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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "modules/audio_coding/neteq/test/result_sink.h"
#include <vector>
#include "rtc_base/ignore_wundef.h"
#include "rtc_base/message_digest.h"
#include "rtc_base/string_encode.h"
#include "test/gtest.h"
#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
RTC_PUSH_IGNORING_WUNDEF()
#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
#include "external/webrtc/webrtc/modules/audio_coding/neteq/neteq_unittest.pb.h"
#else
#include "modules/audio_coding/neteq/neteq_unittest.pb.h"
#endif
RTC_POP_IGNORING_WUNDEF()
#endif
namespace webrtc {
#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
void Convert(const webrtc::NetEqNetworkStatistics& stats_raw,
webrtc::neteq_unittest::NetEqNetworkStatistics* stats) {
stats->set_current_buffer_size_ms(stats_raw.current_buffer_size_ms);
stats->set_preferred_buffer_size_ms(stats_raw.preferred_buffer_size_ms);
stats->set_jitter_peaks_found(stats_raw.jitter_peaks_found);
stats->set_expand_rate(stats_raw.expand_rate);
stats->set_speech_expand_rate(stats_raw.speech_expand_rate);
stats->set_preemptive_rate(stats_raw.preemptive_rate);
stats->set_accelerate_rate(stats_raw.accelerate_rate);
stats->set_secondary_decoded_rate(stats_raw.secondary_decoded_rate);
stats->set_secondary_discarded_rate(stats_raw.secondary_discarded_rate);
stats->set_mean_waiting_time_ms(stats_raw.mean_waiting_time_ms);
stats->set_median_waiting_time_ms(stats_raw.median_waiting_time_ms);
stats->set_min_waiting_time_ms(stats_raw.min_waiting_time_ms);
stats->set_max_waiting_time_ms(stats_raw.max_waiting_time_ms);
}
void AddMessage(FILE* file,
rtc::MessageDigest* digest,
const std::string& message) {
int32_t size = message.length();
if (file)
ASSERT_EQ(1u, fwrite(&size, sizeof(size), 1, file));
digest->Update(&size, sizeof(size));
if (file)
ASSERT_EQ(static_cast<size_t>(size),
fwrite(message.data(), sizeof(char), size, file));
digest->Update(message.data(), sizeof(char) * size);
}
#endif // WEBRTC_NETEQ_UNITTEST_BITEXACT
ResultSink::ResultSink(const std::string& output_file)
: output_fp_(nullptr),
digest_(rtc::MessageDigestFactory::Create(rtc::DIGEST_SHA_1)) {
if (!output_file.empty()) {
output_fp_ = fopen(output_file.c_str(), "wb");
EXPECT_TRUE(output_fp_ != NULL);
}
}
ResultSink::~ResultSink() {
if (output_fp_)
fclose(output_fp_);
}
void ResultSink::AddResult(const NetEqNetworkStatistics& stats_raw) {
#ifdef WEBRTC_NETEQ_UNITTEST_BITEXACT
neteq_unittest::NetEqNetworkStatistics 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<char> buffer;
buffer.resize(digest_->Size());
digest_->Finish(&buffer[0], buffer.size());
const std::string result = rtc::hex_encode(&buffer[0], digest_->Size());
if (checksum.size() == result.size()) {
EXPECT_EQ(checksum, result);
} else {
// Check result is one the '|'-separated checksums.
EXPECT_NE(checksum.find(result), std::string::npos)
<< result << " should be one of these:\n"
<< checksum;
}
}
} // namespace webrtc
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