
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.
343 lines
13 KiB
C++
343 lines
13 KiB
C++
/*
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* Copyright (c) 2018-2020, Andreas Kling <kling@serenityos.org>
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* Copyright (c) 2021, kleines Filmröllchen <filmroellchen@serenityos.org>
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*
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* SPDX-License-Identifier: BSD-2-Clause
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*/
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#include "WavLoader.h"
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#include "LoaderError.h"
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#include <AK/Debug.h>
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#include <AK/Endian.h>
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#include <AK/FixedArray.h>
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#include <AK/MemoryStream.h>
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#include <AK/NumericLimits.h>
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#include <AK/Try.h>
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#include <LibCore/File.h>
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namespace Audio {
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static constexpr size_t const maximum_wav_size = 1 * GiB; // FIXME: is there a more appropriate size limit?
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WavLoaderPlugin::WavLoaderPlugin(NonnullOwnPtr<SeekableStream> stream)
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: LoaderPlugin(move(stream))
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{
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}
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Result<NonnullOwnPtr<WavLoaderPlugin>, LoaderError> WavLoaderPlugin::create(StringView path)
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{
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auto stream = LOADER_TRY(Core::BufferedFile::create(LOADER_TRY(Core::File::open(path, Core::File::OpenMode::Read))));
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auto loader = make<WavLoaderPlugin>(move(stream));
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LOADER_TRY(loader->initialize());
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return loader;
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}
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Result<NonnullOwnPtr<WavLoaderPlugin>, LoaderError> WavLoaderPlugin::create(Bytes buffer)
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{
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auto stream = LOADER_TRY(try_make<FixedMemoryStream>(buffer));
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auto loader = make<WavLoaderPlugin>(move(stream));
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LOADER_TRY(loader->initialize());
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return loader;
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}
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MaybeLoaderError WavLoaderPlugin::initialize()
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{
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LOADER_TRY(parse_header());
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return {};
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}
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template<typename SampleReader>
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MaybeLoaderError WavLoaderPlugin::read_samples_from_stream(Stream& stream, SampleReader read_sample, FixedArray<Sample>& samples) const
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{
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switch (m_num_channels) {
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case 1:
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for (auto& sample : samples)
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sample = Sample(LOADER_TRY(read_sample(stream)));
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break;
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case 2:
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for (auto& sample : samples) {
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auto left_channel_sample = LOADER_TRY(read_sample(stream));
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auto right_channel_sample = LOADER_TRY(read_sample(stream));
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sample = Sample(left_channel_sample, right_channel_sample);
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}
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break;
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default:
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VERIFY_NOT_REACHED();
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}
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return {};
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}
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// There's no i24 type + we need to do the endianness conversion manually anyways.
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static ErrorOr<double> read_sample_int24(Stream& stream)
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{
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i32 sample1 = TRY(stream.read_value<u8>());
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i32 sample2 = TRY(stream.read_value<u8>());
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i32 sample3 = TRY(stream.read_value<u8>());
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i32 value = 0;
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value = sample1;
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value |= sample2 << 8;
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value |= sample3 << 16;
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// Sign extend the value, as it can currently not have the correct sign.
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value = (value << 8) >> 8;
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// Range of value is now -2^23 to 2^23-1 and we can rescale normally.
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return static_cast<double>(value) / static_cast<double>((1 << 23) - 1);
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}
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template<typename T>
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static ErrorOr<double> read_sample(Stream& stream)
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{
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T sample { 0 };
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TRY(stream.read_entire_buffer(Bytes { &sample, sizeof(T) }));
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// Remap integer samples to normalized floating-point range of -1 to 1.
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if constexpr (IsIntegral<T>) {
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if constexpr (NumericLimits<T>::is_signed()) {
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// Signed integer samples are centered around zero, so this division is enough.
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return static_cast<double>(AK::convert_between_host_and_little_endian(sample)) / static_cast<double>(NumericLimits<T>::max());
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} else {
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// Unsigned integer samples, on the other hand, need to be shifted to center them around zero.
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// The first division therefore remaps to the range 0 to 2.
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return static_cast<double>(AK::convert_between_host_and_little_endian(sample)) / (static_cast<double>(NumericLimits<T>::max()) / 2.0) - 1.0;
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}
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} else {
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return static_cast<double>(AK::convert_between_host_and_little_endian(sample));
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}
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}
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LoaderSamples WavLoaderPlugin::samples_from_pcm_data(Bytes const& data, size_t samples_to_read) const
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{
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FixedArray<Sample> samples = LOADER_TRY(FixedArray<Sample>::create(samples_to_read));
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FixedMemoryStream stream { data };
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switch (m_sample_format) {
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case PcmSampleFormat::Uint8:
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TRY(read_samples_from_stream(stream, read_sample<u8>, samples));
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break;
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case PcmSampleFormat::Int16:
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TRY(read_samples_from_stream(stream, read_sample<i16>, samples));
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break;
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case PcmSampleFormat::Int24:
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TRY(read_samples_from_stream(stream, read_sample_int24, samples));
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break;
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case PcmSampleFormat::Float32:
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TRY(read_samples_from_stream(stream, read_sample<float>, samples));
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break;
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case PcmSampleFormat::Float64:
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TRY(read_samples_from_stream(stream, read_sample<double>, samples));
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break;
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default:
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VERIFY_NOT_REACHED();
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}
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return samples;
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}
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ErrorOr<Vector<FixedArray<Sample>>, LoaderError> WavLoaderPlugin::load_chunks(size_t samples_to_read_from_input)
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{
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auto remaining_samples = m_total_samples - m_loaded_samples;
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if (remaining_samples <= 0)
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return Vector<FixedArray<Sample>> {};
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// One "sample" contains data from all channels.
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// In the Wave spec, this is also called a block.
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size_t bytes_per_sample
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= m_num_channels * pcm_bits_per_sample(m_sample_format) / 8;
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auto samples_to_read = min(samples_to_read_from_input, remaining_samples);
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auto bytes_to_read = samples_to_read * bytes_per_sample;
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dbgln_if(AWAVLOADER_DEBUG, "Read {} bytes WAV with num_channels {} sample rate {}, "
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"bits per sample {}, sample format {}",
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bytes_to_read, m_num_channels, m_sample_rate,
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pcm_bits_per_sample(m_sample_format), sample_format_name(m_sample_format));
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auto sample_data = LOADER_TRY(ByteBuffer::create_zeroed(bytes_to_read));
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LOADER_TRY(m_stream->read_entire_buffer(sample_data.bytes()));
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// m_loaded_samples should contain the amount of actually loaded samples
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m_loaded_samples += samples_to_read;
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Vector<FixedArray<Sample>> samples;
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TRY(samples.try_append(TRY(samples_from_pcm_data(sample_data.bytes(), samples_to_read))));
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return samples;
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}
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MaybeLoaderError WavLoaderPlugin::seek(int sample_index)
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{
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dbgln_if(AWAVLOADER_DEBUG, "seek sample_index {}", sample_index);
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if (sample_index < 0 || sample_index >= static_cast<int>(m_total_samples))
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return LoaderError { LoaderError::Category::Internal, m_loaded_samples, "Seek outside the sample range" };
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size_t sample_offset = m_byte_offset_of_data_samples + static_cast<size_t>(sample_index * m_num_channels * (pcm_bits_per_sample(m_sample_format) / 8));
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LOADER_TRY(m_stream->seek(sample_offset, SeekMode::SetPosition));
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m_loaded_samples = sample_index;
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return {};
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}
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// Specification reference: http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/WAVE.html
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MaybeLoaderError WavLoaderPlugin::parse_header()
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{
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bool ok = true;
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size_t bytes_read = 0;
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auto read_u8 = [&]() -> ErrorOr<u8, LoaderError> {
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u8 value = LOADER_TRY(m_stream->read_value<LittleEndian<u8>>());
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bytes_read += 1;
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return value;
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};
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auto read_u16 = [&]() -> ErrorOr<u16, LoaderError> {
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u16 value = LOADER_TRY(m_stream->read_value<LittleEndian<u16>>());
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bytes_read += 2;
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return value;
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};
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auto read_u32 = [&]() -> ErrorOr<u32, LoaderError> {
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u32 value = LOADER_TRY(m_stream->read_value<LittleEndian<u32>>());
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bytes_read += 4;
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return value;
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};
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#define CHECK_OK(category, msg) \
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do { \
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if (!ok) \
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return LoaderError { category, DeprecatedString::formatted("Parsing failed: {}", msg) }; \
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} while (0)
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u32 riff = TRY(read_u32());
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ok = ok && riff == 0x46464952; // "RIFF"
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CHECK_OK(LoaderError::Category::Format, "RIFF header");
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u32 sz = TRY(read_u32());
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ok = ok && sz < maximum_wav_size;
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CHECK_OK(LoaderError::Category::Format, "File size");
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u32 wave = TRY(read_u32());
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ok = ok && wave == 0x45564157; // "WAVE"
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CHECK_OK(LoaderError::Category::Format, "WAVE header");
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u32 fmt_id = TRY(read_u32());
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ok = ok && fmt_id == 0x20746D66; // "fmt "
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CHECK_OK(LoaderError::Category::Format, "FMT header");
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u32 fmt_size = TRY(read_u32());
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ok = ok && (fmt_size == 16 || fmt_size == 18 || fmt_size == 40);
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CHECK_OK(LoaderError::Category::Format, "FMT size");
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u16 audio_format = TRY(read_u16());
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CHECK_OK(LoaderError::Category::Format, "Audio format"); // incomplete read check
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ok = ok && (audio_format == WAVE_FORMAT_PCM || audio_format == WAVE_FORMAT_IEEE_FLOAT || audio_format == WAVE_FORMAT_EXTENSIBLE);
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CHECK_OK(LoaderError::Category::Unimplemented, "Audio format PCM/Float"); // value check
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m_num_channels = TRY(read_u16());
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ok = ok && (m_num_channels == 1 || m_num_channels == 2);
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CHECK_OK(LoaderError::Category::Unimplemented, "Channel count");
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m_sample_rate = TRY(read_u32());
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CHECK_OK(LoaderError::Category::IO, "Sample rate");
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TRY(read_u32());
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CHECK_OK(LoaderError::Category::IO, "Data rate");
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u16 block_size_bytes = TRY(read_u16());
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CHECK_OK(LoaderError::Category::IO, "Block size");
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u16 bits_per_sample = TRY(read_u16());
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CHECK_OK(LoaderError::Category::IO, "Bits per sample");
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if (audio_format == WAVE_FORMAT_EXTENSIBLE) {
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ok = ok && (fmt_size == 40);
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CHECK_OK(LoaderError::Category::Format, "Extensible fmt size"); // value check
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// Discard everything until the GUID.
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// We've already read 16 bytes from the stream. The GUID starts in another 8 bytes.
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TRY(read_u32());
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TRY(read_u32());
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CHECK_OK(LoaderError::Category::IO, "Discard until GUID");
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// Get the underlying audio format from the first two bytes of GUID
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u16 guid_subformat = TRY(read_u16());
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ok = ok && (guid_subformat == WAVE_FORMAT_PCM || guid_subformat == WAVE_FORMAT_IEEE_FLOAT);
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CHECK_OK(LoaderError::Category::Unimplemented, "GUID SubFormat");
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audio_format = guid_subformat;
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}
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if (audio_format == WAVE_FORMAT_PCM) {
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ok = ok && (bits_per_sample == 8 || bits_per_sample == 16 || bits_per_sample == 24);
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CHECK_OK(LoaderError::Category::Unimplemented, "Bits per sample (PCM)"); // value check
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// We only support 8-24 bit audio right now because other formats are uncommon
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if (bits_per_sample == 8) {
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m_sample_format = PcmSampleFormat::Uint8;
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} else if (bits_per_sample == 16) {
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m_sample_format = PcmSampleFormat::Int16;
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} else if (bits_per_sample == 24) {
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m_sample_format = PcmSampleFormat::Int24;
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}
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} else if (audio_format == WAVE_FORMAT_IEEE_FLOAT) {
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ok = ok && (bits_per_sample == 32 || bits_per_sample == 64);
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CHECK_OK(LoaderError::Category::Unimplemented, "Bits per sample (Float)"); // value check
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// Again, only the common 32 and 64 bit
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if (bits_per_sample == 32) {
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m_sample_format = PcmSampleFormat::Float32;
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} else if (bits_per_sample == 64) {
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m_sample_format = PcmSampleFormat::Float64;
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}
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}
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ok = ok && (block_size_bytes == (m_num_channels * (bits_per_sample / 8)));
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CHECK_OK(LoaderError::Category::Format, "Block size sanity check");
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dbgln_if(AWAVLOADER_DEBUG, "WAV format {} at {} bit, {} channels, rate {}Hz ",
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sample_format_name(m_sample_format), pcm_bits_per_sample(m_sample_format), m_num_channels, m_sample_rate);
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// Read chunks until we find DATA
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bool found_data = false;
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u32 data_size = 0;
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u8 search_byte = 0;
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while (true) {
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search_byte = TRY(read_u8());
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CHECK_OK(LoaderError::Category::IO, "Reading byte searching for data");
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if (search_byte != 0x64) // D
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continue;
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search_byte = TRY(read_u8());
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CHECK_OK(LoaderError::Category::IO, "Reading next byte searching for data");
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if (search_byte != 0x61) // A
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continue;
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u16 search_remaining = TRY(read_u16());
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CHECK_OK(LoaderError::Category::IO, "Reading remaining bytes searching for data");
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if (search_remaining != 0x6174) // TA
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continue;
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data_size = TRY(read_u32());
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found_data = true;
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break;
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}
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ok = ok && found_data;
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CHECK_OK(LoaderError::Category::Format, "Found no data chunk");
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ok = ok && data_size < maximum_wav_size;
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CHECK_OK(LoaderError::Category::Format, "Data was too large");
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m_total_samples = data_size / block_size_bytes;
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dbgln_if(AWAVLOADER_DEBUG, "WAV data size {}, bytes per sample {}, total samples {}",
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data_size,
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block_size_bytes,
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m_total_samples);
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m_byte_offset_of_data_samples = bytes_read;
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return {};
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}
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}
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