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webrtc/modules/audio_coding/codecs/opus
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Niels Möller a12c42a6b2 Delete root header file typedef.h. 
Usage replaced with stdint.h, rtc_base/system/arch.h and
rtc_base/system/unused.h, as appropriate.

Bug: webrtc:6854
Change-Id: I97225465d14b969903d92979e2df3c3c05d35f18
Reviewed-on: https://webrtc-review.googlesource.com/90249
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24100}
2018-07-25 14:59:26 +00:00
..
audio_decoder_opus.cc Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
audio_decoder_opus.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_encoder_opus.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
audio_encoder_opus.h Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
audio_encoder_opus_unittest.cc Use absl::make_unique and absl::WrapUnique directly 2018-07-05 10:59:49 +00:00
opus_bandwidth_unittest.cc Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
opus_complexity_unittest.cc Optional: Use nullopt and implicit construction in /modules/audio_coding 2017-11-17 11:58:37 +00:00
opus_fec_test.cc Enable clang::find_bad_constructs for audio_coding (part 1/2). 2018-07-20 13:07:47 +00:00
opus_inst.h Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
opus_interface.c Implement Opus bandwidth adjustment behind a FieldTrial 2017-11-20 20:04:19 +00:00
opus_interface.h Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
opus_speed_test.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
opus_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00