/* * 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 "gstgsmdec.h" GST_DEBUG_CATEGORY_STATIC (gsmdec_debug); #define GST_CAT_DEFAULT (gsmdec_debug) /* GSMDec signals and args */ enum { /* FILL ME */ LAST_SIGNAL }; enum { /* FILL ME */ ARG_0 }; static gboolean gst_gsmdec_start (GstAudioDecoder * dec); static gboolean gst_gsmdec_stop (GstAudioDecoder * dec); static gboolean gst_gsmdec_set_format (GstAudioDecoder * dec, GstCaps * caps); static GstFlowReturn gst_gsmdec_parse (GstAudioDecoder * dec, GstAdapter * adapter, gint * offset, gint * length); static GstFlowReturn gst_gsmdec_handle_frame (GstAudioDecoder * dec, GstBuffer * in_buf); /*static guint gst_gsmdec_signals[LAST_SIGNAL] = { 0 }; */ #define ENCODED_SAMPLES 160 static GstStaticPadTemplate gsmdec_sink_template = GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-gsm, rate = (int) 8000, channels = (int) 1; " "audio/ms-gsm, rate = (int) [1, MAX], channels = (int) 1") ); static GstStaticPadTemplate gsmdec_src_template = GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw, " "format = (string) " GST_AUDIO_NE (S16) ", " "layout = (string) interleaved, " "rate = (int) [1, MAX], channels = (int) 1") ); G_DEFINE_TYPE (GstGSMDec, gst_gsmdec, GST_TYPE_AUDIO_DECODER); static void gst_gsmdec_class_init (GstGSMDecClass * klass) { GstElementClass *element_class; GstAudioDecoderClass *base_class; element_class = (GstElementClass *) klass; base_class = (GstAudioDecoderClass *) klass; gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gsmdec_sink_template)); gst_element_class_add_pad_template (element_class, gst_static_pad_template_get (&gsmdec_src_template)); gst_element_class_set_static_metadata (element_class, "GSM audio decoder", "Codec/Decoder/Audio", "Decodes GSM encoded audio", "Philippe Khalaf <burger@speedy.org>"); base_class->start = GST_DEBUG_FUNCPTR (gst_gsmdec_start); base_class->stop = GST_DEBUG_FUNCPTR (gst_gsmdec_stop); base_class->set_format = GST_DEBUG_FUNCPTR (gst_gsmdec_set_format); base_class->parse = GST_DEBUG_FUNCPTR (gst_gsmdec_parse); base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_gsmdec_handle_frame); GST_DEBUG_CATEGORY_INIT (gsmdec_debug, "gsmdec", 0, "GSM Decoder"); } static void gst_gsmdec_init (GstGSMDec * gsmdec) { gst_audio_decoder_set_needs_format (GST_AUDIO_DECODER (gsmdec), TRUE); gst_audio_decoder_set_use_default_pad_acceptcaps (GST_AUDIO_DECODER_CAST (gsmdec), TRUE); GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_DECODER_SINK_PAD (gsmdec)); } static gboolean gst_gsmdec_start (GstAudioDecoder * dec) { GstGSMDec *gsmdec = GST_GSMDEC (dec); GST_DEBUG_OBJECT (dec, "start"); gsmdec->state = gsm_create (); return TRUE; } static gboolean gst_gsmdec_stop (GstAudioDecoder * dec) { GstGSMDec *gsmdec = GST_GSMDEC (dec); GST_DEBUG_OBJECT (dec, "stop"); gsm_destroy (gsmdec->state); return TRUE; } static gboolean gst_gsmdec_set_format (GstAudioDecoder * dec, GstCaps * caps) { GstGSMDec *gsmdec; GstStructure *s; gboolean ret = FALSE; gint rate; GstAudioInfo info; gsmdec = GST_GSMDEC (dec); s = gst_caps_get_structure (caps, 0); if (s == NULL) goto wrong_caps; /* figure out if we deal with plain or MSGSM */ if (gst_structure_has_name (s, "audio/x-gsm")) gsmdec->use_wav49 = 0; else if (gst_structure_has_name (s, "audio/ms-gsm")) gsmdec->use_wav49 = 1; else goto wrong_caps; gsmdec->needed = 33; if (!gst_structure_get_int (s, "rate", &rate)) { GST_WARNING_OBJECT (gsmdec, "missing sample rate parameter from sink caps"); goto beach; } /* MSGSM needs different framing */ gsm_option (gsmdec->state, GSM_OPT_WAV49, &gsmdec->use_wav49); /* Setting up src caps based on the input sample rate. */ gst_audio_info_init (&info); gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16, rate, 1, NULL); ret = gst_audio_decoder_set_output_format (dec, &info); return ret; /* ERRORS */ wrong_caps: GST_ERROR_OBJECT (gsmdec, "invalid caps received"); beach: return ret; } static GstFlowReturn gst_gsmdec_parse (GstAudioDecoder * dec, GstAdapter * adapter, gint * offset, gint * length) { GstGSMDec *gsmdec = GST_GSMDEC (dec); guint size; size = gst_adapter_available (adapter); /* if input format is TIME each buffer should be self-contained and * the data is presumably packetised, and we should start with a clean * slate/state at the beginning of each buffer (for wav49 case) */ if (dec->input_segment.format == GST_FORMAT_TIME) { *offset = 0; *length = size; gsmdec->needed = 33; return GST_FLOW_OK; } g_return_val_if_fail (size > 0, GST_FLOW_ERROR); if (size < gsmdec->needed) return GST_FLOW_EOS; *offset = 0; *length = gsmdec->needed; /* WAV49 requires alternating 33 and 32 bytes of input */ if (gsmdec->use_wav49) { gsmdec->needed = (gsmdec->needed == 33 ? 32 : 33); } return GST_FLOW_OK; } static guint gst_gsmdec_get_frame_count (GstGSMDec * dec, gsize buffer_size) { guint count; if (dec->use_wav49) { count = (buffer_size / (33 + 32)) * 2; if (buffer_size % (33 + 32) >= dec->needed) ++count; } else { count = buffer_size / 33; } return count; } static GstFlowReturn gst_gsmdec_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer) { GstGSMDec *gsmdec; gsm_signal *out_data; gsm_byte *data; GstFlowReturn ret = GST_FLOW_OK; GstBuffer *outbuf; GstMapInfo map, omap; gsize outsize; guint frames, i, errors = 0; /* no fancy draining */ if (G_UNLIKELY (!buffer)) return GST_FLOW_OK; gsmdec = GST_GSMDEC (dec); gst_buffer_map (buffer, &map, GST_MAP_READ); frames = gst_gsmdec_get_frame_count (gsmdec, map.size); /* always the same amount of output samples (20ms worth per frame) */ outsize = ENCODED_SAMPLES * frames * sizeof (gsm_signal); outbuf = gst_buffer_new_and_alloc (outsize); gst_buffer_map (outbuf, &omap, GST_MAP_WRITE); out_data = (gsm_signal *) omap.data; data = (gsm_byte *) map.data; for (i = 0; i < frames; ++i) { /* now encode frame into the output buffer */ if (gsm_decode (gsmdec->state, data, out_data) < 0) { /* invalid frame */ GST_AUDIO_DECODER_ERROR (gsmdec, 1, STREAM, DECODE, (NULL), ("tried to decode an invalid frame"), ret); memset (out_data, 0, ENCODED_SAMPLES * sizeof (gsm_signal)); ++errors; } out_data += ENCODED_SAMPLES; data += gsmdec->needed; if (gsmdec->use_wav49) gsmdec->needed = (gsmdec->needed == 33 ? 32 : 33); } gst_buffer_unmap (outbuf, &omap); gst_buffer_unmap (buffer, &map); if (errors == frames) { gst_buffer_unref (outbuf); outbuf = NULL; } gst_audio_decoder_finish_frame (dec, outbuf, 1); return ret; }