[FFmpeg-devel] [PATCH v3] GSoC 2023: Add Audio Overlay Filter

* [FFmpeg-devel] [PATCH v3] GSoC 2023: Add Audio Overlay Filter
@ 2023-09-11 19:17 Harshit Karwal
  2023-09-25 18:05 ` Paul B Mahol
  0 siblings, 1 reply; 10+ messages in thread
From: Harshit Karwal @ 2023-09-11 19:17 UTC (permalink / raw)
  To: ffmpeg-devel; +Cc: Harshit Karwal

1. Replaced ring buffer ADT with AVAudioFifo from libavutil/audio_fifo.h
2. Fixed potential freeing of uninitialised pointers in uninit
3. Minor changes like removing unused headers

Signed-off-by: Harshit Karwal <karwalharshit@gmail.com>
---
 doc/filters.texi          |  40 +++
 libavfilter/Makefile      |   1 +
 libavfilter/af_aoverlay.c | 548 ++++++++++++++++++++++++++++++++++++++
 libavfilter/allfilters.c  |   1 +
 4 files changed, 590 insertions(+)
 create mode 100644 libavfilter/af_aoverlay.c

diff --git a/doc/filters.texi b/doc/filters.texi
index cac1ee4381..f6a2ab9743 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -2709,6 +2709,46 @@ This filter supports the same commands as options, excluding option @code{order}
 
 Pass the audio source unchanged to the output.
 
+@section aoverlay
+
+Replace a specified section of an audio stream with another input audio stream.
+
+In case no enable option for timeline editing is specified, the second audio stream will
+be output at sections of the first stream which have a gap in PTS (Presentation TimeStamp) values
+such that the output stream's PTS values are monotonous.
+
+This filter also supports linear cross fading when transitioning from one
+input stream to another.
+
+The filter accepts the following option:
+
+@table @option
+@item cf_duration
+Set duration (in seconds) for cross fade between the inputs. Default value is @code{100} milliseconds.
+@end table
+
+@subsection Examples
+
+@itemize
+@item
+Replace the first stream with the second stream from @code{t=10} seconds to @code{t=20} seconds:
+@example
+ffmpeg -i first.wav -i second.wav -filter_complex "aoverlay=enable='between(t,10,20)'" output.wav
+@end example
+
+@item
+Do the same as above, but with crossfading for @code{2} seconds between the streams:
+@example
+ffmpeg -i first.wav -i second.wav -filter_complex "aoverlay=cf_duration=2:enable='between(t,10,20)'" output.wav
+@end example
+
+@item
+Introduce a PTS gap from @code{t=4} seconds to @code{t=8} seconds in the first stream and output the second stream during this gap:
+@example
+ffmpeg -i first.wav -i second.wav -filter_complex "[0]aselect='not(between(t,4,8))'[temp];[temp][1]aoverlay[out]" -map "[out]" output.wav
+@end example
+@end itemize
+
 @section apad
 
 Pad the end of an audio stream with silence.
diff --git a/libavfilter/Makefile b/libavfilter/Makefile
index 2fe0033b21..c469380038 100644
--- a/libavfilter/Makefile
+++ b/libavfilter/Makefile
@@ -80,6 +80,7 @@ OBJS-$(CONFIG_ANLMDN_FILTER)                 += af_anlmdn.o
 OBJS-$(CONFIG_ANLMF_FILTER)                  += af_anlms.o
 OBJS-$(CONFIG_ANLMS_FILTER)                  += af_anlms.o
 OBJS-$(CONFIG_ANULL_FILTER)                  += af_anull.o
+OBJS-$(CONFIG_AOVERLAY_FILTER)               += af_aoverlay.o
 OBJS-$(CONFIG_APAD_FILTER)                   += af_apad.o
 OBJS-$(CONFIG_APERMS_FILTER)                 += f_perms.o
 OBJS-$(CONFIG_APHASER_FILTER)                += af_aphaser.o generate_wave_table.o
diff --git a/libavfilter/af_aoverlay.c b/libavfilter/af_aoverlay.c
new file mode 100644
index 0000000000..fffacb97e9
--- /dev/null
+++ b/libavfilter/af_aoverlay.c
@@ -0,0 +1,548 @@
+/*
+ * Copyright (c) 2023 Harshit Karwal
+ *
+ * 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 "libavutil/opt.h"
+#include "libavutil/audio_fifo.h"
+
+#include "audio.h"
+#include "avfilter.h"
+#include "filters.h"
+#include "internal.h"
+
+typedef struct AOverlayContext {
+    const AVClass *class;
+    AVFrame *main_input;
+    AVFrame *overlay_input;
+    int64_t pts;
+    int main_eof;
+    int overlay_eof;
+
+    int default_mode;
+    int previous_samples;
+    int64_t pts_gap;
+    int64_t previous_pts;
+    int64_t pts_gap_start;
+    int64_t pts_gap_end;
+
+    int is_disabled;
+    int nb_channels;
+    int crossfade_ready;
+    AVAudioFifo *main_sample_buffers;
+    AVAudioFifo *overlay_sample_buffers;
+    int64_t cf_duration;
+    int64_t cf_samples;
+    void (*crossfade_samples)(uint8_t **dst, uint8_t * const *cf0,
+                              uint8_t * const *cf1,
+                              int nb_samples, int channels);
+
+    int64_t transition_pts;
+    int64_t transition_pts2;
+
+    uint8_t **cf0;
+    uint8_t **cf1;
+    int cf_buffers_unavailable[2];
+} AOverlayContext;
+
+static const enum AVSampleFormat sample_fmts[] = {
+    AV_SAMPLE_FMT_DBLP, AV_SAMPLE_FMT_FLTP,
+    AV_SAMPLE_FMT_S16P, AV_SAMPLE_FMT_S32P,
+    AV_SAMPLE_FMT_NONE
+};
+
+#define SEGMENT_SIZE 1024
+
+#define OFFSET(x) offsetof(AOverlayContext, x)
+
+#define FLAGS AV_OPT_FLAG_AUDIO_PARAM | AV_OPT_FLAG_FILTERING_PARAM
+
+static const AVOption aoverlay_options[] = {
+    { "cf_duration",    "set duration (in seconds) for cross fade between the inputs", OFFSET(cf_duration),    AV_OPT_TYPE_DURATION,   {.i64 = 100000}, 0,  60000000,   FLAGS },
+    { NULL }
+};
+
+AVFILTER_DEFINE_CLASS(aoverlay);
+
+#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)              \
+{                                                                                    \
+    for (int i = 0; i < nb_samples; i++) {                                           \
+        double main_gain = av_clipd(1.0 * (nb_samples - 1 - i) / nb_samples, 0, 1.); \
+        double overlay_gain = av_clipd(1.0 * i / nb_samples, 0, 1.);                 \
+        for (int 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] * main_gain + s1[i] * overlay_gain;                         \
+        }                                                                            \
+    }                                                                                \
+}
+
+CROSSFADE_PLANAR(dbl, double)
+CROSSFADE_PLANAR(flt, float)
+CROSSFADE_PLANAR(s16, int16_t)
+CROSSFADE_PLANAR(s32, int32_t)
+
+static av_cold int init(AVFilterContext *ctx)
+{
+    AOverlayContext *s  = ctx->priv;
+
+    s->is_disabled      = 1;
+    s->transition_pts   = AV_NOPTS_VALUE;
+    s->transition_pts2  = AV_NOPTS_VALUE;
+
+    return 0;
+}
+
+static av_cold void uninit(AVFilterContext *ctx)
+{
+    AOverlayContext *s = ctx->priv;
+
+    av_audio_fifo_free(s->main_sample_buffers);
+    av_audio_fifo_free(s->overlay_sample_buffers);
+
+    for (int i = 0; i < s->nb_channels; i++) {
+        if (!s->cf_buffers_unavailable[0])
+            av_freep(&s->cf0[i]);
+        if (!s->cf_buffers_unavailable[1])
+            av_freep(&s->cf1[i]);
+    }
+    av_freep(&s->cf0);
+    av_freep(&s->cf1);
+
+    av_frame_free(&s->main_input);
+    av_frame_free(&s->overlay_input);
+}
+
+static int crossfade_prepare(AOverlayContext *s, AVFilterLink *main_inlink, AVFilterLink *overlay_inlink, AVFilterLink *outlink,
+                             int nb_samples, AVFrame **main_buffer, AVFrame **overlay_buffer, int mode)
+{
+    int ret;
+
+    *main_buffer = ff_get_audio_buffer(outlink, nb_samples);
+    if (!(*main_buffer))
+        return AVERROR(ENOMEM);
+
+    (*main_buffer)->pts = s->pts;
+    s->pts += av_rescale_q(nb_samples, (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
+
+    if (ret = av_audio_fifo_read(s->main_sample_buffers, (void **) (*main_buffer)->extended_data, nb_samples) < 0)
+        return ret;
+
+    if (mode == 1) {
+        s->previous_samples = (*main_buffer)->nb_samples;
+    } else if (mode == -1 || (mode == 0 && s->is_disabled)) {
+        *overlay_buffer = ff_get_audio_buffer(outlink, nb_samples);
+        if (!(*overlay_buffer))
+            return AVERROR(ENOMEM);
+
+        if (ret = av_audio_fifo_read(s->overlay_sample_buffers, (void **) (*overlay_buffer)->extended_data, nb_samples) < 0)
+            return ret;
+
+        (*overlay_buffer)->pts = (*main_buffer)->pts;
+    }
+
+    s->crossfade_ready = 1;
+
+    return 0;
+}
+
+static int crossfade_samples(AOverlayContext *s, AVFilterLink *main_inlink, AVFilterLink *overlay_inlink, AVFilterLink *outlink,
+                             int nb_samples, AVFrame **out, int mode)
+{
+    int ret;
+
+    *out = ff_get_audio_buffer(outlink, nb_samples);
+    if (!(*out))
+        return AVERROR(ENOMEM);
+
+    if (ret = av_audio_fifo_read(s->main_sample_buffers, (void **) s->cf0, nb_samples) < 0)
+        return ret;
+
+    if (ret = av_audio_fifo_read(s->overlay_sample_buffers, (void **) s->cf1, nb_samples) < 0)
+        return ret;
+
+    if (mode == 0) {
+        s->is_disabled ? s->crossfade_samples((*out)->extended_data, s->cf1, s->cf0, nb_samples, (*out)->ch_layout.nb_channels)
+                       : s->crossfade_samples((*out)->extended_data, s->cf0, s->cf1, nb_samples, (*out)->ch_layout.nb_channels);
+    } else if (mode == -1) {
+        s->crossfade_samples((*out)->extended_data, s->cf1, s->cf0, s->cf_samples, (*out)->ch_layout.nb_channels);
+    } else if (mode == 1) {
+        s->transition_pts2 != AV_NOPTS_VALUE ? s->crossfade_samples((*out)->extended_data, s->cf1, s->cf0, nb_samples, (*out)->ch_layout.nb_channels)
+                                             : s->crossfade_samples((*out)->extended_data, s->cf0, s->cf1, nb_samples, (*out)->ch_layout.nb_channels);
+    }
+
+    (*out)->pts = s->pts;
+    s->pts += av_rescale_q(nb_samples, (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
+    s->transition_pts = AV_NOPTS_VALUE;
+    s->transition_pts2 = AV_NOPTS_VALUE;
+    s->crossfade_ready = 0;
+
+    return 0;
+}
+
+static int consume_samples(AOverlayContext *s, AVFilterLink *overlay_inlink, AVFilterLink *outlink)
+{
+    int ret, status, nb_samples;
+    int64_t pts;
+
+    nb_samples = FFMIN(SEGMENT_SIZE, av_audio_fifo_space(s->overlay_sample_buffers));
+
+    ret = ff_inlink_consume_samples(overlay_inlink, nb_samples, nb_samples, &s->overlay_input);
+    if (ret < 0) {
+        return ret;
+    } else if (ff_inlink_acknowledge_status(overlay_inlink, &status, &pts)) {
+        s->overlay_eof = 1;
+        return 0;
+    } else if (!ret) {
+        if (ff_outlink_frame_wanted(outlink))
+            ff_inlink_request_frame(overlay_inlink);
+        return 0;
+    }
+
+    if (ret = av_audio_fifo_write(s->overlay_sample_buffers, (void **)s->overlay_input->extended_data, nb_samples) < 0)
+        return ret;
+
+    return 1;
+}
+
+static int activate(AVFilterContext *ctx)
+{
+    AOverlayContext *s = ctx->priv;
+    int status, ret, nb_samples;
+    int64_t pts;
+    AVFrame *out = NULL, *main_buffer = NULL, *overlay_buffer = NULL;
+
+    AVFilterLink *main_inlink = ctx->inputs[0];
+    AVFilterLink *overlay_inlink = ctx->inputs[1];
+    AVFilterLink *outlink = ctx->outputs[0];
+
+    FF_FILTER_FORWARD_STATUS_BACK_ALL(outlink, ctx);
+
+    if (s->default_mode && (s->pts_gap_end - s->pts_gap_start <= 0 || s->overlay_eof)) {
+        s->default_mode = 0;
+        s->transition_pts2 = s->pts_gap_end;
+    }
+
+    if (av_audio_fifo_space(s->main_sample_buffers) != 0 && !s->main_eof && !s->default_mode) {
+        nb_samples = FFMIN(SEGMENT_SIZE, av_audio_fifo_space(s->main_sample_buffers));
+
+        ret = ff_inlink_consume_samples(main_inlink, nb_samples, nb_samples, &s->main_input);
+        if (ret > 0) {
+            if (ctx->enable_str && s->is_disabled != ctx->is_disabled && !s->overlay_eof) {
+                s->is_disabled = ctx->is_disabled;
+                s->transition_pts = s->main_input->pts;
+
+                if (s->main_input->nb_samples < av_audio_fifo_space(s->main_sample_buffers))
+                    s->crossfade_ready = 1;
+                if (av_audio_fifo_size(s->main_sample_buffers) == 0) {
+                    s->transition_pts = AV_NOPTS_VALUE;
+                    s->crossfade_ready = 0;
+                }
+            }
+            if (!ctx->enable_str && !s->default_mode) {
+                if (s->previous_pts + av_rescale_q(s->previous_samples, (AVRational){ 1, outlink->sample_rate }, outlink->time_base) >= s->main_input->pts) {
+                    s->default_mode = 0;
+                    s->previous_pts = s->main_input->pts;
+                    s->previous_samples = s->main_input->nb_samples;
+                } else if (!s->overlay_eof) {
+                    s->pts_gap_start = s->previous_pts;
+                    if (s->pts > 0 || av_audio_fifo_size(s->main_sample_buffers) > 0)
+                        s->transition_pts = s->pts_gap_start;
+                    s->pts_gap_end = s->main_input->pts;
+                    s->default_mode = 1;
+                }
+            }
+
+            if (ret = av_audio_fifo_write(s->main_sample_buffers, (void **)s->main_input->extended_data, nb_samples) < 0)
+                return ret;
+        } else if (ret < 0) {
+            return ret;
+        } else if (ff_inlink_acknowledge_status(main_inlink, &status, &pts)) {
+            s->main_eof = 1;
+            s->crossfade_ready = 1;
+        } else if (!ret) {
+            if (ff_outlink_frame_wanted(outlink))
+                ff_inlink_request_frame(main_inlink);
+            return 0;
+        }
+    }
+
+    if (s->main_eof && av_audio_fifo_size(s->main_sample_buffers) == 0 && ff_inlink_acknowledge_status(main_inlink, &status, &pts)) {
+        ff_outlink_set_status(outlink, status, pts);
+        return 0;
+    }
+
+    if (av_audio_fifo_space(s->main_sample_buffers) > 0 &&
+        (s->transition_pts == AV_NOPTS_VALUE || av_audio_fifo_size(s->main_sample_buffers) != s->cf_samples) && !s->default_mode) {
+        if (ff_inlink_acknowledge_status(main_inlink, &status, &pts)) {
+            s->main_eof = 1;
+            s->crossfade_ready = 1;
+        } else {
+            ff_inlink_request_frame(main_inlink);
+            return 0;
+        }
+    }
+
+    if (!s->overlay_eof) {
+        if (av_audio_fifo_space(s->overlay_sample_buffers) > 0) {
+            ret = consume_samples(s, overlay_inlink, outlink);
+            if (ret <= 0) {
+                if (!s->overlay_eof)
+                    return ret;
+            }
+        }
+
+        if (av_audio_fifo_space(s->overlay_sample_buffers) > 0) {
+            if (ff_inlink_acknowledge_status(overlay_inlink, &status, &pts)) {
+                s->overlay_eof = 1;
+                s->transition_pts = s->pts + av_rescale_q(av_audio_fifo_size(s->overlay_sample_buffers) - (s->cf_samples / 2),
+                                                          (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
+                s->is_disabled = 1;
+            } else {
+                ff_inlink_request_frame(overlay_inlink);
+                return 0;
+            }
+        }
+    }
+
+    if (!ctx->enable_str) {
+        if (s->transition_pts != AV_NOPTS_VALUE && av_audio_fifo_size(s->main_sample_buffers) > s->cf_samples + SEGMENT_SIZE) {
+            nb_samples = av_audio_fifo_size(s->main_sample_buffers) + av_audio_fifo_space(s->main_sample_buffers) - s->cf_samples - SEGMENT_SIZE;
+
+            if (ret = crossfade_prepare(s, main_inlink, overlay_inlink, outlink, nb_samples, &main_buffer, &overlay_buffer, 1) < 0)
+                return ret;
+
+            return ff_filter_frame(outlink, main_buffer);
+        } else if (s->transition_pts != AV_NOPTS_VALUE || s->transition_pts2 != AV_NOPTS_VALUE) {
+            nb_samples = FFMIN(s->cf_samples, av_audio_fifo_size(s->main_sample_buffers) - SEGMENT_SIZE);
+
+            if (ret = crossfade_samples(s, main_inlink, overlay_inlink, outlink, nb_samples, &out, 1) < 0)
+                return ret;
+
+            return ff_filter_frame(outlink, out);
+        } else if (!s->default_mode) {
+            nb_samples = FFMIN(av_audio_fifo_size(s->main_sample_buffers), SEGMENT_SIZE);
+
+            main_buffer = ff_get_audio_buffer(outlink, nb_samples);
+            if (!main_buffer)
+                return AVERROR(ENOMEM);
+
+            main_buffer->pts = s->pts;
+            s->pts += av_rescale_q(nb_samples, (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
+
+            if (ret = av_audio_fifo_read(s->main_sample_buffers, (void **)main_buffer->extended_data, nb_samples) < 0)
+                return ret;
+        }
+
+        if (!s->default_mode || s->overlay_eof) {
+            s->previous_samples = main_buffer->nb_samples;
+            return ff_filter_frame(outlink, main_buffer);
+        }
+
+        s->pts_gap = s->pts_gap_end - s->pts_gap_start;
+
+        nb_samples = FFMIN(SEGMENT_SIZE, av_rescale_q(s->pts_gap, outlink->time_base, (AVRational){ 1, outlink->sample_rate }));
+
+        overlay_buffer = ff_get_audio_buffer(outlink, nb_samples);
+        if (!overlay_buffer)
+            return AVERROR(ENOMEM);
+
+        if (ret = av_audio_fifo_read(s->overlay_sample_buffers, (void **)overlay_buffer->extended_data, nb_samples) < 0)
+            return ret;
+
+        s->previous_samples = nb_samples;
+        s->previous_pts += av_rescale_q(nb_samples, (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
+        s->pts_gap_start += av_rescale_q(nb_samples, (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
+
+        overlay_buffer->pts = s->pts;
+        s->pts += av_rescale_q(nb_samples, (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
+
+        av_frame_free(&main_buffer);
+
+        return ff_filter_frame(outlink, overlay_buffer);
+    }
+
+    if (s->overlay_eof && av_audio_fifo_size(s->overlay_sample_buffers) > 0) {
+        if (av_audio_fifo_size(s->overlay_sample_buffers) != s->cf_samples) {
+            nb_samples = av_audio_fifo_size(s->overlay_sample_buffers) - s->cf_samples;
+
+            if (ret = crossfade_prepare(s, main_inlink, overlay_inlink, outlink, nb_samples, &main_buffer, &overlay_buffer, -1) < 0)
+                return ret;
+
+            return ff_filter_frame(outlink, overlay_buffer);
+        } else if (av_audio_fifo_size(s->overlay_sample_buffers) == s->cf_samples) {
+            if (ret = crossfade_samples(s, main_inlink, overlay_inlink, outlink, s->cf_samples, &out, -1) < 0)
+                return ret;
+
+            return ff_filter_frame(outlink, out);
+        }
+    }
+
+    if (s->transition_pts != AV_NOPTS_VALUE && !s->crossfade_ready) {
+        nb_samples = av_rescale_q(s->transition_pts - (s->cf_samples / 2) - s->pts, outlink->time_base, (AVRational) { 1, outlink->sample_rate });
+
+        if (ret = crossfade_prepare(s, main_inlink, overlay_inlink, outlink, nb_samples, &main_buffer, &overlay_buffer, 0) < 0)
+            return ret;
+    } else if (s->transition_pts != AV_NOPTS_VALUE) {
+        nb_samples = s->main_eof ? av_audio_fifo_size(s->main_sample_buffers) : s->cf_samples;
+        if (s->transition_pts < av_rescale_q(s->cf_samples, (AVRational){ 1, outlink->sample_rate }, outlink->time_base)) {
+            nb_samples = av_rescale_q(s->transition_pts, outlink->time_base, (AVRational){ 1, outlink->sample_rate });
+        }
+
+        if (ret = crossfade_samples(s, main_inlink, overlay_inlink, outlink, nb_samples, &out, 0) < 0)
+            return ret;
+
+        return ff_filter_frame(outlink, out);
+    } else {
+        nb_samples = FFMIN(av_audio_fifo_size(s->main_sample_buffers), SEGMENT_SIZE);
+        main_buffer = ff_get_audio_buffer(outlink, nb_samples);
+        if (!main_buffer)
+            return AVERROR(ENOMEM);
+
+        main_buffer->pts = s->pts;
+        s->pts += av_rescale_q(nb_samples, (AVRational){ 1, outlink->sample_rate }, outlink->time_base);
+
+        if (ret = av_audio_fifo_read(s->main_sample_buffers, (void **)main_buffer->extended_data, nb_samples) < 0)
+            return ret;
+    }
+
+    if (!ff_inlink_evaluate_timeline_at_frame(main_inlink, main_buffer) || (s->overlay_eof && av_audio_fifo_size(s->overlay_sample_buffers) == 0)) {
+        return ff_filter_frame(outlink, main_buffer);
+    } else {
+        if (s->transition_pts == AV_NOPTS_VALUE) {
+            nb_samples = FFMIN(av_audio_fifo_size(s->overlay_sample_buffers), SEGMENT_SIZE);
+            overlay_buffer = ff_get_audio_buffer(outlink, nb_samples);
+            if (!overlay_buffer)
+                return AVERROR(ENOMEM);
+
+            if (ret = av_audio_fifo_read(s->overlay_sample_buffers, (void **)overlay_buffer->extended_data, nb_samples) < 0)
+                return ret;
+
+            overlay_buffer->pts = main_buffer->pts;
+        }
+        av_frame_free(&main_buffer);
+        return ff_filter_frame(outlink, overlay_buffer);
+    }
+}
+
+static int config_output(AVFilterLink *outlink)
+{
+    AVFilterContext *ctx = outlink->src;
+    AOverlayContext *s = ctx->priv;
+    int error = 0, size, fifo_size;
+
+    switch (outlink->format) {
+    case AV_SAMPLE_FMT_DBLP: s->crossfade_samples = crossfade_samples_dblp;
+                             size = sizeof(double);
+                             break;
+    case AV_SAMPLE_FMT_FLTP: s->crossfade_samples = crossfade_samples_fltp;
+                             size = sizeof(float);
+                             break;
+    case AV_SAMPLE_FMT_S16P: s->crossfade_samples = crossfade_samples_s16p;
+                             size = sizeof(int16_t);
+                             break;
+    case AV_SAMPLE_FMT_S32P: s->crossfade_samples = crossfade_samples_s32p;
+                             size = sizeof(int32_t);
+                             break;
+    }
+
+    if (s->cf_duration)
+        s->cf_samples = av_rescale(s->cf_duration, outlink->sample_rate, AV_TIME_BASE);
+    else
+        s->cf_samples = av_rescale(100000, outlink->sample_rate, AV_TIME_BASE);
+
+    s->nb_channels = outlink->ch_layout.nb_channels;
+
+    fifo_size = SEGMENT_SIZE + SEGMENT_SIZE * (1 + ((s->cf_samples - 1) / SEGMENT_SIZE));
+
+    s->main_sample_buffers = av_audio_fifo_alloc(outlink->format, s->nb_channels, fifo_size);
+    if (!s->main_sample_buffers)
+        error = 1;
+
+    s->overlay_sample_buffers = av_audio_fifo_alloc(outlink->format, s->nb_channels, fifo_size);
+    if (!s->overlay_sample_buffers)
+        error = 1;
+
+    s->cf0 = av_malloc_array(s->nb_channels, sizeof(uint8_t*));
+    if (!s->cf0) {
+        s->cf_buffers_unavailable[0] = 1;
+        error = 1;
+    }
+
+    s->cf1 = av_malloc_array(s->nb_channels, sizeof(uint8_t*));
+    if (!s->cf1) {
+        s->cf_buffers_unavailable[1] = 1;
+        error = 1;
+    }
+
+    for (int i = 0; i < s->nb_channels; i++) {
+        if (!s->cf_buffers_unavailable[0]) {
+            s->cf0[i] = av_malloc_array(s->cf_samples, size);
+            if (!s->cf0[i])
+                error = 1;
+        }
+        if (!s->cf_buffers_unavailable[1]) {
+            s->cf1[i] = av_malloc_array(s->cf_samples, size);
+            if (!s->cf1[i])
+                error = 1;
+        }
+    }
+
+    if (error)
+        return AVERROR(ENOMEM);
+
+    return 0;
+}
+
+static const AVFilterPad af_aoverlay_inputs[] = {
+    {
+        .name = "main",
+        .type = AVMEDIA_TYPE_AUDIO,
+    },
+    {
+        .name = "overlay",
+        .type = AVMEDIA_TYPE_AUDIO,
+    },
+};
+
+static const AVFilterPad af_aoverlay_outputs[] = {
+    {
+        .name           = "default",
+        .type           = AVMEDIA_TYPE_AUDIO,
+        .config_props   = config_output,
+    },
+};
+
+const AVFilter ff_af_aoverlay = {
+    .name           = "aoverlay",
+    .description    = NULL_IF_CONFIG_SMALL("Replace a specified section of an audio stream with another audio input."),
+    .priv_size      = sizeof(AOverlayContext),
+    .priv_class     = &aoverlay_class,
+    .activate       = activate,
+    .init           = init,
+    .uninit         = uninit,
+    FILTER_INPUTS(af_aoverlay_inputs),
+    FILTER_OUTPUTS(af_aoverlay_outputs),
+    FILTER_SAMPLEFMTS_ARRAY(sample_fmts),
+    .flags          = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
+};
diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
index d4184d6e80..abdeb40fb4 100644
--- a/libavfilter/allfilters.c
+++ b/libavfilter/allfilters.c
@@ -66,6 +66,7 @@ extern const AVFilter ff_af_anlmdn;
 extern const AVFilter ff_af_anlmf;
 extern const AVFilter ff_af_anlms;
 extern const AVFilter ff_af_anull;
+extern const AVFilter ff_af_aoverlay;
 extern const AVFilter ff_af_apad;
 extern const AVFilter ff_af_aperms;
 extern const AVFilter ff_af_aphaser;
-- 
2.39.3 (Apple Git-145)

_______________________________________________
ffmpeg-devel mailing list
ffmpeg-devel@ffmpeg.org
https://ffmpeg.org/mailman/listinfo/ffmpeg-devel

To unsubscribe, visit link above, or email
ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe".

^ permalink raw reply	[flat|nested] 10+ messages in thread