/* 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-amrnbenc * @see_also: #GstAmrnbDec, #GstAmrnbParse * * AMR narrowband encoder based on the * opencore codec implementation. * * * Example launch line * |[ * gst-launch-1.0 filesrc location=abc.wav ! wavparse ! audioconvert ! audioresample ! amrnbenc ! filesink location=abc.amr * ]| * Please note 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 (void) { 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, format = (string) " GST_AUDIO_NE (S16) ", " "layout = (string) interleaved, " "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 gboolean gst_amrnbenc_start (GstAudioEncoder * enc); static gboolean gst_amrnbenc_stop (GstAudioEncoder * enc); static gboolean gst_amrnbenc_set_format (GstAudioEncoder * enc, GstAudioInfo * info); static GstFlowReturn gst_amrnbenc_handle_frame (GstAudioEncoder * enc, GstBuffer * in_buf); #define gst_amrnbenc_parent_class parent_class G_DEFINE_TYPE (GstAmrnbEnc, gst_amrnbenc, GST_TYPE_AUDIO_ENCODER); 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_class_init (GstAmrnbEncClass * klass) { GObjectClass *object_class = G_OBJECT_CLASS (klass); GstElementClass *element_class = GST_ELEMENT_CLASS (klass); GstAudioEncoderClass *base_class = GST_AUDIO_ENCODER_CLASS (klass); object_class->set_property = gst_amrnbenc_set_property; object_class->get_property = gst_amrnbenc_get_property; base_class->start = GST_DEBUG_FUNCPTR (gst_amrnbenc_start); base_class->stop = GST_DEBUG_FUNCPTR (gst_amrnbenc_stop); base_class->set_format = GST_DEBUG_FUNCPTR (gst_amrnbenc_set_format); base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_amrnbenc_handle_frame); 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 | G_PARAM_STATIC_STRINGS)); 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-NB audio encoder", "Codec/Encoder/Audio", "Adaptive Multi-Rate Narrow-Band audio encoder", "Wim Taymans "); GST_DEBUG_CATEGORY_INIT (gst_amrnbenc_debug, "amrnbenc", 0, "AMR-NB audio encoder"); } static void gst_amrnbenc_init (GstAmrnbEnc * amrnbenc) { GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (amrnbenc)); } static gboolean gst_amrnbenc_start (GstAudioEncoder * enc) { GstAmrnbEnc *amrnbenc = GST_AMRNBENC (enc); GST_DEBUG_OBJECT (amrnbenc, "start"); if (!(amrnbenc->handle = Encoder_Interface_init (0))) return FALSE; return TRUE; } static gboolean gst_amrnbenc_stop (GstAudioEncoder * enc) { GstAmrnbEnc *amrnbenc = GST_AMRNBENC (enc); GST_DEBUG_OBJECT (amrnbenc, "stop"); Encoder_Interface_exit (amrnbenc->handle); return TRUE; } static gboolean gst_amrnbenc_set_format (GstAudioEncoder * enc, GstAudioInfo * info) { GstAmrnbEnc *amrnbenc; GstCaps *copy; amrnbenc = GST_AMRNBENC (enc); /* parameters already parsed for us */ amrnbenc->rate = GST_AUDIO_INFO_RATE (info); amrnbenc->channels = GST_AUDIO_INFO_CHANNELS (info); /* we do not really accept other input, but anyway ... */ /* 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); gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (amrnbenc), copy); gst_caps_unref (copy); /* report needs to base class: hand one frame at a time */ gst_audio_encoder_set_frame_samples_min (enc, 160); gst_audio_encoder_set_frame_samples_max (enc, 160); gst_audio_encoder_set_frame_max (enc, 1); return TRUE; } static GstFlowReturn gst_amrnbenc_handle_frame (GstAudioEncoder * enc, GstBuffer * buffer) { GstAmrnbEnc *amrnbenc; GstFlowReturn ret; GstBuffer *out; GstMapInfo in_map, out_map; gsize out_size; amrnbenc = GST_AMRNBENC (enc); g_return_val_if_fail (amrnbenc->handle, GST_FLOW_FLUSHING); /* we don't deal with squeezing remnants, so simply discard those */ if (G_UNLIKELY (buffer == NULL)) { GST_DEBUG_OBJECT (amrnbenc, "no data"); return GST_FLOW_OK; } gst_buffer_map (buffer, &in_map, GST_MAP_READ); if (G_UNLIKELY (in_map.size < 320)) { gst_buffer_unmap (buffer, &in_map); GST_DEBUG_OBJECT (amrnbenc, "discarding trailing data of %" G_GSIZE_FORMAT " bytes", in_map.size); return gst_audio_encoder_finish_frame (enc, NULL, -1); } /* get output, max size is 32 */ out = gst_buffer_new_and_alloc (32); /* AMR encoder actually writes into the source data buffers it gets */ /* should be able to handle that with what we are given */ gst_buffer_map (out, &out_map, GST_MAP_WRITE); /* encode */ out_size = Encoder_Interface_Encode (amrnbenc->handle, amrnbenc->bandmode, (short *) in_map.data, out_map.data, 0); gst_buffer_unmap (out, &out_map); gst_buffer_resize (out, 0, out_size); gst_buffer_unmap (buffer, &in_map); GST_LOG_OBJECT (amrnbenc, "output data size %" G_GSIZE_FORMAT, out_size); if (out_size) { ret = gst_audio_encoder_finish_frame (enc, out, 160); } else { /* should not happen (without dtx or so at least) */ GST_WARNING_OBJECT (amrnbenc, "no encoded data; discarding input"); gst_buffer_unref (out); ret = gst_audio_encoder_finish_frame (enc, NULL, -1); } return ret; }