blob: 24924c9a6d10788883754d54977c3731c4dd6f71 [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 <memory>
#include "webrtc/audio_receive_stream.h"
#include "webrtc/audio_state.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/thread_checker.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
namespace webrtc {
class CongestionController;
class RemoteBitrateEstimator;
class RtcEventLog;
namespace voe {
class ChannelProxy;
} // namespace voe
namespace internal {
class AudioReceiveStream final : public webrtc::AudioReceiveStream {
AudioReceiveStream(CongestionController* congestion_controller,
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;
void SetSink(std::unique_ptr<AudioSinkInterface> sink) override;
void SetGain(float gain) override;
void SignalNetworkState(NetworkState state);
bool DeliverRtcp(const uint8_t* packet, size_t length);
bool DeliverRtp(const uint8_t* packet,
size_t length,
const PacketTime& packet_time);
const webrtc::AudioReceiveStream::Config& config() const;
VoiceEngine* voice_engine() const;
rtc::ThreadChecker thread_checker_;
RemoteBitrateEstimator* remote_bitrate_estimator_ = nullptr;
const webrtc::AudioReceiveStream::Config config_;
rtc::scoped_refptr<webrtc::AudioState> audio_state_;
std::unique_ptr<RtpHeaderParser> rtp_header_parser_;
std::unique_ptr<voe::ChannelProxy> channel_proxy_;
} // namespace internal
} // namespace webrtc