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Delete unused test method in neteq that uses RtcpStatistics

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Bug: webrtc:10678
Change-Id: I759b635037ab7d2d113fbf8359cdbc46e7712ea5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/218843
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34018}
This commit is contained in:
Danil Chapovalov
2021-05-14 17:48:19 +02:00
committed by WebRTC LUCI CQ
parent 4e60937123
commit 36b7d10a1f
3 changed files with 0 additions and 25 deletions
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1
modules/audio_coding/BUILD.gn
View File

@ -1248,7 +1248,6 @@ rtc_library("audio_coding_modules_tests_shared") {
"../../system_wrappers", "../../system_wrappers",
"../../test:fileutils", "../../test:fileutils",
"../../test:test_support", "../../test:test_support",
"../rtp_rtcp:rtp_rtcp_format",
"//testing/gtest", "//testing/gtest",
] ]
absl_deps = [ absl_deps = [

22
modules/audio_coding/neteq/test/result_sink.cc
View File

@ -47,15 +47,6 @@ void Convert(const webrtc::NetEqNetworkStatistics& stats_raw,
stats->set_max_waiting_time_ms(stats_raw.max_waiting_time_ms); 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, void AddMessage(FILE* file,
rtc::MessageDigest* digest, rtc::MessageDigest* digest,
const std::string& message) { const std::string& message) {
@ -99,19 +90,6 @@ void ResultSink::AddResult(const NetEqNetworkStatistics& stats_raw) {
#endif // WEBRTC_NETEQ_UNITTEST_BITEXACT #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) { void ResultSink::VerifyChecksum(const std::string& checksum) {
std::vector<char> buffer; std::vector<char> buffer;
buffer.resize(digest_->Size()); buffer.resize(digest_->Size());

2
modules/audio_coding/neteq/test/result_sink.h
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@ -16,7 +16,6 @@
#include <string> #include <string>
#include "api/neteq/neteq.h" #include "api/neteq/neteq.h"
#include "modules/rtp_rtcp/include/rtcp_statistics.h"
#include "rtc_base/message_digest.h" #include "rtc_base/message_digest.h"
namespace webrtc { namespace webrtc {
@ -30,7 +29,6 @@ class ResultSink {
void AddResult(const T* test_results, size_t length); void AddResult(const T* test_results, size_t length);
void AddResult(const NetEqNetworkStatistics& stats); void AddResult(const NetEqNetworkStatistics& stats);
void AddResult(const RtcpStatistics& stats);
void VerifyChecksum(const std::string& ref_check_sum); void VerifyChecksum(const std::string& ref_check_sum);