A SDK for sending RTP and RTMP streams
Project description
description
功能:将音视频转换成rtp和rtmp并发送
usage examples
源码编译安装ffmpeg和av
apt install -y wget unzip tar pigz rsync
wget http://prod2.wos.58dns.org/uwsRqPoNuVZ/humanvirtual/install.sh && chmod +x install.sh && /bin/bash install.sh # 耗时约10mins
安装StreamSender
pip install StreamSender
RTPSender用法示例:
from streamers.rtp_sender import RTPSender
from pydub import AudioSegment
import cv2
from time import sleep, time
if __name__ == '__main__':
ip_address = "127.0.0.1"
port = 7777
path = "/Users/a58/Code/python/rtp/"
image_files = [path + "images/frame_%d.png" % i for i in range(5)]
audio_file = path + "audios/bgroup.wav"
audio_16k_file = path + "audios/bgroup16k.wav"
audio_48k_file = path + "audios/bgroup48k.wav"
resolution = (1080, 1920) # (width, height)
rtpSender = RTPSender(ip_address, port, resolution, hard_encode=True, open_log=True, days=7)
audio = AudioSegment.from_file(audio_48k_file, format="wav")
audio_data = audio.raw_data
imgs = [cv2.imread(image_file) for image_file in image_files]
i, cnt = 0, 0
frame_size = 1920
t1 = time()
while True:
for img in imgs:
if i >= len(audio_data) - 50 * frame_size:
i = 0
for j in range(25):
rtpSender.send_video_rtp_from_img(img)
rtpSender.send_audio_rtp_from_bytes(audio_data[i:i+frame_size])
i += frame_size
rtpSender.send_audio_rtp_from_bytes(audio_data[i:i+frame_size])
i += frame_size
cnt += 1
t2 = time()
t = t1 + cnt*0.04 - t2
if t > 0:
sleep(t)
RTMPSender用法示例:
from streamers.rtmp_sender import RTMPSender
from pydub import AudioSegment
import cv2
from time import sleep, time
if __name__ == '__main__':
rtmp_url = 'your_rtmp_url'
path = "/Users/a58/Code/python/rtp/"
image_files = [path + "images/frame_%d.png" % i for i in range(5)]
audio_file = path + "audios/bgroup.wav"
audio_16k_file = path + "audios/bgroup16k.wav"
audio_48k_file = path + "audios/bgroup48k.wav"
resolution = (1080, 1920) # (width, height)
sample_rate = 48000
rtmpSender = RTMPSender(resolution, rtmp_url, sample_rate, hard_encode=True, open_log=True, days=7, stdout=False, bit_rate=600000)
audio = AudioSegment.from_file(audio_48k_file, format="wav")
audio_data = audio.raw_data
imgs = [cv2.imread(image_file) for image_file in image_files]
i, cnt = 0, 0
frame_size = 640
t1 = time()
while True:
for img in imgs:
if i >= len(audio_data) - 50 * frame_size:
i = 0
for j in range(25):
rtmpSender.send_video_rtmp_from_img(img)
rtmpSender.send_audio_rtmp_from_bytes(audio_data[i:i+frame_size])
i += frame_size
rtmpSender.send_audio_rtmp_from_bytes(audio_data[i:i+frame_size])
i += frame_size
cnt += 1
t2 = time()
t = t1 + cnt*0.04 - t2
if t > 0:
sleep(t)
Major Releases
| Release Version | Release Date | Updates |
|---|---|---|
| v0.0.8 | 2024-11-19 | 增加音频采样率转换条件 |
| v0.0.6 | 2024-10-31 | 使用PyAV推送rtmp |
| v0.0.4 | 2024-10-16 | 增加音频采样率转换 |
| v0.0.2 | 2024-10-15 | 增加rtmp推流异常处理, 并格式化日志 |
| v0.0.1 | 2024-10-15 | 调整目录结构, 同时支持RTPSender和RTMPSender |
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