mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-23 10:11:08 +00:00
wasapisrc: Port to GstAudioSrc
This commit is contained in:
Sebastian Dröge
parent
1445438a8b
commit
363aa90a10
7 changed files with 226 additions and 202 deletions
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@ -1,6 +1,7 @@
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plugin_LTLIBRARIES = libgstwasapi.la
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libgstwasapi_la_SOURCES = gstwasapi.c \
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gstwasapisrc.c \
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gstwasapisink.c \
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gstwasapiutil.c
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@ -11,7 +12,7 @@ libgstwasapi_la_LIBADD = $(GST_LIBS) $(GST_BASE_LIBS) $(GST_PLUGINS_BASE_LIBS) \
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libgstwasapi_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
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libgstwasapi_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
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noinst_HEADERS = \
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noinst_HEADERS = gstwasapisrc.h \
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gstwasapisink.h \
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gstwasapiutil.h
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@ -22,12 +22,16 @@
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#endif
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#include "gstwasapisink.h"
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#include "gstwasapisrc.h"
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static gboolean
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plugin_init (GstPlugin * plugin)
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{
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gst_element_register (plugin, "wasapisink", GST_RANK_NONE,
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GST_TYPE_WASAPI_SINK);
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gst_element_register (plugin, "wasapisrc", GST_RANK_NONE,
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GST_TYPE_WASAPI_SRC);
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return TRUE;
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}
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@ -1,5 +1,7 @@
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/*
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* Copyright (C) 2008 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
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* Copyright (C) 2013 Collabora Ltd.
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* Author: Sebastian Dröge <sebastian.droege@collabora.co.uk>
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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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@ -62,7 +64,6 @@ static gint gst_wasapi_sink_write (GstAudioSink * asink,
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static guint gst_wasapi_sink_delay (GstAudioSink * asink);
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static void gst_wasapi_sink_reset (GstAudioSink * asink);
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G_DEFINE_TYPE (GstWasapiSink, gst_wasapi_sink, GST_TYPE_AUDIO_SINK);
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static void
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@ -35,7 +35,6 @@
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#endif
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#include "gstwasapisrc.h"
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#include <gst/audio/gstaudioclock.h>
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GST_DEBUG_CATEGORY_STATIC (gst_wasapi_src_debug);
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#define GST_CAT_DEFAULT gst_wasapi_src_debug
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@ -46,23 +45,25 @@ static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) S16LE, "
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"layout = (string) interleaved, "
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"rate = (int) 8000, " "channels = (int) 1"));
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"rate = (int) 44100, " "channels = (int) 1"));
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static void gst_wasapi_src_dispose (GObject * object);
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static void gst_wasapi_src_finalize (GObject * object);
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static GstClock *gst_wasapi_src_provide_clock (GstElement * element);
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static GstCaps * gst_wasapi_src_get_caps (GstBaseSrc * bsrc, GstCaps * filter);
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static gboolean gst_wasapi_src_start (GstBaseSrc * src);
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static gboolean gst_wasapi_src_stop (GstBaseSrc * src);
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static gboolean gst_wasapi_src_query (GstBaseSrc * src, GstQuery * query);
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static GstFlowReturn gst_wasapi_src_create (GstPushSrc * src, GstBuffer ** buf);
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static gboolean gst_wasapi_src_open (GstAudioSrc * asrc);
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static gboolean gst_wasapi_src_close (GstAudioSrc * asrc);
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static gboolean gst_wasapi_src_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec);
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static gboolean gst_wasapi_src_unprepare (GstAudioSrc * asrc);
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static guint gst_wasapi_src_read (GstAudioSrc * asrc, gpointer data, guint length, GstClockTime * timestamp);
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static guint gst_wasapi_src_delay (GstAudioSrc * asrc);
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static void gst_wasapi_src_reset (GstAudioSrc * asrc);
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static GstClockTime gst_wasapi_src_get_time (GstClock * clock,
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gpointer user_data);
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G_DEFINE_TYPE (GstWasapiSrc, gst_wasapi_src, GST_TYPE_PUSH_SRC);
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G_DEFINE_TYPE (GstWasapiSrc, gst_wasapi_src, GST_TYPE_AUDIO_SRC);
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static void
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gst_wasapi_src_class_init (GstWasapiSrcClass * klass)
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@ -70,13 +71,11 @@ gst_wasapi_src_class_init (GstWasapiSrcClass * klass)
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GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
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GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
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GstBaseSrcClass *gstbasesrc_class = GST_BASE_SRC_CLASS (klass);
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GstPushSrcClass *gstpushsrc_class = GST_PUSH_SRC_CLASS (klass);
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GstAudioSrcClass *gstaudiosrc_class = GST_AUDIO_SRC_CLASS (klass);
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gobject_class->dispose = gst_wasapi_src_dispose;
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gobject_class->finalize = gst_wasapi_src_finalize;
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gstelement_class->provide_clock = gst_wasapi_src_provide_clock;
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&src_template));
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gst_element_class_set_static_metadata (gstelement_class, "WasapiSrc",
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@ -84,11 +83,16 @@ gst_wasapi_src_class_init (GstWasapiSrcClass * klass)
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"Stream audio from an audio capture device through WASAPI",
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"Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>");
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gstbasesrc_class->start = gst_wasapi_src_start;
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gstbasesrc_class->stop = gst_wasapi_src_stop;
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gstbasesrc_class->query = gst_wasapi_src_query;
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gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_wasapi_src_get_caps);
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gstpushsrc_class->create = gst_wasapi_src_create;
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gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_wasapi_src_open);
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gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_wasapi_src_close);
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gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_wasapi_src_read);
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gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_wasapi_src_prepare);
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gstaudiosrc_class->unprepare =
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GST_DEBUG_FUNCPTR (gst_wasapi_src_unprepare);
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gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_wasapi_src_delay);
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gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_wasapi_src_reset);
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GST_DEBUG_CATEGORY_INIT (gst_wasapi_src_debug, "wasapisrc",
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0, "Windows audio session API source");
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@ -97,24 +101,16 @@ gst_wasapi_src_class_init (GstWasapiSrcClass * klass)
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static void
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gst_wasapi_src_init (GstWasapiSrc * self)
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{
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GstBaseSrc *basesrc = GST_BASE_SRC (self);
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/* override with a custom clock */
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if (GST_AUDIO_BASE_SRC (self)->clock)
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gst_object_unref (GST_AUDIO_BASE_SRC (self)->clock);
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gst_base_src_set_format (basesrc, GST_FORMAT_TIME);
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gst_base_src_set_live (basesrc, TRUE);
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self->rate = 8000;
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self->buffer_time = 20 * GST_MSECOND;
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self->period_time = 20 * GST_MSECOND;
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self->latency = GST_CLOCK_TIME_NONE;
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self->samples_per_buffer = self->rate / (GST_SECOND / self->period_time);
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self->start_time = GST_CLOCK_TIME_NONE;
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self->next_time = GST_CLOCK_TIME_NONE;
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self->clock = gst_audio_clock_new ("GstWasapiSrcClock",
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GST_AUDIO_BASE_SRC (self)->clock = gst_audio_clock_new ("GstWasapiSrcClock",
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gst_wasapi_src_get_time, gst_object_ref (self),
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(GDestroyNotify) gst_object_unref);
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self->event_handle = CreateEvent (NULL, FALSE, FALSE, NULL);
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CoInitialize (NULL);
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}
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@ -123,9 +119,9 @@ gst_wasapi_src_dispose (GObject * object)
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{
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GstWasapiSrc *self = GST_WASAPI_SRC (object);
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if (self->clock != NULL) {
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gst_object_unref (self->clock);
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self->clock = NULL;
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if (self->event_handle != NULL) {
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CloseHandle (self->event_handle);
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self->event_handle = NULL;
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}
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G_OBJECT_CLASS (gst_wasapi_src_parent_class)->dispose (object);
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@ -139,52 +135,100 @@ gst_wasapi_src_finalize (GObject * object)
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G_OBJECT_CLASS (gst_wasapi_src_parent_class)->finalize (object);
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}
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static GstClock *
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gst_wasapi_src_provide_clock (GstElement * element)
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static GstCaps *
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gst_wasapi_src_get_caps (GstBaseSrc * bsrc, GstCaps * filter)
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{
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GstWasapiSrc *self = GST_WASAPI_SRC (element);
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GstClock *clock;
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GST_OBJECT_LOCK (self);
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if (self->client_clock == NULL)
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goto wrong_state;
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clock = GST_CLOCK (gst_object_ref (self->clock));
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GST_OBJECT_UNLOCK (self);
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return clock;
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/* ERRORS */
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wrong_state:
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{
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GST_OBJECT_UNLOCK (self);
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GST_DEBUG_OBJECT (self, "IAudioClock not acquired");
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/* TODO: Implement */
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return NULL;
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}
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}
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static gboolean
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gst_wasapi_src_start (GstBaseSrc * src)
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gst_wasapi_src_open (GstAudioSrc * asrc)
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{
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GstWasapiSrc *self = GST_WASAPI_SRC (src);
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GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
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gboolean res = FALSE;
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IAudioClient * client = NULL;
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if (!gst_wasapi_util_get_default_device_client (GST_ELEMENT (self), TRUE, &client)) {
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GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ, (NULL),
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("Failed to get default device"));
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goto beach;
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}
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self->client = client;
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res = TRUE;
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beach:
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return res;
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}
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static gboolean
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gst_wasapi_src_close (GstAudioSrc * asrc)
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{
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GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
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if (self->client != NULL) {
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IUnknown_Release (self->client);
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self->client = NULL;
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}
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return TRUE;
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}
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static gboolean
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gst_wasapi_src_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
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{
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GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
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gboolean res = FALSE;
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IAudioClock *client_clock = NULL;
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guint64 client_clock_freq = 0;
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IAudioCaptureClient *capture_client = NULL;
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REFERENCE_TIME latency_rt, def_period, min_period;
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WAVEFORMATEXTENSIBLE format;
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HRESULT hr;
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if (!gst_wasapi_util_get_default_device_client (GST_ELEMENT (self),
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TRUE, self->rate, self->buffer_time, self->period_time, 0, &client,
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&self->latency))
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goto beach;
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hr = IAudioClient_GetService (client, &IID_IAudioClock,
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(void **) &client_clock);
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hr = IAudioClient_GetDevicePeriod (self->client, &def_period, &min_period);
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if (hr != S_OK) {
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GST_ERROR_OBJECT (self, "IAudioClient::GetService (IID_IAudioClock) "
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"failed");
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GST_ERROR_OBJECT (self, "IAudioClient::GetDevicePeriod () failed");
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goto beach;
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}
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gst_wasapi_util_audio_info_to_waveformatex (&spec->info, &format);
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self->info = spec->info;
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hr = IAudioClient_Initialize (self->client, AUDCLNT_SHAREMODE_SHARED, AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
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spec->buffer_time / 100, 0, (WAVEFORMATEX *) & format, NULL);
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if (hr != S_OK) {
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GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ, (NULL),
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("IAudioClient::Initialize () failed: %s",
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gst_wasapi_util_hresult_to_string (hr)));
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goto beach;
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}
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hr = IAudioClient_GetStreamLatency (self->client, &latency_rt);
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if (hr != S_OK) {
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GST_ERROR_OBJECT (self, "IAudioClient::GetStreamLatency () failed");
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goto beach;
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}
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GST_INFO_OBJECT (self, "default period: %d (%d ms), "
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"minimum period: %d (%d ms), "
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"latency: %d (%d ms)",
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(guint32) def_period, (guint32) def_period / 10000,
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(guint32) min_period, (guint32) min_period / 10000,
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(guint32) latency_rt, (guint32) latency_rt / 10000);
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/* FIXME: What to do with the latency? */
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hr = IAudioClient_SetEventHandle (self->client, self->event_handle);
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if (hr != S_OK) {
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GST_ERROR_OBJECT (self, "IAudioClient::SetEventHandle () failed");
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goto beach;
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}
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if (!gst_wasapi_util_get_clock (GST_ELEMENT (self), self->client,
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&client_clock)) {
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goto beach;
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}
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@ -194,21 +238,17 @@ gst_wasapi_src_start (GstBaseSrc * src)
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goto beach;
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}
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hr = IAudioClient_GetService (client, &IID_IAudioCaptureClient,
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(void **) &capture_client);
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if (hr != S_OK) {
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GST_ERROR_OBJECT (self, "IAudioClient::GetService "
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"(IID_IAudioCaptureClient) failed");
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if (!gst_wasapi_util_get_capture_client (GST_ELEMENT (self), self->client,
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&capture_client)) {
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goto beach;
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}
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hr = IAudioClient_Start (client);
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hr = IAudioClient_Start (self->client);
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if (hr != S_OK) {
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GST_ERROR_OBJECT (self, "IAudioClient::Start failed");
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goto beach;
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}
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self->client = client;
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self->client_clock = client_clock;
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self->client_clock_freq = client_clock_freq;
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self->capture_client = capture_client;
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@ -222,18 +262,15 @@ beach:
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if (client_clock != NULL)
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IUnknown_Release (client_clock);
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if (client != NULL)
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IUnknown_Release (client);
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}
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return res;
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}
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static gboolean
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gst_wasapi_src_stop (GstBaseSrc * src)
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gst_wasapi_src_unprepare (GstAudioSrc * asrc)
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{
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GstWasapiSrc *self = GST_WASAPI_SRC (src);
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GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
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if (self->client != NULL) {
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IAudioClient_Stop (self->client);
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@ -249,88 +286,34 @@ gst_wasapi_src_stop (GstBaseSrc * src)
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self->client_clock = NULL;
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}
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if (self->client != NULL) {
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IUnknown_Release (self->client);
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self->client = NULL;
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}
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return TRUE;
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}
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static gboolean
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gst_wasapi_src_query (GstBaseSrc * src, GstQuery * query)
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static guint
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gst_wasapi_src_read (GstAudioSrc * asrc, gpointer data, guint length,
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GstClockTime * timestamp)
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{
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GstWasapiSrc *self = GST_WASAPI_SRC (src);
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gboolean ret = FALSE;
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GST_DEBUG_OBJECT (self, "query for %s",
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gst_query_type_get_name (GST_QUERY_TYPE (query)));
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switch (GST_QUERY_TYPE (query)) {
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case GST_QUERY_LATENCY:{
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GstClockTime min_latency, max_latency;
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min_latency = self->latency + self->period_time;
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max_latency = min_latency;
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GST_DEBUG_OBJECT (self, "reporting latency of min %" GST_TIME_FORMAT
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" max %" GST_TIME_FORMAT,
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GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
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gst_query_set_latency (query, TRUE, min_latency, max_latency);
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ret = TRUE;
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break;
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}
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default:
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ret =
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GST_BASE_SRC_CLASS (gst_wasapi_src_parent_class)->query (src, query);
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break;
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}
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return ret;
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}
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static GstFlowReturn
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gst_wasapi_src_create (GstPushSrc * src, GstBuffer ** buf)
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{
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GstWasapiSrc *self = GST_WASAPI_SRC (src);
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GstFlowReturn ret = GST_FLOW_OK;
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GstClock *clock;
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GstClockTime timestamp, duration = self->period_time;
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GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
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HRESULT hr;
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gint16 *samples = NULL;
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guint32 nsamples_read = 0, nsamples;
|
||||
guint32 nsamples = 0, length_samples;
|
||||
DWORD flags = 0;
|
||||
guint64 devpos;
|
||||
guint i;
|
||||
GstMapInfo minfo;
|
||||
gint16 *dst;
|
||||
|
||||
GST_OBJECT_LOCK (self);
|
||||
clock = GST_ELEMENT_CLOCK (self);
|
||||
if (clock != NULL)
|
||||
gst_object_ref (clock);
|
||||
GST_OBJECT_UNLOCK (self);
|
||||
|
||||
if (clock != NULL && GST_CLOCK_TIME_IS_VALID (self->next_time)) {
|
||||
GstClockID id;
|
||||
|
||||
id = gst_clock_new_single_shot_id (clock, self->next_time);
|
||||
gst_clock_id_wait (id, NULL);
|
||||
gst_clock_id_unref (id);
|
||||
}
|
||||
WaitForSingleObject (self->event_handle, INFINITE);
|
||||
|
||||
do {
|
||||
hr = IAudioCaptureClient_GetBuffer (self->capture_client,
|
||||
(BYTE **) & samples, &nsamples_read, &flags, &devpos, NULL);
|
||||
(BYTE **) & samples, &nsamples, &flags, &devpos, NULL);
|
||||
}
|
||||
while (hr == AUDCLNT_S_BUFFER_EMPTY);
|
||||
|
||||
if (hr != S_OK) {
|
||||
GST_ERROR_OBJECT (self, "IAudioCaptureClient::GetBuffer () failed: %s",
|
||||
gst_wasapi_util_hresult_to_string (hr));
|
||||
ret = GST_FLOW_ERROR;
|
||||
length = 0;
|
||||
goto beach;
|
||||
}
|
||||
|
||||
|
|
@ -339,69 +322,58 @@ gst_wasapi_src_create (GstPushSrc * src, GstBuffer ** buf)
|
|||
devpos, (guint) flags);
|
||||
}
|
||||
|
||||
/* FIXME: Why do we get 1024 sometimes and not a multiple of
|
||||
* samples_per_buffer? Shouldn't WASAPI provide a DISCONT
|
||||
* flag if we read too slow?
|
||||
*/
|
||||
nsamples = nsamples_read;
|
||||
g_assert (nsamples >= self->samples_per_buffer);
|
||||
if (nsamples > self->samples_per_buffer) {
|
||||
GST_WARNING_OBJECT (self,
|
||||
"devpos %" G_GUINT64_FORMAT ": got %d samples, expected %d, clipping!",
|
||||
devpos, nsamples, self->samples_per_buffer);
|
||||
length_samples = length / self->info.bpf;
|
||||
nsamples = MIN (length_samples, nsamples);
|
||||
length = nsamples * self->info.bpf;
|
||||
|
||||
nsamples = self->samples_per_buffer;
|
||||
}
|
||||
|
||||
if (clock == NULL || clock == self->clock) {
|
||||
timestamp =
|
||||
gst_util_uint64_scale (devpos, GST_SECOND, self->client_clock_freq);
|
||||
} else {
|
||||
GstClockTime base_time;
|
||||
|
||||
timestamp = gst_clock_get_time (clock);
|
||||
|
||||
base_time = GST_ELEMENT_CAST (self)->base_time;
|
||||
if (timestamp > base_time)
|
||||
timestamp -= base_time;
|
||||
else
|
||||
timestamp = 0;
|
||||
|
||||
if (timestamp > duration)
|
||||
timestamp -= duration;
|
||||
else
|
||||
timestamp = 0;
|
||||
}
|
||||
|
||||
*buf = gst_buffer_new_and_alloc (nsamples * sizeof (gint16));
|
||||
|
||||
GST_BUFFER_OFFSET_END (*buf) = devpos + self->samples_per_buffer;
|
||||
GST_BUFFER_TIMESTAMP (*buf) = timestamp;
|
||||
GST_BUFFER_DURATION (*buf) = duration;
|
||||
|
||||
gst_buffer_map (*buf, &minfo, GST_MAP_WRITE);
|
||||
dst = (gint16 *) minfo.data;
|
||||
dst = (gint16 *) data;
|
||||
for (i = 0; i < nsamples; i++) {
|
||||
*dst = *samples;
|
||||
|
||||
samples += 2;
|
||||
dst++;
|
||||
}
|
||||
gst_buffer_unmap (*buf, &minfo);
|
||||
|
||||
hr = IAudioCaptureClient_ReleaseBuffer (self->capture_client, nsamples_read);
|
||||
hr = IAudioCaptureClient_ReleaseBuffer (self->capture_client, nsamples);
|
||||
if (hr != S_OK) {
|
||||
GST_ERROR_OBJECT (self, "IAudioCaptureClient::ReleaseBuffer () failed: %s",
|
||||
gst_wasapi_util_hresult_to_string (hr));
|
||||
ret = GST_FLOW_ERROR;
|
||||
goto beach;
|
||||
}
|
||||
|
||||
beach:
|
||||
if (clock != NULL)
|
||||
gst_object_unref (clock);
|
||||
|
||||
return ret;
|
||||
return length;
|
||||
}
|
||||
|
||||
static guint
|
||||
gst_wasapi_src_delay (GstAudioSrc * asrc)
|
||||
{
|
||||
/* FIXME: Implement */
|
||||
return 0;
|
||||
}
|
||||
|
||||
static void
|
||||
gst_wasapi_src_reset (GstAudioSrc * asrc)
|
||||
{
|
||||
GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
|
||||
HRESULT hr;
|
||||
|
||||
if (self->client) {
|
||||
hr = IAudioClient_Stop (self->client);
|
||||
if (hr != S_OK) {
|
||||
GST_ERROR_OBJECT (self, "IAudioClient::Stop () failed: %s",
|
||||
gst_wasapi_util_hresult_to_string (hr));
|
||||
return;
|
||||
}
|
||||
|
||||
hr = IAudioClient_Reset (self->client);
|
||||
if (hr != S_OK) {
|
||||
GST_ERROR_OBJECT (self, "IAudioClient::Reset () failed: %s",
|
||||
gst_wasapi_util_hresult_to_string (hr));
|
||||
return;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
static GstClockTime
|
||||
|
|
|
|||
|
|
@ -40,28 +40,20 @@ typedef struct _GstWasapiSrcClass GstWasapiSrcClass;
|
|||
|
||||
struct _GstWasapiSrc
|
||||
{
|
||||
GstPushSrc audio_src;
|
||||
GstAudioSrc parent;
|
||||
|
||||
GstClock * clock;
|
||||
|
||||
guint rate;
|
||||
GstClockTime buffer_time;
|
||||
GstClockTime period_time;
|
||||
GstClockTime latency;
|
||||
guint samples_per_buffer;
|
||||
GstAudioInfo info;
|
||||
|
||||
IAudioClient * client;
|
||||
IAudioClock * client_clock;
|
||||
guint64 client_clock_freq;
|
||||
IAudioCaptureClient * capture_client;
|
||||
|
||||
GstClockTime start_time;
|
||||
GstClockTime next_time;
|
||||
HANDLE event_handle;
|
||||
};
|
||||
|
||||
struct _GstWasapiSrcClass
|
||||
{
|
||||
GstPushSrcClass parent_class;
|
||||
GstAudioSrcClass parent_class;
|
||||
};
|
||||
|
||||
GType gst_wasapi_src_get_type (void);
|
||||
|
|
|
|||
|
|
@ -207,6 +207,50 @@ beach:
|
|||
return res;
|
||||
}
|
||||
|
||||
gboolean
|
||||
gst_wasapi_util_get_capture_client (GstElement * element, IAudioClient * client,
|
||||
IAudioCaptureClient ** ret_capture_client)
|
||||
{
|
||||
gboolean res = FALSE;
|
||||
HRESULT hr;
|
||||
IAudioCaptureClient *capture_client = NULL;
|
||||
|
||||
hr = IAudioClient_GetService (client, &IID_IAudioCaptureClient,
|
||||
(void **) &capture_client);
|
||||
if (hr != S_OK) {
|
||||
GST_ERROR_OBJECT (element, "IAudioClient::GetService "
|
||||
"(IID_IAudioCaptureClient) failed");
|
||||
goto beach;
|
||||
}
|
||||
|
||||
*ret_capture_client = capture_client;
|
||||
|
||||
beach:
|
||||
return res;
|
||||
}
|
||||
|
||||
gboolean
|
||||
gst_wasapi_util_get_clock (GstElement * element, IAudioClient * client,
|
||||
IAudioClock ** ret_clock)
|
||||
{
|
||||
gboolean res = FALSE;
|
||||
HRESULT hr;
|
||||
IAudioClock *clock = NULL;
|
||||
|
||||
hr = IAudioClient_GetService (client, &IID_IAudioClock,
|
||||
(void **) &clock);
|
||||
if (hr != S_OK) {
|
||||
GST_ERROR_OBJECT (element, "IAudioClient::GetService "
|
||||
"(IID_IAudioClock) failed");
|
||||
goto beach;
|
||||
}
|
||||
|
||||
*ret_clock = clock;
|
||||
|
||||
beach:
|
||||
return res;
|
||||
}
|
||||
|
||||
void
|
||||
gst_wasapi_util_audio_info_to_waveformatex (GstAudioInfo * info,
|
||||
WAVEFORMATEXTENSIBLE * format)
|
||||
|
|
|
|||
|
|
@ -22,6 +22,8 @@
|
|||
|
||||
#include <gst/gst.h>
|
||||
#include <gst/audio/audio.h>
|
||||
#include <gst/audio/gstaudiosrc.h>
|
||||
#include <gst/audio/gstaudiosink.h>
|
||||
|
||||
#include <audioclient.h>
|
||||
|
||||
|
|
@ -37,6 +39,14 @@ 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);
|
||||
|
||||
void
|
||||
gst_wasapi_util_audio_info_to_waveformatex (GstAudioInfo *info,
|
||||
WAVEFORMATEXTENSIBLE *format);
|
||||
|
|
|
|||
Loading…
Reference in a new issue