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Diffstat (limited to 'gst/rtp/gstrtpsbcdepay.c')
-rwxr-xr-xgst/rtp/gstrtpsbcdepay.c287
1 files changed, 287 insertions, 0 deletions
diff --git a/gst/rtp/gstrtpsbcdepay.c b/gst/rtp/gstrtpsbcdepay.c
new file mode 100755
index 0000000..28c00fc
--- /dev/null
+++ b/gst/rtp/gstrtpsbcdepay.c
@@ -0,0 +1,287 @@
+/*
+ * GStreamer RTP SBC depayloader
+ *
+ * Copyright (C) 2012 Collabora Ltd.
+ * @author: Arun Raghavan <arun.raghavan@collabora.co.uk>
+ *
+ * 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 <gst/rtp/gstrtpbuffer.h>
+#include "gstrtpsbcdepay.h"
+
+GST_DEBUG_CATEGORY_STATIC (rtpsbcdepay_debug);
+#define GST_CAT_DEFAULT (rtpsbcdepay_debug)
+
+static GstStaticPadTemplate gst_rtp_sbc_depay_src_template =
+GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-sbc, "
+ "rate = (int) { 16000, 32000, 44100, 48000 }, "
+ "channels = (int) [ 1, 2 ], "
+ "mode = (string) { mono, dual, stereo, joint }, "
+ "blocks = (int) { 4, 8, 12, 16 }, "
+ "subbands = (int) { 4, 8 }, "
+ "allocation-method = (string) { snr, loudness }, "
+ "bitpool = (int) [ 2, 64 ]")
+ );
+
+static GstStaticPadTemplate gst_rtp_sbc_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) { 16000, 32000, 44100, 48000 },"
+ "encoding-name = (string) SBC")
+ );
+
+#define gst_rtp_sbc_depay_parent_class parent_class
+G_DEFINE_TYPE (GstRtpSbcDepay, gst_rtp_sbc_depay, GST_TYPE_RTP_BASE_DEPAYLOAD);
+
+static void gst_rtp_sbc_depay_finalize (GObject * object);
+
+static gboolean gst_rtp_sbc_depay_setcaps (GstRTPBaseDepayload * base,
+ GstCaps * caps);
+static GstBuffer *gst_rtp_sbc_depay_process (GstRTPBaseDepayload * base,
+ GstBuffer * in);
+
+static void
+gst_rtp_sbc_depay_class_init (GstRtpSbcDepayClass * klass)
+{
+ GstRTPBaseDepayloadClass *gstbasertpdepayload_class =
+ GST_RTP_BASE_DEPAYLOAD_CLASS (klass);
+ GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
+ GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
+
+ gobject_class->finalize = gst_rtp_sbc_depay_finalize;
+
+ gstbasertpdepayload_class->set_caps = gst_rtp_sbc_depay_setcaps;
+ gstbasertpdepayload_class->process = gst_rtp_sbc_depay_process;
+
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&gst_rtp_sbc_depay_src_template));
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&gst_rtp_sbc_depay_sink_template));
+
+ GST_DEBUG_CATEGORY_INIT (rtpsbcdepay_debug, "rtpsbcdepay", 0,
+ "SBC Audio RTP Depayloader");
+
+ gst_element_class_set_static_metadata (element_class,
+ "RTP SBC audio depayloader",
+ "Codec/Depayloader/Network/RTP",
+ "Extracts SBC audio from RTP packets",
+ "Arun Raghavan <arun.raghavan@collabora.co.uk>");
+}
+
+static void
+gst_rtp_sbc_depay_init (GstRtpSbcDepay * rtpsbcdepay)
+{
+ rtpsbcdepay->adapter = gst_adapter_new ();
+}
+
+static void
+gst_rtp_sbc_depay_finalize (GObject * object)
+{
+ GstRtpSbcDepay *depay = GST_RTP_SBC_DEPAY (object);
+
+ gst_object_unref (depay->adapter);
+
+ G_OBJECT_CLASS (parent_class)->finalize (object);
+}
+
+/* FIXME: This duplicates similar functionality rtpsbcpay, but there isn't a
+ * simple way to consolidate the two. This is best done by moving the function
+ * to the codec-utils library in gst-plugins-base when these elements move to
+ * GStreamer. */
+static int
+gst_rtp_sbc_depay_get_params (GstRtpSbcDepay * depay, const guint8 * data,
+ gint size, int *framelen, int *samples)
+{
+ int blocks, channel_mode, channels, subbands, bitpool;
+ int length;
+
+ if (size < 3) {
+ /* Not enough data for the header */
+ return -1;
+ }
+
+ /* Sanity check */
+ if (data[0] != 0x9c) {
+ GST_WARNING_OBJECT (depay, "Bad packet: couldn't find syncword");
+ return -2;
+ }
+
+ blocks = (data[1] >> 4) & 0x3;
+ blocks = (blocks + 1) * 4;
+ channel_mode = (data[1] >> 2) & 0x3;
+ channels = channel_mode ? 2 : 1;
+ subbands = (data[1] & 0x1);
+ subbands = (subbands + 1) * 4;
+ bitpool = data[2];
+
+ length = 4 + ((4 * subbands * channels) / 8);
+
+ if (channel_mode == 0 || channel_mode == 1) {
+ /* Mono || Dual channel */
+ length += ((blocks * channels * bitpool)
+ + 4 /* round up */ ) / 8;
+ } else {
+ /* Stereo || Joint stereo */
+ gboolean joint = (channel_mode == 3);
+
+ length += ((joint * subbands) + (blocks * bitpool)
+ + 4 /* round up */ ) / 8;
+ }
+
+ *framelen = length;
+ *samples = blocks * subbands;
+
+ return 0;
+}
+
+static gboolean
+gst_rtp_sbc_depay_setcaps (GstRTPBaseDepayload * base, GstCaps * caps)
+{
+ GstRtpSbcDepay *depay = GST_RTP_SBC_DEPAY (base);
+ GstStructure *structure;
+ GstCaps *outcaps, *oldcaps;
+
+ structure = gst_caps_get_structure (caps, 0);
+
+ if (!gst_structure_get_int (structure, "clock-rate", &depay->rate))
+ goto bad_caps;
+
+ outcaps = gst_caps_new_simple ("audio/x-sbc", "rate", G_TYPE_INT,
+ depay->rate, NULL);
+
+ gst_pad_set_caps (GST_RTP_BASE_DEPAYLOAD_SRCPAD (base), outcaps);
+
+ oldcaps = gst_pad_get_current_caps (GST_RTP_BASE_DEPAYLOAD_SINKPAD (base));
+ if (oldcaps && !gst_caps_can_intersect (oldcaps, caps)) {
+ /* Caps have changed, flush old data */
+ gst_adapter_clear (depay->adapter);
+ }
+
+ gst_caps_unref (outcaps);
+
+ return TRUE;
+
+bad_caps:
+ GST_WARNING_OBJECT (depay, "Can't support the caps we got: %"
+ GST_PTR_FORMAT, caps);
+ return FALSE;
+}
+
+static GstBuffer *
+gst_rtp_sbc_depay_process (GstRTPBaseDepayload * base, GstBuffer * in)
+{
+ GstRtpSbcDepay *depay = GST_RTP_SBC_DEPAY (base);
+ GstBuffer *data = NULL;
+ GstRTPBuffer rtp = { NULL };
+
+ gboolean fragment, start, last;
+ guint8 nframes;
+ guint8 *payload;
+ guint payload_len;
+
+ gst_rtp_buffer_map (in, GST_MAP_READ, &rtp);
+
+ GST_LOG_OBJECT (depay, "Got %" G_GSIZE_FORMAT " bytes",
+ gst_buffer_get_size (in));
+
+ if (gst_rtp_buffer_get_marker (&rtp)) {
+ /* Marker isn't supposed to be set */
+ GST_WARNING_OBJECT (depay, "Marker bit was set");
+ goto bad_packet;
+ }
+
+ payload = gst_rtp_buffer_get_payload (&rtp);
+ payload_len = gst_rtp_buffer_get_payload_len (&rtp);
+
+ fragment = payload[0] & 0x80;
+ start = payload[0] & 0x40;
+ last = payload[0] & 0x20;
+ nframes = payload[0] & 0x0f;
+
+ payload += 1;
+ payload_len -= 1;
+
+ data = gst_rtp_buffer_get_payload_subbuffer (&rtp, 1, -1);
+
+ if (fragment) {
+ /* Got a packet with a fragment */
+ GST_LOG_OBJECT (depay, "Got fragment");
+
+ if (start && gst_adapter_available (depay->adapter)) {
+ GST_WARNING_OBJECT (depay, "Missing last fragment");
+ gst_adapter_clear (depay->adapter);
+
+ } else if (!start && !gst_adapter_available (depay->adapter)) {
+ GST_WARNING_OBJECT (depay, "Missing start fragment");
+ gst_buffer_unref (data);
+ data = NULL;
+ goto out;
+ }
+
+ gst_adapter_push (depay->adapter, data);
+
+ if (last) {
+ data = gst_adapter_take_buffer (depay->adapter,
+ gst_adapter_available (depay->adapter));
+ } else
+ data = NULL;
+
+ } else {
+ /* !fragment */
+ gint framelen, samples;
+
+ GST_LOG_OBJECT (depay, "Got %d frames", nframes);
+
+ if (gst_rtp_sbc_depay_get_params (depay, payload,
+ payload_len, &framelen, &samples) < 0) {
+ gst_adapter_clear (depay->adapter);
+ goto bad_packet;
+ }
+
+ GST_LOG_OBJECT (depay, "Got payload of %d", payload_len);
+
+ if (nframes * framelen > (gint) payload_len) {
+ GST_WARNING_OBJECT (depay, "Short packet");
+ goto bad_packet;
+ } else if (nframes * framelen < (gint) payload_len) {
+ GST_WARNING_OBJECT (depay, "Junk at end of packet");
+ }
+ }
+
+out:
+ gst_rtp_buffer_unmap (&rtp);
+ return data;
+
+bad_packet:
+ GST_ELEMENT_WARNING (depay, STREAM, DECODE,
+ ("Received invalid RTP payload, dropping"), (NULL));
+ goto out;
+}
+
+gboolean
+gst_rtp_sbc_depay_plugin_init (GstPlugin * plugin)
+{
+ return gst_element_register (plugin, "rtpsbcdepay", GST_RANK_SECONDARY,
+ GST_TYPE_RTP_SBC_DEPAY);
+}
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