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webrtc
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5d9726a790a0b86a7c6ba9543924da806f51dc9d
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call
823f908
Switch voice transport to use Call and Stream instead of VoENetwork.
by mflodman
· 9 years ago
29dbd8a
Remove calls to ScopedToUnique and UniqueToScoped
by kwiberg
· 9 years ago
5fb5bd2
#include "webrtc/base/constructormagic.h" where appropriate
by kwiberg
· 9 years ago
413dd56
Fix test.gyp dependency.
by nisse
· 9 years ago
450784a
Refactored CL for moving the output to a separate thread.
by terelius
· 9 years ago
cad4296
Disable SwitchesToASTThenBackToTOFForVideo test completely.
by deadbeef
· 9 years ago
512897c
Update the call when the network route changes
by Honghai Zhang
· 9 years ago
46b2024
Disabling SwitchesToASTThenBackToTOFForVideo test for MSan bot.
by deadbeef
· 9 years ago
b6a1179
Suppress a flaky test: SwitchesToASTThenBackToTOFForVideo.
by minyuel
· 9 years ago
63b8d3f
Add macros for ability to log samples that are added to histograms (RTC_LOGGED_*).
by asapersson
· 9 years ago
a2761a5
Delete class webrtc::VideoRenderer and its header file.
by nisse
· 9 years ago
ee1e4ed
Make the audio channel communicate network state changes to the call.
by skvlad
· 9 years ago
f8e52ce
This is an initial cleanup step, aiming to delete the
by nisse
· 9 years ago
0b54e5a
Reland "Add check_deps rules in DEPS files."
by kjellander@webrtc.org
· 9 years ago
de82d23
Revert of Add check_deps rules in DEPS files. (patchset #2 id:60001 of https://codereview.webrtc.org/1796413002/ )
by kjellander
· 9 years ago
181e867
Add check_deps rules in DEPS files.
by kjellander@webrtc.org
· 9 years ago
cf2fc20
Clean away use of RtpAudioFeedback interface from RTP/RTCP sender code.
by solenberg
· 9 years ago
a637d2c
Add histogram stats for AV sync stream offset: "WebRTC.Video.AVSyncOffsetInMs"
by asapersson
· 9 years ago
4185d51
Replace scoped_ptr with unique_ptr in webrtc/call/
by kwiberg
· 9 years ago
3e089d3
Move BitrateAllocator reference from ViEEncoder to VideoSendStream.
by mflodman
· 9 years ago
06e85e9
Make ReconfigureVideoEncoder void.
by Peter Boström
· 9 years ago
9c0599f
VideoCaptureInput enforce VideoFrame::render_time to be generated by webrtc clock.
by danilchap
· 9 years ago
579eea4
Break out pacer thread from CongestionController to increase testability.
by Stefan Holmer
· 9 years ago
85150f6
Disabled flaky tests
by philipel
· 9 years ago
ead3cf2
Move congestion controller to a separate module.
by Stefan Holmer
· 9 years ago
5ba7bdf
Use CallStats for RTT in Call, rather than VideoSendStream::GetRtt()
by sprang
· 9 years ago
1262b9d
Simplify CongestionController.
by Stefan Holmer
· 9 years ago
54fc3d3
[rtp_rtcp] rtc::scoped_ptr<rtcp::RawPacket> replaced with rtc::Buffer
by danilchap
· 9 years ago
8d1de81
Update bitrate only when we have incoming packet.
by Stefan Holmer
· 9 years ago
a96ea33
removed five redundant avsync tests to make webrtc_perf_test faster
by Danil Chapovalov
· 9 years ago
25b4a62
Initialize VideoSendStream members in constructor.
by Peter Boström
· 9 years ago
a4777ed
Added A/V sync tests with drifting clocks.
by danilchap
· 9 years ago
ed50be1
Clean up of CongestionController.
by Stefan Holmer
· 9 years ago
74c29e2
Add send-side BWE to WebRtcVoiceEngine under a finch experiment.
by stefan
· 9 years ago
3dad57b
Use separate rtp module lists for send and receive in PacketRouter.
by stefan
· 9 years ago
b295e77
Switch to use new implementation in metrics.h.
by asapersson
· 9 years ago
1e5b805
Remove mutable from rtc::CriticalSection members.
by pbos
· 9 years ago
f0c9607
Fixes a bug which incorrectly logs incoming RTCP as outgoing.
by terelius
· 9 years ago
10a9538
Support REMB in combination with send-side BWE.
by stefan
· 9 years ago
e2095e4
Switch use of CriticalSectionWrapper -> rtc::CriticalSection in call/
by tommi
· 9 years ago
d146002
Remove extra_options from VideoCodec.
by Peter Boström
· 9 years ago
8d5f298
[rtp_rtcp] rtcp::SenderReport moved into own file and got Parse function
by danilchap
· 9 years ago
5bfff64
Add ramp-up tests for transport sequence number with and w/o audio.
by Stefan Holmer
· 9 years ago
cbbb3b1
Fix capture ntp time issue introduced with r11187.
by Stefan Holmer
· 9 years ago
5af14d7
Fix test bug introduced in r11101.
by Stefan Holmer
· 9 years ago
87f3db7
Wire-up BWE feedback for audio receive streams.
by Stefan Holmer
· 9 years ago
a21e4b9
Disable RampUpTest.UpDownUp* in webrtc_perf_tests on Mac
by kjellander
· 9 years ago
26f9c18
Add CreateSend/ReceiveTransport() methods to CallTest.
by stefan
· 9 years ago
8c0d4cb
Add audio streams to CallTest and a first A/V call test.
by Stefan Holmer
· 9 years ago
98836c0
Remove DISABLED_ON_ macros.
by Peter Boström
· 9 years ago
19cc283
Step 1 to prepare call_test.* for combined audio/video tests.
by stefan
· 9 years ago
4de01f8
Rename RTC_HISTOGRAM_* macros to RTC_HISTOGRAM_*_SPARSE_* to indicate that these are for infrequent updates.
by asapersson
· 9 years ago
1cddafe
Replace EventWrapper in video/, test/ and call/.
by Peter Boström
· 9 years ago
0739714
Rewrote the PRNG using an xorshift* algorithm and moved the files from test/ to base/.
by terelius
· 9 years ago
e8f0735
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
by Peter Boström
· 9 years ago
a24951b
Nuke TickTime::UseFakeClock.
by Peter Boström
· 9 years ago
32949e5
Revert of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:20001 of https://codereview.webrtc.org/1506773002/ )
by kjellander
· 9 years ago
5be013d
Merge webrtc/video_engine/ into webrtc/video/
by Peter Boström
· 9 years ago
155895c
Implement standalone event tracing in AppRTCDemo.
by Peter Boström
· 9 years ago
f95302f
Prepare the AudioSendStream to be hooked up to send-side BWE.
by Stefan Holmer
· 9 years ago
e0dd44e
Use webrtc/base/logging.h in stefan@'s ownership.
by Peter Boström
· 9 years ago
edbb7ba
Enable cpplint for webrtc/audio and webrtc/call, and fix all uncovered cpplint errors.
by Fredrik Solenberg
· 9 years ago
1834d13
Remove duplicate decoders in BitrateEstimatorTest.
by Peter Boström
· 9 years ago
9780212
Rewrote pacer and bandwidth UMA stats.
by Stefan Holmer
· 9 years ago
0c1546c
audio_coding: remove "main" directory
by kjellander
· 9 years ago
9ab9f46
Add UMA for send bwe and pacer bitrate.
by stefan
· 9 years ago
ffe1ce0
Move some send stream configuration into webrtc::AudioSendStream.
by solenberg
· 9 years ago
273352e
Remove include dirs from modules/{media_file,pacing}
by Henrik Kjellander
· 9 years ago
1b78cc3
Move BitrateAllocator from BitrateController logic to Call.
by mflodman
· 9 years ago
73807f4
Re-add a thread check in Call::Call that was removed by mistake in a rebase.
by solenberg
· 9 years ago
0e2506e
Add receive bitrate UMA stats.
by stefan
· 9 years ago
b0f22c5
Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().
by solenberg
· 9 years ago
e37a013
Removed cname and receiver_reference_time_report from proto and logging code. Changed logging of RTCP to omit messages of type SDES and APP.
by terelius
· 9 years ago
ad8bc01
Change to use local Random object instead of global rand() in the RtcEventLog unit test.
by terelius
· 9 years ago
cfdad3c
Added protobuf message for loss-based BWE events, and wired it up to the send side bandwidth estimator.
by terelius
· 9 years ago
2cded7a
Set pacer target bitrate to max of bwe and bitrate allocation.
by sprang
· 9 years ago
36a14b5
modules: more interface -> include renames
by Henrik Kjellander
· 9 years ago
4f247a6
Changed FakeVoiceEngine into a MockVoiceEngine.
by Fredrik Solenberg
· 9 years ago
602ff1f
audio_coding: rename interface -> include
by Henrik Kjellander
· 9 years ago
78f65d0
system_wrappers: rename interface -> include
by Henrik Kjellander
· 9 years ago
f5a44e8
Call OnSentPacket for all packets sent in the test framework.
by stefan
· 9 years ago
1c28f5c
Implement AudioSendStream::GetStats().
by solenberg
· 9 years ago
18a79dc
Remove network_enabled_crit_ in call.cc.
by mflodman
· 9 years ago
379f274
Set send times in send time history via OnSentPacket.
by stefan
· 9 years ago
10762d3
Re-Land: Implement AudioReceiveStream::GetStats().
by Fredrik Solenberg
· 9 years ago
b62f41a
Rename ChannelGroup to CongestionController and move to webrtc/call/.
by mflodman
· 9 years ago
d5b6d22
ChannelGroup cleanup.
by mflodman
· 9 years ago
512a784
Remove system_wrappers/interface/trace_event.h
by tommi
· 9 years ago
0e9f679
Revert of Implement AudioReceiveStream::GetStats(). (patchset #19 id:360001 of https://codereview.webrtc.org/1390753002/ )
by solenberg
· 9 years ago
bbb922f
Implement AudioReceiveStream::GetStats().
by Fredrik Solenberg
· 9 years ago
fd3f2cd
Remove the video channel id completely.
by mflodman
· 9 years ago
6db0478
Added thread checker to webrtc::Call.
by solenberg
· 9 years ago
7d39383
Move ownership of receive ViEChannel to VideoReceiveStream.
by mflodman
· 9 years ago
f707c68
Add webrtc::AudioSendStream and methods on webrtc::Call to create and delete AudioSendStreams.
by solenberg
· 9 years ago
2676348
Old config events are no longer removed from RtcEventLog.
by terelius
· 9 years ago
922e8ed
Added functions on libjingle API to start and stop the recording of an RtcEventLog.
by ivoc
· 9 years ago
15b2099
Wire up packet_id / send time callbacks to webrtc via libjingle.
by stefan
· 9 years ago
f863304
Log Call {audio, video} stream deletions.
by pbos
· 9 years ago
df65b2e
Pause/resume pacer from Call instead of via SendStreams.
by stefan
· 9 years ago
3939297
Update to the RtcEventLog protobuf to remove the DebugEvent message.
by Ivo Creusen
· 9 years ago
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