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1 /* | 1 /* |
2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. | 2 * Copyright (c) 2012 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_VOICE_ENGINE_CHANNEL_H_ | 11 #ifndef WEBRTC_VOICE_ENGINE_CHANNEL_H_ |
12 #define WEBRTC_VOICE_ENGINE_CHANNEL_H_ | 12 #define WEBRTC_VOICE_ENGINE_CHANNEL_H_ |
13 | 13 |
14 #include <memory> | 14 #include <memory> |
15 | 15 |
16 #include "webrtc/api/audio/audio_mixer.h" | 16 #include "webrtc/api/audio/audio_mixer.h" |
17 #include "webrtc/api/call/audio_sink.h" | 17 #include "webrtc/api/call/audio_sink.h" |
18 #include "webrtc/base/criticalsection.h" | 18 #include "webrtc/base/criticalsection.h" |
19 #include "webrtc/base/optional.h" | 19 #include "webrtc/base/optional.h" |
20 #include "webrtc/common_audio/resampler/include/push_resampler.h" | 20 #include "webrtc/common_audio/resampler/include/push_resampler.h" |
21 #include "webrtc/common_types.h" | 21 #include "webrtc/common_types.h" |
22 #include "webrtc/modules/audio_coding/acm2/codec_manager.h" | 22 #include "webrtc/modules/audio_coding/acm2/codec_manager.h" |
23 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" | 23 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" |
24 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h" | |
24 #include "webrtc/modules/audio_coding/include/audio_coding_module.h" | 25 #include "webrtc/modules/audio_coding/include/audio_coding_module.h" |
25 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_d efines.h" | 26 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_d efines.h" |
26 #include "webrtc/modules/audio_processing/rms_level.h" | 27 #include "webrtc/modules/audio_processing/rms_level.h" |
27 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h" | 28 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h" |
28 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" | 29 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h" |
29 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" | 30 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h" |
30 #include "webrtc/voice_engine/file_player.h" | 31 #include "webrtc/voice_engine/file_player.h" |
31 #include "webrtc/voice_engine/file_recorder.h" | 32 #include "webrtc/voice_engine/file_recorder.h" |
32 #include "webrtc/voice_engine/include/voe_audio_processing.h" | 33 #include "webrtc/voice_engine/include/voe_audio_processing.h" |
33 #include "webrtc/voice_engine/include/voe_base.h" | 34 #include "webrtc/voice_engine/include/voe_base.h" |
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162 int32_t UpdateLocalTimeStamp(); | 163 int32_t UpdateLocalTimeStamp(); |
163 | 164 |
164 void SetSink(std::unique_ptr<AudioSinkInterface> sink); | 165 void SetSink(std::unique_ptr<AudioSinkInterface> sink); |
165 | 166 |
166 // TODO(ossu): Don't use! It's only here to confirm that the decoder factory | 167 // TODO(ossu): Don't use! It's only here to confirm that the decoder factory |
167 // passed into AudioReceiveStream is the same as the one set when creating the | 168 // passed into AudioReceiveStream is the same as the one set when creating the |
168 // ADM. Once Channel creation is moved into Audio{Send,Receive}Stream this can | 169 // ADM. Once Channel creation is moved into Audio{Send,Receive}Stream this can |
169 // go. | 170 // go. |
170 const rtc::scoped_refptr<AudioDecoderFactory>& GetAudioDecoderFactory() const; | 171 const rtc::scoped_refptr<AudioDecoderFactory>& GetAudioDecoderFactory() const; |
171 | 172 |
173 // Send using this encoder, with this payload type. | |
174 virtual bool SetEncoder(int payload_type, | |
the sun
2017/03/10 12:55:43
Remove "virtual"?
| |
175 std::unique_ptr<AudioEncoder> encoder); | |
176 | |
172 // API methods | 177 // API methods |
173 | 178 |
174 // VoEBase | 179 // VoEBase |
175 int32_t StartPlayout(); | 180 int32_t StartPlayout(); |
176 int32_t StopPlayout(); | 181 int32_t StopPlayout(); |
177 int32_t StartSend(); | 182 int32_t StartSend(); |
178 int32_t StopSend(); | 183 int32_t StopSend(); |
179 int32_t RegisterVoiceEngineObserver(VoiceEngineObserver& observer); | 184 int32_t RegisterVoiceEngineObserver(VoiceEngineObserver& observer); |
180 int32_t DeRegisterVoiceEngineObserver(); | 185 int32_t DeRegisterVoiceEngineObserver(); |
181 | 186 |
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513 std::unique_ptr<RateLimiter> retransmission_rate_limiter_; | 518 std::unique_ptr<RateLimiter> retransmission_rate_limiter_; |
514 | 519 |
515 // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed. | 520 // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed. |
516 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; | 521 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; |
517 }; | 522 }; |
518 | 523 |
519 } // namespace voe | 524 } // namespace voe |
520 } // namespace webrtc | 525 } // namespace webrtc |
521 | 526 |
522 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ | 527 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ |
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