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_ |