[FFmpeg-cvslog] miscellaneous typo fixes
Diego Biurrun
git at videolan.org
Fri Jul 26 11:20:25 CEST 2013
ffmpeg | branch: master | Diego Biurrun <diego at biurrun.de> | Sat May 4 21:18:13 2013 +0200| [03039f4c8cdb4f8e5dc69905f3d73e2a531157e8] | committer: Diego Biurrun
miscellaneous typo fixes
> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=03039f4c8cdb4f8e5dc69905f3d73e2a531157e8
---
Changelog | 2 +-
avplay.c | 3 ++-
configure | 2 +-
doc/faq.texi | 4 ++--
libavcodec/aac.h | 2 +-
libavcodec/twinvq.c | 8 ++++----
libavcodec/wma.c | 2 +-
libavcodec/wmaprodec.c | 4 ++--
libavcodec/wmavoice.c | 2 +-
libavformat/avisynth.c | 26 +++++++++++++-------------
libavresample/avresample-test.c | 2 +-
tests/audiogen.c | 4 ++--
12 files changed, 31 insertions(+), 30 deletions(-)
diff --git a/Changelog b/Changelog
index 7a2a7c8..faf0393 100644
--- a/Changelog
+++ b/Changelog
@@ -479,7 +479,7 @@ 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
diff --git a/avplay.c b/avplay.c
index ba59a8c..00becbe 100644
--- a/avplay.c
+++ b/avplay.c
@@ -847,7 +847,8 @@ 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/configure b/configure
index 0bd54dd..adda2cf 100755
--- a/configure
+++ b/configure
@@ -173,7 +173,7 @@ 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]
--enable-bzlib enable bzlib [autodetect]
--enable-frei0r enable frei0r video filtering
--enable-gnutls enable gnutls [no]
diff --git a/doc/faq.texi b/doc/faq.texi
index 088ca03..b400124 100644
--- a/doc/faq.texi
+++ b/doc/faq.texi
@@ -202,8 +202,8 @@ Just create an "input.avs" text file with this single line ...
avconv -i input.avs
@end example
-For ANY other help on Avisynth, please visit the
- at uref{http://www.avisynth.org/, Avisynth homepage}.
+For ANY other help on AviSynth, please visit the
+ at uref{http://www.avisynth.org/, AviSynth homepage}.
@section How can I join video files?
diff --git a/libavcodec/aac.h b/libavcodec/aac.h
index b12d2ba..a7c9995 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 887a88c..8c304c6 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 db27646..f5ea64a 100644
--- a/libavcodec/wma.c
+++ b/libavcodec/wma.c
@@ -304,7 +304,7 @@ 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 d3bd381..bbf3216 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
@@ -449,7 +449,7 @@ 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 219f4e5..0ba2c48 100644
--- a/libavcodec/wmavoice.c
+++ b/libavcodec/wmavoice.c
@@ -620,7 +620,7 @@ 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/libavformat/avisynth.c b/libavformat/avisynth.c
index 43b3da5..b846c9f 100644
--- a/libavformat/avisynth.c
+++ b/libavformat/avisynth.c
@@ -1,5 +1,5 @@
/*
- * AVISynth support
+ * AviSynth support
* Copyright (c) 2006 DivX, Inc.
*
* This file is part of Libav.
@@ -32,29 +32,29 @@ typedef struct {
DWORD read;
LONG chunck_size;
LONG chunck_samples;
-} AVISynthStream;
+} AviSynthStream;
typedef struct {
PAVIFILE file;
AVISynthStream *streams;
int nb_streams;
int next_stream;
-} AVISynthContext;
+} AviSynthContext;
static int avisynth_read_header(AVFormatContext *s)
{
- AVISynthContext *avs = s->priv_data;
+ AviSynthContext *avs = s->priv_data;
HRESULT res;
AVIFILEINFO info;
DWORD id;
AVStream *st;
- AVISynthStream *stream;
+ AviSynthStream *stream;
wchar_t filename_wchar[1024] = { 0 };
char filename_char[1024] = { 0 };
AVIFileInit();
- /* avisynth can't accept UTF-8 filename */
+ /* AviSynth cannot accept UTF-8 file names. */
MultiByteToWideChar(CP_UTF8, 0, s->filename, -1, filename_wchar, 1024);
WideCharToMultiByte(CP_THREAD_ACP, 0, filename_wchar, -1, filename_char, 1024, NULL, NULL);
res = AVIFileOpen(&avs->file, filename_char, OF_READ|OF_SHARE_DENY_WRITE, NULL);
@@ -73,7 +73,7 @@ static int avisynth_read_header(AVFormatContext *s)
return -1;
}
- avs->streams = av_mallocz(info.dwStreams * sizeof(AVISynthStream));
+ avs->streams = av_mallocz(info.dwStreams * sizeof(AviSynthStream));
for (id=0; id<info.dwStreams; id++)
{
@@ -154,9 +154,9 @@ static int avisynth_read_header(AVFormatContext *s)
static int avisynth_read_packet(AVFormatContext *s, AVPacket *pkt)
{
- AVISynthContext *avs = s->priv_data;
+ AviSynthContext *avs = s->priv_data;
HRESULT res;
- AVISynthStream *stream;
+ AviSynthStream *stream;
int stream_id = avs->next_stream;
LONG read_size;
@@ -188,7 +188,7 @@ static int avisynth_read_packet(AVFormatContext *s, AVPacket *pkt)
static int avisynth_read_close(AVFormatContext *s)
{
- AVISynthContext *avs = s->priv_data;
+ AviSynthContext *avs = s->priv_data;
int i;
for (i=0;i<avs->nb_streams;i++)
@@ -204,7 +204,7 @@ static int avisynth_read_close(AVFormatContext *s)
static int avisynth_read_seek(AVFormatContext *s, int stream_index, int64_t pts, int flags)
{
- AVISynthContext *avs = s->priv_data;
+ AviSynthContext *avs = s->priv_data;
int stream_id;
for (stream_id = 0; stream_id < avs->nb_streams; stream_id++)
@@ -217,8 +217,8 @@ static int avisynth_read_seek(AVFormatContext *s, int stream_index, int64_t pts,
AVInputFormat ff_avisynth_demuxer = {
.name = "avisynth",
- .long_name = NULL_IF_CONFIG_SMALL("AVISynth"),
- .priv_data_size = sizeof(AVISynthContext),
+ .long_name = NULL_IF_CONFIG_SMALL("AviSynth"),
+ .priv_data_size = sizeof(AviSynthContext),
.read_header = avisynth_read_header,
.read_packet = avisynth_read_packet,
.read_close = avisynth_read_close,
diff --git a/libavresample/avresample-test.c b/libavresample/avresample-test.c
index 81e9bf0..697b4ba 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 4fa4656..ce6f785 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;
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