webrtc/modules/audio_coding/codecs/opus
Alessio Bazzica d4161a3c9d Moving LappedTransform, Blocker and AudioRingBuffer.
LappedTransform is only used in BandwidthAdaptationTest and therefore it
should not be anymore a visible target under common_audio.
This CL moves LappedTransform and other two classes it depends on (and which
are not used elsewhere) to modules/audio_coding/codecs/opus/test.

Bug: webrtc:9577, webrtc:5298
Change-Id: I1aa8052c2df2b2b150c279c0c9b1001474aed47a
Reviewed-on: https://webrtc-review.googlesource.com/96440
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24509}
2018-08-31 15:27:50 +00:00
..
test Moving LappedTransform, Blocker and AudioRingBuffer. 2018-08-31 15:27:50 +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 Moving LappedTransform, Blocker and AudioRingBuffer. 2018-08-31 15:27:50 +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