common_audio/resampler/sinc_resampler.h - src - Git at Google

 /*
  *  Copyright (c) 2013 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.
  */

 // Modified from the Chromium original here:
 // src/media/base/sinc_resampler.h

 #ifndef COMMON_AUDIO_RESAMPLER_SINC_RESAMPLER_H_
 #define COMMON_AUDIO_RESAMPLER_SINC_RESAMPLER_H_

 #include <stddef.h>

 #include <memory>

 #include "rtc_base/gtest_prod_util.h"
 #include "rtc_base/memory/aligned_malloc.h"
 #include "rtc_base/system/arch.h"

 namespace webrtc {

 // Callback class for providing more data into the resampler.  Expects `frames`
 // of data to be rendered into `destination`; zero padded if not enough frames
 // are available to satisfy the request.
 class SincResamplerCallback {
  public:
   virtual ~SincResamplerCallback() {}
   virtual void Run(size_t frames, float* destination) = 0;
 };

 // SincResampler is a high-quality single-channel sample-rate converter.
 class SincResampler {
  public:
   // The kernel size can be adjusted for quality (higher is better) at the
   // expense of performance.  Must be a multiple of 32.
   // TODO(dalecurtis): Test performance to see if we can jack this up to 64+.
   static const size_t kKernelSize = 32;

   // Default request size.  Affects how often and for how much SincResampler
   // calls back for input.  Must be greater than kKernelSize.
   static const size_t kDefaultRequestSize = 512;

   // The kernel offset count is used for interpolation and is the number of
   // sub-sample kernel shifts.  Can be adjusted for quality (higher is better)
   // at the expense of allocating more memory.
   static const size_t kKernelOffsetCount = 32;
   static const size_t kKernelStorageSize =
       kKernelSize * (kKernelOffsetCount + 1);

   // Constructs a SincResampler with the specified `read_cb`, which is used to
   // acquire audio data for resampling.  `io_sample_rate_ratio` is the ratio
   // of input / output sample rates.  `request_frames` controls the size in
   // frames of the buffer requested by each `read_cb` call.  The value must be
   // greater than kKernelSize.  Specify kDefaultRequestSize if there are no
   // request size constraints.
   SincResampler(double io_sample_rate_ratio,
                 size_t request_frames,
                 SincResamplerCallback* read_cb);
   virtual ~SincResampler();

   SincResampler(const SincResampler&) = delete;
   SincResampler& operator=(const SincResampler&) = delete;

   // Resample `frames` of data from `read_cb_` into `destination`.
   void Resample(size_t frames, float* destination);

   // The maximum size in frames that guarantees Resample() will only make a
   // single call to `read_cb_` for more data.
   size_t ChunkSize() const;

   size_t request_frames() const { return request_frames_; }

   // Flush all buffered data and reset internal indices.  Not thread safe, do
   // not call while Resample() is in progress.
   void Flush();

   // Update `io_sample_rate_ratio_`.  SetRatio() will cause a reconstruction of
   // the kernels used for resampling.  Not thread safe, do not call while
   // Resample() is in progress.
   //
   // TODO(ajm): Use this in PushSincResampler rather than reconstructing
   // SincResampler.  We would also need a way to update `request_frames_`.
   void SetRatio(double io_sample_rate_ratio);

   float* get_kernel_for_testing() { return kernel_storage_.get(); }

  private:
   FRIEND_TEST_ALL_PREFIXES(SincResamplerTest, Convolve);
   FRIEND_TEST_ALL_PREFIXES(SincResamplerTest, ConvolveBenchmark);

   void InitializeKernel();
   void UpdateRegions(bool second_load);

   // Selects runtime specific CPU features like SSE.  Must be called before
   // using SincResampler.
   // TODO(ajm): Currently managed by the class internally. See the note with
   // `convolve_proc_` below.
   void InitializeCPUSpecificFeatures();

   // Compute convolution of `k1` and `k2` over `input_ptr`, resultant sums are
   // linearly interpolated using `kernel_interpolation_factor`.  On x86 and ARM
   // the underlying implementation is chosen at run time.
   static float Convolve_C(const float* input_ptr,
                           const float* k1,
                           const float* k2,
                           double kernel_interpolation_factor);
 #if defined(WEBRTC_ARCH_X86_FAMILY)
   static float Convolve_SSE(const float* input_ptr,
                             const float* k1,
                             const float* k2,
                             double kernel_interpolation_factor);
   static float Convolve_AVX2(const float* input_ptr,
                              const float* k1,
                              const float* k2,
                              double kernel_interpolation_factor);
 #elif defined(WEBRTC_HAS_NEON)
   static float Convolve_NEON(const float* input_ptr,
                              const float* k1,
                              const float* k2,
                              double kernel_interpolation_factor);
 #endif

   // The ratio of input / output sample rates.
   double io_sample_rate_ratio_;

   // An index on the source input buffer with sub-sample precision.  It must be
   // double precision to avoid drift.
   double virtual_source_idx_;

   // The buffer is primed once at the very beginning of processing.
   bool buffer_primed_;

   // Source of data for resampling.
   SincResamplerCallback* read_cb_;

   // The size (in samples) to request from each `read_cb_` execution.
   const size_t request_frames_;

   // The number of source frames processed per pass.
   size_t block_size_;

   // The size (in samples) of the internal buffer used by the resampler.
   const size_t input_buffer_size_;

   // Contains kKernelOffsetCount kernels back-to-back, each of size kKernelSize.
   // The kernel offsets are sub-sample shifts of a windowed sinc shifted from
   // 0.0 to 1.0 sample.
   std::unique_ptr<float[], AlignedFreeDeleter> kernel_storage_;
   std::unique_ptr<float[], AlignedFreeDeleter> kernel_pre_sinc_storage_;
   std::unique_ptr<float[], AlignedFreeDeleter> kernel_window_storage_;

   // Data from the source is copied into this buffer for each processing pass.
   std::unique_ptr<float[], AlignedFreeDeleter> input_buffer_;

   // Stores the runtime selection of which Convolve function to use.
   // TODO(ajm): Move to using a global static which must only be initialized
   // once by the user. We're not doing this initially, because we don't have
   // e.g. a LazyInstance helper in webrtc.
   typedef float (*ConvolveProc)(const float*,
                                 const float*,
                                 const float*,
                                 double);
   ConvolveProc convolve_proc_;

   // Pointers to the various regions inside `input_buffer_`.  See the diagram at
   // the top of the .cc file for more information.
   float* r0_;
   float* const r1_;
   float* const r2_;
   float* r3_;
   float* r4_;
 };

 }  // namespace webrtc

 #endif  // COMMON_AUDIO_RESAMPLER_SINC_RESAMPLER_H_
	/*
	* Copyright (c) 2013 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.
	*/

	// Modified from the Chromium original here:
	// src/media/base/sinc_resampler.h

	#ifndef COMMON_AUDIO_RESAMPLER_SINC_RESAMPLER_H_
	#define COMMON_AUDIO_RESAMPLER_SINC_RESAMPLER_H_

	#include <stddef.h>

	#include <memory>

	#include "rtc_base/gtest_prod_util.h"
	#include "rtc_base/memory/aligned_malloc.h"
	#include "rtc_base/system/arch.h"

	namespace webrtc {

	// Callback class for providing more data into the resampler. Expects `frames`
	// of data to be rendered into `destination`; zero padded if not enough frames
	// are available to satisfy the request.
	class SincResamplerCallback {
	public:
	virtual ~SincResamplerCallback() {}
	virtual void Run(size_t frames, float* destination) = 0;
	};

	// SincResampler is a high-quality single-channel sample-rate converter.
	class SincResampler {
	public:
	// The kernel size can be adjusted for quality (higher is better) at the
	// expense of performance. Must be a multiple of 32.
	// TODO(dalecurtis): Test performance to see if we can jack this up to 64+.
	static const size_t kKernelSize = 32;

	// Default request size. Affects how often and for how much SincResampler
	// calls back for input. Must be greater than kKernelSize.
	static const size_t kDefaultRequestSize = 512;

	// The kernel offset count is used for interpolation and is the number of
	// sub-sample kernel shifts. Can be adjusted for quality (higher is better)
	// at the expense of allocating more memory.
	static const size_t kKernelOffsetCount = 32;
	static const size_t kKernelStorageSize =
	kKernelSize * (kKernelOffsetCount + 1);

	// Constructs a SincResampler with the specified `read_cb`, which is used to
	// acquire audio data for resampling. `io_sample_rate_ratio` is the ratio
	// of input / output sample rates. `request_frames` controls the size in
	// frames of the buffer requested by each `read_cb` call. The value must be
	// greater than kKernelSize. Specify kDefaultRequestSize if there are no
	// request size constraints.
	SincResampler(double io_sample_rate_ratio,
	size_t request_frames,
	SincResamplerCallback* read_cb);
	virtual ~SincResampler();

	SincResampler(const SincResampler&) = delete;
	SincResampler& operator=(const SincResampler&) = delete;

	// Resample `frames` of data from `read_cb_` into `destination`.
	void Resample(size_t frames, float* destination);

	// The maximum size in frames that guarantees Resample() will only make a
	// single call to `read_cb_` for more data.
	size_t ChunkSize() const;

	size_t request_frames() const { return request_frames_; }

	// Flush all buffered data and reset internal indices. Not thread safe, do
	// not call while Resample() is in progress.
	void Flush();

	// Update `io_sample_rate_ratio_`. SetRatio() will cause a reconstruction of
	// the kernels used for resampling. Not thread safe, do not call while
	// Resample() is in progress.
	//
	// TODO(ajm): Use this in PushSincResampler rather than reconstructing
	// SincResampler. We would also need a way to update `request_frames_`.
	void SetRatio(double io_sample_rate_ratio);

	float* get_kernel_for_testing() { return kernel_storage_.get(); }

	private:
	FRIEND_TEST_ALL_PREFIXES(SincResamplerTest, Convolve);
	FRIEND_TEST_ALL_PREFIXES(SincResamplerTest, ConvolveBenchmark);

	void InitializeKernel();
	void UpdateRegions(bool second_load);

	// Selects runtime specific CPU features like SSE. Must be called before
	// using SincResampler.
	// TODO(ajm): Currently managed by the class internally. See the note with
	// `convolve_proc_` below.
	void InitializeCPUSpecificFeatures();

	// Compute convolution of `k1` and `k2` over `input_ptr`, resultant sums are
	// linearly interpolated using `kernel_interpolation_factor`. On x86 and ARM
	// the underlying implementation is chosen at run time.
	static float Convolve_C(const float* input_ptr,
	const float* k1,
	const float* k2,
	double kernel_interpolation_factor);
	#if defined(WEBRTC_ARCH_X86_FAMILY)
	static float Convolve_SSE(const float* input_ptr,
	const float* k1,
	const float* k2,
	double kernel_interpolation_factor);
	static float Convolve_AVX2(const float* input_ptr,
	const float* k1,
	const float* k2,
	double kernel_interpolation_factor);
	#elif defined(WEBRTC_HAS_NEON)
	static float Convolve_NEON(const float* input_ptr,
	const float* k1,
	const float* k2,
	double kernel_interpolation_factor);
	#endif

	// The ratio of input / output sample rates.
	double io_sample_rate_ratio_;

	// An index on the source input buffer with sub-sample precision. It must be
	// double precision to avoid drift.
	double virtual_source_idx_;

	// The buffer is primed once at the very beginning of processing.
	bool buffer_primed_;

	// Source of data for resampling.
	SincResamplerCallback* read_cb_;

	// The size (in samples) to request from each `read_cb_` execution.
	const size_t request_frames_;

	// The number of source frames processed per pass.
	size_t block_size_;

	// The size (in samples) of the internal buffer used by the resampler.
	const size_t input_buffer_size_;

	// Contains kKernelOffsetCount kernels back-to-back, each of size kKernelSize.
	// The kernel offsets are sub-sample shifts of a windowed sinc shifted from
	// 0.0 to 1.0 sample.
	std::unique_ptr<float[], AlignedFreeDeleter> kernel_storage_;
	std::unique_ptr<float[], AlignedFreeDeleter> kernel_pre_sinc_storage_;
	std::unique_ptr<float[], AlignedFreeDeleter> kernel_window_storage_;

	// Data from the source is copied into this buffer for each processing pass.
	std::unique_ptr<float[], AlignedFreeDeleter> input_buffer_;

	// Stores the runtime selection of which Convolve function to use.
	// TODO(ajm): Move to using a global static which must only be initialized
	// once by the user. We're not doing this initially, because we don't have
	// e.g. a LazyInstance helper in webrtc.
	typedef float (ConvolveProc)(const float,
	const float*,
	const float*,
	double);
	ConvolveProc convolve_proc_;

	// Pointers to the various regions inside `input_buffer_`. See the diagram at
	// the top of the .cc file for more information.
	float* r0_;
	float* const r1_;
	float* const r2_;
	float* r3_;
	float* r4_;
	};

	} // namespace webrtc

	#endif // COMMON_AUDIO_RESAMPLER_SINC_RESAMPLER_H_