Webrtc Audio Engine

音频

AudioEngine Class

.
├── voe_audio_processing.h
├── voe_base.h
├── voe_codec.h
├── voe_dtmf.h
├── voe_errors.h
├── voe_external_media.h
├── voe_file.h
├── voe_hardware.h
├── voe_neteq_stats.h
├── voe_network.h
├── voe_rtp_rtcp.h
├── voe_video_sync.h
└── voe_volume_control.h

基础功能 Base

• 使能VoIP双向通话，并不默认使用G711编码[Enables full duplex VoIP sessions via RTP using G.711 (mu-Law or A-Law)].
• 初始化和终止 [Initialization and termination].
• 获取追踪信息 [Trace information on text files or via callbacks].
• 多路混音 [Multi-channel support (mixing, sending to multiple destinations etc.)].

语音处理 Audio Processing

• 降噪 Noise Suppression (NS).
• 增益控制 [Automatic Gain Control (AGC)].
• 回声消除 [Echo Control (EC)].
• RX部分的AGC，VAD，降噪 [Receiving side VAD, NS and AGC].
• 语音，噪音，回声等级的实时检测 [Measurements of instantaneous speech, noise and echo levels].
• 生成保存调试信息 [Generation of AP debug recordings].
• 检测键盘输入控制语音 [Detection of keyboard typing which can disrupt a voice conversation].

支持各个模块相应的配置。

音量控制

• 喇叭音量控制 Speaker volume controls.
• 麦克风音量控制 Microphone volume control.
• 非线性语音控制 Non-linear speech level control.
• 静音控制 Mute functions.
• Additional stereo scaling methods.

网络部分

配置网络接口发送接口 （回调形式），接收packet接口。

RTP/RTCP 配置

• 当网络上SSRC/CSRC变化，产生回调 [Callbacks for RTP and RTCP events such as modified SSRC or CSRC].
• SSRC 处理 [SSRC handling].
• 获取RTCP信息 [Transmission of RTCP sender reports].
• 获取远端RTCP数据 [Obtaining RTCP data from incoming RTCP sender reports].
• RTP and RTCP statistics (jitter, packet loss, RTT etc.).
• Redundant Coding (RED)
• Writing RTP and RTCP packets to binary files for off-line analysis of
  the call quality.

配置编解码

• 支持非默认编码 [Support of non-default codecs (e.g. iLBC, iSAC, etc.)].
• 支持每个通道上的静音检测 [Voice Activity Detection (VAD) on a per channel basis].
• 配置特定编码的为特定payload。[Possibility to specify how to map received payload types to codecs.]

备注 ： 有接口更改payload，码率等信息。

硬件接口

• Audio device handling.
• Device information.
• CPU load monitoring.

备注： 提供音频设备接口切换，设备ID等, 硬件AEC AGC NS 开关接口。

获取NetEQ信息

struct NetworkStatistics           // NETEQ statistics
{
    // 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;
    // Loss rate (network + late); fraction between 0 and 1, scaled to Q14.
    uint16_t currentPacketLossRate;
    // Late loss rate; fraction between 0 and 1, scaled to Q14.
    uint16_t currentDiscardRate;
    // 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;
    // clock-drift in parts-per-million (negative or positive)
    int32_t clockDriftPPM;
    // average packet waiting time in the jitter buffer (ms)
    int meanWaitingTimeMs;
    // median packet waiting time in the jitter buffer (ms)
    int medianWaitingTimeMs;
    // min packet waiting time in the jitter buffer (ms)
    int minWaitingTimeMs;
    // max packet waiting time in the jitter buffer (ms)
    int maxWaitingTimeMs;
    // added samples in off mode due to packet loss
    size_t addedSamples;
};

RFC2833 DTMF

• Telephone event transmission.
• DTMF tone generation.

备注： 支持RFC2833 配置发送。一个接口就搞定。

音视频同步

• RTP header modification (time stamp and sequence number fields).
• Playout delay tuning to synchronize the voice with video.
• Playout delay monitoring.

外部模块添加

需要额外添加部分

• 提供回调函数，可以实现以下几个点添加音频处理模块.

    kPlaybackPerChannel = 0,         //  channel playback
    kPlaybackAllChannelsMixed,       //  混音 playback
    kRecordingPerChannel,            //  channel recorder
    kRecordingAllChannelsMixed,      //  混音 record
    kRecordingPreprocessing          //  Preprocessing  record

音频数据保存播放

从用文件音频数据，替代设备采集的数据发送出去，以及将RX数据保存到文件中。方便调试。

• File playback.
• File recording.
• File conversion.

备注 ： 支持回调，或保存文件等各种形式。

出错信息

// Warnings
#define VE_PORT_NOT_DEFINED 8001
#define VE_CHANNEL_NOT_VALID 8002
#define VE_FUNC_NOT_SUPPORTED 8003
#define VE_INVALID_LISTNR 8004
#define VE_INVALID_ARGUMENT 8005
#define VE_INVALID_PORT_NMBR 8006
#define VE_INVALID_PLNAME 8007
#define VE_INVALID_PLFREQ 8008
#define VE_INVALID_PLTYPE 8009
#define VE_INVALID_PACSIZE 8010
.....