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Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
44 lines
1.3 KiB
C++
44 lines
1.3 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/constructor_magic.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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protected:
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std::bitset<128> filter_; // Payload type is 7 bits in the RFC.
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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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