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e830e683c490937d39cb6a8e765d23a8c521ecff
platform-external-webrtc/pc/peerconnectioninterface_unittest.cc
Zhi Huang e830e683c4 Use new TransportController implementation in PeerConnection. 
The TransportController will be replaced by the JsepTransportController
and JsepTransport will be replace be JsepTransport2.

The JsepTransportController will take the entire SessionDescription
and handle the RtcpMux, Sdes and BUNDLE internally.

The ownership model is also changed. The P2P layer transports are not
ref-counted and will be owned by the JsepTransport2.

In ORTC aspect, RtpTransportAdapter is now a wrapper over RtpTransport
or SrtpTransport and it implements the public and internal interface
by calling the transport underneath.

Bug: webrtc:8587
Change-Id: Ia7fa61288a566f211f8560072ea0eecaf19e48df
Reviewed-on: https://webrtc-review.googlesource.com/59586
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22693}
2018-03-30 18:41:19 +00:00

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/*
* Copyright 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include <algorithm>
#include <memory>
#include <sstream>
#include <string>
#include <utility>
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/jsepsessiondescription.h"
#include "api/mediastreaminterface.h"
#include "api/peerconnectioninterface.h"
#include "api/rtpreceiverinterface.h"
#include "api/rtpsenderinterface.h"
#include "api/test/fakeconstraints.h"
#include "logging/rtc_event_log/output/rtc_event_log_output_file.h"
#include "media/base/fakevideocapturer.h"
#include "media/engine/webrtcmediaengine.h"
#include "media/sctp/sctptransportinternal.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "p2p/base/fakeportallocator.h"
#include "pc/audiotrack.h"
#include "pc/mediasession.h"
#include "pc/mediastream.h"
#include "pc/peerconnection.h"
#include "pc/rtpsender.h"
#include "pc/streamcollection.h"
#include "pc/test/fakeaudiocapturemodule.h"
#include "pc/test/fakertccertificategenerator.h"
#include "pc/test/fakevideotracksource.h"
#include "pc/test/mockpeerconnectionobservers.h"
#include "pc/test/testsdpstrings.h"
#include "pc/videocapturertracksource.h"
#include "pc/videotrack.h"
#include "rtc_base/gunit.h"
#include "rtc_base/ptr_util.h"
#include "rtc_base/stringutils.h"
#include "rtc_base/virtualsocketserver.h"
#include "test/gmock.h"
#include "test/testsupport/fileutils.h"
#ifdef WEBRTC_ANDROID
#include "pc/test/androidtestinitializer.h"
#endif
static const char kStreamId1[] = "local_stream_1";
static const char kStreamId2[] = "local_stream_2";
static const char kStreamId3[] = "local_stream_3";
static const int kDefaultStunPort = 3478;
static const char kStunAddressOnly[] = "stun:address";
static const char kStunInvalidPort[] = "stun:address:-1";
static const char kStunAddressPortAndMore1[] = "stun:address:port:more";
static const char kStunAddressPortAndMore2[] = "stun:address:port more";
static const char kTurnIceServerUri[] = "turn:user@turn.example.org";
static const char kTurnUsername[] = "user";
static const char kTurnPassword[] = "password";
static const char kTurnHostname[] = "turn.example.org";
static const uint32_t kTimeout = 10000U;
static const char kStreams[][8] = {"stream1", "stream2"};
static const char kAudioTracks[][32] = {"audiotrack0", "audiotrack1"};
static const char kVideoTracks[][32] = {"videotrack0", "videotrack1"};
static const char kRecvonly[] = "recvonly";
static const char kSendrecv[] = "sendrecv";
// Reference SDP with a MediaStream with label "stream1" and audio track with
// id "audio_1" and a video track with id "video_1;
static const char kSdpStringWithStream1PlanB[] =
"v=0\r\n"
"o=- 0 0 IN IP4 127.0.0.1\r\n"
"s=-\r\n"
"t=0 0\r\n"
"m=audio 1 RTP/AVPF 103\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:audio\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:103 ISAC/16000\r\n"
"a=ssrc:1 cname:stream1\r\n"
"a=ssrc:1 mslabel:stream1\r\n"
"a=ssrc:1 label:audiotrack0\r\n"
"m=video 1 RTP/AVPF 120\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:video\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:120 VP8/90000\r\n"
"a=ssrc:2 cname:stream1\r\n"
"a=ssrc:2 mslabel:stream1\r\n"
"a=ssrc:2 label:videotrack0\r\n";
// Same string as above but with the MID changed to the Unified Plan default.
// This is needed so that this SDP can be used as an answer for a Unified Plan
// offer.
static const char kSdpStringWithStream1UnifiedPlan[] =
"v=0\r\n"
"o=- 0 0 IN IP4 127.0.0.1\r\n"
"s=-\r\n"
"t=0 0\r\n"
"m=audio 1 RTP/AVPF 103\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:0\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:103 ISAC/16000\r\n"
"a=ssrc:1 cname:stream1\r\n"
"a=ssrc:1 mslabel:stream1\r\n"
"a=ssrc:1 label:audiotrack0\r\n"
"m=video 1 RTP/AVPF 120\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:1\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:120 VP8/90000\r\n"
"a=ssrc:2 cname:stream1\r\n"
"a=ssrc:2 mslabel:stream1\r\n"
"a=ssrc:2 label:videotrack0\r\n";
// Reference SDP with a MediaStream with label "stream1" and audio track with
// id "audio_1";
static const char kSdpStringWithStream1AudioTrackOnly[] =
"v=0\r\n"
"o=- 0 0 IN IP4 127.0.0.1\r\n"
"s=-\r\n"
"t=0 0\r\n"
"m=audio 1 RTP/AVPF 103\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:audio\r\n"
"a=sendrecv\r\n"
"a=rtpmap:103 ISAC/16000\r\n"
"a=ssrc:1 cname:stream1\r\n"
"a=ssrc:1 mslabel:stream1\r\n"
"a=ssrc:1 label:audiotrack0\r\n"
"a=rtcp-mux\r\n";
// Reference SDP with two MediaStreams with label "stream1" and "stream2. Each
// MediaStreams have one audio track and one video track.
// This uses MSID.
static const char kSdpStringWithStream1And2PlanB[] =
"v=0\r\n"
"o=- 0 0 IN IP4 127.0.0.1\r\n"
"s=-\r\n"
"t=0 0\r\n"
"a=msid-semantic: WMS stream1 stream2\r\n"
"m=audio 1 RTP/AVPF 103\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:audio\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:103 ISAC/16000\r\n"
"a=ssrc:1 cname:stream1\r\n"
"a=ssrc:1 msid:stream1 audiotrack0\r\n"
"a=ssrc:3 cname:stream2\r\n"
"a=ssrc:3 msid:stream2 audiotrack1\r\n"
"m=video 1 RTP/AVPF 120\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:video\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:120 VP8/0\r\n"
"a=ssrc:2 cname:stream1\r\n"
"a=ssrc:2 msid:stream1 videotrack0\r\n"
"a=ssrc:4 cname:stream2\r\n"
"a=ssrc:4 msid:stream2 videotrack1\r\n";
static const char kSdpStringWithStream1And2UnifiedPlan[] =
"v=0\r\n"
"o=- 0 0 IN IP4 127.0.0.1\r\n"
"s=-\r\n"
"t=0 0\r\n"
"a=msid-semantic: WMS stream1 stream2\r\n"
"m=audio 1 RTP/AVPF 103\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:0\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:103 ISAC/16000\r\n"
"a=ssrc:1 cname:stream1\r\n"
"a=ssrc:1 msid:stream1 audiotrack0\r\n"
"m=video 1 RTP/AVPF 120\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:1\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:120 VP8/0\r\n"
"a=ssrc:2 cname:stream1\r\n"
"a=ssrc:2 msid:stream1 videotrack0\r\n"
"m=audio 1 RTP/AVPF 103\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:2\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:103 ISAC/16000\r\n"
"a=ssrc:3 cname:stream2\r\n"
"a=ssrc:3 msid:stream2 audiotrack1\r\n"
"m=video 1 RTP/AVPF 120\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:3\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:120 VP8/0\r\n"
"a=ssrc:4 cname:stream2\r\n"
"a=ssrc:4 msid:stream2 videotrack1\r\n";
// Reference SDP without MediaStreams. Msid is not supported.
static const char kSdpStringWithoutStreams[] =
"v=0\r\n"
"o=- 0 0 IN IP4 127.0.0.1\r\n"
"s=-\r\n"
"t=0 0\r\n"
"m=audio 1 RTP/AVPF 103\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:audio\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:103 ISAC/16000\r\n"
"m=video 1 RTP/AVPF 120\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:video\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:120 VP8/90000\r\n";
// Reference SDP without MediaStreams. Msid is supported.
static const char kSdpStringWithMsidWithoutStreams[] =
"v=0\r\n"
"o=- 0 0 IN IP4 127.0.0.1\r\n"
"s=-\r\n"
"t=0 0\r\n"
"a=msid-semantic: WMS\r\n"
"m=audio 1 RTP/AVPF 103\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:audio\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:103 ISAC/16000\r\n"
"m=video 1 RTP/AVPF 120\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:video\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:120 VP8/90000\r\n";
// Reference SDP without MediaStreams and audio only.
static const char kSdpStringWithoutStreamsAudioOnly[] =
"v=0\r\n"
"o=- 0 0 IN IP4 127.0.0.1\r\n"
"s=-\r\n"
"t=0 0\r\n"
"m=audio 1 RTP/AVPF 103\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:audio\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:103 ISAC/16000\r\n";
// Reference SENDONLY SDP without MediaStreams. Msid is not supported.
static const char kSdpStringSendOnlyWithoutStreams[] =
"v=0\r\n"
"o=- 0 0 IN IP4 127.0.0.1\r\n"
"s=-\r\n"
"t=0 0\r\n"
"m=audio 1 RTP/AVPF 103\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:audio\r\n"
"a=sendrecv\r\n"
"a=sendonly\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:103 ISAC/16000\r\n"
"m=video 1 RTP/AVPF 120\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:video\r\n"
"a=sendrecv\r\n"
"a=sendonly\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:120 VP8/90000\r\n";
static const char kSdpStringInit[] =
"v=0\r\n"
"o=- 0 0 IN IP4 127.0.0.1\r\n"
"s=-\r\n"
"t=0 0\r\n"
"a=msid-semantic: WMS\r\n";
static const char kSdpStringAudio[] =
"m=audio 1 RTP/AVPF 103\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:audio\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:103 ISAC/16000\r\n";
static const char kSdpStringVideo[] =
"m=video 1 RTP/AVPF 120\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:video\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:120 VP8/90000\r\n";
static const char kSdpStringMs1Audio0[] =
"a=ssrc:1 cname:stream1\r\n"
"a=ssrc:1 msid:stream1 audiotrack0\r\n";
static const char kSdpStringMs1Video0[] =
"a=ssrc:2 cname:stream1\r\n"
"a=ssrc:2 msid:stream1 videotrack0\r\n";
static const char kSdpStringMs1Audio1[] =
"a=ssrc:3 cname:stream1\r\n"
"a=ssrc:3 msid:stream1 audiotrack1\r\n";
static const char kSdpStringMs1Video1[] =
"a=ssrc:4 cname:stream1\r\n"
"a=ssrc:4 msid:stream1 videotrack1\r\n";
static const char kDtlsSdesFallbackSdp[] =
"v=0\r\n"
"o=xxxxxx 7 2 IN IP4 0.0.0.0\r\n"
"s=-\r\n"
"c=IN IP4 0.0.0.0\r\n"
"t=0 0\r\n"
"a=group:BUNDLE audio\r\n"
"a=msid-semantic: WMS\r\n"
"m=audio 1 RTP/SAVPF 0\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=mid:audio\r\n"
"a=ssrc:1 cname:stream1\r\n"
"a=ssrc:1 mslabel:stream1\r\n"
"a=ssrc:1 label:audiotrack0\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=rtpmap:0 pcmu/8000\r\n"
"a=fingerprint:sha-1 "
"4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB\r\n"
"a=setup:actpass\r\n"
"a=crypto:1 AES_CM_128_HMAC_SHA1_32 "
"inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj|2^20|1:32 "
"dummy_session_params\r\n";
using ::testing::Exactly;
using ::testing::Values;
using cricket::StreamParams;
using webrtc::AudioSourceInterface;
using webrtc::AudioTrack;
using webrtc::AudioTrackInterface;
using webrtc::DataBuffer;
using webrtc::DataChannelInterface;
using webrtc::FakeConstraints;
using webrtc::IceCandidateInterface;
using webrtc::MediaConstraintsInterface;
using webrtc::MediaStream;
using webrtc::MediaStreamInterface;
using webrtc::MediaStreamTrackInterface;
using webrtc::MockCreateSessionDescriptionObserver;
using webrtc::MockDataChannelObserver;
using webrtc::MockPeerConnectionObserver;
using webrtc::MockSetSessionDescriptionObserver;
using webrtc::MockStatsObserver;
using webrtc::NotifierInterface;
using webrtc::ObserverInterface;
using webrtc::PeerConnectionInterface;
using webrtc::PeerConnectionObserver;
using webrtc::RTCError;
using webrtc::RTCErrorType;
using webrtc::RtpReceiverInterface;
using webrtc::RtpSenderInterface;
using webrtc::RtpSenderProxyWithInternal;
using webrtc::RtpSenderInternal;
using webrtc::RtpTransceiverDirection;
using webrtc::SdpParseError;
using webrtc::SdpSemantics;
using webrtc::SdpType;
using webrtc::SessionDescriptionInterface;
using webrtc::StreamCollection;
using webrtc::StreamCollectionInterface;
using webrtc::VideoTrackSourceInterface;
using webrtc::VideoTrack;
using webrtc::VideoTrackInterface;
using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;
using RTCOfferAnswerOptions = PeerConnectionInterface::RTCOfferAnswerOptions;
namespace {
// Gets the first ssrc of given content type from the ContentInfo.
bool GetFirstSsrc(const cricket::ContentInfo* content_info, int* ssrc) {
if (!content_info || !ssrc) {
return false;
}
const cricket::MediaContentDescription* media_desc =
content_info->media_description();
if (!media_desc || media_desc->streams().empty()) {
return false;
}
*ssrc = media_desc->streams().begin()->first_ssrc();
return true;
}
// Get the ufrags out of an SDP blob. Useful for testing ICE restart
// behavior.
std::vector<std::string> GetUfrags(
const webrtc::SessionDescriptionInterface* desc) {
std::vector<std::string> ufrags;
for (const cricket::TransportInfo& info :
desc->description()->transport_infos()) {
ufrags.push_back(info.description.ice_ufrag);
}
return ufrags;
}
void SetSsrcToZero(std::string* sdp) {
const char kSdpSsrcAtribute[] = "a=ssrc:";
const char kSdpSsrcAtributeZero[] = "a=ssrc:0";
size_t ssrc_pos = 0;
while ((ssrc_pos = sdp->find(kSdpSsrcAtribute, ssrc_pos)) !=
std::string::npos) {
size_t end_ssrc = sdp->find(" ", ssrc_pos);
sdp->replace(ssrc_pos, end_ssrc - ssrc_pos, kSdpSsrcAtributeZero);
ssrc_pos = end_ssrc;
}
}
// Check if |streams| contains the specified track.
bool ContainsTrack(const std::vector<cricket::StreamParams>& streams,
const std::string& stream_id,
const std::string& track_id) {
for (const cricket::StreamParams& params : streams) {
if (params.first_stream_id() == stream_id && params.id == track_id) {
return true;
}
}
return false;
}
// Check if |senders| contains the specified sender, by id.
bool ContainsSender(
const std::vector<rtc::scoped_refptr<RtpSenderInterface>>& senders,
const std::string& id) {
for (const auto& sender : senders) {
if (sender->id() == id) {
return true;
}
}
return false;
}
// Check if |senders| contains the specified sender, by id and stream id.
bool ContainsSender(
const std::vector<rtc::scoped_refptr<RtpSenderInterface>>& senders,
const std::string& id,
const std::string& stream_id) {
for (const auto& sender : senders) {
if (sender->id() == id && sender->stream_ids()[0] == stream_id) {
return true;
}
}
return false;
}
// Create a collection of streams.
// CreateStreamCollection(1) creates a collection that
// correspond to kSdpStringWithStream1.
// CreateStreamCollection(2) correspond to kSdpStringWithStream1And2.
rtc::scoped_refptr<StreamCollection> CreateStreamCollection(
int number_of_streams,
int tracks_per_stream) {
rtc::scoped_refptr<StreamCollection> local_collection(
StreamCollection::Create());
for (int i = 0; i < number_of_streams; ++i) {
rtc::scoped_refptr<webrtc::MediaStreamInterface> stream(
webrtc::MediaStream::Create(kStreams[i]));
for (int j = 0; j < tracks_per_stream; ++j) {
// Add a local audio track.
rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
webrtc::AudioTrack::Create(kAudioTracks[i * tracks_per_stream + j],
nullptr));
stream->AddTrack(audio_track);
// Add a local video track.
rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
webrtc::VideoTrack::Create(kVideoTracks[i * tracks_per_stream + j],
webrtc::FakeVideoTrackSource::Create(),
rtc::Thread::Current()));
stream->AddTrack(video_track);
}
local_collection->AddStream(stream);
}
return local_collection;
}
// Check equality of StreamCollections.
bool CompareStreamCollections(StreamCollectionInterface* s1,
StreamCollectionInterface* s2) {
if (s1 == nullptr || s2 == nullptr || s1->count() != s2->count()) {
return false;
}
for (size_t i = 0; i != s1->count(); ++i) {
if (s1->at(i)->id() != s2->at(i)->id()) {
return false;
}
webrtc::AudioTrackVector audio_tracks1 = s1->at(i)->GetAudioTracks();
webrtc::AudioTrackVector audio_tracks2 = s2->at(i)->GetAudioTracks();
webrtc::VideoTrackVector video_tracks1 = s1->at(i)->GetVideoTracks();
webrtc::VideoTrackVector video_tracks2 = s2->at(i)->GetVideoTracks();
if (audio_tracks1.size() != audio_tracks2.size()) {
return false;
}
for (size_t j = 0; j != audio_tracks1.size(); ++j) {
if (audio_tracks1[j]->id() != audio_tracks2[j]->id()) {
return false;
}
}
if (video_tracks1.size() != video_tracks2.size()) {
return false;
}
for (size_t j = 0; j != video_tracks1.size(); ++j) {
if (video_tracks1[j]->id() != video_tracks2[j]->id()) {
return false;
}
}
}
return true;
}
// Helper class to test Observer.
class MockTrackObserver : public ObserverInterface {
public:
explicit MockTrackObserver(NotifierInterface* notifier)
: notifier_(notifier) {
notifier_->RegisterObserver(this);
}
~MockTrackObserver() { Unregister(); }
void Unregister() {
if (notifier_) {
notifier_->UnregisterObserver(this);
notifier_ = nullptr;
}
}
MOCK_METHOD0(OnChanged, void());
private:
NotifierInterface* notifier_;
};
} // namespace
// The PeerConnectionMediaConfig tests below verify that configuration and
// constraints are propagated into the PeerConnection's MediaConfig. These
// settings are intended for MediaChannel constructors, but that is not
// exercised by these unittest.
class PeerConnectionFactoryForTest : public webrtc::PeerConnectionFactory {
public:
static rtc::scoped_refptr<PeerConnectionFactoryForTest>
CreatePeerConnectionFactoryForTest() {
auto audio_encoder_factory = webrtc::CreateBuiltinAudioEncoderFactory();
auto audio_decoder_factory = webrtc::CreateBuiltinAudioDecoderFactory();
// Use fake audio device module since we're only testing the interface
// level, and using a real one could make tests flaky when run in parallel.
auto media_engine = std::unique_ptr<cricket::MediaEngineInterface>(
cricket::WebRtcMediaEngineFactory::Create(
FakeAudioCaptureModule::Create(), audio_encoder_factory,
audio_decoder_factory, nullptr, nullptr, nullptr,
webrtc::AudioProcessingBuilder().Create()));
std::unique_ptr<webrtc::CallFactoryInterface> call_factory =
webrtc::CreateCallFactory();
std::unique_ptr<webrtc::RtcEventLogFactoryInterface> event_log_factory =
webrtc::CreateRtcEventLogFactory();
return new rtc::RefCountedObject<PeerConnectionFactoryForTest>(
rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(),
std::move(media_engine), std::move(call_factory),
std::move(event_log_factory));
}
PeerConnectionFactoryForTest(
rtc::Thread* network_thread,
rtc::Thread* worker_thread,
rtc::Thread* signaling_thread,
std::unique_ptr<cricket::MediaEngineInterface> media_engine,
std::unique_ptr<webrtc::CallFactoryInterface> call_factory,
std::unique_ptr<webrtc::RtcEventLogFactoryInterface> event_log_factory)
: webrtc::PeerConnectionFactory(network_thread,
worker_thread,
signaling_thread,
std::move(media_engine),
std::move(call_factory),
std::move(event_log_factory)) {}
rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
};
// TODO(steveanton): Convert to use the new PeerConnectionWrapper.
class PeerConnectionInterfaceBaseTest : public testing::Test {
protected:
explicit PeerConnectionInterfaceBaseTest(SdpSemantics sdp_semantics)
: vss_(new rtc::VirtualSocketServer()),
main_(vss_.get()),
sdp_semantics_(sdp_semantics) {
#ifdef WEBRTC_ANDROID
webrtc::InitializeAndroidObjects();
#endif
}
virtual void SetUp() {
// Use fake audio capture module since we're only testing the interface
// level, and using a real one could make tests flaky when run in parallel.
fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
pc_factory_ = webrtc::CreatePeerConnectionFactory(
rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(),
fake_audio_capture_module_, webrtc::CreateBuiltinAudioEncoderFactory(),
webrtc::CreateBuiltinAudioDecoderFactory(), nullptr, nullptr);
ASSERT_TRUE(pc_factory_);
pc_factory_for_test_ =
PeerConnectionFactoryForTest::CreatePeerConnectionFactoryForTest();
pc_factory_for_test_->Initialize();
}
void CreatePeerConnection() {
CreatePeerConnection(PeerConnectionInterface::RTCConfiguration(), nullptr);
}
// DTLS does not work in a loopback call, so is disabled for most of the
// tests in this file.
void CreatePeerConnectionWithoutDtls() {
FakeConstraints no_dtls_constraints;
no_dtls_constraints.AddMandatory(
webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, false);
CreatePeerConnection(PeerConnectionInterface::RTCConfiguration(),
&no_dtls_constraints);
}
void CreatePeerConnection(webrtc::MediaConstraintsInterface* constraints) {
CreatePeerConnection(PeerConnectionInterface::RTCConfiguration(),
constraints);
}
void CreatePeerConnectionWithIceTransportsType(
PeerConnectionInterface::IceTransportsType type) {
PeerConnectionInterface::RTCConfiguration config;
config.type = type;
return CreatePeerConnection(config, nullptr);
}
void CreatePeerConnectionWithIceServer(const std::string& uri,
const std::string& password) {
PeerConnectionInterface::RTCConfiguration config;
PeerConnectionInterface::IceServer server;
server.uri = uri;
server.password = password;
config.servers.push_back(server);
CreatePeerConnection(config, nullptr);
}
void CreatePeerConnection(
const PeerConnectionInterface::RTCConfiguration& config,
webrtc::MediaConstraintsInterface* constraints) {
std::unique_ptr<cricket::FakePortAllocator> port_allocator(
new cricket::FakePortAllocator(rtc::Thread::Current(), nullptr));
port_allocator_ = port_allocator.get();
// Create certificate generator unless DTLS constraint is explicitly set to
// false.
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator;
bool dtls;
if (FindConstraint(constraints,
webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
&dtls,
nullptr) && dtls) {
fake_certificate_generator_ = new FakeRTCCertificateGenerator();
cert_generator.reset(fake_certificate_generator_);
}
RTCConfiguration modified_config = config;
modified_config.sdp_semantics = sdp_semantics_;
pc_ = pc_factory_->CreatePeerConnection(
modified_config, constraints, std::move(port_allocator),
std::move(cert_generator), &observer_);
ASSERT_TRUE(pc_.get() != NULL);
observer_.SetPeerConnectionInterface(pc_.get());
EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_);
}
void CreatePeerConnectionExpectFail(const std::string& uri) {
PeerConnectionInterface::RTCConfiguration config;
PeerConnectionInterface::IceServer server;
server.uri = uri;
config.servers.push_back(server);
config.sdp_semantics = sdp_semantics_;
rtc::scoped_refptr<PeerConnectionInterface> pc =
pc_factory_->CreatePeerConnection(config, nullptr, nullptr, nullptr,
&observer_);
EXPECT_EQ(nullptr, pc);
}
void CreatePeerConnectionExpectFail(
PeerConnectionInterface::RTCConfiguration config) {
PeerConnectionInterface::IceServer server;
server.uri = kTurnIceServerUri;
server.password = kTurnPassword;
config.servers.push_back(server);
config.sdp_semantics = sdp_semantics_;
rtc::scoped_refptr<PeerConnectionInterface> pc =
pc_factory_->CreatePeerConnection(config, nullptr, nullptr, &observer_);
EXPECT_EQ(nullptr, pc);
}
void CreatePeerConnectionWithDifferentConfigurations() {
CreatePeerConnectionWithIceServer(kStunAddressOnly, "");
EXPECT_EQ(1u, port_allocator_->stun_servers().size());
EXPECT_EQ(0u, port_allocator_->turn_servers().size());
EXPECT_EQ("address", port_allocator_->stun_servers().begin()->hostname());
EXPECT_EQ(kDefaultStunPort,
port_allocator_->stun_servers().begin()->port());
CreatePeerConnectionExpectFail(kStunInvalidPort);
CreatePeerConnectionExpectFail(kStunAddressPortAndMore1);
CreatePeerConnectionExpectFail(kStunAddressPortAndMore2);
CreatePeerConnectionWithIceServer(kTurnIceServerUri, kTurnPassword);
EXPECT_EQ(0u, port_allocator_->stun_servers().size());
EXPECT_EQ(1u, port_allocator_->turn_servers().size());
EXPECT_EQ(kTurnUsername,
port_allocator_->turn_servers()[0].credentials.username);
EXPECT_EQ(kTurnPassword,
port_allocator_->turn_servers()[0].credentials.password);
EXPECT_EQ(kTurnHostname,
port_allocator_->turn_servers()[0].ports[0].address.hostname());
}
void ReleasePeerConnection() {
pc_ = NULL;
observer_.SetPeerConnectionInterface(NULL);
}
rtc::scoped_refptr<VideoTrackInterface> CreateVideoTrack(
const std::string& label) {
auto video_source = pc_factory_->CreateVideoSource(
rtc::MakeUnique<cricket::FakeVideoCapturer>(), nullptr);
return pc_factory_->CreateVideoTrack(label, video_source);
}
void AddVideoTrack(const std::string& track_label,
const std::vector<std::string>& stream_ids = {}) {
auto sender_or_error =
pc_->AddTrack(CreateVideoTrack(track_label), stream_ids);
ASSERT_EQ(RTCErrorType::NONE, sender_or_error.error().type());
EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
observer_.renegotiation_needed_ = false;
}
void AddVideoStream(const std::string& label) {
rtc::scoped_refptr<MediaStreamInterface> stream(
pc_factory_->CreateLocalMediaStream(label));
stream->AddTrack(CreateVideoTrack(label + "v0"));
ASSERT_TRUE(pc_->AddStream(stream));
EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
observer_.renegotiation_needed_ = false;
}
rtc::scoped_refptr<AudioTrackInterface> CreateAudioTrack(
const std::string& label) {
return pc_factory_->CreateAudioTrack(label, nullptr);
}
void AddAudioTrack(const std::string& track_label,
const std::vector<std::string>& stream_ids = {}) {
auto sender_or_error =
pc_->AddTrack(CreateAudioTrack(track_label), stream_ids);
ASSERT_EQ(RTCErrorType::NONE, sender_or_error.error().type());
EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
observer_.renegotiation_needed_ = false;
}
void AddAudioStream(const std::string& label) {
rtc::scoped_refptr<MediaStreamInterface> stream(
pc_factory_->CreateLocalMediaStream(label));
stream->AddTrack(CreateAudioTrack(label + "a0"));
ASSERT_TRUE(pc_->AddStream(stream));
EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
observer_.renegotiation_needed_ = false;
}
void AddAudioVideoStream(const std::string& stream_id,
const std::string& audio_track_label,
const std::string& video_track_label) {
// Create a local stream.
rtc::scoped_refptr<MediaStreamInterface> stream(
pc_factory_->CreateLocalMediaStream(stream_id));
stream->AddTrack(CreateAudioTrack(audio_track_label));
stream->AddTrack(CreateVideoTrack(video_track_label));
ASSERT_TRUE(pc_->AddStream(stream));
EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
observer_.renegotiation_needed_ = false;
}
rtc::scoped_refptr<RtpReceiverInterface> GetFirstReceiverOfType(
cricket::MediaType media_type) {
for (auto receiver : pc_->GetReceivers()) {
if (receiver->media_type() == media_type) {
return receiver;
}
}
return nullptr;
}
bool DoCreateOfferAnswer(std::unique_ptr<SessionDescriptionInterface>* desc,
bool offer,
MediaConstraintsInterface* constraints) {
rtc::scoped_refptr<MockCreateSessionDescriptionObserver>
observer(new rtc::RefCountedObject<
MockCreateSessionDescriptionObserver>());
if (offer) {
pc_->CreateOffer(observer, constraints);
} else {
pc_->CreateAnswer(observer, constraints);
}
EXPECT_EQ_WAIT(true, observer->called(), kTimeout);
*desc = observer->MoveDescription();
return observer->result();
}
bool DoCreateOffer(std::unique_ptr<SessionDescriptionInterface>* desc,
MediaConstraintsInterface* constraints) {
return DoCreateOfferAnswer(desc, true, constraints);
}
bool DoCreateAnswer(std::unique_ptr<SessionDescriptionInterface>* desc,
MediaConstraintsInterface* constraints) {
return DoCreateOfferAnswer(desc, false, constraints);
}
bool DoSetSessionDescription(
std::unique_ptr<SessionDescriptionInterface> desc,
bool local) {
rtc::scoped_refptr<MockSetSessionDescriptionObserver>
observer(new rtc::RefCountedObject<
MockSetSessionDescriptionObserver>());
if (local) {
pc_->SetLocalDescription(observer, desc.release());
} else {
pc_->SetRemoteDescription(observer, desc.release());
}
if (pc_->signaling_state() != PeerConnectionInterface::kClosed) {
EXPECT_EQ_WAIT(true, observer->called(), kTimeout);
}
return observer->result();
}
bool DoSetLocalDescription(
std::unique_ptr<SessionDescriptionInterface> desc) {
return DoSetSessionDescription(std::move(desc), true);
}
bool DoSetRemoteDescription(
std::unique_ptr<SessionDescriptionInterface> desc) {
return DoSetSessionDescription(std::move(desc), false);
}
// Calls PeerConnection::GetStats and check the return value.
// It does not verify the values in the StatReports since a RTCP packet might
// be required.
bool DoGetStats(MediaStreamTrackInterface* track) {
rtc::scoped_refptr<MockStatsObserver> observer(
new rtc::RefCountedObject<MockStatsObserver>());
if (!pc_->GetStats(
observer, track, PeerConnectionInterface::kStatsOutputLevelStandard))
return false;
EXPECT_TRUE_WAIT(observer->called(), kTimeout);
return observer->called();
}
// Call the standards-compliant GetStats function.
bool DoGetRTCStats() {
rtc::scoped_refptr<webrtc::MockRTCStatsCollectorCallback> callback(
new rtc::RefCountedObject<webrtc::MockRTCStatsCollectorCallback>());
pc_->GetStats(callback);
EXPECT_TRUE_WAIT(callback->called(), kTimeout);
return callback->called();
}
void InitiateCall() {
CreatePeerConnectionWithoutDtls();
// Create a local stream with audio&video tracks.
if (sdp_semantics_ == SdpSemantics::kPlanB) {
AddAudioVideoStream(kStreamId1, "audio_track", "video_track");
} else {
// Unified Plan does not support AddStream, so just add an audio and video
// track.
AddAudioTrack(kAudioTracks[0], {kStreamId1});
AddVideoTrack(kVideoTracks[0], {kStreamId1});
}
CreateOfferReceiveAnswer();
}
// Verify that RTP Header extensions has been negotiated for audio and video.
void VerifyRemoteRtpHeaderExtensions() {
const cricket::MediaContentDescription* desc =
cricket::GetFirstAudioContentDescription(
pc_->remote_description()->description());
ASSERT_TRUE(desc != NULL);
EXPECT_GT(desc->rtp_header_extensions().size(), 0u);
desc = cricket::GetFirstVideoContentDescription(
pc_->remote_description()->description());
ASSERT_TRUE(desc != NULL);
EXPECT_GT(desc->rtp_header_extensions().size(), 0u);
}
void CreateOfferAsRemoteDescription() {
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
std::string sdp;
EXPECT_TRUE(offer->ToString(&sdp));
std::unique_ptr<SessionDescriptionInterface> remote_offer(
webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_offer)));
EXPECT_EQ(PeerConnectionInterface::kHaveRemoteOffer, observer_.state_);
}
void CreateAndSetRemoteOffer(const std::string& sdp) {
std::unique_ptr<SessionDescriptionInterface> remote_offer(
webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_offer)));
EXPECT_EQ(PeerConnectionInterface::kHaveRemoteOffer, observer_.state_);
}
void CreateAnswerAsLocalDescription() {
std::unique_ptr<SessionDescriptionInterface> answer;
ASSERT_TRUE(DoCreateAnswer(&answer, nullptr));
// TODO(perkj): Currently SetLocalDescription fails if any parameters in an
// audio codec change, even if the parameter has nothing to do with
// receiving. Not all parameters are serialized to SDP.
// Since CreatePrAnswerAsLocalDescription serialize/deserialize
// the SessionDescription, it is necessary to do that here to in order to
// get ReceiveOfferCreatePrAnswerAndAnswer and RenegotiateAudioOnly to pass.
// https://code.google.com/p/webrtc/issues/detail?id=1356
std::string sdp;
EXPECT_TRUE(answer->ToString(&sdp));
std::unique_ptr<SessionDescriptionInterface> new_answer(
webrtc::CreateSessionDescription(SdpType::kAnswer, sdp));
EXPECT_TRUE(DoSetLocalDescription(std::move(new_answer)));
EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_);
}
void CreatePrAnswerAsLocalDescription() {
std::unique_ptr<SessionDescriptionInterface> answer;
ASSERT_TRUE(DoCreateAnswer(&answer, nullptr));
std::string sdp;
EXPECT_TRUE(answer->ToString(&sdp));
std::unique_ptr<SessionDescriptionInterface> pr_answer(
webrtc::CreateSessionDescription(SdpType::kPrAnswer, sdp));
EXPECT_TRUE(DoSetLocalDescription(std::move(pr_answer)));
EXPECT_EQ(PeerConnectionInterface::kHaveLocalPrAnswer, observer_.state_);
}
void CreateOfferReceiveAnswer() {
CreateOfferAsLocalDescription();
std::string sdp;
EXPECT_TRUE(pc_->local_description()->ToString(&sdp));
CreateAnswerAsRemoteDescription(sdp);
}
void CreateOfferAsLocalDescription() {
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
// TODO(perkj): Currently SetLocalDescription fails if any parameters in an
// audio codec change, even if the parameter has nothing to do with
// receiving. Not all parameters are serialized to SDP.
// Since CreatePrAnswerAsLocalDescription serialize/deserialize
// the SessionDescription, it is necessary to do that here to in order to
// get ReceiveOfferCreatePrAnswerAndAnswer and RenegotiateAudioOnly to pass.
// https://code.google.com/p/webrtc/issues/detail?id=1356
std::string sdp;
EXPECT_TRUE(offer->ToString(&sdp));
std::unique_ptr<SessionDescriptionInterface> new_offer(
webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
EXPECT_TRUE(DoSetLocalDescription(std::move(new_offer)));
EXPECT_EQ(PeerConnectionInterface::kHaveLocalOffer, observer_.state_);
// Wait for the ice_complete message, so that SDP will have candidates.
EXPECT_TRUE_WAIT(observer_.ice_gathering_complete_, kTimeout);
}
void CreateAnswerAsRemoteDescription(const std::string& sdp) {
std::unique_ptr<SessionDescriptionInterface> answer(
webrtc::CreateSessionDescription(SdpType::kAnswer, sdp));
ASSERT_TRUE(answer);
EXPECT_TRUE(DoSetRemoteDescription(std::move(answer)));
EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_);
}
void CreatePrAnswerAndAnswerAsRemoteDescription(const std::string& sdp) {
std::unique_ptr<SessionDescriptionInterface> pr_answer(
webrtc::CreateSessionDescription(SdpType::kPrAnswer, sdp));
ASSERT_TRUE(pr_answer);
EXPECT_TRUE(DoSetRemoteDescription(std::move(pr_answer)));
EXPECT_EQ(PeerConnectionInterface::kHaveRemotePrAnswer, observer_.state_);
std::unique_ptr<SessionDescriptionInterface> answer(
webrtc::CreateSessionDescription(SdpType::kAnswer, sdp));
ASSERT_TRUE(answer);
EXPECT_TRUE(DoSetRemoteDescription(std::move(answer)));
EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_);
}
// Waits until a remote stream with the given id is signaled. This helper
// function will verify both OnAddTrack and OnAddStream (Plan B only) are
// called with the given stream id and expected number of tracks.
void WaitAndVerifyOnAddStream(const std::string& stream_id,
int expected_num_tracks) {
// Verify that both OnAddStream and OnAddTrack are called.
EXPECT_EQ_WAIT(stream_id, observer_.GetLastAddedStreamId(), kTimeout);
EXPECT_EQ_WAIT(expected_num_tracks,
observer_.CountAddTrackEventsForStream(stream_id), kTimeout);
}
// Creates an offer and applies it as a local session description.
// Creates an answer with the same SDP an the offer but removes all lines
// that start with a:ssrc"
void CreateOfferReceiveAnswerWithoutSsrc() {
CreateOfferAsLocalDescription();
std::string sdp;
EXPECT_TRUE(pc_->local_description()->ToString(&sdp));
SetSsrcToZero(&sdp);
CreateAnswerAsRemoteDescription(sdp);
}
// This function creates a MediaStream with label kStreams[0] and
// |number_of_audio_tracks| and |number_of_video_tracks| tracks and the
// corresponding SessionDescriptionInterface. The SessionDescriptionInterface
// is returned and the MediaStream is stored in
// |reference_collection_|
std::unique_ptr<SessionDescriptionInterface>
CreateSessionDescriptionAndReference(size_t number_of_audio_tracks,
size_t number_of_video_tracks) {
EXPECT_LE(number_of_audio_tracks, 2u);
EXPECT_LE(number_of_video_tracks, 2u);
reference_collection_ = StreamCollection::Create();
std::string sdp_ms1 = std::string(kSdpStringInit);
std::string mediastream_id = kStreams[0];
rtc::scoped_refptr<webrtc::MediaStreamInterface> stream(
webrtc::MediaStream::Create(mediastream_id));
reference_collection_->AddStream(stream);
if (number_of_audio_tracks > 0) {
sdp_ms1 += std::string(kSdpStringAudio);
sdp_ms1 += std::string(kSdpStringMs1Audio0);
AddAudioTrack(kAudioTracks[0], stream);
}
if (number_of_audio_tracks > 1) {
sdp_ms1 += kSdpStringMs1Audio1;
AddAudioTrack(kAudioTracks[1], stream);
}
if (number_of_video_tracks > 0) {
sdp_ms1 += std::string(kSdpStringVideo);
sdp_ms1 += std::string(kSdpStringMs1Video0);
AddVideoTrack(kVideoTracks[0], stream);
}
if (number_of_video_tracks > 1) {
sdp_ms1 += kSdpStringMs1Video1;
AddVideoTrack(kVideoTracks[1], stream);
}
return std::unique_ptr<SessionDescriptionInterface>(
webrtc::CreateSessionDescription(SdpType::kOffer, sdp_ms1));
}
void AddAudioTrack(const std::string& track_id,
MediaStreamInterface* stream) {
rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
webrtc::AudioTrack::Create(track_id, nullptr));
ASSERT_TRUE(stream->AddTrack(audio_track));
}
void AddVideoTrack(const std::string& track_id,
MediaStreamInterface* stream) {
rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
webrtc::VideoTrack::Create(track_id,
webrtc::FakeVideoTrackSource::Create(),
rtc::Thread::Current()));
ASSERT_TRUE(stream->AddTrack(video_track));
}
std::unique_ptr<SessionDescriptionInterface> CreateOfferWithOneAudioTrack() {
CreatePeerConnectionWithoutDtls();
AddAudioTrack(kAudioTracks[0]);
std::unique_ptr<SessionDescriptionInterface> offer;
EXPECT_TRUE(DoCreateOffer(&offer, nullptr));
return offer;
}
std::unique_ptr<SessionDescriptionInterface> CreateOfferWithOneAudioStream() {
CreatePeerConnectionWithoutDtls();
AddAudioStream(kStreamId1);
std::unique_ptr<SessionDescriptionInterface> offer;
EXPECT_TRUE(DoCreateOffer(&offer, nullptr));
return offer;
}
std::unique_ptr<SessionDescriptionInterface> CreateAnswerWithOneAudioTrack() {
EXPECT_TRUE(DoSetRemoteDescription(CreateOfferWithOneAudioTrack()));
std::unique_ptr<SessionDescriptionInterface> answer;
EXPECT_TRUE(DoCreateAnswer(&answer, nullptr));
return answer;
}
std::unique_ptr<SessionDescriptionInterface>
CreateAnswerWithOneAudioStream() {
EXPECT_TRUE(DoSetRemoteDescription(CreateOfferWithOneAudioStream()));
std::unique_ptr<SessionDescriptionInterface> answer;
EXPECT_TRUE(DoCreateAnswer(&answer, nullptr));
return answer;
}
const std::string& GetFirstAudioStreamCname(
const SessionDescriptionInterface* desc) {
const cricket::AudioContentDescription* audio_desc =
cricket::GetFirstAudioContentDescription(desc->description());
return audio_desc->streams()[0].cname;
}
std::unique_ptr<SessionDescriptionInterface> CreateOfferWithOptions(
const RTCOfferAnswerOptions& offer_answer_options) {
RTC_DCHECK(pc_);
rtc::scoped_refptr<MockCreateSessionDescriptionObserver> observer(
new rtc::RefCountedObject<MockCreateSessionDescriptionObserver>());
pc_->CreateOffer(observer, offer_answer_options);
EXPECT_EQ_WAIT(true, observer->called(), kTimeout);
return observer->MoveDescription();
}
void CreateOfferWithOptionsAsRemoteDescription(
std::unique_ptr<SessionDescriptionInterface>* desc,
const RTCOfferAnswerOptions& offer_answer_options) {
*desc = CreateOfferWithOptions(offer_answer_options);
ASSERT_TRUE(desc != nullptr);
std::string sdp;
EXPECT_TRUE((*desc)->ToString(&sdp));
std::unique_ptr<SessionDescriptionInterface> remote_offer(
webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_offer)));
EXPECT_EQ(PeerConnectionInterface::kHaveRemoteOffer, observer_.state_);
}
void CreateOfferWithOptionsAsLocalDescription(
std::unique_ptr<SessionDescriptionInterface>* desc,
const RTCOfferAnswerOptions& offer_answer_options) {
*desc = CreateOfferWithOptions(offer_answer_options);
ASSERT_TRUE(desc != nullptr);
std::string sdp;
EXPECT_TRUE((*desc)->ToString(&sdp));
std::unique_ptr<SessionDescriptionInterface> new_offer(
webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
EXPECT_TRUE(DoSetLocalDescription(std::move(new_offer)));
EXPECT_EQ(PeerConnectionInterface::kHaveLocalOffer, observer_.state_);
}
bool HasCNCodecs(const cricket::ContentInfo* content) {
RTC_DCHECK(content);
RTC_DCHECK(content->media_description());
for (const cricket::AudioCodec& codec :
content->media_description()->as_audio()->codecs()) {
if (codec.name == "CN") {
return true;
}
}
return false;
}
const char* GetSdpStringWithStream1() const {
if (sdp_semantics_ == SdpSemantics::kPlanB) {
return kSdpStringWithStream1PlanB;
} else {
return kSdpStringWithStream1UnifiedPlan;
}
}
const char* GetSdpStringWithStream1And2() const {
if (sdp_semantics_ == SdpSemantics::kPlanB) {
return kSdpStringWithStream1And2PlanB;
} else {
return kSdpStringWithStream1And2UnifiedPlan;
}
}
std::unique_ptr<rtc::VirtualSocketServer> vss_;
rtc::AutoSocketServerThread main_;
rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
cricket::FakePortAllocator* port_allocator_ = nullptr;
FakeRTCCertificateGenerator* fake_certificate_generator_ = nullptr;
rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory_;
rtc::scoped_refptr<PeerConnectionFactoryForTest> pc_factory_for_test_;
rtc::scoped_refptr<PeerConnectionInterface> pc_;
MockPeerConnectionObserver observer_;
rtc::scoped_refptr<StreamCollection> reference_collection_;
const SdpSemantics sdp_semantics_;
};
class PeerConnectionInterfaceTest
: public PeerConnectionInterfaceBaseTest,
public ::testing::WithParamInterface<SdpSemantics> {
protected:
PeerConnectionInterfaceTest() : PeerConnectionInterfaceBaseTest(GetParam()) {}
};
class PeerConnectionInterfaceTestPlanB
: public PeerConnectionInterfaceBaseTest {
protected:
PeerConnectionInterfaceTestPlanB()
: PeerConnectionInterfaceBaseTest(SdpSemantics::kPlanB) {}
};
// Generate different CNAMEs when PeerConnections are created.
// The CNAMEs are expected to be generated randomly. It is possible
// that the test fails, though the possibility is very low.
TEST_P(PeerConnectionInterfaceTest, CnameGenerationInOffer) {
std::unique_ptr<SessionDescriptionInterface> offer1 =
CreateOfferWithOneAudioTrack();
std::unique_ptr<SessionDescriptionInterface> offer2 =
CreateOfferWithOneAudioTrack();
EXPECT_NE(GetFirstAudioStreamCname(offer1.get()),
GetFirstAudioStreamCname(offer2.get()));
}
TEST_P(PeerConnectionInterfaceTest, CnameGenerationInAnswer) {
std::unique_ptr<SessionDescriptionInterface> answer1 =
CreateAnswerWithOneAudioTrack();
std::unique_ptr<SessionDescriptionInterface> answer2 =
CreateAnswerWithOneAudioTrack();
EXPECT_NE(GetFirstAudioStreamCname(answer1.get()),
GetFirstAudioStreamCname(answer2.get()));
}
TEST_P(PeerConnectionInterfaceTest,
CreatePeerConnectionWithDifferentConfigurations) {
CreatePeerConnectionWithDifferentConfigurations();
}
TEST_P(PeerConnectionInterfaceTest,
CreatePeerConnectionWithDifferentIceTransportsTypes) {
CreatePeerConnectionWithIceTransportsType(PeerConnectionInterface::kNone);
EXPECT_EQ(cricket::CF_NONE, port_allocator_->candidate_filter());
CreatePeerConnectionWithIceTransportsType(PeerConnectionInterface::kRelay);
EXPECT_EQ(cricket::CF_RELAY, port_allocator_->candidate_filter());
CreatePeerConnectionWithIceTransportsType(PeerConnectionInterface::kNoHost);
EXPECT_EQ(cricket::CF_ALL & ~cricket::CF_HOST,
port_allocator_->candidate_filter());
CreatePeerConnectionWithIceTransportsType(PeerConnectionInterface::kAll);
EXPECT_EQ(cricket::CF_ALL, port_allocator_->candidate_filter());
}
// Test that when a PeerConnection is created with a nonzero candidate pool
// size, the pooled PortAllocatorSession is created with all the attributes
// in the RTCConfiguration.
TEST_P(PeerConnectionInterfaceTest, CreatePeerConnectionWithPooledCandidates) {
PeerConnectionInterface::RTCConfiguration config;
PeerConnectionInterface::IceServer server;
server.uri = kStunAddressOnly;
config.servers.push_back(server);
config.type = PeerConnectionInterface::kRelay;
config.disable_ipv6 = true;
config.tcp_candidate_policy =
PeerConnectionInterface::kTcpCandidatePolicyDisabled;
config.candidate_network_policy =
PeerConnectionInterface::kCandidateNetworkPolicyLowCost;
config.ice_candidate_pool_size = 1;
CreatePeerConnection(config, nullptr);
const cricket::FakePortAllocatorSession* session =
static_cast<const cricket::FakePortAllocatorSession*>(
port_allocator_->GetPooledSession());
ASSERT_NE(nullptr, session);
EXPECT_EQ(1UL, session->stun_servers().size());
EXPECT_EQ(0U, session->flags() & cricket::PORTALLOCATOR_ENABLE_IPV6);
EXPECT_LT(0U, session->flags() & cricket::PORTALLOCATOR_DISABLE_TCP);
EXPECT_LT(0U,
session->flags() & cricket::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS);
}
// Test that network-related RTCConfiguration members are applied to the
// PortAllocator when CreatePeerConnection is called. Specifically:
// - disable_ipv6_on_wifi
// - max_ipv6_networks
// - tcp_candidate_policy
// - candidate_network_policy
// - prune_turn_ports
//
// Note that the candidate filter (RTCConfiguration::type) is already tested
// above.
TEST_P(PeerConnectionInterfaceTest,
CreatePeerConnectionAppliesNetworkConfigToPortAllocator) {
// Create fake port allocator.
std::unique_ptr<cricket::FakePortAllocator> port_allocator(
new cricket::FakePortAllocator(rtc::Thread::Current(), nullptr));
cricket::FakePortAllocator* raw_port_allocator = port_allocator.get();
// Create RTCConfiguration with some network-related fields relevant to
// PortAllocator populated.
PeerConnectionInterface::RTCConfiguration config;
config.disable_ipv6_on_wifi = true;
config.max_ipv6_networks = 10;
config.tcp_candidate_policy =
PeerConnectionInterface::kTcpCandidatePolicyDisabled;
config.candidate_network_policy =
PeerConnectionInterface::kCandidateNetworkPolicyLowCost;
config.prune_turn_ports = true;
// Create the PC factory and PC with the above config.
rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory(
webrtc::CreatePeerConnectionFactory(
rtc::Thread::Current(), rtc::Thread::Current(),
rtc::Thread::Current(), fake_audio_capture_module_,
webrtc::CreateBuiltinAudioEncoderFactory(),
webrtc::CreateBuiltinAudioDecoderFactory(), nullptr, nullptr));
rtc::scoped_refptr<PeerConnectionInterface> pc(
pc_factory->CreatePeerConnection(
config, nullptr, std::move(port_allocator), nullptr, &observer_));
// Now validate that the config fields set above were applied to the
// PortAllocator, as flags or otherwise.
EXPECT_FALSE(raw_port_allocator->flags() &
cricket::PORTALLOCATOR_ENABLE_IPV6_ON_WIFI);
EXPECT_EQ(10, raw_port_allocator->max_ipv6_networks());
EXPECT_TRUE(raw_port_allocator->flags() & cricket::PORTALLOCATOR_DISABLE_TCP);
EXPECT_TRUE(raw_port_allocator->flags() &
cricket::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS);
EXPECT_TRUE(raw_port_allocator->prune_turn_ports());
}
// Test that the PeerConnection initializes the port allocator passed into it,
// and on the correct thread.
TEST_P(PeerConnectionInterfaceTest,
CreatePeerConnectionInitializesPortAllocatorOnNetworkThread) {
std::unique_ptr<rtc::Thread> network_thread(
rtc::Thread::CreateWithSocketServer());
network_thread->Start();
rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory(
webrtc::CreatePeerConnectionFactory(
network_thread.get(), rtc::Thread::Current(), rtc::Thread::Current(),
fake_audio_capture_module_,
webrtc::CreateBuiltinAudioEncoderFactory(),
webrtc::CreateBuiltinAudioDecoderFactory(), nullptr, nullptr));
std::unique_ptr<cricket::FakePortAllocator> port_allocator(
new cricket::FakePortAllocator(network_thread.get(), nullptr));
cricket::FakePortAllocator* raw_port_allocator = port_allocator.get();
PeerConnectionInterface::RTCConfiguration config;
rtc::scoped_refptr<PeerConnectionInterface> pc(
pc_factory->CreatePeerConnection(
config, nullptr, std::move(port_allocator), nullptr, &observer_));
// FakePortAllocator RTC_CHECKs that it's initialized on the right thread,
// so all we have to do here is check that it's initialized.
EXPECT_TRUE(raw_port_allocator->initialized());
}
// Check that GetConfiguration returns the configuration the PeerConnection was
// constructed with, before SetConfiguration is called.
TEST_P(PeerConnectionInterfaceTest, GetConfigurationAfterCreatePeerConnection) {
PeerConnectionInterface::RTCConfiguration config;
config.type = PeerConnectionInterface::kRelay;
CreatePeerConnection(config, nullptr);
PeerConnectionInterface::RTCConfiguration returned_config =
pc_->GetConfiguration();
EXPECT_EQ(PeerConnectionInterface::kRelay, returned_config.type);
}
// Check that GetConfiguration returns the last configuration passed into
// SetConfiguration.
TEST_P(PeerConnectionInterfaceTest, GetConfigurationAfterSetConfiguration) {
CreatePeerConnection();
PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration();
config.type = PeerConnectionInterface::kRelay;
EXPECT_TRUE(pc_->SetConfiguration(config));
PeerConnectionInterface::RTCConfiguration returned_config =
pc_->GetConfiguration();
EXPECT_EQ(PeerConnectionInterface::kRelay, returned_config.type);
}
TEST_F(PeerConnectionInterfaceTestPlanB, AddStreams) {
CreatePeerConnectionWithoutDtls();
AddVideoStream(kStreamId1);
AddAudioStream(kStreamId2);
ASSERT_EQ(2u, pc_->local_streams()->count());
// Test we can add multiple local streams to one peerconnection.
rtc::scoped_refptr<MediaStreamInterface> stream(
pc_factory_->CreateLocalMediaStream(kStreamId3));
rtc::scoped_refptr<AudioTrackInterface> audio_track(
pc_factory_->CreateAudioTrack(kStreamId3,
static_cast<AudioSourceInterface*>(NULL)));
stream->AddTrack(audio_track.get());
EXPECT_TRUE(pc_->AddStream(stream));
EXPECT_EQ(3u, pc_->local_streams()->count());
// Remove the third stream.
pc_->RemoveStream(pc_->local_streams()->at(2));
EXPECT_EQ(2u, pc_->local_streams()->count());
// Remove the second stream.
pc_->RemoveStream(pc_->local_streams()->at(1));
EXPECT_EQ(1u, pc_->local_streams()->count());
// Remove the first stream.
pc_->RemoveStream(pc_->local_streams()->at(0));
EXPECT_EQ(0u, pc_->local_streams()->count());
}
// Test that the created offer includes streams we added.
// Don't run under Unified Plan since the stream API is not available.
TEST_F(PeerConnectionInterfaceTestPlanB, AddedStreamsPresentInOffer) {
CreatePeerConnectionWithoutDtls();
AddAudioVideoStream(kStreamId1, "audio_track", "video_track");
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
const cricket::AudioContentDescription* audio_desc =
cricket::GetFirstAudioContentDescription(offer->description());
EXPECT_TRUE(ContainsTrack(audio_desc->streams(), kStreamId1, "audio_track"));
const cricket::VideoContentDescription* video_desc =
cricket::GetFirstVideoContentDescription(offer->description());
EXPECT_TRUE(ContainsTrack(video_desc->streams(), kStreamId1, "video_track"));
// Add another stream and ensure the offer includes both the old and new
// streams.
AddAudioVideoStream(kStreamId2, "audio_track2", "video_track2");
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
audio_desc = cricket::GetFirstAudioContentDescription(offer->description());
EXPECT_TRUE(ContainsTrack(audio_desc->streams(), kStreamId1, "audio_track"));
EXPECT_TRUE(ContainsTrack(audio_desc->streams(), kStreamId2, "audio_track2"));
video_desc = cricket::GetFirstVideoContentDescription(offer->description());
EXPECT_TRUE(ContainsTrack(video_desc->streams(), kStreamId1, "video_track"));
EXPECT_TRUE(ContainsTrack(video_desc->streams(), kStreamId2, "video_track2"));
}
// Don't run under Unified Plan since the stream API is not available.
TEST_F(PeerConnectionInterfaceTestPlanB, RemoveStream) {
CreatePeerConnectionWithoutDtls();
AddVideoStream(kStreamId1);
ASSERT_EQ(1u, pc_->local_streams()->count());
pc_->RemoveStream(pc_->local_streams()->at(0));
EXPECT_EQ(0u, pc_->local_streams()->count());
}
// Test for AddTrack and RemoveTrack methods.
// Tests that the created offer includes tracks we added,
// and that the RtpSenders are created correctly.
// Also tests that RemoveTrack removes the tracks from subsequent offers.
// Only tested with Plan B since Unified Plan is covered in more detail by tests
// in peerconnection_jsep_unittests.cc
TEST_F(PeerConnectionInterfaceTestPlanB, AddTrackRemoveTrack) {
CreatePeerConnectionWithoutDtls();
rtc::scoped_refptr<AudioTrackInterface> audio_track(
pc_factory_->CreateAudioTrack("audio_track", nullptr));
rtc::scoped_refptr<VideoTrackInterface> video_track(
pc_factory_->CreateVideoTrack(
"video_track", pc_factory_->CreateVideoSource(
std::unique_ptr<cricket::VideoCapturer>(
new cricket::FakeVideoCapturer()))));
auto audio_sender = pc_->AddTrack(audio_track, {kStreamId1}).MoveValue();
auto video_sender = pc_->AddTrack(video_track, {kStreamId1}).MoveValue();
EXPECT_EQ(1UL, audio_sender->stream_ids().size());
EXPECT_EQ(kStreamId1, audio_sender->stream_ids()[0]);
EXPECT_EQ("audio_track", audio_sender->id());
EXPECT_EQ(audio_track, audio_sender->track());
EXPECT_EQ(1UL, video_sender->stream_ids().size());
EXPECT_EQ(kStreamId1, video_sender->stream_ids()[0]);
EXPECT_EQ("video_track", video_sender->id());
EXPECT_EQ(video_track, video_sender->track());
// Now create an offer and check for the senders.
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
const cricket::ContentInfo* audio_content =
cricket::GetFirstAudioContent(offer->description());
EXPECT_TRUE(ContainsTrack(audio_content->media_description()->streams(),
kStreamId1, "audio_track"));
const cricket::ContentInfo* video_content =
cricket::GetFirstVideoContent(offer->description());
EXPECT_TRUE(ContainsTrack(video_content->media_description()->streams(),
kStreamId1, "video_track"));
EXPECT_TRUE(DoSetLocalDescription(std::move(offer)));
// Now try removing the tracks.
EXPECT_TRUE(pc_->RemoveTrack(audio_sender));
EXPECT_TRUE(pc_->RemoveTrack(video_sender));
// Create a new offer and ensure it doesn't contain the removed senders.
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
audio_content = cricket::GetFirstAudioContent(offer->description());
EXPECT_FALSE(ContainsTrack(audio_content->media_description()->streams(),
kStreamId1, "audio_track"));
video_content = cricket::GetFirstVideoContent(offer->description());
EXPECT_FALSE(ContainsTrack(video_content->media_description()->streams(),
kStreamId1, "video_track"));
EXPECT_TRUE(DoSetLocalDescription(std::move(offer)));
// Calling RemoveTrack on a sender no longer attached to a PeerConnection
// should return false.
EXPECT_FALSE(pc_->RemoveTrack(audio_sender));
EXPECT_FALSE(pc_->RemoveTrack(video_sender));
}
// Test creating senders without a stream specified,
// expecting a random stream ID to be generated.
TEST_P(PeerConnectionInterfaceTest, AddTrackWithoutStream) {
CreatePeerConnectionWithoutDtls();
rtc::scoped_refptr<AudioTrackInterface> audio_track(
pc_factory_->CreateAudioTrack("audio_track", nullptr));
rtc::scoped_refptr<VideoTrackInterface> video_track(
pc_factory_->CreateVideoTrack(
"video_track", pc_factory_->CreateVideoSource(
std::unique_ptr<cricket::VideoCapturer>(
new cricket::FakeVideoCapturer()))));
auto audio_sender =
pc_->AddTrack(audio_track, std::vector<std::string>()).MoveValue();
auto video_sender =
pc_->AddTrack(video_track, std::vector<std::string>()).MoveValue();
EXPECT_EQ("audio_track", audio_sender->id());
EXPECT_EQ(audio_track, audio_sender->track());
EXPECT_EQ("video_track", video_sender->id());
EXPECT_EQ(video_track, video_sender->track());
// If the ID is truly a random GUID, it should be infinitely unlikely they
// will be the same.
EXPECT_NE(video_sender->stream_ids(), audio_sender->stream_ids());
}
// Test that we can call GetStats() after AddTrack but before connecting
// the PeerConnection to a peer.
TEST_P(PeerConnectionInterfaceTest, AddTrackBeforeConnecting) {
CreatePeerConnectionWithoutDtls();
rtc::scoped_refptr<AudioTrackInterface> audio_track(
pc_factory_->CreateAudioTrack("audio_track", nullptr));
rtc::scoped_refptr<VideoTrackInterface> video_track(
pc_factory_->CreateVideoTrack(
"video_track", pc_factory_->CreateVideoSource(
std::unique_ptr<cricket::VideoCapturer>(
new cricket::FakeVideoCapturer()))));
auto audio_sender = pc_->AddTrack(audio_track, std::vector<std::string>());
auto video_sender = pc_->AddTrack(video_track, std::vector<std::string>());
EXPECT_TRUE(DoGetStats(nullptr));
}
TEST_P(PeerConnectionInterfaceTest, AttachmentIdIsSetOnAddTrack) {
CreatePeerConnectionWithoutDtls();
rtc::scoped_refptr<AudioTrackInterface> audio_track(
pc_factory_->CreateAudioTrack("audio_track", nullptr));
rtc::scoped_refptr<VideoTrackInterface> video_track(
pc_factory_->CreateVideoTrack(
"video_track", pc_factory_->CreateVideoSource(
std::unique_ptr<cricket::VideoCapturer>(
new cricket::FakeVideoCapturer()))));
auto audio_sender = pc_->AddTrack(audio_track, std::vector<std::string>());
ASSERT_TRUE(audio_sender.ok());
auto* audio_sender_proxy =
static_cast<RtpSenderProxyWithInternal<RtpSenderInternal>*>(
audio_sender.value().get());
EXPECT_NE(0, audio_sender_proxy->internal()->AttachmentId());
auto video_sender = pc_->AddTrack(video_track, std::vector<std::string>());
ASSERT_TRUE(video_sender.ok());
auto* video_sender_proxy =
static_cast<RtpSenderProxyWithInternal<RtpSenderInternal>*>(
video_sender.value().get());
EXPECT_NE(0, video_sender_proxy->internal()->AttachmentId());
}
// Don't run under Unified Plan since the stream API is not available.
TEST_F(PeerConnectionInterfaceTestPlanB, AttachmentIdIsSetOnAddStream) {
CreatePeerConnectionWithoutDtls();
AddVideoStream(kStreamId1);
auto senders = pc_->GetSenders();
ASSERT_EQ(1u, senders.size());
auto* sender_proxy =
static_cast<RtpSenderProxyWithInternal<RtpSenderInternal>*>(
senders[0].get());
EXPECT_NE(0, sender_proxy->internal()->AttachmentId());
}
TEST_P(PeerConnectionInterfaceTest, CreateOfferReceiveAnswer) {
InitiateCall();
WaitAndVerifyOnAddStream(kStreamId1, 2);
VerifyRemoteRtpHeaderExtensions();
}
TEST_P(PeerConnectionInterfaceTest, CreateOfferReceivePrAnswerAndAnswer) {
CreatePeerConnectionWithoutDtls();
AddVideoTrack(kVideoTracks[0], {kStreamId1});
CreateOfferAsLocalDescription();
std::string offer;
EXPECT_TRUE(pc_->local_description()->ToString(&offer));
CreatePrAnswerAndAnswerAsRemoteDescription(offer);
WaitAndVerifyOnAddStream(kStreamId1, 1);
}
TEST_P(PeerConnectionInterfaceTest, ReceiveOfferCreateAnswer) {
CreatePeerConnectionWithoutDtls();
AddVideoTrack(kVideoTracks[0], {kStreamId1});
CreateOfferAsRemoteDescription();
CreateAnswerAsLocalDescription();
WaitAndVerifyOnAddStream(kStreamId1, 1);
}
TEST_P(PeerConnectionInterfaceTest, ReceiveOfferCreatePrAnswerAndAnswer) {
CreatePeerConnectionWithoutDtls();
AddVideoTrack(kVideoTracks[0], {kStreamId1});
CreateOfferAsRemoteDescription();
CreatePrAnswerAsLocalDescription();
CreateAnswerAsLocalDescription();
WaitAndVerifyOnAddStream(kStreamId1, 1);
}
// Don't run under Unified Plan since the stream API is not available.
TEST_F(PeerConnectionInterfaceTestPlanB, Renegotiate) {
InitiateCall();
ASSERT_EQ(1u, pc_->remote_streams()->count());
pc_->RemoveStream(pc_->local_streams()->at(0));
CreateOfferReceiveAnswer();
EXPECT_EQ(0u, pc_->remote_streams()->count());
AddVideoStream(kStreamId1);
CreateOfferReceiveAnswer();
}
// Tests that after negotiating an audio only call, the respondent can perform a
// renegotiation that removes the audio stream.
TEST_F(PeerConnectionInterfaceTestPlanB, RenegotiateAudioOnly) {
CreatePeerConnectionWithoutDtls();
AddAudioStream(kStreamId1);
CreateOfferAsRemoteDescription();
CreateAnswerAsLocalDescription();
ASSERT_EQ(1u, pc_->remote_streams()->count());
pc_->RemoveStream(pc_->local_streams()->at(0));
CreateOfferReceiveAnswer();
EXPECT_EQ(0u, pc_->remote_streams()->count());
}
// Test that candidates are generated and that we can parse our own candidates.
TEST_P(PeerConnectionInterfaceTest, IceCandidates) {
CreatePeerConnectionWithoutDtls();
EXPECT_FALSE(pc_->AddIceCandidate(observer_.last_candidate()));
// SetRemoteDescription takes ownership of offer.
std::unique_ptr<SessionDescriptionInterface> offer;
AddVideoTrack(kVideoTracks[0]);
EXPECT_TRUE(DoCreateOffer(&offer, nullptr));
EXPECT_TRUE(DoSetRemoteDescription(std::move(offer)));
// SetLocalDescription takes ownership of answer.
std::unique_ptr<SessionDescriptionInterface> answer;
EXPECT_TRUE(DoCreateAnswer(&answer, nullptr));
EXPECT_TRUE(DoSetLocalDescription(std::move(answer)));
EXPECT_TRUE_WAIT(observer_.last_candidate() != nullptr, kTimeout);
EXPECT_TRUE_WAIT(observer_.ice_gathering_complete_, kTimeout);
EXPECT_TRUE(pc_->AddIceCandidate(observer_.last_candidate()));
}
// Test that CreateOffer and CreateAnswer will fail if the track labels are
// not unique.
TEST_P(PeerConnectionInterfaceTest, CreateOfferAnswerWithInvalidStream) {
CreatePeerConnectionWithoutDtls();
// Create a regular offer for the CreateAnswer test later.
std::unique_ptr<SessionDescriptionInterface> offer;
EXPECT_TRUE(DoCreateOffer(&offer, nullptr));
EXPECT_TRUE(offer);
offer.reset();
// Create a local stream with audio&video tracks having same label.
AddAudioTrack("track_label", {kStreamId1});
AddVideoTrack("track_label", {kStreamId1});
// Test CreateOffer
EXPECT_FALSE(DoCreateOffer(&offer, nullptr));
// Test CreateAnswer
std::unique_ptr<SessionDescriptionInterface> answer;
EXPECT_FALSE(DoCreateAnswer(&answer, nullptr));
}
// Test that we will get different SSRCs for each tracks in the offer and answer
// we created.
TEST_P(PeerConnectionInterfaceTest, SsrcInOfferAnswer) {
CreatePeerConnectionWithoutDtls();
// Create a local stream with audio&video tracks having different labels.
AddAudioTrack(kAudioTracks[0], {kStreamId1});
AddVideoTrack(kVideoTracks[0], {kStreamId1});
// Test CreateOffer
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
int audio_ssrc = 0;
int video_ssrc = 0;
EXPECT_TRUE(GetFirstSsrc(GetFirstAudioContent(offer->description()),
&audio_ssrc));
EXPECT_TRUE(GetFirstSsrc(GetFirstVideoContent(offer->description()),
&video_ssrc));
EXPECT_NE(audio_ssrc, video_ssrc);
// Test CreateAnswer
EXPECT_TRUE(DoSetRemoteDescription(std::move(offer)));
std::unique_ptr<SessionDescriptionInterface> answer;
ASSERT_TRUE(DoCreateAnswer(&answer, nullptr));
audio_ssrc = 0;
video_ssrc = 0;
EXPECT_TRUE(GetFirstSsrc(GetFirstAudioContent(answer->description()),
&audio_ssrc));
EXPECT_TRUE(GetFirstSsrc(GetFirstVideoContent(answer->description()),
&video_ssrc));
EXPECT_NE(audio_ssrc, video_ssrc);
}
// Test that it's possible to call AddTrack on a MediaStream after adding
// the stream to a PeerConnection.
// TODO(deadbeef): Remove this test once this behavior is no longer supported.
// Don't run under Unified Plan since the stream API is not available.
TEST_F(PeerConnectionInterfaceTestPlanB, AddTrackAfterAddStream) {
CreatePeerConnectionWithoutDtls();
// Create audio stream and add to PeerConnection.
AddAudioStream(kStreamId1);
MediaStreamInterface* stream = pc_->local_streams()->at(0);
// Add video track to the audio-only stream.
rtc::scoped_refptr<VideoTrackInterface> video_track(
pc_factory_->CreateVideoTrack(
"video_label", pc_factory_->CreateVideoSource(
std::unique_ptr<cricket::VideoCapturer>(
new cricket::FakeVideoCapturer()))));
stream->AddTrack(video_track.get());
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
const cricket::MediaContentDescription* video_desc =
cricket::GetFirstVideoContentDescription(offer->description());
EXPECT_TRUE(video_desc != nullptr);
}
// Test that it's possible to call RemoveTrack on a MediaStream after adding
// the stream to a PeerConnection.
// TODO(deadbeef): Remove this test once this behavior is no longer supported.
// Don't run under Unified Plan since the stream API is not available.
TEST_F(PeerConnectionInterfaceTestPlanB, RemoveTrackAfterAddStream) {
CreatePeerConnectionWithoutDtls();
// Create audio/video stream and add to PeerConnection.
AddAudioVideoStream(kStreamId1, "audio_label", "video_label");
MediaStreamInterface* stream = pc_->local_streams()->at(0);
// Remove the video track.
stream->RemoveTrack(stream->GetVideoTracks()[0]);
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
const cricket::MediaContentDescription* video_desc =
cricket::GetFirstVideoContentDescription(offer->description());
EXPECT_TRUE(video_desc == nullptr);
}
// Verify that CreateDtmfSender only succeeds if called with a valid local
// track. Other aspects of DtmfSenders are tested in
// peerconnection_integrationtest.cc.
TEST_P(PeerConnectionInterfaceTest, CreateDtmfSenderWithInvalidParams) {
CreatePeerConnection();
AddAudioTrack(kAudioTracks[0]);
EXPECT_EQ(nullptr, pc_->CreateDtmfSender(nullptr));
rtc::scoped_refptr<webrtc::AudioTrackInterface> non_localtrack(
pc_factory_->CreateAudioTrack("dummy_track", nullptr));
EXPECT_EQ(nullptr, pc_->CreateDtmfSender(non_localtrack));
}
// Test creating a sender with a stream ID, and ensure the ID is populated
// in the offer.
// Don't run under Unified Plan since the stream API is not available.
TEST_F(PeerConnectionInterfaceTestPlanB, CreateSenderWithStream) {
CreatePeerConnectionWithoutDtls();
pc_->CreateSender("video", kStreamId1);
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
const cricket::MediaContentDescription* video_desc =
cricket::GetFirstVideoContentDescription(offer->description());
ASSERT_TRUE(video_desc != nullptr);
ASSERT_EQ(1u, video_desc->streams().size());
EXPECT_EQ(kStreamId1, video_desc->streams()[0].first_stream_id());
}
// Test that we can specify a certain track that we want statistics about.
TEST_P(PeerConnectionInterfaceTest, GetStatsForSpecificTrack) {
InitiateCall();
ASSERT_LT(0u, pc_->GetSenders().size());
ASSERT_LT(0u, pc_->GetReceivers().size());
rtc::scoped_refptr<MediaStreamTrackInterface> remote_audio =
pc_->GetReceivers()[0]->track();
EXPECT_TRUE(DoGetStats(remote_audio));
// Remove the stream. Since we are sending to our selves the local
// and the remote stream is the same.
pc_->RemoveTrack(pc_->GetSenders()[0]);
// Do a re-negotiation.
CreateOfferReceiveAnswer();
// Test that we still can get statistics for the old track. Even if it is not
// sent any longer.
EXPECT_TRUE(DoGetStats(remote_audio));
}
// Test that we can get stats on a video track.
TEST_P(PeerConnectionInterfaceTest, GetStatsForVideoTrack) {
InitiateCall();
auto video_receiver = GetFirstReceiverOfType(cricket::MEDIA_TYPE_VIDEO);
ASSERT_TRUE(video_receiver);
EXPECT_TRUE(DoGetStats(video_receiver->track()));
}
// Test that we don't get statistics for an invalid track.
TEST_P(PeerConnectionInterfaceTest, GetStatsForInvalidTrack) {
InitiateCall();
rtc::scoped_refptr<AudioTrackInterface> unknown_audio_track(
pc_factory_->CreateAudioTrack("unknown track", NULL));
EXPECT_FALSE(DoGetStats(unknown_audio_track));
}
TEST_P(PeerConnectionInterfaceTest, GetRTCStatsBeforeAndAfterCalling) {
CreatePeerConnectionWithoutDtls();
EXPECT_TRUE(DoGetRTCStats());
// Clearing stats cache is needed now, but should be temporary.
// https://bugs.chromium.org/p/webrtc/issues/detail?id=8693
pc_->ClearStatsCache();
AddAudioTrack(kAudioTracks[0], {kStreamId1});
AddVideoTrack(kVideoTracks[0], {kStreamId1});
EXPECT_TRUE(DoGetRTCStats());
pc_->ClearStatsCache();
CreateOfferReceiveAnswer();
EXPECT_TRUE(DoGetRTCStats());
}
// This test setup two RTP data channels in loop back.
TEST_P(PeerConnectionInterfaceTest, TestDataChannel) {
FakeConstraints constraints;
constraints.SetAllowRtpDataChannels();
CreatePeerConnection(&constraints);
rtc::scoped_refptr<DataChannelInterface> data1 =
pc_->CreateDataChannel("test1", NULL);
rtc::scoped_refptr<DataChannelInterface> data2 =
pc_->CreateDataChannel("test2", NULL);
ASSERT_TRUE(data1 != NULL);
std::unique_ptr<MockDataChannelObserver> observer1(
new MockDataChannelObserver(data1));
std::unique_ptr<MockDataChannelObserver> observer2(
new MockDataChannelObserver(data2));
EXPECT_EQ(DataChannelInterface::kConnecting, data1->state());
EXPECT_EQ(DataChannelInterface::kConnecting, data2->state());
std::string data_to_send1 = "testing testing";
std::string data_to_send2 = "testing something else";
EXPECT_FALSE(data1->Send(DataBuffer(data_to_send1)));
CreateOfferReceiveAnswer();
EXPECT_TRUE_WAIT(observer1->IsOpen(), kTimeout);
EXPECT_TRUE_WAIT(observer2->IsOpen(), kTimeout);
EXPECT_EQ(DataChannelInterface::kOpen, data1->state());
EXPECT_EQ(DataChannelInterface::kOpen, data2->state());
EXPECT_TRUE(data1->Send(DataBuffer(data_to_send1)));
EXPECT_TRUE(data2->Send(DataBuffer(data_to_send2)));
EXPECT_EQ_WAIT(data_to_send1, observer1->last_message(), kTimeout);
EXPECT_EQ_WAIT(data_to_send2, observer2->last_message(), kTimeout);
data1->Close();
EXPECT_EQ(DataChannelInterface::kClosing, data1->state());
CreateOfferReceiveAnswer();
EXPECT_FALSE(observer1->IsOpen());
EXPECT_EQ(DataChannelInterface::kClosed, data1->state());
EXPECT_TRUE(observer2->IsOpen());
data_to_send2 = "testing something else again";
EXPECT_TRUE(data2->Send(DataBuffer(data_to_send2)));
EXPECT_EQ_WAIT(data_to_send2, observer2->last_message(), kTimeout);
}
// This test verifies that sendnig binary data over RTP data channels should
// fail.
TEST_P(PeerConnectionInterfaceTest, TestSendBinaryOnRtpDataChannel) {
FakeConstraints constraints;
constraints.SetAllowRtpDataChannels();
CreatePeerConnection(&constraints);
rtc::scoped_refptr<DataChannelInterface> data1 =
pc_->CreateDataChannel("test1", NULL);
rtc::scoped_refptr<DataChannelInterface> data2 =
pc_->CreateDataChannel("test2", NULL);
ASSERT_TRUE(data1 != NULL);
std::unique_ptr<MockDataChannelObserver> observer1(
new MockDataChannelObserver(data1));
std::unique_ptr<MockDataChannelObserver> observer2(
new MockDataChannelObserver(data2));
EXPECT_EQ(DataChannelInterface::kConnecting, data1->state());
EXPECT_EQ(DataChannelInterface::kConnecting, data2->state());
CreateOfferReceiveAnswer();
EXPECT_TRUE_WAIT(observer1->IsOpen(), kTimeout);
EXPECT_TRUE_WAIT(observer2->IsOpen(), kTimeout);
EXPECT_EQ(DataChannelInterface::kOpen, data1->state());
EXPECT_EQ(DataChannelInterface::kOpen, data2->state());
rtc::CopyOnWriteBuffer buffer("test", 4);
EXPECT_FALSE(data1->Send(DataBuffer(buffer, true)));
}
// This test setup a RTP data channels in loop back and test that a channel is
// opened even if the remote end answer with a zero SSRC.
TEST_P(PeerConnectionInterfaceTest, TestSendOnlyDataChannel) {
FakeConstraints constraints;
constraints.SetAllowRtpDataChannels();
CreatePeerConnection(&constraints);
rtc::scoped_refptr<DataChannelInterface> data1 =
pc_->CreateDataChannel("test1", NULL);
std::unique_ptr<MockDataChannelObserver> observer1(
new MockDataChannelObserver(data1));
CreateOfferReceiveAnswerWithoutSsrc();
EXPECT_TRUE_WAIT(observer1->IsOpen(), kTimeout);
data1->Close();
EXPECT_EQ(DataChannelInterface::kClosing, data1->state());
CreateOfferReceiveAnswerWithoutSsrc();
EXPECT_EQ(DataChannelInterface::kClosed, data1->state());
EXPECT_FALSE(observer1->IsOpen());
}
// This test that if a data channel is added in an answer a receive only channel
// channel is created.
TEST_P(PeerConnectionInterfaceTest, TestReceiveOnlyDataChannel) {
FakeConstraints constraints;
constraints.SetAllowRtpDataChannels();
CreatePeerConnection(&constraints);
std::string offer_label = "offer_channel";
rtc::scoped_refptr<DataChannelInterface> offer_channel =
pc_->CreateDataChannel(offer_label, NULL);
CreateOfferAsLocalDescription();
// Replace the data channel label in the offer and apply it as an answer.
std::string receive_label = "answer_channel";
std::string sdp;
EXPECT_TRUE(pc_->local_description()->ToString(&sdp));
rtc::replace_substrs(offer_label.c_str(), offer_label.length(),
receive_label.c_str(), receive_label.length(),
&sdp);
CreateAnswerAsRemoteDescription(sdp);
// Verify that a new incoming data channel has been created and that
// it is open but can't we written to.
ASSERT_TRUE(observer_.last_datachannel_ != NULL);
DataChannelInterface* received_channel = observer_.last_datachannel_;
EXPECT_EQ(DataChannelInterface::kConnecting, received_channel->state());
EXPECT_EQ(receive_label, received_channel->label());
EXPECT_FALSE(received_channel->Send(DataBuffer("something")));
// Verify that the channel we initially offered has been rejected.
EXPECT_EQ(DataChannelInterface::kClosed, offer_channel->state());
// Do another offer / answer exchange and verify that the data channel is
// opened.
CreateOfferReceiveAnswer();
EXPECT_EQ_WAIT(DataChannelInterface::kOpen, received_channel->state(),
kTimeout);
}
// This test that no data channel is returned if a reliable channel is
// requested.
// TODO(perkj): Remove this test once reliable channels are implemented.
TEST_P(PeerConnectionInterfaceTest, CreateReliableRtpDataChannelShouldFail) {
FakeConstraints constraints;
constraints.SetAllowRtpDataChannels();
CreatePeerConnection(&constraints);
std::string label = "test";
webrtc::DataChannelInit config;
config.reliable = true;
rtc::scoped_refptr<DataChannelInterface> channel =
pc_->CreateDataChannel(label, &config);
EXPECT_TRUE(channel == NULL);
}
// Verifies that duplicated label is not allowed for RTP data channel.
TEST_P(PeerConnectionInterfaceTest, RtpDuplicatedLabelNotAllowed) {
FakeConstraints constraints;
constraints.SetAllowRtpDataChannels();
CreatePeerConnection(&constraints);
std::string label = "test";
rtc::scoped_refptr<DataChannelInterface> channel =
pc_->CreateDataChannel(label, nullptr);
EXPECT_NE(channel, nullptr);
rtc::scoped_refptr<DataChannelInterface> dup_channel =
pc_->CreateDataChannel(label, nullptr);
EXPECT_EQ(dup_channel, nullptr);
}
// This tests that a SCTP data channel is returned using different
// DataChannelInit configurations.
TEST_P(PeerConnectionInterfaceTest, CreateSctpDataChannel) {
FakeConstraints constraints;
constraints.SetAllowDtlsSctpDataChannels();
CreatePeerConnection(&constraints);
webrtc::DataChannelInit config;
rtc::scoped_refptr<DataChannelInterface> channel =
pc_->CreateDataChannel("1", &config);
EXPECT_TRUE(channel != NULL);
EXPECT_TRUE(channel->reliable());
EXPECT_TRUE(observer_.renegotiation_needed_);
observer_.renegotiation_needed_ = false;
config.ordered = false;
channel = pc_->CreateDataChannel("2", &config);
EXPECT_TRUE(channel != NULL);
EXPECT_TRUE(channel->reliable());
EXPECT_FALSE(observer_.renegotiation_needed_);
config.ordered = true;
config.maxRetransmits = 0;
channel = pc_->CreateDataChannel("3", &config);
EXPECT_TRUE(channel != NULL);
EXPECT_FALSE(channel->reliable());
EXPECT_FALSE(observer_.renegotiation_needed_);
config.maxRetransmits = -1;
config.maxRetransmitTime = 0;
channel = pc_->CreateDataChannel("4", &config);
EXPECT_TRUE(channel != NULL);
EXPECT_FALSE(channel->reliable());
EXPECT_FALSE(observer_.renegotiation_needed_);
}
// This tests that no data channel is returned if both maxRetransmits and
// maxRetransmitTime are set for SCTP data channels.
TEST_P(PeerConnectionInterfaceTest,
CreateSctpDataChannelShouldFailForInvalidConfig) {
FakeConstraints constraints;
constraints.SetAllowDtlsSctpDataChannels();
CreatePeerConnection(&constraints);
std::string label = "test";
webrtc::DataChannelInit config;
config.maxRetransmits = 0;
config.maxRetransmitTime = 0;
rtc::scoped_refptr<DataChannelInterface> channel =
pc_->CreateDataChannel(label, &config);
EXPECT_TRUE(channel == NULL);
}
// The test verifies that creating a SCTP data channel with an id already in use
// or out of range should fail.
TEST_P(PeerConnectionInterfaceTest,
CreateSctpDataChannelWithInvalidIdShouldFail) {
FakeConstraints constraints;
constraints.SetAllowDtlsSctpDataChannels();
CreatePeerConnection(&constraints);
webrtc::DataChannelInit config;
rtc::scoped_refptr<DataChannelInterface> channel;
config.id = 1;
channel = pc_->CreateDataChannel("1", &config);
EXPECT_TRUE(channel != NULL);
EXPECT_EQ(1, channel->id());
channel = pc_->CreateDataChannel("x", &config);
EXPECT_TRUE(channel == NULL);
config.id = cricket::kMaxSctpSid;
channel = pc_->CreateDataChannel("max", &config);
EXPECT_TRUE(channel != NULL);
EXPECT_EQ(config.id, channel->id());
config.id = cricket::kMaxSctpSid + 1;
channel = pc_->CreateDataChannel("x", &config);
EXPECT_TRUE(channel == NULL);
}
// Verifies that duplicated label is allowed for SCTP data channel.
TEST_P(PeerConnectionInterfaceTest, SctpDuplicatedLabelAllowed) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
std::string label = "test";
rtc::scoped_refptr<DataChannelInterface> channel =
pc_->CreateDataChannel(label, nullptr);
EXPECT_NE(channel, nullptr);
rtc::scoped_refptr<DataChannelInterface> dup_channel =
pc_->CreateDataChannel(label, nullptr);
EXPECT_NE(dup_channel, nullptr);
}
// This test verifies that OnRenegotiationNeeded is fired for every new RTP
// DataChannel.
TEST_P(PeerConnectionInterfaceTest, RenegotiationNeededForNewRtpDataChannel) {
FakeConstraints constraints;
constraints.SetAllowRtpDataChannels();
CreatePeerConnection(&constraints);
rtc::scoped_refptr<DataChannelInterface> dc1 =
pc_->CreateDataChannel("test1", NULL);
EXPECT_TRUE(observer_.renegotiation_needed_);
observer_.renegotiation_needed_ = false;
rtc::scoped_refptr<DataChannelInterface> dc2 =
pc_->CreateDataChannel("test2", NULL);
EXPECT_TRUE(observer_.renegotiation_needed_);
}
// This test that a data channel closes when a PeerConnection is deleted/closed.
TEST_P(PeerConnectionInterfaceTest, DataChannelCloseWhenPeerConnectionClose) {
FakeConstraints constraints;
constraints.SetAllowRtpDataChannels();
CreatePeerConnection(&constraints);
rtc::scoped_refptr<DataChannelInterface> data1 =
pc_->CreateDataChannel("test1", NULL);
rtc::scoped_refptr<DataChannelInterface> data2 =
pc_->CreateDataChannel("test2", NULL);
ASSERT_TRUE(data1 != NULL);
std::unique_ptr<MockDataChannelObserver> observer1(
new MockDataChannelObserver(data1));
std::unique_ptr<MockDataChannelObserver> observer2(
new MockDataChannelObserver(data2));
CreateOfferReceiveAnswer();
EXPECT_TRUE_WAIT(observer1->IsOpen(), kTimeout);
EXPECT_TRUE_WAIT(observer2->IsOpen(), kTimeout);
ReleasePeerConnection();
EXPECT_EQ(DataChannelInterface::kClosed, data1->state());
EXPECT_EQ(DataChannelInterface::kClosed, data2->state());
}
// This test that data channels can be rejected in an answer.
TEST_P(PeerConnectionInterfaceTest, TestRejectDataChannelInAnswer) {
FakeConstraints constraints;
constraints.SetAllowRtpDataChannels();
CreatePeerConnection(&constraints);
rtc::scoped_refptr<DataChannelInterface> offer_channel(
pc_->CreateDataChannel("offer_channel", NULL));
CreateOfferAsLocalDescription();
// Create an answer where the m-line for data channels are rejected.
std::string sdp;
EXPECT_TRUE(pc_->local_description()->ToString(&sdp));
std::unique_ptr<SessionDescriptionInterface> answer(
webrtc::CreateSessionDescription(SdpType::kAnswer, sdp));
ASSERT_TRUE(answer);
cricket::ContentInfo* data_info =
cricket::GetFirstDataContent(answer->description());
data_info->rejected = true;
DoSetRemoteDescription(std::move(answer));
EXPECT_EQ(DataChannelInterface::kClosed, offer_channel->state());
}
// Test that we can create a session description from an SDP string from
// FireFox, use it as a remote session description, generate an answer and use
// the answer as a local description.
TEST_P(PeerConnectionInterfaceTest, ReceiveFireFoxOffer) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
AddAudioTrack("audio_label");
AddVideoTrack("video_label");
std::unique_ptr<SessionDescriptionInterface> desc(
webrtc::CreateSessionDescription(SdpType::kOffer,
webrtc::kFireFoxSdpOffer, nullptr));
EXPECT_TRUE(DoSetSessionDescription(std::move(desc), false));
CreateAnswerAsLocalDescription();
ASSERT_TRUE(pc_->local_description() != NULL);
ASSERT_TRUE(pc_->remote_description() != NULL);
const cricket::ContentInfo* content =
cricket::GetFirstAudioContent(pc_->local_description()->description());
ASSERT_TRUE(content != NULL);
EXPECT_FALSE(content->rejected);
content =
cricket::GetFirstVideoContent(pc_->local_description()->description());
ASSERT_TRUE(content != NULL);
EXPECT_FALSE(content->rejected);
#ifdef HAVE_SCTP
content =
cricket::GetFirstDataContent(pc_->local_description()->description());
ASSERT_TRUE(content != NULL);
EXPECT_FALSE(content->rejected);
#endif
}
// Test that fallback from DTLS to SDES is not supported.
// The fallback was previously supported but was removed to simplify the code
// and because it's non-standard.
TEST_P(PeerConnectionInterfaceTest, DtlsSdesFallbackNotSupported) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
// Wait for fake certificate to be generated. Previously, this is what caused
// the "a=crypto" lines to be rejected.
AddAudioTrack("audio_label");
AddVideoTrack("video_label");
ASSERT_NE(nullptr, fake_certificate_generator_);
EXPECT_EQ_WAIT(1, fake_certificate_generator_->generated_certificates(),
kTimeout);
std::unique_ptr<SessionDescriptionInterface> desc(
webrtc::CreateSessionDescription(SdpType::kOffer, kDtlsSdesFallbackSdp,
nullptr));
EXPECT_FALSE(DoSetSessionDescription(std::move(desc), /*local=*/false));
}
// Test that we can create an audio only offer and receive an answer with a
// limited set of audio codecs and receive an updated offer with more audio
// codecs, where the added codecs are not supported.
TEST_P(PeerConnectionInterfaceTest, ReceiveUpdatedAudioOfferWithBadCodecs) {
CreatePeerConnectionWithoutDtls();
AddAudioTrack("audio_label");
CreateOfferAsLocalDescription();
const char* answer_sdp =
(sdp_semantics_ == SdpSemantics::kPlanB ? webrtc::kAudioSdpPlanB
: webrtc::kAudioSdpUnifiedPlan);
std::unique_ptr<SessionDescriptionInterface> answer(
webrtc::CreateSessionDescription(SdpType::kAnswer, answer_sdp, nullptr));
EXPECT_TRUE(DoSetSessionDescription(std::move(answer), false));
const char* reoffer_sdp =
(sdp_semantics_ == SdpSemantics::kPlanB
? webrtc::kAudioSdpWithUnsupportedCodecsPlanB
: webrtc::kAudioSdpWithUnsupportedCodecsUnifiedPlan);
std::unique_ptr<SessionDescriptionInterface> updated_offer(
webrtc::CreateSessionDescription(SdpType::kOffer, reoffer_sdp, nullptr));
EXPECT_TRUE(DoSetSessionDescription(std::move(updated_offer), false));
CreateAnswerAsLocalDescription();
}
// Test that if we're receiving (but not sending) a track, subsequent offers
// will have m-lines with a=recvonly.
TEST_P(PeerConnectionInterfaceTest, CreateSubsequentRecvOnlyOffer) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
CreateAndSetRemoteOffer(GetSdpStringWithStream1());
CreateAnswerAsLocalDescription();
// At this point we should be receiving stream 1, but not sending anything.
// A new offer should be recvonly.
std::unique_ptr<SessionDescriptionInterface> offer;
DoCreateOffer(&offer, nullptr);
const cricket::ContentInfo* video_content =
cricket::GetFirstVideoContent(offer->description());
ASSERT_EQ(RtpTransceiverDirection::kRecvOnly,
video_content->media_description()->direction());
const cricket::ContentInfo* audio_content =
cricket::GetFirstAudioContent(offer->description());
ASSERT_EQ(RtpTransceiverDirection::kRecvOnly,
audio_content->media_description()->direction());
}
// Test that if we're receiving (but not sending) a track, and the
// offerToReceiveVideo/offerToReceiveAudio constraints are explicitly set to
// false, the generated m-lines will be a=inactive.
TEST_P(PeerConnectionInterfaceTest, CreateSubsequentInactiveOffer) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
CreateAndSetRemoteOffer(GetSdpStringWithStream1());
CreateAnswerAsLocalDescription();
// At this point we should be receiving stream 1, but not sending anything.
// A new offer would be recvonly, but we'll set the "no receive" constraints
// to make it inactive.
std::unique_ptr<SessionDescriptionInterface> offer;
FakeConstraints offer_constraints;
offer_constraints.AddMandatory(
webrtc::MediaConstraintsInterface::kOfferToReceiveVideo, false);
offer_constraints.AddMandatory(
webrtc::MediaConstraintsInterface::kOfferToReceiveAudio, false);
DoCreateOffer(&offer, &offer_constraints);
const cricket::ContentInfo* video_content =
cricket::GetFirstVideoContent(offer->description());
ASSERT_EQ(RtpTransceiverDirection::kInactive,
video_content->media_description()->direction());
const cricket::ContentInfo* audio_content =
cricket::GetFirstAudioContent(offer->description());
ASSERT_EQ(RtpTransceiverDirection::kInactive,
audio_content->media_description()->direction());
}
// Test that we can use SetConfiguration to change the ICE servers of the
// PortAllocator.
TEST_P(PeerConnectionInterfaceTest, SetConfigurationChangesIceServers) {
CreatePeerConnection();
PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration();
PeerConnectionInterface::IceServer server;
server.uri = "stun:test_hostname";
config.servers.push_back(server);
EXPECT_TRUE(pc_->SetConfiguration(config));
EXPECT_EQ(1u, port_allocator_->stun_servers().size());
EXPECT_EQ("test_hostname",
port_allocator_->stun_servers().begin()->hostname());
}
TEST_P(PeerConnectionInterfaceTest, SetConfigurationChangesCandidateFilter) {
CreatePeerConnection();
PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration();
config.type = PeerConnectionInterface::kRelay;
EXPECT_TRUE(pc_->SetConfiguration(config));
EXPECT_EQ(cricket::CF_RELAY, port_allocator_->candidate_filter());
}
TEST_P(PeerConnectionInterfaceTest, SetConfigurationChangesPruneTurnPortsFlag) {
PeerConnectionInterface::RTCConfiguration config;
config.prune_turn_ports = false;
CreatePeerConnection(config, nullptr);
config = pc_->GetConfiguration();
EXPECT_FALSE(port_allocator_->prune_turn_ports());
config.prune_turn_ports = true;
EXPECT_TRUE(pc_->SetConfiguration(config));
EXPECT_TRUE(port_allocator_->prune_turn_ports());
}
// Test that the ice check interval can be changed. This does not verify that
// the setting makes it all the way to P2PTransportChannel, as that would
// require a very complex set of mocks.
TEST_P(PeerConnectionInterfaceTest, SetConfigurationChangesIceCheckInterval) {
PeerConnectionInterface::RTCConfiguration config;
config.ice_check_min_interval = rtc::nullopt;
CreatePeerConnection(config, nullptr);
config = pc_->GetConfiguration();
config.ice_check_min_interval = 100;
EXPECT_TRUE(pc_->SetConfiguration(config));
PeerConnectionInterface::RTCConfiguration new_config =
pc_->GetConfiguration();
EXPECT_EQ(new_config.ice_check_min_interval, 100);
}
// Test that when SetConfiguration changes both the pool size and other
// attributes, the pooled session is created with the updated attributes.
TEST_P(PeerConnectionInterfaceTest,
SetConfigurationCreatesPooledSessionCorrectly) {
CreatePeerConnection();
PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration();
config.ice_candidate_pool_size = 1;
PeerConnectionInterface::IceServer server;
server.uri = kStunAddressOnly;
config.servers.push_back(server);
config.type = PeerConnectionInterface::kRelay;
EXPECT_TRUE(pc_->SetConfiguration(config));
const cricket::FakePortAllocatorSession* session =
static_cast<const cricket::FakePortAllocatorSession*>(
port_allocator_->GetPooledSession());
ASSERT_NE(nullptr, session);
EXPECT_EQ(1UL, session->stun_servers().size());
}
// Test that after SetLocalDescription, changing the pool size is not allowed,
// and an invalid modification error is returned.
TEST_P(PeerConnectionInterfaceTest,
CantChangePoolSizeAfterSetLocalDescription) {
CreatePeerConnection();
// Start by setting a size of 1.
PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration();
config.ice_candidate_pool_size = 1;
EXPECT_TRUE(pc_->SetConfiguration(config));
// Set remote offer; can still change pool size at this point.
CreateOfferAsRemoteDescription();
config.ice_candidate_pool_size = 2;
EXPECT_TRUE(pc_->SetConfiguration(config));
// Set local answer; now it's too late.
CreateAnswerAsLocalDescription();
config.ice_candidate_pool_size = 3;
RTCError error;
EXPECT_FALSE(pc_->SetConfiguration(config, &error));
EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, error.type());
}
// Test that after setting an answer, extra pooled sessions are discarded. The
// ICE candidate pool is only intended to be used for the first offer/answer.
TEST_P(PeerConnectionInterfaceTest,
ExtraPooledSessionsDiscardedAfterApplyingAnswer) {
CreatePeerConnection();
// Set a larger-than-necessary size.
PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration();
config.ice_candidate_pool_size = 4;
EXPECT_TRUE(pc_->SetConfiguration(config));
// Do offer/answer.
CreateOfferAsRemoteDescription();
CreateAnswerAsLocalDescription();
// Expect no pooled sessions to be left.
const cricket::PortAllocatorSession* session =
port_allocator_->GetPooledSession();
EXPECT_EQ(nullptr, session);
}
// After Close is called, pooled candidates should be discarded so as to not
// waste network resources.
TEST_P(PeerConnectionInterfaceTest, PooledSessionsDiscardedAfterClose) {
CreatePeerConnection();
PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration();
config.ice_candidate_pool_size = 3;
EXPECT_TRUE(pc_->SetConfiguration(config));
pc_->Close();
// Expect no pooled sessions to be left.
const cricket::PortAllocatorSession* session =
port_allocator_->GetPooledSession();
EXPECT_EQ(nullptr, session);
}
// Test that SetConfiguration returns an invalid modification error if
// modifying a field in the configuration that isn't allowed to be modified.
TEST_P(PeerConnectionInterfaceTest,
SetConfigurationReturnsInvalidModificationError) {
PeerConnectionInterface::RTCConfiguration config;
config.bundle_policy = PeerConnectionInterface::kBundlePolicyBalanced;
config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyNegotiate;
config.continual_gathering_policy = PeerConnectionInterface::GATHER_ONCE;
CreatePeerConnection(config, nullptr);
PeerConnectionInterface::RTCConfiguration modified_config =
pc_->GetConfiguration();
modified_config.bundle_policy =
PeerConnectionInterface::kBundlePolicyMaxBundle;
RTCError error;
EXPECT_FALSE(pc_->SetConfiguration(modified_config, &error));
EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, error.type());
modified_config = pc_->GetConfiguration();
modified_config.rtcp_mux_policy =
PeerConnectionInterface::kRtcpMuxPolicyRequire;
error.set_type(RTCErrorType::NONE);
EXPECT_FALSE(pc_->SetConfiguration(modified_config, &error));
EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, error.type());
modified_config = pc_->GetConfiguration();
modified_config.continual_gathering_policy =
PeerConnectionInterface::GATHER_CONTINUALLY;
error.set_type(RTCErrorType::NONE);
EXPECT_FALSE(pc_->SetConfiguration(modified_config, &error));
EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, error.type());
}
// Test that SetConfiguration returns a range error if the candidate pool size
// is negative or larger than allowed by the spec.
TEST_P(PeerConnectionInterfaceTest,
SetConfigurationReturnsRangeErrorForBadCandidatePoolSize) {
PeerConnectionInterface::RTCConfiguration config;
CreatePeerConnection(config, nullptr);
config = pc_->GetConfiguration();
config.ice_candidate_pool_size = -1;
RTCError error;
EXPECT_FALSE(pc_->SetConfiguration(config, &error));
EXPECT_EQ(RTCErrorType::INVALID_RANGE, error.type());
config.ice_candidate_pool_size = INT_MAX;
error.set_type(RTCErrorType::NONE);
EXPECT_FALSE(pc_->SetConfiguration(config, &error));
EXPECT_EQ(RTCErrorType::INVALID_RANGE, error.type());
}
// Test that SetConfiguration returns a syntax error if parsing an ICE server
// URL failed.
TEST_P(PeerConnectionInterfaceTest,
SetConfigurationReturnsSyntaxErrorFromBadIceUrls) {
PeerConnectionInterface::RTCConfiguration config;
CreatePeerConnection(config, nullptr);
config = pc_->GetConfiguration();
PeerConnectionInterface::IceServer bad_server;
bad_server.uri = "stunn:www.example.com";
config.servers.push_back(bad_server);
RTCError error;
EXPECT_FALSE(pc_->SetConfiguration(config, &error));
EXPECT_EQ(RTCErrorType::SYNTAX_ERROR, error.type());
}
// Test that SetConfiguration returns an invalid parameter error if a TURN
// IceServer is missing a username or password.
TEST_P(PeerConnectionInterfaceTest,
SetConfigurationReturnsInvalidParameterIfCredentialsMissing) {
PeerConnectionInterface::RTCConfiguration config;
CreatePeerConnection(config, nullptr);
config = pc_->GetConfiguration();
PeerConnectionInterface::IceServer bad_server;
bad_server.uri = "turn:www.example.com";
// Missing password.
bad_server.username = "foo";
config.servers.push_back(bad_server);
RTCError error;
EXPECT_FALSE(pc_->SetConfiguration(config, &error));
EXPECT_EQ(RTCErrorType::INVALID_PARAMETER, error.type());
}
// Test that PeerConnection::Close changes the states to closed and all remote
// tracks change state to ended.
TEST_P(PeerConnectionInterfaceTest, CloseAndTestStreamsAndStates) {
// Initialize a PeerConnection and negotiate local and remote session
// description.
InitiateCall();
// With Plan B, verify the stream count. The analog with Unified Plan is the
// RtpTransceiver count.
if (sdp_semantics_ == SdpSemantics::kPlanB) {
ASSERT_EQ(1u, pc_->local_streams()->count());
ASSERT_EQ(1u, pc_->remote_streams()->count());
} else {
ASSERT_EQ(2u, pc_->GetTransceivers().size());
}
pc_->Close();
EXPECT_EQ(PeerConnectionInterface::kClosed, pc_->signaling_state());
EXPECT_EQ(PeerConnectionInterface::kIceConnectionClosed,
pc_->ice_connection_state());
EXPECT_EQ(PeerConnectionInterface::kIceGatheringComplete,
pc_->ice_gathering_state());
if (sdp_semantics_ == SdpSemantics::kPlanB) {
EXPECT_EQ(1u, pc_->local_streams()->count());
EXPECT_EQ(1u, pc_->remote_streams()->count());
} else {
// Verify that the RtpTransceivers are still present but all stopped.
EXPECT_EQ(2u, pc_->GetTransceivers().size());
for (auto transceiver : pc_->GetTransceivers()) {
EXPECT_TRUE(transceiver->stopped());
}
}
auto audio_receiver = GetFirstReceiverOfType(cricket::MEDIA_TYPE_AUDIO);
ASSERT_TRUE(audio_receiver);
auto video_receiver = GetFirstReceiverOfType(cricket::MEDIA_TYPE_VIDEO);
ASSERT_TRUE(video_receiver);
// Track state may be updated asynchronously.
EXPECT_EQ_WAIT(MediaStreamTrackInterface::kEnded,
audio_receiver->track()->state(), kTimeout);
EXPECT_EQ_WAIT(MediaStreamTrackInterface::kEnded,
video_receiver->track()->state(), kTimeout);
}
// Test that PeerConnection methods fails gracefully after
// PeerConnection::Close has been called.
// Don't run under Unified Plan since the stream API is not available.
TEST_F(PeerConnectionInterfaceTestPlanB, CloseAndTestMethods) {
CreatePeerConnectionWithoutDtls();
AddAudioVideoStream(kStreamId1, "audio_label", "video_label");
CreateOfferAsRemoteDescription();
CreateAnswerAsLocalDescription();
ASSERT_EQ(1u, pc_->local_streams()->count());
rtc::scoped_refptr<MediaStreamInterface> local_stream =
pc_->local_streams()->at(0);
pc_->Close();
pc_->RemoveStream(local_stream);
EXPECT_FALSE(pc_->AddStream(local_stream));
ASSERT_FALSE(local_stream->GetAudioTracks().empty());
rtc::scoped_refptr<webrtc::DtmfSenderInterface> dtmf_sender(
pc_->CreateDtmfSender(local_stream->GetAudioTracks()[0]));
EXPECT_TRUE(NULL == dtmf_sender); // local stream has been removed.
EXPECT_TRUE(pc_->CreateDataChannel("test", NULL) == NULL);
EXPECT_TRUE(pc_->local_description() != NULL);
EXPECT_TRUE(pc_->remote_description() != NULL);
std::unique_ptr<SessionDescriptionInterface> offer;
EXPECT_FALSE(DoCreateOffer(&offer, nullptr));
std::unique_ptr<SessionDescriptionInterface> answer;
EXPECT_FALSE(DoCreateAnswer(&answer, nullptr));
std::string sdp;
ASSERT_TRUE(pc_->remote_description()->ToString(&sdp));
std::unique_ptr<SessionDescriptionInterface> remote_offer(
webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
EXPECT_FALSE(DoSetRemoteDescription(std::move(remote_offer)));
ASSERT_TRUE(pc_->local_description()->ToString(&sdp));
std::unique_ptr<SessionDescriptionInterface> local_offer(
webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
EXPECT_FALSE(DoSetLocalDescription(std::move(local_offer)));
}
// Test that GetStats can still be called after PeerConnection::Close.
TEST_P(PeerConnectionInterfaceTest, CloseAndGetStats) {
InitiateCall();
pc_->Close();
DoGetStats(NULL);
}
// NOTE: The series of tests below come from what used to be
// mediastreamsignaling_unittest.cc, and are mostly aimed at testing that
// setting a remote or local description has the expected effects.
// This test verifies that the remote MediaStreams corresponding to a received
// SDP string is created. In this test the two separate MediaStreams are
// signaled.
TEST_P(PeerConnectionInterfaceTest, UpdateRemoteStreams) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
CreateAndSetRemoteOffer(GetSdpStringWithStream1());
rtc::scoped_refptr<StreamCollection> reference(CreateStreamCollection(1, 1));
EXPECT_TRUE(
CompareStreamCollections(observer_.remote_streams(), reference.get()));
MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
EXPECT_TRUE(remote_stream->GetVideoTracks()[0]->GetSource() != nullptr);
// Create a session description based on another SDP with another
// MediaStream.
CreateAndSetRemoteOffer(GetSdpStringWithStream1And2());
rtc::scoped_refptr<StreamCollection> reference2(CreateStreamCollection(2, 1));
EXPECT_TRUE(
CompareStreamCollections(observer_.remote_streams(), reference2.get()));
}
// This test verifies that when remote tracks are added/removed from SDP, the
// created remote streams are updated appropriately.
// Don't run under Unified Plan since this test uses Plan B SDP to test Plan B
// specific behavior.
TEST_F(PeerConnectionInterfaceTestPlanB,
AddRemoveTrackFromExistingRemoteMediaStream) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
std::unique_ptr<SessionDescriptionInterface> desc_ms1 =
CreateSessionDescriptionAndReference(1, 1);
EXPECT_TRUE(DoSetRemoteDescription(std::move(desc_ms1)));
EXPECT_TRUE(CompareStreamCollections(observer_.remote_streams(),
reference_collection_));
// Add extra audio and video tracks to the same MediaStream.
std::unique_ptr<SessionDescriptionInterface> desc_ms1_two_tracks =
CreateSessionDescriptionAndReference(2, 2);
EXPECT_TRUE(DoSetRemoteDescription(std::move(desc_ms1_two_tracks)));
EXPECT_TRUE(CompareStreamCollections(observer_.remote_streams(),
reference_collection_));
rtc::scoped_refptr<AudioTrackInterface> audio_track2 =
observer_.remote_streams()->at(0)->GetAudioTracks()[1];
EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, audio_track2->state());
rtc::scoped_refptr<VideoTrackInterface> video_track2 =
observer_.remote_streams()->at(0)->GetVideoTracks()[1];
EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, video_track2->state());
// Remove the extra audio and video tracks.
std::unique_ptr<SessionDescriptionInterface> desc_ms2 =
CreateSessionDescriptionAndReference(1, 1);
MockTrackObserver audio_track_observer(audio_track2);
MockTrackObserver video_track_observer(video_track2);
EXPECT_CALL(audio_track_observer, OnChanged()).Times(Exactly(1));
EXPECT_CALL(video_track_observer, OnChanged()).Times(Exactly(1));
EXPECT_TRUE(DoSetRemoteDescription(std::move(desc_ms2)));
EXPECT_TRUE(CompareStreamCollections(observer_.remote_streams(),
reference_collection_));
// Track state may be updated asynchronously.
EXPECT_EQ_WAIT(webrtc::MediaStreamTrackInterface::kEnded,
audio_track2->state(), kTimeout);
EXPECT_EQ_WAIT(webrtc::MediaStreamTrackInterface::kEnded,
video_track2->state(), kTimeout);
}
// This tests that remote tracks are ended if a local session description is set
// that rejects the media content type.
TEST_P(PeerConnectionInterfaceTest, RejectMediaContent) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
// First create and set a remote offer, then reject its video content in our
// answer.
CreateAndSetRemoteOffer(kSdpStringWithStream1PlanB);
auto audio_receiver = GetFirstReceiverOfType(cricket::MEDIA_TYPE_AUDIO);
ASSERT_TRUE(audio_receiver);
auto video_receiver = GetFirstReceiverOfType(cricket::MEDIA_TYPE_VIDEO);
ASSERT_TRUE(video_receiver);
rtc::scoped_refptr<MediaStreamTrackInterface> remote_audio =
audio_receiver->track();
EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, remote_audio->state());
rtc::scoped_refptr<MediaStreamTrackInterface> remote_video =
video_receiver->track();
EXPECT_EQ(MediaStreamTrackInterface::kLive, remote_video->state());
std::unique_ptr<SessionDescriptionInterface> local_answer;
EXPECT_TRUE(DoCreateAnswer(&local_answer, nullptr));
cricket::ContentInfo* video_info =
local_answer->description()->GetContentByName("video");
video_info->rejected = true;
EXPECT_TRUE(DoSetLocalDescription(std::move(local_answer)));
EXPECT_EQ(MediaStreamTrackInterface::kEnded, remote_video->state());
EXPECT_EQ(MediaStreamTrackInterface::kLive, remote_audio->state());
// Now create an offer where we reject both video and audio.
std::unique_ptr<SessionDescriptionInterface> local_offer;
EXPECT_TRUE(DoCreateOffer(&local_offer, nullptr));
video_info = local_offer->description()->GetContentByName("video");
ASSERT_TRUE(video_info != nullptr);
video_info->rejected = true;
cricket::ContentInfo* audio_info =
local_offer->description()->GetContentByName("audio");
ASSERT_TRUE(audio_info != nullptr);
audio_info->rejected = true;
EXPECT_TRUE(DoSetLocalDescription(std::move(local_offer)));
// Track state may be updated asynchronously.
EXPECT_EQ_WAIT(MediaStreamTrackInterface::kEnded, remote_audio->state(),
kTimeout);
EXPECT_EQ_WAIT(MediaStreamTrackInterface::kEnded, remote_video->state(),
kTimeout);
}
// This tests that we won't crash if the remote track has been removed outside
// of PeerConnection and then PeerConnection tries to reject the track.
// Don't run under Unified Plan since the stream API is not available.
TEST_F(PeerConnectionInterfaceTestPlanB, RemoveTrackThenRejectMediaContent) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
CreateAndSetRemoteOffer(GetSdpStringWithStream1());
MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
remote_stream->RemoveTrack(remote_stream->GetVideoTracks()[0]);
remote_stream->RemoveTrack(remote_stream->GetAudioTracks()[0]);
std::unique_ptr<SessionDescriptionInterface> local_answer(
webrtc::CreateSessionDescription(SdpType::kAnswer,
GetSdpStringWithStream1(), nullptr));
cricket::ContentInfo* video_info =
local_answer->description()->GetContentByName("video");
video_info->rejected = true;
cricket::ContentInfo* audio_info =
local_answer->description()->GetContentByName("audio");
audio_info->rejected = true;
EXPECT_TRUE(DoSetLocalDescription(std::move(local_answer)));
// No crash is a pass.
}
// This tests that if a recvonly remote description is set, no remote streams
// will be created, even if the description contains SSRCs/MSIDs.
// See: https://code.google.com/p/webrtc/issues/detail?id=5054
TEST_P(PeerConnectionInterfaceTest, RecvonlyDescriptionDoesntCreateStream) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
std::string recvonly_offer = GetSdpStringWithStream1();
rtc::replace_substrs(kSendrecv, strlen(kSendrecv), kRecvonly,
strlen(kRecvonly), &recvonly_offer);
CreateAndSetRemoteOffer(recvonly_offer);
EXPECT_EQ(0u, observer_.remote_streams()->count());
}
// This tests that a default MediaStream is created if a remote session
// description doesn't contain any streams and no MSID support.
// It also tests that the default stream is updated if a video m-line is added
// in a subsequent session description.
// Don't run under Unified Plan since this behavior is Plan B specific.
TEST_F(PeerConnectionInterfaceTestPlanB, SdpWithoutMsidCreatesDefaultStream) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
CreateAndSetRemoteOffer(kSdpStringWithoutStreamsAudioOnly);
ASSERT_EQ(1u, observer_.remote_streams()->count());
MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
EXPECT_EQ(1u, remote_stream->GetAudioTracks().size());
EXPECT_EQ(0u, remote_stream->GetVideoTracks().size());
EXPECT_EQ("default", remote_stream->id());
CreateAndSetRemoteOffer(kSdpStringWithoutStreams);
ASSERT_EQ(1u, observer_.remote_streams()->count());
ASSERT_EQ(1u, remote_stream->GetAudioTracks().size());
EXPECT_EQ("defaulta0", remote_stream->GetAudioTracks()[0]->id());
EXPECT_EQ(MediaStreamTrackInterface::kLive,
remote_stream->GetAudioTracks()[0]->state());
ASSERT_EQ(1u, remote_stream->GetVideoTracks().size());
EXPECT_EQ("defaultv0", remote_stream->GetVideoTracks()[0]->id());
EXPECT_EQ(MediaStreamTrackInterface::kLive,
remote_stream->GetVideoTracks()[0]->state());
}
// This tests that a default MediaStream is created if a remote session
// description doesn't contain any streams and media direction is send only.
// Don't run under Unified Plan since this behavior is Plan B specific.
TEST_F(PeerConnectionInterfaceTestPlanB,
SendOnlySdpWithoutMsidCreatesDefaultStream) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
CreateAndSetRemoteOffer(kSdpStringSendOnlyWithoutStreams);
ASSERT_EQ(1u, observer_.remote_streams()->count());
MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
EXPECT_EQ(1u, remote_stream->GetAudioTracks().size());
EXPECT_EQ(1u, remote_stream->GetVideoTracks().size());
EXPECT_EQ("default", remote_stream->id());
}
// This tests that it won't crash when PeerConnection tries to remove
// a remote track that as already been removed from the MediaStream.
// Don't run under Unified Plan since this behavior is Plan B specific.
TEST_F(PeerConnectionInterfaceTestPlanB, RemoveAlreadyGoneRemoteStream) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
CreateAndSetRemoteOffer(GetSdpStringWithStream1());
MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
remote_stream->RemoveTrack(remote_stream->GetAudioTracks()[0]);
remote_stream->RemoveTrack(remote_stream->GetVideoTracks()[0]);
CreateAndSetRemoteOffer(kSdpStringWithoutStreams);
// No crash is a pass.
}
// This tests that a default MediaStream is created if the remote session
// description doesn't contain any streams and don't contain an indication if
// MSID is supported.
// Don't run under Unified Plan since this behavior is Plan B specific.
TEST_F(PeerConnectionInterfaceTestPlanB,
SdpWithoutMsidAndStreamsCreatesDefaultStream) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
CreateAndSetRemoteOffer(kSdpStringWithoutStreams);
ASSERT_EQ(1u, observer_.remote_streams()->count());
MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
EXPECT_EQ(1u, remote_stream->GetAudioTracks().size());
EXPECT_EQ(1u, remote_stream->GetVideoTracks().size());
}
// This tests that a default MediaStream is not created if the remote session
// description doesn't contain any streams but does support MSID.
// Don't run under Unified Plan since this behavior is Plan B specific.
TEST_F(PeerConnectionInterfaceTestPlanB, SdpWithMsidDontCreatesDefaultStream) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
CreateAndSetRemoteOffer(kSdpStringWithMsidWithoutStreams);
EXPECT_EQ(0u, observer_.remote_streams()->count());
}
// This tests that when setting a new description, the old default tracks are
// not destroyed and recreated.
// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5250
// Don't run under Unified Plan since this behavior is Plan B specific.
TEST_F(PeerConnectionInterfaceTestPlanB,
DefaultTracksNotDestroyedAndRecreated) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
CreateAndSetRemoteOffer(kSdpStringWithoutStreamsAudioOnly);
ASSERT_EQ(1u, observer_.remote_streams()->count());
MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
ASSERT_EQ(1u, remote_stream->GetAudioTracks().size());
// Set the track to "disabled", then set a new description and ensure the
// track is still disabled, which ensures it hasn't been recreated.
remote_stream->GetAudioTracks()[0]->set_enabled(false);
CreateAndSetRemoteOffer(kSdpStringWithoutStreamsAudioOnly);
ASSERT_EQ(1u, remote_stream->GetAudioTracks().size());
EXPECT_FALSE(remote_stream->GetAudioTracks()[0]->enabled());
}
// This tests that a default MediaStream is not created if a remote session
// description is updated to not have any MediaStreams.
// Don't run under Unified Plan since this behavior is Plan B specific.
TEST_F(PeerConnectionInterfaceTestPlanB, VerifyDefaultStreamIsNotCreated) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
CreateAndSetRemoteOffer(GetSdpStringWithStream1());
rtc::scoped_refptr<StreamCollection> reference(CreateStreamCollection(1, 1));
EXPECT_TRUE(
CompareStreamCollections(observer_.remote_streams(), reference.get()));
CreateAndSetRemoteOffer(kSdpStringWithoutStreams);
EXPECT_EQ(0u, observer_.remote_streams()->count());
}
// This tests that an RtpSender is created when the local description is set
// after adding a local stream.
// TODO(deadbeef): This test and the one below it need to be updated when
// an RtpSender's lifetime isn't determined by when a local description is set.
// Don't run under Unified Plan since this behavior is Plan B specific.
TEST_F(PeerConnectionInterfaceTestPlanB, LocalDescriptionChanged) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
// Create an offer with 1 stream with 2 tracks of each type.
rtc::scoped_refptr<StreamCollection> stream_collection =
CreateStreamCollection(1, 2);
pc_->AddStream(stream_collection->at(0));
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
EXPECT_TRUE(DoSetLocalDescription(std::move(offer)));
auto senders = pc_->GetSenders();
EXPECT_EQ(4u, senders.size());
EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0]));
EXPECT_TRUE(ContainsSender(senders, kAudioTracks[1]));
EXPECT_TRUE(ContainsSender(senders, kVideoTracks[1]));
// Remove an audio and video track.
pc_->RemoveStream(stream_collection->at(0));
stream_collection = CreateStreamCollection(1, 1);
pc_->AddStream(stream_collection->at(0));
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
EXPECT_TRUE(DoSetLocalDescription(std::move(offer)));
senders = pc_->GetSenders();
EXPECT_EQ(2u, senders.size());
EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0]));
EXPECT_FALSE(ContainsSender(senders, kAudioTracks[1]));
EXPECT_FALSE(ContainsSender(senders, kVideoTracks[1]));
}
// This tests that an RtpSender is created when the local description is set
// before adding a local stream.
// Don't run under Unified Plan since this behavior is Plan B specific.
TEST_F(PeerConnectionInterfaceTestPlanB,
AddLocalStreamAfterLocalDescriptionChanged) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
rtc::scoped_refptr<StreamCollection> stream_collection =
CreateStreamCollection(1, 2);
// Add a stream to create the offer, but remove it afterwards.
pc_->AddStream(stream_collection->at(0));
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
pc_->RemoveStream(stream_collection->at(0));
EXPECT_TRUE(DoSetLocalDescription(std::move(offer)));
auto senders = pc_->GetSenders();
EXPECT_EQ(0u, senders.size());
pc_->AddStream(stream_collection->at(0));
senders = pc_->GetSenders();
EXPECT_EQ(4u, senders.size());
EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0]));
EXPECT_TRUE(ContainsSender(senders, kAudioTracks[1]));
EXPECT_TRUE(ContainsSender(senders, kVideoTracks[1]));
}
// This tests that the expected behavior occurs if the SSRC on a local track is
// changed when SetLocalDescription is called.
TEST_P(PeerConnectionInterfaceTest,
ChangeSsrcOnTrackInLocalSessionDescription) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
AddAudioTrack(kAudioTracks[0]);
AddVideoTrack(kVideoTracks[0]);
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
// Grab a copy of the offer before it gets passed into the PC.
auto modified_offer =
rtc::MakeUnique<webrtc::JsepSessionDescription>(webrtc::SdpType::kOffer);
modified_offer->Initialize(offer->description()->Copy(), offer->session_id(),
offer->session_version());
EXPECT_TRUE(DoSetLocalDescription(std::move(offer)));
auto senders = pc_->GetSenders();
EXPECT_EQ(2u, senders.size());
EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0]));
// Change the ssrc of the audio and video track.
cricket::MediaContentDescription* desc =
cricket::GetFirstAudioContentDescription(modified_offer->description());
ASSERT_TRUE(desc != NULL);
for (StreamParams& stream : desc->mutable_streams()) {
for (unsigned int& ssrc : stream.ssrcs) {
++ssrc;
}
}
desc =
cricket::GetFirstVideoContentDescription(modified_offer->description());
ASSERT_TRUE(desc != NULL);
for (StreamParams& stream : desc->mutable_streams()) {
for (unsigned int& ssrc : stream.ssrcs) {
++ssrc;
}
}
EXPECT_TRUE(DoSetLocalDescription(std::move(modified_offer)));
senders = pc_->GetSenders();
EXPECT_EQ(2u, senders.size());
EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0]));
// TODO(deadbeef): Once RtpSenders expose parameters, check that the SSRC
// changed.
}
// This tests that the expected behavior occurs if a new session description is
// set with the same tracks, but on a different MediaStream.
// Don't run under Unified Plan since the stream API is not available.
TEST_F(PeerConnectionInterfaceTestPlanB,
SignalSameTracksInSeparateMediaStream) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
rtc::scoped_refptr<StreamCollection> stream_collection =
CreateStreamCollection(2, 1);
pc_->AddStream(stream_collection->at(0));
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
EXPECT_TRUE(DoSetLocalDescription(std::move(offer)));
auto senders = pc_->GetSenders();
EXPECT_EQ(2u, senders.size());
EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0], kStreams[0]));
EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0], kStreams[0]));
// Add a new MediaStream but with the same tracks as in the first stream.
rtc::scoped_refptr<webrtc::MediaStreamInterface> stream_1(
webrtc::MediaStream::Create(kStreams[1]));
stream_1->AddTrack(stream_collection->at(0)->GetVideoTracks()[0]);
stream_1->AddTrack(stream_collection->at(0)->GetAudioTracks()[0]);
pc_->AddStream(stream_1);
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
EXPECT_TRUE(DoSetLocalDescription(std::move(offer)));
auto new_senders = pc_->GetSenders();
// Should be the same senders as before, but with updated stream id.
// Note that this behavior is subject to change in the future.
// We may decide the PC should ignore existing tracks in AddStream.
EXPECT_EQ(senders, new_senders);
EXPECT_TRUE(ContainsSender(new_senders, kAudioTracks[0], kStreams[1]));
EXPECT_TRUE(ContainsSender(new_senders, kVideoTracks[0], kStreams[1]));
}
// This tests that PeerConnectionObserver::OnAddTrack is correctly called.
TEST_P(PeerConnectionInterfaceTest, OnAddTrackCallback) {
FakeConstraints constraints;
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
true);
CreatePeerConnection(&constraints);
CreateAndSetRemoteOffer(kSdpStringWithStream1AudioTrackOnly);
EXPECT_EQ(observer_.num_added_tracks_, 1);
EXPECT_EQ(observer_.last_added_track_label_, kAudioTracks[0]);
// Create and set the updated remote SDP.
CreateAndSetRemoteOffer(kSdpStringWithStream1PlanB);
if (sdp_semantics_ == SdpSemantics::kPlanB) {
EXPECT_EQ(observer_.num_added_tracks_, 2);
} else {
// With Unified Plan, OnAddTrack will fire every time SetRemoteDescription
// is called until the offer/answer exchange is complete. So in this case
// OnAddTrack is fired twice for the first audio track plus the one time
// for the video track.
EXPECT_EQ(observer_.num_added_tracks_, 3);
}
EXPECT_EQ(observer_.last_added_track_label_, kVideoTracks[0]);
}
// Test that when SetConfiguration is called and the configuration is
// changing, the next offer causes an ICE restart.
TEST_P(PeerConnectionInterfaceTest, SetConfigurationCausingIceRestart) {
PeerConnectionInterface::RTCConfiguration config;
config.type = PeerConnectionInterface::kRelay;
// Need to pass default constraints to prevent disabling of DTLS...
FakeConstraints default_constraints;
CreatePeerConnection(config, &default_constraints);
config = pc_->GetConfiguration();
AddAudioTrack(kAudioTracks[0], {kStreamId1});
AddVideoTrack(kVideoTracks[0], {kStreamId1});
// Do initial offer/answer so there's something to restart.
CreateOfferAsLocalDescription();
CreateAnswerAsRemoteDescription(GetSdpStringWithStream1());
// Grab the ufrags.
std::vector<std::string> initial_ufrags = GetUfrags(pc_->local_description());
// Change ICE policy, which should trigger an ICE restart on the next offer.
config.type = PeerConnectionInterface::kAll;
EXPECT_TRUE(pc_->SetConfiguration(config));
CreateOfferAsLocalDescription();
// Grab the new ufrags.
std::vector<std::string> subsequent_ufrags =
GetUfrags(pc_->local_description());
// Sanity check.
EXPECT_EQ(initial_ufrags.size(), subsequent_ufrags.size());
// Check that each ufrag is different.
for (int i = 0; i < static_cast<int>(initial_ufrags.size()); ++i) {
EXPECT_NE(initial_ufrags[i], subsequent_ufrags[i]);
}
}
// Test that when SetConfiguration is called and the configuration *isn't*
// changing, the next offer does *not* cause an ICE restart.
TEST_P(PeerConnectionInterfaceTest, SetConfigurationNotCausingIceRestart) {
PeerConnectionInterface::RTCConfiguration config;
config.type = PeerConnectionInterface::kRelay;
// Need to pass default constraints to prevent disabling of DTLS...
FakeConstraints default_constraints;
CreatePeerConnection(config, &default_constraints);
config = pc_->GetConfiguration();
AddAudioTrack(kAudioTracks[0]);
AddVideoTrack(kVideoTracks[0]);
// Do initial offer/answer so there's something to restart.
CreateOfferAsLocalDescription();
CreateAnswerAsRemoteDescription(GetSdpStringWithStream1());
// Grab the ufrags.
std::vector<std::string> initial_ufrags = GetUfrags(pc_->local_description());
// Call SetConfiguration with a config identical to what the PC was
// constructed with.
EXPECT_TRUE(pc_->SetConfiguration(config));
CreateOfferAsLocalDescription();
// Grab the new ufrags.
std::vector<std::string> subsequent_ufrags =
GetUfrags(pc_->local_description());
EXPECT_EQ(initial_ufrags, subsequent_ufrags);
}
// Test for a weird corner case scenario:
// 1. Audio/video session established.
// 2. SetConfiguration changes ICE config; ICE restart needed.
// 3. ICE restart initiated by remote peer, but only for one m= section.
// 4. Next createOffer should initiate an ICE restart, but only for the other
// m= section; it would be pointless to do an ICE restart for the m= section
// that was already restarted.
TEST_P(PeerConnectionInterfaceTest, SetConfigurationCausingPartialIceRestart) {
PeerConnectionInterface::RTCConfiguration config;
config.type = PeerConnectionInterface::kRelay;
// Need to pass default constraints to prevent disabling of DTLS...
FakeConstraints default_constraints;
CreatePeerConnection(config, &default_constraints);
config = pc_->GetConfiguration();
AddAudioTrack(kAudioTracks[0], {kStreamId1});
AddVideoTrack(kVideoTracks[0], {kStreamId1});
// Do initial offer/answer so there's something to restart.
CreateOfferAsLocalDescription();
CreateAnswerAsRemoteDescription(GetSdpStringWithStream1());
// Change ICE policy, which should set the "needs-ice-restart" flag.
config.type = PeerConnectionInterface::kAll;
EXPECT_TRUE(pc_->SetConfiguration(config));
// Do ICE restart for the first m= section, initiated by remote peer.
std::unique_ptr<webrtc::SessionDescriptionInterface> remote_offer(
webrtc::CreateSessionDescription(SdpType::kOffer,
GetSdpStringWithStream1(), nullptr));
ASSERT_TRUE(remote_offer);
remote_offer->description()->transport_infos()[0].description.ice_ufrag =
"modified";
EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_offer)));
CreateAnswerAsLocalDescription();
// Grab the ufrags.
std::vector<std::string> initial_ufrags = GetUfrags(pc_->local_description());
ASSERT_EQ(2, initial_ufrags.size());
// Create offer and grab the new ufrags.
CreateOfferAsLocalDescription();
std::vector<std::string> subsequent_ufrags =
GetUfrags(pc_->local_description());
ASSERT_EQ(2, subsequent_ufrags.size());
// Ensure that only the ufrag for the second m= section changed.
EXPECT_EQ(initial_ufrags[0], subsequent_ufrags[0]);
EXPECT_NE(initial_ufrags[1], subsequent_ufrags[1]);
}
// Tests that the methods to return current/pending descriptions work as
// expected at different points in the offer/answer exchange. This test does
// one offer/answer exchange as the offerer, then another as the answerer.
TEST_P(PeerConnectionInterfaceTest, CurrentAndPendingDescriptions) {
// This disables DTLS so we can apply an answer to ourselves.
CreatePeerConnection();
// Create initial local offer and get SDP (which will also be used as
// answer/pranswer);
std::unique_ptr<SessionDescriptionInterface> local_offer;
ASSERT_TRUE(DoCreateOffer(&local_offer, nullptr));
std::string sdp;
EXPECT_TRUE(local_offer->ToString(&sdp));
// Set local offer.
SessionDescriptionInterface* local_offer_ptr = local_offer.get();
EXPECT_TRUE(DoSetLocalDescription(std::move(local_offer)));
EXPECT_EQ(local_offer_ptr, pc_->pending_local_description());
EXPECT_EQ(nullptr, pc_->pending_remote_description());
EXPECT_EQ(nullptr, pc_->current_local_description());
EXPECT_EQ(nullptr, pc_->current_remote_description());
// Set remote pranswer.
std::unique_ptr<SessionDescriptionInterface> remote_pranswer(
webrtc::CreateSessionDescription(SdpType::kPrAnswer, sdp));
SessionDescriptionInterface* remote_pranswer_ptr = remote_pranswer.get();
EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_pranswer)));
EXPECT_EQ(local_offer_ptr, pc_->pending_local_description());
EXPECT_EQ(remote_pranswer_ptr, pc_->pending_remote_description());
EXPECT_EQ(nullptr, pc_->current_local_description());
EXPECT_EQ(nullptr, pc_->current_remote_description());
// Set remote answer.
std::unique_ptr<SessionDescriptionInterface> remote_answer(
webrtc::CreateSessionDescription(SdpType::kAnswer, sdp));
SessionDescriptionInterface* remote_answer_ptr = remote_answer.get();
EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_answer)));
EXPECT_EQ(nullptr, pc_->pending_local_description());
EXPECT_EQ(nullptr, pc_->pending_remote_description());
EXPECT_EQ(local_offer_ptr, pc_->current_local_description());
EXPECT_EQ(remote_answer_ptr, pc_->current_remote_description());
// Set remote offer.
std::unique_ptr<SessionDescriptionInterface> remote_offer(
webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
SessionDescriptionInterface* remote_offer_ptr = remote_offer.get();
EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_offer)));
EXPECT_EQ(remote_offer_ptr, pc_->pending_remote_description());
EXPECT_EQ(nullptr, pc_->pending_local_description());
EXPECT_EQ(local_offer_ptr, pc_->current_local_description());
EXPECT_EQ(remote_answer_ptr, pc_->current_remote_description());
// Set local pranswer.
std::unique_ptr<SessionDescriptionInterface> local_pranswer(
webrtc::CreateSessionDescription(SdpType::kPrAnswer, sdp));
SessionDescriptionInterface* local_pranswer_ptr = local_pranswer.get();
EXPECT_TRUE(DoSetLocalDescription(std::move(local_pranswer)));
EXPECT_EQ(remote_offer_ptr, pc_->pending_remote_description());
EXPECT_EQ(local_pranswer_ptr, pc_->pending_local_description());
EXPECT_EQ(local_offer_ptr, pc_->current_local_description());
EXPECT_EQ(remote_answer_ptr, pc_->current_remote_description());
// Set local answer.
std::unique_ptr<SessionDescriptionInterface> local_answer(
webrtc::CreateSessionDescription(SdpType::kAnswer, sdp));
SessionDescriptionInterface* local_answer_ptr = local_answer.get();
EXPECT_TRUE(DoSetLocalDescription(std::move(local_answer)));
EXPECT_EQ(nullptr, pc_->pending_remote_description());
EXPECT_EQ(nullptr, pc_->pending_local_description());
EXPECT_EQ(remote_offer_ptr, pc_->current_remote_description());
EXPECT_EQ(local_answer_ptr, pc_->current_local_description());
}
// Tests that it won't crash when calling StartRtcEventLog or StopRtcEventLog
// after the PeerConnection is closed.
// This version tests the StartRtcEventLog version that receives a file.
TEST_P(PeerConnectionInterfaceTest,
StartAndStopLoggingToFileAfterPeerConnectionClosed) {
CreatePeerConnection();
// The RtcEventLog will be reset when the PeerConnection is closed.
pc_->Close();
auto test_info = ::testing::UnitTest::GetInstance()->current_test_info();
std::string filename = webrtc::test::OutputPath() +
test_info->test_case_name() + test_info->name();
rtc::PlatformFile file = rtc::CreatePlatformFile(filename);
constexpr int64_t max_size_bytes = 1024;
EXPECT_FALSE(pc_->StartRtcEventLog(file, max_size_bytes));
pc_->StopRtcEventLog();
// Cleanup.
rtc::ClosePlatformFile(file);
rtc::RemoveFile(filename);
}
// Tests that it won't crash when calling StartRtcEventLog or StopRtcEventLog
// after the PeerConnection is closed.
// This version tests the StartRtcEventLog version that receives an object
// of type |RtcEventLogOutput|.
TEST_P(PeerConnectionInterfaceTest,
StartAndStopLoggingToOutputAfterPeerConnectionClosed) {
CreatePeerConnection();
// The RtcEventLog will be reset when the PeerConnection is closed.
pc_->Close();
rtc::PlatformFile file = 0;
int64_t max_size_bytes = 1024;
EXPECT_FALSE(pc_->StartRtcEventLog(
rtc::MakeUnique<webrtc::RtcEventLogOutputFile>(file, max_size_bytes),
webrtc::RtcEventLog::kImmediateOutput));
pc_->StopRtcEventLog();
}
// Test that generated offers/answers include "ice-option:trickle".
TEST_P(PeerConnectionInterfaceTest, OffersAndAnswersHaveTrickleIceOption) {
CreatePeerConnection();
// First, create an offer with audio/video.
FakeConstraints constraints;
constraints.SetMandatoryReceiveAudio(true);
constraints.SetMandatoryReceiveVideo(true);
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, &constraints));
cricket::SessionDescription* desc = offer->description();
ASSERT_EQ(2u, desc->transport_infos().size());
EXPECT_TRUE(desc->transport_infos()[0].description.HasOption("trickle"));
EXPECT_TRUE(desc->transport_infos()[1].description.HasOption("trickle"));
// Apply the offer as a remote description, then create an answer.
EXPECT_TRUE(DoSetRemoteDescription(std::move(offer)));
std::unique_ptr<SessionDescriptionInterface> answer;
ASSERT_TRUE(DoCreateAnswer(&answer, &constraints));
desc = answer->description();
ASSERT_EQ(2u, desc->transport_infos().size());
EXPECT_TRUE(desc->transport_infos()[0].description.HasOption("trickle"));
EXPECT_TRUE(desc->transport_infos()[1].description.HasOption("trickle"));
}
// Test that ICE renomination isn't offered if it's not enabled in the PC's
// RTCConfiguration.
TEST_P(PeerConnectionInterfaceTest, IceRenominationNotOffered) {
PeerConnectionInterface::RTCConfiguration config;
config.enable_ice_renomination = false;
CreatePeerConnection(config, nullptr);
AddAudioTrack("foo");
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
cricket::SessionDescription* desc = offer->description();
EXPECT_EQ(1u, desc->transport_infos().size());
EXPECT_FALSE(
desc->transport_infos()[0].description.GetIceParameters().renomination);
}
// Test that the ICE renomination option is present in generated offers/answers
// if it's enabled in the PC's RTCConfiguration.
TEST_P(PeerConnectionInterfaceTest, IceRenominationOptionInOfferAndAnswer) {
PeerConnectionInterface::RTCConfiguration config;
config.enable_ice_renomination = true;
CreatePeerConnection(config, nullptr);
AddAudioTrack("foo");
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
cricket::SessionDescription* desc = offer->description();
EXPECT_EQ(1u, desc->transport_infos().size());
EXPECT_TRUE(
desc->transport_infos()[0].description.GetIceParameters().renomination);
// Set the offer as a remote description, then create an answer and ensure it
// has the renomination flag too.
EXPECT_TRUE(DoSetRemoteDescription(std::move(offer)));
std::unique_ptr<SessionDescriptionInterface> answer;
ASSERT_TRUE(DoCreateAnswer(&answer, nullptr));
desc = answer->description();
EXPECT_EQ(1u, desc->transport_infos().size());
EXPECT_TRUE(
desc->transport_infos()[0].description.GetIceParameters().renomination);
}
// Test that if CreateOffer is called with the deprecated "offer to receive
// audio/video" constraints, they're processed and result in an offer with
// audio/video sections just as if RTCOfferAnswerOptions had been used.
TEST_P(PeerConnectionInterfaceTest, CreateOfferWithOfferToReceiveConstraints) {
CreatePeerConnection();
FakeConstraints constraints;
constraints.SetMandatoryReceiveAudio(true);
constraints.SetMandatoryReceiveVideo(true);
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, &constraints));
cricket::SessionDescription* desc = offer->description();
const cricket::ContentInfo* audio = cricket::GetFirstAudioContent(desc);
const cricket::ContentInfo* video = cricket::GetFirstVideoContent(desc);
ASSERT_NE(nullptr, audio);
ASSERT_NE(nullptr, video);
EXPECT_FALSE(audio->rejected);
EXPECT_FALSE(video->rejected);
}
// Test that if CreateAnswer is called with the deprecated "offer to receive
// audio/video" constraints, they're processed and can be used to reject an
// offered m= section just as can be done with RTCOfferAnswerOptions;
// Don't run under Unified Plan since this behavior is not supported.
TEST_F(PeerConnectionInterfaceTestPlanB,
CreateAnswerWithOfferToReceiveConstraints) {
CreatePeerConnection();
// First, create an offer with audio/video and apply it as a remote
// description.
FakeConstraints constraints;
constraints.SetMandatoryReceiveAudio(true);
constraints.SetMandatoryReceiveVideo(true);
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, &constraints));
EXPECT_TRUE(DoSetRemoteDescription(std::move(offer)));
// Now create answer that rejects audio/video.
constraints.SetMandatoryReceiveAudio(false);
constraints.SetMandatoryReceiveVideo(false);
std::unique_ptr<SessionDescriptionInterface> answer;
ASSERT_TRUE(DoCreateAnswer(&answer, &constraints));
cricket::SessionDescription* desc = answer->description();
const cricket::ContentInfo* audio = cricket::GetFirstAudioContent(desc);
const cricket::ContentInfo* video = cricket::GetFirstVideoContent(desc);
ASSERT_NE(nullptr, audio);
ASSERT_NE(nullptr, video);
EXPECT_TRUE(audio->rejected);
EXPECT_TRUE(video->rejected);
}
// Test that negotiation can succeed with a data channel only, and with the max
// bundle policy. Previously there was a bug that prevented this.
#ifdef HAVE_SCTP
TEST_P(PeerConnectionInterfaceTest, DataChannelOnlyOfferWithMaxBundlePolicy) {
#else
TEST_P(PeerConnectionInterfaceTest,
DISABLED_DataChannelOnlyOfferWithMaxBundlePolicy) {
#endif // HAVE_SCTP
PeerConnectionInterface::RTCConfiguration config;
config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
CreatePeerConnection(config, nullptr);
// First, create an offer with only a data channel and apply it as a remote
// description.
pc_->CreateDataChannel("test", nullptr);
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
EXPECT_TRUE(DoSetRemoteDescription(std::move(offer)));
// Create and set answer as well.
std::unique_ptr<SessionDescriptionInterface> answer;
ASSERT_TRUE(DoCreateAnswer(&answer, nullptr));
EXPECT_TRUE(DoSetLocalDescription(std::move(answer)));
}
TEST_P(PeerConnectionInterfaceTest, SetBitrateWithoutMinSucceeds) {
CreatePeerConnection();
PeerConnectionInterface::BitrateParameters bitrate;
bitrate.current_bitrate_bps = 100000;
EXPECT_TRUE(pc_->SetBitrate(bitrate).ok());
}
TEST_P(PeerConnectionInterfaceTest, SetBitrateNegativeMinFails) {
CreatePeerConnection();
PeerConnectionInterface::BitrateParameters bitrate;
bitrate.min_bitrate_bps = -1;
EXPECT_FALSE(pc_->SetBitrate(bitrate).ok());
}
TEST_P(PeerConnectionInterfaceTest, SetBitrateCurrentLessThanMinFails) {
CreatePeerConnection();
PeerConnectionInterface::BitrateParameters bitrate;
bitrate.min_bitrate_bps = 5;
bitrate.current_bitrate_bps = 3;
EXPECT_FALSE(pc_->SetBitrate(bitrate).ok());
}
TEST_P(PeerConnectionInterfaceTest, SetBitrateCurrentNegativeFails) {
CreatePeerConnection();
PeerConnectionInterface::BitrateParameters bitrate;
bitrate.current_bitrate_bps = -1;
EXPECT_FALSE(pc_->SetBitrate(bitrate).ok());
}
TEST_P(PeerConnectionInterfaceTest, SetBitrateMaxLessThanCurrentFails) {
CreatePeerConnection();
PeerConnectionInterface::BitrateParameters bitrate;
bitrate.current_bitrate_bps = 10;
bitrate.max_bitrate_bps = 8;
EXPECT_FALSE(pc_->SetBitrate(bitrate).ok());
}
TEST_P(PeerConnectionInterfaceTest, SetBitrateMaxLessThanMinFails) {
CreatePeerConnection();
PeerConnectionInterface::BitrateParameters bitrate;
bitrate.min_bitrate_bps = 10;
bitrate.max_bitrate_bps = 8;
EXPECT_FALSE(pc_->SetBitrate(bitrate).ok());
}
TEST_P(PeerConnectionInterfaceTest, SetBitrateMaxNegativeFails) {
CreatePeerConnection();
PeerConnectionInterface::BitrateParameters bitrate;
bitrate.max_bitrate_bps = -1;
EXPECT_FALSE(pc_->SetBitrate(bitrate).ok());
}
// ice_regather_interval_range requires WebRTC to be configured for continual
// gathering already.
TEST_P(PeerConnectionInterfaceTest,
SetIceRegatherIntervalRangeWithoutContinualGatheringFails) {
PeerConnectionInterface::RTCConfiguration config;
config.ice_regather_interval_range.emplace(1000, 2000);
config.continual_gathering_policy =
PeerConnectionInterface::ContinualGatheringPolicy::GATHER_ONCE;
CreatePeerConnectionExpectFail(config);
}
// Ensures that there is no error when ice_regather_interval_range is set with
// continual gathering enabled.
TEST_P(PeerConnectionInterfaceTest,
SetIceRegatherIntervalRangeWithContinualGathering) {
PeerConnectionInterface::RTCConfiguration config;
config.ice_regather_interval_range.emplace(1000, 2000);
config.continual_gathering_policy =
PeerConnectionInterface::ContinualGatheringPolicy::GATHER_CONTINUALLY;
CreatePeerConnection(config, nullptr);
}
// The current bitrate from Call's BitrateConstraintsMask is currently clamped
// by Call's BitrateConstraints, which comes from the SDP or a default value.
// This test checks that a call to SetBitrate with a current bitrate that will
// be clamped succeeds.
TEST_P(PeerConnectionInterfaceTest, SetBitrateCurrentLessThanImplicitMin) {
CreatePeerConnection();
PeerConnectionInterface::BitrateParameters bitrate;
bitrate.current_bitrate_bps = 1;
EXPECT_TRUE(pc_->SetBitrate(bitrate).ok());
}
// The following tests verify that the offer can be created correctly.
TEST_P(PeerConnectionInterfaceTest,
CreateOfferFailsWithInvalidOfferToReceiveAudio) {
RTCOfferAnswerOptions rtc_options;
// Setting offer_to_receive_audio to a value lower than kUndefined or greater
// than kMaxOfferToReceiveMedia should be treated as invalid.
rtc_options.offer_to_receive_audio = RTCOfferAnswerOptions::kUndefined - 1;
CreatePeerConnection();
EXPECT_FALSE(CreateOfferWithOptions(rtc_options));
rtc_options.offer_to_receive_audio =
RTCOfferAnswerOptions::kMaxOfferToReceiveMedia + 1;
EXPECT_FALSE(CreateOfferWithOptions(rtc_options));
}
TEST_P(PeerConnectionInterfaceTest,
CreateOfferFailsWithInvalidOfferToReceiveVideo) {
RTCOfferAnswerOptions rtc_options;
// Setting offer_to_receive_video to a value lower than kUndefined or greater
// than kMaxOfferToReceiveMedia should be treated as invalid.
rtc_options.offer_to_receive_video = RTCOfferAnswerOptions::kUndefined - 1;
CreatePeerConnection();
EXPECT_FALSE(CreateOfferWithOptions(rtc_options));
rtc_options.offer_to_receive_video =
RTCOfferAnswerOptions::kMaxOfferToReceiveMedia + 1;
EXPECT_FALSE(CreateOfferWithOptions(rtc_options));
}
// Test that the audio and video content will be added to an offer if both
// |offer_to_receive_audio| and |offer_to_receive_video| options are 1.
TEST_P(PeerConnectionInterfaceTest, CreateOfferWithAudioVideoOptions) {
RTCOfferAnswerOptions rtc_options;
rtc_options.offer_to_receive_audio = 1;
rtc_options.offer_to_receive_video = 1;
std::unique_ptr<SessionDescriptionInterface> offer;
CreatePeerConnection();
offer = CreateOfferWithOptions(rtc_options);
ASSERT_TRUE(offer);
EXPECT_NE(nullptr, GetFirstAudioContent(offer->description()));
EXPECT_NE(nullptr, GetFirstVideoContent(offer->description()));
}
// Test that only audio content will be added to the offer if only
// |offer_to_receive_audio| options is 1.
TEST_P(PeerConnectionInterfaceTest, CreateOfferWithAudioOnlyOptions) {
RTCOfferAnswerOptions rtc_options;
rtc_options.offer_to_receive_audio = 1;
rtc_options.offer_to_receive_video = 0;
std::unique_ptr<SessionDescriptionInterface> offer;
CreatePeerConnection();
offer = CreateOfferWithOptions(rtc_options);
ASSERT_TRUE(offer);
EXPECT_NE(nullptr, GetFirstAudioContent(offer->description()));
EXPECT_EQ(nullptr, GetFirstVideoContent(offer->description()));
}
// Test that only video content will be added if only |offer_to_receive_video|
// options is 1.
TEST_P(PeerConnectionInterfaceTest, CreateOfferWithVideoOnlyOptions) {
RTCOfferAnswerOptions rtc_options;
rtc_options.offer_to_receive_audio = 0;
rtc_options.offer_to_receive_video = 1;
std::unique_ptr<SessionDescriptionInterface> offer;
CreatePeerConnection();
offer = CreateOfferWithOptions(rtc_options);
ASSERT_TRUE(offer);
EXPECT_EQ(nullptr, GetFirstAudioContent(offer->description()));
EXPECT_NE(nullptr, GetFirstVideoContent(offer->description()));
}
// Test that no media content will be added to the offer if using default
// RTCOfferAnswerOptions.
TEST_P(PeerConnectionInterfaceTest, CreateOfferWithDefaultOfferAnswerOptions) {
RTCOfferAnswerOptions rtc_options;
std::unique_ptr<SessionDescriptionInterface> offer;
CreatePeerConnection();
offer = CreateOfferWithOptions(rtc_options);
ASSERT_TRUE(offer);
EXPECT_EQ(nullptr, GetFirstAudioContent(offer->description()));
EXPECT_EQ(nullptr, GetFirstVideoContent(offer->description()));
}
// Test that if |ice_restart| is true, the ufrag/pwd will change, otherwise
// ufrag/pwd will be the same in the new offer.
TEST_P(PeerConnectionInterfaceTest, CreateOfferWithIceRestart) {
CreatePeerConnection();
RTCOfferAnswerOptions rtc_options;
rtc_options.ice_restart = false;
rtc_options.offer_to_receive_audio = 1;
std::unique_ptr<SessionDescriptionInterface> offer;
CreateOfferWithOptionsAsLocalDescription(&offer, rtc_options);
std::string mid = cricket::GetFirstAudioContent(offer->description())->name;
auto ufrag1 =
offer->description()->GetTransportInfoByName(mid)->description.ice_ufrag;
auto pwd1 =
offer->description()->GetTransportInfoByName(mid)->description.ice_pwd;
// |ice_restart| is false, the ufrag/pwd shouldn't change.
CreateOfferWithOptionsAsLocalDescription(&offer, rtc_options);
auto ufrag2 =
offer->description()->GetTransportInfoByName(mid)->description.ice_ufrag;
auto pwd2 =
offer->description()->GetTransportInfoByName(mid)->description.ice_pwd;
// |ice_restart| is true, the ufrag/pwd should change.
rtc_options.ice_restart = true;
CreateOfferWithOptionsAsLocalDescription(&offer, rtc_options);
auto ufrag3 =
offer->description()->GetTransportInfoByName(mid)->description.ice_ufrag;
auto pwd3 =
offer->description()->GetTransportInfoByName(mid)->description.ice_pwd;
EXPECT_EQ(ufrag1, ufrag2);
EXPECT_EQ(pwd1, pwd2);
EXPECT_NE(ufrag2, ufrag3);
EXPECT_NE(pwd2, pwd3);
}
// Test that if |use_rtp_mux| is true, the bundling will be enabled in the
// offer; if it is false, there won't be any bundle group in the offer.
TEST_P(PeerConnectionInterfaceTest, CreateOfferWithRtpMux) {
RTCOfferAnswerOptions rtc_options;
rtc_options.offer_to_receive_audio = 1;
rtc_options.offer_to_receive_video = 1;
std::unique_ptr<SessionDescriptionInterface> offer;
CreatePeerConnection();
rtc_options.use_rtp_mux = true;
offer = CreateOfferWithOptions(rtc_options);
ASSERT_TRUE(offer);
EXPECT_NE(nullptr, GetFirstAudioContent(offer->description()));
EXPECT_NE(nullptr, GetFirstVideoContent(offer->description()));
EXPECT_TRUE(offer->description()->HasGroup(cricket::GROUP_TYPE_BUNDLE));
rtc_options.use_rtp_mux = false;
offer = CreateOfferWithOptions(rtc_options);
ASSERT_TRUE(offer);
EXPECT_NE(nullptr, GetFirstAudioContent(offer->description()));
EXPECT_NE(nullptr, GetFirstVideoContent(offer->description()));
EXPECT_FALSE(offer->description()->HasGroup(cricket::GROUP_TYPE_BUNDLE));
}
// This test ensures OnRenegotiationNeeded is called when we add track with
// MediaStream -> AddTrack in the same way it is called when we add track with
// PeerConnection -> AddTrack.
// The test can be removed once addStream is rewritten in terms of addTrack
// https://bugs.chromium.org/p/webrtc/issues/detail?id=7815
// Don't run under Unified Plan since the stream API is not available.
TEST_F(PeerConnectionInterfaceTestPlanB,
MediaStreamAddTrackRemoveTrackRenegotiate) {
CreatePeerConnectionWithoutDtls();
rtc::scoped_refptr<MediaStreamInterface> stream(
pc_factory_->CreateLocalMediaStream(kStreamId1));
pc_->AddStream(stream);
rtc::scoped_refptr<AudioTrackInterface> audio_track(
pc_factory_->CreateAudioTrack("audio_track", nullptr));
rtc::scoped_refptr<VideoTrackInterface> video_track(
pc_factory_->CreateVideoTrack(
"video_track", pc_factory_->CreateVideoSource(
std::unique_ptr<cricket::VideoCapturer>(
new cricket::FakeVideoCapturer()))));
stream->AddTrack(audio_track);
EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
observer_.renegotiation_needed_ = false;
stream->AddTrack(video_track);
EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
observer_.renegotiation_needed_ = false;
stream->RemoveTrack(audio_track);
EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
observer_.renegotiation_needed_ = false;
stream->RemoveTrack(video_track);
EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
observer_.renegotiation_needed_ = false;
}
// Tests that an error is returned if a description is applied that has fewer
// media sections than the existing description.
TEST_P(PeerConnectionInterfaceTest,
MediaSectionCountEnforcedForSubsequentOffer) {
CreatePeerConnection();
AddAudioTrack("audio_label");
AddVideoTrack("video_label");
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
EXPECT_TRUE(DoSetRemoteDescription(std::move(offer)));
// A remote offer with fewer media sections should be rejected.
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
offer->description()->contents().pop_back();
offer->description()->contents().pop_back();
ASSERT_TRUE(offer->description()->contents().empty());
EXPECT_FALSE(DoSetRemoteDescription(std::move(offer)));
std::unique_ptr<SessionDescriptionInterface> answer;
ASSERT_TRUE(DoCreateAnswer(&answer, nullptr));
EXPECT_TRUE(DoSetLocalDescription(std::move(answer)));
// A subsequent local offer with fewer media sections should be rejected.
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
offer->description()->contents().pop_back();
offer->description()->contents().pop_back();
ASSERT_TRUE(offer->description()->contents().empty());
EXPECT_FALSE(DoSetLocalDescription(std::move(offer)));
}
INSTANTIATE_TEST_CASE_P(PeerConnectionInterfaceTest,
PeerConnectionInterfaceTest,
Values(SdpSemantics::kPlanB,
SdpSemantics::kUnifiedPlan));
class PeerConnectionMediaConfigTest : public testing::Test {
protected:
void SetUp() override {
pcf_ = PeerConnectionFactoryForTest::CreatePeerConnectionFactoryForTest();
pcf_->Initialize();
}
const cricket::MediaConfig TestCreatePeerConnection(
const PeerConnectionInterface::RTCConfiguration& config,
const MediaConstraintsInterface* constraints) {
rtc::scoped_refptr<PeerConnectionInterface> pc(pcf_->CreatePeerConnection(
config, constraints, nullptr, nullptr, &observer_));
EXPECT_TRUE(pc.get());
return pc->GetConfiguration().media_config;
}
rtc::scoped_refptr<PeerConnectionFactoryForTest> pcf_;
MockPeerConnectionObserver observer_;
};
// This test verifies the default behaviour with no constraints and a
// default RTCConfiguration.
TEST_F(PeerConnectionMediaConfigTest, TestDefaults) {
PeerConnectionInterface::RTCConfiguration config;
FakeConstraints constraints;
const cricket::MediaConfig& media_config =
TestCreatePeerConnection(config, &constraints);
EXPECT_FALSE(media_config.enable_dscp);
EXPECT_TRUE(media_config.video.enable_cpu_adaptation);
EXPECT_TRUE(media_config.video.enable_prerenderer_smoothing);
EXPECT_FALSE(media_config.video.suspend_below_min_bitrate);
EXPECT_FALSE(media_config.video.experiment_cpu_load_estimator);
}
// This test verifies the DSCP constraint is recognized and passed to
// the PeerConnection.
TEST_F(PeerConnectionMediaConfigTest, TestDscpConstraintTrue) {
PeerConnectionInterface::RTCConfiguration config;
FakeConstraints constraints;
constraints.AddOptional(webrtc::MediaConstraintsInterface::kEnableDscp, true);
const cricket::MediaConfig& media_config =
TestCreatePeerConnection(config, &constraints);
EXPECT_TRUE(media_config.enable_dscp);
}
// This test verifies the cpu overuse detection constraint is
// recognized and passed to the PeerConnection.
TEST_F(PeerConnectionMediaConfigTest, TestCpuOveruseConstraintFalse) {
PeerConnectionInterface::RTCConfiguration config;
FakeConstraints constraints;
constraints.AddOptional(
webrtc::MediaConstraintsInterface::kCpuOveruseDetection, false);
const cricket::MediaConfig media_config =
TestCreatePeerConnection(config, &constraints);
EXPECT_FALSE(media_config.video.enable_cpu_adaptation);
}
// This test verifies that the enable_prerenderer_smoothing flag is
// propagated from RTCConfiguration to the PeerConnection.
TEST_F(PeerConnectionMediaConfigTest, TestDisablePrerendererSmoothingTrue) {
PeerConnectionInterface::RTCConfiguration config;
FakeConstraints constraints;
config.set_prerenderer_smoothing(false);
const cricket::MediaConfig& media_config =
TestCreatePeerConnection(config, &constraints);
EXPECT_FALSE(media_config.video.enable_prerenderer_smoothing);
}
// This test verifies that the experiment_cpu_load_estimator flag is
// propagated from RTCConfiguration to the PeerConnection.
TEST_F(PeerConnectionMediaConfigTest, TestEnableExperimentCpuLoadEstimator) {
PeerConnectionInterface::RTCConfiguration config;
FakeConstraints constraints;
config.set_experiment_cpu_load_estimator(true);
const cricket::MediaConfig& media_config =
TestCreatePeerConnection(config, &constraints);
EXPECT_TRUE(media_config.video.experiment_cpu_load_estimator);
}
// This test verifies the suspend below min bitrate constraint is
// recognized and passed to the PeerConnection.
TEST_F(PeerConnectionMediaConfigTest,
TestSuspendBelowMinBitrateConstraintTrue) {
PeerConnectionInterface::RTCConfiguration config;
FakeConstraints constraints;
constraints.AddOptional(
webrtc::MediaConstraintsInterface::kEnableVideoSuspendBelowMinBitrate,
true);
const cricket::MediaConfig media_config =
TestCreatePeerConnection(config, &constraints);
EXPECT_TRUE(media_config.video.suspend_below_min_bitrate);
}
// Tests a few random fields being different.
TEST(RTCConfigurationTest, ComparisonOperators) {
PeerConnectionInterface::RTCConfiguration a;
PeerConnectionInterface::RTCConfiguration b;
EXPECT_EQ(a, b);
PeerConnectionInterface::RTCConfiguration c;
c.servers.push_back(PeerConnectionInterface::IceServer());
EXPECT_NE(a, c);
PeerConnectionInterface::RTCConfiguration d;
d.type = PeerConnectionInterface::kRelay;
EXPECT_NE(a, d);
PeerConnectionInterface::RTCConfiguration e;
e.audio_jitter_buffer_max_packets = 5;
EXPECT_NE(a, e);
PeerConnectionInterface::RTCConfiguration f;
f.ice_connection_receiving_timeout = 1337;
EXPECT_NE(a, f);
PeerConnectionInterface::RTCConfiguration g;
g.disable_ipv6 = true;
EXPECT_NE(a, g);
PeerConnectionInterface::RTCConfiguration h(
PeerConnectionInterface::RTCConfigurationType::kAggressive);
EXPECT_NE(a, h);
}
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