On Fri, Apr 29, 2022 at 10:49 PM Wang Cao <wangcao-at-google....@ffmpeg.org>
wrote:
> On Fri, Apr 15, 2022 at 11:50 AM Wang Cao <wang...@google.com> wrote:
>
> > 1. The option also flushes all the valid audio samples in the lookahead
> > buffer so the audio integrity is preserved. Previously the the output
> > audio will lose the amount of audio samples equal to the size of
> > lookahead buffer
> > 2. Add a FATE test to verify that when the filter is working as
> > passthrough filter, all audio samples are properly handled from the
> > input to the output.
> >
> > Signed-off-by: Wang Cao <wang...@google.com>
> > ---
> > doc/filters.texi | 5 +++
> > libavfilter/af_alimiter.c | 74 +++++++++++++++++++++++++++++++++++++
> > tests/fate/filter-audio.mak | 12 ++++++
> > 3 files changed, 91 insertions(+)
> >
> > diff --git a/doc/filters.texi b/doc/filters.texi
> > index a161754233..2af0953c89 100644
> > --- a/doc/filters.texi
> > +++ b/doc/filters.texi
> > @@ -1978,6 +1978,11 @@ in release time while 1 produces higher release
> > times.
> > @item level
> > Auto level output signal. Default is enabled.
> > This normalizes audio back to 0dB if enabled.
> > +
> > +@item comp_delay
> > +Compensate the delay introduced by using the lookahead buffer set with
> > attack
> > +parameter. Also flush the valid audio data in the lookahead buffer when
> > the
> > +stream hits EOF.
> > @end table
> >
> > Depending on picked setting it is recommended to upsample input 2x or 4x
> > times
> > diff --git a/libavfilter/af_alimiter.c b/libavfilter/af_alimiter.c
> > index 133f98f165..d10a90859b 100644
> > --- a/libavfilter/af_alimiter.c
> > +++ b/libavfilter/af_alimiter.c
> > @@ -55,6 +55,12 @@ typedef struct AudioLimiterContext {
> > int *nextpos;
> > double *nextdelta;
> >
> > + int lookahead_delay_samples;
> > + int lookahead_flush_samples;
> > + int64_t output_pts;
> > + int64_t next_output_pts;
> > + int comp_delay;
> > +
> > double delta;
> > int nextiter;
> > int nextlen;
> > @@ -73,6 +79,7 @@ static const AVOption alimiter_options[] = {
> > { "asc", "enable asc", OFFSET(auto_release),
> > AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, AF },
> > { "asc_level", "set asc level", OFFSET(asc_coeff),
> > AV_OPT_TYPE_DOUBLE, {.dbl=0.5}, 0, 1, AF },
> > { "level", "auto level", OFFSET(auto_level),
> > AV_OPT_TYPE_BOOL, {.i64=1}, 0, 1, AF },
> > + { "comp_delay","compensate delay", OFFSET(comp_delay),
> > AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, AF },
> > { NULL }
> > };
> >
> > @@ -129,6 +136,8 @@ static int filter_frame(AVFilterLink *inlink, AVFrame
> > *in)
> > AVFrame *out;
> > double *buf;
> > int n, c, i;
> > + int num_output_samples = in->nb_samples;
> > + int trim_offset;
> >
> > if (av_frame_is_writable(in)) {
> > out = in;
> > @@ -271,10 +280,71 @@ static int filter_frame(AVFilterLink *inlink,
> > AVFrame *in)
> >
> > if (in != out)
> > av_frame_free(&in);
> > +
> > + if (!s->comp_delay) {
> > + return ff_filter_frame(outlink, out);
> > + }
> > +
> > + if (s->output_pts == AV_NOPTS_VALUE) {
> > + s->output_pts = in->pts;
> > + }
> > +
> > + if (s->lookahead_delay_samples > 0) {
> > + // The current output frame is completely silence
> > + if (s->lookahead_delay_samples >= in->nb_samples) {
> > + s->lookahead_delay_samples -= in->nb_samples;
> > + return 0;
> > + }
> > +
> > + // Trim the silence part
> > + trim_offset = av_samples_get_buffer_size(
> > + NULL, inlink->ch_layout.nb_channels,
> > s->lookahead_delay_samples,
> > + (enum AVSampleFormat)out->format, 1);
> > + out->data[0] += trim_offset;
> > + out->nb_samples = in->nb_samples - s->lookahead_delay_samples;
> > + s->lookahead_delay_samples = 0;
> > + num_output_samples = out->nb_samples;
> > + }
> > +
> > + if (s->lookahead_delay_samples < 0) {
> > + return AVERROR_BUG;
> > + }
> > +
> > + out->pts = s->output_pts;
> > + s->next_output_pts = s->output_pts + num_output_samples;
> > + s->output_pts = s->next_output_pts;
> >
> > return ff_filter_frame(outlink, out);
> > }
> >
> > +static int request_frame(AVFilterLink* outlink)
> > +{
> > + AVFilterContext *ctx = outlink->src;
> > + AudioLimiterContext *s = (AudioLimiterContext*)ctx->priv;
> > + int ret;
> > + AVFilterLink *inlink;
> > + AVFrame *silence_frame;
> > +
> > + ret = ff_request_frame(ctx->inputs[0]);
> > +
> > + if (ret != AVERROR_EOF || s->lookahead_flush_samples == 0 ||
> > !s->comp_delay) {
> > + // Not necessarily an error, just not EOF.
> > + return ret;
> > + }
> > +
> > + // We reach here when input filters have finished producing data
> > (i.e. EOF),
> > + // but because of the attack param, s->buffer still has meaningful
> > + // audio content that needs flushing.
> > + inlink = ctx->inputs[0];
> > + // Pushes silence frame to flush valid audio in the s->buffer
> > + silence_frame = ff_get_audio_buffer(inlink,
> > s->lookahead_flush_samples);
> > + ret = filter_frame(inlink, silence_frame);
> > + if (ret < 0) {
> > + return ret;
> > + }
> > + return AVERROR_EOF;
> > +}
> > +
> > static int config_input(AVFilterLink *inlink)
> > {
> > AVFilterContext *ctx = inlink->dst;
> > @@ -294,6 +364,9 @@ static int config_input(AVFilterLink *inlink)
> > memset(s->nextpos, -1, obuffer_size * sizeof(*s->nextpos));
> > s->buffer_size = inlink->sample_rate * s->attack *
> > inlink->ch_layout.nb_channels;
> > s->buffer_size -= s->buffer_size % inlink->ch_layout.nb_channels;
> > + // the current logic outputs the next sample from the lookahead
> > buffer from the beginning so the amount of delay
> > + // compensation is less than the lookahead buffer size by 1 .
> > + s->lookahead_delay_samples = s->lookahead_flush_samples =
> > s->buffer_size / inlink->ch_layout.nb_channels - 1;
> >
> > if (s->buffer_size <= 0) {
> > av_log(ctx, AV_LOG_ERROR, "Attack is too small.\n");
> > @@ -325,6 +398,7 @@ static const AVFilterPad alimiter_outputs[] = {
> > {
> > .name = "default",
> > .type = AVMEDIA_TYPE_AUDIO,
> > + .request_frame = request_frame,
> > },
> > };
> >
> > diff --git a/tests/fate/filter-audio.mak b/tests/fate/filter-audio.mak
> > index eff32b9f81..3a51ca18a6 100644
> > --- a/tests/fate/filter-audio.mak
> > +++ b/tests/fate/filter-audio.mak
> > @@ -63,6 +63,18 @@ fate-filter-agate: tests/data/asynth-44100-2.wav
> > fate-filter-agate: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
> > fate-filter-agate: CMD = framecrc -i $(SRC) -af
> >
> aresample,agate=level_in=10:range=0:threshold=1:ratio=1:attack=1:knee=1:makeup=4,aresample
> >
> > +tests/data/filter-alimiter-passthrough: TAG = GEN
> > +tests/data/filter-alimiter-passthrough: ffmpeg$(PROGSSUF)$(EXESUF) |
> > tests/data
> > + $(M)$(TARGET_EXEC) $(TARGET_PATH)/$< -nostdin \
> > + -i $(TARGET_PATH)/tests/data/asynth-44100-2.wav -af aresample -f
> > crc $(TARGET_PATH)/$@ -y 2>/dev/null
> > +
> > +FATE_AFILTER-$(call FILTERDEMDECENCMUX, AFADE, WAV, PCM_S16LE,
> PCM_S16LE,
> > WAV) += fate-filter-alimiter-passthrough
> > +fate-filter-alimiter-passthrough: tests/data/asynth-44100-2.wav
> > +fate-filter-alimiter-passthrough: tests/data/filter-alimiter-passthrough
> > +fate-filter-alimiter-passthrough: SRC =
> > $(TARGET_PATH)/tests/data/asynth-44100-2.wav
> > +fate-filter-alimiter-passthrough: REF =
> > $(TARGET_PATH)/tests/data/filter-alimiter-passthrough
> > +fate-filter-alimiter-passthrough: CMD = crc -i $(SRC) -af
> >
> aresample,alimiter=level_in=1:level_out=1:limit=1:level=0:comp_delay=1,aresample
> > +
> > FATE_AFILTER-$(call FILTERDEMDECENCMUX, AFADE, WAV, PCM_S16LE,
> PCM_S16LE,
> > WAV) += fate-filter-alimiter
> > fate-filter-alimiter: tests/data/asynth-44100-2.wav
> > fate-filter-alimiter: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav
> > --
> > 2.36.0.rc0.470.gd361397f0d-goog
> >
> > Hello folks, we would really appreciate any feedback on my patch. It
> looks
> confusing to
> me that "FATE" failed on the server while the test I added passed locally.
>
> I use "make fate-filter-alimiter-passthrough" to run the test FYI. Thank
> you!
>
Still not using logic as in af_ladspa.c
Check latest version of FFmpeg source code.
Please remove excessive self-explanatory comments in patch.
I dunno how FATE test can work if non-trivial filter uses floats.
Please follow dev guidelines. And make commit message follow other commits
in repo.
> --
>
> Wang Cao | Software Engineer | wang...@google.com | 650-203-7807
> <(650)%20203-7807>
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