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| 1 /* |
| 2 * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. |
| 3 * |
| 4 * Use of this source code is governed by a BSD-style license |
| 5 * that can be found in the LICENSE file in the root of the source |
| 6 * tree. An additional intellectual property rights grant can be found |
| 7 * in the file PATENTS. All contributing project authors may |
| 8 * be found in the AUTHORS file in the root of the source tree. |
| 9 */ |
| 10 |
| 11 #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_ |
| 12 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_ |
| 13 |
| 14 #include <algorithm> |
| 15 #include <memory> |
| 16 |
| 17 #include "webrtc/base/buffer.h" |
| 18 #include "webrtc/base/optional.h" |
| 19 #include "webrtc/modules/audio_coding/neteq/tools/packet.h" |
| 20 #include "webrtc/modules/audio_coding/neteq/tools/packet_source.h" |
| 21 #include "webrtc/modules/include/module_common_types.h" |
| 22 |
| 23 namespace webrtc { |
| 24 namespace test { |
| 25 |
| 26 // Interface class for input to the NetEqTest class. |
| 27 class NetEqInput { |
| 28 public: |
| 29 struct PacketData { |
| 30 WebRtcRTPHeader header; |
| 31 rtc::Buffer payload; |
| 32 double time_ms; |
| 33 }; |
| 34 |
| 35 virtual ~NetEqInput() = default; |
| 36 |
| 37 // Returns at what time (in ms) NetEq::InsertPacket should be called next, or |
| 38 // empty if the source is out of packets. |
| 39 virtual rtc::Optional<int64_t> NextPacketTime() const = 0; |
| 40 |
| 41 // Returns at what time (in ms) NetEq::GetAudio should be called next, or |
| 42 // empty if no more output events are available. |
| 43 virtual rtc::Optional<int64_t> NextOutputEventTime() const = 0; |
| 44 |
| 45 // Returns the time (in ms) for the next event from either NextPacketTime() |
| 46 // or NextOutputEventTime(), or empty if both are out of events. |
| 47 rtc::Optional<int64_t> NextEventTime() const { |
| 48 const auto a = NextPacketTime(); |
| 49 const auto b = NextOutputEventTime(); |
| 50 // Return the minimum of non-empty |a| and |b|, or empty if both are empty. |
| 51 if (a) { |
| 52 return b ? rtc::Optional<int64_t>(std::min(*a, *b)) : a; |
| 53 } |
| 54 return b ? b : rtc::Optional<int64_t>(); |
| 55 } |
| 56 |
| 57 // Returns the next packet to be inserted into NetEq. The packet following the |
| 58 // returned one is pre-fetched in the NetEqInput object, such that future |
| 59 // calls to NextPacketTime() or NextHeader() will return information from that |
| 60 // packet. |
| 61 virtual std::unique_ptr<PacketData> PopPacket() = 0; |
| 62 |
| 63 // Move to the next output event. This will make NextOutputEventTime() return |
| 64 // a new value (potentially the same if several output events share the same |
| 65 // time). |
| 66 virtual void AdvanceOutputEvent() = 0; |
| 67 |
| 68 // Returns true if the source has come to an end. |
| 69 virtual bool ended() const = 0; |
| 70 |
| 71 // Returns the RTP header for the next packet, i.e., the packet that will be |
| 72 // delivered next by PopPacket(). |
| 73 virtual rtc::Optional<RTPHeader> NextHeader() const = 0; |
| 74 }; |
| 75 |
| 76 } // namespace test |
| 77 } // namespace webrtc |
| 78 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_ |
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