
Prior code in `WavLoader::get_more_samples()` would attempt to read the requested number of samples without actually checking whether that many samples were remaining in the stream. This was the cause of an audible pop at the end of a track, due to reading non-audio data that is sometimes at the end of a Wave file. Now we only attempt to read up to the end of sample data, but no further. Also, added comments to clarify the meaning of "sample", and how it should be independent of the number of channels.
311 lines
9.5 KiB
C++
311 lines
9.5 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 <AK/Debug.h>
|
|
#include <AK/NumericLimits.h>
|
|
#include <AK/OwnPtr.h>
|
|
#include <LibAudio/Buffer.h>
|
|
#include <LibAudio/WavLoader.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;
|
|
|
|
m_resampler = make<ResampleHelper>(m_sample_rate, 44100);
|
|
}
|
|
|
|
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;
|
|
|
|
m_resampler = make<ResampleHelper>(m_sample_rate, 44100);
|
|
}
|
|
|
|
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_resampler,
|
|
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;
|
|
}
|
|
|
|
ResampleHelper::ResampleHelper(double source, double target)
|
|
: m_ratio(source / target)
|
|
{
|
|
}
|
|
|
|
void ResampleHelper::process_sample(double sample_l, double sample_r)
|
|
{
|
|
m_last_sample_l = sample_l;
|
|
m_last_sample_r = sample_r;
|
|
m_current_ratio += 1;
|
|
}
|
|
|
|
bool ResampleHelper::read_sample(double& next_l, double& next_r)
|
|
{
|
|
if (m_current_ratio > 0) {
|
|
m_current_ratio -= m_ratio;
|
|
next_l = m_last_sample_l;
|
|
next_r = m_last_sample_r;
|
|
return true;
|
|
}
|
|
|
|
return false;
|
|
}
|
|
|
|
}
|