| /* |
| * Copyright (C) 2008 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.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-wasapisrc |
| * |
| * Provides audio capture from the Windows Audio Session API available with |
| * Vista and newer. |
| * |
| * <refsect2> |
| * <title>Example pipelines</title> |
| * |[ |
| * gst-launch-1.0 -v wasapisrc ! fakesink |
| * ]| Capture from the default audio device and render to fakesink. |
| * </refsect2> |
| */ |
| #ifdef HAVE_CONFIG_H |
| # include <config.h> |
| #endif |
| |
| #include "gstwasapisrc.h" |
| |
| 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 ("audio/x-raw, " |
| "format = (string) S16LE, " |
| "layout = (string) interleaved, " |
| "rate = (int) 44100, " "channels = (int) 1")); |
| |
| static void gst_wasapi_src_dispose (GObject * object); |
| static void gst_wasapi_src_finalize (GObject * object); |
| |
| 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); |
| |
| 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; |
| |
| 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", |
| "Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>"); |
| |
| 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->event_handle = CreateEvent (NULL, FALSE, FALSE, NULL); |
| |
| 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; |
| } |
| |
| G_OBJECT_CLASS (gst_wasapi_src_parent_class)->dispose (object); |
| } |
| |
| static void |
| gst_wasapi_src_finalize (GObject * object) |
| { |
| CoUninitialize (); |
| |
| G_OBJECT_CLASS (gst_wasapi_src_parent_class)->finalize (object); |
| } |
| |
| static GstCaps * |
| gst_wasapi_src_get_caps (GstBaseSrc * bsrc, GstCaps * filter) |
| { |
| /* TODO: Implement */ |
| return NULL; |
| } |
| |
| static gboolean |
| gst_wasapi_src_open (GstAudioSrc * asrc) |
| { |
| GstWasapiSrc *self = GST_WASAPI_SRC (asrc); |
| gboolean res = FALSE; |
| IAudioClient *client = NULL; |
| |
| if (!gst_wasapi_util_get_default_device_client (GST_ELEMENT (self), TRUE, |
| &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_src_close (GstAudioSrc * asrc) |
| { |
| GstWasapiSrc *self = GST_WASAPI_SRC (asrc); |
| |
| 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; |
| IAudioClock *client_clock = NULL; |
| guint64 client_clock_freq = 0; |
| IAudioCaptureClient *capture_client = NULL; |
| REFERENCE_TIME latency_rt, def_period, min_period; |
| WAVEFORMATEXTENSIBLE format; |
| HRESULT hr; |
| |
| 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_clock (GST_ELEMENT (self), self->client, |
| &client_clock)) { |
| goto beach; |
| } |
| |
| hr = IAudioClock_GetFrequency (client_clock, &client_clock_freq); |
| if (hr != S_OK) { |
| GST_ERROR_OBJECT (self, "IAudioClock::GetFrequency () failed"); |
| goto beach; |
| } |
| |
| if (!gst_wasapi_util_get_capture_client (GST_ELEMENT (self), self->client, |
| &capture_client)) { |
| goto beach; |
| } |
| |
| hr = IAudioClient_Start (self->client); |
| if (hr != S_OK) { |
| GST_ERROR_OBJECT (self, "IAudioClient::Start failed"); |
| goto beach; |
| } |
| |
| self->client_clock = client_clock; |
| self->client_clock_freq = client_clock_freq; |
| self->capture_client = capture_client; |
| |
| res = TRUE; |
| |
| beach: |
| if (!res) { |
| if (capture_client != NULL) |
| IUnknown_Release (capture_client); |
| |
| if (client_clock != NULL) |
| IUnknown_Release (client_clock); |
| } |
| |
| return res; |
| } |
| |
| static gboolean |
| gst_wasapi_src_unprepare (GstAudioSrc * asrc) |
| { |
| GstWasapiSrc *self = GST_WASAPI_SRC (asrc); |
| |
| 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; |
| } |
| |
| 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 *samples = NULL; |
| guint32 nsamples = 0, length_samples; |
| DWORD flags = 0; |
| guint64 devpos; |
| guint i; |
| gint16 *dst; |
| |
| WaitForSingleObject (self->event_handle, INFINITE); |
| |
| do { |
| hr = IAudioCaptureClient_GetBuffer (self->capture_client, |
| (BYTE **) & samples, &nsamples, &flags, &devpos, NULL); |
| } |
| while (hr == AUDCLNT_S_BUFFER_EMPTY); |
| |
| if (hr != S_OK) { |
| GST_ERROR_OBJECT (self, "IAudioCaptureClient::GetBuffer () failed: %s", |
| gst_wasapi_util_hresult_to_string (hr)); |
| length = 0; |
| goto beach; |
| } |
| |
| if (flags != 0) { |
| GST_WARNING_OBJECT (self, "devpos %" G_GUINT64_FORMAT ": flags=0x%08x", |
| devpos, (guint) flags); |
| } |
| |
| length_samples = length / self->info.bpf; |
| nsamples = MIN (length_samples, nsamples); |
| length = nsamples * self->info.bpf; |
| |
| dst = (gint16 *) data; |
| for (i = 0; i < nsamples; i++) { |
| *dst = *samples; |
| |
| samples += 2; |
| dst++; |
| } |
| |
| hr = IAudioCaptureClient_ReleaseBuffer (self->capture_client, nsamples); |
| if (hr != S_OK) { |
| GST_ERROR_OBJECT (self, "IAudioCaptureClient::ReleaseBuffer () failed: %s", |
| gst_wasapi_util_hresult_to_string (hr)); |
| goto beach; |
| } |
| |
| beach: |
| |
| return length; |
| } |
| |
| static guint |
| gst_wasapi_src_delay (GstAudioSrc * asrc) |
| { |
| /* FIXME: Implement */ |
| return 0; |
| } |
| |
| static void |
| gst_wasapi_src_reset (GstAudioSrc * asrc) |
| { |
| GstWasapiSrc *self = GST_WASAPI_SRC (asrc); |
| 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; |
| } |
| } |
| } |
| |
| 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); |
| if (G_UNLIKELY (hr != S_OK)) |
| return 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; |
| } |