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#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 "libaf/reorder_ch.h"
#include "mpbswap.h"
static ad_info_t info =
{
"FFmpeg/libavcodec audio decoders",
"ffmpeg",
"Nick Kurshev",
"ffmpeg.sf.net",
""
};
LIBAD_EXTERN(ffmpeg)
#define assert(x)
#include "libavcodec/avcodec.h"
extern int avcodec_initialized;
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_initialized){
avcodec_init();
avcodec_register_all();
avcodec_initialized=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;
lavc_context->sample_rate = sh_audio->samplerate;
lavc_context->bit_rate = sh_audio->i_bps * 8;
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->request_channels = audio_output_channels;
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;
if (sh_audio->wf->nSamplesPerSec)
sh_audio->samplerate=sh_audio->wf->nSamplesPerSec;
if (sh_audio->wf->nAvgBytesPerSec)
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=maxlen;
double pts;
int x=ds_get_packet_pts(sh_audio->ds,&start, &pts);
if(x<=0) break; // error
if (pts != MP_NOPTS_VALUE) {
sh_audio->pts = pts;
sh_audio->pts_bytes = 0;
}
y=avcodec_decode_audio2(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){
if (((AVCodecContext *)sh_audio->context)->channels >= 5) {
int src_ch_layout = AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT;
const char *codec=((AVCodecContext*)sh_audio->context)->codec->name;
if (!strcasecmp(codec, "ac3"))
src_ch_layout = AF_CHANNEL_LAYOUT_LAVC_AC3_DEFAULT;
else if (!strcasecmp(codec, "dca"))
src_ch_layout = AF_CHANNEL_LAYOUT_LAVC_DCA_DEFAULT;
else if (!strcasecmp(codec, "libfaad")
|| !strcasecmp(codec, "mpeg4aac"))
src_ch_layout = AF_CHANNEL_LAYOUT_AAC_DEFAULT;
else if (!strcasecmp(codec, "liba52"))
src_ch_layout = AF_CHANNEL_LAYOUT_LAVC_LIBA52_DEFAULT;
else if (!strcasecmp(codec, "vorbis"))
src_ch_layout = AF_CHANNEL_LAYOUT_LAVC_VORBIS_DEFAULT;
else
src_ch_layout = AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT;
reorder_channel_nch(buf, src_ch_layout,
AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
((AVCodecContext *)sh_audio->context)->channels,
len2 / 2, 2);
}
//len=len2;break;
if(len<0) len=len2; else len+=len2;
buf+=len2;
maxlen -= len2;
sh_audio->pts_bytes += len2;
}
mp_dbg(MSGT_DECAUDIO,MSGL_DBG2,"Decoded %d -> %d \n",y,len2);
}
return len;
}
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