session-android/jni/webrtc/modules/audio_coding/neteq/tools/packet_source.h
Moxie Marlinspike d83a3d71bc Support for Signal calls.
Merge in RedPhone

// FREEBIE
2015-09-30 14:30:09 -07:00

48 lines
1.3 KiB
C++

/*
* 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 WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_
#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_
#include <bitset>
#include "webrtc/base/constructormagic.h"
#include "webrtc/typedefs.h"
namespace webrtc {
namespace test {
class Packet;
// Interface class for an object delivering RTP packets to test applications.
class PacketSource {
public:
PacketSource() {}
virtual ~PacketSource() {}
// Returns a pointer to the next packet. Returns NULL if the source is
// depleted, or if an error occurred.
virtual Packet* NextPacket() = 0;
virtual void FilterOutPayloadType(uint8_t payload_type) {
filter_.set(payload_type, true);
}
protected:
std::bitset<128> filter_; // Payload type is 7 bits in the RFC.
private:
DISALLOW_COPY_AND_ASSIGN(PacketSource);
};
} // namespace test
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_