mirror of
https://github.com/amiaopensource/ffmprovisr.git
synced 2024-11-06 05:27:24 +01:00
change wording
This commit is contained in:
parent
60d452a431
commit
2d6bf9159f
@ -1225,7 +1225,7 @@
|
||||
<h3>Remove silent portion at the beginning of an audio file</h3>
|
||||
<p><code>ffmpeg -i <em>input_file</em> -af silenceremove=start_threshold=-57dB:start_duration=1:start_periods=1 -c:a <em>your_codec_choice</em> -ar <em>your_sample_rate_choice</em> <em>output_file</em></code></p>
|
||||
<p>This command will automatically remove silence at the beginning of an audio file. The threshold for what qualifies as silence can be changed - this example uses anything under -57 dB, which is a decent level for accounting for analogue hiss.</p>
|
||||
<p><strong>Note:</strong> Since this command uses a filter, the audio stream will be re-encoded for the output. If you do not specify an output sample rate and codec this command will use <a href='#codec-defaults'>the defaults for your output format</a>. Take care that you are getting your intended results!</p>
|
||||
<p><strong>Note:</strong> Since this command uses a filter, the audio stream will be re-encoded for the output. If you do not specify a sample rate or codec, this command will use the sample rate from your input and <a href='#codec-defaults'>the codec defaults for your output format</a>. Take care that you are getting your intended results!</p>
|
||||
<dl>
|
||||
<dt>ffmpeg</dt><dd>starts the command</dd>
|
||||
<dt>-i <em>input_file</em></dt><dd>path, name and extension of the input file (e.g. input_file.wav)</dd>
|
||||
@ -1234,7 +1234,6 @@
|
||||
<dt>start_duration=1</dt><dd>This tells the filter how much non-silent audio must be detected before it stops trimming. With a value of <code>0</code> the filter would stop after detecting any non-silent audio. A setting of <code>1</code> allows it to continue trimming through short 'pops' such as those caused by engaging the playback device, or the recorded sound of a microphone being plugged in.</dd>
|
||||
<dt>start_periods=1</dt><dd>This tells the filter to trim the first example of silence it discovers from the beginning of the file. This value could be increased to remove subsequent silent portions from the file if desired.</dd>
|
||||
<dt>-c:a <em>your_codec_choice</em></dt><dd>This tells the filter what codec to use, and must be specified to avoid defaults. If you want 24 bit PCM, your value would be <code>-c:a pcm_s24le</code>.</dd>
|
||||
<dt>-ar <em>your_sample_rate_choice</em></dt><dd>This tells the filter what sample rate to use, and must be specified to avoid defaults. If you want 96 kHz, your value would be <code>-ar 96000</code>.</dd>
|
||||
<dt><em>output_file</em></dt><dd>path, name and extension of the output file (e.g. output_file.wav).</dd>
|
||||
</dl>
|
||||
</div>
|
||||
@ -1247,7 +1246,7 @@
|
||||
<h3>Remove silent portion from the end of an audio file</h3>
|
||||
<p><code>ffmpeg -i <em>input_file</em> -af areverse,silenceremove=start_threshold=-57dB:start_duration=1:start_periods=1,areverse -c:a <em>your_codec_choice</em> -ar <em>your_sample_rate_choice</em> <em>output_file</em></code></p>
|
||||
<p>This command will automatically remove silence at the end of an audio file. Since the <code>silenceremove</code> filter is best at removing silence from the beginning of files, this command used the <code>areverse</code> filter twice to reverse the input, remove silence and then restore correct orientation.</p>
|
||||
<p><strong>Note:</strong> Since this command uses a filter, the audio stream will be re-encoded for the output. If you do not specify an output sample rate and codec this command will use <a href='#codec-defaults'>the defaults for your output format</a>. Take care that you are getting your intended results!</p>
|
||||
<p><strong>Note:</strong> Since this command uses a filter, the audio stream will be re-encoded for the output. If you do not specify a sample rate or codec, this command will use the sample rate from your input and <a href='#codec-defaults'>the codec defaults for your output format</a>. Take care that you are getting your intended results!</p>
|
||||
<dl>
|
||||
<dt>ffmpeg</dt><dd>starts the command</dd>
|
||||
<dt>-i <em>input_file</em></dt><dd>path, name and extension of the input file (e.g. input_file.wav)</dd>
|
||||
@ -1258,7 +1257,6 @@
|
||||
<dt>start_periods=1</dt><dd>This tells the filter to trim the first example of silence it discovers.</dd>
|
||||
<dt>areverse</dt><dd>applies the audio reverse filter again to restore input to correct orientation.</dd>
|
||||
<dt>-c:a <em>your_codec_choice</em></dt><dd>This tells the filter what codec to use, and must be specified to avoid defaults. If you want 24 bit PCM, your value would be <code>-c:a pcm_s24le</code>.</dd>
|
||||
<dt>-ar <em>your_sample_rate_choice</em></dt><dd>This tells the filter what sample rate to use, and must be specified to avoid defaults. If you want 96 kHz, your value would be <code>-ar 96000</code>.</dd>
|
||||
<dt><em>output_file</em></dt><dd>path, name and extension of the output file (e.g. output_file.wav).</dd>
|
||||
</dl>
|
||||
</div>
|
||||
|
Loading…
Reference in New Issue
Block a user