@@ -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

@@ -1,6 +1,6 @@

 /*

  * 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

@@ -1,7 +1,7 @@

-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

@@ -1,6 +1,6 @@

 /*

  * Autodesk RLE Decoder

- * Copyright (C) 2005 the ffmpeg project

+ * Copyright (C) 2005 The FFmpeg project

  *

  * This file is part of Libav.

  *

@@ -1,5 +1,5 @@

 /*

- * 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)

@@ -1,5 +1,5 @@

 /*

- * Copyright (c) 2001-2003 The ffmpeg Project

+ * Copyright (c) 2001-2003 The FFmpeg project

  *

  * This file is part of Libav.

  *

@@ -1,5 +1,5 @@

 /*

- * Copyright (c) 2001-2003 The ffmpeg Project

+ * Copyright (c) 2001-2003 The FFmpeg project

  *

  * This file is part of Libav.

  *

@@ -1,5 +1,5 @@

 /*

- * Copyright (c) 2001-2003 The ffmpeg Project

+ * Copyright (c) 2001-2003 The FFmpeg project

  *

  * This file is part of Libav.

  *

@@ -1,5 +1,5 @@

 /*

- * 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 frame

- * @param fixed_sparse sparse respresentation of the fixed vector

+ * @param fixed_sparse sparse representation of the fixed vector

  */

 static void pitch_sharpening(AMRContext *p, int subframe, enum Mode mode,

                              AMRFixed *fixed_sparse)

 {

     // The spec suggests the current pitch gain is always used, but in other

-    // modes the pitch and codebook gains are joinly quantized (sec 5.8.2)

+    // modes the pitch and codebook gains are jointly quantized (sec 5.8.2)

     // so the codebook gain cannot depend on the quantized pitch gain.

     if (mode == MODE_12k2)

         p->beta = FFMIN(p->pitch_gain[4], 1.0);

@@ -38,7 +38,7 @@

 #define MIN_ISF_SPACING     (128.0 / 32768.0) ///< minimum isf gap

 #define PRED_FACTOR         (1.0 / 3.0)

-#define MIN_ENERGY         -14.0              ///< initial innnovation energy (dB)

+#define MIN_ENERGY         -14.0              ///< initial innovation energy (dB)

 #define ENERGY_MEAN         30.0              ///< mean innovation energy (dB) in all modes

 #define PREEMPH_FAC         0.68              ///< factor used to de-emphasize synthesis

@@ -189,7 +189,7 @@ static int execute_code(AVCodecContext * avctx, int c)

         s->y = s->nb_args > 0 ? av_clip((s->args[0] - 1)*s->font_height, 0, avctx->height - s->font_height) : 0;

         s->x = s->nb_args > 1 ? av_clip((s->args[1] - 1)*FONT_WIDTH,     0, avctx->width  - FONT_WIDTH) : 0;

         break;

-    case 'h': //set creen mode

+    case 'h': //set screen mode

     case 'l': //reset screen mode

         if (s->nb_args < 2)

             s->args[0] = DEFAULT_SCREEN_MODE;

@@ -359,10 +359,10 @@ static inline void range_dec_normalize(APEContext *ctx)

 }

 /**

- * Calculate culmulative frequency for next symbol. Does NO update!

+ * Calculate cumulative frequency for next symbol. Does NO update!

  * @param ctx decoder context

  * @param tot_f is the total frequency or (code_value)1<<shift

- * @return the culmulative frequency

+ * @return the cumulative frequency

  */

 static inline int range_decode_culfreq(APEContext *ctx, int tot_f)

 {

@@ -240,7 +240,7 @@ DAT3    .req    v4

 DAT4    .req    v5

 DAT5    .req    v6

 DAT6    .req    sl // use these rather than the otherwise unused

-DAT7    .req    fp // ip and lr so that we can load them usinf LDRD

+DAT7    .req    fp // ip and lr so that we can load them using LDRD

  .macro output4words  tail, head, r0, r1, r2, r3, r4, r5, r6, r7, pointer_dead=0

   .if \head

@@ -28,7 +28,7 @@ void ff_dct_unquantize_h263_armv5te(int16_t *block, int qmul, int qadd, int coun

 #ifdef ENABLE_ARM_TESTS

 /**

- * h263 dequantizer supplementary function, it is performance critical and needs to

+ * H.263 dequantizer supplementary function, it is performance critical and needs to

  * have optimized implementations for each architecture. Is also used as a reference

  * implementation in regression tests

  */

@@ -64,7 +64,7 @@ function ff_simple_idct_arm, export=1

 __row_loop:

-        @@ read the row and check if it is null, almost null, or not, according to strongarm specs, it is not necessary to optimize ldr accesses (i.e. split 32bits in 2 16bits words), at least it gives more usable registers :)

+        @@ read the row and check if it is null, almost null, or not, according to strongarm specs, it is not necessary to optimize ldr accesses (i.e. split 32 bits in two 16-bit words), at least it gives more usable registers :)

         ldr r1, [r14, #0]        @ R1=(int32)(R12)[0]=ROWr32[0] (relative row cast to a 32b pointer)

         ldr r2, [r14, #4]        @ R2=(int32)(R12)[1]=ROWr32[1]

         ldr r3, [r14, #8]        @ R3=ROWr32[2]

@@ -234,8 +234,8 @@ __end_a_evaluation:

         @@ row[7] = (a0 - b0) >> ROW_SHIFT;

         add r8, r6, r0           @ R8=a0+b0

         add r9, r2, r1           @ R9=a1+b1

-        @@ put 2 16 bits half-words in a 32bits word

-        @@ ROWr32[0]=ROWr16[0] | (ROWr16[1]<<16) (only Little Endian compliant then!!!)

+        @@ put two 16-bit half-words in a 32-bit word

+        @@ ROWr32[0]=ROWr16[0] | (ROWr16[1]<<16) (only little-endian compliant then!!!)

         ldr r10, =MASK_MSHW      @ R10=0xFFFF0000

         and r9, r10, r9, lsl #ROW_SHIFT2MSHW @ R9=0xFFFF0000 & ((a1+b1)<<5)

         mvn r11, r10             @ R11= NOT R10= 0x0000FFFF

@@ -322,7 +322,7 @@ endfunc

         vmov.i16        q12, #3

         vsubl.s8        q10, d8,  d6            @ QS0 - PS0

-        vsubl.s8        q11, d9,  d7            @   (widened to 16bit)

+        vsubl.s8        q11, d9,  d7            @   (widened to 16 bits)

         veor            q2,  q2,  q13           @ PS1 = P1 ^ 0x80

         veor            q5,  q5,  q13           @ QS1 = Q1 ^ 0x80

         vmul.i16        q10, q10, q12           @ w = 3 * (QS0 - PS0)

@@ -1,5 +1,5 @@

 /*

- * SSA/ASS common funtions