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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
50 lines
1.5 KiB
C++
50 lines
1.5 KiB
C++
/*
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* Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_
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#define MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_
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#include <bitset>
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#include <memory>
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#include "modules/audio_coding/neteq/tools/packet.h"
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#include "rtc_base/constructormagic.h"
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#include "typedefs.h"
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namespace webrtc {
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namespace test {
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// Interface class for an object delivering RTP packets to test applications.
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class PacketSource {
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public:
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PacketSource();
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virtual ~PacketSource();
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// Returns next packet. Returns nullptr if the source is depleted, or if an
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// error occurred.
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virtual std::unique_ptr<Packet> NextPacket() = 0;
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virtual void FilterOutPayloadType(uint8_t payload_type);
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virtual void SelectSsrc(uint32_t ssrc);
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protected:
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std::bitset<128> filter_; // Payload type is 7 bits in the RFC.
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// If SSRC filtering discards all packet that do not match the SSRC.
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bool use_ssrc_filter_; // True when SSRC filtering is active.
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uint32_t ssrc_; // The selected SSRC. All other SSRCs will be discarded.
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private:
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RTC_DISALLOW_COPY_AND_ASSIGN(PacketSource);
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};
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} // namespace test
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_
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