292 lines
13 KiB
C++
292 lines
13 KiB
C++
#include <gst/gst.h>
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#include <gst/audio/audio.h>
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#include <gst/video/gstvideometa.h>
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#include <gst/app/app.h>
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#include <cstring>
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#include <iostream>
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#include <cassert>
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#include <sstream>
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#define CHUNK_SIZE 1024 /* Amount of bytes we are sending in each buffer */
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#define SAMPLE_RATE 44100 /* Samples per second we are sending */
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/* Structure to contain all our information, so we can pass it to callbacks */
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typedef struct _CustomData {
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GstElement *pipeline, *app_source, *tee, *audio_queue, *audio_convert1, *audio_resample, *audio_sink;
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GstElement *video_queue, *audio_convert2, *visual, *video_convert, *video_sink;
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GstElement *app_queue, *app_sink;
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guint64 num_samples; /* Number of samples generated so far (for timestamp generation) */
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gfloat a, b, c, d; /* For waveform generation */
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guint sourceid; /* To control the GSource */
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GMainLoop *main_loop; /* GLib's Main Loop */
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} CustomData;
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/* This method is called by the idle GSource in the mainloop, to feed CHUNK_SIZE bytes into appsrc.
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* The idle handler is added to the mainloop when appsrc requests us to start sending data (need-data signal)
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* and is removed when appsrc has enough data (enough-data signal).
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*/
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static gboolean push_data(CustomData *data) {
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GstBuffer *buffer = nullptr;
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GstFlowReturn ret;
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int i;
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GstMapInfo map;
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gint16 *raw;
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gint num_samples = CHUNK_SIZE / 2; /* Because each sample is 16 bits */
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gfloat freq;
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/* Create a new empty buffer */
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buffer = gst_buffer_new_and_alloc (CHUNK_SIZE);
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/* Set its timestamp and duration */
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GST_BUFFER_TIMESTAMP (buffer) = gst_util_uint64_scale(data->num_samples, GST_SECOND, SAMPLE_RATE);
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GST_BUFFER_DURATION (buffer) = gst_util_uint64_scale(num_samples, GST_SECOND, SAMPLE_RATE);
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static int my_counter_ = 0;
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my_counter_ += 1;
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std::string dont_blame_me = "abc fuck this " + std::to_string(my_counter_) + "X";
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auto my_meta = gst_buffer_add_video_caption_meta(
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buffer, GST_VIDEO_CAPTION_TYPE_CEA608_RAW, (unsigned char *) dont_blame_me.c_str(), dont_blame_me.size());
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// std::cout << "My GstMeta: " << my_meta << std::endl;
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auto tc = gst_video_time_code_new(
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30, 1, nullptr, GST_VIDEO_TIME_CODE_FLAGS_NONE, 12, 12, 12, 12, 0);
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auto my_meta_2 = gst_buffer_add_video_time_code_meta(buffer, tc);
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/* Generate some psychodelic waveforms */
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gst_buffer_map(buffer, &map, GST_MAP_WRITE);
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raw = (gint16 *) map.data;
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data->c += data->d;
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data->d -= data->c / 1000;
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freq = 1100 + 1000 * data->d;
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for (i = 0; i < num_samples; i++) {
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data->a += data->b;
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data->b -= data->a / freq;
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raw[i] = (gint16) (500 * data->a);
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}
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gst_buffer_unmap(buffer, &map);
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data->num_samples += num_samples;
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/* Push the buffer into the appsrc */
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g_signal_emit_by_name(data->app_source, "push-buffer", buffer, &ret);
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/* Free the buffer now that we are done with it */
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gst_buffer_unref(buffer);
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if (ret != GST_FLOW_OK) {
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/* We got some error, stop sending data */
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return FALSE;
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}
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return TRUE;
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}
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/* This signal callback triggers when appsrc needs data. Here, we add an idle handler
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* to the mainloop to start pushing data into the appsrc */
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static void start_feed(GstElement *source, guint size, CustomData *data) {
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if (data->sourceid == 0) {
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g_print("Start feeding\n");
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data->sourceid = g_idle_add((GSourceFunc) push_data, data);
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}
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}
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/* This callback triggers when appsrc has enough data and we can stop sending.
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* We remove the idle handler from the mainloop */
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static void stop_feed(GstElement *source, CustomData *data) {
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if (data->sourceid != 0) {
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g_print("Stop feeding\n");
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g_source_remove(data->sourceid);
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data->sourceid = 0;
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}
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}
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/* The appsink has received a buffer */
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static GstFlowReturn new_sample(GstElement *sink, CustomData *data) {
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GstSample *sample;
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/* Retrieve the buffer */
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g_signal_emit_by_name(sink, "pull-sample", &sample);
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if (sample) {
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/* The only thing we do in this example is print a * to indicate a received buffer */
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g_print("*");
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GstBuffer *buffer = gst_sample_get_buffer(sample);
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const auto &n_memory = gst_buffer_n_memory(buffer);
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std::cout << "n_memory = " << n_memory << std::endl;
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std::cout << "buffer->pts = " << buffer->pts << std::endl;
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std::cout << "buffer->dts = " << buffer->dts << std::endl;
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std::cout << "buffer->duration = " << buffer->duration << std::endl;
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std::cout << "buffer->offset = " << buffer->offset << std::endl;
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std::cout << "buffer->offset_end = " << buffer->offset_end << std::endl;
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GstMeta *gst_meta;
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gpointer state = nullptr;
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while ((gst_meta = gst_buffer_iterate_meta(buffer, &state))) {
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if (gst_meta->info == gst_video_caption_meta_get_info()) {
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auto specific_meta = (GstVideoCaptionMeta *) gst_meta;
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if (specific_meta) {
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auto x = (const char *) (specific_meta->data);
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std::cout << "MetaInfo is recognized to be [GstVideoCaptionMeta]"
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<< "caption = " << std::string(x, specific_meta->size)
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<< std::endl;
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}
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} else if (gst_meta->info == gst_video_time_code_meta_get_info()) {
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auto specific_meta = (GstVideoTimeCodeMeta *) gst_meta;
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if (specific_meta) {
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std::cout << "MetaInfo is recognized to be [GstVideoTimeCodeMeta]"
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<< " h = " << specific_meta->tc.hours
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<< " m = " << specific_meta->tc.minutes
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<< " s = " << specific_meta->tc.seconds
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<< " f = " << specific_meta->tc.frames
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<< std::endl;
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}
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} else {
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std::cout << "GstMetaInfo is not recognized."
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<< " info = " << gst_meta->info
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<< " api = " << gst_meta->info->api
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<< std::endl;
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}
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}
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gst_sample_unref(sample);
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return GST_FLOW_OK;
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}
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return GST_FLOW_ERROR;
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}
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/* This function is called when an error message is posted on the bus */
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static void error_cb(GstBus *bus, GstMessage *msg, CustomData *data) {
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GError *err;
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gchar *debug_info;
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/* Print error details on the screen */
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gst_message_parse_error(msg, &err, &debug_info);
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g_printerr("Error received from element %s: %s\n", GST_OBJECT_NAME (msg->src), err->message);
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g_printerr("Debugging information: %s\n", debug_info ? debug_info : "none");
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g_clear_error(&err);
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g_free(debug_info);
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g_main_loop_quit(data->main_loop);
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}
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int main(int argc, char *argv[]) {
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CustomData data;
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GstPad *tee_audio_pad, *tee_video_pad, *tee_app_pad;
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GstPad *queue_audio_pad, *queue_video_pad, *queue_app_pad;
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GstAudioInfo info;
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GstCaps *audio_caps;
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GstBus *bus;
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/* Initialize custom data structure */
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memset(&data, 0, sizeof(data));
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data.b = 1; /* For waveform generation */
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data.d = 1;
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/* Initialize GStreamer */
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gst_init(&argc, &argv);
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/* Create the elements */
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data.app_source = gst_element_factory_make("appsrc", "audio_source");
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data.tee = gst_element_factory_make("tee", "tee");
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data.audio_queue = gst_element_factory_make("queue", "audio_queue");
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data.audio_convert1 = gst_element_factory_make("audioconvert", "audio_convert1");
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data.audio_resample = gst_element_factory_make("audioresample", "audio_resample");
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data.audio_sink = gst_element_factory_make("autoaudiosink", "audio_sink");
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data.video_queue = gst_element_factory_make("queue", "video_queue");
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data.audio_convert2 = gst_element_factory_make("audioconvert", "audio_convert2");
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data.visual = gst_element_factory_make("wavescope", "visual");
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data.video_convert = gst_element_factory_make("videoconvert", "video_convert");
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data.video_sink = gst_element_factory_make("autovideosink", "video_sink");
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data.app_queue = gst_element_factory_make("queue", "app_queue");
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data.app_sink = gst_element_factory_make("appsink", "app_sink");
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/* Create the empty pipeline */
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data.pipeline = gst_pipeline_new("test-pipeline");
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if (!data.pipeline || !data.app_source || !data.tee || !data.audio_queue || !data.audio_convert1 ||
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!data.audio_resample || !data.audio_sink || !data.video_queue || !data.audio_convert2 || !data.visual ||
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!data.video_convert || !data.video_sink || !data.app_queue || !data.app_sink) {
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g_printerr("Not all elements could be created.\n");
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return -1;
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}
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/* Configure wavescope */
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g_object_set(data.visual, "shader", 0, "style", 0, NULL);
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/* Configure appsrc */
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gst_audio_info_set_format(&info, GST_AUDIO_FORMAT_S16, SAMPLE_RATE, 1, NULL);
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audio_caps = gst_audio_info_to_caps(&info);
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g_object_set(data.app_source, "caps", audio_caps, "format", GST_FORMAT_TIME, NULL);
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g_signal_connect (data.app_source, "need-data", G_CALLBACK(start_feed), &data);
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g_signal_connect (data.app_source, "enough-data", G_CALLBACK(stop_feed), &data);
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/* Configure appsink */
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g_object_set(data.app_sink, "emit-signals", TRUE, "caps", audio_caps, NULL);
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g_signal_connect (data.app_sink, "new-sample", G_CALLBACK(new_sample), &data);
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gst_caps_unref(audio_caps);
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/* Link all elements that can be automatically linked because they have "Always" pads */
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gst_bin_add_many(GST_BIN (data.pipeline), data.app_source, data.tee, data.audio_queue, data.audio_convert1,
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data.audio_resample,
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data.audio_sink, data.video_queue, data.audio_convert2, data.visual, data.video_convert,
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data.video_sink, data.app_queue,
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data.app_sink, NULL);
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if (gst_element_link_many(data.app_source, data.tee, NULL) != TRUE ||
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gst_element_link_many(data.audio_queue, data.audio_convert1, data.audio_resample, data.audio_sink, NULL) !=
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TRUE ||
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gst_element_link_many(data.video_queue, data.audio_convert2, data.visual, data.video_convert, data.video_sink,
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NULL) != TRUE ||
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gst_element_link_many(data.app_queue, data.app_sink, NULL) != TRUE) {
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g_printerr("Elements could not be linked.\n");
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gst_object_unref(data.pipeline);
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return -1;
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}
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/* Manually link the Tee, which has "Request" pads */
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tee_audio_pad = gst_element_get_request_pad(data.tee, "src_%u");
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g_print("Obtained request pad %s for audio branch.\n", gst_pad_get_name (tee_audio_pad));
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queue_audio_pad = gst_element_get_static_pad(data.audio_queue, "sink");
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tee_video_pad = gst_element_get_request_pad(data.tee, "src_%u");
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g_print("Obtained request pad %s for video branch.\n", gst_pad_get_name (tee_video_pad));
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queue_video_pad = gst_element_get_static_pad(data.video_queue, "sink");
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tee_app_pad = gst_element_get_request_pad(data.tee, "src_%u");
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g_print("Obtained request pad %s for app branch.\n", gst_pad_get_name (tee_app_pad));
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queue_app_pad = gst_element_get_static_pad(data.app_queue, "sink");
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if (gst_pad_link(tee_audio_pad, queue_audio_pad) != GST_PAD_LINK_OK ||
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gst_pad_link(tee_video_pad, queue_video_pad) != GST_PAD_LINK_OK ||
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gst_pad_link(tee_app_pad, queue_app_pad) != GST_PAD_LINK_OK) {
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g_printerr("Tee could not be linked\n");
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gst_object_unref(data.pipeline);
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return -1;
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}
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gst_object_unref(queue_audio_pad);
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gst_object_unref(queue_video_pad);
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gst_object_unref(queue_app_pad);
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/* Instruct the bus to emit signals for each received message, and connect to the interesting signals */
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bus = gst_element_get_bus(data.pipeline);
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gst_bus_add_signal_watch(bus);
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g_signal_connect (G_OBJECT(bus), "message::error", (GCallback) error_cb, &data);
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gst_object_unref(bus);
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/* Start playing the pipeline */
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gst_element_set_state(data.pipeline, GST_STATE_PLAYING);
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/* Create a GLib Main Loop and set it to run */
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data.main_loop = g_main_loop_new(NULL, FALSE);
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g_main_loop_run(data.main_loop);
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/* Release the request pads from the Tee, and unref them */
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gst_element_release_request_pad(data.tee, tee_audio_pad);
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gst_element_release_request_pad(data.tee, tee_video_pad);
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gst_element_release_request_pad(data.tee, tee_app_pad);
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gst_object_unref(tee_audio_pad);
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gst_object_unref(tee_video_pad);
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gst_object_unref(tee_app_pad);
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/* Free resources */
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gst_element_set_state(data.pipeline, GST_STATE_NULL);
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gst_object_unref(data.pipeline);
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return 0;
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}
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