diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn index 4f04b35d70..0bf6968cb6 100644 --- a/modules/audio_coding/BUILD.gn +++ b/modules/audio_coding/BUILD.gn @@ -883,6 +883,7 @@ rtc_library("neteq_tools") { "../../api:array_view", "../../api/audio_codecs:audio_codecs_api", "../../rtc_base:checks", + "../../rtc_base:rtc_numerics", "../../rtc_base:safe_conversions", "../../rtc_base:stringutils", "../../rtc_base:timeutils", diff --git a/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc b/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc index 020199e9ac..9e77457775 100644 --- a/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc +++ b/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc @@ -18,8 +18,8 @@ #include #include "absl/strings/string_view.h" -#include "modules/include/module_common_types_public.h" #include "rtc_base/checks.h" +#include "rtc_base/numerics/sequence_number_unwrapper.h" namespace webrtc { namespace test { @@ -140,7 +140,7 @@ void NetEqDelayAnalyzer::CreateGraphs(Delays* arrival_delay_ms, std::vector rtp_timestamps_ms; double offset = std::numeric_limits::max(); - TimestampUnwrapper unwrapper; + RtpTimestampUnwrapper unwrapper; // This loop traverses data_ and populates rtp_timestamps_ms as well as // calculates the base offset. for (auto& d : data_) {