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/*
* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef AUDIO_AUDIO_STATE_H_
#define AUDIO_AUDIO_STATE_H_
#include <map>
#include <memory>
#include "api/sequence_checker.h"
#include "audio/audio_transport_impl.h"
#include "call/audio_state.h"
#include "rtc_base/containers/flat_set.h"
#include "rtc_base/ref_count.h"
#include "rtc_base/task_utils/repeating_task.h"
#include "rtc_base/thread_annotations.h"
namespace webrtc {
class AudioSendStream;
class AudioReceiveStreamInterface;
namespace internal {
class AudioState : public webrtc::AudioState {
public:
explicit AudioState(const AudioState::Config& config);
AudioState() = delete;
AudioState(const AudioState&) = delete;
AudioState& operator=(const AudioState&) = delete;
~AudioState() override;
AudioProcessing* audio_processing() override;
AudioTransport* audio_transport() override;
void SetPlayout(bool enabled) override;
void SetRecording(bool enabled) override;
void SetStereoChannelSwapping(bool enable) override;
AudioDeviceModule* audio_device_module() {
RTC_DCHECK(config_.audio_device_module);
return config_.audio_device_module.get();
}
void AddReceivingStream(webrtc::AudioReceiveStreamInterface* stream);
void RemoveReceivingStream(webrtc::AudioReceiveStreamInterface* stream);
void AddSendingStream(webrtc::AudioSendStream* stream,
int sample_rate_hz,
size_t num_channels);
void RemoveSendingStream(webrtc::AudioSendStream* stream);
private:
void UpdateAudioTransportWithSendingStreams();
void UpdateNullAudioPollerState() RTC_RUN_ON(&thread_checker_);
SequenceChecker thread_checker_;
SequenceChecker process_thread_checker_;
const webrtc::AudioState::Config config_;
bool recording_enabled_ = true;
bool playout_enabled_ = true;
// Transports mixed audio from the mixer to the audio device and
// recorded audio to the sending streams.
AudioTransportImpl audio_transport_;
// Null audio poller is used to continue polling the audio streams if audio
// playout is disabled so that audio processing still happens and the audio
// stats are still updated.
RepeatingTaskHandle null_audio_poller_ RTC_GUARDED_BY(&thread_checker_);
webrtc::flat_set<webrtc::AudioReceiveStreamInterface*> receiving_streams_;
struct StreamProperties {
int sample_rate_hz = 0;
size_t num_channels = 0;
};
std::map<webrtc::AudioSendStream*, StreamProperties> sending_streams_;
};
} // namespace internal
} // namespace webrtc
#endif // AUDIO_AUDIO_STATE_H_
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