Mercurial > repos > blastem
annotate render_audio.c @ 1971:80920c21bb52
Add an event log soft flush and call it twice per frame in between hard flushes to netplay latency when there are insufficient hardware updates to flush packets in the middle of a frame
author | Michael Pavone <pavone@retrodev.com> |
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date | Fri, 08 May 2020 11:40:30 -0700 |
parents | e4671a39d155 |
children | cfd53c94fffb |
rev | line source |
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1865
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1 #include <limits.h> |
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2 #include <string.h> |
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3 #include <stdlib.h> |
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4 #include <math.h> |
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5 #include "render_audio.h" |
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6 #include "util.h" |
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7 #include "config.h" |
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8 #include "blastem.h" |
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9 |
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10 static uint8_t output_channels; |
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11 static uint32_t buffer_samples, sample_rate; |
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12 |
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13 static audio_source *audio_sources[8]; |
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14 static audio_source *inactive_audio_sources[8]; |
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15 static uint8_t num_audio_sources; |
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16 static uint8_t num_inactive_audio_sources; |
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17 |
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18 static float overall_gain_mult, *mix_buf; |
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19 static int sample_size; |
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20 |
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21 typedef void (*conv_func)(float *samples, void *vstream, int sample_count); |
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22 |
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23 static void convert_null(float *samples, void *vstream, int sample_count) |
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24 { |
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25 memset(vstream, 0, sample_count * sample_size); |
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26 } |
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27 |
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28 static void convert_s16(float *samples, void *vstream, int sample_count) |
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29 { |
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30 int16_t *stream = vstream; |
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31 for (int16_t *end = stream + sample_count; stream < end; stream++, samples++) |
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32 { |
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33 float sample = *samples; |
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34 int16_t out_sample; |
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35 if (sample >= 1.0f) { |
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36 out_sample = 0x7FFF; |
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37 } else if (sample <= -1.0f) { |
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38 out_sample = -0x8000; |
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39 } else { |
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40 out_sample = sample * 0x7FFF; |
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41 } |
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42 *stream = out_sample; |
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43 } |
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44 } |
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45 |
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46 static void clamp_f32(float *samples, void *vstream, int sample_count) |
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47 { |
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48 for (; sample_count > 0; sample_count--, samples++) |
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49 { |
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50 float sample = *samples; |
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51 if (sample > 1.0f) { |
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52 sample = 1.0f; |
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53 } else if (sample < -1.0f) { |
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54 sample = -1.0f; |
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55 } |
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56 *samples = sample; |
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57 } |
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58 } |
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59 |
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60 static int32_t mix_f32(audio_source *audio, float *stream, int samples) |
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61 { |
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62 float *end = stream + samples; |
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63 int16_t *src = audio->front; |
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64 uint32_t i = audio->read_start; |
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65 uint32_t i_end = audio->read_end; |
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66 float *cur = stream; |
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67 float gain_mult = audio->gain_mult * overall_gain_mult; |
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68 size_t first_add = output_channels > 1 ? 1 : 0, second_add = output_channels > 1 ? output_channels - 1 : 1; |
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69 if (audio->num_channels == 1) { |
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70 while (cur < end && i != i_end) |
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71 { |
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72 *cur += gain_mult * ((float)src[i]) / 0x7FFF; |
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73 cur += first_add; |
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74 *cur += gain_mult * ((float)src[i++]) / 0x7FFF; |
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75 cur += second_add; |
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76 i &= audio->mask; |
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77 } |
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78 } else { |
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79 while(cur < end && i != i_end) |
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80 { |
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81 *cur += gain_mult * ((float)src[i++]) / 0x7FFF; |
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82 cur += first_add; |
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83 *cur += gain_mult * ((float)src[i++]) / 0x7FFF; |
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84 cur += second_add; |
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85 i &= audio->mask; |
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86 } |
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87 } |
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88 if (!render_is_audio_sync()) { |
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89 audio->read_start = i; |
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90 } |
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91 if (cur != end) { |
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92 debug_message("Underflow of %d samples, read_start: %d, read_end: %d, mask: %X\n", (int)(end-cur)/2, audio->read_start, audio->read_end, audio->mask); |
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93 return (cur-end)/2; |
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94 } else { |
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95 return ((i_end - i) & audio->mask) / audio->num_channels; |
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96 } |
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97 } |
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98 |
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99 static conv_func convert; |
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100 |
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101 |
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102 int mix_and_convert(unsigned char *byte_stream, int len, int *min_remaining_out) |
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103 { |
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104 int samples = len / sample_size; |
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105 float *mix_dest = mix_buf ? mix_buf : (float *)byte_stream; |
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106 memset(mix_dest, 0, samples * sizeof(float)); |
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107 int min_buffered = INT_MAX; |
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108 int min_remaining_buffer = INT_MAX; |
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109 for (uint8_t i = 0; i < num_audio_sources; i++) |
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110 { |
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111 int buffered = mix_f32(audio_sources[i], mix_dest, samples); |
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112 int remaining = (audio_sources[i]->mask + 1) / audio_sources[i]->num_channels - buffered; |
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113 min_buffered = buffered < min_buffered ? buffered : min_buffered; |
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114 min_remaining_buffer = remaining < min_remaining_buffer ? remaining : min_remaining_buffer; |
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115 audio_sources[i]->front_populated = 0; |
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116 render_buffer_consumed(audio_sources[i]); |
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117 } |
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118 convert(mix_dest, byte_stream, samples); |
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119 if (min_remaining_out) { |
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120 *min_remaining_out = min_remaining_buffer; |
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121 } |
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122 return min_buffered; |
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123 } |
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124 |
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125 uint8_t all_sources_ready(void) |
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126 { |
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127 uint8_t num_populated = 0; |
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128 num_populated = 0; |
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129 for (uint8_t i = 0; i < num_audio_sources; i++) |
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130 { |
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131 if (audio_sources[i]->front_populated) { |
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132 num_populated++; |
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133 } |
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134 } |
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135 return num_populated == num_audio_sources; |
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136 } |
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137 |
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138 #define BUFFER_INC_RES 0x40000000UL |
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139 |
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140 void render_audio_adjust_clock(audio_source *src, uint64_t master_clock, uint64_t sample_divider) |
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141 { |
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142 src->buffer_inc = ((BUFFER_INC_RES * (uint64_t)sample_rate) / master_clock) * sample_divider; |
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143 } |
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144 |
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145 void render_audio_adjust_speed(float adjust_ratio) |
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146 { |
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147 for (uint8_t i = 0; i < num_audio_sources; i++) |
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148 { |
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149 audio_sources[i]->buffer_inc = ((double)audio_sources[i]->buffer_inc) + ((double)audio_sources[i]->buffer_inc) * adjust_ratio + 0.5; |
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150 } |
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151 } |
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152 |
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153 audio_source *render_audio_source(uint64_t master_clock, uint64_t sample_divider, uint8_t channels) |
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154 { |
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155 audio_source *ret = NULL; |
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156 uint32_t alloc_size = render_is_audio_sync() ? channels * buffer_samples : nearest_pow2(render_min_buffered() * 4 * channels); |
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157 render_lock_audio(); |
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158 if (num_audio_sources < 8) { |
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159 ret = calloc(1, sizeof(audio_source)); |
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160 ret->back = malloc(alloc_size * sizeof(int16_t)); |
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161 ret->front = render_is_audio_sync() ? malloc(alloc_size * sizeof(int16_t)) : ret->back; |
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162 ret->front_populated = 0; |
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163 ret->opaque = render_new_audio_opaque(); |
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164 ret->num_channels = channels; |
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165 audio_sources[num_audio_sources++] = ret; |
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166 } |
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167 render_unlock_audio(); |
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168 if (!ret) { |
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169 fatal_error("Too many audio sources!"); |
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170 } else { |
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171 render_audio_adjust_clock(ret, master_clock, sample_divider); |
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172 double lowpass_cutoff = get_lowpass_cutoff(config); |
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173 double rc = (1.0 / lowpass_cutoff) / (2.0 * M_PI); |
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174 ret->dt = 1.0 / ((double)master_clock / (double)(sample_divider)); |
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175 double alpha = ret->dt / (ret->dt + rc); |
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176 ret->lowpass_alpha = (int32_t)(((double)0x10000) * alpha); |
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177 ret->buffer_pos = 0; |
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178 ret->buffer_fraction = 0; |
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179 ret->last_left = ret->last_right = 0; |
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180 ret->read_start = 0; |
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181 ret->read_end = render_is_audio_sync() ? buffer_samples * channels : 0; |
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182 ret->mask = render_is_audio_sync() ? 0xFFFFFFFF : alloc_size-1; |
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183 ret->gain_mult = 1.0f; |
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184 } |
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185 render_audio_created(ret); |
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186 |
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187 return ret; |
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188 } |
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189 |
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190 |
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191 static float db_to_mult(float gain) |
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192 { |
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193 return powf(10.0f, gain/20.0f); |
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194 } |
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195 |
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196 void render_audio_source_gaindb(audio_source *src, float gain) |
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197 { |
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198 src->gain_mult = db_to_mult(gain); |
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199 } |
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200 |
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201 void render_pause_source(audio_source *src) |
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202 { |
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203 uint8_t found = 0, remaining_sources; |
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204 render_lock_audio(); |
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205 for (uint8_t i = 0; i < num_audio_sources; i++) |
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206 { |
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207 if (audio_sources[i] == src) { |
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208 audio_sources[i] = audio_sources[--num_audio_sources]; |
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209 found = 1; |
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210 remaining_sources = num_audio_sources; |
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211 break; |
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212 } |
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213 } |
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214 |
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215 render_unlock_audio(); |
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216 if (found) { |
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217 render_source_paused(src, remaining_sources); |
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218 } |
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219 inactive_audio_sources[num_inactive_audio_sources++] = src; |
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220 } |
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221 |
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222 void render_resume_source(audio_source *src) |
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223 { |
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224 render_lock_audio(); |
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225 if (num_audio_sources < 8) { |
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226 audio_sources[num_audio_sources++] = src; |
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227 } |
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228 render_unlock_audio(); |
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229 for (uint8_t i = 0; i < num_inactive_audio_sources; i++) |
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230 { |
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231 if (inactive_audio_sources[i] == src) { |
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232 inactive_audio_sources[i] = inactive_audio_sources[--num_inactive_audio_sources]; |
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233 } |
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234 } |
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235 render_source_resumed(src); |
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236 } |
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237 |
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238 void render_free_source(audio_source *src) |
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239 { |
1870
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240 uint8_t found = 0; |
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241 for (uint8_t i = 0; i < num_inactive_audio_sources; i++) |
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242 { |
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243 if (inactive_audio_sources[i] == src) { |
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244 inactive_audio_sources[i] = inactive_audio_sources[--num_inactive_audio_sources]; |
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245 found = 1; |
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246 break; |
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247 } |
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248 } |
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249 if (!found) { |
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250 render_pause_source(src); |
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251 num_inactive_audio_sources--; |
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252 } |
1865
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253 |
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254 free(src->front); |
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255 if (render_is_audio_sync()) { |
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256 free(src->back); |
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257 render_free_audio_opaque(src->opaque); |
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258 } |
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259 free(src); |
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260 } |
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261 |
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262 static int16_t lowpass_sample(audio_source *src, int16_t last, int16_t current) |
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263 { |
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264 int32_t tmp = current * src->lowpass_alpha + last * (0x10000 - src->lowpass_alpha); |
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265 current = tmp >> 16; |
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266 return current; |
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267 } |
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268 |
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269 static void interp_sample(audio_source *src, int16_t last, int16_t current) |
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270 { |
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271 int64_t tmp = last * ((src->buffer_fraction << 16) / src->buffer_inc); |
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272 tmp += current * (0x10000 - ((src->buffer_fraction << 16) / src->buffer_inc)); |
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273 src->back[src->buffer_pos++] = tmp >> 16; |
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274 } |
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275 |
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276 static uint32_t sync_samples; |
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277 void render_put_mono_sample(audio_source *src, int16_t value) |
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278 { |
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279 value = lowpass_sample(src, src->last_left, value); |
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280 src->buffer_fraction += src->buffer_inc; |
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281 uint32_t base = render_is_audio_sync() ? 0 : src->read_end; |
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282 while (src->buffer_fraction > BUFFER_INC_RES) |
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283 { |
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284 src->buffer_fraction -= BUFFER_INC_RES; |
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285 interp_sample(src, src->last_left, value); |
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286 |
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287 if (((src->buffer_pos - base) & src->mask) >= sync_samples) { |
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288 render_do_audio_ready(src); |
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289 } |
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290 src->buffer_pos &= src->mask; |
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291 } |
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292 src->last_left = value; |
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293 } |
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294 |
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295 void render_put_stereo_sample(audio_source *src, int16_t left, int16_t right) |
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296 { |
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297 left = lowpass_sample(src, src->last_left, left); |
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298 right = lowpass_sample(src, src->last_right, right); |
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299 src->buffer_fraction += src->buffer_inc; |
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300 uint32_t base = render_is_audio_sync() ? 0 : src->read_end; |
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301 while (src->buffer_fraction > BUFFER_INC_RES) |
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302 { |
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303 src->buffer_fraction -= BUFFER_INC_RES; |
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304 |
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305 interp_sample(src, src->last_left, left); |
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306 interp_sample(src, src->last_right, right); |
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307 |
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308 if (((src->buffer_pos - base) & src->mask)/2 >= sync_samples) { |
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309 render_do_audio_ready(src); |
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310 } |
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311 src->buffer_pos &= src->mask; |
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312 } |
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313 src->last_left = left; |
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314 src->last_right = right; |
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315 } |
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316 |
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317 static void update_source(audio_source *src, double rc, uint8_t sync_changed) |
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318 { |
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319 double alpha = src->dt / (src->dt + rc); |
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320 int32_t lowpass_alpha = (int32_t)(((double)0x10000) * alpha); |
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321 src->lowpass_alpha = lowpass_alpha; |
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322 if (sync_changed) { |
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323 uint32_t alloc_size = render_is_audio_sync() ? src->num_channels * buffer_samples : nearest_pow2(render_min_buffered() * 4 * src->num_channels); |
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324 src->back = realloc(src->back, alloc_size * sizeof(int16_t)); |
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325 if (render_is_audio_sync()) { |
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326 src->front = malloc(alloc_size * sizeof(int16_t)); |
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327 } else { |
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328 free(src->front); |
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329 src->front = src->back; |
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330 } |
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331 src->mask = render_is_audio_sync() ? 0xFFFFFFFF : alloc_size-1; |
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332 src->read_start = 0; |
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333 src->read_end = render_is_audio_sync() ? buffer_samples * src->num_channels : 0; |
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334 src->buffer_pos = 0; |
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335 } |
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336 } |
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337 |
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338 uint8_t old_audio_sync; |
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339 void render_audio_initialized(render_audio_format format, uint32_t rate, uint8_t channels, uint32_t buffer_size, int sample_size_in) |
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340 { |
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341 sample_rate = rate; |
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342 output_channels = channels; |
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343 buffer_samples = buffer_size; |
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344 sample_size = sample_size_in; |
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345 if (mix_buf) { |
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346 free(mix_buf); |
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347 mix_buf = NULL; |
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348 } |
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349 switch(format) |
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350 { |
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351 case RENDER_AUDIO_S16: |
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352 convert = convert_s16; |
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353 mix_buf = calloc(output_channels * buffer_samples, sizeof(float)); |
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354 break; |
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355 case RENDER_AUDIO_FLOAT: |
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356 convert = clamp_f32; |
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357 break; |
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358 case RENDER_AUDIO_UNKNOWN: |
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359 convert = convert_null; |
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360 mix_buf = calloc(output_channels * buffer_samples, sizeof(float)); |
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361 break; |
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362 } |
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363 uint32_t syncs = render_audio_syncs_per_sec(); |
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364 if (syncs) { |
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365 sync_samples = rate / syncs; |
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366 } else { |
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367 sync_samples = buffer_samples; |
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368 } |
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369 char * gain_str = tern_find_path(config, "audio\0gain\0", TVAL_PTR).ptrval; |
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370 overall_gain_mult = db_to_mult(gain_str ? atof(gain_str) : 0.0f); |
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371 uint8_t sync_changed = old_audio_sync != render_is_audio_sync(); |
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372 old_audio_sync = render_is_audio_sync(); |
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373 double lowpass_cutoff = get_lowpass_cutoff(config); |
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374 double rc = (1.0 / lowpass_cutoff) / (2.0 * M_PI); |
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375 render_lock_audio(); |
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376 for (uint8_t i = 0; i < num_audio_sources; i++) |
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377 { |
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378 update_source(audio_sources[i], rc, sync_changed); |
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379 } |
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380 render_unlock_audio(); |
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381 for (uint8_t i = 0; i < num_inactive_audio_sources; i++) |
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382 { |
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383 update_source(inactive_audio_sources[i], rc, sync_changed); |
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384 } |
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385 } |