尝试从 Hikvision IP 摄像机正确获取视频和音频数据。
例如，对于 H264 + MP2 来说，一切都运行良好。
当我尝试在 PCM s16le 中获取 RAW 音频时，我脸上的笑容消失了。
这是我抓取相机的方式（你可以尝试将其向世界开放）：
ffmpeg -re -acodec pcm_s16le -ac 1 -rtsp_transport tcp -i rtsp://超级用户:[电子邮件保护]:10554 -vcodec 复制-acodec libfdk_aac -vbr 5 测试.ts
该命令有效并将 RTSP 流打包为 TS 文件。
但是音频和视频的时长不同。例如，我录制了 21 秒，其中音频为 21 秒，视频为 15 秒。
音频被拉长，音调降低。花了几天时间阅读 FFmpeg 文档，并应用了各种选项，如异步、更改采样率等 - 但没有成功。
我希望 Mulvya 或其他 FFmpeg 专家能建议我修复这个问题，以便正确完成工作。
C:\Users\User>d:/ffmpeg/bin/ffmpeg -y -re -acodec pcm_s16le -rtsp_transport
tcp -i rtsp://superuser:[email protected]:10554 -vcodec copy -
acodec aac -b:a 96k d:/ffmpeg/hik_aac.ts
ffmpeg version N-83410-gb1e2192 Copyright (c) 2000-2017 the FFmpeg
developers
built with gcc 5.4.0 (GCC)
configuration: --enable-gpl --enable-version3 --enable-cuda --enable-cuvid -
-enable-d3d11va --enable-dxva2 --enable-libmfx --enable-nvenc --enable-
avisynth --enable-bzlib --enable-fontconfig --enable-frei0r --enable-gnutls
--enable-iconv --enable-libass --enable-libbluray --enable-libbs2b --enable-
libcaca --enable-libfreetype --enable-libgme --enable-libgsm --enable-
libilbc --enable-libmodplug --enable-libmp3lame --enable-libopencore-amrnb -
-enable-libopencore-amrwb --enable-libopenh264 --enable-libopenjpeg --
enable-libopus --enable-librtmp --enable-libsnappy --enable-libsoxr --
enable-libspeex --enable-libtheora --enable-libtwolame --enable-libvidstab -
-enable-libvo-amrwbenc --enable-libvorbis --enable-libvpx --enable-
libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-
libxavs --enable-libxvid --enable-libzimg --enable-lzma --enable-decklink --
enable-zlib
libavutil 55. 46.100 / 55. 46.100
libavcodec 57. 75.100 / 57. 75.100
libavformat 57. 66.101 / 57. 66.101
libavdevice 57. 2.100 / 57. 2.100
libavfilter 6. 72.100 / 6. 72.100
libswscale 4. 3.101 / 4. 3.101
libswresample 2. 4.100 / 2. 4.100
libpostproc 54. 2.100 / 54. 2.100
Guessed Channel Layout for Input Stream #0.1 : mono
Input #0, rtsp, from 'rtsp://superuser:[email protected]:10554':
Metadata:
title : Media Presentation
Duration: N/A, start: 0.000000, bitrate: N/A
Stream #0:0: Video: h264 (Main), yuv420p(progressive), 1920x1080, 16 fps, 25
tbr, 90k tbn, 32.01 tbc
Stream #0:1: Audio: pcm_s16le, 16000 Hz, mono, s16, 256 kb/s
Output #0, mpegts, to 'd:/ffmpeg/hik_aac.ts':
Metadata:
title : Media Presentation
encoder : Lavf57.66.101
Stream #0:0: Video: h264 (Main), yuv420p(progressive), 1920x1080, q=2-31, 16
fps, 25 tbr, 90k tbn, 90k tbc
Stream #0:1: Audio: aac (LC), 16000 Hz, mono, fltp, 96 kb/s
Metadata:
encoder : Lavc57.75.100 aac
Stream mapping:
Stream #0:0 -> #0:0 (copy)
Stream #0:1 -> #0:1 (pcm_s16le (native) -> aac (native))
Press [q] to stop, [?] for help
[mpegts @ 00000000032cf020] Non-monotonous DTS in output stream 0:0;
previous: 33976, current: 7200; changing to 33977. This may result in
incorrect timestamps in the output file.
[mpegts @ 00000000032cf020] Non-monotonous DTS in output stream 0:0;
previous: 33977, current: 14400; changing to 33978. This may result in
incorrect timestamps in the output file.
[mpegts @ 00000000032cf020] Non-monotonous DTS in output stream 0:0;
previous: 33978, current: 18000; changing to 33979. This may result in
incorrect timestamps in the output file.
[mpegts @ 00000000032cf020] Non-monotonous DTS in output stream 0:0;
previous: 33979, current: 25200; changing to 33980. This may result in
incorrect timestamps in the output file.
[mpegts @ 00000000032cf020] Non-monotonous DTS in output stream 0:0;
previous: 33980, current: 28800; changing to 33981. This may result in
incorrect timestamps in the output file.
frame= 85 fps= 11 q=-1.0 Lsize= 1357kB time=00:00:07.42
bitrate=1497.1kbits/s speed=0.997x
video:1196kB audio:51kB subtitle:0kB other streams:0kB global headers:0kB
muxing overhead: 8.805858%
aac @ 00000000030a0a00] Qavg: 63342.980
Exiting normally, received signal 2.
答案1
根据评论，由于实际采样率似乎是 22.05 kHz，我们可以使音频符合该速率。
使用
ffmpeg -y -re -acodec pcm_s16le -rtsp_transport tcp -i rtsp://URL
-vcodec copy -af asetrate=22050 -acodec aac -b:a 96k test.mp4
它asetrate不会对音频进行重新采样，而只是重置采样率上下文。