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ladybird/Userland/Libraries/LibAudio/MP3Loader.cpp
kleines Filmröllchen 49b087f3cd LibAudio+Userland: Use new audio queue in client-server communication 
Previously, we were sending Buffers to the server whenever we had new
audio data for it. This meant that for every audio enqueue action, we
needed to create a new shared memory anonymous buffer, send that
buffer's file descriptor over IPC (+recfd on the other side) and then
map the buffer into the audio server's memory to be able to play it.
This was fine for sending large chunks of audio data, like when playing
existing audio files. However, in the future we want to move to
real-time audio in some applications like Piano. This means that the
size of buffers that are sent need to be very small, as just the size of
a buffer itself is part of the audio latency. If we were to try
real-time audio with the existing system, we would run into problems
really quickly. Dealing with a continuous stream of new anonymous files
like the current audio system is rather expensive, as we need Kernel
help in multiple places. Additionally, every enqueue incurs an IPC call,
which are not optimized for >1000 calls/second (which would be needed
for real-time audio with buffer sizes of ~40 samples). So a fundamental
change in how we handle audio sending in userspace is necessary.

This commit moves the audio sending system onto a shared single producer
circular queue (SSPCQ) (introduced with one of the previous commits).
This queue is intended to live in shared memory and be accessed by
multiple processes at the same time. It was specifically written to
support the audio sending case, so e.g. it only supports a single
producer (the audio client). Now, audio sending follows these general
steps:
- The audio client connects to the audio server.
- The audio client creates a SSPCQ in shared memory.
- The audio client sends the SSPCQ's file descriptor to the audio server
  with the set_buffer() IPC call.
- The audio server receives the SSPCQ and maps it.
- The audio client signals start of playback with start_playback().
- At the same time:
  - The audio client writes its audio data into the shared-memory queue.
  - The audio server reads audio data from the shared-memory queue(s).
  Both sides have additional before-queue/after-queue buffers, depending
  on the exact application.
- Pausing playback is just an IPC call, nothing happens to the buffer
  except that the server stops reading from it until playback is
  resumed.
- Muting has nothing to do with whether audio data is read or not.
- When the connection closes, the queues are unmapped on both sides.

This should already improve audio playback performance in a bunch of
places.

Implementation & commit notes:
- Audio loaders don't create LegacyBuffers anymore. LegacyBuffer is kept
  for WavLoader, see previous commit message.
- Most intra-process audio data passing is done with FixedArray<Sample>
  or Vector<Sample>.
- Improvements to most audio-enqueuing applications. (If necessary I can
  try to extract some of the aplay improvements.)
- New APIs on LibAudio/ClientConnection which allows non-realtime
  applications to enqueue audio in big chunks like before.
- Removal of status APIs from the audio server connection for
  information that can be directly obtained from the shared queue.
- Split the pause playback API into two APIs with more intuitive names.

I know this is a large commit, and you can kinda tell from the commit
message. It's basically impossible to break this up without hacks, so
please forgive me. These are some of the best changes to the audio
subsystem and I hope that that makes up for this :yaktangle: commit.

:yakring:
2022-04-21 13:55:00 +02:00

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/*
* Copyright (c) 2021, Arne Elster <arne@elster.li>
*
* SPDX-License-Identifier: BSD-2-Clause
*/
#include "MP3Loader.h"
#include "MP3HuffmanTables.h"
#include "MP3Tables.h"
#include <AK/FixedArray.h>
#include <LibCore/File.h>
#include <LibCore/FileStream.h>
namespace Audio {
LibDSP::MDCT<12> MP3LoaderPlugin::s_mdct_12;
LibDSP::MDCT<36> MP3LoaderPlugin::s_mdct_36;
MP3LoaderPlugin::MP3LoaderPlugin(StringView path)
: m_file(Core::File::construct(path))
{
if (!m_file->open(Core::OpenMode::ReadOnly)) {
m_error = LoaderError { LoaderError::Category::IO, String::formatted("Can't open file: {}", m_file->error_string()) };
return;
}
off_t file_size = 0;
if (!m_file->seek(0, Core::SeekMode::FromEndPosition, &file_size)) {
m_error = LoaderError { LoaderError::Category::IO, "Could not seek in file." };
return;
}
m_file_size = file_size;
if (!m_file->seek(0, Core::SeekMode::SetPosition)) {
m_error = LoaderError { LoaderError::Category::IO, "Could not seek in file." };
return;
}
m_input_stream = make<Core::InputFileStream>(*m_file);
if (!m_input_stream || m_input_stream->has_any_error()) {
m_error = LoaderError { LoaderError::Category::Internal, "Could not create input stream on file." };
return;
}
m_bitstream = make<InputBitStream>(*m_input_stream);
if (!m_bitstream || m_bitstream->has_any_error()) {
m_error = LoaderError { LoaderError::Category::Internal, "Could not create bit stream on top of input stream" };
return;
}
}
MP3LoaderPlugin::~MP3LoaderPlugin()
{
if (m_bitstream)
m_bitstream->handle_any_error();
if (m_input_stream)
m_input_stream->handle_any_error();
}
MaybeLoaderError MP3LoaderPlugin::initialize()
{
if (m_error.has_value())
return m_error.release_value();
TRY(synchronize());
auto header = TRY(read_header());
if (header.id != 1 || header.layer != 3)
return LoaderError { LoaderError::Category::Format, "Only MPEG-1 layer 3 supported." };
m_sample_rate = header.samplerate;
m_num_channels = header.channel_count();
m_loaded_samples = 0;
if (!m_file->seek(0, Core::SeekMode::SetPosition))
return LoaderError { LoaderError::Category::IO, "Could not seek in file." };
TRY(build_seek_table());
if (!m_file->seek(0, Core::SeekMode::SetPosition))
return LoaderError { LoaderError::Category::IO, "Could not seek in file." };
m_bitstream->handle_any_error();
return {};
}
MaybeLoaderError MP3LoaderPlugin::reset()
{
TRY(seek(0));
m_current_frame = {};
m_current_frame_read = 0;
m_synthesis_buffer = {};
m_loaded_samples = 0;
m_bit_reservoir.discard_or_error(m_bit_reservoir.size());
return {};
}
MaybeLoaderError MP3LoaderPlugin::seek(int const position)
{
for (auto const& seek_entry : m_seek_table) {
if (seek_entry.get<1>() >= position) {
m_file->seek(seek_entry.get<0>(), Core::SeekMode::SetPosition);
m_loaded_samples = seek_entry.get<1>();
break;
}
}
m_current_frame = {};
m_current_frame_read = 0;
m_synthesis_buffer = {};
m_bit_reservoir.discard_or_error(m_bit_reservoir.size());
m_input_stream->handle_any_error();
m_bitstream->handle_any_error();
m_bit_reservoir.handle_any_error();
m_is_first_frame = true;
return {};
}
LoaderSamples MP3LoaderPlugin::get_more_samples(size_t max_samples_to_read_from_input)
{
FixedArray<Sample> samples = LOADER_TRY(FixedArray<Sample>::try_create(max_samples_to_read_from_input));
size_t samples_to_read = max_samples_to_read_from_input;
while (samples_to_read > 0) {
if (!m_current_frame.has_value()) {
auto maybe_frame = read_next_frame();
if (maybe_frame.is_error()) {
if (m_input_stream->unreliable_eof()) {
return FixedArray<Sample> {};
}
return maybe_frame.release_error();
}
m_current_frame = maybe_frame.release_value();
if (!m_current_frame.has_value())
break;
m_is_first_frame = false;
m_current_frame_read = 0;
}
bool const is_stereo = m_current_frame->header.channel_count() == 2;
for (; m_current_frame_read < 576 && samples_to_read > 0; m_current_frame_read++) {
auto const left_sample = m_current_frame->channels[0].granules[0].pcm[m_current_frame_read / 32][m_current_frame_read % 32];
auto const right_sample = is_stereo ? m_current_frame->channels[1].granules[0].pcm[m_current_frame_read / 32][m_current_frame_read % 32] : left_sample;
samples[samples.size() - samples_to_read] = Sample { left_sample, right_sample };
samples_to_read--;
}
for (; m_current_frame_read < 1152 && samples_to_read > 0; m_current_frame_read++) {
auto const left_sample = m_current_frame->channels[0].granules[1].pcm[(m_current_frame_read - 576) / 32][(m_current_frame_read - 576) % 32];
auto const right_sample = is_stereo ? m_current_frame->channels[1].granules[1].pcm[(m_current_frame_read - 576) / 32][(m_current_frame_read - 576) % 32] : left_sample;
samples[samples.size() - samples_to_read] = Sample { left_sample, right_sample };
samples_to_read--;
}
if (m_current_frame_read == 1152) {
m_current_frame = {};
}
}
m_loaded_samples += samples.size();
return samples;
}
MaybeLoaderError MP3LoaderPlugin::build_seek_table()
{
int sample_count = 0;
size_t frame_count = 0;
m_seek_table.clear();
m_bitstream->align_to_byte_boundary();
while (!synchronize().is_error()) {
off_t frame_pos = 0;
if (!m_file->seek(0, Core::SeekMode::FromCurrentPosition, &frame_pos))
return LoaderError { LoaderError::Category::IO, String::formatted("Could not get stream position at frame {}.", frame_count) };
frame_pos -= 2;
auto error_or_header = read_header();
if (error_or_header.is_error() || error_or_header.value().id != 1 || error_or_header.value().layer != 3) {
continue;
}
frame_count++;
sample_count += 1152;
if (frame_count % 10 == 0)
m_seek_table.append({ frame_pos, sample_count });
size_t const next_frame_position = error_or_header.value().frame_size - 6;
if (!m_file->seek(error_or_header.value().frame_size - 6, Core::SeekMode::FromCurrentPosition))
return LoaderError { LoaderError::Category::IO, String::formatted("Could not seek to frame {} (stream position {}).", frame_count, next_frame_position) };
// TODO: This is just here to clear the bitstream buffer.
// Bitstream should have a method to sync its state to the underlying stream.
m_bitstream->align_to_byte_boundary();
}
m_total_samples = sample_count;
return {};
}
ErrorOr<MP3::Header, LoaderError> MP3LoaderPlugin::read_header()
{
MP3::Header header;
header.id = m_bitstream->read_bit_big_endian();
header.layer = MP3::Tables::LayerNumberLookup[m_bitstream->read_bits_big_endian(2)];
if (header.layer <= 0)
return LoaderError { LoaderError::Category::Format, m_loaded_samples, "Frame header contains invalid layer number." };
header.protection_bit = m_bitstream->read_bit_big_endian();
header.bitrate = MP3::Tables::BitratesPerLayerLookup[header.layer - 1][m_bitstream->read_bits_big_endian(4)];
if (header.bitrate <= 0)
return LoaderError { LoaderError::Category::Format, m_loaded_samples, "Frame header contains invalid bitrate." };
header.samplerate = MP3::Tables::SampleratesLookup[m_bitstream->read_bits_big_endian(2)];
if (header.samplerate <= 0)
return LoaderError { LoaderError::Category::Format, m_loaded_samples, "Frame header contains invalid samplerate." };
header.padding_bit = m_bitstream->read_bit_big_endian();
header.private_bit = m_bitstream->read_bit_big_endian();
header.mode = static_cast<MP3::Mode>(m_bitstream->read_bits_big_endian(2));
header.mode_extension = static_cast<MP3::ModeExtension>(m_bitstream->read_bits_big_endian(2));
header.copyright_bit = m_bitstream->read_bit_big_endian();
header.original_bit = m_bitstream->read_bit_big_endian();
header.emphasis = static_cast<MP3::Emphasis>(m_bitstream->read_bits_big_endian(2));
if (!header.protection_bit)
header.crc16 = static_cast<u16>(m_bitstream->read_bits_big_endian(16));
header.frame_size = 144 * header.bitrate * 1000 / header.samplerate + header.padding_bit;
header.slot_count = header.frame_size - ((header.channel_count() == 2 ? 32 : 17) + (header.protection_bit ? 0 : 2) + 4);
return header;
}
MaybeLoaderError MP3LoaderPlugin::synchronize()
{
size_t one_counter = 0;
while (one_counter < 12 && !m_bitstream->has_any_error()) {
bool const bit = m_bitstream->read_bit_big_endian();
one_counter = bit ? one_counter + 1 : 0;
if (!bit) {
m_bitstream->align_to_byte_boundary();
}
}
if (one_counter != 12)
return LoaderError { LoaderError::Category::Format, m_loaded_samples, "Failed to synchronize." };
return {};
}
ErrorOr<MP3::MP3Frame, LoaderError> MP3LoaderPlugin::read_next_frame()
{
while (!m_bitstream->has_any_error()) {
TRY(synchronize());
MP3::Header header = TRY(read_header());
if (header.id != 1 || header.layer != 3) {
continue;
}
return read_frame_data(header, m_is_first_frame);
}
return LoaderError { LoaderError::Category::Internal, m_loaded_samples, "Could not find another frame." };
}
ErrorOr<MP3::MP3Frame, LoaderError> MP3LoaderPlugin::read_frame_data(MP3::Header const& header, bool is_first_frame)
{
MP3::MP3Frame frame { header };
TRY(read_side_information(frame));
if (m_bitstream->has_any_error())
return LoaderError { LoaderError::Category::IO, m_loaded_samples, "Read error" };
auto maybe_buffer = ByteBuffer::create_uninitialized(header.slot_count);
if (maybe_buffer.is_error())
return LoaderError { LoaderError::Category::IO, m_loaded_samples, "Out of memory" };
auto& buffer = maybe_buffer.value();
size_t old_reservoir_size = m_bit_reservoir.size();
if (m_bitstream->read(buffer) != buffer.size())
return LoaderError { LoaderError::Category::IO, m_loaded_samples, "Could not read whole frame." };
if (m_bit_reservoir.write(buffer) != header.slot_count)
return LoaderError { LoaderError::Category::IO, m_loaded_samples, "Could not write frame into bit reservoir." };
if (frame.main_data_begin > 0 && is_first_frame)
return frame;
if (!m_bit_reservoir.discard_or_error(old_reservoir_size - frame.main_data_begin))
return LoaderError { LoaderError::Category::IO, m_loaded_samples, "Could not discard old frame data." };
InputBitStream reservoir_stream(m_bit_reservoir);
ScopeGuard reservoir_guard([&reservoir_stream]() {
if (reservoir_stream.has_any_error()) {
reservoir_stream.handle_any_error();
}
});
for (size_t granule_index = 0; granule_index < 2; granule_index++) {
for (size_t channel_index = 0; channel_index < header.channel_count(); channel_index++) {
size_t scale_factor_size = TRY(read_scale_factors(frame, reservoir_stream, granule_index, channel_index));
TRY(read_huffman_data(frame, reservoir_stream, granule_index, channel_index, scale_factor_size));
if (frame.channels[channel_index].granules[granule_index].block_type == MP3::BlockType::Short) {
reorder_samples(frame.channels[channel_index].granules[granule_index], frame.header.samplerate);
// Only reduce alias for lowest 2 bands as they're long.
// Afaik this is not mentioned in the ISO spec, but it is addressed in the
// changelog for the ISO compliance tests.
if (frame.channels[channel_index].granules[granule_index].mixed_block_flag)
reduce_alias(frame.channels[channel_index].granules[granule_index], 36);
} else {
reduce_alias(frame.channels[channel_index].granules[granule_index]);
}
}
if (header.mode == MP3::Mode::JointStereo) {
process_stereo(frame, granule_index);
}
}
for (size_t granule_index = 0; granule_index < 2; granule_index++) {
for (size_t channel_index = 0; channel_index < header.channel_count(); channel_index++) {
auto& granule = frame.channels[channel_index].granules[granule_index];
for (size_t i = 0; i < 576; i += 18) {
MP3::BlockType block_type = granule.block_type;
if (i < 36 && granule.mixed_block_flag) {
// ISO/IEC 11172-3: if mixed_block_flag is set, the lowest two subbands are transformed with normal window.
block_type = MP3::BlockType::Normal;
}
Array<double, 36> output;
transform_samples_to_time(granule.samples, i, output, block_type);
int const subband_index = i / 18;
for (size_t sample_index = 0; sample_index < 18; sample_index++) {
// overlap add
granule.filter_bank_input[subband_index][sample_index] = output[sample_index] + m_last_values[channel_index][subband_index][sample_index];
m_last_values[channel_index][subband_index][sample_index] = output[sample_index + 18];
// frequency inversion
if (subband_index % 2 == 1 && sample_index % 2 == 1)
granule.filter_bank_input[subband_index][sample_index] *= -1;
}
}
}
}
Array<double, 32> in_samples;
for (size_t channel_index = 0; channel_index < frame.header.channel_count(); channel_index++) {
for (size_t granule_index = 0; granule_index < 2; granule_index++) {
auto& granule = frame.channels[channel_index].granules[granule_index];
for (size_t sample_index = 0; sample_index < 18; sample_index++) {
for (size_t band_index = 0; band_index < 32; band_index++) {
in_samples[band_index] = granule.filter_bank_input[band_index][sample_index];
}
synthesis(m_synthesis_buffer[channel_index], in_samples, granule.pcm[sample_index]);
}
}
}
return frame;
}
MaybeLoaderError MP3LoaderPlugin::read_side_information(MP3::MP3Frame& frame)
{
frame.main_data_begin = m_bitstream->read_bits_big_endian(9);
if (frame.header.channel_count() == 1) {
frame.private_bits = m_bitstream->read_bits_big_endian(5);
} else {
frame.private_bits = m_bitstream->read_bits_big_endian(3);
}
for (size_t channel_index = 0; channel_index < frame.header.channel_count(); channel_index++) {
for (size_t scale_factor_selection_info_band = 0; scale_factor_selection_info_band < 4; scale_factor_selection_info_band++) {
frame.channels[channel_index].scale_factor_selection_info[scale_factor_selection_info_band] = m_bitstream->read_bit_big_endian();
}
}
for (size_t granule_index = 0; granule_index < 2; granule_index++) {
for (size_t channel_index = 0; channel_index < frame.header.channel_count(); channel_index++) {
auto& granule = frame.channels[channel_index].granules[granule_index];
granule.part_2_3_length = m_bitstream->read_bits_big_endian(12);
granule.big_values = m_bitstream->read_bits_big_endian(9);
granule.global_gain = m_bitstream->read_bits_big_endian(8);
granule.scalefac_compress = m_bitstream->read_bits_big_endian(4);
granule.window_switching_flag = m_bitstream->read_bit_big_endian();
if (granule.window_switching_flag) {
granule.block_type = static_cast<MP3::BlockType>(m_bitstream->read_bits_big_endian(2));
granule.mixed_block_flag = m_bitstream->read_bit_big_endian();
for (size_t region = 0; region < 2; region++)
granule.table_select[region] = m_bitstream->read_bits_big_endian(5);
for (size_t window = 0; window < 3; window++)
granule.sub_block_gain[window] = m_bitstream->read_bits_big_endian(3);
granule.region0_count = (granule.block_type == MP3::BlockType::Short && !granule.mixed_block_flag) ? 8 : 7;
granule.region1_count = 36;
} else {
for (size_t region = 0; region < 3; region++)
granule.table_select[region] = m_bitstream->read_bits_big_endian(5);
granule.region0_count = m_bitstream->read_bits_big_endian(4);
granule.region1_count = m_bitstream->read_bits_big_endian(3);
}
granule.preflag = m_bitstream->read_bit_big_endian();
granule.scalefac_scale = m_bitstream->read_bit_big_endian();
granule.count1table_select = m_bitstream->read_bit_big_endian();
}
}
if (m_bitstream->has_any_error())
return LoaderError { LoaderError::Category::IO, m_loaded_samples, "Read error" };
return {};
}
// From ISO/IEC 11172-3 (2.4.3.4.7.1)
Array<double, 576> MP3LoaderPlugin::calculate_frame_exponents(MP3::MP3Frame const& frame, size_t granule_index, size_t channel_index)
{
Array<double, 576> exponents;
auto fill_band = [&exponents](double exponent, size_t start, size_t end) {
for (size_t j = start; j <= end; j++) {
exponents[j] = exponent;
}
};
auto const& channel = frame.channels[channel_index];
auto const& granule = frame.channels[channel_index].granules[granule_index];
auto const scale_factor_bands = get_scalefactor_bands(granule, frame.header.samplerate);
double const scale_factor_multiplier = granule.scalefac_scale ? 1 : 0.5;
int const gain = granule.global_gain - 210;
if (granule.block_type != MP3::BlockType::Short) {
for (size_t band_index = 0; band_index < 22; band_index++) {
double const exponent = gain / 4.0 - (scale_factor_multiplier * (channel.scale_factors[band_index] + granule.preflag * MP3::Tables::Pretab[band_index]));
fill_band(AK::pow(2.0, exponent), scale_factor_bands[band_index].start, scale_factor_bands[band_index].end);
}
} else {
size_t band_index = 0;
size_t sample_count = 0;
if (granule.mixed_block_flag) {
while (sample_count < 36) {
double const exponent = gain / 4.0 - (scale_factor_multiplier * (channel.scale_factors[band_index] + granule.preflag * MP3::Tables::Pretab[band_index]));
fill_band(AK::pow(2.0, exponent), scale_factor_bands[band_index].start, scale_factor_bands[band_index].end);
sample_count += scale_factor_bands[band_index].width;
band_index++;
}
}
double const gain0 = (gain - 8 * granule.sub_block_gain[0]) / 4.0;
double const gain1 = (gain - 8 * granule.sub_block_gain[1]) / 4.0;
double const gain2 = (gain - 8 * granule.sub_block_gain[2]) / 4.0;
while (sample_count < 576 && band_index < scale_factor_bands.size()) {
double const exponent0 = gain0 - (scale_factor_multiplier * channel.scale_factors[band_index + 0]);
double const exponent1 = gain1 - (scale_factor_multiplier * channel.scale_factors[band_index + 1]);
double const exponent2 = gain2 - (scale_factor_multiplier * channel.scale_factors[band_index + 2]);
fill_band(AK::pow(2.0, exponent0), scale_factor_bands[band_index + 0].start, scale_factor_bands[band_index + 0].end);
sample_count += scale_factor_bands[band_index + 0].width;
fill_band(AK::pow(2.0, exponent1), scale_factor_bands[band_index + 1].start, scale_factor_bands[band_index + 1].end);
sample_count += scale_factor_bands[band_index + 1].width;
fill_band(AK::pow(2.0, exponent2), scale_factor_bands[band_index + 2].start, scale_factor_bands[band_index + 2].end);
sample_count += scale_factor_bands[band_index + 2].width;
band_index += 3;
}
while (sample_count < 576)
exponents[sample_count++] = 0;
}
return exponents;
}
ErrorOr<size_t, LoaderError> MP3LoaderPlugin::read_scale_factors(MP3::MP3Frame& frame, InputBitStream& reservoir, size_t granule_index, size_t channel_index)
{
auto& channel = frame.channels[channel_index];
auto const& granule = channel.granules[granule_index];
size_t band_index = 0;
size_t bits_read = 0;
if (granule.window_switching_flag && granule.block_type == MP3::BlockType::Short) {
if (granule.mixed_block_flag) {
for (size_t i = 0; i < 8; i++) {
auto const bits = MP3::Tables::ScalefacCompressSlen1[granule.scalefac_compress];
channel.scale_factors[band_index++] = reservoir.read_bits_big_endian(bits);
bits_read += bits;
}
for (size_t i = 3; i < 12; i++) {
auto const bits = i <= 5 ? MP3::Tables::ScalefacCompressSlen1[granule.scalefac_compress] : MP3::Tables::ScalefacCompressSlen2[granule.scalefac_compress];
channel.scale_factors[band_index++] = reservoir.read_bits_big_endian(bits);
channel.scale_factors[band_index++] = reservoir.read_bits_big_endian(bits);
channel.scale_factors[band_index++] = reservoir.read_bits_big_endian(bits);
bits_read += 3 * bits;
}
} else {
for (size_t i = 0; i < 12; i++) {
auto const bits = i <= 5 ? MP3::Tables::ScalefacCompressSlen1[granule.scalefac_compress] : MP3::Tables::ScalefacCompressSlen2[granule.scalefac_compress];
channel.scale_factors[band_index++] = reservoir.read_bits_big_endian(bits);
channel.scale_factors[band_index++] = reservoir.read_bits_big_endian(bits);
channel.scale_factors[band_index++] = reservoir.read_bits_big_endian(bits);
bits_read += 3 * bits;
}
}
channel.scale_factors[band_index++] = 0;
channel.scale_factors[band_index++] = 0;
channel.scale_factors[band_index++] = 0;
} else {
if ((channel.scale_factor_selection_info[0] == 0) || (granule_index == 0)) {
for (band_index = 0; band_index < 6; band_index++) {
auto const bits = MP3::Tables::ScalefacCompressSlen1[granule.scalefac_compress];
channel.scale_factors[band_index] = reservoir.read_bits_big_endian(bits);
bits_read += bits;
}
}
if ((channel.scale_factor_selection_info[1] == 0) || (granule_index == 0)) {
for (band_index = 6; band_index < 11; band_index++) {
auto const bits = MP3::Tables::ScalefacCompressSlen1[granule.scalefac_compress];
channel.scale_factors[band_index] = reservoir.read_bits_big_endian(bits);
bits_read += bits;
}
}
if ((channel.scale_factor_selection_info[2] == 0) || (granule_index == 0)) {
for (band_index = 11; band_index < 16; band_index++) {
auto const bits = MP3::Tables::ScalefacCompressSlen2[granule.scalefac_compress];
channel.scale_factors[band_index] = reservoir.read_bits_big_endian(bits);
bits_read += bits;
}
}
if ((channel.scale_factor_selection_info[3] == 0) || (granule_index == 0)) {
for (band_index = 16; band_index < 21; band_index++) {
auto const bits = MP3::Tables::ScalefacCompressSlen2[granule.scalefac_compress];
channel.scale_factors[band_index] = reservoir.read_bits_big_endian(bits);
bits_read += bits;
}
}
channel.scale_factors[21] = 0;
}
if (reservoir.has_any_error())
return LoaderError { LoaderError::Category::IO, m_loaded_samples, "Read error" };
return bits_read;
}
MaybeLoaderError MP3LoaderPlugin::read_huffman_data(MP3::MP3Frame& frame, InputBitStream& reservoir, size_t granule_index, size_t channel_index, size_t granule_bits_read)
{
auto const exponents = calculate_frame_exponents(frame, granule_index, channel_index);
auto& granule = frame.channels[channel_index].granules[granule_index];
auto const scale_factor_bands = get_scalefactor_bands(granule, frame.header.samplerate);
size_t const scale_factor_band_index1 = granule.region0_count + 1;
size_t const scale_factor_band_index2 = min(scale_factor_bands.size() - 1, scale_factor_band_index1 + granule.region1_count + 1);
bool const is_short_granule = granule.window_switching_flag && granule.block_type == MP3::BlockType::Short;
size_t const region1_start = is_short_granule ? 36 : scale_factor_bands[scale_factor_band_index1].start;
size_t const region2_start = is_short_granule ? 576 : scale_factor_bands[scale_factor_band_index2].start;
auto requantize = [](int const sample, double const exponent) -> double {
int const sign = sample < 0 ? -1 : 1;
int const magnitude = AK::abs(sample);
return sign * AK::pow(static_cast<double>(magnitude), 4 / 3.0) * exponent;
};
size_t count = 0;
for (; count < granule.big_values * 2; count += 2) {
MP3::Tables::Huffman::HuffmanTreeXY const* tree = nullptr;
if (count < region1_start) {
tree = &MP3::Tables::Huffman::HuffmanTreesXY[granule.table_select[0]];
} else if (count < region2_start) {
tree = &MP3::Tables::Huffman::HuffmanTreesXY[granule.table_select[1]];
} else {
tree = &MP3::Tables::Huffman::HuffmanTreesXY[granule.table_select[2]];
}
if (!tree || tree->nodes.is_empty()) {
return LoaderError { LoaderError::Category::Format, m_loaded_samples, "Frame references invalid huffman table." };
}
// Assumption: There's enough bits to read. 32 is just a placeholder for "unlimited".
// There are no 32 bit long huffman codes in the tables.
auto const entry = MP3::Tables::Huffman::huffman_decode(reservoir, tree->nodes, 32);
granule_bits_read += entry.bits_read;
if (!entry.code.has_value())
return LoaderError { LoaderError::Category::Format, m_loaded_samples, "Frame contains invalid huffman data." };
int x = entry.code->symbol.x;
int y = entry.code->symbol.y;
if (x == 15 && tree->linbits > 0) {
x += reservoir.read_bits_big_endian(tree->linbits);
granule_bits_read += tree->linbits;
}
if (x != 0) {
if (reservoir.read_bit_big_endian())
x = -x;
granule_bits_read++;
}
if (y == 15 && tree->linbits > 0) {
y += reservoir.read_bits_big_endian(tree->linbits);
granule_bits_read += tree->linbits;
}
if (y != 0) {
if (reservoir.read_bit_big_endian())
y = -y;
granule_bits_read++;
}
granule.samples[count + 0] = requantize(x, exponents[count + 0]);
granule.samples[count + 1] = requantize(y, exponents[count + 1]);
}
Span<MP3::Tables::Huffman::HuffmanNode<MP3::Tables::Huffman::HuffmanVWXY> const> count1table = granule.count1table_select ? MP3::Tables::Huffman::TreeB : MP3::Tables::Huffman::TreeA;
// count1 is not known. We have to read huffman encoded values
// until we've exhausted the granule's bits. We know the size of
// the granule from part2_3_length, which is the number of bits
// used for scaleactors and huffman data (in the granule).
while (granule_bits_read < granule.part_2_3_length && count <= 576 - 4) {
auto const entry = MP3::Tables::Huffman::huffman_decode(reservoir, count1table, granule.part_2_3_length - granule_bits_read);
granule_bits_read += entry.bits_read;
if (!entry.code.has_value())
return LoaderError { LoaderError::Category::Format, m_loaded_samples, "Frame contains invalid huffman data." };
int v = entry.code->symbol.v;
if (v != 0) {
if (granule_bits_read >= granule.part_2_3_length)
break;
if (reservoir.read_bit_big_endian())
v = -v;
granule_bits_read++;
}
int w = entry.code->symbol.w;
if (w != 0) {
if (granule_bits_read >= granule.part_2_3_length)
break;
if (reservoir.read_bit_big_endian())
w = -w;
granule_bits_read++;
}
int x = entry.code->symbol.x;
if (x != 0) {
if (granule_bits_read >= granule.part_2_3_length)
break;
if (reservoir.read_bit_big_endian())
x = -x;
granule_bits_read++;
}
int y = entry.code->symbol.y;
if (y != 0) {
if (granule_bits_read >= granule.part_2_3_length)
break;
if (reservoir.read_bit_big_endian())
y = -y;
granule_bits_read++;
}
granule.samples[count + 0] = requantize(v, exponents[count + 0]);
granule.samples[count + 1] = requantize(w, exponents[count + 1]);
granule.samples[count + 2] = requantize(x, exponents[count + 2]);
granule.samples[count + 3] = requantize(y, exponents[count + 3]);
count += 4;
}
if (granule_bits_read > granule.part_2_3_length) {
return LoaderError { LoaderError::Category::Format, m_loaded_samples, "Read too many bits from bit reservoir." };
}
for (size_t i = granule_bits_read; i < granule.part_2_3_length; i++) {
reservoir.read_bit_big_endian();
}
return {};
}
void MP3LoaderPlugin::reorder_samples(MP3::Granule& granule, u32 sample_rate)
{
double tmp[576] = {};
size_t band_index = 0;
size_t subband_index = 0;
auto scale_factor_bands = get_scalefactor_bands(granule, sample_rate);
if (granule.mixed_block_flag) {
while (subband_index < 36) {
for (size_t frequency_line_index = 0; frequency_line_index < scale_factor_bands[band_index].width; frequency_line_index++) {
tmp[subband_index] = granule.samples[subband_index];
subband_index++;
}
band_index++;
}
}
while (subband_index < 576 && band_index <= 36) {
for (size_t frequency_line_index = 0; frequency_line_index < scale_factor_bands[band_index].width; frequency_line_index++) {
tmp[subband_index++] = granule.samples[scale_factor_bands[band_index + 0].start + frequency_line_index];
tmp[subband_index++] = granule.samples[scale_factor_bands[band_index + 1].start + frequency_line_index];
tmp[subband_index++] = granule.samples[scale_factor_bands[band_index + 2].start + frequency_line_index];
}
band_index += 3;
}
for (size_t i = 0; i < 576; i++)
granule.samples[i] = tmp[i];
}
void MP3LoaderPlugin::reduce_alias(MP3::Granule& granule, size_t max_subband_index)
{
for (size_t subband = 0; subband < max_subband_index - 18; subband += 18) {
for (size_t i = 0; i < 8; i++) {
size_t const idx1 = subband + 17 - i;
size_t const idx2 = subband + 18 + i;
auto const d1 = granule.samples[idx1];
auto const d2 = granule.samples[idx2];
granule.samples[idx1] = d1 * MP3::Tables::AliasReductionCs[i] - d2 * MP3::Tables::AliasReductionCa[i];
granule.samples[idx2] = d2 * MP3::Tables::AliasReductionCs[i] + d1 * MP3::Tables::AliasReductionCa[i];
}
}
}
void MP3LoaderPlugin::process_stereo(MP3::MP3Frame& frame, size_t granule_index)
{
size_t band_index_ms_start = 0;
size_t band_index_ms_end = 0;
size_t band_index_intensity_start = 0;
size_t band_index_intensity_end = 0;
auto& granule_left = frame.channels[0].granules[granule_index];
auto& granule_right = frame.channels[1].granules[granule_index];
auto get_last_nonempty_band = [](Span<double> samples, Span<MP3::Tables::ScaleFactorBand const> bands) -> size_t {
size_t last_nonempty_band = 0;
for (size_t i = 0; i < bands.size(); i++) {
bool is_empty = true;
for (size_t l = bands[i].start; l < bands[i].end; l++) {
if (samples[l] != 0) {
is_empty = false;
break;
}
}
if (!is_empty)
last_nonempty_band = i;
}
return last_nonempty_band;
};
auto process_ms_stereo = [&](MP3::Tables::ScaleFactorBand const& band) {
double const SQRT_2 = AK::sqrt(2.0);
for (size_t i = band.start; i <= band.end; i++) {
double const m = granule_left.samples[i];
double const s = granule_right.samples[i];
granule_left.samples[i] = (m + s) / SQRT_2;
granule_right.samples[i] = (m - s) / SQRT_2;
}
};
auto process_intensity_stereo = [&](MP3::Tables::ScaleFactorBand const& band, double intensity_stereo_ratio) {
for (size_t i = band.start; i <= band.end; i++) {
double const sample_left = granule_left.samples[i];
double const coeff_l = intensity_stereo_ratio / (1 + intensity_stereo_ratio);
double const coeff_r = 1 / (1 + intensity_stereo_ratio);
granule_left.samples[i] = sample_left * coeff_l;
granule_right.samples[i] = sample_left * coeff_r;
}
};
auto scale_factor_bands = get_scalefactor_bands(granule_right, frame.header.samplerate);
if (has_flag(frame.header.mode_extension, MP3::ModeExtension::MsStereo)) {
band_index_ms_start = 0;
band_index_ms_end = scale_factor_bands.size();
}
if (has_flag(frame.header.mode_extension, MP3::ModeExtension::IntensityStereo)) {
band_index_intensity_start = get_last_nonempty_band(granule_right.samples, scale_factor_bands);
band_index_intensity_end = scale_factor_bands.size();
band_index_ms_end = band_index_intensity_start;
}
for (size_t band_index = band_index_ms_start; band_index < band_index_ms_end; band_index++) {
process_ms_stereo(scale_factor_bands[band_index]);
}
for (size_t band_index = band_index_intensity_start; band_index < band_index_intensity_end; band_index++) {
auto const intensity_stereo_position = frame.channels[1].scale_factors[band_index];
if (intensity_stereo_position == 7) {
if (has_flag(frame.header.mode_extension, MP3::ModeExtension::MsStereo))
process_ms_stereo(scale_factor_bands[band_index]);
continue;
}
double const intensity_stereo_ratio = AK::tan(intensity_stereo_position * AK::Pi<double> / 12);
process_intensity_stereo(scale_factor_bands[band_index], intensity_stereo_ratio);
}
}
void MP3LoaderPlugin::transform_samples_to_time(Array<double, 576> const& input, size_t input_offset, Array<double, 36>& output, MP3::BlockType block_type)
{
if (block_type == MP3::BlockType::Short) {
size_t const N = 12;
Array<double, N * 3> temp_out;
Array<double, N / 2> temp_in;
for (size_t k = 0; k < N / 2; k++)
temp_in[k] = input[input_offset + 3 * k + 0];
s_mdct_12.transform(temp_in, Span<double>(temp_out).slice(0, N));
for (size_t i = 0; i < N; i++)
temp_out[i + 0] *= MP3::Tables::WindowBlockTypeShort[i];
for (size_t k = 0; k < N / 2; k++)
temp_in[k] = input[input_offset + 3 * k + 1];
s_mdct_12.transform(temp_in, Span<double>(temp_out).slice(12, N));
for (size_t i = 0; i < N; i++)
temp_out[i + 12] *= MP3::Tables::WindowBlockTypeShort[i];
for (size_t k = 0; k < N / 2; k++)
temp_in[k] = input[input_offset + 3 * k + 2];
s_mdct_12.transform(temp_in, Span<double>(temp_out).slice(24, N));
for (size_t i = 0; i < N; i++)
temp_out[i + 24] *= MP3::Tables::WindowBlockTypeShort[i];
Span<double> idmct1 = Span<double>(temp_out).slice(0, 12);
Span<double> idmct2 = Span<double>(temp_out).slice(12, 12);
Span<double> idmct3 = Span<double>(temp_out).slice(24, 12);
for (size_t i = 0; i < 6; i++)
output[i] = 0;
for (size_t i = 6; i < 12; i++)
output[i] = idmct1[i - 6];
for (size_t i = 12; i < 18; i++)
output[i] = idmct1[i - 6] + idmct2[i - 12];
for (size_t i = 18; i < 24; i++)
output[i] = idmct2[i - 12] + idmct3[i - 18];
for (size_t i = 24; i < 30; i++)
output[i] = idmct3[i - 18];
for (size_t i = 30; i < 36; i++)
output[i] = 0;
} else {
s_mdct_36.transform(Span<double const>(input).slice(input_offset, 18), output);
for (size_t i = 0; i < 36; i++) {
switch (block_type) {
case MP3::BlockType::Normal:
output[i] *= MP3::Tables::WindowBlockTypeNormal[i];
break;
case MP3::BlockType::Start:
output[i] *= MP3::Tables::WindowBlockTypeStart[i];
break;
case MP3::BlockType::End:
output[i] *= MP3::Tables::WindowBlockTypeEnd[i];
break;
case MP3::BlockType::Short:
VERIFY_NOT_REACHED();
break;
}
}
}
}
// ISO/IEC 11172-3 (Figure A.2)
void MP3LoaderPlugin::synthesis(Array<double, 1024>& V, Array<double, 32>& samples, Array<double, 32>& result)
{
for (size_t i = 1023; i >= 64; i--) {
V[i] = V[i - 64];
}
for (size_t i = 0; i < 64; i++) {
V[i] = 0;
for (size_t k = 0; k < 32; k++) {
double const N = MP3::Tables::SynthesisSubbandFilterCoefficients[i][k];
V[i] += N * samples[k];
}
}
Array<double, 512> U;
for (size_t i = 0; i < 8; i++) {
for (size_t j = 0; j < 32; j++) {
U[i * 64 + j] = V[i * 128 + j];
U[i * 64 + 32 + j] = V[i * 128 + 96 + j];
}
}
Array<double, 512> W;
for (size_t i = 0; i < 512; i++) {
W[i] = U[i] * MP3::Tables::WindowSynthesis[i];
}
for (size_t j = 0; j < 32; j++) {
result[j] = 0;
for (size_t k = 0; k < 16; k++) {
result[j] += W[j + 32 * k];
}
}
}
Span<MP3::Tables::ScaleFactorBand const> MP3LoaderPlugin::get_scalefactor_bands(MP3::Granule const& granule, int samplerate)
{
switch (granule.block_type) {
case MP3::BlockType::Short:
switch (samplerate) {
case 32000:
return granule.mixed_block_flag ? MP3::Tables::ScaleFactorBandMixed32000 : MP3::Tables::ScaleFactorBandShort32000;
case 44100:
return granule.mixed_block_flag ? MP3::Tables::ScaleFactorBandMixed44100 : MP3::Tables::ScaleFactorBandShort44100;
case 48000:
return granule.mixed_block_flag ? MP3::Tables::ScaleFactorBandMixed48000 : MP3::Tables::ScaleFactorBandShort48000;
}
break;
case MP3::BlockType::Normal:
[[fallthrough]];
case MP3::BlockType::Start:
[[fallthrough]];
case MP3::BlockType::End:
switch (samplerate) {
case 32000:
return MP3::Tables::ScaleFactorBandLong32000;
case 44100:
return MP3::Tables::ScaleFactorBandLong44100;
case 48000:
return MP3::Tables::ScaleFactorBandLong48000;
}
}
VERIFY_NOT_REACHED();
}
}
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