ffmpeg 广播直播转码 mp3/aac 到 g722

Question

我尝试将带有 ffmpeg 的广播流转码为 g722。我让流媒体工作并且无法收听流媒体。问题是输出流的速度比输入流快。所以结果不好。我试过用atempo减慢速度，但没有成功。

喜欢：

size=     241kB time=00:00:28.67 bitrate=  68.8kbits/s speed= 1.4x

从 1.x 到 15.x

控制台输出：

c:\ffmpeg\bin>ffmpeg -i http://lyd.nrk.no/nrk_radio_mp3_mp3_l -ac 1 -acodec g722 -f rtp -ab 64k -ar 16k rtp://192.168.0.99:555
ffmpeg version N-85750-ga75ef15 Copyright (c) 2000-2017 the FFmpeg developers
  built with gcc 6.3.0 (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-zlib
  libavutil      55. 61.100 / 55. 61.100
  libavcodec     57. 93.100 / 57. 93.100
  libavformat    57. 72.101 / 57. 72.101
  libavdevice    57.  7.100 / 57.  7.100
  libavfilter     6. 88.100 /  6. 88.100
  libswscale      4.  7.101 /  4.  7.101
  libswresample   2.  8.100 /  2.  8.100
  libpostproc    54.  6.100 / 54.  6.100
Input #0, mp3, from 'http://lyd.nrk.no/nrk_radio_mp3_mp3_l':
  Metadata:
    icy-name        : NRK mP3
    icy-pub         : 1
  Duration: N/A, start: 0.000000, bitrate: 96 kb/s
    Stream #0:0: Audio: mp3, 48000 Hz, stereo, s16p, 96 kb/s
[udp @ 000000000244bec0] 'circular_buffer_size' option was set but it is not supported on this build (pthread support is required)
[udp @ 00000000024781a0] 'circular_buffer_size' option was set but it is not supported on this build (pthread support is required)
Stream mapping:
  Stream #0:0 -> #0:0 (mp3 (native) -> adpcm_g722 (g722))
Press [q] to stop, [?] for help
Output #0, rtp, to 'rtp://192.168.0.99:555':
  Metadata:
    icy-name        : NRK mP3
    icy-pub         : 1
    encoder         : Lavf57.72.101
    Stream #0:0: Audio: adpcm_g722 (g722), 16000 Hz, mono, s16, 64 kb/s
    Metadata:
      encoder         : Lavc57.93.100 g722
SDP:
v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
c=IN IP4 192.168.0.99
t=0 0
a=tool:libavformat 57.72.101
m=audio 555 RTP/AVP 9
b=AS:64

size=     413kB time=00:00:49.17 bitrate=  68.8kbits/s speed=1.43x

有谁知道我做错了什么？谢谢

Answer 1

The problem is that the output stream have faster speed than the input stream

你确定吗？如果是这样，问题就归结为采样率。播放端（在 FFmpeg 之后）以比您从 FFmpeg 输出的采样率更高的采样率播放。

我怀疑这并没有真正发生，但基于这条评论：

this varies from 1.x to 15.x

当您连接到互联网广播流时，将尽快向您刷新一个大缓冲区。这可以让玩家快速启动。对于你的FFmpeg命令来说，意味着当你第一次连接的时候，FFmpeg也会以最快的速度处理这个数据并发送过来。如果终端播放设备正在缓冲数据，这通常没问题。如果不是，则必须强制 FFmpeg 缓冲数据。

您可以通过在输入前指定 -re 参数来实现。这将根据软件定义的时钟实时强制输入运行。

ffmpeg 广播直播转码 mp3/aac 到 g722

ffmpeg radio live transcoding mp3/aac to g722

audio

ffmpeg

transcoding

live-streaming