view libao2/ao_arts.c @ 27518:e54c9b7eb0d8

Revert bad changes to SSA/ASS subtitle packet format The following commits are reverted partially or completely: "a valid ASS line contains 9 ',' before actual text" "demux_mkv: output correctly formated ASS packets" "libass: add a new ass_process_data() to process demuxed subtitle packets" These commits converted the internal representation of SSA/ASS subtitle packets from the format used by Matroska to a custom format where each packet has contents exactly matching one line in complete SSA script files. AFAIK no files natively use such a format for muxed subtitles. The stated reason for this change was to use a format that could in principle be muxed into a maximal number of containers. SSA subtitles do not have an implicit duration so both start time and duration or end time need to be specified explicitly; the new format moved timing information inside the codec packet data so it could be muxed without modification into containers that can represent only start time at the container level. However such a change is wrong from the viewpoint of program architecture. Timing information belongs to the demuxer level, but these commits moved not only the duration but also the authoritative value of the start time to inside the codec data. Additionally the new format lost the value of the Matroska ReadOrder field which is used by MPlayer. This commit changes the internal packet format back to that used by Matroska and makes the internal Matroska demuxer output that format again. Libavformat still outputs the "new" format; it could be converted back to the Matroska format in demux_lavf.c, but I'm not adding that code at least yet. The current lavf code has similar problems as the reverted code in MPlayer, and it also currently fails to provide any way to access the value of the ReadOrder field. I hope that the lavf side will be improved; if it isn't conversion can be added later. For now I'll make MPlayer default to the internal Matroska demuxer instead of the lavf one in a separate commit.

/*
 * aRts audio output driver for MPlayer
 *
 * copyright (c) 2002 Michele Balistreri <[e-mail]>
 *
 * This file is part of MPlayer.
 *
 * MPlayer is free software; you can redistribute it and/or modify
 * it under the terms of the GNU General Public License as published by
 * the Free Software Foundation; either version 2 of the License, or
 * (at your option) any later version.
 *
 * MPlayer is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
 * GNU General Public License for more details.
 *
 * You should have received a copy of the GNU General Public License along
 * with MPlayer; if not, write to the Free Software Foundation, Inc.,
 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
 */

#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);
}