236 lines
6.2 KiB
C
236 lines
6.2 KiB
C
/* GStreamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-mulawenc
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* @title: mulawenc
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*
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* This element encode mulaw audio. Mulaw coding is also known as G.711.
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <gst/gst.h>
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#include <gst/audio/audio.h>
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#include "mulaw-encode.h"
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#include "mulaw-conversion.h"
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extern GstStaticPadTemplate mulaw_enc_src_factory;
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extern GstStaticPadTemplate mulaw_enc_sink_factory;
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#if G_BYTE_ORDER == G_LITTLE_ENDIAN
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#define INT_FORMAT "S16LE"
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#else
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#define INT_FORMAT "S16BE"
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#endif
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GstStaticPadTemplate mulaw_enc_sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) " INT_FORMAT ", "
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"layout = (string) interleaved, "
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"rate = (int) [ 8000, 192000 ], " "channels = (int) [ 1, 2 ]")
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);
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GstStaticPadTemplate mulaw_enc_src_factory = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-mulaw, "
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"rate = [ 8000 , 192000 ], " "channels = [ 1 , 2 ]")
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);
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/* Stereo signals and args */
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enum
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{
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/* FILL ME */
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LAST_SIGNAL
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};
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enum
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{
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PROP_0
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};
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static gboolean gst_mulawenc_start (GstAudioEncoder * audioenc);
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static gboolean gst_mulawenc_set_format (GstAudioEncoder * enc,
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GstAudioInfo * info);
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static GstFlowReturn gst_mulawenc_handle_frame (GstAudioEncoder * enc,
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GstBuffer * buffer);
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static void gst_mulawenc_set_tags (GstMuLawEnc * mulawenc);
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#define gst_mulawenc_parent_class parent_class
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G_DEFINE_TYPE (GstMuLawEnc, gst_mulawenc, GST_TYPE_AUDIO_ENCODER);
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GST_ELEMENT_REGISTER_DEFINE (mulawenc, "mulawenc", GST_RANK_PRIMARY,
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GST_TYPE_MULAWENC);
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/*static guint gst_stereo_signals[LAST_SIGNAL] = { 0 }; */
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static gboolean
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gst_mulawenc_start (GstAudioEncoder * audioenc)
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{
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GstMuLawEnc *mulawenc = GST_MULAWENC (audioenc);
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mulawenc->channels = 0;
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mulawenc->rate = 0;
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return TRUE;
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}
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static void
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gst_mulawenc_set_tags (GstMuLawEnc * mulawenc)
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{
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GstTagList *taglist;
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guint bitrate;
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/* bitrate of mulaw is 8 bits/sample * sample rate * number of channels */
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bitrate = 8 * mulawenc->rate * mulawenc->channels;
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taglist = gst_tag_list_new_empty ();
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gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE,
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GST_TAG_MAXIMUM_BITRATE, bitrate, NULL);
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gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE,
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GST_TAG_MINIMUM_BITRATE, bitrate, NULL);
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gst_tag_list_add (taglist, GST_TAG_MERGE_REPLACE,
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GST_TAG_BITRATE, bitrate, NULL);
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gst_audio_encoder_merge_tags (GST_AUDIO_ENCODER (mulawenc),
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taglist, GST_TAG_MERGE_REPLACE);
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gst_tag_list_unref (taglist);
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}
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static gboolean
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gst_mulawenc_set_format (GstAudioEncoder * audioenc, GstAudioInfo * info)
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{
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GstCaps *base_caps;
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GstStructure *structure;
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GstMuLawEnc *mulawenc = GST_MULAWENC (audioenc);
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gboolean ret;
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mulawenc->rate = info->rate;
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mulawenc->channels = info->channels;
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base_caps =
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gst_pad_get_pad_template_caps (GST_AUDIO_ENCODER_SRC_PAD (audioenc));
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g_assert (base_caps);
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base_caps = gst_caps_make_writable (base_caps);
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g_assert (base_caps);
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structure = gst_caps_get_structure (base_caps, 0);
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g_assert (structure);
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gst_structure_set (structure, "rate", G_TYPE_INT, mulawenc->rate, NULL);
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gst_structure_set (structure, "channels", G_TYPE_INT, mulawenc->channels,
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NULL);
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gst_mulawenc_set_tags (mulawenc);
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ret = gst_audio_encoder_set_output_format (audioenc, base_caps);
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gst_caps_unref (base_caps);
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return ret;
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}
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static GstFlowReturn
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gst_mulawenc_handle_frame (GstAudioEncoder * audioenc, GstBuffer * buffer)
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{
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GstMuLawEnc *mulawenc;
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GstMapInfo inmap, outmap;
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gint16 *linear_data;
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gsize linear_size;
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guint8 *mulaw_data;
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guint mulaw_size;
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GstBuffer *outbuf;
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GstFlowReturn ret;
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if (!buffer) {
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ret = GST_FLOW_OK;
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goto done;
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}
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mulawenc = GST_MULAWENC (audioenc);
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if (!mulawenc->rate || !mulawenc->channels)
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goto not_negotiated;
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gst_buffer_map (buffer, &inmap, GST_MAP_READ);
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linear_data = (gint16 *) inmap.data;
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linear_size = inmap.size;
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mulaw_size = linear_size / 2;
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outbuf = gst_audio_encoder_allocate_output_buffer (audioenc, mulaw_size);
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g_assert (outbuf);
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gst_buffer_map (outbuf, &outmap, GST_MAP_WRITE);
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mulaw_data = outmap.data;
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mulaw_encode (linear_data, mulaw_data, mulaw_size);
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gst_buffer_unmap (outbuf, &outmap);
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gst_buffer_unmap (buffer, &inmap);
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ret = gst_audio_encoder_finish_frame (audioenc, outbuf, -1);
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done:
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return ret;
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not_negotiated:
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{
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GST_DEBUG_OBJECT (mulawenc, "no format negotiated");
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ret = GST_FLOW_NOT_NEGOTIATED;
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goto done;
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}
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}
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static void
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gst_mulawenc_class_init (GstMuLawEncClass * klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GstAudioEncoderClass *audio_encoder_class = GST_AUDIO_ENCODER_CLASS (klass);
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audio_encoder_class->start = GST_DEBUG_FUNCPTR (gst_mulawenc_start);
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audio_encoder_class->set_format = GST_DEBUG_FUNCPTR (gst_mulawenc_set_format);
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audio_encoder_class->handle_frame =
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GST_DEBUG_FUNCPTR (gst_mulawenc_handle_frame);
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gst_element_class_add_static_pad_template (element_class,
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&mulaw_enc_src_factory);
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gst_element_class_add_static_pad_template (element_class,
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&mulaw_enc_sink_factory);
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gst_element_class_set_static_metadata (element_class, "Mu Law audio encoder",
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"Codec/Encoder/Audio",
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"Convert 16bit PCM to 8bit mu law",
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"Zaheer Abbas Merali <zaheerabbas at merali dot org>");
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}
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static void
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gst_mulawenc_init (GstMuLawEnc * mulawenc)
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{
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GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (mulawenc));
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}
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