Log - 1e48e8095c8bb4daf6554233e8bc3d36563c817a - platform/external/webrtc

04dd176 Reland "Move rtp-specific config out of EncoderSettings." by Niels Möller · 6 years ago
92be1ca Revert "Move rtp-specific config out of EncoderSettings." by Niels Moller · 6 years ago
bc900cb Move rtp-specific config out of EncoderSettings. by Niels Möller · 6 years ago
e62f600 Extend WavReader and WavWriter API. by Artem Titov · 7 years ago
465a5d9 Refactor payload types constants in CallTest by Ilya Nikolaevskiy · 7 years ago
7696bef Remove the public_deps to fileutils from test_support. by Patrik Höglund · 7 years ago
7bd79a0 Split up audio_device build target by Paulina Hensman · 7 years ago
9f64b9c Reland "Remove unnecessary dependency on base." by Patrik Höglund · 7 years ago
b3bac5e Revert "Remove unnecessary dependency on base." by Patrik Höglund · 7 years ago
e0eb13c Remove unnecessary dependency on base. by Patrik Höglund · 7 years ago
0970851 Reland: Add ability to emulate degraded network in Call via field trial by Erik Språng · 7 years ago
16cba5c Revert "Add ability to emulate degraded network in Call via field trial" by Ilya Nikolaevskiy · 7 years ago
31a12c5 Add ability to emulate degraded network in Call via field trial by Erik Språng · 7 years ago
e61bf67 Separate test/fake_audio_device on API and implementation. Step 3. by Artem Titov · 7 years ago
207a75d Remove unused FrameGeneratorCapturer::Create signature by Emircan Uysaler · 7 years ago
3faa832 Separate test/fake_audio_device on API and implementation. Step 2. by Artem Titov · 7 years ago
d6fbf2a Tests: Pass codec ID argument to audio codecs by Karl Wiberg · 7 years ago
03e6ec9 Reland "Add multiplex case to webrtc_perf_tests" by Emircan Uysaler · 7 years ago
dd7e284 Reland "Enable and fix chromium clang warnings in rtp_rtcp test targets" by Danil Chapovalov · 7 years ago
01aa210 Revert "Enable and fix chromium clang warnings in rtp_rtcp test targets" by Oleh Prypin · 7 years ago
8fabab1 CNG fuzzer: avoid long fuzzer runs by limiting generator calls by Henrik Lundin · 7 years ago
9486b11 Enable and fix chromium clang warnings in rtp_rtcp test targets by Danil Chapovalov · 7 years ago
bbf1465 Delete dead code for video quality calculation. by Rasmus Brandt · 7 years ago
081136f Revert "Reland "Add multiplex case to webrtc_perf_tests"" by Taylor Brandstetter · 7 years ago
7c5bc1c Reland "Add multiplex case to webrtc_perf_tests" by Emircan Uysaler · 7 years ago
5aac372 Revert "Add multiplex case to webrtc_perf_tests" by Emircan Uysaler · 7 years ago
d90a7e8 Add multiplex case to webrtc_perf_tests by Emircan Uysaler · 7 years ago
12edf4c Separate build target for rtc_base/numerics/safe_minmax.h by Karl Wiberg · 7 years ago
0f03973 Separate test/fake_audio_device on API and implementation. Step 1. by Artem Titov · 7 years ago
2e1d784 Delete the VideoCodec::plName string. by Niels Möller · 7 years ago
6723cdc Revert "Separate test/fake_audio_device on API and implementation." by Artem Titov · 7 years ago
8ea5f9a Separate test/fake_audio_device on API and implementation. by Artem Titov · 7 years ago
3f693b9 Delete unused method SetPeriodicKeyFrames. by Niels Möller · 7 years ago
38c15d3 Template argument and corpora for Audio Processing Fuzzer. by Alex Loiko · 7 years ago
27e8a3e Revert "Adding gtest-spi.h in webrtc/test/gtest.h" by Philip Eliasson · 7 years ago
68f4904 Adding gtest-spi.h in webrtc/test/gtest.h by Alessio Bazzica · 7 years ago
0efa941 Move EchoCanceller3Factory to api/auido by Gustaf Ullberg · 7 years ago
151be2d comfort_noise_decoder_fuzzer: limit the fuzzer input size to avoid timeout by Henrik Lundin · 7 years ago
06fa153 neteq_rtp_fuzzer: limit the fuzzer input size to avoid timeout by Henrik Lundin · 7 years ago
2a6d864 neteq_signal_fuzzer: limit the fuzzer input size to avoid timeout by Henrik Lundin · 7 years ago
f35c666 Separate build targets for aec3 and aec3_unittests by Gustaf Ullberg · 7 years ago
41f16be Silencing warnings in audio send stream unit tests. by Sebastian Jansson · 7 years ago
64cf731 Roll chromium_revision 2c98648a24..37c4da4be1 (538114:538199) by Mirko Bonadei · 7 years ago
9a03dd8 Removed new calls on RtpTransportControllerSend. by Sebastian Jansson · 7 years ago
5d436ac Removed Die mock from MockAudioEncoder by Sebastian Jansson · 7 years ago
97f61ea Moved bitrate configuration to rtp controller by Sebastian Jansson · 7 years ago
a05ee82 Fixed Digital mode of AGC2 implementation finished. by Alex Loiko · 7 years ago
9d138fc Drop dependency of common_video on api:libjingle_peerconnection_api. by Niels Möller · 7 years ago
61405bc Fix infinite loop in rtp packet parsing by Danil Chapovalov · 7 years ago
0c15a09 Don't use gtest-parallel when running webrtc_perf_tests. by Edward Lemur · 7 years ago
2b304f1 Simplify CodecSettings helper function. by Rasmus Brandt · 7 years ago
1e06289 Delete macro RTC_ACCESS_ON, replaced by RTC_GUARDED_BY. by Niels Möller · 7 years ago
c2dd59c Skip oversized rtp header extension when parsing Rtp Packet. by Danil Chapovalov · 7 years ago
cc7125f Sets sending status for active RtpRtcp modules. by Seth Hampson · 7 years ago
970b088 Reland "Break up rtc_event_log_api to solve circular dependencies." by Qingsi Wang · 7 years ago
edab301 Remove webrtc::test::InitFieldTrialsFromString(const std::string&). by Bjorn Terelius · 7 years ago
75df728 Revert "Break up rtc_event_log_api to solve circular dependencies." by Mirko Bonadei · 7 years ago
001546d Break up rtc_event_log_api to solve circular dependencies. by Qingsi Wang · 7 years ago
2e5966b Store video_quality_loopback_test perf results in Chart JSON format. by Edward Lemur · 7 years ago
4f6e4f0 Increase rtp_file_reader line length to support ipv6. by Stefan Holmer · 7 years ago
d7ae3c3 Reland "Rename stereo video codec to multiplex" by Emircan Uysaler · 7 years ago
e48c61f Delete unused MediaFile module. by Niels Möller · 7 years ago
1204448 Revert "Reland "Rename stereo video codec to multiplex"" by Taylor Brandstetter · 7 years ago
4954a77 Reland "Rename stereo video codec to multiplex" by Emircan Uysaler · 7 years ago
65ce311 Removing useless dependencies on //testing/gmock. by Mirko Bonadei · 7 years ago
6bc7bb6 Revert "Rename stereo video codec to multiplex" by Ivo Creusen · 7 years ago
bbdabe5 Rename stereo video codec to multiplex by Emircan Uysaler · 7 years ago
3a5653a Use FILE* instead of const FILE* in perf_test.h by Edward Lemur · 7 years ago
c9e4522 Add an option to print perf results to a file. by Edward Lemur · 7 years ago
1723cf9 Get rid of packet loss related stuff from videoprocessor. by Sergey Silkin · 7 years ago
393e266 Use correct RTP header length in RED generation for ULPFEC packets. by Rasmus Brandt · 7 years ago
34924c2 Fix warning 4373. by Patrik Höglund · 7 years ago
46e31ba Reland "Enables/disables simulcast streams by allocating a bitrate of 0 to the spatial layer." by Seth Hampson · 7 years ago
2ffe3e8 Reland Use runtime enabled features API to enable dual stream mode by Ilya Nikolaevskiy · 7 years ago
3be2a55 Reland "Updated analysis in videoprocessor." by Sergey Silkin · 7 years ago
c1094eb Revert "Use runtime enabled features API to enable dual stream mode" by Lu Liu · 7 years ago
6f011df Use runtime enabled features API to enable dual stream mode by Ilya Nikolaevskiy · 7 years ago
a8b7c7f Move remaining traces of VoiceEngine by Fredrik Solenberg · 7 years ago
18bc3e1 Revert "Updated analysis in videoprocessor." by Sergey Silkin · 7 years ago
1880c71 Updated analysis in videoprocessor. by Sergey Silkin · 7 years ago
0f17f9c Revert "Enables/disables simulcast streams by allocating a bitrate of 0 to the spatial layer." by Lu Liu · 7 years ago
18c4261 Enables/disables simulcast streams by allocating a bitrate of 0 to the spatial layer. by Seth Hampson · 7 years ago
ab20a60 AEC-m and AEC-2 fuzzing. by Alex Loiko · 7 years ago
3ac67a7 Make aleloi@webrtc.org owner of test/fuzzers by Alex Loiko · 7 years ago
90ea504 Delete Channel::OnRecoveredPacket. by Niels Möller · 7 years ago
75baa49 Stop using public_deps in media/. by Mirko Bonadei · 7 years ago
3f6804d Optional: Use nullopt and implicit construction in /test by Oskar Sundbom · 7 years ago
8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
a7f2d84 Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" by Per Kjellander · 7 years ago
c73e1f4 Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" by Per Kjellander · 7 years ago
588c548 GN rtc_* templates: Set default visibility to webrtc_root + "/*" by Karl Wiberg · 7 years ago
62337e5 Use AudioProcessingBuilder everywhere AudioProcessing is created. by Ivo Creusen · 7 years ago
3460fa6 Use .empty() instead of '!= ""' by Edward Lemur · 7 years ago
24722b3 Reland "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator." by Seth Hampson · 7 years ago
9a0a17f Make it possible to change the amplitude of the pulses generated by PulsedNoiseCapturer. by erikvarga@webrtc.org · 7 years ago
e66572b Reland "iOS: Save perf results under Documents/perf_result.json" by Edward Lemur · 7 years ago
99175c6 Add untracked headers to video_coding. by Patrik Höglund · 7 years ago
c492bf1 Fix JSON format for reporting perf results. by Edward Lemur · 7 years ago
9e19403 Move videosourceinterface to api. by Patrik Höglund · 7 years ago
be214a2 Move videosinkinterface.h to common_video to solve a circular dep. by Patrik Höglund · 7 years ago