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/*
* Copyright (c) 2012 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_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_
#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "webrtc/modules/rtp_rtcp/source/bitrate.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/system_wrappers/interface/scoped_ptr.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class RTPReceiverVideo : public RTPReceiverStrategy {
public:
RTPReceiverVideo(RtpData* data_callback);
virtual ~RTPReceiverVideo();
virtual int32_t ParseRtpPacket(
WebRtcRTPHeader* rtp_header,
const PayloadUnion& specific_payload,
bool is_red,
const uint8_t* packet,
uint16_t packet_length,
int64_t timestamp,
bool is_first_packet) OVERRIDE;
TelephoneEventHandler* GetTelephoneEventHandler() {
return NULL;
}
int GetPayloadTypeFrequency() const OVERRIDE;
virtual RTPAliveType ProcessDeadOrAlive(uint16_t last_payload_length) const
OVERRIDE;
virtual bool ShouldReportCsrcChanges(uint8_t payload_type) const OVERRIDE;
virtual int32_t OnNewPayloadTypeCreated(
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
int8_t payload_type,
uint32_t frequency) OVERRIDE;
virtual int32_t InvokeOnInitializeDecoder(
RtpFeedback* callback,
int32_t id,
int8_t payload_type,
const char payload_name[RTP_PAYLOAD_NAME_SIZE],
const PayloadUnion& specific_payload) const OVERRIDE;
void SetPacketOverHead(uint16_t packet_over_head);
protected:
int32_t ReceiveGenericCodec(WebRtcRTPHeader* rtp_header,
const uint8_t* payload_data,
uint16_t payload_data_length);
int32_t ReceiveVp8Codec(WebRtcRTPHeader* rtp_header,
const uint8_t* payload_data,
uint16_t payload_data_length);
int32_t BuildRTPheader(const WebRtcRTPHeader* rtp_header,
uint8_t* data_buffer) const;
private:
int32_t ParseVideoCodecSpecific(
WebRtcRTPHeader* rtp_header,
const uint8_t* payload_data,
uint16_t payload_data_length,
RtpVideoCodecTypes video_type,
int64_t now_ms,
bool is_first_packet);
};
} // namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_
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