Gstreamer 录制带音频的视频

Posted 2023-02-22

【中文标题】Gstreamer 录制带音频的视频

【英文标题】：Gstreamer recording video with audio

【发布时间】：2019-03-06 18:26:35

【问题描述】：

我正在尝试通过 glib 库在我的 Ubuntu 16 机器上使用 Gstreamer 将来自我的网络摄像头的视频以及音频录制到一个文件中。 我可以通过这些代码行观看来自网络摄像头的视频流

#include <gst/gst.h>

int main(int argc, char *argv[]) 
    GstElement *pipeline, *source, *sink, *convert;
    GstBus *bus;
    GstMessage *msg;
    GstStateChangeReturn ret;

    /* Initialize GStreamer */
    gst_init (&argc, &argv);

    /* Create the elements */
    source = gst_element_factory_make ("v4l2src", "source");
    sink = gst_element_factory_make ("autovideosink", "sink");
    convert =gst_element_factory_make("videoconvert","convert");
    //convert = gst_element_factory_make ("audioconvert", "convert");
    //sink = gst_element_factory_make ("autoaudiosink", "sink");

    /* Create the empty pipeline */
    pipeline = gst_pipeline_new ("test-pipeline");

    if (!pipeline || !source || !sink || !convert) 
        g_printerr ("Not all elements could be created.\n");
        return -1;
    

    /*set der source*/
    g_object_set (source, "device", "/dev/video0", NULL);

    /* Build the pipeline */
    gst_bin_add_many (GST_BIN (pipeline), source, sink, convert, NULL);
    if (gst_element_link (convert, sink) != TRUE) 
        g_printerr ("Elements could not be linked confert sink.\n");
        gst_object_unref (pipeline);
        return -1;
    


    if (gst_element_link (source, convert) != TRUE) 
        g_printerr ("Elements could not be linked source -convert.\n");
        gst_object_unref (pipeline);
        return -1;
    

    /* Start playing */
    ret = gst_element_set_state (pipeline, GST_STATE_PLAYING);
    if (ret == GST_STATE_CHANGE_FAILURE) 
        g_printerr ("Unable to set the pipeline to the playing state.\n");
        gst_object_unref (pipeline);
        return -1;
    

    /* Wait until error or EOS */
    bus = gst_element_get_bus (pipeline);
    msg = gst_bus_timed_pop_filtered (bus, GST_CLOCK_TIME_NONE,(GstMessageType) (GST_MESSAGE_ERROR | GST_MESSAGE_EOS));

    /* Parse message */
    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_EOS:
                g_print ("End-Of-Stream reached.\n");
                break;
            default:
                /* We should not reach here because we only asked for ERRORs and EOS */
                g_printerr ("Unexpected message received.\n");
                break;
        
        gst_message_unref (msg);
    

    /* Free resources */
    gst_object_unref (bus);
    gst_element_set_state (pipeline, GST_STATE_NULL);
    gst_object_unref (pipeline);
    return 0;

并使用这些代码行从麦克风捕获音频并通过扬声器收听

#include <gst/gst.h>
#include <glib.h>

static gboolean
bus_call (GstBus     *bus,
          GstMessage *msg,
          gpointer    data)
  GMainLoop *loop = (GMainLoop *) data;

  switch (GST_MESSAGE_TYPE (msg)) 

    case GST_MESSAGE_EOS:
      g_print ("End of stream\n");
      g_main_loop_quit (loop);
      break;

    case GST_MESSAGE_ERROR: 
      gchar  *debug;
      GError *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;


/* Main function for audio pipeline initialization and looping streaming process  */
gint
main (gint argc, gchar **argv) 
    GMainLoop *loop;
    GstElement *pipeline, *audio_source, *sink; 
    GstBus *bus;
    guint bus_watch_id;
    GstCaps *caps;
    gboolean ret;

    /* Initialization of gstreamer */
    gst_init (&argc, &argv);
    loop = g_main_loop_new (NULL, FALSE);

    /* Elements creation */
    pipeline     = gst_pipeline_new ("audio_stream");
    audio_source = gst_element_factory_make ("alsasrc", "audio_source");
    sink   = gst_element_factory_make ("alsasink", "audio_sink");

    // video_source = gst_element_factory_make ("v4l2src", "source");
    // video_sink   = gst_element_factory_make ("autovideosink", "sink");
    // video_convert= gst_element_factory_make("videoconvert","convert");

    if (!pipeline) 
        g_printerr ("Audio: Pipeline couldn't be created\n");
        return -1;
    
    if (!audio_source) 
        g_printerr ("Audio: alsasrc couldn't be created\n");
        return -1;
    
    if (!sink) 
        g_printerr ("Audio: Output file couldn't be created\n");
        return -1;
    

    g_object_set (G_OBJECT (audio_source), "device", "hw:1,0", NULL);
    g_object_set (G_OBJECT (sink), "device", "hw:1,0", NULL);

    bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
    bus_watch_id = gst_bus_add_watch (bus, bus_call, loop);
    gst_object_unref (bus);

    gst_bin_add_many (GST_BIN(pipeline), audio_source, sink, NULL);

    caps = gst_caps_new_simple ("audio/x-raw", "format", G_TYPE_STRING, "S16LE",  "layout", G_TYPE_STRING, "interleaved", "rate", G_TYPE_INT, (int)44100, "channels", G_TYPE_INT, (int)2, NULL);
    ret = gst_element_link_filtered (audio_source, sink, caps);
    if (!ret) 
        g_print ("audio_source and sink couldn't be linked\n");
        gst_caps_unref (caps);
        return FALSE;
    

    gst_element_set_state (pipeline, GST_STATE_PLAYING);

    g_print ("streaming...\n");
    g_main_loop_run (loop);

    g_print ("Returned, stopping stream\n");
    gst_element_set_state (pipeline, GST_STATE_NULL);

    g_print ("Deleting pipeline\n");
    gst_object_unref (GST_OBJECT (pipeline));
    g_source_remove (bus_watch_id);
    g_main_loop_unref (loop);

    return 0;

我真正不明白的是如何同时从网络摄像头获取视频和从我的 alsa hw 获取音频并将它们保存到文件中（例如 .mp4 for ex）。谁能帮我？我试图找到一些有用的东西，但黑板上什么都没有。此外，如果将视频流或音频流保存在单独的文件中，我将非常感激。

更新 我再次查看了教程和@nayana 提供的 git 链接，所以我尝试自己编写一些代码。我有两个结果：

#include <string.h>
#include <gst/gst.h>
#include <signal.h>
#include <unistd.h>
#include <stdlib.h>

static GMainLoop *loop;
static GstElement *pipeline;
static GstElement *muxer, *sink;
static GstElement *src_video, *encoder_video, *queue_video; 
static GstElement *src_audio, *encoder_audio, *queue_audio;
static GstBus *bus;

static gboolean
message_cb (GstBus * bus, GstMessage * message, gpointer user_data)

  switch (GST_MESSAGE_TYPE (message)) 
    case GST_MESSAGE_ERROR:
      GError *err = NULL;
      gchar *name, *debug = NULL;

      name = gst_object_get_path_string (message->src);
      gst_message_parse_error (message, &err, &debug);

      g_printerr ("ERROR: from element %s: %s\n", name, err->message);
      if (debug != NULL)
        g_printerr ("Additional debug info:\n%s\n", debug);

      g_error_free (err);
      g_free (debug);
      g_free (name);

      g_main_loop_quit (loop);
      break;
    
    case GST_MESSAGE_WARNING:
    GError *err = NULL;
    gchar *name, *debug = NULL;

    name = gst_object_get_path_string (message->src);
    gst_message_parse_warning (message, &err, &debug);

    g_printerr ("ERROR: from element %s: %s\n", name, err->message);
    if (debug != NULL)
    g_printerr ("Additional debug info:\n%s\n", debug);

    g_error_free (err);
    g_free (debug);
    g_free (name);
    break;
    
    case GST_MESSAGE_EOS:
    g_print ("Got EOS\n");
    g_main_loop_quit (loop);
    gst_element_set_state (pipeline, GST_STATE_NULL);
    g_main_loop_unref (loop);
    gst_object_unref (pipeline);
    exit(0);
    break;
  
    default:
    break;
  

  return TRUE;


void sigintHandler(int unused) 
  g_print("You ctrl-c-ed! Sending EoS");
  gst_element_send_event(pipeline, gst_event_new_eos()); 


int main(int argc, char *argv[])

  signal(SIGINT, sigintHandler);
  gst_init (&argc, &argv);

  pipeline = gst_pipeline_new(NULL);

  src_video = gst_element_factory_make("v4l2src", NULL);
  encoder_video = gst_element_factory_make("x264enc", NULL);
  queue_video = gst_element_factory_make("queue", NULL);

  src_audio = gst_element_factory_make ("alsasrc", NULL);
  encoder_audio = gst_element_factory_make("lamemp3enc", NULL);
  queue_audio = gst_element_factory_make("queue", NULL);

  muxer = gst_element_factory_make("mp4mux", NULL);
  sink = gst_element_factory_make("filesink", NULL);

  if (!pipeline || !src_video || !encoder_video || !src_audio || !encoder_audio
        || !queue_video || !queue_audio || !muxer || !sink) 
    g_error("Failed to create elements");
    return -1;
  

  g_object_set(src_audio, "device", "hw:1,0", NULL);
  g_object_set(sink, "location", "video_audio_test.mp4", NULL);


  gst_bin_add_many(GST_BIN(pipeline), src_video, encoder_video, queue_video, 
    src_audio, encoder_audio, queue_audio, muxer, sink, NULL);

  gst_element_link_many (src_video,encoder_video,queue_video, muxer,NULL);

  gst_element_link_many (src_audio,encoder_audio,queue_audio, muxer,NULL);

  if (!gst_element_link_many(muxer, sink, NULL))
    g_error("Failed to link elements");
    return -2;
  

  loop = g_main_loop_new(NULL, FALSE);

  bus = gst_pipeline_get_bus(GST_PIPELINE (pipeline));
  gst_bus_add_signal_watch(bus);
  g_signal_connect(G_OBJECT(bus), "message", G_CALLBACK(message_cb), NULL);
  gst_object_unref(GST_OBJECT(bus));

  gst_element_set_state(pipeline, GST_STATE_PLAYING);

  g_print("Starting loop");
  g_main_loop_run(loop);

  return 0;

有了这个，我就可以从摄像头录制视频了，但是在录制过程中，音频在某处随机录制了一秒钟，这给了我这个错误

ERROR: from element /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: Can't record audio fast enough
Additional debug info:
gstaudiobasesrc.c(869): gst_audio_base_src_create (): /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0:
Dropped 206388 samples. This is most likely because downstream can't keep up and is consuming samples too slowly.<br>

所以我尝试添加一些设置和队列

#include <string.h>
#include <gst/gst.h>
#include <signal.h>
#include <unistd.h>
#include <stdlib.h>

static GMainLoop *loop;
static GstElement *pipeline;
static GstElement *muxer, *sink;
static GstElement *src_video, *encoder_video, *queue_video, *rate_video, *scale_video, *capsfilter_video; 
static GstElement *src_audio, *encoder_audio, *queue_audio, *queue_audio2, *capsfilter_audio, *rate_audio;
static GstBus *bus;
static GstCaps *caps;

static gboolean
message_cb (GstBus * bus, GstMessage * message, gpointer user_data)

  switch (GST_MESSAGE_TYPE (message)) 
    case GST_MESSAGE_ERROR:
      GError *err = NULL;
      gchar *name, *debug = NULL;

      name = gst_object_get_path_string (message->src);
      gst_message_parse_error (message, &err, &debug);

      g_printerr ("ERROR: from element %s: %s\n", name, err->message);
      if (debug != NULL)
        g_printerr ("Additional debug info:\n%s\n", debug);

      g_error_free (err);
      g_free (debug);
      g_free (name);

      g_main_loop_quit (loop);
      break;
    
    case GST_MESSAGE_WARNING:
    GError *err = NULL;
    gchar *name, *debug = NULL;

    name = gst_object_get_path_string (message->src);
    gst_message_parse_warning (message, &err, &debug);

    g_printerr ("ERROR: from element %s: %s\n", name, err->message);
    if (debug != NULL)
    g_printerr ("Additional debug info:\n%s\n", debug);

    g_error_free (err);
    g_free (debug);
    g_free (name);
    break;
    
    case GST_MESSAGE_EOS:
    g_print ("Got EOS\n");
    g_main_loop_quit (loop);
    gst_element_set_state (pipeline, GST_STATE_NULL);
    g_main_loop_unref (loop);
    gst_object_unref (pipeline);
    exit(0);
    break;
  
    default:
    break;
  

  return TRUE;


void sigintHandler(int unused) 
  g_print("You ctrl-c-ed! Sending EoS");
  gst_element_send_event(pipeline, gst_event_new_eos()); 


int main(int argc, char *argv[])

  signal(SIGINT, sigintHandler);
  gst_init (&argc, &argv);

  pipeline = gst_pipeline_new(NULL);

  src_video = gst_element_factory_make("v4l2src", NULL);
  rate_video = gst_element_factory_make ("videorate", NULL);
  scale_video = gst_element_factory_make ("videoscale", NULL);
  capsfilter_video = gst_element_factory_make ("capsfilter", NULL);
  queue_video = gst_element_factory_make("queue", NULL);
  encoder_video = gst_element_factory_make("x264enc", NULL);

  src_audio = gst_element_factory_make ("alsasrc", NULL);
  capsfilter_audio = gst_element_factory_make ("capsfilter", NULL);
  queue_audio = gst_element_factory_make("queue", NULL);
  rate_audio = gst_element_factory_make ("audiorate", NULL);
  queue_audio2 = gst_element_factory_make("queue", NULL);
  encoder_audio = gst_element_factory_make("lamemp3enc", NULL);

  muxer = gst_element_factory_make("mp4mux", NULL);
  sink = gst_element_factory_make("filesink", NULL);

  if (!pipeline || !src_video || !rate_video || !scale_video || !capsfilter_video 
     || !queue_video || !encoder_video || !src_audio || !capsfilter_audio 
     || !queue_audio || !rate_audio || !queue_audio2 || !encoder_audio 
     || !muxer || !sink) 
    g_error("Failed to create elements");
    return -1;
  

  // Set up the pipeline
  g_object_set(src_video, "device", "/dev/video0", NULL); 
  g_object_set(src_audio, "device", "hw:1,0", NULL);
  g_object_set(sink, "location", "video_audio_test.mp4", NULL);

  // video settings
  caps = gst_caps_from_string("video/x-raw,format=(string)I420,width=480,height=384,framerate=(fraction)25/1");
  g_object_set (G_OBJECT (capsfilter_video), "caps", caps, NULL);
  gst_caps_unref (caps); 

  // audio settings
  caps = gst_caps_from_string("audio/x-raw,rate=44100,channels=1");
  g_object_set (G_OBJECT (capsfilter_audio), "caps", caps, NULL);
  gst_caps_unref (caps);

  // add all elements into the pipeline 
  gst_bin_add_many(GST_BIN(pipeline), src_video, rate_video, scale_video, capsfilter_video, 
    queue_video, encoder_video, src_audio, capsfilter_audio, queue_audio, rate_audio, 
    queue_audio2, encoder_audio, muxer, sink, NULL);

  if (!gst_element_link_many (src_video,rate_video,scale_video, capsfilter_video,
    queue_video, encoder_video, muxer,NULL))
  
    g_error("Failed to link video elements");
    return -2;
  

  if (!gst_element_link_many (src_audio, capsfilter_audio, queue_audio, rate_audio, 
    queue_audio2, encoder_audio, muxer,NULL))
  
    g_error("Failed to link audio elements");
    return -2;
  

  if (!gst_element_link_many(muxer, sink, NULL))
  
    g_error("Failed to link elements");
    return -2;
  

  loop = g_main_loop_new(NULL, FALSE);

  bus = gst_pipeline_get_bus(GST_PIPELINE (pipeline));
  gst_bus_add_signal_watch(bus);
  g_signal_connect(G_OBJECT(bus), "message", G_CALLBACK(message_cb), NULL);
  gst_object_unref(GST_OBJECT(bus));

  gst_element_set_state(pipeline, GST_STATE_PLAYING);

  g_print("Starting loop");
  g_main_loop_run(loop);

  return 0;

这次代码没有记录任何东西，并给我以下错误

   ERROR: from element /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0: Internal data flow error.
Additional debug info:
gstbasesrc.c(2948): gst_base_src_loop (): /GstPipeline:pipeline0/GstAlsaSrc:alsasrc0:
streaming task paused, reason not-negotiated (-4)

你能告诉我修复错误吗？ 提前致谢

【问题讨论】：

好的，尝试在第一个解决方案中为 alsasrc 添加do-timestamp=true。您还可以调试使用 env varbiable GST_DEBUG=5 ./your_binary 运行二进制文件时发生的情况。另一件事是尝试 qtmux，它在 mp4mux (iirc) 上有一些快速的时间扩展

据我了解，有必要对视频流执行某种转换以减少繁琐并将其与音频流同步。我仍然面临这个问题，因为我无法为源设置带有大写过滤器的正确管道（我想降低分辨率和帧速率）

我发现了一个非常有趣的帖子，它建议使用 zerolatency，这可以解决 x264enc 跟不上 lamemp3enc 的问题。 gstreamer-devel.966125.n4.nabble.com/… 您应该在编码器之前添加队列（在第一个解决方案中）。第二个太复杂了，我不太理解。再一次，先用 gst-launch 试试。

【参考方案1】：

您需要的是多路复用器 - 可以将两个流合并为一个的 GStreamer 元素。

mp4、mkv、avi..只是一种容器格式，包含多个“数据流”，可以是音频、视频、字幕（并非所有格式都支持）。

我不了解您的用例，但您的工作不需要 C 代码。你可以使用 gst-launch-1.0 工具，它有自己的 GStreamer 类型的脚本语言。

为简单起见，我将使用调试元素 videotestsrc 和 audiotestsrc 来模拟输入（而不是实际的相机等）。

gst-launch-1.0 -e videotestsrc ! x264enc ! mp4mux name=mux ! filesink location="bla.mp4"  audiotestsrc ! lamemp3enc ! mux.

videotestsrc --> x264enc -----\
                               >---> mp4mux ---> filesink
audiotestsrc --> lamemp3enc --/ 

解释：

Videotestsrc 生成原始视频，在 GStreamer 术语中称为“video/x-raw”。

但是 mp4 不能保存原始视频，因此我们需要使用例如 x264enc 对其进行编码，这使得我们的数据为“video/x-h264”。

然后我们最终可以将它与mp4mux 元素混合到我们的mp4 中。

当我们使用 gst-inspect-1.0 mp4mux 查看 GStreamer 文档时，我们发现该元素支持各种格式，其中还有 video/x-h264。

对于 AAC 格式的 faac 或对于 mp3 的 lamemp3enc，我们对音频执行相同的操作。

通过gst-launch-1.0，我做了两个技巧和一个奖励技巧：

能够在一行中有单独的分支。这是通过用空格而不是!分隔这些分支来实现的 能够使用 name=mux 创建别名，然后在名称末尾添加点，如 mux. 。您可以为该元素起任何您喜欢的名称。 按 ctrl+c 后写入 EOS 以停止录制。这是通过参数-e 实现的

最后输出到 filesink，它只会将你给它的任何内容写入文件。

现在做作业：

根据需要使用元素 - v4l2、alsasrc

添加队列元素以添加缓冲和thread separation

【讨论】：

非常感谢您的回答，您给了我更多细节，但我需要在 c 代码程序中完成，这就是我写这个的原因。通过c代码做的任何细节？谢谢！

我的建议是使用 gst-launch，这可以在几分钟的反复试验中完成。然后，当你让它工作时，尝试通过增强 gist.github.com/crearo/8c71729ed58c4c134cac44db74ea1754

将它重写为 C

我在编写代码之前就这样做了（所以设置正确的 gst-launch-1.0 命令行并对其进行测试），但我仍然无法为 C 中的文件接收器设置任何内容。我试过了在网上找到一些东西，但没有什么完全符合我的要求（也在 gstreamer API 官方教程上）。我会查看您的链接并让您知道，提前谢谢！

您可以为文件接收器提出另一个问题并在此处提供链接.. 或者您可以使用您的 C 代码更新此问题，因为您问题的原始主题没有改变