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273 lines
10 KiB
C++
273 lines
10 KiB
C++
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_INTERFACE_NETEQ_H_
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#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_INTERFACE_NETEQ_H_
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#include <string.h> // Provide access to size_t.
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#include <vector>
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#include "webrtc/base/constructormagic.h"
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#include "webrtc/common_types.h"
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#include "webrtc/modules/audio_coding/neteq/interface/audio_decoder.h"
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#include "webrtc/typedefs.h"
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namespace webrtc {
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// Forward declarations.
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struct WebRtcRTPHeader;
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struct NetEqNetworkStatistics {
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uint16_t current_buffer_size_ms; // Current jitter buffer size in ms.
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uint16_t preferred_buffer_size_ms; // Target buffer size in ms.
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uint16_t jitter_peaks_found; // 1 if adding extra delay due to peaky
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// jitter; 0 otherwise.
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uint16_t packet_loss_rate; // Loss rate (network + late) in Q14.
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uint16_t packet_discard_rate; // Late loss rate in Q14.
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uint16_t expand_rate; // Fraction (of original stream) of synthesized
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// speech inserted through expansion (in Q14).
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uint16_t preemptive_rate; // Fraction of data inserted through pre-emptive
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// expansion (in Q14).
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uint16_t accelerate_rate; // Fraction of data removed through acceleration
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// (in Q14).
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int32_t clockdrift_ppm; // Average clock-drift in parts-per-million
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// (positive or negative).
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int added_zero_samples; // Number of zero samples added in "off" mode.
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};
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enum NetEqOutputType {
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kOutputNormal,
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kOutputPLC,
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kOutputCNG,
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kOutputPLCtoCNG,
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kOutputVADPassive
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};
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enum NetEqPlayoutMode {
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kPlayoutOn,
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kPlayoutOff,
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kPlayoutFax,
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kPlayoutStreaming
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};
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// This is the interface class for NetEq.
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class NetEq {
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public:
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enum BackgroundNoiseMode {
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kBgnOn, // Default behavior with eternal noise.
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kBgnFade, // Noise fades to zero after some time.
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kBgnOff // Background noise is always zero.
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};
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struct Config {
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Config()
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: sample_rate_hz(16000),
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enable_audio_classifier(false),
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max_packets_in_buffer(50),
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// |max_delay_ms| has the same effect as calling SetMaximumDelay().
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max_delay_ms(2000),
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background_noise_mode(kBgnOff) {}
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int sample_rate_hz; // Initial vale. Will change with input data.
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bool enable_audio_classifier;
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int max_packets_in_buffer;
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int max_delay_ms;
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BackgroundNoiseMode background_noise_mode;
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};
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enum ReturnCodes {
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kOK = 0,
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kFail = -1,
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kNotImplemented = -2
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};
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enum ErrorCodes {
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kNoError = 0,
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kOtherError,
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kInvalidRtpPayloadType,
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kUnknownRtpPayloadType,
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kCodecNotSupported,
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kDecoderExists,
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kDecoderNotFound,
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kInvalidSampleRate,
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kInvalidPointer,
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kAccelerateError,
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kPreemptiveExpandError,
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kComfortNoiseErrorCode,
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kDecoderErrorCode,
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kOtherDecoderError,
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kInvalidOperation,
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kDtmfParameterError,
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kDtmfParsingError,
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kDtmfInsertError,
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kStereoNotSupported,
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kSampleUnderrun,
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kDecodedTooMuch,
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kFrameSplitError,
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kRedundancySplitError,
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kPacketBufferCorruption,
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kSyncPacketNotAccepted
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};
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// Creates a new NetEq object, with parameters set in |config|. The |config|
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// object will only have to be valid for the duration of the call to this
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// method.
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static NetEq* Create(const NetEq::Config& config);
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virtual ~NetEq() {}
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// Inserts a new packet into NetEq. The |receive_timestamp| is an indication
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// of the time when the packet was received, and should be measured with
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// the same tick rate as the RTP timestamp of the current payload.
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// Returns 0 on success, -1 on failure.
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virtual int InsertPacket(const WebRtcRTPHeader& rtp_header,
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const uint8_t* payload,
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int length_bytes,
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uint32_t receive_timestamp) = 0;
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// Inserts a sync-packet into packet queue. Sync-packets are decoded to
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// silence and are intended to keep AV-sync intact in an event of long packet
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// losses when Video NACK is enabled but Audio NACK is not. Clients of NetEq
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// might insert sync-packet when they observe that buffer level of NetEq is
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// decreasing below a certain threshold, defined by the application.
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// Sync-packets should have the same payload type as the last audio payload
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// type, i.e. they cannot have DTMF or CNG payload type, nor a codec change
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// can be implied by inserting a sync-packet.
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// Returns kOk on success, kFail on failure.
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virtual int InsertSyncPacket(const WebRtcRTPHeader& rtp_header,
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uint32_t receive_timestamp) = 0;
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// Instructs NetEq to deliver 10 ms of audio data. The data is written to
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// |output_audio|, which can hold (at least) |max_length| elements.
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// The number of channels that were written to the output is provided in
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// the output variable |num_channels|, and each channel contains
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// |samples_per_channel| elements. If more than one channel is written,
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// the samples are interleaved.
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// The speech type is written to |type|, if |type| is not NULL.
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// Returns kOK on success, or kFail in case of an error.
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virtual int GetAudio(size_t max_length, int16_t* output_audio,
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int* samples_per_channel, int* num_channels,
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NetEqOutputType* type) = 0;
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// Associates |rtp_payload_type| with |codec| and stores the information in
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// the codec database. Returns 0 on success, -1 on failure.
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virtual int RegisterPayloadType(enum NetEqDecoder codec,
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uint8_t rtp_payload_type) = 0;
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// Provides an externally created decoder object |decoder| to insert in the
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// decoder database. The decoder implements a decoder of type |codec| and
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// associates it with |rtp_payload_type|. Returns kOK on success,
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// kFail on failure.
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virtual int RegisterExternalDecoder(AudioDecoder* decoder,
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enum NetEqDecoder codec,
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uint8_t rtp_payload_type) = 0;
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// Removes |rtp_payload_type| from the codec database. Returns 0 on success,
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// -1 on failure.
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virtual int RemovePayloadType(uint8_t rtp_payload_type) = 0;
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// Sets a minimum delay in millisecond for packet buffer. The minimum is
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// maintained unless a higher latency is dictated by channel condition.
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// Returns true if the minimum is successfully applied, otherwise false is
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// returned.
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virtual bool SetMinimumDelay(int delay_ms) = 0;
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// Sets a maximum delay in milliseconds for packet buffer. The latency will
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// not exceed the given value, even required delay (given the channel
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// conditions) is higher. Calling this method has the same effect as setting
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// the |max_delay_ms| value in the NetEq::Config struct.
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virtual bool SetMaximumDelay(int delay_ms) = 0;
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// The smallest latency required. This is computed bases on inter-arrival
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// time and internal NetEq logic. Note that in computing this latency none of
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// the user defined limits (applied by calling setMinimumDelay() and/or
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// SetMaximumDelay()) are applied.
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virtual int LeastRequiredDelayMs() const = 0;
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// Not implemented.
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virtual int SetTargetDelay() = 0;
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// Not implemented.
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virtual int TargetDelay() = 0;
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// Not implemented.
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virtual int CurrentDelay() = 0;
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// Sets the playout mode to |mode|.
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virtual void SetPlayoutMode(NetEqPlayoutMode mode) = 0;
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// Returns the current playout mode.
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virtual NetEqPlayoutMode PlayoutMode() const = 0;
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// Writes the current network statistics to |stats|. The statistics are reset
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// after the call.
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virtual int NetworkStatistics(NetEqNetworkStatistics* stats) = 0;
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// Writes the last packet waiting times (in ms) to |waiting_times|. The number
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// of values written is no more than 100, but may be smaller if the interface
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// is polled again before 100 packets has arrived.
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virtual void WaitingTimes(std::vector<int>* waiting_times) = 0;
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// Writes the current RTCP statistics to |stats|. The statistics are reset
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// and a new report period is started with the call.
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virtual void GetRtcpStatistics(RtcpStatistics* stats) = 0;
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// Same as RtcpStatistics(), but does not reset anything.
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virtual void GetRtcpStatisticsNoReset(RtcpStatistics* stats) = 0;
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// Enables post-decode VAD. When enabled, GetAudio() will return
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// kOutputVADPassive when the signal contains no speech.
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virtual void EnableVad() = 0;
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// Disables post-decode VAD.
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virtual void DisableVad() = 0;
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// Gets the RTP timestamp for the last sample delivered by GetAudio().
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// Returns true if the RTP timestamp is valid, otherwise false.
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virtual bool GetPlayoutTimestamp(uint32_t* timestamp) = 0;
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// Not implemented.
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virtual int SetTargetNumberOfChannels() = 0;
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// Not implemented.
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virtual int SetTargetSampleRate() = 0;
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// Returns the error code for the last occurred error. If no error has
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// occurred, 0 is returned.
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virtual int LastError() = 0;
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// Returns the error code last returned by a decoder (audio or comfort noise).
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// When LastError() returns kDecoderErrorCode or kComfortNoiseErrorCode, check
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// this method to get the decoder's error code.
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virtual int LastDecoderError() = 0;
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// Flushes both the packet buffer and the sync buffer.
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virtual void FlushBuffers() = 0;
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// Current usage of packet-buffer and it's limits.
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virtual void PacketBufferStatistics(int* current_num_packets,
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int* max_num_packets) const = 0;
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// Get sequence number and timestamp of the latest RTP.
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// This method is to facilitate NACK.
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virtual int DecodedRtpInfo(int* sequence_number,
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uint32_t* timestamp) const = 0;
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protected:
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NetEq() {}
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private:
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DISALLOW_COPY_AND_ASSIGN(NetEq);
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};
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} // namespace webrtc
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#endif // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INTERFACE_NETEQ_H_
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