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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
stefanc1aeaf02015-10-15 07:26:07 -070072 bool SendRtp(const void* data, int len, const rtc::PacketOptions& options) {
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);
stefanc1aeaf02015-10-15 07:26:07 -070078 return Base::SendPacket(&packet, options);
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);
stefanc1aeaf02015-10-15 07:26:07 -070083 return Base::SendRtcp(&packet, rtc::PacketOptions());
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 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000303
304 virtual bool GetActiveStreams(AudioInfo::StreamList* streams) { return true; }
305 virtual int GetOutputLevel() { return 0; }
306 void set_time_since_last_typing(int ms) { time_since_last_typing_ = ms; }
307 virtual int GetTimeSinceLastTyping() { return time_since_last_typing_; }
308 virtual void SetTypingDetectionParameters(
309 int time_window, int cost_per_typing, int reporting_threshold,
310 int penalty_decay, int type_event_delay) {}
311
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000312 virtual bool CanInsertDtmf() {
313 for (std::vector<AudioCodec>::const_iterator it = send_codecs_.begin();
314 it != send_codecs_.end(); ++it) {
315 // Find the DTMF telephone event "codec".
316 if (_stricmp(it->name.c_str(), "telephone-event") == 0) {
317 return true;
318 }
319 }
320 return false;
321 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200322 virtual bool InsertDtmf(uint32_t ssrc,
323 int event_code,
324 int duration,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000325 int flags) {
326 dtmf_info_queue_.push_back(DtmfInfo(ssrc, event_code, duration, flags));
327 return true;
328 }
329
solenberg4bac9c52015-10-09 02:32:53 -0700330 virtual bool SetOutputVolume(uint32_t ssrc, double volume) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000331 if (0 == ssrc) {
solenberg4bac9c52015-10-09 02:32:53 -0700332 std::map<uint32_t, double>::iterator it;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000333 for (it = output_scalings_.begin(); it != output_scalings_.end(); ++it) {
solenberg4bac9c52015-10-09 02:32:53 -0700334 it->second = volume;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000335 }
336 return true;
337 } else if (output_scalings_.find(ssrc) != output_scalings_.end()) {
solenberg4bac9c52015-10-09 02:32:53 -0700338 output_scalings_[ssrc] = volume;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000339 return true;
340 }
341 return false;
342 }
solenberg4bac9c52015-10-09 02:32:53 -0700343 bool GetOutputVolume(uint32_t ssrc, double* volume) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000344 if (output_scalings_.find(ssrc) == output_scalings_.end())
345 return false;
solenberg4bac9c52015-10-09 02:32:53 -0700346 *volume = output_scalings_[ssrc];
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000347 return true;
348 }
349
350 virtual bool GetStats(VoiceMediaInfo* info) { return false; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000351
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000352 private:
henrike@webrtc.orga7b98182014-02-21 15:51:43 +0000353 class VoiceChannelAudioSink : public AudioRenderer::Sink {
354 public:
355 explicit VoiceChannelAudioSink(AudioRenderer* renderer)
356 : renderer_(renderer) {
henrike@webrtc.orga7b98182014-02-21 15:51:43 +0000357 renderer_->SetSink(this);
358 }
359 virtual ~VoiceChannelAudioSink() {
360 if (renderer_) {
henrike@webrtc.orga7b98182014-02-21 15:51:43 +0000361 renderer_->SetSink(NULL);
362 }
363 }
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000364 void OnData(const void* audio_data,
365 int bits_per_sample,
366 int sample_rate,
367 int number_of_channels,
Peter Kastingdce40cf2015-08-24 14:52:23 -0700368 size_t number_of_frames) override {}
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000369 void OnClose() override { renderer_ = NULL; }
henrike@webrtc.orga7b98182014-02-21 15:51:43 +0000370 AudioRenderer* renderer() const { return renderer_; }
371
372 private:
373 AudioRenderer* renderer_;
374 };
375
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200376 bool SetRecvCodecs(const std::vector<AudioCodec>& codecs) {
377 if (fail_set_recv_codecs()) {
378 // Fake the failure in SetRecvCodecs.
379 return false;
380 }
381 recv_codecs_ = codecs;
382 return true;
383 }
384 bool SetSendCodecs(const std::vector<AudioCodec>& codecs) {
385 if (fail_set_send_codecs()) {
386 // Fake the failure in SetSendCodecs.
387 return false;
388 }
389 send_codecs_ = codecs;
390 return true;
391 }
392 bool SetMaxSendBandwidth(int bps) { return true; }
393 bool SetOptions(const AudioOptions& options) {
394 // Does a "merge" of current options and set options.
395 options_.SetAll(options);
396 return true;
397 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200398 bool SetLocalRenderer(uint32_t ssrc, AudioRenderer* renderer) {
solenberg1dd98f32015-09-10 01:57:14 -0700399 auto it = local_renderers_.find(ssrc);
400 if (renderer) {
401 if (it != local_renderers_.end()) {
402 ASSERT(it->second->renderer() == renderer);
403 } else {
404 local_renderers_.insert(std::make_pair(
405 ssrc, new VoiceChannelAudioSink(renderer)));
406 }
407 } else {
408 if (it != local_renderers_.end()) {
409 delete it->second;
410 local_renderers_.erase(it);
411 }
412 }
413 return true;
414 }
henrike@webrtc.orga7b98182014-02-21 15:51:43 +0000415
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000416 FakeVoiceEngine* engine_;
417 std::vector<AudioCodec> recv_codecs_;
418 std::vector<AudioCodec> send_codecs_;
solenberg4bac9c52015-10-09 02:32:53 -0700419 std::map<uint32_t, double> output_scalings_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000420 std::vector<DtmfInfo> dtmf_info_queue_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000421 int time_since_last_typing_;
422 AudioOptions options_;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200423 std::map<uint32_t, VoiceChannelAudioSink*> local_renderers_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000424};
425
426// A helper function to compare the FakeVoiceMediaChannel::DtmfInfo.
427inline bool CompareDtmfInfo(const FakeVoiceMediaChannel::DtmfInfo& info,
Peter Boström0c4e06b2015-10-07 12:23:21 +0200428 uint32_t ssrc,
429 int event_code,
430 int duration,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000431 int flags) {
432 return (info.duration == duration && info.event_code == event_code &&
433 info.flags == flags && info.ssrc == ssrc);
434}
435
436class FakeVideoMediaChannel : public RtpHelper<VideoMediaChannel> {
437 public:
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200438 explicit FakeVideoMediaChannel(FakeVideoEngine* engine,
439 const VideoOptions& options)
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000440 : engine_(engine),
441 sent_intra_frame_(false),
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000442 requested_intra_frame_(false),
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200443 max_bps_(-1) {
444 SetOptions(options);
445 }
buildbot@webrtc.org1ecbe452014-10-14 20:29:28 +0000446
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000447 ~FakeVideoMediaChannel();
448
449 const std::vector<VideoCodec>& recv_codecs() const { return recv_codecs_; }
450 const std::vector<VideoCodec>& send_codecs() const { return send_codecs_; }
451 const std::vector<VideoCodec>& codecs() const { return send_codecs(); }
452 bool rendering() const { return playout(); }
453 const VideoOptions& options() const { return options_; }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200454 const std::map<uint32_t, VideoRenderer*>& renderers() const {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000455 return renderers_;
456 }
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000457 int max_bps() const { return max_bps_; }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200458 bool GetSendStreamFormat(uint32_t ssrc, VideoFormat* format) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000459 if (send_formats_.find(ssrc) == send_formats_.end()) {
460 return false;
461 }
462 *format = send_formats_[ssrc];
463 return true;
464 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200465 virtual bool SetSendStreamFormat(uint32_t ssrc, const VideoFormat& format) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000466 if (send_formats_.find(ssrc) == send_formats_.end()) {
467 return false;
468 }
469 send_formats_[ssrc] = format;
470 return true;
471 }
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200472 virtual bool SetSendParameters(const VideoSendParameters& params) {
473 return (SetSendCodecs(params.codecs) &&
474 SetSendRtpHeaderExtensions(params.extensions) &&
475 SetMaxSendBandwidth(params.max_bandwidth_bps) &&
476 SetOptions(params.options));
477 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000478
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200479 virtual bool SetRecvParameters(const VideoRecvParameters& params) {
480 return (SetRecvCodecs(params.codecs) &&
481 SetRecvRtpHeaderExtensions(params.extensions));
482 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000483 virtual bool AddSendStream(const StreamParams& sp) {
484 if (!RtpHelper<VideoMediaChannel>::AddSendStream(sp)) {
485 return false;
486 }
487 SetSendStreamDefaultFormat(sp.first_ssrc());
488 return true;
489 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200490 virtual bool RemoveSendStream(uint32_t ssrc) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000491 send_formats_.erase(ssrc);
492 return RtpHelper<VideoMediaChannel>::RemoveSendStream(ssrc);
493 }
494
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000495 virtual bool GetSendCodec(VideoCodec* send_codec) {
496 if (send_codecs_.empty()) {
497 return false;
498 }
499 *send_codec = send_codecs_[0];
500 return true;
501 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200502 virtual bool SetRenderer(uint32_t ssrc, VideoRenderer* r) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000503 if (ssrc != 0 && renderers_.find(ssrc) == renderers_.end()) {
504 return false;
505 }
506 if (ssrc != 0) {
507 renderers_[ssrc] = r;
508 }
509 return true;
510 }
511
512 virtual bool SetSend(bool send) { return set_sending(send); }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200513 virtual bool SetVideoSend(uint32_t ssrc, bool enable,
solenberg1dd98f32015-09-10 01:57:14 -0700514 const VideoOptions* options) {
solenbergdfc8f4f2015-10-01 02:31:10 -0700515 if (!RtpHelper<VideoMediaChannel>::MuteStream(ssrc, !enable)) {
solenberg1dd98f32015-09-10 01:57:14 -0700516 return false;
517 }
solenbergdfc8f4f2015-10-01 02:31:10 -0700518 if (enable && options) {
solenberg1dd98f32015-09-10 01:57:14 -0700519 return SetOptions(*options);
solenberg1dd98f32015-09-10 01:57:14 -0700520 }
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200521 return true;
solenberg1dd98f32015-09-10 01:57:14 -0700522 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200523 virtual bool SetCapturer(uint32_t ssrc, VideoCapturer* capturer) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000524 capturers_[ssrc] = capturer;
525 return true;
526 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200527 bool HasCapturer(uint32_t ssrc) const {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000528 return capturers_.find(ssrc) != capturers_.end();
529 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000530 virtual bool AddRecvStream(const StreamParams& sp) {
531 if (!RtpHelper<VideoMediaChannel>::AddRecvStream(sp))
532 return false;
533 renderers_[sp.first_ssrc()] = NULL;
534 return true;
535 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200536 virtual bool RemoveRecvStream(uint32_t ssrc) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000537 if (!RtpHelper<VideoMediaChannel>::RemoveRecvStream(ssrc))
538 return false;
539 renderers_.erase(ssrc);
540 return true;
541 }
542
pbos@webrtc.org058b1f12015-03-04 08:54:32 +0000543 virtual bool GetStats(VideoMediaInfo* info) { return false; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000544 virtual bool SendIntraFrame() {
545 sent_intra_frame_ = true;
546 return true;
547 }
548 virtual bool RequestIntraFrame() {
549 requested_intra_frame_ = true;
550 return true;
551 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000552 virtual void UpdateAspectRatio(int ratio_w, int ratio_h) {}
553 void set_sent_intra_frame(bool v) { sent_intra_frame_ = v; }
554 bool sent_intra_frame() const { return sent_intra_frame_; }
555 void set_requested_intra_frame(bool v) { requested_intra_frame_ = v; }
556 bool requested_intra_frame() const { return requested_intra_frame_; }
557
558 private:
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200559 bool SetRecvCodecs(const std::vector<VideoCodec>& codecs) {
560 if (fail_set_recv_codecs()) {
561 // Fake the failure in SetRecvCodecs.
562 return false;
563 }
564 recv_codecs_ = codecs;
565 return true;
566 }
567 bool SetSendCodecs(const std::vector<VideoCodec>& codecs) {
568 if (fail_set_send_codecs()) {
569 // Fake the failure in SetSendCodecs.
570 return false;
571 }
572 send_codecs_ = codecs;
573
574 for (std::vector<StreamParams>::const_iterator it = send_streams().begin();
575 it != send_streams().end(); ++it) {
576 SetSendStreamDefaultFormat(it->first_ssrc());
577 }
578 return true;
579 }
580 bool SetOptions(const VideoOptions& options) {
581 options_ = options;
582 return true;
583 }
584 bool SetMaxSendBandwidth(int bps) {
585 max_bps_ = bps;
586 return true;
587 }
588
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000589 // Be default, each send stream uses the first send codec format.
Peter Boström0c4e06b2015-10-07 12:23:21 +0200590 void SetSendStreamDefaultFormat(uint32_t ssrc) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000591 if (!send_codecs_.empty()) {
592 send_formats_[ssrc] = VideoFormat(
593 send_codecs_[0].width, send_codecs_[0].height,
594 cricket::VideoFormat::FpsToInterval(send_codecs_[0].framerate),
595 cricket::FOURCC_I420);
596 }
597 }
598
599 FakeVideoEngine* engine_;
600 std::vector<VideoCodec> recv_codecs_;
601 std::vector<VideoCodec> send_codecs_;
Peter Boström0c4e06b2015-10-07 12:23:21 +0200602 std::map<uint32_t, VideoRenderer*> renderers_;
603 std::map<uint32_t, VideoFormat> send_formats_;
604 std::map<uint32_t, VideoCapturer*> capturers_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000605 bool sent_intra_frame_;
606 bool requested_intra_frame_;
607 VideoOptions options_;
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000608 int max_bps_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000609};
610
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000611class FakeDataMediaChannel : public RtpHelper<DataMediaChannel> {
612 public:
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200613 explicit FakeDataMediaChannel(void* unused, const DataOptions& options)
sergeyu@chromium.org4b26e2e2014-01-15 23:15:54 +0000614 : send_blocked_(false), max_bps_(-1) {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000615 ~FakeDataMediaChannel() {}
616 const std::vector<DataCodec>& recv_codecs() const { return recv_codecs_; }
617 const std::vector<DataCodec>& send_codecs() const { return send_codecs_; }
618 const std::vector<DataCodec>& codecs() const { return send_codecs(); }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000619 int max_bps() const { return max_bps_; }
620
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200621 virtual bool SetSendParameters(const DataSendParameters& params) {
622 return (SetSendCodecs(params.codecs) &&
623 SetMaxSendBandwidth(params.max_bandwidth_bps));
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000624 }
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200625 virtual bool SetRecvParameters(const DataRecvParameters& params) {
626 return SetRecvCodecs(params.codecs);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000627 }
628 virtual bool SetSend(bool send) { return set_sending(send); }
629 virtual bool SetReceive(bool receive) {
630 set_playout(receive);
631 return true;
632 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000633 virtual bool AddRecvStream(const StreamParams& sp) {
634 if (!RtpHelper<DataMediaChannel>::AddRecvStream(sp))
635 return false;
636 return true;
637 }
Peter Boström0c4e06b2015-10-07 12:23:21 +0200638 virtual bool RemoveRecvStream(uint32_t ssrc) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000639 if (!RtpHelper<DataMediaChannel>::RemoveRecvStream(ssrc))
640 return false;
641 return true;
642 }
643
644 virtual bool SendData(const SendDataParams& params,
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000645 const rtc::Buffer& payload,
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000646 SendDataResult* result) {
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000647 if (send_blocked_) {
648 *result = SDR_BLOCK;
649 return false;
650 } else {
651 last_sent_data_params_ = params;
Karl Wiberg94784372015-04-20 14:03:07 +0200652 last_sent_data_ = std::string(payload.data<char>(), payload.size());
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000653 return true;
654 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000655 }
656
657 SendDataParams last_sent_data_params() { return last_sent_data_params_; }
658 std::string last_sent_data() { return last_sent_data_; }
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000659 bool is_send_blocked() { return send_blocked_; }
660 void set_send_blocked(bool blocked) { send_blocked_ = blocked; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000661
662 private:
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200663 bool SetRecvCodecs(const std::vector<DataCodec>& codecs) {
664 if (fail_set_recv_codecs()) {
665 // Fake the failure in SetRecvCodecs.
666 return false;
667 }
668 recv_codecs_ = codecs;
669 return true;
670 }
671 bool SetSendCodecs(const std::vector<DataCodec>& codecs) {
672 if (fail_set_send_codecs()) {
673 // Fake the failure in SetSendCodecs.
674 return false;
675 }
676 send_codecs_ = codecs;
677 return true;
678 }
679 bool SetMaxSendBandwidth(int bps) {
680 max_bps_ = bps;
681 return true;
682 }
683
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000684 std::vector<DataCodec> recv_codecs_;
685 std::vector<DataCodec> send_codecs_;
686 SendDataParams last_sent_data_params_;
687 std::string last_sent_data_;
wu@webrtc.orgd64719d2013-08-01 00:00:07 +0000688 bool send_blocked_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000689 int max_bps_;
690};
691
692// A base class for all of the shared parts between FakeVoiceEngine
693// and FakeVideoEngine.
694class FakeBaseEngine {
695 public:
696 FakeBaseEngine()
697 : loglevel_(-1),
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000698 options_changed_(false),
699 fail_create_channel_(false) {}
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000700 void SetLogging(int level, const char* filter) {
701 loglevel_ = level;
702 logfilter_ = filter;
703 }
704
705 void set_fail_create_channel(bool fail) { fail_create_channel_ = fail; }
706
707 const std::vector<RtpHeaderExtension>& rtp_header_extensions() const {
708 return rtp_header_extensions_;
709 }
henrike@webrtc.org79047f92014-03-06 23:46:59 +0000710 void set_rtp_header_extensions(
711 const std::vector<RtpHeaderExtension>& extensions) {
712 rtp_header_extensions_ = extensions;
713 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000714
715 protected:
716 int loglevel_;
717 std::string logfilter_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000718 // Flag used by optionsmessagehandler_unittest for checking whether any
719 // relevant setting has been updated.
720 // TODO(thaloun): Replace with explicit checks of before & after values.
721 bool options_changed_;
722 bool fail_create_channel_;
723 std::vector<RtpHeaderExtension> rtp_header_extensions_;
724};
725
726class FakeVoiceEngine : public FakeBaseEngine {
727 public:
728 FakeVoiceEngine()
solenberg4a3ccad2015-09-24 03:53:08 -0700729 : output_volume_(-1) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000730 // Add a fake audio codec. Note that the name must not be "" as there are
731 // sanity checks against that.
732 codecs_.push_back(AudioCodec(101, "fake_audio_codec", 0, 0, 1, 0));
733 }
Fredrik Solenberg9a416bd2015-05-22 09:04:09 +0200734 bool Init(rtc::Thread* worker_thread) { return true; }
735 void Terminate() {}
Fredrik Solenberg709ed672015-09-15 12:26:33 +0200736 webrtc::VoiceEngine* GetVoE() { return nullptr; }
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000737 AudioOptions GetOptions() const {
738 return options_;
739 }
740 bool SetOptions(const AudioOptions& options) {
741 options_ = options;
742 options_changed_ = true;
743 return true;
744 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000745
Fredrik Solenberg709ed672015-09-15 12:26:33 +0200746 VoiceMediaChannel* CreateChannel(webrtc::Call* call,
747 const AudioOptions& options) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000748 if (fail_create_channel_) {
Jelena Marusicc28a8962015-05-29 15:05:44 +0200749 return nullptr;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000750 }
751
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200752 FakeVoiceMediaChannel* ch = new FakeVoiceMediaChannel(this, options);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000753 channels_.push_back(ch);
754 return ch;
755 }
756 FakeVoiceMediaChannel* GetChannel(size_t index) {
757 return (channels_.size() > index) ? channels_[index] : NULL;
758 }
759 void UnregisterChannel(VoiceMediaChannel* channel) {
760 channels_.erase(std::find(channels_.begin(), channels_.end(), channel));
761 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000762
763 const std::vector<AudioCodec>& codecs() { return codecs_; }
764 void SetCodecs(const std::vector<AudioCodec> codecs) { codecs_ = codecs; }
765
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000766 bool SetDevices(const Device* in_device, const Device* out_device) {
767 in_device_ = (in_device) ? in_device->name : "";
768 out_device_ = (out_device) ? out_device->name : "";
769 options_changed_ = true;
770 return true;
771 }
772
773 bool GetOutputVolume(int* level) {
774 *level = output_volume_;
775 return true;
776 }
777
778 bool SetOutputVolume(int level) {
779 output_volume_ = level;
780 options_changed_ = true;
781 return true;
782 }
783
784 int GetInputLevel() { return 0; }
785
buildbot@webrtc.orgd4e598d2014-07-29 17:36:52 +0000786 bool StartAecDump(rtc::PlatformFile file) { return false; }
wu@webrtc.orga9890802013-12-13 00:21:03 +0000787
ivoc112a3d82015-10-16 02:22:18 -0700788 bool StartRtcEventLog(rtc::PlatformFile file) { return false; }
789
790 void StopRtcEventLog() {}
791
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000792 private:
793 std::vector<FakeVoiceMediaChannel*> channels_;
794 std::vector<AudioCodec> codecs_;
795 int output_volume_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000796 std::string in_device_;
797 std::string out_device_;
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000798 AudioOptions options_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000799
800 friend class FakeMediaEngine;
801};
802
803class FakeVideoEngine : public FakeBaseEngine {
804 public:
Fredrik Solenberg709ed672015-09-15 12:26:33 +0200805 FakeVideoEngine() : capture_(false) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000806 // Add a fake video codec. Note that the name must not be "" as there are
807 // sanity checks against that.
808 codecs_.push_back(VideoCodec(0, "fake_video_codec", 0, 0, 0, 0));
809 }
Fredrik Solenberg9a416bd2015-05-22 09:04:09 +0200810 void Init() {}
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000811 bool SetOptions(const VideoOptions& options) {
812 options_ = options;
813 options_changed_ = true;
814 return true;
815 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000816 bool SetDefaultEncoderConfig(const VideoEncoderConfig& config) {
817 default_encoder_config_ = config;
818 return true;
819 }
820 const VideoEncoderConfig& default_encoder_config() const {
821 return default_encoder_config_;
822 }
823
Fredrik Solenberg709ed672015-09-15 12:26:33 +0200824 VideoMediaChannel* CreateChannel(webrtc::Call* call,
825 const VideoOptions& options) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000826 if (fail_create_channel_) {
827 return NULL;
828 }
829
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200830 FakeVideoMediaChannel* ch = new FakeVideoMediaChannel(this, options);
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000831 channels_.push_back(ch);
832 return ch;
833 }
834 FakeVideoMediaChannel* GetChannel(size_t index) {
835 return (channels_.size() > index) ? channels_[index] : NULL;
836 }
837 void UnregisterChannel(VideoMediaChannel* channel) {
838 channels_.erase(std::find(channels_.begin(), channels_.end(), channel));
839 }
840
841 const std::vector<VideoCodec>& codecs() const { return codecs_; }
842 bool FindCodec(const VideoCodec& in) {
843 for (size_t i = 0; i < codecs_.size(); ++i) {
844 if (codecs_[i].Matches(in)) {
845 return true;
846 }
847 }
848 return false;
849 }
850 void SetCodecs(const std::vector<VideoCodec> codecs) { codecs_ = codecs; }
851
852 bool SetCaptureDevice(const Device* device) {
853 in_device_ = (device) ? device->name : "";
854 options_changed_ = true;
855 return true;
856 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000857 bool SetCapture(bool capture) {
858 capture_ = capture;
859 return true;
860 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000861
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000862 private:
863 std::vector<FakeVideoMediaChannel*> channels_;
864 std::vector<VideoCodec> codecs_;
865 VideoEncoderConfig default_encoder_config_;
866 std::string in_device_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000867 bool capture_;
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000868 VideoOptions options_;
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000869
870 friend class FakeMediaEngine;
871};
872
873class FakeMediaEngine :
874 public CompositeMediaEngine<FakeVoiceEngine, FakeVideoEngine> {
875 public:
876 FakeMediaEngine() {
877 voice_ = FakeVoiceEngine();
878 video_ = FakeVideoEngine();
879 }
880 virtual ~FakeMediaEngine() {}
881
henrike@webrtc.org79047f92014-03-06 23:46:59 +0000882 void SetAudioCodecs(const std::vector<AudioCodec>& codecs) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000883 voice_.SetCodecs(codecs);
884 }
henrike@webrtc.org79047f92014-03-06 23:46:59 +0000885 void SetVideoCodecs(const std::vector<VideoCodec>& codecs) {
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000886 video_.SetCodecs(codecs);
887 }
888
henrike@webrtc.org79047f92014-03-06 23:46:59 +0000889 void SetAudioRtpHeaderExtensions(
890 const std::vector<RtpHeaderExtension>& extensions) {
891 voice_.set_rtp_header_extensions(extensions);
892 }
893 void SetVideoRtpHeaderExtensions(
894 const std::vector<RtpHeaderExtension>& extensions) {
895 video_.set_rtp_header_extensions(extensions);
896 }
897
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000898 FakeVoiceMediaChannel* GetVoiceChannel(size_t index) {
899 return voice_.GetChannel(index);
900 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000901 FakeVideoMediaChannel* GetVideoChannel(size_t index) {
902 return video_.GetChannel(index);
903 }
904
mallinath@webrtc.orga27be8e2013-09-27 23:04:10 +0000905 AudioOptions audio_options() const { return voice_.options_; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000906 int output_volume() const { return voice_.output_volume_; }
907 const VideoEncoderConfig& default_video_encoder_config() const {
908 return video_.default_encoder_config_;
909 }
910 const std::string& audio_in_device() const { return voice_.in_device_; }
911 const std::string& audio_out_device() const { return voice_.out_device_; }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000912 int voice_loglevel() const { return voice_.loglevel_; }
913 const std::string& voice_logfilter() const { return voice_.logfilter_; }
914 int video_loglevel() const { return video_.loglevel_; }
915 const std::string& video_logfilter() const { return video_.logfilter_; }
916 bool capture() const { return video_.capture_; }
917 bool options_changed() const {
918 return voice_.options_changed_ || video_.options_changed_;
919 }
920 void clear_options_changed() {
921 video_.options_changed_ = false;
922 voice_.options_changed_ = false;
923 }
924 void set_fail_create_channel(bool fail) {
925 voice_.set_fail_create_channel(fail);
926 video_.set_fail_create_channel(fail);
927 }
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000928};
929
930// CompositeMediaEngine with FakeVoiceEngine to expose SetAudioCodecs to
931// establish a media connectionwith minimum set of audio codes required
932template <class VIDEO>
933class CompositeMediaEngineWithFakeVoiceEngine :
934 public CompositeMediaEngine<FakeVoiceEngine, VIDEO> {
935 public:
936 CompositeMediaEngineWithFakeVoiceEngine() {}
937 virtual ~CompositeMediaEngineWithFakeVoiceEngine() {}
938
939 virtual void SetAudioCodecs(const std::vector<AudioCodec>& codecs) {
940 CompositeMediaEngine<FakeVoiceEngine, VIDEO>::voice_.SetCodecs(codecs);
941 }
942};
943
944// Have to come afterwards due to declaration order
945inline FakeVoiceMediaChannel::~FakeVoiceMediaChannel() {
946 if (engine_) {
947 engine_->UnregisterChannel(this);
948 }
949}
950
951inline FakeVideoMediaChannel::~FakeVideoMediaChannel() {
952 if (engine_) {
953 engine_->UnregisterChannel(this);
954 }
955}
956
957class FakeDataEngine : public DataEngineInterface {
958 public:
959 FakeDataEngine() : last_channel_type_(DCT_NONE) {}
960
961 virtual DataMediaChannel* CreateChannel(DataChannelType data_channel_type) {
962 last_channel_type_ = data_channel_type;
Fredrik Solenbergb071a192015-09-17 16:42:56 +0200963 FakeDataMediaChannel* ch = new FakeDataMediaChannel(this, DataOptions());
henrike@webrtc.org28e20752013-07-10 00:45:36 +0000964 channels_.push_back(ch);
965 return ch;
966 }
967
968 FakeDataMediaChannel* GetChannel(size_t index) {
969 return (channels_.size() > index) ? channels_[index] : NULL;
970 }
971
972 void UnregisterChannel(DataMediaChannel* channel) {
973 channels_.erase(std::find(channels_.begin(), channels_.end(), channel));
974 }
975
976 virtual void SetDataCodecs(const std::vector<DataCodec>& data_codecs) {
977 data_codecs_ = data_codecs;
978 }
979
980 virtual const std::vector<DataCodec>& data_codecs() { return data_codecs_; }
981
982 DataChannelType last_channel_type() const { return last_channel_type_; }
983
984 private:
985 std::vector<FakeDataMediaChannel*> channels_;
986 std::vector<DataCodec> data_codecs_;
987 DataChannelType last_channel_type_;
988};
989
990} // namespace cricket
991
992#endif // TALK_MEDIA_BASE_FAKEMEDIAENGINE_H_