Chromium Code Reviews| Index: webrtc/api/webrtcsession.cc |
| diff --git a/webrtc/api/webrtcsession.cc b/webrtc/api/webrtcsession.cc |
| index e312c819f82ce4d1c4dd91e818c7bfd5fa745055..c6c7ea23c9843499ea35d6c8f805eddbdbf3dbc5 100644 |
| --- a/webrtc/api/webrtcsession.cc |
| +++ b/webrtc/api/webrtcsession.cc |
| @@ -258,7 +258,7 @@ static bool GetAudioSsrcByTrackId(const SessionDescription* session_description, |
| static bool GetTrackIdBySsrc(const SessionDescription* session_description, |
| uint32_t ssrc, |
| std::string* track_id) { |
| - ASSERT(track_id != NULL); |
| + RTC_DCHECK(track_id != NULL); |
| const cricket::ContentInfo* audio_info = |
| cricket::GetFirstAudioContent(session_description); |
| @@ -502,7 +502,7 @@ WebRtcSession::WebRtcSession( |
| } |
| WebRtcSession::~WebRtcSession() { |
| - ASSERT(signaling_thread()->IsCurrent()); |
| + RTC_DCHECK(signaling_thread()->IsCurrent()); |
| // Destroy video_channel_ first since it may have a pointer to the |
| // voice_channel_. |
| if (video_channel_) { |
| @@ -698,7 +698,7 @@ void WebRtcSession::CreateAnswer( |
| bool WebRtcSession::SetLocalDescription(SessionDescriptionInterface* desc, |
| std::string* err_desc) { |
| - ASSERT(signaling_thread()->IsCurrent()); |
| + RTC_DCHECK(signaling_thread()->IsCurrent()); |
| // Takes the ownership of |desc| regardless of the result. |
| std::unique_ptr<SessionDescriptionInterface> desc_temp(desc); |
| @@ -752,7 +752,7 @@ bool WebRtcSession::SetLocalDescription(SessionDescriptionInterface* desc, |
| bool WebRtcSession::SetRemoteDescription(SessionDescriptionInterface* desc, |
| std::string* err_desc) { |
| - ASSERT(signaling_thread()->IsCurrent()); |
| + RTC_DCHECK(signaling_thread()->IsCurrent()); |
| // Takes the ownership of |desc| regardless of the result. |
| std::unique_ptr<SessionDescriptionInterface> desc_temp(desc); |
| @@ -853,7 +853,7 @@ void WebRtcSession::LogState(State old_state, State new_state) { |
| } |
| void WebRtcSession::SetState(State state) { |
| - ASSERT(signaling_thread_->IsCurrent()); |
| + RTC_DCHECK(signaling_thread_->IsCurrent()); |
| if (state != state_) { |
| LogState(state_, state); |
| state_ = state; |
| @@ -862,7 +862,7 @@ void WebRtcSession::SetState(State state) { |
| } |
| void WebRtcSession::SetError(Error error, const std::string& error_desc) { |
| - ASSERT(signaling_thread_->IsCurrent()); |
| + RTC_DCHECK(signaling_thread_->IsCurrent()); |
| if (error != error_) { |
| error_ = error; |
| error_desc_ = error_desc; |
| @@ -872,11 +872,11 @@ void WebRtcSession::SetError(Error error, const std::string& error_desc) { |
| bool WebRtcSession::UpdateSessionState( |
| Action action, cricket::ContentSource source, |
| std::string* err_desc) { |
| - ASSERT(signaling_thread()->IsCurrent()); |
| + RTC_DCHECK(signaling_thread()->IsCurrent()); |
| // If there's already a pending error then no state transition should happen. |
| // But all call-sites should be verifying this before calling us! |
| - ASSERT(error() == ERROR_NONE); |
| + RTC_DCHECK(error() == ERROR_NONE); |
| std::string td_err; |
| if (action == kOffer) { |
| if (!PushdownTransportDescription(source, cricket::CA_OFFER, &td_err)) { |
| @@ -944,7 +944,7 @@ WebRtcSession::Action WebRtcSession::GetAction(const std::string& type) { |
| } else if (type == SessionDescriptionInterface::kAnswer) { |
| return WebRtcSession::kAnswer; |
| } |
| - ASSERT(false && "unknown action type"); |
| + RTC_DCHECK(false && "unknown action type"); |
|
kwiberg-webrtc
2017/01/12 02:10:41
RTC_NOTREACHED?
(This is the last one I'll commen
nisse-webrtc
2017/01/12 10:53:24
Done. Only three occurrences.
|
| return WebRtcSession::kOffer; |
| } |
| @@ -1240,7 +1240,7 @@ std::string WebRtcSession::BadStateErrMsg(State state) { |
| } |
| bool WebRtcSession::CanInsertDtmf(const std::string& track_id) { |
| - ASSERT(signaling_thread()->IsCurrent()); |
| + RTC_DCHECK(signaling_thread()->IsCurrent()); |
| if (!voice_channel_) { |
| LOG(LS_ERROR) << "CanInsertDtmf: No audio channel exists."; |
| return false; |
| @@ -1259,7 +1259,7 @@ bool WebRtcSession::CanInsertDtmf(const std::string& track_id) { |
| bool WebRtcSession::InsertDtmf(const std::string& track_id, |
| int code, int duration) { |
| - ASSERT(signaling_thread()->IsCurrent()); |
| + RTC_DCHECK(signaling_thread()->IsCurrent()); |
| if (!voice_channel_) { |
| LOG(LS_ERROR) << "InsertDtmf: No audio channel exists."; |
| return false; |
| @@ -1364,7 +1364,7 @@ bool WebRtcSession::ReadyToSendData() const { |
| } |
| std::unique_ptr<SessionStats> WebRtcSession::GetStats_s() { |
| - ASSERT(signaling_thread()->IsCurrent()); |
| + RTC_DCHECK(signaling_thread()->IsCurrent()); |
| ChannelNamePairs channel_name_pairs; |
| if (voice_channel()) { |
| channel_name_pairs.voice = rtc::Optional<ChannelNamePair>(ChannelNamePair( |
| @@ -1524,7 +1524,7 @@ void WebRtcSession::SetIceConnectionReceiving(bool receiving) { |
| void WebRtcSession::OnTransportControllerCandidatesGathered( |
| const std::string& transport_name, |
| const cricket::Candidates& candidates) { |
| - ASSERT(signaling_thread()->IsCurrent()); |
| + RTC_DCHECK(signaling_thread()->IsCurrent()); |
| int sdp_mline_index; |
| if (!GetLocalCandidateMediaIndex(transport_name, &sdp_mline_index)) { |
| LOG(LS_ERROR) << "OnTransportControllerCandidatesGathered: content name " |
| @@ -1547,7 +1547,7 @@ void WebRtcSession::OnTransportControllerCandidatesGathered( |
| void WebRtcSession::OnTransportControllerCandidatesRemoved( |
| const std::vector<cricket::Candidate>& candidates) { |
| - ASSERT(signaling_thread()->IsCurrent()); |
| + RTC_DCHECK(signaling_thread()->IsCurrent()); |
| // Sanity check. |
| for (const cricket::Candidate& candidate : candidates) { |
| if (candidate.transport_name().empty()) { |
| @@ -1872,7 +1872,7 @@ bool WebRtcSession::CreateDataChannel(const cricket::ContentInfo* content, |
| std::unique_ptr<SessionStats> WebRtcSession::GetStats_n( |
| const ChannelNamePairs& channel_name_pairs) { |
| - ASSERT(network_thread()->IsCurrent()); |
| + RTC_DCHECK(network_thread()->IsCurrent()); |
| std::unique_ptr<SessionStats> session_stats(new SessionStats()); |
| for (const auto channel_name_pair : { &channel_name_pairs.voice, |
| &channel_name_pairs.video, |
| @@ -2001,13 +2001,13 @@ bool WebRtcSession::ValidateBundleSettings(const SessionDescription* desc) { |
| const cricket::ContentGroup* bundle_group = |
| desc->GetGroupByName(cricket::GROUP_TYPE_BUNDLE); |
| - ASSERT(bundle_group != NULL); |
| + 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); |
| - ASSERT(content != NULL); |
| + RTC_DCHECK(content != NULL); |
| if (bundle_group->HasContentName(content->name) && |
| !content->rejected && content->type == cricket::NS_JINGLE_RTP) { |
| if (!HasRtcpMuxEnabled(content)) |
| @@ -2160,7 +2160,7 @@ bool WebRtcSession::SrtpRequired() const { |
| void WebRtcSession::OnTransportControllerGatheringState( |
| cricket::IceGatheringState state) { |
| - ASSERT(signaling_thread()->IsCurrent()); |
| + RTC_DCHECK(signaling_thread()->IsCurrent()); |
| if (state == cricket::kIceGatheringGathering) { |
| if (ice_observer_) { |
| ice_observer_->OnIceGatheringChange( |