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source: webrtc/webrtc/modules/audio_coding/neteq4/decision_logic_normal.h @ 0:4bda6873e34c

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Last change on this file since 0:4bda6873e34c was 0:4bda6873e34c, checked in by Michael Prittie <mprittie@…>, 6 years ago

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1/*
2 *  Copyright (c) 2013 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_NETEQ4_DECISION_LOGIC_NORMAL_H_
12#define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DECISION_LOGIC_NORMAL_H_
13
14#include "webrtc/modules/audio_coding/neteq4/decision_logic.h"
15#include "webrtc/system_wrappers/interface/constructor_magic.h"
16#include "webrtc/typedefs.h"
17
18namespace webrtc {
19
20// Implementation of the DecisionLogic class for playout modes kPlayoutOn and
21// kPlayoutStreaming.
22class DecisionLogicNormal : public DecisionLogic {
23 public:
24  // Constructor.
25  DecisionLogicNormal(int fs_hz,
26                      int output_size_samples,
27                      NetEqPlayoutMode playout_mode,
28                      DecoderDatabase* decoder_database,
29                      const PacketBuffer& packet_buffer,
30                      DelayManager* delay_manager,
31                      BufferLevelFilter* buffer_level_filter)
32      : DecisionLogic(fs_hz, output_size_samples, playout_mode,
33                      decoder_database, packet_buffer, delay_manager,
34                      buffer_level_filter) {
35  }
36
37  // Destructor.
38  virtual ~DecisionLogicNormal() {}
39
40 protected:
41  // Returns the operation that should be done next. |sync_buffer| and |expand|
42  // are provided for reference. |decoder_frame_length| is the number of samples
43  // obtained from the last decoded frame. If there is a packet available, the
44  // packet header should be supplied in |packet_header|; otherwise it should
45  // be NULL. The mode resulting form the last call to NetEqImpl::GetAudio is
46  // supplied in |prev_mode|. If there is a DTMF event to play, |play_dtmf|
47  // should be set to true. The output variable |reset_decoder| will be set to
48  // true if a reset is required; otherwise it is left unchanged (i.e., it can
49  // remain true if it was true before the call).
50  virtual Operations GetDecisionSpecialized(const SyncBuffer& sync_buffer,
51                                            const Expand& expand,
52                                            int decoder_frame_length,
53                                            const RTPHeader* packet_header,
54                                            Modes prev_mode, bool play_dtmf,
55                                            bool* reset_decoder);
56
57 private:
58  static const int kAllowMergeWithoutExpandMs = 20;  // 20 ms.
59  static const int kReinitAfterExpands = 100;
60  static const int kMaxWaitForPacket = 10;
61
62  // Returns the operation given that the next available packet is a comfort
63  // noise payload (RFC 3389 only, not codec-internal).
64  Operations CngOperation(Modes prev_mode, uint32_t target_timestamp,
65                          uint32_t available_timestamp);
66
67  // Returns the operation given that no packets are available (except maybe
68  // a DTMF event, flagged by setting |play_dtmf| true).
69  Operations NoPacket(bool play_dtmf);
70
71  // Returns the operation to do given that the expected packet is available.
72  Operations ExpectedPacketAvailable(Modes prev_mode, bool play_dtmf);
73
74  // Returns the operation to do given that the expected packet is not
75  // available, but a packet further into the future is at hand.
76  Operations FuturePacketAvailable(const SyncBuffer& sync_buffer,
77                                   const Expand& expand,
78                                   int decoder_frame_length, Modes prev_mode,
79                                   uint32_t target_timestamp,
80                                   uint32_t available_timestamp,
81                                   bool play_dtmf);
82
83  // Checks if enough time has elapsed since the last successful timescale
84  // operation was done (i.e., accelerate or preemptive expand).
85  bool TimescaleAllowed() const { return timescale_hold_off_ == 0; }
86
87  // Checks if the current (filtered) buffer level is under the target level.
88  bool UnderTargetLevel() const;
89
90  // Checks if |timestamp_leap| is so long into the future that a reset due
91  // to exceeding kReinitAfterExpands will be done.
92  bool ReinitAfterExpands(uint32_t timestamp_leap) const;
93
94  // Checks if we still have not done enough expands to cover the distance from
95  // the last decoded packet to the next available packet, the distance beeing
96  // conveyed in |timestamp_leap|.
97  bool PacketTooEarly(uint32_t timestamp_leap) const;
98
99  // Checks if num_consecutive_expands_ >= kMaxWaitForPacket.
100  bool MaxWaitForPacket() const;
101
102  DISALLOW_COPY_AND_ASSIGN(DecisionLogicNormal);
103};
104
105}  // namespace webrtc
106#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_DECISION_LOGIC_NORMAL_H_
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