Support arbitrary input/output rates and downmixing in AudioProcessing.
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
diff --git a/webrtc/modules/audio_processing/debug.proto b/webrtc/modules/audio_processing/debug.proto
index 7d4e9d1..dce2f79 100644
--- a/webrtc/modules/audio_processing/debug.proto
+++ b/webrtc/modules/audio_processing/debug.proto
@@ -4,11 +4,12 @@
message Init {
optional int32 sample_rate = 1;
- optional int32 device_sample_rate = 2;
+ optional int32 device_sample_rate = 2 [deprecated=true];
optional int32 num_input_channels = 3;
optional int32 num_output_channels = 4;
optional int32 num_reverse_channels = 5;
optional int32 reverse_sample_rate = 6;
+ optional int32 output_sample_rate = 7;
}
// May contain interleaved or deinterleaved data, but don't store both formats.