webrtc/modules/audio_coding
Jesús de Vicente Peña 01cac31d58 Fixes for the neteq_test clock.
The problem occurs when more than one call is made to the method RunToNextGetAudio. Except for the first call to that method, the clock was not properly updated on the first iteration of the inner loop in RunToNextGetAudio.

Pair: lionelk@webrtc.org

Bug: webrtc:14735
Change-Id: If6fb5c2c700b0f715f626fedf95672a56b04ab12
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285942
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38843}
2022-12-08 10:13:00 +00:00
..
acm2 Reland "[ACM] iSAC audio codec removed" 2022-11-17 12:52:35 +00:00
audio_network_adaptor Remove default enabled field trial WebRTC-SendSideBwe-WithOverhead 2022-11-30 20:19:36 +00:00
codecs Remove default enabled field trial WebRTC-SendSideBwe-WithOverhead 2022-11-30 20:19:36 +00:00
g3doc Update links to point at main branch 2021-07-22 16:41:26 +00:00
include Implement RTCInboundRTPStreamStats.JitterBufferMinimumDelay 2022-07-20 09:14:03 +00:00
neteq Fixes for the neteq_test clock. 2022-12-08 10:13:00 +00:00
test Reland "[ACM] iSAC audio codec removed" 2022-11-17 12:52:35 +00:00
audio_coding.gni Reland "[ACM] iSAC audio codec removed" 2022-11-17 12:52:35 +00:00
BUILD.gn Reland "[ACM] iSAC audio codec removed" 2022-11-17 12:52:35 +00:00
DEPS
OWNERS Add jakobi to modules/audio_coding OWNERS 2021-06-18 11:52:58 +00:00