| /* GStreamer |
| * Copyright (C) 2012 Fluendo S.A. <support@fluendo.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-openslessrc |
| * @see_also: openslessink |
| * |
| * This element reads data from default audio input using the OpenSL ES API in Android OS. |
| * |
| * <refsect2> |
| * <title>Example pipelines</title> |
| * |[ |
| * gst-launch -v openslessrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=recorded.ogg |
| * ]| Record from default audio input and encode to Ogg/Vorbis. |
| * </refsect2> |
| * |
| */ |
| |
| #ifdef HAVE_CONFIG_H |
| # include <config.h> |
| #endif |
| |
| #include "openslessrc.h" |
| |
| GST_DEBUG_CATEGORY_STATIC (opensles_src_debug); |
| #define GST_CAT_DEFAULT opensles_src_debug |
| |
| /* *INDENT-OFF* */ |
| static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src", |
| GST_PAD_SRC, |
| GST_PAD_ALWAYS, |
| GST_STATIC_CAPS ("audio/x-raw, " |
| "format = (string) " GST_AUDIO_NE (S16) ", " |
| "rate = (int) 16000, " |
| "channels = (int) 1, " |
| "layout = (string) interleaved") |
| ); |
| /* *INDENT-ON* */ |
| |
| #define _do_init \ |
| GST_DEBUG_CATEGORY_INIT (opensles_src_debug, "openslessrc", 0, \ |
| "OpenSLES Source"); |
| #define parent_class gst_opensles_src_parent_class |
| G_DEFINE_TYPE_WITH_CODE (GstOpenSLESSrc, gst_opensles_src, |
| GST_TYPE_AUDIO_BASE_SRC, _do_init); |
| |
| static GstAudioRingBuffer * |
| gst_opensles_src_create_ringbuffer (GstAudioBaseSrc * base) |
| { |
| GstAudioRingBuffer *rb; |
| |
| rb = gst_opensles_ringbuffer_new (RB_MODE_SRC); |
| |
| return rb; |
| } |
| |
| static void |
| gst_opensles_src_class_init (GstOpenSLESSrcClass * klass) |
| { |
| GstElementClass *gstelement_class; |
| GstAudioBaseSrcClass *gstaudiobasesrc_class; |
| |
| gstelement_class = (GstElementClass *) klass; |
| gstaudiobasesrc_class = (GstAudioBaseSrcClass *) klass; |
| |
| gst_element_class_add_pad_template (gstelement_class, |
| gst_static_pad_template_get (&src_factory)); |
| |
| gst_element_class_set_static_metadata (gstelement_class, "OpenSL ES Src", |
| "Source/Audio", |
| "Input sound using the OpenSL ES APIs", |
| "Josep Torra <support@fluendo.com>"); |
| |
| gstaudiobasesrc_class->create_ringbuffer = |
| GST_DEBUG_FUNCPTR (gst_opensles_src_create_ringbuffer); |
| } |
| |
| static void |
| gst_opensles_src_init (GstOpenSLESSrc * src) |
| { |
| /* Override some default values to fit on the AudioFlinger behaviour of |
| * processing 20ms buffers as minimum buffer size. */ |
| GST_AUDIO_BASE_SRC (src)->buffer_time = 200000; |
| GST_AUDIO_BASE_SRC (src)->latency_time = 20000; |
| } |