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3a65f392a3
This CL allows to trigger related tests when rolling opus (at chromium side). Namely: * TestOpusBitExactness * TestOpusDtxBitExactness This CL also prevents name clash for OpusTest: * modules/audio_coding/test/opus_test.h: Helper class. * modules/audio_coding/neteq/opus_unittest.cc: Local test fixture. Bug: chromium:1002973 Change-Id: If8470b5f64fbdb1f7a84b838bde62d8c90390f2c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159033 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29759}
95 lines
3.2 KiB
C++
95 lines
3.2 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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#ifndef MODULES_AUDIO_CODING_NETEQ_TEST_NETEQ_DECODING_TEST_H_
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#define MODULES_AUDIO_CODING_NETEQ_TEST_NETEQ_DECODING_TEST_H_
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#include <memory>
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#include <set>
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#include <string>
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#include "api/audio/audio_frame.h"
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#include "api/neteq/neteq.h"
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#include "api/rtp_headers.h"
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#include "modules/audio_coding/neteq/tools/packet.h"
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#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
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#include "system_wrappers/include/clock.h"
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#include "test/gtest.h"
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namespace webrtc {
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class NetEqDecodingTest : public ::testing::Test {
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protected:
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// NetEQ must be polled for data once every 10 ms.
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// Thus, none of the constants below can be changed.
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static constexpr int kTimeStepMs = 10;
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static constexpr size_t kBlockSize8kHz = kTimeStepMs * 8;
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static constexpr size_t kBlockSize16kHz = kTimeStepMs * 16;
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static constexpr size_t kBlockSize32kHz = kTimeStepMs * 32;
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static constexpr size_t kBlockSize48kHz = kTimeStepMs * 48;
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static constexpr int kInitSampleRateHz = 8000;
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NetEqDecodingTest();
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virtual void SetUp();
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virtual void TearDown();
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void OpenInputFile(const std::string& rtp_file);
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void Process();
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void DecodeAndCompare(const std::string& rtp_file,
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const std::string& output_checksum,
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const std::string& network_stats_checksum,
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bool gen_ref);
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static void PopulateRtpInfo(int frame_index,
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int timestamp,
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RTPHeader* rtp_info);
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static void PopulateCng(int frame_index,
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int timestamp,
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RTPHeader* rtp_info,
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uint8_t* payload,
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size_t* payload_len);
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void WrapTest(uint16_t start_seq_no,
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uint32_t start_timestamp,
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const std::set<uint16_t>& drop_seq_numbers,
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bool expect_seq_no_wrap,
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bool expect_timestamp_wrap);
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void LongCngWithClockDrift(double drift_factor,
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double network_freeze_ms,
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bool pull_audio_during_freeze,
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int delay_tolerance_ms,
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int max_time_to_speech_ms);
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SimulatedClock clock_;
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std::unique_ptr<NetEq> neteq_;
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NetEq::Config config_;
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std::unique_ptr<test::RtpFileSource> rtp_source_;
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std::unique_ptr<test::Packet> packet_;
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AudioFrame out_frame_;
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int output_sample_rate_;
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int algorithmic_delay_ms_;
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};
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class NetEqDecodingTestTwoInstances : public NetEqDecodingTest {
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public:
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NetEqDecodingTestTwoInstances() : NetEqDecodingTest() {}
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void SetUp() override;
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void CreateSecondInstance();
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protected:
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std::unique_ptr<NetEq> neteq2_;
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NetEq::Config config2_;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_NETEQ_TEST_NETEQ_DECODING_TEST_H_
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