1 files changed, 96 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_decoding_test.h b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_decoding_test.h
new file mode 100644
index 0000000000..456c397fdd
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/test/neteq_decoding_test.h
@@ -0,0 +1,96 @@
+/*
+ *  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_NETEQ_DECODING_TEST_H_
+#define MODULES_AUDIO_CODING_NETEQ_TEST_NETEQ_DECODING_TEST_H_
+
+#include <memory>
+#include <set>
+#include <string>
+
+#include "absl/strings/string_view.h"
+#include "api/audio/audio_frame.h"
+#include "api/neteq/neteq.h"
+#include "api/rtp_headers.h"
+#include "modules/audio_coding/neteq/tools/packet.h"
+#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
+#include "system_wrappers/include/clock.h"
+#include "test/gtest.h"
+
+namespace webrtc {
+
+class NetEqDecodingTest : public ::testing::Test {
+ protected:
+  // NetEQ must be polled for data once every 10 ms.
+  // Thus, none of the constants below can be changed.
+  static constexpr int kTimeStepMs = 10;
+  static constexpr size_t kBlockSize8kHz = kTimeStepMs * 8;
+  static constexpr size_t kBlockSize16kHz = kTimeStepMs * 16;
+  static constexpr size_t kBlockSize32kHz = kTimeStepMs * 32;
+  static constexpr size_t kBlockSize48kHz = kTimeStepMs * 48;
+  static constexpr int kInitSampleRateHz = 8000;
+
+  NetEqDecodingTest();
+  virtual void SetUp();
+  virtual void TearDown();
+  void OpenInputFile(absl::string_view rtp_file);
+  void Process();
+
+  void DecodeAndCompare(absl::string_view rtp_file,
+                        absl::string_view output_checksum,
+                        absl::string_view network_stats_checksum,
+                        bool gen_ref);
+
+  static void PopulateRtpInfo(int frame_index,
+                              int timestamp,
+                              RTPHeader* rtp_info);
+  static void PopulateCng(int frame_index,
+                          int timestamp,
+                          RTPHeader* rtp_info,
+                          uint8_t* payload,
+                          size_t* payload_len);
+
+  void WrapTest(uint16_t start_seq_no,
+                uint32_t start_timestamp,
+                const std::set<uint16_t>& drop_seq_numbers,
+                bool expect_seq_no_wrap,
+                bool expect_timestamp_wrap);
+
+  void LongCngWithClockDrift(double drift_factor,
+                             double network_freeze_ms,
+                             bool pull_audio_during_freeze,
+                             int delay_tolerance_ms,
+                             int max_time_to_speech_ms);
+
+  SimulatedClock clock_;
+  std::unique_ptr<NetEq> neteq_;
+  NetEq::Config config_;
+  std::unique_ptr<test::RtpFileSource> rtp_source_;
+  std::unique_ptr<test::Packet> packet_;
+  AudioFrame out_frame_;
+  int output_sample_rate_;
+  int algorithmic_delay_ms_;
+};
+
+class NetEqDecodingTestTwoInstances : public NetEqDecodingTest {
+ public:
+  NetEqDecodingTestTwoInstances() : NetEqDecodingTest() {}
+
+  void SetUp() override;
+
+  void CreateSecondInstance();
+
+ protected:
+  std::unique_ptr<NetEq> neteq2_;
+  NetEq::Config config2_;
+};
+
+}  // namespace webrtc
+#endif  // MODULES_AUDIO_CODING_NETEQ_TEST_NETEQ_DECODING_TEST_H_