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webrtc/modules/audio_coding
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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
..
acm2 ACM: Properly initialize last_audio_buffer_ array 2018-05-08 11:40:04 +00:00
audio_network_adaptor Replacing rtc::TimeDelta with webrtc::TimeDelta. 2018-05-08 13:22:53 +00:00
codecs Replacing rtc::TimeDelta with webrtc::TimeDelta. 2018-05-08 13:22:53 +00:00
include Remove dependencies on modules:module_api from AudioProcessing. 2018-04-12 22:05:27 +00:00
neteq Plot NetEq stats in RTC event log visualizer. 2018-05-07 17:01:48 +00:00
test Remove incompatiblities with absl::optional in audio_coding 2018-04-17 12:05:13 +00:00
audio_coding.gni Don't select audio codecs depending on GN vars build_with_{chromium|mozilla} 2017-11-01 18:59:27 +00:00
BUILD.gn Break out the part of the iSAC codec that's used for Voice Activity Detection 2018-05-04 08:53:34 +00:00
DEPS Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
OWNERS Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00