apply spelling fixes

Fix spelling issue as reported by Debian's lintian tool:
accomodate -> accommodate
addtional -> additional
auxillary -> auxiliary
bellow -> below
betweeen -> between
Calulate -> Calculate
coefficents -> coefficients
Defalt -> Default
defaul -> default
higer -> higher
neccesary -> necessary
orignal -> original
ouput -> output
precison -> precision
processsing -> processing
substract -> subtract
Transfered -> Transferred
upto -> up to

Also add several of them to the 'common typos' check in patcheck.

Signed-off-by: Diederik de Haas <didi.debian@cknow.org>

doc/demuxers.texi

@@ -777,7 +777,7 @@ error or used to store a negative value for dts correction when treated as signe
 the user set an upper limit, beyond which the delta is clamped to 1. Values greater than the limit if negative when
 cast to int32 are used to adjust onward dts.
 
-Unit is the track time scale. Range is 0 to UINT_MAX. Default is @code{UINT_MAX - 48000*10} which allows upto
+Unit is the track time scale. Range is 0 to UINT_MAX. Default is @code{UINT_MAX - 48000*10} which allows up to
 a 10 second dts correction for 48 kHz audio streams while accommodating 99.9% of @code{uint32} range.
 
 @item interleaved_read

doc/filters.texi

@@ -2306,7 +2306,7 @@ Set transform type of IIR filter.
 @end table
 
 @item precision, r
-Set precison of filtering.
+Set precision of filtering.
 @table @option
 @item auto
 Pick automatic sample format depending on surround filters.
@@ -3678,7 +3678,7 @@ Set order of tilt filter.
 
 @item level
 Set input volume level. Allowed range is from 0 to 4.
-Defalt is 1.
+Default is 1.
 @end table
 
 @subsection Commands
@@ -3846,7 +3846,7 @@ Set transform type of IIR filter.
 @end table
 
 @item precision, r
-Set precison of filtering.
+Set precision of filtering.
 @table @option
 @item auto
 Pick automatic sample format depending on surround filters.
@@ -3943,7 +3943,7 @@ Set transform type of IIR filter.
 @end table
 
 @item precision, r
-Set precison of filtering.
+Set precision of filtering.
 @table @option
 @item auto
 Pick automatic sample format depending on surround filters.
@@ -4050,7 +4050,7 @@ Set transform type of IIR filter.
 @end table
 
 @item precision, r
-Set precison of filtering.
+Set precision of filtering.
 @table @option
 @item auto
 Pick automatic sample format depending on surround filters.
@@ -4142,7 +4142,7 @@ Set transform type of IIR filter.
 @end table
 
 @item precision, r
-Set precison of filtering.
+Set precision of filtering.
 @table @option
 @item auto
 Pick automatic sample format depending on surround filters.
@@ -4583,7 +4583,7 @@ This filter supports the all above options as @ref{commands}.
 @section crystalizer
 Simple algorithm for audio noise sharpening.
 
-This filter linearly increases differences betweeen each audio sample.
+This filter linearly increases differences between each audio sample.
 
 The filter accepts the following options:
 
@@ -4978,7 +4978,7 @@ Set transform type of IIR filter.
 @end table
 
 @item precision, r
-Set precison of filtering.
+Set precision of filtering.
 @table @option
 @item auto
 Pick automatic sample format depending on surround filters.
@@ -5489,7 +5489,7 @@ Set transform type of IIR filter.
 @end table
 
 @item precision, r
-Set precison of filtering.
+Set precision of filtering.
 @table @option
 @item auto
 Pick automatic sample format depending on surround filters.
@@ -5849,7 +5849,7 @@ Set transform type of IIR filter.
 @end table
 
 @item precision, r
-Set precison of filtering.
+Set precision of filtering.
 @table @option
 @item auto
 Pick automatic sample format depending on surround filters.
@@ -7206,7 +7206,7 @@ Set transform type of IIR filter.
 @end table
 
 @item precision, r
-Set precison of filtering.
+Set precision of filtering.
 @table @option
 @item auto
 Pick automatic sample format depending on surround filters.
@@ -7296,7 +7296,7 @@ Set transform type of IIR filter.
 @end table
 
 @item precision, r
-Set precison of filtering.
+Set precision of filtering.
 @table @option
 @item auto
 Pick automatic sample format depending on surround filters.
@@ -7765,7 +7765,7 @@ Set the sample rate, default is 44100.
 Set the number of samples per each frame. Default is 1024.
 
 @item taps, t
-Set the number of filter coefficents in output audio stream.
+Set the number of filter coefficients in output audio stream.
 Default value is 0.
 
 @item channel_layout, c
@@ -7821,7 +7821,7 @@ Bands are separated by white spaces and each band represent frequency in Hz.
 Default is @code{25 40 63 100 160 250 400 630 1000 1600 2500 4000 6300 10000 16000 24000}.
 
 @item taps, t
-Set number of filter coefficents in output audio stream.
+Set number of filter coefficients in output audio stream.
 Default value is @code{4096}.
 
 @item sample_rate, r
@@ -7848,7 +7848,7 @@ The filter accepts the following options:
 
 @table @option
 @item taps, t
-Set number of filter coefficents in output audio stream.
+Set number of filter coefficients in output audio stream.
 Default value is 1025.
 
 @item frequency, f
@@ -16833,7 +16833,7 @@ ffmpeg -init_hw_device vulkan -hwaccel vaapi -hwaccel_output_format vaapi ... -v
 @anchor{libvmaf}
 @section libvmaf
 
-Calulate the VMAF (Video Multi-Method Assessment Fusion) score for a
+Calculate the VMAF (Video Multi-Method Assessment Fusion) score for a
 reference/distorted pair of input videos.
 
 The first input is the distorted video, and the second input is the reference video.
@@ -16889,7 +16889,7 @@ ffmpeg -i distorted.mpg -i reference.mpg -lavfi libvmaf='model=version=vmaf_v0.6
 @end example
 
 @item
-Example with multiple addtional features:
+Example with multiple additional features:
 @example
 ffmpeg -i distorted.mpg -i reference.mpg -lavfi libvmaf='feature=name=psnr|name=ciede' -f null -
 @end example
@@ -20911,7 +20911,7 @@ pixel format is used.
 The filter does not support converting between YUV and RGB pixel formats.
 
 @item passthrough
-If set to 0, every frame is processed, even if no conversion is neccesary.
+If set to 0, every frame is processed, even if no conversion is necessary.
 This mode can be useful to use the filter as a buffer for a downstream
 frame-consumer that exhausts the limited decoder frame pool.
 
@@ -23041,7 +23041,7 @@ The filter accepts the following options:
 @table @option
 
 @item layout
-Set the grid size in the form @code{COLUMNSxROWS}. Range is upto UINT_MAX cells.
+Set the grid size in the form @code{COLUMNSxROWS}. Range is up to UINT_MAX cells.
 Default is @code{6x5}.
 
 @item nb_frames
@@ -27339,7 +27339,7 @@ Stack input videos horizontally.
 
 This is the VA-API variant of the @ref{hstack} filter, each input stream may
 have different height, this filter will scale down/up each input stream while
-keeping the orignal aspect.
+keeping the original aspect.
 
 It accepts the following options:
 
@@ -27360,7 +27360,7 @@ Stack input videos vertically.
 
 This is the VA-API variant of the @ref{vstack} filter, each input stream may
 have different width, this filter will scale down/up each input stream while
-keeping the orignal aspect.
+keeping the original aspect.
 
 It accepts the following options:
 
@@ -27821,7 +27821,7 @@ Stack input videos horizontally.
 
 This is the QSV variant of the @ref{hstack} filter, each input stream may
 have different height, this filter will scale down/up each input stream while
-keeping the orignal aspect.
+keeping the original aspect.
 
 It accepts the following options:
 
@@ -27842,7 +27842,7 @@ Stack input videos vertically.
 
 This is the QSV variant of the @ref{vstack} filter, each input stream may
 have different width, this filter will scale down/up each input stream while
-keeping the orignal aspect.
+keeping the original aspect.
 
 It accepts the following options:
 
@@ -28190,7 +28190,7 @@ It accepts the following values:
 Passes all supported output formats to DDA and returns what DDA decides to use.
 @item 8bit
 @item bgra
-8 Bit formats always work, and DDA will convert to them if neccesary.
+8 Bit formats always work, and DDA will convert to them if necessary.
 @item 10bit
 @item x2bgr10
 Filter initialization will fail if 10 bit format is requested but unavailable.

libavcodec/cbs_bsf.h

@@ -98,7 +98,7 @@ enum {
     // Pass this element through unchanged.
     BSF_ELEMENT_PASS,
     // Insert this element, replacing any existing instances of it.
-    // Associated values may be provided explicitly (as addtional options)
+    // Associated values may be provided explicitly (as additional options)
     // or implicitly (either as side data or deduced from other parts of
     // the stream).
     BSF_ELEMENT_INSERT,

libavdevice/pulse_audio_enc.c

@@ -504,7 +504,7 @@ static av_cold int pulse_write_header(AVFormatContext *h)
         pulse_map_channels_to_pulse(&st->codecpar->ch_layout, &channel_map);
         /* Unknown channel is present in channel_layout, let PulseAudio use its default. */
         if (channel_map.channels != sample_spec.channels) {
-            av_log(s, AV_LOG_WARNING, "Unknown channel. Using defaul channel map.\n");
+            av_log(s, AV_LOG_WARNING, "Unknown channel. Using default channel map.\n");
             channel_map.channels = 0;
         }
     } else

libavfilter/af_aiir.c

@@ -1309,7 +1309,7 @@ static int config_output(AVFilterLink *outlink)
         av_log(ctx, AV_LOG_WARNING, "transfer function coefficients format is not recommended for too high number of zeros/poles.\n");
 
     if (s->format > 0 && s->process == 0) {
-        av_log(ctx, AV_LOG_WARNING, "Direct processsing is not recommended for zp coefficients format.\n");
+        av_log(ctx, AV_LOG_WARNING, "Direct processing is not recommended for zp coefficients format.\n");
 
         ret = convert_zp2tf(ctx, inlink->ch_layout.nb_channels);
         if (ret < 0)

libavfilter/af_surround.c

@@ -1426,7 +1426,7 @@ static const AVOption surround_options[] = {
     { "lfe_high",  "LFE high cut off",          OFFSET(highcutf),               AV_OPT_TYPE_INT,    {.i64=256},   0, 512, FLAGS },
     { "lfe_mode",  "set LFE channel mode",      OFFSET(lfe_mode),               AV_OPT_TYPE_INT,    {.i64=0},     0,   1, TFLAGS, "lfe_mode" },
     {  "add",      "just add LFE channel",                  0,                  AV_OPT_TYPE_CONST,  {.i64=0},     0,   1, TFLAGS, "lfe_mode" },
-    {  "sub",      "substract LFE channel with others",     0,                  AV_OPT_TYPE_CONST,  {.i64=1},     0,   1, TFLAGS, "lfe_mode" },
+    {  "sub",      "subtract LFE channel with others",      0,                  AV_OPT_TYPE_CONST,  {.i64=1},     0,   1, TFLAGS, "lfe_mode" },
     { "smooth",    "set temporal smoothness strength",      OFFSET(smooth),     AV_OPT_TYPE_FLOAT,  {.dbl=0},     0,   1, TFLAGS },
     { "angle",     "set soundfield transform angle",        OFFSET(angle),      AV_OPT_TYPE_FLOAT,  {.dbl=90},    0, 360, TFLAGS },
     { "focus",     "set soundfield transform focus",        OFFSET(focus),      AV_OPT_TYPE_FLOAT,  {.dbl=0},    -1,   1, TFLAGS },

libavfilter/cuda/load_helper.h

@@ -20,7 +20,7 @@
 #define AVFILTER_CUDA_LOAD_HELPER_H
 
 /**
- * Loads a CUDA module and applies any decompression, if neccesary.
+ * Loads a CUDA module and applies any decompression, if necessary.
  */
 int ff_cuda_load_module(void *avctx, AVCUDADeviceContext *hwctx, CUmodule *cu_module,
                         const unsigned char *data, const unsigned int length);

libavfilter/opencl/deshake.cl

@@ -231,7 +231,7 @@ __kernel void harris_response(
         {-1, -2, -1}
     };
 
-    // 8 x 8 local work + 3 pixels around each side (needed to accomodate for the
+    // 8 x 8 local work + 3 pixels around each side (needed to accommodate for the
     // block size radius of 2)
     __local float grayscale_data[196];
 

libavfilter/vf_dedot.c

@@ -111,12 +111,12 @@ static int dedotcrawl##name(AVFilterContext *ctx, void *arg,     \
     for (int y = slice_start; y < slice_end; y++) {              \
         for (int x = 1; x < s->planewidth[0] - 1; x++) {         \
             int above = src[x - src_linesize];                   \
-            int bellow = src[x + src_linesize];                  \
+            int below = src[x + src_linesize];                   \
             int cur = src[x];                                    \
             int left = src[x - 1];                               \
             int right = src[x + 1];                              \
                                                                  \
-            if (FFABS(above + bellow - 2 * cur) <= luma2d &&     \
+            if (FFABS(above + below - 2 * cur) <= luma2d &&      \
                 FFABS(left + right - 2 * cur) <= luma2d)         \
                 continue;                                        \
                                                                  \

libavfilter/vf_transpose_npp.c

@@ -300,7 +300,7 @@ static int npptranspose_rotate(AVFilterContext *ctx, NPPTransposeStageContext *s
 
         // nppRotate uses 0,0 as the rotation point
         // need to shift the image accordingly after rotation
-        // need to substract 1 to get the correct coordinates
+        // need to subtract 1 to get the correct coordinates
         double angle = s->dir == NPP_TRANSPOSE_CLOCK ? -90.0 : s->dir == NPP_TRANSPOSE_CCLOCK ? 90.0 : 180.0;
         int shiftw = (s->dir == NPP_TRANSPOSE_CLOCK  || s->dir == NPP_TRANSPOSE_CLOCK_FLIP) ? ow - 1 : 0;
         int shifth = (s->dir == NPP_TRANSPOSE_CCLOCK || s->dir == NPP_TRANSPOSE_CLOCK_FLIP) ? oh - 1 : 0;