捕获来自Ekiga softphone / Pulseaudio JACK接收器的传入音频(Capturing incoming audio from Ekiga softphone/Pulseaudio JACK sink)

我想用Sphinx进行语音识别。

我正在寻找使用Java或Python捕获Ekiga VOIP软电话的输出/传入音频并将其传递给Sphinx。 现在，输出指向PulseAudio JACK Sink。

对不起，如果我没有使用正确的术语，因为我是一个相当的菜鸟。

如果你能指出我正确的方向，那将非常感谢，谢谢。

I want to do speech recognition using Sphinx.

I'm looking to capture the output/incoming audio of the Ekiga VOIP softphone using Java or Python and pass it on to Sphinx. Right now, the output is directed toward the PulseAudio JACK Sink.

Sorry if I am not using the right terminology as I am quite a noob at this.

If you could point me in the right direction, it would be much appreciated, thanks.

最满意答案

您需要正确地格式化要实现的任务，并为此选择正确的工具。 软件开发人员不会将像Ekiga这样的桌面应用程序用于他们的项目。 这只是因为桌面应用程序是为用户编写的，而不是软件开发人员。 您只能在源代码中修改Ekiga等桌面应用程序，但这是一个完全不同的故事。

如果您想拥有VOIP端点并将其连接到语音识别服务，您需要查看IVR系统和类似的东西。 IVR系统是通过VOIP协议实现复杂交互的特定工具。 一些例子是：

FreeSWITCH的

星号

GNU Gatekeeper - 一个项目基于与Ekiga相同的Opal技术

您可以设置freeswitch等待调用并将它们传递给pcoketsphinx。 你可以用Asterisk做同样的事情。 有关详细信息，请参阅

如何使用Freeswitch的Pocketpshinx

如何使用UniMRCP项目将Pocketsphinx集成到Asterisk中

You need to properly formalate the task you want to achive and select the right tools for that. Software developers do not use desktop applications like Ekiga for their projects. It's just because desktop applications are written for the users, they are not for software developers. You can only modify desktop applications like Ekiga in the source code, but that is a whole different story.

If you want to have a VOIP endpoint and connect it to a speech recognition service, you need to look on IVR systems and similar things. IVR system is a specific tool to implement complex interactions over the VOIP protocols. Some examples are:

Freeswitch

Asterisk

GNU Gatekeeper - a project is built on the same Opal technology as Ekiga

You can setup freeswitch to wait for the calls and pass them to the pcoketsphinx. You can do the same with Asterisk. For more details see

How to use Pocketpshinx from Freeswitch

How to integrate Pocketsphinx in Asterisk using UniMRCP project

捕获来自Ekiga softphone / Pulseaudio JACK接收器的传入音频(Capturing incoming audio from Ekiga softphone/Pulseaudio JACK sink)

我想用Sphinx进行语音识别。

我正在寻找使用Java或Python捕获Ekiga VOIP软电话的输出/传入音频并将其传递给Sphinx。 现在，输出指向PulseAudio JACK Sink。

对不起，如果我没有使用正确的术语，因为我是一个相当的菜鸟。

如果你能指出我正确的方向，那将非常感谢，谢谢。

I want to do speech recognition using Sphinx.

I'm looking to capture the output/incoming audio of the Ekiga VOIP softphone using Java or Python and pass it on to Sphinx. Right now, the output is directed toward the PulseAudio JACK Sink.

Sorry if I am not using the right terminology as I am quite a noob at this.

If you could point me in the right direction, it would be much appreciated, thanks.

最满意答案

您需要正确地格式化要实现的任务，并为此选择正确的工具。 软件开发人员不会将像Ekiga这样的桌面应用程序用于他们的项目。 这只是因为桌面应用程序是为用户编写的，而不是软件开发人员。 您只能在源代码中修改Ekiga等桌面应用程序，但这是一个完全不同的故事。

如果您想拥有VOIP端点并将其连接到语音识别服务，您需要查看IVR系统和类似的东西。 IVR系统是通过VOIP协议实现复杂交互的特定工具。 一些例子是：

FreeSWITCH的

星号

GNU Gatekeeper - 一个项目基于与Ekiga相同的Opal技术

您可以设置freeswitch等待调用并将它们传递给pcoketsphinx。 你可以用Asterisk做同样的事情。 有关详细信息，请参阅

如何使用Freeswitch的Pocketpshinx

如何使用UniMRCP项目将Pocketsphinx集成到Asterisk中

You need to properly formalate the task you want to achive and select the right tools for that. Software developers do not use desktop applications like Ekiga for their projects. It's just because desktop applications are written for the users, they are not for software developers. You can only modify desktop applications like Ekiga in the source code, but that is a whole different story.

If you want to have a VOIP endpoint and connect it to a speech recognition service, you need to look on IVR systems and similar things. IVR system is a specific tool to implement complex interactions over the VOIP protocols. Some examples are:

Freeswitch

Asterisk

GNU Gatekeeper - a project is built on the same Opal technology as Ekiga

You can setup freeswitch to wait for the calls and pass them to the pcoketsphinx. You can do the same with Asterisk. For more details see

How to use Pocketpshinx from Freeswitch

How to integrate Pocketsphinx in Asterisk using UniMRCP project