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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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446 lines
12 KiB
C
446 lines
12 KiB
C
/*
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* Copyright (C) 2008 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-wasapisrc
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*
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* Provides audio capture from the Windows Audio Session API available with
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* Vista and newer.
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*
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* <refsect2>
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* <title>Example pipelines</title>
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* |[
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* gst-launch-0.10 -v wasapisrc ! fakesink
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* ]| Capture from the default audio device and render to fakesink.
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* </refsect2>
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*/
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#include "gstwasapisrc.h"
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#include <gst/audio/gstaudioclock.h>
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GST_DEBUG_CATEGORY_STATIC (gst_wasapi_src_debug);
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#define GST_CAT_DEFAULT gst_wasapi_src_debug
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static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw-int, "
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"width = (int) 16, "
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"depth = (int) 16, "
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"rate = (int) 8000, "
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"channels = (int) 1, "
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"signed = (boolean) TRUE, "
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"endianness = (int) " G_STRINGIFY (G_BYTE_ORDER)));
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static void gst_wasapi_src_dispose (GObject * object);
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static void gst_wasapi_src_finalize (GObject * object);
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static GstClock *gst_wasapi_src_provide_clock (GstElement * element);
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static gboolean gst_wasapi_src_start (GstBaseSrc * src);
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static gboolean gst_wasapi_src_stop (GstBaseSrc * src);
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static gboolean gst_wasapi_src_query (GstBaseSrc * src, GstQuery * query);
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static GstFlowReturn gst_wasapi_src_create (GstPushSrc * src, GstBuffer ** buf);
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static GstClockTime gst_wasapi_src_get_time (GstClock * clock,
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gpointer user_data);
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GST_BOILERPLATE (GstWasapiSrc, gst_wasapi_src, GstPushSrc, GST_TYPE_PUSH_SRC);
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static void
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gst_wasapi_src_base_init (gpointer gclass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (gclass);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&src_template));
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gst_element_class_set_static_metadata (element_class, "WasapiSrc",
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"Source/Audio",
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"Stream audio from an audio capture device through WASAPI",
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"Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>");
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}
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static void
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gst_wasapi_src_class_init (GstWasapiSrcClass * klass)
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{
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GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
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GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
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GstBaseSrcClass *gstbasesrc_class = GST_BASE_SRC_CLASS (klass);
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GstPushSrcClass *gstpushsrc_class = GST_PUSH_SRC_CLASS (klass);
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gobject_class->dispose = gst_wasapi_src_dispose;
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gobject_class->finalize = gst_wasapi_src_finalize;
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gstelement_class->provide_clock = gst_wasapi_src_provide_clock;
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gstbasesrc_class->start = gst_wasapi_src_start;
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gstbasesrc_class->stop = gst_wasapi_src_stop;
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gstbasesrc_class->query = gst_wasapi_src_query;
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gstpushsrc_class->create = gst_wasapi_src_create;
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GST_DEBUG_CATEGORY_INIT (gst_wasapi_src_debug, "wasapisrc",
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0, "Windows audio session API source");
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}
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static void
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gst_wasapi_src_init (GstWasapiSrc * self, GstWasapiSrcClass * gclass)
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{
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GstBaseSrc *basesrc = GST_BASE_SRC (self);
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gst_base_src_set_format (basesrc, GST_FORMAT_TIME);
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gst_base_src_set_live (basesrc, TRUE);
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self->rate = 8000;
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self->buffer_time = 20 * GST_MSECOND;
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self->period_time = 20 * GST_MSECOND;
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self->latency = GST_CLOCK_TIME_NONE;
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self->samples_per_buffer = self->rate / (GST_SECOND / self->period_time);
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self->start_time = GST_CLOCK_TIME_NONE;
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self->next_time = GST_CLOCK_TIME_NONE;
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#if GST_CHECK_VERSION(0, 10, 31) || (GST_CHECK_VERSION(0, 10, 30) && GST_VERSION_NANO > 0)
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self->clock = gst_audio_clock_new_full ("GstWasapiSrcClock",
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gst_wasapi_src_get_time, gst_object_ref (self),
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(GDestroyNotify) gst_object_unref);
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#else
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self->clock = gst_audio_clock_new ("GstWasapiSrcClock",
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gst_wasapi_src_get_time, self);
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#endif
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CoInitialize (NULL);
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}
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static void
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gst_wasapi_src_dispose (GObject * object)
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{
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GstWasapiSrc *self = GST_WASAPI_SRC (object);
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if (self->clock != NULL) {
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gst_object_unref (self->clock);
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self->clock = NULL;
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}
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G_OBJECT_CLASS (parent_class)->dispose (object);
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}
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static void
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gst_wasapi_src_finalize (GObject * object)
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{
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GstWasapiSrc *self = GST_WASAPI_SRC (object);
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CoUninitialize ();
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static GstClock *
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gst_wasapi_src_provide_clock (GstElement * element)
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{
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GstWasapiSrc *self = GST_WASAPI_SRC (element);
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GstClock *clock;
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GST_OBJECT_LOCK (self);
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if (self->client_clock == NULL)
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goto wrong_state;
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clock = GST_CLOCK (gst_object_ref (self->clock));
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GST_OBJECT_UNLOCK (self);
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return clock;
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/* ERRORS */
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wrong_state:
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{
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GST_OBJECT_UNLOCK (self);
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GST_DEBUG_OBJECT (self, "IAudioClock not acquired");
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return NULL;
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}
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}
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static gboolean
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gst_wasapi_src_start (GstBaseSrc * src)
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{
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GstWasapiSrc *self = GST_WASAPI_SRC (src);
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gboolean res = FALSE;
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IAudioClient *client = NULL;
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IAudioClock *client_clock = NULL;
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guint64 client_clock_freq = 0;
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IAudioCaptureClient *capture_client = NULL;
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HRESULT hr;
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if (!gst_wasapi_util_get_default_device_client (GST_ELEMENT (self),
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TRUE, self->rate, self->buffer_time, self->period_time, 0, &client,
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&self->latency))
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goto beach;
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hr = IAudioClient_GetService (client, &IID_IAudioClock, &client_clock);
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if (hr != S_OK) {
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GST_ERROR_OBJECT (self, "IAudioClient::GetService (IID_IAudioClock) "
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"failed");
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goto beach;
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}
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hr = IAudioClock_GetFrequency (client_clock, &client_clock_freq);
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if (hr != S_OK) {
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GST_ERROR_OBJECT (self, "IAudioClock::GetFrequency () failed");
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goto beach;
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}
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hr = IAudioClient_GetService (client, &IID_IAudioCaptureClient,
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&capture_client);
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if (hr != S_OK) {
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GST_ERROR_OBJECT (self, "IAudioClient::GetService "
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"(IID_IAudioCaptureClient) failed");
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goto beach;
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}
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hr = IAudioClient_Start (client);
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if (hr != S_OK) {
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GST_ERROR_OBJECT (self, "IAudioClient::Start failed");
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goto beach;
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}
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self->client = client;
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self->client_clock = client_clock;
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self->client_clock_freq = client_clock_freq;
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self->capture_client = capture_client;
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res = TRUE;
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beach:
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if (!res) {
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if (capture_client != NULL)
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IUnknown_Release (capture_client);
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if (client_clock != NULL)
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IUnknown_Release (client_clock);
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if (client != NULL)
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IUnknown_Release (client);
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}
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return res;
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}
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static gboolean
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gst_wasapi_src_stop (GstBaseSrc * src)
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{
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GstWasapiSrc *self = GST_WASAPI_SRC (src);
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if (self->client != NULL) {
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IAudioClient_Stop (self->client);
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}
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if (self->capture_client != NULL) {
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IUnknown_Release (self->capture_client);
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self->capture_client = NULL;
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}
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if (self->client_clock != NULL) {
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IUnknown_Release (self->client_clock);
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self->client_clock = NULL;
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}
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if (self->client != NULL) {
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IUnknown_Release (self->client);
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self->client = NULL;
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}
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return TRUE;
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}
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static gboolean
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gst_wasapi_src_query (GstBaseSrc * src, GstQuery * query)
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{
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GstWasapiSrc *self = GST_WASAPI_SRC (src);
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gboolean ret = FALSE;
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GST_DEBUG_OBJECT (self, "query for %s",
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gst_query_type_get_name (GST_QUERY_TYPE (query)));
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switch (GST_QUERY_TYPE (query)) {
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case GST_QUERY_LATENCY:{
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GstClockTime min_latency, max_latency;
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min_latency = self->latency + self->period_time;
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max_latency = min_latency;
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GST_DEBUG_OBJECT (self, "reporting latency of min %" GST_TIME_FORMAT
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" max %" GST_TIME_FORMAT,
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GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
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gst_query_set_latency (query, TRUE, min_latency, max_latency);
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ret = TRUE;
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break;
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}
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default:
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ret = GST_BASE_SRC_CLASS (parent_class)->query (src, query);
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break;
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}
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return ret;
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}
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static GstFlowReturn
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gst_wasapi_src_create (GstPushSrc * src, GstBuffer ** buf)
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{
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GstWasapiSrc *self = GST_WASAPI_SRC (src);
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GstFlowReturn ret = GST_FLOW_OK;
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GstClock *clock;
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GstClockTime timestamp, duration = self->period_time;
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HRESULT hr;
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gint16 *samples = NULL;
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guint32 nsamples_read = 0, nsamples;
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DWORD flags = 0;
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guint64 devpos;
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GST_OBJECT_LOCK (self);
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clock = GST_ELEMENT_CLOCK (self);
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if (clock != NULL)
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gst_object_ref (clock);
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GST_OBJECT_UNLOCK (self);
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if (clock != NULL && GST_CLOCK_TIME_IS_VALID (self->next_time)) {
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GstClockID id;
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id = gst_clock_new_single_shot_id (clock, self->next_time);
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gst_clock_id_wait (id, NULL);
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gst_clock_id_unref (id);
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}
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do {
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hr = IAudioCaptureClient_GetBuffer (self->capture_client,
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(BYTE **) & samples, &nsamples_read, &flags, &devpos, NULL);
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}
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while (hr == AUDCLNT_S_BUFFER_EMPTY);
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if (hr != S_OK) {
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GST_ERROR_OBJECT (self, "IAudioCaptureClient::GetBuffer () failed: %s",
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gst_wasapi_util_hresult_to_string (hr));
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ret = GST_FLOW_ERROR;
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goto beach;
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}
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if (flags != 0) {
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GST_WARNING_OBJECT (self, "devpos %" G_GUINT64_FORMAT ": flags=0x%08x",
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devpos, flags);
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}
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/* FIXME: Why do we get 1024 sometimes and not a multiple of
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* samples_per_buffer? Shouldn't WASAPI provide a DISCONT
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* flag if we read too slow?
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*/
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nsamples = nsamples_read;
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g_assert (nsamples >= self->samples_per_buffer);
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if (nsamples > self->samples_per_buffer) {
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GST_WARNING_OBJECT (self,
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"devpos %" G_GUINT64_FORMAT ": got %d samples, expected %d, clipping!",
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devpos, nsamples, self->samples_per_buffer);
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nsamples = self->samples_per_buffer;
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}
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if (clock == NULL || clock == self->clock) {
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timestamp =
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gst_util_uint64_scale (devpos, GST_SECOND, self->client_clock_freq);
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} else {
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GstClockTime base_time;
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timestamp = gst_clock_get_time (clock);
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base_time = GST_ELEMENT_CAST (self)->base_time;
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if (timestamp > base_time)
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timestamp -= base_time;
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else
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timestamp = 0;
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if (timestamp > duration)
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timestamp -= duration;
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else
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timestamp = 0;
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}
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ret = gst_pad_alloc_buffer_and_set_caps (GST_BASE_SRC_PAD (self),
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devpos,
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nsamples * sizeof (gint16), GST_PAD_CAPS (GST_BASE_SRC_PAD (self)), buf);
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if (ret == GST_FLOW_OK) {
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guint i;
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gint16 *dst;
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GST_BUFFER_OFFSET_END (*buf) = devpos + self->samples_per_buffer;
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GST_BUFFER_TIMESTAMP (*buf) = timestamp;
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GST_BUFFER_DURATION (*buf) = duration;
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dst = (gint16 *) GST_BUFFER_DATA (*buf);
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for (i = 0; i < nsamples; i++) {
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*dst = *samples;
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samples += 2;
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dst++;
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}
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}
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hr = IAudioCaptureClient_ReleaseBuffer (self->capture_client, nsamples_read);
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if (hr != S_OK) {
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GST_ERROR_OBJECT (self, "IAudioCaptureClient::ReleaseBuffer () failed: %s",
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gst_wasapi_util_hresult_to_string (hr));
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ret = GST_FLOW_ERROR;
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goto beach;
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}
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beach:
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if (clock != NULL)
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gst_object_unref (clock);
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return ret;
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}
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static GstClockTime
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gst_wasapi_src_get_time (GstClock * clock, gpointer user_data)
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{
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GstWasapiSrc *self = GST_WASAPI_SRC (user_data);
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HRESULT hr;
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guint64 devpos;
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GstClockTime result;
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if (G_UNLIKELY (self->client_clock == NULL))
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return GST_CLOCK_TIME_NONE;
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hr = IAudioClock_GetPosition (self->client_clock, &devpos, NULL);
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if (G_UNLIKELY (hr != S_OK))
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return GST_CLOCK_TIME_NONE;
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result = gst_util_uint64_scale_int (devpos, GST_SECOND,
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self->client_clock_freq);
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/*
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GST_DEBUG_OBJECT (self, "devpos = %" G_GUINT64_FORMAT
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" frequency = %" G_GUINT64_FORMAT
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" result = %" G_GUINT64_FORMAT " ms",
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devpos, self->client_clock_freq, GST_TIME_AS_MSECONDS (result));
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*/
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return result;
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}
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