tor-browser

The Tor Browser
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audio_state.h (3161B)

      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 <cstddef>
     15 #include <map>
     16 
     17 #include "api/audio/audio_device.h"
     18 #include "api/audio/audio_processing.h"
     19 #include "api/sequence_checker.h"
     20 #include "audio/audio_transport_impl.h"
     21 #include "call/audio_state.h"
     22 #include "rtc_base/checks.h"
     23 #include "rtc_base/containers/flat_set.h"
     24 #include "rtc_base/system/no_unique_address.h"
     25 #include "rtc_base/task_utils/repeating_task.h"
     26 #include "rtc_base/thread_annotations.h"
     27 
     28 namespace webrtc {
     29 
     30 class AudioSendStream;
     31 class AudioReceiveStreamInterface;
     32 
     33 namespace internal {
     34 
     35 class AudioState : public webrtc::AudioState {
     36 public:
     37  explicit AudioState(const AudioState::Config& config);
     38 
     39  AudioState() = delete;
     40  AudioState(const AudioState&) = delete;
     41  AudioState& operator=(const AudioState&) = delete;
     42 
     43  ~AudioState() override;
     44 
     45  AudioProcessing* audio_processing() override;
     46  AudioTransport* audio_transport() override;
     47 
     48  void SetPlayout(bool enabled) override;
     49  void SetRecording(bool enabled) override;
     50 
     51  void SetStereoChannelSwapping(bool enable) override;
     52 
     53  AudioDeviceModule* audio_device_module() {
     54    RTC_DCHECK(config_.audio_device_module);
     55    return config_.audio_device_module.get();
     56  }
     57 
     58  void AddReceivingStream(webrtc::AudioReceiveStreamInterface* stream);
     59  void RemoveReceivingStream(webrtc::AudioReceiveStreamInterface* stream);
     60 
     61  void AddSendingStream(webrtc::AudioSendStream* stream,
     62                        int sample_rate_hz,
     63                        size_t num_channels);
     64  void RemoveSendingStream(webrtc::AudioSendStream* stream);
     65 
     66 private:
     67  void UpdateAudioTransportWithSendingStreams();
     68  void UpdateNullAudioPollerState() RTC_RUN_ON(&thread_checker_);
     69 
     70  RTC_NO_UNIQUE_ADDRESS SequenceChecker thread_checker_{
     71      SequenceChecker::kDetached};
     72  RTC_NO_UNIQUE_ADDRESS SequenceChecker process_thread_checker_{
     73      SequenceChecker::kDetached};
     74  const webrtc::AudioState::Config config_;
     75  bool recording_enabled_ = true;
     76  bool playout_enabled_ = true;
     77 
     78  // Transports mixed audio from the mixer to the audio device and
     79  // recorded audio to the sending streams.
     80  AudioTransportImpl audio_transport_;
     81 
     82  // Null audio poller is used to continue polling the audio streams if audio
     83  // playout is disabled so that audio processing still happens and the audio
     84  // stats are still updated.
     85  RepeatingTaskHandle null_audio_poller_ RTC_GUARDED_BY(&thread_checker_);
     86 
     87  webrtc::flat_set<webrtc::AudioReceiveStreamInterface*> receiving_streams_;
     88  struct StreamProperties {
     89    int sample_rate_hz = 0;
     90    size_t num_channels = 0;
     91  };
     92  std::map<webrtc::AudioSendStream*, StreamProperties> sending_streams_;
     93 };
     94 }  // namespace internal
     95 }  // namespace webrtc
     96 
     97 #endif  // AUDIO_AUDIO_STATE_H_