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efeeba0864
Bug: webrtc:15623 Change-Id: Ia184993769f74d51e68a5a536d5fdde26890bcfd Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325481 Commit-Queue: Björn Terelius <terelius@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#41058}
108 lines
3.6 KiB
C++
108 lines
3.6 KiB
C++
/*
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* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_coding/neteq/test/result_sink.h"
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#include <string>
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#include "absl/strings/string_view.h"
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#include "rtc_base/message_digest.h"
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#include "rtc_base/string_encode.h"
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#include "test/gtest.h"
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#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
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#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
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#include "external/webrtc/webrtc/modules/audio_coding/neteq/neteq_unittest.pb.h"
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#else
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#include "modules/audio_coding/neteq/neteq_unittest.pb.h"
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#endif
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#endif
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namespace webrtc {
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#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
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void Convert(const webrtc::NetEqNetworkStatistics& stats_raw,
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webrtc::neteq_unittest::NetEqNetworkStatistics* stats) {
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stats->set_current_buffer_size_ms(stats_raw.current_buffer_size_ms);
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stats->set_preferred_buffer_size_ms(stats_raw.preferred_buffer_size_ms);
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stats->set_jitter_peaks_found(stats_raw.jitter_peaks_found);
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stats->set_expand_rate(stats_raw.expand_rate);
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stats->set_speech_expand_rate(stats_raw.speech_expand_rate);
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stats->set_preemptive_rate(stats_raw.preemptive_rate);
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stats->set_accelerate_rate(stats_raw.accelerate_rate);
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stats->set_secondary_decoded_rate(stats_raw.secondary_decoded_rate);
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stats->set_secondary_discarded_rate(stats_raw.secondary_discarded_rate);
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stats->set_mean_waiting_time_ms(stats_raw.mean_waiting_time_ms);
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stats->set_median_waiting_time_ms(stats_raw.median_waiting_time_ms);
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stats->set_min_waiting_time_ms(stats_raw.min_waiting_time_ms);
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stats->set_max_waiting_time_ms(stats_raw.max_waiting_time_ms);
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}
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void AddMessage(FILE* file,
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rtc::MessageDigest* digest,
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absl::string_view message) {
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int32_t size = message.length();
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if (file)
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ASSERT_EQ(1u, fwrite(&size, sizeof(size), 1, file));
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digest->Update(&size, sizeof(size));
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if (file)
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ASSERT_EQ(static_cast<size_t>(size),
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fwrite(message.data(), sizeof(char), size, file));
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digest->Update(message.data(), sizeof(char) * size);
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}
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#endif // WEBRTC_NETEQ_UNITTEST_BITEXACT
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ResultSink::ResultSink(absl::string_view output_file)
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: output_fp_(nullptr),
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digest_(rtc::MessageDigestFactory::Create(rtc::DIGEST_SHA_1)) {
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if (!output_file.empty()) {
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output_fp_ = fopen(std::string(output_file).c_str(), "wb");
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EXPECT_TRUE(output_fp_ != NULL);
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}
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}
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ResultSink::~ResultSink() {
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if (output_fp_)
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fclose(output_fp_);
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}
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void ResultSink::AddResult(const NetEqNetworkStatistics& stats_raw) {
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#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
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neteq_unittest::NetEqNetworkStatistics stats;
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Convert(stats_raw, &stats);
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std::string stats_string;
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ASSERT_TRUE(stats.SerializeToString(&stats_string));
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AddMessage(output_fp_, digest_.get(), stats_string);
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#else
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FAIL() << "Writing to reference file requires Proto Buffer.";
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#endif // WEBRTC_NETEQ_UNITTEST_BITEXACT
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}
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void ResultSink::VerifyChecksum(absl::string_view checksum) {
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std::string buffer;
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buffer.resize(digest_->Size());
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digest_->Finish(buffer.data(), buffer.size());
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const std::string result = rtc::hex_encode(buffer);
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if (checksum.size() == result.size()) {
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EXPECT_EQ(checksum, result);
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} else {
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// Check result is one the '|'-separated checksums.
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EXPECT_NE(checksum.find(result), absl::string_view::npos)
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<< result << " should be one of these:\n"
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<< checksum;
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}
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}
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} // namespace webrtc
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