Relanding "Setting up an RTP input fuzzer for NetEq"

The original CL (https://codereview.webrtc.org/2315633002) was
reverted since the fuzzer depended on gflags and files in the
resources folder; neither of this is allowed for a fuzzer test in
Chromium. This new version streamlines the dependencies, and changes
the test to generate a sinusoid input audio signal instead of reading
from a file.

Original commit message:
This CL introduces a new fuzzer target neteq_rtp_fuzzer that
manipulates the RTP header fields before inserting the packets into
NetEq. A few helper classes are also introduced.

BUG=webrtc:5447
CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.win:win_chromium_rel_ng;master.tryserver.chromium.android:android_compile_dbg,linux_android_rel_ng;master.tryserver.chromium.linux:linux_chromium_rel_ng;master.tryserver.chromium.mac:mac_chromium_rel_ng,ios-device

Review-Url: https://codereview.webrtc.org/2384423002
Cr-Commit-Position: refs/heads/master@{#14523}

webrtc/modules/audio_coding/BUILD.gn

@@ -868,6 +868,25 @@ rtc_static_library("neteq") {
   }
 }
 
+# Although providing only test support, this target must be outside of the
+# rtc_include_tests conditional. The reason is that it supports fuzzer tests
+# that ultimately are built and run as a part of the Chromium ecosystem, which
+# does not set the rtc_include_tests flag.
+rtc_source_set("neteq_test_minimal") {
+  testonly = true
+  sources = [
+    "neteq/tools/encode_neteq_input.cc",
+    "neteq/tools/encode_neteq_input.h",
+    "neteq/tools/neteq_test.cc",
+    "neteq/tools/neteq_test.h",
+  ]
+
+  if (is_clang) {
+    # Suppress warnings from the Chromium Clang plugins (bugs.webrtc.org/163).
+    suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+  }
+}
+
 if (rtc_include_tests) {
   rtc_source_set("acm_receive_test") {
     testonly = true
@@ -1156,8 +1175,6 @@ if (rtc_include_tests) {
       "neteq/tools/neteq_packet_source_input.h",
       "neteq/tools/neteq_replacement_input.cc",
       "neteq/tools/neteq_replacement_input.h",
-      "neteq/tools/neteq_test.cc",
-      "neteq/tools/neteq_test.h",
       "neteq/tools/output_audio_file.h",
       "neteq/tools/output_wav_file.h",
       "neteq/tools/packet.cc",
@@ -1180,6 +1197,7 @@ if (rtc_include_tests) {
     }
 
     deps = [
+      ":neteq_test_minimal",
       "../../common_audio",
       "../../test:rtp_test_utils",
       "../rtp_rtcp",

webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.cc

@@ -0,0 +1,88 @@
+/*
+ *  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 "webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.h"
+
+#include <utility>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/safe_conversions.h"
+
+namespace webrtc {
+namespace test {
+
+EncodeNetEqInput::EncodeNetEqInput(std::unique_ptr<Generator> generator,
+                                   std::unique_ptr<AudioEncoder> encoder,
+                                   int64_t input_duration_ms)
+    : generator_(std::move(generator)),
+      encoder_(std::move(encoder)),
+      input_duration_ms_(input_duration_ms) {
+  CreatePacket();
+}
+
+rtc::Optional<int64_t> EncodeNetEqInput::NextPacketTime() const {
+  RTC_DCHECK(packet_data_);
+  return rtc::Optional<int64_t>(static_cast<int64_t>(packet_data_->time_ms));
+}
+
+rtc::Optional<int64_t> EncodeNetEqInput::NextOutputEventTime() const {
+  return rtc::Optional<int64_t>(next_output_event_ms_);
+}
+
+std::unique_ptr<NetEqInput::PacketData> EncodeNetEqInput::PopPacket() {
+  RTC_DCHECK(packet_data_);
+  // Grab the packet to return...
+  std::unique_ptr<PacketData> packet_to_return = std::move(packet_data_);
+  // ... and line up the next packet for future use.
+  CreatePacket();
+
+  return packet_to_return;
+}
+
+void EncodeNetEqInput::AdvanceOutputEvent() {
+  next_output_event_ms_ += kOutputPeriodMs;
+}
+
+rtc::Optional<RTPHeader> EncodeNetEqInput::NextHeader() const {
+  RTC_DCHECK(packet_data_);
+  return rtc::Optional<RTPHeader>(packet_data_->header.header);
+}
+
+void EncodeNetEqInput::CreatePacket() {
+  // Create a new PacketData object.
+  RTC_DCHECK(!packet_data_);
+  packet_data_.reset(new NetEqInput::PacketData);
+  RTC_DCHECK_EQ(packet_data_->payload.size(), 0u);
+
+  // Loop until we get a packet.
+  AudioEncoder::EncodedInfo info;
+  RTC_DCHECK(!info.send_even_if_empty);
+  int num_blocks = 0;
+  while (packet_data_->payload.size() == 0 && !info.send_even_if_empty) {
+    const size_t num_samples = rtc::CheckedDivExact(
+        static_cast<int>(encoder_->SampleRateHz() * kOutputPeriodMs), 1000);
+
+    info = encoder_->Encode(rtp_timestamp_, generator_->Generate(num_samples),
+                            &packet_data_->payload);
+
+    rtp_timestamp_ += rtc::checked_cast<uint32_t>(
+        num_samples * encoder_->RtpTimestampRateHz() /
+        encoder_->SampleRateHz());
+    ++num_blocks;
+  }
+  packet_data_->header.header.timestamp = info.encoded_timestamp;
+  packet_data_->header.header.payloadType = info.payload_type;
+  packet_data_->header.header.sequenceNumber = sequence_number_++;
+  packet_data_->time_ms = next_packet_time_ms_;
+  next_packet_time_ms_ += num_blocks * kOutputPeriodMs;
+}
+
+}  // namespace test
+}  // namespace webrtc

webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.h

@@ -0,0 +1,71 @@
+/*
+ *  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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_ENCODE_NETEQ_INPUT_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_ENCODE_NETEQ_INPUT_H_
+
+#include <memory>
+
+#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
+#include "webrtc/modules/audio_coding/neteq/tools/neteq_input.h"
+#include "webrtc/modules/include/module_common_types.h"
+
+namespace webrtc {
+namespace test {
+
+// This class provides a NetEqInput that takes audio from a generator object and
+// encodes it using a given audio encoder.
+class EncodeNetEqInput : public NetEqInput {
+ public:
+  // Generator class, to be provided to the EncodeNetEqInput constructor.
+  class Generator {
+   public:
+    virtual ~Generator() = default;
+    // Returns the next num_samples values from the signal generator.
+    virtual rtc::ArrayView<const int16_t> Generate(size_t num_samples) = 0;
+  };
+
+  // The source will end after the given input duration.
+  EncodeNetEqInput(std::unique_ptr<Generator> generator,
+                   std::unique_ptr<AudioEncoder> encoder,
+                   int64_t input_duration_ms);
+
+  rtc::Optional<int64_t> NextPacketTime() const override;
+
+  rtc::Optional<int64_t> NextOutputEventTime() const override;
+
+  std::unique_ptr<PacketData> PopPacket() override;
+
+  void AdvanceOutputEvent() override;
+
+  bool ended() const override {
+    return next_output_event_ms_ <= input_duration_ms_;
+  }
+
+  rtc::Optional<RTPHeader> NextHeader() const override;
+
+ private:
+  static constexpr int64_t kOutputPeriodMs = 10;
+
+  void CreatePacket();
+
+  std::unique_ptr<Generator> generator_;
+  std::unique_ptr<AudioEncoder> encoder_;
+  std::unique_ptr<PacketData> packet_data_;
+  uint32_t rtp_timestamp_ = 0;
+  int16_t sequence_number_ = 0;
+  int64_t next_packet_time_ms_ = 0;
+  int64_t next_output_event_ms_ = 0;
+  const int64_t input_duration_ms_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_ENCODE_NETEQ_INPUT_H_

webrtc/modules/audio_coding/neteq/tools/neteq_input.h

@@ -65,7 +65,9 @@ class NetEqInput {
   // time).
   virtual void AdvanceOutputEvent() = 0;
 
-  // Returns true if the source has come to an end.
+  // Returns true if the source has come to an end. An implementation must
+  // eventually return true from this method, or the test will end up in an
+  // infinite loop.
   virtual bool ended() const = 0;
 
   // Returns the RTP header for the next packet, i.e., the packet that will be