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chromium_org
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webrtc
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9b5315258a9a931a4fb8b0806a79a4758a1ddc13
9b53152
Change capture interface to use NTP capture time.
by stefan@webrtc.org
· 11 years ago
f272497
Adding playout buffer status to the voe video sync
by pwestin@webrtc.org
· 11 years ago
02f158e
VCM/JB:Removing hybrid and setting a decodable state.
by mikhal@webrtc.org
· 11 years ago
42c7409
Fix issues with incorrect wrap checks when having big buffers and high bitrate.
by stefan@webrtc.org
· 11 years ago
1e43446
Fixes an issue where the start bitrate is stored in kbps instead of bps.
by stefan@webrtc.org
· 11 years ago
e68605a
Fix -Wstring-conversion warnings.
by wu@webrtc.org
· 11 years ago
bda02e4
Re-write the build of the nacklist.
by andresp@webrtc.org
· 11 years ago
1064639
WebRTCDemo: handle stride!=width from first frame.
by fischman@webrtc.org
· 11 years ago
de1c434
Updated WebRTC version number to 3.29
by elham@webrtc.org
· 11 years ago
6e816cb
WebRTCDemo: Enable making multiple calls.
by fischman@webrtc.org
· 11 years ago
980d8ea
Add OWNERS file for channel_transport
by kjellander@webrtc.org
· 11 years ago
45a3434
Replace legacy G_CONST with const.
by pbos@webrtc.org
· 11 years ago
0486a10
Removing remaining WebRtc_Word32 not in typedefs.h
by pbos@webrtc.org
· 11 years ago
d0ee571
WebRTCDemo: no-op out instead of NPEing on destroyed camera.
by fischman@webrtc.org
· 11 years ago
f7e44d6
WebRtc_Word32 -> int32_t in video_capture/
by pbos@webrtc.org
· 11 years ago
e1ca446
WebRtc_Word32 -> int32_t in video_render/
by pbos@webrtc.org
· 11 years ago
3f6d5e0
WebRtc_Word32 -> int32_t in audio_processing/
by pbos@webrtc.org
· 11 years ago
35deccc
Reapply the reverted r3747.
by marpan@webrtc.org
· 11 years ago
74472fe
More trace events
by hclam@chromium.org
· 11 years ago
378a923
Improve how NACK lists are generated before a frame has been decoded.
by stefan@webrtc.org
· 11 years ago
74f9bbb
WebRtc_Word32 -> int32_t in audio_conference_mixer/
by pbos@webrtc.org
· 11 years ago
c49ec13
WebRtc_Word32 -> int32_t in common_audio/
by pbos@webrtc.org
· 11 years ago
73ebe67
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
67879bc
WebRtc_Word32 -> int32_t in video_engine/
by pbos@webrtc.org
· 11 years ago
c57ef84
WebRtc_Word32 -> int32_t in video_processing/
by pbos@webrtc.org
· 11 years ago
65deb26
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
a97bf1c
WebRtc_Word32 -> int32_t in common_video.
by pbos@webrtc.org
· 11 years ago
f85a509
WebRtc_Word32 -> int32_t in utility/
by pbos@webrtc.org
· 11 years ago
283c29a
WebRtc_Word32 -> int32_t in media_file/
by pbos@webrtc.org
· 11 years ago
5d9a1bc
Fixing the flakiness of ThreadWakesTwice.
by hta@webrtc.org
· 11 years ago
91cab71
WebRtc_Word32 -> int32_t in test/
by pbos@webrtc.org
· 11 years ago
64a144f
WebRtc_Word32 -> int32_t in audio_device/
by pbos@webrtc.org
· 11 years ago
54f03bc
WebRtc_Word32 -> int32_t in voice_engine/
by pbos@webrtc.org
· 11 years ago
c0231af
WebRtc_Word32 -> int32_t in system_wrappers
by pbos@webrtc.org
· 11 years ago
208a648
Always set render delay in ViEChannel::RegisterExternalDecoder.
by pbos@webrtc.org
· 11 years ago
fbda0fc
WebRtc_Word32 => int32_t etc. in audio_coding/
by pbos@webrtc.org
· 11 years ago
8ec8955
Remove the old unused udp_transport
by pwestin@webrtc.org
· 11 years ago
7deebae
Reduce execution time of rate control test.
by marpan@webrtc.org
· 11 years ago
715275c
Fixed a bug in isac-fix's entropy coding function: out of bounds acces to array.
by kma@webrtc.org
· 11 years ago
48c4b75
WebRtc_Word32 => int32_t in video_coding/
by pbos@webrtc.org
· 11 years ago
b57da65
WebRtc_Word32 => int32_t for rtp_rtcp/
by pbos@webrtc.org
· 11 years ago
a9f28d5
Clean packets on the network when closing + made loopback test actually run again.
by mflodman@webrtc.org
· 11 years ago
63a1ebd
WebRtc_Word32 => int32_t remote_bitrate_estimator/
by pbos@webrtc.org
· 11 years ago
14e22dd
Fix a crash issue on WinXP where LoadLibrary(TEXT("Kernel32.dll")) may fail.
by wu@webrtc.org
· 11 years ago
a2576cf
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
1d25eac
Resolves TSan v2 reports data races in voe_auto_test.
by henrika@webrtc.org
· 11 years ago
004f462
Add GYP target for WebRTC Video demo for Android.
by kjellander@webrtc.org
· 11 years ago
2ed1cd9
Permit arbitrary payload names for kVideoCodecGeneric.
by pbos@webrtc.org
· 11 years ago
ef91cbf
Remove WEBRTC_*_ENGINE_NETWORK_API use
by pwestin@webrtc.org
· 11 years ago
dded206
Adds event traces and counters for WebRTC receive side.
by edjee@google.com
· 11 years ago
c39749a
Fix no received audio in tests.
by pwestin@webrtc.org
· 11 years ago
84423e9
Disabling MixingTests due to race conditions.
by henrika@webrtc.org
· 11 years ago
c9f8871
Two more sleep calls converted to use SleepMs().
by hta@webrtc.org
· 11 years ago
45ce6a8
TSan v2 reports data races in WebRTCAudioDeviceTest.FullDuplexAudioWithAGC
by henrika@webrtc.org
· 11 years ago
e493218
Remove UDP transport API from VoE
by pwestin@webrtc.org
· 11 years ago
9e8a401
Fixes memory leak in AudioLevel class reported by memory try bots.
by henrika@webrtc.org
· 11 years ago
c4efe71
Fixes data race in WebRTCAudioDeviceTest.StartRecording reported by ThreadSanitizer
by henrika@webrtc.org
· 11 years ago
065b64d
Remove UDP transport API from ViE
by pwestin@webrtc.org
· 11 years ago
dba5f45
Webrtc_Word32 => int32_t in video_coding/main/
by pbos@webrtc.org
· 11 years ago
7873061
Revert of r3747.
by henrike@webrtc.org
· 11 years ago
f535877
Two more sleep calls converted to use SleepMs().
by hta@webrtc.org
· 11 years ago
f46e1fa
Fixes data race in WebRTCAudioDeviceTest.Construct reported by ThreadSanitizer
by henrika@webrtc.org
· 11 years ago
b514117
Fix opus bitrate truncated to 16-bit int. This prevented setting bitrates higher
by justinlin@chromium.org
· 11 years ago
a4b78ff
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
f3ac3ba
Updated Webrtc version to 3.28
by elham@webrtc.org
· 11 years ago
55742e5
Fix for issue: https://code.google.com/p/webrtc/issues/detail?id=1549
by marpan@webrtc.org
· 11 years ago
46144bb
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
3b6f728
Removed CPU APIs from VoEHardware. Code is now only used by test applications.
by henrike@webrtc.org
· 11 years ago
fa2dd22
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
2ffc8bf
Revert 3736 "Removed CPU APIs from VoEHardware. Code is now only..."
by wu@webrtc.org
· 11 years ago
365ca40
Removed CPU APIs from VoEHardware. Code is now only used by test applications.
by henrike@webrtc.org
· 11 years ago
0c0795e
Fix broken audio.
by leozwang@webrtc.org
· 11 years ago
a0bba27
G722-stereo has been missing when creating AudioDecoder.
by turaj@webrtc.org
· 11 years ago
88a7940
NetEq4 fails if the first packets inserted in are out-of-band DTMFs.
by turaj@webrtc.org
· 11 years ago
88f12ab
Fix flakiness in network up/down event tests when running under memcheck.
by stefan@webrtc.org
· 11 years ago
6fc5215
WebRTCDemo: remove unnecessary stop & start during orientation change which isn't necessary since API v14.
by fischman@webrtc.org
· 11 years ago
dca71b2
Add interface to signal a network down event.
by stefan@webrtc.org
· 11 years ago
6e34ceb
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
f386e2b
Remove VoE's default call in Trace::SetLevelFilter.
by andrew@webrtc.org
· 11 years ago
aa0fcd7
Fix potential buffer overrun when checking if a packet is RTCP. Also makes validation slightly more robust.
by solenberg@webrtc.org
· 11 years ago
31b4448
Alphabetize include order in fake_voe_external_media.h.
by andrew@webrtc.org
· 11 years ago
a7a643e
Restart Android capture after orientation change. Also prevent an NPE on exit.
by fischman@webrtc.org
· 11 years ago
13f66d1
Add some VoE and AudioProcessing mocks.
by andrew@webrtc.org
· 11 years ago
8826e34
Refactor unittest trace printouts to a separate class.
by andrew@webrtc.org
· 11 years ago
0c1f10b
Enable the below APIs for iOS.
by sjlee@webrtc.org
· 11 years ago
9522792
Introduced pause and resume to the pacer
by pwestin@webrtc.org
· 11 years ago
f49577f
Updated WebRTC version to 3.27
by elham@webrtc.org
· 11 years ago
7fd368f
Bugfix for extended RTP/RTCP test
by pwestin@webrtc.org
· 11 years ago
c075e25
Move the VIE tests to use external transport instead of the built in udp transport
by pwestin@webrtc.org
· 11 years ago
4c27c03
Add trace printouts to all unit tests.
by andrew@webrtc.org
· 11 years ago
e1198e6
Add min and target bitrate to VideoCodec.
by marpan@webrtc.org
· 11 years ago
aa922de
Move the VoE tests to use external transport instead of the built in udp transport
by pwestin@webrtc.org
· 11 years ago
af6aa7b
Creating a copy of Udp transport under webrtc/test
by pwestin@webrtc.org
· 11 years ago
495c563
Cleanup nanosleep -> SleepMs Remove some leftover stuff
by hta@webrtc.org
· 11 years ago
4a48fd6
WebRtc_Word -> stdint in audio_coding/g711/
by pbos@webrtc.org
· 11 years ago
a2df078
Remove incorrect asserts.
by stefan@webrtc.org
· 11 years ago
e49f252
WebRtc_Word -> stdint in audio_coding/cng/
by pbos@webrtc.org
· 11 years ago
0f2782f
Fix -Wstring-conversion warnings.
by wu@webrtc.org
· 11 years ago
5815b7c
Thread safety issue fix in incoming_video_stream.cc. See issue 1465.
by vikasmarwaha@webrtc.org
· 11 years ago
757bf0f
Account for header inside I420Encoder::InitEncode.
by pbos@webrtc.org
· 11 years ago
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