| /* 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 |
| * |
| * 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. |
| * |
| * <refsect2> |
| * <title>Example launch lines</title> |
| * |[ |
| * gst-launch-1.0 -v rtmpsrc location=rtmp://somehost/someurl ! fakesink |
| * ]| Open an RTMP location and pass its content to fakesink. |
| * </refsect2> |
| */ |
| |
| #ifdef HAVE_CONFIG_H |
| #include "config.h" |
| #endif |
| |
| #include <glib/gi18n-lib.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_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); |
| 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); |
| |
| #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)); |
| |
| static void |
| gst_rtmp_src_class_init (GstRTMPSrcClass * klass) |
| { |
| GObjectClass *gobject_class; |
| GstElementClass *gstelement_class; |
| GstBaseSrcClass *gstbasesrc_class; |
| GstPushSrcClass *gstpushsrc_class; |
| |
| gobject_class = G_OBJECT_CLASS (klass); |
| 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); |
| |
| gst_element_class_set_static_metadata (gstelement_class, |
| "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); |
| 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); |
| |
| GST_DEBUG_CATEGORY_INIT (rtmpsrc_debug, "rtmpsrc", 0, "RTMP Source"); |
| } |
| |
| static void |
| 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; |
| |
| 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 |
| gst_rtmp_src_uri_get_type (GType type) |
| { |
| return GST_URI_SRC; |
| } |
| |
| static const gchar *const * |
| 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; |
| gsize bsize; |
| int size; |
| |
| src = GST_RTMP_SRC (pushsrc); |
| |
| g_return_val_if_fail (src->rtmp != NULL, GST_FLOW_ERROR); |
| |
| size = GST_BASE_SRC_CAST (pushsrc)->blocksize; |
| |
| GST_DEBUG ("reading from %" G_GUINT64_FORMAT |
| ", size %u", src->cur_offset, size); |
| |
| 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)) |
| break; |
| |
| if (G_UNLIKELY (read < 0)) |
| goto read_failed; |
| |
| if (read < todo) { |
| data += read; |
| todo -= read; |
| bsize += read; |
| } else { |
| bsize += todo; |
| todo = 0; |
| } |
| GST_LOG (" got size %d", read); |
| } |
| gst_buffer_unmap (buf, &map); |
| gst_buffer_resize (buf, 0, bsize); |
| |
| 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; |
| 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); |
| |
| 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))); |
| |
| |
| /* we're done, return the buffer */ |
| *buffer = buf; |
| |
| return GST_FLOW_OK; |
| |
| read_failed: |
| { |
| gst_buffer_unref (buf); |
| GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL), ("Failed to read data")); |
| return GST_FLOW_ERROR; |
| } |
| eos: |
| { |
| 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); |
| |
| 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); |
| gst_segment_do_seek (segment, rate, format, flags, cur_type, cur, stop_type, |
| 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; |
| } |
| |
| src->discont = TRUE; |
| |
| /* Initial seek */ |
| if (src->cur_offset == 0 && segment->start == 0) |
| return TRUE; |
| |
| if (!src->seekable) { |
| GST_LOG_OBJECT (src, "Not a seekable stream"); |
| return FALSE; |
| } |
| |
| src->last_timestamp = GST_CLOCK_TIME_NONE; |
| if (!RTMP_SendSeek (src->rtmp, segment->start / GST_MSECOND)) { |
| GST_ERROR_OBJECT (src, "Seeking failed"); |
| src->seekable = FALSE; |
| return FALSE; |
| } |
| |
| GST_DEBUG_OBJECT (src, "Seek to %" GST_TIME_FORMAT " successfull", |
| GST_TIME_ARGS (segment->start)); |
| |
| return TRUE; |
| } |
| |
| #define STR2AVAL(av,str) G_STMT_START { \ |
| av.av_val = str; \ |
| av.av_len = strlen(av.av_val); \ |
| } G_STMT_END; |
| |
| /* 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; |
| 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; |
| } |
| |
| 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)); |
| goto error; |
| } |
| 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)); |
| goto error; |
| } |
| } |
| |
| return TRUE; |
| |
| error: |
| if (src->rtmp) { |
| RTMP_Free (src->rtmp); |
| src->rtmp = NULL; |
| } |
| return FALSE; |
| } |
| |
| #undef STR2AVAL |
| |
| 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; |
| src->last_timestamp = 0; |
| src->discont = TRUE; |
| |
| return TRUE; |
| } |