2 * Copyright (C) 2008 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
4 * This library is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Library General Public
6 * License as published by the Free Software Foundation; either
7 * version 2 of the License, or (at your option) any later version.
9 * This library is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Library General Public License for more details.
14 * You should have received a copy of the GNU Library General Public
15 * License along with this library; if not, write to the
16 * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
17 * Boston, MA 02111-1307, USA.
21 * SECTION:element-wasapisrc
23 * Provides audio capture from the Windows Audio Session API available with
27 * <title>Example pipelines</title>
29 * gst-launch-0.10 -v wasapisrc ! fakesink
30 * ]| Capture from the default audio device and render to fakesink.
34 #include "gstwasapisrc.h"
35 #include <gst/audio/gstaudioclock.h>
37 GST_DEBUG_CATEGORY_STATIC (gst_wasapi_src_debug);
38 #define GST_CAT_DEFAULT gst_wasapi_src_debug
40 static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
43 GST_STATIC_CAPS ("audio/x-raw-int, "
47 "channels = (int) 1, "
48 "signed = (boolean) TRUE, "
49 "endianness = (int) " G_STRINGIFY (G_BYTE_ORDER)));
51 static void gst_wasapi_src_dispose (GObject * object);
52 static void gst_wasapi_src_finalize (GObject * object);
54 static GstClock *gst_wasapi_src_provide_clock (GstElement * element);
56 static gboolean gst_wasapi_src_start (GstBaseSrc * src);
57 static gboolean gst_wasapi_src_stop (GstBaseSrc * src);
58 static gboolean gst_wasapi_src_query (GstBaseSrc * src, GstQuery * query);
60 static GstFlowReturn gst_wasapi_src_create (GstPushSrc * src, GstBuffer ** buf);
62 static GstClockTime gst_wasapi_src_get_time (GstClock * clock,
65 GST_BOILERPLATE (GstWasapiSrc, gst_wasapi_src, GstPushSrc, GST_TYPE_PUSH_SRC);
68 gst_wasapi_src_base_init (gpointer gclass)
70 GstElementClass *element_class = GST_ELEMENT_CLASS (gclass);
72 gst_element_class_add_pad_template (element_class,
73 gst_static_pad_template_get (&src_template));
74 gst_element_class_set_details_simple (element_class, "WasapiSrc",
76 "Stream audio from an audio capture device through WASAPI",
77 "Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>");
81 gst_wasapi_src_class_init (GstWasapiSrcClass * klass)
83 GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
84 GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
85 GstBaseSrcClass *gstbasesrc_class = GST_BASE_SRC_CLASS (klass);
86 GstPushSrcClass *gstpushsrc_class = GST_PUSH_SRC_CLASS (klass);
88 gobject_class->dispose = gst_wasapi_src_dispose;
89 gobject_class->finalize = gst_wasapi_src_finalize;
91 gstelement_class->provide_clock = gst_wasapi_src_provide_clock;
93 gstbasesrc_class->start = gst_wasapi_src_start;
94 gstbasesrc_class->stop = gst_wasapi_src_stop;
95 gstbasesrc_class->query = gst_wasapi_src_query;
97 gstpushsrc_class->create = gst_wasapi_src_create;
99 GST_DEBUG_CATEGORY_INIT (gst_wasapi_src_debug, "wasapisrc",
100 0, "Windows audio session API source");
104 gst_wasapi_src_init (GstWasapiSrc * self, GstWasapiSrcClass * gclass)
106 GstBaseSrc *basesrc = GST_BASE_SRC (self);
108 gst_base_src_set_format (basesrc, GST_FORMAT_TIME);
109 gst_base_src_set_live (basesrc, TRUE);
112 self->buffer_time = 20 * GST_MSECOND;
113 self->period_time = 20 * GST_MSECOND;
114 self->latency = GST_CLOCK_TIME_NONE;
115 self->samples_per_buffer = self->rate / (GST_SECOND / self->period_time);
117 self->start_time = GST_CLOCK_TIME_NONE;
118 self->next_time = GST_CLOCK_TIME_NONE;
120 #if GST_CHECK_VERSION(0, 10, 31) || (GST_CHECK_VERSION(0, 10, 30) && GST_VERSION_NANO > 0)
121 self->clock = gst_audio_clock_new_full ("GstWasapiSrcClock",
122 gst_wasapi_src_get_time, gst_object_ref (self),
123 (GDestroyNotify) gst_object_unref);
125 self->clock = gst_audio_clock_new ("GstWasapiSrcClock",
126 gst_wasapi_src_get_time, self);
133 gst_wasapi_src_dispose (GObject * object)
135 GstWasapiSrc *self = GST_WASAPI_SRC (object);
137 if (self->clock != NULL) {
138 gst_object_unref (self->clock);
142 G_OBJECT_CLASS (parent_class)->dispose (object);
146 gst_wasapi_src_finalize (GObject * object)
148 GstWasapiSrc *self = GST_WASAPI_SRC (object);
152 G_OBJECT_CLASS (parent_class)->finalize (object);
156 gst_wasapi_src_provide_clock (GstElement * element)
158 GstWasapiSrc *self = GST_WASAPI_SRC (element);
161 GST_OBJECT_LOCK (self);
163 if (self->client_clock == NULL)
166 clock = GST_CLOCK (gst_object_ref (self->clock));
168 GST_OBJECT_UNLOCK (self);
174 GST_OBJECT_UNLOCK (self);
175 GST_DEBUG_OBJECT (self, "IAudioClock not acquired");
181 gst_wasapi_src_start (GstBaseSrc * src)
183 GstWasapiSrc *self = GST_WASAPI_SRC (src);
184 gboolean res = FALSE;
185 IAudioClient *client = NULL;
186 IAudioClock *client_clock = NULL;
187 guint64 client_clock_freq = 0;
188 IAudioCaptureClient *capture_client = NULL;
191 if (!gst_wasapi_util_get_default_device_client (GST_ELEMENT (self),
192 TRUE, self->rate, self->buffer_time, self->period_time, 0, &client,
196 hr = IAudioClient_GetService (client, &IID_IAudioClock, &client_clock);
198 GST_ERROR_OBJECT (self, "IAudioClient::GetService (IID_IAudioClock) "
203 hr = IAudioClock_GetFrequency (client_clock, &client_clock_freq);
205 GST_ERROR_OBJECT (self, "IAudioClock::GetFrequency () failed");
209 hr = IAudioClient_GetService (client, &IID_IAudioCaptureClient,
212 GST_ERROR_OBJECT (self, "IAudioClient::GetService "
213 "(IID_IAudioCaptureClient) failed");
217 hr = IAudioClient_Start (client);
219 GST_ERROR_OBJECT (self, "IAudioClient::Start failed");
223 self->client = client;
224 self->client_clock = client_clock;
225 self->client_clock_freq = client_clock_freq;
226 self->capture_client = capture_client;
232 if (capture_client != NULL)
233 IUnknown_Release (capture_client);
235 if (client_clock != NULL)
236 IUnknown_Release (client_clock);
239 IUnknown_Release (client);
246 gst_wasapi_src_stop (GstBaseSrc * src)
248 GstWasapiSrc *self = GST_WASAPI_SRC (src);
250 if (self->client != NULL) {
251 IAudioClient_Stop (self->client);
254 if (self->capture_client != NULL) {
255 IUnknown_Release (self->capture_client);
256 self->capture_client = NULL;
259 if (self->client_clock != NULL) {
260 IUnknown_Release (self->client_clock);
261 self->client_clock = NULL;
264 if (self->client != NULL) {
265 IUnknown_Release (self->client);
273 gst_wasapi_src_query (GstBaseSrc * src, GstQuery * query)
275 GstWasapiSrc *self = GST_WASAPI_SRC (src);
276 gboolean ret = FALSE;
278 GST_DEBUG_OBJECT (self, "query for %s",
279 gst_query_type_get_name (GST_QUERY_TYPE (query)));
281 switch (GST_QUERY_TYPE (query)) {
282 case GST_QUERY_LATENCY:{
283 GstClockTime min_latency, max_latency;
285 min_latency = self->latency + self->period_time;
286 max_latency = min_latency;
288 GST_DEBUG_OBJECT (self, "reporting latency of min %" GST_TIME_FORMAT
289 " max %" GST_TIME_FORMAT,
290 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
292 gst_query_set_latency (query, TRUE, min_latency, max_latency);
298 ret = GST_BASE_SRC_CLASS (parent_class)->query (src, query);
306 gst_wasapi_src_create (GstPushSrc * src, GstBuffer ** buf)
308 GstWasapiSrc *self = GST_WASAPI_SRC (src);
309 GstFlowReturn ret = GST_FLOW_OK;
311 GstClockTime timestamp, duration = self->period_time;
313 gint16 *samples = NULL;
314 guint32 nsamples_read = 0, nsamples;
318 GST_OBJECT_LOCK (self);
319 clock = GST_ELEMENT_CLOCK (self);
321 gst_object_ref (clock);
322 GST_OBJECT_UNLOCK (self);
324 if (clock != NULL && GST_CLOCK_TIME_IS_VALID (self->next_time)) {
327 id = gst_clock_new_single_shot_id (clock, self->next_time);
328 gst_clock_id_wait (id, NULL);
329 gst_clock_id_unref (id);
333 hr = IAudioCaptureClient_GetBuffer (self->capture_client,
334 (BYTE **) & samples, &nsamples_read, &flags, &devpos, NULL);
336 while (hr == AUDCLNT_S_BUFFER_EMPTY);
339 GST_ERROR_OBJECT (self, "IAudioCaptureClient::GetBuffer () failed: %s",
340 gst_wasapi_util_hresult_to_string (hr));
341 ret = GST_FLOW_ERROR;
346 GST_WARNING_OBJECT (self, "devpos %" G_GUINT64_FORMAT ": flags=0x%08x",
350 /* FIXME: Why do we get 1024 sometimes and not a multiple of
351 * samples_per_buffer? Shouldn't WASAPI provide a DISCONT
352 * flag if we read too slow?
354 nsamples = nsamples_read;
355 g_assert (nsamples >= self->samples_per_buffer);
356 if (nsamples > self->samples_per_buffer) {
357 GST_WARNING_OBJECT (self,
358 "devpos %" G_GUINT64_FORMAT ": got %d samples, expected %d, clipping!",
359 devpos, nsamples, self->samples_per_buffer);
361 nsamples = self->samples_per_buffer;
364 if (clock == NULL || clock == self->clock) {
366 gst_util_uint64_scale (devpos, GST_SECOND, self->client_clock_freq);
368 GstClockTime base_time;
370 timestamp = gst_clock_get_time (clock);
372 base_time = GST_ELEMENT_CAST (self)->base_time;
373 if (timestamp > base_time)
374 timestamp -= base_time;
378 if (timestamp > duration)
379 timestamp -= duration;
384 ret = gst_pad_alloc_buffer_and_set_caps (GST_BASE_SRC_PAD (self),
386 nsamples * sizeof (gint16), GST_PAD_CAPS (GST_BASE_SRC_PAD (self)), buf);
388 if (ret == GST_FLOW_OK) {
392 GST_BUFFER_OFFSET_END (*buf) = devpos + self->samples_per_buffer;
393 GST_BUFFER_TIMESTAMP (*buf) = timestamp;
394 GST_BUFFER_DURATION (*buf) = duration;
396 dst = (gint16 *) GST_BUFFER_DATA (*buf);
397 for (i = 0; i < nsamples; i++) {
405 hr = IAudioCaptureClient_ReleaseBuffer (self->capture_client, nsamples_read);
407 GST_ERROR_OBJECT (self, "IAudioCaptureClient::ReleaseBuffer () failed: %s",
408 gst_wasapi_util_hresult_to_string (hr));
409 ret = GST_FLOW_ERROR;
415 gst_object_unref (clock);
421 gst_wasapi_src_get_time (GstClock * clock, gpointer user_data)
423 GstWasapiSrc *self = GST_WASAPI_SRC (user_data);
428 if (G_UNLIKELY (self->client_clock == NULL))
429 return GST_CLOCK_TIME_NONE;
431 hr = IAudioClock_GetPosition (self->client_clock, &devpos, NULL);
432 if (G_UNLIKELY (hr != S_OK))
433 return GST_CLOCK_TIME_NONE;
435 result = gst_util_uint64_scale_int (devpos, GST_SECOND,
436 self->client_clock_freq);
439 GST_DEBUG_OBJECT (self, "devpos = %" G_GUINT64_FORMAT
440 " frequency = %" G_GUINT64_FORMAT
441 " result = %" G_GUINT64_FORMAT " ms",
442 devpos, self->client_clock_freq, GST_TIME_AS_MSECONDS (result));