Commit 46989dca authored by Aaron Boxer's avatar Aaron Boxer Committed by Tim-Philipp Müller

documentation: fix a number of typos

parent 8173596e
Pipeline #68609 passed with stages
in 44 minutes and 32 seconds
......@@ -353,7 +353,7 @@ New element features and additions
- rtpjitterbuffer has improved end-of-stream handling
- rtpmp4vpay will be prefered over rtpmp4gpay for MPEG-4 video in
- rtpmp4vpay will be preferred over rtpmp4gpay for MPEG-4 video in
autoplugging scenarios now
- rtspsrc now allows applications to send RTSP SET_PARAMETER and
......@@ -1208,7 +1208,7 @@ Cerbero has seen a number of improvements:
used in order to re-produce a specific build. To set a manifest, you
can set manifest = 'my_manifest.xml' in your configuration file, or
use the --manifest command line option. The command line option will
take precendence over anything specific in the configuration file.
take precedence over anything specific in the configuration file.
- The new build-deps command can be used to build only the
dependencies of a recipe, without the recipe itself.
......
......@@ -1700,7 +1700,7 @@
},
"properties": {
"drain-on-changes": {
"blurb": "Drains the filter when its coeficients change",
"blurb": "Drains the filter when its coefficients change",
"construct": false,
"construct-only": false,
"default": "true",
......@@ -2120,7 +2120,7 @@
},
"properties": {
"drain-on-changes": {
"blurb": "Drains the filter when its coeficients change",
"blurb": "Drains the filter when its coefficients change",
"construct": false,
"construct-only": false,
"default": "true",
......@@ -2289,7 +2289,7 @@
"writable": true
},
"drain-on-changes": {
"blurb": "Drains the filter when its coeficients change",
"blurb": "Drains the filter when its coefficients change",
"construct": false,
"construct-only": false,
"default": "true",
......@@ -4563,7 +4563,7 @@
"writable": true
},
"min": {
"blurb": "mininum buffer size",
"blurb": "minimum buffer size",
"construct": true,
"construct-only": false,
"default": "1",
......@@ -29808,7 +29808,7 @@
"writable": true
},
"tls-interaction": {
"blurb": "A GTlsInteraction object to promt the user for password or certificate",
"blurb": "A GTlsInteraction object to prompt the user for password or certificate",
"construct": false,
"construct-only": false,
"type-name": "GTlsInteraction",
......@@ -33823,7 +33823,7 @@
"writable": true
},
"multicast-iface": {
"blurb": "The network interface on which to join the multicast group.This allows multiple interfaces seperated by comma. (\"eth0,eth1\")",
"blurb": "The network interface on which to join the multicast group.This allows multiple interfaces separated by comma. (\"eth0,eth1\")",
"construct": false,
"construct-only": false,
"default": "NULL",
......@@ -1121,7 +1121,7 @@ gst_dvdemux_handle_pull_seek (GstDVDemux * demux, GstPad * pad,
gst_dvdemux_push_event (demux, new_event);
}
/* if successfull seek, we update our real segment and push
/* if successful seek, we update our real segment and push
* out the new segment. */
if (res) {
memcpy (&demux->time_segment, &seeksegment, sizeof (GstSegment));
......@@ -1149,7 +1149,7 @@ gst_dvdemux_handle_pull_seek (GstDVDemux * demux, GstPad * pad,
demux->need_segment = FALSE;
}
/* and restart the task in case it got paused explicitely or by
/* and restart the task in case it got paused explicitly or by
* the FLUSH_START event we pushed out. */
gst_pad_start_task (demux->sinkpad, (GstTaskFunction) gst_dvdemux_loop,
demux->sinkpad, NULL);
......@@ -1800,7 +1800,7 @@ gst_dvdemux_loop (GstPad * pad)
if (!gst_dvdemux_handle_pull_seek (dvdemux, dvdemux->videosrcpad,
event)) {
GST_ELEMENT_WARNING (dvdemux, STREAM, DECODE, (NULL),
("Error perfoming initial seek"));
("Error performing initial seek"));
}
gst_event_unref (event);
......
......@@ -31,7 +31,7 @@
* of the FLAC stream.
*
* Applications can set the tags to write using the #GstTagSetter interface.
* Tags contained withing the FLAC bitstream will be picked up
* Tags contained within the FLAC bitstream will be picked up
* automatically (and merged according to the merge mode set via the tag
* setter interface).
*
......
......@@ -370,7 +370,8 @@ gst_gdk_pixbuf_dec_flush (GstGdkPixbufDec * filter)
/* ERRORS */
no_pixbuf:
{
GST_ELEMENT_ERROR (filter, STREAM, DECODE, (NULL), ("error geting pixbuf"));
GST_ELEMENT_ERROR (filter, STREAM, DECODE, (NULL),
("error getting pixbuf"));
return GST_FLOW_ERROR;
}
channels_not_supported:
......
......@@ -200,7 +200,7 @@ gst_gtk_base_sink_get_widget (GstGtkBaseSink * gtk_sink)
"ignore-alpha", G_BINDING_BIDIRECTIONAL | G_BINDING_SYNC_CREATE);
/* Take the floating ref, other wise the destruction of the container will
* make this widget disapear possibly before we are done. */
* make this widget disappear possibly before we are done. */
gst_object_ref_sink (gtk_sink->widget);
gtk_sink->widget_destroy_id = g_signal_connect (gtk_sink->widget, "destroy",
G_CALLBACK (widget_destroy_cb), gtk_sink);
......
......@@ -591,7 +591,7 @@ gst_jack_audio_client_get_client (GstJackAudioClient * client)
* @client: a #GstJackAudioClient
* @active: new mode for the client
*
* Activate or deactive @client. When a client is activated it will receive
* Activate or deactivate @client. When a client is activated it will receive
* callbacks when data should be processed.
*
* Returns: 0 if all ok.
......
......@@ -4,7 +4,7 @@ libgstjpeg_la_SOURCES = \
gstjpeg.c \
gstjpegenc.c \
gstjpegdec.c
# deprected gstsmokeenc.c smokecodec.c gstsmokedec.c
# deprecated gstsmokeenc.c smokecodec.c gstsmokedec.c
libgstjpeg_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS)
libgstjpeg_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) $(GST_BASE_LIBS) $(GST_LIBS) -lgstvideo-$(GST_API_VERSION) \
......
......@@ -154,7 +154,7 @@ static guint mainloop_ref_ct = 0;
static GMutex pa_shared_resource_mutex;
/* We keep a custom ringbuffer that is backed up by data allocated by
* pulseaudio. We must also overide the commit function to write into
* pulseaudio. We must also override the commit function to write into
* pulseaudio memory instead. */
struct _GstPulseRingBuffer
{
......@@ -545,7 +545,7 @@ gst_pulseringbuffer_open_device (GstAudioRingBuffer * buf)
gst_pulsering_context_subscribe_cb, pctx);
/* try to connect to the server and wait for completion, we don't want to
* autospawn a deamon */
* autospawn a daemon */
GST_LOG_OBJECT (psink, "connect to server %s",
GST_STR_NULL (psink->server));
if (pa_context_connect (pctx->context, psink->server,
......@@ -685,7 +685,7 @@ gst_pulsering_stream_request_cb (pa_stream * s, size_t length, void *userdata)
if (pbuf->in_commit && (length >= rbuf->spec.segsize)) {
/* only signal when we are waiting in the commit thread
* and got request for atleast a segment */
* and got request for at least a segment */
pa_threaded_mainloop_signal (mainloop, 0);
}
}
......@@ -2431,7 +2431,7 @@ gst_pulsesink_set_volume (GstPulseSink * psink, gdouble volume)
if (pbuf->is_pcm)
gst_pulse_cvolume_from_linear (&v, pbuf->channels, volume);
else
/* FIXME: this will eventually be superceded by checks to see if the volume
/* FIXME: this will eventually be superseded by checks to see if the volume
* is readable/writable */
goto unlock;
......@@ -3003,7 +3003,7 @@ gst_pulsesink_change_props (GstPulseSink * psink, GstTagList * l)
static const gchar *const map[] = {
GST_TAG_TITLE, PA_PROP_MEDIA_TITLE,
/* might get overriden in the next iteration by GST_TAG_ARTIST */
/* might get overridden in the next iteration by GST_TAG_ARTIST */
GST_TAG_PERFORMER, PA_PROP_MEDIA_ARTIST,
GST_TAG_ARTIST, PA_PROP_MEDIA_ARTIST,
......
......@@ -36,7 +36,7 @@
#include <QtQuick/QQuickWindow>
#include <QOpenGLFramebufferObject>
/* compatability definitions... */
/* compatibility definitions... */
#ifndef GL_READ_FRAMEBUFFER
#define GL_READ_FRAMEBUFFER 0x8CA8
#endif
......
......@@ -50,7 +50,7 @@ typedef struct _GstDV1394SrcClass GstDV1394SrcClass;
struct _GstDV1394Src {
GstPushSrc element;
// consecutive=2, skip=4 will skip 4 frames, then let 2 consecutive ones thru
// consecutive=2, skip=4 will skip 4 frames, then let 2 consecutive ones through
gint consecutive;
gint skip;
gboolean drop_incomplete;
......
......@@ -741,7 +741,7 @@ foreach_add_tag (const GstTagList * list, const gchar * tag, gpointer userdata)
GST_LOG ("Processing tag %s (num=%u)", tag, num_tags);
if (num_tags > 1 && gst_tag_is_fixed (tag)) {
GST_WARNING ("Multiple occurences of fixed tag '%s', ignoring some", tag);
GST_WARNING ("Multiple occurrences of fixed tag '%s', ignoring some", tag);
num_tags = 1;
}
......
......@@ -436,7 +436,7 @@ static void
gst_wavpack_enc_set_wp_config (GstWavpackEnc * enc)
{
enc->wp_config = g_new0 (WavpackConfig, 1);
/* set general stream informations in the WavpackConfig */
/* set general stream information in the WavpackConfig */
enc->wp_config->bytes_per_sample = GST_ROUND_UP_8 (enc->depth) / 8;
enc->wp_config->bits_per_sample = enc->depth;
enc->wp_config->num_channels = enc->channels;
......@@ -708,7 +708,7 @@ gst_wavpack_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
WavpackCloseFile (enc->wp_context);
goto config_failed;
}
GST_DEBUG_OBJECT (enc, "setup of encoding context successfull");
GST_DEBUG_OBJECT (enc, "setup of encoding context successful");
}
if (enc->need_channel_remap) {
......
......@@ -457,7 +457,7 @@ gst_audio_dynamic_transform_soft_knee_compressor_float (GstAudioDynamic *
* f(x) = ax^2 + bx + c
*/
/* FIXME: If treshold is the same as the maximum
/* FIXME: If threshold is the same as the maximum
* we need to raise it a bit to prevent
* division by zero. */
if (threshold == 1.0)
......
......@@ -267,7 +267,7 @@ process_fft_##channels##_##width (GstAudioFXBaseFIRFilter * self, const g##ctype
* plus some more space for the inverse FFT below. \
* \
* The samples are put at offset kernel_length, the inverse FFT \
* overwrites everthing from offset 0 to length-kernel_length+1, keeping \
* overwrites everything from offset 0 to length-kernel_length+1, keeping \
* the last kernel_length-1 samples for copying to the next processing \
* step. \
*/ \
......@@ -558,14 +558,14 @@ gst_audio_fx_base_fir_filter_class_init (GstAudioFXBaseFIRFilterClass * klass)
/**
* GstAudioFXBaseFIRFilter:drain-on-changes:
*
* Whether the filter should be drained when its coeficients change
* Whether the filter should be drained when its coefficients change
*
* Note: Currently this only works if the kernel size is not changed!
* Support for drainless kernel size changes will be added in the future.
*/
g_object_class_install_property (gobject_class, PROP_DRAIN_ON_CHANGES,
g_param_spec_boolean ("drain-on-changes", "Drain on changes",
"Drains the filter when its coeficients change",
"Drains the filter when its coefficients change",
DEFAULT_DRAIN_ON_CHANGES,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
......
......@@ -63,7 +63,7 @@ struct _GstAudioFXBaseFIRFilter {
gboolean low_latency; /* work in slower low latency mode */
gboolean drain_on_changes; /* If the filter should be drained when
* coeficients change */
* coefficients change */
/* < private > */
GstAudioFXBaseFIRFilterProcessFunc process;
......
......@@ -53,7 +53,7 @@
* for the best overlap position. Scaletempo uses a statistical cross
* correlation (roughly a dot-product). Scaletempo consumes most of its CPU
* cycles here. One can use the #GstScaletempo:search propery to tune how far
* the algoritm looks.
* the algorithm looks.
*
*/
......
......@@ -53,7 +53,7 @@
* register the element factories and pad templates
* register the features
*
* exchange the string 'plugin' with your elemnt name
* exchange the string 'plugin' with your element name
*/
static gboolean
......
......@@ -154,7 +154,7 @@ gst_auto_detect_attach_ghost_pad (GstAutoDetect * self)
return res;
}
/* Hack to make initial linking work; ideally, this'd work even when
/* Hack to make initial linking work; ideally, this would work even when
* no target has been assigned to the ghostpad yet. */
static void
gst_auto_detect_reset (GstAutoDetect * self)
......
......@@ -2864,7 +2864,7 @@ gst_avi_demux_stream_index (GstAviDemux * avi)
if (map.size < 8)
goto too_small;
/* check tag first before blindy trying to read 'size' bytes */
/* check tag first before blindly trying to read 'size' bytes */
tag = GST_READ_UINT32_LE (map.data);
size = GST_READ_UINT32_LE (map.data + 4);
if (tag == GST_RIFF_TAG_LIST) {
......@@ -3377,7 +3377,7 @@ gst_avi_demux_stream_header_push (GstAviDemux * avi)
if (!gst_avi_demux_parse_avih (avi, sub, &avi->avih))
goto header_wrong_avih;
GST_DEBUG_OBJECT (avi, "AVI header ok, reading elemnts from header");
GST_DEBUG_OBJECT (avi, "AVI header ok, reading elements from header");
/* now, read the elements from the header until the end */
while (gst_riff_parse_chunk (GST_ELEMENT_CAST (avi), buf, &offset, &tag,
......@@ -5267,7 +5267,7 @@ gst_avi_demux_loop_data (GstAviDemux * avi)
}
if (avi->segment.rate > 0.0) {
/* only check this for fowards playback for now */
/* only check this for forwards playback for now */
if (keyframe && GST_CLOCK_TIME_IS_VALID (avi->segment.stop)
&& (timestamp > avi->segment.stop)) {
goto eos_stop;
......
......@@ -1382,7 +1382,7 @@ gst_avi_mux_riff_get_avi_header (GstAviMux * avimux)
gst_tag_list_foreach (tags, gst_avi_mux_write_tag, &bw);
if (info + 8 == gst_byte_writer_get_pos (&bw)) {
/* no tags writen, remove the empty INFO LIST as it is useless
/* no tags written, remove the empty INFO LIST as it is useless
* and prevents playback in vlc */
gst_byte_writer_set_pos (&bw, info - 4);
} else {
......
......@@ -26,7 +26,7 @@
*
* The progressreport element can be put into a pipeline to report progress,
* which is done by doing upstream duration and position queries in regular
* (real-time) intervals. Both the interval and the prefered query format
* (real-time) intervals. Both the interval and the preferred query format
* can be specified via the #GstProgressReport:update-freq and the
* #GstProgressReport:format property.
*
......
......@@ -137,7 +137,7 @@ gst_rnd_buffer_size_class_init (GstRndBufferSizeClass * klass)
0, G_MAXUINT32, DEFAULT_SEED,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_MINIMUM,
g_param_spec_int ("min", "mininum", "mininum buffer size",
g_param_spec_int ("min", "minimum", "minimum buffer size",
0, G_MAXINT32, DEFAULT_MIN,
G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_MAXIMUM,
......
......@@ -524,7 +524,7 @@ gst_deinterlace_class_init (GstDeinterlaceClass * klass)
*
* Some methods provide parameters which can be set by getting
* the "method" child via the #GstChildProxy interface and
* setting the appropiate properties on it.
* setting the appropriate properties on it.
*
* * tomsmocomp Motion Adaptive: Motion Search
* * greedyh Motion Adaptive: Advanced Detection
......@@ -1444,7 +1444,7 @@ gst_deinterlace_get_pattern_lock (GstDeinterlace * self, gboolean * flush_one)
break;
}
/* make complete matches more signficant */
/* make complete matches more significant */
if (k == length)
k += GST_DEINTERLACE_MAX_BUFFER_STATE_HISTORY;
......
/* sse.h
Streaming SIMD Extenstions (a.k.a. Katmai New Instructions)
Streaming SIMD Extensions (a.k.a. Katmai New Instructions)
GCC interface library for IA32.
To use this library, simply include this header file
......@@ -954,7 +954,7 @@ sse_ok(void)
/* Store FENCE - enforce ordering of stores before fence vs. stores
occuring after fence in source code.
occurring after fence in source code.
*/
#ifdef SSE_TRACE
#define sfence() \
......
......@@ -5,7 +5,7 @@
#ifndef IS_C
#ifdef SKIP_SEARCH
"movq %%mm6, %%mm0\n\t" // just use the results of our wierd bob
"movq %%mm6, %%mm0\n\t" // just use the results of our weird bob
#else
......@@ -114,7 +114,7 @@
return 0;
#else
#ifdef SKIP_SEARCH
out[0] = best[0]; // just use the results of our wierd bob
out[0] = best[0]; // just use the results of our weird bob
out[1] = best[1];
#else
diff[0] = diff[0] - MIN (diff[0], 10) - 4;
......
// -*- c++ -*-
// First, get and save our possible Bob values
// Assume our pixels are layed out as follows with x the calc'd bob value
// Assume our pixels are laid out as follows with x the calc'd bob value
// and the other pixels are from the current field
//
// j a b c k current field
......
// -*- c++ -*-
// First, get and save our possible Bob values
// Assume our pixels are layed out as follows with x the calc'd bob value
// Assume our pixels are laid out as follows with x the calc'd bob value
// and the other pixels are from the current field
//
// j a b c k current field
......
......@@ -57,7 +57,7 @@ typedef GstDeinterlaceSimpleMethodClass GstDeinterlaceMethodVFIRClass;
/*
* The MPEG2 spec uses a slightly harsher filter, they specify
* [-1 8 2 8 -1]. ffmpeg uses a similar filter but with more of
* a tendancy to blur than to use the local information. The
* a tendency to blur than to use the local information. The
* filter taps here are: [-1 4 2 4 -1].
*/
......
......@@ -39,7 +39,7 @@
*
* * `type` (G_TYPE_INT, 0-1): The application uses this field to specify which of the two methods
* specified in RFC 2833 to use. The value should be 0 for tones and 1 for
* named events. Tones are specified by their frequencies and events are specied
* named events. Tones are specified by their frequencies and events are specified
* by their number. This element can only take events as input. Do not confuse
* with "method" which specified the output.
*
......
......@@ -31,7 +31,7 @@
*
* * `type` (G_TYPE_INT, 0-1): Which of the two methods
* specified in RFC 2833 to use. The value should be 0 for tones and 1 for
* named events. Tones are specified by their frequencies and events are specied
* named events. Tones are specified by their frequencies and events are specified
* by their number. This element currently only recognizes events.
* Do not confuse with "method" which specified the output.
*
......
......@@ -37,7 +37,7 @@
*
* * `type` (G_TYPE_INT, 0-1): The application uses this field to specify which of the two methods
* specified in RFC 2833 to use. The value should be 0 for tones and 1 for
* named events. Tones are specified by their frequencies and events are specied
* named events. Tones are specified by their frequencies and events are specified
* by their number. This element can only take events as input. Do not confuse
* with "method" which specified the output.
*
......
......@@ -132,7 +132,7 @@ gst_edgetv_transform_frame (GstVideoFilter * vfilter, GstVideoFrame * in_frame,
g *= g;
b *= b;
r = r >> 5; /* To lack the lower bit for saturated addition, */
g = g >> 5; /* devide the value with 32, instead of 16. It is */
g = g >> 5; /* divide the value with 32, instead of 16. It is */
b = b >> 4; /* same as `v2 &= 0xfefeff' */
if (r > 127)
r = 127;
......
......@@ -48,7 +48,7 @@
#include "gstquark.h"
#include "gsteffectv.h"
/* number of frames of time-buffer. It should be as a configurable paramater */
/* number of frames of time-buffer. It should be as a configurable parameter */
/* This number also must be 2^n just for the speed. */
#define PLANES 16
......
......@@ -1003,7 +1003,7 @@ gst_flv_demux_update_resync (GstFlvDemux * demux, guint32 dts, gboolean discont,
gboolean ret = FALSE;
gint32 ddts = dts - *last;
if (!discont && ddts <= -RESYNC_THRESHOLD) {
/* Theoretically, we should use substract the duration of the last buffer,
/* Theoretically, we should use subtract the duration of the last buffer,
but this demuxer sends no durations on buffers, not sure if it cannot
know, or just does not care to calculate. */
*offset -= ddts * GST_MSECOND;
......@@ -3047,7 +3047,7 @@ gst_flv_demux_handle_seek_pull (GstFlvDemux * demux, GstEvent * event,
demux->seek_event = gst_event_ref (event);
demux->seek_time = seeksegment.position;
demux->state = FLV_STATE_SEEK;
/* do not know about succes yet, but we did care and handled it */
/* do not know about success yet, but we did care and handled it */
ret = TRUE;
goto exit;
}
......@@ -3123,7 +3123,7 @@ wrong_format:
}
}
/* If we can pull that's prefered */
/* If we can pull that's preferred */
static gboolean
gst_flv_demux_sink_activate (GstPad * sinkpad, GstObject * parent)
{
......
......@@ -39,7 +39,7 @@
* The application that wants to index the stream will create a new index object
* using gst_index_new() or gst_index_factory_make(). The index is assigned to a
* specific element, a bin or the whole pipeline. This will cause indexable
* elements to add entires to the index while playing.
* elements to add entries to the index while playing.
*/
/* FIXME: complete gobject annotations */
......@@ -326,7 +326,7 @@ gst_index_new (void)
/**
* gst_index_commit:
* @index: the index to commit
* @id: the writer that commited the index
* @id: the writer that committed the index
*
* Tell the index that the writer with the given id is done
* with this index and is not going to write any more entries
......@@ -787,7 +787,7 @@ gst_index_add_entry (GstIndex * index, GstIndexEntry * entry)
* gst_index_add_associationv:
* @index: the index to add the entry to
* @id: the id of the index writer
* @flags: optinal flags for this entry
* @flags: optional flags for this entry
* @n: number of associations
* @list: list of associations
*
......@@ -826,7 +826,7 @@ gst_index_add_associationv (GstIndex * index, gint id,
* gst_index_add_association:
* @index: the index to add the entry to
* @id: the id of the index writer
* @flags: optinal flags for this entry
* @flags: optional flags for this entry
* @format: the format of the value
* @value: the value
* @...: other format/value pairs or 0 to end the list
......
......@@ -493,7 +493,7 @@ set_caps_failed:
}
info_from_caps_failed:
{
GST_ERROR_OBJECT (self, "coud not get info from caps");
GST_ERROR_OBJECT (self, "could not get info from caps");
return FALSE;
}
}
......@@ -552,7 +552,7 @@ gst_deinterleave_sink_acceptcaps (GstPad * pad, GstObject * parent,
info_from_caps_failed:
{
GST_ERROR_OBJECT (self, "coud not get info from caps");
GST_ERROR_OBJECT (self, "could not get info from caps");
return FALSE;
}
}
......@@ -583,7 +583,7 @@ gst_deinterleave_getcaps (GstPad * pad, GstObject * parent, GstCaps * filter)
* to get all formats that are possible up- and downstream.
*
* For the pad for which the caps are requested we don't remove the channel
* informations as they must be in the returned caps and incompatibilities
* information as they must be in the returned caps and incompatibilities
* will be detected here already
*/
ret = gst_caps_new_any ();
......
......@@ -3209,7 +3209,7 @@ gst_qt_mux_start_file (GstQTMux * qtmux)
atom_moov_get_trak_count (qtmux->moov));
GST_OBJECT_UNLOCK (qtmux);
/* Now that we know how much reserved space is targetted,
/* Now that we know how much reserved space is targeted,
* output a free atom to fill the extra reserved */
ret = gst_qt_mux_send_free_atom (qtmux, &qtmux->header_size,
qtmux->reserved_moov_size - qtmux->base_moov_size, FALSE);
......
......@@ -133,7 +133,7 @@ struct _QtDemuxSample
#define QTSAMPLE_DTS(stream,sample) (QTSTREAMTIME_TO_GSTTIME((stream), (sample)->timestamp))
/* timestamp + offset + cslg_shift is the outgoing PTS */
#define QTSAMPLE_PTS(stream,sample) (QTSTREAMTIME_TO_GSTTIME((stream), (sample)->timestamp + (stream)->cslg_shift + (sample)->pts_offset))
/* timestamp + offset is the PTS used for internal seek calcuations */
/* timestamp + offset is the PTS used for internal seek calculations */
#define QTSAMPLE_PTS_NO_CSLG(stream,sample) (QTSTREAMTIME_TO_GSTTIME((stream), (sample)->timestamp + (sample)->pts_offset))
/* timestamp + duration - dts is the duration */
#define QTSAMPLE_DUR_DTS(stream, sample, dts) (QTSTREAMTIME_TO_GSTTIME ((stream), (sample)->timestamp + (sample)->duration) - (dts))
......@@ -2231,7 +2231,7 @@ gst_qtdemux_reset (GstQTDemux * qtdemux, gboolean hard)
/* Maps the @segment to the qt edts internal segments and pushes
* the correspnding segment event.
* the corresponding segment event.
*
* If it ends up being at a empty segment, a gap will be pushed and the next
* edts segment will be activated in sequence.
......@@ -4270,7 +4270,7 @@ qtdemux_parse_moof (GstQTDemux * qtdemux, const guint8 * buffer, guint length,
if (!qtdemux->upstream_format_is_time && !qtdemux->first_moof_already_parsed
&& !qtdemux->received_seek && GST_CLOCK_TIME_IS_VALID (min_dts)
&& min_dts != 0) {
/* Unless the user has explictly requested another seek, perform an
/* Unless the user has explicitly requested another seek, perform an
* internal seek to the time specified in the tfdt.
*
* This way if the user opens a file where the first tfdt is 1 hour
......@@ -5690,7 +5690,7 @@ extract_cc_from_data (QtDemuxStream * stream, const guint8 * data, gsize size,
GST_DEBUG_OBJECT (stream->pad, "here");
/* Check if we have somethig compatible */
/* Check if we have something compatible */
stsd_entry = CUR_STREAM (stream);
switch (stsd_entry->fourcc) {
case FOURCC_c608:{
......@@ -7229,7 +7229,7 @@ gst_qtdemux_process_adapter (GstQTDemux * demux, gboolean force)
*
* To keep track of the current buffer timestamp and starting point
* we use gst_adapter_prev_pts that gives us the PTS and the distance
* from the beggining of the buffer, with the distance and demux->offset
* from the beginning of the buffer, with the distance and demux->offset
* we know if it is still the same buffer or not.
*/
prev_pts = gst_adapter_prev_pts (demux->adapter, &dist);
......@@ -8062,7 +8062,7 @@ qtdemux_parse_node (GstQTDemux * qtdemux, GNode * node, const guint8 * buffer,
* the same format. */
/* video sample description size is 86 bytes without extension.
* node_length have to be bigger than 86 bytes because video sample
* description can include extenstions such as esds, fiel, glbl, etc. */
* description can include extensions such as esds, fiel, glbl, etc. */
if (node_length < 86) {
GST_WARNING_OBJECT (qtdemux, "%" GST_FOURCC_FORMAT
" sample description length too short (%u < 86)",
......@@ -9336,7 +9336,7 @@ qtdemux_stbl_init (GstQTDemux * qtdemux, QtDemuxStream * stream, GNode * stbl)
}
} else {
/* Ensure the cslg_shift value is consistent so we can use it
* unconditionnally to produce TS and Segment */
* unconditionally to produce TS and Segment */
stream->cslg_shift = 0;
}
......@@ -13813,7 +13813,7 @@ qtdemux_tag_add_blob (GNode * node, GstQtDemuxTagList * qtdemuxtaglist)
else
style = "iso";
/* santize the name for the caps. */
/* sanitize the name for the caps. */
for (i = 0; i < 4; i++) {
guint8 d = data[4 + i];
if (g_ascii_isalnum (d))
......
......@@ -63,7 +63,7 @@ static const gchar qt_lang_map[][4] = {
* 026 Hungarian
* 027 Estonian
* 028 Latvian / Lettish
* 029 Lappish / Saamish (used code for Nothern Sami)
* 029 Lappish / Saamish (used code for Northern Sami)
*/
"urd", "hin", "tha", "kor", "lit", "pol", "hun", "est", "lav", "sme",
......
......@@ -308,7 +308,7 @@ gst_level_get_property (GObject * object, guint prop_id,
* input sample data enters in *in_data and is not modified
* this filter only accepts signed audio data, so mid level is always 0
*
* for integers, this code considers the non-existant positive max value to be
* for integers, this code considers the non-existent positive max value to be
* full-scale; so max-1 will not map to 1.0
*/
......
......@@ -863,7 +863,7 @@ gst_ebml_write_buffer_header (GstEbmlWrite * ebml, guint32 id, guint64 length)
/**
* gst_ebml_write_buffer:
* @ebml: #GstEbmlWrite