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Bug 1530715 - P31: Merge render_input into audiounit_input_callback. r=padenot
1. If AudioUnitRender return kAudioUnitErr_CannotDoInCurrentContext within a input-only stream, the input-only stream will keep feed silence data into the buffer instead of reporting the error. With this change, the error will be rendered as the returned value of the data callback to the underlying CoreAudio framework. 2. By merging the render_input into audiounit_input_callback and adjust the timing to call reinit_async, now the reinit_async is called at the line that is out of the main logic for feeding buffer data. In the scope of the main logic, there will be a critical section created by locking a Rust mutex within AudioUnitStream in the later mutex replacement. Without moving the reinit_async, which borrows AudioUnitStream as a mutable, out of the critical section, there will be a borrowing-twice issue. Differential Revision: https://phabricator.services.mozilla.com/D34064 --HG-- extra : moz-landing-system : lando
This commit is contained in:
Chun-Min Chang
parent
7a27f19d14
commit
f06595279a
@ -3,4 +3,4 @@ git repository using the update.sh script.
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The cubeb-coreaudio-rs git repository is: https://github.com/ChunMinChang/cubeb-coreaudio-rs
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The git commit ID used was 405a7c00ee0a98e396c58461da73b151fd755661 (2019-06-25 11:32:22 -0700)
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The git commit ID used was d7d9b40a6c9385a34412c111d537737f9d09bc37 (2019-06-25 11:32:22 -0700)
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@ -390,57 +390,137 @@ extern "C" fn audiounit_input_callback(
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input_frames: u32,
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_: *mut AudioBufferList,
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) -> OSStatus {
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let stm = unsafe { &mut *(user_ptr as *mut AudioUnitStream) };
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enum ErrorHandle {
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Return(OSStatus),
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Reinit,
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};
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assert!(!stm.input_unit.is_null());
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assert!(input_frames > 0);
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assert_eq!(bus, AU_IN_BUS);
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assert!(!user_ptr.is_null());
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let stm = unsafe { &mut *(user_ptr as *mut AudioUnitStream) };
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if stm.shutdown.load(Ordering::SeqCst) {
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cubeb_log!("({:p}) input shutdown", stm as *const AudioUnitStream);
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return NO_ERR;
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}
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let r = stm.render_input(flags, tstamp, bus, input_frames);
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if r != NO_ERR {
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return r;
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}
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let handler = |stm: &mut AudioUnitStream,
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flags: *mut AudioUnitRenderActionFlags,
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tstamp: *const AudioTimeStamp,
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bus: u32,
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input_frames: u32|
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-> (ErrorHandle, Option<State>) {
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// Create the AudioBufferList to store input.
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let mut input_buffer_list = AudioBufferList::default();
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input_buffer_list.mBuffers[0].mDataByteSize = stm.input_desc.mBytesPerFrame * input_frames;
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input_buffer_list.mBuffers[0].mData = ptr::null_mut();
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input_buffer_list.mBuffers[0].mNumberChannels = stm.input_desc.mChannelsPerFrame;
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input_buffer_list.mNumberBuffers = 1;
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// Full Duplex. We'll call data_callback in the AudioUnit output callback.
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if !stm.output_unit.is_null() {
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return NO_ERR;
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}
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assert!(!stm.input_unit.is_null());
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let status = audio_unit_render(
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stm.input_unit,
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flags,
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tstamp,
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bus,
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input_frames,
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&mut input_buffer_list,
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);
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if (status != NO_ERR)
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&& (status != kAudioUnitErr_CannotDoInCurrentContext || stm.output_unit.is_null())
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{
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return (ErrorHandle::Return(status), None);
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}
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let handle = if status == kAudioUnitErr_CannotDoInCurrentContext {
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assert!(!stm.output_unit.is_null());
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// kAudioUnitErr_CannotDoInCurrentContext is returned when using a BT
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// headset and the profile is changed from A2DP to HFP/HSP. The previous
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// output device is no longer valid and must be reset.
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// For now state that no error occurred and feed silence, stream will be
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// resumed once reinit has completed.
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cubeb_logv!(
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"({:p}) input: reinit pending feeding silence instead",
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stm as *const AudioUnitStream
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);
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stm.input_linear_buffer
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.as_mut()
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.unwrap()
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.push_zeros((input_frames * stm.input_desc.mChannelsPerFrame) as usize);
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ErrorHandle::Reinit
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} else {
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assert_eq!(status, NO_ERR);
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// Copy input data in linear buffer.
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stm.input_linear_buffer.as_mut().unwrap().push(
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input_buffer_list.mBuffers[0].mData,
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(input_frames * stm.input_desc.mChannelsPerFrame) as usize,
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);
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ErrorHandle::Return(status)
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};
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// Input only. Call the user callback through resampler.
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// Resampler will deliver input buffer in the correct rate.
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assert!(
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input_frames as usize
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<= stm.input_linear_buffer.as_ref().unwrap().elements()
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// Advance input frame counter.
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stm.frames_read
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.fetch_add(i64::from(input_frames), atomic::Ordering::SeqCst);
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cubeb_logv!(
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"({:p}) input: buffers {}, size {}, channels {}, rendered frames {}, total frames {}.",
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stm as *const AudioUnitStream,
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input_buffer_list.mNumberBuffers,
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input_buffer_list.mBuffers[0].mDataByteSize,
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input_buffer_list.mBuffers[0].mNumberChannels,
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input_frames,
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stm.input_linear_buffer.as_ref().unwrap().elements()
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/ stm.input_desc.mChannelsPerFrame as usize
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);
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let mut total_input_frames = (stm.input_linear_buffer.as_ref().unwrap().elements()
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/ stm.input_desc.mChannelsPerFrame as usize) as i64;
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assert!(!stm.input_linear_buffer.as_ref().unwrap().as_ptr().is_null());
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let outframes = {
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let mut stream_device = stm.stream_device.lock().unwrap();
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stream_device.resampler.fill(
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stm.input_linear_buffer.as_mut().unwrap().as_mut_ptr(),
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&mut total_input_frames,
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ptr::null_mut(),
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0,
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)
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);
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// Full Duplex. We'll call data_callback in the AudioUnit output callback.
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if !stm.output_unit.is_null() {
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return (handle, None);
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}
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// Input only. Call the user callback through resampler.
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// Resampler will deliver input buffer in the correct rate.
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assert!(
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input_frames as usize
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<= stm.input_linear_buffer.as_ref().unwrap().elements()
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/ stm.input_desc.mChannelsPerFrame as usize
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);
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let mut total_input_frames = (stm.input_linear_buffer.as_ref().unwrap().elements()
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/ stm.input_desc.mChannelsPerFrame as usize)
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as i64;
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assert!(!stm.input_linear_buffer.as_ref().unwrap().as_ptr().is_null());
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let outframes = {
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let mut stream_device = stm.stream_device.lock().unwrap();
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stream_device.resampler.fill(
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stm.input_linear_buffer.as_mut().unwrap().as_mut_ptr(),
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&mut total_input_frames,
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ptr::null_mut(),
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0,
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)
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};
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if outframes < total_input_frames {
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assert_eq!(audio_output_unit_stop(stm.input_unit), NO_ERR);
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return (handle, Some(State::Drained));
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}
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// Reset input buffer
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stm.input_linear_buffer.as_mut().unwrap().clear();
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(handle, None)
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};
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if outframes < total_input_frames {
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assert_eq!(audio_output_unit_stop(stm.input_unit), NO_ERR);
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stm.notify_state_changed(State::Drained);
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return NO_ERR;
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let (handle, notification) = handler(stm, flags, tstamp, bus, input_frames);
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if let Some(state) = notification {
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stm.notify_state_changed(state);
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}
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// Reset input buffer
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stm.input_linear_buffer.as_mut().unwrap().clear();
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NO_ERR
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let status = match handle {
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ErrorHandle::Reinit => {
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stm.reinit_async();
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NO_ERR
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}
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ErrorHandle::Return(s) => s,
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};
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status
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}
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extern "C" fn audiounit_output_callback(
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@ -2570,78 +2650,6 @@ impl<'ctx> AudioUnitStream<'ctx> {
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self.output_stream_params.rate() > 0
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}
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fn render_input(
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&mut self,
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flags: *mut AudioUnitRenderActionFlags,
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tstamp: *const AudioTimeStamp,
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bus: u32,
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input_frames: u32,
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) -> OSStatus {
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// Create the AudioBufferList to store input.
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let mut input_buffer_list = AudioBufferList::default();
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input_buffer_list.mBuffers[0].mDataByteSize = self.input_desc.mBytesPerFrame * input_frames;
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input_buffer_list.mBuffers[0].mData = ptr::null_mut();
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input_buffer_list.mBuffers[0].mNumberChannels = self.input_desc.mChannelsPerFrame;
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input_buffer_list.mNumberBuffers = 1;
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assert!(!self.input_unit.is_null());
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let r = audio_unit_render(
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self.input_unit,
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flags,
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tstamp,
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bus,
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input_frames,
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&mut input_buffer_list,
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);
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if r != NO_ERR {
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cubeb_log!("AudioUnitRender rv={}", r);
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if r != kAudioUnitErr_CannotDoInCurrentContext {
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return r;
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}
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if !self.output_unit.is_null() {
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// kAudioUnitErr_CannotDoInCurrentContext is returned when using a BT
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// headset and the profile is changed from A2DP to HFP/HSP. The previous
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// output device is no longer valid and must be reset.
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self.reinit_async();
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}
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// For now state that no error occurred and feed silence, stream will be
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// resumed once reinit has completed.
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cubeb_logv!(
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"({:p}) input: reinit pending feeding silence instead",
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self as *const AudioUnitStream
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);
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self.input_linear_buffer
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.as_mut()
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.unwrap()
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.push_zeros((input_frames * self.input_desc.mChannelsPerFrame) as usize);
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} else {
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// Copy input data in linear buffer.
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self.input_linear_buffer.as_mut().unwrap().push(
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input_buffer_list.mBuffers[0].mData,
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(input_frames * self.input_desc.mChannelsPerFrame) as usize,
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);
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}
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// Advance input frame counter.
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assert!(input_frames > 0);
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self.frames_read
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.fetch_add(i64::from(input_frames), Ordering::SeqCst);
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cubeb_logv!(
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"({:p}) input: buffers {}, size {}, channels {}, rendered frames {}, total frames {}.",
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self as *const AudioUnitStream,
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input_buffer_list.mNumberBuffers,
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input_buffer_list.mBuffers[0].mDataByteSize,
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input_buffer_list.mBuffers[0].mNumberChannels,
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input_frames,
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self.input_linear_buffer.as_ref().unwrap().elements()
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/ self.input_desc.mChannelsPerFrame as usize
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);
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NO_ERR
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}
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fn minimum_resampling_input_frames(&self, output_frames: i64) -> i64 {
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assert_ne!(self.input_hw_rate, 0_f64);
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assert_ne!(self.output_stream_params.rate(), 0);
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