webrtc_m130/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.h
pbos@webrtc.org b5e6bfc76a Make RTPSender/RTPReceiver generic.
Changes include,
1) Introduce class RtpPacketizerGeneric & RtpDePacketizerGeneric.
2) Introduce class RtpDepacketizerVp8.
3) Make RTPSenderVideo::SendH264 generic and used by all packetizers.
4) Move codec specific functions from RTPSenderVideo/RTPReceiverVideo to
RtpPacketizer/RtpDePacketizer sub-classes.

R=pbos@webrtc.org, stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26399004

Patch from Changbin Shao <changbin.shao@intel.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7163 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-12 11:05:55 +00:00

67 lines
2.3 KiB
C++

/*
* 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:
explicit 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);
private:
int32_t BuildRTPheader(const WebRtcRTPHeader* rtp_header,
uint8_t* data_buffer) const;
};
} // namespace webrtc
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_