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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 "modules/audio_coding/acm2/acm_resampler.h"
#include <string.h>
#include "rtc_base/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,
size_t num_audio_channels,
size_t out_capacity_samples,
int16_t* out_audio) {
size_t in_length = in_freq_hz * num_audio_channels / 100;
if (in_freq_hz == out_freq_hz) {
if (out_capacity_samples < in_length) {
RTC_DCHECK_NOTREACHED();
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) {
RTC_LOG(LS_ERROR) << "InitializeIfNeeded(" << in_freq_hz << ", "
<< out_freq_hz << ", " << num_audio_channels
<< ") failed.";
return -1;
}
int out_length =
resampler_.Resample(in_audio, in_length, out_audio, out_capacity_samples);
if (out_length == -1) {
RTC_LOG(LS_ERROR) << "Resample(" << in_audio << ", " << in_length << ", "
<< out_audio << ", " << out_capacity_samples
<< ") failed.";
return -1;
}
return static_cast<int>(out_length / num_audio_channels);
}
} // namespace acm2
} // namespace webrtc
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