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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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f62b7fd712
This is now handled directly in gstaudiosrc/sink, and we were setting it in the wrong thread anyway. prepare() is not the same thread as sink_write() or src_read().
117 lines
4.1 KiB
C
117 lines
4.1 KiB
C
/*
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* Copyright (C) 2008 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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#ifndef __GST_WASAPI_UTIL_H__
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#define __GST_WASAPI_UTIL_H__
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#include <gst/gst.h>
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#include <gst/audio/audio.h>
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#include <gst/audio/gstaudiosrc.h>
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#include <gst/audio/gstaudiosink.h>
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#ifdef _MSC_VER
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#include <initguid.h>
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#endif
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#include <mmdeviceapi.h>
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#include <audioclient.h>
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#include "gstaudioclient3.h"
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/* Static Caps shared between source, sink, and device provider */
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#define GST_WASAPI_STATIC_CAPS "audio/x-raw, " \
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"format = (string) " GST_AUDIO_FORMATS_ALL ", " \
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"layout = (string) interleaved, " \
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"rate = " GST_AUDIO_RATE_RANGE ", " \
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"channels = " GST_AUDIO_CHANNELS_RANGE
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/* Standard error path */
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#define HR_FAILED_AND(hr,func,and) \
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do { \
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if (FAILED (hr)) { \
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gchar *msg = gst_wasapi_util_hresult_to_string (hr); \
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GST_ERROR_OBJECT (self, #func " failed (%x): %s", (guint) hr, msg); \
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g_free (msg); \
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and; \
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} \
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} while (0)
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#define HR_FAILED_RET(hr,func,ret) HR_FAILED_AND(hr,func,return ret)
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#define HR_FAILED_GOTO(hr,func,where) HR_FAILED_AND(hr,func,res = FALSE; goto where)
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/* Device role enum property */
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typedef enum
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{
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GST_WASAPI_DEVICE_ROLE_CONSOLE,
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GST_WASAPI_DEVICE_ROLE_MULTIMEDIA,
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GST_WASAPI_DEVICE_ROLE_COMMS
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} GstWasapiDeviceRole;
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#define GST_WASAPI_DEVICE_TYPE_ROLE (gst_wasapi_device_role_get_type())
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GType gst_wasapi_device_role_get_type (void);
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/* Utilities */
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gboolean gst_wasapi_util_have_audioclient3 (void);
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gint gst_wasapi_device_role_to_erole (gint role);
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gint gst_wasapi_erole_to_device_role (gint erole);
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gchar *gst_wasapi_util_hresult_to_string (HRESULT hr);
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gboolean gst_wasapi_util_get_devices (GstElement * element, gboolean active,
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GList ** devices);
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gboolean gst_wasapi_util_get_device_client (GstElement * element,
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gint data_flow, gint role, const wchar_t * device_strid,
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IMMDevice ** ret_device, IAudioClient ** ret_client);
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gboolean gst_wasapi_util_get_device_format (GstElement * element,
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gint device_mode, IMMDevice * device, IAudioClient * client,
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WAVEFORMATEX ** ret_format);
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gboolean gst_wasapi_util_get_render_client (GstElement * element,
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IAudioClient * client, IAudioRenderClient ** ret_render_client);
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gboolean gst_wasapi_util_get_capture_client (GstElement * element,
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IAudioClient * client, IAudioCaptureClient ** ret_capture_client);
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gboolean gst_wasapi_util_get_clock (GstElement * element,
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IAudioClient * client, IAudioClock ** ret_clock);
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gboolean gst_wasapi_util_parse_waveformatex (WAVEFORMATEXTENSIBLE * format,
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GstCaps * template_caps, GstCaps ** out_caps,
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GstAudioChannelPosition ** out_positions);
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void gst_wasapi_util_get_best_buffer_sizes (GstAudioRingBufferSpec * spec,
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gboolean exclusive, REFERENCE_TIME default_period,
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REFERENCE_TIME min_period, REFERENCE_TIME * ret_period,
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REFERENCE_TIME * ret_buffer_duration);
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gboolean gst_wasapi_util_initialize_audioclient (GstElement * element,
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GstAudioRingBufferSpec * spec, IAudioClient * client,
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WAVEFORMATEX * format, guint sharemode, gboolean low_latency,
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gboolean loopback, guint * ret_devicep_frames);
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gboolean gst_wasapi_util_initialize_audioclient3 (GstElement * element,
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GstAudioRingBufferSpec * spec, IAudioClient3 * client,
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WAVEFORMATEX * format, gboolean low_latency, gboolean loopback,
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guint * ret_devicep_frames);
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#endif /* __GST_WASAPI_UTIL_H__ */
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