chromium/third_party/webrtc/common_audio/resampler/resampler.cc

/*
 *  Copyright (c) 2011 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.
 */

/*
 * A wrapper for resampling a numerous amount of sampling combinations.
 */

#include "common_audio/resampler/include/resampler.h"

#include <stdint.h>
#include <stdlib.h>
#include <string.h>

#include "common_audio/signal_processing/include/signal_processing_library.h"
#include "rtc_base/logging.h"

namespace webrtc {

Resampler::Resampler()
    : … { … }

Resampler::Resampler(int inFreq, int outFreq, size_t num_channels)
    : … { … }

Resampler::~Resampler() { … }

int Resampler::ResetIfNeeded(int inFreq, int outFreq, size_t num_channels) { … }

int Resampler::Reset(int inFreq, int outFreq, size_t num_channels) { … }

int Resampler::ComputeResamplerMode(int in_freq_hz,
                                    int out_freq_hz,
                                    ResamplerMode* mode) { … }

// Synchronous resampling, all output samples are written to samplesOut
int Resampler::Push(const int16_t* samplesIn,
                    size_t lengthIn,
                    int16_t* samplesOut,
                    size_t maxLen,
                    size_t& outLen) { … }

}  // namespace webrtc