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1 /* | 1 /* |
2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2004 The WebRTC project authors. All Rights Reserved. |
3 * | 3 * |
4 * Use of this source code is governed by a BSD-style license | 4 * Use of this source code is governed by a BSD-style license |
5 * that can be found in the LICENSE file in the root of the source | 5 * that can be found in the LICENSE file in the root of the source |
6 * tree. An additional intellectual property rights grant can be found | 6 * tree. An additional intellectual property rights grant can be found |
7 * in the file PATENTS. All contributing project authors may | 7 * in the file PATENTS. All contributing project authors may |
8 * be found in the AUTHORS file in the root of the source tree. | 8 * be found in the AUTHORS file in the root of the source tree. |
9 */ | 9 */ |
10 | 10 |
11 #ifndef WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_ | 11 #ifndef WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_ |
12 #define WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_ | 12 #define WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_ |
13 | 13 |
14 #include <list> | 14 #include <list> |
15 #include <map> | 15 #include <map> |
16 #include <memory> | 16 #include <memory> |
17 #include <set> | 17 #include <set> |
18 #include <string> | 18 #include <string> |
19 #include <vector> | 19 #include <vector> |
20 | 20 |
21 #include "webrtc/api/call/audio_sink.h" | 21 #include "webrtc/api/call/audio_sink.h" |
| 22 #include "webrtc/base/checks.h" |
22 #include "webrtc/base/copyonwritebuffer.h" | 23 #include "webrtc/base/copyonwritebuffer.h" |
23 #include "webrtc/base/networkroute.h" | 24 #include "webrtc/base/networkroute.h" |
24 #include "webrtc/base/stringutils.h" | 25 #include "webrtc/base/stringutils.h" |
25 #include "webrtc/media/base/audiosource.h" | 26 #include "webrtc/media/base/audiosource.h" |
26 #include "webrtc/media/base/mediaengine.h" | 27 #include "webrtc/media/base/mediaengine.h" |
27 #include "webrtc/media/base/rtputils.h" | 28 #include "webrtc/media/base/rtputils.h" |
28 #include "webrtc/media/base/streamparams.h" | 29 #include "webrtc/media/base/streamparams.h" |
29 #include "webrtc/p2p/base/sessiondescription.h" | 30 #include "webrtc/p2p/base/sessiondescription.h" |
30 | 31 |
31 using webrtc::RtpExtension; | 32 using webrtc::RtpExtension; |
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461 } | 462 } |
462 bool SetOptions(const AudioOptions& options) { | 463 bool SetOptions(const AudioOptions& options) { |
463 // Does a "merge" of current options and set options. | 464 // Does a "merge" of current options and set options. |
464 options_.SetAll(options); | 465 options_.SetAll(options); |
465 return true; | 466 return true; |
466 } | 467 } |
467 bool SetLocalSource(uint32_t ssrc, AudioSource* source) { | 468 bool SetLocalSource(uint32_t ssrc, AudioSource* source) { |
468 auto it = local_sinks_.find(ssrc); | 469 auto it = local_sinks_.find(ssrc); |
469 if (source) { | 470 if (source) { |
470 if (it != local_sinks_.end()) { | 471 if (it != local_sinks_.end()) { |
471 ASSERT(it->second->source() == source); | 472 RTC_DCHECK(it->second->source() == source); |
472 } else { | 473 } else { |
473 local_sinks_.insert( | 474 local_sinks_.insert( |
474 std::make_pair(ssrc, new VoiceChannelAudioSink(source))); | 475 std::make_pair(ssrc, new VoiceChannelAudioSink(source))); |
475 } | 476 } |
476 } else { | 477 } else { |
477 if (it != local_sinks_.end()) { | 478 if (it != local_sinks_.end()) { |
478 delete it->second; | 479 delete it->second; |
479 local_sinks_.erase(it); | 480 local_sinks_.erase(it); |
480 } | 481 } |
481 } | 482 } |
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965 virtual const std::vector<DataCodec>& data_codecs() { return data_codecs_; } | 966 virtual const std::vector<DataCodec>& data_codecs() { return data_codecs_; } |
966 | 967 |
967 private: | 968 private: |
968 std::vector<FakeDataMediaChannel*> channels_; | 969 std::vector<FakeDataMediaChannel*> channels_; |
969 std::vector<DataCodec> data_codecs_; | 970 std::vector<DataCodec> data_codecs_; |
970 }; | 971 }; |
971 | 972 |
972 } // namespace cricket | 973 } // namespace cricket |
973 | 974 |
974 #endif // WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_ | 975 #endif // WEBRTC_MEDIA_BASE_FAKEMEDIAENGINE_H_ |
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