annotate audio.c @ 215:99f4032097e0 libavformat

Compile fix when no CONFIG_ENCODERS by (Glenn Maynard <g_sf at zewt dot org>)
author michaelni
date Thu, 04 Sep 2003 10:30:52 +0000
parents a58a8a53eb46
children 3d92f793fd67
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1 /*
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2 * Linux audio play and grab interface
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3 * Copyright (c) 2000, 2001 Fabrice Bellard.
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4 *
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5 * This library is free software; you can redistribute it and/or
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6 * modify it under the terms of the GNU Lesser General Public
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7 * License as published by the Free Software Foundation; either
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8 * version 2 of the License, or (at your option) any later version.
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9 *
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10 * This library is distributed in the hope that it will be useful,
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11 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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13 * Lesser General Public License for more details.
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14 *
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15 * You should have received a copy of the GNU Lesser General Public
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16 * License along with this library; if not, write to the Free Software
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17 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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18 */
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19 #include "avformat.h"
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20
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21 #include <stdlib.h>
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22 #include <stdio.h>
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23 #include <string.h>
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24 #include <sys/soundcard.h>
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25 #include <unistd.h>
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26 #include <fcntl.h>
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27 #include <sys/ioctl.h>
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28 #include <sys/mman.h>
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29 #include <sys/time.h>
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30
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31 #define AUDIO_BLOCK_SIZE 4096
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32
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33 typedef struct {
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34 int fd;
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35 int sample_rate;
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36 int channels;
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37 int frame_size; /* in bytes ! */
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38 int codec_id;
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39 int flip_left : 1;
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a58a8a53eb46 * UINTX -> uintx_t INTX -> intx_t
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40 uint8_t buffer[AUDIO_BLOCK_SIZE];
0
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41 int buffer_ptr;
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42 } AudioData;
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43
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44 static int audio_open(AudioData *s, int is_output, const char *audio_device)
0
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45 {
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46 int audio_fd;
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47 int tmp, err;
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48 char *flip = getenv("AUDIO_FLIP_LEFT");
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49
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50 /* open linux audio device */
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51 if (!audio_device)
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52 audio_device = "/dev/dsp";
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53
0
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54 if (is_output)
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55 audio_fd = open(audio_device, O_WRONLY);
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56 else
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57 audio_fd = open(audio_device, O_RDONLY);
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58 if (audio_fd < 0) {
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59 perror(audio_device);
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60 return -EIO;
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61 }
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62
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63 if (flip && *flip == '1') {
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64 s->flip_left = 1;
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65 }
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66
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67 /* non blocking mode */
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68 if (!is_output)
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69 fcntl(audio_fd, F_SETFL, O_NONBLOCK);
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70
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71 s->frame_size = AUDIO_BLOCK_SIZE;
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72 #if 0
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73 tmp = (NB_FRAGMENTS << 16) | FRAGMENT_BITS;
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74 err = ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &tmp);
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75 if (err < 0) {
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76 perror("SNDCTL_DSP_SETFRAGMENT");
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77 }
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78 #endif
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79
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80 /* select format : favour native format */
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81 err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);
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82
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83 #ifdef WORDS_BIGENDIAN
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84 if (tmp & AFMT_S16_BE) {
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85 tmp = AFMT_S16_BE;
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86 } else if (tmp & AFMT_S16_LE) {
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87 tmp = AFMT_S16_LE;
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88 } else {
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89 tmp = 0;
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90 }
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91 #else
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92 if (tmp & AFMT_S16_LE) {
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93 tmp = AFMT_S16_LE;
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94 } else if (tmp & AFMT_S16_BE) {
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95 tmp = AFMT_S16_BE;
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96 } else {
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97 tmp = 0;
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98 }
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99 #endif
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100
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101 switch(tmp) {
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102 case AFMT_S16_LE:
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103 s->codec_id = CODEC_ID_PCM_S16LE;
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104 break;
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105 case AFMT_S16_BE:
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106 s->codec_id = CODEC_ID_PCM_S16BE;
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107 break;
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108 default:
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109 fprintf(stderr, "Soundcard does not support 16 bit sample format\n");
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110 close(audio_fd);
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111 return -EIO;
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112 }
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113 err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp);
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114 if (err < 0) {
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115 perror("SNDCTL_DSP_SETFMT");
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116 goto fail;
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117 }
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118
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119 tmp = (s->channels == 2);
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120 err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp);
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121 if (err < 0) {
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122 perror("SNDCTL_DSP_STEREO");
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123 goto fail;
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124 }
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125 if (tmp)
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126 s->channels = 2;
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127
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128 tmp = s->sample_rate;
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129 err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);
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130 if (err < 0) {
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131 perror("SNDCTL_DSP_SPEED");
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132 goto fail;
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133 }
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134 s->sample_rate = tmp; /* store real sample rate */
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135 s->fd = audio_fd;
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136
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137 return 0;
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138 fail:
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139 close(audio_fd);
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140 return -EIO;
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141 }
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142
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143 static int audio_close(AudioData *s)
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144 {
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145 close(s->fd);
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146 return 0;
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147 }
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148
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149 /* sound output support */
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150 static int audio_write_header(AVFormatContext *s1)
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151 {
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152 AudioData *s = s1->priv_data;
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153 AVStream *st;
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154 int ret;
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155
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156 st = s1->streams[0];
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157 s->sample_rate = st->codec.sample_rate;
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158 s->channels = st->codec.channels;
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159 ret = audio_open(s, 1, NULL);
0
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160 if (ret < 0) {
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161 return -EIO;
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162 } else {
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163 return 0;
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164 }
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165 }
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166
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167 static int audio_write_packet(AVFormatContext *s1, int stream_index,
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a58a8a53eb46 * UINTX -> uintx_t INTX -> intx_t
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168 uint8_t *buf, int size, int force_pts)
0
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169 {
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170 AudioData *s = s1->priv_data;
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171 int len, ret;
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172
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173 while (size > 0) {
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174 len = AUDIO_BLOCK_SIZE - s->buffer_ptr;
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175 if (len > size)
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176 len = size;
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177 memcpy(s->buffer + s->buffer_ptr, buf, len);
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178 s->buffer_ptr += len;
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179 if (s->buffer_ptr >= AUDIO_BLOCK_SIZE) {
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180 for(;;) {
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181 ret = write(s->fd, s->buffer, AUDIO_BLOCK_SIZE);
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182 if (ret > 0)
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183 break;
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184 if (ret < 0 && (errno != EAGAIN && errno != EINTR))
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185 return -EIO;
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186 }
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187 s->buffer_ptr = 0;
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188 }
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189 buf += len;
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190 size -= len;
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191 }
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192 return 0;
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193 }
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194
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195 static int audio_write_trailer(AVFormatContext *s1)
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196 {
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197 AudioData *s = s1->priv_data;
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198
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199 audio_close(s);
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200 return 0;
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201 }
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202
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203 /* grab support */
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204
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205 static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap)
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206 {
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207 AudioData *s = s1->priv_data;
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208 AVStream *st;
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209 int ret;
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210
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211 if (!ap || ap->sample_rate <= 0 || ap->channels <= 0)
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212 return -1;
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213
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214 st = av_new_stream(s1, 0);
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215 if (!st) {
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216 return -ENOMEM;
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217 }
05318cf2e886 renamed libav to libavformat
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218 s->sample_rate = ap->sample_rate;
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219 s->channels = ap->channels;
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220
30
90fd30dd68b3 grab device is in AVFormatParameter (at least better than global variable)
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221 ret = audio_open(s, 0, ap->device);
0
05318cf2e886 renamed libav to libavformat
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222 if (ret < 0) {
05318cf2e886 renamed libav to libavformat
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223 av_free(st);
05318cf2e886 renamed libav to libavformat
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224 return -EIO;
05318cf2e886 renamed libav to libavformat
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225 }
05318cf2e886 renamed libav to libavformat
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226
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227 /* take real parameters */
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228 st->codec.codec_type = CODEC_TYPE_AUDIO;
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229 st->codec.codec_id = s->codec_id;
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230 st->codec.sample_rate = s->sample_rate;
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231 st->codec.channels = s->channels;
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232
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233 av_set_pts_info(s1, 48, 1, 1000000); /* 48 bits pts in us */
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234 return 0;
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235 }
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236
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237 static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
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238 {
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239 AudioData *s = s1->priv_data;
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240 int ret, bdelay;
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241 int64_t cur_time;
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242 struct audio_buf_info abufi;
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243
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244 if (av_new_packet(pkt, s->frame_size) < 0)
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245 return -EIO;
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246 for(;;) {
56
01d48dc59dab Fix the 'hard cpu loop' problem when capturing audio from /dev/dsp. This
philipjsg
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247 struct timeval tv;
01d48dc59dab Fix the 'hard cpu loop' problem when capturing audio from /dev/dsp. This
philipjsg
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248 fd_set fds;
01d48dc59dab Fix the 'hard cpu loop' problem when capturing audio from /dev/dsp. This
philipjsg
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249
01d48dc59dab Fix the 'hard cpu loop' problem when capturing audio from /dev/dsp. This
philipjsg
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250 tv.tv_sec = 0;
01d48dc59dab Fix the 'hard cpu loop' problem when capturing audio from /dev/dsp. This
philipjsg
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251 tv.tv_usec = 30 * 1000; /* 30 msecs -- a bit shorter than 1 frame at 30fps */
01d48dc59dab Fix the 'hard cpu loop' problem when capturing audio from /dev/dsp. This
philipjsg
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252
01d48dc59dab Fix the 'hard cpu loop' problem when capturing audio from /dev/dsp. This
philipjsg
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253 FD_ZERO(&fds);
01d48dc59dab Fix the 'hard cpu loop' problem when capturing audio from /dev/dsp. This
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254 FD_SET(s->fd, &fds);
01d48dc59dab Fix the 'hard cpu loop' problem when capturing audio from /dev/dsp. This
philipjsg
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255
01d48dc59dab Fix the 'hard cpu loop' problem when capturing audio from /dev/dsp. This
philipjsg
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diff changeset
256 /* This will block until data is available or we get a timeout */
01d48dc59dab Fix the 'hard cpu loop' problem when capturing audio from /dev/dsp. This
philipjsg
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257 (void) select(s->fd + 1, &fds, 0, 0, &tv);
01d48dc59dab Fix the 'hard cpu loop' problem when capturing audio from /dev/dsp. This
philipjsg
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258
0
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259 ret = read(s->fd, pkt->data, pkt->size);
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260 if (ret > 0)
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261 break;
05318cf2e886 renamed libav to libavformat
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262 if (ret == -1 && (errno == EAGAIN || errno == EINTR)) {
05318cf2e886 renamed libav to libavformat
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263 av_free_packet(pkt);
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264 pkt->size = 0;
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265 return 0;
05318cf2e886 renamed libav to libavformat
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266 }
05318cf2e886 renamed libav to libavformat
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267 if (!(ret == 0 || (ret == -1 && (errno == EAGAIN || errno == EINTR)))) {
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268 av_free_packet(pkt);
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269 return -EIO;
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270 }
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271 }
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272 pkt->size = ret;
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273
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274 /* compute pts of the start of the packet */
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275 cur_time = av_gettime();
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276 bdelay = ret;
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277 if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) {
05318cf2e886 renamed libav to libavformat
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278 bdelay += abufi.bytes;
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279 }
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280 /* substract time represented by the number of bytes in the audio fifo */
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281 cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
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282
05318cf2e886 renamed libav to libavformat
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283 /* convert to wanted units */
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284 pkt->pts = cur_time & ((1LL << 48) - 1);
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285
05318cf2e886 renamed libav to libavformat
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286 if (s->flip_left && s->channels == 2) {
05318cf2e886 renamed libav to libavformat
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287 int i;
05318cf2e886 renamed libav to libavformat
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288 short *p = (short *) pkt->data;
05318cf2e886 renamed libav to libavformat
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289
05318cf2e886 renamed libav to libavformat
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diff changeset
290 for (i = 0; i < ret; i += 4) {
05318cf2e886 renamed libav to libavformat
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291 *p = ~*p;
05318cf2e886 renamed libav to libavformat
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292 p += 2;
05318cf2e886 renamed libav to libavformat
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293 }
05318cf2e886 renamed libav to libavformat
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294 }
05318cf2e886 renamed libav to libavformat
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295 return 0;
05318cf2e886 renamed libav to libavformat
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parents:
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296 }
05318cf2e886 renamed libav to libavformat
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297
05318cf2e886 renamed libav to libavformat
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298 static int audio_read_close(AVFormatContext *s1)
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299 {
05318cf2e886 renamed libav to libavformat
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300 AudioData *s = s1->priv_data;
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301
05318cf2e886 renamed libav to libavformat
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302 audio_close(s);
05318cf2e886 renamed libav to libavformat
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303 return 0;
05318cf2e886 renamed libav to libavformat
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304 }
05318cf2e886 renamed libav to libavformat
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305
05318cf2e886 renamed libav to libavformat
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306 static AVInputFormat audio_in_format = {
05318cf2e886 renamed libav to libavformat
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307 "audio_device",
05318cf2e886 renamed libav to libavformat
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308 "audio grab and output",
05318cf2e886 renamed libav to libavformat
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diff changeset
309 sizeof(AudioData),
05318cf2e886 renamed libav to libavformat
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310 NULL,
05318cf2e886 renamed libav to libavformat
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diff changeset
311 audio_read_header,
05318cf2e886 renamed libav to libavformat
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diff changeset
312 audio_read_packet,
05318cf2e886 renamed libav to libavformat
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313 audio_read_close,
05318cf2e886 renamed libav to libavformat
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diff changeset
314 .flags = AVFMT_NOFILE,
05318cf2e886 renamed libav to libavformat
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315 };
05318cf2e886 renamed libav to libavformat
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316
05318cf2e886 renamed libav to libavformat
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diff changeset
317 static AVOutputFormat audio_out_format = {
05318cf2e886 renamed libav to libavformat
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318 "audio_device",
05318cf2e886 renamed libav to libavformat
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diff changeset
319 "audio grab and output",
05318cf2e886 renamed libav to libavformat
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320 "",
05318cf2e886 renamed libav to libavformat
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diff changeset
321 "",
05318cf2e886 renamed libav to libavformat
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diff changeset
322 sizeof(AudioData),
05318cf2e886 renamed libav to libavformat
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diff changeset
323 /* XXX: we make the assumption that the soundcard accepts this format */
05318cf2e886 renamed libav to libavformat
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diff changeset
324 /* XXX: find better solution with "preinit" method, needed also in
05318cf2e886 renamed libav to libavformat
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325 other formats */
05318cf2e886 renamed libav to libavformat
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diff changeset
326 #ifdef WORDS_BIGENDIAN
05318cf2e886 renamed libav to libavformat
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327 CODEC_ID_PCM_S16BE,
05318cf2e886 renamed libav to libavformat
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diff changeset
328 #else
05318cf2e886 renamed libav to libavformat
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diff changeset
329 CODEC_ID_PCM_S16LE,
05318cf2e886 renamed libav to libavformat
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diff changeset
330 #endif
05318cf2e886 renamed libav to libavformat
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diff changeset
331 CODEC_ID_NONE,
05318cf2e886 renamed libav to libavformat
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diff changeset
332 audio_write_header,
05318cf2e886 renamed libav to libavformat
bellard
parents:
diff changeset
333 audio_write_packet,
05318cf2e886 renamed libav to libavformat
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parents:
diff changeset
334 audio_write_trailer,
05318cf2e886 renamed libav to libavformat
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diff changeset
335 .flags = AVFMT_NOFILE,
05318cf2e886 renamed libav to libavformat
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parents:
diff changeset
336 };
05318cf2e886 renamed libav to libavformat
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diff changeset
337
05318cf2e886 renamed libav to libavformat
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parents:
diff changeset
338 int audio_init(void)
05318cf2e886 renamed libav to libavformat
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parents:
diff changeset
339 {
05318cf2e886 renamed libav to libavformat
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parents:
diff changeset
340 av_register_input_format(&audio_in_format);
05318cf2e886 renamed libav to libavformat
bellard
parents:
diff changeset
341 av_register_output_format(&audio_out_format);
05318cf2e886 renamed libav to libavformat
bellard
parents:
diff changeset
342 return 0;
05318cf2e886 renamed libav to libavformat
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343 }