November 2013 - ffmpeg-user - lists.ffmpeg.org

Audio normalization using "volume" and "compand" filters
by Thierry Lelegard 02 Dec '13

02 Dec '13

Hello, A have a couple of questions regarding the normalization of audio levels in a file. Note that I am not an audio expert. The idea is to bring the RMS level to a given value "out_rms", say -20 dBFS, with a given maximum peak level "out_peak_max", say -1 dBFS. In a first pass, I measure "in_rms" and "in_peak" using the audio filter "volumedetect". My first thought is to use the audio filter "volume" if the input dynamics is less than the output one (ie. in_peak - in_rms < out_peak_max - out_rms). I expect to shift the whole signal to out_rms without distorsion and keep out_peak < out_peak_max. My second thought is to use the audio filter "compand" to adjust the volume and compress the dynamics if the input dynamics is too large. I expect to obtain out_rms for the mean level and out_peak = out_peak_max. First problem: The filter "volume" works but the actual adjustment is always shifted by -0.5dB from the requested value. For -af volume=+5dB, I get an actual +4.5 dB. For -af volume=-10dB, I get -10.5 dB. Etc. See the details below. Second problem: The usage of the filter "compand" is extremely obscure. Its documentation (http://ffmpeg.org/ffmpeg-filters.html#compand) can hardly be understood if you do not already know the meaning of each parameter. See below some tests I made without deeply understanding what they mean. ------------------------ More details on the -0.5dB offset with the filter "volume": Let's take as input an MP3 file with low-level audio volume. > ffmpeg -i test-audio-1.mp3 -af volumedetect -f null -y nul ffmpeg version 2.1.1 Copyright (c) 2000-2013 the FFmpeg developers built on Nov 20 2013 21:13:48 with gcc 4.8.2 (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-libcaca --enable-libfreetype --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-libx264 --enable-libxavs --enable-libxvid --enable-zlib libavutil 52. 48.101 / 52. 48.101 libavcodec 55. 39.101 / 55. 39.101 libavformat 55. 19.104 / 55. 19.104 libavdevice 55. 5.100 / 55. 5.100 libavfilter 3. 90.100 / 3. 90.100 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.104 / 0. 17.104 libpostproc 52. 3.100 / 52. 3.100 Input #0, mp3, from 'test-audio-1.mp3': Metadata: encoder : Lavf55.19.104 Duration: 00:01:00.00, start: 0.000000, bitrate: 96 kb/s Stream #0:0: Audio: mp3, 48000 Hz, stereo, s16p, 96 kb/s Output #0, null, to 'nul': Metadata: encoder : Lavf55.19.104 Stream #0:0: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s Stream mapping: Stream #0:0 -> #0:0 (mp3 -> pcm_s16le) Press [q] to stop, [?] for help size=N/A time=00:01:00.00 bitrate=N/A video:0kB audio:11248kB subtitle:0 global headers:0kB muxing overhead -100.000191% [Parsed_volumedetect_0 @ 0000000002888ba0] n_samples: 5758942 [Parsed_volumedetect_0 @ 0000000002888ba0] mean_volume: -34.6 dB [Parsed_volumedetect_0 @ 0000000002888ba0] max_volume: -14.5 dB [Parsed_volumedetect_0 @ 0000000002888ba0] histogram_14db: 23 [Parsed_volumedetect_0 @ 0000000002888ba0] histogram_15db: 92 [Parsed_volumedetect_0 @ 0000000002888ba0] histogram_16db: 219 [Parsed_volumedetect_0 @ 0000000002888ba0] histogram_17db: 778 [Parsed_volumedetect_0 @ 0000000002888ba0] histogram_18db: 2862 [Parsed_volumedetect_0 @ 0000000002888ba0] histogram_19db: 7000 The RMS level is -34.6 dBFS and the peak level is -14.5 dBFS. I apply the audio filter "volume" with the following command and check the audio levels of the output file using "-af volumedetect" again. ffmpeg -i test-audio-1.mp3 -af "volume=...dB" -y test-audio-2.mp3 ffmpeg -i test-audio-2.mp3 -af volumedetect -f null -y nul Here are the results with various volume options: Input file: mean_volume: -34.6 dB max_volume: -14.5 dB Using -af "volume=+5dB": mean_volume: -30.1 dB => +4.5 max_volume: -9.9 dB => +4.6 Using -af "volume=+13.5dB": mean_volume: -21.6 dB => +13 max_volume: -1.4 dB => +13.1 Using -af "volume=-10dB": mean_volume: -45.1 dB => -10.5 max_volume: -24.9 dB => -10.4 So it seems that the actual adjustment is the requested value - 0.5 for the RMS level and - 0.4 for the peak. Is there any rational explanation for this? How can we compute the right volume adjustment for bringing the input RMS level to a precise value (apart from blindly applying a hardcoded +0.5 to compensate the observed behavior)? ------------------------ Some tests using the filter "compand". Let's use the same input file with in_rms = -34.6 dBFS and in_peak = -14.5 dBFS. Let's try to compress the dynamics and adjust the level so that out_rms = -10 dBFS and out_peak = -1 dBFS. This is too compressed, I know, this is just to test the filter. My understanding is to get a dynamics of peak - rms = 9 dB, so compress the signal centered on the in_rms = -34.6 dBFS with a peak of -25.6 dBFS and then apply a gain of +24.6 dB. Am I right or completely out? Using a mixture of recommended values in the documentation and what I think I understand of the "compand" parameters, I tried the following filters with the following results. Input file mean_volume: -34.6 dB max_volume: -14.5 dB -af "compand=attacks=0.3 0.3:decays=0.8 0.8:points=-90/-900 -70/-70 -34.6/-34.6 -14.5/-25.6 0/-25.6:soft-knee=0.01:gain=24.6:volume=0:delay=0.8" mean_volume: -12.6 dB max_volume: 0.0 dB I though that the points "-90/-900 -70/-70 -34.6/-34.6 -14.5/-25.6 0/-25.6" would keep -34.6 dbFS at its level and compress -14.5 dBFS and higher to -25.6 and then the signal would be shifted up to out_rms = -10 dBFS and out_peak = -1 dBFS. RMS is not that far from -10 dBFS but there is clearly some clipping. Same test without the additional gain: -af "compand=attacks=0.3 0.3:decays=0.8 0.8:points=-90/-900 -70/-70 -34.6/-34.6 -14.5/-25.6 0/-25.6:soft-knee=0.01:gain=0:volume=0:delay=0.8" mean_volume: -36.7 dB max_volume: -17.1 dB The RMS has been lowered and the dynamics is 19.6 dB instead of 9. Could someone please explain how to use "compand" and its parameters? More precisely, how can we compress an input audio with the characteristics in_rms and in_peak into a given target out_rms and out_peak? Thanks for your help. -Thierry

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ffmpeg decoding DCP MXF with SMPTE labels
by Rens Dijkshoorn 01 Dec '13

01 Dec '13

Hi, is decoding of DCP cinema4k jpeg2000.mxf files with SMPTE Labels in the ffmpeg developement pipeline. MXF Interop labels is working. Rens

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Stream #X:Y: Data: none
by Vincent Olivier 30 Nov '13

30 Nov '13

Hi, I am wondering how I can start parsing an unrecognized data stream programatically (I am already using libavcodec, etc. to parse the audio and the video). I couldn’t find any documentation on how to start doing that. Please find the ffprobe dump below. Regards, Vincent ffprobe version 2.0-bb7f3dd Copyright (c) 2007-2013 the FFmpeg developers built on Nov 29 2013 19:43:38 with Apple LLVM version 5.0 (clang-500.2.79) (based on LLVM 3.3svn) configuration: --prefix=/opt/local --enable-swscale --enable-avfilter --enable-avresample --enable-libmp3lame --enable-libvorbis --enable-libopus --enable-libtheora --enable-libschroedinger --enable-libopenjpeg --enable-libmodplug --enable-libvpx --enable-libspeex --enable-libass --enable-libbluray --enable-gnutls --enable-libfreetype --disable-indev=jack --disable-outdev=xv --mandir=/opt/local/share/man --enable-shared --enable-pthreads --cc=/usr/bin/clang --arch=x86_64 --enable-yasm --enable-gpl --enable-postproc --enable-libx264 --enable-libxvid --enable-nonfree --enable-libfaac libavutil 52. 53.100 / 52. 53.100 libavcodec 55. 43.101 / 55. 43.101 libavformat 55. 21.100 / 55. 21.100 libavdevice 55. 5.100 / 55. 5.100 libavfilter 3. 91.100 / 3. 91.100 libavresample 1. 1. 0 / 1. 1. 0 libswscale 2. 5.101 / 2. 5.101 libswresample 0. 17.104 / 0. 17.104 libpostproc 52. 3.100 / 52. 3.100 [mxf @ 0x7fc573054400] could not resolve sub descriptor strong ref [mxf @ 0x7fc573054400] source track 11: stream 9, no descriptor found [mxf @ 0x7fc573054400] decoding for stream 0 failed Input #0, mxf, from ‘input.mxf': Metadata: uid : 22bbd33a-7608-4329-b2bc-6009c08be0e3 generation_uid : f4962248-052a-47a1-a3af-16fa2b74dd78 company_name : Sony product_name : util product_version : 1.00 application_platform: Sony MXF Development Kit (MacOS X) product_uid : cede4403-8280-11de-8a39-08004678031c modification_date: 2013-08-02 04:47:37 timecode : 05:07:35:00 Duration: 00:00:00.08, start: 0.000000, bitrate: 143679 kb/s Stream #0:0: Video: h264 (High 4:2:2 Intra), yuv422p10le, 4096x2160 [SAR 1:1 DAR 256:135], 23.98 tbr, 23.98 tbn, 47.95 tbc Stream #0:1: Audio: pcm_s24le, 48000 Hz, 1 channels, s32, 1152 kb/s Stream #0:2: Audio: pcm_s24le, 48000 Hz, 1 channels, s32, 1152 kb/s Stream #0:3: Audio: pcm_s24le, 48000 Hz, 1 channels, s32, 1152 kb/s Stream #0:4: Audio: pcm_s24le, 48000 Hz, 1 channels, s32, 1152 kb/s Stream #0:5: Audio: pcm_s24le, 48000 Hz, 1 channels, s32, 1152 kb/s Stream #0:6: Audio: pcm_s24le, 48000 Hz, 1 channels, s32, 1152 kb/s Stream #0:7: Audio: pcm_s24le, 48000 Hz, 1 channels, s32, 1152 kb/s Stream #0:8: Audio: pcm_s24le, 48000 Hz, 1 channels, s32, 1152 kb/s Stream #0:9: Data: none Unsupported codec with id 0 for input stream 9

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scale video filter and force_original_aspect_ratio option
by James Darnley 29 Nov '13

29 Nov '13

Does the force_original_aspect_ratio option of the scale filter work as described in the online docs? > One useful instance of this option is that when you know a specific > device’s maximum allowed resolution, you can use this to limit the > output video to that, while retaining the aspect ratio. For example, > device A allows 1280x720 playback, and your video is 1920x800. Using > this option (set it to decrease) and specifying 1280x720 to the > command line makes the output 1280x533. I read that to be "If and only if the video is larger than the size you set it will be scaled to fit within the size you set." Is that reading wrong? I've been trying to get it to work when encoding a random set of files, some smaller and some larger, than what I want but all are resized to fit. In my example below we can clearly see that a video of 704x576 is resized up to 880x720 when I requested 1280x720 with force_original_aspect_ratio=decrease command: > ffmpeg -loglevel verbose -y -i "Q:\test clips\crew_4cif.y4m" -vcodec mpeg2video -q 4 -vframes 1 -vf "scale=1280:720:force_original_aspect_ratio=decrease" temp.vob > ffmpeg version N-58485-ga12b4bd Copyright (c) 2000-2013 the FFmpeg developers > built on Nov 26 2013 22:07:02 with gcc 4.8.2 (GCC) > configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av > isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab > le-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp > e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --ena > ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-l > ibopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libsp > eex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aa > cenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavp > ack --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib > libavutil 52. 55.100 / 52. 55.100 > libavcodec 55. 44.100 / 55. 44.100 > libavformat 55. 21.102 / 55. 21.102 > libavdevice 55. 5.101 / 55. 5.101 > libavfilter 3. 91.100 / 3. 91.100 > libswscale 2. 5.101 / 2. 5.101 > libswresample 0. 17.104 / 0. 17.104 > libpostproc 52. 3.100 / 52. 3.100 > Input #0, yuv4mpegpipe, from 'Q:\test clips\crew_4cif.y4m': > Duration: N/A, bitrate: N/A > Stream #0:0: Video: rawvideo (I420 / 0x30323449), yuv420p, 704x576, SAR 128: > 117 DAR 1408:1053, 60 fps, 60 tbr, 60 tbn, 60 tbc > [Parsed_scale_0 @ 000000000502fa60] w:1280 h:720 flags:'0x4' interl:0 > [graph 0 input from stream 0:0 @ 0000000002568100] w:704 h:576 pixfmt:yuv420p tb > :1/60 fr:60/1 sar:128/117 sws_param:flags=2 > [Parsed_scale_0 @ 000000000502fa60] w:704 h:576 fmt:yuv420p sar:128/117 -> w:880 > h:720 fmt:yuv420p sar:128/117 flags:0x4 > [svcd @ 000000000502d960] VBV buffer size not set, muxing may fail > Output #0, svcd, to 'temp.vob': > Metadata: > encoder : Lavf55.21.102 > Stream #0:0: Video: mpeg2video, yuv420p, 880x720 [SAR 128:117 DAR 1408:1053] > , q=2-31, 200 kb/s, 90k tbn, 60 tbc > Stream mapping: > Stream #0:0 -> #0:0 (rawvideo -> mpeg2video) > Press [q] to stop, [?] for help > No more output streams to write to, finishing. > frame= 1 fps=0.0 q=4.0 Lsize= 50kB time=00:00:00.01 bitrate=24575.5kbits > /s > video:46kB audio:0kB subtitle:0 global headers:0kB muxing overhead 9.096333% And with a value of 1 instead of string "decrease": > ffmpeg -loglevel verbose -y -i "Q:\test clips\crew_4cif.y4m" -vcodec mpeg2video -q 4 -vframes 1 -vf "scale=1280:720:force_original_aspect_ratio=1" temp.vob > ffmpeg version N-58485-ga12b4bd Copyright (c) 2000-2013 the FFmpeg developers > built on Nov 26 2013 22:07:02 with gcc 4.8.2 (GCC) > configuration: --enable-gpl --enable-version3 --disable-w32threads --enable-av > isynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls --enab > le-iconv --enable-libass --enable-libbluray --enable-libcaca --enable-libfreetyp > e --enable-libgsm --enable-libilbc --enable-libmodplug --enable-libmp3lame --ena > ble-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-l > ibopus --enable-librtmp --enable-libschroedinger --enable-libsoxr --enable-libsp > eex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvo-aa > cenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libwavp > ack --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib > libavutil 52. 55.100 / 52. 55.100 > libavcodec 55. 44.100 / 55. 44.100 > libavformat 55. 21.102 / 55. 21.102 > libavdevice 55. 5.101 / 55. 5.101 > libavfilter 3. 91.100 / 3. 91.100 > libswscale 2. 5.101 / 2. 5.101 > libswresample 0. 17.104 / 0. 17.104 > libpostproc 52. 3.100 / 52. 3.100 > Input #0, yuv4mpegpipe, from 'Q:\test clips\crew_4cif.y4m': > Duration: N/A, bitrate: N/A > Stream #0:0: Video: rawvideo (I420 / 0x30323449), yuv420p, 704x576, SAR 128: > 117 DAR 1408:1053, 60 fps, 60 tbr, 60 tbn, 60 tbc > [Parsed_scale_0 @ 000000000502ec40] w:1280 h:720 flags:'0x4' interl:0 > [graph 0 input from stream 0:0 @ 000000000502f040] w:704 h:576 pixfmt:yuv420p tb > :1/60 fr:60/1 sar:128/117 sws_param:flags=2 > [Parsed_scale_0 @ 000000000502ec40] w:704 h:576 fmt:yuv420p sar:128/117 -> w:880 > h:720 fmt:yuv420p sar:128/117 flags:0x4 > [svcd @ 0000000000300a20] VBV buffer size not set, muxing may fail > Output #0, svcd, to 'temp.vob': > Metadata: > encoder : Lavf55.21.102 > Stream #0:0: Video: mpeg2video, yuv420p, 880x720 [SAR 128:117 DAR 1408:1053] > , q=2-31, 200 kb/s, 90k tbn, 60 tbc > Stream mapping: > Stream #0:0 -> #0:0 (rawvideo -> mpeg2video) > Press [q] to stop, [?] for help > No more output streams to write to, finishing. > frame= 1 fps=0.0 q=4.0 Lsize= 50kB time=00:00:00.01 bitrate=24575.5kbits > /s > video:46kB audio:0kB subtitle:0 global headers:0kB muxing overhead 9.096333%

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ffmpeg tool compilation problem
by Francis Kumar 29 Nov '13

29 Nov '13

Dear Sir, I am trying to configure and building the ffmpeg using arm tool chain, It is not building success, Please let us know the Supported software for building and using for webcam. Find the logs ---------------------------------------------------------------------------- ----------------------------------------------------------- root@ubuntu:/media/francis/PDA-A2/480x272# cd ffmpeg-0.10.1 root@ubuntu:/media/francis/PDA-A2/480x272/ffmpeg-0.10.1# ls Changelog common.mak COPYING.LGPLv2.1 Doxyfile ffprobe.c libavdevice libpostproc LICENSE presets tools cmdutils.c configure COPYING.LGPLv3 ffmpeg.c ffserver.c libavfilter library.mak MAINTAINERS README VERSION cmdutils_common_opts.h COPYING.GPLv2 CREDITS ffplay.c INSTALL libavformat libswresample Makefile RELEASE version.sh cmdutils.h COPYING.GPLv3 doc ffpresets libavcodec libavutil libswscale mt-work tests root@ubuntu:/media/francis/PDA-A2/480x272/ffmpeg-0.10.1# ./configure --host=arm-linux-gnueabi CC=arm-none-linux-gnueabi-gcc Unknown option "--host=arm-linux-gnueabi". See ./configure --help for available options. root@ubuntu:/media/francis/PDA-A2/480x272/ffmpeg-0.10.1# ./configure CC=arm-none-linux-gnueabi-gcc Unknown option "CC=arm-none-linux-gnueabi-gcc". See ./configure --help for available options. root@ubuntu:/media/francis/PDA-A2/480x272/ffmpeg-0.10.1# ./configure --prefix=/media/francis/PDA-A2/480x272/ffmpeg --host=arm-linux-gnueabi CC=arm-none-linux-gnueabi-gcc Unknown option "--host=arm-linux-gnueabi". See ./configure --help for available options. root@ubuntu:/media/francis/PDA-A2/480x272/ffmpeg-0.10.1# make Makefile:2: config.mak: No such file or directory Makefile:45: /common.mak: No such file or directory Makefile:84: /libavutil/Makefile: No such file or directory Makefile:84: /library.mak: No such file or directory Makefile:161: /doc/Makefile: No such file or directory Makefile:162: /tests/Makefile: No such file or directory make: *** No rule to make target `/tests/Makefile'. Stop. root@ubuntu:/media/francis/PDA-A2/480x272/ffmpeg-0.10.1# ./configure yasm not found, use --disable-yasm for a crippled build If you think configure made a mistake, make sure you are using the latest version from Git. If the latest version fails, report the problem to the ffmpeg-user(a)ffmpeg.org mailing list or IRC #ffmpeg on irc.freenode.net. Include the log file "config.log" produced by configure as this will help solving the problem. root@ubuntu:/media/francis/PDA-A2/480x272/ffmpeg-0.10.1# root@ubuntu:/media/francis/PDA-A2/480x272/ffmpeg-0.10.1# ./configure yasm not found, use --disable-yasm for a crippled build If you think configure made a mistake, make sure you are using the latest version from Git. If the latest version fails, report the problem to the ffmpeg-user(a)ffmpeg.org mailing list or IRC #ffmpeg on irc.freenode.net. Include the log file "config.log" produced by configure as this will help solving the problem. root@ubuntu:/media/francis/PDA-A2/480x272/ffmpeg-0.10.1# ---------------------------------------------------------------------------- ----------------------------------------------------------- Thanks, Francis

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freior
by ray 29 Nov '13

29 Nov '13

Good morning. I try to run the FREIOR effects with FFMPEG (Windows) I specify impossible to convert the video file here are the problems I have, can you inform me. Excuse my bad english this is a translation, I'm French. Thank you in anticipation. <http://ffmpeg-users.933282.n4.nabble.com/file/n4662484/le_vrai_freior.jpg> -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/freior-tp4662484.html Sent from the FFmpeg-users mailing list archive at Nabble.com.

2 2

Help transcoding TS files
by Young Kim 29 Nov '13

29 Nov '13

Hey all, I've been trying to transcode some TS files with the requirement that the PTS is preserved in both the audio and video. However, I've been having some issues with this. In particular, here's an example of a command I've ran: ffmpeg -re -i raw/4bZe7TBKQ8Q1p3B_1384459221193_1384459221193_094.ts -s 640x480 -vcodec libx264 -vprofile main -preset veryfast -tune animation -acodec copy -y -mpegts_copyts 1 test/4bZe7TBKQ8Q1p3B_1384459221193_1384459221193_094.ts After running this, I ran ffprobe to make sure that the duration and pts of both is similar, but it's a bit off: raw: duration - 9.968 stream_info - {'audio': {'duration_ts': 898610, 'start_pts': 83149425}, 'video': {'duration_ts': 897120, 'start_pts': 83150550}} test: duration - 9.968289 stream_info - {'audio': {'duration_ts': 898610, 'start_pts': 7508}, 'video': {'duration_ts': 897146, 'start_pts': 7508}} I've tried using vsync, async, copyts, and copytb to see if it would fix the problem, but a combination of these either freezes ffmpeg or results in the same results as above. Does anyone have any suggestions? Thanks, Young

2 1

ERROR: Xext not found
by Luis LROR 29 Nov '13

29 Nov '13

hello good morning friends, I wonder if you guys can help me, estoutendo problems in the binary FFMPEG installation on Debian 7 chuego in step 3 when he presents the following ERROR Step 1 git clone - depth 1 git :/ / source.ffmpeg.org / ffmpeg Step2 cd ffmpeg Step 3 . / configure - enable-gpl - enable-libass - enable-libfaac - enable-libfdk-aac - enable-libmp3lame \ - enable-libopencore-amrnb - enable-libopencore-amrwb - enable-libspeex - enable-librtmp - enable-libtheora \ - enable-libvorbis - enable-libvpx - enable-x11grab - enable-libx264 - enable-nonfree - enable-version3 ERROR Presented ERROR: xExt not found If you think configure made a mistake, make sure you are using the latest version from Git. If the latest version fails, report the problem to the ffmpeg-user(a)ffmpeg.org mailing list or IRC # ffmpeg on irc.freenode.net. Include the log file "config.log" produced by configure this will help solving the problem. I thank *_________________________________________________________________________________* *Luis Renato Oliveira* *Operador de Web Tv * *Coopas / Canal Saúde - Fiocruz* *Cood. de Tecnologia da Informação* *Av. Brasil, 4365 - Manguinhos* *Cel.: 021 8685-1094* *Tel.: 021 3194-7765*

3 2

Unable to read certain RTMP streams with FFmpeg\FFplay\FFProbe
by dokosten 28 Nov '13

28 Nov '13

Hi there! It seems like I've been trying everything, but I'm having no luck at all reading the following stream using ffmpeg, ffplay or ffprobe. *rtmp://85.19.218.242:1935/liveonly_video/5207?userId=b337&pasword=8e1e3a26ab2a85b48eaaf129d338fdff&channelId=5207/stream* I'm using the latest windows build. (ffmpeg-20131122-git-fb7d70c-win32-static) I've also tried with older versions of FFmpeg which I've compiled for iOS, with no luck. I know there is nothing wrong with the stream as it works in other players like this one: http://www.wowza.com/resources/3.5.0/examples/LiveVideoStreaming/FlashRTMPP… *Server:* rtmp://85.19.218.242:1935/liveonly_video/5207?userId=b337&pasword=8e1e3a26ab2a85b48eaaf129d338fdff&channelId=5207 *Stream:* stream This is the only error data I get from FFmpeg: *Closing connection: NetStream.Play.StreamNotFound rtmp://85.19.218.242:1935/liveonly_video/5207?userId=b337&pasword=8e1e3a26ab2a85 b48eaaf129d338fdff&channelId=5207/stream: Unknown error occurred* Any help would be greatly appreciated! Thanks. -- View this message in context: http://ffmpeg-users.933282.n4.nabble.com/Unable-to-read-certain-RTMP-stream… Sent from the FFmpeg-users mailing list archive at Nabble.com.

1 1

Preserve bitdepth of input audio file during conversion to wav
by Michal Šmucr 27 Nov '13

27 Nov '13

Hello to all, i would like to ask you. Is there any FFmpeg option or trick to preserve bitdepth of input file during decompression to wave file? For instance during conversion of 24bit FLAC or ALAC audio to WAV. When i use straight "ffmpeg -i whatever-24bit.flac output.wav", resulting file is always coded as PCM_S16LE. I didn't find any way, how to do this, except of some wrapper script, which will run ffmpeg or ffprobe before conversion, parse it's output for audio stream info and set proper format to subsequent conversion. However this approach is quite cumbersome, because it needs to lookup in some table with corresponding formats for particular codec as for instance ALAC with bitdepth > 16bit has format PCM_S32P which usually needs to be adjusted to PCM_S24LE for output wav. And on Windows it also requires to install some additional script interpreter (like cygwin bash and coreutils or python). Thanks Michal

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