#include <math.h>
-#include "libavutil/audio_fifo.h"
#include "libavutil/avstring.h"
#include "libavutil/channel_layout.h"
#include "libavutil/float_dsp.h"
#include "libavcodec/avfft.h"
#include "avfilter.h"
+#include "filters.h"
#include "internal.h"
#include "audio.h"
int have_hrirs;
int eof_hrirs;
- int64_t pts;
int ir_len;
+ int air_len;
int mapping[64];
float *data_ir[2];
float *temp_src[2];
FFTComplex *temp_fft[2];
+ FFTComplex *temp_afft[2];
FFTContext *fft[2], *ifft[2];
FFTComplex *data_hrtf[2];
AVFloatDSPContext *fdsp;
struct headphone_inputs {
- AVAudioFifo *fifo;
AVFrame *frame;
int ir_len;
int delay_l;
float **ringbuffer;
float **temp_src;
FFTComplex **temp_fft;
+ FFTComplex **temp_afft;
} ThreadData;
static int headphone_convolute(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs)
float *ringbuffer = td->ringbuffer[jobnr];
float *temp_src = td->temp_src[jobnr];
const int ir_len = s->ir_len;
+ const int air_len = s->air_len;
const float *src = (const float *)in->data[0];
float *dst = (float *)out->data[0];
const int in_channels = in->channels;
if (l == s->lfe_channel) {
*dst += *(buffer[s->lfe_channel] + wr) * s->gain_lfe;
- temp_ir += FFALIGN(ir_len, 16);
+ temp_ir += air_len;
continue;
}
if (read + ir_len < buffer_length) {
memcpy(temp_src, bptr + read, ir_len * sizeof(*temp_src));
} else {
- int len = FFMIN(ir_len - (read % ir_len), buffer_length - read);
+ int len = FFMIN(air_len - (read % ir_len), buffer_length - read);
memcpy(temp_src, bptr + read, len * sizeof(*temp_src));
- memcpy(temp_src + len, bptr, (ir_len - len) * sizeof(*temp_src));
+ memcpy(temp_src + len, bptr, (air_len - len) * sizeof(*temp_src));
}
- dst[0] += s->fdsp->scalarproduct_float(temp_ir, temp_src, ir_len);
- temp_ir += FFALIGN(ir_len, 16);
+ dst[0] += s->fdsp->scalarproduct_float(temp_ir, temp_src, FFALIGN(ir_len, 32));
+ temp_ir += air_len;
}
- if (fabs(*dst) > 1)
- *n_clippings += 1;
+ if (fabsf(dst[0]) > 1)
+ n_clippings[0]++;
dst += 2;
src += in_channels;
const int buffer_length = s->buffer_length;
const uint32_t modulo = (uint32_t)buffer_length - 1;
FFTComplex *fft_in = s->temp_fft[jobnr];
+ FFTComplex *fft_acc = s->temp_afft[jobnr];
FFTContext *ifft = s->ifft[jobnr];
FFTContext *fft = s->fft[jobnr];
const int n_fft = s->n_fft;
dst += offset;
- n_read = FFMIN(s->ir_len, in->nb_samples);
+ n_read = FFMIN(ir_len, in->nb_samples);
for (j = 0; j < n_read; j++) {
dst[2 * j] = ringbuffer[wr];
ringbuffer[wr] = 0.0;
dst[2 * j] = 0;
}
+ memset(fft_acc, 0, sizeof(FFTComplex) * n_fft);
+
for (i = 0; i < in_channels; i++) {
if (i == s->lfe_channel) {
for (j = 0; j < in->nb_samples; j++) {
const float re = fft_in[j].re;
const float im = fft_in[j].im;
- fft_in[j].re = re * hcomplex->re - im * hcomplex->im;
- fft_in[j].im = re * hcomplex->im + im * hcomplex->re;
+ fft_acc[j].re += re * hcomplex->re - im * hcomplex->im;
+ fft_acc[j].im += re * hcomplex->im + im * hcomplex->re;
}
+ }
- av_fft_permute(ifft, fft_in);
- av_fft_calc(ifft, fft_in);
+ av_fft_permute(ifft, fft_acc);
+ av_fft_calc(ifft, fft_acc);
- for (j = 0; j < in->nb_samples; j++) {
- dst[2 * j] += fft_in[j].re * fft_scale;
- }
+ for (j = 0; j < in->nb_samples; j++) {
+ dst[2 * j] += fft_acc[j].re * fft_scale;
+ }
- for (j = 0; j < ir_len - 1; j++) {
- int write_pos = (wr + j) & modulo;
+ for (j = 0; j < ir_len - 1; j++) {
+ int write_pos = (wr + j) & modulo;
- *(ringbuffer + write_pos) += fft_in[in->nb_samples + j].re * fft_scale;
- }
+ *(ringbuffer + write_pos) += fft_acc[in->nb_samples + j].re * fft_scale;
}
for (i = 0; i < out->nb_samples; i++) {
- if (fabs(*dst) > 1) {
+ if (fabsf(dst[0]) > 1) {
n_clippings[0]++;
}
return 0;
}
-static int read_ir(AVFilterLink *inlink, AVFrame *frame)
+static int check_ir(AVFilterLink *inlink, int input_number)
{
AVFilterContext *ctx = inlink->dst;
HeadphoneContext *s = ctx->priv;
- int ir_len, max_ir_len, input_number, ret;
-
- for (input_number = 0; input_number < s->nb_inputs; input_number++)
- if (inlink == ctx->inputs[input_number])
- break;
-
- ret = av_audio_fifo_write(s->in[input_number].fifo, (void **)frame->extended_data,
- frame->nb_samples);
- av_frame_free(&frame);
-
- if (ret < 0)
- return ret;
+ int ir_len, max_ir_len;
- ir_len = av_audio_fifo_size(s->in[input_number].fifo);
+ ir_len = ff_inlink_queued_samples(inlink);
max_ir_len = 65536;
if (ir_len > max_ir_len) {
av_log(ctx, AV_LOG_ERROR, "Too big length of IRs: %d > %d.\n", ir_len, max_ir_len);
return 0;
}
-static int headphone_frame(HeadphoneContext *s, AVFilterLink *outlink, int max_nb_samples)
+static int headphone_frame(HeadphoneContext *s, AVFrame *in, AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
- AVFrame *in = s->in[0].frame;
int n_clippings[2] = { 0 };
ThreadData td;
AVFrame *out;
- av_audio_fifo_read(s->in[0].fifo, (void **)in->extended_data, s->size);
-
out = ff_get_audio_buffer(outlink, in->nb_samples);
- if (!out)
+ if (!out) {
+ av_frame_free(&in);
return AVERROR(ENOMEM);
- out->pts = s->pts;
- if (s->pts != AV_NOPTS_VALUE)
- s->pts += av_rescale_q(out->nb_samples, (AVRational){1, outlink->sample_rate}, outlink->time_base);
+ }
+ out->pts = in->pts;
td.in = in; td.out = out; td.write = s->write;
td.delay = s->delay; td.ir = s->data_ir; td.n_clippings = n_clippings;
td.ringbuffer = s->ringbuffer; td.temp_src = s->temp_src;
td.temp_fft = s->temp_fft;
+ td.temp_afft = s->temp_afft;
if (s->type == TIME_DOMAIN) {
ctx->internal->execute(ctx, headphone_convolute, &td, NULL, 2);
n_clippings[0] + n_clippings[1], out->nb_samples * 2);
}
- out->nb_samples = max_nb_samples;
+ av_frame_free(&in);
return ff_filter_frame(outlink, out);
}
int n_fft;
int i, j, k;
- s->buffer_length = 1 << (32 - ff_clz(s->ir_len));
- s->n_fft = n_fft = 1 << (32 - ff_clz(s->ir_len + s->size));
+ s->air_len = 1 << (32 - ff_clz(ir_len));
+ s->buffer_length = 1 << (32 - ff_clz(s->air_len));
+ s->n_fft = n_fft = 1 << (32 - ff_clz(ir_len + s->size));
if (s->type == FREQUENCY_DOMAIN) {
fft_in_l = av_calloc(n_fft, sizeof(*fft_in_l));
av_fft_end(s->fft[0]);
av_fft_end(s->fft[1]);
- s->fft[0] = av_fft_init(log2(s->n_fft), 0);
- s->fft[1] = av_fft_init(log2(s->n_fft), 0);
+ s->fft[0] = av_fft_init(av_log2(s->n_fft), 0);
+ s->fft[1] = av_fft_init(av_log2(s->n_fft), 0);
av_fft_end(s->ifft[0]);
av_fft_end(s->ifft[1]);
- s->ifft[0] = av_fft_init(log2(s->n_fft), 1);
- s->ifft[1] = av_fft_init(log2(s->n_fft), 1);
+ s->ifft[0] = av_fft_init(av_log2(s->n_fft), 1);
+ s->ifft[1] = av_fft_init(av_log2(s->n_fft), 1);
if (!s->fft[0] || !s->fft[1] || !s->ifft[0] || !s->ifft[1]) {
av_log(ctx, AV_LOG_ERROR, "Unable to create FFT contexts of size %d.\n", s->n_fft);
}
}
- s->data_ir[0] = av_calloc(FFALIGN(s->ir_len, 16), sizeof(float) * s->nb_irs);
- s->data_ir[1] = av_calloc(FFALIGN(s->ir_len, 16), sizeof(float) * s->nb_irs);
+ s->data_ir[0] = av_calloc(s->air_len, sizeof(float) * s->nb_irs);
+ s->data_ir[1] = av_calloc(s->air_len, sizeof(float) * s->nb_irs);
s->delay[0] = av_calloc(s->nb_irs, sizeof(float));
s->delay[1] = av_calloc(s->nb_irs, sizeof(float));
s->ringbuffer[1] = av_calloc(s->buffer_length, sizeof(float));
s->temp_fft[0] = av_calloc(s->n_fft, sizeof(FFTComplex));
s->temp_fft[1] = av_calloc(s->n_fft, sizeof(FFTComplex));
- if (!s->temp_fft[0] || !s->temp_fft[1]) {
+ s->temp_afft[0] = av_calloc(s->n_fft, sizeof(FFTComplex));
+ s->temp_afft[1] = av_calloc(s->n_fft, sizeof(FFTComplex));
+ if (!s->temp_fft[0] || !s->temp_fft[1] ||
+ !s->temp_afft[0] || !s->temp_afft[1]) {
ret = AVERROR(ENOMEM);
goto fail;
}
goto fail;
}
- s->in[0].frame = ff_get_audio_buffer(ctx->inputs[0], s->size);
- if (!s->in[0].frame) {
- ret = AVERROR(ENOMEM);
- goto fail;
- }
- for (i = 0; i < s->nb_inputs - 1; i++) {
- s->in[i + 1].frame = ff_get_audio_buffer(ctx->inputs[i + 1], s->ir_len);
- if (!s->in[i + 1].frame) {
- ret = AVERROR(ENOMEM);
- goto fail;
- }
- }
-
if (s->type == TIME_DOMAIN) {
- s->temp_src[0] = av_calloc(FFALIGN(ir_len, 16), sizeof(float));
- s->temp_src[1] = av_calloc(FFALIGN(ir_len, 16), sizeof(float));
+ s->temp_src[0] = av_calloc(s->air_len, sizeof(float));
+ s->temp_src[1] = av_calloc(s->air_len, sizeof(float));
- data_ir_l = av_calloc(nb_irs * FFALIGN(ir_len, 16), sizeof(*data_ir_l));
- data_ir_r = av_calloc(nb_irs * FFALIGN(ir_len, 16), sizeof(*data_ir_r));
+ data_ir_l = av_calloc(nb_irs * s->air_len, sizeof(*data_ir_l));
+ data_ir_r = av_calloc(nb_irs * s->air_len, sizeof(*data_ir_r));
if (!data_ir_r || !data_ir_l || !s->temp_src[0] || !s->temp_src[1]) {
ret = AVERROR(ENOMEM);
goto fail;
int delay_r = s->in[i + 1].delay_r;
float *ptr;
- av_audio_fifo_read(s->in[i + 1].fifo, (void **)s->in[i + 1].frame->extended_data, len);
+ ret = ff_inlink_consume_samples(ctx->inputs[i + 1], len, len, &s->in[i + 1].frame);
+ if (ret < 0)
+ goto fail;
ptr = (float *)s->in[i + 1].frame->extended_data[0];
if (s->hrir_fmt == HRIR_STEREO) {
if (idx == -1)
continue;
if (s->type == TIME_DOMAIN) {
- offset = idx * FFALIGN(len, 16);
+ offset = idx * s->air_len;
for (j = 0; j < len; j++) {
data_ir_l[offset + j] = ptr[len * 2 - j * 2 - 2] * gain_lin;
data_ir_r[offset + j] = ptr[len * 2 - j * 2 - 1] * gain_lin;
I = idx * 2;
if (s->type == TIME_DOMAIN) {
- offset = idx * FFALIGN(len, 16);
+ offset = idx * s->air_len;
for (j = 0; j < len; j++) {
data_ir_l[offset + j] = ptr[len * N - j * N - N + I ] * gain_lin;
data_ir_r[offset + j] = ptr[len * N - j * N - N + I + 1] * gain_lin;
}
}
}
+
+ av_frame_free(&s->in[i + 1].frame);
}
if (s->type == TIME_DOMAIN) {
- memcpy(s->data_ir[0], data_ir_l, sizeof(float) * nb_irs * FFALIGN(ir_len, 16));
- memcpy(s->data_ir[1], data_ir_r, sizeof(float) * nb_irs * FFALIGN(ir_len, 16));
+ memcpy(s->data_ir[0], data_ir_l, sizeof(float) * nb_irs * s->air_len);
+ memcpy(s->data_ir[1], data_ir_r, sizeof(float) * nb_irs * s->air_len);
} else {
s->data_hrtf[0] = av_calloc(n_fft * s->nb_irs, sizeof(FFTComplex));
s->data_hrtf[1] = av_calloc(n_fft * s->nb_irs, sizeof(FFTComplex));
fail:
+ for (i = 0; i < s->nb_inputs - 1; i++)
+ av_frame_free(&s->in[i + 1].frame);
+
av_freep(&data_ir_l);
av_freep(&data_ir_r);
return ret;
}
-static int filter_frame(AVFilterLink *inlink, AVFrame *in)
+static int activate(AVFilterContext *ctx)
{
- AVFilterContext *ctx = inlink->dst;
HeadphoneContext *s = ctx->priv;
+ AVFilterLink *inlink = ctx->inputs[0];
AVFilterLink *outlink = ctx->outputs[0];
- int ret = 0;
+ AVFrame *in = NULL;
+ int i, ret;
- ret = av_audio_fifo_write(s->in[0].fifo, (void **)in->extended_data,
- in->nb_samples);
- if (s->pts == AV_NOPTS_VALUE)
- s->pts = in->pts;
+ FF_FILTER_FORWARD_STATUS_BACK_ALL(ctx->outputs[0], ctx);
+ if (!s->eof_hrirs) {
+ for (i = 1; i < s->nb_inputs; i++) {
+ if (s->in[i].eof)
+ continue;
- av_frame_free(&in);
+ if ((ret = check_ir(ctx->inputs[i], i)) < 0)
+ return ret;
- if (ret < 0)
- return ret;
+ if (!s->in[i].eof) {
+ if (ff_outlink_get_status(ctx->inputs[i]) == AVERROR_EOF)
+ s->in[i].eof = 1;
+ }
+ }
+
+ for (i = 1; i < s->nb_inputs; i++) {
+ if (!s->in[i].eof)
+ break;
+ }
+
+ if (i != s->nb_inputs) {
+ if (ff_outlink_frame_wanted(ctx->outputs[0])) {
+ for (i = 1; i < s->nb_inputs; i++) {
+ if (!s->in[i].eof)
+ ff_inlink_request_frame(ctx->inputs[i]);
+ }
+ }
+
+ return 0;
+ } else {
+ s->eof_hrirs = 1;
+ }
+ }
if (!s->have_hrirs && s->eof_hrirs) {
ret = convert_coeffs(ctx, inlink);
return ret;
}
- if (s->have_hrirs) {
- while (av_audio_fifo_size(s->in[0].fifo) >= s->size) {
- ret = headphone_frame(s, outlink, s->size);
- if (ret < 0)
- return ret;
- }
+ if ((ret = ff_inlink_consume_samples(ctx->inputs[0], s->size, s->size, &in)) > 0) {
+ ret = headphone_frame(s, in, outlink);
+ if (ret < 0)
+ return ret;
}
+ if (ret < 0)
+ return ret;
+
+ FF_FILTER_FORWARD_STATUS(ctx->inputs[0], ctx->outputs[0]);
+ if (ff_outlink_frame_wanted(ctx->outputs[0]))
+ ff_inlink_request_frame(ctx->inputs[0]);
+
return 0;
}
.name = "in0",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_input,
- .filter_frame = filter_frame,
};
if ((ret = ff_insert_inpad(ctx, 0, &pad)) < 0)
return ret;
AVFilterPad pad = {
.name = name,
.type = AVMEDIA_TYPE_AUDIO,
- .filter_frame = read_ir,
};
if (!name)
return AVERROR(ENOMEM);
s->fdsp = avpriv_float_dsp_alloc(0);
if (!s->fdsp)
return AVERROR(ENOMEM);
- s->pts = AV_NOPTS_VALUE;
return 0;
}
AVFilterContext *ctx = outlink->src;
HeadphoneContext *s = ctx->priv;
AVFilterLink *inlink = ctx->inputs[0];
- int i;
if (s->hrir_fmt == HRIR_MULTI) {
AVFilterLink *hrir_link = ctx->inputs[1];
}
}
- for (i = 0; i < s->nb_inputs; i++) {
- s->in[i].fifo = av_audio_fifo_alloc(ctx->inputs[i]->format, ctx->inputs[i]->channels, 1024);
- if (!s->in[i].fifo)
- return AVERROR(ENOMEM);
- }
- s->gain_lfe = expf((s->gain - 3 * inlink->channels - 6 + s->lfe_gain) / 20 * M_LN10);
+ s->gain_lfe = expf((s->gain - 3 * inlink->channels + s->lfe_gain) / 20 * M_LN10);
return 0;
}
-static int request_frame(AVFilterLink *outlink)
-{
- AVFilterContext *ctx = outlink->src;
- HeadphoneContext *s = ctx->priv;
- int i, ret;
-
- for (i = 1; !s->eof_hrirs && i < s->nb_inputs; i++) {
- if (!s->in[i].eof) {
- ret = ff_request_frame(ctx->inputs[i]);
- if (ret == AVERROR_EOF) {
- s->in[i].eof = 1;
- ret = 0;
- }
- return ret;
- } else {
- if (i == s->nb_inputs - 1)
- s->eof_hrirs = 1;
- }
- }
-
- ret = ff_request_frame(ctx->inputs[0]);
- if (ret == AVERROR_EOF && av_audio_fifo_size(s->in[0].fifo) > 0 && s->have_hrirs) {
- int nb_samples = av_audio_fifo_size(s->in[0].fifo);
- AVFrame *in = ff_get_audio_buffer(ctx->inputs[0], s->size - nb_samples);
-
- if (!in)
- return AVERROR(ENOMEM);
-
- av_samples_set_silence(in->extended_data, 0,
- in->nb_samples,
- in->channels,
- in->format);
-
- ret = av_audio_fifo_write(s->in[0].fifo, (void **)in->extended_data,
- in->nb_samples);
- av_frame_free(&in);
- if (ret < 0)
- return ret;
- ret = headphone_frame(s, outlink, nb_samples);
-
- av_audio_fifo_drain(s->in[0].fifo, av_audio_fifo_size(s->in[0].fifo));
- }
-
- return ret;
-}
-
static av_cold void uninit(AVFilterContext *ctx)
{
HeadphoneContext *s = ctx->priv;
av_freep(&s->temp_src[1]);
av_freep(&s->temp_fft[0]);
av_freep(&s->temp_fft[1]);
+ av_freep(&s->temp_afft[0]);
+ av_freep(&s->temp_afft[1]);
av_freep(&s->data_hrtf[0]);
av_freep(&s->data_hrtf[1]);
av_freep(&s->fdsp);
for (i = 0; i < s->nb_inputs; i++) {
- av_frame_free(&s->in[i].frame);
- av_audio_fifo_free(s->in[i].fifo);
if (ctx->input_pads && i)
av_freep(&ctx->input_pads[i].name);
}
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
- .request_frame = request_frame,
},
{ NULL }
};
.init = init,
.uninit = uninit,
.query_formats = query_formats,
+ .activate = activate,
.inputs = NULL,
.outputs = outputs,
.flags = AVFILTER_FLAG_SLICE_THREADS | AVFILTER_FLAG_DYNAMIC_INPUTS,
projects / ffmpeg / blobdiff