virtualenv/FFmpeg
1
0
Fork 0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-23 12:43:46 +02:00
Code Issues Releases Activity

Move clipping of audio samples (for those codecs outputting float) from decoder

Browse Source 
to the audio conversion routines.

Originally committed as revision 22937 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Ronald S. Bultje 2010-04-21 17:57:48 +00:00
parent 6858ce2ffc
commit b1078e9fe6
12 changed files with 28 additions and 55 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

12
libavcodec/amrnbdec.c
View File

@ -796,7 +796,7 @@ static int synthesis(AMRContext *p, float *lpc,
float fixed_gain, const float *fixed_vector,
float *samples, uint8_t overflow)
{
int i, overflow_temp = 0;
int i;
float excitation[AMR_SUBFRAME_SIZE];
// if an overflow has been detected, the pitch vector is scaled down by a
@ -831,12 +831,10 @@ static int synthesis(AMRContext *p, float *lpc,
// detect overflow
for (i = 0; i < AMR_SUBFRAME_SIZE; i++)
if (fabsf(samples[i]) > AMR_SAMPLE_BOUND) {
overflow_temp = 1;
samples[i] = av_clipf(samples[i], -AMR_SAMPLE_BOUND,
AMR_SAMPLE_BOUND);
return 1;
}
return overflow_temp;
return 0;
}
/// @}
@ -1048,10 +1046,6 @@ static int amrnb_decode_frame(AVCodecContext *avctx, void *data, int *data_size,
highpass_poles, highpass_gain,
p->high_pass_mem, AMR_BLOCK_SIZE);
for (i = 0; i < AMR_BLOCK_SIZE; i++)
buf_out[i] = av_clipf(buf_out[i] * AMR_SAMPLE_SCALE,
-1.0, 32767.0 / 32768.0);
/* Update averaged lsf vector (used for fixed gain smoothing).
*
* Note that lsf_avg should not incorporate the current frame's LSFs

13
libavcodec/atrac1.c
View File

@ -305,20 +305,15 @@ static int atrac1_decode_frame(AVCodecContext *avctx, void *data,
at1_subband_synthesis(q, su, q->out_samples[ch]);
}
/* round, convert to 16bit and interleave */
/* interleave; FIXME, should create/use a DSP function */
if (q->channels == 1) {
/* mono */
q->dsp.vector_clipf(samples, q->out_samples[0], -32700.0 / (1 << 15),
32700.0 / (1 << 15), AT1_SU_SAMPLES);
memcpy(samples, q->out_samples[0], AT1_SU_SAMPLES * 4);
} else {
/* stereo */
for (i = 0; i < AT1_SU_SAMPLES; i++) {
samples[i * 2] = av_clipf(q->out_samples[0][i],
-32700.0 / (1 << 15),
32700.0 / (1 << 15));
samples[i * 2 + 1] = av_clipf(q->out_samples[1][i],
-32700.0 / (1 << 15),
32700.0 / (1 << 15));
samples[i * 2] = q->out_samples[0][i];
samples[i * 2 + 1] = q->out_samples[1][i];
}
}

14
libavcodec/audioconvert.c
View File

@ -209,7 +209,7 @@ if(ctx->fmt_pair == ofmt + SAMPLE_FMT_NB*ifmt){\
}
//FIXME put things below under ifdefs so we do not waste space for cases no codec will need
//FIXME rounding and clipping ?
//FIXME rounding ?
CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_U8 , *(const uint8_t*)pi)
else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)<<8)
@ -226,14 +226,14 @@ if(ctx->fmt_pair == ofmt + SAMPLE_FMT_NB*ifmt){\
else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_S32, *(const int32_t*)pi)
else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1<<31)))
else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1<<31)))
else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_FLT, lrintf(*(const float*)pi * (1<<7)) + 0x80)
else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_FLT, lrintf(*(const float*)pi * (1<<15)))
else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_FLT, lrintf(*(const float*)pi * (1<<31)))
else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_FLT, av_clip_uint8( lrintf(*(const float*)pi * (1<<7)) + 0x80))
else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_FLT, av_clip_int16( lrintf(*(const float*)pi * (1<<15))))
else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float*)pi * (1U<<31))))
else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_FLT, *(const float*)pi)
else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_FLT, *(const float*)pi)
else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_DBL, lrint(*(const double*)pi * (1<<7)) + 0x80)
else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_DBL, lrint(*(const double*)pi * (1<<15)))
else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_DBL, lrint(*(const double*)pi * (1<<31)))
else CONV(SAMPLE_FMT_U8 , uint8_t, SAMPLE_FMT_DBL, av_clip_uint8( lrint(*(const double*)pi * (1<<7)) + 0x80))
else CONV(SAMPLE_FMT_S16, int16_t, SAMPLE_FMT_DBL, av_clip_int16( lrint(*(const double*)pi * (1<<15))))
else CONV(SAMPLE_FMT_S32, int32_t, SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double*)pi * (1U<<31))))
else CONV(SAMPLE_FMT_FLT, float , SAMPLE_FMT_DBL, *(const double*)pi)
else CONV(SAMPLE_FMT_DBL, double , SAMPLE_FMT_DBL, *(const double*)pi)
else return -1;

10
libavcodec/qcelpdata.h
View File

@ -424,16 +424,6 @@ static const qcelp_vector * const qcelp_lspvq[5] = {
*/
#define QCELP_SCALE 8192.
/**
* the upper boundary of the clipping, depends on QCELP_SCALE
*/
#define QCELP_CLIP_UPPER_BOUND (8191.75/8192.)
/**
* the lower boundary of the clipping, depends on QCELP_SCALE
*/
#define QCELP_CLIP_LOWER_BOUND -1.
/**
* table for computing Ga (decoded linear codebook gain magnitude)
*

4
libavcodec/qcelpdec.c
View File

@ -834,10 +834,6 @@ erasure:
memcpy(q->formant_mem, q->formant_mem + 160, 10 * sizeof(float));
for(i=0; i<160; i++)
outbuffer[i] = av_clipf(outbuffer[i], QCELP_CLIP_LOWER_BOUND,
QCELP_CLIP_UPPER_BOUND);
memcpy(q->prev_lspf, quantized_lspf, sizeof(q->prev_lspf));
q->prev_bitrate = q->bitrate;

4
libavcodec/ra288.c
View File

@ -102,10 +102,6 @@ static void decode(RA288Context *ractx, float gain, int cb_coef)
gain_block[9] = 10 * log10(sum) - 32;
ff_celp_lp_synthesis_filterf(block, ractx->sp_lpc, buffer, 5, 36);
/* output */
for (i=0; i < 5; i++)
block[i] = av_clipf(block[i], -4095./4096., 4095./4096.);
}
/**

3
libavcodec/sipr.c
View File

@ -496,9 +496,6 @@ static void decode_frame(SiprContext *ctx, SiprParameters *params,
0.939805806,
ctx->highpass_filt_mem,
frame_size);
ctx->dsp.vector_clipf(out_data, out_data, -1, 32767./(1<<15), frame_size);
}
static av_cold int sipr_decoder_init(AVCodecContext * avctx)

3
libavcodec/sipr16k.c
View File

@ -264,9 +264,6 @@ void ff_sipr_decode_frame_16k(SiprContext *ctx, SiprParameters *params,
postfilter(out_data, synth, ctx->iir_mem, ctx->filt_mem, ctx->mem_preemph);
memcpy(ctx->iir_mem, Az[1], LP_FILTER_ORDER_16k * sizeof(float));
ctx->dsp.vector_clipf(out_data, out_data, -1, 32767./(1<<15), frame_size);
}
void ff_sipr_init_16k(SiprContext *ctx)

3
libavcodec/twinvq.c
View File

@ -850,9 +850,6 @@ static int twin_decode_frame(AVCodecContext * avctx, void *data,
return buf_size;
}
tctx->dsp.vector_clipf(out, out, -32700./(1<<15), 32700./(1<<15),
avctx->channels * mtab->size);
*data_size = mtab->size*avctx->channels*4;
return buf_size;

3
libavcodec/wmaprodec.c
View File

@ -1351,8 +1351,9 @@ static int decode_frame(WMAProDecodeCtx *s)
float* iptr = s->channel[i].out;
float* iend = iptr + s->samples_per_frame;
// FIXME should create/use a DSP function here
while (iptr < iend) {
*ptr = av_clipf(*iptr++, -1.0, 32767.0 / 32768.0);
*ptr = *iptr++;
ptr += incr;
}

3
libavcodec/wmavoice.c
View File

@ -1117,8 +1117,7 @@ static int synth_frame(AVCodecContext *ctx, GetBitContext *gb,
av_log_missing_feature(ctx, "APF", 0);
s->do_apf = 0;
} //else
for (n = 0; n < 160; n++)
samples[n] = av_clipf(synth[n], -1.0, 1.0);
memcpy(samples, synth, 160 * sizeof(synth[0]));
/* Cache values for next frame */
s->frame_cntr++;

11
libavutil/common.h
View File

@ -144,6 +144,17 @@ static inline av_const int16_t av_clip_int16(int a)
else return a;
}
/**
* Clips a signed 64-bit integer value into the -2147483648,2147483647 range.
* @param a value to clip
* @return clipped value
*/
static inline av_const int32_t av_clipl_int32(int64_t a)
{
if ((a+2147483648) & ~2147483647) return (a>>63) ^ 2147483647;
else return a;
}
/**
* Clips a float value into the amin-amax range.
* @param a value to clip