webrtc/common_audio/resampler/include/push_resampler.h
Tommi 1f3679884c Start using ArrayView in AudioFrame, update PushResampler
Start introducing ArrayView to AudioFrame and code that flows down
from there.  In this first step:
* Add `data_view()` that returns a read-only ArrayView for the
  audio buffer. When AudioFrame is not initialized however, data_view()
  will return a nullptr whereas the current data() method never returns
  nullptr.
* Add `mutable_data()` that requires two arguments for properly setting
  the samples per channel and number of channels that's required for
  accurately reserving the returned mutable ArrayView.
  A notable behavior change is that if the requested number of channels
  is larger than supported or the calculated buffer size is too large,
  the function will trigger a check.
* Add TODOs for following work.

Bug: chromium:335805780
Change-Id: I2937de800422589ebe6a3840b3caadf3d9ff8b00
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347982
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42202}
2024-04-30 15:33:08 +00:00

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2.1 KiB
C++

/*
* 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.
*/
#ifndef COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_
#define COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_
#include <memory>
#include <vector>
#include "api/array_view.h"
namespace webrtc {
class PushSincResampler;
// Wraps PushSincResampler to provide stereo support.
// Note: This implementation assumes 10ms buffer sizes throughout.
// TODO(ajm): add support for an arbitrary number of channels.
template <typename T>
class PushResampler {
public:
PushResampler();
virtual ~PushResampler();
// Must be called whenever the parameters change. Free to be called at any
// time as it is a no-op if parameters have not changed since the last call.
int InitializeIfNeeded(int src_sample_rate_hz,
int dst_sample_rate_hz,
size_t num_channels);
// Returns the total number of samples provided in destination (e.g. 32 kHz,
// 2 channel audio gives 640 samples).
int Resample(rtc::ArrayView<const T> src, rtc::ArrayView<T> dst);
private:
int src_sample_rate_hz_;
int dst_sample_rate_hz_;
size_t num_channels_;
// Vector that is needed to provide the proper inputs and outputs to the
// interleave/de-interleave methods used in Resample. This needs to be
// heap-allocated on the state to support an arbitrary number of channels
// without doing run-time heap-allocations in the Resample method.
std::vector<T*> channel_data_array_;
struct ChannelResampler {
std::unique_ptr<PushSincResampler> resampler;
std::vector<T> source;
std::vector<T> destination;
};
std::vector<ChannelResampler> channel_resamplers_;
};
} // namespace webrtc
#endif // COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_