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621 lines
15 KiB
621 lines
15 KiB
/*
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base class for all mpeglib decoders
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Copyright (C) 2000 Martin Vogt
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This program is free software; you can redistribute it and/or modify
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it under the terms of the GNU Library General Public License as published by
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the Free Software Foundation.
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For more information look at the file COPYRIGHT in this package
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*/
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#include <queue>
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#include <iostream>
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#include <connect.h>
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#include "decoderBaseObject_impl.h"
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#include "../mpeglib/lib/decoder/decoderPlugin.h"
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#include "debug.h"
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// define this to run the playobject without the
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// arts backend. (useful to check if a bug is in arts or mpeglib)
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//#define _STRIP_ZERO
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static int instanceCnt=0;
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DecoderBaseObject_impl::DecoderBaseObject_impl()
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: _speed(1.0f)
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{
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flpos=0.0;
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_blocking = false;
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#ifdef _STRIP_ZERO
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outputStream=NULL;
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#else
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m_outputStream=new ArtsOutputStream(NULL);
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arts_debug("outputStream created");
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decoderPlugin=NULL;
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#endif
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startTime=0.0;
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m_inputStream=NULL;
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setStreamState(_THREADSTATE_INIT);
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_state=Arts::posIdle;
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instance=instanceCnt;
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instanceCnt++;
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m_packetQueue = new std::queue<DataPacket<mcopbyte>*>;
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}
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DecoderBaseObject_impl::~DecoderBaseObject_impl() {
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arts_debug("~DecoderBaseObject_impl -s");
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shudownPlugins();
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if (decoderPlugin != NULL) {
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arts_debug("delete decoderPlugin");
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delete decoderPlugin;
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decoderPlugin=NULL;
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}
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if (m_outputStream != NULL) {
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arts_debug("delete outputStream");
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delete m_outputStream;
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m_outputStream=NULL;
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}
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if (m_streaming)
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m_artsInputStream.streamEnd();
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delete m_packetQueue;
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}
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DecoderPlugin* DecoderBaseObject_impl::createPlugin() {
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arts_fatal("direct virtual call DecoderBaseObject_impl::getPlugin");
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return NULL;
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}
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InputStream* DecoderBaseObject_impl::createInputStream(const char* url) {
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InputStream* back = InputPlugin::createInputStream(url,true);
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return back;
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}
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bool DecoderBaseObject_impl::loadMedia(const string &filename) {
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arts_debug("loadMedia");
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int back=true;
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m_streaming = false;
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if ( m_inputStream != NULL ) {
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arts_fatal("remove resources first with a call to: halt()");
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}
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if (decoderPlugin == NULL) {
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decoderPlugin=createPlugin();
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if(doFloat()) decoderPlugin->config("dofloat",0,0);
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}
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flpos=0.0;
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startTime=0.0;
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lastAudioBufferSize=-1;
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/**
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Note: you can only play one file with a PlayObject !!
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Then you must destroy it.
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A StreamEnd call should do the job.
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*/
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#ifdef _STRIP_ZERO
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return true;
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#endif
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m_inputStream=createInputStream(filename.c_str());
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// the plugin does not open the stream!
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// we do it.
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back=m_inputStream->open((char*)filename.c_str());
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setStreamState(_THREADSTATE_OPENED);
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// we are still in posIdle here
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m_outputStream->audioOpen();
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// watch the order!
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decoderPlugin->setOutputPlugin(m_outputStream);
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decoderPlugin->setInputPlugin(m_inputStream);
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return back;
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}
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#define INPUT_BUFFER_SIZE 32768
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bool DecoderBaseObject_impl::streamMedia(Arts::InputStream instream) {
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arts_debug("DecoderBaseObject_impl::streamMedia -s");
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bool back = true;
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if (m_inputStream != NULL) {
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arts_fatal("resource in use, call halt() first");
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}
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if (decoderPlugin == NULL) {
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decoderPlugin = createPlugin();
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if (doFloat())
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decoderPlugin->config("dofloat", 0, 0);
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// streaming, don't know the length
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decoderPlugin->config("-c", 0, 0);
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}
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flpos = 0.0;
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startTime = 0.0;
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m_streaming = true;
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lastAudioBufferSize = -1;
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m_artsInputStream = instream;
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m_inputStream = new BufferInputStream(INPUT_BUFFER_SIZE, 4096, (char*)"InputStream");
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m_inputStream->open((char*)"InputStream");
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// connect the stream now
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Arts::StreamPlayObject self = Arts::StreamPlayObject::_from_base(_copy());
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connect(m_artsInputStream, "outdata", self);
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setStreamState(_THREADSTATE_OPENED);
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m_outputStream->audioOpen();
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decoderPlugin->setOutputPlugin(m_outputStream);
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decoderPlugin->setInputPlugin(m_inputStream);
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arts_debug("DecoderBaseObject_impl::streamMedia -e");
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return back;
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}
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void DecoderBaseObject_impl::process_indata(DataPacket<mcopbyte> *inpacket) {
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m_packetQueue->push(inpacket);
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processQueue();
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}
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void DecoderBaseObject_impl::processQueue() {
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// early exit if no packets in the queue
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if (m_packetQueue->empty())
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return;
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// see how much space we have in the stream
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BufferInputStream* stream = static_cast<BufferInputStream*>(m_inputStream);
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if (!stream) return;
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int length = stream->getByteLength();
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int freeSpace = INPUT_BUFFER_SIZE - length;
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DataPacket<mcopbyte> *inpacket = m_packetQueue->front();
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if (!inpacket) return;
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if (freeSpace >= inpacket->size) {
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stream->write((char*)inpacket->contents, inpacket->size, 0);
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m_packetQueue->pop();
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inpacket->processed();
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}
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}
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string DecoderBaseObject_impl::description() {
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arts_debug("description");
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string back;
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#ifdef _STRIP_ZERO
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return back;
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#endif
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PluginInfo* pluginInfo=decoderPlugin->getPluginInfo();
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pluginInfo->print();
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return back;
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}
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void DecoderBaseObject_impl::description(const string &) {
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arts_debug("description");
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// what should do this?
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}
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poTime DecoderBaseObject_impl::currentTime() {
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poTime time;
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#ifdef _STRIP_ZERO
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return time;
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#endif
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AudioTime* audioTime=m_outputStream->getAudioTime();
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float currentTime=audioTime->getTime()+(float)startTime;
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time.seconds=(long)(currentTime);
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time.ms=(long) (1000.0*(currentTime-(float)time.seconds));
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return time;
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}
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poTime DecoderBaseObject_impl::overallTime() {
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poTime time;
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#ifdef _STRIP_ZERO
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return time;
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#endif
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PluginInfo* pluginInfo=decoderPlugin->getPluginInfo();
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time.seconds=pluginInfo->getLength();
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time.ms=0;
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return time;
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}
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poCapabilities DecoderBaseObject_impl::capabilities() {
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arts_debug("capabilities");
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#ifdef _STRIP_ZERO
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return capSeek;
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#endif
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PluginInfo* pluginInfo=decoderPlugin->getPluginInfo();
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long len=pluginInfo->getLength();
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if (len == 0) {
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return Arts::capPause; /* no seek supported */
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}
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// seek and pause supported
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return (poCapabilities)(Arts::capSeek | Arts::capPause);
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}
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string DecoderBaseObject_impl::mediaName() {
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arts_debug("mediaName");
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string back;
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// whats a mediaName?
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return back;
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}
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poState DecoderBaseObject_impl::state() {
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return _state;
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}
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void DecoderBaseObject_impl::play() {
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arts_debug("play: %d", (int)streamState);
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if (streamState == _THREADSTATE_OPENED) {
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decoderPlugin->play();
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} else {
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Command cmd(_COMMAND_PLAY);
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decoderPlugin->insertAsyncCommand(&cmd);
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}
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setStreamState(_THREADSTATE_PLAYING);
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_state = Arts::posPlaying;
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}
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void DecoderBaseObject_impl::seek(const class poTime& seekTime) {
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#ifdef _STRIP_ZERO
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return;
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#endif
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long sec=seekTime.seconds;
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arts_debug("sec in plugin is %d:", sec);
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// we send an async command
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Command cmd(_COMMAND_SEEK,sec);
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decoderPlugin->insertAsyncCommand(&cmd);
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// if the thread blocks on the artsOutputstream: kick him out
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// the next command will the the seek command
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m_outputStream->audioClose();
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// thread blocking allowed
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m_outputStream->audioOpen();
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arts_debug("************ reopen");
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// now set a new startTime
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startTime=sec;
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}
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void DecoderBaseObject_impl::pause() {
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arts_debug("pause");
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_state = Arts::posPaused;
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Command cmd(_COMMAND_PAUSE);
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decoderPlugin->insertAsyncCommand(&cmd);
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}
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void DecoderBaseObject_impl::halt() {
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/*
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*
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* halt() (which the normal programmer would probably refer to as stop())
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* should seek to the beginning and go into the posIdle state, like a just
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* opened PlayObject
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*
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*/
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arts_debug("halt");
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_state=Arts::posIdle;
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shudownPlugins();
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}
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void DecoderBaseObject_impl::streamInit() {
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#ifdef _STRIP_ZERO
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return;
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#endif
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}
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void DecoderBaseObject_impl::streamStart() {
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arts_debug("DecoderBaseObject_impl::streamStart");
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}
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int DecoderBaseObject_impl::fillArts(unsigned long samples,
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float* left , float* right) {
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unsigned long haveSamples = 0;
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AudioTime* audioTime=m_outputStream->getAudioTime();
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int wav_samplingRate=audioTime->getSpeed();
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int wav_sampleWidth=audioTime->getSampleSize();
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int wav_channelCount=audioTime->getStereo()+1;
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if(doFloat()) wav_sampleWidth = sizeof(float)*8;
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// here seems to be an error, I have clicks sometimes in the stream
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//int byteMultiplikator=(wav_sampleWidth/8)*wav_channelCount;
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// maybe first multiply, then divide?
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int byteMultiplikator = wav_channelCount * wav_sampleWidth / 8;
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char* buffer;
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int hasBytes = 0;
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int wantBytes = 0;
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int bufferSize=getBufferSize();
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if (bufferSize != lastAudioBufferSize) {
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lastAudioBufferSize=bufferSize;
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m_outputStream->setAudioBufferSize(bufferSize);
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}
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/* difference between the sampling rates in percent */
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float diff = fabs((double)wav_samplingRate - (double)(samplingRateFloat/_speed))
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/ (double)samplingRateFloat;
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/*
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* efficient optimized case:
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* 1. decoder -> float rendering
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* 2. no resampling (i.e. artsd running @ 44100 Hz, playing an 44100 Hz mp3)
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*/
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if(_state == Arts::posPlaying && doFloat() && diff < 0.0005) {
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wantBytes = sizeof(float) * wav_channelCount * samples;
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hasBytes = m_outputStream->read(&buffer,wantBytes);
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float *flptr = (float *)buffer;
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if(wav_channelCount == 1)
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{
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while((int)(haveSamples * sizeof(float)) < hasBytes)
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{
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left[haveSamples] = right[haveSamples] = flptr[haveSamples];
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haveSamples++;
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}
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}
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else if(wav_channelCount == 2)
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{
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while((int)(haveSamples * 2 * sizeof(float)) < hasBytes)
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{
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left[haveSamples] = flptr[haveSamples*2];
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right[haveSamples] = flptr[haveSamples*2+1];
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haveSamples++;
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}
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}
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m_outputStream->forwardReadPtr(haveSamples*sizeof(float)*wav_channelCount);
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}
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else if(_state == Arts::posPlaying) {
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//
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// since the samplingrate of the MP3 and the samplingrate of the output
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// device (soundcard) are not necessarily the same, it's a bit tricky
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//
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// calculate "how fast" we consume input samples (2.0 means, we need 2
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// input samples to generate one output sample)
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double speed = (double)wav_samplingRate / (double)(samplingRateFloat/_speed);
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// calculate how many input samples we need, then to satisfy the request
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// use a larger amount than "really" required, to ensure that samples are
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// available for rounding errors and interpolation
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double wantWavSamples = (double)samples*speed+8.0;
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// convert that into bytes and try to read that many bytes
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wantBytes=(int) (wantWavSamples*byteMultiplikator);
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hasBytes=m_outputStream->read(&buffer,wantBytes);
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int format = doFloat()?Arts::uni_convert_float_ne:wav_sampleWidth;
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// convert those bytes into the suitable output form
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haveSamples = Arts::uni_convert_stereo_2float(samples, (unsigned char *)buffer,
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hasBytes, wav_channelCount,
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format,
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left,right,speed,flpos);
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// calculate where we are now (as floating point position) in our
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// inputsample buffer
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flpos += (double)haveSamples * speed;
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// Good - so how many input samples we won't need anymore (for the
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// next request)? Skip them.
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int skip = (int)floor(flpos);
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// we need to call this even on skip == 0
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// because we must unlock the remoteBuffer
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int forward=skip*byteMultiplikator;
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flpos = flpos - floor(flpos);
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m_outputStream->forwardReadPtr(forward);
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}
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if(haveSamples != samples) {
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unsigned long i;
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for(i=haveSamples;i<samples;i++)
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left[i] = right[i] = 0.0;
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}
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return samples;
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}
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void DecoderBaseObject_impl::calculateBlock(unsigned long samples,
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float* left , float* right) {
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#ifndef _STRIP_ZERO
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int audioState=m_outputStream->waitStreamState(_OUTPUT_WAIT_METHOD_POLL,
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_STREAM_MASK_ALL,
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_STREAMTYPE_AUDIO);
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if (audioState & _STREAM_MASK_IS_INIT) {
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// now check if we already have enough data
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int lenough=false;
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if (audioState & _STREAM_MASK_IS_EOF) {
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if(_state == Arts::posPlaying) {
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arts_debug("eof got in arts********** END");
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_state = Arts::posIdle;
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}
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lenough=true;
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}
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if (m_outputStream->getBufferFillgrade() >= 4096) {
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lenough=true;
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}
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if (_state == Arts::posPlaying) {
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if (m_streaming) {
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// produce more data
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processQueue();
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// check for stream end
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if ( m_inputStream->getByteLength() == 0 ) {
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if ( m_artsInputStream.eof() ) {
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m_inputStream->close();
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m_artsInputStream.streamEnd();
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}
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}
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}
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if (lenough || _blocking) {
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fillArts(samples, left, right);
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return;
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}
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}
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}
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#endif
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// filling with zero , stream not ready(yet)
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unsigned int i;
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for(i=0;i<samples;i++)
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left[i] = right[i] = 0.0;
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}
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void DecoderBaseObject_impl::streamEnd() {
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arts_debug("streamEnd");
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#ifdef _STRIP_ZERO
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return;
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#endif
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halt();
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}
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int DecoderBaseObject_impl::getBufferSize() {
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float hardwareBuffer;
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float fragmentSize;
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float fragmentCount;
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float channels;
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float sampleSize;
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fragmentSize=Arts::AudioSubSystem::the()->fragmentSize();
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fragmentCount=Arts::AudioSubSystem::the()->fragmentCount();
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channels=Arts::AudioSubSystem::the()->channels();
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sampleSize=16.0/8.0;
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hardwareBuffer=fragmentSize*fragmentCount;
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return (int)hardwareBuffer;
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}
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void DecoderBaseObject_impl::shudownPlugins() {
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arts_debug("shudownPlugins -s");
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/**
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The order here is important.
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First we close the audio so that the thread never blocks
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on the ringbuffer.
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Then we are sure thst we can safley call plugin->close,
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because the thread does not block.
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We then have the thread back in the decoder_loop of
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the plugin.
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*/
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// this should theoretically be faster
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if (decoderPlugin != NULL) {
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Command cmd(_COMMAND_CLOSE);
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decoderPlugin->insertAsyncCommand(&cmd);
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}
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if (m_outputStream != NULL) {
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m_outputStream->audioClose();
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}
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// very likely the thread already is closed
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// because of the asyncCommand above.
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if (decoderPlugin) {
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decoderPlugin->close();
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}
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delete m_inputStream;
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m_inputStream=NULL;
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if (m_streaming)
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m_artsInputStream.streamEnd();
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setStreamState(_THREADSTATE_CLOSED);
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arts_debug("shudownPlugins -e");
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}
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void DecoderBaseObject_impl::setStreamState(int state) {
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switch (state) {
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case _THREADSTATE_INIT: {
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streamState=_THREADSTATE_INIT;
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break;
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}
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case _THREADSTATE_OPENED: {
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streamState=_THREADSTATE_OPENED;
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break;
|
|
}
|
|
case _THREADSTATE_PLAYING: {
|
|
streamState=_THREADSTATE_PLAYING;
|
|
break;
|
|
}
|
|
case _THREADSTATE_CLOSED: {
|
|
streamState=_THREADSTATE_INIT;
|
|
break;
|
|
}
|
|
default:
|
|
std::cerr << "unknown streamState:DecoderBaseObject_impl:"<<state<<std::endl;
|
|
}
|
|
|
|
}
|
|
|
|
/** internal stuff for testing **/
|
|
|
|
void DecoderBaseObject_impl::blocking(bool newvalue)
|
|
{
|
|
_blocking = newvalue;
|
|
}
|
|
|
|
bool DecoderBaseObject_impl::blocking()
|
|
{
|
|
return _blocking;
|
|
}
|
|
|
|
void DecoderBaseObject_impl::speed(float newValue)
|
|
{
|
|
_speed= newValue;
|
|
}
|
|
|
|
float DecoderBaseObject_impl::speed()
|
|
{
|
|
return _speed;
|
|
}
|