/* * Farsight * GStreamer GSM encoder * Copyright (C) 2005 Philippe Khalaf <burger@speedy.org> * * 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. */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include <string.h> #include "gstgsmenc.h" GST_DEBUG_CATEGORY_STATIC (gsmenc_debug); #define GST_CAT_DEFAULT (gsmenc_debug) /* GSMEnc signals and args */ enum { /* FILL ME */ LAST_SIGNAL }; enum { /* FILL ME */ ARG_0 }; static gboolean gst_gsmenc_start (GstAudioEncoder * enc); static gboolean gst_gsmenc_stop (GstAudioEncoder * enc); static gboolean gst_gsmenc_set_format (GstAudioEncoder * enc, GstAudioInfo * info); static GstFlowReturn gst_gsmenc_handle_frame (GstAudioEncoder * enc, GstBuffer * in_buf); static GstStaticPadTemplate gsmenc_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-gsm, " "rate = (int) 8000, " "channels = (int) 1") ); static GstStaticPadTemplate gsmenc_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") ); G_DEFINE_TYPE (GstGSMEnc, gst_gsmenc, GST_TYPE_AUDIO_ENCODER); GST_ELEMENT_REGISTER_DEFINE (gsmenc, "gsmenc", GST_RANK_PRIMARY, GST_TYPE_GSMENC); static void gst_gsmenc_class_init (GstGSMEncClass * klass) { GstElementClass *element_class; GstAudioEncoderClass *base_class; element_class = (GstElementClass *) klass; base_class = (GstAudioEncoderClass *) klass; gst_element_class_add_static_pad_template (element_class, &gsmenc_sink_template); gst_element_class_add_static_pad_template (element_class, &gsmenc_src_template); gst_element_class_set_static_metadata (element_class, "GSM audio encoder", "Codec/Encoder/Audio", "Encodes GSM audio", "Philippe Khalaf <burger@speedy.org>"); base_class->start = GST_DEBUG_FUNCPTR (gst_gsmenc_start); base_class->stop = GST_DEBUG_FUNCPTR (gst_gsmenc_stop); base_class->set_format = GST_DEBUG_FUNCPTR (gst_gsmenc_set_format); base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_gsmenc_handle_frame); GST_DEBUG_CATEGORY_INIT (gsmenc_debug, "gsmenc", 0, "GSM Encoder"); } static void gst_gsmenc_init (GstGSMEnc * gsmenc) { GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (gsmenc)); } static gboolean gst_gsmenc_start (GstAudioEncoder * enc) { GstGSMEnc *gsmenc = GST_GSMENC (enc); gint use_wav49; GST_DEBUG_OBJECT (enc, "start"); gsmenc->state = gsm_create (); /* turn off WAV49 handling */ use_wav49 = 0; gsm_option (gsmenc->state, GSM_OPT_WAV49, &use_wav49); return TRUE; } static gboolean gst_gsmenc_stop (GstAudioEncoder * enc) { GstGSMEnc *gsmenc = GST_GSMENC (enc); GST_DEBUG_OBJECT (enc, "stop"); gsm_destroy (gsmenc->state); return TRUE; } static gboolean gst_gsmenc_set_format (GstAudioEncoder * benc, GstAudioInfo * info) { GstCaps *srccaps; srccaps = gst_static_pad_template_get_caps (&gsmenc_src_template); gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (benc), srccaps); gst_caps_unref (srccaps); /* report needs to base class */ gst_audio_encoder_set_frame_samples_min (benc, 160); gst_audio_encoder_set_frame_samples_max (benc, 160); gst_audio_encoder_set_frame_max (benc, 1); return TRUE; } static GstFlowReturn gst_gsmenc_handle_frame (GstAudioEncoder * benc, GstBuffer * buffer) { GstGSMEnc *gsmenc; gsm_signal *data; GstFlowReturn ret = GST_FLOW_OK; GstBuffer *outbuf; GstMapInfo map, omap; gsmenc = GST_GSMENC (benc); /* we don't deal with squeezing remnants, so simply discard those */ if (G_UNLIKELY (buffer == NULL)) { GST_DEBUG_OBJECT (gsmenc, "no data"); goto done; } gst_buffer_map (buffer, &map, GST_MAP_READ); if (G_UNLIKELY (map.size < 320)) { GST_DEBUG_OBJECT (gsmenc, "discarding trailing data %d", (gint) map.size); gst_buffer_unmap (buffer, &map); ret = gst_audio_encoder_finish_frame (benc, NULL, -1); goto done; } outbuf = gst_buffer_new_and_alloc (33 * sizeof (gsm_byte)); gst_buffer_map (outbuf, &omap, GST_MAP_WRITE); /* encode 160 16-bit samples into 33 bytes */ data = (gsm_signal *) map.data; gsm_encode (gsmenc->state, data, (gsm_byte *) omap.data); GST_LOG_OBJECT (gsmenc, "encoded to %d bytes", (gint) omap.size); gst_buffer_unmap (buffer, &map); gst_buffer_unmap (outbuf, &omap); ret = gst_audio_encoder_finish_frame (benc, outbuf, 160); done: return ret; }