Я в полном смятении. Я пытаюсь использовать Gstreamer в C для динамического «разделения» вывода веб-камеры H264.
#include <stdio.h>
#include <string.h>
#include <gst/gst.h>
typedef struct _CameraData
{
GstElement *pipeline;
GstElement *video_source;
GstElement *audio_source;
GstElement *video_capsfilter;
GstElement *audio_capsfilter;
GstElement *h264parse;
GstElement *audio_convert;
GstElement *audio_resample;
GstElement *mux;
GstElement *sink;
gboolean is_recording;
} CameraData;
CameraData camera_data;
void StartRecording(const char *filename)
{
if (camera_data.is_recording)
{
g_print("Recording is already in progress.\n");
return;
}
// Set the output file location to the user-provided filename
g_object_set(G_OBJECT(camera_data.sink), "location", filename, NULL);
GstStateChangeReturn result;
// Set pipeline state to PLAYING
result = gst_element_set_state(camera_data.pipeline, GST_STATE_PLAYING);
if (result == GST_STATE_CHANGE_FAILURE)
{
g_printerr("Failed to set pipeline to PLAYING state.\n");
// Get additional error details
GstBus *bus = gst_element_get_bus(camera_data.pipeline);
GstMessage *msg = gst_bus_poll(bus, GST_MESSAGE_ERROR, -1); // Wait indefinitely for an error message
if (msg != NULL)
{
GError *err;
gchar *debug_info;
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);
gst_message_unref(msg);
}
gst_object_unref(bus);
return;
}
// Wait for the state change to complete
GstBus *bus = gst_element_get_bus(camera_data.pipeline);
GstMessage *msg;
g_print("Starting recording to file: %s\n", filename);
msg = gst_bus_timed_pop_filtered(bus, GST_CLOCK_TIME_NONE,
GST_MESSAGE_ERROR | GST_MESSAGE_ASYNC_DONE | GST_MESSAGE_EOS);
if (msg != NULL)
{
GError *err;
gchar *debug_info;
switch (GST_MESSAGE_TYPE(msg))
{
case GST_MESSAGE_ERROR:
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);
break;
case GST_MESSAGE_ASYNC_DONE:
g_print("Pipeline is now PLAYING.\n");
camera_data.is_recording = TRUE;
break;
case GST_MESSAGE_EOS:
g_print("End-Of-Stream reached.\n");
break;
default:
g_printerr("Unexpected message received.\n");
break;
}
gst_message_unref(msg);
}
gst_object_unref(bus);
}
void StopRecording()
{
if (!camera_data.is_recording)
{
g_print("Recording is not in progress.\n");
return;
}
// Set pipeline state to NULL to stop recording
gst_element_set_state(camera_data.pipeline, GST_STATE_NULL);
g_print("Recording stopped.\n");
camera_data.is_recording = FALSE;
}
void InitializeGStreamerPipeline()
{
// Initialize GStreamer
gst_init(NULL, NULL);
// Create the GStreamer elements
camera_data.video_source = gst_element_factory_make("v4l2src", "video_source");
camera_data.audio_source = gst_element_factory_make("alsasrc", "audio_source");
camera_data.video_capsfilter = gst_element_factory_make("capsfilter", "video_filter");
camera_data.audio_capsfilter = gst_element_factory_make("capsfilter", "audio_filter");
camera_data.h264parse = gst_element_factory_make("h264parse", "h264parse");
camera_data.audio_convert = gst_element_factory_make("audioconvert", "audio_convert");
camera_data.audio_resample = gst_element_factory_make("audioresample", "audio_resample");
camera_data.mux = gst_element_factory_make("matroskamux", "mux");
camera_data.sink = gst_element_factory_make("filesink", "sink");
// Check if elements are created successfully
if (!camera_data.video_source || !camera_data.audio_source || !camera_data.video_capsfilter ||
!camera_data.audio_capsfilter || !camera_data.h264parse || !camera_data.audio_convert ||
!camera_data.audio_resample || !camera_data.mux || !camera_data.sink)
{
g_printerr("Failed to create GStreamer elements.\n");
return;
}
// Create a GStreamer pipeline
camera_data.pipeline = gst_pipeline_new("webcam-pipeline");
if (!camera_data.pipeline)
{
g_printerr("Failed to create GStreamer pipeline.\n");
return;
}
// Set properties for the elements
g_object_set(G_OBJECT(camera_data.video_source), "device", "/dev/video2", NULL); // Change to your webcam device
g_object_set(G_OBJECT(camera_data.audio_source), "device", "hw:1,0", NULL); // Audio device
// Set caps for MJPEG capture
GstCaps *video_caps = gst_caps_new_simple("video/x-h264",
"framerate", GST_TYPE_FRACTION, 30, 1, // Set to 30 fps
"width", G_TYPE_INT, 1920, // Adjust width as needed
"height", G_TYPE_INT, 1080, // Adjust height as needed
NULL);
g_object_set(G_OBJECT(camera_data.video_capsfilter), "caps", video_caps, NULL);
gst_caps_unref(video_caps);
// Set caps for the audio capture to match S16_LE format and 48 kHz
GstCaps *audio_caps = gst_caps_new_simple("audio/x-raw",
"format", G_TYPE_STRING, "S16LE",
"rate", G_TYPE_INT, 48000,
"channels", G_TYPE_INT, 2, // Stereo
NULL);
g_object_set(G_OBJECT(camera_data.audio_capsfilter), "caps", audio_caps, NULL);
gst_caps_unref(audio_caps);
// Add elements to the pipeline
gst_bin_add_many(GST_BIN(camera_data.pipeline), camera_data.video_source, camera_data.audio_source,
camera_data.video_capsfilter, camera_data.audio_capsfilter, camera_data.h264parse,
camera_data.audio_convert, camera_data.audio_resample, camera_data.mux, camera_data.sink, NULL);
// Link video elements in the pipeline
if (!gst_element_link_many(camera_data.video_source, camera_data.video_capsfilter, camera_data.h264parse, camera_data.mux, NULL))
{
g_printerr("Failed to link video elements in the pipeline.\n");
gst_object_unref(camera_data.pipeline);
return;
}
// Link audio elements in the pipeline
if (!gst_element_link_many(camera_data.audio_source, camera_data.audio_capsfilter, camera_data.audio_convert, camera_data.audio_resample, camera_data.mux, NULL))
{
g_printerr("Failed to link audio elements in the pipeline.\n");
gst_object_unref(camera_data.pipeline);
return;
}
// Link muxer to sink
if (!gst_element_link(camera_data.mux, camera_data.sink))
{
g_printerr("Failed to link muxer to sink.\n");
gst_object_unref(camera_data.pipeline);
return;
}
camera_data.is_recording = FALSE;
g_print("GStreamer pipeline initialized.\n");
}
int main(void)
{
InitializeGStreamerPipeline();
char input[100];
while (1)
{
printf("Enter 'start <filename>' to begin recording, 'stop' to end recording, or 'quit' to exit: ");
fgets(input, 100, stdin);
if (strncmp(input, "start", 5) == 0)
{
char *filename = strtok(input + 6, "\n"); // Get filename from input
if (filename)
{
StartRecording(filename);
}
else
{
printf("Please provide a filename.\n");
}
}
else if (strncmp(input, "stop", 4) == 0)
{
StopRecording();
}
else if (strncmp(input, "quit", 4) == 0)
{
StopRecording(); // Ensure the pipeline is properly stopped before quitting
break;
}
else
{
printf("Invalid command. Please enter 'start <filename>', 'stop', or 'quit'.\n");
}
}
return 0;
}
Я заставил его работать для простого ввода (/dev/video2 + alsa) в файловый приемник.
Поскольку камера поддерживает H264, я использую h264parse. Поэтому кодирование не требуется.
Однако я понятия не имею, как использовать вставку тройника в уравнение, и все мои попытки встречали катастрофическое разочарование (теперь я понимаю мем о гусином фермере).
Моей конечной целью было бы динамическое добавление/удаление очередей в тройнике. Но на данный момент я бы просто решил просто настроить один и сохранить вывод в два разных файла. Нет, я не могу использовать multifilesink, потому что я хочу также использовать tcpserversinks.
Спасибо, ребята!
Вы можете использовать такую футболку
gst-launch-1.0 v4l2src ! video/x-h264 ! tee name=t \
t. ! queue ! h264parse ! qtmux ! filesink location=vid0.mp4 async=true sync=false \
t. ! queue ! h264parse ! qtmux ! filesink location=vid1.mp4 async=true sync=false -e
важно добавлять очередь после каждой ветки тройника
В чем именно проблема?