WIP Add timestamp implementation to coreaudio backend

Currently a rough estimate of the required work, not yet tested.
This commit is contained in:
mitchmindtree 2020-04-27 20:29:51 +02:00
parent 777a6b2bd1
commit 54e5e95705
1 changed files with 49 additions and 3 deletions

View File

@ -637,6 +637,7 @@ impl Device {
// Register the callback that is being called by coreaudio whenever it needs data to be
// fed to the audio buffer.
let bytes_per_channel = sample_format.sample_size();
let sample_rate = config.sample_rate;
type Args = render_callback::Args<data::Raw>;
audio_unit.set_input_callback(move |args: Args| unsafe {
let ptr = (*args.data.data).mBuffers.as_ptr() as *const AudioBuffer;
@ -653,7 +654,17 @@ impl Device {
let data = data as *mut ();
let len = (data_byte_size as usize / bytes_per_channel) as usize;
let data = Data::from_parts(data, len, sample_format);
let info = InputCallbackInfo {};
// TODO: Need a better way to get delay, for now we assume a double-buffer offset.
let callback = host_time_to_stream_instant(args.time_stamp.mHostTime);
let buffer_frames = len / channels;
let delay = frames_to_duration(buffer_frames, sample_rate);
let capture = callback
.sub(delay)
.expect("`capture` occurs before origin of alsa `StreamInstant`");
let timestamp = crate::InputStreamTimestamp { callback, capture };
let info = InputCallbackInfo { timestamp };
data_callback(&data, &info);
Ok(())
})?;
@ -691,13 +702,14 @@ impl Device {
// Register the callback that is being called by coreaudio whenever it needs data to be
// fed to the audio buffer.
let bytes_per_channel = sample_format.sample_size();
let sample_rate = config.sample_rate;
type Args = render_callback::Args<data::Raw>;
audio_unit.set_render_callback(move |args: Args| unsafe {
// If `run()` is currently running, then a callback will be available from this list.
// Otherwise, we just fill the buffer with zeroes and return.
let AudioBuffer {
mNumberChannels: _num_channels,
mNumberChannels: channels,
mDataByteSize: data_byte_size,
mData: data,
} = (*args.data.data).mBuffers[0];
@ -705,7 +717,17 @@ impl Device {
let data = data as *mut ();
let len = (data_byte_size as usize / bytes_per_channel) as usize;
let mut data = Data::from_parts(data, len, sample_format);
let info = OutputCallbackInfo {};
let callback = host_time_to_stream_instant(args.time_stamp.mHostTime);
// TODO: Need a better way to get delay, for now we assume a double-buffer offset.
let buffer_frames = len / channels;
let delay = frames_to_duration(buffer_frames, sample_rate);
let playback = callback
.add(delay)
.expect("`playback` occurs beyond representation supported by `StreamInstant`");
let timestamp = crate::OutputStreamTimestamp { callback, playback };
let info = OutputCallbackInfo { timestamp };
data_callback(&mut data, &info);
Ok(())
})?;
@ -720,6 +742,30 @@ impl Device {
}
}
fn host_time_to_stream_instant(
m_host_time: u64,
) -> Result<crate::StreamInstant, BackendSpecificError> {
let mut info: coreaudio::sys::mach_timebase_info_data_t = unimplemented!();
let res = coreaudio::sys::mach_timebase_info(&mut info);
if res != SUCCESS {
let description = unimplemented!();
let err = BackendSpecificError { description };
return Err(err.into());
}
let nanos = m_host_time * info.numer / info.denom;
let secs = nanos / 1_000_000_000;
let subsec_nanos = nanos - secs * 1_000_000_000;
crate::StreamInstant::new(secs, subsec_nanos)
}
// Convert the given duration in frames at the given sample rate to a `std::time::Duration`.
fn frames_to_duration(frames: usize, rate: crate::SampleRate) -> std::time::Duration {
let secsf = frames as f64 / rate.0 as f64;
let secs = secsf as u64;
let nanos = ((secsf - secs as f64) * 1_000_000_000.0) as u32;
std::time::Duration::new(secs, nanos)
}
pub struct Stream {
inner: RefCell<StreamInner>,
}