Resampler modifications in preparation for arbitrary audioproc rates.
- Templatize PushResampler to support int16 and float.
- Add a helper method to PushSincResampler to compute the algorithmic
delay.
This is a prerequisite of:
http://review.webrtc.org/9919004/
BUG=2894
R=turaj@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/12169004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5943 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/voice_engine/utility.h b/webrtc/voice_engine/utility.h
index f6fa35b..127bdba 100644
--- a/webrtc/voice_engine/utility.h
+++ b/webrtc/voice_engine/utility.h
@@ -15,12 +15,12 @@
#ifndef WEBRTC_VOICE_ENGINE_UTILITY_H_
#define WEBRTC_VOICE_ENGINE_UTILITY_H_
+#include "webrtc/common_audio/resampler/include/push_resampler.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class AudioFrame;
-class PushResampler;
namespace voe {
@@ -30,7 +30,7 @@
//
// On failure, returns -1 and copies |src_frame| to |dst_frame|.
void RemixAndResample(const AudioFrame& src_frame,
- PushResampler* resampler,
+ PushResampler<int16_t>* resampler,
AudioFrame* dst_frame);
// Downmix and downsample the audio in |src_data| to |dst_af| as necessary,
@@ -44,7 +44,7 @@
int codec_num_channels,
int codec_rate_hz,
int16_t* mono_buffer,
- PushResampler* resampler,
+ PushResampler<int16_t>* resampler,
AudioFrame* dst_af);
void MixWithSat(int16_t target[],