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libmp3lame: K&R formatting cosmetics

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Signed-off-by: Diego Biurrun <diego@biurrun.de>
This commit is contained in:
Aneesh Dogra 2011-12-30 02:49:55 +05:30 committed by Diego Biurrun
parent 8ca903ebcb
commit c4db344664
1 changed files with 84 additions and 89 deletions
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173
libavcodec/libmp3lame.c
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@ -31,7 +31,7 @@
#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)
typedef struct Mp3AudioContext {
AVClass *class;
lame_global_flags *gfp;
@ -55,17 +55,17 @@ static av_cold int MP3lame_encode_init(AVCodecContext *avctx)
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
@ -75,10 +75,9 @@ static av_cold int MP3lame_encode_init(AVCodecContext *avctx)
if (lame_init_params(s->gfp) < 0)
goto err_close;
avctx->frame_size = lame_get_framesize(s->gfp);
avctx->coded_frame= avcodec_alloc_frame();
avctx->coded_frame->key_frame= 1;
avctx->frame_size = lame_get_framesize(s->gfp);
avctx->coded_frame = avcodec_alloc_frame();
avctx->coded_frame->key_frame = 1;
return 0;
@ -93,60 +92,62 @@ static const int sSampleRates[] = {
};
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;
@ -154,59 +155,52 @@ static int MP3lame_encode_frame(AVCodecContext *avctx,
/* lame 3.91 dies on '1-channel interleaved' data */
if(data){
if (data) {
if (s->stereo) {
lame_result = lame_encode_buffer_interleaved(
s->gfp,
data,
avctx->frame_size,
s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index
);
lame_result = lame_encode_buffer_interleaved(s->gfp, data,
avctx->frame_size,
s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index);
} else {
lame_result = lame_encode_buffer(
s->gfp,
data,
data,
avctx->frame_size,
s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index
);
lame_result = lame_encode_buffer(s->gfp, data, data,
avctx->frame_size, s->buffer +
s->buffer_index, BUFFER_SIZE -
s->buffer_index);
}
}else{
lame_result= lame_encode_flush(
s->gfp,
s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index
);
} else {
lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
BUFFER_SIZE - 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;
}
@ -223,7 +217,7 @@ static av_cold int MP3lame_encode_close(AVCodecContext *avctx)
#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 },
};
@ -235,16 +229,17 @@ static const AVClass libmp3lame_class = {
};
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,AV_SAMPLE_FMT_NONE},
.supported_samplerates= sSampleRates,
.long_name= NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
.priv_class = &libmp3lame_class,
.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,
AV_SAMPLE_FMT_NONE },
.supported_samplerates = sSampleRates,
.long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
.priv_class = &libmp3lame_class,
};