gstreamer/ext/rtmp/gstrtmpsrc.c

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/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2000 Wim Taymans <wtay@chello.be>
* 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:
*
* 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.
*/
/**
* SECTION:element-rtmpsrc
* @title: rtmpsrc
*
* 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, i.e. rtmp, rtmpt, rtmps, rtmpe, rtmfp, rtmpte and rtmpts.
*
* ## Example launch lines
* |[
* gst-launch-1.0 -v rtmpsrc location=rtmp://somehost/someurl ! fakesink
* ]| Open an RTMP location and pass its content to fakesink.
*
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst-i18n-plugin.h>
#include "gstrtmpelements.h"
#include "gstrtmpsrc.h"
#include <stdio.h>
#include <stdlib.h>
#include <string.h>
#include <gst/gst.h>
#ifdef G_OS_WIN32
#include <winsock2.h>
#endif
GST_DEBUG_CATEGORY_STATIC (rtmpsrc_debug);
#define GST_CAT_DEFAULT rtmpsrc_debug
static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS_ANY);
enum
{
PROP_0,
PROP_LOCATION,
PROP_TIMEOUT
#if 0
PROP_SWF_URL,
PROP_PAGE_URL
#endif
};
#define DEFAULT_LOCATION NULL
#define DEFAULT_TIMEOUT 120
static void gst_rtmp_src_uri_handler_init (gpointer g_iface,
gpointer iface_data);
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_connect (GstRTMPSrc * src);
static gboolean gst_rtmp_src_unlock (GstBaseSrc * src);
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);
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static gboolean gst_rtmp_src_prepare_seek_segment (GstBaseSrc * src,
GstEvent * event, GstSegment * segment);
static gboolean gst_rtmp_src_do_seek (GstBaseSrc * src, GstSegment * segment);
static GstFlowReturn gst_rtmp_src_create (GstPushSrc * pushsrc,
GstBuffer ** buffer);
static gboolean gst_rtmp_src_query (GstBaseSrc * src, GstQuery * query);
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#define gst_rtmp_src_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstRTMPSrc, gst_rtmp_src, GST_TYPE_PUSH_SRC,
G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER,
gst_rtmp_src_uri_handler_init));
GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (rtmpsrc, "rtmpsrc", GST_RANK_PRIMARY,
GST_TYPE_RTMP_SRC, rtmp_element_init (plugin));
static void
gst_rtmp_src_class_init (GstRTMPSrcClass * klass)
{
GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
GstBaseSrcClass *gstbasesrc_class;
GstPushSrcClass *gstpushsrc_class;
gobject_class = G_OBJECT_CLASS (klass);
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gstelement_class = GST_ELEMENT_CLASS (klass);
gstbasesrc_class = GST_BASE_SRC_CLASS (klass);
gstpushsrc_class = GST_PUSH_SRC_CLASS (klass);
gobject_class->finalize = gst_rtmp_src_finalize;
gobject_class->set_property = gst_rtmp_src_set_property;
gobject_class->get_property = gst_rtmp_src_get_property;
/* properties */
g_object_class_install_property (gobject_class, PROP_LOCATION,
g_param_spec_string ("location", "RTMP Location",
"Location of the RTMP url to read",
DEFAULT_LOCATION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_TIMEOUT,
g_param_spec_int ("timeout", "RTMP Timeout",
"Time without receiving any data from the server to wait before to timeout the session",
0, G_MAXINT,
DEFAULT_TIMEOUT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_element_class_add_static_pad_template (gstelement_class, &srctemplate);
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gst_element_class_set_static_metadata (gstelement_class,
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"RTMP Source",
"Source/File",
"Read RTMP streams",
"Bastien Nocera <hadess@hadess.net>, "
"Sebastian Dröge <sebastian.droege@collabora.co.uk>");
gstbasesrc_class->start = GST_DEBUG_FUNCPTR (gst_rtmp_src_start);
gstbasesrc_class->stop = GST_DEBUG_FUNCPTR (gst_rtmp_src_stop);
gstbasesrc_class->unlock = GST_DEBUG_FUNCPTR (gst_rtmp_src_unlock);
gstbasesrc_class->is_seekable = GST_DEBUG_FUNCPTR (gst_rtmp_src_is_seekable);
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gstbasesrc_class->prepare_seek_segment =
GST_DEBUG_FUNCPTR (gst_rtmp_src_prepare_seek_segment);
gstbasesrc_class->do_seek = GST_DEBUG_FUNCPTR (gst_rtmp_src_do_seek);
gstpushsrc_class->create = GST_DEBUG_FUNCPTR (gst_rtmp_src_create);
gstbasesrc_class->query = GST_DEBUG_FUNCPTR (gst_rtmp_src_query);
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GST_DEBUG_CATEGORY_INIT (rtmpsrc_debug, "rtmpsrc", 0, "RTMP Source");
}
static void
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gst_rtmp_src_init (GstRTMPSrc * rtmpsrc)
{
#ifdef G_OS_WIN32
WSADATA wsa_data;
if (WSAStartup (MAKEWORD (2, 2), &wsa_data) != 0) {
GST_ERROR_OBJECT (rtmpsrc, "WSAStartup failed: 0x%08x", WSAGetLastError ());
}
#endif
rtmpsrc->cur_offset = 0;
rtmpsrc->last_timestamp = 0;
rtmpsrc->timeout = DEFAULT_TIMEOUT;
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gst_base_src_set_format (GST_BASE_SRC (rtmpsrc), GST_FORMAT_TIME);
}
static void
gst_rtmp_src_finalize (GObject * object)
{
GstRTMPSrc *rtmpsrc = GST_RTMP_SRC (object);
g_free (rtmpsrc->uri);
rtmpsrc->uri = NULL;
#ifdef G_OS_WIN32
WSACleanup ();
#endif
G_OBJECT_CLASS (parent_class)->finalize (object);
}
/*
* URI interface support.
*/
static GstURIType
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gst_rtmp_src_uri_get_type (GType type)
{
return GST_URI_SRC;
}
static const gchar *const *
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gst_rtmp_src_uri_get_protocols (GType type)
{
static const gchar *protocols[] =
{ "rtmp", "rtmpt", "rtmps", "rtmpe", "rtmfp", "rtmpte", "rtmpts", NULL };
return protocols;
}
static gchar *
gst_rtmp_src_uri_get_uri (GstURIHandler * handler)
{
GstRTMPSrc *src = GST_RTMP_SRC (handler);
/* FIXME: make thread-safe */
return g_strdup (src->uri);
}
static gboolean
gst_rtmp_src_uri_set_uri (GstURIHandler * handler, const gchar * uri,
GError ** error)
{
GstRTMPSrc *src = GST_RTMP_SRC (handler);
if (GST_STATE (src) >= GST_STATE_PAUSED) {
g_set_error (error, GST_URI_ERROR, GST_URI_ERROR_BAD_STATE,
"Changing the URI on rtmpsrc when it is running is not supported");
return FALSE;
}
g_free (src->uri);
src->uri = NULL;
if (uri != NULL) {
int protocol;
AVal host;
unsigned int port;
AVal playpath, app;
if (!RTMP_ParseURL (uri, &protocol, &host, &port, &playpath, &app) ||
!host.av_len || !playpath.av_len) {
GST_ERROR_OBJECT (src, "Failed to parse URI %s", uri);
g_set_error (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
"Could not parse RTMP URI");
/* FIXME: we should not be freeing RTMP internals to avoid leaking */
free (playpath.av_val);
return FALSE;
}
free (playpath.av_val);
src->uri = g_strdup (uri);
}
GST_DEBUG_OBJECT (src, "Changed URI to %s", GST_STR_NULL (uri));
return TRUE;
}
static void
gst_rtmp_src_uri_handler_init (gpointer g_iface, gpointer iface_data)
{
GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
iface->get_type = gst_rtmp_src_uri_get_type;
iface->get_protocols = gst_rtmp_src_uri_get_protocols;
iface->get_uri = gst_rtmp_src_uri_get_uri;
iface->set_uri = gst_rtmp_src_uri_set_uri;
}
static void
gst_rtmp_src_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRTMPSrc *src;
src = GST_RTMP_SRC (object);
switch (prop_id) {
case PROP_LOCATION:{
gst_rtmp_src_uri_set_uri (GST_URI_HANDLER (src),
g_value_get_string (value), NULL);
break;
}
case PROP_TIMEOUT:{
src->timeout = g_value_get_int (value);
break;
}
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtmp_src_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstRTMPSrc *src;
src = GST_RTMP_SRC (object);
switch (prop_id) {
case PROP_LOCATION:
g_value_set_string (value, src->uri);
break;
case PROP_TIMEOUT:
g_value_set_int (value, src->timeout);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
/*
* Read a new buffer from src->reqoffset, takes care of events
* and seeking and such.
*/
static GstFlowReturn
gst_rtmp_src_create (GstPushSrc * pushsrc, GstBuffer ** buffer)
{
GstRTMPSrc *src;
GstBuffer *buf;
GstMapInfo map;
guint8 *data;
guint todo;
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gsize bsize;
int size;
src = GST_RTMP_SRC (pushsrc);
g_return_val_if_fail (src->rtmp != NULL, GST_FLOW_ERROR);
if (!RTMP_IsConnected (src->rtmp)) {
GST_DEBUG_OBJECT (src, "reconnecting");
if (!gst_rtmp_src_connect (src))
return GST_FLOW_ERROR;
}
size = GST_BASE_SRC_CAST (pushsrc)->blocksize;
GST_DEBUG ("reading from %" G_GUINT64_FORMAT
", size %u", src->cur_offset, size);
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buf = gst_buffer_new_allocate (NULL, size, NULL);
if (G_UNLIKELY (buf == NULL)) {
GST_ERROR_OBJECT (src, "Failed to allocate %u bytes", size);
return GST_FLOW_ERROR;
}
todo = size;
gst_buffer_map (buf, &map, GST_MAP_WRITE);
data = map.data;
bsize = 0;
while (todo > 0) {
int read = RTMP_Read (src->rtmp, (char *) data, todo);
if (G_UNLIKELY (read == 0 && todo == size))
goto eos;
if (G_UNLIKELY (read == 0))
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break;
if (G_UNLIKELY (read < 0))
goto read_failed;
if (read < todo) {
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data += read;
todo -= read;
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bsize += read;
} else {
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bsize += todo;
todo = 0;
}
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GST_LOG (" got size %d", read);
}
gst_buffer_unmap (buf, &map);
gst_buffer_resize (buf, 0, bsize);
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if (src->discont) {
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
src->discont = FALSE;
}
GST_BUFFER_TIMESTAMP (buf) = src->last_timestamp;
GST_BUFFER_OFFSET (buf) = src->cur_offset;
src->cur_offset += size;
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if (src->last_timestamp == GST_CLOCK_TIME_NONE)
src->last_timestamp = src->rtmp->m_mediaStamp * GST_MSECOND;
else
src->last_timestamp =
MAX (src->last_timestamp, src->rtmp->m_mediaStamp * GST_MSECOND);
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GST_LOG_OBJECT (src, "Created buffer of size %u at %" G_GINT64_FORMAT
" with timestamp %" GST_TIME_FORMAT, size, GST_BUFFER_OFFSET (buf),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
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/* we're done, return the buffer */
*buffer = buf;
return GST_FLOW_OK;
read_failed:
{
gst_buffer_unmap (buf, &map);
gst_buffer_unref (buf);
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GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL), ("Failed to read data"));
return GST_FLOW_ERROR;
}
eos:
{
gst_buffer_unmap (buf, &map);
gst_buffer_unref (buf);
if (src->cur_offset == 0) {
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
("Failed to read any data from stream, check your URL"));
return GST_FLOW_ERROR;
} else {
GST_DEBUG_OBJECT (src, "Reading data gave EOS");
return GST_FLOW_EOS;
}
}
}
static gboolean
gst_rtmp_src_query (GstBaseSrc * basesrc, GstQuery * query)
{
gboolean ret = FALSE;
GstRTMPSrc *src = GST_RTMP_SRC (basesrc);
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_URI:
gst_query_set_uri (query, src->uri);
ret = TRUE;
break;
case GST_QUERY_POSITION:{
GstFormat format;
gst_query_parse_position (query, &format, NULL);
if (format == GST_FORMAT_TIME) {
gst_query_set_position (query, format, src->last_timestamp);
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;
}
case GST_QUERY_SCHEDULING:{
gst_query_set_scheduling (query,
GST_SCHEDULING_FLAG_SEQUENTIAL |
GST_SCHEDULING_FLAG_BANDWIDTH_LIMITED, 1, -1, 0);
gst_query_add_scheduling_mode (query, GST_PAD_MODE_PUSH);
ret = TRUE;
break;
}
default:
ret = FALSE;
break;
}
if (!ret)
ret = GST_BASE_SRC_CLASS (parent_class)->query (basesrc, query);
return ret;
}
static gboolean
gst_rtmp_src_is_seekable (GstBaseSrc * basesrc)
{
GstRTMPSrc *src;
src = GST_RTMP_SRC (basesrc);
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return src->seekable;
}
static gboolean
gst_rtmp_src_prepare_seek_segment (GstBaseSrc * basesrc, GstEvent * event,
GstSegment * segment)
{
GstRTMPSrc *src;
GstSeekType cur_type, stop_type;
gint64 cur, stop;
GstSeekFlags flags;
GstFormat format;
gdouble rate;
src = GST_RTMP_SRC (basesrc);
gst_event_parse_seek (event, &rate, &format, &flags,
&cur_type, &cur, &stop_type, &stop);
if (!src->seekable) {
GST_LOG_OBJECT (src, "Not a seekable stream");
return FALSE;
}
if (!src->rtmp) {
GST_LOG_OBJECT (src, "Not connected yet");
return FALSE;
}
if (format != GST_FORMAT_TIME) {
GST_LOG_OBJECT (src, "Seeking only supported in TIME format");
return FALSE;
}
if (stop_type != GST_SEEK_TYPE_NONE) {
GST_LOG_OBJECT (src, "Setting a stop position is not supported");
return FALSE;
}
gst_segment_init (segment, GST_FORMAT_TIME);
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gst_segment_do_seek (segment, rate, format, flags, cur_type, cur, stop_type,
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stop, NULL);
return TRUE;
}
static gboolean
gst_rtmp_src_do_seek (GstBaseSrc * basesrc, GstSegment * segment)
{
GstRTMPSrc *src;
src = GST_RTMP_SRC (basesrc);
if (segment->format != GST_FORMAT_TIME) {
GST_LOG_OBJECT (src, "Only time based seeks are supported");
return FALSE;
}
if (!src->rtmp) {
GST_LOG_OBJECT (src, "Not connected yet");
return FALSE;
}
/* Initial seek */
if (src->cur_offset == 0 && segment->start == 0)
goto success;
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if (!src->seekable) {
GST_LOG_OBJECT (src, "Not a seekable stream");
return FALSE;
}
/* If we have just disconnected in unlock(), we need to re-connect
* and also let librtmp read some data before sending a seek,
* otherwise it will stall. Calling create() does both. */
if (!RTMP_IsConnected (src->rtmp)) {
GstBuffer *buffer = NULL;
gst_rtmp_src_create (GST_PUSH_SRC (basesrc), &buffer);
gst_buffer_replace (&buffer, NULL);
}
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src->last_timestamp = GST_CLOCK_TIME_NONE;
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if (!RTMP_SendSeek (src->rtmp, segment->start / GST_MSECOND)) {
GST_ERROR_OBJECT (src, "Seeking failed");
src->seekable = FALSE;
return FALSE;
}
success:
/* This is set here so that the call to create() above doesn't clear it */
src->discont = TRUE;
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GST_DEBUG_OBJECT (src, "Seek to %" GST_TIME_FORMAT " successful",
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GST_TIME_ARGS (segment->start));
return TRUE;
}
static gboolean
gst_rtmp_src_connect (GstRTMPSrc * src)
{
RTMP_Init (src->rtmp);
src->rtmp->Link.timeout = src->timeout;
if (!RTMP_SetupURL (src->rtmp, src->uri)) {
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
("Failed to setup URL '%s'", src->uri));
return FALSE;
}
src->seekable = !(src->rtmp->Link.lFlags & RTMP_LF_LIVE);
GST_INFO_OBJECT (src, "seekable %d", src->seekable);
/* open if required */
if (!RTMP_IsConnected (src->rtmp)) {
if (!RTMP_Connect (src->rtmp, NULL)) {
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL),
("Could not connect to RTMP stream \"%s\" for reading", src->uri));
return FALSE;
}
}
return TRUE;
}
/* open the file, do stuff necessary to go to PAUSED state */
static gboolean
gst_rtmp_src_start (GstBaseSrc * basesrc)
{
GstRTMPSrc *src;
src = GST_RTMP_SRC (basesrc);
if (!src->uri) {
GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ, (NULL), ("No filename given"));
return FALSE;
}
src->cur_offset = 0;
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src->last_timestamp = 0;
src->discont = TRUE;
src->rtmp = RTMP_Alloc ();
if (!src->rtmp) {
GST_ERROR_OBJECT (src, "Could not allocate librtmp's RTMP context");
goto error;
}
if (!gst_rtmp_src_connect (src))
goto error;
return TRUE;
error:
if (src->rtmp) {
RTMP_Free (src->rtmp);
src->rtmp = NULL;
}
return FALSE;
}
static gboolean
gst_rtmp_src_unlock (GstBaseSrc * basesrc)
{
GstRTMPSrc *rtmpsrc = GST_RTMP_SRC (basesrc);
GST_DEBUG_OBJECT (rtmpsrc, "unlock");
/* This closes the socket, which means that any pending socket calls
* error out. */
if (rtmpsrc->rtmp) {
RTMP_Close (rtmpsrc->rtmp);
}
return TRUE;
}
static gboolean
gst_rtmp_src_stop (GstBaseSrc * basesrc)
{
GstRTMPSrc *src;
src = GST_RTMP_SRC (basesrc);
if (src->rtmp) {
RTMP_Free (src->rtmp);
src->rtmp = NULL;
}
src->cur_offset = 0;
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src->last_timestamp = 0;
src->discont = TRUE;
return TRUE;
}