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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.
*/
#include "common_audio/resampler/push_sinc_resampler.h"
#include <cstring>
#include "common_audio/include/audio_util.h"
#include "rtc_base/checks.h"
namespace webrtc {
PushSincResampler::PushSincResampler(size_t source_frames,
size_t destination_frames)
: resampler_(new SincResampler(source_frames * 1.0 / destination_frames,
source_frames,
this)),
source_ptr_(nullptr),
source_ptr_int_(nullptr),
destination_frames_(destination_frames),
first_pass_(true),
source_available_(0) {}
PushSincResampler::~PushSincResampler() {}
size_t PushSincResampler::Resample(const int16_t* source,
size_t source_length,
int16_t* destination,
size_t /* destination_capacity */) {
if (!float_buffer_.get())
float_buffer_.reset(new float[destination_frames_]);
source_ptr_int_ = source;
// Pass nullptr as the float source to have Run() read from the int16 source.
Resample(nullptr, source_length, float_buffer_.get(), destination_frames_);
FloatS16ToS16(float_buffer_.get(), destination_frames_, destination);
source_ptr_int_ = nullptr;
return destination_frames_;
}
size_t PushSincResampler::Resample(const float* source,
size_t source_length,
float* destination,
size_t destination_capacity) {
RTC_CHECK_EQ(source_length, resampler_->request_frames());
RTC_CHECK_GE(destination_capacity, destination_frames_);
// Cache the source pointer. Calling Resample() will immediately trigger
// the Run() callback whereupon we provide the cached value.
source_ptr_ = source;
source_available_ = source_length;
// On the first pass, we call Resample() twice. During the first call, we
// provide dummy input and discard the output. This is done to prime the
// SincResampler buffer with the correct delay (half the kernel size), thereby
// ensuring that all later Resample() calls will only result in one input
// request through Run().
//
// If this wasn't done, SincResampler would call Run() twice on the first
// pass, and we'd have to introduce an entire `source_frames` of delay, rather
// than the minimum half kernel.
//
// It works out that ChunkSize() is exactly the amount of output we need to
// request in order to prime the buffer with a single Run() request for
// `source_frames`.
if (first_pass_)
resampler_->Resample(resampler_->ChunkSize(), destination);
resampler_->Resample(destination_frames_, destination);
source_ptr_ = nullptr;
return destination_frames_;
}
void PushSincResampler::Run(size_t frames, float* destination) {
// Ensure we are only asked for the available samples. This would fail if
// Run() was triggered more than once per Resample() call.
RTC_CHECK_EQ(source_available_, frames);
if (first_pass_) {
// Provide dummy input on the first pass, the output of which will be
// discarded, as described in Resample().
std::memset(destination, 0, frames * sizeof(*destination));
first_pass_ = false;
return;
}
if (source_ptr_) {
std::memcpy(destination, source_ptr_, frames * sizeof(*destination));
} else {
for (size_t i = 0; i < frames; ++i)
destination[i] = static_cast<float>(source_ptr_int_[i]);
}
source_available_ -= frames;
}
} // namespace webrtc
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