History log of /external/webrtc/webrtc/modules/rtp_rtcp/source/rtp_sender.cc
Revision Date Author Comments (<<< Hide modified files) (Show modified files >>>)
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
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
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.cc
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
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
545727ecce444320328b825d65b287e844dca7cb 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
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
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 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
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
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
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
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
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
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
d16e839c6d29831e79312180085b6a19149df43f 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
11d8176cb3383a2f96e118ff054e92e97a8d9db4 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
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
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 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
afcc6101d01be8c6cd9cf246dcf5b37b31ce0cd0 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
2f44673d665899ca788ae44247a9a7f4764f5e2b 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
e1a719386935a72d9489fcd7a078bf8fd76eb39f 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
bfacda60be5f816a04bd278d4aa4cd3d8fd01e9f 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
8911ce46a4c76c09b8c58828532836c9cd95549d 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
bda7f305c5d7d675f1c35813bd2b2a5732775bb9 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
becf9c897c41eea3f021f99d87889c32c78b0de9 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
43da54a458a7a992c702d85f0327e1d394ec5cf3 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
a678a3baee2e680bd521f3a6caf97707fffd6093 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
20ed36dada62ad56ec01263fc0eef0ed198f6476 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
c38eef896a483c5d4a2975d76060c9942031a94a 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
571a1c035be6b0afd7f357001bef775c51ec9364 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
c66e8b3f31db39d96bec6dc9ee9439455415a2be 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
e5b49a0472b97fa262b641b78cf4230bd824296f 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
14b43beb7ce4440b30dcea31196de5b4a529cb6b 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