internal Visualization(PlayerEngine.AudioSinkBin audiobin)
{
// The basic pipeline we're constructing is:
// .audiotee ! queue ! audioresample ! audioconvert ! fakesink
Element converter, resampler;
Element audiosinkqueue;
Pad pad;
vis_buffer = null;
vis_fft = new Gst.FFT.FFTF32 (SLICE_SIZE * 2, false);
vis_fft_buffer = new GstFFTF32Complex [SLICE_SIZE + 1];
vis_fft_sample_buffer = new float [SLICE_SIZE];
// Core elements, if something fails here, it's the end of the world
audiosinkqueue = ElementFactory.Make ("queue", "vis-queue");
pad = audiosinkqueue.GetStaticPad ("sink");
pad.AddProbe (PadProbeType.EventDownstream, EventProbe);
resampler = ElementFactory.Make ("audioresample", "vis-resample");
converter = ElementFactory.Make ("audioconvert", "vis-convert");
Element fakesink = ElementFactory.Make ("fakesink", "vis-sink");
if (audiosinkqueue == null || resampler == null || converter == null || fakesink == null) {
Log.Debug ("Could not construct visualization pipeline, a fundamental element could not be created");
return;
}
//http://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-plugins/html/gstreamer-plugins-queue.html#GstQueueLeaky
const int GST_QUEUE_LEAK_DOWNSTREAM = 2;
// Keep around the 5 most recent seconds of audio so that when resuming
// visualization we have something to show right away.
audiosinkqueue ["leaky"] = GST_QUEUE_LEAK_DOWNSTREAM;
audiosinkqueue ["max-size-buffers"] = 0;
audiosinkqueue ["max-size-bytes"] = 0;
audiosinkqueue ["max-size-time"] = ((long)Constants.SECOND) * 5L;
fakesink.Connect ("handoff", PCMHandoff);
// This enables the handoff signal.
fakesink ["signal-handoffs"] = true;
// Synchronize so we see vis at the same time as we hear it.
fakesink ["sync"] = true;
// Drop buffers if they come in too late. This is mainly used when
// thawing the vis pipeline.
fakesink ["max-lateness"] = ((long)Constants.SECOND / 120L);
// Deliver buffers one frame early. This allows for rendering
// time. (TODO: It would be great to calculate this on-the-fly so
// we match the rendering time.
fakesink ["ts-offset"] = -((long)Constants.SECOND / 60L);
// Don't go to PAUSED when we freeze the pipeline.
fakesink ["async"] = false;
audiobin.Add (audiosinkqueue, resampler, converter, fakesink);
pad = audiosinkqueue.GetStaticPad ("sink");
Pad teepad = audiobin.RequestTeePad ();
teepad.Link (pad);
teepad.Dispose ();
pad.Dispose ();
Element.Link (audiosinkqueue, resampler, converter);
converter.LinkFiltered (fakesink, caps);
vis_buffer = new Adapter ();
vis_resampler = resampler;
vis_thawing = false;
active = false;
}
public Visualization (Bin audiobin, Pad teepad)
{
// The basic pipeline we're constructing is:
// .audiotee ! queue ! audioresample ! audioconvert ! fakesink
Element converter, resampler;
Queue audiosinkqueue;
Pad pad;
vis_buffer = null;
vis_fft = gst_fft_f32_new (SLICE_SIZE * 2, false);
vis_fft_buffer = new GstFFTF32Complex [SLICE_SIZE + 1];
vis_fft_sample_buffer = new float [SLICE_SIZE];
// Core elements, if something fails here, it's the end of the world
audiosinkqueue = (Queue)ElementFactory.Make ("queue", "vis-queue");
pad = audiosinkqueue.GetStaticPad ("sink");
pad.AddEventProbe (new PadEventProbeCallback (EventProbe));
resampler = ElementFactory.Make ("audioresample", "vis-resample");
converter = ElementFactory.Make ("audioconvert", "vis-convert");
FakeSink fakesink = ElementFactory.Make ("fakesink", "vis-sink") as FakeSink;
// channels * slice size * float size = size of chunks we want
wanted_size = (uint)(2 * SLICE_SIZE * sizeof(float));
if (audiosinkqueue == null || resampler == null || converter == null || fakesink == null) {
Log.Debug ("Could not construct visualization pipeline, a fundamental element could not be created");
return;
}
// Keep around the 5 most recent seconds of audio so that when resuming
// visualization we have something to show right away.
audiosinkqueue.Leaky = Queue.LeakyType.Downstream;
audiosinkqueue.MaxSizeBuffers = 0;
audiosinkqueue.MaxSizeBytes = 0;
audiosinkqueue.MaxSizeTime = Clock.Second * 5;
fakesink.Handoff += PCMHandoff;
// This enables the handoff signal.
fakesink.SignalHandoffs = true;
// Synchronize so we see vis at the same time as we hear it.
fakesink.Sync = true;
// Drop buffers if they come in too late. This is mainly used when
// thawing the vis pipeline.
fakesink.MaxLateness = (long)(Clock.Second / 120);
// Deliver buffers one frame early. This allows for rendering
// time. (TODO: It would be great to calculate this on-the-fly so
// we match the rendering time.
fakesink.TsOffset = -(long)(Clock.Second / 60);
// Don't go to PAUSED when we freeze the pipeline.
fakesink.Async = false;
audiobin.Add (audiosinkqueue, resampler, converter, fakesink);
pad = audiosinkqueue.GetStaticPad ("sink");
teepad.Link (pad);
Element.Link (audiosinkqueue, resampler, converter);
converter.LinkFiltered (fakesink, caps);
vis_buffer = new Adapter ();
vis_resampler = resampler;
vis_thawing = false;
active = false;
// Disable the pipeline till we hear otherwise from managed land.
Blocked = true;
}
