chromium/third_party/webrtc/common_audio/audio_converter.cc

/*
 *  Copyright (c) 2014 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 "common_audio/audio_converter.h"

#include <cstring>
#include <memory>
#include <utility>
#include <vector>

#include "common_audio/channel_buffer.h"
#include "common_audio/resampler/push_sinc_resampler.h"
#include "rtc_base/checks.h"
#include "rtc_base/numerics/safe_conversions.h"

namespace webrtc {

class CopyConverter : public AudioConverter { … };

class UpmixConverter : public AudioConverter { … };

class DownmixConverter : public AudioConverter { … };

class ResampleConverter : public AudioConverter { … };

// Apply a vector of converters in serial, in the order given. At least two
// converters must be provided.
class CompositionConverter : public AudioConverter { … };

std::unique_ptr<AudioConverter> AudioConverter::Create(size_t src_channels,
                                                       size_t src_frames,
                                                       size_t dst_channels,
                                                       size_t dst_frames) { … }

// For CompositionConverter.
AudioConverter::AudioConverter()
    : … { … }

AudioConverter::AudioConverter(size_t src_channels,
                               size_t src_frames,
                               size_t dst_channels,
                               size_t dst_frames)
    : … { … }

void AudioConverter::CheckSizes(size_t src_size, size_t dst_capacity) const { … }

}  // namespace webrtc