3c089d751e
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition. * DISALLOW_ASSIGN -> RTC_DISALLOW_ASSIGN * DISALLOW_COPY_AND_ASSIGN -> RTC_DISALLOW_COPY_AND_ASSIGN * DISALLOW_IMPLICIT_CONSTRUCTORS -> RTC_DISALLOW_IMPLICIT_CONSTRUCTORS Related CL: https://codereview.webrtc.org/1335923002/ BUG=chromium:468375 NOTRY=true Review URL: https://codereview.webrtc.org/1345433002 Cr-Commit-Position: refs/heads/master@{#9953}
55 lines
1.7 KiB
C++
55 lines
1.7 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 WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_
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#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_
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#include <bitset>
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
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#include "webrtc/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() : use_ssrc_filter_(false), ssrc_(0) {}
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virtual ~PacketSource() {}
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// Returns a pointer to the next packet. Returns NULL if the source is
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// depleted, or if an error occurred.
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virtual Packet* NextPacket() = 0;
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virtual void FilterOutPayloadType(uint8_t payload_type) {
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filter_.set(payload_type, true);
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}
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virtual void SelectSsrc(uint32_t ssrc) {
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use_ssrc_filter_ = true;
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ssrc_ = ssrc;
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}
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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 // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_PACKET_SOURCE_H_
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