Select "processing" rates based on the input and output sampling rates.
Resample the input streams to those rates, and if necessary to the
output rate.
- Remove deprecated stream format APIs.
- Remove deprecated device sample rate APIs.
- Add a ChannelBuffer class to help manage deinterleaved channels.
- Clean up the splitting filter state.
- Add a unit test which verifies the output against known-working
native format output.
BUG=2894
R=aluebs@webrtc.org, bjornv@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9919004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5959 4adac7df-926f-26a2-2b94-8c16560cd09d
This is mainly to support the native audio format in Chrome. Although
this implementation just moves the float->int conversion under the hood,
we will transition AudioProcessing towards supporting this format
throughout.
- Add a test which verifies we get identical output with the float and
int interfaces.
- The float and int wrappers are tasked with conversion to the
AudioBuffer format. A new shared Process/Analyze method does most of
the work.
- Add a new field to the debug.proto to hold deinterleaved data.
- Add helpers to audio_utils.cc, and start using numeric_limits.
- Note that there was no performance difference between numeric_limits
and a literal value when measured on Linux using gcc or clang.
BUG=2894
R=aluebs@webrtc.org, bjornv@webrtc.org, henrikg@webrtc.org, tommi@webrtc.org, turaj@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9179004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5641 4adac7df-926f-26a2-2b94-8c16560cd09d
