Repo created

TMessagesProj/jni/voip/webrtc/modules/audio_processing/audio_buffer.h

@@ -0,0 +1,172 @@
+/*
+ *  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 "common_audio/channel_buffer.h"
+#include "modules/audio_processing/include/audio_processing.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;
+  static const int kMaxSampleRate = 384000;
+  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);
+
+  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_