/* -*- 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/. */ #if !defined(AudioStream_h_) #define AudioStream_h_ #include "AudioSampleFormat.h" #include "nsAutoPtr.h" #include "nsAutoRef.h" #include "nsCOMPtr.h" #include "nsThreadUtils.h" #include "Latency.h" #include "mozilla/dom/AudioChannelBinding.h" #include "mozilla/StaticMutex.h" #include "mozilla/RefPtr.h" #include "cubeb/cubeb.h" template <> class nsAutoRefTraits : public nsPointerRefTraits { public: static void Release(cubeb_stream* aStream) { cubeb_stream_destroy(aStream); } }; namespace soundtouch { class SoundTouch; } namespace mozilla { class AudioStream; class FrameHistory; class AudioClock { public: AudioClock(AudioStream* aStream); // Initialize the clock with the current AudioStream. Need to be called // before querying the clock. Called on the audio thread. void Init(); // Update the number of samples that has been written in the audio backend. // Called on the state machine thread. void UpdateFrameHistory(uint32_t aServiced, uint32_t aUnderrun); // Get the read position of the stream, in microseconds. // Called on the state machine thead. // Assumes the AudioStream lock is held and thus calls Unlocked versions // of AudioStream funcs. int64_t GetPositionUnlocked() const; // Get the read position of the stream, in frames. // Called on the state machine thead. int64_t GetPositionInFrames() const; // Set the playback rate. // Called on the audio thread. // Assumes the AudioStream lock is held and thus calls Unlocked versions // of AudioStream funcs. void SetPlaybackRateUnlocked(double aPlaybackRate); // Get the current playback rate. // Called on the audio thread. double GetPlaybackRate() const; // Set if we are preserving the pitch. // Called on the audio thread. void SetPreservesPitch(bool aPreservesPitch); // Get the current pitch preservation state. // Called on the audio thread. bool GetPreservesPitch() const; private: // This AudioStream holds a strong reference to this AudioClock. This // pointer is garanteed to always be valid. AudioStream* const mAudioStream; // Output rate in Hz (characteristic of the playback rate) int mOutRate; // Input rate in Hz (characteristic of the media being played) int mInRate; // True if the we are timestretching, false if we are resampling. bool mPreservesPitch; // The history of frames sent to the audio engine in each Datacallback. const nsAutoPtr mFrameHistory; }; class CircularByteBuffer { public: CircularByteBuffer() : mBuffer(nullptr), mCapacity(0), mStart(0), mCount(0) {} // Set the capacity of the buffer in bytes. Must be called before any // call to append or pop elements. void SetCapacity(uint32_t aCapacity) { NS_ABORT_IF_FALSE(!mBuffer, "Buffer allocated."); mCapacity = aCapacity; mBuffer = new uint8_t[mCapacity]; } uint32_t Length() { return mCount; } uint32_t Capacity() { return mCapacity; } uint32_t Available() { return Capacity() - Length(); } // Append aLength bytes from aSrc to the buffer. Caller must check that // sufficient space is available. void AppendElements(const uint8_t* aSrc, uint32_t aLength) { NS_ABORT_IF_FALSE(mBuffer && mCapacity, "Buffer not initialized."); NS_ABORT_IF_FALSE(aLength <= Available(), "Buffer full."); uint32_t end = (mStart + mCount) % mCapacity; uint32_t toCopy = std::min(mCapacity - end, aLength); memcpy(&mBuffer[end], aSrc, toCopy); memcpy(&mBuffer[0], aSrc + toCopy, aLength - toCopy); mCount += aLength; } // Remove aSize bytes from the buffer. Caller must check returned size in // aSize{1,2} before using the pointer returned in aData{1,2}. Caller // must not specify an aSize larger than Length(). void PopElements(uint32_t aSize, void** aData1, uint32_t* aSize1, void** aData2, uint32_t* aSize2) { NS_ABORT_IF_FALSE(mBuffer && mCapacity, "Buffer not initialized."); NS_ABORT_IF_FALSE(aSize <= Length(), "Request too large."); *aData1 = &mBuffer[mStart]; *aSize1 = std::min(mCapacity - mStart, aSize); *aData2 = &mBuffer[0]; *aSize2 = aSize - *aSize1; mCount -= *aSize1 + *aSize2; mStart += *aSize1 + *aSize2; mStart %= mCapacity; } // Throw away all but aSize bytes from the buffer. Returns new size, which // may be less than aSize uint32_t ContractTo(uint32_t aSize) { NS_ABORT_IF_FALSE(mBuffer && mCapacity, "Buffer not initialized."); if (aSize >= mCount) { return mCount; } mStart += (mCount - aSize); mCount = aSize; mStart %= mCapacity; return mCount; } size_t SizeOfExcludingThis(MallocSizeOf aMallocSizeOf) const { size_t amount = 0; amount += mBuffer.SizeOfExcludingThis(aMallocSizeOf); return amount; } private: nsAutoArrayPtr mBuffer; uint32_t mCapacity; uint32_t mStart; uint32_t mCount; }; class AudioInitTask; // Access to a single instance of this class must be synchronized by // callers, or made from a single thread. One exception is that access to // GetPosition, GetPositionInFrames, SetVolume, and Get{Rate,Channels} // is thread-safe without external synchronization. class AudioStream MOZ_FINAL { virtual ~AudioStream(); public: // Initialize Audio Library. Some Audio backends require initializing the // library before using it. static void InitLibrary(); // Shutdown Audio Library. Some Audio backends require shutting down the // library after using it. static void ShutdownLibrary(); // Returns the maximum number of channels supported by the audio hardware. static int MaxNumberOfChannels(); // Queries the samplerate the hardware/mixer runs at, and stores it. // Can be called on any thread. When this returns, it is safe to call // PreferredSampleRate without locking. static void InitPreferredSampleRate(); // Get the aformentionned sample rate. Does not lock. static int PreferredSampleRate(); NS_INLINE_DECL_THREADSAFE_REFCOUNTING(AudioStream) AudioStream(); enum LatencyRequest { HighLatency, LowLatency }; // Initialize the audio stream. aNumChannels is the number of audio // channels (1 for mono, 2 for stereo, etc) and aRate is the sample rate // (22050Hz, 44100Hz, etc). nsresult Init(int32_t aNumChannels, int32_t aRate, const dom::AudioChannel aAudioStreamChannel, LatencyRequest aLatencyRequest); // Closes the stream. All future use of the stream is an error. void Shutdown(); // Write audio data to the audio hardware. aBuf is an array of AudioDataValues // AudioDataValue of length aFrames*mChannels. If aFrames is larger // than the result of Available(), the write will block until sufficient // buffer space is available. aTime is the time in ms associated with the first sample // for latency calculations nsresult Write(const AudioDataValue* aBuf, uint32_t aFrames, TimeStamp* aTime = nullptr); // Return the number of audio frames that can be written without blocking. uint32_t Available(); // Set the current volume of the audio playback. This is a value from // 0 (meaning muted) to 1 (meaning full volume). Thread-safe. void SetVolume(double aVolume); // Block until buffered audio data has been consumed. void Drain(); // Break any blocking operation and set the stream to shutdown. void Cancel(); // Start the stream. void Start(); // Return the number of frames written so far in the stream. This allow the // caller to check if it is safe to start the stream, if needed. int64_t GetWritten(); // Pause audio playback. void Pause(); // Resume audio playback. void Resume(); // Return the position in microseconds of the audio frame being played by // the audio hardware, compensated for playback rate change. Thread-safe. int64_t GetPosition(); // Return the position, measured in audio frames played since the stream // was opened, of the audio hardware. Thread-safe. int64_t GetPositionInFrames(); // Returns true when the audio stream is paused. bool IsPaused(); int GetRate() { return mOutRate; } int GetChannels() { return mChannels; } int GetOutChannels() { return mOutChannels; } // Set playback rate as a multiple of the intrinsic playback rate. This is to // be called only with aPlaybackRate > 0.0. nsresult SetPlaybackRate(double aPlaybackRate); // Switch between resampling (if false) and time stretching (if true, default). nsresult SetPreservesPitch(bool aPreservesPitch); size_t SizeOfIncludingThis(MallocSizeOf aMallocSizeOf) const; protected: friend class AudioClock; // Return the position, measured in audio frames played since the stream was // opened, of the audio hardware, not adjusted for the changes of playback // rate or underrun frames. // Caller must own the monitor. int64_t GetPositionInFramesUnlocked(); private: friend class AudioInitTask; // So we can call it asynchronously from AudioInitTask nsresult OpenCubeb(cubeb_stream_params &aParams, LatencyRequest aLatencyRequest); void CheckForStart(); static void PrefChanged(const char* aPref, void* aClosure); static double GetVolumeScale(); static bool GetFirstStream(); static cubeb* GetCubebContext(); static cubeb* GetCubebContextUnlocked(); static uint32_t GetCubebLatency(); static bool CubebLatencyPrefSet(); static long DataCallback_S(cubeb_stream*, void* aThis, void* aBuffer, long aFrames) { return static_cast(aThis)->DataCallback(aBuffer, aFrames); } static void StateCallback_S(cubeb_stream*, void* aThis, cubeb_state aState) { static_cast(aThis)->StateCallback(aState); } long DataCallback(void* aBuffer, long aFrames); void StateCallback(cubeb_state aState); nsresult EnsureTimeStretcherInitializedUnlocked(); // aTime is the time in ms the samples were inserted into MediaStreamGraph long GetUnprocessed(void* aBuffer, long aFrames, int64_t &aTime); long GetTimeStretched(void* aBuffer, long aFrames, int64_t &aTime); long GetUnprocessedWithSilencePadding(void* aBuffer, long aFrames, int64_t &aTime); int64_t GetLatencyInFrames(); void GetBufferInsertTime(int64_t &aTimeMs); void StartUnlocked(); // The monitor is held to protect all access to member variables. Write() // waits while mBuffer is full; DataCallback() notifies as it consumes // data from mBuffer. Drain() waits while mState is DRAINING; // StateCallback() notifies when mState is DRAINED. Monitor mMonitor; // Input rate in Hz (characteristic of the media being played) int mInRate; // Output rate in Hz (characteristic of the playback rate) int mOutRate; int mChannels; int mOutChannels; // Number of frames written to the buffers. int64_t mWritten; AudioClock mAudioClock; nsAutoPtr mTimeStretcher; nsRefPtr mLatencyLog; // copy of Latency logger's starting time for offset calculations TimeStamp mStartTime; // Whether we are playing a low latency stream, or a normal stream. LatencyRequest mLatencyRequest; // Where in the current mInserts[0] block cubeb has read to int64_t mReadPoint; // Keep track of each inserted block of samples and the time it was inserted // so we can estimate the clock time for a specific sample's insertion (for when // we send data to cubeb). Blocks are aged out as needed. struct Inserts { int64_t mTimeMs; int64_t mFrames; }; nsAutoTArray mInserts; // Output file for dumping audio FILE* mDumpFile; // Temporary audio buffer. Filled by Write() and consumed by // DataCallback(). Once mBuffer is full, Write() blocks until sufficient // space becomes available in mBuffer. mBuffer is sized in bytes, not // frames. CircularByteBuffer mBuffer; // Software volume level. Applied during the servicing of DataCallback(). double mVolume; // Owning reference to a cubeb_stream. cubeb_stream_destroy is called by // nsAutoRef's destructor. nsAutoRef mCubebStream; uint32_t mBytesPerFrame; uint32_t BytesToFrames(uint32_t aBytes) { NS_ASSERTION(aBytes % mBytesPerFrame == 0, "Byte count not aligned on frames size."); return aBytes / mBytesPerFrame; } uint32_t FramesToBytes(uint32_t aFrames) { return aFrames * mBytesPerFrame; } enum StreamState { INITIALIZED, // Initialized, playback has not begun. STARTED, // cubeb started, but callbacks haven't started RUNNING, // DataCallbacks have started after STARTED, or after Resume(). STOPPED, // Stopped by a call to Pause(). DRAINING, // Drain requested. DataCallback will indicate end of stream // once the remaining contents of mBuffer are requested by // cubeb, after which StateCallback will indicate drain // completion. DRAINED, // StateCallback has indicated that the drain is complete. ERRORED, // Stream disabled due to an internal error. SHUTDOWN // Shutdown has been called }; StreamState mState; bool mNeedsStart; // needed in case Start() is called before cubeb is open bool mIsFirst; // This mutex protects the static members below. static StaticMutex sMutex; static cubeb* sCubebContext; // Prefered samplerate, in Hz (characteristic of the // hardware/mixer/platform/API used). static uint32_t sPreferredSampleRate; static double sVolumeScale; static uint32_t sCubebLatency; static bool sCubebLatencyPrefSet; }; class AudioInitTask : public nsRunnable { public: AudioInitTask(AudioStream *aStream, AudioStream::LatencyRequest aLatencyRequest, const cubeb_stream_params &aParams) : mAudioStream(aStream) , mLatencyRequest(aLatencyRequest) , mParams(aParams) {} nsresult Dispatch() { // Can't add 'this' as the event to run, since mThread may not be set yet nsresult rv = NS_NewNamedThread("CubebInit", getter_AddRefs(mThread)); if (NS_SUCCEEDED(rv)) { // Note: event must not null out mThread! rv = mThread->Dispatch(this, NS_DISPATCH_NORMAL); } return rv; } protected: virtual ~AudioInitTask() {}; private: NS_IMETHOD Run() MOZ_OVERRIDE MOZ_FINAL; RefPtr mAudioStream; AudioStream::LatencyRequest mLatencyRequest; cubeb_stream_params mParams; nsCOMPtr mThread; }; } // namespace mozilla #endif