| Index: webrtc/call/call.cc
|
| diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
|
| index e8b51f9bb10c5b461cd3caf52e97ffcf6fd76986..b7cd059085ed6262da46e51ebe564ba25c6f4b5d 100644
|
| --- a/webrtc/call/call.cc
|
| +++ b/webrtc/call/call.cc
|
| @@ -236,11 +236,11 @@ class Call : public webrtc::Call,
|
| size_t length,
|
| const PacketTime& packet_time);
|
| void ConfigureSync(const std::string& sync_group)
|
| - EXCLUSIVE_LOCKS_REQUIRED(receive_crit_);
|
| + RTC_EXCLUSIVE_LOCKS_REQUIRED(receive_crit_);
|
|
|
| void NotifyBweOfReceivedPacket(const RtpPacketReceived& packet,
|
| MediaType media_type)
|
| - SHARED_LOCKS_REQUIRED(receive_crit_);
|
| + RTC_SHARED_LOCKS_REQUIRED(receive_crit_);
|
|
|
| rtc::Optional<RtpPacketReceived> ParseRtpPacket(
|
| const uint8_t* packet,
|
| @@ -248,7 +248,7 @@ class Call : public webrtc::Call,
|
| const PacketTime* packet_time) const;
|
|
|
| void UpdateSendHistograms(int64_t first_sent_packet_ms)
|
| - EXCLUSIVE_LOCKS_REQUIRED(&bitrate_crit_);
|
| + RTC_EXCLUSIVE_LOCKS_REQUIRED(&bitrate_crit_);
|
| void UpdateReceiveHistograms();
|
| void UpdateHistograms();
|
| void UpdateAggregateNetworkState();
|
| @@ -274,12 +274,12 @@ class Call : public webrtc::Call,
|
| // Audio, Video, and FlexFEC receive streams are owned by the client that
|
| // creates them.
|
| std::set<AudioReceiveStream*> audio_receive_streams_
|
| - GUARDED_BY(receive_crit_);
|
| + RTC_GUARDED_BY(receive_crit_);
|
| std::set<VideoReceiveStream*> video_receive_streams_
|
| - GUARDED_BY(receive_crit_);
|
| + RTC_GUARDED_BY(receive_crit_);
|
|
|
| std::map<std::string, AudioReceiveStream*> sync_stream_mapping_
|
| - GUARDED_BY(receive_crit_);
|
| + RTC_GUARDED_BY(receive_crit_);
|
|
|
| // TODO(nisse): Should eventually be injected at creation,
|
| // with a single object in the bundled case.
|
| @@ -308,19 +308,21 @@ class Call : public webrtc::Call,
|
| bool use_send_side_bwe = false;
|
| };
|
| std::map<uint32_t, ReceiveRtpConfig> receive_rtp_config_
|
| - GUARDED_BY(receive_crit_);
|
| + RTC_GUARDED_BY(receive_crit_);
|
|
|
| std::unique_ptr<RWLockWrapper> send_crit_;
|
| // Audio and Video send streams are owned by the client that creates them.
|
| - std::map<uint32_t, AudioSendStream*> audio_send_ssrcs_ GUARDED_BY(send_crit_);
|
| - std::map<uint32_t, VideoSendStream*> video_send_ssrcs_ GUARDED_BY(send_crit_);
|
| - std::set<VideoSendStream*> video_send_streams_ GUARDED_BY(send_crit_);
|
| + std::map<uint32_t, AudioSendStream*> audio_send_ssrcs_
|
| + RTC_GUARDED_BY(send_crit_);
|
| + std::map<uint32_t, VideoSendStream*> video_send_ssrcs_
|
| + RTC_GUARDED_BY(send_crit_);
|
| + std::set<VideoSendStream*> video_send_streams_ RTC_GUARDED_BY(send_crit_);
|
|
|
| using RtpStateMap = std::map<uint32_t, RtpState>;
|
| RtpStateMap suspended_audio_send_ssrcs_
|
| - GUARDED_BY(configuration_sequence_checker_);
|
| + RTC_GUARDED_BY(configuration_sequence_checker_);
|
| RtpStateMap suspended_video_send_ssrcs_
|
| - GUARDED_BY(configuration_sequence_checker_);
|
| + RTC_GUARDED_BY(configuration_sequence_checker_);
|
|
|
| webrtc::RtcEventLog* event_log_;
|
|
|
| @@ -340,10 +342,11 @@ class Call : public webrtc::Call,
|
| // TODO(holmer): Remove this lock once BitrateController no longer calls
|
| // OnNetworkChanged from multiple threads.
|
| rtc::CriticalSection bitrate_crit_;
|
| - uint32_t min_allocated_send_bitrate_bps_ GUARDED_BY(&bitrate_crit_);
|
| - uint32_t configured_max_padding_bitrate_bps_ GUARDED_BY(&bitrate_crit_);
|
| - AvgCounter estimated_send_bitrate_kbps_counter_ GUARDED_BY(&bitrate_crit_);
|
| - AvgCounter pacer_bitrate_kbps_counter_ GUARDED_BY(&bitrate_crit_);
|
| + uint32_t min_allocated_send_bitrate_bps_ RTC_GUARDED_BY(&bitrate_crit_);
|
| + uint32_t configured_max_padding_bitrate_bps_ RTC_GUARDED_BY(&bitrate_crit_);
|
| + AvgCounter estimated_send_bitrate_kbps_counter_
|
| + RTC_GUARDED_BY(&bitrate_crit_);
|
| + AvgCounter pacer_bitrate_kbps_counter_ RTC_GUARDED_BY(&bitrate_crit_);
|
|
|
| std::map<std::string, rtc::NetworkRoute> network_routes_;
|
|
|
|
|