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/merge.h b/webrtc/modules/audio_coding/neteq4/merge.h
new file mode 100644
index 0000000..fdb9a16
--- /dev/null
+++ b/webrtc/modules/audio_coding/neteq4/merge.h
@@ -0,0 +1,104 @@
+/*
+ * 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_MERGE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MERGE_H_
+
+#include <assert.h>
+
+#include "webrtc/modules/audio_coding/neteq4/audio_multi_vector.h"
+#include "webrtc/system_wrappers/interface/constructor_magic.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+
+// Forward declarations.
+class Expand;
+class SyncBuffer;
+
+// This class handles the transition from expansion to normal operation.
+// When a packet is not available for decoding when needed, the expand operation
+// is called to generate extrapolation data. If the missing packet arrives,
+// i.e., it was just delayed, it can be decoded and appended directly to the
+// end of the expanded data (thanks to how the Expand class operates). However,
+// if a later packet arrives instead, the loss is a fact, and the new data must
+// be stitched together with the end of the expanded data. This stitching is
+// what the Merge class does.
+class Merge {
+ public:
+ Merge(int fs_hz, size_t num_channels, Expand* expand, SyncBuffer* sync_buffer)
+ : fs_hz_(fs_hz),
+ fs_mult_(fs_hz_ / 8000),
+ num_channels_(num_channels),
+ timestamps_per_call_(fs_hz_ / 100),
+ expand_(expand),
+ sync_buffer_(sync_buffer),
+ expanded_(num_channels_) {
+ assert(num_channels_ > 0);
+ }
+
+ // The main method to produce the audio data. The decoded data is supplied in
+ // |input|, having |input_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|. The values in |external_mute_factor_array| (Q14)
+ // will be used to scale the audio, and is updated in the process. The array
+ // must have |num_channels_| elements.
+ int Process(int16_t* input, int input_length,
+ int16_t* external_mute_factor_array,
+ AudioMultiVector<int16_t>* output);
+
+ private:
+ static const int kMaxSampleRate = 48000;
+ static const int kExpandDownsampLength = 100;
+ static const int kInputDownsampLength = 40;
+ static const int kMaxCorrelationLength = 60;
+
+ // Calls |expand_| to get more expansion data to merge with. The data is
+ // written to |expanded_signal_|. Returns the length of the expanded data,
+ // while |expand_period| will be the number of samples in one expansion period
+ // (typically one pitch period). The value of |old_length| will be the number
+ // of samples that were taken from the |sync_buffer_|.
+ int GetExpandedSignal(int* old_length, int* expand_period);
+
+ // Analyzes |input| and |expanded_signal| to find maximum values. Returns
+ // a muting factor (Q14) to be used on the new data.
+ int16_t SignalScaling(const int16_t* input, int input_length,
+ const int16_t* expanded_signal,
+ int16_t* expanded_max, int16_t* input_max) const;
+
+ // Downsamples |input| (|input_length| samples) and |expanded_signal| to
+ // 4 kHz sample rate. The downsampled signals are written to
+ // |input_downsampled_| and |expanded_downsampled_|, respectively.
+ void Downsample(const int16_t* input, int input_length,
+ const int16_t* expanded_signal, int expanded_length);
+
+ // Calculates cross-correlation between |input_downsampled_| and
+ // |expanded_downsampled_|, and finds the correlation maximum. The maximizing
+ // lag is returned.
+ int16_t CorrelateAndPeakSearch(int16_t expanded_max, int16_t input_max,
+ int start_position, int input_length,
+ int expand_period) const;
+
+ const int fs_hz_;
+ const int fs_mult_; // fs_hz_ / 8000.
+ const size_t num_channels_;
+ const int timestamps_per_call_;
+ Expand* expand_;
+ SyncBuffer* sync_buffer_;
+ int16_t expanded_downsampled_[kExpandDownsampLength];
+ int16_t input_downsampled_[kInputDownsampLength];
+ AudioMultiVector<int16_t> expanded_;
+
+ DISALLOW_COPY_AND_ASSIGN(Merge);
+};
+
+} // namespace webrtc
+#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_MERGE_H_