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!
--
Wang Cao | Software Engineer | wang...@google.com | 650-203-7807
<(650)%20203-7807>
_______________________________________________
ffmpeg-devel mailing list
ffmpeg-devel@ffmpeg.org
https://ffmpeg.org/mailman/listinfo/ffmpeg-devel
To unsubscribe, visit link above, or email
ffmpeg-devel-requ...@ffmpeg.org with subject "unsubscribe".
