audio.py

import sys, gi, os, socket, time, subprocess, psutil

from cam import Camera

gi.require_version('Gst', '1.0')
gi.require_version('GLib', '2.0')
from gi.repository import GObject, Gst, GstRtspServer, GLib, GObject

class AudioStream:
    def __init__(self, name, audio_pipeline, rtp_payloader, encoded_caps):
        self.rtp_payloader = rtp_payloader
        self.name = name

        self.audio = Gst.parse_launch(f'{audio_pipeline} ! '
                   f'interpipesink name={name} blocksize=0 max-lateness=2000000000 max-buffers=0 processing-deadline=5000000000 buffer-list=1 ')

        self.shmsrc_str = f'interpipesrc listen-to={name} is-live=1 format=time stream-sync=restart-ts blocksize=0 max-time=2000000000 leaky-type=downstream ! {encoded_caps} '

    def shutdown(self):
        # Run on driver shutdown, no need to cleanup anything here..
        set_state(Gst.State.NONE)

    def run(self):
        self.audio.get_bus().connect("message", self.snd_message)
        self.audio.get_bus().add_signal_watch()
        self.audio.set_state(Gst.State.PLAYING)

    def snd_message(self, obj, event):
        if event.src == self.audio:
            if event.type == Gst.MessageType.STATE_CHANGED:
                prev, new, pending = event.parse_state_changed()
                if pending == Gst.State.VOID_PENDING:
                    print("New audio state: %s" % new)

    def shmsrc(self):
        return self.shmsrc_str

    def payloader(self):
        return self.rtp_payloader