Re: [FFmpeg-user] Issues deinterlacing DirectShow input with ffplay
On Tuesday, April 19th, 2022 at 2:03 AM, Roger Pack wrote: > Can you replicate it not using dshow? Not in any way that I know of. I recorded raw output from the capture card (using -c copy) to an AVI, and when playing the file with ffplay the yadif filter works fine (also when piping the file through stdin). ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Issues deinterlacing DirectShow input with ffplay
On Sunday, April 10th, 2022 at 1:18 AM, Roger Pack wrote: > Input frame rate is still the same both ways? Yes, 29.97 fps either way (didn't realize that it's covered in the video; the log files should have everything though). Alex ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] Increasing lags when using ffmpeg for streaming audio from microphone
Hi everyone, I'm attempting to use ffmpeg as a DIY baby audio monitor. Although I have a working prototype, there is a delay that gets progressively greater. The lag is of a few seconds in the beginning, but it reaches a few minutes after several hours. My set-up is: - Raspberry Pi 2B with a USB microphone is the streaming server - My client is usually VLC running on Android or on a computer elsewhere in the house - I only have one client at a time, - but clients are different, depending on who watches the child (so the solution to stream directly to a specific IP is not suitable) - All the devices are in the same network - ffmpeg is started by another process with these parameters: ffmpeg -re -f alsa -i plughw:1,0 -vn -acodec libmp3lame -b:a 8k -ac 1 -ar 22050 -f mp3 - The parent process continuously reads stdout and exposes the chunks over HTTP. This makes it convenient, as the stream can be played in a browser too. The tool that does it is micstream: https://github.com/BlackLight/micstream/blob/main/micstream/server.py I've tried tweaking the ffmpeg parameters and have gotten some small improvements, by reducing the bit-rate, for example. However, I believe the approach needs to be reviewed, because the delays still pile up over time. Hypotheses I've had: 1. The client has a buffer of its own However, VLC allows me to set the cache size by specifying a duration, which is currently at 1000ms. I tried lower values too, but there was no noticeable difference. 2. The hardware is not fast enough I doubt it because next to the RasPi 2B streaming the microphone, I have a RasPi ZeroW that streams video from a camera - it is very smooth, and the delay is ~1s even after weeks of uptime. Further, if I inspect what ffmpeg writes to stdout, the text at the bottom says `size= 343273kB time=97:38:43.52 bitrate=8.0kbits/s speed= 1x`, which I suppose means that encoding in real-time works well. 3. The network itself Although both RasPis mentioned above work over Wi-Fi and are in a remote part of the house - the video stream works reliably. Also, if I disable the video, audio still lags. Moreover, if I modify ffmpeg parameters to stream directly to another address over RTP: `ffmpeg -re -f alsa -i plughw:1,0 -vn -acodec libmp3lame -b:a 8k -ac 1 -ar 22050 -f rtp rtp://192.168.1.10:5002` and on 192.168.1.10 (the receiver, not the same system as the streamer) I run netcat to see what I'm getting `nc -u -l 5002` - I do see small chunks of what appears to be mp3 headers and some payload arriving at regular intervals, and there are no periods where these datagrams stop arriving. I am wondering what else I could try to establish the root cause of the problem and reduce the delays. Perhaps the culprit is `micstream`, I'd be happy to replace it with something else that is known to work better. However, on the Internet I found references to `ffserver` which is not available anymore, while tutorials I've found for Apache and nginx are tailored for video streaming. Is the scenario I described feasible at all? What troubleshooting steps could I try? Best wishes to everyone, and I look forward to your feedback, Alex ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Advice on using silence removal
Hi everyone, Thank you for providing valuable feedback about silence removal last month. For the benefit of future archaeologists, I summarize the steps I've taken and the key elements of the solution. Note that while this worked for me, I do not claim that this is the optimal approach. - As Carl pointed out, don't normalize before silence removal. This is obvious in retrospect, but I didn't think of it myself. - The "compand" filter makes a substantial contribution to the quality of the output. - This article provides a clear, step by step explanation of how to use this feature of ffmpeg; there are also illustrations that show how the waveform changes after each step https://medium.com/@jud.dagnall/dynamic-range-compression-for-audio-with-ffmpeg-and-compand-621fe2b1a892 - Use the mean volume as a threshold for the silence detector (in the past I used the maximum value) In case the site above is not available, here is a relevant excerpt: ``` ffmpeg -i in.mp3 -filter_complex "compand=attacks=0:points=-30/-900|-20/-20" out.wav - attacks=0 means that I wanted to measure absolute volume, not averaging the sound over a short (or long period of time) - followed by points, which is a series of "from->to" mappings that are to be interpreted as: - -30/-900, which means that volume below -30db in the original input track gets converted to -900db (completely silent) - -20/-20 means that at -20db the volume remains unchanged ``` In practical terms, here are the steps I currently use in my noise gate function: 1. cut the leading and trailing 200ms of the file (this is where I usually had the sound of a click/tap when users begin/stop the recording) 2. use a combination of a high-pass and low-pass filter for the range 200 .. 4000 that should cover a typical human voice ffmpeg -i out-02-trim-ex.wav -af "highpass=f=200, lowpass=f=4000" out-03-range-filter.wav 3. apply the compand filter ffmpeg -i out-03-range-filter.wav -filter_complex "compand=attacks=0:points=-30/-900|-20/-20" out-04-compand.wav 4. apply the silence removal filter ffmpeg -i out-04-compand.wav -af silenceremove=start_periods=1:start_duration=0: start_threshold=-6dB:start_silence=0.5,areverse,silenceremove=start_periods=1: start_duration=0:start_threshold=-6dB:start_silence=0.5,afade=t=in:st=0: d=0.3,areverse,afade=t=in:st=0:d=0.3 out-05-silence-fade.wav Notes: - the threshold of -6dB in the command line above is not hardcoded, but it is the mean value as detected by `volumedetect` - we remove silence from the beginning, then turn the signal around and repeat the process, then turn it around again - such that both ends are without silence 5. normalize it to the max value returned by `volumedetect` ffmpeg -i out-05-silence-fade.wav -af "volume=18.2 dB" out-06-normalized.wav Thanks again for your assistance, I greatly appreciate it. If anyone comes up with refinements of the describe approach, please share your methodology. Alex ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] Advice on using silence removal
Hi everyone, I am attempting to leverage ffmpeg in a project that involves recording short audio clips. So far I have gotten some mixed results and I'd like to tap into your collective knowledge to ensure my approach is sound. Context: - a person records an audio clip of themselves pronouncing a word (imagine that you read aloud a flash-card that says "tree" or "helicopter") - the recording is usually made on a mobile phone The clip contains some silence at both ends, because there is a delay between the moment the user presses the record button, the moment they pronounce their word, and the moment they press "stop". Depending on the device, there may also be an audible click in the beginning. My objective is to trim the silence at both ends and apply fade-in/out to soften the clicks, if any. The challenges are: - ffmpeg's silenceremove filter needs a threshold value, however, - each user is in their own environment, with different levels of ambient noise - each device is unique in terms of sensitivity Thus, I can achieve my desired result with one specific clip through trial and error, tinkering with thresholds until I get what I need. But I cannot figure out how to detect these thresholds automatically, such that I can replicate the result with a broad range of users, environments and recording devices. Note that there is no expectation to produce perfect results that match the quality of an audio recording studio, I'm more in the "rough, but good enough for practical purposes" territory. Having read the documentation and various forums, I put together this pipeline (actual commands in the appendix): 1. run volumedetect to see what the maximum level is 1a. parse stdout to extract `max_volume` 2. normalize audio to `max_volume` 3. apply silenceremove with 3a. for the beginning of the file 3b. invert the stream and run another silenceremove for the beginning (which is actually the end) 3c. invert it back and save the output What I read in the forums gave me the impression that we need step#2 such that at step#3 we could say the threshold is 0. However, that is not the case, I still had to find a reasonable threshold via trial and error. After I found a value that produces a good result, I assumed that it might be good enough for practical purposes and it would be OK to simply hardcode it into my code as a magic number. However, on the next day I attempted to replicate the results using the same recording device in the same room - but this time ffmpeg would tell me the filtered stream is empty, nothing to write. The environment wasn't 100% identical, since I'm not doing this in a controlled lab, but most of the variables are the same, though perhaps the windows were open and it was a different time of the day, so the baseline noise level outside was somewhat different. Clearly, my approach is not robust. I'd like to understand whether there are any low-hanging fruits that I can try, or if I'm not on the right track. I imagine that the solution I need would somehow determine the silence threshold relative to the rest of the file, instead of using a "one fits all" value. However I did not find such filters or analyzers in ffmpeg. Your guidance will be greatly appreciated, Alex Appendix, pipeline commands 1. ffmpeg -i input.mp3 -af "volumedetect" -f null /dev/null here I parse stdout, looking for something like "[Parsed_volumedetect_0 @ 0x559dbe815f00] max_volume: -15.9 dB" 2. ffmpeg -i input.mp3 -af "volume=15.9dB" out2-normalized.mp3 3. ffmpeg -i out2-normalized.mp3 -af silenceremove=start_periods=1:start_duration=0:start_threshold=-6dB:start_silence=0.5,areverse,silenceremove=start_periods=1:start_duration=0:start_threshold=-6dB:start_silence=0.5,afade=t=in:st=0:d=0.3,areverse,afade=t=in:st=0:d=0.3 out3-trimmed.mp3 An example of an input file is available at railean.net/files/public-temp/in-fresh.mp3, after normalization you can hear some church bells in the distance. I'm totally fine with them remaining audible in the result, as long as the leading and trailing silence is removed. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] Issues deinterlacing DirectShow input with ffplay
I'm using ffplay as a live preview for my capture card, and I'm trying to use the yadif filter in mode 1 (send_field) to deinterlace the 59.94i input to 59.94 fps. Using this command works as expected: ffplay -f dshow -i video="SA7160 PCI, Analog 01 Capture" -vf yadif=1 However, if I try the same command with an audio device included, the frame rate appears capped at 29.97 fps as if I were using mode 0 (send_frame): ffplay -f dshow -i video="SA7160 PCI, Analog 01 Capture":audio="SA7160 PCI, Analog 01 WaveIn" -vf yadif=1 Why does the filter only work properly when there is no audio device on the input? Is there a workaround for this or is it a bug? Here's a screen recording as a demonstration: https://streamable.com/p0v9rk Console output is attached. AlexC:\test> ffplay -f dshow -i video="SA7160 PCI, Analog 01 Capture" -vf yadif=1 ffplay version N-103227-g115f5e8035-20210813 Copyright (c) 2003-2021 the FFmpeg developers built with gcc 10-win32 (GCC) 20210408 configuration: --prefix=/ffbuild/prefix --pkg-config-flags=--static --pkg-config=pkg-config --cross-prefix=x86_64-w64-mingw32- --arch=x86_64 --target-os=mingw32 --enable-gpl --enable-version3 --disable-debug --disable-w32threads --enable-pthreads --enable-iconv --enable-libxml2 --enable-zlib --enable-libfreetype --enable-libfribidi --enable-gmp --enable-lzma --enable-fontconfig --enable-libvorbis --enable-opencl --enable-libvmaf --enable-vulkan --disable-libxcb --disable-xlib --enable-amf --enable-libaom --enable-avisynth --enable-libdav1d --enable-libdavs2 --disable-libfdk-aac --enable-ffnvcodec --enable-cuda-llvm --enable-libglslang --enable-libgme --enable-libass --enable-libbluray --enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvpx --enable-libwebp --enable-lv2 --enable-libmfx --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librav1e --enable-librubberband --enable-schannel --enable-sdl2 --enable-libsoxr --enable-libsrt --enable-libsvtav1 --enable-libtwolame --enable-libuavs3d --disable-libdrm --disable-vaapi --enable-libvidstab --enable-libx264 --enable-libx265 --enable-libxavs2 --enable-libxvid --enable-libzimg --extra-cflags=-DLIBTWOLAME_STATIC --extra-cxxflags= --extra-ldflags=-pthread --extra-ldexeflags= --extra-libs=-lgomp --extra-version=20210813 libavutil 57. 3.100 / 57. 3.100 libavcodec 59. 4.101 / 59. 4.101 libavformat59. 4.101 / 59. 4.101 libavdevice59. 0.100 / 59. 0.100 libavfilter 8. 1.103 / 8. 1.103 libswscale 6. 0.100 / 6. 0.100 libswresample 4. 0.100 / 4. 0.100 libpostproc56. 0.100 / 56. 0.100 Input #0, dshow, from 'video=SA7160 PCI, Analog 01 Capture':f=0/0 Duration: N/A, start: 214853.603000, bitrate: N/A Stream #0:0: Video: rawvideo (YUY2 / 0x32595559), yuyv422, 720x480, 29.97 fps, 29.97 tbr, 1k tbn 214854.83 M-V: -0.000 fd= 0 aq=0KB vq=0KB sq=0B f=0/0 C:\test> ffplay -f dshow -i video="SA7160 PCI, Analog 01 Capture":audio="SA7160 PCI, Analog 01 WaveIn" -vf yadif=1 ffplay version N-103227-g115f5e8035-20210813 Copyright (c) 2003-2021 the FFmpeg developers built with gcc 10-win32 (GCC) 20210408 configuration: --prefix=/ffbuild/prefix --pkg-config-flags=--static --pkg-config=pkg-config --cross-prefix=x86_64-w64-mingw32- --arch=x86_64 --target-os=mingw32 --enable-gpl --enable-version3 --disable-debug --disable-w32threads --enable-pthreads --enable-iconv --enable-libxml2 --enable-zlib --enable-libfreetype --enable-libfribidi --enable-gmp --enable-lzma --enable-fontconfig --enable-libvorbis --enable-opencl --enable-libvmaf --enable-vulkan --disable-libxcb --disable-xlib --enable-amf --enable-libaom --enable-avisynth --enable-libdav1d --enable-libdavs2 --disable-libfdk-aac --enable-ffnvcodec --enable-cuda-llvm --enable-libglslang --enable-libgme --enable-libass --enable-libbluray --enable-libmp3lame --enable-libopus --enable-libtheora --enable-libvpx --enable-libwebp --enable-lv2 --enable-libmfx --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-librav1e --enable-librubberband --enable-schannel --enable-sdl2 --enable-libsoxr --enable-libsrt --enable-libsvtav1 --enable-libtwolame --enable-libuavs3d --disable-libdrm --disable-vaapi --enable-libvidstab --enable-libx264 --enable-libx265 --enable-libxavs2 --enable-libxvid --enable-libzimg --extra-cflags=-DLIBTWOLAME_STATIC --extra-cxxflags= --extra-ldflags=-pthread --extra-ldexeflags= --extra-libs=-lgomp --extra-version=20210813 libavutil 57. 3.100 / 57. 3.100 libavcodec 59. 4.101 / 59. 4.101 libavformat59. 4.101 / 59. 4.101 libavdevice59. 0.100 / 59. 0.100 libavfilter 8. 1.103 / 8. 1.103 libswscale 6. 0.100 / 6. 0.100 libswresample 4. 0.100 / 4. 0.100 libpostproc56. 0.100 / 56. 0.100 Input #0, dshow, from 'video=SA7160 PCI, Analog 01 Capture:audio=SA7160 PCI, Analog 01 WaveIn': Duration:
Re: [FFmpeg-user] Problem with changing options at runtime with a command
thank you for the quick answer when trying this out I notice 3 things *1: the original size is kept* if the starting crop is *crop=h=100 *then using *c crop -1 h 150 *dose noting and the other way, using *c crop -1 h 50 *leaves the actual pixel dimentions at 100 but the lower portion is transparent. So you can't change the actual dimentions with a command? is the a filter i should use after crop to accomplish this? *2: c crop w dosen't work* *c crop -1 h 100 *crop height to 100 but *c crop -1 w 100 *stil returns a full width image but crops to 1 pixel in height i also tried *c crop -1 w ih *and that returns a full width and dosnt crop the height but *c crop -1 w ih-1 *also crops to 1 pixel in height *3: can't set w and h at the same time* using *c drawtext -1 reinit fontcolor=red:text='test' *works but i can't figur out how to do the same with crop *c crop -1 h 50 : w 50 * returns: [Parsed_crop_2 @ 01dde3ffaa80] [Eval @ 0005a89fdd80] Invalid chars ':w50' at the end of expression '50 : w 50' [Parsed_crop_2 @ 01dde3ffaa80] Error when evaluating the expression '50 : w 50' Command reply for stream 0: ret:-22 res: i tried a a few combinations with no luck *@Michael *i agree that the doc could be a bit cleare. as you can see i got my idears from the only example i could find and that was from drawtext. vh Alex On Sat, Jul 3, 2021 at 3:12 PM Michael Koch wrote: > Am 03.07.2021 um 15:06 schrieb Michael Koch: > > Hi Gyan, > > > >> > >> As the docs state, the acceptable commands are w, h, x, ,y > >> > >> so the syntax is > >> > >> c crop -1 w 100 > >> > > > > Is this documented somewhere? I mean typing "c" in the console while > > FFmpeg is running. > > I know that you mentioned it on stackoverflow some time ago, but I > > never found it in the FFmpeg docs. > > > > > https://stackoverflow.com/questions/56058909/ffmpeg-drawtext-and-live-coordinates-with-sendcmd-zmq > > > > > > I think it should be added to chapter 34 in FFmpeg-all.html, "Changing > > options at runtime with a command". > > P.S. In the same chapter could also be mentioned that two other methods > exist: sendcmd and zmq. > > Michael > > ___ > ffmpeg-user mailing list > ffmpeg-user@ffmpeg.org > https://ffmpeg.org/mailman/listinfo/ffmpeg-user > > To unsubscribe, visit link above, or email > ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] Problem with changing options at runtime with a command
Hello i am trying to change options in the crop filter at runtime with a command this it my test command: ./ffmpeg -hide_banner -y -f dshow -rtbufsize 128M -i video="Cam Link 4K" -filter_complex "[0:v]fps=10[rate], [rate]scale=1280:720, crop=in_w, drawtext=fontfile=RobotoMono-Medium.ttf: text=\'test\': fontsize=50: fontcolor=white: box=1: boxcolor=black@0.5: boxborderw=5" -update 1 public\img.png it works and returns: Input #0, dshow, from 'video=Cam Link 4K': Duration: N/A, start: 136428.863000, bitrate: N/A Stream #0:0: Video: rawvideo (YUY2 / 0x32595559), yuyv422, 1920x1080, 60 fps, 60 tbr, 1k tbn Stream mapping: Stream #0:0 (rawvideo) -> fps drawtext -> Stream #0:0 (png) Press [q] to stop, [?] for help Output #0, image2, to 'public\img.png': Metadata: encoder : Lavf59.3.101 Stream #0:0: Video: png, rgb24(pc, progressive), 1280x720, q=2-31, 200 kb/s, 10 fps, 10 tbn Metadata: encoder : Lavc59.2.100 png then sending: c crop -1 w=100 the retun is: Command reply for stream 0: ret:-40 res: and nothing changes instead trying: c drawtext -1 reinit fontcolor=red the retun is: Command reply for stream 0: ret:0 res: and the text turns red what am i missing? -- Bedste hilsner / Best regards Alex ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] ffmpeg http filter?
Server just do post processing of raw rgb image/frame and return it as response for request.Something like so: POST http://localhost:8080 --- Original message --- From: "Edward Park" Date: 8 September 2020, 02:19:21 Hi, > ffmpeg -i test.jpg -vf format=rgb24,http=localhost:8080 -y out.jpg I don't think it's possible using filters, or with a single invocation like that. (The 'http' filter is hypothetical and just meant for illustration right?) Depending on how you're connected to the server, I think pipes or sockets would be a better place to start. And also separate commands for outputting frames for the server to consume, and another that takes the returned images. Regards, Ted Park ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] ffmpeg http filter?
Hi! Thank for reply, but no I need next logic: input.jpg => (ffmpeg decode image/video) => (ffmpeg scale frame to 600x600 pix and convert frame format to rgb24) => (ffmpeg send frame to remote address localhost:8080 for post processing of image/frame) => (ffmpeg get result image from remote server, format + add overlay) => (ffmpeg encode frame/image to output format) => out.jpg --- Original message --- From: "James Darnley" Date: 8 September 2020, 12:44:57 On 08/09/2020, Alex <3.1...@ukr.net> wrote: > I need to send raw frame/image to server for post processing and server > returned new image that I need to complete with ffmpeg. Do any one know how > to do this? > Somethink like that: > ffmpeg -i test.jpg -vf format=rgb24,http=localhost:8080 -y out.jpg If I understand correctly: use two outputs. One for the rawvideo which goes over some network protocol to your server and one for that jpeg output. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] ffmpeg http filter?
I need to send raw frame/image to server for post processing and server returned new image that I need to complete with ffmpeg. Do any one know how to do this? Somethink like that: ffmpeg -i test.jpg -vf format=rgb24,http=localhost:8080 -y out.jpg ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Error extracting srt
Thank you SO much Moritz! On Fri, Aug 7, 2020 at 12:20 PM Moritz Barsnick wrote: > On Wed, Aug 05, 2020 at 22:28:15 +0100, Alex Zachopoulos wrote: > > This is the command I use to extract Stream #0:2 (subtitle) from file > ^^ This is an input stream > specifier, for mapping > > 1.mp4, on both computers: > > > > ffmpeg -i 1.mp4 -vn -an -codec:s:0.2 srt 1.srt > ^^ This is an output stream specifier. > > You want to tell ffmpeg to encode the first *output* subtitle stream as > SRT. That's not "0:2", neither "0.2", that's just "0". > > If you have only one output SRT stream - which should be the case, > since your output SRT file can only include one stream - you can omit > the output stream specifier: "-codec:s" > > In your case, you can omit the "-codec:s" totally, because the suffix > of your output file implies it. > > > *[srt @ 0x7fd3b300ce00] Invalid stream specifier: s:0.2.* > > I assume the stream specifier parser used to just ignore the ".2", and > it no longer does, and errors out instead. > > Cheers, > Moritz > ___ > ffmpeg-user mailing list > ffmpeg-user@ffmpeg.org > https://ffmpeg.org/mailman/listinfo/ffmpeg-user > > To unsubscribe, visit link above, or email > ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] Error extracting srt
Hi all, I want to extract the SRT stream from mp4 files. On my MacPro with High Sierra running ffmpeg 3.3 all is fine. On my MacBook Pro with Catalina running ffmpeg 4.3 I keep getting an error. This is the command I use to extract Stream #0:2 (subtitle) from file 1.mp4, on both computers: ffmpeg -i 1.mp4 -vn -an -codec:s:0.2 srt 1.srt On the MacBook Pro with Catalina and ffmpeg 4.3 is complains, twice: First, somewhere down the middle of the dump it says: *[mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fd3b4009800] stream 0, timescale not set* Then at the end of the dump it says: *[srt @ 0x7fd3b300ce00] Invalid stream specifier: s:0.2.* *Last message repeated 1 times* Like I said, the same command works without a hitch on the MacPro. Anyone can help with any pointers, it'll be much appreciated. Alex ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] Sound level measuring on the 2nd audio stream
is expectable: [mp3float @ 000 Input #0, lavfi, fr Duration: N/A, st Stream #0:0: Au 0.00,-50.248303 0.024000,-53.384511 0.048000,-53.646786 0.072000,-48.221139 0.096000,-51.244874 0.12,-53.856832 0.144000,-54.184842 0.168000,-52.673918 0.192000,-53.253474 0.216000,-49.317896 0.24,-44.537543 0.264000,-43.784163 0.288000,-47.155750 0.312000,-37.40 0.336000,-45.422727 0.36,-39.557636 0.384000,-35.931526 0.408000,-27.129281 0.432000,-27.934319 0.456000,-30.063120 0.48,-34.268751 0.504000,-32.415779 0.528000,-28.112756 ... For the second stream: ffmpeg -hide_banner -i test1.mov -ss 00:01:00 -t 00:01:00 -map 0:a:0 test2.mp3 ffprobe -hide_banner -f lavfi -i amovie=test2.mp3,astats=metadata=1:reset=1 -show_entries frame=pkt_pts_time:frame_tags=lavfi.astats.1.RMS_level -of csv=p=0 0.00,-inf 0.024000,-inf 0.048000,-inf 0.072000,-inf 0.096000,-inf 0.12,-inf 0.144000,-inf 0.168000,-inf 0.192000,-inf 0.216000,-inf 0.24,-inf 0.264000,-inf 0.288000,-inf 0.312000,-inf 0.336000,-inf 0.36,-inf 0.384000,-inf 0.408000,-inf 0.432000,-inf 0.456000,-inf 0.48,-inf 0.504000,-inf 0.528000,-inf ... Yes, this is really empty. But when I had copied both audio streams into WMA and tested then, I saw a lie again. It seems ffprobe cannot measure the 2nd audio stream and this is a bug. Am I right? WBR Alex ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] disabling scene detection
> On Mar 10, 2020, at 1:04 AM, Gyan Doshi wrote:
>
>
>
>> On 10-03-2020 12:49 pm, Alex Teslik wrote:
>> Hello,
>>
>> I have an image sequence of line drawings where the drawings change
>> abruptly. Ffmpeg is resetting the frame numbers that I am burning in using
>> the
>> drawtext option every time there is an abrupt change. Here is the command I
>> am
>> using, and the output:
>>
>> ffmpeg.exe -y -r 10 -i final-%d.png -vf
>> "drawtext=fontfile='C\:/WINDOWS/fonts/arialbd.ttf':text=%
>> {frame_num}:start_number=1:fontcolor=black:fontsize=15:x=10:y=10,fps=10" -
>> pix_fmt rgb24 -c:v qtrle -r 10 test.mov
>> ffmpeg version git-2020-02-18-ebee808 Copyright (c) 2000-2020 the FFmpeg
>> developers
>> built with gcc 9.2.1 (GCC) 20200122
>> configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-
>> fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d
>> --
>> enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-
>> libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-
>> libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-
>> libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-
>> libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg
>> --
>> enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis
>> --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid
>> --enable-libaom --enable-libmfx --enable-ffnvcodec --enable-cuvid --enable-
>> d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --
>> enable-libopenmpt --enable-amf
>> libavutil 56. 41.100 / 56. 41.100
>> libavcodec 58. 70.100 / 58. 70.100
>> libavformat58. 38.101 / 58. 38.101
>> libavdevice58. 9.103 / 58. 9.103
>> libavfilter 7. 76.100 / 7. 76.100
>> libswscale 5. 6.100 / 5. 6.100
>> libswresample 3. 6.100 / 3. 6.100
>> libpostproc55. 6.100 / 55. 6.100
>> Input #0, image2, from 'final-%d.png':
>> Duration: 00:00:04.96, start: 0.00, bitrate: N/A
>> Stream #0:0: Video: png, gray(pc), 240x196 [SAR 7559:7559 DAR 60:49], 25
>> fps, 25 tbr, 25 tbn, 25 tbc
>> Stream mapping:
>> Stream #0:0 -> #0:0 (png (native) -> qtrle (native))
>> Press [q] to stop, [?] for help
>> Output #0, mov, to 'test.mov':
>> Metadata:
>> encoder : Lavf58.38.101
>> Stream #0:0: Video: qtrle (rle / 0x20656C72), rgb24, 240x196 [SAR 1:1
>> DAR
>> 60:49], q=2-31, 200 kb/s, 10 fps, 10240 tbn, 10 tbc
>> Metadata:
>> encoder : Lavc58.70.100 qtrle
>> frame= 124 fps=0.0 q=-0.0 Lsize= 635kB time=00:00:12.30 bitrate=
>> 422.9kbits/s dup=3 drop=0 speed=92.5x
>> video:634kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB
>> muxing
>> overhead: 0.25%
>>
>>
>> The images start at 1 and sequentially increment. But when the image changes
>> dramatically, the frame numbering starts over at 1 again. So out of 124
>> images
>> I get 4 sequences burned in:
>>
>> 1-82
>> 1-5
>> 1-25
>> 1-9
>
>
> This is almost certainly not due to scene detection (neither the filtering
> framework or drawtext filter has any provision for that).
>
> What this is, is the filtergraph being reinitialized when some property of
> the input changes - for video/images, the candidates are resolution or the
> pixel format.
>
> That can be suppressed,
>
> ffmpeg.exe -y -framerate 10 -reinit_filter 0 -i final-%d.png -vf
> "scale,format=rgb24,drawtext=fontfile='C\:/WINDOWS/fonts/arialbd.ttf':text=%
> {frame_num}:start_number=1:fontcolor=black:fontsize=15:x=10:y=10" -
> pix_fmt rgb24 -c:v qtrle -r 10 test.mov
>
>
> (scale filter prefixed since only some filters can handle changing input
> without reinit). For image and capture input, -framerate is preferable.
>
> Gyan
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> ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
>
Hi Gyan,
Thank you so much. That switch was very helpful. I couldn’t find any
documentation on it, so it was a great suggestion. It pointed to a problem
where some of the frames in my image sequence were silently converted to the
Grey colorspace instead of sRGB. This was an issue in imagemagick, which was
generating my frames. I forced imagemagick to output consistently in sRGB, and
that combined with the switch you provided made everything work correctly. All
my frames have the correct number burned in now .
Thanks!
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[FFmpeg-user] disabling scene detection
Hello,
I have an image sequence of line drawings where the drawings change
abruptly. Ffmpeg is resetting the frame numbers that I am burning in using the
drawtext option every time there is an abrupt change. Here is the command I am
using, and the output:
ffmpeg.exe -y -r 10 -i final-%d.png -vf
"drawtext=fontfile='C\:/WINDOWS/fonts/arialbd.ttf':text=%
{frame_num}:start_number=1:fontcolor=black:fontsize=15:x=10:y=10,fps=10" -
pix_fmt rgb24 -c:v qtrle -r 10 test.mov
ffmpeg version git-2020-02-18-ebee808 Copyright (c) 2000-2020 the FFmpeg
developers
built with gcc 9.2.1 (GCC) 20200122
configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-
fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --
enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-
libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-
libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-
libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-
libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --
enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis
--enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid
--enable-libaom --enable-libmfx --enable-ffnvcodec --enable-cuvid --enable-
d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --
enable-libopenmpt --enable-amf
libavutil 56. 41.100 / 56. 41.100
libavcodec 58. 70.100 / 58. 70.100
libavformat58. 38.101 / 58. 38.101
libavdevice58. 9.103 / 58. 9.103
libavfilter 7. 76.100 / 7. 76.100
libswscale 5. 6.100 / 5. 6.100
libswresample 3. 6.100 / 3. 6.100
libpostproc55. 6.100 / 55. 6.100
Input #0, image2, from 'final-%d.png':
Duration: 00:00:04.96, start: 0.00, bitrate: N/A
Stream #0:0: Video: png, gray(pc), 240x196 [SAR 7559:7559 DAR 60:49], 25
fps, 25 tbr, 25 tbn, 25 tbc
Stream mapping:
Stream #0:0 -> #0:0 (png (native) -> qtrle (native))
Press [q] to stop, [?] for help
Output #0, mov, to 'test.mov':
Metadata:
encoder : Lavf58.38.101
Stream #0:0: Video: qtrle (rle / 0x20656C72), rgb24, 240x196 [SAR 1:1 DAR
60:49], q=2-31, 200 kb/s, 10 fps, 10240 tbn, 10 tbc
Metadata:
encoder : Lavc58.70.100 qtrle
frame= 124 fps=0.0 q=-0.0 Lsize= 635kB time=00:00:12.30 bitrate=
422.9kbits/s dup=3 drop=0 speed=92.5x
video:634kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing
overhead: 0.25%
The images start at 1 and sequentially increment. But when the image changes
dramatically, the frame numbering starts over at 1 again. So out of 124 images
I get 4 sequences burned in:
1-82
1-5
1-25
1-9
How can I turn off the scene detection so that it stops resetting the frame
counter?
Thanks.
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[FFmpeg-user] Can't select appropriate encoder
Hi,
I am piping a buffer to ffmpeg in my c code. I am using ffmpeg version
3.2.10-1~deb9u1+rpt1 on the raspberry pi. Ffmpeg is giving me issues with
the following line.
pipeout = popen("ffmpeg -y -f s32be -ar 131072 -ac 1 -i -c:a pmc_s32be
hydro.wav, "w");
When I run the code, ffmpeg does not like the "-c:a pmc_s32e" part. I get
an error stating: "-c:a Protocol not found. Did you mean file:-c:a?". When
I change my code to "file:-c:a" I then get the error: "file:-c:a: No such
file or directory".
The reason I need this piece is because ffmpeg defaults to outputting
pcm_s16le audio when I need pcm_s32be.
I have tried "-c:a copy" as well. No luck.
Any help would be appreciated.
Thanks.
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Re: [FFmpeg-user] Is it vp9_qsv encoder work on i9-9900k CPU?
Thank for answer! --- Исходное сообщение --- От кого: "Dennis Mungai" Дата: 19 ноября 2019, 20:18:35 On Tue, 19 Nov 2019 at 21:02, Alex <3.1...@ukr.net> wrote: > > I'am build on linux the latest git version of ffmpeg but conversion failed, > so is it vp9_qsv encoder must work on i9-9900k or we need to wait before new > next gen CPU will be released by Intel?Alex It won't work on hardware older than Icelake with the iHD driver. If you must use VP9 H/W encoding on CFL and KBL, then use VAAPI with the i915 driver. The same also applies to the HDR tone-mapping feature. It will only be present on ICL+. See https://github.com/intel/media-driver for details. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] Is it vp9_qsv encoder work on i9-9900k CPU?
I'am build on linux the latest git version of ffmpeg but conversion failed, so is it vp9_qsv encoder must work on i9-9900k or we need to wait before new next gen CPU will be released by Intel?Alex ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] overlay_cuda
Ok, thanks for answering.If any one read this email and want to create overlay_cuda filter then contact me please (kirpasaccess...@gmail.com), and I can pay for it! Alex --- Исходное сообщение --- От кого: "JackDesBwa" Дата: 8 ноября 2019, 18:12:33 2019-11-08 16:27, Alex <3.1...@ukr.net>: > Thanks for answering. How much, do you think it will be cost to develop > overlay_cuda filter if we already have scale_npp, etc (cuda) filter for quick > start? As user only, I have no idea of the complexity of the task, and it depends on the rate of the freelance you hire. You first have to find some freelance (I know there exists platforms but never used one, neither know a name of one) who has the abilities to work on such project (which sounds not trivial, expect several days because of environment setup, build times and tests in addition to the implementation itself), and discuss with this person the price. Depending of the developer you manage to find and their abilities and experience, it might charge more or less and take more or less time. I am afraid I could not help more here, I do not have experience in this domain. JackDesBwa ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] overlay_cuda
Hi! Thanks for answering. How much, do you think it will be cost to develop overlay_cuda filter if we already have scale_npp, etc (cuda) filter for quick start? 8 ноября 2019, 16:41:35, от "JackDesBwa" < jackdes...@desbwa.org >: 2019-11-07 22:39, Paul B Mahol : > No. > Nice top-posted answer, clear, concise, but totally empty of meaning. :-/ Do you realize that in mailing-list you are interacting with humans ? Such answer, as well as the one you gave recently for the color specification bug (that I was sarcastic about in my answer), are perhaps acceptable for dictators and assholes, but certainly not in sane social interactions with human beings. A concise rational is the absolute minimum IMHO. By the way Alex, ffmpeg developers are volunteers human beings, not slaves that develop what you might need. It is a tremendous money-equivalent job you asked to do for free (well, actually whatever the feature, most people find it very pricey when they discover the true price it costs). However, I am almost sure that if you develop such filter (or pay a freelance to do so) and propose it to the community, it will be reviewed and integrated in the software if well designed. JackDesBwa ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] overlay_cuda
We have overlay, overlay_qsv, overlay_opencl filters but don't have overlay_cuda for speed up transcoding videos on nvidia GPU only. Using sw overlay filter is slow down the transcoding because frames copied between CPU and GPU ram. Can You implement overlay_cuda filter, please? Alex ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] ffmpeg hwmap SW decoded frame to OpenCL and hwmap back to h264 for SW encode
I trying speed up process and avoid copy frames between GPU and CPU. But got error: "Segmentation fault: 11", so may be I'm doing something wrong? My full ffmpeg command and log here: ./ffmpeg -i ../720.mp4 -init_hw_device opencl=ocl:0.1 -filter_hw_device ocl -filter_complex "hwmap,avgblur_opencl=30,hwmap" -c:v h264 -an -t 10 -y ../out_blur.mp4 -loglevel debug ffmpeg version N-95621-g53c21c2d6b Copyright (c) 2000-2019 the FFmpeg developers built with Apple LLVM version 10.0.0 (clang-1000.10.44.4) configuration: --enable-fontconfig --enable-gpl --enable-libaom --enable-libass --enable-libbluray --enable-libfreetype --enable-libtheora --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265 --disable-ffplay --enable-nonfree --enable-opencl libavutil 56. 35.101 / 56. 35.101 libavcodec 58. 60.100 / 58. 60.100 libavformat 58. 33.100 / 58. 33.100 libavdevice 58. 9.100 / 58. 9.100 libavfilter 7. 66.100 / 7. 66.100 libswscale 5. 6.100 / 5. 6.100 libswresample 3. 6.100 / 3. 6.100 libpostproc 55. 6.100 / 55. 6.100 Splitting the commandline. Reading option '-i' ... matched as input url with argument '../720.mp4'. Reading option '-init_hw_device' ... matched as option 'init_hw_device' (initialise hardware device) with argument 'opencl=ocl:0.1'. Reading option '-filter_hw_device' ... matched as option 'filter_hw_device' (set hardware device used when filtering) with argument 'ocl'. Reading option '-filter_complex' ... matched as option 'filter_complex' (create a complex filtergraph) with argument 'hwmap,unsharp_opencl=lx=17:ly=17:la=5,hwmap'. Reading option '-c:v' ... matched as option 'c' (codec name) with argument 'h264'. Reading option '-an' ... matched as option 'an' (disable audio) with argument '1'. Reading option '-t' ... matched as option 't' (record or transcode "duration" seconds of audio/video) with argument '10'. Reading option '-y' ... matched as option 'y' (overwrite output files) with argument '1'. Reading option '../out_blur.mp4' ... matched as output url. Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'debug'. Finished splitting the commandline. Parsing a group of options: global . Applying option init_hw_device (initialise hardware device) with argument opencl=ocl:0.1. [AVHWDeviceContext @ 0x7fbb1ec08480] 1 OpenCL platforms found. [AVHWDeviceContext @ 0x7fbb1ec08480] 3 OpenCL devices found on platform "Apple". [AVHWDeviceContext @ 0x7fbb1ec08480] 0.1: Apple / HD Graphics 4000 Applying option filter_hw_device (set hardware device used when filtering) with argument ocl. Applying option filter_complex (create a complex filtergraph) with argument hwmap,unsharp_opencl=lx=17:ly=17:la=5,hwmap. Applying option y (overwrite output files) with argument 1. Applying option loglevel (set logging level) with argument debug. Successfully parsed a group of options. Parsing a group of options: input url ../720.mp4. Successfully parsed a group of options. Opening an input file: ../720.mp4. [NULL @ 0x7fbb1f818c00] Opening '../720.mp4' for reading [file @ 0x7fbb1ec3d600] Setting default whitelist 'file,crypto' [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fbb1f818c00] Format mov,mp4,m4a,3gp,3g2,mj2 probed with size=2048 and score=100 [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fbb1f818c00] ISO: File Type Major Brand: isom [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fbb1f818c00] Unknown dref type 0x206c7275 size 12 [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fbb1f818c00] Processing st: 0, edit list 0 - media time: 0, duration: 300300 [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fbb1f818c00] Before avformat_find_stream_info() pos: 4123312 bytes read:34929 seeks:1 nb_streams:1 [h264 @ 0x7fbb1f9ae800] nal_unit_type: 7(SPS), nal_ref_idc: 1 [h264 @ 0x7fbb1f9ae800] nal_unit_type: 8(PPS), nal_ref_idc: 1 [h264 @ 0x7fbb1f9ae800] nal_unit_type: 6(SEI), nal_ref_idc: 0 [h264 @ 0x7fbb1f9ae800] nal_unit_type: 5(IDR), nal_ref_idc: 1 [h264 @ 0x7fbb1f9ae800] Format yuv420p chosen by get_format(). [h264 @ 0x7fbb1f9ae800] Reinit context to 1280x720, pix_fmt: yuv420p [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fbb1f818c00] All info found [mov,mp4,m4a,3gp,3g2,mj2 @ 0x7fbb1f818c00] After avformat_find_stream_info() pos: 38244 bytes read:73125 seeks:2 frames:1 Input #0, mov,mp4,m4a,3gp,3g2,mj2, from '../720.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 encoder : Lavf58.33.100 Duration: 00:00:10.01, start: 0.00, bitrate: 3295 kb/s Stream #0:0(und), 1, 1/3: Video: h264 (High), 1 reference frame (avc1 / 0x31637661), yuv420p(left), 1280x720 [SAR 1:1 DAR 16:9], 0/1, 3293 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 60k tbc (default) Metadata: handler_name : ISO Media file produced by Google Inc. Created on: 10/24/2018. Successfully opened the file. [Parsed_unsharp_opencl_1 @ 0x7fbb1ec45640] Setting 'lx' to value '17' [Parsed_unsharp_opencl_1 @ 0x7fbb1ec45640] Setting 'ly' to value '17'
Re: [FFmpeg-user] TEE muxer and missing mpegts_flags
Hi Gyan, Actually most of the errors were made just because I wrote the command in the email rather than copying the command I was issuing. Actually I was already using columns and not commas, but definitely the error was in how I was using the -latm option. I though latm was related to the previous flag option, so I didn't think to suffix :a:0 Your help was extremely clarifying, thanks a lot. I've end up doing this: /usr/src/ffmpeg-4.1.3/ffmpeg -v verbose -hwaccel cuvid -c:v h264_cuvid -i "udp://226.45.23.147:2001?fifo_size=100" -vf scale_npp=1280:720:format=yuv420p,hwdownload -map 0:v -map 0:a -map 0:a -r 25 -g 50 -c:v h264_nvenc -c:a libfdk_aac -flags:a:0 +global_header -latm:a:0 1 -profile:a:0 aac_he -c:a:1 libfdk_aac -f tee "[f=mpegts:mpegts_flags=latm:select=\'0,1\']udp://239.99.33.33:7000?pkt_size=1316=15|[f=mpegts:select=\'0,2\']udp://239.99.33.33:6000?pkt_size=1316=15" And actually now does exactly what I need. Thank you, really Alex -Original Message- From: ffmpeg-user [mailto:ffmpeg-user-boun...@ffmpeg.org] On Behalf Of Gyan Sent: 26 June 2019 12:38 To: ffmpeg-user@ffmpeg.org Subject: Re: [FFmpeg-user] TEE muxer and missing mpegts_flags On 26-06-2019 03:56 PM, Alex Molon wrote: > Hi All, > > I need to live encode a video and send it to two separate UDP outputs with > different settings, since the only difference is the audio track, I was > thinking to use the TEE muxer but I don't know how to pass a mpegts_flag > > At the moment I'm using two different processes, but I'm wasting CPU > resources: > Process 1: ffmpeg -i input -c:v libx264 -c:a libfdk_aac -flags:a > +global_header -latm 1 -profile:a aac_he -mpegts_flags latm -f mpegts > udp://239.33.33.33:7000 > Process 2: ffmpeg -i input -c:v libx264 -c:a libfdk_aac -f mpegts > udp://239.33.33.33:6000 > > So I've tried with something like this: > > ffmpeg -i input -map 0:v -map 0:a -map 0:a -c:v libx264 -c:a libfdk_aac > -flags:a:0 +global_header -latm 1 -profile:a: aac_he -f tee > "[f=mpegts:mpegts_flags=latm,select=\'0,1\']udp://239.33.33.33:7000|[f=mpegts,select=\'0,2\']udp://239.33.33.33:6000" > > but it sems the option mpegts_flags=latm is totally ignored since I can see > this error: > > [LATM/LOAS muxer @ 0x55f4f9d038c0] Muxing MPEG-4 AOT 21 in LATM is not > supported > [tee @ 0x55f4f6689600] Slave > '[f=mpegts:mpegts_flags=latm:select='0,4']udp://239.5.99.120:7000?pkt_size=1316': > error writing header: Invalid data found when processing input > > Where am I wrong? > Any suggestion? A few things are wrong. Options for a slave muxer are separated by a colon ':' so f=mpegts:mpegts_flags=latm,select=\'0,1\' becomes f=mpegts:mpegts_flags=latm:select=\'0,1\' Same for f=mpegts,select=\'0,2\' Next, -latm 1 -profile:a: aac_he applies these options to both audio encodes. Change to -latm:a:0 1 -profile:a:0 aac_he Gyan ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] TEE muxer and missing mpegts_flags
Hi All, I need to live encode a video and send it to two separate UDP outputs with different settings, since the only difference is the audio track, I was thinking to use the TEE muxer but I don't know how to pass a mpegts_flag At the moment I'm using two different processes, but I'm wasting CPU resources: Process 1: ffmpeg -i input -c:v libx264 -c:a libfdk_aac -flags:a +global_header -latm 1 -profile:a aac_he -mpegts_flags latm -f mpegts udp://239.33.33.33:7000 Process 2: ffmpeg -i input -c:v libx264 -c:a libfdk_aac -f mpegts udp://239.33.33.33:6000 So I've tried with something like this: ffmpeg -i input -map 0:v -map 0:a -map 0:a -c:v libx264 -c:a libfdk_aac -flags:a:0 +global_header -latm 1 -profile:a: aac_he -f tee "[f=mpegts:mpegts_flags=latm,select=\'0,1\']udp://239.33.33.33:7000|[f=mpegts,select=\'0,2\']udp://239.33.33.33:6000" but it sems the option mpegts_flags=latm is totally ignored since I can see this error: [LATM/LOAS muxer @ 0x55f4f9d038c0] Muxing MPEG-4 AOT 21 in LATM is not supported [tee @ 0x55f4f6689600] Slave '[f=mpegts:mpegts_flags=latm:select='0,4']udp://239.5.99.120:7000?pkt_size=1316': error writing header: Invalid data found when processing input Where am I wrong? Any suggestion? Thanks in advance. Alex Molon ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org https://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Streaming overseas
On the receiver side, set the latency around 600ms https://ffmpeg.org/ffmpeg-protocols.html#srt -Original Message- From: ffmpeg-user [mailto:ffmpeg-user-boun...@ffmpeg.org] On Behalf Of Manuel Alejandro Sent: 03 February 2019 16:18 To: FFmpeg user questions Subject: Re: [FFmpeg-user] Streaming overseas Hi Mustafa Al Ani, How do you deal with the delay variations when loss of unrecoverable packets occurs? In my case, the delay decreases. For example, it goes from 400ms to 0ms. The playback jumps forward. At the moment I do not know how to avoid this. On Sun, Feb 3, 2019 at 7:36 AM Mustafa Al Ani wrote: > I second Alex, > > We use SRT to send low latency stream from Copenhagen Denmark to NY, Ohio, > and LA in the USA. > > R, > Mustafa > > On Thu, Dec 20, 2018 at 4:30 PM Alex Molon > wrote: > > > SRT forever! > > > > Simple, stable, low latency, transports any MPEG-TS content. > > It is designed expressely for hi quality streams over internet. > > > > Alex :) > > > > -Original Message- > > From: ffmpeg-user [mailto:ffmpeg-user-boun...@ffmpeg.org] On Behalf Of > > Louis Letourneau > > Sent: 05 November 2018 18:44 > > To: FFmpeg user questions > > Subject: Re: [FFmpeg-user] Streaming overseas > > > > > Hello Louis, > > > > > > Have you take a look at SRT protocol ? > > > > > > Source code : https://github.com/Haivision/srt > > > > > > Latest FFmpeg handle this protocol (if i am not wrong). > > > > > > > > > I didn't know about it. I will try it as soon as i can. It seems > > interesting. > > > > Louis > > > > > > > ___ > > ffmpeg-user mailing list > > ffmpeg-user@ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > > To unsubscribe, visit link above, or email > > ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". > > ___ > > ffmpeg-user mailing list > > ffmpeg-user@ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > > To unsubscribe, visit link above, or email > > ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". > ___ > ffmpeg-user mailing list > ffmpeg-user@ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > To unsubscribe, visit link above, or email > ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] filter_complex and map. Am i confused or bug?
Hi Gyan, Thanks a lot, Seems to work as expected now :) Alex Molon -Original Message- From: ffmpeg-user [mailto:ffmpeg-user-boun...@ffmpeg.org] On Behalf Of Gyan Sent: 31 January 2019 12:59 To: ffmpeg-user@ffmpeg.org Subject: Re: [FFmpeg-user] filter_complex and map. Am i confused or bug? On 31-01-2019 06:11 PM, Alex Molon wrote: > -b:v:vidout1 2M -minrate:v:vidout1 2M -maxrate:v:vidout1 2M \ <- I specify > that [vidout1] has to be encoded in 2M/2M/2M > > -b:v:vidout2 1M -minrate:v:vidout2 1M -maxrate:v:vidout2 1M \ <- I specify > that [vidout2] has to be encoded in 1M/1M/1M Stream specifiers for output stream codec options s e.g. v:vidout1 can refer to ordinal index only, so -b:v:vidout1 becomes -b:v:1 assuming vidout1 is mapped after vidout2. Gyan ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] filter_complex and map. Am i confused or bug?
In this case I used a "real life" scenario, picking up a live stream. The result is the same: Command executed: ffmpeg -i udp://226.45.23.147:2001?fifo_size=100 -filter_complex "[0:v]scale=720:576[vidout1];[0:v]scale=640:480[vidout2]" -c:v libx264 -c:a aac -b:v:vidout1 2M -minrate:v:vidout1 2M -maxrate:v:vidout1 2M -b:v:vidout2 1M -minrate:v:vidout2 1M -maxrate:v:vidout2 1M -map [vidout1] -map [vidout2] -map 0:a -t 10 -y -f mpegts test.ts Expectation: A file called test.ts containing 3 elementary streams: One video stream 720x576 encoded at 2M with minimum and maximum bitrate 2M One video stream 640x480 encoded at 1M with minimum and maximum bitrate 1M One audio stream AAC Result: A file called test ts containing 3 elementary streams: One video stream at 720x576 unexpectedly encoded at 1M One video stream at 640x480 unexpectedly encoded with no specific settings One audio stream AAC Suspected cause: I think the following section of the command is parsed wrongly or I’m totally confused b:v:vidout1 2M -minrate:v:vidout1 2M -maxrate:v:vidout1 2M -b:v:vidout2 1M -minrate:v:vidout2 1M -maxrate:v:vidout2 1M Output: ffmpeg version 4.1 Copyright (c) 2000-2018 the FFmpeg developers built with gcc 7 (Ubuntu 7.3.0-27ubuntu1~18.04) configuration: --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --enable-gpl --disable-stripping --enable-avresample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libass --enable-libbs2b --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libmp3lame --enable-libopenjpeg --enable-libopenmpt --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libzvbi --enable-opengl --enable-sdl2 --enable-libdrm --enable-chromaprint --enable-frei0r --enable-libx264 --disable-shared --enable-static --enable-libnpp --enable-nonfree --enable-nvenc --enable-cuvid --enable-libfdk-aac libavutil 56. 22.100 / 56. 22.100 libavcodec 58. 35.100 / 58. 35.100 libavformat58. 20.100 / 58. 20.100 libavdevice58. 5.100 / 58. 5.100 libavfilter 7. 40.101 / 7. 40.101 libavresample 4. 0. 0 / 4. 0. 0 libswscale 5. 3.100 / 5. 3.100 libswresample 3. 3.100 / 3. 3.100 libpostproc55. 3.100 / 55. 3.100 Input #0, mpegts, from 'udp://226.45.23.147:2001?fifo_size=100': Duration: N/A, start: 17355.533200, bitrate: N/A Program 1408 Metadata: service_name: BTV HD service_provider: Stream #0:0[0x1959]: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p(tv, bt709, top first), 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 50 tbr, 90k tbn, 50 tbc Stream #0:1[0x195a]: Audio: mp2 ([4][0][0][0] / 0x0004), 48000 Hz, stereo, s16p, 128 kb/s Stream mapping: Stream #0:0 (h264) -> scale (graph 0) Stream #0:0 (h264) -> scale (graph 0) scale (graph 0) -> Stream #0:0 (libx264) scale (graph 0) -> Stream #0:1 (libx264) Stream #0:1 -> #0:2 (mp2 (native) -> aac (native)) Press [q] to stop, [?] for help Output #0, mpegts, to 'test.ts': Metadata: encoder : Lavf58.20.100 Stream #0:0: Video: h264 (libx264), yuv420p(top coded first (swapped)), 720x576 [SAR 64:45 DAR 16:9], q=-1--1, 1000 kb/s, 25 fps, 90k tbn, 25 tbc Metadata: encoder : Lavc58.35.100 libx264 Side data: cpb: bitrate max/min/avg: 100/0/100 buffer size: 0 vbv_delay: -1 Stream #0:1: Video: h264 (libx264), yuv420p, 640x480 [SAR 4:3 DAR 16:9], q=-1--1, 25 fps, 90k tbn, 25 tbc Metadata: encoder : Lavc58.35.100 libx264 Side data: cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1 Stream #0:2: Audio: aac (LC), 48000 Hz, stereo, fltp, 128 kb/s Metadata: encoder : Lavc58.35.100 aac frame= 228 fps= 39 q=-1.0 Lq=-1.0 size=1832kB time=00:00:10.00 bitrate=1500.3kbits/s speed=1.69x Thanks in advance, Alex Molon -Original Message- From: ffmpeg-user [mailto:ffmpeg-user-boun...@ffmpeg.org] On Behalf Of Alex Molon Sent: 31 January 2019 12:52 To: FFmpeg user questions Subject: Re: [FFmpeg-user] filter_complex and map. Am i confused or bug? Hi Carl, My problem actually is not the order itself, but how the single tracks are actually encoded The complete command is: ffmpeg -i INPUTFILE -filter_complex "[0:v]scale=720:576[vidout1];[0:v]scale=640:480[vidout2]" -c:v libx264 -c:a aac -b:v:vidout1 2M -minrate:v:vidout1 2M -maxrate:v:vidout1 2M -b:v:vidout2 1M -minrate:v:vidout2 1M -maxrate:v:vidout2 1M -map [vidout2] -map [vidout1] -map 0:a -f mpegts test.ts And the result is that: instead to have a stream with one video track 720x576@2M/2M/2M, one video track 640x480@1M/1M/1m and one audio track as a result I have a stream with one
Re: [FFmpeg-user] filter_complex and map. Am i confused or bug?
Hi Carl, My problem actually is not the order itself, but how the single tracks are actually encoded The complete command is: ffmpeg -i INPUTFILE -filter_complex "[0:v]scale=720:576[vidout1];[0:v]scale=640:480[vidout2]" -c:v libx264 -c:a aac -b:v:vidout1 2M -minrate:v:vidout1 2M -maxrate:v:vidout1 2M -b:v:vidout2 1M -minrate:v:vidout2 1M -maxrate:v:vidout2 1M -map [vidout2] -map [vidout1] -map 0:a -f mpegts test.ts And the result is that: instead to have a stream with one video track 720x576@2M/2M/2M, one video track 640x480@1M/1M/1m and one audio track as a result I have a stream with one video track with one video track 640x480@1M/1M/1M, one video track 720x576 with no specific encoding settings and one audio track. Or, in alternative (if I change the order of the map commands) one video track 720x576@1M/1M/1M, one video track 640x480 with no specific encoding settings and one audio track. Output #0, mpegts, to 'test.ts': Metadata: encoder : Lavf58.20.100 Stream #0:0: Video: h264 (libx264), yuv420p(top coded first (swapped)), 640x480 [SAR 4:3 DAR 16:9], q=-1--1, 1000 kb/s, 25 fps, 90k tbn, 25 tbc Metadata: encoder : Lavc58.35.100 libx264 Side data: cpb: bitrate max/min/avg: 100/0/100 buffer size: 0 vbv_delay: -1 Stream #0:1: Video: h264 (libx264), yuv420p, 720x576 [SAR 64:45 DAR 16:9], q=-1--1, 25 fps, 90k tbn, 25 tbc Metadata: encoder : Lavc58.35.100 libx264 Side data: cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1 Stream #0:2: Audio: aac (LC), 48000 Hz, stereo, fltp, 128 kb/s Metadata: Thanks in advance, Alex -Original Message- From: ffmpeg-user [mailto:ffmpeg-user-boun...@ffmpeg.org] On Behalf Of Carl Eugen Hoyos Sent: 31 January 2019 12:45 To: FFmpeg user questions Subject: Re: [FFmpeg-user] filter_complex and map. Am i confused or bug? 2019-01-31 13:41 GMT+01:00, Alex Molon : > Basically I have a file with a single video track and a single audio track. > > What I want to achieve is: > > > > a) Scale the video track to 720x576 and encode it in H264 @ 2M with 2M > minrate and 2M maxrate > > b) Scale the video track to 640x480 and encode it in H264 @ 1M with 1M > minrate and 1M maxrate > > c) Pick the audio track and encode it in aac > d) Mux all the tracks together in this order: first Video 640x480, second > Video 720x576, third Audio Transport streams do not know about "order", this concept simply doesn't apply. If this does not answer your question, please provide your actual command line (no variables) including the complete, uncut console output if you need support here. Carl Eugen ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] filter_complex and map. Am i confused or bug?
cpb: bitrate max/min/avg: 0/0/0 buffer size: 0 vbv_delay: -1 Stream #0:2: Audio: aac (LC), 48000 Hz, stereo, fltp, delay 1024, 128 kb/s Metadata: encoder : Lavc58.35.100 aac Basically now vout1 is encoded "almost" as expectedexcept for the minrate What am I doing wrong? Did I totally misunderstood how to use filter_complex and map, or is there any possible bug preventing the mechanism to work properly? Thanks in advance, Alex Molon ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] piping raw audio data from c buffer
Hi, I have written C code that subscribes to a UDP multicast group, and stores the raw (hex) incoming audio data into a buffer. I was wondering if there is a way of piping this buffer with the raw audio data to ffmpeg within the C code in order to play the audio. I have seen examples of .wav files being piped to ffmpeg within C files, but not a buffer. I am not experienced with ffmpeg and any help/tips would be greatly appreciated. Thank you. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] down sampling
Hi, Hopefully this is an appropriate question for the forums. My goal is to receive a live audio stream that is being sampled at 131,072 Hz and re-sample it at 44.1 kHz before outputting it through my computers speakers. Is this a task ffmpeg can perform? Thank you. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Streaming overseas
SRT forever! Simple, stable, low latency, transports any MPEG-TS content. It is designed expressely for hi quality streams over internet. Alex :) -Original Message- From: ffmpeg-user [mailto:ffmpeg-user-boun...@ffmpeg.org] On Behalf Of Louis Letourneau Sent: 05 November 2018 18:44 To: FFmpeg user questions Subject: Re: [FFmpeg-user] Streaming overseas > Hello Louis, > > Have you take a look at SRT protocol ? > > Source code : https://github.com/Haivision/srt > > Latest FFmpeg handle this protocol (if i am not wrong). > I didn't know about it. I will try it as soon as i can. It seems interesting. Louis > ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] ffplay RTP: dropping old packet received too late
Did you try to use -f rtp_mpegts ? -Original Message- From: ffmpeg-user [mailto:ffmpeg-user-boun...@ffmpeg.org] On Behalf Of Gáll Péter Sent: 28 September 2018 09:49 To: ffmpeg-user@ffmpeg.org Subject: [FFmpeg-user] ffplay RTP: dropping old packet received too late I have a machine which streams live audio to a multicast address: ffmpeg -f alsa -i hw:0 -acodec libmp3lame -ab 128k -f rtp rtp:// 239.123.13.101:56789. I catch that stream with ffplay: ffplay rtp:// 239.123.13.101:56789.It works fine until I stop ffmpeg and start it again. If I restart ffmpeg (when ffplay still running), I get the error message from ffplay: RTP: dropping old packet recieved too late. Is there a solution to "re-receice" the newly started stream, without restarting ffplay? Thanks, Peter ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] Can't copy ripped DVD stream due to timestamp issues
ut #0, matroska, to 'VTS_01_1.mkv': Metadata: encoder : Lavf58.12.100 Stream #0:0: Video: mpeg2video (Main) (mpg2 / 0x3267706D), yuv420p(tv, top first), 720x480 [SAR 8:9 DAR 4:3], q=2-31, 29.97 fps, 29.97 tbr, 1k tbn, 29.97 tbc Stream #0:1: Audio: ac3 ([0] [0][0] / 0x2000), 48000 Hz, stereo, fltp, 256 kb/s Stream mapping: Stream #0:1 -> #0:0 (copy) Stream #0:2 -> #0:1 (copy) Press [q] to stop, [?] for help [matroska @ 0x565112939d00] Timestamps are unset in a packet for stream 0. This is deprecated and will stop working in the future. Fix your code to set the timestamps properly av_interleaved_write_frame(): Invalid argument frame=1 fps=0.0 q=-1.0 Lsize= 1kB time=00:00:00.00 bitrate=N/A speed= 0x video:52kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown Conversion failed! Is there a way that I can concatenate these videos, even if I have to recompute the timestamps? -- Sincerely, Alex Parrill ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] Hap_q_alpha is it possible?
Hello, I think I'm missing something. ffmpeg -i source.mov -vcodec hap -format hap target.mov OK ffmpeg -i source.mov -vcodec hap -format hap_alpha target.mov OK ffmpeg -i source.mov -vcodec hap -format hap_q target.mov OK This is not working, what's switch for Hap Q Alpha ? ffmpeg -i source.mov -vcodec hap -format hap_q_alpha target.mov I'm on win7 Thanks Alex -- Sent from: http://www.ffmpeg-archive.org/ ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] Alex Mihalev - Question on having output to a Decklink card
Hello there! I am new to FFMPEG and would like to apologize if I am asking an obvious question, though, I looked and didn't find much information on the topic. It is about playing a Multicast stream to Decklink Monitor Card SDI output. I am using Debian 8 with its "apt-get install ffmpeg" sort of FFMPEG, Do i need to make a compile and somehow insert the Decklink into it? Thank you! ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] 4K 60Hz Directshow Video Capture
Thank you for that suggestion Roger, I can't believe I forgot about the preset options. Executing the following on a RAM disk gets me about .9x performance, letting me capture about 5 seconds worth of video before the buffers overflow and I lose frames. Not perfect but enough for what I need. I think this is the best I'm going to get unless the mfg makes a better driver and adds yuv444p support or I get a better CPU. I really appreciate for everyone's input. ffmpeg -f dshow -video_size 3840x2160 -framerate 6/1001 -rtbufsize 21000 -pixel_format bgr24 -i video="MZ0380 PCI, Analog 01 Capture" -c:v libx264rgb -preset ultrafast -crf 0 -pix_fmt bgr24 -t 00:00:10 -r 6/1001 out.avi One more question, what is the command to use the maximum buffer size? -rtbufsize INT_MAX doesn't work. Thanks. -Original Message- From: ffmpeg-user [mailto:ffmpeg-user-boun...@ffmpeg.org] On Behalf Of Roger Pack Sent: Tuesday, February 27, 2018 8:03 PM To: FFmpeg user questions Subject: Re: [FFmpeg-user] 4K 60Hz Directshow Video Capture consider also libx264 "ultrafast" preset, GL! On Tue, Feb 13, 2018 at 7:57 AM, Alex P <ale...@avenview.com> wrote: > I think I've figured it out. When I use nv12 or yuv420p as the input > and output pixel format, I get x1 performance. If I use bgr24/rgb24 as > the input and yuv444p as the output, I get around x0.3. > > But even when I use bgr0 for the input and output, I get less than x1. > Does anyone know what exactly bgr0 is? I can't find any information > about it in my googling. > > In your testing James, what was the pixel format? > > -Original Message- > From: ffmpeg-user [mailto:ffmpeg-user-boun...@ffmpeg.org] On Behalf Of > James Girotti > Sent: Monday, February 12, 2018 7:03 PM > To: FFmpeg user questions > Subject: Re: [FFmpeg-user] 4K 60Hz Directshow Video Capture > > > > > ffmpeg -f dshow -video_size 3840x2160 -framerate 6/1001 > > -rtbufsize > > 21 -pixel_format bgr24 -i video="MZ0380 PCI, Analog 01 Capture" > > -c:v h264_nvenc -preset lossless -f null - Gives me the same error > > > > That's surprising, I can get about 200fps using file-based/ramdisk > "-c:v h264_nvenc -preset -lossless". Have you also tried "-c:v > hevc_nvenc -preset lossless"? What's the encoding FPS that you're > getting? You technically shouldn't be able get much more than 60fps as > that's what your capture card is supplying. Can you monitor the "Video > Engine Utilization" during encoding? In linux it's listed in the > nvidia-settings GUI or "nvidia-smi dmon" on the CLI will show enc/dec%. > > > > ffmpeg -f dshow -video_size 3840x2160 -framerate 6/1001 > > -rtbufsize > > 21 -pixel_format bgr24 -i video="MZ0380 PCI, Analog 01 Capture" > > -c:v rawvideo -f null - > > Gets me nearly x1 performance when executing from a ram disk but > > > > ffmpeg -f dshow -video_size 3840x2160 -framerate 6/1001 > > -rtbufsize > > 21 -pixel_format bgr24 -i video="MZ0380 PCI, Analog 01 Capture" > > -c:v rawvideo raw.nut > > Only gets me x0.5 and the buffer overflows. > > > > Is there a way of accelerating rawvideo decoding? Would using my > > colleagues 1080 make a difference? Thanks. > > > I think raw-video is already decoded. So no way/need to accelerate that. > You might try a different pix_fmt from your capture card while using > hw-encoding, but you'd have to test. I don't know the internals, i.e. > when the pixel format is converted during hw-encoding. So it might > make a difference. > > Changing pixel formats might be a concern if you are trying to achieve > "100% lossless" capture. I've read that yuv444p should be sufficient > colorspace for bgr24. > > There isn't a lot of info out there on encoding speed differences > based on GPU models. It's a complex subject, but from what I have > observed the ASIC is tied to the GPU clock (I have observed that GPU > clock speed increases as ASIC load increases). If that's true, then a > GTX 1080, with it's higher max clock, could have faster encoding, but I have > no data to back that up only. > ___ > ffmpeg-user mailing list > ffmpeg-user@ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > To unsubscribe, visit link above, or email > ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". > > ___ > ffmpeg-user mailing list > ffmpeg-user@ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > To unsubscribe, visit link above, or email > ffmpeg-user-requ...@ffm
Re: [FFmpeg-user] Fisheye-equirectangular in real time
What is the source of the video? From my quick googling, lens information and other metadata is needed for fisheye to equirectangular conversion. -Alex P -Original Message- From: ffmpeg-user [mailto:ffmpeg-user-boun...@ffmpeg.org] On Behalf Of Edward Bellamy Sent: Wednesday, February 28, 2018 7:25 AM To: ffmpeg-user@ffmpeg.org Subject: [FFmpeg-user] Fisheye-equirectangular in real time Hi, I am looking for a way to convert my fisheye video to equirectsangular in realtime or as low latency as possible. I was thinking about using ffmpeg with 2 SDI capture cards for input and output and using the remap filter to relight the pixels, Do you have any experience with a build like this? Is it possible to do input and output via SDI? How do i get this running as low latency as possible? Kind regards Edward Bellamy | Creative & Technical Director AU +61 476 100 722 http://www.staplesvr.com <http://www.staplesvr.com.au/> ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] 4K 60Hz Directshow Video Capture
Thank you for the offer James, but I think I know how to proceed. Looking at the pixel formats for various implementations of x264 and x265, I'm not seeing any that support bgr24, which is the only uncompressed format offered by my capture card, out of yuyv422, yuv420p, nv12, bgr0 and bgr24 Does anyone know of a lossless encoder that inputs and outputs bgr24? -Original Message- From: ffmpeg-user [mailto:ffmpeg-user-boun...@ffmpeg.org] On Behalf Of James Girotti Sent: Tuesday, February 13, 2018 1:33 PM To: FFmpeg user questions Subject: Re: [FFmpeg-user] 4K 60Hz Directshow Video Capture On Tue, Feb 13, 2018 at 6:57 AM, Alex P <ale...@avenview.com> wrote: > I think I've figured it out. When I use nv12 or yuv420p as the input > and output pixel format, I get x1 performance. If I use bgr24/rgb24 as > the input and yuv444p as the output, I get around x0.3. > Looks like switching pixel formats highly impacts performance. It would be better for your capture card to output, for example, yuv444p. I can't tell from the specs if it can do that though. Careful selecting nv12 as format for output, my quick test showed that the final output was yuv420p. > In your testing James, what was the pixel format? I was testing yuv420p samples as that is what was available to me at the time. I have made a yuv444p using testsrc. My poor Thuban cannot decode this FFv1 at realtime and raw-video filesize is gigantic. So I made a lossless hevc yuv444p. Surprisingly (or maybe not) hevc_cuvid can't decode it! Again, my poor Thuban cannot decode real-time, but there's some hope: FFmpeg encoding speed was about 35-40 fps and hw-encoder utilization topped out at 40%. So there's still a lot of headroom in the hw-encoder. Rough theoretical calculation: I could get 100fps hw-encoding which is ~1.7X I got about the same speed for h264_nvenc lossless. I got similar results using a 3 second raw yuv444p video input file. If there are other pix_fmts you would like me to test, let me know. I'll do my best to try. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] 4K 60Hz Directshow Video Capture
I think I've figured it out. When I use nv12 or yuv420p as the input and output pixel format, I get x1 performance. If I use bgr24/rgb24 as the input and yuv444p as the output, I get around x0.3. But even when I use bgr0 for the input and output, I get less than x1. Does anyone know what exactly bgr0 is? I can't find any information about it in my googling. In your testing James, what was the pixel format? -Original Message- From: ffmpeg-user [mailto:ffmpeg-user-boun...@ffmpeg.org] On Behalf Of James Girotti Sent: Monday, February 12, 2018 7:03 PM To: FFmpeg user questions Subject: Re: [FFmpeg-user] 4K 60Hz Directshow Video Capture > > ffmpeg -f dshow -video_size 3840x2160 -framerate 6/1001 -rtbufsize > 21 -pixel_format bgr24 -i video="MZ0380 PCI, Analog 01 Capture" > -c:v h264_nvenc -preset lossless -f null - Gives me the same error > That's surprising, I can get about 200fps using file-based/ramdisk "-c:v h264_nvenc -preset -lossless". Have you also tried "-c:v hevc_nvenc -preset lossless"? What's the encoding FPS that you're getting? You technically shouldn't be able get much more than 60fps as that's what your capture card is supplying. Can you monitor the "Video Engine Utilization" during encoding? In linux it's listed in the nvidia-settings GUI or "nvidia-smi dmon" on the CLI will show enc/dec%. > ffmpeg -f dshow -video_size 3840x2160 -framerate 6/1001 -rtbufsize > 21 -pixel_format bgr24 -i video="MZ0380 PCI, Analog 01 Capture" > -c:v rawvideo -f null - > Gets me nearly x1 performance when executing from a ram disk but > > ffmpeg -f dshow -video_size 3840x2160 -framerate 6/1001 -rtbufsize > 21 -pixel_format bgr24 -i video="MZ0380 PCI, Analog 01 Capture" > -c:v rawvideo raw.nut > Only gets me x0.5 and the buffer overflows. > Is there a way of accelerating rawvideo decoding? Would using my > colleagues 1080 make a difference? Thanks. I think raw-video is already decoded. So no way/need to accelerate that. You might try a different pix_fmt from your capture card while using hw-encoding, but you'd have to test. I don't know the internals, i.e. when the pixel format is converted during hw-encoding. So it might make a difference. Changing pixel formats might be a concern if you are trying to achieve "100% lossless" capture. I've read that yuv444p should be sufficient colorspace for bgr24. There isn't a lot of info out there on encoding speed differences based on GPU models. It's a complex subject, but from what I have observed the ASIC is tied to the GPU clock (I have observed that GPU clock speed increases as ASIC load increases). If that's true, then a GTX 1080, with it's higher max clock, could have faster encoding, but I have no data to back that up only. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] 4K 60Hz Directshow Video Capture
Hi James, I get the same issue with this command: ffmpeg -f dshow -video_size 3840x2160 -framerate 6/1001 -rtbufsize 21 -pixel_format bgr24 -i video="MZ0380 PCI, Analog 01 Capture" -c:v h264_nvenc -preset lossless -f null - Is there a way to accelerate the decoding of rawvideo? ffmpeg -f dshow -video_size 3840x2160 -framerate 6/1001 -rtbufsize 21 -pixel_format bgr24 -i video="MZ0380 PCI, Analog 01 Capture" -c:v rawvideo -f null - Gets me nearly 1x performance. -Original Message- From: ffmpeg-user [mailto:ffmpeg-user-boun...@ffmpeg.org] On Behalf Of James Girotti Sent: Monday, February 12, 2018 3:31 PM To: FFmpeg user questions Subject: Re: [FFmpeg-user] 4K 60Hz Directshow Video Capture Hi Alex, I looked at your attached log. It appears to me that using libx265, your computer cannot encode fast enough. It appears your computer is encoding about 10.5 fps, but your input is about 60fps. So a lot of frames get buffered every second while they're waiting to be encoded. This leads to the full rtbuffer and then dropped frames. You could try increasing your rtbufsize, but using libx265 your computer will not be able to keep up and eventually the buffer will fill and frames will be dropped. > Any help would be appreciated. 1080p capture right to rawvideo is perfect. > I > would like to use NVENC in lossless mode if anyone has experience with > that. > If you look at the fps that you can do at 1080p, it will be much higher and that's why you don't get buffer over-runs. I also have a GTX 1050 Ti that I use for encoding. It will do hevc lossless at about 150fps, input size 3840x2160@60hz (all in RAM): ffmpeg -hwaccel cuvid -c:v hevc_cuvid -i /dev/shm/foo.4k60.hevc.mkv -c:v hevc_cuvid -preset lossless /dev/shm/test.mkv Based on my limited test, your GPU should be able to keep up and not drop frames, YMMV. I'm not sure what impact you will have because your input is raw-video (positive or negative). You shouldn't include "-hwaccel cuvid" on your cli, because you won't be doing hw-decoding/transcoding. HTH, -J ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] 4K 60Hz Directshow Video Capture
Hi James, ffmpeg -f dshow -video_size 3840x2160 -framerate 6/1001 -rtbufsize 21 -pixel_format bgr24 -i video="MZ0380 PCI, Analog 01 Capture" -c:v h264_nvenc -preset lossless -f null - Gives me the same error ffmpeg -f dshow -video_size 3840x2160 -framerate 6/1001 -rtbufsize 21 -pixel_format bgr24 -i video="MZ0380 PCI, Analog 01 Capture" -c:v rawvideo -f null - Gets me nearly x1 performance when executing from a ram disk but ffmpeg -f dshow -video_size 3840x2160 -framerate 6/1001 -rtbufsize 21 -pixel_format bgr24 -i video="MZ0380 PCI, Analog 01 Capture" -c:v rawvideo raw.nut Only gets me x0.5 and the buffer overflows. Is there a way of accelerating rawvideo decoding? Would using my colleagues 1080 make a difference? Thanks. -Alex P -Original Message- From: ffmpeg-user [mailto:ffmpeg-user-boun...@ffmpeg.org] On Behalf Of James Girotti Sent: Monday, February 12, 2018 3:31 PM To: FFmpeg user questions Subject: Re: [FFmpeg-user] 4K 60Hz Directshow Video Capture Hi Alex, I looked at your attached log. It appears to me that using libx265, your computer cannot encode fast enough. It appears your computer is encoding about 10.5 fps, but your input is about 60fps. So a lot of frames get buffered every second while they're waiting to be encoded. This leads to the full rtbuffer and then dropped frames. You could try increasing your rtbufsize, but using libx265 your computer will not be able to keep up and eventually the buffer will fill and frames will be dropped. > Any help would be appreciated. 1080p capture right to rawvideo is perfect. > I > would like to use NVENC in lossless mode if anyone has experience with > that. > If you look at the fps that you can do at 1080p, it will be much higher and that's why you don't get buffer over-runs. I also have a GTX 1050 Ti that I use for encoding. It will do hevc lossless at about 150fps, input size 3840x2160@60hz (all in RAM): ffmpeg -hwaccel cuvid -c:v hevc_cuvid -i /dev/shm/foo.4k60.hevc.mkv -c:v hevc_cuvid -preset lossless /dev/shm/test.mkv Based on my limited test, your GPU should be able to keep up and not drop frames, YMMV. I'm not sure what impact you will have because your input is raw-video (positive or negative). You shouldn't include "-hwaccel cuvid" on your cli, because you won't be doing hw-decoding/transcoding. HTH, -J ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] 4K 60Hz Directshow Video Capture
I've tried all manner of encoders but get the same buffer error. I've actually gotten a 64-bit fork of VirtualDub to work and capture everything but it occasionally drops frames and I like the CLI of ffmpeg. I guess I could use the VD CLI. Can anyone advise? I could use Linux but I'd rather not as I know this company's Linux drivers are less developed than their Windows drivers. -Original Message- From: ffmpeg-user [mailto:ffmpeg-user-boun...@ffmpeg.org] On Behalf Of William Caulfield Sent: Monday, February 12, 2018 1:11 PM To: FFmpeg user questions Subject: Re: [FFmpeg-user] 4K 60Hz Directshow Video Capture On Mon, Feb 12, 2018 at 7:37 AM, Alex P <ale...@avenview.com> wrote: > Windows 10 > > Intel i7-8700K > > GTX 1050Ti > > 16GB DDR4 > > SATA Samsung Evo > > > > I'm using a Yuan 4K60 capture card with the end goal of capturing the > video in a lossless format. I need it to be lossless as the clips will > be used to test hardware h264 encoders. > > > > Product page: > http://www.yuan.com.tw/products/capture/4k/sc560n1_lv_hdmi2.htm > > Directshow is supported. > > > > Command: ffmpeg -f dshow -video_size 3840x2160 -framerate 59.9 > -rtbufsize > 21 -pixel_format bgr24 -i video="MZ0380 PCI, Analog 01 Capture" > -c:v > libx265 -x265-params lossless=1 out1.avi > > > > I have also tried writing to null: ffmpeg -f dshow -video_size > 3840x2160 -framerate 59.9 -rtbufsize 21 -pixel_format bgr24 -i > video="MZ0380 PCI, Analog 01 Capture" -c:v libx265 -x265-params > lossless=1 -f null - > > > > I get the same error (see attatched) and I have to force quit by > pressing Ctrl-C a bunch. Error also occurs if I write to a RAM disk. > > > > [dshow @ 01dda20ca3a0] real-time buffer [MZ0380 PCI, Analog 01 > Capture] [video input] too full or near too full (63% of size: > 21 [rtbufsize parameter])! frame dropped! > > > > Any help would be appreciated. 1080p capture right to rawvideo is perfect. > I > would like to use NVENC in lossless mode if anyone has experience with > that. > > > > haven't used Windows / Directshow for a few years, but back in the day I would have attempted this using VirtualDub and HuffYUV. I believe that HuffYUV is available for ffmpeg, but can't say that I've ever tried it. -- *William Caulfield *| *ContentBridge Systems* ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] 4K 60Hz Directshow Video Capture
Windows 10 Intel i7-8700K GTX 1050Ti 16GB DDR4 SATA Samsung Evo I'm using a Yuan 4K60 capture card with the end goal of capturing the video in a lossless format. I need it to be lossless as the clips will be used to test hardware h264 encoders. Product page: http://www.yuan.com.tw/products/capture/4k/sc560n1_lv_hdmi2.htm Directshow is supported. Command: ffmpeg -f dshow -video_size 3840x2160 -framerate 59.9 -rtbufsize 21 -pixel_format bgr24 -i video="MZ0380 PCI, Analog 01 Capture" -c:v libx265 -x265-params lossless=1 out1.avi I have also tried writing to null: ffmpeg -f dshow -video_size 3840x2160 -framerate 59.9 -rtbufsize 21 -pixel_format bgr24 -i video="MZ0380 PCI, Analog 01 Capture" -c:v libx265 -x265-params lossless=1 -f null - I get the same error (see attatched) and I have to force quit by pressing Ctrl-C a bunch. Error also occurs if I write to a RAM disk. [dshow @ 01dda20ca3a0] real-time buffer [MZ0380 PCI, Analog 01 Capture] [video input] too full or near too full (63% of size: 21 [rtbufsize parameter])! frame dropped! Any help would be appreciated. 1080p capture right to rawvideo is perfect. I would like to use NVENC in lossless mode if anyone has experience with that. C:\Users\alex.p\Desktop\ffmpeg\bin>ffmpeg -f dshow -video_size 3840x2160 -framerate 59.9 -rtbufsize 21 -pixel_format bgr24 -i video="MZ0380 PCI, Analog 01 Capture" -c:v libx265 -x265-params lossless=1 -f null - ffmpeg version git-2017-12-22-e3b2c85 Copyright (c) 2000-2017 the FFmpeg developers built with gcc 7.2.0 (GCC) configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-bzlib --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-amf --enable-cuda --enable-cuvid --enable-d3d11va --enable-nvenc --enable-dxva2 --enable-avisynth --enable-libmfx libavutil 56. 6.100 / 56. 6.100 libavcodec 58. 8.100 / 58. 8.100 libavformat58. 3.100 / 58. 3.100 libavdevice58. 0.100 / 58. 0.100 libavfilter 7. 7.100 / 7. 7.100 libswscale 5. 0.101 / 5. 0.101 libswresample 3. 0.101 / 3. 0.101 libpostproc55. 0.100 / 55. 0.100 Input #0, dshow, from 'video=MZ0380 PCI, Analog 01 Capture': Duration: N/A, start: 433919.57, bitrate: N/A Stream #0:0: Video: rawvideo, bgr24, 3840x2160, 59.90 fps, 59.90 tbr, 1k tbn, 1k tbc Stream mapping: Stream #0:0 -> #0:0 (rawvideo (native) -> hevc (libx265)) Press [q] to stop, [?] for help x265 [info]: HEVC encoder version 2.6+13-6b079854e56e x265 [info]: build info [Windows][GCC 7.2.0][64 bit] 8bit x265 [info]: using cpu capabilities: MMX2 SSE2Fast LZCNT SSSE3 SSE4.2 AVX FMA3 BMI2 AVX2 x265 [info]: Main 4:4:4 profile, Level-8.5 (Main tier) x265 [info]: Thread pool created using 12 threads x265 [info]: Slices : 1 x265 [info]: frame threads / pool features : 3 / wpp(34 rows) x265 [info]: Coding QT: max CU size, min CU size : 64 / 8 x265 [info]: Residual QT: max TU size, max depth : 32 / 1 inter / 1 intra x265 [info]: ME / range / subpel / merge : hex / 57 / 2 / 2 x265 [info]: Keyframe min / max / scenecut / bias: 25 / 250 / 40 / 5.00 x265 [info]: Cb/Cr QP Offset : 6 / 6 x265 [info]: Lookahead / bframes / badapt: 20 / 4 / 2 x265 [info]: b-pyramid / weightp / weightb : 1 / 1 / 0 x265 [info]: References / ref-limit cu / depth : 3 / on / on x265 [info]: Rate Control: Lossless x265 [info]: tools: rd=3 psy-rd=2.00 rskip signhide tmvp strong-intra-smoothing x265 [info]: tools: lslices=8 deblock sao Output #0, null, to 'pipe:': Metadata: encoder : Lavf58.3.100 Stream #0:0: Video: hevc (libx265), gbrp, 3840x2160, q=2-31, 59.90 fps, 59.90 tbn, 59.90 tbc Metadata: encoder : Lavc58.8.100 libx265 [dshow @ 01dda20ca3a0] real-time buffer [MZ0380 PCI, Analog 01 Capture] [video input] too full or near too full (63% of size: 21 [rtbufsize parameter])! frame dropped! [dshow @ 01dda20ca3a0] real-time buffer [MZ0380 PCI, Analog 01 Capture] [video input] too full or near too full (66% of size: 21 [rtbufsize parameter])! frame dropped! [dshow @ 01dda20ca3a0] real-time buffer [MZ0380 PCI, Analog 01 Capture] [video input] too full or near too full (69% of size: 21 [rtbufsize parameter])! frame dropped! [dshow @ 01dda20ca3a0] real-time buffer [MZ0380 PCI,
Re: [FFmpeg-user] Bitrate range when streaming to udp
I'm sorry my explanation may be really unclear. I'll make one more attempt :) I have a *.ts file. I want to stream it via udp. No transcoding or other similar things, just one stream of MPEG TS. And I'd like to have CBR of the stream, as constant as possible. When trying to stream the file (with ffmpeg, vlc, tsplay etc.), I get a terrible bitrate dispersion which was shown at the picture (see link in the first message of this topic). I was very unhappy with this. And I was surprised that all tools I have used give me terrible bitrate dispersion. May the problem be not in tools but in the source file? Is it possible to prepare the file better in order to get better result stream bitrate? Or, is the result stream bitrate independent of the source file? WBR Alex 12.01.2018 16:20, Carl Eugen Hoyos пишет: > 2018-01-12 12:56 GMT+01:00 Alex Alex <win2000...@hotmail.com>: > >> Does multicast bitrate depend on a source file features? Is it >> possible to prepare the file better for getting smoother output >> bitrate? > Are you asking about adaptive bitrate? That either needs files > encoded to the desired bitrates in advance or real-time encoding. > Sorry, I am not sure I understand the question. > > Please do not top-post here, Carl Eugen > ___ > ffmpeg-user mailing list > ffmpeg-user@ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > To unsubscribe, visit link above, or email > ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Bitrate range when streaming to udp
OK thank you. I'm sorry. Just one more question (maybe, stupid one). Does multicast bitrate depend on a source file features? Is it possible to prepare the file better for getting smoother output bitrate? WBR Alex. 10.01.2018 04:36, Carl Eugen Hoyos пишет: > 2018-01-09 23:37 GMT+01:00 Alex Alex <win2000...@hotmail.com>: > >> I try to solve a problem which seems to be very simple. There is a >> source file which is to be streamed as multicast (UDP) infinitely. > If you don't want to reencode the file, there must be a better > tool than FFmpeg (I don't know of any). > Among the reasons is that our mpegts muxer is not bug-free. > > Generally, please never post an excerpt of the command line and > console output when asking for help here, always post the command > line you tested together with the complete, uncut console output. > > Carl Eugen > ___ > ffmpeg-user mailing list > ffmpeg-user@ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > To unsubscribe, visit link above, or email > ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] Bitrate range when streaming to udp
Hello! I try to solve a problem which seems to be very simple. There is a source file which is to be streamed as multicast (UDP) infinitely. Ok, not even infinitely, just once. What is the problem? Terrible bitrate range. Here is my source file: Input #0, mpegts, from 'test.ts': Duration: 00:08:14.46, start: 1.458667, bitrate: 4490 kb/s Program 1 Metadata: service_name : Service01 service_provider: FFmpeg Stream #0:0[0x100]: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p(progressive), 1920x1080 [SAR 1:1 DAR 16:9], 25 fps, 25 tbr, 90k tbn, 50 tbc Stream #0:1[0x101](und): Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000 Hz, stereo, fltp, 139 kb/s I try to stream it: ffmpeg -threads 0 -re -stream_loop -1 -i test.ts -c copy -f mpegts "udp://@235.5.2.193:1234?pkt_size=1316" But when I analyze the result multicast I see bitrate jumps from almost zero to 12 Mbps: http://image.ibb.co/cqrhPw/ffmpeg_stream.png I made a lot of attempts to change some parameters but nothing helped. Is it possible to get smoother bitrate? I don't even dream about CBR but I would be happy to get 3-5 Mbps at least. WBR Alex ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] Encoding 4K 60Hz lossless from a capture card
Windows 10 64-bit Ryzen 7 GTX 1080 32GB RAM Hi all, I'm trying to encode 4K 30/60Hz video in a lossless format from a 4K capture card and everything I've tried gives me a similar error as in the linked image, [real-time buffer too full or near too full frame dropped] https://cloud.githubusercontent.com/assets/4932401/22171307/ef5c9864-df58-11e6-8821-4b74ce3f32d0.png This is the command I've tried most recently: ffmpeg.exe -f dshow -video_size 3840x2160 -framerate 30 -pixel_format bgr24 -rtbufsize INT_MAX -i video="MZ0380 PCI, Analog 01 Capture" -vf fps=30 out%d.BMP With the images dumped to a 10G RAM disk or 850 EVO. I'm doing this to skip the encoding step. I get the same error when encoding with h265 lossless and NVENC h265 lossless. I need the video to be lossless as it will be used to test hardware h265 encoders. Video source is a 4K Blu-ray. Any help would be greatly appreciated. Thank you. -Alex P ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] RTSP to HLS re-stream stops with “No more output streams to write to, finishing.”
Did you try with something like this? ffmpeg -i url://whatever/link.ext?fifo_size=100_nonfatal=1 I don't use RTSP in my environment but I use a lot of UDP streams (coming from outside my network and with recurrent drops even for few seconds) and ffmpeg is able to compensate without dying. Of course the encoding stops... Alex -Original Message- From: ffmpeg-user [mailto:ffmpeg-user-boun...@ffmpeg.org] On Behalf Of Tim Williams Sent: 18 October 2017 14:11 To: ffmpeg-user@ffmpeg.org Subject: Re: [FFmpeg-user] RTSP to HLS re-stream stops with “No more output streams to write to, finishing.” Hi All, Can I assume that nobody knows the answer to this? If that's the case can I ask for some advice on how to move forward and find a solution, it seems to me there is a clear bug in ffmpeg since short interruptions (a few seconds) when using a remote stream causes ffmpeg to stop, effectively preventing it from being used reliably in such cases. I can't see how you are ever going to get that degree of reliability across anything other than a local LAN. I have seen a few other instances of this problem being reported elsewhere, but there is never a solution. So my questions are: - Should I be reporting this to the bug tracker either as bug or a feature request for an option enabling outputs to cope with a discontinuity in the input when it comes from a source which might have gap in the data due to network issues. - Is ffmpeg fundamentally unsuited to the task of re-streaming a remote RTSP feed? - Would running two instances of ffmpeg with the raw video being piped between them allow the interruptions to be tolerated? - Should I be using some other tool to read the RTSP stream and then pipe that into ffmpeg to do the HLS encapsulation? - Would I be better off running the RTSP>HLS encoding on a machine located on the local LAN with the segment data then being automatically synced to the remote server? This seems to defeat the object of having a streaming protocol like RTSP, but it would isolate ffmpeg from short network interruptions, which a file sync process will cope with far better. I'm not expecting ffmpeg to tolerate big interruptions, we're talking about a few seconds here and I'm not worried about frame drops during the interruptions, I just want to avoid having to repeatedly restart ffmpeg, I had about 15 restarts today during ~4 hours of streaming, many of which caused client playback to stall. If ffmpeg could be made to tolerate the interruption, then playback wouldn't stall and all the viewer would see is a glitch in the picture. Any help would be much appreciated, I've spent many hours trying and failing to solve this! Tim W -- Tim Williams BSc MSc MBCS AutoTrain 58 Jacoby Place Priory Road Edgbaston Birmingham B5 7UW United Kingdom Web : http://www.autotrain.org, http://www.utrain.info Tel : +44 (0)844 487 4117 AutoTrain is a trading name of EuroMotor-AutoTrain LLP Registered in the United Kingdom, number: OC317070. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] Multiple live input stops when one input is lost
Hi all, I'd like to remux two live SPTS inputs (in this case two channels downlinked from different dishes) in a single MPTS with two programs in, and I'm doing it with this command: ffmpeg -i udp://226.45.23.147:2001?fifo_size=100 -i udp://229.45.20.40:2000?fifo_size=100 -c copy -map 0:0 -map 0:1 -map 1:0 -map 1:1 -program title=Program_1:st=0:st=1 -program title=Program_2:st=2:st=3 -f mpegts udp://239.86.86.86:8686?pkt_size=1316 Everything works as expected, but if I miss one input ffmpeg just hangs waiting for the input to come back. Is there any way to force ffmpeg to go ahead streaming only the available input until the lost input is back again? Kind regards, Alex Molon CTO alex.mo...@vision247.com<mailto:alex.mo...@vision247.com> Vision247(tm) Chiswick Park 2nd Floor, Building 10 566 Chiswick High Road London, W4 5XS, UK t:+44 20 7636 7474 f:+44 20 7908 7299 www.vision247.com<http://www.vision247.com> Follow us on: twitter<https://twitter.com/vision_247> | linkedin<http://www.linkedin.com/company/vision247?trk=hb_tab_compy_id_2456018> [cid:image001.png@01D33B86.3B97D8E0] Complete broadcast solutions Any opinions expressed in this Email are those of the individual and not necessarily the company. This Email and any files transmitted with it are confidential and intended solely for the use of the intended recipient. If you are not the intended recipient, or the person responsible for delivering to the intended recipient, be advised that you have received this Email in error and that any use of it is strictly prohibited. If you have received this Email in error please notify us via email i...@vision247.com<mailto:%20i...@vision247.com> and then delete the Email, including any attachments, and destroy any copies of it. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] SRT Secure Reliable Transport support.
Unfortunately my development abilities are equal to zero :) But this feature would be absolutely great! Hopefully someone will do it -Original Message- From: ffmpeg-user [mailto:ffmpeg-user-boun...@ffmpeg.org] On Behalf Of Moritz Barsnick Sent: 29 September 2017 12:31 To: FFmpeg user discussions Subject: Re: [FFmpeg-user] SRT Secure Reliable Transport support. On Thu, Sep 28, 2017 at 17:40:03 +0100, Alex Molon wrote: > Any hope for this to be supported in FFMPEG anytime soon? > http://www.srtalliance.org/ > It would be something reeally nice The request is there, but apparently noone has taken the time yet: https://trac.ffmpeg.org/ticket/6348 You can volunteer! Send patches! ;-) Moritz ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] SRT Secure Reliable Transport support.
Hi All, Any hope for this to be supported in FFMPEG anytime soon? http://www.srtalliance.org/ It would be something reeally nice Kind regards, Alex Molon CTO alex.mo...@vision247.com<mailto:alex.mo...@vision247.com> Vision247(tm) Chiswick Park 2nd Floor, Building 10 566 Chiswick High Road London, W4 5XS, UK t:+44 20 7636 7474 f:+44 20 7908 7299 www.vision247.com<http://www.vision247.com> Follow us on: twitter<https://twitter.com/vision_247> | linkedin<http://www.linkedin.com/company/vision247?trk=hb_tab_compy_id_2456018> [cid:image001.png@01D33880.D1019EB0] Complete broadcast solutions Any opinions expressed in this Email are those of the individual and not necessarily the company. This Email and any files transmitted with it are confidential and intended solely for the use of the intended recipient. If you are not the intended recipient, or the person responsible for delivering to the intended recipient, be advised that you have received this Email in error and that any use of it is strictly prohibited. If you have received this Email in error please notify us via email i...@vision247.com<mailto:%20i...@vision247.com> and then delete the Email, including any attachments, and destroy any copies of it. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] missing h264_cuvid
I think the problem is more on the decoder you are using. Apparently your ffmpeg is compiled to support cuvid but if your stream is dvb-s mpeg2 maybe you should use this decoder: V. mpeg2_cuvid Nvidia CUVID MPEG2VIDEO decoder (codec mpeg2video) Alex -Original Message- From: ffmpeg-user [mailto:ffmpeg-user-boun...@ffmpeg.org] On Behalf Of tasos Sent: 08 August 2017 19:28 To: ffmpeg-user@ffmpeg.org Subject: Re: [FFmpeg-user] missing h264_cuvid Hello. I'm not sure but you have to compile at least with --enable-cuda --enable-cuvid --enable-nvenc. Can you try compiling with those enabled? Moreover i don't know if you want/need --enable-opencl On 8/8/2017 8:54 PM, Daniel wrote: > Hello everyone, > > I am trying to decode stream using h264_cuvid decoder but > unfortunately i get the following error : "Unrecognized hwaccel: > h264_cuvid. > Supported hwaccels: vdpau vaapi cuvid " same time if i request > "/usr/local/bin/ffmpeg -decoders |grep -i h264 " i get this result: > > ffmpeg version N-86054-g2171dfa Copyright (c) 2000-2017 the FFmpeg > developers > built with gcc 5.3.1 (Ubuntu 5.3.1-14ubuntu2.1) 20160413 > configuration: --prefix=/usr/src/ffmpeg/ffmpeg_build > --pkg-config-flags=--static > --extra-cflags=-I/usr/src/ffmpeg/ffmpeg_build/include > --extra-ldflags=-L/usr/src/ffmpeg/ffmpeg_build/lib > --bindir=/usr/src/ffmpeg/bin --enable-gpl --enable-libass > --enable-libfdk-aac --enable-libfreetype --enable-libmp3lame > --enable-libopus --enable-libtheora --enable-libvorbis --enable-libvpx > --enable-libx264 --enable-libx265 --enable-nonfree --enable-nvenc > --enable-opencl --enable-librtmp --enable-libv4l2 --enable-libvpx > libavutil 55. 62.100 / 55. 62.100 > libavcodec 57. 95.101 / 57. 95.101 > libavformat57. 72.101 / 57. 72.101 > libavdevice57. 7.100 / 57. 7.100 > libavfilter 6. 89.100 / 6. 89.100 > libswscale 4. 7.101 / 4. 7.101 > libswresample 2. 8.100 / 2. 8.100 > libpostproc54. 6.100 / 54. 6.100 > VFS..D h264 H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 > VD h264_vdpau H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 > (VDPAU acceleration) (codec h264) > V. h264_cuvid Nvidia CUVID H264 decoder (codec h264) > > can you tell me if ffmpeg is missing h264_cuvid decoder or it could be > something else.The stream i am trying to decode is dvb-s mpeg2 that's > why i doubt about the decoder i have to use(i would like to use > hwaccel decoder). > > Thank you > > ___ > ffmpeg-user mailing list > ffmpeg-user@ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > To unsubscribe, visit link above, or email > ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Possible bug in TEE muxer.
It wooorkkks :) Thanks a lot! Really :) Alex -Original Message- From: ffmpeg-user [mailto:ffmpeg-user-boun...@ffmpeg.org] On Behalf Of Moritz Barsnick Sent: 18 July 2017 00:20 To: FFmpeg user discussions Subject: Re: [FFmpeg-user] Possible bug in TEE muxer. On Mon, Jul 17, 2017 at 14:57:37 +0100, Alex Molon wrote: > # ffmpeg -v verbose -i udp://226.45.23.147:2001 -c:v h264_nvenc -c:a > aac -strict 2 -f tee -map 0 > [f=mpegts:hls_segment_filename=sftp://compliance:complitest@localhost/ > home/auser/test/test.ts]sftp://compliance:complitest@localhost/home/au > ser/test/test.m3u8 > And the result is always the same: > > [tee @ 0x556de1794140] No option found near > "//compliance:complitest@localhost/home/auser/test/test.ts]sftp://compliance:complitest@localhost/home/auser/test/test.m3u8; > [tee @ 0x556de1794140] All tee outputs failed. > > I've tried with single, double and triple escape before the ":" signs > (how suggested in https://trac.ffmpeg.org/ticket/5692 regarding > something similar) but the result is always the same :( I tried it like this, and it works: $ ffmpeg -re -f lavfi -i testsrc2 -c:v libx264 -t 33 -map 0 -f tee -use_localtime 1 "[f=hls:use_localtime=1:hls_segment_filename='sftp\://user\:password@sunshine/home/user/tmp/sftp-%Y-%m-%d-%H-%M-%S']sftp://user:password@sunshine/home/user/tmp/sftp.m3u8; This also works - this is your approach, but you don't use *any* quotation marks. That's something I would avoid, as soon as e.g. square brackets come into play. That's probably why you need *even more* (four) backslashes - for the shell this time. $ ffmpeg -re -f lavfi -i testsrc2 -c:v libx264 -t 33 -map 0 -f tee -use_localtime 1 [f=hls:use_localtime=1:hls_segment_filename=sftp://user:password@sunshine/home/user/tmp/sftp-%Y-%m-%d-%H-%M-%S]sftp://user:password@sunshine/home/user/tmp/sftp.m3u8 BTW: - "f=mpegts" is incorrect, as you wanted hls (as in the original, non-tee command line). - "-strict 2" doesn't do anything. The native aac encoder used to require "-strict -2", but no longer does. But that was a "minus two". ;-) So proud that I got it to work :-D Moritz ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Possible bug in TEE muxer.
Update. Well... I'm trying with this command: # ffmpeg -v verbose -i udp://226.45.23.147:2001 -c:v h264_nvenc -c:a aac -strict 2 -f tee -map 0 [f=mpegts:hls_segment_filename=sftp://compliance:complitest@localhost/home/auser/test/test.ts]sftp://compliance:complitest@localhost/home/auser/test/test.m3u8 And the result is always the same: [tee @ 0x556de1794140] No option found near "//compliance:complitest@localhost/home/auser/test/test.ts]sftp://compliance:complitest@localhost/home/auser/test/test.m3u8; [tee @ 0x556de1794140] All tee outputs failed. I've tried with single, double and triple escape before the ":" signs (how suggested in https://trac.ffmpeg.org/ticket/5692 regarding something similar) but the result is always the same :( Same happens if I try to use the ftp:// protocol. -Original Message- From: ffmpeg-user [mailto:ffmpeg-user-boun...@ffmpeg.org] On Behalf Of Alex Molon Sent: 15 July 2017 14:05 To: FFmpeg user discussions; FFmpeg user questions Subject: Re: [FFmpeg-user] Possible bug in TEE muxer. Sorry i wrote ssh:// by mistake actually i'm having the issue with sftp:// urls Get Outlook for Android<https://aka.ms/ghei36> On Fri, Jul 14, 2017 at 7:03 PM +0100, "Moritz Barsnick" <barsn...@gmx.net<mailto:barsn...@gmx.net>> wrote: On Fri, Jul 14, 2017 at 10:33:08 +0100, Alex Molon wrote: > Basically, if I launch ffmpeg in this way: > > # ffmpeg -i input -f hls -hls_segment_filename > ssh://user:password@host/path/filename-%Y-%m-%d-%H-%M-%S > ssh://user:password@host/path/filename.m3u8 > > everything works as expected. Both playlist and chunks are correctly sent to > the ssh server and saved regularly. Could you show us the complete, uncut console output (but blank out the passwords, please)? I can find no indication whatsoever that ffmpeg supports ssh:// URLs. > If I launch ffmpeg in this way: > > # ffmpeg -i input -f tee > [f=whatever]|[f=hls:use_localtime=1:hls_segment_filename=ssh\://user\: > password@host/path/filename-%Y-%m-%d-%H-%M-%S]ssh://user:password@host > /path/filename.m3u8 > > the playlist is correctly sent to the SSH server, but the chunks are not > saved. I'll try to reproduce once I understand how to use ssh://. Perhaps if you add "-loglevel verbose", ffmpeg might show you how it interprets the options you are passing. Moritz ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". On Fri, Jul 14, 2017 at 10:33:08 +0100, Alex Molon wrote: > Basically, if I launch ffmpeg in this way: > > # ffmpeg -i input -f hls -hls_segment_filename > ssh://user:password@host/path/filename-%Y-%m-%d-%H-%M-%S > ssh://user:password@host/path/filename.m3u8 > > everything works as expected. Both playlist and chunks are correctly sent to > the ssh server and saved regularly. Could you show us the complete, uncut console output (but blank out the passwords, please)? I can find no indication whatsoever that ffmpeg supports ssh:// URLs. > If I launch ffmpeg in this way: > > # ffmpeg -i input -f tee > [f=whatever]|[f=hls:use_localtime=1:hls_segment_filename=ssh\://user\: > password@host/path/filename-%Y-%m-%d-%H-%M-%S]ssh://user:password@host > /path/filename.m3u8 > > the playlist is correctly sent to the SSH server, but the chunks are not > saved. I'll try to reproduce once I understand how to use ssh://. Perhaps if you add "-loglevel verbose", ffmpeg might show you how it interprets the options you are passing. Moritz ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Possible bug in TEE muxer.
Sorry i wrote ssh:// by mistake actually i'm having the issue with sftp:// urls Get Outlook for Android<https://aka.ms/ghei36> On Fri, Jul 14, 2017 at 7:03 PM +0100, "Moritz Barsnick" <barsn...@gmx.net<mailto:barsn...@gmx.net>> wrote: On Fri, Jul 14, 2017 at 10:33:08 +0100, Alex Molon wrote: > Basically, if I launch ffmpeg in this way: > > # ffmpeg -i input -f hls -hls_segment_filename > ssh://user:password@host/path/filename-%Y-%m-%d-%H-%M-%S > ssh://user:password@host/path/filename.m3u8 > > everything works as expected. Both playlist and chunks are correctly sent to > the ssh server and saved regularly. Could you show us the complete, uncut console output (but blank out the passwords, please)? I can find no indication whatsoever that ffmpeg supports ssh:// URLs. > If I launch ffmpeg in this way: > > # ffmpeg -i input -f tee > [f=whatever]|[f=hls:use_localtime=1:hls_segment_filename=ssh\://user\:password@host/path/filename-%Y-%m-%d-%H-%M-%S]ssh://user:password@host/path/filename.m3u8 > > the playlist is correctly sent to the SSH server, but the chunks are not > saved. I'll try to reproduce once I understand how to use ssh://. Perhaps if you add "-loglevel verbose", ffmpeg might show you how it interprets the options you are passing. Moritz ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". On Fri, Jul 14, 2017 at 10:33:08 +0100, Alex Molon wrote: > Basically, if I launch ffmpeg in this way: > > # ffmpeg -i input -f hls -hls_segment_filename > ssh://user:password@host/path/filename-%Y-%m-%d-%H-%M-%S > ssh://user:password@host/path/filename.m3u8 > > everything works as expected. Both playlist and chunks are correctly sent to > the ssh server and saved regularly. Could you show us the complete, uncut console output (but blank out the passwords, please)? I can find no indication whatsoever that ffmpeg supports ssh:// URLs. > If I launch ffmpeg in this way: > > # ffmpeg -i input -f tee > [f=whatever]|[f=hls:use_localtime=1:hls_segment_filename=ssh\://user\:password@host/path/filename-%Y-%m-%d-%H-%M-%S]ssh://user:password@host/path/filename.m3u8 > > the playlist is correctly sent to the SSH server, but the chunks are not > saved. I'll try to reproduce once I understand how to use ssh://. Perhaps if you add "-loglevel verbose", ffmpeg might show you how it interprets the options you are passing. Moritz ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] Possible bug in TEE muxer.
Hi All, I was playing around with the tee muxer and I wasn't able to do something, but I actually don't know if it's a bug or I'm doing something wrong. Apparently the muxer is handling umproperly the " : " character. Basically, if I launch ffmpeg in this way: # ffmpeg -i input -f hls -hls_segment_filename ssh://user:password@host/path/filename-%Y-%m-%d-%H-%M-%S ssh://user:password@host/path/filename.m3u8 everything works as expected. Both playlist and chunks are correctly sent to the ssh server and saved regularly. If I launch ffmpeg in this way: # ffmpeg -i input -f tee [f=whatever]|[f=hls:use_localtime=1:hls_segment_filename=ssh\://user\:password@host/path/filename-%Y-%m-%d-%H-%M-%S]ssh://user:password@host/path/filename.m3u8 the playlist is correctly sent to the SSH server, but the chunks are not saved. I also tried to remove the backslashes before the : but nothing happened. I found the same behaviour in different versions of ffmpeg. What am I doing wrong? Thanks in advance. Cheers Alex Molon ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] FPS drop when transcode 3 Full HD channel to SD
How many CPUs / cores are you using? Transcoding FullHD in SD, without using any hardware accelleration for more than two channels seems a quite huge load. It can be simply your CPU cannot go fast enough -Original Message- From: ffmpeg-user [mailto:ffmpeg-user-boun...@ffmpeg.org] On Behalf Of Hendrik Karsimin Sent: 13 June 2017 14:03 To: ffmpeg-user@ffmpeg.org Subject: [FFmpeg-user] FPS drop when transcode 3 Full HD channel to SD Warning: This message has had one or more attachments removed Warning: (encode_cna.out.log, encode_okto.out.log, when 3 ffmpeg running.zip, encode_suria.out.log). Warning: Please read the "vision247.com-Attachment-Warning.txt" attachment(s) for more information. Hi, I've been trying to transcode the Full HD Channel (1920x1080) to SD (720x576). When I transcode 1 channel, the resulting video is no problem. So are when transcode 2 channel. But when the 3rd channel I tried to transcode start. the FPS start to drop on all the channel I transcoding. I'm not sure if it's a computer hardware limitation or the software limitation. But I have tried to change some parameter of ffmpeg, and settle on the last ffmpeg configuration I use now. Attached is the information I collect on the PC I use to transcode. The Ubuntu Version I use is Ubuntu 14.04.5 #lsb_release -a No LSB modules are available. Distributor ID: Ubuntu Description:Ubuntu 14.04.5 LTS Release:14.04 Codename: trusty #uname -a Linux Compaq 3.13.0-119-generic #166-Ubuntu SMP Wed May 3 12:18:55 UTC 2017 x86_64 x86_64 x86_64 GNU/Linux Need advice to resolve this problem. *Thanks & regards,* Hendrik ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Unknown encoder 'libfdk-aac'
And in any case, even if compiled correctly, the encoder is called "libfdk_aac" with the underscore, not libfdk-aac Cheers Alex Molon From: ffmpeg-user [ffmpeg-user-boun...@ffmpeg.org] On Behalf Of Carl Eugen Hoyos [ceffm...@gmail.com] Sent: 24 February 2017 00:46 To: FFmpeg user questions Subject: Re: [FFmpeg-user] Unknown encoder 'libfdk-aac' 2017-02-23 23:45 GMT+01:00 JD <jd1...@gmail.com>: > ~/bin/ffmpeg.d/ffmpeg -i video_Z10.mp4 -movflags +faststart -vb 8000k -c:a > libfdk-aac -ab 384k -s 1920x1080 -y video_Z10-libfdk.mp4 > > ffmpeg version 3.0.2-static http://johnvansickle.com/ffmpeg/ libfdk is non-free, you have to compile FFmpeg yourself if you want to use libfdk-aac. Carl Eugen ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] AAC LATM Encoding
I've used this setup and it works well... # ffmpeg -i input -c:v libx264 -c:a libfdk_aac -flags:a +global_header -latm 1 -strict 2 -profile:a aac_he -mpegts_flags latm -f mpegts output Of course you need to set all the parameters you prefer related to video/audio bandwidth, bitrate and so on. Cheers Alex Molon -Original Message- From: ffmpeg-user [mailto:ffmpeg-user-boun...@ffmpeg.org] On Behalf Of Carl Eugen Hoyos Sent: 23 February 2017 13:24 To: FFmpeg user questions Subject: Re: [FFmpeg-user] AAC LATM Encoding 2017-02-23 10:37 GMT+01:00 Rashed <mail2rashed...@gmail.com>: > I am trying to generate AAC LATM content but was unable to figure out > how to get that done with ffmpeg. The fdk encoder has a latm option that you have to set to 1 to get latm output.. Carl Eugen ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] Scaling a video with multiple different resolutions
I have a webm file that has multiple different resolutions in it (it's a screen capture of a window that changes dimensions). https://www.dropbox.com/s/ptueirabmmht0fr/4be7fdb7-d7e9-41b4-ba26-e20a3eeb6026.webm?dl=0 Is there any way to "normalize" the dimensions and output a video of a constant resolution, having e.g. black borders around the video when the resolutions change If you play the video on the dropbox page above, in chrome or firefox, how that looks is what I'd like to output. Tried messing around with scale, setsar and setdar parameters but I can't figure out how to get output that's not just distorted Any help appreciated, thanks! ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] Question about video compositing
Ah, thanks a lot for the suggestion, but I should have been clearer that I need to do this in an automated fashion for arbitrary sets of videos so it has to be command-line (or a library I guess) so that I can integrate it into an automated pipeline in my app. Thanks, Alex On Tue, Jan 10, 2017 at 3:37 AM Steve Boyer <steveboye...@gmail.com> wrote: > > Any suggestions on if either of these approaches is better, or any > > alternatives? Thanks! > > > > Hi! I've done something similar to doing this, but I ended up using a > non-linear video editor. Specifically, I used kdenlive. It can do keyframe > animation, so combine that with fade-ins/fade from blacks (audio/video > filters) as well as fade-outs/fade to blacks, and you can do multiple > tracks combined into a single output video with animations/fades when a > stream ends. The downsides are that kdenlive, despite being the best video > editor on linux (IMHO), is a little buggy, is all CPU-based when it comes > to rendering and output file, for stability purposes it is recommended to > use a single thread, you will have to manually put things together and time > them, and need to use a GUI to do it all. > > I'd be happy to help with suggestions if you go this route, but understand > if you want to go a different way (and I'd be interested if anyone has > other suggestions how this can be accomplished via FFmpeg or CLI tools). > > Steve > > > > ___ > > ffmpeg-user mailing list > > ffmpeg-user@ffmpeg.org > > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > > > To unsubscribe, visit link above, or email > > ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". > ___ > ffmpeg-user mailing list > ffmpeg-user@ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > > To unsubscribe, visit link above, or email > ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe". ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] Question about video compositing
I have a question about video compositing. I’ve included the text of the question below but I’ve also put it in a gist for easier to read formatting here: https://gist.github.com/alexspeller/aefdd5a6d7100d28d0bbc4838527f797 I have multiple mp4 video files and I want to composite them into a single video. Each stream is an mp4 video. They are of different lengths, and each file also has audio. The tricky thing is, I want the layout to change depending on how many streams are currently visible. As a concrete example, say I have 3 video files: | File | Duration | Start | End | |---|--|---|-| | a.mp4 | 30s | 0s| 30s | | b.mp4 | 10s | 10s | 20s | | c.mp4 | 15s | 15s | 30s | So at t=0 seconds, I want the video to look like this: ``` +-+ | | | | | | | | | a.mp4 | | | | | | | | | | | +-+ ``` At t=10s, I want the video to look like this: ``` +--++ | || | || | | a.mp4 | | || | ++ | b.mp4| | | | | | | | | | | +--+ ``` At t=15s, I want the video to look like this: ``` +--++ | || | || | | a.mp4 | | || | ++ | b.mp4|| | || | | c.mp4 | | || | ++ | | +--+ ``` And at t=20s until the end, I want the video to look like this: ``` +--++ | || | || | | a.mp4 | | || | ++ | c.mp4| | | | | | | | | | | +--+ ``` Ideally there would be some animated transitions between the states, but that's not essential. I have found two possible approaches that might work, but I'm not sure what the best one is. The first is using [filters](https://trac.ffmpeg.org/wiki/Create%20a%20mosaic%20out%20of%20several%20input%20videos) to acheive the result, but I'm not sure if it will cope well with (a) the changing layouts and (b) keeping the audio without any artefacts when the layout changes. The other approach I thought of would be exporting all frames to images, building new frames with imagemagick, and then layering the new frames on top of the audio like in [this blog post](https://broadcasterproject.wordpress.com/2010/05/18/how-to-layerremix-videos-with-free-command-line-tools/). Any suggestions on if either of these approaches is better, or any alternatives? Thanks! ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] Split RTSP stream to file and socket
Hello, As the subject suggests, I have an RTSP stream (from an IP camera). I'm trying to create two outputs: a file recording the stream, and a unix socket. Specifically, the file output is just an MP4 file. For the socket output I want to pass the stream through the 'fps' filter first. I tried this: ffmpeg -i rtsp://ip/stream -filter_complex '[0:v]split=2[in1][in2];[in2]fps=4[out2]' -map '[in1]' out.mp4 -map '[out2]' unix://video_stream I intend to use some python scripts to do something with the 4FPS video coming in from that socket. I thought about (and tried) using a named pipe but couldn't get that working. Plus to my knowledge pipes are blocking and I would like ffmpeg to continue writing to the MP4 file regardless of the python script processing input from the socket or pipe. Is this possible? ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] Invalid data found when processing input with TP-Link tl-sc3430
Hello I'm trying to connect tl-sc3430 to ZoneMinder, and ffmpeg yields an error, not too verbose rtsp://192.168.51.149/video.h264: Invalid data found when processing input setting loglevel to debug shows only command line parsing. VLC on Windows play the stream just fine. rtsp://192.168.51.149/video.3gp works ok, but 160x120 is too little. ffmpeg is built as ffmpeg version 2.8.7 Copyright (c) 2000-2016 the FFmpeg developers built with FreeBSD clang version 3.4.1 (tags/RELEASE_34/dot1-final 208032) 20140512 configuration: --prefix=/usr/local --mandir=/usr/local/man --datadir=/usr/local/share/ffmpeg --pkgconfigdir=/usr/local/libdata/pkgconfig --enable-shared --enable-gpl --enable-postproc --enable-avfilter --enable-avresample --enable-pthreads --disable-libstagefright-h264 --disable-libutvideo --disable-libsoxr --cc=cc --extra-cflags='-msse -I/usr/local/include/vorbis -I/usr/local/include' --extra-ldflags='-L/usr/local/lib ' --extra-libs=-lpthread --disable-libaacplus --disable-indev=alsa --disable-outdev=alsa --disable-libopencore-amrnb --disable-libopencore-amrwb --disable-libass --disable-libbs2b --disable-libcaca --disable-libcdio --disable-libcelt --disable-libdc1394 --disable-debug --enable-htmlpages --disable-libfaac --disable-libfdk-aac --enable-ffserver --disable-libflite --disable-fontconfig --disable-libfreetype --enable-frei0r --disable-libfribidi --disable-libgme --disable-libgsm --enable-iconv --disable-libilbc --disable-indev=jack --disable-ladspa --disable-libmp3lame --disable-libbluray --enable-mmx --disable-libmodplug --disable-openal --disable-indev=openal --disable-opencl --enable-libopencv --disable-opengl --enable-libopenh264 --disable-libopenjpeg --disable-libopus --disable-libpulse --disable-indev=pulse --disable-outdev=pulse --disable-libquvi --enable-runtime-cpudetect --enable-librtmp --enable-libschroedinger --disable-ffplay --disable-outdev=sdl --disable-libsmbclient --disable-libsnappy --disable-libspeex --enable-sse --disable-libssh --enable-libtheora --disable-libtwolame --disable-libv4l2 --disable-indev=v4l2 --disable-outdev=v4l2 --disable-vaapi --disable-vdpau --disable-libvidstab --enable-libvorbis --disable-libvo-aacenc --disable-libvo-amrwbenc --enable-libvpx --disable-libwavpack --disable-libwebp --disable-x11grab --enable-libx264 --disable-libx265 --disable-libxcb --enable-libxvid --disable-outdev=xv --disable-libzmq --disable-libzvbi --disable-gnutls --enable-openssl --enable-version3 --enable-nonfree libavutil 54. 31.100 / 54. 31.100 libavcodec 56. 60.100 / 56. 60.100 libavformat56. 40.101 / 56. 40.101 libavdevice56. 4.100 / 56. 4.100 libavfilter 5. 40.101 / 5. 40.101 libavresample 2. 1. 0 / 2. 1. 0 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 2.101 / 1. 2.101 libpostproc53. 3.100 / 53. 3.100 Alex ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] amerge channel layout
d, use AVStream.codecpar instead." - What does it mean? Did I compile the ffmpeg binary wrong? "Codec AVOption bf (set maximum number of B frames between non-B-frames) specified for output file #0 (file.mp4) has not been used for any stream. The most likely reason is either wrong type (e.g. a video option with no video streams) or that it is a private option of some encoder which was not actually used for any stream." - I want to set the number of bframes manually. It seems that ffmpeg can't apply that to the video stream. "[Parsed_amerge_0 @ 0x3507330] No channel layout for input 1" - You said it's not an error, okay. But isn't it possible to define a fix layout and mapping for the output stream to avoid this warning message? Thanks, Alex -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/amerge-channel-layout-tp4676437p4676459.html Sent from the FFmpeg-users mailing list archive at Nabble.com. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] amerge channel layout
The output file seems to be okay. I was a bit confused regarding the warning messages. Especially this one: "Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead". -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/amerge-channel-layout-tp4676437p4676458.html Sent from the FFmpeg-users mailing list archive at Nabble.com. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] amerge channel layout
Sorry, output again: ffmpeg version N-80123-gd74cc61-static Copyright (c) 2000-2016 the FFmpeg developers built with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-17) configuration: --arch=64 --prefix=/root/ffmpeg-static/ffmpeg-build-script/workspace --extra-cflags=-I/root/ffmpeg-static/ffmpeg-build-script/workspace/include --extra-ldflags=-L/root/ffmpeg-static/ffmpeg-build-script/workspace/lib --extra-version=static --extra-cflags=--static --enable-static --disable-debug --disable-shared --disable-ffplay --disable-ffserver --disable-doc --enable-gpl --enable-version3 --enable-nonfree --enable-pthreads --enable-libvpx --enable-libmp3lame --enable-libtheora --enable-libvorbis --enable-libx264 --enable-libx265 --enable-runtime-cpudetect --enable-libfdk-aac --enable-avfilter --enable-libopencore_amrwb --enable-libopencore_amrnb --enable-filters --enable-libvidstab --enable-libebur128 --enable-bzlib --enable-libopus --enable-libkvazaar --enable-frei0r libavutil 55. 24.100 / 55. 24.100 libavcodec 57. 43.100 / 57. 43.100 libavformat57. 37.101 / 57. 37.101 libavdevice57. 0.101 / 57. 0.101 libavfilter 6. 46.100 / 6. 46.100 libswscale 4. 1.100 / 4. 1.100 libswresample 2. 0.101 / 2. 0.101 libpostproc54. 0.100 / 54. 0.100 Guessed Channel Layout for Input Stream #0.1 : mono Guessed Channel Layout for Input Stream #0.2 : mono Input #0, mxf, from 'file.mxf': Metadata: uid : 42704c88-2d83-11e6-a3f0-002608fe0387 generation_uid : 42704c89-2d83-11e6-a457-002608fe0387 company_name: Adobe Systems Incorporated product_name: Premiere Pro product_version : 7.2.2 application_platform: Mac OS X product_uid : 10ab07a9-e89e-7510-a923-ea9220524153 modification_date: 2016-06-08 14:14:06 material_package_umid: 0x060A2B340101010501010D11130098F70D03477505A50431002608FE0387 timecode: 00:00:00:00 Duration: 00:04:25.96, start: 0.00, bitrate: 51792 kb/s Stream #0:0: Video: mpeg2video (4:2:2), yuv422p(tv, unknown/bt709/bt709), 1920x1080 [SAR 1:1 DAR 16:9], 5 kb/s, 25 fps, 25 tbr, 25 tbn Metadata: file_package_umid: 0x060A2B340101010501010D1213B3B5A098F70D03477505A5C467002608FE0387 file_package_name: Source Package Stream #0:1: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s Metadata: file_package_umid: 0x060A2B340101010501010D1213B3B5A098F70D03477505A5C467002608FE0387 file_package_name: Source Package Stream #0:2: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s Metadata: file_package_umid: 0x060A2B340101010501010D1213B3B5A098F70D03477505A5C467002608FE0387 file_package_name: Source Package Codec AVOption bf (set maximum number of B frames between non-B-frames) specified for output file #0 (file.mp4) has not been used for any stream. The most likely reason is either wrong type (e.g. a video option with no video streams) or that it is a private option of some encoder which was not actually used for any stream. [Parsed_amerge_0 @ 0x37f6330] No channel layout for input 1 [Parsed_amerge_0 @ 0x37f6330] Input channel layouts overlap: output layout will be determined by the number of distinct input channels [mp4 @ 0x3505660] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead. Output #0, mp4, to 'file.mp4': Metadata: uid : 42704c88-2d83-11e6-a3f0-002608fe0387 generation_uid : 42704c89-2d83-11e6-a457-002608fe0387 company_name: Adobe Systems Incorporated product_name: Premiere Pro product_version : 7.2.2 application_platform: Mac OS X product_uid : 10ab07a9-e89e-7510-a923-ea9220524153 modification_date: 2016-06-08 14:14:06 material_package_umid: 0x060A2B340101010501010D11130098F70D03477505A50431002608FE0387 timecode: 00:00:00:00 encoder : Lavf57.37.101 Stream #0:0: Audio: aac ([64][0][0][0] / 0x0040), 48000 Hz, stereo, s16, 384 kb/s (default) Metadata: encoder : Lavc57.43.100 libfdk_aac Stream mapping: Stream #0:1 (pcm_s16le) -> amerge:in0 Stream #0:2 (pcm_s16le) -> amerge:in1 volume -> Stream #0:0 (libfdk_aac) Press [q] to stop, [?] for help [mp4 @ 0x3505660] Starting second pass: moving the moov atom to the beginning of the file -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/amerge-channel-layout-tp4676437p4676455.html Sent from the FFmpeg-users mailing list archive at Nabble.com. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
Re: [FFmpeg-user] amerge channel layout
ut #0, mp4, to 'file.mp4': Metadata: uid : 42704c88-2d83-11e6-a3f0-002608fe0387 generation_uid : 42704c89-2d83-11e6-a457-002608fe0387 company_name: Adobe Systems Incorporated product_name: Premiere Pro product_version : 7.2.2 application_platform: Mac OS X product_uid : 10ab07a9-e89e-7510-a923-ea9220524153 modification_date: 2016-06-08 14:14:06 material_package_umid: 0x060A2B340101010501010D11130098F70D03477505A5043 1002608FE0387 timecode: 00:00:00:00 encoder : Lavf57.37.101 Stream #0:0: Audio: aac ([64][0][0][0] / 0x0040), 48000 Hz, stereo, s16, 384kb/s (default) Metadata: encoder : Lavc57.43.100 libfdk_aac Stream mapping: Stream #0:1 (pcm_s16le) -> amerge:in0 Stream #0:2 (pcm_s16le) -> amerge:in1 volume -> Stream #0:0 (libfdk_aac) Press [q] to stop, [?] for help [mp4 @ 0x2e7c660] Starting second pass: moving the moov atom to the beginning ofthe file Thanks, Alex -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/amerge-channel-layout-tp4676437p4676454.html Sent from the FFmpeg-users mailing list archive at Nabble.com. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] amerge channel layout
Hi all, I have some issues with amerge channel layouts. Source file is a MXF with 2 mono stream. Output should be one stereo stream (L/R). Source File: Input #0, mxf, from 'file.mxf': Duration: 00:04:25.96, start: 0.00, bitrate: 51792 kb/s Stream #0:0: Video: mpeg2video (4:2:2), yuv422p(tv, unknown/bt709/bt709), 1920x1080 [SAR 1:1 DAR 16:9], 5 kb/s, 25 fps, 25 tbr, 25 tbn Stream #0:1: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s Stream #0:2: Audio: pcm_s16le, 48000 Hz, 1 channels, s16, 768 kb/s I tried the following: ffmpeg -y -i file.mxf -filter:v yadif -b:v 8000k -minrate 8000k -maxrate 8000k -bufsize 4000k -vcodec libx264 -bf 2 -flags +cgop -pix_fmt yuv420p -f mp4 -filter_complex "[0:1][0:2]amerge=inputs=2,volume=14.4dB [aout]" -map [aout] -c:a libfdk_aac -b:a 384k -ac 2 -ar 48000 -cutoff 20k -movflags faststart -ss 00:00:00 file.mp4 But: Guessed Channel Layout for Input Stream #0.1 : mono Guessed Channel Layout for Input Stream #0.2 : mono [Parsed_amerge_0 @ 0x2eae4d0] No channel layout for input 1 [Parsed_amerge_0 @ 0x2eae4d0] Input channel layouts overlap: output layout will be determined by the number of distinct input channels So it isn't really a problem because the output audio stream is correct (stereo) "output layout will be determined by the number of distinct input channels". But how can I set a correct channel layout to avoid these warnings. Thanks, Alex -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/amerge-channel-layout-tp4676437.html Sent from the FFmpeg-users mailing list archive at Nabble.com. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] ffmpeg new version, some warning messages
Hi all, I recently switched ffmpeg from version 1.x to 3. Now I got some warning when I transcoding MXF files to youtube compliant MP4s. command: ffmpeg -y -i file.mxf -filter:v yadif -b:v 8000k -minrate 8000k -maxrate 8000k -bufsize 4000k -vcodec libx264 -flags +cgop -pix_fmt yuv420p -f mp4 -filter_complex "amerge,volume=14.4dB [aout]" -map [aout] -c:a libfdk_aac -b:a 384k -ac 2 -ar 48000 -cutoff 20k -movflags faststart -ss 00:00:00 file.mp4 warnings: Codec AVOption bf (set maximum number of B frames between non-B-frames) specified for output file #0 (file.mp4) has not been used for any stream. The most likely reason is either wrong type (e.g. a video option with no video streams) or that it is a private option of some encoder which was not actually used for any stream. [mp4 @ 0x39156b0] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead. Thanks, Alex -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/ffmpeg-new-version-some-warning-messages-tp4676436.html Sent from the FFmpeg-users mailing list archive at Nabble.com. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user To unsubscribe, visit link above, or email ffmpeg-user-requ...@ffmpeg.org with subject "unsubscribe".
[FFmpeg-user] Feeding DVR to ffmpeg server, part 2
Hello I've managed to feed video with /usr/local/bin/tanidvr -m 1 -c 1 -m 1 -t dvr-host -u user -w password | ffmpeg -r 25 -i - -strict -2 http://localhost:8090/feed1.ffm but the picture is of poor quality, with lots of artifacts. How do I determine proper parameters for processing it? === ffserver.conf === HTTPPort 8090 HTTPBindAddress 0.0.0.0 MaxHTTPConnections 2000 MaxClients 1000 MaxBandwidth 1000 # '-' is the standard output. CustomLog - # ffmpeg http://localhost:8090/feed1.ffm File /tmp/feed1.ffm FileMaxSize 200K # Specify launch in order to start ffmpeg automatically. # First ffmpeg must be defined with an appropriate path if needed, # after that options can follow, but avoid adding the http:// field #Launch ffmpeg ACL allow 127.0.0.1 Feed feed1.ffm # Format of the stream : you can choose among: # mpeg : MPEG-1 multiplexed video and audio # mpegvideo : only MPEG-1 video # mp2: MPEG-2 audio (use AudioCodec to select layer 2 and 3 codec) # ogg: Ogg format (Vorbis audio codec) # rm : RealNetworks-compatible stream. Multiplexed audio and video. # ra : RealNetworks-compatible stream. Audio only. # mpjpeg : Multipart JPEG (works with Netscape without any plugin) # jpeg : Generate a single JPEG image. # mjpeg : Generate a M-JPEG stream. # asf: ASF compatible streaming (Windows Media Player format). # swf: Macromedia Flash compatible stream # avi: AVI format (MPEG-4 video, MPEG audio sound) Format mpeg NoAudio VideoBitRate 128 VideoBufferSize 40 VideoFrameRate 25 VideoSize 352x288 VideoGopSize 12 === ffserver.conf === Alex ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Feeding DVR to ffserver
On 24.02.2016 18:04, Moritz Barsnick wrote: > Hi Alex, > although there's a lot of confusion on this thread, there is one thing > I can likely say: > > On Wed, Feb 24, 2016 at 17:41:47 +0300, Alex Povolotsky wrote: > >> [mpeg1video @ 0x808476400] MPEG1/2 does not support 3/1 fps > [...] >> Error while opening encoder for output stream #0:0 - maybe incorrect >> parameters such as bit_rate, rate, width or height > > I'm sure these two are related. The 3 fps comes from this line in your > config: > VideoFrameRate 3 > > Use a different codec or a different fps to circumvent this > restriction (which is not ffmpeg's restriction). Video stream is clearly NOT 3 FPS. Are there any way to find out required params? Alex ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Feeding DVR to ffserver
On 24.02.2016 17:36, Alex Povolotsky wrote: >>> >>> File /tmp/feed1.ffm >> >> Can ffserver write to this location? Yes. Just in case: [16:59] superbook:~ % ls -ls /tmp/feed1.ffm 5 -rw-r--r-- 1 root wheel 4096 Feb 24 17:39 /tmp/feed1.ffm [17:39] superbook:~ % date Wed Feb 24 17:39:33 MSK 2016 Yes I know that running ffserver as root is bad, but I'm doing it only as long as I'm testing the whole thing. >>> AudioChannels 0 >> >> Instead of AudioChannels 0 use NoAudio Done. # /usr/local/bin/tanidvr -m 1 -c 1 -m 1 -t dvr -u admin -w admin | ffmpeg -i - http://localhost:8090/feed1.ffm ... INFO (main): Media container: DHAV Invalid UE golomb code Last message repeated 1 times Input #0, matroska,webm, from 'pipe:': Metadata: encoder : TaniDVR 1.4.1 Duration: 00:00:00.27, start: 0.00, bitrate: N/A Stream #0:0: Video: h264 (Constrained Baseline), yuv420p, 352x288, SAR 12:11 DAR 4:3, 24.98 fps, 24.98 tbr, 1000k tbn, 2000k tbc [mpeg1video @ 0x808476400] bitrate tolerance 21333 too small for bitrate 64000, overriding [mpeg1video @ 0x808476400] too many threads/slices (9), reducing to 8 [mpeg1video @ 0x808476400] MPEG1/2 does not support 3/1 fps Output #0, ffm, to 'http://localhost:8090/feed1.ffm': Metadata: encoder : TaniDVR 1.4.1 creation_time : 2016-02-24 17:41:07 Stream #0:0: Video: mpeg1video, none, 160x128, q=2-31, 64 kb/s, SAR 16:15 DAR 4:3, 25 fps, 3 tbc Metadata: encoder : Lavc56.60.100 mpeg1video Stream #0:1: Video: msmpeg4v3 (msmpeg4), none, 352x240, q=2-31, 256 kb/s, SAR 10:11 DAR 4:3, 24.98 fps, 15 tbc Metadata: encoder : Lavc56.60.100 msmpeg4 Stream mapping: Stream #0:0 -> #0:0 (h264 (native) -> mpeg1video (native)) Stream #0:0 -> #0:1 (h264 (native) -> msmpeg4v3 (msmpeg4)) Error while opening encoder for output stream #0:0 - maybe incorrect parameters such as bit_rate, rate, width or height FATAL ERROR (main): Unable to write to target: -1. Same error here Alex ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Feeding DVR to ffserver
On 24.02.2016 17:28, Jimmy Asher wrote: > FYI: This thread considers top posing “rude" > > > > On 2/24/16, 9:16 AM, "ffmpeg-user on behalf of Alex Povolotsky" > <ffmpeg-user-boun...@ffmpeg.org on behalf of tark...@corp.infotel.ru> wrote: > >> Skipping comments: >> >> HTTPPort 8090 >> HTTPBindAddress 0.0.0.0 >> MaxHTTPConnections 2000 >> MaxClients 1000 >> MaxBandwidth 1000 >> CustomLog - >> >> >> File /tmp/feed1.ffm > > Can ffserver write to this location? > >> FileMaxSize 200K >> ACL allow 127.0.0.1 >> >> >> >> Feed feed1.ffm >> Format mpeg >> AudioChannels 0 > > Instead of AudioChannels 0 use NoAudio > >> VideoBitRate 64 >> VideoBufferSize 40 >> VideoFrameRate 3 >> VideoSize 160x128 >> VideoGopSize 12 >> >> >> >> Feed feed1.ffm >> Format asf >> VideoFrameRate 15 >> VideoSize 352x240 >> VideoBitRate 256 >> VideoBufferSize 40 >> VideoGopSize 30 >> AudioBitRate 64 >> StartSendOnKey >> >> >> Format status >> >> >> >> >> URL http://www.ffmpeg.org/ >> > ___ > ffmpeg-user mailing list > ffmpeg-user@ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > Okay. Sorry for top posting. I'm clearly doing something stupid with ffmpeg. Can't you please point me to my mistake and suggest good codec options for video compressing? Alex ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Feeding DVR to ffserver
Skipping comments: HTTPPort 8090 HTTPBindAddress 0.0.0.0 MaxHTTPConnections 2000 MaxClients 1000 MaxBandwidth 1000 CustomLog - File /tmp/feed1.ffm FileMaxSize 200K ACL allow 127.0.0.1 Feed feed1.ffm Format mpeg AudioChannels 0 VideoBitRate 64 VideoBufferSize 40 VideoFrameRate 3 VideoSize 160x128 VideoGopSize 12 Feed feed1.ffm Format asf VideoFrameRate 15 VideoSize 352x240 VideoBitRate 256 VideoBufferSize 40 VideoGopSize 30 AudioBitRate 64 StartSendOnKey Format status URL http://www.ffmpeg.org/ it is running, stat.html is responding with ffserver Status Available Streams PathServed Conns bytes Format Bit rate kbits/s Video kbits/s Codec Audio kbits/s Codec Feed cam1-2.mpg 1 63 mpeg128 64 mpeg1video 64 mp2 feed1.ffm test.asf0 0 asf_stream 320 256 msmpeg4 64 wmav2 feed1.ffm stat.html 2 2105- - - - index.html 0 3536- - - - Feed feed1.ffm Stream typekbits/s codec Parameters 0 audio 64 mp2 0 channel(s), 22050 Hz 1 video 64 mpeg1video 160x128, q=2-31, fps=3 2 audio 64 wmav2 1 channel(s), 22050 Hz 3 video 256 msmpeg4 352x240, q=2-31, fps=15 Connection Status Number of connections: 1 / 1000 Bandwidth in use: 0k / 1000k # FileIP Proto State Target bits/sec Actual bits/sec Bytes transferred 1 stat.html 10.187.11.6 HTTP/1.1HTTP_WAIT_REQUEST 0 0 0 Generated at Wed Feb 24 17:16:10 2016 On 24.02.2016 17:09, Jimmy Asher wrote: > > > > > > On 2/24/16, 9:04 AM, "ffmpeg-user on behalf of Alex Povolotsky" > <ffmpeg-user-boun...@ffmpeg.org on behalf of tark...@corp.infotel.ru> wrote: > >> Hello >> >> I'm working with pretty aged video recorer for surveillance cams, trying >> to connect it to zoneminder instead of weird propiertary surveillance soft. >> >> tanidvr gets video stream, but all attempts to send it to ffserver >> failed so far. >> >> === ffserver.conf === >> >> File /tmp/feed1.ffm >> FileMaxSize 200K >> ACL allow 127.0.0.1 >> >> === ffserver.conf === > > > If this is the complete ffserver config you do not set up the config file > correctly > >> >> than >> >> >> >> I'm new to video processing and maybe I'm doing lots of dumb errors, but >> googling for error yielded nothing meaningful. >> >> What should I do to feed ffserver with video and what are good codec >> params to compress the video? > > Did you start FFServer before you invoked Ffmpeg? > > >> >> Where can I find good introduction for video processing with ffmpeg? > > FFmpeg Streaming Guide (good place to start) > https://trac.ffmpeg.org/wiki/StreamingGuide > > >> >> Alex >> ___ >> ffmpeg-user mailing list >> ffmpeg-user@ffmpeg.org >> http://ffmpeg.org/mailman/listinfo/ffmpeg-user > ___ > ffmpeg-user mailing list > ffmpeg-user@ffmpeg.org > http://ffmpeg.org/mailman/listinfo/ffmpeg-user > -- Александр Поволоцкий Системный администратор АО "Инфотел" 119021 г. Москва, ул. Льва Толстого 23 стр 3 (адрес юридический и почтовый) 115088 г. Москва, 2-ой Южнопортовый пр-д, д 20А стр.4, 2-й подъезд, 1-й этаж (фактический адрес/ для курьерской доставки) +7(495) 744-0918 (тел.) +7(495) 744-0922 (факс) www.infotel.ru ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
[FFmpeg-user] Feeding DVR to ffserver
3 (msmpeg4)) Error while opening encoder for output stream #0:1 - maybe incorrect parameters such as bit_rate, rate, width or height FATAL ERROR (main): Unable to write to target: -1. DETAIL (buffer): Got SIGTERM|SIGHUP|SIGXCPU. DETAIL (buffer): Write socket closed remotely. DETAIL (buffer): buffered_tunnel_pipe() returned: -5 DETAIL (buffer): Got SIGCHLD. DETAIL (buffer): Intermediate process returned (102). DETAIL (main): Got SIGCHLD. I'm new to video processing and maybe I'm doing lots of dumb errors, but googling for error yielded nothing meaningful. What should I do to feed ffserver with video and what are good codec params to compress the video? Where can I find good introduction for video processing with ffmpeg? Alex ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
[FFmpeg-user] Using drawtext for date/time
Hi, I’d like to burn a date/time using drawtext, based on PTS with the addition of a specified basetime. Looking at the code, expansion=strftime seems to do exactly what I want — but it is marked as deprecated. Alternatives don’t seem to cut it: func_pts doesn’t print date and does not take basetime into account, func_strftime doesn’t take basetime into account, and uses current time rather than PTS. What is the reason the strftime expansion had been deprecated? Is there a current equivalent? Thanks. - Alex ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Using drawtext for date/time
Moritz — thanks. Note, however, that I wanted to derive the text from pts (with
optional application of some base time), rather than from the current time.
Correct me if I’m wrong, but what you’ve described (localtime) can only do the
latter.
I’ve just submitted a patch, which hopefully will introduce that functionality.
--
Alex
On October 8, 2015 at 4:26:24 PM, Moritz Barsnick (barsn...@gmx.net) wrote:
Hi Alex,
On Thu, Oct 08, 2015 at 15:11:46 -0400, Alex Agranovsky wrote:
> I’d like to burn a date/time using drawtext, based on PTS with the
> addition of a specified basetime. Looking at the code,
> expansion=strftime seems to do exactly what I want — but it is marked
> as deprecated. Alternatives don’t seem to cut it: func_pts doesn’t
> print date and does not take basetime into account, func_strftime
> doesn’t take basetime into account, and uses current time rather than
> PTS. What is the reason the strftime expansion had been deprecated?
> Is there a current equivalent?
I think you only need to read a bit further down in the filter's
documentation:
If ‘expansion’ is set to normal (which is the default), the following
expansion mechanism is used.
[...]
Sequence [sic] of the form %{...} are expanded. The text between the
braces is a function name, possibly followed by arguments separated
by ':'.
[...]
The following functions are available:
[...]
localtime
The time at which the filter is running, expressed in the local
time zone. It can accept an argument: a strftime() format string.
So, this boils down to:
-vf 'drawtext=expansion=%{localtime:%H\:%M\:%S}'
or something like this with a lot more escaping. :-)
Warning: escape hell:
https://trac.ffmpeg.org/ticket/3253
(Note: I haven't tried this.)
Moritz
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Re: [FFmpeg-user] Audio Tapes - How to restore audio quality
Use audacity instead On May 29, 2015, at 22:34, jd1008 jd1...@gmail.com wrote: I need to transfer a bunch of audio tapes to digital media. But I would like to 1. improve the signal to noise ratio. 2. get rid of clicks and pops. Is there a way to do this using ffmpeg? ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
[FFmpeg-user] MXF Replace Audio Streams while maintaining Data Closed Caption vbi_vanc_smpte_436M stream
Hello, Attempting to replace first two audio streams within a MXF File with two WAV audio streams. The below works if I remove Data stream but I need CC. Is there a way to copy over the data stream as well? /Applications/DevelopmentTools/ffmpeg/ffmpeg -i ~/Downloads/ExampleCopy.mxf -i -acodec copy -vcodec copy -dcodec copy -map 0:0 -map 1:0 -map 2:0 -map 0:9 Output.mxf HSPOutput.mxfaudio_3.wav audio_6.wav audio_encoded_2.wav audio_encoded_5.wav audio_1.wav audio_4.wav audio_7.wav audio_encoded_3.wav audio_encoded_6.wav audio_2.wav audio_5.wav audio_encoded_1.wav audio_encoded_4.wav audio_encoded_7.wav XXX-MacBook-Air-2:temp xxx$ /Applications/DevelopmentTools/ffmpeg/ffmpeg -i ~/Downloads/ExampleCopy.mxf -i audio_1.wav -i audio_2.wav -acodec copy -vcodec copy -dcodec copy -map 0:0 -map 1:0 -map 2:0 -map 0:9 Output.mxf ffmpeg version N-69382-g038f3a1 Copyright (c) 2000-2015 the FFmpeg developers built on Jan 29 2015 14:02:27 with Apple LLVM version 6.0 (clang-600.0.56) (based on LLVM 3.5svn) configuration: libavutil 54. 18.100 / 54. 18.100 libavcodec 56. 21.101 / 56. 21.101 libavformat56. 19.100 / 56. 19.100 libavdevice56. 4.100 / 56. 4.100 libavfilter 5. 9.101 / 5. 9.101 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 Guessed Channel Layout for Input Stream #0.1 : mono Guessed Channel Layout for Input Stream #0.2 : mono Guessed Channel Layout for Input Stream #0.3 : mono Guessed Channel Layout for Input Stream #0.4 : mono Guessed Channel Layout for Input Stream #0.5 : mono Guessed Channel Layout for Input Stream #0.6 : mono Guessed Channel Layout for Input Stream #0.7 : mono Guessed Channel Layout for Input Stream #0.8 : mono Input #0, mxf, from '/Users/xxx/Downloads/ExampleCopy.mxf': Metadata: uid : 138157ae-9bbe-1b41-871b-1bfc102cb188 generation_uid : 8af54e43-25e4-184a-b85a-87bb9d8a4d8d company_name: AVID product_name: TRMG product_version : 3.01 product_uid : ---- modification_date: 2014-12-31 19:29:53 material_package_umid: 0x060A2B340101010501010D131300E785EB19DB75A340B703A96982AF2957 timecode: 01:00:00;00 Duration: 00:01:00.06, start: 0.00, bitrate: 60846 kb/s Stream #0:0: Video: mpeg2video (4:2:2), yuv422p(tv, bt709), 1280x720 [SAR 1:1 DAR 16:9], 5 kb/s, 59.94 fps, 59.94 tbr, 59.94 tbn, 119.88 tbc Metadata: file_package_umid: 0x060A2B340101010501010D1313006D9F377935593C4280BA79DD0090270A Stream #0:1: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s Metadata: file_package_umid: 0x060A2B340101010501010D1313006D9F377935593C4280BA79DD0090270A Stream #0:2: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s Metadata: file_package_umid: 0x060A2B340101010501010D1313006D9F377935593C4280BA79DD0090270A Stream #0:3: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s Metadata: file_package_umid: 0x060A2B340101010501010D1313006D9F377935593C4280BA79DD0090270A Stream #0:4: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s Metadata: file_package_umid: 0x060A2B340101010501010D1313006D9F377935593C4280BA79DD0090270A Stream #0:5: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s Metadata: file_package_umid: 0x060A2B340101010501010D1313006D9F377935593C4280BA79DD0090270A Stream #0:6: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s Metadata: file_package_umid: 0x060A2B340101010501010D1313006D9F377935593C4280BA79DD0090270A Stream #0:7: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s Metadata: file_package_umid: 0x060A2B340101010501010D1313006D9F377935593C4280BA79DD0090270A Stream #0:8: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s Metadata: file_package_umid: 0x060A2B340101010501010D1313006D9F377935593C4280BA79DD0090270A Stream #0:9: Data: none Metadata: file_package_umid: 0x060A2B340101010501010D1313006D9F377935593C4280BA79DD0090270A data_type : vbi_vanc_smpte_436M Input #1, wav, from 'audio_1.wav': Metadata: encoder : Lavf54.63.104 timecode: 01:00:00;00 Duration: 00:01:00.06, bitrate: 1152 kb/s Stream #1:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s32 (24 bit), 1152 kb/s Input #2, wav, from 'audio_2.wav': Metadata: encoder : Lavf54.63.104 timecode: 01:00:00;00 Duration: 00:01:00.06, bitrate: 1152 kb/s Stream #2:0: Audio: pcm_s24le ([1][0][0][0] / 0x0001), 48000 Hz, mono, s32 (24 bit), 1152 kb/s [mxf @ 0x7fe0b2a42000] track 3: could not find essence container ul, codec not currently supported in container Output #0, mxf, to 'Output.mxf': Metadata: uid :
[FFmpeg-user] Detecting blue frames.
Hello all. Is there a way to detect blue screen (usually generated by analog videotape equipment) using ffmpeg. I know there is a way to detect black screen. Below is a sample of what I have in mind. https://www.youtube.com/watch?v=fC_bO-4uwFk Thanks in advance. -Alex ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Right audio channel shifted
Alex wrote VLC still said: http://s14.directupload.net/images/141014/r7h73d5b.jpg One other thing: I checked my database again and I have to verify over 300 videos which are possibly affected by this bug. Is there a possibility with ffmpeg to repair those files automatically or at least see if a file is affected. Otherwise I am going to do this job manually via a vectorscope. :) Thanks, Alex Okay, when I check the trimmed file with ffmpeg it looks okay: C:\ffmpeg -i test_new.mxf ffmpeg version N-66809-g20df026 Copyright (c) 2000-2014 the FFmpeg developers built on Oct 11 2014 23:42:02 with gcc 4.9.1 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable- libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc -- enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-zlib libavutil 54. 10.100 / 54. 10.100 libavcodec 56. 4.101 / 56. 4.101 libavformat56. 9.100 / 56. 9.100 libavdevice56. 1.100 / 56. 1.100 libavfilter 5. 1.103 / 5. 1.103 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc53. 1.100 / 53. 1.100 [mxf @ 00350280] index entry 5095 + TemporalOffset 1 = 5096, which is out of bounds Guessed Channel Layout for Input Stream #0.1 : mono Guessed Channel Layout for Input Stream #0.2 : mono Input #0, mxf, from 'test_new.mxf': Metadata: uid : adab4424-2f25-4dc7-92ff-29bd000b generation_uid : adab4424-2f25-4dc7-92ff-29bd000b0001 company_name: FFmpeg product_name: OP1a Muxer product_version : 56.9.100 product_uid : adab4424-2f25-4dc7-92ff-29bd000b0002 modification_date: -01-01 00:00:00 timecode: 00:00:00:00 Duration: 00:03:32.55, start: 0.00, bitrate: 52515 kb/s Stream #0:0: Video: mpeg2video (4:2:2), yuv422p(tv, bt709), 1920x1080 [SAR 1:1 DAR 16:9], 5 kb/s, 23.98 fps, 23.98 tbr, 23.98 tbn, 47.95 tbc Stream #0:1: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s Stream #0:2: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s But: [mxf @ 00350280] index entry 5095 + TemporalOffset 1 = 5096, which is out of bounds What does this mean? An why does VLC identifies different channel values? Best, Alex -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Right-audio-channel-shifted-tp4667730p4667781.html Sent from the FFmpeg-users mailing list archive at Nabble.com. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Right audio channel shifted
Okay, I played a bit around with the atrim Filter, but the manual isn't very helpful here. I've tried the following: C:\ffmpeg -i test.mxf -vcodec copy -map 0:v -acodec pcm_s24le -map 0:1 -map 0:2 -filter_complex [a:1]atrim=start=0.035 test_new.mxf ffmpeg version N-66809-g20df026 Copyright (c) 2000-2014 the FFmpeg developers built on Oct 11 2014 23:42:02 with gcc 4.9.1 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable- libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc -- enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-zlib libavutil 54. 10.100 / 54. 10.100 libavcodec 56. 4.101 / 56. 4.101 libavformat56. 9.100 / 56. 9.100 libavdevice56. 1.100 / 56. 1.100 libavfilter 5. 1.103 / 5. 1.103 libswscale 3. 1.101 / 3. 1.101 libswresample 1. 1.100 / 1. 1.100 libpostproc53. 1.100 / 53. 1.100 Guessed Channel Layout for Input Stream #0.1 : mono Guessed Channel Layout for Input Stream #0.2 : mono Input #0, mxf, from 'test.mxf': Metadata: uid : adab4424-2f25-4dc7-92ff-29bd000b generation_uid : adab4424-2f25-4dc7-92ff-29bd000b0001 company_name: FFmbc product_name: OP1a Muxer product_version : 53.6.0 product_uid : adab4424-2f25-4dc7-92ff-29bd000b0002 modification_date: 2014-03-28 11:58:57 timecode: 00:00:00:00 Duration: 00:03:32.59, start: 0.00, bitrate: 52515 kb/s Stream #0:0: Video: mpeg2video (4:2:2), yuv422p(tv, bt709), 1920x1080 [SAR 1:1 DAR 16:9], 5 kb/s, 23.98 fps, 23.98 tbr, 23.98 tbn, 47.95 tbc Stream #0:1: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s Stream #0:2: Audio: pcm_s24le, 48000 Hz, 1 channels, s32 (24 bit), 1152 kb/s File 'test_new.mxf' already exists. Overwrite ? [y/N] y [mxf @ 003b69c0] there must be exactly one video stream and it must be the first one Output #0, mxf, to 'test_new.mxf': Metadata: uid : adab4424-2f25-4dc7-92ff-29bd000b generation_uid : adab4424-2f25-4dc7-92ff-29bd000b0001 company_name: FFmbc product_name: OP1a Muxer product_version : 53.6.0 product_uid : adab4424-2f25-4dc7-92ff-29bd000b0002 modification_date: 2014-03-28 11:58:57 timecode: 00:00:00:00 encoder : Lavf56.9.100 Stream #0:0: Audio: pcm_s24le, 48000 Hz, mono, s32 (24 bit), 1152 kb/s Metadata: encoder : Lavc56.4.101 pcm_s24le Stream #0:1: Video: mpeg2video, yuv422p, 1920x1080 [SAR 1:1 DAR 16:9], q=2-31, 5 kb/s, 23.98 fps, 23.98 tbn, 23.98 tbc Stream mapping: Stream #0:2 (pcm_s24le) - atrim atrim - Stream #0:0 (pcm_s24le) Stream #0:0 - #0:1 (copy) Could not write header for output file #0 (incorrect codec parameters ?): Error number -1 occurred I guess there is somehing wrong with the channel mapping, but without the fliter_complex the mapping seems to be okay. Thanks, Alex -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Right-audio-channel-shifted-tp4667730p4667764.html Sent from the FFmpeg-users mailing list archive at Nabble.com. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Right audio channel shifted
Moritz Barsnick wrote Hi Alex, I guess there is somehing wrong with the channel mapping, but without the fliter_complex the mapping seems to be okay. I think the filter_complex messes up the mapping. Note this mapping from your output: Stream mapping: Stream #0:2 (pcm_s24le) - atrim atrim - Stream #0:0 (pcm_s24le) Stream #0:0 - #0:1 (copy) and this result: Output #0, mxf, to 'test_new.mxf': Stream #0:0: Audio: pcm_s24le, 48000 Hz, mono, s32 (24 bit), 1152 kb/s Stream #0:1: Video: mpeg2video, yuv422p, 1920x1080 [SAR 1:1 DAR 16:9], and this warning: [mxf @ 003b69c0] there must be exactly one video stream and it must be the first one I'm sure that's your issue. The filter seems to be messing up the mapping, at least in this automatic case. Instead of C:\ffmpeg -i test.mxf -vcodec copy -map 0:v -acodec pcm_s24le -map 0:1 -map 0:2 -filter_complex [a:1]atrim=start=0.035 test_new.mxf you probably need something like ffmpeg -i test.mxf -vcodec copy -map 0:v -acodec pcm_s24le -map 0:1 -filter_complex [a:1]atrim=start=0.035[ashifted] -map [ashifted] test_new.mxf (Untested) Note that I'm trying to explicitly map the filter output to a particular stream, avoiding automatic mapping. You may have to play around a bit more with the map options. Is this a bug, BTW? Documentation on -map states: The first -map option on the command line specifies the source for output stream 0, the second -map option specifies the source for output stream 1, etc. Digging further, documentation of -filter_complex clarifies: Output link labels are referred to with ‘-map’. Unlabeled outputs are added to the first output file. So the first documentation section is somewhat unclear, as it is disturbed by the second behavior. (And in the second section: first output file - first output stream, right?) Moritz Hi Moritz, thanks for your help, works great! The mapping thing is still a bit confusing to me. :) One last problem: I would like to set the sample format of the second stream back to s32. I tried the follwoing, but it won't work: ffmpeg -i test.mxf -vcodec copy -map 0:v -map 0:1 -sample_fmt s32 -c:a:0 copy -c:a:1 pcm_s24le -sample_fmt:a:1 s32 -filter_complex [a:1]atrim=start=0.035[ashifted] -map [ashifted] test_new.mxf Thanks, Alex -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Right-audio-channel-shifted-tp4667730p4667768.html Sent from the FFmpeg-users mailing list archive at Nabble.com. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] Right audio channel shifted
Thats absolutely clear, but I want to use ffmpeg to solve that issue. Alex -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Right-audio-channel-shifted-tp4667730p4667732.html Sent from the FFmpeg-users mailing list archive at Nabble.com. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
Re: [FFmpeg-user] ffmpeg read from UDP/Port
Hi Maziar, I finally figured it out. Instead of using udp stream as input, I generated a SDP file and use it as the input. Prior to receiving the audio packets, I receive a SIP packet with SDP payload. The SDP payload describes the IP/port of the video packet as well as the format. I just created a SDP file with the content of the payload, and FFmpeg was able to use it to receive and save the packets into a mpeg file. (or play it) Thank you for your help! AL On Mon, Sep 22, 2014 at 10:05 PM, Maziar Mehrabi mmehr...@abo.fi wrote: Hi, Try testing it over http first, rather than UDP or RTP. What I understand is that the errors somehow are related to audio channel, and RTP divides the media into two streams (one audio and one video) and sends them through separate ports, so I guess you should have some sort of mechanism on the other end to capture and mix these two streams. One other way to test this idea is to use the -an option while you are streaming media, this option means No Audio and will just stream the video channel. if you please, keep me updated about your results. BR, Maziar -- Hälsningar, Maziar Mehrabi On Tue, Sep 23, 2014 at 2:38 AM, Alex Lin op1...@gmail.com wrote: Hi Bill, I might be tempted to set both video and audio codec to copy, and save in a .TS file without -f at all. I did a look search on google about .ts files. My understanding is that, I can use videosnarf to convert a packet trace to .TS file. Is that the way you have in mind as well? Thank you, AL On Fri, Sep 19, 2014 at 12:19 PM, Bill Davidsen david...@tmr.com wrote: Alex Lin wrote: Hi all, I am using Windows 7 64 bit, and I downloaded the 64 bit version of ffmpeg: ffmpeg-20140916-git-b76d613-win64-static.7z I have spent the entire day experimenting with ffmpeg today but I haven't quite figure out if ffmpeg is the right solution to my problem yet, so I would like to get some opinions. I am receiving H264 encoded packets through RTP with sample rate of 90,000, and I need to record these packets. The file format doesn't really matter at this point. Currently, I have a server sending the H264 packets to port 50002 of my machine (192.168.1.200), so I run the following ffmpeg command on my machine: ffmpeg -i udp://192.168.1.200:50002 -f mp4 hello.mp4 I might be tempted to set both video and audio codec to copy, and save in a .TS file without -f at all. If that works you can capture the information and then play with it from the file. I am a Linux user, so you may need guidance for Windows issues from an expert, but if you can capture the data you eliminate some possible problems by not reformatting the data at capture. Then the following shows ffmpeg version N-66289-gb76d613 Copyright (c) 2000-2014 the FFmpeg developers built on Sep 15 2014 22:11:04 with gcc 4.8.3 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable- libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc -- enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-decklink --enable-zlib libavutil 54. 7.100 / 54. 7.100 libavcodec 56. 1.100 / 56. 1.100 libavformat56. 4.101 / 56. 4.101 libavdevice56. 0.100 / 56. 0.100 libavfilter 5. 1.100 / 5. 1.100 libswscale 3. 0.100 / 3. 0.100 libswresample 1. 1.100 / 1. 1.100 libpostproc53. 0.100 / 53. 0.100 The server starts sending video packets, and nothing happens on the command prompt with the ffmpeg command. The server stops sending video packets, and nothing happens still (I waited for at least 5 minutes), so I pressed Ctrl+C, then I see this udp://192.168.1.200:50002: Invalid data found when processing input Received signal 2: terminating. am I using ffmpeg correctly? and are the H264 packets I am receiving not supported? I have looked around and did not see any debug log generated by ffmpeg, so I don't really know where else to look. If it is necessary, I can provide a packet capture with the H264 packets. -- bill davidsen david...@tmr.com CTO TMR Associates, Inc Unsigned numbers may not be negative. However, unsigned numbers may
Re: [FFmpeg-user] ffmpeg read from UDP/Port
Hi Maziar, I am implementing the client. Before the server sends video stream to me, it sends a SIP INVITE with SDP which describes the video stream. Once I get the SIP INVITE, I can create a SDP file according to the SIP packet and call ffmpeg to record the incoming video stream. I have not programmed with html5 before, but I imagine there are packets to describe the incoming video stream before the video actually arrives. Otherwise, the server and the client must have already agreed on the video type ahead of time. Hope this helps, AL On Tue, Sep 23, 2014 at 11:16 AM, Maziar Mehrabi mmehr...@abo.fi wrote: Hi, No problem, it was yourself who figured it out after all. I have a question still though. How do you transmit the SDP file to the client? I mean apparently for every stream you should have an SDP file generated and sent to the client before the client is able to play it. This problem is a major challenge if the client if going to play the stream in a browser using html5. So how do you manage this? Or do you think what solutions are there? Thank you, Maziar -- Hälsningar, Maziar Mehrabi On Tue, Sep 23, 2014 at 8:19 PM, Alex Lin op1...@gmail.com wrote: Hi Maziar, I finally figured it out. Instead of using udp stream as input, I generated a SDP file and use it as the input. Prior to receiving the audio packets, I receive a SIP packet with SDP payload. The SDP payload describes the IP/port of the video packet as well as the format. I just created a SDP file with the content of the payload, and FFmpeg was able to use it to receive and save the packets into a mpeg file. (or play it) Thank you for your help! AL On Mon, Sep 22, 2014 at 10:05 PM, Maziar Mehrabi mmehr...@abo.fi wrote: Hi, Try testing it over http first, rather than UDP or RTP. What I understand is that the errors somehow are related to audio channel, and RTP divides the media into two streams (one audio and one video) and sends them through separate ports, so I guess you should have some sort of mechanism on the other end to capture and mix these two streams. One other way to test this idea is to use the -an option while you are streaming media, this option means No Audio and will just stream the video channel. if you please, keep me updated about your results. BR, Maziar -- Hälsningar, Maziar Mehrabi On Tue, Sep 23, 2014 at 2:38 AM, Alex Lin op1...@gmail.com wrote: Hi Bill, I might be tempted to set both video and audio codec to copy, and save in a .TS file without -f at all. I did a look search on google about .ts files. My understanding is that, I can use videosnarf to convert a packet trace to .TS file. Is that the way you have in mind as well? Thank you, AL On Fri, Sep 19, 2014 at 12:19 PM, Bill Davidsen david...@tmr.com wrote: Alex Lin wrote: Hi all, I am using Windows 7 64 bit, and I downloaded the 64 bit version of ffmpeg: ffmpeg-20140916-git-b76d613-win64-static.7z I have spent the entire day experimenting with ffmpeg today but I haven't quite figure out if ffmpeg is the right solution to my problem yet, so I would like to get some opinions. I am receiving H264 encoded packets through RTP with sample rate of 90,000, and I need to record these packets. The file format doesn't really matter at this point. Currently, I have a server sending the H264 packets to port 50002 of my machine (192.168.1.200), so I run the following ffmpeg command on my machine: ffmpeg -i udp://192.168.1.200:50002 -f mp4 hello.mp4 I might be tempted to set both video and audio codec to copy, and save in a .TS file without -f at all. If that works you can capture the information and then play with it from the file. I am a Linux user, so you may need guidance for Windows issues from an expert, but if you can capture the data you eliminate some possible problems by not reformatting the data at capture. Then the following shows ffmpeg version N-66289-gb76d613 Copyright (c) 2000-2014 the FFmpeg developers built on Sep 15 2014 22:11:04 with gcc 4.8.3 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable- libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable
Re: [FFmpeg-user] ffmpeg read from UDP/Port
Hi Bill, I might be tempted to set both video and audio codec to copy, and save in a .TS file without -f at all. I did a look search on google about .ts files. My understanding is that, I can use videosnarf to convert a packet trace to .TS file. Is that the way you have in mind as well? Thank you, AL On Fri, Sep 19, 2014 at 12:19 PM, Bill Davidsen david...@tmr.com wrote: Alex Lin wrote: Hi all, I am using Windows 7 64 bit, and I downloaded the 64 bit version of ffmpeg: ffmpeg-20140916-git-b76d613-win64-static.7z I have spent the entire day experimenting with ffmpeg today but I haven't quite figure out if ffmpeg is the right solution to my problem yet, so I would like to get some opinions. I am receiving H264 encoded packets through RTP with sample rate of 90,000, and I need to record these packets. The file format doesn't really matter at this point. Currently, I have a server sending the H264 packets to port 50002 of my machine (192.168.1.200), so I run the following ffmpeg command on my machine: ffmpeg -i udp://192.168.1.200:50002 -f mp4 hello.mp4 I might be tempted to set both video and audio codec to copy, and save in a .TS file without -f at all. If that works you can capture the information and then play with it from the file. I am a Linux user, so you may need guidance for Windows issues from an expert, but if you can capture the data you eliminate some possible problems by not reformatting the data at capture. Then the following shows ffmpeg version N-66289-gb76d613 Copyright (c) 2000-2014 the FFmpeg developers built on Sep 15 2014 22:11:04 with gcc 4.8.3 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable- libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc -- enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-decklink --enable-zlib libavutil 54. 7.100 / 54. 7.100 libavcodec 56. 1.100 / 56. 1.100 libavformat56. 4.101 / 56. 4.101 libavdevice56. 0.100 / 56. 0.100 libavfilter 5. 1.100 / 5. 1.100 libswscale 3. 0.100 / 3. 0.100 libswresample 1. 1.100 / 1. 1.100 libpostproc53. 0.100 / 53. 0.100 The server starts sending video packets, and nothing happens on the command prompt with the ffmpeg command. The server stops sending video packets, and nothing happens still (I waited for at least 5 minutes), so I pressed Ctrl+C, then I see this udp://192.168.1.200:50002: Invalid data found when processing input Received signal 2: terminating. am I using ffmpeg correctly? and are the H264 packets I am receiving not supported? I have looked around and did not see any debug log generated by ffmpeg, so I don't really know where else to look. If it is necessary, I can provide a packet capture with the H264 packets. -- bill davidsen david...@tmr.com CTO TMR Associates, Inc Unsigned numbers may not be negative. However, unsigned numbers may be less than zero for suffiently large values of zero. ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
[FFmpeg-user] ffmpeg read from UDP/Port
Hi all, I am using Windows 7 64 bit, and I downloaded the 64 bit version of ffmpeg: ffmpeg-20140916-git-b76d613-win64-static.7z I have spent the entire day experimenting with ffmpeg today but I haven't quite figure out if ffmpeg is the right solution to my problem yet, so I would like to get some opinions. I am receiving H264 encoded packets through RTP with sample rate of 90,000, and I need to record these packets. The file format doesn't really matter at this point. Currently, I have a server sending the H264 packets to port 50002 of my machine (192.168.1.200), so I run the following ffmpeg command on my machine: ffmpeg -i udp://192.168.1.200:50002 -f mp4 hello.mp4 Then the following shows ffmpeg version N-66289-gb76d613 Copyright (c) 2000-2014 the FFmpeg developers built on Sep 15 2014 22:11:04 with gcc 4.8.3 (GCC) configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable- libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aacenc -- enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxavs --enable-libxvid --enable-decklink --enable-zlib libavutil 54. 7.100 / 54. 7.100 libavcodec 56. 1.100 / 56. 1.100 libavformat56. 4.101 / 56. 4.101 libavdevice56. 0.100 / 56. 0.100 libavfilter 5. 1.100 / 5. 1.100 libswscale 3. 0.100 / 3. 0.100 libswresample 1. 1.100 / 1. 1.100 libpostproc53. 0.100 / 53. 0.100 The server starts sending video packets, and nothing happens on the command prompt with the ffmpeg command. The server stops sending video packets, and nothing happens still (I waited for at least 5 minutes), so I pressed Ctrl+C, then I see this udp://192.168.1.200:50002: Invalid data found when processing input Received signal 2: terminating. am I using ffmpeg correctly? and are the H264 packets I am receiving not supported? I have looked around and did not see any debug log generated by ffmpeg, so I don't really know where else to look. If it is necessary, I can provide a packet capture with the H264 packets. Thanks in advance. AL ___ ffmpeg-user mailing list ffmpeg-user@ffmpeg.org http://ffmpeg.org/mailman/listinfo/ffmpeg-user
