lavfi: switch to AVFrame.

Deprecate AVFilterBuffer/AVFilterBufferRef and everything related to it
and use AVFrame instead.

1 files changed, 28 insertions, 30 deletions

diff --git a/libavfilter/af_resample.c b/libavfilter/af_resample.c
index 84ca8f5501..f82a970bb3 100644
--- a/libavfilter/af_resample.c
+++ b/libavfilter/af_resample.c
@@ -174,7 +174,7 @@ static int request_frame(AVFilterLink *outlink)
 
     /* flush the lavr delay buffer */
     if (ret == AVERROR_EOF && s->avr) {
-        AVFilterBufferRef *buf;
+        AVFrame *frame;
         int nb_samples = av_rescale_rnd(avresample_get_delay(s->avr),
                                         outlink->sample_rate,
                                         ctx->inputs[0]->sample_rate,
@@ -183,25 +183,25 @@ static int request_frame(AVFilterLink *outlink)
         if (!nb_samples)
             return ret;
 
-        buf = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
-        if (!buf)
+        frame = ff_get_audio_buffer(outlink, nb_samples);
+        if (!frame)
             return AVERROR(ENOMEM);
 
-        ret = avresample_convert(s->avr, buf->extended_data,
-                                 buf->linesize[0], nb_samples,
+        ret = avresample_convert(s->avr, frame->extended_data,
+                                 frame->linesize[0], nb_samples,
                                  NULL, 0, 0);
         if (ret <= 0) {
-            avfilter_unref_buffer(buf);
+            av_frame_free(&frame);
             return (ret == 0) ? AVERROR_EOF : ret;
         }
 
-        buf->pts = s->next_pts;
-        return ff_filter_frame(outlink, buf);
+        frame->pts = s->next_pts;
+        return ff_filter_frame(outlink, frame);
     }
     return ret;
 }
 
-static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
+static int filter_frame(AVFilterLink *inlink, AVFrame *in)
 {
     AVFilterContext  *ctx = inlink->dst;
     ResampleContext    *s = ctx->priv;
@@ -209,27 +209,26 @@ static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
     int ret;
 
     if (s->avr) {
-        AVFilterBufferRef *buf_out;
+        AVFrame *out;
         int delay, nb_samples;
 
         /* maximum possible samples lavr can output */
         delay      = avresample_get_delay(s->avr);
-        nb_samples = av_rescale_rnd(buf->audio->nb_samples + delay,
+        nb_samples = av_rescale_rnd(in->nb_samples + delay,
                                     outlink->sample_rate, inlink->sample_rate,
                                     AV_ROUND_UP);
 
-        buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
-        if (!buf_out) {
+        out = ff_get_audio_buffer(outlink, nb_samples);
+        if (!out) {
             ret = AVERROR(ENOMEM);
             goto fail;
         }
 
-        ret     = avresample_convert(s->avr, buf_out->extended_data,
-                                     buf_out->linesize[0], nb_samples,
-                                     buf->extended_data, buf->linesize[0],
-                                     buf->audio->nb_samples);
+        ret = avresample_convert(s->avr, out->extended_data, out->linesize[0],
+                                 nb_samples, in->extended_data, in->linesize[0],
+                                 in->nb_samples);
         if (ret <= 0) {
-            avfilter_unref_buffer(buf_out);
+            av_frame_free(&out);
             if (ret < 0)
                 goto fail;
         }
@@ -237,36 +236,36 @@ static int filter_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
         av_assert0(!avresample_available(s->avr));
 
         if (s->next_pts == AV_NOPTS_VALUE) {
-            if (buf->pts == AV_NOPTS_VALUE) {
+            if (in->pts == AV_NOPTS_VALUE) {
                 av_log(ctx, AV_LOG_WARNING, "First timestamp is missing, "
                        "assuming 0.\n");
                 s->next_pts = 0;
             } else
-                s->next_pts = av_rescale_q(buf->pts, inlink->time_base,
+                s->next_pts = av_rescale_q(in->pts, inlink->time_base,
                                            outlink->time_base);
         }
 
         if (ret > 0) {
-            buf_out->audio->nb_samples = ret;
-            if (buf->pts != AV_NOPTS_VALUE) {
-                buf_out->pts = av_rescale_q(buf->pts, inlink->time_base,
+            out->nb_samples = ret;
+            if (in->pts != AV_NOPTS_VALUE) {
+                out->pts = av_rescale_q(in->pts, inlink->time_base,
                                             outlink->time_base) -
                                av_rescale(delay, outlink->sample_rate,
                                           inlink->sample_rate);
             } else
-                buf_out->pts = s->next_pts;
+                out->pts = s->next_pts;
 
-            s->next_pts = buf_out->pts + buf_out->audio->nb_samples;
+            s->next_pts = out->pts + out->nb_samples;
 
-            ret = ff_filter_frame(outlink, buf_out);
+            ret = ff_filter_frame(outlink, out);
             s->got_output = 1;
         }
 
 fail:
-        avfilter_unref_buffer(buf);
+        av_frame_free(&in);
     } else {
-        buf->format = outlink->format;
-        ret = ff_filter_frame(outlink, buf);
+        in->format = outlink->format;
+        ret = ff_filter_frame(outlink, in);
         s->got_output = 1;
     }
 
@@ -278,7 +277,6 @@ static const AVFilterPad avfilter_af_resample_inputs[] = {
         .name           = "default",
         .type           = AVMEDIA_TYPE_AUDIO,
         .filter_frame   = filter_frame,
-        .min_perms      = AV_PERM_READ
     },
     { NULL }
 };