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gerrit-public.fairphone.software / platform / external / webrtc / a4d873786f10eedd72de25ad0d94ad7c53c1f68a / modules / audio_coding / codecs / opus / audio_encoder_opus.cc
  1. 126f2b3 AudioEncoderOpus: Add support for 16 kHz input sample rate by Karl Wiberg · 5 years ago
  2. e0eb325 AudioEncoderOpusImpl: Remove unused static methods by Karl Wiberg · 5 years ago
  3. 7e7c5c3 WebRTC Opus C interface: Add support for non-48 kHz encode sample rate by Karl Wiberg · 5 years ago
  4. eb16697 AudioEncoderOpus: Don't mix up sample rate and RTP timestamp rate by Karl Wiberg · 5 years ago
  5. e5b9416 Decoder for multistream Opus. by Alex Loiko · 5 years ago
  6. e45c688 Remove webrtc::ProtoString. by Mirko Bonadei · 5 years ago
  7. 9dac02d Adding text log on actual opus bitrate. by Minyue Li · 6 years ago
  8. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  9. 41f3a43 Remove CodecInst pt.3 by Fredrik Solenberg · 6 years ago
  10. 8ac05cc Adds trial to use link capacity estimate in Opus encoder. by Sebastian Jansson · 6 years ago
  11. 988cc08 [Cleanup] Add missing #include. Remove useless ones. by Yves Gerey · 6 years ago
  12. 2edab4c Delete use of STR_CASE_CMP, replaced with absl::EqualsIgnoreCase. by Niels Möller · 6 years ago
  13. 83bd37c Add field trials for configuring Opus encoder packet loss rate. by Jakob Ivarsson · 6 years ago
  14. 88b68ac Create field trial for setting a minimum value for Opus encoder packet loss rate by Jakob Ivarsson · 6 years ago
  15. 918f50c Use absl::make_unique and absl::WrapUnique directly by Karl Wiberg · 6 years ago
  16. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  17. b602123 Replace rtc::Optional with absl::optional in modules/audio_coding by Danil Chapovalov · 6 years ago
  18. e40468b Move some numeric utility code from rtc_base/ to rtc_base/numerics/ by Karl Wiberg · 7 years ago
  19. eeb2765 Implement Opus bandwidth adjustment behind a FieldTrial by Alex Luebs · 7 years ago
  20. 36de62e Avoid flagging Opus DTX frames as speech. by Gustaf Ullberg · 7 years ago
  21. 12ab00b Optional: Use nullopt and implicit construction in /modules/audio_coding by Oskar Sundbom · 7 years ago
  22. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
  23. 7275e18 Hide the internal AudioEncoderOpus class by giving it an "Impl" suffix by Karl Wiberg · 7 years ago
  24. 7120742 Adding NOLINT for typedefs.h and common_types.h by Mirko Bonadei · 7 years ago
  25. 92ea95e Fixing WebRTC after moving from src/webrtc to src/ by Mirko Bonadei · 7 years ago
  26. bb54720 Moving src/webrtc into src/. by Mirko Bonadei · 7 years ago[Renamed from webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc]
  27. e1198e0 Add new ANA stats to the old GetStats() to count the number of actions taken by each controller. by ivoc · 7 years ago
  28. bf94fda Renaming probing_interval to bwe_period globally. by minyue-webrtc · 7 years ago
  29. 5d68910 Don't use rvalue reference function arguments in the audio coding module by minyue-webrtc · 7 years ago
  30. ee89e78 Replace CHECK(x && y) with two separate CHECK() calls by kwiberg · 7 years ago
  31. 54348fb Removed an obsolete DCHECK in AudioEncoderOpus. by tschumim · 7 years ago
  32. c20978e Rename webrtc/base -> webrtc/rtc_base by Edward Lemur · 7 years ago
  33. dca1e09 Revert "Update includes for webrtc/{base => rtc_base} rename (1/3)" by Henrik Kjellander · 7 years ago
  34. c8fa692 Update includes for webrtc/{base => rtc_base} rename (1/3) by kjellander · 7 years ago
  35. 96da011 Opus implementation of the AudioEncoderFactoryTemplate API by kwiberg · 7 years ago
  36. 1a610f1 Revert of Opus implementation of the AudioEncoderFactoryTemplate API (patchset #4 id:80001 of https://codereview.webrtc.org/2930243003/ ) by charujain · 7 years ago
  37. fe1aa82 Opus implementation of the AudioEncoderFactoryTemplate API by kwiberg · 7 years ago
  38. 0703856 Add SafeClamp(), which accepts args of different types by kwiberg · 7 years ago
  39. 0d6195d Hooked up Opus CBR support when configured manually or through an SdpAudioFormat. by ossu · 7 years ago
  40. 92aef17 Replace Clock with timeutils in AudioEncoder. by michaelt · 7 years ago
  41. 7c2c843 Reland of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #1 id:1 of https://codereview.webrtc.org/2786363002/ ) by mbonadei · 7 years ago
  42. a1a040a Injectable audio encoders: BuiltinAudioEncoderFactory by ossu · 7 years ago
  43. d00aad5 Revert of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #16 id:300001 of https://codereview.webrtc.org/2747863003/ ) by mbonadei · 7 years ago
  44. 16ab93b To accommodate some downstream WebRTC users we need to loosen by mbonadei · 7 years ago
  45. dadb4dc Allow ANA to receive RPLR (recoverable packet loss rate) indications by elad.alon · 7 years ago
  46. cfd88bb Fix AudioEncoderOpus::RecreateEncoderInstance() referring to old config_ by elad.alon · 7 years ago
  47. c1b57a1 Test field trial group with startswith rather than equals. by sprang · 7 years ago
  48. 6f08d7d Change frame length on very low bandwidth. by michaelt · 7 years ago
  49. a55f021 Add 120ms frame ability to ANA by michaelt · 7 years ago
  50. 54340d8 Change opus min bitrate. by michaelt · 7 years ago
  51. 0fe1216 Move more calls to webrtc::field_trial::FindFullName into ctor, thereby minimizing the non-trivial cost of repeated string comparisons. by elad.alon · 7 years ago
  52. bc5d921 Rename base/analytics/ to base/numerics/ by terelius · 8 years ago
  53. bf279fc Pass event log to ANA. by michaelt · 8 years ago
  54. 566d820 Update smoothed bitrate. by michaelt · 8 years ago
  55. 584c35a Make a the decisions of ANA optional for the opus encoder. by michaelt · 8 years ago
  56. eca373f Adding OnReceivedOverhead to AudioEncoder. by minyue · 8 years ago
  57. 4b9a2cb Reland "Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate." by minyue · 8 years ago
  58. e69b468 Revert of Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate. (patchset #5 id:240001 of https://codereview.webrtc.org/2411613002/ ) by minyue · 8 years ago
  59. 84e56d5 Renaming AudioEncoder::SetTargetBitrate and SetProjectedPacketLossRate. by minyue · 8 years ago
  60. af476c7 RTC_[D]CHECK_op: Remove "u" suffix on integer constants by kwiberg · 8 years ago
  61. 69b627d Move smoothing filter to common audio and exp_filter to base/analytics. by minyue · 8 years ago
  62. 3c3aef4 Revert of Reland "Move smoothing filter to common audio". (patchset #5 id:100001 of https://codereview.webrtc.org/2520003005/ ) by minyue · 8 years ago
  63. 223641f Reland "Move smoothing filter to common audio". by minyue · 8 years ago
  64. 875862c Let Opus increase complexity for low bitrates by henrik.lundin · 8 years ago
  65. a6f495c Simplifying audio network adaptor by moving receiver frame length range to ctor. by minyue · 8 years ago
  66. 7e30432 Hooking up audio network adaptor to VoE. by minyue · 8 years ago
  67. 41b9c80 Adding audio network adaptor to AudioEncoderOpus. by minyue · 8 years ago
  68. 85228d6 Regression test for issue where Opus DTX status was being forgotten. by ivoc · 8 years ago
  69. 2036135 AudioEncoderOpus: Default to 32 kbit/s for mono, 64 kbit/s for stereo by kwiberg · 8 years ago
  70. 8bce67b Added UMA logging for audio codec usage. A histogram statistic named "WebRTC.Audio.Encoder.CodecType" has been created. by aleloi · 8 years ago
  71. 264087f A few small cleanups of stuff caught by lint by ossu · 8 years ago
  72. 2903ba5 Reland Remove the deprecated EncodeInternal interface from AudioEncoder by ossu · 8 years ago
  73. 164bc4b Revert of Remove the deprecated EncodeInternal interface from AudioEncoder (patchset #4 id:60001 of https://codereview.webrtc.org/1864993002/ ) by ossu · 8 years ago
  74. 5222d31 Remove the deprecated EncodeInternal interface from AudioEncoder by ossu · 8 years ago
  75. d44c077 Revert of Safe numeric library: base/numerics (copied from Chromium) (patchset #11 id:250001 of https://codereview.webrtc.org/1753293002/ ) by Tommi · 8 years ago
  76. de1c81b Safe numeric library added: base/numerics (copied from Chromium) by hbos · 8 years ago
  77. 4f43fcf Renamed new EncodeInternal to EncodeImpl to ensure proper backwards compatibility. by ossu · 8 years ago
  78. 10a029e Changed AudioEncoder::Encode to take an rtc::Buffer* instead of uint8_t* and a maximum size. by ossu · 8 years ago
  79. 6955870 Convert channel counts to size_t. by Peter Kasting · 9 years ago
  80. 25702cb Misc. small cleanups. by pkasting · 9 years ago
  81. 3c652b6 modules/audio_coding: Remove some codec include dirs by kjellander@webrtc.org · 9 years ago
  82. 288886b Pass audio to AudioEncoder::Encode() in an ArrayView by kwiberg · 9 years ago
  83. 7464089 audio_coding: rename interface -> include by Henrik Kjellander · 9 years ago
  84. 91d6ede Add RTC_ prefix to (D)CHECKs and related macros. by henrikg · 9 years ago
  85. 3f5f1c2 Change return type of AudioEncoder::SetMaxPlaybackRate to void by kwiberg · 9 years ago
  86. 12cfc9b Fold AudioEncoderMutable into AudioEncoder by kwiberg · 9 years ago
  87. dce40cf Update a ton of audio code to use size_t more correctly and in general reduce by Peter Kasting · 9 years ago
  88. b297c5a Miscellaneous changes split from https://codereview.webrtc.org/1230503003 . by pkasting · 9 years ago
  89. 3e89dbf Add AudioEncoder::GetTargetBitrate by Henrik Lundin · 9 years ago
  90. bba7807 Reland "Upconvert various types to int.", misc. codecs portion. by Peter Kasting · 9 years ago
  91. 728d903 Reformat existing code. There should be no functional effects. by Peter Kasting · 9 years ago
  92. b7e5054 Match existing type usage better. by Peter Kasting · 9 years ago
  93. cb18097 Revert "Upconvert various types to int." by Peter Kasting · 9 years ago
  94. 83ad33a Upconvert various types to int. by Peter Kasting · 9 years ago
  95. 092041c Setting OPUS_SIGNAL_VOICE when enable DTX. by Minyue Li · 9 years ago
  96. dcccab3 New interface: AudioEncoderMutable by Karl Wiberg · 9 years ago
  97. 9afaee7 Reland 8749: AudioEncoder: return EncodedInfo from Encode() and EncodeInternal() by jmarusic@webrtc.org · 9 years ago
  98. 019955d Revert 8749 "We changed Encode() and EncodeInternal() return typ..." by tommi@webrtc.org · 9 years ago
  99. 0cb612b We changed Encode() and EncodeInternal() return type from bool to void in this issue: by jmarusic@webrtc.org · 9 years ago
  100. e16bfde Adding flag to force Opus application and DTX while toggling. by minyue@webrtc.org · 9 years ago