gstreamer/sys/wasapi/gstwasapiutil.h

/*
 * 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"

/* 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 (GstObject * element, gboolean active,
    GList ** devices);

gboolean gst_wasapi_util_get_device (GstElement * self,
    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__ */