89 lines
3.3 KiB
C
89 lines
3.3 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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#ifndef COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_
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#define COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_
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#include <stddef.h>
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#include <stdint.h>
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#include <memory>
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#include "api/audio/audio_view.h"
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#include "common_audio/resampler/sinc_resampler.h"
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namespace webrtc {
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// A thin wrapper over SincResampler to provide a push-based interface as
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// required by WebRTC. SincResampler uses a pull-based interface, and will
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// use SincResamplerCallback::Run() to request data upon a call to Resample().
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// These Run() calls will happen on the same thread Resample() is called on.
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class PushSincResampler : public SincResamplerCallback {
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public:
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// Provide the size of the source and destination blocks in samples. These
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// must correspond to the same time duration (typically 10 ms) as the sample
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// ratio is inferred from them.
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PushSincResampler(size_t source_frames, size_t destination_frames);
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~PushSincResampler() override;
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PushSincResampler(const PushSincResampler&) = delete;
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PushSincResampler& operator=(const PushSincResampler&) = delete;
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// Perform the resampling. `source_frames` must always equal the
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// `source_frames` provided at construction. `destination_capacity` must be
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// at least as large as `destination_frames`. Returns the number of samples
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// provided in destination (for convenience, since this will always be equal
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// to `destination_frames`).
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template <typename S, typename D>
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size_t Resample(const MonoView<S>& source, const MonoView<D>& destination) {
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return Resample(&source[0], SamplesPerChannel(source), &destination[0],
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SamplesPerChannel(destination));
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}
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size_t Resample(const int16_t* source,
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size_t source_frames,
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int16_t* destination,
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size_t destination_capacity);
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size_t Resample(const float* source,
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size_t source_frames,
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float* destination,
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size_t destination_capacity);
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// Delay due to the filter kernel. Essentially, the time after which an input
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// sample will appear in the resampled output.
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static float AlgorithmicDelaySeconds(int source_rate_hz) {
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return 1.f / source_rate_hz * SincResampler::kKernelSize / 2;
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}
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protected:
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// Implements SincResamplerCallback.
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void Run(size_t frames, float* destination) override;
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private:
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friend class PushSincResamplerTest;
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SincResampler* get_resampler_for_testing() { return resampler_.get(); }
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std::unique_ptr<SincResampler> resampler_;
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std::unique_ptr<float[]> float_buffer_;
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const float* source_ptr_;
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const int16_t* source_ptr_int_;
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const size_t destination_frames_;
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// True on the first call to Resample(), to prime the SincResampler buffer.
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bool first_pass_;
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// Used to assert we are only requested for as much data as is available.
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size_t source_available_;
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};
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} // namespace webrtc
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#endif // COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_
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