int abr;
float *samples_flt[2];
AudioFrameQueue afq;
- AVFloatDSPContext fdsp;
+ AVFloatDSPContext *fdsp;
} LAMEContext;
av_freep(&s->samples_flt[0]);
av_freep(&s->samples_flt[1]);
av_freep(&s->buffer);
+ av_freep(&s->fdsp);
ff_af_queue_close(&s->afq);
lame_set_out_samplerate(s->gfp, avctx->sample_rate);
/* algorithmic quality */
- if (avctx->compression_level == FF_COMPRESSION_DEFAULT)
- lame_set_quality(s->gfp, 5);
- else
+ if (avctx->compression_level != FF_COMPRESSION_DEFAULT)
lame_set_quality(s->gfp, avctx->compression_level);
/* rate control */
}
/* get encoder delay */
- avctx->delay = lame_get_encoder_delay(s->gfp) + 528 + 1;
+ avctx->initial_padding = lame_get_encoder_delay(s->gfp) + 528 + 1;
ff_af_queue_init(avctx, &s->afq);
avctx->frame_size = lame_get_framesize(s->gfp);
if (ret < 0)
goto error;
- avpriv_float_dsp_init(&s->fdsp, avctx->flags & CODEC_FLAG_BITEXACT);
+ s->fdsp = avpriv_float_dsp_alloc(avctx->flags & CODEC_FLAG_BITEXACT);
+ if (!s->fdsp) {
+ ret = AVERROR(ENOMEM);
+ goto error;
+ }
+
return 0;
error:
return AVERROR(EINVAL);
}
for (ch = 0; ch < avctx->channels; ch++) {
- s->fdsp.vector_fmul_scalar(s->samples_flt[ch],
+ s->fdsp->vector_fmul_scalar(s->samples_flt[ch],
(const float *)frame->data[ch],
32768.0f,
FFALIGN(frame->nb_samples, 8));