bjornv@webrtc.org 7bb2586c55 audio_processing: Correct sample rate in aec_debug_dump
When writing to wav files in the low level flag aec_debug_dump incorrect sample rates were used for recordings using rates from 32 kHz and above. This since internally inside the AEC we process the data using 16 kHz. Any upper band is processed and combined later on.

This CL adds the correct sample rate to the recording.

BUG=3359
TESTED=locally on 44.1 kHz recordings on Linux
R=aluebs@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23649004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7182 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-09-15 13:23:07 +00:00
2014-09-10 09:29:12 +00:00
2014-09-11 21:12:59 +00:00
2014-06-17 08:54:03 +00:00
2014-08-25 14:41:41 +00:00
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
BSD-3-Clause 446 MiB
Languages
C++ 90.3%
Java 2.9%
C 2.2%
Objective-C++ 2%
Python 1.3%
Other 1%