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Diffstat (limited to 'wearable/gst/rtp/gstrtpg722depay.c')
-rw-r--r--wearable/gst/rtp/gstrtpg722depay.c260
1 files changed, 0 insertions, 260 deletions
diff --git a/wearable/gst/rtp/gstrtpg722depay.c b/wearable/gst/rtp/gstrtpg722depay.c
deleted file mode 100644
index 726426b..0000000
--- a/wearable/gst/rtp/gstrtpg722depay.c
+++ /dev/null
@@ -1,260 +0,0 @@
-/* GStreamer
- * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
- *
- * 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., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#include <string.h>
-#include <stdlib.h>
-
-#include <gst/audio/audio.h>
-#include <gst/audio/multichannel.h>
-
-#include "gstrtpg722depay.h"
-#include "gstrtpchannels.h"
-
-GST_DEBUG_CATEGORY_STATIC (rtpg722depay_debug);
-#define GST_CAT_DEFAULT (rtpg722depay_debug)
-
-static GstStaticPadTemplate gst_rtp_g722_depay_src_template =
-GST_STATIC_PAD_TEMPLATE ("src",
- GST_PAD_SRC,
- GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/G722, "
- "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
- );
-
-static GstStaticPadTemplate gst_rtp_g722_depay_sink_template =
- GST_STATIC_PAD_TEMPLATE ("sink",
- GST_PAD_SINK,
- GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("application/x-rtp, "
- "media = (string) \"audio\", "
- "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
- "clock-rate = (int) 8000, "
- /* "channels = (int) [1, MAX]" */
- /* "channel-order = (string) ANY" */
- "encoding-name = (string) \"G722\";"
- "application/x-rtp, "
- "media = (string) \"audio\", "
- "payload = (int) " GST_RTP_PAYLOAD_G722_STRING ", "
- "clock-rate = (int) [ 1, MAX ]"
- /* "channels = (int) [1, MAX]" */
- /* "emphasis = (string) ANY" */
- /* "channel-order = (string) ANY" */
- )
- );
-
-GST_BOILERPLATE (GstRtpG722Depay, gst_rtp_g722_depay, GstBaseRTPDepayload,
- GST_TYPE_BASE_RTP_DEPAYLOAD);
-
-static gboolean gst_rtp_g722_depay_setcaps (GstBaseRTPDepayload * depayload,
- GstCaps * caps);
-static GstBuffer *gst_rtp_g722_depay_process (GstBaseRTPDepayload * depayload,
- GstBuffer * buf);
-
-static void
-gst_rtp_g722_depay_base_init (gpointer klass)
-{
- GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
-
- gst_element_class_add_static_pad_template (element_class,
- &gst_rtp_g722_depay_src_template);
- gst_element_class_add_static_pad_template (element_class,
- &gst_rtp_g722_depay_sink_template);
-
- gst_element_class_set_details_simple (element_class, "RTP audio depayloader",
- "Codec/Depayloader/Network/RTP",
- "Extracts G722 audio from RTP packets",
- "Wim Taymans <wim.taymans@gmail.com>");
-}
-
-static void
-gst_rtp_g722_depay_class_init (GstRtpG722DepayClass * klass)
-{
- GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
-
- gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
-
- gstbasertpdepayload_class->set_caps = gst_rtp_g722_depay_setcaps;
- gstbasertpdepayload_class->process = gst_rtp_g722_depay_process;
-
- GST_DEBUG_CATEGORY_INIT (rtpg722depay_debug, "rtpg722depay", 0,
- "G722 RTP Depayloader");
-}
-
-static void
-gst_rtp_g722_depay_init (GstRtpG722Depay * rtpg722depay,
- GstRtpG722DepayClass * klass)
-{
- /* needed because of GST_BOILERPLATE */
-}
-
-static gint
-gst_rtp_g722_depay_parse_int (GstStructure * structure, const gchar * field,
- gint def)
-{
- const gchar *str;
- gint res;
-
- if ((str = gst_structure_get_string (structure, field)))
- return atoi (str);
-
- if (gst_structure_get_int (structure, field, &res))
- return res;
-
- return def;
-}
-
-static gboolean
-gst_rtp_g722_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
-{
- GstStructure *structure;
- GstRtpG722Depay *rtpg722depay;
- gint clock_rate, payload, samplerate;
- gint channels;
- GstCaps *srccaps;
- gboolean res;
- const gchar *channel_order;
- const GstRTPChannelOrder *order;
-
- rtpg722depay = GST_RTP_G722_DEPAY (depayload);
-
- structure = gst_caps_get_structure (caps, 0);
-
- payload = 96;
- gst_structure_get_int (structure, "payload", &payload);
- switch (payload) {
- case GST_RTP_PAYLOAD_G722:
- channels = 1;
- clock_rate = 8000;
- samplerate = 16000;
- break;
- default:
- /* no fixed mapping, we need clock-rate */
- channels = 0;
- clock_rate = 0;
- samplerate = 0;
- break;
- }
-
- /* caps can overwrite defaults */
- clock_rate =
- gst_rtp_g722_depay_parse_int (structure, "clock-rate", clock_rate);
- if (clock_rate == 0)
- goto no_clockrate;
-
- if (clock_rate == 8000)
- samplerate = 16000;
-
- if (samplerate == 0)
- samplerate = clock_rate;
-
- channels =
- gst_rtp_g722_depay_parse_int (structure, "encoding-params", channels);
- if (channels == 0) {
- channels = gst_rtp_g722_depay_parse_int (structure, "channels", channels);
- if (channels == 0) {
- /* channels defaults to 1 otherwise */
- channels = 1;
- }
- }
-
- depayload->clock_rate = clock_rate;
- rtpg722depay->rate = samplerate;
- rtpg722depay->channels = channels;
-
- srccaps = gst_caps_new_simple ("audio/G722",
- "rate", G_TYPE_INT, samplerate, "channels", G_TYPE_INT, channels, NULL);
-
- /* add channel positions */
- channel_order = gst_structure_get_string (structure, "channel-order");
-
- order = gst_rtp_channels_get_by_order (channels, channel_order);
- if (order) {
- gst_audio_set_channel_positions (gst_caps_get_structure (srccaps, 0),
- order->pos);
- } else {
- GstAudioChannelPosition *pos;
-
- GST_ELEMENT_WARNING (rtpg722depay, STREAM, DECODE,
- (NULL), ("Unknown channel order '%s' for %d channels",
- GST_STR_NULL (channel_order), channels));
- /* create default NONE layout */
- pos = gst_rtp_channels_create_default (channels);
- gst_audio_set_channel_positions (gst_caps_get_structure (srccaps, 0), pos);
- g_free (pos);
- }
-
- res = gst_pad_set_caps (depayload->srcpad, srccaps);
- gst_caps_unref (srccaps);
-
- return res;
-
- /* ERRORS */
-no_clockrate:
- {
- GST_ERROR_OBJECT (depayload, "no clock-rate specified");
- return FALSE;
- }
-}
-
-static GstBuffer *
-gst_rtp_g722_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
-{
- GstRtpG722Depay *rtpg722depay;
- GstBuffer *outbuf;
- gint payload_len;
- gboolean marker;
-
- rtpg722depay = GST_RTP_G722_DEPAY (depayload);
-
- payload_len = gst_rtp_buffer_get_payload_len (buf);
-
- if (payload_len <= 0)
- goto empty_packet;
-
- GST_DEBUG_OBJECT (rtpg722depay, "got payload of %d bytes", payload_len);
-
- outbuf = gst_rtp_buffer_get_payload_buffer (buf);
- marker = gst_rtp_buffer_get_marker (buf);
-
- if (marker && outbuf) {
- /* mark talk spurt with DISCONT */
- GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
- }
-
- return outbuf;
-
- /* ERRORS */
-empty_packet:
- {
- GST_ELEMENT_WARNING (rtpg722depay, STREAM, DECODE,
- ("Empty Payload."), (NULL));
- return NULL;
- }
-}
-
-gboolean
-gst_rtp_g722_depay_plugin_init (GstPlugin * plugin)
-{
- return gst_element_register (plugin, "rtpg722depay",
- GST_RANK_SECONDARY, GST_TYPE_RTP_G722_DEPAY);
-}
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-27 03:43:26 +0000