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252 lines
7 KiB
C
252 lines
7 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., 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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/**
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* SECTION:element-sirendec
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* @title: 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, format = (string) \"S16LE\", "
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"rate = (int) 16000, " "channels = (int) 1"));
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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 GstFlowReturn 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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G_DEFINE_TYPE (GstSirenDec, gst_siren_dec, GST_TYPE_AUDIO_DECODER);
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GST_ELEMENT_REGISTER_DEFINE (sirendec, "sirendec",
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GST_RANK_MARGINAL, GST_TYPE_SIREN_DEC);
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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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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass);
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GST_DEBUG_CATEGORY_INIT (sirendec_debug, "sirendec", 0, "sirendec");
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gst_element_class_add_static_pad_template (element_class, &srctemplate);
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gst_element_class_add_static_pad_template (element_class, &sinktemplate);
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gst_element_class_set_static_metadata (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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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)
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{
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gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (dec), TRUE);
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gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST
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(dec), TRUE);
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GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (dec));
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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_set_format (GstAudioDecoder * bdec, GstCaps * caps)
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{
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GstAudioInfo info;
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gst_audio_info_init (&info);
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gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16LE, 16000, 1, NULL);
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return gst_audio_decoder_set_output_format (bdec, &info);
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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_EOS;
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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;
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#ifndef GST_DISABLE_GST_DEBUG
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gint in_size;
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#endif
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gint decode_ret;
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GstMapInfo inmap, outmap;
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dec = GST_SIREN_DEC (bdec);
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size = gst_buffer_get_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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#ifndef GST_DISABLE_GST_DEBUG
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in_size = num_frames * 40;
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#endif
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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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out_buf = gst_audio_decoder_allocate_output_buffer (bdec, out_size);
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if (out_buf == NULL)
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goto alloc_failed;
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/* get the input data for all the frames */
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gst_buffer_map (buf, &inmap, GST_MAP_READ);
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gst_buffer_map (out_buf, &outmap, GST_MAP_WRITE);
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in_data = inmap.data;
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out_data = outmap.data;
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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_buffer_unmap (buf, &inmap);
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gst_buffer_unmap (out_buf, &outmap);
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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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