tor-browser

audio_buffer.h (6500B)

---

/*
*  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 = 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_