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/*
* Copyright (c) 2012 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 MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_
#define MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_
#include <stddef.h>
#include <stdint.h>
#include "api/audio/audio_frame.h"
#include "common_audio/resampler/include/push_resampler.h"
namespace webrtc {
namespace acm2 {
class ACMResampler {
public:
ACMResampler();
~ACMResampler();
// TODO: b/335805780 - Change to accept InterleavedView<>.
int Resample10Msec(const int16_t* in_audio,
int in_freq_hz,
int out_freq_hz,
size_t num_audio_channels,
size_t out_capacity_samples,
int16_t* out_audio);
private:
PushResampler<int16_t> resampler_;
};
// Helper class to perform resampling if needed, meant to be used after
// receiving the audio_frame from NetEq. Provides reasonably glitch free
// transitions between different output sample rates from NetEq.
class ResamplerHelper {
public:
ResamplerHelper();
// Resamples audio_frame if it is not already in desired_sample_rate_hz.
bool MaybeResample(int desired_sample_rate_hz, AudioFrame* audio_frame);
private:
ACMResampler resampler_;
bool resampled_last_output_frame_ = true;
std::array<int16_t, AudioFrame::kMaxDataSizeSamples> last_audio_buffer_;
};
} // namespace acm2
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_ACM2_ACM_RESAMPLER_H_