April 2010 - ffmpeg-user - lists.ffmpeg.org

Adding an Audio Stream with silence to a Video Stream
by tobiasbrockamp.lists＠googlemail.com 09 Apr '10

09 Apr '10

Hi there, i found on this homepage (http://www.wlug.org.nz/FFmpeg) an example how to generate an audio stream with silence: ffmpeg -ar 44100 -acodec pcm_s16le -f s16le -ac 2 -i <(dd if=/dev/zero bs=$((44100 * 2 * 2)) count=10) silence.wav My goal is to get one single command line which takes a video (with only a video stream) as input and also take the command above to get an output file with video and a silent audio stream. Has someone an idea? Is it possible using a pipe? Is it possible to generate the "silence" in any other audioformat like AAC? For me it was not possible. Kind regards

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ffmpeg command line expert needed
by afalaschi＠widen.com 09 Apr '10

09 Apr '10

I use ffmpeg to transcode videos into a number of different formats. My company is a SaaS based digital asset management provider. For the past three years, we have been using Carbon Coder as our transcoder, which is integrated into DAM software environment. We are now augmenting that with ffmpeg for the more common formats. Since it is open source, it can scale much more cost effectively. I understand the basics, but when it comes to some of the more intricate, or advanced commands, the documentation on ffmpeg sucks...we end up spinning our wheels for hours. Our need to change settings happen on a regular basis. Instead of breaking a sweat for a couple hours to "figure it out." I am looking for an expert that I can call/pay to give me the command line for the intended output format. This would be a reoccurring consultant project-by-project type situation.

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Adding an Audio Stream with silence to a Video Stream
by tobiasbrockamp.lists＠googlemail.com 09 Apr '10

09 Apr '10

Hi there, i found on this homepage (http://www.wlug.org.nz/FFmpeg) an example how to generate an audio stream with silence: ffmpeg -ar 44100 -acodec pcm_s16le -f s16le -ac 2 -i <(dd if=/dev/zero bs=$((44100 * 2 * 2)) count=10) silence.wav My goal is to get one single command line which takes a video (with only a video stream) as input and also take the command above to get an output file with video and a silent audio stream. Has someone an idea? Is it possible using a pipe? Is it possible to generate the "silence" in any other audioformat like AAC? For me it was not possible. Kind regards

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Adding audio to AVI conversion
by eowen＠wkag.com 09 Apr '10

09 Apr '10

First of all, thanks to all of you for helping earlier to get my AVI's converted to MPEG. Walter, your idea of updating the codec worked like a charm. The summary of what I'm doing is taking an AVI file and converting it to MPEG2. Works fine, but I want to add an audio file to the finished product. I'm assuming I can do it in one move, but I've hit difficulties so far. Here is the command line I'm using for the current conversion: FFMPEG -i <AVI File> -vcodec mpeg2video -b 2000k -r 25 -s 384x288 -an -y <output File> I want to use the 'shortest' command to end the encode when the video ends, and I've tried inserting the following within the previous command line: -i <Audio File> -shortest but so far, I don't get any audio wherever I place these commands, and sometimes it negates the creation of the file totally. Any suggestions? Thanks! ed

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RTP video steaming
by sushanth＠iitb.ac.in 09 Apr '10

09 Apr '10

Hey, I have been trying to streaming webcam video over RTP with the following config file Port 8090 BindAddress 0.0.0.0 RTSPPort 5004 RTSPBindAddress 0.0.0.0 MaxClients 10 MaxBandwidth 40000 NoDaemon <Feed feed3.ffm> File /tmp/feed3.ffm FileMaxSize 500M ACL allow localhost ACL allow 127.0.0.1 #ACL allow 192.168.0.0 192.168.255.255 </Feed> <stream webcamvideo.avi> Feed feed3.ffm Format rtp VideoCodec rawvideo VideoFrameRate 10 VideoBufferSize 40 VideoBitRate 36864 VideoSize 640x480 VideoGopSize 10 NoAudio AVOptionVideo flags +global_header </Stream> <Stream stat.html> Format status ACL allow localhost #ACL allow 127.0.0.1 </Stream> <Redirect index.html> URL http://www.ffmpeg.org/ </Redirect> I started the ffserver using the following command ffserver -d -f ffserverwebcam.conf FFmpeg command ffmpeg -f video4linux2 -s 176x244 -r 10 -i /dev/video0 -an -vcodec rawvideo http://localhost:8090/feed3.ffm FFmpeg version 0.5, Copyright (c) 2000-2009 Fabrice Bellard, et al. configuration: --enable-libmp3lame libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 0 / 52.20. 0 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 built on Oct 21 2009 16:43:19, gcc: 4.3.2 [video4linux2 @ 0xa26a290][3]Capabilities: 4000001 [video4linux2 @ 0xa26a290]The V4L2 driver changed the video from 176x244 to 176x144 Input #0, video4linux2, from '/dev/video0': Duration: N/A, start: 1270723080.673550, bitrate: 4055 kb/s Stream #0.0: Video: rawvideo, yuyv422, 176x144, 4055 kb/s, 10 tbr, 1000k tbn, 10 tbc [imgconvert @ 0xa26c9d0]PIX_FMT_YUV420P will be used as an intermediate format for rescaling Output #0, ffm, to 'http://localhost:8090/feed3.ffm': Stream #0.0: Video: rawvideo, yuv420p, 640x480, q=3-31, 36864 kb/s, 1000k tbn, 10 tbc Stream mapping: Stream #0.0 -> #0.0 Press [q] to stop encoding frame= 6384 fps= 10 q=0.0 Lsize= 2882760kB time=638.40 bitrate=36991.8kbits/s I get the following error when I use ffplay powerbase at powerbase-desktop:~$ ffplay rtsp://localhost:5004/webcamvideo.mpg FFplay version 0.5, Copyright (c) 2003-2009 Fabrice Bellard, et al. configuration: --enable-libmp3lame libavutil 49.15. 0 / 49.15. 0 libavcodec 52.20. 0 / 52.20. 0 libavformat 52.31. 0 / 52.31. 0 libavdevice 52. 1. 0 / 52. 1. 0 built on Oct 21 2009 16:43:19, gcc: 4.3.2 ^C[rtsp @ 0x926b6d0]Could not find codec parameters (Video: 0x0000) rtsp://localhost:5004/webcamvideo.mpg: could not find codec parameters I get the following error when I use mplayer powerbase at powerbase-desktop:~$ mplayer rtsp://localhost:5004/webcamvideo.mpg MPlayer 1.0rc2-4.3.2 (C) 2000-2007 MPlayer Team CPU: Intel(R) Core(TM)2 CPU 6300 @ 1.86GHz (Family: 6, Model: 15, Stepping: 2) CPUflags: MMX: 1 MMX2: 1 3DNow: 0 3DNow2: 0 SSE: 1 SSE2: 1 Compiled with runtime CPU detection. mplayer: could not connect to socket mplayer: No such file or directory Failed to open LIRC support. You will not be able to use your remote control. Playing rtsp://localhost:5004/webcamvideo.mpg. Resolving localhost for AF_INET6... Couldn't resolve name for AF_INET6: localhost Resolving localhost for AF_INET... Connecting to server localhost[127.0.0.1]: 5004... rtsp_session: unsupported RTSP server. Server type is 'unknown'. STREAM_LIVE555, URL: rtsp://localhost:5004/webcamvideo.mpg Stream not seekable! file format detected. Exiting... (End of file) Please let me know if you have any suggestion for the above issues. Thanks & Regards, Sushanth

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Question about ffmpeg use!!!
by enrico.cara＠gmail.com 08 Apr '10

08 Apr '10

Hello, I'm new with ffmpeg and for that I am not very able. I am working with the streaming of a video between two PC which I call here A and C. This streaming goes from A to C going through B, that is another PC. A and C are in different network but thanks to B I am able to ping each other. I set these address on the machines: A : IP ---> 10.0.0.4 default GW ---> 10.0.0.9 per eth0 B : IP ---> 10.0.0.9 default GW ---> 10.0.0.1 per eth0 IP ---> 18.0.0.9 default GW ---> 18.0.0.1 per eth1 C : IP ---> 18.0.0.4 default GW ---> 18.0.0.9 per eth0 What I want to do is to change the bit rate of the video streaming on the B side using ffmpeg. At this moment I don't know how could do that and if it is possible. Whether somebody knows the solution to such issue please sends it to me. Thanks a lot in advance. Enrico

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Audio stays out of sync with video
by ferenc.deak＠adactus.no 08 Apr '10

08 Apr '10

Hi all, I am trying to watch a live source on iphone, with the following setup: some app using ffmpeg libraries grabs the source and cuts it into 10 second long "chunks" and then ffmpeg converts them "live" to 100kbit and 300kbit, the final destination being watching it on the iphone. The following strange situation comes up: the segment files when played separately (with ffplay) are fine and ok, but when they are played continuously on the iphone after some time the audio gets out of sync. Usually the first file played on the iphone is ok, but after 10 - 15 minutes of play I get a ~ 1 second difference between the audio and video, depending on the bitrate, the audio is either before or after the video (on lower bitrates the video is before the audio, on higher bitrates the audio is before the video). Another really annoying "feature" is that for some TS files there is no audio, or the audio is "packed" into the first 0.1 seconds, and I hear a pretty annoying artefact at the beginning of the file and then silence for the remaining seconds. With the next file the audio comes back normally. Have you ever seen this behavior? If yes did you manage to fix it? If yes could you please tell me how? Thanks and regards, Ferenc

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timescale of a container
by sangeetha＠julysystems.com 08 Apr '10

08 Apr '10

Hi, I need to change the timescale of a container from 600 to 90000 because I need to stitch this file with some other file. All the values I need to match with that file. All other values are matching. Can anyone help me out to change that. I tried with MP4Box fps option. It is not reflected. Is there any option in ffmpeg? MP4Box -info score.mp4 * Movie Info * Timescale 600 - Duration 00:00:07.920 Fragmented File no - 1 track(s) File Brand mp42 - version 0 Created: GMT Thu Apr 08 17:03:30 2010 File has root IOD Scene PL 0xff - Graphics PL 0xff - OD PL 0xff Visual PL: AVC/H264 Profile (0x15) Audio PL: No audio capability required (0xff) No streams included in root OD Track # 1 Info - TrackID 1 - TimeScale 25000 - Duration 00:00:07.920 Media Info: Language "Undetermined" - Type "vide:avc1" - 198 samples MPEG-4 Config: Visual Stream - ObjectTypeIndication 0x21 AVC/H264 Video - Visual Size 320 x 240 - Profile High @ Level 3 NAL Unit length bits: 32 Pixel Aspect Ratio 1:1 - Indicated track size 320 x 240 Self-synchronized Thanks, Sangeetha

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What to change to switch stderr with stdout?
by jamiesgreen＠gmail.com 08 Apr '10

08 Apr '10

Hi, Can someone give me a direction of (where and) what I need to change to the source so the output of FFmpeg (frame= time=) can be changed to stdout but the other messages still at stderr? I know I can simply redirect them in the shell but that's not what I'm looking for.. Thanks in advance.

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Recalculate PTS/DTS
by sun_radha＠yahoo.com 07 Apr '10

07 Apr '10

Hi All, I was wondering if some one can point me to function/s in libavformat/codec that can help me recalculate the pts/dts of frames. I am writing a application to re-encode a video, looked at ffmpeg.c but it does a lot of stuff and the pts/dts calculation part if not very intuitive. Thanks for any help Kris

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