提交 d7bcc4ad 编写于 作者: J Justin Ruggles

AC-3 decoder, soc revision 58, Aug 19 14:20:45 2006 UTC by cloud9

Further optimizations.

Originally committed as revision 9666 to svn://svn.ffmpeg.org/ffmpeg/trunk
上级 1d0a6f52
......@@ -1579,35 +1579,21 @@ static void do_imdct_256(AC3DecodeContext *ctx, int chindex)
ff_imdct_calc(&ctx->imdct_256, ctx->tmp_output + 256, x2, ctx->tmp_imdct);
ptr = ctx->output[chindex];
ctx->dsp.vector_fmul_add_add(ptr, ctx->tmp_output, window, ctx->delay[chindex], 0, BLOCK_SIZE, 1);
ctx->dsp.vector_fmul_add_add(ptr, ctx->tmp_output, window, ctx->delay[chindex], 384, BLOCK_SIZE, 1);
ptr = ctx->delay[chindex];
ctx->dsp.vector_fmul_reverse(ptr, ctx->tmp_output + 256, window, BLOCK_SIZE);
/*for (k = 0; k < N / 2; k++) {
ctx->output[chindex][k] = ctx->tmp_output[k] * window[k] + ctx->delay[chindex][k];
//dump_floats("samples", 10, ctx->output[chindex], 256);
ctx->delay[chindex][k] = ctx->tmp_output[N / 2 + k] * window[255 - k];
}*/
}
static void do_imdct_512(AC3DecodeContext *ctx, int chindex)
{
//int k;
float *ptr;
ff_imdct_calc(&ctx->imdct_512, ctx->tmp_output,
ctx->transform_coeffs[chindex], ctx->tmp_imdct);
//ff_imdct_calc_ac3_512(&ctx->imdct_512, ctx->tmp_output, ctx->transform_coeffs[chindex],
// ctx->tmp_imdct, window);
ptr = ctx->output[chindex];
ctx->dsp.vector_fmul_add_add(ptr, ctx->tmp_output, window, ctx->delay[chindex], 0, BLOCK_SIZE, 1);
ctx->dsp.vector_fmul_add_add(ptr, ctx->tmp_output, window, ctx->delay[chindex], 384, BLOCK_SIZE, 1);
ptr = ctx->delay[chindex];
ctx->dsp.vector_fmul_reverse(ptr, ctx->tmp_output + 256, window, BLOCK_SIZE);
/*for (k = 0; k < N / 2; k++) {
ctx->output[chindex][k] = ctx->tmp_output[k] * window[k] + ctx->delay[chindex][k];
//dump_floats("samples", 10, ctx->output[chindex], 256);
ctx->delay[chindex][k] = ctx->tmp_output[N / 2 + k] * window[255 - k];
} */
}
static inline void do_imdct(AC3DecodeContext *ctx)
......@@ -1899,16 +1885,16 @@ static int ac3_parse_audio_block(AC3DecodeContext * ctx)
if (ctx->rematflg)
do_rematrixing(ctx);
do_downmix(ctx);
do_imdct(ctx);
/*for(i = 0; i < nfchans; i++)
dump_floats("channel output", 10, ctx->output[i + 1], BLOCK_SIZE);*/
do_downmix(ctx);
return 0;
}
static inline int16_t convert(float f)
/*static inline int16_t convert(float f)
{
if (f >= 1.0)
return 32767;
......@@ -1916,6 +1902,16 @@ static inline int16_t convert(float f)
return -32768;
else
return (lrintf(f * 32767.0));
}*/
static inline int16_t convert(int32_t i)
{
if (i > 0x43c07fff)
return 32767;
else if (i <= 0x43bf8000)
return -32768;
else
return (i - 0x43c00000);
}
static int frame_count = 0;
......@@ -1925,9 +1921,13 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
AC3DecodeContext *ctx = (AC3DecodeContext *)avctx->priv_data;
int frame_start;
int16_t *out_samples = (int16_t *)data;
int i, j, k, value;
int i, j, k, start;
int32_t *int_ptr[6];
av_log(NULL, AV_LOG_INFO, "decoding frame %d buf_size = %d\n", frame_count++, buf_size);
for (i = 0; i < 6; i++)
int_ptr[i] = (int32_t *)(&ctx->output[i]);
//av_log(NULL, AV_LOG_INFO, "decoding frame %d buf_size = %d\n", frame_count++, buf_size);
//Synchronize the frame.
frame_start = ac3_synchronize(buf, buf_size);
......@@ -1978,7 +1978,7 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
avctx->channels = ctx->nfchans + ctx->lfeon;
}
av_log(avctx, AV_LOG_INFO, "channels = %d \t bit rate = %d \t sampling rate = %d \n", avctx->channels, avctx->bit_rate * 1000, avctx->sample_rate);
//av_log(avctx, AV_LOG_INFO, "channels = %d \t bit rate = %d \t sampling rate = %d \n", avctx->channels, avctx->bit_rate * 1000, avctx->sample_rate);
//Parse the Audio Blocks.
for (i = 0; i < AUDIO_BLOCKS; i++) {
......@@ -1987,13 +1987,10 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
*data_size = 0;
return ctx->frame_size;
}
for (k = 0; k < BLOCK_SIZE; k++) {
j = (ctx->blkoutput & AC3_OUTPUT_LFEON) ? 0 : 1;
for (; j <= avctx->channels; j++) {
value = convert(ctx->output[j][k]);
*(out_samples++) = value;
}
}
start = (ctx->blkoutput & AC3_OUTPUT_LFEON) ? 0 : 1;
for (k = 0; k < BLOCK_SIZE; k++)
for (j = start; j <= avctx->channels; j++)
*(out_samples++) = convert(int_ptr[j][k]);
}
*data_size = AUDIO_BLOCKS * BLOCK_SIZE * avctx->channels * sizeof (int16_t);
return ctx->frame_size;
......
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