--- /dev/null
+/* GStreamer
+ * Copyright (C) <2007> Nokia Corporation
+ * Copyright (C) <2007> Collabora Ltd
+ * @author: Olivier Crete <olivier.crete@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., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+/*
+ * This payloader assumes that the data will ALWAYS come as zero or more
+ * 10 bytes frame of audio followed by 0 or 1 2 byte frame of silence.
+ * Any other buffer format won't work
+ */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <string.h>
+#include <gst/rtp/gstrtpbuffer.h>
+#include <gst/base/gstadapter.h>
+
+#include "gstrtpg723pay.h"
+
+#define GST_RTP_PAYLOAD_G723 4
+#define GST_RTP_PAYLOAD_G723_STRING "4"
+
+/* According to RFC 3551, works only with G723 encoded with 6.3 kb/s high-rate */
+#define G723_FRAME_SIZE 24
+#define G723B_SID_FRAME_SIZE 4
+#define G723_FRAME_DURATION (30 * GST_MSECOND)
+#define G723_FRAME_DURATION_MS (30)
+
+static gboolean
+gst_rtp_g723_pay_set_caps (GstBaseRTPPayload * payload, GstCaps * caps);
+static GstFlowReturn
+gst_rtp_g723_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf);
+
+
+static const GstElementDetails gst_rtp_g723_pay_details =
+GST_ELEMENT_DETAILS ("RTP G.723 payloader",
+ "Codec/Payloader/Network",
+ "Packetize 6.3kb/s G.723 audio into RTP packets",
+ "Tiago Katcipis <tiago.katcipis@digitro.com.br>");
+
+static GstStaticPadTemplate gst_rtp_g723_pay_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/G723, " /* according to RFC 3551 */
+ "channels = (int) 1, " "rate = (int) 8000")
+ );
+
+static GstStaticPadTemplate gst_rtp_g723_pay_src_template =
+ GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("application/x-rtp, "
+ "media = (string) \"audio\", "
+ "payload = (int) " GST_RTP_PAYLOAD_G723_STRING ", "
+ "clock-rate = (int) 8000, "
+ "encoding-name = (string) \"G723\"; "
+ "application/x-rtp, "
+ "media = (string) \"audio\", "
+ "payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
+ "clock-rate = (int) 8000, " "encoding-name = (string) \"G723\"")
+ );
+
+static void
+gst_rtp_g723_pay_init (GstRTPG723Pay * pay, GstRTPG723PayClass * klass);
+
+GST_BOILERPLATE (GstRTPG723Pay, gst_rtp_g723_pay, GstBaseRTPAudioPayload,
+ GST_TYPE_BASE_RTP_AUDIO_PAYLOAD);
+
+static void
+gst_rtp_g723_pay_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_g723_pay_sink_template));
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&gst_rtp_g723_pay_src_template));
+ gst_element_class_set_details (element_class, &gst_rtp_g723_pay_details);
+}
+
+static void
+gst_rtp_g723_pay_class_init (GstRTPG723PayClass * klass)
+{
+ GstBaseRTPPayloadClass *payload_class = GST_BASE_RTP_PAYLOAD_CLASS (klass);
+
+ payload_class->set_caps = gst_rtp_g723_pay_set_caps;
+ payload_class->handle_buffer = gst_rtp_g723_pay_handle_buffer;
+}
+
+static void
+gst_rtp_g723_pay_init (GstRTPG723Pay * pay, GstRTPG723PayClass * klass)
+{
+ GstBaseRTPPayload *payload = GST_BASE_RTP_PAYLOAD (pay);
+ GstBaseRTPAudioPayload *audiopayload = GST_BASE_RTP_AUDIO_PAYLOAD (pay);
+
+ payload->pt = GST_RTP_PAYLOAD_G723;
+ gst_basertppayload_set_options (payload, "audio", FALSE, "G723", 8000);
+
+ gst_base_rtp_audio_payload_set_frame_based (audiopayload);
+ gst_base_rtp_audio_payload_set_frame_options (audiopayload,
+ G723_FRAME_DURATION_MS, G723_FRAME_SIZE);
+
+}
+
+static gboolean
+gst_rtp_g723_pay_set_caps (GstBaseRTPPayload * payload, GstCaps * caps)
+{
+ gboolean res;
+ GstStructure *structure;
+ gint pt;
+
+ structure = gst_caps_get_structure (caps, 0);
+ if (!gst_structure_get_int (structure, "payload", &pt))
+ pt = GST_RTP_PAYLOAD_G723;
+
+ payload->pt = pt;
+ payload->dynamic = pt != GST_RTP_PAYLOAD_G723;
+
+ res = gst_basertppayload_set_outcaps (payload, NULL);
+
+ return res;
+}
+
+static GstFlowReturn
+gst_rtp_g723_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buf)
+{
+ GstFlowReturn ret = GST_FLOW_OK;
+ GstBaseRTPAudioPayload *basertpaudiopayload =
+ GST_BASE_RTP_AUDIO_PAYLOAD (payload);
+ GstAdapter *adapter = NULL;
+ guint payload_len;
+ const guint8 *data = NULL;
+ guint available;
+ guint maxptime_octets = G_MAXUINT;
+ guint minptime_octets = 0;
+ guint min_payload_len;
+ guint max_payload_len;
+ gboolean use_adapter = FALSE;
+
+ available = GST_BUFFER_SIZE (buf);
+
+ if (available % G723_FRAME_SIZE != 0 &&
+ available % G723_FRAME_SIZE != G723B_SID_FRAME_SIZE)
+ goto invalid_size;
+
+ /* max number of bytes based on given ptime, has to be multiple of
+ * frame_duration */
+ if (payload->max_ptime != -1) {
+ guint ptime_ms = payload->max_ptime / 1000000;
+
+ maxptime_octets = G723_FRAME_SIZE *
+ (int) (ptime_ms / G723_FRAME_DURATION_MS);
+
+ if (maxptime_octets < G723_FRAME_SIZE) {
+ GST_WARNING_OBJECT (basertpaudiopayload, "Given ptime %" G_GINT64_FORMAT
+ " is smaller than minimum %d ns, overwriting to minimum",
+ payload->max_ptime, G723_FRAME_DURATION_MS);
+ maxptime_octets = G723_FRAME_SIZE;
+ }
+ }
+
+ max_payload_len = MIN (
+ /* MTU max */
+ (int) (gst_rtp_buffer_calc_payload_len (GST_BASE_RTP_PAYLOAD_MTU
+ (basertpaudiopayload), 0, 0) / G723_FRAME_SIZE) * G723_FRAME_SIZE,
+ /* ptime max */
+ maxptime_octets);
+
+ /* min number of bytes based on a given ptime, has to be a multiple
+ of frame duration */
+ {
+ guint64 min_ptime;
+
+ g_object_get (G_OBJECT (payload), "min-ptime", &min_ptime, NULL);
+
+ min_ptime = min_ptime / 1000000;
+ minptime_octets = G723_FRAME_SIZE *
+ (int) (min_ptime / G723_FRAME_DURATION_MS);
+ }
+
+ min_payload_len = MAX (minptime_octets, G723_FRAME_SIZE);
+
+ if (min_payload_len > max_payload_len) {
+ min_payload_len = max_payload_len;
+ }
+
+ GST_DEBUG_OBJECT (basertpaudiopayload,
+ "Calculated min_payload_len %u and max_payload_len %u",
+ min_payload_len, max_payload_len);
+
+ adapter = gst_base_rtp_audio_payload_get_adapter (basertpaudiopayload);
+
+ if (adapter && gst_adapter_available (adapter)) {
+ /* If there is always data in the adapter, we have to use it */
+ gst_adapter_push (adapter, buf);
+ available = gst_adapter_available (adapter);
+ use_adapter = TRUE;
+ } else {
+ /* let's set the base timestamp */
+ basertpaudiopayload->base_ts = GST_BUFFER_TIMESTAMP (buf);
+
+ /* If buffer fits on an RTP packet, let's just push it through */
+ /* this will check against max_ptime and max_mtu */
+ if (GST_BUFFER_SIZE (buf) >= min_payload_len &&
+ GST_BUFFER_SIZE (buf) <= max_payload_len) {
+ ret = gst_base_rtp_audio_payload_push (basertpaudiopayload,
+ GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf),
+ GST_BUFFER_TIMESTAMP (buf));
+ gst_buffer_unref (buf);
+ return ret;
+ }
+
+ available = GST_BUFFER_SIZE (buf);
+ data = (guint8 *) GST_BUFFER_DATA (buf);
+ }
+
+ /* as long as we have full frames */
+ /* this loop will push all available buffers till the last frame */
+ while (available >= min_payload_len ||
+ available % G723_FRAME_SIZE == G723B_SID_FRAME_SIZE) {
+ guint num;
+
+ /* We send as much as we can */
+ if (available <= max_payload_len) {
+ payload_len = available;
+ } else {
+ payload_len = MIN (max_payload_len,
+ (available / G723_FRAME_SIZE) * G723_FRAME_SIZE);
+ }
+
+ if (use_adapter) {
+ data = gst_adapter_peek (adapter, payload_len);
+ }
+
+ ret = gst_base_rtp_audio_payload_push (basertpaudiopayload, data,
+ payload_len, basertpaudiopayload->base_ts);
+
+ num = payload_len / G723_FRAME_SIZE;
+ basertpaudiopayload->base_ts += G723_FRAME_DURATION * num;
+
+ if (use_adapter) {
+ gst_adapter_flush (adapter, payload_len);
+ available = gst_adapter_available (adapter);
+ } else {
+ available -= payload_len;
+ data += payload_len;
+ }
+ }
+
+ if (!use_adapter) {
+ if (available != 0 && adapter) {
+ GstBuffer *buf2;
+ buf2 = gst_buffer_create_sub (buf,
+ GST_BUFFER_SIZE (buf) - available, available);
+ gst_adapter_push (adapter, buf2);
+ } else {
+ gst_buffer_unref (buf);
+ }
+ }
+
+ if (adapter) {
+ g_object_unref (adapter);
+ }
+
+ return ret;
+
+ /* ERRORS */
+invalid_size:
+ {
+ GST_ELEMENT_ERROR (payload, STREAM, WRONG_TYPE,
+ ("Invalid input buffer size"),
+ ("Invalid buffer size, should be a multiple of"
+ " G723_FRAME_SIZE(24) with an optional G723B_SID_FRAME_SIZE(4)"
+ " added to it, but it is %u", available));
+ gst_buffer_unref (buf);
+ return GST_FLOW_ERROR;
+ }
+}
+
+/*Plugin init functions*/
+gboolean
+gst_rtp_g723_pay_plugin_init (GstPlugin * plugin)
+{
+ return gst_element_register (plugin, "rtpg723pay", GST_RANK_NONE,
+ gst_rtp_g723_pay_get_type ());
+}
--- /dev/null
+/* GStreamer
+ * Copyright (C) <2007> Nokia Corporation
+ * Copyright (C) <2007> Collabora Ltd
+ *
+ * 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.
+ */
+
+#ifndef __GST_RTP_G723_PAY_H__
+#define __GST_RTP_G723_PAY_H__
+
+#include <gst/gst.h>
+#include <gst/rtp/gstbasertpaudiopayload.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RTP_G723_PAY \
+ (gst_rtp_g723_pay_get_type())
+#define GST_RTP_G723_PAY(obj) \
+ (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTP_G723_PAY,GstRTPG723Pay))
+#define GST_RTP_G723_PAY_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RTP_G723_PAY,GstRTPG723PayClass))
+#define GST_IS_RTP_G723_PAY(obj) \
+ (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTP_G723_PAY))
+#define GST_IS_RTP_G723_PAY_CLASS(klass) \
+ (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RTP_G723_PAY))
+
+typedef struct _GstRTPG723Pay GstRTPG723Pay;
+typedef struct _GstRTPG723PayClass GstRTPG723PayClass;
+
+struct _GstRTPG723Pay
+{
+ GstBaseRTPAudioPayload audiopayload;
+};
+
+struct _GstRTPG723PayClass
+{
+ GstBaseRTPAudioPayloadClass parent_class;
+};
+
+gboolean gst_rtp_g723_pay_plugin_init (GstPlugin * plugin);
+
+GType gst_rtp_g723_pay_get_type (void);
+
+G_END_DECLS
+
+#endif /* __GST_RTP_G723_PAY_H__ */
projects / platform / upstream / gst-plugins-good.git / commitdiff