view libmpdemux/demux_aac.c @ 23510:a6c619ee9d30

Teletext support for tv:// (v4l and v4l2 only) modified patch from Otvos Attila oattila at chello dot hu Module uses zvbi library for all low-level VBI operations (like I/O with vbi device, converting vbi pages into usefull vbi_page stuctures, rendering them into RGB32 images). All teletext related stuff (except properties, slave commands and rendering osd in text mode or RGB32 rendered teletext pages in spu mode) is implemented in tvi_vbi.c New properties: teletext_page - switching between pages teletext_mode - switch between on/off/opaque/transparent modes teletext_format - (currently read-only) allows to get format info (black/white,gray,text) teletext_half_page - trivial zooming (displaying top/bottom half of teletext page) New slave commands: teletext_add_dec - user interface for jumping to any page by editing page number interactively teletext_go_link - goes though links, specified on current page

#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 "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;

/// \param srate (out) sample rate
/// \param num (out) number of audio frames in this ADTS frame
/// \return size of the ADTS frame in bytes
/// aac_parse_frames needs a buffer at least 8 bytes long
int aac_parse_frame(uint8_t *buf, int *srate, int *num)
{
	int i = 0, sr, fl = 0, id;
	static int srates[] = {96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000, 0, 0, 0};
	
	if((buf[i] != 0xFF) || ((buf[i+1] & 0xF6) != 0xF0))
		return 0;
	
	id = (buf[i+1] >> 3) & 0x01;	//id=1 mpeg2, 0: mpeg4
	sr = (buf[i+2] >> 2)  & 0x0F;
	if(sr > 11)
		return 0;
	*srate = srates[sr];

	fl = ((buf[i+3] & 0x03) << 11) | (buf[i+4] << 3) | ((buf[i+5] >> 5) & 0x07);
	*num = (buf[i+6] & 0x02) + 1;

	return fl;
}

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 = (uint8_t*) 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;

	if(priv->buf)
		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);
	sh->ds = demuxer->audio;
	sh->format = mmioFOURCC('M', 'P', '4', 'A');
	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 & 1) ? 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;
		}
	}
}


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
};