Encoding raw data from buffere and store encoded data in another buffer
aasim
mdamirraza at gmail.com
Tue May 8 04:42:48 UTC 2018
Hi guys,
I have very less experience in gstreamer api level code.
kindly help me or guide in right path.
Problem:
I want to inject the buffer (two buffer's ) containing YUV data & PCM data
(read from .yuv & .pcm files) to gstreamer-1.0 pipe line.
Error:
I am getting error in linking the elements.
"Elements could not be linked"
#include <gst/gst.h>
#include <gst/audio/audio.h>
#include <gst/app/gstappsrc.h>
#include <gst/base/gstpushsrc.h>
#include <gst/app/gstappsink.h>
#include <string.h>
#include <stdio.h>
#define CHUNK_SIZE 4096 /* Amount of bytes we are sending in each buffer
*/
#define SAMPLE_RATE 48000 /* 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;//*app_decode,*audio_decode;
GstElement *app_queue, *audio_convert2, *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 */
FILE *fp, *fp1;
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, r;
GstMapInfo map;
printf("sujith2222");
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);
r = fread(map.data, 2, CHUNK_SIZE / 2, data->fp);
gst_buffer_unmap(buffer, &map);
data->num_samples += num_samples;
while (r == NULL)
gst_app_src_end_of_stream((GstAppSrc *)(data->app_source));
/* Push the buffer into the appsrc */
g_signal_emit_by_name(data->app_source, "push-buffer", buffer, &ret);
// gst_app_src_end_of_stream (data->app_source);
//gst_app_src_push_buffer (data->app_source, buffer);
/* 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) {
printf("sujith1111111");
GstSample *sample;
///////////////////////////////////////////////////////
GstBuffer *buffer;
GstMapInfo map;
g_signal_emit_by_name(data->app_sink, "pull-sample", &sample, NULL);
if (sample)
{
buffer = gst_sample_get_buffer(sample);
gst_buffer_map(buffer, &map, GST_MAP_READ);
g_print("\n here size=%d\n", map.size);
fwrite(map.data, 1, map.size, data->fp1); ///data is written to a file
gst_buffer_unmap(buffer, &map);
gst_sample_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 main1(int argc, char *argv[]) {
CustomData data;
GstPad *tee_audio_pad, *tee_app_pad;
GstPad *queue_audio_pad, *queue_app_pad;
GstAudioInfo info;
GstCaps *audio_caps;
GstBus *bus;
/* Initialize cumstom data structure */
memset(&data, 0, sizeof(data));
//data.fp=fopen("/songs/ChoosiChudangane.mp3","rb");
data.fp = fopen("E:/pocVR/ConsoleApplication6/x64/Transformers1080p.pcm",
"rb");
if (data.fp == NULL)
{
printf("\n not bale to open input file \n");
}
data.fp1 = fopen("1.raw", "wb");
/* 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.app_decode = gst_element_factory_make ("decodebin", "app_decode");
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.app_queue = gst_element_factory_make("queue", "app_queue");
//data.audio_decode = gst_element_factory_make ("decodebin",
"audio_decode");
data.audio_convert2 = gst_element_factory_make("audioconvert",
"audio_convert2");
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.audio_convert2 ||
!data.app_queue || !data.app_sink) //||!data.audio_decode||
//!data.app_decode
{
g_printerr("Not all elements could be created.\n");
return -1;
}
/* 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.app_queue, data.audio_convert2, data.app_sink,
NULL);//,data.audio_decode,data.app_decode
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.app_queue,
data.audio_convert2, data.app_sink, NULL) != TRUE)//,data.app_decode ,
data.audio_decode
{
g_printerr("Elements could not be linked.\n");
gst_object_unref(data.pipeline);
return -1;
}
/* Manually link the Tee, which has "Request" pads */
tee_audio_pad = gst_element_get_request_pad(data.tee, "src_%u");
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_app_pad = gst_element_get_request_pad(data.tee, "src_%u");
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_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_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_app_pad);
gst_object_unref(tee_audio_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;
}
-----
adi
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