/* GStreamer Adaptive Multi-Rate Wide-Band (AMR-WB) plugin * Copyright (C) 2006 Edgard Lima * * 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., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ /** * SECTION:element-amrwbenc * @see_also: #GstAmrwbDec, #GstAmrwbParse * * AMR wideband encoder based on the * reference codec implementation. * * * Example launch line * |[ * gst-launch filesrc location=abc.wav ! wavparse ! audioresample ! audioconvert ! amrwbenc ! filesink location=abc.amr * ]| * Please not that the above stream misses the header, that is needed to play * the stream. * */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include "gstamrwbenc.h" /* these defines are not in all .h files */ #ifndef MR660 #define MR660 0 #define MR885 1 #define MR1265 2 #define MR1425 2 #define MR1585 3 #define MR1825 4 #define MR1985 5 #define MR2305 6 #define MR2385 7 #define MRDTX 8 #endif static GType gst_amrwbenc_bandmode_get_type () { static GType gst_amrwbenc_bandmode_type = 0; static GEnumValue gst_amrwbenc_bandmode[] = { {MR660, "MR660", "MR660"}, {MR885, "MR885", "MR885"}, {MR1265, "MR1265", "MR1265"}, {MR1425, "MR1425", "MR1425"}, {MR1585, "MR1585", "MR1585"}, {MR1825, "MR1825", "MR1825"}, {MR1985, "MR1985", "MR1985"}, {MR2305, "MR2305", "MR2305"}, {MR2385, "MR2385", "MR2385"}, {MRDTX, "MRDTX", "MRDTX"}, {0, NULL, NULL}, }; if (!gst_amrwbenc_bandmode_type) { gst_amrwbenc_bandmode_type = g_enum_register_static ("GstAmrwbEncBandMode", gst_amrwbenc_bandmode); } return gst_amrwbenc_bandmode_type; } #define GST_AMRWBENC_BANDMODE_TYPE (gst_amrwbenc_bandmode_get_type()) #define BANDMODE_DEFAULT MR660 enum { PROP_0, PROP_BANDMODE }; static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-int, " "width = (int) 16, " "depth = (int) 16, " "signed = (boolean) TRUE, " "endianness = (int) BYTE_ORDER, " "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/AMR-WB, " "rate = (int) 16000, " "channels = (int) 1") ); GST_DEBUG_CATEGORY_STATIC (gst_amrwbenc_debug); #define GST_CAT_DEFAULT gst_amrwbenc_debug static void gst_amrwbenc_finalize (GObject * object); static GstFlowReturn gst_amrwbenc_chain (GstPad * pad, GstBuffer * buffer); static gboolean gst_amrwbenc_setcaps (GstPad * pad, GstCaps * caps); static GstStateChangeReturn gst_amrwbenc_state_change (GstElement * element, GstStateChange transition); static void _do_init (GType object_type) { const GInterfaceInfo preset_interface_info = { NULL, /* interface init */ NULL, /* interface finalize */ NULL /* interface_data */ }; g_type_add_interface_static (object_type, GST_TYPE_PRESET, &preset_interface_info); GST_DEBUG_CATEGORY_INIT (gst_amrnbenc_debug, "amrwbenc", 0, "AMR-WB audio encoder"); } GST_BOILERPLATE_FULL (GstAmrwbEnc, gst_amrwbenc, GstElement, GST_TYPE_ELEMENT, _do_init); static void gst_amrwbenc_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstAmrwbEnc *self = GST_AMRWBENC (object); switch (prop_id) { case PROP_BANDMODE: self->bandmode = g_value_get_enum (value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } return; } static void gst_amrwbenc_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstAmrwbEnc *self = GST_AMRWBENC (object); switch (prop_id) { case PROP_BANDMODE: g_value_set_enum (value, self->bandmode); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } return; } static void gst_amrwbenc_base_init (gpointer klass) { GstElementClass *element_class = GST_ELEMENT_CLASS (klass); GstElementDetails details = GST_ELEMENT_DETAILS ("AMR-WB audio encoder", "Codec/Encoder/Audio", "Adaptive Multi-Rate Wideband audio encoder", "Renato Araujo "); 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_details (element_class, &details); } static void gst_amrwbenc_class_init (GstAmrwbEncClass * klass) { GObjectClass *object_class = G_OBJECT_CLASS (klass); GstElementClass *element_class = GST_ELEMENT_CLASS (klass); object_class->finalize = gst_amrwbenc_finalize; object_class->set_property = gst_amrwbenc_set_property; object_class->get_property = gst_amrwbenc_get_property; g_object_class_install_property (object_class, PROP_BANDMODE, g_param_spec_enum ("band-mode", "Band Mode", "Encoding Band Mode (Kbps)", GST_AMRWBENC_BANDMODE_TYPE, BANDMODE_DEFAULT, G_PARAM_READWRITE | G_PARAM_CONSTRUCT)); element_class->change_state = GST_DEBUG_FUNCPTR (gst_amrwbenc_state_change); } static void gst_amrwbenc_init (GstAmrwbEnc * amrwbenc, GstAmrwbEncClass * klass) { /* create the sink pad */ amrwbenc->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink"); gst_pad_set_setcaps_function (amrwbenc->sinkpad, gst_amrwbenc_setcaps); gst_pad_set_chain_function (amrwbenc->sinkpad, gst_amrwbenc_chain); gst_element_add_pad (GST_ELEMENT (amrwbenc), amrwbenc->sinkpad); /* create the src pad */ amrwbenc->srcpad = gst_pad_new_from_static_template (&src_template, "src"); gst_pad_use_fixed_caps (amrwbenc->srcpad); gst_element_add_pad (GST_ELEMENT (amrwbenc), amrwbenc->srcpad); amrwbenc->adapter = gst_adapter_new (); /* init rest */ amrwbenc->handle = NULL; amrwbenc->channels = 0; amrwbenc->rate = 0; amrwbenc->ts = 0; } static void gst_amrwbenc_finalize (GObject * object) { GstAmrwbEnc *amrwbenc; amrwbenc = GST_AMRWBENC (object); g_object_unref (G_OBJECT (amrwbenc->adapter)); amrwbenc->adapter = NULL; G_OBJECT_CLASS (parent_class)->finalize (object); } static gboolean gst_amrwbenc_setcaps (GstPad * pad, GstCaps * caps) { GstStructure *structure; GstAmrwbEnc *amrwbenc; GstCaps *copy; amrwbenc = GST_AMRWBENC (GST_PAD_PARENT (pad)); structure = gst_caps_get_structure (caps, 0); /* get channel count */ gst_structure_get_int (structure, "channels", &amrwbenc->channels); gst_structure_get_int (structure, "rate", &amrwbenc->rate); /* this is not wrong but will sound bad */ if (amrwbenc->channels != 1) { GST_WARNING ("amrwbdec is only optimized for mono channels"); } if (amrwbenc->rate != 16000) { GST_WARNING ("amrwbdec is only optimized for 16000 Hz samplerate"); } /* create reverse caps */ copy = gst_caps_new_simple ("audio/AMR-WB", "channels", G_TYPE_INT, amrwbenc->channels, "rate", G_TYPE_INT, amrwbenc->rate, NULL); gst_pad_set_caps (amrwbenc->srcpad, copy); gst_caps_unref (copy); return TRUE; } static GstFlowReturn gst_amrwbenc_chain (GstPad * pad, GstBuffer * buffer) { GstAmrwbEnc *amrwbenc; GstFlowReturn ret = GST_FLOW_OK; const int buffer_size = sizeof (Word16) * L_FRAME16k; amrwbenc = GST_AMRWBENC (gst_pad_get_parent (pad)); g_return_val_if_fail (amrwbenc->handle, GST_FLOW_WRONG_STATE); if (amrwbenc->rate == 0 || amrwbenc->channels == 0) { ret = GST_FLOW_NOT_NEGOTIATED; goto done; } /* discontinuity clears adapter, FIXME, maybe we can set some * encoder flag to mask the discont. */ if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT)) { gst_adapter_clear (amrwbenc->adapter); amrwbenc->ts = 0; amrwbenc->discont = TRUE; } if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) amrwbenc->ts = GST_BUFFER_TIMESTAMP (buffer); ret = GST_FLOW_OK; gst_adapter_push (amrwbenc->adapter, buffer); /* Collect samples until we have enough for an output frame */ while (gst_adapter_available (amrwbenc->adapter) >= buffer_size) { GstBuffer *out; guint8 *data; gint outsize; out = gst_buffer_new_and_alloc (buffer_size); GST_BUFFER_DURATION (out) = GST_SECOND * L_FRAME16k / (amrwbenc->rate * amrwbenc->channels); GST_BUFFER_TIMESTAMP (out) = amrwbenc->ts; if (amrwbenc->ts != -1) { amrwbenc->ts += GST_BUFFER_DURATION (out); } if (amrwbenc->discont) { GST_BUFFER_FLAG_SET (out, GST_BUFFER_FLAG_DISCONT); amrwbenc->discont = FALSE; } gst_buffer_set_caps (out, gst_pad_get_caps (amrwbenc->srcpad)); data = (guint8 *) gst_adapter_peek (amrwbenc->adapter, buffer_size); /* encode */ outsize = E_IF_encode (amrwbenc->handle, amrwbenc->bandmode, (Word16 *) data, (UWord8 *) GST_BUFFER_DATA (out), 0); gst_adapter_flush (amrwbenc->adapter, buffer_size); GST_BUFFER_SIZE (out) = outsize; /* play */ if ((ret = gst_pad_push (amrwbenc->srcpad, out)) != GST_FLOW_OK) break; } done: gst_object_unref (amrwbenc); return ret; } static GstStateChangeReturn gst_amrwbenc_state_change (GstElement * element, GstStateChange transition) { GstAmrwbEnc *amrwbenc; GstStateChangeReturn ret; amrwbenc = GST_AMRWBENC (element); switch (transition) { case GST_STATE_CHANGE_NULL_TO_READY: if (!(amrwbenc->handle = E_IF_init ())) return GST_STATE_CHANGE_FAILURE; break; case GST_STATE_CHANGE_READY_TO_PAUSED: amrwbenc->rate = 0; amrwbenc->channels = 0; amrwbenc->ts = 0; amrwbenc->discont = FALSE; gst_adapter_clear (amrwbenc->adapter); break; default: break; } ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); switch (transition) { case GST_STATE_CHANGE_READY_TO_NULL: E_IF_exit (amrwbenc->handle); break; default: break; } return ret; }