gstreamer/gst/rtmp2/gstrtmp2sink.c

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/* GStreamer
* Copyright (C) 2014 David Schleef <ds@schleef.org>
* Copyright (C) 2017 Make.TV, Inc. <info@make.tv>
* Contact: Jan Alexander Steffens (heftig) <jsteffens@make.tv>
*
* 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 Street, Suite 500,
* Boston, MA 02110-1335, USA.
*/
/**
* SECTION:element-gstrtmp2sink
*
* The rtmp2sink element sends audio and video streams to an RTMP
* server.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch -v videotestsrc ! x264enc ! flvmux ! rtmp2sink
* location=rtmp://server.example.com/live/myStream
* ]|
* FIXME Describe what the pipeline does.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstrtmp2sink.h"
#include "gstrtmp2locationhandler.h"
#include "rtmp/rtmpclient.h"
#include "rtmp/rtmpmessage.h"
#include <gst/gst.h>
#include <gst/base/gstbasesink.h>
#include <gio/gnetworking.h>
#include <string.h>
GST_DEBUG_CATEGORY_STATIC (gst_rtmp2_sink_debug_category);
#define GST_CAT_DEFAULT gst_rtmp2_sink_debug_category
/* prototypes */
#define GST_RTMP2_SINK(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RTMP2_SINK,GstRtmp2Sink))
#define GST_IS_RTMP2_SINK(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RTMP2_SINK))
typedef struct
{
GstBaseSink parent_instance;
/* properties */
GstRtmpLocation location;
gboolean async_connect;
guint peak_kbps;
/* stuff */
gboolean running, flushing;
GMutex lock;
GCond cond;
GstTask *task;
GRecMutex task_lock;
GMainLoop *loop;
GMainContext *context;
GCancellable *cancellable;
GstRtmpConnection *connection;
guint32 stream_id;
GPtrArray *headers;
guint64 last_ts, base_ts; /* timestamp fixup */
} GstRtmp2Sink;
typedef struct
{
GstBaseSinkClass parent_class;
} GstRtmp2SinkClass;
/* GObject virtual functions */
static void gst_rtmp2_sink_set_property (GObject * object,
guint property_id, const GValue * value, GParamSpec * pspec);
static void gst_rtmp2_sink_get_property (GObject * object,
guint property_id, GValue * value, GParamSpec * pspec);
static void gst_rtmp2_sink_finalize (GObject * object);
static void gst_rtmp2_sink_uri_handler_init (GstURIHandlerInterface * iface);
/* GstBaseSink virtual functions */
static gboolean gst_rtmp2_sink_start (GstBaseSink * sink);
static gboolean gst_rtmp2_sink_stop (GstBaseSink * sink);
static gboolean gst_rtmp2_sink_unlock (GstBaseSink * sink);
static gboolean gst_rtmp2_sink_unlock_stop (GstBaseSink * sink);
static GstFlowReturn gst_rtmp2_sink_render (GstBaseSink * sink,
GstBuffer * buffer);
static gboolean gst_rtmp2_sink_set_caps (GstBaseSink * sink, GstCaps * caps);
/* Internal API */
static void gst_rtmp2_sink_task_func (gpointer user_data);
static void client_connect_done (GObject * source, GAsyncResult * result,
gpointer user_data);
static void start_publish_done (GObject * source, GAsyncResult * result,
gpointer user_data);
static void connect_task_done (GObject * object, GAsyncResult * result,
gpointer user_data);
static void set_pacing_rate (GstRtmp2Sink * self);
enum
{
PROP_0,
PROP_LOCATION,
PROP_SCHEME,
PROP_HOST,
PROP_PORT,
PROP_APPLICATION,
PROP_STREAM,
PROP_SECURE_TOKEN,
PROP_USERNAME,
PROP_PASSWORD,
PROP_AUTHMOD,
PROP_TIMEOUT,
PROP_TLS_VALIDATION_FLAGS,
PROP_ASYNC_CONNECT,
PROP_PEAK_KBPS,
};
/* pad templates */
static GstStaticPadTemplate gst_rtmp2_sink_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("video/x-flv")
);
/* class initialization */
G_DEFINE_TYPE_WITH_CODE (GstRtmp2Sink, gst_rtmp2_sink, GST_TYPE_BASE_SINK,
G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER,
gst_rtmp2_sink_uri_handler_init);
G_IMPLEMENT_INTERFACE (GST_TYPE_RTMP_LOCATION_HANDLER, NULL));
static void
gst_rtmp2_sink_class_init (GstRtmp2SinkClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstBaseSinkClass *base_sink_class = GST_BASE_SINK_CLASS (klass);
gst_element_class_add_static_pad_template (GST_ELEMENT_CLASS (klass),
&gst_rtmp2_sink_sink_template);
gst_element_class_set_static_metadata (GST_ELEMENT_CLASS (klass),
"RTMP sink element", "Sink", "Sink element for RTMP streams",
"Make.TV, Inc. <info@make.tv>");
gobject_class->set_property = gst_rtmp2_sink_set_property;
gobject_class->get_property = gst_rtmp2_sink_get_property;
gobject_class->finalize = gst_rtmp2_sink_finalize;
base_sink_class->start = GST_DEBUG_FUNCPTR (gst_rtmp2_sink_start);
base_sink_class->stop = GST_DEBUG_FUNCPTR (gst_rtmp2_sink_stop);
base_sink_class->unlock = GST_DEBUG_FUNCPTR (gst_rtmp2_sink_unlock);
base_sink_class->unlock_stop = GST_DEBUG_FUNCPTR (gst_rtmp2_sink_unlock_stop);
base_sink_class->render = GST_DEBUG_FUNCPTR (gst_rtmp2_sink_render);
base_sink_class->set_caps = GST_DEBUG_FUNCPTR (gst_rtmp2_sink_set_caps);
g_object_class_override_property (gobject_class, PROP_LOCATION, "location");
g_object_class_override_property (gobject_class, PROP_SCHEME, "scheme");
g_object_class_override_property (gobject_class, PROP_HOST, "host");
g_object_class_override_property (gobject_class, PROP_PORT, "port");
g_object_class_override_property (gobject_class, PROP_APPLICATION,
"application");
g_object_class_override_property (gobject_class, PROP_STREAM, "stream");
g_object_class_override_property (gobject_class, PROP_SECURE_TOKEN,
"secure-token");
g_object_class_override_property (gobject_class, PROP_USERNAME, "username");
g_object_class_override_property (gobject_class, PROP_PASSWORD, "password");
g_object_class_override_property (gobject_class, PROP_AUTHMOD, "authmod");
g_object_class_override_property (gobject_class, PROP_TIMEOUT, "timeout");
g_object_class_override_property (gobject_class, PROP_TLS_VALIDATION_FLAGS,
"tls-validation-flags");
g_object_class_install_property (gobject_class, PROP_ASYNC_CONNECT,
g_param_spec_boolean ("async-connect", "Async connect",
"Connect on READY, otherwise on first push", TRUE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_PEAK_KBPS,
g_param_spec_uint ("peak-kbps", "Peak bitrate",
"Bitrate in kbit/sec to pace outgoing packets", 0, G_MAXINT / 125, 0,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
GST_PARAM_MUTABLE_PLAYING));
GST_DEBUG_CATEGORY_INIT (gst_rtmp2_sink_debug_category, "rtmp2sink", 0,
"debug category for rtmp2sink element");
}
static void
gst_rtmp2_sink_init (GstRtmp2Sink * self)
{
self->location.flash_ver = g_strdup ("FMLE/3.0 (compatible; FMSc/1.0)");
self->async_connect = TRUE;
g_mutex_init (&self->lock);
g_cond_init (&self->cond);
self->task = gst_task_new (gst_rtmp2_sink_task_func, self, NULL);
g_rec_mutex_init (&self->task_lock);
gst_task_set_lock (self->task, &self->task_lock);
self->headers = g_ptr_array_new_with_free_func
((GDestroyNotify) gst_mini_object_unref);
}
static void
gst_rtmp2_sink_uri_handler_init (GstURIHandlerInterface * iface)
{
gst_rtmp_location_handler_implement_uri_handler (iface, GST_URI_SINK);
}
static void
gst_rtmp2_sink_set_property (GObject * object, guint property_id,
const GValue * value, GParamSpec * pspec)
{
GstRtmp2Sink *self = GST_RTMP2_SINK (object);
switch (property_id) {
case PROP_LOCATION:
gst_rtmp_location_handler_set_uri (GST_RTMP_LOCATION_HANDLER (self),
g_value_get_string (value));
break;
case PROP_SCHEME:
GST_OBJECT_LOCK (self);
self->location.scheme = g_value_get_enum (value);
GST_OBJECT_UNLOCK (self);
break;
case PROP_HOST:
GST_OBJECT_LOCK (self);
g_free (self->location.host);
self->location.host = g_value_dup_string (value);
GST_OBJECT_UNLOCK (self);
break;
case PROP_PORT:
GST_OBJECT_LOCK (self);
self->location.port = g_value_get_int (value);
GST_OBJECT_UNLOCK (self);
break;
case PROP_APPLICATION:
GST_OBJECT_LOCK (self);
g_free (self->location.application);
self->location.application = g_value_dup_string (value);
GST_OBJECT_UNLOCK (self);
break;
case PROP_STREAM:
GST_OBJECT_LOCK (self);
g_free (self->location.stream);
self->location.stream = g_value_dup_string (value);
GST_OBJECT_UNLOCK (self);
break;
case PROP_SECURE_TOKEN:
GST_OBJECT_LOCK (self);
g_free (self->location.secure_token);
self->location.secure_token = g_value_dup_string (value);
GST_OBJECT_UNLOCK (self);
break;
case PROP_USERNAME:
GST_OBJECT_LOCK (self);
g_free (self->location.username);
self->location.username = g_value_dup_string (value);
GST_OBJECT_UNLOCK (self);
break;
case PROP_PASSWORD:
GST_OBJECT_LOCK (self);
g_free (self->location.password);
self->location.password = g_value_dup_string (value);
GST_OBJECT_UNLOCK (self);
break;
case PROP_AUTHMOD:
GST_OBJECT_LOCK (self);
self->location.authmod = g_value_get_enum (value);
GST_OBJECT_UNLOCK (self);
break;
case PROP_TIMEOUT:
GST_OBJECT_LOCK (self);
self->location.timeout = g_value_get_uint (value);
GST_OBJECT_UNLOCK (self);
break;
case PROP_TLS_VALIDATION_FLAGS:
GST_OBJECT_LOCK (self);
self->location.tls_flags = g_value_get_flags (value);
GST_OBJECT_UNLOCK (self);
break;
case PROP_ASYNC_CONNECT:
GST_OBJECT_LOCK (self);
self->async_connect = g_value_get_boolean (value);
GST_OBJECT_UNLOCK (self);
break;
case PROP_PEAK_KBPS:
GST_OBJECT_LOCK (self);
self->peak_kbps = g_value_get_uint (value);
GST_OBJECT_UNLOCK (self);
g_mutex_lock (&self->lock);
set_pacing_rate (self);
g_mutex_unlock (&self->lock);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
break;
}
}
static void
gst_rtmp2_sink_get_property (GObject * object, guint property_id,
GValue * value, GParamSpec * pspec)
{
GstRtmp2Sink *self = GST_RTMP2_SINK (object);
switch (property_id) {
case PROP_LOCATION:
GST_OBJECT_LOCK (self);
g_value_take_string (value, gst_rtmp_location_get_string (&self->location,
TRUE));
GST_OBJECT_UNLOCK (self);
break;
case PROP_SCHEME:
GST_OBJECT_LOCK (self);
g_value_set_enum (value, self->location.scheme);
GST_OBJECT_UNLOCK (self);
break;
case PROP_HOST:
GST_OBJECT_LOCK (self);
g_value_set_string (value, self->location.host);
GST_OBJECT_UNLOCK (self);
break;
case PROP_PORT:
GST_OBJECT_LOCK (self);
g_value_set_int (value, self->location.port);
GST_OBJECT_UNLOCK (self);
break;
case PROP_APPLICATION:
GST_OBJECT_LOCK (self);
g_value_set_string (value, self->location.application);
GST_OBJECT_UNLOCK (self);
break;
case PROP_STREAM:
GST_OBJECT_LOCK (self);
g_value_set_string (value, self->location.stream);
GST_OBJECT_UNLOCK (self);
break;
case PROP_SECURE_TOKEN:
GST_OBJECT_LOCK (self);
g_value_set_string (value, self->location.secure_token);
GST_OBJECT_UNLOCK (self);
break;
case PROP_USERNAME:
GST_OBJECT_LOCK (self);
g_value_set_string (value, self->location.username);
GST_OBJECT_UNLOCK (self);
break;
case PROP_PASSWORD:
GST_OBJECT_LOCK (self);
g_value_set_string (value, self->location.password);
GST_OBJECT_UNLOCK (self);
break;
case PROP_AUTHMOD:
GST_OBJECT_LOCK (self);
g_value_set_enum (value, self->location.authmod);
GST_OBJECT_UNLOCK (self);
break;
case PROP_TIMEOUT:
GST_OBJECT_LOCK (self);
g_value_set_uint (value, self->location.timeout);
GST_OBJECT_UNLOCK (self);
break;
case PROP_TLS_VALIDATION_FLAGS:
GST_OBJECT_LOCK (self);
g_value_set_flags (value, self->location.tls_flags);
GST_OBJECT_UNLOCK (self);
break;
case PROP_ASYNC_CONNECT:
GST_OBJECT_LOCK (self);
g_value_set_boolean (value, self->async_connect);
GST_OBJECT_UNLOCK (self);
break;
case PROP_PEAK_KBPS:
GST_OBJECT_LOCK (self);
g_value_set_uint (value, self->peak_kbps);
GST_OBJECT_UNLOCK (self);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, property_id, pspec);
break;
}
}
static void
gst_rtmp2_sink_finalize (GObject * object)
{
GstRtmp2Sink *self = GST_RTMP2_SINK (object);
g_clear_pointer (&self->headers, g_ptr_array_unref);
g_clear_object (&self->cancellable);
g_clear_object (&self->connection);
g_clear_object (&self->task);
g_rec_mutex_clear (&self->task_lock);
g_mutex_clear (&self->lock);
g_cond_clear (&self->cond);
gst_rtmp_location_clear (&self->location);
G_OBJECT_CLASS (gst_rtmp2_sink_parent_class)->finalize (object);
}
static gboolean
gst_rtmp2_sink_start (GstBaseSink * sink)
{
GstRtmp2Sink *self = GST_RTMP2_SINK (sink);
gboolean async;
GST_OBJECT_LOCK (self);
async = self->async_connect;
GST_OBJECT_UNLOCK (self);
GST_INFO_OBJECT (self, "Starting (%s)", async ? "async" : "delayed");
g_clear_object (&self->cancellable);
self->running = TRUE;
self->cancellable = g_cancellable_new ();
self->stream_id = 0;
self->last_ts = 0;
self->base_ts = 0;
if (async) {
gst_task_start (self->task);
}
return TRUE;
}
static gboolean
quit_invoker (gpointer user_data)
{
g_main_loop_quit (user_data);
return G_SOURCE_REMOVE;
}
static void
stop_task (GstRtmp2Sink * self)
{
gst_task_stop (self->task);
self->running = FALSE;
if (self->cancellable) {
GST_DEBUG_OBJECT (self, "Cancelling");
g_cancellable_cancel (self->cancellable);
}
if (self->loop) {
GST_DEBUG_OBJECT (self, "Stopping loop");
g_main_context_invoke_full (self->context, G_PRIORITY_DEFAULT_IDLE,
quit_invoker, g_main_loop_ref (self->loop),
(GDestroyNotify) g_main_loop_unref);
}
g_cond_broadcast (&self->cond);
}
static gboolean
gst_rtmp2_sink_stop (GstBaseSink * sink)
{
GstRtmp2Sink *self = GST_RTMP2_SINK (sink);
GST_DEBUG_OBJECT (self, "stop");
g_mutex_lock (&self->lock);
stop_task (self);
g_mutex_unlock (&self->lock);
gst_task_join (self->task);
return TRUE;
}
static gboolean
gst_rtmp2_sink_unlock (GstBaseSink * sink)
{
GstRtmp2Sink *self = GST_RTMP2_SINK (sink);
GST_DEBUG_OBJECT (self, "unlock");
g_mutex_lock (&self->lock);
self->flushing = TRUE;
g_cond_broadcast (&self->cond);
g_mutex_unlock (&self->lock);
return TRUE;
}
static gboolean
gst_rtmp2_sink_unlock_stop (GstBaseSink * sink)
{
GstRtmp2Sink *self = GST_RTMP2_SINK (sink);
GST_DEBUG_OBJECT (self, "unlock_stop");
g_mutex_lock (&self->lock);
self->flushing = FALSE;
g_mutex_unlock (&self->lock);
return TRUE;
}
static gboolean
buffer_to_message (GstRtmp2Sink * self, GstBuffer * buffer, GstBuffer ** outbuf)
{
GstBuffer *message;
gsize payload_offset, payload_size;
guint64 timestamp;
guint32 cstream;
GstRtmpMessageType type;
{
GstMapInfo info;
if (G_UNLIKELY (!gst_buffer_map (buffer, &info, GST_MAP_READ))) {
GST_ERROR_OBJECT (self, "map failed: %" GST_PTR_FORMAT, buffer);
return FALSE;
}
/* FIXME: This is ugly and only works behind flvmux.
* Implement true RTMP muxing. */
if (G_UNLIKELY (info.size >= 4 && memcmp (info.data, "FLV", 3) == 0)) {
/* drop the header, we don't need it */
GST_DEBUG_OBJECT (self, "ignoring FLV header: %" GST_PTR_FORMAT, buffer);
gst_buffer_unmap (buffer, &info);
*outbuf = NULL;
return TRUE;
}
if (G_UNLIKELY (info.size < 11 + 4)) {
GST_ERROR_OBJECT (self, "too small: %" GST_PTR_FORMAT, buffer);
gst_buffer_unmap (buffer, &info);
return FALSE;
}
/* payload between 11 byte header and 4 byte size footer */
payload_offset = 11;
payload_size = info.size - 11 - 4;
type = GST_READ_UINT8 (info.data);
timestamp = GST_READ_UINT24_BE (info.data + 4);
timestamp |= (guint32) GST_READ_UINT8 (info.data + 7) << 24;
/* flvmux timestamps roll over after about 49 days */
if (timestamp + self->base_ts + G_MAXINT32 < self->last_ts) {
GST_WARNING_OBJECT (self, "Timestamp regression %" G_GUINT64_FORMAT
" -> %" G_GUINT64_FORMAT "; assuming overflow", self->last_ts,
timestamp + self->base_ts);
self->base_ts += G_MAXUINT32;
self->base_ts += 1;
} else if (timestamp + self->base_ts > self->last_ts + G_MAXINT32) {
GST_WARNING_OBJECT (self, "Timestamp jump %" G_GUINT64_FORMAT
" -> %" G_GUINT64_FORMAT "; assuming underflow", self->last_ts,
timestamp + self->base_ts);
if (self->base_ts > 0) {
self->base_ts -= G_MAXUINT32;
self->base_ts -= 1;
} else {
GST_WARNING_OBJECT (self, "Cannot regress further;"
" forcing timestamp to zero");
timestamp = 0;
}
}
timestamp += self->base_ts;
self->last_ts = timestamp;
gst_buffer_unmap (buffer, &info);
}
switch (type) {
case GST_RTMP_MESSAGE_TYPE_DATA_AMF0:
cstream = 4;
break;
case GST_RTMP_MESSAGE_TYPE_AUDIO:
cstream = 5;
break;
case GST_RTMP_MESSAGE_TYPE_VIDEO:
cstream = 6;
break;
default:
GST_ERROR_OBJECT (self, "unknown tag type %d", type);
return FALSE;
}
/* May not know stream ID yet; set later */
message = gst_rtmp_message_new (type, cstream, 0);
message = gst_buffer_append_region (message, gst_buffer_ref (buffer),
payload_offset, payload_size);
GST_BUFFER_DTS (message) = timestamp * GST_MSECOND;
if (type == GST_RTMP_MESSAGE_TYPE_DATA_AMF0) {
/* FIXME: HACK: Attach a setDataFrame header.
* This should be done using a command. */
static const guint8 header[] = {
0x02, 0x00, 0x0d, 0x40, 0x73, 0x65, 0x74, 0x44,
0x61, 0x74, 0x61, 0x46, 0x72, 0x61, 0x6d, 0x65
};
GstMemory *memory = gst_memory_new_wrapped (GST_MEMORY_FLAG_READONLY,
(guint8 *) header, sizeof header, 0, sizeof header, NULL, NULL);
gst_buffer_prepend_memory (message, memory);
}
*outbuf = message;
return TRUE;
}
static gboolean
should_drop_header (GstRtmp2Sink * self, GstBuffer * buffer)
{
guint len;
if (G_LIKELY (!GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_HEADER))) {
return FALSE;
}
g_mutex_lock (&self->lock);
len = self->headers->len;
g_mutex_unlock (&self->lock);
/* Drop header buffers when we have streamheader caps */
return len > 0;
}
static void
send_message (GstRtmp2Sink * self, GstBuffer * message)
{
GstRtmpMeta *meta = gst_buffer_get_rtmp_meta (message);
g_return_if_fail (self->stream_id != 0);
meta->mstream = self->stream_id;
gst_rtmp_connection_queue_message (self->connection, message);
}
static void
send_streamheader (GstRtmp2Sink * self)
{
guint i;
if (G_LIKELY (self->headers->len == 0)) {
return;
}
GST_DEBUG_OBJECT (self, "Sending %u streamheader messages",
self->headers->len);
for (i = 0; i < self->headers->len; i++) {
send_message (self, g_ptr_array_index (self->headers, i));
}
/* Steal pointers: suppress free */
g_ptr_array_set_free_func (self->headers, NULL);
g_ptr_array_set_size (self->headers, 0);
g_ptr_array_set_free_func (self->headers,
(GDestroyNotify) gst_mini_object_unref);
}
static inline gboolean
is_running (GstRtmp2Sink * self)
{
return G_LIKELY (self->running && !self->flushing);
}
static GstFlowReturn
gst_rtmp2_sink_render (GstBaseSink * sink, GstBuffer * buffer)
{
GstRtmp2Sink *self = GST_RTMP2_SINK (sink);
GstBuffer *message;
GstFlowReturn ret;
if (G_UNLIKELY (should_drop_header (self, buffer))) {
GST_DEBUG_OBJECT (self, "Skipping header %" GST_PTR_FORMAT, buffer);
return GST_FLOW_OK;
}
GST_LOG_OBJECT (self, "render %" GST_PTR_FORMAT, buffer);
if (G_UNLIKELY (!buffer_to_message (self, buffer, &message))) {
GST_ELEMENT_ERROR (self, STREAM, FAILED, ("Failed to convert FLV to RTMP"),
("Failed to convert %" GST_PTR_FORMAT, message));
return GST_FLOW_ERROR;
}
if (G_UNLIKELY (!message)) {
GST_DEBUG_OBJECT (self, "Skipping %" GST_PTR_FORMAT, buffer);
return GST_FLOW_OK;
}
g_mutex_lock (&self->lock);
if (G_UNLIKELY (is_running (self) && self->cancellable &&
gst_task_get_state (self->task) != GST_TASK_STARTED)) {
GST_DEBUG_OBJECT (self, "Starting connect");
gst_task_start (self->task);
}
while (G_UNLIKELY (is_running (self) && !self->connection)) {
GST_DEBUG_OBJECT (self, "Waiting for connection");
g_cond_wait (&self->cond, &self->lock);
}
while (G_UNLIKELY (is_running (self) && self->connection &&
gst_rtmp_connection_get_num_queued (self->connection) > 3)) {
GST_LOG_OBJECT (self, "Waiting for queue");
g_cond_wait (&self->cond, &self->lock);
}
if (G_UNLIKELY (!is_running (self))) {
gst_buffer_unref (message);
ret = GST_FLOW_FLUSHING;
} else if (G_UNLIKELY (!self->connection)) {
gst_buffer_unref (message);
/* send_connect_error has sent an ERROR message */
ret = GST_FLOW_ERROR;
} else {
send_streamheader (self);
send_message (self, message);
ret = GST_FLOW_OK;
}
g_mutex_unlock (&self->lock);
return ret;
}
static gboolean
add_streamheader (GstRtmp2Sink * self, const GValue * value)
{
GstBuffer *buffer, *message;
g_return_val_if_fail (value, FALSE);
if (!GST_VALUE_HOLDS_BUFFER (value)) {
GST_ERROR_OBJECT (self, "'streamheader' item of unexpected type '%s'",
G_VALUE_TYPE_NAME (value));
return FALSE;
}
buffer = gst_value_get_buffer (value);
if (!buffer_to_message (self, buffer, &message)) {
GST_ERROR_OBJECT (self, "Failed to read streamheader %" GST_PTR_FORMAT,
buffer);
return FALSE;
}
if (message) {
GST_DEBUG_OBJECT (self, "Adding streamheader %" GST_PTR_FORMAT, buffer);
g_ptr_array_add (self->headers, message);
} else {
GST_DEBUG_OBJECT (self, "Skipping streamheader %" GST_PTR_FORMAT, buffer);
}
return TRUE;
}
static gboolean
gst_rtmp2_sink_set_caps (GstBaseSink * sink, GstCaps * caps)
{
GstRtmp2Sink *self = GST_RTMP2_SINK (sink);
GstStructure *s;
const GValue *streamheader;
guint i = 0;
GST_DEBUG_OBJECT (self, "setcaps %" GST_PTR_FORMAT, caps);
g_ptr_array_set_size (self->headers, 0);
s = gst_caps_get_structure (caps, 0);
streamheader = gst_structure_get_value (s, "streamheader");
if (!streamheader) {
GST_DEBUG_OBJECT (self, "'streamheader' field not present");
} else if (GST_VALUE_HOLDS_BUFFER (streamheader)) {
GST_DEBUG_OBJECT (self, "'streamheader' field holds buffer");
if (!add_streamheader (self, streamheader)) {
return FALSE;
}
i = 1;
} else if (GST_VALUE_HOLDS_ARRAY (streamheader)) {
guint size = gst_value_array_get_size (streamheader);
GST_DEBUG_OBJECT (self, "'streamheader' field holds array");
for (; i < size; i++) {
const GValue *v = gst_value_array_get_value (streamheader, i);
if (!add_streamheader (self, v)) {
return FALSE;
}
}
} else {
GST_ERROR_OBJECT (self, "'streamheader' field has unexpected type '%s'",
G_VALUE_TYPE_NAME (streamheader));
return FALSE;
}
GST_DEBUG_OBJECT (self, "Collected streamheaders: %u buffers -> %u messages",
i, self->headers->len);
return TRUE;
}
/* Mainloop task */
static void
gst_rtmp2_sink_task_func (gpointer user_data)
{
GstRtmp2Sink *self = GST_RTMP2_SINK (user_data);
GMainContext *context;
GMainLoop *loop;
GTask *connector;
GST_DEBUG_OBJECT (self, "gst_rtmp2_sink_task starting");
g_mutex_lock (&self->lock);
context = self->context = g_main_context_new ();
g_main_context_push_thread_default (context);
loop = self->loop = g_main_loop_new (context, TRUE);
connector = g_task_new (self, self->cancellable, connect_task_done, NULL);
GST_OBJECT_LOCK (self);
gst_rtmp_client_connect_async (&self->location, self->cancellable,
client_connect_done, connector);
GST_OBJECT_UNLOCK (self);
g_mutex_unlock (&self->lock);
g_main_loop_run (loop);
g_mutex_lock (&self->lock);
g_clear_pointer (&self->loop, g_main_loop_unref);
g_clear_pointer (&self->connection, gst_rtmp_connection_close_and_unref);
g_cond_broadcast (&self->cond);
g_mutex_unlock (&self->lock);
while (g_main_context_pending (context)) {
GST_DEBUG_OBJECT (self, "iterating main context to clean up");
g_main_context_iteration (context, FALSE);
}
g_main_context_pop_thread_default (context);
g_mutex_lock (&self->lock);
g_clear_pointer (&self->context, g_main_context_unref);
g_ptr_array_set_size (self->headers, 0);
g_mutex_unlock (&self->lock);
GST_DEBUG_OBJECT (self, "gst_rtmp2_sink_task exiting");
}
static void
client_connect_done (GObject * source, GAsyncResult * result,
gpointer user_data)
{
GTask *task = user_data;
GstRtmp2Sink *self = g_task_get_source_object (task);
GError *error = NULL;
GstRtmpConnection *connection;
connection = gst_rtmp_client_connect_finish (result, &error);
if (!connection) {
g_task_return_error (task, error);
g_object_unref (task);
return;
}
g_task_set_task_data (task, connection, g_object_unref);
if (g_task_return_error_if_cancelled (task)) {
g_object_unref (task);
return;
}
GST_OBJECT_LOCK (self);
gst_rtmp_client_start_publish_async (connection, self->location.stream,
g_task_get_cancellable (task), start_publish_done, task);
GST_OBJECT_UNLOCK (self);
}
static void
start_publish_done (GObject * source, GAsyncResult * result, gpointer user_data)
{
GTask *task = G_TASK (user_data);
GstRtmp2Sink *self = g_task_get_source_object (task);
GstRtmpConnection *connection = g_task_get_task_data (task);
GError *error = NULL;
if (g_task_return_error_if_cancelled (task)) {
g_object_unref (task);
return;
}
if (gst_rtmp_client_start_publish_finish (connection, result,
&self->stream_id, &error)) {
g_task_return_pointer (task, g_object_ref (connection),
gst_rtmp_connection_close_and_unref);
} else {
g_task_return_error (task, error);
}
g_task_set_task_data (task, NULL, NULL);
g_object_unref (task);
}
static void
put_chunk (GstRtmpConnection * connection, gpointer user_data)
{
GstRtmp2Sink *self = GST_RTMP2_SINK (user_data);
g_mutex_lock (&self->lock);
g_cond_signal (&self->cond);
g_mutex_unlock (&self->lock);
}
static void
error_callback (GstRtmpConnection * connection, GstRtmp2Sink * self)
{
g_mutex_lock (&self->lock);
if (self->cancellable) {
g_cancellable_cancel (self->cancellable);
} else if (self->loop) {
GST_ELEMENT_ERROR (self, RESOURCE, WRITE, ("Connection error"), (NULL));
stop_task (self);
}
g_mutex_unlock (&self->lock);
}
static void
send_connect_error (GstRtmp2Sink * self, GError * error)
{
if (!error) {
GST_ERROR_OBJECT (self, "Connect failed with NULL error");
GST_ELEMENT_ERROR (self, RESOURCE, FAILED, ("Failed to connect"), (NULL));
return;
}
if (g_error_matches (error, G_IO_ERROR, G_IO_ERROR_CANCELLED)) {
GST_DEBUG_OBJECT (self, "Connection was cancelled (%s)",
GST_STR_NULL (error->message));
return;
}
GST_ERROR_OBJECT (self, "Failed to connect (%s:%d): %s",
g_quark_to_string (error->domain), error->code,
GST_STR_NULL (error->message));
if (g_error_matches (error, G_IO_ERROR, G_IO_ERROR_PERMISSION_DENIED)) {
GST_ELEMENT_ERROR (self, RESOURCE, NOT_AUTHORIZED,
("Not authorized to connect"), ("%s", GST_STR_NULL (error->message)));
} else if (g_error_matches (error, G_IO_ERROR, G_IO_ERROR_CONNECTION_REFUSED)) {
GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ,
("Could not connect"), ("%s", GST_STR_NULL (error->message)));
} else {
GST_ELEMENT_ERROR (self, RESOURCE, FAILED,
("Failed to connect"),
("error %s:%d: %s", g_quark_to_string (error->domain), error->code,
GST_STR_NULL (error->message)));
}
}
static void
connect_task_done (GObject * object, GAsyncResult * result, gpointer user_data)
{
GstRtmp2Sink *self = GST_RTMP2_SINK (object);
GTask *task = G_TASK (result);
GError *error = NULL;
g_mutex_lock (&self->lock);
g_warn_if_fail (g_task_is_valid (task, object));
if (self->cancellable == g_task_get_cancellable (task)) {
g_clear_object (&self->cancellable);
}
self->connection = g_task_propagate_pointer (task, &error);
if (self->connection) {
set_pacing_rate (self);
gst_rtmp_connection_set_output_handler (self->connection,
put_chunk, g_object_ref (self), g_object_unref);
g_signal_connect_object (self->connection, "error",
G_CALLBACK (error_callback), self, 0);
} else {
send_connect_error (self, error);
stop_task (self);
g_error_free (error);
}
g_cond_broadcast (&self->cond);
g_mutex_unlock (&self->lock);
}
static gboolean
socket_set_pacing_rate (GSocket * socket, gint pacing_rate, GError ** error)
{
#ifdef SO_MAX_PACING_RATE
if (!g_socket_set_option (socket, SOL_SOCKET, SO_MAX_PACING_RATE,
pacing_rate, error)) {
g_prefix_error (error, "setsockopt failed: ");
return FALSE;
}
#else
if (pacing_rate != -1) {
g_set_error (error, G_IO_ERROR, G_IO_ERROR_NOT_SUPPORTED,
"SO_MAX_PACING_RATE is not supported");
return FALSE;
}
#endif
return TRUE;
}
static void
set_pacing_rate (GstRtmp2Sink * self)
{
GError *error = NULL;
gint pacing_rate;
if (!self->connection)
return;
GST_OBJECT_LOCK (self);
pacing_rate = self->peak_kbps ? self->peak_kbps * 125 : -1;
GST_OBJECT_UNLOCK (self);
if (socket_set_pacing_rate (gst_rtmp_connection_get_socket (self->connection),
pacing_rate, &error))
GST_INFO_OBJECT (self, "Set pacing rate to %d Bps", pacing_rate);
else
GST_WARNING_OBJECT (self, "Could not set pacing rate: %s", error->message);
g_clear_error (&error);
}