2 * Copyright (C) 2008 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
3 * Copyright (C) 2018 Centricular Ltd.
4 * Author: Nirbheek Chauhan <nirbheek@centricular.com>
6 * This library is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Library General Public
8 * License as published by the Free Software Foundation; either
9 * version 2 of the License, or (at your option) any later version.
11 * This library is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Library General Public License for more details.
16 * You should have received a copy of the GNU Library General Public
17 * License along with this library; if not, write to the
18 * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
19 * Boston, MA 02110-1301, USA.
23 * SECTION:element-wasapisrc
26 * Provides audio capture from the Windows Audio Session API available with
29 * ## Example pipelines
31 * gst-launch-1.0 -v wasapisrc ! fakesink
32 * ]| Capture from the default audio device and render to fakesink.
39 #include "gstwasapisrc.h"
43 GST_DEBUG_CATEGORY_STATIC (gst_wasapi_src_debug);
44 #define GST_CAT_DEFAULT gst_wasapi_src_debug
46 static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
49 GST_STATIC_CAPS (GST_WASAPI_STATIC_CAPS));
51 #define DEFAULT_ROLE GST_WASAPI_DEVICE_ROLE_CONSOLE
52 #define DEFAULT_EXCLUSIVE FALSE
53 #define DEFAULT_LOW_LATENCY FALSE
64 static void gst_wasapi_src_dispose (GObject * object);
65 static void gst_wasapi_src_finalize (GObject * object);
66 static void gst_wasapi_src_set_property (GObject * object, guint prop_id,
67 const GValue * value, GParamSpec * pspec);
68 static void gst_wasapi_src_get_property (GObject * object, guint prop_id,
69 GValue * value, GParamSpec * pspec);
71 static GstCaps *gst_wasapi_src_get_caps (GstBaseSrc * bsrc, GstCaps * filter);
73 static gboolean gst_wasapi_src_open (GstAudioSrc * asrc);
74 static gboolean gst_wasapi_src_close (GstAudioSrc * asrc);
75 static gboolean gst_wasapi_src_prepare (GstAudioSrc * asrc,
76 GstAudioRingBufferSpec * spec);
77 static gboolean gst_wasapi_src_unprepare (GstAudioSrc * asrc);
78 static guint gst_wasapi_src_read (GstAudioSrc * asrc, gpointer data,
79 guint length, GstClockTime * timestamp);
80 static guint gst_wasapi_src_delay (GstAudioSrc * asrc);
81 static void gst_wasapi_src_reset (GstAudioSrc * asrc);
83 static GstClockTime gst_wasapi_src_get_time (GstClock * clock,
86 #define gst_wasapi_src_parent_class parent_class
87 G_DEFINE_TYPE (GstWasapiSrc, gst_wasapi_src, GST_TYPE_AUDIO_SRC);
90 gst_wasapi_src_class_init (GstWasapiSrcClass * klass)
92 GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
93 GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
94 GstBaseSrcClass *gstbasesrc_class = GST_BASE_SRC_CLASS (klass);
95 GstAudioSrcClass *gstaudiosrc_class = GST_AUDIO_SRC_CLASS (klass);
97 gobject_class->dispose = gst_wasapi_src_dispose;
98 gobject_class->finalize = gst_wasapi_src_finalize;
99 gobject_class->set_property = gst_wasapi_src_set_property;
100 gobject_class->get_property = gst_wasapi_src_get_property;
102 g_object_class_install_property (gobject_class,
104 g_param_spec_enum ("role", "Role",
105 "Role of the device: communications, multimedia, etc",
106 GST_WASAPI_DEVICE_TYPE_ROLE, DEFAULT_ROLE, G_PARAM_READWRITE |
107 G_PARAM_STATIC_STRINGS | GST_PARAM_MUTABLE_READY));
109 g_object_class_install_property (gobject_class,
111 g_param_spec_string ("device", "Device",
112 "WASAPI playback device as a GUID string",
113 NULL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
115 g_object_class_install_property (gobject_class,
117 g_param_spec_boolean ("exclusive", "Exclusive mode",
118 "Open the device in exclusive mode",
119 DEFAULT_EXCLUSIVE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
121 g_object_class_install_property (gobject_class,
123 g_param_spec_boolean ("low-latency", "Low latency",
124 "Optimize all settings for lowest latency",
125 DEFAULT_LOW_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
127 gst_element_class_add_static_pad_template (gstelement_class, &src_template);
128 gst_element_class_set_static_metadata (gstelement_class, "WasapiSrc",
130 "Stream audio from an audio capture device through WASAPI",
131 "Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>");
133 gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_wasapi_src_get_caps);
135 gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_wasapi_src_open);
136 gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_wasapi_src_close);
137 gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_wasapi_src_read);
138 gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_wasapi_src_prepare);
139 gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_wasapi_src_unprepare);
140 gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_wasapi_src_delay);
141 gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_wasapi_src_reset);
143 GST_DEBUG_CATEGORY_INIT (gst_wasapi_src_debug, "wasapisrc",
144 0, "Windows audio session API source");
148 gst_wasapi_src_init (GstWasapiSrc * self)
150 /* override with a custom clock */
151 if (GST_AUDIO_BASE_SRC (self)->clock)
152 gst_object_unref (GST_AUDIO_BASE_SRC (self)->clock);
154 GST_AUDIO_BASE_SRC (self)->clock = gst_audio_clock_new ("GstWasapiSrcClock",
155 gst_wasapi_src_get_time, gst_object_ref (self),
156 (GDestroyNotify) gst_object_unref);
158 self->event_handle = CreateEvent (NULL, FALSE, FALSE, NULL);
164 gst_wasapi_src_dispose (GObject * object)
166 GstWasapiSrc *self = GST_WASAPI_SRC (object);
168 if (self->event_handle != NULL) {
169 CloseHandle (self->event_handle);
170 self->event_handle = NULL;
173 if (self->client_clock != NULL) {
174 IUnknown_Release (self->client_clock);
175 self->client_clock = NULL;
178 if (self->client != NULL) {
179 IUnknown_Release (self->client);
183 if (self->capture_client != NULL) {
184 IUnknown_Release (self->capture_client);
185 self->capture_client = NULL;
188 G_OBJECT_CLASS (parent_class)->dispose (object);
192 gst_wasapi_src_finalize (GObject * object)
194 GstWasapiSrc *self = GST_WASAPI_SRC (object);
196 g_clear_pointer (&self->mix_format, CoTaskMemFree);
200 g_clear_pointer (&self->cached_caps, gst_caps_unref);
201 g_clear_pointer (&self->positions, g_free);
202 g_clear_pointer (&self->device_strid, g_free);
204 G_OBJECT_CLASS (parent_class)->finalize (object);
208 gst_wasapi_src_set_property (GObject * object, guint prop_id,
209 const GValue * value, GParamSpec * pspec)
211 GstWasapiSrc *self = GST_WASAPI_SRC (object);
215 self->role = gst_wasapi_device_role_to_erole (g_value_get_enum (value));
219 const gchar *device = g_value_get_string (value);
220 g_free (self->device_strid);
222 device ? g_utf8_to_utf16 (device, -1, NULL, NULL, NULL) : NULL;
226 self->sharemode = g_value_get_boolean (value)
227 ? AUDCLNT_SHAREMODE_EXCLUSIVE : AUDCLNT_SHAREMODE_SHARED;
229 case PROP_LOW_LATENCY:
230 self->low_latency = g_value_get_boolean (value);
233 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
239 gst_wasapi_src_get_property (GObject * object, guint prop_id,
240 GValue * value, GParamSpec * pspec)
242 GstWasapiSrc *self = GST_WASAPI_SRC (object);
246 g_value_set_enum (value, gst_wasapi_erole_to_device_role (self->role));
249 g_value_take_string (value, self->device_strid ?
250 g_utf16_to_utf8 (self->device_strid, -1, NULL, NULL, NULL) : NULL);
253 g_value_set_boolean (value,
254 self->sharemode == AUDCLNT_SHAREMODE_EXCLUSIVE);
256 case PROP_LOW_LATENCY:
257 g_value_set_boolean (value, self->low_latency);
260 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
266 gst_wasapi_src_get_caps (GstBaseSrc * bsrc, GstCaps * filter)
268 GstWasapiSrc *self = GST_WASAPI_SRC (bsrc);
269 WAVEFORMATEX *format = NULL;
270 GstCaps *caps = NULL;
272 GST_DEBUG_OBJECT (self, "entering get caps");
274 if (self->cached_caps) {
275 caps = gst_caps_ref (self->cached_caps);
277 GstCaps *template_caps;
280 template_caps = gst_pad_get_pad_template_caps (bsrc->srcpad);
283 gst_wasapi_src_open (GST_AUDIO_SRC (bsrc));
285 ret = gst_wasapi_util_get_device_format (GST_ELEMENT (self),
286 self->sharemode, self->device, self->client, &format);
288 GST_ELEMENT_ERROR (self, STREAM, FORMAT, (NULL),
289 ("failed to detect format"));
293 gst_wasapi_util_parse_waveformatex ((WAVEFORMATEXTENSIBLE *) format,
294 template_caps, &caps, &self->positions);
296 GST_ELEMENT_ERROR (self, STREAM, FORMAT, (NULL), ("unknown format"));
301 gchar *pos_str = gst_audio_channel_positions_to_string (self->positions,
303 GST_INFO_OBJECT (self, "positions are: %s", pos_str);
307 self->mix_format = format;
308 gst_caps_replace (&self->cached_caps, caps);
309 gst_caps_unref (template_caps);
314 gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
315 gst_caps_unref (caps);
319 GST_DEBUG_OBJECT (self, "returning caps %" GST_PTR_FORMAT, caps);
326 gst_wasapi_src_open (GstAudioSrc * asrc)
328 GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
329 gboolean res = FALSE;
330 IAudioClient *client = NULL;
331 IMMDevice *device = NULL;
336 /* FIXME: Switching the default device does not switch the stream to it,
337 * even if the old device was unplugged. We need to handle this somehow.
338 * For example, perhaps we should automatically switch to the new device if
339 * the default device is changed and a device isn't explicitly selected. */
340 if (!gst_wasapi_util_get_device_client (GST_ELEMENT (self), TRUE,
341 self->role, self->device_strid, &device, &client)) {
342 if (!self->device_strid)
343 GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ, (NULL),
344 ("Failed to get default device"));
346 GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ, (NULL),
347 ("Failed to open device %S", self->device_strid));
351 self->client = client;
352 self->device = device;
361 gst_wasapi_src_close (GstAudioSrc * asrc)
363 GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
365 if (self->device != NULL) {
366 IUnknown_Release (self->device);
370 if (self->client != NULL) {
371 IUnknown_Release (self->client);
379 gst_wasapi_src_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
381 GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
382 gboolean res = FALSE;
383 REFERENCE_TIME latency_rt;
384 guint bpf, rate, devicep_frames, buffer_frames;
387 if (self->sharemode == AUDCLNT_SHAREMODE_SHARED &&
388 gst_wasapi_util_have_audioclient3 ()) {
389 if (!gst_wasapi_util_initialize_audioclient3 (GST_ELEMENT (self), spec,
390 (IAudioClient3 *) self->client, self->mix_format, self->low_latency,
394 if (!gst_wasapi_util_initialize_audioclient (GST_ELEMENT (self), spec,
395 self->client, self->mix_format, self->sharemode, self->low_latency,
400 bpf = GST_AUDIO_INFO_BPF (&spec->info);
401 rate = GST_AUDIO_INFO_RATE (&spec->info);
403 /* Total size in frames of the allocated buffer that we will read from */
404 hr = IAudioClient_GetBufferSize (self->client, &buffer_frames);
405 HR_FAILED_GOTO (hr, IAudioClient::GetBufferSize, beach);
407 GST_INFO_OBJECT (self, "buffer size is %i frames, device period is %i "
408 "frames, bpf is %i bytes, rate is %i Hz", buffer_frames,
409 devicep_frames, bpf, rate);
411 /* Actual latency-time/buffer-time will be different now */
412 spec->segsize = devicep_frames * bpf;
414 /* We need a minimum of 2 segments to ensure glitch-free playback */
415 spec->segtotal = MAX (self->buffer_frame_count * bpf / spec->segsize, 2);
417 GST_INFO_OBJECT (self, "segsize is %i, segtotal is %i", spec->segsize,
420 /* Get WASAPI latency for logging */
421 hr = IAudioClient_GetStreamLatency (self->client, &latency_rt);
422 HR_FAILED_GOTO (hr, IAudioClient::GetStreamLatency, beach);
424 GST_INFO_OBJECT (self, "wasapi stream latency: %" G_GINT64_FORMAT " (%"
425 G_GINT64_FORMAT " ms)", latency_rt, latency_rt / 10000);
427 /* Set the event handler which will trigger reads */
428 hr = IAudioClient_SetEventHandle (self->client, self->event_handle);
429 HR_FAILED_GOTO (hr, IAudioClient::SetEventHandle, beach);
431 /* Get the clock and the clock freq */
432 if (!gst_wasapi_util_get_clock (GST_ELEMENT (self), self->client,
433 &self->client_clock))
436 hr = IAudioClock_GetFrequency (self->client_clock, &self->client_clock_freq);
437 HR_FAILED_GOTO (hr, IAudioClock::GetFrequency, beach);
439 /* Get capture source client and start it up */
440 if (!gst_wasapi_util_get_capture_client (GST_ELEMENT (self), self->client,
441 &self->capture_client)) {
445 hr = IAudioClient_Start (self->client);
446 HR_FAILED_GOTO (hr, IAudioClock::Start, beach);
448 gst_audio_ring_buffer_set_channel_positions (GST_AUDIO_BASE_SRC
449 (self)->ringbuffer, self->positions);
451 /* Increase the thread priority to reduce glitches */
452 self->thread_priority_handle = gst_wasapi_util_set_thread_characteristics ();
456 /* unprepare() is not called if prepare() fails, but we want it to be, so call
457 * it manually when needed */
459 gst_wasapi_src_unprepare (asrc);
465 gst_wasapi_src_unprepare (GstAudioSrc * asrc)
467 GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
469 if (self->sharemode == AUDCLNT_SHAREMODE_EXCLUSIVE)
472 if (self->thread_priority_handle != NULL) {
473 gst_wasapi_util_revert_thread_characteristics
474 (self->thread_priority_handle);
475 self->thread_priority_handle = NULL;
478 if (self->client != NULL) {
479 IAudioClient_Stop (self->client);
482 if (self->capture_client != NULL) {
483 IUnknown_Release (self->capture_client);
484 self->capture_client = NULL;
487 if (self->client_clock != NULL) {
488 IUnknown_Release (self->client_clock);
489 self->client_clock = NULL;
492 self->client_clock_freq = 0;
498 gst_wasapi_src_read (GstAudioSrc * asrc, gpointer data, guint length,
499 GstClockTime * timestamp)
501 GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
504 guint wanted = length;
508 guint have_frames, n_frames, want_frames, read_len;
510 /* Wait for data to become available */
511 WaitForSingleObject (self->event_handle, INFINITE);
513 hr = IAudioCaptureClient_GetBuffer (self->capture_client,
514 (BYTE **) & from, &have_frames, &flags, NULL, NULL);
516 gchar *msg = gst_wasapi_util_hresult_to_string (hr);
517 if (hr == AUDCLNT_S_BUFFER_EMPTY)
518 GST_WARNING_OBJECT (self, "IAudioCaptureClient::GetBuffer failed: %s"
521 GST_ERROR_OBJECT (self, "IAudioCaptureClient::GetBuffer failed: %s",
529 GST_INFO_OBJECT (self, "buffer flags=%#08x", (guint) flags);
531 /* XXX: How do we handle AUDCLNT_BUFFERFLAGS_SILENT? We're supposed to write
532 * out silence when that flag is set? See:
533 * https://msdn.microsoft.com/en-us/library/windows/desktop/dd370800(v=vs.85).aspx */
535 if (flags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY)
536 GST_WARNING_OBJECT (self, "WASAPI reported glitch in buffer");
538 want_frames = wanted / self->mix_format->nBlockAlign;
540 /* If GetBuffer is returning more frames than we can handle, all we can do is
541 * hope that this is temporary and that things will settle down later. */
542 if (G_UNLIKELY (have_frames > want_frames))
543 GST_WARNING_OBJECT (self, "captured too many frames: have %i, want %i",
544 have_frames, want_frames);
546 /* Only copy data that will fit into the allocated buffer of size @length */
547 n_frames = MIN (have_frames, want_frames);
548 read_len = n_frames * self->mix_format->nBlockAlign;
551 guint bpf = self->mix_format->nBlockAlign;
552 GST_DEBUG_OBJECT (self, "have: %i (%i bytes), can read: %i (%i bytes), "
553 "will read: %i (%i bytes)", have_frames, have_frames * bpf,
554 want_frames, wanted, n_frames, read_len);
557 memcpy (data, from, read_len);
560 /* Always release all captured buffers if we've captured any at all */
561 hr = IAudioCaptureClient_ReleaseBuffer (self->capture_client, have_frames);
562 HR_FAILED_AND (hr, IAudioClock::ReleaseBuffer, goto beach);
572 gst_wasapi_src_delay (GstAudioSrc * asrc)
574 GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
578 hr = IAudioClient_GetCurrentPadding (self->client, &delay);
579 HR_FAILED_RET (hr, IAudioClock::GetCurrentPadding, 0);
585 gst_wasapi_src_reset (GstAudioSrc * asrc)
587 GstWasapiSrc *self = GST_WASAPI_SRC (asrc);
593 hr = IAudioClient_Stop (self->client);
594 HR_FAILED_RET (hr, IAudioClock::Stop,);
596 hr = IAudioClient_Reset (self->client);
597 HR_FAILED_RET (hr, IAudioClock::Reset,);
601 gst_wasapi_src_get_time (GstClock * clock, gpointer user_data)
603 GstWasapiSrc *self = GST_WASAPI_SRC (user_data);
608 if (G_UNLIKELY (self->client_clock == NULL))
609 return GST_CLOCK_TIME_NONE;
611 hr = IAudioClock_GetPosition (self->client_clock, &devpos, NULL);
612 HR_FAILED_RET (hr, IAudioClock::GetPosition, GST_CLOCK_TIME_NONE);
614 result = gst_util_uint64_scale_int (devpos, GST_SECOND,
615 self->client_clock_freq);
618 GST_DEBUG_OBJECT (self, "devpos = %" G_GUINT64_FORMAT
619 " frequency = %" G_GUINT64_FORMAT
620 " result = %" G_GUINT64_FORMAT " ms",
621 devpos, self->client_clock_freq, GST_TIME_AS_MSECONDS (result));