This source file includes following definitions.
- mp3_header_decompress
#include "libavutil/common.h"
#include "libavutil/intreadwrite.h"
#include "bsf.h"
#include "bsf_internal.h"
#include "mpegaudiodecheader.h"
#include "mpegaudiodata.h"
static int mp3_header_decompress(AVBSFContext *ctx, AVPacket *out)
{
AVPacket *in;
uint32_t header;
int sample_rate= ctx->par_in->sample_rate;
int sample_rate_index=0;
int lsf, mpeg25, bitrate_index, frame_size, ret;
uint8_t *buf;
int buf_size;
ret = ff_bsf_get_packet(ctx, &in);
if (ret < 0)
return ret;
buf = in->data;
buf_size = in->size;
header = AV_RB32(buf);
if(ff_mpa_check_header(header) >= 0){
av_packet_move_ref(out, in);
av_packet_free(&in);
return 0;
}
if(ctx->par_in->extradata_size != 15 || strcmp(ctx->par_in->extradata, "FFCMP3 0.0")){
av_log(ctx, AV_LOG_ERROR, "Extradata invalid %d\n", ctx->par_in->extradata_size);
ret = AVERROR(EINVAL);
goto fail;
}
header= AV_RB32(ctx->par_in->extradata+11) & MP3_MASK;
lsf = sample_rate < (24000+32000)/2;
mpeg25 = sample_rate < (12000+16000)/2;
sample_rate_index= (header>>10)&3;
if (sample_rate_index == 3) {
ret = AVERROR_INVALIDDATA;
goto fail;
}
sample_rate= avpriv_mpa_freq_tab[sample_rate_index] >> (lsf + mpeg25);
for(bitrate_index=2; bitrate_index<30; bitrate_index++){
frame_size = avpriv_mpa_bitrate_tab[lsf][2][bitrate_index>>1];
frame_size = (frame_size * 144000) / (sample_rate << lsf) + (bitrate_index&1);
if(frame_size == buf_size + 4)
break;
if(frame_size == buf_size + 6)
break;
}
if(bitrate_index == 30){
av_log(ctx, AV_LOG_ERROR, "Could not find bitrate_index.\n");
ret = AVERROR(EINVAL);
goto fail;
}
header |= (bitrate_index&1)<<9;
header |= (bitrate_index>>1)<<12;
header |= (frame_size == buf_size + 4)<<16;
ret = av_new_packet(out, frame_size);
if (ret < 0)
goto fail;
ret = av_packet_copy_props(out, in);
if (ret < 0) {
av_packet_unref(out);
goto fail;
}
memcpy(out->data + frame_size - buf_size, buf, buf_size + AV_INPUT_BUFFER_PADDING_SIZE);
if(ctx->par_in->channels==2){
uint8_t *p= out->data + frame_size - buf_size;
if(lsf){
FFSWAP(int, p[1], p[2]);
header |= (p[1] & 0xC0)>>2;
p[1] &= 0x3F;
}else{
header |= p[1] & 0x30;
p[1] &= 0xCF;
}
}
AV_WB32(out->data, header);
ret = 0;
fail:
av_packet_free(&in);
return ret;
}
static const enum AVCodecID codec_ids[] = {
AV_CODEC_ID_MP3, AV_CODEC_ID_NONE,
};
const AVBitStreamFilter ff_mp3_header_decompress_bsf = {
.name = "mp3decomp",
.filter = mp3_header_decompress,
.codec_ids = codec_ids,
};