Initial upload of NetEq4
This is the first public upload of the new NetEq, version 4.
It has been through extensive internal review during the course of
the project.
TEST=trybots
Review URL: https://webrtc-codereview.appspot.com/1073005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@3425 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/audio_coding/neteq4/normal.h b/webrtc/modules/audio_coding/neteq4/normal.h
new file mode 100644
index 0000000..aba55c4
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+++ b/webrtc/modules/audio_coding/neteq4/normal.h
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+/*
+ * 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_NETEQ4_NORMAL_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_NORMAL_H_
+
+#include <cstring> // Access to size_t.
+#include <vector>
+
+#include "webrtc/modules/audio_coding/neteq4/audio_multi_vector.h"
+#include "webrtc/modules/audio_coding/neteq4/defines.h"
+#include "webrtc/system_wrappers/interface/constructor_magic.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<int16_t>* output);
+
+ private:
+ int fs_hz_;
+ DecoderDatabase* decoder_database_;
+ const BackgroundNoise& background_noise_;
+ Expand* expand_;
+
+ DISALLOW_COPY_AND_ASSIGN(Normal);
+};
+
+} // namespace webrtc
+#endif // SRC_MODULES_AUDIO_CODING_NETEQ4_NORMAL_H_