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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
kwiberg2d0c3322016-02-14 09:28:33 -080014#include <memory>
henrik.lundin4cf61dd2015-12-09 06:20:58 -080015#include <string>
16
Danil Chapovalovb6021232018-06-19 13:26:36 +020017#include "absl/types/optional.h"
Fredrik Solenbergbbf21a32018-04-12 22:44:09 +020018#include "api/audio/audio_frame.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020019#include "modules/audio_coding/neteq/audio_multi_vector.h"
Yves Gerey988cc082018-10-23 12:03:01 +020020#include "modules/audio_coding/neteq/defines.h" // Modes, Operations
Henrik Lundin3ef3bfc2018-04-10 15:10:26 +020021#include "modules/audio_coding/neteq/expand_uma_logger.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020022#include "modules/audio_coding/neteq/include/neteq.h"
Yves Gerey988cc082018-10-23 12:03:01 +020023#include "modules/audio_coding/neteq/packet.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020024#include "modules/audio_coding/neteq/random_vector.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020025#include "modules/audio_coding/neteq/statistics_calculator.h"
26#include "modules/audio_coding/neteq/tick_timer.h"
Steve Anton10542f22019-01-11 09:11:00 -080027#include "rtc_base/constructor_magic.h"
28#include "rtc_base/critical_section.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020029#include "rtc_base/thread_annotations.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000030
31namespace webrtc {
32
33// Forward declarations.
henrik.lundin@webrtc.org40d3fc62013-09-18 12:19:50 +000034class Accelerate;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000035class BackgroundNoise;
36class BufferLevelFilter;
37class ComfortNoise;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000038class DecisionLogic;
39class DecoderDatabase;
40class DelayManager;
41class DelayPeakDetector;
42class DtmfBuffer;
43class DtmfToneGenerator;
44class Expand;
henrik.lundin@webrtc.org40d3fc62013-09-18 12:19:50 +000045class Merge;
henrik.lundin91951862016-06-08 06:43:41 -070046class NackTracker;
henrik.lundin@webrtc.org40d3fc62013-09-18 12:19:50 +000047class Normal;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000048class PacketBuffer;
ossua70695a2016-09-22 02:06:28 -070049class RedPayloadSplitter;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000050class PostDecodeVad;
henrik.lundin@webrtc.org40d3fc62013-09-18 12:19:50 +000051class PreemptiveExpand;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000052class RandomVector;
53class SyncBuffer;
54class TimestampScaler;
henrik.lundin@webrtc.orgd9faa462014-01-14 10:18:45 +000055struct AccelerateFactory;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000056struct DtmfEvent;
henrik.lundin@webrtc.orgd9faa462014-01-14 10:18:45 +000057struct ExpandFactory;
58struct PreemptiveExpandFactory;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000059
60class NetEqImpl : public webrtc::NetEq {
61 public:
Jonas Olssona4d87372019-07-05 19:08:33 +020062 enum class OutputType { kNormalSpeech, kPLC, kCNG, kPLCCNG, kVadPassive };
henrik.lundin55480f52016-03-08 02:37:57 -080063
Henrik Lundinc417d9e2017-06-14 12:29:03 +020064 enum ErrorCodes {
65 kNoError = 0,
66 kOtherError,
67 kUnknownRtpPayloadType,
68 kDecoderNotFound,
69 kInvalidPointer,
70 kAccelerateError,
71 kPreemptiveExpandError,
72 kComfortNoiseErrorCode,
73 kDecoderErrorCode,
74 kOtherDecoderError,
75 kInvalidOperation,
76 kDtmfParsingError,
77 kDtmfInsertError,
78 kSampleUnderrun,
79 kDecodedTooMuch,
80 kRedundancySplitError,
81 kPacketBufferCorruption
82 };
83
henrik.lundin1d9061e2016-04-26 12:19:34 -070084 struct Dependencies {
85 // The constructor populates the Dependencies struct with the default
86 // implementations of the objects. They can all be replaced by the user
87 // before sending the struct to the NetEqImpl constructor. However, there
88 // are dependencies between some of the classes inside the struct, so
89 // swapping out one may make it necessary to re-create another one.
Alessio Bazzicafab34602019-07-24 16:41:00 +000090 explicit Dependencies(
ossue3525782016-05-25 07:37:43 -070091 const NetEq::Config& config,
92 const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory);
henrik.lundin1d9061e2016-04-26 12:19:34 -070093 ~Dependencies();
94
95 std::unique_ptr<TickTimer> tick_timer;
Jakob Ivarsson44507082019-03-05 16:59:03 +010096 std::unique_ptr<StatisticsCalculator> stats;
henrik.lundin1d9061e2016-04-26 12:19:34 -070097 std::unique_ptr<BufferLevelFilter> buffer_level_filter;
98 std::unique_ptr<DecoderDatabase> decoder_database;
99 std::unique_ptr<DelayPeakDetector> delay_peak_detector;
100 std::unique_ptr<DelayManager> delay_manager;
101 std::unique_ptr<DtmfBuffer> dtmf_buffer;
102 std::unique_ptr<DtmfToneGenerator> dtmf_tone_generator;
103 std::unique_ptr<PacketBuffer> packet_buffer;
ossua70695a2016-09-22 02:06:28 -0700104 std::unique_ptr<RedPayloadSplitter> red_payload_splitter;
henrik.lundin1d9061e2016-04-26 12:19:34 -0700105 std::unique_ptr<TimestampScaler> timestamp_scaler;
106 std::unique_ptr<AccelerateFactory> accelerate_factory;
107 std::unique_ptr<ExpandFactory> expand_factory;
108 std::unique_ptr<PreemptiveExpandFactory> preemptive_expand_factory;
109 };
110
111 // Creates a new NetEqImpl object.
henrik.lundin@webrtc.orgea257842014-08-07 12:27:37 +0000112 NetEqImpl(const NetEq::Config& config,
henrik.lundin1d9061e2016-04-26 12:19:34 -0700113 Dependencies&& deps,
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000114 bool create_components = true);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000115
Karl Wiberg7f6c4d42015-04-09 15:44:22 +0200116 ~NetEqImpl() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000117
118 // Inserts a new packet into NetEq. The |receive_timestamp| is an indication
119 // of the time when the packet was received, and should be measured with
120 // the same tick rate as the RTP timestamp of the current payload.
121 // Returns 0 on success, -1 on failure.
Henrik Lundin70c09bd2017-04-24 15:56:56 +0200122 int InsertPacket(const RTPHeader& rtp_header,
kwibergee2bac22015-11-11 10:34:00 -0800123 rtc::ArrayView<const uint8_t> payload,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000124 uint32_t receive_timestamp) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000125
henrik.lundinb8c55b12017-05-10 07:38:01 -0700126 void InsertEmptyPacket(const RTPHeader& rtp_header) override;
127
Ivo Creusen55de08e2018-09-03 11:49:27 +0200128 int GetAudio(
129 AudioFrame* audio_frame,
130 bool* muted,
131 absl::optional<Operations> action_override = absl::nullopt) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000132
kwiberg1c07c702017-03-27 07:15:49 -0700133 void SetCodecs(const std::map<int, SdpAudioFormat>& codecs) override;
134
kwiberg5adaf732016-10-04 09:33:27 -0700135 bool RegisterPayloadType(int rtp_payload_type,
136 const SdpAudioFormat& audio_format) override;
137
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000138 // Removes |rtp_payload_type| from the codec database. Returns 0 on success,
139 // -1 on failure.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000140 int RemovePayloadType(uint8_t rtp_payload_type) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000141
kwiberg6b19b562016-09-20 04:02:25 -0700142 void RemoveAllPayloadTypes() override;
143
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000144 bool SetMinimumDelay(int delay_ms) override;
turaj@webrtc.orgf1efc572013-08-16 23:44:24 +0000145
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000146 bool SetMaximumDelay(int delay_ms) override;
turaj@webrtc.orgf1efc572013-08-16 23:44:24 +0000147
Ruslan Burakov9bee67c2019-02-05 13:49:26 +0100148 bool SetBaseMinimumDelayMs(int delay_ms) override;
149
150 int GetBaseMinimumDelayMs() const override;
151
Henrik Lundinabbff892017-11-29 09:14:04 +0100152 int TargetDelayMs() const override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000153
henrik.lundinb3f1c5d2016-08-22 15:39:53 -0700154 int FilteredCurrentDelayMs() const override;
155
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000156 // Writes the current network statistics to |stats|. The statistics are reset
157 // after the call.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000158 int NetworkStatistics(NetEqNetworkStatistics* stats) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000159
Steve Anton2dbc69f2017-08-24 17:15:13 -0700160 NetEqLifetimeStatistics GetLifetimeStatistics() const override;
161
Ivo Creusend1c2f782018-09-13 14:39:55 +0200162 NetEqOperationsAndState GetOperationsAndState() const override;
163
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000164 // Enables post-decode VAD. When enabled, GetAudio() will return
165 // kOutputVADPassive when the signal contains no speech.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000166 void EnableVad() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000167
168 // Disables post-decode VAD.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000169 void DisableVad() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000170
Danil Chapovalovb6021232018-06-19 13:26:36 +0200171 absl::optional<uint32_t> GetPlayoutTimestamp() const override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000172
henrik.lundind89814b2015-11-23 06:49:25 -0800173 int last_output_sample_rate_hz() const override;
174
Danil Chapovalovb6021232018-06-19 13:26:36 +0200175 absl::optional<SdpAudioFormat> GetDecoderFormat(
ossuf1b08da2016-09-23 02:19:43 -0700176 int payload_type) const override;
kwibergc4ccd4d2016-09-21 10:55:15 -0700177
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000178 // Flushes both the packet buffer and the sync buffer.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000179 void FlushBuffers() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000180
henrik.lundin48ed9302015-10-29 05:36:24 -0700181 void EnableNack(size_t max_nack_list_size) override;
182
183 void DisableNack() override;
184
185 std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const override;
turaj@webrtc.orgff43c852013-09-25 00:07:27 +0000186
henrik.lundin114c1b32017-04-26 07:47:32 -0700187 std::vector<uint32_t> LastDecodedTimestamps() const override;
188
189 int SyncBufferSizeMs() const override;
190
henrik.lundin@webrtc.orgb287d962014-04-07 21:21:45 +0000191 // This accessor method is only intended for testing purposes.
henrike@webrtc.org47658f12014-09-10 22:14:59 +0000192 const SyncBuffer* sync_buffer_for_test() const;
minyue5bd33972016-05-02 04:46:11 -0700193 Operations last_operation_for_test() const;
henrik.lundin@webrtc.orgb287d962014-04-07 21:21:45 +0000194
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000195 protected:
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000196 static const int kOutputSizeMs = 10;
minyue5bd33972016-05-02 04:46:11 -0700197 static const size_t kMaxFrameSize = 5760; // 120 ms @ 48 kHz.
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000198 // TODO(hlundin): Provide a better value for kSyncBufferSize.
minyue17461792016-05-03 13:32:05 -0700199 // Current value is kMaxFrameSize + 60 ms * 48 kHz, which is enough for
200 // calculating correlations of current frame against history.
201 static const size_t kSyncBufferSize = kMaxFrameSize + 60 * 48;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000202
203 // Inserts a new packet into NetEq. This is used by the InsertPacket method
204 // above. Returns 0 on success, otherwise an error code.
205 // TODO(hlundin): Merge this with InsertPacket above?
Henrik Lundin70c09bd2017-04-24 15:56:56 +0200206 int InsertPacketInternal(const RTPHeader& rtp_header,
kwibergee2bac22015-11-11 10:34:00 -0800207 rtc::ArrayView<const uint8_t> payload,
ossu17e3fa12016-09-08 04:52:55 -0700208 uint32_t receive_timestamp)
danilchap56359be2017-09-07 07:53:45 -0700209 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000210
henrik.lundin6d8e0112016-03-04 10:34:21 -0800211 // Delivers 10 ms of audio data. The data is written to |audio_frame|.
henrik.lundin@webrtc.orge1d468c2013-01-30 07:37:20 +0000212 // Returns 0 on success, otherwise an error code.
Ivo Creusen55de08e2018-09-03 11:49:27 +0200213 int GetAudioInternal(AudioFrame* audio_frame,
214 bool* muted,
215 absl::optional<Operations> action_override)
danilchap56359be2017-09-07 07:53:45 -0700216 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000217
218 // Provides a decision to the GetAudioInternal method. The decision what to
219 // do is written to |operation|. Packets to decode are written to
220 // |packet_list|, and a DTMF event to play is written to |dtmf_event|. When
221 // DTMF should be played, |play_dtmf| is set to true by the method.
222 // Returns 0 on success, otherwise an error code.
223 int GetDecision(Operations* operation,
224 PacketList* packet_list,
225 DtmfEvent* dtmf_event,
Ivo Creusen55de08e2018-09-03 11:49:27 +0200226 bool* play_dtmf,
227 absl::optional<Operations> action_override)
228 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000229
230 // Decodes the speech packets in |packet_list|, and writes the results to
231 // |decoded_buffer|, which is allocated to hold |decoded_buffer_length|
232 // elements. The length of the decoded data is written to |decoded_length|.
233 // The speech type -- speech or (codec-internal) comfort noise -- is written
234 // to |speech_type|. If |packet_list| contains any SID frames for RFC 3389
235 // comfort noise, those are not decoded.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000236 int Decode(PacketList* packet_list,
237 Operations* operation,
238 int* decoded_length,
239 AudioDecoder::SpeechType* speech_type)
danilchap56359be2017-09-07 07:53:45 -0700240 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000241
minyuel6d92bf52015-09-23 15:20:39 +0200242 // Sub-method to Decode(). Performs codec internal CNG.
danilchap56359be2017-09-07 07:53:45 -0700243 int DecodeCng(AudioDecoder* decoder,
244 int* decoded_length,
minyuel6d92bf52015-09-23 15:20:39 +0200245 AudioDecoder::SpeechType* speech_type)
danilchap56359be2017-09-07 07:53:45 -0700246 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
minyuel6d92bf52015-09-23 15:20:39 +0200247
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000248 // Sub-method to Decode(). Performs the actual decoding.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000249 int DecodeLoop(PacketList* packet_list,
minyuel6d92bf52015-09-23 15:20:39 +0200250 const Operations& operation,
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000251 AudioDecoder* decoder,
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
256 // Sub-method which calls the Normal class to perform the normal operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000257 void DoNormal(const int16_t* decoded_buffer,
258 size_t decoded_length,
259 AudioDecoder::SpeechType speech_type,
danilchap56359be2017-09-07 07:53:45 -0700260 bool play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000261
262 // Sub-method which calls the Merge class to perform the merge operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000263 void DoMerge(int16_t* decoded_buffer,
264 size_t decoded_length,
265 AudioDecoder::SpeechType speech_type,
danilchap56359be2017-09-07 07:53:45 -0700266 bool play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000267
Henrik Lundin00eb12a2018-09-05 18:14:52 +0200268 bool DoCodecPlc() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
269
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000270 // Sub-method which calls the Expand class to perform the expand operation.
danilchap56359be2017-09-07 07:53:45 -0700271 int DoExpand(bool play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000272
273 // Sub-method which calls the Accelerate class to perform the accelerate
274 // operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000275 int DoAccelerate(int16_t* decoded_buffer,
276 size_t decoded_length,
277 AudioDecoder::SpeechType speech_type,
Henrik Lundincf808d22015-05-27 14:33:29 +0200278 bool play_dtmf,
danilchap56359be2017-09-07 07:53:45 -0700279 bool fast_accelerate)
280 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000281
282 // Sub-method which calls the PreemptiveExpand class to perform the
283 // preemtive expand operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000284 int DoPreemptiveExpand(int16_t* decoded_buffer,
285 size_t decoded_length,
286 AudioDecoder::SpeechType speech_type,
danilchap56359be2017-09-07 07:53:45 -0700287 bool play_dtmf)
288 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000289
290 // Sub-method which calls the ComfortNoise class to generate RFC 3389 comfort
291 // noise. |packet_list| can either contain one SID frame to update the
292 // noise parameters, or no payload at all, in which case the previously
293 // received parameters are used.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000294 int DoRfc3389Cng(PacketList* packet_list, bool play_dtmf)
danilchap56359be2017-09-07 07:53:45 -0700295 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000296
297 // Calls the audio decoder to generate codec-internal comfort noise when
298 // no packet was received.
minyuel6d92bf52015-09-23 15:20:39 +0200299 void DoCodecInternalCng(const int16_t* decoded_buffer, size_t decoded_length)
danilchap56359be2017-09-07 07:53:45 -0700300 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000301
302 // Calls the DtmfToneGenerator class to generate DTMF tones.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000303 int DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf)
danilchap56359be2017-09-07 07:53:45 -0700304 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000305
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000306 // Overdub DTMF on top of |output|.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000307 int DtmfOverdub(const DtmfEvent& dtmf_event,
308 size_t num_channels,
danilchap56359be2017-09-07 07:53:45 -0700309 int16_t* output) const
310 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000311
312 // Extracts packets from |packet_buffer_| to produce at least
313 // |required_samples| samples. The packets are inserted into |packet_list|.
314 // Returns the number of samples that the packets in the list will produce, or
315 // -1 in case of an error.
Peter Kastingdce40cf2015-08-24 14:52:23 -0700316 int ExtractPackets(size_t required_samples, PacketList* packet_list)
danilchap56359be2017-09-07 07:53:45 -0700317 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000318
319 // Resets various variables and objects to new values based on the sample rate
320 // |fs_hz| and |channels| number audio channels.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000321 void SetSampleRateAndChannels(int fs_hz, size_t channels)
danilchap56359be2017-09-07 07:53:45 -0700322 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000323
324 // Returns the output type for the audio produced by the latest call to
325 // GetAudio().
danilchap56359be2017-09-07 07:53:45 -0700326 OutputType LastOutputType() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000327
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000328 // Updates Expand and Merge.
329 virtual void UpdatePlcComponents(int fs_hz, size_t channels)
danilchap56359be2017-09-07 07:53:45 -0700330 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000331
henrik.lundin@webrtc.org7cbc4f92014-10-07 06:37:39 +0000332 // Creates DecisionLogic object with the mode given by |playout_mode_|.
danilchap56359be2017-09-07 07:53:45 -0700333 virtual void CreateDecisionLogic() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000334
pbos5ad935c2016-01-25 03:52:44 -0800335 rtc::CriticalSection crit_sect_;
danilchap56359be2017-09-07 07:53:45 -0700336 const std::unique_ptr<TickTimer> tick_timer_ RTC_GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800337 const std::unique_ptr<BufferLevelFilter> buffer_level_filter_
danilchap56359be2017-09-07 07:53:45 -0700338 RTC_GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800339 const std::unique_ptr<DecoderDatabase> decoder_database_
danilchap56359be2017-09-07 07:53:45 -0700340 RTC_GUARDED_BY(crit_sect_);
341 const std::unique_ptr<DelayManager> delay_manager_ RTC_GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800342 const std::unique_ptr<DelayPeakDetector> delay_peak_detector_
danilchap56359be2017-09-07 07:53:45 -0700343 RTC_GUARDED_BY(crit_sect_);
344 const std::unique_ptr<DtmfBuffer> dtmf_buffer_ RTC_GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800345 const std::unique_ptr<DtmfToneGenerator> dtmf_tone_generator_
danilchap56359be2017-09-07 07:53:45 -0700346 RTC_GUARDED_BY(crit_sect_);
347 const std::unique_ptr<PacketBuffer> packet_buffer_ RTC_GUARDED_BY(crit_sect_);
ossua70695a2016-09-22 02:06:28 -0700348 const std::unique_ptr<RedPayloadSplitter> red_payload_splitter_
danilchap56359be2017-09-07 07:53:45 -0700349 RTC_GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800350 const std::unique_ptr<TimestampScaler> timestamp_scaler_
danilchap56359be2017-09-07 07:53:45 -0700351 RTC_GUARDED_BY(crit_sect_);
352 const std::unique_ptr<PostDecodeVad> vad_ RTC_GUARDED_BY(crit_sect_);
353 const std::unique_ptr<ExpandFactory> expand_factory_
354 RTC_GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800355 const std::unique_ptr<AccelerateFactory> accelerate_factory_
danilchap56359be2017-09-07 07:53:45 -0700356 RTC_GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800357 const std::unique_ptr<PreemptiveExpandFactory> preemptive_expand_factory_
danilchap56359be2017-09-07 07:53:45 -0700358 RTC_GUARDED_BY(crit_sect_);
Jakob Ivarsson44507082019-03-05 16:59:03 +0100359 const std::unique_ptr<StatisticsCalculator> stats_ RTC_GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000360
danilchap56359be2017-09-07 07:53:45 -0700361 std::unique_ptr<BackgroundNoise> background_noise_ RTC_GUARDED_BY(crit_sect_);
362 std::unique_ptr<DecisionLogic> decision_logic_ RTC_GUARDED_BY(crit_sect_);
363 std::unique_ptr<AudioMultiVector> algorithm_buffer_
364 RTC_GUARDED_BY(crit_sect_);
365 std::unique_ptr<SyncBuffer> sync_buffer_ RTC_GUARDED_BY(crit_sect_);
366 std::unique_ptr<Expand> expand_ RTC_GUARDED_BY(crit_sect_);
367 std::unique_ptr<Normal> normal_ RTC_GUARDED_BY(crit_sect_);
368 std::unique_ptr<Merge> merge_ RTC_GUARDED_BY(crit_sect_);
369 std::unique_ptr<Accelerate> accelerate_ RTC_GUARDED_BY(crit_sect_);
370 std::unique_ptr<PreemptiveExpand> preemptive_expand_
371 RTC_GUARDED_BY(crit_sect_);
372 RandomVector random_vector_ RTC_GUARDED_BY(crit_sect_);
373 std::unique_ptr<ComfortNoise> comfort_noise_ RTC_GUARDED_BY(crit_sect_);
danilchap56359be2017-09-07 07:53:45 -0700374 int fs_hz_ RTC_GUARDED_BY(crit_sect_);
375 int fs_mult_ RTC_GUARDED_BY(crit_sect_);
376 int last_output_sample_rate_hz_ RTC_GUARDED_BY(crit_sect_);
377 size_t output_size_samples_ RTC_GUARDED_BY(crit_sect_);
378 size_t decoder_frame_length_ RTC_GUARDED_BY(crit_sect_);
379 Modes last_mode_ RTC_GUARDED_BY(crit_sect_);
380 Operations last_operation_ RTC_GUARDED_BY(crit_sect_);
danilchap56359be2017-09-07 07:53:45 -0700381 size_t decoded_buffer_length_ RTC_GUARDED_BY(crit_sect_);
382 std::unique_ptr<int16_t[]> decoded_buffer_ RTC_GUARDED_BY(crit_sect_);
383 uint32_t playout_timestamp_ RTC_GUARDED_BY(crit_sect_);
384 bool new_codec_ RTC_GUARDED_BY(crit_sect_);
385 uint32_t timestamp_ RTC_GUARDED_BY(crit_sect_);
386 bool reset_decoder_ RTC_GUARDED_BY(crit_sect_);
Danil Chapovalovb6021232018-06-19 13:26:36 +0200387 absl::optional<uint8_t> current_rtp_payload_type_ RTC_GUARDED_BY(crit_sect_);
388 absl::optional<uint8_t> current_cng_rtp_payload_type_
danilchap56359be2017-09-07 07:53:45 -0700389 RTC_GUARDED_BY(crit_sect_);
danilchap56359be2017-09-07 07:53:45 -0700390 bool first_packet_ RTC_GUARDED_BY(crit_sect_);
danilchap56359be2017-09-07 07:53:45 -0700391 bool enable_fast_accelerate_ RTC_GUARDED_BY(crit_sect_);
392 std::unique_ptr<NackTracker> nack_ RTC_GUARDED_BY(crit_sect_);
393 bool nack_enabled_ RTC_GUARDED_BY(crit_sect_);
394 const bool enable_muted_state_ RTC_GUARDED_BY(crit_sect_);
395 AudioFrame::VADActivity last_vad_activity_ RTC_GUARDED_BY(crit_sect_) =
henrik.lundin500c04b2016-03-08 02:36:04 -0800396 AudioFrame::kVadPassive;
henrik.lundinb1fb72b2016-05-03 08:18:47 -0700397 std::unique_ptr<TickTimer::Stopwatch> generated_noise_stopwatch_
danilchap56359be2017-09-07 07:53:45 -0700398 RTC_GUARDED_BY(crit_sect_);
399 std::vector<uint32_t> last_decoded_timestamps_ RTC_GUARDED_BY(crit_sect_);
Henrik Lundin3ef3bfc2018-04-10 15:10:26 +0200400 ExpandUmaLogger expand_uma_logger_ RTC_GUARDED_BY(crit_sect_);
401 ExpandUmaLogger speech_expand_uma_logger_ RTC_GUARDED_BY(crit_sect_);
Henrik Lundin7687ad52018-07-02 10:14:46 +0200402 bool no_time_stretching_ RTC_GUARDED_BY(crit_sect_); // Only used for test.
Henrik Lundin00eb12a2018-09-05 18:14:52 +0200403 rtc::BufferT<int16_t> concealment_audio_ RTC_GUARDED_BY(crit_sect_);
Jakob Ivarsson39b934b2019-01-10 10:28:23 +0100404 const bool enable_rtx_handling_ RTC_GUARDED_BY(crit_sect_);
minyue@webrtc.orgd7301772013-08-29 00:58:14 +0000405
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000406 private:
henrikg3c089d72015-09-16 05:37:44 -0700407 RTC_DISALLOW_COPY_AND_ASSIGN(NetEqImpl);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000408};
409
410} // namespace webrtc
Mirko Bonadei92ea95e2017-09-15 06:47:31 +0200411#endif // MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_