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v2017-07-1
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v2017-08-0
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@@ -4,6 +4,7 @@ h1, h2, h3, h4, h5, h6 {
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code {
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word-break: break-all;
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word-wrap: break-word;
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}
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dd {
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90
index.html
90
index.html
@@ -188,7 +188,7 @@
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<div class="well">
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<h3>Create FFV1 Version 3 video in a Matroska container with framemd5 of input</h3>
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<p><code>ffmpeg -i <i>input_file</i> -map 0 -dn -c:v ffv1 -level 3 -g 1 -slicecrc 1 -slices 16 -c:a copy <i>output_file</i>.mkv -f framemd5 -an <i>md5_output_file</i></code></p>
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<p>This will losslessly trancode your video with the FFV1 Version 3 codec in a Matroska container. In order to verify losslessness, a framemd5 of the source video is also generated. For more information on FFV1 encoding, <a href="https://trac.ffmpeg.org/wiki/Encode/FFV1" target="_blank">try the ffmpeg wiki</a>.</p>
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<p>This will losslessly transcode your video with the FFV1 Version 3 codec in a Matroska container. In order to verify losslessness, a framemd5 of the source video is also generated. For more information on FFV1 encoding, <a href="https://trac.ffmpeg.org/wiki/Encode/FFV1" target="_blank">try the ffmpeg wiki</a>.</p>
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<dl>
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<dt>ffmpeg</dt><dd>starts the command.</dd>
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<dt>-i <i>input_file</i></dt><dd>path, name and extension of the input file.</dd>
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@@ -668,7 +668,7 @@
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<dt>movie='<i>input.mp3</i>'</dt><dd>declares audio source file on which to apply filter</dd>
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<dt>,</dt><dd>comma signifies the end of audio source section and the beginning of the filter section</dd>
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<dt>astats=metadata=1</dt><dd>tells the astats filter to ouput metadata that can be passed to another filter (in this case adrawgraph)</dd>
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<dt>:</dt><dd>divides beteen options of the same filter</dd>
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<dt>:</dt><dd>divides between options of the same filter</dd>
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<dt>reset=1</dt><dd>tells the filter to calculate the stats on every frame (increasing this number would calculate stats for groups of frames)</dd>
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<dt>,</dt><dd>comma divides one filter in the chain from another</dd>
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<dt>adrawgraph=lavfi.astats.Overall.Peak_level:max=0:min=-30.0</dt><dd>draws a graph using the overall peak volume calculated by the astats filter. It sets the max for the graph to 0 (dB) and the minimum to -30 (dB). For more options on data points that can be graphed see the <a href="https://ffmpeg.org/ffmpeg-filters.html#astats-1" target="_blank">ffmpeg astats documentation</a></dd>
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@@ -1083,13 +1083,13 @@
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<div class="well">
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<h3>Generate two access MP3s from input. One with appended audio (such as copyright notice) and one unmodified.</h3>
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<p> <code>ffmpeg -i <i>input_file</i> -i <i>input_file_to_append</i> -filter_complex "[0:a:0]asplit=2[a][b];[b]afifo[bb];[1:a:0][bb]concat=n=2:v=0:a=1[concatout]" -map "[a]" -codec:a libmp3lame -dither_method modified_e_weighted -qscale:a 2 <i>output_file.mp3</i> -map "[concatout]" -codec:a libmp3lame -dither_method modified_e_weighted -qscale:a 2 <i>output_file_appended.mp3</i></code></p>
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<p>This script allows you to generate two derivative audio files from a master while appending audio from a seperate file (for example a copyright or institutional notice) to one of them.</p>
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<p>This script allows you to generate two derivative audio files from a master while appending audio from a separate file (for example a copyright or institutional notice) to one of them.</p>
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<dl>
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<dt>ffmpeg</dt><dd>starts the command</dd>
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<dt>-i <i>input_file</i></dt><dd>path, name and extension of the input file (the master file)</dd>
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<dt>-i <i>input_file_to_append</i></dt><dd>path, name and extension of the input file (the file to be appended to access file)</dd>
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<dt>-filter_complex</dt><dd>enables the complex filtering to manage splitting the input to two audio streams</dd>
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<dt>[0:a:0]asplit=2[a][b];</dt><dd><code>asplit</code> allows audio streams to be split up for seperate manipulation. This command splits the audio from the first input (the master file) into two streams "a" and "b"</dd>
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<dt>[0:a:0]asplit=2[a][b];</dt><dd><code>asplit</code> allows audio streams to be split up for separate manipulation. This command splits the audio from the first input (the master file) into two streams "a" and "b"</dd>
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<dt>[b]afifo[bb];</dt><dd>this buffers the stream "b" to help prevent dropped samples and renames stream to "bb"</dd>
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<dt>[1:a:0][bb]concat=n=2:v=0:a=1[concatout]</dt><dd><code>concat</code> is used to join files. <code>n=2</code> tells the filter there are two inputs. <code>v=0:a=1</code> Tells the filter there are 0 video outputs and 1 audio output. This command appends the audio from the second input to the beginning of stream "bb" and names the output "concatout"</dd>
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<dt>-map "[a]"</dt><dd>this maps the unmodified audio stream to the first output</dd>
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@@ -1110,6 +1110,30 @@
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<div class="well">
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<h4>Normalize/Equalize Audio</h4>
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<!-- phase shift -->
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<span data-toggle="modal" data-target="#phase_shift"><button type="button" class="btn btn-default" data-toggle="tooltip" data-placement="bottom" title="Inverses the audio phase of the second channel">Flip phase shift</button></span>
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<div id="phase_shift" class="modal fade" tabindex="-1" role="dialog">
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<div class="modal-dialog modal-lg">
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<div class="modal-content">
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<div class="well">
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<h3>Flip audio phase shift</h3>
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<p><code>ffmpeg -i <i>input_file</i> -af pan="stereo|c0=c0|c1=-1*c1" <i>output_file</i></code></p>
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<p>This command inverses the audio phase of the second channel by rotating it 180°.</p>
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<dl>
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<dt>ffmpeg</dt><dd>starts the command</dd>
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<dt>-i <i>input file</i></dt><dd>path, name and extension of the input file</dd>
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<dt>-af</dt><dd>specifies that the next section should be interpreted as an audio filter</dd>
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<dt>pan=</dt><dd>tell the quoted text below to use the <a href="https://ffmpeg.org/ffmpeg-filters.html#pan-1" target="_blank">pan filter</a></dd>
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<dt>"stereo|c0=c0|c1=-1*c1"</dt><dd>maps the output's first channel (c0) to the input's first channel and the output's second channel (c1) to the inverse of the input's second channel</dd>
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<dt><i>output file</i></dt><dd>path, name and extension of the output file</dd>
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</dl>
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<p class="link"></p>
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</div>
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</div>
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</div>
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</div>
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<!-- ends phase shift -->
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<!-- loudnorm metadata -->
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<span data-toggle="modal" data-target="#loudnorm_metadata"><button type="button" class="btn btn-default" data-toggle="tooltip" data-placement="bottom" title="Calculate Loudness Levels">Calculate Loudness Levels</button></span>
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<div id="loudnorm_metadata" class="modal fade" tabindex="-1" role="dialog">
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@@ -1119,8 +1143,8 @@
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<h3>Calculate Loudness Levels</h3>
|
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<p><code>ffmpeg -i <i>input_file</i> -af loudnorm=print_format=json -f null -</code></p>
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<p>This filter calculates and outputs loudness information in json about an input file (labeled input) as well as what the levels would be if loudnorm were applied in its one pass mode (labeled output). The values generated can be used as inputs for a 'second pass' of the loudnorm filter allowing more accurate loudness normalization than if it is used in a single pass.</p>
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<p>These instructions use the loudnorm defaults, which align well with PBS recommendations for target loudness. More information can be found at the <a href=https://ffmpeg.org/ffmpeg-filters.html#loudnorm>loudnorm documentation</a>.</p>
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<p>Information about PBS loudness standards can be found in the <a href=http://www-tc.pbs.org/capt/Producing/TOS-2012-Pt2-Distribution.pdf>PBS Technical Operating Specifications</a> document. Information about EBU loudness standards can be found in the <a href=https://tech.ebu.ch/docs/r/r128-2014.pdf>EBU R 128</a> recommendation document.</p>
|
||||
<p>These instructions use the loudnorm defaults, which align well with PBS recommendations for target loudness. More information can be found at the <a href="https://ffmpeg.org/ffmpeg-filters.html#loudnorm">loudnorm documentation</a>.</p>
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<p>Information about PBS loudness standards can be found in the <a href="http://www-tc.pbs.org/capt/Producing/TOS-2012-Pt2-Distribution.pdf" target="_blank">PBS Technical Operating Specifications</a> document. Information about EBU loudness standards can be found in the <a href="https://tech.ebu.ch/docs/r/r128-2014.pdf" target="_blank">EBU R 128</a> recommendation document.</p>
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<dl>
|
||||
<dt>ffmpeg</dt><dd>starts the command</dd>
|
||||
<dt><i>input_file</i></dt><dd>path, name and extension of the input file</dd>
|
||||
@@ -1142,14 +1166,15 @@
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||||
<div class="modal-content">
|
||||
<div class="well">
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<h3>One Pass Loudness Normalization</h3>
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<p><code>ffmpeg -i <i>input_file</i> -af loudnorm=dual_mono=true <i>output_file</i></code></p>
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<p>This will normalize the loudness of an input using one pass, which is quicker but less accurate than using two passes. This command uses the loudnorm filter defaults for target loudness. These defaults align well with PBS recommendations, but loudnorm does allow targeting of specific loudness levels. More information can be found at the <a href=https://ffmpeg.org/ffmpeg-filters.html#loudnorm>loudnorm documentation</a>.</p>
|
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<p>Information about PBS loudness standards can be found in the <a href=http://www-tc.pbs.org/capt/Producing/TOS-2012-Pt2-Distribution.pdf>PBS Technical Operating Specifications</a> document. Information about EBU loudness standards can be found in the <a href=https://tech.ebu.ch/docs/r/r128-2014.pdf>EBU R 128</a> recommendation document.</p>
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<p><code>ffmpeg -i <i>input_file</i> -af loudnorm=dual_mono=true -ar 48k <i>output_file</i></code></p>
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<p>This will normalize the loudness of an input using one pass, which is quicker but less accurate than using two passes. This command uses the loudnorm filter defaults for target loudness. These defaults align well with PBS recommendations, but loudnorm does allow targeting of specific loudness levels. More information can be found at the <a href="https://ffmpeg.org/ffmpeg-filters.html#loudnorm" target="_blank">loudnorm documentation</a>.</p>
|
||||
<p>Information about PBS loudness standards can be found in the <a href="http://www-tc.pbs.org/capt/Producing/TOS-2012-Pt2-Distribution.pdf" target="_blank">PBS Technical Operating Specifications</a> document. Information about EBU loudness standards can be found in the <a href="https://tech.ebu.ch/docs/r/r128-2014.pdf" target="_blank">EBU R 128</a> recommendation document.</p>
|
||||
<dl>
|
||||
<dt>ffmpeg</dt><dd>starts the command</dd>
|
||||
<dt><i>input_file</i></dt><dd>path, name and extension of the input file</dd>
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||||
<dt>-af loudnorm</dt><dd>activates the loudnorm filter with default settings</dd>
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<dt>dual_mono=true</dt><dd>(optional) Use this for mono files meant to be played back on stereo systems for correct loudness. Not necessary for multi-track inputs.</dd>
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<dt>-ar 48k</dt><dd>Sets the output sample rate to 48 kHz. (The loudnorm filter upsamples to 192 kHz so it is best to manually set a desired output sample rate).</dd>
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||||
<dt><i>output_file</i></dt><dd>path, name and extension for output file</dd>
|
||||
</dl>
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||||
<p class="link"></p>
|
||||
@@ -1166,9 +1191,9 @@
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||||
<div class="modal-content">
|
||||
<div class="well">
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<h3>Two Pass Loudness Normalization</h3>
|
||||
<p><code>ffmpeg -i <i>input_file</i> -af loudnorm=dual_mono=true:measured_I=<i>input_i</i>:measured_TP=<i>input_tp</i>:measured_LRA=<i>input_lra</i>:measured_thresh=<i>input_thresh</i>:offset=<i>target_offset</i>:linear=true <i>output_file</i></code></p>
|
||||
<p>This command allows using the levels calculated using a <a href=#loudnorm_metadata>first pass of the loudnorm filter</a> to more accurately normalize loudness. This command uses the loudnorm filter defaults for target loudness. These defaults align well with PBS recommendations, but loudnorm does allow targeting of specific loudness levels. More information can be found at the <a href=https://ffmpeg.org/ffmpeg-filters.html#loudnorm>loudnorm documentation</a>.</p>
|
||||
<p>Information about PBS loudness standards can be found in the <a href=http://www-tc.pbs.org/capt/Producing/TOS-2012-Pt2-Distribution.pdf>PBS Technical Operating Specifications</a> document. Information about EBU loudness standards can be found in the <a href=https://tech.ebu.ch/docs/r/r128-2014.pdf>EBU R 128</a> recommendation document.</p>
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<p><code>ffmpeg -i <i>input_file</i> -af loudnorm=dual_mono=true:measured_I=<i>input_i</i>:measured_TP=<i>input_tp</i>:measured_LRA=<i>input_lra</i>:measured_thresh=<i>input_thresh</i>:offset=<i>target_offset</i>:linear=true -ar 48k <i>output_file</i></code></p>
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<p>This command allows using the levels calculated using a <a href=#loudnorm_metadata>first pass of the loudnorm filter</a> to more accurately normalize loudness. This command uses the loudnorm filter defaults for target loudness. These defaults align well with PBS recommendations, but loudnorm does allow targeting of specific loudness levels. More information can be found at the <a href="https://ffmpeg.org/ffmpeg-filters.html#loudnorm" target="_blank">loudnorm documentation</a>.</p>
|
||||
<p>Information about PBS loudness standards can be found in the <a href="http://www-tc.pbs.org/capt/Producing/TOS-2012-Pt2-Distribution.pdf" target="_blank">PBS Technical Operating Specifications</a> document. Information about EBU loudness standards can be found in the <a href="https://tech.ebu.ch/docs/r/r128-2014.pdf" target="_blank">EBU R 128</a> recommendation document.</p>
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<dl>
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<dt>ffmpeg</dt><dd>starts the command</dd>
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<dt><i>input_file</i></dt><dd>path, name and extension of the input file</dd>
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@@ -1180,6 +1205,7 @@
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<dt>measured_LRA=<i>input_thresh</i></dt><dd>use the 'input_thresh' value (threshold) from the first pass in place of input_thresh</dd>
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<dt>offset=<i>target_offset</i></dt><dd>use the 'target_offset' value (offset) from the first pass in place of target_offset</dd>
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<dt>linear=true</dt><dd>tells loudnorm to use linear normalization</dd>
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<dt>-ar 48k</dt><dd>Sets the output sample rate to 48 kHz. (The loudnorm filter upsamples to 192 kHz so it is best to manually set a desired output sample rate).</dd>
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<dt><i>output_file</i></dt><dd>path, name and extension for output file</dd>
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</dl>
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<p class="link"></p>
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@@ -1197,7 +1223,7 @@
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<div class="well">
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<h3>RIAA Equalization</h3>
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<p><code>ffmpeg -i <i>input_file</i> -af aemphasis=type=riaa <i>output_file</i></code></p>
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<p>This will apply RIAA equalization to an input file allowing correct listening of audio transferred 'flat' (without EQ) from records that used this EQ curve. For more information about RIAA equalization see the <a href=https://en.wikipedia.org/wiki/RIAA_equalization>Wikipedia page</a> on the subject.</p>
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<p>This will apply RIAA equalization to an input file allowing correct listening of audio transferred 'flat' (without EQ) from records that used this EQ curve. For more information about RIAA equalization see the <a href="https://en.wikipedia.org/wiki/RIAA_equalization" target="_blank">Wikipedia page</a> on the subject.</p>
|
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<dl>
|
||||
<dt>ffmpeg</dt><dd>starts the command</dd>
|
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<dt><i>input_file</i></dt><dd>path, name and extension of the input file</dd>
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@@ -1481,15 +1507,15 @@ foreach ($file in $inputfiles) {
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<div class="well">
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<h3>Read/Extract EIA-608 (Line 21) closed captioning</h3>
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<p><code>ffprobe -f lavfi -i movie=<i>input_file</i>,readeia608 -show_entries frame=pkt_pts_time:frame_tags=lavfi.readeia608.0.line,lavfi.readeia608.0.cc,lavfi.readeia608.1.line,lavfi.readeia608.1.cc -of csv > <i>input_file</i>.csv</code></p>
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<p>This command uses FFmpeg's <a href="https://ffmpeg.org/ffmpeg-filters.html#readeia608" target="_blank">readeia608</a> filter to extract the hexidecimal values hidden within <a href="https://en.wikipedia.org/wiki/EIA-608" target="_blank">EIA-608 (Line 21)</a> Closed Captioning, outputting a csv file. For more information about EIA-608, check out Adobe's <a href="https://www.adobe.com/content/dam/Adobe/en/devnet/video/pdfs/introduction_to_closed_captions.pdf" target="_blank">Introduction to Closed Captions</a>.</p>
|
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<p>This command uses FFmpeg's <a href="https://ffmpeg.org/ffmpeg-filters.html#readeia608" target="_blank">readeia608</a> filter to extract the hexadecimal values hidden within <a href="https://en.wikipedia.org/wiki/EIA-608" target="_blank">EIA-608 (Line 21)</a> Closed Captioning, outputting a csv file. For more information about EIA-608, check out Adobe's <a href="https://www.adobe.com/content/dam/Adobe/en/devnet/video/pdfs/introduction_to_closed_captions.pdf" target="_blank">Introduction to Closed Captions</a>.</p>
|
||||
<p>If hex isn't your thing, closed captioning <a href="http://www.theneitherworld.com/mcpoodle/SCC_TOOLS/DOCS/CC_CHARS.HTML" target="_blank">character</a> and <a href="http://www.theneitherworld.com/mcpoodle/SCC_TOOLS/DOCS/CC_CODES.HTML" target="_blank">code</a> sets can be found in the documentation for SCTools.</p>
|
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<dl>
|
||||
<dt>ffprobe</dt><dd>starts the command</dd>
|
||||
<dt>-f lavfi</dt><dd>tells ffprobe to use the <a href="http://ffmpeg.org/ffmpeg-devices.html#lavfi" target="_blank">libavfilter</a> input virtual device</a></dd>
|
||||
<dt>-f lavfi</dt><dd>tells ffprobe to use the <a href="http://ffmpeg.org/ffmpeg-devices.html#lavfi" target="_blank">libavfilter</a> input virtual device</dd>
|
||||
<dt>-i <i>input_file</i></dt><dd>input file and parameters</dd>
|
||||
<dt>readeia608 -show_entries frame=pkt_pts_time:frame_tags=lavfi.readeia608.0.line,lavfi.readeia608.0.cc,lavfi.readeia608.1.line,lavfi.readeia608.1.cc -of csv</dt><dd>specifies the first two lines of video in which EIA-608 data (hexidecimal byte pairs) are identifiable by ffprobe, outputting comma separated values (CSV)</dt>
|
||||
<dt>></dt><dd>redirects the standard output (the data created by ffprobe about the video)</dt>
|
||||
<dt><i>output_file</i>.csv</dt><dd>names the CSV output file</dt>
|
||||
<dt>readeia608 -show_entries frame=pkt_pts_time:frame_tags=lavfi.readeia608.0.line,lavfi.readeia608.0.cc,lavfi.readeia608.1.line,lavfi.readeia608.1.cc -of csv</dt><dd>specifies the first two lines of video in which EIA-608 data (hexadecimal byte pairs) are identifiable by ffprobe, outputting comma separated values (CSV)</dd>
|
||||
<dt>></dt><dd>redirects the standard output (the data created by ffprobe about the video)</dd>
|
||||
<dt><i>output_file</i>.csv</dt><dd>names the CSV output file</dd>
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</dl>
|
||||
<div class="sample-image">
|
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<h4>Example</h4>
|
||||
@@ -2035,7 +2061,7 @@ e.g.: <code>ffmpeg -f concat -safe 0 -i mylist.txt -c copy <i>output_file</i></c
|
||||
<div class="modal-content">
|
||||
<div class="well">
|
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<h3>Transcode an image sequence into uncompressed 10-bit video</h3>
|
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<p><code>ffmpeg -f image2 -framerate 24 -i <i>input_file_%06d.ext</i> -c:v v210 -an <i>output_file</i></code></p>
|
||||
<p><code>ffmpeg -f image2 -framerate 24 -i <i>input_file_%06d.ext</i> -c:v v210 <i>output_file</i></code></p>
|
||||
<dl>
|
||||
<dt>ffmpeg</dt><dd>starts the command</dd>
|
||||
<dt>-f image2</dt><dd>forces the image file de-muxer for single image files</dd>
|
||||
@@ -2043,7 +2069,6 @@ e.g.: <code>ffmpeg -f concat -safe 0 -i mylist.txt -c copy <i>output_file</i></c
|
||||
<dt>-i <i>input_file</i></dt><dd>path, name and extension of the input file<br>
|
||||
This must match the naming convention actually used! The regex %06d matches six digits long numbers, possibly with leading zeroes. This allows to read in ascending order, one image after the other, the full sequence inside one folder. For image sequences starting with 086400 (i.e. captured with a timecode starting at 01:00:00:00 and at 24 fps), add the flag <code>-start_number 086400</code> before <code>-i input_file_%06d.ext</code>. The extension for TIFF files is .tif or maybe .tiff; the extension for DPX files is .dpx (or eventually .cin for old files).</dd>
|
||||
<dt>-c:v v210</dt><dd>encodes an uncompressed 10-bit video stream</dd>
|
||||
<dt>-an copy</dt><dd>no audio</dd>
|
||||
<dt><i>output_file</i></dt><dd>path, name and extension of the output file</dd>
|
||||
</dl>
|
||||
<p class="link"></p>
|
||||
@@ -2240,7 +2265,31 @@ e.g.: <code>ffmpeg -f concat -safe 0 -i mylist.txt -c copy <i>output_file</i></c
|
||||
</div>
|
||||
<!-- ends Strip metadata -->
|
||||
|
||||
<!-- Scene Detection using YDIF -->
|
||||
<span data-toggle="modal" data-target="#csv-ydif"><button type="button" class="btn btn-default" data-toggle="tooltip" data-placement="bottom" title="Exports CSV for scene detection using YDIF">CSV with timecodes and YDIF</button></span>
|
||||
<div id="csv-ydif" class="modal fade" tabindex="-1" role="dialog">
|
||||
<div class="modal-dialog modal-lg">
|
||||
<div class="modal-content">
|
||||
<div class="well">
|
||||
<h3>Exports CSV for scene detection using YDIF</h3>
|
||||
<p><code>ffprobe -f lavfi -i movie=<i>input_file</i>,signalstats -show_entries frame=pkt_pts_time:frame_tags=lavfi.signalstats.YDIF -of csv</code></p>
|
||||
<p>This ffprobe command prints a CSV correlating timestamps and their YDIF values, useful for determining cuts.</p>
|
||||
<dl>
|
||||
<dt>ffprobe</dt><dd>starts the command</dd>
|
||||
<dt>-f lavfi</dt><dd>uses the <a href="http://ffmpeg.org/ffmpeg-devices.html#lavfi" target="_blank">Libavfilter input virtual device</a> as chosen format</dd>
|
||||
<dt>-i movie=<i>input file</i></dt><dd>path, name and extension of the input video file</dd>
|
||||
<dt>,</dt><dd>comma signifies closing of video source assertion and ready for filter assertion</dd>
|
||||
<dt>signalstats</dt><dd>tells ffprobe to use the signalstats command</dd>
|
||||
<dt>-show_entries</dt><dd>sets list of entries to show per column, determined on the next line</dd>
|
||||
<dt>frame=pkt_pts_time:frame_tags=lavfi.signalstats.YDIF</dt><dd>specifies showing the timecode (<code>pkt_pts_time</code>) in the frame stream and the YDIF section of the frame_tags stream</dd>
|
||||
<dt>-of csv</dt><dd>sets the output printing format to CSV. <code>-of</code> is an alias of <code>-print_format</code>.</dd>
|
||||
</dl>
|
||||
<p class="link"></p>
|
||||
</div>
|
||||
</div>
|
||||
</div>
|
||||
</div>
|
||||
<!-- ends sample Scene Detection using YDIF -->
|
||||
|
||||
|
||||
<!-- sample example -->
|
||||
@@ -2267,6 +2316,7 @@ Change the above data-target field, the hover-over description, the button text,
|
||||
<!-- ends sample example -->
|
||||
|
||||
|
||||
</div>
|
||||
</div> <!-- end "well col-md-6 col-md-offset-2" -->
|
||||
</div> <!-- row -->
|
||||
|
||||
|
||||
Reference in New Issue
Block a user