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apply spelling fixes

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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>
This commit is contained in:
Diederik de Haas via ffmpeg-devel 2023-11-10 16:41:54 +01:00 committed by Stefano Sabatini
parent 5452cbdc15
commit c07ed10b0e
17 changed files with 42 additions and 42 deletions
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2
doc/demuxers.texi
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@ -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 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. 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. a 10 second dts correction for 48 kHz audio streams while accommodating 99.9% of @code{uint32} range.
@item interleaved_read @item interleaved_read

48
doc/filters.texi
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@ -2306,7 +2306,7 @@ Set transform type of IIR filter.
@end table @end table
@item precision, r @item precision, r
Set precison of filtering. Set precision of filtering.
@table @option @table @option
@item auto @item auto
Pick automatic sample format depending on surround filters. Pick automatic sample format depending on surround filters.
@ -3678,7 +3678,7 @@ Set order of tilt filter.
@item level @item level
Set input volume level. Allowed range is from 0 to 4. Set input volume level. Allowed range is from 0 to 4.
Defalt is 1. Default is 1.
@end table @end table
@subsection Commands @subsection Commands
@ -3846,7 +3846,7 @@ Set transform type of IIR filter.
@end table @end table
@item precision, r @item precision, r
Set precison of filtering. Set precision of filtering.
@table @option @table @option
@item auto @item auto
Pick automatic sample format depending on surround filters. Pick automatic sample format depending on surround filters.
@ -3943,7 +3943,7 @@ Set transform type of IIR filter.
@end table @end table
@item precision, r @item precision, r
Set precison of filtering. Set precision of filtering.
@table @option @table @option
@item auto @item auto
Pick automatic sample format depending on surround filters. Pick automatic sample format depending on surround filters.
@ -4050,7 +4050,7 @@ Set transform type of IIR filter.
@end table @end table
@item precision, r @item precision, r
Set precison of filtering. Set precision of filtering.
@table @option @table @option
@item auto @item auto
Pick automatic sample format depending on surround filters. Pick automatic sample format depending on surround filters.
@ -4142,7 +4142,7 @@ Set transform type of IIR filter.
@end table @end table
@item precision, r @item precision, r
Set precison of filtering. Set precision of filtering.
@table @option @table @option
@item auto @item auto
Pick automatic sample format depending on surround filters. Pick automatic sample format depending on surround filters.
@ -4583,7 +4583,7 @@ This filter supports the all above options as @ref{commands}.
@section crystalizer @section crystalizer
Simple algorithm for audio noise sharpening. 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: The filter accepts the following options:
@ -4978,7 +4978,7 @@ Set transform type of IIR filter.
@end table @end table
@item precision, r @item precision, r
Set precison of filtering. Set precision of filtering.
@table @option @table @option
@item auto @item auto
Pick automatic sample format depending on surround filters. Pick automatic sample format depending on surround filters.
@ -5489,7 +5489,7 @@ Set transform type of IIR filter.
@end table @end table
@item precision, r @item precision, r
Set precison of filtering. Set precision of filtering.
@table @option @table @option
@item auto @item auto
Pick automatic sample format depending on surround filters. Pick automatic sample format depending on surround filters.
@ -5849,7 +5849,7 @@ Set transform type of IIR filter.
@end table @end table
@item precision, r @item precision, r
Set precison of filtering. Set precision of filtering.
@table @option @table @option
@item auto @item auto
Pick automatic sample format depending on surround filters. Pick automatic sample format depending on surround filters.
@ -7206,7 +7206,7 @@ Set transform type of IIR filter.
@end table @end table
@item precision, r @item precision, r
Set precison of filtering. Set precision of filtering.
@table @option @table @option
@item auto @item auto
Pick automatic sample format depending on surround filters. Pick automatic sample format depending on surround filters.
@ -7296,7 +7296,7 @@ Set transform type of IIR filter.
@end table @end table
@item precision, r @item precision, r
Set precison of filtering. Set precision of filtering.
@table @option @table @option
@item auto @item auto
Pick automatic sample format depending on surround filters. 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. Set the number of samples per each frame. Default is 1024.
@item taps, t @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. Default value is 0.
@item channel_layout, c @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}. Default is @code{25 40 63 100 160 250 400 630 1000 1600 2500 4000 6300 10000 16000 24000}.
@item taps, t @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}. Default value is @code{4096}.
@item sample_rate, r @item sample_rate, r
@ -7848,7 +7848,7 @@ The filter accepts the following options:
@table @option @table @option
@item taps, t @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. Default value is 1025.
@item frequency, f @item frequency, f
@ -16833,7 +16833,7 @@ ffmpeg -init_hw_device vulkan -hwaccel vaapi -hwaccel_output_format vaapi ... -v
@anchor{libvmaf} @anchor{libvmaf}
@section 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. reference/distorted pair of input videos.
The first input is the distorted video, and the second input is the reference video. 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 @end example
@item @item
Example with multiple addtional features: Example with multiple additional features:
@example @example
ffmpeg -i distorted.mpg -i reference.mpg -lavfi libvmaf='feature=name=psnr|name=ciede' -f null - ffmpeg -i distorted.mpg -i reference.mpg -lavfi libvmaf='feature=name=psnr|name=ciede' -f null -
@end example @end example
@ -20911,7 +20911,7 @@ pixel format is used.
The filter does not support converting between YUV and RGB pixel formats. The filter does not support converting between YUV and RGB pixel formats.
@item passthrough @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 This mode can be useful to use the filter as a buffer for a downstream
frame-consumer that exhausts the limited decoder frame pool. frame-consumer that exhausts the limited decoder frame pool.
@ -23041,7 +23041,7 @@ The filter accepts the following options:
@table @option @table @option
@item layout @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}. Default is @code{6x5}.
@item nb_frames @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 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 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: 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 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 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: 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 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 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: 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 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 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: 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. Passes all supported output formats to DDA and returns what DDA decides to use.
@item 8bit @item 8bit
@item bgra @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 10bit
@item x2bgr10 @item x2bgr10
Filter initialization will fail if 10 bit format is requested but unavailable. Filter initialization will fail if 10 bit format is requested but unavailable.

2
libavcodec/cbs_bsf.h
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@ -98,7 +98,7 @@ enum {
// Pass this element through unchanged. // Pass this element through unchanged.
BSF_ELEMENT_PASS, BSF_ELEMENT_PASS,
// Insert this element, replacing any existing instances of it. // 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 // or implicitly (either as side data or deduced from other parts of
// the stream). // the stream).
BSF_ELEMENT_INSERT, BSF_ELEMENT_INSERT,

2
libavdevice/pulse_audio_enc.c
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@ -504,7 +504,7 @@ static av_cold int pulse_write_header(AVFormatContext *h)
pulse_map_channels_to_pulse(&st->codecpar->ch_layout, &channel_map); pulse_map_channels_to_pulse(&st->codecpar->ch_layout, &channel_map);
/* Unknown channel is present in channel_layout, let PulseAudio use its default. */ /* Unknown channel is present in channel_layout, let PulseAudio use its default. */
if (channel_map.channels != sample_spec.channels) { 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; channel_map.channels = 0;
} }
} else } else

2
libavfilter/af_aiir.c
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@ -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"); 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) { 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); ret = convert_zp2tf(ctx, inlink->ch_layout.nb_channels);
if (ret < 0) if (ret < 0)

2
libavfilter/af_surround.c
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@ -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_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" }, { "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" }, { "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 }, { "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 }, { "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 }, { "focus", "set soundfield transform focus", OFFSET(focus), AV_OPT_TYPE_FLOAT, {.dbl=0}, -1, 1, TFLAGS },

2
libavfilter/cuda/load_helper.h
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@ -20,7 +20,7 @@
#define AVFILTER_CUDA_LOAD_HELPER_H #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, int ff_cuda_load_module(void *avctx, AVCUDADeviceContext *hwctx, CUmodule *cu_module,
const unsigned char *data, const unsigned int length); const unsigned char *data, const unsigned int length);

2
libavfilter/opencl/deshake.cl
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@ -231,7 +231,7 @@ __kernel void harris_response(
{-1, -2, -1} {-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) // block size radius of 2)
__local float grayscale_data[196]; __local float grayscale_data[196];

4
libavfilter/vf_dedot.c
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@ -111,12 +111,12 @@ static int dedotcrawl##name(AVFilterContext *ctx, void *arg, \
for (int y = slice_start; y < slice_end; y++) { \ for (int y = slice_start; y < slice_end; y++) { \
for (int x = 1; x < s->planewidth[0] - 1; x++) { \ for (int x = 1; x < s->planewidth[0] - 1; x++) { \
int above = src[x - src_linesize]; \ int above = src[x - src_linesize]; \
int bellow = src[x + src_linesize]; \ int below = src[x + src_linesize]; \
int cur = src[x]; \ int cur = src[x]; \
int left = src[x - 1]; \ int left = src[x - 1]; \
int right = 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) \ FFABS(left + right - 2 * cur) <= luma2d) \
continue; \ continue; \
\ \

2
libavfilter/vf_transpose_npp.c
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@ -300,7 +300,7 @@ static int npptranspose_rotate(AVFilterContext *ctx, NPPTransposeStageContext *s
// nppRotate uses 0,0 as the rotation point // nppRotate uses 0,0 as the rotation point
// need to shift the image accordingly after rotation // 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; 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 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; int shifth = (s->dir == NPP_TRANSPOSE_CCLOCK || s->dir == NPP_TRANSPOSE_CLOCK_FLIP) ? oh - 1 : 0;

2
libavformat/dashenc.c
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@ -1454,7 +1454,7 @@ static int dash_init(AVFormatContext *s)
} }
if (av_cmp_q(c->max_playback_rate, c->min_playback_rate) < 0) { if (av_cmp_q(c->max_playback_rate, c->min_playback_rate) < 0) {
av_log(s, AV_LOG_WARNING, "Minimum playback rate value is higer than the Maximum. Both will be ignored\n"); av_log(s, AV_LOG_WARNING, "Minimum playback rate value is higher than the Maximum. Both will be ignored\n");
c->min_playback_rate = c->max_playback_rate = (AVRational) {1, 1}; c->min_playback_rate = c->max_playback_rate = (AVRational) {1, 1};
} }

2
libavformat/demux.h
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@ -169,7 +169,7 @@ void ff_rfps_calculate(AVFormatContext *ic);
* Useful to simplify the rescaling of the arguments of AVInputFormat::read_seek2() * Useful to simplify the rescaling of the arguments of AVInputFormat::read_seek2()
* *
* @param[in] tb_in Timebase of the input `min_ts`, `ts` and `max_ts` * @param[in] tb_in Timebase of the input `min_ts`, `ts` and `max_ts`
* @param[in] tb_out Timebase of the ouput `min_ts`, `ts` and `max_ts` * @param[in] tb_out Timebase of the output `min_ts`, `ts` and `max_ts`
* @param[in,out] min_ts Lower bound of the interval * @param[in,out] min_ts Lower bound of the interval
* @param[in,out] ts Timestamp * @param[in,out] ts Timestamp
* @param[in,out] max_ts Upper bound of the interval * @param[in,out] max_ts Upper bound of the interval

2
libavformat/scd.c
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@ -187,7 +187,7 @@ static int scd_read_track(AVFormatContext *s, SCDTrackHeader *track, int index)
/* Not sure what to do with these, it seems to be fine to ignore them. */ /* Not sure what to do with these, it seems to be fine to ignore them. */
if (track->aux_count != 0) if (track->aux_count != 0)
av_log(s, AV_LOG_DEBUG, "[%d] Track has %u auxillary chunk(s).\n", index, track->aux_count); av_log(s, AV_LOG_DEBUG, "[%d] Track has %u auxiliary chunk(s).\n", index, track->aux_count);
if ((st = avformat_new_stream(s, NULL)) == NULL) if ((st = avformat_new_stream(s, NULL)) == NULL)
return AVERROR(ENOMEM); return AVERROR(ENOMEM);

2
libavutil/eval.h
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@ -105,7 +105,7 @@ int av_expr_count_vars(AVExpr *e, unsigned *counter, int size);
* @param e the AVExpr to track user provided functions in * @param e the AVExpr to track user provided functions in
* @param counter a zero-initialized array where the count of each function will be stored * @param counter a zero-initialized array where the count of each function will be stored
* if you passed 5 functions with 2 arguments to av_expr_parse() * if you passed 5 functions with 2 arguments to av_expr_parse()
* then for arg=2 this will use upto 5 entries. * then for arg=2 this will use up to 5 entries.
* @param size size of array * @param size size of array
* @param arg number of arguments the counted functions have * @param arg number of arguments the counted functions have
* @return 0 on success, a negative value indicates that no expression or array was passed * @return 0 on success, a negative value indicates that no expression or array was passed

4
libavutil/hwcontext_vulkan.c
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@ -3072,7 +3072,7 @@ static int vulkan_transfer_data_from_cuda(AVHWFramesContext *hwfc,
CHECK_CU(cu->cuCtxPopCurrent(&dummy)); CHECK_CU(cu->cuCtxPopCurrent(&dummy));
av_log(hwfc, AV_LOG_VERBOSE, "Transfered CUDA image to Vulkan!\n"); av_log(hwfc, AV_LOG_VERBOSE, "Transferred CUDA image to Vulkan!\n");
return err = prepare_frame(hwfc, &fp->upload_exec, dst_f, PREP_MODE_EXTERNAL_IMPORT); return err = prepare_frame(hwfc, &fp->upload_exec, dst_f, PREP_MODE_EXTERNAL_IMPORT);
@ -3648,7 +3648,7 @@ static int vulkan_transfer_data_to_cuda(AVHWFramesContext *hwfc, AVFrame *dst,
CHECK_CU(cu->cuCtxPopCurrent(&dummy)); CHECK_CU(cu->cuCtxPopCurrent(&dummy));
av_log(hwfc, AV_LOG_VERBOSE, "Transfered Vulkan image to CUDA!\n"); av_log(hwfc, AV_LOG_VERBOSE, "Transferred Vulkan image to CUDA!\n");
return prepare_frame(hwfc, &fp->upload_exec, dst_f, PREP_MODE_EXTERNAL_IMPORT); return prepare_frame(hwfc, &fp->upload_exec, dst_f, PREP_MODE_EXTERNAL_IMPORT);

2
tools/enc_recon_frame_test.c
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@ -297,7 +297,7 @@ int main(int argc, char **argv)
return 1; return 1;
} }
if (!(enc->capabilities & AV_CODEC_CAP_ENCODER_RECON_FRAME)) { if (!(enc->capabilities & AV_CODEC_CAP_ENCODER_RECON_FRAME)) {
fprintf(stderr, "Encoder '%s' cannot ouput reconstructed frames\n", fprintf(stderr, "Encoder '%s' cannot output reconstructed frames\n",
enc->name); enc->name);
return 1; return 1;
} }

2
tools/patcheck
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@ -66,7 +66,7 @@ $EGREP $OPT '^\+ *(const *|)static' $*| $EGREP --color=always '[^=]= *(0|NULL)[^
cat $TMP cat $TMP
hiegrep '# *ifdef * (HAVE|CONFIG)_' 'ifdefs that should be #if' $* hiegrep '# *ifdef * (HAVE|CONFIG)_' 'ifdefs that should be #if' $*
hiegrep '\b(awnser|cant|dont|wont|doesnt|usefull|successfull|occured|teh|alot|wether|skiped|skiping|heigth|informations|colums|loosy|loosing|ouput|seperate|preceed|upto|paket|posible|unkown|inpossible|dimention|acheive|funtions|overriden|outputing|seperation|initalize|compatibilty|bistream|knwon|unknwon|choosen|additonal|gurantee|availble|wich|begining|milisecond|missmatch|threshhold)\b' 'common typos' $* hiegrep '\b(awnser|cant|dont|wont|doesnt|usefull|successfull|occured|teh|alot|wether|skiped|skiping|heigth|informations|colums|loosy|loosing|ouput|seperate|preceed|upto|paket|posible|unkown|inpossible|dimention|acheive|funtions|overriden|outputing|seperation|initalize|compatibilty|bistream|knwon|unknwon|choosen|additonal|gurantee|availble|wich|begining|milisecond|missmatch|threshhold|accomodate|processsing|substract|auxillary|coefficents|neccesary|precison)\b' 'common typos' $*
hiegrep 'av_log\( *NULL' 'Missing context in av_log' $* hiegrep 'av_log\( *NULL' 'Missing context in av_log' $*
hiegrep '[^sn]printf' 'Please use av_log' $* hiegrep '[^sn]printf' 'Please use av_log' $*