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_artsplug/splayPlayObject_impl.cpp | 500 ++++++++++++++++++++++++++++++
 1 file changed, 500 insertions(+)
 create mode 100644 mpeglib_artsplug/splayPlayObject_impl.cpp

(limited to 'mpeglib_artsplug/splayPlayObject_impl.cpp')

diff --git a/mpeglib_artsplug/splayPlayObject_impl.cpp b/mpeglib_artsplug/splayPlayObject_impl.cpp
new file mode 100644
index 00000000..d29aa61a
--- /dev/null
+++ b/mpeglib_artsplug/splayPlayObject_impl.cpp
@@ -0,0 +1,500 @@
+/*
+  base class for splay mp3 decoder
+  Copyright (C) 2000  Martin Vogt
+
+  This program is free software; you can redistribute it and/or modify
+  it under the terms of the GNU Library General Public License as published by
+  the Free Software Foundation.
+
+  For more information look at the file COPYRIGHT in this package
+
+ */
+
+#include "splayPlayObject_impl.h"
+#include "../mpeglib/lib/splay/splayDecoder.h"
+#include "../mpeglib/lib/frame/audioFrameQueue.h"
+#include "../mpeglib/lib/splay/mpegAudioFrame.h"
+
+#include "debug.h"
+
+#define INPUT_SIZE  8192
+
+/**
+   Problems with streaming, which must be solved:
+   ==============================================
+
+
+   Deadlocks. I think we cannot have sync streams
+   for input/output.
+
+   This example is an OLD style PlayObject. (it uses sync streams)
+   The task is to convert this into an async PlayObject.
+
+   What does this PlayObject do:
+   -----------------------------
+
+   Current:
+   It reads data from a file and insert it
+   into a queue.
+
+   Future:
+   A "remote" sender of raw data, can test how much
+   bytes he can insert into the "framer".
+   If the framer produces a frame, we decode it,
+   and send an async dataPacket out.
+   (Then the frameQueue is not necessary anymore -IMHO)
+
+*/
+
+
+SplayPlayObject_impl::SplayPlayObject_impl() {
+
+  flpos=0.0;
+  
+  splay=new SplayDecoder();
+  frameQueue= new AudioFrameQueue(10,MP3FRAMESIZE,_FRAME_AUDIO_FLOAT);
+  framer=new MpegAudioFrame();
+
+  arts_debug("outputStream created");
+
+  _state=Arts::posIdle;
+  file=NULL;
+  lStreaming=false;
+  
+  resampleBuffer=NULL;
+  resampleBufferSize=0;
+  currentPacket=NULL;
+  currentPos=0;
+  // ok, we store packets here. Someday I should make this a template.
+  // use STL? maybe.
+  packetQueue=new FrameQueue(10);
+    
+  // Note: The inputbuffer must NOT NOT NOT be on the stack!
+  inputbuffer=new unsigned char[INPUT_SIZE];
+
+}
+
+SplayPlayObject_impl::~SplayPlayObject_impl() {
+  arts_debug("~SplayPlayObject_impl -s");
+  delete splay;
+  delete frameQueue;
+  delete framer; 
+  arts_debug("~SplayPlayObject_impl -e");
+
+  if (resampleBuffer != NULL) {
+    delete resampleBuffer;
+  }
+  //
+  // arts wants to free the packets and packetQueue too.
+  // so we dequeue all packets here, before deleting packetQueue
+  // otherwise we call delete on the packets twice.
+  //
+  while (packetQueue->getFillgrade() > 0) packetQueue->dequeue();
+  // now delete the (empty) queue
+
+  delete packetQueue;
+  delete [] inputbuffer;
+}
+
+
+bool SplayPlayObject_impl::loadMedia(const string &filename) {
+  arts_debug("loadMedia");
+
+  if (file != NULL) {
+    arts_fatal("~SplayPlayObject_impl already loaded");
+  }
+
+  lStreaming=false;
+
+  file=fopen(filename.c_str(),"r");
+  if (file == NULL) {
+    arts_debug("splay cannot open file");
+    return false;
+  }
+  
+  flpos=0.0;
+
+  return true;
+}
+
+bool SplayPlayObject_impl::streamMedia(Arts::InputStream instream) {
+  arts_debug("streamMedia");
+  lStreaming=true;
+  currentStream = instream;
+  Arts::StreamPlayObject self = Arts::StreamPlayObject::_from_base(_copy());
+  connect(currentStream, "outdata", self);
+  return true;
+}
+
+void SplayPlayObject_impl::process_indata(Arts::DataPacket<Arts::mcopbyte> *inpacket) {
+  arts_debug("receiving packet");
+
+
+  packetQueue->enqueue((Frame*)inpacket);
+  if (packetQueue->getFillgrade() == 1) {
+    currentPos=0;
+  }
+
+  processQueue();
+}
+
+
+void SplayPlayObject_impl::processQueue() {
+
+  
+
+  unsigned char* ptr;
+
+  if (packetQueue->getFillgrade() == 0) {
+    return;
+  }
+  Arts::DataPacket<Arts::mcopbyte>* currentPacket;
+
+  currentPacket=(Arts::DataPacket<Arts::mcopbyte>*) packetQueue->peekqueue(0);
+  int rest=currentPacket->size-currentPos;
+
+  int cnt=0;
+  while( (rest > 0)  && (frameQueue->emptyQueueCanRead()) ) {
+
+    int state=framer->getState();
+    switch(state) {
+    case FRAME_NEED: {
+      int bytes=framer->canStore();
+      ptr=currentPacket->contents+currentPos;
+      // don't read beyond the packet.
+      if (bytes >= rest) {
+	bytes=rest;
+	// we must copy this, because we can commit a packet,
+	// which then deletes the pointer to the packet.
+	// but the framer still wants to access the data, later!
+	if (bytes > INPUT_SIZE) {
+	  cout << "inputbuffer too small"<<endl;
+	  exit(0);
+	}
+ 	memcpy(inputbuffer,ptr,bytes);
+	ptr=inputbuffer;
+      }
+      framer->store(ptr,bytes);
+      currentPos+=bytes;
+      rest-=bytes;
+      break;
+    }
+    case FRAME_WORK:
+      framer->work();
+      break;
+    case FRAME_HAS:{
+      AudioFrame* emptyFrame= frameQueue->emptyQueueDequeue();
+      if (splay->decode(framer->outdata(),framer->len(),emptyFrame)==true) {
+         frameQueue->dataQueueEnqueue(emptyFrame);
+         cnt++;
+      }
+      break;
+    }
+    default:
+      cout << "unknown state in mpeg audio framing"<<endl;
+      exit(0);
+    }
+  }
+  //cout << "rest-last:"<<rest<<" cnt:"<<cnt<<endl;
+
+  if (rest == 0) {
+    arts_debug("packet processed");
+    currentPacket->processed();
+    packetQueue->dequeue();
+    currentPos=0;
+  }
+
+}
+
+string SplayPlayObject_impl::description() {
+  arts_debug("description [GET1]");
+  string back;
+  return back;
+}
+
+void SplayPlayObject_impl::description(const string &) {
+  arts_debug("description [GET2]");
+}
+
+Arts::poTime SplayPlayObject_impl::currentTime() {
+  Arts::poTime time;
+  return time;
+}
+
+
+
+Arts::poTime SplayPlayObject_impl::overallTime() {
+  Arts::poTime time;
+  return time;
+}
+
+Arts::poCapabilities SplayPlayObject_impl::capabilities() {
+  arts_debug("capabilities");
+  return (Arts::poCapabilities)(Arts::capPause);
+}
+
+string SplayPlayObject_impl::mediaName() {
+  arts_debug("mediaName");
+  string back;
+  return back;
+}
+
+Arts::poState SplayPlayObject_impl::state() {
+  return _state;
+}
+
+void SplayPlayObject_impl::play() {
+  arts_debug("play:");
+  if (file == NULL) {
+    arts_debug("file is NULL:");
+    if (lStreaming && _state != Arts::posPlaying) {
+      currentStream.streamStart();
+      _state = Arts::posPlaying;
+    }
+    
+  } else {
+    _state = Arts::posPlaying;
+  }
+}
+
+
+void SplayPlayObject_impl::seek(const class Arts::poTime& ) {
+  arts_debug("SEEK - RESET in decoder.");
+  framer->reset();
+  // and remove pre-decoded frame:
+  frameQueue->clear();
+    
+  return;
+}
+
+
+void SplayPlayObject_impl::pause() {
+  arts_debug("pause");
+  _state=Arts::posPaused;
+}
+/*
+ *
+ * halt() (which the normal programmer would probably refer to as stop())
+ * should seek to the beginning and go into the posIdle state, like a just
+ * opened PlayObject
+ *
+ */
+
+void SplayPlayObject_impl::halt() {
+  arts_debug("halt");
+  _state=Arts::posIdle;
+}
+ 
+
+void SplayPlayObject_impl::streamInit() {
+  arts_debug("streamInit");
+  return;
+}
+
+
+void SplayPlayObject_impl::streamStart() {
+  arts_debug("streamStart");
+  return;
+}
+
+
+
+void SplayPlayObject_impl::calculateBlock(unsigned long wantSamples) {
+  unsigned long i;
+  unsigned long haveSamples=frameQueue->getLen();
+
+  // the wantSamples*2 takes care that there is enough data
+  // even if we have stereo.
+
+  
+  if (haveSamples < wantSamples*2) {
+    if (lStreaming) {
+      for(i=0;i<wantSamples;i++) {
+	left[i] = right[i] = 0.0;
+      }
+      return;
+    }
+    // else (file access)
+    getMoreSamples(wantSamples*2);
+  }
+
+  // Yep, we have enough data
+
+
+
+  // now check if we need resampling.
+  AudioFrame* audioFrame=frameQueue->getCurrent();
+  double wav_samplingRate=(double)audioFrame->getFrequenceHZ();
+
+  /* difference between the sampling rates in percent */
+  float diff = fabs(wav_samplingRate-samplingRateFloat) / samplingRateFloat;
+  //cout << "wav_samplingRate:"<<wav_samplingRate<<endl;
+  /*
+   * efficient optimized case:
+   * 1. decoder -> float rendering
+   * 2. no resampling (i.e. artsd running @ 44100 Hz, playing an 44100 Hz mp3)
+   */
+
+  if(diff < 0.02) {
+    // no resample (easy+fast case)
+    haveSamples=frameQueue->copy(left,right,wantSamples);
+
+    for(i=haveSamples;i<wantSamples;i++) {
+      left[i] = right[i] = 0.0;
+    }
+    frameQueue->forwardStreamDouble(haveSamples);
+    
+    if (lStreaming) {
+      processQueue();
+    }
+  } else {
+    /** FIXME: you can use this to implement pitchable playobject **/
+    double _speed = 1.0; 
+
+    // calculate "how fast" we consume input samples (speed = 2.0 means,
+    // we need 2 input samples to generate one output sample)
+    double speed = (double)wav_samplingRate/(double)(samplingRateFloat/_speed);
+
+    // calculate how many samples we need to read from the frameQueue to
+    // satisfy the request
+    long readSamples = long((double)wantSamples*speed+8.0);
+    checkResampleBuffer(readSamples*2);
+
+    // read the samples
+    //  - readSamplesOk contains how much we really got
+    //  - we put the samples non-interleaved into the resampleBuffer,
+    //    first the left channel, then the right channel
+    long readSamplesOk = frameQueue->copy(&resampleBuffer[0],
+					  &resampleBuffer[readSamples],
+					  readSamples);
+    // check how many output samples we will be able to generate from that
+    long samplesToConvert = long((double)readSamplesOk/speed)-4;
+    if(samplesToConvert < 0) samplesToConvert = 0;
+    if(samplesToConvert > wantSamples) samplesToConvert = wantSamples;
+
+    // do the conversion
+    Arts::interpolate_mono_float_float(samplesToConvert,flpos,speed,
+				 &resampleBuffer[0],left);
+    Arts::interpolate_mono_float_float(samplesToConvert,flpos,speed,
+				 &resampleBuffer[readSamples],right);
+
+    // calculate where we are now (as floating point position) in our
+    // inputsample buffer
+    flpos += (double)samplesToConvert * speed;
+
+    // Good - so how many input samples we won't need anymore (for the
+    // next request)? Skip them.
+    int skip = (int)floor(flpos);
+    if(skip)
+    {
+      frameQueue->forwardStreamDouble(skip);
+      flpos = flpos - floor(flpos);
+    }
+
+    for(i=samplesToConvert; i<wantSamples; i++) {
+      left[i] = right[i] = 0.0;
+    }
+
+    if (lStreaming) {
+      processQueue();
+    }
+  }
+ 
+  // ok, eof stop.
+  /*
+  if (haveSamples  < wantSamples) {
+    halt();
+  }
+  */
+}
+
+void SplayPlayObject_impl::streamEnd() {
+  arts_debug("streamEnd");
+  if (file != NULL) {
+    fclose(file);
+    file=NULL;
+  }
+  return;
+}
+
+
+void SplayPlayObject_impl::checkResampleBuffer(int size) {
+ 
+
+  if (resampleBufferSize != size) {
+    if (resampleBuffer != NULL) {
+      delete resampleBuffer;
+    }
+    resampleBuffer = new float[size];
+    resampleBufferSize=size;
+  }
+ 
+}
+
+
+
+//
+// This method shows how we insert data in the framer,
+// decode a frame and store it in the queue.
+// This damned queue is necessary, becasue a sync out
+// stream needs a guranteed streamsize == samples.
+// an async stream would simply send a packet
+// and we are done.
+
+void SplayPlayObject_impl::getMoreSamples(int needLen) {
+
+
+  while( (feof(file) == false) && (frameQueue->getLen() < needLen) ) {
+
+    //
+    // 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=framer->getState();
+    int cnt=0;
+    switch(state) {
+    case FRAME_NEED: {
+      int bytes=framer->canStore();
+      int read=fread(inputbuffer,1,bytes,file);
+      if (read != bytes) {
+	// read error. reset framer
+	framer->reset();
+	continue;
+      }
+      // Note: The inputbuffer must NOT NOT NOT be on the stack!
+      framer->store(inputbuffer,bytes);
+      break;
+    }
+    case FRAME_WORK:
+      framer->work();
+      break;
+    case FRAME_HAS:{
+      AudioFrame* emptyFrame= frameQueue->emptyQueueDequeue();
+      if (splay->decode(framer->outdata(),framer->len(),emptyFrame)==true) {
+         frameQueue->dataQueueEnqueue(emptyFrame);
+         cnt++;
+      }
+      break;
+    }
+    default:
+      cout << "unknown state in mpeg audio framing"<<endl;
+      exit(0);
+    }
+  }
+  if (feof(file)==true) {
+    halt();
+    return;
+  }
+}
+
+REGISTER_IMPLEMENTATION(SplayPlayObject_impl);
+
+// vim:ts=8:sw=2:sts=2
-- 
cgit v1.2.3