| Message ID | 20220505211416.671275-1-wangcao@google.com |
|---|---|
| State | New |
| Headers | show |
| Series | [FFmpeg-devel] avfilter/alimiter:add latency compensation | expand |
| Context | Check | Description |
|---|---|---|
| yinshiyou/commit_msg_loongarch64 | warning | The first line of the commit message must start with a context terminated by a colon and a space, for example "lavu/opt: " or "doc: ". |
| andriy/commit_msg_x86 | warning | The first line of the commit message must start with a context terminated by a colon and a space, for example "lavu/opt: " or "doc: ". |
| yinshiyou/make_loongarch64 | success | Make finished |
| yinshiyou/make_fate_loongarch64 | fail | Make fate failed |
| andriy/make_x86 | success | Make finished |
| andriy/make_fate_x86 | fail | Make fate failed |
On Thu, May 5, 2022 at 2:14 PM Wang Cao <wangcao@google.com> wrote: > Also added 2 FATE tests to verify delay is compenated correctly > > Signed-off-by: Wang Cao <wangcao@google.com> > --- > doc/filters.texi | 5 +++ > libavfilter/af_alimiter.c | 90 +++++++++++++++++++++++++++++++++++++ > tests/fate/filter-audio.mak | 24 ++++++++-- > 3 files changed, 116 insertions(+), 3 deletions(-) > > diff --git a/doc/filters.texi b/doc/filters.texi > index a161754233..75a43edd88 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 latency > +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..01265758d7 100644 > --- a/libavfilter/af_alimiter.c > +++ b/libavfilter/af_alimiter.c > @@ -26,6 +26,7 @@ > > #include "libavutil/channel_layout.h" > #include "libavutil/common.h" > +#include "libavutil/fifo.h" > #include "libavutil/opt.h" > > #include "audio.h" > @@ -33,6 +34,11 @@ > #include "formats.h" > #include "internal.h" > > +typedef struct MetaItem { > + int64_t pts; > + int nb_samples; > +} MetaItem; > + > typedef struct AudioLimiterContext { > const AVClass *class; > > @@ -55,6 +61,14 @@ typedef struct AudioLimiterContext { > int *nextpos; > double *nextdelta; > > + int in_trim; > + int out_pad; > + int64_t next_in_pts; > + int64_t next_out_pts; > + int latency; > + > + AVFifo *fifo; > + > double delta; > int nextiter; > int nextlen; > @@ -73,6 +87,7 @@ 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 }, > + { "latency", "compensate delay", OFFSET(latency), > AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, AF }, > { NULL } > }; > > @@ -129,6 +144,11 @@ static int filter_frame(AVFilterLink *inlink, AVFrame > *in) > AVFrame *out; > double *buf; > int n, c, i; > + int new_out_samples; > + int64_t out_duration; > + int64_t in_duration; > + int64_t in_pts; > + MetaItem meta; > > if (av_frame_is_writable(in)) { > out = in; > @@ -269,12 +289,69 @@ static int filter_frame(AVFilterLink *inlink, > AVFrame *in) > dst += channels; > } > > + in_duration = av_rescale_q(in->nb_samples, inlink->time_base, > av_make_q(1, in->sample_rate)); > + in_pts = in->pts; > + meta = (MetaItem){ in->pts, in->nb_samples }; > + av_fifo_write(s->fifo, &meta, 1); > if (in != out) > av_frame_free(&in); > > + new_out_samples = out->nb_samples; > + if (s->in_trim > 0) { > + int trim = FFMIN(new_out_samples, s->in_trim); > + new_out_samples -= trim; > + s->in_trim -= trim; > + } > + > + if (new_out_samples <= 0) { > + av_frame_free(&out); > + return 0; > + } else if (new_out_samples < out->nb_samples) { > + int offset = out->nb_samples - new_out_samples; > + memmove(out->extended_data[0], out->extended_data[0] + > sizeof(double) * offset * out->ch_layout.nb_channels, > + sizeof(double) * new_out_samples * > out->ch_layout.nb_channels); > + out->nb_samples = new_out_samples; > + s->in_trim = 0; > + } > + > + av_fifo_read(s->fifo, &meta, 1); > + > + out_duration = av_rescale_q(out->nb_samples, inlink->time_base, > av_make_q(1, out->sample_rate)); > + in_duration = av_rescale_q(meta.nb_samples, inlink->time_base, > av_make_q(1, out->sample_rate)); > + in_pts = meta.pts; > + > + if (s->next_out_pts != AV_NOPTS_VALUE && out->pts != s->next_out_pts > && > + s->next_in_pts != AV_NOPTS_VALUE && in_pts == s->next_in_pts) { > + out->pts = s->next_out_pts; > + } else { > + out->pts = in_pts; > + } > + s->next_in_pts = in_pts + in_duration; > + s->next_out_pts = out->pts + out_duration; > + > return ff_filter_frame(outlink, out); > } > > +static int request_frame(AVFilterLink* outlink) > +{ > + AVFilterContext *ctx = outlink->src; > + AudioLimiterContext *s = (AudioLimiterContext*)ctx->priv; > + int ret; > + > + ret = ff_request_frame(ctx->inputs[0]); > + > + if (ret == AVERROR_EOF && s->out_pad > 0) { > + AVFrame *frame = ff_get_audio_buffer(outlink, FFMIN(1024, > s->out_pad)); > + if (!frame) > + return AVERROR(ENOMEM); > + > + s->out_pad -= frame->nb_samples; > + frame->pts = s->next_in_pts; > + return filter_frame(ctx->inputs[0], frame); > + } > + return ret; > +} > + > static int config_input(AVFilterLink *inlink) > { > AVFilterContext *ctx = inlink->dst; > @@ -294,6 +371,16 @@ 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; > + if (s->latency) { > + s->in_trim = s->out_pad = s->buffer_size / > inlink->ch_layout.nb_channels - 1; > + } > + s->next_out_pts = AV_NOPTS_VALUE; > + s->next_in_pts = AV_NOPTS_VALUE; > + > + s->fifo = av_fifo_alloc2(8, sizeof(MetaItem), AV_FIFO_FLAG_AUTO_GROW); > + if (!s->fifo) { > + return AVERROR(ENOMEM); > + } > > if (s->buffer_size <= 0) { > av_log(ctx, AV_LOG_ERROR, "Attack is too small.\n"); > @@ -310,6 +397,8 @@ static av_cold void uninit(AVFilterContext *ctx) > av_freep(&s->buffer); > av_freep(&s->nextdelta); > av_freep(&s->nextpos); > + > + av_fifo_freep2(&s->fifo); > } > > static const AVFilterPad alimiter_inputs[] = { > @@ -325,6 +414,7 @@ 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..e33ffdf37f 100644 > --- a/tests/fate/filter-audio.mak > +++ b/tests/fate/filter-audio.mak > @@ -63,11 +63,29 @@ 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 $(SRC) -af > aresample,agate=level_in=10:range=0:threshold=1:ratio=1:attack=1:knee=1:makeup=4,aresample > > -FATE_AFILTER-$(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 = $(TARGET_PATH)/tests/data/asynth-44100-2.wav > +tests/data/filter-alimiter-passthrough: TAG = GEN > +tests/data/filter-alimiter-passthrough: ffmpeg$(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_ALIMITER += fate-filter-alimiter-passthrough-default-attack > +fate-filter-alimiter-passthrough-default-attack: > tests/data/filter-alimiter-passthrough > +fate-filter-alimiter-passthrough-default-attack: REF = > $(TARGET_PATH)/tests/data/filter-alimiter-passthrough > +fate-filter-alimiter-passthrough-default-attack: CMD = crc -i $(SRC) -af > aresample,alimiter=level_in=1:level_out=1:limit=1:level=0:latency=1,aresample > + > +FATE_ALIMITER += fate-filter-alimiter-passthrough-large-attack > +fate-filter-alimiter-passthrough-large-attack: > tests/data/filter-alimiter-passthrough > +fate-filter-alimiter-passthrough-large-attack: REF = > $(TARGET_PATH)/tests/data/filter-alimiter-passthrough > +fate-filter-alimiter-passthrough-large-attack: CMD = crc -i $(SRC) -af > aresample,alimiter=level_in=1:level_out=1:limit=1:level=0:latency=1:attack=80,aresample > + > +FATE_ALIMITER += fate-filter-alimiter > fate-filter-alimiter: CMD = framecrc -i $(SRC) -af > aresample,alimiter=level_in=1:level_out=2:limit=0.2,aresample > > +$(FATE_ALIMITER): tests/data/asynth-44100-2.wav > +$(FATE_ALIMITER): SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav > + > +FATE_AFILTER-$(call FILTERDEMDECENCMUX, ATRIM, WAV, PCM_S16LE, PCM_S16LE, > WAV) += $(FATE_ALIMITER) > + > FATE_AFILTER-$(call FILTERDEMDECENCMUX, AMERGE, WAV, PCM_S16LE, > PCM_S16LE, WAV) += fate-filter-amerge > fate-filter-amerge: tests/data/asynth-44100-1.wav > fate-filter-amerge: SRC = $(TARGET_PATH)/tests/data/asynth-44100-1.wav > -- > 2.36.0.512.ge40c2bad7a-goog > > Hi folks, I don't know how FATE failed on the server but this specific "make fate-filter-alimiter" and "make tests/data/filter-alimiter-passthrough" passed on my local machine. The code is basically referenced from af_ladspa. Can you advice on this implementation? Thanks!
On Wed, May 11, 2022 at 12:45 AM Wang Cao <wangcao-at-google.com@ffmpeg.org> wrote: > On Thu, May 5, 2022 at 2:14 PM Wang Cao <wangcao@google.com> wrote: > > > Also added 2 FATE tests to verify delay is compenated correctly > Applied with minor fixes and tests removed for now. > > > > Signed-off-by: Wang Cao <wangcao@google.com> > > --- > > doc/filters.texi | 5 +++ > > libavfilter/af_alimiter.c | 90 +++++++++++++++++++++++++++++++++++++ > > tests/fate/filter-audio.mak | 24 ++++++++-- > > 3 files changed, 116 insertions(+), 3 deletions(-) > > > > diff --git a/doc/filters.texi b/doc/filters.texi > > index a161754233..75a43edd88 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 latency > > +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..01265758d7 100644 > > --- a/libavfilter/af_alimiter.c > > +++ b/libavfilter/af_alimiter.c > > @@ -26,6 +26,7 @@ > > > > #include "libavutil/channel_layout.h" > > #include "libavutil/common.h" > > +#include "libavutil/fifo.h" > > #include "libavutil/opt.h" > > > > #include "audio.h" > > @@ -33,6 +34,11 @@ > > #include "formats.h" > > #include "internal.h" > > > > +typedef struct MetaItem { > > + int64_t pts; > > + int nb_samples; > > +} MetaItem; > > + > > typedef struct AudioLimiterContext { > > const AVClass *class; > > > > @@ -55,6 +61,14 @@ typedef struct AudioLimiterContext { > > int *nextpos; > > double *nextdelta; > > > > + int in_trim; > > + int out_pad; > > + int64_t next_in_pts; > > + int64_t next_out_pts; > > + int latency; > > + > > + AVFifo *fifo; > > + > > double delta; > > int nextiter; > > int nextlen; > > @@ -73,6 +87,7 @@ 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 }, > > + { "latency", "compensate delay", OFFSET(latency), > > AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, AF }, > > { NULL } > > }; > > > > @@ -129,6 +144,11 @@ static int filter_frame(AVFilterLink *inlink, > AVFrame > > *in) > > AVFrame *out; > > double *buf; > > int n, c, i; > > + int new_out_samples; > > + int64_t out_duration; > > + int64_t in_duration; > > + int64_t in_pts; > > + MetaItem meta; > > > > if (av_frame_is_writable(in)) { > > out = in; > > @@ -269,12 +289,69 @@ static int filter_frame(AVFilterLink *inlink, > > AVFrame *in) > > dst += channels; > > } > > > > + in_duration = av_rescale_q(in->nb_samples, inlink->time_base, > > av_make_q(1, in->sample_rate)); > > + in_pts = in->pts; > > + meta = (MetaItem){ in->pts, in->nb_samples }; > > + av_fifo_write(s->fifo, &meta, 1); > > if (in != out) > > av_frame_free(&in); > > > > + new_out_samples = out->nb_samples; > > + if (s->in_trim > 0) { > > + int trim = FFMIN(new_out_samples, s->in_trim); > > + new_out_samples -= trim; > > + s->in_trim -= trim; > > + } > > + > > + if (new_out_samples <= 0) { > > + av_frame_free(&out); > > + return 0; > > + } else if (new_out_samples < out->nb_samples) { > > + int offset = out->nb_samples - new_out_samples; > > + memmove(out->extended_data[0], out->extended_data[0] + > > sizeof(double) * offset * out->ch_layout.nb_channels, > > + sizeof(double) * new_out_samples * > > out->ch_layout.nb_channels); > > + out->nb_samples = new_out_samples; > > + s->in_trim = 0; > > + } > > + > > + av_fifo_read(s->fifo, &meta, 1); > > + > > + out_duration = av_rescale_q(out->nb_samples, inlink->time_base, > > av_make_q(1, out->sample_rate)); > > + in_duration = av_rescale_q(meta.nb_samples, inlink->time_base, > > av_make_q(1, out->sample_rate)); > > + in_pts = meta.pts; > > + > > + if (s->next_out_pts != AV_NOPTS_VALUE && out->pts != s->next_out_pts > > && > > + s->next_in_pts != AV_NOPTS_VALUE && in_pts == > s->next_in_pts) { > > + out->pts = s->next_out_pts; > > + } else { > > + out->pts = in_pts; > > + } > > + s->next_in_pts = in_pts + in_duration; > > + s->next_out_pts = out->pts + out_duration; > > + > > return ff_filter_frame(outlink, out); > > } > > > > +static int request_frame(AVFilterLink* outlink) > > +{ > > + AVFilterContext *ctx = outlink->src; > > + AudioLimiterContext *s = (AudioLimiterContext*)ctx->priv; > > + int ret; > > + > > + ret = ff_request_frame(ctx->inputs[0]); > > + > > + if (ret == AVERROR_EOF && s->out_pad > 0) { > > + AVFrame *frame = ff_get_audio_buffer(outlink, FFMIN(1024, > > s->out_pad)); > > + if (!frame) > > + return AVERROR(ENOMEM); > > + > > + s->out_pad -= frame->nb_samples; > > + frame->pts = s->next_in_pts; > > + return filter_frame(ctx->inputs[0], frame); > > + } > > + return ret; > > +} > > + > > static int config_input(AVFilterLink *inlink) > > { > > AVFilterContext *ctx = inlink->dst; > > @@ -294,6 +371,16 @@ 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; > > + if (s->latency) { > > + s->in_trim = s->out_pad = s->buffer_size / > > inlink->ch_layout.nb_channels - 1; > > + } > > + s->next_out_pts = AV_NOPTS_VALUE; > > + s->next_in_pts = AV_NOPTS_VALUE; > > + > > + s->fifo = av_fifo_alloc2(8, sizeof(MetaItem), > AV_FIFO_FLAG_AUTO_GROW); > > + if (!s->fifo) { > > + return AVERROR(ENOMEM); > > + } > > > > if (s->buffer_size <= 0) { > > av_log(ctx, AV_LOG_ERROR, "Attack is too small.\n"); > > @@ -310,6 +397,8 @@ static av_cold void uninit(AVFilterContext *ctx) > > av_freep(&s->buffer); > > av_freep(&s->nextdelta); > > av_freep(&s->nextpos); > > + > > + av_fifo_freep2(&s->fifo); > > } > > > > static const AVFilterPad alimiter_inputs[] = { > > @@ -325,6 +414,7 @@ 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..e33ffdf37f 100644 > > --- a/tests/fate/filter-audio.mak > > +++ b/tests/fate/filter-audio.mak > > @@ -63,11 +63,29 @@ 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 $(SRC) -af > > > aresample,agate=level_in=10:range=0:threshold=1:ratio=1:attack=1:knee=1:makeup=4,aresample > > > > -FATE_AFILTER-$(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 = $(TARGET_PATH)/tests/data/asynth-44100-2.wav > > +tests/data/filter-alimiter-passthrough: TAG = GEN > > +tests/data/filter-alimiter-passthrough: ffmpeg$(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_ALIMITER += fate-filter-alimiter-passthrough-default-attack > > +fate-filter-alimiter-passthrough-default-attack: > > tests/data/filter-alimiter-passthrough > > +fate-filter-alimiter-passthrough-default-attack: REF = > > $(TARGET_PATH)/tests/data/filter-alimiter-passthrough > > +fate-filter-alimiter-passthrough-default-attack: CMD = crc -i $(SRC) -af > > > aresample,alimiter=level_in=1:level_out=1:limit=1:level=0:latency=1,aresample > > + > > +FATE_ALIMITER += fate-filter-alimiter-passthrough-large-attack > > +fate-filter-alimiter-passthrough-large-attack: > > tests/data/filter-alimiter-passthrough > > +fate-filter-alimiter-passthrough-large-attack: REF = > > $(TARGET_PATH)/tests/data/filter-alimiter-passthrough > > +fate-filter-alimiter-passthrough-large-attack: CMD = crc -i $(SRC) -af > > > aresample,alimiter=level_in=1:level_out=1:limit=1:level=0:latency=1:attack=80,aresample > > + > > +FATE_ALIMITER += fate-filter-alimiter > > fate-filter-alimiter: CMD = framecrc -i $(SRC) -af > > aresample,alimiter=level_in=1:level_out=2:limit=0.2,aresample > > > > +$(FATE_ALIMITER): tests/data/asynth-44100-2.wav > > +$(FATE_ALIMITER): SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav > > + > > +FATE_AFILTER-$(call FILTERDEMDECENCMUX, ATRIM, WAV, PCM_S16LE, > PCM_S16LE, > > WAV) += $(FATE_ALIMITER) > > + > > FATE_AFILTER-$(call FILTERDEMDECENCMUX, AMERGE, WAV, PCM_S16LE, > > PCM_S16LE, WAV) += fate-filter-amerge > > fate-filter-amerge: tests/data/asynth-44100-1.wav > > fate-filter-amerge: SRC = $(TARGET_PATH)/tests/data/asynth-44100-1.wav > > -- > > 2.36.0.512.ge40c2bad7a-goog > > > > Hi folks, I don't know how FATE failed on the server but this specific > "make fate-filter-alimiter" and > "make tests/data/filter-alimiter-passthrough" passed on my local machine. > The code is basically referenced from af_ladspa. Can you advice on this > implementation? Thanks! > -- > > Wang Cao | Software Engineer | wangcao@google.com | 650-203-7807 > _______________________________________________ > ffmpeg-devel mailing list > ffmpeg-devel@ffmpeg.org > https://ffmpeg.org/mailman/listinfo/ffmpeg-devel > > To unsubscribe, visit link above, or email > ffmpeg-devel-request@ffmpeg.org with subject "unsubscribe". >
diff --git a/doc/filters.texi b/doc/filters.texi index a161754233..75a43edd88 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 latency +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..01265758d7 100644 --- a/libavfilter/af_alimiter.c +++ b/libavfilter/af_alimiter.c @@ -26,6 +26,7 @@ #include "libavutil/channel_layout.h" #include "libavutil/common.h" +#include "libavutil/fifo.h" #include "libavutil/opt.h" #include "audio.h" @@ -33,6 +34,11 @@ #include "formats.h" #include "internal.h" +typedef struct MetaItem { + int64_t pts; + int nb_samples; +} MetaItem; + typedef struct AudioLimiterContext { const AVClass *class; @@ -55,6 +61,14 @@ typedef struct AudioLimiterContext { int *nextpos; double *nextdelta; + int in_trim; + int out_pad; + int64_t next_in_pts; + int64_t next_out_pts; + int latency; + + AVFifo *fifo; + double delta; int nextiter; int nextlen; @@ -73,6 +87,7 @@ 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 }, + { "latency", "compensate delay", OFFSET(latency), AV_OPT_TYPE_BOOL, {.i64=0}, 0, 1, AF }, { NULL } }; @@ -129,6 +144,11 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in) AVFrame *out; double *buf; int n, c, i; + int new_out_samples; + int64_t out_duration; + int64_t in_duration; + int64_t in_pts; + MetaItem meta; if (av_frame_is_writable(in)) { out = in; @@ -269,12 +289,69 @@ static int filter_frame(AVFilterLink *inlink, AVFrame *in) dst += channels; } + in_duration = av_rescale_q(in->nb_samples, inlink->time_base, av_make_q(1, in->sample_rate)); + in_pts = in->pts; + meta = (MetaItem){ in->pts, in->nb_samples }; + av_fifo_write(s->fifo, &meta, 1); if (in != out) av_frame_free(&in); + new_out_samples = out->nb_samples; + if (s->in_trim > 0) { + int trim = FFMIN(new_out_samples, s->in_trim); + new_out_samples -= trim; + s->in_trim -= trim; + } + + if (new_out_samples <= 0) { + av_frame_free(&out); + return 0; + } else if (new_out_samples < out->nb_samples) { + int offset = out->nb_samples - new_out_samples; + memmove(out->extended_data[0], out->extended_data[0] + sizeof(double) * offset * out->ch_layout.nb_channels, + sizeof(double) * new_out_samples * out->ch_layout.nb_channels); + out->nb_samples = new_out_samples; + s->in_trim = 0; + } + + av_fifo_read(s->fifo, &meta, 1); + + out_duration = av_rescale_q(out->nb_samples, inlink->time_base, av_make_q(1, out->sample_rate)); + in_duration = av_rescale_q(meta.nb_samples, inlink->time_base, av_make_q(1, out->sample_rate)); + in_pts = meta.pts; + + if (s->next_out_pts != AV_NOPTS_VALUE && out->pts != s->next_out_pts && + s->next_in_pts != AV_NOPTS_VALUE && in_pts == s->next_in_pts) { + out->pts = s->next_out_pts; + } else { + out->pts = in_pts; + } + s->next_in_pts = in_pts + in_duration; + s->next_out_pts = out->pts + out_duration; + return ff_filter_frame(outlink, out); } +static int request_frame(AVFilterLink* outlink) +{ + AVFilterContext *ctx = outlink->src; + AudioLimiterContext *s = (AudioLimiterContext*)ctx->priv; + int ret; + + ret = ff_request_frame(ctx->inputs[0]); + + if (ret == AVERROR_EOF && s->out_pad > 0) { + AVFrame *frame = ff_get_audio_buffer(outlink, FFMIN(1024, s->out_pad)); + if (!frame) + return AVERROR(ENOMEM); + + s->out_pad -= frame->nb_samples; + frame->pts = s->next_in_pts; + return filter_frame(ctx->inputs[0], frame); + } + return ret; +} + static int config_input(AVFilterLink *inlink) { AVFilterContext *ctx = inlink->dst; @@ -294,6 +371,16 @@ 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; + if (s->latency) { + s->in_trim = s->out_pad = s->buffer_size / inlink->ch_layout.nb_channels - 1; + } + s->next_out_pts = AV_NOPTS_VALUE; + s->next_in_pts = AV_NOPTS_VALUE; + + s->fifo = av_fifo_alloc2(8, sizeof(MetaItem), AV_FIFO_FLAG_AUTO_GROW); + if (!s->fifo) { + return AVERROR(ENOMEM); + } if (s->buffer_size <= 0) { av_log(ctx, AV_LOG_ERROR, "Attack is too small.\n"); @@ -310,6 +397,8 @@ static av_cold void uninit(AVFilterContext *ctx) av_freep(&s->buffer); av_freep(&s->nextdelta); av_freep(&s->nextpos); + + av_fifo_freep2(&s->fifo); } static const AVFilterPad alimiter_inputs[] = { @@ -325,6 +414,7 @@ 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..e33ffdf37f 100644 --- a/tests/fate/filter-audio.mak +++ b/tests/fate/filter-audio.mak @@ -63,11 +63,29 @@ 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 $(SRC) -af aresample,agate=level_in=10:range=0:threshold=1:ratio=1:attack=1:knee=1:makeup=4,aresample -FATE_AFILTER-$(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 = $(TARGET_PATH)/tests/data/asynth-44100-2.wav +tests/data/filter-alimiter-passthrough: TAG = GEN +tests/data/filter-alimiter-passthrough: ffmpeg$(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_ALIMITER += fate-filter-alimiter-passthrough-default-attack +fate-filter-alimiter-passthrough-default-attack: tests/data/filter-alimiter-passthrough +fate-filter-alimiter-passthrough-default-attack: REF = $(TARGET_PATH)/tests/data/filter-alimiter-passthrough +fate-filter-alimiter-passthrough-default-attack: CMD = crc -i $(SRC) -af aresample,alimiter=level_in=1:level_out=1:limit=1:level=0:latency=1,aresample + +FATE_ALIMITER += fate-filter-alimiter-passthrough-large-attack +fate-filter-alimiter-passthrough-large-attack: tests/data/filter-alimiter-passthrough +fate-filter-alimiter-passthrough-large-attack: REF = $(TARGET_PATH)/tests/data/filter-alimiter-passthrough +fate-filter-alimiter-passthrough-large-attack: CMD = crc -i $(SRC) -af aresample,alimiter=level_in=1:level_out=1:limit=1:level=0:latency=1:attack=80,aresample + +FATE_ALIMITER += fate-filter-alimiter fate-filter-alimiter: CMD = framecrc -i $(SRC) -af aresample,alimiter=level_in=1:level_out=2:limit=0.2,aresample +$(FATE_ALIMITER): tests/data/asynth-44100-2.wav +$(FATE_ALIMITER): SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav + +FATE_AFILTER-$(call FILTERDEMDECENCMUX, ATRIM, WAV, PCM_S16LE, PCM_S16LE, WAV) += $(FATE_ALIMITER) + FATE_AFILTER-$(call FILTERDEMDECENCMUX, AMERGE, WAV, PCM_S16LE, PCM_S16LE, WAV) += fate-filter-amerge fate-filter-amerge: tests/data/asynth-44100-1.wav fate-filter-amerge: SRC = $(TARGET_PATH)/tests/data/asynth-44100-1.wav
Also added 2 FATE tests to verify delay is compenated correctly Signed-off-by: Wang Cao <wangcao@google.com> --- doc/filters.texi | 5 +++ libavfilter/af_alimiter.c | 90 +++++++++++++++++++++++++++++++++++++ tests/fate/filter-audio.mak | 24 ++++++++-- 3 files changed, 116 insertions(+), 3 deletions(-)