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https://github.com/danog/libtgvoip.git
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5caaaafa42
I'm now using the entire audio processing module from WebRTC as opposed to individual DSP algorithms pulled from there before. Seems to work better this way.
81 lines
3.0 KiB
C++
81 lines
3.0 KiB
C++
/*
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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 "common_audio/resampler/sinc_resampler.h"
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#include "rtc_base/constructormagic.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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// 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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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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RTC_DISALLOW_COPY_AND_ASSIGN(PushSincResampler);
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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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