| /* |
| * 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> |
| * |
| * 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. |
| * |
| */ |
| #ifdef HAVE_CONFIG_H |
| # include <config.h> |
| #endif |
| |
| #include "gstwasapisink.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 ("audio/x-raw, " |
| "format = (string) S16LE, " |
| "layout = (string) interleaved, " |
| "rate = (int) 44100, " "channels = (int) 2")); |
| |
| static void gst_wasapi_sink_dispose (GObject * object); |
| static void gst_wasapi_sink_finalize (GObject * object); |
| |
| 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); |
| |
| 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; |
| |
| 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", |
| "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->event_handle = CreateEvent (NULL, FALSE, FALSE, NULL); |
| |
| 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; |
| } |
| |
| G_OBJECT_CLASS (gst_wasapi_sink_parent_class)->dispose (object); |
| } |
| |
| static void |
| gst_wasapi_sink_finalize (GObject * object) |
| { |
| CoUninitialize (); |
| |
| G_OBJECT_CLASS (gst_wasapi_sink_parent_class)->finalize (object); |
| } |
| |
| static GstCaps * |
| gst_wasapi_sink_get_caps (GstBaseSink * bsink, GstCaps * filter) |
| { |
| /* FIXME: Implement */ |
| return NULL; |
| } |
| |
| static gboolean |
| gst_wasapi_sink_open (GstAudioSink * asink) |
| { |
| GstWasapiSink *self = GST_WASAPI_SINK (asink); |
| gboolean res = FALSE; |
| IAudioClient *client = NULL; |
| |
| if (!gst_wasapi_util_get_default_device_client (GST_ELEMENT (self), FALSE, |
| &client)) { |
| GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ, (NULL), |
| ("Failed to get default device")); |
| goto beach; |
| } |
| |
| self->client = client; |
| res = TRUE; |
| |
| beach: |
| |
| return res; |
| } |
| |
| static gboolean |
| gst_wasapi_sink_close (GstAudioSink * asink) |
| { |
| GstWasapiSink *self = GST_WASAPI_SINK (asink); |
| |
| if (self->client != NULL) { |
| IUnknown_Release (self->client); |
| self->client = NULL; |
| } |
| |
| return TRUE; |
| } |
| |
| static gboolean |
| gst_wasapi_sink_prepare (GstAudioSink * asink, GstAudioRingBufferSpec * spec) |
| { |
| GstWasapiSink *self = GST_WASAPI_SINK (asink); |
| gboolean res = FALSE; |
| HRESULT hr; |
| REFERENCE_TIME latency_rt, def_period, min_period; |
| WAVEFORMATEXTENSIBLE format; |
| IAudioRenderClient *render_client = NULL; |
| |
| hr = IAudioClient_GetDevicePeriod (self->client, &def_period, &min_period); |
| if (hr != S_OK) { |
| GST_ERROR_OBJECT (self, "IAudioClient::GetDevicePeriod () failed"); |
| goto beach; |
| } |
| |
| gst_wasapi_util_audio_info_to_waveformatex (&spec->info, &format); |
| self->info = spec->info; |
| |
| hr = IAudioClient_Initialize (self->client, AUDCLNT_SHAREMODE_SHARED, |
| AUDCLNT_STREAMFLAGS_EVENTCALLBACK, |
| spec->buffer_time / 100, 0, (WAVEFORMATEX *) & format, NULL); |
| if (hr != S_OK) { |
| GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ, (NULL), |
| ("IAudioClient::Initialize () failed: %s", |
| gst_wasapi_util_hresult_to_string (hr))); |
| goto beach; |
| } |
| |
| hr = IAudioClient_GetStreamLatency (self->client, &latency_rt); |
| if (hr != S_OK) { |
| GST_ERROR_OBJECT (self, "IAudioClient::GetStreamLatency () failed"); |
| goto beach; |
| } |
| |
| GST_INFO_OBJECT (self, "default period: %d (%d ms), " |
| "minimum period: %d (%d ms), " |
| "latency: %d (%d ms)", |
| (guint32) def_period, (guint32) def_period / 10000, |
| (guint32) min_period, (guint32) min_period / 10000, |
| (guint32) latency_rt, (guint32) latency_rt / 10000); |
| |
| /* FIXME: What to do with the latency? */ |
| |
| hr = IAudioClient_SetEventHandle (self->client, self->event_handle); |
| if (hr != S_OK) { |
| GST_ERROR_OBJECT (self, "IAudioClient::SetEventHandle () failed"); |
| goto beach; |
| } |
| |
| if (!gst_wasapi_util_get_render_client (GST_ELEMENT (self), self->client, |
| &render_client)) { |
| goto beach; |
| } |
| |
| hr = IAudioClient_Start (self->client); |
| if (hr != S_OK) { |
| GST_ERROR_OBJECT (self, "IAudioClient::Start failed"); |
| goto beach; |
| } |
| |
| self->render_client = render_client; |
| render_client = NULL; |
| |
| res = TRUE; |
| |
| beach: |
| if (render_client != NULL) |
| IUnknown_Release (render_client); |
| |
| 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; |
| } |
| |
| 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 nsamples; |
| |
| nsamples = length / self->info.bpf; |
| |
| WaitForSingleObject (self->event_handle, INFINITE); |
| |
| hr = IAudioRenderClient_GetBuffer (self->render_client, nsamples, |
| (BYTE **) & dst); |
| if (hr != S_OK) { |
| GST_ELEMENT_ERROR (self, RESOURCE, WRITE, (NULL), |
| ("IAudioRenderClient::GetBuffer () failed: %s", |
| gst_wasapi_util_hresult_to_string (hr))); |
| length = 0; |
| goto beach; |
| } |
| |
| memcpy (dst, data, length); |
| |
| hr = IAudioRenderClient_ReleaseBuffer (self->render_client, nsamples, 0); |
| if (hr != S_OK) { |
| GST_ERROR_OBJECT (self, "IAudioRenderClient::ReleaseBuffer () failed: %s", |
| gst_wasapi_util_hresult_to_string (hr)); |
| length = 0; |
| goto beach; |
| } |
| |
| beach: |
| |
| return length; |
| } |
| |
| static guint |
| gst_wasapi_sink_delay (GstAudioSink * asink) |
| { |
| /* FIXME: Implement */ |
| return 0; |
| } |
| |
| static void |
| gst_wasapi_sink_reset (GstAudioSink * asink) |
| { |
| GstWasapiSink *self = GST_WASAPI_SINK (asink); |
| HRESULT hr; |
| |
| if (self->client) { |
| hr = IAudioClient_Stop (self->client); |
| if (hr != S_OK) { |
| GST_ERROR_OBJECT (self, "IAudioClient::Stop () failed: %s", |
| gst_wasapi_util_hresult_to_string (hr)); |
| return; |
| } |
| |
| hr = IAudioClient_Reset (self->client); |
| if (hr != S_OK) { |
| GST_ERROR_OBJECT (self, "IAudioClient::Reset () failed: %s", |
| gst_wasapi_util_hresult_to_string (hr)); |
| return; |
| } |
| } |
| } |