diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn index 3820776a81..445b314129 100644 --- a/modules/audio_coding/BUILD.gn +++ b/modules/audio_coding/BUILD.gn @@ -1248,7 +1248,6 @@ rtc_library("audio_coding_modules_tests_shared") { "../../system_wrappers", "../../test:fileutils", "../../test:test_support", - "../rtp_rtcp:rtp_rtcp_format", "//testing/gtest", ] absl_deps = [ diff --git a/modules/audio_coding/neteq/test/result_sink.cc b/modules/audio_coding/neteq/test/result_sink.cc index bb2a59bcfe..b70016180e 100644 --- a/modules/audio_coding/neteq/test/result_sink.cc +++ b/modules/audio_coding/neteq/test/result_sink.cc @@ -47,15 +47,6 @@ void Convert(const webrtc::NetEqNetworkStatistics& stats_raw, stats->set_max_waiting_time_ms(stats_raw.max_waiting_time_ms); } -void Convert(const webrtc::RtcpStatistics& stats_raw, - webrtc::neteq_unittest::RtcpStatistics* stats) { - stats->set_fraction_lost(stats_raw.fraction_lost); - stats->set_cumulative_lost(stats_raw.packets_lost); - stats->set_extended_max_sequence_number( - stats_raw.extended_highest_sequence_number); - stats->set_jitter(stats_raw.jitter); -} - void AddMessage(FILE* file, rtc::MessageDigest* digest, const std::string& message) { @@ -99,19 +90,6 @@ void ResultSink::AddResult(const NetEqNetworkStatistics& stats_raw) { #endif // WEBRTC_NETEQ_UNITTEST_BITEXACT } -void ResultSink::AddResult(const RtcpStatistics& stats_raw) { -#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT - neteq_unittest::RtcpStatistics stats; - Convert(stats_raw, &stats); - - std::string stats_string; - ASSERT_TRUE(stats.SerializeToString(&stats_string)); - AddMessage(output_fp_, digest_.get(), stats_string); -#else - FAIL() << "Writing to reference file requires Proto Buffer."; -#endif // WEBRTC_NETEQ_UNITTEST_BITEXACT -} - void ResultSink::VerifyChecksum(const std::string& checksum) { std::vector buffer; buffer.resize(digest_->Size()); diff --git a/modules/audio_coding/neteq/test/result_sink.h b/modules/audio_coding/neteq/test/result_sink.h index 357b635b08..dcde02d450 100644 --- a/modules/audio_coding/neteq/test/result_sink.h +++ b/modules/audio_coding/neteq/test/result_sink.h @@ -16,7 +16,6 @@ #include #include "api/neteq/neteq.h" -#include "modules/rtp_rtcp/include/rtcp_statistics.h" #include "rtc_base/message_digest.h" namespace webrtc { @@ -30,7 +29,6 @@ class ResultSink { void AddResult(const T* test_results, size_t length); void AddResult(const NetEqNetworkStatistics& stats); - void AddResult(const RtcpStatistics& stats); void VerifyChecksum(const std::string& ref_check_sum);