Files
platform-external-webrtc/test/fuzzers/neteq_signal_fuzzer.cc
Henrik Lundin 1ff41eb784 Revert "NetEq: Deprecate playout modes Fax, Off and Streaming"
This reverts commit 80c4cca4915dbc6094a5bfae749f85f7371eadd1.

Reason for revert: Breaks downstream tests.

Original change's description:
> NetEq: Deprecate playout modes Fax, Off and Streaming
> 
> The playout modes other than Normal have not been reachable for a long
> time, other than through tests. It is time to deprecate them.
> 
> The only meaningful use was that Fax mode was sometimes set from
> tests, in order to avoid time-stretching operations (accelerate and
> pre-emptive expand) from messing with the test results. With this CL,
> a new config is added instead, which lets the user specify exactly
> this: don't do time-stretching.
> 
> As a result of Fax and Off modes being removed, the following code
> clean-up was done:
> - Fold DecisionLogicNormal into DecisionLogic.
> - Remove AudioRepetition and AlternativePlc operations, since they can
>   no longer be reached.
> 
> Bug: webrtc:9421
> Change-Id: I651458e9c1931a99f3b07e242817d303bac119df
> Reviewed-on: https://webrtc-review.googlesource.com/84123
> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
> Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Minyue Li <minyue@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23704}

TBR=henrik.lundin@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org

Change-Id: I555aae8850fc4ac1ea919bfa72c11b5218066f30
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9421
Reviewed-on: https://webrtc-review.googlesource.com/84680
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23706}
2018-06-21 12:36:44 +00:00

209 lines
8.0 KiB
C++

/*
* Copyright (c) 2017 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 <cmath>
#include <limits>
#include <memory>
#include <vector>
#include "api/array_view.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/encode_neteq_input.h"
#include "modules/audio_coding/neteq/tools/neteq_test.h"
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/random.h"
#include "test/fuzzers/fuzz_data_helper.h"
namespace webrtc {
namespace test {
namespace {
// Generate a mixture of sine wave and gaussian noise.
class SineAndNoiseGenerator : public EncodeNetEqInput::Generator {
public:
// The noise generator is seeded with a value from the fuzzer data, but 0 is
// avoided (since it is not allowed by the Random class).
SineAndNoiseGenerator(int sample_rate_hz, FuzzDataHelper* fuzz_data)
: sample_rate_hz_(sample_rate_hz),
fuzz_data_(*fuzz_data),
noise_generator_(fuzz_data_.ReadOrDefaultValueNotZero<uint64_t>(1)) {}
// Generates num_samples of the sine-gaussian mixture.
rtc::ArrayView<const int16_t> Generate(size_t num_samples) override {
if (samples_.size() < num_samples) {
samples_.resize(num_samples);
}
rtc::ArrayView<int16_t> output(samples_.data(), num_samples);
// Randomize an amplitude between 0 and 32768; use 65000/2 if we are out of
// fuzzer data.
const float amplitude = fuzz_data_.ReadOrDefaultValue<uint16_t>(65000) / 2;
// Randomize a noise standard deviation between 0 and 1999.
const float noise_std = fuzz_data_.ReadOrDefaultValue<uint16_t>(0) % 2000;
for (auto& x : output) {
x = rtc::saturated_cast<int16_t>(amplitude * std::sin(phase_) +
noise_generator_.Gaussian(0, noise_std));
phase_ += 2 * kPi * kFreqHz / sample_rate_hz_;
}
return output;
}
private:
static constexpr int kFreqHz = 300; // The sinewave frequency.
const int sample_rate_hz_;
const double kPi = std::acos(-1);
std::vector<int16_t> samples_;
double phase_ = 0.0;
FuzzDataHelper& fuzz_data_;
Random noise_generator_;
};
class FuzzSignalInput : public NetEqInput {
public:
explicit FuzzSignalInput(FuzzDataHelper* fuzz_data,
int sample_rate,
uint8_t payload_type)
: fuzz_data_(*fuzz_data) {
AudioEncoderPcm16B::Config config;
config.payload_type = payload_type;
config.sample_rate_hz = sample_rate;
std::unique_ptr<AudioEncoder> encoder(new AudioEncoderPcm16B(config));
std::unique_ptr<EncodeNetEqInput::Generator> generator(
new SineAndNoiseGenerator(config.sample_rate_hz, fuzz_data));
input_.reset(new EncodeNetEqInput(std::move(generator), std::move(encoder),
std::numeric_limits<int64_t>::max()));
packet_ = input_->PopPacket();
// Select an output event period. This is how long time we wait between each
// call to NetEq::GetAudio. 10 ms is nominal, 9 and 11 ms will both lead to
// clock drift (in different directions).
constexpr int output_event_periods[] = {9, 10, 11};
output_event_period_ms_ = fuzz_data_.SelectOneOf(output_event_periods);
}
absl::optional<int64_t> NextPacketTime() const override {
return packet_->time_ms;
}
absl::optional<int64_t> NextOutputEventTime() const override {
return next_output_event_ms_;
}
std::unique_ptr<PacketData> PopPacket() override {
RTC_DCHECK(packet_);
std::unique_ptr<PacketData> packet_to_return = std::move(packet_);
do {
packet_ = input_->PopPacket();
// If the next value from the fuzzer input is 0, the packet is discarded
// and the next one is pulled from the source.
} while (fuzz_data_.CanReadBytes(1) && fuzz_data_.Read<uint8_t>() == 0);
if (fuzz_data_.CanReadBytes(1)) {
// Generate jitter by setting an offset for the arrival time.
const int8_t arrival_time_offset_ms = fuzz_data_.Read<int8_t>();
// The arrival time can not be before the previous packets.
packet_->time_ms = std::max(packet_to_return->time_ms,
packet_->time_ms + arrival_time_offset_ms);
} else {
// Mark that we are at the end of the test. However, the current packet is
// still valid (but it may not have been fuzzed as expected).
ended_ = true;
}
return packet_to_return;
}
void AdvanceOutputEvent() override {
next_output_event_ms_ += output_event_period_ms_;
}
bool ended() const override { return ended_; }
absl::optional<RTPHeader> NextHeader() const override {
RTC_DCHECK(packet_);
return packet_->header;
}
private:
bool ended_ = false;
FuzzDataHelper& fuzz_data_;
std::unique_ptr<EncodeNetEqInput> input_;
std::unique_ptr<PacketData> packet_;
int64_t next_output_event_ms_ = 0;
int64_t output_event_period_ms_ = 10;
};
} // namespace
void FuzzOneInputTest(const uint8_t* data, size_t size) {
if (size < 1)
return;
// Limit the input size to 100000 bytes to avoid fuzzer timeout.
if (size > 100000)
return;
FuzzDataHelper fuzz_data(rtc::ArrayView<const uint8_t>(data, size));
// Allowed sample rates and payload types used in the test.
std::pair<int, uint8_t> rate_types[] = {
{8000, 93}, {16000, 94}, {32000, 95}, {48000, 96}};
const auto rate_type = fuzz_data.SelectOneOf(rate_types);
const int sample_rate = rate_type.first;
const uint8_t payload_type = rate_type.second;
// Set up the input signal generator.
std::unique_ptr<FuzzSignalInput> input(
new FuzzSignalInput(&fuzz_data, sample_rate, payload_type));
// Output sink for the test.
std::unique_ptr<AudioChecksum> output(new AudioChecksum);
// Configure NetEq and the NetEqTest object.
NetEqTest::Callbacks callbacks;
NetEq::Config config;
config.enable_post_decode_vad = true;
config.enable_fast_accelerate = true;
NetEqTest::DecoderMap codecs;
codecs[0] = std::make_pair(NetEqDecoder::kDecoderPCMu, "pcmu");
codecs[8] = std::make_pair(NetEqDecoder::kDecoderPCMa, "pcma");
codecs[103] = std::make_pair(NetEqDecoder::kDecoderISAC, "isac");
codecs[104] = std::make_pair(NetEqDecoder::kDecoderISACswb, "isac-swb");
codecs[111] = std::make_pair(NetEqDecoder::kDecoderOpus, "opus");
codecs[9] = std::make_pair(NetEqDecoder::kDecoderG722, "g722");
codecs[106] = std::make_pair(NetEqDecoder::kDecoderAVT, "avt");
codecs[114] = std::make_pair(NetEqDecoder::kDecoderAVT16kHz, "avt-16");
codecs[115] = std::make_pair(NetEqDecoder::kDecoderAVT32kHz, "avt-32");
codecs[116] = std::make_pair(NetEqDecoder::kDecoderAVT48kHz, "avt-48");
codecs[117] = std::make_pair(NetEqDecoder::kDecoderRED, "red");
codecs[13] = std::make_pair(NetEqDecoder::kDecoderCNGnb, "cng-nb");
codecs[98] = std::make_pair(NetEqDecoder::kDecoderCNGwb, "cng-wb");
codecs[99] = std::make_pair(NetEqDecoder::kDecoderCNGswb32kHz, "cng-swb32");
codecs[100] = std::make_pair(NetEqDecoder::kDecoderCNGswb48kHz, "cng-swb48");
// One of these payload types will be used for encoding.
codecs[rate_types[0].second] =
std::make_pair(NetEqDecoder::kDecoderPCM16B, "pcm16-nb");
codecs[rate_types[1].second] =
std::make_pair(NetEqDecoder::kDecoderPCM16Bwb, "pcm16-wb");
codecs[rate_types[2].second] =
std::make_pair(NetEqDecoder::kDecoderPCM16Bswb32kHz, "pcm16-swb32");
codecs[rate_types[3].second] =
std::make_pair(NetEqDecoder::kDecoderPCM16Bswb48kHz, "pcm16-swb48");
NetEqTest::ExtDecoderMap ext_codecs;
NetEqTest test(config, codecs, ext_codecs, std::move(input),
std::move(output), callbacks);
test.Run();
}
} // namespace test
void FuzzOneInput(const uint8_t* data, size_t size) {
test::FuzzOneInputTest(data, size);
}
} // namespace webrtc