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Conflicts: NEWS RELEASE common configure.ac docs/libs/gst-plugins-bad-libs-sections.txt docs/plugins/gst-plugins-bad-plugins.args docs/plugins/gst-plugins-bad-plugins.hierarchy docs/plugins/gst-plugins-bad-plugins.interfaces docs/plugins/inspect/plugin-adpcmdec.xml docs/plugins/inspect/plugin-adpcmenc.xml docs/plugins/inspect/plugin-assrender.xml docs/plugins/inspect/plugin-audiovisualizers.xml docs/plugins/inspect/plugin-autoconvert.xml docs/plugins/inspect/plugin-bayer.xml docs/plugins/inspect/plugin-bz2.xml docs/plugins/inspect/plugin-camerabin2.xml docs/plugins/inspect/plugin-celt.xml docs/plugins/inspect/plugin-dataurisrc.xml docs/plugins/inspect/plugin-debugutilsbad.xml docs/plugins/inspect/plugin-dtmf.xml docs/plugins/inspect/plugin-dtsdec.xml docs/plugins/inspect/plugin-dvbsuboverlay.xml docs/plugins/inspect/plugin-dvdspu.xml docs/plugins/inspect/plugin-faac.xml docs/plugins/inspect/plugin-faad.xml docs/plugins/inspect/plugin-gsm.xml docs/plugins/inspect/plugin-h264parse.xml docs/plugins/inspect/plugin-mms.xml docs/plugins/inspect/plugin-modplug.xml docs/plugins/inspect/plugin-mpeg2enc.xml docs/plugins/inspect/plugin-mpegdemux2.xml docs/plugins/inspect/plugin-mpegtsdemux.xml docs/plugins/inspect/plugin-mpegvideoparse.xml docs/plugins/inspect/plugin-mplex.xml docs/plugins/inspect/plugin-pcapparse.xml docs/plugins/inspect/plugin-rawparse.xml docs/plugins/inspect/plugin-rtpmux.xml docs/plugins/inspect/plugin-rtpvp8.xml docs/plugins/inspect/plugin-scaletempo.xml docs/plugins/inspect/plugin-schro.xml docs/plugins/inspect/plugin-sdp.xml docs/plugins/inspect/plugin-segmentclip.xml docs/plugins/inspect/plugin-shm.xml docs/plugins/inspect/plugin-videomaxrate.xml docs/plugins/inspect/plugin-videoparsersbad.xml docs/plugins/inspect/plugin-vp8.xml docs/plugins/inspect/plugin-y4mdec.xml ext/celt/gstceltdec.c ext/dts/gstdtsdec.c ext/modplug/gstmodplug.cc ext/opus/gstopusenc.c gst-libs/gst/video/gstbasevideocodec.c gst-libs/gst/video/gstbasevideocodec.h gst-libs/gst/video/gstbasevideodecoder.c gst-libs/gst/video/gstbasevideodecoder.h gst-libs/gst/video/gstbasevideoencoder.c gst-libs/gst/video/gstbasevideoencoder.h gst/adpcmdec/Makefile.am gst/audiovisualizers/gstbaseaudiovisualizer.c gst/h264parse/gsth264parse.c gst/mpegdemux/mpegtsparse.c gst/mpegtsdemux/mpegtsbase.c gst/mpegtsdemux/mpegtspacketizer.c gst/mpegtsdemux/mpegtsparse.c gst/mpegtsdemux/tsdemux.c gst/mpegtsdemux/tsdemux.h gst/mxf/mxfdemux.c gst/rawparse/gstaudioparse.c gst/videoparsers/gsth263parse.c gst/videoparsers/gsth264parse.c sys/d3dvideosink/d3dvideosink.c sys/decklink/gstdecklinksink.cpp sys/dvb/gstdvbsrc.c sys/shm/gstshmsrc.c sys/vdpau/h264/gstvdph264dec.c sys/vdpau/mpeg/gstvdpmpegdec.c tests/examples/opencv/gst_element_print_properties.c win32/common/config.h
289 lines
7.5 KiB
C
289 lines
7.5 KiB
C
/*
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* Siren Decoder Gst Element
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*
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* @author: Youness Alaoui <kakaroto@kakaroto.homelinux.net>
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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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*
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*/
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/**
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* SECTION:element-sirendec
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*
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* This decodes audio buffers from the Siren 16 codec (a 16khz extension of
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* G.722.1) that is meant to be compatible with the Microsoft Windows Live
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* Messenger(tm) implementation.
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*
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* Ref: http://www.polycom.com/company/about_us/technology/siren_g7221/index.html
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include "gstsirendec.h"
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#include <string.h>
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GST_DEBUG_CATEGORY (sirendec_debug);
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#define GST_CAT_DEFAULT (sirendec_debug)
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#define FRAME_DURATION (20 * GST_MSECOND)
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static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-siren, " "dct-length = (int) 320"));
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static GstStaticPadTemplate srctemplate = 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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"endianness = (int) 1234, "
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"signed = (boolean) true, "
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"rate = (int) 16000, " "channels = (int) 1"));
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/* signals and args */
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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enum
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{
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ARG_0,
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};
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static gboolean gst_siren_dec_start (GstAudioDecoder * dec);
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static gboolean gst_siren_dec_stop (GstAudioDecoder * dec);
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static gboolean gst_siren_dec_set_format (GstAudioDecoder * dec,
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GstCaps * caps);
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static gboolean gst_siren_dec_parse (GstAudioDecoder * dec,
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GstAdapter * adapter, gint * offset, gint * length);
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static GstFlowReturn gst_siren_dec_handle_frame (GstAudioDecoder * dec,
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GstBuffer * buffer);
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static void
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_do_init (GType type)
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{
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GST_DEBUG_CATEGORY_INIT (sirendec_debug, "sirendec", 0, "sirendec");
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}
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GST_BOILERPLATE_FULL (GstSirenDec, gst_siren_dec, GstAudioDecoder,
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GST_TYPE_AUDIO_DECODER, _do_init);
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static void
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gst_siren_dec_base_init (gpointer klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&srctemplate));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&sinktemplate));
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gst_element_class_set_details_simple (element_class, "Siren Decoder element",
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"Codec/Decoder/Audio ",
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"Decode streams encoded with the Siren7 codec into 16bit PCM",
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"Youness Alaoui <kakaroto@kakaroto.homelinux.net>");
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}
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static void
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gst_siren_dec_class_init (GstSirenDecClass * klass)
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{
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GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass);
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GST_DEBUG ("Initializing Class");
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base_class->start = GST_DEBUG_FUNCPTR (gst_siren_dec_start);
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base_class->stop = GST_DEBUG_FUNCPTR (gst_siren_dec_stop);
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base_class->set_format = GST_DEBUG_FUNCPTR (gst_siren_dec_set_format);
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base_class->parse = GST_DEBUG_FUNCPTR (gst_siren_dec_parse);
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base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_siren_dec_handle_frame);
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GST_DEBUG ("Class Init done");
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}
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static void
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gst_siren_dec_init (GstSirenDec * dec, GstSirenDecClass * klass)
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{
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}
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static gboolean
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gst_siren_dec_start (GstAudioDecoder * dec)
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{
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GstSirenDec *sdec = GST_SIREN_DEC (dec);
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GST_DEBUG_OBJECT (dec, "start");
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sdec->decoder = Siren7_NewDecoder (16000);;
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/* no flushing please */
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gst_audio_decoder_set_drainable (dec, FALSE);
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return TRUE;
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}
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static gboolean
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gst_siren_dec_stop (GstAudioDecoder * dec)
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{
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GstSirenDec *sdec = GST_SIREN_DEC (dec);
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GST_DEBUG_OBJECT (dec, "stop");
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Siren7_CloseDecoder (sdec->decoder);
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return TRUE;
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}
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static gboolean
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gst_siren_dec_negotiate (GstSirenDec * dec)
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{
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gboolean res;
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GstCaps *outcaps;
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outcaps = gst_static_pad_template_get_caps (&srctemplate);
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res = gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), outcaps);
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gst_caps_unref (outcaps);
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return res;
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}
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static gboolean
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gst_siren_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps)
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{
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GstSirenDec *dec;
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dec = GST_SIREN_DEC (bdec);
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return gst_siren_dec_negotiate (dec);
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}
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static GstFlowReturn
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gst_siren_dec_parse (GstAudioDecoder * dec, GstAdapter * adapter,
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gint * offset, gint * length)
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{
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gint size;
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GstFlowReturn ret;
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size = gst_adapter_available (adapter);
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g_return_val_if_fail (size > 0, GST_FLOW_ERROR);
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/* accept any multiple of frames */
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if (size > 40) {
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ret = GST_FLOW_OK;
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*offset = 0;
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*length = size - (size % 40);
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} else {
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ret = GST_FLOW_UNEXPECTED;
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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_siren_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf)
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{
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GstSirenDec *dec;
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GstFlowReturn ret = GST_FLOW_OK;
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GstBuffer *out_buf;
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guint8 *in_data, *out_data;
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guint i, size, num_frames;
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gint out_size, in_size;
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gint decode_ret;
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dec = GST_SIREN_DEC (bdec);
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size = GST_BUFFER_SIZE (buf);
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GST_LOG_OBJECT (dec, "Received buffer of size %u", size);
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g_return_val_if_fail (size % 40 == 0, GST_FLOW_ERROR);
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g_return_val_if_fail (size > 0, GST_FLOW_ERROR);
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/* process 40 input bytes into 640 output bytes */
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num_frames = size / 40;
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/* this is the input/output size */
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in_size = num_frames * 40;
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out_size = num_frames * 640;
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GST_LOG_OBJECT (dec, "we have %u frames, %u in, %u out", num_frames, in_size,
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out_size);
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/* allow and handle un-negotiated input */
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if (G_UNLIKELY (GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)) == NULL)) {
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gst_siren_dec_negotiate (dec);
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}
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/* get a buffer */
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ret = gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), -1,
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out_size, GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)), &out_buf);
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if (ret != GST_FLOW_OK)
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goto alloc_failed;
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/* get the input data for all the frames */
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in_data = GST_BUFFER_DATA (buf);
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out_data = GST_BUFFER_DATA (out_buf);
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for (i = 0; i < num_frames; i++) {
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GST_LOG_OBJECT (dec, "Decoding frame %u/%u", i, num_frames);
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/* decode 40 input bytes to 640 output bytes */
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decode_ret = Siren7_DecodeFrame (dec->decoder, in_data, out_data);
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if (decode_ret != 0)
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goto decode_error;
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/* move to next frame */
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out_data += 640;
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in_data += 40;
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}
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GST_LOG_OBJECT (dec, "Finished decoding");
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/* might really be multiple frames,
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* but was treated as one for all purposes here */
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ret = gst_audio_decoder_finish_frame (bdec, out_buf, 1);
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done:
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return ret;
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/* ERRORS */
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alloc_failed:
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{
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GST_DEBUG_OBJECT (dec, "failed to pad_alloc buffer: %d (%s)", ret,
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gst_flow_get_name (ret));
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goto done;
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}
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decode_error:
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{
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GST_AUDIO_DECODER_ERROR (bdec, 1, STREAM, DECODE, (NULL),
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("Error decoding frame: %d", decode_ret), ret);
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if (ret == GST_FLOW_OK)
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gst_audio_decoder_finish_frame (bdec, NULL, 1);
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gst_buffer_unref (out_buf);
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goto done;
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}
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}
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gboolean
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gst_siren_dec_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "sirendec",
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GST_RANK_MARGINAL, GST_TYPE_SIREN_DEC);
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}
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