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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 |
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| 22 #include "webrtc/base/networkroute.h" | 22 #include "webrtc/base/networkroute.h" |
| 23 #include "webrtc/base/scoped_ref_ptr.h" | 23 #include "webrtc/base/scoped_ref_ptr.h" |
| 24 #include "webrtc/base/thread_checker.h" | 24 #include "webrtc/base/thread_checker.h" |
| 25 #include "webrtc/call/audio_state.h" | 25 #include "webrtc/call/audio_state.h" |
| 26 #include "webrtc/call/call.h" | 26 #include "webrtc/call/call.h" |
| 27 #include "webrtc/config.h" | 27 #include "webrtc/config.h" |
| 28 #include "webrtc/media/base/rtputils.h" | 28 #include "webrtc/media/base/rtputils.h" |
| 29 #include "webrtc/media/engine/apm_helpers.h" | 29 #include "webrtc/media/engine/apm_helpers.h" |
| 30 #include "webrtc/media/engine/webrtccommon.h" | 30 #include "webrtc/media/engine/webrtccommon.h" |
| 31 #include "webrtc/media/engine/webrtcvoe.h" | 31 #include "webrtc/media/engine/webrtcvoe.h" |
| 32 #include "webrtc/modules/audio_coding/codecs/audio_encoder_factory.h" |
| 32 #include "webrtc/modules/audio_processing/include/audio_processing.h" | 33 #include "webrtc/modules/audio_processing/include/audio_processing.h" |
| 33 #include "webrtc/pc/channel.h" | 34 #include "webrtc/pc/channel.h" |
| 34 | 35 |
| 35 namespace webrtc { | 36 namespace webrtc { |
| 36 namespace voe { | 37 namespace voe { |
| 37 class TransmitMixer; | 38 class TransmitMixer; |
| 38 } // namespace voe | 39 } // namespace voe |
| 39 } // namespace webrtc | 40 } // namespace webrtc |
| 40 | 41 |
| 41 namespace cricket { | 42 namespace cricket { |
| 42 | 43 |
| 43 class AudioDeviceModule; | 44 class AudioDeviceModule; |
| 44 class AudioMixer; | 45 class AudioMixer; |
| 45 class AudioSource; | 46 class AudioSource; |
| 46 class VoEWrapper; | 47 class VoEWrapper; |
| 47 class WebRtcVoiceMediaChannel; | 48 class WebRtcVoiceMediaChannel; |
| 48 | 49 |
| 49 // WebRtcVoiceEngine is a class to be used with CompositeMediaEngine. | 50 // WebRtcVoiceEngine is a class to be used with CompositeMediaEngine. |
| 50 // It uses the WebRtc VoiceEngine library for audio handling. | 51 // It uses the WebRtc VoiceEngine library for audio handling. |
| 51 class WebRtcVoiceEngine final : public webrtc::TraceCallback { | 52 class WebRtcVoiceEngine final : public webrtc::TraceCallback { |
| 52 friend class WebRtcVoiceMediaChannel; | 53 friend class WebRtcVoiceMediaChannel; |
| 53 public: | 54 public: |
| 54 // Exposed for the WVoE/MC unit test. | |
| 55 static bool ToCodecInst(const AudioCodec& in, webrtc::CodecInst* out); | |
| 56 | |
| 57 WebRtcVoiceEngine( | 55 WebRtcVoiceEngine( |
| 58 webrtc::AudioDeviceModule* adm, | 56 webrtc::AudioDeviceModule* adm, |
| 59 const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory, | 57 const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory, |
| 60 rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer); | 58 rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer); |
| 61 // Dependency injection for testing. | 59 // Dependency injection for testing. |
| 62 WebRtcVoiceEngine( | 60 WebRtcVoiceEngine( |
| 63 webrtc::AudioDeviceModule* adm, | 61 webrtc::AudioDeviceModule* adm, |
| 64 const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory, | 62 const rtc::scoped_refptr<webrtc::AudioDecoderFactory>& decoder_factory, |
| 65 rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer, | 63 rtc::scoped_refptr<webrtc::AudioMixer> audio_mixer, |
| 66 VoEWrapper* voe_wrapper); | 64 VoEWrapper* voe_wrapper); |
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| 107 | 105 |
| 108 // webrtc::TraceCallback: | 106 // webrtc::TraceCallback: |
| 109 void Print(webrtc::TraceLevel level, const char* trace, int length) override; | 107 void Print(webrtc::TraceLevel level, const char* trace, int length) override; |
| 110 | 108 |
| 111 void StartAecDump(const std::string& filename); | 109 void StartAecDump(const std::string& filename); |
| 112 int CreateVoEChannel(); | 110 int CreateVoEChannel(); |
| 113 webrtc::AudioDeviceModule* adm(); | 111 webrtc::AudioDeviceModule* adm(); |
| 114 webrtc::AudioProcessing* apm(); | 112 webrtc::AudioProcessing* apm(); |
| 115 webrtc::voe::TransmitMixer* transmit_mixer(); | 113 webrtc::voe::TransmitMixer* transmit_mixer(); |
| 116 | 114 |
| 117 AudioCodecs CollectRecvCodecs() const; | 115 AudioCodecs CollectCodecs( |
| 116 const std::vector<webrtc::AudioCodecSpec>& specs) const; |
| 118 | 117 |
| 119 rtc::ThreadChecker signal_thread_checker_; | 118 rtc::ThreadChecker signal_thread_checker_; |
| 120 rtc::ThreadChecker worker_thread_checker_; | 119 rtc::ThreadChecker worker_thread_checker_; |
| 121 | 120 |
| 122 // The audio device manager. | 121 // The audio device manager. |
| 123 rtc::scoped_refptr<webrtc::AudioDeviceModule> adm_; | 122 rtc::scoped_refptr<webrtc::AudioDeviceModule> adm_; |
| 123 rtc::scoped_refptr<webrtc::AudioEncoderFactory> encoder_factory_; |
| 124 rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory_; | 124 rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory_; |
| 125 // Reference to the APM, owned by VoE. | 125 // Reference to the APM, owned by VoE. |
| 126 webrtc::AudioProcessing* apm_ = nullptr; | 126 webrtc::AudioProcessing* apm_ = nullptr; |
| 127 // Reference to the TransmitMixer, owned by VoE. | 127 // Reference to the TransmitMixer, owned by VoE. |
| 128 webrtc::voe::TransmitMixer* transmit_mixer_ = nullptr; | 128 webrtc::voe::TransmitMixer* transmit_mixer_ = nullptr; |
| 129 // The primary instance of WebRtc VoiceEngine. | 129 // The primary instance of WebRtc VoiceEngine. |
| 130 std::unique_ptr<VoEWrapper> voe_wrapper_; | 130 std::unique_ptr<VoEWrapper> voe_wrapper_; |
| 131 rtc::scoped_refptr<webrtc::AudioState> audio_state_; | 131 rtc::scoped_refptr<webrtc::AudioState> audio_state_; |
| 132 std::vector<AudioCodec> send_codecs_; | 132 std::vector<AudioCodec> send_codecs_; |
| 133 std::vector<AudioCodec> recv_codecs_; | 133 std::vector<AudioCodec> recv_codecs_; |
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| 284 uint32_t receiver_reports_ssrc_ = 0xFA17FA17u; | 284 uint32_t receiver_reports_ssrc_ = 0xFA17FA17u; |
| 285 | 285 |
| 286 class WebRtcAudioSendStream; | 286 class WebRtcAudioSendStream; |
| 287 std::map<uint32_t, WebRtcAudioSendStream*> send_streams_; | 287 std::map<uint32_t, WebRtcAudioSendStream*> send_streams_; |
| 288 std::vector<webrtc::RtpExtension> send_rtp_extensions_; | 288 std::vector<webrtc::RtpExtension> send_rtp_extensions_; |
| 289 | 289 |
| 290 class WebRtcAudioReceiveStream; | 290 class WebRtcAudioReceiveStream; |
| 291 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; | 291 std::map<uint32_t, WebRtcAudioReceiveStream*> recv_streams_; |
| 292 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; | 292 std::vector<webrtc::RtpExtension> recv_rtp_extensions_; |
| 293 | 293 |
| 294 webrtc::AudioSendStream::Config::SendCodecSpec send_codec_spec_; | 294 rtc::Optional<webrtc::AudioSendStream::Config::SendCodecSpec> |
| 295 send_codec_spec_; |
| 295 | 296 |
| 296 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); | 297 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcVoiceMediaChannel); |
| 297 }; | 298 }; |
| 298 } // namespace cricket | 299 } // namespace cricket |
| 299 | 300 |
| 300 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ | 301 #endif // WEBRTC_MEDIA_ENGINE_WEBRTCVOICEENGINE_H_ |
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