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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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25 #include "modules/audio_conference_mixer/include/audio_conference_mixer_defines.
h" | 25 #include "modules/audio_conference_mixer/include/audio_conference_mixer_defines.
h" |
26 #include "modules/audio_processing/rms_level.h" | 26 #include "modules/audio_processing/rms_level.h" |
27 #include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h" | 27 #include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h" |
28 #include "modules/rtp_rtcp/include/rtp_header_parser.h" | 28 #include "modules/rtp_rtcp/include/rtp_header_parser.h" |
29 #include "modules/rtp_rtcp/include/rtp_receiver.h" | 29 #include "modules/rtp_rtcp/include/rtp_receiver.h" |
30 #include "modules/rtp_rtcp/include/rtp_rtcp.h" | 30 #include "modules/rtp_rtcp/include/rtp_rtcp.h" |
31 #include "rtc_base/criticalsection.h" | 31 #include "rtc_base/criticalsection.h" |
32 #include "rtc_base/event.h" | 32 #include "rtc_base/event.h" |
33 #include "rtc_base/thread_checker.h" | 33 #include "rtc_base/thread_checker.h" |
34 #include "voice_engine/audio_level.h" | 34 #include "voice_engine/audio_level.h" |
35 #include "voice_engine/file_player.h" | |
36 #include "voice_engine/file_recorder.h" | |
37 #include "voice_engine/include/voe_base.h" | 35 #include "voice_engine/include/voe_base.h" |
38 #include "voice_engine/include/voe_network.h" | 36 #include "voice_engine/include/voe_network.h" |
39 #include "voice_engine/shared_data.h" | 37 #include "voice_engine/shared_data.h" |
40 #include "voice_engine/voice_engine_defines.h" | 38 #include "voice_engine/voice_engine_defines.h" |
41 | 39 |
42 namespace rtc { | 40 namespace rtc { |
43 class TimestampWrapAroundHandler; | 41 class TimestampWrapAroundHandler; |
44 } | 42 } |
45 | 43 |
46 namespace webrtc { | 44 namespace webrtc { |
47 | 45 |
48 class AudioDeviceModule; | 46 class AudioDeviceModule; |
49 class FileWrapper; | |
50 class PacketRouter; | 47 class PacketRouter; |
51 class ProcessThread; | 48 class ProcessThread; |
52 class RateLimiter; | 49 class RateLimiter; |
53 class ReceiveStatistics; | 50 class ReceiveStatistics; |
54 class RemoteNtpTimeEstimator; | 51 class RemoteNtpTimeEstimator; |
55 class RtcEventLog; | 52 class RtcEventLog; |
56 class RTPPayloadRegistry; | 53 class RTPPayloadRegistry; |
57 class RTPReceiverAudio; | 54 class RTPReceiverAudio; |
58 class RtpPacketReceived; | 55 class RtpPacketReceived; |
59 class RtpRtcp; | 56 class RtpRtcp; |
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78 class VoERtcpObserver; | 75 class VoERtcpObserver; |
79 | 76 |
80 // Helper class to simplify locking scheme for members that are accessed from | 77 // Helper class to simplify locking scheme for members that are accessed from |
81 // multiple threads. | 78 // multiple threads. |
82 // Example: a member can be set on thread T1 and read by an internal audio | 79 // Example: a member can be set on thread T1 and read by an internal audio |
83 // thread T2. Accessing the member via this class ensures that we are | 80 // thread T2. Accessing the member via this class ensures that we are |
84 // safe and also avoid TSan v2 warnings. | 81 // safe and also avoid TSan v2 warnings. |
85 class ChannelState { | 82 class ChannelState { |
86 public: | 83 public: |
87 struct State { | 84 struct State { |
88 bool output_file_playing = false; | |
89 bool input_file_playing = false; | |
90 bool playing = false; | 85 bool playing = false; |
91 bool sending = false; | 86 bool sending = false; |
92 }; | 87 }; |
93 | 88 |
94 ChannelState() {} | 89 ChannelState() {} |
95 virtual ~ChannelState() {} | 90 virtual ~ChannelState() {} |
96 | 91 |
97 void Reset() { | 92 void Reset() { |
98 rtc::CritScope lock(&lock_); | 93 rtc::CritScope lock(&lock_); |
99 state_ = State(); | 94 state_ = State(); |
100 } | 95 } |
101 | 96 |
102 State Get() const { | 97 State Get() const { |
103 rtc::CritScope lock(&lock_); | 98 rtc::CritScope lock(&lock_); |
104 return state_; | 99 return state_; |
105 } | 100 } |
106 | 101 |
107 void SetOutputFilePlaying(bool enable) { | |
108 rtc::CritScope lock(&lock_); | |
109 state_.output_file_playing = enable; | |
110 } | |
111 | |
112 void SetInputFilePlaying(bool enable) { | |
113 rtc::CritScope lock(&lock_); | |
114 state_.input_file_playing = enable; | |
115 } | |
116 | |
117 void SetPlaying(bool enable) { | 102 void SetPlaying(bool enable) { |
118 rtc::CritScope lock(&lock_); | 103 rtc::CritScope lock(&lock_); |
119 state_.playing = enable; | 104 state_.playing = enable; |
120 } | 105 } |
121 | 106 |
122 void SetSending(bool enable) { | 107 void SetSending(bool enable) { |
123 rtc::CritScope lock(&lock_); | 108 rtc::CritScope lock(&lock_); |
124 state_.sending = enable; | 109 state_.sending = enable; |
125 } | 110 } |
126 | 111 |
127 private: | 112 private: |
128 rtc::CriticalSection lock_; | 113 rtc::CriticalSection lock_; |
129 State state_; | 114 State state_; |
130 }; | 115 }; |
131 | 116 |
132 class Channel | 117 class Channel |
133 : public RtpData, | 118 : public RtpData, |
134 public RtpFeedback, | 119 public RtpFeedback, |
135 public FileCallback, // receiving notification from file player & | |
136 // recorder | |
137 public Transport, | 120 public Transport, |
138 public AudioPacketizationCallback, // receive encoded packets from the | 121 public AudioPacketizationCallback, // receive encoded packets from the |
139 // ACM | 122 // ACM |
140 public MixerParticipant, // supplies output mixer with audio frames | 123 public MixerParticipant, // supplies output mixer with audio frames |
141 public OverheadObserver { | 124 public OverheadObserver { |
142 public: | 125 public: |
143 friend class VoERtcpObserver; | 126 friend class VoERtcpObserver; |
144 | 127 |
145 enum { KNumSocketThreads = 1 }; | 128 enum { KNumSocketThreads = 1 }; |
146 enum { KNumberOfSocketBuffers = 8 }; | 129 enum { KNumberOfSocketBuffers = 8 }; |
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210 // VoENetwork | 193 // VoENetwork |
211 int32_t RegisterExternalTransport(Transport* transport); | 194 int32_t RegisterExternalTransport(Transport* transport); |
212 int32_t DeRegisterExternalTransport(); | 195 int32_t DeRegisterExternalTransport(); |
213 int32_t ReceivedRTPPacket(const uint8_t* received_packet, | 196 int32_t ReceivedRTPPacket(const uint8_t* received_packet, |
214 size_t length, | 197 size_t length, |
215 const PacketTime& packet_time); | 198 const PacketTime& packet_time); |
216 // TODO(nisse, solenberg): Delete when VoENetwork is deleted. | 199 // TODO(nisse, solenberg): Delete when VoENetwork is deleted. |
217 int32_t ReceivedRTCPPacket(const uint8_t* data, size_t length); | 200 int32_t ReceivedRTCPPacket(const uint8_t* data, size_t length); |
218 void OnRtpPacket(const RtpPacketReceived& packet); | 201 void OnRtpPacket(const RtpPacketReceived& packet); |
219 | 202 |
220 // VoEFile | |
221 int StartPlayingFileLocally(const char* fileName, | |
222 bool loop, | |
223 FileFormats format, | |
224 int startPosition, | |
225 float volumeScaling, | |
226 int stopPosition, | |
227 const CodecInst* codecInst); | |
228 int StartPlayingFileLocally(InStream* stream, | |
229 FileFormats format, | |
230 int startPosition, | |
231 float volumeScaling, | |
232 int stopPosition, | |
233 const CodecInst* codecInst); | |
234 int StopPlayingFileLocally(); | |
235 int IsPlayingFileLocally() const; | |
236 int RegisterFilePlayingToMixer(); | |
237 int StartPlayingFileAsMicrophone(const char* fileName, | |
238 bool loop, | |
239 FileFormats format, | |
240 int startPosition, | |
241 float volumeScaling, | |
242 int stopPosition, | |
243 const CodecInst* codecInst); | |
244 int StartPlayingFileAsMicrophone(InStream* stream, | |
245 FileFormats format, | |
246 int startPosition, | |
247 float volumeScaling, | |
248 int stopPosition, | |
249 const CodecInst* codecInst); | |
250 int StopPlayingFileAsMicrophone(); | |
251 int IsPlayingFileAsMicrophone() const; | |
252 int StartRecordingPlayout(const char* fileName, const CodecInst* codecInst); | |
253 int StartRecordingPlayout(OutStream* stream, const CodecInst* codecInst); | |
254 int StopRecordingPlayout(); | |
255 | |
256 void SetMixWithMicStatus(bool mix); | |
257 | |
258 // Muting, Volume and Level. | 203 // Muting, Volume and Level. |
259 void SetInputMute(bool enable); | 204 void SetInputMute(bool enable); |
260 void SetChannelOutputVolumeScaling(float scaling); | 205 void SetChannelOutputVolumeScaling(float scaling); |
261 int GetSpeechOutputLevel() const; | 206 int GetSpeechOutputLevel() const; |
262 int GetSpeechOutputLevelFullRange() const; | 207 int GetSpeechOutputLevelFullRange() const; |
263 // See description of "totalAudioEnergy" in the WebRTC stats spec: | 208 // See description of "totalAudioEnergy" in the WebRTC stats spec: |
264 // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudio
energy | 209 // https://w3c.github.io/webrtc-stats/#dom-rtcmediastreamtrackstats-totalaudio
energy |
265 double GetTotalOutputEnergy() const; | 210 double GetTotalOutputEnergy() const; |
266 double GetTotalOutputDuration() const; | 211 double GetTotalOutputDuration() const; |
267 | 212 |
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341 MixerParticipant::AudioFrameInfo GetAudioFrameWithMuted( | 286 MixerParticipant::AudioFrameInfo GetAudioFrameWithMuted( |
342 int32_t id, | 287 int32_t id, |
343 AudioFrame* audioFrame) override; | 288 AudioFrame* audioFrame) override; |
344 int32_t NeededFrequency(int32_t id) const override; | 289 int32_t NeededFrequency(int32_t id) const override; |
345 | 290 |
346 // From AudioMixer::Source. | 291 // From AudioMixer::Source. |
347 AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo( | 292 AudioMixer::Source::AudioFrameInfo GetAudioFrameWithInfo( |
348 int sample_rate_hz, | 293 int sample_rate_hz, |
349 AudioFrame* audio_frame); | 294 AudioFrame* audio_frame); |
350 | 295 |
351 // From FileCallback | |
352 void PlayNotification(int32_t id, uint32_t durationMs) override; | |
353 void RecordNotification(int32_t id, uint32_t durationMs) override; | |
354 void PlayFileEnded(int32_t id) override; | |
355 void RecordFileEnded(int32_t id) override; | |
356 | |
357 uint32_t InstanceId() const { return _instanceId; } | 296 uint32_t InstanceId() const { return _instanceId; } |
358 int32_t ChannelId() const { return _channelId; } | 297 int32_t ChannelId() const { return _channelId; } |
359 bool Playing() const { return channel_state_.Get().playing; } | 298 bool Playing() const { return channel_state_.Get().playing; } |
360 bool Sending() const { return channel_state_.Get().sending; } | 299 bool Sending() const { return channel_state_.Get().sending; } |
361 bool ExternalTransport() const { | 300 bool ExternalTransport() const { |
362 rtc::CritScope cs(&_callbackCritSect); | 301 rtc::CritScope cs(&_callbackCritSect); |
363 return _externalTransport; | 302 return _externalTransport; |
364 } | 303 } |
365 RtpRtcp* RtpRtcpModulePtr() const { return _rtpRtcpModule.get(); } | 304 RtpRtcp* RtpRtcpModulePtr() const { return _rtpRtcpModule.get(); } |
366 int8_t OutputEnergyLevel() const { return _outputAudioLevel.Level(); } | 305 int8_t OutputEnergyLevel() const { return _outputAudioLevel.Level(); } |
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423 RTPHeader *header); | 362 RTPHeader *header); |
424 bool OnRecoveredPacket(const uint8_t* packet, size_t packet_length); | 363 bool OnRecoveredPacket(const uint8_t* packet, size_t packet_length); |
425 | 364 |
426 bool ReceivePacket(const uint8_t* packet, | 365 bool ReceivePacket(const uint8_t* packet, |
427 size_t packet_length, | 366 size_t packet_length, |
428 const RTPHeader& header, | 367 const RTPHeader& header, |
429 bool in_order); | 368 bool in_order); |
430 bool IsPacketInOrder(const RTPHeader& header) const; | 369 bool IsPacketInOrder(const RTPHeader& header) const; |
431 bool IsPacketRetransmitted(const RTPHeader& header, bool in_order) const; | 370 bool IsPacketRetransmitted(const RTPHeader& header, bool in_order) const; |
432 int ResendPackets(const uint16_t* sequence_numbers, int length); | 371 int ResendPackets(const uint16_t* sequence_numbers, int length); |
433 int32_t MixOrReplaceAudioWithFile(AudioFrame* audio_frame); | |
434 int32_t MixAudioWithFile(AudioFrame& audioFrame, int mixingFrequency); | |
435 void UpdatePlayoutTimestamp(bool rtcp); | 372 void UpdatePlayoutTimestamp(bool rtcp); |
436 void RegisterReceiveCodecsToRTPModule(); | 373 void RegisterReceiveCodecsToRTPModule(); |
437 | 374 |
438 int SetSendRtpHeaderExtension(bool enable, | 375 int SetSendRtpHeaderExtension(bool enable, |
439 RTPExtensionType type, | 376 RTPExtensionType type, |
440 unsigned char id); | 377 unsigned char id); |
441 | 378 |
442 void UpdateOverheadForEncoder() | 379 void UpdateOverheadForEncoder() |
443 RTC_EXCLUSIVE_LOCKS_REQUIRED(overhead_per_packet_lock_); | 380 RTC_EXCLUSIVE_LOCKS_REQUIRED(overhead_per_packet_lock_); |
444 | 381 |
445 int GetRtpTimestampRateHz() const; | 382 int GetRtpTimestampRateHz() const; |
446 int64_t GetRTT(bool allow_associate_channel) const; | 383 int64_t GetRTT(bool allow_associate_channel) const; |
447 | 384 |
448 // Called on the encoder task queue when a new input audio frame is ready | 385 // Called on the encoder task queue when a new input audio frame is ready |
449 // for encoding. | 386 // for encoding. |
450 void ProcessAndEncodeAudioOnTaskQueue(AudioFrame* audio_input); | 387 void ProcessAndEncodeAudioOnTaskQueue(AudioFrame* audio_input); |
451 | 388 |
452 uint32_t _instanceId; | 389 uint32_t _instanceId; |
453 int32_t _channelId; | 390 int32_t _channelId; |
454 | 391 |
455 rtc::CriticalSection _fileCritSect; | |
456 rtc::CriticalSection _callbackCritSect; | 392 rtc::CriticalSection _callbackCritSect; |
457 rtc::CriticalSection volume_settings_critsect_; | 393 rtc::CriticalSection volume_settings_critsect_; |
458 | 394 |
459 ChannelState channel_state_; | 395 ChannelState channel_state_; |
460 | 396 |
461 std::unique_ptr<voe::RtcEventLogProxy> event_log_proxy_; | 397 std::unique_ptr<voe::RtcEventLogProxy> event_log_proxy_; |
462 std::unique_ptr<voe::RtcpRttStatsProxy> rtcp_rtt_stats_proxy_; | 398 std::unique_ptr<voe::RtcpRttStatsProxy> rtcp_rtt_stats_proxy_; |
463 | 399 |
464 std::unique_ptr<RtpHeaderParser> rtp_header_parser_; | 400 std::unique_ptr<RtpHeaderParser> rtp_header_parser_; |
465 std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry_; | 401 std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry_; |
466 std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_; | 402 std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_; |
467 std::unique_ptr<RtpReceiver> rtp_receiver_; | 403 std::unique_ptr<RtpReceiver> rtp_receiver_; |
468 TelephoneEventHandler* telephone_event_handler_; | 404 TelephoneEventHandler* telephone_event_handler_; |
469 std::unique_ptr<RtpRtcp> _rtpRtcpModule; | 405 std::unique_ptr<RtpRtcp> _rtpRtcpModule; |
470 std::unique_ptr<AudioCodingModule> audio_coding_; | 406 std::unique_ptr<AudioCodingModule> audio_coding_; |
471 acm2::CodecManager codec_manager_; | 407 acm2::CodecManager codec_manager_; |
472 acm2::RentACodec rent_a_codec_; | 408 acm2::RentACodec rent_a_codec_; |
473 std::unique_ptr<AudioSinkInterface> audio_sink_; | 409 std::unique_ptr<AudioSinkInterface> audio_sink_; |
474 AudioLevel _outputAudioLevel; | 410 AudioLevel _outputAudioLevel; |
475 bool _externalTransport; | 411 bool _externalTransport; |
476 // Downsamples to the codec rate if necessary. | 412 // Downsamples to the codec rate if necessary. |
477 PushResampler<int16_t> input_resampler_; | 413 PushResampler<int16_t> input_resampler_; |
478 std::unique_ptr<FilePlayer> input_file_player_; | |
479 std::unique_ptr<FilePlayer> output_file_player_; | |
480 std::unique_ptr<FileRecorder> output_file_recorder_; | |
481 int _inputFilePlayerId; | |
482 int _outputFilePlayerId; | |
483 int _outputFileRecorderId; | |
484 bool _outputFileRecording; | |
485 uint32_t _timeStamp RTC_ACCESS_ON(encoder_queue_); | 414 uint32_t _timeStamp RTC_ACCESS_ON(encoder_queue_); |
486 | 415 |
487 RemoteNtpTimeEstimator ntp_estimator_ RTC_GUARDED_BY(ts_stats_lock_); | 416 RemoteNtpTimeEstimator ntp_estimator_ RTC_GUARDED_BY(ts_stats_lock_); |
488 | 417 |
489 // Timestamp of the audio pulled from NetEq. | 418 // Timestamp of the audio pulled from NetEq. |
490 rtc::Optional<uint32_t> jitter_buffer_playout_timestamp_; | 419 rtc::Optional<uint32_t> jitter_buffer_playout_timestamp_; |
491 | 420 |
492 rtc::CriticalSection video_sync_lock_; | 421 rtc::CriticalSection video_sync_lock_; |
493 uint32_t playout_timestamp_rtp_ RTC_GUARDED_BY(video_sync_lock_); | 422 uint32_t playout_timestamp_rtp_ RTC_GUARDED_BY(video_sync_lock_); |
494 uint32_t playout_delay_ms_ RTC_GUARDED_BY(video_sync_lock_); | 423 uint32_t playout_delay_ms_ RTC_GUARDED_BY(video_sync_lock_); |
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508 OutputMixer* _outputMixerPtr; | 437 OutputMixer* _outputMixerPtr; |
509 ProcessThread* _moduleProcessThreadPtr; | 438 ProcessThread* _moduleProcessThreadPtr; |
510 AudioDeviceModule* _audioDeviceModulePtr; | 439 AudioDeviceModule* _audioDeviceModulePtr; |
511 VoiceEngineObserver* _voiceEngineObserverPtr; // owned by base | 440 VoiceEngineObserver* _voiceEngineObserverPtr; // owned by base |
512 rtc::CriticalSection* _callbackCritSectPtr; // owned by base | 441 rtc::CriticalSection* _callbackCritSectPtr; // owned by base |
513 Transport* _transportPtr; // WebRtc socket or external transport | 442 Transport* _transportPtr; // WebRtc socket or external transport |
514 RmsLevel rms_level_ RTC_ACCESS_ON(encoder_queue_); | 443 RmsLevel rms_level_ RTC_ACCESS_ON(encoder_queue_); |
515 bool input_mute_ RTC_GUARDED_BY(volume_settings_critsect_); | 444 bool input_mute_ RTC_GUARDED_BY(volume_settings_critsect_); |
516 bool previous_frame_muted_ RTC_ACCESS_ON(encoder_queue_); | 445 bool previous_frame_muted_ RTC_ACCESS_ON(encoder_queue_); |
517 float _outputGain RTC_GUARDED_BY(volume_settings_critsect_); | 446 float _outputGain RTC_GUARDED_BY(volume_settings_critsect_); |
518 // VoEBase | |
519 bool _mixFileWithMicrophone; | |
520 // VoeRTP_RTCP | 447 // VoeRTP_RTCP |
521 // TODO(henrika): can today be accessed on the main thread and on the | 448 // TODO(henrika): can today be accessed on the main thread and on the |
522 // task queue; hence potential race. | 449 // task queue; hence potential race. |
523 bool _includeAudioLevelIndication; | 450 bool _includeAudioLevelIndication; |
524 size_t transport_overhead_per_packet_ | 451 size_t transport_overhead_per_packet_ |
525 RTC_GUARDED_BY(overhead_per_packet_lock_); | 452 RTC_GUARDED_BY(overhead_per_packet_lock_); |
526 size_t rtp_overhead_per_packet_ RTC_GUARDED_BY(overhead_per_packet_lock_); | 453 size_t rtp_overhead_per_packet_ RTC_GUARDED_BY(overhead_per_packet_lock_); |
527 rtc::CriticalSection overhead_per_packet_lock_; | 454 rtc::CriticalSection overhead_per_packet_lock_; |
528 // VoENetwork | 455 // VoENetwork |
529 AudioFrame::SpeechType _outputSpeechType; | 456 AudioFrame::SpeechType _outputSpeechType; |
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553 | 480 |
554 bool encoder_queue_is_active_ RTC_GUARDED_BY(encoder_queue_lock_) = false; | 481 bool encoder_queue_is_active_ RTC_GUARDED_BY(encoder_queue_lock_) = false; |
555 | 482 |
556 rtc::TaskQueue* encoder_queue_ = nullptr; | 483 rtc::TaskQueue* encoder_queue_ = nullptr; |
557 }; | 484 }; |
558 | 485 |
559 } // namespace voe | 486 } // namespace voe |
560 } // namespace webrtc | 487 } // namespace webrtc |
561 | 488 |
562 #endif // VOICE_ENGINE_CHANNEL_H_ | 489 #endif // VOICE_ENGINE_CHANNEL_H_ |
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