mirror of https://git.ffmpeg.org/ffmpeg.git
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>
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f57f665183
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752dd1952a
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@ -997,7 +997,7 @@ static int residue_encode(vorbis_enc_context *venc, vorbis_enc_residue *rc,
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return 0;
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return 0;
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}
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}
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static int apply_window_and_mdct(vorbis_enc_context *venc, int samples)
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static int apply_window_and_mdct(vorbis_enc_context *venc)
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{
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{
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int channel;
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int channel;
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const float * win = venc->win[1];
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const float * win = venc->win[1];
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@ -1008,13 +1008,13 @@ static int apply_window_and_mdct(vorbis_enc_context *venc, int samples)
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for (channel = 0; channel < venc->channels; channel++) {
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for (channel = 0; channel < venc->channels; channel++) {
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float *offset = venc->samples + channel * window_len * 2;
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float *offset = venc->samples + channel * window_len * 2;
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fdsp->vector_fmul(offset, offset, win, samples);
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fdsp->vector_fmul(offset, offset, win, window_len);
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fdsp->vector_fmul_scalar(offset, offset, 1/n, samples);
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fdsp->vector_fmul_scalar(offset, offset, 1/n, window_len);
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offset += window_len;
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offset += window_len;
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fdsp->vector_fmul_reverse(offset, offset, win, samples);
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fdsp->vector_fmul_reverse(offset, offset, win, window_len);
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fdsp->vector_fmul_scalar(offset, offset, 1/n, samples);
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fdsp->vector_fmul_scalar(offset, offset, 1/n, window_len);
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venc->mdct[1].mdct_calc(&venc->mdct[1], venc->coeffs + channel * window_len,
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venc->mdct[1].mdct_calc(&venc->mdct[1], venc->coeffs + channel * window_len,
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venc->samples + channel * window_len * 2);
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venc->samples + channel * window_len * 2);
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@ -1047,7 +1047,7 @@ static AVFrame *spawn_empty_frame(AVCodecContext *avctx, int channels)
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}
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}
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/* Set up audio samples for psy analysis and window/mdct */
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/* Set up audio samples for psy analysis and window/mdct */
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static void move_audio(vorbis_enc_context *venc, int *samples, int sf_size)
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static void move_audio(vorbis_enc_context *venc, int sf_size)
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{
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{
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AVFrame *cur = NULL;
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AVFrame *cur = NULL;
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int frame_size = 1 << (venc->log2_blocksize[1] - 1);
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int frame_size = 1 << (venc->log2_blocksize[1] - 1);
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@ -1065,7 +1065,6 @@ static void move_audio(vorbis_enc_context *venc, int *samples, int sf_size)
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for (sf = 0; sf < subframes; sf++) {
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for (sf = 0; sf < subframes; sf++) {
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cur = ff_bufqueue_get(&venc->bufqueue);
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cur = ff_bufqueue_get(&venc->bufqueue);
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*samples += cur->nb_samples;
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for (ch = 0; ch < venc->channels; ch++) {
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for (ch = 0; ch < venc->channels; ch++) {
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float *offset = venc->samples + 2 * ch * frame_size + frame_size;
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float *offset = venc->samples + 2 * ch * frame_size + frame_size;
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@ -1087,7 +1086,7 @@ static int vorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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{
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{
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vorbis_enc_context *venc = avctx->priv_data;
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vorbis_enc_context *venc = avctx->priv_data;
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int i, ret, need_more;
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int i, ret, need_more;
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int samples = 0, frame_size = 1 << (venc->log2_blocksize[1] - 1);
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int frame_size = 1 << (venc->log2_blocksize[1] - 1);
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vorbis_enc_mode *mode;
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vorbis_enc_mode *mode;
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vorbis_enc_mapping *mapping;
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vorbis_enc_mapping *mapping;
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PutBitContext pb;
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PutBitContext pb;
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@ -1120,9 +1119,9 @@ static int vorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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}
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}
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}
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}
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move_audio(venc, &samples, avctx->frame_size);
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move_audio(venc, avctx->frame_size);
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if (!apply_window_and_mdct(venc, samples))
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if (!apply_window_and_mdct(venc))
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return 0;
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return 0;
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if ((ret = ff_alloc_packet2(avctx, avpkt, 8192, 0)) < 0)
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if ((ret = ff_alloc_packet2(avctx, avpkt, 8192, 0)) < 0)
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@ -1149,21 +1148,21 @@ static int vorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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for (i = 0; i < venc->channels; i++) {
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for (i = 0; i < venc->channels; i++) {
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vorbis_enc_floor *fc = &venc->floors[mapping->floor[mapping->mux[i]]];
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vorbis_enc_floor *fc = &venc->floors[mapping->floor[mapping->mux[i]]];
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uint16_t posts[MAX_FLOOR_VALUES];
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uint16_t posts[MAX_FLOOR_VALUES];
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floor_fit(venc, fc, &venc->coeffs[i * samples], posts, samples);
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floor_fit(venc, fc, &venc->coeffs[i * frame_size], posts, frame_size);
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if (floor_encode(venc, fc, &pb, posts, &venc->floor[i * samples], samples)) {
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if (floor_encode(venc, fc, &pb, posts, &venc->floor[i * frame_size], frame_size)) {
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av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
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av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
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return AVERROR(EINVAL);
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return AVERROR(EINVAL);
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}
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}
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}
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}
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for (i = 0; i < venc->channels * samples; i++)
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for (i = 0; i < venc->channels * frame_size; i++)
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venc->coeffs[i] /= venc->floor[i];
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venc->coeffs[i] /= venc->floor[i];
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for (i = 0; i < mapping->coupling_steps; i++) {
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for (i = 0; i < mapping->coupling_steps; i++) {
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float *mag = venc->coeffs + mapping->magnitude[i] * samples;
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float *mag = venc->coeffs + mapping->magnitude[i] * frame_size;
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float *ang = venc->coeffs + mapping->angle[i] * samples;
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float *ang = venc->coeffs + mapping->angle[i] * frame_size;
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int j;
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int j;
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for (j = 0; j < samples; j++) {
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for (j = 0; j < frame_size; j++) {
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float a = ang[j];
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float a = ang[j];
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ang[j] -= mag[j];
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ang[j] -= mag[j];
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if (mag[j] > 0)
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if (mag[j] > 0)
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@ -1174,7 +1173,7 @@ static int vorbis_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
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}
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}
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if (residue_encode(venc, &venc->residues[mapping->residue[mapping->mux[0]]],
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if (residue_encode(venc, &venc->residues[mapping->residue[mapping->mux[0]]],
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&pb, venc->coeffs, samples, venc->channels)) {
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&pb, venc->coeffs, frame_size, venc->channels)) {
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av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
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av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
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return AVERROR(EINVAL);
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return AVERROR(EINVAL);
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}
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}
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