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/*
* 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.
*/
#ifndef MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_
#define MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_
#include <stddef.h>
#include <stdint.h>
#include <memory>
#include <vector>
#include "api/audio/audio_processing.h"
#include "api/audio/audio_view.h"
#include "common_audio/channel_buffer.h"
#include "common_audio/include/audio_util.h"
#include "rtc_base/gtest_prod_util.h"
namespace webrtc {
class PushSincResampler;
class SplittingFilter;
enum Band { kBand0To8kHz = 0, kBand8To16kHz = 1, kBand16To24kHz = 2 };
// Stores any audio data in a way that allows the audio processing module to
// operate on it in a controlled manner.
class AudioBuffer {
public:
static const int kSplitBandSize = 160;
// TODO(tommi): Remove this (`AudioBuffer::kMaxSampleRate`) constant.
static const int kMaxSampleRate = webrtc::kMaxSampleRateHz;
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);
virtual ~AudioBuffer();
AudioBuffer(const AudioBuffer&) = delete;
AudioBuffer& operator=(const AudioBuffer&) = delete;
// Specify that downmixing should be done by selecting a single channel.
void set_downmixing_to_specific_channel(size_t channel);
// Specify that downmixing should be done by averaging all channels,.
void set_downmixing_by_averaging();
// Set the number of channels in the buffer. The specified number of channels
// cannot be larger than the specified buffer_num_channels. The number is also
// reset at each call to CopyFrom or InterleaveFrom.
void set_num_channels(size_t num_channels);
// Returns a DeinterleavedView<> over the channel data.
DeinterleavedView<float> view() {
return DeinterleavedView<float>(
num_channels_ && buffer_num_frames_ ? channels() : nullptr,
buffer_num_frames_, num_channels_);
}
size_t num_channels() const { return num_channels_; }
size_t num_frames() const { return buffer_num_frames_; }
size_t num_frames_per_band() const { return num_split_frames_; }
size_t num_bands() const { return num_bands_; }
// Returns pointer arrays to the full-band channels.
// Usage:
// channels()[channel][sample].
// Where:
// 0 <= channel < `buffer_num_channels_`
// 0 <= sample < `buffer_num_frames_`
float* const* channels() { return data_->channels(); }
const float* const* channels_const() const { return data_->channels(); }
// Returns pointer arrays to the bands for a specific channel.
// Usage:
// split_bands(channel)[band][sample].
// Where:
// 0 <= channel < `buffer_num_channels_`
// 0 <= band < `num_bands_`
// 0 <= sample < `num_split_frames_`
const float* const* split_bands_const(size_t channel) const {
return split_data_.get() ? split_data_->bands(channel)
: data_->bands(channel);
}
float* const* split_bands(size_t channel) {
return split_data_.get() ? split_data_->bands(channel)
: data_->bands(channel);
}
// Returns a pointer array to the channels for a specific band.
// Usage:
// split_channels(band)[channel][sample].
// Where:
// 0 <= band < `num_bands_`
// 0 <= channel < `buffer_num_channels_`
// 0 <= sample < `num_split_frames_`
const float* const* split_channels_const(Band band) const {
if (split_data_.get()) {
return split_data_->channels(band);
} else {
return band == kBand0To8kHz ? data_->channels() : nullptr;
}
}
// Copies data into the buffer.
void CopyFrom(const int16_t* const interleaved_data,
const StreamConfig& stream_config);
void CopyFrom(const float* const* stacked_data,
const StreamConfig& stream_config);
// Copies data from the buffer.
void CopyTo(const StreamConfig& stream_config,
int16_t* const interleaved_data);
void CopyTo(const StreamConfig& stream_config, float* const* stacked_data);
void CopyTo(AudioBuffer* buffer) const;
// Splits the buffer data into frequency bands.
void SplitIntoFrequencyBands();
// Recombines the frequency bands into a full-band signal.
void MergeFrequencyBands();
// Copies the split bands data into the integer two-dimensional array.
void ExportSplitChannelData(size_t channel,
int16_t* const* split_band_data) const;
// Copies the data in the integer two-dimensional array into the split_bands
// data.
void ImportSplitChannelData(size_t channel,
const int16_t* const* split_band_data);
static const size_t kMaxSplitFrameLength = 160;
static const size_t kMaxNumBands = 3;
// Deprecated methods, will be removed soon.
float* const* channels_f() { return channels(); }
const float* const* channels_const_f() const { return channels_const(); }
const float* const* split_bands_const_f(size_t channel) const {
return split_bands_const(channel);
}
float* const* split_bands_f(size_t channel) { return split_bands(channel); }
const float* const* split_channels_const_f(Band band) const {
return split_channels_const(band);
}
private:
FRIEND_TEST_ALL_PREFIXES(AudioBufferTest,
SetNumChannelsSetsChannelBuffersNumChannels);
void RestoreNumChannels();
const size_t input_num_frames_;
const size_t input_num_channels_;
const size_t buffer_num_frames_;
const size_t buffer_num_channels_;
const size_t output_num_frames_;
const size_t output_num_channels_;
size_t num_channels_;
size_t num_bands_;
size_t num_split_frames_;
std::unique_ptr<ChannelBuffer<float>> data_;
std::unique_ptr<ChannelBuffer<float>> split_data_;
std::unique_ptr<SplittingFilter> splitting_filter_;
std::vector<std::unique_ptr<PushSincResampler>> input_resamplers_;
std::vector<std::unique_ptr<PushSincResampler>> output_resamplers_;
bool downmix_by_averaging_ = true;
size_t channel_for_downmixing_ = 0;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_
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