blob: 9d4a9ff26966c703065c36f4da367e44562cc06f [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.lundined497212016-04-25 10:11:38 -070027#include "webrtc/modules/audio_coding/neteq/tick_timer.h"
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000028#include "webrtc/typedefs.h"
29
30namespace webrtc {
31
32// Forward declarations.
henrik.lundin@webrtc.org40d3fc62013-09-18 12:19:50 +000033class Accelerate;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000034class BackgroundNoise;
35class BufferLevelFilter;
36class ComfortNoise;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000037class DecisionLogic;
38class DecoderDatabase;
39class DelayManager;
40class DelayPeakDetector;
41class DtmfBuffer;
42class DtmfToneGenerator;
43class Expand;
henrik.lundin@webrtc.org40d3fc62013-09-18 12:19:50 +000044class Merge;
henrik.lundin48ed9302015-10-29 05:36:24 -070045class Nack;
henrik.lundin@webrtc.org40d3fc62013-09-18 12:19:50 +000046class Normal;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000047class PacketBuffer;
48class PayloadSplitter;
49class PostDecodeVad;
henrik.lundin@webrtc.org40d3fc62013-09-18 12:19:50 +000050class PreemptiveExpand;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000051class RandomVector;
52class SyncBuffer;
53class TimestampScaler;
henrik.lundin@webrtc.orgd9faa462014-01-14 10:18:45 +000054struct AccelerateFactory;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000055struct DtmfEvent;
henrik.lundin@webrtc.orgd9faa462014-01-14 10:18:45 +000056struct ExpandFactory;
57struct PreemptiveExpandFactory;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000058
59class NetEqImpl : public webrtc::NetEq {
60 public:
henrik.lundin55480f52016-03-08 02:37:57 -080061 enum class OutputType {
62 kNormalSpeech,
63 kPLC,
64 kCNG,
65 kPLCCNG,
66 kVadPassive
67 };
68
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000069 // Creates a new NetEqImpl object. The object will assume ownership of all
70 // injected dependencies, and will delete them when done.
henrik.lundin@webrtc.orgea257842014-08-07 12:27:37 +000071 NetEqImpl(const NetEq::Config& config,
henrik.lundined497212016-04-25 10:11:38 -070072 std::unique_ptr<TickTimer> tick_timer,
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000073 BufferLevelFilter* buffer_level_filter,
74 DecoderDatabase* decoder_database,
75 DelayManager* delay_manager,
76 DelayPeakDetector* delay_peak_detector,
77 DtmfBuffer* dtmf_buffer,
78 DtmfToneGenerator* dtmf_tone_generator,
79 PacketBuffer* packet_buffer,
80 PayloadSplitter* payload_splitter,
henrik.lundin@webrtc.orgd9faa462014-01-14 10:18:45 +000081 TimestampScaler* timestamp_scaler,
82 AccelerateFactory* accelerate_factory,
83 ExpandFactory* expand_factory,
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +000084 PreemptiveExpandFactory* preemptive_expand_factory,
85 bool create_components = true);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000086
Karl Wiberg7f6c4d42015-04-09 15:44:22 +020087 ~NetEqImpl() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000088
89 // Inserts a new packet into NetEq. The |receive_timestamp| is an indication
90 // of the time when the packet was received, and should be measured with
91 // the same tick rate as the RTP timestamp of the current payload.
92 // Returns 0 on success, -1 on failure.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000093 int InsertPacket(const WebRtcRTPHeader& rtp_header,
kwibergee2bac22015-11-11 10:34:00 -080094 rtc::ArrayView<const uint8_t> payload,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +000095 uint32_t receive_timestamp) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +000096
turaj@webrtc.org7b75ac62013-09-26 00:27:56 +000097 // Inserts a sync-packet into packet queue. Sync-packets are decoded to
98 // silence and are intended to keep AV-sync intact in an event of long packet
99 // losses when Video NACK is enabled but Audio NACK is not. Clients of NetEq
100 // might insert sync-packet when they observe that buffer level of NetEq is
101 // decreasing below a certain threshold, defined by the application.
102 // Sync-packets should have the same payload type as the last audio payload
103 // type, i.e. they cannot have DTMF or CNG payload type, nor a codec change
104 // can be implied by inserting a sync-packet.
105 // Returns kOk on success, kFail on failure.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000106 int InsertSyncPacket(const WebRtcRTPHeader& rtp_header,
107 uint32_t receive_timestamp) override;
turaj@webrtc.org7b75ac62013-09-26 00:27:56 +0000108
henrik.lundin55480f52016-03-08 02:37:57 -0800109 int GetAudio(AudioFrame* audio_frame) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000110
kwibergee1879c2015-10-29 06:20:28 -0700111 int RegisterPayloadType(NetEqDecoder codec,
henrik.lundin4cf61dd2015-12-09 06:20:58 -0800112 const std::string& codec_name,
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000113 uint8_t rtp_payload_type) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000114
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000115 int RegisterExternalDecoder(AudioDecoder* decoder,
kwibergee1879c2015-10-29 06:20:28 -0700116 NetEqDecoder codec,
henrik.lundin4cf61dd2015-12-09 06:20:58 -0800117 const std::string& codec_name,
Karl Wibergd8399e62015-05-25 14:39:56 +0200118 uint8_t rtp_payload_type,
119 int sample_rate_hz) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000120
121 // Removes |rtp_payload_type| from the codec database. Returns 0 on success,
122 // -1 on failure.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000123 int RemovePayloadType(uint8_t rtp_payload_type) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000124
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000125 bool SetMinimumDelay(int delay_ms) override;
turaj@webrtc.orgf1efc572013-08-16 23:44:24 +0000126
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000127 bool SetMaximumDelay(int delay_ms) override;
turaj@webrtc.orgf1efc572013-08-16 23:44:24 +0000128
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000129 int LeastRequiredDelayMs() const override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000130
Karl Wiberg7f6c4d42015-04-09 15:44:22 +0200131 int SetTargetDelay() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000132
Karl Wiberg7f6c4d42015-04-09 15:44:22 +0200133 int TargetDelay() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000134
henrik.lundin9c3efd02015-08-27 13:12:22 -0700135 int CurrentDelayMs() const override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000136
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000137 // Sets the playout mode to |mode|.
henrik.lundin@webrtc.org7cbc4f92014-10-07 06:37:39 +0000138 // Deprecated.
139 // TODO(henrik.lundin) Delete.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000140 void SetPlayoutMode(NetEqPlayoutMode mode) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000141
142 // Returns the current playout mode.
henrik.lundin@webrtc.org7cbc4f92014-10-07 06:37:39 +0000143 // Deprecated.
144 // TODO(henrik.lundin) Delete.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000145 NetEqPlayoutMode PlayoutMode() const override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000146
147 // Writes the current network statistics to |stats|. The statistics are reset
148 // after the call.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000149 int NetworkStatistics(NetEqNetworkStatistics* stats) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000150
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000151 // Writes the current RTCP statistics to |stats|. The statistics are reset
152 // and a new report period is started with the call.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000153 void GetRtcpStatistics(RtcpStatistics* stats) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000154
155 // Same as RtcpStatistics(), but does not reset anything.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000156 void GetRtcpStatisticsNoReset(RtcpStatistics* stats) override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000157
158 // Enables post-decode VAD. When enabled, GetAudio() will return
159 // kOutputVADPassive when the signal contains no speech.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000160 void EnableVad() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000161
162 // Disables post-decode VAD.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000163 void DisableVad() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000164
henrik.lundin15c51e32016-04-06 08:38:56 -0700165 rtc::Optional<uint32_t> GetPlayoutTimestamp() const override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000166
henrik.lundind89814b2015-11-23 06:49:25 -0800167 int last_output_sample_rate_hz() const override;
168
Karl Wiberg7f6c4d42015-04-09 15:44:22 +0200169 int SetTargetNumberOfChannels() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000170
Karl Wiberg7f6c4d42015-04-09 15:44:22 +0200171 int SetTargetSampleRate() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000172
173 // Returns the error code for the last occurred error. If no error has
174 // occurred, 0 is returned.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000175 int LastError() const override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000176
177 // Returns the error code last returned by a decoder (audio or comfort noise).
178 // When LastError() returns kDecoderErrorCode or kComfortNoiseErrorCode, check
179 // this method to get the decoder's error code.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000180 int LastDecoderError() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000181
182 // Flushes both the packet buffer and the sync buffer.
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000183 void FlushBuffers() override;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000184
kjellander@webrtc.org14665ff2015-03-04 12:58:35 +0000185 void PacketBufferStatistics(int* current_num_packets,
186 int* max_num_packets) const override;
turaj@webrtc.org7df97062013-08-02 18:07:13 +0000187
henrik.lundin48ed9302015-10-29 05:36:24 -0700188 void EnableNack(size_t max_nack_list_size) override;
189
190 void DisableNack() override;
191
192 std::vector<uint16_t> GetNackList(int64_t round_trip_time_ms) const override;
turaj@webrtc.orgff43c852013-09-25 00:07:27 +0000193
henrik.lundin@webrtc.orgb287d962014-04-07 21:21:45 +0000194 // This accessor method is only intended for testing purposes.
henrike@webrtc.org47658f12014-09-10 22:14:59 +0000195 const SyncBuffer* sync_buffer_for_test() const;
henrik.lundin@webrtc.orgb287d962014-04-07 21:21:45 +0000196
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000197 protected:
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000198 static const int kOutputSizeMs = 10;
Peter Kastingdce40cf2015-08-24 14:52:23 -0700199 static const size_t kMaxFrameSize = 2880; // 60 ms @ 48 kHz.
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000200 // TODO(hlundin): Provide a better value for kSyncBufferSize.
Peter Kastingdce40cf2015-08-24 14:52:23 -0700201 static const size_t kSyncBufferSize = 2 * kMaxFrameSize;
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000202
203 // Inserts a new packet into NetEq. This is used by the InsertPacket method
204 // above. Returns 0 on success, otherwise an error code.
205 // TODO(hlundin): Merge this with InsertPacket above?
206 int InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
kwibergee2bac22015-11-11 10:34:00 -0800207 rtc::ArrayView<const uint8_t> payload,
turaj@webrtc.org7b75ac62013-09-26 00:27:56 +0000208 uint32_t receive_timestamp,
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000209 bool is_sync_packet)
210 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000211
henrik.lundin6d8e0112016-03-04 10:34:21 -0800212 // Delivers 10 ms of audio data. The data is written to |audio_frame|.
henrik.lundin@webrtc.orge1d468c2013-01-30 07:37:20 +0000213 // Returns 0 on success, otherwise an error code.
henrik.lundin6d8e0112016-03-04 10:34:21 -0800214 int GetAudioInternal(AudioFrame* audio_frame)
Peter Kasting69558702016-01-12 16:26:35 -0800215 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000216
217 // Provides a decision to the GetAudioInternal method. The decision what to
218 // do is written to |operation|. Packets to decode are written to
219 // |packet_list|, and a DTMF event to play is written to |dtmf_event|. When
220 // DTMF should be played, |play_dtmf| is set to true by the method.
221 // Returns 0 on success, otherwise an error code.
222 int GetDecision(Operations* operation,
223 PacketList* packet_list,
224 DtmfEvent* dtmf_event,
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000225 bool* play_dtmf) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000226
227 // Decodes the speech packets in |packet_list|, and writes the results to
228 // |decoded_buffer|, which is allocated to hold |decoded_buffer_length|
229 // elements. The length of the decoded data is written to |decoded_length|.
230 // The speech type -- speech or (codec-internal) comfort noise -- is written
231 // to |speech_type|. If |packet_list| contains any SID frames for RFC 3389
232 // comfort noise, those are not decoded.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000233 int Decode(PacketList* packet_list,
234 Operations* operation,
235 int* decoded_length,
236 AudioDecoder::SpeechType* speech_type)
237 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000238
minyuel6d92bf52015-09-23 15:20:39 +0200239 // Sub-method to Decode(). Performs codec internal CNG.
240 int DecodeCng(AudioDecoder* decoder, int* decoded_length,
241 AudioDecoder::SpeechType* speech_type)
242 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
243
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000244 // Sub-method to Decode(). Performs the actual decoding.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000245 int DecodeLoop(PacketList* packet_list,
minyuel6d92bf52015-09-23 15:20:39 +0200246 const Operations& operation,
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000247 AudioDecoder* decoder,
248 int* decoded_length,
249 AudioDecoder::SpeechType* speech_type)
250 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000251
252 // Sub-method which calls the Normal class to perform the normal operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000253 void DoNormal(const int16_t* decoded_buffer,
254 size_t decoded_length,
255 AudioDecoder::SpeechType speech_type,
256 bool play_dtmf) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000257
258 // Sub-method which calls the Merge class to perform the merge operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000259 void DoMerge(int16_t* decoded_buffer,
260 size_t decoded_length,
261 AudioDecoder::SpeechType speech_type,
262 bool play_dtmf) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000263
264 // Sub-method which calls the Expand class to perform the expand operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000265 int DoExpand(bool play_dtmf) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000266
267 // Sub-method which calls the Accelerate class to perform the accelerate
268 // operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000269 int DoAccelerate(int16_t* decoded_buffer,
270 size_t decoded_length,
271 AudioDecoder::SpeechType speech_type,
Henrik Lundincf808d22015-05-27 14:33:29 +0200272 bool play_dtmf,
273 bool fast_accelerate) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000274
275 // Sub-method which calls the PreemptiveExpand class to perform the
276 // preemtive expand operation.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000277 int DoPreemptiveExpand(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 ComfortNoise class to generate RFC 3389 comfort
283 // noise. |packet_list| can either contain one SID frame to update the
284 // noise parameters, or no payload at all, in which case the previously
285 // received parameters are used.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000286 int DoRfc3389Cng(PacketList* packet_list, bool play_dtmf)
287 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000288
289 // Calls the audio decoder to generate codec-internal comfort noise when
290 // no packet was received.
minyuel6d92bf52015-09-23 15:20:39 +0200291 void DoCodecInternalCng(const int16_t* decoded_buffer, size_t decoded_length)
292 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000293
294 // Calls the DtmfToneGenerator class to generate DTMF tones.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000295 int DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf)
296 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000297
298 // Produces packet-loss concealment using alternative methods. If the codec
299 // has an internal PLC, it is called to generate samples. Otherwise, the
300 // method performs zero-stuffing.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000301 void DoAlternativePlc(bool increase_timestamp)
302 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000303
304 // Overdub DTMF on top of |output|.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000305 int DtmfOverdub(const DtmfEvent& dtmf_event,
306 size_t num_channels,
307 int16_t* output) const EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000308
309 // Extracts packets from |packet_buffer_| to produce at least
310 // |required_samples| samples. The packets are inserted into |packet_list|.
311 // Returns the number of samples that the packets in the list will produce, or
312 // -1 in case of an error.
Peter Kastingdce40cf2015-08-24 14:52:23 -0700313 int ExtractPackets(size_t required_samples, PacketList* packet_list)
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000314 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000315
316 // Resets various variables and objects to new values based on the sample rate
317 // |fs_hz| and |channels| number audio channels.
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000318 void SetSampleRateAndChannels(int fs_hz, size_t channels)
319 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000320
321 // Returns the output type for the audio produced by the latest call to
322 // GetAudio().
henrik.lundin55480f52016-03-08 02:37:57 -0800323 OutputType LastOutputType() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000324
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000325 // Updates Expand and Merge.
326 virtual void UpdatePlcComponents(int fs_hz, size_t channels)
327 EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
328
henrik.lundin@webrtc.org7cbc4f92014-10-07 06:37:39 +0000329 // Creates DecisionLogic object with the mode given by |playout_mode_|.
330 virtual void CreateDecisionLogic() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000331
pbos5ad935c2016-01-25 03:52:44 -0800332 rtc::CriticalSection crit_sect_;
henrik.lundined497212016-04-25 10:11:38 -0700333 const std::unique_ptr<TickTimer> tick_timer_ GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800334 const std::unique_ptr<BufferLevelFilter> buffer_level_filter_
henrik.lundin@webrtc.org2f816bb2014-06-05 10:37:13 +0000335 GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800336 const std::unique_ptr<DecoderDatabase> decoder_database_
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<DelayManager> delay_manager_ GUARDED_BY(crit_sect_);
339 const std::unique_ptr<DelayPeakDetector> delay_peak_detector_
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<DtmfBuffer> dtmf_buffer_ GUARDED_BY(crit_sect_);
342 const std::unique_ptr<DtmfToneGenerator> dtmf_tone_generator_
henrik.lundin@webrtc.org2f816bb2014-06-05 10:37:13 +0000343 GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800344 const std::unique_ptr<PacketBuffer> packet_buffer_ GUARDED_BY(crit_sect_);
345 const std::unique_ptr<PayloadSplitter> payload_splitter_
kwiberg@webrtc.org00b8f6b2015-02-26 14:34:55 +0000346 GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800347 const std::unique_ptr<TimestampScaler> timestamp_scaler_
kwiberg@webrtc.org00b8f6b2015-02-26 14:34:55 +0000348 GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800349 const std::unique_ptr<PostDecodeVad> vad_ GUARDED_BY(crit_sect_);
350 const std::unique_ptr<ExpandFactory> expand_factory_ GUARDED_BY(crit_sect_);
351 const std::unique_ptr<AccelerateFactory> accelerate_factory_
kwiberg@webrtc.org00b8f6b2015-02-26 14:34:55 +0000352 GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800353 const std::unique_ptr<PreemptiveExpandFactory> preemptive_expand_factory_
henrik.lundin@webrtc.org2f816bb2014-06-05 10:37:13 +0000354 GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000355
kwiberg2d0c3322016-02-14 09:28:33 -0800356 std::unique_ptr<BackgroundNoise> background_noise_ GUARDED_BY(crit_sect_);
357 std::unique_ptr<DecisionLogic> decision_logic_ GUARDED_BY(crit_sect_);
358 std::unique_ptr<AudioMultiVector> algorithm_buffer_ GUARDED_BY(crit_sect_);
359 std::unique_ptr<SyncBuffer> sync_buffer_ GUARDED_BY(crit_sect_);
360 std::unique_ptr<Expand> expand_ GUARDED_BY(crit_sect_);
361 std::unique_ptr<Normal> normal_ GUARDED_BY(crit_sect_);
362 std::unique_ptr<Merge> merge_ GUARDED_BY(crit_sect_);
363 std::unique_ptr<Accelerate> accelerate_ GUARDED_BY(crit_sect_);
364 std::unique_ptr<PreemptiveExpand> preemptive_expand_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000365 RandomVector random_vector_ GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800366 std::unique_ptr<ComfortNoise> comfort_noise_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000367 Rtcp rtcp_ GUARDED_BY(crit_sect_);
368 StatisticsCalculator stats_ GUARDED_BY(crit_sect_);
369 int fs_hz_ GUARDED_BY(crit_sect_);
370 int fs_mult_ GUARDED_BY(crit_sect_);
henrik.lundind89814b2015-11-23 06:49:25 -0800371 int last_output_sample_rate_hz_ GUARDED_BY(crit_sect_);
Peter Kastingdce40cf2015-08-24 14:52:23 -0700372 size_t output_size_samples_ GUARDED_BY(crit_sect_);
373 size_t decoder_frame_length_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000374 Modes last_mode_ GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800375 std::unique_ptr<int16_t[]> mute_factor_array_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000376 size_t decoded_buffer_length_ GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800377 std::unique_ptr<int16_t[]> decoded_buffer_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgdcc301b2014-03-18 11:49:22 +0000378 uint32_t playout_timestamp_ GUARDED_BY(crit_sect_);
379 bool new_codec_ GUARDED_BY(crit_sect_);
380 uint32_t timestamp_ GUARDED_BY(crit_sect_);
381 bool reset_decoder_ GUARDED_BY(crit_sect_);
382 uint8_t current_rtp_payload_type_ GUARDED_BY(crit_sect_);
383 uint8_t current_cng_rtp_payload_type_ GUARDED_BY(crit_sect_);
384 uint32_t ssrc_ GUARDED_BY(crit_sect_);
385 bool first_packet_ GUARDED_BY(crit_sect_);
386 int error_code_ GUARDED_BY(crit_sect_); // Store last error code.
387 int decoder_error_code_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.orgea257842014-08-07 12:27:37 +0000388 const BackgroundNoiseMode background_noise_mode_ GUARDED_BY(crit_sect_);
henrik.lundin@webrtc.org7cbc4f92014-10-07 06:37:39 +0000389 NetEqPlayoutMode playout_mode_ GUARDED_BY(crit_sect_);
Henrik Lundincf808d22015-05-27 14:33:29 +0200390 bool enable_fast_accelerate_ GUARDED_BY(crit_sect_);
kwiberg2d0c3322016-02-14 09:28:33 -0800391 std::unique_ptr<Nack> nack_ GUARDED_BY(crit_sect_);
henrik.lundin48ed9302015-10-29 05:36:24 -0700392 bool nack_enabled_ GUARDED_BY(crit_sect_);
henrik.lundin500c04b2016-03-08 02:36:04 -0800393 AudioFrame::VADActivity last_vad_activity_ GUARDED_BY(crit_sect_) =
394 AudioFrame::kVadPassive;
minyue@webrtc.orgd7301772013-08-29 00:58:14 +0000395
turaj@webrtc.org8d1cdaa2014-04-11 18:47:55 +0000396 private:
henrikg3c089d72015-09-16 05:37:44 -0700397 RTC_DISALLOW_COPY_AND_ASSIGN(NetEqImpl);
henrik.lundin@webrtc.orgd94659d2013-01-29 12:09:21 +0000398};
399
400} // namespace webrtc
henrik.lundin@webrtc.org9c55f0f2014-06-09 08:10:28 +0000401#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_