#include "mpegaudio.h"
#include <lame/lame.h>
-#define BUFFER_SIZE (7200 + 2*MPA_FRAME_SIZE + MPA_FRAME_SIZE/4)
+#define BUFFER_SIZE (7200 + 2 * MPA_FRAME_SIZE + MPA_FRAME_SIZE / 4+1000) // FIXME: Buffer size to small? Adding 1000 to make up for it.
typedef struct Mp3AudioContext {
AVClass *class;
lame_global_flags *gfp;
lame_set_in_samplerate(s->gfp, avctx->sample_rate);
lame_set_out_samplerate(s->gfp, avctx->sample_rate);
lame_set_num_channels(s->gfp, avctx->channels);
- if(avctx->compression_level == FF_COMPRESSION_DEFAULT) {
+ if (avctx->compression_level == FF_COMPRESSION_DEFAULT) {
lame_set_quality(s->gfp, 5);
} else {
lame_set_quality(s->gfp, avctx->compression_level);
}
lame_set_mode(s->gfp, s->stereo ? JOINT_STEREO : MONO);
- lame_set_brate(s->gfp, avctx->bit_rate/1000);
- if(avctx->flags & CODEC_FLAG_QSCALE) {
+ lame_set_brate(s->gfp, avctx->bit_rate / 1000);
+ if (avctx->flags & CODEC_FLAG_QSCALE) {
lame_set_brate(s->gfp, 0);
lame_set_VBR(s->gfp, vbr_default);
- lame_set_VBR_quality(s->gfp, avctx->global_quality/(float)FF_QP2LAMBDA);
+ lame_set_VBR_quality(s->gfp, avctx->global_quality / (float)FF_QP2LAMBDA);
}
lame_set_bWriteVbrTag(s->gfp,0);
#if FF_API_LAME_GLOBAL_OPTS
if (lame_init_params(s->gfp) < 0)
goto err_close;
- avctx->frame_size = lame_get_framesize(s->gfp);
+ avctx->frame_size = lame_get_framesize(s->gfp);
if(!(avctx->coded_frame= avcodec_alloc_frame())) {
lame_close(s->gfp);
return AVERROR(ENOMEM);
}
- avctx->coded_frame->key_frame= 1;
+ avctx->coded_frame->key_frame = 1;
if(AV_SAMPLE_FMT_S32 == avctx->sample_fmt && s->stereo) {
int nelem = 2 * avctx->frame_size;
};
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, 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}
+ {
+ { 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 sSamplesPerFrame[2][3] = {
+ { 384, 1152, 1152 },
+ { 384, 1152, 576 }
};
-static const int sBitsPerSlot[3] = {
- 32,
- 8,
- 8
-};
+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);
+ 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 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) {
+ if (((header >> 21) & 0x7ff) != 0x7ff || mode == 3 || layerID == 3 ||
+ sampleRateID == 3) {
return -1;
}
- if(!samplesPerFrame) samplesPerFrame= &temp0;
- if(!sampleRate ) sampleRate = &temp1;
+ if (!samplesPerFrame)
+ samplesPerFrame = &temp0;
+ if (!sampleRate)
+ sampleRate = &temp1;
-// *isMono = ((header >> 6) & 0x03) == 0x03;
+ //*isMono = ((header >> 6) & 0x03) == 0x03;
- *sampleRate = sSampleRates[sampleRateID]>>mode;
- bitRate = sBitRates[mpeg_id][layerID][bitRateID] * 1000;
+ *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);
+ //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)
+static int MP3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
+ int buf_size, void *data)
{
Mp3AudioContext *s = avctx->priv_data;
int len;
/* lame 3.91 dies on '1-channel interleaved' data */
- if(!data){
+ if (!data){
lame_result= lame_encode_flush(
s->gfp,
s->buffer + s->buffer_index,
}
}
- if(lame_result < 0){
- if(lame_result==-1) {
+ if (lame_result < 0) {
+ if (lame_result == -1) {
/* output buffer too small */
- av_log(avctx, AV_LOG_ERROR, "lame: output buffer too small (buffer index: %d, free bytes: %d)\n", s->buffer_index, BUFFER_SIZE - s->buffer_index);
+ av_log(avctx, AV_LOG_ERROR,
+ "lame: output buffer too small (buffer index: %d, free bytes: %d)\n",
+ s->buffer_index, BUFFER_SIZE - s->buffer_index);
}
return -1;
}
s->buffer_index += lame_result;
- if(s->buffer_index<4)
+ if (s->buffer_index < 4)
return 0;
- len= mp3len(s->buffer, NULL, NULL);
-//av_log(avctx, AV_LOG_DEBUG, "in:%d packet-len:%d index:%d\n", avctx->frame_size, len, s->buffer_index);
- if(len <= s->buffer_index){
+ len = mp3len(s->buffer, NULL, NULL);
+ //av_log(avctx, AV_LOG_DEBUG, "in:%d packet-len:%d index:%d\n",
+ // avctx->frame_size, len, s->buffer_index);
+ if (len <= s->buffer_index) {
memcpy(frame, s->buffer, len);
s->buffer_index -= len;
- memmove(s->buffer, s->buffer+len, s->buffer_index);
- //FIXME fix the audio codec API, so we do not need the memcpy()
-/*for(i=0; i<len; i++){
- av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]);
-}*/
+ memmove(s->buffer, s->buffer + len, s->buffer_index);
+ // FIXME fix the audio codec API, so we do not need the memcpy()
+ /*for(i=0; i<len; i++) {
+ av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]);
+ }*/
return len;
- }else
+ } else
return 0;
}
#define OFFSET(x) offsetof(Mp3AudioContext, x)
#define AE AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_ENCODING_PARAM
static const AVOption options[] = {
- { "reservoir", "Use bit reservoir.", OFFSET(reservoir), AV_OPT_TYPE_INT, { 1 }, 0, 1, AE },
+ { "reservoir", "Use bit reservoir.", OFFSET(reservoir), AV_OPT_TYPE_INT, { 1 }, 0, 1, AE },
{ NULL },
};
};
AVCodec ff_libmp3lame_encoder = {
- .name = "libmp3lame",
- .type = AVMEDIA_TYPE_AUDIO,
- .id = CODEC_ID_MP3,
- .priv_data_size = sizeof(Mp3AudioContext),
- .init = MP3lame_encode_init,
- .encode = MP3lame_encode_frame,
- .close = MP3lame_encode_close,
- .capabilities= CODEC_CAP_DELAY,
- .sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,
+ .name = "libmp3lame",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .id = CODEC_ID_MP3,
+ .priv_data_size = sizeof(Mp3AudioContext),
+ .init = MP3lame_encode_init,
+ .encode = MP3lame_encode_frame,
+ .close = MP3lame_encode_close,
+ .capabilities = CODEC_CAP_DELAY,
+ .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
#if 2147483647 == INT_MAX
AV_SAMPLE_FMT_S32,
#endif
- AV_SAMPLE_FMT_NONE},
- .supported_samplerates= sSampleRates,
- .long_name= NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
- .priv_class = &libmp3lame_class,
+ AV_SAMPLE_FMT_NONE },
+ .supported_samplerates = sSampleRates,
+ .long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
+ .priv_class = &libmp3lame_class,
};
projects / ffmpeg / blobdiff