chromium/third_party/webrtc/modules/audio_processing/audio_buffer.cc

/*
 *  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_processing/audio_buffer.h"

#include <string.h>

#include <cstdint>

#include "common_audio/channel_buffer.h"
#include "common_audio/resampler/push_sinc_resampler.h"
#include "modules/audio_processing/splitting_filter.h"
#include "rtc_base/checks.h"

namespace webrtc {
namespace {

constexpr size_t kSamplesPer32kHzChannel = …;
constexpr size_t kSamplesPer48kHzChannel = …;

size_t NumBandsFromFramesPerChannel(size_t num_frames) { … }

}  // namespace

AudioBuffer::AudioBuffer(size_t input_rate,
                         size_t input_num_channels,
                         size_t buffer_rate,
                         size_t buffer_num_channels,
                         size_t output_rate,
                         size_t output_num_channels)
    : … { … }

AudioBuffer::~AudioBuffer() { … }

void AudioBuffer::set_downmixing_to_specific_channel(size_t channel) { … }

void AudioBuffer::set_downmixing_by_averaging() { … }

void AudioBuffer::CopyFrom(const float* const* stacked_data,
                           const StreamConfig& stream_config) { … }

void AudioBuffer::CopyTo(const StreamConfig& stream_config,
                         float* const* stacked_data) { … }

void AudioBuffer::CopyTo(AudioBuffer* buffer) const { … }

void AudioBuffer::RestoreNumChannels() { … }

void AudioBuffer::set_num_channels(size_t num_channels) { … }

// The resampler is only for supporting 48kHz to 16kHz in the reverse stream.
void AudioBuffer::CopyFrom(const int16_t* const interleaved_data,
                           const StreamConfig& stream_config) { … }

void AudioBuffer::CopyTo(const StreamConfig& stream_config,
                         int16_t* const interleaved_data) { … }

void AudioBuffer::SplitIntoFrequencyBands() { … }

void AudioBuffer::MergeFrequencyBands() { … }

void AudioBuffer::ExportSplitChannelData(
    size_t channel,
    int16_t* const* split_band_data) const { … }

void AudioBuffer::ImportSplitChannelData(
    size_t channel,
    const int16_t* const* split_band_data) { … }

}  // namespace webrtc