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authorcrueter <crueter@eden-emu.dev>2025-08-04 04:50:14 +0200
committercrueter <crueter@eden-emu.dev>2025-08-04 04:50:14 +0200
commit51b170b470ed587f99a7c2a773e4464687f0d07a (patch)
tree4c03c510cfb17a5f5d6233df490d78f06dcfa7f7 /externals/oboe/src/common/AudioStream.cpp
parent04e5e6453875ffa5c79cdb421a29bc19985b1049 (diff)
[cmake] refactor: Use CPM over submodules (#143)
Transfers the majority of submodules and large externals to CPM, using source archives rather than full Git clones. Not only does this save massive amounts of clone and configure time, but dependencies are grabbed on-demand rather than being required by default. Additionally, CPM will (generally) automatically search for system dependencies, though certain dependencies have options to control this. Testing shows gains ranging from 5x to 10x in terms of overall clone/configure time. Reviewed-on: https://git.eden-emu.dev/eden-emu/eden/pulls/143 Reviewed-by: CamilleLaVey <camillelavey99@gmail.com>
Diffstat (limited to 'externals/oboe/src/common/AudioStream.cpp')
-rw-r--r--externals/oboe/src/common/AudioStream.cpp232
1 files changed, 0 insertions, 232 deletions
diff --git a/externals/oboe/src/common/AudioStream.cpp b/externals/oboe/src/common/AudioStream.cpp
deleted file mode 100644
index 5eb4e4629e..0000000000
--- a/externals/oboe/src/common/AudioStream.cpp
+++ /dev/null
@@ -1,232 +0,0 @@
-/*
- * Copyright 2015 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include <sys/types.h>
-#include <pthread.h>
-#include <thread>
-
-#include "oboe/AudioClock.h"
-#include "oboe/AudioStream.h"
-#include "oboe/Utilities.h"
-#include "OboeDebug.h"
-
-namespace oboe {
-
-/*
- * AudioStream
- */
-AudioStream::AudioStream(const AudioStreamBuilder &builder)
- : AudioStreamBase(builder) {
- LOGD("Constructor for AudioStream at %p", this);
-}
-
-AudioStream::~AudioStream() {
- // This is to help debug use after free bugs.
- LOGD("Destructor for AudioStream at %p", this);
-}
-
-Result AudioStream::close() {
- closePerformanceHint();
- // Update local counters so they can be read after the close.
- updateFramesWritten();
- updateFramesRead();
- return Result::OK;
-}
-
-// Call this from fireDataCallback() if you want to monitor CPU scheduler.
-void AudioStream::checkScheduler() {
- int scheduler = sched_getscheduler(0) & ~SCHED_RESET_ON_FORK; // for current thread
- if (scheduler != mPreviousScheduler) {
- LOGD("AudioStream::%s() scheduler = %s", __func__,
- ((scheduler == SCHED_FIFO) ? "SCHED_FIFO" :
- ((scheduler == SCHED_OTHER) ? "SCHED_OTHER" :
- ((scheduler == SCHED_RR) ? "SCHED_RR" : "UNKNOWN")))
- );
- mPreviousScheduler = scheduler;
- }
-}
-
-DataCallbackResult AudioStream::fireDataCallback(void *audioData, int32_t numFrames) {
- if (!isDataCallbackEnabled()) {
- LOGW("AudioStream::%s() called with data callback disabled!", __func__);
- return DataCallbackResult::Stop; // Should not be getting called
- }
-
- beginPerformanceHintInCallback();
-
- // Call the app to do the work.
- DataCallbackResult result;
- if (mDataCallback) {
- result = mDataCallback->onAudioReady(this, audioData, numFrames);
- } else {
- result = onDefaultCallback(audioData, numFrames);
- }
- // On Oreo, we might get called after returning stop.
- // So block that here.
- setDataCallbackEnabled(result == DataCallbackResult::Continue);
-
- endPerformanceHintInCallback(numFrames);
-
- return result;
-}
-
-Result AudioStream::waitForStateTransition(StreamState startingState,
- StreamState endingState,
- int64_t timeoutNanoseconds)
-{
- StreamState state;
- {
- std::lock_guard<std::mutex> lock(mLock);
- state = getState();
- if (state == StreamState::Closed) {
- return Result::ErrorClosed;
- } else if (state == StreamState::Disconnected) {
- return Result::ErrorDisconnected;
- }
- }
-
- StreamState nextState = state;
- // TODO Should this be a while()?!
- if (state == startingState && state != endingState) {
- Result result = waitForStateChange(state, &nextState, timeoutNanoseconds);
- if (result != Result::OK) {
- return result;
- }
- }
-
- if (nextState != endingState) {
- return Result::ErrorInvalidState;
- } else {
- return Result::OK;
- }
-}
-
-Result AudioStream::start(int64_t timeoutNanoseconds)
-{
- Result result = requestStart();
- if (result != Result::OK) return result;
- if (timeoutNanoseconds <= 0) return result;
- result = waitForStateTransition(StreamState::Starting,
- StreamState::Started, timeoutNanoseconds);
- if (result != Result::OK) {
- LOGE("AudioStream::%s() timed out before moving from STARTING to STARTED", __func__);
- }
- return result;
-}
-
-Result AudioStream::pause(int64_t timeoutNanoseconds)
-{
- Result result = requestPause();
- if (result != Result::OK) return result;
- if (timeoutNanoseconds <= 0) return result;
- return waitForStateTransition(StreamState::Pausing,
- StreamState::Paused, timeoutNanoseconds);
-}
-
-Result AudioStream::flush(int64_t timeoutNanoseconds)
-{
- Result result = requestFlush();
- if (result != Result::OK) return result;
- if (timeoutNanoseconds <= 0) return result;
- return waitForStateTransition(StreamState::Flushing,
- StreamState::Flushed, timeoutNanoseconds);
-}
-
-Result AudioStream::stop(int64_t timeoutNanoseconds)
-{
- Result result = requestStop();
- if (result != Result::OK) return result;
- if (timeoutNanoseconds <= 0) return result;
- return waitForStateTransition(StreamState::Stopping,
- StreamState::Stopped, timeoutNanoseconds);
-}
-
-int32_t AudioStream::getBytesPerSample() const {
- return convertFormatToSizeInBytes(mFormat);
-}
-
-int64_t AudioStream::getFramesRead() {
- updateFramesRead();
- return mFramesRead;
-}
-
-int64_t AudioStream::getFramesWritten() {
- updateFramesWritten();
- return mFramesWritten;
-}
-
-ResultWithValue<int32_t> AudioStream::getAvailableFrames() {
- int64_t readCounter = getFramesRead();
- if (readCounter < 0) return ResultWithValue<int32_t>::createBasedOnSign(readCounter);
- int64_t writeCounter = getFramesWritten();
- if (writeCounter < 0) return ResultWithValue<int32_t>::createBasedOnSign(writeCounter);
- int32_t framesAvailable = writeCounter - readCounter;
- return ResultWithValue<int32_t>(framesAvailable);
-}
-
-ResultWithValue<int32_t> AudioStream::waitForAvailableFrames(int32_t numFrames,
- int64_t timeoutNanoseconds) {
- if (numFrames == 0) return Result::OK;
- if (numFrames < 0) return Result::ErrorOutOfRange;
-
- // Make sure we don't try to wait for more frames than the buffer can hold.
- // Subtract framesPerBurst because this is often called from a callback
- // and we don't want to be sleeping if the buffer is close to overflowing.
- const int32_t maxAvailableFrames = getBufferCapacityInFrames() - getFramesPerBurst();
- numFrames = std::min(numFrames, maxAvailableFrames);
- // The capacity should never be less than one burst. But clip to zero just in case.
- numFrames = std::max(0, numFrames);
-
- int64_t framesAvailable = 0;
- int64_t burstInNanos = getFramesPerBurst() * kNanosPerSecond / getSampleRate();
- bool ready = false;
- int64_t deadline = AudioClock::getNanoseconds() + timeoutNanoseconds;
- do {
- ResultWithValue<int32_t> result = getAvailableFrames();
- if (!result) return result;
- framesAvailable = result.value();
- ready = (framesAvailable >= numFrames);
- if (!ready) {
- int64_t now = AudioClock::getNanoseconds();
- if (now > deadline) break;
- AudioClock::sleepForNanos(burstInNanos);
- }
- } while (!ready);
- return (!ready)
- ? ResultWithValue<int32_t>(Result::ErrorTimeout)
- : ResultWithValue<int32_t>(framesAvailable);
-}
-
-ResultWithValue<FrameTimestamp> AudioStream::getTimestamp(clockid_t clockId) {
- FrameTimestamp frame;
- Result result = getTimestamp(clockId, &frame.position, &frame.timestamp);
- if (result == Result::OK){
- return ResultWithValue<FrameTimestamp>(frame);
- } else {
- return ResultWithValue<FrameTimestamp>(static_cast<Result>(result));
- }
-}
-
-void AudioStream::calculateDefaultDelayBeforeCloseMillis() {
- // Calculate delay time before close based on burst duration.
- // Start with a burst duration then add 1 msec as a safety margin.
- mDelayBeforeCloseMillis = std::max(kMinDelayBeforeCloseMillis,
- 1 + ((mFramesPerBurst * 1000) / getSampleRate()));
- LOGD("calculateDefaultDelayBeforeCloseMillis() default = %d",
- static_cast<int>(mDelayBeforeCloseMillis));
-}
-
-} // namespace oboe