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gerrit-public.fairphone.software / platform / external / webrtc / f11a0287d6217c629fb381dba6f668db9f560efd / audio / audio_send_stream_tests.cc
  1. 1b4e4bf Migrate several call tests from legacy RtpHeaderParser to RtpPacket parsing. by Danil Chapovalov · 4 years, 10 months ago
  2. d8d3248 Reland "Delete test/constants.h" by Elad Alon · 6 years ago
  3. 4f36b7a Revert "Delete test/constants.h" by Oleh Prypin · 6 years ago
  4. 389b167 Delete test/constants.h by Elad Alon · 6 years ago
  5. 914351d Reland "Always offer transport sequence number header extension for audio"" by Per Kjellander · 6 years ago
  6. 397c06f Revert "Always offer transport sequence number header extension for audio" by Ying Wang · 6 years ago
  7. fd965c0 Always offer transport sequence number header extension for audio by Per Kjellander · 6 years ago
  8. 665174f Reformat the WebRTC code base by Yves Gerey · 6 years ago
  9. 675513b Stop using LOG macros in favor of RTC_ prefixed macros. by Mirko Bonadei · 7 years ago
  10. 18f5427 Remove voe_auto_test and add new tests to cover the missing cases. by solenberg · 7 years ago