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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 |
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83 uint32_t interarrival_jitter; | 83 uint32_t interarrival_jitter; |
84 uint32_t last_SR_timestamp; | 84 uint32_t last_SR_timestamp; |
85 uint32_t delay_since_last_SR; | 85 uint32_t delay_since_last_SR; |
86 }; | 86 }; |
87 | 87 |
88 namespace voe { | 88 namespace voe { |
89 | 89 |
90 class RtcEventLogProxy; | 90 class RtcEventLogProxy; |
91 class RtcpRttStatsProxy; | 91 class RtcpRttStatsProxy; |
92 class RtpPacketSenderProxy; | 92 class RtpPacketSenderProxy; |
93 class Statistics; | |
94 class TransportFeedbackProxy; | 93 class TransportFeedbackProxy; |
95 class TransportSequenceNumberProxy; | 94 class TransportSequenceNumberProxy; |
96 class VoERtcpObserver; | 95 class VoERtcpObserver; |
97 | 96 |
98 // Helper class to simplify locking scheme for members that are accessed from | 97 // Helper class to simplify locking scheme for members that are accessed from |
99 // multiple threads. | 98 // multiple threads. |
100 // Example: a member can be set on thread T1 and read by an internal audio | 99 // Example: a member can be set on thread T1 and read by an internal audio |
101 // thread T2. Accessing the member via this class ensures that we are | 100 // thread T2. Accessing the member via this class ensures that we are |
102 // safe and also avoid TSan v2 warnings. | 101 // safe and also avoid TSan v2 warnings. |
103 class ChannelState { | 102 class ChannelState { |
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150 virtual ~Channel(); | 149 virtual ~Channel(); |
151 static int32_t CreateChannel(Channel*& channel, | 150 static int32_t CreateChannel(Channel*& channel, |
152 int32_t channelId, | 151 int32_t channelId, |
153 uint32_t instanceId, | 152 uint32_t instanceId, |
154 const VoEBase::ChannelConfig& config); | 153 const VoEBase::ChannelConfig& config); |
155 Channel(int32_t channelId, | 154 Channel(int32_t channelId, |
156 uint32_t instanceId, | 155 uint32_t instanceId, |
157 const VoEBase::ChannelConfig& config); | 156 const VoEBase::ChannelConfig& config); |
158 int32_t Init(); | 157 int32_t Init(); |
159 void Terminate(); | 158 void Terminate(); |
160 int32_t SetEngineInformation(Statistics& engineStatistics, | 159 int32_t SetEngineInformation(ProcessThread& moduleProcessThread, |
161 ProcessThread& moduleProcessThread, | |
162 AudioDeviceModule& audioDeviceModule, | 160 AudioDeviceModule& audioDeviceModule, |
163 rtc::CriticalSection* callbackCritSect, | |
164 rtc::TaskQueue* encoder_queue); | 161 rtc::TaskQueue* encoder_queue); |
165 | 162 |
166 void SetSink(std::unique_ptr<AudioSinkInterface> sink); | 163 void SetSink(std::unique_ptr<AudioSinkInterface> sink); |
167 | 164 |
168 // TODO(ossu): Don't use! It's only here to confirm that the decoder factory | 165 // TODO(ossu): Don't use! It's only here to confirm that the decoder factory |
169 // passed into AudioReceiveStream is the same as the one set when creating the | 166 // passed into AudioReceiveStream is the same as the one set when creating the |
170 // ADM. Once Channel creation is moved into Audio{Send,Receive}Stream this can | 167 // ADM. Once Channel creation is moved into Audio{Send,Receive}Stream this can |
171 // go. | 168 // go. |
172 const rtc::scoped_refptr<AudioDecoderFactory>& GetAudioDecoderFactory() const; | 169 const rtc::scoped_refptr<AudioDecoderFactory>& GetAudioDecoderFactory() const; |
173 | 170 |
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190 int32_t GetSendCodec(CodecInst& codec); | 187 int32_t GetSendCodec(CodecInst& codec); |
191 int32_t GetRecCodec(CodecInst& codec); | 188 int32_t GetRecCodec(CodecInst& codec); |
192 int32_t SetSendCodec(const CodecInst& codec); | 189 int32_t SetSendCodec(const CodecInst& codec); |
193 void SetBitRate(int bitrate_bps, int64_t probing_interval_ms); | 190 void SetBitRate(int bitrate_bps, int64_t probing_interval_ms); |
194 bool EnableAudioNetworkAdaptor(const std::string& config_string); | 191 bool EnableAudioNetworkAdaptor(const std::string& config_string); |
195 void DisableAudioNetworkAdaptor(); | 192 void DisableAudioNetworkAdaptor(); |
196 void SetReceiverFrameLengthRange(int min_frame_length_ms, | 193 void SetReceiverFrameLengthRange(int min_frame_length_ms, |
197 int max_frame_length_ms); | 194 int max_frame_length_ms); |
198 | 195 |
199 // Network | 196 // Network |
200 int32_t RegisterExternalTransport(Transport* transport); | 197 void RegisterTransport(Transport* transport); |
201 int32_t DeRegisterExternalTransport(); | |
202 // TODO(nisse, solenberg): Delete when VoENetwork is deleted. | 198 // TODO(nisse, solenberg): Delete when VoENetwork is deleted. |
203 int32_t ReceivedRTCPPacket(const uint8_t* data, size_t length); | 199 int32_t ReceivedRTCPPacket(const uint8_t* data, size_t length); |
204 void OnRtpPacket(const RtpPacketReceived& packet); | 200 void OnRtpPacket(const RtpPacketReceived& packet); |
205 | 201 |
206 // Muting, Volume and Level. | 202 // Muting, Volume and Level. |
207 void SetInputMute(bool enable); | 203 void SetInputMute(bool enable); |
208 void SetChannelOutputVolumeScaling(float scaling); | 204 void SetChannelOutputVolumeScaling(float scaling); |
209 int GetSpeechOutputLevel() const; | 205 int GetSpeechOutputLevel() const; |
210 int GetSpeechOutputLevelFullRange() const; | 206 int GetSpeechOutputLevelFullRange() const; |
211 // See description of "totalAudioEnergy" in the WebRTC stats spec: | 207 // See description of "totalAudioEnergy" in the WebRTC stats spec: |
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279 AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo( | 275 AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo( |
280 int sample_rate_hz, | 276 int sample_rate_hz, |
281 AudioFrame* audio_frame); | 277 AudioFrame* audio_frame); |
282 | 278 |
283 int PreferredSampleRate() const; | 279 int PreferredSampleRate() const; |
284 | 280 |
285 uint32_t InstanceId() const { return _instanceId; } | 281 uint32_t InstanceId() const { return _instanceId; } |
286 int32_t ChannelId() const { return _channelId; } | 282 int32_t ChannelId() const { return _channelId; } |
287 bool Playing() const { return channel_state_.Get().playing; } | 283 bool Playing() const { return channel_state_.Get().playing; } |
288 bool Sending() const { return channel_state_.Get().sending; } | 284 bool Sending() const { return channel_state_.Get().sending; } |
289 bool ExternalTransport() const { | |
290 rtc::CritScope cs(&_callbackCritSect); | |
291 return _externalTransport; | |
292 } | |
293 RtpRtcp* RtpRtcpModulePtr() const { return _rtpRtcpModule.get(); } | 285 RtpRtcp* RtpRtcpModulePtr() const { return _rtpRtcpModule.get(); } |
294 int8_t OutputEnergyLevel() const { return _outputAudioLevel.Level(); } | 286 int8_t OutputEnergyLevel() const { return _outputAudioLevel.Level(); } |
295 | 287 |
296 // ProcessAndEncodeAudio() creates an audio frame copy and posts a task | 288 // ProcessAndEncodeAudio() creates an audio frame copy and posts a task |
297 // on the shared encoder task queue, wich in turn calls (on the queue) | 289 // on the shared encoder task queue, wich in turn calls (on the queue) |
298 // ProcessAndEncodeAudioOnTaskQueue() where the actual processing of the | 290 // ProcessAndEncodeAudioOnTaskQueue() where the actual processing of the |
299 // audio takes place. The processing mainly consists of encoding and preparing | 291 // audio takes place. The processing mainly consists of encoding and preparing |
300 // the result for sending by adding it to a send queue. | 292 // the result for sending by adding it to a send queue. |
301 // The main reason for using a task queue here is to release the native, | 293 // The main reason for using a task queue here is to release the native, |
302 // OS-specific, audio capture thread as soon as possible to ensure that it | 294 // OS-specific, audio capture thread as soon as possible to ensure that it |
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388 std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry_; | 380 std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry_; |
389 std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_; | 381 std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_; |
390 std::unique_ptr<RtpReceiver> rtp_receiver_; | 382 std::unique_ptr<RtpReceiver> rtp_receiver_; |
391 TelephoneEventHandler* telephone_event_handler_; | 383 TelephoneEventHandler* telephone_event_handler_; |
392 std::unique_ptr<RtpRtcp> _rtpRtcpModule; | 384 std::unique_ptr<RtpRtcp> _rtpRtcpModule; |
393 std::unique_ptr<AudioCodingModule> audio_coding_; | 385 std::unique_ptr<AudioCodingModule> audio_coding_; |
394 acm2::CodecManager codec_manager_; | 386 acm2::CodecManager codec_manager_; |
395 acm2::RentACodec rent_a_codec_; | 387 acm2::RentACodec rent_a_codec_; |
396 std::unique_ptr<AudioSinkInterface> audio_sink_; | 388 std::unique_ptr<AudioSinkInterface> audio_sink_; |
397 AudioLevel _outputAudioLevel; | 389 AudioLevel _outputAudioLevel; |
398 bool _externalTransport; | |
399 // Downsamples to the codec rate if necessary. | 390 // Downsamples to the codec rate if necessary. |
400 PushResampler<int16_t> input_resampler_; | 391 PushResampler<int16_t> input_resampler_; |
401 uint32_t _timeStamp RTC_ACCESS_ON(encoder_queue_); | 392 uint32_t _timeStamp RTC_ACCESS_ON(encoder_queue_); |
402 | 393 |
403 RemoteNtpTimeEstimator ntp_estimator_ RTC_GUARDED_BY(ts_stats_lock_); | 394 RemoteNtpTimeEstimator ntp_estimator_ RTC_GUARDED_BY(ts_stats_lock_); |
404 | 395 |
405 // Timestamp of the audio pulled from NetEq. | 396 // Timestamp of the audio pulled from NetEq. |
406 rtc::Optional<uint32_t> jitter_buffer_playout_timestamp_; | 397 rtc::Optional<uint32_t> jitter_buffer_playout_timestamp_; |
407 | 398 |
408 rtc::CriticalSection video_sync_lock_; | 399 rtc::CriticalSection video_sync_lock_; |
409 uint32_t playout_timestamp_rtp_ RTC_GUARDED_BY(video_sync_lock_); | 400 uint32_t playout_timestamp_rtp_ RTC_GUARDED_BY(video_sync_lock_); |
410 uint32_t playout_delay_ms_ RTC_GUARDED_BY(video_sync_lock_); | 401 uint32_t playout_delay_ms_ RTC_GUARDED_BY(video_sync_lock_); |
411 uint16_t send_sequence_number_; | 402 uint16_t send_sequence_number_; |
412 | 403 |
413 rtc::CriticalSection ts_stats_lock_; | 404 rtc::CriticalSection ts_stats_lock_; |
414 | 405 |
415 std::unique_ptr<rtc::TimestampWrapAroundHandler> rtp_ts_wraparound_handler_; | 406 std::unique_ptr<rtc::TimestampWrapAroundHandler> rtp_ts_wraparound_handler_; |
416 // The rtp timestamp of the first played out audio frame. | 407 // The rtp timestamp of the first played out audio frame. |
417 int64_t capture_start_rtp_time_stamp_; | 408 int64_t capture_start_rtp_time_stamp_; |
418 // The capture ntp time (in local timebase) of the first played out audio | 409 // The capture ntp time (in local timebase) of the first played out audio |
419 // frame. | 410 // frame. |
420 int64_t capture_start_ntp_time_ms_ RTC_GUARDED_BY(ts_stats_lock_); | 411 int64_t capture_start_ntp_time_ms_ RTC_GUARDED_BY(ts_stats_lock_); |
421 | 412 |
422 // uses | 413 // uses |
423 Statistics* _engineStatisticsPtr; | |
424 ProcessThread* _moduleProcessThreadPtr; | 414 ProcessThread* _moduleProcessThreadPtr; |
425 AudioDeviceModule* _audioDeviceModulePtr; | 415 AudioDeviceModule* _audioDeviceModulePtr; |
426 rtc::CriticalSection* _callbackCritSectPtr; // owned by base | |
427 Transport* _transportPtr; // WebRtc socket or external transport | 416 Transport* _transportPtr; // WebRtc socket or external transport |
428 RmsLevel rms_level_ RTC_ACCESS_ON(encoder_queue_); | 417 RmsLevel rms_level_ RTC_ACCESS_ON(encoder_queue_); |
429 bool input_mute_ RTC_GUARDED_BY(volume_settings_critsect_); | 418 bool input_mute_ RTC_GUARDED_BY(volume_settings_critsect_); |
430 bool previous_frame_muted_ RTC_ACCESS_ON(encoder_queue_); | 419 bool previous_frame_muted_ RTC_ACCESS_ON(encoder_queue_); |
431 float _outputGain RTC_GUARDED_BY(volume_settings_critsect_); | 420 float _outputGain RTC_GUARDED_BY(volume_settings_critsect_); |
432 // VoeRTP_RTCP | 421 // VoeRTP_RTCP |
433 // TODO(henrika): can today be accessed on the main thread and on the | 422 // TODO(henrika): can today be accessed on the main thread and on the |
434 // task queue; hence potential race. | 423 // task queue; hence potential race. |
435 bool _includeAudioLevelIndication; | 424 bool _includeAudioLevelIndication; |
436 size_t transport_overhead_per_packet_ | 425 size_t transport_overhead_per_packet_ |
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465 | 454 |
466 bool encoder_queue_is_active_ RTC_GUARDED_BY(encoder_queue_lock_) = false; | 455 bool encoder_queue_is_active_ RTC_GUARDED_BY(encoder_queue_lock_) = false; |
467 | 456 |
468 rtc::TaskQueue* encoder_queue_ = nullptr; | 457 rtc::TaskQueue* encoder_queue_ = nullptr; |
469 }; | 458 }; |
470 | 459 |
471 } // namespace voe | 460 } // namespace voe |
472 } // namespace webrtc | 461 } // namespace webrtc |
473 | 462 |
474 #endif // VOICE_ENGINE_CHANNEL_H_ | 463 #endif // VOICE_ENGINE_CHANNEL_H_ |
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