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-rw-r--r--libavfilter/af_afade.c669
1 files changed, 669 insertions, 0 deletions
diff --git a/libavfilter/af_afade.c b/libavfilter/af_afade.c
new file mode 100644
index 0000000000..9acadc51c5
--- /dev/null
+++ b/libavfilter/af_afade.c
@@ -0,0 +1,669 @@
+/*
+ * Copyright (c) 2013-2015 Paul B Mahol
+ *
+ * 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
+ */
+
+/**
+ * @file
+ * fade audio filter
+ */
+
+#include "libavutil/audio_fifo.h"
+#include "libavutil/opt.h"
+#include "audio.h"
+#include "avfilter.h"
+#include "internal.h"
+
+typedef struct {
+ const AVClass *class;
+ int type;
+ int curve, curve2;
+ int nb_samples;
+ int64_t start_sample;
+ int64_t duration;
+ int64_t start_time;
+ int overlap;
+ int cf0_eof;
+ int crossfade_is_over;
+ AVAudioFifo *fifo[2];
+ int64_t pts;
+
+ void (*fade_samples)(uint8_t **dst, uint8_t * const *src,
+ int nb_samples, int channels, int direction,
+ int64_t start, int range, int curve);
+ void (*crossfade_samples)(uint8_t **dst, uint8_t * const *cf0,
+ uint8_t * const *cf1,
+ int nb_samples, int channels,
+ int curve0, int curve1);
+} AudioFadeContext;
+
+enum CurveType { TRI, QSIN, ESIN, HSIN, LOG, IPAR, QUA, CUB, SQU, CBR, PAR, EXP, IQSIN, IHSIN, DESE, DESI, NB_CURVES };
+
+#define OFFSET(x) offsetof(AudioFadeContext, x)
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM|AV_OPT_FLAG_FILTERING_PARAM
+
+static int query_formats(AVFilterContext *ctx)
+{
+ AVFilterFormats *formats;
+ AVFilterChannelLayouts *layouts;
+ static const enum AVSampleFormat sample_fmts[] = {
+ AV_SAMPLE_FMT_S16, AV_SAMPLE_FMT_S16P,
+ AV_SAMPLE_FMT_S32, AV_SAMPLE_FMT_S32P,
+ AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_FLTP,
+ AV_SAMPLE_FMT_DBL, AV_SAMPLE_FMT_DBLP,
+ AV_SAMPLE_FMT_NONE
+ };
+ int ret;
+
+ layouts = ff_all_channel_counts();
+ if (!layouts)
+ return AVERROR(ENOMEM);
+ ret = ff_set_common_channel_layouts(ctx, layouts);
+ if (ret < 0)
+ return ret;
+
+ formats = ff_make_format_list(sample_fmts);
+ if (!formats)
+ return AVERROR(ENOMEM);
+ ret = ff_set_common_formats(ctx, formats);
+ if (ret < 0)
+ return ret;
+
+ formats = ff_all_samplerates();
+ if (!formats)
+ return AVERROR(ENOMEM);
+ return ff_set_common_samplerates(ctx, formats);
+}
+
+static double fade_gain(int curve, int64_t index, int range)
+{
+#define CUBE(a) ((a)*(a)*(a))
+ double gain;
+
+ gain = av_clipd(1.0 * index / range, 0, 1.0);
+
+ switch (curve) {
+ case QSIN:
+ gain = sin(gain * M_PI / 2.0);
+ break;
+ case IQSIN:
+ /* 0.6... = 2 / M_PI */
+ gain = 0.6366197723675814 * asin(gain);
+ break;
+ case ESIN:
+ gain = 1.0 - cos(M_PI / 4.0 * (CUBE(2.0*gain - 1) + 1));
+ break;
+ case HSIN:
+ gain = (1.0 - cos(gain * M_PI)) / 2.0;
+ break;
+ case IHSIN:
+ /* 0.3... = 1 / M_PI */
+ gain = 0.3183098861837907 * acos(1 - 2 * gain);
+ break;
+ case EXP:
+ /* -11.5... = 5*ln(0.1) */
+ gain = exp(-11.512925464970227 * (1 - gain));
+ break;
+ case LOG:
+ gain = av_clipd(1 + 0.2 * log10(gain), 0, 1.0);
+ break;
+ case PAR:
+ gain = 1 - sqrt(1 - gain);
+ break;
+ case IPAR:
+ gain = (1 - (1 - gain) * (1 - gain));
+ break;
+ case QUA:
+ gain *= gain;
+ break;
+ case CUB:
+ gain = CUBE(gain);
+ break;
+ case SQU:
+ gain = sqrt(gain);
+ break;
+ case CBR:
+ gain = cbrt(gain);
+ break;
+ case DESE:
+ gain = gain <= 0.5 ? cbrt(2 * gain) / 2: 1 - cbrt(2 * (1 - gain)) / 2;
+ break;
+ case DESI:
+ gain = gain <= 0.5 ? CUBE(2 * gain) / 2: 1 - CUBE(2 * (1 - gain)) / 2;
+ break;
+ }
+
+ return gain;
+}
+
+#define FADE_PLANAR(name, type) \
+static void fade_samples_## name ##p(uint8_t **dst, uint8_t * const *src, \
+ int nb_samples, int channels, int dir, \
+ int64_t start, int range, int curve) \
+{ \
+ int i, c; \
+ \
+ for (i = 0; i < nb_samples; i++) { \
+ double gain = fade_gain(curve, start + i * dir, range); \
+ for (c = 0; c < channels; c++) { \
+ type *d = (type *)dst[c]; \
+ const type *s = (type *)src[c]; \
+ \
+ d[i] = s[i] * gain; \
+ } \
+ } \
+}
+
+#define FADE(name, type) \
+static void fade_samples_## name (uint8_t **dst, uint8_t * const *src, \
+ int nb_samples, int channels, int dir, \
+ int64_t start, int range, int curve) \
+{ \
+ type *d = (type *)dst[0]; \
+ const type *s = (type *)src[0]; \
+ int i, c, k = 0; \
+ \
+ for (i = 0; i < nb_samples; i++) { \
+ double gain = fade_gain(curve, start + i * dir, range); \
+ for (c = 0; c < channels; c++, k++) \
+ d[k] = s[k] * gain; \
+ } \
+}
+
+FADE_PLANAR(dbl, double)
+FADE_PLANAR(flt, float)
+FADE_PLANAR(s16, int16_t)
+FADE_PLANAR(s32, int32_t)
+
+FADE(dbl, double)
+FADE(flt, float)
+FADE(s16, int16_t)
+FADE(s32, int32_t)
+
+static int config_output(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ AudioFadeContext *s = ctx->priv;
+
+ switch (outlink->format) {
+ case AV_SAMPLE_FMT_DBL: s->fade_samples = fade_samples_dbl; break;
+ case AV_SAMPLE_FMT_DBLP: s->fade_samples = fade_samples_dblp; break;
+ case AV_SAMPLE_FMT_FLT: s->fade_samples = fade_samples_flt; break;
+ case AV_SAMPLE_FMT_FLTP: s->fade_samples = fade_samples_fltp; break;
+ case AV_SAMPLE_FMT_S16: s->fade_samples = fade_samples_s16; break;
+ case AV_SAMPLE_FMT_S16P: s->fade_samples = fade_samples_s16p; break;
+ case AV_SAMPLE_FMT_S32: s->fade_samples = fade_samples_s32; break;
+ case AV_SAMPLE_FMT_S32P: s->fade_samples = fade_samples_s32p; break;
+ }
+
+ if (s->duration)
+ s->nb_samples = av_rescale(s->duration, outlink->sample_rate, AV_TIME_BASE);
+ if (s->start_time)
+ s->start_sample = av_rescale(s->start_time, outlink->sample_rate, AV_TIME_BASE);
+
+ return 0;
+}
+
+#if CONFIG_AFADE_FILTER
+
+static const AVOption afade_options[] = {
+ { "type", "set the fade direction", OFFSET(type), AV_OPT_TYPE_INT, {.i64 = 0 }, 0, 1, FLAGS, "type" },
+ { "t", "set the fade direction", OFFSET(type), AV_OPT_TYPE_INT, {.i64 = 0 }, 0, 1, FLAGS, "type" },
+ { "in", "fade-in", 0, AV_OPT_TYPE_CONST, {.i64 = 0 }, 0, 0, FLAGS, "type" },
+ { "out", "fade-out", 0, AV_OPT_TYPE_CONST, {.i64 = 1 }, 0, 0, FLAGS, "type" },
+ { "start_sample", "set number of first sample to start fading", OFFSET(start_sample), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, FLAGS },
+ { "ss", "set number of first sample to start fading", OFFSET(start_sample), AV_OPT_TYPE_INT64, {.i64 = 0 }, 0, INT64_MAX, FLAGS },
+ { "nb_samples", "set number of samples for fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 44100}, 1, INT32_MAX, FLAGS },
+ { "ns", "set number of samples for fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 44100}, 1, INT32_MAX, FLAGS },
+ { "start_time", "set time to start fading", OFFSET(start_time), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT32_MAX, FLAGS },
+ { "st", "set time to start fading", OFFSET(start_time), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT32_MAX, FLAGS },
+ { "duration", "set fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT32_MAX, FLAGS },
+ { "d", "set fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, INT32_MAX, FLAGS },
+ { "curve", "set fade curve type", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, 0, NB_CURVES - 1, FLAGS, "curve" },
+ { "c", "set fade curve type", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, 0, NB_CURVES - 1, FLAGS, "curve" },
+ { "tri", "linear slope", 0, AV_OPT_TYPE_CONST, {.i64 = TRI }, 0, 0, FLAGS, "curve" },
+ { "qsin", "quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = QSIN }, 0, 0, FLAGS, "curve" },
+ { "esin", "exponential sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = ESIN }, 0, 0, FLAGS, "curve" },
+ { "hsin", "half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = HSIN }, 0, 0, FLAGS, "curve" },
+ { "log", "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64 = LOG }, 0, 0, FLAGS, "curve" },
+ { "ipar", "inverted parabola", 0, AV_OPT_TYPE_CONST, {.i64 = IPAR }, 0, 0, FLAGS, "curve" },
+ { "qua", "quadratic", 0, AV_OPT_TYPE_CONST, {.i64 = QUA }, 0, 0, FLAGS, "curve" },
+ { "cub", "cubic", 0, AV_OPT_TYPE_CONST, {.i64 = CUB }, 0, 0, FLAGS, "curve" },
+ { "squ", "square root", 0, AV_OPT_TYPE_CONST, {.i64 = SQU }, 0, 0, FLAGS, "curve" },
+ { "cbr", "cubic root", 0, AV_OPT_TYPE_CONST, {.i64 = CBR }, 0, 0, FLAGS, "curve" },
+ { "par", "parabola", 0, AV_OPT_TYPE_CONST, {.i64 = PAR }, 0, 0, FLAGS, "curve" },
+ { "exp", "exponential", 0, AV_OPT_TYPE_CONST, {.i64 = EXP }, 0, 0, FLAGS, "curve" },
+ { "iqsin", "inverted quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IQSIN}, 0, 0, FLAGS, "curve" },
+ { "ihsin", "inverted half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IHSIN}, 0, 0, FLAGS, "curve" },
+ { "dese", "double-exponential seat", 0, AV_OPT_TYPE_CONST, {.i64 = DESE }, 0, 0, FLAGS, "curve" },
+ { "desi", "double-exponential sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = DESI }, 0, 0, FLAGS, "curve" },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(afade);
+
+static av_cold int init(AVFilterContext *ctx)
+{
+ AudioFadeContext *s = ctx->priv;
+
+ if (INT64_MAX - s->nb_samples < s->start_sample)
+ return AVERROR(EINVAL);
+
+ return 0;
+}
+
+static int filter_frame(AVFilterLink *inlink, AVFrame *buf)
+{
+ AudioFadeContext *s = inlink->dst->priv;
+ AVFilterLink *outlink = inlink->dst->outputs[0];
+ int nb_samples = buf->nb_samples;
+ AVFrame *out_buf;
+ int64_t cur_sample = av_rescale_q(buf->pts, inlink->time_base, (AVRational){1, inlink->sample_rate});
+
+ if ((!s->type && (s->start_sample + s->nb_samples < cur_sample)) ||
+ ( s->type && (cur_sample + nb_samples < s->start_sample)))
+ return ff_filter_frame(outlink, buf);
+
+ if (av_frame_is_writable(buf)) {
+ out_buf = buf;
+ } else {
+ out_buf = ff_get_audio_buffer(inlink, nb_samples);
+ if (!out_buf)
+ return AVERROR(ENOMEM);
+ av_frame_copy_props(out_buf, buf);
+ }
+
+ if ((!s->type && (cur_sample + nb_samples < s->start_sample)) ||
+ ( s->type && (s->start_sample + s->nb_samples < cur_sample))) {
+ av_samples_set_silence(out_buf->extended_data, 0, nb_samples,
+ av_frame_get_channels(out_buf), out_buf->format);
+ } else {
+ int64_t start;
+
+ if (!s->type)
+ start = cur_sample - s->start_sample;
+ else
+ start = s->start_sample + s->nb_samples - cur_sample;
+
+ s->fade_samples(out_buf->extended_data, buf->extended_data,
+ nb_samples, av_frame_get_channels(buf),
+ s->type ? -1 : 1, start,
+ s->nb_samples, s->curve);
+ }
+
+ if (buf != out_buf)
+ av_frame_free(&buf);
+
+ return ff_filter_frame(outlink, out_buf);
+}
+
+static const AVFilterPad avfilter_af_afade_inputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .filter_frame = filter_frame,
+ },
+ { NULL }
+};
+
+static const AVFilterPad avfilter_af_afade_outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .config_props = config_output,
+ },
+ { NULL }
+};
+
+AVFilter ff_af_afade = {
+ .name = "afade",
+ .description = NULL_IF_CONFIG_SMALL("Fade in/out input audio."),
+ .query_formats = query_formats,
+ .priv_size = sizeof(AudioFadeContext),
+ .init = init,
+ .inputs = avfilter_af_afade_inputs,
+ .outputs = avfilter_af_afade_outputs,
+ .priv_class = &afade_class,
+ .flags = AVFILTER_FLAG_SUPPORT_TIMELINE_GENERIC,
+};
+
+#endif /* CONFIG_AFADE_FILTER */
+
+#if CONFIG_ACROSSFADE_FILTER
+
+static const AVOption acrossfade_options[] = {
+ { "nb_samples", "set number of samples for cross fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 44100}, 1, INT32_MAX/10, FLAGS },
+ { "ns", "set number of samples for cross fade duration", OFFSET(nb_samples), AV_OPT_TYPE_INT, {.i64 = 44100}, 1, INT32_MAX/10, FLAGS },
+ { "duration", "set cross fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, 60, FLAGS },
+ { "d", "set cross fade duration", OFFSET(duration), AV_OPT_TYPE_DURATION, {.i64 = 0. }, 0, 60, FLAGS },
+ { "overlap", "overlap 1st stream end with 2nd stream start", OFFSET(overlap), AV_OPT_TYPE_BOOL, {.i64 = 1 }, 0, 1, FLAGS },
+ { "o", "overlap 1st stream end with 2nd stream start", OFFSET(overlap), AV_OPT_TYPE_BOOL, {.i64 = 1 }, 0, 1, FLAGS },
+ { "curve1", "set fade curve type for 1st stream", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, 0, NB_CURVES - 1, FLAGS, "curve" },
+ { "c1", "set fade curve type for 1st stream", OFFSET(curve), AV_OPT_TYPE_INT, {.i64 = TRI }, 0, NB_CURVES - 1, FLAGS, "curve" },
+ { "tri", "linear slope", 0, AV_OPT_TYPE_CONST, {.i64 = TRI }, 0, 0, FLAGS, "curve" },
+ { "qsin", "quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = QSIN }, 0, 0, FLAGS, "curve" },
+ { "esin", "exponential sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = ESIN }, 0, 0, FLAGS, "curve" },
+ { "hsin", "half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = HSIN }, 0, 0, FLAGS, "curve" },
+ { "log", "logarithmic", 0, AV_OPT_TYPE_CONST, {.i64 = LOG }, 0, 0, FLAGS, "curve" },
+ { "ipar", "inverted parabola", 0, AV_OPT_TYPE_CONST, {.i64 = IPAR }, 0, 0, FLAGS, "curve" },
+ { "qua", "quadratic", 0, AV_OPT_TYPE_CONST, {.i64 = QUA }, 0, 0, FLAGS, "curve" },
+ { "cub", "cubic", 0, AV_OPT_TYPE_CONST, {.i64 = CUB }, 0, 0, FLAGS, "curve" },
+ { "squ", "square root", 0, AV_OPT_TYPE_CONST, {.i64 = SQU }, 0, 0, FLAGS, "curve" },
+ { "cbr", "cubic root", 0, AV_OPT_TYPE_CONST, {.i64 = CBR }, 0, 0, FLAGS, "curve" },
+ { "par", "parabola", 0, AV_OPT_TYPE_CONST, {.i64 = PAR }, 0, 0, FLAGS, "curve" },
+ { "exp", "exponential", 0, AV_OPT_TYPE_CONST, {.i64 = EXP }, 0, 0, FLAGS, "curve" },
+ { "iqsin", "inverted quarter of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IQSIN}, 0, 0, FLAGS, "curve" },
+ { "ihsin", "inverted half of sine wave", 0, AV_OPT_TYPE_CONST, {.i64 = IHSIN}, 0, 0, FLAGS, "curve" },
+ { "dese", "double-exponential seat", 0, AV_OPT_TYPE_CONST, {.i64 = DESE }, 0, 0, FLAGS, "curve" },
+ { "desi", "double-exponential sigmoid", 0, AV_OPT_TYPE_CONST, {.i64 = DESI }, 0, 0, FLAGS, "curve" },
+ { "curve2", "set fade curve type for 2nd stream", OFFSET(curve2), AV_OPT_TYPE_INT, {.i64 = TRI }, 0, NB_CURVES - 1, FLAGS, "curve" },
+ { "c2", "set fade curve type for 2nd stream", OFFSET(curve2), AV_OPT_TYPE_INT, {.i64 = TRI }, 0, NB_CURVES - 1, FLAGS, "curve" },
+ { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(acrossfade);
+
+#define CROSSFADE_PLANAR(name, type) \
+static void crossfade_samples_## name ##p(uint8_t **dst, uint8_t * const *cf0, \
+ uint8_t * const *cf1, \
+ int nb_samples, int channels, \
+ int curve0, int curve1) \
+{ \
+ int i, c; \
+ \
+ for (i = 0; i < nb_samples; i++) { \
+ double gain0 = fade_gain(curve0, nb_samples - 1 - i, nb_samples); \
+ double gain1 = fade_gain(curve1, i, nb_samples); \
+ for (c = 0; c < channels; c++) { \
+ type *d = (type *)dst[c]; \
+ const type *s0 = (type *)cf0[c]; \
+ const type *s1 = (type *)cf1[c]; \
+ \
+ d[i] = s0[i] * gain0 + s1[i] * gain1; \
+ } \
+ } \
+}
+
+#define CROSSFADE(name, type) \
+static void crossfade_samples_## name (uint8_t **dst, uint8_t * const *cf0, \
+ uint8_t * const *cf1, \
+ int nb_samples, int channels, \
+ int curve0, int curve1) \
+{ \
+ type *d = (type *)dst[0]; \
+ const type *s0 = (type *)cf0[0]; \
+ const type *s1 = (type *)cf1[0]; \
+ int i, c, k = 0; \
+ \
+ for (i = 0; i < nb_samples; i++) { \
+ double gain0 = fade_gain(curve0, nb_samples - 1 - i, nb_samples); \
+ double gain1 = fade_gain(curve1, i, nb_samples); \
+ for (c = 0; c < channels; c++, k++) \
+ d[k] = s0[k] * gain0 + s1[k] * gain1; \
+ } \
+}
+
+CROSSFADE_PLANAR(dbl, double)
+CROSSFADE_PLANAR(flt, float)
+CROSSFADE_PLANAR(s16, int16_t)
+CROSSFADE_PLANAR(s32, int32_t)
+
+CROSSFADE(dbl, double)
+CROSSFADE(flt, float)
+CROSSFADE(s16, int16_t)
+CROSSFADE(s32, int32_t)
+
+static int acrossfade_filter_frame(AVFilterLink *inlink, AVFrame *in)
+{
+ AVFilterContext *ctx = inlink->dst;
+ AudioFadeContext *s = ctx->priv;
+ AVFilterLink *outlink = ctx->outputs[0];
+ AVFrame *out, *cf[2] = { NULL };
+ int ret = 0, nb_samples;
+
+ if (s->crossfade_is_over) {
+ in->pts = s->pts;
+ s->pts += av_rescale_q(in->nb_samples,
+ (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
+ return ff_filter_frame(outlink, in);
+ } else if (inlink == ctx->inputs[0]) {
+ av_audio_fifo_write(s->fifo[0], (void **)in->extended_data, in->nb_samples);
+
+ nb_samples = av_audio_fifo_size(s->fifo[0]) - s->nb_samples;
+ if (nb_samples > 0) {
+ out = ff_get_audio_buffer(outlink, nb_samples);
+ if (!out) {
+ ret = AVERROR(ENOMEM);
+ goto fail;
+ }
+ av_audio_fifo_read(s->fifo[0], (void **)out->extended_data, nb_samples);
+ out->pts = s->pts;
+ s->pts += av_rescale_q(nb_samples,
+ (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
+ ret = ff_filter_frame(outlink, out);
+ }
+ } else if (av_audio_fifo_size(s->fifo[1]) < s->nb_samples) {
+ if (!s->overlap && av_audio_fifo_size(s->fifo[0]) > 0) {
+ nb_samples = av_audio_fifo_size(s->fifo[0]);
+
+ cf[0] = ff_get_audio_buffer(outlink, nb_samples);
+ out = ff_get_audio_buffer(outlink, nb_samples);
+ if (!out || !cf[0]) {
+ ret = AVERROR(ENOMEM);
+ goto fail;
+ }
+ av_audio_fifo_read(s->fifo[0], (void **)cf[0]->extended_data, nb_samples);
+
+ s->fade_samples(out->extended_data, cf[0]->extended_data, nb_samples,
+ outlink->channels, -1, nb_samples - 1, nb_samples, s->curve);
+ out->pts = s->pts;
+ s->pts += av_rescale_q(nb_samples,
+ (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
+ ret = ff_filter_frame(outlink, out);
+ if (ret < 0)
+ goto fail;
+ }
+
+ av_audio_fifo_write(s->fifo[1], (void **)in->extended_data, in->nb_samples);
+ } else if (av_audio_fifo_size(s->fifo[1]) >= s->nb_samples) {
+ av_audio_fifo_write(s->fifo[1], (void **)in->extended_data, in->nb_samples);
+
+ if (s->overlap) {
+ cf[0] = ff_get_audio_buffer(outlink, s->nb_samples);
+ cf[1] = ff_get_audio_buffer(outlink, s->nb_samples);
+ out = ff_get_audio_buffer(outlink, s->nb_samples);
+ if (!out || !cf[0] || !cf[1]) {
+ av_frame_free(&out);
+ ret = AVERROR(ENOMEM);
+ goto fail;
+ }
+
+ av_audio_fifo_read(s->fifo[0], (void **)cf[0]->extended_data, s->nb_samples);
+ av_audio_fifo_read(s->fifo[1], (void **)cf[1]->extended_data, s->nb_samples);
+
+ s->crossfade_samples(out->extended_data, cf[0]->extended_data,
+ cf[1]->extended_data,
+ s->nb_samples, av_frame_get_channels(in),
+ s->curve, s->curve2);
+ out->pts = s->pts;
+ s->pts += av_rescale_q(s->nb_samples,
+ (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
+ ret = ff_filter_frame(outlink, out);
+ if (ret < 0)
+ goto fail;
+ } else {
+ out = ff_get_audio_buffer(outlink, s->nb_samples);
+ cf[1] = ff_get_audio_buffer(outlink, s->nb_samples);
+ if (!out || !cf[1]) {
+ ret = AVERROR(ENOMEM);
+ av_frame_free(&out);
+ goto fail;
+ }
+
+ av_audio_fifo_read(s->fifo[1], (void **)cf[1]->extended_data, s->nb_samples);
+
+ s->fade_samples(out->extended_data, cf[1]->extended_data, s->nb_samples,
+ outlink->channels, 1, 0, s->nb_samples, s->curve2);
+ out->pts = s->pts;
+ s->pts += av_rescale_q(s->nb_samples,
+ (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
+ ret = ff_filter_frame(outlink, out);
+ if (ret < 0)
+ goto fail;
+ }
+
+ nb_samples = av_audio_fifo_size(s->fifo[1]);
+ if (nb_samples > 0) {
+ out = ff_get_audio_buffer(outlink, nb_samples);
+ if (!out) {
+ ret = AVERROR(ENOMEM);
+ goto fail;
+ }
+
+ av_audio_fifo_read(s->fifo[1], (void **)out->extended_data, nb_samples);
+ out->pts = s->pts;
+ s->pts += av_rescale_q(nb_samples,
+ (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
+ ret = ff_filter_frame(outlink, out);
+ }
+ s->crossfade_is_over = 1;
+ }
+
+fail:
+ av_frame_free(&in);
+ av_frame_free(&cf[0]);
+ av_frame_free(&cf[1]);
+ return ret;
+}
+
+static int acrossfade_request_frame(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ AudioFadeContext *s = ctx->priv;
+ int ret = 0;
+
+ if (!s->cf0_eof) {
+ AVFilterLink *cf0 = ctx->inputs[0];
+ ret = ff_request_frame(cf0);
+ if (ret < 0 && ret != AVERROR_EOF)
+ return ret;
+ if (ret == AVERROR_EOF) {
+ s->cf0_eof = 1;
+ ret = 0;
+ }
+ } else {
+ AVFilterLink *cf1 = ctx->inputs[1];
+ int nb_samples = av_audio_fifo_size(s->fifo[1]);
+
+ ret = ff_request_frame(cf1);
+ if (ret == AVERROR_EOF && nb_samples > 0) {
+ AVFrame *out = ff_get_audio_buffer(outlink, nb_samples);
+ if (!out)
+ return AVERROR(ENOMEM);
+
+ av_audio_fifo_read(s->fifo[1], (void **)out->extended_data, nb_samples);
+ ret = ff_filter_frame(outlink, out);
+ }
+ }
+
+ return ret;
+}
+
+static int acrossfade_config_output(AVFilterLink *outlink)
+{
+ AVFilterContext *ctx = outlink->src;
+ AudioFadeContext *s = ctx->priv;
+
+ if (ctx->inputs[0]->sample_rate != ctx->inputs[1]->sample_rate) {
+ av_log(ctx, AV_LOG_ERROR,
+ "Inputs must have the same sample rate "
+ "%d for in0 vs %d for in1\n",
+ ctx->inputs[0]->sample_rate, ctx->inputs[1]->sample_rate);
+ return AVERROR(EINVAL);
+ }
+
+ outlink->sample_rate = ctx->inputs[0]->sample_rate;
+ outlink->time_base = ctx->inputs[0]->time_base;
+ outlink->channel_layout = ctx->inputs[0]->channel_layout;
+ outlink->channels = ctx->inputs[0]->channels;
+
+ switch (outlink->format) {
+ case AV_SAMPLE_FMT_DBL: s->crossfade_samples = crossfade_samples_dbl; break;
+ case AV_SAMPLE_FMT_DBLP: s->crossfade_samples = crossfade_samples_dblp; break;
+ case AV_SAMPLE_FMT_FLT: s->crossfade_samples = crossfade_samples_flt; break;
+ case AV_SAMPLE_FMT_FLTP: s->crossfade_samples = crossfade_samples_fltp; break;
+ case AV_SAMPLE_FMT_S16: s->crossfade_samples = crossfade_samples_s16; break;
+ case AV_SAMPLE_FMT_S16P: s->crossfade_samples = crossfade_samples_s16p; break;
+ case AV_SAMPLE_FMT_S32: s->crossfade_samples = crossfade_samples_s32; break;
+ case AV_SAMPLE_FMT_S32P: s->crossfade_samples = crossfade_samples_s32p; break;
+ }
+
+ config_output(outlink);
+
+ s->fifo[0] = av_audio_fifo_alloc(outlink->format, outlink->channels, s->nb_samples);
+ s->fifo[1] = av_audio_fifo_alloc(outlink->format, outlink->channels, s->nb_samples);
+ if (!s->fifo[0] || !s->fifo[1])
+ return AVERROR(ENOMEM);
+
+ return 0;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+ AudioFadeContext *s = ctx->priv;
+
+ av_audio_fifo_free(s->fifo[0]);
+ av_audio_fifo_free(s->fifo[1]);
+}
+
+static const AVFilterPad avfilter_af_acrossfade_inputs[] = {
+ {
+ .name = "crossfade0",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .filter_frame = acrossfade_filter_frame,
+ },
+ {
+ .name = "crossfade1",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .filter_frame = acrossfade_filter_frame,
+ },
+ { NULL }
+};
+
+static const AVFilterPad avfilter_af_acrossfade_outputs[] = {
+ {
+ .name = "default",
+ .type = AVMEDIA_TYPE_AUDIO,
+ .request_frame = acrossfade_request_frame,
+ .config_props = acrossfade_config_output,
+ },
+ { NULL }
+};
+
+AVFilter ff_af_acrossfade = {
+ .name = "acrossfade",
+ .description = NULL_IF_CONFIG_SMALL("Cross fade two input audio streams."),
+ .query_formats = query_formats,
+ .priv_size = sizeof(AudioFadeContext),
+ .uninit = uninit,
+ .priv_class = &acrossfade_class,
+ .inputs = avfilter_af_acrossfade_inputs,
+ .outputs = avfilter_af_acrossfade_outputs,
+};
+
+#endif /* CONFIG_ACROSSFADE_FILTER */
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-13 15:08:52 +0000