| /* | |
| * Copyright (C) 2008 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com> | |
| * | |
| * 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. | |
| */ | |
| #ifndef __GST_WASAPI_UTIL_H__ | |
| #define __GST_WASAPI_UTIL_H__ | |
| #include <gst/gst.h> | |
| #include <gst/audio/audio.h> | |
| #include <gst/audio/gstaudiosrc.h> | |
| #include <gst/audio/gstaudiosink.h> | |
| #include <mmdeviceapi.h> | |
| #include <audioclient.h> | |
| #include "gstaudioclient3.h" | |
| #include "gstmmdeviceenumerator.h" | |
| /* Static Caps shared between source, sink, and device provider */ | |
| #define GST_WASAPI_STATIC_CAPS "audio/x-raw, " \ | |
| "format = (string) " GST_AUDIO_FORMATS_ALL ", " \ | |
| "layout = (string) interleaved, " \ | |
| "rate = " GST_AUDIO_RATE_RANGE ", " \ | |
| "channels = " GST_AUDIO_CHANNELS_RANGE | |
| /* Standard error path */ | |
| #define HR_FAILED_AND(hr,func,and) \ | |
| G_STMT_START { \ | |
| if (FAILED (hr)) { \ | |
| gchar *msg = gst_wasapi_util_hresult_to_string (hr); \ | |
| GST_ERROR_OBJECT (self, #func " failed (%x): %s", (guint) hr, msg); \ | |
| g_free (msg); \ | |
| and; \ | |
| } \ | |
| } G_STMT_END | |
| #define HR_FAILED_RET(hr,func,ret) HR_FAILED_AND(hr,func,return ret) | |
| #define HR_FAILED_GOTO(hr,func,where) HR_FAILED_AND(hr,func,res = FALSE; goto where) | |
| #define HR_FAILED_ELEMENT_ERROR_AND(hr,func,el,and) \ | |
| G_STMT_START { \ | |
| if (FAILED (hr)) { \ | |
| gchar *msg = gst_wasapi_util_hresult_to_string (hr); \ | |
| GST_ERROR_OBJECT (el, #func " failed (%x): %s", (guint) hr, msg); \ | |
| if (GST_IS_AUDIO_SRC (el)) \ | |
| GST_ELEMENT_ERROR(el, RESOURCE, READ, \ | |
| (#func " failed (%x): %s", (guint) hr, msg), (NULL)); \ | |
| else \ | |
| GST_ELEMENT_ERROR(el, RESOURCE, WRITE, \ | |
| (#func " failed (%x): %s", (guint) hr, msg), (NULL)); \ | |
| g_free (msg); \ | |
| and; \ | |
| } \ | |
| } G_STMT_END | |
| #define HR_FAILED_ELEMENT_ERROR_RET(hr,func,el,ret) \ | |
| HR_FAILED_ELEMENT_ERROR_AND(hr,func,el,return ret) | |
| /* Device role enum property */ | |
| typedef enum | |
| { | |
| GST_WASAPI_DEVICE_ROLE_CONSOLE, | |
| GST_WASAPI_DEVICE_ROLE_MULTIMEDIA, | |
| GST_WASAPI_DEVICE_ROLE_COMMS | |
| } GstWasapiDeviceRole; | |
| #define GST_WASAPI_DEVICE_TYPE_ROLE (gst_wasapi_device_role_get_type()) | |
| GType gst_wasapi_device_role_get_type (void); | |
| /* Utilities */ | |
| gboolean gst_wasapi_util_have_audioclient3 (void); | |
| gint gst_wasapi_device_role_to_erole (gint role); | |
| gint gst_wasapi_erole_to_device_role (gint erole); | |
| gchar *gst_wasapi_util_hresult_to_string (HRESULT hr); | |
| gboolean gst_wasapi_util_get_devices (GstMMDeviceEnumerator * enumerator, | |
| gboolean active, | |
| GList ** devices); | |
| gboolean gst_wasapi_util_get_device (GstMMDeviceEnumerator * enumerator, | |
| gint data_flow, gint role, const wchar_t * device_strid, | |
| IMMDevice ** ret_device); | |
| gboolean gst_wasapi_util_get_audio_client (GstElement * self, | |
| IMMDevice * device, IAudioClient ** ret_client); | |
| gboolean gst_wasapi_util_get_device_format (GstElement * element, | |
| gint device_mode, IMMDevice * device, IAudioClient * client, | |
| WAVEFORMATEX ** ret_format); | |
| gboolean gst_wasapi_util_get_render_client (GstElement * element, | |
| IAudioClient * client, IAudioRenderClient ** ret_render_client); | |
| gboolean gst_wasapi_util_get_capture_client (GstElement * element, | |
| IAudioClient * client, IAudioCaptureClient ** ret_capture_client); | |
| gboolean gst_wasapi_util_get_clock (GstElement * element, | |
| IAudioClient * client, IAudioClock ** ret_clock); | |
| gboolean gst_wasapi_util_parse_waveformatex (WAVEFORMATEXTENSIBLE * format, | |
| GstCaps * template_caps, GstCaps ** out_caps, | |
| GstAudioChannelPosition ** out_positions); | |
| void gst_wasapi_util_get_best_buffer_sizes (GstAudioRingBufferSpec * spec, | |
| gboolean exclusive, REFERENCE_TIME default_period, | |
| REFERENCE_TIME min_period, REFERENCE_TIME * ret_period, | |
| REFERENCE_TIME * ret_buffer_duration); | |
| gboolean gst_wasapi_util_initialize_audioclient (GstElement * element, | |
| GstAudioRingBufferSpec * spec, IAudioClient * client, | |
| WAVEFORMATEX * format, guint sharemode, gboolean low_latency, | |
| gboolean loopback, guint * ret_devicep_frames); | |
| gboolean gst_wasapi_util_initialize_audioclient3 (GstElement * element, | |
| GstAudioRingBufferSpec * spec, IAudioClient3 * client, | |
| WAVEFORMATEX * format, gboolean low_latency, gboolean loopback, | |
| guint * ret_devicep_frames); | |
| #endif /* __GST_WASAPI_UTIL_H__ */ |