Commit 14644457 authored by Piotr Fusik's avatar Piotr Fusik Committed by Stefan Sauer

various: typo fixes

Fix typos in code and docs. Fixes. #658984
parent 0cce8ab9
......@@ -82,7 +82,7 @@ Design:
Whenever new samples are to be put into the ringbuffer, the position of the
read pointer is taken. The required write position is taken and the diff
is made between the required qnd actual position. If the defference is <0,
is made between the required and actual position. If the difference is <0,
the sample is too late. If the difference is bigger than segtotal, the
writing part has to wait for the play pointer to advance.
......
......@@ -57,7 +57,7 @@ fine-tune the process.
Get a list of elementfactories for @pad with @caps. This function is used to
instruct decodebin2 of the elements it should try to autoplug. The default
behaviour when this function is not overridern is to get all elements that
behaviour when this function is not overriden is to get all elements that
can handle @caps from the registry sorted by rank.
- 'autoplug-select' :
......@@ -142,7 +142,7 @@ Description:
Multiple input-output data queue
The GstMultiQueue achieves the same functionnality as GstQueue, with a few
The GstMultiQueue achieves the same functionality as GstQueue, with a few
differences:
* Multiple streams handling.
......
......@@ -16,13 +16,13 @@ A. Problems this proposal attempts to solve
* Duplication of pipeline code for gstreamer-based applications
wishing to encode and or mux streams, leading to subtle differences
and inconsistencies accross those applications.
and inconsistencies across those applications.
* No unified system for describing encoding targets for applications
in a user-friendly way.
* No unified system for creating encoding targets for applications,
resulting in duplication of code accross all applications,
resulting in duplication of code across all applications,
differences and inconsistencies that come with that duplication,
and applications hardcoding element names and settings resulting in
poor portability.
......
......@@ -86,7 +86,7 @@ given an input format, channel position manipulation, dithering and
quantizing configuration, and output format, a Orc code generator
would create an OrcProgram, add the appropriate instructions to do
each step based on the configuration, and then compile the program.
Sucessfully compiling the program would return a function pointer
Successfully compiling the program would return a function pointer
that can be called to perform the operation.
This sort of advanced usage requires structural changes to current
......
......@@ -11,7 +11,7 @@ Consider the following use case:
the existing file we are writing to and start writing to a new file.
We want the new file to start with a keyframe so that one can start decoding
the file immediatly.
the file immediately.
Components:
......
......@@ -7,7 +7,7 @@ Status:
Purpose:
Provide an standarized generic way to introduce Video Acceleration APIs in
Provide an standardized generic way to introduce Video Acceleration APIs in
already available elements instead of duplicating those into specialized ones.
Provide a mechanism for a light GstBuffer subclassing in order to be able
......@@ -26,7 +26,7 @@ Proposal:
video/x-raw-va
Light subclassing embeding an structure in the data field of a standard
Light subclassing embedding an structure in the data field of a standard
GstBuffer.
struct {
......
......@@ -800,7 +800,7 @@ gst_alsa_mixer_set_record (GstAlsaMixer * mixer,
snd_mixer_selem_set_capture_switch_all (alsa_track->element,
record ? 1 : 0);
/* update all tracks in same exlusive cswitch group */
/* update all tracks in same exclusive cswitch group */
if (alsa_track->alsa_flags & GST_ALSA_MIXER_TRACK_CSWITCH_EXCL) {
GList *item;
......
......@@ -548,7 +548,7 @@ gst_visual_src_query (GstPad * pad, GstQuery * query)
GST_TIME_FORMAT " max %" GST_TIME_FORMAT,
GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
/* the max samples we must buffer buffer */
/* the max samples we must buffer */
max_samples = MAX (VISUAL_SAMPLES, visual->spf);
our_latency =
gst_util_uint64_scale_int (max_samples, GST_SECOND, visual->rate);
......
......@@ -99,7 +99,7 @@ with great efficiency.
1) the streaming mode.
In this mode, the ogg demuxer receives buffers in the _chain() function which
are then simply submited to the ogg sync layer. Pages are then processed when
are then simply submitted to the ogg sync layer. Pages are then processed when
the sync layer detects them, pads are created for new chains and packets are
sent to the peer elements of the pads.
......
......@@ -572,7 +572,7 @@ gst_ogg_demux_chain_peer (GstOggPad * pad, ogg_packet * packet,
pad->current_granule);
} else if (ogg->segment.rate > 0.0 && pad->current_granule != -1) {
pad->current_granule += duration;
GST_DEBUG_OBJECT (ogg, "interpollating granule %" G_GUINT64_FORMAT,
GST_DEBUG_OBJECT (ogg, "interpolating granule %" G_GUINT64_FORMAT,
pad->current_granule);
}
if (ogg->segment.rate < 0.0 && packet->granulepos == -1) {
......
......@@ -1411,7 +1411,7 @@ theora_dec_flush_decode (GstTheoraDec * dec)
while (dec->queued) {
GstBuffer *buf = GST_BUFFER_CAST (dec->queued->data);
/* iterate ouput queue an push downstream */
/* iterate output queue an push downstream */
res = gst_pad_push (dec->srcpad, buf);
dec->queued = g_list_delete_link (dec->queued, dec->queued);
......
......@@ -73,7 +73,7 @@ struct _GstTheoraDec
gint offset_x, offset_y;
gint output_bpp;
/* telemetry debuging options */
/* telemetry debugging options */
gint telemetry_mv;
gint telemetry_mbmode;
gint telemetry_qi;
......
......@@ -328,7 +328,7 @@ theora_parse_set_streamheader (GstTheoraParse * parse)
parse->shift = parse->info.keyframe_granule_shift;
/* With libtheora-1.0beta1 the granulepos scheme was changed:
* where earlier the granulepos refered to the index/beginning
* where earlier the granulepos referred to the index/beginning
* of a frame, it now refers to the end, which matches the use
* in vorbis/speex. We check the bitstream version from the header so
* we know which way to interpret the incoming granuepos
......
......@@ -552,7 +552,7 @@ vorbis_handle_data_packet (GstVorbisDec * vd, ogg_packet * packet,
/* normal data packet */
/* FIXME, we can skip decoding if the packet is outside of the
* segment, this is however not very trivial as we need a previous
* packet to decode the current one so we must be carefull not to
* packet to decode the current one so we must be careful not to
* throw away too much. For now we decode everything and clip right
* before pushing data. */
......
......@@ -274,9 +274,9 @@ gst_app_sink_class_init (GstAppSinkClass * klass)
/**
* GstAppSink::eos:
* @appsink: the appsink element that emited the signal
* @appsink: the appsink element that emitted the signal
*
* Signal that the end-of-stream has been reached. This signal is emited from
* Signal that the end-of-stream has been reached. This signal is emitted from
* the steaming thread.
*/
gst_app_sink_signals[SIGNAL_EOS] =
......@@ -285,18 +285,18 @@ gst_app_sink_class_init (GstAppSinkClass * klass)
NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
/**
* GstAppSink::new-preroll:
* @appsink: the appsink element that emited the signal
* @appsink: the appsink element that emitted the signal
*
* Signal that a new preroll buffer is available.
*
* This signal is emited from the steaming thread and only when the
* This signal is emitted from the steaming thread and only when the
* "emit-signals" property is %TRUE.
*
* The new preroll buffer can be retrieved with the "pull-preroll" action
* signal or gst_app_sink_pull_preroll() either from this signal callback
* or from any other thread.
*
* Note that this signal is only emited when the "emit-signals" property is
* Note that this signal is only emitted when the "emit-signals" property is
* set to %TRUE, which it is not by default for performance reasons.
*/
gst_app_sink_signals[SIGNAL_NEW_PREROLL] =
......@@ -305,18 +305,18 @@ gst_app_sink_class_init (GstAppSinkClass * klass)
NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
/**
* GstAppSink::new-buffer:
* @appsink: the appsink element that emited the signal
* @appsink: the appsink element that emitted the signal
*
* Signal that a new buffer is available.
*
* This signal is emited from the steaming thread and only when the
* This signal is emitted from the steaming thread and only when the
* "emit-signals" property is %TRUE.
*
* The new buffer can be retrieved with the "pull-buffer" action
* signal or gst_app_sink_pull_buffer() either from this signal callback
* or from any other thread.
*
* Note that this signal is only emited when the "emit-signals" property is
* Note that this signal is only emitted when the "emit-signals" property is
* set to %TRUE, which it is not by default for performance reasons.
*/
gst_app_sink_signals[SIGNAL_NEW_BUFFER] =
......@@ -325,18 +325,18 @@ gst_app_sink_class_init (GstAppSinkClass * klass)
NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
/**
* GstAppSink::new-buffer-list:
* @appsink: the appsink element that emited the signal
* @appsink: the appsink element that emitted the signal
*
* Signal that a new bufferlist is available.
*
* This signal is emited from the steaming thread and only when the
* This signal is emitted from the steaming thread and only when the
* "emit-signals" property is %TRUE.
*
* The new buffer can be retrieved with the "pull-buffer-list" action
* signal or gst_app_sink_pull_buffer_list() either from this signal callback
* or from any other thread.
*
* Note that this signal is only emited when the "emit-signals" property is
* Note that this signal is only emitted when the "emit-signals" property is
* set to %TRUE, which it is not by default for performance reasons.
*/
gst_app_sink_signals[SIGNAL_NEW_BUFFER_LIST] =
......@@ -1066,7 +1066,7 @@ gst_app_sink_set_emit_signals (GstAppSink * appsink, gboolean emit)
*
* Check if appsink will emit the "new-preroll" and "new-buffer" signals.
*
* Returns: %TRUE if @appsink is emiting the "new-preroll" and "new-buffer"
* Returns: %TRUE if @appsink is emitting the "new-preroll" and "new-buffer"
* signals.
*
* Since: 0.10.22
......@@ -1339,7 +1339,7 @@ gst_app_sink_pull_buffer_list (GstAppSink * appsink)
* This is an alternative to using the signals, it has lower overhead and is thus
* less expensive, but also less flexible.
*
* If callbacks are installed, no signals will be emited for performance
* If callbacks are installed, no signals will be emitted for performance
* reasons.
*
* Since: 0.10.23
......
......@@ -37,7 +37,7 @@
* byte buffers.
*
* The main way of handing data to the appsrc element is by calling the
* gst_app_src_push_buffer() method or by emiting the push-buffer action signal.
* gst_app_src_push_buffer() method or by emitting the push-buffer action signal.
* This will put the buffer onto a queue from which appsrc will read from in its
* streaming thread. It is important to note that data transport will not happen
* from the thread that performed the push-buffer call.
......@@ -49,7 +49,7 @@
* block the push-buffer method until free data becomes available again.
*
* When the internal queue is running out of data, the "need-data" signal is
* emited, which signals the application that it should start pushing more data
* emitted, which signals the application that it should start pushing more data
* into appsrc.
*
* In addition to the "need-data" and "enough-data" signals, appsrc can emit the
......@@ -62,7 +62,7 @@
* These signals allow the application to operate the appsrc in two different
* ways:
*
* The push model, in which the application repeadedly calls the push-buffer method
* The push model, in which the application repeatedly calls the push-buffer method
* with a new buffer. Optionally, the queue size in the appsrc can be controlled
* with the enough-data and need-data signals by respectively stopping/starting
* the push-buffer calls. This is a typical mode of operation for the
......@@ -333,7 +333,7 @@ gst_app_src_class_init (GstAppSrcClass * klass)
/**
* GstAppSrc::block
*
* When max-bytes are queued and after the enough-data signal has been emited,
* When max-bytes are queued and after the enough-data signal has been emitted,
* block any further push-buffer calls until the amount of queued bytes drops
* below the max-bytes limit.
*/
......@@ -406,7 +406,7 @@ gst_app_src_class_init (GstAppSrcClass * klass)
/**
* GstAppSrc::need-data:
* @appsrc: the appsrc element that emited the signal
* @appsrc: the appsrc element that emitted the signal
* @length: the amount of bytes needed.
*
* Signal that the source needs more data. In the callback or from another
......@@ -425,11 +425,11 @@ gst_app_src_class_init (GstAppSrcClass * klass)
/**
* GstAppSrc::enough-data:
* @appsrc: the appsrc element that emited the signal
* @appsrc: the appsrc element that emitted the signal
*
* Signal that the source has enough data. It is recommended that the
* application stops calling push-buffer until the need-data signal is
* emited again to avoid excessive buffer queueing.
* emitted again to avoid excessive buffer queueing.
*/
gst_app_src_signals[SIGNAL_ENOUGH_DATA] =
g_signal_new ("enough-data", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST,
......@@ -438,7 +438,7 @@ gst_app_src_class_init (GstAppSrcClass * klass)
/**
* GstAppSrc::seek-data:
* @appsrc: the appsrc element that emited the signal
* @appsrc: the appsrc element that emitted the signal
* @offset: the offset to seek to
*
* Seek to the given offset. The next push-buffer should produce buffers from
......@@ -1010,7 +1010,7 @@ gst_app_src_create (GstBaseSrc * bsrc, guint64 offset, guint size,
* random-access mode (where a buffer is normally pushed in the above
* signal) we can still be empty because the pushed buffer got flushed or
* when the application pushes the requested buffer later, we support both
* possiblities. */
* possibilities. */
if (!g_queue_is_empty (priv->queue))
continue;
......@@ -1391,7 +1391,7 @@ gst_app_src_set_emit_signals (GstAppSrc * appsrc, gboolean emit)
*
* Check if appsrc will emit the "new-preroll" and "new-buffer" signals.
*
* Returns: %TRUE if @appsrc is emiting the "new-preroll" and "new-buffer"
* Returns: %TRUE if @appsrc is emitting the "new-preroll" and "new-buffer"
* signals.
*
* Since: 0.10.23
......@@ -1588,7 +1588,7 @@ flushing:
* This is an alternative to using the signals, it has lower overhead and is thus
* less expensive, but also less flexible.
*
* If callbacks are installed, no signals will be emited for performance
* If callbacks are installed, no signals will be emitted for performance
* reasons.
*
* Since: 0.10.23
......
......@@ -50,7 +50,7 @@ typedef struct _GstAppSrcPrivate GstAppSrcPrivate;
* and when it is set to -1, any number of bytes can be pushed into @appsrc.
* @enough_data: Called when appsrc has enough data. It is recommended that the
* application stops calling push-buffer until the need_data callback is
* emited again to avoid excessive buffer queueing.
* emitted again to avoid excessive buffer queueing.
* @seek_data: Called when a seek should be performed to the offset.
* The next push-buffer should produce buffers from the new @offset.
* This callback is only called for seekable stream types.
......
......@@ -708,7 +708,7 @@ done:
* @rate: sample rate.
* @frame_size: size of one audio frame in bytes.
*
* Clip the the buffer to the given %GstSegment.
* Clip the buffer to the given %GstSegment.
*
* After calling this function the caller does not own a reference to
* @buffer anymore.
......
......@@ -2009,7 +2009,7 @@ gst_audio_encoder_set_perfect_timestamp (GstAudioEncoder * enc,
*
* Queries encoder perfect timestamp behaviour.
*
* Returns: TRUE if pefect timestamp setting enabled.
* Returns: TRUE if perfect timestamp setting enabled.
*
* MT safe.
*
......
......@@ -342,7 +342,7 @@ gst_base_audio_sink_init (GstBaseAudioSink * baseaudiosink,
if (feature) {
if (strcmp (gst_plugin_feature_get_name (feature), "pulsesink") == 0) {
if (!gst_plugin_feature_check_version (feature, 0, 10, 17)) {
/* we're dealing with an old pulsesink, we need to disable time corection */
/* we're dealing with an old pulsesink, we need to disable time correction */
GST_DEBUG ("disable time offset");
baseaudiosink->priv->do_time_offset = FALSE;
}
......@@ -2119,7 +2119,7 @@ gst_base_audio_sink_async_play (GstBaseSink * basesink)
sink->priv->sync_latency = TRUE;
gst_ring_buffer_may_start (sink->ringbuffer, TRUE);
if (basesink->pad_mode == GST_ACTIVATE_PULL) {
/* we always start the ringbuffer in pull mode immediatly */
/* we always start the ringbuffer in pull mode immediately */
gst_ring_buffer_start (sink->ringbuffer);
}
......@@ -2173,7 +2173,7 @@ gst_base_audio_sink_change_state (GstElement * element,
gst_ring_buffer_may_start (sink->ringbuffer, TRUE);
if (GST_BASE_SINK_CAST (sink)->pad_mode == GST_ACTIVATE_PULL ||
g_atomic_int_get (&sink->abidata.ABI.eos_rendering) || eos) {
/* we always start the ringbuffer in pull mode immediatly */
/* we always start the ringbuffer in pull mode immediately */
/* sync rendering on eos needs running clock,
* and others need running clock when finished rendering eos */
gst_ring_buffer_start (sink->ringbuffer);
......@@ -2241,7 +2241,7 @@ gst_base_audio_sink_change_state (GstElement * element,
/* ERRORS */
open_failed:
{
/* subclass must post a meaningfull error message */
/* subclass must post a meaningful error message */
GST_DEBUG_OBJECT (sink, "open failed");
return GST_STATE_CHANGE_FAILURE;
}
......
......@@ -895,7 +895,7 @@ gst_base_audio_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
running_time_sample =
gst_util_uint64_scale_int (running_time, spec->rate, GST_SECOND);
/* the segmentnr corrensponding to running_time, round down */
/* the segmentnr corresponding to running_time, round down */
running_time_segment = running_time_sample / sps;
/* the segment currently read from the ringbuffer */
......@@ -921,7 +921,7 @@ gst_base_audio_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
*
* 1. We are more than the length of the ringbuffer behind.
* The length of the ringbuffer then gets to dictate
* the threshold for what is concidered "too late"
* the threshold for what is considered "too late"
*
* 2. If this is our first buffer.
* We know that we should catch up to running_time
......@@ -1152,7 +1152,7 @@ gst_base_audio_src_change_state (GstElement * element,
/* ERRORS */
open_failed:
{
/* subclass must post a meaningfull error message */
/* subclass must post a meaningful error message */
GST_DEBUG_OBJECT (src, "open failed");
return GST_STATE_CHANGE_FAILURE;
}
......
......@@ -1771,7 +1771,7 @@ not_started:
*
* Commit @in_samples samples pointed to by @data to the ringbuffer @buf.
*
* @in_samples and @out_samples define the rate conversion to perform on the the
* @in_samples and @out_samples define the rate conversion to perform on the
* samples in @data. For negative rates, @out_samples must be negative and
* @in_samples positive.
*
......
......@@ -104,7 +104,7 @@ void gst_audio_set_caps_channel_positions_list
gint num_positions);
/* Custom fixate function. Elements that implement some sort of
* channel conversion algorhithm should use this function for
* channel conversion algorithm should use this function for
* fixating on GstAudioChannelPosition properties. It will take
* care of equal channel positioning (left/right). Caller g_free()s
* the return value. The input properties may be (and are supposed
......
......@@ -31,7 +31,7 @@
*
* #GstFFTF32 provides a FFT implementation and related functions for
* 32 bit float samples. To use this call gst_fft_f32_new() for
* allocating a #GstFFTF32 instance with the appropiate parameters and
* allocating a #GstFFTF32 instance with the appropriate parameters and
* then call gst_fft_f32_fft() or gst_fft_f32_inverse_fft() to perform the
* FFT or inverse FFT on a buffer of samples.
*
......
......@@ -31,7 +31,7 @@
*
* #GstFFTF64 provides a FFT implementation and related functions for
* 64 bit float samples. To use this call gst_fft_f64_new() for
* allocating a #GstFFTF64 instance with the appropiate parameters and
* allocating a #GstFFTF64 instance with the appropriate parameters and
* then call gst_fft_f64_fft() or gst_fft_f64_inverse_fft() to perform the
* FFT or inverse FFT on a buffer of samples.
*
......
......@@ -31,7 +31,7 @@
*
* #GstFFTS16 provides a FFT implementation and related functions for
* signed 16 bit integer samples. To use this call gst_fft_s16_new() for
* allocating a #GstFFTS16 instance with the appropiate parameters and
* allocating a #GstFFTS16 instance with the appropriate parameters and
* then call gst_fft_s16_fft() or gst_fft_s16_inverse_fft() to perform the
* FFT or inverse FFT on a buffer of samples.
*
......
......@@ -31,7 +31,7 @@
*
* #GstFFTS32 provides a FFT implementation and related functions for
* signed 32 bit integer samples. To use this call gst_fft_s32_new() for
* allocating a #GstFFTS32 instance with the appropiate parameters and
* allocating a #GstFFTS32 instance with the appropriate parameters and
* then call gst_fft_s32_fft() or gst_fft_s32_inverse_fft() to perform the
* FFT or inverse FFT on a buffer of samples.
*
......
......@@ -53,7 +53,7 @@
* mouse moving over a clickable region, or the set of available angles changing.
* </para><para>
* The GstNavigation message functions provide functions for creating and parsing
* custom bus messages for signalling GstNavigation changes.
* custom bus messages for signaling GstNavigation changes.
* </para>
* </listitem>
* </itemizedlist>
......
......@@ -501,7 +501,7 @@ gst_x_overlay_expose (GstXOverlay * overlay)
* @handle_events: a #gboolean indicating if events should be handled or not.
*
* Tell an overlay that it should handle events from the window system. These
* events are forwared upstream as navigation events. In some window system,
* events are forwarded upstream as navigation events. In some window system,
* events are not propagated in the window hierarchy if a client is listening
* for them. This method allows you to disable events handling completely
* from the XOverlay.
......
......@@ -288,7 +288,7 @@ gst_netaddress_get_address_bytes (const GstNetAddress * naddr,
* Set just the address bytes stored in @naddr into @address.
*
* Note that @port must be expressed in network byte order, use g_htons() to
* convert it to network byte order order. IP4 address bytes must also be
* convert it to network byte order. IP4 address bytes must also be
* stored in network byte order.
*
* Returns: number of bytes actually copied
......
......@@ -152,7 +152,7 @@ static const FormatInfo formats[] = {
{"video/sp5x", "Sunplus JPEG 5.x", 0},
{"video/vivo", "Vivo", 0},
{"video/x-3ivx", "3ivx", 0},
{"video/x-4xm", "4X Techologies Video", 0},
{"video/x-4xm", "4X Technologies Video", 0},
{"video/x-apple-video", "Apple video", 0},
{"video/x-aasc", "Autodesk Animator", 0},
{"video/x-camtasia", "TechSmith Camtasia", 0},
......
......@@ -531,7 +531,7 @@ gst_encoding_video_profile_set_pass (GstEncodingVideoProfile * prof, guint pass)
* @prof: a #GstEncodingVideoProfile
* @variableframerate: a boolean
*
* If set to %TRUE, then the incoming streamm will be allowed to have non-constant
* If set to %TRUE, then the incoming stream will be allowed to have non-constant
* framerate. If set to %FALSE (default value), then the incoming stream will
* be normalized by dropping/duplicating frames in order to produce a
* constance framerate.
......
......@@ -36,7 +36,7 @@ G_BEGIN_DECLS
* GST_ENCODING_CATEGORY_DEVICE:
*
* #GstEncodingTarget category for device-specific targets.
* The name of the target will usually be the contructor and model of the device,
* The name of the target will usually be the constructor and model of the device,
* and that target will contain #GstEncodingProfiles suitable for that device.
*/
#define GST_ENCODING_CATEGORY_DEVICE "device"
......
......@@ -1018,7 +1018,7 @@ DISCOVERER_INFO_ACCESSOR_CODE (duration, GstClockTime, GST_CLOCK_TIME_NONE);
* gst_discoverer_info_get_seekable:
* @info: a #GstDiscovererInfo
*
* Returns: the wheter the URI is seekable.
* Returns: the whether the URI is seekable.
*
* Since: 0.10.32
*/
......
......@@ -1467,7 +1467,7 @@ gst_discoverer_stop (GstDiscoverer * discoverer)
* A copy of @uri will be made internally, so the caller can safely g_free()
* afterwards.
*
* Returns: %TRUE if the @uri was succesfully appended to the list of pending
* Returns: %TRUE if the @uri was successfully appended to the list of pending
* uris, else %FALSE
*
* Since: 0.10.31
......
......@@ -839,7 +839,7 @@ gst_base_rtp_audio_payload_handle_buffer (GstBaseRTPPayload *
GstClockTime diff;
guint64 bytes;
/* we're only going to apply a positive gap, otherwise we let the marker
* bit do its thing. simply convert to bytes and add the the current
* bit do its thing. simply convert to bytes and add the current
* offset */
diff = timestamp - priv->last_timestamp;
bytes = priv->time_to_bytes (payload, diff);
......
......@@ -599,7 +599,7 @@ gst_rtcp_packet_get_length (GstRTCPPacket * packet)
* @ntptime: result NTP time
* @rtptime: result RTP time
* @packet_count: result packet count
* @octet_count: result octect count
* @octet_count: result octet count
*
* Parse the SR sender info and store the values.
*/
......@@ -641,7 +641,7 @@ gst_rtcp_packet_sr_get_sender_info (GstRTCPPacket * packet, guint32 * ssrc,
* @ntptime: the NTP time
* @rtptime: the RTP time
* @packet_count: the packet count
* @octet_count: the octect count
* @octet_count: the octet count
*
* Set the given values in the SR packet @packet.
*/
......@@ -1137,7 +1137,7 @@ gst_rtcp_packet_sdes_next_entry (GstRTCPPacket * packet)
*
* When @type refers to a text item, @data will point to a UTF8 string. Note
* that this UTF8 string is NOT null-terminated. Use
* gst_rtcp_packet_sdes_copy_entry() to get a null-termined copy of the entry.
* gst_rtcp_packet_sdes_copy_entry() to get a null-terminated copy of the entry.
*
* Returns: %TRUE if there was valid data.
*/
......
......@@ -323,7 +323,7 @@ validate_data (guint8 * data, guint len, guint8 * payload, guint payload_len)
guint8 *extpos;
guint16 extlen;
/* this points to the extenstion bits and header length */
/* this points to the extension bits and header length */
extpos = &data[header_len];
/* skip the header and check that we have enough space */
......
......@@ -1907,7 +1907,7 @@ build_next (GstRTSPBuilder * builder, GstRTSPMessage * message,
goto done;
/* we have the complete body now, store in the message adjusting the
* length to include the traling '\0' */
* length to include the trailing '\0' */
gst_rtsp_message_take_body (message,
(guint8 *) builder->body_data, builder->body_len + 1);
builder->body_data = NULL;
......
......@@ -263,7 +263,7 @@ gst_rtsp_range_to_string (const GstRTSPTimeRange * range)
* gst_rtsp_range_free:
* @range: a #GstRTSPTimeRange
*
* Free the memory alocated by @range.
* Free the memory allocated by @range.
*/
void
gst_rtsp_range_free (GstRTSPTimeRange * range)
......
......@@ -1523,7 +1523,7 @@ write_exif_ifd (const GstTagList * taglist, gboolean byte_order,
else
gst_byte_writer_put_uint16_be (&writer.tagwriter, writer.tags_total);
GST_DEBUG ("Number of tags rewriten to %d", writer.tags_total);
GST_DEBUG ("Number of tags rewritten to %d", writer.tags_total);
/* now that we know the tag headers size, we can add the offsets */
gst_exif_tag_rewrite_offsets (&writer.tagwriter, writer.byte_order,
......@@ -2000,7 +2000,7 @@ deserialize_geo_coordinate (GstExifReader * exif_reader,
}
if (exiftag->exif_tag != next_tagdata.tag) {
GST_WARNING ("This is not a geo cordinate tag");
GST_WARNING ("This is not a geo coordinate tag");
return ret;
}
......
......@@ -604,7 +604,7 @@ gst_tag_to_metadata_block_picture (const gchar * tag,
* Creates a new tag list that contains the information parsed out of a
* vorbiscomment packet.
*
* Returns: A #GList of newly-allowcated key=value strings. Free with
* Returns: A #GList of newly-allocated key=value strings. Free with
* g_list_foreach (list, (GFunc) g_free, NULL) plus g_list_free (list)
*/
GList *
......
......@@ -1403,7 +1403,7 @@ gst_tag_list_from_xmp_buffer (const GstBuffer * buffer)
}
} else {
XmpTag *xmp_tag = NULL;
/* FIXME: eventualy rewrite ns
/* FIXME: eventually rewrite ns
* find ':'
* check if ns before ':' is in ns_map and ns_map[i].gstreamer_ns!=NULL
* do 2 stage filter in tag_matches
......@@ -1459,7 +1459,7 @@ gst_tag_list_from_xmp_buffer (const GstBuffer * buffer)
<dc:type><rdf:Bag><rdf:li>Image</rdf:li></rdf:Bag></dc:type>
<dc:creator><rdf:Seq><rdf:li/></rdf:Seq></dc:creator>
*/
/* FIXME: eventualy rewrite ns */
/* FIXME: eventually rewrite ns */
/* skip rdf tags for now */
if (strncmp (part, "rdf:", 4)) {
......@@ -1840,7 +1840,7 @@ gst_tag_list_to_xmp_buffer_full (const GstTagList * list, gboolean read_only,
g_string_append (data, "</x:xmpmeta>\n");
if (!read_only) {
/* the xmp spec recommand to add 2-4KB padding for in-place editable xmp */
/* the xmp spec recommends to add 2-4KB padding for in-place editable xmp */
guint i;
for (i = 0; i < 32; i++) {
......
......@@ -183,7 +183,7 @@ bits are ignored, so a 257 bytes long tag is represented as $00 00 02 01.
The ID3v2 tag size is the size of the complete tag after unsychronisation,
including padding, excluding the header but not excluding the extended header
(total tag size - 10). Only 28 bits (representing up to 256MB) are used in the
size description to avoid the introducuction of 'false syncsignals'.
size description to avoid the introduction of 'false syncsignals'.
An ID3v2 tag can be detected with the following pattern:
$49 44 33 yy yy xx zz zz zz zz
......@@ -1006,7 +1006,7 @@ Where time stamp format is:
$01 Absolute time, 32 bit sized, using MPEG frames as unit
$02 Absolute time, 32 bit sized, using milliseconds as unit
Abolute time means that every stamp contains the time from the beginning of the
Absolute time means that every stamp contains the time from the beginning of the
file.
Followed by a list of key events in the following format:
......@@ -1111,7 +1111,7 @@ Where time stamp format is:
$01 Absolute time, 32 bit sized, using MPEG frames as unit
$02 Absolute time, 32 bit sized, using milliseconds as unit
Abolute time means that every stamp contains the time from the beginning of the
Absolute time means that every stamp contains the time from the beginning of the
file.
4.9. Unsychronised lyrics/text transcription
......@@ -1167,7 +1167,7 @@ Time stamp format is:
$01 Absolute time, 32 bit sized, using MPEG frames as unit
$02 Absolute time, 32 bit sized, using milliseconds as unit
Abolute time means that every stamp contains the time from the beginning of the
Absolute time means that every stamp contains the time from the beginning of the
file.