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 MODULES_AUDIO_CODING_NETEQ_MERGE_H_ 12 #define MODULES_AUDIO_CODING_NETEQ_MERGE_H_ 13 14 #include "modules/audio_coding/neteq/audio_multi_vector.h" 15 16 namespace webrtc { 17 18 // Forward declarations. 19 class Expand; 20 class SyncBuffer; 21 22 // This class handles the transition from expansion to normal operation. 23 // When a packet is not available for decoding when needed, the expand operation 24 // is called to generate extrapolation data. If the missing packet arrives, 25 // i.e., it was just delayed, it can be decoded and appended directly to the 26 // end of the expanded data (thanks to how the Expand class operates). However, 27 // if a later packet arrives instead, the loss is a fact, and the new data must 28 // be stitched together with the end of the expanded data. This stitching is 29 // what the Merge class does. 30 class Merge { 31 public: 32 Merge(int fs_hz, 33 size_t num_channels, 34 Expand* expand, 35 SyncBuffer* sync_buffer); 36 virtual ~Merge(); 37 38 Merge(const Merge&) = delete; 39 Merge& operator=(const Merge&) = delete; 40 41 // The main method to produce the audio data. The decoded data is supplied in 42 // `input`, having `input_length` samples in total for all channels 43 // (interleaved). The result is written to `output`. The number of channels 44 // allocated in `output` defines the number of channels that will be used when 45 // de-interleaving `input`. 46 virtual size_t Process(int16_t* input, 47 size_t input_length, 48 AudioMultiVector* output); 49 50 virtual size_t RequiredFutureSamples(); 51 52 protected: 53 const int fs_hz_; 54 const size_t num_channels_; 55 56 private: 57 static const int kMaxSampleRate = 48000; 58 static const size_t kExpandDownsampLength = 100; 59 static const size_t kInputDownsampLength = 40; 60 static const size_t kMaxCorrelationLength = 60; 61 62 // Calls `expand_` to get more expansion data to merge with. The data is 63 // written to `expanded_signal_`. Returns the length of the expanded data, 64 // while `expand_period` will be the number of samples in one expansion period 65 // (typically one pitch period). The value of `old_length` will be the number 66 // of samples that were taken from the `sync_buffer_`. 67 size_t GetExpandedSignal(size_t* old_length, size_t* expand_period); 68 69 // Analyzes `input` and `expanded_signal` and returns muting factor (Q14) to 70 // be used on the new data. 71 int16_t SignalScaling(const int16_t* input, 72 size_t input_length, 73 const int16_t* expanded_signal) const; 74 75 // Downsamples `input` (`input_length` samples) and `expanded_signal` to 76 // 4 kHz sample rate. The downsampled signals are written to 77 // `input_downsampled_` and `expanded_downsampled_`, respectively. 78 void Downsample(const int16_t* input, 79 size_t input_length, 80 const int16_t* expanded_signal, 81 size_t expanded_length); 82 83 // Calculates cross-correlation between `input_downsampled_` and 84 // `expanded_downsampled_`, and finds the correlation maximum. The maximizing 85 // lag is returned. 86 size_t CorrelateAndPeakSearch(size_t start_position, 87 size_t input_length, 88 size_t expand_period) const; 89 90 const int fs_mult_; // fs_hz_ / 8000. 91 const size_t timestamps_per_call_; 92 Expand* expand_; 93 SyncBuffer* sync_buffer_; 94 int16_t expanded_downsampled_[kExpandDownsampLength]; 95 int16_t input_downsampled_[kInputDownsampLength]; 96 AudioMultiVector expanded_; 97 std::vector<int16_t> temp_data_; 98 }; 99 100 } // namespace webrtc 101 #endif // MODULES_AUDIO_CODING_NETEQ_MERGE_H_ 102