[libav-stable] afifo: fix request_samples on the last frame in certain cases

Anton Khirnov git at libav.org
Mon Apr 22 02:35:39 CEST 2013


Module: libav
Branch: master
Commit: 9bfc6e02bae9de354fb9ba09a8a140e83eeadf7d

Author:    Anton Khirnov <anton at khirnov.net>
Committer: Diego Biurrun <diego at biurrun.de>
Date:      Tue Apr 16 21:53:56 2013 +0200

afifo: fix request_samples on the last frame in certain cases

The current code can fail to return the last frame if it contains
exactly the requested number of samples.

Fixes the join filter test, which previously did not include the last
408 samples in most cases.

CC:libav-stable at libav.org

Signed-off-by: Diego Biurrun <diego at biurrun.de>

---

 libavfilter/fifo.c          |   36 +++++++++++++++++++-----------------
 tests/fate/filter-audio.mak |    2 +-
 2 files changed, 20 insertions(+), 18 deletions(-)

diff --git a/libavfilter/fifo.c b/libavfilter/fifo.c
index ae46403..8d981ce 100644
--- a/libavfilter/fifo.c
+++ b/libavfilter/fifo.c
@@ -183,8 +183,25 @@ static int return_audio_frame(AVFilterContext *ctx)
         }
 
         while (s->out->nb_samples < s->allocated_samples) {
-            int len = FFMIN(s->allocated_samples - s->out->nb_samples,
-                            head->nb_samples);
+            int len;
+
+            if (!s->root.next) {
+                ret = ff_request_frame(ctx->inputs[0]);
+                if (ret == AVERROR_EOF) {
+                    av_samples_set_silence(s->out->extended_data,
+                                           s->out->nb_samples,
+                                           s->allocated_samples -
+                                           s->out->nb_samples,
+                                           nb_channels, link->format);
+                    s->out->nb_samples = s->allocated_samples;
+                    break;
+                } else if (ret < 0)
+                    return ret;
+            }
+            head = s->root.next->frame;
+
+            len = FFMIN(s->allocated_samples - s->out->nb_samples,
+                        head->nb_samples);
 
             av_samples_copy(s->out->extended_data, head->extended_data,
                             s->out->nb_samples, 0, len, nb_channels,
@@ -194,21 +211,6 @@ static int return_audio_frame(AVFilterContext *ctx)
             if (len == head->nb_samples) {
                 av_frame_free(&head);
                 queue_pop(s);
-
-                if (!s->root.next &&
-                    (ret = ff_request_frame(ctx->inputs[0])) < 0) {
-                    if (ret == AVERROR_EOF) {
-                        av_samples_set_silence(s->out->extended_data,
-                                               s->out->nb_samples,
-                                               s->allocated_samples -
-                                               s->out->nb_samples,
-                                               nb_channels, link->format);
-                        s->out->nb_samples = s->allocated_samples;
-                        break;
-                    }
-                    return ret;
-                }
-                head = s->root.next->frame;
             } else {
                 buffer_offset(link, head, len);
             }
diff --git a/tests/fate/filter-audio.mak b/tests/fate/filter-audio.mak
index 63293d9..7f1a056 100644
--- a/tests/fate/filter-audio.mak
+++ b/tests/fate/filter-audio.mak
@@ -45,7 +45,7 @@ fate-filter-join: SRC2 = $(TARGET_PATH)/tests/data/asynth-44100-3.wav
 fate-filter-join: tests/data/asynth-44100-2.wav tests/data/asynth-44100-3.wav
 fate-filter-join: CMD = md5 -i $(SRC1) -i $(SRC2) -filter_complex join=channel_layout=5 -f s16le
 fate-filter-join: CMP = oneline
-fate-filter-join: REF = 38fa1b18b0c46d77df6f17bfc4f078dd
+fate-filter-join: REF = 409e66fc5ece4ddea4aa16bc47026bb0
 
 FATE_AFILTER-$(call FILTERDEMDECENCMUX, VOLUME, WAV, PCM_S16LE, PCM_S16LE, PCM_S16LE) += fate-filter-volume
 fate-filter-volume: SRC = $(TARGET_PATH)/tests/data/asynth-44100-2.wav



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