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authorJan Alexander Steffens (heftig) <jan.steffens@ltnglobal.com>2020-06-29 20:25:27 +0200
committerGStreamer Merge Bot <gitlab-merge-bot@gstreamer-foundation.org>2020-07-01 18:33:42 +0000
commit1e29c5d52ae0eb71222ddc85529d8beb696855e9 (patch)
tree38022be4615967185056802d63edb92cb1b1f627 /gst
parent2ad3aab1d47c2752c82dde757e9f541f50fd272b (diff)
downloadgstreamer-plugins-bad-1e29c5d52ae0eb71222ddc85529d8beb696855e9.tar.gz
rtmp2: Set connect args like libavformat does
To improve our compatibility. Critically, a server might elide data for codecs we don't advertise. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1384>
Diffstat (limited to 'gst')
-rw-r--r--gst/rtmp2/gstrtmp2sink.c1
-rw-r--r--gst/rtmp2/rtmp/rtmpclient.c108
-rw-r--r--gst/rtmp2/rtmp/rtmpclient.h1
3 files changed, 108 insertions, 2 deletions
diff --git a/gst/rtmp2/gstrtmp2sink.c b/gst/rtmp2/gstrtmp2sink.c
index a5ee7acdd..08937930b 100644
--- a/gst/rtmp2/gstrtmp2sink.c
+++ b/gst/rtmp2/gstrtmp2sink.c
@@ -236,6 +236,7 @@ static void
gst_rtmp2_sink_init (GstRtmp2Sink * self)
{
self->location.flash_ver = g_strdup ("FMLE/3.0 (compatible; FMSc/1.0)");
+ self->location.publish = TRUE;
self->async_connect = TRUE;
self->chunk_size = GST_RTMP_DEFAULT_CHUNK_SIZE;
diff --git a/gst/rtmp2/rtmp/rtmpclient.c b/gst/rtmp2/rtmp/rtmpclient.c
index fadc3c2f1..73f397d3d 100644
--- a/gst/rtmp2/rtmp/rtmpclient.c
+++ b/gst/rtmp2/rtmp/rtmpclient.c
@@ -184,6 +184,7 @@ gst_rtmp_location_copy (GstRtmpLocation * dest, const GstRtmpLocation * src)
dest->timeout = src->timeout;
dest->tls_flags = src->tls_flags;
dest->flash_ver = g_strdup (src->flash_ver);
+ dest->publish = src->publish;
}
void
@@ -199,6 +200,7 @@ gst_rtmp_location_clear (GstRtmpLocation * location)
g_clear_pointer (&location->password, g_free);
g_clear_pointer (&location->secure_token, g_free);
g_clear_pointer (&location->flash_ver, g_free);
+ location->publish = FALSE;
}
gchar *
@@ -229,6 +231,65 @@ gst_rtmp_location_get_string (const GstRtmpLocation * location,
return string;
}
+/* Flag values for the audioCodecs property,
+ * rtmp_specification_1.0.pdf page 32 */
+enum
+{
+ SUPPORT_SND_NONE = 0x001, /* Raw sound, no compression */
+ SUPPORT_SND_ADPCM = 0x002, /* ADPCM compression */
+ SUPPORT_SND_MP3 = 0x004, /* mp3 compression */
+ SUPPORT_SND_INTEL = 0x008, /* Not used */
+ SUPPORT_SND_UNUSED = 0x010, /* Not used */
+ SUPPORT_SND_NELLY8 = 0x020, /* NellyMoser at 8-kHz compression */
+ SUPPORT_SND_NELLY = 0x040, /* NellyMoser compression
+ * (5, 11, 22, and 44 kHz) */
+ SUPPORT_SND_G711A = 0x080, /* G711A sound compression
+ * (Flash Media Server only) */
+ SUPPORT_SND_G711U = 0x100, /* G711U sound compression
+ * (Flash Media Server only) */
+ SUPPORT_SND_NELLY16 = 0x200, /* NellyMoser at 16-kHz compression */
+ SUPPORT_SND_AAC = 0x400, /* Advanced audio coding (AAC) codec */
+ SUPPORT_SND_SPEEX = 0x800, /* Speex Audio */
+ SUPPORT_SND_ALL = 0xFFF, /* All RTMP-supported audio codecs */
+};
+
+/* audioCodecs value sent by libavformat. All "used" codecs. */
+#define GST_RTMP_AUDIOCODECS \
+ (SUPPORT_SND_ALL & ~SUPPORT_SND_INTEL & ~SUPPORT_SND_UNUSED)
+G_STATIC_ASSERT (GST_RTMP_AUDIOCODECS == 4071); /* libavformat's magic number */
+
+/* Flag values for the videoCodecs property,
+ * rtmp_specification_1.0.pdf page 32 */
+enum
+{
+ SUPPORT_VID_UNUSED = 0x01, /* Obsolete value */
+ SUPPORT_VID_JPEG = 0x02, /* Obsolete value */
+ SUPPORT_VID_SORENSON = 0x04, /* Sorenson Flash video */
+ SUPPORT_VID_HOMEBREW = 0x08, /* V1 screen sharing */
+ SUPPORT_VID_VP6 = 0x10, /* On2 video (Flash 8+) */
+ SUPPORT_VID_VP6ALPHA = 0x20, /* On2 video with alpha channel */
+ SUPPORT_VID_HOMEBREWV = 0x40, /* Screen sharing version 2 (Flash 8+) */
+ SUPPORT_VID_H264 = 0x80, /* H264 video */
+ SUPPORT_VID_ALL = 0xFF, /* All RTMP-supported video codecs */
+};
+
+/* videoCodecs value sent by libavformat. All non-obsolete codecs. */
+#define GST_RTMP_VIDEOCODECS \
+ (SUPPORT_VID_ALL & ~SUPPORT_VID_UNUSED & ~SUPPORT_VID_JPEG)
+G_STATIC_ASSERT (GST_RTMP_VIDEOCODECS == 252); /* libavformat's magic number */
+
+/* Flag values for the videoFunction property,
+ * rtmp_specification_1.0.pdf page 32 */
+enum
+{
+ /* Indicates that the client can perform frame-accurate seeks. */
+ SUPPORT_VID_CLIENT_SEEK = 1,
+};
+
+/* videoFunction value sent by libavformat */
+#define GST_RTMP_VIDEOFUNCTION (SUPPORT_VID_CLIENT_SEEK)
+G_STATIC_ASSERT (GST_RTMP_VIDEOFUNCTION == 1); /* libavformat's magic number */
+
static void socket_connect (GTask * task);
static void socket_connect_done (GObject * source, GAsyncResult * result,
gpointer user_data);
@@ -499,10 +560,12 @@ send_connect (GTask * task)
GstAmfNode *node;
const gchar *app, *flash_ver;
gchar *uri, *appstr = NULL, *uristr = NULL;
+ gboolean publish;
node = gst_amf_node_new_object ();
app = data->location.application;
flash_ver = data->location.flash_ver;
+ publish = data->location.publish;
uri = gst_rtmp_location_get_string (&data->location, FALSE);
if (!app) {
@@ -548,11 +611,52 @@ send_connect (GTask * task)
uristr = g_strdup (uri);
}
+ /* Arguments for the connect command.
+ * Most of these are described in rtmp_specification_1.0.pdf page 30 */
+
+ /* "The server application name the client is connected to." */
gst_amf_node_append_field_take_string (node, "app", appstr, -1);
- gst_amf_node_append_field_take_string (node, "tcUrl", uristr, -1);
- gst_amf_node_append_field_string (node, "type", "nonprivate", -1);
+
+ if (publish) {
+ /* Undocumented. Sent by both libavformat and librtmp. */
+ gst_amf_node_append_field_string (node, "type", "nonprivate", -1);
+ }
+
+ /* "Flash Player version. It is the same string as returned by the
+ * ApplicationScript getversion () function." */
gst_amf_node_append_field_string (node, "flashVer", flash_ver, -1);
+ /* "URL of the source SWF file making the connection."
+ * XXX: libavformat sends "swfUrl" here, if provided. */
+
+ /* "URL of the Server. It has the following format.
+ * protocol://servername:port/appName/appInstance" */
+ gst_amf_node_append_field_take_string (node, "tcUrl", uristr, -1);
+
+ if (!publish) {
+ /* "True if proxy is being used." */
+ gst_amf_node_append_field_boolean (node, "fpad", FALSE);
+
+ /* Undocumented. Sent by libavformat. */
+ gst_amf_node_append_field_number (node, "capabilities",
+ 15 /* libavformat's magic number */ );
+
+ /* "Indicates what audio codecs the client supports." */
+ gst_amf_node_append_field_number (node, "audioCodecs",
+ GST_RTMP_AUDIOCODECS);
+
+ /* "Indicates what video codecs are supported." */
+ gst_amf_node_append_field_number (node, "videoCodecs",
+ GST_RTMP_VIDEOCODECS);
+
+ /* "Indicates what special video functions are supported." */
+ gst_amf_node_append_field_number (node, "videoFunction",
+ GST_RTMP_VIDEOFUNCTION);
+
+ /* "URL of the web page from where the SWF file was loaded."
+ * XXX: libavformat sends "pageUrl" here, if provided. */
+ }
+
gst_rtmp_connection_send_command (data->connection, send_connect_done,
task, 0, "connect", node, NULL);
diff --git a/gst/rtmp2/rtmp/rtmpclient.h b/gst/rtmp2/rtmp/rtmpclient.h
index ec75d43f9..10a6e22de 100644
--- a/gst/rtmp2/rtmp/rtmpclient.h
+++ b/gst/rtmp2/rtmp/rtmpclient.h
@@ -70,6 +70,7 @@ typedef struct _GstRtmpLocation
gint timeout;
GTlsCertificateFlags tls_flags;
gchar *flash_ver;
+ gboolean publish;
} GstRtmpLocation;
void gst_rtmp_location_copy (GstRtmpLocation * dest,