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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"
Tommif888bb52015-12-12 01:37:01 +010041#include "webrtc/audio/audio_sink.h"
buildbot@webrtc.orga09a9992014-08-13 17:26:08 +000042#include "webrtc/base/buffer.h"
43#include "webrtc/base/stringutils.h"
Tommif888bb52015-12-12 01:37:01 +010044#include "webrtc/p2p/base/sessiondescription.h"
henrike@webrtc.org28e20752013-07-10 00:45:36 +000045
46namespace cricket {
47
48class FakeMediaEngine;
49class FakeVideoEngine;
50class FakeVoiceEngine;
51
52// A common helper class that handles sending and receiving RTP/RTCP packets.
53template <class Base> class RtpHelper : public Base {
54 public:
55 RtpHelper()
56 : sending_(false),
57 playout_(false),
58 fail_set_send_codecs_(false),
59 fail_set_recv_codecs_(false),
60 send_ssrc_(0),
61 ready_to_send_(false) {}
62 const std::vector<RtpHeaderExtension>& recv_extensions() {
63 return recv_extensions_;
64 }
65 const std::vector<RtpHeaderExtension>& send_extensions() {
66 return send_extensions_;
67 }
68 bool sending() const { return sending_; }
69 bool playout() const { return playout_; }
70 const std::list<std::string>& rtp_packets() const { return rtp_packets_; }
71 const std::list<std::string>& rtcp_packets() const { return rtcp_packets_; }
72
stefanc1aeaf02015-10-15 07:26:07 -070073 bool SendRtp(const void* data, int len, const rtc::PacketOptions& options) {
henrike@webrtc.org1e09a712013-07-26 19:17:59 +000074 if (!sending_) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +000075 return false;
76 }
Karl Wiberg94784372015-04-20 14:03:07 +020077 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
78 kMaxRtpPacketLen);
stefanc1aeaf02015-10-15 07:26:07 -070079 return Base::SendPacket(&packet, options);
henrike@webrtc.org28e20752013-07-10 00:45:36 +000080 }
81 bool SendRtcp(const void* data, int len) {
Karl Wiberg94784372015-04-20 14:03:07 +020082 rtc::Buffer packet(reinterpret_cast<const uint8_t*>(data), len,
83 kMaxRtpPacketLen);
stefanc1aeaf02015-10-15 07:26:07 -070084 return Base::SendRtcp(&packet, rtc::PacketOptions());
henrike@webrtc.org28e20752013-07-10 00:45:36 +000085 }
86
87 bool CheckRtp(const void* data, int len) {
88 bool success = !rtp_packets_.empty();
89 if (success) {
90 std::string packet = rtp_packets_.front();
91 rtp_packets_.pop_front();
92 success = (packet == std::string(static_cast<const char*>(data), len));
93 }
94 return success;
95 }
96 bool CheckRtcp(const void* data, int len) {
97 bool success = !rtcp_packets_.empty();
98 if (success) {
99 std::string packet = rtcp_packets_.front();
100 rtcp_packets_.pop_front();
101 success = (packet == std::string(static_cast<const char*>(data), len));
102 }
103 return success;
104 }
105 bool CheckNoRtp() { return rtp_packets_.empty(); }
106 bool CheckNoRtcp() { return rtcp_packets_.empty(); }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000107 void set_fail_set_send_codecs(bool fail) { fail_set_send_codecs_ = fail; }
108 void set_fail_set_recv_codecs(bool fail) { fail_set_recv_codecs_ = fail; }
109 virtual bool AddSendStream(const StreamParams& sp) {
110 if (std::find(send_streams_.begin(), send_streams_.end(), sp) !=
111 send_streams_.end()) {
112 return false;
113 }
114 send_streams_.push_back(sp);
115 return true;
116 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200117 virtual bool RemoveSendStream(uint32_t ssrc) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000118 return RemoveStreamBySsrc(&send_streams_, ssrc);
119 }
120 virtual bool AddRecvStream(const StreamParams& sp) {
121 if (std::find(receive_streams_.begin(), receive_streams_.end(), sp) !=
122 receive_streams_.end()) {
123 return false;
124 }
125 receive_streams_.push_back(sp);
126 return true;
127 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200128 virtual bool RemoveRecvStream(uint32_t ssrc) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000129 return RemoveStreamBySsrc(&receive_streams_, ssrc);
130 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200131 bool IsStreamMuted(uint32_t ssrc) const {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000132 bool ret = muted_streams_.find(ssrc) != muted_streams_.end();
133 // If |ssrc = 0| check if the first send stream is muted.
134 if (!ret && ssrc == 0 && !send_streams_.empty()) {
135 return muted_streams_.find(send_streams_[0].first_ssrc()) !=
136 muted_streams_.end();
137 }
138 return ret;
139 }
140 const std::vector<StreamParams>& send_streams() const {
141 return send_streams_;
142 }
143 const std::vector<StreamParams>& recv_streams() const {
144 return receive_streams_;
145 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200146 bool HasRecvStream(uint32_t ssrc) const {
tommi@webrtc.org586f2ed2015-01-22 23:00:41 +0000147 return GetStreamBySsrc(receive_streams_, ssrc) != nullptr;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000148 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200149 bool HasSendStream(uint32_t ssrc) const {
tommi@webrtc.org586f2ed2015-01-22 23:00:41 +0000150 return GetStreamBySsrc(send_streams_, ssrc) != nullptr;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000151 }
152 // TODO(perkj): This is to support legacy unit test that only check one
153 // sending stream.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200154 uint32_t send_ssrc() const {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000155 if (send_streams_.empty())
156 return 0;
157 return send_streams_[0].first_ssrc();
158 }
159
160 // TODO(perkj): This is to support legacy unit test that only check one
161 // sending stream.
162 const std::string rtcp_cname() {
163 if (send_streams_.empty())
164 return "";
165 return send_streams_[0].cname;
166 }
167
168 bool ready_to_send() const {
169 return ready_to_send_;
170 }
171
172 protected:
Peter Boström0c4e06b2015-10-07 12:23:21 +0200173 bool MuteStream(uint32_t ssrc, bool mute) {
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200174 if (!HasSendStream(ssrc) && ssrc != 0) {
solenberg1dd98f32015-09-10 01:57:14 -0700175 return false;
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200176 }
177 if (mute) {
solenberg1dd98f32015-09-10 01:57:14 -0700178 muted_streams_.insert(ssrc);
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200179 } else {
solenberg1dd98f32015-09-10 01:57:14 -0700180 muted_streams_.erase(ssrc);
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200181 }
solenberg1dd98f32015-09-10 01:57:14 -0700182 return true;
183 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000184 bool set_sending(bool send) {
185 sending_ = send;
186 return true;
187 }
188 void set_playout(bool playout) { playout_ = playout; }
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200189 bool SetRecvRtpHeaderExtensions(
190 const std::vector<RtpHeaderExtension>& extensions) {
191 recv_extensions_ = extensions;
192 return true;
193 }
194 bool SetSendRtpHeaderExtensions(
195 const std::vector<RtpHeaderExtension>& extensions) {
196 send_extensions_ = extensions;
197 return true;
198 }
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000199 virtual void OnPacketReceived(rtc::Buffer* packet,
200 const rtc::PacketTime& packet_time) {
Karl Wiberg94784372015-04-20 14:03:07 +0200201 rtp_packets_.push_back(std::string(packet->data<char>(), packet->size()));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000202 }
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000203 virtual void OnRtcpReceived(rtc::Buffer* packet,
204 const rtc::PacketTime& packet_time) {
Karl Wiberg94784372015-04-20 14:03:07 +0200205 rtcp_packets_.push_back(std::string(packet->data<char>(), packet->size()));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000206 }
207 virtual void OnReadyToSend(bool ready) {
208 ready_to_send_ = ready;
209 }
210 bool fail_set_send_codecs() const { return fail_set_send_codecs_; }
211 bool fail_set_recv_codecs() const { return fail_set_recv_codecs_; }
212
213 private:
214 bool sending_;
215 bool playout_;
216 std::vector<RtpHeaderExtension> recv_extensions_;
217 std::vector<RtpHeaderExtension> send_extensions_;
218 std::list<std::string> rtp_packets_;
219 std::list<std::string> rtcp_packets_;
220 std::vector<StreamParams> send_streams_;
221 std::vector<StreamParams> receive_streams_;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200222 std::set<uint32_t> muted_streams_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000223 bool fail_set_send_codecs_;
224 bool fail_set_recv_codecs_;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200225 uint32_t send_ssrc_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000226 std::string rtcp_cname_;
227 bool ready_to_send_;
228};
229
230class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
231 public:
232 struct DtmfInfo {
solenberg1d63dd02015-12-02 12:35:09 -0800233 DtmfInfo(uint32_t ssrc, int event_code, int duration)
Peter Boström0c4e06b2015-10-07 12:23:21 +0200234 : ssrc(ssrc),
235 event_code(event_code),
solenberg1d63dd02015-12-02 12:35:09 -0800236 duration(duration) {}
Peter Boström0c4e06b2015-10-07 12:23:21 +0200237 uint32_t ssrc;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000238 int event_code;
239 int duration;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000240 };
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200241 explicit FakeVoiceMediaChannel(FakeVoiceEngine* engine,
242 const AudioOptions& options)
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000243 : engine_(engine),
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000244 time_since_last_typing_(-1) {
solenberg4bac9c52015-10-09 02:32:53 -0700245 output_scalings_[0] = 1.0; // For default channel.
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200246 SetOptions(options);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000247 }
248 ~FakeVoiceMediaChannel();
249 const std::vector<AudioCodec>& recv_codecs() const { return recv_codecs_; }
250 const std::vector<AudioCodec>& send_codecs() const { return send_codecs_; }
251 const std::vector<AudioCodec>& codecs() const { return send_codecs(); }
252 const std::vector<DtmfInfo>& dtmf_info_queue() const {
253 return dtmf_info_queue_;
254 }
255 const AudioOptions& options() const { return options_; }
256
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200257 virtual bool SetSendParameters(const AudioSendParameters& params) {
258 return (SetSendCodecs(params.codecs) &&
259 SetSendRtpHeaderExtensions(params.extensions) &&
260 SetMaxSendBandwidth(params.max_bandwidth_bps) &&
261 SetOptions(params.options));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000262 }
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200263
264 virtual bool SetRecvParameters(const AudioRecvParameters& params) {
265 return (SetRecvCodecs(params.codecs) &&
266 SetRecvRtpHeaderExtensions(params.extensions));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000267 }
268 virtual bool SetPlayout(bool playout) {
269 set_playout(playout);
270 return true;
271 }
272 virtual bool SetSend(SendFlags flag) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000273 return set_sending(flag != SEND_NOTHING);
274 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200275 virtual bool SetAudioSend(uint32_t ssrc,
276 bool enable,
solenberg1dd98f32015-09-10 01:57:14 -0700277 const AudioOptions* options,
278 AudioRenderer* renderer) {
279 if (!SetLocalRenderer(ssrc, renderer)) {
280 return false;
281 }
solenbergdfc8f4f2015-10-01 02:31:10 -0700282 if (!RtpHelper<VoiceMediaChannel>::MuteStream(ssrc, !enable)) {
solenberg1dd98f32015-09-10 01:57:14 -0700283 return false;
284 }
solenbergdfc8f4f2015-10-01 02:31:10 -0700285 if (enable && options) {
solenberg1dd98f32015-09-10 01:57:14 -0700286 return SetOptions(*options);
287 }
288 return true;
289 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000290 virtual bool AddRecvStream(const StreamParams& sp) {
291 if (!RtpHelper<VoiceMediaChannel>::AddRecvStream(sp))
292 return false;
solenberg4bac9c52015-10-09 02:32:53 -0700293 output_scalings_[sp.first_ssrc()] = 1.0;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000294 return true;
295 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200296 virtual bool RemoveRecvStream(uint32_t ssrc) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000297 if (!RtpHelper<VoiceMediaChannel>::RemoveRecvStream(ssrc))
298 return false;
299 output_scalings_.erase(ssrc);
300 return true;
301 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000302
303 virtual bool GetActiveStreams(AudioInfo::StreamList* streams) { return true; }
304 virtual int GetOutputLevel() { return 0; }
305 void set_time_since_last_typing(int ms) { time_since_last_typing_ = ms; }
306 virtual int GetTimeSinceLastTyping() { return time_since_last_typing_; }
307 virtual void SetTypingDetectionParameters(
308 int time_window, int cost_per_typing, int reporting_threshold,
309 int penalty_decay, int type_event_delay) {}
310
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000311 virtual bool CanInsertDtmf() {
312 for (std::vector<AudioCodec>::const_iterator it = send_codecs_.begin();
313 it != send_codecs_.end(); ++it) {
314 // Find the DTMF telephone event "codec".
315 if (_stricmp(it->name.c_str(), "telephone-event") == 0) {
316 return true;
317 }
318 }
319 return false;
320 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200321 virtual bool InsertDtmf(uint32_t ssrc,
322 int event_code,
solenberg1d63dd02015-12-02 12:35:09 -0800323 int duration) {
324 dtmf_info_queue_.push_back(DtmfInfo(ssrc, event_code, duration));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000325 return true;
326 }
327
solenberg4bac9c52015-10-09 02:32:53 -0700328 virtual bool SetOutputVolume(uint32_t ssrc, double volume) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000329 if (0 == ssrc) {
solenberg4bac9c52015-10-09 02:32:53 -0700330 std::map<uint32_t, double>::iterator it;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000331 for (it = output_scalings_.begin(); it != output_scalings_.end(); ++it) {
solenberg4bac9c52015-10-09 02:32:53 -0700332 it->second = volume;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000333 }
334 return true;
335 } else if (output_scalings_.find(ssrc) != output_scalings_.end()) {
solenberg4bac9c52015-10-09 02:32:53 -0700336 output_scalings_[ssrc] = volume;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000337 return true;
338 }
339 return false;
340 }
solenberg4bac9c52015-10-09 02:32:53 -0700341 bool GetOutputVolume(uint32_t ssrc, double* volume) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000342 if (output_scalings_.find(ssrc) == output_scalings_.end())
343 return false;
solenberg4bac9c52015-10-09 02:32:53 -0700344 *volume = output_scalings_[ssrc];
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000345 return true;
346 }
347
348 virtual bool GetStats(VoiceMediaInfo* info) { return false; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000349
Tommif888bb52015-12-12 01:37:01 +0100350 virtual void SetRawAudioSink(
351 uint32_t ssrc,
deadbeef2d110be2016-01-13 12:00:26 -0800352 rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) {
353 sink_ = std::move(sink);
Tommif888bb52015-12-12 01:37:01 +0100354 }
355
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000356 private:
henrike@webrtc.orga7b98182014-02-21 15:51:43 +0000357 class VoiceChannelAudioSink : public AudioRenderer::Sink {
358 public:
359 explicit VoiceChannelAudioSink(AudioRenderer* renderer)
360 : renderer_(renderer) {
henrike@webrtc.orga7b98182014-02-21 15:51:43 +0000361 renderer_->SetSink(this);
362 }
363 virtual ~VoiceChannelAudioSink() {
364 if (renderer_) {
henrike@webrtc.orga7b98182014-02-21 15:51:43 +0000365 renderer_->SetSink(NULL);
366 }
367 }
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000368 void OnData(const void* audio_data,
369 int bits_per_sample,
370 int sample_rate,
Peter Kasting69558702016-01-12 16:26:35 -0800371 size_t number_of_channels,
Peter Kastingdce40cf2015-08-24 14:52:23 -0700372 size_t number_of_frames) override {}
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000373 void OnClose() override { renderer_ = NULL; }
henrike@webrtc.orga7b98182014-02-21 15:51:43 +0000374 AudioRenderer* renderer() const { return renderer_; }
375
376 private:
377 AudioRenderer* renderer_;
378 };
379
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200380 bool SetRecvCodecs(const std::vector<AudioCodec>& codecs) {
381 if (fail_set_recv_codecs()) {
382 // Fake the failure in SetRecvCodecs.
383 return false;
384 }
385 recv_codecs_ = codecs;
386 return true;
387 }
388 bool SetSendCodecs(const std::vector<AudioCodec>& codecs) {
389 if (fail_set_send_codecs()) {
390 // Fake the failure in SetSendCodecs.
391 return false;
392 }
393 send_codecs_ = codecs;
394 return true;
395 }
396 bool SetMaxSendBandwidth(int bps) { return true; }
397 bool SetOptions(const AudioOptions& options) {
398 // Does a "merge" of current options and set options.
399 options_.SetAll(options);
400 return true;
401 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200402 bool SetLocalRenderer(uint32_t ssrc, AudioRenderer* renderer) {
solenberg1dd98f32015-09-10 01:57:14 -0700403 auto it = local_renderers_.find(ssrc);
404 if (renderer) {
405 if (it != local_renderers_.end()) {
406 ASSERT(it->second->renderer() == renderer);
407 } else {
408 local_renderers_.insert(std::make_pair(
409 ssrc, new VoiceChannelAudioSink(renderer)));
410 }
411 } else {
412 if (it != local_renderers_.end()) {
413 delete it->second;
414 local_renderers_.erase(it);
415 }
416 }
417 return true;
418 }
henrike@webrtc.orga7b98182014-02-21 15:51:43 +0000419
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000420 FakeVoiceEngine* engine_;
421 std::vector<AudioCodec> recv_codecs_;
422 std::vector<AudioCodec> send_codecs_;
solenberg4bac9c52015-10-09 02:32:53 -0700423 std::map<uint32_t, double> output_scalings_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000424 std::vector<DtmfInfo> dtmf_info_queue_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000425 int time_since_last_typing_;
426 AudioOptions options_;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200427 std::map<uint32_t, VoiceChannelAudioSink*> local_renderers_;
deadbeef2d110be2016-01-13 12:00:26 -0800428 rtc::scoped_ptr<webrtc::AudioSinkInterface> sink_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000429};
430
431// A helper function to compare the FakeVoiceMediaChannel::DtmfInfo.
432inline bool CompareDtmfInfo(const FakeVoiceMediaChannel::DtmfInfo& info,
Peter Boström0c4e06b2015-10-07 12:23:21 +0200433 uint32_t ssrc,
434 int event_code,
solenberg1d63dd02015-12-02 12:35:09 -0800435 int duration) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000436 return (info.duration == duration && info.event_code == event_code &&
solenberg1d63dd02015-12-02 12:35:09 -0800437 info.ssrc == ssrc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000438}
439
440class FakeVideoMediaChannel : public RtpHelper<VideoMediaChannel> {
441 public:
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200442 explicit FakeVideoMediaChannel(FakeVideoEngine* engine,
443 const VideoOptions& options)
Peter Boströma6c39d92016-02-01 19:30:33 +0100444 : engine_(engine), max_bps_(-1) {
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200445 SetOptions(options);
446 }
buildbot@webrtc.org1ecbe452014-10-14 20:29:28 +0000447
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000448 ~FakeVideoMediaChannel();
449
450 const std::vector<VideoCodec>& recv_codecs() const { return recv_codecs_; }
451 const std::vector<VideoCodec>& send_codecs() const { return send_codecs_; }
452 const std::vector<VideoCodec>& codecs() const { return send_codecs(); }
453 bool rendering() const { return playout(); }
454 const VideoOptions& options() const { return options_; }
nisse08582ff2016-02-04 01:24:52 -0800455 const std::map<uint32_t, rtc::VideoSinkInterface<VideoFrame>*>& sinks()
456 const {
457 return sinks_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000458 }
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000459 int max_bps() const { return max_bps_; }
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200460 virtual bool SetSendParameters(const VideoSendParameters& params) {
461 return (SetSendCodecs(params.codecs) &&
462 SetSendRtpHeaderExtensions(params.extensions) &&
463 SetMaxSendBandwidth(params.max_bandwidth_bps) &&
464 SetOptions(params.options));
465 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000466
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200467 virtual bool SetRecvParameters(const VideoRecvParameters& params) {
468 return (SetRecvCodecs(params.codecs) &&
469 SetRecvRtpHeaderExtensions(params.extensions));
470 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000471 virtual bool AddSendStream(const StreamParams& sp) {
Peter Boströmce23bee2016-02-02 14:14:30 +0100472 return RtpHelper<VideoMediaChannel>::AddSendStream(sp);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000473 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200474 virtual bool RemoveSendStream(uint32_t ssrc) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000475 return RtpHelper<VideoMediaChannel>::RemoveSendStream(ssrc);
476 }
477
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000478 virtual bool GetSendCodec(VideoCodec* send_codec) {
479 if (send_codecs_.empty()) {
480 return false;
481 }
482 *send_codec = send_codecs_[0];
483 return true;
484 }
nisse08582ff2016-02-04 01:24:52 -0800485 bool SetSink(uint32_t ssrc,
486 rtc::VideoSinkInterface<cricket::VideoFrame>* sink) override {
487 if (ssrc != 0 && sinks_.find(ssrc) == sinks_.end()) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000488 return false;
489 }
490 if (ssrc != 0) {
nisse08582ff2016-02-04 01:24:52 -0800491 sinks_[ssrc] = sink;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000492 }
493 return true;
494 }
495
496 virtual bool SetSend(bool send) { return set_sending(send); }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200497 virtual bool SetVideoSend(uint32_t ssrc, bool enable,
solenberg1dd98f32015-09-10 01:57:14 -0700498 const VideoOptions* options) {
solenbergdfc8f4f2015-10-01 02:31:10 -0700499 if (!RtpHelper<VideoMediaChannel>::MuteStream(ssrc, !enable)) {
solenberg1dd98f32015-09-10 01:57:14 -0700500 return false;
501 }
solenbergdfc8f4f2015-10-01 02:31:10 -0700502 if (enable && options) {
solenberg1dd98f32015-09-10 01:57:14 -0700503 return SetOptions(*options);
solenberg1dd98f32015-09-10 01:57:14 -0700504 }
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200505 return true;
solenberg1dd98f32015-09-10 01:57:14 -0700506 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200507 virtual bool SetCapturer(uint32_t ssrc, VideoCapturer* capturer) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000508 capturers_[ssrc] = capturer;
509 return true;
510 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200511 bool HasCapturer(uint32_t ssrc) const {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000512 return capturers_.find(ssrc) != capturers_.end();
513 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000514 virtual bool AddRecvStream(const StreamParams& sp) {
515 if (!RtpHelper<VideoMediaChannel>::AddRecvStream(sp))
516 return false;
nisse08582ff2016-02-04 01:24:52 -0800517 sinks_[sp.first_ssrc()] = NULL;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000518 return true;
519 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200520 virtual bool RemoveRecvStream(uint32_t ssrc) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000521 if (!RtpHelper<VideoMediaChannel>::RemoveRecvStream(ssrc))
522 return false;
nisse08582ff2016-02-04 01:24:52 -0800523 sinks_.erase(ssrc);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000524 return true;
525 }
526
pbos@webrtc.org058b1f12015-03-04 08:54:32 +0000527 virtual bool GetStats(VideoMediaInfo* info) { return false; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000528
529 private:
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200530 bool SetRecvCodecs(const std::vector<VideoCodec>& codecs) {
531 if (fail_set_recv_codecs()) {
532 // Fake the failure in SetRecvCodecs.
533 return false;
534 }
535 recv_codecs_ = codecs;
536 return true;
537 }
538 bool SetSendCodecs(const std::vector<VideoCodec>& codecs) {
539 if (fail_set_send_codecs()) {
540 // Fake the failure in SetSendCodecs.
541 return false;
542 }
543 send_codecs_ = codecs;
544
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200545 return true;
546 }
547 bool SetOptions(const VideoOptions& options) {
548 options_ = options;
549 return true;
550 }
551 bool SetMaxSendBandwidth(int bps) {
552 max_bps_ = bps;
553 return true;
554 }
555
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000556 FakeVideoEngine* engine_;
557 std::vector<VideoCodec> recv_codecs_;
558 std::vector<VideoCodec> send_codecs_;
nisse08582ff2016-02-04 01:24:52 -0800559 std::map<uint32_t, rtc::VideoSinkInterface<VideoFrame>*> sinks_;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200560 std::map<uint32_t, VideoCapturer*> capturers_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000561 VideoOptions options_;
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000562 int max_bps_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000563};
564
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000565class FakeDataMediaChannel : public RtpHelper<DataMediaChannel> {
566 public:
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200567 explicit FakeDataMediaChannel(void* unused, const DataOptions& options)
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000568 : send_blocked_(false), max_bps_(-1) {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000569 ~FakeDataMediaChannel() {}
570 const std::vector<DataCodec>& recv_codecs() const { return recv_codecs_; }
571 const std::vector<DataCodec>& send_codecs() const { return send_codecs_; }
572 const std::vector<DataCodec>& codecs() const { return send_codecs(); }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000573 int max_bps() const { return max_bps_; }
574
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200575 virtual bool SetSendParameters(const DataSendParameters& params) {
576 return (SetSendCodecs(params.codecs) &&
577 SetMaxSendBandwidth(params.max_bandwidth_bps));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000578 }
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200579 virtual bool SetRecvParameters(const DataRecvParameters& params) {
580 return SetRecvCodecs(params.codecs);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000581 }
582 virtual bool SetSend(bool send) { return set_sending(send); }
583 virtual bool SetReceive(bool receive) {
584 set_playout(receive);
585 return true;
586 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000587 virtual bool AddRecvStream(const StreamParams& sp) {
588 if (!RtpHelper<DataMediaChannel>::AddRecvStream(sp))
589 return false;
590 return true;
591 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200592 virtual bool RemoveRecvStream(uint32_t ssrc) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000593 if (!RtpHelper<DataMediaChannel>::RemoveRecvStream(ssrc))
594 return false;
595 return true;
596 }
597
598 virtual bool SendData(const SendDataParams& params,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000599 const rtc::Buffer& payload,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000600 SendDataResult* result) {
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000601 if (send_blocked_) {
602 *result = SDR_BLOCK;
603 return false;
604 } else {
605 last_sent_data_params_ = params;
Karl Wiberg94784372015-04-20 14:03:07 +0200606 last_sent_data_ = std::string(payload.data<char>(), payload.size());
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000607 return true;
608 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000609 }
610
611 SendDataParams last_sent_data_params() { return last_sent_data_params_; }
612 std::string last_sent_data() { return last_sent_data_; }
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000613 bool is_send_blocked() { return send_blocked_; }
614 void set_send_blocked(bool blocked) { send_blocked_ = blocked; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000615
616 private:
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200617 bool SetRecvCodecs(const std::vector<DataCodec>& codecs) {
618 if (fail_set_recv_codecs()) {
619 // Fake the failure in SetRecvCodecs.
620 return false;
621 }
622 recv_codecs_ = codecs;
623 return true;
624 }
625 bool SetSendCodecs(const std::vector<DataCodec>& codecs) {
626 if (fail_set_send_codecs()) {
627 // Fake the failure in SetSendCodecs.
628 return false;
629 }
630 send_codecs_ = codecs;
631 return true;
632 }
633 bool SetMaxSendBandwidth(int bps) {
634 max_bps_ = bps;
635 return true;
636 }
637
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000638 std::vector<DataCodec> recv_codecs_;
639 std::vector<DataCodec> send_codecs_;
640 SendDataParams last_sent_data_params_;
641 std::string last_sent_data_;
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000642 bool send_blocked_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000643 int max_bps_;
644};
645
646// A base class for all of the shared parts between FakeVoiceEngine
647// and FakeVideoEngine.
648class FakeBaseEngine {
649 public:
650 FakeBaseEngine()
solenbergbd138382015-11-20 16:08:07 -0800651 : options_changed_(false),
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000652 fail_create_channel_(false) {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000653 void set_fail_create_channel(bool fail) { fail_create_channel_ = fail; }
654
Stefan Holmer9d69c3f2015-12-07 10:45:43 +0100655 RtpCapabilities GetCapabilities() const { return capabilities_; }
henrike@webrtc.org79047f92014-03-06 23:46:59 +0000656 void set_rtp_header_extensions(
657 const std::vector<RtpHeaderExtension>& extensions) {
Stefan Holmer9d69c3f2015-12-07 10:45:43 +0100658 capabilities_.header_extensions = extensions;
henrike@webrtc.org79047f92014-03-06 23:46:59 +0000659 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000660
661 protected:
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000662 // Flag used by optionsmessagehandler_unittest for checking whether any
663 // relevant setting has been updated.
664 // TODO(thaloun): Replace with explicit checks of before & after values.
665 bool options_changed_;
666 bool fail_create_channel_;
Stefan Holmer9d69c3f2015-12-07 10:45:43 +0100667 RtpCapabilities capabilities_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000668};
669
670class FakeVoiceEngine : public FakeBaseEngine {
671 public:
672 FakeVoiceEngine()
solenberg4a3ccad2015-09-24 03:53:08 -0700673 : output_volume_(-1) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000674 // Add a fake audio codec. Note that the name must not be "" as there are
675 // sanity checks against that.
676 codecs_.push_back(AudioCodec(101, "fake_audio_codec", 0, 0, 1, 0));
677 }
Fredrik Solenberg9a416bd2015-05-22 09:04:09 +0200678 bool Init(rtc::Thread* worker_thread) { return true; }
679 void Terminate() {}
solenberg566ef242015-11-06 15:34:49 -0800680 rtc::scoped_refptr<webrtc::AudioState> GetAudioState() const {
681 return rtc::scoped_refptr<webrtc::AudioState>();
682 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000683
Fredrik Solenberg709ed672015-09-15 12:26:33 +0200684 VoiceMediaChannel* CreateChannel(webrtc::Call* call,
685 const AudioOptions& options) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000686 if (fail_create_channel_) {
Jelena Marusicc28a8962015-05-29 15:05:44 +0200687 return nullptr;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000688 }
689
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200690 FakeVoiceMediaChannel* ch = new FakeVoiceMediaChannel(this, options);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000691 channels_.push_back(ch);
692 return ch;
693 }
694 FakeVoiceMediaChannel* GetChannel(size_t index) {
695 return (channels_.size() > index) ? channels_[index] : NULL;
696 }
697 void UnregisterChannel(VoiceMediaChannel* channel) {
698 channels_.erase(std::find(channels_.begin(), channels_.end(), channel));
699 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000700
701 const std::vector<AudioCodec>& codecs() { return codecs_; }
702 void SetCodecs(const std::vector<AudioCodec> codecs) { codecs_ = codecs; }
703
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000704 bool GetOutputVolume(int* level) {
705 *level = output_volume_;
706 return true;
707 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000708 bool SetOutputVolume(int level) {
709 output_volume_ = level;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000710 return true;
711 }
712
713 int GetInputLevel() { return 0; }
714
ivocd66b44d2016-01-15 03:06:36 -0800715 bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) {
716 return false;
717 }
wu@webrtc.orga9890802013-12-13 00:21:03 +0000718
ivoc797ef122015-10-22 03:25:41 -0700719 void StopAecDump() {}
720
ivoc112a3d82015-10-16 02:22:18 -0700721 bool StartRtcEventLog(rtc::PlatformFile file) { return false; }
722
723 void StopRtcEventLog() {}
724
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000725 private:
726 std::vector<FakeVoiceMediaChannel*> channels_;
727 std::vector<AudioCodec> codecs_;
728 int output_volume_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000729
730 friend class FakeMediaEngine;
731};
732
733class FakeVideoEngine : public FakeBaseEngine {
734 public:
Fredrik Solenberg709ed672015-09-15 12:26:33 +0200735 FakeVideoEngine() : capture_(false) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000736 // Add a fake video codec. Note that the name must not be "" as there are
737 // sanity checks against that.
738 codecs_.push_back(VideoCodec(0, "fake_video_codec", 0, 0, 0, 0));
739 }
Fredrik Solenberg9a416bd2015-05-22 09:04:09 +0200740 void Init() {}
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000741 bool SetOptions(const VideoOptions& options) {
742 options_ = options;
743 options_changed_ = true;
744 return true;
745 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000746
Fredrik Solenberg709ed672015-09-15 12:26:33 +0200747 VideoMediaChannel* CreateChannel(webrtc::Call* call,
748 const VideoOptions& options) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000749 if (fail_create_channel_) {
750 return NULL;
751 }
752
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200753 FakeVideoMediaChannel* ch = new FakeVideoMediaChannel(this, options);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000754 channels_.push_back(ch);
755 return ch;
756 }
757 FakeVideoMediaChannel* GetChannel(size_t index) {
758 return (channels_.size() > index) ? channels_[index] : NULL;
759 }
760 void UnregisterChannel(VideoMediaChannel* channel) {
761 channels_.erase(std::find(channels_.begin(), channels_.end(), channel));
762 }
763
764 const std::vector<VideoCodec>& codecs() const { return codecs_; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000765 void SetCodecs(const std::vector<VideoCodec> codecs) { codecs_ = codecs; }
766
767 bool SetCaptureDevice(const Device* device) {
768 in_device_ = (device) ? device->name : "";
769 options_changed_ = true;
770 return true;
771 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000772 bool SetCapture(bool capture) {
773 capture_ = capture;
774 return true;
775 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000776
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000777 private:
778 std::vector<FakeVideoMediaChannel*> channels_;
779 std::vector<VideoCodec> codecs_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000780 std::string in_device_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000781 bool capture_;
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000782 VideoOptions options_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000783
784 friend class FakeMediaEngine;
785};
786
787class FakeMediaEngine :
788 public CompositeMediaEngine<FakeVoiceEngine, FakeVideoEngine> {
789 public:
solenberg246b8172015-12-08 09:50:23 -0800790 FakeMediaEngine() {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000791 virtual ~FakeMediaEngine() {}
792
henrike@webrtc.org79047f92014-03-06 23:46:59 +0000793 void SetAudioCodecs(const std::vector<AudioCodec>& codecs) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000794 voice_.SetCodecs(codecs);
795 }
henrike@webrtc.org79047f92014-03-06 23:46:59 +0000796 void SetVideoCodecs(const std::vector<VideoCodec>& codecs) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000797 video_.SetCodecs(codecs);
798 }
799
henrike@webrtc.org79047f92014-03-06 23:46:59 +0000800 void SetAudioRtpHeaderExtensions(
801 const std::vector<RtpHeaderExtension>& extensions) {
802 voice_.set_rtp_header_extensions(extensions);
803 }
804 void SetVideoRtpHeaderExtensions(
805 const std::vector<RtpHeaderExtension>& extensions) {
806 video_.set_rtp_header_extensions(extensions);
807 }
808
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000809 FakeVoiceMediaChannel* GetVoiceChannel(size_t index) {
810 return voice_.GetChannel(index);
811 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000812 FakeVideoMediaChannel* GetVideoChannel(size_t index) {
813 return video_.GetChannel(index);
814 }
815
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000816 int output_volume() const { return voice_.output_volume_; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000817 bool capture() const { return video_.capture_; }
818 bool options_changed() const {
solenberg246b8172015-12-08 09:50:23 -0800819 return video_.options_changed_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000820 }
821 void clear_options_changed() {
822 video_.options_changed_ = false;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000823 }
824 void set_fail_create_channel(bool fail) {
825 voice_.set_fail_create_channel(fail);
826 video_.set_fail_create_channel(fail);
827 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000828};
829
830// CompositeMediaEngine with FakeVoiceEngine to expose SetAudioCodecs to
831// establish a media connectionwith minimum set of audio codes required
832template <class VIDEO>
833class CompositeMediaEngineWithFakeVoiceEngine :
834 public CompositeMediaEngine<FakeVoiceEngine, VIDEO> {
835 public:
836 CompositeMediaEngineWithFakeVoiceEngine() {}
837 virtual ~CompositeMediaEngineWithFakeVoiceEngine() {}
838
839 virtual void SetAudioCodecs(const std::vector<AudioCodec>& codecs) {
840 CompositeMediaEngine<FakeVoiceEngine, VIDEO>::voice_.SetCodecs(codecs);
841 }
842};
843
844// Have to come afterwards due to declaration order
845inline FakeVoiceMediaChannel::~FakeVoiceMediaChannel() {
846 if (engine_) {
847 engine_->UnregisterChannel(this);
848 }
849}
850
851inline FakeVideoMediaChannel::~FakeVideoMediaChannel() {
852 if (engine_) {
853 engine_->UnregisterChannel(this);
854 }
855}
856
857class FakeDataEngine : public DataEngineInterface {
858 public:
859 FakeDataEngine() : last_channel_type_(DCT_NONE) {}
860
861 virtual DataMediaChannel* CreateChannel(DataChannelType data_channel_type) {
862 last_channel_type_ = data_channel_type;
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200863 FakeDataMediaChannel* ch = new FakeDataMediaChannel(this, DataOptions());
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000864 channels_.push_back(ch);
865 return ch;
866 }
867
868 FakeDataMediaChannel* GetChannel(size_t index) {
869 return (channels_.size() > index) ? channels_[index] : NULL;
870 }
871
872 void UnregisterChannel(DataMediaChannel* channel) {
873 channels_.erase(std::find(channels_.begin(), channels_.end(), channel));
874 }
875
876 virtual void SetDataCodecs(const std::vector<DataCodec>& data_codecs) {
877 data_codecs_ = data_codecs;
878 }
879
880 virtual const std::vector<DataCodec>& data_codecs() { return data_codecs_; }
881
882 DataChannelType last_channel_type() const { return last_channel_type_; }
883
884 private:
885 std::vector<FakeDataMediaChannel*> channels_;
886 std::vector<DataCodec> data_codecs_;
887 DataChannelType last_channel_type_;
888};
889
890} // namespace cricket
891
892#endif // TALK_MEDIA_BASE_FAKEMEDIAENGINE_H_