hxaudstr_new.cpp

著名的 helix realplayer 基于手机 symbian 系统的 播放器全套源代码

{
    HX_RESULT	    theErr = HXR_OK;
    BOOL	    bInTSRollOver = FALSE;
    HXAudioInfo*    pAinfo = 0;

    /* If buffer is NULL, it means that the user just 
     * wants to know what timestamp should be placed in the next 
     * STREAMED/TIMED audio data
     */
    if (!pAudioData->pData)
    {
	HXAudioInfo* pInfo = NULL;

	if (!m_pDataList->IsEmpty() && 
	    NULL != (pInfo = (HXAudioInfo*) m_pDataList->GetTail()))
	{
	    pAudioData->ulAudioTime = pInfo->ulStartTime + 
				      CalcMs(pInfo->pBuffer->GetSize());
	}
	else
	{
	    pAudioData->ulAudioTime = INT64_TO_UINT32(m_llLastWriteTime - CAST_TO_INT64 (m_ulTSRollOver) * CAST_TO_INT64 (MAX_UINT32));
	}

	return HXR_OK;
    }

    // make sure the renderer does not pass NULL data!!
    HX_ASSERT(pAudioData->pData->GetBuffer() != NULL &&
	      pAudioData->pData->GetSize() != 0);
    if (pAudioData->pData->GetBuffer()	== NULL ||
	pAudioData->pData->GetSize()	== 0)
    {
	return HXR_INVALID_PARAMETER;
    }

    if (m_bIsFirstPacket)
    {
	m_bIsFirstPacket = FALSE;
	
	IHXErrorMessages* pErrMsg = NULL;
	if (HXR_OK == m_Owner->m_pContext->QueryInterface(IID_IHXErrorMessages, (void**)&pErrMsg))
	{
	    DEBUG_OUT(pErrMsg, DOL_GENERIC, (s,"AudioFormatIn: %lu channels %lu SamplesPerSec", m_AudioFmt.uChannels, m_AudioFmt.ulSamplesPerSec));
	    DEBUG_OUT(pErrMsg, DOL_GENERIC, (s,"AudioFormatOut: %lu channels %lu SamplesPerSec", m_DeviceFmt.uChannels, m_DeviceFmt.ulSamplesPerSec));
	}
	HX_RELEASE(pErrMsg);

	if (m_bIsLive)
	{
            // XXX wschildbach why do we update a last write time *here*?
	    m_Owner->UpdateStreamLastWriteTime();
	    UpdateStreamLastWriteTime(TRUE);
	}
    }
//{FILE* f1 = ::fopen("c:\\temp\\audio.txt", "a+"); ::fprintf(f1, "%lu\tAddData\t%p\t%lu\n", HX_GET_BETTERTICKCOUNT(), this, pAudioData->ulAudioTime);::fclose(f1);}
//    ::fwrite(pAudioData->pData->GetBuffer(), pAudioData->pData->GetSize(), 1, fdin);

    UINT32 ulDataTime = CalcMs(pAudioData->pData->GetSize());
    UINT32 ulEndTime = pAudioData->ulAudioTime + ulDataTime;

    if (m_pAvailableBuffers && !m_bDeterminedInitialCacheSize && ulDataTime > 0)
    {
	m_bDeterminedInitialCacheSize = TRUE;
	m_uCacheSize = (UINT16) (m_ulGranularity*2/ulDataTime) + 1;
	/* make sure it is atleast CACHE_INCREMENT_SIZE to begin with */
	m_uCacheSize = m_uCacheSize < CACHE_INCREMENT_SIZE ? 
			    CACHE_INCREMENT_SIZE : m_uCacheSize;
    }

    if (m_ulLastInputStartTime > pAudioData->ulAudioTime &&
	((m_ulLastInputStartTime - pAudioData->ulAudioTime) > MAX_TIMESTAMP_GAP))
    {
	bInTSRollOver = TRUE;
	m_ulTSRollOver++;
    }

    m_ulLastInputStartTime  = pAudioData->ulAudioTime; 
    m_ulLastInputEndTime    = ulEndTime;

    /* even in STREAMING_AUDIO case, it might happen, that the packets
     * written are late. e.g. packets received late on the network 
     */

    // XXX wschildbach : Why do we even care? Just insert the packet into the queue
    // and let the reaper take care of it!

    INT64 llActualTimestamp = CAST_TO_INT64 (pAudioData->ulAudioTime) + CAST_TO_INT64 m_ulTSRollOver * CAST_TO_INT64 MAX_UINT32;
    INT64 llActualEndTime = CAST_TO_INT64 (pAudioData->ulAudioTime) + CAST_TO_INT64 (ulDataTime) +
			    CAST_TO_INT64 (m_ulTSRollOver) * CAST_TO_INT64 (MAX_UINT32);

#if 0 // testing sampling frequency estimation for "inaccurate resampling"
    if (pAudioData->uAudioStreamType == TIMED_AUDIO)
    {
        m_startMeasureTime = llActualTimestamp ;
        m_totSamples = 0 ;
    }
    else if (pAudioData->uAudioStreamType == STREAMING_AUDIO)
    {
        float diffTime = (llActualTimestamp - m_startMeasureTime) / 1000.0f ;
        if (diffTime)
        {
            float frEstimate = m_totSamples / m_AudioFmt.uChannels / diffTime ;
            float frEstErr = frEstimate * 0.001f / diffTime ; // 1 ms inaccuracy
            {FILE *f1 = fopen("c:\\temp\\estimatesr.txt","a+");
             fprintf(f1,"%I64d\t%I64d\t%f\t%f\n",llActualTimestamp,llActualEndTime,frEstimate,frEstErr);
             fclose(f1);
            }
        }
    }
    m_totSamples += CalcSamples(pAudioData->pData->GetSize()) ;
#endif

    if ((pAudioData->uAudioStreamType == STREAMING_AUDIO ||
	 pAudioData->uAudioStreamType == TIMED_AUDIO) &&
       !(llActualTimestamp >= m_llLastWriteTime ||
	 llActualEndTime > m_llLastWriteTime)) 
    {
	/* Too late*/
	m_bTobeTimed	= TRUE;
//{FILE* f1 = ::fopen("e:\\audio.txt", "a+"); ::fprintf(f1, "%lu\t%p\t%d\t%lu\t%lu\tLATE\n", HX_GET_BETTERTICKCOUNT(), this, m_pDataList->GetCount(), pAudioData->ulAudioTime, (INT32)m_llLastWriteTime);::fclose(f1);}

	/*
    {FILE *f1 = fopen("c:\\temp\\mix.txt","a+"); fprintf(f1,"LATE packet\n");
     fclose(f1);
    }
    */  
      
        return HXR_LATE_PACKET;	
    }

    pAinfo = new HXAudioInfo;
    if( !pAinfo )
    {
        return HXR_OUTOFMEMORY;
    }

    pAudioData->pData->AddRef();
    pAinfo->pBuffer 		= pAudioData->pData;
    pAinfo->ulStartTime  	= pAudioData->ulAudioTime;
    pAinfo->pOffset     	= pAudioData->pData->GetBuffer();  
    pAinfo->ulBytesLeft		= pAudioData->pData->GetSize();  
    pAinfo->uAudioStreamType    = pAudioData->uAudioStreamType;

    if (m_bTobeTimed && pAudioData->uAudioStreamType == STREAMING_AUDIO)
    {
	pAinfo->uAudioStreamType = TIMED_AUDIO;
	m_bTobeTimed		 = FALSE;
    }
    else if (m_bTobeTimed && pAudioData->uAudioStreamType == TIMED_AUDIO)
    {
	m_bTobeTimed	= FALSE;
    }

//{FILE* f1 = ::fopen("c:\\temp\\audio.txt", "a+"); ::fprintf(f1, "AddData ulAudioTime: %lu\n", pAudioData->ulAudioTime);::fclose(f1);}

//////////////////////////////////////////////////////////////////////
                // XXX wschildbach
    // the start time of this packet in samples. This may be corrected later on.
    pAinfo->llStartTimeInSamples = CAST_TO_INT64(llActualTimestamp) * m_AudioFmt.ulSamplesPerSec / 1000 * m_AudioFmt.uChannels ;
    pAinfo->llEndTimeInSamples   = pAinfo->llStartTimeInSamples + Bytes2Samples(pAinfo->pBuffer->GetSize(), &m_AudioFmt) ;

//////////////////////////////////////////////////////////////////////

    if (pAinfo->uAudioStreamType == INSTANTANEOUS_AUDIO)
    {
	CHXSimpleList* pList = new CHXSimpleList;
        if( !pList )
        {
            HX_RELEASE(pAudioData->pData);
            HX_DELETE(pAinfo);
            return HXR_OUTOFMEMORY;
        }
	pList->AddHead((void*) pAinfo);
	m_pInstantaneousList->AddTail((void*) pList);
	m_Owner->m_Owner->ToBeRewound();
    }
    else if (pAinfo->uAudioStreamType == STREAMING_INSTANTANEOUS_AUDIO)
    {
	HX_ASSERT(m_pInstantaneousList && m_pInstantaneousList->GetCount() > 0);
	CHXSimpleList* pList = NULL;
	if (m_pInstantaneousList->GetCount() == 0)
	{
	    pList = new CHXSimpleList;
            if( !pList )
            {
                HX_RELEASE(pAudioData->pData);
                HX_DELETE(pAinfo);
                return HXR_OUTOFMEMORY;
            }
	    m_pInstantaneousList->AddTail((void*) pList);

	    // fix for naive users!
	    pAinfo->uAudioStreamType = INSTANTANEOUS_AUDIO;
	    m_Owner->m_Owner->ToBeRewound();
	}

	pList = (CHXSimpleList*) m_pInstantaneousList->GetTail();
	pList->AddTail(pAinfo);
    }
    else if (m_pDataList->IsEmpty())
    {
	/*
    {FILE *f1 = fopen("c:\\temp\\mix.txt","a+"); fprintf(f1,"adding to empty list...\n");
     fclose(f1);
    }
	*/
        m_pDataList->AddTail((void*) pAinfo);
    }
    else
    {
/*
    {FILE *f1 = fopen("c:\\temp\\mix.txt","a+"); fprintf(f1,"adding to non-empty list...\n");
     fclose(f1);
    }
*/
	HXAudioInfo* pInfo = (HXAudioInfo*) m_pDataList->GetTail();

	UINT32	ulActualTSRollOver = m_ulTSRollOver;

	if (bInTSRollOver && ulActualTSRollOver)
	{
	    ulActualTSRollOver--;
	}

	INT64 llActualLastEndTime = CAST_TO_INT64 (pInfo->ulStartTime) + CAST_TO_INT64 (CalcMs(pInfo->pBuffer->GetSize())) +
				    CAST_TO_INT64 ulActualTSRollOver * CAST_TO_INT64 MAX_UINT32;
	INT64 llActualLastStartTime = CAST_TO_INT64 (pInfo->ulStartTime) + CAST_TO_INT64 ulActualTSRollOver * CAST_TO_INT64 MAX_UINT32;
	
	if (llActualTimestamp < llActualLastStartTime)
	{
	    /* Not allowed */
	    theErr = HXR_OUTOFORDER_PACKET; 
	    /* something is fu*#$#up... figure out what?*/

	    HX_ASSERT(!("Packets written out of order"));
	    goto exit;
	}

	if (pAinfo->uAudioStreamType == STREAMING_AUDIO)
	{
	    /* is it a resonable packet to add to the list */
	    if ((llActualTimestamp <= llActualLastEndTime		&&
		 llActualLastEndTime - llActualTimestamp <= m_ulFudge)	|| 
		(llActualTimestamp >= llActualLastEndTime		&&
		 llActualTimestamp - llActualLastEndTime <= m_ulFudge))
	    {

                // XXX wschildbach
                // make 64-bit timestamps contiguous before adding into the queue.
                pAinfo->llEndTimeInSamples   += pInfo->llEndTimeInSamples - pAinfo->llStartTimeInSamples ;
                pAinfo->llStartTimeInSamples  = pInfo->llEndTimeInSamples ;

                m_pDataList->AddTail((void*) pAinfo);
                
	    }
	    else
	    {
		theErr = HXR_NONCONTIGUOUS_PACKET; //HX_LATE_PACKET;
		/* something is fu*#$#up... figure out what?*/
		HX_ASSERT(!("Streaming Audio: Non-Contigous Write"));
		m_bTobeTimed	= TRUE;
		goto exit;
	    }
	}
	else
	{
	    /* see if there is any overlap.. we do not allow any overlap */
	    if (llActualTimestamp < llActualLastEndTime &&
		llActualLastEndTime - llActualTimestamp > m_ulFudge)
	    {
		/* hmmm an overlapped packet */
		theErr = HXR_OVERLAPPED_PACKET;
		/* something is fu*#$#up... figure out what?*/
		HX_ASSERT(!("Timed Audio: Overlapping write"));
		m_bTobeTimed	= TRUE;
		goto exit;
	    }
	    else
	    {
		m_pDataList->AddTail((void*) pAinfo);
	    }
	}
    }

exit:
    if (theErr != HXR_OK && pAinfo)
    {
	pAinfo->pBuffer->Release();
	delete pAinfo;
    }

//{FILE* f1 = ::fopen("e:\\audio.txt", "a+"); ::fprintf(f1, "%lu\t%p\t%d\t%lu\t%lu\n", HX_GET_BETTERTICKCOUNT(), this, m_pDataList->GetCount(), pAudioData->ulAudioTime, (UINT32)m_llLastWriteTime);::fclose(f1);}

    return theErr;
}

HX_RESULT CHXAudioStream::ProcessInfo(void)
{
    HX_RESULT theErr = HXR_OK;

    m_ulSampleFrameSize = m_AudioFmt.uChannels * (m_AudioFmt.uBitsPerSample>>3) ;

    m_ulPendingAudioBytes = 0 ;

    // Calculate the number of bytes per granularity.

    // XXX wschildbach: These formulas are suspect. There is no feedback to the player
    // or audio session that this is what we assume for a size. I believe this is either
    // not needed or too complex.

    m_ulInputBytesPerGran = (ULONG32) 
		(((m_AudioFmt.uChannels * (m_AudioFmt.uBitsPerSample>>3) * m_AudioFmt.ulSamplesPerSec) 
				/ 1000.0) * m_ulGranularity);

    m_ulOutputBytesPerGran  = (ULONG32) 
		(((m_DeviceFmt.uChannels * (m_DeviceFmt.uBitsPerSample>>3) * m_DeviceFmt.ulSamplesPerSec) 
				/ 1000.0) * m_ulGranularity);

    // Make sure that number of bytes per granularity is an even number.
    m_ulInputBytesPerGran  -= m_ulInputBytesPerGran  % ((m_AudioFmt.uBitsPerSample>>3)*m_AudioFmt.uChannels);
    m_ulOutputBytesPerGran -= m_ulOutputBytesPerGran % ((m_DeviceFmt.uBitsPerSample>>3)*m_DeviceFmt.uChannels);

    if (!theErr)
    {
        // set up the mixing engine
        // XXX wschildbach
        theErr = m_pMixEngine->Init(m_AudioFmt.ulSamplesPerSec, m_DeviceFmt.ulSamplesPerSec, m_AudioFmt.uChannels, m_DeviceFmt.uChannels) ;
        if (SUCCEEDED(theErr))
            theErr = m_pMixEngine->SetSampleConverter(this) ;
        if (SUCCEEDED(theErr))
            theErr = m_pMixEngine->SetOutputBytesPerSample(m_DeviceFmt.uBitsPerSample / 8) ;
        // set the volume (somebody might have set it when we did not have an engine)
#ifdef HELIX_FEATURE_GAINTOOL
        if (SUCCEEDED(theErr))
            m_pMixEngine->SetVolume(m_pMixEngine->HXVolume2TenthOfDB(m_bMute ? HX_MIN_VOLUME : m_uVolume)) ;
#endif
    }

    if (!theErr)
    {
	m_bInited = TRUE;
	
	if (m_eState == E_STOPPED)
	{
	    m_eState = E_INITIALIZED;
	} 
    }

    /* Get the current player time to set the last write audio time 
     * If someone creates a stream mid presentation, we ask the player 
     * object for the current write time.
     */
    // set last write time to be the current playback time since
    // this is what other system components(i.e. renderers) based on 
    // fixed b#69847 - loss of push-down-worth of data =
    // m_Owner->GetLastAudioWriteTime() - m_Owner->GetCurrentPlayBackTime()
//    m_llLastWriteTime = m_Owner->GetCurrentPlayBackTime();

    // XXXRA: It is necessary to use last audio write time for any delayed
    // audio streams to work that do not involve any Pause/Rewind logic.
    // To cover the case where a source (and an audio stream) has been added
    // mid-playback by SMIL renderer which has a delay equivalent to the 
    // current playback time, it should result in a player rewind which should 
    // reset the lastaudiowrite time accordingly...so we should be able
    // to use m_Owner->GetLastAudioWriteTime() value in such a use case as well.
    // this change is required to fix PR 79161 and PR 69780.
    // Henry, PR 69847 (the reason for the earlier change) is still busted. 
    // so I am reverting this code to the original code. you will have 
    // to come up with a different fix for PR 69847 since this was clearly not 
    // the correct fix.
    m_llLastWriteTime = m_Owner->GetLastAudioWriteTime();
    m_pMixEngine->ResetTimeLineInMillis(m_llLastWriteTime) ;

    if (!theErr && m_bInited)
    {
	m_Owner->StreamInitialized(this);
    }

    return theErr;
}


/************************************************************************
 *  Method:
 *		IHXAudioStream::GetFormat
 *	Purpose:
 *  	Return the stream's audio format.
 */
HX_RESULT CHXAudioStream::GetFormat
( 
    HXAudioFormat*	pAudioFormat
)
{
    if (!m_bAudioFormatKnown)
    {
	return HXR_NOT_INITIALIZED;
    }

    pAudioFormat->uChannels       = m_AudioFmt.uChannels;
    pAudioFormat->uBitsPerSample  = m_AudioFmt.uBitsPerSample;
    pAudioFormat->ulSamplesPerSec = m_AudioFmt.ulSamplesPerSec;
    pAudioFormat->uMaxBlockSize   = m_AudioFmt.uMaxBlockSize;

    return HXR_OK;
}

/************************************************************************
 *  Method:
 *		IHXAudioStream::Setup
 *	Purpose:
 *		This is called by the player's Setup method. At this
 *		time the audio device format is set and we can now
 *		set up the streams pre-mixing buffer. This buffer
 *		contains data that has been resampled to match the
 *		audio device format.
 */
HX_RESULT CHXAudioStream::Setup(
	HXAudioFormat*	pFormat
,	ULONG32		ulGranularity
)
{
    HX_RESULT theErr = HXR_OK;

    memcpy( &m_DeviceFmt, pFormat, sizeof(HXAudioFormat) );
    m_ulGranularity = ulGranularity;

    m_bSetupDone = TRUE;
    
    /* we have all the info now.. so setup the resampler */
    if (m_bAudioFormatKnown && !m_bInited)
    {
	theErr = ProcessInfo();
    }

    return theErr;