172 lines
6.7 KiB
C
172 lines
6.7 KiB
C
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/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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// Modified from the Chromium original here:
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// src/media/base/sinc_resampler.h
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#ifndef WEBRTC_COMMON_AUDIO_RESAMPLER_SINC_RESAMPLER_H_
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#define WEBRTC_COMMON_AUDIO_RESAMPLER_SINC_RESAMPLER_H_
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#include <memory>
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/base/gtest_prod_util.h"
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#include "webrtc/system_wrappers/include/aligned_malloc.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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// Callback class for providing more data into the resampler. Expects |frames|
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// of data to be rendered into |destination|; zero padded if not enough frames
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// are available to satisfy the request.
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class SincResamplerCallback {
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public:
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virtual ~SincResamplerCallback() {}
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virtual void Run(size_t frames, float* destination) = 0;
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};
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// SincResampler is a high-quality single-channel sample-rate converter.
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class SincResampler {
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public:
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// The kernel size can be adjusted for quality (higher is better) at the
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// expense of performance. Must be a multiple of 32.
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// TODO(dalecurtis): Test performance to see if we can jack this up to 64+.
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static const size_t kKernelSize = 32;
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// Default request size. Affects how often and for how much SincResampler
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// calls back for input. Must be greater than kKernelSize.
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static const size_t kDefaultRequestSize = 512;
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// The kernel offset count is used for interpolation and is the number of
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// sub-sample kernel shifts. Can be adjusted for quality (higher is better)
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// at the expense of allocating more memory.
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static const size_t kKernelOffsetCount = 32;
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static const size_t kKernelStorageSize =
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kKernelSize * (kKernelOffsetCount + 1);
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// Constructs a SincResampler with the specified |read_cb|, which is used to
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// acquire audio data for resampling. |io_sample_rate_ratio| is the ratio
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// of input / output sample rates. |request_frames| controls the size in
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// frames of the buffer requested by each |read_cb| call. The value must be
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// greater than kKernelSize. Specify kDefaultRequestSize if there are no
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// request size constraints.
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SincResampler(double io_sample_rate_ratio,
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size_t request_frames,
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SincResamplerCallback* read_cb);
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virtual ~SincResampler();
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// Resample |frames| of data from |read_cb_| into |destination|.
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void Resample(size_t frames, float* destination);
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// The maximum size in frames that guarantees Resample() will only make a
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// single call to |read_cb_| for more data.
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size_t ChunkSize() const;
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size_t request_frames() const { return request_frames_; }
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// Flush all buffered data and reset internal indices. Not thread safe, do
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// not call while Resample() is in progress.
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void Flush();
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// Update |io_sample_rate_ratio_|. SetRatio() will cause a reconstruction of
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// the kernels used for resampling. Not thread safe, do not call while
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// Resample() is in progress.
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//
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// TODO(ajm): Use this in PushSincResampler rather than reconstructing
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// SincResampler. We would also need a way to update |request_frames_|.
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void SetRatio(double io_sample_rate_ratio);
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float* get_kernel_for_testing() { return kernel_storage_.get(); }
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private:
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FRIEND_TEST_ALL_PREFIXES(SincResamplerTest, Convolve);
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FRIEND_TEST_ALL_PREFIXES(SincResamplerTest, ConvolveBenchmark);
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void InitializeKernel();
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void UpdateRegions(bool second_load);
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// Selects runtime specific CPU features like SSE. Must be called before
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// using SincResampler.
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// TODO(ajm): Currently managed by the class internally. See the note with
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// |convolve_proc_| below.
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void InitializeCPUSpecificFeatures();
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// Compute convolution of |k1| and |k2| over |input_ptr|, resultant sums are
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// linearly interpolated using |kernel_interpolation_factor|. On x86 and ARM
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// the underlying implementation is chosen at run time.
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static float Convolve_C(const float* input_ptr, const float* k1,
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const float* k2, double kernel_interpolation_factor);
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#if defined(WEBRTC_ARCH_X86_FAMILY)
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static float Convolve_SSE(const float* input_ptr, const float* k1,
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const float* k2,
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double kernel_interpolation_factor);
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#elif defined(WEBRTC_HAS_NEON)
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static float Convolve_NEON(const float* input_ptr, const float* k1,
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const float* k2,
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double kernel_interpolation_factor);
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#endif
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// The ratio of input / output sample rates.
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double io_sample_rate_ratio_;
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// An index on the source input buffer with sub-sample precision. It must be
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// double precision to avoid drift.
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double virtual_source_idx_;
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// The buffer is primed once at the very beginning of processing.
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bool buffer_primed_;
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// Source of data for resampling.
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SincResamplerCallback* read_cb_;
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// The size (in samples) to request from each |read_cb_| execution.
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const size_t request_frames_;
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// The number of source frames processed per pass.
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size_t block_size_;
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// The size (in samples) of the internal buffer used by the resampler.
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const size_t input_buffer_size_;
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// Contains kKernelOffsetCount kernels back-to-back, each of size kKernelSize.
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// The kernel offsets are sub-sample shifts of a windowed sinc shifted from
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// 0.0 to 1.0 sample.
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std::unique_ptr<float[], AlignedFreeDeleter> kernel_storage_;
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std::unique_ptr<float[], AlignedFreeDeleter> kernel_pre_sinc_storage_;
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std::unique_ptr<float[], AlignedFreeDeleter> kernel_window_storage_;
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// Data from the source is copied into this buffer for each processing pass.
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std::unique_ptr<float[], AlignedFreeDeleter> input_buffer_;
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// Stores the runtime selection of which Convolve function to use.
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// TODO(ajm): Move to using a global static which must only be initialized
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// once by the user. We're not doing this initially, because we don't have
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// e.g. a LazyInstance helper in webrtc.
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#if defined(WEBRTC_CPU_DETECTION)
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typedef float (*ConvolveProc)(const float*, const float*, const float*,
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double);
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ConvolveProc convolve_proc_;
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#endif
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// Pointers to the various regions inside |input_buffer_|. See the diagram at
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// the top of the .cc file for more information.
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float* r0_;
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float* const r1_;
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float* const r2_;
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float* r3_;
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float* r4_;
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RTC_DISALLOW_COPY_AND_ASSIGN(SincResampler);
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};
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} // namespace webrtc
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#endif // WEBRTC_COMMON_AUDIO_RESAMPLER_SINC_RESAMPLER_H_
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