Mercurial > mplayer.hg
annotate libao2/ao_macosx.c @ 11041:450f7aa81adc
build fixes
author | diego |
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date | Wed, 08 Oct 2003 01:06:50 +0000 |
parents | b252d1b6829e |
children | 99798c3cdb93 |
rev | line source |
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1 /* |
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2 * |
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3 * ao_macosx.c |
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4 * |
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5 * Original Copyright (C) Timothy J. Wood - Aug 2000 |
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6 * |
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7 * This file is part of libao, a cross-platform library. See |
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8 * README for a history of this source code. |
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9 * |
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10 * libao is free software; you can redistribute it and/or modify |
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11 * it under the terms of the GNU General Public License as published by |
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12 * the Free Software Foundation; either version 2, or (at your option) |
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13 * any later version. |
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14 * |
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15 * libao is distributed in the hope that it will be useful, |
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16 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
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17 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the |
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18 * GNU General Public License for more details. |
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19 * |
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20 * You should have received a copy of the GNU General Public License |
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21 * along with GNU Make; see the file COPYING. If not, write to |
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22 * the Free Software Foundation, 675 Mass Ave, Cambridge, MA 02139, USA. |
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23 */ |
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24 |
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25 /* |
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26 * The MacOS X CoreAudio framework doesn't mesh as simply as some |
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27 * simpler frameworks do. This is due to the fact that CoreAudio pulls |
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28 * audio samples rather than having them pushed at it (which is nice |
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29 * when you are wanting to do good buffering of audio). |
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30 */ |
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31 |
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32 /* Change log: |
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33 * |
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34 * 14/5-2003: Ported to MPlayer libao2 by Dan Christiansen |
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35 * |
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36 * AC-3 and MPEG audio passthrough is possible, but I don't have |
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37 * access to a sound card that supports it. |
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38 */ |
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39 |
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40 #include <CoreAudio/AudioHardware.h> |
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41 #include <stdio.h> |
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42 #include <string.h> |
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43 #include <inttypes.h> |
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44 #include <pthread.h> |
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45 |
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46 #include "../mp_msg.h" |
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47 |
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48 #include "audio_out.h" |
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49 #include "audio_out_internal.h" |
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50 #include "afmt.h" |
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51 |
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52 static ao_info_t info = |
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53 { |
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54 "Darwin/Mac OS X native audio output", |
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55 "macosx", |
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56 "Timothy J. Wood & Dan Christiansen", |
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57 "" |
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58 }; |
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59 |
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60 LIBAO_EXTERN(macosx) |
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61 |
10152 | 62 /* Prefix for all mp_msg() calls */ |
63 #define ao_msg(a, b, c...) mp_msg(a, b, "AO: [macosx] " c) | |
64 | |
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65 /* This is large, but best (maybe it should be even larger). |
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66 * CoreAudio supposedly has an internal latency in the order of 2ms */ |
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67 #define NUM_BUFS 128 |
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68 |
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69 typedef struct ao_macosx_s |
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70 { |
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71 /* CoreAudio */ |
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72 AudioDeviceID outputDeviceID; |
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73 AudioStreamBasicDescription outputStreamBasicDescription; |
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74 |
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75 /* Ring-buffer */ |
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76 pthread_mutex_t buffer_mutex; /* mutex covering buffer variables */ |
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77 |
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78 unsigned char *buffer[NUM_BUFS]; |
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79 unsigned int buffer_len; |
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80 |
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81 unsigned int buf_read; |
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82 unsigned int buf_write; |
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83 unsigned int buf_read_pos; |
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84 unsigned int buf_write_pos; |
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85 int full_buffers; |
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86 int buffered_bytes; |
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87 } ao_macosx_t; |
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88 |
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89 static ao_macosx_t *ao; |
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90 |
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91 /* General purpose Ring-buffering routines */ |
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92 static int write_buffer(unsigned char* data,int len){ |
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93 int len2=0; |
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94 int x; |
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95 |
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96 while(len>0){ |
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97 if(ao->full_buffers==NUM_BUFS) { |
10152 | 98 ao_msg(MSGT_AO,MSGL_V, "Buffer overrun\n"); |
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99 break; |
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100 } |
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101 |
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102 x=ao->buffer_len-ao->buf_write_pos; |
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103 if(x>len) x=len; |
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104 memcpy(ao->buffer[ao->buf_write]+ao->buf_write_pos,data+len2,x); |
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105 |
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106 /* accessing common variables, locking mutex */ |
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107 pthread_mutex_lock(&ao->buffer_mutex); |
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108 len2+=x; len-=x; |
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109 ao->buffered_bytes+=x; ao->buf_write_pos+=x; |
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110 if(ao->buf_write_pos>=ao->buffer_len) { |
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111 /* block is full, find next! */ |
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112 ao->buf_write=(ao->buf_write+1)%NUM_BUFS; |
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113 ++ao->full_buffers; |
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114 ao->buf_write_pos=0; |
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115 } |
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116 pthread_mutex_unlock(&ao->buffer_mutex); |
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117 } |
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118 |
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119 return len2; |
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120 } |
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121 |
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122 static int read_buffer(unsigned char* data,int len){ |
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123 int len2=0; |
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124 int x; |
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125 |
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126 while(len>0){ |
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127 if(ao->full_buffers==0) { |
10152 | 128 ao_msg(MSGT_AO,MSGL_V, "Buffer underrun\n"); |
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129 break; |
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130 } |
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131 |
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132 x=ao->buffer_len-ao->buf_read_pos; |
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133 if(x>len) x=len; |
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134 memcpy(data+len2,ao->buffer[ao->buf_read]+ao->buf_read_pos,x); |
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135 len2+=x; len-=x; |
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136 |
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137 /* accessing common variables, locking mutex */ |
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138 pthread_mutex_lock(&ao->buffer_mutex); |
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139 ao->buffered_bytes-=x; ao->buf_read_pos+=x; |
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140 if(ao->buf_read_pos>=ao->buffer_len){ |
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141 /* block is empty, find next! */ |
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142 ao->buf_read=(ao->buf_read+1)%NUM_BUFS; |
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143 --ao->full_buffers; |
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144 ao->buf_read_pos=0; |
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145 } |
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146 pthread_mutex_unlock(&ao->buffer_mutex); |
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147 } |
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148 |
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149 |
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150 return len2; |
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151 } |
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152 |
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153 /* end ring buffer stuff */ |
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154 |
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155 /* The function that the CoreAudio thread calls when it wants more data */ |
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156 static OSStatus audioDeviceIOProc(AudioDeviceID inDevice, const AudioTimeStamp *inNow, const AudioBufferList *inInputData, const AudioTimeStamp *inInputTime, AudioBufferList *outOutputData, const AudioTimeStamp *inOutputTime, void *inClientData) |
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157 { |
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158 outOutputData->mBuffers[0].mDataByteSize = |
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159 read_buffer((char *)outOutputData->mBuffers[0].mData, ao->buffer_len); |
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160 |
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161 return 0; |
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162 } |
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163 |
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164 |
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165 static int control(int cmd,void *arg){ |
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166 switch (cmd) { |
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167 case AOCONTROL_SET_DEVICE: |
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168 case AOCONTROL_GET_DEVICE: |
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169 case AOCONTROL_GET_VOLUME: |
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170 case AOCONTROL_SET_VOLUME: |
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171 /* unimplemented/meaningless */ |
10152 | 172 return CONTROL_FALSE; |
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173 case AOCONTROL_QUERY_FORMAT: |
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174 /* stick with what CoreAudio requests */ |
10152 | 175 return CONTROL_FALSE; |
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176 default: |
10152 | 177 return CONTROL_FALSE; |
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178 } |
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179 |
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180 } |
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181 |
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182 |
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183 static int init(int rate,int channels,int format,int flags) |
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184 { |
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185 OSStatus status; |
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186 UInt32 propertySize; |
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187 int rc; |
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188 int i; |
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189 |
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190 ao = (ao_macosx_t *)malloc(sizeof(ao_macosx_t)); |
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191 |
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192 /* initialise mutex */ |
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193 pthread_mutex_init(&ao->buffer_mutex, NULL); |
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194 pthread_mutex_unlock(&ao->buffer_mutex); |
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195 |
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196 /* get default output device */ |
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197 propertySize = sizeof(ao->outputDeviceID); |
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198 status = AudioHardwareGetProperty(kAudioHardwarePropertyDefaultOutputDevice, &propertySize, &(ao->outputDeviceID)); |
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199 if (status) { |
10152 | 200 ao_msg(MSGT_AO,MSGL_WARN, |
201 "AudioHardwareGetProperty returned %d\n", | |
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202 (int)status); |
10152 | 203 return CONTROL_FALSE; |
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204 } |
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205 |
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206 if (ao->outputDeviceID == kAudioDeviceUnknown) { |
10152 | 207 ao_msg(MSGT_AO,MSGL_WARN, "AudioHardwareGetProperty: ao->outputDeviceID is kAudioDeviceUnknown\n"); |
208 return CONTROL_FALSE; | |
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209 } |
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210 |
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211 /* get default output format |
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212 * TODO: get all support formats and iterate through them |
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213 */ |
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214 propertySize = sizeof(ao->outputStreamBasicDescription); |
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215 status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioDevicePropertyStreamFormat, &propertySize, &ao->outputStreamBasicDescription); |
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216 if (status) { |
10152 | 217 ao_msg(MSGT_AO,MSGL_WARN, "AudioDeviceGetProperty returned %d when getting kAudioDevicePropertyStreamFormat\n", (int)status); |
218 return CONTROL_FALSE; | |
10147
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219 } |
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220 |
10152 | 221 ao_msg(MSGT_AO,MSGL_V, "hardware format...\n"); |
222 ao_msg(MSGT_AO,MSGL_V, "%f mSampleRate\n", ao->outputStreamBasicDescription.mSampleRate); | |
223 ao_msg(MSGT_AO,MSGL_V, " %c%c%c%c mFormatID\n", | |
10147
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224 (int)(ao->outputStreamBasicDescription.mFormatID & 0xff000000) >> 24, |
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225 (int)(ao->outputStreamBasicDescription.mFormatID & 0x00ff0000) >> 16, |
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226 (int)(ao->outputStreamBasicDescription.mFormatID & 0x0000ff00) >> 8, |
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227 (int)(ao->outputStreamBasicDescription.mFormatID & 0x000000ff) >> 0); |
10152 | 228 ao_msg(MSGT_AO,MSGL_V, "%5d mBytesPerPacket\n", |
10147
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229 (int)ao->outputStreamBasicDescription.mBytesPerPacket); |
10152 | 230 ao_msg(MSGT_AO,MSGL_V, "%5d mFramesPerPacket\n", |
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231 (int)ao->outputStreamBasicDescription.mFramesPerPacket); |
10152 | 232 ao_msg(MSGT_AO,MSGL_V, "%5d mBytesPerFrame\n", |
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233 (int)ao->outputStreamBasicDescription.mBytesPerFrame); |
10152 | 234 ao_msg(MSGT_AO,MSGL_V, "%5d mChannelsPerFrame\n", |
10147
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235 (int)ao->outputStreamBasicDescription.mChannelsPerFrame); |
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236 |
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237 /* get requested buffer length */ |
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238 propertySize = sizeof(ao->buffer_len); |
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239 status = AudioDeviceGetProperty(ao->outputDeviceID, 0, false, kAudioDevicePropertyBufferSize, &propertySize, &ao->buffer_len); |
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240 if (status) { |
10152 | 241 ao_msg(MSGT_AO,MSGL_WARN, "AudioDeviceGetProperty returned %d when getting kAudioDevicePropertyBufferSize\n", (int)status); |
242 return CONTROL_FALSE; | |
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243 } |
10152 | 244 ao_msg(MSGT_AO,MSGL_V, "%5d ao->buffer_len\n", (int)ao->buffer_len); |
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245 |
10152 | 246 /* FIXME: |
247 * | |
248 * Resampling of 32-bit float audio is broken in MPlayer. Refuse to | |
249 * handle anything other than the native format until this is fixed | |
250 * or this module is rewritten, whichever comes first. | |
251 */ | |
252 if (ao_data.samplerate == ao->outputStreamBasicDescription.mSampleRate) { | |
253 ao_data.samplerate = (int)ao->outputStreamBasicDescription.mSampleRate; | |
254 } else { | |
255 ao_msg(MSGT_AO,MSGL_WARN, "Resampling not supported yet.\n"); | |
256 return 0; | |
257 } | |
258 | |
10147
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259 ao_data.channels = ao->outputStreamBasicDescription.mChannelsPerFrame; |
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260 ao_data.outburst = ao_data.buffersize = ao->buffer_len; |
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261 ao_data.bps = |
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262 ao_data.samplerate * ao->outputStreamBasicDescription.mBytesPerFrame; |
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263 |
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264 if (ao->outputStreamBasicDescription.mFormatID == kAudioFormatLinearPCM) { |
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265 uint32_t flags = ao->outputStreamBasicDescription.mFormatFlags; |
10152 | 266 if (flags & kAudioFormatFlagIsFloat) { |
267 ao_data.format = AFMT_FLOAT; | |
268 } else { | |
269 ao_msg(MSGT_AO,MSGL_WARN, "Unsupported audio output " | |
270 "format %d. Please report this to the developer\n", | |
271 (int)status); | |
272 return CONTROL_FALSE; | |
273 } | |
274 | |
10147
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275 } else { |
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276 /* TODO: handle AFMT_AC3, AFMT_MPEG & friends */ |
10152 | 277 ao_msg(MSGT_AO,MSGL_WARN, "Default Audio Device doesn't " |
10147
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278 "support Linear PCM!\n"); |
10152 | 279 return CONTROL_FALSE; |
10147
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280 } |
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281 |
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282 /* Allocate ring-buffer memory */ |
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283 for(i=0;i<NUM_BUFS;i++) |
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284 ao->buffer[i]=(unsigned char *) malloc(ao->buffer_len); |
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285 |
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286 |
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287 /* Prepare for playback */ |
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288 |
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289 reset(); |
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290 |
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291 /* Set the IO proc that CoreAudio will call when it needs data */ |
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292 status = AudioDeviceAddIOProc(ao->outputDeviceID, audioDeviceIOProc, NULL); |
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293 if (status) { |
10152 | 294 ao_msg(MSGT_AO,MSGL_WARN, "AudioDeviceAddIOProc returned %d\n", (int)status); |
295 return CONTROL_FALSE; | |
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296 } |
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297 |
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298 /* Start callback */ |
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299 status = AudioDeviceStart(ao->outputDeviceID, audioDeviceIOProc); |
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300 if (status) { |
10152 | 301 ao_msg(MSGT_AO,MSGL_WARN, "AudioDeviceStart returned %d\n", |
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302 (int)status); |
10152 | 303 return CONTROL_FALSE; |
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304 } |
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305 |
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306 return CONTROL_OK; |
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307 } |
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308 |
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309 |
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310 static int play(void* output_samples,int num_bytes,int flags) |
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311 { |
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312 return write_buffer(output_samples, num_bytes); |
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313 } |
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314 |
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315 /* set variables and buffer to initial state */ |
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316 static void reset() |
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317 { |
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318 int i; |
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319 |
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320 pthread_mutex_lock(&ao->buffer_mutex); |
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321 |
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322 /* reset ring-buffer state */ |
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323 ao->buf_read=0; |
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324 ao->buf_write=0; |
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325 ao->buf_read_pos=0; |
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326 ao->buf_write_pos=0; |
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327 |
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328 ao->full_buffers=0; |
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329 ao->buffered_bytes=0; |
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330 |
10152 | 331 /* zero output buffer */ |
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332 for (i = 0; i < NUM_BUFS; i++) |
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333 bzero(ao->buffer[i], ao->buffer_len); |
10152 | 334 |
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335 pthread_mutex_unlock(&ao->buffer_mutex); |
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336 |
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337 return; |
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338 } |
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339 |
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340 |
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341 /* return available space */ |
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342 static int get_space() |
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343 { |
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344 return (NUM_BUFS-ao->full_buffers)*ao_data.buffersize - ao->buf_write_pos; |
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345 } |
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346 |
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347 |
10152 | 348 /* return delay until audio is played */ |
10147
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349 static float get_delay() |
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350 { |
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351 return (float)(ao->buffered_bytes)/(float)ao_data.bps; |
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352 } |
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353 |
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354 |
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355 /* unload plugin and deregister from coreaudio */ |
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356 static void uninit() |
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357 { |
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358 int i; |
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359 OSErr status; |
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360 |
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361 reset(); |
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362 |
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363 status = AudioDeviceRemoveIOProc(ao->outputDeviceID, audioDeviceIOProc); |
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364 if (status) |
10152 | 365 ao_msg(MSGT_AO,MSGL_WARN, "AudioDeviceRemoveIOProc " |
10147
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366 "returned %d\n", (int)status); |
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367 |
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368 for(i=0;i<NUM_BUFS;i++) free(ao->buffer[i]); |
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369 free(ao); |
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370 } |
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371 |
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372 |
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373 /* stop playing, keep buffers (for pause) */ |
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374 static void audio_pause() |
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375 { |
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376 OSErr status; |
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377 |
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378 /* stop callback */ |
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379 status = AudioDeviceStop(ao->outputDeviceID, audioDeviceIOProc); |
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380 if (status) |
10152 | 381 ao_msg(MSGT_AO,MSGL_WARN, "AudioDeviceStop returned %d\n", |
10147
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382 (int)status); |
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383 } |
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384 |
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385 |
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386 /* resume playing, after audio_pause() */ |
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387 static void audio_resume() |
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388 { |
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389 OSErr status = AudioDeviceStart(ao->outputDeviceID, audioDeviceIOProc); |
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390 if (status) |
10152 | 391 ao_msg(MSGT_AO,MSGL_WARN, "AudioDeviceStart returned %d\n", |
10147
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392 (int)status); |
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393 } |