Merge commit '41ed7ab45fc693f7d7fc35664c0233f4c32d69bb'

* commit '41ed7ab45fc693f7d7fc35664c0233f4c32d69bb':
  cosmetics: Fix spelling mistakes

Merged-by: Clément Bœsch <[email protected]>

Changelog

@@ -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

doc/APIchanges

@@ -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.

doc/examples/decoding_encoding.c

@@ -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);

doc/examples/muxing.c

@@ -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) {

doc/ffmpeg.texi

@@ -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

doc/filters.texi

@@ -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

doc/git-howto.texi

@@ -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).
 

doc/nut.texi

@@ -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

doc/optimization.txt

@@ -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

doc/platform.texi

@@ -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
 

doc/swscale.txt

@@ -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

ffmpeg.c

@@ -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;

ffplay.c

@@ -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;
 

libavcodec/aarch64/mpegaudiodsp_neon.S

@@ -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

libavcodec/aasc.c

@@ -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.
  *

libavcodec/adpcm.c

@@ -1,5 +1,5 @@
 /*
- * Copyright (c) 2001-2003 The FFmpeg Project
+ * Copyright (c) 2001-2003 The FFmpeg project
  *
  * first version by Francois Revol ([email protected])
  * fringe ADPCM codecs (e.g., DK3, DK4, Westwood)

libavcodec/adpcm.h

@@ -1,5 +1,5 @@
 /*
- * Copyright (c) 2001-2003 The FFmpeg Project
+ * Copyright (c) 2001-2003 The FFmpeg project
  *
  * This file is part of FFmpeg.
  *

libavcodec/adpcm_data.c

@@ -1,5 +1,5 @@
 /*
- * Copyright (c) 2001-2003 The FFmpeg Project
+ * Copyright (c) 2001-2003 The FFmpeg project
  *
  * This file is part of FFmpeg.
  *

libavcodec/adpcm_data.h

@@ -1,5 +1,5 @@
 /*
- * Copyright (c) 2001-2003 The FFmpeg Project
+ * Copyright (c) 2001-2003 The FFmpeg project
  *
  * This file is part of FFmpeg.
  *

libavcodec/adpcmenc.c

@@ -1,5 +1,5 @@
 /*
- * Copyright (c) 2001-2003 The FFmpeg Project
+ * Copyright (c) 2001-2003 The FFmpeg project
  *
  * first version by Francois Revol ([email protected])
  * fringe ADPCM codecs (e.g., DK3, DK4, Westwood)

libavcodec/alac.c

@@ -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>

libavcodec/amrnbdec.c

@@ -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);

libavcodec/amrwbdata.h

@@ -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
 

libavcodec/ansi.c

@@ -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;

libavcodec/apedec.c

@@ -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)
 {

libavcodec/arm/mlpdsp_armv6.S

@@ -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

libavcodec/arm/mpegvideo_armv5te.c

@@ -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
  */