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webrtc/modules/audio_coding/codecs/isac
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Patrik Höglund 7b201012bc Flip histograms to true by default, fix unit in isac_fix_test. 
Requires downstream changes for all WebRTC perf tests, and
a corresponding recipe change so isac_fix_test starts using the new
flow.

Bug: chromium:1029452
Change-Id: I8918fca9bef003d365037c1c6bf7c55747dfed99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170633
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30906}
2020-03-26 19:36:44 +00:00
..
fix Flip histograms to true by default, fix unit in isac_fix_test. 2020-03-26 19:36:44 +00:00
main iSAC API wrapper unit test fix 2020-02-27 14:27:23 +00:00
audio_decoder_isac_t.h Delete AudioDecoder method IncomingPacket 2019-09-24 08:30:24 +00:00
audio_decoder_isac_t_impl.h Delete AudioDecoder method IncomingPacket 2019-09-24 08:30:24 +00:00
audio_encoder_isac_t.h Implement AudioEncoder::GetFrameLengthRange() for all audio encoders. 2020-03-25 22:19:21 +00:00
audio_encoder_isac_t_impl.h Implement AudioEncoder::GetFrameLengthRange() for all audio encoders. 2020-03-25 22:19:21 +00:00
bandwidth_info.h Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
empty.cc Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
isac_webrtc_api_test.cc iSAC API wrapper unit test fix 2020-02-27 14:27:23 +00:00