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rtmpsrc: Major cleanup and reorganization

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This commit is contained in:
Sebastian Dröge 2010-06-04 22:04:53 +02:00
parent 547f037ea4
commit c3d10ed72a
2 changed files with 127 additions and 315 deletions
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428
ext/rtmp/gstrtmpsrc.c
View file

@ -1,9 +1,10 @@
/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2000 Wim Taymans <wtay@chello.be>
* 2001 Bastien Nocera <hadess@hadess.net>
* 2002 Kristian Rietveld <kris@gtk.org>
* 2002,2003 Colin Walters <walters@gnu.org>
* 2001,2010 Bastien Nocera <hadess@hadess.net>
* 2010 Sebastian Dröge <sebastian.droege@collabora.co.uk>
*
* rtmpsrc.c:
*
@ -28,38 +29,16 @@
*
* This plugin reads data from a local or remote location specified
* by an URI. This location can be specified using any protocol supported by
* the RTMP library. Common protocols are 'file', 'http', 'ftp', or 'smb'.
*
* In case the #GstRTMPSrc:iradio-mode property is set and the
* location is a http resource, rtmpsrc will send special icecast http
* headers to the server to request additional icecast metainformation. If
* the server is not an icecast server, it will display the same behaviour
* as if the #GstRTMPSrc:iradio-mode property was not set. However,
* if the server is in fact an icecast server, rtmpsrc will output
* data with a media type of application/x-icy, in which case you will
* need to use the #GstICYDemux element as follow-up element to extract
* the icecast meta data and to determine the underlying media type.
* the RTMP library, i.e. rtmp, rtmpt, rtmps, rtmpe, rtmfp, rtmpte and rtmpts.
*
* <refsect2>
* <title>Example launch lines</title>
* |[
* gst-launch -v rtmpsrc location=file:///home/joe/foo.xyz ! fakesink
* ]| The above pipeline will simply read a local file and do nothing with the
* data read. Instead of rtmpsrc, we could just as well have used the
* filesrc element here.
* |[
* gst-launch -v rtmpsrc location=smb://othercomputer/foo.xyz ! filesink location=/home/joe/foo.xyz
* ]| The above pipeline will copy a file from a remote host to the local file
* system using the Samba protocol.
* |[
* gst-launch -v rtmpsrc location=http://music.foobar.com/demo.mp3 ! mad ! audioconvert ! audioresample ! alsasink
* ]| The above pipeline will read and decode and play an mp3 file from a
* web server using the http protocol.
* gst-launch -v rtmpsrc location=rtmp://somehost/someurl ! fakesink
* ]| Open an RTMP location and pass its content to fakesink.
* </refsect2>
*/
#define DEFAULT_RTMP_PORT 1935
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
@ -70,21 +49,9 @@
#include <stdio.h>
#include <stdlib.h>
#include <sys/types.h>
#include <sys/socket.h>
#include <sys/time.h>
#include <netinet/in.h>
#include <arpa/inet.h>
#include <netdb.h>
#include <sys/stat.h>
#include <fcntl.h>
#include <unistd.h>
#include <sys/mman.h>
#include <errno.h>
#include <string.h>
#include <gst/gst.h>
#include <gst/tag/tag.h>
GST_DEBUG_CATEGORY_STATIC (rtmpsrc_debug);
#define GST_CAT_DEFAULT rtmpsrc_debug
@ -96,14 +63,10 @@ static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
enum
{
ARG_0,
ARG_LOCATION,
PROP_0,
PROP_LOCATION,
};
static void gst_rtmp_src_base_init (gpointer g_class);
static void gst_rtmp_src_class_init (GstRTMPSrcClass * klass);
static void gst_rtmp_src_init (GstRTMPSrc * rtmpsrc);
static void gst_rtmp_src_finalize (GObject * object);
static void gst_rtmp_src_uri_handler_init (gpointer g_iface,
gpointer iface_data);
@ -111,66 +74,29 @@ static void gst_rtmp_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_rtmp_src_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_rtmp_src_finalize (GObject * object);
static gboolean gst_rtmp_src_stop (GstBaseSrc * src);
static gboolean gst_rtmp_src_start (GstBaseSrc * src);
static gboolean gst_rtmp_src_is_seekable (GstBaseSrc * src);
#if 0
static gboolean gst_rtmp_src_check_get_range (GstBaseSrc * src);
static gboolean gst_rtmp_src_get_size (GstBaseSrc * src, guint64 * size);
#endif
static GstFlowReturn gst_rtmp_src_create (GstBaseSrc * basesrc,
guint64 offset, guint size, GstBuffer ** buffer);
#if 0
static GstFlowReturn gst_rtmp_src_create (GstPushSrc * pushsrc,
GstBuffer ** buffer);
static gboolean gst_rtmp_src_query (GstBaseSrc * src, GstQuery * query);
#endif
static GstElementClass *parent_class = NULL;
static gboolean
plugin_init (GstPlugin * plugin)
static void
_do_init (GType gtype)
{
return gst_element_register (plugin, "rtmpsrc", GST_RANK_NONE,
GST_TYPE_RTMP_SRC);
static const GInterfaceInfo urihandler_info = {
gst_rtmp_src_uri_handler_init,
NULL,
NULL
};
g_type_add_interface_static (gtype, GST_TYPE_URI_HANDLER, &urihandler_info);
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"rtmpsrc",
"flvstreamer sources",
plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);
GType
gst_rtmp_src_get_type (void)
{
static GType rtmpsrc_type = 0;
if (!rtmpsrc_type) {
static const GTypeInfo rtmpsrc_info = {
sizeof (GstRTMPSrcClass),
gst_rtmp_src_base_init,
NULL,
(GClassInitFunc) gst_rtmp_src_class_init,
NULL,
NULL,
sizeof (GstRTMPSrc),
0,
(GInstanceInitFunc) gst_rtmp_src_init,
};
static const GInterfaceInfo urihandler_info = {
gst_rtmp_src_uri_handler_init,
NULL,
NULL
};
rtmpsrc_type =
g_type_register_static (GST_TYPE_BASE_SRC,
"GstRTMPSrc", &rtmpsrc_info, (GTypeFlags) 0);
g_type_add_interface_static (rtmpsrc_type, GST_TYPE_URI_HANDLER,
&urihandler_info);
}
return rtmpsrc_type;
}
GST_BOILERPLATE_FULL (GstRTMPSrc, gst_rtmp_src, GstPushSrc, GST_TYPE_PUSH_SRC,
_do_init);
static void
gst_rtmp_src_base_init (gpointer g_class)
@ -184,10 +110,8 @@ gst_rtmp_src_base_init (gpointer g_class)
"RTMP Source",
"Source/File",
"Read RTMP streams",
"Bastien Nocera <hadess@hadess.net>\n"
"GStreamer maintainers <gstreamer-devel@lists.sourceforge.net>");
GST_DEBUG_CATEGORY_INIT (rtmpsrc_debug, "rtmpsrc", 0, "RTMP Source");
"Bastien Nocera <hadess@hadess.net>, "
"Sebastian Dröge <sebastian.droege@collabora.co.uk>");
}
static void
@ -195,11 +119,11 @@ gst_rtmp_src_class_init (GstRTMPSrcClass * klass)
{
GObjectClass *gobject_class;
GstBaseSrcClass *gstbasesrc_class;
GstPushSrcClass *gstpushsrc_class;
gobject_class = G_OBJECT_CLASS (klass);
gstbasesrc_class = GST_BASE_SRC_CLASS (klass);
parent_class = (GstElementClass *) g_type_class_peek_parent (klass);
gstpushsrc_class = GST_PUSH_SRC_CLASS (klass);
gobject_class->finalize = gst_rtmp_src_finalize;
gobject_class->set_property = gst_rtmp_src_set_property;
@ -207,29 +131,19 @@ gst_rtmp_src_class_init (GstRTMPSrcClass * klass)
/* properties */
gst_element_class_install_std_props (GST_ELEMENT_CLASS (klass),
"location", ARG_LOCATION, G_PARAM_READWRITE, NULL);
"location", PROP_LOCATION, G_PARAM_READWRITE, NULL);
gstbasesrc_class->start = GST_DEBUG_FUNCPTR (gst_rtmp_src_start);
gstbasesrc_class->stop = GST_DEBUG_FUNCPTR (gst_rtmp_src_stop);
#if 0
gstbasesrc_class->get_size = GST_DEBUG_FUNCPTR (gst_rtmp_src_get_size);
#endif
gstbasesrc_class->is_seekable = GST_DEBUG_FUNCPTR (gst_rtmp_src_is_seekable);
#if 0
gstbasesrc_class->check_get_range =
GST_DEBUG_FUNCPTR (gst_rtmp_src_check_get_range);
#endif
gstbasesrc_class->create = GST_DEBUG_FUNCPTR (gst_rtmp_src_create);
#if 0
gstpushsrc_class->create = GST_DEBUG_FUNCPTR (gst_rtmp_src_create);
gstbasesrc_class->query = GST_DEBUG_FUNCPTR (gst_rtmp_src_query);
#endif
}
static void
gst_rtmp_src_init (GstRTMPSrc * rtmpsrc)
gst_rtmp_src_init (GstRTMPSrc * rtmpsrc, GstRTMPSrcClass * klass)
{
rtmpsrc->curoffset = 0;
rtmpsrc->seekable = FALSE;
}
static void
@ -262,7 +176,10 @@ gst_rtmp_src_uri_get_type (void)
static gchar **
gst_rtmp_src_uri_get_protocols (void)
{
static gchar *protocols[] = { (char *) "rtmp", NULL };
static gchar *protocols[] =
{ (char *) "rtmp", (char *) "rtmpt", (char *) "rtmps", (char *) "rtmpe",
(char *) "rtmfp", (char *) "rtmpte", (char *) "rtmpts", NULL
};
return protocols;
}
@ -278,13 +195,40 @@ static gboolean
gst_rtmp_src_uri_set_uri (GstURIHandler * handler, const gchar * uri)
{
GstRTMPSrc *src = GST_RTMP_SRC (handler);
gchar *new_location;
if (GST_STATE (src) == GST_STATE_PLAYING ||
GST_STATE (src) == GST_STATE_PAUSED)
if (GST_STATE (src) >= GST_STATE_PAUSED)
return FALSE;
g_object_set (G_OBJECT (src), "location", uri, NULL);
g_message ("just set uri to %s", uri);
g_free (src->uri);
src->uri = NULL;
if (src->rtmp) {
RTMP_Close (src->rtmp);
RTMP_Free (src->rtmp);
src->rtmp = NULL;
}
if (uri != NULL) {
new_location = g_strdup (uri);
src->rtmp = RTMP_Alloc ();
RTMP_Init (src->rtmp);
if (!RTMP_SetupURL (src->rtmp, new_location)) {
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, NULL,
("Failed to setup URL '%s'", src->uri));
g_free (new_location);
RTMP_Free (src->rtmp);
src->rtmp = NULL;
return FALSE;
} else {
src->uri = g_strdup (uri);
GST_DEBUG_OBJECT (src, "parsed uri '%s' properly", src->uri);
}
}
GST_DEBUG_OBJECT (src, "Changed URI to %s", GST_STR_NULL (uri));
return TRUE;
}
@ -309,36 +253,9 @@ gst_rtmp_src_set_property (GObject * object, guint prop_id,
src = GST_RTMP_SRC (object);
switch (prop_id) {
case ARG_LOCATION:{
char *new_location;
/* the element must be stopped or paused in order to do this */
if (GST_STATE (src) == GST_STATE_PLAYING ||
GST_STATE (src) == GST_STATE_PAUSED)
break;
g_free (src->uri);
src->uri = NULL;
if (src->rtmp) {
RTMP_Close (src->rtmp);
RTMP_Free (src->rtmp);
src->rtmp = NULL;
}
new_location = g_value_dup_string (value);
src->rtmp = RTMP_Alloc ();
RTMP_Init (src->rtmp);
if (!RTMP_SetupURL (src->rtmp, new_location)) {
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, NULL,
("Failed to setup URL '%s'", src->uri));
g_free (new_location);
RTMP_Free (src->rtmp);
src->rtmp = NULL;
} else {
src->uri = g_value_dup_string (value);
g_message ("parsed uri '%s' properly", src->uri);
}
case PROP_LOCATION:{
gst_rtmp_src_uri_set_uri (GST_URI_HANDLER (src),
g_value_get_string (value));
break;
}
default:
@ -356,7 +273,7 @@ gst_rtmp_src_get_property (GObject * object, guint prop_id, GValue * value,
src = GST_RTMP_SRC (object);
switch (prop_id) {
case ARG_LOCATION:
case PROP_LOCATION:
g_value_set_string (value, src->uri);
break;
default:
@ -370,50 +287,26 @@ gst_rtmp_src_get_property (GObject * object, guint prop_id, GValue * value,
* and seeking and such.
*/
static GstFlowReturn
gst_rtmp_src_create (GstBaseSrc * basesrc, guint64 offset, guint size,
GstBuffer ** buffer)
gst_rtmp_src_create (GstPushSrc * pushsrc, GstBuffer ** buffer)
{
GstRTMPSrc *src;
GstBuffer *buf;
guint8 *data;
guint todo;
int read;
int size;
src = GST_RTMP_SRC (basesrc);
src = GST_RTMP_SRC (pushsrc);
g_return_val_if_fail (src->rtmp != NULL, GST_FLOW_ERROR);
GST_DEBUG ("now at %" G_GINT64_FORMAT ", reading from %" G_GUINT64_FORMAT
", size %u", src->curoffset, offset, size);
size = GST_BASE_SRC_CAST (pushsrc)->blocksize;
/* open if required */
if (G_UNLIKELY (!RTMP_IsConnected (src->rtmp))) {
if (!RTMP_Connect (src->rtmp, NULL)) {
GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL),
("Could not connect to RTMP stream \"%s\" for reading: %s (%d)",
src->uri, "FIXME", 0));
return GST_FLOW_ERROR;
}
}
/* seek if required */
if (G_UNLIKELY (src->curoffset != offset)) {
GST_DEBUG ("need to seek");
if (src->seekable) {
#if 0
GST_DEBUG ("seeking to %" G_GUINT64_FORMAT, offset);
res = rtmp_seek (src->handle, RTMP_SEEK_START, offset);
if (res != RTMP_OK)
goto seek_failed;
src->curoffset = offset;
#endif
} else {
goto cannot_seek;
}
}
GST_DEBUG ("reading from %" G_GUINT64_FORMAT
", size %u", src->curoffset, size);
buf = gst_buffer_try_new_and_alloc (size);
if (G_UNLIKELY (buf == NULL && size == 0)) {
if (G_UNLIKELY (buf == NULL)) {
GST_ERROR_OBJECT (src, "Failed to allocate %u bytes", size);
return GST_FLOW_ERROR;
}
@ -425,16 +318,14 @@ gst_rtmp_src_create (GstBaseSrc * basesrc, guint64 offset, guint size,
todo = size;
while (todo > 0) {
read = RTMP_Read (src->rtmp, (char *) &data, todo);
read = RTMP_Read (src->rtmp, (char *) data, todo);
if (G_UNLIKELY (read == -1))
if (G_UNLIKELY (read == 0))
goto eos;
if (G_UNLIKELY (read == -2))
if (G_UNLIKELY (read == -1))
goto read_failed;
/* FIXME handle -3 ? */
if (read < todo) {
data = &data[read];
todo -= read;
@ -449,30 +340,8 @@ gst_rtmp_src_create (GstBaseSrc * basesrc, guint64 offset, guint size,
/* we're done, return the buffer */
*buffer = buf;
#if 0
RTMPFileSize readbytes;
guint todo;
return GST_FLOW_OK;
#endif
return GST_FLOW_OK;
//seek_failed:
{
GST_ELEMENT_ERROR (src, RESOURCE, SEEK, (NULL),
("Failed to seek to requested position %" G_GINT64_FORMAT ": %s",
offset, "FIXME"));
return GST_FLOW_ERROR;
}
cannot_seek:
{
GST_ELEMENT_ERROR (src, RESOURCE, SEEK, (NULL),
("Requested seek from %" G_GINT64_FORMAT " to %" G_GINT64_FORMAT
" on non-seekable stream", src->curoffset, offset));
return GST_FLOW_ERROR;
}
read_failed:
{
gst_buffer_unref (buf);
@ -488,7 +357,6 @@ eos:
}
}
#if 0
static gboolean
gst_rtmp_src_query (GstBaseSrc * basesrc, GstQuery * query)
{
@ -500,6 +368,20 @@ gst_rtmp_src_query (GstBaseSrc * basesrc, GstQuery * query)
gst_query_set_uri (query, src->uri);
ret = TRUE;
break;
case GST_QUERY_DURATION:{
GstFormat format;
gdouble duration;
gst_query_parse_duration (query, &format, NULL);
if (format == GST_FORMAT_TIME && src->rtmp) {
duration = RTMP_GetDuration (src->rtmp);
if (duration != 0.0) {
gst_query_set_duration (query, format, duration * GST_SECOND);
ret = TRUE;
}
}
break;
}
default:
ret = FALSE;
break;
@ -510,7 +392,7 @@ gst_rtmp_src_query (GstBaseSrc * basesrc, GstQuery * query)
return ret;
}
#endif
static gboolean
gst_rtmp_src_is_seekable (GstBaseSrc * basesrc)
{
@ -518,100 +400,9 @@ gst_rtmp_src_is_seekable (GstBaseSrc * basesrc)
src = GST_RTMP_SRC (basesrc);
return src->seekable;
return FALSE;
}
#if 0
static gboolean
gst_rtmp_src_check_get_range (GstBaseSrc * basesrc)
{
GstRTMPSrc *src;
const gchar *protocol;
src = GST_RTMP_SRC (basesrc);
if (src->uri == NULL) {
GST_WARNING_OBJECT (src, "no URI set yet");
return FALSE;
}
if (rtmp_uri_is_local (src->uri)) {
GST_LOG_OBJECT (src, "local URI (%s), assuming random access is possible",
GST_STR_NULL (src->uri_name));
return TRUE;
}
/* blacklist certain protocols we know won't work getrange-based */
protocol = rtmp_uri_get_scheme (src->uri);
if (protocol == NULL)
goto undecided;
if (strcmp (protocol, "http") == 0 || strcmp (protocol, "https") == 0) {
GST_LOG_OBJECT (src, "blacklisted protocol '%s', no random access possible"
" (URI=%s)", protocol, GST_STR_NULL (src->uri_name));
return FALSE;
}
/* fall through to undecided */
undecided:
{
/* don't know what to do, let the basesrc class decide for us */
GST_LOG_OBJECT (src, "undecided about URI '%s', let base class handle it",
GST_STR_NULL (src->uri_name));
if (GST_BASE_SRC_CLASS (parent_class)->check_get_range)
return GST_BASE_SRC_CLASS (parent_class)->check_get_range (basesrc);
return FALSE;
}
}
#endif
#if 0
static gboolean
gst_rtmp_src_get_size (GstBaseSrc * basesrc, guint64 * size)
{
GstRTMPSrc *src;
RTMPFileInfo *info;
RTMPFileInfoOptions options;
RTMPResult res;
src = GST_RTMP_SRC (basesrc);
*size = -1;
info = rtmp_file_info_new ();
options = RTMP_FILE_INFO_DEFAULT | RTMP_FILE_INFO_FOLLOW_LINKS;
res = rtmp_get_file_info_from_handle (src->handle, info, options);
if (res == RTMP_OK) {
if ((info->valid_fields & RTMP_FILE_INFO_FIELDS_SIZE) != 0) {
*size = info->size;
GST_DEBUG_OBJECT (src, "from handle: %" G_GUINT64_FORMAT " bytes", *size);
} else if (src->own_handle && rtmp_uri_is_local (src->uri)) {
GST_DEBUG_OBJECT (src,
"file size not known, file local, trying fallback");
res = rtmp_get_file_info_uri (src->uri, info, options);
if (res == RTMP_OK &&
(info->valid_fields & RTMP_FILE_INFO_FIELDS_SIZE) != 0) {
*size = info->size;
GST_DEBUG_OBJECT (src, "from uri: %" G_GUINT64_FORMAT " bytes", *size);
}
}
} else {
GST_WARNING_OBJECT (src, "getting info failed: %s",
rtmp_result_to_string (res));
}
rtmp_file_info_unref (info);
if (*size == (RTMPFileSize) - 1)
return FALSE;
GST_DEBUG_OBJECT (src, "return size %" G_GUINT64_FORMAT, *size);
return TRUE;
}
#endif
/* open the file, do stuff necessary to go to PAUSED state */
static gboolean
gst_rtmp_src_start (GstBaseSrc * basesrc)
@ -620,13 +411,23 @@ gst_rtmp_src_start (GstBaseSrc * basesrc)
src = GST_RTMP_SRC (basesrc);
g_message ("start called!");
if (!src->uri) {
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL), ("No filename given"));
return FALSE;
}
src->curoffset = 0;
/* open if required */
if (!RTMP_IsConnected (src->rtmp)) {
if (!RTMP_Connect (src->rtmp, NULL)) {
GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, (NULL),
("Could not connect to RTMP stream \"%s\" for reading: %s (%d)",
src->uri, "FIXME", 0));
return FALSE;
}
}
return TRUE;
}
@ -640,13 +441,22 @@ gst_rtmp_src_stop (GstBaseSrc * basesrc)
//FIXME you can't run RTMP_Close multiple times
// RTMP_Close (src->rtmp);
g_message ("stop called!");
src->curoffset = 0;
return TRUE;
}
/*
* vim: sw=2 ts=8 cindent noai bs=2
*/
static gboolean
plugin_init (GstPlugin * plugin)
{
GST_DEBUG_CATEGORY_INIT (rtmpsrc_debug, "rtmpsrc", 0, "RTMP Source");
return gst_element_register (plugin, "rtmpsrc", GST_RANK_PRIMARY,
GST_TYPE_RTMP_SRC);
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"rtmpsrc",
"RTMP source",
plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);

14
ext/rtmp/gstrtmpsrc.h
View file

@ -1,9 +1,10 @@
/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2000 Wim Taymans <wtay@chello.be>
* 2001 Bastien Nocera <hadess@hadess.net>
* 2002 Kristian Rietveld <kris@gtk.org>
* 2002,2003 Colin Walters <walters@gnu.org>
* 2001,2010 Bastien Nocera <hadess@hadess.net>
* 2010 Sebastian Dröge <sebastian.droege@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
@ -25,6 +26,7 @@
#define __GST_RTMP_SRC_H__
#include <gst/base/gstbasesrc.h>
#include <gst/base/gstpushsrc.h>
#include <librtmp/rtmp.h>
#include <librtmp/log.h>
@ -53,19 +55,19 @@ typedef struct _GstRTMPSrcClass GstRTMPSrcClass;
*/
struct _GstRTMPSrc
{
GstBaseSrc basesrc;
char *uri;
GstPushSrc parent;
/* < private > */
gchar *uri;
RTMP *rtmp;
gboolean seekable;
gint64 curoffset;
};
struct _GstRTMPSrcClass
{
GstBaseSrcClass basesrc_class;
GstPushSrcClass parent;
};
GType gst_rtmp_src_get_type (void);