Remove mutable from rtc::CriticalSection members.
rtc::CriticalSection is now lockable from const methods and no longer
need to remain mutable.
BUG=
R=tommi@webrtc.org
Review URL: https://codereview.webrtc.org/1613643004
Cr-Commit-Position: refs/heads/master@{#11367}
diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.h b/webrtc/modules/audio_coding/acm2/acm_receiver.h
index 826cb45..d1ca504 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver.h
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.h
@@ -281,7 +281,7 @@
uint32_t NowInTimestamp(int decoder_sampling_rate) const;
- mutable rtc::CriticalSection crit_sect_;
+ rtc::CriticalSection crit_sect_;
int id_; // TODO(henrik.lundin) Make const.
const Decoder* last_audio_decoder_ GUARDED_BY(crit_sect_);
AudioFrame::VADActivity previous_audio_activity_ GUARDED_BY(crit_sect_);
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h
index 2a3bc61..6750a91 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h
@@ -240,7 +240,7 @@
// to |index|.
int UpdateUponReceivingCodec(int index);
- mutable rtc::CriticalSection acm_crit_sect_;
+ rtc::CriticalSection acm_crit_sect_;
rtc::Buffer encode_buffer_ GUARDED_BY(acm_crit_sect_);
int id_; // TODO(henrik.lundin) Make const.
uint32_t expected_codec_ts_ GUARDED_BY(acm_crit_sect_);
@@ -274,7 +274,7 @@
uint32_t last_timestamp_ GUARDED_BY(acm_crit_sect_);
uint32_t last_rtp_timestamp_ GUARDED_BY(acm_crit_sect_);
- mutable rtc::CriticalSection callback_crit_sect_;
+ rtc::CriticalSection callback_crit_sect_;
AudioPacketizationCallback* packetization_callback_
GUARDED_BY(callback_crit_sect_);
ACMVADCallback* vad_callback_ GUARDED_BY(callback_crit_sect_);
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
index c738d0f..384db86 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
@@ -147,7 +147,7 @@
int last_payload_type_ GUARDED_BY(crit_sect_);
uint32_t last_timestamp_ GUARDED_BY(crit_sect_);
std::vector<uint8_t> last_payload_vec_ GUARDED_BY(crit_sect_);
- mutable rtc::CriticalSection crit_sect_;
+ rtc::CriticalSection crit_sect_;
};
class AudioCodingModuleTestOldApi : public ::testing::Test {
@@ -579,7 +579,7 @@
int send_count_;
int insert_packet_count_;
int pull_audio_count_ GUARDED_BY(crit_sect_);
- mutable rtc::CriticalSection crit_sect_;
+ rtc::CriticalSection crit_sect_;
int64_t next_insert_packet_time_ms_ GUARDED_BY(crit_sect_);
rtc::scoped_ptr<SimulatedClock> fake_clock_;
};
@@ -842,7 +842,7 @@
rtc::PlatformThread receive_thread_;
rtc::PlatformThread codec_registration_thread_;
const rtc::scoped_ptr<EventWrapper> test_complete_;
- mutable rtc::CriticalSection crit_sect_;
+ rtc::CriticalSection crit_sect_;
bool codec_registered_ GUARDED_BY(crit_sect_);
int receive_packet_count_ GUARDED_BY(crit_sect_);
int64_t next_insert_packet_time_ms_ GUARDED_BY(crit_sect_);
diff --git a/webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h b/webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h
index 9996cbd..002af8c 100644
--- a/webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h
+++ b/webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h
@@ -36,7 +36,7 @@
}
private:
- mutable rtc::CriticalSection lock_;
+ rtc::CriticalSection lock_;
IsacBandwidthInfo bwinfo_ GUARDED_BY(lock_);
};
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.h b/webrtc/modules/audio_coding/neteq/neteq_impl.h
index 817b697..02adcd3 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.h
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.h
@@ -338,7 +338,7 @@
// Creates DecisionLogic object with the mode given by |playout_mode_|.
virtual void CreateDecisionLogic() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
- mutable rtc::CriticalSection crit_sect_;
+ rtc::CriticalSection crit_sect_;
const rtc::scoped_ptr<BufferLevelFilter> buffer_level_filter_
GUARDED_BY(crit_sect_);
const rtc::scoped_ptr<DecoderDatabase> decoder_database_
diff --git a/webrtc/modules/audio_coding/test/Channel.h b/webrtc/modules/audio_coding/test/Channel.h
index 3dcd499..5910fad 100644
--- a/webrtc/modules/audio_coding/test/Channel.h
+++ b/webrtc/modules/audio_coding/test/Channel.h
@@ -100,7 +100,7 @@
// 60msec * 32 sample(max)/msec * 2 description (maybe) * 2 bytes/sample
uint8_t _payloadData[60 * 32 * 2 * 2];
- mutable rtc::CriticalSection _channelCritSect;
+ rtc::CriticalSection _channelCritSect;
FILE* _bitStreamFile;
bool _saveBitStream;
int16_t _lastPayloadType;