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-rw-r--r--src/test-native/ffmpeg/api-example.c479
-rw-r--r--src/test-native/ffmpeg/avcodec_sample.c203
-rw-r--r--src/test-native/ffmpeg/avcodec_sample.sh6
-rw-r--r--src/test-native/gst/helloworld-auto.c112
-rw-r--r--src/test-native/gst/helloworld-playbin.c75
-rw-r--r--src/test-native/gst/helloworld-playbin2.c75
-rw-r--r--src/test-native/gst/helloworld.c142
-rw-r--r--src/test-native/gst/make.sh5
8 files changed, 1097 insertions, 0 deletions
diff --git a/src/test-native/ffmpeg/api-example.c b/src/test-native/ffmpeg/api-example.c
new file mode 100644
index 000000000..970a90eab
--- /dev/null
+++ b/src/test-native/ffmpeg/api-example.c
@@ -0,0 +1,479 @@
+/*
+ * copyright (c) 2001 Fabrice Bellard
+ *
+ * This file is part of Libav.
+ *
+ * Libav is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * Libav is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with Libav; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
+ */
+
+/**
+ * @file
+ * libavcodec API use example.
+ *
+ * @example libavcodec/api-example.c
+ * Note that this library only handles codecs (mpeg, mpeg4, etc...),
+ * not file formats (avi, vob, etc...). See library 'libavformat' for the
+ * format handling
+ */
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+
+#ifdef HAVE_AV_CONFIG_H
+#undef HAVE_AV_CONFIG_H
+#endif
+
+#include "libavcodec/avcodec.h"
+#include "libavutil/mathematics.h"
+#include "libavutil/samplefmt.h"
+
+#define INBUF_SIZE 4096
+#define AUDIO_INBUF_SIZE 20480
+#define AUDIO_REFILL_THRESH 4096
+
+/*
+ * Audio encoding example
+ */
+static void audio_encode_example(const char *filename)
+{
+ AVCodec *codec;
+ AVCodecContext *c= NULL;
+ int frame_size, i, j, out_size, outbuf_size;
+ FILE *f;
+ short *samples;
+ float t, tincr;
+ uint8_t *outbuf;
+
+ printf("Audio encoding\n");
+
+ /* find the MP2 encoder */
+ codec = avcodec_find_encoder(CODEC_ID_MP2);
+ if (!codec) {
+ fprintf(stderr, "codec not found\n");
+ exit(1);
+ }
+
+ c = avcodec_alloc_context3(codec);
+
+ /* put sample parameters */
+ c->bit_rate = 64000;
+ c->sample_rate = 44100;
+ c->channels = 2;
+
+ /* open it */
+ if (avcodec_open(c, codec) < 0) {
+ fprintf(stderr, "could not open codec\n");
+ exit(1);
+ }
+
+ /* the codec gives us the frame size, in samples */
+ frame_size = c->frame_size;
+ samples = malloc(frame_size * 2 * c->channels);
+ outbuf_size = 10000;
+ outbuf = malloc(outbuf_size);
+
+ f = fopen(filename, "wb");
+ if (!f) {
+ fprintf(stderr, "could not open %s\n", filename);
+ exit(1);
+ }
+
+ /* encode a single tone sound */
+ t = 0;
+ tincr = 2 * M_PI * 440.0 / c->sample_rate;
+ for(i=0;i<200;i++) {
+ for(j=0;j<frame_size;j++) {
+ samples[2*j] = (int)(sin(t) * 10000);
+ samples[2*j+1] = samples[2*j];
+ t += tincr;
+ }
+ /* encode the samples */
+ out_size = avcodec_encode_audio(c, outbuf, outbuf_size, samples);
+ fwrite(outbuf, 1, out_size, f);
+ }
+ fclose(f);
+ free(outbuf);
+ free(samples);
+
+ avcodec_close(c);
+ av_free(c);
+}
+
+/*
+ * Audio decoding.
+ */
+static void audio_decode_example(const char *outfilename, const char *filename)
+{
+ AVCodec *codec;
+ AVCodecContext *c= NULL;
+ int len;
+ FILE *f, *outfile;
+ uint8_t inbuf[AUDIO_INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
+ AVPacket avpkt;
+ AVFrame *decoded_frame = NULL;
+
+ av_init_packet(&avpkt);
+
+ printf("Audio decoding\n");
+
+ /* find the mpeg audio decoder */
+ codec = avcodec_find_decoder(CODEC_ID_MP2);
+ if (!codec) {
+ fprintf(stderr, "codec not found\n");
+ exit(1);
+ }
+
+ c = avcodec_alloc_context3(codec);
+
+ /* open it */
+ if (avcodec_open(c, codec) < 0) {
+ fprintf(stderr, "could not open codec\n");
+ exit(1);
+ }
+
+ f = fopen(filename, "rb");
+ if (!f) {
+ fprintf(stderr, "could not open %s\n", filename);
+ exit(1);
+ }
+ outfile = fopen(outfilename, "wb");
+ if (!outfile) {
+ av_free(c);
+ exit(1);
+ }
+
+ /* decode until eof */
+ avpkt.data = inbuf;
+ avpkt.size = fread(inbuf, 1, AUDIO_INBUF_SIZE, f);
+
+ while (avpkt.size > 0) {
+ int got_frame = 0;
+
+ if (!decoded_frame) {
+ if (!(decoded_frame = avcodec_alloc_frame())) {
+ fprintf(stderr, "out of memory\n");
+ exit(1);
+ }
+ } else
+ avcodec_get_frame_defaults(decoded_frame);
+
+ len = avcodec_decode_audio4(c, decoded_frame, &got_frame, &avpkt);
+ if (len < 0) {
+ fprintf(stderr, "Error while decoding\n");
+ exit(1);
+ }
+ if (got_frame) {
+ /* if a frame has been decoded, output it */
+ int data_size = av_samples_get_buffer_size(NULL, c->channels,
+ decoded_frame->nb_samples,
+ c->sample_fmt, 1);
+ fwrite(decoded_frame->data[0], 1, data_size, outfile);
+ }
+ avpkt.size -= len;
+ avpkt.data += len;
+ if (avpkt.size < AUDIO_REFILL_THRESH) {
+ /* Refill the input buffer, to avoid trying to decode
+ * incomplete frames. Instead of this, one could also use
+ * a parser, or use a proper container format through
+ * libavformat. */
+ memmove(inbuf, avpkt.data, avpkt.size);
+ avpkt.data = inbuf;
+ len = fread(avpkt.data + avpkt.size, 1,
+ AUDIO_INBUF_SIZE - avpkt.size, f);
+ if (len > 0)
+ avpkt.size += len;
+ }
+ }
+
+ fclose(outfile);
+ fclose(f);
+
+ avcodec_close(c);
+ av_free(c);
+ av_free(decoded_frame);
+}
+
+/*
+ * Video encoding example
+ */
+static void video_encode_example(const char *filename)
+{
+ AVCodec *codec;
+ AVCodecContext *c= NULL;
+ int i, out_size, size, x, y, outbuf_size;
+ FILE *f;
+ AVFrame *picture;
+ uint8_t *outbuf, *picture_buf;
+
+ printf("Video encoding\n");
+
+ /* find the mpeg1 video encoder */
+ codec = avcodec_find_encoder(CODEC_ID_MPEG1VIDEO);
+ if (!codec) {
+ fprintf(stderr, "codec not found\n");
+ exit(1);
+ }
+
+ c = avcodec_alloc_context3(codec);
+ picture= avcodec_alloc_frame();
+
+ /* put sample parameters */
+ c->bit_rate = 400000;
+ /* resolution must be a multiple of two */
+ c->width = 352;
+ c->height = 288;
+ /* frames per second */
+ c->time_base= (AVRational){1,25};
+ c->gop_size = 10; /* emit one intra frame every ten frames */
+ c->max_b_frames=1;
+ c->pix_fmt = PIX_FMT_YUV420P;
+
+ /* open it */
+ if (avcodec_open(c, codec) < 0) {
+ fprintf(stderr, "could not open codec\n");
+ exit(1);
+ }
+
+ f = fopen(filename, "wb");
+ if (!f) {
+ fprintf(stderr, "could not open %s\n", filename);
+ exit(1);
+ }
+
+ /* alloc image and output buffer */
+ outbuf_size = 100000;
+ outbuf = malloc(outbuf_size);
+ size = c->width * c->height;
+ picture_buf = malloc((size * 3) / 2); /* size for YUV 420 */
+
+ picture->data[0] = picture_buf;
+ picture->data[1] = picture->data[0] + size;
+ picture->data[2] = picture->data[1] + size / 4;
+ picture->linesize[0] = c->width;
+ picture->linesize[1] = c->width / 2;
+ picture->linesize[2] = c->width / 2;
+
+ /* encode 1 second of video */
+ for(i=0;i<25;i++) {
+ fflush(stdout);
+ /* prepare a dummy image */
+ /* Y */
+ for(y=0;y<c->height;y++) {
+ for(x=0;x<c->width;x++) {
+ picture->data[0][y * picture->linesize[0] + x] = x + y + i * 3;
+ }
+ }
+
+ /* Cb and Cr */
+ for(y=0;y<c->height/2;y++) {
+ for(x=0;x<c->width/2;x++) {
+ picture->data[1][y * picture->linesize[1] + x] = 128 + y + i * 2;
+ picture->data[2][y * picture->linesize[2] + x] = 64 + x + i * 5;
+ }
+ }
+
+ /* encode the image */
+ out_size = avcodec_encode_video(c, outbuf, outbuf_size, picture);
+ printf("encoding frame %3d (size=%5d)\n", i, out_size);
+ fwrite(outbuf, 1, out_size, f);
+ }
+
+ /* get the delayed frames */
+ for(; out_size; i++) {
+ fflush(stdout);
+
+ out_size = avcodec_encode_video(c, outbuf, outbuf_size, NULL);
+ printf("write frame %3d (size=%5d)\n", i, out_size);
+ fwrite(outbuf, 1, out_size, f);
+ }
+
+ /* add sequence end code to have a real mpeg file */
+ outbuf[0] = 0x00;
+ outbuf[1] = 0x00;
+ outbuf[2] = 0x01;
+ outbuf[3] = 0xb7;
+ fwrite(outbuf, 1, 4, f);
+ fclose(f);
+ free(picture_buf);
+ free(outbuf);
+
+ avcodec_close(c);
+ av_free(c);
+ av_free(picture);
+ printf("\n");
+}
+
+/*
+ * Video decoding example
+ */
+
+static void pgm_save(unsigned char *buf, int wrap, int xsize, int ysize,
+ char *filename)
+{
+ FILE *f;
+ int i;
+
+ f=fopen(filename,"w");
+ fprintf(f,"P5\n%d %d\n%d\n",xsize,ysize,255);
+ for(i=0;i<ysize;i++)
+ fwrite(buf + i * wrap,1,xsize,f);
+ fclose(f);
+}
+
+static void video_decode_example(const char *outfilename, const char *filename)
+{
+ AVCodec *codec;
+ AVCodecContext *c= NULL;
+ int frame, got_picture, len;
+ FILE *f;
+ AVFrame *picture;
+ uint8_t inbuf[INBUF_SIZE + FF_INPUT_BUFFER_PADDING_SIZE];
+ char buf[1024];
+ AVPacket avpkt;
+
+ av_init_packet(&avpkt);
+
+ /* set end of buffer to 0 (this ensures that no overreading happens for damaged mpeg streams) */
+ memset(inbuf + INBUF_SIZE, 0, FF_INPUT_BUFFER_PADDING_SIZE);
+
+ printf("Video decoding\n");
+
+ /* find the mpeg1 video decoder */
+ codec = avcodec_find_decoder(CODEC_ID_MPEG1VIDEO);
+ if (!codec) {
+ fprintf(stderr, "codec not found\n");
+ exit(1);
+ }
+
+ c = avcodec_alloc_context3(codec);
+ picture= avcodec_alloc_frame();
+
+ if(codec->capabilities&CODEC_CAP_TRUNCATED)
+ c->flags|= CODEC_FLAG_TRUNCATED; /* we do not send complete frames */
+
+ /* For some codecs, such as msmpeg4 and mpeg4, width and height
+ MUST be initialized there because this information is not
+ available in the bitstream. */
+
+ /* open it */
+ if (avcodec_open(c, codec) < 0) {
+ fprintf(stderr, "could not open codec\n");
+ exit(1);
+ }
+
+ /* the codec gives us the frame size, in samples */
+
+ f = fopen(filename, "rb");
+ if (!f) {
+ fprintf(stderr, "could not open %s\n", filename);
+ exit(1);
+ }
+
+ frame = 0;
+ for(;;) {
+ avpkt.size = fread(inbuf, 1, INBUF_SIZE, f);
+ if (avpkt.size == 0)
+ break;
+
+ /* NOTE1: some codecs are stream based (mpegvideo, mpegaudio)
+ and this is the only method to use them because you cannot
+ know the compressed data size before analysing it.
+
+ BUT some other codecs (msmpeg4, mpeg4) are inherently frame
+ based, so you must call them with all the data for one
+ frame exactly. You must also initialize 'width' and
+ 'height' before initializing them. */
+
+ /* NOTE2: some codecs allow the raw parameters (frame size,
+ sample rate) to be changed at any frame. We handle this, so
+ you should also take care of it */
+
+ /* here, we use a stream based decoder (mpeg1video), so we
+ feed decoder and see if it could decode a frame */
+ avpkt.data = inbuf;
+ while (avpkt.size > 0) {
+ len = avcodec_decode_video2(c, picture, &got_picture, &avpkt);
+ if (len < 0) {
+ fprintf(stderr, "Error while decoding frame %d\n", frame);
+ exit(1);
+ }
+ if (got_picture) {
+ printf("saving frame %3d\n", frame);
+ fflush(stdout);
+
+ /* the picture is allocated by the decoder. no need to
+ free it */
+ snprintf(buf, sizeof(buf), outfilename, frame);
+ pgm_save(picture->data[0], picture->linesize[0],
+ c->width, c->height, buf);
+ frame++;
+ }
+ avpkt.size -= len;
+ avpkt.data += len;
+ }
+ }
+
+ /* some codecs, such as MPEG, transmit the I and P frame with a
+ latency of one frame. You must do the following to have a
+ chance to get the last frame of the video */
+ avpkt.data = NULL;
+ avpkt.size = 0;
+ len = avcodec_decode_video2(c, picture, &got_picture, &avpkt);
+ if (got_picture) {
+ printf("saving last frame %3d\n", frame);
+ fflush(stdout);
+
+ /* the picture is allocated by the decoder. no need to
+ free it */
+ snprintf(buf, sizeof(buf), outfilename, frame);
+ pgm_save(picture->data[0], picture->linesize[0],
+ c->width, c->height, buf);
+ frame++;
+ }
+
+ fclose(f);
+
+ avcodec_close(c);
+ av_free(c);
+ av_free(picture);
+ printf("\n");
+}
+
+int main(int argc, char **argv)
+{
+ const char *filename;
+
+ /* must be called before using avcodec lib */
+ avcodec_init();
+
+ /* register all the codecs */
+ avcodec_register_all();
+
+ if (argc <= 1) {
+ audio_encode_example("/tmp/test.mp2");
+ audio_decode_example("/tmp/test.sw", "/tmp/test.mp2");
+
+ video_encode_example("/tmp/test.mpg");
+ filename = "/tmp/test.mpg";
+ } else {
+ filename = argv[1];
+ }
+
+ // audio_decode_example("/tmp/test.sw", filename);
+ video_decode_example("/tmp/test%d.pgm", filename);
+
+ return 0;
+}
diff --git a/src/test-native/ffmpeg/avcodec_sample.c b/src/test-native/ffmpeg/avcodec_sample.c
new file mode 100644
index 000000000..f4001b4e6
--- /dev/null
+++ b/src/test-native/ffmpeg/avcodec_sample.c
@@ -0,0 +1,203 @@
+// avcodec_sample.0.5.0.c
+
+// A small sample program that shows how to use libavformat and libavcodec to
+// read video from a file.
+//
+// This version is for the 0.4.9+ release of ffmpeg. This release adds the
+// av_read_frame() API call, which simplifies the reading of video frames
+// considerably.
+//
+// Use
+//
+// gcc -o avcodec_sample.0.5.0 avcodec_sample.0.5.0.c -lavformat -lavcodec -lavutil -lswscale -lz -lbz2
+//
+// to build (assuming libavformat, libavcodec, libavutil, and swscale are correctly installed on
+// your system).
+//
+// Run using
+//
+// avcodec_sample.0.5.0 myvideofile.mpg
+//
+// to write the first five frames from "myvideofile.mpg" to disk in PPM
+// format.
+
+#include <libavcodec/avcodec.h>
+#include <libavformat/avformat.h>
+#include <libswscale/swscale.h>
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <stdbool.h>
+
+static void SaveFrame(AVFrame *pFrame, int width, int height, int iFrame);
+
+int main (int argc, const char * argv[])
+{
+ AVFormatContext *pFormatCtx;
+ int i, videoStream;
+ AVCodecContext *pCodecCtx;
+ AVCodec *pCodec;
+ AVFrame *pFrame;
+ AVFrame *pFrameRGB;
+ AVPacket packet;
+ int frameFinished;
+ int numBytes;
+ uint8_t *buffer;
+
+ // Register all formats and codecs
+ av_register_all();
+
+ avformat_network_init();
+
+ // Open video file
+ if(avformat_open_input(&pFormatCtx, argv[1], NULL, NULL)!=0)
+ return -1; // Couldn't open file
+
+ // Retrieve stream information
+ if(avformat_find_stream_info(pFormatCtx, NULL)<0)
+ return -1; // Couldn't find stream information
+
+ // Dump information about file onto standard error
+ av_dump_format(pFormatCtx, 0, argv[1], false);
+
+ // Find the first video stream
+ videoStream=-1;
+ for(i=0; i<pFormatCtx->nb_streams; i++)
+ if(pFormatCtx->streams[i]->codec->codec_type==AVMEDIA_TYPE_VIDEO)
+ {
+ videoStream=i;
+ break;
+ }
+ if(videoStream==-1)
+ return -1; // Didn't find a video stream
+
+ // Get a pointer to the codec context for the video stream
+ pCodecCtx=pFormatCtx->streams[videoStream]->codec;
+
+ // Find the decoder for the video stream
+ pCodec=avcodec_find_decoder(pCodecCtx->codec_id);
+ if(pCodec==NULL)
+ return -1; // Codec not found
+
+ // Open codec
+ if(avcodec_open2(pCodecCtx, pCodec, NULL)<0)
+ return -1; // Could not open codec
+
+ // Hack to correct wrong frame rates that seem to be generated by some codecs
+ if(pCodecCtx->time_base.num>1000 && pCodecCtx->time_base.den==1)
+ pCodecCtx->time_base.den=1000;
+
+ // Allocate video frame
+ pFrame=avcodec_alloc_frame();
+
+ // Allocate an AVFrame structure
+ pFrameRGB=avcodec_alloc_frame();
+ if(pFrameRGB==NULL)
+ return -1;
+
+ // Determine required buffer size and allocate buffer
+ numBytes=avpicture_get_size(PIX_FMT_RGB24, pCodecCtx->width,
+ pCodecCtx->height);
+
+ buffer=malloc(numBytes);
+
+ // Assign appropriate parts of buffer to image planes in pFrameRGB
+ avpicture_fill((AVPicture *)pFrameRGB, buffer, PIX_FMT_RGB24,
+ pCodecCtx->width, pCodecCtx->height);
+
+ // Read frames and save first five frames to disk
+ i=0;
+ while(av_read_frame(pFormatCtx, &packet)>=0)
+ {
+ // Is this a packet from the video stream?
+ if(packet.stream_index==videoStream)
+ {
+ // Decode video frame
+ avcodec_decode_video2(pCodecCtx, pFrame, &frameFinished, &packet);
+
+ // Did we get a video frame?
+ if(frameFinished)
+ {
+ static struct SwsContext *img_convert_ctx;
+
+#if 0
+ // Older removed code
+ // Convert the image from its native format to RGB swscale
+ img_convert((AVPicture *)pFrameRGB, PIX_FMT_RGB24,
+ (AVPicture*)pFrame, pCodecCtx->pix_fmt, pCodecCtx->width,
+ pCodecCtx->height);
+
+ // function template, for reference
+ int sws_scale(struct SwsContext *context, uint8_t* src[], int srcStride[], int srcSliceY,
+ int srcSliceH, uint8_t* dst[], int dstStride[]);
+#endif
+ // Convert the image into YUV format that SDL uses
+ if(img_convert_ctx == NULL) {
+ int w = pCodecCtx->width;
+ int h = pCodecCtx->height;
+
+ img_convert_ctx = sws_getContext(w, h,
+ pCodecCtx->pix_fmt,
+ w, h, PIX_FMT_RGB24, SWS_BICUBIC,
+ NULL, NULL, NULL);
+ if(img_convert_ctx == NULL) {
+ fprintf(stderr, "Cannot initialize the conversion context!\n");
+ exit(1);
+ }
+ }
+ int ret = sws_scale(img_convert_ctx, pFrame->data, pFrame->linesize, 0,
+ pCodecCtx->height, pFrameRGB->data, pFrameRGB->linesize);
+#if 0 // this use to be true, as of 1/2009, but apparently it is no longer true in 3/2009
+ if(ret) {
+ fprintf(stderr, "SWS_Scale failed [%d]!\n", ret);
+ exit(-1);
+ }
+#endif
+ // Save the frame to disk
+ if(i++<=5)
+ SaveFrame(pFrameRGB, pCodecCtx->width, pCodecCtx->height, i);
+ }
+ }
+
+ // Free the packet that was allocated by av_read_frame
+ av_free_packet(&packet);
+ }
+
+ // Free the RGB image
+ free(buffer);
+ av_free(pFrameRGB);
+
+ // Free the YUV frame
+ av_free(pFrame);
+
+ // Close the codec
+ avcodec_close(pCodecCtx);
+
+ // Close the video file
+ avformat_close_input(&pFormatCtx);
+
+ return 0;
+}
+
+static void SaveFrame(AVFrame *pFrame, int width, int height, int iFrame)
+{
+ FILE *pFile;
+ char szFilename[32];
+ int y;
+
+ // Open file
+ sprintf(szFilename, "frame%d.ppm", iFrame);
+ pFile=fopen(szFilename, "wb");
+ if(pFile==NULL)
+ return;
+
+ // Write header
+ fprintf(pFile, "P6\n%d %d\n255\n", width, height);
+
+ // Write pixel data
+ for(y=0; y<height; y++)
+ fwrite(pFrame->data[0]+y*pFrame->linesize[0], 1, width*3, pFile);
+
+ // Close file
+ fclose(pFile);
+}
diff --git a/src/test-native/ffmpeg/avcodec_sample.sh b/src/test-native/ffmpeg/avcodec_sample.sh
new file mode 100644
index 000000000..c5ccd6caf
--- /dev/null
+++ b/src/test-native/ffmpeg/avcodec_sample.sh
@@ -0,0 +1,6 @@
+# -I/usr/include/libavcodec -I/usr/include/libavformat \
+
+gcc \
+ -o avcodec_sample avcodec_sample.c \
+ -lavformat -lavcodec -lavutil -lswscale
+
diff --git a/src/test-native/gst/helloworld-auto.c b/src/test-native/gst/helloworld-auto.c
new file mode 100644
index 000000000..6381c9c93
--- /dev/null
+++ b/src/test-native/gst/helloworld-auto.c
@@ -0,0 +1,112 @@
+#include <gst/gst.h>
+#include <glib.h>
+
+
+static gboolean
+my_bus_callback (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;
+}
+
+
+static gboolean
+idle_exit_loop (gpointer data)
+{
+ g_main_loop_quit ((GMainLoop *) data);
+
+ /* once */
+ return FALSE;
+}
+
+static void
+cb_typefound (GstElement *typefind,
+ guint probability,
+ GstCaps *caps,
+ gpointer data)
+{
+ GMainLoop *loop = data;
+ gchar *type;
+
+ type = gst_caps_to_string (caps);
+ g_print ("Media type %s found, probability %d%%\n", type, probability);
+ g_free (type);
+
+ /* since we connect to a signal in the pipeline thread context, we need
+ * to set an idle handler to exit the main loop in the mainloop context.
+ * Normally, your app should not need to worry about such things. */
+ g_idle_add (idle_exit_loop, loop);
+}
+
+gint
+main (gint argc,
+ gchar *argv[])
+{
+ GMainLoop *loop;
+ GstElement *pipeline, *filesrc, *typefind, *fakesink;
+ GstBus *bus;
+
+ /* init GStreamer */
+ gst_init (&argc, &argv);
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* check args */
+ if (argc != 2) {
+ g_print ("Usage: %s <filename>\n", argv[0]);
+ return -1;
+ }
+
+ /* create a new pipeline to hold the elements */
+ pipeline = gst_pipeline_new ("pipe");
+
+ bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
+ gst_bus_add_watch (bus, my_bus_callback, NULL);
+ gst_object_unref (bus);
+
+ /* create file source and typefind element */
+ filesrc = gst_element_factory_make ("filesrc", "source");
+ g_object_set (G_OBJECT (filesrc), "location", argv[1], NULL);
+ typefind = gst_element_factory_make ("typefind", "typefinder");
+ g_signal_connect (typefind, "have-type", G_CALLBACK (cb_typefound), loop);
+ fakesink = gst_element_factory_make ("fakesink", "sink");
+
+ /* setup */
+ gst_bin_add_many (GST_BIN (pipeline), filesrc, typefind, fakesink, NULL);
+ gst_element_link_many (filesrc, typefind, fakesink, NULL);
+ gst_element_set_state (GST_ELEMENT (pipeline), GST_STATE_PLAYING);
+ g_main_loop_run (loop);
+
+ /* unset */
+ gst_element_set_state (GST_ELEMENT (pipeline), GST_STATE_NULL);
+ gst_object_unref (GST_OBJECT (pipeline));
+
+ return 0;
+}
+
diff --git a/src/test-native/gst/helloworld-playbin.c b/src/test-native/gst/helloworld-playbin.c
new file mode 100644
index 000000000..c5a42fe6c
--- /dev/null
+++ b/src/test-native/gst/helloworld-playbin.c
@@ -0,0 +1,75 @@
+#include <gst/gst.h>
+#include <glib.h>
+
+
+static gboolean
+my_bus_callback (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;
+}
+
+gint
+main (gint argc,
+ gchar *argv[])
+{
+ GMainLoop *loop;
+ GstElement *play;
+ GstBus *bus;
+
+ /* init GStreamer */
+ gst_init (&argc, &argv);
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* make sure we have a URI */
+ if (argc != 2) {
+ g_print ("Usage: %s <URI>\n", argv[0]);
+ return -1;
+ }
+
+ /* set up */
+ play = gst_element_factory_make ("playbin", "play");
+ g_object_set (G_OBJECT (play), "uri", argv[1], NULL);
+
+ bus = gst_pipeline_get_bus (GST_PIPELINE (play));
+ gst_bus_add_watch (bus, my_bus_callback, loop);
+ gst_object_unref (bus);
+
+ gst_element_set_state (play, GST_STATE_PLAYING);
+
+ /* now run */
+ g_main_loop_run (loop);
+
+ /* also clean up */
+ gst_element_set_state (play, GST_STATE_NULL);
+ gst_object_unref (GST_OBJECT (play));
+
+ return 0;
+}
diff --git a/src/test-native/gst/helloworld-playbin2.c b/src/test-native/gst/helloworld-playbin2.c
new file mode 100644
index 000000000..b31e3e5c6
--- /dev/null
+++ b/src/test-native/gst/helloworld-playbin2.c
@@ -0,0 +1,75 @@
+#include <gst/gst.h>
+#include <glib.h>
+
+
+static gboolean
+my_bus_callback (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;
+}
+
+gint
+main (gint argc,
+ gchar *argv[])
+{
+ GMainLoop *loop;
+ GstElement *play;
+ GstBus *bus;
+
+ /* init GStreamer */
+ gst_init (&argc, &argv);
+ loop = g_main_loop_new (NULL, FALSE);
+
+ /* make sure we have a URI */
+ if (argc != 2) {
+ g_print ("Usage: %s <URI>\n", argv[0]);
+ return -1;
+ }
+
+ /* set up */
+ play = gst_element_factory_make ("playbin2", "play");
+ g_object_set (G_OBJECT (play), "uri", argv[1], NULL);
+
+ bus = gst_pipeline_get_bus (GST_PIPELINE (play));
+ gst_bus_add_watch (bus, my_bus_callback, loop);
+ gst_object_unref (bus);
+
+ gst_element_set_state (play, GST_STATE_PLAYING);
+
+ /* now run */
+ g_main_loop_run (loop);
+
+ /* also clean up */
+ gst_element_set_state (play, GST_STATE_NULL);
+ gst_object_unref (GST_OBJECT (play));
+
+ return 0;
+}
diff --git a/src/test-native/gst/helloworld.c b/src/test-native/gst/helloworld.c
new file mode 100644
index 000000000..6d991898d
--- /dev/null
+++ b/src/test-native/gst/helloworld.c
@@ -0,0 +1,142 @@
+#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;
+}
+
+
+static void
+on_pad_added (GstElement *element,
+ GstPad *pad,
+ gpointer data)
+{
+ GstPad *sinkpad;
+ GstElement *decoder = (GstElement *) data;
+
+ /* We can now link this pad with the vorbis-decoder sink pad */
+ g_print ("Dynamic pad created, linking demuxer/decoder\n");
+
+ sinkpad = gst_element_get_static_pad (decoder, "sink");
+
+ gst_pad_link (pad, sinkpad);
+
+ gst_object_unref (sinkpad);
+}
+
+
+
+int
+main (int argc,
+ char *argv[])
+{
+ GMainLoop *loop;
+
+ GstElement *pipeline, *source, *demuxer, *decoder, *conv, *sink;
+ GstBus *bus;
+
+ /* Initialisation */
+ gst_init (&argc, &argv);
+
+ loop = g_main_loop_new (NULL, FALSE);
+
+
+ /* Check input arguments */
+ if (argc != 2) {
+ g_printerr ("Usage: %s <Ogg/Vorbis filename>\n", argv[0]);
+ return -1;
+ }
+
+
+ /* Create gstreamer elements */
+ pipeline = gst_pipeline_new ("audio-player");
+ source = gst_element_factory_make ("filesrc", "file-source");
+ demuxer = gst_element_factory_make ("oggdemux", "ogg-demuxer");
+ decoder = gst_element_factory_make ("vorbisdec", "vorbis-decoder");
+ conv = gst_element_factory_make ("audioconvert", "converter");
+ sink = gst_element_factory_make ("autoaudiosink", "audio-output");
+
+ if (!pipeline || !source || !demuxer || !decoder || !conv || !sink) {
+ g_printerr ("One element could not be created. Exiting.\n");
+ return -1;
+ }
+
+ /* Set up the pipeline */
+
+ /* we set the input filename to the source element */
+ g_object_set (G_OBJECT (source), "location", argv[1], NULL);
+
+ /* we add a message handler */
+ bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
+ gst_bus_add_watch (bus, bus_call, loop);
+ gst_object_unref (bus);
+
+ /* we add all elements into the pipeline */
+ /* file-source | ogg-demuxer | vorbis-decoder | converter | alsa-output */
+ gst_bin_add_many (GST_BIN (pipeline),
+ source, demuxer, decoder, conv, sink, NULL);
+
+ /* we link the elements together */
+ /* file-source -> ogg-demuxer ~> vorbis-decoder -> converter -> alsa-output */
+ gst_element_link (source, demuxer);
+ gst_element_link_many (decoder, conv, sink, NULL);
+ g_signal_connect (demuxer, "pad-added", G_CALLBACK (on_pad_added), decoder);
+
+ /* note that the demuxer will be linked to the decoder dynamically.
+ The reason is that Ogg may contain various streams (for example
+ audio and video). The source pad(s) will be created at run time,
+ by the demuxer when it detects the amount and nature of streams.
+ Therefore we connect a callback function which will be executed
+ when the "pad-added" is emitted.*/
+
+
+ /* Set the pipeline to "playing" state*/
+ g_print ("Now playing: %s\n", argv[1]);
+ gst_element_set_state (pipeline, GST_STATE_PLAYING);
+
+
+ /* Iterate */
+ g_print ("Running...\n");
+ g_main_loop_run (loop);
+
+
+ /* Out of the main loop, clean up nicely */
+ g_print ("Returned, stopping playback\n");
+ gst_element_set_state (pipeline, GST_STATE_NULL);
+
+ g_print ("Deleting pipeline\n");
+ gst_object_unref (GST_OBJECT (pipeline));
+
+ return 0;
+}
diff --git a/src/test-native/gst/make.sh b/src/test-native/gst/make.sh
new file mode 100644
index 000000000..23e54a272
--- /dev/null
+++ b/src/test-native/gst/make.sh
@@ -0,0 +1,5 @@
+gcc -Wall helloworld.c -o helloworld $(pkg-config --cflags --libs gstreamer-0.10)
+gcc -Wall helloworld-auto.c -o helloworld-auto $(pkg-config --cflags --libs gstreamer-0.10)
+gcc -Wall helloworld-playbin.c -o helloworld-playbin $(pkg-config --cflags --libs gstreamer-0.10)
+gcc -Wall helloworld-playbin2.c -o helloworld-playbin2 $(pkg-config --cflags --libs gstreamer-0.10)
+
generated by cgit v1.2.3 (git 2.39.1) at 2024-12-14 20:45:47 +0000