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• 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



  


• Converting HLS Stream to stream supported by old radio

  29 novembre 2024, par Alberto Faenza

  I have an old internet radio that does not support HLS streams.
Therefore I cannot listen to my favourite radio at this url :
https://streamcdnf31-4c4b867c89244861ac216426883d1ad0.msvdn.net/radiodeejay/radiodeejay/master_ma.m3u8

  


  I found a solution using a paid software https://minimradio.com/ which is based on minimserver and minimstreamer.

  


  This solution works if I install mininmserver and minimstreamer on a local computer and use the internet radio to point to the converter stream but I will have to pay if I want to use this.

  


  Checking the documentation of minimradio and ministreamer I can see the following :

  


  *Some internet radios can play the previous AAC ADTS streams but can't play these new HLS streams
...

  


  If the network stream URL points to an HLS .m3u8 master playlist or media playlist file, MinimStreamer reads this file and uses the HLS protocol to read the stream audio data and send it to the music player as a conventional HTTP stream. This makes the stream playable on music players that don't support the HLS protocol. The audio data in the stream must be encoded in AAC format.*
and not a single destination receiver I should use a streaming (broadcasting) server. What can I use to do that ?

  


  My question is the following :
Is there a way to replicate what minimstreamer is doing using ffmpeg ?
I have tried this :

  


  


  ffmpeg -re -i https://streamcdnf31-4c4b867c89244861ac216426883d1ad0.msvdn.net/radiodeejay/radiodeejay/master_ma.m3u8 -c copy -listen 1 -f mpegts http://192.168.1.9:10000

  


  


  which is playing corrctly in local vlc on the same computer. But when I stop VLC is got this error in ffmpeg :

  


  [https @ 00000291de047400] Cannot reuse HTTP connection for different host: StreamCdnG20-4c4b867c89244861ac216426883d1ad0.msvdn.net:-1 != 4c4b867c89244861ac216426883d1ad0.msvdn.net:-1
[hls @ 00000291dd96d140] keepalive request failed for 'https://4c4b867c89244861ac216426883d1ad0.msvdn.net/radiodeejay/radiodeejay/play1.m3u8' with error: 'Invalid argument' when parsing playlist
[hls @ 00000291dd96d140] Opening 'https://4c4b867c89244861ac216426883d1ad0.msvdn.net/radiodeejay/radiodeejay/play1.m3u8' for reading
[hls @ 00000291dd96d140] Skip ('#EXT-X-DISCONTINUITY-SEQUENCE:0')
[hls @ 00000291dd96d140] Skip ('#EXT-X-PROGRAM-DATE-TIME:2024-11-29T06:36:56.926Z')
[hls @ 00000291dd96d140] Skip ('#EXT-X-PROGRAM-DATE-TIME:2024-11-29T06:37:07.314Z')
[hls @ 00000291dd96d140] Skip ('#EXT-X-PROGRAM-DATE-TIME:2024-11-29T06:37:17.571Z')
[https @ 00000291de4e00c0] Opening 'https://StreamCdnG20-4c4b867c89244861ac216426883d1ad0.msvdn.net/radiodeejay/radiodeejay/20240722T095729_p1s_001086632.ts' for reading
[aost#0:0/copy @ 00000291de1c4f40] Error submitting a packet to the muxer: Error number -10054 occurred
    Last message repeated 1 times
[out#0/mpegts @ 00000291deaa7e40] Error muxing a packet
[out#0/mpegts @ 00000291deaa7e40] Task finished with error code: -10054 (Error number -10054 occurred)
[out#0/mpegts @ 00000291deaa7e40] Terminating thread with return code -10054 (Error number -10054 occurred)
[out#0/mpegts @ 00000291deaa7e40] Error writing trailer: Error number -10054 occurred
[http @ 00000291de8870c0] URL read error: Error number -10054 occurred
[out#0/mpegts @ 00000291deaa7e40] Error closing file: Error number -10054 occurred
[out#0/mpegts @ 00000291deaa7e40] video:0KiB audio:797KiB subtitle:0KiB other streams:0KiB global headers:0KiB muxing overhead: 21.849292%
size=     971KiB time=00:00:50.98 bitrate= 156.0kbits/s speed=1.01x
Conversion failed!


  


  And if I try to connect from my internet radio I immediately got this error :

  


  [aost#0:0/copy @ 0000027081584a40] Error submitting a packet to the muxer: Error number -10053 occurred
    Last message repeated 1 times
[out#0/mpegts @ 0000027081e684c0] Error muxing a packet
[out#0/mpegts @ 0000027081e684c0] Task finished with error code: -10053 (Error number -10053 occurred)
[out#0/mpegts @ 0000027081e684c0] Terminating thread with return code -10053 (Error number -10053 occurred)
[out#0/mpegts @ 0000027081e684c0] Error writing trailer: Error number -10053 occurred
[http @ 0000027081c47680] URL read error: Error number -10053 occurred
[out#0/mpegts @ 0000027081e684c0] Error closing file: Error number -10053 occurred
[out#0/mpegts @ 0000027081e684c0] video:0KiB audio:46KiB subtitle:0KiB other streams:0KiB global headers:0KiB muxing overhead: 13.917515%
size=      52KiB time=00:00:02.94 bitrate= 145.1kbits/s speed= 1.2x
Conversion failed!


  


  What is the correct way to stream this one locally in order to be listened in my internet radio ?
Shall I use ffmpeg or can be done directly with ngnix ? Or shall I use both ?