On Mon, May 2, 2022 at 1:24 AM Paul B Mahol <one...@gmail.com> wrote:
> 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.
>
Hi Paul, I have prepared the patch according to your suggestion. During the
development, I have also found
the issue in af_ladspa, which caused the latency compensation not to work
as intended. It's also fixed in this
patch. FATE tests pass on my local development too. Please take a look,
thank you!
--
Wang Cao | Software Engineer | wang...@google.com | 650-203-7807
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