[Ffmpeg-devel] Question about parameters
Sun Dec 24 18:54:40 CET 2006
Michael Niedermayer wrote:
> On Sat, Dec 23, 2006 at 07:06:39PM -0800, Aaron Williams wrote:
>> Robert Swain wrote:
>>> On 24 Dec 2006, at 01:57, Aaron Williams wrote:
>>>> I am writing a new audio "codec" which basically computes the peak RMS
>>>> volume for normalizing audio in another pass and am wondering if there's
>>>> a standard way for my codec to introduce a new parameter so I can
>>>> specify the RMS window size? What is the best way for me to add this
>>>> parameter to pass to my module? My goal is to use the output from this
>>>> to adjust the -vol parameter in the transcoding pass.
>>>> Sorry about my previous posting.
>>> Did you intend to quote yourself and reply to the old thread again to
>>> make this apology or do you still think you're not hijacking the
>>> thread with this second posting? Don't reply to a thread and change
>>> the subject to start a new thread. Compose a completely new mail and
>>> send it to the appropriate address.
>> I created a new thread and did not post to the old thread again. It is
>> in no way an attempt to hijack the old thread. My previous posting
>> within the other thread was a mistake. My new posting has no reference
>> to my previous posting, hence trying to clean it up and keep it out of
>> the other thread.
>> I consider the previous posting dead and will no longer respond to any
>> more messages regarding my mistake in posting a new thread since I would
>> rather see the discussion continue in a civilized and professional
>> manner and try and keep on the topic of ffmpeg development.
>> I will repeat my question: How does one add codec specific parameters
>> to the command line of ffmpeg without adding a general purpose option
>> and without bloating AVCodecContext with new fields not needed for other
>> codecs? There are a number of parameters I wish to add for
>> normalization such as RMS window size, target volume, thresholds and an
>> output log filename.
>> Another audio feature I would like to add is the ability to adjust the
>> level of individual channels.
> what you describe are audio filters not audio codecs and while it should
> be possible to implement filters as raw-pcm codecs that would cause
> alot of problems (think of chains or networks of codecs) -> its not
> acceptable for ffmpeg svn and as its not acceptable for svn theres
> no point in asking here how to solve the various problems ensuing
> due to this missdesign
One drawback of a filter is that the normalizing pass is basically a
dead-end. By definition, two passes are required. The first pass must
process the entire file and find the peak levels. Also, as was
previously stated, there currently is no audio filter support. If this
were present then I might make use of this. Right now the normalization
pass goes very quickly since it only decodes the audio stream to PCM and
does no video decoding.
If there is a misdesign, it is in the basic ffmpeg itself in that there
is no support for audio filters or for dynamic parameters. Each codec
and/or filter should be able to supply additional command-line
parameters as needed. This will add greater flexibility and eliminate
parameters in core data structures that are only used by one or two modules.
Actually, it fits quite nicely as a codec, though it does not output
anything. The current design lets it easily specify the window size
which works out perfectly. My only requirement is that I be able to
pass enough information to generate a small log file with the results of
As far as modifying the volume output, this can already be accomplished
during the second transcoding pass by using the -vol parameter. I have
made a change so that it can take a value like it currently does (i.e.
defaults to 256) or one can specify the output volume in DB. My goal is
to also be able to specify a log file to be read at startup with the
volume adjustments, which should be simple to do.
More information about the ffmpeg-devel