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Sur d’autres sites (3544)
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Re-creating/matching ffmpeg settings
8 novembre 2013, par Pete HelgrenI have an MP4 file that plays pretty much on every browser using either HTML5 or flash fallback (using mediaelement.js). I also have a series of additional video files that I would like to have "match" the one file that does play consistently but I haven't been able to match the ffmpeg settings so it produces a like file. I used ffprobe and got the following results on the file that plays successfully :
ffprobe version N-53616-g7a2edcf Copyright (c) 2007-2013 the FFmpeg developers
built on May 29 2013 01:21:08 with gcc 4.7.3 (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-libvo-aacenc --enable-libvo-
amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --
enable-libxvid --enable-zlib
libavutil 52. 34.100 / 52. 34.100
libavcodec 55. 12.102 / 55. 12.102
libavformat 55. 7.100 / 55. 7.100
libavdevice 55. 1.101 / 55. 1.101
libavfilter 3. 72.100 / 3. 72.100
libswscale 2. 3.100 / 2. 3.100
libswresample 0. 17.102 / 0. 17.102
libpostproc 52. 3.100 / 52. 3.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'MyFile.mp4':
Metadata:
major_brand : M4V
minor_version : 1
compatible_brands: M4V mp42isom
creation_time : 2013-05-29 16:08:29
Duration: 00:01:58.10, start: 0.000000, bitrate: 1155 kb/s
Stream #0:0(eng): Video: h264 (Main) (avc1 / 0x31637661), yuv420p, 854x480 [SAR 1:1 DAR 427:240], 995 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc
Metadata:
creation_time : 2013-05-29 16:08:29
handler_name : Mainconcept MP4 Video Media Handler
Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 157 kb/s
Metadata:
creation_time : 2013-05-29 16:08:29
handler_name : Mainconcept MP4 Sound Media HandlerFollowing a few guides I found online, I am using the following ffmpeg settings :
ffmpeg -i {input file name} -y -acodec aac -ac 2 -strict experimental -ab 64k -s 320x240 -vcodec libx264 -pix_fmt yuv420p -preset slow -profile:v baseline -level 30 -maxrate 300k -bufsize 600k -b:v 300k -f mp4 -threads 0 -movflags +faststart {output file name}That command produces the following output :
ffmpeg -i "testin.mp4" -y -acodec aac -ac 2 -strict experimental -ab 64k -s 320x240 -vcodec libx264 -pix_fmt yuv420p -preset slow-profile:v baseline -level 30 -maxrate 300k -bufsize 600k -b:v 300k -f mp4 -threads 0 -movflags +faststart testout.mp4
ffmpeg version N-53616-g7a2edcf Copyright (c) 2000-2013 the FFmpeg developers built on May 29 2013 01:21:08 with gcc 4.7.3 (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-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-libvo-aacenc --enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-libx264 --enable-libxavs --enable-libxvid --enable-zlib
libavutil 52. 34.100 / 52. 34.100
libavcodec 55. 12.102 / 55. 12.102
libavformat 55. 7.100 / 55. 7.100
libavdevice 55. 1.101 / 55. 1.101
libavfilter 3. 72.100 / 3. 72.100
libswscale 2. 3.100 / 2. 3.100
libswresample 0. 17.102 / 0. 17.102
libpostproc 52. 3.100 / 52. 3.100
Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'testin.mp4':
Metadata:
major_brand : mp42
minor_version : 0
compatible_brands: mp42mp41
creation_time : 2013-10-01 17:11:42
Duration: 00:04:33.75, start: 0.000000, bitrate: 725 kb/s
Stream #0:0(eng): Video: h264 (Baseline) (avc1 / 0x31637661), yuv420p, 320x240 [SAR 1:1 DAR 4:3], 596 kb/s, 29.97 fps, 29.97 tbr, 30k tbn, 59.94 tbc
Metadata:
creation_time : 2013-10-01 17:11:42
handler_name : ?Mainconcept Video Media Handler
Stream #0:1(eng): Audio: aac (mp4a / 0x6134706D), 48000 Hz, stereo, fltp, 125 kb/s
Metadata:
creation_time : 2013-10-01 17:11:42
handler_name : #Mainconcept MP4 Sound Media Handler
[libx264 @ 0000000000358160] using SAR=1/1
[libx264 @ 0000000000358160] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2AVX
[libx264 @ 0000000000358160] profile Constrained Baseline, level 3.0
[libx264 @ 0000000000358160] 264 - core 133 r2334 a3ac64b - H.264/MPEG-4 AVC codec - Copyleft 2003-2013 - http://www.videolan.org/x264.html - options: cabac=0 ref=5 deblock=1:0:0 analyse=0x1:0x111 me=umh subme=8 psy=1 psy_rd=1.00:0.00 mixed_ref=1 me_range=16 chroma_me=1 trellis=1 8x8dct=0 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=-2 threads=6 lookahead_threads=1 sliced_threads=0 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=0 keyint=250 keyint_min=25 scenecut=40 intra_refresh=0 rc_lookahead=50 rc=cbr mbtree=1 bitrate=300 ratetol=1.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 vbv_maxrate=300 vbv_bufsize=600 nal_hrd=none ip_ratio=1.40 aq=1:1.00
Output #0, mp4, to 'testout.mp4':
Metadata:
major_brand : mp42
minor_version : 0
compatible_brands: mp42mp41
encoder : Lavf55.7.100
Stream #0:0(eng): Video: h264 (libx264) ([33][0][0][0] / 0x0021), yuv420p, 320x240 [SAR 1:1 DAR 4:3], q=-1--1, 300 kb/s, 30k tbn, 29.97 tbc
Metadata:
creation_time : 2013-10-01 17:11:42
handler_name : ?Mainconcept Video Media Handler
Stream #0:1(eng): Audio: aac ([64][0][0][0] / 0x0040), 48000 Hz, stereo, fltp, 64 kb/s
Metadata:
creation_time : 2013-10-01 17:11:42
handler_name : #Mainconcept MP4 Sound Media Handler
Stream mapping:
Stream #0:0 -> #0:0 (h264 -> libx264)
Stream #0:1 -> #0:1 (aac -> aac)
Press [q] to stop, [?] for help
frame= 283 fps=0.0 q=21.0 size= 329kB time=00:00:09.30 bitrate= 289.9kbits/
frame= 565 fps=564 q=20.0 size= 760kB time=00:00:18.64 bitrate= 333.7kbits/
frame= 846 fps=564 q=21.0 size= 1179kB time=00:00:28.20 bitrate= 342.5kbits/
frame= 1123 fps=561 q=22.0 size= 1603kB time=00:00:37.33 bitrate= 351.7kbits/
frame= 1376 fps=550 q=25.0 size= 1986kB time=00:00:45.69 bitrate= 356.1kbits/
frame= 1651 fps=550 q=22.0 size= 2377kB time=00:00:55.01 bitrate= 353.9kbits/
frame= 1914 fps=546 q=24.0 size= 2776kB time=00:01:03.70 bitrate= 357.0kbits/
frame= 2181 fps=545 q=24.0 size= 3163kB time=00:01:12.70 bitrate= 356.4kbits/
frame= 2443 fps=543 q=24.0 size= 3566kB time=00:01:21.38 bitrate= 358.9kbits/
frame= 2702 fps=540 q=24.0 size= 3939kB time=00:01:30.04 bitrate= 358.4kbits/
frame= 2965 fps=539 q=24.0 size= 4338kB time=00:01:38.73 bitrate= 359.9kbits/
frame= 3236 fps=539 q=23.0 size= 4731kB time=00:01:48.03 bitrate= 358.7kbits/
frame= 3505 fps=539 q=24.0 size= 5130kB time=00:01:56.75 bitrate= 360.0kbits/
frame= 3765 fps=537 q=26.0 size= 5521kB time=00:02:05.41 bitrate= 360.6kbits/
frame= 4036 fps=538 q=24.0 size= 5905kB time=00:02:14.59 bitrate= 359.4kbits/
frame= 4326 fps=540 q=24.0 size= 6343kB time=00:02:24.10 bitrate= 360.6kbits/
frame= 4606 fps=542 q=27.0 size= 6783kB time=00:02:33.49 bitrate= 362.0kbits/
frame= 4886 fps=543 q=24.0 size= 7168kB time=00:02:42.81 bitrate= 360.7kbits/
frame= 5166 fps=544 q=27.0 size= 7616kB time=00:02:52.13 bitrate= 362.5kbits/
frame= 5456 fps=545 q=24.0 size= 8020kB time=00:03:01.84 bitrate= 361.3kbits/
frame= 5736 fps=546 q=27.0 size= 8458kB time=00:03:11.18 bitrate= 362.4kbits/
frame= 6001 fps=545 q=30.0 size= 8849kB time=00:03:20.17 bitrate= 362.1kbits/
frame= 6264 fps=544 q=27.0 size= 9233kB time=00:03:28.85 bitrate= 362.2kbits/
frame= 6526 fps=544 q=26.0 size= 9611kB time=00:03:37.55 bitrate= 361.9kbits/
frame= 6793 fps=543 q=26.0 size= 10006kB time=00:03:46.51 bitrate= 361.9kbits/
frame= 7076 fps=544 q=24.0 size= 10410kB time=00:03:55.86 bitrate= 361.6kbits/
frame= 7355 fps=544 q=25.0 size= 10835kB time=00:04:05.20 bitrate= 362.0kbits/
frame= 7615 fps=543 q=27.0 size= 11236kB time=00:04:13.88 bitrate= 362.5kbits/
frame= 7883 fps=543 q=25.0 size= 11625kB time=00:04:22.89 bitrate= 362.3kbits/
frame= 8166 fps=544 q=21.0 size= 12040kB time=00:04:32.31 bitrate= 362.2kbits/
Starting second pass: moving header on top of the file
frame= 8203 fps=540 q=-1.0 Lsize= 12417kB time=00:04:33.74 bitrate= 371.6kbits/s
video:10035kB audio:2150kB subtitle:0 global headers:0kB muxing overhead 1.901929%
[libx264 @ 0000000000358160] frame I:41 Avg QP:14.13 size: 18349
[libx264 @ 0000000000358160] frame P:8162 Avg QP:17.24 size: 1167
[libx264 @ 0000000000358160] mb I I16..4: 24.8% 0.0% 75.2%
[libx264 @ 0000000000358160] mb P I16..4: 0.4% 0.0% 0.3% P16..4: 29.1% 6.4% 3.2% 0.0% 0.0% skip:60.7%
[libx264 @ 0000000000358160] coded y,uvDC,uvAC intra: 66.1% 74.1% 53.1% inter: 15.3% 20.2% 4.0%
[libx264 @ 0000000000358160] i16 v,h,dc,p: 23% 21% 11% 45%
[libx264 @ 0000000000358160] i4 v,h,dc,ddl,ddr,vr,hd,vl,hu: 21% 25% 14% 6% 7% 7% 7% 6% 7%
[libx264 @ 0000000000358160] i8c dc,h,v,p: 43% 28% 19% 10%
[libx264 @ 0000000000358160] ref P L0: 78.4% 12.6% 4.9% 2.4% 1.7%
[libx264 @ 0000000000358160] kb/s:300.33The one problem with the incantation above is that on iPad's and iPhones and some Android tablets all you hear is the audio with no video. But the file listed above (contents of the ffprobe) does not have that problem, it WILL play on an iPad, iPhone, Android tablet. I researched causes for "audio, no video" and the recommendations were all over the deck. But, the file listed in the ffprobe DOES play, so if I can match those settings described by the ffprobe for all of the other videos, they should play consistently as well. My challenge is to match the ffprobe settings using ffmpeg and there doesn't seem to be a 1 to 1 match between what I see in ffprobe and the corresponding settings in ffmpeg (a nice utility would be to have ffprobe output the ffmpeg settings that could be used to create a file of similar format).
I walked through the ffprobe documentation but couldn't find anything that would produce a format that could be used as input to ffmpeg. Suggestions ?
-
ffmpeg cannot grab the audio played through the speaker under windows 10
6 mars 2024, par shangpingI am using ffmpeg to grab screen. It used to work well under win7 using UScreenCapture and virtual-audio-capturer (all x86 version). On win10, the audio is not recorded. (I cannot hear any sound while playing back). I am using ffmpeg/uscreencapture/virtual-audio-capturer all win32 version since virtual-audio-capturer has no x64 release.



So my question is : How can I record the video and the audio correctly using ffmpeg ?



Here is the screen dump out :
first the direct show devices on my system



C:\ffmpeg-20190403-ee16d14-win32-static\bin>ffmpeg -list_devices true -f dshow -i dummy
ffmpeg version N-93528-gee16d14b0a Copyright (c) 2000-2019 the FFmpeg developers
 built with gcc 8.2.1 (GCC) 20190212
 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-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt
 libavutil 56. 26.100 / 56. 26.100
 libavcodec 58. 48.101 / 58. 48.101
 libavformat 58. 26.101 / 58. 26.101
 libavdevice 58. 7.100 / 58. 7.100
 libavfilter 7. 48.100 / 7. 48.100
 libswscale 5. 4.100 / 5. 4.100
 libswresample 3. 4.100 / 3. 4.100
 libpostproc 55. 4.100 / 55. 4.100
[dshow @ 06303ac0] DirectShow video devices (some may be both video and audio devices)
[dshow @ 06303ac0] "UScreenCapture"
[dshow @ 06303ac0] Alternative name "@device_sw_{860BB310-5D01-11D0-BD3B-00A0C911CE86}\UScreenCapture"
[dshow @ 06303ac0] DirectShow audio devices
[dshow @ 06303ac0] "Microphone Array (Realtek(R) Audio)"
[dshow @ 06303ac0] Alternative name "@device_cm_{33D9A762-90C8-11D0-BD43-00A0C911CE86}\wave_{9F102091-1843-44C8-B20E-C6F2E0F11A0F}"
[dshow @ 06303ac0] "virtual-audio-capturer"
[dshow @ 06303ac0] Alternative name "@device_sw_{33D9A762-90C8-11D0-BD43-00A0C911CE86}\{8E14549B-DB61-4309-AFA1-3578E927E935}"
dummy: Immediate exit requested



2nd, when I use virtual audio capturer, ffmpeg just failed :



C:\ffmpeg-20190403-ee16d14-win32-static\bin>ffmpeg -f dshow -i video="UScreenCapture":audio="Microphone Array (Realtek(R) Audio)" -vf crop=1075:717:189:81 -vcodec mpeg4 -qscale 0 -framerate 60 -acodec libmp3lame -ab 128000 -async 1 output.aviffmpeg -f dshow -i video="UScreenCapture":audio="virtual-audio-capturer" -vf crop=1075:717:189:81 -vcodec mpeg4 -qscale 0 -framerate 60 -acodec libmp3lame -ab 128000 -async 1 output.avi
ffmpeg version N-93528-gee16d14b0a Copyright (c) 2000-2019 the FFmpeg developers
 built with gcc 8.2.1 (GCC) 20190212
 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-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt
 libavutil 56. 26.100 / 56. 26.100
 libavcodec 58. 48.101 / 58. 48.101
 libavformat 58. 26.101 / 58. 26.101
 libavdevice 58. 7.100 / 58. 7.100
 libavfilter 7. 48.100 / 7. 48.100
 libswscale 5. 4.100 / 5. 4.100
 libswresample 3. 4.100 / 3. 4.100
 libpostproc 55. 4.100 / 55. 4.100
Guessed Channel Layout for Input Stream #0.1 : stereo
Input #0, dshow, from 'video=UScreenCapture:audio=Microphone Array (Realtek(R) Audio)':
 Duration: N/A, start: 6333.289000, bitrate: N/A
 Stream #0:0: Video: rawvideo, bgr24, 3200x1080, 15 fps, 15 tbr, 10000k tbn, 10000k tbc
 Stream #0:1: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
IMMDeviceEnumerator::GetDefaultAudioEndpoint failed: hr = 0x80070490
[dshow @ 062f4ac0] real-time buffer [UScreenCapture] [video input] too full or near too full (340% of size: 3041280 [rtbufsize parameter])! frame dropped!



When I use the other audio devices, it seems working but there is not audio recorded :



ffmpeg -f dshow -i video="UScreenCapture":audio="Microphone Array (Realtek(R) Audio)" -vf crop=1075:717:189:81 -vcodec mpeg4 -qscale 0 -framerate 60 -acodec libmp3lame -ab 128000 -async 1 output.avi

 C:\ffmpeg-20190403-ee16d14-win32-static\bin>ffmpeg -f dshow -i video="UScreenCapture":audio="Microphone Array (Realtek(R) Audio)" -vf crop=1075:717:189:81 -vcodec mpeg4 -qscale 0 -framerate 60 -acodec libmp3lame -ab 128000 -async 1 output.avi
 ffmpeg version N-93528-gee16d14b0a Copyright (c) 2000-2019 the FFmpeg developers
 built with gcc 8.2.1 (GCC) 20190212
 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-amf --enable-ffnvcodec --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt
 libavutil 56. 26.100 / 56. 26.100
 libavcodec 58. 48.101 / 58. 48.101
 libavformat 58. 26.101 / 58. 26.101
 libavdevice 58. 7.100 / 58. 7.100
 libavfilter 7. 48.100 / 7. 48.100
 libswscale 5. 4.100 / 5. 4.100
 libswresample 3. 4.100 / 3. 4.100
 libpostproc 55. 4.100 / 55. 4.100
 Guessed Channel Layout for Input Stream #0.1 : stereo
 Input #0, dshow, from 'video=UScreenCapture:audio=Microphone Array (Realtek(R) Audio)':
 Duration: N/A, start: 6168.067000, bitrate: N/A
 Stream #0:0: Video: rawvideo, bgr24, 3200x1080, 15 fps, 15 tbr, 10000k tbn, 10000k tbc
 Stream #0:1: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
 Please use -q:a or -q:v, -qscale is ambiguous
 Stream mapping:
 Stream #0:0 -> #0:0 (rawvideo (native) -> mpeg4 (native))
 Stream #0:1 -> #0:1 (pcm_s16le (native) -> mp3 (libmp3lame))
 Press [q] to stop, [?] for help
 -async is forwarded to lavfi similarly to -af aresample=async=1:min_hard_comp=0.100000:first_pts=0.
 Output #0, avi, to 'output.avi':
 Metadata:
 ISFT : Lavf58.26.101
 Stream #0:0: Video: mpeg4 (FMP4 / 0x34504D46), yuv420p(progressive), 1075x717, q=2-31, 200 kb/s, 15 fps, 15 tbn, 15 tbc
 Metadata:
 encoder : Lavc58.48.101 mpeg4
 Side data:
 cpb: bitrate max/min/avg: 0/0/200000 buffer size: 0 vbv_delay: -1
 Stream #0:1: Audio: mp3 (libmp3lame) (U[0][0][0] / 0x0055), 44100 Hz, stereo, s16p, 128 kb/s
 Metadata:
 encoder : Lavc58.48.101 libmp3lame
 frame= 8 fps=0.0 q=0.0 size= 522kB time=00:00:00.53 bitrate=8013.1kbitsframe= 16 fps= 16 q=0.0 size= 1034kB time=00:00:01.06 bitrate=7938.7kbitsframe= 23 fps= 15 q=0.0 size= 1546kB time=00:00:01.53 bitrate=8258.0kbitsframe= 31 fps= 15 q=0.0 size= 2058kB time=00:00:02.06 bitrate=8156.4kbitsframe= 38 fps= 15 q=0.0 size= 2058kB time=00:00:02.53 bitrate=6653.9kbitsframe= 46 fps= 15 q=0.0 size= 2570kB time=00:00:03.06 bitrate=6864.4kbitsframe= 54 fps= 15 q=0.0 size= 3082kB time=00:00:03.60 bitrate=7012.6kbitsframe= 61 fps= 15 q=0.0 size= 3338kB time=00:00:04.06 bitrate=6723.5kbitsframe= 69 fps= 15 q=0.0 size= 3850kB time=00:00:04.60 bitrate=6855.8kbitsframe= 76 fps= 15 q=0.0 size= 4106kB time=00:00:05.06 bitrate=6638.2kbitsframe= 84 fps= 15 q=0.0 size= 4618kB time=00:00:05.60 bitrate=6755.0kbitsframe= 91 fps= 15 q=0.0 size= 4874kB time=00:00:06.06 bitrate=6581.1kbitsframe= 99 fps= 15 q=0.0 size= 5130kB time=00:00:06.60 bitrate=6367.0kbitsframe= 106 fps= 15 q=0.0 size= 5642kB time=00:00:07.06 bitrate=6540.1kbitsframe= 114 fps= 15 q=0.0 size= 6154kB time=00:00:07.66 bitrate=6575.3kbitsframe= 121 fps= 15 q=0.0 size= 6410kB time=00:00:08.13 bitrate=6455.9kbitsframe= 129 fps= 15 q=0.0 size= 6666kB time=00:00:08.66 bitrate=6300.6kbitsframe= 136 fps= 15 q=0.0 size= 6922kB time=00:00:09.13 bitrate=6208.3kbitsframe= 144 fps= 15 q=0.0 size= 7434kB time=00:00:09.66 bitrate=6299.7kbitsframe= 151 fps= 15 q=0.0 size= 7690kB time=00:00:10.13 bitrate=6216.5kbitsframe= 159 fps= 15 q=0.0 size= 7946kB time=00:00:10.66 bitrate=6102.3kbitsframe= 167 fps= 15 q=0.0 size= 8458kB time=00:00:11.20 bitrate=6186.2kbitsframe= 174 fps= 15 q=0.0 size= 8970kB time=00:00:11.66 bitrate=6298.3kbitsframe= 182 fps= 15 q=0.0 size= 8970kB time=00:00:12.20 bitrate=6022.9kbitsframe= 189 fps= 15 q=0.0 size= 9482kB time=00:00:12.66 bitrate=6132.2kbitsframe= 190 fps= 15 q=0.0 Lsize= 9602kB time=00:00:12.73 bitrate=6177.3kbits/s speed= 1x
 video:9382kB audio:194kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 0.265721%
 [dshow @ 06393f40] real-time buffer [UScreenCapture] [video input] too full or near too full (340% of size: 3041280 [rtbufsize parameter])! frame dropped!
 Exiting normally, received signal 2.


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C++ ffmpeg - export to wav error : Invalid PCM packet, data has size 2 but at least a size of 4 was expected
9 septembre 2024, par Chris PC++ code :


AudioSegment AudioSegment::from_file(const std::string& file_path, const std::string& format, const std::string& codec,
 const std::map& parameters, int start_second, int duration) {

 avformat_network_init();
 av_log_set_level(AV_LOG_ERROR); // Adjust logging level as needed

 AVFormatContext* format_ctx = nullptr;
 if (avformat_open_input(&format_ctx, file_path.c_str(), nullptr, nullptr) != 0) {
 std::cerr << "Error: Could not open audio file." << std::endl;
 return AudioSegment(); // Return an empty AudioSegment on failure
 }

 if (avformat_find_stream_info(format_ctx, nullptr) < 0) {
 std::cerr << "Error: Could not find stream information." << std::endl;
 avformat_close_input(&format_ctx);
 return AudioSegment();
 }

 int audio_stream_index = -1;
 for (unsigned int i = 0; i < format_ctx->nb_streams; i++) {
 if (format_ctx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
 audio_stream_index = i;
 break;
 }
 }

 if (audio_stream_index == -1) {
 std::cerr << "Error: Could not find audio stream." << std::endl;
 avformat_close_input(&format_ctx);
 return AudioSegment();
 }

 AVCodecParameters* codec_par = format_ctx->streams[audio_stream_index]->codecpar;
 const AVCodec* my_codec = avcodec_find_decoder(codec_par->codec_id);
 AVCodecContext* codec_ctx = avcodec_alloc_context3(my_codec);

 if (!codec_ctx) {
 std::cerr << "Error: Could not allocate codec context." << std::endl;
 avformat_close_input(&format_ctx);
 return AudioSegment();
 }

 if (avcodec_parameters_to_context(codec_ctx, codec_par) < 0) {
 std::cerr << "Error: Could not initialize codec context." << std::endl;
 avcodec_free_context(&codec_ctx);
 avformat_close_input(&format_ctx);
 return AudioSegment();
 }

 if (avcodec_open2(codec_ctx, my_codec, nullptr) < 0) {
 std::cerr << "Error: Could not open codec." << std::endl;
 avcodec_free_context(&codec_ctx);
 avformat_close_input(&format_ctx);
 return AudioSegment();
 }

 SwrContext* swr_ctx = swr_alloc();
 if (!swr_ctx) {
 std::cerr << "Error: Could not allocate SwrContext." << std::endl;
 avcodec_free_context(&codec_ctx);
 avformat_close_input(&format_ctx);
 return AudioSegment();
 }
 codec_ctx->sample_rate = 44100;
 // Set up resampling context to convert to S16 format with 2 bytes per sample
 av_opt_set_chlayout(swr_ctx, "in_chlayout", &codec_ctx->ch_layout, 0);
 av_opt_set_int(swr_ctx, "in_sample_rate", codec_ctx->sample_rate, 0);
 av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", codec_ctx->sample_fmt, 0);

 AVChannelLayout dst_ch_layout;
 av_channel_layout_copy(&dst_ch_layout, &codec_ctx->ch_layout);
 av_channel_layout_uninit(&dst_ch_layout);
 av_channel_layout_default(&dst_ch_layout, 2);

 av_opt_set_chlayout(swr_ctx, "out_chlayout", &dst_ch_layout, 0);
 av_opt_set_int(swr_ctx, "out_sample_rate", codec_ctx->sample_rate, 0); // Match input sample rate
 av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0); // Force S16 format

 if (swr_init(swr_ctx) < 0) {
 std::cerr << "Error: Failed to initialize the resampling context" << std::endl;
 swr_free(&swr_ctx);
 avcodec_free_context(&codec_ctx);
 avformat_close_input(&format_ctx);
 return AudioSegment();
 }

 AVPacket packet;
 AVFrame* frame = av_frame_alloc();
 if (!frame) {
 std::cerr << "Error: Could not allocate frame." << std::endl;
 swr_free(&swr_ctx);
 avcodec_free_context(&codec_ctx);
 avformat_close_input(&format_ctx);
 return AudioSegment();
 }

 std::vector<char> output;
 while (av_read_frame(format_ctx, &packet) >= 0) {
 if (packet.stream_index == audio_stream_index) {
 if (avcodec_send_packet(codec_ctx, &packet) == 0) {
 while (avcodec_receive_frame(codec_ctx, frame) == 0) {
 if (frame->pts != AV_NOPTS_VALUE) {
 frame->pts = av_rescale_q(frame->pts, codec_ctx->time_base, format_ctx->streams[audio_stream_index]->time_base);
 }

 uint8_t* output_buffer;
 int output_samples = av_rescale_rnd(
 swr_get_delay(swr_ctx, codec_ctx->sample_rate) + frame->nb_samples,
 codec_ctx->sample_rate, codec_ctx->sample_rate, AV_ROUND_UP);

 int output_buffer_size = av_samples_get_buffer_size(
 nullptr, 2, output_samples, AV_SAMPLE_FMT_S16, 1);

 output_buffer = (uint8_t*)av_malloc(output_buffer_size);

 if (output_buffer) {
 memset(output_buffer, 0, output_buffer_size); // Zero padding to avoid random noise
 int converted_samples = swr_convert(swr_ctx, &output_buffer, output_samples,
 (const uint8_t**)frame->extended_data, frame->nb_samples);

 if (converted_samples >= 0) {
 output.insert(output.end(), output_buffer, output_buffer + output_buffer_size);
 }
 else {
 std::cerr << "Error: Failed to convert audio samples." << std::endl;
 }
 // Make sure output_buffer is valid before freeing
 if (output_buffer != nullptr) {
 av_free(output_buffer);
 output_buffer = nullptr; // Prevent double-free
 }
 }
 else {
 std::cerr << "Error: Could not allocate output buffer." << std::endl;
 }
 }
 }
 else {
 std::cerr << "Error: Failed to send packet to codec context." << std::endl;
 }
 }
 av_packet_unref(&packet);
 }

 int frame_width = av_get_bytes_per_sample(AV_SAMPLE_FMT_S16) * 2; // Use 2 bytes per sample and 2 channels

 std::map metadata = {
 {"sample_width", 2}, // S16 format has 2 bytes per sample
 {"frame_rate", codec_ctx->sample_rate}, // Use the input sample rate
 {"channels", 2}, // Assuming stereo output
 {"frame_width", frame_width}
 };

 av_frame_free(&frame);
 swr_free(&swr_ctx);
 avcodec_free_context(&codec_ctx);
 avformat_close_input(&format_ctx);

 return AudioSegment(static_cast<const>(output.data()), output.size(), metadata);
}

std::ofstream AudioSegment::export_segment_to_wav_file(const std::string& out_f) {
 std::cout << this->get_channels() << std::endl;
 av_log_set_level(AV_LOG_ERROR);
 AVCodecContext* codec_ctx = nullptr;
 AVFormatContext* format_ctx = nullptr;
 AVStream* stream = nullptr;
 AVFrame* frame = nullptr;
 AVPacket* pkt = nullptr;
 int ret;

 // Initialize format context for WAV
 if (avformat_alloc_output_context2(&format_ctx, nullptr, "wav", out_f.c_str()) < 0) {
 throw std::runtime_error("Could not allocate format context.");
 }

 // Find encoder for PCM
 const AVCodec* codec_ptr = avcodec_find_encoder(AV_CODEC_ID_PCM_S16LE);
 if (!codec_ptr) {
 throw std::runtime_error("PCM encoder not found.");
 }

 // Add stream
 stream = avformat_new_stream(format_ctx, codec_ptr);
 if (!stream) {
 throw std::runtime_error("Failed to create new stream.");
 }

 // Allocate codec context
 codec_ctx = avcodec_alloc_context3(codec_ptr);
 if (!codec_ctx) {
 throw std::runtime_error("Could not allocate audio codec context.");
 }

 // Set codec parameters for PCM
 codec_ctx->bit_rate = 128000; // Bitrate
 codec_ctx->sample_rate = this->get_frame_rate(); // Use correct sample rate
 codec_ctx->ch_layout.nb_channels = this->get_channels(); // Set the correct channel count

 // Set the channel layout: stereo or mono
 if (this->get_channels() == 2) {
 av_channel_layout_default(&codec_ctx->ch_layout, 2); // Stereo layout
 }
 else {
 av_channel_layout_default(&codec_ctx->ch_layout, 1); // Mono layout
 }

 codec_ctx->sample_fmt = AV_SAMPLE_FMT_S16; // PCM 16-bit format

 // Open codec
 if (avcodec_open2(codec_ctx, codec_ptr, nullptr) < 0) {
 throw std::runtime_error("Could not open codec.");
 }

 // Set codec parameters to the stream
 if (avcodec_parameters_from_context(stream->codecpar, codec_ctx) < 0) {
 throw std::runtime_error("Could not initialize stream codec parameters.");
 }

 // Open output file
 std::ofstream out_file(out_f, std::ios::binary);
 if (!out_file) {
 throw std::runtime_error("Failed to open output file.");
 }

 if (!(format_ctx->oformat->flags & AVFMT_NOFILE)) {
 if (avio_open(&format_ctx->pb, out_f.c_str(), AVIO_FLAG_WRITE) < 0) {
 throw std::runtime_error("Could not open output file.");
 }
 }

 // Write file header
 if (avformat_write_header(format_ctx, nullptr) < 0) {
 throw std::runtime_error("Error occurred when writing file header.");
 }

 // Initialize packet
 pkt = av_packet_alloc();
 if (!pkt) {
 throw std::runtime_error("Could not allocate AVPacket.");
 }

 // Initialize frame
 frame = av_frame_alloc();
 if (!frame) {
 throw std::runtime_error("Could not allocate AVFrame.");
 }

 // Set the frame properties
 frame->format = codec_ctx->sample_fmt;
 frame->ch_layout = codec_ctx->ch_layout;

 // Number of audio samples available in the data buffer
 int total_samples = data_.size() / (av_get_bytes_per_sample(AV_SAMPLE_FMT_S16) * codec_ctx->ch_layout.nb_channels);
 int samples_read = 0;

 // Set the number of samples per frame dynamically based on the input data
 while (samples_read < total_samples) {
 // Determine how many samples to read in this iteration (don't exceed the total sample count)
 int num_samples = std::min(codec_ctx->frame_size, total_samples - samples_read);
 if (num_samples == 0) {
 num_samples = 1024;
 codec_ctx->frame_size = 1024;
 }
 // Ensure num_samples is not zero
 if (num_samples <= 0) {
 throw std::runtime_error("Invalid number of samples in frame.");
 }

 // Set the number of samples in the frame
 frame->nb_samples = num_samples;

 // Allocate the frame buffer based on the number of samples
 ret = av_frame_get_buffer(frame, 0);
 if (ret < 0) {
 std::cerr << "Error allocating frame buffer: " << ret << std::endl;
 throw std::runtime_error("Could not allocate audio data buffers.");
 }

 // Copy the audio data into the frame's buffer (interleaving if necessary)
 /*if (codec_ctx->ch_layout.nb_channels == 2) {
 // If stereo, interleave planar data into packed format
 for (int i = 0; i < num_samples; ++i) {
 ((int16_t*)frame->data[0])[2 * i] = ((int16_t*)data_.data())[i]; // Left channel
 ((int16_t*)frame->data[0])[2 * i + 1] = ((int16_t*)data_.data())[total_samples + i]; // Right channel
 }
 }
 else {
 // For mono or packed data, directly copy the samples
 std::memcpy(frame->data[0], data_.data() + samples_read * av_get_bytes_per_sample(AV_SAMPLE_FMT_S16) * codec_ctx->ch_layout.nb_channels,
 num_samples * av_get_bytes_per_sample(AV_SAMPLE_FMT_S16) * codec_ctx->ch_layout.nb_channels);
 }
 */
 std::memcpy(frame->data[0], data_.data() + samples_read * av_get_bytes_per_sample(AV_SAMPLE_FMT_S16) * codec_ctx->ch_layout.nb_channels,
 num_samples * av_get_bytes_per_sample(AV_SAMPLE_FMT_S16) * codec_ctx->ch_layout.nb_channels);

 // Send the frame for encoding
 ret = avcodec_send_frame(codec_ctx, frame);
 if (ret < 0) {
 std::cerr << "Error sending frame for encoding: " << ret << std::endl;
 throw std::runtime_error("Error sending frame for encoding.");
 }

 // Receive and write encoded packets
 while (ret >= 0) {
 ret = avcodec_receive_packet(codec_ctx, pkt);
 if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF) {
 break;
 }
 else if (ret < 0) {
 throw std::runtime_error("Error during encoding.");
 }

 out_file.write(reinterpret_cast(pkt->data), pkt->size);
 av_packet_unref(pkt);
 }

 samples_read += num_samples;
 }

 // Flush the encoder
 if (avcodec_send_frame(codec_ctx, nullptr) < 0) {
 throw std::runtime_error("Error flushing the encoder.");
 }

 while (avcodec_receive_packet(codec_ctx, pkt) >= 0) {
 out_file.write(reinterpret_cast(pkt->data), pkt->size);
 av_packet_unref(pkt);
 }

 // Write file trailer
 av_write_trailer(format_ctx);

 // Cleanup
 av_frame_free(&frame);
 av_packet_free(&pkt);
 avcodec_free_context(&codec_ctx);

 if (!(format_ctx->oformat->flags & AVFMT_NOFILE)) {
 avio_closep(&format_ctx->pb);
 }
 avformat_free_context(format_ctx);

 out_file.close();
 return out_file;
}

</const></char>

Run code :


#include "audio_segment.h"
#include "effects.h"
#include "playback.h"
#include "cppaudioop.h"
#include "exceptions.h"
#include "generators.h"
#include "silence.h"
#include "utils.h"

#include <iostream>
#include <filesystem>

using namespace cppdub;

int main() {
 try {
 // Load the source audio file
 AudioSegment seg_1 = AudioSegment::from_file("../data/test10.mp3");
 std::string out_file_name = "ah-ah-ah.wav";

 // Export the audio segment to a new file with specified settings
 //seg_1.export_segment(out_file_name, "mp3");
 seg_1.export_segment_to_wav_file(out_file_name);


 // Optionally play the audio segment to verify
 // play(seg_1);

 // Load the exported audio file
 AudioSegment seg_2 = AudioSegment::from_file(out_file_name);

 // Play segments
 //play(seg_1);
 play(seg_2);
 }
 catch (const std::exception& e) {
 std::cerr << "An error occurred: " << e.what() << std::endl;
 }

 return 0;
}
</filesystem></iostream>

Error in second call of from_file function :


[pcm_s16le @ 000002d82ca5bfc0] Invalid PCM packet, data has size 2 but at least a size of 4 was expected


The process continue, i call hear the seg_2 with play(seg_2) call, but i can't directly play seg_2 export wav file (from windows explorer).


I had a guess that error may be because packed vs plannar formats missmatch but i am not quit sure. Maybe a swr_convert is necessary.


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