I'm implementing a voip application using gstreamer, i use the example of
the rtp in the plugin-good! i want to implement echo cancellation, i
couldn't use the speex echo canceller with gstreamer because the input and
the output are not in the same process. So, i want to use pulse audio to
make echo cancellation? can any one help me how to deal with? the sender
voice is
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");
}
and the receiver is :
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);
i tried to change gstreamer proprietes to pulseaudio server in input and
output
and i used "pactl load-module module-echo-cancel aec_method=adrian" but i
still listen to echo!! any one could help please
thanks!!
