| /* |
| * 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); |
| |
| 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_pad_template (element_class, |
| gst_static_pad_template_get (&gsmenc_sink_template)); |
| gst_element_class_add_pad_template (element_class, |
| gst_static_pad_template_get (&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) |
| { |
| } |
| |
| 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 (buffer, &omap); |
| |
| ret = gst_audio_encoder_finish_frame (benc, outbuf, 160); |
| |
| done: |
| return ret; |
| } |