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webrtc/modules/audio_coding/codecs/opus
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Sebastian Jansson 5f83cf0c6d Replacing rtc::TimeDelta with webrtc::TimeDelta. 
This removes the redundant type and replaces all usages. A slight change
in behavior is that we no longer get nanosecond resolution. This should
not matter since no current code requires nanosecond resolution.

Bug: webrtc:9155
Change-Id: I04334e08c686d95731621a6c8a7e40400d0ae3b2
Reviewed-on: https://webrtc-review.googlesource.com/71163
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23174}
2018-05-08 13:22:53 +00:00
..
audio_decoder_opus.cc Optional: Use nullopt and implicit construction in /modules/audio_coding 2017-11-17 11:58:37 +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 Move some numeric utility code from rtc_base/ to rtc_base/numerics/ 2017-11-22 11:21:47 +00:00
audio_encoder_opus.h Implement Opus bandwidth adjustment behind a FieldTrial 2017-11-20 20:04:19 +00:00
audio_encoder_opus_unittest.cc Replacing rtc::TimeDelta with webrtc::TimeDelta. 2018-05-08 13:22:53 +00:00
opus_bandwidth_unittest.cc Implement Opus bandwidth adjustment behind a FieldTrial 2017-11-20 20:04:19 +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 Stop using std::tr1 2017-10-23 22:11:58 +00:00
opus_inst.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
opus_interface.c Implement Opus bandwidth adjustment behind a FieldTrial 2017-11-20 20:04:19 +00:00
opus_interface.h Implement Opus bandwidth adjustment behind a FieldTrial 2017-11-20 20:04:19 +00:00
opus_speed_test.cc Implement Opus bandwidth adjustment behind a FieldTrial 2017-11-20 20:04:19 +00:00
opus_unittest.cc Move some numeric utility code from rtc_base/ to rtc_base/numerics/ 2017-11-22 11:21:47 +00:00