/* * GStreamer * Copyright 2005 Thomas Vander Stichele * Copyright 2005 Ronald S. Bultje * Copyright 2005 Sébastien Moutte * Copyright 2006 Joni Valtanen * * 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 #include #include "gstdirectsoundsrc.h" #include #include GST_DEBUG_CATEGORY_STATIC (directsoundsrc_debug); #define GST_CAT_DEFAULT directsoundsrc_debug static GstElementDetails gst_directsound_src_details = GST_ELEMENT_DETAILS ("Direct Sound Audio Src", "Source/Audio", "Capture from a soundcard via DIRECTSOUND", "Joni Valtanen "); /* defaults here */ #define DEFAULT_DEVICE 0 /* properties */ enum { PROP_0, PROP_DEVICE }; 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) { 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) { G_OBJECT_CLASS (parent_class)->dispose (object); } static void gst_directsound_src_finalise (GObject * object) { GstDirectSoundSrc *dsoundsrc = GST_DIRECTSOUND_SRC (object); g_mutex_free (dsoundsrc->dsound_lock); } 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_details (element_class, &gst_directsound_src_details); 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"); 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); 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); } static GstCaps * gst_directsound_src_getcaps (GstBaseSrc * bsrc) { 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))); 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\n"); switch (prop_id) { #if 0 /* FIXME */ case PROP_DEVICE: src->device = g_value_get_uint (value); break; #endif 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) { #if 0 GstDirectSoundSrc *src = GST_DIRECTSOUND_SRC (object); #endif GST_DEBUG ("get property\n"); switch (prop_id) { #if 0 /* FIXME */ case PROP_DEVICE: g_value_set_uint (value, src->device); break; #endif 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, GstDirectSoundSrcClass * gclass) { GST_DEBUG ("initializing directsoundsrc\n"); 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"); 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; } /* 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)); 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%x\n", HRESULT_FROM_WIN32 (err)); return FALSE; } } static gboolean gst_directsound_src_close (GstAudioSrc * asrc) { 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) { GstDirectSoundSrc *dsoundsrc; WAVEFORMATEX wfx; /* Wave format structure */ HRESULT hRes; /* Result for windows functions */ DSCBUFFERDESC descSecondary; /* Capturebuffer decsiption */ int fmt = 0; /* audio format */ 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: %llu - buffer time: %llu", 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: %d, 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; 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; wrong_format: { GST_ELEMENT_ERROR (dsoundsrc, RESOURCE, OPEN_READ, ("Unable to get format %d", spec->format), (NULL)); return FALSE; } 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", spec->width), (NULL)); return FALSE; } } static gboolean gst_directsound_src_unprepare (GstAudioSrc * asrc) { 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) { 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) { 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) { 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); }