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 20 namespace webrtc { 21 22 // Forward declarations. 23 class Expand; 24 class 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. 34 class Merge { 35 public: Merge(int fs_hz,size_t num_channels,Expand * expand,SyncBuffer * sync_buffer)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 ~Merge()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