henrik.lundin@webrtc.org | d94659d | 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 | |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 11 | #ifndef MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_H_ |
| 12 | #define MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_H_ |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 13 | |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 14 | #include "modules/audio_coding/neteq/defines.h" |
| 15 | #include "modules/audio_coding/neteq/include/neteq.h" |
| 16 | #include "modules/audio_coding/neteq/tick_timer.h" |
| 17 | #include "rtc_base/constructormagic.h" |
Mirko Bonadei | 7120742 | 2017-09-15 13:58:09 +0200 | [diff] [blame] | 18 | #include "typedefs.h" // NOLINT(build/include) |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 19 | |
| 20 | namespace webrtc { |
| 21 | |
| 22 | // Forward declarations. |
| 23 | class BufferLevelFilter; |
| 24 | class DecoderDatabase; |
| 25 | class DelayManager; |
| 26 | class Expand; |
| 27 | class PacketBuffer; |
| 28 | class SyncBuffer; |
ossu | 7a37761 | 2016-10-18 04:06:13 -0700 | [diff] [blame] | 29 | struct Packet; |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 30 | |
| 31 | // This is the base class for the decision tree implementations. Derived classes |
| 32 | // must implement the method GetDecisionSpecialized(). |
| 33 | class DecisionLogic { |
| 34 | public: |
| 35 | // Static factory function which creates different types of objects depending |
| 36 | // on the |playout_mode|. |
| 37 | static DecisionLogic* Create(int fs_hz, |
Peter Kasting | dce40cf | 2015-08-24 14:52:23 -0700 | [diff] [blame] | 38 | size_t output_size_samples, |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 39 | NetEqPlayoutMode playout_mode, |
| 40 | DecoderDatabase* decoder_database, |
| 41 | const PacketBuffer& packet_buffer, |
| 42 | DelayManager* delay_manager, |
Henrik Lundin | 47b17dc | 2016-05-10 10:20:59 +0200 | [diff] [blame] | 43 | BufferLevelFilter* buffer_level_filter, |
| 44 | const TickTimer* tick_timer); |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 45 | |
| 46 | // Constructor. |
| 47 | DecisionLogic(int fs_hz, |
Peter Kasting | dce40cf | 2015-08-24 14:52:23 -0700 | [diff] [blame] | 48 | size_t output_size_samples, |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 49 | NetEqPlayoutMode playout_mode, |
| 50 | DecoderDatabase* decoder_database, |
| 51 | const PacketBuffer& packet_buffer, |
| 52 | DelayManager* delay_manager, |
Henrik Lundin | 47b17dc | 2016-05-10 10:20:59 +0200 | [diff] [blame] | 53 | BufferLevelFilter* buffer_level_filter, |
| 54 | const TickTimer* tick_timer); |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 55 | |
Henrik Lundin | 47b17dc | 2016-05-10 10:20:59 +0200 | [diff] [blame] | 56 | virtual ~DecisionLogic(); |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 57 | |
| 58 | // Resets object to a clean state. |
| 59 | void Reset(); |
| 60 | |
| 61 | // Resets parts of the state. Typically done when switching codecs. |
| 62 | void SoftReset(); |
| 63 | |
| 64 | // Sets the sample rate and the output block size. |
Peter Kasting | dce40cf | 2015-08-24 14:52:23 -0700 | [diff] [blame] | 65 | void SetSampleRate(int fs_hz, size_t output_size_samples); |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 66 | |
| 67 | // Returns the operation that should be done next. |sync_buffer| and |expand| |
| 68 | // are provided for reference. |decoder_frame_length| is the number of samples |
ossu | 7a37761 | 2016-10-18 04:06:13 -0700 | [diff] [blame] | 69 | // obtained from the last decoded frame. If there is a packet available, it |
| 70 | // should be supplied in |next_packet|; otherwise it should be NULL. The mode |
| 71 | // resulting from the last call to NetEqImpl::GetAudio is supplied in |
| 72 | // |prev_mode|. If there is a DTMF event to play, |play_dtmf| should be set to |
| 73 | // true. The output variable |reset_decoder| will be set to true if a reset is |
| 74 | // required; otherwise it is left unchanged (i.e., it can remain true if it |
| 75 | // was true before the call). This method end with calling |
| 76 | // GetDecisionSpecialized to get the actual return value. |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 77 | Operations GetDecision(const SyncBuffer& sync_buffer, |
| 78 | const Expand& expand, |
Peter Kasting | dce40cf | 2015-08-24 14:52:23 -0700 | [diff] [blame] | 79 | size_t decoder_frame_length, |
ossu | 7a37761 | 2016-10-18 04:06:13 -0700 | [diff] [blame] | 80 | const Packet* next_packet, |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 81 | Modes prev_mode, |
| 82 | bool play_dtmf, |
henrik.lundin | b1fb72b | 2016-05-03 08:18:47 -0700 | [diff] [blame] | 83 | size_t generated_noise_samples, |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 84 | bool* reset_decoder); |
| 85 | |
| 86 | // These methods test the |cng_state_| for different conditions. |
| 87 | bool CngRfc3389On() const { return cng_state_ == kCngRfc3389On; } |
| 88 | bool CngOff() const { return cng_state_ == kCngOff; } |
| 89 | |
| 90 | // Resets the |cng_state_| to kCngOff. |
| 91 | void SetCngOff() { cng_state_ = kCngOff; } |
| 92 | |
| 93 | // Reports back to DecisionLogic whether the decision to do expand remains or |
| 94 | // not. Note that this is necessary, since an expand decision can be changed |
| 95 | // to kNormal in NetEqImpl::GetDecision if there is still enough data in the |
| 96 | // sync buffer. |
turaj@webrtc.org | 8d1cdaa | 2014-04-11 18:47:55 +0000 | [diff] [blame] | 97 | virtual void ExpandDecision(Operations operation); |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 98 | |
| 99 | // Adds |value| to |sample_memory_|. |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame^] | 100 | void AddSampleMemory(int32_t value) { sample_memory_ += value; } |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 101 | |
| 102 | // Accessors and mutators. |
| 103 | void set_sample_memory(int32_t value) { sample_memory_ = value; } |
henrik.lundin | b1fb72b | 2016-05-03 08:18:47 -0700 | [diff] [blame] | 104 | size_t noise_fast_forward() const { return noise_fast_forward_; } |
Peter Kasting | dce40cf | 2015-08-24 14:52:23 -0700 | [diff] [blame] | 105 | size_t packet_length_samples() const { return packet_length_samples_; } |
| 106 | void set_packet_length_samples(size_t value) { |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 107 | packet_length_samples_ = value; |
| 108 | } |
| 109 | void set_prev_time_scale(bool value) { prev_time_scale_ = value; } |
| 110 | NetEqPlayoutMode playout_mode() const { return playout_mode_; } |
| 111 | |
| 112 | protected: |
Henrik Lundin | 47b17dc | 2016-05-10 10:20:59 +0200 | [diff] [blame] | 113 | // The value 5 sets maximum time-stretch rate to about 100 ms/s. |
| 114 | static const int kMinTimescaleInterval = 5; |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 115 | |
Yves Gerey | 665174f | 2018-06-19 15:03:05 +0200 | [diff] [blame^] | 116 | enum CngState { kCngOff, kCngRfc3389On, kCngInternalOn }; |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 117 | |
| 118 | // Returns the operation that should be done next. |sync_buffer| and |expand| |
| 119 | // are provided for reference. |decoder_frame_length| is the number of samples |
ossu | 7a37761 | 2016-10-18 04:06:13 -0700 | [diff] [blame] | 120 | // obtained from the last decoded frame. If there is a packet available, it |
| 121 | // should be supplied in |next_packet|; otherwise it should be NULL. The mode |
| 122 | // resulting from the last call to NetEqImpl::GetAudio is supplied in |
| 123 | // |prev_mode|. If there is a DTMF event to play, |play_dtmf| should be set to |
| 124 | // true. The output variable |reset_decoder| will be set to true if a reset is |
| 125 | // required; otherwise it is left unchanged (i.e., it can remain true if it |
| 126 | // was true before the call). Should be implemented by derived classes. |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 127 | virtual Operations GetDecisionSpecialized(const SyncBuffer& sync_buffer, |
| 128 | const Expand& expand, |
Peter Kasting | dce40cf | 2015-08-24 14:52:23 -0700 | [diff] [blame] | 129 | size_t decoder_frame_length, |
ossu | 7a37761 | 2016-10-18 04:06:13 -0700 | [diff] [blame] | 130 | const Packet* next_packet, |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 131 | Modes prev_mode, |
| 132 | bool play_dtmf, |
henrik.lundin | b1fb72b | 2016-05-03 08:18:47 -0700 | [diff] [blame] | 133 | bool* reset_decoder, |
Ivo Creusen | c7f09ad | 2018-05-22 13:21:01 +0200 | [diff] [blame] | 134 | size_t generated_noise_samples, |
| 135 | size_t cur_size_samples) = 0; |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 136 | |
| 137 | // Updates the |buffer_level_filter_| with the current buffer level |
| 138 | // |buffer_size_packets|. |
Peter Kasting | dce40cf | 2015-08-24 14:52:23 -0700 | [diff] [blame] | 139 | void FilterBufferLevel(size_t buffer_size_packets, Modes prev_mode); |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 140 | |
| 141 | DecoderDatabase* decoder_database_; |
| 142 | const PacketBuffer& packet_buffer_; |
| 143 | DelayManager* delay_manager_; |
| 144 | BufferLevelFilter* buffer_level_filter_; |
Henrik Lundin | 47b17dc | 2016-05-10 10:20:59 +0200 | [diff] [blame] | 145 | const TickTimer* tick_timer_; |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 146 | int fs_mult_; |
Peter Kasting | dce40cf | 2015-08-24 14:52:23 -0700 | [diff] [blame] | 147 | size_t output_size_samples_; |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 148 | CngState cng_state_; // Remember if comfort noise is interrupted by other |
| 149 | // event (e.g., DTMF). |
henrik.lundin | b1fb72b | 2016-05-03 08:18:47 -0700 | [diff] [blame] | 150 | size_t noise_fast_forward_ = 0; |
Peter Kasting | dce40cf | 2015-08-24 14:52:23 -0700 | [diff] [blame] | 151 | size_t packet_length_samples_; |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 152 | int sample_memory_; |
| 153 | bool prev_time_scale_; |
Henrik Lundin | 47b17dc | 2016-05-10 10:20:59 +0200 | [diff] [blame] | 154 | std::unique_ptr<TickTimer::Countdown> timescale_countdown_; |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 155 | int num_consecutive_expands_; |
| 156 | const NetEqPlayoutMode playout_mode_; |
| 157 | |
| 158 | private: |
henrikg | 3c089d7 | 2015-09-16 05:37:44 -0700 | [diff] [blame] | 159 | RTC_DISALLOW_COPY_AND_ASSIGN(DecisionLogic); |
henrik.lundin@webrtc.org | d94659d | 2013-01-29 12:09:21 +0000 | [diff] [blame] | 160 | }; |
| 161 | |
| 162 | } // namespace webrtc |
Mirko Bonadei | 92ea95e | 2017-09-15 06:47:31 +0200 | [diff] [blame] | 163 | #endif // MODULES_AUDIO_CODING_NETEQ_DECISION_LOGIC_H_ |