Log - 8fdcac3f06ae09fe17bbed02037ef136f086cc00 - platform/external/webrtc

8fdcac3 Remove clang:find_bad_constructs suppression from call:call. by Mirko Bonadei · 6 years ago
2370b08 Revert "Update packetsLost and jitter stats any time a packet is received." by Qingsi Wang · 6 years ago
4e199e9 Mark DirectTransport subclasses ctors as deprecated and switch from them by Artem Titov · 6 years ago
46c4e60 Introduce SimulatedNetworkReceiverInterface. by Artem Titov · 6 years ago
fa2b2d6 Delete use of RtpPayloadRegistry. by Niels Möller · 6 years ago
30b4839 Refactor voe::Channel to not use RtpReceiver. by Niels Möller · 6 years ago
9701e0c Makes treatment of received reports of packets lost signed. by Sebastian Jansson · 6 years ago
fa4e185 Delete class voe::RtcEventLogProxy by Niels Möller · 6 years ago
848d6d3 Change Channel::GetRtpRtcp to return only RtpRtcp, not RtpReceiver. by Niels Möller · 6 years ago
7008287 Delete struct webrtc::PacketTime. by Niels Möller · 6 years ago
264bee8 Remove memcheck. by Mirko Bonadei · 6 years ago
ab4a530 Delete telephone-event handling from RTPReceiverAudio. by Niels Möller · 6 years ago
a12c42a Delete root header file typedef.h. by Niels Möller · 6 years ago
3890262 Reland "Removing unneeded dependency." by Mirko Bonadei · 6 years ago
a61f7db Revert "Removing unneeded dependency." by Mirko Bonadei · 6 years ago
06f66c7 Removing unneeded dependency. by Mirko Bonadei · 6 years ago
bbbe4e1 Better handle target audio bitrate allocation. by Alex Narest · 6 years ago
918f50c Use absl::make_unique and absl::WrapUnique directly by Karl Wiberg · 6 years ago
64b17c2 Remove StreamStatistician::IsPacketInOrder by Danil Chapovalov · 6 years ago
bcf9180 Allows audio bitrate allocation in video calls without enabling TWCC (Transport Wide Congestion Control as defined at https://tools.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01.html) for audio stream. by Alex Narest · 6 years ago
8491693 Update packetsLost and jitter stats any time a packet is received. by Taylor Brandstetter · 6 years ago
665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
b9b146c Replace rtc::Optional with absl::optional in audio, call and video by Danil Chapovalov · 6 years ago
867e510 Enable send side audio TWCC only if WebRTC-Audio-ForceNoTWCC is not enabled. by Alex Narest · 6 years ago
f782492 Delete RtpFeedback. The ssrc for a receive stream should be known at by Niels Möller · 6 years ago
eda0087 Drop the RTT as input to IsRetransmitOfOldPacket. by Niels Möller · 6 years ago
5f83cf0 Replacing rtc::TimeDelta with webrtc::TimeDelta. by Sebastian Jansson · 6 years ago
24ad720 Uses config struct with bitrate allocator. by Sebastian Jansson · 6 years ago
104ad0b Remove stale dependencies from APM static lib target: by Fredrik Solenberg · 6 years ago
7ce3091 IWYU: Include <string.h> for memcpy(3) after bbf21a3fd. by Raphael Kubo da Costa · 6 years ago
bbf21a3 Remove dependencies on modules:module_api from AudioProcessing. by Fredrik Solenberg · 6 years ago
abbe841 This CL removes all usages of our custom ostream << overloads. by Jonas Olsson · 6 years ago
003930a Fix MID not always getting set with audio by Steve Anton · 6 years ago
ef99888 Delete OnIncomingCSRCChanged and related code. by Niels Möller · 6 years ago
bb50ce5 Wire up MID send value to the PeerConnection API by Steve Anton · 6 years ago
5f22365 Remove unnecessary proxy+lock code around RtcpRttStats pointer by Tommi · 6 years ago
9cfb18c Delete obsolete method RtpFeedback::OnInitializeDecoder. by Niels Möller · 6 years ago
77490b9 Pass a real audio codec pair ID to encoders that we create by Karl Wiberg · 6 years ago
763e947 Reporting packet feedback availability in AudioSendStream by Sebastian Jansson · 6 years ago
0812634 Pass a real audio codec pair ID to decoders that we create by Karl Wiberg · 6 years ago
fe617a3 Adding has_packet_feedback to LimitObserver callback. by Sebastian Jansson · 6 years ago
9c1ee36 Fix low_bandwidth_audio_perf_test resource dependency on Android by Oleh Prypin · 6 years ago
7b2676f Fix low_bandwidth_audio_perf_test binary dependency on Windows by Oleh Prypin · 6 years ago
8cf0a87 Reland "Split perf-test-specific resources in low_bandwidth_audio_test" by Oleh Prypin · 6 years ago
7696bef Remove the public_deps to fileutils from test_support. by Patrik Höglund · 6 years ago
650a826 Revert "Reland "Split perf-test-specific resources in low_bandwidth_audio_test"" by Oleh Prypin · 6 years ago
b3808dc Reland "Split perf-test-specific resources in low_bandwidth_audio_test" by Oleh Prypin · 6 years ago
aaa882c Revert "Split perf-test-specific resources in low_bandwidth_audio_test" by Oleh Prypin · 6 years ago
4bbc150 Split perf-test-specific resources in low_bandwidth_audio_test by Oleh Prypin · 6 years ago
9599fd4 Make num-retries default a string. by Edward Lesmes · 6 years ago
5b9c684 Add num-retries flag to Android perf tests. by Edward Lesmes · 6 years ago
3faa832 Separate test/fake_audio_device on API and implementation. Step 2. by Artem Titov · 6 years ago
d6fbf2a Tests: Pass codec ID argument to audio codecs by Karl Wiberg · 6 years ago
6fed924 Delete RTPPayloadRegistry::SetIncomingPayloadType. by Niels Möller · 6 years ago
881f168 Make SimpleStringBuilder into a non-template by Karl Wiberg · 6 years ago
8493594 Cleanup of TransportFeedbackObserver interface by Erik Språng · 6 years ago
12edf4c Separate build target for rtc_base/numerics/safe_minmax.h by Karl Wiberg · 6 years ago
98cd810 Production code: Pass codec ID argument to audio codecs by Karl Wiberg · 6 years ago
6723cdc Revert "Separate test/fake_audio_device on API and implementation." by Artem Titov · 6 years ago
8ea5f9a Separate test/fake_audio_device on API and implementation. by Artem Titov · 6 years ago
f69e768 Propagating total_bitrate_bps from BitrateAllocator to ProbeController, part 1. by philipel · 6 years ago
3c24ea8 Removed SetTransportOverhead in transport controller. by Sebastian Jansson · 6 years ago
fef0500 Adding a new string utility class: SimpleStringBuilder. by Tommi · 6 years ago
f35c666 Separate build targets for aec3 and aec3_unittests by Gustaf Ullberg · 6 years ago
ef9daee Using mock transport controller in audio unit tests. by Sebastian Jansson · 6 years ago
41f16be Silencing warnings in audio send stream unit tests. by Sebastian Jansson · 6 years ago
97f61ea Moved bitrate configuration to rtp controller by Sebastian Jansson · 6 years ago
1896cec Removed dependencies from audio send stream unit test by Sebastian Jansson · 6 years ago
2ae140a BUILD.gn file for api/audio. by Gustaf Ullberg · 6 years ago
4c1ffb8 Removing access to pacer in rtp controller. by Sebastian Jansson · 6 years ago
e4be6da Removing access to send side cc in rtp controller. by Sebastian Jansson · 6 years ago
1e06289 Delete macro RTC_ACCESS_ON, replaced by RTC_GUARDED_BY. by Niels Möller · 6 years ago
dbbb33c Stop using public_deps in common_audio. by Mirko Bonadei · 6 years ago
970b088 Reland "Break up rtc_event_log_api to solve circular dependencies." by Qingsi Wang · 6 years ago
ed7b4ff Use isolated-script-test-perf-output on low_bandwidth_audio_test. by Edward Lemur · 6 years ago
06953ba Move AudioSendStream lifetime reporting into destructor by Sam Zackrisson · 6 years ago
75df728 Revert "Break up rtc_event_log_api to solve circular dependencies." by Mirko Bonadei · 6 years ago
001546d Break up rtc_event_log_api to solve circular dependencies. by Qingsi Wang · 7 years ago
f120cba Delete AudioMonitor and related code. by Niels Möller · 6 years ago
65ce311 Removing useless dependencies on //testing/gmock. by Mirko Bonadei · 7 years ago
24ea822 Remove logging in audio/* from release builds. by Jonas Olsson · 7 years ago
a8b7c7f Move remaining traces of VoiceEngine by Fredrik Solenberg · 7 years ago
d8b041c Ignore extra arguments in low_bandwidth_audio_test. by Edward Lemur · 7 years ago
649a385 Removes usage of analog AGC. by henrika · 7 years ago
90ea504 Delete Channel::OnRecoveredPacket. by Niels Möller · 7 years ago
98d4036 Make it possible to run low_bandwidth_audio_test on Android swarming. by Edward Lemur · 7 years ago
b401771 Store JSON perf results for low_bandwidth_audio_test. by Edward Lemur · 7 years ago
8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
3b903d0 Reconfigure, not reconstruct, AudioReceiveStreams. 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
24722b3 Reland "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator." by Seth Hampson · 7 years ago
731082c Reland: Add mock_rtc_event_log.h. by Patrik Höglund · 7 years ago
5a25ab2 Revert "Add mock_rtc_event_log.h." by Edward Lemur · 7 years ago
63aea46 Add mock_rtc_event_log.h. by Patrik Höglund · 7 years ago
94dc177 Add mock_bitrate_controller.h. by Patrik Höglund · 7 years ago
6213929 Add missing files to audio_processing. by Patrik Höglund · 7 years ago
8b77aea Revert "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator." by Lu Liu · 7 years ago
d2b912a Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator. by Seth Hampson · 7 years ago