// Copyright 2019 Citra Emulator Project // Licensed under GPLv2 or any later version // Refer to the license.txt file included. #include #include "audio_core/hle/fdk_decoder.h" namespace AudioCore::HLE { class FDKDecoder::Impl { public: explicit Impl(Memory::MemorySystem& memory); ~Impl(); std::optional ProcessRequest(const BinaryMessage& request); bool IsValid() const { return decoder != nullptr; } private: std::optional Initalize(const BinaryMessage& request); std::optional Decode(const BinaryMessage& request); void Clear(); Memory::MemorySystem& memory; HANDLE_AACDECODER decoder = nullptr; }; FDKDecoder::Impl::Impl(Memory::MemorySystem& memory) : memory(memory) { // allocate an array of LIB_INFO structures // if we don't pre-fill the whole segment with zeros, when we call `aacDecoder_GetLibInfo` // it will segfault, upon investigation, there is some code in fdk_aac depends on your initial // values in this array LIB_INFO decoder_info[FDK_MODULE_LAST] = {}; // get library information and fill the struct if (aacDecoder_GetLibInfo(decoder_info) != 0) { LOG_ERROR(Audio_DSP, "Failed to retrieve fdk_aac library information!"); return; } LOG_INFO(Audio_DSP, "Using fdk_aac version {} (build date: {})", decoder_info[0].versionStr, decoder_info[0].build_date); // choose the input format when initializing: 1 layer of ADTS decoder = aacDecoder_Open(TRANSPORT_TYPE::TT_MP4_ADTS, 1); // set maximum output channel to two (stereo) // if the input samples have more channels, fdk_aac will perform a downmix AAC_DECODER_ERROR ret = aacDecoder_SetParam(decoder, AAC_PCM_MAX_OUTPUT_CHANNELS, 2); if (ret != AAC_DEC_OK) { // unable to set this parameter reflects the decoder implementation might be broken // we'd better shuts down everything aacDecoder_Close(decoder); decoder = nullptr; LOG_ERROR(Audio_DSP, "Unable to set downmix parameter: {}", ret); return; } } std::optional FDKDecoder::Impl::Initalize(const BinaryMessage& request) { BinaryMessage response = request; response.header.result = ResultStatus::Success; if (decoder) { LOG_INFO(Audio_DSP, "FDK Decoder initialized"); Clear(); } else { LOG_ERROR(Audio_DSP, "Decoder not initialized"); } return response; } FDKDecoder::Impl::~Impl() { if (decoder) aacDecoder_Close(decoder); } void FDKDecoder::Impl::Clear() { s16 decoder_output[8192]; // flush and re-sync the decoder, discarding the internal buffer // we actually don't care if this succeeds or not // FLUSH - flush internal buffer // INTR - treat the current internal buffer as discontinuous // CONCEAL - try to interpolate and smooth out the samples if (decoder) aacDecoder_DecodeFrame(decoder, decoder_output, 8192, AACDEC_FLUSH & AACDEC_INTR & AACDEC_CONCEAL); } std::optional FDKDecoder::Impl::ProcessRequest(const BinaryMessage& request) { if (request.header.codec != DecoderCodec::DecodeAAC) { LOG_ERROR(Audio_DSP, "FDK AAC Decoder cannot handle such codec: {}", static_cast(request.header.codec)); return {}; } switch (request.header.cmd) { case DecoderCommand::Init: { return Initalize(request); } case DecoderCommand::EncodeDecode: { return Decode(request); } case DecoderCommand::Unknown: { BinaryMessage response = request; response.header.result = 0x0; return response; } default: LOG_ERROR(Audio_DSP, "Got unknown binary request: {}", static_cast(request.header.cmd)); return {}; } } std::optional FDKDecoder::Impl::Decode(const BinaryMessage& request) { BinaryMessages response; response.header.codec = request.header.codec; response.header.cmd = request.header.cmd; response.decode_aac_response.size = request.decode_aac_request.size; if (!decoder) { LOG_DEBUG(Audio_DSP, "Decoder not initalized"); // This is a hack to continue games that are not compiled with the aac codec response.decode_aac_response.num_channels = 2; response.decode_aac_response.num_samples = 1024; return response; } if (request.decode_aac_request.src_addr < Memory::FCRAM_PADDR || request.decode_aac_request.src_addr + request.decode_aac_request.size > Memory::FCRAM_PADDR + Memory::FCRAM_SIZE) { LOG_ERROR(Audio_DSP, "Got out of bounds src_addr {:08x}", request.decode_aac_request.src_addr); return {}; } u8* data = memory.GetFCRAMPointer(request.decode_aac_request.src_addr - Memory::FCRAM_PADDR); std::array, 2> out_streams; std::size_t data_size = request.decode_aac_request.size; // decoding loops AAC_DECODER_ERROR result = AAC_DEC_OK; // Up to 2048 samples, up to 2 channels each s16 decoder_output[4096]; // note that we don't free this pointer as it is automatically freed by fdk_aac CStreamInfo* stream_info; // how many bytes to be queued into the decoder, decrementing from the buffer size u32 buffer_remaining = data_size; // alias the data_size as an u32 u32 input_size = data_size; while (buffer_remaining) { // queue the input buffer, fdk_aac will automatically slice out the buffer it needs // from the input buffer result = aacDecoder_Fill(decoder, &data, &input_size, &buffer_remaining); if (result != AAC_DEC_OK) { // there are some issues when queuing the input buffer LOG_ERROR(Audio_DSP, "Failed to enqueue the input samples"); return std::nullopt; } // get output from decoder result = aacDecoder_DecodeFrame(decoder, decoder_output, sizeof(decoder_output) / sizeof(s16), 0); if (result == AAC_DEC_OK) { // get the stream information stream_info = aacDecoder_GetStreamInfo(decoder); // fill the stream information for binary response response.decode_aac_response.sample_rate = GetSampleRateEnum(stream_info->sampleRate); response.decode_aac_response.num_channels = stream_info->numChannels; response.decode_aac_response.num_samples = stream_info->frameSize; // fill the output // the sample size = frame_size * channel_counts for (int sample = 0; sample < stream_info->frameSize; sample++) { for (int ch = 0; ch < stream_info->numChannels; ch++) { out_streams[ch].push_back( decoder_output[(sample * stream_info->numChannels) + ch]); } } } else if (result == AAC_DEC_TRANSPORT_SYNC_ERROR) { // decoder has some synchronization problems, try again with new samples, // using old samples might trigger this error again continue; } else { LOG_ERROR(Audio_DSP, "Error decoding the sample: {}", result); return std::nullopt; } } // transfer the decoded buffer from vector to the FCRAM for (std::size_t ch = 0; ch < out_streams.size(); ch++) { if (!out_streams[ch].empty()) { auto byte_size = out_streams[ch].size() * sizeof(s16); auto dst = ch == 0 ? request.decode_aac_request.dst_addr_ch0 : request.decode_aac_request.dst_addr_ch1; if (dst < Memory::FCRAM_PADDR || dst + byte_size > Memory::FCRAM_PADDR + Memory::FCRAM_SIZE) { LOG_ERROR(Audio_DSP, "Got out of bounds dst_addr_ch{} {:08x}", ch, dst); return {}; } std::memcpy(memory.GetFCRAMPointer(dst - Memory::FCRAM_PADDR), out_streams[ch].data(), byte_size); } } return response; } FDKDecoder::FDKDecoder(Memory::MemorySystem& memory) : impl(std::make_unique(memory)) {} FDKDecoder::~FDKDecoder() = default; std::optional FDKDecoder::ProcessRequest(const BinaryMessage& request) { return impl->ProcessRequest(request); } bool FDKDecoder::IsValid() const { return impl->IsValid(); } } // namespace AudioCore::HLE