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08.Appsrc 和 Appsink

丁毅桂2026/1/21大约 5 分钟gstreamernotesappsrcappsink

GStreamer学习笔记：08.Appsrc 和 Appsink

本示例尝试使用 appsrc 和 appsink 元素来实现自定义数据源和从 pipeline 中接收数据。这对于需要将 GStreamer 集成到现有应用程序中的场景非常有用。

核心概念

1. Appsrc - 自定义数据源

appsrc 允许应用程序向 GStreamer pipeline 注入自定义数据。

配置 Appsrc

GstAudioInfo info;
GstCaps *audio_caps;

// 设置音频格式信息
gst_audio_info_set_format(&info, GST_AUDIO_FORMAT_S16, SAMPLE_RATE, 1, NULL);

// 将 info 转换为 caps
audio_caps = gst_audio_info_to_caps(&info);

// 配置 appsrc 元素属性
g_object_set(data.app_src,
             "caps", audio_caps,        // 能力描述
             "format", GST_FORMAT_TIME, // 格式为时间（根据提供的纳秒级时间戳来安排缓冲区的播放顺序和时间）
             NULL
);

// 绑定事件回调函数
g_signal_connect(data.app_src, "need-data", G_CALLBACK(start_feed), &data);
g_signal_connect(data.app_src, "enough-data", G_CALLBACK(stop_feed), &data);

gst_caps_unref(audio_caps);

Appsrc 信号

  • need-data：当 appsrc 内部队列数据不足时触发，请求应用程序提供数据
  • enough-data：当 appsrc 有足够数据时触发，应用程序可以停止推送数据

2. Appsink - 数据接收

appsink 允许应用程序从 GStreamer pipeline 接收数据。

配置 Appsink

g_object_set(data.app_sink,
             "emit-signals", TRUE, // 允许发出例如 `new-sample` 的信号
             "caps", audio_caps,   // 能力描述
             NULL
);

g_signal_connect(data.app_sink, "new-sample", G_CALLBACK(new_sample), &data);

Appsink 信号

  • new-sample：当 appsink 收到一个新 sample 时触发

3. 数据推送机制

推送数据函数

static gboolean push_data(CustomData *data)
{
    GstBuffer *buffer;
    GstFlowReturn ret;
    GstMapInfo map;
    gint16 *raw;
    gint num_samples = CHUNK_SIZE / 2; // 每个采样 16 位
    gfloat freq;

    // 创建缓冲区
    buffer = gst_buffer_new_and_alloc(CHUNK_SIZE);

    // 设置时间戳和持续时间
    // buffer->pts = 采样计数 * 单位时间(ns) / 采样率(hz)
    GST_BUFFER_TIMESTAMP(buffer) = gst_util_uint64_scale(data->num_samples, GST_SECOND, SAMPLE_RATE);

    // buffer->duration = 本缓冲区的采样个数 * 单位时间(ns) / 采样率(hz)
    GST_BUFFER_DURATION(buffer) = gst_util_uint64_scale(num_samples, GST_SECOND, SAMPLE_RATE);

    // 映射缓冲区以进行写入
    gst_buffer_map(buffer, &map, GST_MAP_WRITE);
    raw = (gint16 *)map.data;

    // 生成波形数据
    data->c += data->d;
    data->d -= data->c / 1000;
    freq = 1100 + 1000 * data->d;
    for (int i = 0; i < num_samples; i++)
    {
        data->a += data->b;
        data->b -= data->a / freq;
        raw[i] = (gint16)(500 * data->a);
    }

    gst_buffer_unmap(buffer, &map);
    data->num_samples += num_samples;

    // 推送 buffer 到 appsrc
    g_signal_emit_by_name(data->app_src, "push-buffer", buffer, &ret);

    // 释放 buffer
    gst_buffer_unref(buffer);

    return ret == GST_FLOW_OK;
}

时间戳计算

// gst_util_uint64_scale(a, b, c) 使用 uint64 完成缩放计算: res = a * b / c

// 例如：将 A ∈ [0, 2π] 映射到 B ∈ [0, MAX_INT32]
// B = A / 2π * MAX_INT32 = A * MAX_INT32 / 2π
// B = gst_util_uint64_scale(A, MAX_INT32, 2π)

// 每个缓冲区的时间戳
GST_BUFFER_TIMESTAMP(buffer) = gst_util_uint64_scale(data->num_samples, GST_SECOND, SAMPLE_RATE);

// 每个缓冲区的持续时间
GST_BUFFER_DURATION(buffer) = gst_util_uint64_scale(num_samples, GST_SECOND, SAMPLE_RATE);

4. 信号回调处理

开始推送数据

static void start_feed(GstElement *source, guint size, CustomData *data)
{
    if (data->sourceid == 0)
    {
        g_print("Start feeding\n");
        // 注册 GLib 空闲函数，在主循环空闲时执行
        data->sourceid = g_idle_add((GSourceFunc)push_data, data);
    }
}

停止推送数据

static void stop_feed(GstElement *source, CustomData *data)
{
    if (data->sourceid != 0)
    {
        g_print("Stop feeding\n");
        g_source_remove(data->sourceid);
        data->sourceid = 0;
    }
}

接收 Sample

static GstFlowReturn new_sample(GstElement *sink, CustomData *data)
{
    GstSample *sample;

    g_signal_emit_by_name(sink, "pull-sample", &sample);
    if (sample)
    {
        g_print("*"); // 输出 * 表示接收到一个 buffer
        gst_sample_unref(sample);
        return GST_FLOW_OK;
    }
    return GST_FLOW_ERROR;
}

5. GLib 主循环

data.main_loop = g_main_loop_new(NULL, FALSE);
g_main_loop_run(data.main_loop);
  • 创建 GLib 主循环
  • 上下文为 NULL 代表默认上下文
  • 主循环的作用是等待事件触发、调用绑定的事件处理函数
  • 类似于 Node.js 的事件处理机制

6. Pipeline 结构

app_src -> tee
            tee.src_1 -> audio_queue -> audio_convert1 -> audio_resample -> audio_sink
            tee.src_2 -> video_queue -> audio_convert2 -> visual -> video_convert -> video_sink
            tee.src_3 -> app_queue -> app_sink

完整代码

#include <gst/gst.h>
#include <gst/audio/audio.h>
#include <string.h>

#define CHUNK_SIZE 1024
#define SAMPLE_RATE 44100

typedef struct _CustomData
{
    GstElement *pipeline;
    GstElement *app_src, *tee;
    GstElement *audio_queue, *audio_convert1, *audio_resample, *audio_sink;
    GstElement *video_queue, *audio_convert2, *visual, *video_convert, *video_sink;
    GstElement *app_queue, *app_sink;
    guint64 num_samples;
    gfloat a, b, c, d;
    guint sourceid;
    GMainLoop *main_loop;
} CustomData;

static gboolean push_data(CustomData *data);
static void start_feed(GstElement *source, guint size, CustomData *data);
static void stop_feed(GstElement *source, CustomData *data);
static GstFlowReturn new_sample(GstElement *sink, CustomData *data);
static void error_cb(GstBus *bus, GstMessage *msg, CustomData *data);

int main(int argc, char *argv[])
{
    CustomData data;
    GstPad *tee_pad_1, *tee_pad_2, *tee_pad_3;
    GstPad *queue_audio_pad, *queue_video_pad, *queue_app_pad;
    GstAudioInfo info;
    GstCaps *audio_caps;
    GstBus *bus;

    memset(&data, 0, sizeof(data));
    data.b = 1;
    data.d = 1;

    gst_init(&argc, &argv);

    data.app_src = gst_element_factory_make("appsrc", "audio_source");
    data.tee = gst_element_factory_make("tee", "tee");
    data.audio_queue = gst_element_factory_make("queue", "audio_queue");
    data.audio_convert1 = gst_element_factory_make("audioconvert", "audio_convert1");
    data.audio_resample = gst_element_factory_make("audioresample", "audio_resample");
    data.audio_sink = gst_element_factory_make("autoaudiosink", "audio_sink");
    data.video_queue = gst_element_factory_make("queue", "video_queue");
    data.audio_convert2 = gst_element_factory_make("audioconvert", "audio_convert2");
    data.visual = gst_element_factory_make("wavescope", "visual");
    data.video_convert = gst_element_factory_make("videoconvert", "video_convert");
    data.video_sink = gst_element_factory_make("autovideosink", "video_sink");
    data.app_queue = gst_element_factory_make("queue", "app_queue");
    data.app_sink = gst_element_factory_make("appsink", "app_sink");

    data.pipeline = gst_pipeline_new("test-pipeline");

    if (!data.pipeline || !data.app_src || !data.tee ||
        !data.audio_queue || !data.audio_convert1 || !data.audio_resample || !data.audio_sink ||
        !data.video_queue || !data.audio_convert2 || !data.visual || !data.video_convert || !data.video_sink ||
        !data.app_queue || !data.app_sink)
    {
        g_printerr("Not all elements could be created.\n");
        return -1;
    }

    g_object_set(data.visual, "shader", 0, "style", 0, NULL);

    gst_audio_info_set_format(&info, GST_AUDIO_FORMAT_S16, SAMPLE_RATE, 1, NULL);
    audio_caps = gst_audio_info_to_caps(&info);
    g_object_set(data.app_src, "caps", audio_caps, "format", GST_FORMAT_TIME, NULL);
    g_signal_connect(data.app_src, "need-data", G_CALLBACK(start_feed), &data);
    g_signal_connect(data.app_src, "enough-data", G_CALLBACK(stop_feed), &data);

    g_object_set(data.app_sink, "emit-signals", TRUE, "caps", audio_caps, NULL);
    g_signal_connect(data.app_sink, "new-sample", G_CALLBACK(new_sample), &data);

    gst_caps_unref(audio_caps);

    gst_bin_add_many(GST_BIN(data.pipeline),
        data.app_src, data.tee,
        data.audio_queue, data.audio_convert1, data.audio_resample, data.audio_sink,
        data.video_queue, data.audio_convert2, data.visual, data.video_convert, data.video_sink,
        data.app_queue, data.app_sink,
        NULL
    );

    if (gst_element_link_many(data.app_src, data.tee, NULL) != TRUE ||
        gst_element_link_many(data.audio_queue, data.audio_convert1, data.audio_resample, data.audio_sink, NULL) != TRUE ||
        gst_element_link_many(data.video_queue, data.audio_convert2, data.visual, data.video_convert, data.video_sink, NULL) != TRUE ||
        gst_element_link_many(data.app_queue, data.app_sink, NULL) != TRUE)
    {
        g_printerr("Elements could not be linked.\n");
        gst_object_unref(data.pipeline);
        return -1;
    }

    queue_audio_pad = gst_element_get_static_pad(data.audio_queue, "sink");
    queue_video_pad = gst_element_get_static_pad(data.video_queue, "sink");
    queue_app_pad = gst_element_get_static_pad(data.app_queue, "sink");
    tee_pad_1 = gst_element_request_pad_simple(data.tee, "src_%u");
    tee_pad_2 = gst_element_request_pad_simple(data.tee, "src_%u");
    tee_pad_3 = gst_element_request_pad_simple(data.tee, "src_%u");

    if (gst_pad_link(tee_pad_1, queue_audio_pad) != GST_PAD_LINK_OK ||
        gst_pad_link(tee_pad_2, queue_video_pad) != GST_PAD_LINK_OK ||
        gst_pad_link(tee_pad_3, queue_app_pad) != GST_PAD_LINK_OK)
    {
        g_printerr("Tee could not be linked\n");
        gst_object_unref(data.pipeline);
        return -1;
    }
    gst_object_unref(queue_audio_pad);
    gst_object_unref(queue_video_pad);
    gst_object_unref(queue_app_pad);

    bus = gst_element_get_bus(data.pipeline);
    gst_bus_add_signal_watch(bus);
    g_signal_connect(G_OBJECT(bus), "message::error", (GCallback)error_cb, &data);
    gst_object_unref(bus);

    gst_element_set_state(data.pipeline, GST_STATE_PLAYING);

    data.main_loop = g_main_loop_new(NULL, FALSE);
    g_main_loop_run(data.main_loop);

    gst_element_release_request_pad(data.tee, tee_pad_1);
    gst_element_release_request_pad(data.tee, tee_pad_2);
    gst_element_release_request_pad(data.tee, tee_pad_3);
    gst_object_unref(tee_pad_1);
    gst_object_unref(tee_pad_2);
    gst_object_unref(tee_pad_3);

    gst_element_set_state(data.pipeline, GST_STATE_NULL);
    gst_object_unref(data.pipeline);
    return 0;
}

编译和运行

gcc main.c -o main.out $(pkg-config --cflags --libs gstreamer-1.0)
./main.out

总结

本示例展示了：

  1. Appsrc：自定义数据源，向 pipeline 注入数据
  2. Appsink：数据接收，从 pipeline 获取处理后的数据
  3. 时间戳管理：正确设置 buffer 的 PTS 和 Duration
  4. 信号机制：使用 need-data 和 enough-data 控制数据推送
  5. GLib 主循环：事件驱动的异步处理
  6. 多分支 Pipeline：使用 tee 分发数据到多个分支

Appsrc 和 Appsink 是 GStreamer 与应用程序集成的关键组件，允许将 GStreamer 的强大媒体处理能力集成到任何应用程序中。

贡献者: YiguiDing
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