1/*
2 *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3 *
4 *  Use of this source code is governed by a BSD-style license
5 *  that can be found in the LICENSE file in the root of the source
6 *  tree. An additional intellectual property rights grant can be found
7 *  in the file PATENTS.  All contributing project authors may
8 *  be found in the AUTHORS file in the root of the source tree.
9 */
10
11#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_MERGE_H_
12#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_MERGE_H_
13
14#include <assert.h>
15
16#include "webrtc/base/constructormagic.h"
17#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
18#include "webrtc/typedefs.h"
19
20namespace webrtc {
21
22// Forward declarations.
23class Expand;
24class SyncBuffer;
25
26// This class handles the transition from expansion to normal operation.
27// When a packet is not available for decoding when needed, the expand operation
28// is called to generate extrapolation data. If the missing packet arrives,
29// i.e., it was just delayed, it can be decoded and appended directly to the
30// end of the expanded data (thanks to how the Expand class operates). However,
31// if a later packet arrives instead, the loss is a fact, and the new data must
32// be stitched together with the end of the expanded data. This stitching is
33// what the Merge class does.
34class Merge {
35 public:
36  Merge(int fs_hz, size_t num_channels, Expand* expand, SyncBuffer* sync_buffer)
37      : fs_hz_(fs_hz),
38        num_channels_(num_channels),
39        fs_mult_(fs_hz_ / 8000),
40        timestamps_per_call_(fs_hz_ / 100),
41        expand_(expand),
42        sync_buffer_(sync_buffer),
43        expanded_(num_channels_) {
44    assert(num_channels_ > 0);
45  }
46
47  virtual ~Merge() {}
48
49  // The main method to produce the audio data. The decoded data is supplied in
50  // |input|, having |input_length| samples in total for all channels
51  // (interleaved). The result is written to |output|. The number of channels
52  // allocated in |output| defines the number of channels that will be used when
53  // de-interleaving |input|. The values in |external_mute_factor_array| (Q14)
54  // will be used to scale the audio, and is updated in the process. The array
55  // must have |num_channels_| elements.
56  virtual int Process(int16_t* input, size_t input_length,
57                      int16_t* external_mute_factor_array,
58                      AudioMultiVector* output);
59
60  virtual int RequiredFutureSamples();
61
62 protected:
63  const int fs_hz_;
64  const size_t num_channels_;
65
66 private:
67  static const int kMaxSampleRate = 48000;
68  static const int kExpandDownsampLength = 100;
69  static const int kInputDownsampLength = 40;
70  static const int kMaxCorrelationLength = 60;
71
72  // Calls |expand_| to get more expansion data to merge with. The data is
73  // written to |expanded_signal_|. Returns the length of the expanded data,
74  // while |expand_period| will be the number of samples in one expansion period
75  // (typically one pitch period). The value of |old_length| will be the number
76  // of samples that were taken from the |sync_buffer_|.
77  int GetExpandedSignal(int* old_length, int* expand_period);
78
79  // Analyzes |input| and |expanded_signal| to find maximum values. Returns
80  // a muting factor (Q14) to be used on the new data.
81  int16_t SignalScaling(const int16_t* input, int input_length,
82                        const int16_t* expanded_signal,
83                        int16_t* expanded_max, int16_t* input_max) const;
84
85  // Downsamples |input| (|input_length| samples) and |expanded_signal| to
86  // 4 kHz sample rate. The downsampled signals are written to
87  // |input_downsampled_| and |expanded_downsampled_|, respectively.
88  void Downsample(const int16_t* input, int input_length,
89                  const int16_t* expanded_signal, int expanded_length);
90
91  // Calculates cross-correlation between |input_downsampled_| and
92  // |expanded_downsampled_|, and finds the correlation maximum. The maximizing
93  // lag is returned.
94  int16_t CorrelateAndPeakSearch(int16_t expanded_max, int16_t input_max,
95                                 int start_position, int input_length,
96                                 int expand_period) const;
97
98  const int fs_mult_;  // fs_hz_ / 8000.
99  const int timestamps_per_call_;
100  Expand* expand_;
101  SyncBuffer* sync_buffer_;
102  int16_t expanded_downsampled_[kExpandDownsampLength];
103  int16_t input_downsampled_[kInputDownsampLength];
104  AudioMultiVector expanded_;
105
106  DISALLOW_COPY_AND_ASSIGN(Merge);
107};
108
109}  // namespace webrtc
110#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MERGE_H_
111