view libmpdemux/demux_aac.c @ 22830:1d4a455af876

Set CONFIG_EBP_AVAILABLE, CONFIG_EBX_AVAILABLE for FFmpeg After FFmpeg r8549 these variables are used in libavcodec to determine whether x86 inline asm sections using these registers or requiring a certain total number of total free registers are enabled. Because they were not set by MPlayer configure some H264 decoding optimizations were disabled after that FFmpeg version. This change sets the variables to true unconditionally which should restore previous behavior. Adding proper detection is left for later. EBX should always be available because internal libavcodec is never compiled with PIC. However if -fomit-frame-pointer is not used because of --enable-debug then EBP is not available. Thus proper detection would be preferable to fix compilation with --enable-debug on x86. Currently the variables are also set on non-x86 which should be harmless even if somewhat ugly.

#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
};