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Add a wrapper around PushSincResampler and the old Resampler.

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The old resampler is used whenever it supports the requested rates. Otherwise
the sinc resampler is enabled.

Integrated with output_mixer in order to test the change through
output_mixer_unittest. The sinc resampler will not yet be used, since we don't
feed VoE with any rates that trigger it.

BUG=webrtc:1395
R=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1355004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3915 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
andrew@webrtc.org
2013-04-29 17:27:29 +00:00
parent 5b7120c81b
commit 50b2efef6e
12 changed files with 513 additions and 77 deletions
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41
webrtc/common_audio/audio_util.cc Normal file
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/*
* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "webrtc/common_audio/include/audio_util.h"
#include "webrtc/typedefs.h"
namespace webrtc {
void Deinterleave(const int16_t* interleaved, int samples_per_channel,
int num_channels, int16_t** deinterleaved) {
for (int i = 0; i < num_channels; i++) {
int16_t* channel = deinterleaved[i];
int interleaved_idx = i;
for (int j = 0; j < samples_per_channel; j++) {
channel[j] = interleaved[interleaved_idx];
interleaved_idx += num_channels;
}
}
}
void Interleave(const int16_t* const* deinterleaved, int samples_per_channel,
int num_channels, int16_t* interleaved) {
for (int i = 0; i < num_channels; ++i) {
const int16_t* channel = deinterleaved[i];
int interleaved_idx = i;
for (int j = 0; j < samples_per_channel; j++) {
interleaved[interleaved_idx] = channel[j];
interleaved_idx += num_channels;
}
}
}
} // namespace webrtc