
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}
173 lines
6.0 KiB
C++
173 lines
6.0 KiB
C++
/*
|
|
* Copyright (c) 2016 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 <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 "modules/rtp_rtcp/source/byte_io.h"
|
|
|
|
namespace webrtc {
|
|
namespace test {
|
|
namespace {
|
|
constexpr int kPayloadType = 95;
|
|
|
|
class SineGenerator : public EncodeNetEqInput::Generator {
|
|
public:
|
|
explicit SineGenerator(int sample_rate_hz)
|
|
: sample_rate_hz_(sample_rate_hz) {}
|
|
|
|
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);
|
|
for (auto& x : output) {
|
|
x = static_cast<int16_t>(2000.0 * std::sin(phase_));
|
|
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;
|
|
};
|
|
|
|
class FuzzRtpInput : public NetEqInput {
|
|
public:
|
|
explicit FuzzRtpInput(rtc::ArrayView<const uint8_t> data) : data_(data) {
|
|
AudioEncoderPcm16B::Config config;
|
|
config.payload_type = kPayloadType;
|
|
config.sample_rate_hz = 32000;
|
|
std::unique_ptr<AudioEncoder> encoder(new AudioEncoderPcm16B(config));
|
|
std::unique_ptr<EncodeNetEqInput::Generator> generator(
|
|
new SineGenerator(config.sample_rate_hz));
|
|
input_.reset(new EncodeNetEqInput(std::move(generator), std::move(encoder),
|
|
std::numeric_limits<int64_t>::max()));
|
|
packet_ = input_->PopPacket();
|
|
FuzzHeader();
|
|
}
|
|
|
|
absl::optional<int64_t> NextPacketTime() const override {
|
|
return packet_->time_ms;
|
|
}
|
|
|
|
absl::optional<int64_t> NextOutputEventTime() const override {
|
|
return input_->NextOutputEventTime();
|
|
}
|
|
|
|
std::unique_ptr<PacketData> PopPacket() override {
|
|
RTC_DCHECK(packet_);
|
|
std::unique_ptr<PacketData> packet_to_return = std::move(packet_);
|
|
packet_ = input_->PopPacket();
|
|
FuzzHeader();
|
|
return packet_to_return;
|
|
}
|
|
|
|
void AdvanceOutputEvent() override { return input_->AdvanceOutputEvent(); }
|
|
|
|
bool ended() const override { return ended_; }
|
|
|
|
absl::optional<RTPHeader> NextHeader() const override {
|
|
RTC_DCHECK(packet_);
|
|
return packet_->header;
|
|
}
|
|
|
|
private:
|
|
void FuzzHeader() {
|
|
constexpr size_t kNumBytesToFuzz = 11;
|
|
if (data_ix_ + kNumBytesToFuzz > data_.size()) {
|
|
ended_ = true;
|
|
return;
|
|
}
|
|
RTC_DCHECK(packet_);
|
|
const size_t start_ix = data_ix_;
|
|
packet_->header.payloadType =
|
|
ByteReader<uint8_t>::ReadLittleEndian(&data_[data_ix_]);
|
|
packet_->header.payloadType &= 0x7F;
|
|
data_ix_ += sizeof(uint8_t);
|
|
packet_->header.sequenceNumber =
|
|
ByteReader<uint16_t>::ReadLittleEndian(&data_[data_ix_]);
|
|
data_ix_ += sizeof(uint16_t);
|
|
packet_->header.timestamp =
|
|
ByteReader<uint32_t>::ReadLittleEndian(&data_[data_ix_]);
|
|
data_ix_ += sizeof(uint32_t);
|
|
packet_->header.ssrc =
|
|
ByteReader<uint32_t>::ReadLittleEndian(&data_[data_ix_]);
|
|
data_ix_ += sizeof(uint32_t);
|
|
RTC_CHECK_EQ(data_ix_ - start_ix, kNumBytesToFuzz);
|
|
}
|
|
|
|
bool ended_ = false;
|
|
rtc::ArrayView<const uint8_t> data_;
|
|
size_t data_ix_ = 0;
|
|
std::unique_ptr<EncodeNetEqInput> input_;
|
|
std::unique_ptr<PacketData> packet_;
|
|
};
|
|
} // namespace
|
|
|
|
void FuzzOneInputTest(const uint8_t* data, size_t size) {
|
|
// Limit the input size to 100000 bytes to avoid fuzzer timeout.
|
|
if (size > 100000)
|
|
return;
|
|
|
|
std::unique_ptr<FuzzRtpInput> input(
|
|
new FuzzRtpInput(rtc::ArrayView<const uint8_t>(data, size)));
|
|
std::unique_ptr<AudioChecksum> output(new AudioChecksum);
|
|
NetEqTest::Callbacks callbacks;
|
|
NetEq::Config config;
|
|
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[93] = std::make_pair(NetEqDecoder::kDecoderPCM16B, "pcm16-nb");
|
|
codecs[94] = std::make_pair(NetEqDecoder::kDecoderPCM16Bwb, "pcm16-wb");
|
|
codecs[96] =
|
|
std::make_pair(NetEqDecoder::kDecoderPCM16Bswb48kHz, "pcm16-swb48");
|
|
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");
|
|
// This is the payload type that will be used for encoding.
|
|
codecs[kPayloadType] =
|
|
std::make_pair(NetEqDecoder::kDecoderPCM16Bswb32kHz, "pcm16-swb32");
|
|
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
|