2557b86e7648ffebc5781df9f093ca5a84efc219 |
|
18-Nov-2015 |
Henrik Kjellander <kjellander@google.com> |
modules/video_coding refactorings The main purpose was the interface-> include rename, but other files were also moved, eliminating the "main" dir. To avoid breaking downstream, the "interface" directories were copied into a new "video_coding/include" dir. The old headers got pragma warnings added about deprecation (a very short deprecation since I plan to remove them as soon downstream is updated). Other files also moved: video_coding/main/source -> video_coding video_coding/main/test -> video_coding/test BUG=webrtc:5095 TESTED=Passing compile-trybots with --clobber flag: git cl try --clobber --bot=win_compile_rel --bot=linux_compile_rel --bot=android_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417283007 . Cr-Commit-Position: refs/heads/master@{#10694}
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
|
22993e1a0c114122fc1b9de0fc74d4096ec868bd |
|
19-Oct-2015 |
pbos <pbos@webrtc.org> |
Unify FrameType and VideoFrameType. Prevents some heap allocation and frame-type conversion since interfaces mismatch. Also it's less confusing to have one type for this. BUG=webrtc:5042 R=magjed@webrtc.org, mflodman@webrtc.org, henrik.lundin@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/1371043003 Cr-Commit-Position: refs/heads/master@{#10320}
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
|
0c4e06b4c6107a1b94f764e279e4fb4161e905b0 |
|
07-Oct-2015 |
Peter Boström <pbos@webrtc.org> |
Use suffixed {uint,int}{8,16,32,64}_t types. Removes the use of uint8, etc. in favor of uint8_t. BUG=webrtc:5024 R=henrik.lundin@webrtc.org, henrikg@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org Review URL: https://codereview.webrtc.org/1362503003 . Cr-Commit-Position: refs/heads/master@{#10196}
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
|
4b91bd08979fcfb191cdae27ad24936beefce735 |
|
26-Jun-2015 |
Peter Boström <pbos@webrtc.org> |
Move frame input (ViECapturer) to webrtc/video/. Renames ViECapturer to VideoCaptureInput and initializes several parameters on construction instead of setters. Also removes an old deadlock suppression. BUG=1695, 2999 R=asapersson@webrtc.org, mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/53559004. Cr-Commit-Position: refs/heads/master@{#9508}
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
|
4765070b8d6f024509c717c04d9b708750666927 |
|
30-May-2015 |
Miguel Casas-Sanchez <mcasas@webrtc.org> |
Rename I420VideoFrame to VideoFrame. This is a mechanical change since it affects so many files. I420VideoFrame -> VideoFrame and reformatted. Rationale: in the next CL I420VideoFrame will get an indication of Pixel Format (I420 for starters) and of storage type: usually UNOWNED, could be SHMEM, and in the near future will be possibly TEXTURE. See https://codereview.chromium.org/1154153003 for the change that happened in Cr. BUG=4730, chromium:440843 R=jiayl@webrtc.org, niklas.enbom@webrtc.org, pthatcher@webrtc.org Review URL: https://webrtc-codereview.appspot.com/52629004 Cr-Commit-Position: refs/heads/master@{#9339}
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
|
300eeb68f55c5091c7045e377578586733cddf16 |
|
12-May-2015 |
Peter Boström <pbos@webrtc.org> |
Remove VideoEngine interfaces. Removes ViE interfaces, _impl.cc files, managers (such as ViEChannelManager and ViEInputManager) as well as ViESharedData. Interfaces necessary to implement observers have been moved to a corresponding header (such as vie_channel.h). BUG=1695, 4491 R=mflodman@webrtc.org, solenberg@webrtc.org TBR=pthatcher@webrtc.org Review URL: https://webrtc-codereview.appspot.com/55379004 Cr-Commit-Position: refs/heads/master@{#9179}
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
|
d3ddc1b69e9cdfd7c6d38ab02b8d8ab891d30fd1 |
|
07-May-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Consistently use DCHECK, not ASSERT or assert in talk/media/webrtc/. BUG= R=pbos@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/49929004 Cr-Commit-Position: refs/heads/master@{#9156}
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
|
81ea54eaac82b36b7208a02fd37a469d7d0bd9d0 |
|
07-May-2015 |
Peter Boström <pbos@webrtc.org> |
Remove WebRtcVideoEngine. Leaves a stub file for talk/media/webrtc/webrtcvideoengine.cc until build files in Chromium have been modified. BUG=1695,4566 R=mflodman@webrtc.org, pthatcher@webrtc.org Review URL: https://webrtc-codereview.appspot.com/48339004 Cr-Commit-Position: refs/heads/master@{#9148}
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
|
f16fcbec734e1e3303828525c9fd7e13e0803aab |
|
30-Apr-2015 |
Peter Boström <pbos@webrtc.org> |
Remove ViECapture usage in VideoSendStream. Instead a ViECapturer object is allocated and directly operated on. This additionally exposes ViESharedData to Call to access the module ProcessThread, moving towards Call ownership of shared resources. BUG=1695 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/45339004 Cr-Commit-Position: refs/heads/master@{#9119}
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
|
94cc1fe4af57a01a99a1f76f0ad3d48edf981321 |
|
29-Apr-2015 |
Peter Boström <pbos@webrtc.org> |
Remove ViEImageProcess usage in VideoSendStream. Replaces interface usage with direct calls on ViEEncoder removing a layer of indirection. Also removing some methods from ViEImageProcess that were only added for Video{Send,Receive}Stream usage. BUG=1695 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/45319004 Cr-Commit-Position: refs/heads/master@{#9111}
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
|
c4188fd3c74688264621393fc622cb81c042c1ac |
|
24-Apr-2015 |
Peter Boström <pbos@webrtc.org> |
Use IncomingVideoStream in VideoReceiveStream. Decouples VideoReceiveStream further from webrtc/video_engine/ as well as most of webrtc/modules/video_render/ resulting in a simpler setup. BUG=1695 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50749004 Cr-Commit-Position: refs/heads/master@{#9080}
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
|
e62202fedf57b74cc263246c0586ee353978caf8 |
|
21-Apr-2015 |
Shao Changbin <changbin.shao@webrtc.org> |
Support handling multiple RTX but only generate SDP with RTX associated with VP8. This implementation registers RTX-APT map inside RTP sender and receiver. While it only generates SDP with RTX associated with VP8 to make it compatible with previous Chrome versions. Should add following changes after reaches stable, * Use RTX-APT map for building and restoring RTP packets. * Add RTX support for RED or VP9 in Video engine. * Set RTX payload type for RED inside FecConfig in EndToEndTest. BUG=4024 R=mflodman@webrtc.org, pbos@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/36889004 Cr-Commit-Position: refs/heads/master@{#9040}
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
|
352b2d7a19d6313273608c26edf8900e592a3831 |
|
15-Apr-2015 |
Åsa Persson <asapersson@webrtc.org> |
Fix for sent/received RTCP packet counters returned by GetRtcpPacketTypeCounters. The returned counters are incorrect: sent_packets returns stats from a sent stream (and received_packets returns stats from a receive stream). Add separate functions for returning stats from send/receive stream and updated how functions are used. Add test implementation for histogram methods in system_wrappers/interface/metrics.h. BUG=4519 R=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/49639004 Cr-Commit-Position: refs/heads/master@{#9009}
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
|
76c53d36bc455fe89ca1f860d5171633198fe907 |
|
09-Apr-2015 |
Peter Boström <pbos@webrtc.org> |
Remove ViE interface usage from VideoReceiveStream. References channels and underlying objects directly instead of using interfaces referenced with channel id. Channel creation is still done as before for now. BUG=1695 R=stefan@webrtc.org TBR=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/46849004 Cr-Commit-Position: refs/heads/master@{#8958}
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
|
e59041672283a28bde0b043c0c2bc198272f82e1 |
|
26-Mar-2015 |
Stefan Holmer <holmer@google.com> |
Moving the pacer and the pacer thread to ChannelGroup. This means all channels within the same group will share the same pacing queue and scheduler. It also means padding will be computed and sent by a single pacer. To accomplish this I also introduce a PacketRouter which finds the RTP module which owns the packet to be paced out. BUG=4323 R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/45549004 Cr-Commit-Position: refs/heads/master@{#8864}
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
|
9f9ea7e5abc3fa561e6b190b45219f2416c8786b |
|
20-Mar-2015 |
perkj@webrtc.org <perkj@webrtc.org> |
Clean up webrtc external capture. This cl removes the dependency to the external capture module if external capturing is used in webrtc. It also removes two external capture methods that is not needed. Further more it adds I420VideoFrame::Create that takes a pointer to packed memory as input. R=magjed@webrtc.org, mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/43879004 Cr-Commit-Position: refs/heads/master@{#8804} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8804 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
|
143451d2590ef951f6e66a983a38a18fcd4c66a5 |
|
18-Mar-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Base start bitrate on last observed bitrate. Instead of setting bitrates based on codec target settings (which may have previously been capped by a codec max bitrate), fetch the last bandwidth allocated for this channel. This fixes broken low start bitrates due to QCIF being set as default codec in WebRtcVideoEngine2 which caps the max bitrate to 200kbps. BUG=1788 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/43789004 Cr-Commit-Position: refs/heads/master@{#8780} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8780 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
|
af612d5e0769571544952cbe55e675748afa9bdd |
|
18-Mar-2015 |
perkj@webrtc.org <perkj@webrtc.org> |
Reland "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame."" Original cl description: This removes the none const pointer entry and SwapFrame. Since frames delivered using VideoSendStream no longer use the external capture module, VideoSendStream will not get an incoming framerate callback. VideoSendStream now uses a rtc::RateTracker. Also, the video engine must ensure that time stamps are always increasing. With this, time stamps (ntp, render_time and rtp timestamps ) are checked and set in ViECapturer::OnIncomingCapturedFrame This cl was previously reverted in https://webrtc-codereview.appspot.com/46549004/. Patchset 1 contains the original patch after rebase. Patshet 2 fix webrtc_perf_tests reported in chromium:465306 Note that chromium:465287 is being fixed in https://webrtc-codereview.appspot.com/43829004/ BUG=1128 R=magjed@webrtc.org, mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/47629004 Cr-Commit-Position: refs/heads/master@{#8776} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8776 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
|
52cd828e1731266272e671020c353f5f89992a83 |
|
18-Mar-2015 |
pthatcher@webrtc.org <pthatcher@webrtc.org> |
Allow webrtc external encoder factories to declare encoders have internal camera sources. This flag is passed to existing VieExternalCodec API (and others) to denote encoders that don't require/expect frames from the normal capture pipeline. This is the simplest way to allow camera->encoder texture support, until textures are supported through the normal camera pipeline and the lifetime issues are all figured out (I hear this is on the backlog, but not there yet). Ideally, the flag would be on the encoder, but that doesn't work with SimulcastEncoderAdapter, since it doesn't create an encoder right away. Note that this change only affects WebRtcVideoEngine (not WRVE2), since WRVE2 uses video_send_stream, and my hope is that by the time things have switched to WRVE2, textures will be supported with the normal camera pipeline and the dependency on internal sources can be thrown away. BUG= R=pbos@webrtc.org, pthatcher@webrtc.org Review URL: https://webrtc-codereview.appspot.com/42349004 Cr-Commit-Position: refs/heads/master@{#8769} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8769 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
|
fdd10579496123c9a7fdc0bf185e2a26a12ed340 |
|
12-Mar-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Add CVO support to Vie layer. 1. standard plumbing CVO through vie layer. 2. added a rtp_cvo.h which has both conversion functions from rtp header byte to/from VideoRotation. WebRTCVideoEngine will later pass the rotation info in SendFrame() through VieVideoFrameI420. BUG=4145 R=mflodman@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/46429007 Cr-Commit-Position: refs/heads/master@{#8703} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8703 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
|
afdd5dd372d69be7244a3d90d70de9d5ecd60eb9 |
|
12-Mar-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Revert "Revert "Remove frame copy from cricket::VideoFrame to I420VideoFrame"" This reverts r8683 and is a reland of r8682. Reason for revert: The thread checker in Chromium that crashed has been fixed now. BUG=1128 TBR=tommi,pbos,pthatcher Review URL: https://webrtc-codereview.appspot.com/40319004 Cr-Commit-Position: refs/heads/master@{#8696} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8696 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
|
b218ff553148b9a26c82e3b3a46d626c4438cedd |
|
11-Mar-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Revert "Remove frame copy from cricket::VideoFrame to I420VideoFrame" This reverts r8682. Reason for revert: Fails on Chromium FYI content_browsertests BUG=1128 TBR=tommi,pbos,pthatcher Review URL: https://webrtc-codereview.appspot.com/47529004 Cr-Commit-Position: refs/heads/master@{#8683} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8683 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
|
370a72cc3ff928099c6ec6766659ed12155b74df |
|
11-Mar-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Remove frame copy from cricket::VideoFrame to I420VideoFrame BUG=1128 R=pbos@webrtc.org, pthatcher@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/42249004 Cr-Commit-Position: refs/heads/master@{#8682} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8682 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
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3e6e271ec3253e78ae0eb72156e5236d43f8731d |
|
26-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Implement CpuOveruseMetrics as callbacks. Adds avg_encode_ms and encode_usage_percent in WebRtcVideoEngine2 and corresponding stats to VideoSendStream::Stats. BUG=1667, 1788 R=asapersson@webrtc.org, mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/42429004 Cr-Commit-Position: refs/heads/master@{#8513} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8513 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
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1d0fa5d352fe12092201fade249905c7e1ff974b |
|
19-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Add RtcpPacketTypeCounter stats to new API. R=mflodman@webrtc.org, stefan@webrtc.org BUG=1667,1788 Review URL: https://webrtc-codereview.appspot.com/37489004 Cr-Commit-Position: refs/heads/master@{#8429} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8429 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
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f68e186de317abf2fd17e55a5e3cb417a0e50e1f |
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18-Feb-2015 |
magjed@webrtc.org <magjed@webrtc.org> |
Remove EnableMirroring and MirrorRenderStream R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/35239004 Cr-Commit-Position: refs/heads/master@{#8409} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8409 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
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5a7dc39277999cbfa0da053da5eacc7fee5cd307 |
|
13-Feb-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
This is a code clean up. No functional change intended. Consolidate the enum for capturer/frame rotation we use through out the code base. BUG=4145 R=mflodman@webrtc.org, perkj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/39859004 Cr-Commit-Position: refs/heads/master@{#8365} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8365 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
|
40367f984b2922fbfcf58d7e485ac0ef59149768 |
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13-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove default video encoders for new video API. Reduces stream creation time significantly. As a side effect also removes default encoders for receive-only channels. BUG=1788,1667 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/37049004 Cr-Commit-Position: refs/heads/master@{#8356} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8356 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
|
aafbec15f9e71f103587d1379ff12059d5285c48 |
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12-Feb-2015 |
solenberg@webrtc.org <solenberg@webrtc.org> |
Remove ViENetwork::SetBandwidthEstimationConfig() interface since dynamically changing BWE settings isn't necessary now that AIMD is the default. BUG=3735 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/39919005 Cr-Commit-Position: refs/heads/master@{#8351} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8351 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
|
19f3f71c9873cf5f6d647becd3620ddf8fd6ba7c |
|
02-Feb-2015 |
pkasting@chromium.org <pkasting@chromium.org> |
Fix apparent typo: int -> char. The surrounding similar methods all used unsigned char, using unsigned int in this case looks like an accident, especially since the function passes on the value in question to a function expecting a uint8. BUG=none TEST=none R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/40529004 Cr-Commit-Position: refs/heads/master@{#8228} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8228 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
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f18fba2f7b3d1fad7b7b38a9a5dc281bef06c50e |
|
14-Jan-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Implement SimulcastEncoderAdapter support. R=stefan@webrtc.org BUG=1788 Review URL: https://webrtc-codereview.appspot.com/37589004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8061 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
|
16825b1a828bb4ff40f7682040e43a239b7b8ca3 |
|
12-Jan-2015 |
pkasting@chromium.org <pkasting@chromium.org> |
Use int64_t more consistently for times, in particular for RTT values. Existing code was inconsistent about whether to use uint16_t, int, unsigned int, or uint32_t, and sometimes silently truncated one to another, or truncated int64_t. Because most core time-handling functions use int64_t, being consistent about using int64_t unless otherwise necessary minimizes the number of explicit or implicit casts. BUG=chromium:81439 TEST=none R=henrik.lundin@webrtc.org, holmer@google.com, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31349004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8045 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
|
8f27fcce79584378da97f0d84574564799e138d6 |
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09-Jan-2015 |
andrew@webrtc.org <andrew@webrtc.org> |
Revert 8028 "Support associated payload type when registering Rt..." Reasons for revert: 1. glaznev discovered potentially related problems using the Android AppRTCDemo. 2. We're trying to do an M41 webrtc roll in Chromium, and this CL is risky. > Support associated payload type when registering Rtx payload type. > > Major changes include, > - Add associated payload type for SetRtxSendPayloadType & SetRtxReceivePayloadType. > - Receiver: Restore RTP packets by the new RTX-APT map. > - Sender: Send RTP packets by checking RTX-APT map. > - Add RTX payload type for RED in the default codec list. > > BUG=4024 > R=pbos@webrtc.org, stefan@webrtc.org > TBR=mflodman@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/26259004 > > Patch from Changbin Shao <changbin.shao@intel.com>. TBR=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/33829004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8033 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
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2a169640a3225a559f926fe74f1fe1af239e191f |
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09-Jan-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Support associated payload type when registering Rtx payload type. Major changes include, - Add associated payload type for SetRtxSendPayloadType & SetRtxReceivePayloadType. - Receiver: Restore RTP packets by the new RTX-APT map. - Sender: Send RTP packets by checking RTX-APT map. - Add RTX payload type for RED in the default codec list. BUG=4024 R=pbos@webrtc.org, stefan@webrtc.org TBR=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26259004 Patch from Changbin Shao <changbin.shao@intel.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@8028 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
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742386a13670337db6e3bbf4cf54e7cb24a9b717 |
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19-Dec-2014 |
stefan@webrtc.org <stefan@webrtc.org> |
Enable payload-based padding by default and remove the API. BUG=1812 R=mflodman@webrtc.org, pbos@webrtc.org, perkj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/31319004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7964 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
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ce4e9a356200170abcdd44ff2af95f87a6781b8e |
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18-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Refactor some receive-side stats. Removes polling of CName as well as receive codec statistics in favor of internal callbacks keeping a statistics struct up to date. R=mflodman@webrtc.org, stefan@webrtc.org BUG=1667 Review URL: https://webrtc-codereview.appspot.com/28259005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7950 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
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599e299b9dc3dc07fc78cfeaba629566a201b4f1 |
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05-Dec-2014 |
kjellander@webrtc.org <kjellander@webrtc.org> |
cricket::VideoFrame int64 to int64_t. Needed for successful compile of ios arm64. BUG=3898 R=pthatcher@webrtc.org Review URL: https://webrtc-codereview.appspot.com/30359004 Patch from Zeke Chin <tkchin@webrtc.org>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@7817 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
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273a414b0ec2e58fdf3b817ad8b1a02f4ce15287 |
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01-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Report encoded frame size in VideoSendStream. Implements reporting transmitted frame size in WebRtcVideoEngine2. R=mflodman@webrtc.org, stefan@webrtc.org BUG=4033 Review URL: https://webrtc-codereview.appspot.com/33399004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7772 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
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4591fbd09f9cb6e83433c49a12dd8524c2806502 |
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20-Nov-2014 |
pkasting@chromium.org <pkasting@chromium.org> |
Use size_t more consistently for packet/payload lengths. See design doc at https://docs.google.com/a/chromium.org/document/d/1I6nmE9D_BmCY-IoV6MDPY2V6WYpEI-dg2apWXTfZyUI/edit?usp=sharing for more information. This CL was reviewed and approved in pieces in the following CLs: https://webrtc-codereview.appspot.com/24209004/ https://webrtc-codereview.appspot.com/24229004/ https://webrtc-codereview.appspot.com/24259004/ https://webrtc-codereview.appspot.com/25109004/ https://webrtc-codereview.appspot.com/26099004/ https://webrtc-codereview.appspot.com/27069004/ https://webrtc-codereview.appspot.com/27969004/ https://webrtc-codereview.appspot.com/27989004/ https://webrtc-codereview.appspot.com/29009004/ https://webrtc-codereview.appspot.com/30929004/ https://webrtc-codereview.appspot.com/30939004/ https://webrtc-codereview.appspot.com/31999004/ Committing as TBR to the original reviewers. BUG=chromium:81439 TEST=none TBR=pthatcher,henrik.lundin,tina.legrand,stefan,tkchin,glaznev,kjellander,perkj,mflodman,henrika,asapersson,niklas.enbom Review URL: https://webrtc-codereview.appspot.com/23129004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7726 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
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ece3890d3a40fe911ae895e28c329491e795b14d |
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14-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Report total bitrate for all streams in GetStats. This regression wasn't caught because I accidentally disabled multiple streams for EndToEndTest.GetStats in a refactoring. R=stefan@webrtc.org, xians@webrtc.org BUG=1667 Review URL: https://webrtc-codereview.appspot.com/27179004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7701 4adac7df-926f-26a2-2b94-8c16560cd09d
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7fe1e03dd6da66401010119734245f114bf06645 |
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14-Oct-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Wire up external encoders. R=pthatcher@webrtc.org BUG=1788 Review URL: https://webrtc-codereview.appspot.com/30649005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7440 4adac7df-926f-26a2-2b94-8c16560cd09d
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13-Oct-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 77414393-> 77554188 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7428 4adac7df-926f-26a2-2b94-8c16560cd09d
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25-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org> |
(Auto)update libjingle 73927775-> 74032598 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6965 4adac7df-926f-26a2-2b94-8c16560cd09d
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22-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 73794259-> 73891518 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6955 4adac7df-926f-26a2-2b94-8c16560cd09d
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13-Aug-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 73222930-> 73226398 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6891 4adac7df-926f-26a2-2b94-8c16560cd09d
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henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 73072800 -> 73215194 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6887 4adac7df-926f-26a2-2b94-8c16560cd09d
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29-Jul-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 72097588-> 72159069 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6799 4adac7df-926f-26a2-2b94-8c16560cd09d
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26-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69860953-> 70002228 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6542 4adac7df-926f-26a2-2b94-8c16560cd09d
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17-Jun-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69337301-> 69359922 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6457 4adac7df-926f-26a2-2b94-8c16560cd09d
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buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69144530-> 69164179 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6434 4adac7df-926f-26a2-2b94-8c16560cd09d
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buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 69143161-> 69144530 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6432 4adac7df-926f-26a2-2b94-8c16560cd09d
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buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 68985065-> 69005149 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6406 4adac7df-926f-26a2-2b94-8c16560cd09d
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buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 68646004-> 68648993 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6348 4adac7df-926f-26a2-2b94-8c16560cd09d
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07-May-2014 |
buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66388864-> 66406192 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6072 4adac7df-926f-26a2-2b94-8c16560cd09d
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buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 66301332-> 66303009 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6064 4adac7df-926f-26a2-2b94-8c16560cd09d
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(Auto)update libjingle 66236292-> 66294299 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6061 4adac7df-926f-26a2-2b94-8c16560cd09d
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(Auto)update libjingle 66033941-> 66098243 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6044 4adac7df-926f-26a2-2b94-8c16560cd09d
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buildbot@webrtc.org <buildbot@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
(Auto)update libjingle 64709629-> 64813990 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5897 4adac7df-926f-26a2-2b94-8c16560cd09d
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10-Apr-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
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wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 61538839. TBR=mallinath Review URL: https://webrtc-codereview.appspot.com/8669005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5548 4adac7df-926f-26a2-2b94-8c16560cd09d
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wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 60094938. Review URL: https://webrtc-codereview.appspot.com/7489005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5420 4adac7df-926f-26a2-2b94-8c16560cd09d
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13-Dec-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 58174641 together with http://review.webrtc.org/4319005/. R=turaj@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5809004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5287 4adac7df-926f-26a2-2b94-8c16560cd09d
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13-Dec-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 58127566 together with https://webrtc-codereview.appspot.com/5309005/. R=mallinath@webrtc.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5719004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5277 4adac7df-926f-26a2-2b94-8c16560cd09d
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12-Dec-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 5274 "Update talk to 58113193 together with https://webrt..." > Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. > > R=mallinath@webrtc.org, niklas.enbom@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/5719004 TBR=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5729004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5275 4adac7df-926f-26a2-2b94-8c16560cd09d
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12-Dec-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/. R=mallinath@webrtc.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5719004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5274 4adac7df-926f-26a2-2b94-8c16560cd09d
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11-Dec-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 58037405. R=sergeyu@chromium.org Review URL: https://webrtc-codereview.appspot.com/5579004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5267 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
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5bc25c41fc7880545052770dbcfe67f233c9b0c0 |
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05-Dec-2013 |
sergeyu@chromium.org <sergeyu@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 57692857 R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4999004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5217 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
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cecfd1832dc375225da3f5f18ecac63006ed06bf |
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30-Oct-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 55821645. TEST=try bots R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3139004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5053 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
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97077a3ab27259164eb121034b6e0ebe9ba592df |
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25-Oct-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update libjingle to 55618622. Update libyuv to r826. TEST=try bots R=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2889004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5038 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
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1b15f4226ff417095d2146401ca71cd98ab735b3 |
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07-Sep-2013 |
mallinath@webrtc.org <mallinath@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 51960985. R=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2188004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4696 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
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d6fef9d3805233dd34d253036fd95fc3ed1f7113 |
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27-Aug-2013 |
elham@webrtc.org <elham@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fixing SetDecodeErrorMode build error - got introduced when reverting r4562 R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2118004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4624 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
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7666db79fa269c6688651008edd8cf88276c0671 |
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22-Aug-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 51242664. TBR=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2090005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4594 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
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822fbd8b68ffdb481b9557e2950ae8d6657c8ce6 |
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16-Aug-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk to 50918584. Together with Stefan's http://review.webrtc.org/1960004/. R=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2048004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4556 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
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9dba52562725dbaced0d671982201ede753d72e8 |
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05-Aug-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
* Update libjingle to 50389769. * Together with "Add texture support for i420 video frame." from wuchengli@chromium.org. https://webrtc-codereview.appspot.com/1413004 RISK=P1 TESTED=try bots R=fischman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1967004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4489 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
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1e09a711263dd105e6f7a03812250084c64e5fd8 |
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26-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk folder to revision=49952949 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4413 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
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28654cbc2256230c978f41cbaf550bc2e9c2f2db |
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22-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk folder to revision=49713299. TBR=mallinath@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1848004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4380 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
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9de257d00f1f805af28f15fd814a8a84460028e5 |
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17-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk folder to revision=49470012. Same as 375 in libjingle's google code repository. TBR=wu@webrtc.org BUG=N/A Review URL: https://webrtc-codereview.appspot.com/1824004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4364 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
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723d683ecbe6a934885a60712c66ca2c01700a51 |
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12-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update talk folder to revision=49260075. Same as 369 in libjingle's google code repository. TBR=wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1797004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4338 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
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28e20752806a492f5a6a5d343c02f9556f39b1cd |
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10-Jul-2013 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds trunk/talk folder of revision 359 from libjingles google code to trunk/talk git-svn-id: http://webrtc.googlecode.com/svn/trunk@4318 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/talk/media/webrtc/fakewebrtcvideoengine.h
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