[Live-devel] Implement RTSP skip forward in mediaServer

[Steve Malenfant]

I guess it's related to PCR rolling over within that file....

[steve at CentosP4 readable]$ head -10000 1431_20090215193000.ts | dvbsnoop -if
- -s ts -tssubdecode | grep clock
        program_clock_reference:
             ==> program_clock_reference: 2480569434945 (0x2418d75c341)  [=
PCR-Timestamp: 25:31:12.942035]
        program_clock_reference:
             ==> program_clock_reference: 2480570988683 (0x2418d8d788b)  [=
PCR-Timestamp: 25:31:12.999580]
        program_clock_reference:
             ==> program_clock_reference: 2480571369790 (0x2418d93493e)  [=
PCR-Timestamp: 25:31:13.013695]

[steve at CentosP4 readable]$ tail -10000 1431_20090215193000.ts | dvbsnoop -if
- -s ts -tssubdecode | grep clock
        program_clock_reference:
             ==> program_clock_reference: 146458110601 (0x2219956a89)  [=
PCR-Timestamp: 1:30:24.374466]
        program_clock_reference:
             ==> program_clock_reference: 146458843495 (0x2219a09967)  [=
PCR-Timestamp: 1:30:24.401610]
        program_clock_reference:
             ==> program_clock_reference: 146459538699 (0x2219ab350b)  [=
PCR-Timestamp: 1:30:24.427359]


On Thu, Apr 2, 2009 at 5:42 PM, Steve Malenfant <smalenfant at gmail.com>wrote:

> Here some more details to make it obvious:
>
> [steve at CentosP4 readable]$
> /home/steve/Desktop/testMPEG2TransportStreamTrickPlay 1431_20090215193000.ts
> 3600 1 test2.ts
> Writing output file "test2.ts" (start time 3570.539062, scale 1)...
> [steve at CentosP4 readable]$
> /home/steve/Desktop/testMPEG2TransportStreamTrickPlay 1431_20090215193000.ts
> 4600 1 test2.ts
> Writing output file "test2.ts" (start time 3570.539062, scale 1)...
>
> They both start at 3570.539062 start time.
>
>
> On Thu, Apr 2, 2009 at 3:52 PM, Steve Malenfant <smalenfant at gmail.com>wrote:
>
>> Is there a possible bug that after you request something higher than 3570
>> seconds it is not working on transport stream files? It always brings me
>> back to the same scene.
>>
>> Transport file length : 20177187044 (20GB)
>> Index file lenght : 1151033004 (1.1 GB)
>>
>> Thanks.
>>
>>
>> On Thu, Apr 2, 2009 at 11:18 AM, Steve Malenfant <smalenfant at gmail.com>wrote:
>>
>>> I can't use the live555 client, I'm using an Amino STB. My first test
>>> failed (maybe I used the wrong stream with no trick play), but my second
>>> test just worked.
>>>
>>> Thanks.
>>>
>>>
>>> On Wed, Apr 1, 2009 at 5:21 PM, Ross Finlayson <finlayson at live555.com>wrote:
>>>
>>>>  Seems like there is an option in RTSP to specify a range in time that
>>>> you want to start your stream.
>>>>
>>>>
>>>> Yes, and we support it, for both RTSP clients and RTSP servers.
>>>>
>>>> For RTSP clients: Note the "start" and "end" parameters to "RTSPClient::
>>>> playMediaSession()" and "RTSPClient:: playMediaSubsession()".  These
>>>> parameters (if set to non-default values) tell the RTSP client to request a
>>>> specific time range.
>>>>
>>>> For RTSP servers: Our RTSP server implementation (including its use in
>>>> the "LIVE555 Media Server" product) supports these requests, ***provided
>>>> that*** the underlying file type can handle them.  In our current
>>>> implementation, the following file types support this:
>>>> - MP3 audio files
>>>> - MPEG-1or 2 audio/video Program Stream files (but not reliably)
>>>> - MPEG Transport Stream files (provided that they each have an 'index
>>>> file'; see the documentation)
>>>> - WAV audio files
>>>>
>>>> For other file types (including MPEG-4 video files), our implementation
>>>> currently does *not* support seeking.
>>>>
>>>> http://www.myiptv.org/Articles/RTSP/tabid/72/Default.aspx
>>>> "The important bits of this command are Range and Scale. See I said you
>>>> would want to know the range. Range specifies from where and how much of the
>>>> content to play. 0- tells the server to start at the beginning and play to
>>>> the end but you could also start anywhere in the file as we'll see in a
>>>> minute or only play the first 5 minutes of the content. It's up to you."
>>>>
>>>>
>>>> You don't have to worry about the details of the RTSP protocol; we
>>>> implement all of this for you.  Just use our "RTSPClient" class, and pass
>>>> the appropriate parameters (as noted above).
>>>>
>>>> --
>>>>
>>>>
>>>> Ross Finlayson
>>>> Live Networks, Inc.
>>>> http://www.live555.com/
>>>>
>>>> _______________________________________________
>>>> live-devel mailing list
>>>> live-devel at lists.live555.com
>>>> http://lists.live555.com/mailman/listinfo/live-devel
>>>>
>>>>
>>>
>>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.live555.com/pipermail/live-devel/attachments/20090402/b2ede7ab/attachment.html>