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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
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"
kwiberg65cb70d2017-03-03 06:16:28 -080029#include "webrtc/modules/include/module_common_types.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000030#include "webrtc/typedefs.h"
31
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.lundin1d9061e2016-04-26 12:19:34 -070071 struct Dependencies {
72 // The constructor populates the Dependencies struct with the default
73 // implementations of the objects. They can all be replaced by the user
74 // before sending the struct to the NetEqImpl constructor. However, there
75 // are dependencies between some of the classes inside the struct, so
76 // swapping out one may make it necessary to re-create another one.
ossue3525782016-05-25 07:37:43 -070077 explicit Dependencies(
78 const NetEq::Config& config,
79 const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory);
henrik.lundin1d9061e2016-04-26 12:19:34 -070080 ~Dependencies();
81
82 std::unique_ptr<TickTimer> tick_timer;
83 std::unique_ptr<BufferLevelFilter> buffer_level_filter;
84 std::unique_ptr<DecoderDatabase> decoder_database;
85 std::unique_ptr<DelayPeakDetector> delay_peak_detector;
86 std::unique_ptr<DelayManager> delay_manager;
87 std::unique_ptr<DtmfBuffer> dtmf_buffer;
88 std::unique_ptr<DtmfToneGenerator> dtmf_tone_generator;
89 std::unique_ptr<PacketBuffer> packet_buffer;
ossua70695a2016-09-22 02:06:28 -070090 std::unique_ptr<RedPayloadSplitter> red_payload_splitter;
henrik.lundin1d9061e2016-04-26 12:19:34 -070091 std::unique_ptr<TimestampScaler> timestamp_scaler;
92 std::unique_ptr<AccelerateFactory> accelerate_factory;
93 std::unique_ptr<ExpandFactory> expand_factory;
94 std::unique_ptr<PreemptiveExpandFactory> preemptive_expand_factory;
95 };
96
97 // Creates a new NetEqImpl object.
henrik.lundin@webrtc.orgea257842014-08-07 12:27:37 +000098 NetEqImpl(const NetEq::Config& config,
henrik.lundin1d9061e2016-04-26 12:19:34 -070099 Dependencies&& deps,
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000100 bool create_components = true);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000101
Karl Wiberg7f6c4d42015-04-09 15:44:22 +0200102 ~NetEqImpl() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000103
104 // Inserts a new packet into NetEq. The |receive_timestamp| is an indication
105 // of the time when the packet was received, and should be measured with
106 // the same tick rate as the RTP timestamp of the current payload.
107 // Returns 0 on success, -1 on failure.
Henrik Lundin70c09bd2017-04-24 15:56:56 +0200108 int InsertPacket(const RTPHeader& rtp_header,
kwibergee2bac22015-11-11 10:34:00 -0800109 rtc::ArrayView<const uint8_t> payload,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000110 uint32_t receive_timestamp) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000111
henrik.lundin7a926812016-05-12 13:51:28 -0700112 int GetAudio(AudioFrame* audio_frame, bool* muted) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000113
kwiberg1c07c702017-03-27 07:15:49 -0700114 void SetCodecs(const std::map<int, SdpAudioFormat>& codecs) override;
115
kwibergee1879c2015-10-29 06:20:28 -0700116 int RegisterPayloadType(NetEqDecoder codec,
henrik.lundin4cf61dd2015-12-09 06:20:58 -0800117 const std::string& codec_name,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000118 uint8_t rtp_payload_type) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000119
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000120 int RegisterExternalDecoder(AudioDecoder* decoder,
kwibergee1879c2015-10-29 06:20:28 -0700121 NetEqDecoder codec,
henrik.lundin4cf61dd2015-12-09 06:20:58 -0800122 const std::string& codec_name,
kwiberg342f7402016-06-16 03:18:00 -0700123 uint8_t rtp_payload_type) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000124
kwiberg5adaf732016-10-04 09:33:27 -0700125 bool RegisterPayloadType(int rtp_payload_type,
126 const SdpAudioFormat& audio_format) override;
127
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000128 // Removes |rtp_payload_type| from the codec database. Returns 0 on success,
129 // -1 on failure.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000130 int RemovePayloadType(uint8_t rtp_payload_type) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000131
kwiberg6b19b562016-09-20 04:02:25 -0700132 void RemoveAllPayloadTypes() override;
133
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000134 bool SetMinimumDelay(int delay_ms) override;
turaj@webrtc.orgf1efc572013-08-16 23:44:24 +0000135
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000136 bool SetMaximumDelay(int delay_ms) override;
turaj@webrtc.orgf1efc572013-08-16 23:44:24 +0000137
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000138 int LeastRequiredDelayMs() const override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000139
Karl Wiberg7f6c4d42015-04-09 15:44:22 +0200140 int SetTargetDelay() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000141
Karl Wiberg7f6c4d42015-04-09 15:44:22 +0200142 int TargetDelay() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000143
henrik.lundin9c3efd02015-08-27 13:12:22 -0700144 int CurrentDelayMs() const override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000145
henrik.lundinb3f1c5d2016-08-22 15:39:53 -0700146 int FilteredCurrentDelayMs() const override;
147
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000148 // Sets the playout mode to |mode|.
henrik.lundin@webrtc.org7cbc4f92014-10-07 06:37:39 +0000149 // Deprecated.
150 // TODO(henrik.lundin) Delete.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000151 void SetPlayoutMode(NetEqPlayoutMode mode) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000152
153 // Returns the current playout mode.
henrik.lundin@webrtc.org7cbc4f92014-10-07 06:37:39 +0000154 // Deprecated.
155 // TODO(henrik.lundin) Delete.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000156 NetEqPlayoutMode PlayoutMode() const override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000157
158 // Writes the current network statistics to |stats|. The statistics are reset
159 // after the call.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000160 int NetworkStatistics(NetEqNetworkStatistics* stats) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000161
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000162 // Writes the current RTCP statistics to |stats|. The statistics are reset
163 // and a new report period is started with the call.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000164 void GetRtcpStatistics(RtcpStatistics* stats) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000165
166 // Same as RtcpStatistics(), but does not reset anything.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000167 void GetRtcpStatisticsNoReset(RtcpStatistics* stats) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000168
169 // Enables post-decode VAD. When enabled, GetAudio() will return
170 // kOutputVADPassive when the signal contains no speech.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000171 void EnableVad() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000172
173 // Disables post-decode VAD.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000174 void DisableVad() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000175
henrik.lundin15c51e32016-04-06 08:38:56 -0700176 rtc::Optional<uint32_t> GetPlayoutTimestamp() const override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000177
henrik.lundind89814b2015-11-23 06:49:25 -0800178 int last_output_sample_rate_hz() const override;
179
kwiberg6f0f6162016-09-20 03:07:46 -0700180 rtc::Optional<CodecInst> GetDecoder(int payload_type) const override;
181
ossuf1b08da2016-09-23 02:19:43 -0700182 rtc::Optional<SdpAudioFormat> GetDecoderFormat(
183 int payload_type) const override;
kwibergc4ccd4d2016-09-21 10:55:15 -0700184
Karl Wiberg7f6c4d42015-04-09 15:44:22 +0200185 int SetTargetNumberOfChannels() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000186
Karl Wiberg7f6c4d42015-04-09 15:44:22 +0200187 int SetTargetSampleRate() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000188
189 // Returns the error code for the last occurred error. If no error has
190 // occurred, 0 is returned.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000191 int LastError() const override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000192
193 // Returns the error code last returned by a decoder (audio or comfort noise).
194 // When LastError() returns kDecoderErrorCode or kComfortNoiseErrorCode, check
195 // this method to get the decoder's error code.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000196 int LastDecoderError() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000197
198 // Flushes both the packet buffer and the sync buffer.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000199 void FlushBuffers() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000200
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000201 void PacketBufferStatistics(int* current_num_packets,
202 int* max_num_packets) const override;
turaj@webrtc.org7df97062013-08-02 18:07:13 +0000203
henrik.lundin48ed9302015-10-29 05:36:24 -0700204 void EnableNack(size_t max_nack_list_size) override;
205
206 void DisableNack() override;
207
208 std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const override;
turaj@webrtc.orgff43c852013-09-25 00:07:27 +0000209
henrik.lundin@webrtc.orgb287d962014-04-07 21:21:45 +0000210 // This accessor method is only intended for testing purposes.
henrike@webrtc.org47658f12014-09-10 22:14:59 +0000211 const SyncBuffer* sync_buffer_for_test() const;
minyue5bd33972016-05-02 04:46:11 -0700212 Operations last_operation_for_test() const;
henrik.lundin@webrtc.orgb287d962014-04-07 21:21:45 +0000213
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000214 protected:
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000215 static const int kOutputSizeMs = 10;
minyue5bd33972016-05-02 04:46:11 -0700216 static const size_t kMaxFrameSize = 5760; // 120 ms @ 48 kHz.
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000217 // TODO(hlundin): Provide a better value for kSyncBufferSize.
minyue17461792016-05-03 13:32:05 -0700218 // Current value is kMaxFrameSize + 60 ms * 48 kHz, which is enough for
219 // calculating correlations of current frame against history.
220 static const size_t kSyncBufferSize = kMaxFrameSize + 60 * 48;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000221
222 // Inserts a new packet into NetEq. This is used by the InsertPacket method
223 // above. Returns 0 on success, otherwise an error code.
224 // TODO(hlundin): Merge this with InsertPacket above?
Henrik Lundin70c09bd2017-04-24 15:56:56 +0200225 int InsertPacketInternal(const RTPHeader& rtp_header,
kwibergee2bac22015-11-11 10:34:00 -0800226 rtc::ArrayView<const uint8_t> payload,
ossu17e3fa12016-09-08 04:52:55 -0700227 uint32_t receive_timestamp)
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000228 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000229
henrik.lundin6d8e0112016-03-04 10:34:21 -0800230 // Delivers 10 ms of audio data. The data is written to |audio_frame|.
henrik.lundin@webrtc.orge1d468c2013-01-30 07:37:20 +0000231 // Returns 0 on success, otherwise an error code.
henrik.lundin7a926812016-05-12 13:51:28 -0700232 int GetAudioInternal(AudioFrame* audio_frame, bool* muted)
Peter Kasting69558702016-01-12 16:26:35 -0800233 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000234
235 // Provides a decision to the GetAudioInternal method. The decision what to
236 // do is written to |operation|. Packets to decode are written to
237 // |packet_list|, and a DTMF event to play is written to |dtmf_event|. When
238 // DTMF should be played, |play_dtmf| is set to true by the method.
239 // Returns 0 on success, otherwise an error code.
240 int GetDecision(Operations* operation,
241 PacketList* packet_list,
242 DtmfEvent* dtmf_event,
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000243 bool* play_dtmf) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000244
245 // Decodes the speech packets in |packet_list|, and writes the results to
246 // |decoded_buffer|, which is allocated to hold |decoded_buffer_length|
247 // elements. The length of the decoded data is written to |decoded_length|.
248 // The speech type -- speech or (codec-internal) comfort noise -- is written
249 // to |speech_type|. If |packet_list| contains any SID frames for RFC 3389
250 // comfort noise, those are not decoded.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000251 int Decode(PacketList* packet_list,
252 Operations* operation,
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
minyuel6d92bf52015-09-23 15:20:39 +0200257 // Sub-method to Decode(). Performs codec internal CNG.
258 int DecodeCng(AudioDecoder* decoder, int* decoded_length,
259 AudioDecoder::SpeechType* speech_type)
260 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
261
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000262 // Sub-method to Decode(). Performs the actual decoding.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000263 int DecodeLoop(PacketList* packet_list,
minyuel6d92bf52015-09-23 15:20:39 +0200264 const Operations& operation,
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000265 AudioDecoder* decoder,
266 int* decoded_length,
267 AudioDecoder::SpeechType* speech_type)
268 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000269
270 // Sub-method which calls the Normal class to perform the normal operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000271 void DoNormal(const int16_t* decoded_buffer,
272 size_t decoded_length,
273 AudioDecoder::SpeechType speech_type,
274 bool play_dtmf) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000275
276 // Sub-method which calls the Merge class to perform the merge operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000277 void DoMerge(int16_t* decoded_buffer,
278 size_t decoded_length,
279 AudioDecoder::SpeechType speech_type,
280 bool play_dtmf) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000281
282 // Sub-method which calls the Expand class to perform the expand operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000283 int DoExpand(bool play_dtmf) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000284
285 // Sub-method which calls the Accelerate class to perform the accelerate
286 // operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000287 int DoAccelerate(int16_t* decoded_buffer,
288 size_t decoded_length,
289 AudioDecoder::SpeechType speech_type,
Henrik Lundincf808d22015-05-27 14:33:29 +0200290 bool play_dtmf,
291 bool fast_accelerate) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000292
293 // Sub-method which calls the PreemptiveExpand class to perform the
294 // preemtive expand operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000295 int DoPreemptiveExpand(int16_t* decoded_buffer,
296 size_t decoded_length,
297 AudioDecoder::SpeechType speech_type,
298 bool play_dtmf) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000299
300 // Sub-method which calls the ComfortNoise class to generate RFC 3389 comfort
301 // noise. |packet_list| can either contain one SID frame to update the
302 // noise parameters, or no payload at all, in which case the previously
303 // received parameters are used.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000304 int DoRfc3389Cng(PacketList* packet_list, bool play_dtmf)
305 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000306
307 // Calls the audio decoder to generate codec-internal comfort noise when
308 // no packet was received.
minyuel6d92bf52015-09-23 15:20:39 +0200309 void DoCodecInternalCng(const int16_t* decoded_buffer, size_t decoded_length)
310 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000311
312 // Calls the DtmfToneGenerator class to generate DTMF tones.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000313 int DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf)
314 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000315
316 // Produces packet-loss concealment using alternative methods. If the codec
317 // has an internal PLC, it is called to generate samples. Otherwise, the
318 // method performs zero-stuffing.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000319 void DoAlternativePlc(bool increase_timestamp)
320 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000321
322 // Overdub DTMF on top of |output|.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000323 int DtmfOverdub(const DtmfEvent& dtmf_event,
324 size_t num_channels,
325 int16_t* output) const EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000326
327 // Extracts packets from |packet_buffer_| to produce at least
328 // |required_samples| samples. The packets are inserted into |packet_list|.
329 // Returns the number of samples that the packets in the list will produce, or
330 // -1 in case of an error.
Peter Kastingdce40cf2015-08-24 14:52:23 -0700331 int ExtractPackets(size_t required_samples, PacketList* packet_list)
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000332 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000333
334 // Resets various variables and objects to new values based on the sample rate
335 // |fs_hz| and |channels| number audio channels.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000336 void SetSampleRateAndChannels(int fs_hz, size_t channels)
337 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000338
339 // Returns the output type for the audio produced by the latest call to
340 // GetAudio().
henrik.lundin55480f52016-03-08 02:37:57 -0800341 OutputType LastOutputType() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000342
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000343 // Updates Expand and Merge.
344 virtual void UpdatePlcComponents(int fs_hz, size_t channels)
345 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
346
henrik.lundin@webrtc.org7cbc4f92014-10-07 06:37:39 +0000347 // Creates DecisionLogic object with the mode given by |playout_mode_|.
348 virtual void CreateDecisionLogic() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000349
pbos5ad935c2016-01-25 03:52:44 -0800350 rtc::CriticalSection crit_sect_;
henrik.lundined497212016-04-25 10:11:38 -0700351 const std::unique_ptr<TickTimer> tick_timer_ GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800352 const std::unique_ptr<BufferLevelFilter> buffer_level_filter_
henrik.lundin@webrtc.org2f816bb2014-06-05 10:37:13 +0000353 GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800354 const std::unique_ptr<DecoderDatabase> decoder_database_
henrik.lundin@webrtc.org2f816bb2014-06-05 10:37:13 +0000355 GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800356 const std::unique_ptr<DelayManager> delay_manager_ GUARDED_BY(crit_sect_);
357 const std::unique_ptr<DelayPeakDetector> delay_peak_detector_
henrik.lundin@webrtc.org2f816bb2014-06-05 10:37:13 +0000358 GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800359 const std::unique_ptr<DtmfBuffer> dtmf_buffer_ GUARDED_BY(crit_sect_);
360 const std::unique_ptr<DtmfToneGenerator> dtmf_tone_generator_
henrik.lundin@webrtc.org2f816bb2014-06-05 10:37:13 +0000361 GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800362 const std::unique_ptr<PacketBuffer> packet_buffer_ GUARDED_BY(crit_sect_);
ossua70695a2016-09-22 02:06:28 -0700363 const std::unique_ptr<RedPayloadSplitter> red_payload_splitter_
kwiberg@webrtc.org00b8f6b2015-02-26 14:34:55 +0000364 GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800365 const std::unique_ptr<TimestampScaler> timestamp_scaler_
kwiberg@webrtc.org00b8f6b2015-02-26 14:34:55 +0000366 GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800367 const std::unique_ptr<PostDecodeVad> vad_ GUARDED_BY(crit_sect_);
368 const std::unique_ptr<ExpandFactory> expand_factory_ GUARDED_BY(crit_sect_);
369 const std::unique_ptr<AccelerateFactory> accelerate_factory_
kwiberg@webrtc.org00b8f6b2015-02-26 14:34:55 +0000370 GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800371 const std::unique_ptr<PreemptiveExpandFactory> preemptive_expand_factory_
henrik.lundin@webrtc.org2f816bb2014-06-05 10:37:13 +0000372 GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000373
kwiberg2d0c3322016-02-14 09:28:33 -0800374 std::unique_ptr<BackgroundNoise> background_noise_ GUARDED_BY(crit_sect_);
375 std::unique_ptr<DecisionLogic> decision_logic_ GUARDED_BY(crit_sect_);
376 std::unique_ptr<AudioMultiVector> algorithm_buffer_ GUARDED_BY(crit_sect_);
377 std::unique_ptr<SyncBuffer> sync_buffer_ GUARDED_BY(crit_sect_);
378 std::unique_ptr<Expand> expand_ GUARDED_BY(crit_sect_);
379 std::unique_ptr<Normal> normal_ GUARDED_BY(crit_sect_);
380 std::unique_ptr<Merge> merge_ GUARDED_BY(crit_sect_);
381 std::unique_ptr<Accelerate> accelerate_ GUARDED_BY(crit_sect_);
382 std::unique_ptr<PreemptiveExpand> preemptive_expand_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000383 RandomVector random_vector_ GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800384 std::unique_ptr<ComfortNoise> comfort_noise_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000385 Rtcp rtcp_ GUARDED_BY(crit_sect_);
386 StatisticsCalculator stats_ GUARDED_BY(crit_sect_);
387 int fs_hz_ GUARDED_BY(crit_sect_);
388 int fs_mult_ GUARDED_BY(crit_sect_);
henrik.lundind89814b2015-11-23 06:49:25 -0800389 int last_output_sample_rate_hz_ GUARDED_BY(crit_sect_);
Peter Kastingdce40cf2015-08-24 14:52:23 -0700390 size_t output_size_samples_ GUARDED_BY(crit_sect_);
391 size_t decoder_frame_length_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000392 Modes last_mode_ GUARDED_BY(crit_sect_);
minyue5bd33972016-05-02 04:46:11 -0700393 Operations last_operation_ GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800394 std::unique_ptr<int16_t[]> mute_factor_array_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000395 size_t decoded_buffer_length_ GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800396 std::unique_ptr<int16_t[]> decoded_buffer_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000397 uint32_t playout_timestamp_ GUARDED_BY(crit_sect_);
398 bool new_codec_ GUARDED_BY(crit_sect_);
399 uint32_t timestamp_ GUARDED_BY(crit_sect_);
400 bool reset_decoder_ GUARDED_BY(crit_sect_);
henrik.lundinda8bbf62016-08-31 03:14:11 -0700401 rtc::Optional<uint8_t> current_rtp_payload_type_ GUARDED_BY(crit_sect_);
402 rtc::Optional<uint8_t> current_cng_rtp_payload_type_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000403 uint32_t ssrc_ GUARDED_BY(crit_sect_);
404 bool first_packet_ GUARDED_BY(crit_sect_);
405 int error_code_ GUARDED_BY(crit_sect_); // Store last error code.
406 int decoder_error_code_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgea257842014-08-07 12:27:37 +0000407 const BackgroundNoiseMode background_noise_mode_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.org7cbc4f92014-10-07 06:37:39 +0000408 NetEqPlayoutMode playout_mode_ GUARDED_BY(crit_sect_);
Henrik Lundincf808d22015-05-27 14:33:29 +0200409 bool enable_fast_accelerate_ GUARDED_BY(crit_sect_);
henrik.lundin91951862016-06-08 06:43:41 -0700410 std::unique_ptr<NackTracker> nack_ GUARDED_BY(crit_sect_);
henrik.lundin48ed9302015-10-29 05:36:24 -0700411 bool nack_enabled_ GUARDED_BY(crit_sect_);
henrik.lundin7a926812016-05-12 13:51:28 -0700412 const bool enable_muted_state_ GUARDED_BY(crit_sect_);
henrik.lundin500c04b2016-03-08 02:36:04 -0800413 AudioFrame::VADActivity last_vad_activity_ GUARDED_BY(crit_sect_) =
414 AudioFrame::kVadPassive;
henrik.lundinb1fb72b2016-05-03 08:18:47 -0700415 std::unique_ptr<TickTimer::Stopwatch> generated_noise_stopwatch_
416 GUARDED_BY(crit_sect_);
minyue@webrtc.orgd7301772013-08-29 00:58:14 +0000417
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000418 private:
henrikg3c089d72015-09-16 05:37:44 -0700419 RTC_DISALLOW_COPY_AND_ASSIGN(NetEqImpl);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000420};
421
422} // namespace webrtc
henrik.lundin@webrtc.org9c55f0f2014-06-09 08:10:28 +0000423#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_