[asterisk-dev] (no subject)

[dimas at dataart.com]

Guys, you are doing web application. You have millions of tools at your disposal. You can do whatever you want with that BLOB and convert it to any format you wish. So why something needs to be changed in Asterisk?

We also have voicemail in ODBC database (format = wav49). And we also provide web UI with flash player allowing users to listen these messages. Only two commands (sox and lame encoder) needed to convert these to mp3:

/usr/bin/sox -t wav - -t raw -u -c 1 -b - | /usr/local/bin/lame -r -s 8 --bitwidth 8 -m m --preset phone - FILENAME

Send your BLOB into standard input of this pipeline and final MP3 will be placed under FILENAME.

Regards,
Dmitry Andrianov

-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of Atis Lezdins
Sent: Thursday, November 26, 2009 9:16 PM
To: Asterisk Developers Mailing List
Subject: Re: [asterisk-dev] ODBC stored voicemail blob object format

On Thu, Nov 26, 2009 at 7:38 PM, Tilghman Lesher <tlesher at digium.com> wrote:
> On Thursday 26 November 2009 05:58:35 Nir Simionovich wrote:
>>   In general, I gathered the the BLOB object stored in the database
>> isn't a full WAV file format, and actually is
>> a RAW ADPCM file, which is playable from a standard media player.
>> However, the file in non-playable from
>> an ADOBE based flash player - which requires a fully standard WAV file.
>>
>>   Questions that arise in this case are:
>> 1. Is there a method to convert the stored BLOB to a valid WAV format?
>> 2. Adding proper WAV support to Asterisk, is it feasible?
>> 3. Is there an external method of converting from the RAW format to
>> proper WAV? (tried SOX with no success).
>>
>>   I'm not an audiophile, so I'm kind'a of at a loss here.
>
> From my past experience with Flash development, I've ascertained that it
> CANNOT handle the wav49 format (aka "WAV"), which is a compressed format, but
> only the uncompressed "wav" format.  MPEG Layer 3 was another option, though
> of course we don't have a licensed encoder for that in Asterisk.
>

To my knowledge flash based players is real pain.. as it has to be not
only standart wav file, but also exactly 11khz or some other specific
supported sample rate (44khz, etc)

So, as asterisk works with 8khz you have to resample everything before
passing it to flash player (even if it's mp3).

If possible, try out HTML5 audio support and let community know the
results. It would be cool to add ogg support in voicemail, it seems
that firefox supoorts it by default.

Regards,
Atis

_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev