libmp3lame: use avpriv_mpegaudio_decode_header() for output frame parsing

This commit is contained in:
Justin Ruggles 2012-02-16 23:06:54 -05:00
parent e3d2c89e9d
commit 1f516c0451
2 changed files with 8 additions and 57 deletions

View File

@ -579,7 +579,7 @@ OBJS-$(CONFIG_LIBGSM_DECODER) += libgsm.o
OBJS-$(CONFIG_LIBGSM_ENCODER) += libgsm.o
OBJS-$(CONFIG_LIBGSM_MS_DECODER) += libgsm.o
OBJS-$(CONFIG_LIBGSM_MS_ENCODER) += libgsm.o
OBJS-$(CONFIG_LIBMP3LAME_ENCODER) += libmp3lame.o
OBJS-$(CONFIG_LIBMP3LAME_ENCODER) += libmp3lame.o mpegaudiodecheader.o
OBJS-$(CONFIG_LIBOPENCORE_AMRNB_DECODER) += libopencore-amr.o
OBJS-$(CONFIG_LIBOPENCORE_AMRNB_ENCODER) += libopencore-amr.o
OBJS-$(CONFIG_LIBOPENCORE_AMRWB_DECODER) += libopencore-amr.o

View File

@ -29,6 +29,7 @@
#include "libavutil/opt.h"
#include "avcodec.h"
#include "mpegaudio.h"
#include "mpegaudiodecheader.h"
#include <lame/lame.h>
#define BUFFER_SIZE (7200 + 2 * MPA_FRAME_SIZE + MPA_FRAME_SIZE / 4)
@ -100,65 +101,11 @@ static const int sSampleRates[] = {
44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
};
static const int sBitRates[2][3][15] = {
{
{ 0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448 },
{ 0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384 },
{ 0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320 }
},
{
{ 0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256 },
{ 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 },
{ 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 }
},
};
static const int sSamplesPerFrame[2][3] = {
{ 384, 1152, 1152 },
{ 384, 1152, 576 }
};
static const int sBitsPerSlot[3] = { 32, 8, 8 };
static int mp3len(void *data, int *samplesPerFrame, int *sampleRate)
{
uint32_t header = AV_RB32(data);
int layerID = 3 - ((header >> 17) & 0x03);
int bitRateID = ((header >> 12) & 0x0f);
int sampleRateID = ((header >> 10) & 0x03);
int bitsPerSlot = sBitsPerSlot[layerID];
int isPadded = ((header >> 9) & 0x01);
static int const mode_tab[4] = { 2, 3, 1, 0 };
int mode = mode_tab[(header >> 19) & 0x03];
int mpeg_id = mode > 0;
int temp0, temp1, bitRate;
if (((header >> 21) & 0x7ff) != 0x7ff || mode == 3 || layerID == 3 ||
sampleRateID == 3) {
return -1;
}
if (!samplesPerFrame)
samplesPerFrame = &temp0;
if (!sampleRate)
sampleRate = &temp1;
//*isMono = ((header >> 6) & 0x03) == 0x03;
*sampleRate = sSampleRates[sampleRateID] >> mode;
bitRate = sBitRates[mpeg_id][layerID][bitRateID] * 1000;
*samplesPerFrame = sSamplesPerFrame[mpeg_id][layerID];
//av_log(NULL, AV_LOG_DEBUG,
// "sr:%d br:%d spf:%d l:%d m:%d\n",
// *sampleRate, bitRate, *samplesPerFrame, layerID, mode);
return *samplesPerFrame * bitRate / (bitsPerSlot * *sampleRate) + isPadded;
}
static int MP3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
int buf_size, void *data)
{
Mp3AudioContext *s = avctx->priv_data;
MPADecodeHeader hdr;
int len;
int lame_result;
@ -193,7 +140,11 @@ static int MP3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
if (s->buffer_index < 4)
return 0;
len = mp3len(s->buffer, NULL, NULL);
if (avpriv_mpegaudio_decode_header(&hdr, AV_RB32(s->buffer))) {
av_log(avctx, AV_LOG_ERROR, "free format output not supported\n");
return -1;
}
len = hdr.frame_size;
av_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len,
s->buffer_index);
if (len <= s->buffer_index) {