1 /* 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 11 #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_INTERFACE_NETEQ_H_ 12 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_INTERFACE_NETEQ_H_ 13 14 #include <string.h> // Provide access to size_t. 15 16 #include <vector> 17 18 #include "webrtc/base/constructormagic.h" 19 #include "webrtc/common_types.h" 20 #include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h" 21 #include "webrtc/typedefs.h" 22 23 namespace webrtc { 24 25 // Forward declarations. 26 struct WebRtcRTPHeader; 27 28 struct NetEqNetworkStatistics { 29 uint16_t current_buffer_size_ms; // Current jitter buffer size in ms. 30 uint16_t preferred_buffer_size_ms; // Target buffer size in ms. 31 uint16_t jitter_peaks_found; // 1 if adding extra delay due to peaky 32 // jitter; 0 otherwise. 33 uint16_t packet_loss_rate; // Loss rate (network + late) in Q14. 34 uint16_t packet_discard_rate; // Late loss rate in Q14. 35 uint16_t expand_rate; // Fraction (of original stream) of synthesized 36 // audio inserted through expansion (in Q14). 37 uint16_t speech_expand_rate; // Fraction (of original stream) of synthesized 38 // speech inserted through expansion (in Q14). 39 uint16_t preemptive_rate; // Fraction of data inserted through pre-emptive 40 // expansion (in Q14). 41 uint16_t accelerate_rate; // Fraction of data removed through acceleration 42 // (in Q14). 43 uint16_t secondary_decoded_rate; // Fraction of data coming from secondary 44 // decoding (in Q14). 45 int32_t clockdrift_ppm; // Average clock-drift in parts-per-million 46 // (positive or negative). 47 int added_zero_samples; // Number of zero samples added in "off" mode. 48 }; 49 50 enum NetEqOutputType { 51 kOutputNormal, 52 kOutputPLC, 53 kOutputCNG, 54 kOutputPLCtoCNG, 55 kOutputVADPassive 56 }; 57 58 enum NetEqPlayoutMode { 59 kPlayoutOn, 60 kPlayoutOff, 61 kPlayoutFax, 62 kPlayoutStreaming 63 }; 64 65 // This is the interface class for NetEq. 66 class NetEq { 67 public: 68 enum BackgroundNoiseMode { 69 kBgnOn, // Default behavior with eternal noise. 70 kBgnFade, // Noise fades to zero after some time. 71 kBgnOff // Background noise is always zero. 72 }; 73 74 struct Config { 75 Config() 76 : sample_rate_hz(16000), 77 enable_audio_classifier(false), 78 max_packets_in_buffer(50), 79 // |max_delay_ms| has the same effect as calling SetMaximumDelay(). 80 max_delay_ms(2000), 81 background_noise_mode(kBgnOff), 82 playout_mode(kPlayoutOn) {} 83 84 int sample_rate_hz; // Initial vale. Will change with input data. 85 bool enable_audio_classifier; 86 int max_packets_in_buffer; 87 int max_delay_ms; 88 BackgroundNoiseMode background_noise_mode; 89 NetEqPlayoutMode playout_mode; 90 }; 91 92 enum ReturnCodes { 93 kOK = 0, 94 kFail = -1, 95 kNotImplemented = -2 96 }; 97 98 enum ErrorCodes { 99 kNoError = 0, 100 kOtherError, 101 kInvalidRtpPayloadType, 102 kUnknownRtpPayloadType, 103 kCodecNotSupported, 104 kDecoderExists, 105 kDecoderNotFound, 106 kInvalidSampleRate, 107 kInvalidPointer, 108 kAccelerateError, 109 kPreemptiveExpandError, 110 kComfortNoiseErrorCode, 111 kDecoderErrorCode, 112 kOtherDecoderError, 113 kInvalidOperation, 114 kDtmfParameterError, 115 kDtmfParsingError, 116 kDtmfInsertError, 117 kStereoNotSupported, 118 kSampleUnderrun, 119 kDecodedTooMuch, 120 kFrameSplitError, 121 kRedundancySplitError, 122 kPacketBufferCorruption, 123 kSyncPacketNotAccepted 124 }; 125 126 // Creates a new NetEq object, with parameters set in |config|. The |config| 127 // object will only have to be valid for the duration of the call to this 128 // method. 129 static NetEq* Create(const NetEq::Config& config); 130 131 virtual ~NetEq() {} 132 133 // Inserts a new packet into NetEq. The |receive_timestamp| is an indication 134 // of the time when the packet was received, and should be measured with 135 // the same tick rate as the RTP timestamp of the current payload. 136 // Returns 0 on success, -1 on failure. 137 virtual int InsertPacket(const WebRtcRTPHeader& rtp_header, 138 const uint8_t* payload, 139 size_t length_bytes, 140 uint32_t receive_timestamp) = 0; 141 142 // Inserts a sync-packet into packet queue. Sync-packets are decoded to 143 // silence and are intended to keep AV-sync intact in an event of long packet 144 // losses when Video NACK is enabled but Audio NACK is not. Clients of NetEq 145 // might insert sync-packet when they observe that buffer level of NetEq is 146 // decreasing below a certain threshold, defined by the application. 147 // Sync-packets should have the same payload type as the last audio payload 148 // type, i.e. they cannot have DTMF or CNG payload type, nor a codec change 149 // can be implied by inserting a sync-packet. 150 // Returns kOk on success, kFail on failure. 151 virtual int InsertSyncPacket(const WebRtcRTPHeader& rtp_header, 152 uint32_t receive_timestamp) = 0; 153 154 // Instructs NetEq to deliver 10 ms of audio data. The data is written to 155 // |output_audio|, which can hold (at least) |max_length| elements. 156 // The number of channels that were written to the output is provided in 157 // the output variable |num_channels|, and each channel contains 158 // |samples_per_channel| elements. If more than one channel is written, 159 // the samples are interleaved. 160 // The speech type is written to |type|, if |type| is not NULL. 161 // Returns kOK on success, or kFail in case of an error. 162 virtual int GetAudio(size_t max_length, int16_t* output_audio, 163 int* samples_per_channel, int* num_channels, 164 NetEqOutputType* type) = 0; 165 166 // Associates |rtp_payload_type| with |codec| and stores the information in 167 // the codec database. Returns 0 on success, -1 on failure. 168 virtual int RegisterPayloadType(enum NetEqDecoder codec, 169 uint8_t rtp_payload_type) = 0; 170 171 // Provides an externally created decoder object |decoder| to insert in the 172 // decoder database. The decoder implements a decoder of type |codec| and 173 // associates it with |rtp_payload_type|. Returns kOK on success, 174 // kFail on failure. 175 virtual int RegisterExternalDecoder(AudioDecoder* decoder, 176 enum NetEqDecoder codec, 177 uint8_t rtp_payload_type) = 0; 178 179 // Removes |rtp_payload_type| from the codec database. Returns 0 on success, 180 // -1 on failure. 181 virtual int RemovePayloadType(uint8_t rtp_payload_type) = 0; 182 183 // Sets a minimum delay in millisecond for packet buffer. The minimum is 184 // maintained unless a higher latency is dictated by channel condition. 185 // Returns true if the minimum is successfully applied, otherwise false is 186 // returned. 187 virtual bool SetMinimumDelay(int delay_ms) = 0; 188 189 // Sets a maximum delay in milliseconds for packet buffer. The latency will 190 // not exceed the given value, even required delay (given the channel 191 // conditions) is higher. Calling this method has the same effect as setting 192 // the |max_delay_ms| value in the NetEq::Config struct. 193 virtual bool SetMaximumDelay(int delay_ms) = 0; 194 195 // The smallest latency required. This is computed bases on inter-arrival 196 // time and internal NetEq logic. Note that in computing this latency none of 197 // the user defined limits (applied by calling setMinimumDelay() and/or 198 // SetMaximumDelay()) are applied. 199 virtual int LeastRequiredDelayMs() const = 0; 200 201 // Not implemented. 202 virtual int SetTargetDelay() = 0; 203 204 // Not implemented. 205 virtual int TargetDelay() = 0; 206 207 // Not implemented. 208 virtual int CurrentDelay() = 0; 209 210 // Sets the playout mode to |mode|. 211 // Deprecated. Set the mode in the Config struct passed to the constructor. 212 // TODO(henrik.lundin) Delete. 213 virtual void SetPlayoutMode(NetEqPlayoutMode mode) = 0; 214 215 // Returns the current playout mode. 216 // Deprecated. 217 // TODO(henrik.lundin) Delete. 218 virtual NetEqPlayoutMode PlayoutMode() const = 0; 219 220 // Writes the current network statistics to |stats|. The statistics are reset 221 // after the call. 222 virtual int NetworkStatistics(NetEqNetworkStatistics* stats) = 0; 223 224 // Writes the last packet waiting times (in ms) to |waiting_times|. The number 225 // of values written is no more than 100, but may be smaller if the interface 226 // is polled again before 100 packets has arrived. 227 virtual void WaitingTimes(std::vector<int>* waiting_times) = 0; 228 229 // Writes the current RTCP statistics to |stats|. The statistics are reset 230 // and a new report period is started with the call. 231 virtual void GetRtcpStatistics(RtcpStatistics* stats) = 0; 232 233 // Same as RtcpStatistics(), but does not reset anything. 234 virtual void GetRtcpStatisticsNoReset(RtcpStatistics* stats) = 0; 235 236 // Enables post-decode VAD. When enabled, GetAudio() will return 237 // kOutputVADPassive when the signal contains no speech. 238 virtual void EnableVad() = 0; 239 240 // Disables post-decode VAD. 241 virtual void DisableVad() = 0; 242 243 // Gets the RTP timestamp for the last sample delivered by GetAudio(). 244 // Returns true if the RTP timestamp is valid, otherwise false. 245 virtual bool GetPlayoutTimestamp(uint32_t* timestamp) = 0; 246 247 // Not implemented. 248 virtual int SetTargetNumberOfChannels() = 0; 249 250 // Not implemented. 251 virtual int SetTargetSampleRate() = 0; 252 253 // Returns the error code for the last occurred error. If no error has 254 // occurred, 0 is returned. 255 virtual int LastError() const = 0; 256 257 // Returns the error code last returned by a decoder (audio or comfort noise). 258 // When LastError() returns kDecoderErrorCode or kComfortNoiseErrorCode, check 259 // this method to get the decoder's error code. 260 virtual int LastDecoderError() = 0; 261 262 // Flushes both the packet buffer and the sync buffer. 263 virtual void FlushBuffers() = 0; 264 265 // Current usage of packet-buffer and it's limits. 266 virtual void PacketBufferStatistics(int* current_num_packets, 267 int* max_num_packets) const = 0; 268 269 // Get sequence number and timestamp of the latest RTP. 270 // This method is to facilitate NACK. 271 virtual int DecodedRtpInfo(int* sequence_number, 272 uint32_t* timestamp) const = 0; 273 274 protected: 275 NetEq() {} 276 277 private: 278 DISALLOW_COPY_AND_ASSIGN(NetEq); 279 }; 280 281 } // namespace webrtc 282 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INTERFACE_NETEQ_H_ 283