Webrtc音频编码并打包RTP包

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于 2023-10-23 17:49:41 发布

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文章标签：音视频 webrtc android 音频接收流媒体传输

本文链接：https://blog.csdn.net/xiehuanbin/article/details/133995481

版权

创建G722编码器

rtc::scoped_refptr<AudioEncoderFactory> factory = CreateBuiltinAudioEncoderFactory();
std::unique_ptr<AudioEncoder> encoder = factory->MakeAudioEncoder(9, {"G722", 8000, 1}, absl::nullopt);

初始化参数

int    seqNum    = 0;
qint64 timestamp = 0;
int    ssrc      = 100;

编码操作(对pcm数据进行编码)

rtc::Buffer buffer;
AudioEncoder::EncodedInfo info = 
    encoder->Encode(0,
                rtc::ArrayView<const int16_t>((const int16_t *)pcm, pcmSize/2),
                &buffer);
     
/* 数据有效性判断 */           
if (info.encoder_type == AudioEncoder::CodecType::kOther)
    return;

打包为RTP

// 对于连续的音频包，需要连续的timestamp。
timestamp += sizeof(int16_t) * encoder->NumChannels() * encoder->RtpTimestampRateHz()/100;

/* 创建rtp包 */
RtpPacket packet(nullptr);
packet.SetPayloadType(payloadType);
packet.SetSequenceNumber(seqNum);
packet.SetTimestamp(timestamp);
packet.SetSsrc(ssrc);

uint8_t *payload = packet.SetPayloadSize(buffer.size());
/* 装载rtp包的payload内容 */
memcpy(payload, 
       (const uint8_t *)buffer.data(), 
       buffer.size());