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This reverts commit d93b7b91e0
.
Reason for revert: Breaks downstream tests
Original change's description:
> Unifying the handling of the events in NetEqInput.
>
> Bug: webrtc:14763
> Change-Id: I9615a9ce41c9b577c4ebd4cdcc9885bfbc5dac48
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293040
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39706}
Bug: webrtc:14763
Change-Id: If076c8fc59a38f011dfa20829f2dd91dd2f914b2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299420
Owners-Override: Björn Terelius <terelius@webrtc.org>
Auto-Submit: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#39711}
321 lines
12 KiB
C++
321 lines
12 KiB
C++
/*
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* Copyright (c) 2018 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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// Test to verify correct operation when using the decoder-internal PLC.
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#include <memory>
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#include <utility>
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#include <vector>
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#include "absl/types/optional.h"
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#include "modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h"
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#include "modules/audio_coding/neteq/tools/audio_checksum.h"
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#include "modules/audio_coding/neteq/tools/audio_sink.h"
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#include "modules/audio_coding/neteq/tools/encode_neteq_input.h"
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#include "modules/audio_coding/neteq/tools/fake_decode_from_file.h"
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#include "modules/audio_coding/neteq/tools/input_audio_file.h"
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#include "modules/audio_coding/neteq/tools/neteq_test.h"
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#include "rtc_base/numerics/safe_conversions.h"
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#include "test/audio_decoder_proxy_factory.h"
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#include "test/gtest.h"
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#include "test/testsupport/file_utils.h"
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namespace webrtc {
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namespace test {
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namespace {
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constexpr int kSampleRateHz = 32000;
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constexpr int kRunTimeMs = 10000;
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// This class implements a fake decoder. The decoder will read audio from a file
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// and present as output, both for regular decoding and for PLC.
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class AudioDecoderPlc : public AudioDecoder {
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public:
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AudioDecoderPlc(std::unique_ptr<InputAudioFile> input, int sample_rate_hz)
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: input_(std::move(input)), sample_rate_hz_(sample_rate_hz) {}
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void Reset() override {}
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int SampleRateHz() const override { return sample_rate_hz_; }
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size_t Channels() const override { return 1; }
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int DecodeInternal(const uint8_t* /*encoded*/,
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size_t encoded_len,
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int sample_rate_hz,
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int16_t* decoded,
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SpeechType* speech_type) override {
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RTC_CHECK_GE(encoded_len / 2, 10 * sample_rate_hz_ / 1000);
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RTC_CHECK_LE(encoded_len / 2, 2 * 10 * sample_rate_hz_ / 1000);
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RTC_CHECK_EQ(sample_rate_hz, sample_rate_hz_);
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RTC_CHECK(decoded);
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RTC_CHECK(speech_type);
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RTC_CHECK(input_->Read(encoded_len / 2, decoded));
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*speech_type = kSpeech;
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last_was_plc_ = false;
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return encoded_len / 2;
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}
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void GeneratePlc(size_t requested_samples_per_channel,
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rtc::BufferT<int16_t>* concealment_audio) override {
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// Instead of generating random data for GeneratePlc we use the same data as
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// the input, so we can check that we produce the same result independently
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// of the losses.
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RTC_DCHECK_EQ(requested_samples_per_channel, 10 * sample_rate_hz_ / 1000);
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// Must keep a local copy of this since DecodeInternal sets it to false.
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const bool last_was_plc = last_was_plc_;
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std::vector<int16_t> decoded(5760);
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SpeechType speech_type;
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int dec_len = DecodeInternal(nullptr, 2 * 10 * sample_rate_hz_ / 1000,
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sample_rate_hz_, decoded.data(), &speech_type);
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concealment_audio->AppendData(decoded.data(), dec_len);
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concealed_samples_ += rtc::checked_cast<size_t>(dec_len);
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if (!last_was_plc) {
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++concealment_events_;
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}
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last_was_plc_ = true;
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}
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size_t concealed_samples() { return concealed_samples_; }
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size_t concealment_events() { return concealment_events_; }
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private:
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const std::unique_ptr<InputAudioFile> input_;
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const int sample_rate_hz_;
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size_t concealed_samples_ = 0;
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size_t concealment_events_ = 0;
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bool last_was_plc_ = false;
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};
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// An input sample generator which generates only zero-samples.
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class ZeroSampleGenerator : public EncodeNetEqInput::Generator {
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public:
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rtc::ArrayView<const int16_t> Generate(size_t num_samples) override {
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vec.resize(num_samples, 0);
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rtc::ArrayView<const int16_t> view(vec);
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RTC_DCHECK_EQ(view.size(), num_samples);
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return view;
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}
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private:
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std::vector<int16_t> vec;
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};
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// A NetEqInput which connects to another NetEqInput, but drops a number of
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// consecutive packets on the way
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class LossyInput : public NetEqInput {
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public:
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LossyInput(int loss_cadence,
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int burst_length,
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std::unique_ptr<NetEqInput> input)
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: loss_cadence_(loss_cadence),
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burst_length_(burst_length),
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input_(std::move(input)) {}
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absl::optional<int64_t> NextPacketTime() const override {
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return input_->NextPacketTime();
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}
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absl::optional<int64_t> NextOutputEventTime() const override {
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return input_->NextOutputEventTime();
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}
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absl::optional<SetMinimumDelayInfo> NextSetMinimumDelayInfo() const override {
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return input_->NextSetMinimumDelayInfo();
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}
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std::unique_ptr<PacketData> PopPacket() override {
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if (loss_cadence_ != 0 && (++count_ % loss_cadence_) == 0) {
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// Pop `burst_length_` packets to create the loss.
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auto packet_to_return = input_->PopPacket();
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for (int i = 0; i < burst_length_; i++) {
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input_->PopPacket();
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}
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return packet_to_return;
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}
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return input_->PopPacket();
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}
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void AdvanceOutputEvent() override { return input_->AdvanceOutputEvent(); }
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void AdvanceSetMinimumDelay() override {
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return input_->AdvanceSetMinimumDelay();
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}
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bool ended() const override { return input_->ended(); }
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absl::optional<RTPHeader> NextHeader() const override {
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return input_->NextHeader();
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}
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private:
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const int loss_cadence_;
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const int burst_length_;
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int count_ = 0;
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const std::unique_ptr<NetEqInput> input_;
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};
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class AudioChecksumWithOutput : public AudioChecksum {
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public:
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explicit AudioChecksumWithOutput(std::string* output_str)
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: output_str_(*output_str) {}
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~AudioChecksumWithOutput() { output_str_ = Finish(); }
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private:
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std::string& output_str_;
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};
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struct TestStatistics {
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NetEqNetworkStatistics network;
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NetEqLifetimeStatistics lifetime;
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};
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TestStatistics RunTest(int loss_cadence,
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int burst_length,
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std::string* checksum) {
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NetEq::Config config;
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config.for_test_no_time_stretching = true;
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// The input is mostly useless. It sends zero-samples to a PCM16b encoder,
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// but the actual encoded samples will never be used by the decoder in the
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// test. See below about the decoder.
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auto generator = std::make_unique<ZeroSampleGenerator>();
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constexpr int kPayloadType = 100;
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AudioEncoderPcm16B::Config encoder_config;
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encoder_config.sample_rate_hz = kSampleRateHz;
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encoder_config.payload_type = kPayloadType;
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auto encoder = std::make_unique<AudioEncoderPcm16B>(encoder_config);
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auto input = std::make_unique<EncodeNetEqInput>(
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std::move(generator), std::move(encoder), kRunTimeMs);
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// Wrap the input in a loss function.
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auto lossy_input = std::make_unique<LossyInput>(loss_cadence, burst_length,
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std::move(input));
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// Setting up decoders.
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NetEqTest::DecoderMap decoders;
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// Using a fake decoder which simply reads the output audio from a file.
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auto input_file = std::make_unique<InputAudioFile>(
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webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm"));
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AudioDecoderPlc dec(std::move(input_file), kSampleRateHz);
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// Masquerading as a PCM16b decoder.
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decoders.emplace(kPayloadType, SdpAudioFormat("l16", 32000, 1));
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// Output is simply a checksum calculator.
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auto output = std::make_unique<AudioChecksumWithOutput>(checksum);
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// No callback objects.
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NetEqTest::Callbacks callbacks;
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NetEqTest neteq_test(
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config, /*decoder_factory=*/
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rtc::make_ref_counted<test::AudioDecoderProxyFactory>(&dec),
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/*codecs=*/decoders, /*text_log=*/nullptr, /*neteq_factory=*/nullptr,
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/*input=*/std::move(lossy_input), std::move(output), callbacks);
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EXPECT_LE(kRunTimeMs, neteq_test.Run());
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auto lifetime_stats = neteq_test.LifetimeStats();
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EXPECT_EQ(dec.concealed_samples(), lifetime_stats.concealed_samples);
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EXPECT_EQ(dec.concealment_events(), lifetime_stats.concealment_events);
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return {neteq_test.SimulationStats(), neteq_test.LifetimeStats()};
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}
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} // namespace
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// Check that some basic metrics are produced in the right direction. In
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// particular, expand_rate should only increase if there are losses present. Our
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// dummy decoder is designed such as the checksum should always be the same
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// regardless of the losses given that calls are executed in the right order.
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TEST(NetEqDecoderPlc, BasicMetrics) {
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std::string checksum;
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// Drop 1 packet every 10 packets.
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auto stats = RunTest(10, 1, &checksum);
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std::string checksum_no_loss;
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auto stats_no_loss = RunTest(0, 0, &checksum_no_loss);
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EXPECT_EQ(checksum, checksum_no_loss);
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EXPECT_EQ(stats.network.preemptive_rate,
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stats_no_loss.network.preemptive_rate);
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EXPECT_EQ(stats.network.accelerate_rate,
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stats_no_loss.network.accelerate_rate);
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EXPECT_EQ(0, stats_no_loss.network.expand_rate);
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EXPECT_GT(stats.network.expand_rate, 0);
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}
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// Checks that interruptions are not counted in small losses but they are
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// correctly counted in long interruptions.
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TEST(NetEqDecoderPlc, CountInterruptions) {
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std::string checksum;
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std::string checksum_2;
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std::string checksum_3;
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// Half of the packets lost but in short interruptions.
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auto stats_no_interruptions = RunTest(1, 1, &checksum);
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// One lost of 500 ms (250 packets).
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auto stats_one_interruption = RunTest(200, 250, &checksum_2);
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// Two losses of 250ms each (125 packets).
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auto stats_two_interruptions = RunTest(125, 125, &checksum_3);
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EXPECT_EQ(checksum, checksum_2);
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EXPECT_EQ(checksum, checksum_3);
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EXPECT_GT(stats_no_interruptions.network.expand_rate, 0);
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EXPECT_EQ(stats_no_interruptions.lifetime.total_interruption_duration_ms, 0);
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EXPECT_EQ(stats_no_interruptions.lifetime.interruption_count, 0);
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EXPECT_GT(stats_one_interruption.network.expand_rate, 0);
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EXPECT_EQ(stats_one_interruption.lifetime.total_interruption_duration_ms,
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5000);
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EXPECT_EQ(stats_one_interruption.lifetime.interruption_count, 1);
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EXPECT_GT(stats_two_interruptions.network.expand_rate, 0);
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EXPECT_EQ(stats_two_interruptions.lifetime.total_interruption_duration_ms,
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5000);
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EXPECT_EQ(stats_two_interruptions.lifetime.interruption_count, 2);
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}
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// Checks that small losses do not produce interruptions.
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TEST(NetEqDecoderPlc, NoInterruptionsInSmallLosses) {
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std::string checksum_1;
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std::string checksum_4;
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auto stats_1 = RunTest(300, 1, &checksum_1);
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auto stats_4 = RunTest(300, 4, &checksum_4);
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EXPECT_EQ(checksum_1, checksum_4);
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EXPECT_EQ(stats_1.lifetime.interruption_count, 0);
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EXPECT_EQ(stats_1.lifetime.total_interruption_duration_ms, 0);
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EXPECT_EQ(stats_1.lifetime.concealed_samples, 640u); // 20ms of concealment.
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EXPECT_EQ(stats_1.lifetime.concealment_events, 1u); // in just one event.
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EXPECT_EQ(stats_4.lifetime.interruption_count, 0);
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EXPECT_EQ(stats_4.lifetime.total_interruption_duration_ms, 0);
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EXPECT_EQ(stats_4.lifetime.concealed_samples, 2560u); // 80ms of concealment.
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EXPECT_EQ(stats_4.lifetime.concealment_events, 1u); // in just one event.
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}
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// Checks that interruptions of different sizes report correct duration.
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TEST(NetEqDecoderPlc, InterruptionsReportCorrectSize) {
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std::string checksum;
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for (int burst_length = 5; burst_length < 10; burst_length++) {
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auto stats = RunTest(300, burst_length, &checksum);
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auto duration = stats.lifetime.total_interruption_duration_ms;
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if (burst_length < 8) {
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EXPECT_EQ(duration, 0);
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} else {
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EXPECT_EQ(duration, burst_length * 20);
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}
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}
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}
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} // namespace test
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} // namespace webrtc
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