On date Tuesday 2024-01-16 17:46:42 +0530, Harshit Karwal wrote:
> Co-authored-by: Paul B Mahol <one...@gmail.com>
> Signed-off-by: Harshit Karwal <karwalhars...@gmail.com>
> ---
> doc/filters.texi | 40 +++
> libavfilter/Makefile | 1 +
> libavfilter/af_aoverlay.c | 538 ++++++++++++++++++++++++++++++++++++++
> libavfilter/allfilters.c | 1 +
> 4 files changed, 580 insertions(+)
> create mode 100644 libavfilter/af_aoverlay.c
>
> diff --git a/doc/filters.texi b/doc/filters.texi
> index 20c91bab3a..79eb600ae3 100644
> --- a/doc/filters.texi
> +++ b/doc/filters.texi
> @@ -2779,6 +2779,46 @@ 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
nit: @option{enable}
> +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:
nit: options in case we add more
> +
> +@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 bba0219876..0f2b403441 100644
> --- a/libavfilter/Makefile
> +++ b/libavfilter/Makefile
> @@ -81,6 +81,7 @@ 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..f7ac00dda1
> --- /dev/null
> +++ b/libavfilter/af_aoverlay.c
[...]
> +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)) {
it would help to use an enum to describe the mode value
Also would help to introduce some debug log messages to aid
troubleshooting/debugging.
For instance, it would be very useful to show the exact time when the
overlay stream is inserted.
[...]
> +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) {
nit: else if to avoid this evaluation in case the first block is executed
[...]
> +
> +static int config_output(AVFilterLink *outlink)
> +{
> + AVFilterContext *ctx = outlink->src;
> + AOverlayContext *s = ctx->priv;
> + int 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);
is this needed? shouldn't the duration be set also for the default
case?
[...]
Thanks
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