openharmony/third_party_ffmpeg
1
0
Fork 0
mirror of https://gitee.com/openharmony/third_party_ffmpeg synced 2024-11-23 11:19:55 +00:00
Code Issues Actions Packages Projects Releases Wiki 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