Log - 1a58dd79e8f119cde5ad26343fd11fba46e30630 - fp2-dev/platform/external/chromium_org/third_party/webrtc

1a58dd7 Removing vie file related code from vie_custom_call by mikhal@webrtc.org · 11 years ago
b13f394 Fixed remaining nits from Stefan by pwestin@webrtc.org · 11 years ago
dea537b Add a push-based wrapper around SincResampler. by andrew@webrtc.org · 11 years ago
3e20f91 Add comfort noise disabling and routing mode selection to audioproc. by andrew@webrtc.org · 11 years ago
05c25a7 Removing another instance of file api by mikhal@webrtc.org · 11 years ago
3816c52 Fix the encoder pause logic. BUG=1691 by pwestin@webrtc.org · 11 years ago
40bd744 VCM: Adding API for the size(duration) of the jitter buffer. by mikhal@webrtc.org · 11 years ago
b47b7e0 VCM/JB: Using last decoded state for waiting for key by mikhal@webrtc.org · 11 years ago
1886a04 VCM/JB: FrameForDecoding->IncompleteFrameForDecoding by mikhal@webrtc.org · 11 years ago
4eac481 Disabling avi file interface by mikhal@webrtc.org · 11 years ago
e4ae7a2 Avoid adding duplicates in pacer lists. by pwestin@webrtc.org · 11 years ago
2dfffc3 Make sure timestamps are monotonically increasing. by stefan@webrtc.org · 11 years ago
8f97f02 Revert 3892 "VCM/JB: Using last decoded state for waiting for key" by andrew@webrtc.org · 11 years ago
f292306 Adding extra options to interact with external encoder/decoder. by andresp@webrtc.org · 11 years ago
a7be357 VCM/JB: Using last decoded state for waiting for key by mikhal@webrtc.org · 11 years ago
d430f32 Revert "Add a default RTT to CallStats and use different values for buffered/real-time mode." by stefan@webrtc.org · 11 years ago
292ed1d Buf fix for r3883. by turaj@webrtc.org · 11 years ago
2788107 Add a default RTT to CallStats and use different values for buffered/real-time mode. by stefan@webrtc.org · 11 years ago
9ded0b1 VP8: Avoid copying the codec struct on Reset(). by pbos@webrtc.org · 11 years ago
0dae366 BUG=1351 by mflodman@webrtc.org · 11 years ago
4123abf VCM/JB: Skip to the next complete key frame by mikhal@webrtc.org · 11 years ago
f13f1fc Updated the sync module with a slow moving filter by pwestin@webrtc.org · 11 years ago
92aa25b Improve AV-sync when initial delay is set and NetEq has long buffer. by turaj@webrtc.org · 11 years ago
6cb19e1 emove desktop_capture.gypi from modules.gyp by kjellander@webrtc.org · 11 years ago
c11933f Removed unused variable. by mflodman@webrtc.org · 11 years ago
bea854a Fixing Coverity issues. by mflodman@webrtc.org · 11 years ago
a788a4d Update iOS build script to run on bots. by kjellander@webrtc.org · 11 years ago
e07ec09 Revert 3876 by mikhal@webrtc.org · 11 years ago
c2c65ba VCM/Receiver: Only update render time when decoding by mikhal@webrtc.org · 11 years ago
b6e175d Ensure build_demo.py run subprocesses with bash shell. by kjellander@webrtc.org · 11 years ago
4fdfb61 Add the build script of the voice engine for iOS. by sjlee@webrtc.org · 11 years ago
dc6a521 revert r3871 by mikhal@webrtc.org · 11 years ago
20aee3a - Replace the BWE_MIN and BWE_MAX macros with std::min and std::max by solenberg@webrtc.org · 11 years ago
e5117e7 Apply Chromium C++ style to BitRateStats. by solenberg@webrtc.org · 11 years ago
06e8026 New ViE interface. by mflodman@webrtc.org · 11 years ago
1a67b9c Add lock to prevent possible rare race condition in Win coreAudio capture implementation. by braveyao@webrtc.org · 11 years ago
ea2ea7b Add desktop_capture directory for screen and window capturers. by sergeyu@chromium.org · 11 years ago
e7afdc7 Updating delay for first value by mikhal@webrtc.org · 11 years ago
119bc54 Remove libvpx pre-processor conditions and conditional compile of default temporal layers files. by andresp@webrtc.org · 11 years ago
0cd2401 Revert "Updating test file contents to emmastjernloef" by kjellander@webrtc.org · 11 years ago
e8bfe2a Updating test file contents to emmastjernloef by kjellander@webrtc.org · 11 years ago
28fb40d Adding Opus unit test by tina.legrand@webrtc.org · 11 years ago
570c4a5 Fix for "RTP dynamic payload type 100 is reserved" by henrika@webrtc.org · 11 years ago
6b33839 Issue 1647. Avoid unsequenced modification. by turaj@webrtc.org · 11 years ago
237fe4f Remove vim/emacs modelines from .gypi files by pbos@webrtc.org · 11 years ago
4d2a2ec Add support for multiple streams to RtpPlayer: by solenberg@webrtc.org · 11 years ago
dd32d85 Start NACKing as soon as we have the first packet of a key frame. by stefan@webrtc.org · 11 years ago
7cc4a54 Change receive statistics bitrate to be provided in bps instead of kbps. by stefan@webrtc.org · 11 years ago
c6f71c5 Make win_support_condition_variables_primitive global to aligned with |library| by wu@webrtc.org · 11 years ago
56e0484 Elevate NetEq short-term activity statistics to ACM level for logging. by turaj@webrtc.org · 11 years ago
8432603 Disable -Wunsequenced warning in audio_coding_module by kjellander@webrtc.org · 11 years ago
64d5d26 Partial revert of r3844 by mikhal@webrtc.org · 11 years ago
ab33b46 removing redundant calls to cleanframes by mikhal@webrtc.org · 11 years ago
7bc7e02 Adding a payload type for RTX. by mflodman@webrtc.org · 11 years ago
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