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python - FFMPEG Streaming 抽动低比特率

转载 作者:行者123 更新时间:2023-12-04 22:50:12 24 4
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我有一个 python 脚本,它将为视频流生成帧。要将其流式传输到 twitch,我决定使用 ffmpeg(因为它是我找到的唯一选项)。但是,我的流的比特率非常低(70 KB),尽管在 ffmpeg 选项中它设置为 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()
如何提高比特率?也许您可以向我指出一些不同的解决方案,了解如何将 python 帧流式传输到 twitch 或任何其他 rtmp 服务器。
这是完整的日志,音频也坏了,只是噪音:
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

最佳答案

这是流到 twitch 或其他 RTMP 服务器的最低实现:

import threading, signal, os, sys, time, logging
import numpy as np
import cv2
import subprocess as sp

cap = cv2.VideoCapture(0)
frame_width= 1280
frame_height=720
FPS = 30


cap.set(cv2.CAP_PROP_FRAME_WIDTH, frame_width)
cap.set(cv2.CAP_PROP_FRAME_HEIGHT, frame_height)
cap.set(cv2.CAP_PROP_FPS,FPS)

print("************ cv2.CAP_PROP_FPS : ", cap.get(cv2.CAP_PROP_FPS))

CBR='4500k'
command = [
'ffmpeg',
'-f', 'rawvideo',
'-vcodec','rawvideo',
'-s', str(frame_width)+'x'+str(frame_height),
'-pix_fmt', 'bgr24',
'-r', str(FPS),
'-i', '-',
'-stream_loop', '-1',
'-i', 'PATH_TO_YOUR_AUDIO_FILE.mp3',
'-f', 'flv',
'-vcodec', 'libx264',
'-profile:v', 'main',
'-g', '60',
'-keyint_min', '30',
'-b:v', CBR,
'-minrate', CBR,
'-maxrate', CBR,
'-pix_fmt', 'yuv420p',
'-preset', 'ultrafast',
'-tune', 'zerolatency',
'-threads', '0',
'-bufsize', CBR,
'rtmp://RTMP_SERVER_ULR/YOUR_CHANNEL_KEY']

proc = sp.Popen(command, stdin=sp.PIPE, stderr=sp.STDOUT, bufsize=0)

while True:
ret, frame = cap.read()
proc.stdin.write(frame.tostring())


cap.release()
proc.stdin.close()
proc.wait()
几点注意事项:
  • opencv 颜色像素格式为 bgr24不是 rgb24
  • Youtube、Twitch 等不支持自适应码率,只支持恒定码率 CBR , 目前 ffmpeg 不支持开箱即用的恒定码率,所以你需要设置多个参数为你想要的码率 (b:v, minrate, maxrate, )
  • Youtube 要求您的视频流中有音频
  • Twitch 延迟为 翻倍如果您的流中没有音频
  • 在上面的示例中,我在循环中使用 mp3 文件,但您也可以使用音频源
  • 我怀疑您无法达到所需比特率的原因是-keyint_min太低,-g太高了,试试设置-keyint_min=fps-g=fpsx2
  • 关于python - FFMPEG Streaming 抽动低比特率,我们在Stack Overflow上找到一个类似的问题: https://stackoverflow.com/questions/62156992/

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