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vorbisenc: Stop tracking number of samples per frame 

Each frame is now padded with 0 values if not enough samples are
present, and all frames are guaranteed to have exactly
1 << (venc->log2_blocksize[1] - 1) samples.

Signed-off-by: Tyler Jones <tdjones879@gmail.com>
Reviewed-by: Rostislav Pehlivanov <atomnuker@gmail.com>
This commit is contained in:
Tyler Jones 2017-06-14 14:59:07 -06:00 committed by Rostislav Pehlivanov
parent f57f665183
commit 752dd1952a
1 changed files with 16 additions and 17 deletions
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33
libavcodec/vorbisenc.c
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@ -997,7 +997,7 @@ static int residue_encode(vorbis_enc_context *venc, vorbis_enc_residue *rc,
return 0;
}
static int apply_window_and_mdct(vorbis_enc_context *venc, int samples)
static int apply_window_and_mdct(vorbis_enc_context *venc)
{
int channel;
const float * win = venc->win[1];
@ -1008,13 +1008,13 @@ static int apply_window_and_mdct(vorbis_enc_context *venc, int samples)
for (channel = 0; channel < venc->channels; channel++) {
float *offset = venc->samples + channel * window_len * 2;
fdsp->vector_fmul(offset, offset, win, samples);
fdsp->vector_fmul_scalar(offset, offset, 1/n, samples);
fdsp->vector_fmul(offset, offset, win, window_len);
fdsp->vector_fmul_scalar(offset, offset, 1/n, window_len);
offset += window_len;
fdsp->vector_fmul_reverse(offset, offset, win, samples);
fdsp->vector_fmul_scalar(offset, offset, 1/n, samples);
fdsp->vector_fmul_reverse(offset, offset, win, window_len);
fdsp->vector_fmul_scalar(offset, offset, 1/n, window_len);
venc->mdct[1].mdct_calc(&venc->mdct[1], venc->coeffs + channel * window_len,
venc->samples + channel * window_len * 2);
@ -1047,7 +1047,7 @@ static AVFrame *spawn_empty_frame(AVCodecContext *avctx, int channels)
}
/* Set up audio samples for psy analysis and window/mdct */
static void move_audio(vorbis_enc_context *venc, int *samples, int sf_size)
static void move_audio(vorbis_enc_context *venc, int sf_size)
{
AVFrame *cur = NULL;
int frame_size = 1 << (venc->log2_blocksize[1] - 1);
@ -1065,7 +1065,6 @@ static void move_audio(vorbis_enc_context *venc, int *samples, int sf_size)
for (sf = 0; sf < subframes; sf++) {
cur = ff_bufqueue_get(&venc->bufqueue);
*samples += cur->nb_samples;
for (ch = 0; ch < venc->channels; ch++) {
float *offset = venc->samples + 2 * ch * frame_size + frame_size;
@ -1087,7 +1086,7 @@ static int vorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
{
vorbis_enc_context *venc = avctx->priv_data;
int i, ret, need_more;
int samples = 0, frame_size = 1 << (venc->log2_blocksize[1] - 1);
int frame_size = 1 << (venc->log2_blocksize[1] - 1);
vorbis_enc_mode *mode;
vorbis_enc_mapping *mapping;
PutBitContext pb;
@ -1120,9 +1119,9 @@ static int vorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
}
}
move_audio(venc, &samples, avctx->frame_size);
move_audio(venc, avctx->frame_size);
if (!apply_window_and_mdct(venc, samples))
if (!apply_window_and_mdct(venc))
return 0;
if ((ret = ff_alloc_packet2(avctx, avpkt, 8192, 0)) < 0)
@ -1149,21 +1148,21 @@ static int vorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
for (i = 0; i < venc->channels; i++) {
vorbis_enc_floor *fc = &venc->floors[mapping->floor[mapping->mux[i]]];
uint16_t posts[MAX_FLOOR_VALUES];
floor_fit(venc, fc, &venc->coeffs[i * samples], posts, samples);
if (floor_encode(venc, fc, &pb, posts, &venc->floor[i * samples], samples)) {
floor_fit(venc, fc, &venc->coeffs[i * frame_size], posts, frame_size);
if (floor_encode(venc, fc, &pb, posts, &venc->floor[i * frame_size], frame_size)) {
av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
return AVERROR(EINVAL);
}
}
for (i = 0; i < venc->channels * samples; i++)
for (i = 0; i < venc->channels * frame_size; i++)
venc->coeffs[i] /= venc->floor[i];
for (i = 0; i < mapping->coupling_steps; i++) {
float *mag = venc->coeffs + mapping->magnitude[i] * samples;
float *ang = venc->coeffs + mapping->angle[i] * samples;
float *mag = venc->coeffs + mapping->magnitude[i] * frame_size;
float *ang = venc->coeffs + mapping->angle[i] * frame_size;
int j;
for (j = 0; j < samples; j++) {
for (j = 0; j < frame_size; j++) {
float a = ang[j];
ang[j] -= mag[j];
if (mag[j] > 0)
@ -1174,7 +1173,7 @@ static int vorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
}
if (residue_encode(venc, &venc->residues[mapping->residue[mapping->mux[0]]],
&pb, venc->coeffs, samples, venc->channels)) {
&pb, venc->coeffs, frame_size, venc->channels)) {
av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
return AVERROR(EINVAL);
}