henrik.lundin@webrtc.org | 9a40081 | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 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 | |
| 18 | namespace webrtc { |
| 19 | |
| 20 | // Implementation of the DecisionLogic class for playout modes kPlayoutOn and |
| 21 | // kPlayoutStreaming. |
| 22 | class 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_ |