From d8bbc7858622b6d9c278469aab701ca0b609cddf Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Wed, 15 May 2024 05:35:49 +0200 Subject: Merging upstream version 126.0. Signed-off-by: Daniel Baumann --- third_party/libwebrtc/audio/voip/audio_egress.h | 10 +++++----- 1 file changed, 5 insertions(+), 5 deletions(-) (limited to 'third_party/libwebrtc/audio/voip/audio_egress.h') diff --git a/third_party/libwebrtc/audio/voip/audio_egress.h b/third_party/libwebrtc/audio/voip/audio_egress.h index 989e5bda59..6d1489db34 100644 --- a/third_party/libwebrtc/audio/voip/audio_egress.h +++ b/third_party/libwebrtc/audio/voip/audio_egress.h @@ -16,6 +16,7 @@ #include "api/audio_codecs/audio_format.h" #include "api/sequence_checker.h" +#include "api/task_queue/task_queue_base.h" #include "api/task_queue/task_queue_factory.h" #include "audio/audio_level.h" #include "audio/utility/audio_frame_operations.h" @@ -25,7 +26,7 @@ #include "modules/rtp_rtcp/source/rtp_rtcp_interface.h" #include "modules/rtp_rtcp/source/rtp_sender_audio.h" #include "rtc_base/synchronization/mutex.h" -#include "rtc_base/task_queue.h" +#include "rtc_base/system/no_unique_address.h" #include "rtc_base/time_utils.h" namespace webrtc { @@ -146,11 +147,10 @@ class AudioEgress : public AudioSender, public AudioPacketizationCallback { bool previously_muted_ = false; }; - EncoderContext encoder_context_ RTC_GUARDED_BY(encoder_queue_); + EncoderContext encoder_context_ RTC_GUARDED_BY(encoder_queue_checker_); - // Defined last to ensure that there are no running tasks when the other - // members are destroyed. - rtc::TaskQueue encoder_queue_; + std::unique_ptr encoder_queue_; + RTC_NO_UNIQUE_ADDRESS SequenceChecker encoder_queue_checker_; }; } // namespace webrtc -- cgit v1.2.3