ladybird/Userland/Libraries/LibAudio/WavLoader.cpp
kleines Filmröllchen d049626f40 Userland+LibAudio: Make audio applications support dynamic sample rate
All audio applications (aplay, Piano, Sound Player) respect the ability
of the system to have theoretically any sample rate. Therefore, they
resample their own audio into the system sample rate.

LibAudio previously had its loaders resample their own audio, even
though they expose their sample rate. This is now changed. The loaders
output audio data in their file's sample rate, which the user has to
query and resample appropriately. Resampling code from Buffer, WavLoader
and FlacLoader is removed.

Note that these applications only check the sample rate at startup,
which is reasonable (the user has to restart applications when changing
the sample rate). Fully dynamic adaptation could both lead to errors and
will require another IPC interface. This seems to be enough for now.
2021-08-27 23:35:27 +04:30

282 lines
8.8 KiB
C++

/*
* Copyright (c) 2018-2020, Andreas Kling <kling@serenityos.org>
* Copyright (c) 2021, kleines Filmröllchen <malu.bertsch@gmail.com>
*
* SPDX-License-Identifier: BSD-2-Clause
*/
#include "WavLoader.h"
#include "Buffer.h"
#include <AK/Debug.h>
#include <AK/NumericLimits.h>
#include <AK/OwnPtr.h>
#include <LibCore/File.h>
#include <LibCore/FileStream.h>
namespace Audio {
static constexpr size_t maximum_wav_size = 1 * GiB; // FIXME: is there a more appropriate size limit?
WavLoaderPlugin::WavLoaderPlugin(const StringView& path)
: m_file(Core::File::construct(path))
{
if (!m_file->open(Core::OpenMode::ReadOnly)) {
m_error_string = String::formatted("Can't open file: {}", m_file->error_string());
return;
}
m_stream = make<Core::InputFileStream>(*m_file);
valid = parse_header();
if (!valid)
return;
}
WavLoaderPlugin::WavLoaderPlugin(const ByteBuffer& buffer)
{
m_stream = make<InputMemoryStream>(buffer);
if (!m_stream) {
m_error_string = String::formatted("Can't open memory stream");
return;
}
m_memory_stream = static_cast<InputMemoryStream*>(m_stream.ptr());
valid = parse_header();
if (!valid)
return;
}
RefPtr<Buffer> WavLoaderPlugin::get_more_samples(size_t max_bytes_to_read_from_input)
{
if (!m_stream)
return nullptr;
int remaining_samples = m_total_samples - m_loaded_samples;
if (remaining_samples <= 0) {
return nullptr;
}
// One "sample" contains data from all channels.
// In the Wave spec, this is also called a block.
size_t bytes_per_sample = m_num_channels * pcm_bits_per_sample(m_sample_format) / 8;
// Might truncate if not evenly divisible by the sample size
int max_samples_to_read = static_cast<int>(max_bytes_to_read_from_input) / bytes_per_sample;
int samples_to_read = min(max_samples_to_read, remaining_samples);
size_t bytes_to_read = samples_to_read * bytes_per_sample;
dbgln_if(AWAVLOADER_DEBUG, "Read {} bytes WAV with num_channels {} sample rate {}, "
"bits per sample {}, sample format {}",
bytes_to_read, m_num_channels, m_sample_rate,
pcm_bits_per_sample(m_sample_format), sample_format_name(m_sample_format));
ByteBuffer sample_data = ByteBuffer::create_zeroed(bytes_to_read);
m_stream->read_or_error(sample_data.bytes());
if (m_stream->handle_any_error()) {
return nullptr;
}
RefPtr<Buffer> buffer = Buffer::from_pcm_data(
sample_data.bytes(),
m_num_channels,
m_sample_format);
// m_loaded_samples should contain the amount of actually loaded samples
m_loaded_samples += samples_to_read;
return buffer;
}
void WavLoaderPlugin::seek(const int sample_index)
{
dbgln_if(AWAVLOADER_DEBUG, "seek sample_index {}", sample_index);
if (sample_index < 0 || sample_index >= m_total_samples)
return;
size_t sample_offset = m_byte_offset_of_data_samples + (sample_index * m_num_channels * (pcm_bits_per_sample(m_sample_format) / 8));
// AK::InputStream does not define seek, hence the special-cases for file and stream.
if (m_file) {
m_file->seek(sample_offset);
} else {
m_memory_stream->seek(sample_offset);
}
m_loaded_samples = sample_index;
}
// Specification reference: http://www-mmsp.ece.mcgill.ca/Documents/AudioFormats/WAVE/WAVE.html
bool WavLoaderPlugin::parse_header()
{
if (!m_stream)
return false;
bool ok = true;
size_t bytes_read = 0;
auto read_u8 = [&]() -> u8 {
u8 value;
*m_stream >> value;
if (m_stream->handle_any_error())
ok = false;
bytes_read += 1;
return value;
};
auto read_u16 = [&]() -> u16 {
u16 value;
*m_stream >> value;
if (m_stream->handle_any_error())
ok = false;
bytes_read += 2;
return value;
};
auto read_u32 = [&]() -> u32 {
u32 value;
*m_stream >> value;
if (m_stream->handle_any_error())
ok = false;
bytes_read += 4;
return value;
};
#define CHECK_OK(msg) \
do { \
if (!ok) { \
m_error_string = String::formatted("Parsing failed: {}", msg); \
dbgln_if(AWAVLOADER_DEBUG, m_error_string); \
return {}; \
} \
} while (0)
u32 riff = read_u32();
ok = ok && riff == 0x46464952; // "RIFF"
CHECK_OK("RIFF header");
u32 sz = read_u32();
ok = ok && sz < maximum_wav_size;
CHECK_OK("File size");
u32 wave = read_u32();
ok = ok && wave == 0x45564157; // "WAVE"
CHECK_OK("WAVE header");
u32 fmt_id = read_u32();
ok = ok && fmt_id == 0x20746D66; // "fmt "
CHECK_OK("FMT header");
u32 fmt_size = read_u32();
ok = ok && (fmt_size == 16 || fmt_size == 18 || fmt_size == 40);
CHECK_OK("FMT size");
u16 audio_format = read_u16();
CHECK_OK("Audio format"); // incomplete read check
ok = ok && (audio_format == WAVE_FORMAT_PCM || audio_format == WAVE_FORMAT_IEEE_FLOAT || audio_format == WAVE_FORMAT_EXTENSIBLE);
CHECK_OK("Audio format PCM/Float"); // value check
m_num_channels = read_u16();
ok = ok && (m_num_channels == 1 || m_num_channels == 2);
CHECK_OK("Channel count");
m_sample_rate = read_u32();
CHECK_OK("Sample rate");
read_u32();
CHECK_OK("Data rate");
u16 block_size_bytes = read_u16();
CHECK_OK("Block size");
u16 bits_per_sample = read_u16();
CHECK_OK("Bits per sample");
if (audio_format == WAVE_FORMAT_EXTENSIBLE) {
ok = ok && (fmt_size == 40);
CHECK_OK("Extensible fmt size"); // value check
// Discard everything until the GUID.
// We've already read 16 bytes from the stream. The GUID starts in another 8 bytes.
read_u32();
read_u32();
CHECK_OK("Discard until GUID");
// Get the underlying audio format from the first two bytes of GUID
u16 guid_subformat = read_u16();
ok = ok && (guid_subformat == WAVE_FORMAT_PCM || guid_subformat == WAVE_FORMAT_IEEE_FLOAT);
CHECK_OK("GUID SubFormat");
audio_format = guid_subformat;
}
if (audio_format == WAVE_FORMAT_PCM) {
ok = ok && (bits_per_sample == 8 || bits_per_sample == 16 || bits_per_sample == 24);
CHECK_OK("Bits per sample (PCM)"); // value check
// We only support 8-24 bit audio right now because other formats are uncommon
if (bits_per_sample == 8) {
m_sample_format = PcmSampleFormat::Uint8;
} else if (bits_per_sample == 16) {
m_sample_format = PcmSampleFormat::Int16;
} else if (bits_per_sample == 24) {
m_sample_format = PcmSampleFormat::Int24;
}
} else if (audio_format == WAVE_FORMAT_IEEE_FLOAT) {
ok = ok && (bits_per_sample == 32 || bits_per_sample == 64);
CHECK_OK("Bits per sample (Float)"); // value check
// Again, only the common 32 and 64 bit
if (bits_per_sample == 32) {
m_sample_format = PcmSampleFormat::Float32;
} else if (bits_per_sample == 64) {
m_sample_format = PcmSampleFormat::Float64;
}
}
ok = ok && (block_size_bytes == (m_num_channels * (bits_per_sample / 8)));
CHECK_OK("Block size sanity check");
dbgln_if(AWAVLOADER_DEBUG, "WAV format {} at {} bit, {} channels, rate {}Hz ",
sample_format_name(m_sample_format), pcm_bits_per_sample(m_sample_format), m_num_channels, m_sample_rate);
// Read chunks until we find DATA
bool found_data = false;
u32 data_sz = 0;
u8 search_byte = 0;
while (true) {
search_byte = read_u8();
CHECK_OK("Reading byte searching for data");
if (search_byte != 0x64) //D
continue;
search_byte = read_u8();
CHECK_OK("Reading next byte searching for data");
if (search_byte != 0x61) //A
continue;
u16 search_remaining = read_u16();
CHECK_OK("Reading remaining bytes searching for data");
if (search_remaining != 0x6174) //TA
continue;
data_sz = read_u32();
found_data = true;
break;
}
ok = ok && found_data;
CHECK_OK("Found no data chunk");
ok = ok && data_sz < maximum_wav_size;
CHECK_OK("Data was too large");
m_total_samples = data_sz / block_size_bytes;
dbgln_if(AWAVLOADER_DEBUG, "WAV data size {}, bytes per sample {}, total samples {}",
data_sz,
block_size_bytes,
m_total_samples);
m_byte_offset_of_data_samples = bytes_read;
return true;
}
}