henrik.lundin@webrtc.org | 9a40081 | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 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_NETEQ4_EXPAND_H_ |
| 12 | #define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_EXPAND_H_ |
| 13 | |
| 14 | #include <assert.h> |
| 15 | |
| 16 | #include "webrtc/modules/audio_coding/neteq4/audio_multi_vector.h" |
| 17 | #include "webrtc/system_wrappers/interface/constructor_magic.h" |
| 18 | #include "webrtc/system_wrappers/interface/scoped_ptr.h" |
| 19 | #include "webrtc/typedefs.h" |
| 20 | |
| 21 | namespace webrtc { |
| 22 | |
| 23 | // Forward declarations. |
| 24 | class BackgroundNoise; |
| 25 | class RandomVector; |
| 26 | class SyncBuffer; |
| 27 | |
| 28 | // This class handles extrapolation of audio data from the sync_buffer to |
| 29 | // produce packet-loss concealment. |
| 30 | // TODO(hlundin): Refactor this class to divide the long methods into shorter |
| 31 | // ones. |
| 32 | class Expand { |
| 33 | public: |
| 34 | Expand(BackgroundNoise* background_noise, |
| 35 | SyncBuffer* sync_buffer, |
| 36 | RandomVector* random_vector, |
| 37 | int fs, |
| 38 | size_t num_channels) |
| 39 | : background_noise_(background_noise), |
| 40 | sync_buffer_(sync_buffer), |
| 41 | random_vector_(random_vector), |
| 42 | first_expand_(true), |
| 43 | fs_hz_(fs), |
| 44 | num_channels_(num_channels), |
| 45 | overlap_length_(5 * fs / 8000), |
| 46 | lag_index_direction_(0), |
| 47 | current_lag_index_(0), |
| 48 | stop_muting_(false), |
| 49 | channel_parameters_(new ChannelParameters[num_channels_]) { |
| 50 | assert(fs == 8000 || fs == 16000 || fs == 32000 || fs == 48000); |
| 51 | assert(fs <= kMaxSampleRate); // Should not be possible. |
| 52 | assert(num_channels_ > 0); |
| 53 | memset(expand_lags_, 0, sizeof(expand_lags_)); |
| 54 | Reset(); |
| 55 | } |
| 56 | |
| 57 | virtual ~Expand() {} |
| 58 | |
| 59 | // Resets the object. |
| 60 | void Reset(); |
| 61 | |
| 62 | // The main method to produce concealment data. The data is appended to the |
| 63 | // end of |output|. |
henrik.lundin@webrtc.org | 0e9c399 | 2013-09-30 20:38:44 +0000 | [diff] [blame^] | 64 | int Process(AudioMultiVector* output); |
henrik.lundin@webrtc.org | 9a40081 | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 65 | |
| 66 | // Prepare the object to do extra expansion during normal operation following |
| 67 | // a period of expands. |
| 68 | void SetParametersForNormalAfterExpand(); |
| 69 | |
| 70 | // Prepare the object to do extra expansion during merge operation following |
| 71 | // a period of expands. |
| 72 | void SetParametersForMergeAfterExpand(); |
| 73 | |
| 74 | // Sets the mute factor for |channel| to |value|. |
| 75 | void SetMuteFactor(int16_t value, size_t channel) { |
| 76 | assert(channel < num_channels_); |
| 77 | channel_parameters_[channel].mute_factor = value; |
| 78 | } |
| 79 | |
| 80 | // Returns the mute factor for |channel|. |
| 81 | int16_t MuteFactor(size_t channel) { |
| 82 | assert(channel < num_channels_); |
| 83 | return channel_parameters_[channel].mute_factor; |
| 84 | } |
| 85 | |
| 86 | // Accessors and mutators. |
| 87 | size_t overlap_length() const { return overlap_length_; } |
| 88 | int16_t max_lag() const { return max_lag_; } |
| 89 | |
| 90 | private: |
| 91 | static const int kUnvoicedLpcOrder = 6; |
| 92 | static const int kNumCorrelationCandidates = 3; |
| 93 | static const int kDistortionLength = 20; |
| 94 | static const int kLpcAnalysisLength = 160; |
| 95 | static const int kMaxSampleRate = 48000; |
| 96 | static const int kNumLags = 3; |
| 97 | static const int kMaxConsecutiveExpands = 200; |
| 98 | |
| 99 | struct ChannelParameters { |
| 100 | // Constructor. |
| 101 | ChannelParameters() |
| 102 | : mute_factor(16384), |
| 103 | ar_gain(0), |
| 104 | ar_gain_scale(0), |
| 105 | voice_mix_factor(0), |
| 106 | current_voice_mix_factor(0), |
| 107 | onset(false), |
| 108 | mute_slope(0) { |
| 109 | memset(ar_filter, 0, sizeof(ar_filter)); |
| 110 | memset(ar_filter_state, 0, sizeof(ar_filter_state)); |
| 111 | } |
| 112 | int16_t mute_factor; |
| 113 | int16_t ar_filter[kUnvoicedLpcOrder + 1]; |
| 114 | int16_t ar_filter_state[kUnvoicedLpcOrder]; |
| 115 | int16_t ar_gain; |
| 116 | int16_t ar_gain_scale; |
| 117 | int16_t voice_mix_factor; /* Q14 */ |
| 118 | int16_t current_voice_mix_factor; /* Q14 */ |
| 119 | AudioVector<int16_t> expand_vector0; |
| 120 | AudioVector<int16_t> expand_vector1; |
| 121 | bool onset; |
| 122 | int16_t mute_slope; /* Q20 */ |
| 123 | }; |
| 124 | |
| 125 | // Analyze the signal history in |sync_buffer_|, and set up all parameters |
| 126 | // necessary to produce concealment data. |
| 127 | void AnalyzeSignal(int16_t* random_vector); |
| 128 | |
| 129 | // Calculate the auto-correlation of |input|, with length |input_length| |
| 130 | // samples. The correlation is calculated from a downsampled version of |
| 131 | // |input|, and is written to |output|. The scale factor is written to |
| 132 | // |output_scale|. Returns the length of the correlation vector. |
turaj@webrtc.org | 045e45e | 2013-09-20 16:25:28 +0000 | [diff] [blame] | 133 | int16_t Correlation(const int16_t* input, size_t input_length, |
henrik.lundin@webrtc.org | 9a40081 | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 134 | int16_t* output, int16_t* output_scale) const; |
| 135 | |
| 136 | void UpdateLagIndex(); |
| 137 | |
| 138 | BackgroundNoise* background_noise_; |
| 139 | SyncBuffer* sync_buffer_; |
| 140 | RandomVector* random_vector_; |
| 141 | bool first_expand_; |
| 142 | int fs_hz_; |
| 143 | size_t num_channels_; |
| 144 | size_t overlap_length_; |
| 145 | int consecutive_expands_; |
| 146 | int16_t max_lag_; |
| 147 | size_t expand_lags_[kNumLags]; |
| 148 | int lag_index_direction_; |
| 149 | int current_lag_index_; |
| 150 | bool stop_muting_; |
| 151 | scoped_array<ChannelParameters> channel_parameters_; |
| 152 | |
| 153 | DISALLOW_COPY_AND_ASSIGN(Expand); |
| 154 | }; |
| 155 | |
| 156 | } // namespace webrtc |
| 157 | #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_EXPAND_H_ |