/* * 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. */ #ifndef MODULES_AUDIO_CODING_NETEQ_TEST_RESULT_SINK_H_ #define MODULES_AUDIO_CODING_NETEQ_TEST_RESULT_SINK_H_ #include #include #include #include "api/neteq/neteq.h" #include "modules/rtp_rtcp/include/rtcp_statistics.h" #include "rtc_base/message_digest.h" namespace webrtc { class ResultSink { public: explicit ResultSink(const std::string& output_file); ~ResultSink(); template 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); private: FILE* output_fp_; std::unique_ptr digest_; }; template void ResultSink::AddResult(const T* test_results, size_t length) { if (output_fp_) { ASSERT_EQ(length, fwrite(test_results, sizeof(T), length, output_fp_)); } digest_->Update(test_results, sizeof(T) * length); } } // namespace webrtc #endif // MODULES_AUDIO_CODING_NETEQ_TEST_RESULT_SINK_H_