/* GStreamer Adaptive Multi-Rate Narrow-Band (AMR-NB) plugin * Copyright (C) 2004 Ronald Bultje * * 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-amrwbdec * @see_also: #GstAmrwbEnc * * AMR wideband decoder based on the * opencore codec implementation. * * * Example launch line * |[ * gst-launch filesrc location=abc.amr ! amrparse ! amrwbdec ! audioresample ! audioconvert ! alsasink * ]| * */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include #include "amrwbdec.h" static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/AMR-WB, " "rate = (int) 16000, " "channels = (int) 1") ); static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw, " "format = (string) " GST_AUDIO_NE (S16) ", " "layout = (string) interleaved, " "rate = (int) 16000, " "channels = (int) 1") ); GST_DEBUG_CATEGORY_STATIC (gst_amrwbdec_debug); #define GST_CAT_DEFAULT gst_amrwbdec_debug #define L_FRAME16k 320 /* Frame size at 16kHz */ static const unsigned char block_size[16] = { 18, 24, 33, 37, 41, 47, 51, 59, 61, 6, 0, 0, 0, 0, 1, 1 }; static gboolean gst_amrwbdec_start (GstAudioDecoder * dec); static gboolean gst_amrwbdec_stop (GstAudioDecoder * dec); static gboolean gst_amrwbdec_set_format (GstAudioDecoder * dec, GstCaps * caps); static gboolean gst_amrwbdec_parse (GstAudioDecoder * dec, GstAdapter * adapter, gint * offset, gint * length); static GstFlowReturn gst_amrwbdec_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer); #define gst_amrwbdec_parent_class parent_class G_DEFINE_TYPE (GstAmrwbDec, gst_amrwbdec, GST_TYPE_AUDIO_DECODER); static void gst_amrwbdec_class_init (GstAmrwbDecClass * klass) { GstAudioDecoderClass *base_class = GST_AUDIO_DECODER_CLASS (klass); GstElementClass *element_class = GST_ELEMENT_CLASS (klass); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&sink_template)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&src_template)); gst_element_class_set_static_metadata (element_class, "AMR-WB audio decoder", "Codec/Decoder/Audio", "Adaptive Multi-Rate Wideband audio decoder", "Renato Araujo "); base_class->start = GST_DEBUG_FUNCPTR (gst_amrwbdec_start); base_class->stop = GST_DEBUG_FUNCPTR (gst_amrwbdec_stop); base_class->set_format = GST_DEBUG_FUNCPTR (gst_amrwbdec_set_format); base_class->parse = GST_DEBUG_FUNCPTR (gst_amrwbdec_parse); base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_amrwbdec_handle_frame); GST_DEBUG_CATEGORY_INIT (gst_amrwbdec_debug, "amrwbdec", 0, "AMR-WB audio decoder"); } static void gst_amrwbdec_init (GstAmrwbDec * amrwbdec) { gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (amrwbdec), TRUE); } static gboolean gst_amrwbdec_start (GstAudioDecoder * dec) { GstAmrwbDec *amrwbdec = GST_AMRWBDEC (dec); GST_DEBUG_OBJECT (dec, "start"); if (!(amrwbdec->handle = D_IF_init ())) return FALSE; amrwbdec->rate = 0; amrwbdec->channels = 0; return TRUE; } static gboolean gst_amrwbdec_stop (GstAudioDecoder * dec) { GstAmrwbDec *amrwbdec = GST_AMRWBDEC (dec); GST_DEBUG_OBJECT (dec, "stop"); D_IF_exit (amrwbdec->handle); return TRUE; } static gboolean gst_amrwbdec_set_format (GstAudioDecoder * dec, GstCaps * caps) { GstStructure *structure; GstAmrwbDec *amrwbdec; GstAudioInfo info; amrwbdec = GST_AMRWBDEC (dec); structure = gst_caps_get_structure (caps, 0); /* get channel count */ gst_structure_get_int (structure, "channels", &amrwbdec->channels); gst_structure_get_int (structure, "rate", &amrwbdec->rate); /* create reverse caps */ gst_audio_info_init (&info); gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16, amrwbdec->rate, amrwbdec->channels, NULL); gst_audio_decoder_set_output_format (dec, &info); return TRUE; } static GstFlowReturn gst_amrwbdec_parse (GstAudioDecoder * dec, GstAdapter * adapter, gint * offset, gint * length) { GstAmrwbDec *amrwbdec = GST_AMRWBDEC (dec); guint8 header[1]; guint size; gboolean sync, eos; gint block, mode; size = gst_adapter_available (adapter); g_return_val_if_fail (size > 0, GST_FLOW_ERROR); gst_audio_decoder_get_parse_state (dec, &sync, &eos); /* need to peek data to get the size */ if (size < 1) return GST_FLOW_ERROR; gst_adapter_copy (adapter, header, 0, 1); mode = (header[0] >> 3) & 0x0F; block = block_size[mode]; GST_DEBUG_OBJECT (amrwbdec, "mode %d, block %d", mode, block); if (block) { if (block > size) return GST_FLOW_EOS; *offset = 0; *length = block; } else { /* no frame yet, skip one byte */ GST_LOG_OBJECT (amrwbdec, "skipping byte"); *offset = 1; return GST_FLOW_EOS; } return GST_FLOW_OK; } static GstFlowReturn gst_amrwbdec_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer) { GstAmrwbDec *amrwbdec; GstBuffer *out; GstMapInfo inmap, outmap; amrwbdec = GST_AMRWBDEC (dec); /* no fancy flushing */ if (!buffer || !gst_buffer_get_size (buffer)) return GST_FLOW_OK; /* the library seems to write into the source data, hence the copy. */ /* should be no problem */ gst_buffer_map (buffer, &inmap, GST_MAP_READ); /* get output */ out = gst_buffer_new_and_alloc (sizeof (gint16) * L_FRAME16k); gst_buffer_map (out, &outmap, GST_MAP_WRITE); /* decode */ D_IF_decode (amrwbdec->handle, (unsigned char *) inmap.data, (short int *) outmap.data, _good_frame); gst_buffer_unmap (out, &outmap); gst_buffer_unmap (buffer, &inmap); /* send out */ return gst_audio_decoder_finish_frame (dec, out, 1); }