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[Unwrap] Use RtpTimestampUnwrapper in audio_ingress

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Bug: webrtc:13982
Change-Id: I748f8e9d5497eac3335b8a9397199ddf24ecc8a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288746
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39035}
This commit is contained in:
Evan Shrubsole 2023-01-09 13:41:33 +00:00 committed by WebRTC LUCI CQ
parent 5d8b49b233
commit 8267cf31ce
3 changed files with 4 additions and 2 deletions
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1
audio/voip/BUILD.gn
View File

@ -70,6 +70,7 @@ rtc_library("audio_ingress") {
"../../modules/rtp_rtcp:rtp_rtcp_format",
"../../rtc_base:criticalsection",
"../../rtc_base:logging",
"../../rtc_base:rtc_numerics",
"../../rtc_base:safe_minmax",
"../../rtc_base:timeutils",
"../../rtc_base/synchronization:mutex",

1
audio/voip/audio_ingress.cc
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@ -23,6 +23,7 @@
#include "modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_minmax.h"
#include "rtc_base/time_utils.h"
namespace webrtc {

4
audio/voip/audio_ingress.h
View File

@ -30,8 +30,8 @@
#include "modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
#include "rtc_base/numerics/sequence_number_unwrapper.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/time_utils.h"
namespace webrtc {
@ -137,7 +137,7 @@ class AudioIngress : public AudioMixer::Source {
// per payload type set when caller set via SetReceiveCodecs.
std::map<int, int> receive_codec_info_ RTC_GUARDED_BY(lock_);
rtc::TimestampWrapAroundHandler timestamp_wrap_handler_ RTC_GUARDED_BY(lock_);
RtpTimestampUnwrapper timestamp_wrap_handler_ RTC_GUARDED_BY(lock_);
};
} // namespace webrtc