btrfs: Attempt to fix GCC2 build.

[haiku.git] / src / apps / cortex / addons / common / AudioBuffer.cpp

/*
 * Copyright (c) 1999-2000, Eric Moon.
 * All rights reserved.
 *
 * Redistribution and use in source and binary forms, with or without
 * modification, are permitted provided that the following conditions
 * are met:
 *
 * 1. Redistributions of source code must retain the above copyright
 *    notice, this list of conditions, and the following disclaimer.
 *
 * 2. Redistributions in binary form must reproduce the above copyright
 *    notice, this list of conditions, and the following disclaimer in the
 *    documentation and/or other materials provided with the distribution.
 *
 * 3. The name of the author may not be used to endorse or promote products
 *    derived from this software without specific prior written permission.
 *
 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR "AS IS" AND ANY EXPRESS OR
 * IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES
 * OF TITLE, NON-INFRINGEMENT, MERCHANTABILITY AND FITNESS FOR A PARTICULAR
 * PURPOSE ARE DISCLAIMED.  IN NO EVENT SHALL THE AUTHOR BE LIABLE FOR ANY
 * DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES
 * (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES;
 * LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED
 * AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR
 * TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
 * OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
 */


// AudioBuffer.cpp
// e.moon 31mar99
//

#include <Buffer.h>
#include <Debug.h>
#include <RealtimeAlloc.h>
#include "AudioBuffer.h"

#include <cmath>
#include <cstring>

#include "audio_buffer_tools.h"

const media_raw_audio_format AudioBuffer::defaultFormat = {
        44100.0,
        2,
        media_raw_audio_format::B_AUDIO_FLOAT,
        media_raw_audio_format::wildcard.byte_order,
        media_raw_audio_format::wildcard.buffer_size
};

// -------------------------------------------------------- //
// ctor/dtor/accessors
// -------------------------------------------------------- //

AudioBuffer::AudioBuffer(
        const media_raw_audio_format& format,
        rtm_pool* pFromPool) :
        
        RawBuffer(
                (format.format & 0x0f) * format.channel_count,
                0,
                true,
                pFromPool),             
        m_format(format) {}

AudioBuffer::AudioBuffer(
        const media_raw_audio_format& format,
        uint32 frames,
        bool bCircular,
        rtm_pool* pFromPool) :

        RawBuffer(
                (format.format & 0x0f) * format.channel_count,
                0,
                bCircular,
                pFromPool),
        m_format(format) {
        
        resize(frames);
}

AudioBuffer::AudioBuffer(
        const media_raw_audio_format& format,
        void* pData,
        uint32 frames,
        bool bCircular,
        rtm_pool* pFromPool) :
        
        RawBuffer(
                pData,
                (format.format & 0x0f) * format.channel_count,
                frames,
                bCircular,
                pFromPool) {}

AudioBuffer::AudioBuffer(
        const media_raw_audio_format& format,
        BBuffer* pBuffer,
        bool bCircular) :
        
        RawBuffer(),
        m_format(format)
        
        {

        if(pBuffer->Header()->type != B_MEDIA_RAW_AUDIO)
                return;
        
        // reference it:
        m_pData = pBuffer->Data();
        m_frameSize = (m_format.format & 0x0f) * m_format.channel_count;
        m_frames = pBuffer->Header()->size_used / m_frameSize;
        m_allocatedSize = 0;
        m_bOwnData = false;
        m_bCircular = bCircular;
}

// generate a reference (point) to the target's buffer
AudioBuffer::AudioBuffer(const AudioBuffer& clone) :
        RawBuffer(clone),
        m_format(clone.m_format) {}

AudioBuffer& AudioBuffer::operator=(const AudioBuffer& clone) {
        RawBuffer::operator=(clone);
        m_format = clone.m_format;
        return *this;
}

AudioBuffer::~AudioBuffer() {}

// format access

void AudioBuffer::setFormat(const media_raw_audio_format& format) {
        m_format = format;
}

const media_raw_audio_format& AudioBuffer::format() const {
        return m_format;
}

// extra adoption support

void AudioBuffer::adopt(
        const media_raw_audio_format& format,
        void* pData,
        uint32 frames,
        bool bCircular,
        rtm_pool* pFromPool) {
        
        // clean up myself first
        free();
        
        // reference
        operator=(AudioBuffer(format, pData, frames, bCircular, pFromPool));

        // mark ownership
        m_bOwnData = true;
}

// as with RawBuffer::adopt(), returns false if the target
// doesn't own its buffer, but references it anyway

bool AudioBuffer::adopt(AudioBuffer& target) {
        m_format = target.m_format;
        return RawBuffer::adopt(target);
}

// -------------------------------------------------------- //
// operations
// -------------------------------------------------------- //

// test for format equivalence against target buffer
// (ie. determine whether any conversion would be necessary
//  for copy/mix operations)

bool AudioBuffer::formatSameAs(const AudioBuffer& target) const {
        return
                m_format.format == target.m_format.format &&
                m_format.channel_count == target.m_format.channel_count;
}

// copy to target audio buffer, applying any necessary
// format conversions.  behaves like rawCopyTo().
        
uint32 AudioBuffer::copyTo(
        AudioBuffer& target,
        uint32* pioFromFrame,
        uint32* pioTargetFrame,
        uint32 frames) const {

        // simplest case:       
        if(formatSameAs(target))
                return rawCopyTo(target, pioFromFrame, pioTargetFrame, frames);

        // sanity checks
        ASSERT(m_pData);
        ASSERT(m_frames);
        ASSERT(target.m_pData);

        // figure byte offsets & sizes
        uint32 fromOffset = *pioFromFrame * m_frameSize;
        uint32 targetOffset = *pioTargetFrame * m_frameSize;
        
        uint32 size = m_frames * m_frameSize;   
        uint32 targetSize = target.m_frames * target.m_frameSize;

        // figure number of samples to convert
        uint32 toCopy = frames * m_format.channel_count;        
        if(target.m_bCircular) {
                if(toCopy > targetSize)
                        toCopy = targetSize;
        } else {
                if(toCopy > (targetSize-targetOffset))
                        toCopy = (targetSize-targetOffset);
        }
        uint32 remaining = toCopy;
        
        uint32 sampleSize = m_frameSize / m_format.channel_count;

        // convert and copy a sample at a time
        for(; remaining; remaining -= sampleSize) {
                convert_sample(
                        (void*) ((int8*)m_pData + fromOffset),
                        (void*) ((int8*)target.m_pData + targetOffset),
                        m_format.format,
                        target.m_format.format);
                
                fromOffset += sampleSize;
                if(fromOffset == size)
                        fromOffset = 0;
                
                targetOffset += sampleSize;
                if(targetOffset == targetSize)
                        targetOffset = 0;
        }
        
        // write new offsets
        *pioFromFrame = fromOffset / m_frameSize;
        *pioTargetFrame = targetOffset / m_frameSize;
        
        return toCopy;
}

uint32 AudioBuffer::copyTo(
        AudioBuffer& target,
        uint32* pioFromFrame,
        uint32* pioTargetFrame) const {
        
        return copyTo(target, pioFromFrame, pioTargetFrame, m_frames);
}

// mix to target audio buffer, applying any necessary
// format conversions.  behaves like rawCopyTo().
        
uint32 AudioBuffer::mixTo(
        AudioBuffer& target,
        uint32* pioFromFrame,
        uint32* pioTargetFrame,
        uint32 frames,
        float fGain /*=1.0*/) const { return 0; } //nyi         
        
// calculate minimum & maximum peak levels
// (converted/scaled to given type if necessary)
// pMax and pMin must point to arrays with enough room
// for one value per channel.  existing array values aren't
// cleared first.
//
// (if pMin isn't provided, the maximum absolute levels will
// be written to pMax)

void AudioBuffer::findMin(float* pMin, uint32* pAt /*=0*/) const {
        findMin(0, m_frames, pMin, pAt);
}

uint32 AudioBuffer::findMin(uint32 fromFrame, uint32 frameCount,
        float* pMin, uint32* pAt /*=0*/) const {

        size_t channels = m_format.channel_count;
        size_t samples = m_frames * channels;
        size_t bytesPerSample = m_format.format & 0x0f;

        size_t firstSample = fromFrame * channels;
        size_t remaining = frameCount * channels;

        if(!m_pData)
                return fromFrame;
                
        int8* pCur = (int8*)m_pData + (firstSample * bytesPerSample);

        uint32 n;

        if(pAt) {
                // reset pAt
                for(n = 0; n < channels; n++)
                        pAt[n] = UINT32_MAX;
        }
        
        // find minimum for each channel
        for(
                n = firstSample;
                remaining;
                remaining--, n++, pCur += bytesPerSample) {
                
                // wrap around to start of buffer?
                if(n == samples) {
                        pCur = (int8*)m_pData;
                        n = 0;
                }
                                
                float fCur = 0;
                convert_sample(pCur, fCur, m_format.format);
                
                if(fCur < pMin[n % channels]) {
                        pMin[n % channels] = fCur;
                        if(pAt)
                                pAt[n % channels] = n / channels;
                }
        }
        
        // return current frame
        return n / channels;
}

void AudioBuffer::findMax(float* pMax, uint32* pAt /*=0*/) const {
        findMax(0, m_frames, pMax, pAt);
}

uint32 AudioBuffer::findMax(uint32 fromFrame, uint32 frameCount,
        float* pMax, uint32* pAt /*=0*/) const {

        size_t channels = m_format.channel_count;
        size_t samples = m_frames * channels;
        size_t bytesPerSample = m_format.format & 0x0f;

        size_t firstSample = fromFrame * channels;
        size_t remaining = frameCount * channels;

        if(!m_pData)
                return fromFrame;
                
        int8* pCur = (int8*)m_pData + (firstSample * bytesPerSample);

        uint32 n;

        if(pAt) {
                // reset pAt
                for(n = 0; n < channels; n++)
                        pAt[n] = UINT32_MAX;
        }
        
        // find minimum for each channel
        for(
                n = firstSample;
                remaining;
                remaining--, n++, pCur += bytesPerSample) {
                
                // wrap around to start of buffer?
                if(n == samples) {
                        pCur = (int8*)m_pData;
                        n = 0;
                }
                                
                float fCur = 0;
                convert_sample(pCur, fCur, m_format.format);
                
                if(fCur > pMax[n % channels]) {
                        pMax[n % channels] = fCur;
                        if(pAt)
                                pAt[n % channels] = n / channels;
                }
        }
        
        // return current frame
        return n / channels;
}

void AudioBuffer::findPeaks(float* pPeaks, uint32* pAt /*=0*/) const {
        findPeaks(0, m_frames, pPeaks, pAt);
}

uint32 AudioBuffer::findPeaks(uint32 fromFrame, uint32 frameCount,
        float* pPeaks, uint32* pAt) const {
        
        size_t channels = m_format.channel_count;
        size_t samples = m_frames * channels;
        size_t bytesPerSample = m_format.format & 0x0f;

        size_t firstSample = fromFrame * channels;
        size_t remaining = frameCount * channels;

        if(!m_pData)
                return fromFrame;
        int8* pCur = (int8*)m_pData + (firstSample * bytesPerSample);

        uint32 n;

        if(pAt) {
                // reset pAt
                for(n = 0; n < channels; n++)
                        pAt[n] = UINT32_MAX;
        }

        // find peaks in both directions
        for(
                n = firstSample;
                remaining;
                remaining--, n++, pCur += bytesPerSample) {
                
                // wrap around to start of buffer?
                if(n == samples) {
                        pCur = (int8*)m_pData;
                        n = 0;
                }
                                
                float fCur = 0;
                convert_sample(pCur, fCur, m_format.format);
                
                if(fabs(fCur) > pPeaks[n % channels]) {
                        pPeaks[n % channels] = fCur;
                        if(pAt)
                                pAt[n % channels] = n / channels;
                }
        }
        
        // return current frame
        return n / channels;
}
        
// find average level
// (converted/scaled as necessary)
// pAverage must point to an array with enough room
// for one value per channel.
        
void AudioBuffer::average(float* pAverage) const {
        average(0, m_frames, pAverage);
}

uint32 AudioBuffer::average(uint32 fromFrame, uint32 frameCount,
        float* pAverage) const {
        
        size_t channels = m_format.channel_count;
        size_t samples = m_frames * channels;
        size_t bytesPerSample = m_format.format & 0x0f;

        size_t firstSample = fromFrame * channels;
        size_t remaining = frameCount * channels;

        if(!m_pData)
                return fromFrame;
        int8* pCur = (int8*)m_pData + (firstSample * bytesPerSample);

        // clear averages
        memset(pAverage, 0, sizeof(float)*channels);
        
        // calculate
        uint32 n;
        for(
                n = firstSample;
                remaining;
                remaining--, n++, pCur += bytesPerSample) {
                        
                // wrap around to start of buffer
                if(n == samples) {
                        pCur = (int8*)m_pData;
                        n = 0;
                }
                                
                float fCur = 0;
                convert_sample(pCur, fCur, m_format.format);

                pAverage[n%channels] += fCur;
        }

        for(uint32 n = 0; n < channels; n++)
                pAverage[n] /= frameCount;      
        
        // return current frame
        return n / channels;
}

// -------------------------------------------------------- //


// END -- AudioBuffer.h --