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4a9e80720a
Detected by LLVM's CLang static analyzer
360 lines
10 KiB
C
360 lines
10 KiB
C
/* GStreamer
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* Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
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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., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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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 <string.h>
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#include <gst/audio/audio.h>
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#include <gst/audio/multichannel.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include "gstrtpL16pay.h"
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#include "gstrtpchannels.h"
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GST_DEBUG_CATEGORY_STATIC (rtpL16pay_debug);
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#define GST_CAT_DEFAULT (rtpL16pay_debug)
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/* elementfactory information */
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static const GstElementDetails gst_rtp_L16_pay_details =
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GST_ELEMENT_DETAILS ("RTP audio payloader",
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"Codec/Payloader/Network",
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"Payload-encode Raw audio into RTP packets (RFC 3551)",
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"Wim Taymans <wim.taymans@gmail.com>");
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static GstStaticPadTemplate gst_rtp_L16_pay_sink_template =
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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-int, "
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"endianness = (int) BIG_ENDIAN, "
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"signed = (boolean) true, "
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"width = (int) 16, "
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"depth = (int) 16, "
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"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
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);
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static GstStaticPadTemplate gst_rtp_L16_pay_src_template =
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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 ("application/x-rtp, "
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"media = (string) \"audio\", "
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"payload = (int) [ 96, 127 ], "
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"clock-rate = (int) [ 1, MAX ], "
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"encoding-name = (string) \"L16\", "
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"channels = (int) [ 1, MAX ];"
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"application/x-rtp, "
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"media = (string) \"audio\", "
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"encoding-name = (string) \"L16\", "
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"payload = (int) " GST_RTP_PAYLOAD_L16_STEREO_STRING ", "
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"clock-rate = (int) 44100;"
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"application/x-rtp, "
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"media = (string) \"audio\", "
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"encoding-name = (string) \"L16\", "
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"payload = (int) " GST_RTP_PAYLOAD_L16_MONO_STRING ", "
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"clock-rate = (int) 44100")
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);
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static void gst_rtp_L16_pay_class_init (GstRtpL16PayClass * klass);
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static void gst_rtp_L16_pay_base_init (GstRtpL16PayClass * klass);
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static void gst_rtp_L16_pay_init (GstRtpL16Pay * rtpL16pay);
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static void gst_rtp_L16_pay_finalize (GObject * object);
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static gboolean gst_rtp_L16_pay_setcaps (GstBaseRTPPayload * basepayload,
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GstCaps * caps);
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static GstFlowReturn gst_rtp_L16_pay_handle_buffer (GstBaseRTPPayload * pad,
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GstBuffer * buffer);
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static GstCaps *gst_rtp_L16_pay_getcaps (GstBaseRTPPayload * rtppayload,
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GstPad * pad);
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static GstBaseRTPPayloadClass *parent_class = NULL;
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static GType
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gst_rtp_L16_pay_get_type (void)
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{
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static GType rtpL16pay_type = 0;
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if (!rtpL16pay_type) {
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static const GTypeInfo rtpL16pay_info = {
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sizeof (GstRtpL16PayClass),
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(GBaseInitFunc) gst_rtp_L16_pay_base_init,
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NULL,
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(GClassInitFunc) gst_rtp_L16_pay_class_init,
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NULL,
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NULL,
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sizeof (GstRtpL16Pay),
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0,
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(GInstanceInitFunc) gst_rtp_L16_pay_init,
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};
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rtpL16pay_type =
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g_type_register_static (GST_TYPE_BASE_RTP_PAYLOAD, "GstRtpL16Pay",
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&rtpL16pay_info, 0);
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}
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return rtpL16pay_type;
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}
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static void
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gst_rtp_L16_pay_base_init (GstRtpL16PayClass * klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtp_L16_pay_src_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_rtp_L16_pay_sink_template));
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gst_element_class_set_details (element_class, &gst_rtp_L16_pay_details);
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}
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static void
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gst_rtp_L16_pay_class_init (GstRtpL16PayClass * klass)
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{
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GObjectClass *gobject_class;
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GstBaseRTPPayloadClass *gstbasertppayload_class;
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gobject_class = (GObjectClass *) klass;
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gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
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parent_class = g_type_class_peek_parent (klass);
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gobject_class->finalize = gst_rtp_L16_pay_finalize;
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gstbasertppayload_class->set_caps = gst_rtp_L16_pay_setcaps;
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gstbasertppayload_class->get_caps = gst_rtp_L16_pay_getcaps;
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gstbasertppayload_class->handle_buffer = gst_rtp_L16_pay_handle_buffer;
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GST_DEBUG_CATEGORY_INIT (rtpL16pay_debug, "rtpL16pay", 0,
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"L16 RTP Payloader");
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}
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static void
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gst_rtp_L16_pay_init (GstRtpL16Pay * rtpL16pay)
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{
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rtpL16pay->adapter = gst_adapter_new ();
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}
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static void
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gst_rtp_L16_pay_finalize (GObject * object)
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{
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GstRtpL16Pay *rtpL16pay;
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rtpL16pay = GST_RTP_L16_PAY (object);
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g_object_unref (rtpL16pay->adapter);
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rtpL16pay->adapter = NULL;
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static gboolean
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gst_rtp_L16_pay_setcaps (GstBaseRTPPayload * basepayload, GstCaps * caps)
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{
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GstRtpL16Pay *rtpL16pay;
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GstStructure *structure;
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gint channels, rate;
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gboolean res;
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gchar *params;
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GstAudioChannelPosition *pos;
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const GstRTPChannelOrder *order;
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rtpL16pay = GST_RTP_L16_PAY (basepayload);
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structure = gst_caps_get_structure (caps, 0);
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/* first parse input caps */
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if (!gst_structure_get_int (structure, "rate", &rate))
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goto no_rate;
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if (!gst_structure_get_int (structure, "channels", &channels))
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goto no_channels;
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/* get the channel order */
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pos = gst_audio_get_channel_positions (structure);
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if (pos)
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order = gst_rtp_channels_get_by_pos (channels, pos);
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else
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order = NULL;
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gst_basertppayload_set_options (basepayload, "audio", TRUE, "L16", rate);
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params = g_strdup_printf ("%d", channels);
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if (!order && channels > 2) {
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GST_ELEMENT_WARNING (rtpL16pay, STREAM, DECODE,
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(NULL), ("Unknown channel order for %d channels", channels));
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}
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if (order && order->name) {
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res = gst_basertppayload_set_outcaps (basepayload,
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"encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
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channels, "channel-order", G_TYPE_STRING, order->name, NULL);
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} else {
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res = gst_basertppayload_set_outcaps (basepayload,
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"encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
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channels, NULL);
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}
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g_free (params);
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g_free (pos);
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rtpL16pay->rate = rate;
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rtpL16pay->channels = channels;
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return res;
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/* ERRORS */
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no_rate:
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{
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GST_DEBUG_OBJECT (rtpL16pay, "no rate given");
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return FALSE;
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}
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no_channels:
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{
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GST_DEBUG_OBJECT (rtpL16pay, "no channels given");
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return FALSE;
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}
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}
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static GstFlowReturn
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gst_rtp_L16_pay_flush (GstRtpL16Pay * rtpL16pay, guint len)
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{
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GstBuffer *outbuf;
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guint8 *payload;
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GstFlowReturn ret;
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guint samples;
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GstClockTime duration;
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/* calculate the amount of samples and round down the length */
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samples = len / (2 * rtpL16pay->channels);
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len = samples * (2 * rtpL16pay->channels);
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/* now alloc output buffer */
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outbuf = gst_rtp_buffer_new_allocate (len, 0, 0);
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/* get payload, this is now writable */
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payload = gst_rtp_buffer_get_payload (outbuf);
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/* copy and flush data out of adapter into the RTP payload */
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gst_adapter_copy (rtpL16pay->adapter, payload, 0, len);
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gst_adapter_flush (rtpL16pay->adapter, len);
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duration = gst_util_uint64_scale_int (samples, GST_SECOND, rtpL16pay->rate);
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GST_BUFFER_TIMESTAMP (outbuf) = rtpL16pay->first_ts;
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GST_BUFFER_DURATION (outbuf) = duration;
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/* increase count (in ts) of data pushed to basertppayload */
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if (GST_CLOCK_TIME_IS_VALID (rtpL16pay->first_ts))
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rtpL16pay->first_ts += duration;
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ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpL16pay), outbuf);
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return ret;
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}
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static GstFlowReturn
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gst_rtp_L16_pay_handle_buffer (GstBaseRTPPayload * basepayload,
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GstBuffer * buffer)
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{
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GstRtpL16Pay *rtpL16pay;
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GstFlowReturn ret = GST_FLOW_OK;
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guint payload_len;
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GstClockTime timestamp;
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guint mtu, avail;
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rtpL16pay = GST_RTP_L16_PAY (basepayload);
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mtu = GST_BASE_RTP_PAYLOAD_MTU (rtpL16pay);
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timestamp = GST_BUFFER_TIMESTAMP (buffer);
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if (GST_BUFFER_IS_DISCONT (buffer))
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gst_adapter_clear (rtpL16pay->adapter);
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avail = gst_adapter_available (rtpL16pay->adapter);
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if (avail == 0) {
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rtpL16pay->first_ts = timestamp;
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}
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/* push buffer in adapter */
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gst_adapter_push (rtpL16pay->adapter, buffer);
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/* get payload len for MTU */
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payload_len = gst_rtp_buffer_calc_payload_len (mtu, 0, 0);
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/* flush complete MTU while we have enough data in the adapter */
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while (avail >= payload_len) {
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/* flush payload_len bytes */
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ret = gst_rtp_L16_pay_flush (rtpL16pay, payload_len);
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if (ret != GST_FLOW_OK)
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break;
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avail = gst_adapter_available (rtpL16pay->adapter);
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}
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return ret;
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}
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static GstCaps *
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gst_rtp_L16_pay_getcaps (GstBaseRTPPayload * rtppayload, GstPad * pad)
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{
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GstCaps *otherpadcaps;
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GstCaps *caps;
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otherpadcaps = gst_pad_get_allowed_caps (rtppayload->srcpad);
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caps = gst_caps_copy (gst_pad_get_pad_template_caps (pad));
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if (otherpadcaps) {
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if (!gst_caps_is_empty (otherpadcaps)) {
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GstStructure *structure;
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gint channels;
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gint pt;
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gint rate;
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structure = gst_caps_get_structure (otherpadcaps, 0);
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if (gst_structure_get_int (structure, "channels", &channels)) {
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gst_caps_set_simple (caps, "channels", G_TYPE_INT, channels, NULL);
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} else if (gst_structure_get_int (structure, "payload", &pt)) {
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if (pt == 10)
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gst_caps_set_simple (caps, "channels", G_TYPE_INT, 2, NULL);
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else if (pt == 11)
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gst_caps_set_simple (caps, "channels", G_TYPE_INT, 1, NULL);
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}
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if (gst_structure_get_int (structure, "clock-rate", &rate)) {
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gst_caps_set_simple (caps, "rate", G_TYPE_INT, rate, NULL);
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} else if (gst_structure_get_int (structure, "payload", &pt)) {
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if (pt == 10 || pt == 11)
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gst_caps_set_simple (caps, "rate", G_TYPE_INT, 44100, NULL);
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}
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}
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gst_caps_unref (otherpadcaps);
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}
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return caps;
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}
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gboolean
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gst_rtp_L16_pay_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "rtpL16pay",
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GST_RANK_NONE, GST_TYPE_RTP_L16_PAY);
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}
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