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Diffstat (limited to 'chromium/third_party/webrtc/modules/audio_coding/neteq/normal.h')
-rw-r--r-- | chromium/third_party/webrtc/modules/audio_coding/neteq/normal.h | 68 |
1 files changed, 68 insertions, 0 deletions
diff --git a/chromium/third_party/webrtc/modules/audio_coding/neteq/normal.h b/chromium/third_party/webrtc/modules/audio_coding/neteq/normal.h new file mode 100644 index 00000000000..aa24b528af4 --- /dev/null +++ b/chromium/third_party/webrtc/modules/audio_coding/neteq/normal.h @@ -0,0 +1,68 @@ +/* + * 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_ |