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/*
* 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/modules/audio_coding/neteq/tools/packet.h"
#include "webrtc/typedefs.h"
namespace webrtc {
namespace test {
// Interface class for an object delivering RTP packets to test applications.
class PacketSource {
public:
PacketSource() : use_ssrc_filter_(false), ssrc_(0) {}
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);
}
virtual void SelectSsrc(uint32_t ssrc) {
use_ssrc_filter_ = true;
ssrc_ = ssrc;
}
protected:
std::bitset<128> filter_; // Payload type is 7 bits in the RFC.
// If SSRC filtering discards all packet that do not match the SSRC.
bool use_ssrc_filter_; // True when SSRC filtering is active.
uint32_t ssrc_; // The selected SSRC. All other SSRCs will be discarded.
private:
RTC_DISALLOW_COPY_AND_ASSIGN(PacketSource);
};
} // namespace test
} // namespace webrtc
#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_
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