1 /* 2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_ 12 #define MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_ 13 14 #include <bitset> 15 #include <memory> 16 17 #include "modules/audio_coding/neteq/tools/packet.h" 18 19 namespace webrtc { 20 namespace test { 21 22 // Interface class for an object delivering RTP packets to test applications. 23 class PacketSource { 24 public: 25 PacketSource(); 26 virtual ~PacketSource(); 27 28 PacketSource(const PacketSource&) = delete; 29 PacketSource& operator=(const PacketSource&) = delete; 30 31 // Returns next packet. Returns nullptr if the source is depleted, or if an 32 // error occurred. 33 virtual std::unique_ptr<Packet> NextPacket() = 0; 34 35 virtual void FilterOutPayloadType(uint8_t payload_type); 36 37 protected: 38 std::bitset<128> filter_; // Payload type is 7 bits in the RFC. 39 }; 40 41 } // namespace test 42 } // namespace webrtc 43 #endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_ 44