dc80bae2a6
Clean some logs and add asserts in the way. BUG=3153 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/11129004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5861 4adac7df-926f-26a2-2b94-8c16560cd09d
142 lines
5.1 KiB
C++
142 lines
5.1 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_SENDER_VIDEO_H_
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#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_
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#include <list>
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#include "webrtc/common_types.h"
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#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
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#include "webrtc/modules/rtp_rtcp/source/bitrate.h"
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#include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
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#include "webrtc/modules/rtp_rtcp/source/producer_fec.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
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#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
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#include "webrtc/modules/rtp_rtcp/source/video_codec_information.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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class CriticalSectionWrapper;
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struct RtpPacket;
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class RTPSenderVideo
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{
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public:
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RTPSenderVideo(Clock* clock,
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RTPSenderInterface* rtpSender);
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virtual ~RTPSenderVideo();
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virtual RtpVideoCodecTypes VideoCodecType() const;
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uint16_t FECPacketOverhead() const;
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int32_t RegisterVideoPayload(
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const char payloadName[RTP_PAYLOAD_NAME_SIZE],
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const int8_t payloadType,
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const uint32_t maxBitRate,
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ModuleRTPUtility::Payload*& payload);
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int32_t SendVideo(const RtpVideoCodecTypes videoType,
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const FrameType frameType,
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const int8_t payloadType,
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const uint32_t captureTimeStamp,
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int64_t capture_time_ms,
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const uint8_t* payloadData,
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const uint32_t payloadSize,
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const RTPFragmentationHeader* fragmentation,
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VideoCodecInformation* codecInfo,
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const RTPVideoTypeHeader* rtpTypeHdr);
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int32_t SendRTPIntraRequest();
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void SetVideoCodecType(RtpVideoCodecTypes type);
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VideoCodecInformation* CodecInformationVideo();
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void SetMaxConfiguredBitrateVideo(const uint32_t maxBitrate);
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uint32_t MaxConfiguredBitrateVideo() const;
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// FEC
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int32_t SetGenericFECStatus(const bool enable,
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const uint8_t payloadTypeRED,
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const uint8_t payloadTypeFEC);
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int32_t GenericFECStatus(bool& enable,
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uint8_t& payloadTypeRED,
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uint8_t& payloadTypeFEC) const;
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int32_t SetFecParameters(const FecProtectionParams* delta_params,
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const FecProtectionParams* key_params);
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void ProcessBitrate();
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uint32_t VideoBitrateSent() const;
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uint32_t FecOverheadRate() const;
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int SelectiveRetransmissions() const;
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int SetSelectiveRetransmissions(uint8_t settings);
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protected:
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virtual int32_t SendVideoPacket(uint8_t* dataBuffer,
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const uint16_t payloadLength,
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const uint16_t rtpHeaderLength,
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const uint32_t capture_timestamp,
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int64_t capture_time_ms,
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StorageType storage,
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bool protect);
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private:
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int32_t SendGeneric(const FrameType frame_type,
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const int8_t payload_type,
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const uint32_t capture_timestamp,
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int64_t capture_time_ms,
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const uint8_t* payload, const uint32_t size);
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int32_t SendVP8(const FrameType frameType,
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const int8_t payloadType,
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const uint32_t captureTimeStamp,
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int64_t capture_time_ms,
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const uint8_t* payloadData,
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const uint32_t payloadSize,
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const RTPFragmentationHeader* fragmentation,
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const RTPVideoTypeHeader* rtpTypeHdr);
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private:
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RTPSenderInterface& _rtpSender;
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CriticalSectionWrapper* _sendVideoCritsect;
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RtpVideoCodecTypes _videoType;
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VideoCodecInformation* _videoCodecInformation;
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uint32_t _maxBitrate;
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int32_t _retransmissionSettings;
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// FEC
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ForwardErrorCorrection _fec;
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bool _fecEnabled;
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int8_t _payloadTypeRED;
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int8_t _payloadTypeFEC;
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unsigned int _numberFirstPartition;
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FecProtectionParams delta_fec_params_;
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FecProtectionParams key_fec_params_;
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ProducerFec producer_fec_;
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// Bitrate used for FEC payload, RED headers, RTP headers for FEC packets
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// and any padding overhead.
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Bitrate _fecOverheadRate;
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// Bitrate used for video payload and RTP headers
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Bitrate _videoBitrate;
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_
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