blob: 2059c5a795f9f205497713971f9f5600f63c4519 [file] [log] [blame]
* 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.
#include <stdint.h>
#include <string.h> // Access to size_t.
#include "modules/audio_coding/neteq/defines.h"
#include "rtc_base/checks.h"
#include "rtc_base/constructor_magic.h"
#include "rtc_base/numerics/safe_conversions.h"
namespace webrtc {
// Forward declarations.
class AudioMultiVector;
class BackgroundNoise;
class DecoderDatabase;
class Expand;
// This class provides the "Normal" DSP operation, that is performed when
// there is no data loss, no need to stretch the timing of the signal, and
// no other "special circumstances" are at hand.
class Normal {
Normal(int fs_hz,
DecoderDatabase* decoder_database,
const BackgroundNoise& background_noise,
Expand* expand)
: fs_hz_(fs_hz),
samples_per_ms_(rtc::CheckedDivExact(fs_hz_, 1000)),
rtc::dchecked_cast<uint16_t>((1 << 14) / samples_per_ms_)) {}
virtual ~Normal() {}
// Performs the "Normal" operation. The decoder data is supplied in |input|,
// having |length| samples in total for all channels (interleaved). The
// result is written to |output|. The number of channels allocated in
// |output| defines the number of channels that will be used when
// de-interleaving |input|. |last_mode| contains the mode used in the previous
// GetAudio call (i.e., not the current one).
int Process(const int16_t* input,
size_t length,
Modes last_mode,
AudioMultiVector* output);
int fs_hz_;
DecoderDatabase* decoder_database_;
const BackgroundNoise& background_noise_;
Expand* expand_;
const size_t samples_per_ms_;
const int16_t default_win_slope_Q14_;
} // namespace webrtc