[Ffmpeg-devel-irc] ffmpeg.log.20130623

[burek]

[00:22] <vulture> amoxibos: have you checked out coded_frame ?
[00:22] <vulture> http://ffmpeg.org/doxygen/trunk/structAVFrame.html
[00:23] <vulture> attribute_deprecated short * 	dct_coeff
[00:23] <vulture> says deprecated tho idk
[00:23] <vulture> you'd get that from the encoder context ->coded_frame
[00:24] <vulture> (not sure if decoder context fills it too but I'd guess it would)
[02:32] <xreal> is there a channel for rtmpdump/lib ?
[02:33] <durandal_1707> why? native rtmp* support is not working?
[02:37] <dagerik> guys. help me extract this video here; http://www.vgtv.no/#!/video/65687/adelen-bombo
[02:37] <dagerik> i need to play it using omxplayer
[02:37] <dagerik> my raspberry pi cant handle flash
[02:51] <klaxa> dagerik: this is not a general "my problem is video related"-help channel
[04:47] <apapousek> Hello all, I have question about audio and ffmpeg.
[04:48] <apapousek> I'm dumping my BluRay to mkv, then using ffmpeg to change the container to mp4
[04:48] <apapousek> I'm using: ffmpeg -i input.mkv -acodec copy -vcodec copy output.mp4
[04:48] <apapousek> Is it possible to add another audio stream (a stereo downmix) within the same file?
[06:41] <silv3r_m00n> what is the difference between the extension .avi, .mpg and the codec and format ?
[06:42] <vulture> .avi and .mpg are containers
[06:42] <vulture> within them are multiple/different streams of data
[06:43] <vulture> for mpg there are different "layers" or codec schemes
[06:43] <vulture> mpeg layer 3 for example, is for audio
[06:43] <silv3r_m00n> container means ? if the data inside is in different format, then why name all of them as .mpg ?
[06:43] <vulture> .mpg is pretty limited in which codecs it can contain, whereas .avi supports any/all codecs
[06:43] <silv3r_m00n> it should be something like .codec
[06:44] <vulture> because .mpg is a specific container format
[06:44] <vulture> and .avi is a different, but specific container format
[06:44] <vulture> each codec has their own custom format
[06:44] <silv3r_m00n> so what does it mean to have codec A inside .avi ?
[06:45] <silv3r_m00n> the video is in codec A format i guess
[06:45] <silv3r_m00n> and cannot be played unless the system has the decoder for it
[06:46] <vulture> right
[06:47] <apapousek> Question: Does the MPEG-4 support Dolby TrueHD audio?
[06:47] <apapousek> MPEG-4 container*
[06:48] <vulture> no idea
[06:48] <apapousek> Alrighty. Any idea is it supports DTS?
[06:49] <vulture> as an audiophile I only care about one format, 2 channel stereo :D
[06:49] <vulture> dont know anything about what it supports
[06:50] <silv3r_m00n> vulture: i see this, http://www.pitivi.org/manual/codecscontainers.html
[06:50] <silv3r_m00n> means there are 2 things, container format and codec.... can any format be used with any codec ?
[06:50] <apapousek> Alright, thanks. As a pretend audiophile, I care about quality as well. That's why I was wondering about TrueHD
[06:50] <silv3r_m00n> or each format works with only specific codecs
[06:52] <vulture> pretty much any audio format beyond 2 channel stereo is just flashy selling points. it can be 'useful' if you're trying to emulate some kind of 3D environment in a large room but in the end it gets downsamples to 2 channel stereo in your ears. and I always use headphones anyway :D
[06:52] <vulture> so yeah sorry that's why I dont know, because I just havent cared enough to learn about that
[06:53] <vulture> silv3r_m00n: containers might only support certain codecs
[06:53] <vulture> for example .mpg containers as far as I know only support mpeg-based codecs
[06:53] <vulture> but .avi can contain any mpeg codec, as well as a whole lot more
[06:56] <sacarasc> silv3r_m00n: A container is a bit like a box. You put stuff in it.
[07:53] <knIOO> can someone give me a brief rundown on setsar/setdar and how they are actually used in the command line? I'm getting mismatched sar values when concating two videos together and am attempting to correct it
[07:55] <knIOO> I'm getting [Parsed_concat_0 @ 0245e740] Failed to configure output pad on Parsed_concat_0 - full output is here http://pastebin.com/Tzqpinjf
[07:56] <ilove11ven> knIOO:
[07:57] <knIOO> ilove11ven: hi
[14:07] <illusion> how to stream a playlist with ffmpeg ?
[14:49] <illusion_> help me please
[14:59] <spaam> what? stream a playlist?
[15:04] <KuroiTsuki> hello, I compiled latest mplayer/meconder on mingw64 and when I try convert anything with scale (which as far as i understand uses libswscale) mencoder gives me next Assertion firstChrSrcY >= lastInChrBuf - vChrBufSize + 1 failed at libswscale/swscale.c:484. Any ideas what can be causing this?
[15:56] <illusion_> spaam, yes to stream a playlist with ffmpeg
[17:20] <bencc> burek: now I'm using real ffmpeg and I still can't get live mp3 stream from a rtmp stream
[18:11] <bencc> burek: ok.
[18:12] <burek> illusion_, im not sure if ffmpeg can handle playlists (yet), but i know vlc can
[18:12] <burek> so try with vlc
[18:14] <illusion_> can you give me an example to stream a playlist from cvlc to an rtmp server please :)
[18:14] <illusion_> burek,
[18:15] <burek> cvlc playlist.m3u --sout ...
[18:15] <burek> something like that
[18:18] <burek> cvlc --sout='#transcode{vcodec=h264,acodec=mp4a,ab=32,samplerate=44100,channels=2}:std{access=rtmp,mux=rtp,dst=1.2.3.4/something}'
[18:18] <burek> try reading their wiki, there are some examples how to do that
[18:20] <burek> http://www.jpsaman.org/vlc/rtmp
[18:26] <bencc> burek: this is the command and output I'm getting http://dpaste.com/1265189/
[18:27] <bencc> I'm trying to get live stream as mp3 to stdout
[18:29] <burek> well, everything works ok
[18:29] <burek> no errors
[18:29] <bencc> burek: I have a live rtmp stream which I want to convert to live mp3 stream
[18:29] <klaxa> bencc, you should use pipe: instead of -
[18:29] <bencc> and serve it to web clients
[18:29] <bencc> testing with pipe:
[18:30] <bencc> klaxa: I don't see any output in the terminal with pip:1 (except for log messages)
[18:30] <burek> that's also normal
[18:31] <burek> what exactly is your issue bencc?
[18:31] <bencc> I don't understand how to get a live stream of mp3 packets from ffmpeg
[18:31] <burek> did you try ./ffmpeg -loglevel debug -i rtmp://127.0.0.1/audio/test -f mp3 out.mp3
[18:31] <bencc> yes. it works
[18:32] <burek> well, what's the problem then?
[18:32] <bencc> I can play the output
[18:32] <bencc> but I want to stream it live to clients
[18:32] <bencc> not create a file
[18:32] <burek> do you know what "-" means in your ffmpeg cmd line (or pipe:1) ?
[18:32] <klaxa> if i use "ffmpeg -i /home/klaxa/Musik/\[ASL\]\ Various\ Artists\ -\ Ore\ no\ Kanojo\ to\ Osananajimi\ ga\ Shuraba\ Sugiru\ OP\ -\ Girlish\ Lover\ \[FLAC\]/01\ Girlish\ Lover.flac -c:a libmp3lame -b:a 320k pipe:" my terminal gets spammed with mp3 binary
[18:32] <klaxa> you must be doing something wrong
[18:33] <bencc> I'll try finding a public rtmp stream and show you the result
[18:35] <burek> bencc, what exactly do you do with your stream, after you pipe it?
[18:36] <bencc> burek: currently nothing
[18:36] <bencc> I first want to see it in the terminal
[18:37] <bencc> my aim is to pass it to a web server that will serve it to web clients (html5 audio tag)
[18:38] <klaxa> to warn you, that will not work with firefox on linux
[18:38] <klaxa> firefox on linux doesn't support mp3 in html5 audio tags
[18:38] <bencc> for firefox on linux I can use ogg
[18:38] <bencc> once I'll manage to make it work with ffmpeg, using other codecs will be easy
[18:51] <klaxa> bencc, i took this stream: http://www.twitch.tv/theemero
[18:51] <klaxa> got the stream with livestreamer
[18:51] <klaxa> and you can listen to it as mp3 now here: http://klaxa.eu/live_streaming.html
[18:52] <klaxa> er http://klaxa.eu/live_stream.html
[18:53] <bencc> klaxa: is it a live stream or did you just save it to a file?
[18:53] <klaxa> it's live
[18:53] <bencc> how do you stream it?
[18:53] <klaxa> if you open the stream on twich it should be a few seconds ahead
[18:54] <klaxa> livestreamer --fifo -o /tmp/stream http://www.twitch.tv/theemero best
[18:54] <klaxa> ffmpeg -i /tmp/stream -c:a libmp3lame -b:a 320k pipe: | nc klaxa.eu 8080
[18:55] <klaxa> on klaxa.eu my mp3 server is running and on live_stream.html there is a simple html5 audio tag
[18:57] <LordDoskias> is there a tutorial how to use libav to do muxing? the documentation for demuxing was particularly helpful but it is missing the tutorial style presentation for muxing ?
[18:57] <bencc> can you see the result on the terminal without sending it to klaxa.eu 8080 ?
[18:57] <klaxa> yes
[18:57] <klaxa> if i remove that pipe that is
[18:59] <bencc> what is the url of this public rtmp stream?
[18:59] <bencc> can I test it in my terminal?
[19:00] <klaxa> like i said, i'm using livestreamer
[19:00] <klaxa> it's some python code that does all the rtmp stuff
[19:00] <klaxa> it shouldn't be much different with rtmp directly though
[19:01] <bencc> I'm using rtmplite - a python rtmp server
[19:01] <klaxa> what software are you using to get your microphone to rtmp?
[19:01] <bencc> with a simple swf
[19:01] <klaxa> ok
[19:01] <klaxa> also, stop reading my mind
[19:01] <bencc> I can send you this swf if you like
[19:02] <bencc> not getting output in the terminal :(
[19:02] <klaxa> would be much appreciated
[19:02] <bencc> ok. a second
[19:02] <bencc> do you want the mxml source or the compiled swf?
[19:02] <klaxa> i don't know, what do i need to reconstruct your setup?
[19:03] <bencc> if I'll use 'rtmp://localhost/audio' and later ns.publish('test') will it be fine?
[19:04] <bencc> this is the source http://dpaste.com/1265271/
[19:06] <bencc> klaxa: I'm uploading all the files to google drive. a sec
[19:06] <klaxa> alright thanks
[19:06] <klaxa> also, again, stop reading my mind D: i was about to ask how to use it
[19:09] <bencc> https://docs.google.com/file/d/0B12AhxvnYHrAZDlmZ0k5SUI0Nk0/edit?usp=sharing
[19:09] <bencc> you can see html file, swf file and the mxml source
[19:10] <bencc> I'm using rtmplite with the command python rtmp.py -d
[19:10] <bencc> http://code.google.com/p/rtmplite/
[19:10] <bencc> I've tested with a subscriber and I can hear the rtmp stream
[19:10] <bencc> test with ffmpeg outputing to a file and it works
[19:10] <Yagger> I have an mp4 with 2 audio channels, english and german. I would like to "swap" them so the german audio is "first" and english second.
[19:10] <Yagger> Can this be done with ffmpeg?
[19:11] <klaxa> bencc: okay i started rtmp.py -d what now?
[19:13] <klaxa> Yagger: yes, you can use -map for that: http://ffmpeg.org/trac/ffmpeg/wiki/How%20to%20use%20-map%20option
[19:14] <jure> hello
[19:14] <Yagger> Thank you for the hint, I will try that.
[19:14] <klaxa> assuming that your audio streams are at positions 1 and 2 and there are no other streams apart from the video stream: ffmpeg -i my_file.mp4 -map 0:0 -map 0:2 -map 0:1 my_new_file.mp4
[19:14] <jure> is it possible to rip dvd-a to wav with ffmpeg?
[19:15] <klaxa> bencc: i have rtmplite and your subscriber directory, what do i do now?
[19:15] <klaxa> i started rtmp.py -d
[19:16] <bencc> you load the publisher.html file in your browser
[19:16] <bencc> allow mic access
[19:16] <bencc> and it starts broadcasting
[19:16] <klaxa> ah awesome
[19:16] <bencc> now you can use this stream from ffmpeg
[19:18] <klaxa> i get the stream via rtmp://localhost ?
[19:18] <bencc> rtmp://127.0.0.1/audio/test
[19:19] <bencc> klaxa: that's the input I'm using for ffmpeg
[19:19] <klaxa> mplayer2 returns: HandShake: client signature does not match!
[19:20] <bencc> klaxa: that's not good :)
[19:20] <bencc> can you try with a different rtmp server?
[19:20] <klaxa> ffmpeg fails too
[19:21] <klaxa> i don't know of any other rtmp servers
[19:21] <bencc> for me the handshake works. weird
[19:22] <klaxa> my publisher page is pretty blank too
[19:22] <klaxa> is that normal?
[19:22] <bencc> do you see a title Publisher?
[19:23] <bencc> and below it a text publisher
[19:23] <klaxa> http://klaxa.eu/publisher.png
[19:23] <bencc> yes. that's good
[19:23] <bencc> not very informative, I know
[19:23] <klaxa> i didn't have to confirm something about my microphone
[19:23] <bencc> weird
[19:23] <bencc> let me test
[19:24] <klaxa> ah because i already allowed it sometime earlier
[19:24] <bencc> ok
[19:24] <bencc> you sould see in your browser console
[19:24] <bencc> NetConnection.Connect.Success
[19:24] <bencc> NetStream.Publish.Start
[19:26] <klaxa> where do i find that in chromium? sorry, i'm a webdevelopment stupid
[19:26] <klaxa> i barely know how to operate my browser
[19:28] <bencc> Ctrl+Shift+J
[19:28] <klaxa> nothing there
[19:29] <bencc> you need to reload the page to see it
[19:29] <klaxa> already did
[19:30] <bencc> by the way, chrome have a new mic permission
[19:30] <bencc> you need to allow it both in the browser and in the flash app
[19:30] <bencc> you should see something in the console
[19:30] <bencc> either success or failure
[19:31] <klaxa> well the console is empty
[19:31] <klaxa> i allowed it in the browser and the flash app
[19:31] <klaxa> at least as far as i know
[19:32] <bencc> I don't understand why the console is empty
[19:33] <bencc> do you use a local web server or did you just dragged the html to the browser?
[19:33] <LordDoskias> where can i find information how can I mux audio video in a TS container in memory, not in a file ? so that i can then stream this data over the network?
[19:33] <bencc> maybe it's a permission issue
[19:34] <klaxa> i opened the file directly, no httpd
[19:34] <bencc> let me try it too
[19:34] <bencc> maybe that's the issue
[19:35] <bencc> can you serve the files with python?
[19:35] <bencc> python -m SimpleHTTPServer
[19:35] <bencc> than you can see it in http://localhost:8000/publisher.html
[19:36] <klaxa> yeah tried that already
[19:36] <klaxa> same results
[19:37] <bencc> trying
[19:38] <Mavrik> LordDoskias, add the right format and url as an output?
[19:38] <bencc> klaxa: can you try with FF?
[19:38] <LordDoskias> Mavrik, i want to write in a memory buffer
[19:38] <LordDoskias> Mavrik, where can i find a tutorial how can i do the muxing?
[19:39] <bencc> klaxa: the console doesn't work for me either
[19:39] <klaxa> https://gist.github.com/klaxa/5845844
[19:39] <klaxa> stops there
[19:40] <bencc> klaxa: right
[19:40] <bencc> that's what I'm getting
[19:40] <klaxa> why are you using flash inbetween anyways? :X
[19:40] <Mavrik> LordDoskias, so you want to use the libav libraries as an API not ffmpeg executable right?
[19:40] <LordDoskias> yes
[19:40] <xreal> Are there any free RTMP server as online service, where I can stream to using ffmpeg? I don't mean RTMP server software!
[19:40] <LordDoskias> Mavrik, yues
[19:41] <bencc> klaxa: maybe you are not getting the stream
[19:41] <bencc> I'll send you subsriber.html too so you can test if the stream works
[19:41] <klaxa> i would think it's the same error as with mplayer2
[19:41] <bencc> klaxa: do you see packets in the rtmplite terminal?
[19:42] <klaxa> bencc: https://gist.github.com/klaxa/5845855
[19:42] <klaxa> the complete log during testing
[19:42] <Mavrik> LordDoskias, well the best resource you have is doc/examples for muxing and demuxing
[19:42] <Mavrik> LordDoskias, and the ffmpeg doxygen
[19:43] <bencc> klaxa: ok so publisher.html works. you are sending rtmp audio stream
[19:44] <bencc> klaxa: in ffmpeg my output is a bit different than yours
[19:45] <bencc> http://dpaste.com/1265350/
[19:45] <klaxa> >playpath=z010002.stream
[19:45] <bencc> it manage to do the handshake and play the stream
[19:45] <klaxa> where does that come from?
[19:45] <bencc> sorry. wrong command
[19:45] <bencc> ./ffmpeg -loglevel debug -i rtmp://localhost/audio/test playpath=z010002.stream -f mp3 pipe:
[19:46] <bencc> it's probably from testing
[19:46] <LordDoskias> Mavrik, i will check the muxing example but i think it deals with files not raw memory, hope it produces muxed packets and then those packets are written into memory
[19:46] <Mavrik> um
[19:47] <Mavrik> you define your own AVIOContext on the output format
[19:47] <Mavrik> and just grab encoded data when it comes to you
[19:49] <klaxa> bencc how would i go about to "compile" the mxml file? i uploaded everything to my server, but it fails to connect to localhost (client-side) obviously
[19:50] <bencc> klaxa: do you have flex installed?
[19:50] <bencc> maybe simpler. there is a demo app in rtmplite
[19:50] <bencc> let me see
[19:51] <bencc> klaxa: rtmplite/testClient/bin-debug/testClient.swf
[19:51] <bencc> and testClient.html
[19:51] <bencc> better than my publisher because you can publish and watch the stream for testing
[19:56] <klaxa> bencc: http://klaxa.eu/bencc/rtmplite/testClient/bin-debug/testClient.html
[19:57] <klaxa> if something should be streaming try accessing it and tell me how... i have no clue how to use that thing
[19:57] <bencc> 1. set the URL of your server in NetConnection
[19:57] <bencc> 2. click Connect
[19:57] <bencc> 3. click Publish
[19:59] <klaxa> if i click publish nothing happens
[19:59] <bencc> what is the url of your rtmp server?
[19:59] <bencc> I'll try to publish form here
[19:59] <klaxa> klaxa.eu
[19:59] <klaxa> rtmp://klaxa.eu right?
[20:00] <bencc> rtmp://klaxa.eu/myapp
[20:00] <klaxa> yeah tried that too
[20:01] <klaxa> things are happening but it's still not working
[20:01] <bencc> trying locally
[20:02] <bencc> klaxa: locally it works
[20:02] <klaxa> lolweird
[20:03] <bencc> maybe a flash permission issue
[20:03] <bencc> flash is ugly
[20:03] <klaxa> can you get the stream with ffmpeg though?
[20:03] <bencc> klaxa: trying
[20:03] <bencc> I can publish on my local web server
[20:04] <bencc> maybe your rtmplite server is not running or not publically accesible?
[20:05] <klaxa> it should have been though... can you forward your rtmp port or something?
[20:05] <klaxa> arghs nvm that, but can you grab the stream with ffmpeg?
[20:05] <klaxa> can you play it back with ffplay?
[20:05] <bencc> trying
[20:07] <bencc> klaxa: getting the same output
[20:07] <bencc> starts to play the stream but doesn't output packets to the terminal
[20:08] <bencc> http://dpaste.com/1265404/
[20:08] <klaxa> can you play it back with ffplay though?
[20:08] <klaxa> i mean can you hear it? or anything?
[20:10] <bencc> testing
[20:11] <bencc> klaxa: I don't have ffplay installed
[20:11] <klaxa> try any mediaplayer that supports rtmp then
[20:11] <klaxa> mplayer/mplayer2, vlc, anything
[20:12] <bencc> I can use testClient in another browser and play the stream
[20:12] <bencc> that works
[20:12] <klaxa> that is really weir
[20:12] <klaxa> *weird
[20:13] <bencc> what's weird?
[20:13] <bencc> streaming works
[20:13] <bencc> playing works
[20:13] <bencc> transcoding to mp3 file with ffmpeg works
[20:13] <bencc> it just doesn't work in real time
[20:13] <bencc> maybe it buffers everything and write to the file when you stop the stream
[20:14] <bencc> probably a bug
[20:14] <klaxa> stdin should not be buffered
[20:14] <klaxa> stdout, yes, but not stdin :X
[20:16] <bencc> it should "just work"
[20:16] <klaxa> if it would not be too much to ask, i would like to take a look first hand, can you forward an ssh port and add a guest user? (info via query)
[21:33] <LordDoskias> so before starting encoding I have to use avformat_alloc_output_context2, however this function implies that the data will be written to a file, how can i do it so that data is written to a memory buffer ?
[21:35] <Mavrik> LordDoskias, even though the signature implies filenames, that's not strictly true
[21:35] <LordDoskias> i'm reading https://github.com/FFmpeg/FFmpeg/blob/master/doc/examples/muxing.c
[21:35] <Mavrik> LordDoskias, pass your own AVOutputFormat to that function and AVOutputFormat has callbacks
[21:36] <Mavrik> there you have callbacks write_packet, write_header, etc. which will be called by the muxer when it has data ready for you
[21:37] <LordDoskias> i see
[21:38] <Mavrik> LordDoskias, the question is why don't you just pass something like "udp://192.168.1.1" as an output filename and just let ffmpeg handle that ;)
[21:38] <Mavrik> when you said you want to stream over network
[21:38] <LordDoskias> because i'm ecndoing with raspberrypi
[21:38] <LordDoskias> and i waneed hardware acceleration so i have written custom piece of software that does that
[21:39] <LordDoskias> and i get raw encoded data, now i want to feed this data to avlib in order to perform software muxing
[21:41] <Mavrik> and what of that stops you from using ffmpeg's streaming code to do the muxing and streaming part for you?
[21:44] <LordDoskias> i don't know or rather haven't looked at how i can integrate ffmpeg into this
[21:44] <LordDoskias> is there a tutorial ?
[21:45] <LordDoskias> how can i hook up ffmpeg to a memory buffer or something?
[21:45] <LordDoskias> you are the second person from this channel which hints at this idea
[21:45] <LordDoskias> s/which/who
[21:45] <Mavrik> just like you started.
[21:45] <Mavrik> you have your encoded packets
[21:45] <Mavrik> you initialize output context (thatis - the muxer) and as a filename you give it the address of where to stream the output
[21:46] <Mavrik> then you just use av_write_packet to it and it'll mux it and send it over network
[21:46] <Mavrik> these are the libav supported protocols: http://ffmpeg.org/ffmpeg-protocols.html
[21:47] <LordDoskias> is there a raw udp multicast?
[21:47] <LordDoskias> Mavrik, but then again this is exactly the same thing i want to do
[21:47] <Mavrik> that would be udp://<target-address> :)
[21:47] <LordDoskias> oh wait, so you are saying
[21:47] <LordDoskias> it will be exatly like in the muxing example
[21:47] <LordDoskias> but the file name is going to be the udp://mcast address
[21:47] <Mavrik> yup.
[21:47] <Mavrik> :)(
[21:47] <LordDoskias> yeah,
[21:47] <LordDoskias> okay, that's good
[21:48] <LordDoskias> saves shitload of work :D
[21:48] <Mavrik> that's what I'm telling you :P
[21:48] <LordDoskias> but in the demo they are doing: write_video_frame
[21:48] <Mavrik> that's a function in the same file.
[21:48] <LordDoskias> ah, this is a custom functions
[21:48] <LordDoskias> thanks
[21:48] <LordDoskias> cool, cool, cool
[21:50] <LordDoskias> i don't think ther eis av_write_packet isntead avwrite interleaved frame
[21:50] <LordDoskias> which should be ok
[21:57] <Mavrik> mhm.
[22:56] <bencc> klaxa: not sure how to do it
[22:56] <bencc> klaxa: I'm inside a VM behind a router
[22:56] <klaxa> oh jeez
[22:57] <bencc> anyway, it shouldn't be needed. using rtmplite with the testClient should reproduce the issue
[22:57] <klaxa> it still doesn't for me :S
[22:57] <bencc> I can help you fix it if you want
[22:57] <bencc> I can help you help me :)
[22:58] <klaxa> sounds good enough
[22:58] <klaxa> do you need shell access to fix it?
[22:58] <bencc> I don't think so
[22:58] <klaxa> okay
[22:58] <bencc> let's start from the beginning
[22:58] <bencc> we want to be able to stream with one browser and play it in the other
[22:59] <bencc> did you start rtmplite on your local server?
[22:59] <klaxa> yes
[23:00] <bencc> did you start a local python http server inside the testClient/bin-debug folder?
[23:00] <klaxa> yep
[23:00] <AnAnt> Hello, I have an .MPG file from a camera, I want to compress it to mp4, I tried the command: avconv -i file.MPG -acodec ac2 file.mp4, but the resulting file had a bad resolution
[23:01] <AnAnt> can't I do the compression without losing quality
[23:01] <AnAnt> ?
[23:01] <klaxa> AnAnt: this channel is fo ffmpeg not for avconv, try #libav
[23:01] <bencc> klaxa: now, load http://localhost:8000/testClient.html in your browser
[23:01] <AnAnt> ah
[23:01] <bencc> klaxa: click on the connect button. you should see NetConnection.Connect.Success in the log
[23:01] <klaxa> yep
[23:01] <bencc> good. you are able to connect
[23:02] <bencc> now click on publish and test the rtmplite terminal
[23:02] <bencc> you should see hex messages of packets there
[23:02] <klaxa> yes, now i see them
[23:02] <bencc> good
[23:02] <bencc> you can open testClient with another browser, click connect and than play to test if you can see/hear the stream
[23:03] <klaxa> that works too
[23:03] <bencc> now you have all that needed to try and create an mp3 stream from the rtmp stream with ffmpeg
[23:03] <AnAnt> klaxa: how would I do it using ffmpeg ?
[23:03] <llogan> AnAnt: https://ffmpeg.org/trac/ffmpeg/wiki/x264EncodingGuide
[23:04] <llogan> https://ffmpeg.org/trac/ffmpeg/wiki/UbuntuCompilationGuide
[23:04] <llogan> all the links you'll ever need.
[23:05] <klaxa> bencc: nope won't work
[23:05] <bencc> klaxa: what doesn't work?
[23:05] <llogan> AnAnt: now you have three choices: simply use a static build of real ffmpeg, compile it, or go to #libav
[23:06] <klaxa> can't play it with mplayer2, can't open it with ffmpeg, i'll get a more recent version one sec
[23:07] <bencc> klaxa: I wonder if the problem is with rtmplite or with ffmpeg
[23:07] <bencc> klaxa: I'll might need to ask on the rtmplite mailing list
[23:07] <klaxa> https://gist.github.com/klaxa/5846547
[23:07] <klaxa> i'll build the latest ffmpeg from source, may take some minutes
[23:07] <AnAnt> llogan: thanks
[23:08] <bencc> klaxa: aren't you missing the stream name? rtmp://localhost/myapp
[23:08] <klaxa> hm?
[23:08] <bencc> klaxa: I think it should be: rtmp://localhost/myapp/user1
[23:09] <bencc> in testClient you have a textbox which let you enter the stream name
[23:09] <bencc> the default is user1
[23:09] <klaxa> looks better, still won't start encoding
[23:10] <bencc> you see a message that says it started playing ?
[23:10] <klaxa> https://gist.github.com/klaxa/5846555
[23:11] <bencc> klaxa: it should say it is playing something
[23:13] <bencc> klaxa: http://dpaste.com/1265775/
[23:13] <klaxa> nope, not happening for me
[23:14] <bencc> rtmplite is still running?
[23:14] <bencc> are you still publishing audio/video?
[23:14] <klaxa> yes
[23:15] <bencc> so it should work
[23:15] <klaxa> weird shit, we're even using the same binary
[23:15] <bencc> no idea why it doesn't
[23:15] <klaxa> well not exactly
[23:34] <klaxa> haha wow bencc, rmtplite wasn't allowed to bind port 1935 apparently
[23:34] <klaxa> which is why things failed
[23:36] <klaxa> hmm... but ffmpeg is still not reading the stream
[23:36] <klaxa> neither is ffplay
[23:37] <klaxa> neither is vlc
[23:38] <klaxa> exiting with the same error as mplayer2
[23:38] <klaxa> neither is rtmpdump
[23:38] <bencc> klaxa: even on localhost?
[23:38] <klaxa> i start to think that rtmplite is at fault
[23:39] <bencc> what error do you get?
[23:39] <klaxa> https://gist.github.com/klaxa/5846639
[23:40] <bencc> klaxa: are you sure you are publishing?
[23:40] <klaxa> yes i'm seeing the stream in the browser
[23:41] <bencc> I also got this error with gstreamer. not sure what to say
[23:41] <klaxa> with the flash embedd it works
[23:42] <bencc> klaxa: I don't know what the problem is
[23:42] <bencc> :(
[23:43] <klaxa> i would *think* that rtmplite is not 100% compliant to rtmp
[23:45] <klaxa> i still don't understand exactly why you need flash :X
[23:45] <bencc> klaxa: I can try with red5
[23:45] <klaxa> and why you need rtmp
[23:45] <bencc> klaxa: I need users to publish audio stream from the browser and broadcast it as mp3 to browsers without flash
[23:46] <klaxa> so you want to make an application to stream audio from browser to browser, one with flash, one without?
[23:47] <bencc> yes
[23:47] <bencc> one browser to many browsers
[23:54] <bencc> klaxa: I'm able to connect with ffmpeg to rtmplite
[23:54] <klaxa> so it starts encoding audio?
[23:54] <bencc> and sending the play 'test' command
[23:54] <bencc> but than it just hang
[23:54] <bencc> I've already sent you the log
[23:55] <bencc> http://dpaste.com/1265775/
[23:55] <klaxa> yes that's about as far as i get with any player
[23:55] <klaxa> also, looking into recording audio with html5, seems possible
[00:00] --- Mon Jun 24 2013