⭐ 欢迎来到虫虫下载站! | 📦 资源下载 📁 资源专辑 ℹ️ 关于我们
⭐ 虫虫下载站

📄 output.cpp

📁 Windows CE 6.0 BSP for the Beagle Board.
💻 CPP
📖 第 1 页 / 共 2 页
字号:
//
// Copyright (c) Microsoft Corporation.  All rights reserved.
//
//
// Use of this sample source code is subject to the terms of the Microsoft
// license agreement under which you licensed this sample source code. If
// you did not accept the terms of the license agreement, you are not
// authorized to use this sample source code. For the terms of the license,
// please see the license agreement between you and Microsoft or, if applicable,
// see the LICENSE.RTF on your install media or the root of your tools installation.
// THE SAMPLE SOURCE CODE IS PROVIDED "AS IS", WITH NO WARRANTIES.
//
// Portions Copyright (c) Texas Instruments.  All rights reserved.
//
//------------------------------------------------------------------------------
//

#include "wavemain.h"


//------------------------------------------------------------------------------
//
//  Function: Open
//  
//  Open output stream context
//

HRESULT 
OutputStreamContext::Open(DeviceContext *pDeviceContext, LPWAVEOPENDESC lpWOD, DWORD dwFlags)
{
    HRESULT Result;

    Result = WaveStreamContext::Open(pDeviceContext, lpWOD, dwFlags);

    if (Result==MMSYSERR_NOERROR)
    {
        // Note: Output streams should be initialized in the run state.
        Run();
    }

    return Result;
}

//------------------------------------------------------------------------------
//
//  Function: Reset()
//  
//  eset output stream context
//

DWORD 
OutputStreamContext::Reset()
{
    HRESULT Result;

    Result = WaveStreamContext::Reset();

    if (Result==MMSYSERR_NOERROR)
    {
        // Note: Output streams should be reset to the run state.
        Run();
    }

    return Result;
};

//------------------------------------------------------------------------------
//
//  Function: SetRate
//  
//  Init m_DeltaT with (SampleRate/HWSampleRate) calculated in the appropriate fixed point form.
//  Note that we need to hold the result in a 64-bit value until we're done shifting,
//  since the result of the multiply will overflow 32 bits for sample rates greater than
//  or equal to the hardware's sample rate.
//

DWORD 
OutputStreamContext::SetRate(DWORD dwMultiplier)
{
    UINT64 Delta;

    m_dwMultiplier = dwMultiplier;

    // The following code makes a bunch of assumptions on why we don't overflow a 64-bit value:
    // nSamplesPerSec is < 20 bits (e.g. no input data rates greater than ~1 MHz!)
    // m_dwMultiplier is < 24 bits (e.g. no speedup beyond 256x)
    // INVSAMPLERATE is < 20 bits (e.g. no playback rates less than 4kHz)

    Delta = (m_WaveFormat.nSamplesPerSec * m_dwMultiplier);
    Delta = (Delta * INVSAMPLERATE);    // < Note: These two lines are out of order to preserve accuracy
    Delta >>= 16;                       // <
    Delta >>= DELTAINT;  // Convert to x.x format (e.g. 17.15)

    m_DeltaT = (DWORD)Delta;
    return MMSYSERR_NOERROR;
}

//------------------------------------------------------------------------------
//
//  Function: Render
//  
//  Originally, this code used to be in each renderer, and each one would call 
//  GetNextBuffer as needed. Pulling this code out of each low level renderer 
//  allows the inner loop to be in a leaf routine (ie no subroutine calls out 
//  of that routine), which helps the compiler optimize the inner loop.
//

PBYTE 
WaveStreamContext::Render(PBYTE pBuffer, PBYTE pBufferEnd, PBYTE pBufferLast)
{
    if (!m_bRunning || !m_lpCurrData)
    {
        return pBuffer;
    }

    while (pBuffer < pBufferEnd)
    {
        while (m_lpCurrData>=m_lpCurrDataEnd)
        {
            if (!GetNextBuffer())
            {
                return pBuffer;
            }
        }

        _try
        {
            pBuffer = Render2(pBuffer,pBufferEnd,pBufferLast);
        }
        _except (EXCEPTION_EXECUTE_HANDLER)
        {
            DEBUGMSG(ZONE_ERROR, (L"WaveStreamContext::Render: "
                L"EXCEPTION IN IST for stream 0x%x, buffer 0x%x!!!!\r\n", this, m_lpCurrData
            ));
            m_lpCurrData=m_lpCurrDataEnd; // Pretend we finished reading the application buffer
        }
    }

    return pBuffer;
}

//------------------------------------------------------------------------------
//
//  Function: OutputStreamContextM8::Render2
//  
//  StreamContext to reformat Mono/8bit samples to 16bit/stereo.
//  Also resamples to 44.1khz and checks saturation
//

PBYTE 
OutputStreamContextM8::Render2(PBYTE pBuffer, PBYTE pBufferEnd, PBYTE pBufferLast)
{
    LONG CurrT = m_CurrT;
    LONG DeltaT = m_DeltaT;
    LONG CurrSamp0 = m_CurrSamp[0];
    LONG PrevSamp0 = m_PrevSamp[0];
    PBYTE pCurrData = m_lpCurrData;
    PBYTE pCurrDataEnd = m_lpCurrDataEnd;
    LONG fxpGain = m_fxpGain;

    while (pBuffer < pBufferEnd)
    {
        while (CurrT >= DELTA_OVERFLOW)
        {
            if (pCurrData>=pCurrDataEnd)
            {
                goto Exit;
            }

            CurrT -= DELTA_OVERFLOW;

            PrevSamp0 = CurrSamp0;

            PPCM_SAMPLE pSampleSrc = (PPCM_SAMPLE)pCurrData;
            CurrSamp0 = (LONG)pSampleSrc->m8.sample;
            CurrSamp0 = (CurrSamp0 - 128) << 8;
            pCurrData+=1;
        }

        LONG OutSamp0;
        OutSamp0 = PrevSamp0 + (((CurrSamp0 - PrevSamp0) * CurrT) >> DELTAFRAC);
        OutSamp0 = (OutSamp0 * fxpGain) >> VOLSHIFT;
        CurrT += DeltaT;

#if (OUTCHANNELS==2)
        LONG OutSamp1;
        OutSamp1=OutSamp0;
        if (pBuffer < pBufferLast)
        {
            OutSamp0 += ((HWSAMPLE *)pBuffer)[0];
            OutSamp1 += ((HWSAMPLE *)pBuffer)[1];
#if USE_MIX_SATURATE
            // Handle saturation
            if (OutSamp0>AUDIO_SAMPLE_MAX)
            {
                OutSamp0=AUDIO_SAMPLE_MAX;
            }
            else if (OutSamp0<AUDIO_SAMPLE_MIN)
            {
                OutSamp0=AUDIO_SAMPLE_MIN;
            }
            if (OutSamp1>AUDIO_SAMPLE_MAX)
            {
                OutSamp1=AUDIO_SAMPLE_MAX;
            }
            else if (OutSamp1<AUDIO_SAMPLE_MIN)
            {
                OutSamp1=AUDIO_SAMPLE_MIN;
            }
#endif
        }
        ((HWSAMPLE *)pBuffer)[0] = (HWSAMPLE)OutSamp0;
        ((HWSAMPLE *)pBuffer)[1] = (HWSAMPLE)OutSamp1;
        pBuffer += 2*sizeof(HWSAMPLE);
#else
        if (pBuffer < pBufferLast)
        {
            OutSamp0 += ((HWSAMPLE *)pBuffer)[0];
#if USE_MIX_SATURATE
            // Handle saturation
            if (OutSamp0>AUDIO_SAMPLE_MAX)
            {
                OutSamp0=AUDIO_SAMPLE_MAX;
            }
            else if (OutSamp0<AUDIO_SAMPLE_MIN)
            {
                OutSamp0=AUDIO_SAMPLE_MIN;
            }
#endif
        }
        ((HWSAMPLE *)pBuffer)[0] = (HWSAMPLE)OutSamp0;
        pBuffer += sizeof(HWSAMPLE);
#endif
    }

    Exit:

    m_dwByteCount += (pCurrData - m_lpCurrData);
    m_lpCurrData = pCurrData;
    m_CurrT = CurrT;
    m_PrevSamp[0] = PrevSamp0;
    m_CurrSamp[0] = CurrSamp0;
    return pBuffer;
}

//------------------------------------------------------------------------------
//
//  Function: OutputStreamContextM16::Render2
//  
//  StreamContext to reformat Mono/16bit samples to 16bit/stereo.
//  Also resamples to 44.1khz and checks saturation
//

PBYTE 
OutputStreamContextM16::Render2(PBYTE pBuffer, PBYTE pBufferEnd, PBYTE pBufferLast)
{
    LONG CurrT = m_CurrT;
    LONG DeltaT = m_DeltaT;
    LONG CurrSamp0 = m_CurrSamp[0];
    LONG PrevSamp0 = m_PrevSamp[0];
    PBYTE pCurrData = m_lpCurrData;
    PBYTE pCurrDataEnd = m_lpCurrDataEnd;
    LONG fxpGain = m_fxpGain;
    LONG OutSamp0;

    while (pBuffer < pBufferEnd)
    {
        while (CurrT >= DELTA_OVERFLOW)
        {
            if (pCurrData>=pCurrDataEnd)
            {
                goto Exit;
            }

            CurrT -= DELTA_OVERFLOW;

            PrevSamp0 = CurrSamp0;

            PPCM_SAMPLE pSampleSrc = (PPCM_SAMPLE)pCurrData;
            CurrSamp0 = (LONG)pSampleSrc->m16.sample;
            pCurrData+=2;
        }

        OutSamp0 = PrevSamp0 + (((CurrSamp0 - PrevSamp0) * CurrT) >> DELTAFRAC);
        OutSamp0 = (OutSamp0 * fxpGain) >> VOLSHIFT;
        CurrT += DeltaT;

//        DEBUGMSG(ZONE_VERBOSE, (TEXT("PrevSamp0=0x%x, CurrSamp0=0x%x, CurrT=0x%x, OutSamp0=0x%x\r\n"), PrevSamp0,CurrSamp0,CurrT,OutSamp0));

#if (OUTCHANNELS==2)
        LONG OutSamp1;
        OutSamp1=OutSamp0;
        if (pBuffer < pBufferLast)
        {
            OutSamp0 += ((HWSAMPLE *)pBuffer)[0];
            OutSamp1 += ((HWSAMPLE *)pBuffer)[1];
#if USE_MIX_SATURATE
            // Handle saturation
            if (OutSamp0>AUDIO_SAMPLE_MAX)
            {
                OutSamp0=AUDIO_SAMPLE_MAX;
            }
            else if (OutSamp0<AUDIO_SAMPLE_MIN)
            {
                OutSamp0=AUDIO_SAMPLE_MIN;
            }
            if (OutSamp1>AUDIO_SAMPLE_MAX)
            {
                OutSamp1=AUDIO_SAMPLE_MAX;
            }
            else if (OutSamp1<AUDIO_SAMPLE_MIN)
            {
                OutSamp1=AUDIO_SAMPLE_MIN;
            }
#endif
        }
        ((HWSAMPLE *)pBuffer)[0] = (HWSAMPLE)OutSamp0;
        ((HWSAMPLE *)pBuffer)[1] = (HWSAMPLE)OutSamp1;
        pBuffer += 2*sizeof(HWSAMPLE);
#else
        if (pBuffer < pBufferLast)
        {
            OutSamp0 += ((HWSAMPLE *)pBuffer)[0];
#if USE_MIX_SATURATE
            // Handle saturation
            if (OutSamp0>AUDIO_SAMPLE_MAX)
            {
                OutSamp0=AUDIO_SAMPLE_MAX;
            }
            else if (OutSamp0<AUDIO_SAMPLE_MIN)
            {
                OutSamp0=AUDIO_SAMPLE_MIN;
            }
#endif
        }
        ((HWSAMPLE *)pBuffer)[0] = (HWSAMPLE)OutSamp0;
        pBuffer += sizeof(HWSAMPLE);
#endif
    }

    Exit:
    m_dwByteCount += (pCurrData - m_lpCurrData);
    m_lpCurrData = pCurrData;
    m_CurrT = CurrT;
    m_PrevSamp[0] = PrevSamp0;
    m_CurrSamp[0] = CurrSamp0;
    return pBuffer;
}

⌨️ 快捷键说明

复制代码 Ctrl + C
搜索代码 Ctrl + F
全屏模式 F11
切换主题 Ctrl + Shift + D
显示快捷键 ?
增大字号 Ctrl + =
减小字号 Ctrl + -