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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
Mirko Bonadei92ea95e2017-09-15 06:47:31 +020017#include "api/optional.h"
18#include "modules/audio_coding/neteq/audio_multi_vector.h"
19#include "modules/audio_coding/neteq/defines.h"
20#include "modules/audio_coding/neteq/include/neteq.h"
21#include "modules/audio_coding/neteq/packet.h" // Declare PacketList.
22#include "modules/audio_coding/neteq/random_vector.h"
23#include "modules/audio_coding/neteq/rtcp.h"
24#include "modules/audio_coding/neteq/statistics_calculator.h"
25#include "modules/audio_coding/neteq/tick_timer.h"
26#include "modules/include/module_common_types.h"
27#include "rtc_base/constructormagic.h"
28#include "rtc_base/criticalsection.h"
29#include "rtc_base/thread_annotations.h"
Mirko Bonadei71207422017-09-15 13:58:09 +020030#include "typedefs.h" // NOLINT(build/include)
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000031
32namespace webrtc {
33
34// Forward declarations.
henrik.lundin@webrtc.org40d3fc62013-09-18 12:19:50 +000035class Accelerate;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000036class BackgroundNoise;
37class BufferLevelFilter;
38class ComfortNoise;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000039class DecisionLogic;
40class DecoderDatabase;
41class DelayManager;
42class DelayPeakDetector;
43class DtmfBuffer;
44class DtmfToneGenerator;
45class Expand;
henrik.lundin@webrtc.org40d3fc62013-09-18 12:19:50 +000046class Merge;
henrik.lundin91951862016-06-08 06:43:41 -070047class NackTracker;
henrik.lundin@webrtc.org40d3fc62013-09-18 12:19:50 +000048class Normal;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000049class PacketBuffer;
ossua70695a2016-09-22 02:06:28 -070050class RedPayloadSplitter;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000051class PostDecodeVad;
henrik.lundin@webrtc.org40d3fc62013-09-18 12:19:50 +000052class PreemptiveExpand;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000053class RandomVector;
54class SyncBuffer;
55class TimestampScaler;
henrik.lundin@webrtc.orgd9faa462014-01-14 10:18:45 +000056struct AccelerateFactory;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000057struct DtmfEvent;
henrik.lundin@webrtc.orgd9faa462014-01-14 10:18:45 +000058struct ExpandFactory;
59struct PreemptiveExpandFactory;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000060
61class NetEqImpl : public webrtc::NetEq {
62 public:
henrik.lundin55480f52016-03-08 02:37:57 -080063 enum class OutputType {
64 kNormalSpeech,
65 kPLC,
66 kCNG,
67 kPLCCNG,
68 kVadPassive
69 };
70
Henrik Lundinc417d9e2017-06-14 12:29:03 +020071 enum ErrorCodes {
72 kNoError = 0,
73 kOtherError,
74 kUnknownRtpPayloadType,
75 kDecoderNotFound,
76 kInvalidPointer,
77 kAccelerateError,
78 kPreemptiveExpandError,
79 kComfortNoiseErrorCode,
80 kDecoderErrorCode,
81 kOtherDecoderError,
82 kInvalidOperation,
83 kDtmfParsingError,
84 kDtmfInsertError,
85 kSampleUnderrun,
86 kDecodedTooMuch,
87 kRedundancySplitError,
88 kPacketBufferCorruption
89 };
90
henrik.lundin1d9061e2016-04-26 12:19:34 -070091 struct Dependencies {
92 // The constructor populates the Dependencies struct with the default
93 // implementations of the objects. They can all be replaced by the user
94 // before sending the struct to the NetEqImpl constructor. However, there
95 // are dependencies between some of the classes inside the struct, so
96 // swapping out one may make it necessary to re-create another one.
ossue3525782016-05-25 07:37:43 -070097 explicit Dependencies(
98 const NetEq::Config& config,
99 const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory);
henrik.lundin1d9061e2016-04-26 12:19:34 -0700100 ~Dependencies();
101
102 std::unique_ptr<TickTimer> tick_timer;
103 std::unique_ptr<BufferLevelFilter> buffer_level_filter;
104 std::unique_ptr<DecoderDatabase> decoder_database;
105 std::unique_ptr<DelayPeakDetector> delay_peak_detector;
106 std::unique_ptr<DelayManager> delay_manager;
107 std::unique_ptr<DtmfBuffer> dtmf_buffer;
108 std::unique_ptr<DtmfToneGenerator> dtmf_tone_generator;
109 std::unique_ptr<PacketBuffer> packet_buffer;
ossua70695a2016-09-22 02:06:28 -0700110 std::unique_ptr<RedPayloadSplitter> red_payload_splitter;
henrik.lundin1d9061e2016-04-26 12:19:34 -0700111 std::unique_ptr<TimestampScaler> timestamp_scaler;
112 std::unique_ptr<AccelerateFactory> accelerate_factory;
113 std::unique_ptr<ExpandFactory> expand_factory;
114 std::unique_ptr<PreemptiveExpandFactory> preemptive_expand_factory;
115 };
116
117 // Creates a new NetEqImpl object.
henrik.lundin@webrtc.orgea257842014-08-07 12:27:37 +0000118 NetEqImpl(const NetEq::Config& config,
henrik.lundin1d9061e2016-04-26 12:19:34 -0700119 Dependencies&& deps,
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000120 bool create_components = true);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000121
Karl Wiberg7f6c4d42015-04-09 15:44:22 +0200122 ~NetEqImpl() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000123
124 // Inserts a new packet into NetEq. The |receive_timestamp| is an indication
125 // of the time when the packet was received, and should be measured with
126 // the same tick rate as the RTP timestamp of the current payload.
127 // Returns 0 on success, -1 on failure.
Henrik Lundin70c09bd2017-04-24 15:56:56 +0200128 int InsertPacket(const RTPHeader& rtp_header,
kwibergee2bac22015-11-11 10:34:00 -0800129 rtc::ArrayView<const uint8_t> payload,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000130 uint32_t receive_timestamp) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000131
henrik.lundinb8c55b12017-05-10 07:38:01 -0700132 void InsertEmptyPacket(const RTPHeader& rtp_header) override;
133
henrik.lundin7a926812016-05-12 13:51:28 -0700134 int GetAudio(AudioFrame* audio_frame, bool* muted) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000135
kwiberg1c07c702017-03-27 07:15:49 -0700136 void SetCodecs(const std::map<int, SdpAudioFormat>& codecs) override;
137
kwibergee1879c2015-10-29 06:20:28 -0700138 int RegisterPayloadType(NetEqDecoder codec,
henrik.lundin4cf61dd2015-12-09 06:20:58 -0800139 const std::string& codec_name,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000140 uint8_t rtp_payload_type) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000141
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000142 int RegisterExternalDecoder(AudioDecoder* decoder,
kwibergee1879c2015-10-29 06:20:28 -0700143 NetEqDecoder codec,
henrik.lundin4cf61dd2015-12-09 06:20:58 -0800144 const std::string& codec_name,
kwiberg342f7402016-06-16 03:18:00 -0700145 uint8_t rtp_payload_type) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000146
kwiberg5adaf732016-10-04 09:33:27 -0700147 bool RegisterPayloadType(int rtp_payload_type,
148 const SdpAudioFormat& audio_format) override;
149
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000150 // Removes |rtp_payload_type| from the codec database. Returns 0 on success,
151 // -1 on failure.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000152 int RemovePayloadType(uint8_t rtp_payload_type) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000153
kwiberg6b19b562016-09-20 04:02:25 -0700154 void RemoveAllPayloadTypes() override;
155
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000156 bool SetMinimumDelay(int delay_ms) override;
turaj@webrtc.orgf1efc572013-08-16 23:44:24 +0000157
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000158 bool SetMaximumDelay(int delay_ms) override;
turaj@webrtc.orgf1efc572013-08-16 23:44:24 +0000159
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000160 int LeastRequiredDelayMs() const override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000161
Karl Wiberg7f6c4d42015-04-09 15:44:22 +0200162 int SetTargetDelay() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000163
henrik.lundin114c1b32017-04-26 07:47:32 -0700164 int TargetDelayMs() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000165
henrik.lundin9c3efd02015-08-27 13:12:22 -0700166 int CurrentDelayMs() 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 // Sets the playout mode to |mode|.
henrik.lundin@webrtc.org7cbc4f92014-10-07 06:37:39 +0000171 // Deprecated.
172 // TODO(henrik.lundin) Delete.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000173 void SetPlayoutMode(NetEqPlayoutMode mode) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000174
175 // Returns the current playout mode.
henrik.lundin@webrtc.org7cbc4f92014-10-07 06:37:39 +0000176 // Deprecated.
177 // TODO(henrik.lundin) Delete.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000178 NetEqPlayoutMode PlayoutMode() const override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000179
180 // Writes the current network statistics to |stats|. The statistics are reset
181 // after the call.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000182 int NetworkStatistics(NetEqNetworkStatistics* stats) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000183
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000184 // Writes the current RTCP statistics to |stats|. The statistics are reset
185 // and a new report period is started with the call.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000186 void GetRtcpStatistics(RtcpStatistics* stats) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000187
Steve Anton2dbc69f2017-08-24 17:15:13 -0700188 NetEqLifetimeStatistics GetLifetimeStatistics() const override;
189
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000190 // Same as RtcpStatistics(), but does not reset anything.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000191 void GetRtcpStatisticsNoReset(RtcpStatistics* stats) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000192
193 // Enables post-decode VAD. When enabled, GetAudio() will return
194 // kOutputVADPassive when the signal contains no speech.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000195 void EnableVad() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000196
197 // Disables post-decode VAD.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000198 void DisableVad() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000199
henrik.lundin15c51e32016-04-06 08:38:56 -0700200 rtc::Optional<uint32_t> GetPlayoutTimestamp() const override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000201
henrik.lundind89814b2015-11-23 06:49:25 -0800202 int last_output_sample_rate_hz() const override;
203
kwiberg6f0f6162016-09-20 03:07:46 -0700204 rtc::Optional<CodecInst> GetDecoder(int payload_type) const override;
205
ossuf1b08da2016-09-23 02:19:43 -0700206 rtc::Optional<SdpAudioFormat> GetDecoderFormat(
207 int payload_type) const override;
kwibergc4ccd4d2016-09-21 10:55:15 -0700208
Karl Wiberg7f6c4d42015-04-09 15:44:22 +0200209 int SetTargetNumberOfChannels() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000210
Karl Wiberg7f6c4d42015-04-09 15:44:22 +0200211 int SetTargetSampleRate() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000212
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000213 // Flushes both the packet buffer and the sync buffer.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000214 void FlushBuffers() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000215
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000216 void PacketBufferStatistics(int* current_num_packets,
217 int* max_num_packets) const override;
turaj@webrtc.org7df97062013-08-02 18:07:13 +0000218
henrik.lundin48ed9302015-10-29 05:36:24 -0700219 void EnableNack(size_t max_nack_list_size) override;
220
221 void DisableNack() override;
222
223 std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const override;
turaj@webrtc.orgff43c852013-09-25 00:07:27 +0000224
henrik.lundin114c1b32017-04-26 07:47:32 -0700225 std::vector<uint32_t> LastDecodedTimestamps() const override;
226
227 int SyncBufferSizeMs() const override;
228
henrik.lundin@webrtc.orgb287d962014-04-07 21:21:45 +0000229 // This accessor method is only intended for testing purposes.
henrike@webrtc.org47658f12014-09-10 22:14:59 +0000230 const SyncBuffer* sync_buffer_for_test() const;
minyue5bd33972016-05-02 04:46:11 -0700231 Operations last_operation_for_test() const;
henrik.lundin@webrtc.orgb287d962014-04-07 21:21:45 +0000232
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000233 protected:
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000234 static const int kOutputSizeMs = 10;
minyue5bd33972016-05-02 04:46:11 -0700235 static const size_t kMaxFrameSize = 5760; // 120 ms @ 48 kHz.
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000236 // TODO(hlundin): Provide a better value for kSyncBufferSize.
minyue17461792016-05-03 13:32:05 -0700237 // Current value is kMaxFrameSize + 60 ms * 48 kHz, which is enough for
238 // calculating correlations of current frame against history.
239 static const size_t kSyncBufferSize = kMaxFrameSize + 60 * 48;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000240
241 // Inserts a new packet into NetEq. This is used by the InsertPacket method
242 // above. Returns 0 on success, otherwise an error code.
243 // TODO(hlundin): Merge this with InsertPacket above?
Henrik Lundin70c09bd2017-04-24 15:56:56 +0200244 int InsertPacketInternal(const RTPHeader& rtp_header,
kwibergee2bac22015-11-11 10:34:00 -0800245 rtc::ArrayView<const uint8_t> payload,
ossu17e3fa12016-09-08 04:52:55 -0700246 uint32_t receive_timestamp)
danilchap56359be2017-09-07 07:53:45 -0700247 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000248
henrik.lundin6d8e0112016-03-04 10:34:21 -0800249 // Delivers 10 ms of audio data. The data is written to |audio_frame|.
henrik.lundin@webrtc.orge1d468c2013-01-30 07:37:20 +0000250 // Returns 0 on success, otherwise an error code.
henrik.lundin7a926812016-05-12 13:51:28 -0700251 int GetAudioInternal(AudioFrame* audio_frame, bool* muted)
danilchap56359be2017-09-07 07:53:45 -0700252 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000253
254 // Provides a decision to the GetAudioInternal method. The decision what to
255 // do is written to |operation|. Packets to decode are written to
256 // |packet_list|, and a DTMF event to play is written to |dtmf_event|. When
257 // DTMF should be played, |play_dtmf| is set to true by the method.
258 // Returns 0 on success, otherwise an error code.
259 int GetDecision(Operations* operation,
260 PacketList* packet_list,
261 DtmfEvent* dtmf_event,
danilchap56359be2017-09-07 07:53:45 -0700262 bool* play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000263
264 // Decodes the speech packets in |packet_list|, and writes the results to
265 // |decoded_buffer|, which is allocated to hold |decoded_buffer_length|
266 // elements. The length of the decoded data is written to |decoded_length|.
267 // The speech type -- speech or (codec-internal) comfort noise -- is written
268 // to |speech_type|. If |packet_list| contains any SID frames for RFC 3389
269 // comfort noise, those are not decoded.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000270 int Decode(PacketList* packet_list,
271 Operations* operation,
272 int* decoded_length,
273 AudioDecoder::SpeechType* speech_type)
danilchap56359be2017-09-07 07:53:45 -0700274 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000275
minyuel6d92bf52015-09-23 15:20:39 +0200276 // Sub-method to Decode(). Performs codec internal CNG.
danilchap56359be2017-09-07 07:53:45 -0700277 int DecodeCng(AudioDecoder* decoder,
278 int* decoded_length,
minyuel6d92bf52015-09-23 15:20:39 +0200279 AudioDecoder::SpeechType* speech_type)
danilchap56359be2017-09-07 07:53:45 -0700280 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
minyuel6d92bf52015-09-23 15:20:39 +0200281
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000282 // Sub-method to Decode(). Performs the actual decoding.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000283 int DecodeLoop(PacketList* packet_list,
minyuel6d92bf52015-09-23 15:20:39 +0200284 const Operations& operation,
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000285 AudioDecoder* decoder,
286 int* decoded_length,
287 AudioDecoder::SpeechType* speech_type)
danilchap56359be2017-09-07 07:53:45 -0700288 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000289
290 // Sub-method which calls the Normal class to perform the normal operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000291 void DoNormal(const int16_t* decoded_buffer,
292 size_t decoded_length,
293 AudioDecoder::SpeechType speech_type,
danilchap56359be2017-09-07 07:53:45 -0700294 bool play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000295
296 // Sub-method which calls the Merge class to perform the merge operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000297 void DoMerge(int16_t* decoded_buffer,
298 size_t decoded_length,
299 AudioDecoder::SpeechType speech_type,
danilchap56359be2017-09-07 07:53:45 -0700300 bool play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000301
302 // Sub-method which calls the Expand class to perform the expand operation.
danilchap56359be2017-09-07 07:53:45 -0700303 int DoExpand(bool play_dtmf) RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000304
305 // Sub-method which calls the Accelerate class to perform the accelerate
306 // operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000307 int DoAccelerate(int16_t* decoded_buffer,
308 size_t decoded_length,
309 AudioDecoder::SpeechType speech_type,
Henrik Lundincf808d22015-05-27 14:33:29 +0200310 bool play_dtmf,
danilchap56359be2017-09-07 07:53:45 -0700311 bool fast_accelerate)
312 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000313
314 // Sub-method which calls the PreemptiveExpand class to perform the
315 // preemtive expand operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000316 int DoPreemptiveExpand(int16_t* decoded_buffer,
317 size_t decoded_length,
318 AudioDecoder::SpeechType speech_type,
danilchap56359be2017-09-07 07:53:45 -0700319 bool play_dtmf)
320 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000321
322 // Sub-method which calls the ComfortNoise class to generate RFC 3389 comfort
323 // noise. |packet_list| can either contain one SID frame to update the
324 // noise parameters, or no payload at all, in which case the previously
325 // received parameters are used.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000326 int DoRfc3389Cng(PacketList* packet_list, bool play_dtmf)
danilchap56359be2017-09-07 07:53:45 -0700327 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000328
329 // Calls the audio decoder to generate codec-internal comfort noise when
330 // no packet was received.
minyuel6d92bf52015-09-23 15:20:39 +0200331 void DoCodecInternalCng(const int16_t* decoded_buffer, size_t decoded_length)
danilchap56359be2017-09-07 07:53:45 -0700332 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000333
334 // Calls the DtmfToneGenerator class to generate DTMF tones.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000335 int DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf)
danilchap56359be2017-09-07 07:53:45 -0700336 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000337
338 // Produces packet-loss concealment using alternative methods. If the codec
339 // has an internal PLC, it is called to generate samples. Otherwise, the
340 // method performs zero-stuffing.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000341 void DoAlternativePlc(bool increase_timestamp)
danilchap56359be2017-09-07 07:53:45 -0700342 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000343
344 // Overdub DTMF on top of |output|.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000345 int DtmfOverdub(const DtmfEvent& dtmf_event,
346 size_t num_channels,
danilchap56359be2017-09-07 07:53:45 -0700347 int16_t* output) const
348 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000349
350 // Extracts packets from |packet_buffer_| to produce at least
351 // |required_samples| samples. The packets are inserted into |packet_list|.
352 // Returns the number of samples that the packets in the list will produce, or
353 // -1 in case of an error.
Peter Kastingdce40cf2015-08-24 14:52:23 -0700354 int ExtractPackets(size_t required_samples, PacketList* packet_list)
danilchap56359be2017-09-07 07:53:45 -0700355 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000356
357 // Resets various variables and objects to new values based on the sample rate
358 // |fs_hz| and |channels| number audio channels.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000359 void SetSampleRateAndChannels(int fs_hz, size_t channels)
danilchap56359be2017-09-07 07:53:45 -0700360 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000361
362 // Returns the output type for the audio produced by the latest call to
363 // GetAudio().
danilchap56359be2017-09-07 07:53:45 -0700364 OutputType LastOutputType() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000365
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000366 // Updates Expand and Merge.
367 virtual void UpdatePlcComponents(int fs_hz, size_t channels)
danilchap56359be2017-09-07 07:53:45 -0700368 RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000369
henrik.lundin@webrtc.org7cbc4f92014-10-07 06:37:39 +0000370 // Creates DecisionLogic object with the mode given by |playout_mode_|.
danilchap56359be2017-09-07 07:53:45 -0700371 virtual void CreateDecisionLogic() RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000372
pbos5ad935c2016-01-25 03:52:44 -0800373 rtc::CriticalSection crit_sect_;
danilchap56359be2017-09-07 07:53:45 -0700374 const std::unique_ptr<TickTimer> tick_timer_ RTC_GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800375 const std::unique_ptr<BufferLevelFilter> buffer_level_filter_
danilchap56359be2017-09-07 07:53:45 -0700376 RTC_GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800377 const std::unique_ptr<DecoderDatabase> decoder_database_
danilchap56359be2017-09-07 07:53:45 -0700378 RTC_GUARDED_BY(crit_sect_);
379 const std::unique_ptr<DelayManager> delay_manager_ RTC_GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800380 const std::unique_ptr<DelayPeakDetector> delay_peak_detector_
danilchap56359be2017-09-07 07:53:45 -0700381 RTC_GUARDED_BY(crit_sect_);
382 const std::unique_ptr<DtmfBuffer> dtmf_buffer_ RTC_GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800383 const std::unique_ptr<DtmfToneGenerator> dtmf_tone_generator_
danilchap56359be2017-09-07 07:53:45 -0700384 RTC_GUARDED_BY(crit_sect_);
385 const std::unique_ptr<PacketBuffer> packet_buffer_ RTC_GUARDED_BY(crit_sect_);
ossua70695a2016-09-22 02:06:28 -0700386 const std::unique_ptr<RedPayloadSplitter> red_payload_splitter_
danilchap56359be2017-09-07 07:53:45 -0700387 RTC_GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800388 const std::unique_ptr<TimestampScaler> timestamp_scaler_
danilchap56359be2017-09-07 07:53:45 -0700389 RTC_GUARDED_BY(crit_sect_);
390 const std::unique_ptr<PostDecodeVad> vad_ RTC_GUARDED_BY(crit_sect_);
391 const std::unique_ptr<ExpandFactory> expand_factory_
392 RTC_GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800393 const std::unique_ptr<AccelerateFactory> accelerate_factory_
danilchap56359be2017-09-07 07:53:45 -0700394 RTC_GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800395 const std::unique_ptr<PreemptiveExpandFactory> preemptive_expand_factory_
danilchap56359be2017-09-07 07:53:45 -0700396 RTC_GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000397
danilchap56359be2017-09-07 07:53:45 -0700398 std::unique_ptr<BackgroundNoise> background_noise_ RTC_GUARDED_BY(crit_sect_);
399 std::unique_ptr<DecisionLogic> decision_logic_ RTC_GUARDED_BY(crit_sect_);
400 std::unique_ptr<AudioMultiVector> algorithm_buffer_
401 RTC_GUARDED_BY(crit_sect_);
402 std::unique_ptr<SyncBuffer> sync_buffer_ RTC_GUARDED_BY(crit_sect_);
403 std::unique_ptr<Expand> expand_ RTC_GUARDED_BY(crit_sect_);
404 std::unique_ptr<Normal> normal_ RTC_GUARDED_BY(crit_sect_);
405 std::unique_ptr<Merge> merge_ RTC_GUARDED_BY(crit_sect_);
406 std::unique_ptr<Accelerate> accelerate_ RTC_GUARDED_BY(crit_sect_);
407 std::unique_ptr<PreemptiveExpand> preemptive_expand_
408 RTC_GUARDED_BY(crit_sect_);
409 RandomVector random_vector_ RTC_GUARDED_BY(crit_sect_);
410 std::unique_ptr<ComfortNoise> comfort_noise_ RTC_GUARDED_BY(crit_sect_);
411 Rtcp rtcp_ RTC_GUARDED_BY(crit_sect_);
412 StatisticsCalculator stats_ RTC_GUARDED_BY(crit_sect_);
413 int fs_hz_ RTC_GUARDED_BY(crit_sect_);
414 int fs_mult_ RTC_GUARDED_BY(crit_sect_);
415 int last_output_sample_rate_hz_ RTC_GUARDED_BY(crit_sect_);
416 size_t output_size_samples_ RTC_GUARDED_BY(crit_sect_);
417 size_t decoder_frame_length_ RTC_GUARDED_BY(crit_sect_);
418 Modes last_mode_ RTC_GUARDED_BY(crit_sect_);
419 Operations last_operation_ RTC_GUARDED_BY(crit_sect_);
420 std::unique_ptr<int16_t[]> mute_factor_array_ RTC_GUARDED_BY(crit_sect_);
421 size_t decoded_buffer_length_ RTC_GUARDED_BY(crit_sect_);
422 std::unique_ptr<int16_t[]> decoded_buffer_ RTC_GUARDED_BY(crit_sect_);
423 uint32_t playout_timestamp_ RTC_GUARDED_BY(crit_sect_);
424 bool new_codec_ RTC_GUARDED_BY(crit_sect_);
425 uint32_t timestamp_ RTC_GUARDED_BY(crit_sect_);
426 bool reset_decoder_ RTC_GUARDED_BY(crit_sect_);
427 rtc::Optional<uint8_t> current_rtp_payload_type_ RTC_GUARDED_BY(crit_sect_);
428 rtc::Optional<uint8_t> current_cng_rtp_payload_type_
429 RTC_GUARDED_BY(crit_sect_);
430 uint32_t ssrc_ RTC_GUARDED_BY(crit_sect_);
431 bool first_packet_ RTC_GUARDED_BY(crit_sect_);
432 const BackgroundNoiseMode background_noise_mode_ RTC_GUARDED_BY(crit_sect_);
433 NetEqPlayoutMode playout_mode_ RTC_GUARDED_BY(crit_sect_);
434 bool enable_fast_accelerate_ RTC_GUARDED_BY(crit_sect_);
435 std::unique_ptr<NackTracker> nack_ RTC_GUARDED_BY(crit_sect_);
436 bool nack_enabled_ RTC_GUARDED_BY(crit_sect_);
437 const bool enable_muted_state_ RTC_GUARDED_BY(crit_sect_);
438 AudioFrame::VADActivity last_vad_activity_ RTC_GUARDED_BY(crit_sect_) =
henrik.lundin500c04b2016-03-08 02:36:04 -0800439 AudioFrame::kVadPassive;
henrik.lundinb1fb72b2016-05-03 08:18:47 -0700440 std::unique_ptr<TickTimer::Stopwatch> generated_noise_stopwatch_
danilchap56359be2017-09-07 07:53:45 -0700441 RTC_GUARDED_BY(crit_sect_);
442 std::vector<uint32_t> last_decoded_timestamps_ RTC_GUARDED_BY(crit_sect_);
minyue@webrtc.orgd7301772013-08-29 00:58:14 +0000443
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000444 private:
henrikg3c089d72015-09-16 05:37:44 -0700445 RTC_DISALLOW_COPY_AND_ASSIGN(NetEqImpl);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000446};
447
448} // namespace webrtc
Mirko Bonadei92ea95e2017-09-15 06:47:31 +0200449#endif // MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_