Mercurial > mplayer.hg
view libao2/ao_arts.c @ 16056:65acf12dc569
sync 1.66
author | wight |
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date | Fri, 22 Jul 2005 20:48:15 +0000 |
parents | cae0dbeb44bb |
children | f580a7755ac5 |
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/* * ao_arts - aRts audio output driver for MPlayer * * Michele Balistreri <brain87@gmx.net> * * This driver is distribuited under terms of GPL * */ #include <artsc.h> #include <stdio.h> #include "config.h" #include "audio_out.h" #include "audio_out_internal.h" #include "libaf/af_format.h" #include "mp_msg.h" #include "help_mp.h" #define OBTAIN_BITRATE(a) (((a != AF_FORMAT_U8) && (a != AF_FORMAT_S8)) ? 16 : 8) /* Feel free to experiment with the following values: */ #define ARTS_PACKETS 10 /* Number of audio packets */ #define ARTS_PACKET_SIZE_LOG2 11 /* Log2 of audio packet size */ static arts_stream_t stream; static ao_info_t info = { "aRts audio output", "arts", "Michele Balistreri <brain87@gmx.net>", "" }; LIBAO_EXTERN(arts) static int control(int cmd, void *arg) { return(CONTROL_UNKNOWN); } static int init(int rate_hz, int channels, int format, int flags) { int err; int frag_spec; if( (err=arts_init()) ) { mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ARTS_CantInit, arts_error_text(err)); return 0; } mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_ARTS_ServerConnect); /* * arts supports 8bit unsigned and 16bit signed sample formats * (16bit apparently in little endian format, even in the case * when artsd runs on a big endian cpu). * * Unsupported formats are translated to one of these two formats * using mplayer's audio filters. */ switch (format) { case AF_FORMAT_U8: case AF_FORMAT_S8: format = AF_FORMAT_U8; break; default: format = AF_FORMAT_S16_LE; /* artsd always expects little endian?*/ break; } ao_data.format = format; ao_data.channels = channels; ao_data.samplerate = rate_hz; ao_data.bps = (rate_hz*channels); if(format != AF_FORMAT_U8 && format != AF_FORMAT_S8) ao_data.bps*=2; stream=arts_play_stream(rate_hz, OBTAIN_BITRATE(format), channels, "MPlayer"); if(stream == NULL) { mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_ARTS_CantOpenStream); arts_free(); return 0; } /* Set the stream to blocking: it will not block anyway, but it seems */ /* to be working better */ arts_stream_set(stream, ARTS_P_BLOCKING, 1); frag_spec = ARTS_PACKET_SIZE_LOG2 | ARTS_PACKETS << 16; arts_stream_set(stream, ARTS_P_PACKET_SETTINGS, frag_spec); ao_data.buffersize = arts_stream_get(stream, ARTS_P_BUFFER_SIZE); mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_ARTS_StreamOpen); mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_ARTS_BufferSize, ao_data.buffersize); mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_ARTS_BufferSize, arts_stream_get(stream, ARTS_P_PACKET_SIZE)); return 1; } static void uninit(int immed) { arts_close_stream(stream); arts_free(); } static int play(void* data,int len,int flags) { return arts_write(stream, data, len); } static void audio_pause() { } static void audio_resume() { } static void reset() { } static int get_space() { return arts_stream_get(stream, ARTS_P_BUFFER_SPACE); } static float get_delay() { return ((float) (ao_data.buffersize - arts_stream_get(stream, ARTS_P_BUFFER_SPACE))) / ((float) ao_data.bps); }