Commit 41ed7ab4 authored by Vittorio Giovara's avatar Vittorio Giovara Committed by Diego Biurrun

cosmetics: Fix spelling mistakes

Signed-off-by: default avatarDiego Biurrun <diego@biurrun.de>
parent 5c31eaa9
......@@ -377,7 +377,7 @@ version 0.7_beta2:
- DPX image encoder
- SMPTE 302M AES3 audio decoder
- ffmpeg no longer quits after the 'q' key is pressed; use 'ctrl+c' instead
- 9bit and 10bit per sample support in the H.264 decoder
- 9 bits and 10 bits per sample support in the H.264 decoder
version 0.7_beta1:
......@@ -732,7 +732,7 @@ version 0.4.9-pre1:
- rate distorted optimal lambda->qp support
- AAC encoding with libfaac
- Sunplus JPEG codec (SP5X) support
- use Lagrange multipler instead of QP for ratecontrol
- use Lagrange multiplier instead of QP for ratecontrol
- Theora/VP3 decoding support
- XA and ADX ADPCM codecs
- export MPEG-2 active display area / pan scan
......
/*
* avconv main
* Copyright (c) 2000-2011 The libav developers.
* Copyright (c) 2000-2011 The Libav developers
*
* This file is part of Libav.
*
......@@ -1154,7 +1154,7 @@ int guess_input_channel_layout(InputStream *ist)
return 0;
av_get_channel_layout_string(layout_name, sizeof(layout_name),
dec->channels, dec->channel_layout);
av_log(NULL, AV_LOG_WARNING, "Guessed Channel Layout for Input Stream "
av_log(NULL, AV_LOG_WARNING, "Guessed Channel Layout for Input Stream "
"#%d.%d : %s\n", ist->file_index, ist->st->index, layout_name);
}
return 1;
......@@ -1630,7 +1630,7 @@ static int init_output_bsfs(OutputStream *ost)
for (i = 0; i < ost->nb_bitstream_filters; i++) {
ret = av_bsf_alloc(ost->bitstream_filters[i], &ctx);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error allocating a bistream filter context\n");
av_log(NULL, AV_LOG_ERROR, "Error allocating a bitstream filter context\n");
return ret;
}
ost->bsf_ctx[i] = ctx;
......@@ -1644,7 +1644,7 @@ static int init_output_bsfs(OutputStream *ost)
ret = av_bsf_init(ctx);
if (ret < 0) {
av_log(NULL, AV_LOG_ERROR, "Error initializing bistream filter: %s\n",
av_log(NULL, AV_LOG_ERROR, "Error initializing bitstream filter: %s\n",
ost->bitstream_filters[i]->name);
return ret;
}
......
......@@ -760,7 +760,7 @@ static int open_input_file(OptionsContext *o, const char *filename)
ic->flags |= AVFMT_FLAG_NONBLOCK;
ic->interrupt_callback = int_cb;
/* open the input file with generic libav function */
/* open the input file with generic Libav function */
err = avformat_open_input(&ic, filename, file_iformat, &o->g->format_opts);
if (err < 0) {
print_error(filename, err);
......
......@@ -1362,7 +1362,7 @@ static int output_picture2(PlayerState *is, AVFrame *src_frame, double pts1, int
}
/* update video clock for next frame */
frame_delay = av_q2d(is->video_dec->time_base);
/* for MPEG2, the frame can be repeated, so we update the
/* For MPEG-2, the frame can be repeated, so we update the
clock accordingly */
frame_delay += src_frame->repeat_pict * (frame_delay * 0.5);
is->video_clock += frame_delay;
......@@ -2123,7 +2123,7 @@ static int stream_component_open(PlayerState *is, int stream_index)
/* init averaging filter */
is->audio_diff_avg_coef = exp(log(0.01) / AUDIO_DIFF_AVG_NB);
is->audio_diff_avg_count = 0;
/* since we do not have a precise anough audio fifo fullness,
/* since we do not have a precise enough audio FIFO fullness,
we correct audio sync only if larger than this threshold */
is->audio_diff_threshold = 2.0 * SDL_AUDIO_BUFFER_SIZE / avctx->sample_rate;
......
......@@ -62,7 +62,7 @@ static int use_value_sexagesimal_format = 0;
/* globals */
static const OptionDef *options;
/* AVprobe context */
/* avprobe context */
static const char *input_filename;
static AVInputFormat *iformat = NULL;
......
......@@ -23,7 +23,7 @@ API changes, most recent first:
2016-xx-xx - xxxxxxx - lavc 57.15.0 - avcodec.h
Add a new bitstream filtering API working with AVPackets.
Deprecate the old bistream filtering API.
Deprecate the old bitstream filtering API.
2016-xx-xx - xxxxxxx - lavfi 6.3.0 - avfilter.h
Add AVFilterContext.hw_device_ctx.
......@@ -501,7 +501,7 @@ API changes, most recent first:
2013-08-05 - f824535 - lavc 55.13.0 - avcodec.h
Deprecate the bitstream-related members from struct AVVDPAUContext.
The bistream buffers no longer need to be explicitly freed.
The bitstream buffers no longer need to be explicitly freed.
2013-08-05 - 549294f - lavc 55.12.0 - avcodec.h
Deprecate the CODEC_CAP_HWACCEL_VDPAU codec capability. Use CODEC_CAP_HWACCEL
......@@ -591,7 +591,7 @@ lavd 54.0.0, lavfi 3.5.0
* base -- is now stored in AVBufferRef
* reference, type, buffer_hints -- are unnecessary in the new API
* hwaccel_picture_private, owner, thread_opaque -- should not
have been acessed from outside of lavc
have been accessed from outside of lavc
* qscale_table, qstride, qscale_type, mbskip_table, motion_val,
mb_type, dct_coeff, ref_index -- mpegvideo-specific tables,
which are not exported anymore.
......
......@@ -641,7 +641,7 @@ For DXVA2, this option should contain the number of the display adapter to use.
If this option is not specified, the default adapter is used.
@item qsv
For QSV, this option corresponds to the valus of MFX_IMPL_* . Allowed values
For QSV, this option corresponds to the values of MFX_IMPL_* . Allowed values
are:
@table @option
@item auto
......@@ -886,7 +886,7 @@ avconv -i infile -streamid 0:33 -streamid 1:36 out.ts
@end example
@item -bsf[:@var{stream_specifier}] @var{bitstream_filters} (@emph{output,per-stream})
Set bitstream filters for matching streams. @var{bistream_filters} is
Set bitstream filters for matching streams. @var{bitstream_filters} is
a comma-separated list of bitstream filters. Use the @code{-bsfs} option
to get the list of bitstream filters.
@example
......
......@@ -20,7 +20,7 @@ avplay [options] @file{input_file}
@chapter Description
@c man begin DESCRIPTION
AVplay is a very simple and portable media player using the Libav
avplay is a very simple and portable media player using the Libav
libraries and the SDL library. It is mostly used as a testbed for the
various Libav APIs.
@c man end
......@@ -171,7 +171,7 @@ Seek to percentage in file corresponding to fraction of width.
@ignore
@setfilename avplay
@settitle AVplay media player
@settitle avplay media player
@c man begin SEEALSO
avconv(1), avprobe(1) and the Libav HTML documentation
......
......@@ -19,7 +19,7 @@ are used to precisely specify which stream(s) does a given option belong to.
A stream specifier is a string generally appended to the option name and
separated from it by a colon. E.g. @code{-codec:a:1 ac3} option contains
@code{a:1} stream specifer, which matches the second audio stream. Therefore it
@code{a:1} stream specifier, which matches the second audio stream. Therefore it
would select the ac3 codec for the second audio stream.
A stream specifier can match several stream, the option is then applied to all
......
......@@ -15,7 +15,7 @@ md5.c MD5 Message-Digest Algorithm
rational.c code to perform exact calculations with rational numbers
tree.c generic AVL tree
crc.c generic CRC checksumming code
integer.c 128bit integer math
integer.c 128-bit integer math
lls.c
mathematics.c greatest common divisor, integer sqrt, integer log2, ...
mem.c memory allocation routines with guaranteed alignment
......
......@@ -23,7 +23,7 @@ player. See @file{doc/examples/output.c} to use it to generate
audio or video streams.
@end itemize
@section Integrating libav in your program
@section Integrating Libav in your program
Shared libraries should be used whenever is possible in order to reduce
the effort distributors have to pour to support programs and to ensure
......@@ -617,7 +617,7 @@ least make sure that it does not break anything.
If the code changed has already a test present in FATE you should run it,
otherwise it is advised to add it.
Improvements to codec or demuxer might change the FATE results. Make sure
Improvements to a codec or demuxer might change the FATE results. Make sure
to commit the update reference with the change and to explain in the comment
why the expected result changed.
......
......@@ -803,8 +803,8 @@ Use @var{0} to disable alpha plane coding.
@subsection Speed considerations
In the default mode of operation the encoder has to honor frame constraints
(i.e. not produc frames with size bigger than requested) while still making
output picture as good as possible.
(i.e. not produce frames with a size larger than requested) while still making
the output picture as good as possible.
A frame containing a lot of small details is harder to compress and the encoder
would spend more time searching for appropriate quantizers for each slice.
......
......@@ -23,9 +23,9 @@
* libavcodec API use example.
*
* @example avcodec.c
* Note that this library only handles codecs (mpeg, mpeg4, etc...),
* not file formats (avi, vob, etc...). See library 'libavformat' for the
* format handling
* Note that this library only handles codecs (MPEG, MPEG-4, etc...),
* not file formats (AVI, VOB, etc...). See library 'libavformat' for the
* format handling.
*/
#include <stdlib.h>
......@@ -234,7 +234,7 @@ static void audio_decode_example(const char *outfilename, const char *filename)
printf("Audio decoding\n");
/* find the mpeg audio decoder */
/* find the MPEG audio decoder */
codec = avcodec_find_decoder(AV_CODEC_ID_MP2);
if (!codec) {
fprintf(stderr, "codec not found\n");
......@@ -325,7 +325,7 @@ static void video_encode_example(const char *filename)
printf("Video encoding\n");
/* find the mpeg1 video encoder */
/* find the mpeg1video encoder */
codec = avcodec_find_encoder(AV_CODEC_ID_MPEG1VIDEO);
if (!codec) {
fprintf(stderr, "codec not found\n");
......@@ -424,7 +424,7 @@ static void video_encode_example(const char *filename)
}
}
/* add sequence end code to have a real mpeg file */
/* add sequence end code to have a real MPEG file */
fwrite(endcode, 1, sizeof(endcode), f);
fclose(f);
......@@ -465,12 +465,12 @@ static void video_decode_example(const char *outfilename, const char *filename)
av_init_packet(&avpkt);
/* set end of buffer to 0 (this ensures that no overreading happens for damaged mpeg streams) */
/* set end of buffer to 0 (this ensures that no overreading happens for damaged MPEG streams) */
memset(inbuf + INBUF_SIZE, 0, AV_INPUT_BUFFER_PADDING_SIZE);
printf("Video decoding\n");
/* find the mpeg1 video decoder */
/* find the MPEG-1 video decoder */
codec = avcodec_find_decoder(AV_CODEC_ID_MPEG1VIDEO);
if (!codec) {
fprintf(stderr, "codec not found\n");
......@@ -545,9 +545,9 @@ static void video_decode_example(const char *outfilename, const char *filename)
}
}
/* some codecs, such as MPEG, transmit the I and P frame with a
/* Some codecs, such as MPEG, transmit the I- and P-frame with a
latency of one frame. You must do the following to have a
chance to get the last frame of the video */
chance to get the last frame of the video. */
avpkt.data = NULL;
avpkt.size = 0;
len = avcodec_decode_video2(c, picture, &got_picture, &avpkt);
......
......@@ -365,7 +365,7 @@ static void add_video_stream(OutputStream *ost, AVFormatContext *oc,
c->gop_size = 12; /* emit one intra frame every twelve frames at most */
c->pix_fmt = STREAM_PIX_FMT;
if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) {
/* just for testing, we also add B frames */
/* just for testing, we also add B-frames */
c->max_b_frames = 2;
}
if (c->codec_id == AV_CODEC_ID_MPEG1VIDEO) {
......
......@@ -11,7 +11,7 @@
@chapter Introduction
FATE provides a regression testsuite embedded within the Libav build system.
FATE provides a regression test suite embedded within the Libav build system.
It can be run locally and optionally configured to send reports to a web
aggregator and viewer @url{http://fate.libav.org}.
......@@ -24,7 +24,7 @@ and provide new tests when submitting patches to add additional features.
In order to run, FATE needs a large amount of data (samples and references)
that is provided separately from the actual source distribution.
To inform the build system about the testsuite location, pass
To inform the build system about the test suite location, pass
@option{--samples=<path to the samples>} to @command{configure} or set the
@var{SAMPLES} Make variable or the @var{LIBAV_SAMPLES} environment variable
to a suitable value.
......@@ -57,7 +57,7 @@ Specific Makefile targets and Makefile variables are available:
List all fate/regression test targets.
@item fate-rsync
Shortcut to download the fate test samples to the specified testsuite location.
Shortcut to download the fate test samples to the specified test suite location.
@item fate
Run the FATE test suite (requires the fate-suite dataset).
......
......@@ -1541,7 +1541,7 @@ For more information, see
@section gradfun
Fix the banding artifacts that are sometimes introduced into nearly flat
regions by truncation to 8bit colordepth.
regions by truncation to 8-bit colordepth.
Interpolate the gradients that should go where the bands are, and
dither them.
......@@ -1900,7 +1900,7 @@ libopencv function @code{cvSmooth}.
Overlay one video on top of another.
It takes two inputs and has one output. The first input is the "main"
video on which the second input is overlayed.
video on which the second input is overlaid.
It accepts the following parameters:
......@@ -2327,7 +2327,7 @@ select='not(mod(n\,100))'
# Select only frames contained in the 10-20 time interval
select='gte(t\,10)*lte(t\,20)'
# Select only I frames contained in the 10-20 time interval
# Select only I-frames contained in the 10-20 time interval
select='gte(t\,10)*lte(t\,20)*eq(pict_type\,I)'
# Select frames with a minimum distance of 10 seconds
......@@ -2957,7 +2957,7 @@ number or a valid video frame rate abbreviation. The default value is
The following graph description will generate a red source
with an opacity of 0.2, with size "qcif" and a frame rate of 10
frames per second, which will be overlayed over the source connected
frames per second, which will be overlaid over the source connected
to the pad with identifier "in":
@example
......@@ -3070,7 +3070,7 @@ A '|'-separated list of parameters to pass to the frei0r source.
An example:
@example
# Generate a frei0r partik0l source with size 200x200 and framerate 10
# which is overlayed on the overlay filter main input
# which is overlaid on the overlay filter's main input
frei0r_src=size=200x200:framerate=10:filter_name=partik0l:filter_params=1234 [overlay]; [in][overlay] overlay
@end example
......
......@@ -399,7 +399,7 @@ to ensure no local changes still need to be committed and that no local
changes may have thrown off the results of your testing.
@end itemize
Next let the code pass through a full run of our testsuite. Before you do,
Next let the code pass through a full run of our test suite. Before you do,
the command @command{make fate-rsync} will update the test samples. Changes
to the samples set are not very common and commits depending on samples
changes are delayed for at least 24 hours to allow the new samples to
......
......@@ -369,7 +369,7 @@ The syntax is:
-grab_x @var{x_offset} -grab_y @var{y_offset}
@end example
Set the grabing region coordinates. The are expressed as offset from the top left
Set the grabbing region coordinates. The are expressed as offset from the top left
corner of the X11 window. The default value is 0.
@c man end INPUT DEVICES
......@@ -29,7 +29,7 @@ NUT has some variants signaled by using the flags field in its main header.
The BROADCAST variant provides a secondary time reference to facilitate
detecting endpoint latency and network delays.
It assumes all the endpoint clocks are syncronized.
It assumes all the endpoint clocks are synchronized.
To be used in real-time scenarios.
@section PIPE
......
......@@ -7,7 +7,7 @@ If you plan to do non-x86 architecture specific optimizations (SIMD normally),
then take a look in the x86/ directory, as most important functions are
already optimized for MMX.
If you want to do x86 optimizations then you can either try to finetune the
If you want to do x86 optimizations then you can either try to fine-tune the
stuff in the x86 directory or find some other functions in the C source to
optimize, but there aren't many left.
......@@ -163,7 +163,7 @@ general x86 registers (e.g. eax) as well as XMM registers. This last one is
particularly important on Win64, where xmm6-15 are callee-save, and not
restoring their contents leads to undefined results. In external asm (e.g.
yasm), you do this by using:
cglobal functon_name, num_args, num_regs, num_xmm_regs
cglobal function_name, num_args, num_regs, num_xmm_regs
In inline asm, you specify clobbered registers at the end of your asm:
__asm__(".." ::: "%eax").
If gcc is not set to support sse (-msse) it will not accept xmm registers
......
......@@ -62,7 +62,7 @@ bash ./configure
@section Darwin (OS X, iPhone)
The toolchain provided with Xcode is sufficient to build the basic
unacelerated code.
unaccelerated code.
OS X on PowerPC or ARM (iPhone) requires a preprocessor from
@url{git://git.libav.org/gas-preprocessor.git} to build the optimized
......@@ -137,7 +137,7 @@ pacman -S make pkgconf diffutils
pacman -S mingw-w64-x86_64-yasm mingw-w64-x86_64-gcc mingw-w64-x86_64-SDL
@end example
To target 32bit replace the @code{x86_64} with @code{i686} in the command above.
To target 32 bits replace @code{x86_64} with @code{i686} in the command above.
@section Microsoft Visual C++ or Intel C++ Compiler for Windows
......
......@@ -10,12 +10,12 @@ Current (simplified) Architecture:
/ \
special converter [Input to YUV converter]
| |
| (8bit YUV 4:4:4 / 4:2:2 / 4:2:0 / 4:0:0 )
| (8-bit YUV 4:4:4 / 4:2:2 / 4:2:0 / 4:0:0 )
| |
| v
| Horizontal scaler
| |
| (15bit YUV 4:4:4 / 4:2:2 / 4:2:0 / 4:1:1 / 4:0:0 )
| (15-bit YUV 4:4:4 / 4:2:2 / 4:2:0 / 4:1:1 / 4:0:0 )
| |
| v
| Vertical scaler and output converter
......
This is a quick description of the viterbi aka dynamic programing
algorthm.
This is a quick description of the Viterbi aka dynamic programming
algorithm.
Its reason for existence is that wikipedia has become very poor on
Its reason for existence is that Wikipedia has become very poor on
describing algorithms in a way that makes it useable for understanding
them or anything else actually. It tends now to describe the very same
algorithm under 50 different names and pages with few understandable
......@@ -41,7 +41,7 @@ readable)
Our goal is to find a path from left to right through it which
minimizes the sum of the score of all edges.
(and of course left/right is just a convention here it could be top down too)
Similarly the minimum could be the maximum by just fliping the sign,
Similarly the minimum could be the maximum by just flipping the sign,
Example of a path with scores:
O O O O O O O
......@@ -53,7 +53,7 @@ Example of a path with scores:
O O O O O O-1-O---> (sum here is 24)
The viterbi algorthm now solves this simply column by column
The Viterbi algorithm now solves this simply column by column
For the previous column each point has a best path and a associated
score:
......@@ -100,10 +100,10 @@ trivial given we know the previous column best paths and scores:
O 0 4
the viterbi algorthm continues exactly like this column for column until the
the Viterbi algorithm continues exactly like this column for column until the
end and then just picks the path with the best score (above that would be the
one with score 3)
Author: Michael niedermayer
Author: Michael Niedermayer
Copyright LGPL
......@@ -1916,8 +1916,8 @@ static int decode_ics(AACContext *ac, SingleChannelElement *sce,
avpriv_request_sample(ac->avctx, "SSR");
return AVERROR_PATCHWELCOME;
}
// I see no textual basis in the spec for this occuring after SSR gain
// control, but this is what both reference and real implmentations do
// I see no textual basis in the spec for this occurring after SSR gain
// control, but this is what both reference and real implementations do
if (tns->present && er_syntax)
if (decode_tns(ac, tns, gb, ics) < 0)
return AVERROR_INVALIDDATA;
......@@ -3047,7 +3047,7 @@ static av_cold int aac_decode_close(AVCodecContext *avctx)
struct LATMContext {
AACContext aac_ctx; ///< containing AACContext
int initialized; ///< initilized after a valid extradata was seen
int initialized; ///< initialized after a valid extradata was seen
// parser data
int audio_mux_version_A; ///< LATM syntax version
......
......@@ -130,7 +130,7 @@ av_cold void ff_aac_sbr_init(void)
/** Places SBR in pure upsampling mode. */
static void sbr_turnoff(SpectralBandReplication *sbr) {
sbr->start = 0;
// Init defults used in pure upsampling mode
// Init defaults used in pure upsampling mode
sbr->kx[1] = 32; //Typo in spec, kx' inits to 32
sbr->m[1] = 0;
// Reset values for first SBR header
......
......@@ -142,7 +142,7 @@ function ff_mpadsp_apply_window_\type\()_neon, export=1
sub x10, x10, #4<<2
b.gt 1b
// comuting samples[16]
// computing samples[16]
add x6, x1, #32<<2
ld1 {v0.2s}, [x6], x9
ld1 {v1.2s}, [x0], x9
......
/*
* Autodesk RLE Decoder
* Copyright (C) 2005 the ffmpeg project
* Copyright (C) 2005 The FFmpeg project
*
* This file is part of Libav.
*
......
/*
* Copyright (c) 2001-2003 The ffmpeg Project
* Copyright (c) 2001-2003 The FFmpeg project
*
* first version by Francois Revol (revol@free.fr)
* fringe ADPCM codecs (e.g., DK3, DK4, Westwood)
......
/*
* Copyright (c) 2001-2003 The ffmpeg Project
* Copyright (c) 2001-2003 The FFmpeg project
*
* This file is part of Libav.
*
......
/*
* Copyright (c) 2001-2003 The ffmpeg Project
* Copyright (c) 2001-2003 The FFmpeg project
*
* This file is part of Libav.
*
......
/*
* Copyright (c) 2001-2003 The ffmpeg Project
* Copyright (c) 2001-2003 The FFmpeg project
*
* This file is part of Libav.
*
......
/*
* Copyright (c) 2001-2003 The ffmpeg Project
* Copyright (c) 2001-2003 The FFmpeg project
*
* first version by Francois Revol (revol@free.fr)
* fringe ADPCM codecs (e.g., DK3, DK4, Westwood)
......
......@@ -29,20 +29,20 @@
* passed through the extradata[_size] fields. This atom is tacked onto
* the end of an 'alac' stsd atom and has the following format:
*
* 32bit atom size
* 32bit tag ("alac")
* 32bit tag version (0)
* 32bit samples per frame (used when not set explicitly in the frames)
* 8bit compatible version (0)
* 8bit sample size
* 8bit history mult (40)
* 8bit initial history (14)
* 8bit rice param limit (10)
* 8bit channels
* 16bit maxRun (255)
* 32bit max coded frame size (0 means unknown)
* 32bit average bitrate (0 means unknown)
* 32bit samplerate
* 32 bits atom size
* 32 bits tag ("alac")
* 32 bits tag version (0)
* 32 bits samples per frame (used when not set explicitly in the frames)
* 8 bits compatible version (0)
* 8 bits sample size
* 8 bits history mult (40)
* 8 bits initial history (14)
* 8 bits rice param limit (10)
* 8 bits channels
* 16 bits maxRun (255)
* 32 bits max coded frame size (0 means unknown)
* 32 bits average bitrate (0 means unknown)
* 32 bits samplerate
*/
#include <inttypes.h>
......
......@@ -305,7 +305,7 @@ static av_cold int read_specific_config(ALSDecContext *ctx)
skip_bits_long(&gb, 32); // sample rate already known
sconf->samples = get_bits_long(&gb, 32);
avctx->channels = m4ac.channels;
skip_bits(&gb, 16); // number of channels already knwon
skip_bits(&gb, 16); // number of channels already known
skip_bits(&gb, 3); // skip file_type
sconf->resolution = get_bits(&gb, 3);
sconf->floating = get_bits1(&gb);
......
......@@ -532,13 +532,13 @@ static void decode_fixed_sparse(AMRFixed *fixed_sparse, const uint16_t *pulses,
* @param p the context
* @param subframe unpacked amr subframe
* @param mode mode of the current fram