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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 next packet is | |
ivoc
2016/06/21 07:59:58
In the first sentence when you say "next packet" y
hlundin-webrtc
2016/06/21 09:12:20
Good point. I re-wrote this.
| |
58 // pre-fetched in the NetEqInput object, such that future calls to | |
59 // NextPacketTime() or NextHeader() will return information from the next | |
60 // packet in line. | |
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 GetPacket(). | |
ivoc
2016/06/21 07:59:58
GetPacket() -> PopPacket().
hlundin-webrtc
2016/06/21 09:12:20
Done.
| |
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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