summaryrefslogtreecommitdiff
path: root/subprojects/gst-plugins-good/gst/rtp/gstrtpceltdepay.c
diff options
context:
space:
mode:
Diffstat (limited to 'subprojects/gst-plugins-good/gst/rtp/gstrtpceltdepay.c')
-rw-r--r--subprojects/gst-plugins-good/gst/rtp/gstrtpceltdepay.c271
1 files changed, 271 insertions, 0 deletions
diff --git a/subprojects/gst-plugins-good/gst/rtp/gstrtpceltdepay.c b/subprojects/gst-plugins-good/gst/rtp/gstrtpceltdepay.c
new file mode 100644
index 0000000000..9054af7c66
--- /dev/null
+++ b/subprojects/gst-plugins-good/gst/rtp/gstrtpceltdepay.c
@@ -0,0 +1,271 @@
+/* GStreamer
+ * Copyright (C) <2009> 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., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+# include "config.h"
+#endif
+
+#include <string.h>
+#include <stdlib.h>
+#include <gst/rtp/gstrtpbuffer.h>
+#include <gst/audio/audio.h>
+
+#include "gstrtpelements.h"
+#include "gstrtpceltdepay.h"
+#include "gstrtputils.h"
+
+GST_DEBUG_CATEGORY_STATIC (rtpceltdepay_debug);
+#define GST_CAT_DEFAULT (rtpceltdepay_debug)
+
+/* RtpCELTDepay signals and args */
+
+#define DEFAULT_FRAMESIZE 480
+#define DEFAULT_CHANNELS 1
+#define DEFAULT_CLOCKRATE 32000
+
+enum
+{
+ /* FILL ME */
+ LAST_SIGNAL
+};
+
+enum
+{
+ PROP_0
+};
+
+static GstStaticPadTemplate gst_rtp_celt_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) [32000, 48000], "
+ "encoding-name = (string) \"CELT\"")
+ );
+
+static GstStaticPadTemplate gst_rtp_celt_depay_src_template =
+GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-celt")
+ );
+
+static GstBuffer *gst_rtp_celt_depay_process (GstRTPBaseDepayload * depayload,
+ GstRTPBuffer * rtp);
+static gboolean gst_rtp_celt_depay_setcaps (GstRTPBaseDepayload * depayload,
+ GstCaps * caps);
+
+#define gst_rtp_celt_depay_parent_class parent_class
+G_DEFINE_TYPE (GstRtpCELTDepay, gst_rtp_celt_depay,
+ GST_TYPE_RTP_BASE_DEPAYLOAD);
+GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtpceltdepay, "rtpceltdepay",
+ GST_RANK_SECONDARY, GST_TYPE_RTP_CELT_DEPAY, rtp_element_init (plugin));
+static void
+gst_rtp_celt_depay_class_init (GstRtpCELTDepayClass * klass)
+{
+ GstElementClass *gstelement_class;
+ GstRTPBaseDepayloadClass *gstrtpbasedepayload_class;
+
+ GST_DEBUG_CATEGORY_INIT (rtpceltdepay_debug, "rtpceltdepay", 0,
+ "CELT RTP Depayloader");
+
+ gstelement_class = (GstElementClass *) klass;
+ gstrtpbasedepayload_class = (GstRTPBaseDepayloadClass *) klass;
+
+ gst_element_class_add_static_pad_template (gstelement_class,
+ &gst_rtp_celt_depay_src_template);
+ gst_element_class_add_static_pad_template (gstelement_class,
+ &gst_rtp_celt_depay_sink_template);
+
+ gst_element_class_set_static_metadata (gstelement_class,
+ "RTP CELT depayloader", "Codec/Depayloader/Network/RTP",
+ "Extracts CELT audio from RTP packets",
+ "Wim Taymans <wim.taymans@gmail.com>");
+
+ gstrtpbasedepayload_class->process_rtp_packet = gst_rtp_celt_depay_process;
+ gstrtpbasedepayload_class->set_caps = gst_rtp_celt_depay_setcaps;
+}
+
+static void
+gst_rtp_celt_depay_init (GstRtpCELTDepay * rtpceltdepay)
+{
+}
+
+/* len 4 bytes LE,
+ * vendor string (len bytes),
+ * user_len 4 (0) bytes LE
+ */
+static const gchar gst_rtp_celt_comment[] =
+ "\045\0\0\0Depayloaded with GStreamer celtdepay\0\0\0\0";
+
+static gboolean
+gst_rtp_celt_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
+{
+ GstStructure *structure;
+ GstRtpCELTDepay *rtpceltdepay;
+ gint clock_rate, nb_channels = 0, frame_size = 0;
+ GstBuffer *buf;
+ GstMapInfo map;
+ guint8 *ptr;
+ const gchar *params;
+ GstCaps *srccaps;
+ gboolean res;
+
+ rtpceltdepay = GST_RTP_CELT_DEPAY (depayload);
+
+ structure = gst_caps_get_structure (caps, 0);
+
+ if (!gst_structure_get_int (structure, "clock-rate", &clock_rate))
+ goto no_clockrate;
+ depayload->clock_rate = clock_rate;
+
+ if ((params = gst_structure_get_string (structure, "encoding-params")))
+ nb_channels = atoi (params);
+ if (!nb_channels)
+ nb_channels = DEFAULT_CHANNELS;
+
+ if ((params = gst_structure_get_string (structure, "frame-size")))
+ frame_size = atoi (params);
+ if (!frame_size)
+ frame_size = DEFAULT_FRAMESIZE;
+ rtpceltdepay->frame_size = frame_size;
+
+ GST_DEBUG_OBJECT (depayload, "clock-rate=%d channels=%d frame-size=%d",
+ clock_rate, nb_channels, frame_size);
+
+ /* construct minimal header and comment packet for the decoder */
+ buf = gst_buffer_new_and_alloc (60);
+ gst_buffer_map (buf, &map, GST_MAP_WRITE);
+ ptr = map.data;
+ memcpy (ptr, "CELT ", 8);
+ ptr += 8;
+ memcpy (ptr, "1.1.12", 7);
+ ptr += 20;
+ GST_WRITE_UINT32_LE (ptr, 0x80000006); /* version */
+ ptr += 4;
+ GST_WRITE_UINT32_LE (ptr, 56); /* header_size */
+ ptr += 4;
+ GST_WRITE_UINT32_LE (ptr, clock_rate); /* rate */
+ ptr += 4;
+ GST_WRITE_UINT32_LE (ptr, nb_channels); /* channels */
+ ptr += 4;
+ GST_WRITE_UINT32_LE (ptr, frame_size); /* frame-size */
+ ptr += 4;
+ GST_WRITE_UINT32_LE (ptr, -1); /* overlap */
+ ptr += 4;
+ GST_WRITE_UINT32_LE (ptr, -1); /* bytes_per_packet */
+ ptr += 4;
+ GST_WRITE_UINT32_LE (ptr, 0); /* extra headers */
+ gst_buffer_unmap (buf, &map);
+
+ srccaps = gst_caps_new_empty_simple ("audio/x-celt");
+ res = gst_pad_set_caps (depayload->srcpad, srccaps);
+ gst_caps_unref (srccaps);
+
+ gst_rtp_base_depayload_push (GST_RTP_BASE_DEPAYLOAD (rtpceltdepay), buf);
+
+ buf = gst_buffer_new_and_alloc (sizeof (gst_rtp_celt_comment));
+ gst_buffer_fill (buf, 0, gst_rtp_celt_comment, sizeof (gst_rtp_celt_comment));
+
+ gst_rtp_base_depayload_push (GST_RTP_BASE_DEPAYLOAD (rtpceltdepay), buf);
+
+ return res;
+
+ /* ERRORS */
+no_clockrate:
+ {
+ GST_ERROR_OBJECT (depayload, "no clock-rate specified");
+ return FALSE;
+ }
+}
+
+static GstBuffer *
+gst_rtp_celt_depay_process (GstRTPBaseDepayload * depayload, GstRTPBuffer * rtp)
+{
+ GstBuffer *outbuf = NULL;
+ guint8 *payload;
+ guint offset, pos, payload_len, total_size, size;
+ guint8 s;
+ gint clock_rate = 0, frame_size = 0;
+ GstClockTime framesize_ns = 0, timestamp;
+ guint n = 0;
+ GstRtpCELTDepay *rtpceltdepay;
+
+ rtpceltdepay = GST_RTP_CELT_DEPAY (depayload);
+ clock_rate = depayload->clock_rate;
+ frame_size = rtpceltdepay->frame_size;
+ framesize_ns = gst_util_uint64_scale_int (frame_size, GST_SECOND, clock_rate);
+
+ timestamp = GST_BUFFER_PTS (rtp->buffer);
+
+ GST_LOG_OBJECT (depayload,
+ "got %" G_GSIZE_FORMAT " bytes, mark %d ts %u seqn %d",
+ gst_buffer_get_size (rtp->buffer), gst_rtp_buffer_get_marker (rtp),
+ gst_rtp_buffer_get_timestamp (rtp), gst_rtp_buffer_get_seq (rtp));
+
+ GST_LOG_OBJECT (depayload, "got clock-rate=%d, frame_size=%d, "
+ "_ns=%" GST_TIME_FORMAT ", timestamp=%" GST_TIME_FORMAT, clock_rate,
+ frame_size, GST_TIME_ARGS (framesize_ns), GST_TIME_ARGS (timestamp));
+
+ payload = gst_rtp_buffer_get_payload (rtp);
+ payload_len = gst_rtp_buffer_get_payload_len (rtp);
+
+ /* first count how many bytes are consumed by the size headers and make offset
+ * point to the first data byte */
+ total_size = 0;
+ offset = 0;
+ while (total_size < payload_len) {
+ do {
+ s = payload[offset++];
+ total_size += s + 1;
+ } while (s == 0xff);
+ }
+
+ /* offset is now pointing to the payload */
+ total_size = 0;
+ pos = 0;
+ while (total_size < payload_len) {
+ n++;
+ size = 0;
+ do {
+ s = payload[pos++];
+ size += s;
+ total_size += s + 1;
+ } while (s == 0xff);
+
+ outbuf = gst_rtp_buffer_get_payload_subbuffer (rtp, offset, size);
+ offset += size;
+
+ if (frame_size != -1 && clock_rate != -1) {
+ GST_BUFFER_PTS (outbuf) = timestamp + framesize_ns * n;
+ GST_BUFFER_DURATION (outbuf) = framesize_ns;
+ }
+ GST_LOG_OBJECT (depayload, "push timestamp=%"
+ GST_TIME_FORMAT ", duration=%" GST_TIME_FORMAT,
+ GST_TIME_ARGS (GST_BUFFER_PTS (outbuf)),
+ GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)));
+
+ gst_rtp_drop_non_audio_meta (depayload, outbuf);
+
+ gst_rtp_base_depayload_push (depayload, outbuf);
+ }
+
+ return NULL;
+}