lavfi: add error handling to filter_samples().

libavfilter/af_amix.c

@@ -311,9 +311,7 @@ static int output_frame(AVFilterLink *outlink, int nb_samples)
     if (s->next_pts != AV_NOPTS_VALUE)
         s->next_pts += nb_samples;
 
-    ff_filter_samples(outlink, out_buf);
-
-    return 0;
+    return ff_filter_samples(outlink, out_buf);
 }
 
 /**
@@ -454,31 +452,37 @@ static int request_frame(AVFilterLink *outlink)
     return output_frame(outlink, available_samples);
 }
 
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
 {
     AVFilterContext  *ctx = inlink->dst;
     MixContext       *s = ctx->priv;
     AVFilterLink *outlink = ctx->outputs[0];
-    int i;
+    int i, ret = 0;
 
     for (i = 0; i < ctx->nb_inputs; i++)
         if (ctx->inputs[i] == inlink)
             break;
     if (i >= ctx->nb_inputs) {
         av_log(ctx, AV_LOG_ERROR, "unknown input link\n");
-        return;
+        ret = AVERROR(EINVAL);
+        goto fail;
     }
 
     if (i == 0) {
         int64_t pts = av_rescale_q(buf->pts, inlink->time_base,
                                    outlink->time_base);
-        frame_list_add_frame(s->frame_list, buf->audio->nb_samples, pts);
+        ret = frame_list_add_frame(s->frame_list, buf->audio->nb_samples, pts);
+        if (ret < 0)
+            goto fail;
     }
 
-    av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data,
-                        buf->audio->nb_samples);
+    ret = av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data,
+                              buf->audio->nb_samples);
 
+fail:
     avfilter_unref_buffer(buf);
+
+    return ret;
 }
 
 static int init(AVFilterContext *ctx, const char *args)

libavfilter/af_asyncts.c

@@ -136,18 +136,18 @@ static int request_frame(AVFilterLink *link)
         avresample_convert(s->avr, (void**)buf->extended_data, buf->linesize[0],
                            nb_samples, NULL, 0, 0);
         buf->pts = s->pts;
-        ff_filter_samples(link, buf);
-        return 0;
+        return ff_filter_samples(link, buf);
     }
 
     return ret;
 }
 
-static void write_to_fifo(ASyncContext *s, AVFilterBufferRef *buf)
+static int write_to_fifo(ASyncContext *s, AVFilterBufferRef *buf)
 {
-    avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data,
-                       buf->linesize[0], buf->audio->nb_samples);
+    int ret = avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data,
+                                 buf->linesize[0], buf->audio->nb_samples);
     avfilter_unref_buffer(buf);
+    return ret;
 }
 
 /* get amount of data currently buffered, in samples */
@@ -156,7 +156,7 @@ static int64_t get_delay(ASyncContext *s)
     return avresample_available(s->avr) + avresample_get_delay(s->avr);
 }
 
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
 {
     AVFilterContext  *ctx = inlink->dst;
     ASyncContext       *s = ctx->priv;
@@ -164,7 +164,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
     int nb_channels = av_get_channel_layout_nb_channels(buf->audio->channel_layout);
     int64_t pts = (buf->pts == AV_NOPTS_VALUE) ? buf->pts :
                   av_rescale_q(buf->pts, inlink->time_base, outlink->time_base);
-    int out_size;
+    int out_size, ret;
     int64_t delta;
 
     /* buffer data until we get the first timestamp */
@@ -172,14 +172,12 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
         if (pts != AV_NOPTS_VALUE) {
             s->pts = pts - get_delay(s);
         }
-        write_to_fifo(s, buf);
-        return;
+        return write_to_fifo(s, buf);
     }
 
     /* now wait for the next timestamp */
     if (pts == AV_NOPTS_VALUE) {
-        write_to_fifo(s, buf);
-        return;
+        return write_to_fifo(s, buf);
     }
 
     /* when we have two timestamps, compute how many samples would we have
@@ -202,8 +200,10 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
     if (out_size > 0) {
         AVFilterBufferRef *buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE,
                                                          out_size);
-        if (!buf_out)
-            return;
+        if (!buf_out) {
+            ret = AVERROR(ENOMEM);
+            goto fail;
+        }
 
         avresample_read(s->avr, (void**)buf_out->extended_data, out_size);
         buf_out->pts = s->pts;
@@ -212,7 +212,9 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
             av_samples_set_silence(buf_out->extended_data, out_size - delta,
                                    delta, nb_channels, buf->format);
         }
-        ff_filter_samples(outlink, buf_out);
+        ret = ff_filter_samples(outlink, buf_out);
+        if (ret < 0)
+            goto fail;
         s->got_output = 1;
     } else {
         av_log(ctx, AV_LOG_WARNING, "Non-monotonous timestamps, dropping "
@@ -223,9 +225,13 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
     avresample_read(s->avr, NULL, avresample_available(s->avr));
 
     s->pts = pts - avresample_get_delay(s->avr);
-    avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data,
-                       buf->linesize[0], buf->audio->nb_samples);
+    ret = avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data,
+                             buf->linesize[0], buf->audio->nb_samples);
+
+fail:
     avfilter_unref_buffer(buf);
+
+    return ret;
 }
 
 AVFilter avfilter_af_asyncts = {

libavfilter/af_channelmap.c

@@ -313,7 +313,7 @@ static int channelmap_query_formats(AVFilterContext *ctx)
     return 0;
 }
 
-static void channelmap_filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
+static int channelmap_filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
 {
     AVFilterContext  *ctx = inlink->dst;
     AVFilterLink *outlink = ctx->outputs[0];
@@ -330,8 +330,10 @@ static void channelmap_filter_samples(AVFilterLink *inlink, AVFilterBufferRef *b
         if (nch_out > FF_ARRAY_ELEMS(buf->data)) {
             uint8_t **new_extended_data =
                 av_mallocz(nch_out * sizeof(*buf->extended_data));
-            if (!new_extended_data)
-                return;
+            if (!new_extended_data) {
+                avfilter_unref_buffer(buf);
+                return AVERROR(ENOMEM);
+            }
             if (buf->extended_data == buf->data) {
                 buf->extended_data = new_extended_data;
             } else {
@@ -353,7 +355,7 @@ static void channelmap_filter_samples(AVFilterLink *inlink, AVFilterBufferRef *b
         memcpy(buf->data, buf->extended_data,
            FFMIN(FF_ARRAY_ELEMS(buf->data), nch_out) * sizeof(buf->data[0]));
 
-    ff_filter_samples(outlink, buf);
+    return ff_filter_samples(outlink, buf);
 }
 
 static int channelmap_config_input(AVFilterLink *inlink)

libavfilter/af_channelsplit.c

@@ -110,24 +110,29 @@ static int query_formats(AVFilterContext *ctx)
     return 0;
 }
 
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
 {
     AVFilterContext *ctx = inlink->dst;
-    int i;
+    int i, ret = 0;
 
     for (i = 0; i < ctx->nb_outputs; i++) {
         AVFilterBufferRef *buf_out = avfilter_ref_buffer(buf, ~AV_PERM_WRITE);
 
-        if (!buf_out)
-            return;
+        if (!buf_out) {
+            ret = AVERROR(ENOMEM);
+            break;
+        }
 
         buf_out->data[0] = buf_out->extended_data[0] = buf_out->extended_data[i];
         buf_out->audio->channel_layout =
             av_channel_layout_extract_channel(buf->audio->channel_layout, i);
 
-        ff_filter_samples(ctx->outputs[i], buf_out);
+        ret = ff_filter_samples(ctx->outputs[i], buf_out);
+        if (ret < 0)
+            break;
     }
     avfilter_unref_buffer(buf);
+    return ret;
 }
 
 AVFilter avfilter_af_channelsplit = {

libavfilter/af_join.c

@@ -92,7 +92,7 @@ static const AVClass join_class = {
     .version    = LIBAVUTIL_VERSION_INT,
 };
 
-static void filter_samples(AVFilterLink *link, AVFilterBufferRef *buf)
+static int filter_samples(AVFilterLink *link, AVFilterBufferRef *buf)
 {
     AVFilterContext *ctx = link->dst;
     JoinContext       *s = ctx->priv;
@@ -104,6 +104,8 @@ static void filter_samples(AVFilterLink *link, AVFilterBufferRef *buf)
     av_assert0(i < ctx->nb_inputs);
     av_assert0(!s->input_frames[i]);
     s->input_frames[i] = buf;
+
+    return 0;
 }
 
 static int parse_maps(AVFilterContext *ctx)
@@ -468,11 +470,11 @@ static int join_request_frame(AVFilterLink *outlink)
     priv->nb_in_buffers = ctx->nb_inputs;
     buf->buf->priv      = priv;
 
-    ff_filter_samples(outlink, buf);
+    ret = ff_filter_samples(outlink, buf);
 
     memset(s->input_frames, 0, sizeof(*s->input_frames) * ctx->nb_inputs);
 
-    return 0;
+    return ret;
 
 fail:
     avfilter_unref_buffer(buf);

libavfilter/af_resample.c

@@ -157,21 +157,21 @@ static int request_frame(AVFilterLink *outlink)
         }
 
         buf->pts = s->next_pts;
-        ff_filter_samples(outlink, buf);
-        return 0;
+        return ff_filter_samples(outlink, buf);
     }
     return ret;
 }
 
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
 {
     AVFilterContext  *ctx = inlink->dst;
     ResampleContext    *s = ctx->priv;
     AVFilterLink *outlink = ctx->outputs[0];
+    int ret;
 
     if (s->avr) {
         AVFilterBufferRef *buf_out;
-        int delay, nb_samples, ret;
+        int delay, nb_samples;
 
         /* maximum possible samples lavr can output */
         delay      = avresample_get_delay(s->avr);
@@ -180,10 +180,19 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
                                     AV_ROUND_UP);
 
         buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
+        if (!buf_out) {
+            ret = AVERROR(ENOMEM);
+            goto fail;
+        }
+
         ret     = avresample_convert(s->avr, (void**)buf_out->extended_data,
                                      buf_out->linesize[0], nb_samples,
                                      (void**)buf->extended_data, buf->linesize[0],
                                      buf->audio->nb_samples);
+        if (ret < 0) {
+            avfilter_unref_buffer(buf_out);
+            goto fail;
+        }
 
         av_assert0(!avresample_available(s->avr));
 
@@ -209,14 +218,18 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
 
             s->next_pts = buf_out->pts + buf_out->audio->nb_samples;
 
-            ff_filter_samples(outlink, buf_out);
+            ret = ff_filter_samples(outlink, buf_out);
             s->got_output = 1;
         }
+
+fail:
         avfilter_unref_buffer(buf);
     } else {
-        ff_filter_samples(outlink, buf);
+        ret = ff_filter_samples(outlink, buf);
         s->got_output = 1;
     }
+
+    return ret;
 }
 
 AVFilter avfilter_af_resample = {

libavfilter/asink_anullsink.c

@@ -19,7 +19,10 @@
 #include "avfilter.h"
 #include "internal.h"
 
-static void null_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref) { }
+static int null_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
+{
+    return 0;
+}
 
 AVFilter avfilter_asink_anullsink = {
     .name        = "anullsink",

libavfilter/audio.c

@@ -146,15 +146,15 @@ fail:
     return NULL;
 }
 
-static void default_filter_samples(AVFilterLink *link,
-                                   AVFilterBufferRef *samplesref)
+static int default_filter_samples(AVFilterLink *link,
+                                  AVFilterBufferRef *samplesref)
 {
-    ff_filter_samples(link->dst->outputs[0], samplesref);
+    return ff_filter_samples(link->dst->outputs[0], samplesref);
 }
 
-void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
+int ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
 {
-    void (*filter_samples)(AVFilterLink *, AVFilterBufferRef *);
+    int (*filter_samples)(AVFilterLink *, AVFilterBufferRef *);
     AVFilterPad *dst = link->dstpad;
     AVFilterBufferRef *buf_out;
 
@@ -185,6 +185,6 @@ void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
     } else
         buf_out = samplesref;
 
-    filter_samples(link, buf_out);
+    return filter_samples(link, buf_out);
 }
 

libavfilter/audio.h

@@ -49,7 +49,10 @@ AVFilterBufferRef *ff_get_audio_buffer(AVFilterLink *link, int perms,
  * @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.
  */
-void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref);
+int ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref);
 
 #endif /* AVFILTER_AUDIO_H */

libavfilter/avfilter.h

@@ -288,8 +288,12 @@ struct AVFilterPad {
      * and should do its processing.
      *
      * Input audio pads only.
+     *
+     * @return >= 0 on success, a negative AVERROR on error. This function
+     * must ensure that samplesref is properly unreferenced on error if it
+     * hasn't been passed on to another filter.
      */
-    void (*filter_samples)(AVFilterLink *link, AVFilterBufferRef *samplesref);
+    int (*filter_samples)(AVFilterLink *link, AVFilterBufferRef *samplesref);
 
     /**
      * Frame poll callback. This returns the number of immediately available

libavfilter/buffersink.c

@@ -56,6 +56,12 @@ static void start_frame(AVFilterLink *link, AVFilterBufferRef *buf)
     link->cur_buf = NULL;
 };
 
+static int filter_samples(AVFilterLink *link, AVFilterBufferRef *buf)
+{
+    start_frame(link, buf);
+    return 0;
+}
+
 int av_buffersink_read(AVFilterContext *ctx, AVFilterBufferRef **buf)
 {
     BufferSinkContext *s    = ctx->priv;
@@ -160,7 +166,7 @@ AVFilter avfilter_asink_abuffer = {
 
     .inputs    = (AVFilterPad[]) {{ .name           = "default",
                                     .type           = AVMEDIA_TYPE_AUDIO,
-                                    .filter_samples = start_frame,
+                                    .filter_samples = filter_samples,
                                     .min_perms      = AV_PERM_READ,
                                     .needs_fifo     = 1 },
                                   { .name = NULL }},

libavfilter/buffersrc.c

@@ -312,6 +312,7 @@ static int request_frame(AVFilterLink *link)
 {
     BufferSourceContext *c = link->src->priv;
     AVFilterBufferRef *buf;
+    int ret = 0;
 
     if (!av_fifo_size(c->fifo)) {
         if (c->eof)
@@ -327,7 +328,7 @@ static int request_frame(AVFilterLink *link)
         ff_end_frame(link);
         break;
     case AVMEDIA_TYPE_AUDIO:
-        ff_filter_samples(link, avfilter_ref_buffer(buf, ~0));
+        ret = ff_filter_samples(link, avfilter_ref_buffer(buf, ~0));
         break;
     default:
         return AVERROR(EINVAL);
@@ -335,7 +336,7 @@ static int request_frame(AVFilterLink *link)
 
     avfilter_unref_buffer(buf);
 
-    return 0;
+    return ret;
 }
 
 static int poll_frame(AVFilterLink *link)

libavfilter/fifo.c

@@ -72,13 +72,25 @@ static av_cold void uninit(AVFilterContext *ctx)
     avfilter_unref_buffer(fifo->buf_out);
 }
 
-static void add_to_queue(AVFilterLink *inlink, AVFilterBufferRef *buf)
+static int add_to_queue(AVFilterLink *inlink, AVFilterBufferRef *buf)
 {
     FifoContext *fifo = inlink->dst->priv;
 
     fifo->last->next = av_mallocz(sizeof(Buf));
+    if (!fifo->last->next) {
+        avfilter_unref_buffer(buf);
+        return AVERROR(ENOMEM);
+    }
+
     fifo->last = fifo->last->next;
     fifo->last->buf = buf;
+
+    return 0;
+}
+
+static void start_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
+{
+    add_to_queue(inlink, buf);
 }
 
 static void queue_pop(FifoContext *s)
@@ -210,15 +222,13 @@ static int return_audio_frame(AVFilterContext *ctx)
         buf_out = s->buf_out;
         s->buf_out = NULL;
     }
-    ff_filter_samples(link, buf_out);
-
-    return 0;
+    return ff_filter_samples(link, buf_out);
 }
 
 static int request_frame(AVFilterLink *outlink)
 {
     FifoContext *fifo = outlink->src->priv;
-    int ret;
+    int ret = 0;
 
     if (!fifo->root.next) {
         if ((ret = ff_request_frame(outlink->src->inputs[0])) < 0)
@@ -238,7 +248,7 @@ static int request_frame(AVFilterLink *outlink)
         if (outlink->request_samples) {
             return return_audio_frame(outlink->src);
         } else {
-            ff_filter_samples(outlink, fifo->root.next->buf);
+            ret = ff_filter_samples(outlink, fifo->root.next->buf);
             queue_pop(fifo);
         }
         break;
@@ -246,7 +256,7 @@ static int request_frame(AVFilterLink *outlink)
         return AVERROR(EINVAL);
     }
 
-    return 0;
+    return ret;
 }
 
 AVFilter avfilter_vf_fifo = {
@@ -261,7 +271,7 @@ AVFilter avfilter_vf_fifo = {
     .inputs    = (AVFilterPad[]) {{ .name            = "default",
                                     .type            = AVMEDIA_TYPE_VIDEO,
                                     .get_video_buffer= ff_null_get_video_buffer,
-                                    .start_frame     = add_to_queue,
+                                    .start_frame     = start_frame,
                                     .draw_slice      = draw_slice,
                                     .end_frame       = end_frame,
                                     .rej_perms       = AV_PERM_REUSE2, },

libavfilter/internal.h

@@ -111,8 +111,12 @@ struct AVFilterPad {
      * and should do its processing.
      *
      * Input audio pads only.
+     *
+     * @return >= 0 on success, a negative AVERROR on error. This function
+     * must ensure that samplesref is properly unreferenced on error if it
+     * hasn't been passed on to another filter.
      */
-    void (*filter_samples)(AVFilterLink *link, AVFilterBufferRef *samplesref);
+    int (*filter_samples)(AVFilterLink *link, AVFilterBufferRef *samplesref);
 
     /**
      * Frame poll callback. This returns the number of immediately available

libavfilter/split.c

@@ -110,15 +110,19 @@ AVFilter avfilter_vf_split = {
     .outputs   = (AVFilterPad[]) {{ .name = NULL}},
 };
 
-static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref)
+static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref)
 {
     AVFilterContext *ctx = inlink->dst;
-    int i;
+    int i, ret = 0;
 
-    for (i = 0; i < ctx->nb_outputs; i++)
-        ff_filter_samples(inlink->dst->outputs[i],
-                          avfilter_ref_buffer(samplesref, ~AV_PERM_WRITE));
+    for (i = 0; i < ctx->nb_outputs; i++) {
+        ret = ff_filter_samples(inlink->dst->outputs[i],
+                                avfilter_ref_buffer(samplesref, ~AV_PERM_WRITE));
+        if (ret < 0)
+            break;
+    }
     avfilter_unref_buffer(samplesref);
+    return ret;
 }
 
 AVFilter avfilter_af_asplit = {