244251a9cd
The big benefit is we no longer have a circular dependency between the media receiver and the payload registry. The payload registry is starting to take a bit more place on the stage, and now knows how to do different things depending on audio or video. BUG= TESTED=rtp_rtcp_unittests, vie_auto_test, voe_auto_test Review URL: https://webrtc-codereview.appspot.com/1078004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3465 4adac7df-926f-26a2-2b94-8c16560cd09d
115 lines
3.8 KiB
C++
115 lines
3.8 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_
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#include "bitrate.h"
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#include "rtp_receiver_strategy.h"
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#include "rtp_rtcp_defines.h"
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#include "rtp_utility.h"
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#include "scoped_ptr.h"
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#include "typedefs.h"
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namespace webrtc {
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class CriticalSectionWrapper;
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class ModuleRtpRtcpImpl;
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class ReceiverFEC;
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class RTPReceiver;
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class RTPPayloadRegistry;
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class RTPReceiverVideo : public RTPReceiverStrategy {
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public:
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RTPReceiverVideo(const WebRtc_Word32 id,
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const RTPPayloadRegistry* rtp_payload_registry,
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RtpData* data_callback);
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virtual ~RTPReceiverVideo();
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WebRtc_Word32 ParseRtpPacket(
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WebRtcRTPHeader* rtp_header,
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const ModuleRTPUtility::PayloadUnion& specificPayload,
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const bool is_red,
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const WebRtc_UWord8* packet,
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const WebRtc_UWord16 packet_length,
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const WebRtc_Word64 timestamp,
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const bool is_first_packet);
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WebRtc_Word32 GetFrequencyHz() const;
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RTPAliveType ProcessDeadOrAlive(WebRtc_UWord16 lastPayloadLength) const;
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bool ShouldReportCsrcChanges(WebRtc_UWord8 payload_type) const;
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WebRtc_Word32 OnNewPayloadTypeCreated(
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const char payloadName[RTP_PAYLOAD_NAME_SIZE],
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const WebRtc_Word8 payloadType,
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const WebRtc_UWord32 frequency);
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WebRtc_Word32 InvokeOnInitializeDecoder(
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RtpFeedback* callback,
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const WebRtc_Word32 id,
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const WebRtc_Word8 payloadType,
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const char payloadName[RTP_PAYLOAD_NAME_SIZE],
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const ModuleRTPUtility::PayloadUnion& specificPayload) const;
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virtual WebRtc_Word32 ReceiveRecoveredPacketCallback(
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WebRtcRTPHeader* rtpHeader,
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const WebRtc_UWord8* payloadData,
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const WebRtc_UWord16 payloadDataLength);
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void SetPacketOverHead(WebRtc_UWord16 packetOverHead);
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protected:
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WebRtc_Word32 SetCodecType(const RtpVideoCodecTypes videoType,
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WebRtcRTPHeader* rtpHeader) const;
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WebRtc_Word32 ParseVideoCodecSpecificSwitch(
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WebRtcRTPHeader* rtpHeader,
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const WebRtc_UWord8* payloadData,
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const WebRtc_UWord16 payloadDataLength,
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const RtpVideoCodecTypes videoType,
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const bool isFirstPacket);
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WebRtc_Word32 ReceiveGenericCodec(WebRtcRTPHeader *rtpHeader,
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const WebRtc_UWord8* payloadData,
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const WebRtc_UWord16 payloadDataLength);
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WebRtc_Word32 ReceiveVp8Codec(WebRtcRTPHeader *rtpHeader,
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const WebRtc_UWord8* payloadData,
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const WebRtc_UWord16 payloadDataLength);
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WebRtc_Word32 BuildRTPheader(const WebRtcRTPHeader* rtpHeader,
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WebRtc_UWord8* dataBuffer) const;
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private:
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WebRtc_Word32 ParseVideoCodecSpecific(
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WebRtcRTPHeader* rtpHeader,
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const WebRtc_UWord8* payloadData,
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const WebRtc_UWord16 payloadDataLength,
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const RtpVideoCodecTypes videoType,
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const bool isRED,
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const WebRtc_UWord8* incomingRtpPacket,
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const WebRtc_UWord16 incomingRtpPacketSize,
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const WebRtc_Word64 nowMS,
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const bool isFirstPacket);
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WebRtc_Word32 _id;
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const RTPPayloadRegistry* _rtpRtpPayloadRegistry;
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CriticalSectionWrapper* _criticalSectionReceiverVideo;
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// FEC
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bool _currentFecFrameDecoded;
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ReceiverFEC* _receiveFEC;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_
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