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webrtc_m130/modules/audio_coding
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Pablo Barrera González 7541141b99 Fix assert check in neteq_quality_test 
While reading inpùt files until their end, the assert should be
ASSERT_TRUE.

Change-Id: Ib60b68173b58b77d9789c544c7cb647a752a24d1
Bug: webrtc:10690
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140280
Commit-Queue: Pablo Barrera González <barrerap@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28202}
2019-06-10 08:01:26 +00:00
..
acm2
Audio coding: Don't choke when RTP timestamp rate > sample rate
2019-05-21 03:10:49 +00:00
audio_network_adaptor
Stop DCHECK which occurs in ANA BitrateController when overhead is zero.
2019-04-27 00:20:37 +00:00
codecs
AudioDecoderOpus: Add support for 16 kHz output sample rate
2019-05-29 12:42:38 +00:00
include
Expose new audio stats on the API
2019-05-03 10:10:15 +00:00
neteq
Fix assert check in neteq_quality_test
2019-06-10 08:01:26 +00:00
test
WebRTC Opus C interface: Add support for non-48 kHz decode sample rate
2019-05-29 10:33:03 +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
Encoder side of Multistream Opus.
2019-04-25 15:07:38 +00:00
DEPS
Fixing WebRTC after moving from src/webrtc to src/
2017-09-15 05:02:56 +00:00
OWNERS
Make ivoc owner of audio_coding.
2018-10-15 15:08:28 +00:00