blob: e74f090ac6205e24c54d9e41384c30c0dd9b3760 [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
Karl Wiberg7f6c4d42015-04-09 15:44:22 +0200172 int SetTargetNumberOfChannels() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000173
Karl Wiberg7f6c4d42015-04-09 15:44:22 +0200174 int SetTargetSampleRate() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000175
176 // Returns the error code for the last occurred error. If no error has
177 // occurred, 0 is returned.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000178 int LastError() const override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000179
180 // Returns the error code last returned by a decoder (audio or comfort noise).
181 // When LastError() returns kDecoderErrorCode or kComfortNoiseErrorCode, check
182 // this method to get the decoder's error code.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000183 int LastDecoderError() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000184
185 // Flushes both the packet buffer and the sync buffer.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000186 void FlushBuffers() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000187
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000188 void PacketBufferStatistics(int* current_num_packets,
189 int* max_num_packets) const override;
turaj@webrtc.org7df97062013-08-02 18:07:13 +0000190
henrik.lundin48ed9302015-10-29 05:36:24 -0700191 void EnableNack(size_t max_nack_list_size) override;
192
193 void DisableNack() override;
194
195 std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const override;
turaj@webrtc.orgff43c852013-09-25 00:07:27 +0000196
henrik.lundin@webrtc.orgb287d962014-04-07 21:21:45 +0000197 // This accessor method is only intended for testing purposes.
henrike@webrtc.org47658f12014-09-10 22:14:59 +0000198 const SyncBuffer* sync_buffer_for_test() const;
minyue5bd33972016-05-02 04:46:11 -0700199 Operations last_operation_for_test() const;
henrik.lundin@webrtc.orgb287d962014-04-07 21:21:45 +0000200
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000201 protected:
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000202 static const int kOutputSizeMs = 10;
minyue5bd33972016-05-02 04:46:11 -0700203 static const size_t kMaxFrameSize = 5760; // 120 ms @ 48 kHz.
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000204 // TODO(hlundin): Provide a better value for kSyncBufferSize.
minyue17461792016-05-03 13:32:05 -0700205 // Current value is kMaxFrameSize + 60 ms * 48 kHz, which is enough for
206 // calculating correlations of current frame against history.
207 static const size_t kSyncBufferSize = kMaxFrameSize + 60 * 48;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000208
209 // Inserts a new packet into NetEq. This is used by the InsertPacket method
210 // above. Returns 0 on success, otherwise an error code.
211 // TODO(hlundin): Merge this with InsertPacket above?
212 int InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
kwibergee2bac22015-11-11 10:34:00 -0800213 rtc::ArrayView<const uint8_t> payload,
ossu17e3fa12016-09-08 04:52:55 -0700214 uint32_t receive_timestamp)
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000215 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000216
henrik.lundin6d8e0112016-03-04 10:34:21 -0800217 // Delivers 10 ms of audio data. The data is written to |audio_frame|.
henrik.lundin@webrtc.orge1d468c2013-01-30 07:37:20 +0000218 // Returns 0 on success, otherwise an error code.
henrik.lundin7a926812016-05-12 13:51:28 -0700219 int GetAudioInternal(AudioFrame* audio_frame, bool* muted)
Peter Kasting69558702016-01-12 16:26:35 -0800220 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.
227 int GetDecision(Operations* operation,
228 PacketList* packet_list,
229 DtmfEvent* dtmf_event,
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000230 bool* play_dtmf) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000231
232 // Decodes the speech packets in |packet_list|, and writes the results to
233 // |decoded_buffer|, which is allocated to hold |decoded_buffer_length|
234 // elements. The length of the decoded data is written to |decoded_length|.
235 // The speech type -- speech or (codec-internal) comfort noise -- is written
236 // to |speech_type|. If |packet_list| contains any SID frames for RFC 3389
237 // comfort noise, those are not decoded.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000238 int Decode(PacketList* packet_list,
239 Operations* operation,
240 int* decoded_length,
241 AudioDecoder::SpeechType* speech_type)
242 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000243
minyuel6d92bf52015-09-23 15:20:39 +0200244 // Sub-method to Decode(). Performs codec internal CNG.
245 int DecodeCng(AudioDecoder* decoder, int* decoded_length,
246 AudioDecoder::SpeechType* speech_type)
247 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
248
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000249 // Sub-method to Decode(). Performs the actual decoding.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000250 int DecodeLoop(PacketList* packet_list,
minyuel6d92bf52015-09-23 15:20:39 +0200251 const Operations& operation,
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000252 AudioDecoder* decoder,
253 int* decoded_length,
254 AudioDecoder::SpeechType* speech_type)
255 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000256
257 // Sub-method which calls the Normal class to perform the normal operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000258 void DoNormal(const int16_t* decoded_buffer,
259 size_t decoded_length,
260 AudioDecoder::SpeechType speech_type,
261 bool play_dtmf) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000262
263 // Sub-method which calls the Merge class to perform the merge operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000264 void DoMerge(int16_t* decoded_buffer,
265 size_t decoded_length,
266 AudioDecoder::SpeechType speech_type,
267 bool play_dtmf) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000268
269 // Sub-method which calls the Expand class to perform the expand operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000270 int DoExpand(bool play_dtmf) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000271
272 // Sub-method which calls the Accelerate class to perform the accelerate
273 // operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000274 int DoAccelerate(int16_t* decoded_buffer,
275 size_t decoded_length,
276 AudioDecoder::SpeechType speech_type,
Henrik Lundincf808d22015-05-27 14:33:29 +0200277 bool play_dtmf,
278 bool fast_accelerate) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000279
280 // Sub-method which calls the PreemptiveExpand class to perform the
281 // preemtive expand operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000282 int DoPreemptiveExpand(int16_t* decoded_buffer,
283 size_t decoded_length,
284 AudioDecoder::SpeechType speech_type,
285 bool play_dtmf) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000286
287 // Sub-method which calls the ComfortNoise class to generate RFC 3389 comfort
288 // noise. |packet_list| can either contain one SID frame to update the
289 // noise parameters, or no payload at all, in which case the previously
290 // received parameters are used.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000291 int DoRfc3389Cng(PacketList* packet_list, bool play_dtmf)
292 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000293
294 // Calls the audio decoder to generate codec-internal comfort noise when
295 // no packet was received.
minyuel6d92bf52015-09-23 15:20:39 +0200296 void DoCodecInternalCng(const int16_t* decoded_buffer, size_t decoded_length)
297 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000298
299 // Calls the DtmfToneGenerator class to generate DTMF tones.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000300 int DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf)
301 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000302
303 // Produces packet-loss concealment using alternative methods. If the codec
304 // has an internal PLC, it is called to generate samples. Otherwise, the
305 // method performs zero-stuffing.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000306 void DoAlternativePlc(bool increase_timestamp)
307 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000308
309 // Overdub DTMF on top of |output|.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000310 int DtmfOverdub(const DtmfEvent& dtmf_event,
311 size_t num_channels,
312 int16_t* output) const EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000313
314 // Extracts packets from |packet_buffer_| to produce at least
315 // |required_samples| samples. The packets are inserted into |packet_list|.
316 // Returns the number of samples that the packets in the list will produce, or
317 // -1 in case of an error.
Peter Kastingdce40cf2015-08-24 14:52:23 -0700318 int ExtractPackets(size_t required_samples, PacketList* packet_list)
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000319 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000320
321 // Resets various variables and objects to new values based on the sample rate
322 // |fs_hz| and |channels| number audio channels.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000323 void SetSampleRateAndChannels(int fs_hz, size_t channels)
324 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000325
326 // Returns the output type for the audio produced by the latest call to
327 // GetAudio().
henrik.lundin55480f52016-03-08 02:37:57 -0800328 OutputType LastOutputType() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000329
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000330 // Updates Expand and Merge.
331 virtual void UpdatePlcComponents(int fs_hz, size_t channels)
332 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
333
henrik.lundin@webrtc.org7cbc4f92014-10-07 06:37:39 +0000334 // Creates DecisionLogic object with the mode given by |playout_mode_|.
335 virtual void CreateDecisionLogic() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000336
pbos5ad935c2016-01-25 03:52:44 -0800337 rtc::CriticalSection crit_sect_;
henrik.lundined497212016-04-25 10:11:38 -0700338 const std::unique_ptr<TickTimer> tick_timer_ GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800339 const std::unique_ptr<BufferLevelFilter> buffer_level_filter_
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<DecoderDatabase> decoder_database_
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<DelayManager> delay_manager_ GUARDED_BY(crit_sect_);
344 const std::unique_ptr<DelayPeakDetector> delay_peak_detector_
henrik.lundin@webrtc.org2f816bb2014-06-05 10:37:13 +0000345 GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800346 const std::unique_ptr<DtmfBuffer> dtmf_buffer_ GUARDED_BY(crit_sect_);
347 const std::unique_ptr<DtmfToneGenerator> dtmf_tone_generator_
henrik.lundin@webrtc.org2f816bb2014-06-05 10:37:13 +0000348 GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800349 const std::unique_ptr<PacketBuffer> packet_buffer_ GUARDED_BY(crit_sect_);
350 const std::unique_ptr<PayloadSplitter> payload_splitter_
kwiberg@webrtc.org00b8f6b2015-02-26 14:34:55 +0000351 GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800352 const std::unique_ptr<TimestampScaler> timestamp_scaler_
kwiberg@webrtc.org00b8f6b2015-02-26 14:34:55 +0000353 GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800354 const std::unique_ptr<PostDecodeVad> vad_ GUARDED_BY(crit_sect_);
355 const std::unique_ptr<ExpandFactory> expand_factory_ GUARDED_BY(crit_sect_);
356 const std::unique_ptr<AccelerateFactory> accelerate_factory_
kwiberg@webrtc.org00b8f6b2015-02-26 14:34:55 +0000357 GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800358 const std::unique_ptr<PreemptiveExpandFactory> preemptive_expand_factory_
henrik.lundin@webrtc.org2f816bb2014-06-05 10:37:13 +0000359 GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000360
kwiberg2d0c3322016-02-14 09:28:33 -0800361 std::unique_ptr<BackgroundNoise> background_noise_ GUARDED_BY(crit_sect_);
362 std::unique_ptr<DecisionLogic> decision_logic_ GUARDED_BY(crit_sect_);
363 std::unique_ptr<AudioMultiVector> algorithm_buffer_ GUARDED_BY(crit_sect_);
364 std::unique_ptr<SyncBuffer> sync_buffer_ GUARDED_BY(crit_sect_);
365 std::unique_ptr<Expand> expand_ GUARDED_BY(crit_sect_);
366 std::unique_ptr<Normal> normal_ GUARDED_BY(crit_sect_);
367 std::unique_ptr<Merge> merge_ GUARDED_BY(crit_sect_);
368 std::unique_ptr<Accelerate> accelerate_ GUARDED_BY(crit_sect_);
369 std::unique_ptr<PreemptiveExpand> preemptive_expand_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000370 RandomVector random_vector_ GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800371 std::unique_ptr<ComfortNoise> comfort_noise_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000372 Rtcp rtcp_ GUARDED_BY(crit_sect_);
373 StatisticsCalculator stats_ GUARDED_BY(crit_sect_);
374 int fs_hz_ GUARDED_BY(crit_sect_);
375 int fs_mult_ GUARDED_BY(crit_sect_);
henrik.lundind89814b2015-11-23 06:49:25 -0800376 int last_output_sample_rate_hz_ GUARDED_BY(crit_sect_);
Peter Kastingdce40cf2015-08-24 14:52:23 -0700377 size_t output_size_samples_ GUARDED_BY(crit_sect_);
378 size_t decoder_frame_length_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000379 Modes last_mode_ GUARDED_BY(crit_sect_);
minyue5bd33972016-05-02 04:46:11 -0700380 Operations last_operation_ GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800381 std::unique_ptr<int16_t[]> mute_factor_array_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000382 size_t decoded_buffer_length_ GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800383 std::unique_ptr<int16_t[]> decoded_buffer_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000384 uint32_t playout_timestamp_ GUARDED_BY(crit_sect_);
385 bool new_codec_ GUARDED_BY(crit_sect_);
386 uint32_t timestamp_ GUARDED_BY(crit_sect_);
387 bool reset_decoder_ GUARDED_BY(crit_sect_);
henrik.lundinda8bbf62016-08-31 03:14:11 -0700388 rtc::Optional<uint8_t> current_rtp_payload_type_ GUARDED_BY(crit_sect_);
389 rtc::Optional<uint8_t> current_cng_rtp_payload_type_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000390 uint32_t ssrc_ GUARDED_BY(crit_sect_);
391 bool first_packet_ GUARDED_BY(crit_sect_);
392 int error_code_ GUARDED_BY(crit_sect_); // Store last error code.
393 int decoder_error_code_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgea257842014-08-07 12:27:37 +0000394 const BackgroundNoiseMode background_noise_mode_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.org7cbc4f92014-10-07 06:37:39 +0000395 NetEqPlayoutMode playout_mode_ GUARDED_BY(crit_sect_);
Henrik Lundincf808d22015-05-27 14:33:29 +0200396 bool enable_fast_accelerate_ GUARDED_BY(crit_sect_);
henrik.lundin91951862016-06-08 06:43:41 -0700397 std::unique_ptr<NackTracker> nack_ GUARDED_BY(crit_sect_);
henrik.lundin48ed9302015-10-29 05:36:24 -0700398 bool nack_enabled_ GUARDED_BY(crit_sect_);
henrik.lundin7a926812016-05-12 13:51:28 -0700399 const bool enable_muted_state_ GUARDED_BY(crit_sect_);
henrik.lundin500c04b2016-03-08 02:36:04 -0800400 AudioFrame::VADActivity last_vad_activity_ GUARDED_BY(crit_sect_) =
401 AudioFrame::kVadPassive;
henrik.lundinb1fb72b2016-05-03 08:18:47 -0700402 std::unique_ptr<TickTimer::Stopwatch> generated_noise_stopwatch_
403 GUARDED_BY(crit_sect_);
minyue@webrtc.orgd7301772013-08-29 00:58:14 +0000404
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000405 private:
henrikg3c089d72015-09-16 05:37:44 -0700406 RTC_DISALLOW_COPY_AND_ASSIGN(NetEqImpl);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000407};
408
409} // namespace webrtc
henrik.lundin@webrtc.org9c55f0f2014-06-09 08:10:28 +0000410#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_