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gerrit-public.fairphone.software / platform / external / webrtc / c7ea852189c81f20ff6a34880651a334e57f3d7a / audio
  1. 2370b08 Revert "Update packetsLost and jitter stats any time a packet is received." by Qingsi Wang · 6 years ago
  2. 4e199e9 Mark DirectTransport subclasses ctors as deprecated and switch from them by Artem Titov · 6 years ago
  3. 46c4e60 Introduce SimulatedNetworkReceiverInterface. by Artem Titov · 6 years ago
  4. fa2b2d6 Delete use of RtpPayloadRegistry. by Niels Möller · 6 years ago
  5. 30b4839 Refactor voe::Channel to not use RtpReceiver. by Niels Möller · 6 years ago
  6. 9701e0c Makes treatment of received reports of packets lost signed. by Sebastian Jansson · 6 years ago
  7. fa4e185 Delete class voe::RtcEventLogProxy by Niels Möller · 6 years ago
  8. 848d6d3 Change Channel::GetRtpRtcp to return only RtpRtcp, not RtpReceiver. by Niels Möller · 6 years ago
  9. 7008287 Delete struct webrtc::PacketTime. by Niels Möller · 6 years ago
  10. 264bee8 Remove memcheck. by Mirko Bonadei · 6 years ago
  11. ab4a530 Delete telephone-event handling from RTPReceiverAudio. by Niels Möller · 6 years ago
  12. a12c42a Delete root header file typedef.h. by Niels Möller · 6 years ago
  13. 3890262 Reland "Removing unneeded dependency." by Mirko Bonadei · 6 years ago
  14. a61f7db Revert "Removing unneeded dependency." by Mirko Bonadei · 6 years ago
  15. 06f66c7 Removing unneeded dependency. by Mirko Bonadei · 6 years ago
  16. bbbe4e1 Better handle target audio bitrate allocation. by Alex Narest · 6 years ago
  17. 918f50c Use absl::make_unique and absl::WrapUnique directly by Karl Wiberg · 6 years ago
  18. 64b17c2 Remove StreamStatistician::IsPacketInOrder by Danil Chapovalov · 6 years ago
  19. 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
  20. 8491693 Update packetsLost and jitter stats any time a packet is received. by Taylor Brandstetter · 6 years ago
  21. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  22. b9b146c Replace rtc::Optional with absl::optional in audio, call and video by Danil Chapovalov · 6 years ago
  23. 867e510 Enable send side audio TWCC only if WebRTC-Audio-ForceNoTWCC is not enabled. by Alex Narest · 6 years ago
  24. f782492 Delete RtpFeedback. The ssrc for a receive stream should be known at by Niels Möller · 6 years ago
  25. eda0087 Drop the RTT as input to IsRetransmitOfOldPacket. by Niels Möller · 6 years ago
  26. 5f83cf0 Replacing rtc::TimeDelta with webrtc::TimeDelta. by Sebastian Jansson · 6 years ago
  27. 24ad720 Uses config struct with bitrate allocator. by Sebastian Jansson · 7 years ago
  28. 104ad0b Remove stale dependencies from APM static lib target: by Fredrik Solenberg · 7 years ago
  29. 7ce3091 IWYU: Include <string.h> for memcpy(3) after bbf21a3fd. by Raphael Kubo da Costa · 7 years ago
  30. bbf21a3 Remove dependencies on modules:module_api from AudioProcessing. by Fredrik Solenberg · 7 years ago
  31. abbe841 This CL removes all usages of our custom ostream << overloads. by Jonas Olsson · 7 years ago
  32. 003930a Fix MID not always getting set with audio by Steve Anton · 7 years ago
  33. ef99888 Delete OnIncomingCSRCChanged and related code. by Niels Möller · 7 years ago
  34. bb50ce5 Wire up MID send value to the PeerConnection API by Steve Anton · 7 years ago
  35. 5f22365 Remove unnecessary proxy+lock code around RtcpRttStats pointer by Tommi · 7 years ago
  36. 9cfb18c Delete obsolete method RtpFeedback::OnInitializeDecoder. by Niels Möller · 7 years ago
  37. 77490b9 Pass a real audio codec pair ID to encoders that we create by Karl Wiberg · 7 years ago
  38. 763e947 Reporting packet feedback availability in AudioSendStream by Sebastian Jansson · 7 years ago
  39. 0812634 Pass a real audio codec pair ID to decoders that we create by Karl Wiberg · 7 years ago
  40. fe617a3 Adding has_packet_feedback to LimitObserver callback. by Sebastian Jansson · 7 years ago
  41. 9c1ee36 Fix low_bandwidth_audio_perf_test resource dependency on Android by Oleh Prypin · 7 years ago
  42. 7b2676f Fix low_bandwidth_audio_perf_test binary dependency on Windows by Oleh Prypin · 7 years ago
  43. 8cf0a87 Reland "Split perf-test-specific resources in low_bandwidth_audio_test" by Oleh Prypin · 7 years ago
  44. 7696bef Remove the public_deps to fileutils from test_support. by Patrik Höglund · 7 years ago
  45. 650a826 Revert "Reland "Split perf-test-specific resources in low_bandwidth_audio_test"" by Oleh Prypin · 7 years ago
  46. b3808dc Reland "Split perf-test-specific resources in low_bandwidth_audio_test" by Oleh Prypin · 7 years ago
  47. aaa882c Revert "Split perf-test-specific resources in low_bandwidth_audio_test" by Oleh Prypin · 7 years ago
  48. 4bbc150 Split perf-test-specific resources in low_bandwidth_audio_test by Oleh Prypin · 7 years ago
  49. 9599fd4 Make num-retries default a string. by Edward Lesmes · 7 years ago
  50. 5b9c684 Add num-retries flag to Android perf tests. by Edward Lesmes · 7 years ago
  51. 3faa832 Separate test/fake_audio_device on API and implementation. Step 2. by Artem Titov · 7 years ago
  52. d6fbf2a Tests: Pass codec ID argument to audio codecs by Karl Wiberg · 7 years ago
  53. 6fed924 Delete RTPPayloadRegistry::SetIncomingPayloadType. by Niels Möller · 7 years ago
  54. 881f168 Make SimpleStringBuilder into a non-template by Karl Wiberg · 7 years ago
  55. 8493594 Cleanup of TransportFeedbackObserver interface by Erik Språng · 7 years ago
  56. 12edf4c Separate build target for rtc_base/numerics/safe_minmax.h by Karl Wiberg · 7 years ago
  57. 98cd810 Production code: Pass codec ID argument to audio codecs by Karl Wiberg · 7 years ago
  58. 6723cdc Revert "Separate test/fake_audio_device on API and implementation." by Artem Titov · 7 years ago
  59. 8ea5f9a Separate test/fake_audio_device on API and implementation. by Artem Titov · 7 years ago
  60. f69e768 Propagating total_bitrate_bps from BitrateAllocator to ProbeController, part 1. by philipel · 7 years ago
  61. 3c24ea8 Removed SetTransportOverhead in transport controller. by Sebastian Jansson · 7 years ago
  62. fef0500 Adding a new string utility class: SimpleStringBuilder. by Tommi · 7 years ago
  63. f35c666 Separate build targets for aec3 and aec3_unittests by Gustaf Ullberg · 7 years ago
  64. ef9daee Using mock transport controller in audio unit tests. by Sebastian Jansson · 7 years ago
  65. 41f16be Silencing warnings in audio send stream unit tests. by Sebastian Jansson · 7 years ago
  66. 97f61ea Moved bitrate configuration to rtp controller by Sebastian Jansson · 7 years ago
  67. 1896cec Removed dependencies from audio send stream unit test by Sebastian Jansson · 7 years ago
  68. 2ae140a BUILD.gn file for api/audio. by Gustaf Ullberg · 7 years ago
  69. 4c1ffb8 Removing access to pacer in rtp controller. by Sebastian Jansson · 7 years ago
  70. e4be6da Removing access to send side cc in rtp controller. by Sebastian Jansson · 7 years ago
  71. 1e06289 Delete macro RTC_ACCESS_ON, replaced by RTC_GUARDED_BY. by Niels Möller · 7 years ago
  72. dbbb33c Stop using public_deps in common_audio. by Mirko Bonadei · 7 years ago
  73. 970b088 Reland "Break up rtc_event_log_api to solve circular dependencies." by Qingsi Wang · 7 years ago
  74. ed7b4ff Use isolated-script-test-perf-output on low_bandwidth_audio_test. by Edward Lemur · 7 years ago
  75. 06953ba Move AudioSendStream lifetime reporting into destructor by Sam Zackrisson · 7 years ago
  76. 75df728 Revert "Break up rtc_event_log_api to solve circular dependencies." by Mirko Bonadei · 7 years ago
  77. 001546d Break up rtc_event_log_api to solve circular dependencies. by Qingsi Wang · 7 years ago
  78. f120cba Delete AudioMonitor and related code. by Niels Möller · 7 years ago
  79. 65ce311 Removing useless dependencies on //testing/gmock. by Mirko Bonadei · 7 years ago
  80. 24ea822 Remove logging in audio/* from release builds. by Jonas Olsson · 7 years ago
  81. a8b7c7f Move remaining traces of VoiceEngine by Fredrik Solenberg · 7 years ago
  82. d8b041c Ignore extra arguments in low_bandwidth_audio_test. by Edward Lemur · 7 years ago
  83. 649a385 Removes usage of analog AGC. by henrika · 7 years ago
  84. 90ea504 Delete Channel::OnRecoveredPacket. by Niels Möller · 7 years ago
  85. 98d4036 Make it possible to run low_bandwidth_audio_test on Android swarming. by Edward Lemur · 7 years ago
  86. b401771 Store JSON perf results for low_bandwidth_audio_test. by Edward Lemur · 7 years ago
  87. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
  88. 3b903d0 Reconfigure, not reconstruct, AudioReceiveStreams. by Fredrik Solenberg · 7 years ago
  89. a7f2d84 Revert "Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*""" by Per Kjellander · 7 years ago
  90. c73e1f4 Revert "GN rtc_* templates: Set default visibility to webrtc_root + "/*"" by Per Kjellander · 7 years ago
  91. 588c548 GN rtc_* templates: Set default visibility to webrtc_root + "/*" by Karl Wiberg · 7 years ago
  92. 24722b3 Reland "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator." by Seth Hampson · 7 years ago
  93. 731082c Reland: Add mock_rtc_event_log.h. by Patrik Höglund · 7 years ago
  94. 5a25ab2 Revert "Add mock_rtc_event_log.h." by Edward Lemur · 7 years ago
  95. 63aea46 Add mock_rtc_event_log.h. by Patrik Höglund · 7 years ago
  96. 94dc177 Add mock_bitrate_controller.h. by Patrik Höglund · 7 years ago
  97. 6213929 Add missing files to audio_processing. by Patrik Höglund · 7 years ago
  98. 8b77aea Revert "Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator." by Lu Liu · 7 years ago
  99. d2b912a Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator. by Seth Hampson · 7 years ago
  100. f85e31b Don't (re-)configure BitrateObserver unless already sending by Oskar Sundbom · 7 years ago