/* * Copyright (C) 2008 Ole André Vadla Ravnås * Copyright (C) 2013 Collabora Ltd. * Author: Sebastian Dröge * 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-wasapisink * @title: wasapisink * * Provides audio playback using the Windows Audio Session API available with * Vista and newer. * * ## Example pipelines * |[ * gst-launch-1.0 -v audiotestsrc samplesperbuffer=160 ! wasapisink * ]| Generate 20 ms buffers and render to the default audio device. * * |[ * gst-launch-1.0 -v audiotestsrc samplesperbuffer=160 ! wasapisink low-latency=true * ]| Same as above, but with the minimum possible latency * */ #ifdef HAVE_CONFIG_H # include #endif #include "gstwasapisink.h" #include GST_DEBUG_CATEGORY_STATIC (gst_wasapi_sink_debug); #define GST_CAT_DEFAULT gst_wasapi_sink_debug static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS (GST_WASAPI_STATIC_CAPS)); #define DEFAULT_ROLE GST_WASAPI_DEVICE_ROLE_CONSOLE #define DEFAULT_MUTE FALSE #define DEFAULT_EXCLUSIVE FALSE #define DEFAULT_LOW_LATENCY FALSE #define DEFAULT_AUDIOCLIENT3 TRUE enum { PROP_0, PROP_ROLE, PROP_MUTE, PROP_DEVICE, PROP_EXCLUSIVE, PROP_LOW_LATENCY, PROP_AUDIOCLIENT3 }; static void gst_wasapi_sink_dispose (GObject * object); static void gst_wasapi_sink_finalize (GObject * object); static void gst_wasapi_sink_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void gst_wasapi_sink_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static GstCaps *gst_wasapi_sink_get_caps (GstBaseSink * bsink, GstCaps * filter); static gboolean gst_wasapi_sink_prepare (GstAudioSink * asink, GstAudioRingBufferSpec * spec); static gboolean gst_wasapi_sink_unprepare (GstAudioSink * asink); static gboolean gst_wasapi_sink_open (GstAudioSink * asink); static gboolean gst_wasapi_sink_close (GstAudioSink * asink); static gint gst_wasapi_sink_write (GstAudioSink * asink, gpointer data, guint length); static guint gst_wasapi_sink_delay (GstAudioSink * asink); static void gst_wasapi_sink_reset (GstAudioSink * asink); #define gst_wasapi_sink_parent_class parent_class G_DEFINE_TYPE (GstWasapiSink, gst_wasapi_sink, GST_TYPE_AUDIO_SINK); static void gst_wasapi_sink_class_init (GstWasapiSinkClass * klass) { GObjectClass *gobject_class = G_OBJECT_CLASS (klass); GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass); GstBaseSinkClass *gstbasesink_class = GST_BASE_SINK_CLASS (klass); GstAudioSinkClass *gstaudiosink_class = GST_AUDIO_SINK_CLASS (klass); gobject_class->dispose = gst_wasapi_sink_dispose; gobject_class->finalize = gst_wasapi_sink_finalize; gobject_class->set_property = gst_wasapi_sink_set_property; gobject_class->get_property = gst_wasapi_sink_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_MUTE, g_param_spec_boolean ("mute", "Mute", "Mute state of this stream", DEFAULT_MUTE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | GST_PARAM_MUTABLE_PLAYING)); 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_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", "Use the Windows 10 AudioClient3 API when available and if the " "low-latency property is set to TRUE", DEFAULT_AUDIOCLIENT3, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); gst_element_class_add_static_pad_template (gstelement_class, &sink_template); gst_element_class_set_static_metadata (gstelement_class, "WasapiSrc", "Sink/Audio", "Stream audio to an audio capture device through WASAPI", "Nirbheek Chauhan , " "Ole André Vadla Ravnås "); gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_wasapi_sink_get_caps); gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_wasapi_sink_prepare); gstaudiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_wasapi_sink_unprepare); gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_wasapi_sink_open); gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_wasapi_sink_close); gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_wasapi_sink_write); gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_wasapi_sink_delay); gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_wasapi_sink_reset); GST_DEBUG_CATEGORY_INIT (gst_wasapi_sink_debug, "wasapisink", 0, "Windows audio session API sink"); } static void gst_wasapi_sink_init (GstWasapiSink * self) { self->role = DEFAULT_ROLE; self->mute = DEFAULT_MUTE; self->sharemode = AUDCLNT_SHAREMODE_SHARED; 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_sink_dispose (GObject * object) { GstWasapiSink *self = GST_WASAPI_SINK (object); if (self->event_handle != NULL) { CloseHandle (self->event_handle); self->event_handle = NULL; } if (self->client != NULL) { IUnknown_Release (self->client); self->client = NULL; } if (self->render_client != NULL) { IUnknown_Release (self->render_client); self->render_client = NULL; } G_OBJECT_CLASS (gst_wasapi_sink_parent_class)->dispose (object); } static void gst_wasapi_sink_finalize (GObject * object) { GstWasapiSink *self = GST_WASAPI_SINK (object); CoTaskMemFree (self->mix_format); self->mix_format = NULL; CoUninitialize (); if (self->cached_caps != NULL) { gst_caps_unref (self->cached_caps); self->cached_caps = NULL; } g_clear_pointer (&self->positions, g_free); g_clear_pointer (&self->device_strid, g_free); self->mute = FALSE; G_OBJECT_CLASS (gst_wasapi_sink_parent_class)->finalize (object); } static void gst_wasapi_sink_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstWasapiSink *self = GST_WASAPI_SINK (object); switch (prop_id) { case PROP_ROLE: self->role = gst_wasapi_device_role_to_erole (g_value_get_enum (value)); break; case PROP_MUTE: self->mute = g_value_get_boolean (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_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_sink_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstWasapiSink *self = GST_WASAPI_SINK (object); switch (prop_id) { case PROP_ROLE: g_value_set_enum (value, gst_wasapi_erole_to_device_role (self->role)); break; case PROP_MUTE: g_value_set_boolean (value, self->mute); 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_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_sink_can_audioclient3 (GstWasapiSink * self) { /* AudioClient3 API only makes sense in shared mode */ if (self->sharemode != AUDCLNT_SHAREMODE_SHARED) return FALSE; if (!self->try_audioclient3) { GST_INFO_OBJECT (self, "AudioClient3 disabled by user"); return FALSE; } if (!gst_wasapi_util_have_audioclient3 ()) { GST_INFO_OBJECT (self, "AudioClient3 not available on this OS"); return FALSE; } /* Only use audioclient3 when low-latency is requested because otherwise * very slow machines and VMs with 1 CPU allocated will get glitches: * https://bugzilla.gnome.org/show_bug.cgi?id=794497 */ if (!self->low_latency) { GST_INFO_OBJECT (self, "AudioClient3 disabled because low-latency mode " "was not requested"); return FALSE; } return TRUE; } static GstCaps * gst_wasapi_sink_get_caps (GstBaseSink * bsink, GstCaps * filter) { GstWasapiSink *self = GST_WASAPI_SINK (bsink); 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 (bsink->sinkpad); 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_sink_open (GstAudioSink * asink) { GstWasapiSink *self = GST_WASAPI_SINK (asink); gboolean res = FALSE; IMMDevice *device = NULL; IAudioClient *client = NULL; GST_DEBUG_OBJECT (self, "opening device"); 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), eRender, self->role, self->device_strid, &device, &client)) { if (!self->device_strid) GST_ELEMENT_ERROR (self, RESOURCE, OPEN_WRITE, (NULL), ("Failed to get default device")); else GST_ELEMENT_ERROR (self, RESOURCE, OPEN_WRITE, (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_sink_close (GstAudioSink * asink) { GstWasapiSink *self = GST_WASAPI_SINK (asink); if (self->device != NULL) { IUnknown_Release (self->device); self->device = NULL; } if (self->client != NULL) { IUnknown_Release (self->client); self->client = NULL; } return TRUE; } /* Get the empty space in the buffer that we have to write to */ static gint gst_wasapi_sink_get_can_frames (GstWasapiSink * self) { HRESULT hr; guint n_frames_padding; /* There is no padding in exclusive mode since there is no ringbuffer */ if (self->sharemode == AUDCLNT_SHAREMODE_EXCLUSIVE) { GST_DEBUG_OBJECT (self, "exclusive mode, can write: %i", self->buffer_frame_count); return self->buffer_frame_count; } /* Frames the card hasn't rendered yet */ hr = IAudioClient_GetCurrentPadding (self->client, &n_frames_padding); HR_FAILED_ELEMENT_ERROR_RET (hr, IAudioClient::GetCurrentPadding, self, -1); GST_DEBUG_OBJECT (self, "%i unread frames (padding)", n_frames_padding); /* We can write out these many frames */ return self->buffer_frame_count - n_frames_padding; } static gboolean gst_wasapi_sink_prepare (GstAudioSink * asink, GstAudioRingBufferSpec * spec) { GstWasapiSink *self = GST_WASAPI_SINK (asink); gboolean res = FALSE; REFERENCE_TIME latency_rt; guint bpf, rate, devicep_frames; HRESULT hr; CoInitialize (NULL); if (gst_wasapi_sink_can_audioclient3 (self)) { if (!gst_wasapi_util_initialize_audioclient3 (GST_ELEMENT (self), spec, (IAudioClient3 *) self->client, self->mix_format, self->low_latency, FALSE, &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, FALSE, &devicep_frames)) goto beach; } bpf = GST_AUDIO_INFO_BPF (&spec->info); rate = GST_AUDIO_INFO_RATE (&spec->info); /* Total size of the allocated buffer that we will write to */ hr = IAudioClient_GetBufferSize (self->client, &self->buffer_frame_count); 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", self->buffer_frame_count, 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 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 writes */ hr = IAudioClient_SetEventHandle (self->client, self->event_handle); HR_FAILED_GOTO (hr, IAudioClient::SetEventHandle, beach); /* Get render sink client and start it up */ if (!gst_wasapi_util_get_render_client (GST_ELEMENT (self), self->client, &self->render_client)) { goto beach; } GST_INFO_OBJECT (self, "got render client"); /* To avoid start-up glitches, before starting the streaming, we fill the * buffer with silence as recommended by the documentation: * https://msdn.microsoft.com/en-us/library/windows/desktop/dd370879%28v=vs.85%29.aspx */ { gint n_frames, len; gint16 *dst = NULL; n_frames = gst_wasapi_sink_get_can_frames (self); if (n_frames < 1) { GST_ELEMENT_ERROR (self, RESOURCE, WRITE, (NULL), ("should have more than %i frames to write", n_frames)); goto beach; } len = n_frames * self->mix_format->nBlockAlign; hr = IAudioRenderClient_GetBuffer (self->render_client, n_frames, (BYTE **) & dst); HR_FAILED_GOTO (hr, IAudioRenderClient::GetBuffer, beach); GST_DEBUG_OBJECT (self, "pre-wrote %i bytes of silence", len); hr = IAudioRenderClient_ReleaseBuffer (self->render_client, n_frames, AUDCLNT_BUFFERFLAGS_SILENT); HR_FAILED_GOTO (hr, IAudioRenderClient::ReleaseBuffer, beach); } hr = IAudioClient_Start (self->client); HR_FAILED_GOTO (hr, IAudioClient::Start, beach); self->client_needs_restart = FALSE; gst_audio_ring_buffer_set_channel_positions (GST_AUDIO_BASE_SINK (self)->ringbuffer, self->positions); 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_sink_unprepare (asink); return res; } static gboolean gst_wasapi_sink_unprepare (GstAudioSink * asink) { GstWasapiSink *self = GST_WASAPI_SINK (asink); if (self->client != NULL) { IAudioClient_Stop (self->client); } if (self->render_client != NULL) { IUnknown_Release (self->render_client); self->render_client = NULL; } CoUninitialize (); return TRUE; } static gint gst_wasapi_sink_write (GstAudioSink * asink, gpointer data, guint length) { GstWasapiSink *self = GST_WASAPI_SINK (asink); HRESULT hr; gint16 *dst = NULL; DWORD dwWaitResult; guint can_frames, have_frames, n_frames, write_len, written_len = 0; GST_OBJECT_LOCK (self); if (self->client_needs_restart) { hr = IAudioClient_Start (self->client); HR_FAILED_ELEMENT_ERROR_AND (hr, IAudioClient::Start, self, GST_OBJECT_UNLOCK (self); goto err); self->client_needs_restart = FALSE; } GST_OBJECT_UNLOCK (self); /* We have N frames to be written out */ have_frames = length / (self->mix_format->nBlockAlign); if (self->sharemode == AUDCLNT_SHAREMODE_EXCLUSIVE) { /* In exlusive mode we have to wait always */ dwWaitResult = WaitForSingleObject (self->event_handle, INFINITE); if (dwWaitResult != WAIT_OBJECT_0) { GST_ERROR_OBJECT (self, "Error waiting for event handle: %x", (guint) dwWaitResult); goto err; } can_frames = gst_wasapi_sink_get_can_frames (self); if (can_frames < 0) { GST_ERROR_OBJECT (self, "Error getting frames to write to"); goto err; } /* In exclusive mode we need to fill the whole buffer in one go or * GetBuffer will error out */ if (can_frames != have_frames) { GST_ERROR_OBJECT (self, "Need at %i frames to write for exclusive mode, but got %i", can_frames, have_frames); goto err; } } else { /* In shared mode we can write parts of the buffer, so only wait * in case we can't write anything */ can_frames = gst_wasapi_sink_get_can_frames (self); if (can_frames < 0) { GST_ERROR_OBJECT (self, "Error getting frames to write to"); goto err; } if (can_frames == 0) { dwWaitResult = WaitForSingleObject (self->event_handle, INFINITE); if (dwWaitResult != WAIT_OBJECT_0) { GST_ERROR_OBJECT (self, "Error waiting for event handle: %x", (guint) dwWaitResult); goto err; } can_frames = gst_wasapi_sink_get_can_frames (self); if (can_frames < 0) { GST_ERROR_OBJECT (self, "Error getting frames to write to"); goto err; } } } /* We will write out these many frames, and this much length */ n_frames = MIN (can_frames, have_frames); write_len = n_frames * self->mix_format->nBlockAlign; GST_DEBUG_OBJECT (self, "total: %i, have_frames: %i (%i bytes), " "can_frames: %i, will write: %i (%i bytes)", self->buffer_frame_count, have_frames, length, can_frames, n_frames, write_len); hr = IAudioRenderClient_GetBuffer (self->render_client, n_frames, (BYTE **) & dst); HR_FAILED_ELEMENT_ERROR_AND (hr, IAudioRenderClient::GetBuffer, self, goto err); memcpy (dst, data, write_len); hr = IAudioRenderClient_ReleaseBuffer (self->render_client, n_frames, self->mute ? AUDCLNT_BUFFERFLAGS_SILENT : 0); HR_FAILED_ELEMENT_ERROR_AND (hr, IAudioRenderClient::ReleaseBuffer, self, goto err); written_len = write_len; out: return written_len; err: written_len = -1; goto out; } static guint gst_wasapi_sink_delay (GstAudioSink * asink) { GstWasapiSink *self = GST_WASAPI_SINK (asink); guint delay = 0; HRESULT hr; hr = IAudioClient_GetCurrentPadding (self->client, &delay); HR_FAILED_RET (hr, IAudioClient::GetCurrentPadding, 0); return delay; } static void gst_wasapi_sink_reset (GstAudioSink * asink) { GstWasapiSink *self = GST_WASAPI_SINK (asink); HRESULT hr; GST_INFO_OBJECT (self, "reset called"); if (!self->client) return; GST_OBJECT_LOCK (self); hr = IAudioClient_Stop (self->client); HR_FAILED_AND (hr, IAudioClient::Stop,); hr = IAudioClient_Reset (self->client); HR_FAILED_AND (hr, IAudioClient::Reset,); self->client_needs_restart = TRUE; GST_OBJECT_UNLOCK (self); }