58466f6d97
The original CL (https://codereview.webrtc.org/2315633002) was reverted since the fuzzer depended on gflags and files in the resources folder; neither of this is allowed for a fuzzer test in Chromium. This new version streamlines the dependencies, and changes the test to generate a sinusoid input audio signal instead of reading from a file. Original commit message: This CL introduces a new fuzzer target neteq_rtp_fuzzer that manipulates the RTP header fields before inserting the packets into NetEq. A few helper classes are also introduced. BUG=webrtc:5447 CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.win:win_chromium_rel_ng;master.tryserver.chromium.android:android_compile_dbg,linux_android_rel_ng;master.tryserver.chromium.linux:linux_chromium_rel_ng;master.tryserver.chromium.mac:mac_chromium_rel_ng,ios-device Review-Url: https://codereview.webrtc.org/2384423002 Cr-Commit-Position: refs/heads/master@{#14523}
81 lines
2.9 KiB
C++
81 lines
2.9 KiB
C++
/*
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* Copyright (c) 2016 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_NETEQ_INPUT_H_
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#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_
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#include <algorithm>
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#include <memory>
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#include "webrtc/base/buffer.h"
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#include "webrtc/base/optional.h"
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#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
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#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
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#include "webrtc/modules/include/module_common_types.h"
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namespace webrtc {
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namespace test {
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// Interface class for input to the NetEqTest class.
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class NetEqInput {
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public:
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struct PacketData {
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WebRtcRTPHeader header;
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rtc::Buffer payload;
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double time_ms;
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};
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virtual ~NetEqInput() = default;
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// Returns at what time (in ms) NetEq::InsertPacket should be called next, or
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// empty if the source is out of packets.
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virtual rtc::Optional<int64_t> NextPacketTime() const = 0;
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// Returns at what time (in ms) NetEq::GetAudio should be called next, or
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// empty if no more output events are available.
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virtual rtc::Optional<int64_t> NextOutputEventTime() const = 0;
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// Returns the time (in ms) for the next event from either NextPacketTime()
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// or NextOutputEventTime(), or empty if both are out of events.
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rtc::Optional<int64_t> NextEventTime() const {
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const auto a = NextPacketTime();
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const auto b = NextOutputEventTime();
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// Return the minimum of non-empty |a| and |b|, or empty if both are empty.
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if (a) {
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return b ? rtc::Optional<int64_t>(std::min(*a, *b)) : a;
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}
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return b ? b : rtc::Optional<int64_t>();
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}
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// Returns the next packet to be inserted into NetEq. The packet following the
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// returned one is pre-fetched in the NetEqInput object, such that future
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// calls to NextPacketTime() or NextHeader() will return information from that
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// packet.
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virtual std::unique_ptr<PacketData> PopPacket() = 0;
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// Move to the next output event. This will make NextOutputEventTime() return
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// a new value (potentially the same if several output events share the same
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// time).
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virtual void AdvanceOutputEvent() = 0;
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// Returns true if the source has come to an end. An implementation must
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// eventually return true from this method, or the test will end up in an
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// infinite loop.
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virtual bool ended() const = 0;
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// Returns the RTP header for the next packet, i.e., the packet that will be
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// delivered next by PopPacket().
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virtual rtc::Optional<RTPHeader> NextHeader() const = 0;
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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_NETEQ_INPUT_H_
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