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);
71 static GstElementDetails element_details = {
74 "Stream audio from an audio capture device through WASAPI",
75 "Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>"
78 gst_element_class_add_pad_template (element_class,
79 gst_static_pad_template_get (&src_template));
80 gst_element_class_set_details (element_class, &element_details);
84 gst_wasapi_src_class_init (GstWasapiSrcClass * klass)
86 GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
87 GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
88 GstBaseSrcClass *gstbasesrc_class = GST_BASE_SRC_CLASS (klass);
89 GstPushSrcClass *gstpushsrc_class = GST_PUSH_SRC_CLASS (klass);
91 gobject_class->dispose = gst_wasapi_src_dispose;
92 gobject_class->finalize = gst_wasapi_src_finalize;
94 gstelement_class->provide_clock = gst_wasapi_src_provide_clock;
96 gstbasesrc_class->start = gst_wasapi_src_start;
97 gstbasesrc_class->stop = gst_wasapi_src_stop;
98 gstbasesrc_class->query = gst_wasapi_src_query;
100 gstpushsrc_class->create = gst_wasapi_src_create;
102 GST_DEBUG_CATEGORY_INIT (gst_wasapi_src_debug, "wasapisrc",
103 0, "Windows audio session API source");
107 gst_wasapi_src_init (GstWasapiSrc * self, GstWasapiSrcClass * gclass)
109 GstBaseSrc *basesrc = GST_BASE_SRC (self);
111 gst_base_src_set_format (basesrc, GST_FORMAT_TIME);
112 gst_base_src_set_live (basesrc, TRUE);
115 self->buffer_time = 20 * GST_MSECOND;
116 self->period_time = 20 * GST_MSECOND;
117 self->latency = GST_CLOCK_TIME_NONE;
118 self->samples_per_buffer = self->rate / (GST_SECOND / self->period_time);
120 self->start_time = GST_CLOCK_TIME_NONE;
121 self->next_time = GST_CLOCK_TIME_NONE;
123 self->clock = gst_audio_clock_new ("GstWasapiSrcClock",
124 gst_wasapi_src_get_time, self);
130 gst_wasapi_src_dispose (GObject * object)
132 GstWasapiSrc *self = GST_WASAPI_SRC (object);
134 if (self->clock != NULL) {
135 gst_object_unref (self->clock);
139 G_OBJECT_CLASS (parent_class)->dispose (object);
143 gst_wasapi_src_finalize (GObject * object)
145 GstWasapiSrc *self = GST_WASAPI_SRC (object);
149 G_OBJECT_CLASS (parent_class)->finalize (object);
153 gst_wasapi_src_provide_clock (GstElement * element)
155 GstWasapiSrc *self = GST_WASAPI_SRC (element);
158 GST_OBJECT_LOCK (self);
160 if (self->client_clock == NULL)
163 clock = GST_CLOCK (gst_object_ref (self->clock));
165 GST_OBJECT_UNLOCK (self);
171 GST_OBJECT_UNLOCK (self);
172 GST_DEBUG_OBJECT (self, "IAudioClock not acquired");
178 gst_wasapi_src_start (GstBaseSrc * src)
180 GstWasapiSrc *self = GST_WASAPI_SRC (src);
181 gboolean res = FALSE;
182 IAudioClient *client = NULL;
183 IAudioClock *client_clock = NULL;
184 guint64 client_clock_freq = 0;
185 IAudioCaptureClient *capture_client = NULL;
188 if (!gst_wasapi_util_get_default_device_client (GST_ELEMENT (self),
189 TRUE, self->rate, self->buffer_time, self->period_time, 0, &client,
193 hr = IAudioClient_GetService (client, &IID_IAudioClock, &client_clock);
195 GST_ERROR_OBJECT (self, "IAudioClient::GetService (IID_IAudioClock) "
200 hr = IAudioClock_GetFrequency (client_clock, &client_clock_freq);
202 GST_ERROR_OBJECT (self, "IAudioClock::GetFrequency () failed");
206 hr = IAudioClient_GetService (client, &IID_IAudioCaptureClient,
209 GST_ERROR_OBJECT (self, "IAudioClient::GetService "
210 "(IID_IAudioCaptureClient) failed");
214 hr = IAudioClient_Start (client);
216 GST_ERROR_OBJECT (self, "IAudioClient::Start failed");
220 self->client = client;
221 self->client_clock = client_clock;
222 self->client_clock_freq = client_clock_freq;
223 self->capture_client = capture_client;
229 if (capture_client != NULL)
230 IUnknown_Release (capture_client);
232 if (client_clock != NULL)
233 IUnknown_Release (client_clock);
236 IUnknown_Release (client);
243 gst_wasapi_src_stop (GstBaseSrc * src)
245 GstWasapiSrc *self = GST_WASAPI_SRC (src);
247 if (self->client != NULL) {
248 IAudioClient_Stop (self->client);
251 if (self->capture_client != NULL) {
252 IUnknown_Release (self->capture_client);
253 self->capture_client = NULL;
256 if (self->client_clock != NULL) {
257 IUnknown_Release (self->client_clock);
258 self->client_clock = NULL;
261 if (self->client != NULL) {
262 IUnknown_Release (self->client);
270 gst_wasapi_src_query (GstBaseSrc * src, GstQuery * query)
272 GstWasapiSrc *self = GST_WASAPI_SRC (src);
273 gboolean ret = FALSE;
275 GST_DEBUG_OBJECT (self, "query for %s",
276 gst_query_type_get_name (GST_QUERY_TYPE (query)));
278 switch (GST_QUERY_TYPE (query)) {
279 case GST_QUERY_LATENCY:{
280 GstClockTime min_latency, max_latency;
282 min_latency = self->latency + self->period_time;
283 max_latency = min_latency;
285 GST_DEBUG_OBJECT (self, "reporting latency of min %" GST_TIME_FORMAT
286 " max %" GST_TIME_FORMAT,
287 GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
289 gst_query_set_latency (query, TRUE, min_latency, max_latency);
295 ret = GST_BASE_SRC_CLASS (parent_class)->query (src, query);
303 gst_wasapi_src_create (GstPushSrc * src, GstBuffer ** buf)
305 GstWasapiSrc *self = GST_WASAPI_SRC (src);
306 GstFlowReturn ret = GST_FLOW_OK;
308 GstClockTime timestamp, duration = self->period_time;
310 gint16 *samples = NULL;
311 guint32 nsamples_read = 0, nsamples;
315 GST_OBJECT_LOCK (self);
316 clock = GST_ELEMENT_CLOCK (self);
318 gst_object_ref (clock);
319 GST_OBJECT_UNLOCK (self);
321 if (clock != NULL && GST_CLOCK_TIME_IS_VALID (self->next_time)) {
324 id = gst_clock_new_single_shot_id (clock, self->next_time);
325 gst_clock_id_wait (id, NULL);
326 gst_clock_id_unref (id);
330 hr = IAudioCaptureClient_GetBuffer (self->capture_client,
331 (BYTE **) & samples, &nsamples_read, &flags, &devpos, NULL);
333 while (hr == AUDCLNT_S_BUFFER_EMPTY);
336 GST_ERROR_OBJECT (self, "IAudioCaptureClient::GetBuffer () failed: %s",
337 gst_wasapi_util_hresult_to_string (hr));
338 ret = GST_FLOW_ERROR;
343 GST_WARNING_OBJECT (self, "devpos %" G_GUINT64_FORMAT ": flags=0x%08x",
347 /* FIXME: Why do we get 1024 sometimes and not a multiple of
348 * samples_per_buffer? Shouldn't WASAPI provide a DISCONT
349 * flag if we read too slow?
351 nsamples = nsamples_read;
352 g_assert (nsamples >= self->samples_per_buffer);
353 if (nsamples > self->samples_per_buffer) {
354 GST_WARNING_OBJECT (self,
355 "devpos %" G_GUINT64_FORMAT ": got %d samples, expected %d, clipping!",
356 devpos, nsamples, self->samples_per_buffer);
358 nsamples = self->samples_per_buffer;
361 if (clock == NULL || clock == self->clock) {
363 gst_util_uint64_scale (devpos, GST_SECOND, self->client_clock_freq);
365 GstClockTime base_time;
367 timestamp = gst_clock_get_time (clock);
369 base_time = GST_ELEMENT_CAST (self)->base_time;
370 if (timestamp > base_time)
371 timestamp -= base_time;
375 if (timestamp > duration)
376 timestamp -= duration;
381 ret = gst_pad_alloc_buffer_and_set_caps (GST_BASE_SRC_PAD (self),
383 nsamples * sizeof (gint16), GST_PAD_CAPS (GST_BASE_SRC_PAD (self)), buf);
385 if (ret == GST_FLOW_OK) {
389 GST_BUFFER_OFFSET_END (*buf) = devpos + self->samples_per_buffer;
390 GST_BUFFER_TIMESTAMP (*buf) = timestamp;
391 GST_BUFFER_DURATION (*buf) = duration;
393 dst = (gint16 *) GST_BUFFER_DATA (*buf);
394 for (i = 0; i < nsamples; i++) {
402 hr = IAudioCaptureClient_ReleaseBuffer (self->capture_client, nsamples_read);
404 GST_ERROR_OBJECT (self, "IAudioCaptureClient::ReleaseBuffer () failed: %s",
405 gst_wasapi_util_hresult_to_string (hr));
406 ret = GST_FLOW_ERROR;
412 gst_object_unref (clock);
418 gst_wasapi_src_get_time (GstClock * clock, gpointer user_data)
420 GstWasapiSrc *self = GST_WASAPI_SRC (user_data);
425 if (G_UNLIKELY (self->client_clock == NULL))
426 return GST_CLOCK_TIME_NONE;
428 hr = IAudioClock_GetPosition (self->client_clock, &devpos, NULL);
429 if (G_UNLIKELY (hr != S_OK))
430 return GST_CLOCK_TIME_NONE;
432 result = gst_util_uint64_scale_int (devpos, GST_SECOND,
433 self->client_clock_freq);
436 GST_DEBUG_OBJECT (self, "devpos = %" G_GUINT64_FORMAT
437 " frequency = %" G_GUINT64_FORMAT
438 " result = %" G_GUINT64_FORMAT " ms",
439 devpos, self->client_clock_freq, GST_TIME_AS_MSECONDS (result));