michael@0: /* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
michael@0: /* vim:set ts=2 sw=2 sts=2 et cindent: */
michael@0: /* This Source Code Form is subject to the terms of the Mozilla Public
michael@0:  * License, v. 2.0. If a copy of the MPL was not distributed with this
michael@0:  * file, You can obtain one at http://mozilla.org/MPL/2.0/. */
michael@0: #include <stdio.h>
michael@0: #include <math.h>
michael@0: #include "prlog.h"
michael@0: #include "prdtoa.h"
michael@0: #include "AudioStream.h"
michael@0: #include "VideoUtils.h"
michael@0: #include "mozilla/Monitor.h"
michael@0: #include "mozilla/Mutex.h"
michael@0: #include <algorithm>
michael@0: #include "mozilla/Preferences.h"
michael@0: #include "soundtouch/SoundTouch.h"
michael@0: #include "Latency.h"
michael@0: 
michael@0: namespace mozilla {
michael@0: 
michael@0: #ifdef LOG
michael@0: #undef LOG
michael@0: #endif
michael@0: 
michael@0: #ifdef PR_LOGGING
michael@0: PRLogModuleInfo* gAudioStreamLog = nullptr;
michael@0: // For simple logs
michael@0: #define LOG(x) PR_LOG(gAudioStreamLog, PR_LOG_DEBUG, x)
michael@0: #else
michael@0: #define LOG(x)
michael@0: #endif
michael@0: 
michael@0: /**
michael@0:  * When MOZ_DUMP_AUDIO is set in the environment (to anything),
michael@0:  * we'll drop a series of files in the current working directory named
michael@0:  * dumped-audio-<nnn>.wav, one per AudioStream created, containing
michael@0:  * the audio for the stream including any skips due to underruns.
michael@0:  */
michael@0: static int gDumpedAudioCount = 0;
michael@0: 
michael@0: #define PREF_VOLUME_SCALE "media.volume_scale"
michael@0: #define PREF_CUBEB_LATENCY "media.cubeb_latency_ms"
michael@0: 
michael@0: static const uint32_t CUBEB_NORMAL_LATENCY_MS = 100;
michael@0: 
michael@0: StaticMutex AudioStream::sMutex;
michael@0: cubeb* AudioStream::sCubebContext;
michael@0: uint32_t AudioStream::sPreferredSampleRate;
michael@0: double AudioStream::sVolumeScale;
michael@0: uint32_t AudioStream::sCubebLatency;
michael@0: bool AudioStream::sCubebLatencyPrefSet;
michael@0: 
michael@0: /*static*/ void AudioStream::PrefChanged(const char* aPref, void* aClosure)
michael@0: {
michael@0:   if (strcmp(aPref, PREF_VOLUME_SCALE) == 0) {
michael@0:     nsAdoptingString value = Preferences::GetString(aPref);
michael@0:     StaticMutexAutoLock lock(sMutex);
michael@0:     if (value.IsEmpty()) {
michael@0:       sVolumeScale = 1.0;
michael@0:     } else {
michael@0:       NS_ConvertUTF16toUTF8 utf8(value);
michael@0:       sVolumeScale = std::max<double>(0, PR_strtod(utf8.get(), nullptr));
michael@0:     }
michael@0:   } else if (strcmp(aPref, PREF_CUBEB_LATENCY) == 0) {
michael@0:     // Arbitrary default stream latency of 100ms.  The higher this
michael@0:     // value, the longer stream volume changes will take to become
michael@0:     // audible.
michael@0:     sCubebLatencyPrefSet = Preferences::HasUserValue(aPref);
michael@0:     uint32_t value = Preferences::GetUint(aPref, CUBEB_NORMAL_LATENCY_MS);
michael@0:     StaticMutexAutoLock lock(sMutex);
michael@0:     sCubebLatency = std::min<uint32_t>(std::max<uint32_t>(value, 1), 1000);
michael@0:   }
michael@0: }
michael@0: 
michael@0: /*static*/ double AudioStream::GetVolumeScale()
michael@0: {
michael@0:   StaticMutexAutoLock lock(sMutex);
michael@0:   return sVolumeScale;
michael@0: }
michael@0: 
michael@0: /*static*/ cubeb* AudioStream::GetCubebContext()
michael@0: {
michael@0:   StaticMutexAutoLock lock(sMutex);
michael@0:   return GetCubebContextUnlocked();
michael@0: }
michael@0: 
michael@0: /*static*/ void AudioStream::InitPreferredSampleRate()
michael@0: {
michael@0:   StaticMutexAutoLock lock(sMutex);
michael@0:   if (sPreferredSampleRate == 0 &&
michael@0:       cubeb_get_preferred_sample_rate(GetCubebContextUnlocked(),
michael@0:                                       &sPreferredSampleRate) != CUBEB_OK) {
michael@0:     sPreferredSampleRate = 44100;
michael@0:   }
michael@0: }
michael@0: 
michael@0: /*static*/ cubeb* AudioStream::GetCubebContextUnlocked()
michael@0: {
michael@0:   sMutex.AssertCurrentThreadOwns();
michael@0:   if (sCubebContext ||
michael@0:       cubeb_init(&sCubebContext, "AudioStream") == CUBEB_OK) {
michael@0:     return sCubebContext;
michael@0:   }
michael@0:   NS_WARNING("cubeb_init failed");
michael@0:   return nullptr;
michael@0: }
michael@0: 
michael@0: /*static*/ uint32_t AudioStream::GetCubebLatency()
michael@0: {
michael@0:   StaticMutexAutoLock lock(sMutex);
michael@0:   return sCubebLatency;
michael@0: }
michael@0: 
michael@0: /*static*/ bool AudioStream::CubebLatencyPrefSet()
michael@0: {
michael@0:   StaticMutexAutoLock lock(sMutex);
michael@0:   return sCubebLatencyPrefSet;
michael@0: }
michael@0: 
michael@0: #if defined(__ANDROID__) && defined(MOZ_B2G)
michael@0: static cubeb_stream_type ConvertChannelToCubebType(dom::AudioChannel aChannel)
michael@0: {
michael@0:   switch(aChannel) {
michael@0:     case dom::AudioChannel::Normal:
michael@0:       return CUBEB_STREAM_TYPE_SYSTEM;
michael@0:     case dom::AudioChannel::Content:
michael@0:       return CUBEB_STREAM_TYPE_MUSIC;
michael@0:     case dom::AudioChannel::Notification:
michael@0:       return CUBEB_STREAM_TYPE_NOTIFICATION;
michael@0:     case dom::AudioChannel::Alarm:
michael@0:       return CUBEB_STREAM_TYPE_ALARM;
michael@0:     case dom::AudioChannel::Telephony:
michael@0:       return CUBEB_STREAM_TYPE_VOICE_CALL;
michael@0:     case dom::AudioChannel::Ringer:
michael@0:       return CUBEB_STREAM_TYPE_RING;
michael@0:     // Currently Android openSLES library doesn't support FORCE_AUDIBLE yet.
michael@0:     case dom::AudioChannel::Publicnotification:
michael@0:     default:
michael@0:       NS_ERROR("The value of AudioChannel is invalid");
michael@0:       return CUBEB_STREAM_TYPE_MAX;
michael@0:   }
michael@0: }
michael@0: #endif
michael@0: 
michael@0: AudioStream::AudioStream()
michael@0:   : mMonitor("AudioStream")
michael@0:   , mInRate(0)
michael@0:   , mOutRate(0)
michael@0:   , mChannels(0)
michael@0:   , mOutChannels(0)
michael@0:   , mWritten(0)
michael@0:   , mAudioClock(MOZ_THIS_IN_INITIALIZER_LIST())
michael@0:   , mLatencyRequest(HighLatency)
michael@0:   , mReadPoint(0)
michael@0:   , mLostFrames(0)
michael@0:   , mDumpFile(nullptr)
michael@0:   , mVolume(1.0)
michael@0:   , mBytesPerFrame(0)
michael@0:   , mState(INITIALIZED)
michael@0:   , mNeedsStart(false)
michael@0: {
michael@0:   // keep a ref in case we shut down later than nsLayoutStatics
michael@0:   mLatencyLog = AsyncLatencyLogger::Get(true);
michael@0: }
michael@0: 
michael@0: AudioStream::~AudioStream()
michael@0: {
michael@0:   LOG(("AudioStream: delete %p, state %d", this, mState));
michael@0:   Shutdown();
michael@0:   if (mDumpFile) {
michael@0:     fclose(mDumpFile);
michael@0:   }
michael@0: }
michael@0: 
michael@0: size_t
michael@0: AudioStream::SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const
michael@0: {
michael@0:   size_t amount = aMallocSizeOf(this);
michael@0: 
michael@0:   // Possibly add in the future:
michael@0:   // - mTimeStretcher
michael@0:   // - mLatencyLog
michael@0:   // - mCubebStream
michael@0: 
michael@0:   amount += mInserts.SizeOfExcludingThis(aMallocSizeOf);
michael@0:   amount += mBuffer.SizeOfExcludingThis(aMallocSizeOf);
michael@0: 
michael@0:   return amount;
michael@0: }
michael@0: 
michael@0: /*static*/ void AudioStream::InitLibrary()
michael@0: {
michael@0: #ifdef PR_LOGGING
michael@0:   gAudioStreamLog = PR_NewLogModule("AudioStream");
michael@0: #endif
michael@0:   PrefChanged(PREF_VOLUME_SCALE, nullptr);
michael@0:   Preferences::RegisterCallback(PrefChanged, PREF_VOLUME_SCALE);
michael@0:   PrefChanged(PREF_CUBEB_LATENCY, nullptr);
michael@0:   Preferences::RegisterCallback(PrefChanged, PREF_CUBEB_LATENCY);
michael@0: }
michael@0: 
michael@0: /*static*/ void AudioStream::ShutdownLibrary()
michael@0: {
michael@0:   Preferences::UnregisterCallback(PrefChanged, PREF_VOLUME_SCALE);
michael@0:   Preferences::UnregisterCallback(PrefChanged, PREF_CUBEB_LATENCY);
michael@0: 
michael@0:   StaticMutexAutoLock lock(sMutex);
michael@0:   if (sCubebContext) {
michael@0:     cubeb_destroy(sCubebContext);
michael@0:     sCubebContext = nullptr;
michael@0:   }
michael@0: }
michael@0: 
michael@0: nsresult AudioStream::EnsureTimeStretcherInitializedUnlocked()
michael@0: {
michael@0:   mMonitor.AssertCurrentThreadOwns();
michael@0:   if (!mTimeStretcher) {
michael@0:     mTimeStretcher = new soundtouch::SoundTouch();
michael@0:     mTimeStretcher->setSampleRate(mInRate);
michael@0:     mTimeStretcher->setChannels(mOutChannels);
michael@0:     mTimeStretcher->setPitch(1.0);
michael@0:   }
michael@0:   return NS_OK;
michael@0: }
michael@0: 
michael@0: nsresult AudioStream::SetPlaybackRate(double aPlaybackRate)
michael@0: {
michael@0:   NS_ASSERTION(aPlaybackRate > 0.0,
michael@0:                "Can't handle negative or null playbackrate in the AudioStream.");
michael@0:   // Avoid instantiating the resampler if we are not changing the playback rate.
michael@0:   // GetPreservesPitch/SetPreservesPitch don't need locking before calling
michael@0:   if (aPlaybackRate == mAudioClock.GetPlaybackRate()) {
michael@0:     return NS_OK;
michael@0:   }
michael@0: 
michael@0:   // MUST lock since the rate transposer is used from the cubeb callback,
michael@0:   // and rate changes can cause the buffer to be reallocated
michael@0:   MonitorAutoLock mon(mMonitor);
michael@0:   if (EnsureTimeStretcherInitializedUnlocked() != NS_OK) {
michael@0:     return NS_ERROR_FAILURE;
michael@0:   }
michael@0: 
michael@0:   mAudioClock.SetPlaybackRateUnlocked(aPlaybackRate);
michael@0:   mOutRate = mInRate / aPlaybackRate;
michael@0: 
michael@0:   if (mAudioClock.GetPreservesPitch()) {
michael@0:     mTimeStretcher->setTempo(aPlaybackRate);
michael@0:     mTimeStretcher->setRate(1.0f);
michael@0:   } else {
michael@0:     mTimeStretcher->setTempo(1.0f);
michael@0:     mTimeStretcher->setRate(aPlaybackRate);
michael@0:   }
michael@0:   return NS_OK;
michael@0: }
michael@0: 
michael@0: nsresult AudioStream::SetPreservesPitch(bool aPreservesPitch)
michael@0: {
michael@0:   // Avoid instantiating the timestretcher instance if not needed.
michael@0:   if (aPreservesPitch == mAudioClock.GetPreservesPitch()) {
michael@0:     return NS_OK;
michael@0:   }
michael@0: 
michael@0:   // MUST lock since the rate transposer is used from the cubeb callback,
michael@0:   // and rate changes can cause the buffer to be reallocated
michael@0:   MonitorAutoLock mon(mMonitor);
michael@0:   if (EnsureTimeStretcherInitializedUnlocked() != NS_OK) {
michael@0:     return NS_ERROR_FAILURE;
michael@0:   }
michael@0: 
michael@0:   if (aPreservesPitch == true) {
michael@0:     mTimeStretcher->setTempo(mAudioClock.GetPlaybackRate());
michael@0:     mTimeStretcher->setRate(1.0f);
michael@0:   } else {
michael@0:     mTimeStretcher->setTempo(1.0f);
michael@0:     mTimeStretcher->setRate(mAudioClock.GetPlaybackRate());
michael@0:   }
michael@0: 
michael@0:   mAudioClock.SetPreservesPitch(aPreservesPitch);
michael@0: 
michael@0:   return NS_OK;
michael@0: }
michael@0: 
michael@0: int64_t AudioStream::GetWritten()
michael@0: {
michael@0:   return mWritten;
michael@0: }
michael@0: 
michael@0: /*static*/ int AudioStream::MaxNumberOfChannels()
michael@0: {
michael@0:   cubeb* cubebContext = GetCubebContext();
michael@0:   uint32_t maxNumberOfChannels;
michael@0:   if (cubebContext &&
michael@0:       cubeb_get_max_channel_count(cubebContext,
michael@0:                                   &maxNumberOfChannels) == CUBEB_OK) {
michael@0:     return static_cast<int>(maxNumberOfChannels);
michael@0:   }
michael@0: 
michael@0:   return 0;
michael@0: }
michael@0: 
michael@0: /*static*/ int AudioStream::PreferredSampleRate()
michael@0: {
michael@0:   MOZ_ASSERT(sPreferredSampleRate,
michael@0:              "sPreferredSampleRate has not been initialized!");
michael@0:   return sPreferredSampleRate;
michael@0: }
michael@0: 
michael@0: static void SetUint16LE(uint8_t* aDest, uint16_t aValue)
michael@0: {
michael@0:   aDest[0] = aValue & 0xFF;
michael@0:   aDest[1] = aValue >> 8;
michael@0: }
michael@0: 
michael@0: static void SetUint32LE(uint8_t* aDest, uint32_t aValue)
michael@0: {
michael@0:   SetUint16LE(aDest, aValue & 0xFFFF);
michael@0:   SetUint16LE(aDest + 2, aValue >> 16);
michael@0: }
michael@0: 
michael@0: static FILE*
michael@0: OpenDumpFile(AudioStream* aStream)
michael@0: {
michael@0:   if (!getenv("MOZ_DUMP_AUDIO"))
michael@0:     return nullptr;
michael@0:   char buf[100];
michael@0:   sprintf(buf, "dumped-audio-%d.wav", gDumpedAudioCount);
michael@0:   FILE* f = fopen(buf, "wb");
michael@0:   if (!f)
michael@0:     return nullptr;
michael@0:   ++gDumpedAudioCount;
michael@0: 
michael@0:   uint8_t header[] = {
michael@0:     // RIFF header
michael@0:     0x52, 0x49, 0x46, 0x46, 0x00, 0x00, 0x00, 0x00, 0x57, 0x41, 0x56, 0x45,
michael@0:     // fmt chunk. We always write 16-bit samples.
michael@0:     0x66, 0x6d, 0x74, 0x20, 0x10, 0x00, 0x00, 0x00, 0x01, 0x00, 0xFF, 0xFF,
michael@0:     0xFF, 0xFF, 0xFF, 0xFF, 0x00, 0x00, 0x00, 0x00, 0xFF, 0xFF, 0x10, 0x00,
michael@0:     // data chunk
michael@0:     0x64, 0x61, 0x74, 0x61, 0xFE, 0xFF, 0xFF, 0x7F
michael@0:   };
michael@0:   static const int CHANNEL_OFFSET = 22;
michael@0:   static const int SAMPLE_RATE_OFFSET = 24;
michael@0:   static const int BLOCK_ALIGN_OFFSET = 32;
michael@0:   SetUint16LE(header + CHANNEL_OFFSET, aStream->GetChannels());
michael@0:   SetUint32LE(header + SAMPLE_RATE_OFFSET, aStream->GetRate());
michael@0:   SetUint16LE(header + BLOCK_ALIGN_OFFSET, aStream->GetChannels()*2);
michael@0:   fwrite(header, sizeof(header), 1, f);
michael@0: 
michael@0:   return f;
michael@0: }
michael@0: 
michael@0: static void
michael@0: WriteDumpFile(FILE* aDumpFile, AudioStream* aStream, uint32_t aFrames,
michael@0:               void* aBuffer)
michael@0: {
michael@0:   if (!aDumpFile)
michael@0:     return;
michael@0: 
michael@0:   uint32_t samples = aStream->GetOutChannels()*aFrames;
michael@0:   if (AUDIO_OUTPUT_FORMAT == AUDIO_FORMAT_S16) {
michael@0:     fwrite(aBuffer, 2, samples, aDumpFile);
michael@0:     return;
michael@0:   }
michael@0: 
michael@0:   NS_ASSERTION(AUDIO_OUTPUT_FORMAT == AUDIO_FORMAT_FLOAT32, "bad format");
michael@0:   nsAutoTArray<uint8_t, 1024*2> buf;
michael@0:   buf.SetLength(samples*2);
michael@0:   float* input = static_cast<float*>(aBuffer);
michael@0:   uint8_t* output = buf.Elements();
michael@0:   for (uint32_t i = 0; i < samples; ++i) {
michael@0:     SetUint16LE(output + i*2, int16_t(input[i]*32767.0f));
michael@0:   }
michael@0:   fwrite(output, 2, samples, aDumpFile);
michael@0:   fflush(aDumpFile);
michael@0: }
michael@0: 
michael@0: // NOTE: this must not block a LowLatency stream for any significant amount
michael@0: // of time, or it will block the entirety of MSG
michael@0: nsresult
michael@0: AudioStream::Init(int32_t aNumChannels, int32_t aRate,
michael@0:                   const dom::AudioChannel aAudioChannel,
michael@0:                   LatencyRequest aLatencyRequest)
michael@0: {
michael@0:   if (!GetCubebContext() || aNumChannels < 0 || aRate < 0) {
michael@0:     return NS_ERROR_FAILURE;
michael@0:   }
michael@0: 
michael@0:   PR_LOG(gAudioStreamLog, PR_LOG_DEBUG,
michael@0:     ("%s  channels: %d, rate: %d for %p", __FUNCTION__, aNumChannels, aRate, this));
michael@0:   mInRate = mOutRate = aRate;
michael@0:   mChannels = aNumChannels;
michael@0:   mOutChannels = (aNumChannels > 2) ? 2 : aNumChannels;
michael@0:   mLatencyRequest = aLatencyRequest;
michael@0: 
michael@0:   mDumpFile = OpenDumpFile(this);
michael@0: 
michael@0:   cubeb_stream_params params;
michael@0:   params.rate = aRate;
michael@0:   params.channels = mOutChannels;
michael@0: #if defined(__ANDROID__)
michael@0: #if defined(MOZ_B2G)
michael@0:   params.stream_type = ConvertChannelToCubebType(aAudioChannel);
michael@0: #else
michael@0:   params.stream_type = CUBEB_STREAM_TYPE_MUSIC;
michael@0: #endif
michael@0: 
michael@0:   if (params.stream_type == CUBEB_STREAM_TYPE_MAX) {
michael@0:     return NS_ERROR_INVALID_ARG;
michael@0:   }
michael@0: #endif
michael@0:   if (AUDIO_OUTPUT_FORMAT == AUDIO_FORMAT_S16) {
michael@0:     params.format = CUBEB_SAMPLE_S16NE;
michael@0:   } else {
michael@0:     params.format = CUBEB_SAMPLE_FLOAT32NE;
michael@0:   }
michael@0:   mBytesPerFrame = sizeof(AudioDataValue) * mOutChannels;
michael@0: 
michael@0:   mAudioClock.Init();
michael@0: 
michael@0:   // Size mBuffer for one second of audio.  This value is arbitrary, and was
michael@0:   // selected based on the observed behaviour of the existing AudioStream
michael@0:   // implementations.
michael@0:   uint32_t bufferLimit = FramesToBytes(aRate);
michael@0:   NS_ABORT_IF_FALSE(bufferLimit % mBytesPerFrame == 0, "Must buffer complete frames");
michael@0:   mBuffer.SetCapacity(bufferLimit);
michael@0: 
michael@0:   if (aLatencyRequest == LowLatency) {
michael@0:     // Don't block this thread to initialize a cubeb stream.
michael@0:     // When this is done, it will start callbacks from Cubeb.  Those will
michael@0:     // cause us to move from INITIALIZED to RUNNING.  Until then, we
michael@0:     // can't access any cubeb functions.
michael@0:     // Use a RefPtr to avoid leaks if Dispatch fails
michael@0:     RefPtr<AudioInitTask> init = new AudioInitTask(this, aLatencyRequest, params);
michael@0:     init->Dispatch();
michael@0:     return NS_OK;
michael@0:   }
michael@0:   // High latency - open synchronously
michael@0:   nsresult rv = OpenCubeb(params, aLatencyRequest);
michael@0:   // See if we need to start() the stream, since we must do that from this
michael@0:   // thread for now (cubeb API issue)
michael@0:   CheckForStart();
michael@0:   return rv;
michael@0: }
michael@0: 
michael@0: // This code used to live inside AudioStream::Init(), but on Mac (others?)
michael@0: // it has been known to take 300-800 (or even 8500) ms to execute(!)
michael@0: nsresult
michael@0: AudioStream::OpenCubeb(cubeb_stream_params &aParams,
michael@0:                        LatencyRequest aLatencyRequest)
michael@0: {
michael@0:   cubeb* cubebContext = GetCubebContext();
michael@0:   if (!cubebContext) {
michael@0:     MonitorAutoLock mon(mMonitor);
michael@0:     mState = AudioStream::ERRORED;
michael@0:     return NS_ERROR_FAILURE;
michael@0:   }
michael@0: 
michael@0:   // If the latency pref is set, use it. Otherwise, if this stream is intended
michael@0:   // for low latency playback, try to get the lowest latency possible.
michael@0:   // Otherwise, for normal streams, use 100ms.
michael@0:   uint32_t latency;
michael@0:   if (aLatencyRequest == LowLatency && !CubebLatencyPrefSet()) {
michael@0:     if (cubeb_get_min_latency(cubebContext, aParams, &latency) != CUBEB_OK) {
michael@0:       latency = GetCubebLatency();
michael@0:     }
michael@0:   } else {
michael@0:     latency = GetCubebLatency();
michael@0:   }
michael@0: 
michael@0:   {
michael@0:     cubeb_stream* stream;
michael@0:     if (cubeb_stream_init(cubebContext, &stream, "AudioStream", aParams,
michael@0:                           latency, DataCallback_S, StateCallback_S, this) == CUBEB_OK) {
michael@0:       MonitorAutoLock mon(mMonitor);
michael@0:       mCubebStream.own(stream);
michael@0:       // Make sure we weren't shut down while in flight!
michael@0:       if (mState == SHUTDOWN) {
michael@0:         mCubebStream.reset();
michael@0:         LOG(("AudioStream::OpenCubeb() %p Shutdown while opening cubeb", this));
michael@0:         return NS_ERROR_FAILURE;
michael@0:       }
michael@0: 
michael@0:       // We can't cubeb_stream_start() the thread from a transient thread due to
michael@0:       // cubeb API requirements (init can be called from another thread, but
michael@0:       // not start/stop/destroy/etc)
michael@0:     } else {
michael@0:       MonitorAutoLock mon(mMonitor);
michael@0:       mState = ERRORED;
michael@0:       LOG(("AudioStream::OpenCubeb() %p failed to init cubeb", this));
michael@0:       return NS_ERROR_FAILURE;
michael@0:     }
michael@0:   }
michael@0: 
michael@0:   return NS_OK;
michael@0: }
michael@0: 
michael@0: void
michael@0: AudioStream::CheckForStart()
michael@0: {
michael@0:   if (mState == INITIALIZED) {
michael@0:     // Start the stream right away when low latency has been requested. This means
michael@0:     // that the DataCallback will feed silence to cubeb, until the first frames
michael@0:     // are written to this AudioStream.  Also start if a start has been queued.
michael@0:     if (mLatencyRequest == LowLatency || mNeedsStart) {
michael@0:       StartUnlocked(); // mState = STARTED or ERRORED
michael@0:       mNeedsStart = false;
michael@0:       PR_LOG(gAudioStreamLog, PR_LOG_WARNING,
michael@0:              ("Started waiting %s-latency stream",
michael@0:               mLatencyRequest == LowLatency ? "low" : "high"));
michael@0:     } else {
michael@0:       // high latency, not full - OR Pause() was called before we got here
michael@0:       PR_LOG(gAudioStreamLog, PR_LOG_DEBUG,
michael@0:              ("Not starting waiting %s-latency stream",
michael@0:               mLatencyRequest == LowLatency ? "low" : "high"));
michael@0:     }
michael@0:   }
michael@0: }
michael@0: 
michael@0: NS_IMETHODIMP
michael@0: AudioInitTask::Run()
michael@0: {
michael@0:   MOZ_ASSERT(mThread);
michael@0:   if (NS_IsMainThread()) {
michael@0:     mThread->Shutdown(); // can't Shutdown from the thread itself, darn
michael@0:     // Don't null out mThread!
michael@0:     // See bug 999104.  We must hold a ref to the thread across Dispatch()
michael@0:     // since the internal mThread ref could be released while processing
michael@0:     // the Dispatch(), and Dispatch/PutEvent itself doesn't hold a ref; it
michael@0:     // assumes the caller does.
michael@0:     return NS_OK;
michael@0:   }
michael@0: 
michael@0:   nsresult rv = mAudioStream->OpenCubeb(mParams, mLatencyRequest);
michael@0: 
michael@0:   // and now kill this thread
michael@0:   NS_DispatchToMainThread(this);
michael@0:   return rv;
michael@0: }
michael@0: 
michael@0: // aTime is the time in ms the samples were inserted into MediaStreamGraph
michael@0: nsresult
michael@0: AudioStream::Write(const AudioDataValue* aBuf, uint32_t aFrames, TimeStamp *aTime)
michael@0: {
michael@0:   MonitorAutoLock mon(mMonitor);
michael@0:   if (mState == ERRORED) {
michael@0:     return NS_ERROR_FAILURE;
michael@0:   }
michael@0:   NS_ASSERTION(mState == INITIALIZED || mState == STARTED || mState == RUNNING,
michael@0:     "Stream write in unexpected state.");
michael@0: 
michael@0:   // See if we need to start() the stream, since we must do that from this thread
michael@0:   CheckForStart();
michael@0: 
michael@0:   // Downmix to Stereo.
michael@0:   if (mChannels > 2 && mChannels <= 8) {
michael@0:     DownmixAudioToStereo(const_cast<AudioDataValue*> (aBuf), mChannels, aFrames);
michael@0:   }
michael@0:   else if (mChannels > 8) {
michael@0:     return NS_ERROR_FAILURE;
michael@0:   }
michael@0: 
michael@0:   const uint8_t* src = reinterpret_cast<const uint8_t*>(aBuf);
michael@0:   uint32_t bytesToCopy = FramesToBytes(aFrames);
michael@0: 
michael@0:   // XXX this will need to change if we want to enable this on-the-fly!
michael@0:   if (PR_LOG_TEST(GetLatencyLog(), PR_LOG_DEBUG)) {
michael@0:     // Record the position and time this data was inserted
michael@0:     int64_t timeMs;
michael@0:     if (aTime && !aTime->IsNull()) {
michael@0:       if (mStartTime.IsNull()) {
michael@0:         AsyncLatencyLogger::Get(true)->GetStartTime(mStartTime);
michael@0:       }
michael@0:       timeMs = (*aTime - mStartTime).ToMilliseconds();
michael@0:     } else {
michael@0:       timeMs = 0;
michael@0:     }
michael@0:     struct Inserts insert = { timeMs, aFrames};
michael@0:     mInserts.AppendElement(insert);
michael@0:   }
michael@0: 
michael@0:   while (bytesToCopy > 0) {
michael@0:     uint32_t available = std::min(bytesToCopy, mBuffer.Available());
michael@0:     NS_ABORT_IF_FALSE(available % mBytesPerFrame == 0,
michael@0:         "Must copy complete frames.");
michael@0: 
michael@0:     mBuffer.AppendElements(src, available);
michael@0:     src += available;
michael@0:     bytesToCopy -= available;
michael@0: 
michael@0:     if (bytesToCopy > 0) {
michael@0:       // Careful - the CubebInit thread may not have gotten to STARTED yet
michael@0:       if ((mState == INITIALIZED || mState == STARTED) && mLatencyRequest == LowLatency) {
michael@0:         // don't ever block MediaStreamGraph low-latency streams
michael@0:         uint32_t remains = 0; // we presume the buffer is full
michael@0:         if (mBuffer.Length() > bytesToCopy) {
michael@0:           remains = mBuffer.Length() - bytesToCopy; // Free up just enough space
michael@0:         }
michael@0:         // account for dropping samples
michael@0:         PR_LOG(gAudioStreamLog, PR_LOG_WARNING, ("Stream %p dropping %u bytes (%u frames)in Write()",
michael@0:             this, mBuffer.Length() - remains, BytesToFrames(mBuffer.Length() - remains)));
michael@0:         mReadPoint += BytesToFrames(mBuffer.Length() - remains);
michael@0:         mBuffer.ContractTo(remains);
michael@0:       } else { // RUNNING or high latency
michael@0:         // If we are not playing, but our buffer is full, start playing to make
michael@0:         // room for soon-to-be-decoded data.
michael@0:         if (mState != STARTED && mState != RUNNING) {
michael@0:           PR_LOG(gAudioStreamLog, PR_LOG_WARNING, ("Starting stream %p in Write (%u waiting)",
michael@0:                                                  this, bytesToCopy));
michael@0:           StartUnlocked();
michael@0:           if (mState == ERRORED) {
michael@0:             return NS_ERROR_FAILURE;
michael@0:           }
michael@0:         }
michael@0:         PR_LOG(gAudioStreamLog, PR_LOG_WARNING, ("Stream %p waiting in Write() (%u waiting)",
michael@0:                                                  this, bytesToCopy));
michael@0:         mon.Wait();
michael@0:       }
michael@0:     }
michael@0:   }
michael@0: 
michael@0:   mWritten += aFrames;
michael@0:   return NS_OK;
michael@0: }
michael@0: 
michael@0: uint32_t
michael@0: AudioStream::Available()
michael@0: {
michael@0:   MonitorAutoLock mon(mMonitor);
michael@0:   NS_ABORT_IF_FALSE(mBuffer.Length() % mBytesPerFrame == 0, "Buffer invariant violated.");
michael@0:   return BytesToFrames(mBuffer.Available());
michael@0: }
michael@0: 
michael@0: void
michael@0: AudioStream::SetVolume(double aVolume)
michael@0: {
michael@0:   MonitorAutoLock mon(mMonitor);
michael@0:   NS_ABORT_IF_FALSE(aVolume >= 0.0 && aVolume <= 1.0, "Invalid volume");
michael@0:   mVolume = aVolume;
michael@0: }
michael@0: 
michael@0: void
michael@0: AudioStream::Drain()
michael@0: {
michael@0:   MonitorAutoLock mon(mMonitor);
michael@0:   LOG(("AudioStream::Drain() for %p, state %d, avail %u", this, mState, mBuffer.Available()));
michael@0:   if (mState != STARTED && mState != RUNNING) {
michael@0:     NS_ASSERTION(mState == ERRORED || mBuffer.Available() == 0, "Draining without full buffer of unplayed audio");
michael@0:     return;
michael@0:   }
michael@0:   mState = DRAINING;
michael@0:   while (mState == DRAINING) {
michael@0:     mon.Wait();
michael@0:   }
michael@0: }
michael@0: 
michael@0: void
michael@0: AudioStream::Start()
michael@0: {
michael@0:   MonitorAutoLock mon(mMonitor);
michael@0:   StartUnlocked();
michael@0: }
michael@0: 
michael@0: void
michael@0: AudioStream::StartUnlocked()
michael@0: {
michael@0:   mMonitor.AssertCurrentThreadOwns();
michael@0:   if (!mCubebStream) {
michael@0:     mNeedsStart = true;
michael@0:     return;
michael@0:   }
michael@0:   MonitorAutoUnlock mon(mMonitor);
michael@0:   if (mState == INITIALIZED) {
michael@0:     int r = cubeb_stream_start(mCubebStream);
michael@0:     mState = r == CUBEB_OK ? STARTED : ERRORED;
michael@0:     LOG(("AudioStream: started %p, state %s", this, mState == STARTED ? "STARTED" : "ERRORED"));
michael@0:   }
michael@0: }
michael@0: 
michael@0: void
michael@0: AudioStream::Pause()
michael@0: {
michael@0:   MonitorAutoLock mon(mMonitor);
michael@0:   if (!mCubebStream || (mState != STARTED && mState != RUNNING)) {
michael@0:     mNeedsStart = false;
michael@0:     mState = STOPPED; // which also tells async OpenCubeb not to start, just init
michael@0:     return;
michael@0:   }
michael@0: 
michael@0:   int r;
michael@0:   {
michael@0:     MonitorAutoUnlock mon(mMonitor);
michael@0:     r = cubeb_stream_stop(mCubebStream);
michael@0:   }
michael@0:   if (mState != ERRORED && r == CUBEB_OK) {
michael@0:     mState = STOPPED;
michael@0:   }
michael@0: }
michael@0: 
michael@0: void
michael@0: AudioStream::Resume()
michael@0: {
michael@0:   MonitorAutoLock mon(mMonitor);
michael@0:   if (!mCubebStream || mState != STOPPED) {
michael@0:     return;
michael@0:   }
michael@0: 
michael@0:   int r;
michael@0:   {
michael@0:     MonitorAutoUnlock mon(mMonitor);
michael@0:     r = cubeb_stream_start(mCubebStream);
michael@0:   }
michael@0:   if (mState != ERRORED && r == CUBEB_OK) {
michael@0:     mState = STARTED;
michael@0:   }
michael@0: }
michael@0: 
michael@0: void
michael@0: AudioStream::Shutdown()
michael@0: {
michael@0:   LOG(("AudioStream: Shutdown %p, state %d", this, mState));
michael@0:   {
michael@0:     MonitorAutoLock mon(mMonitor);
michael@0:     if (mState == STARTED || mState == RUNNING) {
michael@0:       MonitorAutoUnlock mon(mMonitor);
michael@0:       Pause();
michael@0:     }
michael@0:     MOZ_ASSERT(mState != STARTED && mState != RUNNING); // paranoia
michael@0:     mState = SHUTDOWN;
michael@0:   }
michael@0:   // Must not try to shut down cubeb from within the lock!  wasapi may still
michael@0:   // call our callback after Pause()/stop()!?! Bug 996162
michael@0:   if (mCubebStream) {
michael@0:     mCubebStream.reset();
michael@0:   }
michael@0: }
michael@0: 
michael@0: int64_t
michael@0: AudioStream::GetPosition()
michael@0: {
michael@0:   MonitorAutoLock mon(mMonitor);
michael@0:   return mAudioClock.GetPositionUnlocked();
michael@0: }
michael@0: 
michael@0: // This function is miscompiled by PGO with MSVC 2010.  See bug 768333.
michael@0: #ifdef _MSC_VER
michael@0: #pragma optimize("", off)
michael@0: #endif
michael@0: int64_t
michael@0: AudioStream::GetPositionInFrames()
michael@0: {
michael@0:   return mAudioClock.GetPositionInFrames();
michael@0: }
michael@0: #ifdef _MSC_VER
michael@0: #pragma optimize("", on)
michael@0: #endif
michael@0: 
michael@0: int64_t
michael@0: AudioStream::GetPositionInFramesInternal()
michael@0: {
michael@0:   MonitorAutoLock mon(mMonitor);
michael@0:   return GetPositionInFramesUnlocked();
michael@0: }
michael@0: 
michael@0: int64_t
michael@0: AudioStream::GetPositionInFramesUnlocked()
michael@0: {
michael@0:   mMonitor.AssertCurrentThreadOwns();
michael@0: 
michael@0:   if (!mCubebStream || mState == ERRORED) {
michael@0:     return -1;
michael@0:   }
michael@0: 
michael@0:   uint64_t position = 0;
michael@0:   {
michael@0:     MonitorAutoUnlock mon(mMonitor);
michael@0:     if (cubeb_stream_get_position(mCubebStream, &position) != CUBEB_OK) {
michael@0:       return -1;
michael@0:     }
michael@0:   }
michael@0: 
michael@0:   // Adjust the reported position by the number of silent frames written
michael@0:   // during stream underruns.
michael@0:   uint64_t adjustedPosition = 0;
michael@0:   if (position >= mLostFrames) {
michael@0:     adjustedPosition = position - mLostFrames;
michael@0:   }
michael@0:   return std::min<uint64_t>(adjustedPosition, INT64_MAX);
michael@0: }
michael@0: 
michael@0: int64_t
michael@0: AudioStream::GetLatencyInFrames()
michael@0: {
michael@0:   uint32_t latency;
michael@0:   if (cubeb_stream_get_latency(mCubebStream, &latency)) {
michael@0:     NS_WARNING("Could not get cubeb latency.");
michael@0:     return 0;
michael@0:   }
michael@0:   return static_cast<int64_t>(latency);
michael@0: }
michael@0: 
michael@0: bool
michael@0: AudioStream::IsPaused()
michael@0: {
michael@0:   MonitorAutoLock mon(mMonitor);
michael@0:   return mState == STOPPED;
michael@0: }
michael@0: 
michael@0: void
michael@0: AudioStream::GetBufferInsertTime(int64_t &aTimeMs)
michael@0: {
michael@0:   if (mInserts.Length() > 0) {
michael@0:     // Find the right block, but don't leave the array empty
michael@0:     while (mInserts.Length() > 1 && mReadPoint >= mInserts[0].mFrames) {
michael@0:       mReadPoint -= mInserts[0].mFrames;
michael@0:       mInserts.RemoveElementAt(0);
michael@0:     }
michael@0:     // offset for amount already read
michael@0:     // XXX Note: could misreport if we couldn't find a block in the right timeframe
michael@0:     aTimeMs = mInserts[0].mTimeMs + ((mReadPoint * 1000) / mOutRate);
michael@0:   } else {
michael@0:     aTimeMs = INT64_MAX;
michael@0:   }
michael@0: }
michael@0: 
michael@0: long
michael@0: AudioStream::GetUnprocessed(void* aBuffer, long aFrames, int64_t &aTimeMs)
michael@0: {
michael@0:   uint8_t* wpos = reinterpret_cast<uint8_t*>(aBuffer);
michael@0: 
michael@0:   // Flush the timestretcher pipeline, if we were playing using a playback rate
michael@0:   // other than 1.0.
michael@0:   uint32_t flushedFrames = 0;
michael@0:   if (mTimeStretcher && mTimeStretcher->numSamples()) {
michael@0:     flushedFrames = mTimeStretcher->receiveSamples(reinterpret_cast<AudioDataValue*>(wpos), aFrames);
michael@0:     wpos += FramesToBytes(flushedFrames);
michael@0:   }
michael@0:   uint32_t toPopBytes = FramesToBytes(aFrames - flushedFrames);
michael@0:   uint32_t available = std::min(toPopBytes, mBuffer.Length());
michael@0: 
michael@0:   void* input[2];
michael@0:   uint32_t input_size[2];
michael@0:   mBuffer.PopElements(available, &input[0], &input_size[0], &input[1], &input_size[1]);
michael@0:   memcpy(wpos, input[0], input_size[0]);
michael@0:   wpos += input_size[0];
michael@0:   memcpy(wpos, input[1], input_size[1]);
michael@0: 
michael@0:   // First time block now has our first returned sample
michael@0:   mReadPoint += BytesToFrames(available);
michael@0:   GetBufferInsertTime(aTimeMs);
michael@0: 
michael@0:   return BytesToFrames(available) + flushedFrames;
michael@0: }
michael@0: 
michael@0: // Get unprocessed samples, and pad the beginning of the buffer with silence if
michael@0: // there is not enough data.
michael@0: long
michael@0: AudioStream::GetUnprocessedWithSilencePadding(void* aBuffer, long aFrames, int64_t& aTimeMs)
michael@0: {
michael@0:   uint32_t toPopBytes = FramesToBytes(aFrames);
michael@0:   uint32_t available = std::min(toPopBytes, mBuffer.Length());
michael@0:   uint32_t silenceOffset = toPopBytes - available;
michael@0: 
michael@0:   uint8_t* wpos = reinterpret_cast<uint8_t*>(aBuffer);
michael@0: 
michael@0:   memset(wpos, 0, silenceOffset);
michael@0:   wpos += silenceOffset;
michael@0: 
michael@0:   void* input[2];
michael@0:   uint32_t input_size[2];
michael@0:   mBuffer.PopElements(available, &input[0], &input_size[0], &input[1], &input_size[1]);
michael@0:   memcpy(wpos, input[0], input_size[0]);
michael@0:   wpos += input_size[0];
michael@0:   memcpy(wpos, input[1], input_size[1]);
michael@0: 
michael@0:   GetBufferInsertTime(aTimeMs);
michael@0: 
michael@0:   return aFrames;
michael@0: }
michael@0: 
michael@0: long
michael@0: AudioStream::GetTimeStretched(void* aBuffer, long aFrames, int64_t &aTimeMs)
michael@0: {
michael@0:   long processedFrames = 0;
michael@0: 
michael@0:   // We need to call the non-locking version, because we already have the lock.
michael@0:   if (EnsureTimeStretcherInitializedUnlocked() != NS_OK) {
michael@0:     return 0;
michael@0:   }
michael@0: 
michael@0:   uint8_t* wpos = reinterpret_cast<uint8_t*>(aBuffer);
michael@0:   double playbackRate = static_cast<double>(mInRate) / mOutRate;
michael@0:   uint32_t toPopBytes = FramesToBytes(ceil(aFrames / playbackRate));
michael@0:   uint32_t available = 0;
michael@0:   bool lowOnBufferedData = false;
michael@0:   do {
michael@0:     // Check if we already have enough data in the time stretcher pipeline.
michael@0:     if (mTimeStretcher->numSamples() <= static_cast<uint32_t>(aFrames)) {
michael@0:       void* input[2];
michael@0:       uint32_t input_size[2];
michael@0:       available = std::min(mBuffer.Length(), toPopBytes);
michael@0:       if (available != toPopBytes) {
michael@0:         lowOnBufferedData = true;
michael@0:       }
michael@0:       mBuffer.PopElements(available, &input[0], &input_size[0],
michael@0:                                      &input[1], &input_size[1]);
michael@0:       mReadPoint += BytesToFrames(available);
michael@0:       for(uint32_t i = 0; i < 2; i++) {
michael@0:         mTimeStretcher->putSamples(reinterpret_cast<AudioDataValue*>(input[i]), BytesToFrames(input_size[i]));
michael@0:       }
michael@0:     }
michael@0:     uint32_t receivedFrames = mTimeStretcher->receiveSamples(reinterpret_cast<AudioDataValue*>(wpos), aFrames - processedFrames);
michael@0:     wpos += FramesToBytes(receivedFrames);
michael@0:     processedFrames += receivedFrames;
michael@0:   } while (processedFrames < aFrames && !lowOnBufferedData);
michael@0: 
michael@0:   GetBufferInsertTime(aTimeMs);
michael@0: 
michael@0:   return processedFrames;
michael@0: }
michael@0: 
michael@0: long
michael@0: AudioStream::DataCallback(void* aBuffer, long aFrames)
michael@0: {
michael@0:   MonitorAutoLock mon(mMonitor);
michael@0:   uint32_t available = std::min(static_cast<uint32_t>(FramesToBytes(aFrames)), mBuffer.Length());
michael@0:   NS_ABORT_IF_FALSE(available % mBytesPerFrame == 0, "Must copy complete frames");
michael@0:   AudioDataValue* output = reinterpret_cast<AudioDataValue*>(aBuffer);
michael@0:   uint32_t underrunFrames = 0;
michael@0:   uint32_t servicedFrames = 0;
michael@0:   int64_t insertTime;
michael@0: 
michael@0:   // NOTE: wasapi (others?) can call us back *after* stop()/Shutdown() (mState == SHUTDOWN)
michael@0:   // Bug 996162
michael@0: 
michael@0:   // callback tells us cubeb succeeded initializing
michael@0:   if (mState == STARTED) {
michael@0:     // For low-latency streams, we want to minimize any built-up data when
michael@0:     // we start getting callbacks.
michael@0:     // Simple version - contract on first callback only.
michael@0:     if (mLatencyRequest == LowLatency) {
michael@0: #ifdef PR_LOGGING
michael@0:       uint32_t old_len = mBuffer.Length();
michael@0: #endif
michael@0:       available = mBuffer.ContractTo(FramesToBytes(aFrames));
michael@0: #ifdef PR_LOGGING
michael@0:       TimeStamp now = TimeStamp::Now();
michael@0:       if (!mStartTime.IsNull()) {
michael@0:         int64_t timeMs = (now - mStartTime).ToMilliseconds();
michael@0:         PR_LOG(gAudioStreamLog, PR_LOG_WARNING,
michael@0:                ("Stream took %lldms to start after first Write() @ %u", timeMs, mOutRate));
michael@0:       } else {
michael@0:         PR_LOG(gAudioStreamLog, PR_LOG_WARNING,
michael@0:           ("Stream started before Write() @ %u", mOutRate));
michael@0:       }
michael@0: 
michael@0:       if (old_len != available) {
michael@0:         // Note that we may have dropped samples in Write() as well!
michael@0:         PR_LOG(gAudioStreamLog, PR_LOG_WARNING,
michael@0:                ("AudioStream %p dropped %u + %u initial frames @ %u", this,
michael@0:                  mReadPoint, BytesToFrames(old_len - available), mOutRate));
michael@0:         mReadPoint += BytesToFrames(old_len - available);
michael@0:       }
michael@0: #endif
michael@0:     }
michael@0:     mState = RUNNING;
michael@0:   }
michael@0: 
michael@0:   if (available) {
michael@0:     // When we are playing a low latency stream, and it is the first time we are
michael@0:     // getting data from the buffer, we prefer to add the silence for an
michael@0:     // underrun at the beginning of the buffer, so the first buffer is not cut
michael@0:     // in half by the silence inserted to compensate for the underrun.
michael@0:     if (mInRate == mOutRate) {
michael@0:       if (mLatencyRequest == LowLatency && !mWritten) {
michael@0:         servicedFrames = GetUnprocessedWithSilencePadding(output, aFrames, insertTime);
michael@0:       } else {
michael@0:         servicedFrames = GetUnprocessed(output, aFrames, insertTime);
michael@0:       }
michael@0:     } else {
michael@0:       servicedFrames = GetTimeStretched(output, aFrames, insertTime);
michael@0:     }
michael@0:     float scaled_volume = float(GetVolumeScale() * mVolume);
michael@0: 
michael@0:     ScaleAudioSamples(output, aFrames * mOutChannels, scaled_volume);
michael@0: 
michael@0:     NS_ABORT_IF_FALSE(mBuffer.Length() % mBytesPerFrame == 0, "Must copy complete frames");
michael@0: 
michael@0:     // Notify any blocked Write() call that more space is available in mBuffer.
michael@0:     mon.NotifyAll();
michael@0:   } else {
michael@0:     GetBufferInsertTime(insertTime);
michael@0:   }
michael@0: 
michael@0:   underrunFrames = aFrames - servicedFrames;
michael@0: 
michael@0:   if (mState != DRAINING) {
michael@0:     uint8_t* rpos = static_cast<uint8_t*>(aBuffer) + FramesToBytes(aFrames - underrunFrames);
michael@0:     memset(rpos, 0, FramesToBytes(underrunFrames));
michael@0:     if (underrunFrames) {
michael@0:       PR_LOG(gAudioStreamLog, PR_LOG_WARNING,
michael@0:              ("AudioStream %p lost %d frames", this, underrunFrames));
michael@0:     }
michael@0:     mLostFrames += underrunFrames;
michael@0:     servicedFrames += underrunFrames;
michael@0:   }
michael@0: 
michael@0:   WriteDumpFile(mDumpFile, this, aFrames, aBuffer);
michael@0:   // Don't log if we're not interested or if the stream is inactive
michael@0:   if (PR_LOG_TEST(GetLatencyLog(), PR_LOG_DEBUG) &&
michael@0:       mState != SHUTDOWN &&
michael@0:       insertTime != INT64_MAX && servicedFrames > underrunFrames) {
michael@0:     uint32_t latency = UINT32_MAX;
michael@0:     if (cubeb_stream_get_latency(mCubebStream, &latency)) {
michael@0:       NS_WARNING("Could not get latency from cubeb.");
michael@0:     }
michael@0:     TimeStamp now = TimeStamp::Now();
michael@0: 
michael@0:     mLatencyLog->Log(AsyncLatencyLogger::AudioStream, reinterpret_cast<uint64_t>(this),
michael@0:                      insertTime, now);
michael@0:     mLatencyLog->Log(AsyncLatencyLogger::Cubeb, reinterpret_cast<uint64_t>(mCubebStream.get()),
michael@0:                      (latency * 1000) / mOutRate, now);
michael@0:   }
michael@0: 
michael@0:   mAudioClock.UpdateWritePosition(servicedFrames);
michael@0:   return servicedFrames;
michael@0: }
michael@0: 
michael@0: void
michael@0: AudioStream::StateCallback(cubeb_state aState)
michael@0: {
michael@0:   MonitorAutoLock mon(mMonitor);
michael@0:   if (aState == CUBEB_STATE_DRAINED) {
michael@0:     mState = DRAINED;
michael@0:   } else if (aState == CUBEB_STATE_ERROR) {
michael@0:     LOG(("AudioStream::StateCallback() state %d cubeb error", mState));
michael@0:     mState = ERRORED;
michael@0:   }
michael@0:   mon.NotifyAll();
michael@0: }
michael@0: 
michael@0: AudioClock::AudioClock(AudioStream* aStream)
michael@0:  :mAudioStream(aStream),
michael@0:   mOldOutRate(0),
michael@0:   mBasePosition(0),
michael@0:   mBaseOffset(0),
michael@0:   mOldBaseOffset(0),
michael@0:   mOldBasePosition(0),
michael@0:   mPlaybackRateChangeOffset(0),
michael@0:   mPreviousPosition(0),
michael@0:   mWritten(0),
michael@0:   mOutRate(0),
michael@0:   mInRate(0),
michael@0:   mPreservesPitch(true),
michael@0:   mCompensatingLatency(false)
michael@0: {}
michael@0: 
michael@0: void AudioClock::Init()
michael@0: {
michael@0:   mOutRate = mAudioStream->GetRate();
michael@0:   mInRate = mAudioStream->GetRate();
michael@0:   mOldOutRate = mOutRate;
michael@0: }
michael@0: 
michael@0: void AudioClock::UpdateWritePosition(uint32_t aCount)
michael@0: {
michael@0:   mWritten += aCount;
michael@0: }
michael@0: 
michael@0: uint64_t AudioClock::GetPositionUnlocked()
michael@0: {
michael@0:   // GetPositionInFramesUnlocked() asserts it owns the monitor
michael@0:   int64_t position = mAudioStream->GetPositionInFramesUnlocked();
michael@0:   int64_t diffOffset;
michael@0:   NS_ASSERTION(position < 0 || (mInRate != 0 && mOutRate != 0), "AudioClock not initialized.");
michael@0:   if (position >= 0) {
michael@0:     if (position < mPlaybackRateChangeOffset) {
michael@0:       // See if we are still playing frames pushed with the old playback rate in
michael@0:       // the backend. If we are, use the old output rate to compute the
michael@0:       // position.
michael@0:       mCompensatingLatency = true;
michael@0:       diffOffset = position - mOldBaseOffset;
michael@0:       position = static_cast<uint64_t>(mOldBasePosition +
michael@0:         static_cast<float>(USECS_PER_S * diffOffset) / mOldOutRate);
michael@0:       mPreviousPosition = position;
michael@0:       return position;
michael@0:     }
michael@0: 
michael@0:     if (mCompensatingLatency) {
michael@0:       diffOffset = position - mPlaybackRateChangeOffset;
michael@0:       mCompensatingLatency = false;
michael@0:       mBasePosition = mPreviousPosition;
michael@0:     } else {
michael@0:       diffOffset = position - mPlaybackRateChangeOffset;
michael@0:     }
michael@0:     position =  static_cast<uint64_t>(mBasePosition +
michael@0:       (static_cast<float>(USECS_PER_S * diffOffset) / mOutRate));
michael@0:     return position;
michael@0:   }
michael@0:   return UINT64_MAX;
michael@0: }
michael@0: 
michael@0: uint64_t AudioClock::GetPositionInFrames()
michael@0: {
michael@0:   return (GetPositionUnlocked() * mOutRate) / USECS_PER_S;
michael@0: }
michael@0: 
michael@0: void AudioClock::SetPlaybackRateUnlocked(double aPlaybackRate)
michael@0: {
michael@0:   // GetPositionInFramesUnlocked() asserts it owns the monitor
michael@0:   int64_t position = mAudioStream->GetPositionInFramesUnlocked();
michael@0:   if (position > mPlaybackRateChangeOffset) {
michael@0:     mOldBasePosition = mBasePosition;
michael@0:     mBasePosition = GetPositionUnlocked();
michael@0:     mOldBaseOffset = mPlaybackRateChangeOffset;
michael@0:     mBaseOffset = position;
michael@0:     mPlaybackRateChangeOffset = mWritten;
michael@0:     mOldOutRate = mOutRate;
michael@0:     mOutRate = static_cast<int>(mInRate / aPlaybackRate);
michael@0:   } else {
michael@0:     // The playbackRate has been changed before the end of the latency
michael@0:     // compensation phase. We don't update the mOld* variable. That way, the
michael@0:     // last playbackRate set is taken into account.
michael@0:     mBasePosition = GetPositionUnlocked();
michael@0:     mBaseOffset = position;
michael@0:     mPlaybackRateChangeOffset = mWritten;
michael@0:     mOutRate = static_cast<int>(mInRate / aPlaybackRate);
michael@0:   }
michael@0: }
michael@0: 
michael@0: double AudioClock::GetPlaybackRate()
michael@0: {
michael@0:   return static_cast<double>(mInRate) / mOutRate;
michael@0: }
michael@0: 
michael@0: void AudioClock::SetPreservesPitch(bool aPreservesPitch)
michael@0: {
michael@0:   mPreservesPitch = aPreservesPitch;
michael@0: }
michael@0: 
michael@0: bool AudioClock::GetPreservesPitch()
michael@0: {
michael@0:   return mPreservesPitch;
michael@0: }
michael@0: } // namespace mozilla