mirror of
https://github.com/FFmpeg/FFmpeg.git
synced 2025-01-08 13:22:53 +02:00
lavfi: replace filter_samples by filter_frame
Based on patch by Anton Khirnov Signed-off-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
parent
16af29a7a6
commit
cd7febd33f
@ -52,7 +52,7 @@ Buffer references ownership and permissions
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point to only a part of a video buffer.
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A reference is usually obtained as input to the start_frame or
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filter_samples method or requested using the ff_get_video_buffer or
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filter_frame method or requested using the ff_get_video_buffer or
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ff_get_audio_buffer functions. A new reference on an existing buffer can
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be created with the avfilter_ref_buffer. A reference is destroyed using
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the avfilter_unref_bufferp function.
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@ -68,14 +68,14 @@ Buffer references ownership and permissions
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Here are the (fairly obvious) rules for reference ownership:
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* A reference received by the start_frame or filter_samples method
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* A reference received by the start_frame or filter_frame method
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belong to the corresponding filter.
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Special exception: for video references: the reference may be used
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internally for automatic copying and must not be destroyed before
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end_frame; it can be given away to ff_start_frame.
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* A reference passed to ff_start_frame or ff_filter_samples is given
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* A reference passed to ff_start_frame or ff_filter_frame is given
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away and must no longer be used.
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* A reference created with avfilter_ref_buffer belongs to the code that
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@ -93,16 +93,16 @@ Buffer references ownership and permissions
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The AVFilterLink structure has a few AVFilterBufferRef fields. Here are
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the rules to handle them:
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* cur_buf is set before the start_frame and filter_samples methods to
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* cur_buf is set before the start_frame and filter_frame methods to
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the same reference given as argument to the methods and belongs to the
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destination filter of the link. If it has not been cleared after
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end_frame or filter_samples, libavfilter will automatically destroy
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end_frame or filter_frame, libavfilter will automatically destroy
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the reference; therefore, any filter that needs to keep the reference
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for longer must set cur_buf to NULL.
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* out_buf belongs to the source filter of the link and can be used to
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store a reference to the buffer that has been sent to the destination.
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If it is not NULL after end_frame or filter_samples, libavfilter will
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If it is not NULL after end_frame or filter_frame, libavfilter will
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automatically destroy the reference.
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If a video input pad does not have a start_frame method, the default
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@ -179,7 +179,7 @@ Buffer references ownership and permissions
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with the WRITE permission.
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* Filters that intend to keep a reference after the filtering process
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is finished (after end_frame or filter_samples returns) must have the
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is finished (after end_frame or filter_frame returns) must have the
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PRESERVE permission on it and remove the WRITE permission if they
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create a new reference to give it away.
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@ -198,7 +198,7 @@ Frame scheduling
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Simple filters that output one frame for each input frame should not have
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to worry about it.
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start_frame / filter_samples
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start_frame / filter_frame
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----------------------------
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These methods are called when a frame is pushed to the filter's input.
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@ -233,7 +233,7 @@ Frame scheduling
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This method is called when a frame is wanted on an output.
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For an input, it should directly call start_frame or filter_samples on
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For an input, it should directly call start_frame or filter_frame on
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the corresponding output.
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For a filter, if there are queued frames already ready, one of these
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@ -266,4 +266,4 @@ Frame scheduling
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Note that, except for filters that can have queued frames, request_frame
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does not push frames: it requests them to its input, and as a reaction,
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the start_frame / filter_samples method will be called and do the work.
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the start_frame / filter_frame method will be called and do the work.
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@ -135,7 +135,7 @@ static int config_output(AVFilterLink *outlink)
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return 0;
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}
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static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
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static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
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{
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AConvertContext *aconvert = inlink->dst->priv;
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const int n = insamplesref->audio->nb_samples;
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@ -149,7 +149,7 @@ static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref
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avfilter_copy_buffer_ref_props(outsamplesref, insamplesref);
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outsamplesref->audio->channel_layout = outlink->channel_layout;
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ret = ff_filter_samples(outlink, outsamplesref);
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ret = ff_filter_frame(outlink, outsamplesref);
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avfilter_unref_buffer(insamplesref);
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return ret;
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}
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@ -164,7 +164,7 @@ AVFilter avfilter_af_aconvert = {
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.inputs = (const AVFilterPad[]) {{ .name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_samples = filter_samples,
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.filter_frame = filter_frame,
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.min_perms = AV_PERM_READ, },
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{ .name = NULL}},
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.outputs = (const AVFilterPad[]) {{ .name = "default",
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@ -217,7 +217,7 @@ static inline void copy_samples(int nb_inputs, struct amerge_input in[],
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}
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}
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static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
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static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *insamples)
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{
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AVFilterContext *ctx = inlink->dst;
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AMergeContext *am = ctx->priv;
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@ -290,7 +290,7 @@ static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
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}
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}
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}
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return ff_filter_samples(ctx->outputs[0], outbuf);
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return ff_filter_frame(ctx->outputs[0], outbuf);
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}
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static av_cold int init(AVFilterContext *ctx, const char *args)
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@ -313,7 +313,7 @@ static av_cold int init(AVFilterContext *ctx, const char *args)
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AVFilterPad pad = {
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.name = name,
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_samples = filter_samples,
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.filter_frame = filter_frame,
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.min_perms = AV_PERM_READ | AV_PERM_PRESERVE,
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};
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if (!name)
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@ -309,7 +309,7 @@ static int output_frame(AVFilterLink *outlink, int nb_samples)
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if (s->next_pts != AV_NOPTS_VALUE)
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s->next_pts += nb_samples;
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return ff_filter_samples(outlink, out_buf);
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return ff_filter_frame(outlink, out_buf);
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}
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/**
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@ -450,7 +450,7 @@ static int request_frame(AVFilterLink *outlink)
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return output_frame(outlink, available_samples);
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}
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static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
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static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
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{
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AVFilterContext *ctx = inlink->dst;
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MixContext *s = ctx->priv;
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@ -502,7 +502,7 @@ static int init(AVFilterContext *ctx, const char *args)
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snprintf(name, sizeof(name), "input%d", i);
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pad.type = AVMEDIA_TYPE_AUDIO;
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pad.name = av_strdup(name);
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pad.filter_samples = filter_samples;
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pad.filter_frame = filter_frame;
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ff_insert_inpad(ctx, i, &pad);
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}
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@ -170,7 +170,7 @@ static int config_output(AVFilterLink *outlink)
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return 0;
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}
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static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
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static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
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{
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AResampleContext *aresample = inlink->dst->priv;
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const int n_in = insamplesref->audio->nb_samples;
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@ -205,7 +205,7 @@ static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
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outsamplesref->audio->nb_samples = n_out;
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ret = ff_filter_samples(outlink, outsamplesref);
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ret = ff_filter_frame(outlink, outsamplesref);
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aresample->req_fullfilled= 1;
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avfilter_unref_buffer(insamplesref);
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return ret;
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@ -247,7 +247,7 @@ static int request_frame(AVFilterLink *outlink)
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outsamplesref->pts = ROUNDED_DIV(outsamplesref->pts, inlink->sample_rate);
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#endif
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ff_filter_samples(outlink, outsamplesref);
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ff_filter_frame(outlink, outsamplesref);
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return 0;
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}
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return ret;
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@ -263,7 +263,7 @@ AVFilter avfilter_af_aresample = {
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.inputs = (const AVFilterPad[]) {{ .name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_samples = filter_samples,
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.filter_frame = filter_frame,
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.min_perms = AV_PERM_READ, },
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{ .name = NULL}},
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.outputs = (const AVFilterPad[]) {{ .name = "default",
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@ -125,12 +125,12 @@ static int push_samples(AVFilterLink *outlink)
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if (asns->next_out_pts != AV_NOPTS_VALUE)
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asns->next_out_pts += nb_out_samples;
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ff_filter_samples(outlink, outsamples);
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ff_filter_frame(outlink, outsamples);
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asns->req_fullfilled = 1;
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return nb_out_samples;
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}
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static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
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static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *insamples)
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{
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AVFilterContext *ctx = inlink->dst;
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ASNSContext *asns = ctx->priv;
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@ -186,7 +186,7 @@ AVFilter avfilter_af_asetnsamples = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_samples = filter_samples,
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.filter_frame = filter_frame,
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.min_perms = AV_PERM_READ|AV_PERM_WRITE
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},
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{ .name = NULL }
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@ -54,7 +54,7 @@ static void uninit(AVFilterContext *ctx)
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av_freep(&s->plane_checksums);
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}
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static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
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static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
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{
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AVFilterContext *ctx = inlink->dst;
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AShowInfoContext *s = ctx->priv;
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@ -100,7 +100,7 @@ static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
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av_log(ctx, AV_LOG_INFO, "]\n");
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s->frame++;
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return ff_filter_samples(inlink->dst->outputs[0], buf);
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return ff_filter_frame(inlink->dst->outputs[0], buf);
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}
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static const AVFilterPad inputs[] = {
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@ -108,7 +108,7 @@ static const AVFilterPad inputs[] = {
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.get_audio_buffer = ff_null_get_audio_buffer,
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.filter_samples = filter_samples,
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.filter_frame = filter_frame,
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.min_perms = AV_PERM_READ,
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},
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{ NULL },
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@ -122,7 +122,7 @@ static int send_out(AVFilterContext *ctx, int out_id)
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av_q2d(ctx->outputs[out_id]->time_base) * buf->pts;
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as->var_values[VAR_T1 + out_id] += buf->audio->nb_samples /
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(double)ctx->inputs[out_id]->sample_rate;
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ret = ff_filter_samples(ctx->outputs[out_id], buf);
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ret = ff_filter_frame(ctx->outputs[out_id], buf);
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queue->nb--;
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queue->tail = (queue->tail + 1) % QUEUE_SIZE;
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if (as->req[out_id])
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@ -167,7 +167,7 @@ static int request_frame(AVFilterLink *outlink)
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return 0;
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}
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static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
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static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *insamples)
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{
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AVFilterContext *ctx = inlink->dst;
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AStreamSyncContext *as = ctx->priv;
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@ -191,11 +191,11 @@ AVFilter avfilter_af_astreamsync = {
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.inputs = (const AVFilterPad[]) {
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{ .name = "in1",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_samples = filter_samples,
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.filter_frame = filter_frame,
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.min_perms = AV_PERM_READ | AV_PERM_PRESERVE, },
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{ .name = "in2",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_samples = filter_samples,
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.filter_frame = filter_frame,
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.min_perms = AV_PERM_READ | AV_PERM_PRESERVE, },
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{ .name = NULL }
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},
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@ -39,7 +39,7 @@ typedef struct ASyncContext {
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float min_delta_sec;
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int max_comp;
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/* set by filter_samples() to signal an output frame to request_frame() */
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/* set by filter_frame() to signal an output frame to request_frame() */
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int got_output;
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} ASyncContext;
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@ -135,7 +135,7 @@ static int request_frame(AVFilterLink *link)
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}
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buf->pts = s->pts;
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return ff_filter_samples(link, buf);
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return ff_filter_frame(link, buf);
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}
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return ret;
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@ -155,7 +155,7 @@ static int64_t get_delay(ASyncContext *s)
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return avresample_available(s->avr) + avresample_get_delay(s->avr);
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}
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static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
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static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
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{
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AVFilterContext *ctx = inlink->dst;
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ASyncContext *s = ctx->priv;
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@ -211,7 +211,7 @@ static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
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av_samples_set_silence(buf_out->extended_data, out_size - delta,
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delta, nb_channels, buf->format);
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}
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ret = ff_filter_samples(outlink, buf_out);
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ret = ff_filter_frame(outlink, buf_out);
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if (ret < 0)
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goto fail;
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s->got_output = 1;
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@ -237,7 +237,7 @@ static const AVFilterPad avfilter_af_asyncts_inputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_samples = filter_samples
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.filter_frame = filter_frame
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},
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{ NULL }
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};
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@ -138,7 +138,7 @@ typedef struct {
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RDFTContext *complex_to_real;
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FFTSample *correlation;
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// for managing AVFilterPad.request_frame and AVFilterPad.filter_samples
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// for managing AVFilterPad.request_frame and AVFilterPad.filter_frame
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int request_fulfilled;
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AVFilterBufferRef *dst_buffer;
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uint8_t *dst;
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@ -1033,7 +1033,7 @@ static void push_samples(ATempoContext *atempo,
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(AVRational){ 1, outlink->sample_rate },
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outlink->time_base);
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ff_filter_samples(outlink, atempo->dst_buffer);
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ff_filter_frame(outlink, atempo->dst_buffer);
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atempo->dst_buffer = NULL;
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atempo->dst = NULL;
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atempo->dst_end = NULL;
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@ -1041,7 +1041,7 @@ static void push_samples(ATempoContext *atempo,
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atempo->nsamples_out += n_out;
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}
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static int filter_samples(AVFilterLink *inlink,
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static int filter_frame(AVFilterLink *inlink,
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AVFilterBufferRef *src_buffer)
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{
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AVFilterContext *ctx = inlink->dst;
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@ -1148,7 +1148,7 @@ AVFilter avfilter_af_atempo = {
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.inputs = (const AVFilterPad[]) {
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{ .name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_samples = filter_samples,
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.filter_frame = filter_frame,
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.config_props = config_props,
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.min_perms = AV_PERM_READ, },
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{ .name = NULL}
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@ -312,7 +312,7 @@ static int channelmap_query_formats(AVFilterContext *ctx)
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return 0;
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}
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static int channelmap_filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
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static int channelmap_filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
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{
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AVFilterContext *ctx = inlink->dst;
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AVFilterLink *outlink = ctx->outputs[0];
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@ -354,7 +354,7 @@ static int channelmap_filter_samples(AVFilterLink *inlink, AVFilterBufferRef *bu
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memcpy(buf->data, buf->extended_data,
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FFMIN(FF_ARRAY_ELEMS(buf->data), nch_out) * sizeof(buf->data[0]));
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return ff_filter_samples(outlink, buf);
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return ff_filter_frame(outlink, buf);
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}
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static int channelmap_config_input(AVFilterLink *inlink)
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@ -389,7 +389,7 @@ static const AVFilterPad avfilter_af_channelmap_inputs[] = {
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.min_perms = AV_PERM_READ | AV_PERM_WRITE,
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.filter_samples = channelmap_filter_samples,
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.filter_frame = channelmap_filter_frame,
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.config_props = channelmap_config_input
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},
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{ NULL }
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@ -105,7 +105,7 @@ static int query_formats(AVFilterContext *ctx)
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return 0;
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}
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static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
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static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
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{
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AVFilterContext *ctx = inlink->dst;
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int i, ret = 0;
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@ -122,7 +122,7 @@ static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
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buf_out->audio->channel_layout =
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av_channel_layout_extract_channel(buf->audio->channel_layout, i);
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ret = ff_filter_samples(ctx->outputs[i], buf_out);
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ret = ff_filter_frame(ctx->outputs[i], buf_out);
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if (ret < 0)
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break;
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}
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@ -134,7 +134,7 @@ static const AVFilterPad avfilter_af_channelsplit_inputs[] = {
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{
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.name = "default",
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.type = AVMEDIA_TYPE_AUDIO,
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.filter_samples = filter_samples,
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.filter_frame = filter_frame,
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},
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{ NULL }
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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}},
|
||||
|
@ -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);
|
||||
|
||||
|
@ -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}
|
||||
},
|
||||
|
@ -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 }
|
||||
|
@ -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[]) {
|
||||
|
@ -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}},
|
||||
|
||||
|
@ -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 }
|
||||
},
|
||||
|
@ -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 },
|
||||
};
|
||||
|
@ -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;
|
||||
}
|
||||
|
@ -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;
|
||||
|
@ -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 = {
|
||||
|
@ -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;
|
||||
}
|
||||
}
|
||||
|
@ -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 */
|
||||
|
@ -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);
|
||||
}
|
||||
|
@ -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 }
|
||||
|
@ -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 }
|
||||
|
@ -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)
|
||||
|
@ -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
|
||||
|
@ -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
|
||||
},
|
||||
|
@ -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);
|
||||
|
@ -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,
|
||||
|
@ -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 }
|
||||
},
|
||||
|
@ -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 }
|
||||
},
|
||||
|
@ -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[]) {
|
||||
|
@ -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 }
|
||||
|
@ -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
|
||||
|
@ -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)
|
||||
|
@ -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 }
|
||||
};
|
||||
|
@ -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;
|
||||
}
|
||||
|
||||
|
Loading…
Reference in New Issue
Block a user