Various spelling fixes

Signed-off-by: wm4 <wm4@nowhere>

DOCS/man/input.rst

@@ -1118,7 +1118,7 @@ Property list
 
 ``demuxer-cache-time``
     Approximate time of video buffered in the demuxer, in seconds. Same as
-    ``demuxer-cache-duration`` but returns the last timestamp of bufferred
+    ``demuxer-cache-duration`` but returns the last timestamp of buffered
     data in demuxer.
 
 ``demuxer-cache-idle``

DOCS/man/options.rst

@@ -838,7 +838,7 @@ Audio
     Note that many AOs have a ``device`` sub-option, which overrides the
     device selection of this option (but not the audio output selection).
     Likewise, forcing an AO with ``--ao`` will override the audio output
-    selection of ``--audio-device`` (but not the device selecton).
+    selection of ``--audio-device`` (but not the device selection).
 
     Currently not implemented for most AOs.
 
@@ -3347,7 +3347,7 @@ Miscellaneous
     you should not need to change this option.
 
     :decoder: Use decoder reordering functionality. Unlike in classic MPlayer
-              and mplayer2, this includes a dTS fallback. (Default.)
+              and mplayer2, this includes a DTS fallback. (Default.)
     :sort:    Maintain a buffer of unused pts values and use the lowest value
               for the frame.
     :auto:    Try to pick a working mode from the ones above automatically.

DOCS/mplayer-changes.rst

@@ -397,7 +397,7 @@ Slave mode
 
   Assuming the system supports ``/dev/stdin``.
 
-  (The option was readded in 0.5.1 and sets exactly these options. It was
+  (The option was added back in 0.5.1 and sets exactly these options. It was
   removed in 0.10.x again.)
 
 * A JSON RPC protocol giving access to the client API is also supported. See

DOCS/waf-buildsystem.rst

@@ -1,7 +1,7 @@
 waf build system overview
 =========================
 
-mpv's new build system is based on waf and it should completly replace the
+mpv's new build system is based on waf and it should completely replace the
 custom ./configure + Makefile based system inherited from MPlayer.
 
 Goals and the choice of waf
@@ -65,7 +65,7 @@ This defines a feature called ``vdpau`` which can be enabled or disabled by
 the users with configure flags (that's the meaning of ``--``). This feature
 depends on another feature whose name is ``x11``, and the autodetection check
 consists of running ``pkg-config`` and looking for ``vdpau`` with version
-``>= 0.2``. If the check succeds a ``#define HAVE_VDPAU 1`` will be added to
+``>= 0.2``. If the check succeeds a ``#define HAVE_VDPAU 1`` will be added to
 ``config.h``, if not ``#define HAVE_VDPAU 0`` will be added.
 
 The defines names are automatically prepended with ``HAVE_``, capitalized and

README.md

@@ -171,7 +171,7 @@ list of changes is located [here][mplayer-changes].
 Most activity happens on the IRC channel and the github issue tracker. The
 mailing lists are mostly unused.
 
- - **Github issue tracker**: [issue tracker][issue-tracker] (report bugs here)
+ - **GitHub issue tracker**: [issue tracker][issue-tracker] (report bugs here)
  - **User IRC Channel**: `#mpv` on `irc.freenode.net`
  - **Developer IRC Channel**: `#mpv-devel` on `irc.freenode.net`
  - **Users Mailing List**: `mpv-users@googlegroups.com` ([Archive / Subscribe][mpv-users]).

audio/chmap.h

@@ -47,7 +47,7 @@ enum mp_speaker_id {
     MP_SPEAKER_ID_TBL,          // TOP_BACK_LEFT
     MP_SPEAKER_ID_TBC,          // TOP_BACK_CENTER
     MP_SPEAKER_ID_TBR,          // TOP_BACK_RIGHT
-     // Inofficial/libav* extensions
+     // Unofficial/libav* extensions
     MP_SPEAKER_ID_DL = 29,      // STEREO_LEFT (stereo downmix special speakers)
     MP_SPEAKER_ID_DR,           // STEREO_RIGHT
     MP_SPEAKER_ID_WL,           // WIDE_LEFT

audio/filter/af.c

@@ -646,7 +646,7 @@ struct af_instance *af_add(struct af_stream *s, char *name, char *label,
         return NULL;
     new->label = talloc_strdup(new, label);
 
-    // Reinitalize the filter list
+    // Reinitialize the filter list
     if (af_reinit(s) != AF_OK) {
         af_remove_by_label(s, label);
         return NULL;

audio/filter/af_drc.c

@@ -131,7 +131,7 @@ static void method1_int16(af_drc_t *s, struct mp_audio *c)
     data[i] = tmp;
   }
 
-  // Evaulation of newavg (not 100% accurate because of values clamping)
+  // Evaluation of newavg (not 100% accurate because of values clamping)
   newavg = s->mul * curavg;
 
   // Stores computed values for future smoothing
@@ -168,7 +168,7 @@ static void method1_float(af_drc_t *s, struct mp_audio *c)
   for (i = 0; i < len; i++)
     data[i] *= s->mul;
 
-  // Evaulation of newavg (not 100% accurate because of values clamping)
+  // Evaluation of newavg (not 100% accurate because of values clamping)
   newavg = s->mul * curavg;
 
   // Stores computed values for future smoothing
@@ -216,7 +216,7 @@ static void method2_int16(af_drc_t *s, struct mp_audio *c)
     data[i] = tmp;
   }
 
-  // Evaulation of newavg (not 100% accurate because of values clamping)
+  // Evaluation of newavg (not 100% accurate because of values clamping)
   newavg = s->mul * curavg;
 
   // Stores computed values for future smoothing
@@ -262,7 +262,7 @@ static void method2_float(af_drc_t *s, struct mp_audio *c)
   for (i = 0; i < len; i++)
     data[i] *= s->mul;
 
-  // Evaulation of newavg (not 100% accurate because of values clamping)
+  // Evaluation of newavg (not 100% accurate because of values clamping)
   newavg = s->mul * curavg;
 
   // Stores computed values for future smoothing

audio/filter/af_export.c

@@ -167,7 +167,7 @@ static int filter(struct af_instance *af, struct mp_audio *data)
     return 0;
   struct mp_audio*      c   = data;          // Current working data
   af_export_t*  s   = af->priv;     // Setup for this instance
-  int16_t*      a   = c->planes[0];          // Incomming sound
+  int16_t*      a   = c->planes[0];          // Incoming sound
   int           nch = c->nch;        // Number of channels
   int           len = c->samples*c->nch; // Number of sample in data chunk
   int           sz  = s->sz;         // buffer size (in samples)

audio/filter/af_hrtf.c

@@ -206,7 +206,7 @@ static inline void matrix_decode(short *in, const int k, const int il,
       information about Lt, Rt correlation.  This effectively reshapes
       the front and rear "cones" to concentrate Lt + Rt to C and
       introduce Lt - Rt in L, R. */
-   /* 0.67677 is the emprical lower bound for lpr_gain. */
+   /* 0.67677 is the empirical lower bound for lpr_gain. */
    c_gain = 8 * (*adapt_lpr_gain - 0.67677);
    c_gain = c_gain > 0 ? c_gain : 0;
    /* c_gain should not be too high, not even reaching full

audio/filter/af_ladspa.c

@@ -144,7 +144,7 @@ static int af_ladspa_parse_plugin(struct af_instance *af) {
     LADSPA_PortRangeHint hint;
 
     if (!setup->libhandle)
-        return AF_ERROR; /* only call parse after a succesful load */
+        return AF_ERROR; /* only call parse after a successful load */
     if (!setup->plugin_descriptor)
         return AF_ERROR; /* same as above */
 

audio/mixer.c

@@ -227,7 +227,7 @@ void mixer_setbalance(struct mixer *mixer, float val)
         return;
     }
 
-    /* make all other channels pass thru since by default pan blocks all */
+    /* make all other channels pass through since by default pan blocks all */
     for (int i = 2; i < AF_NCH; i++) {
         float level[AF_NCH] = {0};
         level[i] = 1.f;

audio/out/ao_dsound.c

@@ -288,7 +288,7 @@ static int InitDirectSound(struct ao *ao)
     /* Set DirectSound Cooperative level, ie what control we want over Windows
      * sound device. In our case, DSSCL_EXCLUSIVE means that we can modify the
      * settings of the primary buffer, but also that only the sound of our
-     * application will be hearable when it will have the focus.
+     * application will be audible when it will have the focus.
      * !!! (this is not really working as intended yet because to set the
      * cooperative level you need the window handle of your application, and
      * I don't know of any easy way to get it. Especially since we might play
@@ -616,9 +616,9 @@ static int check_free_buffer_size(struct ao *ao)
     space = p->buffer_size - (p->write_offset - play_offset);
     // |              | <-- const --> |                |                 |
     // buffer start   play_cursor     write_cursor     p->write_offset   buffer end
-    // play_cursor is the actual postion of the play cursor
+    // play_cursor is the actual position of the play cursor
     // write_cursor is the position after which it is assumed to be save to write data
-    // p->write_offset is the postion where we actually write the data to
+    // p->write_offset is the position where we actually write the data to
     if (space > p->buffer_size)
         space -= p->buffer_size;                        // p->write_offset < play_offset
     // Check for buffer underruns. An underrun happens if DirectSound

common/msg.c

@@ -52,7 +52,7 @@ struct mp_log_root {
     bool module;
     bool show_time;
     bool termosd;       // use terminal control codes for status line
-    int blank_lines;    // number of lines useable by status
+    int blank_lines;    // number of lines usable by status
     int status_lines;   // number of current status lines
     bool color;
     int verbose;

demux/demux_subreader.c

@@ -292,7 +292,7 @@ static subtitle *sub_read_line_subviewer(stream_t *st, subtitle *current,
         current->start = a1 * 360000 + a2 * 6000 + a3 * 100 + a4 / 10;
         current->end   = b1 * 360000 + b2 * 6000 + b3 * 100 + b4 / 10;
 
-        /* Concat lines */
+        /* Concatenate lines */
         full_line[0] = 0;
         for (i = 0; i < SUB_MAX_TEXT; i++) {
             int blank = 1, len = 0;

options/m_property.h

@@ -54,7 +54,7 @@ enum mp_property_action {
     //  arg: struct m_property_switch_arg*
     M_PROPERTY_SWITCH,
 
-    // Get a string containing a parsable representation.
+    // Get a string containing a parseable representation.
     // Can't be overridden by property implementations.
     //  arg: char**
     M_PROPERTY_GET_STRING,

osdep/terminal-win.c

@@ -92,7 +92,7 @@ static void read_input(void)
         case KEY_EVENT: {
             KEY_EVENT_RECORD *record = &eventbuffer[i].Event.KeyEvent;
 
-            /*only a pressed key is interresting for us*/
+            /*only a pressed key is interesting for us*/
             if (record->bKeyDown) {
                 UINT vkey = record->wVirtualKeyCode;
                 bool ext = record->dwControlKeyState & ENHANCED_KEY;

video/out/filter_kernels.c

@@ -370,7 +370,7 @@ const struct filter_kernel mp_filter_kernels[] = {
     {{"robidouxsharp",  2,   cubic_bc, .params = {0.2620, 0.3690} }},
     {{"ewa_robidoux",       2,   cubic_bc, .params = {0.3782, 0.3109}}, .polar = true},
     {{"ewa_robidouxsharp",  2,   cubic_bc, .params = {0.2620, 0.3690}}, .polar = true},
-    // Miscalleaneous filters
+    // Miscellaneous filters
     {{"box",            1,   box, .resizable = true}},
     {{"nearest",        0.5, box}},
     {{"triangle",       1,   triangle, .resizable = true}},

video/out/vo_wayland.c

@@ -87,7 +87,7 @@ static const format_t format_table[] = {
 struct priv;
 
 // We only use double buffering but the creation and usage is still open to
-// triple buffering. Tripple buffering is now removed, because double buffering
+// triple buffering. Triple buffering is now removed, because double buffering
 // is now pixel-perfect.
 struct buffer_pool {
     shm_buffer_t **buffers;
@@ -159,7 +159,7 @@ static const format_t* is_wayland_format_supported(struct priv *p,
     return NULL;
 }
 
-// additinal buffer functions
+// additional buffer functions
 
 static void buffer_finalise_front(shm_buffer_t *buf)
 {
@@ -269,7 +269,7 @@ static bool resize(struct priv *p)
     struct vo_wayland_state *wl = p->wl;
 
     if (!p->video_bufpool.back_buffer || SHM_BUFFER_IS_BUSY(p->video_bufpool.back_buffer))
-        return false; // skip resizing if we can't garantuee pixel perfectness!
+        return false; // skip resizing if we can't guarantee pixel perfectness!
 
     int32_t x = wl->window.sh_x;
     int32_t y = wl->window.sh_y;
@@ -446,7 +446,7 @@ static void draw_osd_cb(void *ctx, struct sub_bitmaps *imgs)
         }
         else if (SHM_BUFFER_IS_BUSY(p->osd_buffers[id])) {
             // freed on release in buffer_listener
-            // garantuees pixel perfect resizing of subtitles and osd
+            // guarantees pixel perfect resizing of subtitles and osd
             SHM_BUFFER_SET_ONESHOT(p->osd_buffers[id]);
             p->osd_buffers[id] = shm_buffer_create(width,
                                                    height,
@@ -475,7 +475,7 @@ static void draw_osd_cb(void *ctx, struct sub_bitmaps *imgs)
         wl_surface_commit(s);
     }
     else {
-        // p->osd_buffer, garantueed to exist here
+        // p->osd_buffer, guaranteed to exist here
         assert(p->osd_buffers[id]);
         wl_surface_attach(s, p->osd_buffers[id]->buffer, 0, 0);
         wl_surface_commit(s);
@@ -490,7 +490,7 @@ static void draw_osd(struct vo *vo)
 {
     struct priv *p = vo->priv;
 
-    // deattach all buffers and attach all needed buffers in osd_draw
+    // detach all buffers and attach all needed buffers in osd_draw
     // only the most recent attach & commit is applied once the parent surface
     // is committed
     for (int i = 0; i < MAX_OSD_PARTS; ++i) {
@@ -569,7 +569,7 @@ static int reconfig(struct vo *vo, struct mp_image_params *fmt, int flags)
             p->video_format = &format_table[DEFAULT_FORMAT_ENTRY];
     }
 
-    // overides alpha
+    // overrides alpha
     // use rgb565 if performance is your main concern
     if (p->use_rgb565) {
         MP_INFO(p->wl, "using rgb565\n");
@@ -625,7 +625,7 @@ static int preinit(struct vo *vo)
     wl_display_dispatch(wl->display.display);
 
     // Commits on surfaces bound to a subsurface are cached until the parent
-    // surface is commited, in this case the video surface.
+    // surface is committed, in this case the video surface.
     // Which means we can call commit anywhere.
     struct wl_region *input =
         wl_compositor_create_region(wl->display.compositor);