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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_RECEIVE_STREAM_H_
#define AUDIO_AUDIO_RECEIVE_STREAM_H_
#include <memory>
#include <vector>
#include "api/audio/audio_mixer.h"
#include "audio/audio_state.h"
#include "call/audio_receive_stream.h"
#include "call/rtp_packet_sink_interface.h"
#include "call/syncable.h"
#include "rtc_base/constructormagic.h"
#include "rtc_base/thread_checker.h"
namespace webrtc {
class PacketRouter;
class RtcEventLog;
class RtpPacketReceived;
class RtpStreamReceiverControllerInterface;
class RtpStreamReceiverInterface;
namespace voe {
class ChannelProxy;
} // namespace voe
namespace internal {
class AudioSendStream;
class AudioReceiveStream final : public webrtc::AudioReceiveStream,
public AudioMixer::Source,
public Syncable {
public:
AudioReceiveStream(RtpStreamReceiverControllerInterface* receiver_controller,
PacketRouter* packet_router,
const webrtc::AudioReceiveStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
webrtc::RtcEventLog* event_log);
~AudioReceiveStream() override;
// webrtc::AudioReceiveStream implementation.
void Start() override;
void Stop() override;
webrtc::AudioReceiveStream::Stats GetStats() const override;
int GetOutputLevel() const override;
void SetSink(std::unique_ptr<AudioSinkInterface> sink) override;
void SetGain(float gain) override;
std::vector<webrtc::RtpSource> GetSources() const override;
// TODO(nisse): We don't formally implement RtpPacketSinkInterface, and this
// method shouldn't be needed. But it's currently used by the
// AudioReceiveStreamTest.ReceiveRtpPacket unittest. Figure out if that test
// shuld be refactored or deleted, and then delete this method.
void OnRtpPacket(const RtpPacketReceived& packet);
// AudioMixer::Source
AudioFrameInfo GetAudioFrameWithInfo(int sample_rate_hz,
AudioFrame* audio_frame) override;
int Ssrc() const override;
int PreferredSampleRate() const override;
// Syncable
int id() const override;
rtc::Optional<Syncable::Info> GetInfo() const override;
uint32_t GetPlayoutTimestamp() const override;
void SetMinimumPlayoutDelay(int delay_ms) override;
void AssociateSendStream(AudioSendStream* send_stream);
void SignalNetworkState(NetworkState state);
bool DeliverRtcp(const uint8_t* packet, size_t length);
const webrtc::AudioReceiveStream::Config& config() const;
private:
VoiceEngine* voice_engine() const;
AudioState* audio_state() const;
int SetVoiceEnginePlayout(bool playout);
rtc::ThreadChecker worker_thread_checker_;
rtc::ThreadChecker module_process_thread_checker_;
const webrtc::AudioReceiveStream::Config config_;
rtc::scoped_refptr<webrtc::AudioState> audio_state_;
std::unique_ptr<voe::ChannelProxy> channel_proxy_;
bool playing_ RTC_ACCESS_ON(worker_thread_checker_) = false;
std::unique_ptr<RtpStreamReceiverInterface> rtp_stream_receiver_;
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream);
};
} // namespace internal
} // namespace webrtc
#endif // AUDIO_AUDIO_RECEIVE_STREAM_H_
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