mirror of
https://github.com/mollyim/webrtc.git
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84df1c724e
Replace calls to .str() which copies with .Release which moves in cases where that's safe. This CL was generated by this command: git grep -l 'StringBuilder' | xargs perl -i -0 -pe "s/(rtc::StringBuilder (\S+);.*?return )\\g2.str\(\)/\$1\$2.Release\(\)/sg" Bug: webrtc:8982 Change-Id: If4dadbeb039df010aaaa9e58da81c1971a84fe8f Reviewed-on: https://webrtc-review.googlesource.com/100307 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24790}
6249 lines
240 KiB
C++
6249 lines
240 KiB
C++
/*
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* Copyright 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "pc/peerconnection.h"
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#include <algorithm>
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#include <limits>
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#include <queue>
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#include <set>
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#include <utility>
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#include <vector>
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#include "absl/memory/memory.h"
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#include "api/jsepicecandidate.h"
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#include "api/jsepsessiondescription.h"
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#include "api/mediastreamproxy.h"
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#include "api/mediastreamtrackproxy.h"
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#include "call/call.h"
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#include "logging/rtc_event_log/icelogger.h"
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#include "logging/rtc_event_log/output/rtc_event_log_output_file.h"
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#include "logging/rtc_event_log/rtc_event_log.h"
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#include "media/sctp/sctptransport.h"
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#include "pc/audiotrack.h"
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#include "pc/channel.h"
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#include "pc/channelmanager.h"
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#include "pc/dtmfsender.h"
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#include "pc/mediastream.h"
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#include "pc/mediastreamobserver.h"
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#include "pc/remoteaudiosource.h"
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#include "pc/rtpmediautils.h"
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#include "pc/rtpreceiver.h"
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#include "pc/rtpsender.h"
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#include "pc/sctputils.h"
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#include "pc/sdputils.h"
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#include "pc/streamcollection.h"
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#include "pc/videocapturertracksource.h"
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#include "pc/videotrack.h"
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#include "rtc_base/bind.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/numerics/safe_conversions.h"
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#include "rtc_base/stringencode.h"
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#include "rtc_base/strings/string_builder.h"
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#include "rtc_base/stringutils.h"
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#include "rtc_base/trace_event.h"
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#include "system_wrappers/include/clock.h"
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#include "system_wrappers/include/field_trial.h"
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#include "system_wrappers/include/metrics.h"
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using cricket::ContentInfo;
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using cricket::ContentInfos;
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using cricket::MediaContentDescription;
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using cricket::SessionDescription;
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using cricket::MediaProtocolType;
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using cricket::TransportInfo;
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using cricket::LOCAL_PORT_TYPE;
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using cricket::STUN_PORT_TYPE;
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using cricket::RELAY_PORT_TYPE;
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using cricket::PRFLX_PORT_TYPE;
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namespace webrtc {
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// Error messages
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const char kBundleWithoutRtcpMux[] =
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"rtcp-mux must be enabled when BUNDLE "
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"is enabled.";
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const char kInvalidCandidates[] = "Description contains invalid candidates.";
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const char kInvalidSdp[] = "Invalid session description.";
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const char kMlineMismatchInAnswer[] =
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"The order of m-lines in answer doesn't match order in offer. Rejecting "
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"answer.";
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const char kMlineMismatchInSubsequentOffer[] =
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"The order of m-lines in subsequent offer doesn't match order from "
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"previous offer/answer.";
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const char kSdpWithoutDtlsFingerprint[] =
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"Called with SDP without DTLS fingerprint.";
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const char kSdpWithoutSdesCrypto[] = "Called with SDP without SDES crypto.";
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const char kSdpWithoutIceUfragPwd[] =
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"Called with SDP without ice-ufrag and ice-pwd.";
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const char kSessionError[] = "Session error code: ";
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const char kSessionErrorDesc[] = "Session error description: ";
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const char kDtlsSrtpSetupFailureRtp[] =
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"Couldn't set up DTLS-SRTP on RTP channel.";
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const char kDtlsSrtpSetupFailureRtcp[] =
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"Couldn't set up DTLS-SRTP on RTCP channel.";
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namespace {
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static const char kDefaultStreamId[] = "default";
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static const char kDefaultAudioSenderId[] = "defaulta0";
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static const char kDefaultVideoSenderId[] = "defaultv0";
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// The length of RTCP CNAMEs.
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static const int kRtcpCnameLength = 16;
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enum {
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MSG_SET_SESSIONDESCRIPTION_SUCCESS = 0,
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MSG_SET_SESSIONDESCRIPTION_FAILED,
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MSG_CREATE_SESSIONDESCRIPTION_FAILED,
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MSG_GETSTATS,
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MSG_FREE_DATACHANNELS,
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MSG_REPORT_USAGE_PATTERN,
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};
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static const int REPORT_USAGE_PATTERN_DELAY_MS = 60000;
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struct SetSessionDescriptionMsg : public rtc::MessageData {
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explicit SetSessionDescriptionMsg(
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webrtc::SetSessionDescriptionObserver* observer)
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: observer(observer) {}
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rtc::scoped_refptr<webrtc::SetSessionDescriptionObserver> observer;
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RTCError error;
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};
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struct CreateSessionDescriptionMsg : public rtc::MessageData {
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explicit CreateSessionDescriptionMsg(
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webrtc::CreateSessionDescriptionObserver* observer)
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: observer(observer) {}
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rtc::scoped_refptr<webrtc::CreateSessionDescriptionObserver> observer;
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RTCError error;
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};
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struct GetStatsMsg : public rtc::MessageData {
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GetStatsMsg(webrtc::StatsObserver* observer,
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webrtc::MediaStreamTrackInterface* track)
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: observer(observer), track(track) {}
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rtc::scoped_refptr<webrtc::StatsObserver> observer;
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rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track;
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};
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// Check if we can send |new_stream| on a PeerConnection.
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bool CanAddLocalMediaStream(webrtc::StreamCollectionInterface* current_streams,
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webrtc::MediaStreamInterface* new_stream) {
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if (!new_stream || !current_streams) {
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return false;
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}
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if (current_streams->find(new_stream->id()) != nullptr) {
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RTC_LOG(LS_ERROR) << "MediaStream with ID " << new_stream->id()
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<< " is already added.";
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return false;
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}
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return true;
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}
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// If the direction is "recvonly" or "inactive", treat the description
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// as containing no streams.
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// See: https://code.google.com/p/webrtc/issues/detail?id=5054
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std::vector<cricket::StreamParams> GetActiveStreams(
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const cricket::MediaContentDescription* desc) {
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return RtpTransceiverDirectionHasSend(desc->direction())
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? desc->streams()
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: std::vector<cricket::StreamParams>();
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}
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bool IsValidOfferToReceiveMedia(int value) {
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typedef PeerConnectionInterface::RTCOfferAnswerOptions Options;
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return (value >= Options::kUndefined) &&
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(value <= Options::kMaxOfferToReceiveMedia);
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}
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// Add options to |[audio/video]_media_description_options| from |senders|.
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void AddRtpSenderOptions(
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const std::vector<rtc::scoped_refptr<
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RtpSenderProxyWithInternal<RtpSenderInternal>>>& senders,
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cricket::MediaDescriptionOptions* audio_media_description_options,
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cricket::MediaDescriptionOptions* video_media_description_options,
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int num_sim_layers) {
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for (const auto& sender : senders) {
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if (sender->media_type() == cricket::MEDIA_TYPE_AUDIO) {
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if (audio_media_description_options) {
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audio_media_description_options->AddAudioSender(
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sender->id(), sender->internal()->stream_ids());
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}
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} else {
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RTC_DCHECK(sender->media_type() == cricket::MEDIA_TYPE_VIDEO);
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if (video_media_description_options) {
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video_media_description_options->AddVideoSender(
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sender->id(), sender->internal()->stream_ids(),
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num_sim_layers);
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}
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}
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}
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}
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// Add options to |session_options| from |rtp_data_channels|.
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void AddRtpDataChannelOptions(
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const std::map<std::string, rtc::scoped_refptr<DataChannel>>&
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rtp_data_channels,
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cricket::MediaDescriptionOptions* data_media_description_options) {
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if (!data_media_description_options) {
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return;
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}
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// Check for data channels.
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for (const auto& kv : rtp_data_channels) {
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const DataChannel* channel = kv.second;
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if (channel->state() == DataChannel::kConnecting ||
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channel->state() == DataChannel::kOpen) {
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// Legacy RTP data channels are signaled with the track/stream ID set to
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// the data channel's label.
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data_media_description_options->AddRtpDataChannel(channel->label(),
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channel->label());
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}
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}
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}
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uint32_t ConvertIceTransportTypeToCandidateFilter(
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PeerConnectionInterface::IceTransportsType type) {
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switch (type) {
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case PeerConnectionInterface::kNone:
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return cricket::CF_NONE;
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case PeerConnectionInterface::kRelay:
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return cricket::CF_RELAY;
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case PeerConnectionInterface::kNoHost:
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return (cricket::CF_ALL & ~cricket::CF_HOST);
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case PeerConnectionInterface::kAll:
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return cricket::CF_ALL;
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default:
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RTC_NOTREACHED();
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}
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return cricket::CF_NONE;
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}
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// Helper to set an error and return from a method.
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bool SafeSetError(webrtc::RTCErrorType type, webrtc::RTCError* error) {
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if (error) {
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error->set_type(type);
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}
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return type == webrtc::RTCErrorType::NONE;
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}
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bool SafeSetError(webrtc::RTCError error, webrtc::RTCError* error_out) {
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bool ok = error.ok();
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if (error_out) {
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*error_out = std::move(error);
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}
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return ok;
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}
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std::string GetSignalingStateString(
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PeerConnectionInterface::SignalingState state) {
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switch (state) {
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case PeerConnectionInterface::kStable:
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return "kStable";
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case PeerConnectionInterface::kHaveLocalOffer:
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return "kHaveLocalOffer";
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case PeerConnectionInterface::kHaveLocalPrAnswer:
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return "kHavePrAnswer";
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case PeerConnectionInterface::kHaveRemoteOffer:
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return "kHaveRemoteOffer";
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case PeerConnectionInterface::kHaveRemotePrAnswer:
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return "kHaveRemotePrAnswer";
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case PeerConnectionInterface::kClosed:
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return "kClosed";
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}
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RTC_NOTREACHED();
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return "";
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}
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IceCandidatePairType GetIceCandidatePairCounter(
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const cricket::Candidate& local,
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const cricket::Candidate& remote) {
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const auto& l = local.type();
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const auto& r = remote.type();
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const auto& host = LOCAL_PORT_TYPE;
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const auto& srflx = STUN_PORT_TYPE;
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const auto& relay = RELAY_PORT_TYPE;
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const auto& prflx = PRFLX_PORT_TYPE;
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if (l == host && r == host) {
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bool local_private = IPIsPrivate(local.address().ipaddr());
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bool remote_private = IPIsPrivate(remote.address().ipaddr());
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if (local_private) {
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if (remote_private) {
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return kIceCandidatePairHostPrivateHostPrivate;
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} else {
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return kIceCandidatePairHostPrivateHostPublic;
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}
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} else {
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if (remote_private) {
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return kIceCandidatePairHostPublicHostPrivate;
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} else {
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return kIceCandidatePairHostPublicHostPublic;
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}
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}
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}
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if (l == host && r == srflx)
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return kIceCandidatePairHostSrflx;
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if (l == host && r == relay)
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return kIceCandidatePairHostRelay;
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if (l == host && r == prflx)
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return kIceCandidatePairHostPrflx;
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if (l == srflx && r == host)
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return kIceCandidatePairSrflxHost;
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if (l == srflx && r == srflx)
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return kIceCandidatePairSrflxSrflx;
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if (l == srflx && r == relay)
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return kIceCandidatePairSrflxRelay;
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if (l == srflx && r == prflx)
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return kIceCandidatePairSrflxPrflx;
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if (l == relay && r == host)
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return kIceCandidatePairRelayHost;
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if (l == relay && r == srflx)
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return kIceCandidatePairRelaySrflx;
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if (l == relay && r == relay)
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return kIceCandidatePairRelayRelay;
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if (l == relay && r == prflx)
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return kIceCandidatePairRelayPrflx;
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if (l == prflx && r == host)
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return kIceCandidatePairPrflxHost;
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if (l == prflx && r == srflx)
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return kIceCandidatePairPrflxSrflx;
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if (l == prflx && r == relay)
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return kIceCandidatePairPrflxRelay;
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return kIceCandidatePairMax;
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}
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|
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// Logic to decide if an m= section can be recycled. This means that the new
|
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// m= section is not rejected, but the old local or remote m= section is
|
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// rejected. |old_content_one| and |old_content_two| refer to the m= section
|
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// of the old remote and old local descriptions in no particular order.
|
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// We need to check both the old local and remote because either
|
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// could be the most current from the latest negotation.
|
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bool IsMediaSectionBeingRecycled(SdpType type,
|
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const ContentInfo& content,
|
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const ContentInfo* old_content_one,
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const ContentInfo* old_content_two) {
|
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return type == SdpType::kOffer && !content.rejected &&
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((old_content_one && old_content_one->rejected) ||
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(old_content_two && old_content_two->rejected));
|
||
}
|
||
|
||
// Verify that the order of media sections in |new_desc| matches
|
||
// |current_desc|. The number of m= sections in |new_desc| should be no
|
||
// less than |current_desc|. In the case of checking an answer's
|
||
// |new_desc|, the |current_desc| is the last offer that was set as the
|
||
// local or remote. In the case of checking an offer's |new_desc| we
|
||
// check against the local and remote descriptions stored from the last
|
||
// negotiation, because either of these could be the most up to date for
|
||
// possible rejected m sections. These are the |current_desc| and
|
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// |secondary_current_desc|.
|
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bool MediaSectionsInSameOrder(const SessionDescription& current_desc,
|
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const SessionDescription* secondary_current_desc,
|
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const SessionDescription& new_desc,
|
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const SdpType type) {
|
||
if (current_desc.contents().size() > new_desc.contents().size()) {
|
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return false;
|
||
}
|
||
|
||
for (size_t i = 0; i < current_desc.contents().size(); ++i) {
|
||
const cricket::ContentInfo* secondary_content_info = nullptr;
|
||
if (secondary_current_desc &&
|
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i < secondary_current_desc->contents().size()) {
|
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secondary_content_info = &secondary_current_desc->contents()[i];
|
||
}
|
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if (IsMediaSectionBeingRecycled(type, new_desc.contents()[i],
|
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¤t_desc.contents()[i],
|
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secondary_content_info)) {
|
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// For new offer descriptions, if the media section can be recycled, it's
|
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// valid for the MID and media type to change.
|
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continue;
|
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}
|
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if (new_desc.contents()[i].name != current_desc.contents()[i].name) {
|
||
return false;
|
||
}
|
||
const MediaContentDescription* new_desc_mdesc =
|
||
new_desc.contents()[i].media_description();
|
||
const MediaContentDescription* current_desc_mdesc =
|
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current_desc.contents()[i].media_description();
|
||
if (new_desc_mdesc->type() != current_desc_mdesc->type()) {
|
||
return false;
|
||
}
|
||
}
|
||
return true;
|
||
}
|
||
|
||
bool MediaSectionsHaveSameCount(const SessionDescription& desc1,
|
||
const SessionDescription& desc2) {
|
||
return desc1.contents().size() == desc2.contents().size();
|
||
}
|
||
|
||
void NoteKeyProtocolAndMedia(KeyExchangeProtocolType protocol_type,
|
||
cricket::MediaType media_type) {
|
||
// Array of structs needed to map {KeyExchangeProtocolType,
|
||
// cricket::MediaType} to KeyExchangeProtocolMedia without using std::map in
|
||
// order to avoid -Wglobal-constructors and -Wexit-time-destructors.
|
||
static constexpr struct {
|
||
KeyExchangeProtocolType protocol_type;
|
||
cricket::MediaType media_type;
|
||
KeyExchangeProtocolMedia protocol_media;
|
||
} kEnumCounterKeyProtocolMediaMap[] = {
|
||
{kEnumCounterKeyProtocolDtls, cricket::MEDIA_TYPE_AUDIO,
|
||
kEnumCounterKeyProtocolMediaTypeDtlsAudio},
|
||
{kEnumCounterKeyProtocolDtls, cricket::MEDIA_TYPE_VIDEO,
|
||
kEnumCounterKeyProtocolMediaTypeDtlsVideo},
|
||
{kEnumCounterKeyProtocolDtls, cricket::MEDIA_TYPE_DATA,
|
||
kEnumCounterKeyProtocolMediaTypeDtlsData},
|
||
{kEnumCounterKeyProtocolSdes, cricket::MEDIA_TYPE_AUDIO,
|
||
kEnumCounterKeyProtocolMediaTypeSdesAudio},
|
||
{kEnumCounterKeyProtocolSdes, cricket::MEDIA_TYPE_VIDEO,
|
||
kEnumCounterKeyProtocolMediaTypeSdesVideo},
|
||
{kEnumCounterKeyProtocolSdes, cricket::MEDIA_TYPE_DATA,
|
||
kEnumCounterKeyProtocolMediaTypeSdesData},
|
||
};
|
||
|
||
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.KeyProtocol", protocol_type,
|
||
kEnumCounterKeyProtocolMax);
|
||
|
||
for (const auto& i : kEnumCounterKeyProtocolMediaMap) {
|
||
if (i.protocol_type == protocol_type && i.media_type == media_type) {
|
||
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.KeyProtocolByMedia",
|
||
i.protocol_media,
|
||
kEnumCounterKeyProtocolMediaTypeMax);
|
||
}
|
||
}
|
||
}
|
||
|
||
void NoteAddIceCandidateResult(int result) {
|
||
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.AddIceCandidate", result,
|
||
kAddIceCandidateMax);
|
||
}
|
||
|
||
// Checks that each non-rejected content has SDES crypto keys or a DTLS
|
||
// fingerprint, unless it's in a BUNDLE group, in which case only the
|
||
// BUNDLE-tag section (first media section/description in the BUNDLE group)
|
||
// needs a ufrag and pwd. Mismatches, such as replying with a DTLS fingerprint
|
||
// to SDES keys, will be caught in JsepTransport negotiation, and backstopped
|
||
// by Channel's |srtp_required| check.
|
||
RTCError VerifyCrypto(const SessionDescription* desc, bool dtls_enabled) {
|
||
const cricket::ContentGroup* bundle =
|
||
desc->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
|
||
for (const cricket::ContentInfo& content_info : desc->contents()) {
|
||
if (content_info.rejected) {
|
||
continue;
|
||
}
|
||
// Note what media is used with each crypto protocol, for all sections.
|
||
NoteKeyProtocolAndMedia(dtls_enabled ? webrtc::kEnumCounterKeyProtocolDtls
|
||
: webrtc::kEnumCounterKeyProtocolSdes,
|
||
content_info.media_description()->type());
|
||
const std::string& mid = content_info.name;
|
||
if (bundle && bundle->HasContentName(mid) &&
|
||
mid != *(bundle->FirstContentName())) {
|
||
// This isn't the first media section in the BUNDLE group, so it's not
|
||
// required to have crypto attributes, since only the crypto attributes
|
||
// from the first section actually get used.
|
||
continue;
|
||
}
|
||
|
||
// If the content isn't rejected or bundled into another m= section, crypto
|
||
// must be present.
|
||
const MediaContentDescription* media = content_info.media_description();
|
||
const TransportInfo* tinfo = desc->GetTransportInfoByName(mid);
|
||
if (!media || !tinfo) {
|
||
// Something is not right.
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, kInvalidSdp);
|
||
}
|
||
if (dtls_enabled) {
|
||
if (!tinfo->description.identity_fingerprint) {
|
||
RTC_LOG(LS_WARNING)
|
||
<< "Session description must have DTLS fingerprint if "
|
||
"DTLS enabled.";
|
||
return RTCError(RTCErrorType::INVALID_PARAMETER,
|
||
kSdpWithoutDtlsFingerprint);
|
||
}
|
||
} else {
|
||
if (media->cryptos().empty()) {
|
||
RTC_LOG(LS_WARNING)
|
||
<< "Session description must have SDES when DTLS disabled.";
|
||
return RTCError(RTCErrorType::INVALID_PARAMETER, kSdpWithoutSdesCrypto);
|
||
}
|
||
}
|
||
}
|
||
return RTCError::OK();
|
||
}
|
||
|
||
// Checks that each non-rejected content has ice-ufrag and ice-pwd set, unless
|
||
// it's in a BUNDLE group, in which case only the BUNDLE-tag section (first
|
||
// media section/description in the BUNDLE group) needs a ufrag and pwd.
|
||
bool VerifyIceUfragPwdPresent(const SessionDescription* desc) {
|
||
const cricket::ContentGroup* bundle =
|
||
desc->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
|
||
for (const cricket::ContentInfo& content_info : desc->contents()) {
|
||
if (content_info.rejected) {
|
||
continue;
|
||
}
|
||
const std::string& mid = content_info.name;
|
||
if (bundle && bundle->HasContentName(mid) &&
|
||
mid != *(bundle->FirstContentName())) {
|
||
// This isn't the first media section in the BUNDLE group, so it's not
|
||
// required to have ufrag/password, since only the ufrag/password from
|
||
// the first section actually get used.
|
||
continue;
|
||
}
|
||
|
||
// If the content isn't rejected or bundled into another m= section,
|
||
// ice-ufrag and ice-pwd must be present.
|
||
const TransportInfo* tinfo = desc->GetTransportInfoByName(mid);
|
||
if (!tinfo) {
|
||
// Something is not right.
|
||
RTC_LOG(LS_ERROR) << kInvalidSdp;
|
||
return false;
|
||
}
|
||
if (tinfo->description.ice_ufrag.empty() ||
|
||
tinfo->description.ice_pwd.empty()) {
|
||
RTC_LOG(LS_ERROR) << "Session description must have ice ufrag and pwd.";
|
||
return false;
|
||
}
|
||
}
|
||
return true;
|
||
}
|
||
|
||
bool GetTrackIdBySsrc(const SessionDescription* session_description,
|
||
uint32_t ssrc,
|
||
std::string* track_id) {
|
||
RTC_DCHECK(track_id != NULL);
|
||
|
||
const cricket::AudioContentDescription* audio_desc =
|
||
cricket::GetFirstAudioContentDescription(session_description);
|
||
if (audio_desc) {
|
||
const auto* found = cricket::GetStreamBySsrc(audio_desc->streams(), ssrc);
|
||
if (found) {
|
||
*track_id = found->id;
|
||
return true;
|
||
}
|
||
}
|
||
|
||
const cricket::VideoContentDescription* video_desc =
|
||
cricket::GetFirstVideoContentDescription(session_description);
|
||
if (video_desc) {
|
||
const auto* found = cricket::GetStreamBySsrc(video_desc->streams(), ssrc);
|
||
if (found) {
|
||
*track_id = found->id;
|
||
return true;
|
||
}
|
||
}
|
||
return false;
|
||
}
|
||
|
||
// Get the SCTP port out of a SessionDescription.
|
||
// Return -1 if not found.
|
||
int GetSctpPort(const SessionDescription* session_description) {
|
||
const cricket::DataContentDescription* data_desc =
|
||
GetFirstDataContentDescription(session_description);
|
||
RTC_DCHECK(data_desc);
|
||
if (!data_desc) {
|
||
return -1;
|
||
}
|
||
std::string value;
|
||
cricket::DataCodec match_pattern(cricket::kGoogleSctpDataCodecPlType,
|
||
cricket::kGoogleSctpDataCodecName);
|
||
for (const cricket::DataCodec& codec : data_desc->codecs()) {
|
||
if (!codec.Matches(match_pattern)) {
|
||
continue;
|
||
}
|
||
if (codec.GetParam(cricket::kCodecParamPort, &value)) {
|
||
return rtc::FromString<int>(value);
|
||
}
|
||
}
|
||
return -1;
|
||
}
|
||
|
||
// Returns true if |new_desc| requests an ICE restart (i.e., new ufrag/pwd).
|
||
bool CheckForRemoteIceRestart(const SessionDescriptionInterface* old_desc,
|
||
const SessionDescriptionInterface* new_desc,
|
||
const std::string& content_name) {
|
||
if (!old_desc) {
|
||
return false;
|
||
}
|
||
const SessionDescription* new_sd = new_desc->description();
|
||
const SessionDescription* old_sd = old_desc->description();
|
||
const ContentInfo* cinfo = new_sd->GetContentByName(content_name);
|
||
if (!cinfo || cinfo->rejected) {
|
||
return false;
|
||
}
|
||
// If the content isn't rejected, check if ufrag and password has changed.
|
||
const cricket::TransportDescription* new_transport_desc =
|
||
new_sd->GetTransportDescriptionByName(content_name);
|
||
const cricket::TransportDescription* old_transport_desc =
|
||
old_sd->GetTransportDescriptionByName(content_name);
|
||
if (!new_transport_desc || !old_transport_desc) {
|
||
// No transport description exists. This is not an ICE restart.
|
||
return false;
|
||
}
|
||
if (cricket::IceCredentialsChanged(
|
||
old_transport_desc->ice_ufrag, old_transport_desc->ice_pwd,
|
||
new_transport_desc->ice_ufrag, new_transport_desc->ice_pwd)) {
|
||
RTC_LOG(LS_INFO) << "Remote peer requests ICE restart for " << content_name
|
||
<< ".";
|
||
return true;
|
||
}
|
||
return false;
|
||
}
|
||
|
||
// Generates a string error message for SetLocalDescription/SetRemoteDescription
|
||
// from an RTCError.
|
||
std::string GetSetDescriptionErrorMessage(cricket::ContentSource source,
|
||
SdpType type,
|
||
const RTCError& error) {
|
||
rtc::StringBuilder oss;
|
||
oss << "Failed to set " << (source == cricket::CS_LOCAL ? "local" : "remote")
|
||
<< " " << SdpTypeToString(type) << " sdp: " << error.message();
|
||
return oss.Release();
|
||
}
|
||
|
||
std::string GetStreamIdsString(rtc::ArrayView<const std::string> stream_ids) {
|
||
std::string output = "streams=[";
|
||
const char* separator = "";
|
||
for (const auto& stream_id : stream_ids) {
|
||
output.append(separator).append(stream_id);
|
||
separator = ", ";
|
||
}
|
||
output.append("]");
|
||
return output;
|
||
}
|
||
|
||
absl::optional<int> RTCConfigurationToIceConfigOptionalInt(
|
||
int rtc_configuration_parameter) {
|
||
if (rtc_configuration_parameter ==
|
||
webrtc::PeerConnectionInterface::RTCConfiguration::kUndefined) {
|
||
return absl::nullopt;
|
||
}
|
||
return rtc_configuration_parameter;
|
||
}
|
||
|
||
} // namespace
|
||
|
||
// Upon completion, posts a task to execute the callback of the
|
||
// SetSessionDescriptionObserver asynchronously on the same thread. At this
|
||
// point, the state of the peer connection might no longer reflect the effects
|
||
// of the SetRemoteDescription operation, as the peer connection could have been
|
||
// modified during the post.
|
||
// TODO(hbos): Remove this class once we remove the version of
|
||
// PeerConnectionInterface::SetRemoteDescription() that takes a
|
||
// SetSessionDescriptionObserver as an argument.
|
||
class PeerConnection::SetRemoteDescriptionObserverAdapter
|
||
: public rtc::RefCountedObject<SetRemoteDescriptionObserverInterface> {
|
||
public:
|
||
SetRemoteDescriptionObserverAdapter(
|
||
rtc::scoped_refptr<PeerConnection> pc,
|
||
rtc::scoped_refptr<SetSessionDescriptionObserver> wrapper)
|
||
: pc_(std::move(pc)), wrapper_(std::move(wrapper)) {}
|
||
|
||
// SetRemoteDescriptionObserverInterface implementation.
|
||
void OnSetRemoteDescriptionComplete(RTCError error) override {
|
||
if (error.ok())
|
||
pc_->PostSetSessionDescriptionSuccess(wrapper_);
|
||
else
|
||
pc_->PostSetSessionDescriptionFailure(wrapper_, std::move(error));
|
||
}
|
||
|
||
private:
|
||
rtc::scoped_refptr<PeerConnection> pc_;
|
||
rtc::scoped_refptr<SetSessionDescriptionObserver> wrapper_;
|
||
};
|
||
|
||
bool PeerConnectionInterface::RTCConfiguration::operator==(
|
||
const PeerConnectionInterface::RTCConfiguration& o) const {
|
||
// This static_assert prevents us from accidentally breaking operator==.
|
||
// Note: Order matters! Fields must be ordered the same as RTCConfiguration.
|
||
struct stuff_being_tested_for_equality {
|
||
IceServers servers;
|
||
IceTransportsType type;
|
||
BundlePolicy bundle_policy;
|
||
RtcpMuxPolicy rtcp_mux_policy;
|
||
std::vector<rtc::scoped_refptr<rtc::RTCCertificate>> certificates;
|
||
int ice_candidate_pool_size;
|
||
bool disable_ipv6;
|
||
bool disable_ipv6_on_wifi;
|
||
int max_ipv6_networks;
|
||
bool disable_link_local_networks;
|
||
bool enable_rtp_data_channel;
|
||
absl::optional<int> screencast_min_bitrate;
|
||
absl::optional<bool> combined_audio_video_bwe;
|
||
absl::optional<bool> enable_dtls_srtp;
|
||
TcpCandidatePolicy tcp_candidate_policy;
|
||
CandidateNetworkPolicy candidate_network_policy;
|
||
int audio_jitter_buffer_max_packets;
|
||
bool audio_jitter_buffer_fast_accelerate;
|
||
int ice_connection_receiving_timeout;
|
||
int ice_backup_candidate_pair_ping_interval;
|
||
ContinualGatheringPolicy continual_gathering_policy;
|
||
bool prioritize_most_likely_ice_candidate_pairs;
|
||
struct cricket::MediaConfig media_config;
|
||
bool prune_turn_ports;
|
||
bool presume_writable_when_fully_relayed;
|
||
bool enable_ice_renomination;
|
||
bool redetermine_role_on_ice_restart;
|
||
absl::optional<int> ice_check_interval_strong_connectivity;
|
||
absl::optional<int> ice_check_interval_weak_connectivity;
|
||
absl::optional<int> ice_check_min_interval;
|
||
absl::optional<int> ice_unwritable_timeout;
|
||
absl::optional<int> ice_unwritable_min_checks;
|
||
absl::optional<int> stun_candidate_keepalive_interval;
|
||
absl::optional<rtc::IntervalRange> ice_regather_interval_range;
|
||
webrtc::TurnCustomizer* turn_customizer;
|
||
SdpSemantics sdp_semantics;
|
||
absl::optional<rtc::AdapterType> network_preference;
|
||
bool active_reset_srtp_params;
|
||
};
|
||
static_assert(sizeof(stuff_being_tested_for_equality) == sizeof(*this),
|
||
"Did you add something to RTCConfiguration and forget to "
|
||
"update operator==?");
|
||
return type == o.type && servers == o.servers &&
|
||
bundle_policy == o.bundle_policy &&
|
||
rtcp_mux_policy == o.rtcp_mux_policy &&
|
||
tcp_candidate_policy == o.tcp_candidate_policy &&
|
||
candidate_network_policy == o.candidate_network_policy &&
|
||
audio_jitter_buffer_max_packets == o.audio_jitter_buffer_max_packets &&
|
||
audio_jitter_buffer_fast_accelerate ==
|
||
o.audio_jitter_buffer_fast_accelerate &&
|
||
ice_connection_receiving_timeout ==
|
||
o.ice_connection_receiving_timeout &&
|
||
ice_backup_candidate_pair_ping_interval ==
|
||
o.ice_backup_candidate_pair_ping_interval &&
|
||
continual_gathering_policy == o.continual_gathering_policy &&
|
||
certificates == o.certificates &&
|
||
prioritize_most_likely_ice_candidate_pairs ==
|
||
o.prioritize_most_likely_ice_candidate_pairs &&
|
||
media_config == o.media_config && disable_ipv6 == o.disable_ipv6 &&
|
||
disable_ipv6_on_wifi == o.disable_ipv6_on_wifi &&
|
||
max_ipv6_networks == o.max_ipv6_networks &&
|
||
disable_link_local_networks == o.disable_link_local_networks &&
|
||
enable_rtp_data_channel == o.enable_rtp_data_channel &&
|
||
screencast_min_bitrate == o.screencast_min_bitrate &&
|
||
combined_audio_video_bwe == o.combined_audio_video_bwe &&
|
||
enable_dtls_srtp == o.enable_dtls_srtp &&
|
||
ice_candidate_pool_size == o.ice_candidate_pool_size &&
|
||
prune_turn_ports == o.prune_turn_ports &&
|
||
presume_writable_when_fully_relayed ==
|
||
o.presume_writable_when_fully_relayed &&
|
||
enable_ice_renomination == o.enable_ice_renomination &&
|
||
redetermine_role_on_ice_restart == o.redetermine_role_on_ice_restart &&
|
||
ice_check_interval_strong_connectivity ==
|
||
o.ice_check_interval_strong_connectivity &&
|
||
ice_check_interval_weak_connectivity ==
|
||
o.ice_check_interval_weak_connectivity &&
|
||
ice_check_min_interval == o.ice_check_min_interval &&
|
||
ice_unwritable_timeout == o.ice_unwritable_timeout &&
|
||
ice_unwritable_min_checks == o.ice_unwritable_min_checks &&
|
||
stun_candidate_keepalive_interval ==
|
||
o.stun_candidate_keepalive_interval &&
|
||
ice_regather_interval_range == o.ice_regather_interval_range &&
|
||
turn_customizer == o.turn_customizer &&
|
||
sdp_semantics == o.sdp_semantics &&
|
||
network_preference == o.network_preference &&
|
||
active_reset_srtp_params == o.active_reset_srtp_params;
|
||
}
|
||
|
||
bool PeerConnectionInterface::RTCConfiguration::operator!=(
|
||
const PeerConnectionInterface::RTCConfiguration& o) const {
|
||
return !(*this == o);
|
||
}
|
||
|
||
// Generate a RTCP CNAME when a PeerConnection is created.
|
||
std::string GenerateRtcpCname() {
|
||
std::string cname;
|
||
if (!rtc::CreateRandomString(kRtcpCnameLength, &cname)) {
|
||
RTC_LOG(LS_ERROR) << "Failed to generate CNAME.";
|
||
RTC_NOTREACHED();
|
||
}
|
||
return cname;
|
||
}
|
||
|
||
bool ValidateOfferAnswerOptions(
|
||
const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options) {
|
||
return IsValidOfferToReceiveMedia(rtc_options.offer_to_receive_audio) &&
|
||
IsValidOfferToReceiveMedia(rtc_options.offer_to_receive_video);
|
||
}
|
||
|
||
// From |rtc_options|, fill parts of |session_options| shared by all generated
|
||
// m= sections (in other words, nothing that involves a map/array).
|
||
void ExtractSharedMediaSessionOptions(
|
||
const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
|
||
cricket::MediaSessionOptions* session_options) {
|
||
session_options->vad_enabled = rtc_options.voice_activity_detection;
|
||
session_options->bundle_enabled = rtc_options.use_rtp_mux;
|
||
}
|
||
|
||
PeerConnection::PeerConnection(PeerConnectionFactory* factory,
|
||
std::unique_ptr<RtcEventLog> event_log,
|
||
std::unique_ptr<Call> call)
|
||
: factory_(factory),
|
||
event_log_(std::move(event_log)),
|
||
rtcp_cname_(GenerateRtcpCname()),
|
||
local_streams_(StreamCollection::Create()),
|
||
remote_streams_(StreamCollection::Create()),
|
||
call_(std::move(call)) {}
|
||
|
||
PeerConnection::~PeerConnection() {
|
||
TRACE_EVENT0("webrtc", "PeerConnection::~PeerConnection");
|
||
RTC_DCHECK_RUN_ON(signaling_thread());
|
||
|
||
// Need to stop transceivers before destroying the stats collector because
|
||
// AudioRtpSender has a reference to the StatsCollector it will update when
|
||
// stopping.
|
||
for (auto transceiver : transceivers_) {
|
||
transceiver->Stop();
|
||
}
|
||
|
||
stats_.reset(nullptr);
|
||
if (stats_collector_) {
|
||
stats_collector_->WaitForPendingRequest();
|
||
stats_collector_ = nullptr;
|
||
}
|
||
|
||
// Don't destroy BaseChannels until after stats has been cleaned up so that
|
||
// the last stats request can still read from the channels.
|
||
DestroyAllChannels();
|
||
|
||
RTC_LOG(LS_INFO) << "Session: " << session_id() << " is destroyed.";
|
||
|
||
webrtc_session_desc_factory_.reset();
|
||
sctp_invoker_.reset();
|
||
sctp_factory_.reset();
|
||
transport_controller_.reset();
|
||
|
||
// port_allocator_ lives on the network thread and should be destroyed there.
|
||
network_thread()->Invoke<void>(RTC_FROM_HERE,
|
||
[this] { port_allocator_.reset(); });
|
||
// call_ and event_log_ must be destroyed on the worker thread.
|
||
worker_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
|
||
call_.reset();
|
||
// The event log must outlive call (and any other object that uses it).
|
||
event_log_.reset();
|
||
});
|
||
}
|
||
|
||
void PeerConnection::DestroyAllChannels() {
|
||
// Destroy video channels first since they may have a pointer to a voice
|
||
// channel.
|
||
for (auto transceiver : transceivers_) {
|
||
if (transceiver->media_type() == cricket::MEDIA_TYPE_VIDEO) {
|
||
DestroyTransceiverChannel(transceiver);
|
||
}
|
||
}
|
||
for (auto transceiver : transceivers_) {
|
||
if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
|
||
DestroyTransceiverChannel(transceiver);
|
||
}
|
||
}
|
||
DestroyDataChannel();
|
||
}
|
||
|
||
bool PeerConnection::Initialize(
|
||
const PeerConnectionInterface::RTCConfiguration& configuration,
|
||
PeerConnectionDependencies dependencies) {
|
||
TRACE_EVENT0("webrtc", "PeerConnection::Initialize");
|
||
|
||
RTCError config_error = ValidateConfiguration(configuration);
|
||
if (!config_error.ok()) {
|
||
RTC_LOG(LS_ERROR) << "Invalid configuration: " << config_error.message();
|
||
return false;
|
||
}
|
||
|
||
if (!dependencies.allocator) {
|
||
RTC_LOG(LS_ERROR)
|
||
<< "PeerConnection initialized without a PortAllocator? "
|
||
"This shouldn't happen if using PeerConnectionFactory.";
|
||
return false;
|
||
}
|
||
|
||
if (!dependencies.observer) {
|
||
// TODO(deadbeef): Why do we do this?
|
||
RTC_LOG(LS_ERROR) << "PeerConnection initialized without a "
|
||
"PeerConnectionObserver";
|
||
return false;
|
||
}
|
||
|
||
observer_ = dependencies.observer;
|
||
async_resolver_factory_ = std::move(dependencies.async_resolver_factory);
|
||
port_allocator_ = std::move(dependencies.allocator);
|
||
tls_cert_verifier_ = std::move(dependencies.tls_cert_verifier);
|
||
|
||
cricket::ServerAddresses stun_servers;
|
||
std::vector<cricket::RelayServerConfig> turn_servers;
|
||
|
||
RTCErrorType parse_error =
|
||
ParseIceServers(configuration.servers, &stun_servers, &turn_servers);
|
||
if (parse_error != RTCErrorType::NONE) {
|
||
return false;
|
||
}
|
||
|
||
// The port allocator lives on the network thread and should be initialized
|
||
// there.
|
||
if (!network_thread()->Invoke<bool>(
|
||
RTC_FROM_HERE,
|
||
rtc::Bind(&PeerConnection::InitializePortAllocator_n, this,
|
||
stun_servers, turn_servers, configuration))) {
|
||
return false;
|
||
}
|
||
// If initialization was successful, note if STUN or TURN servers
|
||
// were supplied.
|
||
if (!stun_servers.empty()) {
|
||
NoteUsageEvent(UsageEvent::STUN_SERVER_ADDED);
|
||
}
|
||
if (!turn_servers.empty()) {
|
||
NoteUsageEvent(UsageEvent::TURN_SERVER_ADDED);
|
||
}
|
||
|
||
// Send information about IPv4/IPv6 status.
|
||
PeerConnectionAddressFamilyCounter address_family;
|
||
if (port_allocator_flags_ & cricket::PORTALLOCATOR_ENABLE_IPV6) {
|
||
address_family = kPeerConnection_IPv6;
|
||
} else {
|
||
address_family = kPeerConnection_IPv4;
|
||
}
|
||
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.IPMetrics", address_family,
|
||
kPeerConnectionAddressFamilyCounter_Max);
|
||
|
||
const PeerConnectionFactoryInterface::Options& options = factory_->options();
|
||
|
||
// RFC 3264: The numeric value of the session id and version in the
|
||
// o line MUST be representable with a "64 bit signed integer".
|
||
// Due to this constraint session id |session_id_| is max limited to
|
||
// LLONG_MAX.
|
||
session_id_ = rtc::ToString(rtc::CreateRandomId64() & LLONG_MAX);
|
||
JsepTransportController::Config config;
|
||
config.redetermine_role_on_ice_restart =
|
||
configuration.redetermine_role_on_ice_restart;
|
||
config.ssl_max_version = factory_->options().ssl_max_version;
|
||
config.disable_encryption = options.disable_encryption;
|
||
config.bundle_policy = configuration.bundle_policy;
|
||
config.rtcp_mux_policy = configuration.rtcp_mux_policy;
|
||
config.crypto_options = options.crypto_options;
|
||
config.transport_observer = this;
|
||
config.event_log = event_log_.get();
|
||
#if defined(ENABLE_EXTERNAL_AUTH)
|
||
config.enable_external_auth = true;
|
||
#endif
|
||
config.active_reset_srtp_params = configuration.active_reset_srtp_params;
|
||
transport_controller_.reset(new JsepTransportController(
|
||
signaling_thread(), network_thread(), port_allocator_.get(),
|
||
async_resolver_factory_.get(), config));
|
||
transport_controller_->SignalIceConnectionState.connect(
|
||
this, &PeerConnection::OnTransportControllerConnectionState);
|
||
transport_controller_->SignalIceGatheringState.connect(
|
||
this, &PeerConnection::OnTransportControllerGatheringState);
|
||
transport_controller_->SignalIceCandidatesGathered.connect(
|
||
this, &PeerConnection::OnTransportControllerCandidatesGathered);
|
||
transport_controller_->SignalIceCandidatesRemoved.connect(
|
||
this, &PeerConnection::OnTransportControllerCandidatesRemoved);
|
||
transport_controller_->SignalDtlsHandshakeError.connect(
|
||
this, &PeerConnection::OnTransportControllerDtlsHandshakeError);
|
||
|
||
sctp_factory_ = factory_->CreateSctpTransportInternalFactory();
|
||
|
||
stats_.reset(new StatsCollector(this));
|
||
stats_collector_ = RTCStatsCollector::Create(this);
|
||
|
||
configuration_ = configuration;
|
||
|
||
// Obtain a certificate from RTCConfiguration if any were provided (optional).
|
||
rtc::scoped_refptr<rtc::RTCCertificate> certificate;
|
||
if (!configuration.certificates.empty()) {
|
||
// TODO(hbos,torbjorng): Decide on certificate-selection strategy instead of
|
||
// just picking the first one. The decision should be made based on the DTLS
|
||
// handshake. The DTLS negotiations need to know about all certificates.
|
||
certificate = configuration.certificates[0];
|
||
}
|
||
|
||
transport_controller_->SetIceConfig(ParseIceConfig(configuration));
|
||
|
||
if (options.disable_encryption) {
|
||
dtls_enabled_ = false;
|
||
} else {
|
||
// Enable DTLS by default if we have an identity store or a certificate.
|
||
dtls_enabled_ = (dependencies.cert_generator || certificate);
|
||
// |configuration| can override the default |dtls_enabled_| value.
|
||
if (configuration.enable_dtls_srtp) {
|
||
dtls_enabled_ = *(configuration.enable_dtls_srtp);
|
||
}
|
||
}
|
||
|
||
// Enable creation of RTP data channels if the kEnableRtpDataChannels is set.
|
||
// It takes precendence over the disable_sctp_data_channels
|
||
// PeerConnectionFactoryInterface::Options.
|
||
if (configuration.enable_rtp_data_channel) {
|
||
data_channel_type_ = cricket::DCT_RTP;
|
||
} else {
|
||
// DTLS has to be enabled to use SCTP.
|
||
if (!options.disable_sctp_data_channels && dtls_enabled_) {
|
||
data_channel_type_ = cricket::DCT_SCTP;
|
||
}
|
||
}
|
||
|
||
video_options_.screencast_min_bitrate_kbps =
|
||
configuration.screencast_min_bitrate;
|
||
audio_options_.combined_audio_video_bwe =
|
||
configuration.combined_audio_video_bwe;
|
||
|
||
audio_options_.audio_jitter_buffer_max_packets =
|
||
configuration.audio_jitter_buffer_max_packets;
|
||
|
||
audio_options_.audio_jitter_buffer_fast_accelerate =
|
||
configuration.audio_jitter_buffer_fast_accelerate;
|
||
|
||
// Whether the certificate generator/certificate is null or not determines
|
||
// what PeerConnectionDescriptionFactory will do, so make sure that we give it
|
||
// the right instructions by clearing the variables if needed.
|
||
if (!dtls_enabled_) {
|
||
dependencies.cert_generator.reset();
|
||
certificate = nullptr;
|
||
} else if (certificate) {
|
||
// Favor generated certificate over the certificate generator.
|
||
dependencies.cert_generator.reset();
|
||
}
|
||
|
||
webrtc_session_desc_factory_.reset(new WebRtcSessionDescriptionFactory(
|
||
signaling_thread(), channel_manager(), this, session_id(),
|
||
std::move(dependencies.cert_generator), certificate));
|
||
webrtc_session_desc_factory_->SignalCertificateReady.connect(
|
||
this, &PeerConnection::OnCertificateReady);
|
||
|
||
if (options.disable_encryption) {
|
||
webrtc_session_desc_factory_->SetSdesPolicy(cricket::SEC_DISABLED);
|
||
}
|
||
|
||
webrtc_session_desc_factory_->set_enable_encrypted_rtp_header_extensions(
|
||
options.crypto_options.enable_encrypted_rtp_header_extensions);
|
||
|
||
// Add default audio/video transceivers for Plan B SDP.
|
||
if (!IsUnifiedPlan()) {
|
||
transceivers_.push_back(
|
||
RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(
|
||
signaling_thread(), new RtpTransceiver(cricket::MEDIA_TYPE_AUDIO)));
|
||
transceivers_.push_back(
|
||
RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(
|
||
signaling_thread(), new RtpTransceiver(cricket::MEDIA_TYPE_VIDEO)));
|
||
}
|
||
int delay_ms =
|
||
return_histogram_very_quickly_ ? 0 : REPORT_USAGE_PATTERN_DELAY_MS;
|
||
signaling_thread()->PostDelayed(RTC_FROM_HERE, delay_ms, this,
|
||
MSG_REPORT_USAGE_PATTERN, nullptr);
|
||
return true;
|
||
}
|
||
|
||
RTCError PeerConnection::ValidateConfiguration(
|
||
const RTCConfiguration& config) const {
|
||
if (config.ice_regather_interval_range &&
|
||
config.continual_gathering_policy == GATHER_ONCE) {
|
||
return RTCError(RTCErrorType::INVALID_PARAMETER,
|
||
"ice_regather_interval_range specified but continual "
|
||
"gathering policy is GATHER_ONCE");
|
||
}
|
||
auto result =
|
||
cricket::P2PTransportChannel::ValidateIceConfig(ParseIceConfig(config));
|
||
return result;
|
||
}
|
||
|
||
rtc::scoped_refptr<StreamCollectionInterface> PeerConnection::local_streams() {
|
||
RTC_CHECK(!IsUnifiedPlan()) << "local_streams is not available with Unified "
|
||
"Plan SdpSemantics. Please use GetSenders "
|
||
"instead.";
|
||
return local_streams_;
|
||
}
|
||
|
||
rtc::scoped_refptr<StreamCollectionInterface> PeerConnection::remote_streams() {
|
||
RTC_CHECK(!IsUnifiedPlan()) << "remote_streams is not available with Unified "
|
||
"Plan SdpSemantics. Please use GetReceivers "
|
||
"instead.";
|
||
return remote_streams_;
|
||
}
|
||
|
||
bool PeerConnection::AddStream(MediaStreamInterface* local_stream) {
|
||
RTC_CHECK(!IsUnifiedPlan()) << "AddStream is not available with Unified Plan "
|
||
"SdpSemantics. Please use AddTrack instead.";
|
||
TRACE_EVENT0("webrtc", "PeerConnection::AddStream");
|
||
if (IsClosed()) {
|
||
return false;
|
||
}
|
||
if (!CanAddLocalMediaStream(local_streams_, local_stream)) {
|
||
return false;
|
||
}
|
||
|
||
local_streams_->AddStream(local_stream);
|
||
MediaStreamObserver* observer = new MediaStreamObserver(local_stream);
|
||
observer->SignalAudioTrackAdded.connect(this,
|
||
&PeerConnection::OnAudioTrackAdded);
|
||
observer->SignalAudioTrackRemoved.connect(
|
||
this, &PeerConnection::OnAudioTrackRemoved);
|
||
observer->SignalVideoTrackAdded.connect(this,
|
||
&PeerConnection::OnVideoTrackAdded);
|
||
observer->SignalVideoTrackRemoved.connect(
|
||
this, &PeerConnection::OnVideoTrackRemoved);
|
||
stream_observers_.push_back(std::unique_ptr<MediaStreamObserver>(observer));
|
||
|
||
for (const auto& track : local_stream->GetAudioTracks()) {
|
||
AddAudioTrack(track.get(), local_stream);
|
||
}
|
||
for (const auto& track : local_stream->GetVideoTracks()) {
|
||
AddVideoTrack(track.get(), local_stream);
|
||
}
|
||
|
||
stats_->AddStream(local_stream);
|
||
Observer()->OnRenegotiationNeeded();
|
||
return true;
|
||
}
|
||
|
||
void PeerConnection::RemoveStream(MediaStreamInterface* local_stream) {
|
||
RTC_CHECK(!IsUnifiedPlan()) << "RemoveStream is not available with Unified "
|
||
"Plan SdpSemantics. Please use RemoveTrack "
|
||
"instead.";
|
||
TRACE_EVENT0("webrtc", "PeerConnection::RemoveStream");
|
||
if (!IsClosed()) {
|
||
for (const auto& track : local_stream->GetAudioTracks()) {
|
||
RemoveAudioTrack(track.get(), local_stream);
|
||
}
|
||
for (const auto& track : local_stream->GetVideoTracks()) {
|
||
RemoveVideoTrack(track.get(), local_stream);
|
||
}
|
||
}
|
||
local_streams_->RemoveStream(local_stream);
|
||
stream_observers_.erase(
|
||
std::remove_if(
|
||
stream_observers_.begin(), stream_observers_.end(),
|
||
[local_stream](const std::unique_ptr<MediaStreamObserver>& observer) {
|
||
return observer->stream()->id().compare(local_stream->id()) == 0;
|
||
}),
|
||
stream_observers_.end());
|
||
|
||
if (IsClosed()) {
|
||
return;
|
||
}
|
||
Observer()->OnRenegotiationNeeded();
|
||
}
|
||
|
||
RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::AddTrack(
|
||
rtc::scoped_refptr<MediaStreamTrackInterface> track,
|
||
const std::vector<std::string>& stream_ids) {
|
||
TRACE_EVENT0("webrtc", "PeerConnection::AddTrack");
|
||
if (!track) {
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, "Track is null.");
|
||
}
|
||
if (!(track->kind() == MediaStreamTrackInterface::kAudioKind ||
|
||
track->kind() == MediaStreamTrackInterface::kVideoKind)) {
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
||
"Track has invalid kind: " + track->kind());
|
||
}
|
||
if (IsClosed()) {
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
|
||
"PeerConnection is closed.");
|
||
}
|
||
if (FindSenderForTrack(track)) {
|
||
LOG_AND_RETURN_ERROR(
|
||
RTCErrorType::INVALID_PARAMETER,
|
||
"Sender already exists for track " + track->id() + ".");
|
||
}
|
||
auto sender_or_error =
|
||
(IsUnifiedPlan() ? AddTrackUnifiedPlan(track, stream_ids)
|
||
: AddTrackPlanB(track, stream_ids));
|
||
if (sender_or_error.ok()) {
|
||
Observer()->OnRenegotiationNeeded();
|
||
stats_->AddTrack(track);
|
||
}
|
||
return sender_or_error;
|
||
}
|
||
|
||
RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>>
|
||
PeerConnection::AddTrackPlanB(
|
||
rtc::scoped_refptr<MediaStreamTrackInterface> track,
|
||
const std::vector<std::string>& stream_ids) {
|
||
if (stream_ids.size() > 1u) {
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
|
||
"AddTrack with more than one stream is not "
|
||
"supported with Plan B semantics.");
|
||
}
|
||
std::vector<std::string> adjusted_stream_ids = stream_ids;
|
||
if (adjusted_stream_ids.empty()) {
|
||
adjusted_stream_ids.push_back(rtc::CreateRandomUuid());
|
||
}
|
||
cricket::MediaType media_type =
|
||
(track->kind() == MediaStreamTrackInterface::kAudioKind
|
||
? cricket::MEDIA_TYPE_AUDIO
|
||
: cricket::MEDIA_TYPE_VIDEO);
|
||
auto new_sender =
|
||
CreateSender(media_type, track->id(), track, adjusted_stream_ids);
|
||
if (track->kind() == MediaStreamTrackInterface::kAudioKind) {
|
||
new_sender->internal()->SetVoiceMediaChannel(voice_media_channel());
|
||
GetAudioTransceiver()->internal()->AddSender(new_sender);
|
||
const RtpSenderInfo* sender_info =
|
||
FindSenderInfo(local_audio_sender_infos_,
|
||
new_sender->internal()->stream_ids()[0], track->id());
|
||
if (sender_info) {
|
||
new_sender->internal()->SetSsrc(sender_info->first_ssrc);
|
||
}
|
||
} else {
|
||
RTC_DCHECK_EQ(MediaStreamTrackInterface::kVideoKind, track->kind());
|
||
new_sender->internal()->SetVideoMediaChannel(video_media_channel());
|
||
GetVideoTransceiver()->internal()->AddSender(new_sender);
|
||
const RtpSenderInfo* sender_info =
|
||
FindSenderInfo(local_video_sender_infos_,
|
||
new_sender->internal()->stream_ids()[0], track->id());
|
||
if (sender_info) {
|
||
new_sender->internal()->SetSsrc(sender_info->first_ssrc);
|
||
}
|
||
}
|
||
return rtc::scoped_refptr<RtpSenderInterface>(new_sender);
|
||
}
|
||
|
||
RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>>
|
||
PeerConnection::AddTrackUnifiedPlan(
|
||
rtc::scoped_refptr<MediaStreamTrackInterface> track,
|
||
const std::vector<std::string>& stream_ids) {
|
||
auto transceiver = FindFirstTransceiverForAddedTrack(track);
|
||
if (transceiver) {
|
||
RTC_LOG(LS_INFO) << "Reusing an existing "
|
||
<< cricket::MediaTypeToString(transceiver->media_type())
|
||
<< " transceiver for AddTrack.";
|
||
if (transceiver->direction() == RtpTransceiverDirection::kRecvOnly) {
|
||
transceiver->internal()->set_direction(
|
||
RtpTransceiverDirection::kSendRecv);
|
||
} else if (transceiver->direction() == RtpTransceiverDirection::kInactive) {
|
||
transceiver->internal()->set_direction(
|
||
RtpTransceiverDirection::kSendOnly);
|
||
}
|
||
transceiver->sender()->SetTrack(track);
|
||
transceiver->internal()->sender_internal()->set_stream_ids(stream_ids);
|
||
} else {
|
||
cricket::MediaType media_type =
|
||
(track->kind() == MediaStreamTrackInterface::kAudioKind
|
||
? cricket::MEDIA_TYPE_AUDIO
|
||
: cricket::MEDIA_TYPE_VIDEO);
|
||
RTC_LOG(LS_INFO) << "Adding " << cricket::MediaTypeToString(media_type)
|
||
<< " transceiver in response to a call to AddTrack.";
|
||
std::string sender_id = track->id();
|
||
// Avoid creating a sender with an existing ID by generating a random ID.
|
||
// This can happen if this is the second time AddTrack has created a sender
|
||
// for this track.
|
||
if (FindSenderById(sender_id)) {
|
||
sender_id = rtc::CreateRandomUuid();
|
||
}
|
||
auto sender = CreateSender(media_type, sender_id, track, stream_ids);
|
||
auto receiver = CreateReceiver(media_type, rtc::CreateRandomUuid());
|
||
transceiver = CreateAndAddTransceiver(sender, receiver);
|
||
transceiver->internal()->set_created_by_addtrack(true);
|
||
transceiver->internal()->set_direction(RtpTransceiverDirection::kSendRecv);
|
||
}
|
||
return transceiver->sender();
|
||
}
|
||
|
||
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
||
PeerConnection::FindFirstTransceiverForAddedTrack(
|
||
rtc::scoped_refptr<MediaStreamTrackInterface> track) {
|
||
RTC_DCHECK(track);
|
||
for (auto transceiver : transceivers_) {
|
||
if (!transceiver->sender()->track() &&
|
||
cricket::MediaTypeToString(transceiver->media_type()) ==
|
||
track->kind() &&
|
||
!transceiver->internal()->has_ever_been_used_to_send() &&
|
||
!transceiver->stopped()) {
|
||
return transceiver;
|
||
}
|
||
}
|
||
return nullptr;
|
||
}
|
||
|
||
bool PeerConnection::RemoveTrack(RtpSenderInterface* sender) {
|
||
TRACE_EVENT0("webrtc", "PeerConnection::RemoveTrack");
|
||
return RemoveTrackNew(sender).ok();
|
||
}
|
||
|
||
RTCError PeerConnection::RemoveTrackNew(
|
||
rtc::scoped_refptr<RtpSenderInterface> sender) {
|
||
if (!sender) {
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, "Sender is null.");
|
||
}
|
||
if (IsClosed()) {
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
|
||
"PeerConnection is closed.");
|
||
}
|
||
if (IsUnifiedPlan()) {
|
||
auto transceiver = FindTransceiverBySender(sender);
|
||
if (!transceiver || !sender->track()) {
|
||
return RTCError::OK();
|
||
}
|
||
sender->SetTrack(nullptr);
|
||
if (transceiver->direction() == RtpTransceiverDirection::kSendRecv) {
|
||
transceiver->internal()->set_direction(
|
||
RtpTransceiverDirection::kRecvOnly);
|
||
} else if (transceiver->direction() == RtpTransceiverDirection::kSendOnly) {
|
||
transceiver->internal()->set_direction(
|
||
RtpTransceiverDirection::kInactive);
|
||
}
|
||
} else {
|
||
bool removed;
|
||
if (sender->media_type() == cricket::MEDIA_TYPE_AUDIO) {
|
||
removed = GetAudioTransceiver()->internal()->RemoveSender(sender);
|
||
} else {
|
||
RTC_DCHECK_EQ(cricket::MEDIA_TYPE_VIDEO, sender->media_type());
|
||
removed = GetVideoTransceiver()->internal()->RemoveSender(sender);
|
||
}
|
||
if (!removed) {
|
||
LOG_AND_RETURN_ERROR(
|
||
RTCErrorType::INVALID_PARAMETER,
|
||
"Couldn't find sender " + sender->id() + " to remove.");
|
||
}
|
||
}
|
||
Observer()->OnRenegotiationNeeded();
|
||
return RTCError::OK();
|
||
}
|
||
|
||
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
||
PeerConnection::FindTransceiverBySender(
|
||
rtc::scoped_refptr<RtpSenderInterface> sender) {
|
||
for (auto transceiver : transceivers_) {
|
||
if (transceiver->sender() == sender) {
|
||
return transceiver;
|
||
}
|
||
}
|
||
return nullptr;
|
||
}
|
||
|
||
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
|
||
PeerConnection::AddTransceiver(
|
||
rtc::scoped_refptr<MediaStreamTrackInterface> track) {
|
||
return AddTransceiver(track, RtpTransceiverInit());
|
||
}
|
||
|
||
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
|
||
PeerConnection::AddTransceiver(
|
||
rtc::scoped_refptr<MediaStreamTrackInterface> track,
|
||
const RtpTransceiverInit& init) {
|
||
RTC_CHECK(IsUnifiedPlan())
|
||
<< "AddTransceiver is only available with Unified Plan SdpSemantics";
|
||
if (!track) {
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, "track is null");
|
||
}
|
||
cricket::MediaType media_type;
|
||
if (track->kind() == MediaStreamTrackInterface::kAudioKind) {
|
||
media_type = cricket::MEDIA_TYPE_AUDIO;
|
||
} else if (track->kind() == MediaStreamTrackInterface::kVideoKind) {
|
||
media_type = cricket::MEDIA_TYPE_VIDEO;
|
||
} else {
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
||
"Track kind is not audio or video");
|
||
}
|
||
return AddTransceiver(media_type, track, init);
|
||
}
|
||
|
||
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
|
||
PeerConnection::AddTransceiver(cricket::MediaType media_type) {
|
||
return AddTransceiver(media_type, RtpTransceiverInit());
|
||
}
|
||
|
||
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
|
||
PeerConnection::AddTransceiver(cricket::MediaType media_type,
|
||
const RtpTransceiverInit& init) {
|
||
RTC_CHECK(IsUnifiedPlan())
|
||
<< "AddTransceiver is only available with Unified Plan SdpSemantics";
|
||
if (!(media_type == cricket::MEDIA_TYPE_AUDIO ||
|
||
media_type == cricket::MEDIA_TYPE_VIDEO)) {
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
||
"media type is not audio or video");
|
||
}
|
||
return AddTransceiver(media_type, nullptr, init);
|
||
}
|
||
|
||
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
|
||
PeerConnection::AddTransceiver(
|
||
cricket::MediaType media_type,
|
||
rtc::scoped_refptr<MediaStreamTrackInterface> track,
|
||
const RtpTransceiverInit& init,
|
||
bool fire_callback) {
|
||
RTC_DCHECK((media_type == cricket::MEDIA_TYPE_AUDIO ||
|
||
media_type == cricket::MEDIA_TYPE_VIDEO));
|
||
if (track) {
|
||
RTC_DCHECK_EQ(media_type,
|
||
(track->kind() == MediaStreamTrackInterface::kAudioKind
|
||
? cricket::MEDIA_TYPE_AUDIO
|
||
: cricket::MEDIA_TYPE_VIDEO));
|
||
}
|
||
|
||
// TODO(bugs.webrtc.org/7600): Verify init.
|
||
|
||
RTC_LOG(LS_INFO) << "Adding " << cricket::MediaTypeToString(media_type)
|
||
<< " transceiver in response to a call to AddTransceiver.";
|
||
// Set the sender ID equal to the track ID if the track is specified unless
|
||
// that sender ID is already in use.
|
||
std::string sender_id =
|
||
(track && !FindSenderById(track->id()) ? track->id()
|
||
: rtc::CreateRandomUuid());
|
||
auto sender = CreateSender(media_type, sender_id, track, init.stream_ids);
|
||
auto receiver = CreateReceiver(media_type, rtc::CreateRandomUuid());
|
||
auto transceiver = CreateAndAddTransceiver(sender, receiver);
|
||
transceiver->internal()->set_direction(init.direction);
|
||
|
||
if (fire_callback) {
|
||
Observer()->OnRenegotiationNeeded();
|
||
}
|
||
|
||
return rtc::scoped_refptr<RtpTransceiverInterface>(transceiver);
|
||
}
|
||
|
||
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>
|
||
PeerConnection::CreateSender(
|
||
cricket::MediaType media_type,
|
||
const std::string& id,
|
||
rtc::scoped_refptr<MediaStreamTrackInterface> track,
|
||
const std::vector<std::string>& stream_ids) {
|
||
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> sender;
|
||
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
|
||
RTC_DCHECK(!track ||
|
||
(track->kind() == MediaStreamTrackInterface::kAudioKind));
|
||
sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
|
||
signaling_thread(),
|
||
new AudioRtpSender(worker_thread(), id, stats_.get()));
|
||
NoteUsageEvent(UsageEvent::AUDIO_ADDED);
|
||
} else {
|
||
RTC_DCHECK_EQ(media_type, cricket::MEDIA_TYPE_VIDEO);
|
||
RTC_DCHECK(!track ||
|
||
(track->kind() == MediaStreamTrackInterface::kVideoKind));
|
||
sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
|
||
signaling_thread(), new VideoRtpSender(worker_thread(), id));
|
||
NoteUsageEvent(UsageEvent::VIDEO_ADDED);
|
||
}
|
||
bool set_track_succeeded = sender->SetTrack(track);
|
||
RTC_DCHECK(set_track_succeeded);
|
||
sender->internal()->set_stream_ids(stream_ids);
|
||
return sender;
|
||
}
|
||
|
||
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
|
||
PeerConnection::CreateReceiver(cricket::MediaType media_type,
|
||
const std::string& receiver_id) {
|
||
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
|
||
receiver;
|
||
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
|
||
receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
|
||
signaling_thread(), new AudioRtpReceiver(worker_thread(), receiver_id,
|
||
std::vector<std::string>({})));
|
||
NoteUsageEvent(UsageEvent::AUDIO_ADDED);
|
||
} else {
|
||
RTC_DCHECK_EQ(media_type, cricket::MEDIA_TYPE_VIDEO);
|
||
receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
|
||
signaling_thread(), new VideoRtpReceiver(worker_thread(), receiver_id,
|
||
std::vector<std::string>({})));
|
||
NoteUsageEvent(UsageEvent::VIDEO_ADDED);
|
||
}
|
||
return receiver;
|
||
}
|
||
|
||
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
||
PeerConnection::CreateAndAddTransceiver(
|
||
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> sender,
|
||
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
|
||
receiver) {
|
||
// Ensure that the new sender does not have an ID that is already in use by
|
||
// another sender.
|
||
// Allow receiver IDs to conflict since those come from remote SDP (which
|
||
// could be invalid, but should not cause a crash).
|
||
RTC_DCHECK(!FindSenderById(sender->id()));
|
||
auto transceiver = RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(
|
||
signaling_thread(), new RtpTransceiver(sender, receiver));
|
||
transceivers_.push_back(transceiver);
|
||
transceiver->internal()->SignalNegotiationNeeded.connect(
|
||
this, &PeerConnection::OnNegotiationNeeded);
|
||
return transceiver;
|
||
}
|
||
|
||
void PeerConnection::OnNegotiationNeeded() {
|
||
RTC_DCHECK_RUN_ON(signaling_thread());
|
||
RTC_DCHECK(!IsClosed());
|
||
Observer()->OnRenegotiationNeeded();
|
||
}
|
||
|
||
rtc::scoped_refptr<RtpSenderInterface> PeerConnection::CreateSender(
|
||
const std::string& kind,
|
||
const std::string& stream_id) {
|
||
RTC_CHECK(!IsUnifiedPlan()) << "CreateSender is not available with Unified "
|
||
"Plan SdpSemantics. Please use AddTransceiver "
|
||
"instead.";
|
||
TRACE_EVENT0("webrtc", "PeerConnection::CreateSender");
|
||
if (IsClosed()) {
|
||
return nullptr;
|
||
}
|
||
|
||
// Internally we need to have one stream with Plan B semantics, so we
|
||
// generate a random stream ID if not specified.
|
||
std::vector<std::string> stream_ids;
|
||
if (stream_id.empty()) {
|
||
stream_ids.push_back(rtc::CreateRandomUuid());
|
||
RTC_LOG(LS_INFO)
|
||
<< "No stream_id specified for sender. Generated stream ID: "
|
||
<< stream_ids[0];
|
||
} else {
|
||
stream_ids.push_back(stream_id);
|
||
}
|
||
|
||
// TODO(steveanton): Move construction of the RtpSenders to RtpTransceiver.
|
||
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> new_sender;
|
||
if (kind == MediaStreamTrackInterface::kAudioKind) {
|
||
auto* audio_sender = new AudioRtpSender(
|
||
worker_thread(), rtc::CreateRandomUuid(), stats_.get());
|
||
audio_sender->SetVoiceMediaChannel(voice_media_channel());
|
||
new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
|
||
signaling_thread(), audio_sender);
|
||
GetAudioTransceiver()->internal()->AddSender(new_sender);
|
||
} else if (kind == MediaStreamTrackInterface::kVideoKind) {
|
||
auto* video_sender =
|
||
new VideoRtpSender(worker_thread(), rtc::CreateRandomUuid());
|
||
video_sender->SetVideoMediaChannel(video_media_channel());
|
||
new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
|
||
signaling_thread(), video_sender);
|
||
GetVideoTransceiver()->internal()->AddSender(new_sender);
|
||
} else {
|
||
RTC_LOG(LS_ERROR) << "CreateSender called with invalid kind: " << kind;
|
||
return nullptr;
|
||
}
|
||
new_sender->internal()->set_stream_ids(stream_ids);
|
||
|
||
return new_sender;
|
||
}
|
||
|
||
std::vector<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::GetSenders()
|
||
const {
|
||
std::vector<rtc::scoped_refptr<RtpSenderInterface>> ret;
|
||
for (auto sender : GetSendersInternal()) {
|
||
ret.push_back(sender);
|
||
}
|
||
return ret;
|
||
}
|
||
|
||
std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>
|
||
PeerConnection::GetSendersInternal() const {
|
||
std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>
|
||
all_senders;
|
||
for (auto transceiver : transceivers_) {
|
||
auto senders = transceiver->internal()->senders();
|
||
all_senders.insert(all_senders.end(), senders.begin(), senders.end());
|
||
}
|
||
return all_senders;
|
||
}
|
||
|
||
std::vector<rtc::scoped_refptr<RtpReceiverInterface>>
|
||
PeerConnection::GetReceivers() const {
|
||
std::vector<rtc::scoped_refptr<RtpReceiverInterface>> ret;
|
||
for (const auto& receiver : GetReceiversInternal()) {
|
||
ret.push_back(receiver);
|
||
}
|
||
return ret;
|
||
}
|
||
|
||
std::vector<
|
||
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>>
|
||
PeerConnection::GetReceiversInternal() const {
|
||
std::vector<
|
||
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>>
|
||
all_receivers;
|
||
for (auto transceiver : transceivers_) {
|
||
auto receivers = transceiver->internal()->receivers();
|
||
all_receivers.insert(all_receivers.end(), receivers.begin(),
|
||
receivers.end());
|
||
}
|
||
return all_receivers;
|
||
}
|
||
|
||
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>
|
||
PeerConnection::GetTransceivers() const {
|
||
RTC_CHECK(IsUnifiedPlan())
|
||
<< "GetTransceivers is only supported with Unified Plan SdpSemantics.";
|
||
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> all_transceivers;
|
||
for (auto transceiver : transceivers_) {
|
||
all_transceivers.push_back(transceiver);
|
||
}
|
||
return all_transceivers;
|
||
}
|
||
|
||
bool PeerConnection::GetStats(StatsObserver* observer,
|
||
MediaStreamTrackInterface* track,
|
||
StatsOutputLevel level) {
|
||
TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
|
||
RTC_DCHECK(signaling_thread()->IsCurrent());
|
||
if (!observer) {
|
||
RTC_LOG(LS_ERROR) << "GetStats - observer is NULL.";
|
||
return false;
|
||
}
|
||
|
||
stats_->UpdateStats(level);
|
||
// The StatsCollector is used to tell if a track is valid because it may
|
||
// remember tracks that the PeerConnection previously removed.
|
||
if (track && !stats_->IsValidTrack(track->id())) {
|
||
RTC_LOG(LS_WARNING) << "GetStats is called with an invalid track: "
|
||
<< track->id();
|
||
return false;
|
||
}
|
||
signaling_thread()->Post(RTC_FROM_HERE, this, MSG_GETSTATS,
|
||
new GetStatsMsg(observer, track));
|
||
return true;
|
||
}
|
||
|
||
void PeerConnection::GetStats(RTCStatsCollectorCallback* callback) {
|
||
TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
|
||
RTC_DCHECK(stats_collector_);
|
||
RTC_DCHECK(callback);
|
||
stats_collector_->GetStatsReport(callback);
|
||
}
|
||
|
||
void PeerConnection::GetStats(
|
||
rtc::scoped_refptr<RtpSenderInterface> selector,
|
||
rtc::scoped_refptr<RTCStatsCollectorCallback> callback) {
|
||
TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
|
||
RTC_DCHECK(callback);
|
||
RTC_DCHECK(stats_collector_);
|
||
rtc::scoped_refptr<RtpSenderInternal> internal_sender;
|
||
if (selector) {
|
||
for (const auto& proxy_transceiver : transceivers_) {
|
||
for (const auto& proxy_sender :
|
||
proxy_transceiver->internal()->senders()) {
|
||
if (proxy_sender == selector) {
|
||
internal_sender = proxy_sender->internal();
|
||
break;
|
||
}
|
||
}
|
||
if (internal_sender)
|
||
break;
|
||
}
|
||
}
|
||
// If there is no |internal_sender| then |selector| is either null or does not
|
||
// belong to the PeerConnection (in Plan B, senders can be removed from the
|
||
// PeerConnection). This means that "all the stats objects representing the
|
||
// selector" is an empty set. Invoking GetStatsReport() with a null selector
|
||
// produces an empty stats report.
|
||
stats_collector_->GetStatsReport(internal_sender, callback);
|
||
}
|
||
|
||
void PeerConnection::GetStats(
|
||
rtc::scoped_refptr<RtpReceiverInterface> selector,
|
||
rtc::scoped_refptr<RTCStatsCollectorCallback> callback) {
|
||
TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
|
||
RTC_DCHECK(callback);
|
||
RTC_DCHECK(stats_collector_);
|
||
rtc::scoped_refptr<RtpReceiverInternal> internal_receiver;
|
||
if (selector) {
|
||
for (const auto& proxy_transceiver : transceivers_) {
|
||
for (const auto& proxy_receiver :
|
||
proxy_transceiver->internal()->receivers()) {
|
||
if (proxy_receiver == selector) {
|
||
internal_receiver = proxy_receiver->internal();
|
||
break;
|
||
}
|
||
}
|
||
if (internal_receiver)
|
||
break;
|
||
}
|
||
}
|
||
// If there is no |internal_receiver| then |selector| is either null or does
|
||
// not belong to the PeerConnection (in Plan B, receivers can be removed from
|
||
// the PeerConnection). This means that "all the stats objects representing
|
||
// the selector" is an empty set. Invoking GetStatsReport() with a null
|
||
// selector produces an empty stats report.
|
||
stats_collector_->GetStatsReport(internal_receiver, callback);
|
||
}
|
||
|
||
PeerConnectionInterface::SignalingState PeerConnection::signaling_state() {
|
||
return signaling_state_;
|
||
}
|
||
|
||
PeerConnectionInterface::IceConnectionState
|
||
PeerConnection::ice_connection_state() {
|
||
return ice_connection_state_;
|
||
}
|
||
|
||
PeerConnectionInterface::IceGatheringState
|
||
PeerConnection::ice_gathering_state() {
|
||
return ice_gathering_state_;
|
||
}
|
||
|
||
rtc::scoped_refptr<DataChannelInterface> PeerConnection::CreateDataChannel(
|
||
const std::string& label,
|
||
const DataChannelInit* config) {
|
||
TRACE_EVENT0("webrtc", "PeerConnection::CreateDataChannel");
|
||
|
||
bool first_datachannel = !HasDataChannels();
|
||
|
||
std::unique_ptr<InternalDataChannelInit> internal_config;
|
||
if (config) {
|
||
internal_config.reset(new InternalDataChannelInit(*config));
|
||
}
|
||
rtc::scoped_refptr<DataChannelInterface> channel(
|
||
InternalCreateDataChannel(label, internal_config.get()));
|
||
if (!channel.get()) {
|
||
return nullptr;
|
||
}
|
||
|
||
// Trigger the onRenegotiationNeeded event for every new RTP DataChannel, or
|
||
// the first SCTP DataChannel.
|
||
if (data_channel_type() == cricket::DCT_RTP || first_datachannel) {
|
||
Observer()->OnRenegotiationNeeded();
|
||
}
|
||
NoteUsageEvent(UsageEvent::DATA_ADDED);
|
||
return DataChannelProxy::Create(signaling_thread(), channel.get());
|
||
}
|
||
|
||
void PeerConnection::CreateOffer(CreateSessionDescriptionObserver* observer,
|
||
const RTCOfferAnswerOptions& options) {
|
||
TRACE_EVENT0("webrtc", "PeerConnection::CreateOffer");
|
||
|
||
if (!observer) {
|
||
RTC_LOG(LS_ERROR) << "CreateOffer - observer is NULL.";
|
||
return;
|
||
}
|
||
|
||
if (IsClosed()) {
|
||
std::string error = "CreateOffer called when PeerConnection is closed.";
|
||
RTC_LOG(LS_ERROR) << error;
|
||
PostCreateSessionDescriptionFailure(
|
||
observer, RTCError(RTCErrorType::INVALID_STATE, std::move(error)));
|
||
return;
|
||
}
|
||
|
||
if (!ValidateOfferAnswerOptions(options)) {
|
||
std::string error = "CreateOffer called with invalid options.";
|
||
RTC_LOG(LS_ERROR) << error;
|
||
PostCreateSessionDescriptionFailure(
|
||
observer, RTCError(RTCErrorType::INVALID_PARAMETER, std::move(error)));
|
||
return;
|
||
}
|
||
|
||
// Legacy handling for offer_to_receive_audio and offer_to_receive_video.
|
||
// Specified in WebRTC section 4.4.3.2 "Legacy configuration extensions".
|
||
if (IsUnifiedPlan()) {
|
||
RTCError error = HandleLegacyOfferOptions(options);
|
||
if (!error.ok()) {
|
||
PostCreateSessionDescriptionFailure(observer, std::move(error));
|
||
return;
|
||
}
|
||
}
|
||
|
||
cricket::MediaSessionOptions session_options;
|
||
GetOptionsForOffer(options, &session_options);
|
||
webrtc_session_desc_factory_->CreateOffer(observer, options, session_options);
|
||
}
|
||
|
||
RTCError PeerConnection::HandleLegacyOfferOptions(
|
||
const RTCOfferAnswerOptions& options) {
|
||
RTC_DCHECK(IsUnifiedPlan());
|
||
|
||
if (options.offer_to_receive_audio == 0) {
|
||
RemoveRecvDirectionFromReceivingTransceiversOfType(
|
||
cricket::MEDIA_TYPE_AUDIO);
|
||
} else if (options.offer_to_receive_audio == 1) {
|
||
AddUpToOneReceivingTransceiverOfType(cricket::MEDIA_TYPE_AUDIO);
|
||
} else if (options.offer_to_receive_audio > 1) {
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_PARAMETER,
|
||
"offer_to_receive_audio > 1 is not supported.");
|
||
}
|
||
|
||
if (options.offer_to_receive_video == 0) {
|
||
RemoveRecvDirectionFromReceivingTransceiversOfType(
|
||
cricket::MEDIA_TYPE_VIDEO);
|
||
} else if (options.offer_to_receive_video == 1) {
|
||
AddUpToOneReceivingTransceiverOfType(cricket::MEDIA_TYPE_VIDEO);
|
||
} else if (options.offer_to_receive_video > 1) {
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_PARAMETER,
|
||
"offer_to_receive_video > 1 is not supported.");
|
||
}
|
||
|
||
return RTCError::OK();
|
||
}
|
||
|
||
void PeerConnection::RemoveRecvDirectionFromReceivingTransceiversOfType(
|
||
cricket::MediaType media_type) {
|
||
for (auto transceiver : GetReceivingTransceiversOfType(media_type)) {
|
||
RtpTransceiverDirection new_direction =
|
||
RtpTransceiverDirectionWithRecvSet(transceiver->direction(), false);
|
||
if (new_direction != transceiver->direction()) {
|
||
RTC_LOG(LS_INFO) << "Changing " << cricket::MediaTypeToString(media_type)
|
||
<< " transceiver (MID="
|
||
<< transceiver->mid().value_or("<not set>") << ") from "
|
||
<< RtpTransceiverDirectionToString(
|
||
transceiver->direction())
|
||
<< " to "
|
||
<< RtpTransceiverDirectionToString(new_direction)
|
||
<< " since CreateOffer specified offer_to_receive=0";
|
||
transceiver->internal()->set_direction(new_direction);
|
||
}
|
||
}
|
||
}
|
||
|
||
void PeerConnection::AddUpToOneReceivingTransceiverOfType(
|
||
cricket::MediaType media_type) {
|
||
if (GetReceivingTransceiversOfType(media_type).empty()) {
|
||
RTC_LOG(LS_INFO)
|
||
<< "Adding one recvonly " << cricket::MediaTypeToString(media_type)
|
||
<< " transceiver since CreateOffer specified offer_to_receive=1";
|
||
RtpTransceiverInit init;
|
||
init.direction = RtpTransceiverDirection::kRecvOnly;
|
||
AddTransceiver(media_type, nullptr, init, /*fire_callback=*/false);
|
||
}
|
||
}
|
||
|
||
std::vector<rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
|
||
PeerConnection::GetReceivingTransceiversOfType(cricket::MediaType media_type) {
|
||
std::vector<
|
||
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
|
||
receiving_transceivers;
|
||
for (auto transceiver : transceivers_) {
|
||
if (!transceiver->stopped() && transceiver->media_type() == media_type &&
|
||
RtpTransceiverDirectionHasRecv(transceiver->direction())) {
|
||
receiving_transceivers.push_back(transceiver);
|
||
}
|
||
}
|
||
return receiving_transceivers;
|
||
}
|
||
|
||
void PeerConnection::CreateAnswer(CreateSessionDescriptionObserver* observer,
|
||
const RTCOfferAnswerOptions& options) {
|
||
TRACE_EVENT0("webrtc", "PeerConnection::CreateAnswer");
|
||
if (!observer) {
|
||
RTC_LOG(LS_ERROR) << "CreateAnswer - observer is NULL.";
|
||
return;
|
||
}
|
||
|
||
if (!(signaling_state_ == kHaveRemoteOffer ||
|
||
signaling_state_ == kHaveLocalPrAnswer)) {
|
||
std::string error =
|
||
"PeerConnection cannot create an answer in a state other than "
|
||
"have-remote-offer or have-local-pranswer.";
|
||
RTC_LOG(LS_ERROR) << error;
|
||
PostCreateSessionDescriptionFailure(
|
||
observer, RTCError(RTCErrorType::INVALID_STATE, std::move(error)));
|
||
return;
|
||
}
|
||
|
||
// The remote description should be set if we're in the right state.
|
||
RTC_DCHECK(remote_description());
|
||
|
||
if (IsUnifiedPlan()) {
|
||
if (options.offer_to_receive_audio != RTCOfferAnswerOptions::kUndefined) {
|
||
RTC_LOG(LS_WARNING) << "CreateAnswer: offer_to_receive_audio is not "
|
||
"supported with Unified Plan semantics. Use the "
|
||
"RtpTransceiver API instead.";
|
||
}
|
||
if (options.offer_to_receive_video != RTCOfferAnswerOptions::kUndefined) {
|
||
RTC_LOG(LS_WARNING) << "CreateAnswer: offer_to_receive_video is not "
|
||
"supported with Unified Plan semantics. Use the "
|
||
"RtpTransceiver API instead.";
|
||
}
|
||
}
|
||
|
||
cricket::MediaSessionOptions session_options;
|
||
GetOptionsForAnswer(options, &session_options);
|
||
|
||
webrtc_session_desc_factory_->CreateAnswer(observer, session_options);
|
||
}
|
||
|
||
void PeerConnection::SetLocalDescription(
|
||
SetSessionDescriptionObserver* observer,
|
||
SessionDescriptionInterface* desc_ptr) {
|
||
TRACE_EVENT0("webrtc", "PeerConnection::SetLocalDescription");
|
||
|
||
// The SetLocalDescription contract is that we take ownership of the session
|
||
// description regardless of the outcome, so wrap it in a unique_ptr right
|
||
// away. Ideally, SetLocalDescription's signature will be changed to take the
|
||
// description as a unique_ptr argument to formalize this agreement.
|
||
std::unique_ptr<SessionDescriptionInterface> desc(desc_ptr);
|
||
|
||
if (!observer) {
|
||
RTC_LOG(LS_ERROR) << "SetLocalDescription - observer is NULL.";
|
||
return;
|
||
}
|
||
|
||
if (!desc) {
|
||
PostSetSessionDescriptionFailure(
|
||
observer,
|
||
RTCError(RTCErrorType::INTERNAL_ERROR, "SessionDescription is NULL."));
|
||
return;
|
||
}
|
||
|
||
// If a session error has occurred the PeerConnection is in a possibly
|
||
// inconsistent state so fail right away.
|
||
if (session_error() != SessionError::kNone) {
|
||
std::string error_message = GetSessionErrorMsg();
|
||
RTC_LOG(LS_ERROR) << "SetLocalDescription: " << error_message;
|
||
PostSetSessionDescriptionFailure(
|
||
observer,
|
||
RTCError(RTCErrorType::INTERNAL_ERROR, std::move(error_message)));
|
||
return;
|
||
}
|
||
|
||
RTCError error = ValidateSessionDescription(desc.get(), cricket::CS_LOCAL);
|
||
if (!error.ok()) {
|
||
std::string error_message = GetSetDescriptionErrorMessage(
|
||
cricket::CS_LOCAL, desc->GetType(), error);
|
||
RTC_LOG(LS_ERROR) << error_message;
|
||
PostSetSessionDescriptionFailure(
|
||
observer,
|
||
RTCError(RTCErrorType::INTERNAL_ERROR, std::move(error_message)));
|
||
return;
|
||
}
|
||
|
||
// Grab the description type before moving ownership to ApplyLocalDescription,
|
||
// which may destroy it before returning.
|
||
const SdpType type = desc->GetType();
|
||
|
||
error = ApplyLocalDescription(std::move(desc));
|
||
// |desc| may be destroyed at this point.
|
||
|
||
if (!error.ok()) {
|
||
// If ApplyLocalDescription fails, the PeerConnection could be in an
|
||
// inconsistent state, so act conservatively here and set the session error
|
||
// so that future calls to SetLocalDescription/SetRemoteDescription fail.
|
||
SetSessionError(SessionError::kContent, error.message());
|
||
std::string error_message =
|
||
GetSetDescriptionErrorMessage(cricket::CS_LOCAL, type, error);
|
||
RTC_LOG(LS_ERROR) << error_message;
|
||
PostSetSessionDescriptionFailure(
|
||
observer,
|
||
RTCError(RTCErrorType::INTERNAL_ERROR, std::move(error_message)));
|
||
return;
|
||
}
|
||
RTC_DCHECK(local_description());
|
||
|
||
PostSetSessionDescriptionSuccess(observer);
|
||
|
||
// MaybeStartGathering needs to be called after posting
|
||
// MSG_SET_SESSIONDESCRIPTION_SUCCESS, so that we don't signal any candidates
|
||
// before signaling that SetLocalDescription completed.
|
||
transport_controller_->MaybeStartGathering();
|
||
|
||
if (local_description()->GetType() == SdpType::kAnswer) {
|
||
// TODO(deadbeef): We already had to hop to the network thread for
|
||
// MaybeStartGathering...
|
||
network_thread()->Invoke<void>(
|
||
RTC_FROM_HERE, rtc::Bind(&cricket::PortAllocator::DiscardCandidatePool,
|
||
port_allocator_.get()));
|
||
// Make UMA notes about what was agreed to.
|
||
ReportNegotiatedSdpSemantics(*local_description());
|
||
}
|
||
NoteUsageEvent(UsageEvent::SET_LOCAL_DESCRIPTION_CALLED);
|
||
}
|
||
|
||
RTCError PeerConnection::ApplyLocalDescription(
|
||
std::unique_ptr<SessionDescriptionInterface> desc) {
|
||
RTC_DCHECK_RUN_ON(signaling_thread());
|
||
RTC_DCHECK(desc);
|
||
|
||
// Update stats here so that we have the most recent stats for tracks and
|
||
// streams that might be removed by updating the session description.
|
||
stats_->UpdateStats(kStatsOutputLevelStandard);
|
||
|
||
// Take a reference to the old local description since it's used below to
|
||
// compare against the new local description. When setting the new local
|
||
// description, grab ownership of the replaced session description in case it
|
||
// is the same as |old_local_description|, to keep it alive for the duration
|
||
// of the method.
|
||
const SessionDescriptionInterface* old_local_description =
|
||
local_description();
|
||
std::unique_ptr<SessionDescriptionInterface> replaced_local_description;
|
||
SdpType type = desc->GetType();
|
||
if (type == SdpType::kAnswer) {
|
||
replaced_local_description = pending_local_description_
|
||
? std::move(pending_local_description_)
|
||
: std::move(current_local_description_);
|
||
current_local_description_ = std::move(desc);
|
||
pending_local_description_ = nullptr;
|
||
current_remote_description_ = std::move(pending_remote_description_);
|
||
} else {
|
||
replaced_local_description = std::move(pending_local_description_);
|
||
pending_local_description_ = std::move(desc);
|
||
}
|
||
// The session description to apply now must be accessed by
|
||
// |local_description()|.
|
||
RTC_DCHECK(local_description());
|
||
|
||
RTCError error = PushdownTransportDescription(cricket::CS_LOCAL, type);
|
||
if (!error.ok()) {
|
||
return error;
|
||
}
|
||
|
||
if (IsUnifiedPlan()) {
|
||
RTCError error = UpdateTransceiversAndDataChannels(
|
||
cricket::CS_LOCAL, *local_description(), old_local_description,
|
||
remote_description());
|
||
if (!error.ok()) {
|
||
return error;
|
||
}
|
||
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> remove_list;
|
||
std::vector<rtc::scoped_refptr<MediaStreamInterface>> removed_streams;
|
||
for (auto transceiver : transceivers_) {
|
||
const ContentInfo* content =
|
||
FindMediaSectionForTransceiver(transceiver, local_description());
|
||
if (!content) {
|
||
continue;
|
||
}
|
||
const MediaContentDescription* media_desc = content->media_description();
|
||
// 2.2.7.1.6: If description is of type "answer" or "pranswer", then run
|
||
// the following steps:
|
||
if (type == SdpType::kPrAnswer || type == SdpType::kAnswer) {
|
||
// 2.2.7.1.6.1: If direction is "sendonly" or "inactive", and
|
||
// transceiver's [[FiredDirection]] slot is either "sendrecv" or
|
||
// "recvonly", process the removal of a remote track for the media
|
||
// description, given transceiver, removeList, and muteTracks.
|
||
if (!RtpTransceiverDirectionHasRecv(media_desc->direction()) &&
|
||
(transceiver->internal()->fired_direction() &&
|
||
RtpTransceiverDirectionHasRecv(
|
||
*transceiver->internal()->fired_direction()))) {
|
||
ProcessRemovalOfRemoteTrack(transceiver, &remove_list,
|
||
&removed_streams);
|
||
}
|
||
// 2.2.7.1.6.2: Set transceiver's [[CurrentDirection]] and
|
||
// [[FiredDirection]] slots to direction.
|
||
transceiver->internal()->set_current_direction(media_desc->direction());
|
||
transceiver->internal()->set_fired_direction(media_desc->direction());
|
||
}
|
||
}
|
||
auto observer = Observer();
|
||
for (auto transceiver : remove_list) {
|
||
observer->OnRemoveTrack(transceiver->receiver());
|
||
}
|
||
for (auto stream : removed_streams) {
|
||
observer->OnRemoveStream(stream);
|
||
}
|
||
} else {
|
||
// Media channels will be created only when offer is set. These may use new
|
||
// transports just created by PushdownTransportDescription.
|
||
if (type == SdpType::kOffer) {
|
||
// TODO(bugs.webrtc.org/4676) - Handle CreateChannel failure, as new local
|
||
// description is applied. Restore back to old description.
|
||
RTCError error = CreateChannels(*local_description()->description());
|
||
if (!error.ok()) {
|
||
return error;
|
||
}
|
||
}
|
||
// Remove unused channels if MediaContentDescription is rejected.
|
||
RemoveUnusedChannels(local_description()->description());
|
||
}
|
||
|
||
error = UpdateSessionState(type, cricket::CS_LOCAL,
|
||
local_description()->description());
|
||
if (!error.ok()) {
|
||
return error;
|
||
}
|
||
|
||
if (remote_description()) {
|
||
// Now that we have a local description, we can push down remote candidates.
|
||
UseCandidatesInSessionDescription(remote_description());
|
||
}
|
||
|
||
pending_ice_restarts_.clear();
|
||
if (session_error() != SessionError::kNone) {
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR, GetSessionErrorMsg());
|
||
}
|
||
|
||
// If setting the description decided our SSL role, allocate any necessary
|
||
// SCTP sids.
|
||
rtc::SSLRole role;
|
||
if (data_channel_type() == cricket::DCT_SCTP && GetSctpSslRole(&role)) {
|
||
AllocateSctpSids(role);
|
||
}
|
||
|
||
if (IsUnifiedPlan()) {
|
||
for (auto transceiver : transceivers_) {
|
||
const ContentInfo* content =
|
||
FindMediaSectionForTransceiver(transceiver, local_description());
|
||
if (!content) {
|
||
continue;
|
||
}
|
||
const auto& streams = content->media_description()->streams();
|
||
if (!content->rejected && !streams.empty()) {
|
||
transceiver->internal()->sender_internal()->set_stream_ids(
|
||
streams[0].stream_ids());
|
||
transceiver->internal()->sender_internal()->SetSsrc(
|
||
streams[0].first_ssrc());
|
||
} else {
|
||
// 0 is a special value meaning "this sender has no associated send
|
||
// stream". Need to call this so the sender won't attempt to configure
|
||
// a no longer existing stream and run into DCHECKs in the lower
|
||
// layers.
|
||
transceiver->internal()->sender_internal()->SetSsrc(0);
|
||
}
|
||
}
|
||
} else {
|
||
// Plan B semantics.
|
||
|
||
// Update state and SSRC of local MediaStreams and DataChannels based on the
|
||
// local session description.
|
||
const cricket::ContentInfo* audio_content =
|
||
GetFirstAudioContent(local_description()->description());
|
||
if (audio_content) {
|
||
if (audio_content->rejected) {
|
||
RemoveSenders(cricket::MEDIA_TYPE_AUDIO);
|
||
} else {
|
||
const cricket::AudioContentDescription* audio_desc =
|
||
audio_content->media_description()->as_audio();
|
||
UpdateLocalSenders(audio_desc->streams(), audio_desc->type());
|
||
}
|
||
}
|
||
|
||
const cricket::ContentInfo* video_content =
|
||
GetFirstVideoContent(local_description()->description());
|
||
if (video_content) {
|
||
if (video_content->rejected) {
|
||
RemoveSenders(cricket::MEDIA_TYPE_VIDEO);
|
||
} else {
|
||
const cricket::VideoContentDescription* video_desc =
|
||
video_content->media_description()->as_video();
|
||
UpdateLocalSenders(video_desc->streams(), video_desc->type());
|
||
}
|
||
}
|
||
}
|
||
|
||
const cricket::ContentInfo* data_content =
|
||
GetFirstDataContent(local_description()->description());
|
||
if (data_content) {
|
||
const cricket::DataContentDescription* data_desc =
|
||
data_content->media_description()->as_data();
|
||
if (rtc::starts_with(data_desc->protocol().data(),
|
||
cricket::kMediaProtocolRtpPrefix)) {
|
||
UpdateLocalRtpDataChannels(data_desc->streams());
|
||
}
|
||
}
|
||
|
||
return RTCError::OK();
|
||
}
|
||
|
||
void PeerConnection::SetRemoteDescription(
|
||
SetSessionDescriptionObserver* observer,
|
||
SessionDescriptionInterface* desc) {
|
||
SetRemoteDescription(
|
||
std::unique_ptr<SessionDescriptionInterface>(desc),
|
||
rtc::scoped_refptr<SetRemoteDescriptionObserverInterface>(
|
||
new SetRemoteDescriptionObserverAdapter(this, observer)));
|
||
}
|
||
|
||
void PeerConnection::SetRemoteDescription(
|
||
std::unique_ptr<SessionDescriptionInterface> desc,
|
||
rtc::scoped_refptr<SetRemoteDescriptionObserverInterface> observer) {
|
||
TRACE_EVENT0("webrtc", "PeerConnection::SetRemoteDescription");
|
||
|
||
if (!observer) {
|
||
RTC_LOG(LS_ERROR) << "SetRemoteDescription - observer is NULL.";
|
||
return;
|
||
}
|
||
|
||
if (!desc) {
|
||
observer->OnSetRemoteDescriptionComplete(RTCError(
|
||
RTCErrorType::INVALID_PARAMETER, "SessionDescription is NULL."));
|
||
return;
|
||
}
|
||
|
||
// If a session error has occurred the PeerConnection is in a possibly
|
||
// inconsistent state so fail right away.
|
||
if (session_error() != SessionError::kNone) {
|
||
std::string error_message = GetSessionErrorMsg();
|
||
RTC_LOG(LS_ERROR) << "SetRemoteDescription: " << error_message;
|
||
observer->OnSetRemoteDescriptionComplete(
|
||
RTCError(RTCErrorType::INTERNAL_ERROR, std::move(error_message)));
|
||
return;
|
||
}
|
||
|
||
if (desc->GetType() == SdpType::kOffer) {
|
||
// Report to UMA the format of the received offer.
|
||
ReportSdpFormatReceived(*desc);
|
||
}
|
||
|
||
RTCError error = ValidateSessionDescription(desc.get(), cricket::CS_REMOTE);
|
||
if (!error.ok()) {
|
||
std::string error_message = GetSetDescriptionErrorMessage(
|
||
cricket::CS_REMOTE, desc->GetType(), error);
|
||
RTC_LOG(LS_ERROR) << error_message;
|
||
observer->OnSetRemoteDescriptionComplete(
|
||
RTCError(error.type(), std::move(error_message)));
|
||
return;
|
||
}
|
||
|
||
// Grab the description type before moving ownership to
|
||
// ApplyRemoteDescription, which may destroy it before returning.
|
||
const SdpType type = desc->GetType();
|
||
|
||
error = ApplyRemoteDescription(std::move(desc));
|
||
// |desc| may be destroyed at this point.
|
||
|
||
if (!error.ok()) {
|
||
// If ApplyRemoteDescription fails, the PeerConnection could be in an
|
||
// inconsistent state, so act conservatively here and set the session error
|
||
// so that future calls to SetLocalDescription/SetRemoteDescription fail.
|
||
SetSessionError(SessionError::kContent, error.message());
|
||
std::string error_message =
|
||
GetSetDescriptionErrorMessage(cricket::CS_REMOTE, type, error);
|
||
RTC_LOG(LS_ERROR) << error_message;
|
||
observer->OnSetRemoteDescriptionComplete(
|
||
RTCError(error.type(), std::move(error_message)));
|
||
return;
|
||
}
|
||
RTC_DCHECK(remote_description());
|
||
|
||
if (type == SdpType::kAnswer) {
|
||
// TODO(deadbeef): We already had to hop to the network thread for
|
||
// MaybeStartGathering...
|
||
network_thread()->Invoke<void>(
|
||
RTC_FROM_HERE, rtc::Bind(&cricket::PortAllocator::DiscardCandidatePool,
|
||
port_allocator_.get()));
|
||
// Make UMA notes about what was agreed to.
|
||
ReportNegotiatedSdpSemantics(*remote_description());
|
||
}
|
||
|
||
observer->OnSetRemoteDescriptionComplete(RTCError::OK());
|
||
NoteUsageEvent(UsageEvent::SET_REMOTE_DESCRIPTION_CALLED);
|
||
}
|
||
|
||
RTCError PeerConnection::ApplyRemoteDescription(
|
||
std::unique_ptr<SessionDescriptionInterface> desc) {
|
||
RTC_DCHECK_RUN_ON(signaling_thread());
|
||
RTC_DCHECK(desc);
|
||
|
||
// Update stats here so that we have the most recent stats for tracks and
|
||
// streams that might be removed by updating the session description.
|
||
stats_->UpdateStats(kStatsOutputLevelStandard);
|
||
|
||
// Take a reference to the old remote description since it's used below to
|
||
// compare against the new remote description. When setting the new remote
|
||
// description, grab ownership of the replaced session description in case it
|
||
// is the same as |old_remote_description|, to keep it alive for the duration
|
||
// of the method.
|
||
const SessionDescriptionInterface* old_remote_description =
|
||
remote_description();
|
||
std::unique_ptr<SessionDescriptionInterface> replaced_remote_description;
|
||
SdpType type = desc->GetType();
|
||
if (type == SdpType::kAnswer) {
|
||
replaced_remote_description = pending_remote_description_
|
||
? std::move(pending_remote_description_)
|
||
: std::move(current_remote_description_);
|
||
current_remote_description_ = std::move(desc);
|
||
pending_remote_description_ = nullptr;
|
||
current_local_description_ = std::move(pending_local_description_);
|
||
} else {
|
||
replaced_remote_description = std::move(pending_remote_description_);
|
||
pending_remote_description_ = std::move(desc);
|
||
}
|
||
// The session description to apply now must be accessed by
|
||
// |remote_description()|.
|
||
RTC_DCHECK(remote_description());
|
||
|
||
RTCError error = PushdownTransportDescription(cricket::CS_REMOTE, type);
|
||
if (!error.ok()) {
|
||
return error;
|
||
}
|
||
// Transport and Media channels will be created only when offer is set.
|
||
if (IsUnifiedPlan()) {
|
||
RTCError error = UpdateTransceiversAndDataChannels(
|
||
cricket::CS_REMOTE, *remote_description(), local_description(),
|
||
old_remote_description);
|
||
if (!error.ok()) {
|
||
return error;
|
||
}
|
||
} else {
|
||
// Media channels will be created only when offer is set. These may use new
|
||
// transports just created by PushdownTransportDescription.
|
||
if (type == SdpType::kOffer) {
|
||
// TODO(mallinath) - Handle CreateChannel failure, as new local
|
||
// description is applied. Restore back to old description.
|
||
RTCError error = CreateChannels(*remote_description()->description());
|
||
if (!error.ok()) {
|
||
return error;
|
||
}
|
||
}
|
||
// Remove unused channels if MediaContentDescription is rejected.
|
||
RemoveUnusedChannels(remote_description()->description());
|
||
}
|
||
|
||
// NOTE: Candidates allocation will be initiated only when
|
||
// SetLocalDescription is called.
|
||
error = UpdateSessionState(type, cricket::CS_REMOTE,
|
||
remote_description()->description());
|
||
if (!error.ok()) {
|
||
return error;
|
||
}
|
||
|
||
if (local_description() &&
|
||
!UseCandidatesInSessionDescription(remote_description())) {
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, kInvalidCandidates);
|
||
}
|
||
|
||
if (old_remote_description) {
|
||
for (const cricket::ContentInfo& content :
|
||
old_remote_description->description()->contents()) {
|
||
// Check if this new SessionDescription contains new ICE ufrag and
|
||
// password that indicates the remote peer requests an ICE restart.
|
||
// TODO(deadbeef): When we start storing both the current and pending
|
||
// remote description, this should reset pending_ice_restarts and compare
|
||
// against the current description.
|
||
if (CheckForRemoteIceRestart(old_remote_description, remote_description(),
|
||
content.name)) {
|
||
if (type == SdpType::kOffer) {
|
||
pending_ice_restarts_.insert(content.name);
|
||
}
|
||
} else {
|
||
// We retain all received candidates only if ICE is not restarted.
|
||
// When ICE is restarted, all previous candidates belong to an old
|
||
// generation and should not be kept.
|
||
// TODO(deadbeef): This goes against the W3C spec which says the remote
|
||
// description should only contain candidates from the last set remote
|
||
// description plus any candidates added since then. We should remove
|
||
// this once we're sure it won't break anything.
|
||
WebRtcSessionDescriptionFactory::CopyCandidatesFromSessionDescription(
|
||
old_remote_description, content.name, mutable_remote_description());
|
||
}
|
||
}
|
||
}
|
||
|
||
if (session_error() != SessionError::kNone) {
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR, GetSessionErrorMsg());
|
||
}
|
||
|
||
// Set the the ICE connection state to connecting since the connection may
|
||
// become writable with peer reflexive candidates before any remote candidate
|
||
// is signaled.
|
||
// TODO(pthatcher): This is a short-term solution for crbug/446908. A real fix
|
||
// is to have a new signal the indicates a change in checking state from the
|
||
// transport and expose a new checking() member from transport that can be
|
||
// read to determine the current checking state. The existing SignalConnecting
|
||
// actually means "gathering candidates", so cannot be be used here.
|
||
if (remote_description()->GetType() != SdpType::kOffer &&
|
||
remote_description()->number_of_mediasections() > 0u &&
|
||
ice_connection_state() == PeerConnectionInterface::kIceConnectionNew) {
|
||
SetIceConnectionState(PeerConnectionInterface::kIceConnectionChecking);
|
||
}
|
||
|
||
// If setting the description decided our SSL role, allocate any necessary
|
||
// SCTP sids.
|
||
rtc::SSLRole role;
|
||
if (data_channel_type() == cricket::DCT_SCTP && GetSctpSslRole(&role)) {
|
||
AllocateSctpSids(role);
|
||
}
|
||
|
||
if (IsUnifiedPlan()) {
|
||
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>
|
||
now_receiving_transceivers;
|
||
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> remove_list;
|
||
std::vector<rtc::scoped_refptr<MediaStreamInterface>> added_streams;
|
||
std::vector<rtc::scoped_refptr<MediaStreamInterface>> removed_streams;
|
||
for (auto transceiver : transceivers_) {
|
||
const ContentInfo* content =
|
||
FindMediaSectionForTransceiver(transceiver, remote_description());
|
||
if (!content) {
|
||
continue;
|
||
}
|
||
const MediaContentDescription* media_desc = content->media_description();
|
||
RtpTransceiverDirection local_direction =
|
||
RtpTransceiverDirectionReversed(media_desc->direction());
|
||
// From the WebRTC specification, steps 2.2.8.5/6 of section 4.4.1.6 "Set
|
||
// the RTCSessionDescription: If direction is sendrecv or recvonly, and
|
||
// transceiver's current direction is neither sendrecv nor recvonly,
|
||
// process the addition of a remote track for the media description.
|
||
std::vector<std::string> stream_ids;
|
||
if (!media_desc->streams().empty()) {
|
||
// The remote description has signaled the stream IDs.
|
||
stream_ids = media_desc->streams()[0].stream_ids();
|
||
}
|
||
if (RtpTransceiverDirectionHasRecv(local_direction) &&
|
||
(!transceiver->fired_direction() ||
|
||
!RtpTransceiverDirectionHasRecv(*transceiver->fired_direction()))) {
|
||
RTC_LOG(LS_INFO) << "Processing the addition of a new track for MID="
|
||
<< content->name << " (added to "
|
||
<< GetStreamIdsString(stream_ids) << ").";
|
||
|
||
std::vector<rtc::scoped_refptr<MediaStreamInterface>> media_streams;
|
||
for (const std::string& stream_id : stream_ids) {
|
||
rtc::scoped_refptr<MediaStreamInterface> stream =
|
||
remote_streams_->find(stream_id);
|
||
if (!stream) {
|
||
stream = MediaStreamProxy::Create(rtc::Thread::Current(),
|
||
MediaStream::Create(stream_id));
|
||
remote_streams_->AddStream(stream);
|
||
added_streams.push_back(stream);
|
||
}
|
||
media_streams.push_back(stream);
|
||
}
|
||
// This will add the remote track to the streams.
|
||
// TODO(hbos): When we remove remote_streams(), use set_stream_ids()
|
||
// instead. https://crbug.com/webrtc/9480
|
||
transceiver->internal()->receiver_internal()->SetStreams(media_streams);
|
||
now_receiving_transceivers.push_back(transceiver);
|
||
}
|
||
// 2.2.8.1.7: If direction is "sendonly" or "inactive", and transceiver's
|
||
// [[FiredDirection]] slot is either "sendrecv" or "recvonly", process the
|
||
// removal of a remote track for the media description, given transceiver,
|
||
// removeList, and muteTracks.
|
||
if (!RtpTransceiverDirectionHasRecv(local_direction) &&
|
||
(transceiver->fired_direction() &&
|
||
RtpTransceiverDirectionHasRecv(*transceiver->fired_direction()))) {
|
||
ProcessRemovalOfRemoteTrack(transceiver, &remove_list,
|
||
&removed_streams);
|
||
}
|
||
// 2.2.8.1.8: Set transceiver's [[FiredDirection]] slot to direction.
|
||
transceiver->internal()->set_fired_direction(local_direction);
|
||
// 2.2.8.1.9: If description is of type "answer" or "pranswer", then run
|
||
// the following steps:
|
||
if (type == SdpType::kPrAnswer || type == SdpType::kAnswer) {
|
||
// 2.2.8.1.9.1: Set transceiver's [[CurrentDirection]] slot to
|
||
// direction.
|
||
transceiver->internal()->set_current_direction(local_direction);
|
||
}
|
||
// 2.2.8.1.10: If the media description is rejected, and transceiver is
|
||
// not already stopped, stop the RTCRtpTransceiver transceiver.
|
||
if (content->rejected && !transceiver->stopped()) {
|
||
RTC_LOG(LS_INFO) << "Stopping transceiver for MID=" << content->name
|
||
<< " since the media section was rejected.";
|
||
transceiver->Stop();
|
||
}
|
||
if (!content->rejected &&
|
||
RtpTransceiverDirectionHasRecv(local_direction)) {
|
||
// Set ssrc to 0 in the case of an unsignalled ssrc.
|
||
uint32_t ssrc = 0;
|
||
if (!media_desc->streams().empty() &&
|
||
media_desc->streams()[0].has_ssrcs()) {
|
||
ssrc = media_desc->streams()[0].first_ssrc();
|
||
}
|
||
transceiver->internal()->receiver_internal()->SetupMediaChannel(ssrc);
|
||
}
|
||
}
|
||
// Once all processing has finished, fire off callbacks.
|
||
auto observer = Observer();
|
||
for (auto transceiver : now_receiving_transceivers) {
|
||
stats_->AddTrack(transceiver->receiver()->track());
|
||
observer->OnTrack(transceiver);
|
||
observer->OnAddTrack(transceiver->receiver(),
|
||
transceiver->receiver()->streams());
|
||
}
|
||
for (auto stream : added_streams) {
|
||
observer->OnAddStream(stream);
|
||
}
|
||
for (auto transceiver : remove_list) {
|
||
observer->OnRemoveTrack(transceiver->receiver());
|
||
}
|
||
for (auto stream : removed_streams) {
|
||
observer->OnRemoveStream(stream);
|
||
}
|
||
}
|
||
|
||
const cricket::ContentInfo* audio_content =
|
||
GetFirstAudioContent(remote_description()->description());
|
||
const cricket::ContentInfo* video_content =
|
||
GetFirstVideoContent(remote_description()->description());
|
||
const cricket::AudioContentDescription* audio_desc =
|
||
GetFirstAudioContentDescription(remote_description()->description());
|
||
const cricket::VideoContentDescription* video_desc =
|
||
GetFirstVideoContentDescription(remote_description()->description());
|
||
const cricket::DataContentDescription* data_desc =
|
||
GetFirstDataContentDescription(remote_description()->description());
|
||
|
||
// Check if the descriptions include streams, just in case the peer supports
|
||
// MSID, but doesn't indicate so with "a=msid-semantic".
|
||
if (remote_description()->description()->msid_supported() ||
|
||
(audio_desc && !audio_desc->streams().empty()) ||
|
||
(video_desc && !video_desc->streams().empty())) {
|
||
remote_peer_supports_msid_ = true;
|
||
}
|
||
|
||
// We wait to signal new streams until we finish processing the description,
|
||
// since only at that point will new streams have all their tracks.
|
||
rtc::scoped_refptr<StreamCollection> new_streams(StreamCollection::Create());
|
||
|
||
if (!IsUnifiedPlan()) {
|
||
// TODO(steveanton): When removing RTP senders/receivers in response to a
|
||
// rejected media section, there is some cleanup logic that expects the
|
||
// voice/ video channel to still be set. But in this method the voice/video
|
||
// channel would have been destroyed by the SetRemoteDescription caller
|
||
// above so the cleanup that relies on them fails to run. The RemoveSenders
|
||
// calls should be moved to right before the DestroyChannel calls to fix
|
||
// this.
|
||
|
||
// Find all audio rtp streams and create corresponding remote AudioTracks
|
||
// and MediaStreams.
|
||
if (audio_content) {
|
||
if (audio_content->rejected) {
|
||
RemoveSenders(cricket::MEDIA_TYPE_AUDIO);
|
||
} else {
|
||
bool default_audio_track_needed =
|
||
!remote_peer_supports_msid_ &&
|
||
RtpTransceiverDirectionHasSend(audio_desc->direction());
|
||
UpdateRemoteSendersList(GetActiveStreams(audio_desc),
|
||
default_audio_track_needed, audio_desc->type(),
|
||
new_streams);
|
||
}
|
||
}
|
||
|
||
// Find all video rtp streams and create corresponding remote VideoTracks
|
||
// and MediaStreams.
|
||
if (video_content) {
|
||
if (video_content->rejected) {
|
||
RemoveSenders(cricket::MEDIA_TYPE_VIDEO);
|
||
} else {
|
||
bool default_video_track_needed =
|
||
!remote_peer_supports_msid_ &&
|
||
RtpTransceiverDirectionHasSend(video_desc->direction());
|
||
UpdateRemoteSendersList(GetActiveStreams(video_desc),
|
||
default_video_track_needed, video_desc->type(),
|
||
new_streams);
|
||
}
|
||
}
|
||
|
||
// Update the DataChannels with the information from the remote peer.
|
||
if (data_desc) {
|
||
if (rtc::starts_with(data_desc->protocol().data(),
|
||
cricket::kMediaProtocolRtpPrefix)) {
|
||
UpdateRemoteRtpDataChannels(GetActiveStreams(data_desc));
|
||
}
|
||
}
|
||
|
||
// Iterate new_streams and notify the observer about new MediaStreams.
|
||
auto observer = Observer();
|
||
for (size_t i = 0; i < new_streams->count(); ++i) {
|
||
MediaStreamInterface* new_stream = new_streams->at(i);
|
||
stats_->AddStream(new_stream);
|
||
observer->OnAddStream(
|
||
rtc::scoped_refptr<MediaStreamInterface>(new_stream));
|
||
}
|
||
|
||
UpdateEndedRemoteMediaStreams();
|
||
}
|
||
|
||
return RTCError::OK();
|
||
}
|
||
|
||
void PeerConnection::ProcessRemovalOfRemoteTrack(
|
||
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
||
transceiver,
|
||
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>* remove_list,
|
||
std::vector<rtc::scoped_refptr<MediaStreamInterface>>* removed_streams) {
|
||
RTC_DCHECK(transceiver->mid());
|
||
RTC_LOG(LS_INFO) << "Processing the removal of a track for MID="
|
||
<< *transceiver->mid();
|
||
std::vector<rtc::scoped_refptr<MediaStreamInterface>> media_streams =
|
||
transceiver->internal()->receiver_internal()->streams();
|
||
// This will remove the remote track from the streams.
|
||
transceiver->internal()->receiver_internal()->set_stream_ids({});
|
||
remove_list->push_back(transceiver);
|
||
// Remove any streams that no longer have tracks.
|
||
// TODO(https://crbug.com/webrtc/9480): When we use stream IDs instead
|
||
// of streams, see if the stream was removed by checking if this was the
|
||
// last receiver with that stream ID.
|
||
for (auto stream : media_streams) {
|
||
if (stream->GetAudioTracks().empty() && stream->GetVideoTracks().empty()) {
|
||
remote_streams_->RemoveStream(stream);
|
||
removed_streams->push_back(stream);
|
||
}
|
||
}
|
||
}
|
||
|
||
RTCError PeerConnection::UpdateTransceiversAndDataChannels(
|
||
cricket::ContentSource source,
|
||
const SessionDescriptionInterface& new_session,
|
||
const SessionDescriptionInterface* old_local_description,
|
||
const SessionDescriptionInterface* old_remote_description) {
|
||
RTC_DCHECK(IsUnifiedPlan());
|
||
|
||
const cricket::ContentGroup* bundle_group = nullptr;
|
||
if (new_session.GetType() == SdpType::kOffer) {
|
||
auto bundle_group_or_error =
|
||
GetEarlyBundleGroup(*new_session.description());
|
||
if (!bundle_group_or_error.ok()) {
|
||
return bundle_group_or_error.MoveError();
|
||
}
|
||
bundle_group = bundle_group_or_error.MoveValue();
|
||
}
|
||
|
||
const ContentInfos& new_contents = new_session.description()->contents();
|
||
for (size_t i = 0; i < new_contents.size(); ++i) {
|
||
const cricket::ContentInfo& new_content = new_contents[i];
|
||
cricket::MediaType media_type = new_content.media_description()->type();
|
||
seen_mids_.insert(new_content.name);
|
||
if (media_type == cricket::MEDIA_TYPE_AUDIO ||
|
||
media_type == cricket::MEDIA_TYPE_VIDEO) {
|
||
const cricket::ContentInfo* old_local_content = nullptr;
|
||
if (old_local_description &&
|
||
i < old_local_description->description()->contents().size()) {
|
||
old_local_content =
|
||
&old_local_description->description()->contents()[i];
|
||
}
|
||
const cricket::ContentInfo* old_remote_content = nullptr;
|
||
if (old_remote_description &&
|
||
i < old_remote_description->description()->contents().size()) {
|
||
old_remote_content =
|
||
&old_remote_description->description()->contents()[i];
|
||
}
|
||
auto transceiver_or_error =
|
||
AssociateTransceiver(source, new_session.GetType(), i, new_content,
|
||
old_local_content, old_remote_content);
|
||
if (!transceiver_or_error.ok()) {
|
||
return transceiver_or_error.MoveError();
|
||
}
|
||
auto transceiver = transceiver_or_error.MoveValue();
|
||
RTCError error =
|
||
UpdateTransceiverChannel(transceiver, new_content, bundle_group);
|
||
if (!error.ok()) {
|
||
return error;
|
||
}
|
||
} else if (media_type == cricket::MEDIA_TYPE_DATA) {
|
||
if (GetDataMid() && new_content.name != *GetDataMid()) {
|
||
// Ignore all but the first data section.
|
||
RTC_LOG(LS_INFO) << "Ignoring data media section with MID="
|
||
<< new_content.name;
|
||
continue;
|
||
}
|
||
RTCError error = UpdateDataChannel(source, new_content, bundle_group);
|
||
if (!error.ok()) {
|
||
return error;
|
||
}
|
||
} else {
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
|
||
"Unknown section type.");
|
||
}
|
||
}
|
||
|
||
return RTCError::OK();
|
||
}
|
||
|
||
RTCError PeerConnection::UpdateTransceiverChannel(
|
||
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
||
transceiver,
|
||
const cricket::ContentInfo& content,
|
||
const cricket::ContentGroup* bundle_group) {
|
||
RTC_DCHECK(IsUnifiedPlan());
|
||
RTC_DCHECK(transceiver);
|
||
cricket::BaseChannel* channel = transceiver->internal()->channel();
|
||
if (content.rejected) {
|
||
if (channel) {
|
||
transceiver->internal()->SetChannel(nullptr);
|
||
DestroyBaseChannel(channel);
|
||
}
|
||
} else {
|
||
if (!channel) {
|
||
if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
|
||
channel = CreateVoiceChannel(content.name);
|
||
} else {
|
||
RTC_DCHECK_EQ(cricket::MEDIA_TYPE_VIDEO, transceiver->media_type());
|
||
channel = CreateVideoChannel(content.name);
|
||
}
|
||
if (!channel) {
|
||
LOG_AND_RETURN_ERROR(
|
||
RTCErrorType::INTERNAL_ERROR,
|
||
"Failed to create channel for mid=" + content.name);
|
||
}
|
||
transceiver->internal()->SetChannel(channel);
|
||
}
|
||
}
|
||
return RTCError::OK();
|
||
}
|
||
|
||
RTCError PeerConnection::UpdateDataChannel(
|
||
cricket::ContentSource source,
|
||
const cricket::ContentInfo& content,
|
||
const cricket::ContentGroup* bundle_group) {
|
||
if (data_channel_type_ == cricket::DCT_NONE) {
|
||
// If data channels are disabled, ignore this media section. CreateAnswer
|
||
// will take care of rejecting it.
|
||
return RTCError::OK();
|
||
}
|
||
if (content.rejected) {
|
||
DestroyDataChannel();
|
||
} else {
|
||
if (!rtp_data_channel_ && !sctp_transport_) {
|
||
if (!CreateDataChannel(content.name)) {
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
|
||
"Failed to create data channel.");
|
||
}
|
||
}
|
||
if (source == cricket::CS_REMOTE) {
|
||
const MediaContentDescription* data_desc = content.media_description();
|
||
if (data_desc && cricket::IsRtpProtocol(data_desc->protocol())) {
|
||
UpdateRemoteRtpDataChannels(GetActiveStreams(data_desc));
|
||
}
|
||
}
|
||
}
|
||
return RTCError::OK();
|
||
}
|
||
|
||
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
|
||
PeerConnection::AssociateTransceiver(cricket::ContentSource source,
|
||
SdpType type,
|
||
size_t mline_index,
|
||
const ContentInfo& content,
|
||
const ContentInfo* old_local_content,
|
||
const ContentInfo* old_remote_content) {
|
||
RTC_DCHECK(IsUnifiedPlan());
|
||
// If this is an offer then the m= section might be recycled. If the m=
|
||
// section is being recycled (defined as: rejected in the current local or
|
||
// remote description and not rejected in new description), dissociate the
|
||
// currently associated RtpTransceiver by setting its mid property to null,
|
||
// and discard the mapping between the transceiver and its m= section index.
|
||
if (IsMediaSectionBeingRecycled(type, content, old_local_content,
|
||
old_remote_content)) {
|
||
// We want to dissociate the transceiver that has the rejected mid.
|
||
const std::string& old_mid =
|
||
(old_local_content && old_local_content->rejected)
|
||
? old_local_content->name
|
||
: old_remote_content->name;
|
||
auto old_transceiver = GetAssociatedTransceiver(old_mid);
|
||
if (old_transceiver) {
|
||
RTC_LOG(LS_INFO) << "Dissociating transceiver for MID=" << old_mid
|
||
<< " since the media section is being recycled.";
|
||
old_transceiver->internal()->set_mid(absl::nullopt);
|
||
old_transceiver->internal()->set_mline_index(absl::nullopt);
|
||
}
|
||
}
|
||
const MediaContentDescription* media_desc = content.media_description();
|
||
auto transceiver = GetAssociatedTransceiver(content.name);
|
||
if (source == cricket::CS_LOCAL) {
|
||
// Find the RtpTransceiver that corresponds to this m= section, using the
|
||
// mapping between transceivers and m= section indices established when
|
||
// creating the offer.
|
||
if (!transceiver) {
|
||
transceiver = GetTransceiverByMLineIndex(mline_index);
|
||
}
|
||
if (!transceiver) {
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
||
"Unknown transceiver");
|
||
}
|
||
} else {
|
||
RTC_DCHECK_EQ(source, cricket::CS_REMOTE);
|
||
// If the m= section is sendrecv or recvonly, and there are RtpTransceivers
|
||
// of the same type...
|
||
if (!transceiver &&
|
||
RtpTransceiverDirectionHasRecv(media_desc->direction())) {
|
||
transceiver = FindAvailableTransceiverToReceive(media_desc->type());
|
||
}
|
||
// If no RtpTransceiver was found in the previous step, create one with a
|
||
// recvonly direction.
|
||
if (!transceiver) {
|
||
RTC_LOG(LS_INFO) << "Adding "
|
||
<< cricket::MediaTypeToString(media_desc->type())
|
||
<< " transceiver for MID=" << content.name
|
||
<< " at i=" << mline_index
|
||
<< " in response to the remote description.";
|
||
std::string sender_id = rtc::CreateRandomUuid();
|
||
auto sender = CreateSender(media_desc->type(), sender_id, nullptr, {});
|
||
std::string receiver_id;
|
||
if (!media_desc->streams().empty()) {
|
||
receiver_id = media_desc->streams()[0].id;
|
||
} else {
|
||
receiver_id = rtc::CreateRandomUuid();
|
||
}
|
||
auto receiver = CreateReceiver(media_desc->type(), receiver_id);
|
||
transceiver = CreateAndAddTransceiver(sender, receiver);
|
||
transceiver->internal()->set_direction(
|
||
RtpTransceiverDirection::kRecvOnly);
|
||
}
|
||
}
|
||
RTC_DCHECK(transceiver);
|
||
if (transceiver->media_type() != media_desc->type()) {
|
||
LOG_AND_RETURN_ERROR(
|
||
RTCErrorType::INVALID_PARAMETER,
|
||
"Transceiver type does not match media description type.");
|
||
}
|
||
// Associate the found or created RtpTransceiver with the m= section by
|
||
// setting the value of the RtpTransceiver's mid property to the MID of the m=
|
||
// section, and establish a mapping between the transceiver and the index of
|
||
// the m= section.
|
||
transceiver->internal()->set_mid(content.name);
|
||
transceiver->internal()->set_mline_index(mline_index);
|
||
return std::move(transceiver);
|
||
}
|
||
|
||
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
||
PeerConnection::GetAssociatedTransceiver(const std::string& mid) const {
|
||
RTC_DCHECK(IsUnifiedPlan());
|
||
for (auto transceiver : transceivers_) {
|
||
if (transceiver->mid() == mid) {
|
||
return transceiver;
|
||
}
|
||
}
|
||
return nullptr;
|
||
}
|
||
|
||
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
||
PeerConnection::GetTransceiverByMLineIndex(size_t mline_index) const {
|
||
RTC_DCHECK(IsUnifiedPlan());
|
||
for (auto transceiver : transceivers_) {
|
||
if (transceiver->internal()->mline_index() == mline_index) {
|
||
return transceiver;
|
||
}
|
||
}
|
||
return nullptr;
|
||
}
|
||
|
||
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
||
PeerConnection::FindAvailableTransceiverToReceive(
|
||
cricket::MediaType media_type) const {
|
||
RTC_DCHECK(IsUnifiedPlan());
|
||
// From JSEP section 5.10 (Applying a Remote Description):
|
||
// If the m= section is sendrecv or recvonly, and there are RtpTransceivers of
|
||
// the same type that were added to the PeerConnection by addTrack and are not
|
||
// associated with any m= section and are not stopped, find the first such
|
||
// RtpTransceiver.
|
||
for (auto transceiver : transceivers_) {
|
||
if (transceiver->media_type() == media_type &&
|
||
transceiver->internal()->created_by_addtrack() && !transceiver->mid() &&
|
||
!transceiver->stopped()) {
|
||
return transceiver;
|
||
}
|
||
}
|
||
return nullptr;
|
||
}
|
||
|
||
const cricket::ContentInfo* PeerConnection::FindMediaSectionForTransceiver(
|
||
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
||
transceiver,
|
||
const SessionDescriptionInterface* sdesc) const {
|
||
RTC_DCHECK(transceiver);
|
||
RTC_DCHECK(sdesc);
|
||
if (IsUnifiedPlan()) {
|
||
if (!transceiver->internal()->mid()) {
|
||
// This transceiver is not associated with a media section yet.
|
||
return nullptr;
|
||
}
|
||
return sdesc->description()->GetContentByName(
|
||
*transceiver->internal()->mid());
|
||
} else {
|
||
// Plan B only allows at most one audio and one video section, so use the
|
||
// first media section of that type.
|
||
return cricket::GetFirstMediaContent(sdesc->description()->contents(),
|
||
transceiver->media_type());
|
||
}
|
||
}
|
||
|
||
PeerConnectionInterface::RTCConfiguration PeerConnection::GetConfiguration() {
|
||
return configuration_;
|
||
}
|
||
|
||
bool PeerConnection::SetConfiguration(const RTCConfiguration& configuration,
|
||
RTCError* error) {
|
||
TRACE_EVENT0("webrtc", "PeerConnection::SetConfiguration");
|
||
if (IsClosed()) {
|
||
RTC_LOG(LS_ERROR) << "SetConfiguration: PeerConnection is closed.";
|
||
return SafeSetError(RTCErrorType::INVALID_STATE, error);
|
||
}
|
||
|
||
// According to JSEP, after setLocalDescription, changing the candidate pool
|
||
// size is not allowed, and changing the set of ICE servers will not result
|
||
// in new candidates being gathered.
|
||
if (local_description() && configuration.ice_candidate_pool_size !=
|
||
configuration_.ice_candidate_pool_size) {
|
||
RTC_LOG(LS_ERROR) << "Can't change candidate pool size after calling "
|
||
"SetLocalDescription.";
|
||
return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error);
|
||
}
|
||
|
||
// The simplest (and most future-compatible) way to tell if the config was
|
||
// modified in an invalid way is to copy each property we do support
|
||
// modifying, then use operator==. There are far more properties we don't
|
||
// support modifying than those we do, and more could be added.
|
||
RTCConfiguration modified_config = configuration_;
|
||
modified_config.servers = configuration.servers;
|
||
modified_config.type = configuration.type;
|
||
modified_config.ice_candidate_pool_size =
|
||
configuration.ice_candidate_pool_size;
|
||
modified_config.prune_turn_ports = configuration.prune_turn_ports;
|
||
modified_config.ice_check_min_interval = configuration.ice_check_min_interval;
|
||
modified_config.ice_check_interval_strong_connectivity =
|
||
configuration.ice_check_interval_strong_connectivity;
|
||
modified_config.ice_check_interval_weak_connectivity =
|
||
configuration.ice_check_interval_weak_connectivity;
|
||
modified_config.ice_unwritable_timeout = configuration.ice_unwritable_timeout;
|
||
modified_config.ice_unwritable_min_checks =
|
||
configuration.ice_unwritable_min_checks;
|
||
modified_config.stun_candidate_keepalive_interval =
|
||
configuration.stun_candidate_keepalive_interval;
|
||
modified_config.turn_customizer = configuration.turn_customizer;
|
||
modified_config.network_preference = configuration.network_preference;
|
||
modified_config.active_reset_srtp_params =
|
||
configuration.active_reset_srtp_params;
|
||
if (configuration != modified_config) {
|
||
RTC_LOG(LS_ERROR) << "Modifying the configuration in an unsupported way.";
|
||
return SafeSetError(RTCErrorType::INVALID_MODIFICATION, error);
|
||
}
|
||
|
||
// Validate the modified configuration.
|
||
RTCError validate_error = ValidateConfiguration(modified_config);
|
||
if (!validate_error.ok()) {
|
||
return SafeSetError(std::move(validate_error), error);
|
||
}
|
||
|
||
// Note that this isn't possible through chromium, since it's an unsigned
|
||
// short in WebIDL.
|
||
if (configuration.ice_candidate_pool_size < 0 ||
|
||
configuration.ice_candidate_pool_size > static_cast<int>(UINT16_MAX)) {
|
||
return SafeSetError(RTCErrorType::INVALID_RANGE, error);
|
||
}
|
||
|
||
// Parse ICE servers before hopping to network thread.
|
||
cricket::ServerAddresses stun_servers;
|
||
std::vector<cricket::RelayServerConfig> turn_servers;
|
||
RTCErrorType parse_error =
|
||
ParseIceServers(configuration.servers, &stun_servers, &turn_servers);
|
||
if (parse_error != RTCErrorType::NONE) {
|
||
return SafeSetError(parse_error, error);
|
||
}
|
||
// Note if STUN or TURN servers were supplied.
|
||
if (!stun_servers.empty()) {
|
||
NoteUsageEvent(UsageEvent::STUN_SERVER_ADDED);
|
||
}
|
||
if (!turn_servers.empty()) {
|
||
NoteUsageEvent(UsageEvent::TURN_SERVER_ADDED);
|
||
}
|
||
|
||
// In theory this shouldn't fail.
|
||
if (!network_thread()->Invoke<bool>(
|
||
RTC_FROM_HERE,
|
||
rtc::Bind(&PeerConnection::ReconfigurePortAllocator_n, this,
|
||
stun_servers, turn_servers, modified_config.type,
|
||
modified_config.ice_candidate_pool_size,
|
||
modified_config.prune_turn_ports,
|
||
modified_config.turn_customizer,
|
||
modified_config.stun_candidate_keepalive_interval))) {
|
||
RTC_LOG(LS_ERROR) << "Failed to apply configuration to PortAllocator.";
|
||
return SafeSetError(RTCErrorType::INTERNAL_ERROR, error);
|
||
}
|
||
|
||
// As described in JSEP, calling setConfiguration with new ICE servers or
|
||
// candidate policy must set a "needs-ice-restart" bit so that the next offer
|
||
// triggers an ICE restart which will pick up the changes.
|
||
if (modified_config.servers != configuration_.servers ||
|
||
modified_config.type != configuration_.type ||
|
||
modified_config.prune_turn_ports != configuration_.prune_turn_ports) {
|
||
transport_controller_->SetNeedsIceRestartFlag();
|
||
}
|
||
|
||
transport_controller_->SetIceConfig(ParseIceConfig(modified_config));
|
||
|
||
if (configuration_.active_reset_srtp_params !=
|
||
modified_config.active_reset_srtp_params) {
|
||
transport_controller_->SetActiveResetSrtpParams(
|
||
modified_config.active_reset_srtp_params);
|
||
}
|
||
|
||
configuration_ = modified_config;
|
||
return SafeSetError(RTCErrorType::NONE, error);
|
||
}
|
||
|
||
bool PeerConnection::AddIceCandidate(
|
||
const IceCandidateInterface* ice_candidate) {
|
||
TRACE_EVENT0("webrtc", "PeerConnection::AddIceCandidate");
|
||
if (IsClosed()) {
|
||
RTC_LOG(LS_ERROR) << "AddIceCandidate: PeerConnection is closed.";
|
||
NoteAddIceCandidateResult(kAddIceCandidateFailClosed);
|
||
return false;
|
||
}
|
||
|
||
if (!remote_description()) {
|
||
RTC_LOG(LS_ERROR) << "AddIceCandidate: ICE candidates can't be added "
|
||
"without any remote session description.";
|
||
NoteAddIceCandidateResult(kAddIceCandidateFailNoRemoteDescription);
|
||
return false;
|
||
}
|
||
|
||
if (!ice_candidate) {
|
||
RTC_LOG(LS_ERROR) << "AddIceCandidate: Candidate is null.";
|
||
NoteAddIceCandidateResult(kAddIceCandidateFailNullCandidate);
|
||
return false;
|
||
}
|
||
|
||
bool valid = false;
|
||
bool ready = ReadyToUseRemoteCandidate(ice_candidate, nullptr, &valid);
|
||
if (!valid) {
|
||
NoteAddIceCandidateResult(kAddIceCandidateFailNotValid);
|
||
return false;
|
||
}
|
||
|
||
// Add this candidate to the remote session description.
|
||
if (!mutable_remote_description()->AddCandidate(ice_candidate)) {
|
||
RTC_LOG(LS_ERROR) << "AddIceCandidate: Candidate cannot be used.";
|
||
NoteAddIceCandidateResult(kAddIceCandidateFailInAddition);
|
||
return false;
|
||
}
|
||
|
||
if (ready) {
|
||
bool result = UseCandidate(ice_candidate);
|
||
if (result) {
|
||
NoteUsageEvent(UsageEvent::REMOTE_CANDIDATE_ADDED);
|
||
NoteAddIceCandidateResult(kAddIceCandidateSuccess);
|
||
} else {
|
||
NoteAddIceCandidateResult(kAddIceCandidateFailNotUsable);
|
||
}
|
||
return result;
|
||
} else {
|
||
RTC_LOG(LS_INFO) << "AddIceCandidate: Not ready to use candidate.";
|
||
NoteAddIceCandidateResult(kAddIceCandidateFailNotReady);
|
||
return true;
|
||
}
|
||
}
|
||
|
||
bool PeerConnection::RemoveIceCandidates(
|
||
const std::vector<cricket::Candidate>& candidates) {
|
||
TRACE_EVENT0("webrtc", "PeerConnection::RemoveIceCandidates");
|
||
if (IsClosed()) {
|
||
RTC_LOG(LS_ERROR) << "RemoveIceCandidates: PeerConnection is closed.";
|
||
return false;
|
||
}
|
||
|
||
if (!remote_description()) {
|
||
RTC_LOG(LS_ERROR) << "RemoveIceCandidates: ICE candidates can't be removed "
|
||
"without any remote session description.";
|
||
return false;
|
||
}
|
||
|
||
if (candidates.empty()) {
|
||
RTC_LOG(LS_ERROR) << "RemoveIceCandidates: candidates are empty.";
|
||
return false;
|
||
}
|
||
|
||
size_t number_removed =
|
||
mutable_remote_description()->RemoveCandidates(candidates);
|
||
if (number_removed != candidates.size()) {
|
||
RTC_LOG(LS_ERROR)
|
||
<< "RemoveIceCandidates: Failed to remove candidates. Requested "
|
||
<< candidates.size() << " but only " << number_removed
|
||
<< " are removed.";
|
||
}
|
||
|
||
// Remove the candidates from the transport controller.
|
||
RTCError error = transport_controller_->RemoveRemoteCandidates(candidates);
|
||
if (!error.ok()) {
|
||
RTC_LOG(LS_ERROR)
|
||
<< "RemoveIceCandidates: Error when removing remote candidates: "
|
||
<< error.message();
|
||
}
|
||
return true;
|
||
}
|
||
|
||
RTCError PeerConnection::SetBitrate(const BitrateSettings& bitrate) {
|
||
if (!worker_thread()->IsCurrent()) {
|
||
return worker_thread()->Invoke<RTCError>(
|
||
RTC_FROM_HERE, [&]() { return SetBitrate(bitrate); });
|
||
}
|
||
|
||
const bool has_min = bitrate.min_bitrate_bps.has_value();
|
||
const bool has_start = bitrate.start_bitrate_bps.has_value();
|
||
const bool has_max = bitrate.max_bitrate_bps.has_value();
|
||
if (has_min && *bitrate.min_bitrate_bps < 0) {
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
||
"min_bitrate_bps <= 0");
|
||
}
|
||
if (has_start) {
|
||
if (has_min && *bitrate.start_bitrate_bps < *bitrate.min_bitrate_bps) {
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
||
"start_bitrate_bps < min_bitrate_bps");
|
||
} else if (*bitrate.start_bitrate_bps < 0) {
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
||
"curent_bitrate_bps < 0");
|
||
}
|
||
}
|
||
if (has_max) {
|
||
if (has_start && *bitrate.max_bitrate_bps < *bitrate.start_bitrate_bps) {
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
||
"max_bitrate_bps < start_bitrate_bps");
|
||
} else if (has_min && *bitrate.max_bitrate_bps < *bitrate.min_bitrate_bps) {
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
||
"max_bitrate_bps < min_bitrate_bps");
|
||
} else if (*bitrate.max_bitrate_bps < 0) {
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
||
"max_bitrate_bps < 0");
|
||
}
|
||
}
|
||
|
||
RTC_DCHECK(call_.get());
|
||
call_->GetTransportControllerSend()->SetClientBitratePreferences(bitrate);
|
||
|
||
return RTCError::OK();
|
||
}
|
||
|
||
void PeerConnection::SetBitrateAllocationStrategy(
|
||
std::unique_ptr<rtc::BitrateAllocationStrategy>
|
||
bitrate_allocation_strategy) {
|
||
rtc::Thread* worker_thread = factory_->worker_thread();
|
||
if (!worker_thread->IsCurrent()) {
|
||
rtc::BitrateAllocationStrategy* strategy_raw =
|
||
bitrate_allocation_strategy.release();
|
||
auto functor = [this, strategy_raw]() {
|
||
call_->SetBitrateAllocationStrategy(
|
||
absl::WrapUnique<rtc::BitrateAllocationStrategy>(strategy_raw));
|
||
};
|
||
worker_thread->Invoke<void>(RTC_FROM_HERE, functor);
|
||
return;
|
||
}
|
||
RTC_DCHECK(call_.get());
|
||
call_->SetBitrateAllocationStrategy(std::move(bitrate_allocation_strategy));
|
||
}
|
||
|
||
void PeerConnection::SetAudioPlayout(bool playout) {
|
||
if (!worker_thread()->IsCurrent()) {
|
||
worker_thread()->Invoke<void>(
|
||
RTC_FROM_HERE,
|
||
rtc::Bind(&PeerConnection::SetAudioPlayout, this, playout));
|
||
return;
|
||
}
|
||
auto audio_state =
|
||
factory_->channel_manager()->media_engine()->GetAudioState();
|
||
audio_state->SetPlayout(playout);
|
||
}
|
||
|
||
void PeerConnection::SetAudioRecording(bool recording) {
|
||
if (!worker_thread()->IsCurrent()) {
|
||
worker_thread()->Invoke<void>(
|
||
RTC_FROM_HERE,
|
||
rtc::Bind(&PeerConnection::SetAudioRecording, this, recording));
|
||
return;
|
||
}
|
||
auto audio_state =
|
||
factory_->channel_manager()->media_engine()->GetAudioState();
|
||
audio_state->SetRecording(recording);
|
||
}
|
||
|
||
std::unique_ptr<rtc::SSLCertificate>
|
||
PeerConnection::GetRemoteAudioSSLCertificate() {
|
||
std::unique_ptr<rtc::SSLCertChain> chain = GetRemoteAudioSSLCertChain();
|
||
if (!chain || !chain->GetSize()) {
|
||
return nullptr;
|
||
}
|
||
return chain->Get(0).GetUniqueReference();
|
||
}
|
||
|
||
std::unique_ptr<rtc::SSLCertChain>
|
||
PeerConnection::GetRemoteAudioSSLCertChain() {
|
||
auto audio_transceiver = GetFirstAudioTransceiver();
|
||
if (!audio_transceiver || !audio_transceiver->internal()->channel()) {
|
||
return nullptr;
|
||
}
|
||
return transport_controller_->GetRemoteSSLCertChain(
|
||
audio_transceiver->internal()->channel()->transport_name());
|
||
}
|
||
|
||
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
||
PeerConnection::GetFirstAudioTransceiver() const {
|
||
for (auto transceiver : transceivers_) {
|
||
if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
|
||
return transceiver;
|
||
}
|
||
}
|
||
return nullptr;
|
||
}
|
||
|
||
bool PeerConnection::StartRtcEventLog(rtc::PlatformFile file,
|
||
int64_t max_size_bytes) {
|
||
// TODO(eladalon): It would be better to not allow negative values into PC.
|
||
const size_t max_size = (max_size_bytes < 0)
|
||
? RtcEventLog::kUnlimitedOutput
|
||
: rtc::saturated_cast<size_t>(max_size_bytes);
|
||
return StartRtcEventLog(
|
||
absl::make_unique<RtcEventLogOutputFile>(file, max_size),
|
||
webrtc::RtcEventLog::kImmediateOutput);
|
||
}
|
||
|
||
bool PeerConnection::StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output,
|
||
int64_t output_period_ms) {
|
||
// TODO(eladalon): In C++14, this can be done with a lambda.
|
||
struct Functor {
|
||
bool operator()() {
|
||
return pc->StartRtcEventLog_w(std::move(output), output_period_ms);
|
||
}
|
||
PeerConnection* const pc;
|
||
std::unique_ptr<RtcEventLogOutput> output;
|
||
const int64_t output_period_ms;
|
||
};
|
||
return worker_thread()->Invoke<bool>(
|
||
RTC_FROM_HERE, Functor{this, std::move(output), output_period_ms});
|
||
}
|
||
|
||
void PeerConnection::StopRtcEventLog() {
|
||
worker_thread()->Invoke<void>(
|
||
RTC_FROM_HERE, rtc::Bind(&PeerConnection::StopRtcEventLog_w, this));
|
||
}
|
||
|
||
const SessionDescriptionInterface* PeerConnection::local_description() const {
|
||
return pending_local_description_ ? pending_local_description_.get()
|
||
: current_local_description_.get();
|
||
}
|
||
|
||
const SessionDescriptionInterface* PeerConnection::remote_description() const {
|
||
return pending_remote_description_ ? pending_remote_description_.get()
|
||
: current_remote_description_.get();
|
||
}
|
||
|
||
const SessionDescriptionInterface* PeerConnection::current_local_description()
|
||
const {
|
||
return current_local_description_.get();
|
||
}
|
||
|
||
const SessionDescriptionInterface* PeerConnection::current_remote_description()
|
||
const {
|
||
return current_remote_description_.get();
|
||
}
|
||
|
||
const SessionDescriptionInterface* PeerConnection::pending_local_description()
|
||
const {
|
||
return pending_local_description_.get();
|
||
}
|
||
|
||
const SessionDescriptionInterface* PeerConnection::pending_remote_description()
|
||
const {
|
||
return pending_remote_description_.get();
|
||
}
|
||
|
||
void PeerConnection::Close() {
|
||
TRACE_EVENT0("webrtc", "PeerConnection::Close");
|
||
// Update stats here so that we have the most recent stats for tracks and
|
||
// streams before the channels are closed.
|
||
stats_->UpdateStats(kStatsOutputLevelStandard);
|
||
|
||
ChangeSignalingState(PeerConnectionInterface::kClosed);
|
||
NoteUsageEvent(UsageEvent::CLOSE_CALLED);
|
||
|
||
for (auto transceiver : transceivers_) {
|
||
transceiver->Stop();
|
||
}
|
||
|
||
// Ensure that all asynchronous stats requests are completed before destroying
|
||
// the transport controller below.
|
||
if (stats_collector_) {
|
||
stats_collector_->WaitForPendingRequest();
|
||
}
|
||
|
||
// Don't destroy BaseChannels until after stats has been cleaned up so that
|
||
// the last stats request can still read from the channels.
|
||
DestroyAllChannels();
|
||
|
||
// The event log is used in the transport controller, which must be outlived
|
||
// by the former. CreateOffer by the peer connection is implemented
|
||
// asynchronously and if the peer connection is closed without resetting the
|
||
// WebRTC session description factory, the session description factory would
|
||
// call the transport controller.
|
||
webrtc_session_desc_factory_.reset();
|
||
transport_controller_.reset();
|
||
|
||
network_thread()->Invoke<void>(
|
||
RTC_FROM_HERE, rtc::Bind(&cricket::PortAllocator::DiscardCandidatePool,
|
||
port_allocator_.get()));
|
||
|
||
worker_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
|
||
call_.reset();
|
||
// The event log must outlive call (and any other object that uses it).
|
||
event_log_.reset();
|
||
});
|
||
ReportUsagePattern();
|
||
// The .h file says that observer can be discarded after close() returns.
|
||
// Make sure this is true.
|
||
observer_ = nullptr;
|
||
}
|
||
|
||
void PeerConnection::OnMessage(rtc::Message* msg) {
|
||
switch (msg->message_id) {
|
||
case MSG_SET_SESSIONDESCRIPTION_SUCCESS: {
|
||
SetSessionDescriptionMsg* param =
|
||
static_cast<SetSessionDescriptionMsg*>(msg->pdata);
|
||
param->observer->OnSuccess();
|
||
delete param;
|
||
break;
|
||
}
|
||
case MSG_SET_SESSIONDESCRIPTION_FAILED: {
|
||
SetSessionDescriptionMsg* param =
|
||
static_cast<SetSessionDescriptionMsg*>(msg->pdata);
|
||
param->observer->OnFailure(std::move(param->error));
|
||
delete param;
|
||
break;
|
||
}
|
||
case MSG_CREATE_SESSIONDESCRIPTION_FAILED: {
|
||
CreateSessionDescriptionMsg* param =
|
||
static_cast<CreateSessionDescriptionMsg*>(msg->pdata);
|
||
param->observer->OnFailure(std::move(param->error));
|
||
delete param;
|
||
break;
|
||
}
|
||
case MSG_GETSTATS: {
|
||
GetStatsMsg* param = static_cast<GetStatsMsg*>(msg->pdata);
|
||
StatsReports reports;
|
||
stats_->GetStats(param->track, &reports);
|
||
param->observer->OnComplete(reports);
|
||
delete param;
|
||
break;
|
||
}
|
||
case MSG_FREE_DATACHANNELS: {
|
||
sctp_data_channels_to_free_.clear();
|
||
break;
|
||
}
|
||
case MSG_REPORT_USAGE_PATTERN: {
|
||
ReportUsagePattern();
|
||
break;
|
||
}
|
||
default:
|
||
RTC_NOTREACHED() << "Not implemented";
|
||
break;
|
||
}
|
||
}
|
||
|
||
cricket::VoiceMediaChannel* PeerConnection::voice_media_channel() const {
|
||
RTC_DCHECK(!IsUnifiedPlan());
|
||
auto* voice_channel = static_cast<cricket::VoiceChannel*>(
|
||
GetAudioTransceiver()->internal()->channel());
|
||
if (voice_channel) {
|
||
return voice_channel->media_channel();
|
||
} else {
|
||
return nullptr;
|
||
}
|
||
}
|
||
|
||
cricket::VideoMediaChannel* PeerConnection::video_media_channel() const {
|
||
RTC_DCHECK(!IsUnifiedPlan());
|
||
auto* video_channel = static_cast<cricket::VideoChannel*>(
|
||
GetVideoTransceiver()->internal()->channel());
|
||
if (video_channel) {
|
||
return video_channel->media_channel();
|
||
} else {
|
||
return nullptr;
|
||
}
|
||
}
|
||
|
||
void PeerConnection::CreateAudioReceiver(
|
||
MediaStreamInterface* stream,
|
||
const RtpSenderInfo& remote_sender_info) {
|
||
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams;
|
||
streams.push_back(rtc::scoped_refptr<MediaStreamInterface>(stream));
|
||
// TODO(https://crbug.com/webrtc/9480): When we remove remote_streams(), use
|
||
// the constructor taking stream IDs instead.
|
||
auto* audio_receiver = new AudioRtpReceiver(
|
||
worker_thread(), remote_sender_info.sender_id, streams);
|
||
audio_receiver->SetVoiceMediaChannel(voice_media_channel());
|
||
audio_receiver->SetupMediaChannel(remote_sender_info.first_ssrc);
|
||
auto receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
|
||
signaling_thread(), audio_receiver);
|
||
GetAudioTransceiver()->internal()->AddReceiver(receiver);
|
||
Observer()->OnAddTrack(receiver, std::move(streams));
|
||
NoteUsageEvent(UsageEvent::AUDIO_ADDED);
|
||
}
|
||
|
||
void PeerConnection::CreateVideoReceiver(
|
||
MediaStreamInterface* stream,
|
||
const RtpSenderInfo& remote_sender_info) {
|
||
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams;
|
||
streams.push_back(rtc::scoped_refptr<MediaStreamInterface>(stream));
|
||
// TODO(https://crbug.com/webrtc/9480): When we remove remote_streams(), use
|
||
// the constructor taking stream IDs instead.
|
||
auto* video_receiver = new VideoRtpReceiver(
|
||
worker_thread(), remote_sender_info.sender_id, streams);
|
||
video_receiver->SetVideoMediaChannel(video_media_channel());
|
||
video_receiver->SetupMediaChannel(remote_sender_info.first_ssrc);
|
||
auto receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
|
||
signaling_thread(), video_receiver);
|
||
GetVideoTransceiver()->internal()->AddReceiver(receiver);
|
||
Observer()->OnAddTrack(receiver, std::move(streams));
|
||
NoteUsageEvent(UsageEvent::VIDEO_ADDED);
|
||
}
|
||
|
||
// TODO(deadbeef): Keep RtpReceivers around even if track goes away in remote
|
||
// description.
|
||
rtc::scoped_refptr<RtpReceiverInterface> PeerConnection::RemoveAndStopReceiver(
|
||
const RtpSenderInfo& remote_sender_info) {
|
||
auto receiver = FindReceiverById(remote_sender_info.sender_id);
|
||
if (!receiver) {
|
||
RTC_LOG(LS_WARNING) << "RtpReceiver for track with id "
|
||
<< remote_sender_info.sender_id << " doesn't exist.";
|
||
return nullptr;
|
||
}
|
||
if (receiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
|
||
GetAudioTransceiver()->internal()->RemoveReceiver(receiver);
|
||
} else {
|
||
GetVideoTransceiver()->internal()->RemoveReceiver(receiver);
|
||
}
|
||
return receiver;
|
||
}
|
||
|
||
void PeerConnection::AddAudioTrack(AudioTrackInterface* track,
|
||
MediaStreamInterface* stream) {
|
||
RTC_DCHECK(!IsClosed());
|
||
RTC_DCHECK(track);
|
||
RTC_DCHECK(stream);
|
||
auto sender = FindSenderForTrack(track);
|
||
if (sender) {
|
||
// We already have a sender for this track, so just change the stream_id
|
||
// so that it's correct in the next call to CreateOffer.
|
||
sender->internal()->set_stream_ids({stream->id()});
|
||
return;
|
||
}
|
||
|
||
// Normal case; we've never seen this track before.
|
||
auto new_sender = CreateSender(cricket::MEDIA_TYPE_AUDIO, track->id(), track,
|
||
{stream->id()});
|
||
new_sender->internal()->SetVoiceMediaChannel(voice_media_channel());
|
||
GetAudioTransceiver()->internal()->AddSender(new_sender);
|
||
// If the sender has already been configured in SDP, we call SetSsrc,
|
||
// which will connect the sender to the underlying transport. This can
|
||
// occur if a local session description that contains the ID of the sender
|
||
// is set before AddStream is called. It can also occur if the local
|
||
// session description is not changed and RemoveStream is called, and
|
||
// later AddStream is called again with the same stream.
|
||
const RtpSenderInfo* sender_info =
|
||
FindSenderInfo(local_audio_sender_infos_, stream->id(), track->id());
|
||
if (sender_info) {
|
||
new_sender->internal()->SetSsrc(sender_info->first_ssrc);
|
||
}
|
||
}
|
||
|
||
// TODO(deadbeef): Don't destroy RtpSenders here; they should be kept around
|
||
// indefinitely, when we have unified plan SDP.
|
||
void PeerConnection::RemoveAudioTrack(AudioTrackInterface* track,
|
||
MediaStreamInterface* stream) {
|
||
RTC_DCHECK(!IsClosed());
|
||
auto sender = FindSenderForTrack(track);
|
||
if (!sender) {
|
||
RTC_LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
|
||
<< " doesn't exist.";
|
||
return;
|
||
}
|
||
GetAudioTransceiver()->internal()->RemoveSender(sender);
|
||
}
|
||
|
||
void PeerConnection::AddVideoTrack(VideoTrackInterface* track,
|
||
MediaStreamInterface* stream) {
|
||
RTC_DCHECK(!IsClosed());
|
||
RTC_DCHECK(track);
|
||
RTC_DCHECK(stream);
|
||
auto sender = FindSenderForTrack(track);
|
||
if (sender) {
|
||
// We already have a sender for this track, so just change the stream_id
|
||
// so that it's correct in the next call to CreateOffer.
|
||
sender->internal()->set_stream_ids({stream->id()});
|
||
return;
|
||
}
|
||
|
||
// Normal case; we've never seen this track before.
|
||
auto new_sender = CreateSender(cricket::MEDIA_TYPE_VIDEO, track->id(), track,
|
||
{stream->id()});
|
||
new_sender->internal()->SetVideoMediaChannel(video_media_channel());
|
||
GetVideoTransceiver()->internal()->AddSender(new_sender);
|
||
const RtpSenderInfo* sender_info =
|
||
FindSenderInfo(local_video_sender_infos_, stream->id(), track->id());
|
||
if (sender_info) {
|
||
new_sender->internal()->SetSsrc(sender_info->first_ssrc);
|
||
}
|
||
}
|
||
|
||
void PeerConnection::RemoveVideoTrack(VideoTrackInterface* track,
|
||
MediaStreamInterface* stream) {
|
||
RTC_DCHECK(!IsClosed());
|
||
auto sender = FindSenderForTrack(track);
|
||
if (!sender) {
|
||
RTC_LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
|
||
<< " doesn't exist.";
|
||
return;
|
||
}
|
||
GetVideoTransceiver()->internal()->RemoveSender(sender);
|
||
}
|
||
|
||
void PeerConnection::SetIceConnectionState(IceConnectionState new_state) {
|
||
RTC_DCHECK(signaling_thread()->IsCurrent());
|
||
if (ice_connection_state_ == new_state) {
|
||
return;
|
||
}
|
||
|
||
// After transitioning to "closed", ignore any additional states from
|
||
// TransportController (such as "disconnected").
|
||
if (IsClosed()) {
|
||
return;
|
||
}
|
||
|
||
RTC_LOG(LS_INFO) << "Changing IceConnectionState " << ice_connection_state_
|
||
<< " => " << new_state;
|
||
RTC_DCHECK(ice_connection_state_ !=
|
||
PeerConnectionInterface::kIceConnectionClosed);
|
||
|
||
ice_connection_state_ = new_state;
|
||
Observer()->OnIceConnectionChange(ice_connection_state_);
|
||
}
|
||
|
||
void PeerConnection::OnIceGatheringChange(
|
||
PeerConnectionInterface::IceGatheringState new_state) {
|
||
RTC_DCHECK(signaling_thread()->IsCurrent());
|
||
if (IsClosed()) {
|
||
return;
|
||
}
|
||
ice_gathering_state_ = new_state;
|
||
Observer()->OnIceGatheringChange(ice_gathering_state_);
|
||
}
|
||
|
||
void PeerConnection::OnIceCandidate(
|
||
std::unique_ptr<IceCandidateInterface> candidate) {
|
||
RTC_DCHECK(signaling_thread()->IsCurrent());
|
||
if (IsClosed()) {
|
||
return;
|
||
}
|
||
NoteUsageEvent(UsageEvent::CANDIDATE_COLLECTED);
|
||
if (candidate->candidate().type() == LOCAL_PORT_TYPE &&
|
||
candidate->candidate().address().IsPrivateIP()) {
|
||
NoteUsageEvent(UsageEvent::PRIVATE_CANDIDATE_COLLECTED);
|
||
}
|
||
Observer()->OnIceCandidate(candidate.get());
|
||
}
|
||
|
||
void PeerConnection::OnIceCandidatesRemoved(
|
||
const std::vector<cricket::Candidate>& candidates) {
|
||
RTC_DCHECK(signaling_thread()->IsCurrent());
|
||
if (IsClosed()) {
|
||
return;
|
||
}
|
||
Observer()->OnIceCandidatesRemoved(candidates);
|
||
}
|
||
|
||
void PeerConnection::ChangeSignalingState(
|
||
PeerConnectionInterface::SignalingState signaling_state) {
|
||
RTC_DCHECK(signaling_thread()->IsCurrent());
|
||
if (signaling_state_ == signaling_state) {
|
||
return;
|
||
}
|
||
RTC_LOG(LS_INFO) << "Session: " << session_id() << " Old state: "
|
||
<< GetSignalingStateString(signaling_state_)
|
||
<< " New state: "
|
||
<< GetSignalingStateString(signaling_state);
|
||
signaling_state_ = signaling_state;
|
||
if (signaling_state == kClosed) {
|
||
ice_connection_state_ = kIceConnectionClosed;
|
||
Observer()->OnIceConnectionChange(ice_connection_state_);
|
||
if (ice_gathering_state_ != kIceGatheringComplete) {
|
||
ice_gathering_state_ = kIceGatheringComplete;
|
||
Observer()->OnIceGatheringChange(ice_gathering_state_);
|
||
}
|
||
}
|
||
Observer()->OnSignalingChange(signaling_state_);
|
||
}
|
||
|
||
void PeerConnection::OnAudioTrackAdded(AudioTrackInterface* track,
|
||
MediaStreamInterface* stream) {
|
||
if (IsClosed()) {
|
||
return;
|
||
}
|
||
AddAudioTrack(track, stream);
|
||
Observer()->OnRenegotiationNeeded();
|
||
}
|
||
|
||
void PeerConnection::OnAudioTrackRemoved(AudioTrackInterface* track,
|
||
MediaStreamInterface* stream) {
|
||
if (IsClosed()) {
|
||
return;
|
||
}
|
||
RemoveAudioTrack(track, stream);
|
||
Observer()->OnRenegotiationNeeded();
|
||
}
|
||
|
||
void PeerConnection::OnVideoTrackAdded(VideoTrackInterface* track,
|
||
MediaStreamInterface* stream) {
|
||
if (IsClosed()) {
|
||
return;
|
||
}
|
||
AddVideoTrack(track, stream);
|
||
Observer()->OnRenegotiationNeeded();
|
||
}
|
||
|
||
void PeerConnection::OnVideoTrackRemoved(VideoTrackInterface* track,
|
||
MediaStreamInterface* stream) {
|
||
if (IsClosed()) {
|
||
return;
|
||
}
|
||
RemoveVideoTrack(track, stream);
|
||
Observer()->OnRenegotiationNeeded();
|
||
}
|
||
|
||
void PeerConnection::PostSetSessionDescriptionSuccess(
|
||
SetSessionDescriptionObserver* observer) {
|
||
SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
|
||
signaling_thread()->Post(RTC_FROM_HERE, this,
|
||
MSG_SET_SESSIONDESCRIPTION_SUCCESS, msg);
|
||
}
|
||
|
||
void PeerConnection::PostSetSessionDescriptionFailure(
|
||
SetSessionDescriptionObserver* observer,
|
||
RTCError&& error) {
|
||
RTC_DCHECK(!error.ok());
|
||
SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
|
||
msg->error = std::move(error);
|
||
signaling_thread()->Post(RTC_FROM_HERE, this,
|
||
MSG_SET_SESSIONDESCRIPTION_FAILED, msg);
|
||
}
|
||
|
||
void PeerConnection::PostCreateSessionDescriptionFailure(
|
||
CreateSessionDescriptionObserver* observer,
|
||
RTCError error) {
|
||
RTC_DCHECK(!error.ok());
|
||
CreateSessionDescriptionMsg* msg = new CreateSessionDescriptionMsg(observer);
|
||
msg->error = std::move(error);
|
||
signaling_thread()->Post(RTC_FROM_HERE, this,
|
||
MSG_CREATE_SESSIONDESCRIPTION_FAILED, msg);
|
||
}
|
||
|
||
void PeerConnection::GetOptionsForOffer(
|
||
const PeerConnectionInterface::RTCOfferAnswerOptions& offer_answer_options,
|
||
cricket::MediaSessionOptions* session_options) {
|
||
ExtractSharedMediaSessionOptions(offer_answer_options, session_options);
|
||
|
||
if (IsUnifiedPlan()) {
|
||
GetOptionsForUnifiedPlanOffer(offer_answer_options, session_options);
|
||
} else {
|
||
GetOptionsForPlanBOffer(offer_answer_options, session_options);
|
||
}
|
||
|
||
// Intentionally unset the data channel type for RTP data channel with the
|
||
// second condition. Otherwise the RTP data channels would be successfully
|
||
// negotiated by default and the unit tests in WebRtcDataBrowserTest will fail
|
||
// when building with chromium. We want to leave RTP data channels broken, so
|
||
// people won't try to use them.
|
||
if (!rtp_data_channels_.empty() || data_channel_type() != cricket::DCT_RTP) {
|
||
session_options->data_channel_type = data_channel_type();
|
||
}
|
||
|
||
// Apply ICE restart flag and renomination flag.
|
||
for (auto& options : session_options->media_description_options) {
|
||
options.transport_options.ice_restart = offer_answer_options.ice_restart;
|
||
options.transport_options.enable_ice_renomination =
|
||
configuration_.enable_ice_renomination;
|
||
}
|
||
|
||
session_options->rtcp_cname = rtcp_cname_;
|
||
session_options->crypto_options = factory_->options().crypto_options;
|
||
session_options->is_unified_plan = IsUnifiedPlan();
|
||
}
|
||
|
||
void PeerConnection::GetOptionsForPlanBOffer(
|
||
const PeerConnectionInterface::RTCOfferAnswerOptions& offer_answer_options,
|
||
cricket::MediaSessionOptions* session_options) {
|
||
// Figure out transceiver directional preferences.
|
||
bool send_audio = HasRtpSender(cricket::MEDIA_TYPE_AUDIO);
|
||
bool send_video = HasRtpSender(cricket::MEDIA_TYPE_VIDEO);
|
||
|
||
// By default, generate sendrecv/recvonly m= sections.
|
||
bool recv_audio = true;
|
||
bool recv_video = true;
|
||
|
||
// By default, only offer a new m= section if we have media to send with it.
|
||
bool offer_new_audio_description = send_audio;
|
||
bool offer_new_video_description = send_video;
|
||
bool offer_new_data_description = HasDataChannels();
|
||
|
||
// The "offer_to_receive_X" options allow those defaults to be overridden.
|
||
if (offer_answer_options.offer_to_receive_audio !=
|
||
RTCOfferAnswerOptions::kUndefined) {
|
||
recv_audio = (offer_answer_options.offer_to_receive_audio > 0);
|
||
offer_new_audio_description =
|
||
offer_new_audio_description ||
|
||
(offer_answer_options.offer_to_receive_audio > 0);
|
||
}
|
||
if (offer_answer_options.offer_to_receive_video !=
|
||
RTCOfferAnswerOptions::kUndefined) {
|
||
recv_video = (offer_answer_options.offer_to_receive_video > 0);
|
||
offer_new_video_description =
|
||
offer_new_video_description ||
|
||
(offer_answer_options.offer_to_receive_video > 0);
|
||
}
|
||
|
||
absl::optional<size_t> audio_index;
|
||
absl::optional<size_t> video_index;
|
||
absl::optional<size_t> data_index;
|
||
// If a current description exists, generate m= sections in the same order,
|
||
// using the first audio/video/data section that appears and rejecting
|
||
// extraneous ones.
|
||
if (local_description()) {
|
||
GenerateMediaDescriptionOptions(
|
||
local_description(),
|
||
RtpTransceiverDirectionFromSendRecv(send_audio, recv_audio),
|
||
RtpTransceiverDirectionFromSendRecv(send_video, recv_video),
|
||
&audio_index, &video_index, &data_index, session_options);
|
||
}
|
||
|
||
// Add audio/video/data m= sections to the end if needed.
|
||
if (!audio_index && offer_new_audio_description) {
|
||
session_options->media_description_options.push_back(
|
||
cricket::MediaDescriptionOptions(
|
||
cricket::MEDIA_TYPE_AUDIO, cricket::CN_AUDIO,
|
||
RtpTransceiverDirectionFromSendRecv(send_audio, recv_audio),
|
||
false));
|
||
audio_index = session_options->media_description_options.size() - 1;
|
||
}
|
||
if (!video_index && offer_new_video_description) {
|
||
session_options->media_description_options.push_back(
|
||
cricket::MediaDescriptionOptions(
|
||
cricket::MEDIA_TYPE_VIDEO, cricket::CN_VIDEO,
|
||
RtpTransceiverDirectionFromSendRecv(send_video, recv_video),
|
||
false));
|
||
video_index = session_options->media_description_options.size() - 1;
|
||
}
|
||
if (!data_index && offer_new_data_description) {
|
||
session_options->media_description_options.push_back(
|
||
GetMediaDescriptionOptionsForActiveData(cricket::CN_DATA));
|
||
data_index = session_options->media_description_options.size() - 1;
|
||
}
|
||
|
||
cricket::MediaDescriptionOptions* audio_media_description_options =
|
||
!audio_index ? nullptr
|
||
: &session_options->media_description_options[*audio_index];
|
||
cricket::MediaDescriptionOptions* video_media_description_options =
|
||
!video_index ? nullptr
|
||
: &session_options->media_description_options[*video_index];
|
||
|
||
AddRtpSenderOptions(GetSendersInternal(), audio_media_description_options,
|
||
video_media_description_options,
|
||
offer_answer_options.num_simulcast_layers);
|
||
}
|
||
|
||
// Find a new MID that is not already in |used_mids|, then add it to |used_mids|
|
||
// and return a reference to it.
|
||
// Generated MIDs should be no more than 3 bytes long to take up less space in
|
||
// the RTP packet.
|
||
static const std::string& AllocateMid(std::set<std::string>* used_mids) {
|
||
RTC_DCHECK(used_mids);
|
||
// We're boring: just generate MIDs 0, 1, 2, ...
|
||
size_t i = 0;
|
||
std::set<std::string>::iterator it;
|
||
bool inserted;
|
||
do {
|
||
std::string mid = rtc::ToString(i++);
|
||
auto insert_result = used_mids->insert(mid);
|
||
it = insert_result.first;
|
||
inserted = insert_result.second;
|
||
} while (!inserted);
|
||
return *it;
|
||
}
|
||
|
||
static cricket::MediaDescriptionOptions
|
||
GetMediaDescriptionOptionsForTransceiver(
|
||
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
||
transceiver,
|
||
const std::string& mid) {
|
||
cricket::MediaDescriptionOptions media_description_options(
|
||
transceiver->media_type(), mid, transceiver->direction(),
|
||
transceiver->stopped());
|
||
// This behavior is specified in JSEP. The gist is that:
|
||
// 1. The MSID is included if the RtpTransceiver's direction is sendonly or
|
||
// sendrecv.
|
||
// 2. If the MSID is included, then it must be included in any subsequent
|
||
// offer/answer exactly the same until the RtpTransceiver is stopped.
|
||
if (!transceiver->stopped() &&
|
||
(RtpTransceiverDirectionHasSend(transceiver->direction()) ||
|
||
transceiver->internal()->has_ever_been_used_to_send())) {
|
||
cricket::SenderOptions sender_options;
|
||
sender_options.track_id = transceiver->sender()->id();
|
||
sender_options.stream_ids = transceiver->sender()->stream_ids();
|
||
// TODO(bugs.webrtc.org/7600): Set num_sim_layers to the number of encodings
|
||
// set in the RTP parameters when the transceiver was added.
|
||
sender_options.num_sim_layers = 1;
|
||
media_description_options.sender_options.push_back(sender_options);
|
||
}
|
||
return media_description_options;
|
||
}
|
||
|
||
void PeerConnection::GetOptionsForUnifiedPlanOffer(
|
||
const RTCOfferAnswerOptions& offer_answer_options,
|
||
cricket::MediaSessionOptions* session_options) {
|
||
// Rules for generating an offer are dictated by JSEP sections 5.2.1 (Initial
|
||
// Offers) and 5.2.2 (Subsequent Offers).
|
||
RTC_DCHECK_EQ(session_options->media_description_options.size(), 0);
|
||
const ContentInfos& local_contents =
|
||
(local_description() ? local_description()->description()->contents()
|
||
: ContentInfos());
|
||
const ContentInfos& remote_contents =
|
||
(remote_description() ? remote_description()->description()->contents()
|
||
: ContentInfos());
|
||
// The mline indices that can be recycled. New transceivers should reuse these
|
||
// slots first.
|
||
std::queue<size_t> recycleable_mline_indices;
|
||
// Track the MIDs used in previous offer/answer exchanges and the current
|
||
// offer so that new, unique MIDs are generated.
|
||
std::set<std::string> used_mids = seen_mids_;
|
||
// First, go through each media section that exists in either the local or
|
||
// remote description and generate a media section in this offer for the
|
||
// associated transceiver. If a media section can be recycled, generate a
|
||
// default, rejected media section here that can be later overwritten.
|
||
for (size_t i = 0;
|
||
i < std::max(local_contents.size(), remote_contents.size()); ++i) {
|
||
// Either |local_content| or |remote_content| is non-null.
|
||
const ContentInfo* local_content =
|
||
(i < local_contents.size() ? &local_contents[i] : nullptr);
|
||
const ContentInfo* remote_content =
|
||
(i < remote_contents.size() ? &remote_contents[i] : nullptr);
|
||
bool had_been_rejected = (local_content && local_content->rejected) ||
|
||
(remote_content && remote_content->rejected);
|
||
const std::string& mid =
|
||
(local_content ? local_content->name : remote_content->name);
|
||
cricket::MediaType media_type =
|
||
(local_content ? local_content->media_description()->type()
|
||
: remote_content->media_description()->type());
|
||
if (media_type == cricket::MEDIA_TYPE_AUDIO ||
|
||
media_type == cricket::MEDIA_TYPE_VIDEO) {
|
||
auto transceiver = GetAssociatedTransceiver(mid);
|
||
RTC_CHECK(transceiver);
|
||
// A media section is considered eligible for recycling if it is marked as
|
||
// rejected in either the local or remote description.
|
||
if (had_been_rejected && transceiver->stopped()) {
|
||
session_options->media_description_options.push_back(
|
||
cricket::MediaDescriptionOptions(transceiver->media_type(), mid,
|
||
RtpTransceiverDirection::kInactive,
|
||
/*stopped=*/true));
|
||
recycleable_mline_indices.push(i);
|
||
} else {
|
||
session_options->media_description_options.push_back(
|
||
GetMediaDescriptionOptionsForTransceiver(transceiver, mid));
|
||
// CreateOffer shouldn't really cause any state changes in
|
||
// PeerConnection, but we need a way to match new transceivers to new
|
||
// media sections in SetLocalDescription and JSEP specifies this is done
|
||
// by recording the index of the media section generated for the
|
||
// transceiver in the offer.
|
||
transceiver->internal()->set_mline_index(i);
|
||
}
|
||
} else {
|
||
RTC_CHECK_EQ(cricket::MEDIA_TYPE_DATA, media_type);
|
||
RTC_CHECK(GetDataMid());
|
||
if (had_been_rejected || mid != *GetDataMid()) {
|
||
session_options->media_description_options.push_back(
|
||
GetMediaDescriptionOptionsForRejectedData(mid));
|
||
} else {
|
||
session_options->media_description_options.push_back(
|
||
GetMediaDescriptionOptionsForActiveData(mid));
|
||
}
|
||
}
|
||
}
|
||
// Next, look for transceivers that are newly added (that is, are not stopped
|
||
// and not associated). Reuse media sections marked as recyclable first,
|
||
// otherwise append to the end of the offer. New media sections should be
|
||
// added in the order they were added to the PeerConnection.
|
||
for (auto transceiver : transceivers_) {
|
||
if (transceiver->mid() || transceiver->stopped()) {
|
||
continue;
|
||
}
|
||
size_t mline_index;
|
||
if (!recycleable_mline_indices.empty()) {
|
||
mline_index = recycleable_mline_indices.front();
|
||
recycleable_mline_indices.pop();
|
||
session_options->media_description_options[mline_index] =
|
||
GetMediaDescriptionOptionsForTransceiver(transceiver,
|
||
AllocateMid(&used_mids));
|
||
} else {
|
||
mline_index = session_options->media_description_options.size();
|
||
session_options->media_description_options.push_back(
|
||
GetMediaDescriptionOptionsForTransceiver(transceiver,
|
||
AllocateMid(&used_mids)));
|
||
}
|
||
// See comment above for why CreateOffer changes the transceiver's state.
|
||
transceiver->internal()->set_mline_index(mline_index);
|
||
}
|
||
// Lastly, add a m-section if we have local data channels and an m section
|
||
// does not already exist.
|
||
if (!GetDataMid() && HasDataChannels()) {
|
||
session_options->media_description_options.push_back(
|
||
GetMediaDescriptionOptionsForActiveData(AllocateMid(&used_mids)));
|
||
}
|
||
}
|
||
|
||
void PeerConnection::GetOptionsForAnswer(
|
||
const RTCOfferAnswerOptions& offer_answer_options,
|
||
cricket::MediaSessionOptions* session_options) {
|
||
ExtractSharedMediaSessionOptions(offer_answer_options, session_options);
|
||
|
||
if (IsUnifiedPlan()) {
|
||
GetOptionsForUnifiedPlanAnswer(offer_answer_options, session_options);
|
||
} else {
|
||
GetOptionsForPlanBAnswer(offer_answer_options, session_options);
|
||
}
|
||
|
||
// Intentionally unset the data channel type for RTP data channel. Otherwise
|
||
// the RTP data channels would be successfully negotiated by default and the
|
||
// unit tests in WebRtcDataBrowserTest will fail when building with chromium.
|
||
// We want to leave RTP data channels broken, so people won't try to use them.
|
||
if (!rtp_data_channels_.empty() || data_channel_type() != cricket::DCT_RTP) {
|
||
session_options->data_channel_type = data_channel_type();
|
||
}
|
||
|
||
// Apply ICE renomination flag.
|
||
for (auto& options : session_options->media_description_options) {
|
||
options.transport_options.enable_ice_renomination =
|
||
configuration_.enable_ice_renomination;
|
||
}
|
||
|
||
session_options->rtcp_cname = rtcp_cname_;
|
||
session_options->crypto_options = factory_->options().crypto_options;
|
||
session_options->is_unified_plan = IsUnifiedPlan();
|
||
}
|
||
|
||
void PeerConnection::GetOptionsForPlanBAnswer(
|
||
const PeerConnectionInterface::RTCOfferAnswerOptions& offer_answer_options,
|
||
cricket::MediaSessionOptions* session_options) {
|
||
// Figure out transceiver directional preferences.
|
||
bool send_audio = HasRtpSender(cricket::MEDIA_TYPE_AUDIO);
|
||
bool send_video = HasRtpSender(cricket::MEDIA_TYPE_VIDEO);
|
||
|
||
// By default, generate sendrecv/recvonly m= sections. The direction is also
|
||
// restricted by the direction in the offer.
|
||
bool recv_audio = true;
|
||
bool recv_video = true;
|
||
|
||
// The "offer_to_receive_X" options allow those defaults to be overridden.
|
||
if (offer_answer_options.offer_to_receive_audio !=
|
||
RTCOfferAnswerOptions::kUndefined) {
|
||
recv_audio = (offer_answer_options.offer_to_receive_audio > 0);
|
||
}
|
||
if (offer_answer_options.offer_to_receive_video !=
|
||
RTCOfferAnswerOptions::kUndefined) {
|
||
recv_video = (offer_answer_options.offer_to_receive_video > 0);
|
||
}
|
||
|
||
absl::optional<size_t> audio_index;
|
||
absl::optional<size_t> video_index;
|
||
absl::optional<size_t> data_index;
|
||
|
||
// Generate m= sections that match those in the offer.
|
||
// Note that mediasession.cc will handle intersection our preferred
|
||
// direction with the offered direction.
|
||
GenerateMediaDescriptionOptions(
|
||
remote_description(),
|
||
RtpTransceiverDirectionFromSendRecv(send_audio, recv_audio),
|
||
RtpTransceiverDirectionFromSendRecv(send_video, recv_video), &audio_index,
|
||
&video_index, &data_index, session_options);
|
||
|
||
cricket::MediaDescriptionOptions* audio_media_description_options =
|
||
!audio_index ? nullptr
|
||
: &session_options->media_description_options[*audio_index];
|
||
cricket::MediaDescriptionOptions* video_media_description_options =
|
||
!video_index ? nullptr
|
||
: &session_options->media_description_options[*video_index];
|
||
|
||
AddRtpSenderOptions(GetSendersInternal(), audio_media_description_options,
|
||
video_media_description_options,
|
||
offer_answer_options.num_simulcast_layers);
|
||
}
|
||
|
||
void PeerConnection::GetOptionsForUnifiedPlanAnswer(
|
||
const PeerConnectionInterface::RTCOfferAnswerOptions& offer_answer_options,
|
||
cricket::MediaSessionOptions* session_options) {
|
||
// Rules for generating an answer are dictated by JSEP sections 5.3.1 (Initial
|
||
// Answers) and 5.3.2 (Subsequent Answers).
|
||
RTC_DCHECK(remote_description());
|
||
RTC_DCHECK(remote_description()->GetType() == SdpType::kOffer);
|
||
for (const ContentInfo& content :
|
||
remote_description()->description()->contents()) {
|
||
cricket::MediaType media_type = content.media_description()->type();
|
||
if (media_type == cricket::MEDIA_TYPE_AUDIO ||
|
||
media_type == cricket::MEDIA_TYPE_VIDEO) {
|
||
auto transceiver = GetAssociatedTransceiver(content.name);
|
||
RTC_CHECK(transceiver);
|
||
session_options->media_description_options.push_back(
|
||
GetMediaDescriptionOptionsForTransceiver(transceiver, content.name));
|
||
} else {
|
||
RTC_CHECK_EQ(cricket::MEDIA_TYPE_DATA, media_type);
|
||
// Reject all data sections if data channels are disabled.
|
||
// Reject a data section if it has already been rejected.
|
||
// Reject all data sections except for the first one.
|
||
if (data_channel_type_ == cricket::DCT_NONE || content.rejected ||
|
||
content.name != *GetDataMid()) {
|
||
session_options->media_description_options.push_back(
|
||
GetMediaDescriptionOptionsForRejectedData(content.name));
|
||
} else {
|
||
session_options->media_description_options.push_back(
|
||
GetMediaDescriptionOptionsForActiveData(content.name));
|
||
}
|
||
}
|
||
}
|
||
}
|
||
|
||
void PeerConnection::GenerateMediaDescriptionOptions(
|
||
const SessionDescriptionInterface* session_desc,
|
||
RtpTransceiverDirection audio_direction,
|
||
RtpTransceiverDirection video_direction,
|
||
absl::optional<size_t>* audio_index,
|
||
absl::optional<size_t>* video_index,
|
||
absl::optional<size_t>* data_index,
|
||
cricket::MediaSessionOptions* session_options) {
|
||
for (const cricket::ContentInfo& content :
|
||
session_desc->description()->contents()) {
|
||
if (IsAudioContent(&content)) {
|
||
// If we already have an audio m= section, reject this extra one.
|
||
if (*audio_index) {
|
||
session_options->media_description_options.push_back(
|
||
cricket::MediaDescriptionOptions(
|
||
cricket::MEDIA_TYPE_AUDIO, content.name,
|
||
RtpTransceiverDirection::kInactive, true));
|
||
} else {
|
||
session_options->media_description_options.push_back(
|
||
cricket::MediaDescriptionOptions(
|
||
cricket::MEDIA_TYPE_AUDIO, content.name, audio_direction,
|
||
audio_direction == RtpTransceiverDirection::kInactive));
|
||
*audio_index = session_options->media_description_options.size() - 1;
|
||
}
|
||
} else if (IsVideoContent(&content)) {
|
||
// If we already have an video m= section, reject this extra one.
|
||
if (*video_index) {
|
||
session_options->media_description_options.push_back(
|
||
cricket::MediaDescriptionOptions(
|
||
cricket::MEDIA_TYPE_VIDEO, content.name,
|
||
RtpTransceiverDirection::kInactive, true));
|
||
} else {
|
||
session_options->media_description_options.push_back(
|
||
cricket::MediaDescriptionOptions(
|
||
cricket::MEDIA_TYPE_VIDEO, content.name, video_direction,
|
||
video_direction == RtpTransceiverDirection::kInactive));
|
||
*video_index = session_options->media_description_options.size() - 1;
|
||
}
|
||
} else {
|
||
RTC_DCHECK(IsDataContent(&content));
|
||
// If we already have an data m= section, reject this extra one.
|
||
if (*data_index) {
|
||
session_options->media_description_options.push_back(
|
||
GetMediaDescriptionOptionsForRejectedData(content.name));
|
||
} else {
|
||
session_options->media_description_options.push_back(
|
||
GetMediaDescriptionOptionsForActiveData(content.name));
|
||
*data_index = session_options->media_description_options.size() - 1;
|
||
}
|
||
}
|
||
}
|
||
}
|
||
|
||
cricket::MediaDescriptionOptions
|
||
PeerConnection::GetMediaDescriptionOptionsForActiveData(
|
||
const std::string& mid) const {
|
||
// Direction for data sections is meaningless, but legacy endpoints might
|
||
// expect sendrecv.
|
||
cricket::MediaDescriptionOptions options(cricket::MEDIA_TYPE_DATA, mid,
|
||
RtpTransceiverDirection::kSendRecv,
|
||
/*stopped=*/false);
|
||
AddRtpDataChannelOptions(rtp_data_channels_, &options);
|
||
return options;
|
||
}
|
||
|
||
cricket::MediaDescriptionOptions
|
||
PeerConnection::GetMediaDescriptionOptionsForRejectedData(
|
||
const std::string& mid) const {
|
||
cricket::MediaDescriptionOptions options(cricket::MEDIA_TYPE_DATA, mid,
|
||
RtpTransceiverDirection::kInactive,
|
||
/*stopped=*/true);
|
||
AddRtpDataChannelOptions(rtp_data_channels_, &options);
|
||
return options;
|
||
}
|
||
|
||
absl::optional<std::string> PeerConnection::GetDataMid() const {
|
||
switch (data_channel_type_) {
|
||
case cricket::DCT_RTP:
|
||
if (!rtp_data_channel_) {
|
||
return absl::nullopt;
|
||
}
|
||
return rtp_data_channel_->content_name();
|
||
case cricket::DCT_SCTP:
|
||
return sctp_mid_;
|
||
default:
|
||
return absl::nullopt;
|
||
}
|
||
}
|
||
|
||
void PeerConnection::RemoveSenders(cricket::MediaType media_type) {
|
||
UpdateLocalSenders(std::vector<cricket::StreamParams>(), media_type);
|
||
UpdateRemoteSendersList(std::vector<cricket::StreamParams>(), false,
|
||
media_type, nullptr);
|
||
}
|
||
|
||
void PeerConnection::UpdateRemoteSendersList(
|
||
const cricket::StreamParamsVec& streams,
|
||
bool default_sender_needed,
|
||
cricket::MediaType media_type,
|
||
StreamCollection* new_streams) {
|
||
RTC_DCHECK(!IsUnifiedPlan());
|
||
|
||
std::vector<RtpSenderInfo>* current_senders =
|
||
GetRemoteSenderInfos(media_type);
|
||
|
||
// Find removed senders. I.e., senders where the sender id or ssrc don't match
|
||
// the new StreamParam.
|
||
for (auto sender_it = current_senders->begin();
|
||
sender_it != current_senders->end();
|
||
/* incremented manually */) {
|
||
const RtpSenderInfo& info = *sender_it;
|
||
const cricket::StreamParams* params =
|
||
cricket::GetStreamBySsrc(streams, info.first_ssrc);
|
||
std::string params_stream_id;
|
||
if (params) {
|
||
params_stream_id =
|
||
(!params->first_stream_id().empty() ? params->first_stream_id()
|
||
: kDefaultStreamId);
|
||
}
|
||
bool sender_exists = params && params->id == info.sender_id &&
|
||
params_stream_id == info.stream_id;
|
||
// If this is a default track, and we still need it, don't remove it.
|
||
if ((info.stream_id == kDefaultStreamId && default_sender_needed) ||
|
||
sender_exists) {
|
||
++sender_it;
|
||
} else {
|
||
OnRemoteSenderRemoved(info, media_type);
|
||
sender_it = current_senders->erase(sender_it);
|
||
}
|
||
}
|
||
|
||
// Find new and active senders.
|
||
for (const cricket::StreamParams& params : streams) {
|
||
if (!params.has_ssrcs()) {
|
||
// The remote endpoint has streams, but didn't signal ssrcs. For an active
|
||
// sender, this means it is coming from a Unified Plan endpoint,so we just
|
||
// create a default.
|
||
default_sender_needed = true;
|
||
break;
|
||
}
|
||
|
||
// |params.id| is the sender id and the stream id uses the first of
|
||
// |params.stream_ids|. The remote description could come from a Unified
|
||
// Plan endpoint, with multiple or no stream_ids() signaled. Since this is
|
||
// not supported in Plan B, we just take the first here and create the
|
||
// default stream ID if none is specified.
|
||
const std::string& stream_id =
|
||
(!params.first_stream_id().empty() ? params.first_stream_id()
|
||
: kDefaultStreamId);
|
||
const std::string& sender_id = params.id;
|
||
uint32_t ssrc = params.first_ssrc();
|
||
|
||
rtc::scoped_refptr<MediaStreamInterface> stream =
|
||
remote_streams_->find(stream_id);
|
||
if (!stream) {
|
||
// This is a new MediaStream. Create a new remote MediaStream.
|
||
stream = MediaStreamProxy::Create(rtc::Thread::Current(),
|
||
MediaStream::Create(stream_id));
|
||
remote_streams_->AddStream(stream);
|
||
new_streams->AddStream(stream);
|
||
}
|
||
|
||
const RtpSenderInfo* sender_info =
|
||
FindSenderInfo(*current_senders, stream_id, sender_id);
|
||
if (!sender_info) {
|
||
current_senders->push_back(RtpSenderInfo(stream_id, sender_id, ssrc));
|
||
OnRemoteSenderAdded(current_senders->back(), media_type);
|
||
}
|
||
}
|
||
|
||
// Add default sender if necessary.
|
||
if (default_sender_needed) {
|
||
rtc::scoped_refptr<MediaStreamInterface> default_stream =
|
||
remote_streams_->find(kDefaultStreamId);
|
||
if (!default_stream) {
|
||
// Create the new default MediaStream.
|
||
default_stream = MediaStreamProxy::Create(
|
||
rtc::Thread::Current(), MediaStream::Create(kDefaultStreamId));
|
||
remote_streams_->AddStream(default_stream);
|
||
new_streams->AddStream(default_stream);
|
||
}
|
||
std::string default_sender_id = (media_type == cricket::MEDIA_TYPE_AUDIO)
|
||
? kDefaultAudioSenderId
|
||
: kDefaultVideoSenderId;
|
||
const RtpSenderInfo* default_sender_info =
|
||
FindSenderInfo(*current_senders, kDefaultStreamId, default_sender_id);
|
||
if (!default_sender_info) {
|
||
current_senders->push_back(
|
||
RtpSenderInfo(kDefaultStreamId, default_sender_id, 0));
|
||
OnRemoteSenderAdded(current_senders->back(), media_type);
|
||
}
|
||
}
|
||
}
|
||
|
||
void PeerConnection::OnRemoteSenderAdded(const RtpSenderInfo& sender_info,
|
||
cricket::MediaType media_type) {
|
||
RTC_LOG(LS_INFO) << "Creating " << cricket::MediaTypeToString(media_type)
|
||
<< " receiver for track_id=" << sender_info.sender_id
|
||
<< " and stream_id=" << sender_info.stream_id;
|
||
|
||
MediaStreamInterface* stream = remote_streams_->find(sender_info.stream_id);
|
||
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
|
||
CreateAudioReceiver(stream, sender_info);
|
||
} else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
|
||
CreateVideoReceiver(stream, sender_info);
|
||
} else {
|
||
RTC_NOTREACHED() << "Invalid media type";
|
||
}
|
||
}
|
||
|
||
void PeerConnection::OnRemoteSenderRemoved(const RtpSenderInfo& sender_info,
|
||
cricket::MediaType media_type) {
|
||
RTC_LOG(LS_INFO) << "Removing " << cricket::MediaTypeToString(media_type)
|
||
<< " receiver for track_id=" << sender_info.sender_id
|
||
<< " and stream_id=" << sender_info.stream_id;
|
||
|
||
MediaStreamInterface* stream = remote_streams_->find(sender_info.stream_id);
|
||
|
||
rtc::scoped_refptr<RtpReceiverInterface> receiver;
|
||
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
|
||
// When the MediaEngine audio channel is destroyed, the RemoteAudioSource
|
||
// will be notified which will end the AudioRtpReceiver::track().
|
||
receiver = RemoveAndStopReceiver(sender_info);
|
||
rtc::scoped_refptr<AudioTrackInterface> audio_track =
|
||
stream->FindAudioTrack(sender_info.sender_id);
|
||
if (audio_track) {
|
||
stream->RemoveTrack(audio_track);
|
||
}
|
||
} else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
|
||
// Stopping or destroying a VideoRtpReceiver will end the
|
||
// VideoRtpReceiver::track().
|
||
receiver = RemoveAndStopReceiver(sender_info);
|
||
rtc::scoped_refptr<VideoTrackInterface> video_track =
|
||
stream->FindVideoTrack(sender_info.sender_id);
|
||
if (video_track) {
|
||
// There's no guarantee the track is still available, e.g. the track may
|
||
// have been removed from the stream by an application.
|
||
stream->RemoveTrack(video_track);
|
||
}
|
||
} else {
|
||
RTC_NOTREACHED() << "Invalid media type";
|
||
}
|
||
if (receiver) {
|
||
Observer()->OnRemoveTrack(receiver);
|
||
}
|
||
}
|
||
|
||
void PeerConnection::UpdateEndedRemoteMediaStreams() {
|
||
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams_to_remove;
|
||
for (size_t i = 0; i < remote_streams_->count(); ++i) {
|
||
MediaStreamInterface* stream = remote_streams_->at(i);
|
||
if (stream->GetAudioTracks().empty() && stream->GetVideoTracks().empty()) {
|
||
streams_to_remove.push_back(stream);
|
||
}
|
||
}
|
||
|
||
for (auto& stream : streams_to_remove) {
|
||
remote_streams_->RemoveStream(stream);
|
||
Observer()->OnRemoveStream(std::move(stream));
|
||
}
|
||
}
|
||
|
||
void PeerConnection::UpdateLocalSenders(
|
||
const std::vector<cricket::StreamParams>& streams,
|
||
cricket::MediaType media_type) {
|
||
std::vector<RtpSenderInfo>* current_senders = GetLocalSenderInfos(media_type);
|
||
|
||
// Find removed tracks. I.e., tracks where the track id, stream id or ssrc
|
||
// don't match the new StreamParam.
|
||
for (auto sender_it = current_senders->begin();
|
||
sender_it != current_senders->end();
|
||
/* incremented manually */) {
|
||
const RtpSenderInfo& info = *sender_it;
|
||
const cricket::StreamParams* params =
|
||
cricket::GetStreamBySsrc(streams, info.first_ssrc);
|
||
if (!params || params->id != info.sender_id ||
|
||
params->first_stream_id() != info.stream_id) {
|
||
OnLocalSenderRemoved(info, media_type);
|
||
sender_it = current_senders->erase(sender_it);
|
||
} else {
|
||
++sender_it;
|
||
}
|
||
}
|
||
|
||
// Find new and active senders.
|
||
for (const cricket::StreamParams& params : streams) {
|
||
// The sync_label is the MediaStream label and the |stream.id| is the
|
||
// sender id.
|
||
const std::string& stream_id = params.first_stream_id();
|
||
const std::string& sender_id = params.id;
|
||
uint32_t ssrc = params.first_ssrc();
|
||
const RtpSenderInfo* sender_info =
|
||
FindSenderInfo(*current_senders, stream_id, sender_id);
|
||
if (!sender_info) {
|
||
current_senders->push_back(RtpSenderInfo(stream_id, sender_id, ssrc));
|
||
OnLocalSenderAdded(current_senders->back(), media_type);
|
||
}
|
||
}
|
||
}
|
||
|
||
void PeerConnection::OnLocalSenderAdded(const RtpSenderInfo& sender_info,
|
||
cricket::MediaType media_type) {
|
||
RTC_DCHECK(!IsUnifiedPlan());
|
||
auto sender = FindSenderById(sender_info.sender_id);
|
||
if (!sender) {
|
||
RTC_LOG(LS_WARNING) << "An unknown RtpSender with id "
|
||
<< sender_info.sender_id
|
||
<< " has been configured in the local description.";
|
||
return;
|
||
}
|
||
|
||
if (sender->media_type() != media_type) {
|
||
RTC_LOG(LS_WARNING) << "An RtpSender has been configured in the local"
|
||
" description with an unexpected media type.";
|
||
return;
|
||
}
|
||
|
||
sender->internal()->set_stream_ids({sender_info.stream_id});
|
||
sender->internal()->SetSsrc(sender_info.first_ssrc);
|
||
}
|
||
|
||
void PeerConnection::OnLocalSenderRemoved(const RtpSenderInfo& sender_info,
|
||
cricket::MediaType media_type) {
|
||
auto sender = FindSenderById(sender_info.sender_id);
|
||
if (!sender) {
|
||
// This is the normal case. I.e., RemoveStream has been called and the
|
||
// SessionDescriptions has been renegotiated.
|
||
return;
|
||
}
|
||
|
||
// A sender has been removed from the SessionDescription but it's still
|
||
// associated with the PeerConnection. This only occurs if the SDP doesn't
|
||
// match with the calls to CreateSender, AddStream and RemoveStream.
|
||
if (sender->media_type() != media_type) {
|
||
RTC_LOG(LS_WARNING) << "An RtpSender has been configured in the local"
|
||
" description with an unexpected media type.";
|
||
return;
|
||
}
|
||
|
||
sender->internal()->SetSsrc(0);
|
||
}
|
||
|
||
void PeerConnection::UpdateLocalRtpDataChannels(
|
||
const cricket::StreamParamsVec& streams) {
|
||
std::vector<std::string> existing_channels;
|
||
|
||
// Find new and active data channels.
|
||
for (const cricket::StreamParams& params : streams) {
|
||
// |it->sync_label| is actually the data channel label. The reason is that
|
||
// we use the same naming of data channels as we do for
|
||
// MediaStreams and Tracks.
|
||
// For MediaStreams, the sync_label is the MediaStream label and the
|
||
// track label is the same as |streamid|.
|
||
const std::string& channel_label = params.first_stream_id();
|
||
auto data_channel_it = rtp_data_channels_.find(channel_label);
|
||
if (data_channel_it == rtp_data_channels_.end()) {
|
||
RTC_LOG(LS_ERROR) << "channel label not found";
|
||
continue;
|
||
}
|
||
// Set the SSRC the data channel should use for sending.
|
||
data_channel_it->second->SetSendSsrc(params.first_ssrc());
|
||
existing_channels.push_back(data_channel_it->first);
|
||
}
|
||
|
||
UpdateClosingRtpDataChannels(existing_channels, true);
|
||
}
|
||
|
||
void PeerConnection::UpdateRemoteRtpDataChannels(
|
||
const cricket::StreamParamsVec& streams) {
|
||
std::vector<std::string> existing_channels;
|
||
|
||
// Find new and active data channels.
|
||
for (const cricket::StreamParams& params : streams) {
|
||
// The data channel label is either the mslabel or the SSRC if the mslabel
|
||
// does not exist. Ex a=ssrc:444330170 mslabel:test1.
|
||
std::string label = params.first_stream_id().empty()
|
||
? rtc::ToString(params.first_ssrc())
|
||
: params.first_stream_id();
|
||
auto data_channel_it = rtp_data_channels_.find(label);
|
||
if (data_channel_it == rtp_data_channels_.end()) {
|
||
// This is a new data channel.
|
||
CreateRemoteRtpDataChannel(label, params.first_ssrc());
|
||
} else {
|
||
data_channel_it->second->SetReceiveSsrc(params.first_ssrc());
|
||
}
|
||
existing_channels.push_back(label);
|
||
}
|
||
|
||
UpdateClosingRtpDataChannels(existing_channels, false);
|
||
}
|
||
|
||
void PeerConnection::UpdateClosingRtpDataChannels(
|
||
const std::vector<std::string>& active_channels,
|
||
bool is_local_update) {
|
||
auto it = rtp_data_channels_.begin();
|
||
while (it != rtp_data_channels_.end()) {
|
||
DataChannel* data_channel = it->second;
|
||
if (std::find(active_channels.begin(), active_channels.end(),
|
||
data_channel->label()) != active_channels.end()) {
|
||
++it;
|
||
continue;
|
||
}
|
||
|
||
if (is_local_update) {
|
||
data_channel->SetSendSsrc(0);
|
||
} else {
|
||
data_channel->RemotePeerRequestClose();
|
||
}
|
||
|
||
if (data_channel->state() == DataChannel::kClosed) {
|
||
rtp_data_channels_.erase(it);
|
||
it = rtp_data_channels_.begin();
|
||
} else {
|
||
++it;
|
||
}
|
||
}
|
||
}
|
||
|
||
void PeerConnection::CreateRemoteRtpDataChannel(const std::string& label,
|
||
uint32_t remote_ssrc) {
|
||
rtc::scoped_refptr<DataChannel> channel(
|
||
InternalCreateDataChannel(label, nullptr));
|
||
if (!channel.get()) {
|
||
RTC_LOG(LS_WARNING) << "Remote peer requested a DataChannel but"
|
||
"CreateDataChannel failed.";
|
||
return;
|
||
}
|
||
channel->SetReceiveSsrc(remote_ssrc);
|
||
rtc::scoped_refptr<DataChannelInterface> proxy_channel =
|
||
DataChannelProxy::Create(signaling_thread(), channel);
|
||
Observer()->OnDataChannel(std::move(proxy_channel));
|
||
}
|
||
|
||
rtc::scoped_refptr<DataChannel> PeerConnection::InternalCreateDataChannel(
|
||
const std::string& label,
|
||
const InternalDataChannelInit* config) {
|
||
if (IsClosed()) {
|
||
return nullptr;
|
||
}
|
||
if (data_channel_type() == cricket::DCT_NONE) {
|
||
RTC_LOG(LS_ERROR)
|
||
<< "InternalCreateDataChannel: Data is not supported in this call.";
|
||
return nullptr;
|
||
}
|
||
InternalDataChannelInit new_config =
|
||
config ? (*config) : InternalDataChannelInit();
|
||
if (data_channel_type() == cricket::DCT_SCTP) {
|
||
if (new_config.id < 0) {
|
||
rtc::SSLRole role;
|
||
if ((GetSctpSslRole(&role)) &&
|
||
!sid_allocator_.AllocateSid(role, &new_config.id)) {
|
||
RTC_LOG(LS_ERROR)
|
||
<< "No id can be allocated for the SCTP data channel.";
|
||
return nullptr;
|
||
}
|
||
} else if (!sid_allocator_.ReserveSid(new_config.id)) {
|
||
RTC_LOG(LS_ERROR) << "Failed to create a SCTP data channel "
|
||
"because the id is already in use or out of range.";
|
||
return nullptr;
|
||
}
|
||
}
|
||
|
||
rtc::scoped_refptr<DataChannel> channel(
|
||
DataChannel::Create(this, data_channel_type(), label, new_config));
|
||
if (!channel) {
|
||
sid_allocator_.ReleaseSid(new_config.id);
|
||
return nullptr;
|
||
}
|
||
|
||
if (channel->data_channel_type() == cricket::DCT_RTP) {
|
||
if (rtp_data_channels_.find(channel->label()) != rtp_data_channels_.end()) {
|
||
RTC_LOG(LS_ERROR) << "DataChannel with label " << channel->label()
|
||
<< " already exists.";
|
||
return nullptr;
|
||
}
|
||
rtp_data_channels_[channel->label()] = channel;
|
||
} else {
|
||
RTC_DCHECK(channel->data_channel_type() == cricket::DCT_SCTP);
|
||
sctp_data_channels_.push_back(channel);
|
||
channel->SignalClosed.connect(this,
|
||
&PeerConnection::OnSctpDataChannelClosed);
|
||
}
|
||
|
||
SignalDataChannelCreated_(channel.get());
|
||
return channel;
|
||
}
|
||
|
||
bool PeerConnection::HasDataChannels() const {
|
||
return !rtp_data_channels_.empty() || !sctp_data_channels_.empty();
|
||
}
|
||
|
||
void PeerConnection::AllocateSctpSids(rtc::SSLRole role) {
|
||
for (const auto& channel : sctp_data_channels_) {
|
||
if (channel->id() < 0) {
|
||
int sid;
|
||
if (!sid_allocator_.AllocateSid(role, &sid)) {
|
||
RTC_LOG(LS_ERROR) << "Failed to allocate SCTP sid.";
|
||
continue;
|
||
}
|
||
channel->SetSctpSid(sid);
|
||
}
|
||
}
|
||
}
|
||
|
||
void PeerConnection::OnSctpDataChannelClosed(DataChannel* channel) {
|
||
RTC_DCHECK(signaling_thread()->IsCurrent());
|
||
for (auto it = sctp_data_channels_.begin(); it != sctp_data_channels_.end();
|
||
++it) {
|
||
if (it->get() == channel) {
|
||
if (channel->id() >= 0) {
|
||
// After the closing procedure is done, it's safe to use this ID for
|
||
// another data channel.
|
||
sid_allocator_.ReleaseSid(channel->id());
|
||
}
|
||
// Since this method is triggered by a signal from the DataChannel,
|
||
// we can't free it directly here; we need to free it asynchronously.
|
||
sctp_data_channels_to_free_.push_back(*it);
|
||
sctp_data_channels_.erase(it);
|
||
signaling_thread()->Post(RTC_FROM_HERE, this, MSG_FREE_DATACHANNELS,
|
||
nullptr);
|
||
return;
|
||
}
|
||
}
|
||
}
|
||
|
||
void PeerConnection::OnDataChannelDestroyed() {
|
||
// Use a temporary copy of the RTP/SCTP DataChannel list because the
|
||
// DataChannel may callback to us and try to modify the list.
|
||
std::map<std::string, rtc::scoped_refptr<DataChannel>> temp_rtp_dcs;
|
||
temp_rtp_dcs.swap(rtp_data_channels_);
|
||
for (const auto& kv : temp_rtp_dcs) {
|
||
kv.second->OnTransportChannelDestroyed();
|
||
}
|
||
|
||
std::vector<rtc::scoped_refptr<DataChannel>> temp_sctp_dcs;
|
||
temp_sctp_dcs.swap(sctp_data_channels_);
|
||
for (const auto& channel : temp_sctp_dcs) {
|
||
channel->OnTransportChannelDestroyed();
|
||
}
|
||
}
|
||
|
||
void PeerConnection::OnDataChannelOpenMessage(
|
||
const std::string& label,
|
||
const InternalDataChannelInit& config) {
|
||
rtc::scoped_refptr<DataChannel> channel(
|
||
InternalCreateDataChannel(label, &config));
|
||
if (!channel.get()) {
|
||
RTC_LOG(LS_ERROR) << "Failed to create DataChannel from the OPEN message.";
|
||
return;
|
||
}
|
||
|
||
rtc::scoped_refptr<DataChannelInterface> proxy_channel =
|
||
DataChannelProxy::Create(signaling_thread(), channel);
|
||
Observer()->OnDataChannel(std::move(proxy_channel));
|
||
NoteUsageEvent(UsageEvent::DATA_ADDED);
|
||
}
|
||
|
||
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
||
PeerConnection::GetAudioTransceiver() const {
|
||
// This method only works with Plan B SDP, where there is a single
|
||
// audio/video transceiver.
|
||
RTC_DCHECK(!IsUnifiedPlan());
|
||
for (auto transceiver : transceivers_) {
|
||
if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
|
||
return transceiver;
|
||
}
|
||
}
|
||
RTC_NOTREACHED();
|
||
return nullptr;
|
||
}
|
||
|
||
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
||
PeerConnection::GetVideoTransceiver() const {
|
||
// This method only works with Plan B SDP, where there is a single
|
||
// audio/video transceiver.
|
||
RTC_DCHECK(!IsUnifiedPlan());
|
||
for (auto transceiver : transceivers_) {
|
||
if (transceiver->media_type() == cricket::MEDIA_TYPE_VIDEO) {
|
||
return transceiver;
|
||
}
|
||
}
|
||
RTC_NOTREACHED();
|
||
return nullptr;
|
||
}
|
||
|
||
// TODO(bugs.webrtc.org/7600): Remove this when multiple transceivers with
|
||
// individual transceiver directions are supported.
|
||
bool PeerConnection::HasRtpSender(cricket::MediaType type) const {
|
||
switch (type) {
|
||
case cricket::MEDIA_TYPE_AUDIO:
|
||
return !GetAudioTransceiver()->internal()->senders().empty();
|
||
case cricket::MEDIA_TYPE_VIDEO:
|
||
return !GetVideoTransceiver()->internal()->senders().empty();
|
||
case cricket::MEDIA_TYPE_DATA:
|
||
return false;
|
||
}
|
||
RTC_NOTREACHED();
|
||
return false;
|
||
}
|
||
|
||
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>
|
||
PeerConnection::FindSenderForTrack(MediaStreamTrackInterface* track) const {
|
||
for (auto transceiver : transceivers_) {
|
||
for (auto sender : transceiver->internal()->senders()) {
|
||
if (sender->track() == track) {
|
||
return sender;
|
||
}
|
||
}
|
||
}
|
||
return nullptr;
|
||
}
|
||
|
||
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>
|
||
PeerConnection::FindSenderById(const std::string& sender_id) const {
|
||
for (auto transceiver : transceivers_) {
|
||
for (auto sender : transceiver->internal()->senders()) {
|
||
if (sender->id() == sender_id) {
|
||
return sender;
|
||
}
|
||
}
|
||
}
|
||
return nullptr;
|
||
}
|
||
|
||
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
|
||
PeerConnection::FindReceiverById(const std::string& receiver_id) const {
|
||
for (auto transceiver : transceivers_) {
|
||
for (auto receiver : transceiver->internal()->receivers()) {
|
||
if (receiver->id() == receiver_id) {
|
||
return receiver;
|
||
}
|
||
}
|
||
}
|
||
return nullptr;
|
||
}
|
||
|
||
std::vector<PeerConnection::RtpSenderInfo>*
|
||
PeerConnection::GetRemoteSenderInfos(cricket::MediaType media_type) {
|
||
RTC_DCHECK(media_type == cricket::MEDIA_TYPE_AUDIO ||
|
||
media_type == cricket::MEDIA_TYPE_VIDEO);
|
||
return (media_type == cricket::MEDIA_TYPE_AUDIO)
|
||
? &remote_audio_sender_infos_
|
||
: &remote_video_sender_infos_;
|
||
}
|
||
|
||
std::vector<PeerConnection::RtpSenderInfo>* PeerConnection::GetLocalSenderInfos(
|
||
cricket::MediaType media_type) {
|
||
RTC_DCHECK(media_type == cricket::MEDIA_TYPE_AUDIO ||
|
||
media_type == cricket::MEDIA_TYPE_VIDEO);
|
||
return (media_type == cricket::MEDIA_TYPE_AUDIO) ? &local_audio_sender_infos_
|
||
: &local_video_sender_infos_;
|
||
}
|
||
|
||
const PeerConnection::RtpSenderInfo* PeerConnection::FindSenderInfo(
|
||
const std::vector<PeerConnection::RtpSenderInfo>& infos,
|
||
const std::string& stream_id,
|
||
const std::string sender_id) const {
|
||
for (const RtpSenderInfo& sender_info : infos) {
|
||
if (sender_info.stream_id == stream_id &&
|
||
sender_info.sender_id == sender_id) {
|
||
return &sender_info;
|
||
}
|
||
}
|
||
return nullptr;
|
||
}
|
||
|
||
DataChannel* PeerConnection::FindDataChannelBySid(int sid) const {
|
||
for (const auto& channel : sctp_data_channels_) {
|
||
if (channel->id() == sid) {
|
||
return channel;
|
||
}
|
||
}
|
||
return nullptr;
|
||
}
|
||
|
||
bool PeerConnection::InitializePortAllocator_n(
|
||
const cricket::ServerAddresses& stun_servers,
|
||
const std::vector<cricket::RelayServerConfig>& turn_servers,
|
||
const RTCConfiguration& configuration) {
|
||
port_allocator_->Initialize();
|
||
// To handle both internal and externally created port allocator, we will
|
||
// enable BUNDLE here.
|
||
port_allocator_flags_ = port_allocator_->flags();
|
||
port_allocator_flags_ |= cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET |
|
||
cricket::PORTALLOCATOR_ENABLE_IPV6 |
|
||
cricket::PORTALLOCATOR_ENABLE_IPV6_ON_WIFI;
|
||
// If the disable-IPv6 flag was specified, we'll not override it
|
||
// by experiment.
|
||
if (configuration.disable_ipv6) {
|
||
port_allocator_flags_ &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6);
|
||
} else if (webrtc::field_trial::FindFullName("WebRTC-IPv6Default")
|
||
.find("Disabled") == 0) {
|
||
port_allocator_flags_ &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6);
|
||
}
|
||
|
||
if (configuration.disable_ipv6_on_wifi) {
|
||
port_allocator_flags_ &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6_ON_WIFI);
|
||
RTC_LOG(LS_INFO) << "IPv6 candidates on Wi-Fi are disabled.";
|
||
}
|
||
|
||
if (configuration.tcp_candidate_policy == kTcpCandidatePolicyDisabled) {
|
||
port_allocator_flags_ |= cricket::PORTALLOCATOR_DISABLE_TCP;
|
||
RTC_LOG(LS_INFO) << "TCP candidates are disabled.";
|
||
}
|
||
|
||
if (configuration.candidate_network_policy ==
|
||
kCandidateNetworkPolicyLowCost) {
|
||
port_allocator_flags_ |= cricket::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS;
|
||
RTC_LOG(LS_INFO) << "Do not gather candidates on high-cost networks";
|
||
}
|
||
|
||
if (configuration.disable_link_local_networks) {
|
||
port_allocator_flags_ |= cricket::PORTALLOCATOR_DISABLE_LINK_LOCAL_NETWORKS;
|
||
RTC_LOG(LS_INFO) << "Disable candidates on link-local network interfaces.";
|
||
}
|
||
|
||
port_allocator_->set_flags(port_allocator_flags_);
|
||
// No step delay is used while allocating ports.
|
||
port_allocator_->set_step_delay(cricket::kMinimumStepDelay);
|
||
port_allocator_->set_candidate_filter(
|
||
ConvertIceTransportTypeToCandidateFilter(configuration.type));
|
||
port_allocator_->set_max_ipv6_networks(configuration.max_ipv6_networks);
|
||
|
||
auto turn_servers_copy = turn_servers;
|
||
for (auto& turn_server : turn_servers_copy) {
|
||
turn_server.tls_cert_verifier = tls_cert_verifier_.get();
|
||
}
|
||
// Call this last since it may create pooled allocator sessions using the
|
||
// properties set above.
|
||
port_allocator_->SetConfiguration(
|
||
stun_servers, std::move(turn_servers_copy),
|
||
configuration.ice_candidate_pool_size, configuration.prune_turn_ports,
|
||
configuration.turn_customizer,
|
||
configuration.stun_candidate_keepalive_interval);
|
||
return true;
|
||
}
|
||
|
||
bool PeerConnection::ReconfigurePortAllocator_n(
|
||
const cricket::ServerAddresses& stun_servers,
|
||
const std::vector<cricket::RelayServerConfig>& turn_servers,
|
||
IceTransportsType type,
|
||
int candidate_pool_size,
|
||
bool prune_turn_ports,
|
||
webrtc::TurnCustomizer* turn_customizer,
|
||
absl::optional<int> stun_candidate_keepalive_interval) {
|
||
port_allocator_->set_candidate_filter(
|
||
ConvertIceTransportTypeToCandidateFilter(type));
|
||
// According to JSEP, after setLocalDescription, changing the candidate pool
|
||
// size is not allowed, and changing the set of ICE servers will not result
|
||
// in new candidates being gathered.
|
||
if (local_description()) {
|
||
port_allocator_->FreezeCandidatePool();
|
||
}
|
||
// Add the custom tls turn servers if they exist.
|
||
auto turn_servers_copy = turn_servers;
|
||
for (auto& turn_server : turn_servers_copy) {
|
||
turn_server.tls_cert_verifier = tls_cert_verifier_.get();
|
||
}
|
||
// Call this last since it may create pooled allocator sessions using the
|
||
// candidate filter set above.
|
||
return port_allocator_->SetConfiguration(
|
||
stun_servers, std::move(turn_servers_copy), candidate_pool_size,
|
||
prune_turn_ports, turn_customizer, stun_candidate_keepalive_interval);
|
||
}
|
||
|
||
cricket::ChannelManager* PeerConnection::channel_manager() const {
|
||
return factory_->channel_manager();
|
||
}
|
||
|
||
bool PeerConnection::StartRtcEventLog_w(
|
||
std::unique_ptr<RtcEventLogOutput> output,
|
||
int64_t output_period_ms) {
|
||
if (!event_log_) {
|
||
return false;
|
||
}
|
||
return event_log_->StartLogging(std::move(output), output_period_ms);
|
||
}
|
||
|
||
void PeerConnection::StopRtcEventLog_w() {
|
||
if (event_log_) {
|
||
event_log_->StopLogging();
|
||
}
|
||
}
|
||
|
||
cricket::BaseChannel* PeerConnection::GetChannel(
|
||
const std::string& content_name) {
|
||
for (auto transceiver : transceivers_) {
|
||
cricket::BaseChannel* channel = transceiver->internal()->channel();
|
||
if (channel && channel->content_name() == content_name) {
|
||
return channel;
|
||
}
|
||
}
|
||
if (rtp_data_channel() &&
|
||
rtp_data_channel()->content_name() == content_name) {
|
||
return rtp_data_channel();
|
||
}
|
||
return nullptr;
|
||
}
|
||
|
||
bool PeerConnection::GetSctpSslRole(rtc::SSLRole* role) {
|
||
if (!local_description() || !remote_description()) {
|
||
RTC_LOG(LS_INFO)
|
||
<< "Local and Remote descriptions must be applied to get the "
|
||
"SSL Role of the SCTP transport.";
|
||
return false;
|
||
}
|
||
if (!sctp_transport_) {
|
||
RTC_LOG(LS_INFO) << "Non-rejected SCTP m= section is needed to get the "
|
||
"SSL Role of the SCTP transport.";
|
||
return false;
|
||
}
|
||
|
||
auto dtls_role = transport_controller_->GetDtlsRole(*sctp_mid_);
|
||
if (dtls_role) {
|
||
*role = *dtls_role;
|
||
return true;
|
||
}
|
||
return false;
|
||
}
|
||
|
||
bool PeerConnection::GetSslRole(const std::string& content_name,
|
||
rtc::SSLRole* role) {
|
||
if (!local_description() || !remote_description()) {
|
||
RTC_LOG(LS_INFO)
|
||
<< "Local and Remote descriptions must be applied to get the "
|
||
"SSL Role of the session.";
|
||
return false;
|
||
}
|
||
|
||
auto dtls_role = transport_controller_->GetDtlsRole(content_name);
|
||
if (dtls_role) {
|
||
*role = *dtls_role;
|
||
return true;
|
||
}
|
||
return false;
|
||
}
|
||
|
||
void PeerConnection::SetSessionError(SessionError error,
|
||
const std::string& error_desc) {
|
||
RTC_DCHECK_RUN_ON(signaling_thread());
|
||
if (error != session_error_) {
|
||
session_error_ = error;
|
||
session_error_desc_ = error_desc;
|
||
}
|
||
}
|
||
|
||
RTCError PeerConnection::UpdateSessionState(
|
||
SdpType type,
|
||
cricket::ContentSource source,
|
||
const cricket::SessionDescription* description) {
|
||
RTC_DCHECK_RUN_ON(signaling_thread());
|
||
|
||
// If there's already a pending error then no state transition should happen.
|
||
// But all call-sites should be verifying this before calling us!
|
||
RTC_DCHECK(session_error() == SessionError::kNone);
|
||
|
||
// If this is answer-ish we're ready to let media flow.
|
||
if (type == SdpType::kPrAnswer || type == SdpType::kAnswer) {
|
||
EnableSending();
|
||
}
|
||
|
||
// Update the signaling state according to the specified state machine (see
|
||
// https://w3c.github.io/webrtc-pc/#rtcsignalingstate-enum).
|
||
if (type == SdpType::kOffer) {
|
||
ChangeSignalingState(source == cricket::CS_LOCAL
|
||
? PeerConnectionInterface::kHaveLocalOffer
|
||
: PeerConnectionInterface::kHaveRemoteOffer);
|
||
} else if (type == SdpType::kPrAnswer) {
|
||
ChangeSignalingState(source == cricket::CS_LOCAL
|
||
? PeerConnectionInterface::kHaveLocalPrAnswer
|
||
: PeerConnectionInterface::kHaveRemotePrAnswer);
|
||
} else {
|
||
RTC_DCHECK(type == SdpType::kAnswer);
|
||
ChangeSignalingState(PeerConnectionInterface::kStable);
|
||
}
|
||
|
||
// Update internal objects according to the session description's media
|
||
// descriptions.
|
||
RTCError error = PushdownMediaDescription(type, source);
|
||
if (!error.ok()) {
|
||
return error;
|
||
}
|
||
|
||
return RTCError::OK();
|
||
}
|
||
|
||
RTCError PeerConnection::PushdownMediaDescription(
|
||
SdpType type,
|
||
cricket::ContentSource source) {
|
||
const SessionDescriptionInterface* sdesc =
|
||
(source == cricket::CS_LOCAL ? local_description()
|
||
: remote_description());
|
||
RTC_DCHECK(sdesc);
|
||
|
||
// Push down the new SDP media section for each audio/video transceiver.
|
||
for (auto transceiver : transceivers_) {
|
||
const ContentInfo* content_info =
|
||
FindMediaSectionForTransceiver(transceiver, sdesc);
|
||
cricket::BaseChannel* channel = transceiver->internal()->channel();
|
||
if (!channel || !content_info || content_info->rejected) {
|
||
continue;
|
||
}
|
||
const MediaContentDescription* content_desc =
|
||
content_info->media_description();
|
||
if (!content_desc) {
|
||
continue;
|
||
}
|
||
std::string error;
|
||
bool success = (source == cricket::CS_LOCAL)
|
||
? channel->SetLocalContent(content_desc, type, &error)
|
||
: channel->SetRemoteContent(content_desc, type, &error);
|
||
if (!success) {
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, std::move(error));
|
||
}
|
||
}
|
||
|
||
// If using the RtpDataChannel, push down the new SDP section for it too.
|
||
if (rtp_data_channel_) {
|
||
const ContentInfo* data_content =
|
||
cricket::GetFirstDataContent(sdesc->description());
|
||
if (data_content && !data_content->rejected) {
|
||
const MediaContentDescription* data_desc =
|
||
data_content->media_description();
|
||
if (data_desc) {
|
||
std::string error;
|
||
bool success =
|
||
(source == cricket::CS_LOCAL)
|
||
? rtp_data_channel_->SetLocalContent(data_desc, type, &error)
|
||
: rtp_data_channel_->SetRemoteContent(data_desc, type, &error);
|
||
if (!success) {
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
||
std::move(error));
|
||
}
|
||
}
|
||
}
|
||
}
|
||
|
||
// Need complete offer/answer with an SCTP m= section before starting SCTP,
|
||
// according to https://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-19
|
||
if (sctp_transport_ && local_description() && remote_description() &&
|
||
cricket::GetFirstDataContent(local_description()->description()) &&
|
||
cricket::GetFirstDataContent(remote_description()->description())) {
|
||
bool success = network_thread()->Invoke<bool>(
|
||
RTC_FROM_HERE,
|
||
rtc::Bind(&PeerConnection::PushdownSctpParameters_n, this, source));
|
||
if (!success) {
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
|
||
"Failed to push down SCTP parameters.");
|
||
}
|
||
}
|
||
|
||
return RTCError::OK();
|
||
}
|
||
|
||
bool PeerConnection::PushdownSctpParameters_n(cricket::ContentSource source) {
|
||
RTC_DCHECK(network_thread()->IsCurrent());
|
||
RTC_DCHECK(local_description());
|
||
RTC_DCHECK(remote_description());
|
||
// Apply the SCTP port (which is hidden inside a DataCodec structure...)
|
||
// When we support "max-message-size", that would also be pushed down here.
|
||
return sctp_transport_->Start(
|
||
GetSctpPort(local_description()->description()),
|
||
GetSctpPort(remote_description()->description()));
|
||
}
|
||
|
||
RTCError PeerConnection::PushdownTransportDescription(
|
||
cricket::ContentSource source,
|
||
SdpType type) {
|
||
RTC_DCHECK_RUN_ON(signaling_thread());
|
||
|
||
if (source == cricket::CS_LOCAL) {
|
||
const SessionDescriptionInterface* sdesc = local_description();
|
||
RTC_DCHECK(sdesc);
|
||
return transport_controller_->SetLocalDescription(type,
|
||
sdesc->description());
|
||
} else {
|
||
const SessionDescriptionInterface* sdesc = remote_description();
|
||
RTC_DCHECK(sdesc);
|
||
return transport_controller_->SetRemoteDescription(type,
|
||
sdesc->description());
|
||
}
|
||
}
|
||
|
||
bool PeerConnection::GetTransportDescription(
|
||
const SessionDescription* description,
|
||
const std::string& content_name,
|
||
cricket::TransportDescription* tdesc) {
|
||
if (!description || !tdesc) {
|
||
return false;
|
||
}
|
||
const TransportInfo* transport_info =
|
||
description->GetTransportInfoByName(content_name);
|
||
if (!transport_info) {
|
||
return false;
|
||
}
|
||
*tdesc = transport_info->description;
|
||
return true;
|
||
}
|
||
|
||
cricket::IceConfig PeerConnection::ParseIceConfig(
|
||
const PeerConnectionInterface::RTCConfiguration& config) const {
|
||
cricket::ContinualGatheringPolicy gathering_policy;
|
||
switch (config.continual_gathering_policy) {
|
||
case PeerConnectionInterface::GATHER_ONCE:
|
||
gathering_policy = cricket::GATHER_ONCE;
|
||
break;
|
||
case PeerConnectionInterface::GATHER_CONTINUALLY:
|
||
gathering_policy = cricket::GATHER_CONTINUALLY;
|
||
break;
|
||
default:
|
||
RTC_NOTREACHED();
|
||
gathering_policy = cricket::GATHER_ONCE;
|
||
}
|
||
|
||
cricket::IceConfig ice_config;
|
||
ice_config.receiving_timeout = RTCConfigurationToIceConfigOptionalInt(
|
||
config.ice_connection_receiving_timeout);
|
||
ice_config.prioritize_most_likely_candidate_pairs =
|
||
config.prioritize_most_likely_ice_candidate_pairs;
|
||
ice_config.backup_connection_ping_interval =
|
||
RTCConfigurationToIceConfigOptionalInt(
|
||
config.ice_backup_candidate_pair_ping_interval);
|
||
ice_config.continual_gathering_policy = gathering_policy;
|
||
ice_config.presume_writable_when_fully_relayed =
|
||
config.presume_writable_when_fully_relayed;
|
||
ice_config.ice_check_interval_strong_connectivity =
|
||
config.ice_check_interval_strong_connectivity;
|
||
ice_config.ice_check_interval_weak_connectivity =
|
||
config.ice_check_interval_weak_connectivity;
|
||
ice_config.ice_check_min_interval = config.ice_check_min_interval;
|
||
ice_config.stun_keepalive_interval = config.stun_candidate_keepalive_interval;
|
||
ice_config.regather_all_networks_interval_range =
|
||
config.ice_regather_interval_range;
|
||
ice_config.network_preference = config.network_preference;
|
||
return ice_config;
|
||
}
|
||
|
||
bool PeerConnection::GetLocalTrackIdBySsrc(uint32_t ssrc,
|
||
std::string* track_id) {
|
||
if (!local_description()) {
|
||
return false;
|
||
}
|
||
return webrtc::GetTrackIdBySsrc(local_description()->description(), ssrc,
|
||
track_id);
|
||
}
|
||
|
||
bool PeerConnection::GetRemoteTrackIdBySsrc(uint32_t ssrc,
|
||
std::string* track_id) {
|
||
if (!remote_description()) {
|
||
return false;
|
||
}
|
||
return webrtc::GetTrackIdBySsrc(remote_description()->description(), ssrc,
|
||
track_id);
|
||
}
|
||
|
||
bool PeerConnection::SendData(const cricket::SendDataParams& params,
|
||
const rtc::CopyOnWriteBuffer& payload,
|
||
cricket::SendDataResult* result) {
|
||
if (!rtp_data_channel_ && !sctp_transport_) {
|
||
RTC_LOG(LS_ERROR) << "SendData called when rtp_data_channel_ "
|
||
"and sctp_transport_ are NULL.";
|
||
return false;
|
||
}
|
||
return rtp_data_channel_
|
||
? rtp_data_channel_->SendData(params, payload, result)
|
||
: network_thread()->Invoke<bool>(
|
||
RTC_FROM_HERE,
|
||
Bind(&cricket::SctpTransportInternal::SendData,
|
||
sctp_transport_.get(), params, payload, result));
|
||
}
|
||
|
||
bool PeerConnection::ConnectDataChannel(DataChannel* webrtc_data_channel) {
|
||
if (!rtp_data_channel_ && !sctp_transport_) {
|
||
// Don't log an error here, because DataChannels are expected to call
|
||
// ConnectDataChannel in this state. It's the only way to initially tell
|
||
// whether or not the underlying transport is ready.
|
||
return false;
|
||
}
|
||
if (rtp_data_channel_) {
|
||
rtp_data_channel_->SignalReadyToSendData.connect(
|
||
webrtc_data_channel, &DataChannel::OnChannelReady);
|
||
rtp_data_channel_->SignalDataReceived.connect(webrtc_data_channel,
|
||
&DataChannel::OnDataReceived);
|
||
} else {
|
||
SignalSctpReadyToSendData.connect(webrtc_data_channel,
|
||
&DataChannel::OnChannelReady);
|
||
SignalSctpDataReceived.connect(webrtc_data_channel,
|
||
&DataChannel::OnDataReceived);
|
||
SignalSctpClosingProcedureStartedRemotely.connect(
|
||
webrtc_data_channel, &DataChannel::OnClosingProcedureStartedRemotely);
|
||
SignalSctpClosingProcedureComplete.connect(
|
||
webrtc_data_channel, &DataChannel::OnClosingProcedureComplete);
|
||
}
|
||
return true;
|
||
}
|
||
|
||
void PeerConnection::DisconnectDataChannel(DataChannel* webrtc_data_channel) {
|
||
if (!rtp_data_channel_ && !sctp_transport_) {
|
||
RTC_LOG(LS_ERROR)
|
||
<< "DisconnectDataChannel called when rtp_data_channel_ and "
|
||
"sctp_transport_ are NULL.";
|
||
return;
|
||
}
|
||
if (rtp_data_channel_) {
|
||
rtp_data_channel_->SignalReadyToSendData.disconnect(webrtc_data_channel);
|
||
rtp_data_channel_->SignalDataReceived.disconnect(webrtc_data_channel);
|
||
} else {
|
||
SignalSctpReadyToSendData.disconnect(webrtc_data_channel);
|
||
SignalSctpDataReceived.disconnect(webrtc_data_channel);
|
||
SignalSctpClosingProcedureStartedRemotely.disconnect(webrtc_data_channel);
|
||
SignalSctpClosingProcedureComplete.disconnect(webrtc_data_channel);
|
||
}
|
||
}
|
||
|
||
void PeerConnection::AddSctpDataStream(int sid) {
|
||
if (!sctp_transport_) {
|
||
RTC_LOG(LS_ERROR)
|
||
<< "AddSctpDataStream called when sctp_transport_ is NULL.";
|
||
return;
|
||
}
|
||
network_thread()->Invoke<void>(
|
||
RTC_FROM_HERE, rtc::Bind(&cricket::SctpTransportInternal::OpenStream,
|
||
sctp_transport_.get(), sid));
|
||
}
|
||
|
||
void PeerConnection::RemoveSctpDataStream(int sid) {
|
||
if (!sctp_transport_) {
|
||
RTC_LOG(LS_ERROR) << "RemoveSctpDataStream called when sctp_transport_ is "
|
||
"NULL.";
|
||
return;
|
||
}
|
||
network_thread()->Invoke<void>(
|
||
RTC_FROM_HERE, rtc::Bind(&cricket::SctpTransportInternal::ResetStream,
|
||
sctp_transport_.get(), sid));
|
||
}
|
||
|
||
bool PeerConnection::ReadyToSendData() const {
|
||
return (rtp_data_channel_ && rtp_data_channel_->ready_to_send_data()) ||
|
||
sctp_ready_to_send_data_;
|
||
}
|
||
|
||
absl::optional<std::string> PeerConnection::sctp_transport_name() const {
|
||
if (sctp_mid_ && transport_controller_) {
|
||
auto dtls_transport = transport_controller_->GetDtlsTransport(*sctp_mid_);
|
||
if (dtls_transport) {
|
||
return dtls_transport->transport_name();
|
||
}
|
||
return absl::optional<std::string>();
|
||
}
|
||
return absl::optional<std::string>();
|
||
}
|
||
|
||
cricket::CandidateStatsList PeerConnection::GetPooledCandidateStats() const {
|
||
cricket::CandidateStatsList candidate_states_list;
|
||
network_thread()->Invoke<void>(
|
||
RTC_FROM_HERE,
|
||
rtc::Bind(&cricket::PortAllocator::GetCandidateStatsFromPooledSessions,
|
||
port_allocator_.get(), &candidate_states_list));
|
||
return candidate_states_list;
|
||
}
|
||
|
||
std::map<std::string, std::string> PeerConnection::GetTransportNamesByMid()
|
||
const {
|
||
std::map<std::string, std::string> transport_names_by_mid;
|
||
for (auto transceiver : transceivers_) {
|
||
cricket::BaseChannel* channel = transceiver->internal()->channel();
|
||
if (channel) {
|
||
transport_names_by_mid[channel->content_name()] =
|
||
channel->transport_name();
|
||
}
|
||
}
|
||
if (rtp_data_channel_) {
|
||
transport_names_by_mid[rtp_data_channel_->content_name()] =
|
||
rtp_data_channel_->transport_name();
|
||
}
|
||
if (sctp_transport_) {
|
||
absl::optional<std::string> transport_name = sctp_transport_name();
|
||
RTC_DCHECK(transport_name);
|
||
transport_names_by_mid[*sctp_mid_] = *transport_name;
|
||
}
|
||
return transport_names_by_mid;
|
||
}
|
||
|
||
std::map<std::string, cricket::TransportStats>
|
||
PeerConnection::GetTransportStatsByNames(
|
||
const std::set<std::string>& transport_names) {
|
||
if (!network_thread()->IsCurrent()) {
|
||
return network_thread()
|
||
->Invoke<std::map<std::string, cricket::TransportStats>>(
|
||
RTC_FROM_HERE,
|
||
[&] { return GetTransportStatsByNames(transport_names); });
|
||
}
|
||
std::map<std::string, cricket::TransportStats> transport_stats_by_name;
|
||
for (const std::string& transport_name : transport_names) {
|
||
cricket::TransportStats transport_stats;
|
||
bool success =
|
||
transport_controller_->GetStats(transport_name, &transport_stats);
|
||
if (success) {
|
||
transport_stats_by_name[transport_name] = std::move(transport_stats);
|
||
} else {
|
||
RTC_LOG(LS_ERROR) << "Failed to get transport stats for transport_name="
|
||
<< transport_name;
|
||
}
|
||
}
|
||
return transport_stats_by_name;
|
||
}
|
||
|
||
bool PeerConnection::GetLocalCertificate(
|
||
const std::string& transport_name,
|
||
rtc::scoped_refptr<rtc::RTCCertificate>* certificate) {
|
||
if (!certificate) {
|
||
return false;
|
||
}
|
||
*certificate = transport_controller_->GetLocalCertificate(transport_name);
|
||
return *certificate != nullptr;
|
||
}
|
||
|
||
std::unique_ptr<rtc::SSLCertChain> PeerConnection::GetRemoteSSLCertChain(
|
||
const std::string& transport_name) {
|
||
return transport_controller_->GetRemoteSSLCertChain(transport_name);
|
||
}
|
||
|
||
cricket::DataChannelType PeerConnection::data_channel_type() const {
|
||
return data_channel_type_;
|
||
}
|
||
|
||
bool PeerConnection::IceRestartPending(const std::string& content_name) const {
|
||
return pending_ice_restarts_.find(content_name) !=
|
||
pending_ice_restarts_.end();
|
||
}
|
||
|
||
bool PeerConnection::NeedsIceRestart(const std::string& content_name) const {
|
||
return transport_controller_->NeedsIceRestart(content_name);
|
||
}
|
||
|
||
void PeerConnection::OnCertificateReady(
|
||
const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) {
|
||
transport_controller_->SetLocalCertificate(certificate);
|
||
}
|
||
|
||
void PeerConnection::OnDtlsSrtpSetupFailure(cricket::BaseChannel*, bool rtcp) {
|
||
SetSessionError(SessionError::kTransport,
|
||
rtcp ? kDtlsSrtpSetupFailureRtcp : kDtlsSrtpSetupFailureRtp);
|
||
}
|
||
|
||
void PeerConnection::OnTransportControllerConnectionState(
|
||
cricket::IceConnectionState state) {
|
||
switch (state) {
|
||
case cricket::kIceConnectionConnecting:
|
||
// If the current state is Connected or Completed, then there were
|
||
// writable channels but now there are not, so the next state must
|
||
// be Disconnected.
|
||
// kIceConnectionConnecting is currently used as the default,
|
||
// un-connected state by the TransportController, so its only use is
|
||
// detecting disconnections.
|
||
if (ice_connection_state_ ==
|
||
PeerConnectionInterface::kIceConnectionConnected ||
|
||
ice_connection_state_ ==
|
||
PeerConnectionInterface::kIceConnectionCompleted) {
|
||
SetIceConnectionState(
|
||
PeerConnectionInterface::kIceConnectionDisconnected);
|
||
}
|
||
break;
|
||
case cricket::kIceConnectionFailed:
|
||
SetIceConnectionState(PeerConnectionInterface::kIceConnectionFailed);
|
||
break;
|
||
case cricket::kIceConnectionConnected:
|
||
RTC_LOG(LS_INFO) << "Changing to ICE connected state because "
|
||
"all transports are writable.";
|
||
SetIceConnectionState(PeerConnectionInterface::kIceConnectionConnected);
|
||
NoteUsageEvent(UsageEvent::ICE_STATE_CONNECTED);
|
||
break;
|
||
case cricket::kIceConnectionCompleted:
|
||
RTC_LOG(LS_INFO) << "Changing to ICE completed state because "
|
||
"all transports are complete.";
|
||
if (ice_connection_state_ !=
|
||
PeerConnectionInterface::kIceConnectionConnected) {
|
||
// If jumping directly from "checking" to "connected",
|
||
// signal "connected" first.
|
||
SetIceConnectionState(PeerConnectionInterface::kIceConnectionConnected);
|
||
}
|
||
SetIceConnectionState(PeerConnectionInterface::kIceConnectionCompleted);
|
||
NoteUsageEvent(UsageEvent::ICE_STATE_CONNECTED);
|
||
ReportTransportStats();
|
||
break;
|
||
default:
|
||
RTC_NOTREACHED();
|
||
}
|
||
}
|
||
|
||
void PeerConnection::OnTransportControllerCandidatesGathered(
|
||
const std::string& transport_name,
|
||
const cricket::Candidates& candidates) {
|
||
RTC_DCHECK(signaling_thread()->IsCurrent());
|
||
int sdp_mline_index;
|
||
if (!GetLocalCandidateMediaIndex(transport_name, &sdp_mline_index)) {
|
||
RTC_LOG(LS_ERROR)
|
||
<< "OnTransportControllerCandidatesGathered: content name "
|
||
<< transport_name << " not found";
|
||
return;
|
||
}
|
||
|
||
for (cricket::Candidates::const_iterator citer = candidates.begin();
|
||
citer != candidates.end(); ++citer) {
|
||
// Use transport_name as the candidate media id.
|
||
std::unique_ptr<JsepIceCandidate> candidate(
|
||
new JsepIceCandidate(transport_name, sdp_mline_index, *citer));
|
||
if (local_description()) {
|
||
mutable_local_description()->AddCandidate(candidate.get());
|
||
}
|
||
OnIceCandidate(std::move(candidate));
|
||
}
|
||
}
|
||
|
||
void PeerConnection::OnTransportControllerCandidatesRemoved(
|
||
const std::vector<cricket::Candidate>& candidates) {
|
||
RTC_DCHECK(signaling_thread()->IsCurrent());
|
||
// Sanity check.
|
||
for (const cricket::Candidate& candidate : candidates) {
|
||
if (candidate.transport_name().empty()) {
|
||
RTC_LOG(LS_ERROR) << "OnTransportControllerCandidatesRemoved: "
|
||
"empty content name in candidate "
|
||
<< candidate.ToString();
|
||
return;
|
||
}
|
||
}
|
||
|
||
if (local_description()) {
|
||
mutable_local_description()->RemoveCandidates(candidates);
|
||
}
|
||
OnIceCandidatesRemoved(candidates);
|
||
}
|
||
|
||
void PeerConnection::OnTransportControllerDtlsHandshakeError(
|
||
rtc::SSLHandshakeError error) {
|
||
RTC_HISTOGRAM_ENUMERATION(
|
||
"WebRTC.PeerConnection.DtlsHandshakeError", static_cast<int>(error),
|
||
static_cast<int>(rtc::SSLHandshakeError::MAX_VALUE));
|
||
}
|
||
|
||
void PeerConnection::EnableSending() {
|
||
for (auto transceiver : transceivers_) {
|
||
cricket::BaseChannel* channel = transceiver->internal()->channel();
|
||
if (channel && !channel->enabled()) {
|
||
channel->Enable(true);
|
||
}
|
||
}
|
||
|
||
if (rtp_data_channel_ && !rtp_data_channel_->enabled()) {
|
||
rtp_data_channel_->Enable(true);
|
||
}
|
||
}
|
||
|
||
// Returns the media index for a local ice candidate given the content name.
|
||
bool PeerConnection::GetLocalCandidateMediaIndex(
|
||
const std::string& content_name,
|
||
int* sdp_mline_index) {
|
||
if (!local_description() || !sdp_mline_index) {
|
||
return false;
|
||
}
|
||
|
||
bool content_found = false;
|
||
const ContentInfos& contents = local_description()->description()->contents();
|
||
for (size_t index = 0; index < contents.size(); ++index) {
|
||
if (contents[index].name == content_name) {
|
||
*sdp_mline_index = static_cast<int>(index);
|
||
content_found = true;
|
||
break;
|
||
}
|
||
}
|
||
return content_found;
|
||
}
|
||
|
||
bool PeerConnection::UseCandidatesInSessionDescription(
|
||
const SessionDescriptionInterface* remote_desc) {
|
||
if (!remote_desc) {
|
||
return true;
|
||
}
|
||
bool ret = true;
|
||
|
||
for (size_t m = 0; m < remote_desc->number_of_mediasections(); ++m) {
|
||
const IceCandidateCollection* candidates = remote_desc->candidates(m);
|
||
for (size_t n = 0; n < candidates->count(); ++n) {
|
||
const IceCandidateInterface* candidate = candidates->at(n);
|
||
bool valid = false;
|
||
if (!ReadyToUseRemoteCandidate(candidate, remote_desc, &valid)) {
|
||
if (valid) {
|
||
RTC_LOG(LS_INFO)
|
||
<< "UseCandidatesInSessionDescription: Not ready to use "
|
||
"candidate.";
|
||
}
|
||
continue;
|
||
}
|
||
ret = UseCandidate(candidate);
|
||
if (!ret) {
|
||
break;
|
||
}
|
||
}
|
||
}
|
||
return ret;
|
||
}
|
||
|
||
bool PeerConnection::UseCandidate(const IceCandidateInterface* candidate) {
|
||
size_t mediacontent_index = static_cast<size_t>(candidate->sdp_mline_index());
|
||
size_t remote_content_size =
|
||
remote_description()->description()->contents().size();
|
||
if (mediacontent_index >= remote_content_size) {
|
||
RTC_LOG(LS_ERROR) << "UseCandidate: Invalid candidate media index.";
|
||
return false;
|
||
}
|
||
|
||
cricket::ContentInfo content =
|
||
remote_description()->description()->contents()[mediacontent_index];
|
||
std::vector<cricket::Candidate> candidates;
|
||
candidates.push_back(candidate->candidate());
|
||
// Invoking BaseSession method to handle remote candidates.
|
||
RTCError error =
|
||
transport_controller_->AddRemoteCandidates(content.name, candidates);
|
||
if (error.ok()) {
|
||
// Candidates successfully submitted for checking.
|
||
if (ice_connection_state_ == PeerConnectionInterface::kIceConnectionNew ||
|
||
ice_connection_state_ ==
|
||
PeerConnectionInterface::kIceConnectionDisconnected) {
|
||
// If state is New, then the session has just gotten its first remote ICE
|
||
// candidates, so go to Checking.
|
||
// If state is Disconnected, the session is re-using old candidates or
|
||
// receiving additional ones, so go to Checking.
|
||
// If state is Connected, stay Connected.
|
||
// TODO(bemasc): If state is Connected, and the new candidates are for a
|
||
// newly added transport, then the state actually _should_ move to
|
||
// checking. Add a way to distinguish that case.
|
||
SetIceConnectionState(PeerConnectionInterface::kIceConnectionChecking);
|
||
}
|
||
// TODO(bemasc): If state is Completed, go back to Connected.
|
||
} else {
|
||
RTC_LOG(LS_WARNING) << error.message();
|
||
}
|
||
return true;
|
||
}
|
||
|
||
void PeerConnection::RemoveUnusedChannels(const SessionDescription* desc) {
|
||
// Destroy video channel first since it may have a pointer to the
|
||
// voice channel.
|
||
const cricket::ContentInfo* video_info = cricket::GetFirstVideoContent(desc);
|
||
if (!video_info || video_info->rejected) {
|
||
DestroyTransceiverChannel(GetVideoTransceiver());
|
||
}
|
||
|
||
const cricket::ContentInfo* audio_info = cricket::GetFirstAudioContent(desc);
|
||
if (!audio_info || audio_info->rejected) {
|
||
DestroyTransceiverChannel(GetAudioTransceiver());
|
||
}
|
||
|
||
const cricket::ContentInfo* data_info = cricket::GetFirstDataContent(desc);
|
||
if (!data_info || data_info->rejected) {
|
||
DestroyDataChannel();
|
||
}
|
||
}
|
||
|
||
RTCErrorOr<const cricket::ContentGroup*> PeerConnection::GetEarlyBundleGroup(
|
||
const SessionDescription& desc) const {
|
||
const cricket::ContentGroup* bundle_group = nullptr;
|
||
if (configuration_.bundle_policy ==
|
||
PeerConnectionInterface::kBundlePolicyMaxBundle) {
|
||
bundle_group = desc.GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
|
||
if (!bundle_group) {
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
||
"max-bundle configured but session description "
|
||
"has no BUNDLE group");
|
||
}
|
||
}
|
||
return std::move(bundle_group);
|
||
}
|
||
|
||
RTCError PeerConnection::CreateChannels(const SessionDescription& desc) {
|
||
// Creating the media channels. Transports should already have been created
|
||
// at this point.
|
||
const cricket::ContentInfo* voice = cricket::GetFirstAudioContent(&desc);
|
||
if (voice && !voice->rejected &&
|
||
!GetAudioTransceiver()->internal()->channel()) {
|
||
cricket::VoiceChannel* voice_channel = CreateVoiceChannel(voice->name);
|
||
if (!voice_channel) {
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
|
||
"Failed to create voice channel.");
|
||
}
|
||
GetAudioTransceiver()->internal()->SetChannel(voice_channel);
|
||
}
|
||
|
||
const cricket::ContentInfo* video = cricket::GetFirstVideoContent(&desc);
|
||
if (video && !video->rejected &&
|
||
!GetVideoTransceiver()->internal()->channel()) {
|
||
cricket::VideoChannel* video_channel = CreateVideoChannel(video->name);
|
||
if (!video_channel) {
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
|
||
"Failed to create video channel.");
|
||
}
|
||
GetVideoTransceiver()->internal()->SetChannel(video_channel);
|
||
}
|
||
|
||
const cricket::ContentInfo* data = cricket::GetFirstDataContent(&desc);
|
||
if (data_channel_type_ != cricket::DCT_NONE && data && !data->rejected &&
|
||
!rtp_data_channel_ && !sctp_transport_) {
|
||
if (!CreateDataChannel(data->name)) {
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
|
||
"Failed to create data channel.");
|
||
}
|
||
}
|
||
|
||
return RTCError::OK();
|
||
}
|
||
|
||
// TODO(steveanton): Perhaps this should be managed by the RtpTransceiver.
|
||
cricket::VoiceChannel* PeerConnection::CreateVoiceChannel(
|
||
const std::string& mid) {
|
||
RtpTransportInternal* rtp_transport =
|
||
transport_controller_->GetRtpTransport(mid);
|
||
RTC_DCHECK(rtp_transport);
|
||
cricket::VoiceChannel* voice_channel = channel_manager()->CreateVoiceChannel(
|
||
call_.get(), configuration_.media_config, rtp_transport,
|
||
signaling_thread(), mid, SrtpRequired(),
|
||
factory_->options().crypto_options, audio_options_);
|
||
if (!voice_channel) {
|
||
return nullptr;
|
||
}
|
||
voice_channel->SignalDtlsSrtpSetupFailure.connect(
|
||
this, &PeerConnection::OnDtlsSrtpSetupFailure);
|
||
voice_channel->SignalSentPacket.connect(this,
|
||
&PeerConnection::OnSentPacket_w);
|
||
voice_channel->SetRtpTransport(rtp_transport);
|
||
|
||
return voice_channel;
|
||
}
|
||
|
||
// TODO(steveanton): Perhaps this should be managed by the RtpTransceiver.
|
||
cricket::VideoChannel* PeerConnection::CreateVideoChannel(
|
||
const std::string& mid) {
|
||
RtpTransportInternal* rtp_transport =
|
||
transport_controller_->GetRtpTransport(mid);
|
||
RTC_DCHECK(rtp_transport);
|
||
cricket::VideoChannel* video_channel = channel_manager()->CreateVideoChannel(
|
||
call_.get(), configuration_.media_config, rtp_transport,
|
||
signaling_thread(), mid, SrtpRequired(),
|
||
factory_->options().crypto_options, video_options_);
|
||
if (!video_channel) {
|
||
return nullptr;
|
||
}
|
||
video_channel->SignalDtlsSrtpSetupFailure.connect(
|
||
this, &PeerConnection::OnDtlsSrtpSetupFailure);
|
||
video_channel->SignalSentPacket.connect(this,
|
||
&PeerConnection::OnSentPacket_w);
|
||
video_channel->SetRtpTransport(rtp_transport);
|
||
|
||
return video_channel;
|
||
}
|
||
|
||
bool PeerConnection::CreateDataChannel(const std::string& mid) {
|
||
bool sctp = (data_channel_type_ == cricket::DCT_SCTP);
|
||
if (sctp) {
|
||
if (!sctp_factory_) {
|
||
RTC_LOG(LS_ERROR)
|
||
<< "Trying to create SCTP transport, but didn't compile with "
|
||
"SCTP support (HAVE_SCTP)";
|
||
return false;
|
||
}
|
||
if (!network_thread()->Invoke<bool>(
|
||
RTC_FROM_HERE,
|
||
rtc::Bind(&PeerConnection::CreateSctpTransport_n, this, mid))) {
|
||
return false;
|
||
}
|
||
for (const auto& channel : sctp_data_channels_) {
|
||
channel->OnTransportChannelCreated();
|
||
}
|
||
} else {
|
||
RtpTransportInternal* rtp_transport =
|
||
transport_controller_->GetRtpTransport(mid);
|
||
RTC_DCHECK(rtp_transport);
|
||
rtp_data_channel_ = channel_manager()->CreateRtpDataChannel(
|
||
configuration_.media_config, rtp_transport, signaling_thread(), mid,
|
||
SrtpRequired(), factory_->options().crypto_options);
|
||
if (!rtp_data_channel_) {
|
||
return false;
|
||
}
|
||
rtp_data_channel_->SignalDtlsSrtpSetupFailure.connect(
|
||
this, &PeerConnection::OnDtlsSrtpSetupFailure);
|
||
rtp_data_channel_->SignalSentPacket.connect(
|
||
this, &PeerConnection::OnSentPacket_w);
|
||
rtp_data_channel_->SetRtpTransport(rtp_transport);
|
||
}
|
||
|
||
return true;
|
||
}
|
||
|
||
Call::Stats PeerConnection::GetCallStats() {
|
||
if (!worker_thread()->IsCurrent()) {
|
||
return worker_thread()->Invoke<Call::Stats>(
|
||
RTC_FROM_HERE, rtc::Bind(&PeerConnection::GetCallStats, this));
|
||
}
|
||
if (call_) {
|
||
return call_->GetStats();
|
||
} else {
|
||
return Call::Stats();
|
||
}
|
||
}
|
||
|
||
bool PeerConnection::CreateSctpTransport_n(const std::string& mid) {
|
||
RTC_DCHECK(network_thread()->IsCurrent());
|
||
RTC_DCHECK(sctp_factory_);
|
||
cricket::DtlsTransportInternal* dtls_transport =
|
||
transport_controller_->GetDtlsTransport(mid);
|
||
RTC_DCHECK(dtls_transport);
|
||
sctp_transport_ = sctp_factory_->CreateSctpTransport(dtls_transport);
|
||
RTC_DCHECK(sctp_transport_);
|
||
sctp_invoker_.reset(new rtc::AsyncInvoker());
|
||
sctp_transport_->SignalReadyToSendData.connect(
|
||
this, &PeerConnection::OnSctpTransportReadyToSendData_n);
|
||
sctp_transport_->SignalDataReceived.connect(
|
||
this, &PeerConnection::OnSctpTransportDataReceived_n);
|
||
// TODO(deadbeef): All we do here is AsyncInvoke to fire the signal on
|
||
// another thread. Would be nice if there was a helper class similar to
|
||
// sigslot::repeater that did this for us, eliminating a bunch of boilerplate
|
||
// code.
|
||
sctp_transport_->SignalClosingProcedureStartedRemotely.connect(
|
||
this, &PeerConnection::OnSctpClosingProcedureStartedRemotely_n);
|
||
sctp_transport_->SignalClosingProcedureComplete.connect(
|
||
this, &PeerConnection::OnSctpClosingProcedureComplete_n);
|
||
sctp_mid_ = mid;
|
||
sctp_transport_->SetDtlsTransport(dtls_transport);
|
||
return true;
|
||
}
|
||
|
||
void PeerConnection::DestroySctpTransport_n() {
|
||
RTC_DCHECK(network_thread()->IsCurrent());
|
||
sctp_transport_.reset(nullptr);
|
||
sctp_mid_.reset();
|
||
sctp_invoker_.reset(nullptr);
|
||
sctp_ready_to_send_data_ = false;
|
||
}
|
||
|
||
void PeerConnection::OnSctpTransportReadyToSendData_n() {
|
||
RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP);
|
||
RTC_DCHECK(network_thread()->IsCurrent());
|
||
// Note: Cannot use rtc::Bind here because it will grab a reference to
|
||
// PeerConnection and potentially cause PeerConnection to live longer than
|
||
// expected. It is safe not to grab a reference since the sctp_invoker_ will
|
||
// be destroyed before PeerConnection is destroyed, and at that point all
|
||
// pending tasks will be cleared.
|
||
sctp_invoker_->AsyncInvoke<void>(RTC_FROM_HERE, signaling_thread(), [this] {
|
||
OnSctpTransportReadyToSendData_s(true);
|
||
});
|
||
}
|
||
|
||
void PeerConnection::OnSctpTransportReadyToSendData_s(bool ready) {
|
||
RTC_DCHECK(signaling_thread()->IsCurrent());
|
||
sctp_ready_to_send_data_ = ready;
|
||
SignalSctpReadyToSendData(ready);
|
||
}
|
||
|
||
void PeerConnection::OnSctpTransportDataReceived_n(
|
||
const cricket::ReceiveDataParams& params,
|
||
const rtc::CopyOnWriteBuffer& payload) {
|
||
RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP);
|
||
RTC_DCHECK(network_thread()->IsCurrent());
|
||
// Note: Cannot use rtc::Bind here because it will grab a reference to
|
||
// PeerConnection and potentially cause PeerConnection to live longer than
|
||
// expected. It is safe not to grab a reference since the sctp_invoker_ will
|
||
// be destroyed before PeerConnection is destroyed, and at that point all
|
||
// pending tasks will be cleared.
|
||
sctp_invoker_->AsyncInvoke<void>(
|
||
RTC_FROM_HERE, signaling_thread(), [this, params, payload] {
|
||
OnSctpTransportDataReceived_s(params, payload);
|
||
});
|
||
}
|
||
|
||
void PeerConnection::OnSctpTransportDataReceived_s(
|
||
const cricket::ReceiveDataParams& params,
|
||
const rtc::CopyOnWriteBuffer& payload) {
|
||
RTC_DCHECK(signaling_thread()->IsCurrent());
|
||
if (params.type == cricket::DMT_CONTROL && IsOpenMessage(payload)) {
|
||
// Received OPEN message; parse and signal that a new data channel should
|
||
// be created.
|
||
std::string label;
|
||
InternalDataChannelInit config;
|
||
config.id = params.ssrc;
|
||
if (!ParseDataChannelOpenMessage(payload, &label, &config)) {
|
||
RTC_LOG(LS_WARNING) << "Failed to parse the OPEN message for sid "
|
||
<< params.ssrc;
|
||
return;
|
||
}
|
||
config.open_handshake_role = InternalDataChannelInit::kAcker;
|
||
OnDataChannelOpenMessage(label, config);
|
||
} else {
|
||
// Otherwise just forward the signal.
|
||
SignalSctpDataReceived(params, payload);
|
||
}
|
||
}
|
||
|
||
void PeerConnection::OnSctpClosingProcedureStartedRemotely_n(int sid) {
|
||
RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP);
|
||
RTC_DCHECK(network_thread()->IsCurrent());
|
||
sctp_invoker_->AsyncInvoke<void>(
|
||
RTC_FROM_HERE, signaling_thread(),
|
||
rtc::Bind(&sigslot::signal1<int>::operator(),
|
||
&SignalSctpClosingProcedureStartedRemotely, sid));
|
||
}
|
||
|
||
void PeerConnection::OnSctpClosingProcedureComplete_n(int sid) {
|
||
RTC_DCHECK(data_channel_type_ == cricket::DCT_SCTP);
|
||
RTC_DCHECK(network_thread()->IsCurrent());
|
||
sctp_invoker_->AsyncInvoke<void>(
|
||
RTC_FROM_HERE, signaling_thread(),
|
||
rtc::Bind(&sigslot::signal1<int>::operator(),
|
||
&SignalSctpClosingProcedureComplete, sid));
|
||
}
|
||
|
||
// Returns false if bundle is enabled and rtcp_mux is disabled.
|
||
bool PeerConnection::ValidateBundleSettings(const SessionDescription* desc) {
|
||
bool bundle_enabled = desc->HasGroup(cricket::GROUP_TYPE_BUNDLE);
|
||
if (!bundle_enabled)
|
||
return true;
|
||
|
||
const cricket::ContentGroup* bundle_group =
|
||
desc->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
|
||
RTC_DCHECK(bundle_group != NULL);
|
||
|
||
const cricket::ContentInfos& contents = desc->contents();
|
||
for (cricket::ContentInfos::const_iterator citer = contents.begin();
|
||
citer != contents.end(); ++citer) {
|
||
const cricket::ContentInfo* content = (&*citer);
|
||
RTC_DCHECK(content != NULL);
|
||
if (bundle_group->HasContentName(content->name) && !content->rejected &&
|
||
content->type == MediaProtocolType::kRtp) {
|
||
if (!HasRtcpMuxEnabled(content))
|
||
return false;
|
||
}
|
||
}
|
||
// RTCP-MUX is enabled in all the contents.
|
||
return true;
|
||
}
|
||
|
||
bool PeerConnection::HasRtcpMuxEnabled(const cricket::ContentInfo* content) {
|
||
return content->media_description()->rtcp_mux();
|
||
}
|
||
|
||
RTCError PeerConnection::ValidateSessionDescription(
|
||
const SessionDescriptionInterface* sdesc,
|
||
cricket::ContentSource source) {
|
||
if (session_error() != SessionError::kNone) {
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR, GetSessionErrorMsg());
|
||
}
|
||
|
||
if (!sdesc || !sdesc->description()) {
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, kInvalidSdp);
|
||
}
|
||
|
||
SdpType type = sdesc->GetType();
|
||
if ((source == cricket::CS_LOCAL && !ExpectSetLocalDescription(type)) ||
|
||
(source == cricket::CS_REMOTE && !ExpectSetRemoteDescription(type))) {
|
||
LOG_AND_RETURN_ERROR(
|
||
RTCErrorType::INVALID_STATE,
|
||
"Called in wrong state: " + GetSignalingStateString(signaling_state()));
|
||
}
|
||
|
||
// Verify crypto settings.
|
||
std::string crypto_error;
|
||
if (webrtc_session_desc_factory_->SdesPolicy() == cricket::SEC_REQUIRED ||
|
||
dtls_enabled_) {
|
||
RTCError crypto_error = VerifyCrypto(sdesc->description(), dtls_enabled_);
|
||
if (!crypto_error.ok()) {
|
||
return crypto_error;
|
||
}
|
||
}
|
||
|
||
// Verify ice-ufrag and ice-pwd.
|
||
if (!VerifyIceUfragPwdPresent(sdesc->description())) {
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
||
kSdpWithoutIceUfragPwd);
|
||
}
|
||
|
||
if (!ValidateBundleSettings(sdesc->description())) {
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
||
kBundleWithoutRtcpMux);
|
||
}
|
||
|
||
// TODO(skvlad): When the local rtcp-mux policy is Require, reject any
|
||
// m-lines that do not rtcp-mux enabled.
|
||
|
||
// Verify m-lines in Answer when compared against Offer.
|
||
if (type == SdpType::kPrAnswer || type == SdpType::kAnswer) {
|
||
// With an answer we want to compare the new answer session description with
|
||
// the offer's session description from the current negotiation.
|
||
const cricket::SessionDescription* offer_desc =
|
||
(source == cricket::CS_LOCAL) ? remote_description()->description()
|
||
: local_description()->description();
|
||
if (!MediaSectionsHaveSameCount(*offer_desc, *sdesc->description()) ||
|
||
!MediaSectionsInSameOrder(*offer_desc, nullptr, *sdesc->description(),
|
||
type)) {
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
||
kMlineMismatchInAnswer);
|
||
}
|
||
} else {
|
||
// The re-offers should respect the order of m= sections in current
|
||
// description. See RFC3264 Section 8 paragraph 4 for more details.
|
||
// With a re-offer, either the current local or current remote descriptions
|
||
// could be the most up to date, so we would like to check against both of
|
||
// them if they exist. It could be the case that one of them has a 0 port
|
||
// for a media section, but the other does not. This is important to check
|
||
// against in the case that we are recycling an m= section.
|
||
const cricket::SessionDescription* current_desc = nullptr;
|
||
const cricket::SessionDescription* secondary_current_desc = nullptr;
|
||
if (local_description()) {
|
||
current_desc = local_description()->description();
|
||
if (remote_description()) {
|
||
secondary_current_desc = remote_description()->description();
|
||
}
|
||
} else if (remote_description()) {
|
||
current_desc = remote_description()->description();
|
||
}
|
||
if (current_desc &&
|
||
!MediaSectionsInSameOrder(*current_desc, secondary_current_desc,
|
||
*sdesc->description(), type)) {
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
||
kMlineMismatchInSubsequentOffer);
|
||
}
|
||
}
|
||
|
||
if (IsUnifiedPlan()) {
|
||
// Ensure that each audio and video media section has at most one
|
||
// "StreamParams". This will return an error if receiving a session
|
||
// description from a "Plan B" endpoint which adds multiple tracks of the
|
||
// same type. With Unified Plan, there can only be at most one track per
|
||
// media section.
|
||
for (const ContentInfo& content : sdesc->description()->contents()) {
|
||
const MediaContentDescription& desc = *content.description;
|
||
if ((desc.type() == cricket::MEDIA_TYPE_AUDIO ||
|
||
desc.type() == cricket::MEDIA_TYPE_VIDEO) &&
|
||
desc.streams().size() > 1u) {
|
||
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
||
"Media section has more than one track specified "
|
||
"with a=ssrc lines which is not supported with "
|
||
"Unified Plan.");
|
||
}
|
||
}
|
||
}
|
||
|
||
return RTCError::OK();
|
||
}
|
||
|
||
bool PeerConnection::ExpectSetLocalDescription(SdpType type) {
|
||
PeerConnectionInterface::SignalingState state = signaling_state();
|
||
if (type == SdpType::kOffer) {
|
||
return (state == PeerConnectionInterface::kStable) ||
|
||
(state == PeerConnectionInterface::kHaveLocalOffer);
|
||
} else {
|
||
RTC_DCHECK(type == SdpType::kPrAnswer || type == SdpType::kAnswer);
|
||
return (state == PeerConnectionInterface::kHaveRemoteOffer) ||
|
||
(state == PeerConnectionInterface::kHaveLocalPrAnswer);
|
||
}
|
||
}
|
||
|
||
bool PeerConnection::ExpectSetRemoteDescription(SdpType type) {
|
||
PeerConnectionInterface::SignalingState state = signaling_state();
|
||
if (type == SdpType::kOffer) {
|
||
return (state == PeerConnectionInterface::kStable) ||
|
||
(state == PeerConnectionInterface::kHaveRemoteOffer);
|
||
} else {
|
||
RTC_DCHECK(type == SdpType::kPrAnswer || type == SdpType::kAnswer);
|
||
return (state == PeerConnectionInterface::kHaveLocalOffer) ||
|
||
(state == PeerConnectionInterface::kHaveRemotePrAnswer);
|
||
}
|
||
}
|
||
|
||
const char* PeerConnection::SessionErrorToString(SessionError error) const {
|
||
switch (error) {
|
||
case SessionError::kNone:
|
||
return "ERROR_NONE";
|
||
case SessionError::kContent:
|
||
return "ERROR_CONTENT";
|
||
case SessionError::kTransport:
|
||
return "ERROR_TRANSPORT";
|
||
}
|
||
RTC_NOTREACHED();
|
||
return "";
|
||
}
|
||
|
||
std::string PeerConnection::GetSessionErrorMsg() {
|
||
rtc::StringBuilder desc;
|
||
desc << kSessionError << SessionErrorToString(session_error()) << ". ";
|
||
desc << kSessionErrorDesc << session_error_desc() << ".";
|
||
return desc.Release();
|
||
}
|
||
|
||
void PeerConnection::ReportSdpFormatReceived(
|
||
const SessionDescriptionInterface& remote_offer) {
|
||
int num_audio_mlines = 0;
|
||
int num_video_mlines = 0;
|
||
int num_audio_tracks = 0;
|
||
int num_video_tracks = 0;
|
||
for (const ContentInfo& content : remote_offer.description()->contents()) {
|
||
cricket::MediaType media_type = content.media_description()->type();
|
||
int num_tracks = std::max(
|
||
1, static_cast<int>(content.media_description()->streams().size()));
|
||
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
|
||
num_audio_mlines += 1;
|
||
num_audio_tracks += num_tracks;
|
||
} else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
|
||
num_video_mlines += 1;
|
||
num_video_tracks += num_tracks;
|
||
}
|
||
}
|
||
SdpFormatReceived format = kSdpFormatReceivedNoTracks;
|
||
if (num_audio_mlines > 1 || num_video_mlines > 1) {
|
||
format = kSdpFormatReceivedComplexUnifiedPlan;
|
||
} else if (num_audio_tracks > 1 || num_video_tracks > 1) {
|
||
format = kSdpFormatReceivedComplexPlanB;
|
||
} else if (num_audio_tracks > 0 || num_video_tracks > 0) {
|
||
format = kSdpFormatReceivedSimple;
|
||
}
|
||
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.SdpFormatReceived", format,
|
||
kSdpFormatReceivedMax);
|
||
}
|
||
|
||
void PeerConnection::NoteUsageEvent(UsageEvent event) {
|
||
RTC_DCHECK_RUN_ON(signaling_thread());
|
||
usage_event_accumulator_ |= static_cast<int>(event);
|
||
}
|
||
|
||
void PeerConnection::ReportUsagePattern() const {
|
||
RTC_DLOG(LS_INFO) << "Usage signature is " << usage_event_accumulator_;
|
||
RTC_HISTOGRAM_ENUMERATION_SPARSE("WebRTC.PeerConnection.UsagePattern",
|
||
usage_event_accumulator_,
|
||
static_cast<int>(UsageEvent::MAX_VALUE));
|
||
const int bad_bits =
|
||
static_cast<int>(UsageEvent::SET_LOCAL_DESCRIPTION_CALLED) |
|
||
static_cast<int>(UsageEvent::CANDIDATE_COLLECTED);
|
||
const int good_bits =
|
||
static_cast<int>(UsageEvent::SET_REMOTE_DESCRIPTION_CALLED) |
|
||
static_cast<int>(UsageEvent::REMOTE_CANDIDATE_ADDED) |
|
||
static_cast<int>(UsageEvent::ICE_STATE_CONNECTED);
|
||
if ((usage_event_accumulator_ & bad_bits) == bad_bits &&
|
||
(usage_event_accumulator_ & good_bits) == 0) {
|
||
// If called after close(), we can't report, because observer may have
|
||
// been deallocated, and therefore pointer is null. Write to log instead.
|
||
if (observer_) {
|
||
Observer()->OnInterestingUsage(usage_event_accumulator_);
|
||
} else {
|
||
RTC_LOG(LS_INFO) << "Interesting usage signature "
|
||
<< usage_event_accumulator_
|
||
<< " observed after observer shutdown";
|
||
}
|
||
}
|
||
}
|
||
|
||
void PeerConnection::ReportNegotiatedSdpSemantics(
|
||
const SessionDescriptionInterface& answer) {
|
||
SdpSemanticNegotiated semantics_negotiated;
|
||
switch (answer.description()->msid_signaling()) {
|
||
case 0:
|
||
semantics_negotiated = kSdpSemanticNegotiatedNone;
|
||
break;
|
||
case cricket::kMsidSignalingMediaSection:
|
||
semantics_negotiated = kSdpSemanticNegotiatedUnifiedPlan;
|
||
break;
|
||
case cricket::kMsidSignalingSsrcAttribute:
|
||
semantics_negotiated = kSdpSemanticNegotiatedPlanB;
|
||
break;
|
||
case cricket::kMsidSignalingMediaSection |
|
||
cricket::kMsidSignalingSsrcAttribute:
|
||
semantics_negotiated = kSdpSemanticNegotiatedMixed;
|
||
break;
|
||
default:
|
||
RTC_NOTREACHED();
|
||
}
|
||
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.SdpSemanticNegotiated",
|
||
semantics_negotiated, kSdpSemanticNegotiatedMax);
|
||
}
|
||
|
||
// We need to check the local/remote description for the Transport instead of
|
||
// the session, because a new Transport added during renegotiation may have
|
||
// them unset while the session has them set from the previous negotiation.
|
||
// Not doing so may trigger the auto generation of transport description and
|
||
// mess up DTLS identity information, ICE credential, etc.
|
||
bool PeerConnection::ReadyToUseRemoteCandidate(
|
||
const IceCandidateInterface* candidate,
|
||
const SessionDescriptionInterface* remote_desc,
|
||
bool* valid) {
|
||
*valid = true;
|
||
|
||
const SessionDescriptionInterface* current_remote_desc =
|
||
remote_desc ? remote_desc : remote_description();
|
||
|
||
if (!current_remote_desc) {
|
||
return false;
|
||
}
|
||
|
||
size_t mediacontent_index = static_cast<size_t>(candidate->sdp_mline_index());
|
||
size_t remote_content_size =
|
||
current_remote_desc->description()->contents().size();
|
||
if (mediacontent_index >= remote_content_size) {
|
||
RTC_LOG(LS_ERROR)
|
||
<< "ReadyToUseRemoteCandidate: Invalid candidate media index "
|
||
<< mediacontent_index;
|
||
|
||
*valid = false;
|
||
return false;
|
||
}
|
||
|
||
cricket::ContentInfo content =
|
||
current_remote_desc->description()->contents()[mediacontent_index];
|
||
|
||
const std::string transport_name = GetTransportName(content.name);
|
||
if (transport_name.empty()) {
|
||
return false;
|
||
}
|
||
return true;
|
||
}
|
||
|
||
bool PeerConnection::SrtpRequired() const {
|
||
return dtls_enabled_ ||
|
||
webrtc_session_desc_factory_->SdesPolicy() == cricket::SEC_REQUIRED;
|
||
}
|
||
|
||
void PeerConnection::OnTransportControllerGatheringState(
|
||
cricket::IceGatheringState state) {
|
||
RTC_DCHECK(signaling_thread()->IsCurrent());
|
||
if (state == cricket::kIceGatheringGathering) {
|
||
OnIceGatheringChange(PeerConnectionInterface::kIceGatheringGathering);
|
||
} else if (state == cricket::kIceGatheringComplete) {
|
||
OnIceGatheringChange(PeerConnectionInterface::kIceGatheringComplete);
|
||
}
|
||
}
|
||
|
||
void PeerConnection::ReportTransportStats() {
|
||
std::map<std::string, std::set<cricket::MediaType>>
|
||
media_types_by_transport_name;
|
||
for (auto transceiver : transceivers_) {
|
||
if (transceiver->internal()->channel()) {
|
||
const std::string& transport_name =
|
||
transceiver->internal()->channel()->transport_name();
|
||
media_types_by_transport_name[transport_name].insert(
|
||
transceiver->media_type());
|
||
}
|
||
}
|
||
if (rtp_data_channel()) {
|
||
media_types_by_transport_name[rtp_data_channel()->transport_name()].insert(
|
||
cricket::MEDIA_TYPE_DATA);
|
||
}
|
||
|
||
absl::optional<std::string> transport_name = sctp_transport_name();
|
||
if (transport_name) {
|
||
media_types_by_transport_name[*transport_name].insert(
|
||
cricket::MEDIA_TYPE_DATA);
|
||
}
|
||
|
||
for (const auto& entry : media_types_by_transport_name) {
|
||
const std::string& transport_name = entry.first;
|
||
const std::set<cricket::MediaType> media_types = entry.second;
|
||
cricket::TransportStats stats;
|
||
if (transport_controller_->GetStats(transport_name, &stats)) {
|
||
ReportBestConnectionState(stats);
|
||
ReportNegotiatedCiphers(stats, media_types);
|
||
}
|
||
}
|
||
}
|
||
// Walk through the ConnectionInfos to gather best connection usage
|
||
// for IPv4 and IPv6.
|
||
void PeerConnection::ReportBestConnectionState(
|
||
const cricket::TransportStats& stats) {
|
||
for (const cricket::TransportChannelStats& channel_stats :
|
||
stats.channel_stats) {
|
||
for (const cricket::ConnectionInfo& connection_info :
|
||
channel_stats.connection_infos) {
|
||
if (!connection_info.best_connection) {
|
||
continue;
|
||
}
|
||
|
||
const cricket::Candidate& local = connection_info.local_candidate;
|
||
const cricket::Candidate& remote = connection_info.remote_candidate;
|
||
|
||
// Increment the counter for IceCandidatePairType.
|
||
if (local.protocol() == cricket::TCP_PROTOCOL_NAME ||
|
||
(local.type() == RELAY_PORT_TYPE &&
|
||
local.relay_protocol() == cricket::TCP_PROTOCOL_NAME)) {
|
||
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.CandidatePairType_TCP",
|
||
GetIceCandidatePairCounter(local, remote),
|
||
kIceCandidatePairMax);
|
||
} else if (local.protocol() == cricket::UDP_PROTOCOL_NAME) {
|
||
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.CandidatePairType_UDP",
|
||
GetIceCandidatePairCounter(local, remote),
|
||
kIceCandidatePairMax);
|
||
} else {
|
||
RTC_CHECK(0);
|
||
}
|
||
|
||
// Increment the counter for IP type.
|
||
if (local.address().family() == AF_INET) {
|
||
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.IPMetrics",
|
||
kBestConnections_IPv4,
|
||
kPeerConnectionAddressFamilyCounter_Max);
|
||
} else if (local.address().family() == AF_INET6) {
|
||
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.IPMetrics",
|
||
kBestConnections_IPv6,
|
||
kPeerConnectionAddressFamilyCounter_Max);
|
||
} else {
|
||
RTC_CHECK(0);
|
||
}
|
||
|
||
return;
|
||
}
|
||
}
|
||
}
|
||
|
||
void PeerConnection::ReportNegotiatedCiphers(
|
||
const cricket::TransportStats& stats,
|
||
const std::set<cricket::MediaType>& media_types) {
|
||
if (!dtls_enabled_ || stats.channel_stats.empty()) {
|
||
return;
|
||
}
|
||
|
||
int srtp_crypto_suite = stats.channel_stats[0].srtp_crypto_suite;
|
||
int ssl_cipher_suite = stats.channel_stats[0].ssl_cipher_suite;
|
||
if (srtp_crypto_suite == rtc::SRTP_INVALID_CRYPTO_SUITE &&
|
||
ssl_cipher_suite == rtc::TLS_NULL_WITH_NULL_NULL) {
|
||
return;
|
||
}
|
||
|
||
if (srtp_crypto_suite != rtc::SRTP_INVALID_CRYPTO_SUITE) {
|
||
for (cricket::MediaType media_type : media_types) {
|
||
switch (media_type) {
|
||
case cricket::MEDIA_TYPE_AUDIO:
|
||
RTC_HISTOGRAM_ENUMERATION_SPARSE(
|
||
"WebRTC.PeerConnection.SrtpCryptoSuite.Audio", srtp_crypto_suite,
|
||
rtc::SRTP_CRYPTO_SUITE_MAX_VALUE);
|
||
break;
|
||
case cricket::MEDIA_TYPE_VIDEO:
|
||
RTC_HISTOGRAM_ENUMERATION_SPARSE(
|
||
"WebRTC.PeerConnection.SrtpCryptoSuite.Video", srtp_crypto_suite,
|
||
rtc::SRTP_CRYPTO_SUITE_MAX_VALUE);
|
||
break;
|
||
case cricket::MEDIA_TYPE_DATA:
|
||
RTC_HISTOGRAM_ENUMERATION_SPARSE(
|
||
"WebRTC.PeerConnection.SrtpCryptoSuite.Data", srtp_crypto_suite,
|
||
rtc::SRTP_CRYPTO_SUITE_MAX_VALUE);
|
||
break;
|
||
default:
|
||
RTC_NOTREACHED();
|
||
continue;
|
||
}
|
||
}
|
||
}
|
||
|
||
if (ssl_cipher_suite != rtc::TLS_NULL_WITH_NULL_NULL) {
|
||
for (cricket::MediaType media_type : media_types) {
|
||
switch (media_type) {
|
||
case cricket::MEDIA_TYPE_AUDIO:
|
||
RTC_HISTOGRAM_ENUMERATION_SPARSE(
|
||
"WebRTC.PeerConnection.SslCipherSuite.Audio", ssl_cipher_suite,
|
||
rtc::SSL_CIPHER_SUITE_MAX_VALUE);
|
||
break;
|
||
case cricket::MEDIA_TYPE_VIDEO:
|
||
RTC_HISTOGRAM_ENUMERATION_SPARSE(
|
||
"WebRTC.PeerConnection.SslCipherSuite.Video", ssl_cipher_suite,
|
||
rtc::SSL_CIPHER_SUITE_MAX_VALUE);
|
||
break;
|
||
case cricket::MEDIA_TYPE_DATA:
|
||
RTC_HISTOGRAM_ENUMERATION_SPARSE(
|
||
"WebRTC.PeerConnection.SslCipherSuite.Data", ssl_cipher_suite,
|
||
rtc::SSL_CIPHER_SUITE_MAX_VALUE);
|
||
break;
|
||
default:
|
||
RTC_NOTREACHED();
|
||
continue;
|
||
}
|
||
}
|
||
}
|
||
}
|
||
|
||
void PeerConnection::OnSentPacket_w(const rtc::SentPacket& sent_packet) {
|
||
RTC_DCHECK(worker_thread()->IsCurrent());
|
||
RTC_DCHECK(call_);
|
||
call_->OnSentPacket(sent_packet);
|
||
}
|
||
|
||
const std::string PeerConnection::GetTransportName(
|
||
const std::string& content_name) {
|
||
cricket::BaseChannel* channel = GetChannel(content_name);
|
||
if (channel) {
|
||
return channel->transport_name();
|
||
}
|
||
if (sctp_transport_) {
|
||
RTC_DCHECK(sctp_mid_);
|
||
if (content_name == *sctp_mid_) {
|
||
return *sctp_transport_name();
|
||
}
|
||
}
|
||
// Return an empty string if failed to retrieve the transport name.
|
||
return "";
|
||
}
|
||
|
||
void PeerConnection::DestroyTransceiverChannel(
|
||
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
||
transceiver) {
|
||
RTC_DCHECK(transceiver);
|
||
|
||
cricket::BaseChannel* channel = transceiver->internal()->channel();
|
||
if (channel) {
|
||
transceiver->internal()->SetChannel(nullptr);
|
||
DestroyBaseChannel(channel);
|
||
}
|
||
}
|
||
|
||
void PeerConnection::DestroyDataChannel() {
|
||
if (rtp_data_channel_) {
|
||
OnDataChannelDestroyed();
|
||
DestroyBaseChannel(rtp_data_channel_);
|
||
rtp_data_channel_ = nullptr;
|
||
}
|
||
|
||
// Note: Cannot use rtc::Bind to create a functor to invoke because it will
|
||
// grab a reference to this PeerConnection. If this is called from the
|
||
// PeerConnection destructor, the RefCountedObject vtable will have already
|
||
// been destroyed (since it is a subclass of PeerConnection) and using
|
||
// rtc::Bind will cause "Pure virtual function called" error to appear.
|
||
|
||
if (sctp_transport_) {
|
||
OnDataChannelDestroyed();
|
||
network_thread()->Invoke<void>(RTC_FROM_HERE,
|
||
[this] { DestroySctpTransport_n(); });
|
||
}
|
||
}
|
||
|
||
void PeerConnection::DestroyBaseChannel(cricket::BaseChannel* channel) {
|
||
RTC_DCHECK(channel);
|
||
switch (channel->media_type()) {
|
||
case cricket::MEDIA_TYPE_AUDIO:
|
||
channel_manager()->DestroyVoiceChannel(
|
||
static_cast<cricket::VoiceChannel*>(channel));
|
||
break;
|
||
case cricket::MEDIA_TYPE_VIDEO:
|
||
channel_manager()->DestroyVideoChannel(
|
||
static_cast<cricket::VideoChannel*>(channel));
|
||
break;
|
||
case cricket::MEDIA_TYPE_DATA:
|
||
channel_manager()->DestroyRtpDataChannel(
|
||
static_cast<cricket::RtpDataChannel*>(channel));
|
||
break;
|
||
default:
|
||
RTC_NOTREACHED() << "Unknown media type: " << channel->media_type();
|
||
break;
|
||
}
|
||
}
|
||
|
||
bool PeerConnection::OnTransportChanged(
|
||
const std::string& mid,
|
||
RtpTransportInternal* rtp_transport,
|
||
cricket::DtlsTransportInternal* dtls_transport) {
|
||
bool ret = true;
|
||
auto base_channel = GetChannel(mid);
|
||
if (base_channel) {
|
||
ret = base_channel->SetRtpTransport(rtp_transport);
|
||
}
|
||
if (sctp_transport_ && mid == sctp_mid_) {
|
||
sctp_transport_->SetDtlsTransport(dtls_transport);
|
||
}
|
||
return ret;
|
||
}
|
||
|
||
PeerConnectionObserver* PeerConnection::Observer() const {
|
||
// In earlier production code, the pointer was not cleared on close,
|
||
// which might have led to undefined behavior if the observer was not
|
||
// deallocated, or strange crashes if it was.
|
||
// We use CHECK in order to catch such behavior if it exists.
|
||
// TODO(hta): Remove or replace with DCHECK if nothing is found.
|
||
RTC_CHECK(observer_);
|
||
return observer_;
|
||
}
|
||
|
||
void PeerConnection::ClearStatsCache() {
|
||
if (stats_collector_) {
|
||
stats_collector_->ClearCachedStatsReport();
|
||
}
|
||
}
|
||
|
||
void PeerConnection::RequestUsagePatternReportForTesting() {
|
||
signaling_thread()->Post(RTC_FROM_HERE, this, MSG_REPORT_USAGE_PATTERN,
|
||
nullptr);
|
||
}
|
||
|
||
} // namespace webrtc
|