citra/src/audio_core/hle/fdk_decoder.cpp

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// Copyright 2019 Citra Emulator Project
// Licensed under GPLv2 or any later version
// Refer to the license.txt file included.
#include <fdk-aac/aacdecoder_lib.h>
#include "audio_core/hle/fdk_decoder.h"
namespace AudioCore::HLE {
class FDKDecoder::Impl {
public:
explicit Impl(Memory::MemorySystem& memory);
~Impl();
std::optional<BinaryMessage> ProcessRequest(const BinaryMessage& request);
bool IsValid() const {
return decoder != nullptr;
}
private:
std::optional<BinaryMessage> Initalize(const BinaryMessage& request);
std::optional<BinaryMessage> 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<BinaryMessage> 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<BinaryMessage> 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<u16>(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 = ResultStatus::Success;
return response;
}
default:
LOG_ERROR(Audio_DSP, "Got unknown binary request: {}",
static_cast<u16>(request.header.cmd));
return {};
}
}
std::optional<BinaryMessage> FDKDecoder::Impl::Decode(const BinaryMessage& request) {
BinaryMessage 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<std::vector<s16>, 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<Impl>(memory)) {}
FDKDecoder::~FDKDecoder() = default;
std::optional<BinaryMessage> FDKDecoder::ProcessRequest(const BinaryMessage& request) {
return impl->ProcessRequest(request);
}
bool FDKDecoder::IsValid() const {
return impl->IsValid();
}
} // namespace AudioCore::HLE