8000 Merge pull request #16 from MarshalX/main · MarshalX/python-webrtc@b331293 · GitHub
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MarshalXMarshalX
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Merge pull request #16 from MarshalX/main
0.0.7 dev
2 parents 5ca9dd4 + 3277ff8 commit b331293

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13 files changed

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‎CMakeLists.txt

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@@ -7,7 +7,7 @@ project(python_webrtc
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LANGUAGES C CXX
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DESCRIPTION "a Python extension that provides bindings to WebRTC"
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HOMEPAGE_URL "https://github.com/MarshalX/python-webrtc"
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VERSION "0.0.0.6"
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VERSION "0.0.0.7"
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)
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include(ExternalProject)
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use_glib=false
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rtc_build_tools=false
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)
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if ("$ENV{TARGET_ARCH}" STREQUAL "ia32")
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list(APPEND GN_GEN_ARGS
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target_arch=ia32
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)
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endif()
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endif()
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if (WIN32)

‎Makefile

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# TODO (MarshalX) make it more pretty
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PATH_TO_WRTC_SO = ../../build/lib.macosx-12.1-arm64-3.10/wrtc.cpython-310-darwin.so
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PATH_TO_PY_MODULES = ./python-webrtc/python
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RUN_TESTS = pytest
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RUN_TESTS_OPTS ?= -vv -W ignore:::wrtc -W ignore:::pkg_resources
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TESTS_DIR = tests
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export PATH_TO_LIB=${PATH_TO_WRTC_SO}
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export PYTHONPATH=${PATH_TO_PY_MODULES}
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test:
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@${RUN_TESTS} ${TESTS_DIR} ${RUN_TESTS_OPTS} $(O)

‎README.md

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@@ -20,13 +20,13 @@
2020

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## Python WebRTC
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> Stop making unstandard pure implementations of WebRTC and let's use the native library!
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> Let's use the native WebRTC with strict compatibility and fully implemented stuff!
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This project follows the [W3C specification](https://w3c.github.io/webrtc-pc/) with some modifications and additions to make it work better with Python applications, with useful APIs like programmatic audio and video.
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## DISCLAIMER
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This project is still under development and isn't ready for any serious use! In the current stage it's possible to establish connection and work with audio with a large number of segfaults and sigbuses. ✨
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This project is still under development and isn't ready for any serious use. In the current stage, it's possible to establish connection and work with audio, but many interfaces and methods not implemented yet.
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#### Snippet
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‎build/Dockerfile

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@@ -1,14 +1,17 @@
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ARG MANYLINUX_ARCH
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# celf pathed cpython https://github.com/MarshalX/manylinux-cpython-pathes
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# self pathed cpython https://github.com/MarshalX/manylinux-cpython-pathes
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FROM quay.io/pypa_patched/manylinux_2_24_$MANYLINUX_ARCH AS builder
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RUN apt-get update && apt-get install -y xz-utils libxml2 wget build-essential
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WORKDIR /opt
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RUN wget http://ftp.mirrorservice.org/sites/sourceware.org/pub/gcc/releases/gcc-7.5.0/gcc-7.5.0.tar.gz && \
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tar zxf gcc-7.5.0.tar.gz
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WORKDIR gcc-7.5.0
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RUN ./contrib/download_prerequisites && \
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tar zxf gcc-7.5.0.tar.gz && \
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cd gcc-7.5.0 && \
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./contrib/download_prerequisites && \
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./configure --disable-multilib && \
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make -j $(nproc) && \
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make install
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make install && \
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cd .. && \
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rm -rf gcc-7.5.0 && \
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rm gcc-7.5.0.tar.gz

‎pytest.ini

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[pytest]
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asyncio_mode = strict

‎python-webrtc/cpp/src/functions/functions.cpp

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@@ -12,7 +12,7 @@
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namespace python_webrtc {
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void Functions::Init(pybind11::module &m) {
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m.def("getUserMedia", &GetUserMedia, pybind11::return_value_policy::move);
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m.def("getUserMedia", &GetUserMedia, pybind11::return_value_policy::reference);
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}
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}

‎python-webrtc/cpp/src/functions/get_user_media.cpp

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namespace python_webrtc {
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static std::unique_ptr<MediaStream> GetUserMedia() {
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static MediaStream *GetUserMedia() {
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// TODO GetUserMediaImpl should be asynced method ?
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auto factory = PeerConnectionFactory::GetOrCreateDefault();
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auto stream = factory->factory()->CreateLocalMediaStream(rtc::CreateRandomUuid());
@@ -32,7 +32,7 @@ namespace python_webrtc {
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// stream->AddTrack(track);
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}
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return std::make_unique<MediaStream>(factory, stream);
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return MediaStream::holder()->GetOrCreate(factory, stream);
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}
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} // namespace python_webrtc

‎python-webrtc/python/webrtc/base.py

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@@ -35,3 +35,9 @@ def _wrap_many(cls, items) -> List['WebRTCObject']:
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def __repr__(self):
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return f'<webrtc.{self.__class__.__name__} object at {hex(id(self))}'
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def __eq__(self, other):
40+
if isinstance(other, WebRTCObject):
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return id(self._native_obj) == id(other._native_obj)
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return super().__eq__(other)

‎requirements-dev.txt

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black
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black
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pytest-asyncio

‎tests/__init__.py

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#
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# Copyright 2022 Il`ya (Marshal) <https://github.com/MarshalX>. All rights reserved.
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#
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# Use of this source code is governed by a BSD-style license
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# that can be found in the LICENSE.md file in the root of the project.
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#

‎tests/conftest.py

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#
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# Copyright 2022 Il`ya (Marshal) <https://github.com/MarshalX>. All rights reserved.
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#
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# Use of this source code is governed by a BSD-style license
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# that can be found in the LICENSE.md file in the root of the project.
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#
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import pytest
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import webrtc
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# alias for rtc_peer_connection
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@pytest.fixture
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def pc(rtc_peer_connection):
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return rtc_peer_connection
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@pytest.fixture
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def rtc_peer_connection(request):
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pc = webrtc.RTCPeerConnection()
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def close_pc():
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pc.close()
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request.addfinalizer(close_pc)
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return pc
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def get_stream(constraints, request):
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# TODO pass constraints when cpp part will be ready
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stream = webrtc.get_user_media()
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def stop_tracks():
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for track in stream.get_tracks():
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track.stop()
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request.addfinalizer(stop_tracks)
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return stream
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@pytest.fixture
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def audio_stream(request):
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return get_stream(None, request)
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@pytest.fixture
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def video_stream(request):
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return get_stream(None, request)

‎tests/rtc_peer_connection/__init__.py

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#
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# Copyright 2022 Il`ya (Marshal) <https://github.com/MarshalX>. All rights reserved.
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#
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# Use of this source code is governed by a BSD-style license
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# that can be found in the LICENSE.md file in the root of the project.
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#

‎tests/rtc_peer_connection/test_add_track.py

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#
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# Copyright 2022 Il`ya (Marshal) <https://github.com/MarshalX>. All rights reserved.
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#
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# Use of this source code is governed by a BSD-style license
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# that can be found in the LICENSE.md file in the root of the project.
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#
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import pytest
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import webrtc
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def test_on_closed_pc(pc, audio_stream):
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track, *_ = audio_stream.get_audio_tracks()
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pc.close()
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with pytest.raises(webrtc.PythonWebRTCException):
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pc.add_track(track, audio_stream)
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def test_sender_creation(pc, audio_stream):
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track, *_ = audio_stream.get_audio_tracks()
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sender = pc.add_track(track, audio_stream)
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assert isinstance(sender, webrtc.RTCRtpSender)
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def test_adding_of_the_same_track(pc, audio_stream):
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track, *_ = audio_stream.get_audio_tracks()
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32+
pc.add_track(track, audio_stream)
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with pytest.raises(webrtc.RTCException):
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pc.add_track(track, audio_stream)
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def test_sender_and_transceivers(pc, audio_stream):
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track, *_ = audio_stream.get_audio_tracks()
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sender = pc.add_track(track, audio_stream)
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assert isinstance(sender, webrtc.RTCRtpSender)
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assert track == sender.track
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return
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# when we ready in cpp
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transceivers = pc.get_transceivers()
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assert len(transceivers) == 1
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transceiver, *_ = transceivers
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assert transceiver.sender == server
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assert [sender] == pc.get_senders()
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# assert kind and list of receivers

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