FFMPEG流可降低低比特率 [英] FFMPEG Streaming to twitch low bitrate

查看:136
本文介绍了FFMPEG流可降低低比特率的处理方法,对大家解决问题具有一定的参考价值,需要的朋友们下面随着小编来一起学习吧!

问题描述

我有一个python脚本,可以为视频流生成帧.为了流化它,我决定使用ffmpeg(因为这是我找到的唯一选项).但是,尽管在ffmpeg选项中将其设置为3000K,但我的流的比特率却非常低(70 KB).

I have a python script that will produce frames for a video stream. To stream it to twitch I decided to use ffmpeg (as it is the only option I found). However, the bitrate of my stream is very low (70 KB), although in ffmpeg options it's set to 3000K.

# This script copies the video frame by frame
import cv2
import subprocess as sp
twitch_stream_key = 'MY_TWITCH_STREAM_KEY'
input_file = 'video.mp4'

cap = cv2.VideoCapture(input_file)
ret, frame = cap.read()
height, width, ch = frame.shape

ffmpeg = 'FFMPEG'
dimension = '{}x{}'.format(width, height)

fps = cap.get(cv2.CAP_PROP_FPS)
command = []
command.extend([
    'FFMPEG',
    '-loglevel', 'verbose',
    '-y',  # overwrite previous file/stream
    '-analyzeduration', '1',
    '-f', 'rawvideo',
    '-r', '%d' % fps,  # set a fixed frame rate
    '-vcodec', 'rawvideo',
    # size of one frame
    '-s', '%dx%d' % (width, height),
    '-pix_fmt', 'rgb24',  # The input are raw bytes
    '-thread_queue_size', '1024',
    '-i', '-',  # The input comes from a pipe
])       
command.extend([
    '-ar', '8000',
    '-ac', '1',
    '-f', 's16le',
    '-i', 'work.mp3',
])
command.extend([
    # VIDEO CODEC PARAMETERS
    '-vcodec', 'libx264',
    '-r', '%d' % fps,
    '-b:v', '3000k',
    '-s', '%dx%d' % (width, height),
    '-preset', 'faster', '-tune', 'zerolatency',
    '-crf', '23',
    '-pix_fmt', 'yuv420p',

    '-minrate', '3000k', '-maxrate', '3000k',
    '-bufsize', '12000k',
    '-g', '60',  # key frame distance
    '-keyint_min', '1',

    # AUDIO CODEC PARAMETERS
    '-acodec', 'libmp3lame', '-ar', '44100', '-b:a', '160k',
    # '-bufsize', '8192k',
    '-ac', '1',
    '-map', '0:v', '-map', '1:a',

    '-threads', '2',
    # STREAM TO TWITCH
    '-f', 'flv', 'rtmp://live-hel.twitch.tv/app/%s' %
          twitch_stream_key
])
proc = sp.Popen(command, stdin=sp.PIPE, stderr=sp.PIPE)

while True:
    ret, frame = cap.read()        
    if not ret:        
        break    
    proc.stdin.write(frame.tostring())

cap.release()
proc.stdin.close()
proc.stderr.close()
proc.wait()

如何增加比特率?也许您可以向我介绍一些有关如何将python帧流化为twitch或任何其他rtmp服务器的解决方案.

How can I increase the bitrate? Maybe you can point me towards some different solution on how I can stream python frames to twitch or any other rtmp server.

这是完整的日志,音频也坏了,只是噪音:

Here is the complete log, the audio is also broken, it's just noise:

ffmpeg version git-2020-06-01-dd76226 Copyright (c) 2000-2020 the FFmpeg developers
  built with gcc 9.3.1 (GCC) 20200523
  configuration: --enable-gpl --enable-version3 --enable-sdl2 --enable-fontconfig --enable-gnutls --enable-iconv --enable-libass --enable-libdav1d --enable-libbluray --enable-libfreetype --enable-libmp3lame --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libopenjpeg --enable-libopus --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libsrt --enable-libtheora --enable-libtwolame --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx264 --enable-libx265 --enable-libxml2 --enable-libzimg --enable-lzma --enable-zlib --enable-gmp --enable-libvidstab --enable-libvmaf --enable-libvorbis --enable-libvo-amrwbenc --enable-libmysofa --enable-libspeex --enable-libxvid --enable-libaom --disable-w32threads --enable-libmfx --enable-ffnvcodec --enable-cuda-llvm --enable-cuvid --enable-d3d11va --enable-nvenc --enable-nvdec --enable-dxva2 --enable-avisynth --enable-libopenmpt --enable-amf
  libavutil      56. 49.100 / 56. 49.100
  libavcodec     58. 90.100 / 58. 90.100
  libavformat    58. 44.100 / 58. 44.100
  libavdevice    58.  9.103 / 58.  9.103
  libavfilter     7. 84.100 /  7. 84.100
  libswscale      5.  6.101 /  5.  6.101
  libswresample   3.  6.100 /  3.  6.100
  libpostproc    55.  6.100 / 55.  6.100
Input #0, rawvideo, from 'pipe:':
  Duration: N/A, start: 0.000000, bitrate: 1443225 kb/s
    Stream #0:0: Video: rawvideo, 1 reference frame (RGB[24] / 0x18424752), rgb24, 1920x1080, 1443225 kb/s, 29 tbr, 29 tbn, 29 tbc
[s16le @ 0000026d64eb5340] Estimating duration from bitrate, this may be inaccurate
Guessed Channel Layout for Input Stream #1.0 : mono
Input #1, s16le, from 'work.mp3':
  Metadata:
    encoded_by      : iTunes v7.0
  Duration: 00:09:36.13, bitrate: 128 kb/s
    Stream #1:0: Audio: pcm_s16le, 8000 Hz, mono, s16, 128 kb/s
[tcp @ 0000026d64ee34c0] Starting connection attempt to 99.181.64.78 port 1935
[tcp @ 0000026d64ee34c0] Successfully connected to 99.181.64.78 port 1935
Stream mapping:
  Stream #0:0 -> #0:0 (rawvideo (native) -> h264 (libx264))
  Stream #1:0 -> #0:1 (pcm_s16le (native) -> mp3 (libmp3lame))
[graph 0 input from stream 0:0 @ 0000026d64f47c00] w:1920 h:1080 pixfmt:rgb24 tb:1/29 fr:29/1 sar:0/1
[scaler_out_0_0 @ 0000026d64f4c780] w:1920 h:1080 flags:'bicubic' interl:0
[scaler_out_0_0 @ 0000026d64f4c780] w:1920 h:1080 fmt:rgb24 sar:0/1 -> w:1920 h:1080 fmt:yuv420p sar:0/1 flags:0x4
[libx264 @ 0000026d64edf840] using cpu capabilities: MMX2 SSE2Fast SSSE3 SSE4.2 AVX FMA3 BMI2 AVX2
[libx264 @ 0000026d64edf840] profile High, level 4.0, 4:2:0, 8-bit
[libx264 @ 0000026d64edf840] 264 - core 160 - H.264/MPEG-4 AVC codec - Copyleft 2003-2020 - http://www.videolan.org/x264.html - options: cabac=1 ref=2 deblock=1:0:0 analyse=0x3:0x113 me=hex subme=4 psy=1 psy_rd=1.00:0.00 mixed_ref=0 me_range=16 chroma_me=1 trellis=1 8x8dct=1 cqm=0 deadzone=21,11 fast_pskip=1 chroma_qp_offset=0 threads=2 lookahead_threads=2 sliced_threads=1 slices=2 nr=0 decimate=1 interlaced=0 bluray_compat=0 constrained_intra=0 bframes=0 weightp=1 keyint=60 keyint_min=1 scenecut=40 intra_refresh=0 rc_lookahead=0 rc=crf mbtree=0 crf=23.0 qcomp=0.60 qpmin=0 qpmax=69 qpstep=4 vbv_maxrate=3000 vbv_bufsize=12000 crf_max=0.0 nal_hrd=none filler=0 ip_ratio=1.40 aq=1:1.00
[graph_1_in_1_0 @ 0000026d651319c0] tb:1/8000 samplefmt:s16 samplerate:8000 chlayout:0x4
[format_out_0_1 @ 0000026d65132d80] auto-inserting filter 'auto_resampler_0' between the filter 'Parsed_anull_0' and the filter 'format_out_0_1'
[auto_resampler_0 @ 0000026d651331c0] ch:1 chl:mono fmt:s16 r:8000Hz -> ch:1 chl:mono fmt:s16p r:44100Hz
Output #0, flv, to 'rtmp://live-hel.twitch.tv/app/live_*************':
  Metadata:
    encoder         : Lavf58.44.100
    Stream #0:0: Video: h264 (libx264), 1 reference frame ([7][0][0][0] / 0x0007), yuv420p(progressive), 1920x1080, q=-1--1, 3000 kb/s, 29 fps, 1k tbn, 29 tbc
    Metadata:
      encoder         : Lavc58.90.100 libx264
    Side data:
      cpb: bitrate max/min/avg: 3000000/0/3000000 buffer size: 12000000 vbv_delay: N/A
    Stream #0:1: Audio: mp3 (libmp3lame) ([2][0][0][0] / 0x0002), 44100 Hz, mono, s16p, delay 1105, 160 kb/s
    Metadata:
      encoder         : Lavc58.90.100 libmp3lame

推荐答案

最初的问题是由于使用了两种相互冲突的速率控制模式. -b:v -crf 是互斥的.由于您正在流式传输,并且想要更特定的比特率,因此请删除 -crf .

Original problem is due to using two conflicting modes of rate control. -b:v and -crf are mutually exclusive. Since you are streaming and want a more specific bitrate remove -crf.

这篇关于FFMPEG流可降低低比特率的文章就介绍到这了,希望我们推荐的答案对大家有所帮助,也希望大家多多支持IT屋!

查看全文
登录 关闭
扫码关注1秒登录
发送“验证码”获取 | 15天全站免登陆