- c6ce9c5 New file api/video/BUILD.gn by Niels Möller · 6 years ago
- c6c4426 Moves network control interface to API. by Sebastian Jansson · 6 years ago
- 0c4f7be New api struct BitrateSettings. by Niels Möller · 6 years ago
- e625605 Moving task queue from Call to transport controller. by Sebastian Jansson · 6 years ago
- 0940811 Moving demux from FakeNetworkPipe to DirectTransport. by Sebastian Jansson · 6 years ago
- 8326780 Adds mock bitrate allocator. by Sebastian Jansson · 6 years ago
- 11b34f4 Remove chromium clang style errors affecting sdk/android/media_jni by Paulina Hensman · 7 years ago
- abbe841 This CL removes all usages of our custom ostream << overloads. by Jonas Olsson · 7 years ago
- 88614b0 Pass VideoEncoderFactory from WebrtcVideoEngine to VideoStreamEncoder. by Niels Möller · 7 years ago
- 2b85792 Move rw_lock_wrapper.h to rtc_base/synchronization/ by Karl Wiberg · 7 years ago
- b34556e Added receive time calculator under field trial. by Sebastian Jansson · 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
- 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
- 19bea51 Adding task queue congestion control experiment. by Sebastian Jansson · 7 years ago
- 3faa832 Separate test/fake_audio_device on API and implementation. Step 2. by Artem Titov · 7 years ago
- 19704ec Removing AvailableBandwidth method on transport controller. by Sebastian Jansson · 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
- a646d30 Enables configuration of transmission max bitrate multiplier and fec protection level. by Ying Wang · 7 years ago
- 45087cd Moved retransmission rate limiter to Call class. by Sebastian Jansson · 7 years ago
- c33c0fc Moved pacer and congestion thread from call. by Sebastian Jansson · 7 years ago
- 35dd6cd Added dependencies to mock transport controller send. by Sebastian Jansson · 7 years ago
- 8f83b42 Moved bitrate config interface from Call class. by Sebastian Jansson · 7 years ago
- 91bb667 Moved routes tracking to rtp transport controller. by Sebastian Jansson · 7 years ago
- 97f61ea Moved bitrate configuration to rtp controller by Sebastian Jansson · 7 years ago
- e5447fb Removed fake rtp transport controller send. by Sebastian Jansson · 7 years ago
- df023aa Extracted bitrate configuration from call class. by Sebastian Jansson · 7 years ago
- fc8d26b Reland "Moved BitrateConfig out of Call::Config." by Sebastian Jansson · 7 years ago
- e4bf600 Revert "Moved BitrateConfig out of Call::Config." by Lu Liu · 7 years ago
- 5897fe2 Moved BitrateConfig out of Call::Config. by Sebastian Jansson · 7 years ago
- 2ae140a BUILD.gn file for api/audio. by Gustaf Ullberg · 7 years ago
- 8366e17 Rename Call::Config to CallConfig, keep old name as alias. by Niels Möller · 7 years ago
- 970b088 Reland "Break up rtc_event_log_api to solve circular dependencies." by Qingsi Wang · 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
- 65ce311 Removing useless dependencies on //testing/gmock. by Mirko Bonadei · 7 years ago
- 3b790f3 Make fec controller plug-able. by Ying Wang · 7 years ago
- 2ffe3e8 Reland Use runtime enabled features API to enable dual stream mode by Ilya Nikolaevskiy · 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
- 9c68613 Update gn files to support Mozilla build by Dan Minor · 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
- e66572b Reland "iOS: Save perf results under Documents/perf_result.json" by Edward Lemur · 7 years ago
- 9e19403 Move videosourceinterface to api. by Patrik Höglund · 7 years ago
- 6213929 Add missing files to audio_processing. by Patrik Höglund · 7 years ago
- 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago
- 3e11343 Fix circular dependencies in webrtc_common. by Patrik Höglund · 7 years ago
- 712989d Revert "Reland "iOS: Save perf results under Documents/perf_result.json"" by Mirko Bonadei · 7 years ago
- a8005cf Fix circular dependencies between optional, array_view, and rtc_base. by Patrik Höglund · 7 years ago
- 8b886bb Reland "iOS: Save perf results under Documents/perf_result.json" by Edward Lemur · 7 years ago
- d37709b Revert "Fix circular dependencies between optional, array_view, and rtc_base." by Patrik Höglund · 7 years ago
- 081c651 Revert "iOS: Save perf results under Documents/perf_result.json" by Rasmus Brandt · 7 years ago
- a9e0924 Fix circular dependencies between optional, array_view, and rtc_base. by Patrik Höglund · 7 years ago
- 10a8e7a iOS: Save perf results under Documents/perf_result.json by Edward Lemur · 7 years ago
- a498ae8 Stop using public_deps in system_wrappers. by Mirko Bonadei · 7 years ago
- b5728d9 Stop using public_deps in modules/rtp_rtcp. by Mirko Bonadei · 7 years ago
- 03d6f2f Stop using public_deps in modules/audio_mixer. by Mirko Bonadei · 7 years ago
- a0e1a55 Stop using public_deps in the call module. by Mirko Bonadei · 7 years ago
- ad62792 Fixing hidden dependencies. by Mirko Bonadei · 7 years ago
- 56d4609 Use the new AudioProcessing statistics everywhere. by Ivo Creusen · 7 years ago
- c0e6804 Fix deps of audio:audio_tests. by Patrik Höglund · 7 years ago
- 61a7b14 Removing conditional visibility. by Mirko Bonadei · 7 years ago
- fd6c091 Delete deprecated constructor of SendSideCongestionController. by Niels Möller · 7 years ago
- f3850f6 Voice Engine: Require caller to supply an AudioDecoderFactory by Karl Wiberg · 7 years ago
- 245660a Fix Gn untracked headers in webrtc/call. by Mirko Bonadei · 7 years ago
- bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/call/BUILD.gn]
- 84f6a3f Move optional.h to webrtc/api/ by kwiberg · 7 years ago
- 529662a Move array_view.h to webrtc/api/ by kwiberg · 7 years ago
- 334f9e6 Tracking mock_paced_sender.h with a GN target by mbonadei · 7 years ago
- 1acbd68 Move RtpExtension to api/ directory and config.h/.cc to call/. by Stefan Holmer · 7 years ago
- 95c8f65 Now that https://codereview.webrtc.org/3003643002 is landed we can by mbonadei · 7 years ago
- 440b6d9 Move video send/receive stream headers to webrtc/call. by aleloi · 7 years ago
- f3f5c0e Change ThreadChecker to SequencedTaskChecker in internal::Call by eladalon · 7 years ago
- b332917 Rename RsidResolutionObserver to SsrcBindingObserver. by Steve Anton · 7 years ago
- db2a9fc Wire up RTP keep-alive in ortc api. by sprang · 7 years ago
- 5166e54 Tracking mock_process_thread with a GN target by mbonadei · 7 years ago
- e2173d9 Only one implementation of MockRtpPacketSink once by eladalon · 7 years ago
- f6a861a Remove remains of webrtc/base by ehmaldonado · 7 years ago
- c024740 Use relative paths in GN files. by jianjun.zhu · 7 years ago
- 370dd47 Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ ) by ehmaldonado · 7 years ago
- 9483b49 Remove remains of webrtc/base by ehmaldonado · 7 years ago
- a52722f Reland of Create RtcpDemuxer (patchset #1 id:1 of https://codereview.webrtc.org/2957763002/ ) by eladalon · 7 years ago
- 0e7e786 Revert of Create RtcpDemuxer (patchset #13 id:240001 of https://codereview.webrtc.org/2943693003/ ) by guidou · 7 years ago
- cb83bdf Create RtcpDemuxer. Capabilities: by eladalon · 7 years ago
- 0f15f92 Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface. by nisse · 7 years ago
- 38ede13 Support building WebRTC without audio and video. by zhihuang · 7 years ago
- d76b7b2 New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender. by nisse · 7 years ago
- 760a076 Create unit tests for RtpDemuxer by eladalon · 7 years ago
- c3d4b48 Store/restore RTP state for audio streams with same SSRC within a call by ossu · 7 years ago
- eed52bf New class RtxReceiveStream. by nisse · 7 years ago
- e4bcd6d New class RtpDemuxer and RtpPacketSinkInterface, use in Call. by nisse · 7 years ago
- 2d9d21f Add untracked headers in modules/rtp_rtcp by danilchap · 7 years ago
- 7cb69d5 This will allow me to test that Call invokes SendSideCongestionController::SetBweBitrates as expected (for https://codereview.chromium.org/2793913008). by zstein · 7 years ago