1 files changed, 49 insertions, 6 deletions

diff --git a/libavutil/audio_fifo.c b/libavutil/audio_fifo.c
index b562537cf6..e4d38e0524 100644
--- a/libavutil/audio_fifo.c
+++ b/libavutil/audio_fifo.c
@@ -2,20 +2,20 @@
  * Audio FIFO
  * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
  *
- * This file is part of Libav.
+ * This file is part of FFmpeg.
  *
- * Libav is free software; you can redistribute it and/or
+ * FFmpeg is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
- * Libav is distributed in the hope that it will be useful,
+ * FFmpeg is distributed in the hope that it will be useful,
  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
- * License along with Libav; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
@@ -51,7 +51,7 @@ void av_audio_fifo_free(AVAudioFifo *af)
                 if (af->buf[i])
                     av_fifo_free(af->buf[i]);
             }
-            av_free(af->buf);
+            av_freep(&af->buf);
         }
         av_free(af);
     }
@@ -76,7 +76,7 @@ AVAudioFifo *av_audio_fifo_alloc(enum AVSampleFormat sample_fmt, int channels,
     af->sample_size = buf_size / nb_samples;
     af->nb_buffers  = av_sample_fmt_is_planar(sample_fmt) ? channels : 1;
 
-    af->buf = av_mallocz(af->nb_buffers * sizeof(*af->buf));
+    af->buf = av_mallocz_array(af->nb_buffers, sizeof(*af->buf));
     if (!af->buf)
         goto error;
 
@@ -136,6 +136,49 @@ int av_audio_fifo_write(AVAudioFifo *af, void **data, int nb_samples)
     return nb_samples;
 }
 
+int av_audio_fifo_peek(AVAudioFifo *af, void **data, int nb_samples)
+{
+    int i, ret, size;
+
+    if (nb_samples < 0)
+        return AVERROR(EINVAL);
+    nb_samples = FFMIN(nb_samples, af->nb_samples);
+    if (!nb_samples)
+        return 0;
+
+    size = nb_samples * af->sample_size;
+    for (i = 0; i < af->nb_buffers; i++) {
+        if ((ret = av_fifo_generic_peek(af->buf[i], data[i], size, NULL)) < 0)
+            return AVERROR_BUG;
+    }
+
+    return nb_samples;
+}
+
+int av_audio_fifo_peek_at(AVAudioFifo *af, void **data, int nb_samples, int offset)
+{
+    int i, ret, size;
+
+    if (offset < 0 || offset >= af->nb_samples)
+        return AVERROR(EINVAL);
+    if (nb_samples < 0)
+        return AVERROR(EINVAL);
+    nb_samples = FFMIN(nb_samples, af->nb_samples);
+    if (!nb_samples)
+        return 0;
+    if (offset > af->nb_samples - nb_samples)
+        return AVERROR(EINVAL);
+
+    offset *= af->sample_size;
+    size = nb_samples * af->sample_size;
+    for (i = 0; i < af->nb_buffers; i++) {
+        if ((ret = av_fifo_generic_peek_at(af->buf[i], data[i], offset, size, NULL)) < 0)
+            return AVERROR_BUG;
+    }
+
+    return nb_samples;
+}
+
 int av_audio_fifo_read(AVAudioFifo *af, void **data, int nb_samples)
 {
     int i, ret, size;