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Unified Diff: webrtc/modules/rtp_rtcp/source/rtp_sender_video.h

Issue 2067673004: Style cleanups in RtpSender. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: fix compilation Created 4 years, 6 months ago
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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_

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