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authorJustin Ruggles <justin.ruggles@gmail.com>2012-02-16 23:06:54 -0500
committerJustin Ruggles <justin.ruggles@gmail.com>2012-02-20 12:32:30 -0500
commit1f516c045152e428b64e1c69105b447a6c484355 (patch)
treee743dd048286844a80c8592c995ed5b6830b58ae /libavcodec/libmp3lame.c
parente3d2c89e9d5f3cdfda3258526db1f441f0459a4c (diff)
downloadffmpeg-1f516c045152e428b64e1c69105b447a6c484355.tar.gz
libmp3lame: use avpriv_mpegaudio_decode_header() for output frame parsing
Diffstat (limited to 'libavcodec/libmp3lame.c')
-rw-r--r--libavcodec/libmp3lame.c63
1 files changed, 7 insertions, 56 deletions
diff --git a/libavcodec/libmp3lame.c b/libavcodec/libmp3lame.c
index 1e0beb58eb..a15aed3d16 100644
--- a/libavcodec/libmp3lame.c
+++ b/libavcodec/libmp3lame.c
@@ -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) {