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gerrit-public.fairphone.software / platform / external / webrtc / cd2a92f8e0b64a05668c8b5a7ea2cffe64b02cf6 / modules / audio_coding / codecs
  1. cd2a92f Removes RPLR based FEC controller. by Sebastian Jansson · 4 years, 9 months ago
  2. fb075d5 Removing unused Opus wrapper API: WebRTCOpus_DecodePlc. by Minyue Li · 4 years, 9 months ago
  3. 54d0278 Renaming opus_interface.c to opus_interface.cc. by Minyue Li · 4 years, 9 months ago
  4. 79e653c Apply bitrate boosting depending on field-trial. by Ivo Creusen · 4 years, 9 months ago
  5. 1272dad Reduce log level of Opus bitrate. by Minyue Li · 4 years, 9 months ago
  6. 86d053c Use source_sets in component builds and static_library in release builds. by Mirko Bonadei · 4 years, 9 months ago
  7. 62aee93 Adds trial to calculate audio overhead based on available data. by Sebastian Jansson · 4 years, 10 months ago
  8. f34116e Replacing bandwidth adaptation trial with stable target in Opus encoder. by Sebastian Jansson · 4 years, 10 months ago
  9. ef14f07 Delete AudioDecoder method IncomingPacket by Niels Möller · 4 years, 10 months ago
  10. 544dfb5 Delete isac GetBandwidthInfo/SetBandwidthInfo by Niels Möller · 4 years, 10 months ago
  11. 48b32b7 Delete support for enabling adaptive isac mode by Niels Möller · 4 years, 10 months ago
  12. e0b3167 Delete dead code inside #ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED by Niels Möller · 4 years, 10 months ago
  13. 6a09263 Delete obsolete isac "assign" api by Niels Möller · 4 years, 10 months ago
  14. 317a1f0 Use std::make_unique instead of absl::make_unique. by Mirko Bonadei · 4 years, 10 months ago
  15. 9b29d69 Make ANA frame length controller more robust to encoder frame lengths. by Minyue Li · 5 years ago
  16. 608e6ba Add AudioDecoderIsacT::Config to include sampling rate and BWInfo object by Jiawei Ou · 5 years ago
  17. b168678 Add RTC_ prefix to non-standard format specifier macro "PRIdNS" by Oleh Prypin · 5 years ago
  18. 6982f60 Remove empty OWNERS file. by Mirko Bonadei · 5 years ago
  19. e9ff992 Fix isac_fix_test perf reporting. by Mirko Bonadei · 5 years ago
  20. bb80c13 Guard against clang-format wrong behavior. by Yves Gerey · 5 years ago
  21. a4d8737 Format almost everything. by Jonas Olsson · 5 years ago
  22. e8fbc5d Refactor WebRtcOpus_PacketHasFec. by Minyue Li · 5 years ago
  23. 225842c Initialize signal processing function pointers statically by Karl Wiberg · 5 years ago
  24. 7eb0a5e AudioDecoderOpus: Add support for 16 kHz output sample rate by Karl Wiberg · 5 years ago
  25. a1d1a1e WebRTC Opus C interface: Add support for non-48 kHz decode sample rate by Karl Wiberg · 5 years ago
  26. 126f2b3 AudioEncoderOpus: Add support for 16 kHz input sample rate by Karl Wiberg · 5 years ago
  27. e0eb325 AudioEncoderOpusImpl: Remove unused static methods by Karl Wiberg · 5 years ago
  28. 7e7c5c3 WebRTC Opus C interface: Add support for non-48 kHz encode sample rate by Karl Wiberg · 5 years ago
  29. eb16697 AudioEncoderOpus: Don't mix up sample rate and RTP timestamp rate by Karl Wiberg · 5 years ago
  30. 44c21f4 Encoder side of Multistream Opus. by Alex Loiko · 5 years ago
  31. 40889f3 Removes TimeMicros interface from ThreadProcessingFakeClock. by Sebastian Jansson · 5 years ago
  32. 6a489f2 Fully qualify googletest symbols. by Mirko Bonadei · 5 years ago
  33. e5b9416 Decoder for multistream Opus. by Alex Loiko · 5 years ago
  34. 50b8c39 Generalize the C-language Opus interface. by Alex Loiko · 5 years ago
  35. 94b57c0 Cleanup BUILD.gn files from imports like foo:foo by Artem Titov · 5 years ago
  36. c4dd730 Fix -Wextra-semi warnings. by Mirko Bonadei · 5 years ago
  37. 6543881 2nd reland of https://webrtc-review.googlesource.com/c/src/+/114883 by Alex Loiko · 5 years ago
  38. 22f9925 webrtc: Remove semicolons. by Nico Weber · 5 years ago
  39. 8b3db59 Revert "Reland of https://webrtc-review.googlesource.com/c/src/+/114883" by Alex Loiko · 5 years ago
  40. 5341aac Reland of https://webrtc-review.googlesource.com/c/src/+/114883 by Alex Loiko · 5 years ago
  41. ffd1f93 Revert "Tests for multi-stream Opus." by Mirko Bonadei · 5 years ago
  42. 9c31ac2 Tests for multi-stream Opus. by Alex Loiko · 5 years ago
  43. e45c688 Remove webrtc::ProtoString. by Mirko Bonadei · 5 years ago
  44. 05cf6be [clang-tidy] Apply performance-move-const-arg fixes. by Mirko Bonadei · 5 years ago
  45. c84f661 Stop using Googletest legacy APIs. by Mirko Bonadei · 5 years ago
  46. 7a3e43a Reland of Opus multistream. by Alex Loiko · 5 years ago
  47. 649a4c2 [clang-tidy] Apply performance-inefficient-vector-operation fixes. by Mirko Bonadei · 5 years ago
  48. d970807 Remove rtc_base/scoped_ref_ptr.h. by Mirko Bonadei · 6 years ago
  49. 9dac02d Adding text log on actual opus bitrate. by Minyue Li · 6 years ago
  50. 1fa51d6 Revert "Opus multistream." by Amit Hilbuch · 6 years ago
  51. 83ed89a Opus multistream. by Alex Loiko · 6 years ago
  52. 10542f2 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries by Steve Anton · 6 years ago
  53. 7289906 Delete enum NetEqDecoder. by Niels Möller · 6 years ago
  54. 41f3a43 Remove CodecInst pt.3 by Fredrik Solenberg · 6 years ago
  55. f693bfa Remove CodecInst pt.2 by Fredrik Solenberg · 6 years ago
  56. e10b163 Stop using 'using namespace'. by Mirko Bonadei · 6 years ago
  57. 698d6c4 Change the type of indW32 back to int32_t by Sam Zackrisson · 6 years ago
  58. 622eeda Bump variable sizes in response to fuzzer bug by Jonas Olsson · 6 years ago
  59. 8ac05cc Adds trial to use link capacity estimate in Opus encoder. by Sebastian Jansson · 6 years ago
  60. 254d3db Add missing #include to absl/memory/memory.h from audio_encoder_cng.cc by tzik · 6 years ago
  61. 0070655 Removing ancient and unused test scripts and data files by Henrik Lundin · 6 years ago
  62. 2365936 Hide the AudioEncoderCng class behind a create function by Karl Wiberg · 6 years ago
  63. 988cc08 [Cleanup] Add missing #include. Remove useless ones. by Yves Gerey · 6 years ago
  64. 2edab4c Delete use of STR_CASE_CMP, replaced with absl::EqualsIgnoreCase. by Niels Möller · 6 years ago
  65. 83bd37c Add field trials for configuring Opus encoder packet loss rate. by Jakob Ivarsson · 6 years ago
  66. 88b68ac Create field trial for setting a minimum value for Opus encoder packet loss rate by Jakob Ivarsson · 6 years ago
  67. 366a50c Remove simple stringstream usages. by Jonas Olsson · 6 years ago
  68. d4161a3 Moving LappedTransform, Blocker and AudioRingBuffer. by Alessio Bazzica · 6 years ago
  69. 75caa59 Untangle fft third party lib from dependon WebRTC by Artem Titov · 6 years ago
  70. a15fd0d Add missing include of stdint.h in MIPS code. by Niels Möller · 6 years ago
  71. a12c42a Delete root header file typedef.h. by Niels Möller · 6 years ago
  72. 52b9000 Move g722 to proper third_party directory by Artem Titov · 6 years ago
  73. e095b81 Move g711 to proper third_party directory by Artem Titov · 6 years ago
  74. 8a838fd Move fft to proper third_party directory by Artem Titov · 6 years ago
  75. 682aac5 Enable clang::find_bad_constructs for audio_coding (part 1/2). by Mirko Bonadei · 6 years ago
  76. 918f50c Use absl::make_unique and absl::WrapUnique directly by Karl Wiberg · 6 years ago
  77. 91280e4 Extract third party part of g722 codec into separate target by Artem Titov · 6 years ago
  78. 3ecec17 Extract third party part of g711 codec into separate target by Artem Titov · 6 years ago
  79. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  80. b602123 Replace rtc::Optional with absl::optional in modules/audio_coding by Danil Chapovalov · 6 years ago
  81. 27fe43a Removing warning suppression flags from modules/audio_coding. by Mirko Bonadei · 6 years ago
  82. e058568 iLBC decoding: Ignore a signed overflow by Karl Wiberg · 6 years ago
  83. c7f09ad NetEq fix for repeated audio issue. by Ivo Creusen · 6 years ago
  84. b9fc650 Add min and max allowed bitrate in Opus bitrate tests by Gustaf Ullberg · 6 years ago
  85. ae93f04 Make an energy computation not overflow in iLBC PLC by Sam Zackrisson · 6 years ago
  86. 5f83cf0 Replacing rtc::TimeDelta with webrtc::TimeDelta. by Sebastian Jansson · 6 years ago
  87. 7ba22b8 Break out the part of the iSAC codec that's used for Voice Activity Detection by Karl Wiberg · 6 years ago
  88. 2a35c43 Removing shared_ptr in a unittest in audio coding. by Minyue Li · 6 years ago
  89. 6f3d01c "Fix" signed integer overflow in old code by Karl Wiberg · 6 years ago
  90. 89f645a Add missing header include for filterbanks_neon.c by Jiawei Ou · 6 years ago
  91. 36b096c Ignore overflowing left shift by Karl Wiberg · 6 years ago
  92. bb19fcf Add explicit cast to void to silence -Wcomma warning by Karl Wiberg · 6 years ago
  93. 338f58d iSAC decoder: Don't read past the end of the buffer of encoded bytes by Karl Wiberg · 6 years ago
  94. 7aabd39 Move asm_defines.h to rtc_base/system/ by Karl Wiberg · 6 years ago
  95. d757356 Fixing -Wstrict-prototypes warnings. by Mirko Bonadei · 6 years ago
  96. d6fbf2a Tests: Pass codec ID argument to audio codecs by Karl Wiberg · 6 years ago
  97. 1631dc6 Make isac_fix_test correctly parse --isolated-script-test-perf-output. by Patrik Höglund · 6 years ago
  98. 5d436ac Removed Die mock from MockAudioEncoder by Sebastian Jansson · 6 years ago
  99. 0c15a09 Don't use gtest-parallel when running webrtc_perf_tests. by Edward Lemur · 6 years ago
  100. 96a48ef Reland "Removing forward headers in modules/audio_coding/codecs."" by Mirko Bonadei · 6 years ago