This is accomplished by breaking out RTX and FEC/RED functionality from the RTP module and keeping track of the base payload type, that is the payload type received when not receiving RTX. Enables retransmissions over RTX by default in the loopback test. BUG=1811 TESTS=voe/vie_auto_test --automated and trybots. R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2154004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4692 4adac7df-926f-26a2-2b94-8c16560cd09d
89 lines
2.8 KiB
C++
89 lines
2.8 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:
|
|
RTPReceiverVideo(const int32_t id, 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);
|
|
|
|
int32_t id_;
|
|
};
|
|
} // namespace webrtc
|
|
|
|
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_
|