view libmpdemux/demux_aac.c @ 36441:b75ebb89d803

Workaround VDPAU decode errors on aspect change on NVidia. The NVidia driver seems to expect a decoder reinit on aspect change, otherwise giving a nonsense VDP_STATUS_INVALID_SIZE error. Since decode and display can run out of sync, we do not in fact know when an aspect change will happen during decode but only when we want to display that decoded frame, and with threaded decoding these will differ significantly. So just catch the error and retry decoding instead, this also has the advantage of not affecting (and possibly costing performance) drivers without this issue.
author reimar
date Sun, 08 Dec 2013 15:07:00 +0000
parents 8fa2f43cb760
children 92dd1764392a
line wrap: on
line source

/*
 * 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 <stdio.h>
#include <stdlib.h>
#include <string.h>

#include "config.h"
#include "mp_msg.h"
#include "help_mp.h"

#include "stream/stream.h"
#include "demuxer.h"
#include "parse_es.h"
#include "stheader.h"
#include "aac_hdr.h"
#include "ms_hdr.h"

typedef struct {
	uint8_t *buf;
	uint64_t size;	/// amount of time of data packets pushed to demuxer->audio (in bytes)
	float time;	/// amount of time elapsed based upon samples_per_frame/sample_rate (in milliseconds)
	float last_pts; /// last pts seen
	int bitrate;	/// bitrate computed as size/time
} aac_priv_t;

static int demux_aac_init(demuxer_t *demuxer)
{
	aac_priv_t *priv;

	priv = calloc(1, sizeof(aac_priv_t));
	if(!priv)
		return 0;

	priv->buf = malloc(8);
	if(!priv->buf)
	{
		free(priv);
		return 0;
	}

	demuxer->priv = priv;
	return 1;
}

static void demux_close_aac(demuxer_t *demuxer)
{
	aac_priv_t *priv = (aac_priv_t *) demuxer->priv;

	if(!priv)
		return;

	free(priv->buf);

	free(demuxer->priv);

	return;
}

/// returns DEMUXER_TYPE_AAC if it finds 8 ADTS frames in 32768 bytes, 0 otherwise
static int demux_aac_probe(demuxer_t *demuxer)
{
	int cnt = 0, c, len, srate, num;
	off_t init, probed;
	aac_priv_t *priv;

	if(! demux_aac_init(demuxer))
	{
		mp_msg(MSGT_DEMUX, MSGL_ERR, "COULDN'T INIT aac_demux, exit\n");
		return 0;
	}

	priv = (aac_priv_t *) demuxer->priv;

	init = probed = stream_tell(demuxer->stream);
	while(probed-init <= 32768 && cnt < 8)
	{
		c = 0;
		while(c != 0xFF)
		{
			c = stream_read_char(demuxer->stream);
			if(c < 0)
				goto fail;
		}
		priv->buf[0] = 0xFF;
		if(stream_read(demuxer->stream, &(priv->buf[1]), 7) < 7)
			goto fail;

		len = aac_parse_frame(priv->buf, &srate, &num);
		if(len > 0)
		{
			cnt++;
			stream_skip(demuxer->stream, len - 8);
		}
		probed = stream_tell(demuxer->stream);
	}

	stream_seek(demuxer->stream, init);
	if(cnt < 8)
		goto fail;

	mp_msg(MSGT_DEMUX, MSGL_V, "demux_aac_probe, INIT: %"PRIu64", PROBED: %"PRIu64", cnt: %d\n", init, probed, cnt);
	return DEMUXER_TYPE_AAC;

fail:
	mp_msg(MSGT_DEMUX, MSGL_V, "demux_aac_probe, failed to detect an AAC stream\n");
	return 0;
}

static demuxer_t* demux_aac_open(demuxer_t *demuxer)
{
	sh_audio_t *sh;

	sh = new_sh_audio(demuxer, 0, NULL);
	sh->ds = demuxer->audio;
	sh->format = mmioFOURCC('M', 'P', '4', 'A');
	demuxer->audio->id = 0;
	demuxer->audio->sh = sh;

	demuxer->filepos = stream_tell(demuxer->stream);

	return demuxer;
}

static int demux_aac_fill_buffer(demuxer_t *demuxer, demux_stream_t *ds)
{
	aac_priv_t *priv = (aac_priv_t *) demuxer->priv;
	demux_packet_t *dp;
	int c1, c2, len, srate, num;
	float tm = 0;

	if(demuxer->stream->eof || (demuxer->movi_end && stream_tell(demuxer->stream) >= demuxer->movi_end))
        	return 0;

	while(! demuxer->stream->eof)
	{
		c1 = c2 = 0;
		while(c1 != 0xFF)
		{
			c1 = stream_read_char(demuxer->stream);
			if(c1 < 0)
				return 0;
		}
		c2 = stream_read_char(demuxer->stream);
		if(c2 < 0)
			return 0;
		if((c2 & 0xF6) != 0xF0)
			continue;

		priv->buf[0] = (unsigned char) c1;
		priv->buf[1] = (unsigned char) c2;
		if(stream_read(demuxer->stream, &(priv->buf[2]), 6) < 6)
			return 0;

		len = aac_parse_frame(priv->buf, &srate, &num);
		if(len > 0)
		{
			dp = new_demux_packet(len);
			if(! dp)
			{
				mp_msg(MSGT_DEMUX, MSGL_ERR, "fill_buffer, NEW_ADD_PACKET(%d)FAILED\n", len);
				return 0;
			}


			memcpy(dp->buffer, priv->buf, 8);
			stream_read(demuxer->stream, &(dp->buffer[8]), len-8);
			if(srate)
				tm = (float) (num * 1024.0/srate);
			priv->last_pts += tm;
			dp->pts = priv->last_pts;
			//fprintf(stderr, "\nPTS: %.3f\n", dp->pts);
			ds_add_packet(demuxer->audio, dp);
			priv->size += len;
			priv->time += tm;

			priv->bitrate = (int) (priv->size / priv->time);
			demuxer->filepos = stream_tell(demuxer->stream);

			return len;
		}
		else
			stream_skip(demuxer->stream, -6);
	}

	return 0;
}


//This is an almost verbatim copy of high_res_mp3_seek(), from demux_audio.c
static void demux_aac_seek(demuxer_t *demuxer, float rel_seek_secs, float audio_delay, int flags)
{
	aac_priv_t *priv = (aac_priv_t *) demuxer->priv;
	demux_stream_t *d_audio=demuxer->audio;
	sh_audio_t *sh_audio=d_audio->sh;
	float time;

	ds_free_packs(d_audio);

	time = (flags & SEEK_ABSOLUTE) ? rel_seek_secs - priv->last_pts : rel_seek_secs;
	if(time < 0)
	{
		stream_seek(demuxer->stream, demuxer->movi_start);
		time = priv->last_pts + time;
		priv->last_pts = 0;
	}

	if(time > 0)
	{
		int len, nf, srate, num;

		nf = time * sh_audio->samplerate/1024;

		while(nf > 0)
		{
			if(stream_read(demuxer->stream,priv->buf, 8) < 8)
				break;
			len = aac_parse_frame(priv->buf, &srate, &num);
			if(len <= 0)
			{
				stream_skip(demuxer->stream, -7);
				continue;
			}
			stream_skip(demuxer->stream, len - 8);
			priv->last_pts += (float) (num*1024.0/srate);
			nf -= num;
		}
	}
}


const demuxer_desc_t demuxer_desc_aac = {
  "AAC demuxer",
  "aac",
  "AAC",
  "Nico Sabbi",
  "Raw AAC files ",
  DEMUXER_TYPE_AAC,
  0, // unsafe autodetect
  demux_aac_probe,
  demux_aac_fill_buffer,
  demux_aac_open,
  demux_close_aac,
  demux_aac_seek,
  NULL
};