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mpv/libmpcodecs/ad_ffmpeg.c

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/*
* This file is part of MPlayer.
*
* MPlayer is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* MPlayer is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with MPlayer; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include <stdio.h>
#include <stdlib.h>
#include <unistd.h>
#include "config.h"
#include "mp_msg.h"
#include "options.h"
#include "ad_internal.h"
#include "libaf/reorder_ch.h"
#include "mpbswap.h"
static const 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 setup_format(sh_audio_t *sh_audio, const AVCodecContext *lavc_context)
{
int broken_srate = 0;
int samplerate = lavc_context->sample_rate;
int sample_format = sh_audio->sample_format;
switch (lavc_context->sample_fmt) {
case SAMPLE_FMT_U8: sample_format = AF_FORMAT_U8; break;
case SAMPLE_FMT_S16: sample_format = AF_FORMAT_S16_NE; break;
case SAMPLE_FMT_S32: sample_format = AF_FORMAT_S32_NE; break;
case SAMPLE_FMT_FLT: sample_format = AF_FORMAT_FLOAT_NE; break;
default:
mp_msg(MSGT_DECAUDIO, MSGL_FATAL, "Unsupported sample format\n");
}
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 (lavc_context->codec_id == CODEC_ID_AAC &&
samplerate == 2*sh_audio->wf->nSamplesPerSec) {
broken_srate = 1;
} else if (sh_audio->wf->nSamplesPerSec)
samplerate=sh_audio->wf->nSamplesPerSec;
}
if (lavc_context->channels != sh_audio->channels ||
samplerate != sh_audio->samplerate ||
sample_format != sh_audio->sample_format) {
sh_audio->channels=lavc_context->channels;
sh_audio->samplerate=samplerate;
sh_audio->sample_format = sample_format;
sh_audio->samplesize=af_fmt2bits(sh_audio->sample_format)/ 8;
if (broken_srate)
mp_msg(MSGT_DECAUDIO, MSGL_WARN,
"Ignoring broken container sample rate for AAC with SBR\n");
return 1;
}
return 0;
}
static int init(sh_audio_t *sh_audio)
{
struct MPOpts *opts = sh_audio->opts;
int tries = 0;
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_tmsg(MSGT_DECAUDIO,MSGL_ERR,"Cannot find codec '%s' in libavcodec...\n",sh_audio->codec->dll);
return 0;
}
lavc_context = avcodec_alloc_context();
sh_audio->context=lavc_context;
lavc_context->drc_scale = opts->drc_level;
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_coded_sample = sh_audio->wf->wBitsPerSample;
}
lavc_context->request_channels = audio_output_channels;
lavc_context->codec_tag = sh_audio->format; //FOURCC
lavc_context->codec_type = CODEC_TYPE_AUDIO;
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_tmsg(MSGT_DECAUDIO,MSGL_ERR, "Could not open codec.\n");
return 0;
}
mp_msg(MSGT_DECAUDIO,MSGL_V,"INFO: libavcodec \"%s\" init OK!\n", lavc_codec->name);
// 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)
do {
x=decode_audio(sh_audio,sh_audio->a_buffer,1,sh_audio->a_buffer_size);
} while (x <= 0 && tries++ < 5);
if(x>0) sh_audio->a_buffer_len=x;
sh_audio->i_bps=lavc_context->bit_rate/8;
if (sh_audio->wf && sh_audio->wf->nAvgBytesPerSec)
sh_audio->i_bps=sh_audio->wf->nAvgBytesPerSec;
switch (lavc_context->sample_fmt) {
case SAMPLE_FMT_U8:
case SAMPLE_FMT_S16:
case SAMPLE_FMT_S32:
case SAMPLE_FMT_FLT:
break;
default:
return 0;
}
return 1;
}
static void uninit(sh_audio_t *sh)
{
AVCodecContext *lavc_context = sh->context;
if (avcodec_close(lavc_context) < 0)
mp_tmsg(MSGT_DECVIDEO, MSGL_ERR, "Could not close codec.\n");
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);
ds_clear_parser(sh->ds);
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){
AVPacket pkt;
int len2=maxlen;
double pts;
int x=ds_get_packet_pts(sh_audio->ds,&start, &pts);
if(x<=0) {
start = NULL;
x = 0;
ds_parse(sh_audio->ds, &start, &x, MP_NOPTS_VALUE, 0);
if (x <= 0)
break; // error
} else {
int in_size = x;
int consumed = ds_parse(sh_audio->ds, &start, &x, pts, 0);
sh_audio->ds->buffer_pos -= in_size - consumed;
}
av_init_packet(&pkt);
pkt.data = start;
pkt.size = x;
if (pts != MP_NOPTS_VALUE) {
sh_audio->pts = pts;
sh_audio->pts_bytes = 0;
}
y=avcodec_decode_audio3(sh_audio->context,(int16_t*)buf,&len2,&pkt);
//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(!sh_audio->parser && y<x)
sh_audio->ds->buffer_pos+=y-x; // put back data (HACK!)
if(len2>0){
if (((AVCodecContext *)sh_audio->context)->channels >= 5) {
int samplesize = av_get_bits_per_sample_format(((AVCodecContext *)
sh_audio->context)->sample_fmt) / 8;
reorder_channel_nch(buf, AF_CHANNEL_LAYOUT_LAVC_DEFAULT,
AF_CHANNEL_LAYOUT_MPLAYER_DEFAULT,
((AVCodecContext *)sh_audio->context)->channels,
len2 / samplesize, samplesize);
}
//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);
if (setup_format(sh_audio, sh_audio->context))
break;
}
return len;
}
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