annotate libao2/ao_arts.c @ 9009:f2d4324561b2

1) Removed the code to remove {} from sub_read_line_ssa 2) Put this code in a separated function: sub_pp_ssa 3) After recoding added a call to sub_pp_ssa. To be more acqurate: I added a pointer to function to hold a postprocessing function to be invoked after using iconv. There are one pointer for each format, if this pointer isn.t NULL the post processing function is called. patch by Salvador Eduardo Tropea <[e-mail]> help by <[e-mail]>

/*
 * 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 "audio_out.h"
#include "audio_out_internal.h"
#include "afmt.h"
#include "../config.h"
#include "../mp_msg.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 <[e-mail]>",
    ""
};

LIBAO_EXTERN(arts)

static int control(int cmd, int 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, "AO: [arts] %s\n", arts_error_text(err));
		return 0;
	}
	mp_msg(MSGT_AO, MSGL_INFO, "AO: [arts] Connected to sound server\n");

	/*
	 * 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, "AO: [arts] Unable to open a stream\n");
		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, "AO: [arts] Stream opened\n");

	mp_msg(MSGT_AO, MSGL_INFO,"AO: [arts] buffer size: %d\n",
	    ao_data.buffersize);
	mp_msg(MSGT_AO, MSGL_INFO,"AO: [arts] packet size: %d\n",
	    arts_stream_get(stream, ARTS_P_PACKET_SIZE));

	return 1;
}

static void uninit()
{
	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);
}