Mercurial > mplayer.hg
annotate libao2/ao_arts.c @ 9051:da9d39c59294
bunkus: Fixed and updated the translations
author | mosu |
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date | Tue, 21 Jan 2003 10:42:18 +0000 |
parents | ed132c268686 |
children | 12b1790038b0 |
rev | line source |
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1 /* |
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2 * ao_arts - aRts audio output driver for MPlayer |
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3 * |
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4 * Michele Balistreri <brain87@gmx.net> |
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5 * |
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6 * This driver is distribuited under terms of GPL |
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7 * |
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8 */ |
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9 |
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10 #include <artsc.h> |
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11 #include <stdio.h> |
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12 |
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13 #include "audio_out.h" |
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14 #include "audio_out_internal.h" |
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15 #include "afmt.h" |
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16 #include "../config.h" |
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17 #include "../mp_msg.h" |
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18 |
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19 #define OBTAIN_BITRATE(a) (((a != AFMT_U8) && (a != AFMT_S8)) ? 16 : 8) |
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20 |
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21 /* Feel free to experiment with the following values: */ |
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22 #define ARTS_PACKETS 10 /* Number of audio packets */ |
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23 #define ARTS_PACKET_SIZE_LOG2 11 /* Log2 of audio packet size */ |
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24 |
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25 static arts_stream_t stream; |
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26 |
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27 static ao_info_t info = |
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28 { |
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29 "aRts audio output", |
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30 "arts", |
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31 "Michele Balistreri <brain87@gmx.net>", |
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32 "" |
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33 }; |
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34 |
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35 LIBAO_EXTERN(arts) |
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36 |
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37 static int control(int cmd, int arg) |
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38 { |
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39 return(CONTROL_UNKNOWN); |
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40 } |
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41 |
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42 static int init(int rate_hz, int channels, int format, int flags) |
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43 { |
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44 int err; |
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45 int frag_spec; |
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46 |
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47 if( (err=arts_init()) ) { |
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48 mp_msg(MSGT_AO, MSGL_ERR, "AO: [arts] %s\n", arts_error_text(err)); |
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49 return 0; |
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50 } |
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51 mp_msg(MSGT_AO, MSGL_INFO, "AO: [arts] Connected to sound server\n"); |
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52 |
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53 /* |
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54 * arts supports 8bit unsigned and 16bit signed sample formats |
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55 * (16bit apparently in little endian format, even in the case |
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56 * when artsd runs on a big endian cpu). |
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57 * |
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58 * Unsupported formats are translated to one of these two formats |
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59 * using mplayer's audio filters. |
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60 */ |
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61 switch (format) { |
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62 case AFMT_U8: |
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63 case AFMT_S8: |
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64 format = AFMT_U8; |
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65 break; |
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66 default: |
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67 format = AFMT_S16_LE; /* artsd always expects little endian?*/ |
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68 break; |
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69 } |
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70 |
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71 ao_data.format = format; |
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72 ao_data.channels = channels; |
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73 ao_data.samplerate = rate_hz; |
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74 ao_data.bps = (rate_hz*channels); |
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75 |
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76 if(format != AFMT_U8 && format != AFMT_S8) |
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77 ao_data.bps*=2; |
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78 |
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79 stream=arts_play_stream(rate_hz, OBTAIN_BITRATE(format), channels, "Mplayer"); |
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80 |
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81 if(stream == NULL) { |
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82 mp_msg(MSGT_AO, MSGL_ERR, "AO: [arts] Unable to open a stream\n"); |
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83 arts_free(); |
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84 return 0; |
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85 } |
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86 |
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87 /* Set the stream to blocking: it will not block anyway, but it seems */ |
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88 /* to be working better */ |
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89 arts_stream_set(stream, ARTS_P_BLOCKING, 1); |
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90 frag_spec = ARTS_PACKET_SIZE_LOG2 | ARTS_PACKETS << 16; |
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91 arts_stream_set(stream, ARTS_P_PACKET_SETTINGS, frag_spec); |
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92 ao_data.buffersize = arts_stream_get(stream, ARTS_P_BUFFER_SIZE); |
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93 mp_msg(MSGT_AO, MSGL_INFO, "AO: [arts] Stream opened\n"); |
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94 |
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95 mp_msg(MSGT_AO, MSGL_INFO,"AO: [arts] buffer size: %d\n", |
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96 ao_data.buffersize); |
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97 mp_msg(MSGT_AO, MSGL_INFO,"AO: [arts] packet size: %d\n", |
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98 arts_stream_get(stream, ARTS_P_PACKET_SIZE)); |
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99 |
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100 return 1; |
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101 } |
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102 |
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103 static void uninit() |
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104 { |
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105 arts_close_stream(stream); |
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106 arts_free(); |
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107 } |
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108 |
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109 static int play(void* data,int len,int flags) |
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110 { |
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111 return arts_write(stream, data, len); |
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112 } |
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113 |
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114 static void audio_pause() |
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115 { |
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116 } |
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117 |
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118 static void audio_resume() |
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119 { |
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120 } |
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121 |
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122 static void reset() |
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123 { |
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124 } |
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125 |
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126 static int get_space() |
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127 { |
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128 return arts_stream_get(stream, ARTS_P_BUFFER_SPACE); |
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129 } |
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130 |
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131 static float get_delay() |
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132 { |
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133 return ((float) (ao_data.buffersize - arts_stream_get(stream, |
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134 ARTS_P_BUFFER_SPACE))) / ((float) ao_data.bps); |
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135 } |
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136 |