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gstwebrtc_app.py
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gstwebrtc_app.py
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# This Source Code Form is subject to the terms of the Mozilla Public
# License, v. 2.0. If a copy of the MPL was not distributed with this
# file, You can obtain one at https://mozilla.org/MPL/2.0/.
#
# This file incorporates work covered by the following copyright and
# permission notice:
#
# Copyright 2019 Google LLC
#
# Licensed under the Apache License, Version 2.0 (the "License");
# you may not use this file except in compliance with the License.
# You may obtain a copy of the License at
#
# http://www.apache.org/licenses/LICENSE-2.0
#
# Unless required by applicable law or agreed to in writing, software
# distributed under the License is distributed on an "AS IS" BASIS,
# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
# See the License for the specific language governing permissions and
# limitations under the License.
import asyncio
import base64
import json
import logging
import re
import time
import gi
gi.require_version('GLib', '2.0')
gi.require_version("Gst", "1.0")
gi.require_version('GstSdp', '1.0')
gi.require_version('GstWebRTC', '1.0')
from gi.repository import GLib, Gst, GstSdp, GstWebRTC
logger = logging.getLogger("gstwebrtc_app")
logger.setLevel(logging.INFO)
class GSTWebRTCAppError(Exception):
pass
class GSTWebRTCApp:
def __init__(self, stun_servers=None, turn_servers=None, audio_channels=2, framerate=30, encoder=None, video_bitrate=2000, audio_bitrate=64000):
"""Initialize GStreamer WebRTC app.
Initializes GObjects and checks for required plugins.
Arguments:
stun_servers {[list of string]} -- Optional STUN server uris in the form of:
stun:<host>:<port>
turn_servers {[list of strings]} -- Optional TURN server uris in the form of:
turn://<user>:<password>@<host>:<port>
"""
self.stun_servers = stun_servers
self.turn_servers = turn_servers
self.audio_channels = audio_channels
self.pipeline = None
self.webrtcbin = None
self.data_channel = None
self.encoder = encoder
self.framerate = framerate
self.video_bitrate = video_bitrate
self.audio_bitrate = audio_bitrate
# WebRTC ICE and SDP events
self.on_ice = lambda mlineindex, candidate: logger.warn(
'unhandled ice event')
self.on_sdp = lambda sdp_type, sdp: logger.warn('unhandled sdp event')
# Data channel events
self.on_data_open = lambda: logger.warn('unhandled on_data_open')
self.on_data_close = lambda: logger.warn('unhandled on_data_close')
self.on_data_error = lambda: logger.warn('unhandled on_data_error')
self.on_data_message = lambda msg: logger.warn(
'unhandled on_data_message')
Gst.init(None)
self.check_plugins()
self.ximagesrc = None
self.ximagesrc_caps = None
self.last_cursor_sent = None
def stop_ximagesrc(self):
"""Helper function to stop the ximagesrc, useful when resizing
"""
if self.ximagesrc:
self.ximagesrc.set_state(Gst.State.NULL)
def start_ximagesrc(self):
"""Helper function to start the ximagesrc, useful when resizing
"""
if self.ximagesrc:
self.ximagesrc.set_property("endx", 0)
self.ximagesrc.set_property("endy", 0)
self.ximagesrc.set_state(Gst.State.PLAYING)
# [START build_webrtcbin_pipeline]
def build_webrtcbin_pipeline(self):
"""Adds the webrtcbin elements to the pipeline.
The video and audio pipelines are linked to this in the
build_video_pipeline() and build_audio_pipeline() methods.
"""
# Create webrtcbin element named app
self.webrtcbin = Gst.ElementFactory.make("webrtcbin", "app")
# The bundle policy affects how the SDP is generated.
# This will ultimately determine how many tracks the browser receives.
# Setting this to max-compat will generate separate tracks for
# audio and video.
# See also: https://webrtcstandards.info/sdp-bundle/
self.webrtcbin.set_property("bundle-policy", "max-compat")
# Set jitterbuffer latency to the minimum possible
self.webrtcbin.set_property("latency", 1)
# Connect signal handlers
self.webrtcbin.connect(
'on-negotiation-needed', lambda webrtcbin: self.__on_negotiation_needed(webrtcbin))
self.webrtcbin.connect('on-ice-candidate', lambda webrtcbin, mlineindex,
candidate: self.__send_ice(webrtcbin, mlineindex, candidate))
# Add STUN server
# TODO: figure out how to add more than 1 stun server.
if self.stun_servers:
self.webrtcbin.set_property("stun-server", self.stun_servers[0])
# Add TURN server
if self.turn_servers:
for turn_server in self.turn_servers:
logger.info("adding TURN server: %s" % turn_server)
self.webrtcbin.emit("add-turn-server", turn_server)
# Add element to the pipeline.
self.pipeline.add(self.webrtcbin)
# [END build_webrtcbin_pipeline]
# [START build_video_pipeline]
def build_video_pipeline(self):
"""Adds the RTP video stream to the pipeline.
"""
# Create ximagesrc element named x11
# Note that when using the ximagesrc plugin, ensure that the X11 server was
# started with shared memory support: '+extension MIT-SHM' to achieve
# full frame rates.
# You can check if XSHM is in use with the following command:
# GST_DEBUG=default:5 gst-launch-1.0 ximagesrc ! fakesink num-buffers=1 2>&1 |grep -i xshm
self.ximagesrc = Gst.ElementFactory.make("ximagesrc", "x11")
ximagesrc = self.ximagesrc
# disables display of the pointer using the XFixes extension,
# common when building a remote desktop interface as the clients
# mouse pointer can be used to give the user perceived lower latency.
# This can be programmatically toggled after the pipeline is started
# for example if the user is viewing fullscreen in the browser,
# they may want to revert to seeing the remote cursor when the
# client side cursor disappears.
ximagesrc.set_property("show-pointer", 0)
# Tells GStreamer that you are using an X11 window manager or
# compositor with off-screen buffer. If you are not using a
# window manager this can be set to 0. It's also important to
# make sure that your X11 server is running with the XSHM extension
# to ensure direct memory access to frames which will reduce latency.
ximagesrc.set_property("remote", 1)
# Defines the size in bytes to read per buffer. Increasing this from
# the default of 4096 bytes helps performance when capturing high
# resolutions like 1080P, and 2K.
ximagesrc.set_property("blocksize", 16384)
# The X11 XDamage extension allows the X server to indicate when a
# regions of the screen has changed. While this can significantly
# reduce CPU usage when the screen is idle, it has little effect with
# constant motion. This can also have a negative consequences with H.264
# as the video stream can drop out and take several seconds to recover
# until a valid I-Frame is received.
# Set this to 0 for most streaming use cases.
ximagesrc.set_property("use-damage", 0)
# Create capabilities for ximagesrc
self.ximagesrc_caps = Gst.caps_from_string("video/x-raw")
# Setting the framerate=60/1 capability instructs the ximagesrc element
# to generate buffers at 60 frames per second (FPS).
# The higher the FPS, the lower the latency so this parameter is one
# way to set the overall target latency of the pipeline though keep in
# mind that the pipeline may not always perfom at the full 60 FPS.
self.ximagesrc_caps.set_value("framerate", Gst.Fraction(self.framerate, 1))
# Create a capability filter for the ximagesrc_caps
self.ximagesrc_capsfilter = Gst.ElementFactory.make("capsfilter")
self.ximagesrc_capsfilter.set_property("caps", self.ximagesrc_caps)
if self.encoder in ["nvh264enc"]:
# Upload buffers from ximagesrc directly to CUDA memory where
# the colorspace conversion will be performed.
cudaupload = Gst.ElementFactory.make("cudaupload")
# Convert the colorspace from BGRx to NVENC compatible format.
# This is performed with CUDA which reduces the overall CPU load
# compared to using the software videoconvert element.
cudaconvert = Gst.ElementFactory.make("cudaconvert")
# Convert ximagesrc BGRx format to NV12 using cudaconvert.
# This is a more compatible format for client-side software decoders.
cudaconvert_caps = Gst.caps_from_string("video/x-raw(memory:CUDAMemory)")
cudaconvert_caps.set_value("format", "NV12")
cudaconvert_capsfilter = Gst.ElementFactory.make("capsfilter")
cudaconvert_capsfilter.set_property("caps", cudaconvert_caps)
# Create the nvh264enc element named nvenc.
# This is the heart of the video pipeline that converts the raw
# frame buffers to an H.264 encoded byte-stream on the GPU.
nvh264enc = Gst.ElementFactory.make("nvh264enc", "nvenc")
# The initial bitrate of the encoder in bits per second.
# Setting this to 0 will use the bitrate from the NVENC preset.
# This parameter can be set while the pipeline is running using the
# set_video_bitrate() method. This helps to match the available
# bandwidth. If set too high, the cliend side jitter buffer will
# not be unable to lock on to the stream and it will fail to render.
nvh264enc.set_property("bitrate", self.video_bitrate)
# Rate control mode tells the encoder how to compress the frames to
# reach the target bitrate. A Constant Bit Rate (CBR) setting is best
# for streaming use cases as bitrate is the most important factor.
# A Variable Bit Rate (VBR) setting tells the encoder to adjust the
# compression level based on scene complexity, something not needed
# when streaming in real-time.
nvh264enc.set_property("rc-mode", "cbr")
# Group of Pictures (GOP) size is the distance between I-Frames that
# contain the full frame data needed to render a whole frame.
# Infinite GOP is best for streaming because it reduces the number
# of large I-Frames being transmitted. At higher resolutions, these
# I-Frames can dominate the bandwidth and add additional latency.
# With infinite GOP, you can use a higher bitrate to increase quality
# without a linear increase in total bandwidth.
# A negative consequence when using infinite GOP size is that
# when packets are lost, it may take the decoder longer to recover.
# NVENC supports infinite GOP by setting this to -1.
nvh264enc.set_property("gop-size", -1)
# Instructs encoder to handle Quality of Service (QOS) events from
# the rest of the pipeline. Setting this to true increases
# encoder stability.
nvh264enc.set_property("qos", True)
# The NVENC encoder supports a limited nubmer of encoding presets.
# These presets are different than the open x264 standard.
# The presets control the picture coding technique, bitrate,
# and encoding quality.
# The low-latency-hq is the NVENC preset reccomended for streaming.
#
# See this link for details on each preset:
# https://streamquality.report/docs/report.html#1080p60-nvenc-h264-picture-quality
nvh264enc.set_property("preset", "low-latency-hq")
# Set the capabilities for the nvh264enc element.
nvh264enc_caps = Gst.caps_from_string("video/x-h264")
# Sets the H.264 encoding profile to one compatible with WebRTC.
# The high profile is used for streaming HD video.
# Browsers only support specific H.264 profiles and they are
# coded in the RTP payload type set by the rtph264pay_caps below.
nvh264enc_caps.set_value("profile", "high")
# Create a capability filter for the nvh264enc_caps.
nvh264enc_capsfilter = Gst.ElementFactory.make("capsfilter")
nvh264enc_capsfilter.set_property("caps", nvh264enc_caps)
# Create the rtph264pay element to convert buffers into
# RTP packets that are sent over the connection transport.
rtph264pay = Gst.ElementFactory.make("rtph264pay")
# Set the capabilities for the rtph264pay element.
rtph264pay_caps = Gst.caps_from_string("application/x-rtp")
# Set the payload type to video.
rtph264pay_caps.set_value("media", "video")
# Set the video encoding name to match our encoded format.
rtph264pay_caps.set_value("encoding-name", "H264")
# Set the payload type to one that matches the encoding profile.
# Payload number 123 corresponds to H.264 encoding with the high profile.
# Other payloads can be derived using WebRTC specification:
# https://tools.ietf.org/html/rfc6184#section-8.2.1
rtph264pay_caps.set_value("payload", 123)
# Set caps that help with frame retransmits that will avoid screen freezing on packet loss.
rtph264pay_caps.set_value("rtcp-fb-nack-pli", True)
rtph264pay_caps.set_value("rtcp-fb-ccm-fir", True)
rtph264pay_caps.set_value("rtcp-fb-x-gstreamer-fir-as-repair", True)
# Set aggregate-mode to reduce RTP packetization overhead
rtph264pay_caps.set_value("aggregate-mode", "zero-latency")
# Create a capability filter for the rtph264pay_caps.
rtph264pay_capsfilter = Gst.ElementFactory.make("capsfilter")
rtph264pay_capsfilter.set_property("caps", rtph264pay_caps)
elif self.encoder in ["vah264enc"]:
# colorspace conversion
vapostproc = Gst.ElementFactory.make("vapostproc")
vapostproc_caps = Gst.caps_from_string("video/x-raw(memory:VAMemory)")
vapostproc_caps.set_value("format", "NV12")
vapostproc_capsfilter = Gst.ElementFactory.make("capsfilter")
vapostproc_capsfilter.set_property("caps", vapostproc_caps)
# encoder
vah264enc = Gst.ElementFactory.make("vah264enc", "vaenc")
if vah264enc is None:
vah264enc = Gst.ElementFactory.make("vah264lpenc", "vaenc")
vah264enc.set_property("i-frames", 0)
vah264enc.set_property("b-frames", 0)
vah264enc.set_property("key-int-max", 0)
vah264enc.set_property("bitrate", self.video_bitrate)
# capsfilter
vah264enc_caps = Gst.caps_from_string("video/x-h264")
vah264enc_caps.set_value("stream-format", "byte-stream")
vah264enc_caps.set_value("profile", "high")
vah264enc_capsfilter = Gst.ElementFactory.make("capsfilter")
vah264enc_capsfilter.set_property("caps", vah264enc_caps)
# RTP payload
rtph264pay = Gst.ElementFactory.make("rtph264pay")
rtph264pay_caps = Gst.caps_from_string("application/x-rtp")
rtph264pay_caps.set_value("media", "video")
rtph264pay_caps.set_value("encoding-name", "H264")
rtph264pay_caps.set_value("payload", 123)
rtph264pay_caps.set_value("rtcp-fb-nack-pli", True)
rtph264pay_caps.set_value("rtcp-fb-ccm-fir", True)
rtph264pay_caps.set_value("rtcp-fb-x-gstreamer-fir-as-repair", True)
rtph264pay_caps.set_value("aggregate-mode", "zero-latency")
# Create a capability filter for the rtph264pay_caps.
rtph264pay_capsfilter = Gst.ElementFactory.make("capsfilter")
rtph264pay_capsfilter.set_property("caps", rtph264pay_caps)
elif self.encoder in ["x264enc"]:
# Videoconvert for colorspace conversion
videoconvert = Gst.ElementFactory.make("videoconvert")
videoconvert_caps = Gst.caps_from_string("video/x-raw")
videoconvert_caps.set_value("format", "NV12")
videoconvert_capsfilter = Gst.ElementFactory.make("capsfilter")
videoconvert_capsfilter.set_property("caps", videoconvert_caps)
# encoder
x264enc = Gst.ElementFactory.make("x264enc", "x264enc")
x264enc.set_property("threads", 8)
x264enc.set_property("aud", False)
x264enc.set_property("b-adapt", False)
x264enc.set_property("bframes", 0)
x264enc.set_property("key-int-max", 0)
x264enc.set_property("sliced-threads", True)
x264enc.set_property("byte-stream", True)
x264enc.set_property("tune", "zerolatency")
x264enc.set_property("speed-preset", "veryfast")
x264enc.set_property("bitrate", self.video_bitrate)
# capsfilter
x264enc_caps = Gst.caps_from_string("video/x-h264")
x264enc_caps.set_value("stream-format", "byte-stream")
x264enc_caps.set_value("profile", "high")
x264enc_capsfilter = Gst.ElementFactory.make("capsfilter")
x264enc_capsfilter.set_property("caps", x264enc_caps)
# RTP payload
rtph264pay = Gst.ElementFactory.make("rtph264pay")
rtph264pay_caps = Gst.caps_from_string("application/x-rtp")
rtph264pay_caps.set_value("media", "video")
rtph264pay_caps.set_value("encoding-name", "H264")
rtph264pay_caps.set_value("payload", 123)
rtph264pay_caps.set_value("rtcp-fb-nack-pli", True)
rtph264pay_caps.set_value("rtcp-fb-ccm-fir", True)
rtph264pay_caps.set_value("rtcp-fb-x-gstreamer-fir-as-repair", True)
rtph264pay_caps.set_value("aggregate-mode", "zero-latency")
# Create a capability filter for the rtph264pay_caps.
rtph264pay_capsfilter = Gst.ElementFactory.make("capsfilter")
rtph264pay_capsfilter.set_property("caps", rtph264pay_caps)
elif self.encoder in ["vp8enc", "vp9enc"]:
videoconvert = Gst.ElementFactory.make("videoconvert")
videoconvert_caps = Gst.caps_from_string("video/x-raw")
videoconvert_caps.set_value("format", "I420")
videoconvert_capsfilter = Gst.ElementFactory.make("capsfilter")
videoconvert_capsfilter.set_property("caps", videoconvert_caps)
if self.encoder == "vp8enc":
vpenc = Gst.ElementFactory.make("vp8enc", "vpenc")
vpenc_caps = Gst.caps_from_string("video/x-vp8")
vpenc_capsfilter = Gst.ElementFactory.make("capsfilter")
vpenc_capsfilter.set_property("caps", vpenc_caps)
rtpvppay = Gst.ElementFactory.make("rtpvp8pay")
rtpvppay_caps = Gst.caps_from_string("application/x-rtp")
rtpvppay_caps.set_value("media", "video")
rtpvppay_caps.set_value("encoding-name", "VP8")
rtpvppay_caps.set_value("payload", 123)
rtpvppay_capsfilter = Gst.ElementFactory.make("capsfilter")
rtpvppay_capsfilter.set_property("caps", rtpvppay_caps)
if self.encoder == "vp9enc":
vpenc = Gst.ElementFactory.make("vp9enc", "vpenc")
vpenc_caps = Gst.caps_from_string("video/x-vp9")
vpenc_capsfilter = Gst.ElementFactory.make("capsfilter")
vpenc_capsfilter.set_property("caps", vpenc_caps)
rtpvppay = Gst.ElementFactory.make("rtpvp9pay")
rtpvppay_caps = Gst.caps_from_string("application/x-rtp")
rtpvppay_caps.set_value("media", "video")
rtpvppay_caps.set_value("encoding-name", "VP9")
rtpvppay_caps.set_value("payload", 123)
rtpvppay_capsfilter = Gst.ElementFactory.make("capsfilter")
rtpvppay_capsfilter.set_property("caps", rtpvppay_caps)
# VPX Parameters
# Borrowed from: https://github.com/nurdism/neko/blob/df98368137732b8aaf840e27cdf2bd41067b2161/server/internal/gst/gst.go#L94
vpenc.set_property("threads", 8)
vpenc.set_property("cpu-used", 8)
vpenc.set_property("deadline", 1)
vpenc.set_property("error-resilient", "partitions")
vpenc.set_property("keyframe-max-dist", 10)
vpenc.set_property("auto-alt-ref", True)
vpenc.set_property("target-bitrate", self.video_bitrate*1000)
else:
raise GSTWebRTCAppError("Unsupported encoder for pipeline: %s" % self.encoder)
# Add all elements to the pipeline.
self.pipeline.add(ximagesrc)
self.pipeline.add(self.ximagesrc_capsfilter)
if self.encoder == "nvh264enc":
self.pipeline.add(cudaupload)
self.pipeline.add(cudaconvert)
self.pipeline.add(cudaconvert_capsfilter)
self.pipeline.add(nvh264enc)
self.pipeline.add(nvh264enc_capsfilter)
self.pipeline.add(rtph264pay)
self.pipeline.add(rtph264pay_capsfilter)
elif self.encoder == "vah264enc":
self.pipeline.add(vapostproc)
self.pipeline.add(vapostproc_capsfilter)
self.pipeline.add(vah264enc)
self.pipeline.add(vah264enc_capsfilter)
self.pipeline.add(rtph264pay)
self.pipeline.add(rtph264pay_capsfilter)
elif self.encoder == "x264enc":
self.pipeline.add(videoconvert)
self.pipeline.add(videoconvert_capsfilter)
self.pipeline.add(x264enc)
self.pipeline.add(x264enc_capsfilter)
self.pipeline.add(rtph264pay)
self.pipeline.add(rtph264pay_capsfilter)
elif self.encoder.startswith("vp"):
self.pipeline.add(videoconvert)
self.pipeline.add(videoconvert_capsfilter)
self.pipeline.add(vpenc)
self.pipeline.add(vpenc_capsfilter)
self.pipeline.add(rtpvppay)
self.pipeline.add(rtpvppay_capsfilter)
# Link the pipeline elements and raise exception of linking fails
# due to incompatible element pad capabilities.
if not Gst.Element.link(ximagesrc, self.ximagesrc_capsfilter):
raise GSTWebRTCAppError("Failed to link ximagesrc -> ximagesrc_capsfilter")
if self.encoder == "nvh264enc":
if not Gst.Element.link(self.ximagesrc_capsfilter, cudaupload):
raise GSTWebRTCAppError(
"Failed to link ximagesrc_capsfilter -> cudaupload")
if not Gst.Element.link(cudaupload, cudaconvert):
raise GSTWebRTCAppError(
"Failed to link cudaupload -> cudaconvert")
if not Gst.Element.link(cudaconvert, cudaconvert_capsfilter):
raise GSTWebRTCAppError(
"Failed to link cudaconvert -> cudaconvert_capsfilter")
if not Gst.Element.link(cudaconvert_capsfilter, nvh264enc):
raise GSTWebRTCAppError(
"Failed to link cudaconvert_capsfilter -> nvh264enc")
if not Gst.Element.link(nvh264enc, nvh264enc_capsfilter):
raise GSTWebRTCAppError(
"Failed to link nvh264enc -> nvh264enc_capsfilter")
if not Gst.Element.link(nvh264enc_capsfilter, rtph264pay):
raise GSTWebRTCAppError(
"Failed to link nvh264enc_capsfilter -> rtph264pay")
if not Gst.Element.link(rtph264pay, rtph264pay_capsfilter):
raise GSTWebRTCAppError(
"Failed to link rtph264pay -> rtph264pay_capsfilter")
# Link the last element to the webrtcbin
if not Gst.Element.link(rtph264pay_capsfilter, self.webrtcbin):
raise GSTWebRTCAppError(
"Failed to link rtph264pay_capsfilter -> webrtcbin")
elif self.encoder == "vah264enc":
if not Gst.Element.link(self.ximagesrc_capsfilter, vapostproc):
raise GSTWebRTCAppError(
"Failed to link ximagesrc_capsfilter -> vapostproc")
if not Gst.Element.link(vapostproc, vapostproc_capsfilter):
raise GSTWebRTCAppError(
"Failed to link vapostproc -> vapostproc_capsfilter")
if not Gst.Element.link(vapostproc_capsfilter, vah264enc):
raise GSTWebRTCAppError(
"Failed to link vapostproc_capsfilter -> vah264enc")
if not Gst.Element.link(vah264enc, vah264enc_capsfilter):
raise GSTWebRTCAppError(
"Failed to link vah264enc -> vah264enc_capsfilter")
if not Gst.Element.link(vah264enc_capsfilter, rtph264pay):
raise GSTWebRTCAppError(
"Failed to link vah264enc_capsfilter -> rtph264pay")
if not Gst.Element.link(rtph264pay, rtph264pay_capsfilter):
raise GSTWebRTCAppError(
"Failed to link rtph264pay -> rtph264pay_capsfilter")
# Link the last element to the webrtcbin
if not Gst.Element.link(rtph264pay_capsfilter, self.webrtcbin):
raise GSTWebRTCAppError(
"Failed to link rtph264pay_capsfilter -> webrtcbin")
elif self.encoder == "x264enc":
if not Gst.Element.link(self.ximagesrc_capsfilter, videoconvert):
raise GSTWebRTCAppError(
"Failed to link ximagesrc_capsfilter -> videoconvert")
if not Gst.Element.link(videoconvert, videoconvert_capsfilter):
raise GSTWebRTCAppError(
"Failed to link videoconvert -> videoconvert_capsfilter")
if not Gst.Element.link(videoconvert_capsfilter, x264enc):
raise GSTWebRTCAppError(
"Failed to link videoconvert_capsfilter -> x264enc")
if not Gst.Element.link(x264enc, x264enc_capsfilter):
raise GSTWebRTCAppError(
"Failed to link x264enc -> x264enc_capsfilter")
if not Gst.Element.link(x264enc_capsfilter, rtph264pay):
raise GSTWebRTCAppError(
"Failed to link x264enc_capsfilter -> rtph264pay")
if not Gst.Element.link(rtph264pay, rtph264pay_capsfilter):
raise GSTWebRTCAppError(
"Failed to link rtph264pay -> rtph264pay_capsfilter")
# Link the last element to the webrtcbin
if not Gst.Element.link(rtph264pay_capsfilter, self.webrtcbin):
raise GSTWebRTCAppError(
"Failed to link rtph264pay_capsfilter -> webrtcbin")
elif self.encoder.startswith("vp"):
if not Gst.Element.link(self.ximagesrc_capsfilter, videoconvert):
raise GSTWebRTCAppError(
"Failed to link ximagesrc_capsfilter -> videoconvert")
if not Gst.Element.link(videoconvert, videoconvert_capsfilter):
raise GSTWebRTCAppError(
"Failed to link videoconvert -> videoconvert_capsfilter")
if not Gst.Element.link(videoconvert_capsfilter, vpenc):
raise GSTWebRTCAppError(
"Failed to link videoconvert_capsfilter -> vpenc")
if not Gst.Element.link(vpenc, vpenc_capsfilter):
raise GSTWebRTCAppError(
"Failed to link vpenc -> vpenc_capsfilter")
if not Gst.Element.link(vpenc_capsfilter, rtpvppay):
raise GSTWebRTCAppError(
"Failed to link vpenc_capsfilter -> rtpvppay")
if not Gst.Element.link(rtpvppay, rtpvppay_capsfilter):
raise GSTWebRTCAppError(
"Failed to link rtpvppay -> rtpvppay_capsfilter")
# Link the last element to the webrtcbin
if not Gst.Element.link(rtpvppay_capsfilter, self.webrtcbin):
raise GSTWebRTCAppError(
"Failed to link rtpvppay_capsfilter -> webrtcbin")
# [END build_video_pipeline]
# [START build_audio_pipeline]
def build_audio_pipeline(self):
"""Adds the RTP audio stream to the pipeline.
"""
# Create element for receiving audio from pulseaudio.
pulsesrc = Gst.ElementFactory.make("pulsesrc", "pulsesrc")
# Let the audio source provide the global clock.
# This is important when trying to keep the audio and video
# jitter buffers in sync. If there is skew between the video and audio
# buffers, features like NetEQ will continuously increase the size of the
# jitter buffer to catch up and will never recover.
pulsesrc.set_property("provide-clock", True)
# Create capabilities for pulsesrc and set channels
pulsesrc_caps = Gst.caps_from_string("audio/x-raw")
pulsesrc_caps.set_value("channels", self.audio_channels)
# Create a capability filter for the pulsesrc_caps
pulsesrc_capsfilter = Gst.ElementFactory.make("capsfilter")
pulsesrc_capsfilter.set_property("caps", pulsesrc_caps)
# Apply stream time to buffers, this helps with pipeline synchronization.
# Disabled by default because pulsesrc should not be re-timestamped with the current stream time when pushed out to the GStreamer pipeline and destroy the original synchronization.
# pulsesrc.set_property("do-timestamp", True)
# Encode the raw pulseaudio stream to opus format which is the
# default packetized streaming format for the web.
opusenc = Gst.ElementFactory.make("opusenc", "opusenc")
# Use full band audio bandwidth
opusenc.set_property("bandwidth", "fullband")
# Set audio bitrate to 64kbps.
# This can be dynamically changed using set_audio_bitrate()
opusenc.set_property("bitrate", self.audio_bitrate)
# Create the rtpopuspay element to convert buffers into
# RTP packets that are sent over the connection transport.
rtpopuspay = Gst.ElementFactory.make("rtpopuspay")
# Insert a queue for the RTP packets.
rtpopuspay_queue = Gst.ElementFactory.make("queue", "rtpopuspay_queue")
# Make the queue leaky in the downstream direction, drop packets if the queue is behind.
rtpopuspay_queue.set_property("leaky", "downstream")
# Discard all data in the queue when an EOS event is received
rtpopuspay_queue.set_property("flush-on-eos", True)
# Set the queue max time to 16ms (16000000ns)
# If the pipeline is behind by more than 1s, the packets
# will be dropped.
# This helps buffer out latency in the audio source.
rtpopuspay_queue.set_property("max-size-time", 16000000)
# Set the other queue sizes to 0 to make it only time-based.
rtpopuspay_queue.set_property("max-size-buffers", 0)
rtpopuspay_queue.set_property("max-size-bytes", 0)
# Set the capabilities for the rtpopuspay element.
rtpopuspay_caps = Gst.caps_from_string("application/x-rtp")
# Set the payload type to audio.
rtpopuspay_caps.set_value("media", "audio")
# Set the audio encoding name to match our encoded format.
rtpopuspay_caps.set_value("encoding-name", "OPUS")
# Set the payload type to match the encoding format.
# A value of 96 is the default that most browsers use for Opus.
# See the RFC for details:
# https://tools.ietf.org/html/rfc4566#section-6
rtpopuspay_caps.set_value("payload", 96)
# Create a capability filter for the rtpopuspay_caps.
rtpopuspay_capsfilter = Gst.ElementFactory.make("capsfilter")
rtpopuspay_capsfilter.set_property("caps", rtpopuspay_caps)
# Add all elements to the pipeline.
self.pipeline.add(pulsesrc)
self.pipeline.add(pulsesrc_capsfilter)
self.pipeline.add(opusenc)
self.pipeline.add(rtpopuspay)
self.pipeline.add(rtpopuspay_queue)
self.pipeline.add(rtpopuspay_capsfilter)
# Link the pipeline elements and raise exception of linking fails
# due to incompatible element pad capabilities.
if not Gst.Element.link(pulsesrc, pulsesrc_capsfilter):
raise GSTWebRTCAppError("Failed to link pulsesrc -> pulsesrc_capsfilter")
if not Gst.Element.link(pulsesrc_capsfilter, opusenc):
raise GSTWebRTCAppError("Failed to link pulsesrc_capsfilter -> opusenc")
if not Gst.Element.link(opusenc, rtpopuspay):
raise GSTWebRTCAppError("Failed to link opusenc -> rtpopuspay")
if not Gst.Element.link(rtpopuspay, rtpopuspay_queue):
raise GSTWebRTCAppError("Failed to link rtpopuspay -> rtpopuspay_queue")
if not Gst.Element.link(rtpopuspay_queue, rtpopuspay_capsfilter):
raise GSTWebRTCAppError(
"Failed to link rtpopuspay_queue -> rtpopuspay_capsfilter")
# Link the last element to the webrtcbin
if not Gst.Element.link(rtpopuspay_capsfilter, self.webrtcbin):
raise GSTWebRTCAppError(
"Failed to link rtpopuspay_capsfilter -> webrtcbin")
# [END build_audio_pipeline]
def check_plugins(self):
"""Check for required gstreamer plugins.
Raises:
GSTWebRTCAppError -- thrown if any plugins are missing.
"""
required = ["opus", "nice", "webrtc", "dtls", "srtp", "rtp", "sctp",
"rtpmanager", "ximagesrc"]
supported = ["nvh264enc", "vah264enc", "x264enc", "vp8enc", "vp9enc"]
if self.encoder not in supported:
raise GSTWebRTCAppError('Unsupported encoder, must be one of: ' + ','.join(supported))
if self.encoder.startswith("nv"):
required.append("nvcodec")
if self.encoder.startswith("vp"):
required.append("vpx")
missing = list(
filter(lambda p: Gst.Registry.get().find_plugin(p) is None, required))
if missing:
raise GSTWebRTCAppError('Missing gstreamer plugins:', missing)
def set_sdp(self, sdp_type, sdp):
"""Sets remote SDP received by peer.
Arguments:
sdp_type {string} -- type of sdp, offer or answer
sdp {object} -- SDP object
Raises:
GSTWebRTCAppError -- thrown if SDP is recevied before session has been started.
GSTWebRTCAppError -- thrown if SDP type is not 'answer', this script initiates the call, not the peer.
"""
if not self.webrtcbin:
raise GSTWebRTCAppError('Received SDP before session started')
if sdp_type != 'answer':
raise GSTWebRTCAppError('ERROR: sdp type was not "answer"')
_, sdpmsg = GstSdp.SDPMessage.new_from_text(sdp)
answer = GstWebRTC.WebRTCSessionDescription.new(
GstWebRTC.WebRTCSDPType.ANSWER, sdpmsg)
promise = Gst.Promise.new()
self.webrtcbin.emit('set-remote-description', answer, promise)
promise.interrupt()
def set_ice(self, mlineindex, candidate):
"""Adds ice candidate received from signalling server
Arguments:
mlineindex {integer} -- the mlineindex
candidate {string} -- the candidate
Raises:
GSTWebRTCAppError -- thrown if called before session is started.
"""
logger.info("setting ICE candidate: %d, %s" % (mlineindex, candidate))
if not self.webrtcbin:
raise GSTWebRTCAppError('Received ICE before session started')
self.webrtcbin.emit('add-ice-candidate', mlineindex, candidate)
def set_framerate(self, framerate):
"""Set pipeline framerate in fps
Arguments:
framerate {integer} -- framerate in frames per second, for example, 15, 30, 60.
"""
self.framerate = framerate
self.ximagesrc_caps = Gst.caps_from_string("video/x-raw")
self.ximagesrc_caps.set_value("framerate", Gst.Fraction(self.framerate, 1))
self.ximagesrc_capsfilter.set_property("caps", self.ximagesrc_caps)
logger.info("framerate set to: %d" % framerate)
def set_video_bitrate(self, bitrate):
"""Set video encoder target bitrate in bps
Arguments:
bitrate {integer} -- bitrate in bits per second, for example, 2000 for 2kbits/s or 10000 for 1mbit/sec.
"""
if self.encoder.startswith("nv"):
element = Gst.Bin.get_by_name(self.pipeline, "nvenc")
element.set_property("bitrate", bitrate)
elif self.encoder.startswith("va"):
element = Gst.Bin.get_by_name(self.pipeline, "vaenc")
element.set_property("bitrate", bitrate)
elif self.encoder.startswith("x264"):
element = Gst.Bin.get_by_name(self.pipeline, "x264enc")
element.set_property("bitrate", bitrate)
elif self.encoder.startswith("vp"):
element = Gst.Bin.get_by_name(self.pipeline, "vpenc")
element.set_property("target-bitrate", bitrate*1000)
else:
logger.warning("set_video_bitrate not supported with encoder: %s" % self.encoder)
logger.info("video bitrate set to: %d" % bitrate)
self.video_bitrate = bitrate
self.__send_data_channel_message(
"pipeline", {"status": "Video bitrate set to: %d" % bitrate})
def set_audio_bitrate(self, bitrate):
"""Set Opus encoder target bitrate in bps
Arguments:
bitrate {integer} -- bitrate in bits per second, for example, 96000 for 96kbits/s.
"""
if self.pipeline:
element = Gst.Bin.get_by_name(self.pipeline, "opusenc")
element.set_property("bitrate", bitrate)
logger.info("audio bitrate set to: %d" % bitrate)
self.audio_bitrate = bitrate
self.__send_data_channel_message(
"pipeline", {"status": "Audio bitrate set to: %d" % bitrate})
def set_pointer_visible(self, visible):
"""Set pointer visibiltiy on the ximagesrc element
Arguments:
visible {bool} -- True to enable pointer visibility
"""
element = Gst.Bin.get_by_name(self.pipeline, "x11")
element.set_property("show-pointer", visible)
self.__send_data_channel_message(
"pipeline", {"status": "Set pointer visibility to: %d" % visible})
def send_clipboard_data(self, data):
# TODO: WebRTC DataChannel accepts a maximum length of 65489 (= 65535 - 46 for '{"type": "clipboard", "data": {"content": ""}}'), remove this restriction after implementing DataChannel chunking
CLIPBOARD_RESTRICTION = 65400
clipboard_message = base64.b64encode(data.encode()).decode("utf-8")
clipboard_length = len(clipboard_message)
if clipboard_length <= CLIPBOARD_RESTRICTION:
self.__send_data_channel_message(
"clipboard", {"content": clipboard_message})
else:
logger.warning("clipboard may not be sent to the client because the base64 message length {} is above the maximum length of {}".format(clipboard_length, CLIPBOARD_RESTRICTION))
def send_cursor_data(self, data):
self.last_cursor_sent = data
self.__send_data_channel_message(
"cursor", data)
def send_gpu_stats(self, load, memory_total, memory_used):
"""Sends GPU stats to the data channel
Arguments:
load {float} -- utilization of GPU between 0 and 1
memory_total {float} -- total memory on GPU in MB
memory_used {float} -- memor used on GPU in MB
"""
self.__send_data_channel_message("gpu_stats", {
"load": load,
"memory_total": memory_total,
"memory_used": memory_used,
})
def send_reload_window(self):
"""Sends reload window command to the data channel
"""
logger.info("sending window reload")
self.__send_data_channel_message(
"system", {"action": "reload"})
def send_framerate(self, framerate):
"""Sends the current framerate to the data channel
"""
logger.info("sending framerate")
self.__send_data_channel_message(
"system", {"action": "framerate,"+str(framerate)})
def send_video_bitrate(self, bitrate):
"""Sends the current video bitrate to the data channel
"""
logger.info("sending video bitrate")
self.__send_data_channel_message(
"system", {"action": "video_bitrate,%d" % bitrate})
def send_audio_bitrate(self, bitrate):
"""Sends the current audio bitrate to the data channel
"""
logger.info("sending audio bitrate")
self.__send_data_channel_message(
"system", {"action": "audio_bitrate,%d" % bitrate})
def send_encoder(self, encoder):
"""Sends the encoder name to the data channel
"""
logger.info("sending encoder: " + encoder)
self.__send_data_channel_message(
"system", {"action": "encoder,%s" % encoder})
def send_resize_enabled(self, resize_enabled):
"""Sends the current resize enabled state
"""
logger.info("sending resize enabled state")
self.__send_data_channel_message(
"system", {"action": "resize,"+str(resize_enabled)})
def send_remote_resolution(self, res):
"""sends the current remote resolution to the client
"""
logger.info("sending remote resolution of: " + res)
self.__send_data_channel_message(
"system", {"action": "resolution," + res})
def send_ping(self, t):
"""Sends a ping request over the data channel to measure latency
"""
self.__send_data_channel_message(
"ping", {"start_time": float("%.3f" % t)})
def send_latency_time(self, latency):
"""Sends measured latency response time in ms
"""
self.__send_data_channel_message(
"latency_measurement", {"latency_ms": latency})
def send_system_stats(self, cpu_percent, mem_total, mem_used):
"""Sends system stats
"""
self.__send_data_channel_message(
"system_stats", {
"cpu_percent": cpu_percent,
"mem_total": mem_total,
"mem_used": mem_used,
})
def is_data_channel_ready(self):
"""Checks to see if the data channel is open.
Returns:
[bool] -- true if data channel is open
"""
return self.data_channel and self.data_channel.get_property("ready-state") == GstWebRTC.WebRTCDataChannelState.OPEN
def __send_data_channel_message(self, msg_type, data):
"""Sends message to the peer through the data channel
Message is dropped if the channel is not open.
Arguments:
msg_type {string} -- the type of message being sent
data {dict} -- data to send, this is JSON serialized.
"""
if not self.is_data_channel_ready():
logger.debug(
"skipping message because data channel is not ready: %s" % msg_type)
return
msg = {"type": msg_type, "data": data}
self.data_channel.emit("send-string", json.dumps(msg))
def __on_offer_created(self, promise, _, __):
"""Handles on-offer-created promise resolution
The offer contains the local description.
Generate a set-local-description action with the offer.
Sends the offer to the on_sdp handler.