blob: 3977466412bac3203e62a4a0fe0a342cc2cd813e [file] [log] [blame]
/*
* GStreamer
* Copyright 2005 Thomas Vander Stichele <thomas@apestaart.org>
* Copyright 2005 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* Copyright 2005 S´┐Żbastien 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/*
TODO: add mixer device init for selection by device-guid
*/
/**
* SECTION:element-directsoundsrc
*
* Reads audio data using the DirectSound API.
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch-1.0 -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/audio.h>
#include <gst/audio/gstaudiobasesrc.h>
#include "gstdirectsoundsrc.h"
#include <windows.h>
#include <dsound.h>
#include <mmsystem.h>
GST_DEBUG_CATEGORY_STATIC (directsoundsrc_debug);
#define GST_CAT_DEFAULT directsoundsrc_debug
/* defaults here */
#define DEFAULT_DEVICE 0
#define DEFAULT_MUTE FALSE
/* properties */
enum
{
PROP_0,
PROP_DEVICE_NAME,
PROP_DEVICE,
PROP_VOLUME,
PROP_MUTE
};
static HRESULT (WINAPI * pDSoundCaptureCreate) (LPGUID,
LPDIRECTSOUNDCAPTURE *, LPUNKNOWN);
static void gst_directsound_src_finalize (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,
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,
GstCaps * filter);
static guint gst_directsound_src_read (GstAudioSrc * asrc,
gpointer data, guint length, GstClockTime * timestamp);
static void gst_directsound_src_dispose (GObject * object);
static guint gst_directsound_src_delay (GstAudioSrc * asrc);
static gboolean gst_directsound_src_mixer_find (GstDirectSoundSrc * dsoundsrc,
MIXERCAPS * mixer_caps);
static void gst_directsound_src_mixer_init (GstDirectSoundSrc * dsoundsrc);
static gdouble gst_directsound_src_get_volume (GstDirectSoundSrc * dsoundsrc);
static void gst_directsound_src_set_volume (GstDirectSoundSrc * dsoundsrc,
gdouble volume);
static gboolean gst_directsound_src_get_mute (GstDirectSoundSrc * dsoundsrc);
static void gst_directsound_src_set_mute (GstDirectSoundSrc * dsoundsrc,
gboolean mute);
static const gchar *gst_directsound_src_get_device (GstDirectSoundSrc *
dsoundsrc);
static void gst_directsound_src_set_device (GstDirectSoundSrc * dsoundsrc,
const gchar * device_id);
static GstStaticPadTemplate directsound_src_src_factory =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) { S16LE, S8 }, "
"layout = (string) interleaved, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ]"));
#define gst_directsound_src_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstDirectSoundSrc, gst_directsound_src,
GST_TYPE_AUDIO_SRC, G_IMPLEMENT_INTERFACE (GST_TYPE_STREAM_VOLUME, NULL)
);
static void
gst_directsound_src_dispose (GObject * object)
{
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_directsound_src_finalize (GObject * object)
{
GstDirectSoundSrc *dsoundsrc = GST_DIRECTSOUND_SRC (object);
g_mutex_clear (&dsoundsrc->dsound_lock);
g_free (dsoundsrc->device_name);
g_free (dsoundsrc->device_id);
g_free (dsoundsrc->device_guid);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_directsound_src_class_init (GstDirectSoundSrcClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSrcClass *gstbasesrc_class;
GstAudioSrcClass *gstaudiosrc_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesrc_class = (GstBaseSrcClass *) klass;
gstaudiosrc_class = (GstAudioSrcClass *) klass;
GST_DEBUG_CATEGORY_INIT (directsoundsrc_debug, "directsoundsrc", 0,
"DirectSound Src");
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);
gobject_class->get_property =
GST_DEBUG_FUNCPTR (gst_directsound_src_get_property);
gobject_class->set_property =
GST_DEBUG_FUNCPTR (gst_directsound_src_set_property);
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);
gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_directsound_src_delay);
gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_directsound_src_reset);
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_static_pad_template (gstelement_class,
&directsound_src_src_factory);
g_object_class_install_property
(gobject_class, PROP_DEVICE_NAME,
g_param_spec_string ("device-name", "Device name",
"Human-readable name of the sound device", NULL, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class,
PROP_DEVICE,
g_param_spec_string ("device", "Device",
"DirectSound playback device as a GUID string (volume and mute will not work!)",
NULL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property
(gobject_class, PROP_VOLUME,
g_param_spec_double ("volume", "Volume",
"Volume of this stream", 0.0, 1.0, 1.0,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property
(gobject_class, PROP_MUTE,
g_param_spec_boolean ("mute", "Mute",
"Mute state of this stream", DEFAULT_MUTE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
}
static GstCaps *
gst_directsound_src_getcaps (GstBaseSrc * bsrc, GstCaps * filter)
{
GstCaps *caps = NULL;
GST_DEBUG_OBJECT (bsrc, "get caps");
caps = gst_pad_get_pad_template_caps (GST_BASE_SRC_PAD (bsrc));
return caps;
}
static void
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");
switch (prop_id) {
case PROP_DEVICE_NAME:
if (src->device_name) {
g_free (src->device_name);
src->device_name = NULL;
}
if (g_value_get_string (value)) {
src->device_name = g_strdup (g_value_get_string (value));
}
break;
case PROP_VOLUME:
gst_directsound_src_set_volume (src, g_value_get_double (value));
break;
case PROP_MUTE:
gst_directsound_src_set_mute (src, g_value_get_boolean (value));
break;
case PROP_DEVICE:
gst_directsound_src_set_device (src, g_value_get_string (value));
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_directsound_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstDirectSoundSrc *src = GST_DIRECTSOUND_SRC (object);
GST_DEBUG ("get property");
switch (prop_id) {
case PROP_DEVICE_NAME:
g_value_set_string (value, src->device_name);
break;
case PROP_DEVICE:
g_value_set_string (value, gst_directsound_src_get_device (src));
break;
case PROP_VOLUME:
g_value_set_double (value, gst_directsound_src_get_volume (src));
break;
case PROP_MUTE:
g_value_set_boolean (value, gst_directsound_src_get_mute (src));
break;
default:
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)
{
GST_DEBUG_OBJECT (src, "initializing directsoundsrc");
g_mutex_init (&src->dsound_lock);
src->device_guid = NULL;
src->device_id = NULL;
src->device_name = NULL;
src->mixer = NULL;
src->control_id_mute = -1;
src->control_id_volume = -1;
src->volume = 100;
src->mute = FALSE;
}
/* Enumeration callback called by DirectSoundCaptureEnumerate.
* Gets the GUID of request audio device
*/
static BOOL CALLBACK
gst_directsound_enum_callback (GUID * pGUID, TCHAR * strDesc,
TCHAR * strDrvName, VOID * pContext)
{
GstDirectSoundSrc *dsoundsrc = GST_DIRECTSOUND_SRC (pContext);
gchar *driver, *description;
description = g_locale_to_utf8 (strDesc, -1, NULL, NULL, NULL);
if (!description) {
GST_ERROR_OBJECT (dsoundsrc,
"Failed to convert description from locale encoding to UTF8");
return TRUE;
}
driver = g_locale_to_utf8 (strDrvName, -1, NULL, NULL, NULL);
if (pGUID && dsoundsrc && dsoundsrc->device_name &&
!g_strcmp0 (dsoundsrc->device_name, description)) {
g_free (dsoundsrc->device_guid);
dsoundsrc->device_guid = (GUID *) g_malloc0 (sizeof (GUID));
memcpy (dsoundsrc->device_guid, pGUID, sizeof (GUID));
GST_INFO_OBJECT (dsoundsrc, "found the requested audio device :%s",
dsoundsrc->device_name);
g_free (description);
g_free (driver);
return FALSE;
}
GST_INFO_OBJECT (dsoundsrc, "sound device names: %s, %s, requested device:%s",
description, driver, dsoundsrc->device_name);
g_free (description);
g_free (driver);
return TRUE;
}
static LPGUID
string_to_guid (const gchar * str)
{
HRESULT ret;
gunichar2 *wstr;
LPGUID out;
wstr = g_utf8_to_utf16 (str, -1, NULL, NULL, NULL);
if (!wstr)
return NULL;
out = g_new (GUID, 1);
ret = CLSIDFromString ((LPOLESTR) wstr, out);
g_free (wstr);
if (ret != NOERROR) {
g_free (out);
return NULL;
}
return out;
}
static gboolean
gst_directsound_src_open (GstAudioSrc * asrc)
{
GstDirectSoundSrc *dsoundsrc;
HRESULT hRes; /* Result for windows functions */
GST_DEBUG_OBJECT (asrc, "opening directsoundsrc");
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,
TEXT ("DirectSoundCaptureCreate"));
/* If everything is not ok */
if (!pDSoundCaptureCreate) {
goto capture_function;
}
if (dsoundsrc->device_id) {
GST_DEBUG_OBJECT (asrc, "device id set to: %s ", dsoundsrc->device_id);
dsoundsrc->device_guid = string_to_guid (dsoundsrc->device_id);
if (dsoundsrc->device_guid == NULL) {
GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ,
("gst_directsound_src_open: device set, but guid not found: %s",
dsoundsrc->device_id), (NULL));
g_free (dsoundsrc->device_guid);
return FALSE;
}
} else {
hRes = DirectSoundCaptureEnumerate ((LPDSENUMCALLBACK)
gst_directsound_enum_callback, (VOID *) dsoundsrc);
if (FAILED (hRes)) {
goto capture_enumerate;
}
}
/* Create capture object */
hRes = pDSoundCaptureCreate (dsoundsrc->device_guid, &dsoundsrc->pDSC, NULL);
if (FAILED (hRes)) {
goto capture_object;
}
// mixer is only supported when device-id is not set
if (!dsoundsrc->device_id) {
gst_directsound_src_mixer_init (dsoundsrc);
}
return TRUE;
capture_function:
{
FreeLibrary (dsoundsrc->DSoundDLL);
GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ,
("Unable to get capturecreate function"), (NULL));
return FALSE;
}
capture_enumerate:
{
FreeLibrary (dsoundsrc->DSoundDLL);
GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ,
("Unable to enumerate audio capture devices"), (NULL));
return FALSE;
}
capture_object:
{
FreeLibrary (dsoundsrc->DSoundDLL);
GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ,
("Unable to create capture object"), (NULL));
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));
return FALSE;
}
}
static gboolean
gst_directsound_src_close (GstAudioSrc * asrc)
{
GstDirectSoundSrc *dsoundsrc;
GST_DEBUG_OBJECT (asrc, "closing directsoundsrc");
dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
/* Release capture handler */
IDirectSoundCapture_Release (dsoundsrc->pDSC);
/* Close library */
FreeLibrary (dsoundsrc->DSoundDLL);
if (dsoundsrc->mixer)
mixerClose (dsoundsrc->mixer);
return TRUE;
}
static gboolean
gst_directsound_src_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
{
GstDirectSoundSrc *dsoundsrc;
WAVEFORMATEX wfx; /* Wave format structure */
HRESULT hRes; /* Result for windows functions */
DSCBUFFERDESC descSecondary; /* Capturebuffer description */
dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
GST_DEBUG_OBJECT (asrc, "preparing directsoundsrc");
/* Define buffer */
memset (&wfx, 0, sizeof (WAVEFORMATEX));
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;
/* Ignored for WAVE_FORMAT_PCM. */
wfx.cbSize = 0;
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;
GST_DEBUG_OBJECT (asrc, "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;
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 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");
}
/* 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 = GST_AUDIO_INFO_BPF (&spec->info) * 8 *
(wfx.nAvgBytesPerSec / spec->segsize);
GST_DEBUG_OBJECT (asrc,
"bytes/sec: %lu, buffer size: %d, segsize: %d, segtotal: %d",
wfx.nAvgBytesPerSec, dsoundsrc->buffer_size, spec->segsize,
spec->segtotal);
/* Not read anything yet */
dsoundsrc->current_circular_offset = 0;
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;
capture_buffer:
{
GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ,
("Unable to create capturebuffer"), (NULL));
return FALSE;
}
dodgy_width:
{
GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ,
("Unexpected width %d", wfx.wBitsPerSample), (NULL));
return FALSE;
}
}
static gboolean
gst_directsound_src_unprepare (GstAudioSrc * asrc)
{
GstDirectSoundSrc *dsoundsrc;
GST_DEBUG_OBJECT (asrc, "unpreparing directsoundsrc");
dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
GST_DSOUND_LOCK (dsoundsrc);
/* Stop capturing */
IDirectSoundCaptureBuffer_Stop (dsoundsrc->pDSBSecondary);
/* Release buffer */
IDirectSoundCaptureBuffer_Release (dsoundsrc->pDSBSecondary);
GST_DSOUND_UNLOCK (dsoundsrc);
return TRUE;
}
/*
return number of readed bytes */
static guint
gst_directsound_src_read (GstAudioSrc * asrc, gpointer data, guint length,
GstClockTime * timestamp)
{
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_OBJECT (asrc, "reading directsoundsrc");
dsoundsrc = GST_DIRECTSOUND_SRC (asrc);
GST_DSOUND_LOCK (dsoundsrc);
/* Get current buffer status */
hRes = IDirectSoundCaptureBuffer_GetStatus (dsoundsrc->pDSBSecondary,
&dwStatus);
if (FAILED (hRes)) {
GST_DSOUND_UNLOCK (dsoundsrc);
return -1;
}
/* Starting capturing if not already */
if (!(dwStatus & DSCBSTATUS_CAPTURING)) {
hRes = IDirectSoundCaptureBuffer_Start (dsoundsrc->pDSBSecondary,
DSCBSTART_LOOPING);
// Sleep (dsoundsrc->latency_time/1000);
GST_DEBUG_OBJECT (asrc, "capture started");
}
// calculate_buffersize:
while (length > dwBufferSize) {
Sleep (dsoundsrc->latency_time / 1000);
hRes =
IDirectSoundCaptureBuffer_GetCurrentPosition (dsoundsrc->pDSBSecondary,
&dwCurrentCaptureCursor, NULL);
if (FAILED (hRes)) {
GST_DSOUND_UNLOCK (dsoundsrc);
return -1;
}
/* 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)
{
GstDirectSoundSrc *dsoundsrc;
HRESULT hRes;
DWORD dwCurrentCaptureCursor;
DWORD dwBytesInQueue = 0;
gint nNbSamplesInQueue = 0;
GST_DEBUG_OBJECT (asrc, "Delay");
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)
{
GstDirectSoundSrc *dsoundsrc;
LPVOID pLockedBuffer = NULL;
DWORD dwSizeBuffer = 0;
GST_DEBUG_OBJECT (asrc, "reset directsoundsrc");
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);
}
/* If the PROP_DEVICE_NAME is set, find the mixer related to device;
* otherwise we get the default input mixer. */
static gboolean
gst_directsound_src_mixer_find (GstDirectSoundSrc * dsoundsrc,
MIXERCAPS * mixer_caps)
{
MMRESULT mmres;
guint i, num_mixers;
num_mixers = mixerGetNumDevs ();
for (i = 0; i < num_mixers; i++) {
mmres = mixerOpen (&dsoundsrc->mixer, i, 0L, 0L,
MIXER_OBJECTF_MIXER | MIXER_OBJECTF_WAVEIN);
if (mmres != MMSYSERR_NOERROR)
continue;
mmres = mixerGetDevCaps ((UINT_PTR)dsoundsrc->mixer,
mixer_caps, sizeof (MIXERCAPS));
if (mmres != MMSYSERR_NOERROR) {
mixerClose (dsoundsrc->mixer);
continue;
}
/* Get default mixer */
if (dsoundsrc->device_name == NULL) {
GST_DEBUG ("Got default input mixer: %s", mixer_caps->szPname);
return TRUE;
}
if (g_strstr_len (dsoundsrc->device_name, -1, mixer_caps->szPname) != NULL) {
GST_DEBUG ("Got requested input mixer: %s", mixer_caps->szPname);
return TRUE;
}
/* Wrong mixer */
mixerClose (dsoundsrc->mixer);
}
GST_DEBUG ("Can't find input mixer");
return FALSE;
}
static void
gst_directsound_src_mixer_init (GstDirectSoundSrc * dsoundsrc)
{
gint i, k;
gboolean found_mic;
MMRESULT mmres;
MIXERCAPS mixer_caps;
MIXERLINE mixer_line;
MIXERLINECONTROLS ml_ctrl;
PMIXERCONTROL pamixer_ctrls;
if (!gst_directsound_src_mixer_find (dsoundsrc, &mixer_caps))
goto mixer_init_fail;
/* Find the MIXERLINE related to MICROPHONE */
found_mic = FALSE;
for (i = 0; i < mixer_caps.cDestinations && !found_mic; i++) {
gint j, num_connections;
mixer_line.cbStruct = sizeof (mixer_line);
mixer_line.dwDestination = i;
mmres = mixerGetLineInfo ((HMIXEROBJ) dsoundsrc->mixer,
&mixer_line, MIXER_GETLINEINFOF_DESTINATION);
if (mmres != MMSYSERR_NOERROR)
goto mixer_init_fail;
num_connections = mixer_line.cConnections;
for (j = 0; j < num_connections && !found_mic; j++) {
mixer_line.cbStruct = sizeof (mixer_line);
mixer_line.dwDestination = i;
mixer_line.dwSource = j;
mmres = mixerGetLineInfo ((HMIXEROBJ) dsoundsrc->mixer,
&mixer_line, MIXER_GETLINEINFOF_SOURCE);
if (mmres != MMSYSERR_NOERROR)
goto mixer_init_fail;
if (mixer_line.dwComponentType == MIXERLINE_COMPONENTTYPE_SRC_MICROPHONE
|| mixer_line.dwComponentType == MIXERLINE_COMPONENTTYPE_SRC_LINE)
found_mic = TRUE;
}
}
if (found_mic == FALSE) {
GST_DEBUG ("Can't find mixer line related to input");
goto mixer_init_fail;
}
/* Get control associated with microphone audio line */
pamixer_ctrls = g_malloc (sizeof (MIXERCONTROL) * mixer_line.cControls);
ml_ctrl.cbStruct = sizeof (ml_ctrl);
ml_ctrl.dwLineID = mixer_line.dwLineID;
ml_ctrl.cControls = mixer_line.cControls;
ml_ctrl.cbmxctrl = sizeof (MIXERCONTROL);
ml_ctrl.pamxctrl = pamixer_ctrls;
mmres = mixerGetLineControls ((HMIXEROBJ) dsoundsrc->mixer,
&ml_ctrl, MIXER_GETLINECONTROLSF_ALL);
/* Find control associated with volume and mute */
for (k = 0; k < mixer_line.cControls; k++) {
if (strstr (pamixer_ctrls[k].szName, "Volume") != NULL) {
dsoundsrc->control_id_volume = pamixer_ctrls[k].dwControlID;
dsoundsrc->dw_vol_max = pamixer_ctrls[k].Bounds.dwMaximum;
dsoundsrc->dw_vol_min = pamixer_ctrls[k].Bounds.dwMinimum;
} else if (strstr (pamixer_ctrls[k].szName, "Mute") != NULL) {
dsoundsrc->control_id_mute = pamixer_ctrls[k].dwControlID;
} else {
GST_DEBUG ("Control not handled: %s", pamixer_ctrls[k].szName);
}
}
g_free (pamixer_ctrls);
if (dsoundsrc->control_id_volume < 0 && dsoundsrc->control_id_mute < 0)
goto mixer_init_fail;
/* Save cChannels information to properly changes in volume */
dsoundsrc->mixerline_cchannels = mixer_line.cChannels;
return;
mixer_init_fail:
GST_WARNING ("Failed to get Volume and Mute controls");
if (dsoundsrc->mixer != NULL) {
mixerClose (dsoundsrc->mixer);
dsoundsrc->mixer = NULL;
}
}
static gdouble
gst_directsound_src_get_volume (GstDirectSoundSrc * dsoundsrc)
{
return (gdouble) dsoundsrc->volume / 100;
}
static gboolean
gst_directsound_src_get_mute (GstDirectSoundSrc * dsoundsrc)
{
return dsoundsrc->mute;
}
static void
gst_directsound_src_set_volume (GstDirectSoundSrc * dsoundsrc, gdouble volume)
{
MMRESULT mmres;
MIXERCONTROLDETAILS details;
MIXERCONTROLDETAILS_UNSIGNED details_unsigned;
glong dwvolume;
if (dsoundsrc->mixer == NULL || dsoundsrc->control_id_volume < 0) {
GST_WARNING ("mixer not initialized");
return;
}
dwvolume = volume * dsoundsrc->dw_vol_max;
dwvolume = CLAMP (dwvolume, dsoundsrc->dw_vol_min, dsoundsrc->dw_vol_max);
GST_DEBUG ("max volume %ld | min volume %ld",
dsoundsrc->dw_vol_max, dsoundsrc->dw_vol_min);
GST_DEBUG ("set volume to %f (%ld)", volume, dwvolume);
details.cbStruct = sizeof (details);
details.dwControlID = dsoundsrc->control_id_volume;
details.cChannels = dsoundsrc->mixerline_cchannels;
details.cMultipleItems = 0;
details_unsigned.dwValue = dwvolume;
details.cbDetails = sizeof (MIXERCONTROLDETAILS_UNSIGNED);
details.paDetails = &details_unsigned;
mmres = mixerSetControlDetails ((HMIXEROBJ) dsoundsrc->mixer,
&details, MIXER_OBJECTF_HMIXER | MIXER_SETCONTROLDETAILSF_VALUE);
if (mmres != MMSYSERR_NOERROR)
GST_WARNING ("Failed to set volume");
else
dsoundsrc->volume = volume * 100;
}
static void
gst_directsound_src_set_mute (GstDirectSoundSrc * dsoundsrc, gboolean mute)
{
MMRESULT mmres;
MIXERCONTROLDETAILS details;
MIXERCONTROLDETAILS_BOOLEAN details_boolean;
if (dsoundsrc->mixer == NULL || dsoundsrc->control_id_mute < 0) {
GST_WARNING ("mixer not initialized");
return;
}
details.cbStruct = sizeof (details);
details.dwControlID = dsoundsrc->control_id_mute;
details.cChannels = dsoundsrc->mixerline_cchannels;
details.cMultipleItems = 0;
details_boolean.fValue = mute;
details.cbDetails = sizeof (MIXERCONTROLDETAILS_BOOLEAN);
details.paDetails = &details_boolean;
mmres = mixerSetControlDetails ((HMIXEROBJ) dsoundsrc->mixer,
&details, MIXER_OBJECTF_HMIXER | MIXER_SETCONTROLDETAILSF_VALUE);
if (mmres != MMSYSERR_NOERROR)
GST_WARNING ("Failed to set mute");
else
dsoundsrc->mute = mute;
}
static const gchar *
gst_directsound_src_get_device (GstDirectSoundSrc * dsoundsrc)
{
return dsoundsrc->device_id;
}
static void
gst_directsound_src_set_device (GstDirectSoundSrc * dsoundsrc,
const gchar * device_id)
{
g_free (dsoundsrc->device_id);
dsoundsrc->device_id = g_strdup (device_id);
}