tor-browser

AudioStream.cpp (25359B)

---

/* -*- 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 "AudioStream.h"

#include <math.h>
#include <stdio.h>
#include <string.h>

#include <algorithm>

#include "AudioConverter.h"
#include "CubebUtils.h"
#include "UnderrunHandler.h"
#include "VideoUtils.h"
#include "mozilla/Logging.h"
#include "mozilla/Monitor.h"
#include "mozilla/Mutex.h"
#include "mozilla/Sprintf.h"
#include "mozilla/dom/AudioDeviceInfo.h"
#include "nsNativeCharsetUtils.h"
#include "nsPrintfCString.h"
#include "prdtoa.h"
#if defined(XP_WIN)
#  include "nsXULAppAPI.h"
#endif
#include "CallbackThreadRegistry.h"
#include "RLBoxSoundTouch.h"
#include "Tracing.h"
#include "mozilla/StaticPrefs_media.h"
#include "webaudio/blink/DenormalDisabler.h"

namespace mozilla {

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) {
   auto timestretcher = MakeUnique<RLBoxSoundTouch>();
   if (!timestretcher || !timestretcher->Init()) {
     return NS_ERROR_FAILURE;
   }
   mTimeStretcher = timestretcher.release();

   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);
 params.input_params = CUBEB_INPUT_PROCESSING_PARAM_NONE;

 // 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);
 int r = InvokeCubeb(cubeb_stream_set_name, aRawStreamName.get());
 if (r && r != CUBEB_ERROR_NOT_SUPPORTED) {
   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());

 if (!mTimeStretcher) {
   return;
 }

 // 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 (!mTimeStretcher) {
   return;
 }

 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_FRAME_COUNT("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; }

#undef LOG
#undef LOGW
#undef LOGE

}  // namespace mozilla