wasapisrc: Port to GstAudioSrc

This commit is contained in:
Sebastian Dröge 2013-03-28 16:52:26 +01:00
parent 1445438a8b
commit 363aa90a10
7 changed files with 226 additions and 202 deletions

View file

@ -1,6 +1,7 @@
plugin_LTLIBRARIES = libgstwasapi.la plugin_LTLIBRARIES = libgstwasapi.la
libgstwasapi_la_SOURCES = gstwasapi.c \ libgstwasapi_la_SOURCES = gstwasapi.c \
gstwasapisrc.c \
gstwasapisink.c \ gstwasapisink.c \
gstwasapiutil.c gstwasapiutil.c
@ -11,7 +12,7 @@ libgstwasapi_la_LIBADD = $(GST_LIBS) $(GST_BASE_LIBS) $(GST_PLUGINS_BASE_LIBS) \
libgstwasapi_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) libgstwasapi_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstwasapi_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS) libgstwasapi_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
noinst_HEADERS = \ noinst_HEADERS = gstwasapisrc.h \
gstwasapisink.h \ gstwasapisink.h \
gstwasapiutil.h gstwasapiutil.h

View file

@ -22,12 +22,16 @@
#endif #endif
#include "gstwasapisink.h" #include "gstwasapisink.h"
#include "gstwasapisrc.h"
static gboolean static gboolean
plugin_init (GstPlugin * plugin) plugin_init (GstPlugin * plugin)
{ {
gst_element_register (plugin, "wasapisink", GST_RANK_NONE, gst_element_register (plugin, "wasapisink", GST_RANK_NONE,
GST_TYPE_WASAPI_SINK); GST_TYPE_WASAPI_SINK);
gst_element_register (plugin, "wasapisrc", GST_RANK_NONE,
GST_TYPE_WASAPI_SRC);
return TRUE; return TRUE;
} }

View file

@ -1,5 +1,7 @@
/* /*
* Copyright (C) 2008 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com> * Copyright (C) 2008 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
* Copyright (C) 2013 Collabora Ltd.
* Author: Sebastian Dröge <sebastian.droege@collabora.co.uk>
* *
* This library is free software; you can redistribute it and/or * This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public * modify it under the terms of the GNU Library General Public
@ -62,7 +64,6 @@ static gint gst_wasapi_sink_write (GstAudioSink * asink,
static guint gst_wasapi_sink_delay (GstAudioSink * asink); static guint gst_wasapi_sink_delay (GstAudioSink * asink);
static void gst_wasapi_sink_reset (GstAudioSink * asink); static void gst_wasapi_sink_reset (GstAudioSink * asink);
G_DEFINE_TYPE (GstWasapiSink, gst_wasapi_sink, GST_TYPE_AUDIO_SINK); G_DEFINE_TYPE (GstWasapiSink, gst_wasapi_sink, GST_TYPE_AUDIO_SINK);
static void static void

View file

@ -35,7 +35,6 @@
#endif #endif
#include "gstwasapisrc.h" #include "gstwasapisrc.h"
#include <gst/audio/gstaudioclock.h>
GST_DEBUG_CATEGORY_STATIC (gst_wasapi_src_debug); GST_DEBUG_CATEGORY_STATIC (gst_wasapi_src_debug);
#define GST_CAT_DEFAULT gst_wasapi_src_debug #define GST_CAT_DEFAULT gst_wasapi_src_debug
@ -46,23 +45,25 @@ static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_STATIC_CAPS ("audio/x-raw, " GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) S16LE, " "format = (string) S16LE, "
"layout = (string) interleaved, " "layout = (string) interleaved, "
"rate = (int) 8000, " "channels = (int) 1")); "rate = (int) 44100, " "channels = (int) 1"));
static void gst_wasapi_src_dispose (GObject * object); static void gst_wasapi_src_dispose (GObject * object);
static void gst_wasapi_src_finalize (GObject * object); static void gst_wasapi_src_finalize (GObject * object);
static GstClock *gst_wasapi_src_provide_clock (GstElement * element); static GstCaps * gst_wasapi_src_get_caps (GstBaseSrc * bsrc, GstCaps * filter);
static gboolean gst_wasapi_src_start (GstBaseSrc * src); static gboolean gst_wasapi_src_open (GstAudioSrc * asrc);
static gboolean gst_wasapi_src_stop (GstBaseSrc * src); static gboolean gst_wasapi_src_close (GstAudioSrc * asrc);
static gboolean gst_wasapi_src_query (GstBaseSrc * src, GstQuery * query); static gboolean gst_wasapi_src_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec);
static gboolean gst_wasapi_src_unprepare (GstAudioSrc * asrc);
static GstFlowReturn gst_wasapi_src_create (GstPushSrc * src, GstBuffer ** buf); static guint gst_wasapi_src_read (GstAudioSrc * asrc, gpointer data, guint length, GstClockTime * timestamp);
static guint gst_wasapi_src_delay (GstAudioSrc * asrc);
static void gst_wasapi_src_reset (GstAudioSrc * asrc);
static GstClockTime gst_wasapi_src_get_time (GstClock * clock, static GstClockTime gst_wasapi_src_get_time (GstClock * clock,
gpointer user_data); gpointer user_data);
G_DEFINE_TYPE (GstWasapiSrc, gst_wasapi_src, GST_TYPE_PUSH_SRC); G_DEFINE_TYPE (GstWasapiSrc, gst_wasapi_src, GST_TYPE_AUDIO_SRC);
static void static void
gst_wasapi_src_class_init (GstWasapiSrcClass * klass) gst_wasapi_src_class_init (GstWasapiSrcClass * klass)
@ -70,13 +71,11 @@ gst_wasapi_src_class_init (GstWasapiSrcClass * klass)
GObjectClass *gobject_class = G_OBJECT_CLASS (klass); GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass); GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
GstBaseSrcClass *gstbasesrc_class = GST_BASE_SRC_CLASS (klass); GstBaseSrcClass *gstbasesrc_class = GST_BASE_SRC_CLASS (klass);
GstPushSrcClass *gstpushsrc_class = GST_PUSH_SRC_CLASS (klass); GstAudioSrcClass *gstaudiosrc_class = GST_AUDIO_SRC_CLASS (klass);
gobject_class->dispose = gst_wasapi_src_dispose; gobject_class->dispose = gst_wasapi_src_dispose;
gobject_class->finalize = gst_wasapi_src_finalize; gobject_class->finalize = gst_wasapi_src_finalize;
gstelement_class->provide_clock = gst_wasapi_src_provide_clock;
gst_element_class_add_pad_template (gstelement_class, gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&src_template)); gst_static_pad_template_get (&src_template));
gst_element_class_set_static_metadata (gstelement_class, "WasapiSrc", gst_element_class_set_static_metadata (gstelement_class, "WasapiSrc",
@ -84,11 +83,16 @@ gst_wasapi_src_class_init (GstWasapiSrcClass * klass)
"Stream audio from an audio capture device through WASAPI", "Stream audio from an audio capture device through WASAPI",
"Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>"); "Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>");
gstbasesrc_class->start = gst_wasapi_src_start; gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_wasapi_src_get_caps);
gstbasesrc_class->stop = gst_wasapi_src_stop;
gstbasesrc_class->query = gst_wasapi_src_query;
gstpushsrc_class->create = gst_wasapi_src_create; gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_wasapi_src_open);
gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_wasapi_src_close);
gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_wasapi_src_read);
gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_wasapi_src_prepare);
gstaudiosrc_class->unprepare =
GST_DEBUG_FUNCPTR (gst_wasapi_src_unprepare);
gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_wasapi_src_delay);
gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_wasapi_src_reset);
GST_DEBUG_CATEGORY_INIT (gst_wasapi_src_debug, "wasapisrc", GST_DEBUG_CATEGORY_INIT (gst_wasapi_src_debug, "wasapisrc",
0, "Windows audio session API source"); 0, "Windows audio session API source");
@ -97,24 +101,16 @@ gst_wasapi_src_class_init (GstWasapiSrcClass * klass)
static void static void
gst_wasapi_src_init (GstWasapiSrc * self) gst_wasapi_src_init (GstWasapiSrc * self)
{ {
GstBaseSrc *basesrc = GST_BASE_SRC (self); /* override with a custom clock */
if (GST_AUDIO_BASE_SRC (self)->clock)
gst_object_unref (GST_AUDIO_BASE_SRC (self)->clock);
gst_base_src_set_format (basesrc, GST_FORMAT_TIME); GST_AUDIO_BASE_SRC (self)->clock = gst_audio_clock_new ("GstWasapiSrcClock",
gst_base_src_set_live (basesrc, TRUE);
self->rate = 8000;
self->buffer_time = 20 * GST_MSECOND;
self->period_time = 20 * GST_MSECOND;
self->latency = GST_CLOCK_TIME_NONE;
self->samples_per_buffer = self->rate / (GST_SECOND / self->period_time);
self->start_time = GST_CLOCK_TIME_NONE;
self->next_time = GST_CLOCK_TIME_NONE;
self->clock = gst_audio_clock_new ("GstWasapiSrcClock",
gst_wasapi_src_get_time, gst_object_ref (self), gst_wasapi_src_get_time, gst_object_ref (self),
(GDestroyNotify) gst_object_unref); (GDestroyNotify) gst_object_unref);
self->event_handle = CreateEvent (NULL, FALSE, FALSE, NULL);
CoInitialize (NULL); CoInitialize (NULL);
} }
@ -123,9 +119,9 @@ gst_wasapi_src_dispose (GObject * object)
{ {
GstWasapiSrc *self = GST_WASAPI_SRC (object); GstWasapiSrc *self = GST_WASAPI_SRC (object);
if (self->clock != NULL) { if (self->event_handle != NULL) {
gst_object_unref (self->clock); CloseHandle (self->event_handle);
self->clock = NULL; self->event_handle = NULL;
} }
G_OBJECT_CLASS (gst_wasapi_src_parent_class)->dispose (object); G_OBJECT_CLASS (gst_wasapi_src_parent_class)->dispose (object);
@ -139,52 +135,100 @@ gst_wasapi_src_finalize (GObject * object)
G_OBJECT_CLASS (gst_wasapi_src_parent_class)->finalize (object); G_OBJECT_CLASS (gst_wasapi_src_parent_class)->finalize (object);
} }
static GstClock * static GstCaps *
gst_wasapi_src_provide_clock (GstElement * element) gst_wasapi_src_get_caps (GstBaseSrc * bsrc, GstCaps * filter)
{ {
GstWasapiSrc *self = GST_WASAPI_SRC (element); /* TODO: Implement */
GstClock *clock; return NULL;
GST_OBJECT_LOCK (self);
if (self->client_clock == NULL)
goto wrong_state;
clock = GST_CLOCK (gst_object_ref (self->clock));
GST_OBJECT_UNLOCK (self);
return clock;
/* ERRORS */
wrong_state:
{
GST_OBJECT_UNLOCK (self);
GST_DEBUG_OBJECT (self, "IAudioClock not acquired");
return NULL;
}
} }
static gboolean static gboolean
gst_wasapi_src_start (GstBaseSrc * src) gst_wasapi_src_open (GstAudioSrc * asrc)
{ {
GstWasapiSrc *self = GST_WASAPI_SRC (src); GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
gboolean res = FALSE;
IAudioClient * client = NULL;
if (!gst_wasapi_util_get_default_device_client (GST_ELEMENT (self), TRUE, &client)) {
GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ, (NULL),
("Failed to get default device"));
goto beach;
}
self->client = client;
res = TRUE;
beach:
return res;
}
static gboolean
gst_wasapi_src_close (GstAudioSrc * asrc)
{
GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
if (self->client != NULL) {
IUnknown_Release (self->client);
self->client = NULL;
}
return TRUE;
}
static gboolean
gst_wasapi_src_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
{
GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
gboolean res = FALSE; gboolean res = FALSE;
IAudioClient *client = NULL;
IAudioClock *client_clock = NULL; IAudioClock *client_clock = NULL;
guint64 client_clock_freq = 0; guint64 client_clock_freq = 0;
IAudioCaptureClient *capture_client = NULL; IAudioCaptureClient *capture_client = NULL;
REFERENCE_TIME latency_rt, def_period, min_period;
WAVEFORMATEXTENSIBLE format;
HRESULT hr; HRESULT hr;
if (!gst_wasapi_util_get_default_device_client (GST_ELEMENT (self), hr = IAudioClient_GetDevicePeriod (self->client, &def_period, &min_period);
TRUE, self->rate, self->buffer_time, self->period_time, 0, &client,
&self->latency))
goto beach;
hr = IAudioClient_GetService (client, &IID_IAudioClock,
(void **) &client_clock);
if (hr != S_OK) { if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::GetService (IID_IAudioClock) " GST_ERROR_OBJECT (self, "IAudioClient::GetDevicePeriod () failed");
"failed"); goto beach;
}
gst_wasapi_util_audio_info_to_waveformatex (&spec->info, &format);
self->info = spec->info;
hr = IAudioClient_Initialize (self->client, AUDCLNT_SHAREMODE_SHARED, AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
spec->buffer_time / 100, 0, (WAVEFORMATEX *) & format, NULL);
if (hr != S_OK) {
GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ, (NULL),
("IAudioClient::Initialize () failed: %s",
gst_wasapi_util_hresult_to_string (hr)));
goto beach;
}
hr = IAudioClient_GetStreamLatency (self->client, &latency_rt);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::GetStreamLatency () failed");
goto beach;
}
GST_INFO_OBJECT (self, "default period: %d (%d ms), "
"minimum period: %d (%d ms), "
"latency: %d (%d ms)",
(guint32) def_period, (guint32) def_period / 10000,
(guint32) min_period, (guint32) min_period / 10000,
(guint32) latency_rt, (guint32) latency_rt / 10000);
/* FIXME: What to do with the latency? */
hr = IAudioClient_SetEventHandle (self->client, self->event_handle);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::SetEventHandle () failed");
goto beach;
}
if (!gst_wasapi_util_get_clock (GST_ELEMENT (self), self->client,
&client_clock)) {
goto beach; goto beach;
} }
@ -194,21 +238,17 @@ gst_wasapi_src_start (GstBaseSrc * src)
goto beach; goto beach;
} }
hr = IAudioClient_GetService (client, &IID_IAudioCaptureClient, if (!gst_wasapi_util_get_capture_client (GST_ELEMENT (self), self->client,
(void **) &capture_client); &capture_client)) {
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::GetService "
"(IID_IAudioCaptureClient) failed");
goto beach; goto beach;
} }
hr = IAudioClient_Start (client); hr = IAudioClient_Start (self->client);
if (hr != S_OK) { if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::Start failed"); GST_ERROR_OBJECT (self, "IAudioClient::Start failed");
goto beach; goto beach;
} }
self->client = client;
self->client_clock = client_clock; self->client_clock = client_clock;
self->client_clock_freq = client_clock_freq; self->client_clock_freq = client_clock_freq;
self->capture_client = capture_client; self->capture_client = capture_client;
@ -222,18 +262,15 @@ beach:
if (client_clock != NULL) if (client_clock != NULL)
IUnknown_Release (client_clock); IUnknown_Release (client_clock);
if (client != NULL)
IUnknown_Release (client);
} }
return res; return res;
} }
static gboolean static gboolean
gst_wasapi_src_stop (GstBaseSrc * src) gst_wasapi_src_unprepare (GstAudioSrc * asrc)
{ {
GstWasapiSrc *self = GST_WASAPI_SRC (src); GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
if (self->client != NULL) { if (self->client != NULL) {
IAudioClient_Stop (self->client); IAudioClient_Stop (self->client);
@ -249,88 +286,34 @@ gst_wasapi_src_stop (GstBaseSrc * src)
self->client_clock = NULL; self->client_clock = NULL;
} }
if (self->client != NULL) {
IUnknown_Release (self->client);
self->client = NULL;
}
return TRUE; return TRUE;
} }
static gboolean static guint
gst_wasapi_src_query (GstBaseSrc * src, GstQuery * query) gst_wasapi_src_read (GstAudioSrc * asrc, gpointer data, guint length,
GstClockTime * timestamp)
{ {
GstWasapiSrc *self = GST_WASAPI_SRC (src); GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
gboolean ret = FALSE;
GST_DEBUG_OBJECT (self, "query for %s",
gst_query_type_get_name (GST_QUERY_TYPE (query)));
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_LATENCY:{
GstClockTime min_latency, max_latency;
min_latency = self->latency + self->period_time;
max_latency = min_latency;
GST_DEBUG_OBJECT (self, "reporting latency of min %" GST_TIME_FORMAT
" max %" GST_TIME_FORMAT,
GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
gst_query_set_latency (query, TRUE, min_latency, max_latency);
ret = TRUE;
break;
}
default:
ret =
GST_BASE_SRC_CLASS (gst_wasapi_src_parent_class)->query (src, query);
break;
}
return ret;
}
static GstFlowReturn
gst_wasapi_src_create (GstPushSrc * src, GstBuffer ** buf)
{
GstWasapiSrc *self = GST_WASAPI_SRC (src);
GstFlowReturn ret = GST_FLOW_OK;
GstClock *clock;
GstClockTime timestamp, duration = self->period_time;
HRESULT hr; HRESULT hr;
gint16 *samples = NULL; gint16 *samples = NULL;
guint32 nsamples_read = 0, nsamples; guint32 nsamples = 0, length_samples;
DWORD flags = 0; DWORD flags = 0;
guint64 devpos; guint64 devpos;
guint i; guint i;
GstMapInfo minfo;
gint16 *dst; gint16 *dst;
GST_OBJECT_LOCK (self); WaitForSingleObject (self->event_handle, INFINITE);
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);
}
do { do {
hr = IAudioCaptureClient_GetBuffer (self->capture_client, 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); while (hr == AUDCLNT_S_BUFFER_EMPTY);
if (hr != S_OK) { if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioCaptureClient::GetBuffer () failed: %s", GST_ERROR_OBJECT (self, "IAudioCaptureClient::GetBuffer () failed: %s",
gst_wasapi_util_hresult_to_string (hr)); gst_wasapi_util_hresult_to_string (hr));
ret = GST_FLOW_ERROR; length = 0;
goto beach; goto beach;
} }
@ -339,69 +322,58 @@ gst_wasapi_src_create (GstPushSrc * src, GstBuffer ** buf)
devpos, (guint) flags); devpos, (guint) flags);
} }
/* FIXME: Why do we get 1024 sometimes and not a multiple of length_samples = length / self->info.bpf;
* samples_per_buffer? Shouldn't WASAPI provide a DISCONT nsamples = MIN (length_samples, nsamples);
* flag if we read too slow? length = nsamples * self->info.bpf;
*/
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);
nsamples = self->samples_per_buffer; dst = (gint16 *) data;
}
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;
for (i = 0; i < nsamples; i++) { for (i = 0; i < nsamples; i++) {
*dst = *samples; *dst = *samples;
samples += 2; samples += 2;
dst++; 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) { if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioCaptureClient::ReleaseBuffer () failed: %s", GST_ERROR_OBJECT (self, "IAudioCaptureClient::ReleaseBuffer () failed: %s",
gst_wasapi_util_hresult_to_string (hr)); gst_wasapi_util_hresult_to_string (hr));
ret = GST_FLOW_ERROR;
goto beach; goto beach;
} }
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 static GstClockTime

View file

@ -40,28 +40,20 @@ typedef struct _GstWasapiSrcClass GstWasapiSrcClass;
struct _GstWasapiSrc struct _GstWasapiSrc
{ {
GstPushSrc audio_src; GstAudioSrc parent;
GstClock * clock; GstAudioInfo info;
guint rate;
GstClockTime buffer_time;
GstClockTime period_time;
GstClockTime latency;
guint samples_per_buffer;
IAudioClient * client; IAudioClient * client;
IAudioClock * client_clock; IAudioClock * client_clock;
guint64 client_clock_freq; guint64 client_clock_freq;
IAudioCaptureClient * capture_client; IAudioCaptureClient * capture_client;
HANDLE event_handle;
GstClockTime start_time;
GstClockTime next_time;
}; };
struct _GstWasapiSrcClass struct _GstWasapiSrcClass
{ {
GstPushSrcClass parent_class; GstAudioSrcClass parent_class;
}; };
GType gst_wasapi_src_get_type (void); GType gst_wasapi_src_get_type (void);

View file

@ -207,6 +207,50 @@ beach:
return res; 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 void
gst_wasapi_util_audio_info_to_waveformatex (GstAudioInfo * info, gst_wasapi_util_audio_info_to_waveformatex (GstAudioInfo * info,
WAVEFORMATEXTENSIBLE * format) WAVEFORMATEXTENSIBLE * format)

View file

@ -22,6 +22,8 @@
#include <gst/gst.h> #include <gst/gst.h>
#include <gst/audio/audio.h> #include <gst/audio/audio.h>
#include <gst/audio/gstaudiosrc.h>
#include <gst/audio/gstaudiosink.h>
#include <audioclient.h> #include <audioclient.h>
@ -37,6 +39,14 @@ gboolean gst_wasapi_util_get_render_client (GstElement * element,
IAudioClient *client, IAudioClient *client,
IAudioRenderClient ** ret_render_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 void
gst_wasapi_util_audio_info_to_waveformatex (GstAudioInfo *info, gst_wasapi_util_audio_info_to_waveformatex (GstAudioInfo *info,
WAVEFORMATEXTENSIBLE *format); WAVEFORMATEXTENSIBLE *format);