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webrtc
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b28e522b4f5b6d467a70e7b623fe1a97e09aaa59
b28e522
WebRTCDemo: Enable making multiple calls.
by fischman@webrtc.org
· 11 years ago
7793e44
Add OWNERS file for channel_transport
by kjellander@webrtc.org
· 11 years ago
bb48e9c
Replace legacy G_CONST with const.
by pbos@webrtc.org
· 11 years ago
76076ec
Removing remaining WebRtc_Word32 not in typedefs.h
by pbos@webrtc.org
· 11 years ago
919738e
WebRTCDemo: no-op out instead of NPEing on destroyed camera.
by fischman@webrtc.org
· 11 years ago
e0e4035
WebRtc_Word32 -> int32_t in video_capture/
by pbos@webrtc.org
· 11 years ago
470cb87
WebRtc_Word32 -> int32_t in video_render/
by pbos@webrtc.org
· 11 years ago
211b771
WebRtc_Word32 -> int32_t in audio_processing/
by pbos@webrtc.org
· 11 years ago
b28b83e
Reapply the reverted r3747.
by marpan@webrtc.org
· 11 years ago
bffd956
More trace events
by hclam@chromium.org
· 11 years ago
d1c6dde
Improve how NACK lists are generated before a frame has been decoded.
by stefan@webrtc.org
· 11 years ago
4b41852
WebRtc_Word32 -> int32_t in audio_conference_mixer/
by pbos@webrtc.org
· 11 years ago
1727dc7
WebRtc_Word32 -> int32_t in common_audio/
by pbos@webrtc.org
· 11 years ago
41e3677
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."
by stefan@webrtc.org
· 11 years ago
2a5d229
WebRtc_Word32 -> int32_t in video_engine/
by pbos@webrtc.org
· 11 years ago
5691648
WebRtc_Word32 -> int32_t in video_processing/
by pbos@webrtc.org
· 11 years ago
82e0d35
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.
by stefan@webrtc.org
· 11 years ago
66a0ab3
WebRtc_Word32 -> int32_t in common_video.
by pbos@webrtc.org
· 11 years ago
896b1e1
WebRtc_Word32 -> int32_t in utility/
by pbos@webrtc.org
· 11 years ago
14353cc
WebRtc_Word32 -> int32_t in media_file/
by pbos@webrtc.org
· 11 years ago
6c604ea
Fixing the flakiness of ThreadWakesTwice.
by hta@webrtc.org
· 11 years ago
b9ada57
WebRtc_Word32 -> int32_t in test/
by pbos@webrtc.org
· 11 years ago
c404426
WebRtc_Word32 -> int32_t in audio_device/
by pbos@webrtc.org
· 11 years ago
1d46b92
WebRtc_Word32 -> int32_t in voice_engine/
by pbos@webrtc.org
· 11 years ago
acf4b69
WebRtc_Word32 -> int32_t in system_wrappers
by pbos@webrtc.org
· 11 years ago
51868ad
Always set render delay in ViEChannel::RegisterExternalDecoder.
by pbos@webrtc.org
· 11 years ago
3e3f84a
WebRtc_Word32 => int32_t etc. in audio_coding/
by pbos@webrtc.org
· 11 years ago
713488f
Remove the old unused udp_transport
by pwestin@webrtc.org
· 11 years ago
d51934d
Reduce execution time of rate control test.
by marpan@webrtc.org
· 11 years ago
ef32e92
Fixed a bug in isac-fix's entropy coding function: out of bounds acces to array.
by kma@webrtc.org
· 11 years ago
2708412
WebRtc_Word32 => int32_t in video_coding/
by pbos@webrtc.org
· 11 years ago
771774f
WebRtc_Word32 => int32_t for rtp_rtcp/
by pbos@webrtc.org
· 11 years ago
98e70d4
Clean packets on the network when closing + made loopback test actually run again.
by mflodman@webrtc.org
· 11 years ago
14d016a
WebRtc_Word32 => int32_t remote_bitrate_estimator/
by pbos@webrtc.org
· 11 years ago
dd78d46
Fix a crash issue on WinXP where LoadLibrary(TEXT("Kernel32.dll")) may fail.
by wu@webrtc.org
· 11 years ago
50fb4ed
In streaming mode it is preferable to fade to silence when sender stops sending, or long period of packet loss.
by turaj@webrtc.org
· 11 years ago
2cc0155
Resolves TSan v2 reports data races in voe_auto_test.
by henrika@webrtc.org
· 11 years ago
ad45772
Add GYP target for WebRTC Video demo for Android.
by kjellander@webrtc.org
· 11 years ago
3c48614
Permit arbitrary payload names for kVideoCodecGeneric.
by pbos@webrtc.org
· 11 years ago
47e4f00
Remove WEBRTC_*_ENGINE_NETWORK_API use
by pwestin@webrtc.org
· 11 years ago
0b8adb4
Adds event traces and counters for WebRTC receive side.
by edjee@google.com
· 11 years ago
34dac64
Fix no received audio in tests.
by pwestin@webrtc.org
· 11 years ago
fe3a907
Disabling MixingTests due to race conditions.
by henrika@webrtc.org
· 11 years ago
5bea712
Two more sleep calls converted to use SleepMs().
by hta@webrtc.org
· 11 years ago
ebc0331
TSan v2 reports data races in WebRTCAudioDeviceTest.FullDuplexAudioWithAGC
by henrika@webrtc.org
· 11 years ago
22f789f
Remove UDP transport API from VoE
by pwestin@webrtc.org
· 11 years ago
25dda04
Fixes memory leak in AudioLevel class reported by memory try bots.
by henrika@webrtc.org
· 11 years ago
63ef6e2
Fixes data race in WebRTCAudioDeviceTest.StartRecording reported by ThreadSanitizer
by henrika@webrtc.org
· 11 years ago
e561f8c
Remove UDP transport API from ViE
by pwestin@webrtc.org
· 11 years ago
45d75a4
Webrtc_Word32 => int32_t in video_coding/main/
by pbos@webrtc.org
· 11 years ago
1562c72
Revert of r3747.
by henrike@webrtc.org
· 11 years ago
d393127
Two more sleep calls converted to use SleepMs().
by hta@webrtc.org
· 11 years ago
d042a17
Fixes data race in WebRTCAudioDeviceTest.Construct reported by ThreadSanitizer
by henrika@webrtc.org
· 11 years ago
d8322b9
Fix opus bitrate truncated to 16-bit int. This prevented setting bitrates higher
by justinlin@chromium.org
· 11 years ago
435b50c
For VGA (640x360), currently 1 thread is used. This change increases it to 2 threads. For HD, 4 threads are enabled.
by fbarchard@google.com
· 11 years ago
2379013
Updated Webrtc version to 3.28
by elham@webrtc.org
· 11 years ago
bbf5086
Fix for issue: https://code.google.com/p/webrtc/issues/detail?id=1549
by marpan@webrtc.org
· 11 years ago
bcce6df
Fixes build break in previous cl (https://code.google.com/p/webrtc/source/detail?r=3739) found by Android bots.
by henrike@webrtc.org
· 11 years ago
18881d5
Removed CPU APIs from VoEHardware. Code is now only used by test applications.
by henrike@webrtc.org
· 11 years ago
1ca9d42
Removed all code enclosed in WEBRTC_SRTP #ifdefs, and the unsupported VoE SRTP APIs. Test stubs are left in place as we still have the (De)RegisterExternalEncryption() APIs, although they are currently untested.
by solenberg@webrtc.org
· 11 years ago
e148532
Revert 3736 "Removed CPU APIs from VoEHardware. Code is now only..."
by wu@webrtc.org
· 11 years ago
90edf85
Removed CPU APIs from VoEHardware. Code is now only used by test applications.
by henrike@webrtc.org
· 11 years ago
fece2f5
Fix broken audio.
by leozwang@webrtc.org
· 11 years ago
11552e9
G722-stereo has been missing when creating AudioDecoder.
by turaj@webrtc.org
· 11 years ago
3e00311
NetEq4 fails if the first packets inserted in are out-of-band DTMFs.
by turaj@webrtc.org
· 11 years ago
c3ab830
Fix flakiness in network up/down event tests when running under memcheck.
by stefan@webrtc.org
· 11 years ago
09e8463
WebRTCDemo: remove unnecessary stop & start during orientation change which isn't necessary since API v14.
by fischman@webrtc.org
· 11 years ago
e3eea1b
Add interface to signal a network down event.
by stefan@webrtc.org
· 11 years ago
fb6a7c4
Split condition_variable_win.cc into native (for Vista and newer OS versions) and generic implementation (based on events).
by henrike@webrtc.org
· 11 years ago
41419d9
Remove VoE's default call in Trace::SetLevelFilter.
by andrew@webrtc.org
· 11 years ago
eefab4e
Fix potential buffer overrun when checking if a packet is RTCP. Also makes validation slightly more robust.
by solenberg@webrtc.org
· 11 years ago
6fc92b4
Alphabetize include order in fake_voe_external_media.h.
by andrew@webrtc.org
· 11 years ago
6666b90
Restart Android capture after orientation change. Also prevent an NPE on exit.
by fischman@webrtc.org
· 11 years ago
58a5924
Add some VoE and AudioProcessing mocks.
by andrew@webrtc.org
· 11 years ago
f658278
Refactor unittest trace printouts to a separate class.
by andrew@webrtc.org
· 11 years ago
8cfba7e
Enable the below APIs for iOS.
by sjlee@webrtc.org
· 11 years ago
60c8100
Introduced pause and resume to the pacer
by pwestin@webrtc.org
· 11 years ago
e760243
Updated WebRTC version to 3.27
by elham@webrtc.org
· 11 years ago
d3eb512
Bugfix for extended RTP/RTCP test
by pwestin@webrtc.org
· 11 years ago
9c3b7bd
Move the VIE tests to use external transport instead of the built in udp transport
by pwestin@webrtc.org
· 11 years ago
680fbc5
Add trace printouts to all unit tests.
by andrew@webrtc.org
· 11 years ago
90fa4a1
Add min and target bitrate to VideoCodec.
by marpan@webrtc.org
· 11 years ago
9c0b169
Move the VoE tests to use external transport instead of the built in udp transport
by pwestin@webrtc.org
· 11 years ago
ce2d125
Creating a copy of Udp transport under webrtc/test
by pwestin@webrtc.org
· 11 years ago
ed6b4c8
Cleanup nanosleep -> SleepMs Remove some leftover stuff
by hta@webrtc.org
· 11 years ago
e3abb18
WebRtc_Word -> stdint in audio_coding/g711/
by pbos@webrtc.org
· 11 years ago
48ec040
Remove incorrect asserts.
by stefan@webrtc.org
· 11 years ago
326becd
WebRtc_Word -> stdint in audio_coding/cng/
by pbos@webrtc.org
· 11 years ago
c226567
Fix -Wstring-conversion warnings.
by wu@webrtc.org
· 11 years ago
f99f63f
Thread safety issue fix in incoming_video_stream.cc. See issue 1465.
by vikasmarwaha@webrtc.org
· 11 years ago
d579a2a
Account for header inside I420Encoder::InitEncode.
by pbos@webrtc.org
· 11 years ago
06d1e8f
Follow-up fix for r3681.
by stefan@webrtc.org
· 11 years ago
6f1f826
Fixed initialization of SPL in echo_control_mobile.
by kma@webrtc.org
· 11 years ago
aef22a7
Android: rename android_build_type gyp variable.
by wjia@webrtc.org
· 11 years ago
035c96a
Updated WebRTC version number to 3.26
by elham@webrtc.org
· 11 years ago
ebdc04d
Fix framerate sent to account for actually sent frames.
by stefan@webrtc.org
· 11 years ago
3be5a98
Change VCM interface to take target bitrate in bits per second.
by stefan@webrtc.org
· 11 years ago
a2e9124
Generic video-codec support.
by pbos@webrtc.org
· 11 years ago
857be46
Revert the deletion of test_api_nack.cc in r3674.
by stefan@webrtc.org
· 11 years ago
1f71c06
Truncated delay quality to avoid negative return values
by bjornv@webrtc.org
· 11 years ago
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