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// Copyright (c) 2015 The Chromium Authors. All rights reserved.
// Use of this source code is governed by a BSD-style license that can be
// found in the LICENSE file.
// Modifications Copyright 2019 The Cobalt Authors. All Rights Reserved.
// Licensed under the Apache License, Version 2.0 (the "License");
// you may not use this file except in compliance with the License.
// You may obtain a copy of the License at
// Unless required by applicable law or agreed to in writing, software
// distributed under the License is distributed on an "AS IS" BASIS,
// See the License for the specific language governing permissions and
// limitations under the License.
#include <math.h>
#include <queue>
#include "starboard/common/ref_counted.h"
#include "starboard/common/scoped_ptr.h"
#include "starboard/shared/internal_only.h"
namespace starboard {
namespace shared {
namespace starboard {
namespace player {
namespace filter {
class InterleavedSincResampler {
// |io_sample_rate_ratio| is the ratio of input / output sample rates.
// |channel_count| is the number of channels in the interleaved audio stream.
InterleavedSincResampler(double io_sample_rate_ratio, int channel_count);
// Append a buffer to the queue. The samples in the buffer has to be floats.
void QueueBuffer(const float* data, int frames);
// Resample |frames| of data from enqueued buffers. Return false if no sample
// is read. Return true if all requested samples have been written into
// |destination|. It will never do a partial read. After the stream reaches
// the end, the function will fill the rest of buffer with 0.
void Resample(float* destination, int frames);
// Flush all buffered data and reset internal indices.
void Flush();
// Return false if we shouldn't queue more buffers to the resampler.
bool CanQueueBuffer() const;
// Return true if the enqueued buffers have enough data for us to resample the
// requested number of frames.
bool HasEnoughData(int frames_to_resample) const;
double GetSampleRateRatio() const { return io_sample_rate_ratio_; }
int GetNumberOfCashedFrames() const;
int channels() const { return channel_count_; }
class Buffer : public ::starboard::RefCountedThreadSafe<Buffer> {
Buffer() {}
Buffer(const float* data, int frames, int channel_count) : frames_(frames) {
data_.reset(new float[frames * channel_count]);
SbMemoryCopy(data_.get(), data, frames * channel_count * sizeof(float));
Buffer(scoped_array<float> data, int frames)
: data_(data.Pass()), frames_(frames) {}
const float* GetData() const { return data_.get(); }
int GetNumFrames() const { return frames_; }
bool IsEndOfStream() const { return GetData() == NULL; }
friend class ::starboard::RefCountedThreadSafe<Buffer>;
scoped_array<float> data_;
int frames_ = 0;
// The kernel size can be adjusted for quality (higher is better) at the
// expense of performance. Must be a multiple of 32.
static const int kKernelSize = 32;
// The number of destination frames generated per processing pass. Affects
// how often and for how much AudioResampler calls back for input.
// Must be greater than kKernelSize.
static const int kBlockSize = 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 int kKernelOffsetCount = 32;
static const int kKernelStorageSize = kKernelSize * (kKernelOffsetCount + 1);
// The size (in samples) of the internal buffer used by the resampler.
static const int kBufferSize = kBlockSize + kKernelSize;
// The maximum numbers of buffer can be queued.
static const int kMaximumPendingBuffers = 8;
static const int kMaxChannels = 8;
static const int kInputBufferSize = kMaxChannels * kBufferSize;
void InitializeKernel();
void Read(float* destination, int frames);
float Convolve(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_ = 0.0;
// An index on the source input buffer with sub-sample precision. It must be
// double precision to avoid drift.
double virtual_source_idx_ = 0.0;
// The buffer is primed once at the very beginning of processing.
bool buffer_primed_ = false;
// Number of audio channels.
int channel_count_ = 0;
// The size of bytes for an audio frame.
const int frame_size_in_bytes_;
// 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.
float kernel_storage_[kKernelStorageSize] = {0};
// Data from the source is copied into this buffer for each processing pass.
float input_buffer_[kInputBufferSize] = {0};
// A queue of buffers to be resampled.
std::queue<scoped_refptr<Buffer>> pending_buffers_;
// The current offset to read when reading from the first pending buffer.
int offset_in_frames_ = 0;
// The following two variables are used to calculate EOS and in HasEnoughData.
int frames_resampled_ = 0;
int frames_queued_ = 0;
// Pointers to the various regions inside |input_buffer_|. See the diagram at
// the top of the .cc file for more information.
float* r0_;
float* r1_;
float* r2_;
float* r3_;
float* r4_;
float* r5_;
} // namespace filter
} // namespace player
} // namespace starboard
} // namespace shared
} // namespace starboard