[Bug 759389] fakesrc ! wasapisink and any other combination raises The stream is in the wrong format

[GStreamer (GNOME Bugzilla)]

https://bugzilla.gnome.org/show_bug.cgi?id=759389

Thomas Roos <thomas at roosesweb.de> changed:

           What    |Removed                     |Added
----------------------------------------------------------------------------
 Attachment #325467|needs-work                  |none
             status|                            |
 Attachment #325467|0                           |1
        is obsolete|                            |

--- Comment #11 from Thomas Roos <thomas at roosesweb.de> ---
Created attachment 325478
  --> https://bugzilla.gnome.org/attachment.cgi?id=325478&action=edit
fix basic stuff - 2nd try

+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&sink_template));

Why this change? It was intentionally changed just a while ago
-> otherwise I will get an warning
"C:/mingw-w64/i686-4.8.1-posix-sjlj-rt_v3-rev2/mingw32/i686-w64-mingw32/include/gstreamer-1.0/gst/gstelement.h:660:25:
note: expected 'struct GstPadTemplate *' but argument is of type 'struct
GstStaticPadTemplat
e *'
 void                    gst_element_class_add_pad_template     
(GstElementClass *klass, GstPadTemplate *templ);"

@@ +111,3 @@
+  if (FAILED(hr)) {
+        GST_ELEMENT_ERROR(self, RESOURCE, OPEN_READ,
+                ("CoInitializeEx: %ld", hr), (NULL));

GST_ELEMENT_ERROR() does not work during init()

-> dont know the correct gst way of printing err msg then

@@ +144,3 @@
+
+  GST_DEBUG ("gst_wasapi_sink_get_caps filter: %s", gst_caps_to_string
(filter));
+  GST_DEBUG ("gst_wasapi_sink_get_caps: %s", gst_caps_to_string (caps));

You need to filter it against the filter caps if filter!=NULL
-> added comment TODO


@@ +155,3 @@
   IAudioClient *client = NULL;

+  GST_DEBUG ("gst_wasapi_sink_open");

Use GST_DEBUG_OBJECT(sink, ...), also in all the other places
-> fixed that

@@ +225,3 @@
+  GST_DEBUG ("gst_wasapi_sink_prepare - buffer time in us: %"G_GUINT64_FORMAT,
spec->buffer_time );
+  GST_DEBUG ("IAudioClient::GetBufferSize (number of audio frames that the
buffer can hold) :%i", NumBufferFrames );
+  newTime =  NumBufferFrames / self->info.channels * self->info.rate / 1000;

You should update spec->buffer_time
-> fixed that

@@ +341,2 @@
+  //TODO: unsure if the calculation is correct
+  hr = IAudioClient_GetCurrentPadding ((IAudioClient*)self->client,
&nbOfSamplesInQueue);

Padding sounds wrong. We're interested in how many samples are currently queued
in WASAPI but not played out yet
-> I know it sounds wrong, but for me description sounds correct
https://msdn.microsoft.com/de-de/library/windows/desktop/dd370868%28v=vs.85%29.aspx

::: sys/wasapi/gstwasapisrc.c
@@ +205,3 @@

+  hr = IAudioClient_Initialize (self->client, AUDCLNT_SHAREMODE_SHARED,
AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
+         spec->buffer_time / 100, 0, (WAVEFORMATEX *) &format, NULL);

In the sink you do * 10, here you do / 100. What is correct now?

-> you're right - microsoft use hns (100ns) as unit

@@ +347,3 @@
+      GST_DEBUG ("gst_wasapi_src_delay");
+
+      hr = IAudioCaptureClient_GetNextPacketSize (self->capture_client,
&nsamples);

Also doesn't seem to be the correct API. Maybe here you could return the
difference between how much data you read and the audio clock position? See old
get_time() function

-> I know it sounds wrong, but for me description sounds correct

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