d3289d2ec0
This is a reland of commit 9671d60925b81baefd4a0d6b05ad539fa4a782d7 after fixing more downstream dependencies Original change's description: > stats: remove RTCRtpInboundRTPStream and RTCRtpoutboundRTPStream aliases > > after upgrading downstream projects > > BUG=webrtc:14973 > > Change-Id: I5df8e95a1c70b1d6078e255166c36ed01f868b6a > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296820 > Reviewed-by: Christoffer Jansson <jansson@webrtc.org> > Reviewed-by: Henrik Boström <hbos@webrtc.org> > Commit-Queue: Philipp Hancke <phancke@microsoft.com> > Cr-Commit-Position: refs/heads/main@{#39526} No-Try: True Bug: webrtc:14973 Change-Id: I33bd99ca211a82ed77e3e8676e00256915fde168 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296881 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39533}
184 lines
6.7 KiB
C++
184 lines
6.7 KiB
C++
/*
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* Copyright 2012 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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// Integration tests for PeerConnection to exercise the options of
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// either splitting or not splitting the MediaChannel object.
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// These tests exercise a full stack over a simulated network.
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// TODO(bugs.webrtc.org/13931): Delete these tests when split is landed.
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#include <string>
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#include <vector>
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#include "absl/types/optional.h"
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#include "api/media_stream_interface.h"
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#include "api/peer_connection_interface.h"
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#include "api/rtc_error.h"
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#include "api/scoped_refptr.h"
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#include "api/stats/rtc_stats.h"
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#include "api/stats/rtc_stats_report.h"
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#include "api/stats/rtcstats_objects.h"
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#include "media/base/codec.h"
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#include "media/base/media_constants.h"
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#include "pc/session_description.h"
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#include "pc/test/integration_test_helpers.h"
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#include "rtc_base/gunit.h"
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#include "rtc_base/virtual_socket_server.h"
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#include "test/gtest.h"
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namespace webrtc {
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namespace {
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class PeerConnectionMediaChannelSplitTest
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: public PeerConnectionIntegrationBaseTest,
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public ::testing::WithParamInterface<std::string> {
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protected:
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PeerConnectionMediaChannelSplitTest()
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: PeerConnectionIntegrationBaseTest(SdpSemantics::kUnifiedPlan,
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/* field_trials = */ GetParam()) {}
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};
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int NacksReceivedCount(PeerConnectionIntegrationWrapper& pc) {
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rtc::scoped_refptr<const webrtc::RTCStatsReport> report = pc.NewGetStats();
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auto sender_stats = report->GetStatsOfType<RTCOutboundRtpStreamStats>();
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if (sender_stats.size() != 1) {
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ADD_FAILURE();
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return 0;
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}
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if (!sender_stats[0]->nack_count.is_defined()) {
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return 0;
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}
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return *sender_stats[0]->nack_count;
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}
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int NacksSentCount(PeerConnectionIntegrationWrapper& pc) {
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rtc::scoped_refptr<const webrtc::RTCStatsReport> report = pc.NewGetStats();
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auto receiver_stats = report->GetStatsOfType<RTCInboundRtpStreamStats>();
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if (receiver_stats.size() != 1) {
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ADD_FAILURE();
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return 0;
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}
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if (!receiver_stats[0]->nack_count.is_defined()) {
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return 0;
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}
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return *receiver_stats[0]->nack_count;
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}
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// Test disabled because it is flaky.
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TEST_P(PeerConnectionMediaChannelSplitTest,
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DISABLED_AudioPacketLossCausesNack) {
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RTCConfiguration config;
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ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config));
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ConnectFakeSignaling();
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auto audio_transceiver_or_error =
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caller()->pc()->AddTransceiver(caller()->CreateLocalAudioTrack());
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ASSERT_TRUE(audio_transceiver_or_error.ok());
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auto send_transceiver = audio_transceiver_or_error.MoveValue();
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// Munge the SDP to include NACK and RRTR on Opus, and remove all other
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// codecs.
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caller()->SetGeneratedSdpMunger([](cricket::SessionDescription* desc) {
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for (ContentInfo& content : desc->contents()) {
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cricket::AudioContentDescription* media =
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content.media_description()->as_audio();
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std::vector<cricket::AudioCodec> codecs = media->codecs();
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std::vector<cricket::AudioCodec> codecs_out;
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for (cricket::AudioCodec codec : codecs) {
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if (codec.name == "opus") {
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codec.AddFeedbackParam(cricket::FeedbackParam(
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cricket::kRtcpFbParamNack, cricket::kParamValueEmpty));
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codec.AddFeedbackParam(cricket::FeedbackParam(
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cricket::kRtcpFbParamRrtr, cricket::kParamValueEmpty));
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codecs_out.push_back(codec);
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}
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}
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EXPECT_FALSE(codecs_out.empty());
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media->set_codecs(codecs_out);
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}
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});
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caller()->CreateAndSetAndSignalOffer();
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// Check for failure in helpers
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ASSERT_FALSE(HasFailure());
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MediaExpectations media_expectations;
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media_expectations.CalleeExpectsSomeAudio(1);
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ExpectNewFrames(media_expectations);
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ASSERT_FALSE(HasFailure());
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virtual_socket_server()->set_drop_probability(0.2);
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// Wait until callee has sent at least one NACK.
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// Note that due to stats caching, this might only be visible 50 ms
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// after the nack was in fact sent.
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EXPECT_TRUE_WAIT(NacksSentCount(*callee()) > 0, kDefaultTimeout);
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ASSERT_FALSE(HasFailure());
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virtual_socket_server()->set_drop_probability(0.0);
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// Wait until caller has received at least one NACK
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EXPECT_TRUE_WAIT(NacksReceivedCount(*caller()) > 0, kDefaultTimeout);
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}
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TEST_P(PeerConnectionMediaChannelSplitTest, VideoPacketLossCausesNack) {
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RTCConfiguration config;
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ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config));
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ConnectFakeSignaling();
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auto video_transceiver_or_error =
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caller()->pc()->AddTransceiver(caller()->CreateLocalVideoTrack());
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ASSERT_TRUE(video_transceiver_or_error.ok());
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auto send_transceiver = video_transceiver_or_error.MoveValue();
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// Munge the SDP to include NACK and RRTR on VP8, and remove all other
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// codecs.
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caller()->SetGeneratedSdpMunger([](cricket::SessionDescription* desc) {
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for (ContentInfo& content : desc->contents()) {
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cricket::VideoContentDescription* media =
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content.media_description()->as_video();
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std::vector<cricket::VideoCodec> codecs = media->codecs();
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std::vector<cricket::VideoCodec> codecs_out;
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for (cricket::VideoCodec codec : codecs) {
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if (codec.name == "VP8") {
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ASSERT_TRUE(codec.HasFeedbackParam(cricket::FeedbackParam(
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cricket::kRtcpFbParamNack, cricket::kParamValueEmpty)));
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codecs_out.push_back(codec);
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}
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}
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EXPECT_FALSE(codecs_out.empty());
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media->set_codecs(codecs_out);
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}
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});
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caller()->CreateAndSetAndSignalOffer();
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// Check for failure in helpers
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ASSERT_FALSE(HasFailure());
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MediaExpectations media_expectations;
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media_expectations.CalleeExpectsSomeVideo(1);
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ExpectNewFrames(media_expectations);
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ASSERT_FALSE(HasFailure());
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virtual_socket_server()->set_drop_probability(0.2);
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// Wait until callee has sent at least one NACK.
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// Note that due to stats caching, this might only be visible 50 ms
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// after the nack was in fact sent.
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EXPECT_TRUE_WAIT(NacksSentCount(*callee()) > 0, kDefaultTimeout);
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ASSERT_FALSE(HasFailure());
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// Wait until caller has received at least one NACK
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EXPECT_TRUE_WAIT(NacksReceivedCount(*caller()) > 0, kDefaultTimeout);
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}
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INSTANTIATE_TEST_SUITE_P(PeerConnectionMediaChannelSplitTest,
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PeerConnectionMediaChannelSplitTest,
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Values("WebRTC-SplitMediaChannel/Disabled/",
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"WebRTC-SplitMediaChannel/Enabled/"));
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} // namespace
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} // namespace webrtc
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