| /* |
| * Copyright (c) 2015 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. |
| */ |
| |
| #include <memory> |
| |
| #include "absl/memory/memory.h" |
| #include "api/audio/audio_frame.h" |
| #include "api/audio_codecs/audio_decoder.h" |
| #include "common_types.h" // NOLINT(build/include) |
| |
| #include "modules/audio_coding/neteq/include/neteq.h" |
| #include "modules/audio_coding/neteq/tools/rtp_generator.h" |
| #include "rtc_base/ref_counted_object.h" |
| #include "test/audio_decoder_proxy_factory.h" |
| #include "test/gmock.h" |
| |
| namespace webrtc { |
| namespace test { |
| |
| using ::testing::_; |
| using ::testing::SetArgPointee; |
| using ::testing::Return; |
| |
| class MockAudioDecoder final : public AudioDecoder { |
| public: |
| // TODO(nisse): Valid overrides commented out, because the gmock |
| // methods don't use any override declarations, and we want to avoid |
| // warnings from -Winconsistent-missing-override. See |
| // http://crbug.com/428099. |
| static const int kPacketDuration = 960; // 48 kHz * 20 ms |
| |
| MockAudioDecoder(int sample_rate_hz, size_t num_channels) |
| : sample_rate_hz_(sample_rate_hz), |
| num_channels_(num_channels), |
| fec_enabled_(false) {} |
| ~MockAudioDecoder() /* override */ { Die(); } |
| MOCK_METHOD0(Die, void()); |
| |
| MOCK_METHOD0(Reset, void()); |
| |
| class MockFrame : public AudioDecoder::EncodedAudioFrame { |
| public: |
| MockFrame(size_t num_channels) : num_channels_(num_channels) {} |
| |
| size_t Duration() const override { return kPacketDuration; } |
| |
| absl::optional<DecodeResult> Decode( |
| rtc::ArrayView<int16_t> decoded) const override { |
| const size_t output_size = |
| sizeof(int16_t) * kPacketDuration * num_channels_; |
| if (decoded.size() >= output_size) { |
| memset(decoded.data(), 0, |
| sizeof(int16_t) * kPacketDuration * num_channels_); |
| return DecodeResult{kPacketDuration * num_channels_, kSpeech}; |
| } else { |
| ADD_FAILURE() << "Expected decoded.size() to be >= output_size (" |
| << decoded.size() << " vs. " << output_size << ")"; |
| return absl::nullopt; |
| } |
| } |
| |
| private: |
| const size_t num_channels_; |
| }; |
| |
| std::vector<ParseResult> ParsePayload(rtc::Buffer&& payload, |
| uint32_t timestamp) /* override */ { |
| std::vector<ParseResult> results; |
| if (fec_enabled_) { |
| std::unique_ptr<MockFrame> fec_frame(new MockFrame(num_channels_)); |
| results.emplace_back(timestamp - kPacketDuration, 1, |
| std::move(fec_frame)); |
| } |
| |
| std::unique_ptr<MockFrame> frame(new MockFrame(num_channels_)); |
| results.emplace_back(timestamp, 0, std::move(frame)); |
| return results; |
| } |
| |
| int PacketDuration(const uint8_t* encoded, size_t encoded_len) const |
| /* override */ { |
| ADD_FAILURE() << "Since going through ParsePayload, PacketDuration should " |
| "never get called."; |
| return kPacketDuration; |
| } |
| |
| bool PacketHasFec(const uint8_t* encoded, size_t encoded_len) const |
| /* override */ { |
| ADD_FAILURE() << "Since going through ParsePayload, PacketHasFec should " |
| "never get called."; |
| return fec_enabled_; |
| } |
| |
| int SampleRateHz() const /* override */ { return sample_rate_hz_; } |
| |
| size_t Channels() const /* override */ { return num_channels_; } |
| |
| void set_fec_enabled(bool enable_fec) { fec_enabled_ = enable_fec; } |
| |
| bool fec_enabled() const { return fec_enabled_; } |
| |
| protected: |
| int DecodeInternal(const uint8_t* encoded, |
| size_t encoded_len, |
| int sample_rate_hz, |
| int16_t* decoded, |
| SpeechType* speech_type) /* override */ { |
| ADD_FAILURE() << "Since going through ParsePayload, DecodeInternal should " |
| "never get called."; |
| return -1; |
| } |
| |
| private: |
| const int sample_rate_hz_; |
| const size_t num_channels_; |
| bool fec_enabled_; |
| }; |
| |
| class NetEqNetworkStatsTest { |
| public: |
| static const int kPayloadSizeByte = 30; |
| static const int kFrameSizeMs = 20; |
| static const uint8_t kPayloadType = 95; |
| static const int kOutputLengthMs = 10; |
| |
| enum logic { |
| kIgnore, |
| kEqual, |
| kSmallerThan, |
| kLargerThan, |
| }; |
| |
| struct NetEqNetworkStatsCheck { |
| logic current_buffer_size_ms; |
| logic preferred_buffer_size_ms; |
| logic jitter_peaks_found; |
| logic packet_loss_rate; |
| logic expand_rate; |
| logic speech_expand_rate; |
| logic preemptive_rate; |
| logic accelerate_rate; |
| logic secondary_decoded_rate; |
| logic secondary_discarded_rate; |
| logic clockdrift_ppm; |
| logic added_zero_samples; |
| NetEqNetworkStatistics stats_ref; |
| }; |
| |
| NetEqNetworkStatsTest(const SdpAudioFormat& format, MockAudioDecoder* decoder) |
| : decoder_(decoder), |
| decoder_factory_( |
| new rtc::RefCountedObject<AudioDecoderProxyFactory>(decoder)), |
| samples_per_ms_(format.clockrate_hz / 1000), |
| frame_size_samples_(kFrameSizeMs * samples_per_ms_), |
| rtp_generator_(new RtpGenerator(samples_per_ms_)), |
| last_lost_time_(0), |
| packet_loss_interval_(0xffffffff) { |
| NetEq::Config config; |
| config.sample_rate_hz = format.clockrate_hz; |
| neteq_ = absl::WrapUnique(NetEq::Create(config, decoder_factory_)); |
| neteq_->RegisterPayloadType(kPayloadType, format); |
| } |
| |
| bool Lost(uint32_t send_time) { |
| if (send_time - last_lost_time_ >= packet_loss_interval_) { |
| last_lost_time_ = send_time; |
| return true; |
| } |
| return false; |
| } |
| |
| void SetPacketLossRate(double loss_rate) { |
| packet_loss_interval_ = |
| (loss_rate >= 1e-3 ? static_cast<double>(kFrameSizeMs) / loss_rate |
| : 0xffffffff); |
| } |
| |
| // |stats_ref| |
| // expects.x = -1, do not care |
| // expects.x = 0, 'x' in current stats should equal 'x' in |stats_ref| |
| // expects.x = 1, 'x' in current stats should < 'x' in |stats_ref| |
| // expects.x = 2, 'x' in current stats should > 'x' in |stats_ref| |
| void CheckNetworkStatistics(NetEqNetworkStatsCheck expects) { |
| NetEqNetworkStatistics stats; |
| neteq_->NetworkStatistics(&stats); |
| |
| #define CHECK_NETEQ_NETWORK_STATS(x) \ |
| switch (expects.x) { \ |
| case kEqual: \ |
| EXPECT_EQ(stats.x, expects.stats_ref.x); \ |
| break; \ |
| case kSmallerThan: \ |
| EXPECT_LT(stats.x, expects.stats_ref.x); \ |
| break; \ |
| case kLargerThan: \ |
| EXPECT_GT(stats.x, expects.stats_ref.x); \ |
| break; \ |
| default: \ |
| break; \ |
| } |
| |
| CHECK_NETEQ_NETWORK_STATS(current_buffer_size_ms); |
| CHECK_NETEQ_NETWORK_STATS(preferred_buffer_size_ms); |
| CHECK_NETEQ_NETWORK_STATS(jitter_peaks_found); |
| CHECK_NETEQ_NETWORK_STATS(packet_loss_rate); |
| CHECK_NETEQ_NETWORK_STATS(expand_rate); |
| CHECK_NETEQ_NETWORK_STATS(speech_expand_rate); |
| CHECK_NETEQ_NETWORK_STATS(preemptive_rate); |
| CHECK_NETEQ_NETWORK_STATS(accelerate_rate); |
| CHECK_NETEQ_NETWORK_STATS(secondary_decoded_rate); |
| CHECK_NETEQ_NETWORK_STATS(secondary_discarded_rate); |
| CHECK_NETEQ_NETWORK_STATS(clockdrift_ppm); |
| CHECK_NETEQ_NETWORK_STATS(added_zero_samples); |
| |
| #undef CHECK_NETEQ_NETWORK_STATS |
| } |
| |
| void RunTest(int num_loops, NetEqNetworkStatsCheck expects) { |
| uint32_t time_now; |
| uint32_t next_send_time; |
| |
| // Initiate |last_lost_time_|. |
| time_now = next_send_time = last_lost_time_ = rtp_generator_->GetRtpHeader( |
| kPayloadType, frame_size_samples_, &rtp_header_); |
| for (int k = 0; k < num_loops; ++k) { |
| // Delay by one frame such that the FEC can come in. |
| while (time_now + kFrameSizeMs >= next_send_time) { |
| next_send_time = rtp_generator_->GetRtpHeader( |
| kPayloadType, frame_size_samples_, &rtp_header_); |
| if (!Lost(next_send_time)) { |
| static const uint8_t payload[kPayloadSizeByte] = {0}; |
| ASSERT_EQ(NetEq::kOK, |
| neteq_->InsertPacket(rtp_header_, payload, next_send_time)); |
| } |
| } |
| bool muted = true; |
| EXPECT_EQ(NetEq::kOK, neteq_->GetAudio(&output_frame_, &muted)); |
| ASSERT_FALSE(muted); |
| EXPECT_EQ(decoder_->Channels(), output_frame_.num_channels_); |
| EXPECT_EQ(static_cast<size_t>(kOutputLengthMs * samples_per_ms_), |
| output_frame_.samples_per_channel_); |
| EXPECT_EQ(48000, neteq_->last_output_sample_rate_hz()); |
| |
| time_now += kOutputLengthMs; |
| } |
| CheckNetworkStatistics(expects); |
| neteq_->FlushBuffers(); |
| } |
| |
| void DecodeFecTest() { |
| decoder_->set_fec_enabled(false); |
| NetEqNetworkStatsCheck expects = {kIgnore, // current_buffer_size_ms |
| kIgnore, // preferred_buffer_size_ms |
| kIgnore, // jitter_peaks_found |
| kEqual, // packet_loss_rate |
| kEqual, // expand_rate |
| kEqual, // voice_expand_rate |
| kIgnore, // preemptive_rate |
| kEqual, // accelerate_rate |
| kEqual, // decoded_fec_rate |
| kEqual, // discarded_fec_rate |
| kIgnore, // clockdrift_ppm |
| kEqual, // added_zero_samples |
| {0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0}}; |
| RunTest(50, expects); |
| |
| // Next we introduce packet losses. |
| SetPacketLossRate(0.1); |
| expects.stats_ref.packet_loss_rate = 1337; |
| expects.stats_ref.expand_rate = expects.stats_ref.speech_expand_rate = 1065; |
| RunTest(50, expects); |
| |
| // Next we enable FEC. |
| decoder_->set_fec_enabled(true); |
| // If FEC fills in the lost packets, no packet loss will be counted. |
| expects.stats_ref.packet_loss_rate = 0; |
| expects.stats_ref.expand_rate = expects.stats_ref.speech_expand_rate = 0; |
| expects.stats_ref.secondary_decoded_rate = 2006; |
| expects.stats_ref.secondary_discarded_rate = 14336; |
| RunTest(50, expects); |
| } |
| |
| void NoiseExpansionTest() { |
| NetEqNetworkStatsCheck expects = {kIgnore, // current_buffer_size_ms |
| kIgnore, // preferred_buffer_size_ms |
| kIgnore, // jitter_peaks_found |
| kEqual, // packet_loss_rate |
| kEqual, // expand_rate |
| kEqual, // speech_expand_rate |
| kIgnore, // preemptive_rate |
| kEqual, // accelerate_rate |
| kEqual, // decoded_fec_rate |
| kEqual, // discard_fec_rate |
| kIgnore, // clockdrift_ppm |
| kEqual, // added_zero_samples |
| {0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0}}; |
| RunTest(50, expects); |
| |
| SetPacketLossRate(1); |
| expects.stats_ref.expand_rate = 16384; |
| expects.stats_ref.speech_expand_rate = 5324; |
| RunTest(10, expects); // Lost 10 * 20ms in a row. |
| } |
| |
| private: |
| MockAudioDecoder* decoder_; |
| rtc::scoped_refptr<AudioDecoderProxyFactory> decoder_factory_; |
| std::unique_ptr<NetEq> neteq_; |
| |
| const int samples_per_ms_; |
| const size_t frame_size_samples_; |
| std::unique_ptr<RtpGenerator> rtp_generator_; |
| RTPHeader rtp_header_; |
| uint32_t last_lost_time_; |
| uint32_t packet_loss_interval_; |
| AudioFrame output_frame_; |
| }; |
| |
| TEST(NetEqNetworkStatsTest, DecodeFec) { |
| MockAudioDecoder decoder(48000, 1); |
| NetEqNetworkStatsTest test(SdpAudioFormat("opus", 48000, 2), &decoder); |
| test.DecodeFecTest(); |
| EXPECT_CALL(decoder, Die()).Times(1); |
| } |
| |
| TEST(NetEqNetworkStatsTest, StereoDecodeFec) { |
| MockAudioDecoder decoder(48000, 2); |
| NetEqNetworkStatsTest test(SdpAudioFormat("opus", 48000, 2), &decoder); |
| test.DecodeFecTest(); |
| EXPECT_CALL(decoder, Die()).Times(1); |
| } |
| |
| TEST(NetEqNetworkStatsTest, NoiseExpansionTest) { |
| MockAudioDecoder decoder(48000, 1); |
| NetEqNetworkStatsTest test(SdpAudioFormat("opus", 48000, 2), &decoder); |
| test.NoiseExpansionTest(); |
| EXPECT_CALL(decoder, Die()).Times(1); |
| } |
| |
| } // namespace test |
| } // namespace webrtc |