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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/call/audio_sink.h" | 17 #include "webrtc/api/call/audio_sink.h" |
17 #include "webrtc/base/criticalsection.h" | 18 #include "webrtc/base/criticalsection.h" |
18 #include "webrtc/base/optional.h" | 19 #include "webrtc/base/optional.h" |
19 #include "webrtc/common_audio/resampler/include/push_resampler.h" | 20 #include "webrtc/common_audio/resampler/include/push_resampler.h" |
20 #include "webrtc/common_types.h" | 21 #include "webrtc/common_types.h" |
21 #include "webrtc/modules/audio_coding/acm2/codec_manager.h" | 22 #include "webrtc/modules/audio_coding/acm2/codec_manager.h" |
22 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" | 23 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h" |
23 #include "webrtc/modules/audio_coding/include/audio_coding_module.h" | 24 #include "webrtc/modules/audio_coding/include/audio_coding_module.h" |
24 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_d
efines.h" | 25 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_d
efines.h" |
25 #include "webrtc/modules/audio_processing/rms_level.h" | 26 #include "webrtc/modules/audio_processing/rms_level.h" |
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370 size_t len, | 371 size_t len, |
371 const PacketOptions& packet_options) override; | 372 const PacketOptions& packet_options) override; |
372 bool SendRtcp(const uint8_t* data, size_t len) override; | 373 bool SendRtcp(const uint8_t* data, size_t len) override; |
373 | 374 |
374 // From MixerParticipant | 375 // From MixerParticipant |
375 MixerParticipant::AudioFrameInfo GetAudioFrameWithMuted( | 376 MixerParticipant::AudioFrameInfo GetAudioFrameWithMuted( |
376 int32_t id, | 377 int32_t id, |
377 AudioFrame* audioFrame) override; | 378 AudioFrame* audioFrame) override; |
378 int32_t NeededFrequency(int32_t id) const override; | 379 int32_t NeededFrequency(int32_t id) const override; |
379 | 380 |
| 381 // From AudioMixer::Source. |
| 382 AudioMixer::Source::AudioFrameWithInfo GetAudioFrameWithInfo( |
| 383 int sample_rate_hz); |
| 384 |
380 // From FileCallback | 385 // From FileCallback |
381 void PlayNotification(int32_t id, uint32_t durationMs) override; | 386 void PlayNotification(int32_t id, uint32_t durationMs) override; |
382 void RecordNotification(int32_t id, uint32_t durationMs) override; | 387 void RecordNotification(int32_t id, uint32_t durationMs) override; |
383 void PlayFileEnded(int32_t id) override; | 388 void PlayFileEnded(int32_t id) override; |
384 void RecordFileEnded(int32_t id) override; | 389 void RecordFileEnded(int32_t id) override; |
385 | 390 |
386 uint32_t InstanceId() const { return _instanceId; } | 391 uint32_t InstanceId() const { return _instanceId; } |
387 int32_t ChannelId() const { return _channelId; } | 392 int32_t ChannelId() const { return _channelId; } |
388 bool Playing() const { return channel_state_.Get().playing; } | 393 bool Playing() const { return channel_state_.Get().playing; } |
389 bool Sending() const { return channel_state_.Get().sending; } | 394 bool Sending() const { return channel_state_.Get().sending; } |
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463 std::unique_ptr<RtpReceiver> rtp_receiver_; | 468 std::unique_ptr<RtpReceiver> rtp_receiver_; |
464 TelephoneEventHandler* telephone_event_handler_; | 469 TelephoneEventHandler* telephone_event_handler_; |
465 std::unique_ptr<RtpRtcp> _rtpRtcpModule; | 470 std::unique_ptr<RtpRtcp> _rtpRtcpModule; |
466 std::unique_ptr<AudioCodingModule> audio_coding_; | 471 std::unique_ptr<AudioCodingModule> audio_coding_; |
467 acm2::CodecManager codec_manager_; | 472 acm2::CodecManager codec_manager_; |
468 acm2::RentACodec rent_a_codec_; | 473 acm2::RentACodec rent_a_codec_; |
469 std::unique_ptr<AudioSinkInterface> audio_sink_; | 474 std::unique_ptr<AudioSinkInterface> audio_sink_; |
470 AudioLevel _outputAudioLevel; | 475 AudioLevel _outputAudioLevel; |
471 bool _externalTransport; | 476 bool _externalTransport; |
472 AudioFrame _audioFrame; | 477 AudioFrame _audioFrame; |
| 478 AudioFrame mix_audio_frame_; |
473 // Downsamples to the codec rate if necessary. | 479 // Downsamples to the codec rate if necessary. |
474 PushResampler<int16_t> input_resampler_; | 480 PushResampler<int16_t> input_resampler_; |
475 std::unique_ptr<FilePlayer> input_file_player_; | 481 std::unique_ptr<FilePlayer> input_file_player_; |
476 std::unique_ptr<FilePlayer> output_file_player_; | 482 std::unique_ptr<FilePlayer> output_file_player_; |
477 std::unique_ptr<FileRecorder> output_file_recorder_; | 483 std::unique_ptr<FileRecorder> output_file_recorder_; |
478 int _inputFilePlayerId; | 484 int _inputFilePlayerId; |
479 int _outputFilePlayerId; | 485 int _outputFilePlayerId; |
480 int _outputFileRecorderId; | 486 int _outputFileRecorderId; |
481 bool _outputFileRecording; | 487 bool _outputFileRecording; |
482 bool _outputExternalMedia; | 488 bool _outputExternalMedia; |
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548 std::unique_ptr<RateLimiter> retransmission_rate_limiter_; | 554 std::unique_ptr<RateLimiter> retransmission_rate_limiter_; |
549 | 555 |
550 // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed. | 556 // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed. |
551 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; | 557 rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_; |
552 }; | 558 }; |
553 | 559 |
554 } // namespace voe | 560 } // namespace voe |
555 } // namespace webrtc | 561 } // namespace webrtc |
556 | 562 |
557 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ | 563 #endif // WEBRTC_VOICE_ENGINE_CHANNEL_H_ |
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