[FFmpeg-cvslog] cosmetics: various spelling fixes

Lou Logan git at videolan.org
Fri Jul 6 17:13:35 CEST 2012


ffmpeg | branch: master | Lou Logan <lou at lrcd.com> | Thu Jul  5 15:38:53 2012 -0800| [b22ecbc6a57cb721960783af3168c1a9332ea3db] | committer: Michael Niedermayer

cosmetics: various spelling fixes

Signed-off-by: Michael Niedermayer <michaelni at gmx.at>

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=b22ecbc6a57cb721960783af3168c1a9332ea3db
---

 doc/git-howto.texi         |    2 +-
 ffprobe.c                  |    4 ++--
 libavdevice/dv1394.h       |    2 +-
 libavformat/avformat.h     |    4 ++--
 libavformat/aviobuf.c      |    2 +-
 libavformat/hls.c          |    2 +-
 libavformat/hlsproto.c     |    2 +-
 libavformat/http.h         |    4 ++--
 libavformat/rtpdec_mpeg4.c |    2 +-
 libavformat/wtvdec.c       |    2 +-
 libavformat/xmv.c          |    2 +-
 libavutil/pixfmt.h         |    2 +-
 12 files changed, 15 insertions(+), 15 deletions(-)

diff --git a/doc/git-howto.texi b/doc/git-howto.texi
index 56a0cda..f9acad5 100644
--- a/doc/git-howto.texi
+++ b/doc/git-howto.texi
@@ -374,7 +374,7 @@ Next let the code pass through a full run of our testsuite.
 Make sure all your changes have been checked before pushing them, the
 testsuite only checks against regressions and that only to some extend. It does
 obviously not check newly added features/code to be working unless you have
-added a test for that (which is recommanded btw).
+added a test for that (which is recommended).
 
 Also note that every single commit should pass the test suite, not just
 the result of a series of patches.
diff --git a/ffprobe.c b/ffprobe.c
index f65e460..14cb054 100644
--- a/ffprobe.c
+++ b/ffprobe.c
@@ -783,7 +783,7 @@ typedef struct FlatContext {
 static const AVOption flat_options[]= {
     {"sep_char", "set separator",    OFFSET(sep_str),    AV_OPT_TYPE_STRING, {.str="."},  CHAR_MIN, CHAR_MAX },
     {"s",        "set separator",    OFFSET(sep_str),    AV_OPT_TYPE_STRING, {.str="."},  CHAR_MIN, CHAR_MAX },
-    {"hierachical", "specify if the section specification should be hierarchical", OFFSET(hierarchical), AV_OPT_TYPE_INT, {.dbl=1}, 0, 1 },
+    {"hierarchical", "specify if the section specification should be hierarchical", OFFSET(hierarchical), AV_OPT_TYPE_INT, {.dbl=1}, 0, 1 },
     {"h",           "specify if the section specification should be hierarchical", OFFSET(hierarchical), AV_OPT_TYPE_INT, {.dbl=1}, 0, 1 },
     {NULL},
 };
@@ -939,7 +939,7 @@ typedef struct {
 #define OFFSET(x) offsetof(INIContext, x)
 
 static const AVOption ini_options[] = {
-    {"hierachical", "specify if the section specification should be hierarchical", OFFSET(hierarchical), AV_OPT_TYPE_INT, {.dbl=1}, 0, 1 },
+    {"hierarchical", "specify if the section specification should be hierarchical", OFFSET(hierarchical), AV_OPT_TYPE_INT, {.dbl=1}, 0, 1 },
     {"h",           "specify if the section specification should be hierarchical", OFFSET(hierarchical), AV_OPT_TYPE_INT, {.dbl=1}, 0, 1 },
     {NULL},
 };
diff --git a/libavdevice/dv1394.h b/libavdevice/dv1394.h
index 3d6033a..b76d633 100644
--- a/libavdevice/dv1394.h
+++ b/libavdevice/dv1394.h
@@ -186,7 +186,7 @@
    where copy_DV_frame() reads or writes on the dv1394 file descriptor
    (read/write mode) or copies data to/from the mmap ringbuffer and
    then calls ioctl(DV1394_SUBMIT_FRAMES) to notify dv1394 that new
-   frames are availble (mmap mode).
+   frames are available (mmap mode).
 
    reset_dv1394() is called in the event of a buffer
    underflow/overflow or a halt in the DV stream (e.g. due to a 1394
diff --git a/libavformat/avformat.h b/libavformat/avformat.h
index 807c7ac..a435d51 100644
--- a/libavformat/avformat.h
+++ b/libavformat/avformat.h
@@ -946,7 +946,7 @@ typedef struct AVFormatContext {
 #define AVFMT_FLAG_MP4A_LATM    0x8000 ///< Enable RTP MP4A-LATM payload
 #define AVFMT_FLAG_SORT_DTS    0x10000 ///< try to interleave outputted packets by dts (using this flag can slow demuxing down)
 #define AVFMT_FLAG_PRIV_OPT    0x20000 ///< Enable use of private options by delaying codec open (this could be made default once all code is converted)
-#define AVFMT_FLAG_KEEP_SIDE_DATA 0x40000 ///< Dont merge side data but keep it separate.
+#define AVFMT_FLAG_KEEP_SIDE_DATA 0x40000 ///< Don't merge side data but keep it separate.
 
     /**
      * decoding: size of data to probe; encoding: unused.
@@ -1739,7 +1739,7 @@ int av_get_output_timestamp(struct AVFormatContext *s, int stream,
  * @ingroup libavf
  * @{
  *
- * Miscelaneous utility functions related to both muxing and demuxing
+ * Miscellaneous utility functions related to both muxing and demuxing
  * (or neither).
  */
 
diff --git a/libavformat/aviobuf.c b/libavformat/aviobuf.c
index 9154075..05df001 100644
--- a/libavformat/aviobuf.c
+++ b/libavformat/aviobuf.c
@@ -389,7 +389,7 @@ static void fill_buffer(AVIOContext *s)
     int len= s->buffer_size - (dst - s->buffer);
     int max_buffer_size = s->max_packet_size ? s->max_packet_size : IO_BUFFER_SIZE;
 
-    /* can't fill the buffer without read_packet, just set EOF if appropiate */
+    /* can't fill the buffer without read_packet, just set EOF if appropriate */
     if (!s->read_packet && s->buf_ptr >= s->buf_end)
         s->eof_reached = 1;
 
diff --git a/libavformat/hls.c b/libavformat/hls.c
index 1449f40..a51a616 100644
--- a/libavformat/hls.c
+++ b/libavformat/hls.c
@@ -42,7 +42,7 @@
  * An apple http stream consists of a playlist with media segment files,
  * played sequentially. There may be several playlists with the same
  * video content, in different bandwidth variants, that are played in
- * parallel (preferrably only one bandwidth variant at a time). In this case,
+ * parallel (preferably only one bandwidth variant at a time). In this case,
  * the user supplied the url to a main playlist that only lists the variant
  * playlists.
  *
diff --git a/libavformat/hlsproto.c b/libavformat/hlsproto.c
index 95bd047..a290c88 100644
--- a/libavformat/hlsproto.c
+++ b/libavformat/hlsproto.c
@@ -36,7 +36,7 @@
  * An apple http stream consists of a playlist with media segment files,
  * played sequentially. There may be several playlists with the same
  * video content, in different bandwidth variants, that are played in
- * parallel (preferrably only one bandwidth variant at a time). In this case,
+ * parallel (preferably only one bandwidth variant at a time). In this case,
  * the user supplied the url to a main playlist that only lists the variant
  * playlists.
  *
diff --git a/libavformat/http.h b/libavformat/http.h
index aaedbd6..a19ad8e 100644
--- a/libavformat/http.h
+++ b/libavformat/http.h
@@ -38,9 +38,9 @@ void ff_http_init_auth_state(URLContext *dest, const URLContext *src);
 /**
  * Send a new HTTP request, reusing the old connection.
  *
- * @param h pointer to the ressource
+ * @param h pointer to the resource
  * @param uri uri used to perform the request
- * @return a negative value if an error condition occured, 0
+ * @return a negative value if an error condition occurred, 0
  * otherwise
  */
 int ff_http_do_new_request(URLContext *h, const char *uri);
diff --git a/libavformat/rtpdec_mpeg4.c b/libavformat/rtpdec_mpeg4.c
index b11cc79..ab405c0 100644
--- a/libavformat/rtpdec_mpeg4.c
+++ b/libavformat/rtpdec_mpeg4.c
@@ -138,7 +138,7 @@ static int rtp_parse_mp4_au(PayloadContext *data, const uint8_t *buf)
 
     init_get_bits(&getbitcontext, buf, data->au_headers_length_bytes * 8);
 
-    /* XXX: Wrong if optionnal additional sections are present (cts, dts etc...) */
+    /* XXX: Wrong if optional additional sections are present (cts, dts etc...) */
     au_header_size = data->sizelength + data->indexlength;
     if (au_header_size <= 0 || (au_headers_length % au_header_size != 0))
         return -1;
diff --git a/libavformat/wtvdec.c b/libavformat/wtvdec.c
index bbd7c23..b21b097 100644
--- a/libavformat/wtvdec.c
+++ b/libavformat/wtvdec.c
@@ -208,7 +208,7 @@ static AVIOContext * wtvfile_open_sector(int first_sector, uint64_t length, int
     }
     wf->length = length;
 
-    /* seek to intial sector */
+    /* seek to initial sector */
     wf->position = 0;
     if (avio_seek(s->pb, (int64_t)wf->sectors[0] << WTV_SECTOR_BITS, SEEK_SET) < 0) {
         av_free(wf->sectors);
diff --git a/libavformat/xmv.c b/libavformat/xmv.c
index cb61e15..b4accdf 100644
--- a/libavformat/xmv.c
+++ b/libavformat/xmv.c
@@ -295,7 +295,7 @@ static int xmv_process_packet_header(AVFormatContext *s)
      * short for every audio track. But as playing around with XMV files with
      * ADPCM audio showed, taking the extra 4 bytes from the audio data gives
      * you either completely distorted audio or click (when skipping the
-     * remaining 68 bytes of the ADPCM block). Substracting 4 bytes for every
+     * remaining 68 bytes of the ADPCM block). Subtracting 4 bytes for every
      * audio track from the video data works at least for the audio. Probably
      * some alignment thing?
      * The video data has (always?) lots of padding, so it should work out...
diff --git a/libavutil/pixfmt.h b/libavutil/pixfmt.h
index d90108a..fa771bc 100644
--- a/libavutil/pixfmt.h
+++ b/libavutil/pixfmt.h
@@ -142,7 +142,7 @@ enum PixelFormat {
     PIX_FMT_BGR48LE,   ///< packed RGB 16:16:16, 48bpp, 16B, 16G, 16R, the 2-byte value for each R/G/B component is stored as little-endian
 
     //the following 10 formats have the disadvantage of needing 1 format for each bit depth, thus
-    //If you want to support multiple bit depths, then using PIX_FMT_YUV420P16* with the bpp stored seperately
+    //If you want to support multiple bit depths, then using PIX_FMT_YUV420P16* with the bpp stored separately
     //is better
     PIX_FMT_YUV420P9BE, ///< planar YUV 4:2:0, 13.5bpp, (1 Cr & Cb sample per 2x2 Y samples), big-endian
     PIX_FMT_YUV420P9LE, ///< planar YUV 4:2:0, 13.5bpp, (1 Cr & Cb sample per 2x2 Y samples), little-endian



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