Log - 40220194bb03637cbc5b21ef36b6abd4e20cbd75 - fp2-dev/platform/external/chromium_org/third_party/webrtc

4022019 Update makefiles after merge of Chromium at facf66e09bf8 by Android Chromium Automerger · 10 years ago
b7e5b27 Update makefiles after merge of Chromium at 457b0a1c9412 by Android Chromium Automerger · 10 years ago
cb45b28 Update makefiles after merge of Chromium at 041843cbf814 by Android Chromium Automerger · 10 years ago
f1234f3 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 69370488385c14d73e6ae8a3d5001c42884f9275 by Android Chromium Automerger · 10 years ago
6937048 Revert 7041 " Audio codecs to include webrtc/typedefs.h" by henrike@webrtc.org · 10 years ago
fc9d779 Audio codecs to include webrtc/typedefs.h by andresp@webrtc.org · 10 years ago
5191730 Partial revert of r7014 (Android APK refactor) by kjellander@webrtc.org · 10 years ago
83d0456 Use the sample rate as a temporary solution to unpack aecdumps with wrong sizes by aluebs@webrtc.org · 10 years ago
b0aac71 Renaming SetOpusMaxBandwidth to SetOpusMaxPlaybackRate by minyue@webrtc.org · 10 years ago
6e9432f Setting marker bit on DTMF correctly by stefan@webrtc.org · 10 years ago
f28ab51 Fix issues in audioproc for float aecdumps by aluebs@webrtc.org · 10 years ago
f81734b audio_processing/nsx: Bug fix that could cause divide by zero by bjornv@webrtc.org · 10 years ago
79426b9 Disable video_engine_tests and webrtc_perf_tests on Android. by kjellander@webrtc.org · 10 years ago
fb9abb2 Divide-by-zero problem in NetEq's Normal::Process fixed by henrik.lundin@webrtc.org · 10 years ago
3de5692 Disable video_capture_tests for Android. by kjellander@webrtc.org · 10 years ago
95d2195 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at f8698ce1dacfdcf804809638483adb702760469c by Android Chromium Automerger · 10 years ago
606c1cd Remove build_with_chromium==1 conditions for Android by kjellander@webrtc.org · 10 years ago
02b30eb Unpacking aecdumps generates wav files by aluebs@webrtc.org · 10 years ago
457cef9 Fix audio_decoder_unittests.isolate by kjellander@webrtc.org · 10 years ago
19e3186 Adding more codecs to the AcmSenderBitExactness by henrik.lundin@webrtc.org · 10 years ago
b9d6b2b Android APK tests built from a normal WebRTC checkout. by kjellander@webrtc.org · 10 years ago
f346864 GN: Audio device module by kjellander@webrtc.org · 10 years ago
05f7eb6 GN: Implement voice engine, common audio, audio coding and audio processing by kjellander@webrtc.org · 10 years ago
af33b90 MIPS optimizations for AEC audio processing module by andrew@webrtc.org · 10 years ago
e2285d5 Add LTO support for Android Chromium. by andrew@webrtc.org · 10 years ago
06c375d Allow same src and dst in InputAudioFile::DuplicateInterleaved by henrik.lundin@webrtc.org · 10 years ago
54ade8b Add CHECK and friends from Chromium. by andrew@webrtc.org · 10 years ago
0de7d38 GN: Implement video_engine, video_capture and video_render. by kjellander@webrtc.org · 10 years ago
5b3e14c common_audio: Removed macro WEBRTC_SPL_DIV by bjornv@webrtc.org · 10 years ago
ec0c58c Fix the different samples per channel in aecdump by aluebs@webrtc.org · 10 years ago
21d508b Add unit tests to rtcp_receiver_test. by asapersson@webrtc.org · 10 years ago
2b1b7b7 Update makefiles after merge of Chromium at b241671f0248 by Android Chromium Automerger · 10 years ago
00c31d6 Expose setPayloadType on the rtp_sender. Thus allowing other users of this module by andresp@webrtc.org · 10 years ago
724e3f8 Refactoring common_audio/signal_processing: Remove macro WEBRTC_SPL_UDIV by bjornv@webrtc.org · 10 years ago
0e2b7ec Remove Android.mk build files. by pbos@webrtc.org · 10 years ago
7a2cfc5 Remove former team members from OWNERS and WATCHLISTS by kjellander@webrtc.org · 10 years ago
5a2b2f1 GN: Disable Chromium clang plugins for standalone build. by kjellander@webrtc.org · 10 years ago
9cfe803 Refactoring common_audio: Replace trivial multiplication macro by bjornv@webrtc.org · 10 years ago
62772e9 Re-landing r6961 by bjornv@webrtc.org · 10 years ago
62f3144 Revert 6961 "common_audio/signal_processing: Remove macro WEBRTC..." by bjornv@webrtc.org · 10 years ago
fe8c1e6 common_audio/signal_processing: Remove macro WEBRTC_SPL_MEMCPY_W8 by bjornv@webrtc.org · 10 years ago
347671c Refactoring common_audio/signal_processing: Replaces trivial macros by bjornv@webrtc.org · 10 years ago
999bcbc Fix WEBRTC_AEC_DEBUG_DUMP (broken by int16->float conversion) by kwiberg@webrtc.org · 10 years ago
53e892f Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at f854f30d7981795f687f9b4379100c037934535d by Android Chromium Automerger · 10 years ago
430518e Update makefiles after merge of Chromium at 291084 by Android Chromium Automerger · 10 years ago
e0ff458 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at 67afd1fc176021f625e064f20ae747e23d87d727 by Android Chromium Automerger · 10 years ago
b9a0168 Add send-side bit-exactness test for AudioCoding Module by henrik.lundin@webrtc.org · 10 years ago
67afd1f Use a deterministic input in NetEqBgnTest by henrik.lundin@webrtc.org · 10 years ago
2a900d1 Convert nsx_core_neon.S to unified syntax. by thakis@chromium.org · 10 years ago
d68cf32 Refactoring common_audio: Remove macro WEBRTC_SPL_MEMMOVE_W16 by bjornv@webrtc.org · 10 years ago
ddd80b5 New utility class for easy debug dumping to WAV files by kwiberg@webrtc.org · 10 years ago
39377b8 Minor bug fix and cosmetic changes in AEC MIPS optimizations. by andrew@webrtc.org · 10 years ago
11a9b6d Update makefiles after merge of Chromium at 290040 by Torne (Richard Coles) · 10 years ago
68fe1fc Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at c1696da9a74c7ed4ed793ce993352bd370cfc414 by Torne (Richard Coles) · 10 years ago
7c6e38e Remove __inline from WebRtcIsacfix_Log2Q8. by pbos@webrtc.org · 10 years ago
3eef73d Remove trailing null character from std::string by jiayl@webrtc.org · 10 years ago
d4bf540 Precompute the AEC FFT tables, rather than initializing at run-time. by andrew@webrtc.org · 10 years ago
00966cf GN: Fixes for Chromium builds. by kjellander@webrtc.org · 10 years ago
b5f0569 replace inline assembly WebRtcNsx_PrepareSpectrumNeon by intrinsics. by andrew@webrtc.org · 10 years ago
7b12812 MIPS optimizations for ISAC (patch #3) by andrew@webrtc.org · 10 years ago
0b97a1e Removing macro in acm_opus.cc by minyue@webrtc.org · 10 years ago
06d45ba Log the Android Audio API choice correctly. by braveyao@webrtc.org · 10 years ago
5526027 Suppress deprecation warnings in video_capture for iOS by kjellander@webrtc.org · 10 years ago
32b0f08 Roll chromium_revision 288251:289723 by kjellander@webrtc.org · 10 years ago
8150ff1 Set updated_rect for frames generated by WindowCapturer implementationsw by sergeyu@chromium.org · 10 years ago
ee3ec04 common_audio/signal_processing: Remove macro WEBRTC_SPL_UMUL_32_16_RSFT16 by bjornv@webrtc.org · 10 years ago
c1696da Small refactor on ViE to remove redudant conditions and long ifdefs. by andresp@webrtc.org · 10 years ago
f694796 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at c2ef523233552340785557abce1129a0f61537eb by Android Chromium Automerger · 10 years ago
d1d198b Return an aggregated report from ViERtpRtcp::GetSentRTCPStatistics(). by stefan@webrtc.org · 10 years ago
3e70717 Adding a 5% as packet loss level for Opus by minyue@webrtc.org · 10 years ago
b2150e5 Fix TimeToSendPadding return to be 0 if no padding bytes are sent. by andresp@webrtc.org · 10 years ago
73e62e7 Removing TODOs related to AcmReceiverBitExactness checksums by henrik.lundin@webrtc.org · 10 years ago
d245091 Update checksums for AcmReceiverBitExactness on android by henrik.lundin@webrtc.org · 10 years ago
8c5063e NetEq background noise generation off by default by henrik.lundin@webrtc.org · 10 years ago
0ee3e19 Fix STAP-A bug where we might overflow the packet buffer due to not accounting for the length of the length field. by stefan@webrtc.org · 10 years ago
bad8a13 Make a int64 constant use ULL suffix so it wont get truncated. by fbarchard@google.com · 10 years ago
5f19242 Update makefiles after merge of Chromium at 288938 by Android Chromium Automerger · 10 years ago
ea0a7d8 Fixing uninitialized variable in file_audio_device.cc. by phoglund@webrtc.org · 10 years ago
b5cd1aa common_audio/signal_processing: Removes macro WEBRTC_SPL_MUL_32_32_RSFT32 by bjornv@webrtc.org · 10 years ago
1bfd540 Adding SetOpusMaxBandwidth in VoE and ACM by minyue@webrtc.org · 10 years ago
22c283b Remove more dependencies on openssl, add dependency on boringssl. Continues on r6798 by henrike@webrtc.org · 10 years ago
8eef1c2 modules/audio_processing: Moves declaration of kDelayDiffOffsetSamples by bjornv@webrtc.org · 10 years ago
619d16a Merge NetEqDecodingTest.TestBitExactnesst and .TestNetworkStatistics by henrik.lundin@webrtc.org · 10 years ago
655819e Use test::Packet test::PacketSource classes in neteq_rtpplay by henrik.lundin@webrtc.org · 10 years ago
3cd40e9 Revert 6860 "SSE2 version of SubbandCoherence()" by bjornv@webrtc.org · 10 years ago
f5caff9 SSE2 version of SubbandCoherence() by bjornv@webrtc.org · 10 years ago
841ee42 Remove the old H264 code now that a new H.264 packetizer has been implemented. by stefan@webrtc.org · 10 years ago
280c829 Fix single nalu packetization bug. by stefan@webrtc.org · 10 years ago
81b53f5 Since the packet loss rate cannot be estimated accurately, there is always a mismatch between the estimated packet loss rate and the true one. Such a mismatch will make Opus FEC suboptimal. by minyue@webrtc.org · 10 years ago
d8b9cd1 Change how background noise mode in NetEq is set by henrik.lundin@webrtc.org · 10 years ago
febc613 Update makefiles after merge of Chromium at 287789 by Android Chromium Automerger · 10 years ago
ed01936 Merge third_party/webrtc from https://chromium.googlesource.com/external/webrtc/trunk/webrtc.git at a288b8cbb568cbf1735e6d5d0012524f4f8e5f74 by Android Chromium Automerger · 10 years ago
ac772a4 RTP video playback tool using Call APIs. by pbos@webrtc.org · 10 years ago
dcc85c0 Change the way we reference enumerators in H.264 packetization code to be standard C++ compliant. by stefan@webrtc.org · 10 years ago
284ac14 initialize packet len in NETEQTEST_DummyRTPpacket.cc and NETEQTEST_RTPpacket.cc to fix build error on vs2013 by fbarchard@google.com · 10 years ago
b0512f9 Fix implicite cast from signed int to unsigned int in unittest.cc by fbarchard@google.com · 10 years ago
a288b8c Fix potential crash when depacketizing VP8. by stefan@webrtc.org · 10 years ago
96c18e0 This is a setup to solve https://code.google.com/p/webrtc/issues/detail?id=1906 by minyue@webrtc.org · 10 years ago
7425710 Fix for retransmission. Base layer packets were not retransmitted. by asapersson@webrtc.org · 10 years ago
8661714 Update makefiles after merge of Chromium at 287308 by Android Chromium Automerger · 10 years ago