blob: 8795ef5210d87937dd04a282272efcf3ac5e4591 [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"
henrik.lundinda8bbf62016-08-31 03:14:11 -070019#include "webrtc/base/optional.h"
pbos@webrtc.org38344ed2014-09-24 06:05:00 +000020#include "webrtc/base/thread_annotations.h"
henrik.lundin@webrtc.org9c55f0f2014-06-09 08:10:28 +000021#include "webrtc/modules/audio_coding/neteq/audio_multi_vector.h"
22#include "webrtc/modules/audio_coding/neteq/defines.h"
Henrik Kjellander74640892015-10-29 11:31:02 +010023#include "webrtc/modules/audio_coding/neteq/include/neteq.h"
henrik.lundin@webrtc.org9c55f0f2014-06-09 08:10:28 +000024#include "webrtc/modules/audio_coding/neteq/packet.h" // Declare PacketList.
25#include "webrtc/modules/audio_coding/neteq/random_vector.h"
26#include "webrtc/modules/audio_coding/neteq/rtcp.h"
27#include "webrtc/modules/audio_coding/neteq/statistics_calculator.h"
henrik.lundined497212016-04-25 10:11:38 -070028#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000029#include "webrtc/typedefs.h"
30
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;
49class PayloadSplitter;
50class 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:
henrik.lundin55480f52016-03-08 02:37:57 -080062 enum class OutputType {
63 kNormalSpeech,
64 kPLC,
65 kCNG,
66 kPLCCNG,
67 kVadPassive
68 };
69
henrik.lundin1d9061e2016-04-26 12:19:34 -070070 struct Dependencies {
71 // The constructor populates the Dependencies struct with the default
72 // implementations of the objects. They can all be replaced by the user
73 // before sending the struct to the NetEqImpl constructor. However, there
74 // are dependencies between some of the classes inside the struct, so
75 // swapping out one may make it necessary to re-create another one.
ossue3525782016-05-25 07:37:43 -070076 explicit Dependencies(
77 const NetEq::Config& config,
78 const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory);
henrik.lundin1d9061e2016-04-26 12:19:34 -070079 ~Dependencies();
80
81 std::unique_ptr<TickTimer> tick_timer;
82 std::unique_ptr<BufferLevelFilter> buffer_level_filter;
83 std::unique_ptr<DecoderDatabase> decoder_database;
84 std::unique_ptr<DelayPeakDetector> delay_peak_detector;
85 std::unique_ptr<DelayManager> delay_manager;
86 std::unique_ptr<DtmfBuffer> dtmf_buffer;
87 std::unique_ptr<DtmfToneGenerator> dtmf_tone_generator;
88 std::unique_ptr<PacketBuffer> packet_buffer;
89 std::unique_ptr<PayloadSplitter> payload_splitter;
90 std::unique_ptr<TimestampScaler> timestamp_scaler;
91 std::unique_ptr<AccelerateFactory> accelerate_factory;
92 std::unique_ptr<ExpandFactory> expand_factory;
93 std::unique_ptr<PreemptiveExpandFactory> preemptive_expand_factory;
94 };
95
96 // Creates a new NetEqImpl object.
henrik.lundin@webrtc.orgea257842014-08-07 12:27:37 +000097 NetEqImpl(const NetEq::Config& config,
henrik.lundin1d9061e2016-04-26 12:19:34 -070098 Dependencies&& deps,
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +000099 bool create_components = true);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000100
Karl Wiberg7f6c4d42015-04-09 15:44:22 +0200101 ~NetEqImpl() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000102
103 // Inserts a new packet into NetEq. The |receive_timestamp| is an indication
104 // of the time when the packet was received, and should be measured with
105 // the same tick rate as the RTP timestamp of the current payload.
106 // Returns 0 on success, -1 on failure.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000107 int InsertPacket(const WebRtcRTPHeader& rtp_header,
kwibergee2bac22015-11-11 10:34:00 -0800108 rtc::ArrayView<const uint8_t> payload,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000109 uint32_t receive_timestamp) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000110
henrik.lundin7a926812016-05-12 13:51:28 -0700111 int GetAudio(AudioFrame* audio_frame, bool* muted) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000112
kwibergee1879c2015-10-29 06:20:28 -0700113 int RegisterPayloadType(NetEqDecoder codec,
henrik.lundin4cf61dd2015-12-09 06:20:58 -0800114 const std::string& codec_name,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000115 uint8_t rtp_payload_type) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000116
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000117 int RegisterExternalDecoder(AudioDecoder* decoder,
kwibergee1879c2015-10-29 06:20:28 -0700118 NetEqDecoder codec,
henrik.lundin4cf61dd2015-12-09 06:20:58 -0800119 const std::string& codec_name,
kwiberg342f7402016-06-16 03:18:00 -0700120 uint8_t rtp_payload_type) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000121
122 // Removes |rtp_payload_type| from the codec database. Returns 0 on success,
123 // -1 on failure.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000124 int RemovePayloadType(uint8_t rtp_payload_type) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000125
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000126 bool SetMinimumDelay(int delay_ms) override;
turaj@webrtc.orgf1efc572013-08-16 23:44:24 +0000127
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000128 bool SetMaximumDelay(int delay_ms) override;
turaj@webrtc.orgf1efc572013-08-16 23:44:24 +0000129
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000130 int LeastRequiredDelayMs() const override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000131
Karl Wiberg7f6c4d42015-04-09 15:44:22 +0200132 int SetTargetDelay() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000133
Karl Wiberg7f6c4d42015-04-09 15:44:22 +0200134 int TargetDelay() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000135
henrik.lundin9c3efd02015-08-27 13:12:22 -0700136 int CurrentDelayMs() const override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000137
henrik.lundinb3f1c5d2016-08-22 15:39:53 -0700138 int FilteredCurrentDelayMs() const override;
139
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000140 // Sets the playout mode to |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 void SetPlayoutMode(NetEqPlayoutMode mode) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000144
145 // Returns the current playout mode.
henrik.lundin@webrtc.org7cbc4f92014-10-07 06:37:39 +0000146 // Deprecated.
147 // TODO(henrik.lundin) Delete.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000148 NetEqPlayoutMode PlayoutMode() const override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000149
150 // Writes the current network statistics to |stats|. The statistics are reset
151 // after the call.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000152 int NetworkStatistics(NetEqNetworkStatistics* stats) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000153
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000154 // Writes the current RTCP statistics to |stats|. The statistics are reset
155 // and a new report period is started with the call.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000156 void GetRtcpStatistics(RtcpStatistics* stats) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000157
158 // Same as RtcpStatistics(), but does not reset anything.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000159 void GetRtcpStatisticsNoReset(RtcpStatistics* stats) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000160
161 // Enables post-decode VAD. When enabled, GetAudio() will return
162 // kOutputVADPassive when the signal contains no speech.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000163 void EnableVad() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000164
165 // Disables post-decode VAD.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000166 void DisableVad() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000167
henrik.lundin15c51e32016-04-06 08:38:56 -0700168 rtc::Optional<uint32_t> GetPlayoutTimestamp() const override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000169
henrik.lundind89814b2015-11-23 06:49:25 -0800170 int last_output_sample_rate_hz() const override;
171
kwiberg6f0f6162016-09-20 03:07:46 -0700172 rtc::Optional<CodecInst> GetDecoder(int payload_type) const override;
173
Karl Wiberg7f6c4d42015-04-09 15:44:22 +0200174 int SetTargetNumberOfChannels() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000175
Karl Wiberg7f6c4d42015-04-09 15:44:22 +0200176 int SetTargetSampleRate() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000177
178 // Returns the error code for the last occurred error. If no error has
179 // occurred, 0 is returned.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000180 int LastError() const override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000181
182 // Returns the error code last returned by a decoder (audio or comfort noise).
183 // When LastError() returns kDecoderErrorCode or kComfortNoiseErrorCode, check
184 // this method to get the decoder's error code.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000185 int LastDecoderError() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000186
187 // Flushes both the packet buffer and the sync buffer.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000188 void FlushBuffers() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000189
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000190 void PacketBufferStatistics(int* current_num_packets,
191 int* max_num_packets) const override;
turaj@webrtc.org7df97062013-08-02 18:07:13 +0000192
henrik.lundin48ed9302015-10-29 05:36:24 -0700193 void EnableNack(size_t max_nack_list_size) override;
194
195 void DisableNack() override;
196
197 std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const override;
turaj@webrtc.orgff43c852013-09-25 00:07:27 +0000198
henrik.lundin@webrtc.orgb287d962014-04-07 21:21:45 +0000199 // This accessor method is only intended for testing purposes.
henrike@webrtc.org47658f12014-09-10 22:14:59 +0000200 const SyncBuffer* sync_buffer_for_test() const;
minyue5bd33972016-05-02 04:46:11 -0700201 Operations last_operation_for_test() const;
henrik.lundin@webrtc.orgb287d962014-04-07 21:21:45 +0000202
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000203 protected:
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000204 static const int kOutputSizeMs = 10;
minyue5bd33972016-05-02 04:46:11 -0700205 static const size_t kMaxFrameSize = 5760; // 120 ms @ 48 kHz.
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000206 // TODO(hlundin): Provide a better value for kSyncBufferSize.
minyue17461792016-05-03 13:32:05 -0700207 // Current value is kMaxFrameSize + 60 ms * 48 kHz, which is enough for
208 // calculating correlations of current frame against history.
209 static const size_t kSyncBufferSize = kMaxFrameSize + 60 * 48;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000210
211 // Inserts a new packet into NetEq. This is used by the InsertPacket method
212 // above. Returns 0 on success, otherwise an error code.
213 // TODO(hlundin): Merge this with InsertPacket above?
214 int InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
kwibergee2bac22015-11-11 10:34:00 -0800215 rtc::ArrayView<const uint8_t> payload,
ossu17e3fa12016-09-08 04:52:55 -0700216 uint32_t receive_timestamp)
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000217 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000218
henrik.lundin6d8e0112016-03-04 10:34:21 -0800219 // Delivers 10 ms of audio data. The data is written to |audio_frame|.
henrik.lundin@webrtc.orge1d468c2013-01-30 07:37:20 +0000220 // Returns 0 on success, otherwise an error code.
henrik.lundin7a926812016-05-12 13:51:28 -0700221 int GetAudioInternal(AudioFrame* audio_frame, bool* muted)
Peter Kasting69558702016-01-12 16:26:35 -0800222 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000223
224 // Provides a decision to the GetAudioInternal method. The decision what to
225 // do is written to |operation|. Packets to decode are written to
226 // |packet_list|, and a DTMF event to play is written to |dtmf_event|. When
227 // DTMF should be played, |play_dtmf| is set to true by the method.
228 // Returns 0 on success, otherwise an error code.
229 int GetDecision(Operations* operation,
230 PacketList* packet_list,
231 DtmfEvent* dtmf_event,
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000232 bool* play_dtmf) 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,
241 Operations* operation,
242 int* decoded_length,
243 AudioDecoder::SpeechType* speech_type)
244 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.
247 int DecodeCng(AudioDecoder* decoder, int* decoded_length,
248 AudioDecoder::SpeechType* speech_type)
249 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
250
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000251 // Sub-method to Decode(). Performs the actual decoding.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000252 int DecodeLoop(PacketList* packet_list,
minyuel6d92bf52015-09-23 15:20:39 +0200253 const Operations& operation,
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000254 AudioDecoder* decoder,
255 int* decoded_length,
256 AudioDecoder::SpeechType* speech_type)
257 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000258
259 // Sub-method which calls the Normal class to perform the normal operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000260 void DoNormal(const int16_t* decoded_buffer,
261 size_t decoded_length,
262 AudioDecoder::SpeechType speech_type,
263 bool play_dtmf) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000264
265 // Sub-method which calls the Merge class to perform the merge operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000266 void DoMerge(int16_t* decoded_buffer,
267 size_t decoded_length,
268 AudioDecoder::SpeechType speech_type,
269 bool play_dtmf) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000270
271 // Sub-method which calls the Expand class to perform the expand operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000272 int DoExpand(bool play_dtmf) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000273
274 // Sub-method which calls the Accelerate class to perform the accelerate
275 // operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000276 int DoAccelerate(int16_t* decoded_buffer,
277 size_t decoded_length,
278 AudioDecoder::SpeechType speech_type,
Henrik Lundincf808d22015-05-27 14:33:29 +0200279 bool play_dtmf,
280 bool fast_accelerate) 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,
287 bool play_dtmf) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000288
289 // Sub-method which calls the ComfortNoise class to generate RFC 3389 comfort
290 // noise. |packet_list| can either contain one SID frame to update the
291 // noise parameters, or no payload at all, in which case the previously
292 // received parameters are used.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000293 int DoRfc3389Cng(PacketList* packet_list, bool play_dtmf)
294 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000295
296 // Calls the audio decoder to generate codec-internal comfort noise when
297 // no packet was received.
minyuel6d92bf52015-09-23 15:20:39 +0200298 void DoCodecInternalCng(const int16_t* decoded_buffer, size_t decoded_length)
299 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000300
301 // Calls the DtmfToneGenerator class to generate DTMF tones.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000302 int DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf)
303 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000304
305 // Produces packet-loss concealment using alternative methods. If the codec
306 // has an internal PLC, it is called to generate samples. Otherwise, the
307 // method performs zero-stuffing.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000308 void DoAlternativePlc(bool increase_timestamp)
309 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000310
311 // Overdub DTMF on top of |output|.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000312 int DtmfOverdub(const DtmfEvent& dtmf_event,
313 size_t num_channels,
314 int16_t* output) const 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)
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000321 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)
326 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().
henrik.lundin55480f52016-03-08 02:37:57 -0800330 OutputType LastOutputType() 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)
334 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
335
henrik.lundin@webrtc.org7cbc4f92014-10-07 06:37:39 +0000336 // Creates DecisionLogic object with the mode given by |playout_mode_|.
337 virtual void CreateDecisionLogic() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000338
pbos5ad935c2016-01-25 03:52:44 -0800339 rtc::CriticalSection crit_sect_;
henrik.lundined497212016-04-25 10:11:38 -0700340 const std::unique_ptr<TickTimer> tick_timer_ GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800341 const std::unique_ptr<BufferLevelFilter> buffer_level_filter_
henrik.lundin@webrtc.org2f816bb2014-06-05 10:37:13 +0000342 GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800343 const std::unique_ptr<DecoderDatabase> decoder_database_
henrik.lundin@webrtc.org2f816bb2014-06-05 10:37:13 +0000344 GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800345 const std::unique_ptr<DelayManager> delay_manager_ GUARDED_BY(crit_sect_);
346 const std::unique_ptr<DelayPeakDetector> delay_peak_detector_
henrik.lundin@webrtc.org2f816bb2014-06-05 10:37:13 +0000347 GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800348 const std::unique_ptr<DtmfBuffer> dtmf_buffer_ GUARDED_BY(crit_sect_);
349 const std::unique_ptr<DtmfToneGenerator> dtmf_tone_generator_
henrik.lundin@webrtc.org2f816bb2014-06-05 10:37:13 +0000350 GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800351 const std::unique_ptr<PacketBuffer> packet_buffer_ GUARDED_BY(crit_sect_);
352 const std::unique_ptr<PayloadSplitter> payload_splitter_
kwiberg@webrtc.org00b8f6b2015-02-26 14:34:55 +0000353 GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800354 const std::unique_ptr<TimestampScaler> timestamp_scaler_
kwiberg@webrtc.org00b8f6b2015-02-26 14:34:55 +0000355 GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800356 const std::unique_ptr<PostDecodeVad> vad_ GUARDED_BY(crit_sect_);
357 const std::unique_ptr<ExpandFactory> expand_factory_ GUARDED_BY(crit_sect_);
358 const std::unique_ptr<AccelerateFactory> accelerate_factory_
kwiberg@webrtc.org00b8f6b2015-02-26 14:34:55 +0000359 GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800360 const std::unique_ptr<PreemptiveExpandFactory> preemptive_expand_factory_
henrik.lundin@webrtc.org2f816bb2014-06-05 10:37:13 +0000361 GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000362
kwiberg2d0c3322016-02-14 09:28:33 -0800363 std::unique_ptr<BackgroundNoise> background_noise_ GUARDED_BY(crit_sect_);
364 std::unique_ptr<DecisionLogic> decision_logic_ GUARDED_BY(crit_sect_);
365 std::unique_ptr<AudioMultiVector> algorithm_buffer_ GUARDED_BY(crit_sect_);
366 std::unique_ptr<SyncBuffer> sync_buffer_ GUARDED_BY(crit_sect_);
367 std::unique_ptr<Expand> expand_ GUARDED_BY(crit_sect_);
368 std::unique_ptr<Normal> normal_ GUARDED_BY(crit_sect_);
369 std::unique_ptr<Merge> merge_ GUARDED_BY(crit_sect_);
370 std::unique_ptr<Accelerate> accelerate_ GUARDED_BY(crit_sect_);
371 std::unique_ptr<PreemptiveExpand> preemptive_expand_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000372 RandomVector random_vector_ GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800373 std::unique_ptr<ComfortNoise> comfort_noise_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000374 Rtcp rtcp_ GUARDED_BY(crit_sect_);
375 StatisticsCalculator stats_ GUARDED_BY(crit_sect_);
376 int fs_hz_ GUARDED_BY(crit_sect_);
377 int fs_mult_ GUARDED_BY(crit_sect_);
henrik.lundind89814b2015-11-23 06:49:25 -0800378 int last_output_sample_rate_hz_ GUARDED_BY(crit_sect_);
Peter Kastingdce40cf2015-08-24 14:52:23 -0700379 size_t output_size_samples_ GUARDED_BY(crit_sect_);
380 size_t decoder_frame_length_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000381 Modes last_mode_ GUARDED_BY(crit_sect_);
minyue5bd33972016-05-02 04:46:11 -0700382 Operations last_operation_ GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800383 std::unique_ptr<int16_t[]> mute_factor_array_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000384 size_t decoded_buffer_length_ GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800385 std::unique_ptr<int16_t[]> decoded_buffer_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000386 uint32_t playout_timestamp_ GUARDED_BY(crit_sect_);
387 bool new_codec_ GUARDED_BY(crit_sect_);
388 uint32_t timestamp_ GUARDED_BY(crit_sect_);
389 bool reset_decoder_ GUARDED_BY(crit_sect_);
henrik.lundinda8bbf62016-08-31 03:14:11 -0700390 rtc::Optional<uint8_t> current_rtp_payload_type_ GUARDED_BY(crit_sect_);
391 rtc::Optional<uint8_t> current_cng_rtp_payload_type_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000392 uint32_t ssrc_ GUARDED_BY(crit_sect_);
393 bool first_packet_ GUARDED_BY(crit_sect_);
394 int error_code_ GUARDED_BY(crit_sect_); // Store last error code.
395 int decoder_error_code_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgea257842014-08-07 12:27:37 +0000396 const BackgroundNoiseMode background_noise_mode_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.org7cbc4f92014-10-07 06:37:39 +0000397 NetEqPlayoutMode playout_mode_ GUARDED_BY(crit_sect_);
Henrik Lundincf808d22015-05-27 14:33:29 +0200398 bool enable_fast_accelerate_ GUARDED_BY(crit_sect_);
henrik.lundin91951862016-06-08 06:43:41 -0700399 std::unique_ptr<NackTracker> nack_ GUARDED_BY(crit_sect_);
henrik.lundin48ed9302015-10-29 05:36:24 -0700400 bool nack_enabled_ GUARDED_BY(crit_sect_);
henrik.lundin7a926812016-05-12 13:51:28 -0700401 const bool enable_muted_state_ GUARDED_BY(crit_sect_);
henrik.lundin500c04b2016-03-08 02:36:04 -0800402 AudioFrame::VADActivity last_vad_activity_ GUARDED_BY(crit_sect_) =
403 AudioFrame::kVadPassive;
henrik.lundinb1fb72b2016-05-03 08:18:47 -0700404 std::unique_ptr<TickTimer::Stopwatch> generated_noise_stopwatch_
405 GUARDED_BY(crit_sect_);
minyue@webrtc.orgd7301772013-08-29 00:58:14 +0000406
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000407 private:
henrikg3c089d72015-09-16 05:37:44 -0700408 RTC_DISALLOW_COPY_AND_ASSIGN(NetEqImpl);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000409};
410
411} // namespace webrtc
henrik.lundin@webrtc.org9c55f0f2014-06-09 08:10:28 +0000412#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_