Audio Formats

Our audio only conversion service enables you to strip the audio out of a video file to create a standalone audio file. You can easily do this via our API or customer area: http://www.encoding.com/login

via API:
For this example, we will use the MP3 template. in your POST, call for the following task: http://www.encoding.com/help/article/aac_audio_template

via web UI:
In the task viewer, please add the MP3, Windows Media Audio, or AAC audio template.

For more information:
http://www.encoding.com/features/audio_images

Interested in viewing the templates for API usage or web UI:
http://www.encoding.com/help/article/aac_audio_template
http://www.encoding.com/help/article/windows_media_audio_template
http://www.encoding.com/help/article/mp3_audio_template

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<metadata>
<copy>yes</copy>
</metadata>

Need to pass supported metadata information in your mp3 --> mp3 encodes. Use this addition to your XML.  Valuable information, such as title, artist and album will be passed through on your encodes. 

<metadata>
<copy>yes</copy>
</metadata>

Need to inject other metadata? Please check out our API documentation on metadata insertion:

http://www.encoding.com/api/category/category/complete_api_documentation#metadata_fields

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 <audio_sync>1</audio_sync> 

You can easily correct audio sync issues by setting the audio_sync tag in your XML, as follows: 

<audio_sync>1</audio_sync>

This works by adjusting the output audio stream to match the source timestamps, in # of samples per second. Consider this a great for fixing sync when source is variable framerate, or has timescale problems.

The most common instance of this is webcam video recorded with Nellymoser audio codec from the free flash plugin.

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<audio_codec>copy</audio_codec>

If you specify the audio_codec tag to equal 'copy', the options audio_channels_number, audio_sample_rate, audio_bitrate will be ignored and their values will be copied from your source file.

Example:

<audio_codec>copy</audio_codec>

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<format>
    <audio_bitrate>64k</audio_bitrate>
    <audio_channels_number>2</audio_channels_number>
    <audio_codec>libfaac</audio_codec>
    <audio_volume>100</audio_volume>
    <output>m4a</output>
    <file_extension>m4a</file_extension>
</format>

Standard AAC Audio Template

 

<format>
    <audio_bitrate>64k</audio_bitrate>
    <audio_channels_number>2</audio_channels_number>
    <audio_codec>libfaac</audio_codec>
    <audio_volume>100</audio_volume>
    <output>m4a</output>
    <file_extension>m4a</file_extension>
</format>

 

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<format>
    <audio_bitrate>64k</audio_bitrate>
    <audio_sample_rate>44100</audio_sample_rate>
    <audio_channels_number>2</audio_channels_number>
    <audio_codec>wmav2</audio_codec>
    <audio_volume>100</audio_volume>
    <output>wma</output>
    <file_extension>wma</file_extension>
</format>

Standard Windows Media Audio Template

 

<format>
    <audio_bitrate>64k</audio_bitrate>
    <audio_sample_rate>44100</audio_sample_rate>
    <audio_channels_number>2</audio_channels_number>
    <audio_codec>wmav2</audio_codec>
    <audio_volume>100</audio_volume>
    <output>wma</output>
    <file_extension>wma</file_extension>
</format>

 

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<format> 
    <audio_bitrate>64k</audio_bitrate>
    <audio_sample_rate>44100</audio_sample_rate>
    <audio_channels_number>2</audio_channels_number>
    <audio_codec>libmp3lame</audio_codec>
    <audio_volume>100</audio_volume>
    <output>mp3</output>
    <file_extension>mp3</file_extension>
</format>

Standard MP3 Audio Template

 

<format> 
    <audio_bitrate>64k</audio_bitrate>
    <audio_sample_rate>44100</audio_sample_rate>
    <audio_channels_number>2</audio_channels_number>
    <audio_codec>libmp3lame</audio_codec>
    <audio_volume>100</audio_volume>
    <output>mp3</output>
    <file_extension>mp3</file_extension>
</format>

 

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Sometimes it is required to use Constant bitrate (CBR) instead of Variable bitrate (VBR), which is default.

Usually it is helpful for video streaming.  

To switch on this option, just add <cbr>yes</cbr> within the <format> section of your XML query. 

Please note that you also MUST explicitly specify  <bitrate> in this case.

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The maximum is 2822400 Hz

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In most cases you can use the default audio bitrate of 64k. If you need quality stereo sound, use 128k or 192k. For home DVD systems use up to 320k, assuming that your source video has appropriate audio quality. Bit rate/filesize mapping is simple. The size of the audio stream is the audio bitrate multiplied by the duration and divided by 8: size (bytes) = bitrate (bps) * duration (sec) / 8 .

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Some audio codecs (e.g. AAC or MPEG-Layer-3) have their own limitation for options values. When you leave these options blank in your API request, encoding.com will get them from the source file, and not always their set will be correct for particular codec. So, it's better to always set audio codec parameters explicitly.

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