- 086055d Reland "Only include overhead if using send side bandwidth estimation." by Sebastian Jansson · 4 years, 8 months ago
- c709412 Revert "Only include overhead if using send side bandwidth estimation." by Sebastian Jansson · 4 years, 8 months ago
- 8c79c6e Only include overhead if using send side bandwidth estimation. by Sebastian Jansson · 4 years, 8 months ago
- ff0e4db Reland "Send absolute capture time through audio coding module." by Minyue Li · 4 years, 8 months ago
- 4175914 Revert "Send absolute capture time through audio coding module." by Minyue Li · 4 years, 8 months ago
- 48655cf Send absolute capture time through audio coding module. by Minyue Li · 4 years, 8 months ago
- ccbe95f Reformat GN files. by Mirko Bonadei · 4 years, 8 months ago
- 6298b56 Cleanup: Using RtpRtcp directly from AudioSendStream by Sebastian Jansson · 4 years, 8 months ago
- b2b2031 Concatenate string literals at compile time. by Jonas Olsson · 4 years, 8 months ago
- b8c775a Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api by Tim Na · 4 years, 8 months ago
- fae6400 Add saza@ and peah@ to OWNERS of some audio files by Sam Zackrisson · 4 years, 8 months ago
- 4db28b5 Cleanup: Removes redundant includes on message_queue.h by Sebastian Jansson · 4 years, 8 months ago
- 1b4e4bf Migrate several call tests from legacy RtpHeaderParser to RtpPacket parsing. by Danil Chapovalov · 4 years, 9 months ago
- f2c0818 Minor fixes to ChannelSend. by Mirko Bonadei · 4 years, 10 months ago
- 7a9a092 Delete media transport integration. by Bjorn A Mellem · 4 years, 10 months ago
- 5b82ba3 Adding VoIP specific channel adjustments by Per Åhgren · 4 years, 10 months ago
- 662678d Adds injectable trials from peerconnection down to transport controller. by Erik Språng · 4 years, 10 months ago
- 39bab5a Add missing assert.h for win no-test build by Jerome Humbert · 4 years, 11 months ago
- c3d1f9b Enable injection of a custom NetEqFactory into PeerConnectionFactory. by Ivo Creusen · 4 years, 10 months ago
- cd2a92f Removes RPLR based FEC controller. by Sebastian Jansson · 4 years, 10 months ago
- fcf79cc Add estimatedPlayoutTimestamp to RTCInboundRTPStreamStats. by Åsa Persson · 4 years, 11 months ago
- 85a1000 Use deprecated SingleThreadedTaskQueueForTesting as regular task queue by Danil Chapovalov · 4 years, 11 months ago
- 55d19e5 Add gustaf to audio/OWNERS by Gustaf Ullberg · 5 years ago
- 86d053c Use source_sets in component builds and static_library in release builds. by Mirko Bonadei · 5 years ago
- dabdde6 Avoid running NullAudioPoller without receiving streams by Gustaf Ullberg · 5 years ago
- 9429888 Delete deprecated bytes_sent/bytes_rcvd stat values by Niels Möller · 5 years ago
- f39c815 Cleanup: Replacing set extension status bool with CHECK. by Sebastian Jansson · 5 years ago
- ac0a4cb Reland "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Niels Möller · 5 years ago
- eb90e6f Merge SendTask implementation for SingleThreadedTaskQueueForTesting and TaskQueueForTest by Danil Chapovalov · 5 years ago
- ef0627f Revert "Fix GetStats bytesSent/Received, wireup headerBytesSent/Received" by Mirko Bonadei · 5 years ago
- fbde32e Fix GetStats bytesSent/Received, wireup headerBytesSent/Received by Niels Möller · 5 years ago
- 4b64411 NetEqImpl::GetDecoderFormat: Return RTP clockrate, not codec sample rate by Karl Wiberg · 5 years ago
- cd0eedb Don't allocate audio if we have no transport sequence number. by Sebastian Jansson · 5 years ago
- 0a6510d Removes rtp_transport checks in AudioSendStream by Sebastian Jansson · 5 years ago
- 35cf9e7 Replaces static modifier functions in AudioSendStream. by Sebastian Jansson · 5 years ago
- ea55b08 Adds support for passing a vector of packets to the paced sender. by Erik Språng · 5 years ago
- 0429f78 Base overhead calculation for audio priority rate on available data. by Sebastian Jansson · 5 years ago
- f23131f Removing AudioAllocationSettings moving functionality to AudioSendStream. by Sebastian Jansson · 5 years ago
- 62aee93 Adds trial to calculate audio overhead based on available data. by Sebastian Jansson · 5 years ago
- 44db436 Propagate task queue to create test::DirectTransport by TaskQueueBase interface by Danil Chapovalov · 5 years ago
- 01dd885 Moves contents of bitrate_controller to goog_cc by Sebastian Jansson · 5 years ago
- 40de3cc Propagating TargetRate struct to BitrateAllocator. by Sebastian Jansson · 5 years ago
- 93b1ea2 Using struct for bitrate allocation limits. by Sebastian Jansson · 5 years ago
- ee5ec9a Replacing local closure classes with C++14 moving capture lambdas. by Sebastian Jansson · 5 years ago
- 317a1f0 Use std::make_unique instead of absl::make_unique. by Mirko Bonadei · 5 years ago
- eaaaf41 Introduce api/crypto/BUILD.gn. by Mirko Bonadei · 5 years ago
- 65f17ca Move MediaTransportInterface out of the libjingle_peerconnection_api target by Niels Möller · 5 years ago
- 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
- cbc91ef 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 · 5 years ago
- 3472b9a Delete RTCInboundRTPStreamStats::fraction_lost by Niels Möller · 5 years ago