LLHLS stream does not work when origin is in a push stream from the same server since v0.15.16

Describe the bug My usage is a bit particular. I have a "source" application and an "app" application. I send stream to the source receives (srt stream with 3 audio tracks or a rtmp stream) and once this stream is received, we send this stream to the app application via the "startPush" API:

let res = await this.ome.startPushPublishing('default', 'source', {
     id: 'xxxx',
     stream: {
         name: 'yyyy',
         variantNames: ['bypass_video', 'bypass_audio']
     },
     protocol: 'srt',
     url: 'srt://127.0.0.1:9999?streamid=' + encodeURI(streamId)
})

I have also try to send stream with rtmp but there is the same issue

The response does not show error:

{ 
    app: 'source',
    createdTime: '2023-11-29T16:18:40.444+00:00',
    finishTime: '1970-01-01T00:00:00.000+00:00',
    id: 'xxxx',
    protocol: 'srt',
    sentBytes: 0,
    sentTime: 0,
    sequence: 0,
    startTime: '1970-01-01T00:00:00.000+00:00',
    state: 'ready',
    stream: { 
                name: 'yyyy',
        trackIds: [],
        variantNames: [ 'bypass_video', 'bypass_audio' ] 
        },
    totalsentBytes: 0,
    totalsentTime: 0,
    url: 'srt://127.0.0.1:9999?streamid=srt://127.0.0.1:9999/app/xxxx',
    vhost: 'default' 
}

The stream is created on the app side, the llhls.m3u8 file is available but I have these errors appearing: [StreamWorker:123] LLHLS Publisher | llhls_session.cpp:327 | Invalid chunklist file name requested: chunklist_0_video_xxxx_llhls.m3u8 On the other hand webrtc stream is OK. When I send the stream directly to the application "app" from obs, I don't have this issue, the LLHLS stream can be played.

When I'm using v0.15.15, there is no error and I can access to the LLHLS stream on "app" application

A part of my server.xml:

<Applications>
    <Application>
        <Name>source</Name>
        <!-- Application type (live/vod) -->
        <Type>live</Type>
        <OutputProfiles>
            <OutputProfile>
                <Name>bypass_stream</Name>
                <OutputStreamName>${OriginStreamName}</OutputStreamName>    
                <Playlist>
                    <Name>Webrtc</Name>
                    <FileName>webrtc</FileName>
                    <Options>
                        <WebRtcAutoAbr>true</WebRtcAutoAbr>
                    </Options>                                  
                    <!--<Rendition>
                        <Name>360p</Name>
                        <Video>360p</Video>
                        <Audio>opus</Audio>
                    </Rendition>-->
                    <Rendition>
                        <Name>bypass</Name>
                        <Video>bypass_video</Video>
                        <Audio>opus</Audio>
                    </Rendition>
                </Playlist>
                <Encodes>                                                               
                    <Video>
                        <Name>bypass_video</Name>
                        <Bypass>true</Bypass>
                    </Video>
                    <Audio>
                        <Name>opus</Name>
                        <Codec>opus</Codec>
                        <Bitrate>128000</Bitrate>
                        <Samplerate>48000</Samplerate>
                        <Channel>2</Channel>
                        <BypassIfMatch>
                        <Codec>eq</Codec>
                        </BypassIfMatch>
                    </Audio>                                
                    <Audio>
                        <Name>bypass_audio</Name>
                        <Bypass>true</Bypass>
                    </Audio>                                
                </Encodes>                          
            </OutputProfile>
        </OutputProfiles>
        <Providers>
            <OVT />
            <WebRTC>
                <Timeout>30000</Timeout>
                <CrossDomains>
                    <Url>*</Url>
                </CrossDomains>
            </WebRTC>
            <RTMP />
            <SRT>
                <AudioMap>
                    <Item>
                        <Name>English</Name>
                        <Language>en</Language> <!-- RFC 5646 -->
                    </Item>
                    <Item>
                        <Name>Korean</Name>
                        <Language>ko</Language> <!-- RFC 5646 -->
                    </Item>
                    <Item>
                        <Name>Japanese</Name>
                        <Language>ja</Language> <!-- RFC 5646 -->
                    </Item>
                </AudioMap>
            </SRT>
            <RTSPPull />
        </Providers>
        <Publishers>
            <AppWorkerCount>1</AppWorkerCount>
            <StreamWorkerCount>1</StreamWorkerCount>
            <OVT />
            <WebRTC>
                <Timeout>30000</Timeout>
                <Rtx>false</Rtx>
                <Ulpfec>false</Ulpfec>
                <JitterBuffer>false</JitterBuffer>
            </WebRTC>
            <LLHLS>
                <OriginMode>true</OriginMode>
                <ChunkDuration>0.5</ChunkDuration>
                <!-- PartHoldBack SHOULD be at least three times the Part Target Duration -->
                <PartHoldBack>1.5</PartHoldBack>
                <SegmentDuration>6</SegmentDuration>
                <SegmentCount>10</SegmentCount>
                <CrossDomains>
                    <Url>*</Url>
                </CrossDomains>
            </LLHLS>
            <RTMPPush>
            </RTMPPush>
            <MPEGTSPush>
            </MPEGTSPush>
            <SRTPush>
            </SRTPush>

            <FILE>
                <RootPath>/opt/ovenmediaengine/bin/dumps/recordings/files</RootPath>
                <FilePath>/${VirtualHost}/${Application}/${Stream}/${StartTime:YYYYMMDDhhmmss}_${EndTime:YYYYMMDDhhmmss}.mp4</FilePath>
                <InfoPath>/${VirtualHost}/${Application}/${Stream}.xml</InfoPath>
            </FILE> 

        </Publishers>
    </Application>
    <Application>
        <Name>app</Name>
        <!-- Application type (live/vod) -->
        <Type>live</Type>
        <OutputProfiles>
            <OutputProfile>
                <Name>bypass_stream</Name>
                <OutputStreamName>${OriginStreamName}</OutputStreamName>
                <Playlist>
                    <Name>ll-HLS</Name>
                    <FileName>llhls</FileName>
                    <Rendition>
                        <Name>Origin</Name>
                        <Video>bypass_video</Video>
                        <Audio>aac</Audio>
                    </Rendition>
                    <Rendition>
                        <Name>480p</Name>
                        <Video>480p</Video>
                        <Audio>aac</Audio>
                    </Rendition>
                </Playlist>
                <Playlist>
                    <Name>Webrtc</Name>
                    <FileName>webrtc</FileName>
                    <Options>
                        <WebRtcAutoAbr>true</WebRtcAutoAbr>
                    </Options>                              
                    <Rendition>
                        <Name>720p bypass</Name>
                        <Video>bypass_video</Video>
                        <Audio>opus</Audio>
                    </Rendition>                                
                </Playlist>
                <Encodes>
                    <Video>
                        <Name>bypass_video</Name>
                        <Bypass>true</Bypass>
                    </Video>
                    <Audio>
                        <Name>bypass_audio</Name>
                        <Bypass>true</Bypass>
                    </Audio>    
                    <Video>
                        <Name>480p</Name>
                        <Codec>h264</Codec>
                        <Bitrate>1000000</Bitrate>
                        <Framerate>30</Framerate>
                        <Width>854</Width>
                        <Height>480</Height>
                        <Preset>faster</Preset>
                    </Video>                                                                
                    <Audio>
                        <Name>aac</Name>
                        <Codec>aac</Codec>
                        <Bitrate>128000</Bitrate>
                        <Samplerate>48000</Samplerate>
                        <Channel>2</Channel>
                        <BypassIfMatch>
                            <Codec>eq</Codec>
                        </BypassIfMatch>
                    </Audio>
                    <Audio>
                        <Name>opus</Name>
                        <Codec>opus</Codec>
                        <Bitrate>128000</Bitrate>
                        <Samplerate>48000</Samplerate>
                        <Channel>2</Channel>
                        <BypassIfMatch>
                            <Codec>eq</Codec>
                        </BypassIfMatch>
                    </Audio>
                </Encodes>
            </OutputProfile>
        </OutputProfiles>
        <Providers>
            <OVT />
            <WebRTC>
                <Timeout>30000</Timeout>
                <CrossDomains>
                    <Url>*</Url>
                </CrossDomains>
            </WebRTC>
            <RTMP />
            <SRT>
                <AudioMap>
                    <Item>
                        <Name>English</Name>
                        <Language>en</Language> <!-- RFC 5646 -->
                    </Item>
                    <Item>
                        <Name>Korean</Name>
                        <Language>ko</Language> <!-- RFC 5646 -->
                    </Item>
                    <Item>
                        <Name>Japanese</Name>
                        <Language>ja</Language> <!-- RFC 5646 -->
                    </Item>
                </AudioMap>
            </SRT>
            <RTSPPull />
        </Providers>
        <Publishers>
            <AppWorkerCount>1</AppWorkerCount>
            <StreamWorkerCount>8</StreamWorkerCount>
            <OVT />
            <WebRTC>
                <Timeout>30000</Timeout>
                <Rtx>false</Rtx>
                <Ulpfec>false</Ulpfec>
                <JitterBuffer>false</JitterBuffer>
            </WebRTC>
            <LLHLS>
                <OriginMode>true</OriginMode>
                <Dumps>
                    <Dump>
                        <Enable>false</Enable>
                        <TargetStreamName>*</TargetStreamName>
                        <Playlists>
                            <Playlist>llhls.m3u8</Playlist>
                        </Playlists>
                        <OutputPath>/opt/ovenmediaengine/bin/dumps/recordings/hls/${VHostName}_${AppName}_${StreamName}/${YYYY}_${MM}_${DD}_${hh}_${mm}_${ss}</OutputPath>
                    </Dump>
                </Dumps>
                <DVR>
                    <Enable>true</Enable>
                    <TempStoragePath>/opt/ovenmediaengine/bin/dumps/ome_dvr/</TempStoragePath>
                    <MaxDuration>36000</MaxDuration>
                </DVR>                          
                <ChunkDuration>0.5</ChunkDuration>
                <!-- PartHoldBack SHOULD be at least three times the Part Target Duration -->
                <PartHoldBack>1.5</PartHoldBack>
                <SegmentDuration>6</SegmentDuration>
                <SegmentCount>10</SegmentCount>
                <CrossDomains>
                    <Url>*</Url>
                </CrossDomains>
            </LLHLS>
            <RTMPPush>
            </RTMPPush>
            <MPEGTSPush>
            </MPEGTSPush>
            <SRTPush>
            </SRTPush>
        </Publishers>
    </Application>
</Applications>

AirenSoft / OvenMediaEngine

LLHLS stream does not work when origin is in a push stream from the same server since v0.15.16 #1464