From 29e39080324a4a5a3b4d5bbc3fc79213090c6b0a Mon Sep 17 00:00:00 2001 From: Iago Toral Date: Mon, 27 Jul 2009 19:55:27 +0200 Subject: amrwb: Remove AMR-WB parser and decoder and rename encoder plugin from amrwb to amrwbenc Partially fixes bug #584890. --- ext/amrwbenc/gstamrwbenc.c | 385 +++++++++++++++++++++++++++++++++++++++++++++ 1 file changed, 385 insertions(+) create mode 100644 ext/amrwbenc/gstamrwbenc.c (limited to 'ext/amrwbenc/gstamrwbenc.c') diff --git a/ext/amrwbenc/gstamrwbenc.c b/ext/amrwbenc/gstamrwbenc.c new file mode 100644 index 00000000..89659e53 --- /dev/null +++ b/ext/amrwbenc/gstamrwbenc.c @@ -0,0 +1,385 @@ +/* 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; +} -- cgit v1.2.1