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-rw-r--r--dom/media/AudioStream.cpp756
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diff --git a/dom/media/AudioStream.cpp b/dom/media/AudioStream.cpp
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+++ b/dom/media/AudioStream.cpp
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+/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
+/* vim:set ts=2 sw=2 sts=2 et cindent: */
+/* This Source Code Form is subject to the terms of the Mozilla Public
+ * License, v. 2.0. If a copy of the MPL was not distributed with this
+ * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
+#include <stdio.h>
+#include <math.h>
+#include <string.h>
+#include "mozilla/Logging.h"
+#include "prdtoa.h"
+#include "AudioStream.h"
+#include "VideoUtils.h"
+#include "mozilla/dom/AudioDeviceInfo.h"
+#include "mozilla/Monitor.h"
+#include "mozilla/Mutex.h"
+#include "mozilla/Sprintf.h"
+#include "mozilla/Unused.h"
+#include <algorithm>
+#include "mozilla/Telemetry.h"
+#include "CubebUtils.h"
+#include "nsNativeCharsetUtils.h"
+#include "nsPrintfCString.h"
+#include "AudioConverter.h"
+#include "UnderrunHandler.h"
+#if defined(XP_WIN)
+# include "nsXULAppAPI.h"
+#endif
+#include "Tracing.h"
+#include "webaudio/blink/DenormalDisabler.h"
+#include "CallbackThreadRegistry.h"
+#include "mozilla/StaticPrefs_media.h"
+
+#include "RLBoxSoundTouch.h"
+
+namespace mozilla {
+
+#undef LOG
+#undef LOGW
+#undef LOGE
+
+LazyLogModule gAudioStreamLog("AudioStream");
+// For simple logs
+#define LOG(x, ...) \
+ MOZ_LOG(gAudioStreamLog, mozilla::LogLevel::Debug, \
+ ("%p " x, this, ##__VA_ARGS__))
+#define LOGW(x, ...) \
+ MOZ_LOG(gAudioStreamLog, mozilla::LogLevel::Warning, \
+ ("%p " x, this, ##__VA_ARGS__))
+#define LOGE(x, ...) \
+ NS_DebugBreak(NS_DEBUG_WARNING, \
+ nsPrintfCString("%p " x, this, ##__VA_ARGS__).get(), nullptr, \
+ __FILE__, __LINE__)
+
+/**
+ * Keep a list of frames sent to the audio engine in each DataCallback along
+ * with the playback rate at the moment. Since the playback rate and number of
+ * underrun frames can vary in each callback. We need to keep the whole history
+ * in order to calculate the playback position of the audio engine correctly.
+ */
+class FrameHistory {
+ struct Chunk {
+ uint32_t servicedFrames;
+ uint32_t totalFrames;
+ uint32_t rate;
+ };
+
+ template <typename T>
+ static T FramesToUs(uint32_t frames, uint32_t rate) {
+ return static_cast<T>(frames) * USECS_PER_S / rate;
+ }
+
+ public:
+ FrameHistory() : mBaseOffset(0), mBasePosition(0) {}
+
+ void Append(uint32_t aServiced, uint32_t aUnderrun, uint32_t aRate) {
+ /* In most case where playback rate stays the same and we don't underrun
+ * frames, we are able to merge chunks to avoid lose of precision to add up
+ * in compressing chunks into |mBaseOffset| and |mBasePosition|.
+ */
+ if (!mChunks.IsEmpty()) {
+ Chunk& c = mChunks.LastElement();
+ // 2 chunks (c1 and c2) can be merged when rate is the same and
+ // adjacent frames are zero. That is, underrun frames in c1 are zero
+ // or serviced frames in c2 are zero.
+ if (c.rate == aRate &&
+ (c.servicedFrames == c.totalFrames || aServiced == 0)) {
+ c.servicedFrames += aServiced;
+ c.totalFrames += aServiced + aUnderrun;
+ return;
+ }
+ }
+ Chunk* p = mChunks.AppendElement();
+ p->servicedFrames = aServiced;
+ p->totalFrames = aServiced + aUnderrun;
+ p->rate = aRate;
+ }
+
+ /**
+ * @param frames The playback position in frames of the audio engine.
+ * @return The playback position in microseconds of the audio engine,
+ * adjusted by playback rate changes and underrun frames.
+ */
+ int64_t GetPosition(int64_t frames) {
+ // playback position should not go backward.
+ MOZ_ASSERT(frames >= mBaseOffset);
+ while (true) {
+ if (mChunks.IsEmpty()) {
+ return static_cast<int64_t>(mBasePosition);
+ }
+ const Chunk& c = mChunks[0];
+ if (frames <= mBaseOffset + c.totalFrames) {
+ uint32_t delta = frames - mBaseOffset;
+ delta = std::min(delta, c.servicedFrames);
+ return static_cast<int64_t>(mBasePosition) +
+ FramesToUs<int64_t>(delta, c.rate);
+ }
+ // Since the playback position of the audio engine will not go backward,
+ // we are able to compress chunks so that |mChunks| won't grow
+ // unlimitedly. Note that we lose precision in converting integers into
+ // floats and inaccuracy will accumulate over time. However, for a 24hr
+ // long, sample rate = 44.1k file, the error will be less than 1
+ // microsecond after playing 24 hours. So we are fine with that.
+ mBaseOffset += c.totalFrames;
+ mBasePosition += FramesToUs<double>(c.servicedFrames, c.rate);
+ mChunks.RemoveElementAt(0);
+ }
+ }
+
+ private:
+ AutoTArray<Chunk, 7> mChunks;
+ int64_t mBaseOffset;
+ double mBasePosition;
+};
+
+AudioStream::AudioStream(DataSource& aSource, uint32_t aInRate,
+ uint32_t aOutputChannels,
+ AudioConfig::ChannelLayout::ChannelMap aChannelMap)
+ : mTimeStretcher(nullptr),
+ mAudioClock(aInRate),
+ mChannelMap(aChannelMap),
+ mMonitor("AudioStream"),
+ mOutChannels(aOutputChannels),
+ mState(INITIALIZED),
+ mDataSource(aSource),
+ mAudioThreadId(ProfilerThreadId{}),
+ mSandboxed(CubebUtils::SandboxEnabled()),
+ mPlaybackComplete(false),
+ mPlaybackRate(1.0f),
+ mPreservesPitch(true),
+ mCallbacksStarted(false) {}
+
+AudioStream::~AudioStream() {
+ LOG("deleted, state %d", mState.load());
+ MOZ_ASSERT(mState == SHUTDOWN && !mCubebStream,
+ "Should've called ShutDown() before deleting an AudioStream");
+}
+
+size_t AudioStream::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const {
+ size_t amount = aMallocSizeOf(this);
+
+ // Possibly add in the future:
+ // - mTimeStretcher
+ // - mCubebStream
+
+ return amount;
+}
+
+nsresult AudioStream::EnsureTimeStretcherInitialized() {
+ AssertIsOnAudioThread();
+ if (!mTimeStretcher) {
+ mTimeStretcher = new RLBoxSoundTouch();
+ mTimeStretcher->setSampleRate(mAudioClock.GetInputRate());
+ mTimeStretcher->setChannels(mOutChannels);
+ mTimeStretcher->setPitch(1.0);
+
+ // SoundTouch v2.1.2 uses automatic time-stretch settings with the following
+ // values:
+ // Tempo 0.5: 90ms sequence, 20ms seekwindow, 8ms overlap
+ // Tempo 2.0: 40ms sequence, 15ms seekwindow, 8ms overlap
+ // We are going to use a smaller 10ms sequence size to improve speech
+ // clarity, giving more resolution at high tempo and less reverb at low
+ // tempo. Maintain 15ms seekwindow and 8ms overlap for smoothness.
+ mTimeStretcher->setSetting(
+ SETTING_SEQUENCE_MS,
+ StaticPrefs::media_audio_playbackrate_soundtouch_sequence_ms());
+ mTimeStretcher->setSetting(
+ SETTING_SEEKWINDOW_MS,
+ StaticPrefs::media_audio_playbackrate_soundtouch_seekwindow_ms());
+ mTimeStretcher->setSetting(
+ SETTING_OVERLAP_MS,
+ StaticPrefs::media_audio_playbackrate_soundtouch_overlap_ms());
+ }
+ return NS_OK;
+}
+
+nsresult AudioStream::SetPlaybackRate(double aPlaybackRate) {
+ TRACE_COMMENT("AudioStream::SetPlaybackRate", "%f", aPlaybackRate);
+ NS_ASSERTION(
+ aPlaybackRate > 0.0,
+ "Can't handle negative or null playbackrate in the AudioStream.");
+ if (aPlaybackRate == mPlaybackRate) {
+ return NS_OK;
+ }
+
+ mPlaybackRate = static_cast<float>(aPlaybackRate);
+
+ return NS_OK;
+}
+
+nsresult AudioStream::SetPreservesPitch(bool aPreservesPitch) {
+ TRACE_COMMENT("AudioStream::SetPreservesPitch", "%d", aPreservesPitch);
+ if (aPreservesPitch == mPreservesPitch) {
+ return NS_OK;
+ }
+
+ mPreservesPitch = aPreservesPitch;
+
+ return NS_OK;
+}
+
+template <typename Function, typename... Args>
+int AudioStream::InvokeCubeb(Function aFunction, Args&&... aArgs) {
+ mMonitor.AssertCurrentThreadOwns();
+ MonitorAutoUnlock mon(mMonitor);
+ return aFunction(mCubebStream.get(), std::forward<Args>(aArgs)...);
+}
+
+nsresult AudioStream::Init(AudioDeviceInfo* aSinkInfo)
+ MOZ_NO_THREAD_SAFETY_ANALYSIS {
+ auto startTime = TimeStamp::Now();
+ TRACE("AudioStream::Init");
+
+ LOG("%s channels: %d, rate: %d", __FUNCTION__, mOutChannels,
+ mAudioClock.GetInputRate());
+
+ mSinkInfo = aSinkInfo;
+
+ cubeb_stream_params params;
+ params.rate = mAudioClock.GetInputRate();
+ params.channels = mOutChannels;
+ params.layout = static_cast<uint32_t>(mChannelMap);
+ params.format = CubebUtils::ToCubebFormat<AUDIO_OUTPUT_FORMAT>::value;
+ params.prefs = CubebUtils::GetDefaultStreamPrefs(CUBEB_DEVICE_TYPE_OUTPUT);
+
+ // This is noop if MOZ_DUMP_AUDIO is not set.
+ mDumpFile.Open("AudioStream", mOutChannels, mAudioClock.GetInputRate());
+
+ RefPtr<CubebUtils::CubebHandle> handle = CubebUtils::GetCubeb();
+ if (!handle) {
+ LOGE("Can't get cubeb context!");
+ CubebUtils::ReportCubebStreamInitFailure(true);
+ return NS_ERROR_DOM_MEDIA_CUBEB_INITIALIZATION_ERR;
+ }
+
+ mCubeb = handle;
+ return OpenCubeb(handle->Context(), params, startTime,
+ CubebUtils::GetFirstStream());
+}
+
+nsresult AudioStream::OpenCubeb(cubeb* aContext, cubeb_stream_params& aParams,
+ TimeStamp aStartTime, bool aIsFirst) {
+ TRACE("AudioStream::OpenCubeb");
+ MOZ_ASSERT(aContext);
+
+ cubeb_stream* stream = nullptr;
+ /* Convert from milliseconds to frames. */
+ uint32_t latency_frames =
+ CubebUtils::GetCubebPlaybackLatencyInMilliseconds() * aParams.rate / 1000;
+ cubeb_devid deviceID = nullptr;
+ if (mSinkInfo && mSinkInfo->DeviceID()) {
+ deviceID = mSinkInfo->DeviceID();
+ }
+ if (CubebUtils::CubebStreamInit(aContext, &stream, "AudioStream", nullptr,
+ nullptr, deviceID, &aParams, latency_frames,
+ DataCallback_S, StateCallback_S,
+ this) == CUBEB_OK) {
+ mCubebStream.reset(stream);
+ CubebUtils::ReportCubebBackendUsed();
+ } else {
+ LOGE("OpenCubeb() failed to init cubeb");
+ CubebUtils::ReportCubebStreamInitFailure(aIsFirst);
+ return NS_ERROR_FAILURE;
+ }
+
+ TimeDuration timeDelta = TimeStamp::Now() - aStartTime;
+ LOG("creation time %sfirst: %u ms", aIsFirst ? "" : "not ",
+ (uint32_t)timeDelta.ToMilliseconds());
+
+ return NS_OK;
+}
+
+void AudioStream::SetVolume(double aVolume) {
+ TRACE_COMMENT("AudioStream::SetVolume", "%f", aVolume);
+ MOZ_ASSERT(aVolume >= 0.0 && aVolume <= 1.0, "Invalid volume");
+
+ MOZ_ASSERT(mState != SHUTDOWN, "Don't set volume after shutdown.");
+ if (mState == ERRORED) {
+ return;
+ }
+
+ MonitorAutoLock mon(mMonitor);
+ if (InvokeCubeb(cubeb_stream_set_volume,
+ aVolume * CubebUtils::GetVolumeScale()) != CUBEB_OK) {
+ LOGE("Could not change volume on cubeb stream.");
+ }
+}
+
+void AudioStream::SetStreamName(const nsAString& aStreamName) {
+ TRACE("AudioStream::SetStreamName");
+
+ nsAutoCString aRawStreamName;
+ nsresult rv = NS_CopyUnicodeToNative(aStreamName, aRawStreamName);
+
+ if (NS_FAILED(rv) || aStreamName.IsEmpty()) {
+ return;
+ }
+
+ MonitorAutoLock mon(mMonitor);
+ if (InvokeCubeb(cubeb_stream_set_name, aRawStreamName.get()) != CUBEB_OK) {
+ LOGE("Could not set cubeb stream name.");
+ }
+}
+
+RefPtr<MediaSink::EndedPromise> AudioStream::Start() {
+ TRACE("AudioStream::Start");
+ MOZ_ASSERT(mState == INITIALIZED);
+ mState = STARTED;
+ RefPtr<MediaSink::EndedPromise> promise;
+ {
+ MonitorAutoLock mon(mMonitor);
+ // As cubeb might call audio stream's state callback very soon after we
+ // start cubeb, we have to create the promise beforehand in order to handle
+ // the case where we immediately get `drained`.
+ promise = mEndedPromise.Ensure(__func__);
+ mPlaybackComplete = false;
+
+ if (InvokeCubeb(cubeb_stream_start) != CUBEB_OK) {
+ mState = ERRORED;
+ mEndedPromise.RejectIfExists(NS_ERROR_FAILURE, __func__);
+ }
+
+ LOG("started, state %s", mState == STARTED ? "STARTED"
+ : mState == DRAINED ? "DRAINED"
+ : "ERRORED");
+ }
+ return promise;
+}
+
+void AudioStream::Pause() {
+ TRACE("AudioStream::Pause");
+ MOZ_ASSERT(mState != INITIALIZED, "Must be Start()ed.");
+ MOZ_ASSERT(mState != STOPPED, "Already Pause()ed.");
+ MOZ_ASSERT(mState != SHUTDOWN, "Already ShutDown()ed.");
+
+ // Do nothing if we are already drained or errored.
+ if (mState == DRAINED || mState == ERRORED) {
+ return;
+ }
+
+ MonitorAutoLock mon(mMonitor);
+ if (InvokeCubeb(cubeb_stream_stop) != CUBEB_OK) {
+ mState = ERRORED;
+ } else if (mState != DRAINED && mState != ERRORED) {
+ // Don't transition to other states if we are already
+ // drained or errored.
+ mState = STOPPED;
+ }
+}
+
+void AudioStream::Resume() {
+ TRACE("AudioStream::Resume");
+ MOZ_ASSERT(mState != INITIALIZED, "Must be Start()ed.");
+ MOZ_ASSERT(mState != STARTED, "Already Start()ed.");
+ MOZ_ASSERT(mState != SHUTDOWN, "Already ShutDown()ed.");
+
+ // Do nothing if we are already drained or errored.
+ if (mState == DRAINED || mState == ERRORED) {
+ return;
+ }
+
+ MonitorAutoLock mon(mMonitor);
+ if (InvokeCubeb(cubeb_stream_start) != CUBEB_OK) {
+ mState = ERRORED;
+ } else if (mState != DRAINED && mState != ERRORED) {
+ // Don't transition to other states if we are already
+ // drained or errored.
+ mState = STARTED;
+ }
+}
+
+void AudioStream::ShutDown() {
+ TRACE("AudioStream::ShutDown");
+ LOG("ShutDown, state %d", mState.load());
+
+ MonitorAutoLock mon(mMonitor);
+ if (mCubebStream) {
+ // Force stop to put the cubeb stream in a stable state before deletion.
+ InvokeCubeb(cubeb_stream_stop);
+ // Must not try to shut down cubeb from within the lock! wasapi may still
+ // call our callback after Pause()/stop()!?! Bug 996162
+ cubeb_stream* cubeb = mCubebStream.release();
+ MonitorAutoUnlock unlock(mMonitor);
+ cubeb_stream_destroy(cubeb);
+ }
+
+ // After `cubeb_stream_stop` has been called, there is no audio thread
+ // anymore. We can delete the time stretcher.
+ if (mTimeStretcher) {
+ delete mTimeStretcher;
+ mTimeStretcher = nullptr;
+ }
+
+ mState = SHUTDOWN;
+ mEndedPromise.ResolveIfExists(true, __func__);
+}
+
+int64_t AudioStream::GetPosition() {
+ TRACE("AudioStream::GetPosition");
+#ifndef XP_MACOSX
+ MonitorAutoLock mon(mMonitor);
+#endif
+ int64_t frames = GetPositionInFramesUnlocked();
+ return frames >= 0 ? mAudioClock.GetPosition(frames) : -1;
+}
+
+int64_t AudioStream::GetPositionInFrames() {
+ TRACE("AudioStream::GetPositionInFrames");
+#ifndef XP_MACOSX
+ MonitorAutoLock mon(mMonitor);
+#endif
+ int64_t frames = GetPositionInFramesUnlocked();
+
+ return frames >= 0 ? mAudioClock.GetPositionInFrames(frames) : -1;
+}
+
+int64_t AudioStream::GetPositionInFramesUnlocked() {
+ TRACE("AudioStream::GetPositionInFramesUnlocked");
+#ifndef XP_MACOSX
+ mMonitor.AssertCurrentThreadOwns();
+#endif
+
+ if (mState == ERRORED) {
+ return -1;
+ }
+
+ uint64_t position = 0;
+ int rv;
+
+#ifndef XP_MACOSX
+ rv = InvokeCubeb(cubeb_stream_get_position, &position);
+#else
+ rv = cubeb_stream_get_position(mCubebStream.get(), &position);
+#endif
+
+ if (rv != CUBEB_OK) {
+ return -1;
+ }
+ return static_cast<int64_t>(std::min<uint64_t>(position, INT64_MAX));
+}
+
+bool AudioStream::IsValidAudioFormat(Chunk* aChunk) {
+ if (aChunk->Rate() != mAudioClock.GetInputRate()) {
+ LOGW("mismatched sample %u, mInRate=%u", aChunk->Rate(),
+ mAudioClock.GetInputRate());
+ return false;
+ }
+
+ return aChunk->Channels() <= 8;
+}
+
+void AudioStream::GetUnprocessed(AudioBufferWriter& aWriter) {
+ TRACE("AudioStream::GetUnprocessed");
+ AssertIsOnAudioThread();
+ // Flush the timestretcher pipeline, if we were playing using a playback rate
+ // other than 1.0.
+ if (mTimeStretcher) {
+ // Get number of samples and based on this either receive samples or write
+ // silence. At worst, the attacker can supply weird sound samples or
+ // result in us writing silence.
+ auto numSamples = mTimeStretcher->numSamples().unverified_safe_because(
+ "We only use this to decide whether to receive samples or write "
+ "silence.");
+ if (numSamples) {
+ RLBoxSoundTouch* timeStretcher = mTimeStretcher;
+ aWriter.Write(
+ [timeStretcher](AudioDataValue* aPtr, uint32_t aFrames) {
+ return timeStretcher->receiveSamples(aPtr, aFrames);
+ },
+ aWriter.Available());
+
+ // TODO: There might be still unprocessed samples in the stretcher.
+ // We should either remove or flush them so they won't be in the output
+ // next time we switch a playback rate other than 1.0.
+ mTimeStretcher->numUnprocessedSamples().copy_and_verify([](auto samples) {
+ NS_WARNING_ASSERTION(samples == 0, "no samples");
+ });
+ } else {
+ // Don't need it anymore: playbackRate is 1.0, and the time stretcher has
+ // been flushed.
+ delete mTimeStretcher;
+ mTimeStretcher = nullptr;
+ }
+ }
+
+ while (aWriter.Available() > 0) {
+ uint32_t count = mDataSource.PopFrames(aWriter.Ptr(), aWriter.Available(),
+ mAudioThreadChanged);
+ if (count == 0) {
+ break;
+ }
+ aWriter.Advance(count);
+ }
+}
+
+void AudioStream::GetTimeStretched(AudioBufferWriter& aWriter) {
+ TRACE("AudioStream::GetTimeStretched");
+ AssertIsOnAudioThread();
+ if (EnsureTimeStretcherInitialized() != NS_OK) {
+ return;
+ }
+
+ uint32_t toPopFrames =
+ ceil(aWriter.Available() * mAudioClock.GetPlaybackRate());
+
+ // At each iteration, get number of samples and (based on this) write from
+ // the data source or silence. At worst, if the number of samples is a lie
+ // (i.e., under attacker control) we'll either not write anything or keep
+ // writing noise. This is safe because all the memory operations within the
+ // loop (and after) are checked.
+ while (mTimeStretcher->numSamples().unverified_safe_because(
+ "Only used to decide whether to put samples.") <
+ aWriter.Available()) {
+ // pop into a temp buffer, and put into the stretcher.
+ AutoTArray<AudioDataValue, 1000> buf;
+ auto size = CheckedUint32(mOutChannels) * toPopFrames;
+ if (!size.isValid()) {
+ // The overflow should not happen in normal case.
+ LOGW("Invalid member data: %d channels, %d frames", mOutChannels,
+ toPopFrames);
+ return;
+ }
+ buf.SetLength(size.value());
+ // ensure no variable channel count or something like that
+ uint32_t count =
+ mDataSource.PopFrames(buf.Elements(), toPopFrames, mAudioThreadChanged);
+ if (count == 0) {
+ break;
+ }
+ mTimeStretcher->putSamples(buf.Elements(), count);
+ }
+
+ auto* timeStretcher = mTimeStretcher;
+ aWriter.Write(
+ [timeStretcher](AudioDataValue* aPtr, uint32_t aFrames) {
+ return timeStretcher->receiveSamples(aPtr, aFrames);
+ },
+ aWriter.Available());
+}
+
+bool AudioStream::CheckThreadIdChanged() {
+ ProfilerThreadId id = profiler_current_thread_id();
+ if (id != mAudioThreadId) {
+ mAudioThreadId = id;
+ mAudioThreadChanged = true;
+ return true;
+ }
+ mAudioThreadChanged = false;
+ return false;
+}
+
+void AudioStream::AssertIsOnAudioThread() const {
+ // This can be called right after CheckThreadIdChanged, because the audio
+ // thread can change when not sandboxed.
+ MOZ_ASSERT(mAudioThreadId.load() == profiler_current_thread_id());
+}
+
+void AudioStream::UpdatePlaybackRateIfNeeded() {
+ AssertIsOnAudioThread();
+ if (mAudioClock.GetPreservesPitch() == mPreservesPitch &&
+ mAudioClock.GetPlaybackRate() == mPlaybackRate) {
+ return;
+ }
+
+ EnsureTimeStretcherInitialized();
+
+ mAudioClock.SetPlaybackRate(mPlaybackRate);
+ mAudioClock.SetPreservesPitch(mPreservesPitch);
+
+ if (mPreservesPitch) {
+ mTimeStretcher->setTempo(mPlaybackRate);
+ mTimeStretcher->setRate(1.0f);
+ } else {
+ mTimeStretcher->setTempo(1.0f);
+ mTimeStretcher->setRate(mPlaybackRate);
+ }
+}
+
+long AudioStream::DataCallback(void* aBuffer, long aFrames) {
+ if (CheckThreadIdChanged() && !mSandboxed) {
+ CallbackThreadRegistry::Get()->Register(mAudioThreadId,
+ "NativeAudioCallback");
+ }
+ WebCore::DenormalDisabler disabler;
+ if (!mCallbacksStarted) {
+ mCallbacksStarted = true;
+ }
+
+ TRACE_AUDIO_CALLBACK_BUDGET("AudioStream real-time budget", aFrames,
+ mAudioClock.GetInputRate());
+ TRACE("AudioStream::DataCallback");
+ MOZ_ASSERT(mState != SHUTDOWN, "No data callback after shutdown");
+
+ if (SoftRealTimeLimitReached()) {
+ DemoteThreadFromRealTime();
+ }
+
+ UpdatePlaybackRateIfNeeded();
+
+ auto writer = AudioBufferWriter(
+ Span<AudioDataValue>(reinterpret_cast<AudioDataValue*>(aBuffer),
+ mOutChannels * aFrames),
+ mOutChannels, aFrames);
+
+ if (mAudioClock.GetInputRate() == mAudioClock.GetOutputRate()) {
+ GetUnprocessed(writer);
+ } else {
+ GetTimeStretched(writer);
+ }
+
+ // Always send audible frames first, and silent frames later.
+ // Otherwise it will break the assumption of FrameHistory.
+ if (!mDataSource.Ended()) {
+#ifndef XP_MACOSX
+ MonitorAutoLock mon(mMonitor);
+#endif
+ mAudioClock.UpdateFrameHistory(aFrames - writer.Available(),
+ writer.Available(), mAudioThreadChanged);
+ if (writer.Available() > 0) {
+ TRACE_COMMENT("AudioStream::DataCallback", "Underrun: %d frames missing",
+ writer.Available());
+ LOGW("lost %d frames", writer.Available());
+ writer.WriteZeros(writer.Available());
+ }
+ } else {
+ // No more new data in the data source, and the drain has completed. We
+ // don't need the time stretcher anymore at this point.
+ if (mTimeStretcher && writer.Available()) {
+ delete mTimeStretcher;
+ mTimeStretcher = nullptr;
+ }
+#ifndef XP_MACOSX
+ MonitorAutoLock mon(mMonitor);
+#endif
+ mAudioClock.UpdateFrameHistory(aFrames - writer.Available(), 0,
+ mAudioThreadChanged);
+ }
+
+ mDumpFile.Write(static_cast<const AudioDataValue*>(aBuffer),
+ aFrames * mOutChannels);
+
+ if (!mSandboxed && writer.Available() != 0) {
+ CallbackThreadRegistry::Get()->Unregister(mAudioThreadId);
+ }
+ return aFrames - writer.Available();
+}
+
+void AudioStream::StateCallback(cubeb_state aState) {
+ MOZ_ASSERT(mState != SHUTDOWN, "No state callback after shutdown");
+ LOG("StateCallback, mState=%d cubeb_state=%d", mState.load(), aState);
+
+ MonitorAutoLock mon(mMonitor);
+ if (aState == CUBEB_STATE_DRAINED) {
+ LOG("Drained");
+ mState = DRAINED;
+ mPlaybackComplete = true;
+ mEndedPromise.ResolveIfExists(true, __func__);
+ } else if (aState == CUBEB_STATE_ERROR) {
+ LOGE("StateCallback() state %d cubeb error", mState.load());
+ mState = ERRORED;
+ mPlaybackComplete = true;
+ mEndedPromise.RejectIfExists(NS_ERROR_FAILURE, __func__);
+ }
+}
+
+bool AudioStream::IsPlaybackCompleted() const { return mPlaybackComplete; }
+
+AudioClock::AudioClock(uint32_t aInRate)
+ : mOutRate(aInRate),
+ mInRate(aInRate),
+ mPreservesPitch(true),
+ mFrameHistory(new FrameHistory()) {}
+
+// Audio thread only
+void AudioClock::UpdateFrameHistory(uint32_t aServiced, uint32_t aUnderrun,
+ bool aAudioThreadChanged) {
+#ifdef XP_MACOSX
+ if (aAudioThreadChanged) {
+ mCallbackInfoQueue.ResetProducerThreadId();
+ }
+ // Flush the local items, if any, and then attempt to enqueue the current
+ // item. This is only a fallback mechanism, under non-critical load this is
+ // just going to enqueue an item in the queue.
+ while (!mAudioThreadCallbackInfo.IsEmpty()) {
+ CallbackInfo& info = mAudioThreadCallbackInfo[0];
+ // If still full, keep it audio-thread side for now.
+ if (mCallbackInfoQueue.Enqueue(info) != 1) {
+ break;
+ }
+ mAudioThreadCallbackInfo.RemoveElementAt(0);
+ }
+ CallbackInfo info(aServiced, aUnderrun, mOutRate);
+ if (mCallbackInfoQueue.Enqueue(info) != 1) {
+ NS_WARNING(
+ "mCallbackInfoQueue full, storing the values in the audio thread.");
+ mAudioThreadCallbackInfo.AppendElement(info);
+ }
+#else
+ MutexAutoLock lock(mMutex);
+ mFrameHistory->Append(aServiced, aUnderrun, mOutRate);
+#endif
+}
+
+int64_t AudioClock::GetPositionInFrames(int64_t aFrames) {
+ CheckedInt64 v = UsecsToFrames(GetPosition(aFrames), mInRate);
+ return v.isValid() ? v.value() : -1;
+}
+
+int64_t AudioClock::GetPosition(int64_t frames) {
+#ifdef XP_MACOSX
+ // Dequeue all history info, and apply them before returning the position
+ // based on frame history.
+ CallbackInfo info;
+ while (mCallbackInfoQueue.Dequeue(&info, 1)) {
+ mFrameHistory->Append(info.mServiced, info.mUnderrun, info.mOutputRate);
+ }
+#else
+ MutexAutoLock lock(mMutex);
+#endif
+ return mFrameHistory->GetPosition(frames);
+}
+
+void AudioClock::SetPlaybackRate(double aPlaybackRate) {
+ mOutRate = static_cast<uint32_t>(mInRate / aPlaybackRate);
+}
+
+double AudioClock::GetPlaybackRate() const {
+ return static_cast<double>(mInRate) / mOutRate;
+}
+
+void AudioClock::SetPreservesPitch(bool aPreservesPitch) {
+ mPreservesPitch = aPreservesPitch;
+}
+
+bool AudioClock::GetPreservesPitch() const { return mPreservesPitch; }
+
+} // namespace mozilla