The Tor Browser: content/media/AudioStream.cpp@6474c204b198 (annotated)

content/media/AudioStream.cpp@6474c204b198 (annotated)

content/media/AudioStream.cpp

Wed, 31 Dec 2014 06:09:35 +0100

author: Michael Schloh von Bennewitz <michael@schloh.com>
date: Wed, 31 Dec 2014 06:09:35 +0100
changeset 0: 6474c204b198
permissions: -rw-r--r--

Cloned upstream origin tor-browser at tor-browser-31.3.0esr-4.5-1-build1
revision ID fc1c9ff7c1b2defdbc039f12214767608f46423f for hacking purpose.

 /* -*- 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 "prlog.h"
 #include "prdtoa.h"
 #include "AudioStream.h"
 #include "VideoUtils.h"
 #include "mozilla/Monitor.h"
 #include "mozilla/Mutex.h"
 #include <algorithm>
 #include "mozilla/Preferences.h"
 #include "soundtouch/SoundTouch.h"
 #include "Latency.h"
 namespace mozilla {
 #ifdef LOG
 #undef LOG
 #endif
 #ifdef PR_LOGGING
 PRLogModuleInfo* gAudioStreamLog = nullptr;
 // For simple logs
 #define LOG(x) PR_LOG(gAudioStreamLog, PR_LOG_DEBUG, x)
 #else
 #define LOG(x)
 #endif
 /**
  * When MOZ_DUMP_AUDIO is set in the environment (to anything),
  * we'll drop a series of files in the current working directory named
  * dumped-audio-<nnn>.wav, one per AudioStream created, containing
  * the audio for the stream including any skips due to underruns.
  */
 static int gDumpedAudioCount = 0;
 #define PREF_VOLUME_SCALE "media.volume_scale"
 #define PREF_CUBEB_LATENCY "media.cubeb_latency_ms"
 static const uint32_t CUBEB_NORMAL_LATENCY_MS = 100;
 StaticMutex AudioStream::sMutex;
 cubeb* AudioStream::sCubebContext;
 uint32_t AudioStream::sPreferredSampleRate;
 double AudioStream::sVolumeScale;
 uint32_t AudioStream::sCubebLatency;
 bool AudioStream::sCubebLatencyPrefSet;
 /*static*/ void AudioStream::PrefChanged(const char* aPref, void* aClosure)
 {
   if (strcmp(aPref, PREF_VOLUME_SCALE) == 0) {
     nsAdoptingString value = Preferences::GetString(aPref);
     StaticMutexAutoLock lock(sMutex);
     if (value.IsEmpty()) {
       sVolumeScale = 1.0;
     } else {
       NS_ConvertUTF16toUTF8 utf8(value);
       sVolumeScale = std::max<double>(0, PR_strtod(utf8.get(), nullptr));
     }
   } else if (strcmp(aPref, PREF_CUBEB_LATENCY) == 0) {
     // Arbitrary default stream latency of 100ms.  The higher this
     // value, the longer stream volume changes will take to become
     // audible.
     sCubebLatencyPrefSet = Preferences::HasUserValue(aPref);
     uint32_t value = Preferences::GetUint(aPref, CUBEB_NORMAL_LATENCY_MS);
     StaticMutexAutoLock lock(sMutex);
     sCubebLatency = std::min<uint32_t>(std::max<uint32_t>(value, 1), 1000);
   }
 }
 /*static*/ double AudioStream::GetVolumeScale()
 {
   StaticMutexAutoLock lock(sMutex);
   return sVolumeScale;
 }
 /*static*/ cubeb* AudioStream::GetCubebContext()
 {
   StaticMutexAutoLock lock(sMutex);
   return GetCubebContextUnlocked();
 }
 /*static*/ void AudioStream::InitPreferredSampleRate()
 {
   StaticMutexAutoLock lock(sMutex);
   if (sPreferredSampleRate == 0 &&
       cubeb_get_preferred_sample_rate(GetCubebContextUnlocked(),
                                       &sPreferredSampleRate) != CUBEB_OK) {
     sPreferredSampleRate = 44100;
   }
 }
 /*static*/ cubeb* AudioStream::GetCubebContextUnlocked()
 {
   sMutex.AssertCurrentThreadOwns();
   if (sCubebContext ||
       cubeb_init(&sCubebContext, "AudioStream") == CUBEB_OK) {
     return sCubebContext;
   }
   NS_WARNING("cubeb_init failed");
   return nullptr;
 }
 /*static*/ uint32_t AudioStream::GetCubebLatency()
 {
   StaticMutexAutoLock lock(sMutex);
   return sCubebLatency;
 }
 /*static*/ bool AudioStream::CubebLatencyPrefSet()
 {
   StaticMutexAutoLock lock(sMutex);
   return sCubebLatencyPrefSet;
 }
 #if defined(__ANDROID__) && defined(MOZ_B2G)
 static cubeb_stream_type ConvertChannelToCubebType(dom::AudioChannel aChannel)
 {
   switch(aChannel) {
     case dom::AudioChannel::Normal:
       return CUBEB_STREAM_TYPE_SYSTEM;
     case dom::AudioChannel::Content:
       return CUBEB_STREAM_TYPE_MUSIC;
     case dom::AudioChannel::Notification:
       return CUBEB_STREAM_TYPE_NOTIFICATION;
     case dom::AudioChannel::Alarm:
       return CUBEB_STREAM_TYPE_ALARM;
     case dom::AudioChannel::Telephony:
       return CUBEB_STREAM_TYPE_VOICE_CALL;
     case dom::AudioChannel::Ringer:
       return CUBEB_STREAM_TYPE_RING;
     // Currently Android openSLES library doesn't support FORCE_AUDIBLE yet.
     case dom::AudioChannel::Publicnotification:
     default:
       NS_ERROR("The value of AudioChannel is invalid");
       return CUBEB_STREAM_TYPE_MAX;
   }
 }
 #endif
 AudioStream::AudioStream()
   : mMonitor("AudioStream")
   , mInRate(0)
   , mOutRate(0)
   , mChannels(0)
   , mOutChannels(0)
   , mWritten(0)
   , mAudioClock(MOZ_THIS_IN_INITIALIZER_LIST())
   , mLatencyRequest(HighLatency)
   , mReadPoint(0)
   , mLostFrames(0)
   , mDumpFile(nullptr)
   , mVolume(1.0)
   , mBytesPerFrame(0)
   , mState(INITIALIZED)
   , mNeedsStart(false)
 {
   // keep a ref in case we shut down later than nsLayoutStatics
   mLatencyLog = AsyncLatencyLogger::Get(true);
 }
 AudioStream::~AudioStream()
 {
   LOG(("AudioStream: delete %p, state %d", this, mState));
   Shutdown();
   if (mDumpFile) {
     fclose(mDumpFile);
   }
 }
 size_t
 AudioStream::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
 {
   size_t amount = aMallocSizeOf(this);
   // Possibly add in the future:
   // - mTimeStretcher
   // - mLatencyLog
   // - mCubebStream
   amount += mInserts.SizeOfExcludingThis(aMallocSizeOf);
   amount += mBuffer.SizeOfExcludingThis(aMallocSizeOf);
   return amount;
 }
 /*static*/ void AudioStream::InitLibrary()
 {
 #ifdef PR_LOGGING
   gAudioStreamLog = PR_NewLogModule("AudioStream");
 #endif
   PrefChanged(PREF_VOLUME_SCALE, nullptr);
   Preferences::RegisterCallback(PrefChanged, PREF_VOLUME_SCALE);
   PrefChanged(PREF_CUBEB_LATENCY, nullptr);
   Preferences::RegisterCallback(PrefChanged, PREF_CUBEB_LATENCY);
 }
 /*static*/ void AudioStream::ShutdownLibrary()
 {
   Preferences::UnregisterCallback(PrefChanged, PREF_VOLUME_SCALE);
   Preferences::UnregisterCallback(PrefChanged, PREF_CUBEB_LATENCY);
   StaticMutexAutoLock lock(sMutex);
   if (sCubebContext) {
     cubeb_destroy(sCubebContext);
     sCubebContext = nullptr;
   }
 }
 nsresult AudioStream::EnsureTimeStretcherInitializedUnlocked()
 {
   mMonitor.AssertCurrentThreadOwns();
   if (!mTimeStretcher) {
     mTimeStretcher = new soundtouch::SoundTouch();
     mTimeStretcher->setSampleRate(mInRate);
     mTimeStretcher->setChannels(mOutChannels);
     mTimeStretcher->setPitch(1.0);
   }
   return NS_OK;
 }
 nsresult AudioStream::SetPlaybackRate(double aPlaybackRate)
 {
   NS_ASSERTION(aPlaybackRate > 0.0,
                "Can't handle negative or null playbackrate in the AudioStream.");
   // Avoid instantiating the resampler if we are not changing the playback rate.
   // GetPreservesPitch/SetPreservesPitch don't need locking before calling
   if (aPlaybackRate == mAudioClock.GetPlaybackRate()) {
     return NS_OK;
   }
   // MUST lock since the rate transposer is used from the cubeb callback,
   // and rate changes can cause the buffer to be reallocated
   MonitorAutoLock mon(mMonitor);
   if (EnsureTimeStretcherInitializedUnlocked() != NS_OK) {
     return NS_ERROR_FAILURE;
   }
   mAudioClock.SetPlaybackRateUnlocked(aPlaybackRate);
   mOutRate = mInRate / aPlaybackRate;
   if (mAudioClock.GetPreservesPitch()) {
     mTimeStretcher->setTempo(aPlaybackRate);
     mTimeStretcher->setRate(1.0f);
   } else {
     mTimeStretcher->setTempo(1.0f);
     mTimeStretcher->setRate(aPlaybackRate);
   }
   return NS_OK;
 }
 nsresult AudioStream::SetPreservesPitch(bool aPreservesPitch)
 {
   // Avoid instantiating the timestretcher instance if not needed.
   if (aPreservesPitch == mAudioClock.GetPreservesPitch()) {
     return NS_OK;
   }
   // MUST lock since the rate transposer is used from the cubeb callback,
   // and rate changes can cause the buffer to be reallocated
   MonitorAutoLock mon(mMonitor);
   if (EnsureTimeStretcherInitializedUnlocked() != NS_OK) {
     return NS_ERROR_FAILURE;
   }
   if (aPreservesPitch == true) {
     mTimeStretcher->setTempo(mAudioClock.GetPlaybackRate());
     mTimeStretcher->setRate(1.0f);
   } else {
     mTimeStretcher->setTempo(1.0f);
     mTimeStretcher->setRate(mAudioClock.GetPlaybackRate());
   }
   mAudioClock.SetPreservesPitch(aPreservesPitch);
   return NS_OK;
 }
 int64_t AudioStream::GetWritten()
 {
   return mWritten;
 }
 /*static*/ int AudioStream::MaxNumberOfChannels()
 {
   cubeb* cubebContext = GetCubebContext();
   uint32_t maxNumberOfChannels;
   if (cubebContext &&
       cubeb_get_max_channel_count(cubebContext,
                                   &maxNumberOfChannels) == CUBEB_OK) {
     return static_cast<int>(maxNumberOfChannels);
   }
   return 0;
 }
 /*static*/ int AudioStream::PreferredSampleRate()
 {
   MOZ_ASSERT(sPreferredSampleRate,
              "sPreferredSampleRate has not been initialized!");
   return sPreferredSampleRate;
 }
 static void SetUint16LE(uint8_t* aDest, uint16_t aValue)
 {
   aDest[0] = aValue & 0xFF;
   aDest[1] = aValue >> 8;
 }
 static void SetUint32LE(uint8_t* aDest, uint32_t aValue)
 {
   SetUint16LE(aDest, aValue & 0xFFFF);
   SetUint16LE(aDest + 2, aValue >> 16);
 }
 static FILE*
 OpenDumpFile(AudioStream* aStream)
 {
   if (!getenv("MOZ_DUMP_AUDIO"))
     return nullptr;
   char buf[100];
   sprintf(buf, "dumped-audio-%d.wav", gDumpedAudioCount);
   FILE* f = fopen(buf, "wb");
   if (!f)
     return nullptr;
   ++gDumpedAudioCount;
   uint8_t header[] = {
     // RIFF header
 x52, 0x49, 0x46, 0x46, 0x00, 0x00, 0x00, 0x00, 0x57, 0x41, 0x56, 0x45,
     // fmt chunk. We always write 16-bit samples.
 x66, 0x6d, 0x74, 0x20, 0x10, 0x00, 0x00, 0x00, 0x01, 0x00, 0xFF, 0xFF,
 xFF, 0xFF, 0xFF, 0xFF, 0x00, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x10, 0x00,
     // data chunk
 x64, 0x61, 0x74, 0x61, 0xFE, 0xFF, 0xFF, 0x7F
   };
   static const int CHANNEL_OFFSET = 22;
   static const int SAMPLE_RATE_OFFSET = 24;
   static const int BLOCK_ALIGN_OFFSET = 32;
   SetUint16LE(header + CHANNEL_OFFSET, aStream->GetChannels());
   SetUint32LE(header + SAMPLE_RATE_OFFSET, aStream->GetRate());
   SetUint16LE(header + BLOCK_ALIGN_OFFSET, aStream->GetChannels()*2);
   fwrite(header, sizeof(header), 1, f);
   return f;
 }
 static void
 WriteDumpFile(FILE* aDumpFile, AudioStream* aStream, uint32_t aFrames,
               void* aBuffer)
 {
   if (!aDumpFile)
     return;
   uint32_t samples = aStream->GetOutChannels()*aFrames;
   if (AUDIO_OUTPUT_FORMAT == AUDIO_FORMAT_S16) {
     fwrite(aBuffer, 2, samples, aDumpFile);
     return;
   }
   NS_ASSERTION(AUDIO_OUTPUT_FORMAT == AUDIO_FORMAT_FLOAT32, "bad format");
   nsAutoTArray<uint8_t, 1024*2> buf;
   buf.SetLength(samples*2);
   float* input = static_cast<float*>(aBuffer);
   uint8_t* output = buf.Elements();
   for (uint32_t i = 0; i < samples; ++i) {
     SetUint16LE(output + i*2, int16_t(input[i]*32767.0f));
   }
   fwrite(output, 2, samples, aDumpFile);
   fflush(aDumpFile);
 }
 // NOTE: this must not block a LowLatency stream for any significant amount
 // of time, or it will block the entirety of MSG
 nsresult
 AudioStream::Init(int32_t aNumChannels, int32_t aRate,
                   const dom::AudioChannel aAudioChannel,
                   LatencyRequest aLatencyRequest)
 {
   if (!GetCubebContext() || aNumChannels < 0 || aRate < 0) {
     return NS_ERROR_FAILURE;
   }
   PR_LOG(gAudioStreamLog, PR_LOG_DEBUG,
     ("%s  channels: %d, rate: %d for %p", __FUNCTION__, aNumChannels, aRate, this));
   mInRate = mOutRate = aRate;
   mChannels = aNumChannels;
   mOutChannels = (aNumChannels > 2) ? 2 : aNumChannels;
   mLatencyRequest = aLatencyRequest;
   mDumpFile = OpenDumpFile(this);
   cubeb_stream_params params;
   params.rate = aRate;
   params.channels = mOutChannels;
 #if defined(__ANDROID__)
 #if defined(MOZ_B2G)
   params.stream_type = ConvertChannelToCubebType(aAudioChannel);
 #else
   params.stream_type = CUBEB_STREAM_TYPE_MUSIC;
 #endif
   if (params.stream_type == CUBEB_STREAM_TYPE_MAX) {
     return NS_ERROR_INVALID_ARG;
   }
 #endif
   if (AUDIO_OUTPUT_FORMAT == AUDIO_FORMAT_S16) {
     params.format = CUBEB_SAMPLE_S16NE;
   } else {
     params.format = CUBEB_SAMPLE_FLOAT32NE;
   }
   mBytesPerFrame = sizeof(AudioDataValue) * mOutChannels;
   mAudioClock.Init();
   // Size mBuffer for one second of audio.  This value is arbitrary, and was
   // selected based on the observed behaviour of the existing AudioStream
   // implementations.
   uint32_t bufferLimit = FramesToBytes(aRate);
   NS_ABORT_IF_FALSE(bufferLimit % mBytesPerFrame == 0, "Must buffer complete frames");
   mBuffer.SetCapacity(bufferLimit);
   if (aLatencyRequest == LowLatency) {
     // Don't block this thread to initialize a cubeb stream.
     // When this is done, it will start callbacks from Cubeb.  Those will
     // cause us to move from INITIALIZED to RUNNING.  Until then, we
     // can't access any cubeb functions.
     // Use a RefPtr to avoid leaks if Dispatch fails
     RefPtr<AudioInitTask> init = new AudioInitTask(this, aLatencyRequest, params);
     init->Dispatch();
     return NS_OK;
   }
   // High latency - open synchronously
   nsresult rv = OpenCubeb(params, aLatencyRequest);
   // See if we need to start() the stream, since we must do that from this
   // thread for now (cubeb API issue)
   CheckForStart();
   return rv;
 }
 // This code used to live inside AudioStream::Init(), but on Mac (others?)
 // it has been known to take 300-800 (or even 8500) ms to execute(!)
 nsresult
 AudioStream::OpenCubeb(cubeb_stream_params &aParams,
                        LatencyRequest aLatencyRequest)
 {
   cubeb* cubebContext = GetCubebContext();
   if (!cubebContext) {
     MonitorAutoLock mon(mMonitor);
     mState = AudioStream::ERRORED;
     return NS_ERROR_FAILURE;
   }
   // If the latency pref is set, use it. Otherwise, if this stream is intended
   // for low latency playback, try to get the lowest latency possible.
   // Otherwise, for normal streams, use 100ms.
   uint32_t latency;
   if (aLatencyRequest == LowLatency && !CubebLatencyPrefSet()) {
     if (cubeb_get_min_latency(cubebContext, aParams, &latency) != CUBEB_OK) {
       latency = GetCubebLatency();
     }
   } else {
     latency = GetCubebLatency();
   }
   {
     cubeb_stream* stream;
     if (cubeb_stream_init(cubebContext, &stream, "AudioStream", aParams,
                           latency, DataCallback_S, StateCallback_S, this) == CUBEB_OK) {
       MonitorAutoLock mon(mMonitor);
       mCubebStream.own(stream);
       // Make sure we weren't shut down while in flight!
       if (mState == SHUTDOWN) {
         mCubebStream.reset();
         LOG(("AudioStream::OpenCubeb() %p Shutdown while opening cubeb", this));
         return NS_ERROR_FAILURE;
       }
       // We can't cubeb_stream_start() the thread from a transient thread due to
       // cubeb API requirements (init can be called from another thread, but
       // not start/stop/destroy/etc)
     } else {
       MonitorAutoLock mon(mMonitor);
       mState = ERRORED;
       LOG(("AudioStream::OpenCubeb() %p failed to init cubeb", this));
       return NS_ERROR_FAILURE;
     }
   }
   return NS_OK;
 }
 void
 AudioStream::CheckForStart()
 {
   if (mState == INITIALIZED) {
     // Start the stream right away when low latency has been requested. This means
     // that the DataCallback will feed silence to cubeb, until the first frames
     // are written to this AudioStream.  Also start if a start has been queued.
     if (mLatencyRequest == LowLatency || mNeedsStart) {
       StartUnlocked(); // mState = STARTED or ERRORED
       mNeedsStart = false;
       PR_LOG(gAudioStreamLog, PR_LOG_WARNING,
              ("Started waiting %s-latency stream",
               mLatencyRequest == LowLatency ? "low" : "high"));
     } else {
       // high latency, not full - OR Pause() was called before we got here
       PR_LOG(gAudioStreamLog, PR_LOG_DEBUG,
              ("Not starting waiting %s-latency stream",
               mLatencyRequest == LowLatency ? "low" : "high"));
     }
   }
 }
 NS_IMETHODIMP
 AudioInitTask::Run()
 {
   MOZ_ASSERT(mThread);
   if (NS_IsMainThread()) {
     mThread->Shutdown(); // can't Shutdown from the thread itself, darn
     // Don't null out mThread!
     // See bug 999104.  We must hold a ref to the thread across Dispatch()
     // since the internal mThread ref could be released while processing
     // the Dispatch(), and Dispatch/PutEvent itself doesn't hold a ref; it
     // assumes the caller does.
     return NS_OK;
   }
   nsresult rv = mAudioStream->OpenCubeb(mParams, mLatencyRequest);
   // and now kill this thread
   NS_DispatchToMainThread(this);
   return rv;
 }
 // aTime is the time in ms the samples were inserted into MediaStreamGraph
 nsresult
 AudioStream::Write(const AudioDataValue* aBuf, uint32_t aFrames, TimeStamp *aTime)
 {
   MonitorAutoLock mon(mMonitor);
   if (mState == ERRORED) {
     return NS_ERROR_FAILURE;
   }
   NS_ASSERTION(mState == INITIALIZED || mState == STARTED || mState == RUNNING,
     "Stream write in unexpected state.");
   // See if we need to start() the stream, since we must do that from this thread
   CheckForStart();
   // Downmix to Stereo.
   if (mChannels > 2 && mChannels <= 8) {
     DownmixAudioToStereo(const_cast<AudioDataValue*> (aBuf), mChannels, aFrames);
   }
   else if (mChannels > 8) {
     return NS_ERROR_FAILURE;
   }
   const uint8_t* src = reinterpret_cast<const uint8_t*>(aBuf);
   uint32_t bytesToCopy = FramesToBytes(aFrames);
   // XXX this will need to change if we want to enable this on-the-fly!
   if (PR_LOG_TEST(GetLatencyLog(), PR_LOG_DEBUG)) {
     // Record the position and time this data was inserted
     int64_t timeMs;
     if (aTime && !aTime->IsNull()) {
       if (mStartTime.IsNull()) {
         AsyncLatencyLogger::Get(true)->GetStartTime(mStartTime);
       }
       timeMs = (*aTime - mStartTime).ToMilliseconds();
     } else {
       timeMs = 0;
     }
     struct Inserts insert = { timeMs, aFrames};
     mInserts.AppendElement(insert);
   }
   while (bytesToCopy > 0) {
     uint32_t available = std::min(bytesToCopy, mBuffer.Available());
     NS_ABORT_IF_FALSE(available % mBytesPerFrame == 0,
         "Must copy complete frames.");
     mBuffer.AppendElements(src, available);
     src += available;
     bytesToCopy -= available;
     if (bytesToCopy > 0) {
       // Careful - the CubebInit thread may not have gotten to STARTED yet
       if ((mState == INITIALIZED || mState == STARTED) && mLatencyRequest == LowLatency) {
         // don't ever block MediaStreamGraph low-latency streams
         uint32_t remains = 0; // we presume the buffer is full
         if (mBuffer.Length() > bytesToCopy) {
           remains = mBuffer.Length() - bytesToCopy; // Free up just enough space
         }
         // account for dropping samples
         PR_LOG(gAudioStreamLog, PR_LOG_WARNING, ("Stream %p dropping %u bytes (%u frames)in Write()",
             this, mBuffer.Length() - remains, BytesToFrames(mBuffer.Length() - remains)));
         mReadPoint += BytesToFrames(mBuffer.Length() - remains);
         mBuffer.ContractTo(remains);
       } else { // RUNNING or high latency
         // If we are not playing, but our buffer is full, start playing to make
         // room for soon-to-be-decoded data.
         if (mState != STARTED && mState != RUNNING) {
           PR_LOG(gAudioStreamLog, PR_LOG_WARNING, ("Starting stream %p in Write (%u waiting)",
                                                  this, bytesToCopy));
           StartUnlocked();
           if (mState == ERRORED) {
             return NS_ERROR_FAILURE;
           }
         }
         PR_LOG(gAudioStreamLog, PR_LOG_WARNING, ("Stream %p waiting in Write() (%u waiting)",
                                                  this, bytesToCopy));
         mon.Wait();
       }
     }
   }
   mWritten += aFrames;
   return NS_OK;
 }
 uint32_t
 AudioStream::Available()
 {
   MonitorAutoLock mon(mMonitor);
   NS_ABORT_IF_FALSE(mBuffer.Length() % mBytesPerFrame == 0, "Buffer invariant violated.");
   return BytesToFrames(mBuffer.Available());
 }
 void
 AudioStream::SetVolume(double aVolume)
 {
   MonitorAutoLock mon(mMonitor);
   NS_ABORT_IF_FALSE(aVolume >= 0.0 && aVolume <= 1.0, "Invalid volume");
   mVolume = aVolume;
 }
 void
 AudioStream::Drain()
 {
   MonitorAutoLock mon(mMonitor);
   LOG(("AudioStream::Drain() for %p, state %d, avail %u", this, mState, mBuffer.Available()));
   if (mState != STARTED && mState != RUNNING) {
     NS_ASSERTION(mState == ERRORED || mBuffer.Available() == 0, "Draining without full buffer of unplayed audio");
     return;
   }
   mState = DRAINING;
   while (mState == DRAINING) {
     mon.Wait();
   }
 }
 void
 AudioStream::Start()
 {
   MonitorAutoLock mon(mMonitor);
   StartUnlocked();
 }
 void
 AudioStream::StartUnlocked()
 {
   mMonitor.AssertCurrentThreadOwns();
   if (!mCubebStream) {
     mNeedsStart = true;
     return;
   }
   MonitorAutoUnlock mon(mMonitor);
   if (mState == INITIALIZED) {
     int r = cubeb_stream_start(mCubebStream);
     mState = r == CUBEB_OK ? STARTED : ERRORED;
     LOG(("AudioStream: started %p, state %s", this, mState == STARTED ? "STARTED" : "ERRORED"));
   }
 }
 void
 AudioStream::Pause()
 {
   MonitorAutoLock mon(mMonitor);
   if (!mCubebStream || (mState != STARTED && mState != RUNNING)) {
     mNeedsStart = false;
     mState = STOPPED; // which also tells async OpenCubeb not to start, just init
     return;
   }
   int r;
   {
     MonitorAutoUnlock mon(mMonitor);
     r = cubeb_stream_stop(mCubebStream);
   }
   if (mState != ERRORED && r == CUBEB_OK) {
     mState = STOPPED;
   }
 }
 void
 AudioStream::Resume()
 {
   MonitorAutoLock mon(mMonitor);
   if (!mCubebStream || mState != STOPPED) {
     return;
   }
   int r;
   {
     MonitorAutoUnlock mon(mMonitor);
     r = cubeb_stream_start(mCubebStream);
   }
   if (mState != ERRORED && r == CUBEB_OK) {
     mState = STARTED;
   }
 }
 void
 AudioStream::Shutdown()
 {
   LOG(("AudioStream: Shutdown %p, state %d", this, mState));
   {
     MonitorAutoLock mon(mMonitor);
     if (mState == STARTED || mState == RUNNING) {
       MonitorAutoUnlock mon(mMonitor);
       Pause();
     }
     MOZ_ASSERT(mState != STARTED && mState != RUNNING); // paranoia
     mState = SHUTDOWN;
   }
   // Must not try to shut down cubeb from within the lock!  wasapi may still
   // call our callback after Pause()/stop()!?! Bug 996162
   if (mCubebStream) {
     mCubebStream.reset();
   }
 }
 int64_t
 AudioStream::GetPosition()
 {
   MonitorAutoLock mon(mMonitor);
   return mAudioClock.GetPositionUnlocked();
 }
 // This function is miscompiled by PGO with MSVC 2010.  See bug 768333.
 #ifdef _MSC_VER
 #pragma optimize("", off)
 #endif
 int64_t
 AudioStream::GetPositionInFrames()
 {
   return mAudioClock.GetPositionInFrames();
 }
 #ifdef _MSC_VER
 #pragma optimize("", on)
 #endif
 int64_t
 AudioStream::GetPositionInFramesInternal()
 {
   MonitorAutoLock mon(mMonitor);
   return GetPositionInFramesUnlocked();
 }
 int64_t
 AudioStream::GetPositionInFramesUnlocked()
 {
   mMonitor.AssertCurrentThreadOwns();
   if (!mCubebStream || mState == ERRORED) {
     return -1;
   }
   uint64_t position = 0;
   {
     MonitorAutoUnlock mon(mMonitor);
     if (cubeb_stream_get_position(mCubebStream, &position) != CUBEB_OK) {
       return -1;
     }
   }
   // Adjust the reported position by the number of silent frames written
   // during stream underruns.
   uint64_t adjustedPosition = 0;
   if (position >= mLostFrames) {
     adjustedPosition = position - mLostFrames;
   }
   return std::min<uint64_t>(adjustedPosition, INT64_MAX);
 }
 int64_t
 AudioStream::GetLatencyInFrames()
 {
   uint32_t latency;
   if (cubeb_stream_get_latency(mCubebStream, &latency)) {
     NS_WARNING("Could not get cubeb latency.");
     return 0;
   }
   return static_cast<int64_t>(latency);
 }
 bool
 AudioStream::IsPaused()
 {
   MonitorAutoLock mon(mMonitor);
   return mState == STOPPED;
 }
 void
 AudioStream::GetBufferInsertTime(int64_t &aTimeMs)
 {
   if (mInserts.Length() > 0) {
     // Find the right block, but don't leave the array empty
     while (mInserts.Length() > 1 && mReadPoint >= mInserts[0].mFrames) {
       mReadPoint -= mInserts[0].mFrames;
       mInserts.RemoveElementAt(0);
     }
     // offset for amount already read
     // XXX Note: could misreport if we couldn't find a block in the right timeframe
     aTimeMs = mInserts[0].mTimeMs + ((mReadPoint * 1000) / mOutRate);
   } else {
     aTimeMs = INT64_MAX;
   }
 }
 long
 AudioStream::GetUnprocessed(void* aBuffer, long aFrames, int64_t &aTimeMs)
 {
   uint8_t* wpos = reinterpret_cast<uint8_t*>(aBuffer);
   // Flush the timestretcher pipeline, if we were playing using a playback rate
   // other than 1.0.
   uint32_t flushedFrames = 0;
   if (mTimeStretcher && mTimeStretcher->numSamples()) {
     flushedFrames = mTimeStretcher->receiveSamples(reinterpret_cast<AudioDataValue*>(wpos), aFrames);
     wpos += FramesToBytes(flushedFrames);
   }
   uint32_t toPopBytes = FramesToBytes(aFrames - flushedFrames);
   uint32_t available = std::min(toPopBytes, mBuffer.Length());
   void* input[2];
   uint32_t input_size[2];
   mBuffer.PopElements(available, &input[0], &input_size[0], &input[1], &input_size[1]);
   memcpy(wpos, input[0], input_size[0]);
   wpos += input_size[0];
   memcpy(wpos, input[1], input_size[1]);
   // First time block now has our first returned sample
   mReadPoint += BytesToFrames(available);
   GetBufferInsertTime(aTimeMs);
   return BytesToFrames(available) + flushedFrames;
 }
 // Get unprocessed samples, and pad the beginning of the buffer with silence if
 // there is not enough data.
 long
 AudioStream::GetUnprocessedWithSilencePadding(void* aBuffer, long aFrames, int64_t& aTimeMs)
 {
   uint32_t toPopBytes = FramesToBytes(aFrames);
   uint32_t available = std::min(toPopBytes, mBuffer.Length());
   uint32_t silenceOffset = toPopBytes - available;
   uint8_t* wpos = reinterpret_cast<uint8_t*>(aBuffer);
   memset(wpos, 0, silenceOffset);
   wpos += silenceOffset;
   void* input[2];
   uint32_t input_size[2];
   mBuffer.PopElements(available, &input[0], &input_size[0], &input[1], &input_size[1]);
   memcpy(wpos, input[0], input_size[0]);
   wpos += input_size[0];
   memcpy(wpos, input[1], input_size[1]);
   GetBufferInsertTime(aTimeMs);
   return aFrames;
 }
 long
 AudioStream::GetTimeStretched(void* aBuffer, long aFrames, int64_t &aTimeMs)
 {
   long processedFrames = 0;
   // We need to call the non-locking version, because we already have the lock.
   if (EnsureTimeStretcherInitializedUnlocked() != NS_OK) {
     return 0;
   }
   uint8_t* wpos = reinterpret_cast<uint8_t*>(aBuffer);
   double playbackRate = static_cast<double>(mInRate) / mOutRate;
   uint32_t toPopBytes = FramesToBytes(ceil(aFrames / playbackRate));
   uint32_t available = 0;
   bool lowOnBufferedData = false;
   do {
     // Check if we already have enough data in the time stretcher pipeline.
     if (mTimeStretcher->numSamples() <= static_cast<uint32_t>(aFrames)) {
       void* input[2];
       uint32_t input_size[2];
       available = std::min(mBuffer.Length(), toPopBytes);
       if (available != toPopBytes) {
         lowOnBufferedData = true;
       }
       mBuffer.PopElements(available, &input[0], &input_size[0],
                                      &input[1], &input_size[1]);
       mReadPoint += BytesToFrames(available);
       for(uint32_t i = 0; i < 2; i++) {
         mTimeStretcher->putSamples(reinterpret_cast<AudioDataValue*>(input[i]), BytesToFrames(input_size[i]));
       }
     }
     uint32_t receivedFrames = mTimeStretcher->receiveSamples(reinterpret_cast<AudioDataValue*>(wpos), aFrames - processedFrames);
     wpos += FramesToBytes(receivedFrames);
     processedFrames += receivedFrames;
   } while (processedFrames < aFrames && !lowOnBufferedData);
   GetBufferInsertTime(aTimeMs);
   return processedFrames;
 }
 long
 AudioStream::DataCallback(void* aBuffer, long aFrames)
 {
   MonitorAutoLock mon(mMonitor);
   uint32_t available = std::min(static_cast<uint32_t>(FramesToBytes(aFrames)), mBuffer.Length());
   NS_ABORT_IF_FALSE(available % mBytesPerFrame == 0, "Must copy complete frames");
   AudioDataValue* output = reinterpret_cast<AudioDataValue*>(aBuffer);
   uint32_t underrunFrames = 0;
   uint32_t servicedFrames = 0;
   int64_t insertTime;
   // NOTE: wasapi (others?) can call us back *after* stop()/Shutdown() (mState == SHUTDOWN)
   // Bug 996162
   // callback tells us cubeb succeeded initializing
   if (mState == STARTED) {
     // For low-latency streams, we want to minimize any built-up data when
     // we start getting callbacks.
     // Simple version - contract on first callback only.
     if (mLatencyRequest == LowLatency) {
 #ifdef PR_LOGGING
       uint32_t old_len = mBuffer.Length();
 #endif
       available = mBuffer.ContractTo(FramesToBytes(aFrames));
 #ifdef PR_LOGGING
       TimeStamp now = TimeStamp::Now();
       if (!mStartTime.IsNull()) {
         int64_t timeMs = (now - mStartTime).ToMilliseconds();
         PR_LOG(gAudioStreamLog, PR_LOG_WARNING,
                ("Stream took %lldms to start after first Write() @ %u", timeMs, mOutRate));
       } else {
         PR_LOG(gAudioStreamLog, PR_LOG_WARNING,
           ("Stream started before Write() @ %u", mOutRate));
       }
       if (old_len != available) {
         // Note that we may have dropped samples in Write() as well!
         PR_LOG(gAudioStreamLog, PR_LOG_WARNING,
                ("AudioStream %p dropped %u + %u initial frames @ %u", this,
                  mReadPoint, BytesToFrames(old_len - available), mOutRate));
         mReadPoint += BytesToFrames(old_len - available);
       }
 #endif
     }
     mState = RUNNING;
   }
   if (available) {
     // When we are playing a low latency stream, and it is the first time we are
     // getting data from the buffer, we prefer to add the silence for an
     // underrun at the beginning of the buffer, so the first buffer is not cut
     // in half by the silence inserted to compensate for the underrun.
     if (mInRate == mOutRate) {
       if (mLatencyRequest == LowLatency && !mWritten) {
         servicedFrames = GetUnprocessedWithSilencePadding(output, aFrames, insertTime);
       } else {
         servicedFrames = GetUnprocessed(output, aFrames, insertTime);
       }
     } else {
       servicedFrames = GetTimeStretched(output, aFrames, insertTime);
     }
     float scaled_volume = float(GetVolumeScale() * mVolume);
     ScaleAudioSamples(output, aFrames * mOutChannels, scaled_volume);
     NS_ABORT_IF_FALSE(mBuffer.Length() % mBytesPerFrame == 0, "Must copy complete frames");
     // Notify any blocked Write() call that more space is available in mBuffer.
     mon.NotifyAll();
   } else {
     GetBufferInsertTime(insertTime);
   }
   underrunFrames = aFrames - servicedFrames;
   if (mState != DRAINING) {
     uint8_t* rpos = static_cast<uint8_t*>(aBuffer) + FramesToBytes(aFrames - underrunFrames);
     memset(rpos, 0, FramesToBytes(underrunFrames));
     if (underrunFrames) {
       PR_LOG(gAudioStreamLog, PR_LOG_WARNING,
              ("AudioStream %p lost %d frames", this, underrunFrames));
     }
     mLostFrames += underrunFrames;
     servicedFrames += underrunFrames;
   }
   WriteDumpFile(mDumpFile, this, aFrames, aBuffer);
   // Don't log if we're not interested or if the stream is inactive
   if (PR_LOG_TEST(GetLatencyLog(), PR_LOG_DEBUG) &&
       mState != SHUTDOWN &&
       insertTime != INT64_MAX && servicedFrames > underrunFrames) {
     uint32_t latency = UINT32_MAX;
     if (cubeb_stream_get_latency(mCubebStream, &latency)) {
       NS_WARNING("Could not get latency from cubeb.");
     }
     TimeStamp now = TimeStamp::Now();
     mLatencyLog->Log(AsyncLatencyLogger::AudioStream, reinterpret_cast<uint64_t>(this),
                      insertTime, now);
     mLatencyLog->Log(AsyncLatencyLogger::Cubeb, reinterpret_cast<uint64_t>(mCubebStream.get()),
                      (latency * 1000) / mOutRate, now);
   }
   mAudioClock.UpdateWritePosition(servicedFrames);
   return servicedFrames;
 }
 void
 AudioStream::StateCallback(cubeb_state aState)
 {
   MonitorAutoLock mon(mMonitor);
   if (aState == CUBEB_STATE_DRAINED) {
     mState = DRAINED;
   } else if (aState == CUBEB_STATE_ERROR) {
     LOG(("AudioStream::StateCallback() state %d cubeb error", mState));
     mState = ERRORED;
   }
   mon.NotifyAll();
 }
 AudioClock::AudioClock(AudioStream* aStream)
  :mAudioStream(aStream),
   mOldOutRate(0),
   mBasePosition(0),
   mBaseOffset(0),
   mOldBaseOffset(0),
   mOldBasePosition(0),
   mPlaybackRateChangeOffset(0),
   mPreviousPosition(0),
   mWritten(0),
   mOutRate(0),
   mInRate(0),
   mPreservesPitch(true),
   mCompensatingLatency(false)
 {}
 void AudioClock::Init()
 {
   mOutRate = mAudioStream->GetRate();
   mInRate = mAudioStream->GetRate();
   mOldOutRate = mOutRate;
 }
 void AudioClock::UpdateWritePosition(uint32_t aCount)
 {
   mWritten += aCount;
 }
 uint64_t AudioClock::GetPositionUnlocked()
 {
   // GetPositionInFramesUnlocked() asserts it owns the monitor
   int64_t position = mAudioStream->GetPositionInFramesUnlocked();
   int64_t diffOffset;
   NS_ASSERTION(position < 0 || (mInRate != 0 && mOutRate != 0), "AudioClock not initialized.");
   if (position >= 0) {
     if (position < mPlaybackRateChangeOffset) {
       // See if we are still playing frames pushed with the old playback rate in
       // the backend. If we are, use the old output rate to compute the
       // position.
       mCompensatingLatency = true;
       diffOffset = position - mOldBaseOffset;
       position = static_cast<uint64_t>(mOldBasePosition +
         static_cast<float>(USECS_PER_S * diffOffset) / mOldOutRate);
       mPreviousPosition = position;
       return position;
     }
     if (mCompensatingLatency) {
       diffOffset = position - mPlaybackRateChangeOffset;
       mCompensatingLatency = false;
       mBasePosition = mPreviousPosition;
     } else {
       diffOffset = position - mPlaybackRateChangeOffset;
     }
     position =  static_cast<uint64_t>(mBasePosition +
       (static_cast<float>(USECS_PER_S * diffOffset) / mOutRate));
     return position;
   }
   return UINT64_MAX;
 }
 uint64_t AudioClock::GetPositionInFrames()
 {
   return (GetPositionUnlocked() * mOutRate) / USECS_PER_S;
 }
 void AudioClock::SetPlaybackRateUnlocked(double aPlaybackRate)
 {
   // GetPositionInFramesUnlocked() asserts it owns the monitor
   int64_t position = mAudioStream->GetPositionInFramesUnlocked();
   if (position > mPlaybackRateChangeOffset) {
     mOldBasePosition = mBasePosition;
     mBasePosition = GetPositionUnlocked();
     mOldBaseOffset = mPlaybackRateChangeOffset;
     mBaseOffset = position;
     mPlaybackRateChangeOffset = mWritten;
     mOldOutRate = mOutRate;
     mOutRate = static_cast<int>(mInRate / aPlaybackRate);
   } else {
     // The playbackRate has been changed before the end of the latency
     // compensation phase. We don't update the mOld* variable. That way, the
     // last playbackRate set is taken into account.
     mBasePosition = GetPositionUnlocked();
     mBaseOffset = position;
     mPlaybackRateChangeOffset = mWritten;
     mOutRate = static_cast<int>(mInRate / aPlaybackRate);
   }
 }
 double AudioClock::GetPlaybackRate()
 {
   return static_cast<double>(mInRate) / mOutRate;
 }
 void AudioClock::SetPreservesPitch(bool aPreservesPitch)
 {
   mPreservesPitch = aPreservesPitch;
 }
 bool AudioClock::GetPreservesPitch()
 {
   return mPreservesPitch;
 }
 } // namespace mozilla

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content/media/AudioStream.cpp@6474c204b198 (annotated)

content/media/AudioStream.cpp