LibAudio: Create a playback class with a PulseAudio implementation

This adds an abstract `Audio::PlaybackStream` class to allow cross-
platform audio playback to be done in an opaque manner by applications
in both Serenity and Lagom.

Currently, the only supported audio API is PulseAudio, but a Serenity
implementation should be added shortly as well.
This commit is contained in:
Zaggy1024 2023-07-04 04:55:53 -05:00 committed by Andrew Kaster
parent fe672989a9
commit bc4d4f0f95
Notes: sideshowbarker 2024-07-17 01:06:10 +09:00
9 changed files with 1041 additions and 0 deletions

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@ -143,6 +143,8 @@ if (ENABLE_LAGOM_LADYBIRD AND (ENABLE_FUZZERS OR ENABLE_COMPILER_EXPLORER_BUILD)
)
endif()
CHECK_INCLUDE_FILE(pulse/pulseaudio.h HAVE_PULSEAUDIO)
if (CMAKE_CXX_COMPILER_ID MATCHES "Clang$")
add_compile_options(-Wno-overloaded-virtual)
# FIXME: Re-enable this check when the warning stops triggering, or document why we can't stop it from triggering.

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@ -8,6 +8,7 @@ set(SOURCES
WavWriter.cpp
Metadata.cpp
MP3Loader.cpp
PlaybackStream.cpp
QOALoader.cpp
QOATypes.cpp
UserSampleQueue.cpp
@ -25,5 +26,17 @@ if (SERENITYOS)
)
endif()
if (HAVE_PULSEAUDIO)
list(APPEND SOURCES
PlaybackStreamPulseAudio.cpp
PulseAudioWrappers.cpp
)
endif()
serenity_lib(LibAudio audio)
target_link_libraries(LibAudio PRIVATE LibCore LibIPC LibThreading LibUnicode LibCrypto)
if (HAVE_PULSEAUDIO)
target_link_libraries(LibAudio PRIVATE pulse)
target_compile_definitions(LibAudio PRIVATE HAVE_PULSEAUDIO=1)
endif()

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@ -10,6 +10,7 @@ namespace Audio {
class ConnectionToServer;
class Loader;
class PlaybackStream;
struct Sample;
template<typename SampleType>

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@ -0,0 +1,39 @@
/*
* Copyright (c) 2023, Gregory Bertilson <zaggy1024@gmail.com>
*
* SPDX-License-Identifier: BSD-2-Clause
*/
#include "PlaybackStream.h"
#include <LibCore/ThreadedPromise.h>
#if defined(HAVE_PULSEAUDIO)
# include <LibAudio/PlaybackStreamPulseAudio.h>
#endif
namespace Audio {
#define TRY_OR_REJECT_AND_STOP(expression, promise) \
({ \
auto&& __temporary_result = (expression); \
if (__temporary_result.is_error()) [[unlikely]] { \
(promise)->reject(__temporary_result.release_error()); \
return 1; \
} \
__temporary_result.release_value(); \
})
ErrorOr<NonnullRefPtr<PlaybackStream>> PlaybackStream::create(OutputState initial_output_state, u32 sample_rate, u8 channels, u32 target_latency_ms, AudioDataRequestCallback&& data_request_callback)
{
VERIFY(data_request_callback);
// Create the platform-specific implementation for this stream.
#if defined(HAVE_PULSEAUDIO)
return PlaybackStreamPulseAudio::create(initial_output_state, sample_rate, channels, target_latency_ms, move(data_request_callback));
#else
(void)initial_output_state, (void)sample_rate, (void)channels, (void)target_latency_ms;
return Error::from_string_literal("Audio output is not available for this platform");
#endif
}
}

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@ -0,0 +1,72 @@
/*
* Copyright (c) 2023, Gregory Bertilson <zaggy1024@gmail.com>
*
* SPDX-License-Identifier: BSD-2-Clause
*/
#pragma once
#include <AK/AtomicRefCounted.h>
#include <AK/Function.h>
#include <AK/Queue.h>
#include <AK/Time.h>
#include <LibAudio/SampleFormats.h>
#include <LibCore/Forward.h>
#include <LibThreading/ConditionVariable.h>
#include <LibThreading/MutexProtected.h>
#include <LibThreading/Thread.h>
namespace Audio {
enum class OutputState {
Playing,
Suspended,
};
// This class implements high-level audio playback behavior. It is primarily intended as an abstract cross-platform
// interface to be used by Ladybird (and its dependent libraries) for playback.
//
// The interface is designed to be simple and robust. All control functions can be called safely from any thread.
// Timing information provided by the class should allow audio timestamps to be tracked with the best accuracy possible.
class PlaybackStream : public AtomicRefCounted<PlaybackStream> {
public:
using AudioDataRequestCallback = Function<ReadonlyBytes(Bytes buffer, PcmSampleFormat format, size_t sample_count)>;
// Creates a new audio Output class.
//
// The initial_output_state parameter determines whether it will begin playback immediately.
//
// The AudioDataRequestCallback will be called when the Output needs more audio data to fill
// its buffers and continue playback.
static ErrorOr<NonnullRefPtr<PlaybackStream>> create(OutputState initial_output_state, u32 sample_rate, u8 channels, u32 target_latency_ms, AudioDataRequestCallback&&);
virtual ~PlaybackStream() = default;
// Sets the callback function that will be fired whenever the server consumes more data than is made available
// by the data request callback. It will fire when either the data request runs too long, or the data request
// returns no data. If all the input data has been exhausted and this event fires, that means that playback
// has ended.
virtual void set_underrun_callback(Function<void()>) = 0;
// Resume playback from the suspended state, requesting new data for audio buffers as soon as possible.
//
// The value provided to the promise resolution will match the `total_time_played()` at the exact moment that
// the stream was resumed.
virtual NonnullRefPtr<Core::ThreadedPromise<Duration>> resume() = 0;
// Completes playback of any buffered audio data and then suspends playback and buffering.
virtual NonnullRefPtr<Core::ThreadedPromise<void>> drain_buffer_and_suspend() = 0;
// Drops any buffered audio data and then suspends playback and buffering. This can used be to stop playback
// as soon as possible instead of waiting for remaining audio to play.
virtual NonnullRefPtr<Core::ThreadedPromise<void>> discard_buffer_and_suspend() = 0;
// Returns a accurate monotonically-increasing time duration that is based on the number of samples that have
// been played by the output device. The value is interpolated and takes into account latency to the speakers
// whenever possible.
//
// This function should be able to run from any thread safely.
virtual ErrorOr<Duration> total_time_played() = 0;
virtual NonnullRefPtr<Core::ThreadedPromise<void>> set_volume(double volume) = 0;
};
}

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@ -0,0 +1,192 @@
/*
* Copyright (c) 2023, Gregory Bertilson <zaggy1024@gmail.com>
*
* SPDX-License-Identifier: BSD-2-Clause
*/
#include "PlaybackStreamPulseAudio.h"
#include <LibCore/ThreadedPromise.h>
namespace Audio {
#define TRY_OR_EXIT_THREAD(expression) \
({ \
auto&& __temporary_result = (expression); \
if (__temporary_result.is_error()) [[unlikely]] { \
warnln("Failure in PulseAudio control thread: {}", __temporary_result.error().string_literal()); \
internal_state->exit(); \
return 1; \
} \
__temporary_result.release_value(); \
})
ErrorOr<NonnullRefPtr<PlaybackStream>> PlaybackStreamPulseAudio::create(OutputState initial_state, u32 sample_rate, u8 channels, u32 target_latency_ms, AudioDataRequestCallback&& data_request_callback)
{
VERIFY(data_request_callback);
// Create an internal state for the control thread to hold on to.
auto internal_state = TRY(adopt_nonnull_ref_or_enomem(new (nothrow) InternalState()));
auto playback_stream = TRY(adopt_nonnull_ref_or_enomem(new (nothrow) PlaybackStreamPulseAudio(internal_state)));
// Create the control thread and start it.
auto thread = TRY(Threading::Thread::try_create([=, data_request_callback = move(data_request_callback)]() mutable {
auto context = TRY_OR_EXIT_THREAD(PulseAudioContext::instance());
internal_state->set_stream(TRY_OR_EXIT_THREAD(context->create_stream(initial_state, sample_rate, channels, target_latency_ms, [data_request_callback = move(data_request_callback)](PulseAudioStream&, Bytes buffer, size_t sample_count) {
return data_request_callback(buffer, PcmSampleFormat::Float32, sample_count);
})));
// PulseAudio retains the last volume it sets for an application. We want to consistently
// start at 100% volume instead.
TRY_OR_EXIT_THREAD(internal_state->stream()->set_volume(1.0));
internal_state->thread_loop();
return 0;
},
"Audio::PlaybackStream"sv));
internal_state->set_thread(thread);
thread->start();
thread->detach();
return playback_stream;
}
PlaybackStreamPulseAudio::PlaybackStreamPulseAudio(NonnullRefPtr<InternalState> state)
: m_state(move(state))
{
}
PlaybackStreamPulseAudio::~PlaybackStreamPulseAudio()
{
m_state->exit();
}
#define TRY_OR_REJECT(expression, ...) \
({ \
auto&& __temporary_result = (expression); \
if (__temporary_result.is_error()) [[unlikely]] { \
promise->reject(__temporary_result.release_error()); \
return __VA_ARGS__; \
} \
__temporary_result.release_value(); \
})
void PlaybackStreamPulseAudio::set_underrun_callback(Function<void()> callback)
{
m_state->enqueue([this, callback = move(callback)]() mutable {
m_state->stream()->set_underrun_callback(move(callback));
});
}
NonnullRefPtr<Core::ThreadedPromise<Duration>> PlaybackStreamPulseAudio::resume()
{
auto promise = Core::ThreadedPromise<Duration>::create();
TRY_OR_REJECT(m_state->check_is_running(), promise);
m_state->enqueue([this, promise]() {
TRY_OR_REJECT(m_state->stream()->resume());
promise->resolve(TRY_OR_REJECT(m_state->stream()->total_time_played()));
});
return promise;
}
NonnullRefPtr<Core::ThreadedPromise<void>> PlaybackStreamPulseAudio::drain_buffer_and_suspend()
{
auto promise = Core::ThreadedPromise<void>::create();
TRY_OR_REJECT(m_state->check_is_running(), promise);
m_state->enqueue([this, promise]() {
TRY_OR_REJECT(m_state->stream()->drain_and_suspend());
promise->resolve();
});
return promise;
}
NonnullRefPtr<Core::ThreadedPromise<void>> PlaybackStreamPulseAudio::discard_buffer_and_suspend()
{
auto promise = Core::ThreadedPromise<void>::create();
TRY_OR_REJECT(m_state->check_is_running(), promise);
m_state->enqueue([this, promise]() {
TRY_OR_REJECT(m_state->stream()->flush_and_suspend());
promise->resolve();
});
return promise;
}
ErrorOr<Duration> PlaybackStreamPulseAudio::total_time_played()
{
if (m_state->stream() != nullptr)
return m_state->stream()->total_time_played();
return Duration::zero();
}
NonnullRefPtr<Core::ThreadedPromise<void>> PlaybackStreamPulseAudio::set_volume(double volume)
{
auto promise = Core::ThreadedPromise<void>::create();
TRY_OR_REJECT(m_state->check_is_running(), promise);
m_state->enqueue([this, promise, volume]() {
TRY_OR_REJECT(m_state->stream()->set_volume(volume));
promise->resolve();
});
return promise;
}
ErrorOr<void> PlaybackStreamPulseAudio::InternalState::check_is_running()
{
if (m_exit)
return Error::from_string_literal("PulseAudio control thread loop is not running");
return {};
}
void PlaybackStreamPulseAudio::InternalState::set_thread(NonnullRefPtr<Threading::Thread> const& thread)
{
Threading::MutexLocker locker { m_mutex };
m_thread = thread;
}
void PlaybackStreamPulseAudio::InternalState::set_stream(NonnullRefPtr<PulseAudioStream> const& stream)
{
m_stream = stream;
}
RefPtr<PulseAudioStream> PlaybackStreamPulseAudio::InternalState::stream()
{
return m_stream;
}
void PlaybackStreamPulseAudio::InternalState::enqueue(Function<void()>&& task)
{
Threading::MutexLocker locker { m_mutex };
m_tasks.enqueue(forward<Function<void()>>(task));
m_wake_condition.signal();
}
void PlaybackStreamPulseAudio::InternalState::thread_loop()
{
while (true) {
auto task = [this]() -> Function<void()> {
Threading::MutexLocker locker { m_mutex };
while (m_tasks.is_empty() && !m_exit)
m_wake_condition.wait();
if (m_exit)
return nullptr;
return m_tasks.dequeue();
}();
if (!task) {
VERIFY(m_exit);
break;
}
task();
}
// Stop holding onto our thread so it can be deleted.
Threading::MutexLocker locker { m_mutex };
m_thread = nullptr;
}
void PlaybackStreamPulseAudio::InternalState::exit()
{
m_exit = true;
m_wake_condition.signal();
}
}

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@ -0,0 +1,62 @@
/*
* Copyright (c) 2023, Gregory Bertilson <zaggy1024@gmail.com>
*
* SPDX-License-Identifier: BSD-2-Clause
*/
#pragma once
#include <LibAudio/PlaybackStream.h>
#include <LibAudio/PulseAudioWrappers.h>
namespace Audio {
class PlaybackStreamPulseAudio final
: public PlaybackStream {
public:
static ErrorOr<NonnullRefPtr<PlaybackStream>> create(OutputState initial_state, u32 sample_rate, u8 channels, u32 target_latency_ms, AudioDataRequestCallback&& data_request_callback);
virtual void set_underrun_callback(Function<void()>) override;
virtual NonnullRefPtr<Core::ThreadedPromise<Duration>> resume() override;
virtual NonnullRefPtr<Core::ThreadedPromise<void>> drain_buffer_and_suspend() override;
virtual NonnullRefPtr<Core::ThreadedPromise<void>> discard_buffer_and_suspend() override;
virtual ErrorOr<Duration> total_time_played() override;
virtual NonnullRefPtr<Core::ThreadedPromise<void>> set_volume(double) override;
private:
// This struct is kept alive until the control thread exits to prevent a use-after-free without blocking on
// the UI thread.
class InternalState : public AtomicRefCounted<InternalState> {
public:
void set_thread(NonnullRefPtr<Threading::Thread> const&);
void set_stream(NonnullRefPtr<PulseAudioStream> const&);
RefPtr<PulseAudioStream> stream();
void enqueue(Function<void()>&&);
void thread_loop();
ErrorOr<void> check_is_running();
void exit();
private:
RefPtr<PulseAudioStream> m_stream { nullptr };
Queue<Function<void()>> m_tasks;
Threading::Mutex m_mutex;
Threading::ConditionVariable m_wake_condition { m_mutex };
Atomic<bool> m_exit { false };
RefPtr<Threading::Thread> m_thread { nullptr };
};
PlaybackStreamPulseAudio(NonnullRefPtr<InternalState>);
~PlaybackStreamPulseAudio();
RefPtr<InternalState> m_state;
};
}

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@ -0,0 +1,476 @@
/*
* Copyright (c) 2023, Gregory Bertilson <zaggy1024@gmail.com>
*
* SPDX-License-Identifier: BSD-2-Clause
*/
#include "PulseAudioWrappers.h"
#include <AK/WeakPtr.h>
#include <LibThreading/Mutex.h>
namespace Audio {
ErrorOr<NonnullRefPtr<PulseAudioContext>> PulseAudioContext::instance()
{
// Use a weak pointer to allow the context to be shut down if we stop outputting audio.
static WeakPtr<PulseAudioContext> the_instance;
static Threading::Mutex instantiation_mutex;
auto instantiation_locker = Threading::MutexLocker(instantiation_mutex);
RefPtr<PulseAudioContext> strong_instance_pointer = the_instance.strong_ref();
if (strong_instance_pointer == nullptr) {
auto* main_loop = pa_threaded_mainloop_new();
if (main_loop == nullptr)
return Error::from_string_literal("Failed to create PulseAudio main loop");
auto* api = pa_threaded_mainloop_get_api(main_loop);
if (api == nullptr)
return Error::from_string_literal("Failed to get PulseAudio API");
auto* context = pa_context_new(api, "Ladybird");
if (context == nullptr)
return Error::from_string_literal("Failed to get PulseAudio connection context");
strong_instance_pointer = make_ref_counted<PulseAudioContext>(main_loop, api, context);
// Set a callback to signal ourselves to wake when the state changes, so that we can
// synchronously wait for the connection.
pa_context_set_state_callback(
context, [](pa_context*, void* user_data) {
static_cast<PulseAudioContext*>(user_data)->signal_to_wake();
},
strong_instance_pointer.ptr());
if (auto error = pa_context_connect(context, nullptr, PA_CONTEXT_NOFLAGS, nullptr); error < 0) {
warnln("Starting PulseAudio context connection failed with error: {}", pulse_audio_error_to_string(static_cast<PulseAudioErrorCode>(-error)));
return Error::from_string_literal("Error while starting PulseAudio daemon connection");
}
if (auto error = pa_threaded_mainloop_start(main_loop); error < 0) {
warnln("Starting PulseAudio main loop failed with error: {}", pulse_audio_error_to_string(static_cast<PulseAudioErrorCode>(-error)));
return Error::from_string_literal("Failed to start PulseAudio main loop");
}
{
auto locker = strong_instance_pointer->main_loop_locker();
while (true) {
bool is_ready = false;
switch (strong_instance_pointer->get_connection_state()) {
case PulseAudioContextState::Connecting:
case PulseAudioContextState::Authorizing:
case PulseAudioContextState::SettingName:
break;
case PulseAudioContextState::Ready:
is_ready = true;
break;
case PulseAudioContextState::Failed:
warnln("PulseAudio server connection failed with error: {}", pulse_audio_error_to_string(strong_instance_pointer->get_last_error()));
return Error::from_string_literal("Failed to connect to PulseAudio server");
case PulseAudioContextState::Unconnected:
case PulseAudioContextState::Terminated:
VERIFY_NOT_REACHED();
break;
}
if (is_ready)
break;
strong_instance_pointer->wait_for_signal();
}
pa_context_set_state_callback(context, nullptr, nullptr);
}
the_instance = strong_instance_pointer;
}
return strong_instance_pointer.release_nonnull();
}
PulseAudioContext::PulseAudioContext(pa_threaded_mainloop* main_loop, pa_mainloop_api* api, pa_context* context)
: m_main_loop(main_loop)
, m_api(api)
, m_context(context)
{
}
PulseAudioContext::~PulseAudioContext()
{
pa_context_disconnect(m_context);
pa_context_unref(m_context);
pa_threaded_mainloop_stop(m_main_loop);
pa_threaded_mainloop_free(m_main_loop);
}
bool PulseAudioContext::current_thread_is_main_loop_thread()
{
return static_cast<bool>(pa_threaded_mainloop_in_thread(m_main_loop));
}
void PulseAudioContext::lock_main_loop()
{
if (!current_thread_is_main_loop_thread())
pa_threaded_mainloop_lock(m_main_loop);
}
void PulseAudioContext::unlock_main_loop()
{
if (!current_thread_is_main_loop_thread())
pa_threaded_mainloop_unlock(m_main_loop);
}
void PulseAudioContext::wait_for_signal()
{
pa_threaded_mainloop_wait(m_main_loop);
}
void PulseAudioContext::signal_to_wake()
{
pa_threaded_mainloop_signal(m_main_loop, 0);
}
PulseAudioContextState PulseAudioContext::get_connection_state()
{
return static_cast<PulseAudioContextState>(pa_context_get_state(m_context));
}
bool PulseAudioContext::connection_is_good()
{
return PA_CONTEXT_IS_GOOD(pa_context_get_state(m_context));
}
PulseAudioErrorCode PulseAudioContext::get_last_error()
{
return static_cast<PulseAudioErrorCode>(pa_context_errno(m_context));
}
#define STREAM_SIGNAL_CALLBACK(stream) \
[](auto*, int, void* user_data) { \
static_cast<PulseAudioStream*>(user_data)->m_context->signal_to_wake(); \
}, \
(stream)
ErrorOr<NonnullRefPtr<PulseAudioStream>> PulseAudioContext::create_stream(OutputState initial_state, u32 sample_rate, u8 channels, u32 target_latency_ms, PulseAudioDataRequestCallback write_callback)
{
auto locker = main_loop_locker();
VERIFY(get_connection_state() == PulseAudioContextState::Ready);
pa_sample_spec sample_specification {
// FIXME: Support more audio sample types.
__BYTE_ORDER__ == __ORDER_LITTLE_ENDIAN__ ? PA_SAMPLE_FLOAT32LE : PA_SAMPLE_FLOAT32BE,
sample_rate,
channels,
};
// Check the sample specification and channel map here. These are also checked by stream_new(),
// but we can return a more accurate error if we check beforehand.
if (pa_sample_spec_valid(&sample_specification) == 0)
return Error::from_string_literal("PulseAudio sample specification is invalid");
pa_channel_map channel_map;
if (pa_channel_map_init_auto(&channel_map, sample_specification.channels, PA_CHANNEL_MAP_DEFAULT) == 0) {
warnln("Getting default PulseAudio channel map failed with error: {}", pulse_audio_error_to_string(get_last_error()));
return Error::from_string_literal("Failed to get default PulseAudio channel map");
}
// Create the stream object and set a callback to signal ourselves to wake when the stream changes states,
// allowing us to wait synchronously for it to become Ready or Failed.
auto* stream = pa_stream_new_with_proplist(m_context, "Audio Stream", &sample_specification, &channel_map, nullptr);
if (stream == nullptr) {
warnln("Instantiating PulseAudio stream failed with error: {}", pulse_audio_error_to_string(get_last_error()));
return Error::from_string_literal("Failed to create PulseAudio stream");
}
pa_stream_set_state_callback(
stream, [](pa_stream*, void* user_data) {
static_cast<PulseAudioContext*>(user_data)->signal_to_wake();
},
this);
auto stream_wrapper = TRY(adopt_nonnull_ref_or_enomem(new (nothrow) PulseAudioStream(NonnullRefPtr(*this), stream)));
stream_wrapper->m_write_callback = move(write_callback);
pa_stream_set_write_callback(
stream, [](pa_stream* stream, size_t bytes_to_write, void* user_data) {
auto& stream_wrapper = *static_cast<PulseAudioStream*>(user_data);
VERIFY(stream_wrapper.m_stream == stream);
stream_wrapper.on_write_requested(bytes_to_write);
},
stream_wrapper.ptr());
// Borrowing logic from cubeb to set reasonable buffer sizes for a target latency:
// https://searchfox.org/mozilla-central/rev/3b707c8fd7e978eebf24279ee51ccf07895cfbcb/third_party/rust/cubeb-sys/libcubeb/src/cubeb_pulse.c#910-927
pa_buffer_attr buffer_attributes;
buffer_attributes.maxlength = -1;
buffer_attributes.prebuf = -1;
buffer_attributes.tlength = target_latency_ms * sample_rate / 1000;
buffer_attributes.minreq = buffer_attributes.tlength / 4;
buffer_attributes.fragsize = buffer_attributes.minreq;
auto flags = static_cast<pa_stream_flags>(PA_STREAM_AUTO_TIMING_UPDATE | PA_STREAM_INTERPOLATE_TIMING | PA_STREAM_ADJUST_LATENCY | PA_STREAM_RELATIVE_VOLUME);
if (initial_state == OutputState::Suspended) {
stream_wrapper->m_suspended = true;
flags = static_cast<pa_stream_flags>(static_cast<u32>(flags) | PA_STREAM_START_CORKED);
}
// This is a workaround for an issue with starting the stream corked, see PulseAudioPlaybackStream::total_time_played().
pa_stream_set_started_callback(
stream, [](pa_stream* stream, void* user_data) {
static_cast<PulseAudioStream*>(user_data)->m_started_playback = true;
pa_stream_set_started_callback(stream, nullptr, nullptr);
},
stream_wrapper.ptr());
pa_stream_set_underflow_callback(
stream, [](pa_stream*, void* user_data) {
auto& stream = *static_cast<PulseAudioStream*>(user_data);
if (stream.m_underrun_callback)
stream.m_underrun_callback();
},
stream_wrapper.ptr());
if (auto error = pa_stream_connect_playback(stream, nullptr, &buffer_attributes, flags, nullptr, nullptr); error != 0) {
warnln("PulseAudio stream connection failed with error: {}", pulse_audio_error_to_string(static_cast<PulseAudioErrorCode>(error)));
return Error::from_string_literal("Error while connecting the PulseAudio stream");
}
// FIXME: This should be asynchronous if connection can take longer than a fraction of a second.
while (true) {
bool is_ready = false;
switch (stream_wrapper->get_connection_state()) {
case PulseAudioStreamState::Creating:
break;
case PulseAudioStreamState::Ready:
is_ready = true;
break;
case PulseAudioStreamState::Failed:
return Error::from_string_literal("Failed to connect to PulseAudio daemon");
case PulseAudioStreamState::Unconnected:
case PulseAudioStreamState::Terminated:
VERIFY_NOT_REACHED();
break;
}
if (is_ready)
break;
wait_for_signal();
}
return stream_wrapper;
}
PulseAudioStream::~PulseAudioStream()
{
pa_stream_unref(m_stream);
}
PulseAudioStreamState PulseAudioStream::get_connection_state()
{
return static_cast<PulseAudioStreamState>(pa_stream_get_state(m_stream));
}
bool PulseAudioStream::connection_is_good()
{
return PA_STREAM_IS_GOOD(pa_stream_get_state(m_stream));
}
void PulseAudioStream::set_underrun_callback(Function<void()> callback)
{
auto locker = m_context->main_loop_locker();
m_underrun_callback = move(callback);
}
u32 PulseAudioStream::sample_rate()
{
return pa_stream_get_sample_spec(m_stream)->rate;
}
size_t PulseAudioStream::sample_size()
{
return pa_sample_size(pa_stream_get_sample_spec(m_stream));
}
size_t PulseAudioStream::frame_size()
{
return pa_frame_size(pa_stream_get_sample_spec(m_stream));
}
u8 PulseAudioStream::channel_count()
{
return pa_stream_get_sample_spec(m_stream)->channels;
}
void PulseAudioStream::on_write_requested(size_t bytes_to_write)
{
VERIFY(m_write_callback);
if (m_suspended)
return;
while (bytes_to_write > 0) {
auto buffer = begin_write(bytes_to_write).release_value_but_fixme_should_propagate_errors();
auto frame_size = this->frame_size();
VERIFY(buffer.size() % frame_size == 0);
auto written_buffer = m_write_callback(*this, buffer, buffer.size() / frame_size);
if (written_buffer.size() == 0) {
cancel_write().release_value_but_fixme_should_propagate_errors();
break;
}
bytes_to_write -= written_buffer.size();
write(written_buffer).release_value_but_fixme_should_propagate_errors();
}
}
ErrorOr<Bytes> PulseAudioStream::begin_write(size_t bytes_to_write)
{
void* data_pointer;
size_t data_size = bytes_to_write;
if (pa_stream_begin_write(m_stream, &data_pointer, &data_size) != 0 || data_pointer == nullptr)
return Error::from_string_literal("Failed to get the playback stream's write buffer from PulseAudio");
return Bytes { data_pointer, data_size };
}
ErrorOr<void> PulseAudioStream::write(ReadonlyBytes data)
{
if (pa_stream_write(m_stream, data.data(), data.size(), nullptr, 0, PA_SEEK_RELATIVE) != 0)
return Error::from_string_literal("Failed to write data to PulseAudio playback stream");
return {};
}
ErrorOr<void> PulseAudioStream::cancel_write()
{
if (pa_stream_cancel_write(m_stream) != 0)
return Error::from_string_literal("Failed to get the playback stream's write buffer from PulseAudio");
return {};
}
bool PulseAudioStream::is_suspended() const
{
return m_suspended;
}
StringView pulse_audio_error_to_string(PulseAudioErrorCode code)
{
if (code < PulseAudioErrorCode::OK || code >= PulseAudioErrorCode::Sentinel)
return "Unknown error code"sv;
char const* string = pa_strerror(static_cast<int>(code));
return StringView { string, strlen(string) };
}
ErrorOr<void> PulseAudioStream::wait_for_operation(pa_operation* operation, StringView error_message)
{
while (pa_operation_get_state(operation) == PA_OPERATION_RUNNING)
m_context->wait_for_signal();
if (!m_context->connection_is_good() || !this->connection_is_good()) {
auto pulse_audio_error_name = pulse_audio_error_to_string(m_context->get_last_error());
warnln("Encountered stream error: {}", pulse_audio_error_name);
return Error::from_string_view(error_message);
}
pa_operation_unref(operation);
return {};
}
ErrorOr<void> PulseAudioStream::drain_and_suspend()
{
auto locker = m_context->main_loop_locker();
if (m_suspended)
return {};
m_suspended = true;
if (pa_stream_is_corked(m_stream) > 0)
return {};
TRY(wait_for_operation(pa_stream_drain(m_stream, STREAM_SIGNAL_CALLBACK(this)), "Draining PulseAudio stream failed"sv));
TRY(wait_for_operation(pa_stream_cork(m_stream, 1, STREAM_SIGNAL_CALLBACK(this)), "Corking PulseAudio stream after drain failed"sv));
return {};
}
ErrorOr<void> PulseAudioStream::flush_and_suspend()
{
auto locker = m_context->main_loop_locker();
if (m_suspended)
return {};
m_suspended = true;
if (pa_stream_is_corked(m_stream) > 0)
return {};
TRY(wait_for_operation(pa_stream_flush(m_stream, STREAM_SIGNAL_CALLBACK(this)), "Flushing PulseAudio stream failed"sv));
TRY(wait_for_operation(pa_stream_cork(m_stream, 1, STREAM_SIGNAL_CALLBACK(this)), "Corking PulseAudio stream after flush failed"sv));
return {};
}
ErrorOr<void> PulseAudioStream::resume()
{
auto locker = m_context->main_loop_locker();
if (!m_suspended)
return {};
m_suspended = false;
TRY(wait_for_operation(pa_stream_cork(m_stream, 0, STREAM_SIGNAL_CALLBACK(this)), "Uncorking PulseAudio stream failed"sv));
// Defer a write to the playback buffer on the PulseAudio main loop. Otherwise, playback will not
// begin again, despite the fact that we uncorked.
// NOTE: We ref here and then unref in the callback so that this stream will not be deleted until
// it finishes.
ref();
pa_mainloop_api_once(
m_context->m_api, [](pa_mainloop_api*, void* user_data) {
auto& stream = *static_cast<PulseAudioStream*>(user_data);
// NOTE: writable_size() returns -1 in case of an error. However, the value is still safe
// since begin_write() will interpret -1 as a default parameter and choose a good size.
auto bytes_to_write = pa_stream_writable_size(stream.m_stream);
stream.on_write_requested(bytes_to_write);
stream.unref();
},
this);
return {};
}
ErrorOr<Duration> PulseAudioStream::total_time_played()
{
auto locker = m_context->main_loop_locker();
// NOTE: This is a workaround for a PulseAudio issue. When a stream is started corked,
// the time smoother doesn't seem to be aware of it, so it will return the time
// since the stream was connected. Once the playback actually starts, the time
// resets back to zero. However, since we request monotonically-increasing time,
// this means that the smoother will register that it had a larger time before,
// and return that time instead, until we reach a timestamp greater than the
// last-returned time. If we never call pa_stream_get_time() until after giving
// the stream its first samples, the issue never occurs.
if (!m_started_playback)
return Duration::zero();
pa_usec_t time = 0;
auto error = pa_stream_get_time(m_stream, &time);
if (error == -PA_ERR_NODATA)
return Duration::zero();
if (error != 0)
return Error::from_string_literal("Failed to get time from PulseAudio stream");
if (time > NumericLimits<i64>::max()) {
warnln("WARNING: Audio time is too large!");
time -= NumericLimits<i64>::max();
}
return Duration::from_microseconds(static_cast<i64>(time));
}
ErrorOr<void> PulseAudioStream::set_volume(double volume)
{
auto locker = m_context->main_loop_locker();
auto index = pa_stream_get_index(m_stream);
if (index == PA_INVALID_INDEX)
return Error::from_string_literal("Failed to get PulseAudio stream index while setting volume");
auto pulse_volume = pa_sw_volume_from_linear(volume);
pa_cvolume per_channel_volumes;
pa_cvolume_set(&per_channel_volumes, channel_count(), pulse_volume);
auto* operation = pa_context_set_sink_input_volume(m_context->m_context, index, &per_channel_volumes, STREAM_SIGNAL_CALLBACK(this));
return wait_for_operation(operation, "Failed to set PulseAudio stream volume"sv);
}
}

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@ -0,0 +1,184 @@
/*
* Copyright (c) 2023, Gregory Bertilson <zaggy1024@gmail.com>
*
* SPDX-License-Identifier: BSD-2-Clause
*/
#pragma once
#include <AK/AtomicRefCounted.h>
#include <AK/Error.h>
#include <AK/NonnullRefPtr.h>
#include <AK/Time.h>
#include <LibAudio/Forward.h>
#include <LibAudio/PlaybackStream.h>
#include <LibAudio/SampleFormats.h>
#include <LibThreading/Thread.h>
#include <pulse/pulseaudio.h>
namespace Audio {
class PulseAudioStream;
enum class PulseAudioContextState {
Unconnected = PA_CONTEXT_UNCONNECTED,
Connecting = PA_CONTEXT_CONNECTING,
Authorizing = PA_CONTEXT_AUTHORIZING,
SettingName = PA_CONTEXT_SETTING_NAME,
Ready = PA_CONTEXT_READY,
Failed = PA_CONTEXT_FAILED,
Terminated = PA_CONTEXT_TERMINATED,
};
enum class PulseAudioErrorCode;
using PulseAudioDataRequestCallback = Function<ReadonlyBytes(PulseAudioStream&, Bytes buffer, size_t sample_count)>;
// A wrapper around the PulseAudio main loop and context structs.
// Generally, only one instance of this should be needed for a single process.
class PulseAudioContext
: public AtomicRefCounted<PulseAudioContext>
, public Weakable<PulseAudioContext> {
public:
static ErrorOr<NonnullRefPtr<PulseAudioContext>> instance();
explicit PulseAudioContext(pa_threaded_mainloop*, pa_mainloop_api*, pa_context*);
PulseAudioContext(PulseAudioContext const& other) = delete;
~PulseAudioContext();
bool current_thread_is_main_loop_thread();
void lock_main_loop();
void unlock_main_loop();
[[nodiscard]] auto main_loop_locker()
{
lock_main_loop();
return ScopeGuard([this]() { unlock_main_loop(); });
}
// Waits for signal_to_wake() to be called.
// This must be called with the main loop locked.
void wait_for_signal();
// Signals to wake all threads from calls to signal_to_wake()
void signal_to_wake();
PulseAudioContextState get_connection_state();
bool connection_is_good();
PulseAudioErrorCode get_last_error();
ErrorOr<NonnullRefPtr<PulseAudioStream>> create_stream(OutputState initial_state, u32 sample_rate, u8 channels, u32 target_latency_ms, PulseAudioDataRequestCallback write_callback);
private:
friend class PulseAudioStream;
pa_threaded_mainloop* m_main_loop { nullptr };
pa_mainloop_api* m_api { nullptr };
pa_context* m_context;
};
enum class PulseAudioStreamState {
Unconnected = PA_STREAM_UNCONNECTED,
Creating = PA_STREAM_CREATING,
Ready = PA_STREAM_READY,
Failed = PA_STREAM_FAILED,
Terminated = PA_STREAM_TERMINATED,
};
class PulseAudioStream : public AtomicRefCounted<PulseAudioStream> {
public:
static constexpr bool start_corked = true;
~PulseAudioStream();
PulseAudioStreamState get_connection_state();
bool connection_is_good();
// Sets the callback to be run when the server consumes more of the buffer than
// has been written yet.
void set_underrun_callback(Function<void()>);
u32 sample_rate();
size_t sample_size();
size_t frame_size();
u8 channel_count();
// Gets a data buffer that can be written to and then passed back to PulseAudio through
// the write() function. This avoids a copy vs directly calling write().
ErrorOr<Bytes> begin_write(size_t bytes_to_write = NumericLimits<size_t>::max());
// Writes a data buffer to the playback stream.
ErrorOr<void> write(ReadonlyBytes data);
// Cancels the previous begin_write() call.
ErrorOr<void> cancel_write();
bool is_suspended() const;
// Plays back all buffered data and corks the stream. Until resume() is called, no data
// will be written to the stream.
ErrorOr<void> drain_and_suspend();
// Drops all buffered data and corks the stream. Until resume() is called, no data will
// be written to the stream.
ErrorOr<void> flush_and_suspend();
// Uncorks the stream and forces data to be written to the buffers to force playback to
// resume as soon as possible.
ErrorOr<void> resume();
ErrorOr<Duration> total_time_played();
ErrorOr<void> set_volume(double volume);
PulseAudioContext& context() { return *m_context; }
private:
friend class PulseAudioContext;
explicit PulseAudioStream(NonnullRefPtr<PulseAudioContext>&& context, pa_stream* stream)
: m_context(context)
, m_stream(stream)
{
}
PulseAudioStream(PulseAudioStream const& other) = delete;
ErrorOr<void> wait_for_operation(pa_operation*, StringView error_message);
void on_write_requested(size_t bytes_to_write);
NonnullRefPtr<PulseAudioContext> m_context;
pa_stream* m_stream { nullptr };
bool m_started_playback { false };
PulseAudioDataRequestCallback m_write_callback { nullptr };
// Determines whether we will allow the write callback to run. This should only be true
// if the stream is becoming or is already corked.
bool m_suspended { false };
Function<void()> m_underrun_callback;
};
enum class PulseAudioErrorCode {
OK = 0,
AccessFailure,
UnknownCommand,
InvalidArgument,
EntityExists,
NoSuchEntity,
ConnectionRefused,
ProtocolError,
Timeout,
NoAuthenticationKey,
InternalError,
ConnectionTerminated,
EntityKilled,
InvalidServer,
NoduleInitFailed,
BadState,
NoData,
IncompatibleProtocolVersion,
DataTooLarge,
NotSupported,
Unknown,
NoExtension,
Obsolete,
NotImplemented,
CalledFromFork,
IOError,
Busy,
Sentinel
};
StringView pulse_audio_error_to_string(PulseAudioErrorCode code);
}