Moving src/webrtc into src/.

In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}

modules/audio_processing/audio_buffer.h

@@ -0,0 +1,166 @@
+/*
+ *  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 WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_
+
+#include <memory>
+#include <vector>
+
+#include "webrtc/common_audio/channel_buffer.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/modules/audio_processing/splitting_filter.h"
+#include "webrtc/modules/include/module_common_types.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+class PushSincResampler;
+class IFChannelBuffer;
+
+enum Band {
+  kBand0To8kHz = 0,
+  kBand8To16kHz = 1,
+  kBand16To24kHz = 2
+};
+
+class AudioBuffer {
+ public:
+  // TODO(ajm): Switch to take ChannelLayouts.
+  AudioBuffer(size_t input_num_frames,
+              size_t num_input_channels,
+              size_t process_num_frames,
+              size_t num_process_channels,
+              size_t output_num_frames);
+  virtual ~AudioBuffer();
+
+  size_t num_channels() const;
+  void set_num_channels(size_t num_channels);
+  size_t num_frames() const;
+  size_t num_frames_per_band() const;
+  size_t num_keyboard_frames() const;
+  size_t num_bands() const;
+
+  // Returns a pointer array to the full-band channels.
+  // Usage:
+  // channels()[channel][sample].
+  // Where:
+  // 0 <= channel < |num_proc_channels_|
+  // 0 <= sample < |proc_num_frames_|
+  int16_t* const* channels();
+  const int16_t* const* channels_const() const;
+  float* const* channels_f();
+  const float* const* channels_const_f() const;
+
+  // Returns a pointer array to the bands for a specific channel.
+  // Usage:
+  // split_bands(channel)[band][sample].
+  // Where:
+  // 0 <= channel < |num_proc_channels_|
+  // 0 <= band < |num_bands_|
+  // 0 <= sample < |num_split_frames_|
+  int16_t* const* split_bands(size_t channel);
+  const int16_t* const* split_bands_const(size_t channel) const;
+  float* const* split_bands_f(size_t channel);
+  const float* const* split_bands_const_f(size_t channel) const;
+
+  // Returns a pointer array to the channels for a specific band.
+  // Usage:
+  // split_channels(band)[channel][sample].
+  // Where:
+  // 0 <= band < |num_bands_|
+  // 0 <= channel < |num_proc_channels_|
+  // 0 <= sample < |num_split_frames_|
+  int16_t* const* split_channels(Band band);
+  const int16_t* const* split_channels_const(Band band) const;
+  float* const* split_channels_f(Band band);
+  const float* const* split_channels_const_f(Band band) const;
+
+  // Returns a pointer to the ChannelBuffer that encapsulates the full-band
+  // data.
+  ChannelBuffer<int16_t>* data();
+  const ChannelBuffer<int16_t>* data() const;
+  ChannelBuffer<float>* data_f();
+  const ChannelBuffer<float>* data_f() const;
+
+  // Returns a pointer to the ChannelBuffer that encapsulates the split data.
+  ChannelBuffer<int16_t>* split_data();
+  const ChannelBuffer<int16_t>* split_data() const;
+  ChannelBuffer<float>* split_data_f();
+  const ChannelBuffer<float>* split_data_f() const;
+
+  // Returns a pointer to the low-pass data downmixed to mono. If this data
+  // isn't already available it re-calculates it.
+  const int16_t* mixed_low_pass_data();
+  const int16_t* low_pass_reference(int channel) const;
+
+  const float* keyboard_data() const;
+
+  void set_activity(AudioFrame::VADActivity activity);
+  AudioFrame::VADActivity activity() const;
+
+  // Use for int16 interleaved data.
+  void DeinterleaveFrom(AudioFrame* audioFrame);
+  // If |data_changed| is false, only the non-audio data members will be copied
+  // to |frame|.
+  void InterleaveTo(AudioFrame* frame, bool data_changed) const;
+
+  // Use for float deinterleaved data.
+  void CopyFrom(const float* const* data, const StreamConfig& stream_config);
+  void CopyTo(const StreamConfig& stream_config, float* const* data);
+  void CopyLowPassToReference();
+
+  // Splits the signal into different bands.
+  void SplitIntoFrequencyBands();
+  // Recombine the different bands into one signal.
+  void MergeFrequencyBands();
+
+ private:
+  FRIEND_TEST_ALL_PREFIXES(AudioBufferTest,
+                           SetNumChannelsSetsChannelBuffersNumChannels);
+  // Called from DeinterleaveFrom() and CopyFrom().
+  void InitForNewData();
+
+  // The audio is passed into DeinterleaveFrom() or CopyFrom() with input
+  // format (samples per channel and number of channels).
+  const size_t input_num_frames_;
+  const size_t num_input_channels_;
+  // The audio is stored by DeinterleaveFrom() or CopyFrom() with processing
+  // format.
+  const size_t proc_num_frames_;
+  const size_t num_proc_channels_;
+  // The audio is returned by InterleaveTo() and CopyTo() with output samples
+  // per channels and the current number of channels. This last one can be
+  // changed at any time using set_num_channels().
+  const size_t output_num_frames_;
+  size_t num_channels_;
+
+  size_t num_bands_;
+  size_t num_split_frames_;
+  bool mixed_low_pass_valid_;
+  bool reference_copied_;
+  AudioFrame::VADActivity activity_;
+
+  const float* keyboard_data_;
+  std::unique_ptr<IFChannelBuffer> data_;
+  std::unique_ptr<IFChannelBuffer> split_data_;
+  std::unique_ptr<SplittingFilter> splitting_filter_;
+  std::unique_ptr<ChannelBuffer<int16_t> > mixed_low_pass_channels_;
+  std::unique_ptr<ChannelBuffer<int16_t> > low_pass_reference_channels_;
+  std::unique_ptr<IFChannelBuffer> input_buffer_;
+  std::unique_ptr<IFChannelBuffer> output_buffer_;
+  std::unique_ptr<ChannelBuffer<float> > process_buffer_;
+  std::vector<std::unique_ptr<PushSincResampler>> input_resamplers_;
+  std::vector<std::unique_ptr<PushSincResampler>> output_resamplers_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_