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-rw-r--r--gst/rtp/gstrtpmp4adepay.c367
1 files changed, 367 insertions, 0 deletions
diff --git a/gst/rtp/gstrtpmp4adepay.c b/gst/rtp/gstrtpmp4adepay.c
new file mode 100644
index 00000000..395db196
--- /dev/null
+++ b/gst/rtp/gstrtpmp4adepay.c
@@ -0,0 +1,367 @@
+/* GStreamer
+ * Copyright (C) <2007> Nokia Corporation (contact <stefan.kost@nokia.com>)
+ * <2007> Wim Taymans <wim@fluendo.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License version 2 as published by the Free Software Foundation.
+ *
+ * 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 <gst/rtp/gstrtpbuffer.h>
+
+#include <string.h>
+#include "gstrtpmp4adepay.h"
+
+GST_DEBUG_CATEGORY_STATIC (rtpmp4adepay_debug);
+#define GST_CAT_DEFAULT (rtpmp4adepay_debug)
+
+/* elementfactory information */
+static const GstElementDetails gst_rtp_mp4adepay_details =
+GST_ELEMENT_DETAILS ("RTP packet parser",
+ "Codec/Depayloader/Network",
+ "Extracts MPEG4 audio from RTP packets (RFC 3016)",
+ "Nokia Corporation (contact <stefan.kost@nokia.com>), "
+ "Wim Taymans <wim@fluendo.com>");
+
+/* RtpMP4ADepay signals and args */
+enum
+{
+ LAST_SIGNAL
+};
+
+enum
+{
+ PROP_0,
+};
+
+static GstStaticPadTemplate gst_rtp_mp4a_depay_src_template =
+GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/mpeg,"
+ "mpegversion = (int) 4," "framed = (boolean) false")
+ );
+
+static GstStaticPadTemplate gst_rtp_mp4a_depay_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("application/x-rtp, "
+ "media = (string) \"audio\", "
+ "clock-rate = (int) [1, MAX ], "
+ "encoding-name = (string) \"MP4A-LATM\""
+ /* All optional parameters
+ *
+ * "profile-level-id=[1,MAX]"
+ * "config="
+ */
+ )
+ );
+
+GST_BOILERPLATE (GstRtpMP4ADepay, gst_rtp_mp4a_depay, GstBaseRTPDepayload,
+ GST_TYPE_BASE_RTP_DEPAYLOAD);
+
+static void gst_rtp_mp4a_depay_finalize (GObject * object);
+
+static gboolean gst_rtp_mp4a_depay_setcaps (GstBaseRTPDepayload * depayload,
+ GstCaps * caps);
+static GstBuffer *gst_rtp_mp4a_depay_process (GstBaseRTPDepayload * depayload,
+ GstBuffer * buf);
+
+static void gst_rtp_mp4a_depay_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec);
+static void gst_rtp_mp4a_depay_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec);
+
+static GstStateChangeReturn gst_rtp_mp4a_depay_change_state (GstElement *
+ element, GstStateChange transition);
+
+
+static void
+gst_rtp_mp4a_depay_base_init (gpointer klass)
+{
+ GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&gst_rtp_mp4a_depay_src_template));
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&gst_rtp_mp4a_depay_sink_template));
+
+ gst_element_class_set_details (element_class, &gst_rtp_mp4adepay_details);
+}
+
+static void
+gst_rtp_mp4a_depay_class_init (GstRtpMP4ADepayClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstElementClass *gstelement_class;
+ GstBaseRTPDepayloadClass *gstbasertpdepayload_class;
+
+ gobject_class = (GObjectClass *) klass;
+ gstelement_class = (GstElementClass *) klass;
+ gstbasertpdepayload_class = (GstBaseRTPDepayloadClass *) klass;
+
+ gobject_class->finalize = gst_rtp_mp4a_depay_finalize;
+ gobject_class->set_property = gst_rtp_mp4a_depay_set_property;
+ gobject_class->get_property = gst_rtp_mp4a_depay_get_property;
+
+ gstelement_class->change_state = gst_rtp_mp4a_depay_change_state;
+
+ gstbasertpdepayload_class->process = gst_rtp_mp4a_depay_process;
+ gstbasertpdepayload_class->set_caps = gst_rtp_mp4a_depay_setcaps;
+
+ GST_DEBUG_CATEGORY_INIT (rtpmp4adepay_debug, "rtpmp4adepay", 0,
+ "MPEG4 audio RTP Depayloader");
+}
+
+static void
+gst_rtp_mp4a_depay_init (GstRtpMP4ADepay * rtpmp4adepay,
+ GstRtpMP4ADepayClass * klass)
+{
+ rtpmp4adepay->adapter = gst_adapter_new ();
+}
+
+static void
+gst_rtp_mp4a_depay_finalize (GObject * object)
+{
+ GstRtpMP4ADepay *rtpmp4adepay;
+
+ rtpmp4adepay = GST_RTP_MP4A_DEPAY (object);
+
+ g_object_unref (rtpmp4adepay->adapter);
+ rtpmp4adepay->adapter = NULL;
+
+ G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+
+static gboolean
+gst_rtp_mp4a_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
+{
+ GstStructure *structure;
+ GstRtpMP4ADepay *rtpmp4adepay;
+ GstCaps *srccaps;
+ const gchar *str;
+ gint clock_rate = 90000; /* default */
+ gint object_type = 2; /* AAC LC default */
+ gint channels = 2; /* default */
+
+ rtpmp4adepay = GST_RTP_MP4A_DEPAY (depayload);
+
+ structure = gst_caps_get_structure (caps, 0);
+
+ if (gst_structure_has_field (structure, "clock-rate"))
+ gst_structure_get_int (structure, "clock-rate", &clock_rate);
+ depayload->clock_rate = clock_rate;
+
+ if (gst_structure_has_field (structure, "object"))
+ gst_structure_get_int (structure, "object", &object_type);
+
+ srccaps = gst_caps_new_simple ("audio/mpeg",
+ "mpegversion", G_TYPE_INT, 4,
+ "framed", G_TYPE_BOOLEAN, FALSE, "channels", G_TYPE_INT, channels, NULL);
+
+ if ((str = gst_structure_get_string (structure, "config"))) {
+ GValue v = { 0 };
+
+ g_value_init (&v, GST_TYPE_BUFFER);
+ if (gst_value_deserialize (&v, str)) {
+ GstBuffer *buffer;
+ guint8 *data;
+ guint size;
+ gint i;
+
+ buffer = gst_value_get_buffer (&v);
+ gst_buffer_ref (buffer);
+ g_value_unset (&v);
+
+ data = GST_BUFFER_DATA (buffer);
+ size = GST_BUFFER_SIZE (buffer);
+ if (size < 2) {
+ GST_WARNING_OBJECT (depayload, "config too short (%d < 2)", size);
+ goto bad_config;
+ }
+
+ /* Parse StreamMuxConfig according to ISO/IEC 14496-3:
+ *
+ * audioMuxVersion == 0 (1 bit)
+ * allStreamsSameTimeFraming == 1 (1 bit)
+ * numSubFrames == 0 (6 bits)
+ * numProgram == 0 (4 bits)
+ * numLayer == 0 (3 bits)
+ *
+ * We only require audioMuxVersion == 0;
+ *
+ * The remaining bit of the second byte and the rest of the bits are used
+ * for audioSpecificConfig which we need to set in codec_info.
+ */
+ if ((data[0] & 0x80) != 0x00) {
+ GST_WARNING_OBJECT (depayload, "unknown audioMuxVersion 1");
+ goto bad_config;
+ }
+
+ /* shift rest of string 15 bits down */
+ size -= 2;
+ for (i = 0; i < size; i++) {
+ data[i] = ((data[i + 1] & 1) << 7) | ((data[i + 2] & 0xfe) >> 1);
+ }
+ /* last bit, this is probably not needed. */
+ data[i] = ((data[i + 1] & 1) << 7);
+ GST_BUFFER_SIZE (buffer) = size + 1;
+
+ gst_caps_set_simple (srccaps,
+ "codec_data", GST_TYPE_BUFFER, buffer, NULL);
+ gst_buffer_unref (buffer);
+ } else {
+ g_warning ("cannot convert config to buffer");
+ }
+ }
+bad_config:
+ gst_pad_set_caps (depayload->srcpad, srccaps);
+ gst_caps_unref (srccaps);
+
+ return TRUE;
+}
+
+static GstBuffer *
+gst_rtp_mp4a_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
+{
+ GstRtpMP4ADepay *rtpmp4adepay;
+ GstBuffer *outbuf;
+
+ rtpmp4adepay = GST_RTP_MP4A_DEPAY (depayload);
+
+ if (!gst_rtp_buffer_validate (buf))
+ goto bad_packet;
+
+ /* flush remaining data on discont */
+ if (GST_BUFFER_IS_DISCONT (buf))
+ gst_adapter_clear (rtpmp4adepay->adapter);
+
+ outbuf = gst_rtp_buffer_get_payload_buffer (buf);
+
+ gst_adapter_push (rtpmp4adepay->adapter, outbuf);
+
+ /* RTP marker bit indicates the last packet of the AudioMuxElement => create
+ * and push a buffer */
+ if (gst_rtp_buffer_get_marker (buf)) {
+ guint avail;
+ guint latm_header_len;
+ guint data_len;
+ guint8 *data;
+
+ avail = gst_adapter_available (rtpmp4adepay->adapter);
+
+ outbuf = gst_adapter_take_buffer (rtpmp4adepay->adapter, avail);
+
+ /* determine payload length and set buffer data pointer accordingly */
+ /* FIXME, check for overrun */
+ latm_header_len = 0;
+ data_len = 0;
+ data = GST_BUFFER_DATA (outbuf);
+ do {
+ data_len += data[latm_header_len];
+ } while (data[latm_header_len++] == 0xff);
+
+ /* just a check that lengths match, possibly there can be more than one
+ * audioMuxElement in the payload? */
+ if ((data_len + latm_header_len) != avail) {
+ GST_WARNING_OBJECT (depayload, "not all payload consumed");
+ }
+
+ GST_BUFFER_SIZE (outbuf) = avail - latm_header_len;
+ GST_BUFFER_DATA (outbuf) += latm_header_len;
+
+ gst_buffer_set_caps (outbuf, GST_PAD_CAPS (depayload->srcpad));
+
+ GST_DEBUG ("gst_rtp_mp4a_depay_process: pushing buffer of size %d",
+ GST_BUFFER_SIZE (outbuf));
+
+ return outbuf;
+ }
+ return NULL;
+
+bad_packet:
+ {
+ GST_ELEMENT_WARNING (rtpmp4adepay, STREAM, DECODE,
+ ("Packet did not validate"), (NULL));
+ return NULL;
+ }
+}
+
+static void
+gst_rtp_mp4a_depay_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstRtpMP4ADepay *rtpmp4adepay;
+
+ rtpmp4adepay = GST_RTP_MP4A_DEPAY (object);
+
+ switch (prop_id) {
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static void
+gst_rtp_mp4a_depay_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec)
+{
+ GstRtpMP4ADepay *rtpmp4adepay;
+
+ rtpmp4adepay = GST_RTP_MP4A_DEPAY (object);
+
+ switch (prop_id) {
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static GstStateChangeReturn
+gst_rtp_mp4a_depay_change_state (GstElement * element,
+ GstStateChange transition)
+{
+ GstRtpMP4ADepay *rtpmp4adepay;
+ GstStateChangeReturn ret;
+
+ rtpmp4adepay = GST_RTP_MP4A_DEPAY (element);
+
+ switch (transition) {
+ case GST_STATE_CHANGE_READY_TO_PAUSED:
+ gst_adapter_clear (rtpmp4adepay->adapter);
+ break;
+ default:
+ break;
+ }
+
+ ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+
+ switch (transition) {
+ default:
+ break;
+ }
+ return ret;
+}
+
+gboolean
+gst_rtp_mp4a_depay_plugin_init (GstPlugin * plugin)
+{
+ return gst_element_register (plugin, "rtpmp4adepay",
+ GST_RANK_NONE, GST_TYPE_RTP_MP4A_DEPAY);
+}
generated by cgit (git 2.34.1) at 2024-09-26 22:07:33 +0000