mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-28 19:20:35 +00:00
540 lines
23 KiB
Markdown
540 lines
23 KiB
Markdown
# Basic tutorial 8: Short-cutting the pipeline
|
|
|
|
## Goal
|
|
|
|
Pipelines constructed with GStreamer do not need to be completely
|
|
closed. Data can be injected into the pipeline and extracted from it at
|
|
any time, in a variety of ways. This tutorial shows:
|
|
|
|
- How to inject external data into a general GStreamer pipeline.
|
|
|
|
- How to extract data from a general GStreamer pipeline.
|
|
|
|
- How to access and manipulate this data.
|
|
|
|
[](tutorials/playback/short-cutting-the-pipeline.md) explains
|
|
how to achieve the same goals in a playbin-based pipeline.
|
|
|
|
## Introduction
|
|
|
|
Applications can interact with the data flowing through a GStreamer
|
|
pipeline in several ways. This tutorial describes the easiest one, since
|
|
it uses elements that have been created for this sole purpose.
|
|
|
|
The element used to inject application data into a GStreamer pipeline is
|
|
`appsrc`, and its counterpart, used to extract GStreamer data back to
|
|
the application is `appsink`. To avoid confusing the names, think of it
|
|
from GStreamer's point of view: `appsrc` is just a regular source, that
|
|
provides data magically fallen from the sky (provided by the
|
|
application, actually). `appsink` is a regular sink, where the data
|
|
flowing through a GStreamer pipeline goes to die (it is recovered by the
|
|
application, actually).
|
|
|
|
`appsrc` and `appsink` are so versatile that they offer their own API
|
|
(see their documentation), which can be accessed by linking against the
|
|
`gstreamer-app` library. In this tutorial, however, we will use a
|
|
simpler approach and control them through signals.
|
|
|
|
`appsrc` can work in a variety of modes: in **pull** mode, it requests
|
|
data from the application every time it needs it. In **push** mode, the
|
|
application pushes data at its own pace. Furthermore, in push mode, the
|
|
application can choose to be blocked in the push function when enough
|
|
data has already been provided, or it can listen to the
|
|
`enough-data` and `need-data` signals to control flow. This example
|
|
implements the latter approach. Information regarding the other methods
|
|
can be found in the `appsrc` documentation.
|
|
|
|
### Buffers
|
|
|
|
Data travels through a GStreamer pipeline in chunks called **buffers**.
|
|
Since this example produces and consumes data, we need to know about
|
|
`GstBuffer`s.
|
|
|
|
Source Pads produce buffers, that are consumed by Sink Pads; GStreamer
|
|
takes these buffers and passes them from element to element.
|
|
|
|
A buffer simply represents a unit of data, do not assume that all
|
|
buffers will have the same size, or represent the same amount of time.
|
|
Neither should you assume that if a single buffer enters an element, a
|
|
single buffer will come out. Elements are free to do with the received
|
|
buffers as they please. `GstBuffer`s may also contain more than one
|
|
actual memory buffer. Actual memory buffers are abstracted away using
|
|
`GstMemory` objects, and a `GstBuffer` can contain multiple `GstMemory` objects.
|
|
|
|
Every buffer has attached time-stamps and duration, that describe in
|
|
which moment the content of the buffer should be decoded, rendered or
|
|
displayed. Time stamping is a very complex and delicate subject, but
|
|
this simplified vision should suffice for now.
|
|
|
|
As an example, a `filesrc` (a GStreamer element that reads files)
|
|
produces buffers with the “ANY” caps and no time-stamping information.
|
|
After demuxing (see [](tutorials/basic/dynamic-pipelines.md))
|
|
buffers can have some specific caps, for example “video/x-h264”. After
|
|
decoding, each buffer will contain a single video frame with raw caps
|
|
(for example, “video/x-raw-yuv”) and very precise time stamps indicating
|
|
when should that frame be displayed.
|
|
|
|
### This tutorial
|
|
|
|
This tutorial expands [](tutorials/basic/multithreading-and-pad-availability.md) in
|
|
two ways: firstly, the `audiotestsrc` is replaced by an `appsrc` that
|
|
will generate the audio data. Secondly, a new branch is added to the
|
|
`tee` so data going into the audio sink and the wave display is also
|
|
replicated into an `appsink`. The `appsink` uploads the information back
|
|
into the application, which then just notifies the user that data has
|
|
been received, but it could obviously perform more complex tasks.
|
|
|
|
|
|
|
|
## A crude waveform generator
|
|
|
|
Copy this code into a text file named `basic-tutorial-8.c` (or find it
|
|
in your GStreamer installation).
|
|
|
|
``` c
|
|
#include <gst/gst.h>
|
|
#include <gst/audio/audio.h>
|
|
#include <string.h>
|
|
|
|
#define CHUNK_SIZE 1024 /* Amount of bytes we are sending in each buffer */
|
|
#define SAMPLE_RATE 44100 /* Samples per second we are sending */
|
|
|
|
/* Structure to contain all our information, so we can pass it to callbacks */
|
|
typedef struct _CustomData {
|
|
GstElement *pipeline, *app_source, *tee, *audio_queue, *audio_convert1, *audio_resample, *audio_sink;
|
|
GstElement *video_queue, *audio_convert2, *visual, *video_convert, *video_sink;
|
|
GstElement *app_queue, *app_sink;
|
|
|
|
guint64 num_samples; /* Number of samples generated so far (for timestamp generation) */
|
|
gfloat a, b, c, d; /* For waveform generation */
|
|
|
|
guint sourceid; /* To control the GSource */
|
|
|
|
GMainLoop *main_loop; /* GLib's Main Loop */
|
|
} CustomData;
|
|
|
|
/* This method is called by the idle GSource in the mainloop, to feed CHUNK_SIZE bytes into appsrc.
|
|
* The ide handler is added to the mainloop when appsrc requests us to start sending data (need-data signal)
|
|
* and is removed when appsrc has enough data (enough-data signal).
|
|
*/
|
|
static gboolean push_data (CustomData *data) {
|
|
GstBuffer *buffer;
|
|
GstFlowReturn ret;
|
|
int i;
|
|
GstMapInfo map;
|
|
gint16 *raw;
|
|
gint num_samples = CHUNK_SIZE / 2; /* Because each sample is 16 bits */
|
|
gfloat freq;
|
|
|
|
/* Create a new empty buffer */
|
|
buffer = gst_buffer_new_and_alloc (CHUNK_SIZE);
|
|
|
|
/* Set its timestamp and duration */
|
|
GST_BUFFER_TIMESTAMP (buffer) = gst_util_uint64_scale (data->num_samples, GST_SECOND, SAMPLE_RATE);
|
|
GST_BUFFER_DURATION (buffer) = gst_util_uint64_scale (CHUNK_SIZE, GST_SECOND, SAMPLE_RATE);
|
|
|
|
/* Generate some psychodelic waveforms */
|
|
gst_buffer_map (buffer, &map, GST_MAP_WRITE);
|
|
raw = (gint16 *)map.data;
|
|
data->c += data->d;
|
|
data->d -= data->c / 1000;
|
|
freq = 1100 + 1000 * data->d;
|
|
for (i = 0; i < num_samples; i++) {
|
|
data->a += data->b;
|
|
data->b -= data->a / freq;
|
|
raw[i] = (gint16)(500 * data->a);
|
|
}
|
|
gst_buffer_unmap (buffer, &map);
|
|
data->num_samples += num_samples;
|
|
|
|
/* Push the buffer into the appsrc */
|
|
g_signal_emit_by_name (data->app_source, "push-buffer", buffer, &ret);
|
|
|
|
/* Free the buffer now that we are done with it */
|
|
gst_buffer_unref (buffer);
|
|
|
|
if (ret != GST_FLOW_OK) {
|
|
/* We got some error, stop sending data */
|
|
return FALSE;
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/* This signal callback triggers when appsrc needs data. Here, we add an idle handler
|
|
* to the mainloop to start pushing data into the appsrc */
|
|
static void start_feed (GstElement *source, guint size, CustomData *data) {
|
|
if (data->sourceid == 0) {
|
|
g_print ("Start feeding\n");
|
|
data->sourceid = g_idle_add ((GSourceFunc) push_data, data);
|
|
}
|
|
}
|
|
|
|
/* This callback triggers when appsrc has enough data and we can stop sending.
|
|
* We remove the idle handler from the mainloop */
|
|
static void stop_feed (GstElement *source, CustomData *data) {
|
|
if (data->sourceid != 0) {
|
|
g_print ("Stop feeding\n");
|
|
g_source_remove (data->sourceid);
|
|
data->sourceid = 0;
|
|
}
|
|
}
|
|
|
|
/* The appsink has received a buffer */
|
|
static void new_sample (GstElement *sink, CustomData *data) {
|
|
GstSample *sample;
|
|
|
|
/* Retrieve the buffer */
|
|
g_signal_emit_by_name (sink, "pull-sample", &sample);
|
|
if (sample) {
|
|
/* The only thing we do in this example is print a * to indicate a received buffer */
|
|
g_print ("*");
|
|
gst_buffer_unref (sample);
|
|
}
|
|
}
|
|
|
|
/* This function is called when an error message is posted on the bus */
|
|
static void error_cb (GstBus *bus, GstMessage *msg, CustomData *data) {
|
|
GError *err;
|
|
gchar *debug_info;
|
|
|
|
/* Print error details on the screen */
|
|
gst_message_parse_error (msg, &err, &debug_info);
|
|
g_printerr ("Error received from element %s: %s\n", GST_OBJECT_NAME (msg->src), err->message);
|
|
g_printerr ("Debugging information: %s\n", debug_info ? debug_info : "none");
|
|
g_clear_error (&err);
|
|
g_free (debug_info);
|
|
|
|
g_main_loop_quit (data->main_loop);
|
|
}
|
|
|
|
int main(int argc, char *argv[]) {
|
|
CustomData data;
|
|
GstPadTemplate *tee_src_pad_template;
|
|
GstPad *tee_audio_pad, *tee_video_pad, *tee_app_pad;
|
|
GstPad *queue_audio_pad, *queue_video_pad, *queue_app_pad;
|
|
GstAudioInfo info;
|
|
GstCaps *audio_caps;
|
|
GstBus *bus;
|
|
|
|
/* Initialize cumstom data structure */
|
|
memset (&data, 0, sizeof (data));
|
|
data.b = 1; /* For waveform generation */
|
|
data.d = 1;
|
|
|
|
/* Initialize GStreamer */
|
|
gst_init (&argc, &argv);
|
|
|
|
/* Create the elements */
|
|
data.app_source = gst_element_factory_make ("appsrc", "audio_source");
|
|
data.tee = gst_element_factory_make ("tee", "tee");
|
|
data.audio_queue = gst_element_factory_make ("queue", "audio_queue");
|
|
data.audio_convert1 = gst_element_factory_make ("audioconvert", "audio_convert1");
|
|
data.audio_resample = gst_element_factory_make ("audioresample", "audio_resample");
|
|
data.audio_sink = gst_element_factory_make ("autoaudiosink", "audio_sink");
|
|
data.video_queue = gst_element_factory_make ("queue", "video_queue");
|
|
data.audio_convert2 = gst_element_factory_make ("audioconvert", "audio_convert2");
|
|
data.visual = gst_element_factory_make ("wavescope", "visual");
|
|
data.video_convert = gst_element_factory_make ("videoconvert", "csp");
|
|
data.video_sink = gst_element_factory_make ("autovideosink", "video_sink");
|
|
data.app_queue = gst_element_factory_make ("queue", "app_queue");
|
|
data.app_sink = gst_element_factory_make ("appsink", "app_sink");
|
|
|
|
/* Create the empty pipeline */
|
|
data.pipeline = gst_pipeline_new ("test-pipeline");
|
|
|
|
if (!data.pipeline || !data.app_source || !data.tee || !data.audio_queue || !data.audio_convert1 ||
|
|
!data.audio_resample || !data.audio_sink || !data.video_queue || !data.audio_convert2 || !data.visual ||
|
|
!data.video_convert || !data.video_sink || !data.app_queue || !data.app_sink) {
|
|
g_printerr ("Not all elements could be created.\n");
|
|
return -1;
|
|
}
|
|
|
|
/* Configure wavescope */
|
|
g_object_set (data.visual, "shader", 0, "style", 0, NULL);
|
|
|
|
/* Configure appsrc */
|
|
gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16, SAMPLE_RATE, 1, NULL);
|
|
audio_caps = gst_audio_info_to_caps (&info);
|
|
g_object_set (data.app_source, "caps", audio_caps, "format", GST_FORMAT_TIME, NULL);
|
|
g_signal_connect (data.app_source, "need-data", G_CALLBACK (start_feed), &data);
|
|
g_signal_connect (data.app_source, "enough-data", G_CALLBACK (stop_feed), &data);
|
|
|
|
/* Configure appsink */
|
|
g_object_set (data.app_sink, "emit-signals", TRUE, "caps", audio_caps, NULL);
|
|
g_signal_connect (data.app_sink, "new-sample", G_CALLBACK (new_sample), &data);
|
|
gst_caps_unref (audio_caps);
|
|
g_free (audio_caps_text);
|
|
|
|
/* Link all elements that can be automatically linked because they have "Always" pads */
|
|
gst_bin_add_many (GST_BIN (data.pipeline), data.app_source, data.tee, data.audio_queue, data.audio_convert1, data.audio_resample,
|
|
data.audio_sink, data.video_queue, data.audio_convert2, data.visual, data.video_convert, data.video_sink, data.app_queue,
|
|
data.app_sink, NULL);
|
|
if (gst_element_link_many (data.app_source, data.tee, NULL) != TRUE ||
|
|
gst_element_link_many (data.audio_queue, data.audio_convert1, data.audio_resample, data.audio_sink, NULL) != TRUE ||
|
|
gst_element_link_many (data.video_queue, data.audio_convert2, data.visual, data.video_convert, data.video_sink, NULL) != TRUE ||
|
|
gst_element_link_many (data.app_queue, data.app_sink, NULL) != TRUE) {
|
|
g_printerr ("Elements could not be linked.\n");
|
|
gst_object_unref (data.pipeline);
|
|
return -1;
|
|
}
|
|
|
|
/* Manually link the Tee, which has "Request" pads */
|
|
tee_src_pad_template = gst_element_class_get_pad_template (GST_ELEMENT_GET_CLASS (data.tee), "src_%d");
|
|
tee_audio_pad = gst_element_request_pad (data.tee, tee_src_pad_template, NULL, NULL);
|
|
g_print ("Obtained request pad %s for audio branch.\n", gst_pad_get_name (tee_audio_pad));
|
|
queue_audio_pad = gst_element_get_static_pad (data.audio_queue, "sink");
|
|
tee_video_pad = gst_element_request_pad (data.tee, tee_src_pad_template, NULL, NULL);
|
|
g_print ("Obtained request pad %s for video branch.\n", gst_pad_get_name (tee_video_pad));
|
|
queue_video_pad = gst_element_get_static_pad (data.video_queue, "sink");
|
|
tee_app_pad = gst_element_request_pad (data.tee, tee_src_pad_template, NULL, NULL);
|
|
g_print ("Obtained request pad %s for app branch.\n", gst_pad_get_name (tee_app_pad));
|
|
queue_app_pad = gst_element_get_static_pad (data.app_queue, "sink");
|
|
if (gst_pad_link (tee_audio_pad, queue_audio_pad) != GST_PAD_LINK_OK ||
|
|
gst_pad_link (tee_video_pad, queue_video_pad) != GST_PAD_LINK_OK ||
|
|
gst_pad_link (tee_app_pad, queue_app_pad) != GST_PAD_LINK_OK) {
|
|
g_printerr ("Tee could not be linked\n");
|
|
gst_object_unref (data.pipeline);
|
|
return -1;
|
|
}
|
|
gst_object_unref (queue_audio_pad);
|
|
gst_object_unref (queue_video_pad);
|
|
gst_object_unref (queue_app_pad);
|
|
|
|
/* Instruct the bus to emit signals for each received message, and connect to the interesting signals */
|
|
bus = gst_element_get_bus (data.pipeline);
|
|
gst_bus_add_signal_watch (bus);
|
|
g_signal_connect (G_OBJECT (bus), "message::error", (GCallback)error_cb, &data);
|
|
gst_object_unref (bus);
|
|
|
|
/* Start playing the pipeline */
|
|
gst_element_set_state (data.pipeline, GST_STATE_PLAYING);
|
|
|
|
/* Create a GLib Main Loop and set it to run */
|
|
data.main_loop = g_main_loop_new (NULL, FALSE);
|
|
g_main_loop_run (data.main_loop);
|
|
|
|
/* Release the request pads from the Tee, and unref them */
|
|
gst_element_release_request_pad (data.tee, tee_audio_pad);
|
|
gst_element_release_request_pad (data.tee, tee_video_pad);
|
|
gst_element_release_request_pad (data.tee, tee_app_pad);
|
|
gst_object_unref (tee_audio_pad);
|
|
gst_object_unref (tee_video_pad);
|
|
gst_object_unref (tee_app_pad);
|
|
|
|
/* Free resources */
|
|
gst_element_set_state (data.pipeline, GST_STATE_NULL);
|
|
gst_object_unref (data.pipeline);
|
|
return 0;
|
|
}
|
|
```
|
|
|
|
> 
|
|
> Need help?
|
|
>
|
|
> If you need help to compile this code, refer to the **Building the tutorials** section for your platform: [Linux](installing/on-linux.md#InstallingonLinux-Build), [Mac OS X](installing/on-mac-osx.md#InstallingonMacOSX-Build) or [Windows](installing/on-windows.md#InstallingonWindows-Build), or use this specific command on Linux:
|
|
>
|
|
> `` gcc basic-tutorial-8.c -o basic-tutorial-8 `pkg-config --cflags --libs gstreamer-1.0 gst-audio-1.0` ``
|
|
>
|
|
>If you need help to run this code, refer to the **Running the tutorials** section for your platform: [Linux](installing/on-linux.md#InstallingonLinux-Run), [Mac OS X](installing/on-mac-osx.md#InstallingonMacOSX-Run) or [Windows](installing/on-windows.md#InstallingonWindows-Run).
|
|
>
|
|
> This tutorial plays an audible tone for varying frequency through the audio card and opens a window with a waveform representation of the tone. The waveform should be a sinusoid, but due to the refreshing of the window might not appear so.
|
|
>
|
|
> Required libraries: `gstreamer-1.0`
|
|
|
|
## Walkthrough
|
|
|
|
The code to create the pipeline (Lines 131 to 205) is an enlarged
|
|
version of [Basic tutorial 7: Multithreading and Pad
|
|
Availability](tutorials/basic/multithreading-and-pad-availability.md).
|
|
It involves instantiating all the elements, link the elements with
|
|
Always Pads, and manually link the Request Pads of the `tee` element.
|
|
|
|
Regarding the configuration of the `appsrc` and `appsink` elements:
|
|
|
|
``` c
|
|
/* Configure appsrc */
|
|
audio_caps_text = g_strdup_printf (AUDIO_CAPS, SAMPLE_RATE);
|
|
audio_caps = gst_caps_from_string (audio_caps_text);
|
|
g_object_set (data.app_source, "caps", audio_caps, NULL);
|
|
g_signal_connect (data.app_source, "need-data", G_CALLBACK (start_feed), &data);
|
|
g_signal_connect (data.app_source, "enough-data", G_CALLBACK (stop_feed), &data);
|
|
```
|
|
|
|
The first property that needs to be set on the `appsrc` is `caps`. It
|
|
specifies the kind of data that the element is going to produce, so
|
|
GStreamer can check if linking with downstream elements is possible
|
|
(this is, if the downstream elements will understand this kind of data).
|
|
This property must be a `GstCaps` object, which is easily built from a
|
|
string with `gst_caps_from_string()`.
|
|
|
|
We then connect to the `need-data` and `enough-data` signals. These are
|
|
fired by `appsrc` when its internal queue of data is running low or
|
|
almost full, respectively. We will use these signals to start and stop
|
|
(respectively) our signal generation process.
|
|
|
|
``` c
|
|
/* Configure appsink */
|
|
g_object_set (data.app_sink, "emit-signals", TRUE, "caps", audio_caps, NULL);
|
|
g_signal_connect (data.app_sink, "new-sample", G_CALLBACK (new_sample), &data);
|
|
gst_caps_unref (audio_caps);
|
|
g_free (audio_caps_text);
|
|
```
|
|
|
|
Regarding the `appsink` configuration, we connect to the
|
|
`new-sample` signal, which is emitted every time the sink receives a
|
|
buffer. Also, the signal emission needs to be enabled through the
|
|
`emit-signals` property, because, by default, it is disabled.
|
|
|
|
Starting the pipeline, waiting for messages and final cleanup is done as
|
|
usual. Let's review the callbacks we have just
|
|
registered:
|
|
|
|
``` c
|
|
/* This signal callback triggers when appsrc needs data. Here, we add an idle handler
|
|
* to the mainloop to start pushing data into the appsrc */
|
|
static void start_feed (GstElement *source, guint size, CustomData *data) {
|
|
if (data->sourceid == 0) {
|
|
g_print ("Start feeding\n");
|
|
data->sourceid = g_idle_add ((GSourceFunc) push_data, data);
|
|
}
|
|
}
|
|
```
|
|
|
|
This function is called when the internal queue of `appsrc` is about to
|
|
starve (run out of data). The only thing we do here is register a GLib
|
|
idle function with `g_idle_add()` that feeds data to `appsrc` until it
|
|
is full again. A GLib idle function is a method that GLib will call from
|
|
its main loop whenever it is “idle”, this is, when it has no
|
|
higher-priority tasks to perform. It requires a GLib `GMainLoop` to be
|
|
instantiated and running, obviously.
|
|
|
|
This is only one of the multiple approaches that `appsrc` allows. In
|
|
particular, buffers do not need to be fed into `appsrc` from the main
|
|
thread using GLib, and you do not need to use the `need-data` and
|
|
`enough-data` signals to synchronize with `appsrc` (although this is
|
|
allegedly the most convenient).
|
|
|
|
We take note of the sourceid that `g_idle_add()` returns, so we can
|
|
disable it
|
|
later.
|
|
|
|
``` c
|
|
/* This callback triggers when appsrc has enough data and we can stop sending.
|
|
* We remove the idle handler from the mainloop */
|
|
static void stop_feed (GstElement *source, CustomData *data) {
|
|
if (data->sourceid != 0) {
|
|
g_print ("Stop feeding\n");
|
|
g_source_remove (data->sourceid);
|
|
data->sourceid = 0;
|
|
}
|
|
}
|
|
```
|
|
|
|
This function is called when the internal queue of `appsrc` is full
|
|
enough so we stop pushing data. Here we simply remove the idle function
|
|
by using `g_source_remove()` (The idle function is implemented as a
|
|
`GSource`).
|
|
|
|
``` c
|
|
/* This method is called by the idle GSource in the mainloop, to feed CHUNK_SIZE bytes into appsrc.
|
|
* The ide handler is added to the mainloop when appsrc requests us to start sending data (need-data signal)
|
|
* and is removed when appsrc has enough data (enough-data signal).
|
|
*/
|
|
static gboolean push_data (CustomData *data) {
|
|
GstBuffer *buffer;
|
|
GstFlowReturn ret;
|
|
int i;
|
|
gint16 *raw;
|
|
gint num_samples = CHUNK_SIZE / 2; /* Because each sample is 16 bits */
|
|
gfloat freq;
|
|
|
|
/* Create a new empty buffer */
|
|
buffer = gst_buffer_new_and_alloc (CHUNK_SIZE);
|
|
|
|
/* Set its timestamp and duration */
|
|
GST_BUFFER_TIMESTAMP (buffer) = gst_util_uint64_scale (data->num_samples, GST_SECOND, SAMPLE_RATE);
|
|
GST_BUFFER_DURATION (buffer) = gst_util_uint64_scale (CHUNK_SIZE, GST_SECOND, SAMPLE_RATE);
|
|
|
|
/* Generate some psychodelic waveforms */
|
|
raw = (gint16 *)GST_BUFFER_DATA (buffer);
|
|
```
|
|
|
|
This is the function that feeds `appsrc`. It will be called by GLib at
|
|
times and rates which are out of our control, but we know that we will
|
|
disable it when its job is done (when the queue in `appsrc` is full).
|
|
|
|
Its first task is to create a new buffer with a given size (in this
|
|
example, it is arbitrarily set to 1024 bytes) with
|
|
`gst_buffer_new_and_alloc()`.
|
|
|
|
We count the number of samples that we have generated so far with the
|
|
`CustomData.num_samples` variable, so we can time-stamp this buffer
|
|
using the `GST_BUFFER_TIMESTAMP` macro in `GstBuffer`.
|
|
|
|
Since we are producing buffers of the same size, their duration is the
|
|
same and is set using the `GST_BUFFER_DURATION` in `GstBuffer`.
|
|
|
|
`gst_util_uint64_scale()` is a utility function that scales (multiply
|
|
and divide) numbers which can be large, without fear of overflows.
|
|
|
|
The bytes that for the buffer can be accessed with GST\_BUFFER\_DATA in
|
|
`GstBuffer` (Be careful not to write past the end of the buffer: you
|
|
allocated it, so you know its size).
|
|
|
|
We will skip over the waveform generation, since it is outside the scope
|
|
of this tutorial (it is simply a funny way of generating a pretty
|
|
psychedelic wave).
|
|
|
|
``` c
|
|
/* Push the buffer into the appsrc */
|
|
g_signal_emit_by_name (data->app_source, "push-buffer", buffer, &ret);
|
|
|
|
/* Free the buffer now that we are done with it */
|
|
gst_buffer_unref (buffer);
|
|
```
|
|
|
|
Once we have the buffer ready, we pass it to `appsrc` with the
|
|
`push-buffer` action signal (see information box at the end of [](tutorials/playback/playbin-usage.md)), and then
|
|
`gst_buffer_unref()` it since we no longer need it.
|
|
|
|
``` c
|
|
/* The appsink has received a buffer */
|
|
static void new_sample (GstElement *sink, CustomData *data) {
|
|
GstSample *sample;
|
|
/* Retrieve the buffer */
|
|
g_signal_emit_by_name (sink, "pull-sample", &sample);
|
|
if (sample) {
|
|
/* The only thing we do in this example is print a * to indicate a received buffer */
|
|
g_print ("*");
|
|
gst_sample_unref (sample);
|
|
}
|
|
}
|
|
```
|
|
|
|
Finally, this is the function that gets called when the
|
|
`appsink` receives a buffer. We use the `pull-sample` action signal to
|
|
retrieve the buffer and then just print some indicator on the screen. We
|
|
can retrieve the data pointer using the `GST_BUFFER_DATA` macro and the
|
|
data size using the `GST_BUFFER_SIZE` macro in `GstBuffer`. Remember
|
|
that this buffer does not have to match the buffer that we produced in
|
|
the `push_data` function, any element in the path could have altered the
|
|
buffers in any way (Not in this example: there is only a `tee` in the
|
|
path between `appsrc` and `appsink`, and it does not change the content
|
|
of the buffers).
|
|
|
|
We then `gst_buffer_unref()` the buffer, and this tutorial is done.
|
|
|
|
## Conclusion
|
|
|
|
This tutorial has shown how applications can:
|
|
|
|
- Inject data into a pipeline using the `appsrc`element.
|
|
- Retrieve data from a pipeline using the `appsink` element.
|
|
- Manipulate this data by accessing the `GstBuffer`.
|
|
|
|
In a playbin-based pipeline, the same goals are achieved in a slightly
|
|
different way. [](tutorials/playback/short-cutting-the-pipeline.md) shows
|
|
how to do it.
|
|
|
|
It has been a pleasure having you here, and see you soon\!
|