blob: a321ec27a57160f670ea5cbe402c8fe0975a3fc7 [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.
*/
#ifndef WEBRTC_VIDEO_AUDIO_RECEIVE_STREAM_H_
#define WEBRTC_VIDEO_AUDIO_RECEIVE_STREAM_H_
#include "webrtc/audio_receive_stream.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
namespace webrtc {
class RemoteBitrateEstimator;
namespace internal {
class AudioReceiveStream : public webrtc::AudioReceiveStream {
public:
AudioReceiveStream(RemoteBitrateEstimator* remote_bitrate_estimator,
const webrtc::AudioReceiveStream::Config& config);
~AudioReceiveStream() override {}
bool DeliverRtcp(const uint8_t* packet, size_t length);
bool DeliverRtp(const uint8_t* packet, size_t length);
const webrtc::AudioReceiveStream::Config& config() const {
return config_;
}
private:
RemoteBitrateEstimator* const remote_bitrate_estimator_;
const webrtc::AudioReceiveStream::Config config_;
rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
};
} // namespace internal
} // namespace webrtc
#endif // WEBRTC_VIDEO_AUDIO_RECEIVE_STREAM_H_