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

1. Added af_aoverlay.c
2. Updated filter documentation
3. Included the filter in libavfilter/Makefile and libavfilter/allfilters.c

Signed-off-by: Harshit Karwal <karwalharshit@gmail.com>
---
 doc/filters.texi          |  40 +++
 libavfilter/Makefile      |   1 +
 libavfilter/af_aoverlay.c | 623 ++++++++++++++++++++++++++++++++++++++
 libavfilter/allfilters.c  |   1 +
 4 files changed, 665 insertions(+)
 create mode 100644 libavfilter/af_aoverlay.c

On Wed, Aug 23, 2023 at 12:16 PM Harshit Karwal <karwalharshit@gmail.com>
wrote:

> 1. Added af_aoverlay.c
> 2. Updated filter documentation
> 3. Included the filter in libavfilter/Makefile and libavfilter/allfilters.c
>
> Signed-off-by: Harshit Karwal <karwalharshit@gmail.com>
> ---
>  doc/filters.texi          |  40 +++
>  libavfilter/Makefile      |   1 +
>  libavfilter/af_aoverlay.c | 623 ++++++++++++++++++++++++++++++++++++++
>  libavfilter/allfilters.c  |   1 +
>  4 files changed, 665 insertions(+)
>  create mode 100644 libavfilter/af_aoverlay.c
>
> diff --git a/doc/filters.texi b/doc/filters.texi
> index cac1ee43810..f6a2ab9743e 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 2fe0033b218..c469380038f 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 00000000000..ea0759fc856
> --- /dev/null
> +++ b/libavfilter/af_aoverlay.c
> @@ -0,0 +1,623 @@
> +/*
> + * 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/log.h"
> +
> +#include "audio.h"
> +#include "avfilter.h"
> +#include "filters.h"
> +#include "internal.h"
> +#include "formats.h"
> +
> +typedef struct FrameRingBuffer {
> +    uint8_t *extended_data_buffer;
> +    int capacity;
> +    int start;
> +    int end;
> +    int size;
> +} FrameRingBuffer;
> +
> +static int ring_init(FrameRingBuffer **ring, unsigned int capacity, int
> size)
> +{
> +    *ring = av_malloc(sizeof(FrameRingBuffer));
> +    if (!*ring)
> +        return AVERROR(ENOMEM);
> +
> +    (*ring)->extended_data_buffer = av_malloc_array(capacity, size);
> +
> +    if (!(*ring)->extended_data_buffer)
> +        return AVERROR(ENOMEM);
> +
> +    (*ring)->capacity = capacity;
> +    (*ring)->start = 0;
> +    (*ring)->end = 0;
> +    (*ring)->size = 0;
> +
> +    return 0;
> +}
> +
> +static int ring_empty(FrameRingBuffer *ring)
> +{
> +    return ring->size == 0;
> +}
> +
> +static int ring_full(FrameRingBuffer *ring)
> +{
> +    return ring->size == ring->capacity;
> +}
> +
> +static int ring_insert(FrameRingBuffer **sample_buffers, AVFrame *frame,
> AVFilterLink *inlink)
> +{
> +    uint8_t *dst;
> +
> +    for (int c = 0; c < inlink->ch_layout.nb_channels; c++) {
> +        for (int i = 0; i < frame->nb_samples; i++) {
> +            if (ring_full(sample_buffers[c]))
> +                return AVERROR(EPERM);
> +
> +            dst = sample_buffers[c]->extended_data_buffer +
> +                  sample_buffers[c]->end *
> av_get_bytes_per_sample(inlink->format);
> +
> +            sample_buffers[c]->end = (sample_buffers[c]->end + 1) %
> sample_buffers[c]->capacity;
> +            sample_buffers[c]->size++;
> +
> +            memcpy(dst, frame->extended_data[c] + i *
> av_get_bytes_per_sample(inlink->format),
> +                   av_get_bytes_per_sample(inlink->format));
> +        }
> +    }
> +
> +    return 0;
> +}
> +
> +static int ring_remove(FrameRingBuffer **sample_buffers, AVFilterLink
> *inlink, uint8_t **dest, int nb_samples)
> +{
> +    uint8_t *src;
> +
> +    for (int c = 0; c < inlink->ch_layout.nb_channels; c++) {
> +        for (int i = 0; i < nb_samples; i++) {
> +            if (ring_empty(sample_buffers[c]))
> +                return AVERROR(EPERM);
> +
> +            src = sample_buffers[c]->extended_data_buffer +
> +                  sample_buffers[c]->start *
> av_get_bytes_per_sample(inlink->format);
> +
> +            sample_buffers[c]->start = (sample_buffers[c]->start + 1) %
> sample_buffers[c]->capacity;
> +            sample_buffers[c]->size--;
> +
> +            memcpy(dest[c] + i * av_get_bytes_per_sample(inlink->format),
> src,
> +                   av_get_bytes_per_sample(inlink->format));
> +        }
> +    }
> +
> +    return 0;
> +}
> +
> +static void ring_free(FrameRingBuffer *ring)
> +{
> +    av_freep(&ring->extended_data_buffer);
> +    av_freep(&ring);
> +}
> +
> +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;
> +    FrameRingBuffer **main_sample_buffers;
> +    FrameRingBuffer **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;
> +} AOverlayContext;
> +
> +#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;
> +
> +    for (int i = 0; i < s->nb_channels; i++) {
> +        ring_free(s->main_sample_buffers[i]);
> +        ring_free(s->overlay_sample_buffers[i]);
> +        av_freep(&s->cf0[i]);
> +        av_freep(&s->cf1[i]);
> +    }
> +    av_freep(&s->cf0);
> +    av_freep(&s->cf1);
> +
> +    av_freep(&s->main_sample_buffers);
> +    av_freep(&s->overlay_sample_buffers);
> +
> +    av_frame_free(&s->main_input);
> +    av_frame_free(&s->overlay_input);
> +}
> +
> +static int query_formats(AVFilterContext *ctx)
> +{
> +    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
> +    };
> +
> +    return ff_set_common_formats_from_list(ctx, sample_fmts);
>

There is now simpler way to do this, by just setting array in filter
structure, plenty of examples in other audio filters.


> +}
> +
> +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 = ring_remove(s->main_sample_buffers, main_inlink,
> (*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 = ring_remove(s->overlay_sample_buffers, overlay_inlink,
> (*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 = ring_remove(s->main_sample_buffers, main_inlink, s->cf0,
> nb_samples) < 0)
> +        return ret;
> +    if (ret = ring_remove(s->overlay_sample_buffers, overlay_inlink,
> 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,
> s->overlay_sample_buffers[0]->capacity -
> s->overlay_sample_buffers[0]->size);
> +
> +    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 = ring_insert(s->overlay_sample_buffers, s->overlay_input,
> overlay_inlink) < 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 (s->main_sample_buffers[0]->size !=
> s->main_sample_buffers[0]->capacity && !s->main_eof && !s->default_mode) {
> +        nb_samples = FFMIN(SEGMENT_SIZE,
> s->main_sample_buffers[0]->capacity - s->main_sample_buffers[0]->size);
> +
> +        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_sample_buffers[0]->size +
> s->main_input->nb_samples < s->main_sample_buffers[0]->capacity)
> +                    s->crossfade_ready = 1;
> +                if (s->main_sample_buffers[0]->size == 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 || s->main_sample_buffers[0]->size > 0)
> +                        s->transition_pts = s->pts_gap_start;
> +                    s->pts_gap_end = s->main_input->pts;
> +                    s->default_mode = 1;
> +                }
> +            }
> +
> +            if (ret = ring_insert(s->main_sample_buffers, s->main_input,
> main_inlink) < 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 && s->main_sample_buffers[0]->size == 0 &&
> ff_inlink_acknowledge_status(main_inlink, &status, &pts)) {
> +        ff_outlink_set_status(outlink, status, pts);
> +        return 0;
> +    }
> +
> +    if (s->main_sample_buffers[0]->size <
> s->main_sample_buffers[0]->capacity &&
> +        (s->transition_pts == AV_NOPTS_VALUE ||
> s->main_sample_buffers[0]->size != 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 (s->overlay_sample_buffers[0]->size <
> s->overlay_sample_buffers[0]->capacity) {
> +            ret = consume_samples(s, overlay_inlink, outlink);
> +            if (ret <= 0) {
> +                if (!s->overlay_eof)
> +                    return ret;
> +            }
> +        }
> +
> +        if (s->overlay_sample_buffers[0]->size <
> s->overlay_sample_buffers[0]->capacity) {
> +            if (ff_inlink_acknowledge_status(overlay_inlink, &status,
> &pts)) {
> +                s->overlay_eof = 1;
> +                s->transition_pts = s->pts +
> av_rescale_q(s->overlay_sample_buffers[0]->size - (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 &&
> s->main_sample_buffers[0]->size > s->cf_samples + SEGMENT_SIZE) {
> +            nb_samples = s->main_sample_buffers[0]->capacity -
> 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,
> s->main_sample_buffers[0]->size - 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(s->main_sample_buffers[0]->size,
> 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 = ring_remove(s->main_sample_buffers, main_inlink,
> 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 = ring_remove(s->overlay_sample_buffers, overlay_inlink,
> 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 && s->overlay_sample_buffers[0]->size > 0) {
> +        if (s->overlay_sample_buffers[0]->size != s->cf_samples) {
> +            nb_samples = s->overlay_sample_buffers[0]->size -
> 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 (s->overlay_sample_buffers[0]->size == 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 ? s->main_sample_buffers[0]->size :
> 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(s->main_sample_buffers[0]->size, 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 = ring_remove(s->main_sample_buffers, main_inlink,
> main_buffer->extended_data, nb_samples) < 0)
> +            return ret;
> +    }
> +
> +    if (!ff_inlink_evaluate_timeline_at_frame(main_inlink, main_buffer)
> || (s->overlay_eof && s->overlay_sample_buffers[0]->size == 0)) {
> +        return ff_filter_frame(outlink, main_buffer);
> +    } else {
> +        if (s->transition_pts == AV_NOPTS_VALUE) {
> +            nb_samples = FFMIN(s->overlay_sample_buffers[0]->size,
> SEGMENT_SIZE);
> +            overlay_buffer = ff_get_audio_buffer(outlink, nb_samples);
> +            if (!overlay_buffer)
> +                return AVERROR(ENOMEM);
> +            if (ret = ring_remove(s->overlay_sample_buffers,
> overlay_inlink, 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 ret, size, ring_buffer_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;
> +
> +    s->cf0 = av_malloc_array(s->nb_channels, sizeof(uint8_t*));
> +    s->cf1 = av_malloc_array(s->nb_channels, sizeof(uint8_t*));
> +
> +    ring_buffer_size = SEGMENT_SIZE + SEGMENT_SIZE * (1 + ((s->cf_samples
> - 1) / SEGMENT_SIZE));
> +
> +    s->main_sample_buffers = av_malloc_array(s->nb_channels,
> sizeof(FrameRingBuffer*));
> +    for (int i = 0; i < s->nb_channels; i++) {
> +        s->cf0[i] = av_malloc_array(s->cf_samples, size);
> +        ret = ring_init(&s->main_sample_buffers[i], ring_buffer_size,
> size);
> +        if (ret < 0)
> +            return ret;
> +    }
> +
> +    s->overlay_sample_buffers = av_malloc_array(s->nb_channels,
> sizeof(FrameRingBuffer*));
> +    for (int i = 0; i < s->nb_channels; i++) {
> +        s->cf1[i] = av_malloc_array(s->cf_samples, size);
> +        ret = ring_init(&s->overlay_sample_buffers[i], ring_buffer_size,
> size);
> +        if (ret < 0)
> +            return ret;
> +    }
> +
> +    return 0;
> +}
> +
> +static const AVFilterPad avfilter_af_aoverlay_inputs[] = {
> +    {
> +        .name = "main",
> +        .type = AVMEDIA_TYPE_AUDIO,
> +    },
> +    {
> +        .name = "s->overlay_input",
> +        .type = AVMEDIA_TYPE_AUDIO,
> +    },
> +};
> +
> +static const AVFilterPad avfilter_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(avfilter_af_aoverlay_inputs),
> +    FILTER_OUTPUTS(avfilter_af_aoverlay_outputs),
> +    FILTER_QUERY_FUNC(query_formats),
> +    .flags          = AVFILTER_FLAG_SUPPORT_TIMELINE_INTERNAL,
> +};
> diff --git a/libavfilter/allfilters.c b/libavfilter/allfilters.c
> index d4184d6e808..abdeb40fb49 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)
>
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