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gerrit.openfyde.cn / webrtc.googlesource.com / src / 0cd0dd3b07261cb15e821622b351b5ce80f98ea6 / modules / audio_coding / acm2 / audio_coding_module.cc
  1. 714e3cb Adopt absl::string_view in modules/audio_coding/ by Ali Tofigh · 3 years ago
  2. d325196 Prepare to rename RTC_NOTREACHED to RTC_DCHECK_NOTREACHED by Artem Titov · 3 years, 8 months ago
  3. bf08745 Implement RTCOutboundRtpStreamStats.targetBitrate for audio. by Jakob Ivarsson · 3 years, 8 months ago
  4. cfea218 Use backticks not vertical bars to denote variables in comments by Artem Titov · 4 years ago
  5. d00ce74 Use backticks not vertical bars to denote variables in comments for /modules/audio_coding by Artem Titov · 4 years ago
  6. 190244b Remove all #include <assert.h>/<cassert> and usage in Obj-C code. by Mirko Bonadei · 4 years ago
  7. 25ab322 Replace assert() with RTC_DCHECK(). by Mirko Bonadei · 4 years ago
  8. 0df0fae Migrate modules/audio_coding, audio_mixer/ and audio_processing/ to webrtc::Mutex. by Markus Handell · 5 years ago
  9. d82a02c ACM: Corrected temporary buffer size by Per Åhgren · 5 years ago
  10. 3bdc5e9 Delete ACMVADCallback by Niels Möller · 5 years ago
  11. dea73ee Pass absolute capture time from WebRtcVoiceEngine to ACM. by Minyue Li · 5 years ago
  12. ff0e4db Reland "Send absolute capture time through audio coding module." by Minyue Li · 5 years ago
  13. 4175914 Revert "Send absolute capture time through audio coding module." by Minyue Li · 5 years ago
  14. 48655cf Send absolute capture time through audio coding module. by Minyue Li · 5 years ago
  15. 4dd56a3 ACM: Adding unittests for the remixing functionality by Per Åhgren · 6 years ago
  16. bb55c5e Correct the upmixing of mono to stereo in ACM2 by Per Åhgren · 6 years ago
  17. 048b10a Correcting the ACM upmixing from mono/stereo to surround by Per Åhgren · 6 years ago
  18. 4f2e940 ACM: Adding support for more than 2 channels in the send pipeline by Per Åhgren · 6 years ago
  19. 3354157 Add support for 192kHz input audio sample rate. by henrika · 6 years ago
  20. 3247244 Delete unused method AudioCodingModule::GetDecodingCallStatistics by Niels Möller · 6 years ago
  21. 5ceb4ac Delete some unused AudioCodingModule methods by Niels Möller · 6 years ago
  22. dc5ed5c Delete NACK-related methods from AudioCodingModule by Niels Möller · 6 years ago
  23. b90d38a Delete unused Opus-specific methods of AudioCodingModule by Niels Möller · 6 years ago
  24. 75caef7 Delete unused ACM members isac_decoder_16k_ and isac_decoder_32k_ by Niels Möller · 6 years ago
  25. c653172 Delete obsolete method AudioCodingModule::SetBitRate by Niels Möller · 6 years ago
  26. a4d8737 Format almost everything. by Jonas Olsson · 6 years ago
  27. 053c371 Audio coding: Don't choke when RTP timestamp rate > sample rate by Karl Wiberg · 6 years ago
  28. c35b6e6 Deprecate RTPFragmentationHeader argument to AudioPacketizationCallback::SendData by Niels Möller · 6 years ago
  29. c936cb6 Make AudioFrameType an enum class, and move to audio_coding_module_typedefs.h by Niels Möller · 6 years ago
  30. 87e2d78 Prepare for splitting FrameType into AudioFrameType and VideoFrameType by Niels Möller · 6 years ago
  31. 6543881 2nd reland of https://webrtc-review.googlesource.com/c/src/+/114883 by Alex Loiko · 6 years ago
  32. 8b3db59 Revert "Reland of https://webrtc-review.googlesource.com/c/src/+/114883" by Alex Loiko · 6 years ago
  33. 5341aac Reland of https://webrtc-review.googlesource.com/c/src/+/114883 by Alex Loiko · 6 years ago
  34. b35bacc Fix NetEq minimum and maximum delay always reset on acm creation. by Ruslan Burakov · 6 years ago
  35. ffd1f93 Revert "Tests for multi-stream Opus." by Mirko Bonadei · 6 years ago
  36. 9c31ac2 Tests for multi-stream Opus. by Alex Loiko · 6 years ago
  37. afb5dbb Update ACM to use RTPHeader instead of WebRtcRTPHeader by Niels Möller · 6 years ago
  38. 3b50f9f Propagate base minimum delay to audio_receiver_stream by Ruslan Burakov · 6 years ago
  39. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 7 years ago
  40. f693bfa Remove CodecInst pt.2 by Fredrik Solenberg · 7 years ago
  41. 49c33ce AudioCodingModule: Remove support for creating encoders by Karl Wiberg · 7 years ago
  42. 988cc08 [Cleanup] Add missing #include. Remove useless ones. by Yves Gerey · 7 years ago
  43. 2edab4c Delete use of STR_CASE_CMP, replaced with absl::EqualsIgnoreCase. by Niels Möller · 7 years ago
  44. eddd366 Delete unused method AudioCodingModuleImpl::SetOpusApplication. by Niels Möller · 7 years ago
  45. 764c14c8 Delete unused AudioCodingModule methods. by Niels Möller · 7 years ago
  46. 18f1adc Delete AudioCodingModule::LeastRequiredDelayMs and related NetEq code. by Niels Möller · 7 years ago
  47. ec93075 Delete deprecated methods on AudioCodingModule by Niels Möller · 7 years ago
  48. 665174f Reformat the WebRTC code base by Yves Gerey · 7 years ago
  49. b602123 Replace rtc::Optional with absl::optional in modules/audio_coding by Danil Chapovalov · 7 years ago
  50. bbf21a3 Remove dependencies on modules:module_api from AudioProcessing. by Fredrik Solenberg · 7 years ago
  51. 5817d3d AudioCodingModule::Create(): Require caller to supply an AudioDecoderFactory by Karl Wiberg · 7 years ago
  52. abbff89 Add new UMA metric for NetEq target buffer delay by Henrik Lundin · 8 years ago
  53. e40468b Move some numeric utility code from rtc_base/ to rtc_base/numerics/ by Karl Wiberg · 8 years ago
  54. 12ab00b Optional: Use nullopt and implicit construction in /modules/audio_coding by Oskar Sundbom · 8 years ago
  55. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 8 years ago
  56. d4a790f Remove AudioCodingModule::IncomingPayload by Henrik Lundin · 8 years ago
  57. c7b4a45 Remove various IDs: by solenberg · 8 years ago
  58. e423a9de Revert of Remove various IDs (patchset #7 id:120001 of https://codereview.webrtc.org/3019543002/ ) by solenberg · 8 years ago
  59. 2d0f775 Remove various IDs: by solenberg · 8 years ago
  60. 48d96c0 Corrected upper limits of NetEq minimum and maximum delay. by Gustaf Ullberg · 8 years ago
  61. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 8 years ago
  62. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 8 years ago[Renamed from webrtc/modules/audio_coding/acm2/audio_coding_module.cc]
  63. e1198e0 Add new ANA stats to the old GetStats() to count the number of actions taken by each controller. by ivoc · 8 years ago
  64. 56359be Update thread annotiation macros in modules to use RTC_ prefix by danilchap · 8 years ago
  65. 36344a0 Fix incorrect memset on muted frames. by Jonathan Yu · 8 years ago
  66. bc8ee33 Remove verbose logs from audio_coding_module.cc. by Noah Richards · 8 years ago
  67. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 8 years ago
  68. dca1e09 Revert "Update includes for webrtc/{base => rtc_base} rename (1/3)" by Henrik Kjellander · 8 years ago
  69. c8fa692 Update includes for webrtc/{base => rtc_base} rename (1/3) by kjellander · 8 years ago
  70. 300ec8c Remove WEBRTC_TRACE from webrtc/modules/audio_coding by Alex Loiko · 8 years ago
  71. 36b1a5f Add mute state field to AudioFrame and switch some callers to use it. Also make AudioFrame::data_ private and instead provide: by yujo · 8 years ago
  72. b8c55b1 Handle padded audio packets correctly by henrik.lundin · 8 years ago
  73. 368f5cf Replace use of system_wrappers/include/logging.h by base/logging.h. by nisse · 8 years ago
  74. 1c07c70 Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType" by kwiberg · 8 years ago
  75. 670a7f3 Revert of WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType (patchset #13 id:260001 of https://codereview.webrtc.org/2686043006/ ) by kwiberg · 8 years ago
  76. 1724cfb WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType by kwiberg · 8 years ago
  77. d3edd77 Introduce dchecked_cast, and start using it by kwiberg · 8 years ago
  78. 087bd34 Move AudioDecoder and related stuff to the api/ directory by kwiberg · 8 years ago
  79. 566d820 Update smoothed bitrate. by michaelt · 9 years ago
  80. 4b9a2cb Reland "Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate." by minyue · 9 years ago
  81. e69b468 Revert of Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate. (patchset #5 id:240001 of https://codereview.webrtc.org/2411613002/ ) by minyue · 9 years ago
  82. 84e56d5 Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate. by minyue · 9 years ago
  83. af476c7 RTC_[D]CHECK_op: Remove "u" suffix on integer constants by kwiberg · 9 years ago
  84. e280cde Voe::Channel: Turned GetPlayoutFrequency into GetRtpTimestampRateHz. by ossu · 9 years ago
  85. 5adaf73 AudioCodingModule: Specify decoders using SdpAudioFormat by kwiberg · 9 years ago
  86. 24c7c12 Move FunctionView from AudioCodingModule to the rtc namespace by kwiberg · 9 years ago
  87. 6b19b56 AcmReceiver: Ask NetEq to delete all decoders at once instead of one by one by kwiberg · 9 years ago
  88. bfb78d1 Revert of AcmReceiver: Ask NetEq to delete all decoders at once instead of one by one (patchset #2 id:20001 of https://codereview.webrtc.org/2342313002/ ) by kwiberg · 9 years ago
  89. f6232b4 AcmReceiver: Ask NetEq to delete all decoders at once instead of one by one by kwiberg · 9 years ago
  90. 36a4388 Fix Chromium clang plugin warnings by kwiberg · 9 years ago
  91. fcada90 Fixing timestamp comparison assert. by deadbeef · 9 years ago
  92. b3f1c5d Add NetEq::FilteredCurrentDelayMs() and use it in VoiceEngine by henrik.lundin · 9 years ago
  93. 85228d6 Regression test for issue where Opus DTX status was being forgotten. by ivoc · 9 years ago
  94. 63fb95a Fixed time moving backwards in the AudioCodingModule. by ossu · 9 years ago
  95. c13ded5 Move AudioCodingModuleImpl to anonymous namespace in audio_coding_module.cc by kwiberg · 9 years ago
  96. e352578 Moved injection of AudioDecoderFactory into voe::Channel. by ossu · 9 years ago
  97. 6955870 Convert channel counts to size_t. by Peter Kasting · 10 years ago
  98. 3e6db23 audio_coding: remove "main" directory by kjellander · 10 years ago[Renamed (92%) from webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc]
  99. be57983 Rename Maybe to Optional by Karl Wiberg · 10 years ago
  100. 7464089 audio_coding: rename interface -> include by Henrik Kjellander · 10 years ago