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avfilter/af_sofalizer: add fltp sample format support

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pull/304/head^2
Paul B Mahol 6 years ago
parent 70c86deb8e
commit 530fc345ec
1 changed files with 61 additions and 28 deletions
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  1. 89
      libavfilter/af_sofalizer.c

89
libavfilter/af_sofalizer.c
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@ -340,8 +340,10 @@ static int sofalizer_convolute(AVFilterContext *ctx, void *arg, int jobnr, int n
float *temp_src = td->temp_src[jobnr];
const int ir_samples = s->sofa.ir_samples; /* length of one IR */
const int n_samples = s->sofa.n_samples;
const float *src = (const float *)in->data[0]; /* get pointer to audio input buffer */
float *dst = (float *)out->data[0]; /* get pointer to audio output buffer */
const int planar = in->format == AV_SAMPLE_FMT_FLTP;
const int mult = 1 + !planar;
const float *src = (const float *)in->extended_data[0]; /* get pointer to audio input buffer */
float *dst = (float *)out->extended_data[jobnr * planar]; /* get pointer to audio output buffer */
const int in_channels = s->n_conv; /* number of input channels */
/* ring buffer length is: longest IR plus max. delay -> next power of 2 */
const int buffer_length = s->buffer_length;
@ -352,7 +354,9 @@ static int sofalizer_convolute(AVFilterContext *ctx, void *arg, int jobnr, int n
int read;
int i, l;
dst += offset;
if (!planar)
dst += offset;
for (l = 0; l < in_channels; l++) {
/* get starting address of ringbuffer for each input channel */
buffer[l] = ringbuffer + l * buffer_length;
@ -362,9 +366,18 @@ static int sofalizer_convolute(AVFilterContext *ctx, void *arg, int jobnr, int n
const float *temp_ir = ir; /* using same set of IRs for each sample */
dst[0] = 0;
for (l = 0; l < in_channels; l++) {
/* write current input sample to ringbuffer (for each channel) */
buffer[l][wr] = src[l];
if (planar) {
for (l = 0; l < in_channels; l++) {
const float *srcp = (const float *)in->extended_data[l];
/* write current input sample to ringbuffer (for each channel) */
buffer[l][wr] = srcp[i];
}
} else {
for (l = 0; l < in_channels; l++) {
/* write current input sample to ringbuffer (for each channel) */
buffer[l][wr] = src[l];
}
}
/* loop goes through all channels to be convolved */
@ -374,7 +387,7 @@ static int sofalizer_convolute(AVFilterContext *ctx, void *arg, int jobnr, int n
if (l == s->lfe_channel) {
/* LFE is an input channel but requires no convolution */
/* apply gain to LFE signal and add to output buffer */
*dst += *(buffer[s->lfe_channel] + wr) * s->gain_lfe;
dst[0] += *(buffer[s->lfe_channel] + wr) * s->gain_lfe;
temp_ir += n_samples;
continue;
}
@ -403,7 +416,7 @@ static int sofalizer_convolute(AVFilterContext *ctx, void *arg, int jobnr, int n
n_clippings[0]++;
/* move output buffer pointer by +2 to get to next sample of processed channel: */
dst += 2;
dst += mult;
src += in_channels;
wr = (wr + 1) & modulo; /* update ringbuffer write position */
}
@ -424,8 +437,9 @@ static int sofalizer_fast_convolute(AVFilterContext *ctx, void *arg, int jobnr,
int *n_clippings = &td->n_clippings[jobnr];
float *ringbuffer = td->ringbuffer[jobnr];
const int n_samples = s->sofa.n_samples; /* length of one IR */
const float *src = (const float *)in->data[0]; /* get pointer to audio input buffer */
float *dst = (float *)out->data[0]; /* get pointer to audio output buffer */
const int planar = in->format == AV_SAMPLE_FMT_FLTP;
const int mult = 1 + !planar;
float *dst = (float *)out->extended_data[jobnr * planar]; /* get pointer to audio output buffer */
const int in_channels = s->n_conv; /* number of input channels */
/* ring buffer length is: longest IR plus max. delay -> next power of 2 */
const int buffer_length = s->buffer_length;
@ -443,14 +457,15 @@ static int sofalizer_fast_convolute(AVFilterContext *ctx, void *arg, int jobnr,
int n_read;
int i, j;
dst += offset;
if (!planar)
dst += offset;
/* find minimum between number of samples and output buffer length:
* (important, if one IR is longer than the output buffer) */
n_read = FFMIN(s->sofa.n_samples, in->nb_samples);
for (j = 0; j < n_read; j++) {
/* initialize output buf with saved signal from overflow buf */
dst[2 * j] = ringbuffer[wr];
dst[mult * j] = ringbuffer[wr];
ringbuffer[wr] = 0.0; /* re-set read samples to zero */
/* update ringbuffer read/write position */
wr = (wr + 1) & modulo;
@ -458,17 +473,26 @@ static int sofalizer_fast_convolute(AVFilterContext *ctx, void *arg, int jobnr,
/* initialize rest of output buffer with 0 */
for (j = n_read; j < in->nb_samples; j++) {
dst[2 * j] = 0;
dst[mult * j] = 0;
}
/* fill FFT accumulation with 0 */
memset(fft_acc, 0, sizeof(FFTComplex) * n_fft);
for (i = 0; i < n_conv; i++) {
const float *src = (const float *)in->extended_data[i * planar]; /* get pointer to audio input buffer */
if (i == s->lfe_channel) { /* LFE */
for (j = 0; j < in->nb_samples; j++) {
/* apply gain to LFE signal and add to output buffer */
dst[2 * j] += src[i + j * in_channels] * s->gain_lfe;
if (in->format == AV_SAMPLE_FMT_FLT) {
for (j = 0; j < in->nb_samples; j++) {
/* apply gain to LFE signal and add to output buffer */
dst[2 * j] += src[i + j * in_channels] * s->gain_lfe;
}
} else {
for (j = 0; j < in->nb_samples; j++) {
/* apply gain to LFE signal and add to output buffer */
dst[j] += src[j] * s->gain_lfe;
}
}
continue;
}
@ -480,10 +504,18 @@ static int sofalizer_fast_convolute(AVFilterContext *ctx, void *arg, int jobnr,
/* fill FFT input with 0 (we want to zero-pad) */
memset(fft_in, 0, sizeof(FFTComplex) * n_fft);
for (j = 0; j < in->nb_samples; j++) {
/* prepare input for FFT */
/* write all samples of current input channel to FFT input array */
fft_in[j].re = src[j * in_channels + i];
if (in->format == AV_SAMPLE_FMT_FLT) {
for (j = 0; j < in->nb_samples; j++) {
/* prepare input for FFT */
/* write all samples of current input channel to FFT input array */
fft_in[j].re = src[j * in_channels + i];
}
} else {
for (j = 0; j < in->nb_samples; j++) {
/* prepare input for FFT */
/* write all samples of current input channel to FFT input array */
fft_in[j].re = src[j];
}
}
/* transform input signal of current channel to frequency domain */
@ -508,7 +540,7 @@ static int sofalizer_fast_convolute(AVFilterContext *ctx, void *arg, int jobnr,
for (j = 0; j < in->nb_samples; j++) {
/* write output signal of current channel to output buffer */
dst[2 * j] += fft_acc[j].re * fft_scale;
dst[mult * j] += fft_acc[j].re * fft_scale;
}
for (j = 0; j < n_samples - 1; j++) { /* overflow length is IR length - 1 */
@ -521,12 +553,9 @@ static int sofalizer_fast_convolute(AVFilterContext *ctx, void *arg, int jobnr,
/* go through all samples of current output buffer: count clippings */
for (i = 0; i < out->nb_samples; i++) {
/* clippings counter */
if (fabsf(dst[0]) > 1) { /* if current output sample > 1 */
if (fabsf(dst[i * mult]) > 1) { /* if current output sample > 1 */
n_clippings[0]++;
}
/* move output buffer pointer by +2 to get to next sample of processed channel: */
dst += 2;
}
/* remember read/write position in ringbuffer for next call */
@ -580,10 +609,14 @@ static int query_formats(AVFilterContext *ctx)
AVFilterFormats *formats = NULL;
AVFilterChannelLayouts *layouts = NULL;
int ret, sample_rates[] = { 48000, -1 };
static const enum AVSampleFormat sample_fmts[] = {
AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE
};
ret = ff_add_format(&formats, AV_SAMPLE_FMT_FLT);
if (ret)
return ret;
formats = ff_make_format_list(sample_fmts);
if (!formats)
return AVERROR(ENOMEM);
ret = ff_set_common_formats(ctx, formats);
if (ret)
return ret;

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