Hello,
I am seeing a fluctuating bitrate when I attempt to convert an AC3 audio stream
that is 5.1 Dolby to AAC in stereo. I thought that to do the proper downmux you
simply need to add the -ac 2 option to your audio output command but I must be
missing something. The target bitrate is 128 but I see it dipping as low as 114
and peak as high as 140. Am I missing something or is this expected behavior
for what I am doing?
Here is the command and output:
user@gpu-transcoder:~/scripts$ ffmpeg -i
'udp://@239.1.1.92:59092?fifo_size=1000000&overrun_nonfatal=1' -vcodec
h264_nvenc -bf:v 2 -g 60 -rc cbr -b:v 6000K -minrate 6000k -maxrate 6000k
-bufsize 12000k -profile:v high -level 4.0 -acodec libfdk_aac -ac 2 -b:a 128k
-f mpegts 'udp://@233.1.1.100:59100?pkt_size=1316'
ffmpeg version N-93005-gd92f06e Copyright (c) 2000-2019 the FFmpeg developers
built with gcc 7 (Ubuntu 7.3.0-27ubuntu1~18.04)
configuration: --prefix=/home/circle/ffmpeg_build --pkg-config-flags=--static
--extra-cflags=-I/home/circle/ffmpeg_build/include
--extra-ldflags=-L/home/circle/ffmpeg_build/lib --extra-libs='-lpthread -lm'
--bindir=/home/circle/bin --enable-gpl --enable-libaom --enable-libass
--enable-libfdk-aac --enable-libfreetype --enable-libmp3lame --enable-libopus
--enable-libvorbis --enable-libvpx --enable-libx264 --enable-libx265
--enable-nonfree --enable-nvenc
libavutil 56. 26.100 / 56. 26.100
libavcodec 58. 44.100 / 58. 44.100
libavformat 58. 26.100 / 58. 26.100
libavdevice 58. 6.101 / 58. 6.101
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
Input #0, mpegts, from
'udp://@239.1.1.92:59092?fifo_size=1000000&overrun_nonfatal=1':
Duration: N/A, start: 58320.721844, bitrate: N/A
Program 3
Stream #0:0[0x31]: Video: mpeg2video (Main) ([2][0][0][0] / 0x0002),
yuv420p(tv, top first), 1920x1080 [SAR 1:1 DAR 16:9], Closed Captions, 29.97
fps, 29.97 tbr, 90k tbn, 59.94 tbc
Stream #0:1[0x34](eng): Audio: ac3 ([129][0][0][0] / 0x0081), 48000 Hz,
5.1(side), fltp, 384 kb/s
Stream #0:2[0x35](spa): Audio: ac3 ([129][0][0][0] / 0x0081), 48000 Hz,
stereo, fltp, 96 kb/s (visual impaired) (descriptions)
Stream mapping:
Stream #0:0 -> #0:0 (mpeg2video (native) -> h264 (h264_nvenc))
Stream #0:1 -> #0:1 (ac3 (native) -> aac (libfdk_aac))
Press [q] to stop, [?] for help
Output #0, mpegts, to 'udp://@239.129.2.220:59220?pkt_size=1316':
Metadata:
encoder : Lavf58.26.100
Stream #0:0: Video: h264 (h264_nvenc) (High), yuv420p, 1920x1080 [SAR 1:1
DAR 16:9], q=-1--1, 6000 kb/s, 29.97 fps, 90k tbn, 29.97 tbc
Metadata:
encoder : Lavc58.44.100 h264_nvenc
Side data:
cpb: bitrate max/min/avg: 6000000/0/6000000 buffer size: 12000000
vbv_delay: -1
Stream #0:1(eng): Audio: aac (libfdk_aac), 48000 Hz, stereo, s16, 128 kb/s
Metadata:
encoder : Lavc58.44.100 libfdk_aac
frame= 1622 fps= 33 q=34.0 size= 43862kB time=00:00:54.42
bitrate=6602.5kbits/s speed=1.11x
Here are the FFPROBE outputs to show the audio bitrate changes:
Input #0, mpegts, from 'udp://@233.1.1.100:59100':
Duration: N/A, start: 47.096589, bitrate: N/A
Program 1
Metadata:
service_name : Service01
service_provider: FFmpeg
Stream #0:0[0x100]: Video: h264 (High) ([27][0][0][0] / 0x001B),
yuv420p(progressive), 1920x1080 [SAR 1:1 DAR 16:9], Closed Captions, 29.97 fps,
29.97 tbr, 90k tbn, 59.94 tbc
Stream #0:1[0x101](eng): Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000
Hz, stereo, fltp, 127 kb/s
And another:
Input #0, mpegts, from 'udp://@239.129.2.100:59100':
Duration: N/A, start: 206.629456, bitrate: N/A
Program 1
Metadata:
service_name : Service01
service_provider: FFmpeg
Stream #0:0[0x100]: Video: h264 (High) ([27][0][0][0] / 0x001B),
yuv420p(progressive), 1920x1080 [SAR 1:1 DAR 16:9], Closed Captions, 29.97 fps,
29.97 tbr, 90k tbn, 59.94 tbc
Stream #0:1[0x101](eng): Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000
Hz, stereo, fltp, 120 kb/s
Let me know what else you may need.
Thanks,
Jimbo
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