d6ef33e59b
This switches from accepting a sample rate and convert to channel size over to accepting the channel size. Instead of InitializeIfNeeded: * Offer a way to explicitly initialize PushResampler via the ctor (needed for VoiceActivityDetectorWrapper) * Implicitly check for the right configuration from within Resample(). (All calls to Resample() were preceded by a call to Initialize) As part of this, refactor VoiceActivityDetectorWrapper (VADW): * VADW is now initialized in the constructor and more const. * Remove VADW::Initialize() and instead reconstruct VADW if needed. Add constants for max sample rate and num channels to audio_util.h In many cases the numbers for these values are embedded in the code which has led to some inconsistency. Bug: chromium:335805780 Change-Id: Iead0d52eb1b261a8d64e93f51401147c8fba32f0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/353360 Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42587}
59 lines
2.0 KiB
C++
59 lines
2.0 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_INCLUDE_PUSH_RESAMPLER_H_
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#define COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_
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#include <memory>
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#include <vector>
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#include "api/audio/audio_view.h"
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namespace webrtc {
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class PushSincResampler;
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// Wraps PushSincResampler to provide stereo support.
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// Note: This implementation assumes 10ms buffer sizes throughout.
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template <typename T>
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class PushResampler final {
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public:
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PushResampler();
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PushResampler(size_t src_samples_per_channel,
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size_t dst_samples_per_channel,
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size_t num_channels);
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~PushResampler();
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// Returns the total number of samples provided in destination (e.g. 32 kHz,
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// 2 channel audio gives 640 samples).
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int Resample(InterleavedView<const T> src, InterleavedView<T> dst);
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// For when a deinterleaved/mono channel already exists and we can skip the
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// deinterleaved operation.
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int Resample(MonoView<const T> src, MonoView<T> dst);
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private:
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// Ensures that source and destination buffers for deinterleaving are
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// correctly configured prior to resampling that requires deinterleaving.
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void EnsureInitialized(size_t src_samples_per_channel,
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size_t dst_samples_per_channel,
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size_t num_channels);
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// Buffers used for when a deinterleaving step is necessary.
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std::unique_ptr<T[]> source_;
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std::unique_ptr<T[]> destination_;
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DeinterleavedView<T> source_view_;
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DeinterleavedView<T> destination_view_;
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std::vector<std::unique_ptr<PushSincResampler>> resamplers_;
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};
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} // namespace webrtc
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#endif // COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_
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