I'm attempting to transcode a DVD to a single MKV file. I've had success in the past with other DVDs, but I'm running into an error I haven't seen before.
First I concatenate the VOB files I want to transcode:
cat VTS_02_1.VOB VTS_02_2.VOB VTS_02_3.VOB > WMAV.VOB
ffprobe output:
$ ffprobe -analyzeduration 100M -probesize 100M WMAV.VOB Input #0, mpeg, from 'WMAV.VOB':
Duration: 01:05:19.42, start: 0.300300, bitrate: 5686 kb/s
Stream #0:0[0x1bf]: Data: dvd_nav_packet
Stream #0:1[0x1e0]: Video: mpeg2video (Main), yuv420p(tv, smpte170m, top first), 720x480 [SAR 32:27 DAR 16:9], 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc
Stream #0:2[0x80]: Audio: ac3, 48000 Hz, stereo, fltp, 192 kb/s
Unsupported codec with id 100357 for input stream 0
Then I run this command to transcode the file:
ffmpeg -analyzeduration 100M -probesize 100M \
-i WMAV.VOB \
-map 0:1 -map 0:2 \
-c:v libx264 -preset slow -tune film -crf 21 \
-c:a aac -b:a 192k \
wmav.mkv
However, when I include -c:a aac, I get thousands of errors like this:
Error while decoding stream #0:2: Error number -16976906 occurred
[ac3 # 000002bd24d8eec0] expacc 127 is out-of-range
[ac3 # 000002bd24d8eec0] error decoding the audio block
There doesn't seem to be any issue with the audio stream since it plays back fine in VLC. The transcode succeeds if I use -c:a copy.
What is causing this error and how could I fix the problem?
Related
I've been using this command to convert my public rtmp audio/video stream to a local mp3 audio icecast2 stream, but I have been unable to do the same for both video and audio.
[Audio Only] (This works fine)
ffmpeg -re -i rtmp://162.142.xx.xxx:xxx/stream -vn -codec:a libmp3lame -b:a 128k -f mp3 -content_type audio/mpeg icecast://source:password#192.168.1.xxx:80/live
I've tried to re-write in order to support video, but I keep hitting dead ends
[Audio & Video Attempt] (this does not work)
ffmpeg -re -i rtmp://162.142.xx.xxx:xxx/stream -codec:v -f mpeg4 -b:v -f mpeg4 -content_type video/mpeg4 icecast://source:password#192.168.1.xxx:80/live
When I run this command, it gives me the error below asking for a suitable format.
$ ffmpeg -re -i rtmp://162.142.xx.xxx:xxx/stream -codec:v -f mpeg4 -b:v -f mpeg4 -content_type video/mpeg4 icecast://source:password#192.168.1.xxx:80/live
[h264 # 0x5598ffbb8980] co located POCs unavailable
[h264 # 0x5598ffbb8980] mmco: unref short failure
Input #0, flv, from 'rtmp://162.142.xx.xxx:xxx/stream':
Metadata:
|RtmpSampleAccess: true
Server : NGINX RTMP (github.com/sergey-dryabzhinsky/nginx-rtmp-module)
displayWidth : 1280
displayHeight : 720
fps : 48
videokeyframe_frequency: 0
profile :
level :
Duration: 00:00:00.00, start: 28117.779000, bitrate: N/A
Stream #0:0: Audio: aac (LC), 48000 Hz, stereo, fltp, 327 kb/s
Stream #0:1: Video: h264 (High), yuv420p(tv, bt709, progressive), 1280x720 [SAR 1:1 DAR 16:9], 2560 kb/s, 48 fps, 48 tbr, 1k tbn
[NULL # 0x5598ffb8bec0] Unable to find a suitable output format for 'mpeg4'
mpeg4: Invalid argument
I am positive that icecast2 can support video streams, however on the few occasions that I was able to actively stream successfully to it, it only showed an empty video embed.
I've re-written the command for AV multiple times while referencing ffmpeg documentation, however my above attempt seems to be the closest (concept-wise) that I have gotten.
What flags/formatting might I be missing which are causing the stream not to work?
i have .ts file looks like the following
Input #0, mpegts, from 'i.ts':
Duration: 00:00:36.32, start: 28752.398067, bitrate: 57694 kb/s
Program 50
Metadata:
service_name : aaa HD
service_provider:
Stream #0:51[0x1f5]: Video: h264 ([27][0][0][0] / 0x001B), none, 90k tbr, 90k tbn, 180k tbc
Stream #0:52[0x1f6]: Audio: mp3 ([3][0][0][0] / 0x0003), 0 channels, fltp
Program 51
Metadata:
service_name : b Music HD
service_provider:
Stream #0:16[0x1ff]: Video: h264 ([27][0][0][0] / 0x001B), none, 90k tbr, 90k tbn, 180k tbc
Stream #0:17[0x200]: Audio: mp3 ([3][0][0][0] / 0x0003), 0 channels, fltp
Program 52
Metadata:
service_name : c ch HD
service_provider:
Stream #0:14[0x209]: Video: h264 ([27][0][0][0] / 0x001B), none, 90k tbr, 90k tbn, 180k tbc
Stream #0:15[0x20a]: Audio: mp3 ([3][0][0][0] / 0x0003), 0 channels, fltp
Program 1510
Metadata:
service_name : asd
service_provider: xyz
Stream #0:18[0x5e7]: Video: h264 ([27][0][0][0] / 0x001B), none, 90k tbr, 90k tbn, 180k tbc
Stream #0:19[0x5e8]: Audio: mp3 ([3][0][0][0] / 0x0003), 0 channels, fltp
i need to extract one video stream and its audio stream from this file for example related to program 50 i tried
ffmpeg -i i.ts -map 0:51 output.mp4
but i got this error
Stream mapping:
Stream #0:51 -> #0:0 (h264 (native) -> h264 (libx264))
Press [q] to stop, [?] for help
Cannot determine format of input stream 0:51 after EOF
Error marking filters as finished
Conversion failed!
I found a solution here
For mapping the whole programs, the syntax is:
ffmpeg -i i.ts -c:v copy -c:a copy -map 0:p:51 output.mp4
I can't verify that solution is actually working with your .ts file.
I created the following sample, that builds a .ts file with two programs, and then extracts each program to .mp4 file:
ffmpeg -y -r 10 -f lavfi -i testsrc=rate=10:size=160x120 -f lavfi -i sine=frequency=1000 -t 5 -c:v libx264 -c:a aac in1.mp4
ffmpeg -y -r 10 -f lavfi -i mandelbrot=rate=10:size=160x120 -f lavfi -i sine=frequency=300 -t 5 -c:v libx264 -c:a aac in2.mp4
ffmpeg -y -i in1.mp4 -i in2.mp4 -map 0:0 -map 0:1 -map 1:0 -map 1:1 -program title=ProgOne:st=0:st=1 -program title=ProgTwo:st=2:st=3 -c:v copy -c:a copy in.ts
ffmpeg -y -i in.ts -c:v copy -c:a copy -map 0:p:1 output1.mp4
ffmpeg -y -i in.ts -c:v copy -c:a copy -map 0:p:2 output2.mp4
1st command builds video test pattern with high frequency beep (output: in1.mp4).
2nd command builds video Mandelbrot pattern with low frequency beep (output: in2.mp4).
3rd command builds transport stream with two programs (output: in.ts).
4th command extracts first program (output: output1.mp4).
5th command extracts second program (output: output2.mp4).
EDIT: It already works. My new router seems to block rtmp traffic. I edited the firewall config and now it works!
I'm trying to stream to YouTube with ffmpeg with the following command:
ffmpeg -f alsa -ac 2 -i hw:0,0 -f v4l2 -s 1280x720 -r 10 -i /dev/video0 -vcodec libx264 -pix_fmt yuv420p -preset ultrafast -strict experimental -r 25 -g 20 -b:v 2500k -codec:a libmp3lame -ar 44100 -b:a 11025 -bufsize 512k -f flv rtmp://a.rtmp.youtube.com/live2/45ee-qka9-0djm-796z
My ffmpeg version 2.8.11 on Linux Mint 18.2 then says:
Input #0, alsa, from 'hw:0,0':
Duration: N/A, start: 1511081240.166016, bitrate: 1536 kb/s
Stream #0:0: Audio: pcm_s16le, 48000 Hz, 2 channels, s16, 1536 kb/s
[video4linux2,v4l2 # 0xacba60] The driver changed the time per frame from 1/10 to 2/15
Input #1, video4linux2,v4l2, from '/dev/video0':
Duration: N/A, start: 5907.755626, bitrate: 110592 kb/s
Stream #1:0: Video: rawvideo (YUY2 / 0x32595559), yuyv422, 1280x720, 110592 kb/s, 7.50 fps, 7.50 tbr, 1000k tbn, 1000k tbc
After that it does nothing for about 3 minutes,then it finally returns the following error:
RTMP_Connect0, failed to connect socket. 110 (Connection timed out)
rtmp://a.rtmp.youtube.com/live2: Unknown error occurred
I used these instructions:
https://gist.github.com/laurenarcher/4644aacef51e734d33d5
FFMPEG to Youtube Live
Finally, I found an easy solution. Quite unexpectedly, the firewall of my router blocks outgoing RTMP traffic by default. By adding a port trigger rule for the well-known RTMP port (1935-TCP), I managed to work around this issue.
I am aware that this is not the most elegant fix, but my ISP's router does not exactly boast with configuration options, so the method above was the only possible way to address this problem.
i have a problem about transcode with ffmpeg
i want to cover m3u8 to mp4, so i transcode every ts file first, and then concat them to a mp4, but i found that the duration will be bigger than source file.
source file is :
http://oc7iy3eta.bkt.clouddn.com/src_20.ts
after transcode, test file is:
http://oc7iy3eta.bkt.clouddn.com/test_20.ts
i use the command as bellow to change to 5fps, and 400k bitrate:
sudo ffmpeg -analyzeduration 2147483647 -probesize 2147483647 -nostdin -y -v warning -i ./src_20.ts -threads 3 -movflags faststart -metadata:s:v rotate=0 -chunk_duration 520000 -video_track_timescale 25000 -pix_fmt yuv420p -copytb 1 -vcodec libx264 -b:v 400000 -minrate 400000 -maxrate 400000 -bufsize 500k -force_key_frames "expr:gte(t,n_forced*2)" -vsync 1 -r 5 -s 544*960 -acodec libfaac -async 1 ./test_20.ts
i use ffprobe command to see video info:
source file info:
Duration: 00:00:01.26, start: 28.346989, bitrate: 921 kb/s
Program 1
Metadata:
service_name : Service01
service_provider: FFmpeg
Stream #0:0[0x100]: Audio: aac ([15][0][0][0] / 0x000F), 44100 Hz, stereo, fltp, 23 kb/s
Stream #0:1[0x101]: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p, 544x960, 10.67 tbr, 90k tbn, 180k tbc
test file:
Input #0, mpegts, from 'test_20.ts':
Duration: 00:00:01.62, start: 1.576778, bitrate: 447 kb/s
Program 1
Metadata:
service_name : Service01
service_provider: FFmpeg
Stream #0:0[0x100]: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p, 544x960, 5 fps, 5 tbr, 90k tbn, 10 tbc
Stream #0:1[0x101]: Audio: aac ([15][0][0][0] / 0x000F), 44100 Hz, stereo, fltp, 5 kb/s
=======================================================================
question
so , we can see that the duration of src file is 1.26s , but after transcode, the test file is 1.62s.
why? can anybody help
I suggest you save the m3u8 to a single TS and then transcode that to MP4.
ffmpeg -i in.m3u8 -c copy src.ts
Your current command is transcoding each TS to CFR at half the rate but your source timestamps have some jitter, so due to PTS quantization, there will be a mismatch. A single file transcode will minimize it.
I have two videos (.mp4) files. One uploads to whatsapp and another does not.
Using ffmpeg I checked their properties:
a) Properties of video which uploads:
Duration: 00:00:56.45, start: 0.148000, bitrate: 1404 kb/s
Stream #0:0(und): Video: h264 (High) (avc1 / 0x31637661), yuv420p, 1080x1080, 1359 kb/s, 23.98 fps, 23.98 tbr, 90k tbn, 47.95 tbc (default)
Metadata:
handler_name : VideoHandler
Stream #0:1(eng): Audio: aac (HE-AACv2) (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 47 kb/s (default)
Metadata:
handler_name : SoundHandler
At least one output file must be specified
b) video which does not upload to whatsapp (because its says format not supported)
Duration: 00:00:56.10, start: 0.000000, bitrate: 543 kb/s
Stream #0:0: Video: h264 (High) (H264 / 0x34363248), yuv420p, 1080x1080 [SAR 1:1 DAR 1:1], 464 kb/s, 23.98 fps, 23.98 tbr, 23.98 tbn, 47.95 tbc
Stream #0:1: Audio: aac (LC) ([255][0][0][0] / 0x00FF), 48000 Hz, stereo, fltp, 56 kb/s
The difference in video I noticed:
(avc1 / 0x31637661) vs (H264 / 0x34363248)
1359 kb/s vs 464 kb/s
90k tbn vs 23.98 tbn
What can be the reason?
Also the second video is not being played in Android.
The link for the video is https://drive.google.com/open?id=0B4UM6vTHw4pyMExQQ1lxZGp0N2c
There are some options for a better compatibility:
ffmpeg -i broken.mp4 -c:v libx264 -profile:v baseline -level 3.0 -pix_fmt yuv420p working.mp4
With -profile:v baseline -level 3.0 you make the file more compatible with most older players, including WhatsApp ;). Although, this disables some advanced features.
-pix_fmt yuv420p is necessary to compile to baseline (YUV planar color space with 4:2:0 chroma subsampling).
Also, you can adjust other options as bitrate, framerate, audio, etc.
Source: H.264 docs
Copied from https://www.reddit.com/r/ffmpeg/comments/564kyc/ffmpeg_whatsapp_video_format_not_supported/?st=ivjxdi0v&sh=848ce7eb
ffmpeg -i brokenvideo.mp4 -c:v libx264 -c:a aac fixedvideo.mp4
Also had to apply this fix: FFMPEG (libx264) "height not divisible by 2"
This is worked for me in 2020
ffmpeg -i broken.mp4 -c:v libx264 -profile:v high -level 3.0 -pix_fmt yuv420p -brand mp42 fixed.mp4
I tried all previous commands and I got some errors. I was able to encode my video using this command and here is the explanation and why I set it up like this for a better compatibility:
ffmpeg -i input.mp4 \
-c:v libx264 -pix_fmt yuv420p \
-profile:v baseline -level 3.0 \
-vf "pad=ceil(iw/2)*2:ceil(ih/2)*2" -vb 1024k \
-acodec aac -ar 44100 -ac 2\
-minrate 1024k -maxrate 1024k -bufsize 1024k \
-movflags +faststart \
output.mp4
If your input contains AAC audio you can stream copy instead of re-encoding by changing -acodec aac -ar 44100 -ac 2 to -acodec copy to preserve the audio quality.
option
explanation
-vcodec libx264
Chooses video encoder libx264
-pix_fmt yuv420p
Ensures YUV 4:2:0 chroma subsampling for compatibility
-profile:v baseline
Set the encoding profile to baseline. Used primarily for low-cost applications that require additional data loss robustness
-level 3.0
Set the operating point level to 3.0 which is necessary to have compatibility with WhatsApp
-vf "pad=ceil(iw/2)*2:ceil(ih/2)*2"
If you get not divisible by 2 error see
-acodec aac
Chooses audio encoder aac
-minrate 1024k
set min bitrate tolerance to 1024k (in bits/s). It is of little use elsewise
-maxrate 1024k
set max bitrate tolerance to 1024k (in bits/s). Requires bufsize to be set
-bufsize 1024k
set rate-control buffer size to 1024k (in bits)
-movflags +faststart
enables fast start for streaming
Note about faststart
Normally, a MP4 file has all its metadata packets stored at the end of the file, in data units named atoms. The mdat atom is located before the moov atom. If the file is created by adding the -movflags faststart, the moov atom is moved at the beginning of the MP4 file. By using this option, the moov atom is located before the mdat atom. This allows video playback to begin before the file has been completely downloaded.
2023-01-22 I used the most upvoted answer format and it worked for video, but audio was not working on iPhones. Here's what worked for me:
ffmpeg -i broken.mp4 -c:v libx264 -profile:v baseline -level 3.0 -pix_fmt yuv420p -ac 2 working.mp4
I had to add -ac 2 because the audio format I had wasn't seen as stereo by iOS.