int flags;
241 };
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200242 explicit FakeVoiceMediaChannel(FakeVoiceEngine* engine,
243 const AudioOptions& options)
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000244 : engine_(engine),
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000245 time_since_last_typing_(-1) {
solenberg4bac9c52015-10-09 02:32:53 -0700246 output_scalings_[0] = 1.0; // For default channel.
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200247 SetOptions(options);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000248 }
249 ~FakeVoiceMediaChannel();
250 const std::vector<AudioCodec>& recv_codecs() const { return recv_codecs_; }
251 const std::vector<AudioCodec>& send_codecs() const { return send_codecs_; }
252 const std::vector<AudioCodec>& codecs() const { return send_codecs(); }
253 const std::vector<DtmfInfo>& dtmf_info_queue() const {
254 return dtmf_info_queue_;
255 }
256 const AudioOptions& options() const { return options_; }
257
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200258 virtual bool SetSendParameters(const AudioSendParameters& params) {
259 return (SetSendCodecs(params.codecs) &&
260 SetSendRtpHeaderExtensions(params.extensions) &&
261 SetMaxSendBandwidth(params.max_bandwidth_bps) &&
262 SetOptions(params.options));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000263 }
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200264
265 virtual bool SetRecvParameters(const AudioRecvParameters& params) {
266 return (SetRecvCodecs(params.codecs) &&
267 SetRecvRtpHeaderExtensions(params.extensions));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000268 }
269 virtual bool SetPlayout(bool playout) {
270 set_playout(playout);
271 return true;
272 }
273 virtual bool SetSend(SendFlags flag) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000274 return set_sending(flag != SEND_NOTHING);
275 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200276 virtual bool SetAudioSend(uint32_t ssrc,
277 bool enable,
solenberg1dd98f32015-09-10 01:57:14 -0700278 const AudioOptions* options,
279 AudioRenderer* renderer) {
280 if (!SetLocalRenderer(ssrc, renderer)) {
281 return false;
282 }
solenbergdfc8f4f2015-10-01 02:31:10 -0700283 if (!RtpHelper<VoiceMediaChannel>::MuteStream(ssrc, !enable)) {
solenberg1dd98f32015-09-10 01:57:14 -0700284 return false;
285 }
solenbergdfc8f4f2015-10-01 02:31:10 -0700286 if (enable && options) {
solenberg1dd98f32015-09-10 01:57:14 -0700287 return SetOptions(*options);
288 }
289 return true;
290 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000291 virtual bool AddRecvStream(const StreamParams& sp) {
292 if (!RtpHelper<VoiceMediaChannel>::AddRecvStream(sp))
293 return false;
solenberg4bac9c52015-10-09 02:32:53 -0700294 output_scalings_[sp.first_ssrc()] = 1.0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000295 return true;
296 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200297 virtual bool RemoveRecvStream(uint32_t ssrc) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000298 if (!RtpHelper<VoiceMediaChannel>::RemoveRecvStream(ssrc))
299 return false;
300 output_scalings_.erase(ssrc);
301 return true;
302 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200303 virtual bool SetRemoteRenderer(uint32_t ssrc, AudioRenderer* renderer) {
304 std::map<uint32_t, AudioRenderer*>::iterator it =
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000305 remote_renderers_.find(ssrc);
306 if (renderer) {
307 if (it != remote_renderers_.end()) {
308 ASSERT(it->second == renderer);
309 } else {
310 remote_renderers_.insert(std::make_pair(ssrc, renderer));
torbjorngeefbc3b2015-10-08 13:10:36 -0700311 renderer->AddChannel(0);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000312 }
313 } else {
314 if (it != remote_renderers_.end()) {
torbjorngeefbc3b2015-10-08 13:10:36 -0700315 it->second->RemoveChannel(0);
henrike@webrtc.org1e09a712013-07-26 19:17:59 +0000316 remote_renderers_.erase(it);
317 } else {
318 return false;
319 }
320 }
321 return true;
322 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000323
324 virtual bool GetActiveStreams(AudioInfo::StreamList* streams) { return true; }
325 virtual int GetOutputLevel() { return 0; }
326 void set_time_since_last_typing(int ms) { time_since_last_typing_ = ms; }
327 virtual int GetTimeSinceLastTyping() { return time_since_last_typing_; }
328 virtual void SetTypingDetectionParameters(
329 int time_window, int cost_per_typing, int reporting_threshold,
330 int penalty_decay, int type_event_delay) {}
331
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000332 virtual bool CanInsertDtmf() {
333 for (std::vector<AudioCodec>::const_iterator it = send_codecs_.begin();
334 it != send_codecs_.end(); ++it) {
335 // Find the DTMF telephone event "codec".
336 if (_stricmp(it->name.c_str(), "telephone-event") == 0) {
337 return true;
338 }
339 }
340 return false;
341 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200342 virtual bool InsertDtmf(uint32_t ssrc,
343 int event_code,
344 int duration,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000345 int flags) {
346 dtmf_info_queue_.push_back(DtmfInfo(ssrc, event_code, duration, flags));
347 return true;
348 }
349
solenberg4bac9c52015-10-09 02:32:53 -0700350 virtual bool SetOutputVolume(uint32_t ssrc, double volume) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000351 if (0 == ssrc) {
solenberg4bac9c52015-10-09 02:32:53 -0700352 std::map<uint32_t, double>::iterator it;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000353 for (it = output_scalings_.begin(); it != output_scalings_.end(); ++it) {
solenberg4bac9c52015-10-09 02:32:53 -0700354 it->second = volume;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000355 }
356 return true;
357 } else if (output_scalings_.find(ssrc) != output_scalings_.end()) {
solenberg4bac9c52015-10-09 02:32:53 -0700358 output_scalings_[ssrc] = volume;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000359 return true;
360 }
361 return false;
362 }
solenberg4bac9c52015-10-09 02:32:53 -0700363 bool GetOutputVolume(uint32_t ssrc, double* volume) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000364 if (output_scalings_.find(ssrc) == output_scalings_.end())
365 return false;
solenberg4bac9c52015-10-09 02:32:53 -0700366 *volume = output_scalings_[ssrc];
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000367 return true;
368 }
369
370 virtual bool GetStats(VoiceMediaInfo* info) { return false; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000371
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000372 private:
henrike@webrtc.orga7b98182014-02-21 15:51:43 +0000373 class VoiceChannelAudioSink : public AudioRenderer::Sink {
374 public:
375 explicit VoiceChannelAudioSink(AudioRenderer* renderer)
376 : renderer_(renderer) {
torbjorngeefbc3b2015-10-08 13:10:36 -0700377 renderer_->AddChannel(0);
henrike@webrtc.orga7b98182014-02-21 15:51:43 +0000378 renderer_->SetSink(this);
379 }
380 virtual ~VoiceChannelAudioSink() {
381 if (renderer_) {
torbjorngeefbc3b2015-10-08 13:10:36 -0700382 renderer_->RemoveChannel(0);
henrike@webrtc.orga7b98182014-02-21 15:51:43 +0000383 renderer_->SetSink(NULL);
384 }
385 }
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000386 void OnData(const void* audio_data,
387 int bits_per_sample,
388 int sample_rate,
389 int number_of_channels,
Peter Kastingdce40cf2015-08-24 14:52:23 -0700390 size_t number_of_frames) override {}
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000391 void OnClose() override { renderer_ = NULL; }
henrike@webrtc.orga7b98182014-02-21 15:51:43 +0000392 AudioRenderer* renderer() const { return renderer_; }
393
394 private:
395 AudioRenderer* renderer_;
396 };
397
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200398 bool SetRecvCodecs(const std::vector<AudioCodec>& codecs) {
399 if (fail_set_recv_codecs()) {
400 // Fake the failure in SetRecvCodecs.
401 return false;
402 }
403 recv_codecs_ = codecs;
404 return true;
405 }
406 bool SetSendCodecs(const std::vector<AudioCodec>& codecs) {
407 if (fail_set_send_codecs()) {
408 // Fake the failure in SetSendCodecs.
409 return false;
410 }
411 send_codecs_ = codecs;
412 return true;
413 }
414 bool SetMaxSendBandwidth(int bps) { return true; }
415 bool SetOptions(const AudioOptions& options) {
416 // Does a "merge" of current options and set options.
417 options_.SetAll(options);
418 return true;
419 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200420 bool SetLocalRenderer(uint32_t ssrc, AudioRenderer* renderer) {
solenberg1dd98f32015-09-10 01:57:14 -0700421 auto it = local_renderers_.find(ssrc);
422 if (renderer) {
423 if (it != local_renderers_.end()) {
424 ASSERT(it->second->renderer() == renderer);
425 } else {
426 local_renderers_.insert(std::make_pair(
427 ssrc, new VoiceChannelAudioSink(renderer)));
428 }
429 } else {
430 if (it != local_renderers_.end()) {
431 delete it->second;
432 local_renderers_.erase(it);
433 }
434 }
435 return true;
436 }
henrike@webrtc.orga7b98182014-02-21 15:51:43 +0000437
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000438 FakeVoiceEngine* engine_;
439 std::vector<AudioCodec> recv_codecs_;
440 std::vector<AudioCodec> send_codecs_;
solenberg4bac9c52015-10-09 02:32:53 -0700441 std::map<uint32_t, double> output_scalings_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000442 std::vector<DtmfInfo> dtmf_info_queue_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000443 int time_since_last_typing_;
444 AudioOptions options_;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200445 std::map<uint32_t, VoiceChannelAudioSink*> local_renderers_;
446 std::map<uint32_t, AudioRenderer*> remote_renderers_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000447};
448
449// A helper function to compare the FakeVoiceMediaChannel::DtmfInfo.
450inline bool CompareDtmfInfo(const FakeVoiceMediaChannel::DtmfInfo& info,
Peter Boström0c4e06b2015-10-07 12:23:21 +0200451 uint32_t ssrc,
452 int event_code,
453 int duration,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000454 int flags) {
455 return (info.duration == duration && info.event_code == event_code &&
456 info.flags == flags && info.ssrc == ssrc);
457}
458
459class FakeVideoMediaChannel : public RtpHelper<VideoMediaChannel> {
460 public:
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200461 explicit FakeVideoMediaChannel(FakeVideoEngine* engine,
462 const VideoOptions& options)
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000463 : engine_(engine),
464 sent_intra_frame_(false),
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000465 requested_intra_frame_(false),
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200466 max_bps_(-1) {
467 SetOptions(options);
468 }
buildbot@webrtc.org1ecbe452014-10-14 20:29:28 +0000469
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000470 ~FakeVideoMediaChannel();
471
472 const std::vector<VideoCodec>& recv_codecs() const { return recv_codecs_; }
473 const std::vector<VideoCodec>& send_codecs() const { return send_codecs_; }
474 const std::vector<VideoCodec>& codecs() const { return send_codecs(); }
475 bool rendering() const { return playout(); }
476 const VideoOptions& options() const { return options_; }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200477 const std::map<uint32_t, VideoRenderer*>& renderers() const {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000478 return renderers_;
479 }
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000480 int max_bps() const { return max_bps_; }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200481 bool GetSendStreamFormat(uint32_t ssrc, VideoFormat* format) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000482 if (send_formats_.find(ssrc) == send_formats_.end()) {
483 return false;
484 }
485 *format = send_formats_[ssrc];
486 return true;
487 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200488 virtual bool SetSendStreamFormat(uint32_t ssrc, const VideoFormat& format) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000489 if (send_formats_.find(ssrc) == send_formats_.end()) {
490 return false;
491 }
492 send_formats_[ssrc] = format;
493 return true;
494 }
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200495 virtual bool SetSendParameters(const VideoSendParameters& params) {
496 return (SetSendCodecs(params.codecs) &&
497 SetSendRtpHeaderExtensions(params.extensions) &&
498 SetMaxSendBandwidth(params.max_bandwidth_bps) &&
499 SetOptions(params.options));
500 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000501
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200502 virtual bool SetRecvParameters(const VideoRecvParameters& params) {
503 return (SetRecvCodecs(params.codecs) &&
504 SetRecvRtpHeaderExtensions(params.extensions));
505 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000506 virtual bool AddSendStream(const StreamParams& sp) {
507 if (!RtpHelper<VideoMediaChannel>::AddSendStream(sp)) {
508 return false;
509 }
510 SetSendStreamDefaultFormat(sp.first_ssrc());
511 return true;
512 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200513 virtual bool RemoveSendStream(uint32_t ssrc) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000514 send_formats_.erase(ssrc);
515 return RtpHelper<VideoMediaChannel>::RemoveSendStream(ssrc);
516 }
517
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000518 virtual bool GetSendCodec(VideoCodec* send_codec) {
519 if (send_codecs_.empty()) {
520 return false;
521 }
522 *send_codec = send_codecs_[0];
523 return true;
524 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200525 virtual bool SetRenderer(uint32_t ssrc, VideoRenderer* r) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000526 if (ssrc != 0 && renderers_.find(ssrc) == renderers_.end()) {
527 return false;
528 }
529 if (ssrc != 0) {
530 renderers_[ssrc] = r;
531 }
532 return true;
533 }
534
535 virtual bool SetSend(bool send) { return set_sending(send); }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200536 virtual bool SetVideoSend(uint32_t ssrc, bool enable,
solenberg1dd98f32015-09-10 01:57:14 -0700537 const VideoOptions* options) {
solenbergdfc8f4f2015-10-01 02:31:10 -0700538 if (!RtpHelper<VideoMediaChannel>::MuteStream(ssrc, !enable)) {
solenberg1dd98f32015-09-10 01:57:14 -0700539 return false;
540 }
solenbergdfc8f4f2015-10-01 02:31:10 -0700541 if (enable && options) {
solenberg1dd98f32015-09-10 01:57:14 -0700542 return SetOptions(*options);
solenberg1dd98f32015-09-10 01:57:14 -0700543 }
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200544 return true;
solenberg1dd98f32015-09-10 01:57:14 -0700545 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200546 virtual bool SetCapturer(uint32_t ssrc, VideoCapturer* capturer) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000547 capturers_[ssrc] = capturer;
548 return true;
549 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200550 bool HasCapturer(uint32_t ssrc) const {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000551 return capturers_.find(ssrc) != capturers_.end();
552 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000553 virtual bool AddRecvStream(const StreamParams& sp) {
554 if (!RtpHelper<VideoMediaChannel>::AddRecvStream(sp))
555 return false;
556 renderers_[sp.first_ssrc()] = NULL;
557 return true;
558 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200559 virtual bool RemoveRecvStream(uint32_t ssrc) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000560 if (!RtpHelper<VideoMediaChannel>::RemoveRecvStream(ssrc))
561 return false;
562 renderers_.erase(ssrc);
563 return true;
564 }
565
pbos@webrtc.org058b1f12015-03-04 08:54:32 +0000566 virtual bool GetStats(VideoMediaInfo* info) { return false; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000567 virtual bool SendIntraFrame() {
568 sent_intra_frame_ = true;
569 return true;
570 }
571 virtual bool RequestIntraFrame() {
572 requested_intra_frame_ = true;
573 return true;
574 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000575 virtual void UpdateAspectRatio(int ratio_w, int ratio_h) {}
576 void set_sent_intra_frame(bool v) { sent_intra_frame_ = v; }
577 bool sent_intra_frame() const { return sent_intra_frame_; }
578 void set_requested_intra_frame(bool v) { requested_intra_frame_ = v; }
579 bool requested_intra_frame() const { return requested_intra_frame_; }
580
581 private:
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200582 bool SetRecvCodecs(const std::vector<VideoCodec>& codecs) {
583 if (fail_set_recv_codecs()) {
584 // Fake the failure in SetRecvCodecs.
585 return false;
586 }
587 recv_codecs_ = codecs;
588 return true;
589 }
590 bool SetSendCodecs(const std::vector<VideoCodec>& codecs) {
591 if (fail_set_send_codecs()) {
592 // Fake the failure in SetSendCodecs.
593 return false;
594 }
595 send_codecs_ = codecs;
596
597 for (std::vector<StreamParams>::const_iterator it = send_streams().begin();
598 it != send_streams().end(); ++it) {
599 SetSendStreamDefaultFormat(it->first_ssrc());
600 }
601 return true;
602 }
603 bool SetOptions(const VideoOptions& options) {
604 options_ = options;
605 return true;
606 }
607 bool SetMaxSendBandwidth(int bps) {
608 max_bps_ = bps;
609 return true;
610 }
611
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000612 // Be default, each send stream uses the first send codec format.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200613 void SetSendStreamDefaultFormat(uint32_t ssrc) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000614 if (!send_codecs_.empty()) {
615 send_formats_[ssrc] = VideoFormat(
616 send_codecs_[0].width, send_codecs_[0].height,
617 cricket::VideoFormat::FpsToInterval(send_codecs_[0].framerate),
618 cricket::FOURCC_I420);
619 }
620 }
621
622 FakeVideoEngine* engine_;
623 std::vector<VideoCodec> recv_codecs_;
624 std::vector<VideoCodec> send_codecs_;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200625 std::map<uint32_t, VideoRenderer*> renderers_;
626 std::map<uint32_t, VideoFormat> send_formats_;
627 std::map<uint32_t, VideoCapturer*> capturers_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000628 bool sent_intra_frame_;
629 bool requested_intra_frame_;
630 VideoOptions options_;
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000631 int max_bps_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000632};
633
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000634class FakeDataMediaChannel : public RtpHelper<DataMediaChannel> {
635 public:
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200636 explicit FakeDataMediaChannel(void* unused, const DataOptions& options)
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000637 : send_blocked_(false), max_bps_(-1) {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000638 ~FakeDataMediaChannel() {}
639 const std::vector<DataCodec>& recv_codecs() const { return recv_codecs_; }
640 const std::vector<DataCodec>& send_codecs() const { return send_codecs_; }
641 const std::vector<DataCodec>& codecs() const { return send_codecs(); }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000642 int max_bps() const { return max_bps_; }
643
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200644 virtual bool SetSendParameters(const DataSendParameters& params) {
645 return (SetSendCodecs(params.codecs) &&
646 SetMaxSendBandwidth(params.max_bandwidth_bps));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000647 }
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200648 virtual bool SetRecvParameters(const DataRecvParameters& params) {
649 return SetRecvCodecs(params.codecs);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000650 }
651 virtual bool SetSend(bool send) { return set_sending(send); }
652 virtual bool SetReceive(bool receive) {
653 set_playout(receive);
654 return true;
655 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000656 virtual bool AddRecvStream(const StreamParams& sp) {
657 if (!RtpHelper<DataMediaChannel>::AddRecvStream(sp))
658 return false;
659 return true;
660 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200661 virtual bool RemoveRecvStream(uint32_t ssrc) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000662 if (!RtpHelper<DataMediaChannel>::RemoveRecvStream(ssrc))
663 return false;
664 return true;
665 }
666
667 virtual bool SendData(const SendDataParams& params,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000668 const rtc::Buffer& payload,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000669 SendDataResult* result) {
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000670 if (send_blocked_) {
671 *result = SDR_BLOCK;
672 return false;
673 } else {
674 last_sent_data_params_ = params;
Karl Wiberg94784372015-04-20 14:03:07 +0200675 last_sent_data_ = std::string(payload.data<char>(), payload.size());
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000676 return true;
677 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000678 }
679
680 SendDataParams last_sent_data_params() { return last_sent_data_params_; }
681 std::string last_sent_data() { return last_sent_data_; }
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000682 bool is_send_blocked() { return send_blocked_; }
683 void set_send_blocked(bool blocked) { send_blocked_ = blocked; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000684
685 private:
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200686 bool SetRecvCodecs(const std::vector<DataCodec>& codecs) {
687 if (fail_set_recv_codecs()) {
688 // Fake the failure in SetRecvCodecs.
689 return false;
690 }
691 recv_codecs_ = codecs;
692 return true;
693 }
694 bool SetSendCodecs(const std::vector<DataCodec>& codecs) {
695 if (fail_set_send_codecs()) {
696 // Fake the failure in SetSendCodecs.
697 return false;
698 }
699 send_codecs_ = codecs;
700 return true;
701 }
702 bool SetMaxSendBandwidth(int bps) {
703 max_bps_ = bps;
704 return true;
705 }
706
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000707 std::vector<DataCodec> recv_codecs_;
708 std::vector<DataCodec> send_codecs_;
709 SendDataParams last_sent_data_params_;
710 std::string last_sent_data_;
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000711 bool send_blocked_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000712 int max_bps_;
713};
714
715// A base class for all of the shared parts between FakeVoiceEngine
716// and FakeVideoEngine.
717class FakeBaseEngine {
718 public:
719 FakeBaseEngine()
720 : loglevel_(-1),
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000721 options_changed_(false),
722 fail_create_channel_(false) {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000723 void SetLogging(int level, const char* filter) {
724 loglevel_ = level;
725 logfilter_ = filter;
726 }
727
728 void set_fail_create_channel(bool fail) { fail_create_channel_ = fail; }
729
730 const std::vector<RtpHeaderExtension>& rtp_header_extensions() const {
731 return rtp_header_extensions_;
732 }
henrike@webrtc.org79047f92014-03-06 23:46:59 +0000733 void set_rtp_header_extensions(
734 const std::vector<RtpHeaderExtension>& extensions) {
735 rtp_header_extensions_ = extensions;
736 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000737
738 protected:
739 int loglevel_;
740 std::string logfilter_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000741 // Flag used by optionsmessagehandler_unittest for checking whether any
742 // relevant setting has been updated.
743 // TODO(thaloun): Replace with explicit checks of before & after values.
744 bool options_changed_;
745 bool fail_create_channel_;
746 std::vector<RtpHeaderExtension> rtp_header_extensions_;
747};
748
749class FakeVoiceEngine : public FakeBaseEngine {
750 public:
751 FakeVoiceEngine()
solenberg4a3ccad2015-09-24 03:53:08 -0700752 : output_volume_(-1) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000753 // Add a fake audio codec. Note that the name must not be "" as there are
754 // sanity checks against that.
755 codecs_.push_back(AudioCodec(101, "fake_audio_codec", 0, 0, 1, 0));
756 }
Fredrik Solenberg9a416bd2015-05-22 09:04:09 +0200757 bool Init(rtc::Thread* worker_thread) { return true; }
758 void Terminate() {}
Fredrik Solenberg709ed672015-09-15 12:26:33 +0200759 webrtc::VoiceEngine* GetVoE() { return nullptr; }
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000760 AudioOptions GetOptions() const {
761 return options_;
762 }
763 bool SetOptions(const AudioOptions& options) {
764 options_ = options;
765 options_changed_ = true;
766 return true;
767 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000768
Fredrik Solenberg709ed672015-09-15 12:26:33 +0200769 VoiceMediaChannel* CreateChannel(webrtc::Call* call,
770 const AudioOptions& options) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000771 if (fail_create_channel_) {
Jelena Marusicc28a8962015-05-29 15:05:44 +0200772 return nullptr;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000773 }
774
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200775 FakeVoiceMediaChannel* ch = new FakeVoiceMediaChannel(this, options);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000776 channels_.push_back(ch);
777 return ch;
778 }
779 FakeVoiceMediaChannel* GetChannel(size_t index) {
780 return (channels_.size() > index) ? channels_[index] : NULL;
781 }
782 void UnregisterChannel(VoiceMediaChannel* channel) {
783 channels_.erase(std::find(channels_.begin(), channels_.end(), channel));
784 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000785
786 const std::vector<AudioCodec>& codecs() { return codecs_; }
787 void SetCodecs(const std::vector<AudioCodec> codecs) { codecs_ = codecs; }
788
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000789 bool SetDevices(const Device* in_device, const Device* out_device) {
790 in_device_ = (in_device) ? in_device->name : "";
791 out_device_ = (out_device) ? out_device->name : "";
792 options_changed_ = true;
793 return true;
794 }
795
796 bool GetOutputVolume(int* level) {
797 *level = output_volume_;
798 return true;
799 }
800
801 bool SetOutputVolume(int level) {
802 output_volume_ = level;
803 options_changed_ = true;
804 return true;
805 }
806
807 int GetInputLevel() { return 0; }
808
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000809 bool StartAecDump(rtc::PlatformFile file) { return false; }
wu@webrtc.orga9890802013-12-13 00:21:03 +0000810
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000811 private:
812 std::vector<FakeVoiceMediaChannel*> channels_;
813 std::vector<AudioCodec> codecs_;
814 int output_volume_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000815 std::string in_device_;
816 std::string out_device_;
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000817 AudioOptions options_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000818
819 friend class FakeMediaEngine;
820};
821
822class FakeVideoEngine : public FakeBaseEngine {
823 public:
Fredrik Solenberg709ed672015-09-15 12:26:33 +0200824 FakeVideoEngine() : capture_(false) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000825 // Add a fake video codec. Note that the name must not be "" as there are
826 // sanity checks against that.
827 codecs_.push_back(VideoCodec(0, "fake_video_codec", 0, 0, 0, 0));
828 }
Fredrik Solenberg9a416bd2015-05-22 09:04:09 +0200829 void Init() {}
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000830 bool SetOptions(const VideoOptions& options) {
831 options_ = options;
832 options_changed_ = true;
833 return true;
834 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000835 bool SetDefaultEncoderConfig(const VideoEncoderConfig& config) {
836 default_encoder_config_ = config;
837 return true;
838 }
839 const VideoEncoderConfig& default_encoder_config() const {
840 return default_encoder_config_;
841 }
842
Fredrik Solenberg709ed672015-09-15 12:26:33 +0200843 VideoMediaChannel* CreateChannel(webrtc::Call* call,
844 const VideoOptions& options) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000845 if (fail_create_channel_) {
846 return NULL;
847 }
848
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200849 FakeVideoMediaChannel* ch = new FakeVideoMediaChannel(this, options);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000850 channels_.push_back(ch);
851 return ch;
852 }
853 FakeVideoMediaChannel* GetChannel(size_t index) {
854 return (channels_.size() > index) ? channels_[index] : NULL;
855 }
856 void UnregisterChannel(VideoMediaChannel* channel) {
857 channels_.erase(std::find(channels_.begin(), channels_.end(), channel));
858 }
859
860 const std::vector<VideoCodec>& codecs() const { return codecs_; }
861 bool FindCodec(const VideoCodec& in) {
862 for (size_t i = 0; i < codecs_.size(); ++i) {
863 if (codecs_[i].Matches(in)) {
864 return true;
865 }
866 }
867 return false;
868 }
869 void SetCodecs(const std::vector<VideoCodec> codecs) { codecs_ = codecs; }
870
871 bool SetCaptureDevice(const Device* device) {
872 in_device_ = (device) ? device->name : "";
873 options_changed_ = true;
874 return true;
875 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000876 bool SetCapture(bool capture) {
877 capture_ = capture;
878 return true;
879 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000880
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000881 private:
882 std::vector<FakeVideoMediaChannel*> channels_;
883 std::vector<VideoCodec> codecs_;
884 VideoEncoderConfig default_encoder_config_;
885 std::string in_device_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000886 bool capture_;
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000887 VideoOptions options_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000888
889 friend class FakeMediaEngine;
890};
891
892class FakeMediaEngine :
893 public CompositeMediaEngine<FakeVoiceEngine, FakeVideoEngine> {
894 public:
895 FakeMediaEngine() {
896 voice_ = FakeVoiceEngine();
897 video_ = FakeVideoEngine();
898 }
899 virtual ~FakeMediaEngine() {}
900
henrike@webrtc.org79047f92014-03-06 23:46:59 +0000901 void SetAudioCodecs(const std::vector<AudioCodec>& codecs) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000902 voice_.SetCodecs(codecs);
903 }
henrike@webrtc.org79047f92014-03-06 23:46:59 +0000904 void SetVideoCodecs(const std::vector<VideoCodec>& codecs) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000905 video_.SetCodecs(codecs);
906 }
907
henrike@webrtc.org79047f92014-03-06 23:46:59 +0000908 void SetAudioRtpHeaderExtensions(
909 const std::vector<RtpHeaderExtension>& extensions) {
910 voice_.set_rtp_header_extensions(extensions);
911 }
912 void SetVideoRtpHeaderExtensions(
913 const std::vector<RtpHeaderExtension>& extensions) {
914 video_.set_rtp_header_extensions(extensions);
915 }
916
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000917 FakeVoiceMediaChannel* GetVoiceChannel(size_t index) {
918 return voice_.GetChannel(index);
919 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000920 FakeVideoMediaChannel* GetVideoChannel(size_t index) {
921 return video_.GetChannel(index);
922 }
923
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000924 AudioOptions audio_options() const { return voice_.options_; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000925 int output_volume() const { return voice_.output_volume_; }
926 const VideoEncoderConfig& default_video_encoder_config() const {
927 return video_.default_encoder_config_;
928 }
929 const std::string& audio_in_device() const { return voice_.in_device_; }
930 const std::string& audio_out_device() const { return voice_.out_device_; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000931 int voice_loglevel() const { return voice_.loglevel_; }
932 const std::string& voice_logfilter() const { return voice_.logfilter_; }
933 int video_loglevel() const { return video_.loglevel_; }
934 const std::string& video_logfilter() const { return video_.logfilter_; }
935 bool capture() const { return video_.capture_; }
936 bool options_changed() const {
937 return voice_.options_changed_ || video_.options_changed_;
938 }
939 void clear_options_changed() {
940 video_.options_changed_ = false;
941 voice_.options_changed_ = false;
942 }
943 void set_fail_create_channel(bool fail) {
944 voice_.set_fail_create_channel(fail);
945 video_.set_fail_create_channel(fail);
946 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000947};
948
949// CompositeMediaEngine with FakeVoiceEngine to expose SetAudioCodecs to
950// establish a media connectionwith minimum set of audio codes required
951template <class VIDEO>
952class CompositeMediaEngineWithFakeVoiceEngine :
953 public CompositeMediaEngine<FakeVoiceEngine, VIDEO> {
954 public:
955 CompositeMediaEngineWithFakeVoiceEngine() {}
956 virtual ~CompositeMediaEngineWithFakeVoiceEngine() {}
957
958 virtual void SetAudioCodecs(const std::vector<AudioCodec>& codecs) {
959 CompositeMediaEngine<FakeVoiceEngine, VIDEO>::voice_.SetCodecs(codecs);
960 }
961};
962
963// Have to come afterwards due to declaration order
964inline FakeVoiceMediaChannel::~FakeVoiceMediaChannel() {
965 if (engine_) {
966 engine_->UnregisterChannel(this);
967 }
968}
969
970inline FakeVideoMediaChannel::~FakeVideoMediaChannel() {
971 if (engine_) {
972 engine_->UnregisterChannel(this);
973 }
974}
975
976class FakeDataEngine : public DataEngineInterface {
977 public:
978 FakeDataEngine() : last_channel_type_(DCT_NONE) {}
979
980 virtual DataMediaChannel* CreateChannel(DataChannelType data_channel_type) {
981 last_channel_type_ = data_channel_type;
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200982 FakeDataMediaChannel* ch = new FakeDataMediaChannel(this, DataOptions());
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000983 channels_.push_back(ch);
984 return ch;
985 }
986
987 FakeDataMediaChannel* GetChannel(size_t index) {
988 return (channels_.size() > index) ? channels_[index] : NULL;
989 }
990
991 void UnregisterChannel(DataMediaChannel* channel) {
992 channels_.erase(std::find(channels_.begin(), channels_.end(), channel));
993 }
994
995 virtual void SetDataCodecs(const std::vector<DataCodec>& data_codecs) {
996 data_codecs_ = data_codecs;
997 }
998
999 virtual const std::vector<DataCodec>& data_codecs() { return data_codecs_; }
1000
1001 DataChannelType last_channel_type() const { return last_channel_type_; }
1002
1003 private:
1004 std::vector<FakeDataMediaChannel*> channels_;
1005 std::vector<DataCodec> data_codecs_;
1006 DataChannelType last_channel_type_;
1007};
1008
1009} // namespace cricket
1010
1011#endif // TALK_MEDIA_BASE_FAKEMEDIAENGINE_H_