6955870806624479723addfae6dcf5d13968796c |
|
13-Jan-2016 |
Peter Kasting <pkasting@google.com> |
Convert channel counts to size_t. IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan. BUG=chromium:81439 TEST=none R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org Review URL: https://codereview.webrtc.org/1316523002 . Cr-Commit-Position: refs/heads/master@{#11229}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
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.cc
|
1227e8b3451b1a2f2a765bf6101cb0862f625568 |
|
21-Dec-2015 |
danilchap <danilchap@webrtc.org> |
[rtp_rtcp] time helper functions RTP timestams helper functions moved from rtp_utility added functions to deal with CompactNtp timestamps R=åsapersson BUG=webrtc:5260 Review URL: https://codereview.webrtc.org/1535113002 Cr-Commit-Position: refs/heads/master@{#11106}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.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.cc
|
47a740bc5e36bcaf19385f9d4c0afb0cad070a05 |
|
15-Dec-2015 |
danilchap <danilchap@webrtc.org> |
[rtp_rtcp] lint errors about rand() usage fixed. rand() usage replaced with new Random class, which also makes it clearer what interval random number is in. BUG=webrtc:5277 R=mflodman Review URL: https://codereview.webrtc.org/1519503002 Cr-Commit-Position: refs/heads/master@{#11019}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.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.cc
|
b86d4e4a8dec1eb1b801244a2a97cda66f561d8e |
|
07-Dec-2015 |
Stefan Holmer <stefan@webrtc.org> |
Prepare the AudioSendStream to be hooked up to send-side BWE. This CL contains three changes as a preparation for adding audio send streams to the send-side BWE: 1. Audio packets are passed through the pacer with high priority. This is needed to be able to set transport sequence numbers on the packets. 2. A feedback observer is passed to the audio stream's rtcp receiver so that the BWE can get notified of any BWE feedback being received on the audio feedback channel. 3. Support for the transport sequence number header extension is added to audio send streams. BUG=webrtc:5263,webrtc:5307 R=mflodman@webrtc.org, solenberg@webrtc.org Review URL: https://codereview.webrtc.org/1479023002 . Cr-Commit-Position: refs/heads/master@{#10909}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.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.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.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_sender.cc
|
f116bd0d7a3cdad20bb638d5a87427bd920c8904 |
|
27-Oct-2015 |
stefan <stefan@webrtc.org> |
Call OnSentPacket for all packets sent in the test framework. Required a bit of refactoring to make it possible to pass a Call to DirectTransport on construction. This also lead to me having to remove the shared lock between PacketTransport and RtpRtcpObserver. Now RtpRtcpObserver has a SetTransports method instead of a SetReceivers method. BUG=webrtc:4173 Review URL: https://codereview.webrtc.org/1419193002 Cr-Commit-Position: refs/heads/master@{#10430}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
bbe876f0d30ec806c7c4a12629eb1f19ab45fb86 |
|
23-Oct-2015 |
stefan <stefan@webrtc.org> |
Set send times in send time history via OnSentPacket. BUG=webrtc:4173 Review URL: https://codereview.webrtc.org/1419503004 Cr-Commit-Position: refs/heads/master@{#10384}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.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.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_sender.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.cc
|
861c55e58311383b7f4f61af463ddea53eb3f30f |
|
16-Oct-2015 |
sprang <sprang@webrtc.org> |
Transport sequence number should be set also for retransmissions. This is a reland of https://codereview.webrtc.org/1385563005 which was reverted since the test was flaky. The reason was a race condition (in the test) and that NACK wasn't properly set up. BUG= Review URL: https://codereview.webrtc.org/1406193002 Cr-Commit-Position: refs/heads/master@{#10303}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
c1aeaf0dc37d96f31c92f893f4e30e7a5f7cc2b7 |
|
15-Oct-2015 |
stefan <stefan@webrtc.org> |
Wire up packet_id / send time callbacks to webrtc via libjingle. BUG=webrtc:4173 Review URL: https://codereview.webrtc.org/1363573002 Cr-Commit-Position: refs/heads/master@{#10289}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.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.cc
|
10950692d67af5cfdcb19d93b40f25193d1db8c6 |
|
06-Oct-2015 |
Alejandro Luebs <aluebs@webrtc.org> |
Revert "Transport sequence number should be set also for retransmissions." After this CL, video_engine_test started failing flakily in different bots for different CLs. TBR=sprang@webrtc.org Review URL: https://codereview.webrtc.org/1393553003 . Cr-Commit-Position: refs/heads/master@{#10188}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
af4ced986bc62c263fbdb6eab68aef5c0d4e7c78 |
|
06-Oct-2015 |
sprang <sprang@webrtc.org> |
Transport sequence number should be set also for retransmissions. When fetching a packet from the rtp packet history, cuased by a retransmission, the transport seq extension header is enabled but the sequence number is set to 0. A new transport seq should be assigned in this case. BUG= Review URL: https://codereview.webrtc.org/1385563005 Cr-Commit-Position: refs/heads/master@{#10183}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.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.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.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.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.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.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_sender.cc
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5e023eb337eed9746ecea7fc6fbb0fca386f1961 |
|
14-Sep-2015 |
sprang <sprang@webrtc.org> |
Add TransportFeedback adapter, adapting remote feedback to bwe estiamtor When using send-side bandwidth estimation, the inter-packet delay is reported back to the sender using RTCP TransportFeedback messages. Theis data needs to be translated into a format which the bandwidth estimator (now instantiated on the send side) can use, including looking up the local absolute send time from the send time history. BUG=webrtc:4173 Review URL: https://codereview.webrtc.org/1329083005 Cr-Commit-Position: refs/heads/master@{#9929}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
c32d2db69bc94480ecb312268c6e6769d4a1cac6 |
|
11-Sep-2015 |
pbos <pbos@webrtc.org> |
Refactor RTPPacketHistory to use a packet struct. Collects packet information within a struct instead of spreading it out over different vectors. Adds a fixed-size buffer to the stored packet instead of using vectors. BUG= R=stefan@webrtc.org Review URL: https://codereview.webrtc.org/1340573002 Cr-Commit-Position: refs/heads/master@{#9926}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
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867fb5224e1ba6a1c2cd523c005499a93ed61a08 |
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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.cc
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ba8c15b857c0f341d9c1e02d41b6ccd56f9f1030 |
|
14-Jul-2015 |
pbos <pbos@webrtc.org> |
Merge methods for configuring NACK/FEC/hybrid. BUG=webrtc:1695 R=stefan@webrtc.org Review URL: https://codereview.webrtc.org/1226143013 Cr-Commit-Position: refs/heads/master@{#9580}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
d6f1a38165455d743fbe61f6980f22be6a3c4de9 |
|
14-Jul-2015 |
Peter Boström <pbos@webrtc.org> |
Remove ViEChannel simulcast lock. Since the number of streams is now known on construction we can initialize all RTP modules on construction. They are internally locked so we don't nede a simulcast lock anymore. BUG=1695 R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/52639004 . Cr-Commit-Position: refs/heads/master@{#9577}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
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d436298332c7a7ecb51241f3a66588539c2ece83 |
|
07-Jul-2015 |
pbos <pbos@webrtc.org> |
Remove ResetStatistics from RTP feedback. BUG= R=asapersson@webrtc.org, stefan@webrtc.org Review URL: https://codereview.webrtc.org/1213603002 Cr-Commit-Position: refs/heads/master@{#9548}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
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545727ecce444320328b825d65b287e844dca7cb |
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01-Jul-2015 |
pbos <pbos@webrtc.org> |
Move early-return in TimeToSendPadding. Prevents taking send_critsect_ for checking sending status when not actually intending to send padding. BUG= R=stefan@webrtc.org Review URL: https://codereview.webrtc.org/1218093002 Cr-Commit-Position: refs/heads/master@{#9526}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
71861a0e2039e1729ad34758474d5e569012fd2f |
|
28-May-2015 |
Peter Boström <pbos@webrtc.org> |
Remove GetSendSideDelay from RtpRtcp. These stats are reported using a callback either way, removing a getter + an old related deadlock suppression. BUG=1695, 2999 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/50119004 Cr-Commit-Position: refs/heads/master@{#9314}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
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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.cc
|
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.cc
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6ae2572fa6dc29349b946e3cfd926289e54d9371 |
|
13-Apr-2015 |
Åsa Persson <asapersson@webrtc.org> |
Add missing configuration of rtx payload type for rtp/rtcp module. BUG=4528 R=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/51639004 Cr-Commit-Position: refs/heads/master@{#8989}
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.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_sender.cc
|
31331cfd2d3d17958942b67190c8b943c05b084f |
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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.cc
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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.cc
|
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.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.cc
|
779c3d16b9623f38a72439bc013102aeb0077a62 |
|
17-Mar-2015 |
sprang@webrtc.org <sprang@webrtc.org> |
Use ByteReader/ByteWriter instead of rtputility and manual shift/add. BUG= R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/41289004 Cr-Commit-Position: refs/heads/master@{#8761} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8761 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
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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.cc
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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.cc
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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.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.cc
|
49096de442f6131e90925daff6dc9888d085de00 |
|
24-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
DCHECK send DataCountersUpdated for valid SSRCs. Also updates RTPSender to not update RTX stats when RTX is disabled. BUG= R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/42399004 Cr-Commit-Position: refs/heads/master@{#8489} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8489 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
d324546ced76d4e792338af4f7d02a5cd8819f92 |
|
23-Feb-2015 |
pkasting@chromium.org <pkasting@chromium.org> |
Misc. cleanup split out of https://webrtc-codereview.appspot.com/37699004/ : * Move constants into the files/functions that use them * Declare variables in the narrowest scope possible * Use correct (expected, actual) order for gtest macros * Remove unused functions * Untabify * 80-column limit * Avoid C-style casts * Prefer true typed constants to "enum hack" constants * Print size_t using the right format macro * Shorten and simplify code * Other random cleanup bits and style fixes BUG=none TEST=none R=henrik.lundin@webrtc.org, tommi@webrtc.org Review URL: https://webrtc-codereview.appspot.com/36179004 Cr-Commit-Position: refs/heads/master@{#8467} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8467 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
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_sender.cc
|
7c4d20fd6c95f76cf909669b94effdbef05ecb54 |
|
12-Feb-2015 |
pbos@webrtc.org <pbos@webrtc.org> |
Remove potential deadlock in RTPSenderAudio. Removes lock-order inversion formed by RTPSenderAudio->RTPSender calls by doing a lot shorter locking which fetches a current state of RTPSenderAudio variables before sending. Thread annotates locked variables and removes one lock in RTPSenderAudio, bonus fixes data races reported in voe_auto_test --automated under TSan (DTMF data race). Also includes some bonus cleanup of RTPSenderVideo which removes the send critsect completely as all methods using it was always called from RTPSender under its send_critsect. R=henrik.lundin@webrtc.org, stefan@webrtc.org, tommi@webrtc.org BUG=3001, chromium:454654 Review URL: https://webrtc-codereview.appspot.com/41869004 Cr-Commit-Position: refs/heads/master@{#8348} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8348 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.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.cc
|
c957ffc6dc36879e5ad72d7f0af2a014707d70fa |
|
02-Feb-2015 |
sprang@webrtc.org <sprang@webrtc.org> |
Fixed potential crash if rtp packet history is completely full. Also performance enhanecement in rtp_sender (don't lookup if kDontStore) BUG=4171 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/39759004 Cr-Commit-Position: refs/heads/master@{#8226} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8226 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
43c883954f5edc84bd8e0e901ef770fead218ed5 |
|
29-Jan-2015 |
sprang@webrtc.org <sprang@webrtc.org> |
Allow rtp packet history to dynamically expand in size. When using the paced sender, packets will be put into the rtp packet history and then retreived from there again when it is time to send. In some cases (low send bitrate and very large frames created) this may overflow, causing packets to be overwritten in the packet history before they have been sent. Check this condition and expand history size if needed. This is primarily triggered during screenshare, when switching to a large picture with lots of high frequency details in it. BUG=4171 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/34879004 Cr-Commit-Position: refs/heads/master@{#8195} git-svn-id: http://webrtc.googlecode.com/svn/trunk@8195 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
273fbbb921e61273c3d83eb494d0a68db7834d3d |
|
27-Jan-2015 |
asapersson@webrtc.org <asapersson@webrtc.org> |
Update StreamDataCounter with FEC bytes. Add histograms stats for send/receive FEC bitrate: - "WebRTC.Video.FecBitrateReceivedInKbps" - "WebRTC.Video.FecBitrateSentInKbps" Correct media payload bytes in StreamDataCounter to not include FEC bytes. Fix stats for rtcp packets sent/received per minute (regression from r7910). BUG=crbug/419657 R=holmer@google.com, mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/34789004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@8164 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.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.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.cc
|
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/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
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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.cc
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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.cc
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d16e839c6d29831e79312180085b6a19149df43f |
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19-Dec-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Rtp-Rtcp sender cleanup. Some setter functions from Rtp and Rtcp Sender never return negative values. Remove return results from those functions. Also removed const on non-pointer/reference types for related files. BUG= R=henrika@webrtc.org, pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/34469004 Patch from Changbin Shao <changbin.shao@intel.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@7962 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
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11d8176cb3383a2f96e118ff054e92e97a8d9db4 |
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19-Dec-2014 |
stefan@webrtc.org <stefan@webrtc.org> |
Move updating nack bitrate inside UpdateNACKBitRate. BUG= R=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/32819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7960 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
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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.cc
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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.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.cc
|
9334ac2d78f760b37f512ef6c12bff220d1654c1 |
|
24-Nov-2014 |
pbos@webrtc.org <pbos@webrtc.org> |
Use vector of CSRCs for DeliverFrame & SetCSRCs. BUG= R=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/28029004 Patch from Changbin Shao <changbin.shao@intel.com>. git-svn-id: http://webrtc.googlecode.com/svn/trunk@7734 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.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.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.cc
|
dcebf2daa76aebd021dbb778f3908375b819e59a |
|
04-Nov-2014 |
sprang@webrtc.org <sprang@webrtc.org> |
Reworked paced sender queue Packet queue in the paced sender is now based on a priority queue rather than having a separate fifo-queue per priority level. This allows more flexible sorting and cleaner usage. Packets with earlier capture times are now prioritized higher. In situations with high packet loss, the queue might contain packets from several subsequent frames. Retransmit packets from the earlier frames first, since the later ones will probably be dependent on these. Also, don't force sending of packets after a certain time of inactivity or when packets grow too old, since this was causing consistent overuse on poor connections. Instead, drop frames in vie encoder if pacer queue is too long. BUG= R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/27869004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7617 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
76960d5f742194ca2de6c900603dc72124bdcf4d |
|
22-Oct-2014 |
stefan@webrtc.org <stefan@webrtc.org> |
For FIR packet, payload length is zero, so SendToNetwork function is failing. R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/23059004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7490 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
c3c29113d1733a4f97ca4b8e212f22f718a876b7 |
|
27-Aug-2014 |
andresp@webrtc.org <andresp@webrtc.org> |
Expose setPayloadType on the rtp_sender. Thus allowing other users of this module to set the payload type to be used without having to call SendOutgoingData. BUG=3694 R=asapersson@webrtc.org Review URL: https://webrtc-codereview.appspot.com/18289004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6988 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
817a034cf25ea2232c54ac2f3afcffe85bd50c47 |
|
14-Aug-2014 |
andresp@webrtc.org <andresp@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix TimeToSendPadding return to be 0 if no padding bytes are sent. BUG=3694 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/15149005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6900 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
8b94e3da0f35638529d6640e4dfcd7f04057d3f4 |
|
17-Jul-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix issue where padding is sent before media with undefined timestamps if not abs-send-time is enabled. This broke bandwidth estimation for calls without abs-send-time is enabled, but where RTX was. BUG= R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/16929004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6719 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
63c60ed22457d45444d29b33a622ea2bedd12ea5 |
|
16-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Remove old padding path in RTPSender. Removing RTPSender::SendPaddingAccordingToBitrate() as well as a couple of arguments from SendPadData(). BUG= R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/14989004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6703 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.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.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.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.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.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.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.cc
|
2bb1bdab8d11f5445693c028335fb3ace631f636 |
|
07-Jul-2014 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Preserve RTP states for restarted VideoSendStreams. A restarted VideoSendStream would previously be completely reset, causing gaps in sequence numbers and potentially RTP timestamps as well. This broke SRTP which requires fairly sequential sequence numbers. Presumably, were this sent without SRTP, we'd still have problems on the receiving end as the corresponding receiver is unaware of this reset. Also adding annotation to RTPSender and addressing some unlocked access to ssrc_, ssrc_rtx_ and rtx_. BUG= R=mflodman@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/20819004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6612 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
88e0dda475e1f6a5fa5855eec0be111bddbf00ac |
|
04-Jul-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Introduces PacedVideoSender to test framework and moves the Pacer to use Clock. R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/14729004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6600 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
fe526ff10fea5cc9f456f9a9313499f19bd7c8d0 |
|
25-Jun-2014 |
henrike@webrtc.org <henrike@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
fix after r6472 in rtp_sender, comparison between signed and unsigned integer expressions. BUG=N/A R=pwestin@webrtc.org, wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/15909004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6539 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.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.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.cc
|
420b2567f38241099907d30d8bece1c4db50262d |
|
30-May-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix bug where RTP headers in the packet history were replaced with the RTX wrapped headers. This caused only the first retransmission to be successful. Introduced with https://code.google.com/p/webrtc/source/detail?r=5728. BUG=1811 R=asapersson@webrtc.org, mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12609005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@6284 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
2c3f1abb69376e66cf15e5fb6fe5bcd88f185aca |
|
15-Apr-2014 |
andrew@webrtc.org <andrew@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Replace flooding logs in rtp_sender.cc with a comment. Started occurring after: https://webrtc-codereview.appspot.com/11129004 BUG=3153 R=andresp@webrtc.org, mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/11439004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5916 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
2f8d5f330279f42ac79174dbbc2e4722f5cf535e |
|
15-Apr-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Check if a header extension is registered before updating it and fail silently if it's not. BUG= R=andresp@webrtc.org Review URL: https://webrtc-codereview.appspot.com/12039004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5909 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
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_sender.cc
|
d09d0748270f40c35330837069523245839b7258 |
|
26-Mar-2014 |
andresp@webrtc.org <andresp@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Protect write of send_target_bitrate. This issue was catch by tsan bot. BUG=3065 R=stefan@webrtc.org, andrew Review URL: https://webrtc-codereview.appspot.com/10619004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5790 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
efcad39f778276296ef45e2f14427154841e911f |
|
25-Mar-2014 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix race condition in RTPSEnder. In RTPSender::SendPayloadType(), payload_type_ should not be read without owning send_critsect_. BUG= R=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8199004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5778 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
16395228f5a6ae6f5d4f85441873d8408f5c11d6 |
|
19-Mar-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Properly account for retransmitted packets when not using the pacer. This regression was introduced in r5728. TBR=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/10269004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5729 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
7c6ff2da261699628e7253d9d10068bc531fe0f8 |
|
19-Mar-2014 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fixes RTX related bugs. - An RTX packet with no payload should be dropped prior to parsing RTX header since it doesn't have an RTX header. This can for example happen when sending padding-only packets over the RTX stream. - The retransmit code path when the pacer is disabled doesn't properly update the abs-send-time and ts-offset header extensions. TEST=trybots R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9189004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5728 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
5a320fb06fadb8378b76112556473af7b9e0c82a |
|
13-Mar-2014 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Race condition in RTPSender RTPSender::sending_media_ should be guarded by send_critsect_. Fix this in GetSendSideDelay, SendPadData and TimeToSendPadding. Also add appropriate thread annotations. BUG=3029 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/9779004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5697 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.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.cc
|
346094cb01ef2ffbf0398f465d61c9a4f77b465c |
|
18-Feb-2014 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Incorrect overhead calculation when using FEC + RTP extension headers. When frames are fragmented inte multiple RTP packets in order to not exceed a maximum packet size, the header overhead calculation must take into account that FEC redundancy packets may use more than the 12 bytes of the basic RTP header. For example, a csrc list or extension headers may be present. BUG=2899 R=phoglund@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/8769005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5562 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
5314e859262c03e8c212fee53245e91851c1e5cc |
|
27-Jan-2014 |
sprang@webrtc.org <sprang@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Race condition in RTPSender::UpdateRtpStats The ssrc should not be access directly from the ssrc_ field, without holding the send_critsect_ lock. A better way is to just use the SSRC() getter method. BUG= R=pbos@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/7539006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5439 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.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.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.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.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.cc
|
0a3c1471b873ea7f81bff2faa7cf0d9e563c7d53 |
|
05-Dec-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add API to query video engine for the send-side delay. R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/4559005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5225 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
71f055fb41336316324942f828e022e2f7d93ec7 |
|
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.cc
|
79b63206b99912d9a5f97a35b546409886a8fed2 |
|
04-Dec-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fixes a crash in fullstack tests introduced with r5209. TBR=mflodman@webrtc.org BUG=1812 Review URL: https://webrtc-codereview.appspot.com/4689005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@5211 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
7e9315b42ebe8f7df860030af93618de81326503 |
|
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.cc
|
6e95d7afab12dcc6cd3893210baf56d49df74ea0 |
|
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.cc
|
9b82f5a6ed2ceb04f72b66c1d3cca67a2bbcec3a |
|
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.cc
|
e07049f19f7ca7a9ab7bc91acbfa24cbac3f8031 |
|
10-Sep-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Lock RTPSender statistics. Suppressing these errors in TSan has become tedious. It's better to just lock them. BUG=2349 R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2197004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4713 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
59f20bb735562d245357609799578edeed46be32 |
|
09-Sep-2013 |
pbos@webrtc.org <pbos@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Break out RTCPSender dependency on ModuleRtpRtcpImpl. BUG= R=stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2191004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4706 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
d4f607e70ad85102b77cf0beba8f11e2599e8f99 |
|
19-Aug-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fixes to padding when driven by encoder. - Allow padding to be sent on an ssrc which doesn't produce video, therefore never having the last_packet_marker_bit_ set. - Add the random timestamp offset to all padding packets. - Store the capture time of padding packets to properly create an offset. BUG=2258 TEST=trybots and a new test which will be committed separately. R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/2060005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4566 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
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_sender.cc
|
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_sender.cc
|
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_sender.cc
|
1a7b9b94be10119224c58edcddebf9ad398331ce |
|
08-Jul-2013 |
hclam@chromium.org <hclam@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Cleanup WebRTC tracing The goal of this change is to: 1. Remove unused tracing events. 2. Organize tracing events to facilitate measurement of end to end latency. The major change in this CL is to use ASYNC_STEP such that operation flow can be traced for the same frame. R=marpan@webrtc.org, pwestin@webrtc.org, turaj@webrtc.org, wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1761004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4308 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
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_sender.cc
|
a4c5abb52a4677ea576c5076ce36df33bb6c9cba |
|
25-Jun-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Make sure padding packets are sent. BUG=1837 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1717006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4260 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
2e402ce873f48e0848468345d848bd3fff75dd3e |
|
20-Jun-2013 |
hclam@chromium.org <hclam@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Enqueue packet in pacer if sending fails If a packet cannot be sent while pacer is in use it should be queued. This avoid packet loss due to congestion. BUG=1930 R=pwestin@webrtc.org, wu@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1693004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4250 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
8ccb9f9716f306dd1ec284b4f61f0b6c82c08c3c |
|
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.cc
|
508a84b25597a8d12177eabed002b71f5730d4c8 |
|
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.cc
|
a817962bab1602a0229cb1d450bae55f22d9bd74 |
|
04-Jun-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Refactor padding and rtp header functionality. BUG=1837 R=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1611004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4172 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
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_sender.cc
|
c74c3c244784fc1d6cea53ecb2dccfe353394e6a |
|
23-May-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adds integration test for RTX and fixes bugs found. BUG=1811 R=mflodman@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1529004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4096 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
5c58f63d3fbce3f894a583a438c164b00c0b15dc |
|
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.cc
|
c0352d566a4291cf587c25ca023e44b52ad7484e |
|
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.cc
|
7ebbea14a956c87f6f6aebb839486b9f12fcdf52 |
|
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.cc
|
7bfb3a322738fdf79a8d2498fd35c00bcc4617a7 |
|
14-May-2013 |
justinlin@chromium.org <justinlin@chromium.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add more tracing for key frames. R=mallinath@webrtc.org, stefan@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1428004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@4015 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
3004c79c6ad0ca4b4df27d0ca76c2eb29735e267 |
|
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.cc
|
52b4e8871a7c43a12177cb9a717baff3fb2680c0 |
|
02-May-2013 |
pwestin@webrtc.org <pwestin@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adding trace and changing pacing constants BUG=1721,1722 R=mikhal@webrtc.org, niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1380005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3940 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
b0061f94b23062aa10c45f967dff622287bd68dc |
|
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.cc
|
9f5ebb525130f207229dfa350ce8c2bdd22163c7 |
|
12-Apr-2013 |
mflodman@webrtc.org <mflodman@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adding a payload type for RTX. BUG=736 TEST=Modified RTP unittests. Review URL: https://webrtc-codereview.appspot.com/1278004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3843 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
|
806dc3b0e62ec68f594e9aadab601b2db7e6c6d5 |
|
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_sender.cc
|
7da3459b2ac83923c1ccbf11ad24d3f700305feb |
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09-Apr-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Revert "With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps." This reverts commit 4954b3650192d78037714138a5c519ef08f2670e. Reverts r3799 TBR=pbos@webrtc.org Review URL: https://webrtc-codereview.appspot.com/1308004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3802 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
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afcc6101d01be8c6cd9cf246dcf5b37b31ce0cd0 |
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09-Apr-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
With these changes we will assume that the capture time of a frame is based on NTP time. This makes the interface of video engine more well defined and makes it easier and cleaner to handle user provided capture timestamps. We should consider making the same change to the render timestamps generated at the receiver. BUG=1563 Review URL: https://webrtc-codereview.appspot.com/1283005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3799 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.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.cc
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e1a719386935a72d9489fcd7a078bf8fd76eb39f |
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27-Mar-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix flakiness in network up/down event tests when running under memcheck. TBR=pwestin@webrtc.org BUG=1524 Review URL: https://webrtc-codereview.appspot.com/1261005 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3732 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
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bfacda60be5f816a04bd278d4aa4cd3d8fd01e9f |
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27-Mar-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Add interface to signal a network down event. - In real-time mode encoding will be paused until the network is back up. - In buffering mode the encoder will keep encoding, and packets will be buffered at the sender. When the buffer grows above the target delay encoding will be paused. - Fixes a couple of issues related to pacing which was found with the new test. - Introduces different max bitrates for pacing and for encoding. This allows the pacer to faster get rid of the queue after a network down event. (Work based on issue 1237004) BUG=1524 TESTS=trybots,vie_auto_test Review URL: https://webrtc-codereview.appspot.com/1258004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3730 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.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.cc
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bda7f305c5d7d675f1c35813bd2b2a5732775bb9 |
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16-Mar-2013 |
mikhal@webrtc.org <mikhal@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Adding RTX on source Review URL: https://webrtc-codereview.appspot.com/1190004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3674 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
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becf9c897c41eea3f021f99d87889c32c78b0de9 |
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01-Feb-2013 |
stefan@webrtc.org <stefan@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Fix mismatch between different NACK list lengths and packet buffers. This is a second version of http://review.webrtc.org/1065006/ which passes the parameters via methods instead of via constructors. BUG=1289 Review URL: https://webrtc-codereview.appspot.com/1065007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3456 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
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43da54a458a7a992c702d85f0327e1d394ec5cf3 |
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25-Jan-2013 |
phoglund@webrtc.org <phoglund@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Reformatted rtp_sender: made lint clean. TESTED=rtp_rtcp_unittests BUG= Review URL: https://webrtc-codereview.appspot.com/1062004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3412 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.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.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.cc
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c38eef896a483c5d4a2975d76060c9942031a94a |
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07-Jan-2013 |
phoglund@webrtc.org <phoglund@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Reformatted RTPReceiver. This is a pure reformat patch, with the exception that I also fixed all comments and moved a constant. I did not change the types in this patch since I though that is more risky, so I'll do that in a separate patch later (perhaps we could purge the types from the whole module in one go?) BUG= TEST=Trybots, vie_ & voe_auto_test --automated Review URL: https://webrtc-codereview.appspot.com/998007 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3338 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.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.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.cc
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e5b49a0472b97fa262b641b78cf4230bd824296f |
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06-Nov-2012 |
asapersson@webrtc.org <asapersson@webrtc.org@4adac7df-926f-26a2-2b94-8c16560cd09d> |
Update timestamp offset for re-transmitted packets. BUG=1059 Review URL: https://webrtc-codereview.appspot.com/930011 git-svn-id: http://webrtc.googlecode.com/svn/trunk@3046 4adac7df-926f-26a2-2b94-8c16560cd09d
/external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.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.cc
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