2 * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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.
11 #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ4_INTERFACE_NETEQ_H_
12 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ4_INTERFACE_NETEQ_H_
14 #include <string.h> // Provide access to size_t.
18 #include "webrtc/common_types.h"
19 #include "webrtc/modules/audio_coding/neteq4/interface/audio_decoder.h"
20 #include "webrtc/system_wrappers/interface/constructor_magic.h"
21 #include "webrtc/typedefs.h"
25 // Forward declarations.
26 struct WebRtcRTPHeader;
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 // speech inserted through expansion (in Q14).
37 uint16_t preemptive_rate; // Fraction of data inserted through pre-emptive
38 // expansion (in Q14).
39 uint16_t accelerate_rate; // Fraction of data removed through acceleration
41 int32_t clockdrift_ppm; // Average clock-drift in parts-per-million
42 // (positive or negative).
43 int added_zero_samples; // Number of zero samples added in "off" mode.
46 enum NetEqOutputType {
54 enum NetEqPlayoutMode {
61 enum NetEqBackgroundNoiseMode {
62 kBgnOn, // Default behavior with eternal noise.
63 kBgnFade, // Noise fades to zero after some time.
64 kBgnOff // Background noise is always zero.
67 // This is the interface class for NetEq.
72 : sample_rate_hz(16000),
73 enable_audio_classifier(false),
74 max_packets_in_buffer(50),
75 // |max_delay_ms| has the same effect as calling SetMaximumDelay().
78 int sample_rate_hz; // Initial vale. Will change with input data.
79 bool enable_audio_classifier;
80 int max_packets_in_buffer;
93 kInvalidRtpPayloadType,
94 kUnknownRtpPayloadType,
101 kPreemptiveExpandError,
102 kComfortNoiseErrorCode,
113 kRedundancySplitError,
114 kPacketBufferCorruption,
115 kSyncPacketNotAccepted
118 // Creates a new NetEq object, with parameters set in |config|. The |config|
119 // object will only have to be valid for the duration of the call to this
121 static NetEq* Create(const NetEq::Config& config);
125 // Inserts a new packet into NetEq. The |receive_timestamp| is an indication
126 // of the time when the packet was received, and should be measured with
127 // the same tick rate as the RTP timestamp of the current payload.
128 // Returns 0 on success, -1 on failure.
129 virtual int InsertPacket(const WebRtcRTPHeader& rtp_header,
130 const uint8_t* payload,
132 uint32_t receive_timestamp) = 0;
134 // Inserts a sync-packet into packet queue. Sync-packets are decoded to
135 // silence and are intended to keep AV-sync intact in an event of long packet
136 // losses when Video NACK is enabled but Audio NACK is not. Clients of NetEq
137 // might insert sync-packet when they observe that buffer level of NetEq is
138 // decreasing below a certain threshold, defined by the application.
139 // Sync-packets should have the same payload type as the last audio payload
140 // type, i.e. they cannot have DTMF or CNG payload type, nor a codec change
141 // can be implied by inserting a sync-packet.
142 // Returns kOk on success, kFail on failure.
143 virtual int InsertSyncPacket(const WebRtcRTPHeader& rtp_header,
144 uint32_t receive_timestamp) = 0;
146 // Instructs NetEq to deliver 10 ms of audio data. The data is written to
147 // |output_audio|, which can hold (at least) |max_length| elements.
148 // The number of channels that were written to the output is provided in
149 // the output variable |num_channels|, and each channel contains
150 // |samples_per_channel| elements. If more than one channel is written,
151 // the samples are interleaved.
152 // The speech type is written to |type|, if |type| is not NULL.
153 // Returns kOK on success, or kFail in case of an error.
154 virtual int GetAudio(size_t max_length, int16_t* output_audio,
155 int* samples_per_channel, int* num_channels,
156 NetEqOutputType* type) = 0;
158 // Associates |rtp_payload_type| with |codec| and stores the information in
159 // the codec database. Returns 0 on success, -1 on failure.
160 virtual int RegisterPayloadType(enum NetEqDecoder codec,
161 uint8_t rtp_payload_type) = 0;
163 // Provides an externally created decoder object |decoder| to insert in the
164 // decoder database. The decoder implements a decoder of type |codec| and
165 // associates it with |rtp_payload_type|. Returns kOK on success,
167 virtual int RegisterExternalDecoder(AudioDecoder* decoder,
168 enum NetEqDecoder codec,
169 uint8_t rtp_payload_type) = 0;
171 // Removes |rtp_payload_type| from the codec database. Returns 0 on success,
173 virtual int RemovePayloadType(uint8_t rtp_payload_type) = 0;
175 // Sets a minimum delay in millisecond for packet buffer. The minimum is
176 // maintained unless a higher latency is dictated by channel condition.
177 // Returns true if the minimum is successfully applied, otherwise false is
179 virtual bool SetMinimumDelay(int delay_ms) = 0;
181 // Sets a maximum delay in milliseconds for packet buffer. The latency will
182 // not exceed the given value, even required delay (given the channel
183 // conditions) is higher. Calling this method has the same effect as setting
184 // the |max_delay_ms| value in the NetEq::Config struct.
185 virtual bool SetMaximumDelay(int delay_ms) = 0;
187 // The smallest latency required. This is computed bases on inter-arrival
188 // time and internal NetEq logic. Note that in computing this latency none of
189 // the user defined limits (applied by calling setMinimumDelay() and/or
190 // SetMaximumDelay()) are applied.
191 virtual int LeastRequiredDelayMs() const = 0;
194 virtual int SetTargetDelay() = 0;
197 virtual int TargetDelay() = 0;
200 virtual int CurrentDelay() = 0;
202 // Sets the playout mode to |mode|.
203 virtual void SetPlayoutMode(NetEqPlayoutMode mode) = 0;
205 // Returns the current playout mode.
206 virtual NetEqPlayoutMode PlayoutMode() const = 0;
208 // Writes the current network statistics to |stats|. The statistics are reset
210 virtual int NetworkStatistics(NetEqNetworkStatistics* stats) = 0;
212 // Writes the last packet waiting times (in ms) to |waiting_times|. The number
213 // of values written is no more than 100, but may be smaller if the interface
214 // is polled again before 100 packets has arrived.
215 virtual void WaitingTimes(std::vector<int>* waiting_times) = 0;
217 // Writes the current RTCP statistics to |stats|. The statistics are reset
218 // and a new report period is started with the call.
219 virtual void GetRtcpStatistics(RtcpStatistics* stats) = 0;
221 // Same as RtcpStatistics(), but does not reset anything.
222 virtual void GetRtcpStatisticsNoReset(RtcpStatistics* stats) = 0;
224 // Enables post-decode VAD. When enabled, GetAudio() will return
225 // kOutputVADPassive when the signal contains no speech.
226 virtual void EnableVad() = 0;
228 // Disables post-decode VAD.
229 virtual void DisableVad() = 0;
231 // Returns the RTP timestamp for the last sample delivered by GetAudio().
232 virtual uint32_t PlayoutTimestamp() = 0;
235 virtual int SetTargetNumberOfChannels() = 0;
238 virtual int SetTargetSampleRate() = 0;
240 // Returns the error code for the last occurred error. If no error has
241 // occurred, 0 is returned.
242 virtual int LastError() = 0;
244 // Returns the error code last returned by a decoder (audio or comfort noise).
245 // When LastError() returns kDecoderErrorCode or kComfortNoiseErrorCode, check
246 // this method to get the decoder's error code.
247 virtual int LastDecoderError() = 0;
249 // Flushes both the packet buffer and the sync buffer.
250 virtual void FlushBuffers() = 0;
252 // Current usage of packet-buffer and it's limits.
253 virtual void PacketBufferStatistics(int* current_num_packets,
254 int* max_num_packets) const = 0;
256 // Get sequence number and timestamp of the latest RTP.
257 // This method is to facilitate NACK.
258 virtual int DecodedRtpInfo(int* sequence_number,
259 uint32_t* timestamp) const = 0;
261 // Sets the background noise mode.
262 virtual void SetBackgroundNoiseMode(NetEqBackgroundNoiseMode mode) = 0;
264 // Gets the background noise mode.
265 virtual NetEqBackgroundNoiseMode BackgroundNoiseMode() const = 0;
271 DISALLOW_COPY_AND_ASSIGN(NetEq);
274 } // namespace webrtc
275 #endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ4_INTERFACE_NETEQ_H_