[Libav-user] ffmpeg Question
I'm not sure if this is the right place to ask. If not could you please tell me where to ask? My company is searching for a replacement for VKC Media Player. Can you tell me if your app meets the following requirements? Operation Used For Supporting both IPv4 and IPv6 We need player to support cameras over both IPv4 and IPv6 network protocols Integration to WPF Player should be able to easily integrate into our current C# WPF/WinForms applications Sixteen Cameras We need to support displaying upto 16 cameras while keeping optimal performance Taking Snapshots We use the option to take snapshots while displaying Audio selection in Live Streaming Support audio and video codecs ( H.264 / H.265 and AAC / MP3 ) Required for playing with live streaming and playback mode RTSP and HLS Stream Playing RTSP and HLS stream in Client and Live Streaming Playing DVR Data Playing for DVR data in Live Streaming, we should be able to jump back and forth with ease while watching We also need accurate duration/time of the DVR data (which increases every X seconds while the live stream is ongoing) Start/Stop Be able to start and stop playing the streams Debug Logs We have been relying on VLC logs for debugging issues from time to time Custom Parameters We change player options for controlling the behaviour of player such as changing Network caching which helps in reducing delay from camera to player Custom Menu for player We display custom right click menu on player Set volume We change volume level for players and mute audio as needed Disable Keyboard/Mouse input We disable the player for taking input from Keyboard and Mouse on player Custom Menu for player We display custom right click menu on player Player Events Click, DoubleClick, Mouse, Down Mediaplayer Events Opening, EncounteredError, Buffering, LengthChanged, TimeChanged, PositionChanged, EndReached Track/Stream Info Allow to get track/stream info to get details about track Set Crop Rectangle Used for digital zoom in live streaming Select channel to play the audio (e.g., Left vs Right) Nice to have: Ability to select which channel for audio Thank you, Kevin Marois kmar...@axon.com ___ Libav-user mailing list Libav-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/libav-user To unsubscribe, visit link above, or email libav-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [Libav-user] FFMpeg Question about paket pts, dts and duration parameters
*./ffmpeg -i rtsp://192.168.1.11:554/onvif1 test.mpg* it works, with this output: Input #0, rtsp, from 'rtsp://192.168.1.11:554/onvif1': Metadata: title : H.264 Video, RtspServer_0.0.0.2 Duration: N/A, start: 0.00, bitrate: N/A Stream #0:0: Video: h264 (Baseline), yuv420p(progressive), 1280x720, 24.92 tbr, 90k tbn, 180k tbc Stream #0:1: Audio: pcm_alaw, 8000 Hz, mono, s16, 64 kb/s Stream mapping: Stream #0:0 -> #0:0 (h264 (native) -> mpeg1video (native)) Stream #0:1 -> #0:1 (pcm_alaw (native) -> mp2 (native)) Press [q] to stop, [?] for help [mpeg @ 0x55c6280f5c00] VBV buffer size not set, using default size of 230KB If you want the mpeg file to be compliant to some specification Like DVD, VCD or others, make sure you set the correct buffer size Output #0, mpeg, to 'test.mpg': Metadata: title : H.264 Video, RtspServer_0.0.0.2 encoder : Lavf58.29.100 Stream #0:0: Video: mpeg1video, yuv420p, 1280x720, q=2-31, 200 kb/s, 25 fps, 90k tbn, 25 tbc Metadata: encoder : Lavc58.54.100 mpeg1video Side data: cpb: bitrate max/min/avg: 0/0/20 buffer size: 0 vbv_delay: -1 Stream #0:1: Audio: mp2, 16000 Hz, mono, s16, 160 kb/s Metadata: encoder : Lavc58.54.100 mp2 frame= 789 fps= 28 q=31.0 Lsize=2382kB time=00:00:32.65 bitrate= 597.5kbits/s dup=635 drop=0 speed=1.18x In data martedì 28 aprile 2020 18:14:45 CEST, Carl Eugen Hoyos ha scritto: > Am Di., 28. Apr. 2020 um 16:57 Uhr schrieb Denis Gottardello > > : > > Hi, after having implemented the remuxing.c example my program works with > > file to file but not with rtsp camera to file. > Does it work with ffmpeg (the application)? > > Does the rtsp demuxer returns any useful timestamps at all for video > if you read more packets but if you do not try to write anything? > > Carl Eugen > ___ > Libav-user mailing list > Libav-user@ffmpeg.org > https://ffmpeg.org/mailman/listinfo/libav-user > > To unsubscribe, visit link above, or email > libav-user-requ...@ffmpeg.org with subject "unsubscribe".-- +39.347.4070897 http://www.labcsp.com[1] http://www.denisgottardello.it[2] GMT+1 Skype: mrdebug [1] http://www.labcsp.com [2] http://www.denisgottardello.it ___ Libav-user mailing list Libav-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/libav-user To unsubscribe, visit link above, or email libav-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [Libav-user] FFMpeg Question about paket pts, dts and duration parameters
Am Di., 28. Apr. 2020 um 16:57 Uhr schrieb Denis Gottardello : > > > > > > Hi, after having implemented the remuxing.c example my program works with > file to file but not with rtsp camera to file. Does it work with ffmpeg (the application)? Does the rtsp demuxer returns any useful timestamps at all for video if you read more packets but if you do not try to write anything? Carl Eugen ___ Libav-user mailing list Libav-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/libav-user To unsubscribe, visit link above, or email libav-user-requ...@ffmpeg.org with subject "unsubscribe".
[Libav-user] FFMpeg Question about paket pts, dts and duration parameters
Hi, after having implemented the remuxing.c example my program works with file to file but not with rtsp camera to file. I think that the problem is related to the following lines of code: /Packet.pts= av_rescale_q_rnd(Packet.pts, pAVStreamIn->time_base, pAVStreamOut->time_base, static_cast(AV_ROUND_NEAR_INF | AV_ROUND_PASS_MINMAX));/ /Packet.dts= av_rescale_q_rnd(Packet.dts, pAVStreamIn->time_base, pAVStreamOut->time_base, static_cast(AV_ROUND_NEAR_INF | AV_ROUND_PASS_MINMAX));/ /Packet.duration= av_rescale_q(Packet.duration, pAVStreamIn->time_base, pAVStreamOut->time_base);/ The code works with file to file but not using an rtsp camera as source to file. I obtain an error at av_interleaved_write_frame function. Please have a look at the following log report: AudioIndex Packet.pts: 0, Packet.dts: 0, Packet.duration: 160 AudioIndex Packet.pts: 120, Packet.dts: 120, Packet.duration: 160 AudioIndex Packet.pts: 312, Packet.dts: 312, Packet.duration: 160 AudioIndex Packet.pts: 432, Packet.dts: 432, Packet.duration: 160 AudioIndex Packet.pts: 632, Packet.dts: 632, Packet.duration: 160 AudioIndex Packet.pts: 752, Packet.dts: 752, Packet.duration: 160 AudioIndex Packet.pts: 952, Packet.dts: 952, Packet.duration: 160 AudioIndex Packet.pts: 1072, Packet.dts: 1072, Packet.duration: 160 AudioIndex Packet.pts: 1272, Packet.dts: 1272, Packet.duration: 160 AudioIndex Packet.pts: 1392, Packet.dts: 1392, Packet.duration: 160 AudioIndex Packet.pts: 1592, Packet.dts: 1592, Packet.duration: 160 AudioIndex Packet.pts: 1712, Packet.dts: 1712, Packet.duration: 160 *VideoIndex Packet.pts: 9223372036854775808, Packet.dts: 9223372036854775808, Packet.duration: 0* As you can see the pts, dts and duration parameters relative to audio stream is ok, the video parameters abolutally not. How can I do to rewrite the video parameters? Can anyone explain? How can I calculate pts and dts using current time stamp? Have you got a example for me? I have tryed in this way but does not work *Packet.pts= Packet.dts= VideoPacketNumber++;* Many thanks. +39.347.4070897 http://www.labcsp.com[1] http://www.denisgottardello.it[2] GMT+1 Skype: mrdebug [1] http://www.labcsp.com [2] http://www.denisgottardello.it ___ Libav-user mailing list Libav-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/libav-user To unsubscribe, visit link above, or email libav-user-requ...@ffmpeg.org with subject "unsubscribe".
