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ffmpeg.wasm-core
Commit c4db3446 authored by Aneesh Dogra's avatar Aneesh Dogra Committed by Diego Biurrun
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libmp3lame: K&R formatting cosmetics 

Signed-off-by: 's avatarDiego Biurrun <diego@biurrun.de>
parent 8ca903eb
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Showing with 84 additions and 89 deletions
libavcodec/libmp3lame.c
View file @ c4db3446
...@@ -31,7 +31,7 @@ ...@@ -31,7 +31,7 @@
#include "mpegaudio.h" #include "mpegaudio.h"
#include <lame/lame.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 { typedef struct Mp3AudioContext {
AVClass *class; AVClass *class;
lame_global_flags *gfp; lame_global_flags *gfp;
...@@ -55,17 +55,17 @@ static av_cold int MP3lame_encode_init(AVCodecContext *avctx) ...@@ -55,17 +55,17 @@ static av_cold int MP3lame_encode_init(AVCodecContext *avctx)
lame_set_in_samplerate(s->gfp, avctx->sample_rate); lame_set_in_samplerate(s->gfp, avctx->sample_rate);
lame_set_out_samplerate(s->gfp, avctx->sample_rate); lame_set_out_samplerate(s->gfp, avctx->sample_rate);
lame_set_num_channels(s->gfp, avctx->channels); 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); lame_set_quality(s->gfp, 5);
} else { } else {
lame_set_quality(s->gfp, avctx->compression_level); lame_set_quality(s->gfp, avctx->compression_level);
} }
lame_set_mode(s->gfp, s->stereo ? JOINT_STEREO : MONO); lame_set_mode(s->gfp, s->stereo ? JOINT_STEREO : MONO);
lame_set_brate(s->gfp, avctx->bit_rate/1000); lame_set_brate(s->gfp, avctx->bit_rate / 1000);
if(avctx->flags & CODEC_FLAG_QSCALE) { if (avctx->flags & CODEC_FLAG_QSCALE) {
lame_set_brate(s->gfp, 0); lame_set_brate(s->gfp, 0);
lame_set_VBR(s->gfp, vbr_default); 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); lame_set_bWriteVbrTag(s->gfp,0);
#if FF_API_LAME_GLOBAL_OPTS #if FF_API_LAME_GLOBAL_OPTS
...@@ -76,9 +76,8 @@ static av_cold int MP3lame_encode_init(AVCodecContext *avctx) ...@@ -76,9 +76,8 @@ static av_cold int MP3lame_encode_init(AVCodecContext *avctx)
goto err_close; goto err_close;
avctx->frame_size = lame_get_framesize(s->gfp); avctx->frame_size = lame_get_framesize(s->gfp);
avctx->coded_frame = avcodec_alloc_frame();
avctx->coded_frame= avcodec_alloc_frame(); avctx->coded_frame->key_frame = 1;
avctx->coded_frame->key_frame= 1;
return 0; return 0;
...@@ -93,27 +92,24 @@ static const int sSampleRates[] = { ...@@ -93,27 +92,24 @@ static const int sSampleRates[] = {
}; };
static const int sBitRates[2][3][15] = { 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, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448 },
{ 0, 32, 40, 48, 56, 64, 80, 96,112,128,160,192,224,256,320} { 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, 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, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 }
}, },
}; };
static const int sSamplesPerFrame[2][3] = static const int sSamplesPerFrame[2][3] = {
{
{ 384, 1152, 1152 }, { 384, 1152, 1152 },
{ 384, 1152, 576 } { 384, 1152, 576 }
}; };
static const int sBitsPerSlot[3] = { static const int sBitsPerSlot[3] = { 32, 8, 8 };
32,
8,
8
};
static int mp3len(void *data, int *samplesPerFrame, int *sampleRate) static int mp3len(void *data, int *samplesPerFrame, int *sampleRate)
{ {
...@@ -123,30 +119,35 @@ static int mp3len(void *data, int *samplesPerFrame, int *sampleRate) ...@@ -123,30 +119,35 @@ static int mp3len(void *data, int *samplesPerFrame, int *sampleRate)
int sampleRateID = ((header >> 10) & 0x03); int sampleRateID = ((header >> 10) & 0x03);
int bitsPerSlot = sBitsPerSlot[layerID]; int bitsPerSlot = sBitsPerSlot[layerID];
int isPadded = ((header >> 9) & 0x01); int isPadded = ((header >> 9) & 0x01);
static int const mode_tab[4]= {2,3,1,0}; static int const mode_tab[4] = { 2, 3, 1, 0 };
int mode= mode_tab[(header >> 19) & 0x03]; int mode = mode_tab[(header >> 19) & 0x03];
int mpeg_id= mode>0; int mpeg_id = mode > 0;
int temp0, temp1, bitRate; 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; return -1;
} }
if(!samplesPerFrame) samplesPerFrame= &temp0; if (!samplesPerFrame)
if(!sampleRate ) sampleRate = &temp1; samplesPerFrame = &temp0;
if (!sampleRate)
sampleRate = &temp1;
// *isMono = ((header >> 6) & 0x03) == 0x03; //*isMono = ((header >> 6) & 0x03) == 0x03;
*sampleRate = sSampleRates[sampleRateID]>>mode; *sampleRate = sSampleRates[sampleRateID] >> mode;
bitRate = sBitRates[mpeg_id][layerID][bitRateID] * 1000; bitRate = sBitRates[mpeg_id][layerID][bitRateID] * 1000;
*samplesPerFrame = sSamplesPerFrame[mpeg_id][layerID]; *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; return *samplesPerFrame * bitRate / (bitsPerSlot * *sampleRate) + isPadded;
} }
static int MP3lame_encode_frame(AVCodecContext *avctx, static int MP3lame_encode_frame(AVCodecContext *avctx, unsigned char *frame,
unsigned char *frame, int buf_size, void *data) int buf_size, void *data)
{ {
Mp3AudioContext *s = avctx->priv_data; Mp3AudioContext *s = avctx->priv_data;
int len; int len;
...@@ -154,59 +155,52 @@ static int MP3lame_encode_frame(AVCodecContext *avctx, ...@@ -154,59 +155,52 @@ static int MP3lame_encode_frame(AVCodecContext *avctx,
/* lame 3.91 dies on '1-channel interleaved' data */ /* lame 3.91 dies on '1-channel interleaved' data */
if(data){ if (data) {
if (s->stereo) { if (s->stereo) {
lame_result = lame_encode_buffer_interleaved( lame_result = lame_encode_buffer_interleaved(s->gfp, data,
s->gfp,
data,
avctx->frame_size, avctx->frame_size,
s->buffer + s->buffer_index, s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index BUFFER_SIZE - s->buffer_index);
);
} else { } else {
lame_result = lame_encode_buffer( lame_result = lame_encode_buffer(s->gfp, data, data,
s->gfp, avctx->frame_size, s->buffer +
data, s->buffer_index, BUFFER_SIZE -
data, s->buffer_index);
avctx->frame_size,
s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index
);
} }
}else{ } else {
lame_result= lame_encode_flush( lame_result = lame_encode_flush(s->gfp, s->buffer + s->buffer_index,
s->gfp, BUFFER_SIZE - s->buffer_index);
s->buffer + s->buffer_index,
BUFFER_SIZE - s->buffer_index
);
} }
if(lame_result < 0){ if (lame_result < 0) {
if(lame_result==-1) { if (lame_result == -1) {
/* output buffer too small */ /* 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; return -1;
} }
s->buffer_index += lame_result; s->buffer_index += lame_result;
if(s->buffer_index<4) if (s->buffer_index < 4)
return 0; return 0;
len= mp3len(s->buffer, NULL, NULL); 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); //av_log(avctx, AV_LOG_DEBUG, "in:%d packet-len:%d index:%d\n",
if(len <= s->buffer_index){ // avctx->frame_size, len, s->buffer_index);
if (len <= s->buffer_index) {
memcpy(frame, s->buffer, len); memcpy(frame, s->buffer, len);
s->buffer_index -= len; s->buffer_index -= len;
memmove(s->buffer, s->buffer+len, s->buffer_index); memmove(s->buffer, s->buffer + len, s->buffer_index);
//FIXME fix the audio codec API, so we do not need the memcpy() // FIXME fix the audio codec API, so we do not need the memcpy()
/*for(i=0; i<len; i++){ /*for(i=0; i<len; i++) {
av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]); av_log(avctx, AV_LOG_DEBUG, "%2X ", frame[i]);
}*/ }*/
return len; return len;
}else } else
return 0; return 0;
} }
...@@ -242,9 +236,10 @@ AVCodec ff_libmp3lame_encoder = { ...@@ -242,9 +236,10 @@ AVCodec ff_libmp3lame_encoder = {
.init = MP3lame_encode_init, .init = MP3lame_encode_init,
.encode = MP3lame_encode_frame, .encode = MP3lame_encode_frame,
.close = MP3lame_encode_close, .close = MP3lame_encode_close,
.capabilities= CODEC_CAP_DELAY, .capabilities = CODEC_CAP_DELAY,
.sample_fmts = (const enum AVSampleFormat[]){AV_SAMPLE_FMT_S16,AV_SAMPLE_FMT_NONE}, .sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16,
.supported_samplerates= sSampleRates, AV_SAMPLE_FMT_NONE },
.long_name= NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"), .supported_samplerates = sSampleRates,
.long_name = NULL_IF_CONFIG_SMALL("libmp3lame MP3 (MPEG audio layer 3)"),
.priv_class = &libmp3lame_class, .priv_class = &libmp3lame_class,
}; };
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