Replace ffmpeg references with more accurate libav* references.

5 files changed, 5 insertions, 5 deletions

diff --git a/libavformat/asfdec.c b/libavformat/asfdec.c
index ed06e8a897..02453407b5 100644
--- a/libavformat/asfdec.c
+++ b/libavformat/asfdec.c
@@ -362,7 +362,7 @@ static int asf_read_stream_properties(AVFormatContext *s, int64_t size)
 
         /* Extract palette from extradata if bpp <= 8 */
         /* This code assumes that extradata contains only palette */
-        /* This is true for all paletted codecs implemented in ffmpeg */
+        /* This is true for all paletted codecs implemented in libavcodec */
         if (st->codec->extradata_size && (st->codec->bits_per_coded_sample <= 8)) {
             int av_unused i;
 #if HAVE_BIGENDIAN
diff --git a/libavformat/au.c b/libavformat/au.c
index 5121574c3c..4f7677347b 100644
--- a/libavformat/au.c
+++ b/libavformat/au.c
@@ -35,7 +35,7 @@
 /* if we don't know the size in advance */
 #define AU_UNKNOWN_SIZE ((uint32_t)(~0))
 
-/* The ffmpeg codecs we support, and the IDs they have in the file */
+/* The libavcodec codecs we support, and the IDs they have in the file */
 static const AVCodecTag codec_au_tags[] = {
     { CODEC_ID_PCM_MULAW, 1 },
     { CODEC_ID_PCM_S8, 2 },
diff --git a/libavformat/cutils.c b/libavformat/cutils.c
index f94442f1bd..995524400e 100644
--- a/libavformat/cutils.c
+++ b/libavformat/cutils.c
@@ -1,5 +1,5 @@
 /*
- * Various simple utilities for ffmpeg system
+ * various simple utilities for libavformat
  * Copyright (c) 2000, 2001, 2002 Fabrice Bellard
  *
  * This file is part of Libav.
diff --git a/libavformat/dv.c b/libavformat/dv.c
index 2813bc3b67..21823752f5 100644
--- a/libavformat/dv.c
+++ b/libavformat/dv.c
@@ -96,7 +96,7 @@ static const uint8_t* dv_extract_pack(uint8_t* frame, enum dv_pack_type t)
 /*
  * There's a couple of assumptions being made here:
  * 1. By default we silence erroneous (0x8000/16bit 0x800/12bit) audio samples.
- *    We can pass them upwards when ffmpeg will be ready to deal with them.
+ *    We can pass them upwards when libavcodec will be ready to deal with them.
  * 2. We don't do software emphasis.
  * 3. Audio is always returned as 16bit linear samples: 12bit nonlinear samples
  *    are converted into 16bit linear ones.
diff --git a/libavformat/rso.h b/libavformat/rso.h
index bdb39e8d7f..e3e88ea644 100644
--- a/libavformat/rso.h
+++ b/libavformat/rso.h
@@ -26,7 +26,7 @@
 
 #define RSO_HEADER_SIZE 8
 
-/* The ffmpeg codecs we support, and the IDs they have in the file */
+/* The libavcodec codecs we support, and the IDs they have in the file */
 extern const AVCodecTag ff_codec_rso_tags[];
 
 #endif /* AVFORMAT_RSO_H */