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_; |