GST_DEBUG_DUMP_DOT_DIR=. GST_DEBUG=3 gst-launch-1...

Tạo vào: 20 tháng 6, 2025

Trả lời bằng O3 bởi Chat01

Câu hỏi

GST_DEBUG_DUMP_DOT_DIR=. GST_DEBUG=3 gst-launch-1.0 urisourcebin uri=http://devimages.apple.com.edgekey.net/streaming/examples/bipbop_4x3/gear2/prog_index.m3u8 name=src ! decodebin3 name=dec dec. ! queue ! videoconvert ! videoscale ! autovideosink dec. ! queue ! audioconvert ! audioresample ! volume name=volume ! level ! autoaudiosink
设置暂停管道 ...
管道正在 PREROLLING ...
0:00:00.026710012 22626 0x55d1a8d25350 WARN structure gststructure.c:3100:priv_gst_structure_append_to_gstring: No value transform to serialize field 'session' of type 'GstSoupSession'
从组件“souphttpsrc0”获取上下文:gst.soup.session=context, session=(GstSoupSession)NULL;
重新分配延迟时间...
0:00:01.335211756 22626 0x7f033404e4c0 FIXME decodebin3 gstdecodebin3.c:1186:gst_decodebin_input_unblock_streams:<dec> Re-use existing input streams if/when possible
0:00:01.335293014 22626 0x7f033404e4c0 FIXME decodebin3 gstdecodebin3.c:1186:gst_decodebin_input_unblock_streams:<dec> Re-use existing input streams if/when possible
重新分配延迟时间...
重新分配延迟时间...
重新分配延迟时间...
设置播放管道 ..
现象:播放只有一幀图像,没有声音,播放进度为0

Suy nghĩ

Troubleshooting GStreamer pipeline

The user is describing an issue with a GStreamer pipeline where only one frame is shown, no sound, and the progress gets stuck. They mention using urisourcebin, decodebin3, and other elements while trying to play an HLS stream. It seems they're looking for help troubleshooting why this is happening. I'll consider potential issues, like incorrect pipeline setup or missing/incorrect elements, and provide suggestions to fix the issue. I'll also think about whether I need to look up any specific resources or guides.

Debugging GStreamer pipeline

The user is experiencing an issue with a GStreamer pipeline involving urisourcebin, decodebin3, and HLS streams. The pipeline shows the first frame but no sound, and playback is stuck. It seems they missed linking the correct pads for urisourcebin. In the pipeline, src was not connected properly to decodebin3. The fix requires the proper connection (src. to dec). There is also a possibility that the hlsdemux plugin might be missing, but the rest of the pipeline looks mostly correct.

Debugging GStreamer playback

The issue seems related to how the pipeline is set up and which elements are used. There may be a missing queue after the audio path or issues related to the decodebin3's handling of dynamic pads. Also, urisourcebin is not properly linking to decodebin3, causing the pipeline to stall. Best advice: use gst-play-1.0 or playbin3 for HLS streams to handle this more smoothly. Alternatively, try using uridecodebin instead of urisourcebin along with corrected linking and necessary plugins.

Thought for 28 seconds

Trả lời

问题根源

urisourcebin 只负责把网络流取下来并解析成 动态 输出 pad,而不是把它们自动连到解码器。
gst-launch-1.0 里,只有当两个元素在 启动时就已经有 pad 时,用 ! 才能一次性把它们“焊”在一起。
HLS 源在一开始是没有 pad 的 → urisourcebin ! decodebin3 实际上 没连上
结果管线能取得头一个切片并完成 preroll,画面停在首帧,音轨也没有真正流进来,进度保持 0。


最省事的做法:直接用 会自己处理动态 pad 的高层元素

bash
# 最简单的播放测试 gst-play-1.0 \ http://devimages.apple.com.edgekey.net/streaming/examples/bipbop_4x3/gear2/prog_index.m3u8

bash
gst-launch-1.0 \ playbin3 uri=\ http://devimages.apple.com.edgekey.net/streaming/examples/bipbop_4x3/gear2/prog_index.m3u8

如果仍想手工拼管线,用 uridecodebin

uridecodebin = urisourcebin + decodebin3,并且会自己把动态 pad 连好:

bash
GST_DEBUG=3 gst-launch-1.0 \ uridecodebin uri=http://devimages.apple.com.edgekey.net/streaming/examples/bipbop_4x3/gear2/prog_index.m3u8 name=dec \ dec. ! queue ! videoconvert ! videoscale ! autovideosink \ dec. ! queue ! audioconvert ! audioresample ! autoaudiosink

要点

  • dec. 前面的空格和点号告诉 gst-launch:等 uridecodebin 产生 pad 后再去连。
  • 每条支路加 queue,避免视频/音频互相阻塞。

想继续用 urisourcebin?那就得自己写代码

在 C/GLib、Python、Rust… 里捕捉 pad-added 信号

c
g_signal_connect (src, "pad-added", G_CALLBACK (on_src_pad_added), decodebin);

然后在回调里把新 pad 手动 link 到 decodebin3,或直接 link 到下游 element。
gst-launch-1.0 无法做这种“运行时布线”,因此不适合用 urisourcebin + decodebin3 的组合来播 HLS。


其它排查建议

检查项说明
插件是否齐全gst-inspect-1.0 adaptivedemux2 hlsdemux avdec_h264 faad 若有缺失请安装对应插件集(gst-plugins-bad, gst-libav 等)。
时钟同步如果只要视频首帧、音频缺失,有时是音频解码/输出插件缺失导致时钟挂起。
版本问题gst-launch-1.0 --version 建议 1.22+;低版本对 HLS 支持不稳定。
更细日志GST_DEBUG=*adaptive*:4,*hls*:4 可看出 segment/playlist 下载、切换与 demux 情况。

照以上方法换成 uridecodebinplaybin3 后,通常就能连播、声音视频都正常,播放进度也会向前走。祝调试顺利!

Câu hỏi

如果是自己写的gstreamer插件呢,请问 playbin3 如何使用私有插件播放

Suy nghĩ

If the plugin is not in the registry, you can load its path programmatically using gst_registry_scan_path after gst_init, or by using gst_plugin_load_file. For Android, set the GST_PLUGIN_PATH environment variable or compile the plugin statically, noting the differences. After building the plugin, you might need to remove the plugin cache (/.cache/gstreamer-1.0/registry.x86_64.bin) and run gst-inspect to update the cache. I'll also search for more sources on the playbin3 video-sink property and provide example commands.

To ensure your custom plugin is used with playbin3, you'll need to either set the GST_PLUGIN_PATH environment variable or directly specify the plugin path with --gst-plugin-path. You can use gst-inspect to confirm the plugin is loaded. If you're using a custom source, decoder, or sink, you’ll need to ensure the plugin is correctly classified and ranked so that uridecodebin picks it properly. I’ll also include debugging tips and examples for static linking and runtime loading.

Thought for 46 seconds

Trả lời

下面这几个步骤就能让 playbin3 顺利找到并使用你自己写的私有插件。按需要任选其一或组合使用即可。


1 把插件放到 GStreamer 的搜索路径里

  1. 把编译好的 .so 文件放到一个单独目录,例如 ~/gst-private
  2. 运行前设环境变量,或直接在命令行里指定:
bash
# 方法 A:环境变量 export GST_PLUGIN_PATH=$HOME/gst-private # 可以设多个目录,用 : 分隔 gst-inspect-1.0 myelement # 先确认插件已被注册 # 方法 B:一次性 gst-launch-1.0 --gst-plugin-path=$HOME/gst-private \ playbin3 uri=file:///tmp/test.mp4

GStreamer 会扫描 GST_PLUGIN_PATH 中的目录并把插件登记进运行时插件表 stackoverflow.com

调试小贴士:
GST_DEBUG=GST_PLUGIN_LOADING:6 能打印每个插件的加载过程。


2 让 playbin3 “点名” 使用你的元素

2.1 你的元素是 音/视频渲染器(sink)

直接把实例(或一个自定义 Bin)塞给 video-sink / audio-sink 属性:

bash
GST_PLUGIN_PATH=~/gst-private \ gst-launch-1.0 playbin3 \ uri=file:///tmp/test.mp4 \ video-sink="myvideosink" \ audio-sink="myaudiosink"

playbin3 默认会用 autovideosink/autoaudiosink,手动指定即可覆盖 gstreamer.freedesktop.org stackoverflow.com

2.2 你的元素是 解码器 / 解析器 / 协议源

  • GST_PLUGIN_DEFINE / gst_element_register() 里给元素设置合适的 classification(如 Decoder/Video)和 pad caps,这样 uridecodebin3 能识别它。
  • rank 设为 GST_RANK_PRIMARY(256)或更高;rank 越高,自动插件器越优先选它 gstreamer.freedesktop.org
    c
    gst_element_register (plugin, "myh264dec", GST_RANK_PRIMARY + 1, GST_TYPE_MY_H264_DEC);
  • 也可以在程序里动态提升 rank:
    c
    GstPluginFeature *f = gst_element_factory_find ("myh264dec"); gst_plugin_feature_set_rank (f, GST_RANK_PRIMARY + 1); ``` [developer.ridgerun.com](https://developer.ridgerun.com/wiki/index.php/GStreamer_modify_the_elements_rank?utm_source=chatgpt.com)

2.3 想把自定义滤镜强插到解码链

创建一个包含你滤镜的 GstBin,然后通过 video-filter / audio-filter 属性插进去:

c
GstElement *flt = gst_parse_bin_from_description ("myfilter ! videoconvert", TRUE, NULL); g_object_set (playbin, "video-filter", flt, NULL);

3 不用 .so,也能让 playbin3 找到元素——静态注册

把插件代码直接和应用链接,在初始化里调用 gst_plugin_register_static()

c
gst_plugin_register_static (GST_VERSION_MAJOR, GST_VERSION_MINOR, "my-static", "Private elements", plugin_init_cb, "1.0", "LGPL", "myapp", "myapp", "https://example.com");

静态注册后,gst_element_factory_make("myvideosink", NULL) 就能正常工作,完全不依赖文件系统扫描 gstreamer.freedesktop.org


4 纯代码方式按需加载动态库

如果你不想改环境变量,可以在 gst_init() 之后手动:

c
GstRegistry *r = gst_registry_get (); gst_registry_scan_path (r, "/opt/myplugins"); // 扫描目录 gst_plugin_load_file ("/opt/myplugins/libgstmy.so", NULL); // 直接加载单个 .so

FAQ & 调试清单

现象排查要点
gst-inspect-1.0 看不到元素路径没进 GST_PLUGIN_PATH;或者 .so 名必须形如 libgstXXX.so
playbin3 还是没用到你的解码器rank 太低;或 pad caps/classification 不匹配
代码修改后仍加载旧插件删除 ~/.cache/gstreamer-1.0/registry*.bin 后重试

按照上面流程,把路径加进来 + 保证元素 rank/分类正确,playbin3 就能像使用系统插件一样自动挑选并使用你的私有插件。祝编译顺利、调试顺心!

Câu hỏi

playbin3 如何设置音量和重采样呢

Suy nghĩ

Explaining playbin3 settings

To set volume in playbin3, you can adjust the "volume" property directly, which is a value between 0 and 1, or use the "audio-filter" property to insert elements like "volume" or "audioresample." Alternatively, you can define an audio-sink that includes a chain like "volume ! audioresample ! autoaudiosink." Playbin3 also has a "mute" boolean to control sound. Regarding resampling, using "audioresample" can help with adjusting sample rates in the audio pipeline. Simple adjustments like these give you fine control over both volume and audio processing in GStreamer.

Thought for 7 seconds

Trả lời

playbin3 里有两条常用途径来控制「音量」和「重采样(audioresample)」——

方案适用场景做法示例(gst-launch-1.0)要点
直接用 playbin3 自带属性只想调整体响度,别的保持自动gst-launch-1.0 playbin3 uri=file:///tmp/demo.mp4 volume=0.4volume 属性是 gdouble,0 = 静音,1 = 100 %。<br>• 可在运行中通过 g_object_set(playbin, "volume", 0.8, NULL); 动态调节。<br>• 还有 mute (gboolean) 可直接开关静音。<br>• 采样率转换由自动协商完成,无需手动插 audioresample
在音频链插自定义滤镜想显式加 volumeaudioresample、capsfilter 等```bash
gst-launch-1.0 playbin3 \
uri=file:///tmp/demo.mp4 \
audio-filter="volume name=vol ! audioresample ! audio/x-raw,rate=48000"
```audio-filter / video-filter 属性接受 bin 描述字符串<br>• 这段 bin 会被插在解码器之后、audio-sink 之前。<br>• 可给元件取名 (name=vol) 方便后续用 gst-launch-1.0remote-control 或在代码里找回调节。<br>• 想指定声道数、格式等,同样在 audio-filter 里加 capsfilter 即可:… ! audio/x-raw,channels=2,format=S16LE
完全替换 audio-sink想把自定义链放到最末端(含 sink)```bash
gst-launch-1.0 playbin3 \
uri=file:///tmp/demo.mp4 \
audio-sink="volume ! audioresample ! autoaudiosink"
```audio-sink 期望的是 元素实例bin<br>• 因为已经在最末端,必须自己包含一个真正的 sink(例如 autoaudiosink / pulsesink 等)。

在 C / C++ / Python (GObject) 代码里

c
/* 创建并初始化 playbin3 */ GstElement *playbin = gst_element_factory_make ("playbin3", NULL); g_object_set (playbin, "uri", "file:///tmp/demo.mp4", "volume", 0.7, /* 方案 1 */ NULL); /* 方案 2:插 audio-filter */ GstElement *aflt = gst_parse_bin_from_description ( "volume name=vol ! audioresample ! audio/x-raw,rate=48000", TRUE, NULL); g_object_set (playbin, "audio-filter", aflt, NULL); /* 随时动态调节音量 */ GstElement *vol = gst_bin_get_by_name (GST_BIN (aflt), "vol"); g_object_set (vol, "volume", 0.3, NULL);

FAQ - 常见疑问

问题答案
audioresample 必须手动加吗?一般不用。只要上游和下游协商不一致,GStreamer 会自动在 pipeline 里插入 audioresample。只有当你想 强制 输出固定采样率或需要精确控制时才显式加入。
能同时用 audio-filteraudio-sink 吗?可以,两者互不冲突。常见做法是把全局处理(EQ、限幅器等)放在 audio-filter,然后在 audio-sink 里替换最终输出(比如自定义的 ALSA sink)。
Python 中怎么拿到 volume 元件?```python
vol = playbin.get_property("audio-filter").get_by_name("vol")
vol.set_property("volume", 0.5)
text
| **playbin 与 playbin3 的差异?** | `playbin3` 使用 `urisourcebin`+`decodebin3`,对自适应流(HLS/DASH)更友好,并且支持 `audio-filter` / `video-filter` 这类新属性;在 1.22+ 建议直接用 `playbin3`。 | 只要记住 **“简单调音量 → 用 playbin3 自带属性;想自定义链 → 用 audio-filter/audio-sink”** 这条原则,就能灵活地在 `playbin3` 里加上任意音频处理元素。祝开发顺利!

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