From fbaf0bb26397aa498eb9156f06d5a6fe34dd7dd8 Mon Sep 17 00:00:00 2001
From: Daniel Baumann <daniel.baumann@progress-linux.org>
Date: Fri, 19 Apr 2024 03:14:29 +0200
Subject: Merging upstream version 125.0.1.

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
---
 third_party/libwebrtc/pc/rtc_stats_collector.cc | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'third_party/libwebrtc/pc/rtc_stats_collector.cc')

diff --git a/third_party/libwebrtc/pc/rtc_stats_collector.cc b/third_party/libwebrtc/pc/rtc_stats_collector.cc
index 0797ba2a76..2bac176aac 100644
--- a/third_party/libwebrtc/pc/rtc_stats_collector.cc
+++ b/third_party/libwebrtc/pc/rtc_stats_collector.cc
@@ -336,7 +336,7 @@ const char* QualityLimitationReasonToRTCQualityLimitationReason(
 
 std::map<std::string, double>
 QualityLimitationDurationToRTCQualityLimitationDuration(
-    std::map<webrtc::QualityLimitationReason, int64_t> durations_ms) {
+    std::map<QualityLimitationReason, int64_t> durations_ms) {
   std::map<std::string, double> result;
   // The internal duration is defined in milliseconds while the spec defines
   // the value in seconds:
@@ -513,7 +513,7 @@ std::unique_ptr<RTCInboundRtpStreamStats> CreateInboundAudioStreamStats(
 
 std::unique_ptr<RTCAudioPlayoutStats> CreateAudioPlayoutStats(
     const AudioDeviceModule::Stats& audio_device_stats,
-    webrtc::Timestamp timestamp) {
+    Timestamp timestamp) {
   auto stats = std::make_unique<RTCAudioPlayoutStats>(
       /*id=*/kAudioPlayoutSingletonId, timestamp);
   stats->synthesized_samples_duration =
-- 
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