kjellander c8fa692ec4 Update includes for webrtc/{base => rtc_base} rename (1/3)
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`

The only manual edit is to add an include of webrtc/rtc_base/checks.h in
webrtc/modules/audio_device/android/opensles_common.h, which likely
was needed due to changed include paths due to 'git cl format'.

BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True

Review-Url: https://codereview.webrtc.org/2969653002
Cr-Commit-Position: refs/heads/master@{#18871}
2017-06-30 21:02:00 +00:00

51 lines
1.6 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 <memory>
#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
#include "webrtc/rtc_base/constructormagic.h"
#include "webrtc/typedefs.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);
virtual void SelectSsrc(uint32_t 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_