mirror of
https://github.com/intrepidcs/libicsneo.git
synced 2026-08-05 01:18:36 +02:00
Device: Refactor A2B APIs
* Removes features in `A2BMessage` class to support API for reading 16, 24, and 32 bit samples from A2B channels * Re-organizes WAV receiving and transmitting code and API * Creates API for mapping message channels to WAV channels and vice versa for transmitting and receiving * Fixes `icsneo::Network::NetID::ExtendedData` VnetID bug for `icsneo::ExtendedDataMessage` decoding * Creates RAD-A2B sequence chart example * Fixes coremini uploading for certain devices in EEPROM by introducing `icsneo::Device::supportsEraseMemory`
This commit is contained in:
Yasser Yassine
committed by
Kyle Schwarz
Kyle Schwarz
parent
06f6861130
commit
cb22e622b3
@@ -0,0 +1,258 @@
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#include "icsneo/communication/message/a2bmessage.h"
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#include "icsneo/communication/message/callback/streamoutput/streamoutput.h"
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using namespace icsneo;
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// Read a 16 bit sample from the audio buffer, which is stored as little endian
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#define SAMPLE_FROM_BYTES_16(audioData) (((audioData)[0]) | ((audioData)[1] << 8))
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// Read a 32 bit sample from the audio buffer
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#define SAMPLE_FROM_BYTES_32(audioData) (((audioData)[0]) | ((audioData)[1] << 8) | ((audioData)[2] << 16) | ((audioData)[3] << 24))
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// Read the most significant bytes of a sample stored in a 32 bit unsigned integer into audioData
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#define SAMPLE_TO_BYTES_16(audioData, offset, sample) {\
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(audioData)[(offset)++] = static_cast<uint8_t>(((sample) & 0x00FF0000u) >> 16);\
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(audioData)[(offset)++] = static_cast<uint8_t>(((sample) & 0xFF000000u) >> 24);\
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}
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// Read little endian a 32 bit unsigned integer into audioData
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#define SAMPLE_TO_BYTES_32(audioData, offset, sample) {\
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(audioData)[(offset)++] = static_cast<uint8_t>(((sample) & 0x000000FFu));\
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(audioData)[(offset)++] = static_cast<uint8_t>(((sample) & 0x0000FF00u) >> 8);\
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(audioData)[(offset)++] = static_cast<uint8_t>(((sample) & 0x00FF0000u) >> 16);\
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(audioData)[(offset)++] = static_cast<uint8_t>(((sample) & 0xFF000000u) >> 24);\
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}
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uint8_t A2BMessage::tdmToChannelNum(TDMMode tdm) {
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switch(tdm) {
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case TDMMode::TDM2:
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return 4;
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case TDMMode::TDM4:
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return 8;
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case TDMMode::TDM8:
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return 16;
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case TDMMode::TDM12:
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return 24;
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case TDMMode::TDM16:
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return 32;
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case TDMMode::TDM20:
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return 40;
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case TDMMode::TDM24:
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return 48;
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case TDMMode::TDM32:
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return 64;
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}
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return 0;
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}
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uint8_t A2BMessage::getBytesPerChannel() const {
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return channelSize16 ? 2u : 4u;
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}
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size_t A2BMessage::getFrameSize() const {
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return static_cast<size_t>(2 * numChannels * getBytesPerChannel());
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}
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size_t A2BMessage::getSampleOffset(Direction dir, uint8_t channel, size_t frame) const {
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size_t frameSize = getFrameSize();
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size_t sampleOffset = static_cast<size_t>(frameSize * frame + 2 * channel * getBytesPerChannel());
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if(dir == Direction::Upstream) {
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sampleOffset++;
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}
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return sampleOffset;
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}
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size_t A2BMessage::getNumFrames() const {
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size_t frameSize = getFrameSize();
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if(frameSize == 0) {
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return 0;
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}
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return data.size() / frameSize;
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}
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A2BMessage::A2BMessage(size_t numFrames, TDMMode tdm, bool chSize16) : channelSize16(chSize16) {
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numChannels = static_cast<uint8_t>(tdmToChannelNum(tdm) / 2);
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size_t frameSize = static_cast<size_t>(2 * numChannels * (chSize16 ? 2u : 4u));
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size_t audioBufferSize = frameSize * numFrames;
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if(audioBufferSize > maxAudioBufferSize) {
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size_t maxNumFrames = maxAudioBufferSize / frameSize;
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audioBufferSize = maxNumFrames * frameSize;
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}
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data.resize(std::min<size_t>(maxAudioBufferSize, audioBufferSize), 0);
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}
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A2BMessage::A2BMessage(TDMMode tdm, bool chSize16) : channelSize16(chSize16) {
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numChannels = static_cast<uint8_t>(tdmToChannelNum(tdm) / 2);
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size_t frameSize = static_cast<size_t>(2 * numChannels * (chSize16 ? 2u : 4u));
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size_t maxNumFrames = maxAudioBufferSize / frameSize;
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size_t audioBufferSize = maxNumFrames * frameSize;
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data.resize(audioBufferSize, 0);
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}
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PCMSample A2BMessage::getChannelSample(Direction dir, uint8_t channel, size_t frame, PCMType pcmType) const {
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size_t sampleOffset = getSampleOffset(dir, channel, frame);
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const uint8_t* audioData = &data[sampleOffset];
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PCMSample result = 0;
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// Samples coming from the device will either come from a 16 bit channel or 32 bit channel
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if(channelSize16) {
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int16_t sample16 = 0;
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uint16_t& uSample16 = *reinterpret_cast<uint16_t*>(&sample16);
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// Read little endian from the audio buffer
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uSample16 = SAMPLE_FROM_BYTES_16(audioData);
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// Scale the sample up according to the desired PCM size by
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// multiplying using logical shifting
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switch(pcmType) {
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case PCMType::L16:
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result = static_cast<PCMSample>(sample16);
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break;
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case PCMType::L24:
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result = static_cast<PCMSample>(sample16) << 8;
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break;
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case PCMType::L32:
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result = static_cast<PCMSample>(sample16) << 16;
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break;
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}
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} else {
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PCMSample sample32 = 0;
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uint32_t& uSample32 = *reinterpret_cast<uint32_t*>(&sample32);
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// Read little endian
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uSample32 = SAMPLE_FROM_BYTES_32(audioData);
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// Scale the sample down according to the desired PCM size by dividing using
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// logical shifting, if the A2B network was set up with the desired pcmType
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// there should be a clean division and no loss in PCM resolution.
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switch(pcmType) {
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case PCMType::L16:
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result = sample32 >> 16;
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break;
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case PCMType::L24:
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result = sample32 >> 8;
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break;
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case PCMType::L32:
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result = sample32;
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break;
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}
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}
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return result;
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}
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void A2BMessage::setChannelSample(Direction dir, uint8_t channel, size_t frame, PCMSample sampleToSet, PCMType pcmType) {
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size_t sampleOffset = getSampleOffset(dir, channel, frame);
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uint8_t* audioData = data.data();
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uint32_t& uSample = *reinterpret_cast<uint32_t*>(&sampleToSet);
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// Align the bytes towards the most significant bit by multiplying using
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// left shifts
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switch(pcmType) {
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case PCMType::L16:
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sampleToSet = sampleToSet << 16;
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break;
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case PCMType::L24:
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sampleToSet = sampleToSet << 8;
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break;
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}
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if(channelSize16) {
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// Read the 2 most significant bytes of the sample
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SAMPLE_TO_BYTES_16(audioData, sampleOffset, uSample)
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} else {
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// Read the entire sample
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SAMPLE_TO_BYTES_32(audioData, sampleOffset, uSample);
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}
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}
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bool A2BMessage::loadAudioBuffer(IWAVStream& wavStream, const ChannelMap& channelMap) {
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if(!wavStream) {
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return false;
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}
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size_t totalMessageChannels = numChannels * 2; // Multiply by two inorder to include both down and upstream channels
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size_t bytesPerChannel = static_cast<size_t>(getBytesPerChannel()); // Number of bytes per message channel
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size_t frameSize = getFrameSize();
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size_t numFrames = getNumFrames();
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size_t bytesPerSampleWAV = static_cast<size_t>(wavStream.header.bitsPerSample / 8); // Number of bytes per sample in the WAV data-stream
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size_t numWAVChannels = static_cast<size_t>(wavStream.header.numChannels);
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size_t wavFrameSize = numWAVChannels * bytesPerSampleWAV;
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if(bytesPerSampleWAV != 2 && bytesPerSampleWAV != 3 && bytesPerSampleWAV != 4) {
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return false;
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}
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if(numFrames == 0) {
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return false;
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}
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uint8_t* audioBuffer = data.data();
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std::vector<uint8_t> wavFrame(wavFrameSize, 0);
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for(size_t frame = 0; frame < numFrames; frame++) {
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// Read one frame of data from the input stream
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if(!wavStream.read(reinterpret_cast<char*>(wavFrame.data()), wavFrame.size())) {
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break;
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}
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// Iterate through each mapping and set a message channel to a channel in the WAV frame above
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for(const auto& [messageChannel, wavChannel] : channelMap) {
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if(messageChannel >= totalMessageChannels || wavChannel >= numWAVChannels) {
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return false;
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}
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size_t frameOffset = wavChannel * bytesPerSampleWAV; // Offset in the read WAV frame
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size_t audioBufferOffset = frame * frameSize + messageChannel * bytesPerChannel; // Offset in the message audio buffer
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if(bytesPerChannel < bytesPerSampleWAV) {
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// In this case, the message channels are smaller than the samples in the input WAV
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// samples in both the message channel and WAV are little endian, so we write only the
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// most significant bytes of the WAV
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// Align to most significant bytes of wav frame
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size_t align = bytesPerSampleWAV - bytesPerChannel;
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for(
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size_t frameByte = frameOffset + align;
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frameByte < frameOffset + bytesPerSampleWAV;
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frameByte++,
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audioBufferOffset++
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) {
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audioBuffer[audioBufferOffset] = wavFrame[frameByte];
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}
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} else {
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// The message channel is greater than or equal to the sample in the WAV
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// I2S specifies that the sample in this case is right aligned to the most significant
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// byte of the message channel
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// Align to most significant byte of audio buffer channel
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size_t align = bytesPerChannel - bytesPerSampleWAV;
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for(
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size_t audioByte = audioBufferOffset + align;
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audioByte < audioBufferOffset + bytesPerChannel;
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audioByte++,
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frameOffset++
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) {
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audioBuffer[audioByte] = wavFrame[frameOffset];
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}
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}
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}
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}
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return true;
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}
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