ff761fba8274d93bd73e76c8b8a1f2d0776dd840 |
|
04-Nov-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
modules: more interface -> include renames This changes the following module directories: * webrtc/modules/audio_conference_mixer/interface * webrtc/modules/interface * webrtc/modules/media_file/interface * webrtc/modules/rtp_rtcp/interface * webrtc/modules/utility/interface To avoid breaking downstream, I followed this recipe: 1. Copy the interface dir to a new sibling directory: include 2. Update the header guards in the include directory to match the style guide. 3. Update the header guards in the interface directory to match the ones in include. This is required to avoid getting redefinitions in the not-yet-updated downstream code. 4. Add a pragma warning in the header files in the interface dir. Example: #pragma message("WARNING: webrtc/modules/interface is DEPRECATED; " "use webrtc/modules/include") 5. Search for all source references to webrtc/modules/interface and update them to webrtc/modules/include (*.c*,*.h,*.mm,*.S) 6. Update all GYP+GN files. This required manual inspection since many subdirectories of webrtc/modules referenced the interface dir using ../interface etc(*.gyp*,*.gn*) BUG=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 -m tryserver.webrtc R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1417683006 . Cr-Commit-Position: refs/heads/master@{#10500}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
|
98f53510b222f71fdd8b799b2f33737ceeb28c61 |
|
28-Oct-2015 |
Henrik Kjellander <kjellander@webrtc.org> |
system_wrappers: rename interface -> include BUG=webrtc:5095 R=tommi@webrtc.org Review URL: https://codereview.webrtc.org/1413333002 . Cr-Commit-Position: refs/heads/master@{#10438}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
|
ebc0b4e99365443111857a0c7cfcc8944d8f1b6e |
|
28-Oct-2015 |
Peter Boström <pbos@webrtc.org> |
Use webrtc/base/logging.h for rtp_rtcp. BUG=webrtc:5118 R=stefan@webrtc.org Review URL: https://codereview.webrtc.org/1422023002 . Cr-Commit-Position: refs/heads/master@{#10437}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
|
e4f96501fc5b3e6de0d1ccd262372afcda1f5b4f |
|
21-Oct-2015 |
tommi <tommi@webrtc.org> |
Remove system_wrappers/interface/trace_event.h BUG= Review URL: https://codereview.webrtc.org/1417773002 Cr-Commit-Position: refs/heads/master@{#10346}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
|
ac547a653862744d0aae560713f8418ad2852085 |
|
17-Sep-2015 |
Peter Boström <pbos@webrtc.org> |
Remove channel ids from various interfaces. Starts by removing channel/engine id from ViEChannel which propagates down to the RTP/RTCP module as well as the transport class. IncomingVideoStream::RenderFrame() is untouched for now but receives a fake id instead of the previous channel id. Added a TODO to remove it later but the RenderFrame call is implemented in a lot of platform-dependent files and should probably remove the "manager" aspect of renderers, so preferring to do it separately BUG=webrtc:1695 R=henrika@webrtc.org, mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1335353005 . Cr-Commit-Position: refs/heads/master@{#9978}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
|
91d6edef35e7275879c30ce16ecb8b6dc73c6e4a |
|
17-Sep-2015 |
henrikg <henrikg@webrtc.org> |
Add RTC_ prefix to (D)CHECKs and related macros. We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition. Alternative solutions: * Check if we already have defined e.g. CHECK, and don't define them in that case. This makes us depend on include order in Chromium, which is not acceptable. * Don't allow using the macros in WebRTC headers. Error prone since if someone adds it there by mistake it may compile fine, but later break if a header in added or order is changed in Chromium. That will be confusing and hard to enforce. * Ensure that headers that are included by an embedder don't include our macros. This would require some heavy refactoring to be maintainable and enforcable. * Changes in Chromium for this is obviously not an option. BUG=chromium:468375 NOTRY=true Review URL: https://codereview.webrtc.org/1335923002 Cr-Commit-Position: refs/heads/master@{#9964}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
|
bd2522abf75891f34da6f83c247c47ca95641cee |
|
01-Jul-2015 |
pbos <pbos@webrtc.org> |
Fail RTP parsing on excessive padding length. BUG=webrtc:4771 R=stefan@webrtc.org Review URL: https://codereview.webrtc.org/1220863002 Cr-Commit-Position: refs/heads/master@{#9525}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
|
64c1e8cda5cb4db85c5c296bf2f6a8181af7de9d |
|
02-Apr-2015 |
Guo-wei Shieh <guoweis@chromium.org> |
Enable CVO by default through webrtc pipeline. All RTP packets from sender side will carry the rotation info. (will file a bug to track this) On the receiving side, only packets with marker bit set will be examined. Tests completed: 1. android standalone to android standalone 2. android standalone to chrome (with and without this change) 3. android on chrome BUG=4145 R=glaznev@webrtc.org, mflodman@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org Committed: https://crrev.com/1b1c15cad16de57053bb6aa8a916079e0534bdae Cr-Commit-Position: refs/heads/master@{#8905} Review URL: https://webrtc-codereview.appspot.com/47399004 Cr-Commit-Position: refs/heads/master@{#8917}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
|
31331cfd2d3d17958942b67190c8b943c05b084f |
|
01-Apr-2015 |
Minyue <minyue@webrtc.org> |
Revert "Enable CVO by default through webrtc pipeline." This reverts commit 1b1c15cad16de57053bb6aa8a916079e0534bdae. Due to failure on http://build.chromium.org/p/client.webrtc/builders/Linux64%20Release%20%5Blarge%20tests%5D/builds/4092 and following builds (the test hangs and never finishes). R=kjellander@webrtc.org TBR=guoweis@chromium.org TESTED=Local revert + execution of libjingle_peerconnection_java_unittest show that this is the culprit. Review URL: https://webrtc-codereview.appspot.com/47909004 Cr-Commit-Position: refs/heads/master@{#8911}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
|
1b1c15cad16de57053bb6aa8a916079e0534bdae |
|
01-Apr-2015 |
Guo-wei Shieh <guoweis@chromium.org> |
Enable CVO by default through webrtc pipeline. All RTP packets from sender side will carry the rotation info. (will file a bug to track this) On the receiving side, only packets with marker bit set will be examined. Tests completed: 1. android standalone to android standalone 2. android standalone to chrome (with and without this change) 3. android on chrome BUG=4145 R=glaznev@webrtc.org, mflodman@webrtc.org, perkj@webrtc.org, pthatcher@webrtc.org Review URL: https://webrtc-codereview.appspot.com/47399004 Cr-Commit-Position: refs/heads/master@{#8905}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
|
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/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
|
1b32bbe0a78adfe5f2d38561ba6d90b754239cd4 |
|
09-Mar-2015 |
mflodman@webrtc.org <mflodman@webrtc.org> |
Removing private and unused method in RTPReceiver. R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/42269004 Cr-Commit-Position: refs/heads/master@{#8650} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8650 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
|
00b8f6b3643332cce1ee711715f7fbb824d793ca |
|
26-Feb-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away BUG= R=andrew@webrtc.org Review URL: https://webrtc-codereview.appspot.com/36229004 Cr-Commit-Position: refs/heads/master@{#8517} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8517 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
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0200f70792982c4b5987cf4364dcd53f8aa94779 |
|
16-Feb-2015 |
sprang@webrtc.org <sprang@webrtc.org> |
Set webrtc_rtp category to be disabled by default. Should not affect webrtc standalone. For chromium, disabling helps mitigate viewing performance problems. BUG=chromium:441440 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/41909004 Cr-Commit-Position: refs/heads/master@{#8375} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8375 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
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026b892e724c3f47bde92d773d84099768e57ec8 |
|
30-Jan-2015 |
pkasting@chromium.org <pkasting@chromium.org> |
Using << on an int8_t or uint8_t will output a character rather than a number. Places that do this need to cast to int to get the desired behavior. BUG=none TEST=none R=henrik.lundin@webrtc.org, pthatcher@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/40579004 Cr-Commit-Position: refs/heads/master@{#8223} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8223 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
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4591fbd09f9cb6e83433c49a12dd8524c2806502 |
|
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/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
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d42a3adf429ad27779bea1789f53b76e52388583 |
|
07-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove partially defined WebRtcRTPHeader from Parse(). It' bit ugly that RtpDepacketizer::ParsedPayload partially defines WebRtcRTPHeader, and then sent to Parse() function for internal change. To make it clearer, the CL gets rid of using partially-defined WebRtcRTPHeader. BUG= R=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/28919004 Patch from Changbin Shao <changbin.shao@intel.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@7660 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
|
730d2707713c4240070af17e56edd10d039bafd2 |
|
29-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove callback from RtpDepacketizer::Parse(). BUG= R=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/30489004 Patch from Changbin Shao <changbin.shao@intel.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@7318 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
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b5e6bfc76a32a588da2400636688d34a71a2f47d |
|
12-Sep-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Make RTPSender/RTPReceiver generic. Changes include, 1) Introduce class RtpPacketizerGeneric & RtpDePacketizerGeneric. 2) Introduce class RtpDepacketizerVp8. 3) Make RTPSenderVideo::SendH264 generic and used by all packetizers. 4) Move codec specific functions from RTPSenderVideo/RTPReceiverVideo to RtpPacketizer/RtpDePacketizer sub-classes. R=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26399004 Patch from Changbin Shao <changbin.shao@intel.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@7163 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
|
2ec560606be6519dc4e32a1e6855b0f362ca498d |
|
31-Jul-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add H.264 packetization. This also includes: - Creating new packetizer and depacketizer interfaces. - Moved VP8 packetization was H264 packetization and depacketization to these interfaces. This is a work in progress and should be continued to get this 100% generic. This also required changing the return type for RtpFormatVp8::NextPacket(), which now returns bool instead of the index of the first partition. - Created a Create() factory method for packetizers and depacketizers. R=niklas.enbom@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21009004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6804 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
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62bafae6618fe3aefbd18657062abc98a40c3375 |
|
08-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Some refactoring inside rtp_rtcp/. Renaming ModuleRTPUtility -> RtpUtility. Renaming RTPHeaderParser -> RtpHeaderParser. Making RtpHeaderParser accept size_t instead of int for packet length. Making RtpUtility::RtpHeaderParser accept size_t for packet length. BUG= R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19899004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6623 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
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b9f5453e2997253addb87706a43b4484e1139972 |
|
04-Jul-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add boilerplate code for H.264. R=mflodman@webrtc.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/17849005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6603 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
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dc80bae2a62a1bdbe0d342b3260a7e5b2cb958df |
|
08-Apr-2014 |
andresp@webrtc.org <andresp@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Convert logs in rtp rtcp module from WEBRTC_TRACE into LOG. Clean some logs and add asserts in the way. BUG=3153 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/11129004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5861 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
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30e055c4dd708636df46c6d76964c7f984dbec46 |
|
08-Sep-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Handle empty RTP video packets agnostic to codec. Sending empty RTP packets caused a crash when using a generic codec instead of VP8. This fix moves handling of empty RTP packets out of ReceiveVp8Codec and into ParseRtpPacket. BUG= R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2185004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4701 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
|
7bb8f02274ecbfa1f7ef134d708369a369a78c83 |
|
06-Sep-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds support for combining RTX and FEC/RED. This is accomplished by breaking out RTX and FEC/RED functionality from the RTP module and keeping track of the base payload type, that is the payload type received when not receiving RTX. Enables retransmissions over RTX by default in the loopback test. BUG=1811 TESTS=voe/vie_auto_test --automated and trybots. R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2154004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4692 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
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822fbd8b68ffdb481b9557e2950ae8d6657c8ce6 |
|
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/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
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12dc1a38ca54a000e4fecfbc6d41138b895c9ca5 |
|
05-Aug-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Switch C++-style C headers with their C equivalents. The C++ headers define the C functions within the std:: namespace, but we mainly don't use the std:: namespace for C functions. Therefore we should include the C headers. BUG=1833 R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1917004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4486 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
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aa4d96a134a03f998d52fb9699845d9c644eb24b |
|
16-Jul-2013 |
tnakamura@webrtc.org <tnakamura@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert r4301 R=mikhal@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1809004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4357 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
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66b2e5c05a3f2a93d634d1dbbcbb283fb218ca4f |
|
05-Jul-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Breaking out receive-stats, rtp-payload-registry and rtp-receiver from the rtp_rtcp implementation. This refactoring significantly reduces the receive-side RTP parser and receiver complexity, and makes it possible to implement RTX correctly by having two instances of receive-statistics. With this change the dead-or-alive and packet timeout APIs are removed. TEST=trybots, vie_auto_test, voe_auto_test BUG=1811 R=mflodman@webrtc.org, pbos@webrtc.org, xians@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1745004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4301 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
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a5cb98cbbd11e93cb6d0a6232387814aac168c7d |
|
29-May-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Breaking out RTP header parsing from the RTP module. This is the first step in order to move bandwidth estimation closer to the network. The goal is to have RTP header parsing and bandwidth estimation before voice and video engine, and have a joint estimate for audio and video. Moving bandwidth estimation before the RTP module is also required for RTX. TEST=vie_auto_test, voe_auto_test, trybots. BUG=1811 R=andresp@webrtc.org, henrika@webrtc.org, mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1545004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4129 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
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806dc3b0e62ec68f594e9aadab601b2db7e6c6d5 |
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09-Apr-2013 |
hclam@chromium.org <hclam@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
More trace events The goal of this change is to unify tracing events styles and add trace events for all RTP traffic. BUG=1555 Review URL: https://webrtc-codereview.appspot.com/1290007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3806 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
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2f44673d665899ca788ae44247a9a7f4764f5e2b |
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08-Apr-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
WebRtc_Word32 => int32_t for rtp_rtcp/ BUG=314 Review URL: https://webrtc-codereview.appspot.com/1279007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3777 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
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79b0289bfc9f425d15442b1ecd73c2ae69646326 |
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04-Apr-2013 |
edjee@google.com <edjee@google.com@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds event traces and counters for WebRTC receive side. Review URL: https://webrtc-codereview.appspot.com/1279005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3766 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
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8911ce46a4c76c09b8c58828532836c9cd95549d |
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18-Mar-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Generic video-codec support. Labels frames as key/delta, also marks the first RTP packet of a frame as such, to allow proper reconstruction even if packets are received out of order. BUG=1442 TBR=ajm@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1207004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3680 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
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03e3117d87e7b70d2658cdd4141b1bc5239ba11d |
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12-Mar-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Removed redundant VP8 width/height and made sure the generic width/height is set. Review URL: https://webrtc-codereview.appspot.com/1158005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3656 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
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a7303bdfb5c2f16e1c8d7189a2a315a6f0b5373f |
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05-Feb-2013 |
phoglund@webrtc.org <phoglund@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Lint-cleaned video and audio receivers. BUG= TESTED=trybots Review URL: https://webrtc-codereview.appspot.com/1093004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3471 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
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244251a9cd283575b27b0b4ab3beddb069e6fa9d |
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04-Feb-2013 |
phoglund@webrtc.org <phoglund@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Moved almost all payload-related stuff to the payload registry. The big benefit is we no longer have a circular dependency between the media receiver and the payload registry. The payload registry is starting to take a bit more place on the stage, and now knows how to do different things depending on audio or video. BUG= TESTED=rtp_rtcp_unittests, vie_auto_test, voe_auto_test Review URL: https://webrtc-codereview.appspot.com/1078004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3465 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
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5accd370e70b94517a39e622c75b794cc7a28829 |
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22-Jan-2013 |
phoglund@webrtc.org <phoglund@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
RTP Receiver is now only deals with a receiver strategy. Cleaned up dependencies. BUG= TESTED=vie/voe_auto_test, rtp_rtcp_unittests Review URL: https://webrtc-codereview.appspot.com/1058004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3397 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
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efae5d59016ebdf959bf5970e36edcd31c9d9867 |
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17-Jan-2013 |
phoglund@webrtc.org <phoglund@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Extracted rtp receiver payload management to its own class, made video receiver depend on that instead. Eliminated need for video receiver to talk to its parent. Also we will now determine if the packet is the first one already in the rtp general receiver. The possible downside would be that recovered video packets no longer can be flagged as the first packet, but I don't think that can happen. Even if it can happen, maybe the bit was set anyway at an earlier stage. The tests run fine. BUG= TEST=rtp_rtcp_unittests, vie_auto_test, voe_auto_test Review URL: https://webrtc-codereview.appspot.com/1022011 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3382 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
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a22a9bd9ca66e98f2d51ea082dec8481f2f39e6e |
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14-Jan-2013 |
phoglund@webrtc.org <phoglund@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Cleaned up the data path for payload data, made callbacks to rtp_receiver nonoptional. The audio receiver is now completely independent of rtp_receiver: video will hopefully be too in the next patch. BUG= TEST=vie & voe_auto_test full runs Review URL: https://webrtc-codereview.appspot.com/1014006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3372 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
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07bf43c67303db4ab64b44f5b849465ec7dfef75 |
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18-Dec-2012 |
phoglund@webrtc.org <phoglund@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Replaced the _audio parameter with a strategy. The purpose is to make _rtpReceiver mostly agnostic to if it processes audio or video, and make its delegates responsible for that. This patch makes the actual interfaces and interactions between the classes a lot clearer which will probably help straighten out the rather convoluted business logic in here. There are a number of rough edges I hope to address in coming patches. In particular, I think there are a lot of audio-specific hacks, especially when it comes to telephone event handling. I think we will see a lot of benefit once that stuff moves out of rtp_receiver altogether. The new strategy I introduced doesn't quite pull its own weight yet, but I think I will be able to remove a lot of that interface later once the responsibilities of the classes becomes move cohesive (e.g. that audio specific stuff actually lives in the audio class, and so on). Also I think it should be possible to extract payload type management to a helper class later on. BUG= TEST=vie/voe_auto_test, trybots Review URL: https://webrtc-codereview.appspot.com/1001006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3306 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
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7659d914bb201d65d1829ed0f0344adeac2fac49 |
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14-Dec-2012 |
phoglund@webrtc.org <phoglund@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Decoupled video rtp receiver from rtp receiver. BUG= Review URL: https://webrtc-codereview.appspot.com/995005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3292 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
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ef90c3227ebd4008bbcfabd17a9f422965f11a25 |
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26-Nov-2012 |
phoglund@webrtc.org <phoglund@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Will now correctly identify the first-ever received packet as the first packet in its frame. We used to flag the _second_ packet in the first frame as the first. Subsequent frames worked as intended. BUG=1103 TEST=vie_auto_test --automated, rtp_rtcp_unittests Review URL: https://webrtc-codereview.appspot.com/964020 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3164 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
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1726661ca26245c4b871d9144b64f605f52862b6 |
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13-Nov-2012 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update parsed non ref frame info. Review URL: https://webrtc-codereview.appspot.com/932015 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3084 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
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14b43beb7ce4440b30dcea31196de5b4a529cb6b |
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22-Oct-2012 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move src/ -> webrtc/ TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/915006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_receiver_video.cc
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