| 3842c5c7f73639527e653f41c65334245d2317a1 |
|
12-Jan-2016 |
Stefan Holmer <stefan@webrtc.org> |
Wire-up BWE feedback for audio receive streams.
Also wires up receiving transport sequence numbers.
BUG=webrtc:5263
R=mflodman@webrtc.org, pbos@webrtc.org, solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1535963002 .
Cr-Commit-Position: refs/heads/master@{#11220}
/external/webrtc/webrtc/call/call.cc
|
| 53805324c0fa904d796cc0b333868c591f2c5f2c |
|
21-Dec-2015 |
asapersson <asapersson@webrtc.org> |
Rename RTC_HISTOGRAM_* macros to RTC_HISTOGRAM_*_SPARSE_* to indicate that these are for infrequent updates.
This implementation will be replaced by a faster one and sparse will be removed.
BUG=webrtc:5283
Review URL: https://codereview.webrtc.org/1530913002
Cr-Commit-Position: refs/heads/master@{#11099}
/external/webrtc/webrtc/call/call.cc
|
| 7623ce4aeb9130c937ba5836490cbb3a35679e79 |
|
09-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
Reason for revert:
Bot breakage caused by TickTime::UseFakeClock has been removed.
Original issue's description:
> Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ )
>
> Reason for revert:
> Breaks Dr Memory Light https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Light/builds/4643 and all the Android Tests bots.
>
> Original issue's description:
> > Merge webrtc/video_engine/ into webrtc/video/
> >
> > BUG=webrtc:1695
> > R=mflodman@webrtc.org
> >
> > Committed: https://crrev.com/03ef053202bc5d5ab43460eebf5403232f157646
> > Cr-Commit-Position: refs/heads/master@{#10926}
>
> TBR=mflodman@webrtc.org,pbos@webrtc.org
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:1695
>
> Committed: https://crrev.com/8237abf563bf4782ee104408b53cc8e55ce44518
> Cr-Commit-Position: refs/heads/master@{#10937}
BUG=webrtc:1695
TBR=mflodman@webrtc.org,kjellander@webrtc.org
NOPRESUBMIT=true
Review URL: https://codereview.webrtc.org/1510183002 .
Cr-Commit-Position: refs/heads/master@{#10948}
/external/webrtc/webrtc/call/call.cc
|
| d3c944755ec546f46d5bdd84bff359fe6c4639e9 |
|
09-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
Nuke TickTime::UseFakeClock.
Removes the global simulated time that affects (or breaks) following
tests in the same binary and replaces it with SimulatedClock.
BUG=webrtc:5318
R=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1512853002 .
Cr-Commit-Position: refs/heads/master@{#10947}
/external/webrtc/webrtc/call/call.cc
|
| 8237abf563bf4782ee104408b53cc8e55ce44518 |
|
08-Dec-2015 |
kjellander <kjellander@webrtc.org> |
Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ )
Reason for revert:
Breaks Dr Memory Light https://build.chromium.org/p/client.webrtc/builders/Win%20DrMemory%20Light/builds/4643 and all the Android Tests bots.
Original issue's description:
> Merge webrtc/video_engine/ into webrtc/video/
>
> BUG=webrtc:1695
> R=mflodman@webrtc.org
>
> Committed: https://crrev.com/03ef053202bc5d5ab43460eebf5403232f157646
> Cr-Commit-Position: refs/heads/master@{#10926}
TBR=mflodman@webrtc.org,pbos@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:1695
Review URL: https://codereview.webrtc.org/1507903005
Cr-Commit-Position: refs/heads/master@{#10937}
/external/webrtc/webrtc/call/call.cc
|
| 03ef053202bc5d5ab43460eebf5403232f157646 |
|
08-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
Merge webrtc/video_engine/ into webrtc/video/
BUG=webrtc:1695
R=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1506773002 .
Cr-Commit-Position: refs/heads/master@{#10926}
/external/webrtc/webrtc/call/call.cc
|
| 6f28cf0b951a9d41246f022f48a6cd035fad151d |
|
07-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
Implement standalone event tracing in AppRTCDemo.
Logs tracing events (TRACE_EVENT0 and friends) to storage in a format
compatible with chrome://tracing which can be used for performance
evaluation, finding lock contention and other sweet things). Tracing is
still basic and doesn't contain thread metadata or logging of tracing
arguments.
BUG=webrtc:5158
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1457383002 .
Cr-Commit-Position: refs/heads/master@{#10921}
/external/webrtc/webrtc/call/call.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/call/call.cc
|
| 7c704b82893bbe7fc206b004fb9dfe6e69a986ef |
|
04-Dec-2015 |
Peter Boström <pbos@webrtc.org> |
Use webrtc/base/logging.h in stefan@'s ownership.
Replaces system_wrappers' logging in call/, bitrate_controller/, pacing/
and remote_bitrate_estimator/.
BUG=webrtc:5118
R=stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1484503002 .
Cr-Commit-Position: refs/heads/master@{#10896}
/external/webrtc/webrtc/call/call.cc
|
| ea07373a2eb46f2732a8b5acef06a9b5078f37f8 |
|
01-Dec-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Enable cpplint for webrtc/audio and webrtc/call, and fix all uncovered cpplint errors.
BUG=webrtc:5268,webrtc:5273
TESTED=Fixed issues reported by:
find webrtc/audio -type f -name *.cc -o -name *.h | xargs cpplint.py
find webrtc/call -type f -name *.cc -o -name *.h | xargs cpplint.py
followed by 'git cl presubmit'.
R=kjellander@webrtc.org, pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1483323002 .
Cr-Commit-Position: refs/heads/master@{#10853}
/external/webrtc/webrtc/call/call.cc
|
| 226befecfb5e56287482a2d6250f01d019761884 |
|
26-Nov-2015 |
Stefan Holmer <stefan@webrtc.org> |
Rewrote pacer and bandwidth UMA stats.
The new version measures receive bitrates from time of first packet to
time of last packet, and send/pacer BWE as the average BWE reported
while we have send streams.
R=asapersson@webrtc.org, pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1470373004 .
Cr-Commit-Position: refs/heads/master@{#10810}
/external/webrtc/webrtc/call/call.cc
|
| 18adf0a79d4a0ea124c7f27fd553382d0b0cb7ac |
|
17-Nov-2015 |
stefan <stefan@webrtc.org> |
Add UMA for send bwe and pacer bitrate.
Review URL: https://codereview.webrtc.org/1434403004
Cr-Commit-Position: refs/heads/master@{#10675}
/external/webrtc/webrtc/call/call.cc
|
| 0b9e29c87da2d9c1a3792d2c87197b0688b68e4e |
|
16-Nov-2015 |
Henrik Kjellander <kjellander@google.com> |
Remove include dirs from modules/{media_file,pacing}
Also move files out of media_file/source.
BUG=webrtc:5095
TESTED=git cl try -c --bot=android_compile_rel --bot=linux_compile_rel --bot=win_compile_rel --bot=mac_compile_rel --bot=ios_rel --bot=linux_gn_rel --bot=win_x64_gn_rel --bot=mac_x64_gn_rel --bot=android_gn_rel -m tryserver.webrtc
R=asapersson@webrtc.org, perkj@webrtc.org, tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1435093002 .
Cr-Commit-Position: refs/heads/master@{#10647}
/external/webrtc/webrtc/call/call.cc
|
| 0e7e259ebd69993bb5670a991f43aa1b06c9bf9e |
|
13-Nov-2015 |
mflodman <mflodman@webrtc.org> |
Move BitrateAllocator from BitrateController logic to Call.
This is a step on the way to have variable bitrate for audio and is
intended to be as much of a no-op as possible.
BUG=webrtc:5079
Review URL: https://codereview.webrtc.org/1441673002
Cr-Commit-Position: refs/heads/master@{#10630}
/external/webrtc/webrtc/call/call.cc
|
| 56a34df92807d95a2660765be10abef7c779666f |
|
12-Nov-2015 |
solenberg <solenberg@webrtc.org> |
Re-add a thread check in Call::Call that was removed by mistake in a rebase.
BUG=
Review URL: https://codereview.webrtc.org/1434263002
Cr-Commit-Position: refs/heads/master@{#10623}
/external/webrtc/webrtc/call/call.cc
|
| 91d926038f7cebf889ef843f2f087d72bc8c60c2 |
|
11-Nov-2015 |
stefan <stefan@webrtc.org> |
Add receive bitrate UMA stats.
Review URL: https://codereview.webrtc.org/1440603002
Cr-Commit-Position: refs/heads/master@{#10605}
/external/webrtc/webrtc/call/call.cc
|
| 566ef247b9779f6c9d0e7ec9eea6b037f4682c53 |
|
07-Nov-2015 |
solenberg <solenberg@webrtc.org> |
Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1403363003
Cr-Commit-Position: refs/heads/master@{#10548}
/external/webrtc/webrtc/call/call.cc
|
| 006d93d3c679ffd15223c327d649066a72400639 |
|
05-Nov-2015 |
terelius <terelius@webrtc.org> |
Added protobuf message for loss-based BWE events, and wired it up to the send side bandwidth estimator.
BUG=
Review URL: https://codereview.webrtc.org/1411673003
Cr-Commit-Position: refs/heads/master@{#10531}
/external/webrtc/webrtc/call/call.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/call/call.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/call/call.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/call/call.cc
|
| 85a0496b8c4ac01da7c716ea7950093659864c8e |
|
27-Oct-2015 |
solenberg <solenberg@webrtc.org> |
Implement AudioSendStream::GetStats().
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1414743004
Cr-Commit-Position: refs/heads/master@{#10424}
/external/webrtc/webrtc/call/call.cc
|
| 717432f13016d2668a584bfd864338ecffd106b2 |
|
26-Oct-2015 |
mflodman <mflodman@webrtc.org> |
Remove network_enabled_crit_ in call.cc.
After #10321 (5a289393928c18af580c6339ba77600fb67006e2) I don't see that
we still need this lock.
R=pbos@webrtc.org, solenberg@webrtc.org
Review URL: https://codereview.webrtc.org/1409193003 .
Cr-Commit-Position: refs/heads/master@{#10410}
/external/webrtc/webrtc/call/call.cc
|
| 4f4ec0a9270a8cefadfa12e9fa3b979b58b15392 |
|
22-Oct-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Re-Land: Implement AudioReceiveStream::GetStats().
R=tommi@webrtc.org
BUG=webrtc:4690
Committed: https://chromium.googlesource.com/external/webrtc/+/a457752f4afc496ed7f4d6b584b08d8635f18cc0
Review URL: https://codereview.webrtc.org/1390753002 .
Cr-Commit-Position: refs/heads/master@{#10369}
/external/webrtc/webrtc/call/call.cc
|
| 0c478b3d75be3c026e68f03a11cb558c3655c926 |
|
21-Oct-2015 |
mflodman <mflodman@webrtc.org> |
Rename ChannelGroup to CongestionController and move to webrtc/call/.
BUG=webrtc:5079
R=pbos@webrtc.org
Review URL: https://codereview.webrtc.org/1419803002 .
Cr-Commit-Position: refs/heads/master@{#10358}
/external/webrtc/webrtc/call/call.cc
|
| e37870297fc45f1253dff4b1c59c85a50d2a8a97 |
|
21-Oct-2015 |
mflodman <mflodman@webrtc.org> |
ChannelGroup cleanup.
Move CallStats to Call, EncoderStateFeedback to VideoSendStream and
remove last ViEChannel dependency from ChannelGroup.
BUG=webrtc:5079
R=pbos@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1418613002 .
Cr-Commit-Position: refs/heads/master@{#10355}
/external/webrtc/webrtc/call/call.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/call/call.cc
|
| 43e83d44f01683fbd304e37d47d2f6db0d52660d |
|
20-Oct-2015 |
solenberg <solenberg@webrtc.org> |
Revert of Implement AudioReceiveStream::GetStats(). (patchset #19 id:360001 of https://codereview.webrtc.org/1390753002/ )
Reason for revert:
webrtc_perf_tests started failing on Win32 Release, Mac32 Release and Linux64 Release (all running large tests). These were not caught by try bots.
Original issue's description:
> Implement AudioReceiveStream::GetStats().
>
> R=tommi@webrtc.org
> TBR=hta@webrtc.org
> BUG=webrtc:4690
>
> Committed: https://chromium.googlesource.com/external/webrtc/+/a457752f4afc496ed7f4d6b584b08d8635f18cc0
TBR=tommi@webrtc.org
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1411083006
Cr-Commit-Position: refs/heads/master@{#10340}
/external/webrtc/webrtc/call/call.cc
|
| a457752f4afc496ed7f4d6b584b08d8635f18cc0 |
|
20-Oct-2015 |
Fredrik Solenberg <solenberg@webrtc.org> |
Implement AudioReceiveStream::GetStats().
R=tommi@webrtc.org
TBR=hta@webrtc.org
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1390753002 .
Cr-Commit-Position: refs/heads/master@{#10338}
/external/webrtc/webrtc/call/call.cc
|
| 0dbf0090a961c5e5fb7362937108337564b4a91f |
|
19-Oct-2015 |
mflodman <mflodman@webrtc.org> |
Remove the video channel id completely.
BUG=webrtc:5079
Review URL: https://codereview.webrtc.org/1412143002
Cr-Commit-Position: refs/heads/master@{#10324}
/external/webrtc/webrtc/call/call.cc
|
| 5a289393928c18af580c6339ba77600fb67006e2 |
|
19-Oct-2015 |
solenberg <solenberg@webrtc.org> |
Added thread checker to webrtc::Call.
BUG=
Review URL: https://codereview.webrtc.org/1403353003
Cr-Commit-Position: refs/heads/master@{#10321}
/external/webrtc/webrtc/call/call.cc
|
| a20de2030f7f3a3c5e252ccc76a467109f5a93dc |
|
19-Oct-2015 |
mflodman <mflodman@webrtc.org> |
Move ownership of receive ViEChannel to VideoReceiveStream.
This CL changes as little as possible and I'll follow up later with
ownership of the other members in ChannelGroup.
The next step is to remove the id used for channels.
BUG=webrtc:5079
Review URL: https://codereview.webrtc.org/1411723002
Cr-Commit-Position: refs/heads/master@{#10318}
/external/webrtc/webrtc/call/call.cc
|
| c7a8b08a7cd8d8f37d7f5fb9930d0cdc74baba35 |
|
16-Oct-2015 |
solenberg <solenberg@webrtc.org> |
Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams.
AudioSendStream will be replacing the send side of VoiceEngine channels and associated APIs. Hence, they will be used transform recorded audio into RTP/RTCP packets that can be transmitted to another party, according to given parameters.
BUG=webrtc:4690
Review URL: https://codereview.webrtc.org/1397123003
Cr-Commit-Position: refs/heads/master@{#10307}
/external/webrtc/webrtc/call/call.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/call/call.cc
|
| a2f30deea342896ee40cc4d90567f091efbe0fc9 |
|
15-Oct-2015 |
pbos <pbos@webrtc.org> |
Log Call {audio, video} stream deletions.
BUG=
R=solenberg@webrtc.org, stefan@webrtc.org
Review URL: https://codereview.webrtc.org/1400333002
Cr-Commit-Position: refs/heads/master@{#10286}
/external/webrtc/webrtc/call/call.cc
|
| 457a61db616f17be54b32bc7d8bb781d53396f69 |
|
14-Oct-2015 |
stefan <stefan@webrtc.org> |
Pause/resume pacer from Call instead of via SendStreams.
BUG=webrtc:5073
Review URL: https://codereview.webrtc.org/1398443007
Cr-Commit-Position: refs/heads/master@{#10271}
/external/webrtc/webrtc/call/call.cc
|
| 4fbd145dcefd23169a9b1612d5ca92dace8196d6 |
|
28-Sep-2015 |
stefan <stefan@webrtc.org> |
Fix suspend below min bitrate in new API by making it possible to set min bitrate at the receive-side.
In addition to this the ramp-up tests are refactored to use a receive call instead of only a remote bitrate estimator, and to make use of BaseTest.
BUG=webrtc:4836
Review URL: https://codereview.webrtc.org/1368943002
Cr-Commit-Position: refs/heads/master@{#10087}
/external/webrtc/webrtc/call/call.cc
|
| 5c389d3e09646c0e2ed76d5ccb37a3419a09eb6a |
|
25-Sep-2015 |
Peter Boström <pbos@webrtc.org> |
Split webrtc/video into webrtc/{audio,call,video}.
Moves audio_receive_stream.{h,cc} into webrtc/audio, and common parts
into webrtc/call, splitting out audio/shared components with separate
OWNERS files.
BUG=webrtc:4690
R=solenberg@webrtc.org, tina.legrand@webrtc.org
TBR=mflodman@webrtc.org
Review URL: https://codereview.webrtc.org/1227923005 .
Cr-Commit-Position: refs/heads/master@{#10073}
/external/webrtc/webrtc/call/call.cc
|