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4349 | Serge | 1 | /* |
2 | * Copyright (c) 2012 Justin Ruggles |
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3 | * |
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4 | * This file is part of FFmpeg. |
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5 | * |
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6 | * FFmpeg is free software; you can redistribute it and/or |
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7 | * modify it under the terms of the GNU Lesser General Public |
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8 | * License as published by the Free Software Foundation; either |
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9 | * version 2.1 of the License, or (at your option) any later version. |
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10 | * |
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11 | * FFmpeg is distributed in the hope that it will be useful, |
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12 | * but WITHOUT ANY WARRANTY; without even the implied warranty of |
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13 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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14 | * Lesser General Public License for more details. |
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15 | * |
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16 | * You should have received a copy of the GNU Lesser General Public |
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17 | * License along with FFmpeg; if not, write to the Free Software |
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18 | * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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19 | */ |
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20 | |||
21 | #include |
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22 | #include |
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23 | |||
24 | #include "libavutil/mem.h" |
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25 | #include "audio_data.h" |
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26 | |||
27 | static const AVClass audio_data_class = { |
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28 | .class_name = "AudioData", |
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29 | .item_name = av_default_item_name, |
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30 | .version = LIBAVUTIL_VERSION_INT, |
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31 | }; |
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32 | |||
33 | /* |
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34 | * Calculate alignment for data pointers. |
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35 | */ |
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36 | static void calc_ptr_alignment(AudioData *a) |
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37 | { |
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38 | int p; |
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39 | int min_align = 128; |
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40 | |||
41 | for (p = 0; p < a->planes; p++) { |
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42 | int cur_align = 128; |
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43 | while ((intptr_t)a->data[p] % cur_align) |
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44 | cur_align >>= 1; |
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45 | if (cur_align < min_align) |
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46 | min_align = cur_align; |
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47 | } |
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48 | a->ptr_align = min_align; |
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49 | } |
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50 | |||
51 | int ff_audio_data_set_channels(AudioData *a, int channels) |
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52 | { |
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53 | if (channels < 1 || channels > AVRESAMPLE_MAX_CHANNELS || |
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54 | channels > a->allocated_channels) |
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55 | return AVERROR(EINVAL); |
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56 | |||
57 | a->channels = channels; |
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58 | a->planes = a->is_planar ? channels : 1; |
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59 | |||
60 | calc_ptr_alignment(a); |
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61 | |||
62 | return 0; |
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63 | } |
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64 | |||
65 | int ff_audio_data_init(AudioData *a, uint8_t **src, int plane_size, int channels, |
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66 | int nb_samples, enum AVSampleFormat sample_fmt, |
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67 | int read_only, const char *name) |
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68 | { |
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69 | int p; |
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70 | |||
71 | memset(a, 0, sizeof(*a)); |
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72 | a->class = &audio_data_class; |
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73 | |||
74 | if (channels < 1 || channels > AVRESAMPLE_MAX_CHANNELS) { |
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75 | av_log(a, AV_LOG_ERROR, "invalid channel count: %d\n", channels); |
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76 | return AVERROR(EINVAL); |
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77 | } |
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78 | |||
79 | a->sample_size = av_get_bytes_per_sample(sample_fmt); |
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80 | if (!a->sample_size) { |
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81 | av_log(a, AV_LOG_ERROR, "invalid sample format\n"); |
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82 | return AVERROR(EINVAL); |
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83 | } |
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84 | a->is_planar = av_sample_fmt_is_planar(sample_fmt); |
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85 | a->planes = a->is_planar ? channels : 1; |
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86 | a->stride = a->sample_size * (a->is_planar ? 1 : channels); |
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87 | |||
88 | for (p = 0; p < (a->is_planar ? channels : 1); p++) { |
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89 | if (!src[p]) { |
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90 | av_log(a, AV_LOG_ERROR, "invalid NULL pointer for src[%d]\n", p); |
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91 | return AVERROR(EINVAL); |
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92 | } |
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93 | a->data[p] = src[p]; |
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94 | } |
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95 | a->allocated_samples = nb_samples * !read_only; |
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96 | a->nb_samples = nb_samples; |
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97 | a->sample_fmt = sample_fmt; |
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98 | a->channels = channels; |
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99 | a->allocated_channels = channels; |
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100 | a->read_only = read_only; |
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101 | a->allow_realloc = 0; |
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102 | a->name = name ? name : "{no name}"; |
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103 | |||
104 | calc_ptr_alignment(a); |
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105 | a->samples_align = plane_size / a->stride; |
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106 | |||
107 | return 0; |
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108 | } |
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109 | |||
110 | AudioData *ff_audio_data_alloc(int channels, int nb_samples, |
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111 | enum AVSampleFormat sample_fmt, const char *name) |
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112 | { |
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113 | AudioData *a; |
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114 | int ret; |
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115 | |||
116 | if (channels < 1 || channels > AVRESAMPLE_MAX_CHANNELS) |
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117 | return NULL; |
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118 | |||
119 | a = av_mallocz(sizeof(*a)); |
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120 | if (!a) |
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121 | return NULL; |
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122 | |||
123 | a->sample_size = av_get_bytes_per_sample(sample_fmt); |
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124 | if (!a->sample_size) { |
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125 | av_free(a); |
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126 | return NULL; |
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127 | } |
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128 | a->is_planar = av_sample_fmt_is_planar(sample_fmt); |
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129 | a->planes = a->is_planar ? channels : 1; |
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130 | a->stride = a->sample_size * (a->is_planar ? 1 : channels); |
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131 | |||
132 | a->class = &audio_data_class; |
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133 | a->sample_fmt = sample_fmt; |
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134 | a->channels = channels; |
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135 | a->allocated_channels = channels; |
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136 | a->read_only = 0; |
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137 | a->allow_realloc = 1; |
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138 | a->name = name ? name : "{no name}"; |
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139 | |||
140 | if (nb_samples > 0) { |
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141 | ret = ff_audio_data_realloc(a, nb_samples); |
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142 | if (ret < 0) { |
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143 | av_free(a); |
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144 | return NULL; |
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145 | } |
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146 | return a; |
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147 | } else { |
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148 | calc_ptr_alignment(a); |
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149 | return a; |
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150 | } |
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151 | } |
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152 | |||
153 | int ff_audio_data_realloc(AudioData *a, int nb_samples) |
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154 | { |
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155 | int ret, new_buf_size, plane_size, p; |
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156 | |||
157 | /* check if buffer is already large enough */ |
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158 | if (a->allocated_samples >= nb_samples) |
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159 | return 0; |
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160 | |||
161 | /* validate that the output is not read-only and realloc is allowed */ |
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162 | if (a->read_only || !a->allow_realloc) |
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163 | return AVERROR(EINVAL); |
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164 | |||
165 | new_buf_size = av_samples_get_buffer_size(&plane_size, |
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166 | a->allocated_channels, nb_samples, |
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167 | a->sample_fmt, 0); |
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168 | if (new_buf_size < 0) |
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169 | return new_buf_size; |
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170 | |||
171 | /* if there is already data in the buffer and the sample format is planar, |
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172 | allocate a new buffer and copy the data, otherwise just realloc the |
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173 | internal buffer and set new data pointers */ |
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174 | if (a->nb_samples > 0 && a->is_planar) { |
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175 | uint8_t *new_data[AVRESAMPLE_MAX_CHANNELS] = { NULL }; |
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176 | |||
177 | ret = av_samples_alloc(new_data, &plane_size, a->allocated_channels, |
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178 | nb_samples, a->sample_fmt, 0); |
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179 | if (ret < 0) |
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180 | return ret; |
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181 | |||
182 | for (p = 0; p < a->planes; p++) |
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183 | memcpy(new_data[p], a->data[p], a->nb_samples * a->stride); |
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184 | |||
185 | av_freep(&a->buffer); |
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186 | memcpy(a->data, new_data, sizeof(new_data)); |
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187 | a->buffer = a->data[0]; |
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188 | } else { |
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189 | av_freep(&a->buffer); |
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190 | a->buffer = av_malloc(new_buf_size); |
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191 | if (!a->buffer) |
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192 | return AVERROR(ENOMEM); |
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193 | ret = av_samples_fill_arrays(a->data, &plane_size, a->buffer, |
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194 | a->allocated_channels, nb_samples, |
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195 | a->sample_fmt, 0); |
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196 | if (ret < 0) |
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197 | return ret; |
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198 | } |
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199 | a->buffer_size = new_buf_size; |
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200 | a->allocated_samples = nb_samples; |
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201 | |||
202 | calc_ptr_alignment(a); |
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203 | a->samples_align = plane_size / a->stride; |
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204 | |||
205 | return 0; |
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206 | } |
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207 | |||
208 | void ff_audio_data_free(AudioData **a) |
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209 | { |
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210 | if (!*a) |
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211 | return; |
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212 | av_free((*a)->buffer); |
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213 | av_freep(a); |
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214 | } |
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215 | |||
216 | int ff_audio_data_copy(AudioData *dst, AudioData *src, ChannelMapInfo *map) |
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217 | { |
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218 | int ret, p; |
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219 | |||
220 | /* validate input/output compatibility */ |
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221 | if (dst->sample_fmt != src->sample_fmt || dst->channels < src->channels) |
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222 | return AVERROR(EINVAL); |
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223 | |||
224 | if (map && !src->is_planar) { |
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225 | av_log(src, AV_LOG_ERROR, "cannot remap packed format during copy\n"); |
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226 | return AVERROR(EINVAL); |
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227 | } |
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228 | |||
229 | /* if the input is empty, just empty the output */ |
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230 | if (!src->nb_samples) { |
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231 | dst->nb_samples = 0; |
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232 | return 0; |
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233 | } |
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234 | |||
235 | /* reallocate output if necessary */ |
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236 | ret = ff_audio_data_realloc(dst, src->nb_samples); |
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237 | if (ret < 0) |
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238 | return ret; |
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239 | |||
240 | /* copy data */ |
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241 | if (map) { |
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242 | if (map->do_remap) { |
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243 | for (p = 0; p < src->planes; p++) { |
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244 | if (map->channel_map[p] >= 0) |
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245 | memcpy(dst->data[p], src->data[map->channel_map[p]], |
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246 | src->nb_samples * src->stride); |
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247 | } |
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248 | } |
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249 | if (map->do_copy || map->do_zero) { |
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250 | for (p = 0; p < src->planes; p++) { |
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251 | if (map->channel_copy[p]) |
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252 | memcpy(dst->data[p], dst->data[map->channel_copy[p]], |
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253 | src->nb_samples * src->stride); |
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254 | else if (map->channel_zero[p]) |
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255 | av_samples_set_silence(&dst->data[p], 0, src->nb_samples, |
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256 | 1, dst->sample_fmt); |
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257 | } |
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258 | } |
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259 | } else { |
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260 | for (p = 0; p < src->planes; p++) |
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261 | memcpy(dst->data[p], src->data[p], src->nb_samples * src->stride); |
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262 | } |
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263 | |||
264 | dst->nb_samples = src->nb_samples; |
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265 | |||
266 | return 0; |
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267 | } |
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268 | |||
269 | int ff_audio_data_combine(AudioData *dst, int dst_offset, AudioData *src, |
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270 | int src_offset, int nb_samples) |
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271 | { |
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272 | int ret, p, dst_offset2, dst_move_size; |
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273 | |||
274 | /* validate input/output compatibility */ |
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275 | if (dst->sample_fmt != src->sample_fmt || dst->channels != src->channels) { |
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276 | av_log(src, AV_LOG_ERROR, "sample format mismatch\n"); |
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277 | return AVERROR(EINVAL); |
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278 | } |
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279 | |||
280 | /* validate offsets are within the buffer bounds */ |
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281 | if (dst_offset < 0 || dst_offset > dst->nb_samples || |
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282 | src_offset < 0 || src_offset > src->nb_samples) { |
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283 | av_log(src, AV_LOG_ERROR, "offset out-of-bounds: src=%d dst=%d\n", |
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284 | src_offset, dst_offset); |
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285 | return AVERROR(EINVAL); |
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286 | } |
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287 | |||
288 | /* check offsets and sizes to see if we can just do nothing and return */ |
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289 | if (nb_samples > src->nb_samples - src_offset) |
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290 | nb_samples = src->nb_samples - src_offset; |
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291 | if (nb_samples <= 0) |
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292 | return 0; |
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293 | |||
294 | /* validate that the output is not read-only */ |
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295 | if (dst->read_only) { |
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296 | av_log(dst, AV_LOG_ERROR, "dst is read-only\n"); |
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297 | return AVERROR(EINVAL); |
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298 | } |
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299 | |||
300 | /* reallocate output if necessary */ |
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301 | ret = ff_audio_data_realloc(dst, dst->nb_samples + nb_samples); |
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302 | if (ret < 0) { |
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303 | av_log(dst, AV_LOG_ERROR, "error reallocating dst\n"); |
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304 | return ret; |
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305 | } |
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306 | |||
307 | dst_offset2 = dst_offset + nb_samples; |
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308 | dst_move_size = dst->nb_samples - dst_offset; |
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309 | |||
310 | for (p = 0; p < src->planes; p++) { |
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311 | if (dst_move_size > 0) { |
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312 | memmove(dst->data[p] + dst_offset2 * dst->stride, |
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313 | dst->data[p] + dst_offset * dst->stride, |
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314 | dst_move_size * dst->stride); |
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315 | } |
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316 | memcpy(dst->data[p] + dst_offset * dst->stride, |
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317 | src->data[p] + src_offset * src->stride, |
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318 | nb_samples * src->stride); |
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319 | } |
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320 | dst->nb_samples += nb_samples; |
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321 | |||
322 | return 0; |
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323 | } |
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324 | |||
325 | void ff_audio_data_drain(AudioData *a, int nb_samples) |
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326 | { |
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327 | if (a->nb_samples <= nb_samples) { |
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328 | /* drain the whole buffer */ |
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329 | a->nb_samples = 0; |
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330 | } else { |
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331 | int p; |
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332 | int move_offset = a->stride * nb_samples; |
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333 | int move_size = a->stride * (a->nb_samples - nb_samples); |
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334 | |||
335 | for (p = 0; p < a->planes; p++) |
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336 | memmove(a->data[p], a->data[p] + move_offset, move_size); |
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337 | |||
338 | a->nb_samples -= nb_samples; |
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339 | } |
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340 | } |
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341 | |||
342 | int ff_audio_data_add_to_fifo(AVAudioFifo *af, AudioData *a, int offset, |
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343 | int nb_samples) |
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344 | { |
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345 | uint8_t *offset_data[AVRESAMPLE_MAX_CHANNELS]; |
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346 | int offset_size, p; |
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347 | |||
348 | if (offset >= a->nb_samples) |
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349 | return 0; |
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350 | offset_size = offset * a->stride; |
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351 | for (p = 0; p < a->planes; p++) |
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352 | offset_data[p] = a->data[p] + offset_size; |
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353 | |||
354 | return av_audio_fifo_write(af, (void **)offset_data, nb_samples); |
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355 | } |
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356 | |||
357 | int ff_audio_data_read_from_fifo(AVAudioFifo *af, AudioData *a, int nb_samples) |
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358 | { |
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359 | int ret; |
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360 | |||
361 | if (a->read_only) |
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362 | return AVERROR(EINVAL); |
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363 | |||
364 | ret = ff_audio_data_realloc(a, nb_samples); |
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365 | if (ret < 0) |
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366 | return ret; |
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367 | |||
368 | ret = av_audio_fifo_read(af, (void **)a->data, nb_samples); |
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369 | if (ret >= 0) |
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370 | a->nb_samples = ret; |
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371 | return ret; |
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372 | }>>>=>>>=>>>>>>>>>>>>>>>>>>> |