- 2406aaf Add accounting of actual audio bit usage by Dan Tan · 7 months ago
- 33582ea Pass Environment instead of just clock to AcmReceiver at construction by Danil Chapovalov · 7 months ago
- 3732b84 Pass Clock and RtcEventLog as Environment into AudioReceiveStream by Danil Chapovalov · 8 months ago
- 943828b Pass Clock and TaskQueueFactory as Environment in voip audio channel by Danil Chapovalov · 7 months ago
- b446251 Pass receive_time through frame transformer by Lionel Koenig Gélas · 7 months ago
- 96c1b9c Add variables to lend unused audio bits to video by Dan Tan · 7 months ago
- f065ff8 Cleanup expired field trial WebRTC-VoIPChannelRemixingAdjustmentKillSwitch by Danil Chapovalov · 8 months ago
- 954e72b Update MockAudioEncoderFactory to override Create instead of MakeAudioEncoder by Danil Chapovalov · 8 months ago
- d6ef33e Remove PushResampler<T>::InitializeIfNeeded by Tommi · 8 months ago
- 32c3398 Update RemixAndResample to use audio views by Tommi · 8 months ago
- 1030eaa Provide Environment to create an audio encoder in tests by Danil Chapovalov · 9 months ago
- 578905e Provide Environment to create audio encoders in both prod code paths by Danil Chapovalov · 9 months ago
- fc6df05 Computing and propagating the audio stats totalprocessingdelay. by Jesús de Vicente Peña · 9 months ago
- 77ffbd3 Include-what-you-use api/rtc_event_log/ by Björn Terelius · 9 months ago
- 03ebfdf Create Environment for VoipCore by Danil Chapovalov · 9 months ago
- c74412b Deprecate rtc::RefCountInterface by Harald Alvestrand · 9 months ago
- 6431a64 Reland "Run IWYU on some files I intend to work on" by Harald Alvestrand · 9 months ago
- fe34363 Revert "Run IWYU on some files I intend to work on" by Mirko Bonadei · 9 months ago
- 827da15 Run IWYU on some files I intend to work on by Harald Alvestrand · 9 months ago
- 5889cf5 Propagate arrival time inside NetEq by Lionel Koenig · 10 months ago
- 61dc3ac Revert "Propagate arrival time inside NetEq" by Lionel Koenig Gélas · 9 months ago
- 0a23279 Propagate arrival time inside NetEq by Lionel Koenig · 10 months ago
- 19c51ea Use std::array<> consistently for reusable audio buffers. by Tommi · 9 months ago
- c157f29 Pass Environment into audio ChannelSend by Danil Chapovalov · 9 months ago
- 61fff58 Split out time_util to separate target ntp_time_util by Per K · 9 months ago
- 0121ff4 Revert "Propagate arrival time inside NetEq" by Manashi Sarkar · 10 months ago
- 5237cbb Propagate arrival time inside NetEq by Lionel Koenig · 10 months ago
- 5d3e680 Add audio view classes by Tommi · 10 months ago
- 99c519b Mass removal of absl_deps in all BUILD.gn files by Florent Castelli · 10 months ago
- bad99ab RTCP: implement reduced size RTCP for audio by Philipp Hancke · 10 months ago
- a45c705 Add passkey to TransformableFrameInterface to prevent external impls by Tony Herre · 10 months ago
- 8d07046 Pass the absolute capture timestamp to rtcp by Lionel Koenig · 10 months ago
- 1e5f88c Make muted param in GetAudio optional. by Jakob Ivarsson · 10 months ago
- 1a436f7 Remove AudioFrameOperations::Add, ApplyHalfGain and Scale. by Tommi · 10 months ago
- 57b09ec Update AudioFrameOperations to require ArrayView by Tommi · 10 months ago
- 1f36798 Start using ArrayView in AudioFrame, update PushResampler by Tommi · 10 months ago
- b2b6166 Make AudioFrame::channel_layout_ private and check for valid values by Tommi · 10 months ago
- 64437e8 Calculate the audio level of audio packets before encoded transforms by Tony Herre · 10 months ago
- 569849e Move call/simulated_network to test/network by Per K · 10 months ago
- 3703b35 Using Ntp times for the absolute send time. by Jesús de Vicente Peña · 11 months ago
- f4673f9 Move webrtc::AudioDeviceModule include to api/ folder by Florent Castelli · 11 months ago
- cca6cee Remove a couple of deprecated and unused AudioFrameOperations methods by Tommi · 11 months ago
- 0afde76 Move webrtc::AudioProcessing include to api/ folder by Florent Castelli · 11 months ago
- 02af840 PacketRouter directly notify RtpTransportControllerSender when sending by Per K · 12 months ago
- 5075cb4 Expose AudioLevel as an absl::optional struct in api/rtp_headers.h by Joachim Reiersen · 12 months ago
- afaae4e Remove remaining .cc files from rtc_media_base by Harald Alvestrand · 12 months ago
- 4a97488 Rename AudioLevel to AudioLevelExtension in rtp_header_extensions.h by Joachim Reiersen · 1 year, 1 month ago
- 3e613c2 Change `vector<const T>` with `const vector<T>` by Dor Hen · 1 year, 1 month ago
- 68831d2 Remove extraneous partial re-initialization of NetEq in the ChannelReceive ctor by Joachim Reiersen · 1 year, 1 month ago
- 9d9b3a3 Add option for the audio encoder to allocate a bitrate range. by Jakob Ivarsson · 1 year, 1 month ago
- 68e85b8 Adds WebRTC-Audio-PriorityBitrate for controlling audio/video rate allocation by Dan Tan · 1 year, 1 month ago
- 7aa7972 Propagate sequence number to cloned encoded audio frames by Tony Herre · 1 year, 1 month ago
- 9c687460 Consolidate encoded transform mocks into api/test/ by Tony Herre · 1 year, 1 month ago
- 340d6c0 Remove packet overhead lock and cached bitrate constraints. by Jakob Ivarsson · 1 year, 2 months ago
- 4c335b7 Record audio timestamps from iOS. by Olov Brändström · 1 year, 2 months ago
- b1799b0 Cleanup usage of the rtc::TaskQueue in audio/ by Danil Chapovalov · 1 year, 2 months ago
- 0f1b9a9 Replace rtc::TaskQueue* with TaskQueueBase* in audio channel send frame transformer by Danil Chapovalov · 1 year, 2 months ago
- ee27f38 Use Environment in RtpTransportyControllerSend by Danil Chapovalov · 1 year, 3 months ago
- 871af92 Log audio stream start/stop. by Jakob Ivarsson · 1 year, 3 months ago
- 5f3ac43 Ensure cloning and then sending audio encoded frames propagates CSRCs by Tony Herre · 1 year, 3 months ago
- f921d25 Remove DCHECK on setting audio rcvr encoded transform twice by Tony Herre · 1 year, 3 months ago
- 6e95605 Support shortcircuiting encoded transforms by Tony Herre · 1 year, 4 months ago
- d209893 Expose audio mimeType for insertable streams by Philipp Hancke · 1 year, 4 months ago
- f8feedf Make field trial string DisableRtxRateLimiter enabled by default. by Ying Wang · 1 year, 4 months ago
- c941579 Move field trial check WebRTC-DisableRtxRateLimiter by Danil Chapovalov · 1 year, 5 months ago
- 0505115 Pass the correct abs_capture_timestamp while cloning audio frame by Palak Agarwal · 1 year, 6 months ago
- c951d1b audio: fix some typos by Alfred E. Heggestad · 1 year, 6 months ago
- ad12dc5 Change ChannelReceive::GetAudioFrameWithInfo to use new Converts method by Olov Brändström · 1 year, 6 months ago
- 156facb change from unsigned to signed function (since offset can be negative) by Olov Brändström · 1 year, 6 months ago
- 4c55621 Cleanup RTPSenderAudio::SendAudio by Danil Chapovalov · 1 year, 6 months ago
- 36500ab Move RTPTimestamp offset handling out of encoded transform delegate by Tony Herre · 1 year, 6 months ago
- 14e5d4c Support sending IncomingFrames in audio by Palak Agarwal · 1 year, 7 months ago
- f263e1e Support receiving cloned encoded audio frames by Palak Agarwal · 1 year, 7 months ago
- 392e471 Remove deprecated TransformableAudioFrameInterface::getHeader() method by Tony Herre · 1 year, 7 months ago
- d43af91 Remove internal overrides using old SendRtp and SendRtcp interfaces. by Harald Alvestrand · 1 year, 7 months ago
- 48a2af3 Connected jitter_buffer_min_delay_ms to DelayManager's min_delay_ms by setting the neteq_config by anurag · 1 year, 9 months ago
- fc68f1f Stop using TransformableAudioFrameInterface::GetHeader() within webrtc by Tony Herre · 1 year, 9 months ago
- c4e0254 Fix capture_clock_offset_updater_ data race. by Jeremy Leconte · 1 year, 9 months ago
- 097a4de Make all encodedaudioframes inherit from TransformableAudioFrameI'face by Tony Herre · 1 year, 9 months ago
- 48c44e3 Ensure RtpSenderEgress run on worker queue by Per K · 1 year, 9 months ago
- fc260a18 Add method SetTimestamp in TransformableAudioFrameInterface by Palak Agarwal · 1 year, 9 months ago
- 54e95bc Propagate time of the last received packet with Timestamp type by Danil Chapovalov · 1 year, 9 months ago
- 3d6e88e Remove low_bandwidth_audio_test. by Jeremy Leconte · 1 year, 9 months ago
- 3e39254 Pass rtcp message to RtpTransportController through newer interface by Danil Chapovalov · 1 year, 10 months ago
- a2cf8ee Simplify handling rtcp messages in audio send channel by Danil Chapovalov · 1 year, 10 months ago
- 8095d02 Add RtpRtcpInterface::LastRtt function to replace RtpRtcpInterface::RTT by Danil Chapovalov · 1 year, 10 months ago
- 00ff2bb Cleanup usasge of ReportBlockData::report_block accessor in audio by Danil Chapovalov · 1 year, 10 months ago
- a9b9d4e Delete audio specific struct ReportBlock in favor of ReportBlockData by Danil Chapovalov · 1 year, 10 months ago
- 8a9f3a8 Reland "Remove dependency of video_replay on TestADM." by Artem Titov · 1 year, 11 months ago
- cde4b67 [SourceTracker] Move state to the worker thread, remove mutex. by Tommi · 1 year, 11 months ago
- f9e3bdd Revert "Remove dependency of video_replay on TestADM." by Jeremy Leconte · 1 year, 11 months ago
- 6a7bf10 Replace "rcvd" with "received" for readability by Philipp Hancke · 1 year, 11 months ago
- 0171666 Remove dependency of video_replay on TestADM. by Artem Titov · 1 year, 11 months ago
- eba7cee Extract TestADM into a separate target by Artem Titov · 1 year, 11 months ago
- fb8e3de Use AudioDeviceModule instead of TestAudioDeviceModule. by Artem Titov · 1 year, 11 months ago
- ec2670e Cleanup ReportBlockData class: use Timestamp and TimeDelta by Danil Chapovalov · 1 year, 11 months ago
- 50b0a76 Reland "[WebRTC-SendPacketsOnWorkerThread] Cleanup AudioSendStream" by Per Kjellander · 1 year, 11 months ago
- 73f048d Revert "[WebRTC-SendPacketsOnWorkerThread] Cleanup AudioSendStream" by Tomas Gunnarsson · 1 year, 11 months ago
- dd557fd [WebRTC-SendPacketsOnWorkerThread] Cleanup AudioSendStream by Per K · 1 year, 11 months ago
- 40a0e31 Remove AudioConfig::Mode. by Jeremy Leconte · 2 years ago