mirror of
https://github.com/oxen-io/session-android.git
synced 2024-12-19 14:37:32 +00:00
69 lines
2.3 KiB
C
69 lines
2.3 KiB
C
|
/*
|
||
|
|
* 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 WEBRTC_MODULES_AUDIO_CODING_NETEQ_NORMAL_H_
|
||
|
|
#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_NORMAL_H_
|
||
|
|
|
||
|
|
#include <string.h> // Access to size_t.
|
||
|
|
|
||
|
|
#include <vector>
|
||
|
|
|
||
|
|
#include "webrtc/base/constructormagic.h"
|
||
|
|
#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
|
||
|
|
#include "webrtc/modules/audio_coding/neteq/defines.h"
|
||
|
|
#include "webrtc/typedefs.h"
|
||
|
|
|
||
|
|
namespace webrtc {
|
||
|
|
|
||
|
|
// Forward declarations.
|
||
|
|
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 {
|
||
|
|
public:
|
||
|
|
Normal(int fs_hz, DecoderDatabase* decoder_database,
|
||
|
|
const BackgroundNoise& background_noise,
|
||
|
|
Expand* expand)
|
||
|
|
: fs_hz_(fs_hz),
|
||
|
|
decoder_database_(decoder_database),
|
||
|
|
background_noise_(background_noise),
|
||
|
|
expand_(expand) {
|
||
|
|
}
|
||
|
|
|
||
|
|
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), and |external_mute_factor| is
|
||
|
|
// a pointer to the mute factor in the NetEqImpl class.
|
||
|
|
int Process(const int16_t* input, size_t length,
|
||
|
|
Modes last_mode,
|
||
|
|
int16_t* external_mute_factor_array,
|
||
|
|
AudioMultiVector* output);
|
||
|
|
|
||
|
|
private:
|
||
|
|
int fs_hz_;
|
||
|
|
DecoderDatabase* decoder_database_;
|
||
|
|
const BackgroundNoise& background_noise_;
|
||
|
|
Expand* expand_;
|
||
|
|
|
||
|
|
DISALLOW_COPY_AND_ASSIGN(Normal);
|
||
|
|
};
|
||
|
|
|
||
|
|
} // namespace webrtc
|
||
|
|
#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_NORMAL_H_
|