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/*
* 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 WEBRTC_COMMON_AUDIO_RESAMPLER_SINC_RESAMPLER_H_
#define WEBRTC_COMMON_AUDIO_RESAMPLER_SINC_RESAMPLER_H_
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/system_wrappers/include/aligned_malloc.h"
#ifndef WEBRTC_AUDIO_PROCESSING_ONLY_BUILD
#include "webrtc/test/testsupport/gtest_prod_util.h"
#endif
#include "webrtc/typedefs.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();
// 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:
#ifndef WEBRTC_AUDIO_PROCESSING_ONLY_BUILD
FRIEND_TEST_ALL_PREFIXES(SincResamplerTest, Convolve);
FRIEND_TEST_ALL_PREFIXES(SincResamplerTest, ConvolveBenchmark);
#endif
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);
#elif defined(WEBRTC_DETECT_NEON) || 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.
rtc::scoped_ptr<float[], AlignedFreeDeleter> kernel_storage_;
rtc::scoped_ptr<float[], AlignedFreeDeleter> kernel_pre_sinc_storage_;
rtc::scoped_ptr<float[], AlignedFreeDeleter> kernel_window_storage_;
// Data from the source is copied into this buffer for each processing pass.
rtc::scoped_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.
#if defined(WEBRTC_CPU_DETECTION)
typedef float (*ConvolveProc)(const float*, const float*, const float*,
double);
ConvolveProc convolve_proc_;
#endif
// 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_;
RTC_DISALLOW_COPY_AND_ASSIGN(SincResampler);
};
} // namespace webrtc
#endif // WEBRTC_COMMON_AUDIO_RESAMPLER_SINC_RESAMPLER_H_
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