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/*
* Copyright (c) 2012 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/modules/audio_coding/main/acm2/acm_resampler.h"
#include <assert.h>
#include <string.h>
#include "webrtc/common_audio/resampler/include/resampler.h"
#include "webrtc/system_wrappers/include/logging.h"
namespace webrtc {
namespace acm2 {
ACMResampler::ACMResampler() {
}
ACMResampler::~ACMResampler() {
}
int ACMResampler::Resample10Msec(const int16_t* in_audio,
int in_freq_hz,
int out_freq_hz,
int num_audio_channels,
size_t out_capacity_samples,
int16_t* out_audio) {
size_t in_length = static_cast<size_t>(in_freq_hz * num_audio_channels / 100);
int out_length = out_freq_hz * num_audio_channels / 100;
if (in_freq_hz == out_freq_hz) {
if (out_capacity_samples < in_length) {
assert(false);
return -1;
}
memcpy(out_audio, in_audio, in_length * sizeof(int16_t));
return static_cast<int>(in_length / num_audio_channels);
}
if (resampler_.InitializeIfNeeded(in_freq_hz, out_freq_hz,
num_audio_channels) != 0) {
LOG_FERR3(LS_ERROR, InitializeIfNeeded, in_freq_hz, out_freq_hz,
num_audio_channels);
return -1;
}
out_length =
resampler_.Resample(in_audio, in_length, out_audio, out_capacity_samples);
if (out_length == -1) {
LOG_FERR4(LS_ERROR,
Resample,
in_audio,
in_length,
out_audio,
out_capacity_samples);
return -1;
}
return out_length / num_audio_channels;
}
} // namespace acm2
} // namespace webrtc
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