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webrtc_m130/webrtc/modules/audio_coding
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ossu e280cdeb74 Voe::Channel: Turned GetPlayoutFrequency into GetRtpTimestampRateHz.
This gets rid of a bit of codec-specific code in VoE.

BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2355483003
Cr-Commit-Position: refs/heads/master@{#14614}
2016-10-12 18:04:16 +00:00
..
acm2
Voe::Channel: Turned GetPlayoutFrequency into GetRtpTimestampRateHz.
2016-10-12 18:04:16 +00:00
audio_network_adaptor
Adding audio network adaptor to AudioEncoderOpus.
2016-10-06 14:13:59 +00:00
codecs
Hooking up audio network adaptor to VoE.
2016-10-12 12:01:01 +00:00
include
Voe::Channel: Turned GetPlayoutFrequency into GetRtpTimestampRateHz.
2016-10-12 18:04:16 +00:00
neteq
Fix bug in DTMF generation where events with level > 36 would be ignored.
2016-10-11 11:35:40 +00:00
test
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
audio_coding_tests.gypi
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
audio_coding.gni
GN conversion of audio_decoder_unittests
2016-08-01 14:49:50 +00:00
audio_coding.gypi
Moved RtcEventLog files from call/ to logging/
2016-10-04 01:31:32 +00:00
BUILD.gn
Cleaning build file for audio network adaptor.
2016-10-07 14:59:36 +00:00
DEPS
Moved RtcEventLog files from call/ to logging/
2016-10-04 01:31:32 +00:00
OWNERS
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
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