view libao2/ao_arts.c @ 11619:179138947307

This patch contains bugfixes for the esd audio output driver that I uncovered while trying to send sound to a remote esd server over a wireless (11 mbs, just enough to handle to sound) link. First, the sound was full "ticking" sounds. I found a bug that prevented the "send the remainder of this block" code from ever being called - so large chunks of audio were simply being ignored. Fixing this bug removed the "ticking" from audio streams. Fixing this bug, however, uncovered another problem - when the socket buffer was full, doing a blocking write to finish the buffer would take far too long and would turn video into a chunky mess. I'd imagine this blocking write would be fine for an audio-only stream, but it turns out to hold up the video far too much. The solution in this patch is to write as much data as possible to the socket, and then return as soon as possible, reporting the number of bytes actually written accurately back to mplayer. I've tested it on both local and remote esd servers, and it works well. Patch by Benjamin Osheroff <[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, 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, "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);
}