经过一番努力,我终于得到了Microsoft.SpeechRecognitionEngine
接受 WAVE 音频流。过程如下:
在 Pi 上,我运行 ffmpeg。我使用此命令流式传输音频
ffmpeg -ac 1 -f alsa -i hw:1,0 -ar 16000 -acodec pcm_s16le -f rtp rtp://XXX.XXX.XXX.XXX:1234
在服务器端,我创建了一个UDPClient
并监听端口 1234。我在单独的线程上接收数据包。首先,我去掉 RTP 标头(标头格式在这里解释 http://en.wikipedia.org/wiki/Real-time_Transport_Protocol#Packet_header)并将有效负载写入特殊流。我不得不使用SpeechStreamer
class 肖恩的回应中描述 https://stackoverflow.com/questions/1682902/streaming-input-to-system-speech-recognition-speechrecognitionengine为了让 SpeechRecognitionEngine 正常工作。它不符合标准Memory Stream
.
我在语音识别方面要做的唯一一件事是将输入设置为音频流而不是默认音频设备。
recognizer.SetInputToAudioStream( rtpClient.AudioStream,
new SpeechAudioFormatInfo(WAVFile.SAMPLE_RATE, AudioBitsPerSample.Sixteen, AudioChannel.Mono));
我还没有对它进行广泛的测试(即让它流几天并看看它是否仍然有效),但我可以将音频样本保存在SpeechRecognized
听起来很棒。我使用的采样率为 16 KHz。我可能会将其降低到 8 KHz 以减少数据传输量,但一旦它成为问题我就会担心。
我还应该提到,响应速度非常快。我可以说出整个句子并在不到一秒的时间内得到答复。 RTP 连接似乎给该过程增加了很少的开销。我必须尝试一个基准测试并将其与仅使用 MIC 输入进行比较。
编辑:这是我的 RTPClient 类。
/// <summary>
/// Connects to an RTP stream and listens for data
/// </summary>
public class RTPClient
{
private const int AUDIO_BUFFER_SIZE = 65536;
private UdpClient client;
private IPEndPoint endPoint;
private SpeechStreamer audioStream;
private bool writeHeaderToConsole = false;
private bool listening = false;
private int port;
private Thread listenerThread;
/// <summary>
/// Returns a reference to the audio stream
/// </summary>
public SpeechStreamer AudioStream
{
get { return audioStream; }
}
/// <summary>
/// Gets whether the client is listening for packets
/// </summary>
public bool Listening
{
get { return listening; }
}
/// <summary>
/// Gets the port the RTP client is listening on
/// </summary>
public int Port
{
get { return port; }
}
/// <summary>
/// RTP Client for receiving an RTP stream containing a WAVE audio stream
/// </summary>
/// <param name="port">The port to listen on</param>
public RTPClient(int port)
{
Console.WriteLine(" [RTPClient] Loading...");
this.port = port;
// Initialize the audio stream that will hold the data
audioStream = new SpeechStreamer(AUDIO_BUFFER_SIZE);
Console.WriteLine(" Done");
}
/// <summary>
/// Creates a connection to the RTP stream
/// </summary>
public void StartClient()
{
// Create new UDP client. The IP end point tells us which IP is sending the data
client = new UdpClient(port);
endPoint = new IPEndPoint(IPAddress.Any, port);
listening = true;
listenerThread = new Thread(ReceiveCallback);
listenerThread.Start();
Console.WriteLine(" [RTPClient] Listening for packets on port " + port + "...");
}
/// <summary>
/// Tells the UDP client to stop listening for packets.
/// </summary>
public void StopClient()
{
// Set the boolean to false to stop the asynchronous packet receiving
listening = false;
Console.WriteLine(" [RTPClient] Stopped listening on port " + port);
}
/// <summary>
/// Handles the receiving of UDP packets from the RTP stream
/// </summary>
/// <param name="ar">Contains packet data</param>
private void ReceiveCallback()
{
// Begin looking for the next packet
while (listening)
{
// Receive packet
byte[] packet = client.Receive(ref endPoint);
// Decode the header of the packet
int version = GetRTPHeaderValue(packet, 0, 1);
int padding = GetRTPHeaderValue(packet, 2, 2);
int extension = GetRTPHeaderValue(packet, 3, 3);
int csrcCount = GetRTPHeaderValue(packet, 4, 7);
int marker = GetRTPHeaderValue(packet, 8, 8);
int payloadType = GetRTPHeaderValue(packet, 9, 15);
int sequenceNum = GetRTPHeaderValue(packet, 16, 31);
int timestamp = GetRTPHeaderValue(packet, 32, 63);
int ssrcId = GetRTPHeaderValue(packet, 64, 95);
if (writeHeaderToConsole)
{
Console.WriteLine("{0} {1} {2} {3} {4} {5} {6} {7} {8}",
version,
padding,
extension,
csrcCount,
marker,
payloadType,
sequenceNum,
timestamp,
ssrcId);
}
// Write the packet to the audio stream
audioStream.Write(packet, 12, packet.Length - 12);
}
}
/// <summary>
/// Grabs a value from the RTP header in Big-Endian format
/// </summary>
/// <param name="packet">The RTP packet</param>
/// <param name="startBit">Start bit of the data value</param>
/// <param name="endBit">End bit of the data value</param>
/// <returns>The value</returns>
private int GetRTPHeaderValue(byte[] packet, int startBit, int endBit)
{
int result = 0;
// Number of bits in value
int length = endBit - startBit + 1;
// Values in RTP header are big endian, so need to do these conversions
for (int i = startBit; i <= endBit; i++)
{
int byteIndex = i / 8;
int bitShift = 7 - (i % 8);
result += ((packet[byteIndex] >> bitShift) & 1) * (int)Math.Pow(2, length - i + startBit - 1);
}
return result;
}
}