| /* |
| * Copyright (C) 2008 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com> |
| * Copyright (C) 2013 Collabora Ltd. |
| * Author: Sebastian Dröge <sebastian.droege@collabora.co.uk> |
| * Copyright (C) 2018 Centricular Ltd. |
| * Author: Nirbheek Chauhan <nirbheek@centricular.com> |
| * |
| * 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 <config.h> |
| #endif |
| |
| #include "gstwasapisink.h" |
| |
| #include <avrt.h> |
| |
| 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 <nirbheek@centricular.com>, " |
| "Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>"); |
| |
| 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); |
| |
| g_clear_pointer (&self->mix_format, CoTaskMemFree); |
| |
| 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), FALSE, |
| 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_RET (hr, IAudioClient::GetCurrentPadding, -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; |
| |
| 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, |
| &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, |
| &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); |
| |
| gst_audio_ring_buffer_set_channel_positions (GST_AUDIO_BASE_SINK |
| (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_sink_unprepare (asink); |
| |
| return res; |
| } |
| |
| static gboolean |
| gst_wasapi_sink_unprepare (GstAudioSink * asink) |
| { |
| GstWasapiSink *self = GST_WASAPI_SINK (asink); |
| |
| if (self->sharemode == AUDCLNT_SHAREMODE_EXCLUSIVE && |
| !gst_wasapi_sink_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->render_client != NULL) { |
| IUnknown_Release (self->render_client); |
| self->render_client = NULL; |
| } |
| |
| 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; |
| guint pending = length; |
| |
| 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 (pending > 0) { |
| DWORD dwWaitResult; |
| guint can_frames, have_frames, n_frames, write_len; |
| |
| dwWaitResult = WaitForSingleObject (self->event_handle, INFINITE); |
| if (dwWaitResult != WAIT_OBJECT_0) { |
| GST_ERROR_OBJECT (self, "Error waiting for event handle: %x", |
| (guint) dwWaitResult); |
| length -= pending; |
| goto beach; |
| } |
| |
| /* We have N frames to be written out */ |
| have_frames = pending / (self->mix_format->nBlockAlign); |
| /* We have can_frames space in the output buffer */ |
| can_frames = gst_wasapi_sink_get_can_frames (self); |
| /* 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, pending, can_frames, n_frames, write_len); |
| |
| hr = IAudioRenderClient_GetBuffer (self->render_client, n_frames, |
| (BYTE **) & dst); |
| HR_FAILED_AND (hr, IAudioRenderClient::GetBuffer, length = 0; goto beach); |
| |
| memcpy (dst, data, write_len); |
| |
| hr = IAudioRenderClient_ReleaseBuffer (self->render_client, n_frames, |
| self->mute ? AUDCLNT_BUFFERFLAGS_SILENT : 0); |
| HR_FAILED_AND (hr, IAudioRenderClient::ReleaseBuffer, length = 0; |
| goto beach); |
| |
| pending -= write_len; |
| } |
| |
| beach: |
| |
| return length; |
| } |
| |
| 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); |
| } |