f6975f46131981f83e0c88d276dee6b6c5753180 |
|
28-Dec-2015 |
danilchap <danilchap@webrtc.org> |
[rtp_rtcp] Lint errors cleaned from rtp_utility R=åsapersson BUG=webrtc:5277 Review URL: https://codereview.webrtc.org/1539423003 Cr-Commit-Position: refs/heads/master@{#11131}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
6db6cdc604f9a866991ecf8454eb7f7aa69918ea |
|
15-Dec-2015 |
danilchap <danilchap@webrtc.org> |
[rtp_rtcp] fixed lint errors in rtp_rtcp module that are not fixed in other CLs BUG=webrtc:5277 R=mflodman Review URL: https://codereview.webrtc.org/1513303003 Cr-Commit-Position: refs/heads/master@{#11025}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
b8b6fbb7a5d2f5a14f7f6f81c253747aa28e4c7f |
|
10-Dec-2015 |
danilchap <danilchap@webrtc.org> |
lint build/include errors fixed in rtp_rtcp module BUG=webrtc:5277 R=mflodman Review URL: https://codereview.webrtc.org/1505993003 Cr-Commit-Position: refs/heads/master@{#10971}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
162abd3562d7b08ab36569800d757b52739b9249 |
|
10-Dec-2015 |
danilchap <danilchap@webrtc.org> |
lint whitespace warning removed from most rtp_rtcp/source/ files rtcp_utility, rtp_utility, tmmbr_help, rtcp_receiver, rtcp_receiver_help are explicetly excluded from the cleanup becaues there are short plans (or cls) to do a deeper cleaning there. BUG=webrtc:5277 R=pbos@webrtc.org, mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1512493002 Cr-Commit-Position: refs/heads/master@{#10966}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
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_sender_unittest.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_sender_unittest.cc
|
0a87ffcaad6a5e8cd4ead9c4d4957bd8a77fd7f2 |
|
21-Oct-2015 |
Stefan Holmer <stefan@webrtc.org> |
Fix bug in how send timestamps are converted to 24 bits. BUG=webrtc:4173 R=sprang@webrtc.org Review URL: https://codereview.webrtc.org/1412683004 . Cr-Commit-Position: refs/heads/master@{#10356}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
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/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
e23e737177cf5d131a6d4a4d229aa513c5270a59 |
|
08-Oct-2015 |
Peter Boström <pbos@webrtc.org> |
Disable pacer disabling. Since the pacer is always enabled, removing enable/disable which makes all packet queueing succeed. Also renaming one of the ::SendPackets ::InsertPacket to avoid confusion. BUG=webrtc:1695, webrtc:2629 R=stefan@webrtc.org Review URL: https://codereview.webrtc.org/1392513002 . Cr-Commit-Position: refs/heads/master@{#10211}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
1d8a506405734d0cef9653704b036ca4f1388960 |
|
02-Oct-2015 |
stefan <stefan@webrtc.org> |
Add a PacketOptions struct to webrtc::Transport. This allows us to pass packet meta data, such as transport sequence number, to libjingle and further down to the socket implementation. A similar struct already exist in libjingle, see rtc::PacketOptions in asyncpacketsocket.h. BUG=4173 Review URL: https://codereview.webrtc.org/1376673004 Cr-Commit-Position: refs/heads/master@{#10144}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
38778b046f058565bd4bae266f79c46cde806aa1 |
|
29-Sep-2015 |
sprang <sprang@webrtc.org> |
Add unit test for nack bandwidth constraint. BUG= Review URL: https://codereview.webrtc.org/1341743002 Cr-Commit-Position: refs/heads/master@{#10111}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
2d566686a23fe93ada58f1c38a0d4b9a0d68556e |
|
28-Sep-2015 |
pbos <pbos@webrtc.org> |
Unify Transport and newapi::Transport interfaces. BUG=webrtc:1695 R=stefan@webrtc.org TBR=mflodman@webrtc.org Review URL: https://codereview.webrtc.org/1369263002 Cr-Commit-Position: refs/heads/master@{#10096}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
ebbf8a805b45613b4cb118e4eb0cebe7eeee69ac |
|
22-Sep-2015 |
sprang <sprang@webrtc.org> |
Make sure rtp_rtcp module doesn't directly reference anything in the pacer module, and remove build dependencies on it. BUG= Review URL: https://codereview.webrtc.org/1350163005 Cr-Commit-Position: refs/heads/master@{#10005}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
586b19bdb615dde34cdcf107272d8857fe2f5631 |
|
18-Sep-2015 |
Stefan Holmer <stefan@webrtc.org> |
Enable probing with repeated payload packets by default. To make this possible padding only packets will have the same timestamp as the previously sent media packet, as long as RTX is not enabled. This has the side effect that if we send only padding for a long time without sending media, a receive-side jitter buffer could potentially overflow. In practice this shouldn't be an issue, partly because RTX is recommended and used by default, but also because padding typically is terminated before being received by a client. It is also not an issue for bandwidth estimation as long as abs-send-time is used instead of toffset. BUG=chromium:425925 R=mflodman@webrtc.org, sprang@webrtc.org, tommi@webrtc.org Review URL: https://codereview.webrtc.org/1327933003 . Cr-Commit-Position: refs/heads/master@{#9984}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.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_sender_unittest.cc
|
4cee419e0777dcbfbd0837e26bed202e35e696a9 |
|
10-Aug-2015 |
Minyue <minyue@webrtc.org> |
Separating voice activity flag from audio level in RtpHeaderExtension. VAD flag was embedded in RtpHeaderExtension.audioLevel, which is not easy to interpret. This CL tries to separate the flag with the actual audio level. BUG= R=andrew@webrtc.org, henrik.lundin@webrtc.org Review URL: https://codereview.webrtc.org/1272343003 . Cr-Commit-Position: refs/heads/master@{#9691}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
867fb5224e1ba6a1c2cd523c005499a93ed61a08 |
|
03-Aug-2015 |
sprang <sprang@webrtc.org> |
Add support for transport wide sequence numbers Also refactor packet router to use a map rather than iterate over all rtp modules for each packet sent. BUG=webrtc:4311 Review URL: https://codereview.webrtc.org/1247293002 Cr-Commit-Position: refs/heads/master@{#9670}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.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_sender_unittest.cc
|
c56ac1ec298630ba95e44a9da9efeb9d1a6d43d4 |
|
04-May-2015 |
Karl Wiberg <kwiberg@webrtc.org> |
rtc::Buffer: Remove backwards compatibility band-aids This CL makes two changes to rtc::Buffer that have had to wait for Chromium's use of it to be modernized: 1. Change default return type of rtc::Buffer::data() from char* to uint8_t*. uint8_t is a more natural type for bytes, and won't accidentally convert to a string. (Chromium previously expected the default return type to be char, which is why rtc::Buffer::data() initially got char as default return type in 9478437f, but that's been fixed now.) 2. Stop accepting void* inputs in constructors and methods. While this is convenient, it's also dangerous since any pointer type will implicitly convert to void*. (This was previously committed (9e1a6d7c) but had to be reverted (cbf09274) because Chromium on Android wasn't quite ready for it). TBR=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/47109004 Cr-Commit-Position: refs/heads/master@{#9132}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
cbf0927473c10a0a25bbf55707f1ca2b2fd57708 |
|
30-Apr-2015 |
Karl Wiberg <kwiberg@webrtc.org> |
Revert "rtc::Buffer: Remove backwards compatibility band-aids" This reverts commit 9e1a6d7c236c9a8a322bef54d4ec2a087e5baa07, because Chromium for Android still isn't happy with it. TBR=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/49869004 Cr-Commit-Position: refs/heads/master@{#9122}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
9e1a6d7c236c9a8a322bef54d4ec2a087e5baa07 |
|
30-Apr-2015 |
Karl Wiberg <kwiberg@webrtc.org> |
rtc::Buffer: Remove backwards compatibility band-aids This CL makes two changes to rtc::Buffer that have had to wait for Chromium's use of it to be modernized: 1. Change default return type of rtc::Buffer::data() from char* to uint8_t*. uint8_t is a more natural type for bytes, and won't accidentally convert to a string. (Chromium previously expected the default return type to be char, which is why rtc::Buffer::data() initially got char as default return type in 9478437f, but that's been fixed now.) 2. Stop accepting void* inputs in constructors and methods. While this is convenient, it's also dangerous since any pointer type will implicitly convert to void*. R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/44269004 Cr-Commit-Position: refs/heads/master@{#9121}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
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/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
9478437fdea4eb31b92ffe0c10368fe5bc9b9e16 |
|
20-Apr-2015 |
Karl Wiberg <kwiberg@webrtc.org> |
rtc::Buffer improvements 1. Constructors, SetData(), and AppendData() now accept uint8_t*, int8_t*, and char*. Previously, they accepted void*, meaning that any kind of pointer was accepted. I think requiring an explicit cast in cases where the input array isn't already of a byte-sized type is a better compromise between convenience and safety. 2. data() can now return a uint8_t* instead of a char*, which seems more appropriate for a byte array, and is harder to mix up with zero-terminated C strings. data<int8_t>() is also available so that callers that want that type instead won't have to cast, as is data<char>() (which remains the default until all existing callers have been fixed). 3. Constructors, SetData(), and AppendData() now accept arrays natively, not just decayed to pointers. The advantage of this is that callers don't have to pass the size separately. 4. There are new constructors that allow setting size and capacity without initializing the array. Previously, this had to be done separately after construction. 5. Instead of TransferTo(), Buffer now supports swap(), and move construction and assignment, and has a Pass() method that works just like std::move(). (The Pass method is modeled after scoped_ptr::Pass().) R=jmarusic@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/42989004 Cr-Commit-Position: refs/heads/master@{#9033}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
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61c2a6f241ac9db626aeab755e49897030b289e1 |
|
16-Apr-2015 |
Karl Wiberg <kwiberg@webrtc.org> |
Remove rtc::Buffer::length(), since no one uses it anymore Chromium now uses size() instead, just like WebRTC. This CL also fixes a new length() call that had crept in. R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/44119004 Cr-Commit-Position: refs/heads/master@{#9024}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
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fcf54bdabbdf495cef7aa587b5cabdcb899ba24f |
|
14-Apr-2015 |
mflodman <mflodman@webrtc.org> |
Reland "Avoid critsect for protection- and qm setting callbacks in VideoSender." The original Cl is uploaded as patch set 1, the fix in ps#2 and I'll rebase in ps#3. BUG=4534 R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/46769004 Cr-Commit-Position: refs/heads/master@{#9000}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
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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_sender_unittest.cc
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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_sender_unittest.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_sender_unittest.cc
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0828a0c09440cb7edbfacc94d362bf08414c7655 |
|
31-Mar-2015 |
mflodman <mflodman@webrtc.org> |
Revert "Avoid critsect for protection- and qm setting callbacks in VideoSender." This reverts commit 903c0f2e7649a2b98659286dc228447facd49bb7, aka #8899. TBR=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/46759004 Cr-Commit-Position: refs/heads/master@{#8901}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
903c0f2e7649a2b98659286dc228447facd49bb7 |
|
31-Mar-2015 |
mflodman <mflodman@webrtc.org> |
Avoid critsect for protection- and qm setting callbacks in VideoSender. This CL avoids changing the mentioned callbacks during a call, to avoid a potential deadlock when acquiring _sendCritSect and calling _mediaOpt.SetTargetRates. Moving the critsect revealed a race for the FEC parameters in RtpVideoSender, so the CL grew a bit to avoid this. I also cleaned up some code here at the same time, but tried to keep it at a minimum since this CL had already increased a lot in size. BUG=769 R=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/42939004 Cr-Commit-Position: refs/heads/master@{#8899}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
eebcab5ce99d3e8641dd92a569916b0d24e29fca |
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24-Mar-2015 |
kwiberg@webrtc.org <kwiberg@webrtc.org> |
rtc::Buffer: Rename length to size, for conformance with the STL And add a constructor for creating an uninitialized Buffer of a specified size. (I intend to follow up with more Buffer changes, but since it's rather widely used, the rename is quite noisy and works better as a separate CL.) R=tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/48579004 Cr-Commit-Position: refs/heads/master@{#8841} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8841 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
30933904797ab220a7a1532a535904f9d8ee3149 |
|
17-Mar-2015 |
sprang@webrtc.org <sprang@webrtc.org> |
Parsing of transport wide sequence number rtp extension header. Plus some refactoring to correctly handle padding. BUG=4311 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/45429004 Cr-Commit-Position: refs/heads/master@{#8757} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8757 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.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_sender_unittest.cc
|
4536289353cdcc315cc5e6218893e4843cf528e6 |
|
04-Mar-2015 |
guoweis@webrtc.org <guoweis@webrtc.org> |
Add CVO support to RTP sender side. According to http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/ts_126114v120700p.pdf, CVO byte should only be added in the last packet of each key frame or when the rotation changes. Currently, we're adding this byte in each frame to start with. BUG=4145 R=mflodman@webrtc.org, pthatcher@webrtc.org Review URL: https://webrtc-codereview.appspot.com/42439004 Cr-Commit-Position: refs/heads/master@{#8606} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8606 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
14665ff7d4024d07e58622f498b23fd980001871 |
|
04-Mar-2015 |
kjellander@webrtc.org <kjellander@webrtc.org> |
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro Clang version changed 223108:230914 Details: https://chromium.googlesource.com/chromium/src/+/e144d30..6fdb142/tools/clang/scripts/update.sh Removes the OVERRIDE macro defined in: * webrtc/base/common.h * webrtc/typedefs.h The majority of the source changes were done by running this in src/: perl -0pi -e "s/virtual\s([^({;]*(\([^({;]*\)[^({;]*))(OVERRIDE|override)/\1override/sg" `find {talk,webrtc} -name "*.h" -o -name "*.cc*" -o -name "*.mm*"` which converted all: virtual Foo() OVERRIDE functions to: Foo() override Then I manually edited: * talk/media/webrtc/fakewebrtccommon.h * webrtc/test/fake_common.h Remaining uses of OVERRIDE was fixed by search+replace. Manual edits were done to fix virtual destructors that were overriding inherited ones. Finally a build error related to the pure virtual definitions of Read, Write and Rewind in common_types.h required a bit of refactoring in: * webrtc/common_types.cc * webrtc/common_types.h * webrtc/system_wrappers/interface/file_wrapper.h * webrtc/system_wrappers/source/file_impl.cc This roll should make it possible for us to finally re-enable deadlock detection for TSan on the buildbots. BUG=4106 R=pbos@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/41069004 Cr-Commit-Position: refs/heads/master@{#8596} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8596 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.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_sender_unittest.cc
|
4414939954fd908b6490ce1bb88271e161219aa3 |
|
04-Feb-2015 |
asapersson@webrtc.org <asapersson@webrtc.org> |
Add method for incrementing RtpPacketCounter. Removes duplicate code. Correction to check if rtx is enabled on send-side (and not receive) when updating rtx send bitrate stat. Remove unneeded guarded by annotations. BUG= R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/41729004 Cr-Commit-Position: refs/heads/master@{#8239} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8239 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
cfd82dfc1156f6610388bec0ebbdeacaf47e9719 |
|
22-Jan-2015 |
asapersson@webrtc.org <asapersson@webrtc.org> |
Split packets/bytes in StreamDataCounter into RtpPacketCounter struct. Prepares for adding FEC bytes to the StreamDataCounter. R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/37579004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8122 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
0b0c24177bac6eaa27cd520595ba799e48e84a0c |
|
13-Jan-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Only return Rtx mode in RTXSendStatus(). There is no need to return 'ssrc' and 'payloadtype' inside this function since they are never used. BUG= R=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/38569004 Patch from Changbin Shao <changbin.shao@intel.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@8049 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
8f27fcce79584378da97f0d84574564799e138d6 |
|
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/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
2a169640a3225a559f926fe74f1fe1af239e191f |
|
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/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
ce4e9a356200170abcdd44ff2af95f87a6781b8e |
|
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/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
d08d389ce836238030ec31e45c5f9a5535e0855f |
|
16-Dec-2014 |
asapersson@webrtc.org <asapersson@webrtc.org> |
Add field to counters for when first rtp/rtcp packet is sent/received. Use this time for histogram statistics (send/receive bitrates, sent/received rtcp fir/nack packets/min). BUG=crbug/419657 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/32219004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7910 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
97d0489058ae7a983f7306f32cfd49a2356c6488 |
|
09-Dec-2014 |
asapersson@webrtc.org <asapersson@webrtc.org> |
Add video send bitrates to histogram stats: - total bitrate ("WebRTC.Video.BitrateSentInKbps") - media bitrate ("WebRTC.Video.MediaBitrateSentInKbps") - rtx bitrate ("WebRTC.Video.RtxBitrateSentInKbps") - padding bitrate ("WebRTC.Video.PaddingBitrateSentInKbps") - retransmitted bitrate ("WebRTC.Video.RetransmittedBitrateInKbps") Add retransmitted bytes to StreamDataCounters. Change in UpdateRtpStats to also update counters for retransmitted packet. BUG=crbug/419657 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/30199004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7838 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
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_sender_unittest.cc
|
0bae1fab4adb9bb8164e53142bf419049eafec38 |
|
05-Nov-2014 |
stefan@webrtc.org <stefan@webrtc.org> |
Wire up bandwidth stats to the new API and webrtcvideoengine2. Adds stats to verify bandwidth and pacer stats. BUG=1788 R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/24969004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7634 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
3cefbc99f4cc2db744cb130ca629768401a59eb4 |
|
10-Oct-2014 |
xians@webrtc.org <xians@webrtc.org> |
Mark all virtual overrides in the hierarchy of Transport as virtual + OVERRIDE. This also marks all virtual overrides of other classes in the same files. This will make a subsequent change I intend to do safer, where I'll change the argument types of the base Transport functions, by breaking the compile if I miss any overrides. This also highlighted a number of unused functions. I've removed some of these. TBR=mflodman@webrtc.org, pkasting@chromium.org BUG=none TEST=none Review URL: https://webrtc-codereview.appspot.com/28709004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7421 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
0a214ffa8ad5c2d52d0f2d20bf5f1d686994f552 |
|
03-Sep-2014 |
stefan@webrtc.org <stefan@webrtc.org> |
Setting marker bit on DTMF correctly BUG=1157 R=braveyao@webrtc.org, pbos@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/22469004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7037 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
2f4b14e3f31b34a50310357c6c7be86c3bca1537 |
|
15-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make RTCP sender report send media bytes. r6654 changed RtpSender::Bytes() to return the number of bytes sent instead of number of media bytes. This is used by VideoEngine for stats. This change broke RTCP which sends this same count as the number of payload bytes sent (excluding headers and padding). BUG= R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/14959004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6691 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
168f23faa5b8a49d4dd709c6649e77d5fecf36bf |
|
11-Jul-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move pacer to fully use webrtc::Clock instead of webrtc::TickTime. This required rewriting the send-side delay stats api to be callback based, as otherwise the SuspendBelowMinBitrate test started flaking much more frequently since it had lock order inversion problems. R=pbos@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21869005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6664 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
72491b9a90bfd4e2339f42e353560c9c33875151 |
|
10-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Count total bytes sent in RTPSender::Bytes(). Previously only media bytes were included, this adds header bytes and padding bytes to the calculation. BUG= R=stefan@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/19939004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6654 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
8f1512140ed57ce57635a1cd561b631dfdc5e05f |
|
10-Jul-2014 |
andresp@webrtc.org <andresp@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Refactor registerable callbacks for FrameCountObserver from rtp_rtcp module into vie_channel. R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16839004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6649 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
d11bec40b25e5990bf05b410676587f6f38b9b8c |
|
08-Jul-2014 |
andresp@webrtc.org <andresp@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Refactor registerable callbacks for VideoBitrateObserver from rtp_rtcp module into vie_channel. R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20879004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6626 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
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_sender_unittest.cc
|
a15fbfdcdee391bd87bb1c2721f0fbb824f5fbfb |
|
17-Jun-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add round-robin selection of send stream to pad on. BUG=1812 R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/21669004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6472 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
ef92755780253c6a7940c89598a206e58e05b810 |
|
05-Jun-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Have RTX be enabled by setting an RTX payload type instead of by setting an RTX SSRC. This makes it easier to disable RTX by filtering out the RTX codec during call setup/signaling, and won't require that also the SSRCs are filtered out. BUG=1811 R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/15629005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6335 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
ebdb0e3ad0a787bee066d12cdcd115a38b0a10d1 |
|
07-Mar-2014 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Help to land 7969005 on behalf of solenberg. The review and try is done in 7969005. - Add ability to VoE to send Absolute Sender Time header extension. - Refactor handling of RTP header extensions in VoE to work the same as in ViE. - Add API to enable receiving Absolute Sender Time in VoE. This is part of the work to include audio packets in bandwidth estimation, for better accuracy in estimates. BUG= TBR=solenberg@webrtc.org,henrikg@webrtc.org,stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9509004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5654 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
6811b6e308d16f160ba4c32650f195d5d3d9a2b1 |
|
13-Dec-2013 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Callback for send bitrate estimates - new roll Issue https://webrtc-codereview.appspot.com/4459004/ was commited as r5259, after which flakiness was detected and a rollback was performed at r5261. Patch Set 1 of this issue is the code submitted in r5259. Subsequent patch sets fixes a race condition which caused the seen problems. The root cause was a dead lock between a thread sending rtp packets and and a timed module processing thread: webrtc::RTPSender::BitrateUpdated() // Get RTPSender stats lock webrtc::Bitrate::Process() // Get Bitrate lock webrtc::RTPSender::ProcessBitrate() webrtc::ModuleRtpRtcpImpl::Process() ... webrtc::Bitrate::Update() // Get Bitrate lock webrtc::RTPSender::UpdateRtpStats() // Get RTPSender stats lock webrtc::RTPSender::SendToNetwork() ... This is fixed in Bitrate::Process() by releasing the lock before calling the callback. BUG=2235 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5619004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5281 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
096e8d9f944abeee5fb75d165d91f7a68258f073 |
|
11-Dec-2013 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert 5259 "Callback for send bitrate estimates" CL is causing flakiness in RampUpTest.WithoutPacing. > Callback for send bitrate estimates > > BUG=2235 > R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org > > Review URL: https://webrtc-codereview.appspot.com/4459004 R=mflodman@webrtc.org, pbos@webrtc.org TBR=mflodman Review URL: https://webrtc-codereview.appspot.com/5579005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5261 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
2656cf9f4c37fe1360e2392a5b0101df38660403 |
|
11-Dec-2013 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Callback for send bitrate estimates BUG=2235 R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4459004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5259 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
88615f0659948ff0cb87e6e467ea650b304b030d |
|
06-Dec-2013 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix use of uninitialized memory in RtpSenderTest::StreamDataCountersCallbacks BUG= R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/5229004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5236 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
ebad765ee00b90c48507bff1997ea8c1070a9316 |
|
05-Dec-2013 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add callbacks for send channel rtp statistics BUG=2235 R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4449004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5227 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
71f055fb41336316324942f828e022e2f7d93ec7 |
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04-Dec-2013 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add send frame rate statistics callback BUG=2235 R=mflodman@webrtc.org, pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4479005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5213 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
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7e9315b42ebe8f7df860030af93618de81326503 |
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04-Dec-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds support for sending redundant payloads over RTX. TEST=trybots BUG=1812 R=mflodman@webrtc.org, pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4169004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5209 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
|
6e95d7afab12dcc6cd3893210baf56d49df74ea0 |
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15-Nov-2013 |
henrik.lundin@webrtc.org <henrik.lundin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Increment RTP timestamps for padding packets This CL makes the padding packets get their own RTP timestamps, rather than having the same timestamp as the last sent video packet. The purpose is to solve Issue 2611, where the overuse- detector does not react to padding packets. A test was implemented to verify that the padding packets do get their own timestamps. BUG=2611 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3869004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5125 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
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9b82f5a6ed2ceb04f72b66c1d3cca67a2bbcec3a |
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13-Nov-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix for RTX in combination with pacing. Retransmissions didn't get sent over RTX when pacing was enabled since the pacer didn't keep track of whether a packet was a retransmit or not. BUG=1811 TEST=trybots R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/3779004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5117 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
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7bb8f02274ecbfa1f7ef134d708369a369a78c83 |
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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_sender_unittest.cc
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8ccb9f9716f306dd1ec284b4f61f0b6c82c08c3c |
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19-Jun-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fixes some pacer/padding issues found while testing. - A bug was introduced in r4234 causing no paced packets to be sent. - Only update the sequence number counter if a padding packet is actually going to be sent, to avoid packet loss. - Have all packets go through the pacer if pacing is enabled to avoid reordering. - Fix race condition on reading capture_time_ms_/timestamp_ in rtp_sender.cc. BUG=1837 TEST=trybots and vie_auto_test --automated R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1682004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4246 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
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508a84b25597a8d12177eabed002b71f5730d4c8 |
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17-Jun-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Wire up pacer-based padding. This connects the pacer-based padding with the RTP modules, which will generate padding packets roughly according to what the pacer suggests. It will only generate padding packets of maximum size to keep the number off padding packets as small as possible. This also sets a limit of how much padding + media bitrate which the pacer is allowed to "request" from the RTP modules. Padding will for now only be generated by the first sending RTP module. BUG=1837 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1612005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4234 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
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fa64a595adef6beefa07caaf65e2dcde44d0be04 |
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03-Jun-2013 |
wu@webrtc.org <wu@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Change SetRTPAudioLevelIndicationStatus to ignore the id in the case of disabling. This makes it easier for the users of the interface, i.e. doesn't need to remember the id in order to disable audio level indication later. BUG=1828 TEST=unit tests R=henrika@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1598005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4157 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
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a048d7cb0a5bad5ca49bbcc5273cb4cca28c1710 |
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29-May-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Include files from webrtc/.. paths in rtp_rtcp/ BUG=1662 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1557004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4135 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
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a5cb98cbbd11e93cb6d0a6232387814aac168c7d |
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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_sender_unittest.cc
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5c58f63d3fbce3f894a583a438c164b00c0b15dc |
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23-May-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix regression where retransmission bitrate is no longer estimated. BUG=1813 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1530004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4095 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
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c0352d566a4291cf587c25ca023e44b52ad7484e |
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20-May-2013 |
solenberg@webrtc.org <solenberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix assertions in rtp_header_extension.h caused by not handling the AudioLevel extension. Added unit tests to do basic checks of the AudioLevel extension. BUG= R=asapersson@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1510004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4069 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
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7ebbea14a956c87f6f6aebb839486b9f12fcdf52 |
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16-May-2013 |
solenberg@webrtc.org <solenberg@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add handling of the absolute send time header extension to the rtp_rtcp module. BUG= R=asapersson@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1480004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4041 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
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3004c79c6ad0ca4b4df27d0ca76c2eb29735e267 |
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07-May-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix clang errors in non-GYP_DEFINES=clang=1 build BUG=1623 R=stefan@webrtc.org, tina.legrand@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1368004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3968 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
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b0061f94b23062aa10c45f967dff622287bd68dc |
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27-Apr-2013 |
pwestin@webrtc.org <pwestin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Enable Nack pacing. Review URL: https://webrtc-codereview.appspot.com/1357004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3912 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.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_sender_unittest.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_sender_unittest.cc
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a678a3baee2e680bd521f3a6caf97707fffd6093 |
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21-Jan-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Move video_coding to new Clock interface and remove fake clock implementations from RTP module tests. TEST=video_coding_unittests, video_coding_integrationtests, rtp_rtcp_unittests, trybots Review URL: https://webrtc-codereview.appspot.com/1044004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3393 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
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20ed36dada62ad56ec01263fc0eef0ed198f6476 |
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17-Jan-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Break out RtpClock to system_wrappers and make it more generic. The goal with this new clock interface is to have something which is used all over WebRTC to make it easier to switch clock implementation depending on where the components are used. This is a first step in that direction. Next steps will be to, step by step, move all modules, video engine and voice engine over to the new interface, effectively deprecating the old clock interfaces. Long-term my vision is that we should be able to deprecate the clock of WebRTC and rely on the user providing the implementation. TEST=vie_auto_test, rtp_rtcp_unittests, trybots Review URL: https://webrtc-codereview.appspot.com/1041004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3381 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
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571a1c035be6b0afd7f357001bef775c51ec9364 |
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13-Nov-2012 |
pwestin@webrtc.org <pwestin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Enable paced sender. Review URL: https://webrtc-codereview.appspot.com/965016 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3089 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
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c66e8b3f31db39d96bec6dc9ee9439455415a2be |
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07-Nov-2012 |
pwestin@webrtc.org <pwestin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
pre-factor cleanup pre-work. Review URL: https://webrtc-codereview.appspot.com/938010 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3054 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.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_sender_unittest.cc
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