Mercurial > mplayer.hg
annotate libao2/ao_arts.c @ 31738:781904280cd7
Remove pointless casts.
author | reimar |
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date | Sun, 25 Jul 2010 11:18:04 +0000 |
parents | 32725ca88fed |
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rev | line source |
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1 /* |
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2 * aRts audio output driver for MPlayer |
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3 * |
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4 * copyright (c) 2002 Michele Balistreri <brain87@gmx.net> |
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5 * |
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6 * This file is part of MPlayer. |
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7 * |
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8 * MPlayer is free software; you can redistribute it and/or modify |
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9 * it under the terms of the GNU General Public License as published by |
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10 * the Free Software Foundation; either version 2 of the License, or |
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11 * (at your option) any later version. |
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12 * |
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13 * MPlayer is distributed in the hope that it will be useful, |
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14 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
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15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the |
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16 * GNU General Public License for more details. |
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17 * |
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18 * You should have received a copy of the GNU General Public License along |
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19 * with MPlayer; if not, write to the Free Software Foundation, Inc., |
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20 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. |
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21 */ |
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22 |
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23 #include <artsc.h> |
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24 #include <stdio.h> |
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25 |
14479 | 26 #include "config.h" |
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27 #include "audio_out.h" |
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28 #include "audio_out_internal.h" |
14245 | 29 #include "libaf/af_format.h" |
14123 | 30 #include "mp_msg.h" |
31 #include "help_mp.h" | |
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32 |
14245 | 33 #define OBTAIN_BITRATE(a) (((a != AF_FORMAT_U8) && (a != AF_FORMAT_S8)) ? 16 : 8) |
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34 |
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35 /* Feel free to experiment with the following values: */ |
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36 #define ARTS_PACKETS 10 /* Number of audio packets */ |
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37 #define ARTS_PACKET_SIZE_LOG2 11 /* Log2 of audio packet size */ |
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38 |
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39 static arts_stream_t stream; |
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40 |
28823 | 41 static const ao_info_t info = |
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42 { |
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43 "aRts audio output", |
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44 "arts", |
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45 "Michele Balistreri <brain87@gmx.net>", |
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46 "" |
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47 }; |
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48 |
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49 LIBAO_EXTERN(arts) |
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50 |
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51 static int control(int cmd, void *arg) |
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52 { |
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53 return CONTROL_UNKNOWN; |
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54 } |
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55 |
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56 static int init(int rate_hz, int channels, int format, int flags) |
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57 { |
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58 int err; |
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59 int frag_spec; |
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60 |
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61 if( (err=arts_init()) ) { |
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62 mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ARTS_CantInit, arts_error_text(err)); |
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63 return 0; |
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64 } |
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65 mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_ARTS_ServerConnect); |
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66 |
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67 /* |
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68 * arts supports 8bit unsigned and 16bit signed sample formats |
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69 * (16bit apparently in little endian format, even in the case |
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70 * when artsd runs on a big endian cpu). |
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71 * |
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72 * Unsupported formats are translated to one of these two formats |
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73 * using mplayer's audio filters. |
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74 */ |
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75 switch (format) { |
14245 | 76 case AF_FORMAT_U8: |
77 case AF_FORMAT_S8: | |
78 format = AF_FORMAT_U8; | |
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79 break; |
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80 default: |
14245 | 81 format = AF_FORMAT_S16_LE; /* artsd always expects little endian?*/ |
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82 break; |
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83 } |
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84 |
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85 ao_data.format = format; |
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86 ao_data.channels = channels; |
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87 ao_data.samplerate = rate_hz; |
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88 ao_data.bps = (rate_hz*channels); |
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89 |
14245 | 90 if(format != AF_FORMAT_U8 && format != AF_FORMAT_S8) |
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91 ao_data.bps*=2; |
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92 |
12984 | 93 stream=arts_play_stream(rate_hz, OBTAIN_BITRATE(format), channels, "MPlayer"); |
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94 |
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95 if(stream == NULL) { |
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96 mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ARTS_CantOpenStream); |
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97 arts_free(); |
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98 return 0; |
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99 } |
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100 |
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101 /* Set the stream to blocking: it will not block anyway, but it seems */ |
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102 /* to be working better */ |
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103 arts_stream_set(stream, ARTS_P_BLOCKING, 1); |
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104 frag_spec = ARTS_PACKET_SIZE_LOG2 | ARTS_PACKETS << 16; |
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105 arts_stream_set(stream, ARTS_P_PACKET_SETTINGS, frag_spec); |
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106 ao_data.buffersize = arts_stream_get(stream, ARTS_P_BUFFER_SIZE); |
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107 mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_ARTS_StreamOpen); |
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108 |
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109 mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_ARTS_BufferSize, |
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110 ao_data.buffersize); |
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111 mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_ARTS_BufferSize, |
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112 arts_stream_get(stream, ARTS_P_PACKET_SIZE)); |
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113 |
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114 return 1; |
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115 } |
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116 |
12145 | 117 static void uninit(int immed) |
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118 { |
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119 arts_close_stream(stream); |
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120 arts_free(); |
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121 } |
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122 |
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123 static int play(void* data,int len,int flags) |
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124 { |
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125 return arts_write(stream, data, len); |
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126 } |
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127 |
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128 static void audio_pause(void) |
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129 { |
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130 } |
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131 |
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132 static void audio_resume(void) |
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133 { |
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134 } |
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135 |
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136 static void reset(void) |
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137 { |
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138 } |
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139 |
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140 static int get_space(void) |
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141 { |
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142 return arts_stream_get(stream, ARTS_P_BUFFER_SPACE); |
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143 } |
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144 |
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145 static float get_delay(void) |
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146 { |
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147 return ((float) (ao_data.buffersize - arts_stream_get(stream, |
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148 ARTS_P_BUFFER_SPACE))) / ((float) ao_data.bps); |
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149 } |