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mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-23 12:43:46 +02:00

avfilter/avf_showspectrum: stop using AVAudioFifo to keep samples

Fixes limitation of input duration that showspectrumpic can process.
This commit is contained in:
Paul B Mahol 2022-02-14 21:02:03 +01:00
parent 9f4dff61f7
commit 81df787b53

View File

@ -30,7 +30,6 @@
#include <math.h>
#include "libavutil/tx.h"
#include "libavutil/audio_fifo.h"
#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/channel_layout.h"
@ -53,6 +52,8 @@ enum ColorMode { CHANNEL, INTENSITY, RAINBOW, MORELAND, NEBULAE, FIRE, FIERY,
enum SlideMode { REPLACE, SCROLL, FULLFRAME, RSCROLL, LREPLACE, NB_SLIDES };
enum Orientation { VERTICAL, HORIZONTAL, NB_ORIENTATIONS };
#define DEFAULT_LENGTH 300
typedef struct ShowSpectrumContext {
const AVClass *class;
int w, h;
@ -60,6 +61,7 @@ typedef struct ShowSpectrumContext {
AVRational auto_frame_rate;
AVRational frame_rate;
AVFrame *outpicref;
AVFrame *in_frame;
int nb_display_channels;
int orientation;
int channel_width;
@ -95,7 +97,6 @@ typedef struct ShowSpectrumContext {
int hop_size;
float *combine_buffer; ///< color combining buffer (3 * h items)
float **color_buffer; ///< color buffer (3 * h * ch items)
AVAudioFifo *fifo;
int64_t pts;
int64_t old_pts;
int old_len;
@ -104,7 +105,12 @@ typedef struct ShowSpectrumContext {
int start_x, start_y;
float drange, limit;
float dmin, dmax;
uint64_t samples;
int (*plot_channel)(AVFilterContext *ctx, void *arg, int jobnr, int nb_jobs);
AVFrame **frames;
unsigned int nb_frames;
unsigned int frames_size;
} ShowSpectrumContext;
#define OFFSET(x) offsetof(ShowSpectrumContext, x)
@ -330,12 +336,19 @@ static av_cold void uninit(AVFilterContext *ctx)
}
av_freep(&s->magnitudes);
av_frame_free(&s->outpicref);
av_audio_fifo_free(s->fifo);
av_frame_free(&s->in_frame);
if (s->phases) {
for (i = 0; i < s->nb_display_channels; i++)
av_freep(&s->phases[i]);
}
av_freep(&s->phases);
while (s->nb_frames > 0) {
av_frame_free(&s->frames[s->nb_frames - 1]);
s->nb_frames--;
}
av_freep(&s->frames);
}
static int query_formats(AVFilterContext *ctx)
@ -380,6 +393,13 @@ static int run_channel_fft(AVFilterContext *ctx, void *arg, int jobnr, int nb_jo
/* fill FFT input with the number of samples available */
const float *p = (float *)fin->extended_data[ch];
float *in_frame = (float *)s->in_frame->extended_data[ch];
memmove(in_frame, in_frame + s->hop_size, (s->fft_size - s->hop_size) * sizeof(float));
memcpy(in_frame + s->fft_size - s->hop_size, p, fin->nb_samples * sizeof(float));
for (int i = fin->nb_samples; i < s->hop_size; i++)
in_frame[i + s->fft_size - s->hop_size] = 0.f;
if (s->stop) {
float theta, phi, psi, a, b, S, c;
@ -391,7 +411,7 @@ static int run_channel_fft(AVFilterContext *ctx, void *arg, int jobnr, int nb_jo
int M = s->win_size / 2;
for (n = 0; n < s->win_size; n++) {
s->fft_data[ch][n].re = p[n] * window_func_lut[n];
s->fft_data[ch][n].re = in_frame[n] * window_func_lut[n];
s->fft_data[ch][n].im = 0;
}
@ -458,7 +478,7 @@ static int run_channel_fft(AVFilterContext *ctx, void *arg, int jobnr, int nb_jo
}
} else {
for (n = 0; n < s->win_size; n++) {
s->fft_in[ch][n].re = p[n] * window_func_lut[n];
s->fft_in[ch][n].re = in_frame[n] * window_func_lut[n];
s->fft_in[ch][n].im = 0;
}
@ -726,7 +746,7 @@ static float get_iscale(AVFilterContext *ctx, int scale, float a)
return a;
}
static int draw_legend(AVFilterContext *ctx, int samples)
static int draw_legend(AVFilterContext *ctx, uint64_t samples)
{
ShowSpectrumContext *s = ctx->priv;
AVFilterLink *inlink = ctx->inputs[0];
@ -1239,10 +1259,15 @@ static int config_output(AVFilterLink *outlink)
av_log(ctx, AV_LOG_VERBOSE, "s:%dx%d FFT window size:%d\n",
s->w, s->h, s->win_size);
av_audio_fifo_free(s->fifo);
s->fifo = av_audio_fifo_alloc(inlink->format, inlink->channels, s->win_size);
if (!s->fifo)
s->in_frame = ff_get_audio_buffer(inlink, s->win_size);
if (!s->in_frame)
return AVERROR(ENOMEM);
s->frames = av_fast_realloc(NULL, &s->frames_size,
DEFAULT_LENGTH * sizeof(*(s->frames)));
if (!s->frames)
return AVERROR(ENOMEM);
return 0;
}
@ -1439,7 +1464,7 @@ static int plot_spectrum_column(AVFilterLink *inlink, AVFrame *insamples)
}
if (s->sliding != FULLFRAME || s->xpos == 0)
outpicref->pts = av_rescale_q(insamples->pts, inlink->time_base, outlink->time_base);
s->pts = outpicref->pts = av_rescale_q(insamples->pts, inlink->time_base, outlink->time_base);
if (s->sliding == LREPLACE) {
s->xpos--;
@ -1507,65 +1532,40 @@ static int activate(AVFilterContext *ctx)
AVFilterLink *inlink = ctx->inputs[0];
AVFilterLink *outlink = ctx->outputs[0];
ShowSpectrumContext *s = ctx->priv;
int ret;
int ret, status;
int64_t pts;
FF_FILTER_FORWARD_STATUS_BACK(outlink, inlink);
if (av_audio_fifo_size(s->fifo) < s->win_size) {
AVFrame *frame = NULL;
if (s->outpicref) {
AVFrame *fin;
ret = ff_inlink_consume_frame(inlink, &frame);
ret = ff_inlink_consume_samples(inlink, s->hop_size, s->hop_size, &fin);
if (ret < 0)
return ret;
if (ret > 0) {
s->pts = frame->pts;
s->consumed = 0;
s->consumed += fin->nb_samples;
ff_filter_execute(ctx, run_channel_fft, fin, NULL, s->nb_display_channels);
av_audio_fifo_write(s->fifo, (void **)frame->extended_data, frame->nb_samples);
av_frame_free(&frame);
}
}
if (s->data == D_MAGNITUDE)
ff_filter_execute(ctx, calc_channel_magnitudes, NULL, NULL, s->nb_display_channels);
if (s->outpicref && (av_audio_fifo_size(s->fifo) >= s->win_size ||
ff_outlink_get_status(inlink))) {
AVFrame *fin = ff_get_audio_buffer(inlink, s->win_size);
if (!fin)
return AVERROR(ENOMEM);
if (s->data == D_PHASE)
ff_filter_execute(ctx, calc_channel_phases, NULL, NULL, s->nb_display_channels);
fin->pts = s->pts + s->consumed;
s->consumed += s->hop_size;
ret = av_audio_fifo_peek(s->fifo, (void **)fin->extended_data,
FFMIN(s->win_size, av_audio_fifo_size(s->fifo)));
if (ret < 0) {
if (s->data == D_UPHASE)
ff_filter_execute(ctx, calc_channel_uphases, NULL, NULL, s->nb_display_channels);
ret = plot_spectrum_column(inlink, fin);
av_frame_free(&fin);
return ret;
if (ret <= 0)
return ret;
}
av_assert0(fin->nb_samples == s->win_size);
ff_filter_execute(ctx, run_channel_fft, fin, NULL, s->nb_display_channels);
if (s->data == D_MAGNITUDE)
ff_filter_execute(ctx, calc_channel_magnitudes, NULL, NULL, s->nb_display_channels);
if (s->data == D_PHASE)
ff_filter_execute(ctx, calc_channel_phases, NULL, NULL, s->nb_display_channels);
if (s->data == D_UPHASE)
ff_filter_execute(ctx, calc_channel_uphases, NULL, NULL, s->nb_display_channels);
ret = plot_spectrum_column(inlink, fin);
av_frame_free(&fin);
av_audio_fifo_drain(s->fifo, s->hop_size);
if (ret <= 0 && !ff_outlink_get_status(inlink))
return ret;
}
if (ff_outlink_get_status(inlink) == AVERROR_EOF &&
s->sliding == FULLFRAME &&
s->xpos > 0 && s->outpicref) {
int64_t pts;
if (s->orientation == VERTICAL) {
for (int i = 0; i < outlink->h; i++) {
@ -1588,17 +1588,19 @@ static int activate(AVFilterContext *ctx)
return 0;
}
FF_FILTER_FORWARD_STATUS(inlink, outlink);
if (av_audio_fifo_size(s->fifo) >= s->win_size ||
ff_inlink_queued_frames(inlink) > 0 ||
ff_outlink_get_status(inlink) == AVERROR_EOF) {
if (ff_inlink_acknowledge_status(inlink, &status, &pts)) {
if (status == AVERROR_EOF) {
ff_outlink_set_status(outlink, status, s->pts);
return 0;
}
}
if (ff_inlink_queued_samples(inlink) >= s->hop_size) {
ff_filter_set_ready(ctx, 10);
return 0;
}
if (ff_outlink_frame_wanted(outlink) && av_audio_fifo_size(s->fifo) < s->win_size &&
ff_inlink_queued_frames(inlink) == 0 &&
ff_outlink_get_status(inlink) != AVERROR_EOF) {
if (ff_outlink_frame_wanted(outlink)) {
ff_inlink_request_frame(inlink);
return 0;
}
@ -1691,39 +1693,53 @@ static int showspectrumpic_request_frame(AVFilterLink *outlink)
AVFilterContext *ctx = outlink->src;
ShowSpectrumContext *s = ctx->priv;
AVFilterLink *inlink = ctx->inputs[0];
int ret, samples;
int ret;
ret = ff_request_frame(inlink);
samples = av_audio_fifo_size(s->fifo);
if (ret == AVERROR_EOF && s->outpicref && samples > 0) {
if (ret == AVERROR_EOF && s->outpicref && s->samples > 0) {
int consumed = 0;
int x = 0, sz = s->orientation == VERTICAL ? s->w : s->h;
unsigned int nb_frame = 0;
int ch, spf, spb;
int src_offset = 0;
AVFrame *fin;
spf = s->win_size * (samples / ((s->win_size * sz) * ceil(samples / (float)(s->win_size * sz))));
spf = s->win_size * (s->samples / ((s->win_size * sz) * ceil(s->samples / (float)(s->win_size * sz))));
spf = FFMAX(1, spf);
spb = (samples / (spf * sz)) * spf;
spb = (s->samples / (spf * sz)) * spf;
fin = ff_get_audio_buffer(inlink, s->win_size);
fin = ff_get_audio_buffer(inlink, spf);
if (!fin)
return AVERROR(ENOMEM);
while (x < sz) {
ret = av_audio_fifo_peek(s->fifo, (void **)fin->extended_data, s->win_size);
if (ret < 0) {
av_frame_free(&fin);
return ret;
}
int acc_samples = 0;
int dst_offset = 0;
av_audio_fifo_drain(s->fifo, spf);
while (nb_frame <= s->nb_frames) {
AVFrame *cur_frame = s->frames[nb_frame];
int cur_frame_samples = cur_frame->nb_samples;
int nb_samples = 0;
if (ret < s->win_size) {
for (ch = 0; ch < s->nb_display_channels; ch++) {
memset(fin->extended_data[ch] + ret * sizeof(float), 0,
(s->win_size - ret) * sizeof(float));
if (acc_samples < spf) {
nb_samples = FFMIN(spf - acc_samples, cur_frame_samples - src_offset);
acc_samples += nb_samples;
av_samples_copy(fin->extended_data, cur_frame->extended_data,
dst_offset, src_offset, nb_samples,
cur_frame->channels, AV_SAMPLE_FMT_FLTP);
}
src_offset += nb_samples;
dst_offset += nb_samples;
if (cur_frame_samples <= src_offset) {
av_frame_free(&s->frames[nb_frame]);
nb_frame++;
src_offset = 0;
}
if (acc_samples == spf)
break;
}
ff_filter_execute(ctx, run_channel_fft, fin, NULL, s->nb_display_channels);
@ -1746,7 +1762,7 @@ static int showspectrumpic_request_frame(AVFilterLink *outlink)
s->outpicref->pts = 0;
if (s->legend)
draw_legend(ctx, samples);
draw_legend(ctx, s->samples);
ret = ff_filter_frame(outlink, s->outpicref);
s->outpicref = NULL;
@ -1759,11 +1775,20 @@ static int showspectrumpic_filter_frame(AVFilterLink *inlink, AVFrame *insamples
{
AVFilterContext *ctx = inlink->dst;
ShowSpectrumContext *s = ctx->priv;
int ret;
void *ptr;
ret = av_audio_fifo_write(s->fifo, (void **)insamples->extended_data, insamples->nb_samples);
av_frame_free(&insamples);
return ret;
if (s->nb_frames + 1ULL > s->frames_size / sizeof(*(s->frames))) {
ptr = av_fast_realloc(s->frames, &s->frames_size, s->frames_size * 2);
if (!ptr)
return AVERROR(ENOMEM);
s->frames = ptr;
}
s->frames[s->nb_frames] = insamples;
s->samples += insamples->nb_samples;
s->nb_frames++;
return 0;
}
static const AVFilterPad showspectrumpic_inputs[] = {