gstreamer学习笔记：通过gst-launch工具抓取播放的音频数据并通过upd传输

假设现在有两台虚拟机 A 和 B， A正在播放音乐，B想抓取A所播放的音乐。操作如下：

(1) A 播放音乐：

    gst-launch-1.0 filesrc location=xxxx.wav ! wavparse ! autoaudiosink

(2) A 获取pulseaudio正在播放的音频数据，通过udp发送

     gst-launch-1.0 -v rtpbin name=rtpbin latency=100 pulsesrc device=alsa_output.pci-0000_00_05.0.analog-stereo.monitor ! queue2 ! audioconvert ! avenc_aac ! rtpmp4apay pt=96 ! queue2 ! rtpbin.send_rtp_sink_1 rtpbin.send_rtp_src_1 ! udpsink host=127.0.0.1 port=5002 async=false

    注意：1、device后面指定的为pulseaudio monitor 的名字，可以通过pactl list 命令查看。如下图：

              2、host后面指定的Ip地址为B的ip地址。

附上C代码：

#include <stdio.h>
#include <string.h>
#include <math.h>
#include <getopt.h>

#include <gst/gst.h>
#define HOST_IP "172.26.178.103"
#define PORT 8888
#define DEVICE "alsa_output.pci-0000_00_05.0.analog-stereo.monitor"

/* 终端执行的命令:
gst-launch-1.0 -v rtpbin name=rtpbin latency=100 pulsesrc \
device=alsa_output.pci-0000_00_05.0.analog-stereo.monitor ! queue ! \
audioconvert ! avenc_aac ! rtpmp4apay pt=96 ! queue ! \
rtpbin.send_rtp_sink_0 rtpbin.send_rtp_src_0 ! udpsink host=127.0.0.1 port=8888 async=false

*/

/* 编译
gcc -Wall music_server.c -o music_server $(pkg-config --cflags --libs gstreamer-1.0)
*/

int main(int argc,char *argv[])
{
    if(argc != 1 && argc != 3
        && argc != 5 && argc != 7){
        printf("invalide args, usage:  ./music_server [-d device_name] [-i hostip] [-p port_number]\n");
        return -1;
    }
    
    char *short_options = "d:i:p:"; 
    static struct option long_options[] = {  
       //{"reqarg", required_argument, NULL, 'r'},  
       //{"noarg",  no_argument,       NULL, 'n'},  
       //{"optarg", optional_argument, NULL, 'o'}, 
       {"device", required_argument, NULL, 'd'+'l'},
       {"host", required_argument, NULL, 'i'+'l'},
       {"port", required_argument, NULL, 'p'+'l'},
       {0, 0, 0, 0}  
   };  

    int opt = 0;
    char device[256] = {'\0'};
    char host[30] = {'\0'};
    int port = 0; 
    while ( (opt = getopt_long(argc, argv, short_options, long_options, NULL)) != -1){
        //printf("opt = %d\n", opt);
        switch(opt){
            case 'd':
            case 'd'+'l':
                if(optarg){
                    char* deviceName = optarg;
                    int len = strlen(deviceName);
                    printf("device name = %s\n", deviceName);
                    strncpy(device, deviceName, len);
                    device[len] = '\0';
                }
                else
                    printf("device name optarg is null\n");
                break;
            case 'i':
            case 'i'+'l':
                if(optarg){
                    char* hostIp = optarg;
                    int len = strlen(hostIp);
                    strncpy(host, hostIp, len);
                    host[len] = '\0';
                    printf("hostIp = %s\n", hostIp);
                }
                else
                    printf("host ip optarg is null\n");
                break;
            case 'p':
            case 'p'+'l':
                if(optarg){
                    char* portNum = optarg;
                    port = atoi(portNum);
                    printf("portNum = %s\n", portNum);
                }
                else
                    printf("port number optarg is null\n");
                break;
            default:
                break;
        }
    }
    
    GstElement *audioSource,*queue0,*audioConvert,*aacEnc;
    GstElement *rtpMp4apay,*queue1,*rtpBin, *udpSink;	
    GstElement *pipeline;
    GMainLoop *loop;
    GstPad *srcpad,*sinkpad;

    gst_init(&argc, &argv);

    pipeline = gst_pipeline_new(NULL);
    g_assert(pipeline);

    audioSource = gst_element_factory_make("pulsesrc","pulseaudio_source");
    g_assert(audioSource);
    if(device[0] == '\0'){
        g_object_set(G_OBJECT(audioSource),"device", DEVICE, NULL);
        printf("pulseaudio device name: %s\n", DEVICE);
    }
    else{
        g_object_set(G_OBJECT(audioSource),"device", device, NULL);
        printf("pulseaudio device name: %s\n", device);
    }
       
    queue0 = gst_element_factory_make("queue","queue_0");
    g_assert(queue0);

    audioConvert = gst_element_factory_make("audioconvert","audio_convert");
    g_assert(audioConvert);

    aacEnc = gst_element_factory_make("avenc_aac","aac_enc");
    g_assert(aacEnc);
    //g_object_set(G_OBJECT(aacEnc),"compliance", -2, NULL);

    rtpMp4apay = gst_element_factory_make("rtpmp4apay","rtp_Mp4_apay");
    g_assert(rtpMp4apay);
    g_object_set(G_OBJECT(rtpMp4apay), "pt", 96, NULL);

    queue1 = gst_element_factory_make("queue","queue_1");
    g_assert(queue1);

    rtpBin = gst_element_factory_make("rtpbin","rtp_bin");
    g_assert(rtpBin);
    g_object_set(G_OBJECT(rtpBin), "latency", 100, NULL);

    udpSink = gst_element_factory_make("udpsink","udp_sink");
    g_assert(udpSink);
    gboolean async = FALSE;
    if(host[0] == '\0' && port == 0){
        g_object_set(G_OBJECT(udpSink),"port",PORT,"host",HOST_IP,"async",async,NULL);
        printf("(1)host ip = %s, port = %d\n", HOST_IP, PORT);
    }
    else if(host[0] == '\0'){
        g_object_set(G_OBJECT(udpSink),"port",port,"host",HOST_IP,"async",async,NULL);
        printf("(2)host ip = %s, port = %d\n", HOST_IP, port);
    }
    else if(port == 0){
        g_object_set(G_OBJECT(udpSink),"port",PORT,"host",host,"async",async,NULL);
        printf("(3)host ip = %s, port = %d\n", host, PORT);
    }
    else{
        g_object_set(G_OBJECT(udpSink),"port",port,"host",host,"async",async,NULL);
        printf("(4)host ip = %s, port = %d\n", host, port);
    }

    gst_bin_add_many(GST_BIN(pipeline),audioSource,queue0,audioConvert,aacEnc,rtpMp4apay,queue1,rtpBin,udpSink,NULL);

    if(!gst_element_link_many(audioSource,queue0,audioConvert,aacEnc,rtpMp4apay,queue1,NULL)){
        g_error("Failed to link \n");
        return -1;
    }

    sinkpad=gst_element_get_request_pad(rtpBin,"send_rtp_sink_0");
    srcpad=gst_element_get_static_pad(queue1,"src");
    if(gst_pad_link(srcpad,sinkpad)!=GST_PAD_LINK_OK)
        g_error("Failed to link video payloader to vrtpbin");
    gst_object_unref(srcpad);
    srcpad=gst_element_get_static_pad(rtpBin,"send_rtp_src_0");
    sinkpad=gst_element_get_static_pad(udpSink,"sink");
    if(gst_pad_link(srcpad,sinkpad)!=GST_PAD_LINK_OK)
        g_error("Failed to link vrtpbin to vrtpsink");
    gst_object_unref(srcpad);
    gst_object_unref(sinkpad);


    g_print("starting sender pipeline\n");
    //	gst_element_set_state(pipeline,SGT_STATE_PLAYING);
    gst_element_set_state (pipeline, GST_STATE_PLAYING);
    loop=g_main_loop_new(NULL,FALSE);
    g_main_loop_run(loop);
    g_print("stopping sender pipeline\n");
    gst_element_set_state(pipeline,GST_STATE_NULL);
    return 0;	
}

（3） B 接收数据,并储存为xxxx.pcm文件：

gst-launch-1.0 -v rtpbin name=rtpbin latency=100 udpsrc caps="application/x-rtp,media=(string)audio,clock-rate=(int)44100,encoding-name=(string)MP4A-LATM,cpresent=(string)0,config=(string)40002420adca00,ssrc=(uint)3235970720,payload=(int)96,clock-base=(uint)2595934541,seqnum-base=(uint)37499" port=5002 ! rtpbin.recv_rtp_sink_0 rtpbin. ! rtpmp4adepay ! queue ! avdec_aac ! filesink location=xxxx.pcm

（4）B播放xxx.pcm文件：

aplay -r 44100 -f s16_le -c 2 xxx.pcm

补充：直接从pulseaudio抓数据并存为文件

(1) 抓取raw数据，存为pcm文件

gst-launch-1.0 pulsesrc device=alsa_output.pci-0000_00_05.0.analog-stereo.monitor ! filesink location=xxx.pcm

（2）转换为aac文件

gst-launch-1.0 pulsesrc device=alsa_output.pci-0000_00_05.0.analog-stereo.monitor ! "audio/x-raw,layout=(string)interleaved,rate=(int)44100,channels=(int)2" ! avenc_aac ! avmux_adts ! filesink location=1.aac

执行可能会出现：ERROR: from element /GstPipeline:pipeline0/avenc_aac:avenc_aac0: Codec is experimental, but settings don't allow encoders to produce output of experimental quality的错误

解决办法是：在avenc_aac 后面添加compliance=-2，问题就解决了。

（3）播放抓取的aac文件

gst-launch-1.0 filesrc location=./1.aac ! aacparse ! avdec_aac ! autoaudiosink

（4）音视频合成TS流
gst-launch -e mpegtsmux name="mux" ! filesink location=mux.ts videotestsrc ! video/x-raw-yuv, framerate=25/1, width=640, height=360 ! \
x264enc ! mux. filesrc location=./1.aac ! aacparse ! mux.

gst-launch -e mpegtsmux name="mux" ! filesink location=mux.ts videotestsrc ! video/x-raw-yuv, framerate=25/1, width=640, height=360 ! \
x264enc ! mux. pulsesrc device=alsa_output.pci-0000_00_05.0.analog-stereo.monitor ! ffenc_aac ! aacparse ! mux.

gst-launch-1.0 -e avmux_mpegts name="mux" ! filesink location=mux.ts videotestsrc pattern=ball ! "video/x-raw,framerate=60/1,width=720,height=480" ! \
avenc_mpeg2video ! mux. pulsesrc device=alsa_output.pci-0000_00_05.0.analog-stereo.monitor ! avenc_aac compliance=-2 ! aacparse ! mux.