Add const or GUARDED_BY on a few ChannelSend members
Bug: webrtc:9719
Change-Id: I537775b3ca7ebdb06d43b2cca911a221add7d7c9
Reviewed-on: https://webrtc-review.googlesource.com/c/111382
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25706}
diff --git a/audio/channel_send.cc b/audio/channel_send.cc
index 715c93f..b37a535 100644
--- a/audio/channel_send.cc
+++ b/audio/channel_send.cc
@@ -259,7 +259,7 @@
uint16_t send_sequence_number_;
// uses
- ProcessThread* _moduleProcessThreadPtr;
+ ProcessThread* const _moduleProcessThreadPtr;
Transport* _transportPtr; // WebRtc socket or external transport
RmsLevel rms_level_ RTC_GUARDED_BY(encoder_queue_);
bool input_mute_ RTC_GUARDED_BY(volume_settings_critsect_);
@@ -273,13 +273,14 @@
size_t rtp_overhead_per_packet_ RTC_GUARDED_BY(overhead_per_packet_lock_);
rtc::CriticalSection overhead_per_packet_lock_;
// RtcpBandwidthObserver
- std::unique_ptr<VoERtcpObserver> rtcp_observer_;
+ const std::unique_ptr<VoERtcpObserver> rtcp_observer_;
- PacketRouter* packet_router_ = nullptr;
- std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_;
- std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_;
- std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_;
- std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
+ PacketRouter* packet_router_ RTC_GUARDED_BY(&worker_thread_checker_) =
+ nullptr;
+ const std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_;
+ const std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_;
+ const std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_;
+ const std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
rtc::ThreadChecker construction_thread_;
@@ -287,7 +288,7 @@
rtc::CriticalSection encoder_queue_lock_;
bool encoder_queue_is_active_ RTC_GUARDED_BY(encoder_queue_lock_) = false;
- rtc::TaskQueue* encoder_queue_ = nullptr;
+ rtc::TaskQueue* const encoder_queue_ = nullptr;
MediaTransportInterface* const media_transport_;
int media_transport_sequence_number_ RTC_GUARDED_BY(encoder_queue_) = 0;
@@ -306,7 +307,7 @@
// E2EE Audio Frame Encryption
rtc::scoped_refptr<FrameEncryptorInterface> frame_encryptor_;
// E2EE Frame Encryption Options
- webrtc::CryptoOptions crypto_options_;
+ const webrtc::CryptoOptions crypto_options_;
rtc::CriticalSection bitrate_crit_section_;
int configured_bitrate_bps_ RTC_GUARDED_BY(bitrate_crit_section_) = 0;