/* * Copyright (C) 2008 Ole André Vadla Ravnås * Copyright (C) 2018 Centricular Ltd. * Author: Nirbheek Chauhan * * This library is free software; you can redistribute it and/or * modify it under the terms of the GNU Library General Public * License as published by the Free Software Foundation; either * version 2 of the License, or (at your option) any later version. * * This library is distributed in the hope that it will be useful, * but WITHOUT ANY WARRANTY; without even the implied warranty of * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU * Library General Public License for more details. * * You should have received a copy of the GNU Library General Public * License along with this library; if not, write to the * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor, * Boston, MA 02110-1301, USA. */ /** * SECTION:element-wasapisrc * @title: wasapisrc * * Provides audio capture from the Windows Audio Session API available with * Vista and newer. * * ## Example pipelines * |[ * gst-launch-1.0 -v wasapisrc ! fakesink * ]| Capture from the default audio device and render to fakesink. * * |[ * gst-launch-1.0 -v wasapisrc low-latency=true ! fakesink * ]| Capture from the default audio device with the minimum possible latency and render to fakesink. * */ #ifdef HAVE_CONFIG_H # include #endif #include "gstwasapisrc.h" #include GST_DEBUG_CATEGORY_STATIC (gst_wasapi_src_debug); #define GST_CAT_DEFAULT gst_wasapi_src_debug static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS (GST_WASAPI_STATIC_CAPS)); #define DEFAULT_ROLE GST_WASAPI_DEVICE_ROLE_CONSOLE #define DEFAULT_LOOPBACK FALSE #define DEFAULT_EXCLUSIVE FALSE #define DEFAULT_LOW_LATENCY FALSE #define DEFAULT_AUDIOCLIENT3 FALSE enum { PROP_0, PROP_ROLE, PROP_DEVICE, PROP_LOOPBACK, PROP_EXCLUSIVE, PROP_LOW_LATENCY, PROP_AUDIOCLIENT3 }; static void gst_wasapi_src_dispose (GObject * object); static void gst_wasapi_src_finalize (GObject * object); static void gst_wasapi_src_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_wasapi_src_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static GstCaps *gst_wasapi_src_get_caps (GstBaseSrc * bsrc, GstCaps * filter); static gboolean gst_wasapi_src_open (GstAudioSrc * asrc); static gboolean gst_wasapi_src_close (GstAudioSrc * asrc); static gboolean gst_wasapi_src_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec); static gboolean gst_wasapi_src_unprepare (GstAudioSrc * asrc); 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, gpointer user_data); #define gst_wasapi_src_parent_class parent_class G_DEFINE_TYPE (GstWasapiSrc, gst_wasapi_src, GST_TYPE_AUDIO_SRC); static void gst_wasapi_src_class_init (GstWasapiSrcClass * klass) { GObjectClass *gobject_class = G_OBJECT_CLASS (klass); GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass); GstBaseSrcClass *gstbasesrc_class = GST_BASE_SRC_CLASS (klass); GstAudioSrcClass *gstaudiosrc_class = GST_AUDIO_SRC_CLASS (klass); gobject_class->dispose = gst_wasapi_src_dispose; gobject_class->finalize = gst_wasapi_src_finalize; gobject_class->set_property = gst_wasapi_src_set_property; gobject_class->get_property = gst_wasapi_src_get_property; g_object_class_install_property (gobject_class, PROP_ROLE, g_param_spec_enum ("role", "Role", "Role of the device: communications, multimedia, etc", GST_WASAPI_DEVICE_TYPE_ROLE, DEFAULT_ROLE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | GST_PARAM_MUTABLE_READY)); g_object_class_install_property (gobject_class, PROP_DEVICE, g_param_spec_string ("device", "Device", "WASAPI playback device as a GUID string", NULL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_LOOPBACK, g_param_spec_boolean ("loopback", "Loopback recording", "Open the sink device for loopback recording", DEFAULT_LOOPBACK, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_EXCLUSIVE, g_param_spec_boolean ("exclusive", "Exclusive mode", "Open the device in exclusive mode", DEFAULT_EXCLUSIVE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_LOW_LATENCY, g_param_spec_boolean ("low-latency", "Low latency", "Optimize all settings for lowest latency. Always safe to enable.", DEFAULT_LOW_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_AUDIOCLIENT3, g_param_spec_boolean ("use-audioclient3", "Use the AudioClient3 API", "Whether to use the Windows 10 AudioClient3 API when available", DEFAULT_AUDIOCLIENT3, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); gst_element_class_add_static_pad_template (gstelement_class, &src_template); gst_element_class_set_static_metadata (gstelement_class, "WasapiSrc", "Source/Audio", "Stream audio from an audio capture device through WASAPI", "Nirbheek Chauhan , " "Ole André Vadla Ravnås "); gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_wasapi_src_get_caps); 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", 0, "Windows audio session API source"); } static void gst_wasapi_src_init (GstWasapiSrc * self) { /* override with a custom clock */ if (GST_AUDIO_BASE_SRC (self)->clock) gst_object_unref (GST_AUDIO_BASE_SRC (self)->clock); GST_AUDIO_BASE_SRC (self)->clock = gst_audio_clock_new ("GstWasapiSrcClock", gst_wasapi_src_get_time, gst_object_ref (self), (GDestroyNotify) gst_object_unref); self->role = DEFAULT_ROLE; self->sharemode = AUDCLNT_SHAREMODE_SHARED; self->loopback = DEFAULT_LOOPBACK; self->low_latency = DEFAULT_LOW_LATENCY; self->try_audioclient3 = DEFAULT_AUDIOCLIENT3; self->event_handle = CreateEvent (NULL, FALSE, FALSE, NULL); self->client_needs_restart = FALSE; CoInitialize (NULL); } static void gst_wasapi_src_dispose (GObject * object) { GstWasapiSrc *self = GST_WASAPI_SRC (object); if (self->event_handle != NULL) { CloseHandle (self->event_handle); self->event_handle = NULL; } if (self->client_clock != NULL) { IUnknown_Release (self->client_clock); self->client_clock = NULL; } if (self->client != NULL) { IUnknown_Release (self->client); self->client = NULL; } if (self->capture_client != NULL) { IUnknown_Release (self->capture_client); self->capture_client = NULL; } G_OBJECT_CLASS (parent_class)->dispose (object); } static void gst_wasapi_src_finalize (GObject * object) { GstWasapiSrc *self = GST_WASAPI_SRC (object); g_clear_pointer (&self->mix_format, CoTaskMemFree); CoUninitialize (); g_clear_pointer (&self->cached_caps, gst_caps_unref); g_clear_pointer (&self->positions, g_free); g_clear_pointer (&self->device_strid, g_free); G_OBJECT_CLASS (parent_class)->finalize (object); } static void gst_wasapi_src_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstWasapiSrc *self = GST_WASAPI_SRC (object); switch (prop_id) { case PROP_ROLE: self->role = gst_wasapi_device_role_to_erole (g_value_get_enum (value)); break; case PROP_DEVICE: { const gchar *device = g_value_get_string (value); g_free (self->device_strid); self->device_strid = device ? g_utf8_to_utf16 (device, -1, NULL, NULL, NULL) : NULL; break; } case PROP_LOOPBACK: self->loopback = g_value_get_boolean (value); break; case PROP_EXCLUSIVE: self->sharemode = g_value_get_boolean (value) ? AUDCLNT_SHAREMODE_EXCLUSIVE : AUDCLNT_SHAREMODE_SHARED; break; case PROP_LOW_LATENCY: self->low_latency = g_value_get_boolean (value); break; case PROP_AUDIOCLIENT3: self->try_audioclient3 = g_value_get_boolean (value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void gst_wasapi_src_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstWasapiSrc *self = GST_WASAPI_SRC (object); switch (prop_id) { case PROP_ROLE: g_value_set_enum (value, gst_wasapi_erole_to_device_role (self->role)); break; case PROP_DEVICE: g_value_take_string (value, self->device_strid ? g_utf16_to_utf8 (self->device_strid, -1, NULL, NULL, NULL) : NULL); break; case PROP_LOOPBACK: g_value_set_boolean (value, self->loopback); break; case PROP_EXCLUSIVE: g_value_set_boolean (value, self->sharemode == AUDCLNT_SHAREMODE_EXCLUSIVE); break; case PROP_LOW_LATENCY: g_value_set_boolean (value, self->low_latency); break; case PROP_AUDIOCLIENT3: g_value_set_boolean (value, self->try_audioclient3); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static gboolean gst_wasapi_src_can_audioclient3 (GstWasapiSrc * self) { if (self->sharemode == AUDCLNT_SHAREMODE_SHARED && self->try_audioclient3 && gst_wasapi_util_have_audioclient3 ()) return TRUE; return FALSE; } static GstCaps * gst_wasapi_src_get_caps (GstBaseSrc * bsrc, GstCaps * filter) { GstWasapiSrc *self = GST_WASAPI_SRC (bsrc); WAVEFORMATEX *format = NULL; GstCaps *caps = NULL; GST_DEBUG_OBJECT (self, "entering get caps"); if (self->cached_caps) { caps = gst_caps_ref (self->cached_caps); } else { GstCaps *template_caps; gboolean ret; template_caps = gst_pad_get_pad_template_caps (bsrc->srcpad); if (!self->client) { caps = template_caps; goto out; } ret = gst_wasapi_util_get_device_format (GST_ELEMENT (self), self->sharemode, self->device, self->client, &format); if (!ret) { GST_ELEMENT_ERROR (self, STREAM, FORMAT, (NULL), ("failed to detect format")); gst_caps_unref (template_caps); return NULL; } gst_wasapi_util_parse_waveformatex ((WAVEFORMATEXTENSIBLE *) format, template_caps, &caps, &self->positions); if (caps == NULL) { GST_ELEMENT_ERROR (self, STREAM, FORMAT, (NULL), ("unknown format")); gst_caps_unref (template_caps); return NULL; } { gchar *pos_str = gst_audio_channel_positions_to_string (self->positions, format->nChannels); GST_INFO_OBJECT (self, "positions are: %s", pos_str); g_free (pos_str); } self->mix_format = format; gst_caps_replace (&self->cached_caps, caps); gst_caps_unref (template_caps); } if (filter) { GstCaps *filtered = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST); gst_caps_unref (caps); caps = filtered; } out: GST_DEBUG_OBJECT (self, "returning caps %" GST_PTR_FORMAT, caps); return caps; } static gboolean gst_wasapi_src_open (GstAudioSrc * asrc) { GstWasapiSrc *self = GST_WASAPI_SRC (asrc); gboolean res = FALSE; IAudioClient *client = NULL; IMMDevice *device = NULL; if (self->client) return TRUE; /* FIXME: Switching the default device does not switch the stream to it, * even if the old device was unplugged. We need to handle this somehow. * For example, perhaps we should automatically switch to the new device if * the default device is changed and a device isn't explicitly selected. */ if (!gst_wasapi_util_get_device_client (GST_ELEMENT (self), self->loopback ? eRender : eCapture, self->role, self->device_strid, &device, &client)) { if (!self->device_strid) GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ, (NULL), ("Failed to get default device")); else GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ, (NULL), ("Failed to open device %S", self->device_strid)); goto beach; } self->client = client; self->device = device; res = TRUE; beach: return res; } static gboolean gst_wasapi_src_close (GstAudioSrc * asrc) { GstWasapiSrc *self = GST_WASAPI_SRC (asrc); if (self->device != NULL) { IUnknown_Release (self->device); self->device = NULL; } 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; REFERENCE_TIME latency_rt; guint bpf, rate, devicep_frames, buffer_frames; HRESULT hr; if (gst_wasapi_src_can_audioclient3 (self)) { if (!gst_wasapi_util_initialize_audioclient3 (GST_ELEMENT (self), spec, (IAudioClient3 *) self->client, self->mix_format, self->low_latency, self->loopback, &devicep_frames)) goto beach; } else { if (!gst_wasapi_util_initialize_audioclient (GST_ELEMENT (self), spec, self->client, self->mix_format, self->sharemode, self->low_latency, self->loopback, &devicep_frames)) goto beach; } bpf = GST_AUDIO_INFO_BPF (&spec->info); rate = GST_AUDIO_INFO_RATE (&spec->info); /* Total size in frames of the allocated buffer that we will read from */ hr = IAudioClient_GetBufferSize (self->client, &buffer_frames); HR_FAILED_GOTO (hr, IAudioClient::GetBufferSize, beach); GST_INFO_OBJECT (self, "buffer size is %i frames, device period is %i " "frames, bpf is %i bytes, rate is %i Hz", buffer_frames, devicep_frames, bpf, rate); /* Actual latency-time/buffer-time will be different now */ spec->segsize = devicep_frames * bpf; /* We need a minimum of 2 segments to ensure glitch-free playback */ spec->segtotal = MAX (self->buffer_frame_count * bpf / spec->segsize, 2); GST_INFO_OBJECT (self, "segsize is %i, segtotal is %i", spec->segsize, spec->segtotal); /* Get WASAPI latency for logging */ hr = IAudioClient_GetStreamLatency (self->client, &latency_rt); HR_FAILED_GOTO (hr, IAudioClient::GetStreamLatency, beach); GST_INFO_OBJECT (self, "wasapi stream latency: %" G_GINT64_FORMAT " (%" G_GINT64_FORMAT " ms)", latency_rt, latency_rt / 10000); /* Set the event handler which will trigger reads */ hr = IAudioClient_SetEventHandle (self->client, self->event_handle); HR_FAILED_GOTO (hr, IAudioClient::SetEventHandle, beach); /* Get the clock and the clock freq */ if (!gst_wasapi_util_get_clock (GST_ELEMENT (self), self->client, &self->client_clock)) goto beach; hr = IAudioClock_GetFrequency (self->client_clock, &self->client_clock_freq); HR_FAILED_GOTO (hr, IAudioClock::GetFrequency, beach); /* Get capture source client and start it up */ if (!gst_wasapi_util_get_capture_client (GST_ELEMENT (self), self->client, &self->capture_client)) { goto beach; } hr = IAudioClient_Start (self->client); HR_FAILED_GOTO (hr, IAudioClock::Start, beach); gst_audio_ring_buffer_set_channel_positions (GST_AUDIO_BASE_SRC (self)->ringbuffer, self->positions); /* Increase the thread priority to reduce glitches */ self->thread_priority_handle = gst_wasapi_util_set_thread_characteristics (); res = TRUE; beach: /* unprepare() is not called if prepare() fails, but we want it to be, so call * it manually when needed */ if (!res) gst_wasapi_src_unprepare (asrc); return res; } static gboolean gst_wasapi_src_unprepare (GstAudioSrc * asrc) { GstWasapiSrc *self = GST_WASAPI_SRC (asrc); if (self->sharemode == AUDCLNT_SHAREMODE_EXCLUSIVE && !gst_wasapi_src_can_audioclient3 (self)) CoUninitialize (); if (self->thread_priority_handle != NULL) { gst_wasapi_util_revert_thread_characteristics (self->thread_priority_handle); self->thread_priority_handle = NULL; } if (self->client != NULL) { IAudioClient_Stop (self->client); } if (self->capture_client != NULL) { IUnknown_Release (self->capture_client); self->capture_client = NULL; } if (self->client_clock != NULL) { IUnknown_Release (self->client_clock); self->client_clock = NULL; } self->client_clock_freq = 0; return TRUE; } static guint gst_wasapi_src_read (GstAudioSrc * asrc, gpointer data, guint length, GstClockTime * timestamp) { GstWasapiSrc *self = GST_WASAPI_SRC (asrc); HRESULT hr; gint16 *from = NULL; guint wanted = length; DWORD flags; GST_OBJECT_LOCK (self); if (self->client_needs_restart) { hr = IAudioClient_Start (self->client); HR_FAILED_AND (hr, IAudioClient::Start, length = 0; goto beach); self->client_needs_restart = FALSE; } GST_OBJECT_UNLOCK (self); while (wanted > 0) { DWORD dwWaitResult; guint have_frames, n_frames, want_frames, read_len; /* Wait for data to become available */ dwWaitResult = WaitForSingleObject (self->event_handle, INFINITE); if (dwWaitResult != WAIT_OBJECT_0) { GST_ERROR_OBJECT (self, "Error waiting for event handle: %x", (guint) dwWaitResult); length = 0; goto beach; } hr = IAudioCaptureClient_GetBuffer (self->capture_client, (BYTE **) & from, &have_frames, &flags, NULL, NULL); if (hr != S_OK) { gchar *msg = gst_wasapi_util_hresult_to_string (hr); if (hr == AUDCLNT_S_BUFFER_EMPTY) GST_WARNING_OBJECT (self, "IAudioCaptureClient::GetBuffer failed: %s" ", retrying", msg); else GST_ERROR_OBJECT (self, "IAudioCaptureClient::GetBuffer failed: %s", msg); g_free (msg); length = 0; goto beach; } if (flags != 0) GST_INFO_OBJECT (self, "buffer flags=%#08x", (guint) flags); /* XXX: How do we handle AUDCLNT_BUFFERFLAGS_SILENT? We're supposed to write * out silence when that flag is set? See: * https://msdn.microsoft.com/en-us/library/windows/desktop/dd370800(v=vs.85).aspx */ if (flags & AUDCLNT_BUFFERFLAGS_DATA_DISCONTINUITY) GST_WARNING_OBJECT (self, "WASAPI reported glitch in buffer"); want_frames = wanted / self->mix_format->nBlockAlign; /* If GetBuffer is returning more frames than we can handle, all we can do is * hope that this is temporary and that things will settle down later. */ if (G_UNLIKELY (have_frames > want_frames)) GST_WARNING_OBJECT (self, "captured too many frames: have %i, want %i", have_frames, want_frames); /* Only copy data that will fit into the allocated buffer of size @length */ n_frames = MIN (have_frames, want_frames); read_len = n_frames * self->mix_format->nBlockAlign; { guint bpf = self->mix_format->nBlockAlign; GST_DEBUG_OBJECT (self, "have: %i (%i bytes), can read: %i (%i bytes), " "will read: %i (%i bytes)", have_frames, have_frames * bpf, want_frames, wanted, n_frames, read_len); } memcpy (data, from, read_len); wanted -= read_len; /* Always release all captured buffers if we've captured any at all */ hr = IAudioCaptureClient_ReleaseBuffer (self->capture_client, have_frames); HR_FAILED_AND (hr, IAudioClock::ReleaseBuffer, goto beach); } beach: return length; } static guint gst_wasapi_src_delay (GstAudioSrc * asrc) { GstWasapiSrc *self = GST_WASAPI_SRC (asrc); guint delay = 0; HRESULT hr; hr = IAudioClient_GetCurrentPadding (self->client, &delay); HR_FAILED_RET (hr, IAudioClock::GetCurrentPadding, 0); return delay; } static void gst_wasapi_src_reset (GstAudioSrc * asrc) { GstWasapiSrc *self = GST_WASAPI_SRC (asrc); HRESULT hr; if (!self->client) return; GST_OBJECT_LOCK (self); hr = IAudioClient_Stop (self->client); HR_FAILED_RET (hr, IAudioClock::Stop,); hr = IAudioClient_Reset (self->client); HR_FAILED_RET (hr, IAudioClock::Reset,); self->client_needs_restart = TRUE; GST_OBJECT_UNLOCK (self); } static GstClockTime gst_wasapi_src_get_time (GstClock * clock, gpointer user_data) { GstWasapiSrc *self = GST_WASAPI_SRC (user_data); HRESULT hr; guint64 devpos; GstClockTime result; if (G_UNLIKELY (self->client_clock == NULL)) return GST_CLOCK_TIME_NONE; hr = IAudioClock_GetPosition (self->client_clock, &devpos, NULL); HR_FAILED_RET (hr, IAudioClock::GetPosition, GST_CLOCK_TIME_NONE); result = gst_util_uint64_scale_int (devpos, GST_SECOND, self->client_clock_freq); /* GST_DEBUG_OBJECT (self, "devpos = %" G_GUINT64_FORMAT " frequency = %" G_GUINT64_FORMAT " result = %" G_GUINT64_FORMAT " ms", devpos, self->client_clock_freq, GST_TIME_AS_MSECONDS (result)); */ return result; }