blob: 60250da89a414b20491445bbfa8b5a0e39313ded [file] [log] [blame]
* 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.
#include <map>
#include <memory>
#include <unordered_set>
#include "audio/audio_transport_impl.h"
#include "audio/null_audio_poller.h"
#include "call/audio_state.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/critical_section.h"
#include "rtc_base/ref_count.h"
#include "rtc_base/thread_checker.h"
namespace webrtc {
class AudioSendStream;
class AudioReceiveStream;
namespace internal {
class AudioState : public webrtc::AudioState {
explicit AudioState(const AudioState::Config& config);
~AudioState() override;
AudioProcessing* audio_processing() override;
AudioTransport* audio_transport() override;
void SetPlayout(bool enabled) override;
void SetRecording(bool enabled) override;
Stats GetAudioInputStats() const override;
void SetStereoChannelSwapping(bool enable) override;
AudioDeviceModule* audio_device_module() {
return config_.audio_device_module.get();
bool typing_noise_detected() const;
void AddReceivingStream(webrtc::AudioReceiveStream* stream);
void RemoveReceivingStream(webrtc::AudioReceiveStream* stream);
void AddSendingStream(webrtc::AudioSendStream* stream,
int sample_rate_hz,
size_t num_channels);
void RemoveSendingStream(webrtc::AudioSendStream* stream);
void UpdateAudioTransportWithSendingStreams();
rtc::ThreadChecker thread_checker_;
rtc::ThreadChecker 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.
std::unique_ptr<NullAudioPoller> null_audio_poller_;
std::unordered_set<webrtc::AudioReceiveStream*> 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