[FFmpeg-devel] [PATCH][BULK][again] new multimedia filter avf_showcqt.c
Muhammad Faiz
mfcc64 at gmail.com
Sun Jun 1 16:29:14 CEST 2014
On Fri, May 30, 2014 at 6:09 PM, Michael Niedermayer <michaelni at gmx.at>
wrote:
>
> you can try libavutil/qsort.h which would avoid some overhead of
> qsort()
> also the abs() does not need to be done in the compare function,
> it could be done outside
>
> or if its still slow, this can be done using a radix sort
> (this would require the floats to be read as integers)
>
>
> [...]
> > + for (x = 0; x < VIDEO_WIDTH; x += VIDEO_WIDTH/10)
> > + {
> > + int u;
>
> > + const char *str = "EF G A BC D ";
>
> static const
>
> and SIMD optimizations of the most speed critical part would make
> sense if you know asm
>
> Also you probably want to add yourself to the MAINTAINERs file for
> showcqt
>
SIMD optimizations, not yet.
Thank's.
-------------- next part --------------
diff --git a/Changelog b/Changelog
index 3d416c4..97a7d1c 100644
--- a/Changelog
+++ b/Changelog
@@ -26,6 +26,7 @@ version <next>:
- native Opus decoder
- display matrix export and rotation api
- WebVTT encoder
+- showcqt multimedia filter
version 2.2:
diff --git a/doc/filters.texi b/doc/filters.texi
index e004c44..9dc97d7 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -10133,6 +10133,76 @@ settb=AVTB
@end example
@end itemize
+ at section showcqt
+Convert input audio to a video output (at full HD resolution), representing
+frequency spectrum logarithmically (using constant Q transform with
+Brown-Puckette algorithm), with musical tone scale, from E0 to D#10 (10 octaves).
+
+The filter accepts the following options:
+
+ at table @option
+ at item volume
+Specify the transform volume (multiplier). Acceptable value is [1.0, 100.0].
+Default value is @code{16.0}.
+
+ at item timeclamp
+Specify the transform timeclamp. At low frequency, there is trade-off between
+accuracy in time domain and frequency domain. If timeclamp is lower,
+event in time domain is represented more accurately (such as fast bass drum),
+otherwise event in frequency domain is represented more accurately
+(such as bass guitar). Acceptable value is [0.1, 1.0]. Default value is @code{0.17}.
+
+ at item coeffclamp
+Specify the transform coeffclamp. If coeffclamp is lower, transform is
+more accurate, otherwise transform is faster. Acceptable value is [0.1, 10.0].
+Default value is @code{1.0}.
+
+ at item gamma
+Specify gamma. Lower gamma makes the spectrum more contrast, higher gamma
+makes the spectrum having more range. Acceptable value is [1.0, 7.0].
+Default value is @code{3.0}.
+
+ at item fps
+Specify video fps. Default value is @code{25}.
+
+ at item count
+Specify number of transform per frame, so there are fps*count transforms
+per second. Note tha audio data rate must be divisible by fps*count.
+Default value is @code{6}.
+
+ at end table
+
+ at subsection Examples
+
+ at itemize
+ at item
+Playing audio while showing the spectrum:
+ at example
+ffplay -f lavfi 'amovie=a.mp3, asplit [a][out1]; [a] showcqt [out0]'
+ at end example
+
+ at item
+Same as above, but with frame rate 30 fps:
+ at example
+ffplay -f lavfi 'amovie=a.mp3, asplit [a][out1]; [a] showcqt=fps=30:count=5 [out0]'
+ at end example
+
+ at item
+A1 and its harmonics: A1, A2, (near)E3, A3:
+ at example
+ffplay -f lavfi 'aevalsrc=0.1*sin(2*PI*55*t)+0.1*sin(4*PI*55*t)+0.1*sin(6*PI*55*t)+0.1*sin(8*PI*55*t),
+ asplit[a][out1]; [a] showcqt [out0]'
+ at end example
+
+ at item
+Same as above, but with more accuracy in frequency domain (and slower):
+ at example
+ffplay -f lavfi 'aevalsrc=0.1*sin(2*PI*55*t)+0.1*sin(4*PI*55*t)+0.1*sin(6*PI*55*t)+0.1*sin(8*PI*55*t),
+ asplit[a][out1]; [a] showcqt=timeclamp=0.5 [out0]'
+ at end example
+
+ at end itemize
+
@section showspectrum
Convert input audio to a video output, representing the audio frequency
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index f981dfa..8ba0312 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -224,6 +224,7 @@ OBJS-$(CONFIG_MP_FILTER) += libmpcodecs/vf_uspp.o
# multimedia filters
OBJS-$(CONFIG_AVECTORSCOPE_FILTER) += avf_avectorscope.o
OBJS-$(CONFIG_CONCAT_FILTER) += avf_concat.o
+OBJS-$(CONFIG_SHOWCQT_FILTER) += avf_showcqt.o
OBJS-$(CONFIG_SHOWSPECTRUM_FILTER) += avf_showspectrum.o
OBJS-$(CONFIG_SHOWWAVES_FILTER) += avf_showwaves.o
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index 22d643d..55d505a 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -231,6 +231,7 @@ void avfilter_register_all(void)
/* multimedia filters */
REGISTER_FILTER(AVECTORSCOPE, avectorscope, avf);
REGISTER_FILTER(CONCAT, concat, avf);
+ REGISTER_FILTER(SHOWCQT, showcqt, avf);
REGISTER_FILTER(SHOWSPECTRUM, showspectrum, avf);
REGISTER_FILTER(SHOWWAVES, showwaves, avf);
diff --git b/libavfilter/avf_showcqt.c b/libavfilter/avf_showcqt.c
new file mode 100644
index 0000000..eaa632d
--- /dev/null
+++ b/libavfilter/avf_showcqt.c
@@ -0,0 +1,585 @@
+/*
+ * Copyright (c) 2014 Muhammad Faiz <mfcc64 at gmail.com>
+ *
+ * This file is part of FFmpeg.
+ *
+ * FFmpeg is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * FFmpeg is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with FFmpeg; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+#include "libavcodec/avfft.h"
+#include "libavutil/avassert.h"
+#include "libavutil/channel_layout.h"
+#include "libavutil/opt.h"
+#include "libavutil/xga_font_data.h"
+#include "libavutil/qsort.h"
+#include "libavutil/time.h"
+#include "avfilter.h"
+#include "internal.h"
+
+#include <math.h>
+#include <stdlib.h>
+
+/* this filter is designed to do 16 bins/semitones constant Q transform with Brown-Puckette algorithm
+ * start from E0 to D#10 (10 octaves)
+ * so there are 16 bins/semitones * 12 semitones/octaves * 10 octaves = 1920 bins
+ * match with full HD resolution */
+
+#define VIDEO_WIDTH 1920
+#define VIDEO_HEIGHT 1080
+#define FONT_HEIGHT 32
+#define SPECTOGRAM_HEIGHT ((VIDEO_HEIGHT-FONT_HEIGHT)/2)
+#define SPECTOGRAM_START (VIDEO_HEIGHT-SPECTOGRAM_HEIGHT)
+#define BASE_FREQ 20.051392800492
+#define COEFF_CLAMP 1.0e-4
+
+typedef struct {
+ FFTSample value;
+ int index;
+} SparseCoeff;
+
+static inline int qsort_sparsecoeff(const SparseCoeff *a, const SparseCoeff *b)
+{
+ if (fabsf(a->value) >= fabsf(b->value))
+ return 1;
+ else
+ return -1;
+}
+
+typedef struct {
+ const AVClass *class;
+ AVFrame *outpicref;
+ FFTContext *fft_context;
+ FFTComplex *fft_data;
+ FFTComplex *fft_result_left;
+ FFTComplex *fft_result_right;
+ SparseCoeff *coeff_sort;
+ SparseCoeff *coeffs[VIDEO_WIDTH];
+ int coeffs_len[VIDEO_WIDTH];
+ uint8_t font_color[VIDEO_WIDTH];
+ uint8_t spectogram[SPECTOGRAM_HEIGHT][VIDEO_WIDTH][3];
+ int64_t frame_count;
+ int spectogram_count;
+ int spectogram_index;
+ int fft_bits;
+ int req_fullfilled;
+ int remaining_fill;
+ double volume;
+ double timeclamp; /* lower timeclamp, time-accurate, higher timeclamp, freq-accurate (at low freq)*/
+ float coeffclamp; /* lower coeffclamp, more precise, higher coeffclamp, faster */
+ float gamma; /* lower gamma, more contrast, higher gamma, more range */
+ int fps; /* the required fps is so strict, so it's enough to be int, but 24000/1001 etc cannot be encoded */
+ int count; /* fps * count = transform rate */
+} ShowCQTContext;
+
+#define OFFSET(x) offsetof(ShowCQTContext, x)
+#define FLAGS AV_OPT_FLAG_FILTERING_PARAM|AV_OPT_FLAG_VIDEO_PARAM
+
+static const AVOption showcqt_options[] = {
+ { "volume", "set volume", OFFSET(volume), AV_OPT_TYPE_DOUBLE, { .dbl = 16 }, 0.1, 100, FLAGS },
+ { "timeclamp", "set timeclamp", OFFSET(timeclamp), AV_OPT_TYPE_DOUBLE, { .dbl = 0.17 }, 0.1, 1.0, FLAGS },
+ { "coeffclamp", "set coeffclamp", OFFSET(coeffclamp), AV_OPT_TYPE_FLOAT, { .dbl = 1 }, 0.1, 10, FLAGS },
+ { "gamma", "set gamma", OFFSET(gamma), AV_OPT_TYPE_FLOAT, { .dbl = 3 }, 1, 7, FLAGS },
+ { "fps", "set video fps", OFFSET(fps), AV_OPT_TYPE_INT, { .i64 = 25 }, 10, 100, FLAGS },
+ { "count", "set number of transform per frame", OFFSET(count), AV_OPT_TYPE_INT, { .i64 = 6 }, 1, 30, FLAGS },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(showcqt);
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ int k;
+ ShowCQTContext *s = ctx->priv;
+ av_fft_end(s->fft_context);
+ s->fft_context = NULL;
+ for (k = 0; k < VIDEO_WIDTH; k++)
+ av_freep(&s->coeffs[k]);
+ av_freep(&s->fft_data);
+ av_freep(&s->fft_result_left);
+ av_freep(&s->fft_result_right);
+ av_freep(&s->coeff_sort);
+ av_frame_free(&s->outpicref);
+}
+
+static int query_formats(AVFilterContext *ctx)
+{
+ AVFilterFormats *formats = NULL;
+ AVFilterChannelLayouts *layouts = NULL;
+ AVFilterLink *inlink = ctx->inputs[0];
+ AVFilterLink *outlink = ctx->outputs[0];
+ static const enum AVSampleFormat sample_fmts[] = { AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_NONE };
+ static const enum AVPixelFormat pix_fmts[] = { AV_PIX_FMT_RGB24, AV_PIX_FMT_NONE };
+ static const int64_t channel_layouts[] = { AV_CH_LAYOUT_STEREO, AV_CH_LAYOUT_STEREO_DOWNMIX, -1 };
+ static const int samplerates[] = { 44100, 48000, -1 };
+
+ /* set input audio formats */
+ formats = ff_make_format_list(sample_fmts);
+ if (!formats)
+ return AVERROR(ENOMEM);
+ ff_formats_ref(formats, &inlink->out_formats);
+
+ layouts = avfilter_make_format64_list(channel_layouts);
+ if (!layouts)
+ return AVERROR(ENOMEM);
+ ff_channel_layouts_ref(layouts, &inlink->out_channel_layouts);
+
+ formats = ff_make_format_list(samplerates);
+ if (!formats)
+ return AVERROR(ENOMEM);
+ ff_formats_ref(formats, &inlink->out_samplerates);
+
+ /* set output video format */
+ formats = ff_make_format_list(pix_fmts);
+ if (!formats)
+ return AVERROR(ENOMEM);
+ ff_formats_ref(formats, &outlink->in_formats);
+
+ return 0;
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ AVFilterLink *inlink = ctx->inputs[0];
+ ShowCQTContext *s = ctx->priv;
+ int fft_len, k, x, y;
+ int num_coeffs = 0;
+ int rate = inlink->sample_rate;
+ double max_len = rate * (double) s->timeclamp;
+ int64_t start_time, end_time;
+ s->fft_bits = ceil(log2(max_len));
+ fft_len = 1 << s->fft_bits;
+
+ if (rate % (s->fps * s->count))
+ {
+ av_log(ctx, AV_LOG_ERROR, "Rate (%u) is not divisible by fps*count (%u*%u)\n", rate, s->fps, s->count);
+ return AVERROR(EINVAL);
+ }
+
+ s->fft_data = av_malloc_array(fft_len, sizeof(*s->fft_data));
+ s->coeff_sort = av_malloc_array(fft_len, sizeof(*s->coeff_sort));
+ s->fft_result_left = av_malloc_array(fft_len, sizeof(*s->fft_result_left));
+ s->fft_result_right = av_malloc_array(fft_len, sizeof(*s->fft_result_right));
+ s->fft_context = av_fft_init(s->fft_bits, 0);
+
+ if (!s->fft_data || !s->coeff_sort || !s->fft_result_left || !s->fft_result_right || !s->fft_context)
+ return AVERROR(ENOMEM);
+
+ /* initializing font */
+ for (x = 0; x < VIDEO_WIDTH; x++)
+ {
+ if (x >= (12*3+8)*16 && x < (12*4+8)*16)
+ {
+ float fx = (x-(12*3+8)*16) * (1.0f/192.0f);
+ float sv = sinf(M_PI*fx);
+ s->font_color[x] = sv*sv*255.0f + 0.5f;
+ }
+ else
+ s->font_color[x] = 0;
+ }
+
+ av_log(ctx, AV_LOG_INFO, "Calculating spectral kernel, please wait\n");
+ start_time = av_gettime_relative();
+ for (k = 0; k < VIDEO_WIDTH; k++)
+ {
+ int hlen = fft_len >> 1;
+ float total = 0;
+ float partial = 0;
+ double freq = BASE_FREQ * exp2(k * (1.0/192.0));
+ double tlen = rate * (24.0 * 16.0) /freq;
+ /* a window function from Albert H. Nuttall,
+ * "Some Windows with Very Good Sidelobe Behavior"
+ * -93.32 dB peak sidelobe and 18 dB/octave asymptotic decay
+ * coefficient normalized to a0 = 1 */
+ double a0 = 0.355768;
+ double a1 = 0.487396/a0;
+ double a2 = 0.144232/a0;
+ double a3 = 0.012604/a0;
+ double sv_step, cv_step, sv, cv;
+ double sw_step, cw_step, sw, cw, w;
+
+ tlen = tlen * max_len / (tlen + max_len);
+ s->fft_data[0].re = 0;
+ s->fft_data[0].im = 0;
+ s->fft_data[hlen].re = (1.0 + a1 + a2 + a3) * (1.0/tlen) * s->volume * (1.0/fft_len);
+ s->fft_data[hlen].im = 0;
+ sv_step = sv = sin(2.0*M_PI*freq*(1.0/rate));
+ cv_step = cv = cos(2.0*M_PI*freq*(1.0/rate));
+ /* also optimizing window func */
+ sw_step = sw = sin(2.0*M_PI*(1.0/tlen));
+ cw_step = cw = cos(2.0*M_PI*(1.0/tlen));
+ for (x = 1; x < 0.5 * tlen; x++)
+ {
+ double cv_tmp, cw_tmp;
+ double cw2, cw3, sw2;
+
+ cw2 = cw * cw - sw * sw;
+ sw2 = cw * sw + sw * cw;
+ cw3 = cw * cw2 - sw * sw2;
+ w = (1.0 + a1 * cw + a2 * cw2 + a3 * cw3) * (1.0/tlen) * s->volume * (1.0/fft_len);
+ s->fft_data[hlen + x].re = w * cv;
+ s->fft_data[hlen + x].im = w * sv;
+ s->fft_data[hlen - x].re = s->fft_data[hlen + x].re;
+ s->fft_data[hlen - x].im = -s->fft_data[hlen + x].im;
+
+ cv_tmp = cv * cv_step - sv * sv_step;
+ sv = sv * cv_step + cv * sv_step;
+ cv = cv_tmp;
+ cw_tmp = cw * cw_step - sw * sw_step;
+ sw = sw * cw_step + cw * sw_step;
+ cw = cw_tmp;
+ }
+ for (; x < hlen; x++)
+ {
+ s->fft_data[hlen + x].re = 0;
+ s->fft_data[hlen + x].im = 0;
+ s->fft_data[hlen - x].re = 0;
+ s->fft_data[hlen - x].im = 0;
+ }
+ av_fft_permute(s->fft_context, s->fft_data);
+ av_fft_calc(s->fft_context, s->fft_data);
+
+ for (x = 0; x < fft_len; x++)
+ {
+ s->coeff_sort[x].index = x;
+ s->coeff_sort[x].value = s->fft_data[x].re;
+ }
+
+ AV_QSORT(s->coeff_sort, fft_len, SparseCoeff, qsort_sparsecoeff);
+ for (x = 0; x < fft_len; x++)
+ total += fabsf(s->coeff_sort[x].value);
+
+ for (x = 0; x < fft_len; x++)
+ {
+ partial += fabsf(s->coeff_sort[x].value);
+ if (partial > (total * s->coeffclamp * COEFF_CLAMP))
+ {
+ s->coeffs_len[k] = fft_len - x;
+ num_coeffs += s->coeffs_len[k];
+ s->coeffs[k] = av_malloc_array(s->coeffs_len[k], sizeof(*s->coeffs[k]));
+ if (!s->coeffs[k])
+ return AVERROR(ENOMEM);
+ for (y = 0; y < s->coeffs_len[k]; y++)
+ s->coeffs[k][y] = s->coeff_sort[x+y];
+ break;
+ }
+ }
+ }
+ end_time = av_gettime_relative();
+ av_log(ctx, AV_LOG_INFO, "Elapsed time %.6f s (fft_len=%u, num_coeffs=%u)\n", 1e-6 * (end_time-start_time), fft_len, num_coeffs);
+
+ outlink->w = VIDEO_WIDTH;
+ outlink->h = VIDEO_HEIGHT;
+
+ s->req_fullfilled = 0;
+ s->spectogram_index = 0;
+ s->frame_count = 0;
+ s->spectogram_count = 0;
+ s->remaining_fill = fft_len >> 1;
+ memset(s->spectogram, 0, VIDEO_WIDTH * SPECTOGRAM_HEIGHT * 3);
+ memset(s->fft_data, 0, fft_len * sizeof(*s->fft_data));
+
+ s->outpicref = ff_get_video_buffer(outlink, outlink->w, outlink->h);
+ if (!s->outpicref)
+ return AVERROR(ENOMEM);
+
+ outlink->sample_aspect_ratio = av_make_q(1, 1);
+ outlink->time_base = av_make_q(1, s->fps);
+ outlink->frame_rate = av_make_q(s->fps, 1);
+ return 0;
+}
+
+static int plot_cqt(AVFilterLink *inlink)
+{
+ AVFilterContext *ctx = inlink->dst;
+ ShowCQTContext *s = ctx->priv;
+ AVFilterLink *outlink = ctx->outputs[0];
+ int fft_len = 1 << s->fft_bits;
+ FFTSample result[VIDEO_WIDTH][4];
+ int x, y, ret = 0;
+
+ /* real part contains left samples, imaginary part contains right samples */
+ memcpy(s->fft_result_left, s->fft_data, fft_len * sizeof(*s->fft_data));
+ av_fft_permute(s->fft_context, s->fft_result_left);
+ av_fft_calc(s->fft_context, s->fft_result_left);
+
+ /* separate left and right, (and multiply by 2.0) */
+ s->fft_result_right[0].re = 2.0f * s->fft_result_left[0].im;
+ s->fft_result_right[0].im = 0;
+ s->fft_result_left[0].re = 2.0f * s->fft_result_left[0].re;
+ s->fft_result_left[0].im = 0;
+ for (x = 1; x <= (fft_len >> 1); x++)
+ {
+ FFTSample tmpy = s->fft_result_left[fft_len-x].im - s->fft_result_left[x].im;
+
+ s->fft_result_right[x].re = s->fft_result_left[x].im + s->fft_result_left[fft_len-x].im;
+ s->fft_result_right[x].im = s->fft_result_left[x].re - s->fft_result_left[fft_len-x].re;
+ s->fft_result_right[fft_len-x].re = s->fft_result_right[x].re;
+ s->fft_result_right[fft_len-x].im = -s->fft_result_right[x].im;
+
+ s->fft_result_left[x].re = s->fft_result_left[x].re + s->fft_result_left[fft_len-x].re;
+ s->fft_result_left[x].im = tmpy;
+ s->fft_result_left[fft_len-x].re = s->fft_result_left[x].re;
+ s->fft_result_left[fft_len-x].im = -s->fft_result_left[x].im;
+ }
+
+ /* calculating cqt */
+ for (x = 0; x < VIDEO_WIDTH; x++)
+ {
+ int u;
+ float g = 1.0f / s->gamma;
+ FFTComplex l = {0,0};
+ FFTComplex r = {0,0};
+
+ for (u = 0; u < s->coeffs_len[x]; u++)
+ {
+ FFTSample value = s->coeffs[x][u].value;
+ int index = s->coeffs[x][u].index;
+ l.re += value * s->fft_result_left[index].re;
+ l.im += value * s->fft_result_left[index].im;
+ r.re += value * s->fft_result_right[index].re;
+ r.im += value * s->fft_result_right[index].im;
+ }
+ /* result is power, not amplitude */
+ result[x][0] = l.re * l.re + l.im * l.im;
+ result[x][2] = r.re * r.re + r.im * r.im;
+ result[x][1] = 0.5f * (result[x][0] + result[x][2]);
+ result[x][3] = result[x][1];
+ result[x][0] = 255.0f * powf(fminf(1.0f,result[x][0]), g);
+ result[x][1] = 255.0f * powf(fminf(1.0f,result[x][1]), g);
+ result[x][2] = 255.0f * powf(fminf(1.0f,result[x][2]), g);
+ }
+
+ for (x = 0; x < VIDEO_WIDTH; x++)
+ {
+ s->spectogram[s->spectogram_index][x][0] = result[x][0] + 0.5f;
+ s->spectogram[s->spectogram_index][x][1] = result[x][1] + 0.5f;
+ s->spectogram[s->spectogram_index][x][2] = result[x][2] + 0.5f;
+ }
+
+ /* drawing */
+ if (!s->spectogram_count)
+ {
+ uint8_t *data = (uint8_t*) s->outpicref->data[0];
+ int linesize = s->outpicref->linesize[0];
+ float rcp_result[VIDEO_WIDTH];
+
+ for (x = 0; x < VIDEO_WIDTH; x++)
+ rcp_result[x] = 1.0f / (result[x][3]+0.0001f);
+
+ /* drawing bar */
+ for (y = 0; y < SPECTOGRAM_HEIGHT; y++)
+ {
+ float height = (SPECTOGRAM_HEIGHT - y) * (1.0f/SPECTOGRAM_HEIGHT);
+ uint8_t *lineptr = data + y * linesize;
+ for (x = 0; x < VIDEO_WIDTH; x++)
+ {
+ float mul;
+ if (result[x][3] <= height)
+ {
+ *lineptr++ = 0;
+ *lineptr++ = 0;
+ *lineptr++ = 0;
+ }
+ else
+ {
+ mul = (result[x][3] - height) * rcp_result[x];
+ *lineptr++ = mul * result[x][0] + 0.5f;
+ *lineptr++ = mul * result[x][1] + 0.5f;
+ *lineptr++ = mul * result[x][2] + 0.5f;
+ }
+ }
+
+ }
+
+ /* drawing font */
+ for (y = 0; y < FONT_HEIGHT; y++)
+ {
+ uint8_t *lineptr = data + (SPECTOGRAM_HEIGHT + y) * linesize;
+ memcpy(lineptr, s->spectogram[s->spectogram_index], VIDEO_WIDTH*3);
+ }
+ for (x = 0; x < VIDEO_WIDTH; x += VIDEO_WIDTH/10)
+ {
+ int u;
+ static const char str[] = "EF G A BC D ";
+ uint8_t *startptr = data + SPECTOGRAM_HEIGHT * linesize + x * 3;
+ for (u = 0; str[u]; u++)
+ {
+ int v;
+ for (v = 0; v < 16; v++)
+ {
+ uint8_t *p = startptr + 2 * v * linesize + 16 * 3 * u;
+ int ux = x + 16 * u;
+ int mask;
+ for (mask = 0x80; mask; mask >>= 1)
+ {
+ if (mask & avpriv_vga16_font[str[u] * 16 + v])
+ {
+ p[0] = p[linesize] = 255 - s->font_color[ux];
+ p[1] = p[linesize+1] = 0;
+ p[2] = p[linesize+2] = s->font_color[ux];
+ p[3] = p[linesize+3] = 255 - s->font_color[ux+1];
+ p[4] = p[linesize+4] = 0;
+ p[5] = p[linesize+5] = s->font_color[ux+1];
+ }
+ p += 6;
+ ux += 2;
+ }
+ }
+ }
+
+ }
+
+ /* drawing spectogram/sonogram */
+ if (linesize == VIDEO_WIDTH * 3)
+ {
+ int total_length = VIDEO_WIDTH * SPECTOGRAM_HEIGHT * 3;
+ int back_length = VIDEO_WIDTH * s->spectogram_index * 3;
+ data += SPECTOGRAM_START * VIDEO_WIDTH * 3;
+ memcpy(data, s->spectogram[s->spectogram_index], total_length - back_length);
+ data += total_length - back_length;
+ if(back_length)
+ memcpy(data, s->spectogram[0], back_length);
+ }
+ else
+ {
+ for (y = 0; y < SPECTOGRAM_HEIGHT; y++)
+ memcpy(data + (SPECTOGRAM_START + y) * linesize, s->spectogram[(s->spectogram_index + y) % SPECTOGRAM_HEIGHT], VIDEO_WIDTH * 3);
+ }
+
+ s->outpicref->pts = s->frame_count;
+ ret = ff_filter_frame(outlink, av_frame_clone(s->outpicref));
+ s->req_fullfilled = 1;
+ s->frame_count++;
+ }
+ s->spectogram_count = (s->spectogram_count + 1) % s->count;
+ s->spectogram_index = (s->spectogram_index + SPECTOGRAM_HEIGHT - 1) % SPECTOGRAM_HEIGHT;
+ return ret;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *insamples)
+{
+ AVFilterContext *ctx = inlink->dst;
+ ShowCQTContext *s = ctx->priv;
+ int step = inlink->sample_rate / (s->fps * s->count);
+ int fft_len = 1 << s->fft_bits;
+ int remaining;
+ float *audio_data;
+
+ if (!insamples)
+ {
+ while (s->remaining_fill < (fft_len >> 1))
+ {
+ int ret, x;
+ memset(&s->fft_data[fft_len - s->remaining_fill], 0, sizeof(*s->fft_data) * s->remaining_fill);
+ ret = plot_cqt(inlink);
+ if (ret < 0)
+ return ret;
+ for (x = 0; x < (fft_len-step); x++)
+ s->fft_data[x] = s->fft_data[x+step];
+ s->remaining_fill += step;
+ }
+ return AVERROR(EOF);
+ }
+
+ remaining = insamples->nb_samples;
+ audio_data = (float*) insamples->data[0];
+
+ while (remaining)
+ {
+ if (remaining >= s->remaining_fill)
+ {
+ int i = insamples->nb_samples - remaining;
+ int j = fft_len - s->remaining_fill;
+ int m, ret;
+ for (m = 0; m < s->remaining_fill; m++)
+ {
+ s->fft_data[j+m].re = audio_data[2*(i+m)];
+ s->fft_data[j+m].im = audio_data[2*(i+m)+1];
+ }
+ ret = plot_cqt(inlink);
+ if (ret < 0)
+ {
+ av_frame_free(&insamples);
+ return ret;
+ }
+ remaining -= s->remaining_fill;
+ for (m = 0; m < fft_len-step; m++)
+ s->fft_data[m] = s->fft_data[m+step];
+ s->remaining_fill = step;
+ }
+ else
+ {
+ int i = insamples->nb_samples - remaining;
+ int j = fft_len - s->remaining_fill;
+ int m;
+ for (m = 0; m < remaining; m++)
+ {
+ s->fft_data[m+j].re = audio_data[2*(i+m)];
+ s->fft_data[m+j].im = audio_data[2*(i+m)+1];
+ }
+ s->remaining_fill -= remaining;
+ remaining = 0;
+ }
+ }
+ av_frame_free(&insamples);
+ return 0;
+}
+
+static int request_frame(AVFilterLink *outlink)
+{
+ ShowCQTContext *s = outlink->src->priv;
+ AVFilterLink *inlink = outlink->src->inputs[0];
+ int ret;
+
+ s->req_fullfilled = 0;
+ do {
+ ret = ff_request_frame(inlink);
+ } while (!s->req_fullfilled && ret >= 0);
+
+ if (ret == AVERROR_EOF && s->outpicref)
+ filter_frame(inlink, NULL);
+ return ret;
+}
+
+static const AVFilterPad showcqt_inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .filter_frame = filter_frame,
+ },
+ { NULL }
+};
+
+static const AVFilterPad showcqt_outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_VIDEO,
+ .config_props = config_output,
+ .request_frame = request_frame,
+ },
+ { NULL }
+};
+
+AVFilter ff_avf_showcqt = {
+ .name = "showcqt",
+ .description = NULL_IF_CONFIG_SMALL("Convert input audio to a CQT (Constant Q Transform) spectrum video output."),
+ .uninit = uninit,
+ .query_formats = query_formats,
+ .priv_size = sizeof(ShowCQTContext),
+ .inputs = showcqt_inputs,
+ .outputs = showcqt_outputs,
+ .priv_class = &showcqt_class,
+};
diff --git a/libavfilter/version.h b/libavfilter/version.h
index 7a6321e..d7acf72 100644
--- a/libavfilter/version.h
+++ b/libavfilter/version.h
@@ -30,7 +30,7 @@
#include "libavutil/version.h"
#define LIBAVFILTER_VERSION_MAJOR 4
-#define LIBAVFILTER_VERSION_MINOR 5
+#define LIBAVFILTER_VERSION_MINOR 6
#define LIBAVFILTER_VERSION_MICRO 100
#define LIBAVFILTER_VERSION_INT AV_VERSION_INT(LIBAVFILTER_VERSION_MAJOR, \
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