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lavfi: replace filter_samples by filter_frame

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Based on patch by Anton Khirnov
Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
pull/8/head
Michael Niedermayer 12 years ago
parent 16af29a7a6
commit cd7febd33f
41 changed files with 134 additions and 134 deletions
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  1. 20
      doc/filter_design.txt
  2. 6
      libavfilter/af_aconvert.c
  3. 6
      libavfilter/af_amerge.c
  4. 6
      libavfilter/af_amix.c
  5. 8
      libavfilter/af_aresample.c
  6. 6
      libavfilter/af_asetnsamples.c
  7. 6
      libavfilter/af_ashowinfo.c
  8. 8
      libavfilter/af_astreamsync.c
  9. 10
      libavfilter/af_asyncts.c
  10. 8
      libavfilter/af_atempo.c
  11. 6
      libavfilter/af_channelmap.c
  12. 6
      libavfilter/af_channelsplit.c
  13. 6
      libavfilter/af_earwax.c
  14. 6
      libavfilter/af_join.c
  15. 6
      libavfilter/af_pan.c
  16. 12
      libavfilter/af_resample.c
  17. 6
      libavfilter/af_silencedetect.c
  18. 6
      libavfilter/af_volume.c
  19. 6
      libavfilter/af_volumedetect.c
  20. 4
      libavfilter/asink_anullsink.c
  21. 2
      libavfilter/asrc_aevalsrc.c
  22. 2
      libavfilter/asrc_anullsrc.c
  23. 2
      libavfilter/asrc_flite.c
  24. 22
      libavfilter/audio.c
  25. 4
      libavfilter/audio.h
  26. 8
      libavfilter/avf_concat.c
  27. 4
      libavfilter/avf_showspectrum.c
  28. 4
      libavfilter/avf_showwaves.c
  29. 2
      libavfilter/avfilter.c
  30. 8
      libavfilter/avfilter.h
  31. 2
      libavfilter/buffersink.c
  32. 2
      libavfilter/buffersrc.c
  33. 6
      libavfilter/f_ebur128.c
  34. 4
      libavfilter/f_sendcmd.c
  35. 4
      libavfilter/f_setpts.c
  36. 6
      libavfilter/f_settb.c
  37. 6
      libavfilter/fifo.c
  38. 2
      libavfilter/internal.h
  39. 22
      libavfilter/sink_buffer.c
  40. 6
      libavfilter/split.c
  41. 2
      libavfilter/src_movie.c

20
doc/filter_design.txt
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@ -52,7 +52,7 @@ Buffer references ownership and permissions
point to only a part of a video buffer.
A reference is usually obtained as input to the start_frame or
filter_samples method or requested using the ff_get_video_buffer or
filter_frame method or requested using the ff_get_video_buffer or
ff_get_audio_buffer functions. A new reference on an existing buffer can
be created with the avfilter_ref_buffer. A reference is destroyed using
the avfilter_unref_bufferp function.
@ -68,14 +68,14 @@ Buffer references ownership and permissions
Here are the (fairly obvious) rules for reference ownership:
* A reference received by the start_frame or filter_samples method
* A reference received by the start_frame or filter_frame method
belong to the corresponding filter.
Special exception: for video references: the reference may be used
internally for automatic copying and must not be destroyed before
end_frame; it can be given away to ff_start_frame.
* A reference passed to ff_start_frame or ff_filter_samples is given
* A reference passed to ff_start_frame or ff_filter_frame is given
away and must no longer be used.
* A reference created with avfilter_ref_buffer belongs to the code that
@ -93,16 +93,16 @@ Buffer references ownership and permissions
The AVFilterLink structure has a few AVFilterBufferRef fields. Here are
the rules to handle them:
* cur_buf is set before the start_frame and filter_samples methods to
* cur_buf is set before the start_frame and filter_frame methods to
the same reference given as argument to the methods and belongs to the
destination filter of the link. If it has not been cleared after
end_frame or filter_samples, libavfilter will automatically destroy
end_frame or filter_frame, libavfilter will automatically destroy
the reference; therefore, any filter that needs to keep the reference
for longer must set cur_buf to NULL.
* out_buf belongs to the source filter of the link and can be used to
store a reference to the buffer that has been sent to the destination.
If it is not NULL after end_frame or filter_samples, libavfilter will
If it is not NULL after end_frame or filter_frame, libavfilter will
automatically destroy the reference.
If a video input pad does not have a start_frame method, the default
@ -179,7 +179,7 @@ Buffer references ownership and permissions
with the WRITE permission.
* Filters that intend to keep a reference after the filtering process
is finished (after end_frame or filter_samples returns) must have the
is finished (after end_frame or filter_frame returns) must have the
PRESERVE permission on it and remove the WRITE permission if they
create a new reference to give it away.
@ -198,7 +198,7 @@ Frame scheduling
Simple filters that output one frame for each input frame should not have
to worry about it.
start_frame / filter_samples
start_frame / filter_frame
----------------------------
These methods are called when a frame is pushed to the filter's input.
@ -233,7 +233,7 @@ Frame scheduling
This method is called when a frame is wanted on an output.
For an input, it should directly call start_frame or filter_samples on
For an input, it should directly call start_frame or filter_frame on
the corresponding output.
For a filter, if there are queued frames already ready, one of these
@ -266,4 +266,4 @@ Frame scheduling
Note that, except for filters that can have queued frames, request_frame
does not push frames: it requests them to its input, and as a reaction,
the start_frame / filter_samples method will be called and do the work.
the start_frame / filter_frame method will be called and do the work.

6
libavfilter/af_aconvert.c
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@ -135,7 +135,7 @@ static int config_output(AVFilterLink *outlink)
return 0;
}
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
{
AConvertContext *aconvert = inlink->dst->priv;
const int n = insamplesref->audio->nb_samples;
@ -149,7 +149,7 @@ static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref
avfilter_copy_buffer_ref_props(outsamplesref, insamplesref);
outsamplesref->audio->channel_layout = outlink->channel_layout;
ret = ff_filter_samples(outlink, outsamplesref);
ret = ff_filter_frame(outlink, outsamplesref);
avfilter_unref_buffer(insamplesref);
return ret;
}
@ -164,7 +164,7 @@ AVFilter avfilter_af_aconvert = {
.inputs = (const AVFilterPad[]) {{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples,
.filter_frame = filter_frame,
.min_perms = AV_PERM_READ, },
{ .name = NULL}},
.outputs = (const AVFilterPad[]) {{ .name = "default",

6
libavfilter/af_amerge.c
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@ -217,7 +217,7 @@ static inline void copy_samples(int nb_inputs, struct amerge_input in[],
}
}
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
AVFilterContext *ctx = inlink->dst;
AMergeContext *am = ctx->priv;
@ -290,7 +290,7 @@ static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
}
}
}
return ff_filter_samples(ctx->outputs[0], outbuf);
return ff_filter_frame(ctx->outputs[0], outbuf);
}
static av_cold int init(AVFilterContext *ctx, const char *args)
@ -313,7 +313,7 @@ static av_cold int init(AVFilterContext *ctx, const char *args)
AVFilterPad pad = {
.name = name,
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples,
.filter_frame = filter_frame,
.min_perms = AV_PERM_READ | AV_PERM_PRESERVE,
};
if (!name)

6
libavfilter/af_amix.c
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@ -309,7 +309,7 @@ static int output_frame(AVFilterLink *outlink, int nb_samples)
if (s->next_pts != AV_NOPTS_VALUE)
s->next_pts += nb_samples;
return ff_filter_samples(outlink, out_buf);
return ff_filter_frame(outlink, out_buf);
}
/**
@ -450,7 +450,7 @@ static int request_frame(AVFilterLink *outlink)
return output_frame(outlink, available_samples);
}
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
{
AVFilterContext *ctx = inlink->dst;
MixContext *s = ctx->priv;
@ -502,7 +502,7 @@ static int init(AVFilterContext *ctx, const char *args)
snprintf(name, sizeof(name), "input%d", i);
pad.type = AVMEDIA_TYPE_AUDIO;
pad.name = av_strdup(name);
pad.filter_samples = filter_samples;
pad.filter_frame = filter_frame;
ff_insert_inpad(ctx, i, &pad);
}

8
libavfilter/af_aresample.c
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@ -170,7 +170,7 @@ static int config_output(AVFilterLink *outlink)
return 0;
}
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
{
AResampleContext *aresample = inlink->dst->priv;
const int n_in = insamplesref->audio->nb_samples;
@ -205,7 +205,7 @@ static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
outsamplesref->audio->nb_samples = n_out;
ret = ff_filter_samples(outlink, outsamplesref);
ret = ff_filter_frame(outlink, outsamplesref);
aresample->req_fullfilled= 1;
avfilter_unref_buffer(insamplesref);
return ret;
@ -247,7 +247,7 @@ static int request_frame(AVFilterLink *outlink)
outsamplesref->pts = ROUNDED_DIV(outsamplesref->pts, inlink->sample_rate);
#endif
ff_filter_samples(outlink, outsamplesref);
ff_filter_frame(outlink, outsamplesref);
return 0;
}
return ret;
@ -263,7 +263,7 @@ AVFilter avfilter_af_aresample = {
.inputs = (const AVFilterPad[]) {{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples,
.filter_frame = filter_frame,
.min_perms = AV_PERM_READ, },
{ .name = NULL}},
.outputs = (const AVFilterPad[]) {{ .name = "default",

6
libavfilter/af_asetnsamples.c
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@ -125,12 +125,12 @@ static int push_samples(AVFilterLink *outlink)
if (asns->next_out_pts != AV_NOPTS_VALUE)
asns->next_out_pts += nb_out_samples;
ff_filter_samples(outlink, outsamples);
ff_filter_frame(outlink, outsamples);
asns->req_fullfilled = 1;
return nb_out_samples;
}
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
AVFilterContext *ctx = inlink->dst;
ASNSContext *asns = ctx->priv;
@ -186,7 +186,7 @@ AVFilter avfilter_af_asetnsamples = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples,
.filter_frame = filter_frame,
.min_perms = AV_PERM_READ|AV_PERM_WRITE
},
{ .name = NULL }

6
libavfilter/af_ashowinfo.c
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@ -54,7 +54,7 @@ static void uninit(AVFilterContext *ctx)
av_freep(&s->plane_checksums);
}
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
{
AVFilterContext *ctx = inlink->dst;
AShowInfoContext *s = ctx->priv;
@ -100,7 +100,7 @@ static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
av_log(ctx, AV_LOG_INFO, "]\n");
s->frame++;
return ff_filter_samples(inlink->dst->outputs[0], buf);
return ff_filter_frame(inlink->dst->outputs[0], buf);
}
static const AVFilterPad inputs[] = {
@ -108,7 +108,7 @@ static const AVFilterPad inputs[] = {
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.get_audio_buffer = ff_null_get_audio_buffer,
.filter_samples = filter_samples,
.filter_frame = filter_frame,
.min_perms = AV_PERM_READ,
},
{ NULL },

8
libavfilter/af_astreamsync.c
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@ -122,7 +122,7 @@ static int send_out(AVFilterContext *ctx, int out_id)
av_q2d(ctx->outputs[out_id]->time_base) * buf->pts;
as->var_values[VAR_T1 + out_id] += buf->audio->nb_samples /
(double)ctx->inputs[out_id]->sample_rate;
ret = ff_filter_samples(ctx->outputs[out_id], buf);
ret = ff_filter_frame(ctx->outputs[out_id], buf);
queue->nb--;
queue->tail = (queue->tail + 1) % QUEUE_SIZE;
if (as->req[out_id])
@ -167,7 +167,7 @@ static int request_frame(AVFilterLink *outlink)
return 0;
}
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
AVFilterContext *ctx = inlink->dst;
AStreamSyncContext *as = ctx->priv;
@ -191,11 +191,11 @@ AVFilter avfilter_af_astreamsync = {
.inputs = (const AVFilterPad[]) {
{ .name = "in1",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples,
.filter_frame = filter_frame,
.min_perms = AV_PERM_READ | AV_PERM_PRESERVE, },
{ .name = "in2",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples,
.filter_frame = filter_frame,
.min_perms = AV_PERM_READ | AV_PERM_PRESERVE, },
{ .name = NULL }
},

10
libavfilter/af_asyncts.c
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@ -39,7 +39,7 @@ typedef struct ASyncContext {
float min_delta_sec;
int max_comp;
/* set by filter_samples() to signal an output frame to request_frame() */
/* set by filter_frame() to signal an output frame to request_frame() */
int got_output;
} ASyncContext;
@ -135,7 +135,7 @@ static int request_frame(AVFilterLink *link)
}
buf->pts = s->pts;
return ff_filter_samples(link, buf);
return ff_filter_frame(link, buf);
}
return ret;
@ -155,7 +155,7 @@ static int64_t get_delay(ASyncContext *s)
return avresample_available(s->avr) + avresample_get_delay(s->avr);
}
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
{
AVFilterContext *ctx = inlink->dst;
ASyncContext *s = ctx->priv;
@ -211,7 +211,7 @@ static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
av_samples_set_silence(buf_out->extended_data, out_size - delta,
delta, nb_channels, buf->format);
}
ret = ff_filter_samples(outlink, buf_out);
ret = ff_filter_frame(outlink, buf_out);
if (ret < 0)
goto fail;
s->got_output = 1;
@ -237,7 +237,7 @@ static const AVFilterPad avfilter_af_asyncts_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples
.filter_frame = filter_frame
},
{ NULL }
};

8
libavfilter/af_atempo.c
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@ -138,7 +138,7 @@ typedef struct {
RDFTContext *complex_to_real;
FFTSample *correlation;
// for managing AVFilterPad.request_frame and AVFilterPad.filter_samples
// for managing AVFilterPad.request_frame and AVFilterPad.filter_frame
int request_fulfilled;
AVFilterBufferRef *dst_buffer;
uint8_t *dst;
@ -1033,7 +1033,7 @@ static void push_samples(ATempoContext *atempo,
(AVRational){ 1, outlink->sample_rate },
outlink->time_base);
ff_filter_samples(outlink, atempo->dst_buffer);
ff_filter_frame(outlink, atempo->dst_buffer);
atempo->dst_buffer = NULL;
atempo->dst = NULL;
atempo->dst_end = NULL;
@ -1041,7 +1041,7 @@ static void push_samples(ATempoContext *atempo,
atempo->nsamples_out += n_out;
}
static int filter_samples(AVFilterLink *inlink,
static int filter_frame(AVFilterLink *inlink,
AVFilterBufferRef *src_buffer)
{
AVFilterContext *ctx = inlink->dst;
@ -1148,7 +1148,7 @@ AVFilter avfilter_af_atempo = {
.inputs = (const AVFilterPad[]) {
{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples,
.filter_frame = filter_frame,
.config_props = config_props,
.min_perms = AV_PERM_READ, },
{ .name = NULL}

6
libavfilter/af_channelmap.c
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@ -312,7 +312,7 @@ static int channelmap_query_formats(AVFilterContext *ctx)
return 0;
}
static int channelmap_filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
static int channelmap_filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
@ -354,7 +354,7 @@ static int channelmap_filter_samples(AVFilterLink *inlink, AVFilterBufferRef *bu
memcpy(buf->data, buf->extended_data,
FFMIN(FF_ARRAY_ELEMS(buf->data), nch_out) * sizeof(buf->data[0]));
return ff_filter_samples(outlink, buf);
return ff_filter_frame(outlink, buf);
}
static int channelmap_config_input(AVFilterLink *inlink)
@ -389,7 +389,7 @@ static const AVFilterPad avfilter_af_channelmap_inputs[] = {
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.min_perms = AV_PERM_READ | AV_PERM_WRITE,
.filter_samples = channelmap_filter_samples,
.filter_frame = channelmap_filter_frame,
.config_props = channelmap_config_input
},
{ NULL }

6
libavfilter/af_channelsplit.c
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@ -105,7 +105,7 @@ static int query_formats(AVFilterContext *ctx)
return 0;
}
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
{
AVFilterContext *ctx = inlink->dst;
int i, ret = 0;
@ -122,7 +122,7 @@ static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
buf_out->audio->channel_layout =
av_channel_layout_extract_channel(buf->audio->channel_layout, i);
ret = ff_filter_samples(ctx->outputs[i], buf_out);
ret = ff_filter_frame(ctx->outputs[i], buf_out);
if (ret < 0)
break;
}
@ -134,7 +134,7 @@ static const AVFilterPad avfilter_af_channelsplit_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples,
.filter_frame = filter_frame,
},
{ NULL }
};

6
libavfilter/af_earwax.c
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@ -120,7 +120,7 @@ static inline int16_t *scalarproduct(const int16_t *in, const int16_t *endin, in
return out;
}
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
AVFilterLink *outlink = inlink->dst->outputs[0];
int16_t *taps, *endin, *in, *out;
@ -148,7 +148,7 @@ static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
// save part of input for next round
memcpy(taps, endin, NUMTAPS * sizeof(*taps));
ret = ff_filter_samples(outlink, outsamples);
ret = ff_filter_frame(outlink, outsamples);
avfilter_unref_buffer(insamples);
return ret;
}
@ -160,7 +160,7 @@ AVFilter avfilter_af_earwax = {
.priv_size = sizeof(EarwaxContext),
.inputs = (const AVFilterPad[]) {{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples,
.filter_frame = filter_frame,
.config_props = config_input,
.min_perms = AV_PERM_READ, },
{ .name = NULL}},

6
libavfilter/af_join.c
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@ -94,7 +94,7 @@ static const AVClass join_class = {
.version = LIBAVUTIL_VERSION_INT,
};
static int filter_samples(AVFilterLink *link, AVFilterBufferRef *buf)
static int filter_frame(AVFilterLink *link, AVFilterBufferRef *buf)
{
AVFilterContext *ctx = link->dst;
JoinContext *s = ctx->priv;
@ -229,7 +229,7 @@ static int join_init(AVFilterContext *ctx, const char *args)
snprintf(name, sizeof(name), "input%d", i);
pad.type = AVMEDIA_TYPE_AUDIO;
pad.name = av_strdup(name);
pad.filter_samples = filter_samples;
pad.filter_frame = filter_frame;
pad.needs_fifo = 1;
@ -470,7 +470,7 @@ static int join_request_frame(AVFilterLink *outlink)
priv->nb_in_buffers = ctx->nb_inputs;
buf->buf->priv = priv;
ret = ff_filter_samples(outlink, buf);
ret = ff_filter_frame(outlink, buf);
memset(s->input_frames, 0, sizeof(*s->input_frames) * ctx->nb_inputs);

6
libavfilter/af_pan.c
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@ -353,7 +353,7 @@ static int config_props(AVFilterLink *link)
return 0;
}
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
int ret;
int n = insamples->audio->nb_samples;
@ -365,7 +365,7 @@ static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
avfilter_copy_buffer_ref_props(outsamples, insamples);
outsamples->audio->channel_layout = outlink->channel_layout;
ret = ff_filter_samples(outlink, outsamples);
ret = ff_filter_frame(outlink, outsamples);
avfilter_unref_buffer(insamples);
return ret;
}
@ -388,7 +388,7 @@ AVFilter avfilter_af_pan = {
{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_props,
.filter_samples = filter_samples,
.filter_frame = filter_frame,
.min_perms = AV_PERM_READ, },
{ .name = NULL}
},

12
libavfilter/af_resample.c
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@ -40,7 +40,7 @@ typedef struct ResampleContext {
int64_t next_pts;
/* set by filter_samples() to signal an output frame to request_frame() */
/* set by filter_frame() to signal an output frame to request_frame() */
int got_output;
} ResampleContext;
@ -162,12 +162,12 @@ static int request_frame(AVFilterLink *outlink)
}
buf->pts = s->next_pts;
return ff_filter_samples(outlink, buf);
return ff_filter_frame(outlink, buf);
}
return ret;
}
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
{
AVFilterContext *ctx = inlink->dst;
ResampleContext *s = ctx->priv;
@ -224,7 +224,7 @@ static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
s->next_pts = buf_out->pts + buf_out->audio->nb_samples;
ret = ff_filter_samples(outlink, buf_out);
ret = ff_filter_frame(outlink, buf_out);
s->got_output = 1;
}
@ -232,7 +232,7 @@ fail:
avfilter_unref_buffer(buf);
} else {
buf->format = outlink->format;
ret = ff_filter_samples(outlink, buf);
ret = ff_filter_frame(outlink, buf);
s->got_output = 1;
}
@ -243,7 +243,7 @@ static const AVFilterPad avfilter_af_resample_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples,
.filter_frame = filter_frame,
.min_perms = AV_PERM_READ
},
{ NULL }

6
libavfilter/af_silencedetect.c
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@ -84,7 +84,7 @@ static char *get_metadata_val(AVFilterBufferRef *insamples, const char *key)
return e && e->value ? e->value : NULL;
}
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
int i;
SilenceDetectContext *silence = inlink->dst->priv;
@ -132,7 +132,7 @@ static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
}
}
return ff_filter_samples(inlink->dst->outputs[0], insamples);
return ff_filter_frame(inlink->dst->outputs[0], insamples);
}
static int query_formats(AVFilterContext *ctx)
@ -173,7 +173,7 @@ AVFilter avfilter_af_silencedetect = {
{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.get_audio_buffer = ff_null_get_audio_buffer,
.filter_samples = filter_samples, },
.filter_frame = filter_frame, },
{ .name = NULL }
},
.outputs = (const AVFilterPad[]) {

6
libavfilter/af_volume.c
Unescape View File

@ -110,7 +110,7 @@ static int query_formats(AVFilterContext *ctx)
return 0;
}
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
VolumeContext *vol = inlink->dst->priv;
AVFilterLink *outlink = inlink->dst->outputs[0];
@ -169,7 +169,7 @@ static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
}
}
}
return ff_filter_samples(outlink, insamples);
return ff_filter_frame(outlink, insamples);
}
AVFilter avfilter_af_volume = {
@ -181,7 +181,7 @@ AVFilter avfilter_af_volume = {
.inputs = (const AVFilterPad[]) {{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples,
.filter_frame = filter_frame,
.min_perms = AV_PERM_READ|AV_PERM_WRITE},
{ .name = NULL}},

6
libavfilter/af_volumedetect.c
Unescape View File

@ -49,7 +49,7 @@ static int query_formats(AVFilterContext *ctx)
return 0;
}
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samples)
static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *samples)
{
AVFilterContext *ctx = inlink->dst;
VolDetectContext *vd = ctx->priv;
@ -70,7 +70,7 @@ static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samples)
vd->histogram[pcm[i] + 0x8000]++;
}
return ff_filter_samples(inlink->dst->outputs[0], samples);
return ff_filter_frame(inlink->dst->outputs[0], samples);
}
#define MAX_DB 91
@ -143,7 +143,7 @@ AVFilter avfilter_af_volumedetect = {
{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.get_audio_buffer = ff_null_get_audio_buffer,
.filter_samples = filter_samples,
.filter_frame = filter_frame,
.min_perms = AV_PERM_READ, },
{ .name = NULL }
},

4
libavfilter/asink_anullsink.c
Unescape View File

@ -22,7 +22,7 @@
#include "avfilter.h"
#include "internal.h"
static int null_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
static int null_filter_frame(AVFilterLink *link, AVFilterBufferRef *samplesref)
{
avfilter_unref_bufferp(&samplesref);
return 0;
@ -32,7 +32,7 @@ static const AVFilterPad avfilter_asink_anullsink_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = null_filter_samples,
.filter_frame = null_filter_frame,
},
{ NULL },
};

2
libavfilter/asrc_aevalsrc.c
Unescape View File

@ -237,7 +237,7 @@ static int request_frame(AVFilterLink *outlink)
samplesref->audio->sample_rate = eval->sample_rate;
eval->pts += eval->nb_samples;
ff_filter_samples(outlink, samplesref);
ff_filter_frame(outlink, samplesref);
return 0;
}

2
libavfilter/asrc_anullsrc.c
Unescape View File

@ -111,7 +111,7 @@ static int request_frame(AVFilterLink *outlink)
samplesref->audio->channel_layout = null->channel_layout;
samplesref->audio->sample_rate = outlink->sample_rate;
ff_filter_samples(outlink, avfilter_ref_buffer(samplesref, ~0));
ff_filter_frame(outlink, avfilter_ref_buffer(samplesref, ~0));
avfilter_unref_buffer(samplesref);
null->pts += null->nb_samples;

2
libavfilter/asrc_flite.c
Unescape View File

@ -265,7 +265,7 @@ static int request_frame(AVFilterLink *outlink)
flite->wave_samples += nb_samples * flite->wave->num_channels;
flite->wave_nb_samples -= nb_samples;
return ff_filter_samples(outlink, samplesref);
return ff_filter_frame(outlink, samplesref);
}
AVFilter avfilter_asrc_flite = {

22
libavfilter/audio.c
Unescape View File

@ -157,30 +157,30 @@ fail:
return NULL;
}
static int default_filter_samples(AVFilterLink *link,
static int default_filter_frame(AVFilterLink *link,
AVFilterBufferRef *samplesref)
{
return ff_filter_samples(link->dst->outputs[0], samplesref);
return ff_filter_frame(link->dst->outputs[0], samplesref);
}
int ff_filter_samples_framed(AVFilterLink *link, AVFilterBufferRef *samplesref)
int ff_filter_frame_framed(AVFilterLink *link, AVFilterBufferRef *samplesref)
{
int (*filter_samples)(AVFilterLink *, AVFilterBufferRef *);
int (*filter_frame)(AVFilterLink *, AVFilterBufferRef *);
AVFilterPad *src = link->srcpad;
AVFilterPad *dst = link->dstpad;
int64_t pts;
AVFilterBufferRef *buf_out;
int ret;
FF_TPRINTF_START(NULL, filter_samples); ff_tlog_link(NULL, link, 1);
FF_TPRINTF_START(NULL, filter_frame); ff_tlog_link(NULL, link, 1);
if (link->closed) {
avfilter_unref_buffer(samplesref);
return AVERROR_EOF;
}
if (!(filter_samples = dst->filter_samples))
filter_samples = default_filter_samples;
if (!(filter_frame = dst->filter_frame))
filter_frame = default_filter_frame;
av_assert1((samplesref->perms & src->min_perms) == src->min_perms);
samplesref->perms &= ~ src->rej_perms;
@ -213,12 +213,12 @@ int ff_filter_samples_framed(AVFilterLink *link, AVFilterBufferRef *samplesref)
link->cur_buf = buf_out;
pts = buf_out->pts;
ret = filter_samples(link, buf_out);
ret = filter_frame(link, buf_out);
ff_update_link_current_pts(link, pts);
return ret;
}
int ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
int ff_filter_frame(AVFilterLink *link, AVFilterBufferRef *samplesref)
{
int insamples = samplesref->audio->nb_samples, inpos = 0, nb_samples;
AVFilterBufferRef *pbuf = link->partial_buf;
@ -232,7 +232,7 @@ int ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
if (!link->min_samples ||
(!pbuf &&
insamples >= link->min_samples && insamples <= link->max_samples)) {
return ff_filter_samples_framed(link, samplesref);
return ff_filter_frame_framed(link, samplesref);
}
/* Handle framing (min_samples, max_samples) */
while (insamples) {
@ -259,7 +259,7 @@ int ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
insamples -= nb_samples;
pbuf->audio->nb_samples += nb_samples;
if (pbuf->audio->nb_samples >= link->min_samples) {
ret = ff_filter_samples_framed(link, pbuf);
ret = ff_filter_frame_framed(link, pbuf);
pbuf = NULL;
}
}

4
libavfilter/audio.h
Unescape View File

@ -74,13 +74,13 @@ AVFilterBufferRef *ff_get_audio_buffer(AVFilterLink *link, int perms,
* @return >= 0 on success, a negative AVERROR on error. The receiving filter
* is responsible for unreferencing samplesref in case of error.
*/
int ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref);
int ff_filter_frame(AVFilterLink *link, AVFilterBufferRef *samplesref);
/**
* Send a buffer of audio samples to the next link, without checking
* min_samples.
*/
int ff_filter_samples_framed(AVFilterLink *link,
int ff_filter_frame_framed(AVFilterLink *link,
AVFilterBufferRef *samplesref);
#endif /* AVFILTER_AUDIO_H */

8
libavfilter/avf_concat.c
Unescape View File

@ -185,7 +185,7 @@ static void push_frame(AVFilterContext *ctx, unsigned in_no,
ff_end_frame(outlink);
break;
case AVMEDIA_TYPE_AUDIO:
ff_filter_samples(outlink, buf);
ff_filter_frame(outlink, buf);
break;
}
}
@ -244,7 +244,7 @@ static int end_frame(AVFilterLink *inlink)
return 0;
}
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
{
process_frame(inlink, buf);
return 0; /* enhancement: handle error return */
@ -297,7 +297,7 @@ static void send_silence(AVFilterContext *ctx, unsigned in_no, unsigned out_no)
av_samples_set_silence(buf->extended_data, 0, frame_nb_samples,
nb_channels, outlink->format);
buf->pts = base_pts + av_rescale_q(sent, rate_tb, outlink->time_base);
ff_filter_samples(outlink, buf);
ff_filter_frame(outlink, buf);
sent += frame_nb_samples;
nb_samples -= frame_nb_samples;
}
@ -397,7 +397,7 @@ static av_cold int init(AVFilterContext *ctx, const char *args)
pad.draw_slice = draw_slice;
pad.end_frame = end_frame;
} else {
pad.filter_samples = filter_samples;
pad.filter_frame = filter_frame;
}
ff_insert_inpad(ctx, ctx->nb_inputs, &pad);
}

4
libavfilter/avf_showspectrum.c
Unescape View File

@ -281,7 +281,7 @@ static int plot_spectrum_column(AVFilterLink *inlink, AVFilterBufferRef *insampl
return add_samples;
}
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
AVFilterContext *ctx = inlink->dst;
ShowSpectrumContext *showspectrum = ctx->priv;
@ -310,7 +310,7 @@ AVFilter avfilter_avf_showspectrum = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples,
.filter_frame = filter_frame,
.min_perms = AV_PERM_READ,
},
{ .name = NULL }

4
libavfilter/avf_showwaves.c
Unescape View File

@ -179,7 +179,7 @@ static int request_frame(AVFilterLink *outlink)
#define MAX_INT16 ((1<<15) -1)
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
@ -240,7 +240,7 @@ AVFilter avfilter_avf_showwaves = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples,
.filter_frame = filter_frame,
.min_perms = AV_PERM_READ,
},
{ .name = NULL }

2
libavfilter/avfilter.c
Unescape View File

@ -343,7 +343,7 @@ int ff_request_frame(AVFilterLink *link)
if (ret == AVERROR_EOF && link->partial_buf) {
AVFilterBufferRef *pbuf = link->partial_buf;
link->partial_buf = NULL;
ff_filter_samples_framed(link, pbuf);
ff_filter_frame_framed(link, pbuf);
return 0;
}
if (ret == AVERROR_EOF)

8
libavfilter/avfilter.h
Unescape View File

@ -339,7 +339,7 @@ struct AVFilterPad {
* must ensure that samplesref is properly unreferenced on error if it
* hasn't been passed on to another filter.
*/
int (*filter_samples)(AVFilterLink *link, AVFilterBufferRef *samplesref);
int (*filter_frame)(AVFilterLink *link, AVFilterBufferRef *samplesref);
/**
* Frame poll callback. This returns the number of immediately available
@ -678,7 +678,7 @@ struct AVFilterLink {
int partial_buf_size;
/**
* Minimum number of samples to filter at once. If filter_samples() is
* Minimum number of samples to filter at once. If filter_frame() is
* called with fewer samples, it will accumulate them in partial_buf.
* This field and the related ones must not be changed after filtering
* has started.
@ -687,7 +687,7 @@ struct AVFilterLink {
int min_samples;
/**
* Maximum number of samples to filter at once. If filter_samples() is
* Maximum number of samples to filter at once. If filter_frame() is
* called with more samples, it will split them.
*/
int max_samples;
@ -703,7 +703,7 @@ struct AVFilterLink {
/**
* True if the link is closed.
* If set, all attemps of start_frame, filter_samples or request_frame
* If set, all attemps of start_frame, filter_frame or request_frame
* will fail with AVERROR_EOF, and if necessary the reference will be
* destroyed.
* If request_frame returns AVERROR_EOF, this flag is set on the

2
libavfilter/buffersink.c
Unescape View File

@ -169,7 +169,7 @@ static const AVFilterPad avfilter_asink_abuffer_inputs[] = {
{
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = start_frame,
.filter_frame = start_frame,
.min_perms = AV_PERM_READ,
.needs_fifo = 1
},

2
libavfilter/buffersrc.c
Unescape View File

@ -379,7 +379,7 @@ static int request_frame(AVFilterLink *link)
return ret;
break;
case AVMEDIA_TYPE_AUDIO:
ret = ff_filter_samples(link, buf);
ret = ff_filter_frame(link, buf);
break;
default:
avfilter_unref_bufferp(&buf);

6
libavfilter/f_ebur128.c
Unescape View File

@ -436,7 +436,7 @@ static int gate_update(struct integrator *integ, double power,
return gate_hist_pos;
}
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
int i, ch;
AVFilterContext *ctx = inlink->dst;
@ -638,7 +638,7 @@ static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
}
}
return ff_filter_samples(ctx->outputs[ebur128->do_video], insamples);
return ff_filter_frame(ctx->outputs[ebur128->do_video], insamples);
}
static int query_formats(AVFilterContext *ctx)
@ -740,7 +740,7 @@ AVFilter avfilter_af_ebur128 = {
{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.get_audio_buffer = ff_null_get_audio_buffer,
.filter_samples = filter_samples, },
.filter_frame = filter_frame, },
{ .name = NULL }
},
.outputs = NULL,

4
libavfilter/f_sendcmd.c
Unescape View File

@ -511,7 +511,7 @@ end:
switch (inlink->type) {
case AVMEDIA_TYPE_VIDEO: return ff_start_frame (inlink->dst->outputs[0], ref);
case AVMEDIA_TYPE_AUDIO: return ff_filter_samples(inlink->dst->outputs[0], ref);
case AVMEDIA_TYPE_AUDIO: return ff_filter_frame(inlink->dst->outputs[0], ref);
}
return AVERROR(ENOSYS);
}
@ -562,7 +562,7 @@ AVFilter avfilter_af_asendcmd = {
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.get_audio_buffer = ff_null_get_audio_buffer,
.filter_samples = process_frame,
.filter_frame = process_frame,
},
{ .name = NULL }
},

4
libavfilter/f_setpts.c
Unescape View File

@ -174,7 +174,7 @@ static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *inpicref)
setpts->var_values[VAR_N] += 1.0;
if (setpts->type == AVMEDIA_TYPE_AUDIO) {
setpts->var_values[VAR_NB_CONSUMED_SAMPLES] += inpicref->audio->nb_samples;
return ff_filter_samples(inlink->dst->outputs[0], outpicref);
return ff_filter_frame(inlink->dst->outputs[0], outpicref);
} else
return ff_start_frame (inlink->dst->outputs[0], outpicref);
}
@ -201,7 +201,7 @@ AVFilter avfilter_af_asetpts = {
.type = AVMEDIA_TYPE_AUDIO,
.get_audio_buffer = ff_null_get_audio_buffer,
.config_props = config_input,
.filter_samples = filter_frame,
.filter_frame = filter_frame,
},
{ .name = NULL }
},

6
libavfilter/f_settb.c
Unescape View File

@ -120,7 +120,7 @@ static int start_frame(AVFilterLink *inlink, AVFilterBufferRef *picref)
return ff_start_frame(outlink, picref);
}
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref)
static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *samplesref)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
@ -133,7 +133,7 @@ static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref)
outlink->time_base.num, outlink->time_base.den, samplesref->pts);
}
return ff_filter_samples(outlink, samplesref);
return ff_filter_frame(outlink, samplesref);
}
#if CONFIG_SETTB_FILTER
@ -181,7 +181,7 @@ AVFilter avfilter_af_asettb = {
{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.get_audio_buffer = ff_null_get_audio_buffer,
.filter_samples = filter_samples, },
.filter_frame = filter_frame, },
{ .name = NULL }
},
.outputs = (const AVFilterPad[]) {

6
libavfilter/fifo.c
Unescape View File

@ -228,7 +228,7 @@ static int return_audio_frame(AVFilterContext *ctx)
buf_out = s->buf_out;
s->buf_out = NULL;
}
return ff_filter_samples(link, buf_out);
return ff_filter_frame(link, buf_out);
}
static int request_frame(AVFilterLink *outlink)
@ -257,7 +257,7 @@ static int request_frame(AVFilterLink *outlink)
if (outlink->request_samples) {
return return_audio_frame(outlink->src);
} else {
ret = ff_filter_samples(outlink, fifo->root.next->buf);
ret = ff_filter_frame(outlink, fifo->root.next->buf);
queue_pop(fifo);
}
break;
@ -308,7 +308,7 @@ static const AVFilterPad avfilter_af_afifo_inputs[] = {
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.get_audio_buffer = ff_null_get_audio_buffer,
.filter_samples = add_to_queue,
.filter_frame = add_to_queue,
.min_perms = AV_PERM_PRESERVE,
},
{ NULL }

2
libavfilter/internal.h
Unescape View File

@ -147,7 +147,7 @@ struct AVFilterPad {
* must ensure that samplesref is properly unreferenced on error if it
* hasn't been passed on to another filter.
*/
int (*filter_samples)(AVFilterLink *link, AVFilterBufferRef *samplesref);
int (*filter_frame)(AVFilterLink *link, AVFilterBufferRef *samplesref);
/**
* Frame poll callback. This returns the number of immediately available

22
libavfilter/sink_buffer.c
Unescape View File

@ -268,7 +268,7 @@ AVFilter avfilter_vsink_buffersink = {
.outputs = (const AVFilterPad[]) {{ .name = NULL }},
};
static int filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
static int filter_frame(AVFilterLink *link, AVFilterBufferRef *samplesref)
{
end_frame(link);
return 0;
@ -338,7 +338,7 @@ AVFilter avfilter_asink_ffabuffersink = {
.inputs = (const AVFilterPad[]) {{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples,
.filter_frame = filter_frame,
.min_perms = AV_PERM_READ | AV_PERM_PRESERVE, },
{ .name = NULL }},
.outputs = (const AVFilterPad[]) {{ .name = NULL }},
@ -354,7 +354,7 @@ AVFilter avfilter_asink_abuffersink = {
.inputs = (const AVFilterPad[]) {{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples,
.filter_frame = filter_frame,
.min_perms = AV_PERM_READ | AV_PERM_PRESERVE, },
{ .name = NULL }},
.outputs = (const AVFilterPad[]) {{ .name = NULL }},
@ -372,13 +372,13 @@ int av_buffersink_read(AVFilterContext *ctx, AVFilterBufferRef **buf)
if (ctx->filter-> inputs[0].start_frame ==
avfilter_vsink_buffer. inputs[0].start_frame ||
ctx->filter-> inputs[0].filter_samples ==
avfilter_asink_abuffer.inputs[0].filter_samples)
ctx->filter-> inputs[0].filter_frame ==
avfilter_asink_abuffer.inputs[0].filter_frame)
return ff_buffersink_read_compat(ctx, buf);
av_assert0(ctx->filter-> inputs[0].end_frame ==
avfilter_vsink_ffbuffersink. inputs[0].end_frame ||
ctx->filter-> inputs[0].filter_samples ==
avfilter_asink_ffabuffersink.inputs[0].filter_samples);
ctx->filter-> inputs[0].filter_frame ==
avfilter_asink_ffabuffersink.inputs[0].filter_frame);
ret = av_buffersink_get_buffer_ref(ctx, &tbuf,
buf ? 0 : AV_BUFFERSINK_FLAG_PEEK);
@ -399,11 +399,11 @@ int av_buffersink_read_samples(AVFilterContext *ctx, AVFilterBufferRef **buf,
AVFilterLink *link = ctx->inputs[0];
int nb_channels = av_get_channel_layout_nb_channels(link->channel_layout);
if (ctx->filter-> inputs[0].filter_samples ==
avfilter_asink_abuffer.inputs[0].filter_samples)
if (ctx->filter-> inputs[0].filter_frame ==
avfilter_asink_abuffer.inputs[0].filter_frame)
return ff_buffersink_read_samples_compat(ctx, buf, nb_samples);
av_assert0(ctx->filter-> inputs[0].filter_samples ==
avfilter_asink_ffabuffersink.inputs[0].filter_samples);
av_assert0(ctx->filter-> inputs[0].filter_frame ==
avfilter_asink_ffabuffersink.inputs[0].filter_frame);
tbuf = ff_get_audio_buffer(link, AV_PERM_WRITE, nb_samples);
if (!tbuf)

6
libavfilter/split.c
Unescape View File

@ -142,7 +142,7 @@ AVFilter avfilter_vf_split = {
.outputs = NULL,
};
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref)
static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *samplesref)
{
AVFilterContext *ctx = inlink->dst;
int i, ret = 0;
@ -155,7 +155,7 @@ static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref)
break;
}
ret = ff_filter_samples(inlink->dst->outputs[i], buf_out);
ret = ff_filter_frame(inlink->dst->outputs[i], buf_out);
if (ret < 0)
break;
}
@ -168,7 +168,7 @@ static const AVFilterPad avfilter_af_asplit_inputs[] = {
.name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.get_audio_buffer = ff_null_get_audio_buffer,
.filter_samples = filter_samples
.filter_frame = filter_frame
},
{ NULL }
};

2
libavfilter/src_movie.c
Unescape View File

@ -577,7 +577,7 @@ static int movie_push_frame(AVFilterContext *ctx, unsigned out_id)
ff_end_frame(outlink);
break;
case AVMEDIA_TYPE_AUDIO:
ff_filter_samples(outlink, buf);
ff_filter_frame(outlink, buf);
break;
}

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