Traneptora/FFmpeg
1
Fork 0
forked from FFmpeg/FFmpeg
Code Pull requests Activity

use float_to_int16_interleave in ac3

Browse source 
Originally committed as revision 14706 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Loren Merritt 2008-08-12 03:01:17 +00:00
parent 916d5d6c32
commit 72745cff20
2 changed files with 9 additions and 18 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

24
libavcodec/ac3dec.c
View file

@ -201,7 +201,7 @@ static av_cold int ac3_decode_init(AVCodecContext *avctx)
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 @@ static inline void do_imdct(AC3DecodeContext *s, int channels)
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 @@ static int decode_audio_block(AC3DecodeContext *s, int blk)
do_imdct(s, s->out_channels);
}
/* convert float to 16-bit integer */
for(ch=0; ch<s->out_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 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
{
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 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
/* 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];
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;

1
libavcodec/ac3dec.h
View file

@ -167,7 +167,6 @@ typedef struct {
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;