Log - 69a0ecad64a82a6915aa589fe46134e259a965b2 - fp2-dev/platform/external/chromium_org/third_party/webrtc

4b1817f Add configuration for cpu overuse detection to video send stream. by asapersson@webrtc.org · 11 years ago
48ac0da Drop early packets when not sending in TransportAdapter. by sprang@webrtc.org · 11 years ago
c766775 Wire up RTX in VideoReceiveStream. by pbos@webrtc.org · 11 years ago
df9f099 Permitting double start/stopping of streams. by pbos@webrtc.org · 11 years ago
ca72300 Wire up statistics in video send stream of new video engine api by sprang@webrtc.org · 11 years ago
39139dc Revert r5294 to re-roll r5293. by pbos@webrtc.org · 11 years ago
0af1d21 Revert 5293 "Auto instantiate RBE depending on whether AST or TO..." by turaj@webrtc.org · 11 years ago
ee867fa Auto instantiate RBE depending on whether AST or TOF is available in incoming packet stream. by solenberg@webrtc.org · 11 years ago
ab6ccbc Adding REMB to receive stream configuration, the send side will always by mflodman@webrtc.org · 11 years ago
e4d538a Make sure channels in the same call are in the same channel group. by mflodman@webrtc.org · 11 years ago
7123a80 Add SwapFrame() to VideoSendStreamInput. by pbos@webrtc.org · 11 years ago
9105cbd Set up SSRCs correctly after switching codec. by pbos@webrtc.org · 11 years ago
4a9843f Implement and test EncodedImageCallback in new ViE API. by sprang@webrtc.org · 11 years ago
d7d60c8 Connect pacer/padding to SuspendBelowMinBitrate by henrik.lundin@webrtc.org · 11 years ago
60108c2 Rename RTP-extension constants. by pbos@webrtc.org · 11 years ago
48cc9dc Rename video streams' start/stop methods. by pbos@webrtc.org · 11 years ago
8fdf191 Rename AutoMute to SuspendBelowMinBitrate by henrik.lundin@webrtc.org · 11 years ago
8f2997c Implement VideoSendStream::SetCodec(). by pbos@webrtc.org · 11 years ago
b581c90 Separate Call API/build files from video_engine/. by pbos@webrtc.org · 11 years ago[Renamed (93%) from video_engine/internal/video_send_stream.cc]
d4ec1f5 Removing the threshold from the auto-mute APIs by henrik.lundin@webrtc.org · 11 years ago
ce21c82 Porting auto mute to new ViE API by henrik.lundin@webrtc.org · 11 years ago
3ba57eb Implement I420FrameCallbacks in Call. by pbos@webrtc.org · 11 years ago
6133dd5 Fixed issue with how MTU is calculated. by sprang@webrtc.org · 11 years ago
4fe8543 Wired up max packet size and added simple test. by sprang@webrtc.org · 11 years ago
aa693dd Enable FEC for VideoSendStream. by pbos@webrtc.org · 11 years ago
26d75f3 Break out glue for old->new Transport. by pbos@webrtc.org · 11 years ago
f952fce Implement NACK over RTX for VideoSendStream. by pbos@webrtc.org · 11 years ago
e22b761 Implement 'abs-send-time' extension in VideoSendStream. by pbos@webrtc.org · 11 years ago
905cebd Implement 'toffset' extension in VideoSendStream. by pbos@webrtc.org · 11 years ago
d8e92c9 Remove newapi:: namespace for typenames without overlap. by pbos@webrtc.org · 11 years ago
debc672 Enabling and testing RTCP CNAME in new API. by pbos@webrtc.org · 11 years ago
a0a91d8 Adds two tests for verifying padding and ramp-up behavior. by stefan@webrtc.org · 11 years ago
30c741a Fix implicit int->bool conversion in VideoSendStream::DeliverRtcp. by pbos@webrtc.org · 11 years ago
bf9bc32 Glue code and tests for NACK in new VideoEngine API. by pbos@webrtc.org · 11 years ago
ecbeb2b Hooking up first simple CPU adaptation version. by mflodman@webrtc.org · 11 years ago
12d5ede Initial port of FullStackTest to new VideoEngine API. by pbos@webrtc.org · 11 years ago
2c343fc Stats+Config moved into VideoSend/ReceiveStreams. by pbos@webrtc.org · 11 years ago
d9f9185 FrameGenerator class for future fake capture device. by pbos@webrtc.org · 11 years ago
dc8c883 New VideoEngine API implementation on top of old one, first steps. by pbos@webrtc.org · 11 years ago