Dynamically load FFmpeg and libfdk-aac if available. (#6570)
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@ -2,8 +2,10 @@
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// Licensed under GPLv2 or any later version
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// Refer to the license.txt file included.
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#include <fdk-aac/aacdecoder_lib.h>
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#include "audio_core/hle/fdk_decoder.h"
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#include "common/dynamic_library/fdk-aac.h"
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using namespace DynamicLibrary;
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namespace AudioCore::HLE {
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@ -29,13 +31,17 @@ private:
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};
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FDKDecoder::Impl::Impl(Memory::MemorySystem& memory) : memory(memory) {
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if (!FdkAac::LoadFdkAac()) {
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return;
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}
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// allocate an array of LIB_INFO structures
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// if we don't pre-fill the whole segment with zeros, when we call `aacDecoder_GetLibInfo`
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// it will segfault, upon investigation, there is some code in fdk_aac depends on your initial
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// values in this array
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LIB_INFO decoder_info[FDK_MODULE_LAST] = {};
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// get library information and fill the struct
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if (aacDecoder_GetLibInfo(decoder_info) != 0) {
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if (FdkAac::aacDecoder_GetLibInfo(decoder_info) != 0) {
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LOG_ERROR(Audio_DSP, "Failed to retrieve fdk_aac library information!");
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return;
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}
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@ -44,14 +50,14 @@ FDKDecoder::Impl::Impl(Memory::MemorySystem& memory) : memory(memory) {
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decoder_info[0].build_date);
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// choose the input format when initializing: 1 layer of ADTS
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decoder = aacDecoder_Open(TRANSPORT_TYPE::TT_MP4_ADTS, 1);
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decoder = FdkAac::aacDecoder_Open(TRANSPORT_TYPE::TT_MP4_ADTS, 1);
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// set maximum output channel to two (stereo)
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// if the input samples have more channels, fdk_aac will perform a downmix
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AAC_DECODER_ERROR ret = aacDecoder_SetParam(decoder, AAC_PCM_MAX_OUTPUT_CHANNELS, 2);
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AAC_DECODER_ERROR ret = FdkAac::aacDecoder_SetParam(decoder, AAC_PCM_MAX_OUTPUT_CHANNELS, 2);
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if (ret != AAC_DEC_OK) {
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// unable to set this parameter reflects the decoder implementation might be broken
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// we'd better shuts down everything
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aacDecoder_Close(decoder);
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FdkAac::aacDecoder_Close(decoder);
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decoder = nullptr;
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LOG_ERROR(Audio_DSP, "Unable to set downmix parameter: {}", ret);
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return;
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@ -73,8 +79,9 @@ std::optional<BinaryMessage> FDKDecoder::Impl::Initalize(const BinaryMessage& re
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}
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FDKDecoder::Impl::~Impl() {
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if (decoder)
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aacDecoder_Close(decoder);
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if (decoder) {
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FdkAac::aacDecoder_Close(decoder);
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}
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}
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void FDKDecoder::Impl::Clear() {
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@ -84,9 +91,10 @@ void FDKDecoder::Impl::Clear() {
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// FLUSH - flush internal buffer
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// INTR - treat the current internal buffer as discontinuous
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// CONCEAL - try to interpolate and smooth out the samples
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if (decoder)
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aacDecoder_DecodeFrame(decoder, decoder_output, 8192,
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AACDEC_FLUSH & AACDEC_INTR & AACDEC_CONCEAL);
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if (decoder) {
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FdkAac::aacDecoder_DecodeFrame(decoder, decoder_output, 8192,
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AACDEC_FLUSH & AACDEC_INTR & AACDEC_CONCEAL);
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}
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}
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std::optional<BinaryMessage> FDKDecoder::Impl::ProcessRequest(const BinaryMessage& request) {
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@ -140,7 +148,7 @@ std::optional<BinaryMessage> FDKDecoder::Impl::Decode(const BinaryMessage& reque
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std::array<std::vector<s16>, 2> out_streams;
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std::size_t data_size = request.decode_aac_request.size;
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u32 data_size = request.decode_aac_request.size;
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// decoding loops
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AAC_DECODER_ERROR result = AAC_DEC_OK;
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@ -156,18 +164,18 @@ std::optional<BinaryMessage> FDKDecoder::Impl::Decode(const BinaryMessage& reque
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while (buffer_remaining) {
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// queue the input buffer, fdk_aac will automatically slice out the buffer it needs
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// from the input buffer
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result = aacDecoder_Fill(decoder, &data, &input_size, &buffer_remaining);
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result = FdkAac::aacDecoder_Fill(decoder, &data, &input_size, &buffer_remaining);
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if (result != AAC_DEC_OK) {
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// there are some issues when queuing the input buffer
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LOG_ERROR(Audio_DSP, "Failed to enqueue the input samples");
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return std::nullopt;
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}
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// get output from decoder
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result = aacDecoder_DecodeFrame(decoder, decoder_output,
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sizeof(decoder_output) / sizeof(s16), 0);
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result = FdkAac::aacDecoder_DecodeFrame(decoder, decoder_output,
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sizeof(decoder_output) / sizeof(s16), 0);
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if (result == AAC_DEC_OK) {
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// get the stream information
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stream_info = aacDecoder_GetStreamInfo(decoder);
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stream_info = FdkAac::aacDecoder_GetStreamInfo(decoder);
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// fill the stream information for binary response
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response.decode_aac_response.sample_rate = GetSampleRateEnum(stream_info->sampleRate);
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response.decode_aac_response.num_channels = stream_info->numChannels;
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