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rtp: import rtpsbcpay from bluez and port to 1.0

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Compiles, but not tested yet (sbc elements still need to be ported).

https://bugzilla.gnome.org/show_bug.cgi?id=690582
This commit is contained in:
Tim-Philipp Müller 2013-01-10 12:38:13 +00:00
parent 4196feb659
commit 39ef892938
4 changed files with 59 additions and 63 deletions
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2
gst/rtp/Makefile.am
View file

@ -62,6 +62,7 @@ libgstrtp_la_SOURCES = \
gstrtpmp4apay.c \
gstrtpqcelpdepay.c \
gstrtpqdmdepay.c \
gstrtpsbcpay.c \
gstrtpsirenpay.c \
gstrtpsirendepay.c \
gstrtpspeexdepay.c \
@ -150,6 +151,7 @@ noinst_HEADERS = \
gstasteriskh263.h \
gstrtpqcelpdepay.h \
gstrtpqdmdepay.h \
gstrtpsbcpay.h \
gstrtpsirenpay.h \
gstrtpsirendepay.h \
gstrtpspeexdepay.h \

4
gst/rtp/gstrtp.c
View file

@ -78,6 +78,7 @@
#include "gstrtpmp4gpay.h"
#include "gstrtpqcelpdepay.h"
#include "gstrtpqdmdepay.h"
#include "gstrtpsbcpay.h"
#include "gstrtpsirenpay.h"
#include "gstrtpsirendepay.h"
#include "gstrtpspeexpay.h"
@ -266,6 +267,9 @@ plugin_init (GstPlugin * plugin)
if (!gst_rtp_qdm2_depay_plugin_init (plugin))
return FALSE;
if (!gst_rtp_sbc_pay_plugin_init (plugin))
return FALSE;
if (!gst_rtp_siren_pay_plugin_init (plugin))
return FALSE;

106
gst/rtp/gstrtpsbcpay.c
View file

@ -1,10 +1,8 @@
/*
*
/* GStreamer RTP SBC payloader
* BlueZ - Bluetooth protocol stack for Linux
*
* Copyright (C) 2004-2010 Marcel Holtmann <marcel@holtmann.org>
*
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
@ -25,7 +23,6 @@
#include <config.h>
#endif
#include "gstpragma.h"
#include "gstrtpsbcpay.h"
#include <math.h>
#include <string.h>
@ -34,8 +31,9 @@
#define DEFAULT_MIN_FRAMES 0
#define RTP_SBC_HEADER_TOTAL (12 + RTP_SBC_PAYLOAD_HEADER_SIZE)
#if __BYTE_ORDER == __LITTLE_ENDIAN
#if G_BYTE_ORDER == G_LITTLE_ENDIAN
/* FIXME: this seems all a bit over the top for a single byte.. */
struct rtp_payload
{
guint8 frame_count:4;
@ -45,7 +43,7 @@ struct rtp_payload
guint8 is_fragmented:1;
} __attribute__ ((packed));
#elif __BYTE_ORDER == __BIG_ENDIAN
#elif G_BYTE_ORDER == G_BIG_ENDIAN
struct rtp_payload
{
@ -69,34 +67,27 @@ enum
GST_DEBUG_CATEGORY_STATIC (gst_rtp_sbc_pay_debug);
#define GST_CAT_DEFAULT gst_rtp_sbc_pay_debug
GST_BOILERPLATE (GstRtpSBCPay, gst_rtp_sbc_pay, GstBaseRTPPayload,
GST_TYPE_BASE_RTP_PAYLOAD);
static const GstElementDetails gst_rtp_sbc_pay_details =
GST_ELEMENT_DETAILS ("RTP packet payloader",
"Codec/Payloader/Network",
"Payload SBC audio as RTP packets",
"Thiago Sousa Santos " "<thiagoss@lcc.ufcg.edu.br>");
#define parent_class gst_rtp_sbc_pay_parent_class
G_DEFINE_TYPE (GstRtpSBCPay, gst_rtp_sbc_pay, GST_TYPE_RTP_BASE_PAYLOAD);
static GstStaticPadTemplate gst_rtp_sbc_pay_sink_factory =
GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, 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\" }, "
"mode = (string) { mono, dual, stereo, joint }, "
"blocks = (int) { 4, 8, 12, 16 }, "
"subbands = (int) { 4, 8 }, "
"allocation = (string) { \"snr\", \"loudness\" }, "
"bitpool = (int) [ 2, 64 ]")
"allocation = (string) { snr, loudness }, " "bitpool = (int) [ 2, 64 ]")
);
static GstStaticPadTemplate gst_rtp_sbc_pay_src_factory =
GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/x-rtp, "
"media = (string) \"audio\","
"media = (string) audio,"
"payload = (int) " GST_RTP_PAYLOAD_DYNAMIC_STRING ", "
"clock-rate = (int) { 16000, 32000, 44100, 48000 },"
"encoding-name = (string) \"SBC\"")
"encoding-name = (string) SBC")
);
static void gst_rtp_sbc_pay_set_property (GObject * object, guint prop_id,
@ -124,7 +115,7 @@ gst_rtp_sbc_pay_get_frame_len (gint subbands, gint channels,
}
static gboolean
gst_rtp_sbc_pay_set_caps (GstBaseRTPPayload * payload, GstCaps * caps)
gst_rtp_sbc_pay_set_caps (GstRTPBasePayload * payload, GstCaps * caps)
{
GstRtpSBCPay *sbcpay;
gint rate, subbands, channels, blocks, bitpool;
@ -155,16 +146,17 @@ gst_rtp_sbc_pay_set_caps (GstBaseRTPPayload * payload, GstCaps * caps)
sbcpay->frame_length = frame_len;
gst_basertppayload_set_options (payload, "audio", TRUE, "SBC", rate);
gst_rtp_base_payload_set_options (payload, "audio", TRUE, "SBC", rate);
GST_DEBUG_OBJECT (payload, "calculated frame length: %d ", frame_len);
return gst_basertppayload_set_outcaps (payload, NULL);
return gst_rtp_base_payload_set_outcaps (payload, NULL);
}
static GstFlowReturn
gst_rtp_sbc_pay_flush_buffers (GstRtpSBCPay * sbcpay)
{
GstRTPBuffer rtp;
guint available;
guint max_payload;
GstBuffer *outbuf;
@ -181,7 +173,7 @@ gst_rtp_sbc_pay_flush_buffers (GstRtpSBCPay * sbcpay)
available = gst_adapter_available (sbcpay->adapter);
max_payload =
gst_rtp_buffer_calc_payload_len (GST_BASE_RTP_PAYLOAD_MTU (sbcpay) -
gst_rtp_buffer_calc_payload_len (GST_RTP_BASE_PAYLOAD_MTU (sbcpay) -
RTP_SBC_PAYLOAD_HEADER_SIZE, 0, 0);
max_payload = MIN (max_payload, available);
@ -193,25 +185,33 @@ gst_rtp_sbc_pay_flush_buffers (GstRtpSBCPay * sbcpay)
outbuf = gst_rtp_buffer_new_allocate (payload_length +
RTP_SBC_PAYLOAD_HEADER_SIZE, 0, 0);
gst_rtp_buffer_set_payload_type (outbuf, GST_BASE_RTP_PAYLOAD_PT (sbcpay));
/* get payload */
gst_rtp_buffer_map (outbuf, GST_MAP_WRITE, &rtp);
payload_data = gst_rtp_buffer_get_payload (outbuf);
gst_rtp_buffer_set_payload_type (&rtp, GST_RTP_BASE_PAYLOAD_PT (sbcpay));
/* write header and copy data into payload */
payload_data = gst_rtp_buffer_get_payload (&rtp);
payload = (struct rtp_payload *) payload_data;
memset (payload, 0, sizeof (struct rtp_payload));
payload->frame_count = frame_count;
gst_adapter_copy (sbcpay->adapter, payload_data +
RTP_SBC_PAYLOAD_HEADER_SIZE, 0, payload_length);
gst_rtp_buffer_unmap (&rtp);
gst_adapter_flush (sbcpay->adapter, payload_length);
/* FIXME: what about duration? */
GST_BUFFER_TIMESTAMP (outbuf) = sbcpay->timestamp;
GST_DEBUG_OBJECT (sbcpay, "Pushing %d bytes", payload_length);
return gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (sbcpay), outbuf);
return gst_rtp_base_payload_push (GST_RTP_BASE_PAYLOAD (sbcpay), outbuf);
}
static GstFlowReturn
gst_rtp_sbc_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buffer)
gst_rtp_sbc_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buffer)
{
GstRtpSBCPay *sbcpay;
guint available;
@ -225,7 +225,7 @@ gst_rtp_sbc_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buffer)
available = gst_adapter_available (sbcpay->adapter);
if (available + RTP_SBC_HEADER_TOTAL >=
GST_BASE_RTP_PAYLOAD_MTU (sbcpay) ||
GST_RTP_BASE_PAYLOAD_MTU (sbcpay) ||
(available > (sbcpay->min_frames * sbcpay->frame_length)))
return gst_rtp_sbc_pay_flush_buffers (sbcpay);
@ -233,9 +233,9 @@ gst_rtp_sbc_pay_handle_buffer (GstBaseRTPPayload * payload, GstBuffer * buffer)
}
static gboolean
gst_rtp_sbc_pay_handle_event (GstPad * pad, GstEvent * event)
gst_rtp_sbc_pay_sink_event (GstRTPBasePayload * payload, GstEvent * event)
{
GstRtpSBCPay *sbcpay = GST_RTP_SBC_PAY (GST_PAD_PARENT (pad));
GstRtpSBCPay *sbcpay = GST_RTP_SBC_PAY (payload);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
@ -245,26 +245,14 @@ gst_rtp_sbc_pay_handle_event (GstPad * pad, GstEvent * event)
break;
}
return FALSE;
}
static void
gst_rtp_sbc_pay_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_sbc_pay_sink_factory));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_sbc_pay_src_factory));
gst_element_class_set_details (element_class, &gst_rtp_sbc_pay_details);
return GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->sink_event (payload, event);
}
static void
gst_rtp_sbc_pay_finalize (GObject * object)
{
GstRtpSBCPay *sbcpay = GST_RTP_SBC_PAY (object);
g_object_unref (sbcpay->adapter);
GST_CALL_PARENT (G_OBJECT_CLASS, finalize, (object));
@ -273,23 +261,18 @@ gst_rtp_sbc_pay_finalize (GObject * object)
static void
gst_rtp_sbc_pay_class_init (GstRtpSBCPayClass * klass)
{
GObjectClass *gobject_class;
GstBaseRTPPayloadClass *payload_class = GST_BASE_RTP_PAYLOAD_CLASS (klass);
GstRTPBasePayloadClass *payload_class = GST_RTP_BASE_PAYLOAD_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
gobject_class = G_OBJECT_CLASS (klass);
parent_class = g_type_class_peek_parent (klass);
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_rtp_sbc_pay_finalize);
gobject_class->set_property =
GST_DEBUG_FUNCPTR (gst_rtp_sbc_pay_set_property);
gobject_class->get_property =
GST_DEBUG_FUNCPTR (gst_rtp_sbc_pay_get_property);
gobject_class->finalize = gst_rtp_sbc_pay_finalize;
gobject_class->set_property = gst_rtp_sbc_pay_set_property;
gobject_class->get_property = gst_rtp_sbc_pay_get_property;
payload_class->set_caps = GST_DEBUG_FUNCPTR (gst_rtp_sbc_pay_set_caps);
payload_class->handle_buffer =
GST_DEBUG_FUNCPTR (gst_rtp_sbc_pay_handle_buffer);
payload_class->handle_event =
GST_DEBUG_FUNCPTR (gst_rtp_sbc_pay_handle_event);
payload_class->sink_event = GST_DEBUG_FUNCPTR (gst_rtp_sbc_pay_sink_event);
/* properties */
g_object_class_install_property (G_OBJECT_CLASS (klass),
@ -299,6 +282,15 @@ gst_rtp_sbc_pay_class_init (GstRtpSBCPayClass * klass)
"(-1 for maximum allowed by the mtu)",
-1, G_MAXINT, DEFAULT_MIN_FRAMES, G_PARAM_READWRITE));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_sbc_pay_sink_factory));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gst_rtp_sbc_pay_src_factory));
gst_element_class_set_static_metadata (element_class, "RTP packet payloader",
"Codec/Payloader/Network", "Payload SBC audio as RTP packets",
"Thiago Sousa Santos <thiagoss@lcc.ufcg.edu.br>");
GST_DEBUG_CATEGORY_INIT (gst_rtp_sbc_pay_debug, "rtpsbcpay", 0,
"RTP SBC payloader");
}
@ -340,7 +332,7 @@ gst_rtp_sbc_pay_get_property (GObject * object, guint prop_id,
}
static void
gst_rtp_sbc_pay_init (GstRtpSBCPay * self, GstRtpSBCPayClass * klass)
gst_rtp_sbc_pay_init (GstRtpSBCPay * self)
{
self->adapter = gst_adapter_new ();
self->frame_length = 0;

10
gst/rtp/gstrtpsbcpay.h
View file

@ -1,10 +1,8 @@
/*
*
/* GStreamer RTP SBC payloader
* BlueZ - Bluetooth protocol stack for Linux
*
* Copyright (C) 2004-2010 Marcel Holtmann <marcel@holtmann.org>
*
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
@ -22,8 +20,8 @@
*/
#include <gst/gst.h>
#include <gst/rtp/gstbasertppayload.h>
#include <gst/base/gstadapter.h>
#include <gst/rtp/gstrtpbasepayload.h>
#include <gst/rtp/gstrtpbuffer.h>
G_BEGIN_DECLS
@ -45,7 +43,7 @@ typedef struct _GstRtpSBCPay GstRtpSBCPay;
typedef struct _GstRtpSBCPayClass GstRtpSBCPayClass;
struct _GstRtpSBCPay {
GstBaseRTPPayload base;
GstRTPBasePayload base;
GstAdapter *adapter;
GstClockTime timestamp;
@ -56,7 +54,7 @@ struct _GstRtpSBCPay {
};
struct _GstRtpSBCPayClass {
GstBaseRTPPayloadClass parent_class;
GstRTPBasePayloadClass parent_class;
};
GType gst_rtp_sbc_pay_get_type(void);