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webrtc / src / 6e1402b25f386af9ed1c42dcb440864c3fb4145f / . / pc / peer_connection_interface_unittest.cc
blob: 06d077a0792a67eca2c52def583bf80d1fb5f446 [file] [log] [blame]
/*
* Copyright 2012 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include <limits.h>
#include <stdint.h>
#include <string.h>
#include <memory>
#include <string>
#include <utility>
#include <vector>
#include "absl/memory/memory.h"
#include "absl/strings/str_replace.h"
#include "absl/types/optional.h"
#include "api/audio/audio_mixer.h"
#include "api/audio_codecs/audio_decoder_factory.h"
#include "api/audio_codecs/audio_encoder_factory.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/call/call_factory_interface.h"
#include "api/create_peerconnection_factory.h"
#include "api/data_channel_interface.h"
#include "api/jsep.h"
#include "api/jsep_session_description.h"
#include "api/media_stream_interface.h"
#include "api/media_types.h"
#include "api/peer_connection_interface.h"
#include "api/rtc_error.h"
#include "api/rtc_event_log_output.h"
#include "api/rtp_receiver_interface.h"
#include "api/rtp_sender_interface.h"
#include "api/rtp_transceiver_interface.h"
#include "api/scoped_refptr.h"
#include "api/video_codecs/builtin_video_decoder_factory.h"
#include "api/video_codecs/builtin_video_encoder_factory.h"
#include "api/video_codecs/video_decoder_factory.h"
#include "api/video_codecs/video_encoder_factory.h"
#include "logging/rtc_event_log/output/rtc_event_log_output_file.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "logging/rtc_event_log/rtc_event_log_factory.h"
#include "logging/rtc_event_log/rtc_event_log_factory_interface.h"
#include "media/base/codec.h"
#include "media/base/media_config.h"
#include "media/base/media_engine.h"
#include "media/base/stream_params.h"
#include "media/engine/webrtc_media_engine.h"
#include "media/sctp/sctp_transport_internal.h"
#include "modules/audio_device/include/audio_device.h"
#include "modules/audio_processing/include/audio_processing.h"
#include "p2p/base/fake_port_allocator.h"
#include "p2p/base/p2p_constants.h"
#include "p2p/base/port.h"
#include "p2p/base/port_allocator.h"
#include "p2p/base/transport_description.h"
#include "p2p/base/transport_info.h"
#include "pc/audio_track.h"
#include "pc/media_session.h"
#include "pc/media_stream.h"
#include "pc/peer_connection.h"
#include "pc/peer_connection_factory.h"
#include "pc/rtc_stats_collector.h"
#include "pc/rtp_sender.h"
#include "pc/session_description.h"
#include "pc/stream_collection.h"
#include "pc/test/fake_audio_capture_module.h"
#include "pc/test/fake_rtc_certificate_generator.h"
#include "pc/test/fake_video_track_source.h"
#include "pc/test/mock_peer_connection_observers.h"
#include "pc/test/test_sdp_strings.h"
#include "pc/video_track.h"
#include "rtc_base/checks.h"
#include "rtc_base/copy_on_write_buffer.h"
#include "rtc_base/gunit.h"
#include "rtc_base/platform_file.h"
#include "rtc_base/ref_counted_object.h"
#include "rtc_base/rtc_certificate_generator.h"
#include "rtc_base/socket_address.h"
#include "rtc_base/thread.h"
#include "rtc_base/time_utils.h"
#include "rtc_base/virtual_socket_server.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/testsupport/file_utils.h"
#ifdef WEBRTC_ANDROID
#include "pc/test/android_test_initializer.h"
#endif
namespace webrtc {
namespace {
static const char kStreamId1[] = "local_stream_1";
static const char kStreamId2[] = "local_stream_2";
static const char kStreamId3[] = "local_stream_3";
static const int kDefaultStunPort = 3478;
static const char kStunAddressOnly[] = "stun:address";
static const char kStunInvalidPort[] = "stun:address:-1";
static const char kStunAddressPortAndMore1[] = "stun:address:port:more";
static const char kStunAddressPortAndMore2[] = "stun:address:port more";
static const char kTurnIceServerUri[] = "turn:user@turn.example.org";
static const char kTurnUsername[] = "user";
static const char kTurnPassword[] = "password";
static const char kTurnHostname[] = "turn.example.org";
static const uint32_t kTimeout = 10000U;
static const char kStreams[][8] = {"stream1", "stream2"};
static const char kAudioTracks[][32] = {"audiotrack0", "audiotrack1"};
static const char kVideoTracks[][32] = {"videotrack0", "videotrack1"};
static const char kRecvonly[] = "recvonly";
static const char kSendrecv[] = "sendrecv";
// Reference SDP with a MediaStream with label "stream1" and audio track with
// id "audio_1" and a video track with id "video_1;
static const char kSdpStringWithStream1PlanB[] =
"v=0\r\n"
"o=- 0 0 IN IP4 127.0.0.1\r\n"
"s=-\r\n"
"t=0 0\r\n"
"m=audio 1 RTP/AVPF 103\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:audio\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:103 ISAC/16000\r\n"
"a=ssrc:1 cname:stream1\r\n"
"a=ssrc:1 mslabel:stream1\r\n"
"a=ssrc:1 label:audiotrack0\r\n"
"m=video 1 RTP/AVPF 120\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:video\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:120 VP8/90000\r\n"
"a=ssrc:2 cname:stream1\r\n"
"a=ssrc:2 mslabel:stream1\r\n"
"a=ssrc:2 label:videotrack0\r\n";
// Same string as above but with the MID changed to the Unified Plan default.
// This is needed so that this SDP can be used as an answer for a Unified Plan
// offer.
static const char kSdpStringWithStream1UnifiedPlan[] =
"v=0\r\n"
"o=- 0 0 IN IP4 127.0.0.1\r\n"
"s=-\r\n"
"t=0 0\r\n"
"m=audio 1 RTP/AVPF 103\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:0\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:103 ISAC/16000\r\n"
"a=ssrc:1 cname:stream1\r\n"
"a=ssrc:1 mslabel:stream1\r\n"
"a=ssrc:1 label:audiotrack0\r\n"
"m=video 1 RTP/AVPF 120\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:1\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:120 VP8/90000\r\n"
"a=ssrc:2 cname:stream1\r\n"
"a=ssrc:2 mslabel:stream1\r\n"
"a=ssrc:2 label:videotrack0\r\n";
// Reference SDP with a MediaStream with label "stream1" and audio track with
// id "audio_1";
static const char kSdpStringWithStream1AudioTrackOnly[] =
"v=0\r\n"
"o=- 0 0 IN IP4 127.0.0.1\r\n"
"s=-\r\n"
"t=0 0\r\n"
"m=audio 1 RTP/AVPF 103\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:audio\r\n"
"a=sendrecv\r\n"
"a=rtpmap:103 ISAC/16000\r\n"
"a=ssrc:1 cname:stream1\r\n"
"a=ssrc:1 mslabel:stream1\r\n"
"a=ssrc:1 label:audiotrack0\r\n"
"a=rtcp-mux\r\n";
// Reference SDP with two MediaStreams with label "stream1" and "stream2. Each
// MediaStreams have one audio track and one video track.
// This uses MSID.
static const char kSdpStringWithStream1And2PlanB[] =
"v=0\r\n"
"o=- 0 0 IN IP4 127.0.0.1\r\n"
"s=-\r\n"
"t=0 0\r\n"
"a=msid-semantic: WMS stream1 stream2\r\n"
"m=audio 1 RTP/AVPF 103\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:audio\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:103 ISAC/16000\r\n"
"a=ssrc:1 cname:stream1\r\n"
"a=ssrc:1 msid:stream1 audiotrack0\r\n"
"a=ssrc:3 cname:stream2\r\n"
"a=ssrc:3 msid:stream2 audiotrack1\r\n"
"m=video 1 RTP/AVPF 120\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:video\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:120 VP8/0\r\n"
"a=ssrc:2 cname:stream1\r\n"
"a=ssrc:2 msid:stream1 videotrack0\r\n"
"a=ssrc:4 cname:stream2\r\n"
"a=ssrc:4 msid:stream2 videotrack1\r\n";
static const char kSdpStringWithStream1And2UnifiedPlan[] =
"v=0\r\n"
"o=- 0 0 IN IP4 127.0.0.1\r\n"
"s=-\r\n"
"t=0 0\r\n"
"a=msid-semantic: WMS stream1 stream2\r\n"
"m=audio 1 RTP/AVPF 103\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:0\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:103 ISAC/16000\r\n"
"a=ssrc:1 cname:stream1\r\n"
"a=ssrc:1 msid:stream1 audiotrack0\r\n"
"m=video 1 RTP/AVPF 120\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:1\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:120 VP8/0\r\n"
"a=ssrc:2 cname:stream1\r\n"
"a=ssrc:2 msid:stream1 videotrack0\r\n"
"m=audio 1 RTP/AVPF 103\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:2\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:103 ISAC/16000\r\n"
"a=ssrc:3 cname:stream2\r\n"
"a=ssrc:3 msid:stream2 audiotrack1\r\n"
"m=video 1 RTP/AVPF 120\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:3\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:120 VP8/0\r\n"
"a=ssrc:4 cname:stream2\r\n"
"a=ssrc:4 msid:stream2 videotrack1\r\n";
// Reference SDP without MediaStreams. Msid is not supported.
static const char kSdpStringWithoutStreams[] =
"v=0\r\n"
"o=- 0 0 IN IP4 127.0.0.1\r\n"
"s=-\r\n"
"t=0 0\r\n"
"m=audio 1 RTP/AVPF 103\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:audio\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:103 ISAC/16000\r\n"
"m=video 1 RTP/AVPF 120\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:video\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:120 VP8/90000\r\n";
// Reference SDP without MediaStreams. Msid is supported.
static const char kSdpStringWithMsidWithoutStreams[] =
"v=0\r\n"
"o=- 0 0 IN IP4 127.0.0.1\r\n"
"s=-\r\n"
"t=0 0\r\n"
"a=msid-semantic: WMS\r\n"
"m=audio 1 RTP/AVPF 103\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:audio\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:103 ISAC/16000\r\n"
"m=video 1 RTP/AVPF 120\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:video\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:120 VP8/90000\r\n";
// Reference SDP without MediaStreams and audio only.
static const char kSdpStringWithoutStreamsAudioOnly[] =
"v=0\r\n"
"o=- 0 0 IN IP4 127.0.0.1\r\n"
"s=-\r\n"
"t=0 0\r\n"
"m=audio 1 RTP/AVPF 103\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:audio\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:103 ISAC/16000\r\n";
// Reference SENDONLY SDP without MediaStreams. Msid is not supported.
static const char kSdpStringSendOnlyWithoutStreams[] =
"v=0\r\n"
"o=- 0 0 IN IP4 127.0.0.1\r\n"
"s=-\r\n"
"t=0 0\r\n"
"m=audio 1 RTP/AVPF 103\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:audio\r\n"
"a=sendrecv\r\n"
"a=sendonly\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:103 ISAC/16000\r\n"
"m=video 1 RTP/AVPF 120\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:video\r\n"
"a=sendrecv\r\n"
"a=sendonly\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:120 VP8/90000\r\n";
static const char kSdpStringInit[] =
"v=0\r\n"
"o=- 0 0 IN IP4 127.0.0.1\r\n"
"s=-\r\n"
"t=0 0\r\n"
"a=msid-semantic: WMS\r\n";
static const char kSdpStringAudio[] =
"m=audio 1 RTP/AVPF 103\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:audio\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:103 ISAC/16000\r\n";
static const char kSdpStringVideo[] =
"m=video 1 RTP/AVPF 120\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=fingerprint:sha-256 58:AB:6E:F5:F1:E4:57:B7:E9:46:F4:86:04:28:F9:A7:ED:"
"BD:AB:AE:40:EF:CE:9A:51:2C:2A:B1:9B:8B:78:84\r\n"
"a=mid:video\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=rtpmap:120 VP8/90000\r\n";
static const char kSdpStringMs1Audio0[] =
"a=ssrc:1 cname:stream1\r\n"
"a=ssrc:1 msid:stream1 audiotrack0\r\n";
static const char kSdpStringMs1Video0[] =
"a=ssrc:2 cname:stream1\r\n"
"a=ssrc:2 msid:stream1 videotrack0\r\n";
static const char kSdpStringMs1Audio1[] =
"a=ssrc:3 cname:stream1\r\n"
"a=ssrc:3 msid:stream1 audiotrack1\r\n";
static const char kSdpStringMs1Video1[] =
"a=ssrc:4 cname:stream1\r\n"
"a=ssrc:4 msid:stream1 videotrack1\r\n";
static const char kDtlsSdesFallbackSdp[] =
"v=0\r\n"
"o=xxxxxx 7 2 IN IP4 0.0.0.0\r\n"
"s=-\r\n"
"c=IN IP4 0.0.0.0\r\n"
"t=0 0\r\n"
"a=group:BUNDLE audio\r\n"
"a=msid-semantic: WMS\r\n"
"m=audio 1 RTP/SAVPF 0\r\n"
"a=sendrecv\r\n"
"a=rtcp-mux\r\n"
"a=mid:audio\r\n"
"a=ssrc:1 cname:stream1\r\n"
"a=ssrc:1 mslabel:stream1\r\n"
"a=ssrc:1 label:audiotrack0\r\n"
"a=ice-ufrag:e5785931\r\n"
"a=ice-pwd:36fb7878390db89481c1d46daa4278d8\r\n"
"a=rtpmap:0 pcmu/8000\r\n"
"a=fingerprint:sha-1 "
"4A:AD:B9:B1:3F:82:18:3B:54:02:12:DF:3E:5D:49:6B:19:E5:7C:AB\r\n"
"a=setup:actpass\r\n"
"a=crypto:0 AES_CM_128_HMAC_SHA1_80 "
"inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj|2^20|1:32 "
"dummy_session_params\r\n";
using ::cricket::StreamParams;
using ::testing::Exactly;
using ::testing::Values;
using RTCConfiguration = PeerConnectionInterface::RTCConfiguration;
using RTCOfferAnswerOptions = PeerConnectionInterface::RTCOfferAnswerOptions;
// Gets the first ssrc of given content type from the ContentInfo.
bool GetFirstSsrc(const cricket::ContentInfo* content_info, int* ssrc) {
if (!content_info || !ssrc) {
return false;
}
const cricket::MediaContentDescription* media_desc =
content_info->media_description();
if (!media_desc || media_desc->streams().empty()) {
return false;
}
*ssrc = media_desc->streams().begin()->first_ssrc();
return true;
}
// Get the ufrags out of an SDP blob. Useful for testing ICE restart
// behavior.
std::vector<std::string> GetUfrags(
const webrtc::SessionDescriptionInterface* desc) {
std::vector<std::string> ufrags;
for (const cricket::TransportInfo& info :
desc->description()->transport_infos()) {
ufrags.push_back(info.description.ice_ufrag);
}
return ufrags;
}
void SetSsrcToZero(std::string* sdp) {
const char kSdpSsrcAtribute[] = "a=ssrc:";
const char kSdpSsrcAtributeZero[] = "a=ssrc:0";
size_t ssrc_pos = 0;
while ((ssrc_pos = sdp->find(kSdpSsrcAtribute, ssrc_pos)) !=
std::string::npos) {
size_t end_ssrc = sdp->find(" ", ssrc_pos);
sdp->replace(ssrc_pos, end_ssrc - ssrc_pos, kSdpSsrcAtributeZero);
ssrc_pos = end_ssrc;
}
}
// Check if |streams| contains the specified track.
bool ContainsTrack(const std::vector<cricket::StreamParams>& streams,
const std::string& stream_id,
const std::string& track_id) {
for (const cricket::StreamParams& params : streams) {
if (params.first_stream_id() == stream_id && params.id == track_id) {
return true;
}
}
return false;
}
// Check if |senders| contains the specified sender, by id.
bool ContainsSender(
const std::vector<rtc::scoped_refptr<RtpSenderInterface>>& senders,
const std::string& id) {
for (const auto& sender : senders) {
if (sender->id() == id) {
return true;
}
}
return false;
}
// Check if |senders| contains the specified sender, by id and stream id.
bool ContainsSender(
const std::vector<rtc::scoped_refptr<RtpSenderInterface>>& senders,
const std::string& id,
const std::string& stream_id) {
for (const auto& sender : senders) {
if (sender->id() == id && sender->stream_ids()[0] == stream_id) {
return true;
}
}
return false;
}
// Create a collection of streams.
// CreateStreamCollection(1) creates a collection that
// correspond to kSdpStringWithStream1.
// CreateStreamCollection(2) correspond to kSdpStringWithStream1And2.
rtc::scoped_refptr<StreamCollection> CreateStreamCollection(
int number_of_streams,
int tracks_per_stream) {
rtc::scoped_refptr<StreamCollection> local_collection(
StreamCollection::Create());
for (int i = 0; i < number_of_streams; ++i) {
rtc::scoped_refptr<webrtc::MediaStreamInterface> stream(
webrtc::MediaStream::Create(kStreams[i]));
for (int j = 0; j < tracks_per_stream; ++j) {
// Add a local audio track.
rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
webrtc::AudioTrack::Create(kAudioTracks[i * tracks_per_stream + j],
nullptr));
stream->AddTrack(audio_track);
// Add a local video track.
rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
webrtc::VideoTrack::Create(kVideoTracks[i * tracks_per_stream + j],
webrtc::FakeVideoTrackSource::Create(),
rtc::Thread::Current()));
stream->AddTrack(video_track);
}
local_collection->AddStream(stream);
}
return local_collection;
}
// Check equality of StreamCollections.
bool CompareStreamCollections(StreamCollectionInterface* s1,
StreamCollectionInterface* s2) {
if (s1 == nullptr || s2 == nullptr || s1->count() != s2->count()) {
return false;
}
for (size_t i = 0; i != s1->count(); ++i) {
if (s1->at(i)->id() != s2->at(i)->id()) {
return false;
}
webrtc::AudioTrackVector audio_tracks1 = s1->at(i)->GetAudioTracks();
webrtc::AudioTrackVector audio_tracks2 = s2->at(i)->GetAudioTracks();
webrtc::VideoTrackVector video_tracks1 = s1->at(i)->GetVideoTracks();
webrtc::VideoTrackVector video_tracks2 = s2->at(i)->GetVideoTracks();
if (audio_tracks1.size() != audio_tracks2.size()) {
return false;
}
for (size_t j = 0; j != audio_tracks1.size(); ++j) {
if (audio_tracks1[j]->id() != audio_tracks2[j]->id()) {
return false;
}
}
if (video_tracks1.size() != video_tracks2.size()) {
return false;
}
for (size_t j = 0; j != video_tracks1.size(); ++j) {
if (video_tracks1[j]->id() != video_tracks2[j]->id()) {
return false;
}
}
}
return true;
}
// Helper class to test Observer.
class MockTrackObserver : public ObserverInterface {
public:
explicit MockTrackObserver(NotifierInterface* notifier)
: notifier_(notifier) {
notifier_->RegisterObserver(this);
}
~MockTrackObserver() { Unregister(); }
void Unregister() {
if (notifier_) {
notifier_->UnregisterObserver(this);
notifier_ = nullptr;
}
}
MOCK_METHOD0(OnChanged, void());
private:
NotifierInterface* notifier_;
};
// The PeerConnectionMediaConfig tests below verify that configuration and
// constraints are propagated into the PeerConnection's MediaConfig. These
// settings are intended for MediaChannel constructors, but that is not
// exercised by these unittest.
class PeerConnectionFactoryForTest : public webrtc::PeerConnectionFactory {
public:
static rtc::scoped_refptr<PeerConnectionFactoryForTest>
CreatePeerConnectionFactoryForTest() {
auto audio_encoder_factory = webrtc::CreateBuiltinAudioEncoderFactory();
auto audio_decoder_factory = webrtc::CreateBuiltinAudioDecoderFactory();
auto video_encoder_factory = webrtc::CreateBuiltinVideoEncoderFactory();
auto video_decoder_factory = webrtc::CreateBuiltinVideoDecoderFactory();
// Use fake audio device module since we're only testing the interface
// level, and using a real one could make tests flaky when run in parallel.
auto media_engine = std::unique_ptr<cricket::MediaEngineInterface>(
cricket::WebRtcMediaEngineFactory::Create(
FakeAudioCaptureModule::Create(), audio_encoder_factory,
audio_decoder_factory, std::move(video_encoder_factory),
std::move(video_decoder_factory), nullptr,
webrtc::AudioProcessingBuilder().Create()));
std::unique_ptr<webrtc::CallFactoryInterface> call_factory =
webrtc::CreateCallFactory();
std::unique_ptr<webrtc::RtcEventLogFactoryInterface> event_log_factory =
webrtc::CreateRtcEventLogFactory();
return new rtc::RefCountedObject<PeerConnectionFactoryForTest>(
rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(),
std::move(media_engine), std::move(call_factory),
std::move(event_log_factory));
}
PeerConnectionFactoryForTest(
rtc::Thread* network_thread,
rtc::Thread* worker_thread,
rtc::Thread* signaling_thread,
std::unique_ptr<cricket::MediaEngineInterface> media_engine,
std::unique_ptr<webrtc::CallFactoryInterface> call_factory,
std::unique_ptr<webrtc::RtcEventLogFactoryInterface> event_log_factory)
: webrtc::PeerConnectionFactory(network_thread,
worker_thread,
signaling_thread,
std::move(media_engine),
std::move(call_factory),
std::move(event_log_factory)) {}
rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
};
// TODO(steveanton): Convert to use the new PeerConnectionWrapper.
class PeerConnectionInterfaceBaseTest : public testing::Test {
protected:
explicit PeerConnectionInterfaceBaseTest(SdpSemantics sdp_semantics)
: vss_(new rtc::VirtualSocketServer()),
main_(vss_.get()),
sdp_semantics_(sdp_semantics) {
#ifdef WEBRTC_ANDROID
webrtc::InitializeAndroidObjects();
#endif
}
virtual void SetUp() {
// Use fake audio capture module since we're only testing the interface
// level, and using a real one could make tests flaky when run in parallel.
fake_audio_capture_module_ = FakeAudioCaptureModule::Create();
pc_factory_ = webrtc::CreatePeerConnectionFactory(
rtc::Thread::Current(), rtc::Thread::Current(), rtc::Thread::Current(),
rtc::scoped_refptr<webrtc::AudioDeviceModule>(
fake_audio_capture_module_),
webrtc::CreateBuiltinAudioEncoderFactory(),
webrtc::CreateBuiltinAudioDecoderFactory(),
webrtc::CreateBuiltinVideoEncoderFactory(),
webrtc::CreateBuiltinVideoDecoderFactory(), nullptr /* audio_mixer */,
nullptr /* audio_processing */);
ASSERT_TRUE(pc_factory_);
pc_factory_for_test_ =
PeerConnectionFactoryForTest::CreatePeerConnectionFactoryForTest();
pc_factory_for_test_->Initialize();
}
void CreatePeerConnection() {
CreatePeerConnection(PeerConnectionInterface::RTCConfiguration());
}
// DTLS does not work in a loopback call, so is disabled for most of the
// tests in this file.
void CreatePeerConnectionWithoutDtls() {
RTCConfiguration config;
config.enable_dtls_srtp = false;
CreatePeerConnection(config);
}
void CreatePeerConnectionWithIceTransportsType(
PeerConnectionInterface::IceTransportsType type) {
PeerConnectionInterface::RTCConfiguration config;
config.type = type;
return CreatePeerConnection(config);
}
void CreatePeerConnectionWithIceServer(const std::string& uri,
const std::string& password) {
PeerConnectionInterface::RTCConfiguration config;
PeerConnectionInterface::IceServer server;
server.uri = uri;
server.password = password;
config.servers.push_back(server);
CreatePeerConnection(config);
}
void CreatePeerConnection(const RTCConfiguration& config) {
std::unique_ptr<cricket::FakePortAllocator> port_allocator(
new cricket::FakePortAllocator(rtc::Thread::Current(), nullptr));
port_allocator_ = port_allocator.get();
// Create certificate generator unless DTLS constraint is explicitly set to
// false.
std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator;
if (config.enable_dtls_srtp.value_or(true)) {
fake_certificate_generator_ = new FakeRTCCertificateGenerator();
cert_generator.reset(fake_certificate_generator_);
}
RTCConfiguration modified_config = config;
modified_config.sdp_semantics = sdp_semantics_;
pc_ = pc_factory_->CreatePeerConnection(
modified_config, std::move(port_allocator), std::move(cert_generator),
&observer_);
ASSERT_TRUE(pc_.get() != NULL);
observer_.SetPeerConnectionInterface(pc_.get());
EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_);
}
void CreatePeerConnectionExpectFail(const std::string& uri) {
PeerConnectionInterface::RTCConfiguration config;
PeerConnectionInterface::IceServer server;
server.uri = uri;
config.servers.push_back(server);
config.sdp_semantics = sdp_semantics_;
rtc::scoped_refptr<PeerConnectionInterface> pc =
pc_factory_->CreatePeerConnection(config, nullptr, nullptr, &observer_);
EXPECT_EQ(nullptr, pc);
}
void CreatePeerConnectionExpectFail(
PeerConnectionInterface::RTCConfiguration config) {
PeerConnectionInterface::IceServer server;
server.uri = kTurnIceServerUri;
server.password = kTurnPassword;
config.servers.push_back(server);
config.sdp_semantics = sdp_semantics_;
rtc::scoped_refptr<PeerConnectionInterface> pc =
pc_factory_->CreatePeerConnection(config, nullptr, nullptr, &observer_);
EXPECT_EQ(nullptr, pc);
}
void CreatePeerConnectionWithDifferentConfigurations() {
CreatePeerConnectionWithIceServer(kStunAddressOnly, "");
EXPECT_EQ(1u, port_allocator_->stun_servers().size());
EXPECT_EQ(0u, port_allocator_->turn_servers().size());
EXPECT_EQ("address", port_allocator_->stun_servers().begin()->hostname());
EXPECT_EQ(kDefaultStunPort,
port_allocator_->stun_servers().begin()->port());
CreatePeerConnectionExpectFail(kStunInvalidPort);
CreatePeerConnectionExpectFail(kStunAddressPortAndMore1);
CreatePeerConnectionExpectFail(kStunAddressPortAndMore2);
CreatePeerConnectionWithIceServer(kTurnIceServerUri, kTurnPassword);
EXPECT_EQ(0u, port_allocator_->stun_servers().size());
EXPECT_EQ(1u, port_allocator_->turn_servers().size());
EXPECT_EQ(kTurnUsername,
port_allocator_->turn_servers()[0].credentials.username);
EXPECT_EQ(kTurnPassword,
port_allocator_->turn_servers()[0].credentials.password);
EXPECT_EQ(kTurnHostname,
port_allocator_->turn_servers()[0].ports[0].address.hostname());
}
void ReleasePeerConnection() {
pc_ = NULL;
observer_.SetPeerConnectionInterface(NULL);
}
rtc::scoped_refptr<VideoTrackInterface> CreateVideoTrack(
const std::string& label) {
return pc_factory_->CreateVideoTrack(label, FakeVideoTrackSource::Create());
}
void AddVideoTrack(const std::string& track_label,
const std::vector<std::string>& stream_ids = {}) {
auto sender_or_error =
pc_->AddTrack(CreateVideoTrack(track_label), stream_ids);
ASSERT_EQ(RTCErrorType::NONE, sender_or_error.error().type());
EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
observer_.renegotiation_needed_ = false;
}
void AddVideoStream(const std::string& label) {
rtc::scoped_refptr<MediaStreamInterface> stream(
pc_factory_->CreateLocalMediaStream(label));
stream->AddTrack(CreateVideoTrack(label + "v0"));
ASSERT_TRUE(pc_->AddStream(stream));
EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
observer_.renegotiation_needed_ = false;
}
rtc::scoped_refptr<AudioTrackInterface> CreateAudioTrack(
const std::string& label) {
return pc_factory_->CreateAudioTrack(label, nullptr);
}
void AddAudioTrack(const std::string& track_label,
const std::vector<std::string>& stream_ids = {}) {
auto sender_or_error =
pc_->AddTrack(CreateAudioTrack(track_label), stream_ids);
ASSERT_EQ(RTCErrorType::NONE, sender_or_error.error().type());
EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
observer_.renegotiation_needed_ = false;
}
void AddAudioStream(const std::string& label) {
rtc::scoped_refptr<MediaStreamInterface> stream(
pc_factory_->CreateLocalMediaStream(label));
stream->AddTrack(CreateAudioTrack(label + "a0"));
ASSERT_TRUE(pc_->AddStream(stream));
EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
observer_.renegotiation_needed_ = false;
}
void AddAudioVideoStream(const std::string& stream_id,
const std::string& audio_track_label,
const std::string& video_track_label) {
// Create a local stream.
rtc::scoped_refptr<MediaStreamInterface> stream(
pc_factory_->CreateLocalMediaStream(stream_id));
stream->AddTrack(CreateAudioTrack(audio_track_label));
stream->AddTrack(CreateVideoTrack(video_track_label));
ASSERT_TRUE(pc_->AddStream(stream));
EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
observer_.renegotiation_needed_ = false;
}
rtc::scoped_refptr<RtpReceiverInterface> GetFirstReceiverOfType(
cricket::MediaType media_type) {
for (auto receiver : pc_->GetReceivers()) {
if (receiver->media_type() == media_type) {
return receiver;
}
}
return nullptr;
}
bool DoCreateOfferAnswer(std::unique_ptr<SessionDescriptionInterface>* desc,
const RTCOfferAnswerOptions* options,
bool offer) {
rtc::scoped_refptr<MockCreateSessionDescriptionObserver> observer(
new rtc::RefCountedObject<MockCreateSessionDescriptionObserver>());
if (offer) {
pc_->CreateOffer(observer, options ? *options : RTCOfferAnswerOptions());
} else {
pc_->CreateAnswer(observer, options ? *options : RTCOfferAnswerOptions());
}
EXPECT_EQ_WAIT(true, observer->called(), kTimeout);
*desc = observer->MoveDescription();
return observer->result();
}
bool DoCreateOffer(std::unique_ptr<SessionDescriptionInterface>* desc,
const RTCOfferAnswerOptions* options) {
return DoCreateOfferAnswer(desc, options, true);
}
bool DoCreateAnswer(std::unique_ptr<SessionDescriptionInterface>* desc,
const RTCOfferAnswerOptions* options) {
return DoCreateOfferAnswer(desc, options, false);
}
bool DoSetSessionDescription(
std::unique_ptr<SessionDescriptionInterface> desc,
bool local) {
rtc::scoped_refptr<MockSetSessionDescriptionObserver> observer(
new rtc::RefCountedObject<MockSetSessionDescriptionObserver>());
if (local) {
pc_->SetLocalDescription(observer, desc.release());
} else {
pc_->SetRemoteDescription(observer, desc.release());
}
if (pc_->signaling_state() != PeerConnectionInterface::kClosed) {
EXPECT_EQ_WAIT(true, observer->called(), kTimeout);
}
return observer->result();
}
bool DoSetLocalDescription(
std::unique_ptr<SessionDescriptionInterface> desc) {
return DoSetSessionDescription(std::move(desc), true);
}
bool DoSetRemoteDescription(
std::unique_ptr<SessionDescriptionInterface> desc) {
return DoSetSessionDescription(std::move(desc), false);
}
// Calls PeerConnection::GetStats and check the return value.
// It does not verify the values in the StatReports since a RTCP packet might
// be required.
bool DoGetStats(MediaStreamTrackInterface* track) {
rtc::scoped_refptr<MockStatsObserver> observer(
new rtc::RefCountedObject<MockStatsObserver>());
if (!pc_->GetStats(observer, track,
PeerConnectionInterface::kStatsOutputLevelStandard))
return false;
EXPECT_TRUE_WAIT(observer->called(), kTimeout);
return observer->called();
}
// Call the standards-compliant GetStats function.
bool DoGetRTCStats() {
rtc::scoped_refptr<webrtc::MockRTCStatsCollectorCallback> callback(
new rtc::RefCountedObject<webrtc::MockRTCStatsCollectorCallback>());
pc_->GetStats(callback);
EXPECT_TRUE_WAIT(callback->called(), kTimeout);
return callback->called();
}
void InitiateCall() {
CreatePeerConnectionWithoutDtls();
// Create a local stream with audio&video tracks.
if (sdp_semantics_ == SdpSemantics::kPlanB) {
AddAudioVideoStream(kStreamId1, "audio_track", "video_track");
} else {
// Unified Plan does not support AddStream, so just add an audio and video
// track.
AddAudioTrack(kAudioTracks[0], {kStreamId1});
AddVideoTrack(kVideoTracks[0], {kStreamId1});
}
CreateOfferReceiveAnswer();
}
// Verify that RTP Header extensions has been negotiated for audio and video.
void VerifyRemoteRtpHeaderExtensions() {
const cricket::MediaContentDescription* desc =
cricket::GetFirstAudioContentDescription(
pc_->remote_description()->description());
ASSERT_TRUE(desc != NULL);
EXPECT_GT(desc->rtp_header_extensions().size(), 0u);
desc = cricket::GetFirstVideoContentDescription(
pc_->remote_description()->description());
ASSERT_TRUE(desc != NULL);
EXPECT_GT(desc->rtp_header_extensions().size(), 0u);
}
void CreateOfferAsRemoteDescription() {
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
std::string sdp;
EXPECT_TRUE(offer->ToString(&sdp));
std::unique_ptr<SessionDescriptionInterface> remote_offer(
webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_offer)));
EXPECT_EQ(PeerConnectionInterface::kHaveRemoteOffer, observer_.state_);
}
void CreateAndSetRemoteOffer(const std::string& sdp) {
std::unique_ptr<SessionDescriptionInterface> remote_offer(
webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_offer)));
EXPECT_EQ(PeerConnectionInterface::kHaveRemoteOffer, observer_.state_);
}
void CreateAnswerAsLocalDescription() {
std::unique_ptr<SessionDescriptionInterface> answer;
ASSERT_TRUE(DoCreateAnswer(&answer, nullptr));
// TODO(perkj): Currently SetLocalDescription fails if any parameters in an
// audio codec change, even if the parameter has nothing to do with
// receiving. Not all parameters are serialized to SDP.
// Since CreatePrAnswerAsLocalDescription serialize/deserialize
// the SessionDescription, it is necessary to do that here to in order to
// get ReceiveOfferCreatePrAnswerAndAnswer and RenegotiateAudioOnly to pass.
// https://code.google.com/p/webrtc/issues/detail?id=1356
std::string sdp;
EXPECT_TRUE(answer->ToString(&sdp));
std::unique_ptr<SessionDescriptionInterface> new_answer(
webrtc::CreateSessionDescription(SdpType::kAnswer, sdp));
EXPECT_TRUE(DoSetLocalDescription(std::move(new_answer)));
EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_);
}
void CreatePrAnswerAsLocalDescription() {
std::unique_ptr<SessionDescriptionInterface> answer;
ASSERT_TRUE(DoCreateAnswer(&answer, nullptr));
std::string sdp;
EXPECT_TRUE(answer->ToString(&sdp));
std::unique_ptr<SessionDescriptionInterface> pr_answer(
webrtc::CreateSessionDescription(SdpType::kPrAnswer, sdp));
EXPECT_TRUE(DoSetLocalDescription(std::move(pr_answer)));
EXPECT_EQ(PeerConnectionInterface::kHaveLocalPrAnswer, observer_.state_);
}
void CreateOfferReceiveAnswer() {
CreateOfferAsLocalDescription();
std::string sdp;
EXPECT_TRUE(pc_->local_description()->ToString(&sdp));
CreateAnswerAsRemoteDescription(sdp);
}
void CreateOfferAsLocalDescription() {
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
// TODO(perkj): Currently SetLocalDescription fails if any parameters in an
// audio codec change, even if the parameter has nothing to do with
// receiving. Not all parameters are serialized to SDP.
// Since CreatePrAnswerAsLocalDescription serialize/deserialize
// the SessionDescription, it is necessary to do that here to in order to
// get ReceiveOfferCreatePrAnswerAndAnswer and RenegotiateAudioOnly to pass.
// https://code.google.com/p/webrtc/issues/detail?id=1356
std::string sdp;
EXPECT_TRUE(offer->ToString(&sdp));
std::unique_ptr<SessionDescriptionInterface> new_offer(
webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
EXPECT_TRUE(DoSetLocalDescription(std::move(new_offer)));
EXPECT_EQ(PeerConnectionInterface::kHaveLocalOffer, observer_.state_);
// Wait for the ice_complete message, so that SDP will have candidates.
EXPECT_TRUE_WAIT(observer_.ice_gathering_complete_, kTimeout);
}
void CreateAnswerAsRemoteDescription(const std::string& sdp) {
std::unique_ptr<SessionDescriptionInterface> answer(
webrtc::CreateSessionDescription(SdpType::kAnswer, sdp));
ASSERT_TRUE(answer);
EXPECT_TRUE(DoSetRemoteDescription(std::move(answer)));
EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_);
}
void CreatePrAnswerAndAnswerAsRemoteDescription(const std::string& sdp) {
std::unique_ptr<SessionDescriptionInterface> pr_answer(
webrtc::CreateSessionDescription(SdpType::kPrAnswer, sdp));
ASSERT_TRUE(pr_answer);
EXPECT_TRUE(DoSetRemoteDescription(std::move(pr_answer)));
EXPECT_EQ(PeerConnectionInterface::kHaveRemotePrAnswer, observer_.state_);
std::unique_ptr<SessionDescriptionInterface> answer(
webrtc::CreateSessionDescription(SdpType::kAnswer, sdp));
ASSERT_TRUE(answer);
EXPECT_TRUE(DoSetRemoteDescription(std::move(answer)));
EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_);
}
// Waits until a remote stream with the given id is signaled. This helper
// function will verify both OnAddTrack and OnAddStream (Plan B only) are
// called with the given stream id and expected number of tracks.
void WaitAndVerifyOnAddStream(const std::string& stream_id,
int expected_num_tracks) {
// Verify that both OnAddStream and OnAddTrack are called.
EXPECT_EQ_WAIT(stream_id, observer_.GetLastAddedStreamId(), kTimeout);
EXPECT_EQ_WAIT(expected_num_tracks,
observer_.CountAddTrackEventsForStream(stream_id), kTimeout);
}
// Creates an offer and applies it as a local session description.
// Creates an answer with the same SDP an the offer but removes all lines
// that start with a:ssrc"
void CreateOfferReceiveAnswerWithoutSsrc() {
CreateOfferAsLocalDescription();
std::string sdp;
EXPECT_TRUE(pc_->local_description()->ToString(&sdp));
SetSsrcToZero(&sdp);
CreateAnswerAsRemoteDescription(sdp);
}
// This function creates a MediaStream with label kStreams[0] and
// |number_of_audio_tracks| and |number_of_video_tracks| tracks and the
// corresponding SessionDescriptionInterface. The SessionDescriptionInterface
// is returned and the MediaStream is stored in
// |reference_collection_|
std::unique_ptr<SessionDescriptionInterface>
CreateSessionDescriptionAndReference(size_t number_of_audio_tracks,
size_t number_of_video_tracks) {
EXPECT_LE(number_of_audio_tracks, 2u);
EXPECT_LE(number_of_video_tracks, 2u);
reference_collection_ = StreamCollection::Create();
std::string sdp_ms1 = std::string(kSdpStringInit);
std::string mediastream_id = kStreams[0];
rtc::scoped_refptr<webrtc::MediaStreamInterface> stream(
webrtc::MediaStream::Create(mediastream_id));
reference_collection_->AddStream(stream);
if (number_of_audio_tracks > 0) {
sdp_ms1 += std::string(kSdpStringAudio);
sdp_ms1 += std::string(kSdpStringMs1Audio0);
AddAudioTrack(kAudioTracks[0], stream);
}
if (number_of_audio_tracks > 1) {
sdp_ms1 += kSdpStringMs1Audio1;
AddAudioTrack(kAudioTracks[1], stream);
}
if (number_of_video_tracks > 0) {
sdp_ms1 += std::string(kSdpStringVideo);
sdp_ms1 += std::string(kSdpStringMs1Video0);
AddVideoTrack(kVideoTracks[0], stream);
}
if (number_of_video_tracks > 1) {
sdp_ms1 += kSdpStringMs1Video1;
AddVideoTrack(kVideoTracks[1], stream);
}
return std::unique_ptr<SessionDescriptionInterface>(
webrtc::CreateSessionDescription(SdpType::kOffer, sdp_ms1));
}
void AddAudioTrack(const std::string& track_id,
MediaStreamInterface* stream) {
rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
webrtc::AudioTrack::Create(track_id, nullptr));
ASSERT_TRUE(stream->AddTrack(audio_track));
}
void AddVideoTrack(const std::string& track_id,
MediaStreamInterface* stream) {
rtc::scoped_refptr<webrtc::VideoTrackInterface> video_track(
webrtc::VideoTrack::Create(track_id,
webrtc::FakeVideoTrackSource::Create(),
rtc::Thread::Current()));
ASSERT_TRUE(stream->AddTrack(video_track));
}
std::unique_ptr<SessionDescriptionInterface> CreateOfferWithOneAudioTrack() {
CreatePeerConnectionWithoutDtls();
AddAudioTrack(kAudioTracks[0]);
std::unique_ptr<SessionDescriptionInterface> offer;
EXPECT_TRUE(DoCreateOffer(&offer, nullptr));
return offer;
}
std::unique_ptr<SessionDescriptionInterface> CreateOfferWithOneAudioStream() {
CreatePeerConnectionWithoutDtls();
AddAudioStream(kStreamId1);
std::unique_ptr<SessionDescriptionInterface> offer;
EXPECT_TRUE(DoCreateOffer(&offer, nullptr));
return offer;
}
std::unique_ptr<SessionDescriptionInterface> CreateAnswerWithOneAudioTrack() {
EXPECT_TRUE(DoSetRemoteDescription(CreateOfferWithOneAudioTrack()));
std::unique_ptr<SessionDescriptionInterface> answer;
EXPECT_TRUE(DoCreateAnswer(&answer, nullptr));
return answer;
}
std::unique_ptr<SessionDescriptionInterface>
CreateAnswerWithOneAudioStream() {
EXPECT_TRUE(DoSetRemoteDescription(CreateOfferWithOneAudioStream()));
std::unique_ptr<SessionDescriptionInterface> answer;
EXPECT_TRUE(DoCreateAnswer(&answer, nullptr));
return answer;
}
const std::string& GetFirstAudioStreamCname(
const SessionDescriptionInterface* desc) {
const cricket::AudioContentDescription* audio_desc =
cricket::GetFirstAudioContentDescription(desc->description());
return audio_desc->streams()[0].cname;
}
std::unique_ptr<SessionDescriptionInterface> CreateOfferWithOptions(
const RTCOfferAnswerOptions& offer_answer_options) {
RTC_DCHECK(pc_);
rtc::scoped_refptr<MockCreateSessionDescriptionObserver> observer(
new rtc::RefCountedObject<MockCreateSessionDescriptionObserver>());
pc_->CreateOffer(observer, offer_answer_options);
EXPECT_EQ_WAIT(true, observer->called(), kTimeout);
return observer->MoveDescription();
}
void CreateOfferWithOptionsAsRemoteDescription(
std::unique_ptr<SessionDescriptionInterface>* desc,
const RTCOfferAnswerOptions& offer_answer_options) {
*desc = CreateOfferWithOptions(offer_answer_options);
ASSERT_TRUE(desc != nullptr);
std::string sdp;
EXPECT_TRUE((*desc)->ToString(&sdp));
std::unique_ptr<SessionDescriptionInterface> remote_offer(
webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_offer)));
EXPECT_EQ(PeerConnectionInterface::kHaveRemoteOffer, observer_.state_);
}
void CreateOfferWithOptionsAsLocalDescription(
std::unique_ptr<SessionDescriptionInterface>* desc,
const RTCOfferAnswerOptions& offer_answer_options) {
*desc = CreateOfferWithOptions(offer_answer_options);
ASSERT_TRUE(desc != nullptr);
std::string sdp;
EXPECT_TRUE((*desc)->ToString(&sdp));
std::unique_ptr<SessionDescriptionInterface> new_offer(
webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
EXPECT_TRUE(DoSetLocalDescription(std::move(new_offer)));
EXPECT_EQ(PeerConnectionInterface::kHaveLocalOffer, observer_.state_);
}
bool HasCNCodecs(const cricket::ContentInfo* content) {
RTC_DCHECK(content);
RTC_DCHECK(content->media_description());
for (const cricket::AudioCodec& codec :
content->media_description()->as_audio()->codecs()) {
if (codec.name == "CN") {
return true;
}
}
return false;
}
const char* GetSdpStringWithStream1() const {
if (sdp_semantics_ == SdpSemantics::kPlanB) {
return kSdpStringWithStream1PlanB;
} else {
return kSdpStringWithStream1UnifiedPlan;
}
}
const char* GetSdpStringWithStream1And2() const {
if (sdp_semantics_ == SdpSemantics::kPlanB) {
return kSdpStringWithStream1And2PlanB;
} else {
return kSdpStringWithStream1And2UnifiedPlan;
}
}
std::unique_ptr<rtc::VirtualSocketServer> vss_;
rtc::AutoSocketServerThread main_;
rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
cricket::FakePortAllocator* port_allocator_ = nullptr;
FakeRTCCertificateGenerator* fake_certificate_generator_ = nullptr;
rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory_;
rtc::scoped_refptr<PeerConnectionFactoryForTest> pc_factory_for_test_;
rtc::scoped_refptr<PeerConnectionInterface> pc_;
MockPeerConnectionObserver observer_;
rtc::scoped_refptr<StreamCollection> reference_collection_;
const SdpSemantics sdp_semantics_;
};
class PeerConnectionInterfaceTest
: public PeerConnectionInterfaceBaseTest,
public ::testing::WithParamInterface<SdpSemantics> {
protected:
PeerConnectionInterfaceTest() : PeerConnectionInterfaceBaseTest(GetParam()) {}
};
class PeerConnectionInterfaceTestPlanB
: public PeerConnectionInterfaceBaseTest {
protected:
PeerConnectionInterfaceTestPlanB()
: PeerConnectionInterfaceBaseTest(SdpSemantics::kPlanB) {}
};
// Generate different CNAMEs when PeerConnections are created.
// The CNAMEs are expected to be generated randomly. It is possible
// that the test fails, though the possibility is very low.
TEST_P(PeerConnectionInterfaceTest, CnameGenerationInOffer) {
std::unique_ptr<SessionDescriptionInterface> offer1 =
CreateOfferWithOneAudioTrack();
std::unique_ptr<SessionDescriptionInterface> offer2 =
CreateOfferWithOneAudioTrack();
EXPECT_NE(GetFirstAudioStreamCname(offer1.get()),
GetFirstAudioStreamCname(offer2.get()));
}
TEST_P(PeerConnectionInterfaceTest, CnameGenerationInAnswer) {
std::unique_ptr<SessionDescriptionInterface> answer1 =
CreateAnswerWithOneAudioTrack();
std::unique_ptr<SessionDescriptionInterface> answer2 =
CreateAnswerWithOneAudioTrack();
EXPECT_NE(GetFirstAudioStreamCname(answer1.get()),
GetFirstAudioStreamCname(answer2.get()));
}
TEST_P(PeerConnectionInterfaceTest,
CreatePeerConnectionWithDifferentConfigurations) {
CreatePeerConnectionWithDifferentConfigurations();
}
TEST_P(PeerConnectionInterfaceTest,
CreatePeerConnectionWithDifferentIceTransportsTypes) {
CreatePeerConnectionWithIceTransportsType(PeerConnectionInterface::kNone);
EXPECT_EQ(cricket::CF_NONE, port_allocator_->candidate_filter());
CreatePeerConnectionWithIceTransportsType(PeerConnectionInterface::kRelay);
EXPECT_EQ(cricket::CF_RELAY, port_allocator_->candidate_filter());
CreatePeerConnectionWithIceTransportsType(PeerConnectionInterface::kNoHost);
EXPECT_EQ(cricket::CF_ALL & ~cricket::CF_HOST,
port_allocator_->candidate_filter());
CreatePeerConnectionWithIceTransportsType(PeerConnectionInterface::kAll);
EXPECT_EQ(cricket::CF_ALL, port_allocator_->candidate_filter());
}
// Test that when a PeerConnection is created with a nonzero candidate pool
// size, the pooled PortAllocatorSession is created with all the attributes
// in the RTCConfiguration.
TEST_P(PeerConnectionInterfaceTest, CreatePeerConnectionWithPooledCandidates) {
PeerConnectionInterface::RTCConfiguration config;
PeerConnectionInterface::IceServer server;
server.uri = kStunAddressOnly;
config.servers.push_back(server);
config.type = PeerConnectionInterface::kRelay;
config.disable_ipv6 = true;
config.tcp_candidate_policy =
PeerConnectionInterface::kTcpCandidatePolicyDisabled;
config.candidate_network_policy =
PeerConnectionInterface::kCandidateNetworkPolicyLowCost;
config.ice_candidate_pool_size = 1;
CreatePeerConnection(config);
const cricket::FakePortAllocatorSession* session =
static_cast<const cricket::FakePortAllocatorSession*>(
port_allocator_->GetPooledSession());
ASSERT_NE(nullptr, session);
EXPECT_EQ(1UL, session->stun_servers().size());
EXPECT_EQ(0U, session->flags() & cricket::PORTALLOCATOR_ENABLE_IPV6);
EXPECT_LT(0U, session->flags() & cricket::PORTALLOCATOR_DISABLE_TCP);
EXPECT_LT(0U,
session->flags() & cricket::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS);
}
// Test that network-related RTCConfiguration members are applied to the
// PortAllocator when CreatePeerConnection is called. Specifically:
// - disable_ipv6_on_wifi
// - max_ipv6_networks
// - tcp_candidate_policy
// - candidate_network_policy
// - prune_turn_ports
//
// Note that the candidate filter (RTCConfiguration::type) is already tested
// above.
TEST_P(PeerConnectionInterfaceTest,
CreatePeerConnectionAppliesNetworkConfigToPortAllocator) {
// Create fake port allocator.
std::unique_ptr<cricket::FakePortAllocator> port_allocator(
new cricket::FakePortAllocator(rtc::Thread::Current(), nullptr));
cricket::FakePortAllocator* raw_port_allocator = port_allocator.get();
// Create RTCConfiguration with some network-related fields relevant to
// PortAllocator populated.
PeerConnectionInterface::RTCConfiguration config;
config.disable_ipv6_on_wifi = true;
config.max_ipv6_networks = 10;
config.tcp_candidate_policy =
PeerConnectionInterface::kTcpCandidatePolicyDisabled;
config.candidate_network_policy =
PeerConnectionInterface::kCandidateNetworkPolicyLowCost;
config.prune_turn_ports = true;
// Create the PC factory and PC with the above config.
rtc::scoped_refptr<webrtc::PeerConnectionFactoryInterface> pc_factory(
webrtc::CreatePeerConnectionFactory(
rtc::Thread::Current(), rtc::Thread::Current(),
rtc::Thread::Current(), fake_audio_capture_module_,
webrtc::CreateBuiltinAudioEncoderFactory(),
webrtc::CreateBuiltinAudioDecoderFactory(),
webrtc::CreateBuiltinVideoEncoderFactory(),
webrtc::CreateBuiltinVideoDecoderFactory(), nullptr /* audio_mixer */,
nullptr /* audio_processing */));
rtc::scoped_refptr<PeerConnectionInterface> pc(
pc_factory->CreatePeerConnection(config, std::move(port_allocator),
nullptr, &observer_));
EXPECT_TRUE(pc.get());
observer_.SetPeerConnectionInterface(pc.get());
// Now validate that the config fields set above were applied to the
// PortAllocator, as flags or otherwise.
EXPECT_FALSE(raw_port_allocator->flags() &
cricket::PORTALLOCATOR_ENABLE_IPV6_ON_WIFI);
EXPECT_EQ(10, raw_port_allocator->max_ipv6_networks());
EXPECT_TRUE(raw_port_allocator->flags() & cricket::PORTALLOCATOR_DISABLE_TCP);
EXPECT_TRUE(raw_port_allocator->flags() &
cricket::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS);
EXPECT_TRUE(raw_port_allocator->prune_turn_ports());
}
// Check that GetConfiguration returns the configuration the PeerConnection was
// constructed with, before SetConfiguration is called.
TEST_P(PeerConnectionInterfaceTest, GetConfigurationAfterCreatePeerConnection) {
PeerConnectionInterface::RTCConfiguration config;
config.type = PeerConnectionInterface::kRelay;
CreatePeerConnection(config);
PeerConnectionInterface::RTCConfiguration returned_config =
pc_->GetConfiguration();
EXPECT_EQ(PeerConnectionInterface::kRelay, returned_config.type);
}
// Check that GetConfiguration returns the last configuration passed into
// SetConfiguration.
TEST_P(PeerConnectionInterfaceTest, GetConfigurationAfterSetConfiguration) {
PeerConnectionInterface::RTCConfiguration starting_config;
starting_config.bundle_policy =
webrtc::PeerConnection::kBundlePolicyMaxBundle;
CreatePeerConnection(starting_config);
PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration();
config.type = PeerConnectionInterface::kRelay;
config.use_media_transport = true;
config.use_media_transport_for_data_channels = true;
EXPECT_TRUE(pc_->SetConfiguration(config));
PeerConnectionInterface::RTCConfiguration returned_config =
pc_->GetConfiguration();
EXPECT_EQ(PeerConnectionInterface::kRelay, returned_config.type);
EXPECT_TRUE(returned_config.use_media_transport);
EXPECT_TRUE(returned_config.use_media_transport_for_data_channels);
}
TEST_P(PeerConnectionInterfaceTest, SetConfigurationFailsAfterClose) {
CreatePeerConnection();
pc_->Close();
EXPECT_FALSE(
pc_->SetConfiguration(PeerConnectionInterface::RTCConfiguration()));
}
TEST_F(PeerConnectionInterfaceTestPlanB, AddStreams) {
CreatePeerConnectionWithoutDtls();
AddVideoStream(kStreamId1);
AddAudioStream(kStreamId2);
ASSERT_EQ(2u, pc_->local_streams()->count());
// Test we can add multiple local streams to one peerconnection.
rtc::scoped_refptr<MediaStreamInterface> stream(
pc_factory_->CreateLocalMediaStream(kStreamId3));
rtc::scoped_refptr<AudioTrackInterface> audio_track(
pc_factory_->CreateAudioTrack(kStreamId3,
static_cast<AudioSourceInterface*>(NULL)));
stream->AddTrack(audio_track.get());
EXPECT_TRUE(pc_->AddStream(stream));
EXPECT_EQ(3u, pc_->local_streams()->count());
// Remove the third stream.
pc_->RemoveStream(pc_->local_streams()->at(2));
EXPECT_EQ(2u, pc_->local_streams()->count());
// Remove the second stream.
pc_->RemoveStream(pc_->local_streams()->at(1));
EXPECT_EQ(1u, pc_->local_streams()->count());
// Remove the first stream.
pc_->RemoveStream(pc_->local_streams()->at(0));
EXPECT_EQ(0u, pc_->local_streams()->count());
}
// Test that the created offer includes streams we added.
// Don't run under Unified Plan since the stream API is not available.
TEST_F(PeerConnectionInterfaceTestPlanB, AddedStreamsPresentInOffer) {
CreatePeerConnectionWithoutDtls();
AddAudioVideoStream(kStreamId1, "audio_track", "video_track");
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
const cricket::AudioContentDescription* audio_desc =
cricket::GetFirstAudioContentDescription(offer->description());
EXPECT_TRUE(ContainsTrack(audio_desc->streams(), kStreamId1, "audio_track"));
const cricket::VideoContentDescription* video_desc =
cricket::GetFirstVideoContentDescription(offer->description());
EXPECT_TRUE(ContainsTrack(video_desc->streams(), kStreamId1, "video_track"));
// Add another stream and ensure the offer includes both the old and new
// streams.
AddAudioVideoStream(kStreamId2, "audio_track2", "video_track2");
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
audio_desc = cricket::GetFirstAudioContentDescription(offer->description());
EXPECT_TRUE(ContainsTrack(audio_desc->streams(), kStreamId1, "audio_track"));
EXPECT_TRUE(ContainsTrack(audio_desc->streams(), kStreamId2, "audio_track2"));
video_desc = cricket::GetFirstVideoContentDescription(offer->description());
EXPECT_TRUE(ContainsTrack(video_desc->streams(), kStreamId1, "video_track"));
EXPECT_TRUE(ContainsTrack(video_desc->streams(), kStreamId2, "video_track2"));
}
// Don't run under Unified Plan since the stream API is not available.
TEST_F(PeerConnectionInterfaceTestPlanB, RemoveStream) {
CreatePeerConnectionWithoutDtls();
AddVideoStream(kStreamId1);
ASSERT_EQ(1u, pc_->local_streams()->count());
pc_->RemoveStream(pc_->local_streams()->at(0));
EXPECT_EQ(0u, pc_->local_streams()->count());
}
// Test for AddTrack and RemoveTrack methods.
// Tests that the created offer includes tracks we added,
// and that the RtpSenders are created correctly.
// Also tests that RemoveTrack removes the tracks from subsequent offers.
// Only tested with Plan B since Unified Plan is covered in more detail by tests
// in peerconnection_jsep_unittests.cc
TEST_F(PeerConnectionInterfaceTestPlanB, AddTrackRemoveTrack) {
CreatePeerConnectionWithoutDtls();
rtc::scoped_refptr<AudioTrackInterface> audio_track(
CreateAudioTrack("audio_track"));
rtc::scoped_refptr<VideoTrackInterface> video_track(
CreateVideoTrack("video_track"));
auto audio_sender = pc_->AddTrack(audio_track, {kStreamId1}).MoveValue();
auto video_sender = pc_->AddTrack(video_track, {kStreamId1}).MoveValue();
EXPECT_EQ(1UL, audio_sender->stream_ids().size());
EXPECT_EQ(kStreamId1, audio_sender->stream_ids()[0]);
EXPECT_EQ("audio_track", audio_sender->id());
EXPECT_EQ(audio_track, audio_sender->track());
EXPECT_EQ(1UL, video_sender->stream_ids().size());
EXPECT_EQ(kStreamId1, video_sender->stream_ids()[0]);
EXPECT_EQ("video_track", video_sender->id());
EXPECT_EQ(video_track, video_sender->track());
// Now create an offer and check for the senders.
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
const cricket::ContentInfo* audio_content =
cricket::GetFirstAudioContent(offer->description());
EXPECT_TRUE(ContainsTrack(audio_content->media_description()->streams(),
kStreamId1, "audio_track"));
const cricket::ContentInfo* video_content =
cricket::GetFirstVideoContent(offer->description());
EXPECT_TRUE(ContainsTrack(video_content->media_description()->streams(),
kStreamId1, "video_track"));
EXPECT_TRUE(DoSetLocalDescription(std::move(offer)));
// Now try removing the tracks.
EXPECT_TRUE(pc_->RemoveTrack(audio_sender));
EXPECT_TRUE(pc_->RemoveTrack(video_sender));
// Create a new offer and ensure it doesn't contain the removed senders.
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
audio_content = cricket::GetFirstAudioContent(offer->description());
EXPECT_FALSE(ContainsTrack(audio_content->media_description()->streams(),
kStreamId1, "audio_track"));
video_content = cricket::GetFirstVideoContent(offer->description());
EXPECT_FALSE(ContainsTrack(video_content->media_description()->streams(),
kStreamId1, "video_track"));
EXPECT_TRUE(DoSetLocalDescription(std::move(offer)));
// Calling RemoveTrack on a sender no longer attached to a PeerConnection
// should return false.
EXPECT_FALSE(pc_->RemoveTrack(audio_sender));
EXPECT_FALSE(pc_->RemoveTrack(video_sender));
}
// Test creating senders without a stream specified,
// expecting a random stream ID to be generated.
TEST_P(PeerConnectionInterfaceTest, AddTrackWithoutStream) {
CreatePeerConnectionWithoutDtls();
rtc::scoped_refptr<AudioTrackInterface> audio_track(
CreateAudioTrack("audio_track"));
rtc::scoped_refptr<VideoTrackInterface> video_track(
CreateVideoTrack("video_track"));
auto audio_sender =
pc_->AddTrack(audio_track, std::vector<std::string>()).MoveValue();
auto video_sender =
pc_->AddTrack(video_track, std::vector<std::string>()).MoveValue();
EXPECT_EQ("audio_track", audio_sender->id());
EXPECT_EQ(audio_track, audio_sender->track());
EXPECT_EQ("video_track", video_sender->id());
EXPECT_EQ(video_track, video_sender->track());
if (sdp_semantics_ == SdpSemantics::kPlanB) {
// If the ID is truly a random GUID, it should be infinitely unlikely they
// will be the same.
EXPECT_NE(video_sender->stream_ids(), audio_sender->stream_ids());
} else {
// We allows creating tracks without stream ids under Unified Plan
// semantics.
EXPECT_EQ(0u, video_sender->stream_ids().size());
EXPECT_EQ(0u, audio_sender->stream_ids().size());
}
}
// Test that we can call GetStats() after AddTrack but before connecting
// the PeerConnection to a peer.
TEST_P(PeerConnectionInterfaceTest, AddTrackBeforeConnecting) {
CreatePeerConnectionWithoutDtls();
rtc::scoped_refptr<AudioTrackInterface> audio_track(
CreateAudioTrack("audio_track"));
rtc::scoped_refptr<VideoTrackInterface> video_track(
CreateVideoTrack("video_track"));
auto audio_sender = pc_->AddTrack(audio_track, std::vector<std::string>());
auto video_sender = pc_->AddTrack(video_track, std::vector<std::string>());
EXPECT_TRUE(DoGetStats(nullptr));
}
TEST_P(PeerConnectionInterfaceTest, AttachmentIdIsSetOnAddTrack) {
CreatePeerConnectionWithoutDtls();
rtc::scoped_refptr<AudioTrackInterface> audio_track(
CreateAudioTrack("audio_track"));
rtc::scoped_refptr<VideoTrackInterface> video_track(
CreateVideoTrack("video_track"));
auto audio_sender = pc_->AddTrack(audio_track, std::vector<std::string>());
ASSERT_TRUE(audio_sender.ok());
auto* audio_sender_proxy =
static_cast<RtpSenderProxyWithInternal<RtpSenderInternal>*>(
audio_sender.value().get());
EXPECT_NE(0, audio_sender_proxy->internal()->AttachmentId());
auto video_sender = pc_->AddTrack(video_track, std::vector<std::string>());
ASSERT_TRUE(video_sender.ok());
auto* video_sender_proxy =
static_cast<RtpSenderProxyWithInternal<RtpSenderInternal>*>(
video_sender.value().get());
EXPECT_NE(0, video_sender_proxy->internal()->AttachmentId());
}
// Don't run under Unified Plan since the stream API is not available.
TEST_F(PeerConnectionInterfaceTestPlanB, AttachmentIdIsSetOnAddStream) {
CreatePeerConnectionWithoutDtls();
AddVideoStream(kStreamId1);
auto senders = pc_->GetSenders();
ASSERT_EQ(1u, senders.size());
auto* sender_proxy =
static_cast<RtpSenderProxyWithInternal<RtpSenderInternal>*>(
senders[0].get());
EXPECT_NE(0, sender_proxy->internal()->AttachmentId());
}
TEST_P(PeerConnectionInterfaceTest, CreateOfferReceiveAnswer) {
InitiateCall();
WaitAndVerifyOnAddStream(kStreamId1, 2);
VerifyRemoteRtpHeaderExtensions();
}
TEST_P(PeerConnectionInterfaceTest, CreateOfferReceivePrAnswerAndAnswer) {
CreatePeerConnectionWithoutDtls();
AddVideoTrack(kVideoTracks[0], {kStreamId1});
CreateOfferAsLocalDescription();
std::string offer;
EXPECT_TRUE(pc_->local_description()->ToString(&offer));
CreatePrAnswerAndAnswerAsRemoteDescription(offer);
WaitAndVerifyOnAddStream(kStreamId1, 1);
}
TEST_P(PeerConnectionInterfaceTest, ReceiveOfferCreateAnswer) {
CreatePeerConnectionWithoutDtls();
AddVideoTrack(kVideoTracks[0], {kStreamId1});
CreateOfferAsRemoteDescription();
CreateAnswerAsLocalDescription();
WaitAndVerifyOnAddStream(kStreamId1, 1);
}
TEST_P(PeerConnectionInterfaceTest, ReceiveOfferCreatePrAnswerAndAnswer) {
CreatePeerConnectionWithoutDtls();
AddVideoTrack(kVideoTracks[0], {kStreamId1});
CreateOfferAsRemoteDescription();
CreatePrAnswerAsLocalDescription();
CreateAnswerAsLocalDescription();
WaitAndVerifyOnAddStream(kStreamId1, 1);
}
// Don't run under Unified Plan since the stream API is not available.
TEST_F(PeerConnectionInterfaceTestPlanB, Renegotiate) {
InitiateCall();
ASSERT_EQ(1u, pc_->remote_streams()->count());
pc_->RemoveStream(pc_->local_streams()->at(0));
CreateOfferReceiveAnswer();
EXPECT_EQ(0u, pc_->remote_streams()->count());
AddVideoStream(kStreamId1);
CreateOfferReceiveAnswer();
}
// Tests that after negotiating an audio only call, the respondent can perform a
// renegotiation that removes the audio stream.
TEST_F(PeerConnectionInterfaceTestPlanB, RenegotiateAudioOnly) {
CreatePeerConnectionWithoutDtls();
AddAudioStream(kStreamId1);
CreateOfferAsRemoteDescription();
CreateAnswerAsLocalDescription();
ASSERT_EQ(1u, pc_->remote_streams()->count());
pc_->RemoveStream(pc_->local_streams()->at(0));
CreateOfferReceiveAnswer();
EXPECT_EQ(0u, pc_->remote_streams()->count());
}
// Test that candidates are generated and that we can parse our own candidates.
TEST_P(PeerConnectionInterfaceTest, IceCandidates) {
CreatePeerConnectionWithoutDtls();
EXPECT_FALSE(pc_->AddIceCandidate(observer_.last_candidate()));
// SetRemoteDescription takes ownership of offer.
std::unique_ptr<SessionDescriptionInterface> offer;
AddVideoTrack(kVideoTracks[0]);
EXPECT_TRUE(DoCreateOffer(&offer, nullptr));
EXPECT_TRUE(DoSetRemoteDescription(std::move(offer)));
// SetLocalDescription takes ownership of answer.
std::unique_ptr<SessionDescriptionInterface> answer;
EXPECT_TRUE(DoCreateAnswer(&answer, nullptr));
EXPECT_TRUE(DoSetLocalDescription(std::move(answer)));
EXPECT_TRUE_WAIT(observer_.last_candidate() != nullptr, kTimeout);
EXPECT_TRUE_WAIT(observer_.ice_gathering_complete_, kTimeout);
EXPECT_TRUE(pc_->AddIceCandidate(observer_.last_candidate()));
}
// Test that CreateOffer and CreateAnswer will fail if the track labels are
// not unique.
TEST_F(PeerConnectionInterfaceTestPlanB, CreateOfferAnswerWithInvalidStream) {
CreatePeerConnectionWithoutDtls();
// Create a regular offer for the CreateAnswer test later.
std::unique_ptr<SessionDescriptionInterface> offer;
EXPECT_TRUE(DoCreateOffer(&offer, nullptr));
EXPECT_TRUE(offer);
offer.reset();
// Create a local stream with audio&video tracks having same label.
AddAudioTrack("track_label", {kStreamId1});
AddVideoTrack("track_label", {kStreamId1});
// Test CreateOffer
EXPECT_FALSE(DoCreateOffer(&offer, nullptr));
// Test CreateAnswer
std::unique_ptr<SessionDescriptionInterface> answer;
EXPECT_FALSE(DoCreateAnswer(&answer, nullptr));
}
// Test that we will get different SSRCs for each tracks in the offer and answer
// we created.
TEST_P(PeerConnectionInterfaceTest, SsrcInOfferAnswer) {
CreatePeerConnectionWithoutDtls();
// Create a local stream with audio&video tracks having different labels.
AddAudioTrack(kAudioTracks[0], {kStreamId1});
AddVideoTrack(kVideoTracks[0], {kStreamId1});
// Test CreateOffer
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
int audio_ssrc = 0;
int video_ssrc = 0;
EXPECT_TRUE(
GetFirstSsrc(GetFirstAudioContent(offer->description()), &audio_ssrc));
EXPECT_TRUE(
GetFirstSsrc(GetFirstVideoContent(offer->description()), &video_ssrc));
EXPECT_NE(audio_ssrc, video_ssrc);
// Test CreateAnswer
EXPECT_TRUE(DoSetRemoteDescription(std::move(offer)));
std::unique_ptr<SessionDescriptionInterface> answer;
ASSERT_TRUE(DoCreateAnswer(&answer, nullptr));
audio_ssrc = 0;
video_ssrc = 0;
EXPECT_TRUE(
GetFirstSsrc(GetFirstAudioContent(answer->description()), &audio_ssrc));
EXPECT_TRUE(
GetFirstSsrc(GetFirstVideoContent(answer->description()), &video_ssrc));
EXPECT_NE(audio_ssrc, video_ssrc);
}
// Test that it's possible to call AddTrack on a MediaStream after adding
// the stream to a PeerConnection.
// TODO(deadbeef): Remove this test once this behavior is no longer supported.
// Don't run under Unified Plan since the stream API is not available.
TEST_F(PeerConnectionInterfaceTestPlanB, AddTrackAfterAddStream) {
CreatePeerConnectionWithoutDtls();
// Create audio stream and add to PeerConnection.
AddAudioStream(kStreamId1);
MediaStreamInterface* stream = pc_->local_streams()->at(0);
// Add video track to the audio-only stream.
rtc::scoped_refptr<VideoTrackInterface> video_track(
CreateVideoTrack("video_label"));
stream->AddTrack(video_track.get());
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
const cricket::MediaContentDescription* video_desc =
cricket::GetFirstVideoContentDescription(offer->description());
EXPECT_TRUE(video_desc != nullptr);
}
// Test that it's possible to call RemoveTrack on a MediaStream after adding
// the stream to a PeerConnection.
// TODO(deadbeef): Remove this test once this behavior is no longer supported.
// Don't run under Unified Plan since the stream API is not available.
TEST_F(PeerConnectionInterfaceTestPlanB, RemoveTrackAfterAddStream) {
CreatePeerConnectionWithoutDtls();
// Create audio/video stream and add to PeerConnection.
AddAudioVideoStream(kStreamId1, "audio_label", "video_label");
MediaStreamInterface* stream = pc_->local_streams()->at(0);
// Remove the video track.
stream->RemoveTrack(stream->GetVideoTracks()[0]);
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
const cricket::MediaContentDescription* video_desc =
cricket::GetFirstVideoContentDescription(offer->description());
EXPECT_TRUE(video_desc == nullptr);
}
// Test creating a sender with a stream ID, and ensure the ID is populated
// in the offer.
// Don't run under Unified Plan since the stream API is not available.
TEST_F(PeerConnectionInterfaceTestPlanB, CreateSenderWithStream) {
CreatePeerConnectionWithoutDtls();
pc_->CreateSender("video", kStreamId1);
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
const cricket::MediaContentDescription* video_desc =
cricket::GetFirstVideoContentDescription(offer->description());
ASSERT_TRUE(video_desc != nullptr);
ASSERT_EQ(1u, video_desc->streams().size());
EXPECT_EQ(kStreamId1, video_desc->streams()[0].first_stream_id());
}
// Test that we can specify a certain track that we want statistics about.
TEST_P(PeerConnectionInterfaceTest, GetStatsForSpecificTrack) {
InitiateCall();
ASSERT_LT(0u, pc_->GetSenders().size());
ASSERT_LT(0u, pc_->GetReceivers().size());
rtc::scoped_refptr<MediaStreamTrackInterface> remote_audio =
pc_->GetReceivers()[0]->track();
EXPECT_TRUE(DoGetStats(remote_audio));
// Remove the stream. Since we are sending to our selves the local
// and the remote stream is the same.
pc_->RemoveTrack(pc_->GetSenders()[0]);
// Do a re-negotiation.
CreateOfferReceiveAnswer();
// Test that we still can get statistics for the old track. Even if it is not
// sent any longer.
EXPECT_TRUE(DoGetStats(remote_audio));
}
// Test that we can get stats on a video track.
TEST_P(PeerConnectionInterfaceTest, GetStatsForVideoTrack) {
InitiateCall();
auto video_receiver = GetFirstReceiverOfType(cricket::MEDIA_TYPE_VIDEO);
ASSERT_TRUE(video_receiver);
EXPECT_TRUE(DoGetStats(video_receiver->track()));
}
// Test that we don't get statistics for an invalid track.
TEST_P(PeerConnectionInterfaceTest, GetStatsForInvalidTrack) {
InitiateCall();
rtc::scoped_refptr<AudioTrackInterface> unknown_audio_track(
pc_factory_->CreateAudioTrack("unknown track", NULL));
EXPECT_FALSE(DoGetStats(unknown_audio_track));
}
TEST_P(PeerConnectionInterfaceTest, GetRTCStatsBeforeAndAfterCalling) {
CreatePeerConnectionWithoutDtls();
EXPECT_TRUE(DoGetRTCStats());
// Clearing stats cache is needed now, but should be temporary.
// https://bugs.chromium.org/p/webrtc/issues/detail?id=8693
pc_->ClearStatsCache();
AddAudioTrack(kAudioTracks[0], {kStreamId1});
AddVideoTrack(kVideoTracks[0], {kStreamId1});
EXPECT_TRUE(DoGetRTCStats());
pc_->ClearStatsCache();
CreateOfferReceiveAnswer();
EXPECT_TRUE(DoGetRTCStats());
}
// This test setup two RTP data channels in loop back.
TEST_P(PeerConnectionInterfaceTest, TestDataChannel) {
RTCConfiguration config;
config.enable_rtp_data_channel = true;
config.enable_dtls_srtp = false;
CreatePeerConnection(config);
rtc::scoped_refptr<DataChannelInterface> data1 =
pc_->CreateDataChannel("test1", NULL);
rtc::scoped_refptr<DataChannelInterface> data2 =
pc_->CreateDataChannel("test2", NULL);
ASSERT_TRUE(data1 != NULL);
std::unique_ptr<MockDataChannelObserver> observer1(
new MockDataChannelObserver(data1));
std::unique_ptr<MockDataChannelObserver> observer2(
new MockDataChannelObserver(data2));
EXPECT_EQ(DataChannelInterface::kConnecting, data1->state());
EXPECT_EQ(DataChannelInterface::kConnecting, data2->state());
std::string data_to_send1 = "testing testing";
std::string data_to_send2 = "testing something else";
EXPECT_FALSE(data1->Send(DataBuffer(data_to_send1)));
CreateOfferReceiveAnswer();
EXPECT_TRUE_WAIT(observer1->IsOpen(), kTimeout);
EXPECT_TRUE_WAIT(observer2->IsOpen(), kTimeout);
EXPECT_EQ(DataChannelInterface::kOpen, data1->state());
EXPECT_EQ(DataChannelInterface::kOpen, data2->state());
EXPECT_TRUE(data1->Send(DataBuffer(data_to_send1)));
EXPECT_TRUE(data2->Send(DataBuffer(data_to_send2)));
EXPECT_EQ_WAIT(data_to_send1, observer1->last_message(), kTimeout);
EXPECT_EQ_WAIT(data_to_send2, observer2->last_message(), kTimeout);
data1->Close();
EXPECT_EQ(DataChannelInterface::kClosing, data1->state());
CreateOfferReceiveAnswer();
EXPECT_FALSE(observer1->IsOpen());
EXPECT_EQ(DataChannelInterface::kClosed, data1->state());
EXPECT_TRUE(observer2->IsOpen());
data_to_send2 = "testing something else again";
EXPECT_TRUE(data2->Send(DataBuffer(data_to_send2)));
EXPECT_EQ_WAIT(data_to_send2, observer2->last_message(), kTimeout);
}
// This test verifies that sendnig binary data over RTP data channels should
// fail.
TEST_P(PeerConnectionInterfaceTest, TestSendBinaryOnRtpDataChannel) {
RTCConfiguration config;
config.enable_rtp_data_channel = true;
config.enable_dtls_srtp = false;
CreatePeerConnection(config);
rtc::scoped_refptr<DataChannelInterface> data1 =
pc_->CreateDataChannel("test1", NULL);
rtc::scoped_refptr<DataChannelInterface> data2 =
pc_->CreateDataChannel("test2", NULL);
ASSERT_TRUE(data1 != NULL);
std::unique_ptr<MockDataChannelObserver> observer1(
new MockDataChannelObserver(data1));
std::unique_ptr<MockDataChannelObserver> observer2(
new MockDataChannelObserver(data2));
EXPECT_EQ(DataChannelInterface::kConnecting, data1->state());
EXPECT_EQ(DataChannelInterface::kConnecting, data2->state());
CreateOfferReceiveAnswer();
EXPECT_TRUE_WAIT(observer1->IsOpen(), kTimeout);
EXPECT_TRUE_WAIT(observer2->IsOpen(), kTimeout);
EXPECT_EQ(DataChannelInterface::kOpen, data1->state());
EXPECT_EQ(DataChannelInterface::kOpen, data2->state());
rtc::CopyOnWriteBuffer buffer("test", 4);
EXPECT_FALSE(data1->Send(DataBuffer(buffer, true)));
}
// This test setup a RTP data channels in loop back and test that a channel is
// opened even if the remote end answer with a zero SSRC.
TEST_P(PeerConnectionInterfaceTest, TestSendOnlyDataChannel) {
RTCConfiguration config;
config.enable_rtp_data_channel = true;
config.enable_dtls_srtp = false;
CreatePeerConnection(config);
rtc::scoped_refptr<DataChannelInterface> data1 =
pc_->CreateDataChannel("test1", NULL);
std::unique_ptr<MockDataChannelObserver> observer1(
new MockDataChannelObserver(data1));
CreateOfferReceiveAnswerWithoutSsrc();
EXPECT_TRUE_WAIT(observer1->IsOpen(), kTimeout);
data1->Close();
EXPECT_EQ(DataChannelInterface::kClosing, data1->state());
CreateOfferReceiveAnswerWithoutSsrc();
EXPECT_EQ(DataChannelInterface::kClosed, data1->state());
EXPECT_FALSE(observer1->IsOpen());
}
// This test that if a data channel is added in an answer a receive only channel
// channel is created.
TEST_P(PeerConnectionInterfaceTest, TestReceiveOnlyDataChannel) {
RTCConfiguration config;
config.enable_rtp_data_channel = true;
config.enable_dtls_srtp = false;
CreatePeerConnection(config);
std::string offer_label = "offer_channel";
rtc::scoped_refptr<DataChannelInterface> offer_channel =
pc_->CreateDataChannel(offer_label, NULL);
CreateOfferAsLocalDescription();
// Replace the data channel label in the offer and apply it as an answer.
std::string receive_label = "answer_channel";
std::string sdp;
EXPECT_TRUE(pc_->local_description()->ToString(&sdp));
absl::StrReplaceAll({{offer_label, receive_label}}, &sdp);
CreateAnswerAsRemoteDescription(sdp);
// Verify that a new incoming data channel has been created and that
// it is open but can't we written to.
ASSERT_TRUE(observer_.last_datachannel_ != NULL);
DataChannelInterface* received_channel = observer_.last_datachannel_;
EXPECT_EQ(DataChannelInterface::kConnecting, received_channel->state());
EXPECT_EQ(receive_label, received_channel->label());
EXPECT_FALSE(received_channel->Send(DataBuffer("something")));
// Verify that the channel we initially offered has been rejected.
EXPECT_EQ(DataChannelInterface::kClosed, offer_channel->state());
// Do another offer / answer exchange and verify that the data channel is
// opened.
CreateOfferReceiveAnswer();
EXPECT_EQ_WAIT(DataChannelInterface::kOpen, received_channel->state(),
kTimeout);
}
// This test that no data channel is returned if a reliable channel is
// requested.
// TODO(perkj): Remove this test once reliable channels are implemented.
TEST_P(PeerConnectionInterfaceTest, CreateReliableRtpDataChannelShouldFail) {
RTCConfiguration rtc_config;
rtc_config.enable_rtp_data_channel = true;
CreatePeerConnection(rtc_config);
std::string label = "test";
webrtc::DataChannelInit config;
config.reliable = true;
rtc::scoped_refptr<DataChannelInterface> channel =
pc_->CreateDataChannel(label, &config);
EXPECT_TRUE(channel == NULL);
}
// Verifies that duplicated label is not allowed for RTP data channel.
TEST_P(PeerConnectionInterfaceTest, RtpDuplicatedLabelNotAllowed) {
RTCConfiguration config;
config.enable_rtp_data_channel = true;
CreatePeerConnection(config);
std::string label = "test";
rtc::scoped_refptr<DataChannelInterface> channel =
pc_->CreateDataChannel(label, nullptr);
EXPECT_NE(channel, nullptr);
rtc::scoped_refptr<DataChannelInterface> dup_channel =
pc_->CreateDataChannel(label, nullptr);
EXPECT_EQ(dup_channel, nullptr);
}
// This tests that a SCTP data channel is returned using different
// DataChannelInit configurations.
TEST_P(PeerConnectionInterfaceTest, CreateSctpDataChannel) {
RTCConfiguration rtc_config;
rtc_config.enable_dtls_srtp = true;
CreatePeerConnection(rtc_config);
webrtc::DataChannelInit config;
rtc::scoped_refptr<DataChannelInterface> channel =
pc_->CreateDataChannel("1", &config);
EXPECT_TRUE(channel != NULL);
EXPECT_TRUE(channel->reliable());
EXPECT_TRUE(observer_.renegotiation_needed_);
observer_.renegotiation_needed_ = false;
config.ordered = false;
channel = pc_->CreateDataChannel("2", &config);
EXPECT_TRUE(channel != NULL);
EXPECT_TRUE(channel->reliable());
EXPECT_FALSE(observer_.renegotiation_needed_);
config.ordered = true;
config.maxRetransmits = 0;
channel = pc_->CreateDataChannel("3", &config);
EXPECT_TRUE(channel != NULL);
EXPECT_FALSE(channel->reliable());
EXPECT_FALSE(observer_.renegotiation_needed_);
config.maxRetransmits = -1;
config.maxRetransmitTime = 0;
channel = pc_->CreateDataChannel("4", &config);
EXPECT_TRUE(channel != NULL);
EXPECT_FALSE(channel->reliable());
EXPECT_FALSE(observer_.renegotiation_needed_);
}
// This tests that no data channel is returned if both maxRetransmits and
// maxRetransmitTime are set for SCTP data channels.
TEST_P(PeerConnectionInterfaceTest,
CreateSctpDataChannelShouldFailForInvalidConfig) {
RTCConfiguration rtc_config;
rtc_config.enable_dtls_srtp = true;
CreatePeerConnection(rtc_config);
std::string label = "test";
webrtc::DataChannelInit config;
config.maxRetransmits = 0;
config.maxRetransmitTime = 0;
rtc::scoped_refptr<DataChannelInterface> channel =
pc_->CreateDataChannel(label, &config);
EXPECT_TRUE(channel == NULL);
}
// The test verifies that creating a SCTP data channel with an id already in use
// or out of range should fail.
TEST_P(PeerConnectionInterfaceTest,
CreateSctpDataChannelWithInvalidIdShouldFail) {
RTCConfiguration rtc_config;
rtc_config.enable_dtls_srtp = true;
CreatePeerConnection(rtc_config);
webrtc::DataChannelInit config;
rtc::scoped_refptr<DataChannelInterface> channel;
config.id = 1;
channel = pc_->CreateDataChannel("1", &config);
EXPECT_TRUE(channel != NULL);
EXPECT_EQ(1, channel->id());
channel = pc_->CreateDataChannel("x", &config);
EXPECT_TRUE(channel == NULL);
config.id = cricket::kMaxSctpSid;
channel = pc_->CreateDataChannel("max", &config);
EXPECT_TRUE(channel != NULL);
EXPECT_EQ(config.id, channel->id());
config.id = cricket::kMaxSctpSid + 1;
channel = pc_->CreateDataChannel("x", &config);
EXPECT_TRUE(channel == NULL);
}
// Verifies that duplicated label is allowed for SCTP data channel.
TEST_P(PeerConnectionInterfaceTest, SctpDuplicatedLabelAllowed) {
RTCConfiguration rtc_config;
rtc_config.enable_dtls_srtp = true;
CreatePeerConnection(rtc_config);
std::string label = "test";
rtc::scoped_refptr<DataChannelInterface> channel =
pc_->CreateDataChannel(label, nullptr);
EXPECT_NE(channel, nullptr);
rtc::scoped_refptr<DataChannelInterface> dup_channel =
pc_->CreateDataChannel(label, nullptr);
EXPECT_NE(dup_channel, nullptr);
}
// This test verifies that OnRenegotiationNeeded is fired for every new RTP
// DataChannel.
TEST_P(PeerConnectionInterfaceTest, RenegotiationNeededForNewRtpDataChannel) {
RTCConfiguration rtc_config;
rtc_config.enable_rtp_data_channel = true;
rtc_config.enable_dtls_srtp = false;
CreatePeerConnection(rtc_config);
rtc::scoped_refptr<DataChannelInterface> dc1 =
pc_->CreateDataChannel("test1", NULL);
EXPECT_TRUE(observer_.renegotiation_needed_);
observer_.renegotiation_needed_ = false;
rtc::scoped_refptr<DataChannelInterface> dc2 =
pc_->CreateDataChannel("test2", NULL);
EXPECT_TRUE(observer_.renegotiation_needed_);
}
// This test that a data channel closes when a PeerConnection is deleted/closed.
TEST_P(PeerConnectionInterfaceTest, DataChannelCloseWhenPeerConnectionClose) {
RTCConfiguration rtc_config;
rtc_config.enable_rtp_data_channel = true;
rtc_config.enable_dtls_srtp = false;
CreatePeerConnection(rtc_config);
rtc::scoped_refptr<DataChannelInterface> data1 =
pc_->CreateDataChannel("test1", NULL);
rtc::scoped_refptr<DataChannelInterface> data2 =
pc_->CreateDataChannel("test2", NULL);
ASSERT_TRUE(data1 != NULL);
std::unique_ptr<MockDataChannelObserver> observer1(
new MockDataChannelObserver(data1));
std::unique_ptr<MockDataChannelObserver> observer2(
new MockDataChannelObserver(data2));
CreateOfferReceiveAnswer();
EXPECT_TRUE_WAIT(observer1->IsOpen(), kTimeout);
EXPECT_TRUE_WAIT(observer2->IsOpen(), kTimeout);
ReleasePeerConnection();
EXPECT_EQ(DataChannelInterface::kClosed, data1->state());
EXPECT_EQ(DataChannelInterface::kClosed, data2->state());
}
// This tests that RTP data channels can be rejected in an answer.
TEST_P(PeerConnectionInterfaceTest, TestRejectRtpDataChannelInAnswer) {
RTCConfiguration rtc_config;
rtc_config.enable_rtp_data_channel = true;
rtc_config.enable_dtls_srtp = false;
CreatePeerConnection(rtc_config);
rtc::scoped_refptr<DataChannelInterface> offer_channel(
pc_->CreateDataChannel("offer_channel", NULL));
CreateOfferAsLocalDescription();
// Create an answer where the m-line for data channels are rejected.
std::string sdp;
EXPECT_TRUE(pc_->local_description()->ToString(&sdp));
std::unique_ptr<SessionDescriptionInterface> answer(
webrtc::CreateSessionDescription(SdpType::kAnswer, sdp));
ASSERT_TRUE(answer);
cricket::ContentInfo* data_info =
cricket::GetFirstDataContent(answer->description());
data_info->rejected = true;
DoSetRemoteDescription(std::move(answer));
EXPECT_EQ(DataChannelInterface::kClosed, offer_channel->state());
}
#ifdef HAVE_SCTP
// This tests that SCTP data channels can be rejected in an answer.
TEST_P(PeerConnectionInterfaceTest, TestRejectSctpDataChannelInAnswer)
#else
TEST_P(PeerConnectionInterfaceTest, DISABLED_TestRejectSctpDataChannelInAnswer)
#endif
{
RTCConfiguration rtc_config;
CreatePeerConnection(rtc_config);
rtc::scoped_refptr<DataChannelInterface> offer_channel(
pc_->CreateDataChannel("offer_channel", NULL));
CreateOfferAsLocalDescription();
// Create an answer where the m-line for data channels are rejected.
std::string sdp;
EXPECT_TRUE(pc_->local_description()->ToString(&sdp));
std::unique_ptr<SessionDescriptionInterface> answer(
webrtc::CreateSessionDescription(SdpType::kAnswer, sdp));
ASSERT_TRUE(answer);
cricket::ContentInfo* data_info =
cricket::GetFirstDataContent(answer->description());
data_info->rejected = true;
DoSetRemoteDescription(std::move(answer));
EXPECT_EQ(DataChannelInterface::kClosed, offer_channel->state());
}
// Test that we can create a session description from an SDP string from
// FireFox, use it as a remote session description, generate an answer and use
// the answer as a local description.
TEST_P(PeerConnectionInterfaceTest, ReceiveFireFoxOffer) {
RTCConfiguration rtc_config;
rtc_config.enable_dtls_srtp = true;
CreatePeerConnection(rtc_config);
AddAudioTrack("audio_label");
AddVideoTrack("video_label");
std::unique_ptr<SessionDescriptionInterface> desc(
webrtc::CreateSessionDescription(SdpType::kOffer,
webrtc::kFireFoxSdpOffer, nullptr));
EXPECT_TRUE(DoSetSessionDescription(std::move(desc), false));
CreateAnswerAsLocalDescription();
ASSERT_TRUE(pc_->local_description() != NULL);
ASSERT_TRUE(pc_->remote_description() != NULL);
const cricket::ContentInfo* content =
cricket::GetFirstAudioContent(pc_->local_description()->description());
ASSERT_TRUE(content != NULL);
EXPECT_FALSE(content->rejected);
content =
cricket::GetFirstVideoContent(pc_->local_description()->description());
ASSERT_TRUE(content != NULL);
EXPECT_FALSE(content->rejected);
#ifdef HAVE_SCTP
content =
cricket::GetFirstDataContent(pc_->local_description()->description());
ASSERT_TRUE(content != NULL);
EXPECT_FALSE(content->rejected);
#endif
}
// Test that fallback from DTLS to SDES is not supported.
// The fallback was previously supported but was removed to simplify the code
// and because it's non-standard.
TEST_P(PeerConnectionInterfaceTest, DtlsSdesFallbackNotSupported) {
RTCConfiguration rtc_config;
rtc_config.enable_dtls_srtp = true;
CreatePeerConnection(rtc_config);
// Wait for fake certificate to be generated. Previously, this is what caused
// the "a=crypto" lines to be rejected.
AddAudioTrack("audio_label");
AddVideoTrack("video_label");
ASSERT_NE(nullptr, fake_certificate_generator_);
EXPECT_EQ_WAIT(1, fake_certificate_generator_->generated_certificates(),
kTimeout);
std::unique_ptr<SessionDescriptionInterface> desc(
webrtc::CreateSessionDescription(SdpType::kOffer, kDtlsSdesFallbackSdp,
nullptr));
EXPECT_FALSE(DoSetSessionDescription(std::move(desc), /*local=*/false));
}
// Test that we can create an audio only offer and receive an answer with a
// limited set of audio codecs and receive an updated offer with more audio
// codecs, where the added codecs are not supported.
TEST_P(PeerConnectionInterfaceTest, ReceiveUpdatedAudioOfferWithBadCodecs) {
CreatePeerConnectionWithoutDtls();
AddAudioTrack("audio_label");
CreateOfferAsLocalDescription();
const char* answer_sdp =
(sdp_semantics_ == SdpSemantics::kPlanB ? webrtc::kAudioSdpPlanB
: webrtc::kAudioSdpUnifiedPlan);
std::unique_ptr<SessionDescriptionInterface> answer(
webrtc::CreateSessionDescription(SdpType::kAnswer, answer_sdp, nullptr));
EXPECT_TRUE(DoSetSessionDescription(std::move(answer), false));
const char* reoffer_sdp =
(sdp_semantics_ == SdpSemantics::kPlanB
? webrtc::kAudioSdpWithUnsupportedCodecsPlanB
: webrtc::kAudioSdpWithUnsupportedCodecsUnifiedPlan);
std::unique_ptr<SessionDescriptionInterface> updated_offer(
webrtc::CreateSessionDescription(SdpType::kOffer, reoffer_sdp, nullptr));
EXPECT_TRUE(DoSetSessionDescription(std::move(updated_offer), false));
CreateAnswerAsLocalDescription();
}
// Test that if we're receiving (but not sending) a track, subsequent offers
// will have m-lines with a=recvonly.
TEST_P(PeerConnectionInterfaceTest, CreateSubsequentRecvOnlyOffer) {
RTCConfiguration rtc_config;
rtc_config.enable_dtls_srtp = true;
CreatePeerConnection(rtc_config);
CreateAndSetRemoteOffer(GetSdpStringWithStream1());
CreateAnswerAsLocalDescription();
// At this point we should be receiving stream 1, but not sending anything.
// A new offer should be recvonly.
std::unique_ptr<SessionDescriptionInterface> offer;
DoCreateOffer(&offer, nullptr);
const cricket::ContentInfo* video_content =
cricket::GetFirstVideoContent(offer->description());
ASSERT_EQ(RtpTransceiverDirection::kRecvOnly,
video_content->media_description()->direction());
const cricket::ContentInfo* audio_content =
cricket::GetFirstAudioContent(offer->description());
ASSERT_EQ(RtpTransceiverDirection::kRecvOnly,
audio_content->media_description()->direction());
}
// Test that if we're receiving (but not sending) a track, and the
// offerToReceiveVideo/offerToReceiveAudio constraints are explicitly set to
// false, the generated m-lines will be a=inactive.
TEST_P(PeerConnectionInterfaceTest, CreateSubsequentInactiveOffer) {
RTCConfiguration rtc_config;
rtc_config.enable_dtls_srtp = true;
CreatePeerConnection(rtc_config);
CreateAndSetRemoteOffer(GetSdpStringWithStream1());
CreateAnswerAsLocalDescription();
// At this point we should be receiving stream 1, but not sending anything.
// A new offer would be recvonly, but we'll set the "no receive" constraints
// to make it inactive.
std::unique_ptr<SessionDescriptionInterface> offer;
RTCOfferAnswerOptions options;
options.offer_to_receive_audio = 0;
options.offer_to_receive_video = 0;
DoCreateOffer(&offer, &options);
const cricket::ContentInfo* video_content =
cricket::GetFirstVideoContent(offer->description());
ASSERT_EQ(RtpTransceiverDirection::kInactive,
video_content->media_description()->direction());
const cricket::ContentInfo* audio_content =
cricket::GetFirstAudioContent(offer->description());
ASSERT_EQ(RtpTransceiverDirection::kInactive,
audio_content->media_description()->direction());
}
// Test that we can use SetConfiguration to change the ICE servers of the
// PortAllocator.
TEST_P(PeerConnectionInterfaceTest, SetConfigurationChangesIceServers) {
CreatePeerConnection();
PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration();
PeerConnectionInterface::IceServer server;
server.uri = "stun:test_hostname";
config.servers.push_back(server);
EXPECT_TRUE(pc_->SetConfiguration(config));
EXPECT_EQ(1u, port_allocator_->stun_servers().size());
EXPECT_EQ("test_hostname",
port_allocator_->stun_servers().begin()->hostname());
}
TEST_P(PeerConnectionInterfaceTest, SetConfigurationChangesCandidateFilter) {
CreatePeerConnection();
PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration();
config.type = PeerConnectionInterface::kRelay;
EXPECT_TRUE(pc_->SetConfiguration(config));
EXPECT_EQ(cricket::CF_RELAY, port_allocator_->candidate_filter());
}
TEST_P(PeerConnectionInterfaceTest, SetConfigurationChangesPruneTurnPortsFlag) {
PeerConnectionInterface::RTCConfiguration config;
config.prune_turn_ports = false;
CreatePeerConnection(config);
config = pc_->GetConfiguration();
EXPECT_FALSE(port_allocator_->prune_turn_ports());
config.prune_turn_ports = true;
EXPECT_TRUE(pc_->SetConfiguration(config));
EXPECT_TRUE(port_allocator_->prune_turn_ports());
}
// Test that the ice check interval can be changed. This does not verify that
// the setting makes it all the way to P2PTransportChannel, as that would
// require a very complex set of mocks.
TEST_P(PeerConnectionInterfaceTest, SetConfigurationChangesIceCheckInterval) {
PeerConnectionInterface::RTCConfiguration config;
config.ice_check_min_interval = absl::nullopt;
CreatePeerConnection(config);
config = pc_->GetConfiguration();
config.ice_check_min_interval = 100;
EXPECT_TRUE(pc_->SetConfiguration(config));
PeerConnectionInterface::RTCConfiguration new_config =
pc_->GetConfiguration();
EXPECT_EQ(new_config.ice_check_min_interval, 100);
}
// Test that when SetConfiguration changes both the pool size and other
// attributes, the pooled session is created with the updated attributes.
TEST_P(PeerConnectionInterfaceTest,
SetConfigurationCreatesPooledSessionCorrectly) {
CreatePeerConnection();
PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration();
config.ice_candidate_pool_size = 1;
PeerConnectionInterface::IceServer server;
server.uri = kStunAddressOnly;
config.servers.push_back(server);
config.type = PeerConnectionInterface::kRelay;
EXPECT_TRUE(pc_->SetConfiguration(config));
const cricket::FakePortAllocatorSession* session =
static_cast<const cricket::FakePortAllocatorSession*>(
port_allocator_->GetPooledSession());
ASSERT_NE(nullptr, session);
EXPECT_EQ(1UL, session->stun_servers().size());
}
// Test that after SetLocalDescription, changing the pool size is not allowed,
// and an invalid modification error is returned.
TEST_P(PeerConnectionInterfaceTest,
CantChangePoolSizeAfterSetLocalDescription) {
CreatePeerConnection();
// Start by setting a size of 1.
PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration();
config.ice_candidate_pool_size = 1;
EXPECT_TRUE(pc_->SetConfiguration(config));
// Set remote offer; can still change pool size at this point.
CreateOfferAsRemoteDescription();
config.ice_candidate_pool_size = 2;
EXPECT_TRUE(pc_->SetConfiguration(config));
// Set local answer; now it's too late.
CreateAnswerAsLocalDescription();
config.ice_candidate_pool_size = 3;
RTCError error;
EXPECT_FALSE(pc_->SetConfiguration(config, &error));
EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, error.type());
}
// Test that after setting an answer, extra pooled sessions are discarded. The
// ICE candidate pool is only intended to be used for the first offer/answer.
TEST_P(PeerConnectionInterfaceTest,
ExtraPooledSessionsDiscardedAfterApplyingAnswer) {
CreatePeerConnection();
// Set a larger-than-necessary size.
PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration();
config.ice_candidate_pool_size = 4;
EXPECT_TRUE(pc_->SetConfiguration(config));
// Do offer/answer.
CreateOfferAsRemoteDescription();
CreateAnswerAsLocalDescription();
// Expect no pooled sessions to be left.
const cricket::PortAllocatorSession* session =
port_allocator_->GetPooledSession();
EXPECT_EQ(nullptr, session);
}
// After Close is called, pooled candidates should be discarded so as to not
// waste network resources.
TEST_P(PeerConnectionInterfaceTest, PooledSessionsDiscardedAfterClose) {
CreatePeerConnection();
PeerConnectionInterface::RTCConfiguration config = pc_->GetConfiguration();
config.ice_candidate_pool_size = 3;
EXPECT_TRUE(pc_->SetConfiguration(config));
pc_->Close();
// Expect no pooled sessions to be left.
const cricket::PortAllocatorSession* session =
port_allocator_->GetPooledSession();
EXPECT_EQ(nullptr, session);
}
// Test that SetConfiguration returns an invalid modification error if
// modifying a field in the configuration that isn't allowed to be modified.
TEST_P(PeerConnectionInterfaceTest,
SetConfigurationReturnsInvalidModificationError) {
PeerConnectionInterface::RTCConfiguration config;
config.bundle_policy = PeerConnectionInterface::kBundlePolicyBalanced;
config.rtcp_mux_policy = PeerConnectionInterface::kRtcpMuxPolicyNegotiate;
config.continual_gathering_policy = PeerConnectionInterface::GATHER_ONCE;
CreatePeerConnection(config);
PeerConnectionInterface::RTCConfiguration modified_config =
pc_->GetConfiguration();
modified_config.bundle_policy =
PeerConnectionInterface::kBundlePolicyMaxBundle;
RTCError error;
EXPECT_FALSE(pc_->SetConfiguration(modified_config, &error));
EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, error.type());
modified_config = pc_->GetConfiguration();
modified_config.rtcp_mux_policy =
PeerConnectionInterface::kRtcpMuxPolicyRequire;
error.set_type(RTCErrorType::NONE);
EXPECT_FALSE(pc_->SetConfiguration(modified_config, &error));
EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, error.type());
modified_config = pc_->GetConfiguration();
modified_config.continual_gathering_policy =
PeerConnectionInterface::GATHER_CONTINUALLY;
error.set_type(RTCErrorType::NONE);
EXPECT_FALSE(pc_->SetConfiguration(modified_config, &error));
EXPECT_EQ(RTCErrorType::INVALID_MODIFICATION, error.type());
}
// Test that SetConfiguration returns a range error if the candidate pool size
// is negative or larger than allowed by the spec.
TEST_P(PeerConnectionInterfaceTest,
SetConfigurationReturnsRangeErrorForBadCandidatePoolSize) {
PeerConnectionInterface::RTCConfiguration config;
CreatePeerConnection(config);
config = pc_->GetConfiguration();
config.ice_candidate_pool_size = -1;
RTCError error;
EXPECT_FALSE(pc_->SetConfiguration(config, &error));
EXPECT_EQ(RTCErrorType::INVALID_RANGE, error.type());
config.ice_candidate_pool_size = INT_MAX;
error.set_type(RTCErrorType::NONE);
EXPECT_FALSE(pc_->SetConfiguration(config, &error));
EXPECT_EQ(RTCErrorType::INVALID_RANGE, error.type());
}
// Test that SetConfiguration returns a syntax error if parsing an ICE server
// URL failed.
TEST_P(PeerConnectionInterfaceTest,
SetConfigurationReturnsSyntaxErrorFromBadIceUrls) {
PeerConnectionInterface::RTCConfiguration config;
CreatePeerConnection(config);
config = pc_->GetConfiguration();
PeerConnectionInterface::IceServer bad_server;
bad_server.uri = "stunn:www.example.com";
config.servers.push_back(bad_server);
RTCError error;
EXPECT_FALSE(pc_->SetConfiguration(config, &error));
EXPECT_EQ(RTCErrorType::SYNTAX_ERROR, error.type());
}
// Test that SetConfiguration returns an invalid parameter error if a TURN
// IceServer is missing a username or password.
TEST_P(PeerConnectionInterfaceTest,
SetConfigurationReturnsInvalidParameterIfCredentialsMissing) {
PeerConnectionInterface::RTCConfiguration config;
CreatePeerConnection(config);
config = pc_->GetConfiguration();
PeerConnectionInterface::IceServer bad_server;
bad_server.uri = "turn:www.example.com";
// Missing password.
bad_server.username = "foo";
config.servers.push_back(bad_server);
RTCError error;
EXPECT_FALSE(pc_->SetConfiguration(config, &error));
EXPECT_EQ(RTCErrorType::INVALID_PARAMETER, error.type());
}
// Test that PeerConnection::Close changes the states to closed and all remote
// tracks change state to ended.
TEST_P(PeerConnectionInterfaceTest, CloseAndTestStreamsAndStates) {
// Initialize a PeerConnection and negotiate local and remote session
// description.
InitiateCall();
// With Plan B, verify the stream count. The analog with Unified Plan is the
// RtpTransceiver count.
if (sdp_semantics_ == SdpSemantics::kPlanB) {
ASSERT_EQ(1u, pc_->local_streams()->count());
ASSERT_EQ(1u, pc_->remote_streams()->count());
} else {
ASSERT_EQ(2u, pc_->GetTransceivers().size());
}
pc_->Close();
EXPECT_EQ(PeerConnectionInterface::kClosed, pc_->signaling_state());
EXPECT_EQ(PeerConnectionInterface::kIceConnectionClosed,
pc_->ice_connection_state());
EXPECT_EQ(PeerConnectionInterface::kIceGatheringComplete,
pc_->ice_gathering_state());
if (sdp_semantics_ == SdpSemantics::kPlanB) {
EXPECT_EQ(1u, pc_->local_streams()->count());
EXPECT_EQ(1u, pc_->remote_streams()->count());
} else {
// Verify that the RtpTransceivers are still present but all stopped.
EXPECT_EQ(2u, pc_->GetTransceivers().size());
for (const auto& transceiver : pc_->GetTransceivers()) {
EXPECT_TRUE(transceiver->stopped());
}
}
auto audio_receiver = GetFirstReceiverOfType(cricket::MEDIA_TYPE_AUDIO);
ASSERT_TRUE(audio_receiver);
auto video_receiver = GetFirstReceiverOfType(cricket::MEDIA_TYPE_VIDEO);
ASSERT_TRUE(video_receiver);
// Track state may be updated asynchronously.
EXPECT_EQ_WAIT(MediaStreamTrackInterface::kEnded,
audio_receiver->track()->state(), kTimeout);
EXPECT_EQ_WAIT(MediaStreamTrackInterface::kEnded,
video_receiver->track()->state(), kTimeout);
}
// Test that PeerConnection methods fails gracefully after
// PeerConnection::Close has been called.
// Don't run under Unified Plan since the stream API is not available.
TEST_F(PeerConnectionInterfaceTestPlanB, CloseAndTestMethods) {
CreatePeerConnectionWithoutDtls();
AddAudioVideoStream(kStreamId1, "audio_label", "video_label");
CreateOfferAsRemoteDescription();
CreateAnswerAsLocalDescription();
ASSERT_EQ(1u, pc_->local_streams()->count());
rtc::scoped_refptr<MediaStreamInterface> local_stream =
pc_->local_streams()->at(0);
pc_->Close();
pc_->RemoveStream(local_stream);
EXPECT_FALSE(pc_->AddStream(local_stream));
EXPECT_TRUE(pc_->CreateDataChannel("test", NULL) == NULL);
EXPECT_TRUE(pc_->local_description() != NULL);
EXPECT_TRUE(pc_->remote_description() != NULL);
std::unique_ptr<SessionDescriptionInterface> offer;
EXPECT_FALSE(DoCreateOffer(&offer, nullptr));
std::unique_ptr<SessionDescriptionInterface> answer;
EXPECT_FALSE(DoCreateAnswer(&answer, nullptr));
std::string sdp;
ASSERT_TRUE(pc_->remote_description()->ToString(&sdp));
std::unique_ptr<SessionDescriptionInterface> remote_offer(
webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
EXPECT_FALSE(DoSetRemoteDescription(std::move(remote_offer)));
ASSERT_TRUE(pc_->local_description()->ToString(&sdp));
std::unique_ptr<SessionDescriptionInterface> local_offer(
webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
EXPECT_FALSE(DoSetLocalDescription(std::move(local_offer)));
}
// Test that GetStats can still be called after PeerConnection::Close.
TEST_P(PeerConnectionInterfaceTest, CloseAndGetStats) {
InitiateCall();
pc_->Close();
DoGetStats(NULL);
}
// NOTE: The series of tests below come from what used to be
// mediastreamsignaling_unittest.cc, and are mostly aimed at testing that
// setting a remote or local description has the expected effects.
// This test verifies that the remote MediaStreams corresponding to a received
// SDP string is created. In this test the two separate MediaStreams are
// signaled.
TEST_P(PeerConnectionInterfaceTest, UpdateRemoteStreams) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
CreateAndSetRemoteOffer(GetSdpStringWithStream1());
rtc::scoped_refptr<StreamCollection> reference(CreateStreamCollection(1, 1));
EXPECT_TRUE(
CompareStreamCollections(observer_.remote_streams(), reference.get()));
MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
EXPECT_TRUE(remote_stream->GetVideoTracks()[0]->GetSource() != nullptr);
// Create a session description based on another SDP with another
// MediaStream.
CreateAndSetRemoteOffer(GetSdpStringWithStream1And2());
rtc::scoped_refptr<StreamCollection> reference2(CreateStreamCollection(2, 1));
EXPECT_TRUE(
CompareStreamCollections(observer_.remote_streams(), reference2.get()));
}
// This test verifies that when remote tracks are added/removed from SDP, the
// created remote streams are updated appropriately.
// Don't run under Unified Plan since this test uses Plan B SDP to test Plan B
// specific behavior.
TEST_F(PeerConnectionInterfaceTestPlanB,
AddRemoveTrackFromExistingRemoteMediaStream) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
std::unique_ptr<SessionDescriptionInterface> desc_ms1 =
CreateSessionDescriptionAndReference(1, 1);
EXPECT_TRUE(DoSetRemoteDescription(std::move(desc_ms1)));
EXPECT_TRUE(CompareStreamCollections(observer_.remote_streams(),
reference_collection_));
// Add extra audio and video tracks to the same MediaStream.
std::unique_ptr<SessionDescriptionInterface> desc_ms1_two_tracks =
CreateSessionDescriptionAndReference(2, 2);
EXPECT_TRUE(DoSetRemoteDescription(std::move(desc_ms1_two_tracks)));
EXPECT_TRUE(CompareStreamCollections(observer_.remote_streams(),
reference_collection_));
rtc::scoped_refptr<AudioTrackInterface> audio_track2 =
observer_.remote_streams()->at(0)->GetAudioTracks()[1];
EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, audio_track2->state());
rtc::scoped_refptr<VideoTrackInterface> video_track2 =
observer_.remote_streams()->at(0)->GetVideoTracks()[1];
EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, video_track2->state());
// Remove the extra audio and video tracks.
std::unique_ptr<SessionDescriptionInterface> desc_ms2 =
CreateSessionDescriptionAndReference(1, 1);
MockTrackObserver audio_track_observer(audio_track2);
MockTrackObserver video_track_observer(video_track2);
EXPECT_CALL(audio_track_observer, OnChanged()).Times(Exactly(1));
EXPECT_CALL(video_track_observer, OnChanged()).Times(Exactly(1));
EXPECT_TRUE(DoSetRemoteDescription(std::move(desc_ms2)));
EXPECT_TRUE(CompareStreamCollections(observer_.remote_streams(),
reference_collection_));
// Track state may be updated asynchronously.
EXPECT_EQ_WAIT(webrtc::MediaStreamTrackInterface::kEnded,
audio_track2->state(), kTimeout);
EXPECT_EQ_WAIT(webrtc::MediaStreamTrackInterface::kEnded,
video_track2->state(), kTimeout);
}
// This tests that remote tracks are ended if a local session description is set
// that rejects the media content type.
TEST_P(PeerConnectionInterfaceTest, RejectMediaContent) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
// First create and set a remote offer, then reject its video content in our
// answer.
CreateAndSetRemoteOffer(kSdpStringWithStream1PlanB);
auto audio_receiver = GetFirstReceiverOfType(cricket::MEDIA_TYPE_AUDIO);
ASSERT_TRUE(audio_receiver);
auto video_receiver = GetFirstReceiverOfType(cricket::MEDIA_TYPE_VIDEO);
ASSERT_TRUE(video_receiver);
rtc::scoped_refptr<MediaStreamTrackInterface> remote_audio =
audio_receiver->track();
EXPECT_EQ(webrtc::MediaStreamTrackInterface::kLive, remote_audio->state());
rtc::scoped_refptr<MediaStreamTrackInterface> remote_video =
video_receiver->track();
EXPECT_EQ(MediaStreamTrackInterface::kLive, remote_video->state());
std::unique_ptr<SessionDescriptionInterface> local_answer;
EXPECT_TRUE(DoCreateAnswer(&local_answer, nullptr));
cricket::ContentInfo* video_info =
local_answer->description()->GetContentByName("video");
video_info->rejected = true;
EXPECT_TRUE(DoSetLocalDescription(std::move(local_answer)));
EXPECT_EQ(MediaStreamTrackInterface::kEnded, remote_video->state());
EXPECT_EQ(MediaStreamTrackInterface::kLive, remote_audio->state());
// Now create an offer where we reject both video and audio.
std::unique_ptr<SessionDescriptionInterface> local_offer;
EXPECT_TRUE(DoCreateOffer(&local_offer, nullptr));
video_info = local_offer->description()->GetContentByName("video");
ASSERT_TRUE(video_info != nullptr);
video_info->rejected = true;
cricket::ContentInfo* audio_info =
local_offer->description()->GetContentByName("audio");
ASSERT_TRUE(audio_info != nullptr);
audio_info->rejected = true;
EXPECT_TRUE(DoSetLocalDescription(std::move(local_offer)));
// Track state may be updated asynchronously.
EXPECT_EQ_WAIT(MediaStreamTrackInterface::kEnded, remote_audio->state(),
kTimeout);
EXPECT_EQ_WAIT(MediaStreamTrackInterface::kEnded, remote_video->state(),
kTimeout);
}
// This tests that we won't crash if the remote track has been removed outside
// of PeerConnection and then PeerConnection tries to reject the track.
// Don't run under Unified Plan since the stream API is not available.
TEST_F(PeerConnectionInterfaceTestPlanB, RemoveTrackThenRejectMediaContent) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
CreateAndSetRemoteOffer(GetSdpStringWithStream1());
MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
remote_stream->RemoveTrack(remote_stream->GetVideoTracks()[0]);
remote_stream->RemoveTrack(remote_stream->GetAudioTracks()[0]);
std::unique_ptr<SessionDescriptionInterface> local_answer(
webrtc::CreateSessionDescription(SdpType::kAnswer,
GetSdpStringWithStream1(), nullptr));
cricket::ContentInfo* video_info =
local_answer->description()->GetContentByName("video");
video_info->rejected = true;
cricket::ContentInfo* audio_info =
local_answer->description()->GetContentByName("audio");
audio_info->rejected = true;
EXPECT_TRUE(DoSetLocalDescription(std::move(local_answer)));
// No crash is a pass.
}
// This tests that if a recvonly remote description is set, no remote streams
// will be created, even if the description contains SSRCs/MSIDs.
// See: https://code.google.com/p/webrtc/issues/detail?id=5054
TEST_P(PeerConnectionInterfaceTest, RecvonlyDescriptionDoesntCreateStream) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
std::string recvonly_offer = GetSdpStringWithStream1();
absl::StrReplaceAll({{kSendrecv, kRecvonly}}, &recvonly_offer);
CreateAndSetRemoteOffer(recvonly_offer);
EXPECT_EQ(0u, observer_.remote_streams()->count());
}
// This tests that a default MediaStream is created if a remote session
// description doesn't contain any streams and no MSID support.
// It also tests that the default stream is updated if a video m-line is added
// in a subsequent session description.
// Don't run under Unified Plan since this behavior is Plan B specific.
TEST_F(PeerConnectionInterfaceTestPlanB, SdpWithoutMsidCreatesDefaultStream) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
CreateAndSetRemoteOffer(kSdpStringWithoutStreamsAudioOnly);
ASSERT_EQ(1u, observer_.remote_streams()->count());
MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
EXPECT_EQ(1u, remote_stream->GetAudioTracks().size());
EXPECT_EQ(0u, remote_stream->GetVideoTracks().size());
EXPECT_EQ("default", remote_stream->id());
CreateAndSetRemoteOffer(kSdpStringWithoutStreams);
ASSERT_EQ(1u, observer_.remote_streams()->count());
ASSERT_EQ(1u, remote_stream->GetAudioTracks().size());
EXPECT_EQ("defaulta0", remote_stream->GetAudioTracks()[0]->id());
EXPECT_EQ(MediaStreamTrackInterface::kLive,
remote_stream->GetAudioTracks()[0]->state());
ASSERT_EQ(1u, remote_stream->GetVideoTracks().size());
EXPECT_EQ("defaultv0", remote_stream->GetVideoTracks()[0]->id());
EXPECT_EQ(MediaStreamTrackInterface::kLive,
remote_stream->GetVideoTracks()[0]->state());
}
// This tests that a default MediaStream is created if a remote session
// description doesn't contain any streams and media direction is send only.
// Don't run under Unified Plan since this behavior is Plan B specific.
TEST_F(PeerConnectionInterfaceTestPlanB,
SendOnlySdpWithoutMsidCreatesDefaultStream) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
CreateAndSetRemoteOffer(kSdpStringSendOnlyWithoutStreams);
ASSERT_EQ(1u, observer_.remote_streams()->count());
MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
EXPECT_EQ(1u, remote_stream->GetAudioTracks().size());
EXPECT_EQ(1u, remote_stream->GetVideoTracks().size());
EXPECT_EQ("default", remote_stream->id());
}
// This tests that it won't crash when PeerConnection tries to remove
// a remote track that as already been removed from the MediaStream.
// Don't run under Unified Plan since this behavior is Plan B specific.
TEST_F(PeerConnectionInterfaceTestPlanB, RemoveAlreadyGoneRemoteStream) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
CreateAndSetRemoteOffer(GetSdpStringWithStream1());
MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
remote_stream->RemoveTrack(remote_stream->GetAudioTracks()[0]);
remote_stream->RemoveTrack(remote_stream->GetVideoTracks()[0]);
CreateAndSetRemoteOffer(kSdpStringWithoutStreams);
// No crash is a pass.
}
// This tests that a default MediaStream is created if the remote session
// description doesn't contain any streams and don't contain an indication if
// MSID is supported.
// Don't run under Unified Plan since this behavior is Plan B specific.
TEST_F(PeerConnectionInterfaceTestPlanB,
SdpWithoutMsidAndStreamsCreatesDefaultStream) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
CreateAndSetRemoteOffer(kSdpStringWithoutStreams);
ASSERT_EQ(1u, observer_.remote_streams()->count());
MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
EXPECT_EQ(1u, remote_stream->GetAudioTracks().size());
EXPECT_EQ(1u, remote_stream->GetVideoTracks().size());
}
// This tests that a default MediaStream is not created if the remote session
// description doesn't contain any streams but does support MSID.
// Don't run under Unified Plan since this behavior is Plan B specific.
TEST_F(PeerConnectionInterfaceTestPlanB, SdpWithMsidDontCreatesDefaultStream) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
CreateAndSetRemoteOffer(kSdpStringWithMsidWithoutStreams);
EXPECT_EQ(0u, observer_.remote_streams()->count());
}
// This tests that when setting a new description, the old default tracks are
// not destroyed and recreated.
// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=5250
// Don't run under Unified Plan since this behavior is Plan B specific.
TEST_F(PeerConnectionInterfaceTestPlanB,
DefaultTracksNotDestroyedAndRecreated) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
CreateAndSetRemoteOffer(kSdpStringWithoutStreamsAudioOnly);
ASSERT_EQ(1u, observer_.remote_streams()->count());
MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
ASSERT_EQ(1u, remote_stream->GetAudioTracks().size());
// Set the track to "disabled", then set a new description and ensure the
// track is still disabled, which ensures it hasn't been recreated.
remote_stream->GetAudioTracks()[0]->set_enabled(false);
CreateAndSetRemoteOffer(kSdpStringWithoutStreamsAudioOnly);
ASSERT_EQ(1u, remote_stream->GetAudioTracks().size());
EXPECT_FALSE(remote_stream->GetAudioTracks()[0]->enabled());
}
// This tests that a default MediaStream is not created if a remote session
// description is updated to not have any MediaStreams.
// Don't run under Unified Plan since this behavior is Plan B specific.
TEST_F(PeerConnectionInterfaceTestPlanB, VerifyDefaultStreamIsNotCreated) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
CreateAndSetRemoteOffer(GetSdpStringWithStream1());
rtc::scoped_refptr<StreamCollection> reference(CreateStreamCollection(1, 1));
EXPECT_TRUE(
CompareStreamCollections(observer_.remote_streams(), reference.get()));
CreateAndSetRemoteOffer(kSdpStringWithoutStreams);
EXPECT_EQ(0u, observer_.remote_streams()->count());
}
// This tests that a default MediaStream is created if a remote SDP comes from
// an endpoint that doesn't signal SSRCs, but signals media stream IDs.
TEST_F(PeerConnectionInterfaceTestPlanB,
SdpWithMsidWithoutSsrcCreatesDefaultStream) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
std::string sdp_string = kSdpStringWithoutStreamsAudioOnly;
// Add a=msid lines to simulate a Unified Plan endpoint that only
// signals stream IDs with a=msid lines.
sdp_string.append("a=msid:audio_stream_id audio_track_id\n");
CreateAndSetRemoteOffer(sdp_string);
ASSERT_EQ(1u, observer_.remote_streams()->count());
MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
EXPECT_EQ("default", remote_stream->id());
ASSERT_EQ(1u, remote_stream->GetAudioTracks().size());
}
// This tests that when a Plan B endpoint receives an SDP that signals no media
// stream IDs indicated by the special character "-" in the a=msid line, that
// a default stream ID will be used for the MediaStream ID. This can occur
// when a Unified Plan endpoint signals no media stream IDs, but signals both
// a=ssrc msid and a=msid lines for interop signaling with Plan B.
TEST_F(PeerConnectionInterfaceTestPlanB,
SdpWithEmptyMsidAndSsrcCreatesDefaultStreamId) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
// Add a a=msid line to the SDP. This is prioritized when parsing the SDP, so
// the sender's stream ID will be interpreted as no stream IDs.
std::string sdp_string = kSdpStringWithStream1AudioTrackOnly;
sdp_string.append("a=msid:- audiotrack0\n");
CreateAndSetRemoteOffer(sdp_string);
ASSERT_EQ(1u, observer_.remote_streams()->count());
// Because SSRCs are signaled the track ID will be what was signaled in the
// a=msid line.
EXPECT_EQ("audiotrack0", observer_.last_added_track_label_);
MediaStreamInterface* remote_stream = observer_.remote_streams()->at(0);
EXPECT_EQ("default", remote_stream->id());
ASSERT_EQ(1u, remote_stream->GetAudioTracks().size());
// Previously a bug ocurred when setting the remote description a second time.
// This is because we checked equality of the remote StreamParams stream ID
// (empty), and the previously set stream ID for the remote sender
// ("default"). This cause a track to be removed, then added, when really
// nothing should occur because it is the same track.
CreateAndSetRemoteOffer(sdp_string);
EXPECT_EQ(0u, observer_.remove_track_events_.size());
EXPECT_EQ(1u, observer_.add_track_events_.size());
EXPECT_EQ("audiotrack0", observer_.last_added_track_label_);
remote_stream = observer_.remote_streams()->at(0);
EXPECT_EQ("default", remote_stream->id());
ASSERT_EQ(1u, remote_stream->GetAudioTracks().size());
}
// This tests that an RtpSender is created when the local description is set
// after adding a local stream.
// TODO(deadbeef): This test and the one below it need to be updated when
// an RtpSender's lifetime isn't determined by when a local description is set.
// Don't run under Unified Plan since this behavior is Plan B specific.
TEST_F(PeerConnectionInterfaceTestPlanB, LocalDescriptionChanged) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
// Create an offer with 1 stream with 2 tracks of each type.
rtc::scoped_refptr<StreamCollection> stream_collection =
CreateStreamCollection(1, 2);
pc_->AddStream(stream_collection->at(0));
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
EXPECT_TRUE(DoSetLocalDescription(std::move(offer)));
auto senders = pc_->GetSenders();
EXPECT_EQ(4u, senders.size());
EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0]));
EXPECT_TRUE(ContainsSender(senders, kAudioTracks[1]));
EXPECT_TRUE(ContainsSender(senders, kVideoTracks[1]));
// Remove an audio and video track.
pc_->RemoveStream(stream_collection->at(0));
stream_collection = CreateStreamCollection(1, 1);
pc_->AddStream(stream_collection->at(0));
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
EXPECT_TRUE(DoSetLocalDescription(std::move(offer)));
senders = pc_->GetSenders();
EXPECT_EQ(2u, senders.size());
EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0]));
EXPECT_FALSE(ContainsSender(senders, kAudioTracks[1]));
EXPECT_FALSE(ContainsSender(senders, kVideoTracks[1]));
}
// This tests that an RtpSender is created when the local description is set
// before adding a local stream.
// Don't run under Unified Plan since this behavior is Plan B specific.
TEST_F(PeerConnectionInterfaceTestPlanB,
AddLocalStreamAfterLocalDescriptionChanged) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
rtc::scoped_refptr<StreamCollection> stream_collection =
CreateStreamCollection(1, 2);
// Add a stream to create the offer, but remove it afterwards.
pc_->AddStream(stream_collection->at(0));
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
pc_->RemoveStream(stream_collection->at(0));
EXPECT_TRUE(DoSetLocalDescription(std::move(offer)));
auto senders = pc_->GetSenders();
EXPECT_EQ(0u, senders.size());
pc_->AddStream(stream_collection->at(0));
senders = pc_->GetSenders();
EXPECT_EQ(4u, senders.size());
EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0]));
EXPECT_TRUE(ContainsSender(senders, kAudioTracks[1]));
EXPECT_TRUE(ContainsSender(senders, kVideoTracks[1]));
}
// This tests that the expected behavior occurs if the SSRC on a local track is
// changed when SetLocalDescription is called.
TEST_P(PeerConnectionInterfaceTest,
ChangeSsrcOnTrackInLocalSessionDescription) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
AddAudioTrack(kAudioTracks[0]);
AddVideoTrack(kVideoTracks[0]);
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
// Grab a copy of the offer before it gets passed into the PC.
std::unique_ptr<SessionDescriptionInterface> modified_offer =
webrtc::CreateSessionDescription(
webrtc::SdpType::kOffer, offer->session_id(),
offer->session_version(),
absl::WrapUnique(offer->description()->Copy()));
EXPECT_TRUE(DoSetLocalDescription(std::move(offer)));
auto senders = pc_->GetSenders();
EXPECT_EQ(2u, senders.size());
EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0]));
// Change the ssrc of the audio and video track.
cricket::MediaContentDescription* desc =
cricket::GetFirstAudioContentDescription(modified_offer->description());
ASSERT_TRUE(desc != NULL);
for (StreamParams& stream : desc->mutable_streams()) {
for (unsigned int& ssrc : stream.ssrcs) {
++ssrc;
}
}
desc =
cricket::GetFirstVideoContentDescription(modified_offer->description());
ASSERT_TRUE(desc != NULL);
for (StreamParams& stream : desc->mutable_streams()) {
for (unsigned int& ssrc : stream.ssrcs) {
++ssrc;
}
}
EXPECT_TRUE(DoSetLocalDescription(std::move(modified_offer)));
senders = pc_->GetSenders();
EXPECT_EQ(2u, senders.size());
EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0]));
EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0]));
// TODO(deadbeef): Once RtpSenders expose parameters, check that the SSRC
// changed.
}
// This tests that the expected behavior occurs if a new session description is
// set with the same tracks, but on a different MediaStream.
// Don't run under Unified Plan since the stream API is not available.
TEST_F(PeerConnectionInterfaceTestPlanB,
SignalSameTracksInSeparateMediaStream) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
rtc::scoped_refptr<StreamCollection> stream_collection =
CreateStreamCollection(2, 1);
pc_->AddStream(stream_collection->at(0));
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
EXPECT_TRUE(DoSetLocalDescription(std::move(offer)));
auto senders = pc_->GetSenders();
EXPECT_EQ(2u, senders.size());
EXPECT_TRUE(ContainsSender(senders, kAudioTracks[0], kStreams[0]));
EXPECT_TRUE(ContainsSender(senders, kVideoTracks[0], kStreams[0]));
// Add a new MediaStream but with the same tracks as in the first stream.
rtc::scoped_refptr<webrtc::MediaStreamInterface> stream_1(
webrtc::MediaStream::Create(kStreams[1]));
stream_1->AddTrack(stream_collection->at(0)->GetVideoTracks()[0]);
stream_1->AddTrack(stream_collection->at(0)->GetAudioTracks()[0]);
pc_->AddStream(stream_1);
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
EXPECT_TRUE(DoSetLocalDescription(std::move(offer)));
auto new_senders = pc_->GetSenders();
// Should be the same senders as before, but with updated stream id.
// Note that this behavior is subject to change in the future.
// We may decide the PC should ignore existing tracks in AddStream.
EXPECT_EQ(senders, new_senders);
EXPECT_TRUE(ContainsSender(new_senders, kAudioTracks[0], kStreams[1]));
EXPECT_TRUE(ContainsSender(new_senders, kVideoTracks[0], kStreams[1]));
}
// This tests that PeerConnectionObserver::OnAddTrack is correctly called.
TEST_P(PeerConnectionInterfaceTest, OnAddTrackCallback) {
RTCConfiguration config;
config.enable_dtls_srtp = true;
CreatePeerConnection(config);
CreateAndSetRemoteOffer(kSdpStringWithStream1AudioTrackOnly);
EXPECT_EQ(observer_.num_added_tracks_, 1);
EXPECT_EQ(observer_.last_added_track_label_, kAudioTracks[0]);
// Create and set the updated remote SDP.
CreateAndSetRemoteOffer(kSdpStringWithStream1PlanB);
EXPECT_EQ(observer_.num_added_tracks_, 2);
EXPECT_EQ(observer_.last_added_track_label_, kVideoTracks[0]);
}
// Test that when SetConfiguration is called and the configuration is
// changing, the next offer causes an ICE restart.
TEST_P(PeerConnectionInterfaceTest, SetConfigurationCausingIceRestart) {
PeerConnectionInterface::RTCConfiguration config;
config.type = PeerConnectionInterface::kRelay;
CreatePeerConnection(config);
config = pc_->GetConfiguration();
AddAudioTrack(kAudioTracks[0], {kStreamId1});
AddVideoTrack(kVideoTracks[0], {kStreamId1});
// Do initial offer/answer so there's something to restart.
CreateOfferAsLocalDescription();
CreateAnswerAsRemoteDescription(GetSdpStringWithStream1());
// Grab the ufrags.
std::vector<std::string> initial_ufrags = GetUfrags(pc_->local_description());
// Change ICE policy, which should trigger an ICE restart on the next offer.
config.type = PeerConnectionInterface::kAll;
EXPECT_TRUE(pc_->SetConfiguration(config));
CreateOfferAsLocalDescription();
// Grab the new ufrags.
std::vector<std::string> subsequent_ufrags =
GetUfrags(pc_->local_description());
// Sanity check.
EXPECT_EQ(initial_ufrags.size(), subsequent_ufrags.size());
// Check that each ufrag is different.
for (int i = 0; i < static_cast<int>(initial_ufrags.size()); ++i) {
EXPECT_NE(initial_ufrags[i], subsequent_ufrags[i]);
}
}
// Test that when SetConfiguration is called and the configuration *isn't*
// changing, the next offer does *not* cause an ICE restart.
TEST_P(PeerConnectionInterfaceTest, SetConfigurationNotCausingIceRestart) {
PeerConnectionInterface::RTCConfiguration config;
config.type = PeerConnectionInterface::kRelay;
CreatePeerConnection(config);
config = pc_->GetConfiguration();
AddAudioTrack(kAudioTracks[0]);
AddVideoTrack(kVideoTracks[0]);
// Do initial offer/answer so there's something to restart.
CreateOfferAsLocalDescription();
CreateAnswerAsRemoteDescription(GetSdpStringWithStream1());
// Grab the ufrags.
std::vector<std::string> initial_ufrags = GetUfrags(pc_->local_description());
// Call SetConfiguration with a config identical to what the PC was
// constructed with.
EXPECT_TRUE(pc_->SetConfiguration(config));
CreateOfferAsLocalDescription();
// Grab the new ufrags.
std::vector<std::string> subsequent_ufrags =
GetUfrags(pc_->local_description());
EXPECT_EQ(initial_ufrags, subsequent_ufrags);
}
// Test for a weird corner case scenario:
// 1. Audio/video session established.
// 2. SetConfiguration changes ICE config; ICE restart needed.
// 3. ICE restart initiated by remote peer, but only for one m= section.
// 4. Next createOffer should initiate an ICE restart, but only for the other
// m= section; it would be pointless to do an ICE restart for the m= section
// that was already restarted.
TEST_P(PeerConnectionInterfaceTest, SetConfigurationCausingPartialIceRestart) {
PeerConnectionInterface::RTCConfiguration config;
config.type = PeerConnectionInterface::kRelay;
CreatePeerConnection(config);
config = pc_->GetConfiguration();
AddAudioTrack(kAudioTracks[0], {kStreamId1});
AddVideoTrack(kVideoTracks[0], {kStreamId1});
// Do initial offer/answer so there's something to restart.
CreateOfferAsLocalDescription();
CreateAnswerAsRemoteDescription(GetSdpStringWithStream1());
// Change ICE policy, which should set the "needs-ice-restart" flag.
config.type = PeerConnectionInterface::kAll;
EXPECT_TRUE(pc_->SetConfiguration(config));
// Do ICE restart for the first m= section, initiated by remote peer.
std::unique_ptr<webrtc::SessionDescriptionInterface> remote_offer(
webrtc::CreateSessionDescription(SdpType::kOffer,
GetSdpStringWithStream1(), nullptr));
ASSERT_TRUE(remote_offer);
remote_offer->description()->transport_infos()[0].description.ice_ufrag =
"modified";
EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_offer)));
CreateAnswerAsLocalDescription();
// Grab the ufrags.
std::vector<std::string> initial_ufrags = GetUfrags(pc_->local_description());
ASSERT_EQ(2U, initial_ufrags.size());
// Create offer and grab the new ufrags.
CreateOfferAsLocalDescription();
std::vector<std::string> subsequent_ufrags =
GetUfrags(pc_->local_description());
ASSERT_EQ(2U, subsequent_ufrags.size());
// Ensure that only the ufrag for the second m= section changed.
EXPECT_EQ(initial_ufrags[0], subsequent_ufrags[0]);
EXPECT_NE(initial_ufrags[1], subsequent_ufrags[1]);
}
// Tests that the methods to return current/pending descriptions work as
// expected at different points in the offer/answer exchange. This test does
// one offer/answer exchange as the offerer, then another as the answerer.
TEST_P(PeerConnectionInterfaceTest, CurrentAndPendingDescriptions) {
// This disables DTLS so we can apply an answer to ourselves.
CreatePeerConnection();
// Create initial local offer and get SDP (which will also be used as
// answer/pranswer);
std::unique_ptr<SessionDescriptionInterface> local_offer;
ASSERT_TRUE(DoCreateOffer(&local_offer, nullptr));
std::string sdp;
EXPECT_TRUE(local_offer->ToString(&sdp));
// Set local offer.
SessionDescriptionInterface* local_offer_ptr = local_offer.get();
EXPECT_TRUE(DoSetLocalDescription(std::move(local_offer)));
EXPECT_EQ(local_offer_ptr, pc_->pending_local_description());
EXPECT_EQ(nullptr, pc_->pending_remote_description());
EXPECT_EQ(nullptr, pc_->current_local_description());
EXPECT_EQ(nullptr, pc_->current_remote_description());
// Set remote pranswer.
std::unique_ptr<SessionDescriptionInterface> remote_pranswer(
webrtc::CreateSessionDescription(SdpType::kPrAnswer, sdp));
SessionDescriptionInterface* remote_pranswer_ptr = remote_pranswer.get();
EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_pranswer)));
EXPECT_EQ(local_offer_ptr, pc_->pending_local_description());
EXPECT_EQ(remote_pranswer_ptr, pc_->pending_remote_description());
EXPECT_EQ(nullptr, pc_->current_local_description());
EXPECT_EQ(nullptr, pc_->current_remote_description());
// Set remote answer.
std::unique_ptr<SessionDescriptionInterface> remote_answer(
webrtc::CreateSessionDescription(SdpType::kAnswer, sdp));
SessionDescriptionInterface* remote_answer_ptr = remote_answer.get();
EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_answer)));
EXPECT_EQ(nullptr, pc_->pending_local_description());
EXPECT_EQ(nullptr, pc_->pending_remote_description());
EXPECT_EQ(local_offer_ptr, pc_->current_local_description());
EXPECT_EQ(remote_answer_ptr, pc_->current_remote_description());
// Set remote offer.
std::unique_ptr<SessionDescriptionInterface> remote_offer(
webrtc::CreateSessionDescription(SdpType::kOffer, sdp));
SessionDescriptionInterface* remote_offer_ptr = remote_offer.get();
EXPECT_TRUE(DoSetRemoteDescription(std::move(remote_offer)));
EXPECT_EQ(remote_offer_ptr, pc_->pending_remote_description());
EXPECT_EQ(nullptr, pc_->pending_local_description());
EXPECT_EQ(local_offer_ptr, pc_->current_local_description());
EXPECT_EQ(remote_answer_ptr, pc_->current_remote_description());
// Set local pranswer.
std::unique_ptr<SessionDescriptionInterface> local_pranswer(
webrtc::CreateSessionDescription(SdpType::kPrAnswer, sdp));
SessionDescriptionInterface* local_pranswer_ptr = local_pranswer.get();
EXPECT_TRUE(DoSetLocalDescription(std::move(local_pranswer)));
EXPECT_EQ(remote_offer_ptr, pc_->pending_remote_description());
EXPECT_EQ(local_pranswer_ptr, pc_->pending_local_description());
EXPECT_EQ(local_offer_ptr, pc_->current_local_description());
EXPECT_EQ(remote_answer_ptr, pc_->current_remote_description());
// Set local answer.
std::unique_ptr<SessionDescriptionInterface> local_answer(
webrtc::CreateSessionDescription(SdpType::kAnswer, sdp));
SessionDescriptionInterface* local_answer_ptr = local_answer.get();
EXPECT_TRUE(DoSetLocalDescription(std::move(local_answer)));
EXPECT_EQ(nullptr, pc_->pending_remote_description());
EXPECT_EQ(nullptr, pc_->pending_local_description());
EXPECT_EQ(remote_offer_ptr, pc_->current_remote_description());
EXPECT_EQ(local_answer_ptr, pc_->current_local_description());
}
// Tests that it won't crash when calling StartRtcEventLog or StopRtcEventLog
// after the PeerConnection is closed.
// This version tests the StartRtcEventLog version that receives a file.
TEST_P(PeerConnectionInterfaceTest,
StartAndStopLoggingToFileAfterPeerConnectionClosed) {
CreatePeerConnection();
// The RtcEventLog will be reset when the PeerConnection is closed.
pc_->Close();
auto test_info = ::testing::UnitTest::GetInstance()->current_test_info();
std::string filename = webrtc::test::OutputPath() +
test_info->test_case_name() + test_info->name();
rtc::PlatformFile file = rtc::CreatePlatformFile(filename);
constexpr int64_t max_size_bytes = 1024;
EXPECT_FALSE(pc_->StartRtcEventLog(file, max_size_bytes));
pc_->StopRtcEventLog();
// Cleanup.
rtc::ClosePlatformFile(file);
rtc::RemoveFile(filename);
}
// Tests that it won't crash when calling StartRtcEventLog or StopRtcEventLog
// after the PeerConnection is closed.
// This version tests the StartRtcEventLog version that receives an object
// of type |RtcEventLogOutput|.
TEST_P(PeerConnectionInterfaceTest,
StartAndStopLoggingToOutputAfterPeerConnectionClosed) {
CreatePeerConnection();
// The RtcEventLog will be reset when the PeerConnection is closed.
pc_->Close();
rtc::PlatformFile file = 0;
int64_t max_size_bytes = 1024;
EXPECT_FALSE(pc_->StartRtcEventLog(
absl::make_unique<webrtc::RtcEventLogOutputFile>(file, max_size_bytes),
webrtc::RtcEventLog::kImmediateOutput));
pc_->StopRtcEventLog();
}
// Test that generated offers/answers include "ice-option:trickle".
TEST_P(PeerConnectionInterfaceTest, OffersAndAnswersHaveTrickleIceOption) {
CreatePeerConnection();
// First, create an offer with audio/video.
RTCOfferAnswerOptions options;
options.offer_to_receive_audio = 1;
options.offer_to_receive_video = 1;
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, &options));
cricket::SessionDescription* desc = offer->description();
ASSERT_EQ(2u, desc->transport_infos().size());
EXPECT_TRUE(desc->transport_infos()[0].description.HasOption("trickle"));
EXPECT_TRUE(desc->transport_infos()[1].description.HasOption("trickle"));
// Apply the offer as a remote description, then create an answer.
EXPECT_TRUE(DoSetRemoteDescription(std::move(offer)));
std::unique_ptr<SessionDescriptionInterface> answer;
ASSERT_TRUE(DoCreateAnswer(&answer, &options));
desc = answer->description();
ASSERT_EQ(2u, desc->transport_infos().size());
EXPECT_TRUE(desc->transport_infos()[0].description.HasOption("trickle"));
EXPECT_TRUE(desc->transport_infos()[1].description.HasOption("trickle"));
}
// Test that ICE renomination isn't offered if it's not enabled in the PC's
// RTCConfiguration.
TEST_P(PeerConnectionInterfaceTest, IceRenominationNotOffered) {
PeerConnectionInterface::RTCConfiguration config;
config.enable_ice_renomination = false;
CreatePeerConnection(config);
AddAudioTrack("foo");
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
cricket::SessionDescription* desc = offer->description();
EXPECT_EQ(1u, desc->transport_infos().size());
EXPECT_FALSE(
desc->transport_infos()[0].description.GetIceParameters().renomination);
}
// Test that the ICE renomination option is present in generated offers/answers
// if it's enabled in the PC's RTCConfiguration.
TEST_P(PeerConnectionInterfaceTest, IceRenominationOptionInOfferAndAnswer) {
PeerConnectionInterface::RTCConfiguration config;
config.enable_ice_renomination = true;
CreatePeerConnection(config);
AddAudioTrack("foo");
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
cricket::SessionDescription* desc = offer->description();
EXPECT_EQ(1u, desc->transport_infos().size());
EXPECT_TRUE(
desc->transport_infos()[0].description.GetIceParameters().renomination);
// Set the offer as a remote description, then create an answer and ensure it
// has the renomination flag too.
EXPECT_TRUE(DoSetRemoteDescription(std::move(offer)));
std::unique_ptr<SessionDescriptionInterface> answer;
ASSERT_TRUE(DoCreateAnswer(&answer, nullptr));
desc = answer->description();
EXPECT_EQ(1u, desc->transport_infos().size());
EXPECT_TRUE(
desc->transport_infos()[0].description.GetIceParameters().renomination);
}
// Test that if CreateOffer is called with the deprecated "offer to receive
// audio/video" constraints, they're processed and result in an offer with
// audio/video sections just as if RTCOfferAnswerOptions had been used.
TEST_P(PeerConnectionInterfaceTest, CreateOfferWithOfferToReceiveConstraints) {
CreatePeerConnection();
RTCOfferAnswerOptions options;
options.offer_to_receive_audio = 1;
options.offer_to_receive_video = 1;
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, &options));
cricket::SessionDescription* desc = offer->description();
const cricket::ContentInfo* audio = cricket::GetFirstAudioContent(desc);
const cricket::ContentInfo* video = cricket::GetFirstVideoContent(desc);
ASSERT_NE(nullptr, audio);
ASSERT_NE(nullptr, video);
EXPECT_FALSE(audio->rejected);
EXPECT_FALSE(video->rejected);
}
// Test that if CreateAnswer is called with the deprecated "offer to receive
// audio/video" constraints, they're processed and can be used to reject an
// offered m= section just as can be done with RTCOfferAnswerOptions;
// Don't run under Unified Plan since this behavior is not supported.
TEST_F(PeerConnectionInterfaceTestPlanB,
CreateAnswerWithOfferToReceiveConstraints) {
CreatePeerConnection();
// First, create an offer with audio/video and apply it as a remote
// description.
RTCOfferAnswerOptions options;
options.offer_to_receive_audio = 1;
options.offer_to_receive_video = 1;
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, &options));
EXPECT_TRUE(DoSetRemoteDescription(std::move(offer)));
// Now create answer that rejects audio/video.
options.offer_to_receive_audio = 0;
options.offer_to_receive_video = 0;
std::unique_ptr<SessionDescriptionInterface> answer;
ASSERT_TRUE(DoCreateAnswer(&answer, &options));
cricket::SessionDescription* desc = answer->description();
const cricket::ContentInfo* audio = cricket::GetFirstAudioContent(desc);
const cricket::ContentInfo* video = cricket::GetFirstVideoContent(desc);
ASSERT_NE(nullptr, audio);
ASSERT_NE(nullptr, video);
EXPECT_TRUE(audio->rejected);
EXPECT_TRUE(video->rejected);
}
// Test that negotiation can succeed with a data channel only, and with the max
// bundle policy. Previously there was a bug that prevented this.
#ifdef HAVE_SCTP
TEST_P(PeerConnectionInterfaceTest, DataChannelOnlyOfferWithMaxBundlePolicy) {
#else
TEST_P(PeerConnectionInterfaceTest,
DISABLED_DataChannelOnlyOfferWithMaxBundlePolicy) {
#endif // HAVE_SCTP
PeerConnectionInterface::RTCConfiguration config;
config.bundle_policy = PeerConnectionInterface::kBundlePolicyMaxBundle;
CreatePeerConnection(config);
// First, create an offer with only a data channel and apply it as a remote
// description.
pc_->CreateDataChannel("test", nullptr);
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
EXPECT_TRUE(DoSetRemoteDescription(std::move(offer)));
// Create and set answer as well.
std::unique_ptr<SessionDescriptionInterface> answer;
ASSERT_TRUE(DoCreateAnswer(&answer, nullptr));
EXPECT_TRUE(DoSetLocalDescription(std::move(answer)));
}
TEST_P(PeerConnectionInterfaceTest, SetBitrateWithoutMinSucceeds) {
CreatePeerConnection();
PeerConnectionInterface::BitrateParameters bitrate;
bitrate.current_bitrate_bps = 100000;
EXPECT_TRUE(pc_->SetBitrate(bitrate).ok());
}
TEST_P(PeerConnectionInterfaceTest, SetBitrateNegativeMinFails) {
CreatePeerConnection();
PeerConnectionInterface::BitrateParameters bitrate;
bitrate.min_bitrate_bps = -1;
EXPECT_FALSE(pc_->SetBitrate(bitrate).ok());
}
TEST_P(PeerConnectionInterfaceTest, SetBitrateCurrentLessThanMinFails) {
CreatePeerConnection();
PeerConnectionInterface::BitrateParameters bitrate;
bitrate.min_bitrate_bps = 5;
bitrate.current_bitrate_bps = 3;
EXPECT_FALSE(pc_->SetBitrate(bitrate).ok());
}
TEST_P(PeerConnectionInterfaceTest, SetBitrateCurrentNegativeFails) {
CreatePeerConnection();
PeerConnectionInterface::BitrateParameters bitrate;
bitrate.current_bitrate_bps = -1;
EXPECT_FALSE(pc_->SetBitrate(bitrate).ok());
}
TEST_P(PeerConnectionInterfaceTest, SetBitrateMaxLessThanCurrentFails) {
CreatePeerConnection();
PeerConnectionInterface::BitrateParameters bitrate;
bitrate.current_bitrate_bps = 10;
bitrate.max_bitrate_bps = 8;
EXPECT_FALSE(pc_->SetBitrate(bitrate).ok());
}
TEST_P(PeerConnectionInterfaceTest, SetBitrateMaxLessThanMinFails) {
CreatePeerConnection();
PeerConnectionInterface::BitrateParameters bitrate;
bitrate.min_bitrate_bps = 10;
bitrate.max_bitrate_bps = 8;
EXPECT_FALSE(pc_->SetBitrate(bitrate).ok());
}
TEST_P(PeerConnectionInterfaceTest, SetBitrateMaxNegativeFails) {
CreatePeerConnection();
PeerConnectionInterface::BitrateParameters bitrate;
bitrate.max_bitrate_bps = -1;
EXPECT_FALSE(pc_->SetBitrate(bitrate).ok());
}
// ice_regather_interval_range requires WebRTC to be configured for continual
// gathering already.
TEST_P(PeerConnectionInterfaceTest,
SetIceRegatherIntervalRangeWithoutContinualGatheringFails) {
PeerConnectionInterface::RTCConfiguration config;
config.ice_regather_interval_range.emplace(1000, 2000);
config.continual_gathering_policy =
PeerConnectionInterface::ContinualGatheringPolicy::GATHER_ONCE;
CreatePeerConnectionExpectFail(config);
}
// Ensures that there is no error when ice_regather_interval_range is set with
// continual gathering enabled.
TEST_P(PeerConnectionInterfaceTest,
SetIceRegatherIntervalRangeWithContinualGathering) {
PeerConnectionInterface::RTCConfiguration config;
config.ice_regather_interval_range.emplace(1000, 2000);
config.continual_gathering_policy =
PeerConnectionInterface::ContinualGatheringPolicy::GATHER_CONTINUALLY;
CreatePeerConnection(config);
}
// The current bitrate from BitrateSettings is currently clamped
// by Call's BitrateConstraints, which comes from the SDP or a default value.
// This test checks that a call to SetBitrate with a current bitrate that will
// be clamped succeeds.
TEST_P(PeerConnectionInterfaceTest, SetBitrateCurrentLessThanImplicitMin) {
CreatePeerConnection();
PeerConnectionInterface::BitrateParameters bitrate;
bitrate.current_bitrate_bps = 1;
EXPECT_TRUE(pc_->SetBitrate(bitrate).ok());
}
// The following tests verify that the offer can be created correctly.
TEST_P(PeerConnectionInterfaceTest,
CreateOfferFailsWithInvalidOfferToReceiveAudio) {
RTCOfferAnswerOptions rtc_options;
// Setting offer_to_receive_audio to a value lower than kUndefined or greater
// than kMaxOfferToReceiveMedia should be treated as invalid.
rtc_options.offer_to_receive_audio = RTCOfferAnswerOptions::kUndefined - 1;
CreatePeerConnection();
EXPECT_FALSE(CreateOfferWithOptions(rtc_options));
rtc_options.offer_to_receive_audio =
RTCOfferAnswerOptions::kMaxOfferToReceiveMedia + 1;
EXPECT_FALSE(CreateOfferWithOptions(rtc_options));
}
TEST_P(PeerConnectionInterfaceTest,
CreateOfferFailsWithInvalidOfferToReceiveVideo) {
RTCOfferAnswerOptions rtc_options;
// Setting offer_to_receive_video to a value lower than kUndefined or greater
// than kMaxOfferToReceiveMedia should be treated as invalid.
rtc_options.offer_to_receive_video = RTCOfferAnswerOptions::kUndefined - 1;
CreatePeerConnection();
EXPECT_FALSE(CreateOfferWithOptions(rtc_options));
rtc_options.offer_to_receive_video =
RTCOfferAnswerOptions::kMaxOfferToReceiveMedia + 1;
EXPECT_FALSE(CreateOfferWithOptions(rtc_options));
}
// Test that the audio and video content will be added to an offer if both
// |offer_to_receive_audio| and |offer_to_receive_video| options are 1.
TEST_P(PeerConnectionInterfaceTest, CreateOfferWithAudioVideoOptions) {
RTCOfferAnswerOptions rtc_options;
rtc_options.offer_to_receive_audio = 1;
rtc_options.offer_to_receive_video = 1;
std::unique_ptr<SessionDescriptionInterface> offer;
CreatePeerConnection();
offer = CreateOfferWithOptions(rtc_options);
ASSERT_TRUE(offer);
EXPECT_NE(nullptr, GetFirstAudioContent(offer->description()));
EXPECT_NE(nullptr, GetFirstVideoContent(offer->description()));
}
// Test that only audio content will be added to the offer if only
// |offer_to_receive_audio| options is 1.
TEST_P(PeerConnectionInterfaceTest, CreateOfferWithAudioOnlyOptions) {
RTCOfferAnswerOptions rtc_options;
rtc_options.offer_to_receive_audio = 1;
rtc_options.offer_to_receive_video = 0;
std::unique_ptr<SessionDescriptionInterface> offer;
CreatePeerConnection();
offer = CreateOfferWithOptions(rtc_options);
ASSERT_TRUE(offer);
EXPECT_NE(nullptr, GetFirstAudioContent(offer->description()));
EXPECT_EQ(nullptr, GetFirstVideoContent(offer->description()));
}
// Test that only video content will be added if only |offer_to_receive_video|
// options is 1.
TEST_P(PeerConnectionInterfaceTest, CreateOfferWithVideoOnlyOptions) {
RTCOfferAnswerOptions rtc_options;
rtc_options.offer_to_receive_audio = 0;
rtc_options.offer_to_receive_video = 1;
std::unique_ptr<SessionDescriptionInterface> offer;
CreatePeerConnection();
offer = CreateOfferWithOptions(rtc_options);
ASSERT_TRUE(offer);
EXPECT_EQ(nullptr, GetFirstAudioContent(offer->description()));
EXPECT_NE(nullptr, GetFirstVideoContent(offer->description()));
}
// Test that no media content will be added to the offer if using default
// RTCOfferAnswerOptions.
TEST_P(PeerConnectionInterfaceTest, CreateOfferWithDefaultOfferAnswerOptions) {
RTCOfferAnswerOptions rtc_options;
std::unique_ptr<SessionDescriptionInterface> offer;
CreatePeerConnection();
offer = CreateOfferWithOptions(rtc_options);
ASSERT_TRUE(offer);
EXPECT_EQ(nullptr, GetFirstAudioContent(offer->description()));
EXPECT_EQ(nullptr, GetFirstVideoContent(offer->description()));
}
// Test that if |ice_restart| is true, the ufrag/pwd will change, otherwise
// ufrag/pwd will be the same in the new offer.
TEST_P(PeerConnectionInterfaceTest, CreateOfferWithIceRestart) {
CreatePeerConnection();
RTCOfferAnswerOptions rtc_options;
rtc_options.ice_restart = false;
rtc_options.offer_to_receive_audio = 1;
std::unique_ptr<SessionDescriptionInterface> offer;
CreateOfferWithOptionsAsLocalDescription(&offer, rtc_options);
std::string mid = cricket::GetFirstAudioContent(offer->description())->name;
auto ufrag1 =
offer->description()->GetTransportInfoByName(mid)->description.ice_ufrag;
auto pwd1 =
offer->description()->GetTransportInfoByName(mid)->description.ice_pwd;
// |ice_restart| is false, the ufrag/pwd shouldn't change.
CreateOfferWithOptionsAsLocalDescription(&offer, rtc_options);
auto ufrag2 =
offer->description()->GetTransportInfoByName(mid)->description.ice_ufrag;
auto pwd2 =
offer->description()->GetTransportInfoByName(mid)->description.ice_pwd;
// |ice_restart| is true, the ufrag/pwd should change.
rtc_options.ice_restart = true;
CreateOfferWithOptionsAsLocalDescription(&offer, rtc_options);
auto ufrag3 =
offer->description()->GetTransportInfoByName(mid)->description.ice_ufrag;
auto pwd3 =
offer->description()->GetTransportInfoByName(mid)->description.ice_pwd;
EXPECT_EQ(ufrag1, ufrag2);
EXPECT_EQ(pwd1, pwd2);
EXPECT_NE(ufrag2, ufrag3);
EXPECT_NE(pwd2, pwd3);
}
// Test that if |use_rtp_mux| is true, the bundling will be enabled in the
// offer; if it is false, there won't be any bundle group in the offer.
TEST_P(PeerConnectionInterfaceTest, CreateOfferWithRtpMux) {
RTCOfferAnswerOptions rtc_options;
rtc_options.offer_to_receive_audio = 1;
rtc_options.offer_to_receive_video = 1;
std::unique_ptr<SessionDescriptionInterface> offer;
CreatePeerConnection();
rtc_options.use_rtp_mux = true;
offer = CreateOfferWithOptions(rtc_options);
ASSERT_TRUE(offer);
EXPECT_NE(nullptr, GetFirstAudioContent(offer->description()));
EXPECT_NE(nullptr, GetFirstVideoContent(offer->description()));
EXPECT_TRUE(offer->description()->HasGroup(cricket::GROUP_TYPE_BUNDLE));
rtc_options.use_rtp_mux = false;
offer = CreateOfferWithOptions(rtc_options);
ASSERT_TRUE(offer);
EXPECT_NE(nullptr, GetFirstAudioContent(offer->description()));
EXPECT_NE(nullptr, GetFirstVideoContent(offer->description()));
EXPECT_FALSE(offer->description()->HasGroup(cricket::GROUP_TYPE_BUNDLE));
}
// This test ensures OnRenegotiationNeeded is called when we add track with
// MediaStream -> AddTrack in the same way it is called when we add track with
// PeerConnection -> AddTrack.
// The test can be removed once addStream is rewritten in terms of addTrack
// https://bugs.chromium.org/p/webrtc/issues/detail?id=7815
// Don't run under Unified Plan since the stream API is not available.
TEST_F(PeerConnectionInterfaceTestPlanB,
MediaStreamAddTrackRemoveTrackRenegotiate) {
CreatePeerConnectionWithoutDtls();
rtc::scoped_refptr<MediaStreamInterface> stream(
pc_factory_->CreateLocalMediaStream(kStreamId1));
pc_->AddStream(stream);
rtc::scoped_refptr<AudioTrackInterface> audio_track(
CreateAudioTrack("audio_track"));
rtc::scoped_refptr<VideoTrackInterface> video_track(
CreateVideoTrack("video_track"));
stream->AddTrack(audio_track);
EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
observer_.renegotiation_needed_ = false;
stream->AddTrack(video_track);
EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
observer_.renegotiation_needed_ = false;
stream->RemoveTrack(audio_track);
EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
observer_.renegotiation_needed_ = false;
stream->RemoveTrack(video_track);
EXPECT_TRUE_WAIT(observer_.renegotiation_needed_, kTimeout);
observer_.renegotiation_needed_ = false;
}
// Tests that an error is returned if a description is applied that has fewer
// media sections than the existing description.
TEST_P(PeerConnectionInterfaceTest,
MediaSectionCountEnforcedForSubsequentOffer) {
CreatePeerConnection();
AddAudioTrack("audio_label");
AddVideoTrack("video_label");
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
EXPECT_TRUE(DoSetRemoteDescription(std::move(offer)));
// A remote offer with fewer media sections should be rejected.
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
offer->description()->contents().pop_back();
offer->description()->contents().pop_back();
ASSERT_TRUE(offer->description()->contents().empty());
EXPECT_FALSE(DoSetRemoteDescription(std::move(offer)));
std::unique_ptr<SessionDescriptionInterface> answer;
ASSERT_TRUE(DoCreateAnswer(&answer, nullptr));
EXPECT_TRUE(DoSetLocalDescription(std::move(answer)));
// A subsequent local offer with fewer media sections should be rejected.
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
offer->description()->contents().pop_back();
offer->description()->contents().pop_back();
ASSERT_TRUE(offer->description()->contents().empty());
EXPECT_FALSE(DoSetLocalDescription(std::move(offer)));
}
TEST_P(PeerConnectionInterfaceTest, ExtmapAllowMixedIsConfigurable) {
RTCConfiguration config;
// Default behavior is false.
CreatePeerConnection(config);
std::unique_ptr<SessionDescriptionInterface> offer;
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
EXPECT_FALSE(offer->description()->extmap_allow_mixed());
// Possible to set to true.
config.offer_extmap_allow_mixed = true;
CreatePeerConnection(config);
offer.release();
ASSERT_TRUE(DoCreateOffer(&offer, nullptr));
EXPECT_TRUE(offer->description()->extmap_allow_mixed());
}
INSTANTIATE_TEST_SUITE_P(PeerConnectionInterfaceTest,
PeerConnectionInterfaceTest,
Values(SdpSemantics::kPlanB,
SdpSemantics::kUnifiedPlan));
class PeerConnectionMediaConfigTest : public testing::Test {
protected:
void SetUp() override {
pcf_ = PeerConnectionFactoryForTest::CreatePeerConnectionFactoryForTest();
pcf_->Initialize();
}
const cricket::MediaConfig TestCreatePeerConnection(
const RTCConfiguration& config) {
rtc::scoped_refptr<PeerConnectionInterface> pc(
pcf_->CreatePeerConnection(config, nullptr, nullptr, &observer_));
EXPECT_TRUE(pc.get());
observer_.SetPeerConnectionInterface(pc.get());
return pc->GetConfiguration().media_config;
}
rtc::scoped_refptr<PeerConnectionFactoryForTest> pcf_;
MockPeerConnectionObserver observer_;
};
// This sanity check validates the test infrastructure itself.
TEST_F(PeerConnectionMediaConfigTest, TestCreateAndClose) {
PeerConnectionInterface::RTCConfiguration config;
rtc::scoped_refptr<PeerConnectionInterface> pc(
pcf_->CreatePeerConnection(config, nullptr, nullptr, &observer_));
EXPECT_TRUE(pc.get());
observer_.SetPeerConnectionInterface(pc.get()); // Required.
pc->Close(); // No abort -> ok.
SUCCEED();
}
// This test verifies the default behaviour with no constraints and a
// default RTCConfiguration.
TEST_F(PeerConnectionMediaConfigTest, TestDefaults) {
PeerConnectionInterface::RTCConfiguration config;
const cricket::MediaConfig& media_config = TestCreatePeerConnection(config);
EXPECT_FALSE(media_config.enable_dscp);
EXPECT_TRUE(media_config.video.enable_cpu_adaptation);
EXPECT_TRUE(media_config.video.enable_prerenderer_smoothing);
EXPECT_FALSE(media_config.video.suspend_below_min_bitrate);
EXPECT_FALSE(media_config.video.experiment_cpu_load_estimator);
}
// This test verifies that the enable_prerenderer_smoothing flag is
// propagated from RTCConfiguration to the PeerConnection.
TEST_F(PeerConnectionMediaConfigTest, TestDisablePrerendererSmoothingTrue) {
PeerConnectionInterface::RTCConfiguration config;
config.set_prerenderer_smoothing(false);
const cricket::MediaConfig& media_config = TestCreatePeerConnection(config);
EXPECT_FALSE(media_config.video.enable_prerenderer_smoothing);
}
// This test verifies that the experiment_cpu_load_estimator flag is
// propagated from RTCConfiguration to the PeerConnection.
TEST_F(PeerConnectionMediaConfigTest, TestEnableExperimentCpuLoadEstimator) {
PeerConnectionInterface::RTCConfiguration config;
config.set_experiment_cpu_load_estimator(true);
const cricket::MediaConfig& media_config = TestCreatePeerConnection(config);
EXPECT_TRUE(media_config.video.experiment_cpu_load_estimator);
}
// Tests a few random fields being different.
TEST(RTCConfigurationTest, ComparisonOperators) {
PeerConnectionInterface::RTCConfiguration a;
PeerConnectionInterface::RTCConfiguration b;
EXPECT_EQ(a, b);
PeerConnectionInterface::RTCConfiguration c;
c.servers.push_back(PeerConnectionInterface::IceServer());
EXPECT_NE(a, c);
PeerConnectionInterface::RTCConfiguration d;
d.type = PeerConnectionInterface::kRelay;
EXPECT_NE(a, d);
PeerConnectionInterface::RTCConfiguration e;
e.audio_jitter_buffer_max_packets = 5;
EXPECT_NE(a, e);
PeerConnectionInterface::RTCConfiguration f;
f.ice_connection_receiving_timeout = 1337;
EXPECT_NE(a, f);
PeerConnectionInterface::RTCConfiguration g;
g.disable_ipv6 = true;
EXPECT_NE(a, g);
PeerConnectionInterface::RTCConfiguration h(
PeerConnectionInterface::RTCConfigurationType::kAggressive);
EXPECT_NE(a, h);
}
} // namespace
} // namespace webrtc