C++ sampleRate函数代码示例

PassRefPtr<MediaStreamAudioSourceNode> AudioContext::createMediaStreamSource(MediaStream* mediaStream, ExceptionState& es)
{
    ASSERT(mediaStream);
    if (!mediaStream) {
        es.throwDOMException(InvalidStateError);
        return 0;
    }

    ASSERT(isMainThread());
    lazyInitialize();

    AudioSourceProvider* provider = 0;

    MediaStreamTrackVector audioTracks = mediaStream->getAudioTracks();

    // FIXME: get a provider for non-local MediaStreams (like from a remote peer).
    for (size_t i = 0; i < audioTracks.size(); ++i) {
        RefPtr<MediaStreamTrack> localAudio = audioTracks[i];
        if (localAudio->component()->audioSourceProvider()) {
            provider = localAudio->component()->audioSourceProvider();
            break;
        }
    }

    RefPtr<MediaStreamAudioSourceNode> node = MediaStreamAudioSourceNode::create(this, mediaStream, provider);

    // FIXME: Only stereo streams are supported right now. We should be able to accept multi-channel streams.
    node->setFormat(2, sampleRate());

    refNode(node.get()); // context keeps reference until node is disconnected
    return node;
}

void Minim::Delay::channelCountChanged()
{
	if ( delayBuffer )
	{
		delete[] delayBuffer;
        delayBuffer = NULL;
	}
	
	if ( sampleRate() > 0 )
	{
		maxBufferSize = (int)( maxDelayTime*sampleRate()*getAudioChannelCount() );
		delayBuffer = new float[maxBufferSize];
		memset(delayBuffer, 0, sizeof(float)*maxBufferSize);
		bufferSizeChanged();
	}
}

ssize_t AudioStreamInStub::read(void* buffer, ssize_t bytes)
{
    // fake timing for audio input
    usleep(bytes * 1000000 / sizeof(int16_t) / AudioSystem::popCount(channels()) / sampleRate());
    memset(buffer, 0, bytes);
    return bytes;
}

status_t AudioStreamInMotorola::dump(int fd, const Vector<String16>& args)
{
    const size_t SIZE = 256;
    char buffer[SIZE];
    String8 result;
    result.append("AudioStreamInMotorola::dump\n");
    snprintf(buffer, SIZE, "\tsample rate: %d\n", sampleRate());
    result.append(buffer);
    snprintf(buffer, SIZE, "\tbuffer size: %d\n", bufferSize());
    result.append(buffer);
    snprintf(buffer, SIZE, "\tchannels: %d\n", channels());
    result.append(buffer);
    snprintf(buffer, SIZE, "\tformat: %d\n", format());
    result.append(buffer);
    snprintf(buffer, SIZE, "\tmHardware: %p\n", mHardware);
    result.append(buffer);
    snprintf(buffer, SIZE, "\tmFd count: %d\n", mFd);
    result.append(buffer);
    snprintf(buffer, SIZE, "\tmStandby: %d\n", mStandby);
    result.append(buffer);
    snprintf(buffer, SIZE, "\tmRetryCount: %d\n", mRetryCount);
    result.append(buffer);
    ::write(fd, result.string(), result.size());
    return NO_ERROR;
}

double AudioBufferSourceNode::totalPitchRate()
{
    double dopplerRate = 1.0;
    if (m_pannerNode)
        dopplerRate = m_pannerNode->dopplerRate();

    // Incorporate buffer's sample-rate versus AudioContext's sample-rate.
    // Normally it's not an issue because buffers are loaded at the AudioContext's sample-rate, but we can handle it in any case.
    double sampleRateFactor = 1.0;
    if (buffer())
        sampleRateFactor = buffer()->sampleRate() / sampleRate();

    double basePitchRate = playbackRate()->value();

    double totalRate = dopplerRate * sampleRateFactor * basePitchRate;

    totalRate = std::max(-MaxRate, std::min(MaxRate, totalRate));

    bool isTotalRateValid = !std::isnan(totalRate) && !std::isinf(totalRate);
    ASSERT(isTotalRateValid);
    if (!isTotalRateValid)
        totalRate = 1.0;

    return totalRate;
}

double AudioBufferSourceNode::totalPitchRate()
{
    double dopplerRate = 1.0;
    if (m_pannerNode)
        dopplerRate = m_pannerNode->dopplerRate();

    // Incorporate buffer's sample-rate versus AudioContext's sample-rate.
    // Normally it's not an issue because buffers are loaded at the AudioContext's sample-rate, but we can handle it in any case.
    double sampleRateFactor = 1.0;
    if (buffer())
        sampleRateFactor = buffer()->sampleRate() / sampleRate();

    double basePitchRate = playbackRate()->value();

    double totalRate = dopplerRate * sampleRateFactor * basePitchRate;

    // Sanity check the total rate.  It's very important that the resampler not get any bad rate values.
    totalRate = std::max(0.0, totalRate);
    if (!totalRate)
        totalRate = 1; // zero rate is considered illegal
    totalRate = std::min(MaxRate, totalRate);

    bool isTotalRateValid = !std::isnan(totalRate) && !std::isinf(totalRate);
    ASSERT(isTotalRateValid);
    if (!isTotalRateValid)
        totalRate = 1.0;

    return totalRate;
}

void MediaStreamAudioSourceNode::setFormat(size_t numberOfChannels, float sourceSampleRate)
{
    if (numberOfChannels != m_sourceNumberOfChannels || sourceSampleRate != sampleRate()) {
        // The sample-rate must be equal to the context's sample-rate.
        if (!numberOfChannels || numberOfChannels > AudioContext::maxNumberOfChannels() || sourceSampleRate != sampleRate()) {
            // process() will generate silence for these uninitialized values.
            LOG(Media, "MediaStreamAudioSourceNode::setFormat(%u, %f) - unhandled format change", static_cast<unsigned>(numberOfChannels), sourceSampleRate);
            m_sourceNumberOfChannels = 0;
            return;
        }

        // Synchronize with process().
        MutexLocker locker(m_processLock);

        m_sourceNumberOfChannels = numberOfChannels;

        {
            // The context must be locked when changing the number of output channels.
            AudioContext::AutoLocker contextLocker(context());

            // Do any necesssary re-configuration to the output's number of channels.
            output(0)->setNumberOfChannels(numberOfChannels);
        }
    }
}

void Minim::MoogFilter::uGenerate( float * out, const int numChannels )
{
	// Set coefficients given frequency & resonance [0.0...1.0]
	float t1, t2; // temporary buffers
	float normFreq = frequency.getLastValue() / ( sampleRate() * 0.5f );
	float rez = constrain( resonance.getLastValue(), 0.f, 1.f );
	
	float q = 1.0f - normFreq;
	float p = normFreq + 0.8f * normFreq * q;
	float f = p + p - 1.0f;
	q = rez * ( 1.0f + 0.5f * q * ( 1.0f - q + 5.6f * q * q ) );
	
	const float* input = audio.getLastValues();
	
	for ( int i = 0; i < numChannels; ++i )
	{
		// Filter (in [-1.0...+1.0])
		float* b = coeff[i];
		float in = constrain( input[i], -1, 1 ); // hard clip
		//float in = atanf(input[i]) * M_2_PI; // soft clip
		
		in -= q * b[4]; // feedback
		
		t1 = b[1];
		b[1] = ( in + b[0] ) * p - b[1] * f;
		
		t2 = b[2];
		b[2] = ( b[1] + t1 ) * p - b[2] * f;
		
		t1 = b[3];
		b[3] = ( b[2] + t2 ) * p - b[3] * f;
		
		b[4] = ( b[3] + t1 ) * p - b[4] * f;
		b[4] = b[4] - b[4] * b[4] * b[4] * 0.166667f; // clipping
        
        // inelegantly squash denormals
        if ( isnan(b[4]) )
        {
            memset(b, 0, sizeof(float)*5);
        }
		
		b[0] = in;
        
        switch( type )
        {
            case HP:
                out[i] = in - b[4];
                break;
                
            case LP:
                out[i] = b[4];
                break;
                
            case BP:
                out[i] = 3.0f * (b[3] - b[4]);
                break;
        }
	}
}

/*!
    Returns the number of microseconds represented by \a bytes in this format.

    Returns 0 if this format is not valid->

    Note that some rounding may occur if \a bytes is not an exact multiple
    of the number of bytes per frame.

    \sa bytesForDuration()
*/
qint64 AudioFormat::durationForBytes(qint32 bytes) const
{
    if (!isValid() || bytes <= 0)
        return 0;

    // We round the byte count to ensure whole frames
    return qint64(kHz * (bytes / bytesPerFrame())) / sampleRate();
}

void ADSREnvOld::setReleaseTime( FLOAT time )
{
	if( time < 0.0 ) {
		negativeRatesError();
		time *= -1;
	}
	releaseRate_ = sustainLevel_ / ( time * sampleRate() );
}

void DynamicsCompressorHandler::initialize()
{
    if (isInitialized())
        return;

    AudioHandler::initialize();
    m_dynamicsCompressor = wrapUnique(new DynamicsCompressor(sampleRate(), defaultNumberOfOutputChannels));
}

void AudioContext::decodeAudioData(ArrayBuffer* audioData, PassRefPtr<AudioBufferCallback> successCallback, PassRefPtr<AudioBufferCallback> errorCallback, ExceptionState& es)
{
    if (!audioData) {
        es.throwDOMException(SyntaxError);
        return;
    }
    m_audioDecoder.decodeAsync(audioData, sampleRate(), successCallback, errorCallback);
}

double HRTFPanner::tailTime() const {
  // Because HRTFPanner is implemented with a DelayKernel and a FFTConvolver,
  // the tailTime of the HRTFPanner is the sum of the tailTime of the
  // DelayKernel and the tailTime of the FFTConvolver, which is
  // MaxDelayTimeSeconds and fftSize() / 2, respectively.
  return MaxDelayTimeSeconds +
         (fftSize() / 2) / static_cast<double>(sampleRate());
}

void AudioContext::decodeAudioData(ArrayBuffer* audioData, PassRefPtr<AudioBufferCallback> successCallback, PassRefPtr<AudioBufferCallback> errorCallback, ExceptionCode& ec)
{
    if (!audioData) {
        ec = SYNTAX_ERR;
        return;
    }
    m_audioDecoder.decodeAsync(audioData, sampleRate(), successCallback, errorCallback);
}

ssize_t AudioStreamInStub::read(void* buffer, ssize_t bytes)
{
    // fake timing for audio input
    usleep(bytes * 1000000 / sizeof(int16_t) /
           audio_channel_count_from_in_mask(channels()) / sampleRate());
    memset(buffer, 0, bytes);
    return bytes;
}