mirror of
https://github.com/RGBCube/serenity
synced 2025-05-20 02:25:07 +00:00
264cc76ab4
Before, some loader plugins implemented their own buffering (FLAC&MP3), some didn't require any (WAV), and some didn't buffer at all (QOA). This meant that in practice, while you could load arbitrary amounts of samples from some loader plugins, you couldn't do that with some others. Also, it was ill-defined how many samples you would actually get back from a get_more_samples call. This commit fixes that by introducing a layer of abstraction between the loader and its plugins (because that's the whole point of having the extra class!). The plugins now only implement a load_chunks() function, which is much simpler to implement and allows plugins to play fast and loose with what they actually return. Basically, they can return many chunks of samples, where one chunk is simply a convenient block of samples to load. In fact, some loaders such as FLAC and QOA have separate internal functions for loading exactly one chunk. The loaders *should* load as many chunks as necessary for the sample count to be reached or surpassed (the latter simplifies loading loops in the implementations, since you don't need to know how large your next chunk is going to be; a problem for e.g. FLAC). If a plugin has no problems returning data of arbitrary size (currently WAV), it can return a single chunk that exactly (or roughly) matches the requested sample count. If a plugin is at the stream end, it can also return less samples than was requested! The loader can handle all of these cases and may call into load_chunk multiple times. If the plugin returns an empty chunk list (or only empty chunks; again, they can play fast and loose), the loader takes that as a stream end signal. Otherwise, the loader will always return exactly as many samples as the user requested. Buffering is handled by the loader, allowing any underlying plugin to deal with any weird sample count requirement the user throws at it (looking at you, SoundPlayer!). This (not accidentally!) makes QOA work in SoundPlayer.
77 lines
3.2 KiB
C++
77 lines
3.2 KiB
C++
/*
|
|
* Copyright (c) 2021, Arne Elster <arne@elster.li>
|
|
*
|
|
* SPDX-License-Identifier: BSD-2-Clause
|
|
*/
|
|
|
|
#pragma once
|
|
|
|
#include "Loader.h"
|
|
#include "MP3Types.h"
|
|
#include <AK/BitStream.h>
|
|
#include <AK/MemoryStream.h>
|
|
#include <AK/Tuple.h>
|
|
#include <LibDSP/MDCT.h>
|
|
|
|
namespace Audio {
|
|
|
|
namespace MP3::Tables {
|
|
struct ScaleFactorBand;
|
|
}
|
|
|
|
class MP3LoaderPlugin : public LoaderPlugin {
|
|
public:
|
|
explicit MP3LoaderPlugin(NonnullOwnPtr<SeekableStream> stream);
|
|
virtual ~MP3LoaderPlugin() = default;
|
|
|
|
static Result<NonnullOwnPtr<MP3LoaderPlugin>, LoaderError> create(StringView path);
|
|
static Result<NonnullOwnPtr<MP3LoaderPlugin>, LoaderError> create(Bytes buffer);
|
|
|
|
virtual ErrorOr<Vector<FixedArray<Sample>>, LoaderError> load_chunks(size_t samples_to_read_from_input) override;
|
|
|
|
virtual MaybeLoaderError reset() override;
|
|
virtual MaybeLoaderError seek(int const position) override;
|
|
|
|
virtual int loaded_samples() override { return m_loaded_samples; }
|
|
virtual int total_samples() override { return m_total_samples; }
|
|
virtual u32 sample_rate() override { return m_sample_rate; }
|
|
virtual u16 num_channels() override { return m_num_channels; }
|
|
virtual PcmSampleFormat pcm_format() override { return m_sample_format; }
|
|
virtual DeprecatedString format_name() override { return "MP3 (.mp3)"; }
|
|
|
|
private:
|
|
MaybeLoaderError initialize();
|
|
MaybeLoaderError synchronize();
|
|
MaybeLoaderError build_seek_table();
|
|
ErrorOr<MP3::Header, LoaderError> read_header();
|
|
ErrorOr<MP3::MP3Frame, LoaderError> read_next_frame();
|
|
ErrorOr<MP3::MP3Frame, LoaderError> read_frame_data(MP3::Header const&);
|
|
MaybeLoaderError read_side_information(MP3::MP3Frame&);
|
|
ErrorOr<size_t, LoaderError> read_scale_factors(MP3::MP3Frame&, BigEndianInputBitStream& reservoir, size_t granule_index, size_t channel_index);
|
|
MaybeLoaderError read_huffman_data(MP3::MP3Frame&, BigEndianInputBitStream& reservoir, size_t granule_index, size_t channel_index, size_t granule_bits_read);
|
|
static AK::Array<float, 576> calculate_frame_exponents(MP3::MP3Frame const&, size_t granule_index, size_t channel_index);
|
|
static void reorder_samples(MP3::Granule&, u32 sample_rate);
|
|
static void reduce_alias(MP3::Granule&, size_t max_subband_index = 576);
|
|
static void process_stereo(MP3::MP3Frame&, size_t granule_index);
|
|
static void transform_samples_to_time(Array<float, 576> const& input, size_t input_offset, Array<float, 36>& output, MP3::BlockType block_type);
|
|
static void synthesis(Array<float, 1024>& V, Array<float, 32>& samples, Array<float, 32>& result);
|
|
static ReadonlySpan<MP3::Tables::ScaleFactorBand> get_scalefactor_bands(MP3::Granule const&, int samplerate);
|
|
|
|
AK::Vector<AK::Tuple<size_t, int>> m_seek_table;
|
|
AK::Array<AK::Array<AK::Array<float, 18>, 32>, 2> m_last_values {};
|
|
AK::Array<AK::Array<float, 1024>, 2> m_synthesis_buffer {};
|
|
static DSP::MDCT<36> s_mdct_36;
|
|
static DSP::MDCT<12> s_mdct_12;
|
|
|
|
u32 m_sample_rate { 0 };
|
|
u8 m_num_channels { 0 };
|
|
PcmSampleFormat m_sample_format { PcmSampleFormat::Int16 };
|
|
int m_total_samples { 0 };
|
|
size_t m_loaded_samples { 0 };
|
|
|
|
AK::Optional<MP3::MP3Frame> m_current_frame;
|
|
OwnPtr<BigEndianInputBitStream> m_bitstream;
|
|
AllocatingMemoryStream m_bit_reservoir;
|
|
};
|
|
|
|
}
|