The duration of source video and subtracted wav audio is different , why?
I'm recorgnizing subtitle from audio, and I need to add subtitle back to video. So I want the duration of audio and video the same.
ffmpeg -i http://cdn.live.360.cn/huikan_news/vod-media/_XW_203286B417B7C6466B3B_20160627185953.m3u8 -vn test.wav -y
My CLI:
[zhangpengcheng#mobiledev03v ifly]$ ffprobe http://cdn.live.360.cn/huikan_news/vod-media/_XW_203286B417B7C6466B3B_20160627185953.m3u8 2>&1 | grep Duration
Duration: 00:06:51.99, start: 1.400000, bitrate: 0 kb/s
[zhangpengcheng#mobiledev03v ifly]$ ffmpeg -i http://cdn.live.360.cn/huikan_news/vod-media/_XW_203286B417B7C6466B3B_20160627185953.m3u8 -vn test.wav -y
ffmpeg version 3.1.3 Copyright (c) 2000-2016 the FFmpeg developers
built with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-17)
configuration: --prefix=./build/ --enable-shared --enable-static --enable-libx264 --enable-avisynth --enable-libass --enable-libfdk-aac --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libopencv --enable-librtmp --enable-gpl --enable-nonfree
libavutil 55. 28.100 / 55. 28.100
libavcodec 57. 48.101 / 57. 48.101
libavformat 57. 41.100 / 57. 41.100
libavdevice 57. 0.101 / 57. 0.101
libavfilter 6. 47.100 / 6. 47.100
libswscale 4. 1.100 / 4. 1.100
libswresample 2. 1.100 / 2. 1.100
libpostproc 54. 0.100 / 54. 0.100
Input #0, hls,applehttp, from 'http://cdn.live.360.cn/huikan_news/vod-media/_XW_203286B417B7C6466B3B_20160627185953.m3u8':
Duration: 00:06:51.99, start: 1.400000, bitrate: 0 kb/s
Program 0
Metadata:
variant_bitrate : 0
Stream #0:0: Video: h264 (Baseline) ([27][0][0][0] / 0x001B), yuv420p(tv, smpte170m/bt709/bt709), 668x376, 15 tbr, 90k tbn, 180k tbc
Stream #0:1: Audio: aac (LC) ([15][0][0][0] / 0x000F), 44100 Hz, mono, fltp, 65 kb/s
[wav # 0x10182e0] Using AVStream.codec to pass codec parameters to muxers is deprecated, use AVStream.codecpar instead.
Output #0, wav, to 'test.wav':
Metadata:
ISFT : Lavf57.41.100
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, mono, s16, 705 kb/s
Metadata:
encoder : Lavc57.48.101 pcm_s16le
Stream mapping:
Stream #0:1 -> #0:0 (aac (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
size= 31904kB time=00:06:51.84 bitrate= 634.6kbits/s speed= 146x
video:0kB audio:31904kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.000239%
[zhangpengcheng#mobiledev03v ifly]$ ffprobe test.wav 2>&1 | grep Duration
Duration: 00:06:10.40, bitrate: 705 kb/s
Looks like there are gaps in the original audio.
Use
ffmpeg -i http://cdn.live.360.cn/huikan_news/vod-media/_XW_203286B417B7C6466B3B_20160627185953.m3u8 -af aresample=async=1 -vn test.wav -y
The audio filter will fill in the gaps.
Related
"volume" filter does not work in complex filter. The volume is not affected in the converted audio.
env: Mac OS 10.12.6
ffmpeg -i /path/bg.mp3 -y -filter_complex [0:0]volume=0[output] -map
[output] -acodec libmp3lame -write_xing 0 /path/mixed.mp3
full log =====================================================================
liqideMacBook-Pro:pinyin-api work$ ffmpeg -i
/Users/work/dev/codebase/pinyin/pinyin-api/test/resources/bg_audio2.mp3
-y -filter_complex [0:0]volume=0.1[output] -map [output] -acodec
libmp3lame -f mp3 -write_xing 0 /Users/work/dev/codebase/pinyin/pinyin-
api/test/resources/mixed3.mp3
ffmpeg version 3.4.1 Copyright (c) 2000-2017 the FFmpeg developers
built with Apple LLVM version 9.0.0 (clang-900.0.39.2)
configuration: --prefix=/usr/local/Cellar/ffmpeg/3.4.1 --enable-
shared --enable-pthreads --enable-version3 --enable-hardcoded-tables --
enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-
gpl --enable-libmp3lame --enable-libx264 --enable-libxvid --enable-
opencl --enable-videotoolbox --disable-lzma
libavutil 55. 78.100 / 55. 78.100
libavcodec 57.107.100 / 57.107.100
libavformat 57. 83.100 / 57. 83.100
libavdevice 57. 10.100 / 57. 10.100
libavfilter 6.107.100 / 6.107.100
libavresample 3. 7. 0 / 3. 7. 0
libswscale 4. 8.100 / 4. 8.100
libswresample 2. 9.100 / 2. 9.100
libpostproc 54. 7.100 / 54. 7.100
Input #0, mp3, from '/Users/work/dev/codebase/pinyin/pinyin-
api/test/resources/bg_audio2.mp3':
Metadata:
title : 喜洋洋
album : 贺岁新年音乐会
encoder : Lavf56.4.101
Duration: 00:02:43.58, start: 0.025057, bitrate: 128 kb/s
Stream #0:0: Audio: mp3, 44100 Hz, stereo, s16p, 128 kb/s
Stream mapping:
Stream #0:0 (mp3) -> volume
volume -> Stream #0:0 (libmp3lame)
Press [q] to stop, [?] for help
Output #0, mp3, to '/Users/work/dev/codebase/pinyin/pinyin-
api/test/resources/mixed3.mp3':
Metadata:
TIT2 : 喜洋洋
TALB : 贺岁新年音乐会
TSSE : Lavf57.83.100
Stream #0:0: Audio: mp3 (libmp3lame), 44100 Hz, stereo, fltp
Metadata:
encoder : Lavc57.107.100 libmp3lame
[libmp3lame # 0x7fad80001a00] Trying to remove 1152 samples, but the
queue is empty
size= 2556kB time=00:02:43.57 bitrate= 128.0kbits/s speed=51.7x
video:0kB audio:2556kB subtitle:0kB other streams:0kB global
headers:0kB muxing overhead: 0.003782%
liqideMacBook-Pro:pinyin-api work$
=====================================================================
I am developing an application.
People upload videos from their mobile, from other places.
Using a CMS in PHP (it is the language with which the application is developed) I need to generate a unique video with these partial uploads.
Through FFmpeg I am doing tests, from the command line:
ffmpeg -i concat:IMG_1916.mp4\|IMG_1917.mp4 -c copy videoLoop.mp4
This code when I run it says:
ffmpeg version 3.2.4 Copyright (c) 2000-2017 the FFmpeg developers
built with Apple LLVM version 8.0.0 (clang-800.0.42.1)
configuration: --prefix=/usr/local/Cellar/ffmpeg/3.2.4 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-libmp3lame --enable-libx264 --enable-libxvid --enable-opencl --disable-lzma --enable-vda
libavutil 55. 34.101 / 55. 34.101
libavcodec 57. 64.101 / 57. 64.101
libavformat 57. 56.101 / 57. 56.101
libavdevice 57. 1.100 / 57. 1.100
libavfilter 6. 65.100 / 6. 65.100
libavresample 3. 1. 0 / 3. 1. 0
libswscale 4. 2.100 / 4. 2.100
libswresample 2. 3.100 / 2. 3.100
libpostproc 54. 1.100 / 54. 1.100
[mov,mp4,m4a,3gp,3g2,mj2 # 0x7f8515000000] Found duplicated MOOV Atom. Skipped it Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'concat:IMG_1916.mp4|IMG_1917.mp4':
Metadata:
encoder : Lavf57.66.102
major_brand : isom
minor_version : 512
compatible_brands: isomiso2avc1mp41
Duration: 00:00:04.27, start: 0.000000, bitrate: 26792 kb/s
Stream #0:0(und): Video: h264 (Constrained Baseline) (avc1 / 0x31637661), yuv420p, 1920x1080, 11978 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc (default)
Metadata:
handler_name : VideoHandler
Stream #0:1(und): Audio: aac (LC) (mp4a / 0x6134706D), 44100 Hz, mono, fltp, 120 kb/s (default)
Metadata:
handler_name : SoundHandler
Output #0, mp4, to 'videoLoop.mp4':
Metadata:
compatible_brands: isomiso2avc1mp41
major_brand : isom
minor_version : 512
encoder : Lavf57.56.101
Stream #0:0(und): Video: h264 (Constrained Baseline) ([33][0][0][0] / 0x0021), yuv420p, 1920x1080, q=2-31, 11978 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 30k tbc (default)
Metadata:
handler_name : VideoHandler
Stream #0:1(und): Audio: aac (LC) ([64][0][0][0] / 0x0040), 44100 Hz, mono, 120 kb/s (default)
Metadata:
handler_name : SoundHandler
Stream mapping:
Stream #0:0 -> #0:0 (copy)
Stream #0:1 -> #0:1 (copy)
Press [q] to stop, [?] for help
frame= 127 fps=0.0 q=-1.0 Lsize= 6264kB time=00:00:04.22 bitrate=12142.8kbits/s speed= 376x
video:6196kB audio:63kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.076698%
This execution generates a video, but not concatenated with the 2 specified, only with the first one.
Why not join the 2?
The videos to upload, will be of very different formats so I can not define codec.
You will have to make all inputs similar before concatenation, then use the concat filter. A rough example (you will of course have to customize it to your needs):
ffmpeg -i input0 -i input1 -filter_complex \
"[0:v]fps=25,scale=1280:720,format=yuv420p,setsar=1,setpts=PTS-STARTPTS[v0]; \
[1:v]fps=25,scale=1280:720,format=yuv420p,setsar=1,setpts=PTS-STARTPTS[v1]; \
[0:a]aformat=channel_layouts=stereo:sample_rates=44100,asetpts=PTS-STARTPTS[a0]; \
[1:a]aformat=channel_layouts=stereo:sample_rates=44100,asetpts=PTS-STARTPTS[a1]; \
[v0][a0][v1][a1]concat=n=2:v=1:a=1[v][a]" \
-map "[v]" -map "[a]" -c:v libx264 -c:a aac -movflags +faststart output.mp4
Using this adaptation of code, i can generate a video with two sources.
ffmpeg -i IMG_1916.mp4 -i IMG_1917.mp4 \
-filter_complex \
"[0:v:0] [0:a:0] \
[1:v:0] [1:a:0] \
concat=n=2:v=1:a=1 [v] [a]" \
-map "[v]" -map "[a]" videoLoop.mp4
I'm not sure if I can concatenate any video format, from any device of any source / format with this code.
While I am trying to convert wav file to wma file size is becomes much larger than original source file:
input.wav - 11M
output.wma - 16M
Is there any way to reduce file size of output file?
Command and output:
$ ffmpeg -i input.wav -c copy output.wma
ffmpeg version N-80797-g8b4d6cc Copyright (c) 2000-2016 the FFmpeg developers
built with gcc 6.1.1 (GCC) 20160501
configuration: --enable-gpl
libavutil 55. 27.100 / 55. 27.100
libavcodec 57. 48.101 / 57. 48.101
libavformat 57. 40.101 / 57. 40.101
libavdevice 57. 0.102 / 57. 0.102
libavfilter 6. 46.102 / 6. 46.102
libswscale 4. 1.100 / 4. 1.100
libswresample 2. 1.100 / 2. 1.100
libpostproc 54. 0.100 / 54. 0.100
Guessed Channel Layout for Input Stream #0.0 : stereo
Input #0, wav, from 'input.wav':
Metadata:
encoder : Lavf57.40.101
Duration: 00:01:00.00, bitrate: 1411 kb/s
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, 2 channels, s16, 1411 kb/s
Output #0, asf, to 'output.wma':
Metadata:
WM/EncodingSettings: Lavf57.40.101
Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 44100 Hz, stereo, 1411 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (copy)
Press [q] to stop, [?] for help
size= 16150kB time=00:01:00.00 bitrate=2205.1kbits/s speed=1.9e+03x
video:0kB audio:10336kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 56.255745%
WMA is crappy. Try changing the packet size. From FFmpeg ASF muxer documentation:
-packet_size
Set the muxer packet size. By tuning this setting you
may reduce data fragmentation or muxer overhead depending on your
source. Default value is 3200, minimum is 100, maximum is 64k.
Example:
ffmpeg -i input.wav -c copy -packet_size 65536 output_64k.wma
File sizes:
input.wav - 11M
output_64k.wma - 11M
output_default.wma - 16M
I have a .ts file (Download files here: http://dropcanvas.com/2gmsg/1).
I want to split this video while I expect ALL other properties remain same including pts time.
Here is what I try to achieve this:
ffmpeg -ss 0.000 -i sample.ts -y -c copy -t 3 splitted.ts
Expected start time: 94678.950389
New start time: 1.402367
I expect the above command should only take first 3 seconds of the .ts file and all other stuff to stay same. I've seen copyts and copytb options from the documentation but I wasn't able to use them.
So how do I do this?
Thank you
Here are the logs for copyts. It creates a 0 byte splitted.ts file:
ffmpeg -ss 0:00:00 -i sample.ts -to 00:00:03 -y -c copy -copyts splitted.ts
ffmpeg version 3.0 Copyright (c) 2000-2016 the FFmpeg developers
built with Apple LLVM version 7.0.0 (clang-700.0.72)
configuration: --prefix=/usr/local/Cellar/ffmpeg/3.0 --enable-shared --enable-pthreads --enable-gpl --enable-version3 --enable-hardcoded-tables --enable-avresample --cc=clang --host-cflags= --host-ldflags= --enable-opencl --enable-libx264 --enable-libmp3lame --enable-libxvid --enable-libfreetype --enable-libvorbis --enable-libvpx --enable-libass --enable-ffplay --enable-libfdk-aac --enable-libopus --enable-libx265 --enable-nonfree --enable-vda
libavutil 55. 17.103 / 55. 17.103
libavcodec 57. 24.102 / 57. 24.102
libavformat 57. 25.100 / 57. 25.100
libavdevice 57. 0.101 / 57. 0.101
libavfilter 6. 31.100 / 6. 31.100
libavresample 3. 0. 0 / 3. 0. 0
libswscale 4. 0.100 / 4. 0.100
libswresample 2. 0.101 / 2. 0.101
libpostproc 54. 0.100 / 54. 0.100
[NULL # 0x7fafac02fc00] start time for stream 2 is not set in estimate_timings_from_pts
Input #0, mpegts, from 'sample.ts':
Duration: 00:00:10.07, start: 94678.950389, bitrate: 934 kb/s
Program 1
Stream #0:0[0x100]: Video: h264 (Main) ([27][0][0][0] / 0x001B), yuv420p, 854x480 [SAR 1280:1281 DAR 16:9], 29.97 fps, 29.97 tbr, 90k tbn, 59.94 tbc
Stream #0:1[0x101]: Audio: aac (LC) ([15][0][0][0] / 0x000F), 44100 Hz, stereo, fltp, 140 kb/s
Stream #0:2[0x102]: Data: timed_id3 (ID3 / 0x20334449)
Output #0, mpegts, to 'splitted.ts':
Metadata:
encoder : Lavf57.25.100
Stream #0:0: Video: h264 ([27][0][0][0] / 0x001B), yuv420p, 854x480 [SAR 1280:1281 DAR 16:9], q=2-31, 29.97 fps, 29.97 tbr, 90k tbn, 90k tbc
Stream #0:1: Audio: aac (LC) ([15][0][0][0] / 0x000F), 44100 Hz, stereo, 140 kb/s
Stream mapping:
Stream #0:0 -> #0:0 (copy)
Stream #0:1 -> #0:1 (copy)
Press [q] to stop, [?] for help
frame= 0 fps=0.0 q=-1.0 Lsize= 0kB time=00:00:00.00 bitrate=N/A speed= 0x
video:0kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
Output file is empty, nothing was encoded (check -ss / -t / -frames parameters if used)
This works for me
ffmpeg -seek_timestamp 1 -ss 94678.950389 -i sample.ts -y -c copy -copyts -to 94681.950389 -muxdelay 0 splitted.ts
Your original command can work if you use the frames flag.
ffmpeg -ss 0.000 -i sample.ts -y -c copy -copyts -muxdelay 0 -vframes 90 splitted.ts
Where 90 represents amount of frames in t seconds.
I have two video streams from which I'd like to take one frame of each, both of them are RTSP. I'm using the same FFMPEG instruction for both of them but changing the URL of the stream, the first one works but the second one throws the error:
method SETUP failed: 455 Method Not Valid In This State
Can anyone tell me what could be the reason for this error and how to solve it?
WORKING
ffmpeg -ss 1 -i rtsp://streamreader:trudat55#69.84.126.216:88/videoMain -an -vcodec mjpeg -vframes 1 -aspect 16:9 -q:v 2 -y test.jpg
ffmpeg version 2.8.2 Copyright (c) 2000-2015 the FFmpeg developers
built with gcc 4.9.2 (Debian 4.9.2-10)
configuration: --enable-gpl --enable-avresample --enable-libopencore-amrnb --enable-libx264 --enable-libxvid --enable-postproc --enable-version3 --enable-shared --enable-pic --extra-ldexeflags=-pie
libavutil 54. 31.100 / 54. 31.100
libavcodec 56. 60.100 / 56. 60.100
libavformat 56. 40.101 / 56. 40.101
libavdevice 56. 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
libpostproc 53. 3.100 / 53. 3.100
[rtsp # 0x55a60a0a7420] UDP timeout, retrying with TCP
Guessed Channel Layout for Input Stream #0.1 : mono
Input #0, rtsp, from 'rtsp://streamreader:password#69.84.126.216:88/videoMain':
Metadata:
title : IP Camera Video
comment : videoMain
Duration: N/A, start: 0.200044, bitrate: N/A
Stream #0:0: Video: h264 (Baseline), yuv420p, 640x480, 50 tbr, 90k tbn, 180k tbc
Stream #0:1: Audio: pcm_mulaw, 8000 Hz, 1 channels, s16, 64 kb/s
[swscaler # 0x55a60a102460] deprecated pixel format used, make sure you did set range correctly
Output #0, image2, to 'test.jpg':
Metadata:
title : IP Camera Video
comment : videoMain
encoder : Lavf56.40.101
Stream #0:0: Video: mjpeg, yuvj420p(pc), 640x480 [SAR 4:3 DAR 16:9], q=2-31, 200 kb/s, 50 fps, 50 tbn, 50 tbc
Metadata:
encoder : Lavc56.60.100 mjpeg
Stream mapping:
Stream #0:0 -> #0:0 (h264 (native) -> mjpeg (native))
Press [q] to stop, [?] for help
frame= 1 fps=0.5 q=2.0 Lsize=N/A time=00:00:00.02 bitrate=N/A dup=1 drop=1
video:66kB audio:0kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: unknown
NOT WORKING
ffmpeg -ss 1 -i rtsp://camaras.corredorautomotriz.cl:554/live.sdp -an -vcodec mjpeg -vframes 1 -aspect 16:9 -q:v 2 -y test.jpg
ffmpeg version 2.8.2 Copyright (c) 2000-2015 the FFmpeg developers
built with gcc 4.9.2 (Debian 4.9.2-10)
configuration: --enable-gpl --enable-avresample --enable-libopencore-amrnb --enable-libx264 --enable-libxvid --enable-postproc --enable-version3 --enable-shared --enable-pic --extra-ldexeflags=-pie
libavutil 54. 31.100 / 54. 31.100
libavcodec 56. 60.100 / 56. 60.100
libavformat 56. 40.101 / 56. 40.101
libavdevice 56. 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
libpostproc 53. 3.100 / 53. 3.100
[rtsp # 0x55b98f2de420] UDP timeout, retrying with TCP
[rtsp # 0x55b98f2de420] method SETUP failed: 455 Method Not Valid In This State
[rtsp # 0x55b98f2de420] Could not find codec parameters for stream 0 (Video: mjpeg, none(bt470bg/unknown/unknown)): unspecified size
Consider increasing the value for the 'analyzeduration' and 'probesize' options
Input #0, rtsp, from 'rtsp://camaras.corredorautomotriz.cl:554/live.sdp':
Metadata:
title : RTSP server
Duration: N/A, start: 0.000000, bitrate: N/A
Stream #0:0: Video: mjpeg, none(bt470bg/unknown/unknown), 90k tbr, 90k tbn, 90k tbc
Stream #0:1: Data: none
Output #0, image2, to 'test.jpg':
Output file #0 does not contain any stream
I found that if I specify that the protocol is TCP with the instruction -rtsp_transport tcp then i don't get an error:
ffmpeg -ss 5 -rtsp_transport tcp -i rtsp://camaras.corredorautomotriz.cl:554/live.sdp -s 640x480 -aspect 16:9 -b:v 800k -r 24 video.flv
But I would still like to know the reason and solution of the problem I got before, because I am not sure I can use this parameter. Anyone knows?
Thanks
As I see in your output, ffmpeg failed to determine input stream type in the second case (rtsp://camaras.corredorautomotriz.cl:554/live.sdp).
I checked it, and it got me mpeg4:
Input #0, rtsp, from 'rtsp://camaras.corredorautomotriz.cl:554/live.sdp':
Metadata:
title : RTSP server
Duration: N/A, start: 0.000000, bitrate: N/A
Stream #0:0: Video: mpeg4 (Simple Profile), yuv420p, 720x480 [SAR 1:1 DAR 3:2], 30 tbr, 30k tbn
Thus possibly you compiled ffmpeg without mpeg4 decoder (or with broken decoder).