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gerrit-public.fairphone.software / platform / external / webrtc / fe3dfeeb3c93e7d9a9df9461150c32c03b4925ac / audio / BUILD.gn
  1. 5c2f1f0 Add some missing includes and dependencies. by Bjorn Terelius · 6 years ago
  2. 40d5533 Include absl/memory/memory.h if absl::make_unique is used by Steve Anton · 6 years ago
  3. 31d8b52 Delete unneeded includes of rtc_base/stringutils.h. by Niels Möller · 6 years ago
  4. f693bfa Remove CodecInst pt.2 by Fredrik Solenberg · 6 years ago
  5. ff05816 Delete the WebRTC.Call.TimeSendingAudioRtpPacketsInSeconds metric by Sam Zackrisson · 6 years ago
  6. e3abb81 Decouple //rtc_base:rtc_base_tests_utils from gunit. by Mirko Bonadei · 6 years ago
  7. 2222a80 Delete unneeded includes of common_types.h and gn deps on webrtc_common. by Niels Möller · 6 years ago
  8. dced9f6 Delete class ChannelSendProxy by Niels Möller · 6 years ago
  9. 349ade3 Delete class ChannelReceiveProxy. by Niels Möller · 6 years ago
  10. b768e88 Reland "Isolating APM API build target: making :api an actual target." by Alessio Bazzica · 6 years ago
  11. 61c6e56 Revert "Isolating APM API build target: making :api an actual target." by Alessio Bazzica · 6 years ago
  12. a7f77a7 Isolating APM API build target: making :api an actual target. by Alessio Bazzica · 6 years ago
  13. 2365936 Hide the AudioEncoderCng class behind a create function by Karl Wiberg · 6 years ago
  14. 56ef305 Move event logging of config into AudioSendStream. by Oskar Sundbom · 6 years ago
  15. 21cddff Harmonize paths to dependent targets. by Yves Gerey · 6 years ago
  16. 7d76a31 Use MediaTransportInterface, for audio streams. by Niels Möller · 6 years ago
  17. 78410ad Fixes use after free error when setting a new FrameEncryptor on ChannelSend. by Benjamin Wright · 6 years ago
  18. 40a7a35 Extract functionality of test_main into separate library. by Artem Titov · 6 years ago
  19. 530ead4 Split voe::Channel into ChannelSend and ChannelReceive by Niels Möller · 6 years ago
  20. b222f49 Split ChannelProxy into send and receive classes. by Niels Möller · 6 years ago
  21. 17f4878 Remove deprecated field_trial_default and metrics_default. by Mirko Bonadei · 6 years ago
  22. 4e199e9 Mark DirectTransport subclasses ctors as deprecated and switch from them by Artem Titov · 6 years ago
  23. 46c4e60 Introduce SimulatedNetworkReceiverInterface. by Artem Titov · 6 years ago
  24. 264bee8 Remove memcheck. by Mirko Bonadei · 6 years ago
  25. 3890262 Reland "Removing unneeded dependency." by Mirko Bonadei · 6 years ago
  26. a61f7db Revert "Removing unneeded dependency." by Mirko Bonadei · 6 years ago
  27. 06f66c7 Removing unneeded dependency. by Mirko Bonadei · 6 years ago
  28. 918f50c Use absl::make_unique and absl::WrapUnique directly by Karl Wiberg · 6 years ago
  29. b9b146c Replace rtc::Optional with absl::optional in audio, call and video by Danil Chapovalov · 6 years ago
  30. 5f83cf0 Replacing rtc::TimeDelta with webrtc::TimeDelta. by Sebastian Jansson · 6 years ago
  31. bbf21a3 Remove dependencies on modules:module_api from AudioProcessing. by Fredrik Solenberg · 7 years ago
  32. abbe841 This CL removes all usages of our custom ostream << overloads. by Jonas Olsson · 7 years ago
  33. 9c1ee36 Fix low_bandwidth_audio_perf_test resource dependency on Android by Oleh Prypin · 7 years ago
  34. 7b2676f Fix low_bandwidth_audio_perf_test binary dependency on Windows by Oleh Prypin · 7 years ago
  35. 8cf0a87 Reland "Split perf-test-specific resources in low_bandwidth_audio_test" by Oleh Prypin · 7 years ago
  36. 7696bef Remove the public_deps to fileutils from test_support. by Patrik Höglund · 7 years ago
  37. 650a826 Revert "Reland "Split perf-test-specific resources in low_bandwidth_audio_test"" by Oleh Prypin · 7 years ago
  38. b3808dc Reland "Split perf-test-specific resources in low_bandwidth_audio_test" by Oleh Prypin · 7 years ago
  39. aaa882c Revert "Split perf-test-specific resources in low_bandwidth_audio_test" by Oleh Prypin · 7 years ago
  40. 4bbc150 Split perf-test-specific resources in low_bandwidth_audio_test by Oleh Prypin · 7 years ago
  41. 3faa832 Separate test/fake_audio_device on API and implementation. Step 2. by Artem Titov · 7 years ago
  42. 12edf4c Separate build target for rtc_base/numerics/safe_minmax.h by Karl Wiberg · 7 years ago
  43. 6723cdc Revert "Separate test/fake_audio_device on API and implementation." by Artem Titov · 7 years ago
  44. 8ea5f9a Separate test/fake_audio_device on API and implementation. by Artem Titov · 7 years ago
  45. fef0500 Adding a new string utility class: SimpleStringBuilder. by Tommi · 7 years ago
  46. f35c666 Separate build targets for aec3 and aec3_unittests by Gustaf Ullberg · 7 years ago
  47. ef9daee Using mock transport controller in audio unit tests. by Sebastian Jansson · 7 years ago
  48. 1896cec Removed dependencies from audio send stream unit test by Sebastian Jansson · 7 years ago
  49. 2ae140a BUILD.gn file for api/audio. by Gustaf Ullberg · 7 years ago
  50. e4be6da Removing access to send side cc in rtp controller. by Sebastian Jansson · 7 years ago
  51. dbbb33c Stop using public_deps in common_audio. by Mirko Bonadei · 7 years ago
  52. 970b088 Reland "Break up rtc_event_log_api to solve circular dependencies." by Qingsi Wang · 7 years ago
  53. 75df728 Revert "Break up rtc_event_log_api to solve circular dependencies." by Mirko Bonadei · 7 years ago
  54. 001546d Break up rtc_event_log_api to solve circular dependencies. by Qingsi Wang · 7 years ago
  55. 65ce311 Removing useless dependencies on //testing/gmock. by Mirko Bonadei · 7 years ago
  56. a8b7c7f Move remaining traces of VoiceEngine by Fredrik Solenberg · 7 years ago
  57. 98d4036 Make it possible to run low_bandwidth_audio_test on Android swarming. by Edward Lemur · 7 years ago
  58. 8f5787a Move ownership of voe::Channel into Audio[Receive|Send]Stream. by Fredrik Solenberg · 7 years ago
  59. 731082c Reland: Add mock_rtc_event_log.h. by Patrik Höglund · 7 years ago
  60. 5a25ab2 Revert "Add mock_rtc_event_log.h." by Edward Lemur · 7 years ago
  61. 63aea46 Add mock_rtc_event_log.h. by Patrik Höglund · 7 years ago
  62. 94dc177 Add mock_bitrate_controller.h. by Patrik Höglund · 7 years ago
  63. 6213929 Add missing files to audio_processing. by Patrik Höglund · 7 years ago
  64. 2a87797 Remove voe::TransmitMixer by Fredrik Solenberg · 7 years ago
  65. a8005cf Fix circular dependencies between optional, array_view, and rtc_base. by Patrik Höglund · 7 years ago
  66. d37709b Revert "Fix circular dependencies between optional, array_view, and rtc_base." by Patrik Höglund · 7 years ago
  67. a9e0924 Fix circular dependencies between optional, array_view, and rtc_base. by Patrik Höglund · 7 years ago
  68. cedd351 Do not add audio bitrate observer if TWCC sending is not supported by Alex Narest · 7 years ago
  69. b5728d9 Stop using public_deps in modules/rtp_rtcp. by Mirko Bonadei · 7 years ago
  70. 56d4609 Use the new AudioProcessing statistics everywhere. by Ivo Creusen · 7 years ago
  71. c0e6804 Fix deps of audio:audio_tests. by Patrik Höglund · 7 years ago
  72. 61a7b14 Removing conditional visibility. by Mirko Bonadei · 7 years ago
  73. 5f6bf24 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API (II) by henrika · 7 years ago
  74. 990d6b8 Revert "Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API" by Mirko Bonadei · 7 years ago
  75. 90bace0 Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API by henrika · 7 years ago
  76. 245660a Fix Gn untracked headers in webrtc/call. by Mirko Bonadei · 7 years ago
  77. 2011075 MB: Add support for isolating scripts + isolate low_bandwidth_audio_test.py. by Edward Lemur · 7 years ago
  78. 18f5427 Remove voe_auto_test and add new tests to cover the missing cases. by solenberg · 7 years ago
  79. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  80. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/audio/BUILD.gn]
  81. 73276ad - Removes voe_conference_test. by Fredrik Solenberg · 7 years ago
  82. 84f6a3f Move optional.h to webrtc/api/ by kwiberg · 7 years ago
  83. 9b2f20c Replace gflags usages with rtc_base/flags in all targets based on test_main by oprypin · 7 years ago
  84. 413ee9a Use SingleThreadedTaskQueue in DirectTransport by eladalon · 7 years ago
  85. 037f3e4 Replace absolute path with relative path for GN files. by Jianjun Zhu · 7 years ago
  86. f6a861a Remove remains of webrtc/base by ehmaldonado · 7 years ago
  87. c58f8c0 Adds a histogram metric tracking for how long audio RTP packets are sent by saza · 7 years ago
  88. 9d11764 Reimplemeted "Test and fix for huge bwe drop after alr state" by tschumim · 7 years ago
  89. c024740 Use relative paths in GN files. by jianjun.zhu · 7 years ago
  90. 370dd47 Revert of Remove remains of webrtc/base (patchset #7 id:120001 of https://codereview.webrtc.org/2973183002/ ) by ehmaldonado · 7 years ago
  91. 9483b49 Remove remains of webrtc/base by ehmaldonado · 7 years ago
  92. e75d96b Revert of Test and fix for huge bwe drop after alr state. (patchset #13 id:320001 of https://codereview.webrtc.org/2931873002/ ) by terelius · 7 years ago
  93. 0f15f92 Introduce RtpStreamReceiverInterface and RtpStreamReceiverControllerInterface. by nisse · 7 years ago
  94. 37aa8ba Test and fix for huge bwe drop after alr state. by tschumim · 7 years ago
  95. d76b7b2 New targets call:rtp_interfaces, call:rtp_receiver, call:rtp_sender. by nisse · 7 years ago
  96. 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
  97. eb1fde4 Injectable audio encoders: Moved audio encoder, factory and builtin factory to api/. by ossu · 7 years ago
  98. 20a4b3f Injectable audio encoders: WebRtcVoiceEngine and company by ossu · 7 years ago
  99. e0629c0 GN: Tighten up test target visibility + refactorings by kjellander · 7 years ago
  100. f250100 Add POLQA to low bandwidth audio test by oprypin · 8 years ago