lavfi: merge start_frame/draw_slice/end_frame

Any alleged performance benefits gained from the split are purely mythological and do not justify added code complexity.
afedchin · Nov 28, 2012 · 565e499 · 565e499
1 parent bb6c67b
commit 565e499
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Showing 55 changed files with 632 additions and 1,458 deletions.
diff --git a/libavfilter/af_amix.c b/libavfilter/af_amix.c
@@ -314,7 +314,7 @@ 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);
 }
 
 /**
@@ -455,7 +455,7 @@ 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;
@@ -509,7 +509,7 @@ 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);
     }

diff --git a/libavfilter/af_ashowinfo.c b/libavfilter/af_ashowinfo.c
@@ -34,6 +34,7 @@
 
 #include "audio.h"
 #include "avfilter.h"
+#include "internal.h"
 
 typedef struct AShowInfoContext {
     /**
@@ -64,7 +65,7 @@ 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;
@@ -103,7 +104,7 @@ 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[] = {
@@ -112,7 +113,7 @@ static const AVFilterPad inputs[] = {
         .type             = AVMEDIA_TYPE_AUDIO,
         .get_audio_buffer = ff_null_get_audio_buffer,
         .config_props     = config_input,
-        .filter_samples   = filter_samples,
+        .filter_frame     = filter_frame,
         .min_perms        = AV_PERM_READ,
     },
     { NULL },

diff --git a/libavfilter/af_asyncts.c b/libavfilter/af_asyncts.c
@@ -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;
 
@@ -141,7 +141,7 @@ static int request_frame(AVFilterLink *link)
         }
 
         buf->pts = s->pts;
-        return ff_filter_samples(link, buf);
+        return ff_filter_frame(link, buf);
     }
 
     return ret;
@@ -161,7 +161,7 @@ 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;
@@ -217,7 +217,7 @@ 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;
@@ -243,7 +243,7 @@ static const AVFilterPad avfilter_af_asyncts_inputs[] = {
     {
         .name           = "default",
         .type           = AVMEDIA_TYPE_AUDIO,
-        .filter_samples = filter_samples
+        .filter_frame   = filter_frame,
     },
     { NULL }
 };

diff --git a/libavfilter/af_channelmap.c b/libavfilter/af_channelmap.c
@@ -313,7 +313,7 @@ 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];
@@ -355,7 +355,7 @@ 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,
-        .filter_samples = channelmap_filter_samples,
+        .filter_frame   = channelmap_filter_frame,
         .config_props   = channelmap_config_input
     },
     { NULL }

diff --git a/libavfilter/af_channelsplit.c b/libavfilter/af_channelsplit.c
@@ -111,7 +111,7 @@ 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;
@@ -128,7 +128,7 @@ 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;
     }
@@ -140,7 +140,7 @@ static const AVFilterPad avfilter_af_channelsplit_inputs[] = {
     {
         .name           = "default",
         .type           = AVMEDIA_TYPE_AUDIO,
-        .filter_samples = filter_samples,
+        .filter_frame   = filter_frame,
     },
     { NULL }
 };

diff --git a/libavfilter/af_join.c b/libavfilter/af_join.c
@@ -93,7 +93,7 @@ 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;
@@ -230,7 +230,7 @@ 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;
 
@@ -471,7 +471,7 @@ 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);
 

diff --git a/libavfilter/af_resample.c b/libavfilter/af_resample.c
@@ -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;
@@ -224,15 +224,15 @@ 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;
         }
 
 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 }

diff --git a/libavfilter/asink_anullsink.c b/libavfilter/asink_anullsink.c
@@ -20,7 +20,7 @@
 #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;
@@ -30,7 +30,7 @@ static const AVFilterPad avfilter_asink_anullsink_inputs[] = {
     {
         .name           = "default",
         .type           = AVMEDIA_TYPE_AUDIO,
-        .filter_samples = null_filter_samples,
+        .filter_frame   = null_filter_frame,
     },
     { NULL },
 };

diff --git a/libavfilter/audio.c b/libavfilter/audio.c
@@ -146,50 +146,3 @@ AVFilterBufferRef* avfilter_get_audio_buffer_ref_from_arrays(uint8_t **data,
     av_freep(&samples);
     return NULL;
 }
-
-static int default_filter_samples(AVFilterLink *link,
-                                  AVFilterBufferRef *samplesref)
-{
-    return ff_filter_samples(link->dst->outputs[0], samplesref);
-}
-
-int ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
-{
-    int (*filter_samples)(AVFilterLink *, AVFilterBufferRef *);
-    AVFilterPad *dst = link->dstpad;
-    AVFilterBufferRef *buf_out;
-
-    FF_DPRINTF_START(NULL, filter_samples); ff_dlog_link(NULL, link, 1);
-
-    if (!(filter_samples = dst->filter_samples))
-        filter_samples = default_filter_samples;
-
-    /* prepare to copy the samples if the buffer has insufficient permissions */
-    if ((dst->min_perms & samplesref->perms) != dst->min_perms ||
-        dst->rej_perms & samplesref->perms) {
-        av_log(link->dst, AV_LOG_DEBUG,
-               "Copying audio data in avfilter (have perms %x, need %x, reject %x)\n",
-               samplesref->perms, link->dstpad->min_perms, link->dstpad->rej_perms);
-
-        buf_out = ff_default_get_audio_buffer(link, dst->min_perms,
-                                              samplesref->audio->nb_samples);
-        if (!buf_out) {
-            avfilter_unref_buffer(samplesref);
-            return AVERROR(ENOMEM);
-        }
-        buf_out->pts                = samplesref->pts;
-        buf_out->audio->sample_rate = samplesref->audio->sample_rate;
-
-        /* Copy actual data into new samples buffer */
-        av_samples_copy(buf_out->extended_data, samplesref->extended_data,
-                        0, 0, samplesref->audio->nb_samples,
-                        av_get_channel_layout_nb_channels(link->channel_layout),
-                        link->format);
-
-        avfilter_unref_buffer(samplesref);
-    } else
-        buf_out = samplesref;
-
-    return filter_samples(link, buf_out);
-}
-
diff --git a/libavfilter/audio.h b/libavfilter/audio.h
@@ -42,17 +42,4 @@ AVFilterBufferRef *ff_null_get_audio_buffer(AVFilterLink *link, int perms,
 AVFilterBufferRef *ff_get_audio_buffer(AVFilterLink *link, int perms,
                                              int nb_samples);
 
-/**
- * Send a buffer of audio samples to the next filter.
- *
- * @param link       the output link over which the audio samples are being sent
- * @param samplesref a reference to the buffer of audio samples being sent. The
- *                   receiving filter will free this reference when it no longer
- *                   needs it or pass it on to the next filter.
- *
- * @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);
-
 #endif /* AVFILTER_AUDIO_H */