comparison atrac1.c @ 11652:8b6f3d3b55cb libavcodec

Move clipping of audio samples (for those codecs outputting float) from decoder to the audio conversion routines.

11651:926ad89ae57a

        if (ret < 0)
            return ret;
        at1_subband_synthesis(q, su, q->out_samples[ch]);
    }

    /* round, convert to 16bit and interleave */
    if (q->channels == 1) {
        /* mono */
        q->dsp.vector_clipf(samples, q->out_samples[0], -32700.0 / (1 << 15),
                            32700.0 / (1 << 15), AT1_SU_SAMPLES);
    } else {
        /* stereo */
        for (i = 0; i < AT1_SU_SAMPLES; i++) {
            samples[i * 2]     = av_clipf(q->out_samples[0][i],
                                          -32700.0 / (1 << 15),
                                           32700.0 / (1 << 15));
            samples[i * 2 + 1] = av_clipf(q->out_samples[1][i],
                                          -32700.0 / (1 << 15),
                                           32700.0 / (1 << 15));
        }
    }

    *data_size = q->channels * AT1_SU_SAMPLES * sizeof(*samples);
    return avctx->block_align;

11652:8b6f3d3b55cb

        if (ret < 0)
            return ret;
        at1_subband_synthesis(q, su, q->out_samples[ch]);
    }

    /* interleave; FIXME, should create/use a DSP function */
    if (q->channels == 1) {
        /* mono */
        memcpy(samples, q->out_samples[0], AT1_SU_SAMPLES * 4);
    } else {
        /* stereo */
        for (i = 0; i < AT1_SU_SAMPLES; i++) {
            samples[i * 2]     = q->out_samples[0][i];
            samples[i * 2 + 1] = q->out_samples[1][i];
        }
    }

    *data_size = q->channels * AT1_SU_SAMPLES * sizeof(*samples);
    return avctx->block_align;