blob: 75055a7b47fe54e18b4069e6506bd6abc1fc75d1 [file] [log] [blame]
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
henrik.lundin@webrtc.org9c55f0f2014-06-09 08:10:28 +000011#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_
12#define WEBRTC_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
henrike@webrtc.org88fbb2d2014-05-21 21:18:46 +000017#include "webrtc/base/constructormagic.h"
Tommi9090e0b2016-01-20 13:39:36 +010018#include "webrtc/base/criticalsection.h"
pbos@webrtc.org38344ed2014-09-24 06:05:00 +000019#include "webrtc/base/thread_annotations.h"
henrik.lundin@webrtc.org9c55f0f2014-06-09 08:10:28 +000020#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
21#include "webrtc/modules/audio_coding/neteq/defines.h"
Henrik Kjellander74640892015-10-29 11:31:02 +010022#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
henrik.lundin@webrtc.org9c55f0f2014-06-09 08:10:28 +000023#include "webrtc/modules/audio_coding/neteq/packet.h" // Declare PacketList.
24#include "webrtc/modules/audio_coding/neteq/random_vector.h"
25#include "webrtc/modules/audio_coding/neteq/rtcp.h"
26#include "webrtc/modules/audio_coding/neteq/statistics_calculator.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000027#include "webrtc/typedefs.h"
28
29namespace webrtc {
30
31// Forward declarations.
henrik.lundin@webrtc.org40d3fc62013-09-18 12:19:50 +000032class Accelerate;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000033class BackgroundNoise;
34class BufferLevelFilter;
35class ComfortNoise;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000036class DecisionLogic;
37class DecoderDatabase;
38class DelayManager;
39class DelayPeakDetector;
40class DtmfBuffer;
41class DtmfToneGenerator;
42class Expand;
henrik.lundin@webrtc.org40d3fc62013-09-18 12:19:50 +000043class Merge;
henrik.lundin48ed9302015-10-29 05:36:24 -070044class Nack;
henrik.lundin@webrtc.org40d3fc62013-09-18 12:19:50 +000045class Normal;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000046class PacketBuffer;
47class PayloadSplitter;
48class PostDecodeVad;
henrik.lundin@webrtc.org40d3fc62013-09-18 12:19:50 +000049class PreemptiveExpand;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000050class RandomVector;
51class SyncBuffer;
52class TimestampScaler;
henrik.lundin@webrtc.orgd9faa462014-01-14 10:18:45 +000053struct AccelerateFactory;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000054struct DtmfEvent;
henrik.lundin@webrtc.orgd9faa462014-01-14 10:18:45 +000055struct ExpandFactory;
56struct PreemptiveExpandFactory;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000057
58class NetEqImpl : public webrtc::NetEq {
59 public:
henrik.lundin55480f52016-03-08 02:37:57 -080060 enum class OutputType {
61 kNormalSpeech,
62 kPLC,
63 kCNG,
64 kPLCCNG,
65 kVadPassive
66 };
67
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000068 // Creates a new NetEqImpl object. The object will assume ownership of all
69 // injected dependencies, and will delete them when done.
henrik.lundin@webrtc.orgea257842014-08-07 12:27:37 +000070 NetEqImpl(const NetEq::Config& config,
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000071 BufferLevelFilter* buffer_level_filter,
72 DecoderDatabase* decoder_database,
73 DelayManager* delay_manager,
74 DelayPeakDetector* delay_peak_detector,
75 DtmfBuffer* dtmf_buffer,
76 DtmfToneGenerator* dtmf_tone_generator,
77 PacketBuffer* packet_buffer,
78 PayloadSplitter* payload_splitter,
henrik.lundin@webrtc.orgd9faa462014-01-14 10:18:45 +000079 TimestampScaler* timestamp_scaler,
80 AccelerateFactory* accelerate_factory,
81 ExpandFactory* expand_factory,
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +000082 PreemptiveExpandFactory* preemptive_expand_factory,
83 bool create_components = true);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000084
Karl Wiberg7f6c4d42015-04-09 15:44:22 +020085 ~NetEqImpl() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000086
87 // Inserts a new packet into NetEq. The |receive_timestamp| is an indication
88 // of the time when the packet was received, and should be measured with
89 // the same tick rate as the RTP timestamp of the current payload.
90 // Returns 0 on success, -1 on failure.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000091 int InsertPacket(const WebRtcRTPHeader& rtp_header,
kwibergee2bac22015-11-11 10:34:00 -080092 rtc::ArrayView<const uint8_t> payload,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000093 uint32_t receive_timestamp) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000094
turaj@webrtc.org7b75ac62013-09-26 00:27:56 +000095 // Inserts a sync-packet into packet queue. Sync-packets are decoded to
96 // silence and are intended to keep AV-sync intact in an event of long packet
97 // losses when Video NACK is enabled but Audio NACK is not. Clients of NetEq
98 // might insert sync-packet when they observe that buffer level of NetEq is
99 // decreasing below a certain threshold, defined by the application.
100 // Sync-packets should have the same payload type as the last audio payload
101 // type, i.e. they cannot have DTMF or CNG payload type, nor a codec change
102 // can be implied by inserting a sync-packet.
103 // Returns kOk on success, kFail on failure.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000104 int InsertSyncPacket(const WebRtcRTPHeader& rtp_header,
105 uint32_t receive_timestamp) override;
turaj@webrtc.org7b75ac62013-09-26 00:27:56 +0000106
henrik.lundin55480f52016-03-08 02:37:57 -0800107 int GetAudio(AudioFrame* audio_frame) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000108
kwibergee1879c2015-10-29 06:20:28 -0700109 int RegisterPayloadType(NetEqDecoder codec,
henrik.lundin4cf61dd2015-12-09 06:20:58 -0800110 const std::string& codec_name,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000111 uint8_t rtp_payload_type) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000112
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000113 int RegisterExternalDecoder(AudioDecoder* decoder,
kwibergee1879c2015-10-29 06:20:28 -0700114 NetEqDecoder codec,
henrik.lundin4cf61dd2015-12-09 06:20:58 -0800115 const std::string& codec_name,
Karl Wibergd8399e62015-05-25 14:39:56 +0200116 uint8_t rtp_payload_type,
117 int sample_rate_hz) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000118
119 // Removes |rtp_payload_type| from the codec database. Returns 0 on success,
120 // -1 on failure.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000121 int RemovePayloadType(uint8_t rtp_payload_type) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000122
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000123 bool SetMinimumDelay(int delay_ms) override;
turaj@webrtc.orgf1efc572013-08-16 23:44:24 +0000124
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000125 bool SetMaximumDelay(int delay_ms) override;
turaj@webrtc.orgf1efc572013-08-16 23:44:24 +0000126
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000127 int LeastRequiredDelayMs() const override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000128
Karl Wiberg7f6c4d42015-04-09 15:44:22 +0200129 int SetTargetDelay() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000130
Karl Wiberg7f6c4d42015-04-09 15:44:22 +0200131 int TargetDelay() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000132
henrik.lundin9c3efd02015-08-27 13:12:22 -0700133 int CurrentDelayMs() const override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000134
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000135 // Sets the playout mode to |mode|.
henrik.lundin@webrtc.org7cbc4f92014-10-07 06:37:39 +0000136 // Deprecated.
137 // TODO(henrik.lundin) Delete.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000138 void SetPlayoutMode(NetEqPlayoutMode mode) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000139
140 // Returns the current playout mode.
henrik.lundin@webrtc.org7cbc4f92014-10-07 06:37:39 +0000141 // Deprecated.
142 // TODO(henrik.lundin) Delete.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000143 NetEqPlayoutMode PlayoutMode() const override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000144
145 // Writes the current network statistics to |stats|. The statistics are reset
146 // after the call.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000147 int NetworkStatistics(NetEqNetworkStatistics* stats) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000148
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000149 // Writes the current RTCP statistics to |stats|. The statistics are reset
150 // and a new report period is started with the call.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000151 void GetRtcpStatistics(RtcpStatistics* stats) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000152
153 // Same as RtcpStatistics(), but does not reset anything.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000154 void GetRtcpStatisticsNoReset(RtcpStatistics* stats) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000155
156 // Enables post-decode VAD. When enabled, GetAudio() will return
157 // kOutputVADPassive when the signal contains no speech.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000158 void EnableVad() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000159
160 // Disables post-decode VAD.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000161 void DisableVad() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000162
henrik.lundin15c51e32016-04-06 08:38:56 -0700163 rtc::Optional<uint32_t> GetPlayoutTimestamp() const override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000164
henrik.lundind89814b2015-11-23 06:49:25 -0800165 int last_output_sample_rate_hz() const override;
166
Karl Wiberg7f6c4d42015-04-09 15:44:22 +0200167 int SetTargetNumberOfChannels() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000168
Karl Wiberg7f6c4d42015-04-09 15:44:22 +0200169 int SetTargetSampleRate() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000170
171 // Returns the error code for the last occurred error. If no error has
172 // occurred, 0 is returned.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000173 int LastError() const override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000174
175 // Returns the error code last returned by a decoder (audio or comfort noise).
176 // When LastError() returns kDecoderErrorCode or kComfortNoiseErrorCode, check
177 // this method to get the decoder's error code.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000178 int LastDecoderError() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000179
180 // Flushes both the packet buffer and the sync buffer.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000181 void FlushBuffers() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000182
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000183 void PacketBufferStatistics(int* current_num_packets,
184 int* max_num_packets) const override;
turaj@webrtc.org7df97062013-08-02 18:07:13 +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.lundin@webrtc.orgb287d962014-04-07 21:21:45 +0000192 // This accessor method is only intended for testing purposes.
henrike@webrtc.org47658f12014-09-10 22:14:59 +0000193 const SyncBuffer* sync_buffer_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;
Peter Kastingdce40cf2015-08-24 14:52:23 -0700197 static const size_t kMaxFrameSize = 2880; // 60 ms @ 48 kHz.
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000198 // TODO(hlundin): Provide a better value for kSyncBufferSize.
Peter Kastingdce40cf2015-08-24 14:52:23 -0700199 static const size_t kSyncBufferSize = 2 * kMaxFrameSize;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000200
201 // Inserts a new packet into NetEq. This is used by the InsertPacket method
202 // above. Returns 0 on success, otherwise an error code.
203 // TODO(hlundin): Merge this with InsertPacket above?
204 int InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
kwibergee2bac22015-11-11 10:34:00 -0800205 rtc::ArrayView<const uint8_t> payload,
turaj@webrtc.org7b75ac62013-09-26 00:27:56 +0000206 uint32_t receive_timestamp,
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000207 bool is_sync_packet)
208 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000209
henrik.lundin6d8e0112016-03-04 10:34:21 -0800210 // Delivers 10 ms of audio data. The data is written to |audio_frame|.
henrik.lundin@webrtc.orge1d468c2013-01-30 07:37:20 +0000211 // Returns 0 on success, otherwise an error code.
henrik.lundin6d8e0112016-03-04 10:34:21 -0800212 int GetAudioInternal(AudioFrame* audio_frame)
Peter Kasting69558702016-01-12 16:26:35 -0800213 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000214
215 // Provides a decision to the GetAudioInternal method. The decision what to
216 // do is written to |operation|. Packets to decode are written to
217 // |packet_list|, and a DTMF event to play is written to |dtmf_event|. When
218 // DTMF should be played, |play_dtmf| is set to true by the method.
219 // Returns 0 on success, otherwise an error code.
220 int GetDecision(Operations* operation,
221 PacketList* packet_list,
222 DtmfEvent* dtmf_event,
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000223 bool* play_dtmf) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000224
225 // Decodes the speech packets in |packet_list|, and writes the results to
226 // |decoded_buffer|, which is allocated to hold |decoded_buffer_length|
227 // elements. The length of the decoded data is written to |decoded_length|.
228 // The speech type -- speech or (codec-internal) comfort noise -- is written
229 // to |speech_type|. If |packet_list| contains any SID frames for RFC 3389
230 // comfort noise, those are not decoded.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000231 int Decode(PacketList* packet_list,
232 Operations* operation,
233 int* decoded_length,
234 AudioDecoder::SpeechType* speech_type)
235 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000236
minyuel6d92bf52015-09-23 15:20:39 +0200237 // Sub-method to Decode(). Performs codec internal CNG.
238 int DecodeCng(AudioDecoder* decoder, int* decoded_length,
239 AudioDecoder::SpeechType* speech_type)
240 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
241
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000242 // Sub-method to Decode(). Performs the actual decoding.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000243 int DecodeLoop(PacketList* packet_list,
minyuel6d92bf52015-09-23 15:20:39 +0200244 const Operations& operation,
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000245 AudioDecoder* decoder,
246 int* decoded_length,
247 AudioDecoder::SpeechType* speech_type)
248 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000249
250 // Sub-method which calls the Normal class to perform the normal operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000251 void DoNormal(const int16_t* decoded_buffer,
252 size_t decoded_length,
253 AudioDecoder::SpeechType speech_type,
254 bool play_dtmf) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000255
256 // Sub-method which calls the Merge class to perform the merge operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000257 void DoMerge(int16_t* decoded_buffer,
258 size_t decoded_length,
259 AudioDecoder::SpeechType speech_type,
260 bool play_dtmf) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000261
262 // Sub-method which calls the Expand class to perform the expand operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000263 int DoExpand(bool play_dtmf) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000264
265 // Sub-method which calls the Accelerate class to perform the accelerate
266 // operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000267 int DoAccelerate(int16_t* decoded_buffer,
268 size_t decoded_length,
269 AudioDecoder::SpeechType speech_type,
Henrik Lundincf808d22015-05-27 14:33:29 +0200270 bool play_dtmf,
271 bool fast_accelerate) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000272
273 // Sub-method which calls the PreemptiveExpand class to perform the
274 // preemtive expand operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000275 int DoPreemptiveExpand(int16_t* decoded_buffer,
276 size_t decoded_length,
277 AudioDecoder::SpeechType speech_type,
278 bool play_dtmf) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000279
280 // Sub-method which calls the ComfortNoise class to generate RFC 3389 comfort
281 // noise. |packet_list| can either contain one SID frame to update the
282 // noise parameters, or no payload at all, in which case the previously
283 // received parameters are used.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000284 int DoRfc3389Cng(PacketList* packet_list, bool play_dtmf)
285 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000286
287 // Calls the audio decoder to generate codec-internal comfort noise when
288 // no packet was received.
minyuel6d92bf52015-09-23 15:20:39 +0200289 void DoCodecInternalCng(const int16_t* decoded_buffer, size_t decoded_length)
290 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000291
292 // Calls the DtmfToneGenerator class to generate DTMF tones.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000293 int DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf)
294 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000295
296 // Produces packet-loss concealment using alternative methods. If the codec
297 // has an internal PLC, it is called to generate samples. Otherwise, the
298 // method performs zero-stuffing.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000299 void DoAlternativePlc(bool increase_timestamp)
300 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000301
302 // Overdub DTMF on top of |output|.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000303 int DtmfOverdub(const DtmfEvent& dtmf_event,
304 size_t num_channels,
305 int16_t* output) const EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000306
307 // Extracts packets from |packet_buffer_| to produce at least
308 // |required_samples| samples. The packets are inserted into |packet_list|.
309 // Returns the number of samples that the packets in the list will produce, or
310 // -1 in case of an error.
Peter Kastingdce40cf2015-08-24 14:52:23 -0700311 int ExtractPackets(size_t required_samples, PacketList* packet_list)
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000312 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000313
314 // Resets various variables and objects to new values based on the sample rate
315 // |fs_hz| and |channels| number audio channels.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000316 void SetSampleRateAndChannels(int fs_hz, size_t channels)
317 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000318
319 // Returns the output type for the audio produced by the latest call to
320 // GetAudio().
henrik.lundin55480f52016-03-08 02:37:57 -0800321 OutputType LastOutputType() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000322
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000323 // Updates Expand and Merge.
324 virtual void UpdatePlcComponents(int fs_hz, size_t channels)
325 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
326
henrik.lundin@webrtc.org7cbc4f92014-10-07 06:37:39 +0000327 // Creates DecisionLogic object with the mode given by |playout_mode_|.
328 virtual void CreateDecisionLogic() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000329
pbos5ad935c2016-01-25 03:52:44 -0800330 rtc::CriticalSection crit_sect_;
kwiberg2d0c3322016-02-14 09:28:33 -0800331 const std::unique_ptr<BufferLevelFilter> buffer_level_filter_
henrik.lundin@webrtc.org2f816bb2014-06-05 10:37:13 +0000332 GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800333 const std::unique_ptr<DecoderDatabase> decoder_database_
henrik.lundin@webrtc.org2f816bb2014-06-05 10:37:13 +0000334 GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800335 const std::unique_ptr<DelayManager> delay_manager_ GUARDED_BY(crit_sect_);
336 const std::unique_ptr<DelayPeakDetector> delay_peak_detector_
henrik.lundin@webrtc.org2f816bb2014-06-05 10:37:13 +0000337 GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800338 const std::unique_ptr<DtmfBuffer> dtmf_buffer_ GUARDED_BY(crit_sect_);
339 const std::unique_ptr<DtmfToneGenerator> dtmf_tone_generator_
henrik.lundin@webrtc.org2f816bb2014-06-05 10:37:13 +0000340 GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800341 const std::unique_ptr<PacketBuffer> packet_buffer_ GUARDED_BY(crit_sect_);
342 const std::unique_ptr<PayloadSplitter> payload_splitter_
kwiberg@webrtc.org00b8f6b2015-02-26 14:34:55 +0000343 GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800344 const std::unique_ptr<TimestampScaler> timestamp_scaler_
kwiberg@webrtc.org00b8f6b2015-02-26 14:34:55 +0000345 GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800346 const std::unique_ptr<PostDecodeVad> vad_ GUARDED_BY(crit_sect_);
347 const std::unique_ptr<ExpandFactory> expand_factory_ GUARDED_BY(crit_sect_);
348 const std::unique_ptr<AccelerateFactory> accelerate_factory_
kwiberg@webrtc.org00b8f6b2015-02-26 14:34:55 +0000349 GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800350 const std::unique_ptr<PreemptiveExpandFactory> preemptive_expand_factory_
henrik.lundin@webrtc.org2f816bb2014-06-05 10:37:13 +0000351 GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000352
kwiberg2d0c3322016-02-14 09:28:33 -0800353 std::unique_ptr<BackgroundNoise> background_noise_ GUARDED_BY(crit_sect_);
354 std::unique_ptr<DecisionLogic> decision_logic_ GUARDED_BY(crit_sect_);
355 std::unique_ptr<AudioMultiVector> algorithm_buffer_ GUARDED_BY(crit_sect_);
356 std::unique_ptr<SyncBuffer> sync_buffer_ GUARDED_BY(crit_sect_);
357 std::unique_ptr<Expand> expand_ GUARDED_BY(crit_sect_);
358 std::unique_ptr<Normal> normal_ GUARDED_BY(crit_sect_);
359 std::unique_ptr<Merge> merge_ GUARDED_BY(crit_sect_);
360 std::unique_ptr<Accelerate> accelerate_ GUARDED_BY(crit_sect_);
361 std::unique_ptr<PreemptiveExpand> preemptive_expand_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000362 RandomVector random_vector_ GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800363 std::unique_ptr<ComfortNoise> comfort_noise_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000364 Rtcp rtcp_ GUARDED_BY(crit_sect_);
365 StatisticsCalculator stats_ GUARDED_BY(crit_sect_);
366 int fs_hz_ GUARDED_BY(crit_sect_);
367 int fs_mult_ GUARDED_BY(crit_sect_);
henrik.lundind89814b2015-11-23 06:49:25 -0800368 int last_output_sample_rate_hz_ GUARDED_BY(crit_sect_);
Peter Kastingdce40cf2015-08-24 14:52:23 -0700369 size_t output_size_samples_ GUARDED_BY(crit_sect_);
370 size_t decoder_frame_length_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000371 Modes last_mode_ GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800372 std::unique_ptr<int16_t[]> mute_factor_array_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000373 size_t decoded_buffer_length_ GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800374 std::unique_ptr<int16_t[]> decoded_buffer_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000375 uint32_t playout_timestamp_ GUARDED_BY(crit_sect_);
376 bool new_codec_ GUARDED_BY(crit_sect_);
377 uint32_t timestamp_ GUARDED_BY(crit_sect_);
378 bool reset_decoder_ GUARDED_BY(crit_sect_);
379 uint8_t current_rtp_payload_type_ GUARDED_BY(crit_sect_);
380 uint8_t current_cng_rtp_payload_type_ GUARDED_BY(crit_sect_);
381 uint32_t ssrc_ GUARDED_BY(crit_sect_);
382 bool first_packet_ GUARDED_BY(crit_sect_);
383 int error_code_ GUARDED_BY(crit_sect_); // Store last error code.
384 int decoder_error_code_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgea257842014-08-07 12:27:37 +0000385 const BackgroundNoiseMode background_noise_mode_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.org7cbc4f92014-10-07 06:37:39 +0000386 NetEqPlayoutMode playout_mode_ GUARDED_BY(crit_sect_);
Henrik Lundincf808d22015-05-27 14:33:29 +0200387 bool enable_fast_accelerate_ GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800388 std::unique_ptr<Nack> nack_ GUARDED_BY(crit_sect_);
henrik.lundin48ed9302015-10-29 05:36:24 -0700389 bool nack_enabled_ GUARDED_BY(crit_sect_);
henrik.lundin500c04b2016-03-08 02:36:04 -0800390 AudioFrame::VADActivity last_vad_activity_ GUARDED_BY(crit_sect_) =
391 AudioFrame::kVadPassive;
minyue@webrtc.orgd7301772013-08-29 00:58:14 +0000392
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000393 private:
henrikg3c089d72015-09-16 05:37:44 -0700394 RTC_DISALLOW_COPY_AND_ASSIGN(NetEqImpl);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000395};
396
397} // namespace webrtc
henrik.lundin@webrtc.org9c55f0f2014-06-09 08:10:28 +0000398#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_