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65881de6c81cf3f33b0f2f5f8f4a19716ee93fdc
platform-external-webrtc/webrtc/modules/audio_coding
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henrik.lundin 65881de6c8 NetEq: Limit payload size for replacement audio input
With this fix, the size of the fake encoded payload is limited to 120
ms at 48000 samples/second.

BUG=webrtc:7467

Review-Url: https://codereview.webrtc.org/2838353002
Cr-Commit-Position: refs/heads/master@{#17891}
2017-04-26 15:23:35 +00:00
..
acm2
Reland of Change NetEq::InsertPacket to take an RTPHeader (patchset #1 id:1 of https://codereview.webrtc.org/2812933002/ )
2017-04-24 13:56:57 +00:00
audio_network_adaptor
Replace Clock with timeutils in AudioEncoder.
2017-04-18 07:11:48 +00:00
codecs
Replace Clock with timeutils in AudioEncoder.
2017-04-18 07:11:48 +00:00
include
Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType"
2017-03-27 14:15:49 +00:00
neteq
NetEq: Limit payload size for replacement audio input
2017-04-26 15:23:35 +00:00
test
Delete unneeded includes of deprecated system_wrappers include files.
2017-03-30 07:31:15 +00:00
audio_coding.gni
Adding build switch for Opus that supports 120ms ptime.
2017-02-02 01:31:11 +00:00
BUILD.gn
Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ )
2017-04-26 10:38:35 +00:00
DEPS
Moved RtcEventLog files from call/ to logging/
2016-10-04 01:31:32 +00:00
OWNERS
Add ossu@ to OWNERS of audio/ and modules/audio_coding/
2016-12-15 15:52:14 +00:00
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