Chromium Code Reviews| Index: webrtc/modules/rtp_rtcp/source/rtp_sender_video.h |
| diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h |
| index 8307b83864de53302f9bf5cbe2973ad568c01073..c904d87de42c3e9676e5720464f2d0e41ec3b19a 100644 |
| --- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h |
| +++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h |
| @@ -40,15 +40,15 @@ class RTPSenderVideo { |
| static RtpUtility::Payload* CreateVideoPayload( |
| const char payloadName[RTP_PAYLOAD_NAME_SIZE], |
| - const int8_t payloadType); |
| + const int8_t payload_type); |
|
danilchap
2016/06/15 13:40:01
while chanching this lines, may be also change typ
Sergey Ulanov
2016/06/15 18:27:53
Done.
|
| - int32_t SendVideo(const RtpVideoCodecTypes videoType, |
| - const FrameType frameType, |
| - const int8_t payloadType, |
| - const uint32_t captureTimeStamp, |
| + int32_t SendVideo(const RtpVideoCodecTypes video_type, |
| + const FrameType frame_type, |
| + const int8_t payload_type, |
| + const uint32_t capture_timestamp, |
| int64_t capture_time_ms, |
| - const uint8_t* payloadData, |
| - const size_t payloadSize, |
| + const uint8_t* payload_data, |
| + const size_t payload_size, |
| const RTPFragmentationHeader* fragmentation, |
| const RTPVideoHeader* video_header); |
| @@ -58,12 +58,12 @@ class RTPSenderVideo { |
| // FEC |
| void SetGenericFECStatus(const bool enable, |
| - const uint8_t payloadTypeRED, |
| - const uint8_t payloadTypeFEC); |
| + const uint8_t payload_type_red, |
| + const uint8_t payload_type_fec); |
| void GenericFECStatus(bool* enable, |
| - uint8_t* payloadTypeRED, |
| - uint8_t* payloadTypeFEC) const; |
| + uint8_t* payload_type_red, |
| + uint8_t* payload_type_fec) const; |
| void SetFecParameters(const FecProtectionParams* delta_params, |
| const FecProtectionParams* key_params); |
| @@ -77,47 +77,48 @@ class RTPSenderVideo { |
| void SetSelectiveRetransmissions(uint8_t settings); |
| private: |
| - void SendVideoPacket(uint8_t* dataBuffer, |
| - const size_t payloadLength, |
| - const size_t rtpHeaderLength, |
| + void SendVideoPacket(uint8_t* data_buffer, |
| + const size_t payload_length, |
| + const size_t rtp_header_length, |
| 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, |
| + void SendVideoPacketAsRed(uint8_t* data_buffer, |
| + const size_t payload_length, |
| + const size_t rtp_header_length, |
| uint16_t video_seq_num, |
| const uint32_t capture_timestamp, |
| int64_t capture_time_ms, |
| StorageType media_packet_storage, |
| bool protect); |
| - RTPSenderInterface& _rtpSender; |
| + RTPSenderInterface* rtp_sender_; |
|
danilchap
2016/06/15 13:40:01
RTPSenderInterface* const
Sergey Ulanov
2016/06/15 18:27:53
Done.
|
| // Should never be held when calling out of this class. |
| const rtc::CriticalSection crit_; |
| - RtpVideoCodecTypes _videoType; |
| - int32_t _retransmissionSettings GUARDED_BY(crit_); |
| + RtpVideoCodecTypes video_type_ = kRtpVideoGeneric; |
| + int32_t retransmission_settings_ GUARDED_BY(crit_) = kRetransmitBaseLayer; |
| // 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_); |
| + 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 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 fec_overhead_rate_; |
| // Bitrate used for video payload and RTP headers |
| - Bitrate _videoBitrate; |
| + Bitrate video_bitrate_; |
| OneTimeEvent first_frame_sent_; |
| }; |
| + |
| } // namespace webrtc |
| #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_ |