c38eef896a
This is a pure reformat patch, with the exception that I also fixed all comments and moved a constant. I did not change the types in this patch since I though that is more risky, so I'll do that in a separate patch later (perhaps we could purge the types from the whole module in one go?) BUG= TEST=Trybots, vie_ & voe_auto_test --automated Review URL: https://webrtc-codereview.appspot.com/998007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3338 4adac7df-926f-26a2-2b94-8c16560cd09d
121 lines
4.0 KiB
C++
121 lines
4.0 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 "bitrate.h"
|
|
#include "rtp_receiver.h"
|
|
#include "rtp_receiver_strategy.h"
|
|
#include "rtp_rtcp_defines.h"
|
|
#include "rtp_utility.h"
|
|
#include "scoped_ptr.h"
|
|
#include "typedefs.h"
|
|
|
|
namespace webrtc {
|
|
class CriticalSectionWrapper;
|
|
class ModuleRtpRtcpImpl;
|
|
class ReceiverFEC;
|
|
|
|
class RTPReceiverVideo : public RTPReceiverStrategy {
|
|
public:
|
|
RTPReceiverVideo(const WebRtc_Word32 id,
|
|
RTPReceiver* parent,
|
|
ModuleRtpRtcpImpl* owner);
|
|
|
|
virtual ~RTPReceiverVideo();
|
|
|
|
WebRtc_Word32 ParseRtpPacket(
|
|
WebRtcRTPHeader* rtp_header,
|
|
const ModuleRTPUtility::PayloadUnion& specificPayload,
|
|
const bool is_red,
|
|
const WebRtc_UWord8* packet,
|
|
const WebRtc_UWord16 packet_length,
|
|
const WebRtc_Word64 timestamp);
|
|
|
|
WebRtc_Word32 GetFrequencyHz() const;
|
|
|
|
RTPAliveType ProcessDeadOrAlive(WebRtc_UWord16 lastPayloadLength) const;
|
|
|
|
bool PayloadIsCompatible(
|
|
const ModuleRTPUtility::Payload& payload,
|
|
const WebRtc_UWord32 frequency,
|
|
const WebRtc_UWord8 channels,
|
|
const WebRtc_UWord32 rate) const;
|
|
|
|
void UpdatePayloadRate(
|
|
ModuleRTPUtility::Payload* payload,
|
|
const WebRtc_UWord32 rate) const;
|
|
|
|
ModuleRTPUtility::Payload* CreatePayloadType(
|
|
const char payloadName[RTP_PAYLOAD_NAME_SIZE],
|
|
const WebRtc_Word8 payloadType,
|
|
const WebRtc_UWord32 frequency,
|
|
const WebRtc_UWord8 channels,
|
|
const WebRtc_UWord32 rate);
|
|
|
|
WebRtc_Word32 InvokeOnInitializeDecoder(
|
|
RtpFeedback* callback,
|
|
const WebRtc_Word32 id,
|
|
const WebRtc_Word8 payloadType,
|
|
const char payloadName[RTP_PAYLOAD_NAME_SIZE],
|
|
const ModuleRTPUtility::PayloadUnion& specificPayload) const;
|
|
|
|
virtual WebRtc_Word32 ReceiveRecoveredPacketCallback(
|
|
WebRtcRTPHeader* rtpHeader,
|
|
const WebRtc_UWord8* payloadData,
|
|
const WebRtc_UWord16 payloadDataLength);
|
|
|
|
void SetPacketOverHead(WebRtc_UWord16 packetOverHead);
|
|
|
|
protected:
|
|
WebRtc_Word32 SetCodecType(const RtpVideoCodecTypes videoType,
|
|
WebRtcRTPHeader* rtpHeader) const;
|
|
|
|
WebRtc_Word32 ParseVideoCodecSpecificSwitch(
|
|
WebRtcRTPHeader* rtpHeader,
|
|
const WebRtc_UWord8* payloadData,
|
|
const WebRtc_UWord16 payloadDataLength,
|
|
const RtpVideoCodecTypes videoType);
|
|
|
|
WebRtc_Word32 ReceiveGenericCodec(WebRtcRTPHeader *rtpHeader,
|
|
const WebRtc_UWord8* payloadData,
|
|
const WebRtc_UWord16 payloadDataLength);
|
|
|
|
WebRtc_Word32 ReceiveVp8Codec(WebRtcRTPHeader *rtpHeader,
|
|
const WebRtc_UWord8* payloadData,
|
|
const WebRtc_UWord16 payloadDataLength);
|
|
|
|
WebRtc_Word32 BuildRTPheader(const WebRtcRTPHeader* rtpHeader,
|
|
WebRtc_UWord8* dataBuffer) const;
|
|
|
|
private:
|
|
WebRtc_Word32 ParseVideoCodecSpecific(
|
|
WebRtcRTPHeader* rtpHeader,
|
|
const WebRtc_UWord8* payloadData,
|
|
const WebRtc_UWord16 payloadDataLength,
|
|
const RtpVideoCodecTypes videoType,
|
|
const bool isRED,
|
|
const WebRtc_UWord8* incomingRtpPacket,
|
|
const WebRtc_UWord16 incomingRtpPacketSize,
|
|
const WebRtc_Word64 nowMS);
|
|
|
|
WebRtc_Word32 _id;
|
|
RTPReceiver* _parent;
|
|
|
|
CriticalSectionWrapper* _criticalSectionReceiverVideo;
|
|
|
|
// FEC
|
|
bool _currentFecFrameDecoded;
|
|
ReceiverFEC* _receiveFEC;
|
|
};
|
|
} // namespace webrtc
|
|
#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_
|