Mercurial > repos > blastem
annotate render_audio.c @ 2237:f82c090c1e89
Implement MMAP_ONLY_ODD/EVEN in combination with MMAP_PTR_IDX. Fixes games that have SRAM when a system with TMSS is selected
author | Michael Pavone <pavone@retrodev.com> |
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date | Sat, 17 Sep 2022 15:38:40 -0700 |
parents | 46ee354f29bd |
children | b3832f73444f |
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 uint8_t audio_deadlock_hack(void) |
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139 { |
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140 uint32_t min_buffer_pos = 0xFFFFFFFFU; |
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141 for (uint8_t i = 0; i < num_audio_sources; i++) |
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142 { |
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143 if (audio_sources[i]->front_populated) { |
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144 uint32_t buffer_pos = audio_sources[i]->buffer_pos; |
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145 if (audio_sources[i]->num_channels == 1) { |
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146 buffer_pos *= 2; |
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147 } |
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148 if (buffer_pos < min_buffer_pos) { |
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149 min_buffer_pos = buffer_pos; |
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150 } |
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151 } |
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152 } |
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153 uint8_t do_signal = 0; |
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154 for (uint8_t i = 0; i < num_audio_sources; i++) |
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155 { |
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156 if (!audio_sources[i]->front_populated) { |
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157 audio_sources[i]->front_populated = 1; |
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158 int16_t *tmp = audio_sources[i]->front; |
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159 audio_sources[i]->front = audio_sources[i]->back; |
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160 audio_sources[i]->back = tmp; |
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161 if (audio_sources[i]->num_channels == 2) { |
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162 audio_sources[i]->buffer_pos = min_buffer_pos; |
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163 } else { |
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164 audio_sources[i]->buffer_pos = min_buffer_pos / 2; |
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165 } |
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166 do_signal = 1; |
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167 } |
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168 } |
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169 return do_signal; |
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170 } |
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171 |
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172 #define BUFFER_INC_RES 0x40000000UL |
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173 |
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174 void render_audio_adjust_clock(audio_source *src, uint64_t master_clock, uint64_t sample_divider) |
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175 { |
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176 src->buffer_inc = ((BUFFER_INC_RES * sample_divider * (uint64_t)sample_rate) / master_clock); |
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177 } |
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178 |
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179 void render_audio_adjust_speed(float adjust_ratio) |
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180 { |
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181 for (uint8_t i = 0; i < num_audio_sources; i++) |
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182 { |
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183 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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184 } |
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185 } |
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186 |
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187 audio_source *render_audio_source(const char *name, uint64_t master_clock, uint64_t sample_divider, uint8_t channels) |
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188 { |
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189 audio_source *ret = NULL; |
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190 uint32_t alloc_size = render_is_audio_sync() ? channels * buffer_samples : nearest_pow2(render_min_buffered() * 4 * channels); |
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191 render_lock_audio(); |
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192 if (num_audio_sources < 8) { |
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193 ret = calloc(1, sizeof(audio_source)); |
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194 ret->name = name; |
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195 ret->back = malloc(alloc_size * sizeof(int16_t)); |
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196 ret->front = render_is_audio_sync() ? malloc(alloc_size * sizeof(int16_t)) : ret->back; |
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197 ret->front_populated = 0; |
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198 ret->opaque = render_new_audio_opaque(); |
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199 ret->num_channels = channels; |
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200 audio_sources[num_audio_sources++] = ret; |
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201 } |
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202 render_unlock_audio(); |
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203 if (!ret) { |
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204 fatal_error("Too many audio sources!"); |
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205 } else { |
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206 render_audio_adjust_clock(ret, master_clock, sample_divider); |
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207 double lowpass_cutoff = get_lowpass_cutoff(config); |
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208 double rc = (1.0 / lowpass_cutoff) / (2.0 * M_PI); |
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209 ret->dt = 1.0 / ((double)master_clock / (double)(sample_divider)); |
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210 double alpha = ret->dt / (ret->dt + rc); |
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211 ret->lowpass_alpha = (int32_t)(((double)0x10000) * alpha); |
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212 ret->buffer_pos = 0; |
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213 ret->buffer_fraction = 0; |
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214 ret->last_left = ret->last_right = 0; |
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215 ret->read_start = 0; |
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216 ret->read_end = render_is_audio_sync() ? buffer_samples * channels : 0; |
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217 ret->mask = render_is_audio_sync() ? 0xFFFFFFFF : alloc_size-1; |
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218 ret->gain_mult = 1.0f; |
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219 } |
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220 render_audio_created(ret); |
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221 |
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222 return ret; |
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223 } |
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224 |
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225 |
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226 static float db_to_mult(float gain) |
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227 { |
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228 return powf(10.0f, gain/20.0f); |
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229 } |
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230 |
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231 void render_audio_source_gaindb(audio_source *src, float gain) |
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232 { |
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233 src->gain_mult = db_to_mult(gain); |
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234 } |
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235 |
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236 void render_pause_source(audio_source *src) |
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237 { |
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238 uint8_t found = 0, remaining_sources; |
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239 render_lock_audio(); |
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240 for (uint8_t i = 0; i < num_audio_sources; i++) |
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241 { |
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242 if (audio_sources[i] == src) { |
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243 audio_sources[i] = audio_sources[--num_audio_sources]; |
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244 found = 1; |
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245 remaining_sources = num_audio_sources; |
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246 break; |
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247 } |
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248 } |
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Initial stab at RF5C164 emulation
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249 |
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250 render_unlock_audio(); |
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251 if (found) { |
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252 render_source_paused(src, remaining_sources); |
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253 } |
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254 inactive_audio_sources[num_inactive_audio_sources++] = src; |
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255 } |
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256 |
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257 void render_resume_source(audio_source *src) |
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258 { |
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259 render_lock_audio(); |
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260 if (num_audio_sources < 8) { |
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261 audio_sources[num_audio_sources++] = src; |
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262 } |
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263 render_unlock_audio(); |
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264 for (uint8_t i = 0; i < num_inactive_audio_sources; i++) |
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265 { |
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266 if (inactive_audio_sources[i] == src) { |
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267 inactive_audio_sources[i] = inactive_audio_sources[--num_inactive_audio_sources]; |
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268 } |
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269 } |
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270 render_source_resumed(src); |
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271 } |
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272 |
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273 void render_free_source(audio_source *src) |
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274 { |
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275 uint8_t found = 0; |
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276 for (uint8_t i = 0; i < num_inactive_audio_sources; i++) |
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277 { |
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278 if (inactive_audio_sources[i] == src) { |
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279 inactive_audio_sources[i] = inactive_audio_sources[--num_inactive_audio_sources]; |
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280 found = 1; |
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281 break; |
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282 } |
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283 } |
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284 if (!found) { |
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285 render_pause_source(src); |
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286 num_inactive_audio_sources--; |
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287 } |
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cfd53c94fffb
Initial stab at RF5C164 emulation
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288 |
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289 free(src->front); |
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290 if (render_is_audio_sync()) { |
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291 free(src->back); |
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292 render_free_audio_opaque(src->opaque); |
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293 } |
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294 free(src); |
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295 } |
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296 |
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297 static int16_t lowpass_sample(audio_source *src, int16_t last, int16_t current) |
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298 { |
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299 int32_t tmp = current * src->lowpass_alpha + last * (0x10000 - src->lowpass_alpha); |
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300 current = tmp >> 16; |
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301 return current; |
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302 } |
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303 |
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304 static void interp_sample(audio_source *src, int16_t last, int16_t current) |
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305 { |
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306 int64_t tmp = last * ((src->buffer_fraction << 16) / src->buffer_inc); |
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307 tmp += current * (0x10000 - ((src->buffer_fraction << 16) / src->buffer_inc)); |
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308 src->back[src->buffer_pos++] = tmp >> 16; |
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309 } |
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310 |
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311 static uint32_t sync_samples; |
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312 void render_put_mono_sample(audio_source *src, int16_t value) |
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313 { |
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314 value = lowpass_sample(src, src->last_left, value); |
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315 src->buffer_fraction += src->buffer_inc; |
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316 uint32_t base = render_is_audio_sync() ? 0 : src->read_end; |
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317 while (src->buffer_fraction > BUFFER_INC_RES) |
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318 { |
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319 src->buffer_fraction -= BUFFER_INC_RES; |
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320 interp_sample(src, src->last_left, value); |
2081
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Initial stab at RF5C164 emulation
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321 |
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322 if (((src->buffer_pos - base) & src->mask) >= sync_samples) { |
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|
323 render_do_audio_ready(src); |
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324 } |
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|
325 src->buffer_pos &= src->mask; |
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326 } |
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Split generic part of audio code into a separate file so it can be used in other targets besides SDL
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327 src->last_left = value; |
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328 } |
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329 |
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330 void render_put_stereo_sample(audio_source *src, int16_t left, int16_t right) |
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331 { |
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332 left = lowpass_sample(src, src->last_left, left); |
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333 right = lowpass_sample(src, src->last_right, right); |
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334 src->buffer_fraction += src->buffer_inc; |
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335 uint32_t base = render_is_audio_sync() ? 0 : src->read_end; |
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336 while (src->buffer_fraction > BUFFER_INC_RES) |
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337 { |
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338 src->buffer_fraction -= BUFFER_INC_RES; |
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Initial stab at RF5C164 emulation
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339 |
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340 interp_sample(src, src->last_left, left); |
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341 interp_sample(src, src->last_right, right); |
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342 |
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343 if (((src->buffer_pos - base) & src->mask)/2 >= sync_samples) { |
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344 render_do_audio_ready(src); |
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345 } |
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346 src->buffer_pos &= src->mask; |
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347 } |
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348 src->last_left = left; |
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349 src->last_right = right; |
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350 } |
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351 |
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352 static void update_source(audio_source *src, double rc, uint8_t sync_changed) |
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353 { |
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354 double alpha = src->dt / (src->dt + rc); |
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355 int32_t lowpass_alpha = (int32_t)(((double)0x10000) * alpha); |
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356 src->lowpass_alpha = lowpass_alpha; |
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357 if (sync_changed) { |
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358 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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359 src->back = realloc(src->back, alloc_size * sizeof(int16_t)); |
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360 if (render_is_audio_sync()) { |
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361 src->front = malloc(alloc_size * sizeof(int16_t)); |
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362 } else { |
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363 free(src->front); |
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364 src->front = src->back; |
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365 } |
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366 src->mask = render_is_audio_sync() ? 0xFFFFFFFF : alloc_size-1; |
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367 src->read_start = 0; |
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368 src->read_end = render_is_audio_sync() ? buffer_samples * src->num_channels : 0; |
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369 src->buffer_pos = 0; |
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370 } |
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371 } |
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372 |
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373 uint8_t old_audio_sync; |
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374 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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375 { |
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376 sample_rate = rate; |
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377 output_channels = channels; |
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378 buffer_samples = buffer_size; |
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379 sample_size = sample_size_in; |
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380 if (mix_buf) { |
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381 free(mix_buf); |
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382 mix_buf = NULL; |
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383 } |
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384 switch(format) |
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385 { |
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386 case RENDER_AUDIO_S16: |
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387 convert = convert_s16; |
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388 mix_buf = calloc(output_channels * buffer_samples, sizeof(float)); |
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389 break; |
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390 case RENDER_AUDIO_FLOAT: |
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391 convert = clamp_f32; |
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392 break; |
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393 case RENDER_AUDIO_UNKNOWN: |
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394 convert = convert_null; |
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395 mix_buf = calloc(output_channels * buffer_samples, sizeof(float)); |
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396 break; |
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397 } |
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398 uint32_t syncs = render_audio_syncs_per_sec(); |
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399 if (syncs) { |
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400 sync_samples = rate / syncs; |
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401 } else { |
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402 sync_samples = buffer_samples; |
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403 } |
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Split generic part of audio code into a separate file so it can be used in other targets besides SDL
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404 char * gain_str = tern_find_path(config, "audio\0gain\0", TVAL_PTR).ptrval; |
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405 overall_gain_mult = db_to_mult(gain_str ? atof(gain_str) : 0.0f); |
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406 uint8_t sync_changed = old_audio_sync != render_is_audio_sync(); |
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407 old_audio_sync = render_is_audio_sync(); |
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408 double lowpass_cutoff = get_lowpass_cutoff(config); |
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409 double rc = (1.0 / lowpass_cutoff) / (2.0 * M_PI); |
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|
410 render_lock_audio(); |
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411 for (uint8_t i = 0; i < num_audio_sources; i++) |
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|
412 { |
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Split generic part of audio code into a separate file so it can be used in other targets besides SDL
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|
413 update_source(audio_sources[i], rc, sync_changed); |
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|
414 } |
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Split generic part of audio code into a separate file so it can be used in other targets besides SDL
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|
415 render_unlock_audio(); |
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416 for (uint8_t i = 0; i < num_inactive_audio_sources; i++) |
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417 { |
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|
418 update_source(inactive_audio_sources[i], rc, sync_changed); |
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Split generic part of audio code into a separate file so it can be used in other targets besides SDL
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|
419 } |
2081
cfd53c94fffb
Initial stab at RF5C164 emulation
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|
420 } |