Log - 5b4a29777dab756293048ea4de1a37dbafab5c79 - fp2-dev/platform/external/chromium_org/third_party/webrtc

49e6306 Preserve RTP states for restarted VideoSendStreams. by pbos@webrtc.org · 10 years ago
77cf8de Have RTX be enabled by setting an RTX payload type instead of by setting an RTX SSRC. by stefan@webrtc.org · 10 years ago
2fa9f7e Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR). by asapersson@webrtc.org · 10 years ago
0e4512b Callback for send bitrate estimates - new roll by sprang@webrtc.org · 11 years ago
66e84b0 Revert 5259 "Callback for send bitrate estimates" by sprang@webrtc.org · 11 years ago
f1d22d4 Callback for send bitrate estimates by sprang@webrtc.org · 11 years ago
cf5c552 Add callbacks for send channel rtp statistics by sprang@webrtc.org · 11 years ago
8db148e Add API to query video engine for the send-side delay. by stefan@webrtc.org · 11 years ago
8911937 Add callbacks for send channel rtcp statistics by sprang@webrtc.org · 11 years ago
9435a17 Add send frame rate statistics callback by sprang@webrtc.org · 11 years ago
da3ae7c Adds support for sending redundant payloads over RTX. by stefan@webrtc.org · 11 years ago
3fe2e7f Added API for enabling/disabling RTCP Receiver Reference Time extension. by asapersson@webrtc.org · 11 years ago
04281a4 Fix for RTX in combination with pacing. by stefan@webrtc.org · 11 years ago
65e4415 Removed unused code. by asapersson@webrtc.org · 11 years ago
f1262f3 Propagate estimated RTT from receivers to rtt observer. by asapersson@webrtc.org · 11 years ago
5ee7139 Improving padding rules and breaking out bw allocation to ViEEncoder. by stefan@webrtc.org · 11 years ago
55055d2 Update talk to 50918584. by wu@webrtc.org · 11 years ago
1628267 Revert r4301 by tnakamura@webrtc.org · 11 years ago
a430fef Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the by stefan@webrtc.org · 11 years ago
19f7ac1 Enqueue packet in pacer if sending fails by hclam@chromium.org · 11 years ago
39f2547 Wire up pacer-based padding. by stefan@webrtc.org · 11 years ago
6c0fab5 Include files from webrtc/.. paths in rtp_rtcp/ by pbos@webrtc.org · 11 years ago
028ec72 Breaking out RTP header parsing from the RTP module. by stefan@webrtc.org · 11 years ago
c22830f Revert 3934 "Revert 3933 "Remove traces of deprecated WebRtc_Wor..." by pbos@webrtc.org · 11 years ago
2d6a699 Revert 3933 "Remove traces of deprecated WebRtc_Word types." by pbos@webrtc.org · 11 years ago
bbb54b3 Remove traces of deprecated WebRtc_Word types. by pbos@webrtc.org · 11 years ago
06077c9 Adding a payload type for RTX. by mflodman@webrtc.org · 11 years ago
41e3677 Revert "With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps." by stefan@webrtc.org · 11 years ago
82e0d35 With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps. by stefan@webrtc.org · 11 years ago
771774f WebRtc_Word32 => int32_t for rtp_rtcp/ by pbos@webrtc.org · 11 years ago
072c9b6 Adding RTX on source by mikhal@webrtc.org · 11 years ago
d4caede Remove DTMF detection. Talk team has been in the loop and there is no need for by turaj@webrtc.org · 11 years ago
a7761c7 Fix mismatch between different NACK list lengths and packet buffers. by stefan@webrtc.org · 11 years ago
6318790 Wire up CallStats to provide modules with correct RTT. by mflodman@webrtc.org · 12 years ago
32f05a7 Enable paced sender. Review URL: https://webrtc-codereview.appspot.com/965016 by pwestin@webrtc.org · 12 years ago
a7b57da Move src/ -> webrtc/ by andrew@webrtc.org · 12 years ago