| /* |
| * Copyright (c) 2018 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 when using the decoder-internal PLC. |
| |
| #include <algorithm> |
| #include <utility> |
| #include <vector> |
| |
| #include "absl/memory/memory.h" |
| #include "absl/types/optional.h" |
| #include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h" |
| #include "modules/audio_coding/neteq/tools/audio_checksum.h" |
| #include "modules/audio_coding/neteq/tools/audio_sink.h" |
| #include "modules/audio_coding/neteq/tools/encode_neteq_input.h" |
| #include "modules/audio_coding/neteq/tools/fake_decode_from_file.h" |
| #include "modules/audio_coding/neteq/tools/input_audio_file.h" |
| #include "modules/audio_coding/neteq/tools/neteq_test.h" |
| #include "rtc_base/numerics/safe_conversions.h" |
| #include "rtc_base/ref_counted_object.h" |
| #include "test/audio_decoder_proxy_factory.h" |
| #include "test/gtest.h" |
| #include "test/testsupport/file_utils.h" |
| |
| namespace webrtc { |
| namespace test { |
| namespace { |
| |
| // This class implements a fake decoder. The decoder will read audio from a file |
| // and present as output, both for regular decoding and for PLC. |
| class AudioDecoderPlc : public AudioDecoder { |
| public: |
| AudioDecoderPlc(std::unique_ptr<InputAudioFile> input, int sample_rate_hz) |
| : input_(std::move(input)), sample_rate_hz_(sample_rate_hz) {} |
| |
| void Reset() override {} |
| int SampleRateHz() const override { return sample_rate_hz_; } |
| size_t Channels() const override { return 1; } |
| int DecodeInternal(const uint8_t* /*encoded*/, |
| size_t encoded_len, |
| int sample_rate_hz, |
| int16_t* decoded, |
| SpeechType* speech_type) override { |
| RTC_CHECK_EQ(encoded_len / 2, 20 * sample_rate_hz_ / 1000); |
| RTC_CHECK_EQ(sample_rate_hz, sample_rate_hz_); |
| RTC_CHECK(decoded); |
| RTC_CHECK(speech_type); |
| RTC_CHECK(input_->Read(encoded_len / 2, decoded)); |
| *speech_type = kSpeech; |
| last_was_plc_ = false; |
| return encoded_len / 2; |
| } |
| |
| void GeneratePlc(size_t requested_samples_per_channel, |
| rtc::BufferT<int16_t>* concealment_audio) override { |
| // Must keep a local copy of this since DecodeInternal sets it to false. |
| const bool last_was_plc = last_was_plc_; |
| SpeechType speech_type; |
| std::vector<int16_t> decoded(5760); |
| int dec_len = DecodeInternal(nullptr, 2 * 20 * sample_rate_hz_ / 1000, |
| sample_rate_hz_, decoded.data(), &speech_type); |
| // This fake decoder can only generate 20 ms of PLC data each time. Make |
| // sure the caller didn't ask for more. |
| RTC_CHECK_GE(dec_len, requested_samples_per_channel); |
| concealment_audio->AppendData(decoded.data(), dec_len); |
| concealed_samples_ += rtc::checked_cast<size_t>(dec_len); |
| if (!last_was_plc) { |
| ++concealment_events_; |
| } |
| last_was_plc_ = true; |
| } |
| |
| size_t concealed_samples() { return concealed_samples_; } |
| size_t concealment_events() { return concealment_events_; } |
| |
| private: |
| const std::unique_ptr<InputAudioFile> input_; |
| const int sample_rate_hz_; |
| size_t concealed_samples_ = 0; |
| size_t concealment_events_ = 0; |
| bool last_was_plc_ = false; |
| }; |
| |
| // An input sample generator which generates only zero-samples. |
| class ZeroSampleGenerator : public EncodeNetEqInput::Generator { |
| public: |
| rtc::ArrayView<const int16_t> Generate(size_t num_samples) override { |
| vec.resize(num_samples, 0); |
| rtc::ArrayView<const int16_t> view(vec); |
| RTC_DCHECK_EQ(view.size(), num_samples); |
| return view; |
| } |
| |
| private: |
| std::vector<int16_t> vec; |
| }; |
| |
| // A NetEqInput which connects to another NetEqInput, but drops a number of |
| // packets on the way. |
| class LossyInput : public NetEqInput { |
| public: |
| LossyInput(int loss_cadence, std::unique_ptr<NetEqInput> input) |
| : loss_cadence_(loss_cadence), input_(std::move(input)) {} |
| |
| absl::optional<int64_t> NextPacketTime() const override { |
| return input_->NextPacketTime(); |
| } |
| |
| absl::optional<int64_t> NextOutputEventTime() const override { |
| return input_->NextOutputEventTime(); |
| } |
| |
| std::unique_ptr<PacketData> PopPacket() override { |
| if (loss_cadence_ != 0 && (++count_ % loss_cadence_) == 0) { |
| // Pop one extra packet to create the loss. |
| input_->PopPacket(); |
| } |
| return input_->PopPacket(); |
| } |
| |
| void AdvanceOutputEvent() override { return input_->AdvanceOutputEvent(); } |
| |
| bool ended() const override { return input_->ended(); } |
| |
| absl::optional<RTPHeader> NextHeader() const override { |
| return input_->NextHeader(); |
| } |
| |
| private: |
| const int loss_cadence_; |
| int count_ = 0; |
| const std::unique_ptr<NetEqInput> input_; |
| }; |
| |
| class AudioChecksumWithOutput : public AudioChecksum { |
| public: |
| explicit AudioChecksumWithOutput(std::string* output_str) |
| : output_str_(*output_str) {} |
| ~AudioChecksumWithOutput() { output_str_ = Finish(); } |
| |
| private: |
| std::string& output_str_; |
| }; |
| |
| NetEqNetworkStatistics RunTest(int loss_cadence, std::string* checksum) { |
| NetEq::Config config; |
| config.for_test_no_time_stretching = true; |
| |
| // The input is mostly useless. It sends zero-samples to a PCM16b encoder, |
| // but the actual encoded samples will never be used by the decoder in the |
| // test. See below about the decoder. |
| auto generator = absl::make_unique<ZeroSampleGenerator>(); |
| constexpr int kSampleRateHz = 32000; |
| constexpr int kPayloadType = 100; |
| AudioEncoderPcm16B::Config encoder_config; |
| encoder_config.sample_rate_hz = kSampleRateHz; |
| encoder_config.payload_type = kPayloadType; |
| auto encoder = absl::make_unique<AudioEncoderPcm16B>(encoder_config); |
| constexpr int kRunTimeMs = 10000; |
| auto input = absl::make_unique<EncodeNetEqInput>( |
| std::move(generator), std::move(encoder), kRunTimeMs); |
| // Wrap the input in a loss function. |
| auto lossy_input = |
| absl::make_unique<LossyInput>(loss_cadence, std::move(input)); |
| |
| // Settinng up decoders. |
| NetEqTest::DecoderMap decoders; |
| // Using a fake decoder which simply reads the output audio from a file. |
| auto input_file = absl::make_unique<InputAudioFile>( |
| webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm")); |
| AudioDecoderPlc dec(std::move(input_file), kSampleRateHz); |
| // Masquerading as a PCM16b decoder. |
| decoders.emplace(kPayloadType, SdpAudioFormat("l16", 32000, 1)); |
| |
| // Output is simply a checksum calculator. |
| auto output = absl::make_unique<AudioChecksumWithOutput>(checksum); |
| |
| // No callback objects. |
| NetEqTest::Callbacks callbacks; |
| |
| NetEqTest neteq_test( |
| config, new rtc::RefCountedObject<test::AudioDecoderProxyFactory>(&dec), |
| decoders, nullptr, std::move(lossy_input), std::move(output), callbacks); |
| EXPECT_LE(kRunTimeMs, neteq_test.Run()); |
| |
| auto lifetime_stats = neteq_test.LifetimeStats(); |
| EXPECT_EQ(dec.concealed_samples(), lifetime_stats.concealed_samples); |
| EXPECT_EQ(dec.concealment_events(), lifetime_stats.concealment_events); |
| |
| return neteq_test.SimulationStats(); |
| } |
| } // namespace |
| |
| TEST(NetEqDecoderPlc, Test) { |
| std::string checksum; |
| auto stats = RunTest(10, &checksum); |
| |
| std::string checksum_no_loss; |
| auto stats_no_loss = RunTest(0, &checksum_no_loss); |
| |
| EXPECT_EQ(checksum, checksum_no_loss); |
| |
| EXPECT_EQ(stats.preemptive_rate, stats_no_loss.preemptive_rate); |
| EXPECT_EQ(stats.accelerate_rate, stats_no_loss.accelerate_rate); |
| EXPECT_EQ(0, stats_no_loss.expand_rate); |
| EXPECT_GT(stats.expand_rate, 0); |
| } |
| |
| } // namespace test |
| } // namespace webrtc |