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webrtc/modules/audio_coding/codecs
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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
..
cng Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
g711 Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
g722 Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
ilbc Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
isac Untangle fft third party lib from dependon WebRTC 2018-07-26 13:44:30 +00:00
opus Moving LappedTransform, Blocker and AudioRingBuffer. 2018-08-31 15:27:50 +00:00
pcm16b Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
red Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
tools Delete root header file typedef.h. 2018-07-25 14:59:26 +00:00
audio_decoder.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_encoder.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_format_conversion.cc Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
audio_format_conversion.h Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
builtin_audio_decoder_factory_unittest.cc Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
builtin_audio_encoder_factory_unittest.cc Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
legacy_encoded_audio_frame.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
legacy_encoded_audio_frame.h Replace rtc::Optional with absl::optional in modules/audio_coding 2018-06-19 12:46:20 +00:00
legacy_encoded_audio_frame_unittest.cc Reformat the WebRTC code base 2018-06-19 14:00:39 +00:00
OWNERS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00