| /* |
| * 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_SENDER_VIDEO_H_ |
| #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_ |
| |
| #include <list> |
| |
| #include "webrtc/base/criticalsection.h" |
| #include "webrtc/base/onetimeevent.h" |
| #include "webrtc/base/thread_annotations.h" |
| #include "webrtc/common_types.h" |
| #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
| #include "webrtc/modules/rtp_rtcp/source/bitrate.h" |
| #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h" |
| #include "webrtc/modules/rtp_rtcp/source/producer_fec.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" |
| #include "webrtc/modules/rtp_rtcp/source/video_codec_information.h" |
| #include "webrtc/typedefs.h" |
| |
| namespace webrtc { |
| |
| class RTPSenderVideo { |
| public: |
| RTPSenderVideo(Clock* clock, RTPSenderInterface* rtpSender); |
| virtual ~RTPSenderVideo(); |
| |
| virtual RtpVideoCodecTypes VideoCodecType() const; |
| |
| size_t FECPacketOverhead() const; |
| |
| static RtpUtility::Payload* CreateVideoPayload( |
| const char payloadName[RTP_PAYLOAD_NAME_SIZE], |
| const int8_t payloadType); |
| |
| int32_t SendVideo(const RtpVideoCodecTypes videoType, |
| const FrameType frameType, |
| const int8_t payloadType, |
| const uint32_t captureTimeStamp, |
| int64_t capture_time_ms, |
| const uint8_t* payloadData, |
| const size_t payloadSize, |
| const RTPFragmentationHeader* fragmentation, |
| const RTPVideoHeader* video_header); |
| |
| int32_t SendRTPIntraRequest(); |
| |
| void SetVideoCodecType(RtpVideoCodecTypes type); |
| |
| // FEC |
| void SetGenericFECStatus(const bool enable, |
| const uint8_t payloadTypeRED, |
| const uint8_t payloadTypeFEC); |
| |
| void GenericFECStatus(bool* enable, |
| uint8_t* payloadTypeRED, |
| uint8_t* payloadTypeFEC) const; |
| |
| void SetFecParameters(const FecProtectionParams* delta_params, |
| const FecProtectionParams* key_params); |
| |
| void ProcessBitrate(); |
| |
| uint32_t VideoBitrateSent() const; |
| uint32_t FecOverheadRate() const; |
| |
| int SelectiveRetransmissions() const; |
| void SetSelectiveRetransmissions(uint8_t settings); |
| |
| private: |
| void SendVideoPacket(uint8_t* dataBuffer, |
| const size_t payloadLength, |
| const size_t rtpHeaderLength, |
| uint16_t seq_num, |
| const uint32_t capture_timestamp, |
| int64_t capture_time_ms, |
| StorageType storage); |
| |
| void SendVideoPacketAsRed(uint8_t* dataBuffer, |
| const size_t payloadLength, |
| const size_t rtpHeaderLength, |
| uint16_t video_seq_num, |
| const uint32_t capture_timestamp, |
| int64_t capture_time_ms, |
| StorageType media_packet_storage, |
| bool protect); |
| |
| RTPSenderInterface& _rtpSender; |
| |
| // Should never be held when calling out of this class. |
| const rtc::CriticalSection crit_; |
| |
| RtpVideoCodecTypes _videoType; |
| int32_t _retransmissionSettings GUARDED_BY(crit_); |
| |
| // FEC |
| ForwardErrorCorrection fec_; |
| bool fec_enabled_ GUARDED_BY(crit_); |
| int8_t red_payload_type_ GUARDED_BY(crit_); |
| int8_t fec_payload_type_ GUARDED_BY(crit_); |
| FecProtectionParams delta_fec_params_ GUARDED_BY(crit_); |
| FecProtectionParams key_fec_params_ GUARDED_BY(crit_); |
| ProducerFec producer_fec_ GUARDED_BY(crit_); |
| |
| // Bitrate used for FEC payload, RED headers, RTP headers for FEC packets |
| // and any padding overhead. |
| Bitrate _fecOverheadRate; |
| // Bitrate used for video payload and RTP headers |
| Bitrate _videoBitrate; |
| OneTimeEvent first_frame_sent_; |
| }; |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_ |