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IN NO EVENT SHALL APPLE BE LIABLE FOR ANY SPECIAL, INDIRECT, INCIDENTAL OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) ARISING IN ANY WAY OUT OF THE USE, REPRODUCTION, MODIFICATION AND/OR DISTRIBUTION OF THE APPLE SOFTWARE, HOWEVER CAUSED AND WHETHER UNDER THEORY OF CONTRACT, TORT (INCLUDING NEGLIGENCE), STRICT LIABILITY OR OTHERWISE, EVEN IF APPLE HAS BEEN ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. */ #include "SampleAudioEngine.h" // The function clipOutputSamples() is called to clip and convert samples from the float mix buffer into the actual // hardware sample buffer. The samples to be clipped, are guaranteed not to wrap from the end of the buffer to the // beginning. // This implementation is very inefficient, but illustrates the clip and conversion process that must take place. // Each floating-point sample must be clipped to a range of -1.0 to 1.0 and then converted to the hardware buffer // format // The parameters are as follows: // mixBuf - a pointer to the beginning of the float mix buffer - its size is based on the number of sample frames // times the number of channels for the stream // sampleBuf - a pointer to the beginning of the hardware formatted sample buffer - this is the same buffer passed // to the IOAudioStream using setSampleBuffer() // firstSampleFrame - this is the index of the first sample frame to perform the clipping and conversion on // numSampleFrames - the total number of sample frames to clip and convert // streamFormat - the current format of the IOAudioStream this function is operating on // audioStream - the audio stream this function is operating on IOReturn SampleAudioEngine::clipOutputSamples(const void *mixBuf, void *sampleBuf, UInt32 firstSampleFrame, UInt32 numSampleFrames, const IOAudioStreamFormat *streamFormat, IOAudioStream *audioStream) { UInt32 sampleIndex, maxSampleIndex; float *floatMixBuf; SInt16 *outputBuf; // Start by casting the void * mix and sample buffers to the appropriate types - float * for the mix buffer // and SInt16 * for the sample buffer (because our sample hardware uses signed 16-bit samples) floatMixBuf = (float *)mixBuf; outputBuf = (SInt16 *)sampleBuf; // We calculate the maximum sample index we are going to clip and convert // This is an index into the entire sample and mix buffers maxSampleIndex = (firstSampleFrame + numSampleFrames) * streamFormat->fNumChannels; // Loop through the mix/sample buffers one sample at a time and perform the clip and conversion operations for (sampleIndex = (firstSampleFrame * streamFormat->fNumChannels); sampleIndex < maxSampleIndex; sampleIndex++) { float inSample; // Fetch the floating point mix sample inSample = floatMixBuf[sampleIndex]; // Clip that sample to a range of -1.0 to 1.0 // A softer clipping operation could be done here if (inSample > 1.0) { inSample = 1.0; } else if (inSample < -1.0) { inSample = -1.0; } // Scale the -1.0 to 1.0 range to the appropriate scale for signed 16-bit samples and then // convert to SInt16 and store in the hardware sample buffer if (inSample >= 0) { outputBuf[sampleIndex] = (SInt16) (inSample * 32767.0); } else { outputBuf[sampleIndex] = (SInt16) (inSample * 32768.0); } } return kIOReturnSuccess; } // The function convertInputSamples() is responsible for converting from the hardware format // in the input sample buffer to float samples in the destination buffer and scale the samples // to a range of -1.0 to 1.0. This function is guaranteed not to have the samples wrapped // from the end of the buffer to the beginning. // This function only needs to be implemented if the device has any input IOAudioStreams // This implementation is very inefficient, but illustrates the conversion and scaling that needs to take place. // The parameters are as follows: // sampleBuf - a pointer to the beginning of the hardware formatted sample buffer - this is the same buffer passed // to the IOAudioStream using setSampleBuffer() // destBuf - a pointer to the float destination buffer - this is the buffer that the CoreAudio.framework uses // its size is numSampleFrames * numChannels * sizeof(float) // firstSampleFrame - this is the index of the first sample frame to the input conversion on // numSampleFrames - the total number of sample frames to convert and scale // streamFormat - the current format of the IOAudioStream this function is operating on // audioStream - the audio stream this function is operating on IOReturn SampleAudioEngine::convertInputSamples(const void *sampleBuf, void *destBuf, UInt32 firstSampleFrame, UInt32 numSampleFrames, const IOAudioStreamFormat *streamFormat, IOAudioStream *audioStream) { UInt32 numSamplesLeft; float *floatDestBuf; SInt16 *inputBuf; // Start by casting the destination buffer to a float * floatDestBuf = (float *)destBuf; // Determine the starting point for our input conversion inputBuf = &(((SInt16 *)sampleBuf)[firstSampleFrame * streamFormat->fNumChannels]); // Calculate the number of actual samples to convert numSamplesLeft = numSampleFrames * streamFormat->fNumChannels; // Loop through each sample and scale and convert them while (numSamplesLeft > 0) { SInt16 inputSample; // Fetch the SInt16 input sample inputSample = *inputBuf; // Scale that sample to a range of -1.0 to 1.0, convert to float and store in the destination buffer // at the proper location if (inputSample >= 0) { *floatDestBuf = inputSample / 32767.0; } else { *floatDestBuf = inputSample / 32768.0; } // Move on to the next sample ++inputBuf; ++floatDestBuf; --numSamplesLeft; } return kIOReturnSuccess; }