chromium/third_party/webrtc/modules/rtp_rtcp/source/rtcp_transceiver_config.h

/*
 *  Copyright (c) 2017 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 MODULES_RTP_RTCP_SOURCE_RTCP_TRANSCEIVER_CONFIG_H_
#define MODULES_RTP_RTCP_SOURCE_RTCP_TRANSCEIVER_CONFIG_H_

#include <string>

#include "api/array_view.h"
#include "api/rtp_headers.h"
#include "api/task_queue/task_queue_base.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "api/video/video_bitrate_allocation.h"
#include "modules/rtp_rtcp/include/report_block_data.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/ntp_time.h"

namespace webrtc {
class ReceiveStatisticsProvider;

// Interface to watch incoming rtcp packets by media (rtp) receiver.
// All message handlers have default empty implementation. This way users only
// need to implement the ones they are interested in.
class MediaReceiverRtcpObserver { … };

// Handles RTCP related messages for a single RTP stream (i.e. single SSRC)
class RtpStreamRtcpHandler { … };

struct RtcpTransceiverConfig { … };

}  // namespace webrtc

#endif  // MODULES_RTP_RTCP_SOURCE_RTCP_TRANSCEIVER_CONFIG_H_