diff options
| -rw-r--r-- | DOCS/man/input.rst | 2 | ||||
| -rw-r--r-- | DOCS/man/options.rst | 4 | ||||
| -rw-r--r-- | DOCS/mplayer-changes.rst | 2 | ||||
| -rw-r--r-- | DOCS/waf-buildsystem.rst | 4 | ||||
| -rw-r--r-- | README.md | 2 | ||||
| -rw-r--r-- | audio/chmap.h | 2 | ||||
| -rw-r--r-- | audio/filter/af.c | 2 | ||||
| -rw-r--r-- | audio/filter/af_drc.c | 8 | ||||
| -rw-r--r-- | audio/filter/af_export.c | 2 | ||||
| -rw-r--r-- | audio/filter/af_hrtf.c | 2 | ||||
| -rw-r--r-- | audio/filter/af_ladspa.c | 2 | ||||
| -rw-r--r-- | audio/mixer.c | 2 | ||||
| -rw-r--r-- | audio/out/ao_dsound.c | 6 | ||||
| -rw-r--r-- | common/msg.c | 2 | ||||
| -rw-r--r-- | demux/demux_subreader.c | 2 | ||||
| -rw-r--r-- | options/m_property.h | 2 | ||||
| -rw-r--r-- | osdep/terminal-win.c | 2 | ||||
| -rw-r--r-- | video/out/filter_kernels.c | 2 | ||||
| -rw-r--r-- | video/out/vo_wayland.c | 16 |
19 files changed, 33 insertions, 33 deletions
diff --git a/DOCS/man/input.rst b/DOCS/man/input.rst index 0b0ccae1ec..aa0c0593f1 100644 --- a/DOCS/man/input.rst +++ b/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`` diff --git a/DOCS/man/options.rst b/DOCS/man/options.rst index 1c35aca9eb..4086475e77 100644 --- a/DOCS/man/options.rst +++ b/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. diff --git a/DOCS/mplayer-changes.rst b/DOCS/mplayer-changes.rst index ed916c14d4..b0389a73f3 100644 --- a/DOCS/mplayer-changes.rst +++ b/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 diff --git a/DOCS/waf-buildsystem.rst b/DOCS/waf-buildsystem.rst index fa71cbbcd6..adb1bc2f3c 100644 --- a/DOCS/waf-buildsystem.rst +++ b/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 @@ -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]). diff --git a/audio/chmap.h b/audio/chmap.h index adb7481665..ba1072547b 100644 --- a/audio/chmap.h +++ b/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 diff --git a/audio/filter/af.c b/audio/filter/af.c index 5a686e813c..e67fc29203 100644 --- a/audio/filter/af.c +++ b/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; diff --git a/audio/filter/af_drc.c b/audio/filter/af_drc.c index 4344766349..472758c4c7 100644 --- a/audio/filter/af_drc.c +++ b/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 diff --git a/audio/filter/af_export.c b/audio/filter/af_export.c index f2613530e5..6020d9d98e 100644 --- a/audio/filter/af_export.c +++ b/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) diff --git a/audio/filter/af_hrtf.c b/audio/filter/af_hrtf.c index 94a1599cd0..3c8a89c665 100644 --- a/audio/filter/af_hrtf.c +++ b/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 diff --git a/audio/filter/af_ladspa.c b/audio/filter/af_ladspa.c index bd54dbb267..edde6a68b1 100644 --- a/audio/filter/af_ladspa.c +++ b/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 */ diff --git a/audio/mixer.c b/audio/mixer.c index 7ecd97449d..29727918f6 100644 --- a/audio/mixer.c +++ b/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; diff --git a/audio/out/ao_dsound.c b/audio/out/ao_dsound.c index 9d216e673b..8b1e10a10b 100644 --- a/audio/out/ao_dsound.c +++ b/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 diff --git a/common/msg.c b/common/msg.c index 3a441d0bcf..a2b0cf2169 100644 --- a/common/msg.c +++ b/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; diff --git a/demux/demux_subreader.c b/demux/demux_subreader.c index 55a85429c6..87a5e9aeb8 100644 --- a/demux/demux_subreader.c +++ b/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; diff --git a/options/m_property.h b/options/m_property.h index 7c5f924c27..93a4a73578 100644 --- a/options/m_property.h +++ b/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, diff --git a/osdep/terminal-win.c b/osdep/terminal-win.c index e6b17fd595..8dd2258bcf 100644 --- a/osdep/terminal-win.c +++ b/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; diff --git a/video/out/filter_kernels.c b/video/out/filter_kernels.c index ef69903df7..a748ac10ab 100644 --- a/video/out/filter_kernels.c +++ b/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}}, diff --git a/video/out/vo_wayland.c b/video/out/vo_wayland.c index 40f06aa7eb..3367e46fe6 100644 --- a/video/out/vo_wayland.c +++ b/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); |