diff options
| -rw-r--r-- | Changelog | 2 | ||||
| -rwxr-xr-x | configure | 2 | ||||
| -rw-r--r-- | doc/faq.texi | 4 | ||||
| -rw-r--r-- | ffplay.c | 3 | ||||
| -rw-r--r-- | libavcodec/aac.h | 2 | ||||
| -rw-r--r-- | libavcodec/twinvq.c | 8 | ||||
| -rw-r--r-- | libavcodec/wma.c | 2 | ||||
| -rw-r--r-- | libavcodec/wmaprodec.c | 4 | ||||
| -rw-r--r-- | libavcodec/wmavoice.c | 2 | ||||
| -rw-r--r-- | libavresample/avresample-test.c | 2 | ||||
| -rw-r--r-- | tests/audiogen.c | 4 |
11 files changed, 18 insertions, 17 deletions
@@ -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 @@ -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 diff --git a/doc/faq.texi b/doc/faq.texi index 4b0b09ccf2..a0ae537444 100644 --- a/doc/faq.texi +++ b/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? @@ -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])); diff --git a/libavcodec/aac.h b/libavcodec/aac.h index d586e271da..209c715e36 100644 --- a/libavcodec/aac.h +++ b/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; diff --git a/libavcodec/twinvq.c b/libavcodec/twinvq.c index ea80becb3e..7c449f3299 100644 --- a/libavcodec/twinvq.c +++ b/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, diff --git a/libavcodec/wma.c b/libavcodec/wma.c index b6dab4b65d..0122ee6fe3 100644 --- a/libavcodec/wma.c +++ b/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]; diff --git a/libavcodec/wmaprodec.c b/libavcodec/wmaprodec.c index ca57f64cda..4bd2d3631b 100644 --- a/libavcodec/wmaprodec.c +++ b/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); diff --git a/libavcodec/wmavoice.c b/libavcodec/wmavoice.c index ae1b70fd53..c0fd8233a4 100644 --- a/libavcodec/wmavoice.c +++ b/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); diff --git a/libavresample/avresample-test.c b/libavresample/avresample-test.c index 81e9bf0f50..697b4ba799 100644 --- a/libavresample/avresample-test.c +++ b/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; diff --git a/tests/audiogen.c b/tests/audiogen.c index 09cf429a71..e705061008 100644 --- a/tests/audiogen.c +++ b/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; |