- 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
- 88a7940 NetEq4 fails if the first packets inserted in are out-of-band DTMFs. by turaj@webrtc.org · 11 years ago
- 4a48fd6 WebRtc_Word -> stdint in audio_coding/g711/ by pbos@webrtc.org · 11 years ago
- e49f252 WebRtc_Word -> stdint in audio_coding/cng/ by pbos@webrtc.org · 11 years ago
- cf9ab12 Adding Opus frame length test by tina.legrand@webrtc.org · 11 years ago
- 0c46af6 Change intrinsic code in isac fix to let it pass chrome clang compiler. by kma@webrtc.org · 11 years ago
- d451969 Fixed issue 1497 in iSAC fixed point. by kma@webrtc.org · 11 years ago
- 6b12f97 Optimized EstCodeLpcCoef() for iSAC with intrinsics in Android-Neon platform. by kma@webrtc.org · 11 years ago
- e9bb4e5 Changing non-const reference arguments to pointers, ACM by tina.legrand@webrtc.org · 11 years ago
- 39ac132 Revert 3543 by tina.legrand@webrtc.org · 11 years ago
- 01ee1ba Changing non-const reference arguments to pointers, ACM by tina.legrand@webrtc.org · 11 years ago
- 00cc4a6 Fixing lint warnings from previous commit by tina.legrand@webrtc.org · 11 years ago