int av_audio_convert(AVAudioConvert *ctx, void * const out[6], const int out_stride[6], const void * const in[6], const int in_stride[6], int len) { int ch; //FIXME optimize common cases for(ch=0; ch<ctx->out_channels; ch++){ const int is= in_stride[ch]; const int os= out_stride[ch]; const uint8_t *pi= in[ch]; uint8_t *po= out[ch]; uint8_t *end= po + os*len; if(!out[ch]) continue; #define CONV(ofmt, otype, ifmt, expr)\ if(ctx->fmt_pair == ofmt + AV_SAMPLE_FMT_NB*ifmt){\ do{\ *(otype*)po = expr; pi += is; po += os;\ }while(po < end);\ } //FIXME put things below under ifdefs so we do not waste space for cases no codec will need //FIXME rounding ? CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_U8 , *(const uint8_t*)pi) else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)<<8) else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)<<24) else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7))) else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7))) else CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_S16, (*(const int16_t*)pi>>8) + 0x80) else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S16, *(const int16_t*)pi) else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S16, *(const int16_t*)pi<<16) else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0 / (1<<15))) else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0 / (1<<15))) else CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_S32, (*(const int32_t*)pi>>24) + 0x80) else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S32, *(const int32_t*)pi>>16) else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S32, *(const int32_t*)pi) else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1U<<31))) else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1U<<31))) else CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8( lrintf(*(const float*)pi * (1<<7)) + 0x80)) else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16( lrintf(*(const float*)pi * (1<<15)))) else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float*)pi * (1U<<31)))) else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_FLT, *(const float*)pi) else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_FLT, *(const float*)pi) else CONV(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8( lrint(*(const double*)pi * (1<<7)) + 0x80)) else CONV(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16( lrint(*(const double*)pi * (1<<15)))) else CONV(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double*)pi * (1U<<31)))) else CONV(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_DBL, *(const double*)pi) else CONV(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_DBL, *(const double*)pi) else return -1; } return 0; }
int ff_audio_mix_set_matrix(AudioMix *am, const double *matrix, int stride) { int i, o; if ( am->in_channels <= 0 || am->in_channels > AVRESAMPLE_MAX_CHANNELS || am->out_channels <= 0 || am->out_channels > AVRESAMPLE_MAX_CHANNELS) { av_log(am->avr, AV_LOG_ERROR, "Invalid channel counts\n"); return AVERROR(EINVAL); } if (am->matrix) { av_free(am->matrix[0]); am->matrix = NULL; } #define CONVERT_MATRIX(type, expr) \ am->matrix_## type[0] = av_mallocz(am->out_channels * am->in_channels * \ sizeof(*am->matrix_## type[0])); \ if (!am->matrix_## type[0]) \ return AVERROR(ENOMEM); \ for (o = 0; o < am->out_channels; o++) { \ if (o > 0) \ am->matrix_## type[o] = am->matrix_## type[o - 1] + \ am->in_channels; \ for (i = 0; i < am->in_channels; i++) { \ double v = matrix[o * stride + i]; \ am->matrix_## type[o][i] = expr; \ } \ } \ am->matrix = (void **)am->matrix_## type; switch (am->coeff_type) { case AV_MIX_COEFF_TYPE_Q8: CONVERT_MATRIX(q8, av_clip_int16(lrint(256.0 * v))) break; case AV_MIX_COEFF_TYPE_Q15: CONVERT_MATRIX(q15, av_clipl_int32(llrint(32768.0 * v))) break; case AV_MIX_COEFF_TYPE_FLT: CONVERT_MATRIX(flt, v) break; default: av_log(am->avr, AV_LOG_ERROR, "Invalid mix coeff type\n"); return AVERROR(EINVAL); } return mix_function_init(am); }
static void set(uint8_t *a[], int ch, int index, int ch_count, enum AVSampleFormat f, double v){ uint8_t *p; if(av_sample_fmt_is_planar(f)){ f= av_get_alt_sample_fmt(f, 0); p= a[ch]; }else{ p= a[0]; index= ch + index*ch_count; } switch(f){ case AV_SAMPLE_FMT_U8 : ((uint8_t*)p)[index]= av_clip_uint8 (lrint((v+1.0)*127)); break; case AV_SAMPLE_FMT_S16: ((int16_t*)p)[index]= av_clip_int16 (lrint(v*32767)); break; case AV_SAMPLE_FMT_S32: ((int32_t*)p)[index]= av_clipl_int32(llrint(v*2147483647)); break; case AV_SAMPLE_FMT_FLT: ((float *)p)[index]= v; break; case AV_SAMPLE_FMT_DBL: ((double *)p)[index]= v; break; default: av_assert2(0); } }
CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80U)<<24) CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0f/ (1<<7))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_U8 , (*(const uint8_t*)pi - 0x80)*(1.0 / (1<<7))) CONV_FUNC(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_S16, (*(const int16_t*)pi>>8) + 0x80) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S16, *(const int16_t*)pi) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S16, *(const int16_t*)pi<<16) CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0f/ (1<<15))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_S16, *(const int16_t*)pi*(1.0 / (1<<15))) CONV_FUNC(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_S32, (*(const int32_t*)pi>>24) + 0x80) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_S32, *(const int32_t*)pi>>16) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_S32, *(const int32_t*)pi) CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0f/ (1U<<31))) CONV_FUNC(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_S32, *(const int32_t*)pi*(1.0 / (1U<<31))) CONV_FUNC(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_FLT, av_clip_uint8( lrintf(*(const float*)pi * (1<<7)) + 0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_FLT, av_clip_int16( lrintf(*(const float*)pi * (1<<15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_FLT, av_clipl_int32(llrintf(*(const float*)pi * (1U<<31)))) CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_FLT, *(const float*)pi) CONV_FUNC(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_FLT, *(const float*)pi) CONV_FUNC(AV_SAMPLE_FMT_U8 , uint8_t, AV_SAMPLE_FMT_DBL, av_clip_uint8( lrint(*(const double*)pi * (1<<7)) + 0x80)) CONV_FUNC(AV_SAMPLE_FMT_S16, int16_t, AV_SAMPLE_FMT_DBL, av_clip_int16( lrint(*(const double*)pi * (1<<15)))) CONV_FUNC(AV_SAMPLE_FMT_S32, int32_t, AV_SAMPLE_FMT_DBL, av_clipl_int32(llrint(*(const double*)pi * (1U<<31)))) CONV_FUNC(AV_SAMPLE_FMT_FLT, float , AV_SAMPLE_FMT_DBL, *(const double*)pi) CONV_FUNC(AV_SAMPLE_FMT_DBL, double , AV_SAMPLE_FMT_DBL, *(const double*)pi) #define FMT_PAIR_FUNC(out, in) [(out) + AV_SAMPLE_FMT_NB*(in)] = CONV_FUNC_NAME(out, in) static conv_func_type * const fmt_pair_to_conv_functions[AV_SAMPLE_FMT_NB*AV_SAMPLE_FMT_NB] = { FMT_PAIR_FUNC(AV_SAMPLE_FMT_U8 , AV_SAMPLE_FMT_U8 ), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_U8 ), FMT_PAIR_FUNC(AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_U8 ), FMT_PAIR_FUNC(AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_U8 ),
/** * builds a polyphase filterbank. * @param factor resampling factor * @param scale wanted sum of coefficients for each filter * @param filter_type filter type * @param kaiser_beta kaiser window beta * @return 0 on success, negative on error */ static int build_filter(ResampleContext *c, void *filter, double factor, int tap_count, int alloc, int phase_count, int scale, int filter_type, double kaiser_beta){ int ph, i; int ph_nb = phase_count % 2 ? phase_count : phase_count / 2 + 1; double x, y, w, t, s; double *tab = av_malloc_array(tap_count+1, sizeof(*tab)); double *sin_lut = av_malloc_array(ph_nb, sizeof(*sin_lut)); const int center= (tap_count-1)/2; int ret = AVERROR(ENOMEM); if (!tab || !sin_lut) goto fail; /* if upsampling, only need to interpolate, no filter */ if (factor > 1.0) factor = 1.0; if (factor == 1.0) { for (ph = 0; ph < ph_nb; ph++) sin_lut[ph] = sin(M_PI * ph / phase_count); } for(ph = 0; ph < ph_nb; ph++) { double norm = 0; s = sin_lut[ph]; for(i=0;i<=tap_count;i++) { x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor; if (x == 0) y = 1.0; else if (factor == 1.0) y = s / x; else y = sin(x) / x; switch(filter_type){ case SWR_FILTER_TYPE_CUBIC:{ const float d= -0.5; //first order derivative = -0.5 x = fabs(((double)(i - center) - (double)ph / phase_count) * factor); if(x<1.0) y= 1 - 3*x*x + 2*x*x*x + d*( -x*x + x*x*x); else y= d*(-4 + 8*x - 5*x*x + x*x*x); break;} case SWR_FILTER_TYPE_BLACKMAN_NUTTALL: w = 2.0*x / (factor*tap_count); t = -cos(w); y *= 0.3635819 - 0.4891775 * t + 0.1365995 * (2*t*t-1) - 0.0106411 * (4*t*t*t - 3*t); break; case SWR_FILTER_TYPE_KAISER: w = 2.0*x / (factor*tap_count*M_PI); y *= bessel(kaiser_beta*sqrt(FFMAX(1-w*w, 0))); break; default: av_assert0(0); } tab[i] = y; s = -s; if (i < tap_count) norm += y; } /* normalize so that an uniform color remains the same */ switch(c->format){ case AV_SAMPLE_FMT_S16P: for(i=0;i<tap_count;i++) ((int16_t*)filter)[ph * alloc + i] = av_clip_int16(lrintf(tab[i] * scale / norm)); if (phase_count % 2) break; if (tap_count % 2 == 0 || tap_count == 1) { for (i = 0; i < tap_count; i++) ((int16_t*)filter)[(phase_count-ph) * alloc + tap_count-1-i] = ((int16_t*)filter)[ph * alloc + i]; } else { for (i = 1; i <= tap_count; i++) ((int16_t*)filter)[(phase_count-ph) * alloc + tap_count-i] = av_clip_int16(lrintf(tab[i] * scale / (norm - tab[0] + tab[tap_count]))); } break; case AV_SAMPLE_FMT_S32P: for(i=0;i<tap_count;i++) ((int32_t*)filter)[ph * alloc + i] = av_clipl_int32(llrint(tab[i] * scale / norm)); if (phase_count % 2) break; if (tap_count % 2 == 0 || tap_count == 1) { for (i = 0; i < tap_count; i++) ((int32_t*)filter)[(phase_count-ph) * alloc + tap_count-1-i] = ((int32_t*)filter)[ph * alloc + i]; } else { for (i = 1; i <= tap_count; i++) ((int32_t*)filter)[(phase_count-ph) * alloc + tap_count-i] = av_clipl_int32(llrint(tab[i] * scale / (norm - tab[0] + tab[tap_count]))); } break; case AV_SAMPLE_FMT_FLTP: for(i=0;i<tap_count;i++) ((float*)filter)[ph * alloc + i] = tab[i] * scale / norm; if (phase_count % 2) break; if (tap_count % 2 == 0 || tap_count == 1) { for (i = 0; i < tap_count; i++) ((float*)filter)[(phase_count-ph) * alloc + tap_count-1-i] = ((float*)filter)[ph * alloc + i]; } else { for (i = 1; i <= tap_count; i++) ((float*)filter)[(phase_count-ph) * alloc + tap_count-i] = tab[i] * scale / (norm - tab[0] + tab[tap_count]); } break; case AV_SAMPLE_FMT_DBLP: for(i=0;i<tap_count;i++) ((double*)filter)[ph * alloc + i] = tab[i] * scale / norm; if (phase_count % 2) break; if (tap_count % 2 == 0 || tap_count == 1) { for (i = 0; i < tap_count; i++) ((double*)filter)[(phase_count-ph) * alloc + tap_count-1-i] = ((double*)filter)[ph * alloc + i]; } else { for (i = 1; i <= tap_count; i++) ((double*)filter)[(phase_count-ph) * alloc + tap_count-i] = tab[i] * scale / (norm - tab[0] + tab[tap_count]); } break; } } #if 0 { #define LEN 1024 int j,k; double sine[LEN + tap_count]; double filtered[LEN]; double maxff=-2, minff=2, maxsf=-2, minsf=2; for(i=0; i<LEN; i++){ double ss=0, sf=0, ff=0; for(j=0; j<LEN+tap_count; j++) sine[j]= cos(i*j*M_PI/LEN); for(j=0; j<LEN; j++){ double sum=0; ph=0; for(k=0; k<tap_count; k++) sum += filter[ph * tap_count + k] * sine[k+j]; filtered[j]= sum / (1<<FILTER_SHIFT); ss+= sine[j + center] * sine[j + center]; ff+= filtered[j] * filtered[j]; sf+= sine[j + center] * filtered[j]; } ss= sqrt(2*ss/LEN); ff= sqrt(2*ff/LEN); sf= 2*sf/LEN; maxff= FFMAX(maxff, ff); minff= FFMIN(minff, ff); maxsf= FFMAX(maxsf, sf); minsf= FFMIN(minsf, sf); if(i%11==0){ av_log(NULL, AV_LOG_ERROR, "i:%4d ss:%f ff:%13.6e-%13.6e sf:%13.6e-%13.6e\n", i, ss, maxff, minff, maxsf, minsf); minff=minsf= 2; maxff=maxsf= -2; } } } #endif ret = 0; fail: av_free(tab); av_free(sin_lut); return ret; }
int ff_audio_mix_set_matrix(AudioMix *am, const double *matrix, int stride) { int i, o, i0, o0, ret; char in_layout_name[128]; char out_layout_name[128]; if ( am->in_channels <= 0 || am->in_channels > AVRESAMPLE_MAX_CHANNELS || am->out_channels <= 0 || am->out_channels > AVRESAMPLE_MAX_CHANNELS) { av_log(am->avr, AV_LOG_ERROR, "Invalid channel counts\n"); return AVERROR(EINVAL); } if (am->matrix) { av_free(am->matrix[0]); am->matrix = NULL; } am->in_matrix_channels = am->in_channels; am->out_matrix_channels = am->out_channels; reduce_matrix(am, matrix, stride); #define CONVERT_MATRIX(type, expr) \ am->matrix_## type[0] = av_mallocz(am->out_matrix_channels * \ am->in_matrix_channels * \ sizeof(*am->matrix_## type[0])); \ if (!am->matrix_## type[0]) \ return AVERROR(ENOMEM); \ for (o = 0, o0 = 0; o < am->out_channels; o++) { \ if (am->output_zero[o] || am->output_skip[o]) \ continue; \ if (o0 > 0) \ am->matrix_## type[o0] = am->matrix_## type[o0 - 1] + \ am->in_matrix_channels; \ for (i = 0, i0 = 0; i < am->in_channels; i++) { \ double v; \ if (am->input_skip[i]) \ continue; \ v = matrix[o * stride + i]; \ am->matrix_## type[o0][i0] = expr; \ i0++; \ } \ o0++; \ } \ am->matrix = (void **)am->matrix_## type; if (am->in_matrix_channels && am->out_matrix_channels) { switch (am->coeff_type) { case AV_MIX_COEFF_TYPE_Q8: CONVERT_MATRIX(q8, av_clip_int16(lrint(256.0 * v))) break; case AV_MIX_COEFF_TYPE_Q15: CONVERT_MATRIX(q15, av_clipl_int32(llrint(32768.0 * v))) break; case AV_MIX_COEFF_TYPE_FLT: CONVERT_MATRIX(flt, v) break; default: av_log(am->avr, AV_LOG_ERROR, "Invalid mix coeff type\n"); return AVERROR(EINVAL); } } ret = mix_function_init(am); if (ret < 0) return ret; av_get_channel_layout_string(in_layout_name, sizeof(in_layout_name), am->in_channels, am->in_layout); av_get_channel_layout_string(out_layout_name, sizeof(out_layout_name), am->out_channels, am->out_layout); av_log(am->avr, AV_LOG_DEBUG, "audio_mix: %s to %s\n", in_layout_name, out_layout_name); av_log(am->avr, AV_LOG_DEBUG, "matrix size: %d x %d\n", am->in_matrix_channels, am->out_matrix_channels); for (o = 0; o < am->out_channels; o++) { for (i = 0; i < am->in_channels; i++) { if (am->output_zero[o]) av_log(am->avr, AV_LOG_DEBUG, " (ZERO)"); else if (am->input_skip[i] || am->output_skip[o]) av_log(am->avr, AV_LOG_DEBUG, " (SKIP)"); else av_log(am->avr, AV_LOG_DEBUG, " %0.3f ", matrix[o * am->in_channels + i]); } av_log(am->avr, AV_LOG_DEBUG, "\n"); } return 0; }
static int filter_frame(AVFilterLink *inlink, AVFrame *buf) { AVFilterContext *ctx = inlink->dst; ASyncContext *s = ctx->priv; AVFilterLink *outlink = ctx->outputs[0]; int nb_channels = av_get_channel_layout_nb_channels(buf->channel_layout); int64_t pts = (buf->pts == AV_NOPTS_VALUE) ? buf->pts : av_rescale_q(buf->pts, inlink->time_base, outlink->time_base); int out_size, ret; int64_t delta; int64_t new_pts; /* buffer data until we get the next timestamp */ if (s->pts == AV_NOPTS_VALUE || pts == AV_NOPTS_VALUE) { if (pts != AV_NOPTS_VALUE) { s->pts = pts - get_delay(s); } return write_to_fifo(s, buf); } if (s->first_pts != AV_NOPTS_VALUE) { handle_trimming(ctx); if (!avresample_available(s->avr)) return write_to_fifo(s, buf); } /* when we have two timestamps, compute how many samples would we have * to add/remove to get proper sync between data and timestamps */ delta = pts - s->pts - get_delay(s); out_size = avresample_available(s->avr); if (llabs(delta) > s->min_delta || (s->first_frame && delta && s->first_pts != AV_NOPTS_VALUE)) { av_log(ctx, AV_LOG_VERBOSE, "Discontinuity - %"PRId64" samples.\n", delta); out_size = av_clipl_int32((int64_t)out_size + delta); } else { if (s->resample) { // adjust the compensation if delta is non-zero int delay = get_delay(s); int comp = s->comp + av_clip(delta * inlink->sample_rate / delay, -s->max_comp, s->max_comp); if (comp != s->comp) { av_log(ctx, AV_LOG_VERBOSE, "Compensating %d samples per second.\n", comp); if (avresample_set_compensation(s->avr, comp, inlink->sample_rate) == 0) { s->comp = comp; } } } // adjust PTS to avoid monotonicity errors with input PTS jitter pts -= delta; delta = 0; } if (out_size > 0) { AVFrame *buf_out = ff_get_audio_buffer(outlink, out_size); if (!buf_out) { ret = AVERROR(ENOMEM); goto fail; } if (s->first_frame && delta > 0) { int planar = av_sample_fmt_is_planar(buf_out->format); int planes = planar ? nb_channels : 1; int block_size = av_get_bytes_per_sample(buf_out->format) * (planar ? 1 : nb_channels); int ch; av_samples_set_silence(buf_out->extended_data, 0, delta, nb_channels, buf->format); for (ch = 0; ch < planes; ch++) buf_out->extended_data[ch] += delta * block_size; avresample_read(s->avr, buf_out->extended_data, out_size); for (ch = 0; ch < planes; ch++) buf_out->extended_data[ch] -= delta * block_size; } else { avresample_read(s->avr, buf_out->extended_data, out_size); if (delta > 0) { av_samples_set_silence(buf_out->extended_data, out_size - delta, delta, nb_channels, buf->format); } } buf_out->pts = s->pts; ret = ff_filter_frame(outlink, buf_out); if (ret < 0) goto fail; s->got_output = 1; } else if (avresample_available(s->avr)) { av_log(ctx, AV_LOG_WARNING, "Non-monotonous timestamps, dropping " "whole buffer.\n"); } /* drain any remaining buffered data */ avresample_read(s->avr, NULL, avresample_available(s->avr)); new_pts = pts - avresample_get_delay(s->avr); /* check for s->pts monotonicity */ if (new_pts > s->pts) { s->pts = new_pts; ret = avresample_convert(s->avr, NULL, 0, 0, buf->extended_data, buf->linesize[0], buf->nb_samples); } else { av_log(ctx, AV_LOG_WARNING, "Non-monotonous timestamps, dropping " "whole buffer.\n"); ret = 0; } s->first_frame = 0; fail: av_frame_free(&buf); return ret; }
int ff_audio_mix_set_matrix(AudioMix *am, const double *matrix, int stride) { int i, o, i0, o0; if ( am->in_channels <= 0 || am->in_channels > AVRESAMPLE_MAX_CHANNELS || am->out_channels <= 0 || am->out_channels > AVRESAMPLE_MAX_CHANNELS) { av_log(am->avr, AV_LOG_ERROR, "Invalid channel counts\n"); return AVERROR(EINVAL); } if (am->matrix) { av_free(am->matrix[0]); am->matrix = NULL; } memset(am->output_zero, 0, sizeof(am->output_zero)); memset(am->input_skip, 0, sizeof(am->input_skip)); memset(am->output_skip, 0, sizeof(am->output_zero)); am->in_matrix_channels = am->in_channels; am->out_matrix_channels = am->out_channels; /* exclude output channels if they can be zeroed instead of mixed */ for (o = 0; o < am->out_channels; o++) { int zero = 1; /* check if the output is always silent */ for (i = 0; i < am->in_channels; i++) { if (matrix[o * stride + i] != 0.0) { zero = 0; break; } } /* check if the corresponding input channel makes a contribution to any output channel */ if (o < am->in_channels) { for (i = 0; i < am->out_channels; i++) { if (matrix[i * stride + o] != 0.0) { zero = 0; break; } } } if (zero) { am->output_zero[o] = 1; am->out_matrix_channels--; } } if (am->out_matrix_channels == 0) { am->in_matrix_channels = 0; return 0; } /* skip input channels that contribute fully only to the corresponding output channel */ for (i = 0; i < FFMIN(am->in_channels, am->out_channels); i++) { int skip = 1; for (o = 0; o < am->out_channels; o++) { if ((o != i && matrix[o * stride + i] != 0.0) || (o == i && matrix[o * stride + i] != 1.0)) { skip = 0; break; } } if (skip) { am->input_skip[i] = 1; am->in_matrix_channels--; } } /* skip input channels that do not contribute to any output channel */ for (; i < am->in_channels; i++) { int contrib = 0; for (o = 0; o < am->out_channels; o++) { if (matrix[o * stride + i] != 0.0) { contrib = 1; break; } } if (!contrib) { am->input_skip[i] = 1; am->in_matrix_channels--; } } if (am->in_matrix_channels == 0) { am->out_matrix_channels = 0; return 0; } /* skip output channels that only get full contribution from the corresponding input channel */ for (o = 0; o < FFMIN(am->in_channels, am->out_channels); o++) { int skip = 1; for (i = 0; i < am->in_channels; i++) { if ((o != i && matrix[o * stride + i] != 0.0) || (o == i && matrix[o * stride + i] != 1.0)) { skip = 0; break; } } if (skip) { am->output_skip[o] = 1; am->out_matrix_channels--; } } if (am->out_matrix_channels == 0) { am->in_matrix_channels = 0; return 0; } #define CONVERT_MATRIX(type, expr) \ am->matrix_## type[0] = av_mallocz(am->out_matrix_channels * \ am->in_matrix_channels * \ sizeof(*am->matrix_## type[0])); \ if (!am->matrix_## type[0]) \ return AVERROR(ENOMEM); \ for (o = 0, o0 = 0; o < am->out_channels; o++) { \ if (am->output_zero[o] || am->output_skip[o]) \ continue; \ if (o0 > 0) \ am->matrix_## type[o0] = am->matrix_## type[o0 - 1] + \ am->in_matrix_channels; \ for (i = 0, i0 = 0; i < am->in_channels; i++) { \ double v; \ if (am->input_skip[i]) \ continue; \ v = matrix[o * stride + i]; \ am->matrix_## type[o0][i0] = expr; \ i0++; \ } \ o0++; \ } \ am->matrix = (void **)am->matrix_## type; switch (am->coeff_type) { case AV_MIX_COEFF_TYPE_Q8: CONVERT_MATRIX(q8, av_clip_int16(lrint(256.0 * v))) break; case AV_MIX_COEFF_TYPE_Q15: CONVERT_MATRIX(q15, av_clipl_int32(llrint(32768.0 * v))) break; case AV_MIX_COEFF_TYPE_FLT: CONVERT_MATRIX(flt, v) break; default: av_log(am->avr, AV_LOG_ERROR, "Invalid mix coeff type\n"); return AVERROR(EINVAL); } return mix_function_init(am); }
int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix, int stride) { int in_channels, out_channels, i, o; in_channels = av_get_channel_layout_nb_channels(avr->in_channel_layout); out_channels = av_get_channel_layout_nb_channels(avr->out_channel_layout); if ( in_channels <= 0 || in_channels > AVRESAMPLE_MAX_CHANNELS || out_channels <= 0 || out_channels > AVRESAMPLE_MAX_CHANNELS) { av_log(avr, AV_LOG_ERROR, "Invalid channel layouts\n"); return AVERROR(EINVAL); } if (avr->am->matrix) { av_free(avr->am->matrix[0]); avr->am->matrix = NULL; } #define CONVERT_MATRIX(type, expr) \ avr->am->matrix_## type[0] = av_mallocz(out_channels * in_channels * \ sizeof(*avr->am->matrix_## type[0])); \ if (!avr->am->matrix_## type[0]) \ return AVERROR(ENOMEM); \ for (o = 0; o < out_channels; o++) { \ if (o > 0) \ avr->am->matrix_## type[o] = avr->am->matrix_## type[o - 1] + \ in_channels; \ for (i = 0; i < in_channels; i++) { \ double v = matrix[o * stride + i]; \ avr->am->matrix_## type[o][i] = expr; \ } \ } \ avr->am->matrix = (void **)avr->am->matrix_## type; switch (avr->mix_coeff_type) { case AV_MIX_COEFF_TYPE_Q8: CONVERT_MATRIX(q8, av_clip_int16(lrint(256.0 * v))) break; case AV_MIX_COEFF_TYPE_Q15: CONVERT_MATRIX(q15, av_clipl_int32(llrint(32768.0 * v))) break; case AV_MIX_COEFF_TYPE_FLT: CONVERT_MATRIX(flt, v) break; default: av_log(avr, AV_LOG_ERROR, "Invalid mix coeff type\n"); return AVERROR(EINVAL); } /* TODO: detect situations where we can just swap around pointers instead of doing matrix multiplications with 0.0 and 1.0 */ /* set AudioMix params */ avr->am->in_layout = avr->in_channel_layout; avr->am->out_layout = avr->out_channel_layout; avr->am->in_channels = in_channels; avr->am->out_channels = out_channels; return 0; }