void AudioFileReader::plugDeinterleave(GstPad* pad)
{
// A decodebin pad was added, plug in a deinterleave element to
// separate each planar channel. Sub pipeline looks like
// ... decodebin2 ! audioconvert ! audioresample ! capsfilter ! deinterleave.
GstElement* audioConvert = gst_element_factory_make("audioconvert", 0);
GstElement* audioResample = gst_element_factory_make("audioresample", 0);
GstElement* capsFilter = gst_element_factory_make("capsfilter", 0);
m_deInterleave = gst_element_factory_make("deinterleave", "deinterleave");
g_object_set(m_deInterleave.get(), "keep-positions", TRUE, NULL);
g_signal_connect(m_deInterleave.get(), "pad-added", G_CALLBACK(onGStreamerDeinterleavePadAddedCallback), this);
g_signal_connect(m_deInterleave.get(), "no-more-pads", G_CALLBACK(onGStreamerDeinterleaveReadyCallback), this);
GstCaps* caps = getGstAudioCaps(2, m_sampleRate);
g_object_set(capsFilter, "caps", caps, NULL);
gst_caps_unref(caps);
gst_bin_add_many(GST_BIN(m_pipeline), audioConvert, audioResample, capsFilter, m_deInterleave.get(), NULL);
GstPad* sinkPad = gst_element_get_static_pad(audioConvert, "sink");
gst_pad_link_full(pad, sinkPad, GST_PAD_LINK_CHECK_NOTHING);
gst_object_unref(GST_OBJECT(sinkPad));
gst_element_link_pads_full(audioConvert, "src", audioResample, "sink", GST_PAD_LINK_CHECK_NOTHING);
gst_element_link_pads_full(audioResample, "src", capsFilter, "sink", GST_PAD_LINK_CHECK_NOTHING);
gst_element_link_pads_full(capsFilter, "src", m_deInterleave.get(), "sink", GST_PAD_LINK_CHECK_NOTHING);
gst_element_sync_state_with_parent(audioConvert);
gst_element_sync_state_with_parent(audioResample);
gst_element_sync_state_with_parent(capsFilter);
gst_element_sync_state_with_parent(m_deInterleave.get());
}
//#define AUDIO_FAKE_INPUT
void AudioLiveInputPipeline::buildInputPipeline()
{
// Sub pipeline looks like:
// ... autoaudiosrc ! audioconvert ! capsfilter ! deinterleave.
m_pipeline = gst_pipeline_new("live-input");
#ifndef AUDIO_FAKE_INPUT
// FIXME: Use autoaudiosrc instead of pulsesrc and set properties using
// gstproxychild, as is being done in AudioDestinatio::configureSinkDevice.
GstElement *source = gst_element_factory_make("pulsesrc", "liveinputsrc");
g_object_set(source, "blocksize", (gint64)1024, nullptr);
g_object_set(source, "buffer-time", (gint64) 1451, nullptr);
g_object_set(source, "latency-time", (gint64) 1451, nullptr);
#else
GstElement *source = gst_element_factory_make("audiotestsrc", "fakeinput");
g_object_set(source, "is-live", TRUE, nullptr);
g_object_set(source, "blocksize", 2048, nullptr);
g_object_set(source, "buffer-time", (gint64) 1451, nullptr);
g_object_set(source, "latency-time", (guint64) 1451, nullptr);
#endif
m_source = source;
GstElement* audioConvert = gst_element_factory_make("audioconvert", nullptr);
GstElement* capsFilter = gst_element_factory_make("capsfilter", nullptr);
m_deInterleave = gst_element_factory_make("deinterleave", "deinterleave");
g_object_set(m_deInterleave, "keep-positions", TRUE, nullptr);
g_signal_connect(m_deInterleave, "pad-added", G_CALLBACK(onGStreamerDeinterleavePadAddedCallback), this);
g_signal_connect(m_deInterleave, "no-more-pads", G_CALLBACK(onGStreamerDeinterleaveReadyCallback), this);
GstCaps* caps = getGstAudioCaps(2, m_sampleRate);
g_object_set(capsFilter, "caps", caps, nullptr);
gst_bin_add_many(GST_BIN(m_pipeline), source, audioConvert, capsFilter, m_deInterleave, nullptr);
gst_element_link_pads_full(source, "src", audioConvert, "sink", GST_PAD_LINK_CHECK_NOTHING);
gst_element_link_pads_full(audioConvert, "src", capsFilter, "sink", GST_PAD_LINK_CHECK_NOTHING);
gst_element_link_pads_full(capsFilter, "src", m_deInterleave, "sink", GST_PAD_LINK_CHECK_NOTHING);
GstPad* pad = gst_element_get_static_pad(m_deInterleave, "sink");
gst_pad_set_caps(pad, caps);
m_ready = true;
gst_element_sync_state_with_parent(source);
gst_element_sync_state_with_parent(audioConvert);
gst_element_sync_state_with_parent(capsFilter);
gst_element_sync_state_with_parent(m_deInterleave);
}
