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/*
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_
#define MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_
#include <map>
namespace webrtc {
///////////////////////////////////////////////////////////////////////////
// enum ACMVADMode
// An enumerator for aggressiveness of VAD
// -VADNormal : least aggressive mode.
// -VADLowBitrate : more aggressive than "VADNormal" to save on
// bit-rate.
// -VADAggr : an aggressive mode.
// -VADVeryAggr : the most agressive mode.
//
enum ACMVADMode {
VADNormal = 0,
VADLowBitrate = 1,
VADAggr = 2,
VADVeryAggr = 3
};
enum class AudioFrameType {
kEmptyFrame = 0,
kAudioFrameSpeech = 1,
kAudioFrameCN = 2,
};
///////////////////////////////////////////////////////////////////////////
//
// Enumeration of Opus mode for intended application.
//
// kVoip : optimized for voice signals.
// kAudio : optimized for non-voice signals like music.
//
enum OpusApplicationMode {
kVoip = 0,
kAudio = 1,
};
// Statistics for calls to AudioCodingModule::PlayoutData10Ms().
struct AudioDecodingCallStats {
AudioDecodingCallStats()
: calls_to_silence_generator(0),
calls_to_neteq(0),
decoded_normal(0),
decoded_neteq_plc(0),
decoded_codec_plc(0),
decoded_cng(0),
decoded_plc_cng(0),
decoded_muted_output(0) {}
int calls_to_silence_generator; // Number of calls where silence generated,
// and NetEq was disengaged from decoding.
int calls_to_neteq; // Number of calls to NetEq.
int decoded_normal; // Number of calls where audio RTP packet decoded.
int decoded_neteq_plc; // Number of calls resulted in NetEq PLC.
int decoded_codec_plc; // Number of calls resulted in codec PLC.
int decoded_cng; // Number of calls where comfort noise generated due to DTX.
int decoded_plc_cng; // Number of calls resulted where PLC faded to CNG.
int decoded_muted_output; // Number of calls returning a muted state output.
};
// NETEQ statistics.
struct NetworkStatistics {
// current jitter buffer size in ms
uint16_t currentBufferSize;
// preferred (optimal) buffer size in ms
uint16_t preferredBufferSize;
// adding extra delay due to "peaky jitter"
bool jitterPeaksFound;
// Stats below correspond to similarly-named fields in the WebRTC stats spec.
// https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats
uint64_t totalSamplesReceived;
uint64_t concealedSamples;
uint64_t silentConcealedSamples;
uint64_t concealmentEvents;
uint64_t jitterBufferDelayMs;
uint64_t jitterBufferTargetDelayMs;
uint64_t jitterBufferMinimumDelayMs;
uint64_t jitterBufferEmittedCount;
uint64_t insertedSamplesForDeceleration;
uint64_t removedSamplesForAcceleration;
uint64_t fecPacketsReceived;
uint64_t fecPacketsDiscarded;
// Stats below correspond to similarly-named fields in the WebRTC stats spec.
// https://w3c.github.io/webrtc-stats/#dom-rtcreceivedrtpstreamstats
uint64_t packetsDiscarded;
// Stats below DO NOT correspond directly to anything in the WebRTC stats
// fraction (of original stream) of synthesized audio inserted through
// expansion (in Q14)
uint16_t currentExpandRate;
// fraction (of original stream) of synthesized speech inserted through
// expansion (in Q14)
uint16_t currentSpeechExpandRate;
// fraction of synthesized speech inserted through pre-emptive expansion
// (in Q14)
uint16_t currentPreemptiveRate;
// fraction of data removed through acceleration (in Q14)
uint16_t currentAccelerateRate;
// fraction of data coming from secondary decoding (in Q14)
uint16_t currentSecondaryDecodedRate;
// Fraction of secondary data, including FEC and RED, that is discarded (in
// Q14). Discarding of secondary data can be caused by the reception of the
// primary data, obsoleting the secondary data. It can also be caused by early
// or late arrival of secondary data.
uint16_t currentSecondaryDiscardedRate;
// average packet waiting time in the jitter buffer (ms)
int meanWaitingTimeMs;
// max packet waiting time in the jitter buffer (ms)
int maxWaitingTimeMs;
// count of the number of buffer flushes
uint64_t packetBufferFlushes;
// number of samples expanded due to delayed packets
uint64_t delayedPacketOutageSamples;
// arrival delay of incoming packets
uint64_t relativePacketArrivalDelayMs;
// number of audio interruptions
int32_t interruptionCount;
// total duration of audio interruptions
int32_t totalInterruptionDurationMs;
};
} // namespace webrtc
#endif // MODULES_AUDIO_CODING_INCLUDE_AUDIO_CODING_MODULE_TYPEDEFS_H_
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