problem when injecting source file mp3/m3u8 data to appsrc element.
Luca Bacci
luca.bacci982 at gmail.com
Fri Mar 16 11:41:52 UTC 2018
in main, audio_caps is set to GST_AUDIO_FORMAT_S16. then you assign it to
the "caps" property of the appsrc element, I think that's wrong because
appsrc sends compressed (MP3) data to the decodebin, not raw PCM data.
I think audio_caps should instead be set to the "sink-caps" property of
decodebin.
So try changing the line
g_object_set (data.app_source, "caps", audio_caps, "format",
GST_FORMAT_TIME, NULL);
to
g_object_set (data.app_source, "format", GST_FORMAT_TIME, NULL);
g_object_set (data.app_decode, "sink-caps", audio_caps, NULL);
Luca
2018-03-16 7:39 GMT+01:00 Sujith reddy <Sujithreddy6192 at gmail.com>:
> Hi All,
>
> Here i wanted to know how to inject a source mp3/m3u8 file to pipeline
> appsrc.
>
>
> my code is working for raw data i.e wav(PCM) file as injecting source.
> for mp3/m3u8 it is giving noise .
>
>
> Here is mycode can anyone help on this to play mp3/m3u8 by injecting data
> to
> appsrc.
>
> /*****************
>
>
> gcc llll.c -o playback-tutorial-7 `pkg-config --cflags --libs gstreamer-1.0
> gstreamer-audio-1.0 gstreamer-app-1.0`
> *******************/
>
> #include <gstreamer-1.0/gst/gst.h>
> #include <gst/audio/audio.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;
> 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 (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 main(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("./Deviceconnected.raw","rb");
> 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;
> }
>
> Thanks
> Sujith
>
>
>
>
> --
> Sent from: http://gstreamer-devel.966125.n4.nabble.com/
> _______________________________________________
> gstreamer-devel mailing list
> gstreamer-devel at lists.freedesktop.org
> https://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>
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