Update:
I have a video player in browser which plays mp4 videos though websocket. The player only supports mp4 file. When i checked normal mp4 fiels does not play in the player, a mp4 file with a "moovflags faststart " will only play on that player. For a allready stored file , this will work properly.
But In case of an livestream(RTSP), using ffmpeg will only work once the RTSP connection has terminated since the "moovflags faststart " flags will work once a connection has terminated properly.
Hope the above statements makes more sense.
Due to this behavior, am checking if there is any way to get the moovflasg at first or something
I am having RTSP live source and i need it to convert the RTSP to a mp4 file which has moov flags in the begining of the file.
I have checked with openrtsp to take a mp4 dump of the rtsp, but it only adds moov flags and other info on the footer of the mp4(onlky when openrtsp has closes the rtsp stream).
Ffmpeg has " -movflags faststart" to move the footer info to the header of the mp4 container.
Since i am having a RTSP live source, the video data will be comming back to back and there wont be any termination. The above ffmpeg command only works once the rtsp stream has terminated.
Is there any way we can make a mp4 container which contains the mp4 footer info present in the header itself so that i can use it for a live source?
EDIT #1
I have video player which plays mp4 video files , it only support playback of a recorded mp4 file which is createtd using "-movflags faststart" , normal mp4 files does not play in that.
This is the player
https://github.com/sonysuqin/WasmVideoPlayer.
Since i am tryng to stream live video to the player, its not possible to use movflags faststart.
The mp4 header can not be added to the file before it is complete. It’s not possible because of how mp4 files are structured. The header needs to know the frame type, timestamp, size, and file offset of every frame in the file. That can’t be known until the file is complete. You can not stream an mp4 while it is being created. You need to use a protocol such as HLS or DASH to accomplish this.
Related
I am able to upload the AMR file to SIM800C successfully.
When I play the uploaded audio file during the call using the below command :
#if CALL_RECORDED_AUDIO
Serial1.print("AT+CMEDPLAY=1,C:\\REC\\");
// "Command Media PLAY" -> to play an audio if it is a recorded audio
#else
Serial1.print("AT+CMEDPLAY=1,C:\\User\\");
// "Command Media PLAY" -> to play an audio if it is a uploaded audio
#endif
Played audio always has noise, from C:\User\.
However if I record the audio during call and save it. Play the recorded audio during next call then there is no noise. ( By defining CALL_RECORDED_AUDIO in above code snippet)
according to the documentation of the sim800 it is necessary to play a sound wav during the call
Note
. mode 2 and 3 are not supported when playing audio file during call.
. The audio file can not be played duirng incoming call or outgoing call.
. Only support WAV, PCM, AMR and MP3 format.
. Only support WAV format with 8K 16bit during call.
page 201/202 of the sim800 guide
personally i did not suck having no sim800
I think the recording of a call must be in .WAV format
let me know if it works
The thing which I am doing right now is that I am playing RTMP streaming on media server using ffmpeg command and also creating an audio file using google text to speech.
So I want to update mp3 file with silence if there is no content, so that it will keep will keep stream running.
I have tried 2 approaches:
By writing raw binary data to mp3 file but not working as it says content is not accurate.
Concatenate the audio content with the silence data and export file. In this scenario, I am able to update file but stream broken at the point while we are exporting file.
I have tried to write the audio file with binary data and also tried to concatenate audio content with silence and then export file but it break stream while we export the file.
I try to stream my video file using VLC player. I choose http transfer protocol and MP4 encoder (H.264 + MP3(MP4)). And automatically I get next command line presets:
:sout=#transcode{vcodec=h264,acodec=mpga,ab=128,channels=2,samplerate=44100}:http{mux=ffmpeg{mux=flv},dst=:8080/} :sout-all :sout-keep
Streaming is works great but no audio sound.
I launched it on my PC localhost and local networks on Windows, and I have no
results.If I change encoder to H.264 + MP3 TS:
:sout=#transcode{vcodec=h264,vb=800,acodec=mpga,ab=128,channels=2,samplerate=44100}:http{mux=ts,dst=:9000/}
If I change transfer protocol to RTSP(or RTP), the sound starts to play with any types of encoders. F.e:
:sout=#transcode{vcodec=h264,scale=auto,acodec=mpga,ab=128,channels=2,samplerate=44100}:rtp{sdp=rtsp://:9000/test} :sout-all :sout-keep
Why sound does't play whith encoder (H.264 + MP3(MP4))?
Streaming is works great but no audio sound...
Try using acodec=mp3 or acodec=aac since they're supported formats for FLV containers.
example:
:sout=#transcode{vcodec=h264,acodec=mp3,ab=128,channels=2,samplerate=44100}:http{mux=ffmpeg{mux=flv},dst=:8080/} :sout-all :sout-keep
Here is the input manifest:
$ curl 'https://example.net/ipadlive/index_new.m3u8?sessionid=81893121496608402793&ipaddress=x.x.x.x&callsign=YYYY&hubid=51&zipcode='
#EXTM3U
#EXT-X-VERSION:4
#EXT-X-MEDIA:TYPE=AUDIO,GROUP-ID="group",NAME="eng",DEFAULT=YES,AUTOSELECT=YES,LANGUAGE="en",URI="https://example.net/ipadlive/06_new.m3u8?cdnHost=da148.cdn.iptv.example.net&sessionid=81893121496608402793&ipaddress=x.x.x.x&callsign=CHAN&hubid=51&zipcode=&countycode=null&fta=null&optimumid=null&devicename=&devicetype=0&osver=&res=&fps="
#EXT-X-MEDIA:TYPE=AUDIO,GROUP-ID="group",NAME="spa",DEFAULT=NO,AUTOSELECT=YES,LANGUAGE="en",URI="https://example.net/ipadlive/07_new.m3u8?cdnHost=da148.cdn.iptv.example.net&sessionid=81893121496608402793&ipaddress=x.x.x.x&callsign=CHAN&hubid=51&zipcode=&countycode=null&fta=null&optimumid=null&devicename=&devicetype=0&osver=&res=&fps="
#EXT-X-STREAM-INF:PROGRAM-ID=1,BANDWIDTH=479776,RESOLUTION=240x180,CODECS="avc1.42c00c,mp4a.40.2",AUDIO="group"
https://example.net/ipadlive/01_new.m3u8?cdnHost=da148.cdn.iptv.example.net&sessionid=81893121496608402793&ipaddress=x.x.x.x&callsign=CHAN&hubid=51&zipcode=&countycode=null&fta=null&optimumid=null&devicename=&devicetype=0&osver=&res=&fps=
#EXT-X-STREAM-INF:PROGRAM-ID=1,BANDWIDTH=780576,RESOLUTION=320x240,CODECS="avc1.42c00d,mp4a.40.2",AUDIO="group"
https://example.net/ipadlive/02_new.m3u8?cdnHost=da148.cdn.iptv.example.net&sessionid=81893121496608402793&ipaddress=x.x.x.x&callsign=CHAN&hubid=51&zipcode=&countycode=null&fta=null&optimumid=null&devicename=&devicetype=0&osver=&res=&fps=
#EXT-X-STREAM-INF:PROGRAM-ID=1,BANDWIDTH=1079872,RESOLUTION=480x360,CODECS="avc1.42c01e,mp4a.40.2",AUDIO="group"
https://example.net/ipadlive/03_new.m3u8?cdnHost=da148.cdn.iptv.example.net&sessionid=81893121496608402793&ipaddress=x.x.x.x&callsign=CHAN&hubid=51&zipcode=&countycode=null&fta=null&optimumid=null&devicename=&devicetype=0&osver=&res=&fps=
#EXT-X-STREAM-INF:PROGRAM-ID=1,BANDWIDTH=1682976,RESOLUTION=640x480,CODECS="avc1.42c01e,mp4a.40.2",AUDIO="group"
https://example.net/ipadlive/04_new.m3u8?cdnHost=da148.cdn.iptv.example.net&sessionid=81893121496608402793&ipaddress=x.x.x.x&callsign=CHAN&hubid=51&zipcode=&countycode=null&fta=null&optimumid=null&devicename=&devicetype=0&osver=&res=&fps=
I've never seen this before where the audio stream is a separate url than a video stream listed in the manifest.
Is there a way I can combine an audio stream and a specific video stream to produce a new stream that has both audio and video in it?
I was doing something like this:
ffmpeg -i <manifest> -c copy test.m3u8 and I don't get any audio.
I've tried changing <manifest> to an individual video stream, but then no audio. If I change it to an AUDIO stream I get no video.
I recently had the problem of combining an audio .ts file with its accompanying video .ts file. I was able to solve it using the following method for Windows users. [see - Video resource ]
1) You will need to download the ffmpeg library that will allow Windows to combine both files together. In my case I was running Windows 8 (32 bit OS) and chose a static build:
2) I then opened notepad and wrote the following code once ffmpeg was installed:
ffmpeg -i VIDEO.ts -i AUDIO.ts -c:v copy -c:a copy OUTPUT.mp4
I saved the notepad file as "joiner.bat"
NB: this bat file must present in the same folder as your separate audio and video ts files in order to combine them!!!
3) Once the bat file is in the same folder as your audio and video ts files you can double click on the joiner.bat file to combine the audio and video ts files into a single mp4 (OUTPUT.mp4) file.
I hope this helps the more novice types among us. Yes I'm still a n00b after many years - don't worry! ;)
I am new to ffmpeg, i have spend more than 10 days on finding any way to do muxing in mp4 format with audio and vedio in streams buffer not in file.
What i want is to mux mp4 format audio & vedio in a streams.
I am able to do muxing mp4 format in file. But not able to get mux mp4 format in streams buffer.
Till now i have tried this:
avio_alloc_context(avio_ctx_buffer, avio_ctx_buffer_size, 1, &bd, NULL, &write_packet, NULL);
By calling this avio_alloc_context and passing reference of write_packet function. I am able to get call write_packet. But when i write the data coming in write_packet in a file, and making mp4 file with that. The resultant mp4 file is not working. There is no vedio or audio information available by watching this file in Media Info.
The header is written , then loop is written and finally trailer is written, but not getting success in running final file.
Is there any good way to do mux in mp4 format in streams, so please tell me.
Kindly help to me to do this.
Thanks in advance.