# HG changeset patch # User lorenm # Date 1218510077 0 # Node ID f9e4afa46993c08517285376a33e0141bf84688e # Parent 60b5c1e5a7ee7dd14ee5de1e6d7515692316486d use float_to_int16_interleave in ac3 diff -r 60b5c1e5a7ee -r f9e4afa46993 ac3dec.c --- a/ac3dec.c Tue Aug 12 01:30:24 2008 +0000 +++ b/ac3dec.c Tue Aug 12 03:01:17 2008 +0000 @@ -201,7 +201,7 @@ av_init_random(0, &s->dith_state); /* set bias values for float to int16 conversion */ - if(s->dsp.float_to_int16 == ff_float_to_int16_c) { + if(s->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) { s->add_bias = 385.0f; s->mul_bias = 1.0f; } else { @@ -604,13 +604,13 @@ for(i=0; i<128; i++) x[i] = s->transform_coeffs[ch][2*i]; ff_imdct_half(&s->imdct_256, s->tmp_output, x); - s->dsp.vector_fmul_window(s->output[ch-1], s->delay[ch-1], s->tmp_output, s->window, 0, 128); + s->dsp.vector_fmul_window(s->output[ch-1], s->delay[ch-1], s->tmp_output, s->window, s->add_bias, 128); for(i=0; i<128; i++) x[i] = s->transform_coeffs[ch][2*i+1]; ff_imdct_half(&s->imdct_256, s->delay[ch-1], x); } else { ff_imdct_half(&s->imdct_512, s->tmp_output, s->transform_coeffs[ch]); - s->dsp.vector_fmul_window(s->output[ch-1], s->delay[ch-1], s->tmp_output, s->window, 0, 128); + s->dsp.vector_fmul_window(s->output[ch-1], s->delay[ch-1], s->tmp_output, s->window, s->add_bias, 128); memcpy(s->delay[ch-1], s->tmp_output+128, 128*sizeof(float)); } } @@ -1018,14 +1018,6 @@ do_imdct(s, s->out_channels); } - /* convert float to 16-bit integer */ - for(ch=0; chout_channels; ch++) { - for(i=0; i<256; i++) { - s->output[ch][i] += s->add_bias; - } - s->dsp.float_to_int16(s->int_output[ch], s->output[ch], 256); - } - return 0; } @@ -1037,7 +1029,7 @@ { AC3DecodeContext *s = avctx->priv_data; int16_t *out_samples = (int16_t *)data; - int i, blk, ch, err; + int blk, ch, err; /* initialize the GetBitContext with the start of valid AC-3 Frame */ if (s->input_buffer) { @@ -1127,14 +1119,14 @@ /* decode the audio blocks */ for (blk = 0; blk < s->num_blocks; blk++) { + const float *output[s->out_channels]; if (!err && decode_audio_block(s, blk)) { av_log(avctx, AV_LOG_ERROR, "error decoding the audio block\n"); } - - /* interleave output samples */ - for (i = 0; i < 256; i++) - for (ch = 0; ch < s->out_channels; ch++) - *(out_samples++) = s->int_output[ch][i]; + for (ch = 0; ch < s->out_channels; ch++) + output[ch] = s->output[ch]; + s->dsp.float_to_int16_interleave(out_samples, output, 256, s->out_channels); + out_samples += 256 * s->out_channels; } *data_size = s->num_blocks * 256 * avctx->channels * sizeof (int16_t); return s->frame_size; diff -r 60b5c1e5a7ee -r f9e4afa46993 ac3dec.h --- a/ac3dec.h Tue Aug 12 01:30:24 2008 +0000 +++ b/ac3dec.h Tue Aug 12 03:01:17 2008 +0000 @@ -167,7 +167,6 @@ DECLARE_ALIGNED_16(float, window[AC3_BLOCK_SIZE]); ///< window coefficients DECLARE_ALIGNED_16(float, tmp_output[AC3_BLOCK_SIZE]); ///< temporary storage for output before windowing DECLARE_ALIGNED_16(float, output[AC3_MAX_CHANNELS][AC3_BLOCK_SIZE]); ///< output after imdct transform and windowing - DECLARE_ALIGNED_16(short, int_output[AC3_MAX_CHANNELS-1][AC3_BLOCK_SIZE]); ///< final 16-bit integer output ///@} } AC3DecodeContext;