Adding upstream version 115.7.0esr.upstream/115.7.0esrupstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 181 insertions, 0 deletions

diff --git a/third_party/libwebrtc/common_audio/resampler/sinc_resampler.h b/third_party/libwebrtc/common_audio/resampler/sinc_resampler.h
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index 0000000000..b89bba7ab4
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+++ b/third_party/libwebrtc/common_audio/resampler/sinc_resampler.h
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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 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_