50b2efef6e
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
42 lines
1.3 KiB
C++
42 lines
1.3 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "webrtc/common_audio/include/audio_util.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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void Deinterleave(const int16_t* interleaved, int samples_per_channel,
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int num_channels, int16_t** deinterleaved) {
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for (int i = 0; i < num_channels; i++) {
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int16_t* channel = deinterleaved[i];
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int interleaved_idx = i;
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for (int j = 0; j < samples_per_channel; j++) {
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channel[j] = interleaved[interleaved_idx];
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interleaved_idx += num_channels;
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}
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}
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}
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void Interleave(const int16_t* const* deinterleaved, int samples_per_channel,
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int num_channels, int16_t* interleaved) {
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for (int i = 0; i < num_channels; ++i) {
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const int16_t* channel = deinterleaved[i];
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int interleaved_idx = i;
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for (int j = 0; j < samples_per_channel; j++) {
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interleaved[interleaved_idx] = channel[j];
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interleaved_idx += num_channels;
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}
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}
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}
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} // namespace webrtc
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