From c4db34466424f9853c3e6b3bbe8a430da14da7e7 Mon Sep 17 00:00:00 2001 From: Aneesh Dogra Date: Fri, 30 Dec 2011 02:49:55 +0530 Subject: libmp3lame: K&R formatting cosmetics Signed-off-by: Diego Biurrun --- libavcodec/libmp3lame.c | 173 +++++++++++++++++++++++------------------------- 1 file changed, 84 insertions(+), 89 deletions(-) (limited to 'libavcodec/libmp3lame.c') diff --git a/libavcodec/libmp3lame.c b/libavcodec/libmp3lame.c index 6da6d71134..f3c4528c07 100644 --- a/libavcodec/libmp3lame.c +++ b/libavcodec/libmp3lame.c @@ -31,7 +31,7 @@ #include "mpegaudio.h" #include -#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; ibuffer, s->buffer + len, s->buffer_index); + // FIXME fix the audio codec API, so we do not need the memcpy() + /*for(i=0; i