Mercurial > libavcodec.hg
changeset 11652:8b6f3d3b55cb libavcodec
Move clipping of audio samples (for those codecs outputting float) from decoder
to the audio conversion routines.
| author | rbultje |
|---|---|
| date | Wed, 21 Apr 2010 17:57:48 +0000 |
| parents | 926ad89ae57a |
| children | 28638e0d6e63 |
| files | amrnbdec.c atrac1.c audioconvert.c qcelpdata.h qcelpdec.c ra288.c sipr.c sipr16k.c twinvq.c wmaprodec.c wmavoice.c |
| diffstat | 11 files changed, 17 insertions(+), 55 deletions(-) [+] |
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--- a/amrnbdec.c Wed Apr 21 17:51:37 2010 +0000 +++ b/amrnbdec.c Wed Apr 21 17:57:48 2010 +0000 @@ -796,7 +796,7 @@ 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 @@ // 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 @@ 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
--- a/atrac1.c Wed Apr 21 17:51:37 2010 +0000 +++ b/atrac1.c Wed Apr 21 17:57:48 2010 +0000 @@ -305,20 +305,15 @@ 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]; } }
--- a/audioconvert.c Wed Apr 21 17:51:37 2010 +0000 +++ b/audioconvert.c Wed Apr 21 17:57:48 2010 +0000 @@ -209,7 +209,7 @@ } //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 @@ 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;
--- a/qcelpdata.h Wed Apr 21 17:51:37 2010 +0000 +++ b/qcelpdata.h Wed Apr 21 17:57:48 2010 +0000 @@ -425,16 +425,6 @@ #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) * * @note The table could fit in int16_t in x*8 form, but it seems
--- a/qcelpdec.c Wed Apr 21 17:51:37 2010 +0000 +++ b/qcelpdec.c Wed Apr 21 17:57:48 2010 +0000 @@ -834,10 +834,6 @@ 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;
--- a/ra288.c Wed Apr 21 17:51:37 2010 +0000 +++ b/ra288.c Wed Apr 21 17:57:48 2010 +0000 @@ -102,10 +102,6 @@ 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.); } /**
--- a/sipr.c Wed Apr 21 17:51:37 2010 +0000 +++ b/sipr.c Wed Apr 21 17:57:48 2010 +0000 @@ -496,9 +496,6 @@ 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)
--- a/sipr16k.c Wed Apr 21 17:51:37 2010 +0000 +++ b/sipr16k.c Wed Apr 21 17:57:48 2010 +0000 @@ -264,9 +264,6 @@ 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)
--- a/twinvq.c Wed Apr 21 17:51:37 2010 +0000 +++ b/twinvq.c Wed Apr 21 17:57:48 2010 +0000 @@ -850,9 +850,6 @@ 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;
--- a/wmaprodec.c Wed Apr 21 17:51:37 2010 +0000 +++ b/wmaprodec.c Wed Apr 21 17:57:48 2010 +0000 @@ -1351,8 +1351,9 @@ 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; }
--- a/wmavoice.c Wed Apr 21 17:51:37 2010 +0000 +++ b/wmavoice.c Wed Apr 21 17:57:48 2010 +0000 @@ -1117,8 +1117,7 @@ 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++;