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|
// Copyright (C) 2018 Sebastian Dröge <sebastian@centricular.com>
//
// Licensed under the Apache License, Version 2.0 <LICENSE-APACHE or
// http://www.apache.org/licenses/LICENSE-2.0> or the MIT license
// <LICENSE-MIT or http://opensource.org/licenses/MIT>, at your
// option. This file may not be copied, modified, or distributed
// except according to those terms.
use glib::subclass::prelude::*;
use gst::prelude::*;
use gst::subclass::prelude::*;
use gst::{gst_debug, gst_error, gst_info, gst_log};
use gst_base::prelude::*;
use gst_base::subclass::prelude::*;
use byte_slice_cast::*;
use std::ops::Rem;
use std::sync::Mutex;
use std::{i32, u32};
use num_traits::cast::NumCast;
use num_traits::float::Float;
use once_cell::sync::Lazy;
// This module contains the private implementation details of our element
static CAT: Lazy<gst::DebugCategory> = Lazy::new(|| {
gst::DebugCategory::new(
"rssinesrc",
gst::DebugColorFlags::empty(),
Some("Rust Sine Wave Source"),
)
});
// Default values of properties
const DEFAULT_SAMPLES_PER_BUFFER: u32 = 1024;
const DEFAULT_FREQ: u32 = 440;
const DEFAULT_VOLUME: f64 = 0.8;
const DEFAULT_MUTE: bool = false;
const DEFAULT_IS_LIVE: bool = false;
// Property value storage
#[derive(Debug, Clone, Copy)]
struct Settings {
samples_per_buffer: u32,
freq: u32,
volume: f64,
mute: bool,
is_live: bool,
}
impl Default for Settings {
fn default() -> Self {
Settings {
samples_per_buffer: DEFAULT_SAMPLES_PER_BUFFER,
freq: DEFAULT_FREQ,
volume: DEFAULT_VOLUME,
mute: DEFAULT_MUTE,
is_live: DEFAULT_IS_LIVE,
}
}
}
// Stream-specific state, i.e. audio format configuration
// and sample offset
struct State {
info: Option<gst_audio::AudioInfo>,
sample_offset: u64,
sample_stop: Option<u64>,
accumulator: f64,
}
impl Default for State {
fn default() -> State {
State {
info: None,
sample_offset: 0,
sample_stop: None,
accumulator: 0.0,
}
}
}
struct ClockWait {
clock_id: Option<gst::SingleShotClockId>,
flushing: bool,
}
impl Default for ClockWait {
fn default() -> ClockWait {
ClockWait {
clock_id: None,
flushing: true,
}
}
}
// Struct containing all the element data
#[derive(Default)]
pub struct SineSrc {
settings: Mutex<Settings>,
state: Mutex<State>,
clock_wait: Mutex<ClockWait>,
}
impl SineSrc {
fn process<F: Float + FromByteSlice>(
data: &mut [u8],
accumulator_ref: &mut f64,
freq: u32,
rate: u32,
channels: u32,
vol: f64,
) {
use std::f64::consts::PI;
// Reinterpret our byte-slice as a slice containing elements of the type
// we're interested in. GStreamer requires for raw audio that the alignment
// of memory is correct, so this will never ever fail unless there is an
// actual bug elsewhere.
let data = data.as_mut_slice_of::<F>().unwrap();
// Convert all our parameters to the target type for calculations
let vol: F = NumCast::from(vol).unwrap();
let freq = freq as f64;
let rate = rate as f64;
let two_pi = 2.0 * PI;
// We're carrying a accumulator with up to 2pi around instead of working
// on the sample offset. High sample offsets cause too much inaccuracy when
// converted to floating point numbers and then iterated over in 1-steps
let mut accumulator = *accumulator_ref;
let step = two_pi * freq / rate;
for chunk in data.chunks_exact_mut(channels as usize) {
let value = vol * F::sin(NumCast::from(accumulator).unwrap());
for sample in chunk {
*sample = value;
}
accumulator += step;
if accumulator >= two_pi {
accumulator -= two_pi;
}
}
*accumulator_ref = accumulator;
}
}
// This trait registers our type with the GObject object system and
// provides the entry points for creating a new instance and setting
// up the class data
#[glib::object_subclass]
impl ObjectSubclass for SineSrc {
const NAME: &'static str = "RsSineSrc";
type Type = super::SineSrc;
type ParentType = gst_base::PushSrc;
}
// Implementation of glib::Object virtual methods
impl ObjectImpl for SineSrc {
// Metadata for the properties
fn properties() -> &'static [glib::ParamSpec] {
static PROPERTIES: Lazy<Vec<glib::ParamSpec>> = Lazy::new(|| {
vec![
glib::ParamSpec::uint(
"samples-per-buffer",
"Samples Per Buffer",
"Number of samples per output buffer",
1,
u32::MAX,
DEFAULT_SAMPLES_PER_BUFFER,
glib::ParamFlags::READWRITE | gst::PARAM_FLAG_MUTABLE_READY,
),
glib::ParamSpec::uint(
"freq",
"Frequency",
"Frequency",
1,
u32::MAX,
DEFAULT_FREQ,
glib::ParamFlags::READWRITE | gst::PARAM_FLAG_MUTABLE_PLAYING,
),
glib::ParamSpec::double(
"volume",
"Volume",
"Output volume",
0.0,
10.0,
DEFAULT_VOLUME,
glib::ParamFlags::READWRITE | gst::PARAM_FLAG_MUTABLE_PLAYING,
),
glib::ParamSpec::boolean(
"mute",
"Mute",
"Mute",
DEFAULT_MUTE,
glib::ParamFlags::READWRITE | gst::PARAM_FLAG_MUTABLE_PLAYING,
),
glib::ParamSpec::boolean(
"is-live",
"Is Live",
"(Pseudo) live output",
DEFAULT_IS_LIVE,
glib::ParamFlags::READWRITE | gst::PARAM_FLAG_MUTABLE_READY,
),
]
});
PROPERTIES.as_ref()
}
// Called right after construction of a new instance
fn constructed(&self, obj: &Self::Type) {
// Call the parent class' ::constructed() implementation first
self.parent_constructed(obj);
// Initialize live-ness and notify the base class that
// we'd like to operate in Time format
obj.set_live(DEFAULT_IS_LIVE);
obj.set_format(gst::Format::Time);
}
// Called whenever a value of a property is changed. It can be called
// at any time from any thread.
fn set_property(
&self,
obj: &Self::Type,
_id: usize,
value: &glib::Value,
pspec: &glib::ParamSpec,
) {
match pspec.name() {
"samples-per-buffer" => {
let mut settings = self.settings.lock().unwrap();
let samples_per_buffer = value.get_some().expect("type checked upstream");
gst_info!(
CAT,
obj: obj,
"Changing samples-per-buffer from {} to {}",
settings.samples_per_buffer,
samples_per_buffer
);
settings.samples_per_buffer = samples_per_buffer;
drop(settings);
let _ = obj.post_message(gst::message::Latency::builder().src(obj).build());
}
"freq" => {
let mut settings = self.settings.lock().unwrap();
let freq = value.get_some().expect("type checked upstream");
gst_info!(
CAT,
obj: obj,
"Changing freq from {} to {}",
settings.freq,
freq
);
settings.freq = freq;
}
"volume" => {
let mut settings = self.settings.lock().unwrap();
let volume = value.get_some().expect("type checked upstream");
gst_info!(
CAT,
obj: obj,
"Changing volume from {} to {}",
settings.volume,
volume
);
settings.volume = volume;
}
"mute" => {
let mut settings = self.settings.lock().unwrap();
let mute = value.get_some().expect("type checked upstream");
gst_info!(
CAT,
obj: obj,
"Changing mute from {} to {}",
settings.mute,
mute
);
settings.mute = mute;
}
"is-live" => {
let mut settings = self.settings.lock().unwrap();
let is_live = value.get_some().expect("type checked upstream");
gst_info!(
CAT,
obj: obj,
"Changing is-live from {} to {}",
settings.is_live,
is_live
);
settings.is_live = is_live;
}
_ => unimplemented!(),
}
}
// Called whenever a value of a property is read. It can be called
// at any time from any thread.
fn get_property(&self, _obj: &Self::Type, _id: usize, pspec: &glib::ParamSpec) -> glib::Value {
match pspec.name() {
"samples-per-buffer" => {
let settings = self.settings.lock().unwrap();
settings.samples_per_buffer.to_value()
}
"freq" => {
let settings = self.settings.lock().unwrap();
settings.freq.to_value()
}
"volume" => {
let settings = self.settings.lock().unwrap();
settings.volume.to_value()
}
"mute" => {
let settings = self.settings.lock().unwrap();
settings.mute.to_value()
}
"is-live" => {
let settings = self.settings.lock().unwrap();
settings.is_live.to_value()
}
_ => unimplemented!(),
}
}
}
// Implementation of gst::Element virtual methods
impl ElementImpl for SineSrc {
// Set the element specific metadata. This information is what
// is visible from gst-inspect-1.0 and can also be programatically
// retrieved from the gst::Registry after initial registration
// without having to load the plugin in memory.
fn metadata() -> Option<&'static gst::subclass::ElementMetadata> {
static ELEMENT_METADATA: Lazy<gst::subclass::ElementMetadata> = Lazy::new(|| {
gst::subclass::ElementMetadata::new(
"Sine Wave Source",
"Source/Audio",
"Creates a sine wave",
"Sebastian Dröge <sebastian@centricular.com>",
)
});
Some(&*ELEMENT_METADATA)
}
// Create and add pad templates for our sink and source pad. These
// are later used for actually creating the pads and beforehand
// already provide information to GStreamer about all possible
// pads that could exist for this type.
fn pad_templates() -> &'static [gst::PadTemplate] {
static PAD_TEMPLATES: Lazy<Vec<gst::PadTemplate>> = Lazy::new(|| {
// On the src pad, we can produce F32/F64 with any sample rate
// and any number of channels
let caps = gst::Caps::new_simple(
"audio/x-raw",
&[
(
"format",
&gst::List::new(&[
&gst_audio::AUDIO_FORMAT_F32.to_str(),
&gst_audio::AUDIO_FORMAT_F64.to_str(),
]),
),
("layout", &"interleaved"),
("rate", &gst::IntRange::<i32>::new(1, i32::MAX)),
("channels", &gst::IntRange::<i32>::new(1, i32::MAX)),
],
);
// The src pad template must be named "src" for basesrc
// and specific a pad that is always there
let src_pad_template = gst::PadTemplate::new(
"src",
gst::PadDirection::Src,
gst::PadPresence::Always,
&caps,
)
.unwrap();
vec![src_pad_template]
});
PAD_TEMPLATES.as_ref()
}
// Called whenever the state of the element should be changed. This allows for
// starting up the element, allocating/deallocating resources or shutting down
// the element again.
fn change_state(
&self,
element: &Self::Type,
transition: gst::StateChange,
) -> Result<gst::StateChangeSuccess, gst::StateChangeError> {
// Configure live'ness once here just before starting the source
if let gst::StateChange::ReadyToPaused = transition {
element.set_live(self.settings.lock().unwrap().is_live);
}
// Call the parent class' implementation of ::change_state()
self.parent_change_state(element, transition)
}
}
// Implementation of gst_base::BaseSrc virtual methods
impl BaseSrcImpl for SineSrc {
// Called whenever the input/output caps are changing, i.e. in the very beginning before data
// flow happens and whenever the situation in the pipeline is changing. All buffers after this
// call have the caps given here.
//
// We simply remember the resulting AudioInfo from the caps to be able to use this for knowing
// the sample rate, etc. when creating buffers
fn set_caps(&self, element: &Self::Type, caps: &gst::Caps) -> Result<(), gst::LoggableError> {
use std::f64::consts::PI;
let info = gst_audio::AudioInfo::from_caps(caps).map_err(|_| {
gst::loggable_error!(CAT, "Failed to build `AudioInfo` from caps {}", caps)
})?;
gst_debug!(CAT, obj: element, "Configuring for caps {}", caps);
element.set_blocksize(info.bpf() * (*self.settings.lock().unwrap()).samples_per_buffer);
let settings = *self.settings.lock().unwrap();
let mut state = self.state.lock().unwrap();
// If we have no caps yet, any old sample_offset and sample_stop will be
// in nanoseconds
let old_rate = match state.info {
Some(ref info) => info.rate() as u64,
None => gst::SECOND_VAL,
};
// Update sample offset and accumulator based on the previous values and the
// sample rate change, if any
let old_sample_offset = state.sample_offset;
let sample_offset = old_sample_offset
.mul_div_floor(info.rate() as u64, old_rate)
.unwrap();
let old_sample_stop = state.sample_stop;
let sample_stop =
old_sample_stop.map(|v| v.mul_div_floor(info.rate() as u64, old_rate).unwrap());
let accumulator =
(sample_offset as f64).rem(2.0 * PI * (settings.freq as f64) / (info.rate() as f64));
*state = State {
info: Some(info),
sample_offset,
sample_stop,
accumulator,
};
drop(state);
let _ = element.post_message(gst::message::Latency::builder().src(element).build());
Ok(())
}
// Called when starting, so we can initialize all stream-related state to its defaults
fn start(&self, element: &Self::Type) -> Result<(), gst::ErrorMessage> {
// Reset state
*self.state.lock().unwrap() = Default::default();
self.unlock_stop(element)?;
gst_info!(CAT, obj: element, "Started");
Ok(())
}
// Called when shutting down the element so we can release all stream-related state
fn stop(&self, element: &Self::Type) -> Result<(), gst::ErrorMessage> {
// Reset state
*self.state.lock().unwrap() = Default::default();
self.unlock(element)?;
gst_info!(CAT, obj: element, "Stopped");
Ok(())
}
fn query(&self, element: &Self::Type, query: &mut gst::QueryRef) -> bool {
use gst::QueryView;
match query.view_mut() {
// In Live mode we will have a latency equal to the number of samples in each buffer.
// We can't output samples before they were produced, and the last sample of a buffer
// is produced that much after the beginning, leading to this latency calculation
QueryView::Latency(ref mut q) => {
let settings = *self.settings.lock().unwrap();
let state = self.state.lock().unwrap();
if let Some(ref info) = state.info {
let latency = gst::SECOND
.mul_div_floor(settings.samples_per_buffer as u64, info.rate() as u64)
.unwrap();
gst_debug!(CAT, obj: element, "Returning latency {}", latency);
q.set(settings.is_live, latency, gst::CLOCK_TIME_NONE);
true
} else {
false
}
}
_ => BaseSrcImplExt::parent_query(self, element, query),
}
}
fn fixate(&self, element: &Self::Type, mut caps: gst::Caps) -> gst::Caps {
// Fixate the caps. BaseSrc will do some fixation for us, but
// as we allow any rate between 1 and MAX it would fixate to 1. 1Hz
// is generally not a useful sample rate.
//
// We fixate to the closest integer value to 48kHz that is possible
// here, and for good measure also decide that the closest value to 1
// channel is good.
caps.truncate();
{
let caps = caps.make_mut();
let s = caps.get_mut_structure(0).unwrap();
s.fixate_field_nearest_int("rate", 48_000);
s.fixate_field_nearest_int("channels", 1);
}
// Let BaseSrc fixate anything else for us. We could've alternatively have
// called caps.fixate() here
self.parent_fixate(element, caps)
}
fn is_seekable(&self, _element: &Self::Type) -> bool {
true
}
fn do_seek(&self, element: &Self::Type, segment: &mut gst::Segment) -> bool {
// Handle seeking here. For Time and Default (sample offset) seeks we can
// do something and have to update our sample offset and accumulator accordingly.
//
// Also we should remember the stop time (so we can stop at that point), and if
// reverse playback is requested. These values will all be used during buffer creation
// and for calculating the timestamps, etc.
if segment.rate() < 0.0 {
gst_error!(CAT, obj: element, "Reverse playback not supported");
return false;
}
let settings = *self.settings.lock().unwrap();
let mut state = self.state.lock().unwrap();
// We store sample_offset and sample_stop in nanoseconds if we
// don't know any sample rate yet. It will be converted correctly
// once a sample rate is known.
let rate = match state.info {
None => gst::SECOND_VAL,
Some(ref info) => info.rate() as u64,
};
if let Some(segment) = segment.downcast_ref::<gst::format::Time>() {
use std::f64::consts::PI;
let sample_offset = segment
.start()
.unwrap()
.mul_div_floor(rate, gst::SECOND_VAL)
.unwrap();
let sample_stop = segment
.stop()
.map(|v| v.mul_div_floor(rate, gst::SECOND_VAL).unwrap());
let accumulator =
(sample_offset as f64).rem(2.0 * PI * (settings.freq as f64) / (rate as f64));
gst_debug!(
CAT,
obj: element,
"Seeked to {}-{:?} (accum: {}) for segment {:?}",
sample_offset,
sample_stop,
accumulator,
segment
);
*state = State {
info: state.info.clone(),
sample_offset,
sample_stop,
accumulator,
};
true
} else if let Some(segment) = segment.downcast_ref::<gst::format::Default>() {
use std::f64::consts::PI;
if state.info.is_none() {
gst_error!(
CAT,
obj: element,
"Can only seek in Default format if sample rate is known"
);
return false;
}
let sample_offset = segment.start().unwrap();
let sample_stop = segment.stop().0;
let accumulator =
(sample_offset as f64).rem(2.0 * PI * (settings.freq as f64) / (rate as f64));
gst_debug!(
CAT,
obj: element,
"Seeked to {}-{:?} (accum: {}) for segment {:?}",
sample_offset,
sample_stop,
accumulator,
segment
);
*state = State {
info: state.info.clone(),
sample_offset,
sample_stop,
accumulator,
};
true
} else {
gst_error!(
CAT,
obj: element,
"Can't seek in format {:?}",
segment.format()
);
false
}
}
fn unlock(&self, element: &Self::Type) -> Result<(), gst::ErrorMessage> {
// This should unblock the create() function ASAP, so we
// just unschedule the clock it here, if any.
gst_debug!(CAT, obj: element, "Unlocking");
let mut clock_wait = self.clock_wait.lock().unwrap();
if let Some(clock_id) = clock_wait.clock_id.take() {
clock_id.unschedule();
}
clock_wait.flushing = true;
Ok(())
}
fn unlock_stop(&self, element: &Self::Type) -> Result<(), gst::ErrorMessage> {
// This signals that unlocking is done, so we can reset
// all values again.
gst_debug!(CAT, obj: element, "Unlock stop");
let mut clock_wait = self.clock_wait.lock().unwrap();
clock_wait.flushing = false;
Ok(())
}
}
impl PushSrcImpl for SineSrc {
// Creates the audio buffers
fn create(&self, element: &Self::Type) -> Result<gst::Buffer, gst::FlowError> {
// Keep a local copy of the values of all our properties at this very moment. This
// ensures that the mutex is never locked for long and the application wouldn't
// have to block until this function returns when getting/setting property values
let settings = *self.settings.lock().unwrap();
// Get a locked reference to our state, i.e. the input and output AudioInfo
let mut state = self.state.lock().unwrap();
let info = match state.info {
None => {
gst::element_error!(element, gst::CoreError::Negotiation, ["Have no caps yet"]);
return Err(gst::FlowError::NotNegotiated);
}
Some(ref info) => info.clone(),
};
// If a stop position is set (from a seek), only produce samples up to that
// point but at most samples_per_buffer samples per buffer
let n_samples = if let Some(sample_stop) = state.sample_stop {
if sample_stop <= state.sample_offset {
gst_log!(CAT, obj: element, "At EOS");
return Err(gst::FlowError::Eos);
}
sample_stop - state.sample_offset
} else {
settings.samples_per_buffer as u64
};
// Allocate a new buffer of the required size, update the metadata with the
// current timestamp and duration and then fill it according to the current
// caps
let mut buffer =
gst::Buffer::with_size((n_samples as usize) * (info.bpf() as usize)).unwrap();
{
let buffer = buffer.get_mut().unwrap();
// Calculate the current timestamp (PTS) and the next one,
// and calculate the duration from the difference instead of
// simply the number of samples to prevent rounding errors
let pts = state
.sample_offset
.mul_div_floor(gst::SECOND_VAL, info.rate() as u64)
.unwrap()
.into();
let next_pts: gst::ClockTime = (state.sample_offset + n_samples)
.mul_div_floor(gst::SECOND_VAL, info.rate() as u64)
.unwrap()
.into();
buffer.set_pts(pts);
buffer.set_duration(next_pts - pts);
// Map the buffer writable and create the actual samples
let mut map = buffer.map_writable().unwrap();
let data = map.as_mut_slice();
if info.format() == gst_audio::AUDIO_FORMAT_F32 {
Self::process::<f32>(
data,
&mut state.accumulator,
settings.freq,
info.rate(),
info.channels(),
settings.volume,
);
} else {
Self::process::<f64>(
data,
&mut state.accumulator,
settings.freq,
info.rate(),
info.channels(),
settings.volume,
);
}
}
state.sample_offset += n_samples;
drop(state);
// If we're live, we are waiting until the time of the last sample in our buffer has
// arrived. This is the very reason why we have to report that much latency.
// A real live-source would of course only allow us to have the data available after
// that latency, e.g. when capturing from a microphone, and no waiting from our side
// would be necessary..
//
// Waiting happens based on the pipeline clock, which means that a real live source
// with its own clock would require various translations between the two clocks.
// This is out of scope for the tutorial though.
if element.is_live() {
let clock = match element.clock() {
None => return Ok(buffer),
Some(clock) => clock,
};
let segment = element.segment().downcast::<gst::format::Time>().unwrap();
let base_time = element.base_time();
let running_time = segment.to_running_time(buffer.pts() + buffer.duration());
// The last sample's clock time is the base time of the element plus the
// running time of the last sample
let wait_until = running_time + base_time;
if wait_until.is_none() {
return Ok(buffer);
}
// Store the clock ID in our struct unless we're flushing anyway.
// This allows to asynchronously cancel the waiting from unlock()
// so that we immediately stop waiting on e.g. shutdown.
let mut clock_wait = self.clock_wait.lock().unwrap();
if clock_wait.flushing {
gst_debug!(CAT, obj: element, "Flushing");
return Err(gst::FlowError::Flushing);
}
let id = clock.new_single_shot_id(wait_until);
clock_wait.clock_id = Some(id.clone());
drop(clock_wait);
gst_log!(
CAT,
obj: element,
"Waiting until {}, now {}",
wait_until,
clock.time()
);
let (res, jitter) = id.wait();
gst_log!(CAT, obj: element, "Waited res {:?} jitter {}", res, jitter);
self.clock_wait.lock().unwrap().clock_id.take();
// If the clock ID was unscheduled, unlock() was called
// and we should return Flushing immediately.
if res == Err(gst::ClockError::Unscheduled) {
gst_debug!(CAT, obj: element, "Flushing");
return Err(gst::FlowError::Flushing);
}
}
gst_debug!(CAT, obj: element, "Produced buffer {:?}", buffer);
Ok(buffer)
}
}
|