view libmpdemux/demux_aac.c @ 25957:0a3b2b2cc1c3

Implement test for system byteswap.h header file. The result of this check is required by libavutil library. If it is not defined the library would try to implement its own byte swapping routines in bswap.h . As the routines are with same names, if included, the system definition would replace the function names with the macros. The result can not be compiled and looks like this: # 42 "../libavutil/bswap.h" -static av_always_inline uint16_t bswap_16(uint16_t x) +static __attribute__((always_inline)) inline uint16_t (__extension__ ({ register unsigned short int __v, __x = (uint16_t x); if (__builtin_constant_p (__x)) __v = ((((__x) >> 8) & 0xff) | (((__x) & 0xff) << 8)); else __asm__ ("rorw $8, %w0" : "=r" (__v) : "0" (__x) : "cc"); __v; }))

#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 & 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
};