From 035ee11f78c99cc8cb0607b6d21087b82ba9dd15 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Niels=20M=C3=B6ller?= Date: Thu, 9 May 2019 09:09:48 +0200 Subject: [PATCH] Delete left-over tests NetEqExternalDecoderUnitTest Related code was deleted in https://webrtc-review.googlesource.com/c/112081. Bug: webrtc:10080 Change-Id: I3adc1238df6e80380cae3403c108403a59fd4a05 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135740 Reviewed-by: Ivo Creusen Commit-Queue: Niels Moller Cr-Commit-Position: refs/heads/master@{#27890} --- .../neteq/neteq_external_decoder_unittest.cc | 451 ------------------ 1 file changed, 451 deletions(-) delete mode 100644 modules/audio_coding/neteq/neteq_external_decoder_unittest.cc diff --git a/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc b/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc deleted file mode 100644 index 8214ed2159..0000000000 --- a/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc +++ /dev/null @@ -1,451 +0,0 @@ -/* - * Copyright (c) 2013 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. - */ - -// Test to verify correct operation for externally created decoders. - -#include - -#include "api/audio/audio_frame.h" -#include "api/audio_codecs/builtin_audio_decoder_factory.h" -#include "modules/audio_coding/neteq/mock/mock_external_decoder_pcm16b.h" -#include "modules/audio_coding/neteq/tools/input_audio_file.h" -#include "modules/audio_coding/neteq/tools/neteq_external_decoder_test.h" -#include "modules/audio_coding/neteq/tools/rtp_generator.h" -#include "rtc_base/strings/string_builder.h" -#include "test/gmock.h" -#include "test/testsupport/file_utils.h" - -namespace webrtc { - -using ::testing::_; -using ::testing::Return; - -class NetEqExternalDecoderUnitTest : public test::NetEqExternalDecoderTest { - protected: - static const int kFrameSizeMs = 10; // Frame size of Pcm16B. - - NetEqExternalDecoderUnitTest(NetEqDecoder codec, - int sample_rate_hz, - MockExternalPcm16B* decoder) - : NetEqExternalDecoderTest(codec, sample_rate_hz, decoder), - external_decoder_(decoder), - samples_per_ms_(sample_rate_hz / 1000), - frame_size_samples_(kFrameSizeMs * samples_per_ms_), - rtp_generator_(new test::RtpGenerator(samples_per_ms_)), - input_(new int16_t[frame_size_samples_]), - // Payload should be no larger than input. - encoded_(new uint8_t[2 * frame_size_samples_]), - payload_size_bytes_(0), - last_send_time_(0), - last_arrival_time_(0) { - // NetEq is not allowed to delete the external decoder (hence Times(0)). - EXPECT_CALL(*external_decoder_, Die()).Times(0); - Init(); - - const std::string file_name = - webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"); - input_file_.reset(new test::InputAudioFile(file_name)); - } - - virtual ~NetEqExternalDecoderUnitTest() { - delete[] input_; - delete[] encoded_; - // ~NetEqExternalDecoderTest() will delete |external_decoder_|, so expecting - // Die() to be called. - EXPECT_CALL(*external_decoder_, Die()).Times(1); - } - - // Method to draw kFrameSizeMs audio and verify the output. - // Use gTest methods. e.g. ASSERT_EQ() inside to trigger errors. - virtual void GetAndVerifyOutput() = 0; - - // Method to get the number of calls to the Decode() method of the external - // decoder. - virtual int NumExpectedDecodeCalls(int num_loops) = 0; - - // Method to generate packets and return the send time of the packet. - int GetNewPacket() { - if (!input_file_->Read(frame_size_samples_, input_)) { - return -1; - } - payload_size_bytes_ = - WebRtcPcm16b_Encode(input_, frame_size_samples_, encoded_); - - int next_send_time = rtp_generator_->GetRtpHeader( - kPayloadType, frame_size_samples_, &rtp_header_); - return next_send_time; - } - - // Method to decide packet losses. - virtual bool Lost() { return false; } - - // Method to calculate packet arrival time. - int GetArrivalTime(int send_time) { - int arrival_time = last_arrival_time_ + (send_time - last_send_time_); - last_send_time_ = send_time; - last_arrival_time_ = arrival_time; - return arrival_time; - } - - void RunTest(int num_loops) { - // Get next input packets (mono and multi-channel). - uint32_t next_send_time; - uint32_t next_arrival_time; - do { - next_send_time = GetNewPacket(); - next_arrival_time = GetArrivalTime(next_send_time); - } while (Lost()); // If lost, immediately read the next packet. - - EXPECT_CALL( - *external_decoder_, - DecodeInternal(_, payload_size_bytes_, 1000 * samples_per_ms_, _, _)) - .Times(NumExpectedDecodeCalls(num_loops)); - - uint32_t time_now = 0; - for (int k = 0; k < num_loops; ++k) { - while (time_now >= next_arrival_time) { - InsertPacket( - rtp_header_, - rtc::ArrayView(encoded_, payload_size_bytes_), - next_arrival_time); - // Get next input packet. - do { - next_send_time = GetNewPacket(); - next_arrival_time = GetArrivalTime(next_send_time); - } while (Lost()); // If lost, immediately read the next packet. - } - - rtc::StringBuilder ss; - ss << "Lap number " << k << "."; - SCOPED_TRACE(ss.str()); // Print out the parameter values on failure. - // Compare mono and multi-channel. - ASSERT_NO_FATAL_FAILURE(GetAndVerifyOutput()); - - time_now += kOutputLengthMs; - } - } - - void InsertPacket(RTPHeader rtp_header, - rtc::ArrayView payload, - uint32_t receive_timestamp) override { - EXPECT_CALL(*external_decoder_, - IncomingPacket(_, payload.size(), rtp_header.sequenceNumber, - rtp_header.timestamp, receive_timestamp)); - NetEqExternalDecoderTest::InsertPacket(rtp_header, payload, - receive_timestamp); - } - - MockExternalPcm16B* external_decoder() { return external_decoder_.get(); } - - void ResetRtpGenerator(test::RtpGenerator* rtp_generator) { - rtp_generator_.reset(rtp_generator); - } - - int samples_per_ms() const { return samples_per_ms_; } - - private: - std::unique_ptr external_decoder_; - int samples_per_ms_; - size_t frame_size_samples_; - std::unique_ptr rtp_generator_; - int16_t* input_; - uint8_t* encoded_; - size_t payload_size_bytes_; - uint32_t last_send_time_; - uint32_t last_arrival_time_; - std::unique_ptr input_file_; - RTPHeader rtp_header_; -}; - -// This test encodes a few packets of PCM16b 32 kHz data and inserts it into two -// different NetEq instances. The first instance uses the internal version of -// the decoder object, while the second one uses an externally created decoder -// object (ExternalPcm16B wrapped in MockExternalPcm16B, both defined above). -// The test verifies that the output from both instances match. -class NetEqExternalVsInternalDecoderTest : public NetEqExternalDecoderUnitTest, - public ::testing::Test { - protected: - static const size_t kMaxBlockSize = 480; // 10 ms @ 48 kHz. - - NetEqExternalVsInternalDecoderTest() - : NetEqExternalDecoderUnitTest(NetEqDecoder::kDecoderPCM16Bswb32kHz, - 32000, - new MockExternalPcm16B(32000)), - sample_rate_hz_(32000) { - NetEq::Config config; - config.sample_rate_hz = sample_rate_hz_; - neteq_internal_.reset( - NetEq::Create(config, CreateBuiltinAudioDecoderFactory())); - } - - void SetUp() override { - ASSERT_EQ(true, neteq_internal_->RegisterPayloadType( - kPayloadType, SdpAudioFormat("L16", 32000, 1))); - } - - void GetAndVerifyOutput() override { - // Get audio from internal decoder instance. - bool muted; - EXPECT_EQ(NetEq::kOK, neteq_internal_->GetAudio(&output_internal_, &muted)); - ASSERT_FALSE(muted); - EXPECT_EQ(1u, output_internal_.num_channels_); - EXPECT_EQ(static_cast(kOutputLengthMs * sample_rate_hz_ / 1000), - output_internal_.samples_per_channel_); - - // Get audio from external decoder instance. - GetOutputAudio(&output_); - - const int16_t* output_data = output_.data(); - const int16_t* output_internal_data = output_internal_.data(); - for (size_t i = 0; i < output_.samples_per_channel_; ++i) { - ASSERT_EQ(output_data[i], output_internal_data[i]) - << "Diff in sample " << i << "."; - } - } - - void InsertPacket(RTPHeader rtp_header, - rtc::ArrayView payload, - uint32_t receive_timestamp) override { - // Insert packet in internal decoder. - ASSERT_EQ(NetEq::kOK, neteq_internal_->InsertPacket(rtp_header, payload, - receive_timestamp)); - - // Insert packet in external decoder instance. - NetEqExternalDecoderUnitTest::InsertPacket(rtp_header, payload, - receive_timestamp); - } - - int NumExpectedDecodeCalls(int num_loops) override { return num_loops; } - - private: - int sample_rate_hz_; - std::unique_ptr neteq_internal_; - AudioFrame output_internal_; - AudioFrame output_; -}; - -TEST_F(NetEqExternalVsInternalDecoderTest, RunTest) { - RunTest(100); // Run 100 laps @ 10 ms each in the test loop. -} - -class LargeTimestampJumpTest : public NetEqExternalDecoderUnitTest, - public ::testing::Test { - protected: - static const size_t kMaxBlockSize = 480; // 10 ms @ 48 kHz. - - enum TestStates { - kInitialPhase, - kNormalPhase, - kExpandPhase, - kFadedExpandPhase, - kRecovered - }; - - LargeTimestampJumpTest() - : NetEqExternalDecoderUnitTest(NetEqDecoder::kDecoderPCM16B, - 8000, - new MockExternalPcm16B(8000)), - test_state_(kInitialPhase) { - EXPECT_CALL(*external_decoder(), HasDecodePlc()) - .WillRepeatedly(Return(false)); - } - - virtual void UpdateState(AudioFrame::SpeechType output_type) { - switch (test_state_) { - case kInitialPhase: { - if (output_type == AudioFrame::kNormalSpeech) { - test_state_ = kNormalPhase; - } - break; - } - case kNormalPhase: { - if (output_type == AudioFrame::kPLC) { - test_state_ = kExpandPhase; - } - break; - } - case kExpandPhase: { - if (output_type == AudioFrame::kPLCCNG) { - test_state_ = kFadedExpandPhase; - } else if (output_type == AudioFrame::kNormalSpeech) { - test_state_ = kRecovered; - } - break; - } - case kFadedExpandPhase: { - if (output_type == AudioFrame::kNormalSpeech) { - test_state_ = kRecovered; - } - break; - } - case kRecovered: { - break; - } - } - } - - void GetAndVerifyOutput() override { - AudioFrame output; - GetOutputAudio(&output); - UpdateState(output.speech_type_); - - if (test_state_ == kExpandPhase || test_state_ == kFadedExpandPhase) { - // Don't verify the output in this phase of the test. - return; - } - - ASSERT_EQ(1u, output.num_channels_); - const int16_t* output_data = output.data(); - for (size_t i = 0; i < output.samples_per_channel_; ++i) { - if (output_data[i] != 0) - return; - } - EXPECT_TRUE(false) - << "Expected at least one non-zero sample in each output block."; - } - - int NumExpectedDecodeCalls(int num_loops) override { - // Some packets at the end of the stream won't be decoded. When the jump in - // timestamp happens, NetEq will do Expand during one GetAudio call. In the - // next call it will decode the packet after the jump, but the net result is - // that the delay increased by 1 packet. In another call, a Pre-emptive - // Expand operation is performed, leading to delay increase by 1 packet. In - // total, the test will end with a 2-packet delay, which results in the 2 - // last packets not being decoded. - return num_loops - 2; - } - - TestStates test_state_; -}; - -TEST_F(LargeTimestampJumpTest, JumpLongerThanHalfRange) { - // Set the timestamp series to start at 2880, increase to 7200, then jump to - // 2869342376. The sequence numbers start at 42076 and increase by 1 for each - // packet, also when the timestamp jumps. - static const uint16_t kStartSeqeunceNumber = 42076; - static const uint32_t kStartTimestamp = 2880; - static const uint32_t kJumpFromTimestamp = 7200; - static const uint32_t kJumpToTimestamp = 2869342376; - static_assert(kJumpFromTimestamp < kJumpToTimestamp, - "timestamp jump should not result in wrap"); - static_assert( - static_cast(kJumpToTimestamp - kJumpFromTimestamp) > 0x7FFFFFFF, - "jump should be larger than half range"); - // Replace the default RTP generator with one that jumps in timestamp. - ResetRtpGenerator(new test::TimestampJumpRtpGenerator( - samples_per_ms(), kStartSeqeunceNumber, kStartTimestamp, - kJumpFromTimestamp, kJumpToTimestamp)); - - RunTest(130); // Run 130 laps @ 10 ms each in the test loop. - EXPECT_EQ(kRecovered, test_state_); -} - -TEST_F(LargeTimestampJumpTest, JumpLongerThanHalfRangeAndWrap) { - // Make a jump larger than half the 32-bit timestamp range. Set the start - // timestamp such that the jump will result in a wrap around. - static const uint16_t kStartSeqeunceNumber = 42076; - // Set the jump length slightly larger than 2^31. - static const uint32_t kStartTimestamp = 3221223116; - static const uint32_t kJumpFromTimestamp = 3221223216; - static const uint32_t kJumpToTimestamp = 1073744278; - static_assert(kJumpToTimestamp < kJumpFromTimestamp, - "timestamp jump should result in wrap"); - static_assert( - static_cast(kJumpToTimestamp - kJumpFromTimestamp) > 0x7FFFFFFF, - "jump should be larger than half range"); - // Replace the default RTP generator with one that jumps in timestamp. - ResetRtpGenerator(new test::TimestampJumpRtpGenerator( - samples_per_ms(), kStartSeqeunceNumber, kStartTimestamp, - kJumpFromTimestamp, kJumpToTimestamp)); - - RunTest(130); // Run 130 laps @ 10 ms each in the test loop. - EXPECT_EQ(kRecovered, test_state_); -} - -class ShortTimestampJumpTest : public LargeTimestampJumpTest { - protected: - void UpdateState(AudioFrame::SpeechType output_type) override { - switch (test_state_) { - case kInitialPhase: { - if (output_type == AudioFrame::kNormalSpeech) { - test_state_ = kNormalPhase; - } - break; - } - case kNormalPhase: { - if (output_type == AudioFrame::kPLC) { - test_state_ = kExpandPhase; - } - break; - } - case kExpandPhase: { - if (output_type == AudioFrame::kNormalSpeech) { - test_state_ = kRecovered; - } - break; - } - case kRecovered: { - break; - } - default: { FAIL(); } - } - } - - int NumExpectedDecodeCalls(int num_loops) override { - // Some packets won't be decoded because of the timestamp jump. - return num_loops - 2; - } -}; - -TEST_F(ShortTimestampJumpTest, JumpShorterThanHalfRange) { - // Make a jump shorter than half the 32-bit timestamp range. Set the start - // timestamp such that the jump will not result in a wrap around. - static const uint16_t kStartSeqeunceNumber = 42076; - // Set the jump length slightly smaller than 2^31. - static const uint32_t kStartTimestamp = 4711; - static const uint32_t kJumpFromTimestamp = 4811; - static const uint32_t kJumpToTimestamp = 2147483747; - static_assert(kJumpFromTimestamp < kJumpToTimestamp, - "timestamp jump should not result in wrap"); - static_assert( - static_cast(kJumpToTimestamp - kJumpFromTimestamp) < 0x7FFFFFFF, - "jump should be smaller than half range"); - // Replace the default RTP generator with one that jumps in timestamp. - ResetRtpGenerator(new test::TimestampJumpRtpGenerator( - samples_per_ms(), kStartSeqeunceNumber, kStartTimestamp, - kJumpFromTimestamp, kJumpToTimestamp)); - - RunTest(130); // Run 130 laps @ 10 ms each in the test loop. - EXPECT_EQ(kRecovered, test_state_); -} - -TEST_F(ShortTimestampJumpTest, JumpShorterThanHalfRangeAndWrap) { - // Make a jump shorter than half the 32-bit timestamp range. Set the start - // timestamp such that the jump will result in a wrap around. - static const uint16_t kStartSeqeunceNumber = 42076; - // Set the jump length slightly smaller than 2^31. - static const uint32_t kStartTimestamp = 3221227827; - static const uint32_t kJumpFromTimestamp = 3221227927; - static const uint32_t kJumpToTimestamp = 1073739567; - static_assert(kJumpToTimestamp < kJumpFromTimestamp, - "timestamp jump should result in wrap"); - static_assert( - static_cast(kJumpToTimestamp - kJumpFromTimestamp) < 0x7FFFFFFF, - "jump should be smaller than half range"); - // Replace the default RTP generator with one that jumps in timestamp. - ResetRtpGenerator(new test::TimestampJumpRtpGenerator( - samples_per_ms(), kStartSeqeunceNumber, kStartTimestamp, - kJumpFromTimestamp, kJumpToTimestamp)); - - RunTest(130); // Run 130 laps @ 10 ms each in the test loop. - EXPECT_EQ(kRecovered, test_state_); -} - -} // namespace webrtc