| Index: webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h
|
| diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h
|
| index 7cfebd91a8baee74aba521845b7b066235e5f90f..a2cd52736f3a13ed61b6d90a6c70eae528441543 100644
|
| --- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h
|
| +++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h
|
| @@ -13,36 +13,37 @@
|
|
|
| // Configuration file for RTP utilities (RTPSender, RTPReceiver ...)
|
| namespace webrtc {
|
| -enum { NACK_BYTECOUNT_SIZE = 60}; // size of our NACK history
|
| +enum { NACK_BYTECOUNT_SIZE = 60 }; // size of our NACK history
|
| // A sanity for the NACK list parsing at the send-side.
|
| enum { kSendSideNackListSizeSanity = 20000 };
|
| enum { kDefaultMaxReorderingThreshold = 50 }; // In sequence numbers.
|
| enum { kRtcpMaxNackFields = 253 };
|
|
|
| -enum { RTCP_INTERVAL_VIDEO_MS = 1000 };
|
| -enum { RTCP_INTERVAL_AUDIO_MS = 5000 };
|
| -enum { RTCP_SEND_BEFORE_KEY_FRAME_MS= 100 };
|
| -enum { RTCP_MAX_REPORT_BLOCKS = 31}; // RFC 3550 page 37
|
| -enum { RTCP_MIN_FRAME_LENGTH_MS = 17};
|
| -enum { kRtcpAppCode_DATA_SIZE = 32*4}; // multiple of 4, this is not a limitation of the size
|
| -enum { RTCP_RPSI_DATA_SIZE = 30};
|
| -enum { RTCP_NUMBER_OF_SR = 60 };
|
| -
|
| -enum { MAX_NUMBER_OF_TEMPORAL_ID = 8 }; // RFC
|
| -enum { MAX_NUMBER_OF_DEPENDENCY_QUALITY_ID = 128 };// RFC
|
| +enum { RTCP_INTERVAL_VIDEO_MS = 1000 };
|
| +enum { RTCP_INTERVAL_AUDIO_MS = 5000 };
|
| +enum { RTCP_SEND_BEFORE_KEY_FRAME_MS = 100 };
|
| +enum { RTCP_MAX_REPORT_BLOCKS = 31 }; // RFC 3550 page 37
|
| +enum { RTCP_MIN_FRAME_LENGTH_MS = 17 };
|
| +enum {
|
| + kRtcpAppCode_DATA_SIZE = 32 * 4
|
| +}; // multiple of 4, this is not a limitation of the size
|
| +enum { RTCP_RPSI_DATA_SIZE = 30 };
|
| +enum { RTCP_NUMBER_OF_SR = 60 };
|
| +
|
| +enum { MAX_NUMBER_OF_TEMPORAL_ID = 8 }; // RFC
|
| +enum { MAX_NUMBER_OF_DEPENDENCY_QUALITY_ID = 128 }; // RFC
|
| enum { MAX_NUMBER_OF_REMB_FEEDBACK_SSRCS = 255 };
|
|
|
| -enum { BW_HISTORY_SIZE = 35};
|
| +enum { BW_HISTORY_SIZE = 35 };
|
|
|
| -#define MIN_AUDIO_BW_MANAGEMENT_BITRATE 6
|
| -#define MIN_VIDEO_BW_MANAGEMENT_BITRATE 30
|
| +#define MIN_AUDIO_BW_MANAGEMENT_BITRATE 6
|
| +#define MIN_VIDEO_BW_MANAGEMENT_BITRATE 30
|
|
|
| -enum { DTMF_OUTBAND_MAX = 20};
|
| +enum { DTMF_OUTBAND_MAX = 20 };
|
|
|
| enum { RTP_MAX_BURST_SLEEP_TIME = 500 };
|
| enum { RTP_AUDIO_LEVEL_UNIQUE_ID = 0xbede };
|
| -enum { RTP_MAX_PACKETS_PER_FRAME= 512 }; // must be multiple of 32
|
| +enum { RTP_MAX_PACKETS_PER_FRAME = 512 }; // must be multiple of 32
|
| } // namespace webrtc
|
|
|
| -
|
| -#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_CONFIG_H_
|
| +#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_CONFIG_H_
|
|
|