/* * Siren Decoder Gst Element * * @author: Youness Alaoui <kakaroto@kakaroto.homelinux.net> * * 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-sirendec * @title: sirendec * * This decodes audio buffers from the Siren 16 codec (a 16khz extension of * G.722.1) that is meant to be compatible with the Microsoft Windows Live * Messenger(tm) implementation. * * Ref: http://www.polycom.com/company/about_us/technology/siren_g7221/index.html */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include "gstsirendec.h" #include <string.h> GST_DEBUG_CATEGORY (sirendec_debug); #define GST_CAT_DEFAULT (sirendec_debug) #define FRAME_DURATION (20 * GST_MSECOND) static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-siren, " "dct-length = (int) 320")); static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw, format = (string) \"S16LE\", " "rate = (int) 16000, " "channels = (int) 1")); static gboolean gst_siren_dec_start (GstAudioDecoder * dec); static gboolean gst_siren_dec_stop (GstAudioDecoder * dec); static gboolean gst_siren_dec_set_format (GstAudioDecoder * dec, GstCaps * caps); static GstFlowReturn gst_siren_dec_parse (GstAudioDecoder * dec, GstAdapter * adapter, gint * offset, gint * length); static GstFlowReturn gst_siren_dec_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer); G_DEFINE_TYPE (GstSirenDec, gst_siren_dec, GST_TYPE_AUDIO_DECODER); GST_ELEMENT_REGISTER_DEFINE (sirendec, "sirendec", GST_RANK_MARGINAL, GST_TYPE_SIREN_DEC); static void gst_siren_dec_class_init (GstSirenDecClass * klass) { GstElementClass *element_class = GST_ELEMENT_CLASS (klass); GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass); GST_DEBUG_CATEGORY_INIT (sirendec_debug, "sirendec", 0, "sirendec"); gst_element_class_add_static_pad_template (element_class, &srctemplate); gst_element_class_add_static_pad_template (element_class, &sinktemplate); gst_element_class_set_static_metadata (element_class, "Siren Decoder element", "Codec/Decoder/Audio ", "Decode streams encoded with the Siren7 codec into 16bit PCM", "Youness Alaoui <kakaroto@kakaroto.homelinux.net>"); base_class->start = GST_DEBUG_FUNCPTR (gst_siren_dec_start); base_class->stop = GST_DEBUG_FUNCPTR (gst_siren_dec_stop); base_class->set_format = GST_DEBUG_FUNCPTR (gst_siren_dec_set_format); base_class->parse = GST_DEBUG_FUNCPTR (gst_siren_dec_parse); base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_siren_dec_handle_frame); GST_DEBUG ("Class Init done"); } static void gst_siren_dec_init (GstSirenDec * dec) { gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (dec), TRUE); gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST (dec), TRUE); GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (dec)); } static gboolean gst_siren_dec_start (GstAudioDecoder * dec) { GstSirenDec *sdec = GST_SIREN_DEC (dec); GST_DEBUG_OBJECT (dec, "start"); sdec->decoder = Siren7_NewDecoder (16000); /* no flushing please */ gst_audio_decoder_set_drainable (dec, FALSE); return TRUE; } static gboolean gst_siren_dec_stop (GstAudioDecoder * dec) { GstSirenDec *sdec = GST_SIREN_DEC (dec); GST_DEBUG_OBJECT (dec, "stop"); Siren7_CloseDecoder (sdec->decoder); return TRUE; } static gboolean gst_siren_dec_set_format (GstAudioDecoder * bdec, GstCaps * caps) { GstAudioInfo info; gst_audio_info_init (&info); gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16LE, 16000, 1, NULL); return gst_audio_decoder_set_output_format (bdec, &info); } static GstFlowReturn gst_siren_dec_parse (GstAudioDecoder * dec, GstAdapter * adapter, gint * offset, gint * length) { gint size; GstFlowReturn ret; size = gst_adapter_available (adapter); g_return_val_if_fail (size > 0, GST_FLOW_ERROR); /* accept any multiple of frames */ if (size > 40) { ret = GST_FLOW_OK; *offset = 0; *length = size - (size % 40); } else { ret = GST_FLOW_EOS; } return ret; } static GstFlowReturn gst_siren_dec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buf) { GstSirenDec *dec; GstFlowReturn ret = GST_FLOW_OK; GstBuffer *out_buf; guint8 *in_data, *out_data; guint i, size, num_frames; gint out_size; #ifndef GST_DISABLE_GST_DEBUG gint in_size; #endif gint decode_ret; GstMapInfo inmap, outmap; dec = GST_SIREN_DEC (bdec); size = gst_buffer_get_size (buf); GST_LOG_OBJECT (dec, "Received buffer of size %u", size); g_return_val_if_fail (size % 40 == 0, GST_FLOW_ERROR); g_return_val_if_fail (size > 0, GST_FLOW_ERROR); /* process 40 input bytes into 640 output bytes */ num_frames = size / 40; /* this is the input/output size */ #ifndef GST_DISABLE_GST_DEBUG in_size = num_frames * 40; #endif out_size = num_frames * 640; GST_LOG_OBJECT (dec, "we have %u frames, %u in, %u out", num_frames, in_size, out_size); out_buf = gst_audio_decoder_allocate_output_buffer (bdec, out_size); if (out_buf == NULL) goto alloc_failed; /* get the input data for all the frames */ gst_buffer_map (buf, &inmap, GST_MAP_READ); gst_buffer_map (out_buf, &outmap, GST_MAP_WRITE); in_data = inmap.data; out_data = outmap.data; for (i = 0; i < num_frames; i++) { GST_LOG_OBJECT (dec, "Decoding frame %u/%u", i, num_frames); /* decode 40 input bytes to 640 output bytes */ decode_ret = Siren7_DecodeFrame (dec->decoder, in_data, out_data); if (decode_ret != 0) goto decode_error; /* move to next frame */ out_data += 640; in_data += 40; } gst_buffer_unmap (buf, &inmap); gst_buffer_unmap (out_buf, &outmap); GST_LOG_OBJECT (dec, "Finished decoding"); /* might really be multiple frames, * but was treated as one for all purposes here */ ret = gst_audio_decoder_finish_frame (bdec, out_buf, 1); done: return ret; /* ERRORS */ alloc_failed: { GST_DEBUG_OBJECT (dec, "failed to pad_alloc buffer: %d (%s)", ret, gst_flow_get_name (ret)); goto done; } decode_error: { GST_AUDIO_DECODER_ERROR (bdec, 1, STREAM, DECODE, (NULL), ("Error decoding frame: %d", decode_ret), ret); if (ret == GST_FLOW_OK) gst_audio_decoder_finish_frame (bdec, NULL, 1); gst_buffer_unref (out_buf); goto done; } }