| /* |
| * 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_; |
| |
| RTC_DISALLOW_COPY_AND_ASSIGN(Normal); |
| }; |
| |
| } // namespace webrtc |
| #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_NORMAL_H_ |