libmp3lame: use avpriv_mpegaudio_decode_header() for output frame parsing
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@ -579,7 +579,7 @@ OBJS-$(CONFIG_LIBGSM_DECODER) += libgsm.o
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OBJS-$(CONFIG_LIBGSM_ENCODER) += libgsm.o
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OBJS-$(CONFIG_LIBGSM_MS_DECODER) += libgsm.o
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OBJS-$(CONFIG_LIBGSM_MS_ENCODER) += libgsm.o
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OBJS-$(CONFIG_LIBMP3LAME_ENCODER) += libmp3lame.o
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OBJS-$(CONFIG_LIBMP3LAME_ENCODER) += libmp3lame.o mpegaudiodecheader.o
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OBJS-$(CONFIG_LIBOPENCORE_AMRNB_DECODER) += libopencore-amr.o
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OBJS-$(CONFIG_LIBOPENCORE_AMRNB_ENCODER) += libopencore-amr.o
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OBJS-$(CONFIG_LIBOPENCORE_AMRWB_DECODER) += libopencore-amr.o
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@ -29,6 +29,7 @@
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#include "libavutil/opt.h"
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#include "avcodec.h"
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#include "mpegaudio.h"
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#include "mpegaudiodecheader.h"
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#include <lame/lame.h>
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#define BUFFER_SIZE (7200 + 2 * MPA_FRAME_SIZE + MPA_FRAME_SIZE / 4)
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@ -100,65 +101,11 @@ static const int sSampleRates[] = {
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44100, 48000, 32000, 22050, 24000, 16000, 11025, 12000, 8000, 0
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};
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static const int sBitRates[2][3][15] = {
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{
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{ 0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448 },
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{ 0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384 },
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{ 0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320 }
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},
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{
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{ 0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256 },
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{ 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 },
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{ 0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 }
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},
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};
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static const int sSamplesPerFrame[2][3] = {
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{ 384, 1152, 1152 },
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{ 384, 1152, 576 }
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};
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static const int sBitsPerSlot[3] = { 32, 8, 8 };
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static int mp3len(void *data, int *samplesPerFrame, int *sampleRate)
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{
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uint32_t header = AV_RB32(data);
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int layerID = 3 - ((header >> 17) & 0x03);
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int bitRateID = ((header >> 12) & 0x0f);
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int sampleRateID = ((header >> 10) & 0x03);
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int bitsPerSlot = sBitsPerSlot[layerID];
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int isPadded = ((header >> 9) & 0x01);
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static int const mode_tab[4] = { 2, 3, 1, 0 };
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int mode = mode_tab[(header >> 19) & 0x03];
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int mpeg_id = mode > 0;
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int temp0, temp1, bitRate;
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if (((header >> 21) & 0x7ff) != 0x7ff || mode == 3 || layerID == 3 ||
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sampleRateID == 3) {
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return -1;
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}
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if (!samplesPerFrame)
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samplesPerFrame = &temp0;
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if (!sampleRate)
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sampleRate = &temp1;
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//*isMono = ((header >> 6) & 0x03) == 0x03;
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*sampleRate = sSampleRates[sampleRateID] >> mode;
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bitRate = sBitRates[mpeg_id][layerID][bitRateID] * 1000;
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*samplesPerFrame = sSamplesPerFrame[mpeg_id][layerID];
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//av_log(NULL, AV_LOG_DEBUG,
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// "sr:%d br:%d spf:%d l:%d m:%d\n",
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// *sampleRate, bitRate, *samplesPerFrame, layerID, mode);
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return *samplesPerFrame * bitRate / (bitsPerSlot * *sampleRate) + isPadded;
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}
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static int MP3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
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int buf_size, void *data)
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{
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Mp3AudioContext *s = avctx->priv_data;
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MPADecodeHeader hdr;
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int len;
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int lame_result;
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@ -193,7 +140,11 @@ static int MP3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
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if (s->buffer_index < 4)
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return 0;
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len = mp3len(s->buffer, NULL, NULL);
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if (avpriv_mpegaudio_decode_header(&hdr, AV_RB32(s->buffer))) {
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av_log(avctx, AV_LOG_ERROR, "free format output not supported\n");
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return -1;
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}
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len = hdr.frame_size;
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av_dlog(avctx, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len,
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s->buffer_index);
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if (len <= s->buffer_index) {
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