Log - 3bd659f1040420b7ca90d0e7df368b083148d0e5 - fp2-dev/platform/external/chromium_org/third_party/webrtc

510ee1b Remove deprecated AudioCodingModule::Destroy. by andrew@webrtc.org · 11 years ago
2529558 Enable SetInitialPlayoutDelay on Android. by dwkang@webrtc.org · 11 years ago
717267a VAD changes ported to ACM2. by turaj@webrtc.org · 11 years ago
045e45e Address Windows 64-bits warnings. by turaj@webrtc.org · 11 years ago
7ab577d Use link_settings instead of all_dependent_settings to pacify xcode gyp generator by fischman@webrtc.org · 11 years ago
6876512 Roll webrtc's chromium_revision 217707:224141 by fischman@webrtc.org · 11 years ago
f5013c0 Heap-use-after-free in WebRtcNetEQ_RecInRTPStruct by tina.legrand@webrtc.org · 11 years ago
4d08199 Compile ACM2 and ACM1. by turaj@webrtc.org · 11 years ago
ab34f11 NetEq4: Making a few more members scoped_ptrs by henrik.lundin@webrtc.org · 11 years ago
05dd6c0 Dedicated speed test for NetEq3 by henrik.lundin@webrtc.org · 11 years ago
ec09fcb Revert r4772 "Compile ACM1 and ACM2." by stefan@webrtc.org · 11 years ago
671d90b NetEq4: Make some DSP operation classes member variables by henrik.lundin@webrtc.org · 11 years ago
7556d2d Compile ACM1 and ACM2. by turaj@webrtc.org · 11 years ago
79d3355 Dedicated speed test for NetEq4 by henrik.lundin@webrtc.org · 11 years ago
ed0b4fb Prepare to compile ACM1 and ACM2. by turaj@webrtc.org · 11 years ago
3f39c00 ACM2 integration with NetEq 4. by turaj@webrtc.org · 11 years ago
4489c51 This issue is related to https://chromereviews.googleplex.com/9908014/ by minyue@webrtc.org · 11 years ago
5e3379e Make the destructor of AudioCodingModule public. by andrew@webrtc.org · 11 years ago
9fea95a Removes function that is not used anywhere but somehow still causing library load issues on Android Release build. by henrike@webrtc.org · 11 years ago
66dbbd9 Adding APIs. These APIs are not implemented yet, they are to help developement of ACM. by turaj@webrtc.org · 11 years ago
5cf83f4 Remove redundant STR_CASE_CMP macro definitions. by andrew@webrtc.org · 11 years ago
6f458ed Recognize armv7 target_arch for ios support in webrtc common.gyp by fischman@webrtc.org · 11 years ago
06eaa54 Restore severity precondition to logging.h. by andrew@webrtc.org · 11 years ago
0e2cb29 Add clockdrift to RtpGenerator by henrik.lundin@webrtc.org · 11 years ago
787364c NetEq4: Small change to reduce allocs in AudioMultiVector by henrik.lundin@webrtc.org · 11 years ago
0f62690 Revert 4671 "Enable SetInitialPlayoutDelay on Android." by mflodman@webrtc.org · 11 years ago
0fe8944 Enable SetInitialPlayoutDelay on Android. by dwkang@webrtc.org · 11 years ago
1c9d3fe Correcting two nits in InputAudioFile by henrik.lundin@webrtc.org · 11 years ago
88a2327 Disable all LS_VERBOSE logging in NetEq4 by henrik.lundin@webrtc.org · 11 years ago
797eb64 NetEq4: Make the algorithm buffer a member variable by henrik.lundin@webrtc.org · 11 years ago
1aa0938 Zero comfort noise for stereo insted of assertion. by turaj@webrtc.org · 11 years ago
b22fe00 Reorder and add critical section to the public method NetEqImpl::PacketBufferStatistics(). by turaj@webrtc.org · 11 years ago
42758b3 update neteq 4 to facilitate NACK by minyue@webrtc.org · 11 years ago
8c6633c Add isolate configuration for Android for all tests. by kjellander@webrtc.org · 11 years ago
584b688 Re-organizing ACM tests by tina.legrand@webrtc.org · 11 years ago
3540c82 Isolate GYP target and .isolate files for tests by kjellander@webrtc.org · 11 years ago
662ded4 Implementing APIs to set maximum and minimum for latency. by turaj@webrtc.org · 11 years ago
1e817c3 Roll chromium_revision 214260:217707 and gflags 45:84 by fischman@webrtc.org · 11 years ago
e155918 Revert 4547 "Isolate GYP target and .isolate files for tests" by kjellander@webrtc.org · 11 years ago
298bbdb Isolate GYP target and .isolate files for tests by kjellander@webrtc.org · 11 years ago
73acde2 To allow the propagation of under-run in NetEq. by minyue@webrtc.org · 11 years ago
bb6151d Added Opus stereo support by minyue@webrtc.org · 11 years ago
705b38d Remove include_dirs from test/test.gyp. by pbos@webrtc.org · 11 years ago
03931c6 Remove unused unreferenced code in webrtc/ by pbos@webrtc.org · 11 years ago
ea5f28b Adding call to Opus PLC by tina.legrand@webrtc.org · 11 years ago
d7b06ec Code formatting on files touched in r4447. by pbos@webrtc.org · 11 years ago
280c0b9 Added configuration of max delay to ACM and NetEq by pwestin@webrtc.org · 11 years ago
3166042 Add turaj@webrtc.org to NetEq owners. by turaj@webrtc.org · 11 years ago
71ffa0c Better error treatment in NetEqImpl::InsertPacketInternal() by minyue@webrtc.org · 11 years ago
2d3071f removed NetEq::EnableDtmf() by minyue@webrtc.org · 11 years ago
3f45c2e Switch C++-style C headers with their C equivalents. by pbos@webrtc.org · 11 years ago
4b8077b Add one API for implementing Initial delay. by turaj@webrtc.org · 11 years ago
acb00f5 Adds all unittests to android NDK-APK framework. by henrike@webrtc.org · 11 years ago
60bf21e Handel zero correlation if at the same time distortion is also zero. by turaj@webrtc.org · 11 years ago
dd1b19d Add some virtual and OVERRIDEs in webrtc/modules/audio_coding/ by pbos@webrtc.org · 11 years ago
e39e35f Correcting Turaj's email. by turaj@webrtc.org · 11 years ago
9c0f14d Cleanup WebRTC tracing by hclam@chromium.org · 11 years ago
7537dde Disables unit tests that don't work on Android for Android. by henrike@webrtc.org · 11 years ago
3b89e10 Proper spacing for end-of-namespace comments. by pbos@webrtc.org · 11 years ago
cbb3966 In call to Opus decoder: frame length too large by tina.legrand@webrtc.org · 11 years ago
1bd3140 Possible divide by 0 in ACM. by tina.legrand@webrtc.org · 11 years ago
cbb535a Error in update of read index in ACM by tina.legrand@webrtc.org · 11 years ago
8dbc8ab Merge more tests into modules_{unit,integration}tests. by kjellander@webrtc.org · 11 years ago
7b2c430 Risk of division by zero. by turaj@webrtc.org · 11 years ago
6ab2b1f G722_1/G722_1C codecs won't instantiate by tina.legrand@webrtc.org · 11 years ago
d0631e3 Nack for audio. by turaj@webrtc.org · 11 years ago
9b82368 Issue 1847, memcopy is wrong and unnecessary, it is sufficient to store the pointer before clearing the instance, and write back the pointer. by turaj@webrtc.org · 11 years ago
ae05178 resolve b9050210. Avoid pushing sync packet before any packet received. Do not turn on AV-sync if initial delay is zero. by turaj@webrtc.org · 11 years ago
1cc4ed7 Disable neteq_unittests on Win x64 in code. by kjellander@webrtc.org · 11 years ago
ec5caf3 Disable audio_decoder_unittests on Win x64 in code. by kjellander@webrtc.org · 11 years ago
f4fc8ba Disable audio_coding_unittests on Win x64 in code. by kjellander@webrtc.org · 11 years ago
92bfbbd Replace the old resampler with SincResampler in the voice engine signal path. by andrew@webrtc.org · 11 years ago
379dce7 Remove ancient and unused CNG test. by andrew@webrtc.org · 11 years ago
50a4d9f Remove #pragma once by pbos@webrtc.org · 11 years ago
d557734 API to control target delay in NetEq jitter buffer. NetEq maintains the given delay unless channel conditions require a higher delay. by turaj@webrtc.org · 11 years ago
8787048 Guarding certain operations, e.g. bandwidth estimation, RTCP statistics update etc., not to be run on sync RTPS. by turaj@webrtc.org · 11 years ago
f6e0404 Address sanitizer out of bounds read in iSAC by turaj@webrtc.org · 11 years ago
ee6f8a2 Replace ExtraCodecOptions with new Config class that supports multiple settings at once. by andresp@webrtc.org · 11 years ago
98b2011 Fix clang errors in non-GYP_DEFINES=clang=1 build by pbos@webrtc.org · 11 years ago
27f61e2 Fix off-by-one buffer overflow in WebRtcNetEQ_PacketBufferInsert(). by sergeyu@chromium.org · 11 years ago
b31332e Formatting ACM tests by tina.legrand@webrtc.org · 11 years ago
49d151e Fix when SetMinimumPlayoutDelay is configured to 0 by pwestin@webrtc.org · 11 years ago
166153e Consolidate common_audio into a single target. by andrew@webrtc.org · 11 years ago
292ed1d Buf fix for r3883. by turaj@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
28fb40d Adding Opus unit test by tina.legrand@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
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
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
74472fe More trace events by hclam@chromium.org · 11 years ago
fbda0fc WebRtc_Word32 => int32_t etc. in audio_coding/ by pbos@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
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
dded206 Adds event traces and counters for WebRTC receive side. by edjee@google.com · 11 years ago
45ce6a8 TSan v2 reports data races in WebRTCAudioDeviceTest.FullDuplexAudioWithAGC 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
a0bba27 G722-stereo has been missing when creating AudioDecoder. by turaj@webrtc.org · 11 years ago