From e2de64d6f1beb9e492daf5b886e19933c1fa41dd Mon Sep 17 00:00:00 2001
From: toma <toma@283d02a7-25f6-0310-bc7c-ecb5cbfe19da>
Date: Wed, 25 Nov 2009 17:56:58 +0000
Subject: Copy the KDE 3.5 branch to branches/trinity for new KDE 3.5 features.
 BUG:215923

git-svn-id: svn://anonsvn.kde.org/home/kde/branches/trinity/kdemultimedia@1054174 283d02a7-25f6-0310-bc7c-ecb5cbfe19da
---
 mpeglib/example/splay/Makefile.am    |  41 ++++++
 mpeglib/example/splay/mp3framing.cpp | 274 +++++++++++++++++++++++++++++++++++
 mpeglib/example/splay/splay.cpp      |  64 ++++++++
 3 files changed, 379 insertions(+)
 create mode 100644 mpeglib/example/splay/Makefile.am
 create mode 100644 mpeglib/example/splay/mp3framing.cpp
 create mode 100644 mpeglib/example/splay/splay.cpp

(limited to 'mpeglib/example/splay')

diff --git a/mpeglib/example/splay/Makefile.am b/mpeglib/example/splay/Makefile.am
new file mode 100644
index 00000000..76e04cff
--- /dev/null
+++ b/mpeglib/example/splay/Makefile.am
@@ -0,0 +1,41 @@
+#  splay-yaf - Makefile.am
+
+INCLUDES		=	-I../../include $(all_includes)
+
+EXTRA_DIST		=	
+
+noinst_PROGRAMS		=	splay mp3framing
+
+splay_SOURCES		=	splay.cpp
+
+mp3framing_SOURCES      =       mp3framing.cpp
+
+noinst_HEADERS		=	
+
+splay_LDFLAGS		=	$(all_libraries) 
+
+splay_LDADD		=	../../lib/libmpeg.la \
+				$(THIS_LIB_LIBS)
+
+mp3framing_LDFLAGS      =       $(all_libraries)
+mp3framing_LDADD        =       ../../lib/libmpeg.la \
+				$(THIS_LIB_LIBS)
+
+
+
+
+
+
+
+
+
+				
+
+
+
+
+
+
+
+
+
diff --git a/mpeglib/example/splay/mp3framing.cpp b/mpeglib/example/splay/mp3framing.cpp
new file mode 100644
index 00000000..586899a3
--- /dev/null
+++ b/mpeglib/example/splay/mp3framing.cpp
@@ -0,0 +1,274 @@
+/*
+
+  Example how frame creation works.
+
+  For the input we have std. file input.
+  We convert it to mpeg I encoded frames (mpegAudioFrame's job)
+
+  The decoded pcmFrames are stores in a queue.
+  
+  If the queue if full, we "stream" the packets to /dev/dsp.
+  (The difference between directly playing the packets and "streaming"
+  is, that while streaming you can say: I want x bytes I don't care
+  about frame boundaries)
+  This is audioFrameQueue's job.
+
+
+ */
+
+
+#include "../../lib/splay/mpegAudioFrame.h"
+#include "../../lib/splay/splayDecoder.h"
+#include "../../lib/util/audio/dspWrapper.h"
+#include "../../lib/frame/audioFrameQueue.h"
+
+#include <iostream>
+
+using namespace std;
+
+// used for getopt
+#include <unistd.h>
+
+#define INPUT_SIZE  8192
+#define _QUEUE_FLOAT 1
+#define _QUEUE_INT   2
+
+
+void usage() {
+  printf("Usage:\n\n");
+  printf("mp3framing [hf] filename\n\n");
+  printf("             -f use float queue\n");
+  printf("             -c [samples] checkout samples\n");
+}
+
+
+void initDSP(DSPWrapper* dsp) {
+  if (dsp->openDevice()==false) {
+    cout << "cannot open /dev/dsp"<<endl;
+    exit(-1);
+  }
+}
+  
+
+
+int main(int argc, char** argv) {
+  int queueType=_QUEUE_INT;
+  int samples=8192;
+
+  PCMFrame* outFrame;
+
+  while(1) { 
+    int c = getopt (argc, argv, "hfc:");
+    if (c == -1) break;
+    switch(c) {
+    case 'h': {    
+      usage();
+      exit(-1);
+      break;
+    }
+    case 'f': {    
+      queueType=_QUEUE_FLOAT;
+      break;
+    }
+    case 'c': {    
+      samples=atoi(optarg);
+      cout << "checking out samples:samples"<<endl;
+      break;
+    }
+    default:
+      printf ("?? getopt returned character code 0%o ??\n", c);
+      exit(-1);
+    }
+  }
+  // if we have nothing left, there is no filename
+  if (optind >= argc ) {
+    usage();
+    exit(-1);
+  }
+  FILE* file=fopen(argv[optind],"r");
+  if (file == NULL) {
+    cout << "cannot open:"<<argv[1]<<endl;
+    exit(-1);
+  }
+  DSPWrapper* dsp=new DSPWrapper();
+  outFrame=new PCMFrame(samples);
+
+  float* left=new float[samples];
+  float* right=new float[samples];
+
+
+  // create queue with 500 frames. We want output to dsp(later)
+  // so we need a _FRAME_AUDIO_PCM type and we need
+  // for splay at least MP3FRAMESIZE
+  AudioFrameQueue* frameQueue;
+  if (queueType == _QUEUE_INT) {
+    frameQueue=new AudioFrameQueue(500,MP3FRAMESIZE,_FRAME_AUDIO_PCM);
+  }
+  if (queueType == _QUEUE_FLOAT) {
+    frameQueue=new AudioFrameQueue(100,MP3FRAMESIZE,_FRAME_AUDIO_FLOAT);
+  }
+   
+
+  initDSP(dsp);
+  SplayDecoder* splay=new SplayDecoder();
+
+
+  //
+  // MpegAudioFrame has the ability to convert "raw" file
+  // data into mpeg I frames. These are "valid" frames,
+  // which means: header is ok, and the packet has the
+  // correct length. splay decodes these frames _at once_
+  // so we need no callbacks for the decoder.
+  MpegAudioFrame* frame=new MpegAudioFrame();
+
+
+  // Note: The inputbuffer must NOT NOT NOT be on the stack!
+  // correction: it should not be corrupted by allocation
+  // on the stack. This is true, if allocation+decoding
+  // or in the _same_ function.
+  unsigned char inputbuffer[INPUT_SIZE];
+
+
+  int cnt=0;
+  // we proceed until eof, or our "queue" is full.
+  while( (feof(file) == false) && (frameQueue->dataQueueCanWrite() == true) ) {
+
+    //
+    // This switch/case statement "partitions" the fileinput
+    // into mp3frames. We directly decode them to pcmFrames.
+    //
+
+    //
+    // There are three states. One is "I want Input, give me input"
+    // The other is "I have enough input, dont' bother me and let
+    // me do my work"
+    // The last ist "I have a frame here, take care of this, or
+    // I will continue with my work"
+ 
+    int state=frame->getState();
+    switch(state) {
+    case FRAME_NEED: {
+      int bytes=frame->canStore();
+      int read=fread(inputbuffer,1,bytes,file);
+      if (read != bytes) {
+	// read error. reset framer
+	frame->reset();
+	continue;
+      }
+      frame->store(inputbuffer,bytes);
+      break;
+    }
+    case FRAME_WORK:
+      frame->work();
+      break;
+    case FRAME_HAS:{
+      AudioFrame* emptyFrame= frameQueue->emptyQueueDequeue();
+      splay->decode(frame->outdata(),frame->len(),emptyFrame);
+      frameQueue->dataQueueEnqueue(emptyFrame);
+      cout << "storing decodec frame:"<<cnt<<endl;
+      cnt++;
+      break;
+    }
+    default:
+      cout << "unknown state in mpeg audio framing"<<endl;
+      exit(0);
+    }
+  }
+
+
+  //
+  // Fine we put all decoded frames in the buffer.
+  // Now we want to "stream" from this buffer.
+  // 
+
+  //
+  // create our "local" stream
+  short int playbuf[INPUT_SIZE];
+   
+  //
+  // we need to setup the dsp manually, when we do not play pcmFrames directly
+  //
+  if (queueType == _QUEUE_INT) {
+    AudioFrame* audioFrame=frameQueue->getCurrent();
+    dsp->audioSetup(audioFrame);
+  }
+  
+  if (queueType == _QUEUE_FLOAT) {
+    AudioFrame* audioFrame=frameQueue->getCurrent();
+    dsp->audioSetup(audioFrame->getStereo(),16,
+		    audioFrame->getSigned(),
+		    audioFrame->getBigEndian(),
+		    audioFrame->getFrequenceHZ());
+    outFrame->setFrameFormat(audioFrame->getStereo(),
+			     audioFrame->getFrequenceHZ());
+  }
+  
+
+
+  //
+  // read from stream demo:
+  //
+  
+  while(frameQueue->getLen() > 0) {
+    int hasLen= frameQueue->getLen();
+
+    cout << "hasLen:"<<hasLen<<endl;
+    int hasRead;
+    if (hasLen < samples) {
+      samples=hasLen;
+    }
+
+    
+
+    if (queueType == _QUEUE_INT) {
+      cout << "reading :"<<samples<<" data"<<endl;
+      hasRead=frameQueue->copy(outFrame->getData(),samples);
+      dsp->audioPlay((char*)outFrame->getData(),hasRead*sizeof(short int));
+      frameQueue->forwardStreamSingle(hasRead);
+    }
+
+
+    if (queueType == _QUEUE_FLOAT) {
+      int n;
+
+      hasRead=frameQueue->copy(left,right,samples);
+      frameQueue->forwardStreamDouble(hasRead);
+      n=hasRead;
+
+      /*
+      FloatFrame* floatFrame=( FloatFrame*) frameQueue->dataQueueDequeue();
+      float* left=floatFrame->getData();
+      n=floatFrame->getLen();
+      */
+
+      int i=0;
+      while(i<n) {
+	int val=(int)(32768.0*left[i]);
+	if (val > 32767) val=32767;
+	if (val < -32768) val=-32768;
+	dsp->audioPlay((char*)&val,2);
+	
+	if (outFrame->getStereo()) {
+	  val=(int)(32768.0*right[i]);
+	  if (val > 32767) val=32767;
+	  if (val < -32768) val=-32768;
+	  dsp->audioPlay((char*)&val,2);
+	}
+
+	i++;
+      }
+
+    }
+  }
+
+
+  delete frameQueue;
+  delete splay;
+  delete frame;
+  delete dsp;
+  delete left;
+  delete right;
+  delete outFrame;
+  return(0);  
+}
+
diff --git a/mpeglib/example/splay/splay.cpp b/mpeglib/example/splay/splay.cpp
new file mode 100644
index 00000000..75254360
--- /dev/null
+++ b/mpeglib/example/splay/splay.cpp
@@ -0,0 +1,64 @@
+/*
+
+  Example how the plugin interface works.(LINUX Open Sound System only!)
+
+
+ */
+
+
+#include "../../lib/decoder/splayPlugin.h"
+#include "../../lib/decoder/tplayPlugin.h"
+#include "../../lib/decoder/vorbisPlugin.h"
+#include "../../lib/input/bufferInputStream.h"
+#include "../../lib/output/avSyncer.h"
+
+#include <iostream>
+
+using namespace std;
+
+int main(int argc, char** argv) {
+
+
+
+  if (argc <= 1) {
+    printf("Usage:\n\n");
+    printf("%s filename\n\n",argv[0]);
+    exit(0);
+  }
+
+  //
+  // The order is important !!!!
+  // 1. construct
+  // 2. set Output
+  // 3. open input
+  // 4. set input
+  // 
+  // you cannot set the input _before_ the output 
+  // in fact you can, but this gives you a segfault!
+
+  SplayPlugin* plugin=new SplayPlugin();
+  OutputStream* out=OutPlugin::createOutputStream(_OUTPUT_LOCAL,true);
+  InputStream* in=InputPlugin::createInputStream(argv[1],true);
+
+  in->open(argv[1]);
+  // watch the order!
+  plugin->setOutputPlugin(out);
+  plugin->setInputPlugin(in);
+  plugin->play();
+
+  while(plugin->getStreamState() != _STREAM_STATE_EOF) {
+    cout << "check state"<<endl;
+    TimeWrapper::sleep(1);
+
+  }
+
+  cout << "****************plugin eof"<<endl;
+  //out->audioClose();
+  plugin->close();
+
+  delete plugin;
+  delete in;
+  delete out;
+  return(0);  
+}
+  
-- 
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