/*
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* libjingle
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* Copyright 2012 Google Inc.
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*
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* Redistribution and use in source and binary forms, with or without
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* modification, are permitted provided that the following conditions are met:
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*
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* 1. Redistributions of source code must retain the above copyright notice,
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* this list of conditions and the following disclaimer.
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* 2. Redistributions in binary form must reproduce the above copyright notice,
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* this list of conditions and the following disclaimer in the documentation
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* and/or other materials provided with the distribution.
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* 3. The name of the author may not be used to endorse or promote products
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* derived from this software without specific prior written permission.
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*
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* THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
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* WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
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* MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
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* EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
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* SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
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* PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
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* OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
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* WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
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* OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
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* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
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*/
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#include <string>
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#include <utility>
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#include "talk/app/webrtc/audiotrack.h"
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#include "talk/app/webrtc/jsepsessiondescription.h"
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#include "talk/app/webrtc/mediastream.h"
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#include "talk/app/webrtc/mediastreaminterface.h"
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#include "talk/app/webrtc/peerconnection.h"
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#include "talk/app/webrtc/peerconnectioninterface.h"
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#include "talk/app/webrtc/rtpreceiverinterface.h"
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#include "talk/app/webrtc/rtpsenderinterface.h"
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#include "talk/app/webrtc/streamcollection.h"
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#ifdef WEBRTC_ANDROID
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#include "talk/app/webrtc/test/androidtestinitializer.h"
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#endif
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#include "talk/app/webrtc/test/fakeconstraints.h"
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#include "talk/app/webrtc/test/fakedtlsidentitystore.h"
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#include "talk/app/webrtc/test/mockpeerconnectionobservers.h"
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#include "talk/app/webrtc/test/testsdpstrings.h"
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#include "talk/app/webrtc/videosource.h"
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#include "talk/app/webrtc/videotrack.h"
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#include "talk/media/base/fakevideocapturer.h"
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#include "talk/media/sctp/sctpdataengine.h"
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#include "talk/session/media/mediasession.h"
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#include "webrtc/base/gunit.h"
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#include "webrtc/base/scoped_ptr.h"
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#include "webrtc/base/ssladapter.h"
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#include "webrtc/base/sslstreamadapter.h"
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#include "webrtc/base/stringutils.h"
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#include "webrtc/base/thread.h"
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#include "webrtc/p2p/client/fakeportallocator.h"
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static const char kStreamLabel1[] = "local_stream_1";
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static const char kStreamLabel2[] = "local_stream_2";
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static const char kStreamLabel3[] = "local_stream_3";
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static const int kDefaultStunPort = 3478;
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static const char kStunAddressOnly[] = "stun:address";
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static const char kStunInvalidPort[] = "stun:address:-1";
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static const char kStunAddressPortAndMore1[] = "stun:address:port:more";
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static const char kStunAddressPortAndMore2[] = "stun:address:port more";
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static const char kTurnIceServerUri[] = "turn:user@turn.example.org";
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static const char kTurnUsername[] = "user";
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static const char kTurnPassword[] = "password";
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static const char kTurnHostname[] = "turn.example.org";
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static const uint32_t kTimeout = 10000U;
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static const char kStreams[][8] = {"stream1", "stream2"};
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static const char kAudioTracks[][32] = {"audiotrack0", "audiotrack1"};
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static const char kVideoTracks[][32] = {"videotrack0", "videotrack1"};
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static const char kRecvonly[] = "recvonly";
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static const char kSendrecv[] = "sendrecv";
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// Reference SDP with a MediaStream with label "stream1" and audio track with
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// id "audio_1" and a video track with id "video_1;
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static const char kSdpStringWithStream1[] =
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"v=0\r\n"
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"o=- 0 0 IN IP4 127.0.0.1\r\n"
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"s=-\r\n"
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"t=0 0\r\n"
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"a=ice-ufrag:e5785931\r\n"
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"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
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"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
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"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
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"m=audio 1 RTP/AVPF 103\r\n"
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"a=mid:audio\r\n"
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"a=sendrecv\r\n"
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"a=rtpmap:103 ISAC/16000\r\n"
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"a=ssrc:1 cname:stream1\r\n"
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"a=ssrc:1 mslabel:stream1\r\n"
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"a=ssrc:1 label:audiotrack0\r\n"
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"m=video 1 RTP/AVPF 120\r\n"
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"a=mid:video\r\n"
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"a=sendrecv\r\n"
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"a=rtpmap:120 VP8/90000\r\n"
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"a=ssrc:2 cname:stream1\r\n"
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"a=ssrc:2 mslabel:stream1\r\n"
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"a=ssrc:2 label:videotrack0\r\n";
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// Reference SDP with two MediaStreams with label "stream1" and "stream2. Each
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// MediaStreams have one audio track and one video track.
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// This uses MSID.
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static const char kSdpStringWithStream1And2[] =
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"v=0\r\n"
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"o=- 0 0 IN IP4 127.0.0.1\r\n"
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"s=-\r\n"
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"t=0 0\r\n"
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"a=ice-ufrag:e5785931\r\n"
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"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
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"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
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"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
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"a=msid-semantic: WMS stream1 stream2\r\n"
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"m=audio 1 RTP/AVPF 103\r\n"
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"a=mid:audio\r\n"
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"a=sendrecv\r\n"
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"a=rtpmap:103 ISAC/16000\r\n"
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"a=ssrc:1 cname:stream1\r\n"
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"a=ssrc:1 msid:stream1 audiotrack0\r\n"
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"a=ssrc:3 cname:stream2\r\n"
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"a=ssrc:3 msid:stream2 audiotrack1\r\n"
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"m=video 1 RTP/AVPF 120\r\n"
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"a=mid:video\r\n"
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"a=sendrecv\r\n"
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"a=rtpmap:120 VP8/0\r\n"
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"a=ssrc:2 cname:stream1\r\n"
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"a=ssrc:2 msid:stream1 videotrack0\r\n"
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"a=ssrc:4 cname:stream2\r\n"
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"a=ssrc:4 msid:stream2 videotrack1\r\n";
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// Reference SDP without MediaStreams. Msid is not supported.
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static const char kSdpStringWithoutStreams[] =
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"v=0\r\n"
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"o=- 0 0 IN IP4 127.0.0.1\r\n"
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"s=-\r\n"
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"t=0 0\r\n"
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"a=ice-ufrag:e5785931\r\n"
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"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
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"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
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"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
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"m=audio 1 RTP/AVPF 103\r\n"
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"a=mid:audio\r\n"
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"a=sendrecv\r\n"
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"a=rtpmap:103 ISAC/16000\r\n"
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"m=video 1 RTP/AVPF 120\r\n"
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"a=mid:video\r\n"
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"a=sendrecv\r\n"
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"a=rtpmap:120 VP8/90000\r\n";
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// Reference SDP without MediaStreams. Msid is supported.
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static const char kSdpStringWithMsidWithoutStreams[] =
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"v=0\r\n"
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"o=- 0 0 IN IP4 127.0.0.1\r\n"
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"s=-\r\n"
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"t=0 0\r\n"
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"a=ice-ufrag:e5785931\r\n"
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"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
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"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
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"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
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"a=msid-semantic: WMS\r\n"
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"m=audio 1 RTP/AVPF 103\r\n"
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"a=mid:audio\r\n"
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"a=sendrecv\r\n"
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"a=rtpmap:103 ISAC/16000\r\n"
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"m=video 1 RTP/AVPF 120\r\n"
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"a=mid:video\r\n"
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"a=sendrecv\r\n"
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"a=rtpmap:120 VP8/90000\r\n";
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// Reference SDP without MediaStreams and audio only.
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static const char kSdpStringWithoutStreamsAudioOnly[] =
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"v=0\r\n"
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"o=- 0 0 IN IP4 127.0.0.1\r\n"
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"s=-\r\n"
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"t=0 0\r\n"
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"a=ice-ufrag:e5785931\r\n"
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"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
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"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
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"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
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"m=audio 1 RTP/AVPF 103\r\n"
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"a=mid:audio\r\n"
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"a=sendrecv\r\n"
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"a=rtpmap:103 ISAC/16000\r\n";
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// Reference SENDONLY SDP without MediaStreams. Msid is not supported.
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static const char kSdpStringSendOnlyWithoutStreams[] =
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"v=0\r\n"
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"o=- 0 0 IN IP4 127.0.0.1\r\n"
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"s=-\r\n"
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"t=0 0\r\n"
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"a=ice-ufrag:e5785931\r\n"
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"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
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"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
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"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
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"m=audio 1 RTP/AVPF 103\r\n"
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"a=mid:audio\r\n"
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"a=sendrecv\r\n"
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"a=sendonly\r\n"
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"a=rtpmap:103 ISAC/16000\r\n"
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"m=video 1 RTP/AVPF 120\r\n"
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"a=mid:video\r\n"
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"a=sendrecv\r\n"
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"a=sendonly\r\n"
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"a=rtpmap:120 VP8/90000\r\n";
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static const char kSdpStringInit[] =
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"v=0\r\n"
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"o=- 0 0 IN IP4 127.0.0.1\r\n"
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"s=-\r\n"
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"t=0 0\r\n"
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"a=ice-ufrag:e5785931\r\n"
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"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
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"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
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"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
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"a=msid-semantic: WMS\r\n";
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static const char kSdpStringAudio[] =
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"m=audio 1 RTP/AVPF 103\r\n"
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"a=mid:audio\r\n"
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"a=sendrecv\r\n"
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"a=rtpmap:103 ISAC/16000\r\n";
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static const char kSdpStringVideo[] =
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"m=video 1 RTP/AVPF 120\r\n"
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"a=mid:video\r\n"
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"a=sendrecv\r\n"
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"a=rtpmap:120 VP8/90000\r\n";
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static const char kSdpStringMs1Audio0[] =
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"a=ssrc:1 cname:stream1\r\n"
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"a=ssrc:1 msid:stream1 audiotrack0\r\n";
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static const char kSdpStringMs1Video0[] =
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"a=ssrc:2 cname:stream1\r\n"
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"a=ssrc:2 msid:stream1 videotrack0\r\n";
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static const char kSdpStringMs1Audio1[] =
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"a=ssrc:3 cname:stream1\r\n"
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"a=ssrc:3 msid:stream1 audiotrack1\r\n";
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static const char kSdpStringMs1Video1[] =
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"a=ssrc:4 cname:stream1\r\n"
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"a=ssrc:4 msid:stream1 videotrack1\r\n";
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#define MAYBE_SKIP_TEST(feature) \
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if (!(feature())) { \
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LOG(LS_INFO) << "Feature disabled... skipping"; \
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return; \
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}
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using rtc::scoped_ptr;
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using rtc::scoped_refptr;
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using webrtc::AudioSourceInterface;
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using webrtc::AudioTrack;
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using webrtc::AudioTrackInterface;
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using webrtc::DataBuffer;
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using webrtc::DataChannelInterface;
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using webrtc::FakeConstraints;
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using webrtc::IceCandidateInterface;
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using webrtc::MediaConstraintsInterface;
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using webrtc::MediaStream;
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using webrtc::MediaStreamInterface;
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using webrtc::MediaStreamTrackInterface;
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using webrtc::MockCreateSessionDescriptionObserver;
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using webrtc::MockDataChannelObserver;
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using webrtc::MockSetSessionDescriptionObserver;
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using webrtc::MockStatsObserver;
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using webrtc::PeerConnectionInterface;
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using webrtc::PeerConnectionObserver;
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using webrtc::RtpReceiverInterface;
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using webrtc::RtpSenderInterface;
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using webrtc::SdpParseError;
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using webrtc::SessionDescriptionInterface;
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using webrtc::StreamCollection;
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using webrtc::StreamCollectionInterface;
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using webrtc::VideoSourceInterface;
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using webrtc::VideoTrack;
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using webrtc::VideoTrackInterface;
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typedef PeerConnectionInterface::RTCOfferAnswerOptions RTCOfferAnswerOptions;
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namespace {
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// Gets the first ssrc of given content type from the ContentInfo.
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bool GetFirstSsrc(const cricket::ContentInfo* content_info, int* ssrc) {
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if (!content_info || !ssrc) {
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return false;
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}
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const cricket::MediaContentDescription* media_desc =
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static_cast<const cricket::MediaContentDescription*>(
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content_info->description);
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if (!media_desc || media_desc->streams().empty()) {
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return false;
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}
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*ssrc = media_desc->streams().begin()->first_ssrc();
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return true;
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}
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void SetSsrcToZero(std::string* sdp) {
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const char kSdpSsrcAtribute[] = "a=ssrc:";
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const char kSdpSsrcAtributeZero[] = "a=ssrc:0";
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size_t ssrc_pos = 0;
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while ((ssrc_pos = sdp->find(kSdpSsrcAtribute, ssrc_pos)) !=
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std::string::npos) {
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size_t end_ssrc = sdp->find(" ", ssrc_pos);
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sdp->replace(ssrc_pos, end_ssrc - ssrc_pos, kSdpSsrcAtributeZero);
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ssrc_pos = end_ssrc;
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}
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}
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// Check if |streams| contains the specified track.
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bool ContainsTrack(const std::vector<cricket::StreamParams>& streams,
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const std::string& stream_label,
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const std::string& track_id) {
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for (const cricket::StreamParams& params : streams) {
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if (params.sync_label == stream_label && params.id == track_id) {
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return true;
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}
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}
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return false;
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}
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// Check if |senders| contains the specified sender, by id.
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bool ContainsSender(
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const std::vector<rtc::scoped_refptr<RtpSenderInterface>>& senders,
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const std::string& id) {
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for (const auto& sender : senders) {
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if (sender->id() == id) {
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return true;
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}
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}
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return false;
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}
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// Create a collection of streams.
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// CreateStreamCollection(1) creates a collection that
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// correspond to kSdpStringWithStream1.
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// CreateStreamCollection(2) correspond to kSdpStringWithStream1And2.
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rtc::scoped_refptr<StreamCollection> CreateStreamCollection(
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int number_of_streams) {
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rtc::scoped_refptr<StreamCollection> local_collection(
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StreamCollection::Create());
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for (int i = 0; i < number_of_streams; ++i) {
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rtc::scoped_refptr<webrtc::MediaStreamInterface> stream(
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webrtc::MediaStream::Create(kStreams[i]));
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// Add a local audio track.
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rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
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webrtc::AudioTrack::Create(kAudioTracks[i], nullptr));
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stream->AddTrack(audio_track);
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// Add a local video track.
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rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
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webrtc::VideoTrack::Create(kVideoTracks[i], nullptr));
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stream->AddTrack(video_track);
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local_collection->AddStream(stream);
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}
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return local_collection;
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}
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// Check equality of StreamCollections.
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bool CompareStreamCollections(StreamCollectionInterface* s1,
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StreamCollectionInterface* s2) {
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if (s1 == nullptr || s2 == nullptr || s1->count() != s2->count()) {
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return false;
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}
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for (size_t i = 0; i != s1->count(); ++i) {
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if (s1->at(i)->label() != s2->at(i)->label()) {
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return false;
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}
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webrtc::AudioTrackVector audio_tracks1 = s1->at(i)->GetAudioTracks();
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webrtc::AudioTrackVector audio_tracks2 = s2->at(i)->GetAudioTracks();
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webrtc::VideoTrackVector video_tracks1 = s1->at(i)->GetVideoTracks();
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webrtc::VideoTrackVector video_tracks2 = s2->at(i)->GetVideoTracks();
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if (audio_tracks1.size() != audio_tracks2.size()) {
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return false;
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}
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for (size_t j = 0; j != audio_tracks1.size(); ++j) {
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if (audio_tracks1[j]->id() != audio_tracks2[j]->id()) {
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return false;
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}
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}
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if (video_tracks1.size() != video_tracks2.size()) {
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return false;
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}
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for (size_t j = 0; j != video_tracks1.size(); ++j) {
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if (video_tracks1[j]->id() != video_tracks2[j]->id()) {
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return false;
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}
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}
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}
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return true;
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}
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class MockPeerConnectionObserver : public PeerConnectionObserver {
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public:
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MockPeerConnectionObserver() : remote_streams_(StreamCollection::Create()) {}
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~MockPeerConnectionObserver() {
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}
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void SetPeerConnectionInterface(PeerConnectionInterface* pc) {
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pc_ = pc;
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if (pc) {
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state_ = pc_->signaling_state();
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}
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}
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virtual void OnSignalingChange(
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PeerConnectionInterface::SignalingState new_state) {
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EXPECT_EQ(pc_->signaling_state(), new_state);
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state_ = new_state;
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}
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// TODO(bemasc): Remove this once callers transition to OnIceGatheringChange.
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virtual void OnStateChange(StateType state_changed) {
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if (pc_.get() == NULL)
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return;
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switch (state_changed) {
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case kSignalingState:
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// OnSignalingChange and OnStateChange(kSignalingState) should always
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// be called approximately simultaneously. To ease testing, we require
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// that they always be called in that order. This check verifies
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// that OnSignalingChange has just been called.
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EXPECT_EQ(pc_->signaling_state(), state_);
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break;
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case kIceState:
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ADD_FAILURE();
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break;
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default:
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ADD_FAILURE();
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break;
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}
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}
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MediaStreamInterface* RemoteStream(const std::string& label) {
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return remote_streams_->find(label);
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}
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StreamCollectionInterface* remote_streams() const { return remote_streams_; }
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virtual void OnAddStream(MediaStreamInterface* stream) {
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last_added_stream_ = stream;
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remote_streams_->AddStream(stream);
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}
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virtual void OnRemoveStream(MediaStreamInterface* stream) {
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last_removed_stream_ = stream;
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remote_streams_->RemoveStream(stream);
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}
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virtual void OnRenegotiationNeeded() {
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renegotiation_needed_ = true;
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}
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virtual void OnDataChannel(DataChannelInterface* data_channel) {
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last_datachannel_ = data_channel;
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}
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virtual void OnIceConnectionChange(
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PeerConnectionInterface::IceConnectionState new_state) {
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EXPECT_EQ(pc_->ice_connection_state(), new_state);
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}
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virtual void OnIceGatheringChange(
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PeerConnectionInterface::IceGatheringState new_state) {
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EXPECT_EQ(pc_->ice_gathering_state(), new_state);
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}
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virtual void OnIceCandidate(const webrtc::IceCandidateInterface* candidate) {
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EXPECT_NE(PeerConnectionInterface::kIceGatheringNew,
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pc_->ice_gathering_state());
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std::string sdp;
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EXPECT_TRUE(candidate->ToString(&sdp));
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EXPECT_LT(0u, sdp.size());
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last_candidate_.reset(webrtc::CreateIceCandidate(candidate->sdp_mid(),
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candidate->sdp_mline_index(), sdp, NULL));
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EXPECT_TRUE(last_candidate_.get() != NULL);
|
}
|
// TODO(bemasc): Remove this once callers transition to OnSignalingChange.
|
virtual void OnIceComplete() {
|
ice_complete_ = true;
|
// OnIceGatheringChange(IceGatheringCompleted) and OnIceComplete() should
|
// be called approximately simultaneously. For ease of testing, this
|
// check additionally requires that they be called in the above order.
|
EXPECT_EQ(PeerConnectionInterface::kIceGatheringComplete,
|
pc_->ice_gathering_state());
|
}
|
|
// Returns the label of the last added stream.
|
// Empty string if no stream have been added.
|
std::string GetLastAddedStreamLabel() {
|
if (last_added_stream_.get())
|
return last_added_stream_->label();
|
return "";
|
}
|
std::string GetLastRemovedStreamLabel() {
|
if (last_removed_stream_.get())
|
return last_removed_stream_->label();
|
return "";
|
}
|
|
scoped_refptr<PeerConnectionInterface> pc_;
|
PeerConnectionInterface::SignalingState state_;
|
scoped_ptr<IceCandidateInterface> last_candidate_;
|
scoped_refptr<DataChannelInterface> last_datachannel_;
|
rtc::scoped_refptr<StreamCollection> remote_streams_;
|
bool renegotiation_needed_ = false;
|
bool ice_complete_ = false;
|
|
private:
|
scoped_refptr<MediaStreamInterface> last_added_stream_;
|
scoped_refptr<MediaStreamInterface> last_removed_stream_;
|
};
|
|
} // namespace
|
|
class PeerConnectionInterfaceTest : public testing::Test {
|
protected:
|
PeerConnectionInterfaceTest() {
|
#ifdef WEBRTC_ANDROID
|
webrtc::InitializeAndroidObjects();
|
#endif
|
}
|
|
virtual void SetUp() {
|
pc_factory_ = webrtc::CreatePeerConnectionFactory(
|
rtc::Thread::Current(), rtc::Thread::Current(), NULL, NULL,
|
NULL);
|
ASSERT_TRUE(pc_factory_.get() != NULL);
|
}
|
|
void CreatePeerConnection() {
|
CreatePeerConnection("", "", NULL);
|
}
|
|
void CreatePeerConnection(webrtc::MediaConstraintsInterface* constraints) {
|
CreatePeerConnection("", "", constraints);
|
}
|
|
void CreatePeerConnection(const std::string& uri,
|
const std::string& password,
|
webrtc::MediaConstraintsInterface* constraints) {
|
PeerConnectionInterface::RTCConfiguration config;
|
PeerConnectionInterface::IceServer server;
|
if (!uri.empty()) {
|
server.uri = uri;
|
server.password = password;
|
config.servers.push_back(server);
|
}
|
|
rtc::scoped_ptr<cricket::FakePortAllocator> port_allocator(
|
new cricket::FakePortAllocator(rtc::Thread::Current(), nullptr));
|
port_allocator_ = port_allocator.get();
|
|
// DTLS does not work in a loopback call, so is disabled for most of the
|
// tests in this file. We only create a FakeIdentityService if the test
|
// explicitly sets the constraint.
|
FakeConstraints default_constraints;
|
if (!constraints) {
|
constraints = &default_constraints;
|
|
default_constraints.AddMandatory(
|
webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, false);
|
}
|
|
scoped_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store;
|
bool dtls;
|
if (FindConstraint(constraints,
|
webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
|
&dtls,
|
nullptr) && dtls) {
|
dtls_identity_store.reset(new FakeDtlsIdentityStore());
|
}
|
pc_ = pc_factory_->CreatePeerConnection(
|
config, constraints, std::move(port_allocator),
|
std::move(dtls_identity_store), &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);
|
|
scoped_refptr<PeerConnectionInterface> pc;
|
pc = pc_factory_->CreatePeerConnection(config, nullptr, nullptr, nullptr,
|
&observer_);
|
EXPECT_EQ(nullptr, pc);
|
}
|
|
void CreatePeerConnectionWithDifferentConfigurations() {
|
CreatePeerConnection(kStunAddressOnly, "", NULL);
|
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);
|
|
CreatePeerConnection(kTurnIceServerUri, kTurnPassword, NULL);
|
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);
|
}
|
|
void AddVideoStream(const std::string& label) {
|
// Create a local stream.
|
scoped_refptr<MediaStreamInterface> stream(
|
pc_factory_->CreateLocalMediaStream(label));
|
scoped_refptr<VideoSourceInterface> video_source(
|
pc_factory_->CreateVideoSource(new cricket::FakeVideoCapturer(), NULL));
|
scoped_refptr<VideoTrackInterface> video_track(
|
pc_factory_->CreateVideoTrack(label + "v0", video_source));
|
stream->AddTrack(video_track.get());
|
EXPECT_TRUE(pc_->AddStream(stream));
|
EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
|
observer_.renegotiation_needed_ = false;
|
}
|
|
void AddVoiceStream(const std::string& label) {
|
// Create a local stream.
|
scoped_refptr<MediaStreamInterface> stream(
|
pc_factory_->CreateLocalMediaStream(label));
|
scoped_refptr<AudioTrackInterface> audio_track(
|
pc_factory_->CreateAudioTrack(label + "a0", NULL));
|
stream->AddTrack(audio_track.get());
|
EXPECT_TRUE(pc_->AddStream(stream));
|
EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
|
observer_.renegotiation_needed_ = false;
|
}
|
|
void AddAudioVideoStream(const std::string& stream_label,
|
const std::string& audio_track_label,
|
const std::string& video_track_label) {
|
// Create a local stream.
|
scoped_refptr<MediaStreamInterface> stream(
|
pc_factory_->CreateLocalMediaStream(stream_label));
|
scoped_refptr<AudioTrackInterface> audio_track(
|
pc_factory_->CreateAudioTrack(
|
audio_track_label, static_cast<AudioSourceInterface*>(NULL)));
|
stream->AddTrack(audio_track.get());
|
scoped_refptr<VideoTrackInterface> video_track(
|
pc_factory_->CreateVideoTrack(video_track_label, NULL));
|
stream->AddTrack(video_track.get());
|
EXPECT_TRUE(pc_->AddStream(stream));
|
EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
|
observer_.renegotiation_needed_ = false;
|
}
|
|
bool DoCreateOfferAnswer(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->release_desc();
|
return observer->result();
|
}
|
|
bool DoCreateOffer(SessionDescriptionInterface** desc,
|
MediaConstraintsInterface* constraints) {
|
return DoCreateOfferAnswer(desc, true, constraints);
|
}
|
|
bool DoCreateAnswer(SessionDescriptionInterface** desc,
|
MediaConstraintsInterface* constraints) {
|
return DoCreateOfferAnswer(desc, false, constraints);
|
}
|
|
bool DoSetSessionDescription(SessionDescriptionInterface* desc, bool local) {
|
rtc::scoped_refptr<MockSetSessionDescriptionObserver>
|
observer(new rtc::RefCountedObject<
|
MockSetSessionDescriptionObserver>());
|
if (local) {
|
pc_->SetLocalDescription(observer, desc);
|
} else {
|
pc_->SetRemoteDescription(observer, desc);
|
}
|
EXPECT_EQ_WAIT(true, observer->called(), kTimeout);
|
return observer->result();
|
}
|
|
bool DoSetLocalDescription(SessionDescriptionInterface* desc) {
|
return DoSetSessionDescription(desc, true);
|
}
|
|
bool DoSetRemoteDescription(SessionDescriptionInterface* desc) {
|
return DoSetSessionDescription(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();
|
}
|
|
void InitiateCall() {
|
CreatePeerConnection();
|
// Create a local stream with audio&video tracks.
|
AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
|
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() {
|
rtc::scoped_ptr<SessionDescriptionInterface> offer;
|
ASSERT_TRUE(DoCreateOffer(offer.use(), nullptr));
|
std::string sdp;
|
EXPECT_TRUE(offer->ToString(&sdp));
|
SessionDescriptionInterface* remote_offer =
|
webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
|
sdp, NULL);
|
EXPECT_TRUE(DoSetRemoteDescription(remote_offer));
|
EXPECT_EQ(PeerConnectionInterface::kHaveRemoteOffer, observer_.state_);
|
}
|
|
void CreateAndSetRemoteOffer(const std::string& sdp) {
|
SessionDescriptionInterface* remote_offer =
|
webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
|
sdp, nullptr);
|
EXPECT_TRUE(DoSetRemoteDescription(remote_offer));
|
EXPECT_EQ(PeerConnectionInterface::kHaveRemoteOffer, observer_.state_);
|
}
|
|
void CreateAnswerAsLocalDescription() {
|
scoped_ptr<SessionDescriptionInterface> answer;
|
ASSERT_TRUE(DoCreateAnswer(answer.use(), 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));
|
SessionDescriptionInterface* new_answer =
|
webrtc::CreateSessionDescription(SessionDescriptionInterface::kAnswer,
|
sdp, NULL);
|
EXPECT_TRUE(DoSetLocalDescription(new_answer));
|
EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_);
|
}
|
|
void CreatePrAnswerAsLocalDescription() {
|
scoped_ptr<SessionDescriptionInterface> answer;
|
ASSERT_TRUE(DoCreateAnswer(answer.use(), nullptr));
|
|
std::string sdp;
|
EXPECT_TRUE(answer->ToString(&sdp));
|
SessionDescriptionInterface* pr_answer =
|
webrtc::CreateSessionDescription(SessionDescriptionInterface::kPrAnswer,
|
sdp, NULL);
|
EXPECT_TRUE(DoSetLocalDescription(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() {
|
rtc::scoped_ptr<SessionDescriptionInterface> offer;
|
ASSERT_TRUE(DoCreateOffer(offer.use(), 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));
|
SessionDescriptionInterface* new_offer =
|
webrtc::CreateSessionDescription(
|
SessionDescriptionInterface::kOffer,
|
sdp, NULL);
|
|
EXPECT_TRUE(DoSetLocalDescription(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_complete_, kTimeout);
|
}
|
|
void CreateAnswerAsRemoteDescription(const std::string& sdp) {
|
webrtc::JsepSessionDescription* answer = new webrtc::JsepSessionDescription(
|
SessionDescriptionInterface::kAnswer);
|
EXPECT_TRUE(answer->Initialize(sdp, NULL));
|
EXPECT_TRUE(DoSetRemoteDescription(answer));
|
EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_);
|
}
|
|
void CreatePrAnswerAndAnswerAsRemoteDescription(const std::string& sdp) {
|
webrtc::JsepSessionDescription* pr_answer =
|
new webrtc::JsepSessionDescription(
|
SessionDescriptionInterface::kPrAnswer);
|
EXPECT_TRUE(pr_answer->Initialize(sdp, NULL));
|
EXPECT_TRUE(DoSetRemoteDescription(pr_answer));
|
EXPECT_EQ(PeerConnectionInterface::kHaveRemotePrAnswer, observer_.state_);
|
webrtc::JsepSessionDescription* answer =
|
new webrtc::JsepSessionDescription(
|
SessionDescriptionInterface::kAnswer);
|
EXPECT_TRUE(answer->Initialize(sdp, NULL));
|
EXPECT_TRUE(DoSetRemoteDescription(answer));
|
EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_);
|
}
|
|
// Help function used for waiting until a the last signaled remote stream has
|
// the same label as |stream_label|. In a few of the tests in this file we
|
// answer with the same session description as we offer and thus we can
|
// check if OnAddStream have been called with the same stream as we offer to
|
// send.
|
void WaitAndVerifyOnAddStream(const std::string& stream_label) {
|
EXPECT_EQ_WAIT(stream_label, observer_.GetLastAddedStreamLabel(), 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 in |desc| and the MediaStream is stored in
|
// |reference_collection_|
|
void CreateSessionDescriptionAndReference(
|
size_t number_of_audio_tracks,
|
size_t number_of_video_tracks,
|
SessionDescriptionInterface** desc) {
|
ASSERT_TRUE(desc != nullptr);
|
ASSERT_LE(number_of_audio_tracks, 2u);
|
ASSERT_LE(number_of_video_tracks, 2u);
|
|
reference_collection_ = StreamCollection::Create();
|
std::string sdp_ms1 = std::string(kSdpStringInit);
|
|
std::string mediastream_label = kStreams[0];
|
|
rtc::scoped_refptr<webrtc::MediaStreamInterface> stream(
|
webrtc::MediaStream::Create(mediastream_label));
|
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);
|
}
|
|
*desc = webrtc::CreateSessionDescription(
|
SessionDescriptionInterface::kOffer, sdp_ms1, nullptr);
|
}
|
|
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, nullptr));
|
ASSERT_TRUE(stream->AddTrack(video_track));
|
}
|
|
cricket::FakePortAllocator* port_allocator_ = nullptr;
|
scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory_;
|
scoped_refptr<PeerConnectionInterface> pc_;
|
MockPeerConnectionObserver observer_;
|
rtc::scoped_refptr<StreamCollection> reference_collection_;
|
};
|
|
TEST_F(PeerConnectionInterfaceTest,
|
CreatePeerConnectionWithDifferentConfigurations) {
|
CreatePeerConnectionWithDifferentConfigurations();
|
}
|
|
TEST_F(PeerConnectionInterfaceTest, AddStreams) {
|
CreatePeerConnection();
|
AddVideoStream(kStreamLabel1);
|
AddVoiceStream(kStreamLabel2);
|
ASSERT_EQ(2u, pc_->local_streams()->count());
|
|
// Test we can add multiple local streams to one peerconnection.
|
scoped_refptr<MediaStreamInterface> stream(
|
pc_factory_->CreateLocalMediaStream(kStreamLabel3));
|
scoped_refptr<AudioTrackInterface> audio_track(
|
pc_factory_->CreateAudioTrack(
|
kStreamLabel3, 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.
|
TEST_F(PeerConnectionInterfaceTest, AddedStreamsPresentInOffer) {
|
CreatePeerConnection();
|
AddAudioVideoStream(kStreamLabel1, "audio_track", "video_track");
|
scoped_ptr<SessionDescriptionInterface> offer;
|
ASSERT_TRUE(DoCreateOffer(offer.accept(), nullptr));
|
|
const cricket::ContentInfo* audio_content =
|
cricket::GetFirstAudioContent(offer->description());
|
const cricket::AudioContentDescription* audio_desc =
|
static_cast<const cricket::AudioContentDescription*>(
|
audio_content->description);
|
EXPECT_TRUE(
|
ContainsTrack(audio_desc->streams(), kStreamLabel1, "audio_track"));
|
|
const cricket::ContentInfo* video_content =
|
cricket::GetFirstVideoContent(offer->description());
|
const cricket::VideoContentDescription* video_desc =
|
static_cast<const cricket::VideoContentDescription*>(
|
video_content->description);
|
EXPECT_TRUE(
|
ContainsTrack(video_desc->streams(), kStreamLabel1, "video_track"));
|
|
// Add another stream and ensure the offer includes both the old and new
|
// streams.
|
AddAudioVideoStream(kStreamLabel2, "audio_track2", "video_track2");
|
ASSERT_TRUE(DoCreateOffer(offer.accept(), nullptr));
|
|
audio_content = cricket::GetFirstAudioContent(offer->description());
|
audio_desc = static_cast<const cricket::AudioContentDescription*>(
|
audio_content->description);
|
EXPECT_TRUE(
|
ContainsTrack(audio_desc->streams(), kStreamLabel1, "audio_track"));
|
EXPECT_TRUE(
|
ContainsTrack(audio_desc->streams(), kStreamLabel2, "audio_track2"));
|
|
video_content = cricket::GetFirstVideoContent(offer->description());
|
video_desc = static_cast<const cricket::VideoContentDescription*>(
|
video_content->description);
|
EXPECT_TRUE(
|
ContainsTrack(video_desc->streams(), kStreamLabel1, "video_track"));
|
EXPECT_TRUE(
|
ContainsTrack(video_desc->streams(), kStreamLabel2, "video_track2"));
|
}
|
|
TEST_F(PeerConnectionInterfaceTest, RemoveStream) {
|
CreatePeerConnection();
|
AddVideoStream(kStreamLabel1);
|
ASSERT_EQ(1u, pc_->local_streams()->count());
|
pc_->RemoveStream(pc_->local_streams()->at(0));
|
EXPECT_EQ(0u, pc_->local_streams()->count());
|
}
|
|
TEST_F(PeerConnectionInterfaceTest, CreateOfferReceiveAnswer) {
|
InitiateCall();
|
WaitAndVerifyOnAddStream(kStreamLabel1);
|
VerifyRemoteRtpHeaderExtensions();
|
}
|
|
TEST_F(PeerConnectionInterfaceTest, CreateOfferReceivePrAnswerAndAnswer) {
|
CreatePeerConnection();
|
AddVideoStream(kStreamLabel1);
|
CreateOfferAsLocalDescription();
|
std::string offer;
|
EXPECT_TRUE(pc_->local_description()->ToString(&offer));
|
CreatePrAnswerAndAnswerAsRemoteDescription(offer);
|
WaitAndVerifyOnAddStream(kStreamLabel1);
|
}
|
|
TEST_F(PeerConnectionInterfaceTest, ReceiveOfferCreateAnswer) {
|
CreatePeerConnection();
|
AddVideoStream(kStreamLabel1);
|
|
CreateOfferAsRemoteDescription();
|
CreateAnswerAsLocalDescription();
|
|
WaitAndVerifyOnAddStream(kStreamLabel1);
|
}
|
|
TEST_F(PeerConnectionInterfaceTest, ReceiveOfferCreatePrAnswerAndAnswer) {
|
CreatePeerConnection();
|
AddVideoStream(kStreamLabel1);
|
|
CreateOfferAsRemoteDescription();
|
CreatePrAnswerAsLocalDescription();
|
CreateAnswerAsLocalDescription();
|
|
WaitAndVerifyOnAddStream(kStreamLabel1);
|
}
|
|
TEST_F(PeerConnectionInterfaceTest, 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(kStreamLabel1);
|
CreateOfferReceiveAnswer();
|
}
|
|
// Tests that after negotiating an audio only call, the respondent can perform a
|
// renegotiation that removes the audio stream.
|
TEST_F(PeerConnectionInterfaceTest, RenegotiateAudioOnly) {
|
CreatePeerConnection();
|
AddVoiceStream(kStreamLabel1);
|
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_F(PeerConnectionInterfaceTest, IceCandidates) {
|
CreatePeerConnection();
|
|
EXPECT_FALSE(pc_->AddIceCandidate(observer_.last_candidate_.get()));
|
// SetRemoteDescription takes ownership of offer.
|
SessionDescriptionInterface* offer = NULL;
|
AddVideoStream(kStreamLabel1);
|
EXPECT_TRUE(DoCreateOffer(&offer, nullptr));
|
EXPECT_TRUE(DoSetRemoteDescription(offer));
|
|
// SetLocalDescription takes ownership of answer.
|
SessionDescriptionInterface* answer = NULL;
|
EXPECT_TRUE(DoCreateAnswer(&answer, nullptr));
|
EXPECT_TRUE(DoSetLocalDescription(answer));
|
|
EXPECT_TRUE_WAIT(observer_.last_candidate_.get() != NULL, kTimeout);
|
EXPECT_TRUE_WAIT(observer_.ice_complete_, kTimeout);
|
|
EXPECT_TRUE(pc_->AddIceCandidate(observer_.last_candidate_.get()));
|
}
|
|
// Test that CreateOffer and CreateAnswer will fail if the track labels are
|
// not unique.
|
TEST_F(PeerConnectionInterfaceTest, CreateOfferAnswerWithInvalidStream) {
|
CreatePeerConnection();
|
// Create a regular offer for the CreateAnswer test later.
|
SessionDescriptionInterface* offer = NULL;
|
EXPECT_TRUE(DoCreateOffer(&offer, nullptr));
|
EXPECT_TRUE(offer != NULL);
|
delete offer;
|
offer = NULL;
|
|
// Create a local stream with audio&video tracks having same label.
|
AddAudioVideoStream(kStreamLabel1, "track_label", "track_label");
|
|
// Test CreateOffer
|
EXPECT_FALSE(DoCreateOffer(&offer, nullptr));
|
|
// Test CreateAnswer
|
SessionDescriptionInterface* answer = NULL;
|
EXPECT_FALSE(DoCreateAnswer(&answer, nullptr));
|
}
|
|
// Test that we will get different SSRCs for each tracks in the offer and answer
|
// we created.
|
TEST_F(PeerConnectionInterfaceTest, SsrcInOfferAnswer) {
|
CreatePeerConnection();
|
// Create a local stream with audio&video tracks having different labels.
|
AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
|
|
// Test CreateOffer
|
scoped_ptr<SessionDescriptionInterface> offer;
|
ASSERT_TRUE(DoCreateOffer(offer.use(), 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(offer.release()));
|
scoped_ptr<SessionDescriptionInterface> answer;
|
ASSERT_TRUE(DoCreateAnswer(answer.use(), 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.
|
TEST_F(PeerConnectionInterfaceTest, AddTrackAfterAddStream) {
|
CreatePeerConnection();
|
// Create audio stream and add to PeerConnection.
|
AddVoiceStream(kStreamLabel1);
|
MediaStreamInterface* stream = pc_->local_streams()->at(0);
|
|
// Add video track to the audio-only stream.
|
scoped_refptr<VideoTrackInterface> video_track(
|
pc_factory_->CreateVideoTrack("video_label", nullptr));
|
stream->AddTrack(video_track.get());
|
|
scoped_ptr<SessionDescriptionInterface> offer;
|
ASSERT_TRUE(DoCreateOffer(offer.use(), 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.
|
TEST_F(PeerConnectionInterfaceTest, RemoveTrackAfterAddStream) {
|
CreatePeerConnection();
|
// Create audio/video stream and add to PeerConnection.
|
AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
|
MediaStreamInterface* stream = pc_->local_streams()->at(0);
|
|
// Remove the video track.
|
stream->RemoveTrack(stream->GetVideoTracks()[0]);
|
|
scoped_ptr<SessionDescriptionInterface> offer;
|
ASSERT_TRUE(DoCreateOffer(offer.use(), nullptr));
|
|
const cricket::MediaContentDescription* video_desc =
|
cricket::GetFirstVideoContentDescription(offer->description());
|
EXPECT_TRUE(video_desc == nullptr);
|
}
|
|
// Test creating a sender with a stream ID, and ensure the ID is populated
|
// in the offer.
|
TEST_F(PeerConnectionInterfaceTest, CreateSenderWithStream) {
|
CreatePeerConnection();
|
pc_->CreateSender("video", kStreamLabel1);
|
|
scoped_ptr<SessionDescriptionInterface> offer;
|
ASSERT_TRUE(DoCreateOffer(offer.use(), nullptr));
|
|
const cricket::MediaContentDescription* video_desc =
|
cricket::GetFirstVideoContentDescription(offer->description());
|
ASSERT_TRUE(video_desc != nullptr);
|
ASSERT_EQ(1u, video_desc->streams().size());
|
EXPECT_EQ(kStreamLabel1, video_desc->streams()[0].sync_label);
|
}
|
|
// Test that we can specify a certain track that we want statistics about.
|
TEST_F(PeerConnectionInterfaceTest, GetStatsForSpecificTrack) {
|
InitiateCall();
|
ASSERT_LT(0u, pc_->remote_streams()->count());
|
ASSERT_LT(0u, pc_->remote_streams()->at(0)->GetAudioTracks().size());
|
scoped_refptr<MediaStreamTrackInterface> remote_audio =
|
pc_->remote_streams()->at(0)->GetAudioTracks()[0];
|
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_->RemoveStream(pc_->local_streams()->at(0));
|
// Do a re-negotiation.
|
CreateOfferReceiveAnswer();
|
|
ASSERT_EQ(0u, pc_->remote_streams()->count());
|
|
// 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_F(PeerConnectionInterfaceTest, GetStatsForVideoTrack) {
|
InitiateCall();
|
ASSERT_LT(0u, pc_->remote_streams()->count());
|
ASSERT_LT(0u, pc_->remote_streams()->at(0)->GetVideoTracks().size());
|
scoped_refptr<MediaStreamTrackInterface> remote_video =
|
pc_->remote_streams()->at(0)->GetVideoTracks()[0];
|
EXPECT_TRUE(DoGetStats(remote_video));
|
}
|
|
// Test that we don't get statistics for an invalid track.
|
// TODO(tommi): Fix this test. DoGetStats will return true
|
// for the unknown track (since GetStats is async), but no
|
// data is returned for the track.
|
TEST_F(PeerConnectionInterfaceTest, DISABLED_GetStatsForInvalidTrack) {
|
InitiateCall();
|
scoped_refptr<AudioTrackInterface> unknown_audio_track(
|
pc_factory_->CreateAudioTrack("unknown track", NULL));
|
EXPECT_FALSE(DoGetStats(unknown_audio_track));
|
}
|
|
// This test setup two RTP data channels in loop back.
|
TEST_F(PeerConnectionInterfaceTest, TestDataChannel) {
|
FakeConstraints constraints;
|
constraints.SetAllowRtpDataChannels();
|
CreatePeerConnection(&constraints);
|
scoped_refptr<DataChannelInterface> data1 =
|
pc_->CreateDataChannel("test1", NULL);
|
scoped_refptr<DataChannelInterface> data2 =
|
pc_->CreateDataChannel("test2", NULL);
|
ASSERT_TRUE(data1 != NULL);
|
rtc::scoped_ptr<MockDataChannelObserver> observer1(
|
new MockDataChannelObserver(data1));
|
rtc::scoped_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_F(PeerConnectionInterfaceTest, TestSendBinaryOnRtpDataChannel) {
|
FakeConstraints constraints;
|
constraints.SetAllowRtpDataChannels();
|
CreatePeerConnection(&constraints);
|
scoped_refptr<DataChannelInterface> data1 =
|
pc_->CreateDataChannel("test1", NULL);
|
scoped_refptr<DataChannelInterface> data2 =
|
pc_->CreateDataChannel("test2", NULL);
|
ASSERT_TRUE(data1 != NULL);
|
rtc::scoped_ptr<MockDataChannelObserver> observer1(
|
new MockDataChannelObserver(data1));
|
rtc::scoped_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::Buffer 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_F(PeerConnectionInterfaceTest, TestSendOnlyDataChannel) {
|
FakeConstraints constraints;
|
constraints.SetAllowRtpDataChannels();
|
CreatePeerConnection(&constraints);
|
scoped_refptr<DataChannelInterface> data1 =
|
pc_->CreateDataChannel("test1", NULL);
|
rtc::scoped_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_F(PeerConnectionInterfaceTest, TestReceiveOnlyDataChannel) {
|
FakeConstraints constraints;
|
constraints.SetAllowRtpDataChannels();
|
CreatePeerConnection(&constraints);
|
|
std::string offer_label = "offer_channel";
|
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_F(PeerConnectionInterfaceTest, CreateReliableRtpDataChannelShouldFail) {
|
FakeConstraints constraints;
|
constraints.SetAllowRtpDataChannels();
|
CreatePeerConnection(&constraints);
|
|
std::string label = "test";
|
webrtc::DataChannelInit config;
|
config.reliable = true;
|
scoped_refptr<DataChannelInterface> channel =
|
pc_->CreateDataChannel(label, &config);
|
EXPECT_TRUE(channel == NULL);
|
}
|
|
// Verifies that duplicated label is not allowed for RTP data channel.
|
TEST_F(PeerConnectionInterfaceTest, RtpDuplicatedLabelNotAllowed) {
|
FakeConstraints constraints;
|
constraints.SetAllowRtpDataChannels();
|
CreatePeerConnection(&constraints);
|
|
std::string label = "test";
|
scoped_refptr<DataChannelInterface> channel =
|
pc_->CreateDataChannel(label, nullptr);
|
EXPECT_NE(channel, nullptr);
|
|
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_F(PeerConnectionInterfaceTest, CreateSctpDataChannel) {
|
FakeConstraints constraints;
|
constraints.SetAllowDtlsSctpDataChannels();
|
CreatePeerConnection(&constraints);
|
|
webrtc::DataChannelInit config;
|
|
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_F(PeerConnectionInterfaceTest,
|
CreateSctpDataChannelShouldFailForInvalidConfig) {
|
FakeConstraints constraints;
|
constraints.SetAllowDtlsSctpDataChannels();
|
CreatePeerConnection(&constraints);
|
|
std::string label = "test";
|
webrtc::DataChannelInit config;
|
config.maxRetransmits = 0;
|
config.maxRetransmitTime = 0;
|
|
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_F(PeerConnectionInterfaceTest,
|
CreateSctpDataChannelWithInvalidIdShouldFail) {
|
FakeConstraints constraints;
|
constraints.SetAllowDtlsSctpDataChannels();
|
CreatePeerConnection(&constraints);
|
|
webrtc::DataChannelInit config;
|
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_F(PeerConnectionInterfaceTest, SctpDuplicatedLabelAllowed) {
|
FakeConstraints constraints;
|
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
|
true);
|
CreatePeerConnection(&constraints);
|
|
std::string label = "test";
|
scoped_refptr<DataChannelInterface> channel =
|
pc_->CreateDataChannel(label, nullptr);
|
EXPECT_NE(channel, nullptr);
|
|
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_F(PeerConnectionInterfaceTest, RenegotiationNeededForNewRtpDataChannel) {
|
FakeConstraints constraints;
|
constraints.SetAllowRtpDataChannels();
|
CreatePeerConnection(&constraints);
|
|
scoped_refptr<DataChannelInterface> dc1 =
|
pc_->CreateDataChannel("test1", NULL);
|
EXPECT_TRUE(observer_.renegotiation_needed_);
|
observer_.renegotiation_needed_ = false;
|
|
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_F(PeerConnectionInterfaceTest, DataChannelCloseWhenPeerConnectionClose) {
|
FakeConstraints constraints;
|
constraints.SetAllowRtpDataChannels();
|
CreatePeerConnection(&constraints);
|
|
scoped_refptr<DataChannelInterface> data1 =
|
pc_->CreateDataChannel("test1", NULL);
|
scoped_refptr<DataChannelInterface> data2 =
|
pc_->CreateDataChannel("test2", NULL);
|
ASSERT_TRUE(data1 != NULL);
|
rtc::scoped_ptr<MockDataChannelObserver> observer1(
|
new MockDataChannelObserver(data1));
|
rtc::scoped_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_F(PeerConnectionInterfaceTest, TestRejectDataChannelInAnswer) {
|
FakeConstraints constraints;
|
constraints.SetAllowRtpDataChannels();
|
CreatePeerConnection(&constraints);
|
|
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));
|
webrtc::JsepSessionDescription* answer = new webrtc::JsepSessionDescription(
|
SessionDescriptionInterface::kAnswer);
|
EXPECT_TRUE(answer->Initialize(sdp, NULL));
|
cricket::ContentInfo* data_info =
|
answer->description()->GetContentByName("data");
|
data_info->rejected = true;
|
|
DoSetRemoteDescription(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_F(PeerConnectionInterfaceTest, ReceiveFireFoxOffer) {
|
MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
|
FakeConstraints constraints;
|
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
|
true);
|
CreatePeerConnection(&constraints);
|
AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
|
SessionDescriptionInterface* desc =
|
webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
|
webrtc::kFireFoxSdpOffer, nullptr);
|
EXPECT_TRUE(DoSetSessionDescription(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_TRUE(content->rejected);
|
#endif
|
}
|
|
// 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_F(PeerConnectionInterfaceTest, ReceiveUpdatedAudioOfferWithBadCodecs) {
|
CreatePeerConnection();
|
AddVoiceStream("audio_label");
|
CreateOfferAsLocalDescription();
|
|
SessionDescriptionInterface* answer =
|
webrtc::CreateSessionDescription(SessionDescriptionInterface::kAnswer,
|
webrtc::kAudioSdp, nullptr);
|
EXPECT_TRUE(DoSetSessionDescription(answer, false));
|
|
SessionDescriptionInterface* updated_offer =
|
webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
|
webrtc::kAudioSdpWithUnsupportedCodecs,
|
nullptr);
|
EXPECT_TRUE(DoSetSessionDescription(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_F(PeerConnectionInterfaceTest, CreateSubsequentRecvOnlyOffer) {
|
FakeConstraints constraints;
|
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
|
true);
|
CreatePeerConnection(&constraints);
|
CreateAndSetRemoteOffer(kSdpStringWithStream1);
|
CreateAnswerAsLocalDescription();
|
|
// At this point we should be receiving stream 1, but not sending anything.
|
// A new offer should be recvonly.
|
SessionDescriptionInterface* offer;
|
DoCreateOffer(&offer, nullptr);
|
|
const cricket::ContentInfo* video_content =
|
cricket::GetFirstVideoContent(offer->description());
|
const cricket::VideoContentDescription* video_desc =
|
static_cast<const cricket::VideoContentDescription*>(
|
video_content->description);
|
ASSERT_EQ(cricket::MD_RECVONLY, video_desc->direction());
|
|
const cricket::ContentInfo* audio_content =
|
cricket::GetFirstAudioContent(offer->description());
|
const cricket::AudioContentDescription* audio_desc =
|
static_cast<const cricket::AudioContentDescription*>(
|
audio_content->description);
|
ASSERT_EQ(cricket::MD_RECVONLY, audio_desc->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_F(PeerConnectionInterfaceTest, CreateSubsequentInactiveOffer) {
|
FakeConstraints constraints;
|
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
|
true);
|
CreatePeerConnection(&constraints);
|
CreateAndSetRemoteOffer(kSdpStringWithStream1);
|
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.
|
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());
|
const cricket::VideoContentDescription* video_desc =
|
static_cast<const cricket::VideoContentDescription*>(
|
video_content->description);
|
ASSERT_EQ(cricket::MD_INACTIVE, video_desc->direction());
|
|
const cricket::ContentInfo* audio_content =
|
cricket::GetFirstAudioContent(offer->description());
|
const cricket::AudioContentDescription* audio_desc =
|
static_cast<const cricket::AudioContentDescription*>(
|
audio_content->description);
|
ASSERT_EQ(cricket::MD_INACTIVE, audio_desc->direction());
|
}
|
|
// Test that we can use SetConfiguration to change the ICE servers of the
|
// PortAllocator.
|
TEST_F(PeerConnectionInterfaceTest, SetConfigurationChangesIceServers) {
|
CreatePeerConnection();
|
|
PeerConnectionInterface::RTCConfiguration config;
|
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 that PeerConnection::Close changes the states to closed and all remote
|
// tracks change state to ended.
|
TEST_F(PeerConnectionInterfaceTest, CloseAndTestStreamsAndStates) {
|
// Initialize a PeerConnection and negotiate local and remote session
|
// description.
|
InitiateCall();
|
ASSERT_EQ(1u, pc_->local_streams()->count());
|
ASSERT_EQ(1u, pc_->remote_streams()->count());
|
|
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());
|
|
EXPECT_EQ(1u, pc_->local_streams()->count());
|
EXPECT_EQ(1u, pc_->remote_streams()->count());
|
|
scoped_refptr<MediaStreamInterface> remote_stream =
|
pc_->remote_streams()->at(0);
|
EXPECT_EQ(MediaStreamTrackInterface::kEnded,
|
remote_stream->GetVideoTracks()[0]->state());
|
EXPECT_EQ(MediaStreamTrackInterface::kEnded,
|
remote_stream->GetAudioTracks()[0]->state());
|
}
|
|
// Test that PeerConnection methods fails gracefully after
|
// PeerConnection::Close has been called.
|
TEST_F(PeerConnectionInterfaceTest, CloseAndTestMethods) {
|
CreatePeerConnection();
|
AddAudioVideoStream(kStreamLabel1, "audio_label", "video_label");
|
CreateOfferAsRemoteDescription();
|
CreateAnswerAsLocalDescription();
|
|
ASSERT_EQ(1u, pc_->local_streams()->count());
|
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);
|
|
rtc::scoped_ptr<SessionDescriptionInterface> offer;
|
EXPECT_TRUE(DoCreateOffer(offer.use(), nullptr));
|
rtc::scoped_ptr<SessionDescriptionInterface> answer;
|
EXPECT_TRUE(DoCreateAnswer(answer.use(), nullptr));
|
|
std::string sdp;
|
ASSERT_TRUE(pc_->remote_description()->ToString(&sdp));
|
SessionDescriptionInterface* remote_offer =
|
webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
|
sdp, NULL);
|
EXPECT_FALSE(DoSetRemoteDescription(remote_offer));
|
|
ASSERT_TRUE(pc_->local_description()->ToString(&sdp));
|
SessionDescriptionInterface* local_offer =
|
webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer,
|
sdp, NULL);
|
EXPECT_FALSE(DoSetLocalDescription(local_offer));
|
}
|
|
// Test that GetStats can still be called after PeerConnection::Close.
|
TEST_F(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_F(PeerConnectionInterfaceTest, UpdateRemoteStreams) {
|
FakeConstraints constraints;
|
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
|
true);
|
CreatePeerConnection(&constraints);
|
CreateAndSetRemoteOffer(kSdpStringWithStream1);
|
|
rtc::scoped_refptr<StreamCollection> reference(CreateStreamCollection(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(kSdpStringWithStream1And2);
|
|
rtc::scoped_refptr<StreamCollection> reference2(CreateStreamCollection(2));
|
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.
|
TEST_F(PeerConnectionInterfaceTest,
|
AddRemoveTrackFromExistingRemoteMediaStream) {
|
FakeConstraints constraints;
|
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
|
true);
|
CreatePeerConnection(&constraints);
|
rtc::scoped_ptr<SessionDescriptionInterface> desc_ms1;
|
CreateSessionDescriptionAndReference(1, 1, desc_ms1.accept());
|
EXPECT_TRUE(DoSetRemoteDescription(desc_ms1.release()));
|
EXPECT_TRUE(CompareStreamCollections(observer_.remote_streams(),
|
reference_collection_));
|
|
// Add extra audio and video tracks to the same MediaStream.
|
rtc::scoped_ptr<SessionDescriptionInterface> desc_ms1_two_tracks;
|
CreateSessionDescriptionAndReference(2, 2, desc_ms1_two_tracks.accept());
|
EXPECT_TRUE(DoSetRemoteDescription(desc_ms1_two_tracks.release()));
|
EXPECT_TRUE(CompareStreamCollections(observer_.remote_streams(),
|
reference_collection_));
|
|
// Remove the extra audio and video tracks.
|
rtc::scoped_ptr<SessionDescriptionInterface> desc_ms2;
|
CreateSessionDescriptionAndReference(1, 1, desc_ms2.accept());
|
EXPECT_TRUE(DoSetRemoteDescription(desc_ms2.release()));
|
EXPECT_TRUE(CompareStreamCollections(observer_.remote_streams(),
|
reference_collection_));
|
}
|
|
// This tests that remote tracks are ended if a local session description is set
|
// that rejects the media content type.
|
TEST_F(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(kSdpStringWithStream1);
|
ASSERT_EQ(1u, observer_.remote_streams()->count());
|
MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
|
ASSERT_EQ(1u, remote_stream->GetVideoTracks().size());
|
ASSERT_EQ(1u, remote_stream->GetAudioTracks().size());
|
|
rtc::scoped_refptr<webrtc::VideoTrackInterface> remote_video =
|
remote_stream->GetVideoTracks()[0];
|
EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, remote_video->state());
|
rtc::scoped_refptr<webrtc::AudioTrackInterface> remote_audio =
|
remote_stream->GetAudioTracks()[0];
|
EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, remote_audio->state());
|
|
rtc::scoped_ptr<SessionDescriptionInterface> local_answer;
|
EXPECT_TRUE(DoCreateAnswer(local_answer.accept(), nullptr));
|
cricket::ContentInfo* video_info =
|
local_answer->description()->GetContentByName("video");
|
video_info->rejected = true;
|
EXPECT_TRUE(DoSetLocalDescription(local_answer.release()));
|
EXPECT_EQ(webrtc::MediaStreamTrackInterface::kEnded, remote_video->state());
|
EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, remote_audio->state());
|
|
// Now create an offer where we reject both video and audio.
|
rtc::scoped_ptr<SessionDescriptionInterface> local_offer;
|
EXPECT_TRUE(DoCreateOffer(local_offer.accept(), 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(local_offer.release()));
|
EXPECT_EQ(webrtc::MediaStreamTrackInterface::kEnded, remote_video->state());
|
EXPECT_EQ(webrtc::MediaStreamTrackInterface::kEnded, remote_audio->state());
|
}
|
|
// 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.
|
TEST_F(PeerConnectionInterfaceTest, RemoveTrackThenRejectMediaContent) {
|
FakeConstraints constraints;
|
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
|
true);
|
CreatePeerConnection(&constraints);
|
CreateAndSetRemoteOffer(kSdpStringWithStream1);
|
MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
|
remote_stream->RemoveTrack(remote_stream->GetVideoTracks()[0]);
|
remote_stream->RemoveTrack(remote_stream->GetAudioTracks()[0]);
|
|
rtc::scoped_ptr<SessionDescriptionInterface> local_answer(
|
webrtc::CreateSessionDescription(SessionDescriptionInterface::kAnswer,
|
kSdpStringWithStream1, 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(local_answer.release()));
|
|
// 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_F(PeerConnectionInterfaceTest, RecvonlyDescriptionDoesntCreateStream) {
|
FakeConstraints constraints;
|
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
|
true);
|
CreatePeerConnection(&constraints);
|
|
std::string recvonly_offer = kSdpStringWithStream1;
|
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.
|
TEST_F(PeerConnectionInterfaceTest, 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->label());
|
|
CreateAndSetRemoteOffer(kSdpStringWithoutStreams);
|
ASSERT_EQ(1u, observer_.remote_streams()->count());
|
ASSERT_EQ(1u, remote_stream->GetAudioTracks().size());
|
EXPECT_EQ("defaulta0", remote_stream->GetAudioTracks()[0]->id());
|
ASSERT_EQ(1u, remote_stream->GetVideoTracks().size());
|
EXPECT_EQ("defaultv0", remote_stream->GetVideoTracks()[0]->id());
|
}
|
|
// This tests that a default MediaStream is created if a remote session
|
// description doesn't contain any streams and media direction is send only.
|
TEST_F(PeerConnectionInterfaceTest,
|
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->label());
|
}
|
|
// This tests that it won't crash when PeerConnection tries to remove
|
// a remote track that as already been removed from the MediaStream.
|
TEST_F(PeerConnectionInterfaceTest, RemoveAlreadyGoneRemoteStream) {
|
FakeConstraints constraints;
|
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
|
true);
|
CreatePeerConnection(&constraints);
|
CreateAndSetRemoteOffer(kSdpStringWithStream1);
|
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.
|
TEST_F(PeerConnectionInterfaceTest,
|
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.
|
TEST_F(PeerConnectionInterfaceTest, 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
|
TEST_F(PeerConnectionInterfaceTest, 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.
|
TEST_F(PeerConnectionInterfaceTest, VerifyDefaultStreamIsNotCreated) {
|
FakeConstraints constraints;
|
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
|
true);
|
CreatePeerConnection(&constraints);
|
CreateAndSetRemoteOffer(kSdpStringWithStream1);
|
rtc::scoped_refptr<StreamCollection> reference(CreateStreamCollection(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.
|
TEST_F(PeerConnectionInterfaceTest, LocalDescriptionChanged) {
|
FakeConstraints constraints;
|
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
|
true);
|
CreatePeerConnection(&constraints);
|
// Create an offer just to ensure we have an identity before we manually
|
// call SetLocalDescription.
|
rtc::scoped_ptr<SessionDescriptionInterface> throwaway;
|
ASSERT_TRUE(DoCreateOffer(throwaway.accept(), nullptr));
|
|
rtc::scoped_ptr<SessionDescriptionInterface> desc_1;
|
CreateSessionDescriptionAndReference(2, 2, desc_1.accept());
|
|
pc_->AddStream(reference_collection_->at(0));
|
EXPECT_TRUE(DoSetLocalDescription(desc_1.release()));
|
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(reference_collection_->at(0));
|
rtc::scoped_ptr<SessionDescriptionInterface> desc_2;
|
CreateSessionDescriptionAndReference(1, 1, desc_2.accept());
|
pc_->AddStream(reference_collection_->at(0));
|
EXPECT_TRUE(DoSetLocalDescription(desc_2.release()));
|
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.
|
TEST_F(PeerConnectionInterfaceTest,
|
AddLocalStreamAfterLocalDescriptionChanged) {
|
FakeConstraints constraints;
|
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
|
true);
|
CreatePeerConnection(&constraints);
|
// Create an offer just to ensure we have an identity before we manually
|
// call SetLocalDescription.
|
rtc::scoped_ptr<SessionDescriptionInterface> throwaway;
|
ASSERT_TRUE(DoCreateOffer(throwaway.accept(), nullptr));
|
|
rtc::scoped_ptr<SessionDescriptionInterface> desc_1;
|
CreateSessionDescriptionAndReference(2, 2, desc_1.accept());
|
|
EXPECT_TRUE(DoSetLocalDescription(desc_1.release()));
|
auto senders = pc_->GetSenders();
|
EXPECT_EQ(0u, senders.size());
|
|
pc_->AddStream(reference_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_F(PeerConnectionInterfaceTest,
|
ChangeSsrcOnTrackInLocalSessionDescription) {
|
FakeConstraints constraints;
|
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
|
true);
|
CreatePeerConnection(&constraints);
|
// Create an offer just to ensure we have an identity before we manually
|
// call SetLocalDescription.
|
rtc::scoped_ptr<SessionDescriptionInterface> throwaway;
|
ASSERT_TRUE(DoCreateOffer(throwaway.accept(), nullptr));
|
|
rtc::scoped_ptr<SessionDescriptionInterface> desc;
|
CreateSessionDescriptionAndReference(1, 1, desc.accept());
|
std::string sdp;
|
desc->ToString(&sdp);
|
|
pc_->AddStream(reference_collection_->at(0));
|
EXPECT_TRUE(DoSetLocalDescription(desc.release()));
|
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.
|
std::string ssrc_org = "a=ssrc:1";
|
std::string ssrc_to = "a=ssrc:97";
|
rtc::replace_substrs(ssrc_org.c_str(), ssrc_org.length(), ssrc_to.c_str(),
|
ssrc_to.length(), &sdp);
|
ssrc_org = "a=ssrc:2";
|
ssrc_to = "a=ssrc:98";
|
rtc::replace_substrs(ssrc_org.c_str(), ssrc_org.length(), ssrc_to.c_str(),
|
ssrc_to.length(), &sdp);
|
rtc::scoped_ptr<SessionDescriptionInterface> updated_desc(
|
webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer, sdp,
|
nullptr));
|
|
EXPECT_TRUE(DoSetLocalDescription(updated_desc.release()));
|
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.
|
TEST_F(PeerConnectionInterfaceTest, SignalSameTracksInSeparateMediaStream) {
|
FakeConstraints constraints;
|
constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp,
|
true);
|
CreatePeerConnection(&constraints);
|
// Create an offer just to ensure we have an identity before we manually
|
// call SetLocalDescription.
|
rtc::scoped_ptr<SessionDescriptionInterface> throwaway;
|
ASSERT_TRUE(DoCreateOffer(throwaway.accept(), nullptr));
|
|
rtc::scoped_ptr<SessionDescriptionInterface> desc;
|
CreateSessionDescriptionAndReference(1, 1, desc.accept());
|
std::string sdp;
|
desc->ToString(&sdp);
|
|
pc_->AddStream(reference_collection_->at(0));
|
EXPECT_TRUE(DoSetLocalDescription(desc.release()));
|
auto senders = pc_->GetSenders();
|
EXPECT_EQ(2u, senders.size());
|
EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
|
EXPECT_TRUE(ContainsSender(senders, kVideoTracks[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(reference_collection_->at(0)->GetVideoTracks()[0]);
|
stream_1->AddTrack(reference_collection_->at(0)->GetAudioTracks()[0]);
|
pc_->AddStream(stream_1);
|
|
// Replace msid in the original SDP.
|
rtc::replace_substrs(kStreams[0], strlen(kStreams[0]), kStreams[1],
|
strlen(kStreams[1]), &sdp);
|
|
rtc::scoped_ptr<SessionDescriptionInterface> updated_desc(
|
webrtc::CreateSessionDescription(SessionDescriptionInterface::kOffer, sdp,
|
nullptr));
|
|
EXPECT_TRUE(DoSetLocalDescription(updated_desc.release()));
|
senders = pc_->GetSenders();
|
EXPECT_EQ(2u, senders.size());
|
EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
|
EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0]));
|
}
|
|
// The following tests verify that session options are created correctly.
|
// TODO(deadbeef): Convert these tests to be more end-to-end. Instead of
|
// "verify options are converted correctly", should be "pass options into
|
// CreateOffer and verify the correct offer is produced."
|
|
TEST(CreateSessionOptionsTest, GetOptionsForOfferWithInvalidAudioOption) {
|
RTCOfferAnswerOptions rtc_options;
|
rtc_options.offer_to_receive_audio = RTCOfferAnswerOptions::kUndefined - 1;
|
|
cricket::MediaSessionOptions options;
|
EXPECT_FALSE(ConvertRtcOptionsForOffer(rtc_options, &options));
|
|
rtc_options.offer_to_receive_audio =
|
RTCOfferAnswerOptions::kMaxOfferToReceiveMedia + 1;
|
EXPECT_FALSE(ConvertRtcOptionsForOffer(rtc_options, &options));
|
}
|
|
TEST(CreateSessionOptionsTest, GetOptionsForOfferWithInvalidVideoOption) {
|
RTCOfferAnswerOptions rtc_options;
|
rtc_options.offer_to_receive_video = RTCOfferAnswerOptions::kUndefined - 1;
|
|
cricket::MediaSessionOptions options;
|
EXPECT_FALSE(ConvertRtcOptionsForOffer(rtc_options, &options));
|
|
rtc_options.offer_to_receive_video =
|
RTCOfferAnswerOptions::kMaxOfferToReceiveMedia + 1;
|
EXPECT_FALSE(ConvertRtcOptionsForOffer(rtc_options, &options));
|
}
|
|
// Test that a MediaSessionOptions is created for an offer if
|
// OfferToReceiveAudio and OfferToReceiveVideo options are set.
|
TEST(CreateSessionOptionsTest, GetMediaSessionOptionsForOfferWithAudioVideo) {
|
RTCOfferAnswerOptions rtc_options;
|
rtc_options.offer_to_receive_audio = 1;
|
rtc_options.offer_to_receive_video = 1;
|
|
cricket::MediaSessionOptions options;
|
EXPECT_TRUE(ConvertRtcOptionsForOffer(rtc_options, &options));
|
EXPECT_TRUE(options.has_audio());
|
EXPECT_TRUE(options.has_video());
|
EXPECT_TRUE(options.bundle_enabled);
|
}
|
|
// Test that a correct MediaSessionOptions is created for an offer if
|
// OfferToReceiveAudio is set.
|
TEST(CreateSessionOptionsTest, GetMediaSessionOptionsForOfferWithAudio) {
|
RTCOfferAnswerOptions rtc_options;
|
rtc_options.offer_to_receive_audio = 1;
|
|
cricket::MediaSessionOptions options;
|
EXPECT_TRUE(ConvertRtcOptionsForOffer(rtc_options, &options));
|
EXPECT_TRUE(options.has_audio());
|
EXPECT_FALSE(options.has_video());
|
EXPECT_TRUE(options.bundle_enabled);
|
}
|
|
// Test that a correct MediaSessionOptions is created for an offer if
|
// the default OfferOptions are used.
|
TEST(CreateSessionOptionsTest, GetDefaultMediaSessionOptionsForOffer) {
|
RTCOfferAnswerOptions rtc_options;
|
|
cricket::MediaSessionOptions options;
|
EXPECT_TRUE(ConvertRtcOptionsForOffer(rtc_options, &options));
|
EXPECT_TRUE(options.has_audio());
|
EXPECT_FALSE(options.has_video());
|
EXPECT_TRUE(options.bundle_enabled);
|
EXPECT_TRUE(options.vad_enabled);
|
EXPECT_FALSE(options.audio_transport_options.ice_restart);
|
EXPECT_FALSE(options.video_transport_options.ice_restart);
|
EXPECT_FALSE(options.data_transport_options.ice_restart);
|
}
|
|
// Test that a correct MediaSessionOptions is created for an offer if
|
// OfferToReceiveVideo is set.
|
TEST(CreateSessionOptionsTest, GetMediaSessionOptionsForOfferWithVideo) {
|
RTCOfferAnswerOptions rtc_options;
|
rtc_options.offer_to_receive_audio = 0;
|
rtc_options.offer_to_receive_video = 1;
|
|
cricket::MediaSessionOptions options;
|
EXPECT_TRUE(ConvertRtcOptionsForOffer(rtc_options, &options));
|
EXPECT_FALSE(options.has_audio());
|
EXPECT_TRUE(options.has_video());
|
EXPECT_TRUE(options.bundle_enabled);
|
}
|
|
// Test that a correct MediaSessionOptions is created for an offer if
|
// UseRtpMux is set to false.
|
TEST(CreateSessionOptionsTest,
|
GetMediaSessionOptionsForOfferWithBundleDisabled) {
|
RTCOfferAnswerOptions rtc_options;
|
rtc_options.offer_to_receive_audio = 1;
|
rtc_options.offer_to_receive_video = 1;
|
rtc_options.use_rtp_mux = false;
|
|
cricket::MediaSessionOptions options;
|
EXPECT_TRUE(ConvertRtcOptionsForOffer(rtc_options, &options));
|
EXPECT_TRUE(options.has_audio());
|
EXPECT_TRUE(options.has_video());
|
EXPECT_FALSE(options.bundle_enabled);
|
}
|
|
// Test that a correct MediaSessionOptions is created to restart ice if
|
// IceRestart is set. It also tests that subsequent MediaSessionOptions don't
|
// have |audio_transport_options.ice_restart| etc. set.
|
TEST(CreateSessionOptionsTest, GetMediaSessionOptionsForOfferWithIceRestart) {
|
RTCOfferAnswerOptions rtc_options;
|
rtc_options.ice_restart = true;
|
|
cricket::MediaSessionOptions options;
|
EXPECT_TRUE(ConvertRtcOptionsForOffer(rtc_options, &options));
|
EXPECT_TRUE(options.audio_transport_options.ice_restart);
|
EXPECT_TRUE(options.video_transport_options.ice_restart);
|
EXPECT_TRUE(options.data_transport_options.ice_restart);
|
|
rtc_options = RTCOfferAnswerOptions();
|
EXPECT_TRUE(ConvertRtcOptionsForOffer(rtc_options, &options));
|
EXPECT_FALSE(options.audio_transport_options.ice_restart);
|
EXPECT_FALSE(options.video_transport_options.ice_restart);
|
EXPECT_FALSE(options.data_transport_options.ice_restart);
|
}
|
|
// Test that the MediaConstraints in an answer don't affect if audio and video
|
// is offered in an offer but that if kOfferToReceiveAudio or
|
// kOfferToReceiveVideo constraints are true in an offer, the media type will be
|
// included in subsequent answers.
|
TEST(CreateSessionOptionsTest, MediaConstraintsInAnswer) {
|
FakeConstraints answer_c;
|
answer_c.SetMandatoryReceiveAudio(true);
|
answer_c.SetMandatoryReceiveVideo(true);
|
|
cricket::MediaSessionOptions answer_options;
|
EXPECT_TRUE(ParseConstraintsForAnswer(&answer_c, &answer_options));
|
EXPECT_TRUE(answer_options.has_audio());
|
EXPECT_TRUE(answer_options.has_video());
|
|
RTCOfferAnswerOptions rtc_offer_options;
|
|
cricket::MediaSessionOptions offer_options;
|
EXPECT_TRUE(ConvertRtcOptionsForOffer(rtc_offer_options, &offer_options));
|
EXPECT_TRUE(offer_options.has_audio());
|
EXPECT_FALSE(offer_options.has_video());
|
|
RTCOfferAnswerOptions updated_rtc_offer_options;
|
updated_rtc_offer_options.offer_to_receive_audio = 1;
|
updated_rtc_offer_options.offer_to_receive_video = 1;
|
|
cricket::MediaSessionOptions updated_offer_options;
|
EXPECT_TRUE(ConvertRtcOptionsForOffer(updated_rtc_offer_options,
|
&updated_offer_options));
|
EXPECT_TRUE(updated_offer_options.has_audio());
|
EXPECT_TRUE(updated_offer_options.has_video());
|
|
// Since an offer has been created with both audio and video, subsequent
|
// offers and answers should contain both audio and video.
|
// Answers will only contain the media types that exist in the offer
|
// regardless of the value of |updated_answer_options.has_audio| and
|
// |updated_answer_options.has_video|.
|
FakeConstraints updated_answer_c;
|
answer_c.SetMandatoryReceiveAudio(false);
|
answer_c.SetMandatoryReceiveVideo(false);
|
|
cricket::MediaSessionOptions updated_answer_options;
|
EXPECT_TRUE(
|
ParseConstraintsForAnswer(&updated_answer_c, &updated_answer_options));
|
EXPECT_TRUE(updated_answer_options.has_audio());
|
EXPECT_TRUE(updated_answer_options.has_video());
|
}
|
