gstreamer pulseaudio 回声消除

Question

我正在使用 gstreamer 实现 voip 应用程序，我使用插件中的 rtp 示例 - 很好！我想实现回声消除，我无法将 speex 回声消除器与 gstreamer 一起使用，因为输入和输出不在同一个进程中。那么，我想使用脉冲音频进行回声消除？有人能帮我处理吗？发送方的声音是

pipeline = gst_pipeline_new (NULL);
  g_assert (pipeline);

  /* the audio capture and format conversion */
  audiosrc = gst_element_factory_make (pulsesrc, "audiosrc");
  g_assert (audiosrc);
  audioconv = gst_element_factory_make ("audioconvert", "audioconv");
  g_assert (audioconv);
  audiores = gst_element_factory_make ("audioresample", "audiores");
  g_assert (audiores);
  /* the encoding and payloading */
  audioenc = gst_element_factory_make (AUDIO_ENC, "audioenc");
  g_assert (audioenc);
  audiopay = gst_element_factory_make (AUDIO_PAY, "audiopay");
  g_assert (audiopay);

  /* add capture and payloading to the pipeline and link */
  gst_bin_add_many (GST_BIN (pipeline), audiosrc, audioconv, audiores,
      audioenc, audiopay, NULL);

  if (!gst_element_link_many (audiosrc, audioconv, audiores, audioenc,
          audiopay, NULL)) {
    g_error ("Failed to link audiosrc, audioconv, audioresample, "
        "audio encoder and audio payloader");
  }

接收者是：

gst_bin_add_many (GST_BIN (pipeline), rtpsrc, rtcpsrc, rtcpsink, NULL);

  /* the depayloading and decoding */
  audiodepay = gst_element_factory_make (AUDIO_DEPAY, "audiodepay");
  g_assert (audiodepay);
  audiodec = gst_element_factory_make (AUDIO_DEC, "audiodec");
  g_assert (audiodec);
  /* the audio playback and format conversion */
  audioconv = gst_element_factory_make ("audioconvert", "audioconv");
  g_assert (audioconv);
  audiores = gst_element_factory_make ("audioresample", "audiores");
  g_assert (audiores);
  audiosink = gst_element_factory_make (pulsesink, "audiosink");
  g_assert (audiosink);

  /* add depayloading and playback to the pipeline and link */
  gst_bin_add_many (GST_BIN (pipeline), audiodepay, audiodec, audioconv,
      audiores, audiosink, NULL);

  res = gst_element_link_many (audiodepay, audiodec, audioconv, audiores,
      audiosink, NULL);
  g_assert (res == TRUE);

我尝试在输入和输出中将 gstreamer 属性更改为 pulseaudio 服务器，并使用了“pactl load-module module-echo-cancel aec_method=adrian”，但我仍然听到回声！！有谁能帮忙，谢谢！！