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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"
Ivo Creusen3ce44a32019-10-31 14:38:11 +010022#include "api/neteq/neteq.h"
23#include "api/neteq/neteq_controller.h"
24#include "api/neteq/neteq_controller_factory.h"
25#include "api/neteq/tick_timer.h"
Alessio Bazzica8f319a32019-07-24 16:47:02 +000026#include "api/rtp_packet_info.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020027#include "modules/audio_coding/neteq/audio_multi_vector.h"
Henrik Lundin3ef3bfc2018-04-10 15:10:26 +020028#include "modules/audio_coding/neteq/expand_uma_logger.h"
Yves Gerey988cc082018-10-23 12:03:01 +020029#include "modules/audio_coding/neteq/packet.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020030#include "modules/audio_coding/neteq/random_vector.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020031#include "modules/audio_coding/neteq/statistics_calculator.h"
Steve Anton10542f22019-01-11 09:11:00 -080032#include "rtc_base/constructor_magic.h"
33#include "rtc_base/critical_section.h"
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020034#include "rtc_base/thread_annotations.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000035
36namespace webrtc {
37
38// Forward declarations.
henrik.lundin@webrtc.org40d3fc62013-09-18 12:19:50 +000039class Accelerate;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000040class BackgroundNoise;
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 DecoderDatabase;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000044class DtmfBuffer;
45class DtmfToneGenerator;
46class Expand;
henrik.lundin@webrtc.org40d3fc62013-09-18 12:19:50 +000047class Merge;
henrik.lundin91951862016-06-08 06:43:41 -070048class NackTracker;
henrik.lundin@webrtc.org40d3fc62013-09-18 12:19:50 +000049class Normal;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000050class PacketBuffer;
ossua70695a2016-09-22 02:06:28 -070051class RedPayloadSplitter;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000052class PostDecodeVad;
henrik.lundin@webrtc.org40d3fc62013-09-18 12:19:50 +000053class PreemptiveExpand;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000054class RandomVector;
55class SyncBuffer;
56class TimestampScaler;
henrik.lundin@webrtc.orgd9faa462014-01-14 10:18:45 +000057struct AccelerateFactory;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000058struct DtmfEvent;
henrik.lundin@webrtc.orgd9faa462014-01-14 10:18:45 +000059struct ExpandFactory;
60struct PreemptiveExpandFactory;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000061
62class NetEqImpl : public webrtc::NetEq {
63 public:
Alex Narest5b5d97c2019-08-07 18:15:08 +020064 enum class OutputType {
65 kNormalSpeech,
66 kPLC,
67 kCNG,
68 kPLCCNG,
69 kVadPassive,
70 kCodecPLC
71 };
henrik.lundin55480f52016-03-08 02:37:57 -080072
Henrik Lundinc417d9e2017-06-14 12:29:03 +020073 enum ErrorCodes {
74 kNoError = 0,
75 kOtherError,
76 kUnknownRtpPayloadType,
77 kDecoderNotFound,
78 kInvalidPointer,
79 kAccelerateError,
80 kPreemptiveExpandError,
81 kComfortNoiseErrorCode,
82 kDecoderErrorCode,
83 kOtherDecoderError,
84 kInvalidOperation,
85 kDtmfParsingError,
86 kDtmfInsertError,
87 kSampleUnderrun,
88 kDecodedTooMuch,
89 kRedundancySplitError,
90 kPacketBufferCorruption
91 };
92
henrik.lundin1d9061e2016-04-26 12:19:34 -070093 struct Dependencies {
94 // The constructor populates the Dependencies struct with the default
95 // implementations of the objects. They can all be replaced by the user
96 // before sending the struct to the NetEqImpl constructor. However, there
97 // are dependencies between some of the classes inside the struct, so
98 // swapping out one may make it necessary to re-create another one.
Ivo Creusen3ce44a32019-10-31 14:38:11 +010099 Dependencies(const NetEq::Config& config,
100 Clock* clock,
101 const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory,
102 const NetEqControllerFactory& controller_factory);
henrik.lundin1d9061e2016-04-26 12:19:34 -0700103 ~Dependencies();
104
Alessio Bazzica8f319a32019-07-24 16:47:02 +0000105 Clock* const clock;
henrik.lundin1d9061e2016-04-26 12:19:34 -0700106 std::unique_ptr<TickTimer> tick_timer;
Jakob Ivarsson44507082019-03-05 16:59:03 +0100107 std::unique_ptr<StatisticsCalculator> stats;
henrik.lundin1d9061e2016-04-26 12:19:34 -0700108 std::unique_ptr<DecoderDatabase> decoder_database;
henrik.lundin1d9061e2016-04-26 12:19:34 -0700109 std::unique_ptr<DtmfBuffer> dtmf_buffer;
110 std::unique_ptr<DtmfToneGenerator> dtmf_tone_generator;
111 std::unique_ptr<PacketBuffer> packet_buffer;
Ivo Creusen53a31f72019-10-24 15:20:39 +0200112 std::unique_ptr<NetEqController> neteq_controller;
ossua70695a2016-09-22 02:06:28 -0700113 std::unique_ptr<RedPayloadSplitter> red_payload_splitter;
henrik.lundin1d9061e2016-04-26 12:19:34 -0700114 std::unique_ptr<TimestampScaler> timestamp_scaler;
115 std::unique_ptr<AccelerateFactory> accelerate_factory;
116 std::unique_ptr<ExpandFactory> expand_factory;
117 std::unique_ptr<PreemptiveExpandFactory> preemptive_expand_factory;
118 };
119
120 // Creates a new NetEqImpl object.
henrik.lundin@webrtc.orgea257842014-08-07 12:27:37 +0000121 NetEqImpl(const NetEq::Config& config,
henrik.lundin1d9061e2016-04-26 12:19:34 -0700122 Dependencies&& deps,
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000123 bool create_components = true);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000124
Karl Wiberg7f6c4d42015-04-09 15:44:22 +0200125 ~NetEqImpl() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000126
Karl Wiberg45eb1352019-10-10 14:23:00 +0200127 // Inserts a new packet into NetEq. Returns 0 on success, -1 on failure.
Henrik Lundin70c09bd2017-04-24 15:56:56 +0200128 int InsertPacket(const RTPHeader& rtp_header,
Karl Wiberg45eb1352019-10-10 14:23:00 +0200129 rtc::ArrayView<const uint8_t> payload) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000130
henrik.lundinb8c55b12017-05-10 07:38:01 -0700131 void InsertEmptyPacket(const RTPHeader& rtp_header) override;
132
Ivo Creusen55de08e2018-09-03 11:49:27 +0200133 int GetAudio(
134 AudioFrame* audio_frame,
135 bool* muted,
Ivo Creusen3ce44a32019-10-31 14:38:11 +0100136 absl::optional<Operation> action_override = absl::nullopt) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000137
kwiberg1c07c702017-03-27 07:15:49 -0700138 void SetCodecs(const std::map<int, SdpAudioFormat>& codecs) override;
139
kwiberg5adaf732016-10-04 09:33:27 -0700140 bool RegisterPayloadType(int rtp_payload_type,
141 const SdpAudioFormat& audio_format) override;
142
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000143 // Removes |rtp_payload_type| from the codec database. Returns 0 on success,
144 // -1 on failure.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000145 int RemovePayloadType(uint8_t rtp_payload_type) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000146
kwiberg6b19b562016-09-20 04:02:25 -0700147 void RemoveAllPayloadTypes() override;
148
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000149 bool SetMinimumDelay(int delay_ms) override;
turaj@webrtc.orgf1efc572013-08-16 23:44:24 +0000150
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000151 bool SetMaximumDelay(int delay_ms) override;
turaj@webrtc.orgf1efc572013-08-16 23:44:24 +0000152
Ruslan Burakov9bee67c2019-02-05 13:49:26 +0100153 bool SetBaseMinimumDelayMs(int delay_ms) override;
154
155 int GetBaseMinimumDelayMs() const override;
156
Henrik Lundinabbff892017-11-29 09:14:04 +0100157 int TargetDelayMs() const override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000158
henrik.lundinb3f1c5d2016-08-22 15:39:53 -0700159 int FilteredCurrentDelayMs() const override;
160
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000161 // Writes the current network statistics to |stats|. The statistics are reset
162 // after the call.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000163 int NetworkStatistics(NetEqNetworkStatistics* stats) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000164
Steve Anton2dbc69f2017-08-24 17:15:13 -0700165 NetEqLifetimeStatistics GetLifetimeStatistics() const override;
166
Ivo Creusend1c2f782018-09-13 14:39:55 +0200167 NetEqOperationsAndState GetOperationsAndState() const override;
168
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000169 // Enables post-decode VAD. When enabled, GetAudio() will return
170 // kOutputVADPassive when the signal contains no speech.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000171 void EnableVad() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000172
173 // Disables post-decode VAD.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000174 void DisableVad() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000175
Danil Chapovalovb6021232018-06-19 13:26:36 +0200176 absl::optional<uint32_t> GetPlayoutTimestamp() const override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000177
henrik.lundind89814b2015-11-23 06:49:25 -0800178 int last_output_sample_rate_hz() const override;
179
Karl Wiberg4b644112019-10-11 09:37:42 +0200180 absl::optional<DecoderFormat> GetDecoderFormat(
ossuf1b08da2016-09-23 02:19:43 -0700181 int payload_type) const override;
kwibergc4ccd4d2016-09-21 10:55:15 -0700182
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000183 // Flushes both the packet buffer and the sync buffer.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000184 void FlushBuffers() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000185
henrik.lundin48ed9302015-10-29 05:36:24 -0700186 void EnableNack(size_t max_nack_list_size) override;
187
188 void DisableNack() override;
189
190 std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const override;
turaj@webrtc.orgff43c852013-09-25 00:07:27 +0000191
henrik.lundin114c1b32017-04-26 07:47:32 -0700192 std::vector<uint32_t> LastDecodedTimestamps() const override;
193
194 int SyncBufferSizeMs() const override;
195
henrik.lundin@webrtc.orgb287d962014-04-07 21:21:45 +0000196 // This accessor method is only intended for testing purposes.
henrike@webrtc.org47658f12014-09-10 22:14:59 +0000197 const SyncBuffer* sync_buffer_for_test() const;
Ivo Creusen3ce44a32019-10-31 14:38:11 +0100198 Operation last_operation_for_test() const;
henrik.lundin@webrtc.orgb287d962014-04-07 21:21:45 +0000199
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000200 protected:
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000201 static const int kOutputSizeMs = 10;
minyue5bd33972016-05-02 04:46:11 -0700202 static const size_t kMaxFrameSize = 5760; // 120 ms @ 48 kHz.
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000203 // TODO(hlundin): Provide a better value for kSyncBufferSize.
minyue17461792016-05-03 13:32:05 -0700204 // Current value is kMaxFrameSize + 60 ms * 48 kHz, which is enough for
205 // calculating correlations of current frame against history.
206 static const size_t kSyncBufferSize = kMaxFrameSize + 60 * 48;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000207
208 // Inserts a new packet into NetEq. This is used by the InsertPacket method
209 // above. Returns 0 on success, otherwise an error code.
210 // TODO(hlundin): Merge this with InsertPacket above?
Henrik Lundin70c09bd2017-04-24 15:56:56 +0200211 int InsertPacketInternal(const RTPHeader& rtp_header,
Karl Wiberg45eb1352019-10-10 14:23:00 +0200212 rtc::ArrayView<const uint8_t> payload)
danilchap56359be2017-09-07 07:53:45 -0700213 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000214
henrik.lundin6d8e0112016-03-04 10:34:21 -0800215 // Delivers 10 ms of audio data. The data is written to |audio_frame|.
henrik.lundin@webrtc.orge1d468c2013-01-30 07:37:20 +0000216 // Returns 0 on success, otherwise an error code.
Ivo Creusen55de08e2018-09-03 11:49:27 +0200217 int GetAudioInternal(AudioFrame* audio_frame,
218 bool* muted,
Ivo Creusen3ce44a32019-10-31 14:38:11 +0100219 absl::optional<Operation> action_override)
danilchap56359be2017-09-07 07:53:45 -0700220 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000221
222 // Provides a decision to the GetAudioInternal method. The decision what to
223 // do is written to |operation|. Packets to decode are written to
224 // |packet_list|, and a DTMF event to play is written to |dtmf_event|. When
225 // DTMF should be played, |play_dtmf| is set to true by the method.
226 // Returns 0 on success, otherwise an error code.
Ivo Creusen3ce44a32019-10-31 14:38:11 +0100227 int GetDecision(Operation* operation,
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000228 PacketList* packet_list,
229 DtmfEvent* dtmf_event,
Ivo Creusen55de08e2018-09-03 11:49:27 +0200230 bool* play_dtmf,
Ivo Creusen3ce44a32019-10-31 14:38:11 +0100231 absl::optional<Operation> action_override)
Ivo Creusen55de08e2018-09-03 11:49:27 +0200232 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000233
234 // Decodes the speech packets in |packet_list|, and writes the results to
235 // |decoded_buffer|, which is allocated to hold |decoded_buffer_length|
236 // elements. The length of the decoded data is written to |decoded_length|.
237 // The speech type -- speech or (codec-internal) comfort noise -- is written
238 // to |speech_type|. If |packet_list| contains any SID frames for RFC 3389
239 // comfort noise, those are not decoded.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000240 int Decode(PacketList* packet_list,
Ivo Creusen3ce44a32019-10-31 14:38:11 +0100241 Operation* operation,
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000242 int* decoded_length,
243 AudioDecoder::SpeechType* speech_type)
danilchap56359be2017-09-07 07:53:45 -0700244 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000245
minyuel6d92bf52015-09-23 15:20:39 +0200246 // Sub-method to Decode(). Performs codec internal CNG.
danilchap56359be2017-09-07 07:53:45 -0700247 int DecodeCng(AudioDecoder* decoder,
248 int* decoded_length,
minyuel6d92bf52015-09-23 15:20:39 +0200249 AudioDecoder::SpeechType* speech_type)
danilchap56359be2017-09-07 07:53:45 -0700250 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
minyuel6d92bf52015-09-23 15:20:39 +0200251
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000252 // Sub-method to Decode(). Performs the actual decoding.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000253 int DecodeLoop(PacketList* packet_list,
Ivo Creusen3ce44a32019-10-31 14:38:11 +0100254 const Operation& operation,
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000255 AudioDecoder* decoder,
256 int* decoded_length,
257 AudioDecoder::SpeechType* speech_type)
danilchap56359be2017-09-07 07:53:45 -0700258 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000259
260 // Sub-method which calls the Normal class to perform the normal operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000261 void DoNormal(const int16_t* decoded_buffer,
262 size_t decoded_length,
263 AudioDecoder::SpeechType speech_type,
danilchap56359be2017-09-07 07:53:45 -0700264 bool play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000265
266 // Sub-method which calls the Merge class to perform the merge operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000267 void DoMerge(int16_t* decoded_buffer,
268 size_t decoded_length,
269 AudioDecoder::SpeechType speech_type,
danilchap56359be2017-09-07 07:53:45 -0700270 bool play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000271
Henrik Lundin00eb12a2018-09-05 18:14:52 +0200272 bool DoCodecPlc() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
273
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000274 // Sub-method which calls the Expand class to perform the expand operation.
danilchap56359be2017-09-07 07:53:45 -0700275 int DoExpand(bool play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000276
277 // Sub-method which calls the Accelerate class to perform the accelerate
278 // operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000279 int DoAccelerate(int16_t* decoded_buffer,
280 size_t decoded_length,
281 AudioDecoder::SpeechType speech_type,
Henrik Lundincf808d22015-05-27 14:33:29 +0200282 bool play_dtmf,
danilchap56359be2017-09-07 07:53:45 -0700283 bool fast_accelerate)
284 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000285
286 // Sub-method which calls the PreemptiveExpand class to perform the
287 // preemtive expand operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000288 int DoPreemptiveExpand(int16_t* decoded_buffer,
289 size_t decoded_length,
290 AudioDecoder::SpeechType speech_type,
danilchap56359be2017-09-07 07:53:45 -0700291 bool play_dtmf)
292 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000293
294 // Sub-method which calls the ComfortNoise class to generate RFC 3389 comfort
295 // noise. |packet_list| can either contain one SID frame to update the
296 // noise parameters, or no payload at all, in which case the previously
297 // received parameters are used.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000298 int DoRfc3389Cng(PacketList* packet_list, bool play_dtmf)
danilchap56359be2017-09-07 07:53:45 -0700299 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000300
301 // Calls the audio decoder to generate codec-internal comfort noise when
302 // no packet was received.
minyuel6d92bf52015-09-23 15:20:39 +0200303 void DoCodecInternalCng(const int16_t* decoded_buffer, size_t decoded_length)
danilchap56359be2017-09-07 07:53:45 -0700304 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000305
306 // Calls the DtmfToneGenerator class to generate DTMF tones.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000307 int DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf)
danilchap56359be2017-09-07 07:53:45 -0700308 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000309
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000310 // Overdub DTMF on top of |output|.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000311 int DtmfOverdub(const DtmfEvent& dtmf_event,
312 size_t num_channels,
danilchap56359be2017-09-07 07:53:45 -0700313 int16_t* output) const
314 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000315
316 // Extracts packets from |packet_buffer_| to produce at least
317 // |required_samples| samples. The packets are inserted into |packet_list|.
318 // Returns the number of samples that the packets in the list will produce, or
319 // -1 in case of an error.
Peter Kastingdce40cf2015-08-24 14:52:23 -0700320 int ExtractPackets(size_t required_samples, PacketList* packet_list)
danilchap56359be2017-09-07 07:53:45 -0700321 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000322
323 // Resets various variables and objects to new values based on the sample rate
324 // |fs_hz| and |channels| number audio channels.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000325 void SetSampleRateAndChannels(int fs_hz, size_t channels)
danilchap56359be2017-09-07 07:53:45 -0700326 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000327
328 // Returns the output type for the audio produced by the latest call to
329 // GetAudio().
danilchap56359be2017-09-07 07:53:45 -0700330 OutputType LastOutputType() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000331
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000332 // Updates Expand and Merge.
333 virtual void UpdatePlcComponents(int fs_hz, size_t channels)
danilchap56359be2017-09-07 07:53:45 -0700334 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000335
Alessio Bazzica8f319a32019-07-24 16:47:02 +0000336 Clock* const clock_;
337
pbos5ad935c2016-01-25 03:52:44 -0800338 rtc::CriticalSection crit_sect_;
danilchap56359be2017-09-07 07:53:45 -0700339 const std::unique_ptr<TickTimer> tick_timer_ RTC_GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800340 const std::unique_ptr<DecoderDatabase> decoder_database_
danilchap56359be2017-09-07 07:53:45 -0700341 RTC_GUARDED_BY(crit_sect_);
danilchap56359be2017-09-07 07:53:45 -0700342 const std::unique_ptr<DtmfBuffer> dtmf_buffer_ RTC_GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800343 const std::unique_ptr<DtmfToneGenerator> dtmf_tone_generator_
danilchap56359be2017-09-07 07:53:45 -0700344 RTC_GUARDED_BY(crit_sect_);
345 const std::unique_ptr<PacketBuffer> packet_buffer_ RTC_GUARDED_BY(crit_sect_);
ossua70695a2016-09-22 02:06:28 -0700346 const std::unique_ptr<RedPayloadSplitter> red_payload_splitter_
danilchap56359be2017-09-07 07:53:45 -0700347 RTC_GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800348 const std::unique_ptr<TimestampScaler> timestamp_scaler_
danilchap56359be2017-09-07 07:53:45 -0700349 RTC_GUARDED_BY(crit_sect_);
350 const std::unique_ptr<PostDecodeVad> vad_ RTC_GUARDED_BY(crit_sect_);
351 const std::unique_ptr<ExpandFactory> expand_factory_
352 RTC_GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800353 const std::unique_ptr<AccelerateFactory> accelerate_factory_
danilchap56359be2017-09-07 07:53:45 -0700354 RTC_GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800355 const std::unique_ptr<PreemptiveExpandFactory> preemptive_expand_factory_
danilchap56359be2017-09-07 07:53:45 -0700356 RTC_GUARDED_BY(crit_sect_);
Jakob Ivarsson44507082019-03-05 16:59:03 +0100357 const std::unique_ptr<StatisticsCalculator> stats_ RTC_GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000358
danilchap56359be2017-09-07 07:53:45 -0700359 std::unique_ptr<BackgroundNoise> background_noise_ RTC_GUARDED_BY(crit_sect_);
Ivo Creusen53a31f72019-10-24 15:20:39 +0200360 std::unique_ptr<NetEqController> controller_ RTC_GUARDED_BY(crit_sect_);
danilchap56359be2017-09-07 07:53:45 -0700361 std::unique_ptr<AudioMultiVector> algorithm_buffer_
362 RTC_GUARDED_BY(crit_sect_);
363 std::unique_ptr<SyncBuffer> sync_buffer_ RTC_GUARDED_BY(crit_sect_);
364 std::unique_ptr<Expand> expand_ RTC_GUARDED_BY(crit_sect_);
365 std::unique_ptr<Normal> normal_ RTC_GUARDED_BY(crit_sect_);
366 std::unique_ptr<Merge> merge_ RTC_GUARDED_BY(crit_sect_);
367 std::unique_ptr<Accelerate> accelerate_ RTC_GUARDED_BY(crit_sect_);
368 std::unique_ptr<PreemptiveExpand> preemptive_expand_
369 RTC_GUARDED_BY(crit_sect_);
370 RandomVector random_vector_ RTC_GUARDED_BY(crit_sect_);
371 std::unique_ptr<ComfortNoise> comfort_noise_ RTC_GUARDED_BY(crit_sect_);
danilchap56359be2017-09-07 07:53:45 -0700372 int fs_hz_ RTC_GUARDED_BY(crit_sect_);
373 int fs_mult_ RTC_GUARDED_BY(crit_sect_);
374 int last_output_sample_rate_hz_ RTC_GUARDED_BY(crit_sect_);
375 size_t output_size_samples_ RTC_GUARDED_BY(crit_sect_);
376 size_t decoder_frame_length_ RTC_GUARDED_BY(crit_sect_);
Ivo Creusen3ce44a32019-10-31 14:38:11 +0100377 Mode last_mode_ RTC_GUARDED_BY(crit_sect_);
378 Operation last_operation_ RTC_GUARDED_BY(crit_sect_);
danilchap56359be2017-09-07 07:53:45 -0700379 size_t decoded_buffer_length_ RTC_GUARDED_BY(crit_sect_);
380 std::unique_ptr<int16_t[]> decoded_buffer_ RTC_GUARDED_BY(crit_sect_);
381 uint32_t playout_timestamp_ RTC_GUARDED_BY(crit_sect_);
382 bool new_codec_ RTC_GUARDED_BY(crit_sect_);
383 uint32_t timestamp_ RTC_GUARDED_BY(crit_sect_);
384 bool reset_decoder_ RTC_GUARDED_BY(crit_sect_);
Danil Chapovalovb6021232018-06-19 13:26:36 +0200385 absl::optional<uint8_t> current_rtp_payload_type_ RTC_GUARDED_BY(crit_sect_);
386 absl::optional<uint8_t> current_cng_rtp_payload_type_
danilchap56359be2017-09-07 07:53:45 -0700387 RTC_GUARDED_BY(crit_sect_);
danilchap56359be2017-09-07 07:53:45 -0700388 bool first_packet_ RTC_GUARDED_BY(crit_sect_);
danilchap56359be2017-09-07 07:53:45 -0700389 bool enable_fast_accelerate_ RTC_GUARDED_BY(crit_sect_);
390 std::unique_ptr<NackTracker> nack_ RTC_GUARDED_BY(crit_sect_);
391 bool nack_enabled_ RTC_GUARDED_BY(crit_sect_);
392 const bool enable_muted_state_ RTC_GUARDED_BY(crit_sect_);
393 AudioFrame::VADActivity last_vad_activity_ RTC_GUARDED_BY(crit_sect_) =
henrik.lundin500c04b2016-03-08 02:36:04 -0800394 AudioFrame::kVadPassive;
henrik.lundinb1fb72b2016-05-03 08:18:47 -0700395 std::unique_ptr<TickTimer::Stopwatch> generated_noise_stopwatch_
danilchap56359be2017-09-07 07:53:45 -0700396 RTC_GUARDED_BY(crit_sect_);
397 std::vector<uint32_t> last_decoded_timestamps_ RTC_GUARDED_BY(crit_sect_);
Alessio Bazzica8f319a32019-07-24 16:47:02 +0000398 std::vector<RtpPacketInfo> last_decoded_packet_infos_
399 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_