mirror of
https://github.com/RGBCube/serenity
synced 2025-07-27 13:57:35 +00:00
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.
This commit is contained in:
parent
d707c0a2f5
commit
264cc76ab4
15 changed files with 146 additions and 128 deletions
|
@ -15,12 +15,6 @@
|
|||
|
||||
namespace Audio {
|
||||
|
||||
// Experimentally determined to be a decent buffer size on i686:
|
||||
// 4K (the default) is slightly worse, and 64K is much worse.
|
||||
// At sufficiently large buffer sizes, the advantage of infrequent read() calls is outweighed by the memmove() overhead.
|
||||
// There was no intensive fine-tuning done to determine this value, so improvements may definitely be possible.
|
||||
constexpr size_t FLAC_BUFFER_SIZE = 8 * KiB;
|
||||
|
||||
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);
|
||||
|
@ -49,7 +43,7 @@ public:
|
|||
static Result<NonnullOwnPtr<FlacLoaderPlugin>, LoaderError> create(StringView path);
|
||||
static Result<NonnullOwnPtr<FlacLoaderPlugin>, LoaderError> create(Bytes buffer);
|
||||
|
||||
virtual LoaderSamples get_more_samples(size_t max_bytes_to_read_from_input = 128 * KiB) override;
|
||||
virtual ErrorOr<Vector<FixedArray<Sample>>, LoaderError> load_chunks(size_t samples_to_read_from_input) override;
|
||||
|
||||
virtual MaybeLoaderError reset() override;
|
||||
virtual MaybeLoaderError seek(int sample_index) override;
|
||||
|
@ -70,8 +64,8 @@ private:
|
|||
// 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);
|
||||
// Fetches and writes the next FLAC frame
|
||||
MaybeLoaderError next_frame(Span<Sample>);
|
||||
// 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
|
||||
|
@ -108,8 +102,6 @@ private:
|
|||
// 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;
|
||||
// Whatever the last get_more_samples() call couldn't return gets stored here.
|
||||
Vector<Sample, FLAC_BUFFER_SIZE> m_unread_data;
|
||||
u64 m_current_sample_or_frame { 0 };
|
||||
Vector<FlacSeekPoint> m_seektable;
|
||||
};
|
||||
|
|
Loading…
Add table
Add a link
Reference in a new issue