annotate libao2/ao_arts.c @ 24131:30028bbcb9e8

Use a single select() for both key and slave input Previous code used two separate select() calls one after another, so that whenever it was running select() on one set of fds events in the other set would go unnoticed until later. Now there's a single select() which allows reacting immediately to any input source. The behavior of the new code differs somewhat from the old; for example multiple fds that stay readable are no longer handled in a round-robin fashion and the total amount the process sleeps can differ. Some tuning might be required later.

/*
 * ao_arts - aRts audio output driver for MPlayer
 *
 * Michele Balistreri <[e-mail]>
 *
 * 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 <[e-mail]>",
    ""
};

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(void)
{
}

static void audio_resume(void)
{
}

static void reset(void)
{
}

static int get_space(void)
{
	return arts_stream_get(stream, ARTS_P_BUFFER_SPACE);
}

static float get_delay(void)
{
	return ((float) (ao_data.buffersize - arts_stream_get(stream,
		ARTS_P_BUFFER_SPACE))) / ((float) ao_data.bps);
}