From: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
To: ffmpeg-devel@ffmpeg.org
Subject: Re: [FFmpeg-devel] [PATCH v2] GSoC 2023: Add Audio Overlay Filter
Date: Sat, 9 Sep 2023 23:04:29 +0200
Message-ID: <AS8P250MB0744BEAF42C343005CAF53428FECA@AS8P250MB0744.EURP250.PROD.OUTLOOK.COM> (raw)
In-Reply-To: <20230909151911.36202-1-karwalharshit@gmail.com>
Harshit Karwal:
> Replaced FILTER_QUERY_FUNC with FILTER_SAMPLEFMTS_ARRAY
>
> Signed-off-by: Harshit Karwal <karwalharshit@gmail.com>
> ---
> doc/filters.texi | 40 +++
> libavfilter/Makefile | 1 +
> libavfilter/af_aoverlay.c | 618 ++++++++++++++++++++++++++++++++++++++
> libavfilter/allfilters.c | 1 +
> 4 files changed, 660 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..3cd82aaf9e
> --- /dev/null
> +++ b/libavfilter/af_aoverlay.c
> @@ -0,0 +1,618 @@
> +/*
> + * 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"
You do not need this header when using FILTER_SAMPLEFMTS_ARRAY
> +
> +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));
Why do you allocate this separately (i.e. why do you use
FrameRingBuffer** instead of using FrameRingBuffer* below)? It just
makes everything more complicated.
> + 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));
Do I see this correctly? You are copying samples one sample at a time
instead of memcpy()'ing as much as you can in one go? Copying into a
ringbuffer should need two memcpy's max per channel.
> + }
> + }
> +
> + 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);
> +}
Could all this FrameRingBuffer be replaced by the audio fifo from
libavutil/audio_fifo.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;
> + 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;
> +
> +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;
> +
> + 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 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*));
You are missing quite a few checks here.
> +
> + 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*));
You need to use av_calloc() here and above
> + 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;
or otherwise uninit may try to free uninitialized pointers (it will do
so if an error happened before the last channel).
> + }
> +
> + 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[] = {
Please don't use the avfilter_ prefix, it is actually intended for stuff
that gets exported from libavfilter (yes, I know that lots of other
filters do it that way).
> + {
> + .name = "main",
> + .type = AVMEDIA_TYPE_AUDIO,
> + },
> + {
> + .name = "s->overlay_input",
The "s->" should not be part of the name (users are not supposed to
concern themselves with such things as the naming of our internal
variables).
> + .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_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;
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prev parent reply other threads:[~2023-09-09 21:03 UTC|newest]
Thread overview: 2+ messages / expand[flat|nested] mbox.gz Atom feed top
2023-09-09 15:19 Harshit Karwal
2023-09-09 21:04 ` Andreas Rheinhardt [this message]
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