Log - 16cb1f61c018ddbaa5d80b600bd5f82b8a29804c - platform/external/webrtc

6516f76 Deprecate SingleThreadedTaskQueueForTesting class. by Yves Gerey · 5 years ago
a837030 Split out RtpSource from libjingle_peerconnection_api by Niels Möller · 5 years ago
65024d9 Remove clock drift metric from NetEq. by Jakob Ivarsson · 5 years ago
b6220d9 Delete unused logic for audio RtcpMode::kOff by Niels Möller · 5 years ago
f13df86 Delete audio methods SignalNetworkState by Niels Möller · 5 years ago
b4a6128 Delete unneeded dependencies on libjingle_peerconnection_api by Niels Möller · 5 years ago
6dcd4dc New target for api/rtp_parameters.h and api/media_types.h. by Niels Möller · 5 years ago
fac7e31 Removes TransportSequenceNumberAllocator by Erik Språng · 5 years ago
4208a13 Removes deprecated InsertPacket/TimeToSendPacket/TimeToSendPadding by Erik Språng · 5 years ago
d77cc24 New const method StreamStatistician::GetStats by Niels Möller · 5 years ago
224c69d Delete ext_seqnum member from VoiceSenderInfo and VoiceReceiverInfo by Niels Möller · 5 years ago
70efdde Set local ssrc at construction of Rtp module by Erik Språng · 5 years ago
54d5d2c Rename RtpRtcp::Configuration::media_send_ssrc to local_media_ssrc by Erik Språng · 5 years ago
71c6b56 Allow sending abs-send-time for audio streams. by Sebastian Jansson · 5 years ago
58b496b Let StreamStatistician::GetReceiveStreamDataCounters return counters by value by Niels Möller · 5 years ago
5b5d97c Reland of "Reporting of decoding_codec_plc events"" by Alex Narest · 5 years ago
b168678 Add RTC_ prefix to non-standard format specifier macro "PRIdNS" by Oleh Prypin · 5 years ago
83bbe91 Delete deprecated rtc_event_log header by Danil Chapovalov · 5 years ago
ed44f54 In ChannelReceive, use AcmReceiver directly, not AudioCodingModule by Niels Möller · 5 years ago
fedd625 Change 2g network pc audio test to more realistic network by Artem Titov · 5 years ago
054e3bb Reland "Replace the implementation of `GetContributingSources()` on the audio side." by Chen Xing · 5 years ago
da4f093 Reland "Only include payload in bytes sent/received." by Bjorn A Mellem · 5 years ago
bedb7a8 Revert "Reporting of decoding_codec_plc events" by Mirko Bonadei · 5 years ago
bcd068d Revert "Only include payload in bytes sent/received." by Bjorn Mellem · 5 years ago
0a88ea0 Reporting of decoding_codec_plc events by Alex Narest · 5 years ago
1704801 Prevent concurrent access to AudioSendStream's configuration. by Yves Gerey · 5 years ago
8f319a3 Reland "Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."" by Alessio Bazzica · 5 years ago
fab3460 Revert "Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker."" by Alessio Bazzica · 5 years ago
9973933 Reland "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker." by Chen Xing · 5 years ago
aa59eca Move RtpPacketSender and merge it with RtpPacketPacer. by Erik Språng · 5 years ago
74a1b4b Only include payload in bytes sent/received. by Bjorn A Mellem · 5 years ago
cbc91efa Improve low bandwidth audio test instrumentatin, fix PC test by Artem Titov · 5 years ago
2ab97f6 Migrate WebRTC test infra to ABSL_FLAG. by Mirko Bonadei · 5 years ago
0182a03 Reland "Remove the injectable bitrate allocation strategy API." by Jonas Olsson · 5 years ago
4c2c412 Set local ssrc at construction (audio) by Erik Språng · 5 years ago
24192c2 Revert "Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker." by Ivo Creusen · 5 years ago
e95b57c Revert "Remove the injectable bitrate allocation strategy API." by Mirko Bonadei · 5 years ago
52e240e Use 16000Hz audio in PC test when specified by Artem Titov · 5 years ago
b1f2d60 Reland "Fix collection of audio metrics from PC test framework for audio test" by Artem Titov · 5 years ago
80cb3f6 Remove the injectable bitrate allocation strategy API. by Jonas Olsson · 5 years ago
4876cb2 Revert "Fix collection of audio metrics from PC test framework for audio test" by Mirko Bonadei · 5 years ago
d0679bd Enables usage of ChannelMixer in WebRTC's output mixer. by henrika · 5 years ago
2d0880b Fix collection of audio metrics from PC test framework for audio test by Artem Titov · 5 years ago
4a126e4 Rename tests to prevent clashing with old audio test by Artem Titov · 5 years ago
a4d8737 Format almost everything. by Jonas Olsson · 5 years ago
c8263e0 Introduce PC level audio quality test. by Artem Titov · 5 years ago
2250b05 Adding support for channel mixing between different channel layouts. by henrika · 5 years ago
d2c336f [getStats] Implement "media-source" audio levels, fixing Chrome bug. by Henrik Boström · 5 years ago
67008df Revert "Replace the implementation of `GetContributingSources()` on the audio side." by Artem Titov · 5 years ago
8fa7151 Replace the implementation of `GetContributingSources()` on the audio side. by Chen Xing · 5 years ago
3e8ef94 Add plumbing of RtpPacketInfos to each AudioFrame as input for SourceTracker. by Chen Xing · 5 years ago
225842c Initialize signal processing function pointers statically by Karl Wiberg · 6 years ago
3472b9a Delete RTCInboundRTPStreamStats::fraction_lost by Niels Möller · 6 years ago
f48bca7 Avoid triggering a false error logging when using encryptor and sending DTX. by Minyue Li · 6 years ago
59b8654 Switch from RtpPacketSender to RtpPacketPacer interface usage. by Erik Språng · 6 years ago
08fa953 Reland "Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory" by Danil Chapovalov · 6 years ago
fd5166c Revert "Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory" by Philip Eliasson · 6 years ago
fc96135 Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory by Danil Chapovalov · 6 years ago
9ab520e Reland "Avoid encrypting empty audio packet." by Minyue Li · 6 years ago
6e436d1 [audio] Plumbing of ReportBlockData from RTCPReceiver to MediaSenderInfo by Henrik Boström · 6 years ago
87da109 Make ReceiveStatisticsImpl::SetMaxReorderingThreshold apply per ssrc by Niels Möller · 6 years ago
a352248 Add a config flag to disable the audio ALR probing request. by Christoffer Rodbro · 6 years ago
b32f2c7 Publish rtc event log api and default factory for it in api/ by Danil Chapovalov · 6 years ago
b5d9183 Add RTP timestamp to contributing sources by Johannes Kron · 6 years ago
4f08faa Introduce MediaTransportConfig by Anton Sukhanov · 6 years ago
d703cd0 Revert "Avoid encrypting empty audio packet." by Minyue Li · 6 years ago
b0ac943 Avoid encrypting empty audio packet. by Minyue Li · 6 years ago
8f119ca Enable experiments with audio bitrate priority. by Jonas Olsson · 6 years ago
9356252 Ensure that we always set values for min and max audio bitrate. by Daniel Lee · 6 years ago
8d8ffdb Expose new audio stats on the API by Ivo Creusen · 6 years ago
44125fa Reland "Piping audio interruption metrics to API layer" by Henrik Lundin · 6 years ago
fc02a79 Revert "Piping audio interruption metrics to API layer" by Henrik Andreassson · 6 years ago
413ccc4 Stop DCHECK which occurs in ANA BitrateController when overhead is zero. by Bjorn A Mellem · 6 years ago
299c4e6 Piping audio interruption metrics to API layer by Henrik Lundin · 6 years ago
c35b6e6 Deprecate RTPFragmentationHeader argument to AudioPacketizationCallback::SendData by Niels Möller · 6 years ago
30a276b Add RTP sequence number to TransportFeedbackObserver::AddPacket() by Erik Språng · 6 years ago
63658d0 Revert "Ensure that we always set values for min and max audio bitrate." by Daniel Lee · 6 years ago
e47aee3 Ensure that we always set values for min and max audio bitrate. by Daniel Lee · 6 years ago
cf96e0f Implement RTCOutboundRtpStreamStats.retransmitted[Bytes/Packets]Sent. by Henrik Boström · 6 years ago
01738c6 Wire up RTCInboundRtpStreamStats.lastPacketReceivedTimestamp. by Henrik Boström · 6 years ago
0810a7c Add base class NetworkPredictor and NetworkPredictorFactory and wire up. by Ying Wang · 6 years ago
2af5dcb Reland "Refactor FrameDecryptorInterface::Decrypt to use new API." by Benjamin Wright · 6 years ago
6a489f2 Fully qualify googletest symbols. by Mirko Bonadei · 6 years ago
7dd83e2 Revert "Refactor FrameDecryptorInterface::Decrypt to use new API." by Henrik Boström · 6 years ago
642aa81 Refactor FrameDecryptorInterface::Decrypt to use new API. by Benjamin Wright · 6 years ago
c01367d Deprecating ThreadChecker specific interface. by Sebastian Jansson · 6 years ago
31660fd Avoid using global task queue factory in audio/ unittests by Danil Chapovalov · 6 years ago
741daaf Move rtc::FunctionView to the public API by Artem Titov · 6 years ago
94b57c0 Cleanup BUILD.gn files from imports like foo:foo by Artem Titov · 6 years ago
53de725 Fix outdated android sdk path in tests. by Oleksandr Iakovenko · 6 years ago
ef1052a Reland "Move api/rtp_headers.h to its own build target." by Niels Möller · 6 years ago
2baef35 Revert "Move api/rtp_headers.h to its own build target." by Steve Anton · 6 years ago
a67050d Move api/rtp_headers.h to its own build target. by Niels Möller · 6 years ago
c936cb6 Make AudioFrameType an enum class, and move to audio_coding_module_typedefs.h by Niels Möller · 6 years ago
f0b8dee Qualify cmath functions. by Mirko Bonadei · 6 years ago
17b050f Fixes ClangTidy errors in audio/ by Benjamin Wright · 6 years ago
471783f Remove rtc::QueuedTask alias, use webrtc::QueuedTask directly by Danil Chapovalov · 6 years ago
9ffb5df Removes unused mock_bitrate_controller. by Sebastian Jansson · 6 years ago
ad89528 Reland "Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current" by Danil Chapovalov · 6 years ago
42d8c93 Revert "Delete rtc::TaskQueue::Current in favor of webrtc::TaskQueueBase::Current" by Yves Gerey · 6 years ago