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Unified Diff: webrtc/call/call.cc

Issue 3012853002: Update thread annotiation macros to use RTC_ prefix (Closed)
Patch Set: Rebase Created 3 years, 3 months ago
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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_;
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