|  | /* | 
|  | *  Copyright (c) 2016 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_NETEQ_INPUT_H_ | 
|  | #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_ | 
|  |  | 
|  | #include <algorithm> | 
|  | #include <memory> | 
|  |  | 
|  | #include "webrtc/base/buffer.h" | 
|  | #include "webrtc/base/optional.h" | 
|  | #include "webrtc/modules/audio_coding/neteq/tools/packet.h" | 
|  | #include "webrtc/modules/audio_coding/neteq/tools/packet_source.h" | 
|  | #include "webrtc/modules/include/module_common_types.h" | 
|  |  | 
|  | namespace webrtc { | 
|  | namespace test { | 
|  |  | 
|  | // Interface class for input to the NetEqTest class. | 
|  | class NetEqInput { | 
|  | public: | 
|  | struct PacketData { | 
|  | WebRtcRTPHeader header; | 
|  | rtc::Buffer payload; | 
|  | double time_ms; | 
|  | }; | 
|  |  | 
|  | virtual ~NetEqInput() = default; | 
|  |  | 
|  | // Returns at what time (in ms) NetEq::InsertPacket should be called next, or | 
|  | // empty if the source is out of packets. | 
|  | virtual rtc::Optional<int64_t> NextPacketTime() const = 0; | 
|  |  | 
|  | // Returns at what time (in ms) NetEq::GetAudio should be called next, or | 
|  | // empty if no more output events are available. | 
|  | virtual rtc::Optional<int64_t> NextOutputEventTime() const = 0; | 
|  |  | 
|  | // Returns the time (in ms) for the next event from either NextPacketTime() | 
|  | // or NextOutputEventTime(), or empty if both are out of events. | 
|  | rtc::Optional<int64_t> NextEventTime() const { | 
|  | const auto a = NextPacketTime(); | 
|  | const auto b = NextOutputEventTime(); | 
|  | // Return the minimum of non-empty |a| and |b|, or empty if both are empty. | 
|  | if (a) { | 
|  | return b ? rtc::Optional<int64_t>(std::min(*a, *b)) : a; | 
|  | } | 
|  | return b ? b : rtc::Optional<int64_t>(); | 
|  | } | 
|  |  | 
|  | // Returns the next packet to be inserted into NetEq. The packet following the | 
|  | // returned one is pre-fetched in the NetEqInput object, such that future | 
|  | // calls to NextPacketTime() or NextHeader() will return information from that | 
|  | // packet. | 
|  | virtual std::unique_ptr<PacketData> PopPacket() = 0; | 
|  |  | 
|  | // Move to the next output event. This will make NextOutputEventTime() return | 
|  | // a new value (potentially the same if several output events share the same | 
|  | // time). | 
|  | virtual void AdvanceOutputEvent() = 0; | 
|  |  | 
|  | // Returns true if the source has come to an end. An implementation must | 
|  | // eventually return true from this method, or the test will end up in an | 
|  | // infinite loop. | 
|  | virtual bool ended() const = 0; | 
|  |  | 
|  | // Returns the RTP header for the next packet, i.e., the packet that will be | 
|  | // delivered next by PopPacket(). | 
|  | virtual rtc::Optional<RTPHeader> NextHeader() const = 0; | 
|  | }; | 
|  |  | 
|  | }  // namespace test | 
|  | }  // namespace webrtc | 
|  | #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_ |