diff options
Diffstat (limited to 'gst-libs/gst')
32 files changed, 68 insertions, 68 deletions
diff --git a/gst-libs/gst/app/gstappsink.c b/gst-libs/gst/app/gstappsink.c index fed4dd823..a1a20a5b0 100644 --- a/gst-libs/gst/app/gstappsink.c +++ b/gst-libs/gst/app/gstappsink.c @@ -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 diff --git a/gst-libs/gst/app/gstappsrc.c b/gst-libs/gst/app/gstappsrc.c index 18e357300..6ae59e593 100644 --- a/gst-libs/gst/app/gstappsrc.c +++ b/gst-libs/gst/app/gstappsrc.c @@ -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 diff --git a/gst-libs/gst/app/gstappsrc.h b/gst-libs/gst/app/gstappsrc.h index 041cb6821..d452ebfae 100644 --- a/gst-libs/gst/app/gstappsrc.h +++ b/gst-libs/gst/app/gstappsrc.h @@ -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. diff --git a/gst-libs/gst/audio/audio.c b/gst-libs/gst/audio/audio.c index 33ab3963b..d2d15c394 100644 --- a/gst-libs/gst/audio/audio.c +++ b/gst-libs/gst/audio/audio.c @@ -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. diff --git a/gst-libs/gst/audio/gstaudioencoder.c b/gst-libs/gst/audio/gstaudioencoder.c index fee5d2221..55f8d8381 100644 --- a/gst-libs/gst/audio/gstaudioencoder.c +++ b/gst-libs/gst/audio/gstaudioencoder.c @@ -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. * diff --git a/gst-libs/gst/audio/gstbaseaudiosink.c b/gst-libs/gst/audio/gstbaseaudiosink.c index 54b092219..a653ebf31 100644 --- a/gst-libs/gst/audio/gstbaseaudiosink.c +++ b/gst-libs/gst/audio/gstbaseaudiosink.c @@ -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; } diff --git a/gst-libs/gst/audio/gstbaseaudiosrc.c b/gst-libs/gst/audio/gstbaseaudiosrc.c index 7718747c5..a6d142b43 100644 --- a/gst-libs/gst/audio/gstbaseaudiosrc.c +++ b/gst-libs/gst/audio/gstbaseaudiosrc.c @@ -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; } diff --git a/gst-libs/gst/audio/gstringbuffer.c b/gst-libs/gst/audio/gstringbuffer.c index 87df45b1c..ab1880c68 100644 --- a/gst-libs/gst/audio/gstringbuffer.c +++ b/gst-libs/gst/audio/gstringbuffer.c @@ -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. * diff --git a/gst-libs/gst/audio/multichannel.h b/gst-libs/gst/audio/multichannel.h index ffd29ae2b..8bf92d7d4 100644 --- a/gst-libs/gst/audio/multichannel.h +++ b/gst-libs/gst/audio/multichannel.h @@ -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 diff --git a/gst-libs/gst/fft/gstfftf32.c b/gst-libs/gst/fft/gstfftf32.c index fd574e0b1..4f78080b1 100644 --- a/gst-libs/gst/fft/gstfftf32.c +++ b/gst-libs/gst/fft/gstfftf32.c @@ -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. * diff --git a/gst-libs/gst/fft/gstfftf64.c b/gst-libs/gst/fft/gstfftf64.c index e737854be..cdf7d4c5a 100644 --- a/gst-libs/gst/fft/gstfftf64.c +++ b/gst-libs/gst/fft/gstfftf64.c @@ -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. * diff --git a/gst-libs/gst/fft/gstffts16.c b/gst-libs/gst/fft/gstffts16.c index 212e93f1d..a204aaa65 100644 --- a/gst-libs/gst/fft/gstffts16.c +++ b/gst-libs/gst/fft/gstffts16.c @@ -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. * diff --git a/gst-libs/gst/fft/gstffts32.c b/gst-libs/gst/fft/gstffts32.c index 56ea543b8..6fc864b0a 100644 --- a/gst-libs/gst/fft/gstffts32.c +++ b/gst-libs/gst/fft/gstffts32.c @@ -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. * diff --git a/gst-libs/gst/interfaces/navigation.c b/gst-libs/gst/interfaces/navigation.c index 2df4a7baf..c8edbe874 100644 --- a/gst-libs/gst/interfaces/navigation.c +++ b/gst-libs/gst/interfaces/navigation.c @@ -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> diff --git a/gst-libs/gst/interfaces/xoverlay.c b/gst-libs/gst/interfaces/xoverlay.c index 8e7ef07a8..cf6a6bbd2 100644 --- a/gst-libs/gst/interfaces/xoverlay.c +++ b/gst-libs/gst/interfaces/xoverlay.c @@ -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. diff --git a/gst-libs/gst/netbuffer/gstnetbuffer.c b/gst-libs/gst/netbuffer/gstnetbuffer.c index 6328a7690..9998dc85d 100644 --- a/gst-libs/gst/netbuffer/gstnetbuffer.c +++ b/gst-libs/gst/netbuffer/gstnetbuffer.c @@ -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 diff --git a/gst-libs/gst/pbutils/descriptions.c b/gst-libs/gst/pbutils/descriptions.c index 5e7de0377..63f5aa6c5 100644 --- a/gst-libs/gst/pbutils/descriptions.c +++ b/gst-libs/gst/pbutils/descriptions.c @@ -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}, diff --git a/gst-libs/gst/pbutils/encoding-profile.c b/gst-libs/gst/pbutils/encoding-profile.c index 65af1b13f..e5e7a7ec7 100644 --- a/gst-libs/gst/pbutils/encoding-profile.c +++ b/gst-libs/gst/pbutils/encoding-profile.c @@ -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. diff --git a/gst-libs/gst/pbutils/encoding-target.h b/gst-libs/gst/pbutils/encoding-target.h index 70c049db3..48d3d8089 100644 --- a/gst-libs/gst/pbutils/encoding-target.h +++ b/gst-libs/gst/pbutils/encoding-target.h @@ -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" diff --git a/gst-libs/gst/pbutils/gstdiscoverer-types.c b/gst-libs/gst/pbutils/gstdiscoverer-types.c index ee357bab0..0d48a3d41 100644 --- a/gst-libs/gst/pbutils/gstdiscoverer-types.c +++ b/gst-libs/gst/pbutils/gstdiscoverer-types.c @@ -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 */ diff --git a/gst-libs/gst/pbutils/gstdiscoverer.c b/gst-libs/gst/pbutils/gstdiscoverer.c index bd5ad95b2..bce956edb 100644 --- a/gst-libs/gst/pbutils/gstdiscoverer.c +++ b/gst-libs/gst/pbutils/gstdiscoverer.c @@ -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 diff --git a/gst-libs/gst/rtp/gstbasertpaudiopayload.c b/gst-libs/gst/rtp/gstbasertpaudiopayload.c index d1a43a90c..c8d0b57f5 100644 --- a/gst-libs/gst/rtp/gstbasertpaudiopayload.c +++ b/gst-libs/gst/rtp/gstbasertpaudiopayload.c @@ -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); diff --git a/gst-libs/gst/rtp/gstrtcpbuffer.c b/gst-libs/gst/rtp/gstrtcpbuffer.c index 3b37c6fc7..fbd928c92 100644 --- a/gst-libs/gst/rtp/gstrtcpbuffer.c +++ b/gst-libs/gst/rtp/gstrtcpbuffer.c @@ -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. */ diff --git a/gst-libs/gst/rtp/gstrtpbuffer.c b/gst-libs/gst/rtp/gstrtpbuffer.c index d99f6462e..cb63827b2 100644 --- a/gst-libs/gst/rtp/gstrtpbuffer.c +++ b/gst-libs/gst/rtp/gstrtpbuffer.c @@ -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 */ diff --git a/gst-libs/gst/rtsp/gstrtspconnection.c b/gst-libs/gst/rtsp/gstrtspconnection.c index 2de21b977..da39c21cd 100644 --- a/gst-libs/gst/rtsp/gstrtspconnection.c +++ b/gst-libs/gst/rtsp/gstrtspconnection.c @@ -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; diff --git a/gst-libs/gst/rtsp/gstrtsprange.c b/gst-libs/gst/rtsp/gstrtsprange.c index 0ad75c8ef..39593ec79 100644 --- a/gst-libs/gst/rtsp/gstrtsprange.c +++ b/gst-libs/gst/rtsp/gstrtsprange.c @@ -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) diff --git a/gst-libs/gst/tag/gstexiftag.c b/gst-libs/gst/tag/gstexiftag.c index 3e5e53b0b..448943dc1 100644 --- a/gst-libs/gst/tag/gstexiftag.c +++ b/gst-libs/gst/tag/gstexiftag.c @@ -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; } diff --git a/gst-libs/gst/tag/gstvorbistag.c b/gst-libs/gst/tag/gstvorbistag.c index 8fb2f8553..1c3d554bb 100644 --- a/gst-libs/gst/tag/gstvorbistag.c +++ b/gst-libs/gst/tag/gstvorbistag.c @@ -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 * diff --git a/gst-libs/gst/tag/gstxmptag.c b/gst-libs/gst/tag/gstxmptag.c index 6ae5d995b..abe359724 100644 --- a/gst-libs/gst/tag/gstxmptag.c +++ b/gst-libs/gst/tag/gstxmptag.c @@ -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++) { diff --git a/gst-libs/gst/tag/id3v2.3.0.txt b/gst-libs/gst/tag/id3v2.3.0.txt index 5b26d638e..5b57850b7 100644 --- a/gst-libs/gst/tag/id3v2.3.0.txt +++ b/gst-libs/gst/tag/id3v2.3.0.txt @@ -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. The text that follows the frame header differs from that of the unsynchronised lyrics/text transcription in one major way. Each syllable (or whatever size of @@ -1463,7 +1463,7 @@ frame in each tag. 4.20. Audio encryption This frame indicates if the actual audio stream is encrypted, and by whom. -Since standardisation of such encrypion scheme is beyond this document, all +Since standardisation of such encryption scheme is beyond this document, all "AENC" frames begin with a terminated string with a URL containing an email address, or a link to a location where an email address can be found, that belongs to the organisation responsible for this specific encrypted audio file. diff --git a/gst-libs/gst/tag/id3v2.4.0-frames.txt b/gst-libs/gst/tag/id3v2.4.0-frames.txt index 74a21bed3..d27b5166b 100644 --- a/gst-libs/gst/tag/id3v2.4.0-frames.txt +++ b/gst-libs/gst/tag/id3v2.4.0-frames.txt @@ -255,7 +255,7 @@ Abstract one text information frame of its kind in an tag. All text information frames supports multiple strings, stored as a null separated list, where null is reperesented by the termination code - for the charater encoding. All text frame identifiers begin with "T". + for the character encoding. All text frame identifiers begin with "T". Only text frame identifiers begin with "T", with the exception of the "TXXX" frame. All the text information frames have the following format: diff --git a/gst-libs/gst/tag/id3v2.4.0-structure.txt b/gst-libs/gst/tag/id3v2.4.0-structure.txt index 5fa156a0a..5d3a6145c 100644 --- a/gst-libs/gst/tag/id3v2.4.0-structure.txt +++ b/gst-libs/gst/tag/id3v2.4.0-structure.txt @@ -411,7 +411,7 @@ Abstract byte indicates that extra information is added to the header. These fields of extra information is ordered as the flags that indicates them. The flags field is defined as follows (l and o left out because - ther resemblence to one and zero): + their resemblence to one and zero): %0abc0000 %0h00kmnp |