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• avfilter : remove deprecated FF_API_LINK_PUBLIC

  19 février, par James Almer

  avfilter : remove deprecated FF_API_LINK_PUBLIC
  Deprecated since 2024-03-08.
  Signed-off-by : James Almer <jamrial@gmail.com>
  

  • [DH] libavfilter/avfilter.c
  • [DH] libavfilter/avfilter.h
  • [DH] libavfilter/version_major.h

• Increased File Size When Converting MP4 to WebM using FFmpeg

  23 décembre 2024, par kimgijeong

  I am using FFmpeg to convert MP4 to WebM with the following command :

  


  ffmpeg -y -hide_banner -nostats \
-f mov,mp4,m4a,3gp,3g2,mj2 -i "http://127.0.0.1:80/lotteon-low-bitrate.mp4" \
-threads auto -f webm -acodec libopus -b:a 96.059k -vcodec libsvtav1 -preset 11 -pix_fmt yuv420p \
-vf "scale='min(-1, iw)':'min(-1,ih)':force_original_aspect_ratio=decrease,crop=trunc(iw/2)*2:trunc(ih/2)*2" \
"/usr/local/m2/m2temp/xcdrtmp/2052_1.webm"


  


  However, the output webm file size is larger than the source MP4 file. For example :

  


  


  • Source MP4 : 4.6 MB (bit rate : 994,053 bps)

    


  


  • Output WebM : 16 MB (bit rate : 3,902,037 bps)

    


  


  


  I know SVT-AV1 encoder defaults to CRF mode. Due to not specifying the bitrate explicitly, the SVT-AV1 encoder automatically sets the bit_rate. It appears that the encoder is setting it to a much higher value (3,323,104 bps), causing the increase in file size compared to the source MP4 (994,053 bps). Here are the methods i tried to reduce the WebM file size compared to the source MP4 :

  


  

  1. -b:v 994k

  


  


  I tried to match the target bitrate with the source MP4's bitrate to reduce the output size, but encountered the following error :

  


  Svt[error]: Instance 1: Force key frames is not supported for VBR mode Last message r
epeated 2 times [libsvtav1 @ 0x239dd100] Error setting encoder parameters: bad parameter (0x80001005)


  


  Looking at the official documentation, this mode change (from CRF to VBR when setting a target bitrate) appears to be expected behavior. However, the error is puzzling since I haven't set any force keyframe parameters in the FFmpeg command.

  


  

  2. svtav1-params "mbr=994k"

  


  


  The second method i tried was using the svtav1-params "mbr=994k" option to set the maxrate while maintaining CRF mode This method showed some improvement, but the output file size was still larger than the source MP4.

  


  

  • Output WebM : 5MB (bit rate : 1,209,877 bps)

  


  


  The more critical reason why we can't adopt the second method (using svtav1-params "mbr=994k") is that even for the same MP4 source file, the output file size varies slightly with each encoding.

  


  

  3. -b:v 994k -svtav1-params “rc=2:pred-struct=1”(CBR, low delay)

  


  


  The final method I tried was setting the target bitrate while using CBR (Constant Bit Rate) and low-delay mode The default value for pred-structure is 2(random access), but I had to use low-delay mode due to the following error :

  


  Svt[error]: CBR Rate control is currently not supported for SVT_AV1_PRED_RANDOM_ACCESS, use VBR mode


  


  This way was the only approach among those i tried that successfully reduced the output size.

  


  

  • Output WebM : 4.3MB (bit rate : 1,032,373 bps)

  


  


  In summary, I have three questions about this MP4 to WebM conversion issue :

  


  


  1. When setting the target bitrate with -b:v 994k, the switch to VBR mode is expected behavior. However, why does the force keyframe error occur when we haven't explicitly set any force keyframe parameters ?

     


  


  2. Why does the output WebM file size fluctuate when setting maxrate either through -maxrate or svtav1-params "mbr=994k", even when using the same MP4 source file ?

     


  


  3. Besides using -b:v 994k -svtav1-params "rc=2:pred-struct=1" (CBR with low delay), are there any other methods that can guarantee a WebM output size smaller than the source MP4 when converting from MP4 to WebM ?

     


  


  


  I am using a recent version of the SVT-AV1 codec :

  


  Svt[info]: SVT [version]:       SVT-AV1 Encoder Lib 58146ca
Svt[info]: SVT [build]  :       GCC 11.5.0 20240719 (Red Hat 11.5.0-2)   64 bit
Svt[info]: LIB Build date: Oct 28 2024 07:40:59
ffmpeg video svt-av1


  


• ffprobe newer version detect audio codec incorrectly

  16 janvier, par alancc

  I find a strange problem.

  


  I have a test video with h264 video codec and aac audio codec. It is at https://drive.google.com/file/d/1YAyz5cO0kb9r0MgahCpISR4bZ_1_n8PL/view?usp=sharing

  


  I build a ffmpeg version by myself, its version is :

  


  ffprobe version 7.0.2 Copyright (c) 2007-2024 the FFmpeg developers
  built with gcc 14.1.0 (Rev3, Built by MSYS2 project)
  configuration: --enable-shared
  libavutil      59.  8.100 / 59.  8.100
  libavcodec     61.  3.100 / 61.  3.100
  libavformat    61.  1.100 / 61.  1.100
  libavdevice    61.  1.100 / 61.  1.100
  libavfilter    10.  1.100 / 10.  1.100
  libswscale      8.  1.100 /  8.  1.100
  libswresample   5.  1.100 /  5.  1.100


  


  I then use ffprobe to get its info :

  


  ffprobe -v quiet -print_format ini -show_streams -show_packets test_h264.mp4 > test_h264.ini


  


  Then I get an ini file which shows the audio codec as MP2 :

  


  [streams.stream.0]
index=0
codec_name=mp2
codec_long_name=MP2 (MPEG audio layer 2)
profile=unknown
codec_type=audio
codec_tag_string=mp4a
codec_tag=0x6134706d
sample_fmt=fltp
sample_rate=44100
channels=2
channel_layout=stereo
bits_per_sample=0
initial_padding=0
id=0x1
r_frame_rate=0/0
avg_frame_rate=0/0
time_base=1/44100
start_pts=2788
start_time=0.063220
duration_ts=435455
duration=9.874263
bit_rate=127706
max_bit_rate=N/A
bits_per_raw_sample=N/A
nb_frames=378
nb_read_frames=N/A
nb_read_packets=378


  


  Another developer he uses his version of ffprobe :

  


  ffprobe version 2023-02-22-git-d5cc7acff1-full_build-www.gyan.dev Copyright (c) 2007-2023 the FFmpeg developers  


  


  Based on the year, my version(2024) should be newer than his(2023), but his version of ffprobe can get the audio codec properly :

  


  [streams.stream.1]
index=1
codec_name=aac
codec_long_name=AAC (Advanced Audio Coding)
profile=LC
codec_type=audio
codec_tag_string=mp4a
codec_tag=0x6134706d
sample_fmt=fltp
sample_rate=44100
channels=2
channel_layout=stereo
bits_per_sample=0
initial_padding=0
id=0x2
r_frame_rate=0/0
avg_frame_rate=0/0
time_base=1/44100
start_pts=1764
start_time=0.040000
duration_ts=436480
duration=9.897506
bit_rate=111733
max_bit_rate=N/A
bits_per_raw_sample=N/A
nb_frames=427
nb_read_frames=N/A
nb_read_packets=427
extradata_size=5


  


  Why ?

  


  I also tried a ffprobe version on ubuntu with the following version :

  


  ffprobe version 6.1.1-3ubuntu5 Copyright (c) 2007-2023 the FFmpeg developers
  built with gcc 13 (Ubuntu 13.2.0-23ubuntu3)
  configuration: --prefix=/usr --extra-version=3ubuntu5 --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --disable-omx --enable-gnutls --enable-libaom --enable-libass --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libdav1d --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libglslang --enable-libgme --enable-libgsm --enable-libharfbuzz --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzimg --enable-openal --enable-opencl --enable-opengl --disable-sndio --enable-libvpl --disable-libmfx --enable-libdc1394 --enable-libdrm --enable-libiec61883 --enable-chromaprint --enable-frei0r --enable-ladspa --enable-libbluray --enable-libjack --enable-libpulse --enable-librabbitmq --enable-librist --enable-libsrt --enable-libssh --enable-libsvtav1 --enable-libx264 --enable-libzmq --enable-libzvbi --enable-lv2 --enable-sdl2 --enable-libplacebo --enable-librav1e --enable-pocketsphinx --enable-librsvg --enable-libjxl --enable-shared
  libavutil      58. 29.100 / 58. 29.100
  libavcodec     60. 31.102 / 60. 31.102
  libavformat    60. 16.100 / 60. 16.100
  libavdevice    60.  3.100 / 60.  3.100
  libavfilter     9. 12.100 /  9. 12.100
  libswscale      7.  5.100 /  7.  5.100
  libswresample   4. 12.100 /  4. 12.100
  libpostproc    57.  3.100 / 57.  3.100


  


  It will detect the audio as aac properly, but with different parameters, for example, bit_rate is 111733(developer) but 110399(ubuntu). But this parameter comes from the same file so should be the same.

  


  [streams.stream.1]
index=1
codec_name=aac
codec_long_name=AAC (Advanced Audio Coding)
profile=LC
codec_type=audio
codec_tag_string=mp4a
codec_tag=0x6134706d
sample_fmt=fltp
sample_rate=44100
channels=2
channel_layout=stereo
bits_per_sample=0
initial_padding=0
id=0x2
r_frame_rate=0/0
avg_frame_rate=0/0
time_base=1/44100
start_pts=0
start_time=0.000000
duration_ts=441353
duration=10.008005
bit_rate=110399
max_bit_rate=N/A
bits_per_raw_sample=N/A
nb_frames=432
nb_read_frames=N/A
nb_read_packets=432
extradata_size=5