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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 "audio_out.h"
#include "audio_out_internal.h"
#include "afmt.h"
#include "../config.h"
#include "../mp_msg.h"
#include "../help_mp.h"
#define OBTAIN_BITRATE(a) (((a != AFMT_U8) && (a != AFMT_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 AFMT_U8:
case AFMT_S8:
format = AFMT_U8;
break;
default:
format = AFMT_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 != AFMT_U8 && format != AFMT_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);
}
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