doc: fix spelling errors

Thanks to Mathieu Malaterre <malat@debian.org> for reporting the
Que/Queue typo. (https://bugs.debian.org/839542)

Reviewed-by: Lou Logan <lou@lrcd.com>
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>

18 files changed, 25 insertions, 25 deletions

diff --git a/doc/Doxyfile b/doc/Doxyfile
index fb5cdd3677..0891899505 100644
--- a/doc/Doxyfile
+++ b/doc/Doxyfile
@@ -1637,7 +1637,7 @@ EXTRA_PACKAGES         =
 # following commands have a special meaning inside the header: $title,
 # $datetime, $date, $doxygenversion, $projectname, $projectnumber,
 # $projectbrief, $projectlogo. Doxygen will replace $title with the empy string,
-# for the replacement values of the other commands the user is refered to
+# for the replacement values of the other commands the user is referred to
 # HTML_HEADER.
 # This tag requires that the tag GENERATE_LATEX is set to YES.
 
diff --git a/doc/encoders.texi b/doc/encoders.texi
index 09e1b9e884..5a60e7ee5a 100644
--- a/doc/encoders.texi
+++ b/doc/encoders.texi
@@ -1773,7 +1773,7 @@ Enable CAVLC and disable CABAC. It generates the same effect as
 @end table
 
 @item cmp
-Set full pixel motion estimation comparation algorithm. Possible values:
+Set full pixel motion estimation comparison algorithm. Possible values:
 
 @table @samp
 @item chroma
diff --git a/doc/ffprobe.texi b/doc/ffprobe.texi
index 2024eed4e5..1069ae32de 100644
--- a/doc/ffprobe.texi
+++ b/doc/ffprobe.texi
@@ -245,7 +245,7 @@ continue reading from that.
 Each interval is specified by two optional parts, separated by "%".
 
 The first part specifies the interval start position. It is
-interpreted as an abolute position, or as a relative offset from the
+interpreted as an absolute position, or as a relative offset from the
 current position if it is preceded by the "+" character. If this first
 part is not specified, no seeking will be performed when reading this
 interval.
diff --git a/doc/filters.texi b/doc/filters.texi
index b470f40f31..c37fa295ce 100644
--- a/doc/filters.texi
+++ b/doc/filters.texi
@@ -449,7 +449,7 @@ This filter is bit crusher with enhanced functionality. A bit crusher
 is used to audibly reduce number of bits an audio signal is sampled
 with. This doesn't change the bit depth at all, it just produces the
 effect. Material reduced in bit depth sounds more harsh and "digital".
-This filter is able to even round to continous values instead of discrete
+This filter is able to even round to continuous values instead of discrete
 bit depths.
 Additionally it has a D/C offset which results in different crushing of
 the lower and the upper half of the signal.
@@ -475,7 +475,7 @@ Set level out.
 Set bit reduction.
 
 @item mix
-Set mixing ammount.
+Set mixing amount.
 
 @item mode
 Can be linear: @code{lin} or logarithmic: @code{log}.
@@ -1203,7 +1203,7 @@ Set video stream size. Only useful if curves option is activated.
 
 @item mgain
 Set max gain that will be displayed. Only useful if curves option is activated.
-Setting this to reasonable value allows to display gain which is derived from
+Setting this to a reasonable value makes it possible to display gain which is derived from
 neighbour bands which are too close to each other and thus produce higher gain
 when both are activated.
 
@@ -8858,7 +8858,7 @@ value.
 @section hysteresis
 
 Grow first stream into second stream by connecting components.
-This allows to build more robust edge masks.
+This makes it possible to build more robust edge masks.
 
 This filter accepts the following options:
 
@@ -17670,7 +17670,7 @@ magnitude across time and second represents phase across time.
 The filter will transform from frequency domain as displayed in videos back
 to time domain as presented in audio output.
 
-This filter is primarly created for reversing processed @ref{showspectrum}
+This filter is primarily created for reversing processed @ref{showspectrum}
 filter outputs, but can synthesize sound from other spectrograms too.
 But in such case results are going to be poor if the phase data is not
 available, because in such cases phase data need to be recreated, usually
diff --git a/doc/muxers.texi b/doc/muxers.texi
index dbe53f5caf..0d856dbf79 100644
--- a/doc/muxers.texi
+++ b/doc/muxers.texi
@@ -1513,14 +1513,14 @@ as a list of @var{key}=@var{value} pairs separated by ':'.
 
 @item drop_pkts_on_overflow @var{bool}
 If set to 1 (true), in case the fifo queue fills up, packets will be dropped
-rather than blocking the encoder. This allows to continue streaming without
-delaying the input, at the cost of ommiting part of the stream. By default
+rather than blocking the encoder. This makes it possible to continue streaming without
+delaying the input, at the cost of omitting part of the stream. By default
 this option is set to 0 (false), so in such cases the encoder will be blocked
 until the muxer processes some of the packets and none of them is lost.
 
 @item attempt_recovery @var{bool}
 If failure occurs, attempt to recover the output. This is especially useful
-when used with network output, allows to restart streaming transparently.
+when used with network output, since it makes it possible to restart streaming transparently.
 By default this option is set to 0 (false).
 
 @item max_recovery_attempts
diff --git a/ffmpeg.c b/ffmpeg.c
index edb27c8cde..3b9171090f 100644
--- a/ffmpeg.c
+++ b/ffmpeg.c
@@ -3771,7 +3771,7 @@ static int check_keyboard_interaction(int64_t cur_time)
                         "+      increase verbosity\n"
                         "-      decrease verbosity\n"
                         "c      Send command to first matching filter supporting it\n"
-                        "C      Send/Que command to all matching filters\n"
+                        "C      Send/Queue command to all matching filters\n"
                         "D      cycle through available debug modes\n"
                         "h      dump packets/hex press to cycle through the 3 states\n"
                         "q      quit\n"
diff --git a/ffmpeg_cuvid.c b/ffmpeg_cuvid.c
index bd2d97517f..766878f58b 100644
--- a/ffmpeg_cuvid.c
+++ b/ffmpeg_cuvid.c
@@ -129,7 +129,7 @@ int cuvid_transcode_init(OutputStream *ost)
         ist->hwaccel_uninit = cuvid_uninit;
 
         /* This is a bit hacky, av_hwframe_ctx_init is called by the cuvid decoder
-         * once it has probed the neccesary format information. But as filters/nvenc
+         * once it has probed the necessary format information. But as filters/nvenc
          * need to know the format/sw_format, set them here so they are happy.
          * This is fine as long as CUVID doesn't add another supported pix_fmt.
          */
@@ -147,7 +147,7 @@ error:
 
 cancel:
     if (ist->hwaccel_id == HWACCEL_CUVID) {
-        av_log(NULL, AV_LOG_ERROR, "CUVID hwaccel requested, but impossible to achive.\n");
+        av_log(NULL, AV_LOG_ERROR, "CUVID hwaccel requested, but impossible to achieve.\n");
         return AVERROR(EINVAL);
     }
 
diff --git a/libavcodec/aaccoder_twoloop.h b/libavcodec/aaccoder_twoloop.h
index fb9849e317..f175c5f8b8 100644
--- a/libavcodec/aaccoder_twoloop.h
+++ b/libavcodec/aaccoder_twoloop.h
@@ -87,7 +87,7 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
      * will keep iterating until it fails to lower it or it reaches
      * ulimit * rdlambda. Keeping it low increases quality on difficult
      * signals, but lower it too much, and bits will be taken from weak
-     * signals, creating "holes". A balance is necesary.
+     * signals, creating "holes". A balance is necessary.
      * rdmax and rdmin specify the relative deviation from rdlambda
      * allowed for tonality compensation
      */
diff --git a/libavcodec/cabac.c b/libavcodec/cabac.c
index c0abe834cf..dd2b057c6d 100644
--- a/libavcodec/cabac.c
+++ b/libavcodec/cabac.c
@@ -182,7 +182,7 @@ int ff_init_cabac_decoder(CABACContext *c, const uint8_t *buf, int buf_size){
 #if CABAC_BITS == 16
     c->low =  (*c->bytestream++)<<18;
     c->low+=  (*c->bytestream++)<<10;
-    // Keep our fetches on a 2-byte boundry as this should avoid ever having to
+    // Keep our fetches on a 2-byte boundary as this should avoid ever having to
     // do unaligned loads if the compiler (or asm) optimises the double byte
     // load into a single instruction
     if(((uintptr_t)c->bytestream & 1) == 0) {
diff --git a/libavcodec/ffjni.c b/libavcodec/ffjni.c
index 0597400ba6..3f4c380673 100644
--- a/libavcodec/ffjni.c
+++ b/libavcodec/ffjni.c
@@ -224,7 +224,7 @@ int ff_jni_exception_get_summary(JNIEnv *env, jthrowable exception, char **error
     } else if (!name && message) {
         av_bprintf(&bp, "Exception: %s", message);
     } else {
-        av_log(log_ctx, AV_LOG_WARNING, "Could not retreive exception name and message\n");
+        av_log(log_ctx, AV_LOG_WARNING, "Could not retrieve exception name and message\n");
         av_bprintf(&bp, "Exception occurred");
     }
 
diff --git a/libavcodec/mediacodec_wrapper.h b/libavcodec/mediacodec_wrapper.h
index cddd420c26..1b4f3a9492 100644
--- a/libavcodec/mediacodec_wrapper.h
+++ b/libavcodec/mediacodec_wrapper.h
@@ -44,7 +44,7 @@
  * implementation.
  *
  * The API around MediaCodecList is not part of the NDK (and is lacking as
- * we still need to retreive the codec name to work around faulty decoders
+ * we still need to retrieve the codec name to work around faulty decoders
  * and encoders).
  *
  * For documentation, please refers to NdkMediaCodec.h NdkMediaFormat.h and
diff --git a/libavcodec/psymodel.h b/libavcodec/psymodel.h
index 35d184c710..582f040c7a 100644
--- a/libavcodec/psymodel.h
+++ b/libavcodec/psymodel.h
@@ -29,7 +29,7 @@
 /** maximum number of channels */
 #define PSY_MAX_CHANS 20
 
-/* cutoff for VBR is purposedly increased, since LP filtering actually
+/* cutoff for VBR is purposely increased, since LP filtering actually
  * hinders VBR performance rather than the opposite
  */
 #define AAC_CUTOFF_FROM_BITRATE(bit_rate,channels,sample_rate) (bit_rate ? FFMIN3(FFMIN3( \
diff --git a/libavcodec/x86/vp9lpf_16bpp.asm b/libavcodec/x86/vp9lpf_16bpp.asm
index c15437b8ba..c0888170c9 100644
--- a/libavcodec/x86/vp9lpf_16bpp.asm
+++ b/libavcodec/x86/vp9lpf_16bpp.asm
@@ -78,7 +78,7 @@ SECTION .text
 %endif
 %endmacro
 
-; calulate p or q portion of flat8out
+; calculate p or q portion of flat8out
 %macro FLAT8OUT_HALF 0
     psubw               m4, m0                      ; q4-q0
     psubw               m5, m0                      ; q5-q0
diff --git a/libavfilter/af_hdcd.c b/libavfilter/af_hdcd.c
index 91f84aea22..caee470c2e 100644
--- a/libavfilter/af_hdcd.c
+++ b/libavfilter/af_hdcd.c
@@ -867,7 +867,7 @@ typedef struct {
      *  steps of 0.5, but no value below -6.0 dB should appear. */
     int gain_counts[16];
     int max_gain;
-    /** occurences of code detect timer expiring without detecting
+    /** occurrences of code detect timer expiring without detecting
      *  a code. -1 for timer never set. */
     int count_sustain_expired;
 
diff --git a/libavfilter/f_ebur128.c b/libavfilter/f_ebur128.c
index 59eaedd926..983c889017 100644
--- a/libavfilter/f_ebur128.c
+++ b/libavfilter/f_ebur128.c
@@ -141,7 +141,7 @@ typedef struct {
     int loglevel;                   ///< log level for frame logging
     int metadata;                   ///< whether or not to inject loudness results in frames
     int dual_mono;                  ///< whether or not to treat single channel input files as dual-mono
-    double pan_law;                 ///< pan law value used to calulate dual-mono measurements
+    double pan_law;                 ///< pan law value used to calculate dual-mono measurements
 } EBUR128Context;
 
 enum {
diff --git a/libavformat/internal.h b/libavformat/internal.h
index 49244faf23..95776a09f5 100644
--- a/libavformat/internal.h
+++ b/libavformat/internal.h
@@ -650,7 +650,7 @@ int ff_bprint_to_codecpar_extradata(AVCodecParameters *par, struct AVBPrint *buf
  * The packet is not removed from the interleaving queue, but only
  * a pointer to it is returned.
  *
- * @param ts_offset the ts difference between packet in the que and the muxer.
+ * @param ts_offset the ts difference between packet in the queue and the muxer.
  *
  * @return a pointer to the next packet, or NULL if no packet is queued
  *         for this stream.
diff --git a/libavutil/frame.h b/libavutil/frame.h
index a19da2f489..8e51361e29 100644
--- a/libavutil/frame.h
+++ b/libavutil/frame.h
@@ -178,7 +178,7 @@ typedef struct AVFrameSideData {
  * without breaking compatibility with each other.
  *
  * Fields can be accessed through AVOptions, the name string used, matches the
- * C structure field name for fields accessable through AVOptions. The AVClass
+ * C structure field name for fields accessible through AVOptions. The AVClass
  * for AVFrame can be obtained from avcodec_get_frame_class()
  */
 typedef struct AVFrame {
diff --git a/libavutil/tree.h b/libavutil/tree.h
index 9a9e11b92c..d5e0aebfbd 100644
--- a/libavutil/tree.h
+++ b/libavutil/tree.h
@@ -58,7 +58,7 @@ struct AVTreeNode *av_tree_node_alloc(void);
  *             then the corresponding entry in next is unchanged.
  * @param cmp compare function used to compare elements in the tree,
  *            API identical to that of Standard C's qsort
- *            It is guranteed that the first and only the first argument to cmp()
+ *            It is guaranteed that the first and only the first argument to cmp()
  *            will be the key parameter to av_tree_find(), thus it could if the
  *            user wants, be a different type (like an opaque context).
  * @return An element with cmp(key, elem) == 0 or NULL if no such element