Prophecy version: 14.5.2
I am using the following CCXML line to playback audio files over a phone call:
<dialogstart src="'NoResponse.wav'" type="'audio/wav'"/>
This works the majority of the time. However sometime a few threads will begin having errors during the playback of an audio file, this doesn't seem to be tied to any specific audio file. In the logs I get a SIP event error with the following details (only showing relavent bits from log):
CTStream/Failed to read from VIO accessor.
CTStream/Transcoder:failed to read from backbone.
Reading from playback stream has failed.
name="error.dialog.notstarted" reason="audio playback conflict"
Any information the leads to the resolution of this issue would be greatly appreciated. I have not found any information on this in the Prophecy documents.
Related
I'm really stumped here. I'm trying to use the ONVIF protocol to enable/disable audio on RTSP streams from an IP camera. If you wish to disable audio on a profile, you should first send 'removeAudioEncoderConfiguration' to remove the audio codec, then send 'removeAudioSourceConfiguration' on the same profile.
So I do this for ALL profiles on the camera (there happen to be 5). However, the RTSP stream still has audio in it.
This is happening on a Uniview camera. If I do the same sequence of messages on a Techwin device I have, then the audio is disabled.
Similarly if I try to enable audio on a profile, I do the reverse operations. Again, the Techwin adds the audio, but the Uniview does not.
The responses I received to my ONVIF messages are always 'OK'.
Any idea what I'm doing wrong and how to fix it?
I'm developing a web receiver for chromecast.
According to chromecast documentation, HLS with AC-3 passthrough is supported. It is to be checked with canDisplayType and following that, I assume it should be okay to use the codec.
I cannot seem to get the chromecast to accept HLS manifests with AC-3 in the codec list. I have tried this on chromecast generation gen2 and gen3 and they both reject it. canDisplayType will return true or false depending on the settings made in the Google Home app. When I have the server actually send ac-3, it throws cast.player.api.ErrorCode.MANIFEST/411, aborts and does not request anything else from the server.
I also cannot seem to get any detailed reason for this, it would be helpful if I was told in the logs why the codec presumably got rejected.
Here's a typical manifest.m3u8 file that I have experimented with:
#EXTM3U
#EXT-X-STREAM-INF:BANDWIDTH=5499178,AVERAGE-BANDWIDTH=5499178,CODECS="avc1.640028,mp4a.a5",RESOLUTION=720x576,FRAME-RATE=50
main.m3u8?<snip>&VideoCodec=h264&AudioCodec=ac3,aac,mp3&<snip>&TranscodingMaxAudioChannels=6&<snip>&h264-profile=high,main,baseline,constrainedbaseline&h264-level=41&<snip>
From my experimentation, mp4a.a5, mp4a.A5 and ac-3 all pass the canDisplayType test but fail at this stage.
For instance:
> context.canDisplayType("video/mp4", "avc1.640028,mp4a.a5", 1920, 1080, 50)
< true
I have also tried to trick the player by substituting the value with mp4a.40.2 or just pretending like there is no audio. In this case the results vary a bit but generally it requests some .ts files before it gives up, or it does not play an audio track at all. In comparison, the same media transcoded using aac-lc 2ch works fine.
Is there anything special one has to do to prepare the receiver to use passthrough audio?
i have an RTSP server which is re-streaming the A/V stream from the camera to clients.
Client side we are using MF to play the stream.
I can successfully play the video but not able to play the audio from the server. However when i use vlc to play, it can play both A/V.
Currently i am implementing IMFMediaStream and have created my customize media stream. I have also created a separate IMFStreamDescriptor for audio and added all the required attributes. When i run , everything goes fine but my RequestSample method never gets called.
Please let me know if i am doing it wrong or if there is any other way to play the audio in MF.
Thanks,
Prateek
Media Foundation support for RTSP is limited to a small number of payload formats. VLC supports more (AFAIR through Live555 library). Most likely, your payload is not supported in Media Foundation.
I have a server (linux) program that generates audio files (mp3). What
I need is to broadcast these files using http stream. The tricky part
is that the broadcast starts when the file to be transmitted is not
fully generated.
I tried to do this using mpd+mpc but once I use the "mpc play" command
only already existing part of the file is buffered and transmitted,
and the player disregards the part that appears after beginning of
playback.
Is there any way to send a mp3 http stream (using mpd or any other
server-side player) so that the player won't stop the playback as it
reaches the end of the part that was buffered initially?
Any ideas, please.
http://streamripper.sourceforge.net/ can record and broadcast the same stream
shotcast(or icecast, dont remember) was designed especially for this, and could re-encode your stream on the fly
I'm using MediaElement to play audio mp3 stream,
everything goes ok, but now I have mp3 stream that does not end with .mp3,
( http://server2.fmstreams.com:8011/spin103) and I'm getting
AG_E_NETWORK_ERROR
I found solution to add ?ext=mp3, but it didn't work for me, any ideas?
If you are streaming live radio, the stream may be encoded by an IceCast server or ShoutCast server. To read these streams, you will need to decode the stream in memomry and pass it to the MediaElement once it has been decoded.
have a look at Mp3MediaStreamSource : http://archive.msdn.microsoft.com/ManagedMediaHelpers
and Audio output from Silverlight
I lost tons of time on this, and this is the best solution I found so far.
You also have to be sure that while you are testing, the device must be unplugged from the computer.