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author | Aneesh Dogra <lionaneesh@gmail.com> | 2011-12-30 02:49:55 +0530 |
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committer | Diego Biurrun <diego@biurrun.de> | 2011-12-29 23:30:20 +0100 |
commit | c4db34466424f9853c3e6b3bbe8a430da14da7e7 (patch) | |
tree | e07a1bdd7c7d445a6b9d53290e8524c50ceffc19 /libavcodec | |
parent | 8ca903ebcbe437f0448e675aa0288ca35ee1d450 (diff) | |
download | ffmpeg-streaming-c4db34466424f9853c3e6b3bbe8a430da14da7e7.zip ffmpeg-streaming-c4db34466424f9853c3e6b3bbe8a430da14da7e7.tar.gz |
libmp3lame: K&R formatting cosmetics
Signed-off-by: Diego Biurrun <diego@biurrun.de>
Diffstat (limited to 'libavcodec')
-rw-r--r-- | libavcodec/libmp3lame.c | 173 |
1 files changed, 84 insertions, 89 deletions
diff --git a/libavcodec/libmp3lame.c b/libavcodec/libmp3lame.c index 6da6d71..f3c4528 100644 --- a/libavcodec/libmp3lame.c +++ b/libavcodec/libmp3lame.c @@ -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, }; 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