1 /* 2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. 3 * 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 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef AUDIO_AUDIO_STATE_H_ 12 #define AUDIO_AUDIO_STATE_H_ 13 14 #include <map> 15 #include <memory> 16 #include <unordered_set> 17 18 #include "audio/audio_transport_impl.h" 19 #include "audio/null_audio_poller.h" 20 #include "call/audio_state.h" 21 #include "rtc_base/constructor_magic.h" 22 #include "rtc_base/ref_count.h" 23 #include "rtc_base/thread_checker.h" 24 25 namespace webrtc { 26 27 class AudioSendStream; 28 class AudioReceiveStream; 29 30 namespace internal { 31 32 class AudioState : public webrtc::AudioState { 33 public: 34 explicit AudioState(const AudioState::Config& config); 35 ~AudioState() override; 36 37 AudioProcessing* audio_processing() override; 38 AudioTransport* audio_transport() override; 39 40 void SetPlayout(bool enabled) override; 41 void SetRecording(bool enabled) override; 42 43 void SetStereoChannelSwapping(bool enable) override; 44 audio_device_module()45 AudioDeviceModule* audio_device_module() { 46 RTC_DCHECK(config_.audio_device_module); 47 return config_.audio_device_module.get(); 48 } 49 50 bool typing_noise_detected() const; 51 52 void AddReceivingStream(webrtc::AudioReceiveStream* stream); 53 void RemoveReceivingStream(webrtc::AudioReceiveStream* stream); 54 55 void AddSendingStream(webrtc::AudioSendStream* stream, 56 int sample_rate_hz, 57 size_t num_channels); 58 void RemoveSendingStream(webrtc::AudioSendStream* stream); 59 60 private: 61 void UpdateAudioTransportWithSendingStreams(); 62 void UpdateNullAudioPollerState(); 63 64 rtc::ThreadChecker thread_checker_; 65 rtc::ThreadChecker process_thread_checker_; 66 const webrtc::AudioState::Config config_; 67 bool recording_enabled_ = true; 68 bool playout_enabled_ = true; 69 70 // Transports mixed audio from the mixer to the audio device and 71 // recorded audio to the sending streams. 72 AudioTransportImpl audio_transport_; 73 74 // Null audio poller is used to continue polling the audio streams if audio 75 // playout is disabled so that audio processing still happens and the audio 76 // stats are still updated. 77 std::unique_ptr<NullAudioPoller> null_audio_poller_; 78 79 std::unordered_set<webrtc::AudioReceiveStream*> receiving_streams_; 80 struct StreamProperties { 81 int sample_rate_hz = 0; 82 size_t num_channels = 0; 83 }; 84 std::map<webrtc::AudioSendStream*, StreamProperties> sending_streams_; 85 86 RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioState); 87 }; 88 } // namespace internal 89 } // namespace webrtc 90 91 #endif // AUDIO_AUDIO_STATE_H_ 92