提交 1402ee72 编写于 作者: B Benjamin Larsson

Fix dca decoder with non simd float2int16 conversion

Originally committed as revision 15714 to svn://svn.ffmpeg.org/ffmpeg/trunk
上级 49c91c34
......@@ -67,7 +67,6 @@ enum DCAMode {
#define DCA_LFE 0x80
#define HEADER_SIZE 14
#define CONVERT_BIAS 384
#define DCA_MAX_FRAME_SIZE 16384
......@@ -159,7 +158,8 @@ typedef struct {
int hist_index[DCA_PRIM_CHANNELS_MAX];
int output; ///< type of output
int bias; ///< output bias
float add_bias; ///< output bias
float scale_bias; ///< output scale
DECLARE_ALIGNED_16(float, samples[1536]); /* 6 * 256 = 1536, might only need 5 */
const float *samples_chanptr[6];
......@@ -230,8 +230,6 @@ static int dca_parse_frame_header(DCAContext * s)
static const int bitlen[11] = { 0, 1, 2, 2, 2, 2, 3, 3, 3, 3, 3 };
static const int thr[11] = { 0, 1, 3, 3, 3, 3, 7, 7, 7, 7, 7 };
s->bias = CONVERT_BIAS;
init_get_bits(&s->gb, s->dca_buffer, s->dca_buffer_size * 8);
/* Sync code */
......@@ -748,7 +746,7 @@ static void lfe_interpolation_fir(int decimation_select,
//FIXME the coeffs are symetric, fix that
for (j = 0; j < 512 / decifactor; j++)
rTmp += samples_in[deciindex - j] * prCoeff[k + j * decifactor];
samples_out[interp_index++] = rTmp / scale + bias;
samples_out[interp_index++] = (rTmp * scale) + bias;
}
}
}
......@@ -984,8 +982,8 @@ static int dca_subsubframe(DCAContext * s)
/* static float pcm_to_double[8] =
{32768.0, 32768.0, 524288.0, 524288.0, 0, 8388608.0, 8388608.0};*/
qmf_32_subbands(s, k, subband_samples[k], &s->samples[256 * k],
M_SQRT1_2 /*pcm_to_double[s->source_pcm_res] */ ,
0 /*s->bias */ );
M_SQRT1_2*s->scale_bias /*pcm_to_double[s->source_pcm_res] */ ,
s->add_bias );
}
/* Down mixing */
......@@ -1003,7 +1001,7 @@ static int dca_subsubframe(DCAContext * s)
s->lfe_data + lfe_samples +
2 * s->lfe * subsubframe,
&s->samples[256 * i_channels],
256.0, 0 /* s->bias */);
(1.0/256.0)*s->scale_bias, s->add_bias);
/* Outputs 20bits pcm samples */
}
......@@ -1214,6 +1212,16 @@ static av_cold int dca_decode_init(AVCodecContext * avctx)
for(i = 0; i < 6; i++)
s->samples_chanptr[i] = s->samples + i * 256;
avctx->sample_fmt = SAMPLE_FMT_S16;
if(s->dsp.float_to_int16 == ff_float_to_int16_c) {
s->add_bias = 385.0f;
s->scale_bias = 1.0 / 32768.0;
} else {
s->add_bias = 0.0f;
s->scale_bias = 1.0;
}
return 0;
}
......
Markdown is supported
0% .
You are about to add 0 people to the discussion. Proceed with caution.
先完成此消息的编辑!
想要评论请 注册