ladybird/Userland/Libraries/LibAudio/FlacLoader.h

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/*
* Copyright (c) 2021, kleines Filmröllchen <filmroellchen@serenityos.org>
*
* SPDX-License-Identifier: BSD-2-Clause
*/
#pragma once
#include "FlacTypes.h"
#include "Loader.h"
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#include <AK/BitStream.h>
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#include <AK/Error.h>
#include <AK/Span.h>
#include <AK/Types.h>
namespace Audio {
ALWAYS_INLINE u8 frame_channel_type_to_channel_count(FlacFrameChannelType channel_type);
// Sign-extend an arbitrary-size signed number to 64 bit signed
ALWAYS_INLINE i64 sign_extend(u32 n, u8 size);
// Decodes the sign representation method used in Rice coding.
// Numbers alternate between positive and negative: 0, 1, -1, 2, -2, 3, -3, 4, -4, 5, -5, ...
ALWAYS_INLINE i32 rice_to_signed(u32 x);
// decoders
// read a UTF-8 encoded number, even if it is not a valid codepoint
ALWAYS_INLINE ErrorOr<u64> read_utf8_char(BigEndianInputBitStream& input);
// decode a single number encoded with exponential golomb encoding of the specified order
ALWAYS_INLINE ErrorOr<i32> decode_unsigned_exp_golomb(u8 order, BigEndianInputBitStream& bit_input);
// Loader for the Free Lossless Audio Codec (FLAC)
// This loader supports all audio features of FLAC, although audio from more than two channels is discarded.
// The loader currently supports the STREAMINFO, PADDING, and SEEKTABLE metadata blocks.
// See: https://xiph.org/flac/documentation_format_overview.html
// https://xiph.org/flac/format.html (identical to IETF draft version 2)
// https://datatracker.ietf.org/doc/html/draft-ietf-cellar-flac-02 (all section numbers refer to this specification)
// https://datatracker.ietf.org/doc/html/draft-ietf-cellar-flac-03 (newer IETF draft that uses incompatible numberings and names)
class FlacLoaderPlugin : public LoaderPlugin {
public:
explicit FlacLoaderPlugin(NonnullOwnPtr<SeekableStream> stream);
virtual ~FlacLoaderPlugin() override = default;
static bool sniff(SeekableStream& stream);
static ErrorOr<NonnullOwnPtr<LoaderPlugin>, LoaderError> create(NonnullOwnPtr<SeekableStream>);
LibAudio: Move audio stream buffering into the loader 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.
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virtual ErrorOr<Vector<FixedArray<Sample>>, LoaderError> load_chunks(size_t samples_to_read_from_input) override;
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virtual MaybeLoaderError reset() override;
virtual MaybeLoaderError seek(int sample_index) override;
virtual int loaded_samples() override { return static_cast<int>(m_loaded_samples); }
virtual int total_samples() override { return static_cast<int>(m_total_samples); }
virtual u32 sample_rate() override { return m_sample_rate; }
virtual u16 num_channels() override { return m_num_channels; }
virtual ByteString format_name() override { return "FLAC (.flac)"; }
virtual PcmSampleFormat pcm_format() override { return m_sample_format; }
bool is_fixed_blocksize_stream() const { return m_min_block_size == m_max_block_size; }
bool sample_count_unknown() const { return m_total_samples == 0; }
private:
MaybeLoaderError initialize();
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MaybeLoaderError parse_header();
// Either returns the metadata block or sets error message.
// Additionally, increments m_data_start_location past the read meta block.
ErrorOr<FlacRawMetadataBlock, LoaderError> next_meta_block(BigEndianInputBitStream& bit_input);
LibAudio: Move audio stream buffering into the loader 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.
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// Fetches and returns the next FLAC frame.
LoaderSamples next_frame();
// Helper of next_frame that fetches a sub frame's header
ErrorOr<FlacSubframeHeader, LoaderError> next_subframe_header(BigEndianInputBitStream& bit_input, u8 channel_index);
// Helper of next_frame that decompresses a subframe
ErrorOr<void, LoaderError> parse_subframe(Vector<i64>& samples, FlacSubframeHeader& subframe_header, BigEndianInputBitStream& bit_input);
// Subframe-internal data decoders (heavy lifting)
ErrorOr<Vector<i64>, LoaderError> decode_fixed_lpc(FlacSubframeHeader& subframe, BigEndianInputBitStream& bit_input);
ErrorOr<Vector<i64>, LoaderError> decode_verbatim(FlacSubframeHeader& subframe, BigEndianInputBitStream& bit_input);
ErrorOr<void, LoaderError> decode_custom_lpc(Vector<i64>& decoded, FlacSubframeHeader& subframe, BigEndianInputBitStream& bit_input);
MaybeLoaderError decode_residual(Vector<i64>& decoded, FlacSubframeHeader& subframe, BigEndianInputBitStream& bit_input);
// decode a single rice partition that has its own rice parameter
ALWAYS_INLINE ErrorOr<Vector<i64>, LoaderError> decode_rice_partition(u8 partition_type, u32 partitions, u32 partition_index, FlacSubframeHeader& subframe, BigEndianInputBitStream& bit_input);
MaybeLoaderError load_seektable(FlacRawMetadataBlock&);
// Note that failing to read a Vorbis comment block is not treated as an error of the FLAC loader, since metadata is optional.
void load_vorbis_comment(FlacRawMetadataBlock&);
MaybeLoaderError load_picture(FlacRawMetadataBlock&);
// Converters for special coding used in frame headers
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ALWAYS_INLINE ErrorOr<u32, LoaderError> convert_sample_count_code(u8 sample_count_code);
ALWAYS_INLINE ErrorOr<u32, LoaderError> convert_sample_rate_code(u8 sample_rate_code);
ALWAYS_INLINE ErrorOr<u8, LoaderError> convert_bit_depth_code(u8 bit_depth_code);
bool should_insert_seekpoint_at(u64 sample_index) const;
// Data obtained directly from the FLAC metadata: many values have specific bit counts
u32 m_sample_rate { 0 }; // 20 bit
u8 m_num_channels { 0 }; // 3 bit
u8 m_bits_per_sample { 0 }; // 5 bits for the integer bit depth
// Externally visible format; the smallest integer format that's larger than the precise bit depth.
PcmSampleFormat m_sample_format;
// Blocks are units of decoded audio data
u16 m_min_block_size { 0 };
u16 m_max_block_size { 0 };
// Frames are units of encoded audio data, both of these are 24-bit
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u32 m_min_frame_size { 0 }; // 24 bit
u32 m_max_frame_size { 0 }; // 24 bit
u64 m_total_samples { 0 }; // 36 bit
u8 m_md5_checksum[128 / 8]; // 128 bit (!)
size_t m_loaded_samples { 0 };
// keep track of the start of the data in the FLAC stream to seek back more easily
u64 m_data_start_location { 0 };
Optional<FlacFrameHeader> m_current_frame;
u64 m_current_sample_or_frame { 0 };
SeekTable m_seektable;
// Keep around a few temporary buffers whose allocated space can be reused.
// This is an empirical optimization since allocations and deallocations take a lot of time in the decoder.
mutable Vector<Vector<i64>, 2> m_subframe_buffers;
};
}