Monthly Archives: November 2019

音视频

基于Gstreamer的rtp转rtmp代码

rtp2rtmp_video.c
========

1. 推流到rtmp服务器

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$ ffmpeg -re -i ./fulankelin-hd.mp4 -an -vcodec h264 -f rtp rtp://127.0.0.1:5004 -vn -acodec libopus -f rtp rtp://127.0.0.1:5003
$

2. SDP信息

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SDP:
 
SDP:
v=0
o=- 0 0 IN IP4 127.0.0.1
s=No Name
t=0 0
a=tool:libavformat 58.29.100
m=video 5004 RTP/AVP 96
c=IN IP4 127.0.0.1
a=rtpmap:96 H264/90000
a=fmtp:96 packetization-mode=1
m=audio 5003 RTP/AVP 97
c=IN IP4 127.0.0.1
b=AS:96
a=rtpmap:97 opus/48000/2
a=fmtp:97 sprop-stereo=1
 
.

3. gst-launch测试

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gst-launch-1.0 -em \
  rtpbin name=rtpbin latency=5 \
  udpsrc port=5003 caps="application/x-rtp,media=(string)audio,clock-rate=(int)48000,encoding-name=(string)OPUS" ! rtpbin.recv_rtp_sink_0 \
    rtpbin.  ! rtpopusdepay ! opusdec ! audioconvert ! audioresample ! avenc_aac ! mux. \
  udpsrc port=5004 caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264" ! rtpbin.recv_rtp_sink_1 \
    rtpbin.  ! rtph264depay ! h264parse ! mux. \
  flvmux name=mux streamable=true ! rtmpsink sync=false location=rtmp://u1802/live/demo
 
.

4. 程序代码

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#include <string.h>
#include <math.h>
 
#include <gst/gst.h>
 
#define VIDEO_CAPS "application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H264"
#define AUDIO_CAPS "application/x-rtp,media=(string)audio,clock-rate=(int)48000,encoding-name=(string)OPUS"
 
/* will be called when rtpbin has validated a payload that we can depayload */
static void
pad_added_cb(GstElement *rtpbin, GstPad *new_pad, GstElement *depay)
{
    char *pad_name = GST_PAD_NAME(new_pad);
    char *depay_name = gst_element_get_name(depay);
    if (strstr(pad_name, "recv_rtp_src_0_") && strstr(depay_name, "audiodepay"))
    {
        GstPad *sinkpad;
        GstPadLinkReturn lres;
 
        g_print("new payload on rtpbin: %s %s %s\n",
                gst_element_get_name(rtpbin), GST_PAD_NAME(new_pad), gst_element_get_name(depay));
 
        sinkpad = gst_element_get_static_pad(depay, "sink");
        g_assert(sinkpad);
 
        lres = gst_pad_link(new_pad, sinkpad);
        g_assert(lres == GST_PAD_LINK_OK);
        gst_object_unref(sinkpad);
    }
    else if (strstr(pad_name, "recv_rtp_src_1_") && strstr(depay_name, "videodepay"))
    {
        GstPad *sinkpad;
        GstPadLinkReturn lres;
 
        g_print("new payload on rtpbin: %s %s %s\n",
                gst_element_get_name(rtpbin), GST_PAD_NAME(new_pad), gst_element_get_name(depay));
 
        sinkpad = gst_element_get_static_pad(depay, "sink");
        g_assert(sinkpad);
 
        lres = gst_pad_link(new_pad, sinkpad);
        g_assert(lres == GST_PAD_LINK_OK);
        gst_object_unref(sinkpad);
    }
}
 
int main(int argc, char *argv[])
{
    GMainLoop *loop;
    GstElement *pipeline;
 
    GstElement *rtpbin;
    GstElement *audiosrc, *audiodepay, *audiodec, *audiores, *audioconv, *audiosink;
    GstElement *videosrc, *videodepay, *videosink;
    GstElement *flvmux, *rtmpsink;
 
    gboolean res;
    GstCaps *caps;
    GstPadLinkReturn lres;
    GstPad *srcpad, *audio_sinkpad, *video_sinkpad;
 
    gst_init(&argc, &argv);
    pipeline = gst_pipeline_new(NULL);
    g_assert(pipeline);
 
    /* the rtpbin element */
    rtpbin = gst_element_factory_make("rtpbin", "rtpbin");
    g_assert(rtpbin);
    gst_bin_add(GST_BIN(pipeline), rtpbin);
    // 001 源
    audiosrc = gst_element_factory_make("udpsrc", "audiosrc");
    g_assert(audiosrc);
    g_object_set(audiosrc, "port", 5003, NULL);
    caps = gst_caps_from_string(AUDIO_CAPS);
    g_object_set(audiosrc, "caps", caps, NULL);
    gst_caps_unref(caps);
    gst_bin_add(GST_BIN(pipeline), audiosrc);
 
    videosrc = gst_element_factory_make("udpsrc", "videosrc");
    g_assert(videosrc);
    g_object_set(videosrc, "port", 5004, NULL);
    caps = gst_caps_from_string(VIDEO_CAPS);
    g_object_set(videosrc, "caps", caps, NULL);
    gst_caps_unref(caps);
    gst_bin_add(GST_BIN(pipeline), videosrc);
 
    /* now link all to the rtpbin, start by getting an RTP sinkpad for session 0 */
    srcpad = gst_element_get_static_pad(audiosrc, "src");
    audio_sinkpad = gst_element_get_request_pad(rtpbin, "recv_rtp_sink_0");
    lres = gst_pad_link(srcpad, audio_sinkpad);
    g_assert(lres == GST_PAD_LINK_OK);
    gst_object_unref(srcpad);
 
    srcpad = gst_element_get_static_pad(videosrc, "src");
    video_sinkpad = gst_element_get_request_pad(rtpbin, "recv_rtp_sink_1");
    lres = gst_pad_link(srcpad, video_sinkpad);
    g_assert(lres == GST_PAD_LINK_OK);
    gst_object_unref(srcpad);
 
    /* the depayloading and decoding */
    audiodepay = gst_element_factory_make("rtpopusdepay", "audiodepay");
    g_assert(audiodepay);
    audiodec = gst_element_factory_make("opusdec", "audiodec");
    g_assert(audiodepay);
    /* the audio playback and format conversion */
    audioconv = gst_element_factory_make("audioconvert", "audioconv");
    g_assert(audioconv);
    audiores = gst_element_factory_make("audioresample", "audiores");
    g_assert(audiores);
    audiosink = gst_element_factory_make("avenc_aac", "audiosink"); // autoaudiosink voaacenc avenc_aac avenc_opus
    g_assert(audiosink);
    /* add depayloading and playback to the pipeline and link */
    gst_bin_add_many(GST_BIN(pipeline), audiodepay, audiodec, audioconv,
                     audiores, audiosink, NULL);
    res = gst_element_link_many(audiodepay, audiodec, audioconv, audiores,
                                audiosink, NULL);
    g_assert(res == TRUE);
 
    videodepay = gst_element_factory_make("rtph264depay", "videodepay");
    g_assert(videodepay);
    videosink = gst_element_factory_make("h264parse", "videosink");
    g_assert(videosink);
    gst_bin_add_many(GST_BIN(pipeline), videodepay, videosink, NULL);
    res = gst_element_link_many(videodepay, videosink, NULL);
    g_assert(res == TRUE);
 
    // flvmux
    flvmux = gst_element_factory_make("flvmux", "flvmux");
    g_assert(flvmux);
    g_object_set(flvmux, "streamable", TRUE, NULL);
    gst_bin_add(GST_BIN(pipeline), flvmux);
 
    res = gst_element_link(audiosink, flvmux);
    g_assert(res == TRUE);
    res = gst_element_link(videosink, flvmux);
    g_assert(res == TRUE);
 
    rtmpsink = gst_element_factory_make("rtmpsink", "rtmpsink");
    g_assert(rtmpsink);
    g_object_set(rtmpsink, "sync", FALSE, NULL);
    g_object_set(rtmpsink, "location", "rtmp://u1802/live/demo2", NULL);
    gst_bin_add(GST_BIN(pipeline), rtmpsink);
    res = gst_element_link(flvmux, rtmpsink);
    g_assert(res == TRUE);
 
    /* the RTP pad that we have to connect to the depayloader will be created
   * dynamically so we connect to the pad-added signal, pass the depayloader as
   * user_data so that we can link to it. */
    g_signal_connect(rtpbin, "pad-added", G_CALLBACK(pad_added_cb), audiodepay);
    g_signal_connect(rtpbin, "pad-added", G_CALLBACK(pad_added_cb), videodepay);
 
    /* set the pipeline to playing */
    g_print("starting receiver pipeline\n");
    gst_element_set_state(pipeline, GST_STATE_PLAYING);
 
    /* we need to run a GLib main loop to get the messages */
    loop = g_main_loop_new(NULL, FALSE);
    g_main_loop_run(loop);
 
    g_print("stopping receiver pipeline\n");
    gst_element_set_state(pipeline, GST_STATE_NULL);
 
    gst_object_unref(loop);
    gst_object_unref(pipeline);
    gst_object_unref(audio_sinkpad);
    gst_object_unref(video_sinkpad);
    gst_object_unref(rtmpsink);
    gst_object_unref(flvmux);
    gst_object_unref(rtpbin);
    gst_object_unref(audiosrc);
    gst_object_unref(audiodepay);
    gst_object_unref(audiodec);
    gst_object_unref(audiores);
    gst_object_unref(audioconv);
    gst_object_unref(audiosink);
    gst_object_unref(videosrc);
    gst_object_unref(videodepay);
    gst_object_unref(videosink);
    return 0;
}
 
//