LibMedia: Remove the now-unused audio loader plugins

This commit is contained in:
Zaggy1024
2025-12-15 17:11:38 -06:00
committed by Gregory Bertilson
parent 07b5ac5db0
commit 9d6bc89ed7
Notes: github-actions[bot] 2025-12-16 00:04:01 +00:00
15 changed files with 0 additions and 922 deletions

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@@ -1,293 +0,0 @@
/*
* Copyright (c) 2024, Jelle Raaijmakers <jelle@ladybird.org>
*
* SPDX-License-Identifier: BSD-2-Clause
*/
#include "FFmpegLoader.h"
#include <AK/NumericLimits.h>
#include <LibCore/System.h>
#if LIBAVCODEC_VERSION_INT >= AV_VERSION_INT(59, 24, 100)
# define USE_FFMPEG_CH_LAYOUT
#endif
#if LIBAVFORMAT_VERSION_INT >= AV_VERSION_INT(59, 0, 100)
# define USE_CONSTIFIED_POINTERS
#endif
namespace Audio {
static constexpr int BUFFER_MAX_PROBE_SIZE = 64 * KiB;
FFmpegLoaderPlugin::FFmpegLoaderPlugin(NonnullOwnPtr<SeekableStream> stream, NonnullOwnPtr<Media::FFmpeg::FFmpegIOContext> io_context)
: LoaderPlugin(move(stream))
, m_io_context(move(io_context))
{
}
FFmpegLoaderPlugin::~FFmpegLoaderPlugin()
{
if (m_frame != nullptr)
av_frame_free(&m_frame);
if (m_packet != nullptr)
av_packet_free(&m_packet);
if (m_codec_context != nullptr)
avcodec_free_context(&m_codec_context);
if (m_format_context != nullptr)
avformat_close_input(&m_format_context);
}
ErrorOr<NonnullOwnPtr<LoaderPlugin>> FFmpegLoaderPlugin::create(NonnullOwnPtr<SeekableStream> stream)
{
auto io_context = TRY(Media::FFmpeg::FFmpegIOContext::create(*stream));
auto loader = make<FFmpegLoaderPlugin>(move(stream), move(io_context));
TRY(loader->initialize());
return loader;
}
ErrorOr<void> FFmpegLoaderPlugin::initialize()
{
// Open the container
m_format_context = avformat_alloc_context();
if (m_format_context == nullptr)
return Error::from_string_literal("Failed to allocate format context");
m_format_context->pb = m_io_context->avio_context();
if (avformat_open_input(&m_format_context, nullptr, nullptr, nullptr) < 0)
return Error::from_string_literal("Failed to open input for format parsing");
// Read stream info; doing this is required for headerless formats like MPEG
if (avformat_find_stream_info(m_format_context, nullptr) < 0)
return Error::from_string_literal("Failed to find stream info");
#ifdef USE_CONSTIFIED_POINTERS
AVCodec const* codec {};
#else
AVCodec* codec {};
#endif
// Find the best stream to play within the container
int best_stream_index = av_find_best_stream(m_format_context, AVMediaType::AVMEDIA_TYPE_AUDIO, -1, -1, &codec, 0);
if (best_stream_index == AVERROR_STREAM_NOT_FOUND)
return Error::from_string_literal("No audio stream found in container");
if (best_stream_index == AVERROR_DECODER_NOT_FOUND)
return Error::from_string_literal("No suitable decoder found for stream");
if (best_stream_index < 0)
return Error::from_string_literal("Failed to find an audio stream");
m_audio_stream = m_format_context->streams[best_stream_index];
// Set up the context to decode the audio stream
m_codec_context = avcodec_alloc_context3(codec);
if (m_codec_context == nullptr)
return Error::from_string_literal("Failed to allocate the codec context");
if (avcodec_parameters_to_context(m_codec_context, m_audio_stream->codecpar) < 0)
return Error::from_string_literal("Failed to copy codec parameters");
m_codec_context->pkt_timebase = m_audio_stream->time_base;
m_codec_context->thread_count = AK::min(static_cast<int>(Core::System::hardware_concurrency()), 4);
if (avcodec_open2(m_codec_context, codec, nullptr) < 0)
return Error::from_string_literal("Failed to open input for decoding");
// This is an initial estimate of the total number of samples in the stream.
// During decoding, we might need to increase the number as more frames come in.
auto duration_in_seconds = TRY([this] -> ErrorOr<double> {
if (m_audio_stream->duration >= 0) {
auto time_base = av_q2d(m_audio_stream->time_base);
return static_cast<double>(m_audio_stream->duration) * time_base;
}
// If the stream doesn't specify the duration, fallback to what the container says the duration is.
// If the container doesn't know the duration, then we're out of luck. Return an error.
if (m_format_context->duration < 0)
return Error::from_string_literal("Negative stream duration");
return static_cast<double>(m_format_context->duration) / AV_TIME_BASE;
}());
m_total_samples = AK::round_to<decltype(m_total_samples)>(sample_rate() * duration_in_seconds);
// Allocate packet (logical chunk of data) and frame (video / audio frame) buffers
m_packet = av_packet_alloc();
if (m_packet == nullptr)
return Error::from_string_literal("Failed to allocate packet");
m_frame = av_frame_alloc();
if (m_frame == nullptr)
return Error::from_string_literal("Failed to allocate frame");
return {};
}
double FFmpegLoaderPlugin::time_base() const
{
return av_q2d(m_audio_stream->time_base);
}
bool FFmpegLoaderPlugin::sniff(SeekableStream& stream)
{
auto io_context = MUST(Media::FFmpeg::FFmpegIOContext::create(stream));
#ifdef USE_CONSTIFIED_POINTERS
AVInputFormat const* detected_format {};
#else
AVInputFormat* detected_format {};
#endif
auto score = av_probe_input_buffer2(io_context->avio_context(), &detected_format, nullptr, nullptr, 0, BUFFER_MAX_PROBE_SIZE);
return score > 0;
}
static ErrorOr<FixedArray<Sample>> extract_samples_from_frame(AVFrame& frame)
{
size_t number_of_samples = frame.nb_samples;
VERIFY(number_of_samples > 0);
#ifdef USE_FFMPEG_CH_LAYOUT
size_t number_of_channels = frame.ch_layout.nb_channels;
#else
size_t number_of_channels = frame.channels;
#endif
auto format = static_cast<AVSampleFormat>(frame.format);
auto packed_format = av_get_packed_sample_fmt(format);
auto is_planar = av_sample_fmt_is_planar(format) == 1;
// FIXME: handle number_of_channels > 2
if (number_of_channels != 1 && number_of_channels != 2)
return Error::from_string_view("Unsupported number of channels"sv);
switch (format) {
case AV_SAMPLE_FMT_FLTP:
case AV_SAMPLE_FMT_S16:
case AV_SAMPLE_FMT_S32:
break;
default:
// FIXME: handle other formats
return Error::from_string_view("Unsupported sample format"sv);
}
auto get_plane_pointer = [&](size_t channel_index) -> uint8_t* {
return is_planar ? frame.extended_data[channel_index] : frame.extended_data[0];
};
auto index_in_plane = [&](size_t sample_index, size_t channel_index) {
if (is_planar)
return sample_index;
return sample_index * number_of_channels + channel_index;
};
auto read_sample = [&](uint8_t* data, size_t index) -> float {
switch (packed_format) {
case AV_SAMPLE_FMT_FLT:
return reinterpret_cast<float*>(data)[index];
case AV_SAMPLE_FMT_S16:
return reinterpret_cast<i16*>(data)[index] / static_cast<float>(NumericLimits<i16>::max());
case AV_SAMPLE_FMT_S32:
return reinterpret_cast<i32*>(data)[index] / static_cast<float>(NumericLimits<i32>::max());
default:
VERIFY_NOT_REACHED();
}
};
auto samples = TRY(FixedArray<Sample>::create(number_of_samples));
for (size_t sample = 0; sample < number_of_samples; ++sample) {
if (number_of_channels == 1) {
samples.unchecked_at(sample) = Sample { read_sample(get_plane_pointer(0), index_in_plane(sample, 0)) };
} else {
samples.unchecked_at(sample) = Sample {
read_sample(get_plane_pointer(0), index_in_plane(sample, 0)),
read_sample(get_plane_pointer(1), index_in_plane(sample, 1)),
};
}
}
return samples;
}
ErrorOr<Vector<FixedArray<Sample>>> FFmpegLoaderPlugin::load_chunks(size_t samples_to_read_from_input)
{
Vector<FixedArray<Sample>> chunks {};
do {
// Obtain a packet
auto read_frame_error = av_read_frame(m_format_context, m_packet);
if (read_frame_error < 0) {
if (read_frame_error == AVERROR_EOF)
break;
return Error::from_string_literal("Failed to read frame");
}
if (m_packet->stream_index != m_audio_stream->index) {
av_packet_unref(m_packet);
continue;
}
// Send the packet to the decoder
if (avcodec_send_packet(m_codec_context, m_packet) < 0)
return Error::from_string_literal("Failed to send packet");
av_packet_unref(m_packet);
// Ask the decoder for a new frame. We might not have sent enough data yet
auto receive_frame_error = avcodec_receive_frame(m_codec_context, m_frame);
if (receive_frame_error != 0) {
if (receive_frame_error == AVERROR(EAGAIN))
continue;
if (receive_frame_error == AVERROR_EOF)
break;
return Error::from_string_literal("Failed to receive frame");
}
chunks.append(TRY(extract_samples_from_frame(*m_frame)));
// Use the frame's presentation timestamp to set the number of loaded samples
m_loaded_samples = static_cast<int>(m_frame->pts * sample_rate() * time_base());
if (m_loaded_samples > m_total_samples) [[unlikely]]
m_total_samples = m_loaded_samples;
samples_to_read_from_input -= AK::min(samples_to_read_from_input, m_frame->nb_samples);
} while (samples_to_read_from_input > 0);
return chunks;
}
ErrorOr<void> FFmpegLoaderPlugin::reset()
{
return seek(0);
}
ErrorOr<void> FFmpegLoaderPlugin::seek(int sample_index)
{
auto sample_position_in_seconds = static_cast<double>(sample_index) / sample_rate();
auto sample_timestamp = AK::round_to<int64_t>(sample_position_in_seconds / time_base());
if (av_seek_frame(m_format_context, m_audio_stream->index, sample_timestamp, AVSEEK_FLAG_ANY) < 0)
return Error::from_string_literal("Failed to seek");
avcodec_flush_buffers(m_codec_context);
m_loaded_samples = sample_index;
return {};
}
u32 FFmpegLoaderPlugin::sample_rate()
{
VERIFY(m_codec_context != nullptr);
return m_codec_context->sample_rate;
}
u16 FFmpegLoaderPlugin::num_channels()
{
VERIFY(m_codec_context != nullptr);
#ifdef USE_FFMPEG_CH_LAYOUT
return m_codec_context->ch_layout.nb_channels;
#else
return m_codec_context->channels;
#endif
}
PcmSampleFormat FFmpegLoaderPlugin::pcm_format()
{
// FIXME: pcm_format() is unused, always return Float for now
return PcmSampleFormat::Float32;
}
ByteString FFmpegLoaderPlugin::format_name()
{
if (!m_format_context)
return "unknown";
return m_format_context->iformat->name;
}
}

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@@ -1,55 +0,0 @@
/*
* Copyright (c) 2024, Jelle Raaijmakers <jelle@ladybird.org>
*
* SPDX-License-Identifier: BSD-2-Clause
*/
#pragma once
#include "Loader.h"
#include <AK/Error.h>
#include <AK/NonnullOwnPtr.h>
#include <LibMedia/FFmpeg/FFmpegIOContext.h>
extern "C" {
#include <libavcodec/avcodec.h>
#include <libavformat/avformat.h>
}
namespace Audio {
class FFmpegLoaderPlugin : public LoaderPlugin {
public:
explicit FFmpegLoaderPlugin(NonnullOwnPtr<SeekableStream>, NonnullOwnPtr<Media::FFmpeg::FFmpegIOContext>);
virtual ~FFmpegLoaderPlugin();
static bool sniff(SeekableStream& stream);
static ErrorOr<NonnullOwnPtr<LoaderPlugin>> create(NonnullOwnPtr<SeekableStream>);
virtual ErrorOr<Vector<FixedArray<Sample>>> load_chunks(size_t samples_to_read_from_input) override;
virtual ErrorOr<void> reset() override;
virtual ErrorOr<void> seek(int sample_index) override;
virtual int loaded_samples() override { return m_loaded_samples; }
virtual int total_samples() override { return m_total_samples; }
virtual u32 sample_rate() override;
virtual u16 num_channels() override;
virtual PcmSampleFormat pcm_format() override;
virtual ByteString format_name() override;
private:
ErrorOr<void> initialize();
double time_base() const;
AVStream* m_audio_stream;
AVCodecContext* m_codec_context { nullptr };
AVFormatContext* m_format_context { nullptr };
AVFrame* m_frame { nullptr };
NonnullOwnPtr<Media::FFmpeg::FFmpegIOContext> m_io_context;
int m_loaded_samples { 0 };
AVPacket* m_packet { nullptr };
int m_total_samples { 0 };
};
}

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@@ -9,8 +9,6 @@
namespace Audio {
class AudioConverter;
class Loader;
class PlaybackStream;
struct Sample;
}

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/*
* Copyright (c) 2018-2023, the SerenityOS developers.
* Copyright (c) 2024, Jelle Raaijmakers <jelle@ladybird.org>
*
* SPDX-License-Identifier: BSD-2-Clause
*/
#include "Loader.h"
#include "FFmpegLoader.h"
#include <AK/TypedTransfer.h>
#include <LibCore/MappedFile.h>
namespace Audio {
LoaderPlugin::LoaderPlugin(NonnullOwnPtr<SeekableStream> stream)
: m_stream(move(stream))
{
}
Loader::Loader(NonnullOwnPtr<LoaderPlugin> plugin)
: m_plugin(move(plugin))
{
}
struct LoaderPluginInitializer {
bool (*sniff)(SeekableStream&);
ErrorOr<NonnullOwnPtr<LoaderPlugin>> (*create)(NonnullOwnPtr<SeekableStream>);
};
static constexpr LoaderPluginInitializer s_initializers[] = {
{ FFmpegLoaderPlugin::sniff, FFmpegLoaderPlugin::create },
};
ErrorOr<NonnullRefPtr<Loader>> Loader::create(StringView path)
{
auto stream = TRY(Core::MappedFile::map(path, Core::MappedFile::Mode::ReadOnly));
auto plugin = TRY(Loader::create_plugin(move(stream)));
return adopt_ref(*new (nothrow) Loader(move(plugin)));
}
ErrorOr<NonnullRefPtr<Loader>> Loader::create(ReadonlyBytes buffer)
{
auto stream = TRY(try_make<FixedMemoryStream>(buffer));
auto plugin = TRY(Loader::create_plugin(move(stream)));
return adopt_ref(*new (nothrow) Loader(move(plugin)));
}
ErrorOr<NonnullOwnPtr<LoaderPlugin>> Loader::create_plugin(NonnullOwnPtr<SeekableStream> stream)
{
for (auto const& loader : s_initializers) {
if (loader.sniff(*stream)) {
TRY(stream->seek(0, SeekMode::SetPosition));
return loader.create(move(stream));
}
TRY(stream->seek(0, SeekMode::SetPosition));
}
return Error::from_string_literal("No loader plugin available");
}
ErrorOr<Samples> Loader::get_more_samples(size_t samples_to_read_from_input)
{
if (m_plugin_at_end_of_stream && m_buffer.is_empty())
return Samples {};
size_t remaining_samples = total_samples() - loaded_samples();
size_t samples_to_read = min(remaining_samples, samples_to_read_from_input);
auto samples = TRY(Samples::create(samples_to_read));
size_t sample_index = 0;
if (m_buffer.size() > 0) {
size_t to_transfer = min(m_buffer.size(), samples_to_read);
AK::TypedTransfer<Sample>::move(samples.data(), m_buffer.data(), to_transfer);
if (to_transfer < m_buffer.size())
m_buffer.remove(0, to_transfer);
else
m_buffer.clear_with_capacity();
sample_index += to_transfer;
}
while (sample_index < samples_to_read) {
auto chunk_data = TRY(m_plugin->load_chunks(samples_to_read - sample_index));
chunk_data.remove_all_matching([](auto& chunk) { return chunk.is_empty(); });
if (chunk_data.is_empty()) {
m_plugin_at_end_of_stream = true;
break;
}
for (auto& chunk : chunk_data) {
if (sample_index < samples_to_read) {
auto count = min(samples_to_read - sample_index, chunk.size());
AK::TypedTransfer<Sample>::move(samples.span().offset(sample_index), chunk.data(), count);
// We didn't read all of the chunk; transfer the rest into the buffer.
if (count < chunk.size()) {
auto remaining_samples_count = chunk.size() - count;
// We will always have an empty buffer at this point!
TRY(m_buffer.try_append(chunk.span().offset(count), remaining_samples_count));
}
} else {
// We're now past what the user requested. Transfer the entirety of the data into the buffer.
TRY(m_buffer.try_append(chunk.data(), chunk.size()));
}
sample_index += chunk.size();
}
}
return samples;
}
}

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/*
* Copyright (c) 2018-2022, the SerenityOS developers.
*
* SPDX-License-Identifier: BSD-2-Clause
*/
#pragma once
#include "Sample.h"
#include "SampleFormats.h"
#include <AK/ByteString.h>
#include <AK/Error.h>
#include <AK/FixedArray.h>
#include <AK/NonnullOwnPtr.h>
#include <AK/NonnullRefPtr.h>
#include <AK/RefCounted.h>
#include <AK/Stream.h>
#include <AK/StringView.h>
#include <AK/Vector.h>
#include <LibMedia/Export.h>
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 const loader_buffer_size = 8 * KiB;
// Two seek points should ideally not be farther apart than this.
// This variable is a heuristic for seek table-constructing loaders.
constexpr u64 const maximum_seekpoint_distance_ms = 1000;
// Seeking should be at least as precise as this.
// That means: The actual achieved seek position must not be more than this amount of time before the requested seek position.
constexpr u64 const seek_tolerance_ms = 5000;
using Samples = FixedArray<Sample>;
class LoaderPlugin {
public:
explicit LoaderPlugin(NonnullOwnPtr<SeekableStream> stream);
virtual ~LoaderPlugin() = default;
// Load as many audio chunks as necessary to get up to the required samples.
// A chunk can be anything that is convenient for the plugin to load in one go without requiring to move samples around different buffers.
// For example: A FLAC, MP3 or QOA frame.
// The chunks are returned in a vector, so the loader can simply add chunks until the requested sample amount is reached.
// The sample count MAY be surpassed, but only as little as possible. It CAN be undershot when the end of the stream is reached.
// If the loader has no chunking limitations (e.g. WAV), it may return a single exact-sized chunk.
virtual ErrorOr<Vector<FixedArray<Sample>>> load_chunks(size_t samples_to_read_from_input) = 0;
virtual ErrorOr<void> reset() = 0;
virtual ErrorOr<void> seek(int const sample_index) = 0;
// total_samples() and loaded_samples() should be independent
// of the number of channels.
//
// For example, with a three-second-long, stereo, 44.1KHz audio file:
// num_channels() should return 2
// sample_rate() should return 44100 (each channel is sampled at this rate)
// total_samples() should return 132300 (sample_rate * three seconds)
virtual int loaded_samples() = 0;
virtual int total_samples() = 0;
virtual u32 sample_rate() = 0;
virtual u16 num_channels() = 0;
// Human-readable name of the file format, of the form <full abbreviation> (.<ending>)
virtual ByteString format_name() = 0;
virtual PcmSampleFormat pcm_format() = 0;
protected:
NonnullOwnPtr<SeekableStream> m_stream;
};
class MEDIA_API Loader : public RefCounted<Loader> {
public:
static ErrorOr<NonnullRefPtr<Loader>> create(StringView path);
static ErrorOr<NonnullRefPtr<Loader>> create(ReadonlyBytes buffer);
// Will only read less samples if we're at the end of the stream.
ErrorOr<Samples> get_more_samples(size_t samples_to_read_from_input = 128 * KiB);
ErrorOr<void> reset() const
{
m_plugin_at_end_of_stream = false;
return m_plugin->reset();
}
ErrorOr<void> seek(int const position) const
{
m_buffer.clear_with_capacity();
m_plugin_at_end_of_stream = false;
return m_plugin->seek(position);
}
int loaded_samples() const { return m_plugin->loaded_samples() - (int)m_buffer.size(); }
int total_samples() const { return m_plugin->total_samples(); }
u32 sample_rate() const { return m_plugin->sample_rate(); }
u16 num_channels() const { return m_plugin->num_channels(); }
ByteString format_name() const { return m_plugin->format_name(); }
u16 bits_per_sample() const { return pcm_bits_per_sample(m_plugin->pcm_format()); }
PcmSampleFormat pcm_format() const { return m_plugin->pcm_format(); }
private:
static ErrorOr<NonnullOwnPtr<LoaderPlugin>> create_plugin(NonnullOwnPtr<SeekableStream> stream);
explicit Loader(NonnullOwnPtr<LoaderPlugin>);
mutable NonnullOwnPtr<LoaderPlugin> m_plugin;
// The plugin can signal an end of stream by returning no (or only empty) chunks.
mutable bool m_plugin_at_end_of_stream { false };
mutable Vector<Sample, loader_buffer_size> m_buffer;
};
}

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@@ -6,7 +6,6 @@
#pragma once
#include "SampleFormats.h"
#include <AK/AtomicRefCounted.h>
#include <AK/Function.h>
#include <AK/Time.h>

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/*
* Copyright (c) 2018-2020, Andreas Kling <andreas@ladybird.org>
* Copyright (c) 2021, kleines Filmröllchen <filmroellchen@serenityos.org>
*
* SPDX-License-Identifier: BSD-2-Clause
*/
#pragma once
#include <AK/Format.h>
#include <AK/Math.h>
namespace Audio {
using AK::Exponentials::exp;
using AK::Exponentials::log;
// Constants for logarithmic volume. See Sample::linear_to_log
// Corresponds to 60dB
constexpr float DYNAMIC_RANGE = 1000;
constexpr float VOLUME_A = 1 / DYNAMIC_RANGE;
float const VOLUME_B = log(DYNAMIC_RANGE);
// A single sample in an audio buffer.
// Values are floating point, and should range from -1.0 to +1.0
struct Sample {
constexpr Sample() = default;
// For mono
constexpr explicit Sample(float left)
: left(left)
, right(left)
{
}
// For stereo
constexpr Sample(float left, float right)
: left(left)
, right(right)
{
}
// Returns the absolute maximum range (separate per channel) of the given sample buffer.
// For example { 0.8, 0 } means that samples on the left channel occupy the range { -0.8, 0.8 },
// while all samples on the right channel are 0.
static Sample max_range(ReadonlySpan<Sample> span)
{
Sample result { NumericLimits<float>::min_normal(), NumericLimits<float>::min_normal() };
for (Sample sample : span) {
result.left = max(result.left, AK::fabs(sample.left));
result.right = max(result.right, AK::fabs(sample.right));
}
return result;
}
void clip()
{
if (left > 1)
left = 1;
else if (left < -1)
left = -1;
if (right > 1)
right = 1;
else if (right < -1)
right = -1;
}
// Logarithmic scaling, as audio should ALWAYS do.
// Reference: https://www.dr-lex.be/info-stuff/volumecontrols.html
// We use the curve `factor = a * exp(b * change)`,
// where change is the input fraction we want to change by,
// a = 1/1000, b = ln(1000) = 6.908 and factor is the multiplier used.
// The value 1000 represents the dynamic range in sound pressure, which corresponds to 60 dB(A).
// This is a good dynamic range because it can represent all loudness values from
// 30 dB(A) (barely hearable with background noise)
// to 90 dB(A) (almost too loud to hear and about the reasonable limit of actual sound equipment).
//
// Format ranges:
// - Linear: 0.0 to 1.0
// - Logarithmic: 0.0 to 1.0
ALWAYS_INLINE float linear_to_log(float const change) const
{
// TODO: Add linear slope around 0
return VOLUME_A * exp(VOLUME_B * change);
}
ALWAYS_INLINE float log_to_linear(float const val) const
{
// TODO: Add linear slope around 0
return log(val / VOLUME_A) / VOLUME_B;
}
ALWAYS_INLINE Sample& log_multiply(float const change)
{
float factor = linear_to_log(change);
left *= factor;
right *= factor;
return *this;
}
ALWAYS_INLINE Sample log_multiplied(float const volume_change) const
{
Sample new_frame { left, right };
new_frame.log_multiply(volume_change);
return new_frame;
}
// Constant power panning
ALWAYS_INLINE Sample& pan(float const position)
{
float const pi_over_2 = AK::Pi<float> * 0.5f;
float const root_over_2 = AK::sqrt<float>(2.0) * 0.5f;
float const angle = position * pi_over_2 * 0.5f;
float s, c;
AK::sincos<float>(angle, s, c);
left *= root_over_2 * (c - s);
right *= root_over_2 * (c + s);
return *this;
}
ALWAYS_INLINE Sample panned(float const position) const
{
Sample new_sample { left, right };
new_sample.pan(position);
return new_sample;
}
constexpr Sample& operator*=(float const mult)
{
left *= mult;
right *= mult;
return *this;
}
constexpr Sample operator*(float const mult) const
{
return { left * mult, right * mult };
}
constexpr Sample& operator+=(Sample const& other)
{
left += other.left;
right += other.right;
return *this;
}
constexpr Sample& operator+=(float other)
{
left += other;
right += other;
return *this;
}
constexpr Sample operator+(Sample const& other) const
{
return { left + other.left, right + other.right };
}
float left { 0 };
float right { 0 };
};
}
namespace AK {
template<>
struct Formatter<Audio::Sample> : Formatter<FormatString> {
ErrorOr<void> format(FormatBuilder& builder, Audio::Sample const& value)
{
return Formatter<FormatString>::format(builder, "[{}, {}]"sv, value.left, value.right);
}
};
}

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@@ -1,31 +0,0 @@
/*
* Copyright (c) 2022, kleines Filmröllchen <filmroellchen@serenityos.org>.
*
* SPDX-License-Identifier: BSD-2-Clause
*/
#include "SampleFormats.h"
#include <AK/Assertions.h>
namespace Audio {
u16 pcm_bits_per_sample(PcmSampleFormat format)
{
switch (format) {
case PcmSampleFormat::Uint8:
return 8;
case PcmSampleFormat::Int16:
return 16;
case PcmSampleFormat::Int24:
return 24;
case PcmSampleFormat::Int32:
case PcmSampleFormat::Float32:
return 32;
case PcmSampleFormat::Float64:
return 64;
default:
VERIFY_NOT_REACHED();
}
}
}

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@@ -1,27 +0,0 @@
/*
* Copyright (c) 2022, kleines Filmröllchen <filmroellchen@serenityos.org>.
*
* SPDX-License-Identifier: BSD-2-Clause
*/
#pragma once
#include <AK/Types.h>
#include <LibMedia/Export.h>
namespace Audio {
// Supported PCM sample formats.
enum class PcmSampleFormat : u8 {
Uint8,
Int16,
Int24,
Int32,
Float32,
Float64,
};
// Most of the read code only cares about how many bits to read or write
MEDIA_API u16 pcm_bits_per_sample(PcmSampleFormat format);
}

View File

@@ -3,8 +3,6 @@ include(audio)
include(ffmpeg)
set(SOURCES
Audio/Loader.cpp
Audio/SampleFormats.cpp
Color/ColorConverter.cpp
Color/ColorPrimaries.cpp
Color/TransferCharacteristics.cpp
@@ -24,7 +22,6 @@ ladybird_lib(LibMedia media EXPLICIT_SYMBOL_EXPORT)
target_link_libraries(LibMedia PRIVATE LibCore LibCrypto LibIPC LibGfx LibThreading LibUnicode)
target_sources(LibMedia PRIVATE
Audio/FFmpegLoader.cpp
FFmpeg/FFmpegAudioConverter.cpp
FFmpeg/FFmpegAudioDecoder.cpp
FFmpeg/FFmpegDemuxer.cpp

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@@ -11,7 +11,6 @@
#include <AK/RefPtr.h>
#include <LibCore/EventLoop.h>
#include <LibMedia/Audio/Forward.h>
#include <LibMedia/Audio/SampleFormats.h>
#include <LibMedia/Export.h>
#include <LibMedia/Forward.h>
#include <LibMedia/Providers/MediaTimeProvider.h>

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@@ -1,39 +0,0 @@
/*
* Copyright (c) 2022, Luke Wilde <lukew@serenityos.org>
* Copyright (c) 2023, kleines Filmröllchen <filmroellchen@serenityos.org>
* Copyright (c) 2021-2023, the SerenityOS developers.
*
* SPDX-License-Identifier: BSD-2-Clause
*/
#pragma once
#include <AK/Concepts.h>
#include <AK/MemoryStream.h>
#include <LibMedia/Audio/Loader.h>
#include <stddef.h>
#include <stdint.h>
template<typename LoaderPluginType>
requires(IsBaseOf<Audio::LoaderPlugin, LoaderPluginType>)
int fuzz_audio_loader(uint8_t const* data, size_t size)
{
auto const bytes = ReadonlyBytes { data, size };
auto stream = try_make<FixedMemoryStream>(bytes).release_value();
auto audio_or_error = LoaderPluginType::create(move(stream));
if (audio_or_error.is_error())
return 0;
auto audio = audio_or_error.release_value();
for (;;) {
auto samples = audio->load_chunks(4 * KiB);
if (samples.is_error())
return 0;
if (samples.value().size() == 0)
break;
}
return 0;
}

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@@ -4,7 +4,6 @@ import("//Meta/gn/build/libs/pulse/enable.gni")
shared_library("LibMedia") {
include_dirs = [ "//Userland/Libraries" ]
sources = [
"Audio/Loader.cpp",
"Audio/PlaybackStream.cpp",
"Audio/SampleFormats.cpp",
"Color/ColorConverter.cpp",
@@ -29,7 +28,6 @@ shared_library("LibMedia") {
}
if (enable_ffmpeg) {
sources += [
"Audio/FFmpegLoader.cpp",
"FFmpeg/FFmpegAudioConverter.cpp",
"FFmpeg/FFmpegAudioDecoder.cpp",
"FFmpeg/FFmpegHelpers.cpp",

View File

@@ -8,7 +8,6 @@ else()
endif()
lagom_utility(xml SOURCES xml.cpp LIBS LibFileSystem LibMain LibXML LibURL)
lagom_utility(abench SOURCES abench.cpp LIBS LibMain LibFileSystem LibMedia)
lagom_utility(dns SOURCES dns.cpp LIBS LibDNS LibMain LibTLS LibCrypto)
if (ENABLE_GUI_TARGETS)

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@@ -1,66 +0,0 @@
/*
* Copyright (c) 2021, the SerenityOS developers.
*
* SPDX-License-Identifier: BSD-2-Clause
*/
#include <AK/NumericLimits.h>
#include <AK/Types.h>
#include <LibCore/ArgsParser.h>
#include <LibCore/ElapsedTimer.h>
#include <LibFileSystem/FileSystem.h>
#include <LibMain/Main.h>
#include <LibMedia/Audio/Loader.h>
#include <stdio.h>
// The Kernel has problems with large anonymous buffers, so let's limit sample reads ourselves.
static constexpr size_t MAX_CHUNK_SIZE = 1 * MiB / 2;
ErrorOr<int> ladybird_main(Main::Arguments arguments)
{
StringView path {};
int sample_count = -1;
Core::ArgsParser args_parser;
args_parser.set_general_help("Benchmark audio loading");
args_parser.add_positional_argument(path, "Path to audio file", "path");
args_parser.add_option(sample_count, "How many samples to load at maximum", "sample-count", 's', "samples");
args_parser.parse(arguments);
auto maybe_loader = Audio::Loader::create(path);
if (maybe_loader.is_error()) {
warnln("Failed to load audio file: {}", maybe_loader.error());
return 1;
}
auto loader = maybe_loader.release_value();
Core::ElapsedTimer sample_timer { Core::TimerType::Precise };
i64 total_loader_time = 0;
int remaining_samples = sample_count > 0 ? sample_count : NumericLimits<int>::max();
unsigned total_loaded_samples = 0;
for (;;) {
if (remaining_samples > 0) {
sample_timer = sample_timer.start_new();
auto samples = loader->get_more_samples(min(MAX_CHUNK_SIZE, remaining_samples));
total_loader_time += sample_timer.elapsed_milliseconds();
if (!samples.is_error()) {
remaining_samples -= samples.value().size();
total_loaded_samples += samples.value().size();
if (samples.value().size() == 0)
break;
} else {
warnln("Error while loading audio: {}", samples.error());
return 1;
}
} else
break;
}
auto time_per_sample = static_cast<double>(total_loader_time) / static_cast<double>(total_loaded_samples) * 1000.;
auto playback_time_per_sample = (1. / static_cast<double>(loader->sample_rate())) * 1000'000.;
outln("Loaded {:10d} samples in {:06.3f} s, {:9.3f} µs/sample, {:6.1f}% speed (realtime {:9.3f} µs/sample)", total_loaded_samples, static_cast<double>(total_loader_time) / 1000., time_per_sample, playback_time_per_sample / time_per_sample * 100., playback_time_per_sample);
return 0;
}