Audio Formats

 

There are 3 situations where we have seen audio sync issues originate. In each of these cases, the <audio_sync> feature can be useful:

1. Apple Final Cut Pro 7 sequences with mixed timecode. This was was fixed in FCP X, so FCP7 and versions below can create outputs that have incorrect timescale.
2. MPEG live capture streams with variable timescale or framerates.
3. Adobe Flash video recorder with Nellymoser audio (variable timescale).

NOTE: If you are submitting QuickTime (.mov) source, a more reliable workaround is to export MP4 (.mp4) from your computer locally. We recommend using Handbrake or MPEG Streamclip, since these free apps will correct broken timescale using native Apple QuickTime core media 64-bit processing. Please send the corrected MP4 thru as source and your job should be fine. If you are still having problems, please reach out for help. 

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 <format> 
<!-- Format fields --> 
 <audio_normalization>[0-100]</audio_normalization> 
 ... 

Varying content and varying edit styles can cause unexpected results during compression. Correct any and all issues with audio_normalization. This will correct variances in dialogue, natural sound and background music. There is no default value for this tag. Please specify a non-negative integer between 0 and 100 for this value.

<format> 
<!-- Format fields --> 
    <audio_normalization>[0-100]</audio_normalization> 
    ... 
</format>

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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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A common issue with video encoding involves the video and audio going out of sync. This can happen right at the beginning of a video, or slowly over time. Here's some possible fixes: 1. Make sure the output framerate setting matches the source file's framerate. For example: if source is 29.97 fps video into encoding.com, put 29.97 into the frame rate setting for output. 2. Set the audio sample rate in the format request. As with frame rate, ensure this value matches the source input video's audio sample rate. 3. If the audio is going out over time, or slowly loses, then corrects itself every few seconds: Set the keyframes value to a lower figure. Default for encoding.com is generally 300, which in 30 fps material, is one keyframe every 10 seconds. Try lowering the value to every 3 seconds, 90 frames per second. 4. QuickTime sometimes has timescale issues when bringing MPEG source into Avid or Final Cut Pro. Try converting it to your native edit codec using Apple Compressor or Quicktime Player export. This should correct the A/V sync and allow you to edit the file normally. This issue can occur for a variety of reasons, and in some cases, it may be impossible in the video encoder to properly encode the video without engineering resources. If this should be your case, after trying the steps listed here, please contact support at helpdesk@encoding.com and have your MediaID(s), source files and desired output formats ready.

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