gstreamer/sys/directsound/gstdirectsoundsrc.c

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/*
* GStreamer
* Copyright 2005 Thomas Vander Stichele <thomas@apestaart.org>
* Copyright 2005 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* Copyright 2005 S<EFBFBD>bastien Moutte <sebastien@moutte.net>
* Copyright 2006 Joni Valtanen <joni.valtanen@movial.fi>
*
* Permission is hereby granted, free of charge, to any person obtaining a
* copy of this software and associated documentation files (the "Software"),
* to deal in the Software without restriction, including without limitation
* the rights to use, copy, modify, merge, publish, distribute, sublicense,
* and/or sell copies of the Software, and to permit persons to whom the
* Software is furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
* AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
* DEALINGS IN THE SOFTWARE.
*
* Alternatively, the contents of this file may be used under the
* GNU Lesser General Public License Version 2.1 (the "LGPL"), in
* which case the following provisions apply instead of the ones
* mentioned above:
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/*
TODO: add device selection and check rate etc.
*/
#ifdef HAVE_CONFIG_H
# include "config.h"
#endif
#include <gst/gst.h>
#include <gst/audio/gstbaseaudiosrc.h>
#include "gstdirectsoundsrc.h"
#include <windows.h>
#include <dsound.h>
GST_DEBUG_CATEGORY_STATIC (directsoundsrc_debug);
#define GST_CAT_DEFAULT directsoundsrc_debug
/* defaults here */
#define DEFAULT_DEVICE 0
/* properties */
enum
{
PROP_0,
PROP_DEVICE
};
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static HRESULT (WINAPI * pDSoundCaptureCreate) (LPGUID,
LPDIRECTSOUNDCAPTURE *, LPUNKNOWN);
static void gst_directsound_src_finalise (GObject * object);
static void gst_directsound_src_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_directsound_src_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
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);
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 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_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, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]"));
static void
gst_directsound_src_do_init (GType type)
{
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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);
static void
gst_directsound_src_dispose (GObject * object)
{
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G_OBJECT_CLASS (parent_class)->dispose (object);
}
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static void
gst_directsound_src_finalise (GObject * object)
{
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GstDirectSoundSrc *dsoundsrc = GST_DIRECTSOUND_SRC (object);
g_mutex_free (dsoundsrc->dsound_lock);
}
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static void
gst_directsound_src_base_init (gpointer g_class)
{
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
GST_DEBUG ("initializing directsoundsrc base\n");
gst_element_class_set_details_simple (element_class, "Direct Sound Audio Src",
"Source/Audio",
"Capture from a soundcard via DIRECTSOUND",
"Joni Valtanen <joni.valtanen@movial.fi>");
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gst_element_class_add_static_pad_template (element_class,
&directsound_src_src_factory);
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}
/* initialize the plugin's class */
static void
gst_directsound_src_class_init (GstDirectSoundSrcClass * klass)
{
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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");
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_directsound_src_finalise);
gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_directsound_src_dispose);
gobject_class->get_property =
GST_DEBUG_FUNCPTR (gst_directsound_src_get_property);
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gobject_class->set_property =
GST_DEBUG_FUNCPTR (gst_directsound_src_set_property);
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gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_directsound_src_getcaps);
gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_directsound_src_open);
gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_directsound_src_close);
gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_directsound_src_read);
gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_directsound_src_prepare);
gstaudiosrc_class->unprepare =
GST_DEBUG_FUNCPTR (gst_directsound_src_unprepare);
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gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_directsound_src_delay);
gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_directsound_src_reset);
}
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static GstCaps *
gst_directsound_src_getcaps (GstBaseSrc * bsrc)
{
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GstDirectSoundSrc *dsoundsrc;
GstCaps *caps = NULL;
GST_DEBUG ("get caps\n");
dsoundsrc = GST_DIRECTSOUND_SRC (bsrc);
caps = gst_caps_copy (gst_pad_get_pad_template_caps (GST_BASE_SRC_PAD
(bsrc)));
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return caps;
}
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static void
gst_directsound_src_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
{
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// GstDirectSoundSrc *src = GST_DIRECTSOUND_SRC (object);
GST_DEBUG ("set property\n");
switch (prop_id) {
#if 0
/* FIXME */
case PROP_DEVICE:
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src->device = g_value_get_uint (value);
break;
#endif
default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
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static void
gst_directsound_src_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
{
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#if 0
GstDirectSoundSrc *src = GST_DIRECTSOUND_SRC (object);
#endif
GST_DEBUG ("get property\n");
switch (prop_id) {
#if 0
/* FIXME */
case PROP_DEVICE:
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g_value_set_uint (value, src->device);
break;
#endif
default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
/* initialize the new element
* instantiate pads and add them to element
* set functions
* initialize structure
*/
static void
gst_directsound_src_init (GstDirectSoundSrc * src,
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GstDirectSoundSrcClass * gclass)
{
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GST_DEBUG ("initializing directsoundsrc\n");
src->dsound_lock = g_mutex_new ();
}
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static gboolean
gst_directsound_src_open (GstAudioSrc * asrc)
{
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GstDirectSoundSrc *dsoundsrc;
HRESULT hRes; /* Result for windows functions */
GST_DEBUG ("initializing directsoundsrc\n");
dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
/* Open dsound.dll */
dsoundsrc->DSoundDLL = LoadLibrary ("dsound.dll");
if (!dsoundsrc->DSoundDLL) {
goto dsound_open;
}
/* Building the DLL Calls */
pDSoundCaptureCreate =
(void *) GetProcAddress (dsoundsrc->DSoundDLL,
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TEXT ("DirectSoundCaptureCreate"));
/* If everything is not ok */
if (!pDSoundCaptureCreate) {
goto capture_function;
}
/* FIXME: add here device selection */
/* Create capture object */
hRes = pDSoundCaptureCreate (NULL, &dsoundsrc->pDSC, NULL);
if (FAILED (hRes)) {
goto capture_object;
}
return TRUE;
capture_function:
{
FreeLibrary (dsoundsrc->DSoundDLL);
GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ,
("Unable to get capturecreate function"), (NULL));
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return FALSE;
}
capture_object:
{
FreeLibrary (dsoundsrc->DSoundDLL);
GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ,
("Unable to create capture object"), (NULL));
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return FALSE;
}
dsound_open:
{
DWORD err = GetLastError ();
GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ,
("Unable to open dsound.dll"), (NULL));
g_print ("0x%lx\n", HRESULT_FROM_WIN32 (err));
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return FALSE;
}
}
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static gboolean
gst_directsound_src_close (GstAudioSrc * asrc)
{
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GstDirectSoundSrc *dsoundsrc;
HRESULT hRes; /* Result for windows functions */
GST_DEBUG ("initializing directsoundsrc\n");
dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
/* Release capture handler */
hRes = IDirectSoundCapture_Release (dsoundsrc->pDSC);
/* Close library */
FreeLibrary (dsoundsrc->DSoundDLL);
return TRUE;
}
static gboolean
gst_directsound_src_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec)
{
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GstDirectSoundSrc *dsoundsrc;
WAVEFORMATEX wfx; /* Wave format structure */
HRESULT hRes; /* Result for windows functions */
DSCBUFFERDESC descSecondary; /* Capturebuffer decsiption */
dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
GST_DEBUG ("initializing directsoundsrc\n");
/* 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.nAvgBytesPerSec = wfx.nSamplesPerSec * wfx.nBlockAlign;
wfx.cbSize = 0; /* This size is allways for PCM-format */
/* 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;
}
/* 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);
/* Init secondary buffer desciption */
memset (&descSecondary, 0, sizeof (DSCBUFFERDESC));
descSecondary.dwSize = sizeof (DSCBUFFERDESC);
descSecondary.dwFlags = 0;
descSecondary.dwReserved = 0;
/* This is not primary buffer so have to set size */
descSecondary.dwBufferBytes = dsoundsrc->buffer_size;
descSecondary.lpwfxFormat = &wfx;
/* Create buffer */
hRes = IDirectSoundCapture_CreateCaptureBuffer (dsoundsrc->pDSC,
&descSecondary, &dsoundsrc->pDSBSecondary, NULL);
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;
GST_DEBUG ("latency time: %" G_GUINT64_FORMAT " - buffer time: %" G_GUINT64_FORMAT,
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spec->latency_time, spec->buffer_time);
/* Buffer-time should be allways 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");
}
/* Save the times */
dsoundsrc->buffer_time = spec->buffer_time;
dsoundsrc->latency_time = spec->latency_time;
dsoundsrc->latency_size = (gint) wfx.nAvgBytesPerSec *
dsoundsrc->latency_time / 1000000.0;
spec->segsize = (guint) (((double) spec->buffer_time / 1000000.0) *
wfx.nAvgBytesPerSec);
/* just in case */
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",
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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;
if (spec->width != 16 && spec->width != 8)
goto dodgy_width;
/* Not readed 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);
return TRUE;
capture_buffer:
{
GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ,
("Unable to create capturebuffer"), (NULL));
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return FALSE;
}
dodgy_width:
{
GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ,
("Unexpected width %d", spec->width), (NULL));
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return FALSE;
}
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}
static gboolean
gst_directsound_src_unprepare (GstAudioSrc * asrc)
{
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GstDirectSoundSrc *dsoundsrc;
HRESULT hRes; /* Result for windows functions */
/* Resets */
GST_DEBUG ("unpreparing directsoundsrc");
dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
/* Stop capturing */
hRes = IDirectSoundCaptureBuffer_Stop (dsoundsrc->pDSBSecondary);
/* Release buffer */
hRes = IDirectSoundCaptureBuffer_Release (dsoundsrc->pDSBSecondary);
return TRUE;
}
/*
return number of readed bytes */
static guint
gst_directsound_src_read (GstAudioSrc * asrc, gpointer data, guint length)
{
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GstDirectSoundSrc *dsoundsrc;
HRESULT hRes; /* Result for windows functions */
DWORD dwCurrentCaptureCursor = 0;
DWORD dwBufferSize = 0;
LPVOID pLockedBuffer1 = NULL;
LPVOID pLockedBuffer2 = NULL;
DWORD dwSizeBuffer1 = 0;
DWORD dwSizeBuffer2 = 0;
DWORD dwStatus = 0;
GST_DEBUG ("reading directsoundsrc\n");
dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
GST_DSOUND_LOCK (dsoundsrc);
/* Get current buffer status */
hRes = IDirectSoundCaptureBuffer_GetStatus (dsoundsrc->pDSBSecondary,
&dwStatus);
/* Starting capturing if not allready */
if (!(dwStatus & DSCBSTATUS_CAPTURING)) {
hRes = IDirectSoundCaptureBuffer_Start (dsoundsrc->pDSBSecondary,
DSCBSTART_LOOPING);
// Sleep (dsoundsrc->latency_time/1000);
GST_DEBUG ("capture started");
}
// calculate_buffersize:
while (length > dwBufferSize) {
Sleep (dsoundsrc->latency_time / 1000);
hRes =
IDirectSoundCaptureBuffer_GetCurrentPosition (dsoundsrc->pDSBSecondary,
&dwCurrentCaptureCursor, NULL);
/* calculate the buffer */
if (dwCurrentCaptureCursor < dsoundsrc->current_circular_offset) {
dwBufferSize = dsoundsrc->buffer_size -
(dsoundsrc->current_circular_offset - dwCurrentCaptureCursor);
} else {
dwBufferSize =
dwCurrentCaptureCursor - dsoundsrc->current_circular_offset;
}
} // while (...
/* Lock the buffer */
hRes = IDirectSoundCaptureBuffer_Lock (dsoundsrc->pDSBSecondary,
dsoundsrc->current_circular_offset,
length,
&pLockedBuffer1, &dwSizeBuffer1, &pLockedBuffer2, &dwSizeBuffer2, 0L);
/* Copy buffer data to another buffer */
if (hRes == DS_OK) {
memcpy (data, pLockedBuffer1, dwSizeBuffer1);
}
/* ...and if something is in another buffer */
if (pLockedBuffer2 != NULL) {
memcpy (((guchar *) data + dwSizeBuffer1), pLockedBuffer2, dwSizeBuffer2);
}
dsoundsrc->current_circular_offset += dwSizeBuffer1 + dwSizeBuffer2;
dsoundsrc->current_circular_offset %= dsoundsrc->buffer_size;
IDirectSoundCaptureBuffer_Unlock (dsoundsrc->pDSBSecondary,
pLockedBuffer1, dwSizeBuffer1, pLockedBuffer2, dwSizeBuffer2);
GST_DSOUND_UNLOCK (dsoundsrc);
/* return length (readed data size in bytes) */
return length;
}
static guint
gst_directsound_src_delay (GstAudioSrc * asrc)
{
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GstDirectSoundSrc *dsoundsrc;
HRESULT hRes;
DWORD dwCurrentCaptureCursor;
DWORD dwBytesInQueue = 0;
gint nNbSamplesInQueue = 0;
GST_DEBUG ("Delay\n");
dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
/* evaluate the number of samples in queue in the circular buffer */
hRes =
IDirectSoundCaptureBuffer_GetCurrentPosition (dsoundsrc->pDSBSecondary,
&dwCurrentCaptureCursor, NULL);
/* FIXME: Check is this calculated right */
if (hRes == S_OK) {
if (dwCurrentCaptureCursor < dsoundsrc->current_circular_offset) {
dwBytesInQueue =
dsoundsrc->buffer_size - (dsoundsrc->current_circular_offset -
dwCurrentCaptureCursor);
} else {
dwBytesInQueue =
dwCurrentCaptureCursor - dsoundsrc->current_circular_offset;
}
nNbSamplesInQueue = dwBytesInQueue / dsoundsrc->bytes_per_sample;
}
return nNbSamplesInQueue;
}
static void
gst_directsound_src_reset (GstAudioSrc * asrc)
{
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GstDirectSoundSrc *dsoundsrc;
LPVOID pLockedBuffer = NULL;
DWORD dwSizeBuffer = 0;
GST_DEBUG ("reset directsoundsrc\n");
dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
#if 0
IDirectSoundCaptureBuffer_Stop (dsoundsrc->pDSBSecondary);
#endif
GST_DSOUND_LOCK (dsoundsrc);
if (dsoundsrc->pDSBSecondary) {
/*stop capturing */
HRESULT hRes = IDirectSoundCaptureBuffer_Stop (dsoundsrc->pDSBSecondary);
/*reset position */
/* hRes = IDirectSoundCaptureBuffer_SetCurrentPosition (dsoundsrc->pDSBSecondary, 0); */
/*reset the buffer */
hRes = IDirectSoundCaptureBuffer_Lock (dsoundsrc->pDSBSecondary,
dsoundsrc->current_circular_offset, dsoundsrc->buffer_size,
pLockedBuffer, &dwSizeBuffer, NULL, NULL, 0L);
if (SUCCEEDED (hRes)) {
memset (pLockedBuffer, 0, dwSizeBuffer);
hRes =
IDirectSoundCaptureBuffer_Unlock (dsoundsrc->pDSBSecondary,
pLockedBuffer, dwSizeBuffer, NULL, 0);
}
dsoundsrc->current_circular_offset = 0;
}
GST_DSOUND_UNLOCK (dsoundsrc);
}