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Libavcodec AC3/E-AC3/DTS decoders now output floating point data.

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git-svn-id: https://ffdshow-tryout.svn.sourceforge.net/svnroot/ffdshow-tryout@3769 3b938f2f-1a1a-0410-8054-a526ea5ff92c
This commit is contained in:
clsid2 2011-03-07 00:08:34 +00:00 committed by Michael Niedermayer
parent 361fa0ed40
commit 0e09997fa4
4 changed files with 75 additions and 3 deletions
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22
libavcodec/ac3dec.c
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@ -188,8 +188,13 @@ static av_cold int ac3_decode_init(AVCodecContext *avctx)
ff_fmt_convert_init(&s->fmt_conv, avctx);
av_lfg_init(&s->dith_state, 0);
/* ffdshow custom code */
#if CONFIG_AUDIO_FLOAT
s->mul_bias = 1.0f;
#else
/* set scale value for float to int16 conversion */
s->mul_bias = 32767.0f;
#endif
/* allow downmixing to stereo or mono */
if (avctx->channels > 0 && avctx->request_channels > 0 &&
@ -204,7 +209,12 @@ static av_cold int ac3_decode_init(AVCodecContext *avctx)
if (!s->input_buffer)
return AVERROR(ENOMEM);
/* ffdshow custom code */
#if CONFIG_AUDIO_FLOAT
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
#else
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
#endif
return 0;
}
@ -1299,7 +1309,12 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
const uint8_t *buf = avpkt->data;
int buf_size = avpkt->size;
AC3DecodeContext *s = avctx->priv_data;
/* ffdshow custom code */
#if CONFIG_AUDIO_FLOAT
float *out_samples = (float *)data;
#else
int16_t *out_samples = (int16_t *)data;
#endif
int blk, ch, err;
const uint8_t *channel_map;
const float *output[AC3_MAX_CHANNELS];
@ -1405,10 +1420,15 @@ static int ac3_decode_frame(AVCodecContext * avctx, void *data, int *data_size,
av_log(avctx, AV_LOG_ERROR, "error decoding the audio block\n");
err = 1;
}
/* ffdshow custom code */
#if CONFIG_AUDIO_FLOAT
float_interleave_noscale(out_samples, output, 256, s->out_channels);
#else
s->fmt_conv.float_to_int16_interleave(out_samples, output, 256, s->out_channels);
#endif
out_samples += 256 * s->out_channels;
}
*data_size = s->num_blocks * 256 * avctx->channels * sizeof (int16_t);
*data_size = s->num_blocks * 256 * avctx->channels * sizeof (out_samples[0]); /* ffdshow custom code */
return FFMIN(buf_size, s->frame_size);
}

21
libavcodec/dca.c
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@ -1626,7 +1626,12 @@ static int dca_decode_frame(AVCodecContext * avctx,
int lfe_samples;
int num_core_channels = 0;
int i;
/* ffdshow custom code */
#if CONFIG_AUDIO_FLOAT
float *samples = data;
#else
int16_t *samples = data;
#endif
DCAContext *s = avctx->priv_data;
int channels;
int core_ss_end;
@ -1812,9 +1817,10 @@ static int dca_decode_frame(AVCodecContext * avctx,
return -1;
}
if (*data_size < (s->sample_blocks / 8) * 256 * sizeof(int16_t) * channels)
/* ffdshow custom code */
if (*data_size < (s->sample_blocks / 8) * 256 * sizeof(samples[0]) * channels)
return -1;
*data_size = 256 / 8 * s->sample_blocks * sizeof(int16_t) * channels;
*data_size = 256 / 8 * s->sample_blocks * sizeof(samples[0]) * channels;
/* filter to get final output */
for (i = 0; i < (s->sample_blocks / 8); i++) {
@ -1833,7 +1839,13 @@ static int dca_decode_frame(AVCodecContext * avctx,
}
}
/* interleave samples */
#if CONFIG_AUDIO_FLOAT
/* ffdshow custom code */
float_interleave(samples, s->samples_chanptr, 256, channels);
#else
s->fmt_conv.float_to_int16_interleave(samples, s->samples_chanptr, 256, channels);
#endif
samples += 256 * channels;
}
@ -1870,7 +1882,12 @@ static av_cold int dca_decode_init(AVCodecContext * avctx)
for (i = 0; i < DCA_PRIM_CHANNELS_MAX+1; i++)
s->samples_chanptr[i] = s->samples + i * 256;
/* ffdshow custom code */
#if CONFIG_AUDIO_FLOAT
avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
#else
avctx->sample_fmt = AV_SAMPLE_FMT_S16;
#endif
s->scale_bias = 1.0;

31
libavcodec/fmtconvert.c
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@ -66,3 +66,34 @@ av_cold void ff_fmt_convert_init(FmtConvertContext *c, AVCodecContext *avctx)
if (HAVE_ALTIVEC) ff_fmt_convert_init_altivec(c, avctx);
if (HAVE_MMX) ff_fmt_convert_init_x86(c, avctx);
}
/* ffdshow custom code */
void float_interleave(float *dst, const float **src, long len, int channels)
{
int i,j,c;
if(channels==2){
for(i=0; i<len; i++){
dst[2*i] = src[0][i] / 32768.0f;
dst[2*i+1] = src[1][i] / 32768.0f;
}
}else{
for(c=0; c<channels; c++)
for(i=0, j=c; i<len; i++, j+=channels)
dst[j] = src[c][i] / 32768.0f;
}
}
void float_interleave_noscale(float *dst, const float **src, long len, int channels)
{
int i,j,c;
if(channels==2){
for(i=0; i<len; i++){
dst[2*i] = src[0][i];
dst[2*i+1] = src[1][i];
}
}else{
for(c=0; c<channels; c++)
for(i=0, j=c; i<len; i++, j+=channels)
dst[j] = src[c][i];
}
}

4
libavcodec/fmtconvert.h
View File

@ -76,4 +76,8 @@ void ff_fmt_convert_init_arm(FmtConvertContext *c, AVCodecContext *avctx);
void ff_fmt_convert_init_altivec(FmtConvertContext *c, AVCodecContext *avctx);
void ff_fmt_convert_init_x86(FmtConvertContext *c, AVCodecContext *avctx);
/* ffdshow custom code */
void float_interleave(float *dst, const float **src, long len, int channels);
void float_interleave_noscale(float *dst, const float **src, long len, int channels);
#endif /* AVCODEC_FMTCONVERT_H */