| /* |
| * 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/rate_statistics.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/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, RTPSender* rtpSender); |
| virtual ~RTPSenderVideo(); |
| |
| virtual RtpVideoCodecTypes VideoCodecType() const; |
| |
| size_t FECPacketOverhead() const; |
| |
| static RtpUtility::Payload* CreateVideoPayload( |
| const char payload_name[RTP_PAYLOAD_NAME_SIZE], |
| int8_t payload_type); |
| |
| bool SendVideo(RtpVideoCodecTypes video_type, |
| FrameType frame_type, |
| int8_t payload_type, |
| uint32_t capture_timestamp, |
| int64_t capture_time_ms, |
| const uint8_t* payload_data, |
| size_t payload_size, |
| const RTPFragmentationHeader* fragmentation, |
| const RTPVideoHeader* video_header); |
| |
| int32_t SendRTPIntraRequest(); |
| |
| void SetVideoCodecType(RtpVideoCodecTypes type); |
| |
| // FEC |
| void SetGenericFECStatus(bool enable, |
| uint8_t payload_type_red, |
| uint8_t payload_type_fec); |
| |
| void GenericFECStatus(bool* enable, |
| uint8_t* payload_type_red, |
| uint8_t* payload_type_fec) const; |
| |
| void SetFecParameters(const FecProtectionParams* delta_params, |
| const FecProtectionParams* key_params); |
| |
| uint32_t VideoBitrateSent() const; |
| uint32_t FecOverheadRate() const; |
| |
| int SelectiveRetransmissions() const; |
| void SetSelectiveRetransmissions(uint8_t settings); |
| |
| private: |
| void SendVideoPacket(uint8_t* data_buffer, |
| size_t payload_length, |
| size_t rtp_header_length, |
| uint16_t seq_num, |
| uint32_t capture_timestamp, |
| int64_t capture_time_ms, |
| StorageType storage); |
| |
| void SendVideoPacketAsRed(uint8_t* data_buffer, |
| size_t payload_length, |
| size_t rtp_header_length, |
| uint16_t video_seq_num, |
| uint32_t capture_timestamp, |
| int64_t capture_time_ms, |
| StorageType media_packet_storage, |
| bool protect); |
| |
| RTPSender* const rtp_sender_; |
| Clock* const clock_; |
| |
| // Should never be held when calling out of this class. |
| rtc::CriticalSection crit_; |
| |
| RtpVideoCodecTypes video_type_ = kRtpVideoGeneric; |
| int32_t retransmission_settings_ GUARDED_BY(crit_) = kRetransmitBaseLayer; |
| |
| // FEC |
| ForwardErrorCorrection fec_; |
| bool fec_enabled_ GUARDED_BY(crit_) = false; |
| int8_t red_payload_type_ GUARDED_BY(crit_) = 0; |
| int8_t fec_payload_type_ GUARDED_BY(crit_) = 0; |
| FecProtectionParams delta_fec_params_ GUARDED_BY(crit_) = FecProtectionParams{ |
| 0, 1, kFecMaskRandom}; |
| FecProtectionParams key_fec_params_ GUARDED_BY(crit_) = FecProtectionParams{ |
| 0, 1, kFecMaskRandom}; |
| ProducerFec producer_fec_ GUARDED_BY(crit_); |
| |
| rtc::CriticalSection stats_crit_; |
| // Bitrate used for FEC payload, RED headers, RTP headers for FEC packets |
| // and any padding overhead. |
| RateStatistics fec_bitrate_ GUARDED_BY(stats_crit_); |
| // Bitrate used for video payload and RTP headers. |
| RateStatistics video_bitrate_ GUARDED_BY(stats_crit_); |
| OneTimeEvent first_frame_sent_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_ |