directsoundsrc: port to 1.0

https://bugzilla.gnome.org/show_bug.cgi?id=673414

1 files changed, 78 insertions, 127 deletions

diff --git a/sys/directsound/gstdirectsoundsrc.c b/sys/directsound/gstdirectsoundsrc.c
index 169073b88..a372681d5 100644
--- a/sys/directsound/gstdirectsoundsrc.c
+++ b/sys/directsound/gstdirectsoundsrc.c
@@ -48,12 +48,25 @@
   TODO: add device selection and check rate etc.
 */
 
+/**
+ * SECTION:element-directsoundsrc
+ *
+ * Reads audio data using the DirectSound API.
+ *
+ * <refsect2>
+ * <title>Example pipelines</title>
+ * |[
+ * gst-launch -v directsoundsrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=dsound.ogg
+ * ]| Record from DirectSound and encode to Ogg/Vorbis.
+ * </refsect2>
+ */
+
 #ifdef HAVE_CONFIG_H
 #  include "config.h"
 #endif
 
 #include <gst/gst.h>
-#include <gst/audio/gstbaseaudiosrc.h>
+#include <gst/audio/gstaudiobasesrc.h>
 
 #include "gstdirectsoundsrc.h"
 
@@ -73,7 +86,6 @@ enum
   PROP_DEVICE
 };
 
-
 static HRESULT (WINAPI * pDSoundCaptureCreate) (LPGUID,
     LPDIRECTSOUNDCAPTURE *, LPUNKNOWN);
 
@@ -88,47 +100,30 @@ static void gst_directsound_src_get_property (GObject * object,
 static gboolean gst_directsound_src_open (GstAudioSrc * asrc);
 static gboolean gst_directsound_src_close (GstAudioSrc * asrc);
 static gboolean gst_directsound_src_prepare (GstAudioSrc * asrc,
-    GstRingBufferSpec * spec);
+    GstAudioRingBufferSpec * spec);
 static gboolean gst_directsound_src_unprepare (GstAudioSrc * asrc);
 static void gst_directsound_src_reset (GstAudioSrc * asrc);
-static GstCaps *gst_directsound_src_getcaps (GstBaseSrc * bsrc);
+static GstCaps *gst_directsound_src_getcaps (GstBaseSrc * bsrc,
+    GstCaps * filter);
 
 static guint gst_directsound_src_read (GstAudioSrc * asrc,
     gpointer data, guint length);
 
 static void gst_directsound_src_dispose (GObject * object);
 
-static void gst_directsound_src_do_init (GType type);
-
 static guint gst_directsound_src_delay (GstAudioSrc * asrc);
 
 static GstStaticPadTemplate directsound_src_src_factory =
-    GST_STATIC_PAD_TEMPLATE ("src",
+GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_SRC,
     GST_PAD_ALWAYS,
-    GST_STATIC_CAPS ("audio/x-raw-int, "
-        "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, "
-        "signed = (boolean) { TRUE, FALSE }, "
-        "width = (int) 16, "
-        "depth = (int) 16, "
-        "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]; "
-        "audio/x-raw-int, "
-        "signed = (boolean) { TRUE, FALSE }, "
-        "width = (int) 8, "
-        "depth = (int) 8, "
+    GST_STATIC_CAPS ("audio/x-raw, "
+        "format = (string) { S16LE, S8 }, "
+        "layout = (string) interleaved, "
         "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]"));
 
-static void
-gst_directsound_src_do_init (GType type)
-{
-  GST_DEBUG_CATEGORY_INIT (directsoundsrc_debug, "directsoundsrc", 0,
-      "DirectSound Src");
-
-
-}
-
-GST_BOILERPLATE_FULL (GstDirectSoundSrc, gst_directsound_src, GstAudioSrc,
-    GST_TYPE_AUDIO_SRC, gst_directsound_src_do_init);
+#define gst_directsound_src_parent_class parent_class
+G_DEFINE_TYPE (GstDirectSoundSrc, gst_directsound_src, GST_TYPE_AUDIO_SRC);
 
 static void
 gst_directsound_src_dispose (GObject * object)
@@ -147,39 +142,19 @@ gst_directsound_src_finalize (GObject * object)
 }
 
 static void
-gst_directsound_src_base_init (gpointer g_class)
-{
-  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
-
-  GST_DEBUG ("initializing directsoundsrc base\n");
-
-  gst_element_class_set_metadata (element_class, "Direct Sound Audio Src",
-      "Source/Audio",
-      "Capture from a soundcard via DIRECTSOUND",
-      "Joni Valtanen <joni.valtanen@movial.fi>");
-
-  gst_element_class_add_pad_template (element_class,
-      gst_static_pad_template_get (&directsound_src_src_factory));
-}
-
-
-/* initialize the plugin's class */
-static void
 gst_directsound_src_class_init (GstDirectSoundSrcClass * klass)
 {
   GObjectClass *gobject_class;
   GstElementClass *gstelement_class;
   GstBaseSrcClass *gstbasesrc_class;
-  GstBaseAudioSrcClass *gstbaseaudiosrc_class;
   GstAudioSrcClass *gstaudiosrc_class;
 
   gobject_class = (GObjectClass *) klass;
   gstelement_class = (GstElementClass *) klass;
   gstbasesrc_class = (GstBaseSrcClass *) klass;
-  gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass;
   gstaudiosrc_class = (GstAudioSrcClass *) klass;
 
-  GST_DEBUG ("initializing directsoundsrc class\n");
+  GST_DEBUG ("initializing directsoundsrc class");
 
   gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_directsound_src_finalize);
   gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_directsound_src_dispose);
@@ -199,23 +174,30 @@ gst_directsound_src_class_init (GstDirectSoundSrcClass * klass)
   gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_directsound_src_delay);
   gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_directsound_src_reset);
 
+  GST_DEBUG_CATEGORY_INIT (directsoundsrc_debug, "directsoundsrc", 0,
+      "DirectSound Src");
 
+  gst_element_class_set_static_metadata (gstelement_class,
+      "DirectSound audio source", "Source/Audio",
+      "Capture from a soundcard via DirectSound",
+      "Joni Valtanen <joni.valtanen@movial.fi>");
+
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&directsound_src_src_factory));
 }
 
 static GstCaps *
-gst_directsound_src_getcaps (GstBaseSrc * bsrc)
+gst_directsound_src_getcaps (GstBaseSrc * bsrc, GstCaps * filter)
 {
   GstDirectSoundSrc *dsoundsrc;
   GstCaps *caps = NULL;
-  GST_DEBUG ("get caps\n");
+  GST_DEBUG_OBJECT (bsrc, "get caps");
 
   dsoundsrc = GST_DIRECTSOUND_SRC (bsrc);
 
-
   caps = gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_SRC_PAD
           (bsrc)));
   return caps;
-
 }
 
 static void
@@ -223,7 +205,7 @@ gst_directsound_src_set_property (GObject * object, guint prop_id,
     const GValue * value, GParamSpec * pspec)
 {
   //  GstDirectSoundSrc *src = GST_DIRECTSOUND_SRC (object);
-  GST_DEBUG ("set property\n");
+  GST_DEBUG ("set property");
 
   switch (prop_id) {
 #if 0
@@ -246,7 +228,7 @@ gst_directsound_src_get_property (GObject * object, guint prop_id,
   GstDirectSoundSrc *src = GST_DIRECTSOUND_SRC (object);
 #endif
 
-  GST_DEBUG ("get property\n");
+  GST_DEBUG ("get property");
 
   switch (prop_id) {
 #if 0
@@ -268,22 +250,19 @@ gst_directsound_src_get_property (GObject * object, guint prop_id,
  * initialize structure
  */
 static void
-gst_directsound_src_init (GstDirectSoundSrc * src,
-    GstDirectSoundSrcClass * gclass)
+gst_directsound_src_init (GstDirectSoundSrc * src)
 {
-  GST_DEBUG ("initializing directsoundsrc\n");
+  GST_DEBUG_OBJECT (src, "initializing directsoundsrc");
   src->dsound_lock = g_mutex_new ();
 }
 
-
-
 static gboolean
 gst_directsound_src_open (GstAudioSrc * asrc)
 {
   GstDirectSoundSrc *dsoundsrc;
   HRESULT hRes;                 /* Result for windows functions */
 
-  GST_DEBUG ("initializing directsoundsrc\n");
+  GST_DEBUG_OBJECT (asrc, "opening directsoundsrc");
 
   dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
 
@@ -342,7 +321,7 @@ gst_directsound_src_close (GstAudioSrc * asrc)
   GstDirectSoundSrc *dsoundsrc;
   HRESULT hRes;                 /* Result for windows functions */
 
-  GST_DEBUG ("initializing directsoundsrc\n");
+  GST_DEBUG_OBJECT (asrc, "closing directsoundsrc");
 
   dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
 
@@ -356,49 +335,37 @@ gst_directsound_src_close (GstAudioSrc * asrc)
 }
 
 static gboolean
-gst_directsound_src_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
+gst_directsound_src_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
 {
-
   GstDirectSoundSrc *dsoundsrc;
   WAVEFORMATEX wfx;             /* Wave format structure */
   HRESULT hRes;                 /* Result for windows functions */
-  DSCBUFFERDESC descSecondary;  /* Capturebuffer decsiption */
+  DSCBUFFERDESC descSecondary;  /* Capturebuffer description */
 
   dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
 
-  GST_DEBUG ("initializing directsoundsrc\n");
+  GST_DEBUG_OBJECT (asrc, "preparing directsoundsrc");
 
   /* Define buffer */
   memset (&wfx, 0, sizeof (WAVEFORMATEX));
-  wfx.wFormatTag = WAVE_FORMAT_PCM;     /* should be WAVE_FORMAT_PCM */
-  wfx.nChannels = spec->channels;
-  wfx.nSamplesPerSec = spec->rate;      /* 8000|11025|22050|44100 */
-  wfx.wBitsPerSample = spec->width;     // 8|16;
-
-  wfx.nBlockAlign = wfx.nChannels * (wfx.wBitsPerSample / 8);
+  wfx.wFormatTag = WAVE_FORMAT_PCM;
+  wfx.nChannels = GST_AUDIO_INFO_CHANNELS (&spec->info);
+  wfx.nSamplesPerSec = GST_AUDIO_INFO_RATE (&spec->info);
+  wfx.wBitsPerSample = GST_AUDIO_INFO_BPF (&spec->info) * 8 / wfx.nChannels;
+  wfx.nBlockAlign = GST_AUDIO_INFO_BPF (&spec->info);
   wfx.nAvgBytesPerSec = wfx.nSamplesPerSec * wfx.nBlockAlign;
-  wfx.cbSize = 0;               /* This size is allways for PCM-format */
+  /* Ignored for WAVE_FORMAT_PCM. */
+  wfx.cbSize = 0;
 
-  /* 1 or 2 Channels etc...
-     FIXME: Never really tested. Is this ok?
-   */
-  if (spec->width == 16 && spec->channels == 1) {
-    spec->format = GST_S16_LE;
-  } else if (spec->width == 16 && spec->channels == 2) {
-    spec->format = GST_U16_LE;
-  } else if (spec->width == 8 && spec->channels == 1) {
-    spec->format = GST_S8;
-  } else if (spec->width == 8 && spec->channels == 2) {
-    spec->format = GST_U8;
-  }
+  if (wfx.wBitsPerSample != 16 && wfx.wBitsPerSample != 8)
+    goto dodgy_width;
 
   /* Set the buffer size to two seconds. 
      This should never reached. 
    */
   dsoundsrc->buffer_size = wfx.nAvgBytesPerSec * 2;
 
-  //notifysize * 16; //spec->width; /*original 16*/
-  GST_DEBUG ("Buffer size: %d", dsoundsrc->buffer_size);
+  GST_DEBUG_OBJECT (asrc, "Buffer size: %d", dsoundsrc->buffer_size);
 
   /* Init secondary buffer desciption */
   memset (&descSecondary, 0, sizeof (DSCBUFFERDESC));
@@ -413,19 +380,15 @@ gst_directsound_src_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
   /* Create buffer */
   hRes = IDirectSoundCapture_CreateCaptureBuffer (dsoundsrc->pDSC,
       &descSecondary, &dsoundsrc->pDSBSecondary, NULL);
-  if (hRes != DS_OK) {
+  if (hRes != DS_OK)
     goto capture_buffer;
-  }
 
-  spec->channels = wfx.nChannels;
-  spec->rate = wfx.nSamplesPerSec;
-  spec->bytes_per_sample = (spec->width / 8) * spec->channels;
-  dsoundsrc->bytes_per_sample = spec->bytes_per_sample;
+  dsoundsrc->bytes_per_sample = GST_AUDIO_INFO_BPF (&spec->info);
 
   GST_DEBUG ("latency time: %" G_GUINT64_FORMAT " - buffer time: %"
       G_GUINT64_FORMAT, spec->latency_time, spec->buffer_time);
 
-  /* Buffer-time should be allways more than 2*latency */
+  /* Buffer-time should be always more than 2*latency */
   if (spec->buffer_time < spec->latency_time * 2) {
     spec->buffer_time = spec->latency_time * 2;
     GST_WARNING ("buffer-time was less than latency");
@@ -438,7 +401,6 @@ gst_directsound_src_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
   dsoundsrc->latency_size = (gint) wfx.nAvgBytesPerSec *
       dsoundsrc->latency_time / 1000000.0;
 
-
   spec->segsize = (guint) (((double) spec->buffer_time / 1000000.0) *
       wfx.nAvgBytesPerSec);
 
@@ -446,27 +408,23 @@ gst_directsound_src_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
   if (spec->segsize < 1)
     spec->segsize = 1;
 
-  spec->segtotal = spec->width * (wfx.nAvgBytesPerSec / spec->segsize);
-
-  GST_DEBUG ("bytes/sec: %lu, buffer size: %d, segsize: %d, segtotal: %d",
-      wfx.nAvgBytesPerSec,
-      dsoundsrc->buffer_size, spec->segsize, spec->segtotal);
-
-  spec->silence_sample[0] = 0;
-  spec->silence_sample[1] = 0;
-  spec->silence_sample[2] = 0;
-  spec->silence_sample[3] = 0;
+  spec->segtotal = GST_AUDIO_INFO_BPF (&spec->info) * 8 *
+      (wfx.nAvgBytesPerSec / spec->segsize);
 
-  if (spec->width != 16 && spec->width != 8)
-    goto dodgy_width;
+  GST_DEBUG_OBJECT (asrc,
+      "bytes/sec: %lu, buffer size: %d, segsize: %d, segtotal: %d",
+      wfx.nAvgBytesPerSec, dsoundsrc->buffer_size, spec->segsize,
+      spec->segtotal);
 
-  /* Not readed anything yet */
+  /* Not read anything yet */
   dsoundsrc->current_circular_offset = 0;
 
-  GST_DEBUG ("GstRingBufferSpec->channels: %d, GstRingBufferSpec->rate: %d, \
-GstRingBufferSpec->bytes_per_sample: %d\n\
-WAVEFORMATEX.nSamplesPerSec: %ld, WAVEFORMATEX.wBitsPerSample: %d, \
-WAVEFORMATEX.nBlockAlign: %d, WAVEFORMATEX.nAvgBytesPerSec: %ld\n", spec->channels, spec->rate, spec->bytes_per_sample, wfx.nSamplesPerSec, wfx.wBitsPerSample, wfx.nBlockAlign, wfx.nAvgBytesPerSec);
+  GST_DEBUG_OBJECT (asrc, "channels: %d, rate: %d, bytes_per_sample: %d"
+      " WAVEFORMATEX.nSamplesPerSec: %ld, WAVEFORMATEX.wBitsPerSample: %d,"
+      " WAVEFORMATEX.nBlockAlign: %d, WAVEFORMATEX.nAvgBytesPerSec: %ld",
+      GST_AUDIO_INFO_CHANNELS (&spec->info), GST_AUDIO_INFO_RATE (&spec->info),
+      GST_AUDIO_INFO_BPF (&spec->info), wfx.nSamplesPerSec, wfx.wBitsPerSample,
+      wfx.nBlockAlign, wfx.nAvgBytesPerSec);
 
   return TRUE;
 
@@ -479,21 +437,18 @@ capture_buffer:
 dodgy_width:
   {
     GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ,
-        ("Unexpected width %d", spec->width), (NULL));
+        ("Unexpected width %d", wfx.wBitsPerSample), (NULL));
     return FALSE;
   }
-
 }
 
 static gboolean
 gst_directsound_src_unprepare (GstAudioSrc * asrc)
 {
   GstDirectSoundSrc *dsoundsrc;
+  HRESULT hRes;
 
-  HRESULT hRes;                 /* Result for windows functions */
-
-  /* Resets */
-  GST_DEBUG ("unpreparing directsoundsrc");
+  GST_DEBUG_OBJECT (asrc, "unpreparing directsoundsrc");
 
   dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
 
@@ -504,7 +459,6 @@ gst_directsound_src_unprepare (GstAudioSrc * asrc)
   hRes = IDirectSoundCaptureBuffer_Release (dsoundsrc->pDSBSecondary);
 
   return TRUE;
-
 }
 
 /* 
@@ -525,7 +479,7 @@ gst_directsound_src_read (GstAudioSrc * asrc, gpointer data, guint length)
 
   DWORD dwStatus = 0;
 
-  GST_DEBUG ("reading directsoundsrc\n");
+  GST_DEBUG_OBJECT (asrc, "reading directsoundsrc");
 
   dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
 
@@ -535,12 +489,12 @@ gst_directsound_src_read (GstAudioSrc * asrc, gpointer data, guint length)
   hRes = IDirectSoundCaptureBuffer_GetStatus (dsoundsrc->pDSBSecondary,
       &dwStatus);
 
-  /* Starting capturing if not allready */
+  /* Starting capturing if not already */
   if (!(dwStatus & DSCBSTATUS_CAPTURING)) {
     hRes = IDirectSoundCaptureBuffer_Start (dsoundsrc->pDSBSecondary,
         DSCBSTART_LOOPING);
     //    Sleep (dsoundsrc->latency_time/1000);
-    GST_DEBUG ("capture started");
+    GST_DEBUG_OBJECT (asrc, "capture started");
   }
   //  calculate_buffersize:
   while (length > dwBufferSize) {
@@ -558,8 +512,6 @@ gst_directsound_src_read (GstAudioSrc * asrc, gpointer data, guint length)
       dwBufferSize =
           dwCurrentCaptureCursor - dsoundsrc->current_circular_offset;
     }
-
-
   }                             // while (...
 
   /* Lock the buffer */
@@ -588,7 +540,6 @@ gst_directsound_src_read (GstAudioSrc * asrc, gpointer data, guint length)
 
   /* return length (readed data size in bytes) */
   return length;
-
 }
 
 static guint
@@ -600,7 +551,7 @@ gst_directsound_src_delay (GstAudioSrc * asrc)
   DWORD dwBytesInQueue = 0;
   gint nNbSamplesInQueue = 0;
 
-  GST_DEBUG ("Delay\n");
+  GST_DEBUG_OBJECT (asrc, "Delay");
 
   dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
 
@@ -632,7 +583,7 @@ gst_directsound_src_reset (GstAudioSrc * asrc)
   LPVOID pLockedBuffer = NULL;
   DWORD dwSizeBuffer = 0;
 
-  GST_DEBUG ("reset directsoundsrc\n");
+  GST_DEBUG_OBJECT (asrc, "reset directsoundsrc");
 
   dsoundsrc = GST_DIRECTSOUND_SRC (asrc);