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[Unwrap] Use RtpTimestampUnwrapper in audio/channel_receive

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Bug: webrtc:13982
Change-Id: I02ef68cdda97585a543a1430f19959b589e82002
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288745
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Jakob Ivarsson‎ <jakobi@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39070}
This commit is contained in:
Evan Shrubsole 2023-01-09 13:41:24 +00:00 committed by WebRTC LUCI CQ
parent fa962ffc69
commit 57e5562c3f
2 changed files with 4 additions and 3 deletions
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1
audio/BUILD.gn
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@ -97,6 +97,7 @@ rtc_library("audio") {
"../rtc_base:rate_limiter",
"../rtc_base:refcount",
"../rtc_base:rtc_event",
"../rtc_base:rtc_numerics",
"../rtc_base:rtc_task_queue",
"../rtc_base:safe_conversions",
"../rtc_base:safe_minmax",

6
audio/channel_receive.cc
View File

@ -44,6 +44,7 @@
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_minmax.h"
#include "rtc_base/numerics/sequence_number_unwrapper.h"
#include "rtc_base/race_checker.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/system/no_unique_address.h"
@ -265,7 +266,7 @@ class ChannelReceive : public ChannelReceiveInterface,
mutable Mutex ts_stats_lock_;
std::unique_ptr<rtc::TimestampWrapAroundHandler> rtp_ts_wraparound_handler_;
webrtc::RtpTimestampUnwrapper rtp_ts_wraparound_handler_;
// The rtp timestamp of the first played out audio frame.
int64_t capture_start_rtp_time_stamp_;
// The capture ntp time (in local timebase) of the first played out audio
@ -448,7 +449,7 @@ AudioMixer::Source::AudioFrameInfo ChannelReceive::GetAudioFrameWithInfo(
// audio_frame.timestamp_ should be valid from now on.
// Compute elapsed time.
int64_t unwrap_timestamp =
rtp_ts_wraparound_handler_->Unwrap(audio_frame->timestamp_);
rtp_ts_wraparound_handler_.Unwrap(audio_frame->timestamp_);
audio_frame->elapsed_time_ms_ =
(unwrap_timestamp - capture_start_rtp_time_stamp_) /
(GetRtpTimestampRateHz() / 1000);
@ -554,7 +555,6 @@ ChannelReceive::ChannelReceive(
ntp_estimator_(clock),
playout_timestamp_rtp_(0),
playout_delay_ms_(0),
rtp_ts_wraparound_handler_(new rtc::TimestampWrapAroundHandler()),
capture_start_rtp_time_stamp_(-1),
capture_start_ntp_time_ms_(-1),
_audioDeviceModulePtr(audio_device_module),