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: 36 Merge(int fs_hz, 37 size_t num_channels, 38 Expand* expand, 39 SyncBuffer* sync_buffer); ~Merge()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