gstreamer读取USB摄像头H264帧并用rtmp推流

看见就非常 2020-10-30

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文章目录

gstreamer命令行实现rtmp推流
gstreamer代码实现rtmp推流

因为要在嵌入式端使用rtmp推流，目前我知道的有三种办法，ffmpeg、gstreamer、librtmp，每一种都需要移植到嵌入式平台，还是从我最熟悉的gstreamer开始验证吧。
现在我的嵌入式平台gstreamer库没有rtmp元件，因此只能先在Ubuntu16.04系统的PC上测试，然后再移植带有rtmp元件的gstreamer库。
Ubuntu16.04系统已经自带了gstreamer-1.0的库，并且已经包含rtmp元件，不用移植可以直接测试了。
注意：我使用的USB摄像头可以直接输出H264帧，因此不需要使用编码元件。

gstreamer命令行实现rtmp推流

首先用命令行工具测试：

gst-launch-1.0 -v v4l2src device=/dev/video0 ! 'video/x-h264, width=1280, height=720, framerate=30/1' ! queue !  h264parse ! flvmux ! rtmpsink location='rtmp://192.168.1.102/live'

这个命令行执行后，就可以在192.168.1.102地址的PC上打开流媒体服务端观看。可以使用nginx或者srs流媒体服务端，创建一个html文件打开网页观看。

<h1>01</h1><object width='640' height='377' id='SampleMediaPlayback' name='SampleMediaPlayback' type='application/x-shockwave-flash' classid='clsid:d27cdb6e-ae6d-11cf-96b8-444553540000' ><param name='movie' value='swfs/SampleMediaPlayback.swf' /> <param name='quality' value='high' /> <param name='bgcolor' value='#000000' /> <param name='allowfullscreen' value='true' /> <embed src='SampleMediaPlayback.swf' width='640' height='377' id='SampleMediaPlayback' quality='high' bgcolor='#000000' name='SampleMediaPlayback' allowfullscreen='true' pluginspage='http://www.adobe.com/go/getflashplayer' flashvars='&src=rtmp://192.168.1.102:1935/live&autoHideControlBar=true&streamType=live&autoPlay=true&verbose=true' type='application/x-shockwave-flash'> </embed></object>1
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gstreamer代码实现rtmp推流

#include <string.h>#include <gst/gst.h>#include <signal.h>#include <unistd.h>#include <stdlib.h>#include <stdio.h>#include <string.h>//gst-launch-1.0 -v v4l2src device=/dev/video0 ! 'video/x-h264, width=640, height=360, framerate=30/1' ! queue !  h264parse ! flvmux ! rtmpsink location='rtmp://192.168.1.102/live'typedef struct _GstDataStruct{GstElement *pipeline;GstElement *v4l2src;GstElement *queue;GstElement *h264parse;GstElement *flvmux;GstElement *rtmpsink;GstBus *bus;guint bus_watch_id;guint sourceid;        /* To control the GSource */GMainLoop *loop;  /* GLib's Main Loop */} GstDataStruct;static GstDataStruct GstData;static unsigned int frame_width;static unsigned int frame_height;static unsigned int frame_rate;static unsigned int frame_bps;static char devname[32] = {0};gboolean bus_msg_call(GstBus *bus, GstMessage *msg, GstDataStruct *pGstData){gchar *debug;GError *error;GMainLoop *loop = pGstData->loop;GST_DEBUG ('got message %s',gst_message_type_get_name (GST_MESSAGE_TYPE (msg)));switch (GST_MESSAGE_TYPE(msg)){case GST_MESSAGE_EOS:printf('End of stream\n');g_main_loop_quit(loop);break;case GST_MESSAGE_ERROR:gst_message_parse_error(msg, &error, &debug);g_free(debug);g_printerr('Error: %s\n', error->message);g_error_free(error);g_main_loop_quit(loop);break;default:break;}return TRUE;}int main(int argc, char *argv[]){if(argc != 6){frame_width = 1280;frame_height = 720;frame_rate = 30;frame_bps = 1500000;sprintf(devname, '%s', '/dev/video0');}else{frame_width = atoi(argv[2]);frame_height = atoi(argv[3]);frame_rate = atoi(argv[4]);frame_bps = atoi(argv[5]);sprintf(devname, '%s', argv[1]);}printf('width:%d, height:%d, rate:%d, bps:%d, dev:%s\n', frame_width, frame_height, frame_rate, frame_bps, devname);printf('============= v4l2 rtmp gst init start ============\n');gst_init (NULL, NULL);printf('=========== create v4l2 rtmp pipeline =============\n');GstData.pipeline           = gst_pipeline_new ('v4l2_rtmp');GstData.pipeline           = gst_pipeline_new ('v4l2_rtmp');GstData.v4l2src           = gst_element_factory_make ('v4l2src',      'v4l2src');GstData.queue         = gst_element_factory_make ('queue',  'queue');GstData.h264parse         = gst_element_factory_make ('h264parse','h264parse');GstData.flvmux           = gst_element_factory_make ('flvmux',      'flvmux');GstData.rtmpsink            = gst_element_factory_make ('rtmpsink',     'rtmpsink');if (!GstData.pipeline || !GstData.v4l2src || !GstData.queue ||!GstData.h264parse || !GstData.flvmux || !GstData.rtmpsink){g_printerr ('One element could not be created... Exit\n');return -1;}printf('============ link v4l2 rtmp pipeline ==============\n');GstCaps *caps_v4l2src;caps_v4l2src = gst_caps_new_simple('video/x-h264', 'stream-format', G_TYPE_STRING,'byte-stream',   'alignment', G_TYPE_STRING, 'au',   'width', G_TYPE_INT, frame_width,   'height', G_TYPE_INT, frame_height,   'framerate',GST_TYPE_FRACTION, frame_rate, 1, NULL);GstCaps *caps_flv_sink;caps_flv_sink = gst_caps_new_simple('video/x-h264', 'stream-format', G_TYPE_STRING,'avc',    'alignment', G_TYPE_STRING, 'au',    'width', G_TYPE_INT, frame_width,    'height', G_TYPE_INT, frame_height,    'framerate',GST_TYPE_FRACTION, frame_rate, 1, NULL);g_object_set(G_OBJECT(GstData.v4l2src), 'device', devname, NULL);g_object_set(G_OBJECT(GstData.rtmpsink), 'location', 'rtmp://192.168.1.102/live', NULL);//注意：此处的location参数代表rtmp的url，其取值必须与html文件的rtmp的URL保持一致，才可观看视频。GstData.bus = gst_pipeline_get_bus(GST_PIPELINE(GstData.pipeline));GstData.bus_watch_id = gst_bus_add_watch(GstData.bus, (GstBusFunc)bus_msg_call, (gpointer)&GstData);gst_object_unref(GstData.bus);gst_bin_add_many(GST_BIN(GstData.pipeline), GstData.v4l2src, GstData.queue, GstData.h264parse, GstData.flvmux, GstData.rtmpsink,NULL);if(gst_element_link_filtered(GstData.v4l2src, GstData.queue, caps_v4l2src) != TRUE){g_printerr ('GstData.v4l2src could not link GstData.queue\n');gst_object_unref (GstData.pipeline);return -1;}gst_caps_unref (caps_v4l2src);if(gst_element_link(GstData.queue, GstData.h264parse) != TRUE){g_printerr ('GstData.queue could not link GstData.h264parse\n');gst_object_unref (GstData.pipeline);return -1;}if(gst_element_link_filtered(GstData.h264parse, GstData.flvmux, caps_flv_sink) != TRUE){g_printerr ('GstData.h264parse could not link GstData.flvmux\n');gst_object_unref (GstData.pipeline);return -1;}gst_caps_unref (caps_flv_sink);if(gst_element_link(GstData.flvmux, GstData.rtmpsink) != TRUE){g_printerr ('GstData.h264parse could not link GstData.flvmux\n');gst_object_unref (GstData.pipeline);return -1;}printf('========= link v4l2 rtmp pipeline running ==========\n');gst_element_set_state (GstData.pipeline, GST_STATE_PLAYING);GstData.loop = g_main_loop_new(NULL, FALSE);// Create gstreamer loopg_main_loop_run(GstData.loop);// Loop will run until receiving EOS (end-of-stream), will block hereprintf('g_main_loop_run returned, stopping rtmp!\n');gst_element_set_state (GstData.pipeline, GST_STATE_NULL);// Stop pipeline to be releasedprintf('Deleting pipeline\n');gst_object_unref (GstData.pipeline);// THis will also delete all pipeline elementsg_source_remove(GstData.bus_watch_id);g_main_loop_unref(GstData.loop);return 0;}

此代码在Ubuntu16.04系统下使用gcc编译，makefile如下：
需要将系统目录下的关于gstreamer的库文件拷贝到当前目录的libs_x86目录下，另外系统的gstreamer库链接文件都带有.so.0后缀，去掉最后的.0，保留到.so即可。

CFLAGS = -v -g -Wall -Wno-shift-count-overflow -I./includeLDFLAGS = -L./libs_x86CC = gccEXTRA_LIBS = -lstdc++ -lm -lpthread -lgstreamer-1.0 -lgstbase-1.0 -lgobject-2.0 -lgmodule-2.0 -lglib-2.0 -lpcre -lrtSRC = v4l2_rtmp.cTARGET = v4l2_rtmpALL:$(CC) $(CFLAGS) $(LDFLAGS) $(SRC) -o $(TARGET) $(EXTRA_LIBS) clean:rm v4l2_rtmp *.raw *.mp4 *.wav -rf1
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