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gerrit-public.fairphone.software / platform / external / webrtc / c58c8a54224d4e2269b69c6218d2a0d89a2f650f / audio
  1. 8fb5746 Delete obsolete interface class RtpData by Niels Möller · 6 years ago
  2. 5571812 Adding rtcp report interval into RTCConfiguration. by Jiawei Ou · 6 years ago
  3. 80c6762 Tweak ChannelReceive interface, to make it closer to ChannelReceiveProxy by Niels Möller · 6 years ago
  4. b768e88 Reland "Isolating APM API build target: making :api an actual target." by Alessio Bazzica · 6 years ago
  5. 61c6e56 Revert "Isolating APM API build target: making :api an actual target." by Alessio Bazzica · 6 years ago
  6. a7f77a7 Isolating APM API build target: making :api an actual target. by Alessio Bazzica · 6 years ago
  7. c572ff3 Add default constructor for rtc::Event by Niels Möller · 6 years ago
  8. 967f7d5 Add audio level to CSRC class by Jonas Oreland · 6 years ago
  9. fd1a2fb Set RtpRtcp config receive_only in voe::ChannelReceive by Niels Möller · 6 years ago
  10. 273d029 Implement data channel methods in LoopbackMediaTransport. by Bjorn Mellem · 6 years ago
  11. 2365936 Hide the AudioEncoderCng class behind a create function by Karl Wiberg · 6 years ago
  12. 56ef305 Move event logging of config into AudioSendStream. by Oskar Sundbom · 6 years ago
  13. 21cddff Harmonize paths to dependent targets. by Yves Gerey · 6 years ago
  14. fcc3981 Revert "Use only first payload timestamp for RTCP SR generation for audio" by Ilya Nikolaevskiy · 6 years ago
  15. 9a0662a Use only first payload timestamp for RTCP SR generation for audio by Ilya Nikolaevskiy · 6 years ago
  16. 9190b82 Propagate SDP negotiation of extmap-allow-mixed to RtpHeaderExtensionMap by Johannes Kron · 6 years ago
  17. 7d76a31 Use MediaTransportInterface, for audio streams. by Niels Möller · 6 years ago
  18. 78410ad Fixes use after free error when setting a new FrameEncryptor on ChannelSend. by Benjamin Wright · 6 years ago
  19. 359d60a Adds target rate to audio send stream stats. by Sebastian Jansson · 6 years ago
  20. c0e4d45 Adds BitrateAllocation struct to OnBitrateUpdated. by Sebastian Jansson · 6 years ago
  21. 988cc08 [Cleanup] Add missing #include. Remove useless ones. by Yves Gerey · 6 years ago
  22. 67b011d Use BitrateAllocatorInterface in AudioSendStream and VideoSendStream by Niels Möller · 6 years ago
  23. 648d28a Media engine and channel support for per-channel dscp values, specified by RtpParameter by Tim Haloun · 6 years ago
  24. 2dfa998 Reland "Prefix flag macros with WEBRTC_." by Mirko Bonadei · 6 years ago
  25. c538fc7 Revert "Prefix flag macros with WEBRTC_." by Mirko Bonadei · 6 years ago
  26. 5ccdc13 Prefix flag macros with WEBRTC_. by Mirko Bonadei · 6 years ago
  27. bfb444c Adds new CryptoOption crypto_options.frame.require_frame_encryption. by Benjamin Wright · 6 years ago
  28. 40a7a35 Extract functionality of test_main into separate library. by Artem Titov · 6 years ago
  29. b686396 Makes AudioSendStream signal that it's part of allocation. by Sebastian Jansson · 6 years ago
  30. 75e3647 Switch usages of DefaultNetworkSimulationConfig to BuiltInNetworkBehaviorConfig by Artem Titov · 6 years ago
  31. 2e00abc Reland "[cleanup] Remove useless includes." by Yves Gerey · 6 years ago
  32. 64be7fa Move FecController to RtpVideoSender. by Stefan Holmer · 6 years ago
  33. 96a0f61 Revert "[cleanup] Remove useless includes." by Oleh Prypin · 6 years ago
  34. be8b534 [cleanup] Remove useless includes. by Yves Gerey · 6 years ago
  35. ae4237e Set ChannelReceive transport at construction time. by Niels Möller · 6 years ago
  36. 84583f6 Enable End-to-End Encrypted Audio Payloads. by Benjamin Wright · 6 years ago
  37. 530ead4 Split voe::Channel into ChannelSend and ChannelReceive by Niels Möller · 6 years ago
  38. 4a72ba9 Delete RtpReceiver and related code. by Niels Möller · 6 years ago
  39. b222f49 Split ChannelProxy into send and receive classes. by Niels Möller · 6 years ago
  40. 35fa280 Adds allocated rate without feedback to new congestion controller. by Sebastian Jansson · 6 years ago
  41. 1f3206c Change ReceiveStatistics to implement RtpPacketSinkInterface, part 1 by Niels Möller · 6 years ago
  42. 17f4878 Remove deprecated field_trial_default and metrics_default. by Mirko Bonadei · 6 years ago
  43. 7988e5c Remove echo_cancellation() and echo_control_mobile() interface access outside APM by Sam Zackrisson · 6 years ago
  44. 637b0b5 Make Python-based performance tests output an empty result output.json by Oleh Prypin · 6 years ago
  45. db12856 Cleanup modules_common_types by Danil Chapovalov · 6 years ago
  46. 6151828 Delete always true member voe::Channel::pacing_enabled_ by Niels Möller · 6 years ago
  47. 5304a32 Delete StreamStatistician::IsRetransmitOfOldPacket by Niels Möller · 6 years ago
  48. 8fdcac3 Remove clang:find_bad_constructs suppression from call:call. by Mirko Bonadei · 6 years ago
  49. 2370b08 Revert "Update packetsLost and jitter stats any time a packet is received." by Qingsi Wang · 6 years ago
  50. 4e199e9 Mark DirectTransport subclasses ctors as deprecated and switch from them by Artem Titov · 6 years ago
  51. 46c4e60 Introduce SimulatedNetworkReceiverInterface. by Artem Titov · 6 years ago
  52. fa2b2d6 Delete use of RtpPayloadRegistry. by Niels Möller · 6 years ago
  53. 30b4839 Refactor voe::Channel to not use RtpReceiver. by Niels Möller · 6 years ago
  54. 9701e0c Makes treatment of received reports of packets lost signed. by Sebastian Jansson · 6 years ago
  55. fa4e185 Delete class voe::RtcEventLogProxy by Niels Möller · 6 years ago
  56. 848d6d3 Change Channel::GetRtpRtcp to return only RtpRtcp, not RtpReceiver. by Niels Möller · 6 years ago
  57. 7008287 Delete struct webrtc::PacketTime. by Niels Möller · 6 years ago
  58. 264bee8 Remove memcheck. by Mirko Bonadei · 6 years ago
  59. ab4a530 Delete telephone-event handling from RTPReceiverAudio. by Niels Möller · 6 years ago
  60. a12c42a Delete root header file typedef.h. by Niels Möller · 6 years ago
  61. 3890262 Reland "Removing unneeded dependency." by Mirko Bonadei · 6 years ago
  62. a61f7db Revert "Removing unneeded dependency." by Mirko Bonadei · 6 years ago
  63. 06f66c7 Removing unneeded dependency. by Mirko Bonadei · 6 years ago
  64. bbbe4e1 Better handle target audio bitrate allocation. by Alex Narest · 6 years ago
  65. 918f50c Use absl::make_unique and absl::WrapUnique directly by Karl Wiberg · 6 years ago
  66. 64b17c2 Remove StreamStatistician::IsPacketInOrder by Danil Chapovalov · 6 years ago
  67. 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
  68. 8491693 Update packetsLost and jitter stats any time a packet is received. by Taylor Brandstetter · 6 years ago
  69. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  70. b9b146c Replace rtc::Optional with absl::optional in audio, call and video by Danil Chapovalov · 6 years ago
  71. 867e510 Enable send side audio TWCC only if WebRTC-Audio-ForceNoTWCC is not enabled. by Alex Narest · 6 years ago
  72. f782492 Delete RtpFeedback. The ssrc for a receive stream should be known at by Niels Möller · 6 years ago
  73. eda0087 Drop the RTT as input to IsRetransmitOfOldPacket. by Niels Möller · 6 years ago
  74. 5f83cf0 Replacing rtc::TimeDelta with webrtc::TimeDelta. by Sebastian Jansson · 6 years ago
  75. 24ad720 Uses config struct with bitrate allocator. by Sebastian Jansson · 6 years ago
  76. 104ad0b Remove stale dependencies from APM static lib target: by Fredrik Solenberg · 6 years ago
  77. 7ce3091 IWYU: Include <string.h> for memcpy(3) after bbf21a3fd. by Raphael Kubo da Costa · 6 years ago
  78. bbf21a3 Remove dependencies on modules:module_api from AudioProcessing. by Fredrik Solenberg · 6 years ago
  79. abbe841 This CL removes all usages of our custom ostream << overloads. by Jonas Olsson · 6 years ago
  80. 003930a Fix MID not always getting set with audio by Steve Anton · 6 years ago
  81. ef99888 Delete OnIncomingCSRCChanged and related code. by Niels Möller · 6 years ago
  82. bb50ce5 Wire up MID send value to the PeerConnection API by Steve Anton · 6 years ago
  83. 5f22365 Remove unnecessary proxy+lock code around RtcpRttStats pointer by Tommi · 6 years ago
  84. 9cfb18c Delete obsolete method RtpFeedback::OnInitializeDecoder. by Niels Möller · 6 years ago
  85. 77490b9 Pass a real audio codec pair ID to encoders that we create by Karl Wiberg · 6 years ago
  86. 763e947 Reporting packet feedback availability in AudioSendStream by Sebastian Jansson · 6 years ago
  87. 0812634 Pass a real audio codec pair ID to decoders that we create by Karl Wiberg · 6 years ago
  88. fe617a3 Adding has_packet_feedback to LimitObserver callback. by Sebastian Jansson · 6 years ago
  89. 9c1ee36 Fix low_bandwidth_audio_perf_test resource dependency on Android by Oleh Prypin · 6 years ago
  90. 7b2676f Fix low_bandwidth_audio_perf_test binary dependency on Windows by Oleh Prypin · 6 years ago
  91. 8cf0a87 Reland "Split perf-test-specific resources in low_bandwidth_audio_test" by Oleh Prypin · 6 years ago
  92. 7696bef Remove the public_deps to fileutils from test_support. by Patrik Höglund · 6 years ago
  93. 650a826 Revert "Reland "Split perf-test-specific resources in low_bandwidth_audio_test"" by Oleh Prypin · 6 years ago
  94. b3808dc Reland "Split perf-test-specific resources in low_bandwidth_audio_test" by Oleh Prypin · 6 years ago
  95. aaa882c Revert "Split perf-test-specific resources in low_bandwidth_audio_test" by Oleh Prypin · 6 years ago
  96. 4bbc150 Split perf-test-specific resources in low_bandwidth_audio_test by Oleh Prypin · 6 years ago
  97. 9599fd4 Make num-retries default a string. by Edward Lesmes · 6 years ago
  98. 5b9c684 Add num-retries flag to Android perf tests. by Edward Lesmes · 6 years ago
  99. 3faa832 Separate test/fake_audio_device on API and implementation. Step 2. by Artem Titov · 6 years ago
  100. d6fbf2a Tests: Pass codec ID argument to audio codecs by Karl Wiberg · 6 years ago