首先，我想首先说我真的很新 Gstreamer及其能力如此赦免我的无知，如果我的理解或 实施是错误的，我还在学习。
我想用QOS在PYTHON或JAVA（因为我不精通C）中构建一个小型流媒体应用程序
集成在其中，特别是对于包丢弃计数统计和RTP
和RTCP似乎是完美的匹配。
为此，我实施了一个 服务器
#! /usr/bin/env python
import gi
import sys
gi.require_version('Gst', '1.0')
from gi.repository import GObject, Gst
#gst-launch -v rtpbin name=rtpbin audiotestsrc ! audioconvert ! alawenc ! rtppcmapay ! rtpbin.send_rtp_sink_0 \
# rtpbin.send_rtp_src_0 ! udpsink port=10000 host=xxx.xxx.xxx.xxx \
# rtpbin.send_rtcp_src_0 ! udpsink port=10001 host=xxx.xxx.xxx.xxx sync=false async=false \
# udpsrc port=10002 ! rtpbin.recv_rtcp_sink_0
DEST_HOST = '127.0.0.1'
AUDIO_SRC = 'audiotestsrc'
AUDIO_ENC = 'alawenc'
AUDIO_PAY = 'rtppcmapay'
RTP_SEND_PORT = 5002
RTCP_SEND_PORT = 5003
RTCP_RECV_PORT = 5007
GObject.threads_init()
Gst.init(sys.argv)
# the pipeline to hold everything
pipeline = Gst.Pipeline.new('rtp_server')
# the pipeline to hold everything
audiosrc = Gst.ElementFactory.make(AUDIO_SRC, 'audiosrc')
audioconv = Gst.ElementFactory.make('audioconvert', 'audioconv')
audiores = Gst.ElementFactory.make('audioresample', 'audiores')
# the pipeline to hold everything
audioenc = Gst.ElementFactory.make(AUDIO_ENC, 'audioenc')
audiopay = Gst.ElementFactory.make(AUDIO_PAY, 'audiopay')
# add capture and payloading to the pipeline and link
pipeline.add(audiosrc)
pipeline.add(audioconv)
pipeline.add(audiores)
pipeline.add(audioenc)
pipeline.add(audiopay)
audiosrc.link(audioconv)
audioconv.link(audiores)
audiores.link(audioenc)
audioenc.link(audiopay)
# the rtpbin element
rtpbin = Gst.ElementFactory.make('rtpbin', 'rtpbin')
pipeline.add(rtpbin)
# the udp sinks and source we will use for RTP and RTCP
rtpsink = Gst.ElementFactory.make('udpsink', 'rtpsink')
rtpsink.set_property('port', RTP_SEND_PORT)
rtpsink.set_property('host', DEST_HOST)
rtcpsink = Gst.ElementFactory.make('udpsink', 'rtcpsink')
rtcpsink.set_property('port', RTCP_SEND_PORT)
rtcpsink.set_property('host', DEST_HOST)
# no need for synchronisation or preroll on the RTCP sink
rtcpsink.set_property('async', False)
rtcpsink.set_property('sync', False)
rtcpsrc = Gst.ElementFactory.make('udpsrc', 'rtcpsrc')
rtcpsrc.set_property('port', RTCP_RECV_PORT)
pipeline.add(rtpsink)
pipeline.add(rtcpsink)
pipeline.add(rtcpsrc)
# now link all to the rtpbin, start by getting an RTP sinkpad for session 0
sinkpad = Gst.Element.get_request_pad(rtpbin, 'send_rtp_sink_0')
srcpad = Gst.Element.get_static_pad(audiopay, 'src')
lres = Gst.Pad.link(srcpad, sinkpad)
# get the RTP srcpad that was created when we requested the sinkpad above and
# link it to the rtpsink sinkpad
srcpad = Gst.Element.get_static_pad(rtpbin, 'send_rtp_src_0')
sinkpad = Gst.Element.get_static_pad(rtpsink, 'sink')
lres = Gst.Pad.link(srcpad, sinkpad)
# get an RTCP srcpad for sending RTCP to the receiver
srcpad = Gst.Element.get_request_pad(rtpbin, 'send_rtcp_src_0')
sinkpad = Gst.Element.get_static_pad(rtcpsink, 'sink')
lres = Gst.Pad.link(srcpad, sinkpad)
# we also want to receive RTCP, request an RTCP sinkpad for session 0 and
# link it to the srcpad of the udpsrc for RTCP
srcpad = Gst.Element.get_static_pad(rtcpsrc, 'src')
sinkpad = Gst.Element.get_request_pad(rtpbin, 'recv_rtcp_sink_0')
lres = Gst.Pad.link(srcpad, sinkpad)
# set the pipeline to playing
Gst.Element.set_state(pipeline, Gst.State.PLAYING)
# we need to run a GLib main loop to get the messages
mainloop = GObject.MainLoop()
mainloop.run()
Gst.Element.set_state(pipeline, Gst.State.NULL)
和客户
#! /usr/bin/env python
import gi
import sys
gi.require_version('Gst', '1.0')
from gi.repository import GObject, Gst
#
# A simple RTP receiver
#
# receives alaw encoded RTP audio on port 5002, RTCP is received on port 5003.
# the receiver RTCP reports are sent to port 5007
#
# .-------. .----------. .---------. .-------. .--------.
# RTP |udpsrc | | rtpbin | |pcmadepay| |alawdec| |alsasink|
# port=5002 | src->recv_rtp recv_rtp->sink src->sink src->sink |
# '-------' | | '---------' '-------' '--------'
# | |
# | | .-------.
# | | |udpsink| RTCP
# | send_rtcp->sink | port=5007
# .-------. | | '-------' sync=false
# RTCP |udpsrc | | | async=false
# port=5003 | src->recv_rtcp |
# '-------' '----------'
AUDIO_CAPS = 'application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)PCMA'
AUDIO_DEPAY = 'rtppcmadepay'
AUDIO_DEC = 'alawdec'
AUDIO_SINK = 'autoaudiosink'
DEST = '127.0.0.1'
RTP_RECV_PORT = 5002
RTCP_RECV_PORT = 5003
RTCP_SEND_PORT = 5007
GObject.threads_init()
Gst.init(sys.argv)
#gst-launch -v rtpbin name=rtpbin \
# udpsrc caps=$AUDIO_CAPS port=$RTP_RECV_PORT ! rtpbin.recv_rtp_sink_0 \
# rtpbin. ! rtppcmadepay ! alawdec ! audioconvert ! audioresample ! autoaudiosink \
# udpsrc port=$RTCP_RECV_PORT ! rtpbin.recv_rtcp_sink_0 \
# rtpbin.send_rtcp_src_0 ! udpsink port=$RTCP_SEND_PORT host=$DEST sync=false async=false
def pad_added_cb(rtpbin, new_pad, depay):
sinkpad = Gst.Element.get_static_pad(depay, 'sink')
lres = Gst.Pad.link(new_pad, sinkpad)
# the pipeline to hold eveything
pipeline = Gst.Pipeline.new('rtp_client')
# the udp src and source we will use for RTP and RTCP
rtpsrc = Gst.ElementFactory.make('udpsrc', 'rtpsrc')
rtpsrc.set_property('port', RTP_RECV_PORT)
# we need to set caps on the udpsrc for the RTP data
caps = Gst.caps_from_string(AUDIO_CAPS)
rtpsrc.set_property('caps', caps)
rtcpsrc = Gst.ElementFactory.make('udpsrc', 'rtcpsrc')
rtcpsrc.set_property('port', RTCP_RECV_PORT)
rtcpsink = Gst.ElementFactory.make('udpsink', 'rtcpsink')
rtcpsink.set_property('port', RTCP_SEND_PORT)
rtcpsink.set_property('host', DEST)
# no need for synchronisation or preroll on the RTCP sink
rtcpsink.set_property('async', False)
rtcpsink.set_property('sync', False)
pipeline.add(rtpsrc)
pipeline.add(rtcpsrc)
pipeline.add(rtcpsink)
# the depayloading and decoding
audiodepay = Gst.ElementFactory.make(AUDIO_DEPAY, 'audiodepay')
audiodec = Gst.ElementFactory.make(AUDIO_DEC, 'audiodec')
# the audio playback and format conversion
audioconv = Gst.ElementFactory.make('audioconvert', 'audioconv')
audiores = Gst.ElementFactory.make('audioresample', 'audiores')
audiosink = Gst.ElementFactory.make(AUDIO_SINK, 'audiosink')
# add depayloading and playback to the pipeline and link
pipeline.add(audiodepay)
pipeline.add(audiodec)
pipeline.add(audioconv)
pipeline.add(audiores)
pipeline.add(audiosink)
audiodepay.link(audiodec)
audiodec.link(audioconv)
audioconv.link(audiores)
audiores.link(audiosink)
# the rtpbin element
rtpbin = Gst.ElementFactory.make('rtpbin', 'rtpbin')
pipeline.add(rtpbin)
# now link all to the rtpbin, start by getting an RTP sinkpad for session 0
srcpad = Gst.Element.get_static_pad(rtpsrc, 'src')
sinkpad = Gst.Element.get_request_pad(rtpbin, 'recv_rtp_sink_0')
lres = Gst.Pad.link(srcpad, sinkpad)
# get an RTCP sinkpad in session 0
srcpad = Gst.Element.get_static_pad(rtcpsrc, 'src')
sinkpad = Gst.Element.get_request_pad(rtpbin, 'recv_rtcp_sink_0')
lres = Gst.Pad.link(srcpad, sinkpad)
# get an RTCP srcpad for sending RTCP back to the sender
srcpad = Gst.Element.get_request_pad(rtpbin, 'send_rtcp_src_0')
sinkpad = Gst.Element.get_static_pad(rtcpsink, 'sink')
lres = Gst.Pad.link(srcpad, sinkpad)
rtpbin.connect('pad-added', pad_added_cb, audiodepay)
# def newManager():
# rtpbin.connect('on-ssrc-active', onSSRCActive)
# def onSSRCActive(self):
# print (self)
# rtpsrc.connect('new-manager', newManager)
Gst.Element.set_state(pipeline, Gst.State.PLAYING)
mainloop = GObject.MainLoop()
mainloop.run()
Gst.Element.set_state(pipeline, Gst.State.NULL)
虽然服务器和客户端工作没有问题我找不到方法或 有关如何从RTCP检索此RTP流的已删除包统计信息的信息，即使它位于命令行中也是如此。
在这方面的任何帮助将不胜感激！
答案 0 :(得分：1)
这是一个模拟，在Linux上，只是为了说明一种收听总线消息的方法
我没有时间整理QOS Gst.Message，所以我在那里攻击了一些强迫QOS的东西，所以你至少看到了它。
希望这足以让你朝着正确的方向前进。
#!/usr/bin/python3
from os import path
import time
import gi
gi.require_version('Gst', '1.0')
gi.require_version('Gtk', '3.0')
gi.require_version('GstVideo', '1.0')
from gi.repository import GObject, Gst, Gtk
from gi.repository import GdkX11, GstVideo
GObject.threads_init()
Gst.init(None)
filename = path.join(path.dirname(path.abspath(__file__)), '../H.mkv')
uri = 'file://' + filename
class Player(object):
def __init__(self):
self.window = Gtk.Window()
self.window.connect('destroy', self.quit)
self.window.set_default_size(800, 450)
self.drawingarea = Gtk.DrawingArea()
self.window.add(self.drawingarea)
# Create GStreamer pipeline
self.pipeline = Gst.Pipeline()
# Create bus to get events from GStreamer pipeline
self.bus = self.pipeline.get_bus()
self.bus.add_signal_watch()
self.bus.connect('message::eos', self.on_eos)
self.bus.connect('message::error', self.on_error)
self.bus.connect('message::qos', self.on_quality_of_service)
self.bus.enable_sync_message_emission()
self.bus.connect('sync-message::element', self.on_sync_message)
# Create GStreamer elements
self.playbin = Gst.ElementFactory.make('playbin', None)
# Add playbin to the pipeline
self.pipeline.add(self.playbin)
# Set properties
self.playbin.set_property('uri', uri)
def run(self):
self.window.show_all()
self.xid = self.drawingarea.get_property('window').get_xid()
self.pipeline.set_state(Gst.State.PLAYING)
time.sleep(2)
self.bus.emit('message::qos',Gst.Message('xxxxx'))
Gtk.main()
def quit(self, window):
self.pipeline.set_state(Gst.State.NULL)
Gtk.main_quit()
def on_sync_message(self, bus, msg):
if msg.get_structure().get_name() == 'prepare-window-handle':
print('prepare-window-handle')
msg.src.set_window_handle(self.xid)
def on_eos(self, bus, msg):
print('End of Service')
def on_error(self, bus, msg):
print('Error', msg.parse_error())
def on_quality_of_service(self, bus, msg):
print('Qos Message:', msg.parse_qos())
p = Player()
p.run()