/* * 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/system_wrappers/interface/aligned_malloc.h" #include "webrtc/system_wrappers/interface/constructor_magic.h" #include "webrtc/system_wrappers/interface/scoped_ptr.h" #include "webrtc/test/testsupport/gtest_prod_util.h" namespace webrtc { // Callback class to provide SincResampler with input. class SincResamplerCallback { public: virtual ~SincResamplerCallback() {} virtual void Run(float* destination, int frames) = 0; }; // SincResampler is a high-quality single-channel sample-rate converter. class SincResampler { public: // 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. If desired, the number of destination frames // generated per processing pass can be specified through |block_size|. SincResampler(double io_sample_rate_ratio, SincResamplerCallback* read_cb); SincResampler(double io_sample_rate_ratio, SincResamplerCallback* read_cb, int block_size); virtual ~SincResampler(); // Resample |frames| of data from |read_cb_| into |destination|. void Resample(float* destination, int frames); // The maximum size in frames that guarantees Resample() will only make a // single call to |read_cb_| for more data. int ChunkSize(); // Flush all buffered data and reset internal indices. void Flush(); private: FRIEND_TEST_ALL_PREFIXES(SincResamplerTest, Convolve); FRIEND_TEST_ALL_PREFIXES(SincResamplerTest, ConvolveBenchmark); void Initialize(); void InitializeKernel(); // Compute convolution of |k1| and |k2| over |input_ptr|, resultant sums are // linearly interpolated using |kernel_interpolation_factor|. On x86, the // underlying implementation is chosen at run time based on SSE support. On // ARM, NEON support is chosen at compile time based on compilation flags. static float Convolve(const float* input_ptr, const float* k1, const float* k2, double kernel_interpolation_factor); static float Convolve_C(const float* input_ptr, const float* k1, const float* k2, double kernel_interpolation_factor); static float Convolve_SSE(const float* input_ptr, const float* k1, const float* k2, double kernel_interpolation_factor); static float Convolve_NEON(const float* input_ptr, const float* k1, const float* k2, double kernel_interpolation_factor); // 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_; // See kDefaultBlockSize. int block_size_; // See kDefaultBufferSize. int 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. scoped_ptr_malloc kernel_storage_; // Data from the source is copied into this buffer for each processing pass. scoped_ptr_malloc input_buffer_; // Pointers to the various regions inside |input_buffer_|. See the diagram at // the top of the .cc file for more information. float* const r0_; float* const r1_; float* const r2_; float* const r3_; float* const r4_; float* const r5_; DISALLOW_COPY_AND_ASSIGN(SincResampler); }; } // namespace webrtc #endif // WEBRTC_COMMON_AUDIO_RESAMPLER_SINC_RESAMPLER_H_