@@ -914,8 +914,8 @@ version 0.8:

 - showinfo filter added

 - SMPTE 302M AES3 audio decoder

 - Apple Core Audio Format muxer

-- 9bit and 10bit per sample support in the H.264 decoder

-- 9bit and 10bit FFV1 encoding / decoding

+- 9 bits and 10 bits per sample support in the H.264 decoder

+- 9 bits and 10 bits FFV1 encoding / decoding

 - split filter added

 - select filter added

 - sdl output device added

@@ -1208,7 +1208,7 @@ 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

@@ -1175,7 +1175,7 @@ lavd 54.4.100 / 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.

@@ -25,9 +25,9 @@

  * libavcodec API use example.

  *

  * @example decoding_encoding.c

- * Note that libavcodec only handles codecs (mpeg, mpeg4, etc...),

- * not file formats (avi, vob, mp4, mov, mkv, mxf, flv, mpegts, mpegps, etc...). See library 'libavformat' for the

- * format handling

+ * Note that libavcodec only handles codecs (MPEG, MPEG-4, etc...),

+ * not file formats (AVI, VOB, MP4, MOV, MKV, MXF, FLV, MPEG-TS, MPEG-PS, etc...).

+ * See library 'libavformat' for the format handling

  */

 #include <math.h>

@@ -253,7 +253,7 @@ static void audio_decode_example(const char *outfilename, const char *filename)

     printf("Decode audio file %s to %s\n", filename, outfilename);

-    /* 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");

@@ -356,7 +356,7 @@ static void video_encode_example(const char *filename, int codec_id)

     printf("Encode video file %s\n", filename);

-    /* find the mpeg1 video encoder */

+    /* find the video encoder */

     codec = avcodec_find_encoder(codec_id);

     if (!codec) {

         fprintf(stderr, "Codec not found\n");

@@ -475,7 +475,7 @@ static void video_encode_example(const char *filename, int codec_id)

         }

     }

-    /* 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);

@@ -543,12 +543,12 @@ 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("Decode video file %s to %s\n", filename, outfilename);

-    /* 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");

@@ -613,9 +613,9 @@ static void video_decode_example(const char *outfilename, const char *filename)

                 exit(1);

     }

-    /* 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;

     decode_write_frame(outfilename, c, frame, &frame_count, &avpkt, 1);

@@ -161,7 +161,7 @@ static void add_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) {

@@ -720,7 +720,7 @@ 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

@@ -5667,7 +5667,7 @@ It accepts the following parameters:

 @item limit

 Set higher black value threshold, which can be optionally specified

-from nothing (0) to everything (255 for 8bit based formats). An intensity

+from nothing (0) to everything (255 for 8-bit based formats). An intensity

 value greater to the set value is considered non-black. It defaults to 24.

 You can also specify a value between 0.0 and 1.0 which will be scaled depending

 on the bitdepth of the pixel format.

@@ -8305,7 +8305,7 @@ geq=lum=255*gauss((X/W-0.5)*3)*gauss((Y/H-0.5)*3)/gauss(0)/gauss(0),format=gray

 @section gradfun

 Fix the banding artifacts that are sometimes introduced into nearly flat

-regions by truncation to 8bit color depth.

+regions by truncation to 8-bit color depth.

 Interpolate the gradients that should go where the bands are, and

 dither them.

@@ -15588,7 +15588,7 @@ select=between(t\,10\,20)

 @end example

 @item

-Select only I frames contained in the 10-20 time interval:

+Select only I-frames contained in the 10-20 time interval:

 @example

 select=between(t\,10\,20)*eq(pict_type\,I)

 @end example

@@ -408,7 +408,7 @@ with @option{--dry-run} first. And then inspecting the commits listed with

 @command{git log -p 1234567..987654}. The @command{git status} command

 may help in finding local changes that have been forgotten to be added.

-Next let the code pass through a full run of our testsuite.

+Next let the code pass through a full run of our test suite.

 @itemize

 @item @command{make distclean}

@@ -418,7 +418,7 @@ Next let the code pass through a full run of our testsuite.

 @end itemize

 Make sure all your changes have been checked before pushing them, the

-testsuite only checks against regressions and that only to some extend. It does

+test suite only checks against regressions and that only to some extend. It does

 obviously not check newly added features/code to be working unless you have

 added a test for that (which is recommended).

@@ -34,7 +34,7 @@ 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

@@ -63,7 +63,7 @@ bash ./configure

 @section Darwin (Mac OS X, iPhone)

 The toolchain provided with Xcode is sufficient to build the basic

-unacelerated code.

+unaccelerated code.

 Mac OS X on PowerPC or ARM (iPhone) requires a preprocessor from

 @url{https://github.com/FFmpeg/gas-preprocessor} or

@@ -144,7 +144,7 @@ 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

@@ -1922,7 +1922,7 @@ 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;

@@ -2745,7 +2745,7 @@ static int stream_component_open(VideoState *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 anough audio FIFO fullness,

            we correct audio sync only if larger than this threshold */

         is->audio_diff_threshold = (double)(is->audio_hw_buf_size) / is->audio_tgt.bytes_per_sec;

@@ -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 FFmpeg.

  *

@@ -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 FFmpeg.

  *

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

 /*

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

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

  *

  * This file is part of FFmpeg.

  *

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

 /*

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

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

  *

  * This file is part of FFmpeg.

  *

@@ -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      (10)

- *  8bit  rice param limit     (14)

- *  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      (10)

+ *  8 bits  rice param limit     (14)

+ *  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>

@@ -547,13 +547,13 @@ 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

@@ -206,7 +206,7 @@ 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;

@@ -339,10 +339,10 @@ 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

@@ -29,7 +29,7 @@ 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)

@@ -880,7 +880,7 @@ typedef struct RcOverride{

  * Use only bitexact stuff (except (I)DCT).

  */

 #define AV_CODEC_FLAG_BITEXACT        (1 << 23)

-/* Fx : Flag for h263+ extra options */

+/* Fx : Flag for H.263+ extra options */

 /**

  * H.263 advanced intra coding / MPEG-4 AC prediction

  */

@@ -997,7 +997,7 @@ typedef struct RcOverride{

  * are connected to a parser to split what they return into proper frames.

  * This flag is reserved to the very rare category of codecs which have a

  * bitstream that cannot be split into frames without timeconsuming

- * operations like full decoding. Demuxers carring such bitstreams thus

+ * operations like full decoding. Demuxers carrying such bitstreams thus

  * may return multiple frames in a packet. This has many disadvantages like

  * prohibiting stream copy in many cases thus it should only be considered

  * as a last resort.

@@ -1177,7 +1177,7 @@ typedef struct RcOverride{

  * are connected to a parser to split what they return into proper frames.

  * This flag is reserved to the very rare category of codecs which have a

  * bitstream that cannot be split into frames without timeconsuming

- * operations like full decoding. Demuxers carring such bitstreams thus

+ * operations like full decoding. Demuxers carrying such bitstreams thus

  * may return multiple frames in a packet. This has many disadvantages like

  * prohibiting stream copy in many cases thus it should only be considered

  * as a last resort.

@@ -1257,7 +1257,7 @@ typedef struct RcOverride{

 #define MB_TYPE_L0L1       (MB_TYPE_L0   | MB_TYPE_L1)

 #define MB_TYPE_QUANT      0x00010000

 #define MB_TYPE_CBP        0x00020000

-//Note bits 24-31 are reserved for codec specific use (h264 ref0, mpeg1 0mv, ...)

+// Note bits 24-31 are reserved for codec specific use (H.264 ref0, MPEG-1 0mv, ...)

 #endif

 /**

@@ -1751,9 +1751,9 @@ typedef struct AVCodecContext {

     /**

      * some codecs need / can use extradata like Huffman tables.

-     * mjpeg: Huffman tables

+     * MJPEG: Huffman tables

      * rv10: additional flags

-     * mpeg4: global headers (they can be in the bitstream or here)

+     * MPEG-4: global headers (they can be in the bitstream or here)

      * The allocated memory should be AV_INPUT_BUFFER_PADDING_SIZE bytes larger

      * than extradata_size to avoid problems if it is read with the bitstream reader.

      * The bytewise contents of extradata must not depend on the architecture or CPU endianness.

@@ -1823,7 +1823,7 @@ typedef struct AVCodecContext {

      * picture width / height.

      *

      * @note Those fields may not match the values of the last

-     * AVFrame outputted by avcodec_decode_video2 due frame

+     * AVFrame output by avcodec_decode_video2 due frame

      * reordering.

      *

      * - encoding: MUST be set by user.

@@ -1839,7 +1839,7 @@ typedef struct AVCodecContext {

      * the decoded frame is cropped before being output or lowres is enabled.

      *

      * @note Those field may not match the value of the last

-     * AVFrame outputted by avcodec_receive_frame() due frame

+     * AVFrame output by avcodec_receive_frame() due frame

      * reordering.

      *

      * - encoding: unused

@@ -1866,7 +1866,7 @@ typedef struct AVCodecContext {

      * May be overridden by the decoder if it knows better.

      *

      * @note This field may not match the value of the last

-     * AVFrame outputted by avcodec_receive_frame() due frame

+     * AVFrame output by avcodec_receive_frame() due frame

      * reordering.

      *

      * - encoding: Set by user.

@@ -1976,8 +1976,8 @@ typedef struct AVCodecContext {

 #endif

     /**

-     * qscale factor between P and I-frames

-     * If > 0 then the last p frame quantizer will be used (q= lastp_q*factor+offset).

+     * qscale factor between P- and I-frames

+     * If > 0 then the last P-frame quantizer will be used (q = lastp_q * factor + offset).

      * If < 0 then normal ratecontrol will be done (q= -normal_q*factor+offset).

      * - encoding: Set by user.

      * - decoding: unused

@@ -2189,7 +2189,7 @@ typedef struct AVCodecContext {

      */

     int slice_flags;

 #define SLICE_FLAG_CODED_ORDER    0x0001 ///< draw_horiz_band() is called in coded order instead of display

-#define SLICE_FLAG_ALLOW_FIELD    0x0002 ///< allow draw_horiz_band() with field slices (MPEG2 field pics)

+#define SLICE_FLAG_ALLOW_FIELD    0x0002 ///< allow draw_horiz_band() with field slices (MPEG-2 field pics)

 #define SLICE_FLAG_ALLOW_PLANE    0x0004 ///< allow draw_horiz_band() with 1 component at a time (SVQ1)

 #if FF_API_XVMC

@@ -2280,14 +2280,14 @@ typedef struct AVCodecContext {

 #endif

     /**

-     * minimum MB lagrange multipler

+     * minimum MB Lagrange multiplier

      * - encoding: Set by user.

      * - decoding: unused

      */

     int mb_lmin;

     /**

-     * maximum MB lagrange multipler

+     * maximum MB Lagrange multiplier

      * - encoding: Set by user.

      * - decoding: unused

      */

@@ -2507,7 +2507,7 @@ typedef struct AVCodecContext {

      *     to all data planes. data[] must hold as many pointers as it can.

      *     extended_data must be allocated with av_malloc() and will be freed in

      *     av_frame_unref().

-     *   * otherwise exended_data must point to data

+     *   * otherwise extended_data must point to data

      * - buf[] must contain one or more pointers to AVBufferRef structures. Each of

      *   the frame's data and extended_data pointers must be contained in these. That

      *   is, one AVBufferRef for each allocated chunk of memory, not necessarily one

@@ -2851,7 +2851,7 @@ typedef struct AVCodecContext {

 #define FF_BUG_TRUNCATED       16384

     /**

-     * strictly follow the standard (MPEG4, ...).

+     * strictly follow the standard (MPEG-4, ...).

      * - encoding: Set by user.

      * - decoding: Set by user.

      * Setting this to STRICT or higher means the encoder and decoder will

@@ -2922,9 +2922,9 @@ typedef struct AVCodecContext {

      * - decoding: Set by user.

      */

     int debug_mv;

-#define FF_DEBUG_VIS_MV_P_FOR  0x00000001 //visualize forward predicted MVs of P frames

-#define FF_DEBUG_VIS_MV_B_FOR  0x00000002 //visualize forward predicted MVs of B frames

-#define FF_DEBUG_VIS_MV_B_BACK 0x00000004 //visualize backward predicted MVs of B frames

+#define FF_DEBUG_VIS_MV_P_FOR  0x00000001 // visualize forward predicted MVs of P-frames

+#define FF_DEBUG_VIS_MV_B_FOR  0x00000002 // visualize forward predicted MVs of B-frames

+#define FF_DEBUG_VIS_MV_B_BACK 0x00000004 // visualize backward predicted MVs of B-frames

 #endif

     /**

@@ -2952,7 +2952,7 @@ typedef struct AVCodecContext {

     /**

-     * opaque 64bit number (generally a PTS) that will be reordered and

+     * opaque 64-bit number (generally a PTS) that will be reordered and

      * output in AVFrame.reordered_opaque

      * - encoding: unused

      * - decoding: Set by user.

@@ -5950,7 +5950,7 @@ unsigned int av_xiphlacing(unsigned char *s, unsigned int v);

  * a pointer to an AVClass struct

  * @param[in] feature string containing the name of the missing feature

  * @param[in] want_sample indicates if samples are wanted which exhibit this feature.

- * If want_sample is non-zero, additional verbage will be added to the log

+ * If want_sample is non-zero, additional verbiage will be added to the log

  * message which tells the user how to report samples to the development

  * mailing list.

  * @deprecated Use avpriv_report_missing_feature() instead.

@@ -1240,7 +1240,7 @@ static int decode_frame(AVCodecContext *avctx, void *data, int *got_frame, AVPac

 }

 /**

- * Caclulate quantization tables for version b

+ * Calculate quantization tables for version b

  */

 static av_cold void binkb_calc_quant(void)

 {

@@ -150,7 +150,7 @@ static int compare_vlcspec(const void *a, const void *b)

 /**

  * Build VLC decoding tables suitable for use with get_vlc().

  *

- * @param vlc            the context to be initted

+ * @param vlc            the context to be initialized

  *

  * @param table_nb_bits  max length of vlc codes to store directly in this table

  *                       (Longer codes are delegated to subtables.)

@@ -248,7 +248,7 @@ static int build_table(VLC *vlc, int table_nb_bits, int nb_codes,

 /* Build VLC decoding tables suitable for use with get_vlc().

-   'nb_bits' set the decoding table size (2^nb_bits) entries. The

+   'nb_bits' sets the decoding table size (2^nb_bits) entries. The

    bigger it is, the faster is the decoding. But it should not be too

    big to save memory and L1 cache. '9' is a good compromise.

@@ -22,7 +22,7 @@

 #include "avcodec.h"

 /**

- * Called by the biststream filters to get the next packet for filtering.

+ * Called by the bitstream filters to get the next packet for filtering.

  * The filter is responsible for either freeing the packet or passing it to the

  * caller.

  */

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

 /*

  * Cinepak Video Decoder

- * Copyright (c) 2003 The FFmpeg Project

+ * Copyright (C) 2003 The FFmpeg project

  *

  * This file is part of FFmpeg.

  *

@@ -141,7 +141,7 @@ typedef struct cook {

     VLC                 envelope_quant_index[13];

     VLC                 sqvh[7];          // scalar quantization

-    /* generatable tables and related variables */

+    /* generate tables and related variables */

     int                 gain_size_factor;

     float               gain_table[23];

@@ -1187,7 +1187,7 @@ static av_cold int cook_decode_init(AVCodecContext *avctx)

         /* Initialize variable relations */

         q->subpacket[s].numvector_size = (1 << q->subpacket[s].log2_numvector_size);

-        /* Try to catch some obviously faulty streams, othervise it might be exploitable */

+        /* Try to catch some obviously faulty streams, otherwise it might be exploitable */

         if (q->subpacket[s].total_subbands > 53) {

             avpriv_request_sample(avctx, "total_subbands > 53");

             return AVERROR_PATCHWELCOME;

@@ -1260,7 +1260,7 @@ static av_cold int cook_decode_init(AVCodecContext *avctx)

         q->saturate_output = saturate_output_float;

     }

-    /* Try to catch some obviously faulty streams, othervise it might be exploitable */

+    /* Try to catch some obviously faulty streams, otherwise it might be exploitable */

     if (q->samples_per_channel != 256 && q->samples_per_channel != 512 &&

         q->samples_per_channel != 1024) {

         avpriv_request_sample(avctx, "samples_per_channel = %d",

@@ -22,7 +22,7 @@

 /**

  * @file

- * Cook AKA RealAudio G2 compatible decoderdata

+ * Cook AKA RealAudio G2 compatible decoder data

  */

 #ifndef AVCODEC_COOKDATA_H

@@ -4,7 +4,7 @@

  * based on "Creative YUV (CYUV) stream format for AVI":

  *   http://www.csse.monash.edu.au/~timf/videocodec/cyuv.txt

  *

- * Copyright (c) 2003 The FFmpeg Project

+ * Copyright (C) 2003 The FFmpeg project

  *

  * This file is part of FFmpeg.

  *

@@ -52,7 +52,7 @@ const uint8_t ff_dca_dmix_primary_nch[8] = {

 /* ADPCM data */

-/* 16bits signed fractional Q13 binary codes */

+/* 16 bits signed fractional Q13 binary codes */

 const int16_t ff_dca_adpcm_vb[4096][4] = {

     {   9928,  -2618,  -1093, -1263 },

     {  11077,  -2876,  -1747,  -308 },

@@ -4215,7 +4215,7 @@ const uint32_t ff_dca_quant_levels[32] = {

     2097152, 4194304, 8388608,     0,      0,      0,      0,       0

 };

-/* 20bits unsigned fractional binary codes */

+/* 20 bits unsigned fractional binary codes */

 const uint32_t ff_dca_lossy_quant[32] = {

          0, 6710886, 4194304, 3355443, 2474639, 2097152, 1761608, 1426063,

     796918,  461373,  251658,  146801,   79692,   46137,   27263,   16777,

@@ -4223,7 +4223,7 @@ const uint32_t ff_dca_lossy_quant[32] = {

         84,      42,      21,       0,       0,       0,       0,       0

 };

-/* 20bits unsigned fractional binary codes */

+/* 20 bits unsigned fractional binary codes */

 const uint32_t ff_dca_lossless_quant[32] = {

          0, 4194304, 2097152, 1384120, 1048576, 696254, 524288, 348127,

     262144,  131072,   65431,   33026,   16450,   8208,   4100,   2049,

@@ -442,7 +442,7 @@ static void help(void)

     printf("dct-test [-i] [<test-number>] [<bits>]\n"

            "test-number 0 -> test with random matrixes\n"

            "            1 -> test with random sparse matrixes\n"

-           "            2 -> do 3. test from mpeg4 std\n"

+           "            2 -> do 3. test from MPEG-4 std\n"

            "bits        Number of time domain bits to use, 8 is default\n"

            "-i          test IDCT implementations\n"

            "-4          test IDCT248 implementations\n"

@@ -36,8 +36,7 @@

 #include "pixblockdsp.h"

 #include "dnxhdenc.h"

-

-// The largest value that will not lead to overflow for 10bit samples.

+// The largest value that will not lead to overflow for 10-bit samples.

 #define DNX10BIT_QMAT_SHIFT 18