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https://github.com/FFmpeg/FFmpeg.git
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avfilter/af_sofalizer: fix non-power of 2 IR length filtering in time domain
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d62cb29716
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@ -43,7 +43,8 @@
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typedef struct MySofa { /* contains data of one SOFA file */
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struct MYSOFA_EASY *easy;
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int n_samples; /* length of one impulse response (IR) */
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int ir_samples; /* length of one impulse response (IR) */
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int n_samples; /* ir_samples to next power of 2 */
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float *lir, *rir; /* IRs (time-domain) */
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int max_delay;
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} MySofa;
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@ -126,7 +127,8 @@ static int preload_sofa(AVFilterContext *ctx, char *filename, int *samplingrate)
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if (mysofa->DataSamplingRate.elements != 1)
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return AVERROR(EINVAL);
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*samplingrate = mysofa->DataSamplingRate.values[0];
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s->sofa.n_samples = mysofa->N;
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s->sofa.ir_samples = mysofa->N;
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s->sofa.n_samples = 1 << (32 - ff_clz(s->sofa.ir_samples));
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license = mysofa_getAttribute(mysofa->attributes, (char *)"License");
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if (license)
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av_log(ctx, AV_LOG_INFO, "SOFA license: %s\n", license);
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@ -291,7 +293,8 @@ static int sofalizer_convolute(AVFilterContext *ctx, void *arg, int jobnr, int n
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int *n_clippings = &td->n_clippings[jobnr];
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float *ringbuffer = td->ringbuffer[jobnr];
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float *temp_src = td->temp_src[jobnr];
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const int n_samples = s->sofa.n_samples; /* length of one IR */
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const int ir_samples = s->sofa.ir_samples; /* length of one IR */
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const int n_samples = s->sofa.n_samples;
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const float *src = (const float *)in->data[0]; /* get pointer to audio input buffer */
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float *dst = (float *)out->data[0]; /* get pointer to audio output buffer */
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const int in_channels = s->n_conv; /* number of input channels */
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@ -327,7 +330,7 @@ static int sofalizer_convolute(AVFilterContext *ctx, void *arg, int jobnr, int n
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/* LFE is an input channel but requires no convolution */
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/* apply gain to LFE signal and add to output buffer */
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*dst += *(buffer[s->lfe_channel] + wr) * s->gain_lfe;
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temp_ir += FFALIGN(n_samples, 32);
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temp_ir += n_samples;
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continue;
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}
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@ -346,8 +349,8 @@ static int sofalizer_convolute(AVFilterContext *ctx, void *arg, int jobnr, int n
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}
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/* multiply signal and IR, and add up the results */
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dst[0] += s->fdsp->scalarproduct_float(temp_ir, temp_src, n_samples);
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temp_ir += FFALIGN(n_samples, 32);
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dst[0] += s->fdsp->scalarproduct_float(temp_ir, temp_src, FFALIGN(ir_samples, 32));
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temp_ir += n_samples;
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}
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/* clippings counter */
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@ -563,6 +566,7 @@ static int load_data(AVFilterContext *ctx, int azim, int elev, float radius, int
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{
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struct SOFAlizerContext *s = ctx->priv;
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int n_samples;
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int ir_samples;
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int n_conv = s->n_conv; /* no. channels to convolve */
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int n_fft;
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float delay_l; /* broadband delay for each IR */
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@ -588,9 +592,10 @@ static int load_data(AVFilterContext *ctx, int azim, int elev, float radius, int
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}
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n_samples = s->sofa.n_samples;
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ir_samples = s->sofa.ir_samples;
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s->data_ir[0] = av_calloc(FFALIGN(n_samples, 32), sizeof(float) * s->n_conv);
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s->data_ir[1] = av_calloc(FFALIGN(n_samples, 32), sizeof(float) * s->n_conv);
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s->data_ir[0] = av_calloc(n_samples, sizeof(float) * s->n_conv);
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s->data_ir[1] = av_calloc(n_samples, sizeof(float) * s->n_conv);
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s->delay[0] = av_calloc(s->n_conv, sizeof(int));
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s->delay[1] = av_calloc(s->n_conv, sizeof(int));
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@ -600,16 +605,16 @@ static int load_data(AVFilterContext *ctx, int azim, int elev, float radius, int
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}
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/* get temporary IR for L and R channel */
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data_ir_l = av_calloc(n_conv * FFALIGN(n_samples, 32), sizeof(*data_ir_l));
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data_ir_r = av_calloc(n_conv * FFALIGN(n_samples, 32), sizeof(*data_ir_r));
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data_ir_l = av_calloc(n_conv * n_samples, sizeof(*data_ir_l));
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data_ir_r = av_calloc(n_conv * n_samples, sizeof(*data_ir_r));
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if (!data_ir_r || !data_ir_l) {
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ret = AVERROR(ENOMEM);
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goto fail;
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}
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if (s->type == TIME_DOMAIN) {
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s->temp_src[0] = av_calloc(FFALIGN(n_samples, 32), sizeof(float));
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s->temp_src[1] = av_calloc(FFALIGN(n_samples, 32), sizeof(float));
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s->temp_src[0] = av_calloc(n_samples, sizeof(float));
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s->temp_src[1] = av_calloc(n_samples, sizeof(float));
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if (!s->temp_src[0] || !s->temp_src[1]) {
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ret = AVERROR(ENOMEM);
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goto fail;
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@ -644,8 +649,8 @@ static int load_data(AVFilterContext *ctx, int azim, int elev, float radius, int
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/* get id of IR closest to desired position */
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mysofa_getfilter_float(s->sofa.easy, coordinates[0], coordinates[1], coordinates[2],
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data_ir_l + FFALIGN(n_samples, 32) * i,
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data_ir_r + FFALIGN(n_samples, 32) * i,
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data_ir_l + n_samples * i,
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data_ir_r + n_samples * i,
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&delay_l, &delay_r);
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s->delay[0][i] = delay_l * sample_rate;
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@ -656,7 +661,7 @@ static int load_data(AVFilterContext *ctx, int azim, int elev, float radius, int
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/* get size of ringbuffer (longest IR plus max. delay) */
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/* then choose next power of 2 for performance optimization */
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n_current = s->sofa.n_samples + s->sofa.max_delay;
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n_current = n_samples + s->sofa.max_delay;
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/* length of longest IR plus max. delay */
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n_max = FFMAX(n_max, n_current);
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@ -721,24 +726,24 @@ static int load_data(AVFilterContext *ctx, int azim, int elev, float radius, int
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for (i = 0; i < s->n_conv; i++) {
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float *lir, *rir;
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offset = i * FFALIGN(n_samples, 32); /* no. samples already written */
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offset = i * n_samples; /* no. samples already written */
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lir = data_ir_l + offset;
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rir = data_ir_r + offset;
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if (s->type == TIME_DOMAIN) {
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for (j = 0; j < n_samples; j++) {
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for (j = 0; j < ir_samples; j++) {
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/* load reversed IRs of the specified source position
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* sample-by-sample for left and right ear; and apply gain */
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s->data_ir[0][offset + j] = lir[n_samples - 1 - j] * gain_lin;
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s->data_ir[1][offset + j] = rir[n_samples - 1 - j] * gain_lin;
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s->data_ir[0][offset + j] = lir[ir_samples - 1 - j] * gain_lin;
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s->data_ir[1][offset + j] = rir[ir_samples - 1 - j] * gain_lin;
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}
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} else {
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memset(fft_in_l, 0, n_fft * sizeof(*fft_in_l));
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memset(fft_in_r, 0, n_fft * sizeof(*fft_in_r));
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offset = i * n_fft; /* no. samples already written */
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for (j = 0; j < n_samples; j++) {
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for (j = 0; j < ir_samples; j++) {
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/* load non-reversed IRs of the specified source position
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* sample-by-sample and apply gain,
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* L channel is loaded to real part, R channel to imag part,
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