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,
37        size_t num_channels,
38        Expand* expand,
39        SyncBuffer* sync_buffer);
40  virtual ~Merge() {}
41
42  // The main method to produce the audio data. The decoded data is supplied in
43  // |input|, having |input_length| samples in total for all channels
44  // (interleaved). The result is written to |output|. The number of channels
45  // allocated in |output| defines the number of channels that will be used when
46  // de-interleaving |input|. The values in |external_mute_factor_array| (Q14)
47  // will be used to scale the audio, and is updated in the process. The array
48  // must have |num_channels_| elements.
49  virtual size_t Process(int16_t* input, size_t input_length,
50                         int16_t* external_mute_factor_array,
51                         AudioMultiVector* output);
52
53  virtual size_t RequiredFutureSamples();
54
55 protected:
56  const int fs_hz_;
57  const size_t num_channels_;
58
59 private:
60  static const int kMaxSampleRate = 48000;
61  static const size_t kExpandDownsampLength = 100;
62  static const size_t kInputDownsampLength = 40;
63  static const size_t kMaxCorrelationLength = 60;
64
65  // Calls |expand_| to get more expansion data to merge with. The data is
66  // written to |expanded_signal_|. Returns the length of the expanded data,
67  // while |expand_period| will be the number of samples in one expansion period
68  // (typically one pitch period). The value of |old_length| will be the number
69  // of samples that were taken from the |sync_buffer_|.
70  size_t GetExpandedSignal(size_t* old_length, size_t* expand_period);
71
72  // Analyzes |input| and |expanded_signal| to find maximum values. Returns
73  // a muting factor (Q14) to be used on the new data.
74  int16_t SignalScaling(const int16_t* input, size_t input_length,
75                        const int16_t* expanded_signal,
76                        int16_t* expanded_max, int16_t* input_max) const;
77
78  // Downsamples |input| (|input_length| samples) and |expanded_signal| to
79  // 4 kHz sample rate. The downsampled signals are written to
80  // |input_downsampled_| and |expanded_downsampled_|, respectively.
81  void Downsample(const int16_t* input, size_t input_length,
82                  const int16_t* expanded_signal, size_t expanded_length);
83
84  // Calculates cross-correlation between |input_downsampled_| and
85  // |expanded_downsampled_|, and finds the correlation maximum. The maximizing
86  // lag is returned.
87  size_t CorrelateAndPeakSearch(int16_t expanded_max, int16_t input_max,
88                                size_t start_position, size_t input_length,
89                                size_t expand_period) const;
90
91  const int fs_mult_;  // fs_hz_ / 8000.
92  const size_t timestamps_per_call_;
93  Expand* expand_;
94  SyncBuffer* sync_buffer_;
95  int16_t expanded_downsampled_[kExpandDownsampLength];
96  int16_t input_downsampled_[kInputDownsampLength];
97  AudioMultiVector expanded_;
98
99  RTC_DISALLOW_COPY_AND_ASSIGN(Merge);
100};
101
102}  // namespace webrtc
103#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_MERGE_H_
104