| /* |
| * Copyright (c) 2014 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 MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_ |
| #define MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_ |
| |
| #include <bitset> |
| #include <memory> |
| |
| #include "modules/audio_coding/neteq/tools/packet.h" |
| #include "rtc_base/constructormagic.h" |
| |
| namespace webrtc { |
| namespace test { |
| |
| // Interface class for an object delivering RTP packets to test applications. |
| class PacketSource { |
| public: |
| PacketSource(); |
| virtual ~PacketSource(); |
| |
| // Returns next packet. Returns nullptr if the source is depleted, or if an |
| // error occurred. |
| virtual std::unique_ptr<Packet> NextPacket() = 0; |
| |
| virtual void FilterOutPayloadType(uint8_t payload_type); |
| |
| protected: |
| std::bitset<128> filter_; // Payload type is 7 bits in the RFC. |
| |
| private: |
| RTC_DISALLOW_COPY_AND_ASSIGN(PacketSource); |
| }; |
| |
| } // namespace test |
| } // namespace webrtc |
| #endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_ |