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doc: fix spelling errors
Thanks to Mathieu Malaterre <malat@debian.org> for reporting the Que/Queue typo. (https://bugs.debian.org/839542) Reviewed-by: Lou Logan <lou@lrcd.com> Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
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@ -1637,7 +1637,7 @@ EXTRA_PACKAGES =
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# following commands have a special meaning inside the header: $title,
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# $datetime, $date, $doxygenversion, $projectname, $projectnumber,
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# $projectbrief, $projectlogo. Doxygen will replace $title with the empy string,
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# for the replacement values of the other commands the user is refered to
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# for the replacement values of the other commands the user is referred to
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# HTML_HEADER.
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# This tag requires that the tag GENERATE_LATEX is set to YES.
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@ -1773,7 +1773,7 @@ Enable CAVLC and disable CABAC. It generates the same effect as
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@end table
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@item cmp
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Set full pixel motion estimation comparation algorithm. Possible values:
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Set full pixel motion estimation comparison algorithm. Possible values:
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@table @samp
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@item chroma
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@ -245,7 +245,7 @@ continue reading from that.
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Each interval is specified by two optional parts, separated by "%".
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The first part specifies the interval start position. It is
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interpreted as an abolute position, or as a relative offset from the
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interpreted as an absolute position, or as a relative offset from the
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current position if it is preceded by the "+" character. If this first
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part is not specified, no seeking will be performed when reading this
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interval.
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@ -449,7 +449,7 @@ This filter is bit crusher with enhanced functionality. A bit crusher
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is used to audibly reduce number of bits an audio signal is sampled
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with. This doesn't change the bit depth at all, it just produces the
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effect. Material reduced in bit depth sounds more harsh and "digital".
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This filter is able to even round to continous values instead of discrete
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This filter is able to even round to continuous values instead of discrete
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bit depths.
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Additionally it has a D/C offset which results in different crushing of
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the lower and the upper half of the signal.
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@ -475,7 +475,7 @@ Set level out.
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Set bit reduction.
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@item mix
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Set mixing ammount.
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Set mixing amount.
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@item mode
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Can be linear: @code{lin} or logarithmic: @code{log}.
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@ -1203,7 +1203,7 @@ Set video stream size. Only useful if curves option is activated.
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@item mgain
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Set max gain that will be displayed. Only useful if curves option is activated.
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Setting this to reasonable value allows to display gain which is derived from
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Setting this to a reasonable value makes it possible to display gain which is derived from
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neighbour bands which are too close to each other and thus produce higher gain
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when both are activated.
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@ -8858,7 +8858,7 @@ value.
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@section hysteresis
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Grow first stream into second stream by connecting components.
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This allows to build more robust edge masks.
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This makes it possible to build more robust edge masks.
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This filter accepts the following options:
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@ -17670,7 +17670,7 @@ magnitude across time and second represents phase across time.
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The filter will transform from frequency domain as displayed in videos back
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to time domain as presented in audio output.
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This filter is primarly created for reversing processed @ref{showspectrum}
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This filter is primarily created for reversing processed @ref{showspectrum}
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filter outputs, but can synthesize sound from other spectrograms too.
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But in such case results are going to be poor if the phase data is not
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available, because in such cases phase data need to be recreated, usually
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@ -1513,14 +1513,14 @@ as a list of @var{key}=@var{value} pairs separated by ':'.
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@item drop_pkts_on_overflow @var{bool}
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If set to 1 (true), in case the fifo queue fills up, packets will be dropped
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rather than blocking the encoder. This allows to continue streaming without
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delaying the input, at the cost of ommiting part of the stream. By default
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rather than blocking the encoder. This makes it possible to continue streaming without
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delaying the input, at the cost of omitting part of the stream. By default
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this option is set to 0 (false), so in such cases the encoder will be blocked
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until the muxer processes some of the packets and none of them is lost.
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@item attempt_recovery @var{bool}
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If failure occurs, attempt to recover the output. This is especially useful
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when used with network output, allows to restart streaming transparently.
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when used with network output, since it makes it possible to restart streaming transparently.
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By default this option is set to 0 (false).
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@item max_recovery_attempts
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2
ffmpeg.c
2
ffmpeg.c
@ -3771,7 +3771,7 @@ static int check_keyboard_interaction(int64_t cur_time)
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"+ increase verbosity\n"
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"- decrease verbosity\n"
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"c Send command to first matching filter supporting it\n"
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"C Send/Que command to all matching filters\n"
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"C Send/Queue command to all matching filters\n"
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"D cycle through available debug modes\n"
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"h dump packets/hex press to cycle through the 3 states\n"
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"q quit\n"
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@ -129,7 +129,7 @@ int cuvid_transcode_init(OutputStream *ost)
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ist->hwaccel_uninit = cuvid_uninit;
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/* This is a bit hacky, av_hwframe_ctx_init is called by the cuvid decoder
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* once it has probed the neccesary format information. But as filters/nvenc
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* once it has probed the necessary format information. But as filters/nvenc
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* need to know the format/sw_format, set them here so they are happy.
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* This is fine as long as CUVID doesn't add another supported pix_fmt.
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*/
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@ -147,7 +147,7 @@ error:
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cancel:
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if (ist->hwaccel_id == HWACCEL_CUVID) {
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av_log(NULL, AV_LOG_ERROR, "CUVID hwaccel requested, but impossible to achive.\n");
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av_log(NULL, AV_LOG_ERROR, "CUVID hwaccel requested, but impossible to achieve.\n");
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return AVERROR(EINVAL);
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}
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@ -87,7 +87,7 @@ static void search_for_quantizers_twoloop(AVCodecContext *avctx,
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* will keep iterating until it fails to lower it or it reaches
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* ulimit * rdlambda. Keeping it low increases quality on difficult
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* signals, but lower it too much, and bits will be taken from weak
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* signals, creating "holes". A balance is necesary.
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* signals, creating "holes". A balance is necessary.
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* rdmax and rdmin specify the relative deviation from rdlambda
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* allowed for tonality compensation
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*/
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@ -182,7 +182,7 @@ int ff_init_cabac_decoder(CABACContext *c, const uint8_t *buf, int buf_size){
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#if CABAC_BITS == 16
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c->low = (*c->bytestream++)<<18;
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c->low+= (*c->bytestream++)<<10;
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// Keep our fetches on a 2-byte boundry as this should avoid ever having to
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// Keep our fetches on a 2-byte boundary as this should avoid ever having to
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// do unaligned loads if the compiler (or asm) optimises the double byte
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// load into a single instruction
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if(((uintptr_t)c->bytestream & 1) == 0) {
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@ -224,7 +224,7 @@ int ff_jni_exception_get_summary(JNIEnv *env, jthrowable exception, char **error
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} else if (!name && message) {
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av_bprintf(&bp, "Exception: %s", message);
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} else {
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av_log(log_ctx, AV_LOG_WARNING, "Could not retreive exception name and message\n");
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av_log(log_ctx, AV_LOG_WARNING, "Could not retrieve exception name and message\n");
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av_bprintf(&bp, "Exception occurred");
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}
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@ -44,7 +44,7 @@
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* implementation.
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*
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* The API around MediaCodecList is not part of the NDK (and is lacking as
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* we still need to retreive the codec name to work around faulty decoders
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* we still need to retrieve the codec name to work around faulty decoders
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* and encoders).
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*
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* For documentation, please refers to NdkMediaCodec.h NdkMediaFormat.h and
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@ -29,7 +29,7 @@
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/** maximum number of channels */
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#define PSY_MAX_CHANS 20
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/* cutoff for VBR is purposedly increased, since LP filtering actually
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/* cutoff for VBR is purposely increased, since LP filtering actually
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* hinders VBR performance rather than the opposite
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*/
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#define AAC_CUTOFF_FROM_BITRATE(bit_rate,channels,sample_rate) (bit_rate ? FFMIN3(FFMIN3( \
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@ -78,7 +78,7 @@ SECTION .text
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%endif
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%endmacro
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; calulate p or q portion of flat8out
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; calculate p or q portion of flat8out
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%macro FLAT8OUT_HALF 0
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psubw m4, m0 ; q4-q0
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psubw m5, m0 ; q5-q0
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@ -867,7 +867,7 @@ typedef struct {
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* steps of 0.5, but no value below -6.0 dB should appear. */
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int gain_counts[16];
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int max_gain;
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/** occurences of code detect timer expiring without detecting
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/** occurrences of code detect timer expiring without detecting
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* a code. -1 for timer never set. */
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int count_sustain_expired;
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int loglevel; ///< log level for frame logging
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int metadata; ///< whether or not to inject loudness results in frames
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int dual_mono; ///< whether or not to treat single channel input files as dual-mono
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double pan_law; ///< pan law value used to calulate dual-mono measurements
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double pan_law; ///< pan law value used to calculate dual-mono measurements
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} EBUR128Context;
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enum {
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@ -650,7 +650,7 @@ int ff_bprint_to_codecpar_extradata(AVCodecParameters *par, struct AVBPrint *buf
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* The packet is not removed from the interleaving queue, but only
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* a pointer to it is returned.
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*
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* @param ts_offset the ts difference between packet in the que and the muxer.
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* @param ts_offset the ts difference between packet in the queue and the muxer.
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*
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* @return a pointer to the next packet, or NULL if no packet is queued
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* for this stream.
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@ -178,7 +178,7 @@ typedef struct AVFrameSideData {
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* without breaking compatibility with each other.
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*
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* Fields can be accessed through AVOptions, the name string used, matches the
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* C structure field name for fields accessable through AVOptions. The AVClass
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* C structure field name for fields accessible through AVOptions. The AVClass
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* for AVFrame can be obtained from avcodec_get_frame_class()
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*/
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typedef struct AVFrame {
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* then the corresponding entry in next is unchanged.
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* @param cmp compare function used to compare elements in the tree,
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* API identical to that of Standard C's qsort
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* It is guranteed that the first and only the first argument to cmp()
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* It is guaranteed that the first and only the first argument to cmp()
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* will be the key parameter to av_tree_find(), thus it could if the
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* user wants, be a different type (like an opaque context).
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* @return An element with cmp(key, elem) == 0 or NULL if no such element
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