gst/rtp/gstrtpspeexpay.c - mtk-gst-plugins-good - Git at Google

 /* GStreamer
  * Copyright (C) <2005> Edgard Lima <edgard.lima@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 <stdlib.h>
 #include <string.h>
 #include <gst/rtp/gstrtpbuffer.h>
 #include <gst/audio/audio.h>

 #include "gstrtpspeexpay.h"
 #include "gstrtputils.h"

 GST_DEBUG_CATEGORY_STATIC (rtpspeexpay_debug);
 #define GST_CAT_DEFAULT (rtpspeexpay_debug)

 static GstStaticPadTemplate gst_rtp_speex_pay_sink_template =
 GST_STATIC_PAD_TEMPLATE ("sink",
     GST_PAD_SINK,
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("audio/x-speex, "
         "rate = (int) [ 6000, 48000 ], " "channels = (int) 1")
     );

 static GstStaticPadTemplate gst_rtp_speex_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_DYNAMIC_STRING ", "
         "clock-rate =  (int) [ 6000, 48000 ], "
         "encoding-name = (string) \"SPEEX\", "
         "encoding-params = (string) \"1\"")
     );

 static GstStateChangeReturn gst_rtp_speex_pay_change_state (GstElement *
     element, GstStateChange transition);

 static gboolean gst_rtp_speex_pay_setcaps (GstRTPBasePayload * payload,
     GstCaps * caps);
 static GstCaps *gst_rtp_speex_pay_getcaps (GstRTPBasePayload * payload,
     GstPad * pad, GstCaps * filter);
 static GstFlowReturn gst_rtp_speex_pay_handle_buffer (GstRTPBasePayload *
     payload, GstBuffer * buffer);

 #define gst_rtp_speex_pay_parent_class parent_class
 G_DEFINE_TYPE (GstRtpSPEEXPay, gst_rtp_speex_pay, GST_TYPE_RTP_BASE_PAYLOAD);

 static void
 gst_rtp_speex_pay_class_init (GstRtpSPEEXPayClass * klass)
 {
   GstElementClass *gstelement_class;
   GstRTPBasePayloadClass *gstrtpbasepayload_class;

   gstelement_class = (GstElementClass *) klass;
   gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;

   gstelement_class->change_state = gst_rtp_speex_pay_change_state;

   gstrtpbasepayload_class->set_caps = gst_rtp_speex_pay_setcaps;
   gstrtpbasepayload_class->get_caps = gst_rtp_speex_pay_getcaps;
   gstrtpbasepayload_class->handle_buffer = gst_rtp_speex_pay_handle_buffer;

   gst_element_class_add_static_pad_template (gstelement_class,
       &gst_rtp_speex_pay_sink_template);
   gst_element_class_add_static_pad_template (gstelement_class,
       &gst_rtp_speex_pay_src_template);
   gst_element_class_set_static_metadata (gstelement_class,
       "RTP Speex payloader", "Codec/Payloader/Network/RTP",
       "Payload-encodes Speex audio into a RTP packet",
       "Edgard Lima <edgard.lima@gmail.com>");

   GST_DEBUG_CATEGORY_INIT (rtpspeexpay_debug, "rtpspeexpay", 0,
       "Speex RTP Payloader");
 }

 static void
 gst_rtp_speex_pay_init (GstRtpSPEEXPay * rtpspeexpay)
 {
   GST_RTP_BASE_PAYLOAD (rtpspeexpay)->clock_rate = 8000;
   GST_RTP_BASE_PAYLOAD_PT (rtpspeexpay) = 110;  /* Create String */
 }

 static gboolean
 gst_rtp_speex_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
 {
   /* don't configure yet, we wait for the ident packet */
   return TRUE;
 }


 static GstCaps *
 gst_rtp_speex_pay_getcaps (GstRTPBasePayload * payload, GstPad * pad,
     GstCaps * filter)
 {
   GstCaps *otherpadcaps;
   GstCaps *caps;

   otherpadcaps = gst_pad_get_allowed_caps (payload->srcpad);
   caps = gst_pad_get_pad_template_caps (pad);

   if (otherpadcaps) {
     if (!gst_caps_is_empty (otherpadcaps)) {
       GstStructure *ps;
       GstStructure *s;
       gint clock_rate;

       ps = gst_caps_get_structure (otherpadcaps, 0);
       caps = gst_caps_make_writable (caps);
       s = gst_caps_get_structure (caps, 0);

       if (gst_structure_get_int (ps, "clock-rate", &clock_rate)) {
         gst_structure_fixate_field_nearest_int (s, "rate", clock_rate);
       }
     }
     gst_caps_unref (otherpadcaps);
   }

   if (filter) {
     GstCaps *tcaps = caps;

     caps = gst_caps_intersect_full (filter, tcaps, GST_CAPS_INTERSECT_FIRST);
     gst_caps_unref (tcaps);
   }

   return caps;
 }

 static gboolean
 gst_rtp_speex_pay_parse_ident (GstRtpSPEEXPay * rtpspeexpay,
     const guint8 * data, guint size)
 {
   guint32 version, header_size, rate, mode, nb_channels;
   GstRTPBasePayload *payload;
   gchar *cstr;
   gboolean res;

   /* we need the header string (8), the version string (20), the version
    * and the header length. */
   if (size < 36)
     goto too_small;

   if (!g_str_has_prefix ((const gchar *) data, "Speex   "))
     goto wrong_header;

   /* skip header and version string */
   data += 28;

   version = GST_READ_UINT32_LE (data);
   if (version != 1)
     goto wrong_version;

   data += 4;
   /* ensure sizes */
   header_size = GST_READ_UINT32_LE (data);
   if (header_size < 80)
     goto header_too_small;

   if (size < header_size)
     goto payload_too_small;

   data += 4;
   rate = GST_READ_UINT32_LE (data);
   data += 4;
   mode = GST_READ_UINT32_LE (data);
   data += 8;
   nb_channels = GST_READ_UINT32_LE (data);

   GST_DEBUG_OBJECT (rtpspeexpay, "rate %d, mode %d, nb_channels %d",
       rate, mode, nb_channels);

   payload = GST_RTP_BASE_PAYLOAD (rtpspeexpay);

   gst_rtp_base_payload_set_options (payload, "audio", FALSE, "SPEEX", rate);
   cstr = g_strdup_printf ("%d", nb_channels);
   res = gst_rtp_base_payload_set_outcaps (payload, "encoding-params",
       G_TYPE_STRING, cstr, NULL);
   g_free (cstr);

   return res;

   /* ERRORS */
 too_small:
   {
     GST_DEBUG_OBJECT (rtpspeexpay,
         "ident packet too small, need at least 32 bytes");
     return FALSE;
   }
 wrong_header:
   {
     GST_DEBUG_OBJECT (rtpspeexpay,
         "ident packet does not start with \"Speex   \"");
     return FALSE;
   }
 wrong_version:
   {
     GST_DEBUG_OBJECT (rtpspeexpay, "can only handle version 1, have version %d",
         version);
     return FALSE;
   }
 header_too_small:
   {
     GST_DEBUG_OBJECT (rtpspeexpay,
         "header size too small, need at least 80 bytes, " "got only %d",
         header_size);
     return FALSE;
   }
 payload_too_small:
   {
     GST_DEBUG_OBJECT (rtpspeexpay,
         "payload too small, need at least %d bytes, got only %d", header_size,
         size);
     return FALSE;
   }
 }

 static GstFlowReturn
 gst_rtp_speex_pay_handle_buffer (GstRTPBasePayload * basepayload,
     GstBuffer * buffer)
 {
   GstRtpSPEEXPay *rtpspeexpay;
   GstMapInfo map;
   GstBuffer *outbuf;
   GstClockTime timestamp, duration;
   GstFlowReturn ret;

   rtpspeexpay = GST_RTP_SPEEX_PAY (basepayload);

   gst_buffer_map (buffer, &map, GST_MAP_READ);

   switch (rtpspeexpay->packet) {
     case 0:
       /* ident packet. We need to parse the headers to construct the RTP
        * properties. */
       if (!gst_rtp_speex_pay_parse_ident (rtpspeexpay, map.data, map.size)) {
         gst_buffer_unmap (buffer, &map);
         goto parse_error;
       }

       ret = GST_FLOW_OK;
       gst_buffer_unmap (buffer, &map);
       goto done;
     case 1:
       /* comment packet, we ignore it */
       ret = GST_FLOW_OK;
       gst_buffer_unmap (buffer, &map);
       goto done;
     default:
       /* other packets go in the payload */
       break;
   }
   gst_buffer_unmap (buffer, &map);

   if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_GAP)) {
     ret = GST_FLOW_OK;
     goto done;
   }

   timestamp = GST_BUFFER_PTS (buffer);
   duration = GST_BUFFER_DURATION (buffer);

   /* FIXME, only one SPEEX frame per RTP packet for now */

   outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
   /* FIXME, assert for now */
   g_assert (gst_buffer_get_size (buffer) <=
       GST_RTP_BASE_PAYLOAD_MTU (rtpspeexpay));

   /* copy timestamp and duration */
   GST_BUFFER_PTS (outbuf) = timestamp;
   GST_BUFFER_DURATION (outbuf) = duration;

   gst_rtp_copy_audio_meta (basepayload, outbuf, buffer);
   outbuf = gst_buffer_append (outbuf, buffer);
   buffer = NULL;

   ret = gst_rtp_base_payload_push (basepayload, outbuf);

 done:
   if (buffer)
     gst_buffer_unref (buffer);

   rtpspeexpay->packet++;

   return ret;

   /* ERRORS */
 parse_error:
   {
     GST_ELEMENT_ERROR (rtpspeexpay, STREAM, DECODE, (NULL),
         ("Error parsing first identification packet."));
     gst_buffer_unref (buffer);
     return GST_FLOW_ERROR;
   }
 }

 static GstStateChangeReturn
 gst_rtp_speex_pay_change_state (GstElement * element, GstStateChange transition)
 {
   GstRtpSPEEXPay *rtpspeexpay;
   GstStateChangeReturn ret;

   rtpspeexpay = GST_RTP_SPEEX_PAY (element);

   switch (transition) {
     case GST_STATE_CHANGE_NULL_TO_READY:
       break;
     case GST_STATE_CHANGE_READY_TO_PAUSED:
       rtpspeexpay->packet = 0;
       break;
     default:
       break;
   }

   ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);

   switch (transition) {
     case GST_STATE_CHANGE_READY_TO_NULL:
       break;
     default:
       break;
   }
   return ret;
 }

 gboolean
 gst_rtp_speex_pay_plugin_init (GstPlugin * plugin)
 {
   return gst_element_register (plugin, "rtpspeexpay",
       GST_RANK_SECONDARY, GST_TYPE_RTP_SPEEX_PAY);
 }
	/* GStreamer
	* Copyright (C) <2005> Edgard Lima <edgard.lima@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 <stdlib.h>
	#include <string.h>
	#include <gst/rtp/gstrtpbuffer.h>
	#include <gst/audio/audio.h>

	#include "gstrtpspeexpay.h"
	#include "gstrtputils.h"

	GST_DEBUG_CATEGORY_STATIC (rtpspeexpay_debug);
	#define GST_CAT_DEFAULT (rtpspeexpay_debug)

	static GstStaticPadTemplate gst_rtp_speex_pay_sink_template =
	GST_STATIC_PAD_TEMPLATE ("sink",
	GST_PAD_SINK,
	GST_PAD_ALWAYS,
	GST_STATIC_CAPS ("audio/x-speex, "
	"rate = (int) [ 6000, 48000 ], " "channels = (int) 1")
	);

	static GstStaticPadTemplate gst_rtp_speex_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_DYNAMIC_STRING ", "
	"clock-rate = (int) [ 6000, 48000 ], "
	"encoding-name = (string) \"SPEEX\", "
	"encoding-params = (string) \"1\"")
	);

	static GstStateChangeReturn gst_rtp_speex_pay_change_state (GstElement *
	element, GstStateChange transition);

	static gboolean gst_rtp_speex_pay_setcaps (GstRTPBasePayload * payload,
	GstCaps * caps);
	static GstCaps gst_rtp_speex_pay_getcaps (GstRTPBasePayload payload,
	GstPad * pad, GstCaps * filter);
	static GstFlowReturn gst_rtp_speex_pay_handle_buffer (GstRTPBasePayload *
	payload, GstBuffer * buffer);

	#define gst_rtp_speex_pay_parent_class parent_class
	G_DEFINE_TYPE (GstRtpSPEEXPay, gst_rtp_speex_pay, GST_TYPE_RTP_BASE_PAYLOAD);

	static void
	gst_rtp_speex_pay_class_init (GstRtpSPEEXPayClass * klass)
	{
	GstElementClass *gstelement_class;
	GstRTPBasePayloadClass *gstrtpbasepayload_class;

	gstelement_class = (GstElementClass *) klass;
	gstrtpbasepayload_class = (GstRTPBasePayloadClass *) klass;

	gstelement_class->change_state = gst_rtp_speex_pay_change_state;

	gstrtpbasepayload_class->set_caps = gst_rtp_speex_pay_setcaps;
	gstrtpbasepayload_class->get_caps = gst_rtp_speex_pay_getcaps;
	gstrtpbasepayload_class->handle_buffer = gst_rtp_speex_pay_handle_buffer;

	gst_element_class_add_static_pad_template (gstelement_class,
	&gst_rtp_speex_pay_sink_template);
	gst_element_class_add_static_pad_template (gstelement_class,
	&gst_rtp_speex_pay_src_template);
	gst_element_class_set_static_metadata (gstelement_class,
	"RTP Speex payloader", "Codec/Payloader/Network/RTP",
	"Payload-encodes Speex audio into a RTP packet",
	"Edgard Lima <edgard.lima@gmail.com>");

	GST_DEBUG_CATEGORY_INIT (rtpspeexpay_debug, "rtpspeexpay", 0,
	"Speex RTP Payloader");
	}

	static void
	gst_rtp_speex_pay_init (GstRtpSPEEXPay * rtpspeexpay)
	{
	GST_RTP_BASE_PAYLOAD (rtpspeexpay)->clock_rate = 8000;
	GST_RTP_BASE_PAYLOAD_PT (rtpspeexpay) = 110; /* Create String */
	}

	static gboolean
	gst_rtp_speex_pay_setcaps (GstRTPBasePayload * payload, GstCaps * caps)
	{
	/* don't configure yet, we wait for the ident packet */
	return TRUE;
	}


	static GstCaps *
	gst_rtp_speex_pay_getcaps (GstRTPBasePayload * payload, GstPad * pad,
	GstCaps * filter)
	{
	GstCaps *otherpadcaps;
	GstCaps *caps;

	otherpadcaps = gst_pad_get_allowed_caps (payload->srcpad);
	caps = gst_pad_get_pad_template_caps (pad);

	if (otherpadcaps) {
	if (!gst_caps_is_empty (otherpadcaps)) {
	GstStructure *ps;
	GstStructure *s;
	gint clock_rate;

	ps = gst_caps_get_structure (otherpadcaps, 0);
	caps = gst_caps_make_writable (caps);
	s = gst_caps_get_structure (caps, 0);

	if (gst_structure_get_int (ps, "clock-rate", &clock_rate)) {
	gst_structure_fixate_field_nearest_int (s, "rate", clock_rate);
	}
	}
	gst_caps_unref (otherpadcaps);
	}

	if (filter) {
	GstCaps *tcaps = caps;

	caps = gst_caps_intersect_full (filter, tcaps, GST_CAPS_INTERSECT_FIRST);
	gst_caps_unref (tcaps);
	}

	return caps;
	}

	static gboolean
	gst_rtp_speex_pay_parse_ident (GstRtpSPEEXPay * rtpspeexpay,
	const guint8 * data, guint size)
	{
	guint32 version, header_size, rate, mode, nb_channels;
	GstRTPBasePayload *payload;
	gchar *cstr;
	gboolean res;

	/* we need the header string (8), the version string (20), the version
	* and the header length. */
	if (size < 36)
	goto too_small;

	if (!g_str_has_prefix ((const gchar *) data, "Speex "))
	goto wrong_header;

	/* skip header and version string */
	data += 28;

	version = GST_READ_UINT32_LE (data);
	if (version != 1)
	goto wrong_version;

	data += 4;
	/* ensure sizes */
	header_size = GST_READ_UINT32_LE (data);
	if (header_size < 80)
	goto header_too_small;

	if (size < header_size)
	goto payload_too_small;

	data += 4;
	rate = GST_READ_UINT32_LE (data);
	data += 4;
	mode = GST_READ_UINT32_LE (data);
	data += 8;
	nb_channels = GST_READ_UINT32_LE (data);

	GST_DEBUG_OBJECT (rtpspeexpay, "rate %d, mode %d, nb_channels %d",
	rate, mode, nb_channels);

	payload = GST_RTP_BASE_PAYLOAD (rtpspeexpay);

	gst_rtp_base_payload_set_options (payload, "audio", FALSE, "SPEEX", rate);
	cstr = g_strdup_printf ("%d", nb_channels);
	res = gst_rtp_base_payload_set_outcaps (payload, "encoding-params",
	G_TYPE_STRING, cstr, NULL);
	g_free (cstr);

	return res;

	/* ERRORS */
	too_small:
	{
	GST_DEBUG_OBJECT (rtpspeexpay,
	"ident packet too small, need at least 32 bytes");
	return FALSE;
	}
	wrong_header:
	{
	GST_DEBUG_OBJECT (rtpspeexpay,
	"ident packet does not start with \"Speex \"");
	return FALSE;
	}
	wrong_version:
	{
	GST_DEBUG_OBJECT (rtpspeexpay, "can only handle version 1, have version %d",
	version);
	return FALSE;
	}
	header_too_small:
	{
	GST_DEBUG_OBJECT (rtpspeexpay,
	"header size too small, need at least 80 bytes, " "got only %d",
	header_size);
	return FALSE;
	}
	payload_too_small:
	{
	GST_DEBUG_OBJECT (rtpspeexpay,
	"payload too small, need at least %d bytes, got only %d", header_size,
	size);
	return FALSE;
	}
	}

	static GstFlowReturn
	gst_rtp_speex_pay_handle_buffer (GstRTPBasePayload * basepayload,
	GstBuffer * buffer)
	{
	GstRtpSPEEXPay *rtpspeexpay;
	GstMapInfo map;
	GstBuffer *outbuf;
	GstClockTime timestamp, duration;
	GstFlowReturn ret;

	rtpspeexpay = GST_RTP_SPEEX_PAY (basepayload);

	gst_buffer_map (buffer, &map, GST_MAP_READ);

	switch (rtpspeexpay->packet) {
	case 0:
	/* ident packet. We need to parse the headers to construct the RTP
	* properties. */
	if (!gst_rtp_speex_pay_parse_ident (rtpspeexpay, map.data, map.size)) {
	gst_buffer_unmap (buffer, &map);
	goto parse_error;
	}

	ret = GST_FLOW_OK;
	gst_buffer_unmap (buffer, &map);
	goto done;
	case 1:
	/* comment packet, we ignore it */
	ret = GST_FLOW_OK;
	gst_buffer_unmap (buffer, &map);
	goto done;
	default:
	/* other packets go in the payload */
	break;
	}
	gst_buffer_unmap (buffer, &map);

	if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_GAP)) {
	ret = GST_FLOW_OK;
	goto done;
	}

	timestamp = GST_BUFFER_PTS (buffer);
	duration = GST_BUFFER_DURATION (buffer);

	/* FIXME, only one SPEEX frame per RTP packet for now */

	outbuf = gst_rtp_buffer_new_allocate (0, 0, 0);
	/* FIXME, assert for now */
	g_assert (gst_buffer_get_size (buffer) <=
	GST_RTP_BASE_PAYLOAD_MTU (rtpspeexpay));

	/* copy timestamp and duration */
	GST_BUFFER_PTS (outbuf) = timestamp;
	GST_BUFFER_DURATION (outbuf) = duration;

	gst_rtp_copy_audio_meta (basepayload, outbuf, buffer);
	outbuf = gst_buffer_append (outbuf, buffer);
	buffer = NULL;

	ret = gst_rtp_base_payload_push (basepayload, outbuf);

	done:
	if (buffer)
	gst_buffer_unref (buffer);

	rtpspeexpay->packet++;

	return ret;

	/* ERRORS */
	parse_error:
	{
	GST_ELEMENT_ERROR (rtpspeexpay, STREAM, DECODE, (NULL),
	("Error parsing first identification packet."));
	gst_buffer_unref (buffer);
	return GST_FLOW_ERROR;
	}
	}

	static GstStateChangeReturn
	gst_rtp_speex_pay_change_state (GstElement * element, GstStateChange transition)
	{
	GstRtpSPEEXPay *rtpspeexpay;
	GstStateChangeReturn ret;

	rtpspeexpay = GST_RTP_SPEEX_PAY (element);

	switch (transition) {
	case GST_STATE_CHANGE_NULL_TO_READY:
	break;
	case GST_STATE_CHANGE_READY_TO_PAUSED:
	rtpspeexpay->packet = 0;
	break;
	default:
	break;
	}

	ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);

	switch (transition) {
	case GST_STATE_CHANGE_READY_TO_NULL:
	break;
	default:
	break;
	}
	return ret;
	}

	gboolean
	gst_rtp_speex_pay_plugin_init (GstPlugin * plugin)
	{
	return gst_element_register (plugin, "rtpspeexpay",
	GST_RANK_SECONDARY, GST_TYPE_RTP_SPEEX_PAY);
	}