@@ -36,7 +36,7 @@

  * a little more compression by exploiting the fact that adjacent pixels

  * tend to be similar.

  *

- * Note that this decoder could use ffmpeg's optimized VLC facilities

+ * Note that this decoder could use libavcodec's optimized VLC facilities

  * rather than naive, tree-based Huffman decoding. However, there are 256

  * Huffman tables. Plus, the VLC bit coding order is right -> left instead

  * or left -> right, so all of the bits would have to be reversed. Further,

@@ -241,7 +241,7 @@ static av_cold int encode_init(AVCodecContext* avc_context)

         header, comment, and tables.

         Each one is prefixed with a 16bit size, then they

-        are concatenated together into ffmpeg's extradata.

+        are concatenated together into libavcodec's extradata.

     */

     offset = 0;

@@ -27,7 +27,7 @@

 #include "avcodec.h"

 /*

- * Adapted to ffmpeg by Francois Revol <revol@free.fr>

+ * Adapted to libavcodec by Francois Revol <revol@free.fr>

  * (removed 68k REG stuff, changed types, added some statics and consts,

  * libavcodec api, context stuff, interlaced stereo out).

  */

@@ -1854,7 +1854,7 @@ static int decode_user_data(MpegEncContext *s, GetBitContext *gb){

         }

     }

-    /* ffmpeg detection */

+    /* libavcodec detection */

     e=sscanf(buf, "FFmpe%*[^b]b%d", &build)+3;

     if(e!=4)

         e=sscanf(buf, "FFmpeg v%d.%d.%d / libavcodec build: %d", &ver, &ver2, &ver3, &build);

@@ -30,8 +30,8 @@

  * Note that this decoder reads big endian RGB555 pixel values from the

  * bytestream, arranges them in the host's endian order, and outputs

  * them to the final rendered map in the same host endian order. This is

- * intended behavior as the ffmpeg documentation states that RGB555 pixels

- * shall be stored in native CPU endianness.

+ * intended behavior as the libavcodec documentation states that RGB555

+ * pixels shall be stored in native CPU endianness.

  */

 #include <stdio.h>

@@ -19,7 +19,7 @@

  */

 /* The *no_round* functions have been added by James A. Morrison, 2003,2004.

-   The vis code from libmpeg2 was adapted for ffmpeg by James A. Morrison.

+   The vis code from libmpeg2 was adapted for libavcodec by James A. Morrison.

  */

 #include "config.h"

@@ -2,9 +2,9 @@

  * Duck Truemotion v1 Decoding Tables

  *

  * Data in this file was originally part of VpVision from On2 which is

- * distributed under the GNU GPL. It is redistributed with ffmpeg under the

- * GNU LGPL using the common understanding that data tables necessary for

- * decoding algorithms are not necessarily licensable.

+ * distributed under the GNU GPL. It is redistributed with libavcodec under

+ * the GNU LGPL using the common understanding that data tables necessary

+ * for decoding algorithms are not necessarily copyrightable.

  *

  * This file is part of Libav.

  *

@@ -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

@@ -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 },

@@ -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.

@@ -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.

@@ -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 */