/* 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., 59 Temple Place - Suite 330, * Boston, MA 02111-1307, USA. */ /** * SECTION:element-amrnbenc * @see_also: #GstAmrnbDec, #GstAmrnbParse * * * * This is an AMR narrowband encoder based on the * reference codec implementation. * * Example launch line * * * gst-launch filesrc location=abc.wav ! wavparse ! audioresample ! audioconvert ! amrnbenc ! 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 "amrnbenc.h" static GType gst_amrnbenc_bandmode_get_type () { static GType gst_amrnbenc_bandmode_type = 0; static const GEnumValue gst_amrnbenc_bandmode[] = { {MR475, "MR475", "MR475"}, {MR515, "MR515", "MR515"}, {MR59, "MR59", "MR59"}, {MR67, "MR67", "MR67"}, {MR74, "MR74", "MR74"}, {MR795, "MR795", "MR795"}, {MR102, "MR102", "MR102"}, {MR122, "MR122", "MR122"}, {MRDTX, "MRDTX", "MRDTX"}, {0, NULL, NULL}, }; if (!gst_amrnbenc_bandmode_type) { gst_amrnbenc_bandmode_type = g_enum_register_static ("GstAmrnbEncBandMode", gst_amrnbenc_bandmode); } return gst_amrnbenc_bandmode_type; } #define GST_AMRNBENC_BANDMODE_TYPE (gst_amrnbenc_bandmode_get_type()) #define BANDMODE_DEFAULT MR122 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) 8000," "channels = (int) 1") ); static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/AMR, " "rate = (int) 8000, " "channels = (int) 1") ); GST_DEBUG_CATEGORY_STATIC (gst_amrnbenc_debug); #define GST_CAT_DEFAULT gst_amrnbenc_debug static void gst_amrnbenc_finalize (GObject * object); static GstFlowReturn gst_amrnbenc_chain (GstPad * pad, GstBuffer * buffer); static gboolean gst_amrnbenc_setcaps (GstPad * pad, GstCaps * caps); static GstStateChangeReturn gst_amrnbenc_state_change (GstElement * element, GstStateChange transition); #define _do_init(bla) \ GST_DEBUG_CATEGORY_INIT (gst_amrnbenc_debug, "amrnbenc", 0, "AMR-NB audio encoder"); GST_BOILERPLATE_FULL (GstAmrnbEnc, gst_amrnbenc, GstElement, GST_TYPE_ELEMENT, _do_init); static void gst_amrnbenc_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { GstAmrnbEnc *self = GST_AMRNBENC (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_amrnbenc_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { GstAmrnbEnc *self = GST_AMRNBENC (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_amrnbenc_base_init (gpointer klass) { GstElementClass *element_class = GST_ELEMENT_CLASS (klass); GstElementDetails details = GST_ELEMENT_DETAILS ("AMR-NB audio encoder", "Codec/Encoder/Audio", "Adaptive Multi-Rate Narrow-Band audio encoder", "Ronald Bultje , " "Wim Taymans "); 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_amrnbenc_class_init (GstAmrnbEncClass * klass) { GObjectClass *object_class = G_OBJECT_CLASS (klass); GstElementClass *element_class = GST_ELEMENT_CLASS (klass); object_class->set_property = gst_amrnbenc_set_property; object_class->get_property = gst_amrnbenc_get_property; object_class->finalize = gst_amrnbenc_finalize; g_object_class_install_property (object_class, PROP_BANDMODE, g_param_spec_enum ("band-mode", "Band Mode", "Encoding Band Mode (Kbps)", GST_AMRNBENC_BANDMODE_TYPE, BANDMODE_DEFAULT, G_PARAM_READWRITE | G_PARAM_CONSTRUCT)); element_class->change_state = GST_DEBUG_FUNCPTR (gst_amrnbenc_state_change); } static void gst_amrnbenc_init (GstAmrnbEnc * amrnbenc, GstAmrnbEncClass * klass) { /* create the sink pad */ amrnbenc->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink"); gst_pad_set_setcaps_function (amrnbenc->sinkpad, gst_amrnbenc_setcaps); gst_pad_set_chain_function (amrnbenc->sinkpad, gst_amrnbenc_chain); gst_element_add_pad (GST_ELEMENT (amrnbenc), amrnbenc->sinkpad); /* create the src pad */ amrnbenc->srcpad = gst_pad_new_from_static_template (&src_template, "src"); gst_pad_use_fixed_caps (amrnbenc->srcpad); gst_element_add_pad (GST_ELEMENT (amrnbenc), amrnbenc->srcpad); amrnbenc->adapter = gst_adapter_new (); /* init rest */ amrnbenc->handle = NULL; } static void gst_amrnbenc_finalize (GObject * object) { GstAmrnbEnc *amrnbenc; amrnbenc = GST_AMRNBENC (object); g_object_unref (G_OBJECT (amrnbenc->adapter)); amrnbenc->adapter = NULL; G_OBJECT_CLASS (parent_class)->finalize (object); } static gboolean gst_amrnbenc_setcaps (GstPad * pad, GstCaps * caps) { GstStructure *structure; GstAmrnbEnc *amrnbenc; GstCaps *copy; amrnbenc = GST_AMRNBENC (GST_PAD_PARENT (pad)); structure = gst_caps_get_structure (caps, 0); /* get channel count */ gst_structure_get_int (structure, "channels", &amrnbenc->channels); gst_structure_get_int (structure, "rate", &amrnbenc->rate); /* this is not wrong but will sound bad */ if (amrnbenc->channels != 1) { g_warning ("amrnbdec is only optimized for mono channels"); } if (amrnbenc->rate != 8000) { g_warning ("amrnbdec is only optimized for 8000 Hz samplerate"); } /* create reverse caps */ copy = gst_caps_new_simple ("audio/AMR", "channels", G_TYPE_INT, amrnbenc->channels, "rate", G_TYPE_INT, amrnbenc->rate, NULL); /* precalc duration as it's constant now */ amrnbenc->duration = gst_util_uint64_scale_int (160, GST_SECOND, amrnbenc->rate * amrnbenc->channels); gst_pad_set_caps (amrnbenc->srcpad, copy); gst_caps_unref (copy); return TRUE; } static GstFlowReturn gst_amrnbenc_chain (GstPad * pad, GstBuffer * buffer) { GstAmrnbEnc *amrnbenc; GstFlowReturn ret; amrnbenc = GST_AMRNBENC (GST_PAD_PARENT (pad)); g_return_val_if_fail (amrnbenc->handle, GST_FLOW_WRONG_STATE); if (amrnbenc->rate == 0 || amrnbenc->channels == 0) goto not_negotiated; /* 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 (amrnbenc->adapter); amrnbenc->ts = 0; } /* take latest timestamp, FIXME timestamp is the one of the * first buffer in the adapter. */ if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) amrnbenc->ts = GST_BUFFER_TIMESTAMP (buffer); ret = GST_FLOW_OK; gst_adapter_push (amrnbenc->adapter, buffer); /* Collect samples until we have enough for an output frame */ while (gst_adapter_available (amrnbenc->adapter) >= 320) { GstBuffer *out; guint8 *data; gint outsize; /* get output, max size is 32 */ out = gst_buffer_new_and_alloc (32); GST_BUFFER_DURATION (out) = amrnbenc->duration; GST_BUFFER_TIMESTAMP (out) = amrnbenc->ts; if (amrnbenc->ts != -1) amrnbenc->ts += amrnbenc->duration; gst_buffer_set_caps (out, GST_PAD_CAPS (amrnbenc->srcpad)); /* The AMR encoder actually writes into the source data buffers it gets */ data = gst_adapter_take (amrnbenc->adapter, 320); /* encode */ outsize = Encoder_Interface_Encode (amrnbenc->handle, amrnbenc->bandmode, (short *) data, (guint8 *) GST_BUFFER_DATA (out), 0); g_free (data); GST_BUFFER_SIZE (out) = outsize; /* play */ if ((ret = gst_pad_push (amrnbenc->srcpad, out)) != GST_FLOW_OK) break; } return ret; /* ERRORS */ not_negotiated: { GST_ELEMENT_ERROR (amrnbenc, STREAM, TYPE_NOT_FOUND, (NULL), ("unknown type")); return GST_FLOW_NOT_NEGOTIATED; } } static GstStateChangeReturn gst_amrnbenc_state_change (GstElement * element, GstStateChange transition) { GstAmrnbEnc *amrnbenc; GstStateChangeReturn ret; amrnbenc = GST_AMRNBENC (element); switch (transition) { case GST_STATE_CHANGE_NULL_TO_READY: if (!(amrnbenc->handle = Encoder_Interface_init (0))) return GST_STATE_CHANGE_FAILURE; break; case GST_STATE_CHANGE_READY_TO_PAUSED: amrnbenc->rate = 0; amrnbenc->channels = 0; amrnbenc->ts = 0; gst_adapter_clear (amrnbenc->adapter); break; default: break; } ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); switch (transition) { case GST_STATE_CHANGE_READY_TO_NULL: Encoder_Interface_exit (amrnbenc->handle); break; default: break; } return ret; }