Setting up an RTP input fuzzer for NetEq

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

Index: webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.h
diff --git a/webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.h b/webrtc/modules/audio_coding/neteq/tools/encode_neteq_input.h
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index 0000000000000000000000000000000000000000..ab28fd96cf4b388d0cd33aba6da18812410d15e8
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+/*
+ *  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/input_audio_file.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 an input file and
+// encodes it using a given audio encoder.
+class EncodeNetEqInput : public NetEqInput {
+ public:
+  // The source will end after the given input duration.
+  EncodeNetEqInput(std::unique_ptr<InputAudioFile> input,
+                   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<InputAudioFile> input_;
+  std::unique_ptr<AudioEncoder> encoder_;
+  std::unique_ptr<PacketData> packet_data_;
+  int32_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_