Log - 8c9d79a29d9127d4ff8aa4ae386630c72cfb1808 - platform/external/webrtc

8c9d79a Add adapter_type into Candidate object. by guoweis@webrtc.org · 10 years ago
c57310b Switch kStatsValueName* constants to be enums instead of char*. by tommi@webrtc.org · 10 years ago
3b79daf Moving encoded_bytes into EncodedInfo by henrik.lundin@webrtc.org · 10 years ago
c8bc717 Fix webrtc gn windows build. by kjellander@webrtc.org · 10 years ago
f68faa5 Removing manual test pages because they have been moved to github. by jansson@webrtc.org · 10 years ago
40b276e Cleanup little things found when refactoring. by pthatcher@webrtc.org · 10 years ago
27d106b Move the downmixing out of AudioBuffer by aluebs@webrtc.org · 10 years ago
0ca768b Adding DTX to WebRTC Opus wrapper (relanding). by minyue@webrtc.org · 10 years ago
5f162c8 Merge AEC changes. by pbos@webrtc.org · 10 years ago
2b19f06 Wire up RTT statistics to webrtc::Call. by pbos@webrtc.org · 10 years ago
1351895 Remove old_factory from WebRtcVideoEngine. by pbos@webrtc.org · 10 years ago
128faba Revert "Revert 7826 "Change Android PeerConnectionUnittest to build usin..."" by perkj@webrtc.org · 10 years ago
626c09f Move isolate path into webrtc/build/android/test_runner.py by kjellander@webrtc.org · 10 years ago
817e50d Make an AudioEncoder subclass for PCM16B by henrik.lundin@webrtc.org · 10 years ago
b3ad8cf Make an AudioEncoder subclass for iSAC by kwiberg@webrtc.org · 10 years ago
abe3f18 Checking whether ACM uses codec internal or WebRTC DTX. by minyue@webrtc.org · 10 years ago
55d42c3 DCHECK: Reference condition parameter in release builds by kwiberg@webrtc.org · 10 years ago
cd5b209 Deleting quality dashboard code. by phoglund@webrtc.org · 10 years ago
3c31e6e Add NEON intrinsics version for WebRtcSpl_MinValueW16Neon by andrew@webrtc.org · 10 years ago
f4c1948 Remove jitter_estimate_test.h by mflodman@webrtc.org · 10 years ago
c5ebbd9 Support 48kHz in Noise Suppression by aluebs@webrtc.org · 10 years ago
d8ca723 Remove CELT support from audio_coding. by pbos@webrtc.org · 10 years ago
8084f95 Change LastProcessedRtt (used in the rtp/rtcp module) to return the average RTT (instead of max RTT) to get a smooth estimate of the nack interval. by asapersson@webrtc.org · 10 years ago
85bd53e Add AbsSendTime unittests to rampup_tests.cc. by pbos@webrtc.org · 10 years ago
0df3715 Cast payload type to int in logs. by asapersson@webrtc.org · 10 years ago
a853077 (Auto)update libjingle 81702493-> 81755413 by buildbot@webrtc.org · 10 years ago
3cd26b6 Revert r7858 ("DCHECK: Reference condition parameter in release builds") by kwiberg@webrtc.org · 10 years ago
3148060 DCHECK: Reference condition parameter in release builds by kwiberg@webrtc.org · 10 years ago
ff1a3e3 Make an AudioEncoder subclass for comfort noise by henrik.lundin@webrtc.org · 10 years ago
6fd52f3 Add NEON intrinsics version for WebRtcSpl_DownsampleFastNeon. by andrew@webrtc.org · 10 years ago
ae20d3b Add NEON intrinsics version for WebRtcSpl_CrossCorrelationNeon. by andrew@webrtc.org · 10 years ago
aa2c342 Add back a constructor to fix FYI build. by tommi@webrtc.org · 10 years ago
5c32a84 Attempt to fix FYI bots. by tommi@webrtc.org · 10 years ago
87776a8 iAppRTCDemo: WebSocket based signaling. by tkchin@webrtc.org · 10 years ago
0babb4a Fix a comment. by pthatcher@webrtc.org · 10 years ago
c9d155f Move implementation of types in statstypes. to its cc file. by tommi@webrtc.org · 10 years ago
a954c07 AppRTCDemo (Android): built-in AEC should be enabled if device supports it and in combination with Java-based audio layer by henrika@webrtc.org · 10 years ago
19dd129 Revert 7846 "Adding DTX to WebRTC Opus wrapper" by minyue@webrtc.org · 10 years ago
f244760 Add histograms for receive statistics: by asapersson@webrtc.org · 10 years ago
4321f17 Adding DTX to WebRTC Opus wrapper by minyue@webrtc.org · 10 years ago
5c3ee4b Add empty implementation file that will hold statstypes.h implementation. by tommi@webrtc.org · 10 years ago
1784d7c Adding an codec interal CNG test in NetEq. by minyue@webrtc.org · 10 years ago
9115cde Merge VP8 changes. by pbos@webrtc.org · 10 years ago
e04a93b Move the AudioDecoder interface out of NetEq by kwiberg@webrtc.org · 10 years ago
97d0489 Add video send bitrates to histogram stats: by asapersson@webrtc.org · 10 years ago
7ba9f27 Set CHECKOUT_SOURCE_ROOT environment variable for Android test wrapper. by kjellander@webrtc.org · 10 years ago
eef8538 Fix AppRTCDemo closing error for KK and JB Android devices. by glaznev@webrtc.org · 10 years ago
86b6d65 Remove no longer used video codec test framework. by stefan@webrtc.org · 10 years ago
8911bc5 Add AudioEncoder::Max10MsFramesInAPacket by henrik.lundin@webrtc.org · 10 years ago
130fef8 Bugfix in AudioDecoderTest by henrik.lundin@webrtc.org · 10 years ago
edeea91 Change all system clock types to int64_t in bitrate_controller. by stefan@webrtc.org · 10 years ago
fcbe36a Add const qualifier to WebRtcPcm16b_Encode by henrik.lundin@webrtc.org · 10 years ago
a1ef7bf ATTRIBUTE_UNUSED expanded to empty on MSVS, so be sure to use the variable. by kwiberg@webrtc.org · 10 years ago
3b3c406 Revert 7826 "Change Android PeerConnectionUnittest to build usin..." by andrew@webrtc.org · 10 years ago
cb858ba Make an AudioEncoder subclass for iLBC by kwiberg@webrtc.org · 10 years ago
ee43263 Cleaned up real_fft APIs due to non-existing NEON code by bjornv@webrtc.org · 10 years ago
ed7824b Change Android PeerConnectionUnittest to build using Chrome macros. by perkj@webrtc.org · 10 years ago
ba8138b Change type of nack_last_time_sent_full_ from uint32_t to int64_t. by asapersson@webrtc.org · 10 years ago
aefe61a PRESUBMIT: Add check for checkdeps. by kjellander@webrtc.org · 10 years ago
7db359b Roll chromium_revision 24b4c73..8e72e1d by kjellander@webrtc.org · 10 years ago
d91d359 PRESUBMIT: Add iOS ARM64 trybots to default set. by kjellander@webrtc.org · 10 years ago
fb01376 Adjust some parameters for VP9 tests. by marpan@webrtc.org · 10 years ago
e2a9261 Improve AppRTCDemo connection speed by sending all by glaznev@webrtc.org · 10 years ago
bd8cc0b Add codereview.settings to the /talk subdirectory by kjellander@webrtc.org · 10 years ago
5af8cd7 Add codereview.settings to the /webrtc subdirectory by kjellander@webrtc.org · 10 years ago
599e299 cricket::VideoFrame int64 to int64_t. by kjellander@webrtc.org · 10 years ago
9b5467e Fix assertion failure when closing data channel, and add a unit test. by bemasc@webrtc.org · 10 years ago
4b407aa Update AppRTCDemo README with information on 3-dot-apprtc server by glaznev@webrtc.org · 10 years ago
7169afd With IPv6 enabled, it's important to know whether IPv6 is really used or not. BestConnection is tracked for this purpose. Also added a test case to verify the end to end behavior. by guoweis@webrtc.org · 10 years ago
369746b Support new WebSocket signaling format. by glaznev@webrtc.org · 10 years ago
0b38478 Add support for parsing header only RTP dumps with bwe_rtp_play. by stefan@webrtc.org · 10 years ago
9f79fe6 Merge remote bitrate estimator changes. by pbos@webrtc.org · 10 years ago
33ccdfa Relanding r7807. by minyue@webrtc.org · 10 years ago
52bc4f4 Revert 7807 "Removing unused opus wrapper APIs." by minyue@webrtc.org · 10 years ago
c0991fe Roll chromium_revision 24b4c73..f27c369 by kjellander@webrtc.org · 10 years ago
e54a634 Removing unused opus wrapper APIs. by minyue@webrtc.org · 10 years ago
8c9ff20 Redo the change of https://webrtc-codereview.appspot.com/30949004/ by guoweis@webrtc.org · 10 years ago
fd84229 Revert "Implement GetState() for channel's connectivity check state." by guoweis@webrtc.org · 10 years ago
ff72f9e Implement GetState() for channel's connectivity check state. by guoweis@webrtc.org · 10 years ago
fd4acf6 Adding WebRtcSpl_MaxAbsValueW16 intrinsics version by andrew@webrtc.org · 10 years ago
3a52458 add WebRtcIsacfix_AutocorrNeon's intrinsics version by andrew@webrtc.org · 10 years ago
8dc21dc Rename internal AudioEncoder::Encode method to EncodeInternal by henrik.lundin@webrtc.org · 10 years ago
d1fac61 Remove need for assembly offset generation in aecm and ns module. by andrew@webrtc.org · 10 years ago
3800e13 Revert r7798 ("Move the AudioDecoder interface out of NetEq") by kwiberg@webrtc.org · 10 years ago
00ba1a7 Move the AudioDecoder interface out of NetEq by kwiberg@webrtc.org · 10 years ago
0fb6ad2 Check if cpu_monitor_ exists before Stop(). by pbos@webrtc.org · 10 years ago
fa914e2 Adding a duration printout to neteq_rtpplay by henrik.lundin@webrtc.org · 10 years ago
d8aed6b Verify that cpu_monitor exists before calling Stop(). by asapersson@webrtc.org · 10 years ago
c3e097c Add Android test runner script for WebRTC. by kjellander@webrtc.org · 10 years ago
8e5c814 Convert DEPS to only reference Git repos by kjellander@webrtc.org · 10 years ago
511f8a8 TurnPort should ignore STUN binding reponses when using shared socket. by jiayl@webrtc.org · 10 years ago
001f3b9 Adjust parameter in videoprocessor_integration_test for vp9. by marpan@webrtc.org · 10 years ago
a7384a1 Simplify audio_buffer APIs by aluebs@webrtc.org · 10 years ago
ceca014 Re-enable test: VideoProcessorIntegrationTest.ProcessNoLossChangeBitRateVP9. by marpan@webrtc.org · 10 years ago
eb09542 Don't reset sequence number for a stream on deactivate/reactivate. by pthatcher@webrtc.org · 10 years ago
d019551 Change minimum video encoder initialization resolution to by glaznev@webrtc.org · 10 years ago
1751ee7 Remove -flax-vector-conversions flag for ARM NEON building. by andrew@webrtc.org · 10 years ago
ac68ef9 Clear 2 unused functions in audio processing aecm module. by andrew@webrtc.org · 10 years ago
beee9ce Change back so that Android ApprtcDemo only use one MediaStream containing both audio and video. by perkj@webrtc.org · 10 years ago
7f1dfa5 Adding a payload type to AudioEncoder objects by henrik.lundin@webrtc.org · 10 years ago