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henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +00001/*
2 * Copyright (c) 2012 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 Bonadei92ea95e2017-09-15 06:47:31 +020011#ifndef MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_
12#define MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000013
Alessio Bazzica8f319a32019-07-24 16:47:02 +000014#include <map>
kwiberg2d0c3322016-02-14 09:28:33 -080015#include <memory>
henrik.lundin4cf61dd2015-12-09 06:20:58 -080016#include <string>
Alessio Bazzica8f319a32019-07-24 16:47:02 +000017#include <utility>
18#include <vector>
henrik.lundin4cf61dd2015-12-09 06:20:58 -080019
Danil Chapovalovb6021232018-06-19 13:26:36 +020020#include "absl/types/optional.h"
Fredrik Solenbergbbf21a32018-04-12 22:44:09 +020021#include "api/audio/audio_frame.h"
Alessio Bazzica8f319a32019-07-24 16:47:02 +000022#include "api/rtp_packet_info.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020023#include "modules/audio_coding/neteq/audio_multi_vector.h"
Yves Gerey988cc082018-10-23 12:03:01 +020024#include "modules/audio_coding/neteq/defines.h" // Modes, Operations
Henrik Lundin3ef3bfc2018-04-10 15:10:26 +020025#include "modules/audio_coding/neteq/expand_uma_logger.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020026#include "modules/audio_coding/neteq/include/neteq.h"
Yves Gerey988cc082018-10-23 12:03:01 +020027#include "modules/audio_coding/neteq/packet.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020028#include "modules/audio_coding/neteq/random_vector.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020029#include "modules/audio_coding/neteq/statistics_calculator.h"
30#include "modules/audio_coding/neteq/tick_timer.h"
Steve Anton10542f22019-01-11 09:11:00 -080031#include "rtc_base/constructor_magic.h"
32#include "rtc_base/critical_section.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020033#include "rtc_base/thread_annotations.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000034
35namespace webrtc {
36
37// Forward declarations.
henrik.lundin@webrtc.org40d3fc62013-09-18 12:19:50 +000038class Accelerate;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000039class BackgroundNoise;
40class BufferLevelFilter;
Alessio Bazzica8f319a32019-07-24 16:47:02 +000041class Clock;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000042class ComfortNoise;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000043class DecisionLogic;
44class DecoderDatabase;
45class DelayManager;
46class DelayPeakDetector;
47class DtmfBuffer;
48class DtmfToneGenerator;
49class Expand;
henrik.lundin@webrtc.org40d3fc62013-09-18 12:19:50 +000050class Merge;
henrik.lundin91951862016-06-08 06:43:41 -070051class NackTracker;
henrik.lundin@webrtc.org40d3fc62013-09-18 12:19:50 +000052class Normal;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000053class PacketBuffer;
ossua70695a2016-09-22 02:06:28 -070054class RedPayloadSplitter;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000055class PostDecodeVad;
henrik.lundin@webrtc.org40d3fc62013-09-18 12:19:50 +000056class PreemptiveExpand;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000057class RandomVector;
58class SyncBuffer;
59class TimestampScaler;
henrik.lundin@webrtc.orgd9faa462014-01-14 10:18:45 +000060struct AccelerateFactory;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000061struct DtmfEvent;
henrik.lundin@webrtc.orgd9faa462014-01-14 10:18:45 +000062struct ExpandFactory;
63struct PreemptiveExpandFactory;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000064
65class NetEqImpl : public webrtc::NetEq {
66 public:
Alex Narest0a88ea02019-07-29 18:07:36 +020067 enum class OutputType {
68 kNormalSpeech,
69 kPLC,
70 kCNG,
71 kPLCCNG,
72 kVadPassive,
73 kCodecPLC
74 };
henrik.lundin55480f52016-03-08 02:37:57 -080075
Henrik Lundinc417d9e2017-06-14 12:29:03 +020076 enum ErrorCodes {
77 kNoError = 0,
78 kOtherError,
79 kUnknownRtpPayloadType,
80 kDecoderNotFound,
81 kInvalidPointer,
82 kAccelerateError,
83 kPreemptiveExpandError,
84 kComfortNoiseErrorCode,
85 kDecoderErrorCode,
86 kOtherDecoderError,
87 kInvalidOperation,
88 kDtmfParsingError,
89 kDtmfInsertError,
90 kSampleUnderrun,
91 kDecodedTooMuch,
92 kRedundancySplitError,
93 kPacketBufferCorruption
94 };
95
henrik.lundin1d9061e2016-04-26 12:19:34 -070096 struct Dependencies {
97 // The constructor populates the Dependencies struct with the default
98 // implementations of the objects. They can all be replaced by the user
99 // before sending the struct to the NetEqImpl constructor. However, there
100 // are dependencies between some of the classes inside the struct, so
101 // swapping out one may make it necessary to re-create another one.
Alessio Bazzica8f319a32019-07-24 16:47:02 +0000102 Dependencies(
ossue3525782016-05-25 07:37:43 -0700103 const NetEq::Config& config,
Alessio Bazzica8f319a32019-07-24 16:47:02 +0000104 Clock* clock,
ossue3525782016-05-25 07:37:43 -0700105 const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory);
henrik.lundin1d9061e2016-04-26 12:19:34 -0700106 ~Dependencies();
107
Alessio Bazzica8f319a32019-07-24 16:47:02 +0000108 Clock* const clock;
henrik.lundin1d9061e2016-04-26 12:19:34 -0700109 std::unique_ptr<TickTimer> tick_timer;
Jakob Ivarsson44507082019-03-05 16:59:03 +0100110 std::unique_ptr<StatisticsCalculator> stats;
henrik.lundin1d9061e2016-04-26 12:19:34 -0700111 std::unique_ptr<BufferLevelFilter> buffer_level_filter;
112 std::unique_ptr<DecoderDatabase> decoder_database;
113 std::unique_ptr<DelayPeakDetector> delay_peak_detector;
114 std::unique_ptr<DelayManager> delay_manager;
115 std::unique_ptr<DtmfBuffer> dtmf_buffer;
116 std::unique_ptr<DtmfToneGenerator> dtmf_tone_generator;
117 std::unique_ptr<PacketBuffer> packet_buffer;
ossua70695a2016-09-22 02:06:28 -0700118 std::unique_ptr<RedPayloadSplitter> red_payload_splitter;
henrik.lundin1d9061e2016-04-26 12:19:34 -0700119 std::unique_ptr<TimestampScaler> timestamp_scaler;
120 std::unique_ptr<AccelerateFactory> accelerate_factory;
121 std::unique_ptr<ExpandFactory> expand_factory;
122 std::unique_ptr<PreemptiveExpandFactory> preemptive_expand_factory;
123 };
124
125 // Creates a new NetEqImpl object.
henrik.lundin@webrtc.orgea257842014-08-07 12:27:37 +0000126 NetEqImpl(const NetEq::Config& config,
henrik.lundin1d9061e2016-04-26 12:19:34 -0700127 Dependencies&& deps,
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000128 bool create_components = true);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000129
Karl Wiberg7f6c4d42015-04-09 15:44:22 +0200130 ~NetEqImpl() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000131
132 // Inserts a new packet into NetEq. The |receive_timestamp| is an indication
133 // of the time when the packet was received, and should be measured with
134 // the same tick rate as the RTP timestamp of the current payload.
135 // Returns 0 on success, -1 on failure.
Henrik Lundin70c09bd2017-04-24 15:56:56 +0200136 int InsertPacket(const RTPHeader& rtp_header,
kwibergee2bac22015-11-11 10:34:00 -0800137 rtc::ArrayView<const uint8_t> payload,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000138 uint32_t receive_timestamp) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000139
henrik.lundinb8c55b12017-05-10 07:38:01 -0700140 void InsertEmptyPacket(const RTPHeader& rtp_header) override;
141
Ivo Creusen55de08e2018-09-03 11:49:27 +0200142 int GetAudio(
143 AudioFrame* audio_frame,
144 bool* muted,
145 absl::optional<Operations> action_override = absl::nullopt) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000146
kwiberg1c07c702017-03-27 07:15:49 -0700147 void SetCodecs(const std::map<int, SdpAudioFormat>& codecs) override;
148
kwiberg5adaf732016-10-04 09:33:27 -0700149 bool RegisterPayloadType(int rtp_payload_type,
150 const SdpAudioFormat& audio_format) override;
151
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000152 // Removes |rtp_payload_type| from the codec database. Returns 0 on success,
153 // -1 on failure.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000154 int RemovePayloadType(uint8_t rtp_payload_type) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000155
kwiberg6b19b562016-09-20 04:02:25 -0700156 void RemoveAllPayloadTypes() override;
157
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000158 bool SetMinimumDelay(int delay_ms) override;
turaj@webrtc.orgf1efc572013-08-16 23:44:24 +0000159
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000160 bool SetMaximumDelay(int delay_ms) override;
turaj@webrtc.orgf1efc572013-08-16 23:44:24 +0000161
Ruslan Burakov9bee67c2019-02-05 13:49:26 +0100162 bool SetBaseMinimumDelayMs(int delay_ms) override;
163
164 int GetBaseMinimumDelayMs() const override;
165
Henrik Lundinabbff892017-11-29 09:14:04 +0100166 int TargetDelayMs() const override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000167
henrik.lundinb3f1c5d2016-08-22 15:39:53 -0700168 int FilteredCurrentDelayMs() const override;
169
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000170 // Writes the current network statistics to |stats|. The statistics are reset
171 // after the call.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000172 int NetworkStatistics(NetEqNetworkStatistics* stats) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000173
Steve Anton2dbc69f2017-08-24 17:15:13 -0700174 NetEqLifetimeStatistics GetLifetimeStatistics() const override;
175
Ivo Creusend1c2f782018-09-13 14:39:55 +0200176 NetEqOperationsAndState GetOperationsAndState() const override;
177
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000178 // Enables post-decode VAD. When enabled, GetAudio() will return
179 // kOutputVADPassive when the signal contains no speech.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000180 void EnableVad() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000181
182 // Disables post-decode VAD.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000183 void DisableVad() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000184
Danil Chapovalovb6021232018-06-19 13:26:36 +0200185 absl::optional<uint32_t> GetPlayoutTimestamp() const override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000186
henrik.lundind89814b2015-11-23 06:49:25 -0800187 int last_output_sample_rate_hz() const override;
188
Danil Chapovalovb6021232018-06-19 13:26:36 +0200189 absl::optional<SdpAudioFormat> GetDecoderFormat(
ossuf1b08da2016-09-23 02:19:43 -0700190 int payload_type) const override;
kwibergc4ccd4d2016-09-21 10:55:15 -0700191
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000192 // Flushes both the packet buffer and the sync buffer.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000193 void FlushBuffers() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000194
henrik.lundin48ed9302015-10-29 05:36:24 -0700195 void EnableNack(size_t max_nack_list_size) override;
196
197 void DisableNack() override;
198
199 std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const override;
turaj@webrtc.orgff43c852013-09-25 00:07:27 +0000200
henrik.lundin114c1b32017-04-26 07:47:32 -0700201 std::vector<uint32_t> LastDecodedTimestamps() const override;
202
203 int SyncBufferSizeMs() const override;
204
henrik.lundin@webrtc.orgb287d962014-04-07 21:21:45 +0000205 // This accessor method is only intended for testing purposes.
henrike@webrtc.org47658f12014-09-10 22:14:59 +0000206 const SyncBuffer* sync_buffer_for_test() const;
minyue5bd33972016-05-02 04:46:11 -0700207 Operations last_operation_for_test() const;
henrik.lundin@webrtc.orgb287d962014-04-07 21:21:45 +0000208
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000209 protected:
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000210 static const int kOutputSizeMs = 10;
minyue5bd33972016-05-02 04:46:11 -0700211 static const size_t kMaxFrameSize = 5760; // 120 ms @ 48 kHz.
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000212 // TODO(hlundin): Provide a better value for kSyncBufferSize.
minyue17461792016-05-03 13:32:05 -0700213 // Current value is kMaxFrameSize + 60 ms * 48 kHz, which is enough for
214 // calculating correlations of current frame against history.
215 static const size_t kSyncBufferSize = kMaxFrameSize + 60 * 48;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000216
217 // Inserts a new packet into NetEq. This is used by the InsertPacket method
218 // above. Returns 0 on success, otherwise an error code.
219 // TODO(hlundin): Merge this with InsertPacket above?
Henrik Lundin70c09bd2017-04-24 15:56:56 +0200220 int InsertPacketInternal(const RTPHeader& rtp_header,
kwibergee2bac22015-11-11 10:34:00 -0800221 rtc::ArrayView<const uint8_t> payload,
ossu17e3fa12016-09-08 04:52:55 -0700222 uint32_t receive_timestamp)
danilchap56359be2017-09-07 07:53:45 -0700223 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000224
henrik.lundin6d8e0112016-03-04 10:34:21 -0800225 // Delivers 10 ms of audio data. The data is written to |audio_frame|.
henrik.lundin@webrtc.orge1d468c2013-01-30 07:37:20 +0000226 // Returns 0 on success, otherwise an error code.
Ivo Creusen55de08e2018-09-03 11:49:27 +0200227 int GetAudioInternal(AudioFrame* audio_frame,
228 bool* muted,
229 absl::optional<Operations> action_override)
danilchap56359be2017-09-07 07:53:45 -0700230 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000231
232 // Provides a decision to the GetAudioInternal method. The decision what to
233 // do is written to |operation|. Packets to decode are written to
234 // |packet_list|, and a DTMF event to play is written to |dtmf_event|. When
235 // DTMF should be played, |play_dtmf| is set to true by the method.
236 // Returns 0 on success, otherwise an error code.
237 int GetDecision(Operations* operation,
238 PacketList* packet_list,
239 DtmfEvent* dtmf_event,
Ivo Creusen55de08e2018-09-03 11:49:27 +0200240 bool* play_dtmf,
241 absl::optional<Operations> action_override)
242 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000243
244 // Decodes the speech packets in |packet_list|, and writes the results to
245 // |decoded_buffer|, which is allocated to hold |decoded_buffer_length|
246 // elements. The length of the decoded data is written to |decoded_length|.
247 // The speech type -- speech or (codec-internal) comfort noise -- is written
248 // to |speech_type|. If |packet_list| contains any SID frames for RFC 3389
249 // comfort noise, those are not decoded.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000250 int Decode(PacketList* packet_list,
251 Operations* operation,
252 int* decoded_length,
253 AudioDecoder::SpeechType* speech_type)
danilchap56359be2017-09-07 07:53:45 -0700254 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000255
minyuel6d92bf52015-09-23 15:20:39 +0200256 // Sub-method to Decode(). Performs codec internal CNG.
danilchap56359be2017-09-07 07:53:45 -0700257 int DecodeCng(AudioDecoder* decoder,
258 int* decoded_length,
minyuel6d92bf52015-09-23 15:20:39 +0200259 AudioDecoder::SpeechType* speech_type)
danilchap56359be2017-09-07 07:53:45 -0700260 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
minyuel6d92bf52015-09-23 15:20:39 +0200261
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000262 // Sub-method to Decode(). Performs the actual decoding.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000263 int DecodeLoop(PacketList* packet_list,
minyuel6d92bf52015-09-23 15:20:39 +0200264 const Operations& operation,
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000265 AudioDecoder* decoder,
266 int* decoded_length,
267 AudioDecoder::SpeechType* speech_type)
danilchap56359be2017-09-07 07:53:45 -0700268 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000269
270 // Sub-method which calls the Normal class to perform the normal operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000271 void DoNormal(const int16_t* decoded_buffer,
272 size_t decoded_length,
273 AudioDecoder::SpeechType speech_type,
danilchap56359be2017-09-07 07:53:45 -0700274 bool play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000275
276 // Sub-method which calls the Merge class to perform the merge operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000277 void DoMerge(int16_t* decoded_buffer,
278 size_t decoded_length,
279 AudioDecoder::SpeechType speech_type,
danilchap56359be2017-09-07 07:53:45 -0700280 bool play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000281
Henrik Lundin00eb12a2018-09-05 18:14:52 +0200282 bool DoCodecPlc() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
283
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000284 // Sub-method which calls the Expand class to perform the expand operation.
danilchap56359be2017-09-07 07:53:45 -0700285 int DoExpand(bool play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000286
287 // Sub-method which calls the Accelerate class to perform the accelerate
288 // operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000289 int DoAccelerate(int16_t* decoded_buffer,
290 size_t decoded_length,
291 AudioDecoder::SpeechType speech_type,
Henrik Lundincf808d22015-05-27 14:33:29 +0200292 bool play_dtmf,
danilchap56359be2017-09-07 07:53:45 -0700293 bool fast_accelerate)
294 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000295
296 // Sub-method which calls the PreemptiveExpand class to perform the
297 // preemtive expand operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000298 int DoPreemptiveExpand(int16_t* decoded_buffer,
299 size_t decoded_length,
300 AudioDecoder::SpeechType speech_type,
danilchap56359be2017-09-07 07:53:45 -0700301 bool play_dtmf)
302 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000303
304 // Sub-method which calls the ComfortNoise class to generate RFC 3389 comfort
305 // noise. |packet_list| can either contain one SID frame to update the
306 // noise parameters, or no payload at all, in which case the previously
307 // received parameters are used.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000308 int DoRfc3389Cng(PacketList* packet_list, bool play_dtmf)
danilchap56359be2017-09-07 07:53:45 -0700309 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000310
311 // Calls the audio decoder to generate codec-internal comfort noise when
312 // no packet was received.
minyuel6d92bf52015-09-23 15:20:39 +0200313 void DoCodecInternalCng(const int16_t* decoded_buffer, size_t decoded_length)
danilchap56359be2017-09-07 07:53:45 -0700314 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000315
316 // Calls the DtmfToneGenerator class to generate DTMF tones.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000317 int DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf)
danilchap56359be2017-09-07 07:53:45 -0700318 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000319
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000320 // Overdub DTMF on top of |output|.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000321 int DtmfOverdub(const DtmfEvent& dtmf_event,
322 size_t num_channels,
danilchap56359be2017-09-07 07:53:45 -0700323 int16_t* output) const
324 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000325
326 // Extracts packets from |packet_buffer_| to produce at least
327 // |required_samples| samples. The packets are inserted into |packet_list|.
328 // Returns the number of samples that the packets in the list will produce, or
329 // -1 in case of an error.
Peter Kastingdce40cf2015-08-24 14:52:23 -0700330 int ExtractPackets(size_t required_samples, PacketList* packet_list)
danilchap56359be2017-09-07 07:53:45 -0700331 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000332
333 // Resets various variables and objects to new values based on the sample rate
334 // |fs_hz| and |channels| number audio channels.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000335 void SetSampleRateAndChannels(int fs_hz, size_t channels)
danilchap56359be2017-09-07 07:53:45 -0700336 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000337
338 // Returns the output type for the audio produced by the latest call to
339 // GetAudio().
danilchap56359be2017-09-07 07:53:45 -0700340 OutputType LastOutputType() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000341
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000342 // Updates Expand and Merge.
343 virtual void UpdatePlcComponents(int fs_hz, size_t channels)
danilchap56359be2017-09-07 07:53:45 -0700344 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000345
henrik.lundin@webrtc.org7cbc4f92014-10-07 06:37:39 +0000346 // Creates DecisionLogic object with the mode given by |playout_mode_|.
danilchap56359be2017-09-07 07:53:45 -0700347 virtual void CreateDecisionLogic() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000348
Alessio Bazzica8f319a32019-07-24 16:47:02 +0000349 Clock* const clock_;
350
pbos5ad935c2016-01-25 03:52:44 -0800351 rtc::CriticalSection crit_sect_;
danilchap56359be2017-09-07 07:53:45 -0700352 const std::unique_ptr<TickTimer> tick_timer_ RTC_GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800353 const std::unique_ptr<BufferLevelFilter> buffer_level_filter_
danilchap56359be2017-09-07 07:53:45 -0700354 RTC_GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800355 const std::unique_ptr<DecoderDatabase> decoder_database_
danilchap56359be2017-09-07 07:53:45 -0700356 RTC_GUARDED_BY(crit_sect_);
357 const std::unique_ptr<DelayManager> delay_manager_ RTC_GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800358 const std::unique_ptr<DelayPeakDetector> delay_peak_detector_
danilchap56359be2017-09-07 07:53:45 -0700359 RTC_GUARDED_BY(crit_sect_);
360 const std::unique_ptr<DtmfBuffer> dtmf_buffer_ RTC_GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800361 const std::unique_ptr<DtmfToneGenerator> dtmf_tone_generator_
danilchap56359be2017-09-07 07:53:45 -0700362 RTC_GUARDED_BY(crit_sect_);
363 const std::unique_ptr<PacketBuffer> packet_buffer_ RTC_GUARDED_BY(crit_sect_);
ossua70695a2016-09-22 02:06:28 -0700364 const std::unique_ptr<RedPayloadSplitter> red_payload_splitter_
danilchap56359be2017-09-07 07:53:45 -0700365 RTC_GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800366 const std::unique_ptr<TimestampScaler> timestamp_scaler_
danilchap56359be2017-09-07 07:53:45 -0700367 RTC_GUARDED_BY(crit_sect_);
368 const std::unique_ptr<PostDecodeVad> vad_ RTC_GUARDED_BY(crit_sect_);
369 const std::unique_ptr<ExpandFactory> expand_factory_
370 RTC_GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800371 const std::unique_ptr<AccelerateFactory> accelerate_factory_
danilchap56359be2017-09-07 07:53:45 -0700372 RTC_GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800373 const std::unique_ptr<PreemptiveExpandFactory> preemptive_expand_factory_
danilchap56359be2017-09-07 07:53:45 -0700374 RTC_GUARDED_BY(crit_sect_);
Jakob Ivarsson44507082019-03-05 16:59:03 +0100375 const std::unique_ptr<StatisticsCalculator> stats_ RTC_GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000376
danilchap56359be2017-09-07 07:53:45 -0700377 std::unique_ptr<BackgroundNoise> background_noise_ RTC_GUARDED_BY(crit_sect_);
378 std::unique_ptr<DecisionLogic> decision_logic_ RTC_GUARDED_BY(crit_sect_);
379 std::unique_ptr<AudioMultiVector> algorithm_buffer_
380 RTC_GUARDED_BY(crit_sect_);
381 std::unique_ptr<SyncBuffer> sync_buffer_ RTC_GUARDED_BY(crit_sect_);
382 std::unique_ptr<Expand> expand_ RTC_GUARDED_BY(crit_sect_);
383 std::unique_ptr<Normal> normal_ RTC_GUARDED_BY(crit_sect_);
384 std::unique_ptr<Merge> merge_ RTC_GUARDED_BY(crit_sect_);
385 std::unique_ptr<Accelerate> accelerate_ RTC_GUARDED_BY(crit_sect_);
386 std::unique_ptr<PreemptiveExpand> preemptive_expand_
387 RTC_GUARDED_BY(crit_sect_);
388 RandomVector random_vector_ RTC_GUARDED_BY(crit_sect_);
389 std::unique_ptr<ComfortNoise> comfort_noise_ RTC_GUARDED_BY(crit_sect_);
danilchap56359be2017-09-07 07:53:45 -0700390 int fs_hz_ RTC_GUARDED_BY(crit_sect_);
391 int fs_mult_ RTC_GUARDED_BY(crit_sect_);
392 int last_output_sample_rate_hz_ RTC_GUARDED_BY(crit_sect_);
393 size_t output_size_samples_ RTC_GUARDED_BY(crit_sect_);
394 size_t decoder_frame_length_ RTC_GUARDED_BY(crit_sect_);
395 Modes last_mode_ RTC_GUARDED_BY(crit_sect_);
396 Operations last_operation_ RTC_GUARDED_BY(crit_sect_);
danilchap56359be2017-09-07 07:53:45 -0700397 size_t decoded_buffer_length_ RTC_GUARDED_BY(crit_sect_);
398 std::unique_ptr<int16_t[]> decoded_buffer_ RTC_GUARDED_BY(crit_sect_);
399 uint32_t playout_timestamp_ RTC_GUARDED_BY(crit_sect_);
400 bool new_codec_ RTC_GUARDED_BY(crit_sect_);
401 uint32_t timestamp_ RTC_GUARDED_BY(crit_sect_);
402 bool reset_decoder_ RTC_GUARDED_BY(crit_sect_);
Danil Chapovalovb6021232018-06-19 13:26:36 +0200403 absl::optional<uint8_t> current_rtp_payload_type_ RTC_GUARDED_BY(crit_sect_);
404 absl::optional<uint8_t> current_cng_rtp_payload_type_
danilchap56359be2017-09-07 07:53:45 -0700405 RTC_GUARDED_BY(crit_sect_);
danilchap56359be2017-09-07 07:53:45 -0700406 bool first_packet_ RTC_GUARDED_BY(crit_sect_);
danilchap56359be2017-09-07 07:53:45 -0700407 bool enable_fast_accelerate_ RTC_GUARDED_BY(crit_sect_);
408 std::unique_ptr<NackTracker> nack_ RTC_GUARDED_BY(crit_sect_);
409 bool nack_enabled_ RTC_GUARDED_BY(crit_sect_);
410 const bool enable_muted_state_ RTC_GUARDED_BY(crit_sect_);
411 AudioFrame::VADActivity last_vad_activity_ RTC_GUARDED_BY(crit_sect_) =
henrik.lundin500c04b2016-03-08 02:36:04 -0800412 AudioFrame::kVadPassive;
henrik.lundinb1fb72b2016-05-03 08:18:47 -0700413 std::unique_ptr<TickTimer::Stopwatch> generated_noise_stopwatch_
danilchap56359be2017-09-07 07:53:45 -0700414 RTC_GUARDED_BY(crit_sect_);
415 std::vector<uint32_t> last_decoded_timestamps_ RTC_GUARDED_BY(crit_sect_);
Alessio Bazzica8f319a32019-07-24 16:47:02 +0000416 std::vector<RtpPacketInfo> last_decoded_packet_infos_
417 RTC_GUARDED_BY(crit_sect_);
Henrik Lundin3ef3bfc2018-04-10 15:10:26 +0200418 ExpandUmaLogger expand_uma_logger_ RTC_GUARDED_BY(crit_sect_);
419 ExpandUmaLogger speech_expand_uma_logger_ RTC_GUARDED_BY(crit_sect_);
Henrik Lundin7687ad52018-07-02 10:14:46 +0200420 bool no_time_stretching_ RTC_GUARDED_BY(crit_sect_); // Only used for test.
Henrik Lundin00eb12a2018-09-05 18:14:52 +0200421 rtc::BufferT<int16_t> concealment_audio_ RTC_GUARDED_BY(crit_sect_);
Jakob Ivarsson39b934b2019-01-10 10:28:23 +0100422 const bool enable_rtx_handling_ RTC_GUARDED_BY(crit_sect_);
minyue@webrtc.orgd7301772013-08-29 00:58:14 +0000423
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000424 private:
henrikg3c089d72015-09-16 05:37:44 -0700425 RTC_DISALLOW_COPY_AND_ASSIGN(NetEqImpl);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000426};
427
428} // namespace webrtc
Mirko Bonadei92ea95e2017-09-15 06:47:31 +0200429#endif // MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_