使用GStreamer播放原始PCM数据

时间:2016-11-16 19:12:59

标签: c gstreamer

好的,我已经编写了一些应该打开和音频输出设备的代码,每当实际的PCM数据可用时,我会调用Sound_WriteFrame()并堆积更多要播放的数据。

数据是原始的,没有标题,所以当我调用Sound_Open()时,我会传递这些信息,让GStreamer知道将会到达什么样的数据。

这段代码不起作用 - 坦率地说我不知道​​我在做什么,我发现GStreamer有点难以使用,我希望最终可以改变。

我正在使用GStreamer 1.0。感谢帮助。

#include <gstreamer-1.0/gst/gst.h>
#include <gstreamer-1.0/gst/gstelement.h>
#include <gtk/gtk.h>
#include <stdlib.h>
#include <string.h>

typedef struct _sound_t {
    GstElement *source, *sink, *pipeline;
    GMemoryInputStream *giostream;
    GstPad *sourcepad;
    GstCaps *gcaps;
    int bDeviceOpen;
} SOUND_CTX;

int Sound_Close(SOUND_CTX *p){
    printf("Closing\n");
    gst_element_set_state(p->pipeline, GST_STATE_NULL);
    return 1;
}

void Sound_SetState(SOUND_CTX *p, GstState state){
    GstStateChangeReturn r = gst_element_set_state(p->pipeline, state);
    switch(r){
        case GST_STATE_CHANGE_FAILURE: printf("GST_STATE_CHANGE_FAILURE\n"); break;
        case GST_STATE_CHANGE_SUCCESS: printf("GST_STATE_CHANGE_SUCCESS\n"); break;
        case GST_STATE_CHANGE_ASYNC: printf("GST_STATE_CHANGE_ASYNC\n"); break;
        case GST_STATE_CHANGE_NO_PREROLL: printf("GST_STATE_CHANGE_NO_PREROLL\n"); break;
        default: printf("Unknown state\n"); break;
    }
    return;
}

int Sound_Open(SOUND_CTX *p, int nSamplesPerSec, int nChannels){

    p->source = gst_element_factory_make("giostreamsrc", "source");
    p->giostream = G_MEMORY_INPUT_STREAM(g_memory_input_stream_new());
    g_object_set(G_OBJECT(p->source), "stream", G_INPUT_STREAM(p->giostream), NULL);
    p->sourcepad = gst_element_get_static_pad(p->source, "src");

    p->gcaps = gst_caps_new_simple(
        "audio/x-raw",
        "rate", G_TYPE_INT, nSamplesPerSec,
        "channels", G_TYPE_INT, nChannels,
        "width", G_TYPE_INT, 16,
        "depth", G_TYPE_INT, 16,
        "signed", G_TYPE_BOOLEAN, TRUE,
        NULL
    );

    gst_pad_set_caps(p->sourcepad, p->gcaps);
    gst_object_unref(p->sourcepad);

    p->sink = gst_element_factory_make("alsasink", "sink");
    p->pipeline = gst_pipeline_new("pipeline_name");
    gst_bin_add_many(GST_BIN(p->pipeline), p->source, p->sink, NULL);
    gst_element_link_many(p->source, p->sink, NULL);

    Sound_SetState(p, GST_STATE_PLAYING);

    return 1;
}

int Sound_WriteFrame(SOUND_CTX *p, void *lpData, unsigned int size){
    g_memory_input_stream_add_data(
        G_MEMORY_INPUT_STREAM(p->giostream),
        lpData, size, NULL
    );
    return 0;
}

int timer_callback(const void *data){
    g_main_loop_quit((GMainLoop *)data);
    return FALSE;
}



int main(int argc, char *argv[]){
    gst_init(&argc, &argv);
    GMainLoop *loop = NULL;
    SOUND_CTX a;
    memset(&a, 0x00, sizeof(SOUND_CTX));
    if(Sound_Open(&a, 44100, 2)){
        FILE *handle;
        unsigned char tmp[4096];
        if((handle = fopen("test.pcm", "rb")) != NULL){
            while(fread(tmp, 1, sizeof(tmp), handle) == sizeof(tmp)){
                Sound_WriteFrame(&a, tmp, sizeof(tmp));
            }
            fclose(handle);
        }
    }

    loop = g_main_loop_new(NULL, FALSE);
    g_timeout_add(5500, (GSourceFunc)timer_callback, loop);
    g_main_loop_run(loop);
    Sound_Close(&a);
    g_main_loop_unref(loop);

    return 0;
}

好的,所以这里有一个更新的来源,但扬声器没有声音

#include <gstreamer-1.0/gst/gst.h>
#include <gstreamer-1.0/gst/gstelement.h>
#include <gstreamer-1.0/gst/app/gstappsrc.h>
#include <gtk/gtk.h>
#include <stdlib.h>
#include <string.h>

typedef struct _sound_t {
    GstElement *sink, *pipeline;
    GstAppSrc *src;
    GstCaps *pcm_caps;
} SOUND_CTX;

int Sound_Close(SOUND_CTX *p){
    printf("Closing\n");
    gst_element_set_state(p->pipeline, GST_STATE_NULL);
    return 1;
}

void Sound_SetState(SOUND_CTX *p, GstState state){
    GstStateChangeReturn r = gst_element_set_state(p->pipeline, state);
    switch(r){
        case GST_STATE_CHANGE_FAILURE: printf("GST_STATE_CHANGE_FAILURE\n"); break;
        case GST_STATE_CHANGE_SUCCESS: printf("GST_STATE_CHANGE_SUCCESS\n"); break;
        case GST_STATE_CHANGE_ASYNC: printf("GST_STATE_CHANGE_ASYNC\n"); break;
        case GST_STATE_CHANGE_NO_PREROLL: printf("GST_STATE_CHANGE_NO_PREROLL\n"); break;
        default: printf("Unknown state\n"); break;
    }
    return;
}

int Sound_Open(SOUND_CTX *p, int nSamplesPerSec, int nChannels){

    p->pipeline = gst_pipeline_new("pipeline_name");
    p->sink = gst_element_factory_make("alsasink", "sink");

    p->src = (GstAppSrc*) gst_element_factory_make("appsrc", "source");
    gst_app_src_set_stream_type(p->src, GST_APP_STREAM_TYPE_STREAM);

    // I am hardcoding the format, channels, and rate for now
    p->pcm_caps = gst_caps_from_string("audio/x-raw,format=S16LE,rate=44100,channels=2");
    gst_app_src_set_caps(p->src, p->pcm_caps);

    gst_bin_add_many(GST_BIN(p->pipeline), (GstElement*)p->src, p->sink, NULL);
    gst_element_link_many((GstElement*)p->src, p->sink, NULL);

    Sound_SetState(p, GST_STATE_PLAYING);

    return 1;
}

int Sound_WriteFrame(SOUND_CTX *p, void *lpData, unsigned int size){
    GstBuffer *buf = NULL;
    void *lpHeapData = NULL;
    if((lpHeapData = g_malloc(size)) == NULL) return 0;
    memcpy(lpHeapData, lpData, size);
    buf = gst_buffer_new_wrapped(lpHeapData, size);
    if(buf == NULL){
        g_free(lpHeapData);
        return 0;
    }
    gst_app_src_push_buffer(p->src, buf);
    return 0;
}

int timer_callback(const void *data){
    g_main_loop_quit((GMainLoop *)data);
    return FALSE;
}



int main(int argc, char *argv[]){
    gst_init(&argc, &argv);
    GMainLoop *loop = NULL;
    SOUND_CTX a;
    memset(&a, 0x00, sizeof(SOUND_CTX));
    if(Sound_Open(&a, 44100, 2)){
        FILE *handle;
        unsigned char tmp[4096];
        if((handle = fopen("test.pcm", "rb")) != NULL){
            while(fread(tmp, 1, sizeof(tmp), handle) == sizeof(tmp)){
                Sound_WriteFrame(&a, tmp, sizeof(tmp));
            }
            fclose(handle);
        }
    }

    loop = g_main_loop_new(NULL, FALSE);
    g_timeout_add(5500, (GSourceFunc)timer_callback, loop);
    g_main_loop_run(loop);
    Sound_Close(&a);
    g_main_loop_unref(loop);

    return 0;
}

1 个答案:

答案 0 :(得分:0)

您正在使用GStreamer 0.10 Caps字段,其内容类似于GStreamer 1.0 Code。感兴趣的是,宽度/深度/有符号已被称为格式的单个字段所取代。有关完整列表,请参阅文档,大写字母中的名称与枚举相同,但没有名称空间GST_AUDIO_FORMAT _。

https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstaudio.html#GstAudioFormat