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604fd2f1ab
Bug: webrtc:13555, webrtc:13082 Change-Id: I2c2cbcbd918f0cfa970c1a964893220ba11d4b41 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/247960 Reviewed-by: Artem Titov <titovartem@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Commit-Queue: (Daniel.L) Byoungchan Lee <daniel.l@hpcnt.com> Cr-Commit-Position: refs/heads/main@{#35771}
76 lines
2.5 KiB
C++
76 lines
2.5 KiB
C++
/*
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* Copyright (c) 2012 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 MODULES_AUDIO_CODING_NETEQ_NORMAL_H_
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#define MODULES_AUDIO_CODING_NETEQ_NORMAL_H_
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#include <stdint.h>
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#include <string.h> // Access to size_t.
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#include "api/neteq/neteq.h"
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#include "modules/audio_coding/neteq/statistics_calculator.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/numerics/safe_conversions.h"
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namespace webrtc {
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// Forward declarations.
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class AudioMultiVector;
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class BackgroundNoise;
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class DecoderDatabase;
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class Expand;
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// This class provides the "Normal" DSP operation, that is performed when
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// there is no data loss, no need to stretch the timing of the signal, and
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// no other "special circumstances" are at hand.
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class Normal {
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public:
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Normal(int fs_hz,
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DecoderDatabase* decoder_database,
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const BackgroundNoise& background_noise,
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Expand* expand,
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StatisticsCalculator* statistics)
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: fs_hz_(fs_hz),
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decoder_database_(decoder_database),
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background_noise_(background_noise),
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expand_(expand),
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samples_per_ms_(rtc::CheckedDivExact(fs_hz_, 1000)),
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default_win_slope_Q14_(
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rtc::dchecked_cast<uint16_t>((1 << 14) / samples_per_ms_)),
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statistics_(statistics) {}
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virtual ~Normal() {}
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Normal(const Normal&) = delete;
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Normal& operator=(const Normal&) = delete;
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// Performs the "Normal" operation. The decoder data is supplied in `input`,
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// having `length` samples in total for all channels (interleaved). The
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// result is written to `output`. The number of channels allocated in
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// `output` defines the number of channels that will be used when
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// de-interleaving `input`. `last_mode` contains the mode used in the previous
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// GetAudio call (i.e., not the current one).
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int Process(const int16_t* input,
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size_t length,
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NetEq::Mode last_mode,
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AudioMultiVector* output);
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private:
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int fs_hz_;
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DecoderDatabase* decoder_database_;
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const BackgroundNoise& background_noise_;
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Expand* expand_;
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const size_t samples_per_ms_;
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const int16_t default_win_slope_Q14_;
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StatisticsCalculator* const statistics_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_NETEQ_NORMAL_H_
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