Merge remote-tracking branch 'qatar/master'

* qatar/master:
  miscellaneous typo fixes

Conflicts:
	configure
	libavformat/avisynth.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>

Changelog

@@ -713,7 +713,7 @@ version 0.5:
 - MXF demuxer
 - VC-1/WMV3/WMV9 video decoder
 - MacIntel support
-- AVISynth support
+- AviSynth support
 - VMware video decoder
 - VP5 video decoder
 - VP6 video decoder

configure

@@ -186,7 +186,7 @@ Individual component options:
   --disable-filters        disable all filters
 
 External library support:
-  --enable-avisynth        enable reading of AVISynth script files [no]
+  --enable-avisynth        enable reading of AviSynth script files [no]
   --disable-bzlib          disable bzlib [autodetect]
   --enable-fontconfig      enable fontconfig
   --enable-frei0r          enable frei0r video filtering

doc/faq.texi

@@ -234,8 +234,8 @@ Just create an "input.avs" text file with this single line ...
   ffmpeg -i input.avs
 @end example
 
-For ANY other help on Avisynth, please visit the
-@uref{http://www.avisynth.org/, Avisynth homepage}.
+For ANY other help on AviSynth, please visit the
+@uref{http://www.avisynth.org/, AviSynth homepage}.
 
 @section How can I join video files?
 

ffplay.c

@@ -975,7 +975,8 @@ static void video_audio_display(VideoState *s)
                 }
                 av_rdft_calc(s->rdft, data[ch]);
             }
-            // least efficient way to do this, we should of course directly access it but its more than fast enough
+            /* Least efficient way to do this, we should of course
+             * directly access it but it is more than fast enough. */
             for (y = 0; y < s->height; y++) {
                 double w = 1 / sqrt(nb_freq);
                 int a = sqrt(w * sqrt(data[0][2 * y + 0] * data[0][2 * y + 0] + data[0][2 * y + 1] * data[0][2 * y + 1]));

libavcodec/aac.h

@@ -157,7 +157,7 @@ typedef struct LongTermPrediction {
 typedef struct IndividualChannelStream {
     uint8_t max_sfb;            ///< number of scalefactor bands per group
     enum WindowSequence window_sequence[2];
-    uint8_t use_kb_window[2];   ///< If set, use Kaiser-Bessel window, otherwise use a sinus window.
+    uint8_t use_kb_window[2];   ///< If set, use Kaiser-Bessel window, otherwise use a sine window.
     int num_window_groups;
     uint8_t group_len[8];
     LongTermPrediction ltp;

libavcodec/twinvq.c

@@ -230,7 +230,7 @@ static void memset_float(float *buf, float val, int size)
  * Evaluate a single LPC amplitude spectrum envelope coefficient from the line
  * spectrum pairs.
  *
- * @param lsp a vector of the cosinus of the LSP values
+ * @param lsp a vector of the cosine of the LSP values
  * @param cos_val cos(PI*i/N) where i is the index of the LPC amplitude
  * @param order the order of the LSP (and the size of the *lsp buffer). Must
  *        be a multiple of four.
@@ -302,9 +302,9 @@ static inline float get_cos(int idx, int part, const float *cos_tab, int size)
  * unexplained condition.
  *
  * @param step the size of a block "siiiibiiii"
- * @param in the cosinus of the LSP data
- * @param part is 0 for 0...PI (positive cossinus values) and 1 for PI...2PI
- *        (negative cossinus values)
+ * @param in the cosine of the LSP data
+ * @param part is 0 for 0...PI (positive cosine values) and 1 for PI...2PI
+ *        (negative cosine values)
  * @param size the size of the whole output
  */
 static inline void eval_lpcenv_or_interp(TwinContext *tctx,

libavcodec/wma.c

@@ -308,7 +308,7 @@ av_cold int ff_wma_init(AVCodecContext *avctx, int flags2)
     }
 #endif
 
-    /* init MDCT windows : simple sinus window */
+    /* init MDCT windows : simple sine window */
     for (i = 0; i < s->nb_block_sizes; i++) {
         ff_init_ff_sine_windows(s->frame_len_bits - i);
         s->windows[i] = ff_sine_windows[s->frame_len_bits - i];

libavcodec/wmaprodec.c

@@ -125,7 +125,7 @@ static VLC              vec4_vlc;         ///< 4 coefficients per symbol
 static VLC              vec2_vlc;         ///< 2 coefficients per symbol
 static VLC              vec1_vlc;         ///< 1 coefficient per symbol
 static VLC              coef_vlc[2];      ///< coefficient run length vlc codes
-static float            sin64[33];        ///< sinus table for decorrelation
+static float            sin64[33];        ///< sine table for decorrelation
 
 /**
  * @brief frame specific decoder context for a single channel
@@ -458,7 +458,7 @@ static av_cold int decode_init(AVCodecContext *avctx)
                      1.0 / (1 << (WMAPRO_BLOCK_MIN_BITS + i - 1))
                      / (1 << (s->bits_per_sample - 1)));
 
-    /** init MDCT windows: simple sinus window */
+    /** init MDCT windows: simple sine window */
     for (i = 0; i < WMAPRO_BLOCK_SIZES; i++) {
         const int win_idx = WMAPRO_BLOCK_MAX_BITS - i;
         ff_init_ff_sine_windows(win_idx);

libavcodec/wmavoice.c

@@ -618,7 +618,7 @@ static void calc_input_response(WMAVoiceContext *s, float *lpcs,
     }
 
     /* calculate the Hilbert transform of the gains, which we do (since this
-     * is a sinus input) by doing a phase shift (in theory, H(sin())=cos()).
+     * is a sine input) by doing a phase shift (in theory, H(sin())=cos()).
      * Hilbert_Transform(RDFT(x)) = Laplace_Transform(x), which calculates the
      * "moment" of the LPCs in this filter. */
     s->dct.dct_calc(&s->dct, lpcs);

libavresample/avresample-test.c

@@ -91,7 +91,7 @@ static void audiogen(AVLFG *rnd, void **data, enum AVSampleFormat sample_fmt,
 
     k = 0;
 
-    /* 1 second of single freq sinus at 1000 Hz */
+    /* 1 second of single freq sine at 1000 Hz */
     a = 0;
     for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) {
         v = sin(a) * 0.30;

tests/audiogen.c

@@ -48,7 +48,7 @@ static unsigned int myrnd(unsigned int *seed_ptr, int n)
 
 #define COS_TABLE_BITS 7
 
-/* integer cosinus */
+/* integer cosine */
 static const unsigned short cos_table[(1 << COS_TABLE_BITS) + 2] = {
     0x8000, 0x7ffe, 0x7ff6, 0x7fea, 0x7fd9, 0x7fc2, 0x7fa7, 0x7f87,
     0x7f62, 0x7f38, 0x7f0a, 0x7ed6, 0x7e9d, 0x7e60, 0x7e1e, 0x7dd6,
@@ -180,7 +180,7 @@ int main(int argc, char **argv)
     if ((ext = strrchr(argv[1], '.')) != NULL && !strcmp(ext, ".wav"))
         put_wav_header(sample_rate, nb_channels, 6 * sample_rate);
 
-    /* 1 second of single freq sinus at 1000 Hz */
+    /* 1 second of single freq sine at 1000 Hz */
     a = 0;
     for (i = 0; i < 1 * sample_rate; i++) {
         v = (int_cos(a) * 10000) >> FRAC_BITS;