f0ea56a0a2
as described in https://www.rfc-editor.org/rfc/rfc8843#name-payload-type-pt-value-reuse ... all codecs associated with the payload type number MUST share an identical codec configuration See also https://github.com/w3c/webrtc-stats/issues/664 Measure how much this would break in UMA first BUG=webrtc:14420,webrtc:12716 Change-Id: Iafdc70248aa22bc37c15cc88a0c244398cb58176 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273881 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Philipp Hancke <phancke@microsoft.com> Cr-Commit-Position: refs/heads/main@{#38759}
242 lines
8.2 KiB
C++
242 lines
8.2 KiB
C++
/*
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* Copyright 2017 The WebRTC project authors. 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 file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include <memory>
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#include <utility>
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#include <vector>
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#include "api/audio/audio_mixer.h"
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#include "api/audio_codecs/builtin_audio_decoder_factory.h"
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#include "api/audio_codecs/builtin_audio_encoder_factory.h"
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#include "api/create_peerconnection_factory.h"
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#include "api/media_types.h"
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#include "api/peer_connection_interface.h"
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#include "api/rtp_transceiver_interface.h"
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#include "api/scoped_refptr.h"
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#include "api/video_codecs/builtin_video_decoder_factory.h"
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#include "api/video_codecs/builtin_video_encoder_factory.h"
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#include "modules/audio_device/include/audio_device.h"
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#include "modules/audio_processing/include/audio_processing.h"
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#include "p2p/base/port_allocator.h"
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#include "pc/peer_connection_wrapper.h"
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#include "pc/test/fake_audio_capture_module.h"
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#include "pc/test/mock_peer_connection_observers.h"
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#include "rtc_base/rtc_certificate_generator.h"
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#include "rtc_base/thread.h"
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#include "system_wrappers/include/metrics.h"
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#include "test/gtest.h"
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// This file contains unit tests that relate to the behavior of the
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// SdpOfferAnswer module.
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// Tests are writen as integration tests with PeerConnection, since the
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// behaviors are still linked so closely that it is hard to test them in
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// isolation.
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namespace webrtc {
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using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;
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namespace {
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std::unique_ptr<rtc::Thread> CreateAndStartThread() {
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auto thread = rtc::Thread::Create();
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thread->Start();
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return thread;
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}
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} // namespace
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class SdpOfferAnswerTest : public ::testing::Test {
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public:
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SdpOfferAnswerTest()
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// Note: We use a PeerConnectionFactory with a distinct
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// signaling thread, so that thread handling can be tested.
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: signaling_thread_(CreateAndStartThread()),
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pc_factory_(
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CreatePeerConnectionFactory(nullptr,
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nullptr,
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signaling_thread_.get(),
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FakeAudioCaptureModule::Create(),
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CreateBuiltinAudioEncoderFactory(),
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CreateBuiltinAudioDecoderFactory(),
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CreateBuiltinVideoEncoderFactory(),
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CreateBuiltinVideoDecoderFactory(),
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nullptr /* audio_mixer */,
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nullptr /* audio_processing */)) {
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webrtc::metrics::Reset();
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}
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std::unique_ptr<PeerConnectionWrapper> CreatePeerConnection() {
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RTCConfiguration config;
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config.sdp_semantics = SdpSemantics::kUnifiedPlan;
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return CreatePeerConnection(config);
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}
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std::unique_ptr<PeerConnectionWrapper> CreatePeerConnection(
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const RTCConfiguration& config) {
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auto observer = std::make_unique<MockPeerConnectionObserver>();
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auto result = pc_factory_->CreatePeerConnectionOrError(
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config, PeerConnectionDependencies(observer.get()));
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EXPECT_TRUE(result.ok());
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observer->SetPeerConnectionInterface(result.value().get());
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return std::make_unique<PeerConnectionWrapper>(
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pc_factory_, result.MoveValue(), std::move(observer));
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}
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protected:
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std::unique_ptr<rtc::Thread> signaling_thread_;
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rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_;
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private:
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rtc::AutoThread main_thread_;
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};
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TEST_F(SdpOfferAnswerTest, OnTrackReturnsProxiedObject) {
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auto caller = CreatePeerConnection();
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auto callee = CreatePeerConnection();
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auto audio_transceiver = caller->AddTransceiver(cricket::MEDIA_TYPE_AUDIO);
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ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
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// Verify that caller->observer->OnTrack() has been called with a
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// proxied transceiver object.
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ASSERT_EQ(callee->observer()->on_track_transceivers_.size(), 1u);
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auto transceiver = callee->observer()->on_track_transceivers_[0];
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// Since the signaling thread is not the current thread,
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// this will DCHECK if the transceiver is not proxied.
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transceiver->stopped();
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}
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TEST_F(SdpOfferAnswerTest, BundleRejectsCodecCollisionsAudioVideo) {
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auto pc = CreatePeerConnection();
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std::string sdp =
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"v=0\r\n"
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"o=- 0 3 IN IP4 127.0.0.1\r\n"
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"s=-\r\n"
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"t=0 0\r\n"
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"a=group:BUNDLE 0 1\r\n"
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"a=fingerprint:sha-1 "
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"4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB\r\n"
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"a=setup:actpass\r\n"
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"a=ice-ufrag:ETEn\r\n"
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"a=ice-pwd:OtSK0WpNtpUjkY4+86js7Z/l\r\n"
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"m=audio 9 UDP/TLS/RTP/SAVPF 111\r\n"
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"c=IN IP4 0.0.0.0\r\n"
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"a=rtcp-mux\r\n"
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"a=sendonly\r\n"
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"a=mid:0\r\n"
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"a=rtpmap:111 opus/48000/2\r\n"
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"m=video 9 UDP/TLS/RTP/SAVPF 111\r\n"
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"c=IN IP4 0.0.0.0\r\n"
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"a=rtcp-mux\r\n"
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"a=sendonly\r\n"
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"a=mid:1\r\n"
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"a=rtpmap:111 H264/90000\r\n"
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"a=fmtp:111 "
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"level-asymmetry-allowed=1;packetization-mode=0;profile-level-id="
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"42e01f\r\n";
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auto desc = CreateSessionDescription(SdpType::kOffer, sdp);
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ASSERT_NE(desc, nullptr);
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RTCError error;
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pc->SetRemoteDescription(std::move(desc), &error);
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EXPECT_TRUE(error.ok());
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EXPECT_METRIC_EQ(
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1, webrtc::metrics::NumEvents(
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"WebRTC.PeerConnection.ValidBundledPayloadTypes", false));
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}
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TEST_F(SdpOfferAnswerTest, BundleRejectsCodecCollisionsVideoFmtp) {
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auto pc = CreatePeerConnection();
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std::string sdp =
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"v=0\r\n"
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"o=- 0 3 IN IP4 127.0.0.1\r\n"
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"s=-\r\n"
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"t=0 0\r\n"
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"a=group:BUNDLE 0 1\r\n"
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"a=fingerprint:sha-1 "
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"4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB\r\n"
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"a=setup:actpass\r\n"
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"a=ice-ufrag:ETEn\r\n"
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"a=ice-pwd:OtSK0WpNtpUjkY4+86js7Z/l\r\n"
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"m=video 9 UDP/TLS/RTP/SAVPF 111\r\n"
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"c=IN IP4 0.0.0.0\r\n"
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"a=rtcp-mux\r\n"
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"a=sendonly\r\n"
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"a=mid:0\r\n"
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"a=rtpmap:111 H264/90000\r\n"
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"a=fmtp:111 "
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"level-asymmetry-allowed=1;packetization-mode=0;profile-level-id="
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"42e01f\r\n"
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"m=video 9 UDP/TLS/RTP/SAVPF 111\r\n"
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"c=IN IP4 0.0.0.0\r\n"
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"a=rtcp-mux\r\n"
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"a=sendonly\r\n"
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"a=mid:1\r\n"
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"a=rtpmap:111 H264/90000\r\n"
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"a=fmtp:111 "
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"level-asymmetry-allowed=1;packetization-mode=1;profile-level-id="
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"42e01f\r\n";
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auto desc = CreateSessionDescription(SdpType::kOffer, sdp);
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ASSERT_NE(desc, nullptr);
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RTCError error;
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pc->SetRemoteDescription(std::move(desc), &error);
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EXPECT_TRUE(error.ok());
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EXPECT_METRIC_EQ(
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1, webrtc::metrics::NumEvents(
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"WebRTC.PeerConnection.ValidBundledPayloadTypes", false));
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}
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TEST_F(SdpOfferAnswerTest, BundleCodecCollisionInDifferentBundlesAllowed) {
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auto pc = CreatePeerConnection();
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std::string sdp =
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"v=0\r\n"
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"o=- 0 3 IN IP4 127.0.0.1\r\n"
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"s=-\r\n"
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"t=0 0\r\n"
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"a=group:BUNDLE 0\r\n"
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"a=group:BUNDLE 1\r\n"
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"a=fingerprint:sha-1 "
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"4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB\r\n"
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"a=setup:actpass\r\n"
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"a=ice-ufrag:ETEn\r\n"
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"a=ice-pwd:OtSK0WpNtpUjkY4+86js7Z/l\r\n"
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"m=video 9 UDP/TLS/RTP/SAVPF 111\r\n"
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"c=IN IP4 0.0.0.0\r\n"
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"a=rtcp-mux\r\n"
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"a=sendonly\r\n"
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"a=mid:0\r\n"
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"a=rtpmap:111 H264/90000\r\n"
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"a=fmtp:111 "
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"level-asymmetry-allowed=1;packetization-mode=0;profile-level-id="
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"42e01f\r\n"
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"m=video 9 UDP/TLS/RTP/SAVPF 111\r\n"
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"c=IN IP4 0.0.0.0\r\n"
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"a=rtcp-mux\r\n"
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"a=sendonly\r\n"
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"a=mid:1\r\n"
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"a=rtpmap:111 H264/90000\r\n"
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"a=fmtp:111 "
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"level-asymmetry-allowed=1;packetization-mode=1;profile-level-id="
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"42e01f\r\n";
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auto desc = CreateSessionDescription(SdpType::kOffer, sdp);
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ASSERT_NE(desc, nullptr);
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RTCError error;
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pc->SetRemoteDescription(std::move(desc), &error);
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EXPECT_TRUE(error.ok());
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EXPECT_METRIC_EQ(
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0, webrtc::metrics::NumEvents(
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"WebRTC.PeerConnection.ValidBundledPayloadTypes", false));
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}
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} // namespace webrtc
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