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Original Commit: r59 | ods15 | 2006-09-24 17:40:40 +0300 (Sun, 24 Sep 2006) | 2 lines

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cosmetic, reorganize

Originally committed as revision 6464 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Oded Shimon 2006-10-02 06:07:44 +00:00
parent 8a202cb3a7
commit 6f17618af9
1 changed files with 69 additions and 69 deletions
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138
libavcodec/vorbis_enc.c
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@ -598,75 +598,6 @@ static int put_main_header(venc_context_t * venc, uint8_t ** out) {
return p - *out; return p - *out;
} }
static int vorbis_encode_init(AVCodecContext * avccontext)
{
venc_context_t * venc = avccontext->priv_data;
create_vorbis_context(venc, avccontext);
//if (avccontext->flags & CODEC_FLAG_QSCALE) avccontext->global_quality / (float)FF_QP2LAMBDA); else avccontext->bit_rate;
//if(avccontext->cutoff > 0) cfreq = avccontext->cutoff / 1000.0;
avccontext->extradata_size = put_main_header(venc, (uint8_t**)&avccontext->extradata);
avccontext->frame_size = 1 << (venc->blocksize[0] - 1);
avccontext->coded_frame = avcodec_alloc_frame();
avccontext->coded_frame->key_frame = 1;
return 0;
}
static int window(venc_context_t * venc, signed short * audio, int samples) {
int i, j, channel;
const float * win = venc->win[0];
int window_len = 1 << (venc->blocksize[0] - 1);
float n = (float)(1 << venc->blocksize[0]) / 4.;
// FIXME use dsp
if (!venc->have_saved && !samples) return 0;
if (venc->have_saved) {
for (channel = 0; channel < venc->channels; channel++) {
memcpy(venc->samples + channel*window_len*2, venc->saved + channel*window_len, sizeof(float)*window_len);
}
} else {
for (channel = 0; channel < venc->channels; channel++) {
memset(venc->samples + channel*window_len*2, 0, sizeof(float)*window_len);
}
}
if (samples) {
for (channel = 0; channel < venc->channels; channel++) {
float * offset = venc->samples + channel*window_len*2 + window_len;
j = channel;
for (i = 0; i < samples; i++, j += venc->channels)
offset[i] = audio[j] / 32768. * win[window_len - i] / n;
}
} else {
for (channel = 0; channel < venc->channels; channel++) {
memset(venc->samples + channel*window_len*2 + window_len, 0, sizeof(float)*window_len);
}
}
for (channel = 0; channel < venc->channels; channel++) {
ff_mdct_calc(&venc->mdct[0], venc->coeffs + channel*window_len, venc->samples + channel*window_len*2, venc->floor/*tmp*/);
}
if (samples) {
for (channel = 0; channel < venc->channels; channel++) {
float * offset = venc->saved + channel*window_len;
j = channel;
for (i = 0; i < samples; i++, j += venc->channels)
offset[i] = audio[j] / 32768. * win[i] / n;
}
venc->have_saved = 1;
} else {
venc->have_saved = 0;
}
return 1;
}
static float * put_vector(codebook_t * book, PutBitContext * pb, float * num) { static float * put_vector(codebook_t * book, PutBitContext * pb, float * num) {
int i; int i;
int entry = -1; int entry = -1;
@ -728,6 +659,75 @@ static void residue_encode(venc_context_t * venc, residue_t * rc, PutBitContext
} }
} }
static int window(venc_context_t * venc, signed short * audio, int samples) {
int i, j, channel;
const float * win = venc->win[0];
int window_len = 1 << (venc->blocksize[0] - 1);
float n = (float)(1 << venc->blocksize[0]) / 4.;
// FIXME use dsp
if (!venc->have_saved && !samples) return 0;
if (venc->have_saved) {
for (channel = 0; channel < venc->channels; channel++) {
memcpy(venc->samples + channel*window_len*2, venc->saved + channel*window_len, sizeof(float)*window_len);
}
} else {
for (channel = 0; channel < venc->channels; channel++) {
memset(venc->samples + channel*window_len*2, 0, sizeof(float)*window_len);
}
}
if (samples) {
for (channel = 0; channel < venc->channels; channel++) {
float * offset = venc->samples + channel*window_len*2 + window_len;
j = channel;
for (i = 0; i < samples; i++, j += venc->channels)
offset[i] = audio[j] / 32768. * win[window_len - i] / n;
}
} else {
for (channel = 0; channel < venc->channels; channel++) {
memset(venc->samples + channel*window_len*2 + window_len, 0, sizeof(float)*window_len);
}
}
for (channel = 0; channel < venc->channels; channel++) {
ff_mdct_calc(&venc->mdct[0], venc->coeffs + channel*window_len, venc->samples + channel*window_len*2, venc->floor/*tmp*/);
}
if (samples) {
for (channel = 0; channel < venc->channels; channel++) {
float * offset = venc->saved + channel*window_len;
j = channel;
for (i = 0; i < samples; i++, j += venc->channels)
offset[i] = audio[j] / 32768. * win[i] / n;
}
venc->have_saved = 1;
} else {
venc->have_saved = 0;
}
return 1;
}
static int vorbis_encode_init(AVCodecContext * avccontext)
{
venc_context_t * venc = avccontext->priv_data;
create_vorbis_context(venc, avccontext);
//if (avccontext->flags & CODEC_FLAG_QSCALE) avccontext->global_quality / (float)FF_QP2LAMBDA); else avccontext->bit_rate;
//if(avccontext->cutoff > 0) cfreq = avccontext->cutoff / 1000.0;
avccontext->extradata_size = put_main_header(venc, (uint8_t**)&avccontext->extradata);
avccontext->frame_size = 1 << (venc->blocksize[0] - 1);
avccontext->coded_frame = avcodec_alloc_frame();
avccontext->coded_frame->key_frame = 1;
return 0;
}
static int vorbis_encode_frame(AVCodecContext * avccontext, unsigned char * packets, int buf_size, void *data) static int vorbis_encode_frame(AVCodecContext * avccontext, unsigned char * packets, int buf_size, void *data)
{ {
venc_context_t * venc = avccontext->priv_data; venc_context_t * venc = avccontext->priv_data;