Files
mpv/libmpcodecs/ad_ffmpeg.c
T
diego d4e9855511 Using channel count, samplerate and input bps values from the container
instead of the decoder breaks some DTS samples where the container says
the audio has 6 channels but the decoder gives 2. In this case take the
number of channels from the decoder instead, the output will almost
certainly be badly garbled anyway if the number of channels is wrong.
patch by Uoti Urpala, uoti <<.>> urpala <<@>> pp1 <<.>> inet <<.>> fi


git-svn-id: svn://svn.mplayerhq.hu/mplayer/trunk@18151 b3059339-0415-0410-9bf9-f77b7e298cf2
2006-04-19 20:12:01 +00:00

167 lines
4.7 KiB
C

#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include "config.h"
#include "mp_msg.h"
#include "help_mp.h"
#include "ad_internal.h"
#include "bswap.h"
static ad_info_t info =
{
"FFmpeg/libavcodec audio decoders",
"ffmpeg",
"Nick Kurshev",
"ffmpeg.sf.net",
""
};
LIBAD_EXTERN(ffmpeg)
#define assert(x)
#ifdef USE_LIBAVCODEC_SO
#include <ffmpeg/avcodec.h>
#else
#include "avcodec.h"
#endif
extern int avcodec_inited;
static int preinit(sh_audio_t *sh)
{
sh->audio_out_minsize=AVCODEC_MAX_AUDIO_FRAME_SIZE;
return 1;
}
static int init(sh_audio_t *sh_audio)
{
int x;
AVCodecContext *lavc_context;
AVCodec *lavc_codec;
mp_msg(MSGT_DECAUDIO,MSGL_V,"FFmpeg's libavcodec audio codec\n");
if(!avcodec_inited){
avcodec_init();
avcodec_register_all();
avcodec_inited=1;
}
lavc_codec = (AVCodec *)avcodec_find_decoder_by_name(sh_audio->codec->dll);
if(!lavc_codec){
mp_msg(MSGT_DECAUDIO,MSGL_ERR,MSGTR_MissingLAVCcodec,sh_audio->codec->dll);
return 0;
}
lavc_context = avcodec_alloc_context();
sh_audio->context=lavc_context;
if(sh_audio->wf){
lavc_context->channels = sh_audio->wf->nChannels;
lavc_context->sample_rate = sh_audio->wf->nSamplesPerSec;
lavc_context->bit_rate = sh_audio->wf->nAvgBytesPerSec * 8;
lavc_context->block_align = sh_audio->wf->nBlockAlign;
lavc_context->bits_per_sample = sh_audio->wf->wBitsPerSample;
}
lavc_context->codec_tag = sh_audio->format; //FOURCC
lavc_context->codec_id = lavc_codec->id; // not sure if required, imho not --A'rpi
/* alloc extra data */
if (sh_audio->wf && sh_audio->wf->cbSize > 0) {
lavc_context->extradata = av_mallocz(sh_audio->wf->cbSize + FF_INPUT_BUFFER_PADDING_SIZE);
lavc_context->extradata_size = sh_audio->wf->cbSize;
memcpy(lavc_context->extradata, (char *)sh_audio->wf + sizeof(WAVEFORMATEX),
lavc_context->extradata_size);
}
// for QDM2
if (sh_audio->codecdata_len && sh_audio->codecdata && !lavc_context->extradata)
{
lavc_context->extradata = av_malloc(sh_audio->codecdata_len);
lavc_context->extradata_size = sh_audio->codecdata_len;
memcpy(lavc_context->extradata, (char *)sh_audio->codecdata,
lavc_context->extradata_size);
}
/* open it */
if (avcodec_open(lavc_context, lavc_codec) < 0) {
mp_msg(MSGT_DECAUDIO,MSGL_ERR, MSGTR_CantOpenCodec);
return 0;
}
mp_msg(MSGT_DECAUDIO,MSGL_V,"INFO: libavcodec init OK!\n");
// printf("\nFOURCC: 0x%X\n",sh_audio->format);
if(sh_audio->format==0x3343414D){
// MACE 3:1
sh_audio->ds->ss_div = 2*3; // 1 samples/packet
sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet
} else
if(sh_audio->format==0x3643414D){
// MACE 6:1
sh_audio->ds->ss_div = 2*6; // 1 samples/packet
sh_audio->ds->ss_mul = 2*sh_audio->wf->nChannels; // 1 byte*ch/packet
}
// Decode at least 1 byte: (to get header filled)
x=decode_audio(sh_audio,sh_audio->a_buffer,1,sh_audio->a_buffer_size);
if(x>0) sh_audio->a_buffer_len=x;
sh_audio->channels=lavc_context->channels;
sh_audio->samplerate=lavc_context->sample_rate;
sh_audio->i_bps=lavc_context->bit_rate/8;
if(sh_audio->wf){
// If the decoder uses the wrong number of channels all is lost anyway.
// sh_audio->channels=sh_audio->wf->nChannels;
sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec;
}
sh_audio->samplesize=2;
return 1;
}
static void uninit(sh_audio_t *sh)
{
AVCodecContext *lavc_context = sh->context;
if (avcodec_close(lavc_context) < 0)
mp_msg(MSGT_DECVIDEO, MSGL_ERR, MSGTR_CantCloseCodec);
av_freep(&lavc_context->extradata);
av_freep(&lavc_context);
}
static int control(sh_audio_t *sh,int cmd,void* arg, ...)
{
AVCodecContext *lavc_context = sh->context;
switch(cmd){
case ADCTRL_RESYNC_STREAM:
avcodec_flush_buffers(lavc_context);
return CONTROL_TRUE;
}
return CONTROL_UNKNOWN;
}
static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
{
unsigned char *start=NULL;
int y,len=-1;
while(len<minlen){
int len2=0;
int x=ds_get_packet(sh_audio->ds,&start);
if(x<=0) break; // error
y=avcodec_decode_audio(sh_audio->context,(int16_t*)buf,&len2,start,x);
//printf("return:%d samples_out:%d bitstream_in:%d sample_sum:%d\n", y, len2, x, len); fflush(stdout);
if(y<0){ mp_msg(MSGT_DECAUDIO,MSGL_V,"lavc_audio: error\n");break; }
if(y<x) sh_audio->ds->buffer_pos+=y-x; // put back data (HACK!)
if(len2>0){
//len=len2;break;
if(len<0) len=len2; else len+=len2;
buf+=len2;
}
mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"Decoded %d -> %d \n",y,len2);
}
return len;
}