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webrtc
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dbb9f9bf2684ac070e7f6b8807526498bfb4a115
dbb9f9b
Move thread_ conditional back under defines.
by Peter Boström
· 9 years ago
fe2d50c
Skip setting thread priorities in NaCl.
by Peter Boström
· 9 years ago
ebb00d1
Improve documentation for ArrayView
by kwiberg
· 9 years ago
60c5d9b
Remove duplicated headers after updating downstream code.
by kjellander
· 9 years ago
4799095
Work around data race in TransmitMixer.
by solenberg
· 9 years ago
827b2d3
Remove VIDEOCODEC_* from engine_configurations.h.
by Peter Boström
· 9 years ago
6ec6cfb
Inline ConvertToSystemPriority.
by Peter Boström
· 9 years ago
d419613
Add option to capture to texture in AppRTCDemo for Android.
by Per
· 9 years ago
a5161c2
First part of the preparatory work before the actual work for solving the ducking problem starts.
by peah
· 9 years ago
3db1033
GN: Fix iOS error in audio_device and rtc_base
by kjellander
· 9 years ago
9410e01
Move ThreadWrapper to ProcessThread in base.
by pbos
· 9 years ago
3adfcea
Test case for CL 1437933002.
by guoweis
· 9 years ago
6d76de2
Add new method AcmReceiver::last_packet_sample_rate_hz()
by henrik.lundin
· 9 years ago
22dae1a
Remove the special case for std::vector in rtc::ArrayView
by kwiberg
· 9 years ago
0bd578b
NetEq: Add new method last_output_sample_rate_hz
by henrik.lundin
· 9 years ago
ee616be
Remove ThreadWrapper::GetThreadId. The method just calls rtc::CurrentThreadId(), which also has a more descriptive name.
by Tommi
· 9 years ago
fe3129a
Implement fuzzing of VP9 depacketization.
by Peter Boström
· 9 years ago
2b1da16
Add screenshare perf tests with lossy links
by sprang
· 9 years ago
0e2b794
Extract the parameters for the encoder stack from the CodecManager
by kwiberg
· 9 years ago
fc9226f
Request keyframe if too many packets are missing and NACK is disabled.
by jbauch
· 9 years ago
8674f94
Remove <iostream> include from file_audio_device.cc
by kjellander@webrtc.org
· 9 years ago
f1b5d20
RTCP Bye packet moved to own file
by danilchap
· 9 years ago
3e38ef9
Increase transport feedback frequency to 20 Hz.
by stefan
· 9 years ago
2db00ce
Require negotiation to send transport cc feedback over RTCP.
by stefan
· 9 years ago
8d85ad5
NetEq: Remove overly verbose logging
by henrik.lundin
· 9 years ago
697d03b
Revert of Adding the ability to create an RtpSender without a track. (patchset #3 id:300001 of https://codereview.webrtc.org/1413983004/ )
by deadbeef
· 9 years ago
775e132
Move some receive stream configuration into webrtc::AudioReceiveStream.
by solenberg
· 9 years ago
ecbf8f5
Reland of Adding the ability to create an RtpSender without a track. (patchset #1 id:1 of https://codereview.webrtc.org/1426443007/ )
by deadbeef
· 9 years ago
0cd2148
Allow pacer to boost bitrate in order to meet time constraints.
by sprang
· 9 years ago
732b339
Improved error handling in iOS ADM to avoid race during init
by henrika
· 9 years ago
9119471
Avoids hitting DCHECK in OpenSL ES player
by henrika
· 9 years ago
dba3e45
iOS: Set enable_protobuf=1 by default.
by kjellander@webrtc.org
· 9 years ago
568ca73
Add aecdump support to audioproc_f
by aluebs
· 9 years ago
0e34004
Re-enable mistakenly disabled PEM tests. Misc cleanup and alignment fixes.
by torbjorng
· 9 years ago
a67ca87
Remove dead code (we no longer support SILK)
by kwiberg
· 9 years ago
d0586f8
Disable PhysicalSocketTest.TestUdpReadyToSendIPv4 on linux due to flakiness on the Linux64 Debug bot.
by ivoc
· 9 years ago
dc67f78
Remove build_with_libjingle and exclude failing iOS tests from 'All' target.
by kjellander@webrtc.org
· 9 years ago
8448fcf
Fix DTLS packet boundary handling in SSLStreamAdapterTests.
by jbauch
· 9 years ago
0f714d5
Revert of Adding thread timeout for audio recorer thread in Java (patchset #2 id:20001 of https://codereview.webrtc.org/1444313002/ )
by henrika
· 9 years ago
838c3b5
Reland Convert internal representation of Srtp cryptos from string to int
by Guo-wei Shieh
· 9 years ago
0eaad21
Revert of Convert internal representation of Srtp cryptos from string to int. (patchset #10 id:180001 of https://codereview.webrtc.org/1416673006/ )
by guoweis
· 9 years ago
279937b
Convert internal representation of Srtp cryptos from string to int.
by guoweis
· 9 years ago
9592d2a
modules/audio_coding: Remove some codec include dirs
by kjellander@webrtc.org
· 9 years ago
8c330b6
modules/video_coding/utility: Remove include
by kjellander@webrtc.org
· 9 years ago
82064b0
modules/video_processing: refactor interface->include + more.
by Henrik Kjellander
· 9 years ago
1a4dbb4
WebRTC: Add compability header for video_coding refactoring.
by Henrik Kjellander
· 9 years ago
fe7633e
modules/video_coding refactorings
by Henrik Kjellander
· 9 years ago
262cad5
Remove dead code
by kwiberg
· 9 years ago
6c7e6c2
Move CNG/RED payload type extraction to Rent-A-Codec
by kwiberg
· 9 years ago
f7e7c2a
Fixed the render queue item size preallocation and verification, moved constant declarations, removed redundant queue allocation
by peah
· 9 years ago
2ba093c
rtcp::App moved into own file and got Parse function
by danilchap
· 9 years ago
e12951e
So long and thanks for all the code reviews!
by andrew
· 9 years ago
82cc96f
Set temporal up switch bit to false for flexible mode (one temporal layer is configured currently).
by asapersson
· 9 years ago
df7f9e5
Fix active tcp port to 9
by Guo-wei Shieh
· 9 years ago
12b7819
Several Tick counter improvements try #2."
by thaloun
· 9 years ago
7edbc3a
Update references to TLS1_CK_ECDHE_RSA_CHACHA20_POLY1305, etc.
by davidben
· 9 years ago
2f4c471
Re-apply change https://codereview.webrtc.org/1426673007/
by honghaiz
· 9 years ago
b1e1f5d
Add OpenSL ES enable setting to AppRTCDemo (part 2).
by henrika
· 9 years ago
87c6742
Remove ViEEncoder::ScaleInputImage.
by Peter Boström
· 9 years ago
63c862a
Unconditionally build VP9 support.
by Peter Boström
· 9 years ago
9ab9f46
Add UMA for send bwe and pacer bitrate.
by stefan
· 9 years ago
30dd8bd
Trace encoding/decoding time in a generic way.
by pbos
· 9 years ago
963832b
Deactivate the audio session after a call ends using the AVAudioSessionSetActiveOptionNotifyOthersOnDeactivation constant
by henrika
· 9 years ago
3c173a9
Adding thread timeout for audio recorer thread in Java
by henrika
· 9 years ago
80b7950
Add OpenSL ES enable setting to AppRTCDemo.
by glaznev
· 9 years ago
1b5ad57
Revert of Create rtc::AtomicInt POD struct. (patchset #12 id:220001 of https://codereview.webrtc.org/1420043008/ )
by pbos
· 9 years ago
568fbab
Preparational work before introducing the locks in order to harmonize the code:
by peah
· 9 years ago
9323716
Applied the render queueing to the agc.
by peah
· 9 years ago
2c58add
Remove packet initializer in RtpRtcpRtxNackTest.
by pbos
· 9 years ago
f8290aa
Use webrtc/base/logging.h for video coding/processing.
by pbos
· 9 years ago
f3ae889
Revert of Several Tick counter improvements. (patchset #8 id:140001 of https://codereview.webrtc.org/1415923010/ )
by thaloun
· 9 years ago
1506348
Introduced the render sample queue for the aec and aecm.
by peah
· 9 years ago
45d1ec1
Several Tick counter improvements.
by Tim Haloun
· 9 years ago
b799b43
Fix VP9 support in AppRTCDemo.
by Alex Glaznev
· 9 years ago
cd66254
common_video: rename interface -> include
by kjellander
· 9 years ago
3492dab
Create rtc::AtomicInt POD struct.
by pbos
· 9 years ago
2778875
Flesh out webrtc/.gitignore
by brucedawson
· 9 years ago
03d4810
Converted a bunch of error checking in Audio[Receive|Send]Stream to RTC_CHECKs instead. They should never fail.
by solenberg
· 9 years ago
9b5cde7
Adding stddef.h to opus_inst.h.
by minyue
· 9 years ago
ffe1ce0
Move some send stream configuration into webrtc::AudioSendStream.
by solenberg
· 9 years ago
3c69ade
Move CNG and RED management into the Rent-A-Codec
by kwiberg
· 9 years ago
74bf484
Revert of Do not delete the turn port entry right away when the respective connection is deleted. (patchset #5 id:260001 of https://codereview.webrtc.org/1426673007/ )
by tommi
· 9 years ago
273352e
Remove include dirs from modules/{media_file,pacing}
by Henrik Kjellander
· 9 years ago
62f8929
Fix a data race in the thread unit tests.
by nisse
· 9 years ago
2533f78
Add limit for minimum number of required samples before recording input and sent framerate stats.
by asapersson
· 9 years ago
3ecbb85
Do not delete the turn port entry right away when the respective connection is deleted.
by honghaiz
· 9 years ago
124daee
cleanup: get rid of basicdefs.h include
by tfarina
· 9 years ago
41f42f6
Fix flaky tests
by honghaiz
· 9 years ago
ce58e44
rtcp::Ij renamed to rtcp::ExtendedJitterReport
by danilchap
· 9 years ago
516dcbf
Remove scoped_ptrs for VCM sender_ and receiver_.
by pbos
· 9 years ago
2950018
rtcp::ReportBlock refactored to contain parsing
by danilchap
· 9 years ago
78b51e8
Remove BitrateController dependency fromVideoReceiveStream.
by mflodman
· 9 years ago
5303026
Rename screenshare test.
by philipel
· 9 years ago
1b78cc3
Move BitrateAllocator from BitrateController logic to Call.
by mflodman
· 9 years ago
aeae9d6
Do not delete a connection until it has not received anything for 30 seconds.
by Honghai Zhang
· 9 years ago
3347b88
Schedule a CreatePermissionRequest after the success of a previous request
by Honghai Zhang
· 9 years ago
a9c3720
Introduces Android API level linting, fixes all current API lint errors.
by Patrik Höglund
· 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
e22c7db
Remove webrtc/modules/video_{capture,render}/include
by Henrik Kjellander
· 9 years ago
cebb833
OpenSL ES stability improvements.
by henrika
· 9 years ago
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