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vorbis: reduce memory by reusing buffers 

Originally committed as revision 14251 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Loren Merritt 2008-07-15 23:04:28 +00:00
parent 1a32536737
commit 633d9def9d
1 changed files with 9 additions and 15 deletions
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24
libavcodec/vorbis_dec.c
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@ -153,9 +153,6 @@ typedef struct vorbis_context_s {
float *channel_residues;
float *channel_floors;
float *saved;
float *ret;
float *buf;
float *buf_tmp;
uint_fast32_t add_bias; // for float->int conversion
uint_fast32_t exp_bias;
} vorbis_context;
@ -181,9 +178,6 @@ static void vorbis_free(vorbis_context *vc) {
av_freep(&vc->channel_residues);
av_freep(&vc->channel_floors);
av_freep(&vc->saved);
av_freep(&vc->ret);
av_freep(&vc->buf);
av_freep(&vc->buf_tmp);
av_freep(&vc->residues);
av_freep(&vc->modes);
@ -900,9 +894,6 @@ static int vorbis_parse_id_hdr(vorbis_context *vc){
vc->channel_residues= av_malloc((vc->blocksize[1]/2)*vc->audio_channels * sizeof(float));
vc->channel_floors = av_malloc((vc->blocksize[1]/2)*vc->audio_channels * sizeof(float));
vc->saved = av_mallocz((vc->blocksize[1]/4)*vc->audio_channels * sizeof(float));
vc->ret = av_malloc((vc->blocksize[1]/2)*vc->audio_channels * sizeof(float));
vc->buf = av_malloc( vc->blocksize[1]/2 * sizeof(float));
vc->buf_tmp = av_malloc( vc->blocksize[1]/2 * sizeof(float));
vc->previous_window=0;
ff_mdct_init(&vc->mdct[0], bl0, 1);
@ -1423,7 +1414,7 @@ static int vorbis_parse_audio_packet(vorbis_context *vc) {
uint_fast8_t mode_number;
uint_fast8_t blockflag;
uint_fast16_t blocksize;
int_fast32_t i,j;
int_fast32_t i,j,dir;
uint_fast8_t no_residue[vc->audio_channels];
uint_fast8_t do_not_decode[vc->audio_channels];
vorbis_mapping *mapping;
@ -1525,15 +1516,18 @@ static int vorbis_parse_audio_packet(vorbis_context *vc) {
// MDCT, overlap/add, save data for next overlapping FPMATH
retlen = (blocksize + vc->blocksize[previous_window])/4;
for(j=0;j<vc->audio_channels;++j) {
dir = retlen <= blocksize/2; // pick an order so that ret[] can reuse residues[] without stepping on any data we need
for(j=dir?0:vc->audio_channels-1; (unsigned)j<vc->audio_channels; j+=dir*2-1) {
uint_fast16_t bs0=vc->blocksize[0];
uint_fast16_t bs1=vc->blocksize[1];
float *residue=vc->channel_residues+res_chan[j]*blocksize/2;
float *floor=vc->channel_floors+j*blocksize/2;
float *saved=vc->saved+j*bs1/4;
float *ret=vc->ret+j*retlen;
float *buf=vc->buf;
float *ret=vc->channel_residues+j*retlen;
float *buf=floor;
const float *win=vc->win[blockflag&previous_window];
vc->mdct[0].fft.imdct_half(&vc->mdct[blockflag], buf, vc->channel_floors+j*blocksize/2, vc->buf_tmp);
vc->mdct[0].fft.imdct_half(&vc->mdct[blockflag], buf, floor, residue);
if(blockflag == previous_window) {
vc->dsp.vector_fmul_window(ret, saved, buf, win, fadd_bias, blocksize/4);
@ -1585,7 +1579,7 @@ static int vorbis_decode_frame(AVCodecContext *avccontext,
AV_DEBUG("parsed %d bytes %d bits, returned %d samples (*ch*bits) \n", get_bits_count(gb)/8, get_bits_count(gb)%8, len);
vc->dsp.float_to_int16_interleave(data, vc->ret, len, vc->audio_channels);
vc->dsp.float_to_int16_interleave(data, vc->channel_residues, len, vc->audio_channels);
*data_size=len*2*vc->audio_channels;
return buf_size ;